# Benchmark DAC1 now available with USB



## majid

I got an email from them announcing an updated DAC1 with 24/96 USB input:

http://www.benchmarkmedia.com/eupdate







 They claim no special drivers (e.g. ASIO) are required for it. You pay dearly for the privilege of USB, though, an extra $300...

  Quote:


 FOR IMMEDIATE RELEASE

 Benchmark Media Systems Announces First Native 96-kHz 24-bit "Advanced USB Audio"

 SYRACUSE, New York, February 15, 2007 - Benchmark Media Systems reinforces its 
 position as a leading manufacturer of precision audio electronics with today's 
 announcement of the DAC1 USB - a 192-kHz, 24-bit, digital-to-analog converter 
 featuring "Advanced USB Audio", the world's first native USB audio solution that 
 supports 96-kHz, 24-bit audio.

 Until now, special drivers were required for 24-bit word lengths and sample rates 
 above 48-kHz. Benchmark's Advanced USB Audio provides high-resolution, 
 bit-transparent playback just seconds after plugging into a computer's USB port 
 for the first time. There is no software to install, and there are no system 
 settings that need to be changed. Benchmark's Advanced USB Audio is compatible 
 with Microsoft Windows Vista/XP/2000 and Mac OS X.

 "The ground-breaking performance and elegant simplicity of Benchmark's Advanced 
 USB Audio technology is the ideal addition to the DAC1's jitter-immune UltraLock™ 
 system", noted John Siau, Director of Engineering at Benchmark Media Systems. 
 "We conducted stringent audio-quality tests on four different operating systems 
 using a variety of audio playback applications. Our system delivers stunning 
 performance without the compromises normally associated with computer-based 
 audio playback."

 "The DAC1 USB is the only PC audio interface device that is fully satisfactory; 
 in terms of bit-accuracy, sample-rate, and word-length performance", added Allen 
 Burdick, President of Benchmark Media Systems. "It's intuitive, it's easy - it's 
 USB audio done right."

 The new DAC1 USB, which begins shipping worldwide on March 1, 2007 at a price of 
 $1275 USD, also includes high-current output drivers that can be configured to 
 mute upon headphone insertion. The classic DAC1, which does not include the USB 
 option and special high-current drivers, is still available for $975 USD. Both 
 options are available in the silver, black, and, black rack-mountable chassis.

 Benchmark's DAC1 - a 2-Channel 192-kHz 24-bit digital-to-analog converter - is 
 the recipient of a multitude of awards and a consistent top-seller in both the 
 pro-audio and audiophile markets worldwide.

 "The DAC1 set the mark for digital audio conversion - uncompromised sonic 
 integrity at an incredible value - the DAC1 USB does the same for PC audio", 
 emphasized Allen Burdick. "The importance of high-resolution, bit-transparent 
 computer audio playback simply cannot be overstated. Computer-based audio is 
 used almost exclusively in the broadcast world, it's commonplace in recording 
 studios, and it's revolutionizing home audio systems."

 More information about the DAC1 USB is available on Benchmark's website at:

http://www.benchmarkmedia.com/eupdate

 High resolution photos are available at:

http://www.benchmarkmedia.com/dac1/photos

 For all the latest product announcements, news, and information about Benchmark 
 Media Systems, subscribe to the RSS feed:

http://www.benchmarkmedia.com/press/index.rss

 Benchmark Media Systems, Inc. is a privately-held manufacturer of precision audio 
 electronics providing leading-edge performance to broadcasters, recording studios, 
 sound reinforcement contractors, and home audiophiles since 1978.

 For more information, call Matthew Martin at: 1-800-262-4675 or email:

press@benchmarkmedia.com

 BENCHMARK MEDIA SYSTEMS, INC.
 5925 Court Street Road
 Syracuse, NY, USA 13206-1707
 Phone 800-262-4675, 315-437-6300 
 FAX 315-437-8119
http://www.benchmarkmedia.com


----------



## JSTpt1022

I saw this and immediately hoped that they had reduced the price of the standard version. No such luck of course, but I certainly wasn't expecting a $300 premium! Hopefully the DAC1 price will come down in the future. Or perhaps we might see a rash of sales of the old version? I would sure love to get my hands on one but I can't justify that kind of cost right now.


----------



## lextek

I just got an email about this. Looks pretty, interesting. Lately I'e been consider a computer as a source.


----------



## Jon L

I don't know what they're smoking at Benchmark, but clearly their DAC-1 success has gone to their head. Such baseless, ridiculous hyperbole is sad to witness. It might help if they revealed what exactly is so stupendous about their USB implementation. 

 Their USB solution sounds like the usual run-of-the-mill USB/spdif chip that supports 24/96 (such as one in M-Audio Transit) that feeds their spdif receiver with the "ultralock" asynchronous upsampling circuit, which will feed their AD DAC chip. Big deal, and NO WAY is it worth $300 premium.

 Unless they start releasing some specific info about some new, wonder USB to I2S technology with custom software optimization, the hot air will definitely escape.


----------



## Andrew_WOT

Quote:


  Originally Posted by *Jon L* /img/forum/go_quote.gif 
_I don't know what they're smoking at Benchmark, but clearly their DAC-1 success has gone to their head. Such baseless, ridiculous hyperbole is sad to witness. It might help if they revealed what exactly is so stupendous about their USB implementation. 

 Their USB solution sounds like the usual run-of-the-mill USB/spdif chip that supports 24/96 (such as one in M-Audio Transit) that feeds their spdif receiver with the "ultralock" asynchronous upsampling circuit, which will feed their AD DAC chip. Big deal, and NO WAY is it worth $300 premium.

 Unless they start releasing some specific info about some new, wonder USB to I2S technology with custom software optimization, the hot air will definitely escape._

 

Some more info
http://www.benchmarkmedia.com/comput...usb_audio.html
 They mostly emphasize on being the first driver less 24/96 bit-transparent device.


----------



## kin0kin

Didn't think a PCM270x circuit feeding I2S is worth 300 bucks extra...that makes the Stello DA100 super worthy.


----------



## thomaspf

There is of course the obvious question how they can be bit transparent with the standard Windows USB audio driver while nobody else can. I assume they test this with ASIO4all or kernel streaming at which point all the other USB solutions are bit perfect as well.

 The whole section about pseudo-random samples is just a bunch of crap.

 We will see. 

 Cheers

 Thomas


----------



## Thelonious Monk

$1300 for a DAC is insane. can it really be THAT much better than, say, the PreSonus Centralstation? (that's what i plan to get for my first good external DAC...)

 edit: also, i know that the central station isn't a usb device


----------



## Andrew_WOT

Quote:


  Originally Posted by *thomaspf* /img/forum/go_quote.gif 
_There is of course the obvious question how they can be bit transparent with the standard Windows USB audio driver while nobody else can. I assume they test this with ASIO4all or kernel streaming at which point all the other USB solutions are bit perfect as well.

 The whole section about pseudo-random samples is just a bunch of crap.

 We will see. 

 Cheers

 Thomas_

 

Standard windows drivers do not support 24/96, how they accomplished that without any proprietary drivers is beyond me.


----------



## audioengr

Quote:


  Originally Posted by *thomaspf* /img/forum/go_quote.gif 
_There is of course the obvious question how they can be bit transparent with the standard Windows USB audio driver while nobody else can. I assume they test this with ASIO4all or kernel streaming at which point all the other USB solutions are bit perfect as well.

 The whole section about pseudo-random samples is just a bunch of crap.

 We will see. 

 Cheers

 Thomas_

 


 They claim no special software, so no fair to use Kernel-streaming or ASIO4ALL!

 Steve N.


----------



## The Monkey

Frankly, if I'm resorting to USB, I'll just use my MicroDAC. My Mac has optical out and the DAC1 handles that just fine.

 I'm disappointed that this was their big news. I thought we'd be hearing about a discrete headphone amp in the DAC1.


----------



## Davesrose

I'd be interested to hear the reviews. I had bought a USB to spdif converter so that I could do computer audio with my DAC-1. At first I thought mp3s sounded great through the Benchmark. Then I got into SACD and could hear a difference with a good CD transport. It's redbook playback for CDs sounds very good...so it would be interesting to see if there will be much 24bit audio sources for the USB version: seems like it might be a way to get 24bit DVD-Audio through the Benchmark. But since most albums out are for CD, I guess I'll just be using my current Benchmark as a CD DAC for sometime now. Would be great if I could get SACD out to it though.


----------



## thomaspf

Quote:


  Originally Posted by *Andrew_WOT* 
_Standard windows drivers do not support 24/96, how they accomplished that without any proprietary drivers is beyond me._

 

Hmm, that is news to me. My Audiotrak Optoplay seems to work just fine in 24/96 with the Windows standard driver.


----------



## Iron_Dreamer

I guess they got tired of seeing someone making a boatload of money off the fact that their DAC1 didn't come with USB input. Even if their implementation of the USB input isn't as good as it might be, they will likely get quite a bit of sales from this move. Even though $300 isn't chump change, they can still claim that the DAC1 is pretty reasonably priced for what it offers, and USB inputs on DAC's, even if not ideally designed, do seem to be a big selling point these days.


----------



## humanflyz

wrong thread


----------



## granodemostasa

I'm not entirely sure what the benefit is of not having to install drivers, other than it makes up less hard drive space. the M-Audio audiophile usb converts 24/96 and doesn't cost nearly as much, all one has to sacrifice is using M-Audio's ASIO drivers.


----------



## gregeas

I'm a fan of the DAC1 but coincidentally sold mine this week to fund a Slimdevices Transporter purchase. 

 I would pay an extra $300 for a good USB input on the DAC1. I have a small Dell laptop (XPS M1210) with a 120GB hard drive storing lossless albums, and I love the idea of simply adding the new Benchmark for a high-end two-box solution. I used a Transit in the past, but the extra box (albeit small) and cabling always bugged me. I looked around for a ExpressCard sound card with coax digital output, but none exist yet. (The M1210 only has an ExpressCard slot.)

 I can fit the laptop, DAC1, and my E500s in small laptop bag and have 200 lossless albums at my fingertips. Cool.


----------



## gregeas

>The new DAC1 USB, which begins shipping worldwide on March 1, 2007 
 >at a price of $1275 USD, also includes high-current output drivers that 
 >can be configured to mute upon headphone insertion. 

 What the heck are high-current output drivers?


----------



## Herandu

Quote:


 >The new DAC1 USB, which begins shipping worldwide on March 1, 2007 
 >at a price of $1275 USD, also includes high-current output drivers that 
 >can be configured to mute upon headphone insertion. 
 

Oh boy. More snake oil. The line output is voltage dependent, not current dependent. The input to your amplifier is specified in terms of Volts. Like 2V RMS. SO why do they need a high current going in t perhaps a 47K input on your amp? To blow your amp input up? 
 I pass.


----------



## The Monkey

Are there any other balanced output USB DACs on the market?


----------



## humanflyz

Quote:


  Originally Posted by *The Monkey* /img/forum/go_quote.gif 
_Are there any other balanced output USB DACs on the market?_

 

The Stello DA220 MKII, the Aqvox USB2 D/A MKII, Apogee Mini


----------



## Dept_of_Alchemy

Quote:


  Originally Posted by *The Monkey* /img/forum/go_quote.gif 
_Are there any other balanced output USB DACs on the market?_

 

Does HeadRoom's implementation of their DACs in the balanced Home and Max count?

 So what can the DAC1 USB input ($300) do that can't be accomplished by the M-Audio Transit ($60)? They're both bit-perfect at 24/96... do ya'll think Benchmark will publish jitter measurements?


----------



## thomaspf

I think we know the M-audio transit supports 24/96 and is bit perfect but it comes with a custom driver.

 The standard driver does support 24/96 but unless you use kernel streaming either direct or with ASIO4all it is not bit perfect.


 Cheers

 Thomas


----------



## audioengr

Quote:


  Originally Posted by *Herandu* /img/forum/go_quote.gif 
_Oh boy. More snake oil. The line output is voltage dependent, not current dependent. The input to your amplifier is specified in terms of Volts. Like 2V RMS. SO why do they need a high current going in t perhaps a 47K input on your amp? To blow your amp input up? 
 I pass._

 

High-current output drivers just really means low output impedance. This is ALWAYS a good thing. Most expensive preamps have around 7-10 ohms output impedance.

 Steve N.


----------



## audioengr

Quote:


  Originally Posted by *thomaspf* /img/forum/go_quote.gif 
_I think we know the M-audio transit supports 24/96 and is bit perfect but it comes with a custom driver.

 The standard driver does support 24/96 but unless you use kernel streaming either direct or with ASIO4all it is not bit perfect.


 Cheers

 Thomas_

 

Right, that is my undestanding too. Also, read this:

  Quote:


 All audio streams go through Windows' Kmixer to the USB device. To be sure Kmixer does not affect the quality of audio, we tested it extensively and found it to perform exactly as it should. 
 

They evidently don't hear the effect of KMIXER with Win2K or XP either
	

	
	
		
		

		
		
	


	










 Steve N.


----------



## Davesrose

Quote:


  Originally Posted by *Dept_of_Alchemy* /img/forum/go_quote.gif 
_So what can the DAC1 USB input ($300) do that can't be accomplished by the M-Audio Transit ($60)? They're both bit-perfect at 24/96... do ya'll think Benchmark will publish jitter measurements? 
	

	
	
		
		

		
		
	


	


_

 

Measurements from a 3rd party...would assume USB implimentation would yield similar results as their traditional inputs:

http://www.stereophile.com/digitalpr...86/index4.html


----------



## Joey_V

Quote:


  Originally Posted by *humanflyz* /img/forum/go_quote.gif 
_The Stello DA220 MKII, the Aqvox USB2 D/A MKII, Apogee Mini_

 

And PS Audio Digital Link III.


----------



## J-Pak

Slightly worrying they can't tell the difference with kmixer in the signal path and without 
	

	
	
		
		

		
		
	


	




 Anyone that has any experience, would easily be able to discern a difference.


----------



## clarkc

Quote:


  Originally Posted by *gregeas* /img/forum/go_quote.gif 
_I'm a fan of the DAC1 but coincidentally sold mine this week to fund a Slimdevices Transporter purchase._

 

Gregeas, interesting comment. I am planning to move to PC-based audio and am also contemplating the Transporter or a good USB DAC (was considering Stello 220 MkII, now also DAC1). What prompted you to go for the change?

 Thanks, Clark


----------



## darkless

For what it's worth, I decided to go with the Stello DA220 MkII, coming from my AQVOX USB 2 D/A MkI. It's currently on backorder, so I should hopefully receive it in 3 weeks.


----------



## lextek

So are you guys saying this DAC isn't worth consideration for USB use? As a "normal" DAC it was rated pretty high.


----------



## gregeas

Quote:


  Originally Posted by *clarkc* /img/forum/go_quote.gif 
_Gregeas, interesting comment. I am planning to move to PC-based audio and am also contemplating the Transporter or a good USB DAC (was considering Stello 220 MkII, now also DAC1). What prompted you to go for the change?

 Thanks, Clark_

 

Hey Clark, 

 My thinking was that the price of the Transporter was less than my Arcam CD33, DAC1, and Squeezebox. So off they went, and now I can use the Transporter as a high-end source for my PC music AND as a DAC for cable box and PS3. 

 I'm sold on PC audio and have been using Squeezeboxes in secondary systems for a quite a while. But until the Transporter arrived I was not ready to give up shiny disks in my main rig. Frankly, I'm surprised there isn't more talk about the Transporter. Very nice component.


----------



## edisonwu

Quote:


  Originally Posted by *Jon L* /img/forum/go_quote.gif 
_I don't know what they're smoking at Benchmark, but clearly their DAC-1 success has gone to their head. Such baseless, ridiculous hyperbole is sad to witness. It might help if they revealed what exactly is so stupendous about their USB implementation. 

 Their USB solution sounds like the usual run-of-the-mill USB/spdif chip that supports 24/96 (such as one in M-Audio Transit) that feeds their spdif receiver with the "ultralock" asynchronous upsampling circuit, which will feed their AD DAC chip. Big deal, and NO WAY is it worth $300 premium.

 Unless they start releasing some specific info about some new, wonder USB to I2S technology with custom software optimization, the hot air will definitely escape._

 

worthless to me as well. people buy DAC cost that much are expecting something designed to contribute to the sound quality.


----------



## Joey_V

Just wanted to add some premature impressions of the PS Audio Digital Link III DAC that my local audiophile friend just loaned me this afternoon....

 So far so good... the DAC is indeed a solid upgrade from the sound I'm getting straight from the Squeezebox3, I'm very happy about the improvement! I'm going to listen more to this DAC tomorrow and get better impressions, but as of now, the impressions are very favorable.

 I would say that this DAC at 1000$ is a good value especially since it has the USB inputs that many on this forum value. Plus this thing is hefty! Atleast 15 to 20lbs just for a DAC. Not too shabby.


----------



## Tunz

I found an ad on Audiogon for preorders of the Dac1 w/usb for $975

http://cls.audiogon.com/cgi-bin/cls....7386&demo&3&4&


----------



## educator

Quote:


  Originally Posted by *Tunz* /img/forum/go_quote.gif 
_I found an ad on Audiogon for preorders of the Dac1 w/usb for $975

http://cls.audiogon.com/cgi-bin/cls....7386&demo&3&4&_

 

In the title of the ad, it says that the price for preorders of the usb benchmark is $1275.


----------



## A.Thorsen

Quote:


  Originally Posted by *Davesrose* /img/forum/go_quote.gif 
_I'd be interested to hear the reviews. I had bought a USB to spdif converter so that I could do computer audio with my DAC-1. At first I thought mp3s sounded great through the Benchmark. Then I got into SACD and could hear a difference with a good CD transport. It's redbook playback for CDs sounds very good...so it would be interesting to see if there will be much 24bit audio sources for the USB version: seems like it might be a way to get 24bit DVD-Audio through the Benchmark. But since most albums out are for CD, I guess I'll just be using my current Benchmark as a CD DAC for sometime now. Would be great if I could get SACD out to it though._

 

I must be completely misreading this post.

 Going by your sig: 
 You're using the Benchmark DAC in your food chain right now to that Maverick SACD player and it only handles CD and nothing else?


----------



## Tunz

Under the asking price it says $975. May have to email for clarification.


----------



## EliasGwinn

Hi there! My name is Elias Gwinn, I'm an engineer at Benchmark Media Systems. I'm glad to see such a lively discussion about the new product!!

 I would like to answer some of the questions about the new DAC1 USB. 

 When we decided to add a USB interface to the DAC1, we purchased a 'boat-load' of USB audio interfaces to test and use. Our goal was to get familiar with the various technologies available and determine what we would want for our solution. 

 The testing consisted of the 'psuedo-random' bit-test that was mentioned in the press release. This is, quiet simply, testing "what-bits-go-in-and-what-bits-come-out". This is a standard test developed by Audio Precision, the leading audio electronics testing equipment manufacturer. When the Audio Precision (AP) sends a digital audio signal into a device, it checks to see if the exact same bits come out. So, for example, if the AP sends in 101100111000, a 'bit-transparent' data path will output the exact same bits: 101100111000. This was our testing proceedure. 

 An ideal transport will deliver to the DAC the original digital audio data bit-for-bit without a single bit changed. This is true for CD/DVD transports as much as computers. 

 Through our testing, we found that USB audio devices with custom drivers were rarely, if ever, bit-transparent. This includes ASIO devices. 

 Conversely, 'native' (without custom driver) devices were ALWAYS transparent. This was true with Windows and Mac. The problem with native solutions was the lack of 24-bit 96-kHz capablities. 

 At first we suspected this was a limitation with Windows USB audio. However, Microsoft audio software engineers assured us 24/96 was possible using the native USB driver (usbaudio.sys). Both Microsoft and Apple had a DAC1 USB prototype during the development period and our engineering team was in constant communication with their engineers. 

 After walking down the long, unpaved road towards a truely native 24/96 USB audio solution, it was finally achieved!!

 -Elias Gwinn


----------



## granodemostasa

part of my weariness of direct usb connectsion has been frequent complaints about digital noise and interference. for example, there are plenty of accounts of this occuring on Stello and wavelength products. what does the dac-1 do to keep this from happening? 
 have you done testing between usb>spdif converters v. your native USB dac? 

 thankyou


----------



## TreAdidas

Quote:


  Originally Posted by *The Monkey* /img/forum/go_quote.gif 
_Frankly, if I'm resorting to USB, I'll just use my MicroDAC. My Mac has optical out and the DAC1 handles that just fine.

 I'm disappointed that this was their big news. I thought we'd be hearing about a discrete headphone amp in the DAC1._

 

My thoughts as well. Anyone come up with a compelling enough reason to need to go over USB in the instance you have a computer equipped with an optical output?


----------



## The Monkey

Quote:


  Originally Posted by *EliasGwinn* /img/forum/go_quote.gif 
_Hi there! My name is Elias Gwinn, I'm an engineer at Benchmark Media Systems. I'm glad to see such a lively discussion about the new product!!
_

 

Elias,

 Thanks for taking the time to post in this thread. Welcome to Head-Fi and, er, sorry for what you do to our wallets. 
	

	
	
		
		

		
		
	


	




 Interesting point about working with the MS and Apple folks. How seriously do their engineers take their audio at the respective companies?


----------



## melomaniac

Quote:


  Originally Posted by *TreAdidas* /img/forum/go_quote.gif 
_My thoughts as well. Anyone come up with a compelling enough reason to need to go over USB in the instance you have a computer equipped with an optical output?_

 

hm... still depends on what you do downstream. I'm willing to bet that a nice (tubed?) USB DAC like one from wavelength is going to trounce whatever a cheaper (three-figure) DAC can do with your optical in. and all things being equal (i.e. let's set a hypothetical cost/quality inflection point for both DACs) I am further convinced that jitter is something that we ought to take seriously, even and especially in computer audio. USB has a firmer lock than many SPDIF solutions (especially if it means longer signal paths). for that reason, I own three amps that take USB input, and don't own a separate DAC anymore.


----------



## EliasGwinn

Yes, jitter is a very serious matter with computer audio. In fact, one interesting piece of information we found (though we did not confirm) is that computer manufacturers will ADD JITTER to their clocks for the purpose of spreading the energy across a wider bandwidth to pass emissions testing. Apparently the spike at the fundamental clock frequency was causing interference of some sort.

 This is detrimental to audio, as most digital audio connoisseurs will attest to. In the simplest terms, jitter causes the digital audio converter to misfire. It’s like a bad timing system on a car's engine causing the spark plugs to fire when the pistons are not quite ready for it.

 The Benchmark DAC1 and DAC1 USB are designed with a special clock recovery system that makes jitter irrelevant. The way it works, in simple terms, is - the DAC1 clock is not discretely attached to the signal clock, but instead monitors it and replicates it with an isolated clock which is extremely stable. This is described in more detail including performance curves in the manual (http://www.benchmarkmedia.com/dac1/DAC1-Manual.pdf).

 With this clocking system, the DAC1 will perform exactly the same no matter how much jitter is on the clock...quite literally. There aren't too many converters that can make that claim, but we encourage you to compare. It is for this reason that we tell our customers that it is not necessary to buy extremely expensive (<$100) digital cables. We've tested the DAC1 with cables of varying quality, and it made no difference whatsoever.

 What does this all have to do with USB, you may be asking (please excuse my rambling...can you tell I'm not in marketing)? We haven't yet tested the jitter on USB interfaces, but from everything we know about computer hardware architecture, all signs point to it being a very serious problem. 

 USB protocol was not designed to stream data fluently and consistent; it was designed to transfer 'bursts' of data. This is the reason why audio drop out and 'ticks' are common complaints from users of USB audio devices. The DAC1 USB interface software and buffers were designed with this in mind; it 'monitors' the data flow to maintain a consistent and fluid audio stream.

 The reason we announced this as our 'big news', is because this type of USB audio solution is the first of its kind with regard to the scope of the solution. It addresses more concerns then any interface of its kind. Even engineers at Microsoft and Apple were impressed!!


----------



## EliasGwinn

Quote:


  Originally Posted by *The Monkey* /img/forum/go_quote.gif 
_Elias,

 Thanks for taking the time to post in this thread. Welcome to Head-Fi and, er, sorry for what you do to our wallets. 
	

	
	
		
		

		
		
	


	




 Interesting point about working with the MS and Apple folks. How seriously do their engineers take their audio at the respective companies?_

 

Monkey (if that is, in fact, your real name...),

 MS and Apple folks take their audio very seriously. However, the beast that is called 'product developement' sometimes prevents a designer from optimizing every aspect of every function of their product. This can be for many reasons: cost, development time (time to market), stability, and intra-compatibility. 

 The latter (intra-compatibility) refers to something very common in computer hardware and software. That is, when a product performs such a wide array of functions as a computer does, the designers are often limited by other functions within the product. A quick and dirty example is: traffic build-up on a road may not be limited to how many lanes make up the road, but actually by an intersection with another road. Another example: the USB port is built to feed printers and audio devices!!

 -Elias

 ps. Monkey, I'm sorry about your wallet, but my 176,000 mile, 12-year old car can attest that it goes into the product, not our wallets 
	

	
	
		
		

		
		
	


	




 .


----------



## EliasGwinn

An addition to the last post:

 The Vista designers were VERY serious about audio. The sample-rate conversion in Vista has distortion artifacts no higher then -120dBFS!!


----------



## EliasGwinn

Quote:


  Originally Posted by *granodemostasa* /img/forum/go_quote.gif 
_part of my weariness of direct usb connectsion has been frequent complaints about digital noise and interference. for example, there are plenty of accounts of this occuring on Stello and wavelength products. what does the dac-1 do to keep this from happening? 
 have you done testing between usb>spdif converters v. your native USB dac? 

 thankyou_

 

Sorry for posting a million times in a row, but I just saw Granodemostasa's question. 

 Here's from a post I left about 10 min ago. It answers your question about typical USB audio problems...

  Quote:


  Originally Posted by *EliasGwinn* /img/forum/go_quote.gif 
_
 USB protocol was not designed to stream data fluently and consistent; it was designed to transfer 'bursts' of data. This is the reason why audio drop out and 'ticks' are common complaints from users of USB audio devices. The DAC1 USB interface software and buffers were designed with this in mind; it 'monitors' the data flow to maintain a consistent and fluid audio stream.

 The reason we announced this as our 'big news', is because this type of USB audio solution is the first of its kind with regard to the scope of the solution. It addresses more concerns then any interface of its kind. Even engineers at Microsoft and Apple were impressed!!_


----------



## gregeas

Very interesting thread. I recently sold my trusty Arcam CD player and now listen to PC audio exclusively. I've also had good experiences with the DAC1, but recently sold it because of its lack of USB input. For reasons I can't explain, I never really trusted the M-Audio Transit with an optical cable to the DAC1.

 Two questions for Benchmark:

 * Is it not true that Microsoft's Kmixer alters the bits as audio is output through it? If so, how does the DAC1 handle this without a driver?

 * How does USB audio on the DAC1 compare to other high-end PC audio products such as the Slimdevices Transporter or the Empirical Audio USB converters? Is jitter as low as on these other products?


----------



## EliasGwinn

hello gregeas!

 The short answer to the first question is: Kmixer will not affect the audio at all unless multiple audio streams are being played simultaneously. 

 Here is a summary of what kmixer does to audio:

 CONDITION 1: One audio stream is sent through audio through kmixer at a time

 RESULT: kmixer streams the audio bit-transparently - that is, bit-for-bit, what goes in, also comes out. We have tested and proven this using a test function called 'Bittest' by Audio Precision.

 CONDITION 2: Two audio streams of same sample rate are sent through kmixer

 RESULT: kmixer streams both without problems, ASSUMING THE SUM OF THE AUDIO STREAMS DOES NOT ECLIPSE 0 dBFS!! Just like any digital mixer, if the sum of the audio eclipses 0 dBFS, digital clipping will occur, which is not popular among audio enthusiasts. However, if it does not eclipse 0 dBFS, there should be no problems. This was confirmed by playing a 'Bittest' stream with one app and a silence (all 0's) stream with another. The result was bit-transparency. NOTE: When multiple audio streams are summed in kmixer (even 16-bit audio streams), the result will be a 24-bit audio stream. THIS IS WHAT WE WANT, assuming we have a 24-bit device to recieve it.

 CONDITION 3: Two audio streams with different sample rates are sent through kmixer

 RESULT: kmixer will up-sample the lower sample rate to the higher one. The higher one remains unaffected. This conversion is not very good though, and should be avoided. It is easily avoidable, however, as long as only one audio stream is playing at a time, or they are of equal sample-rates. But, who listens to more then one CD at a time anymore these days? (ok, Flaming Lips fans aside!!)

 To answer your 2nd question, Gregeas, we don't have either of the products you mentioned to test in our facility. I would have to see Jitter measurement curves to check. We have our measurement curves posted in the manual of the DAC1 (http://www.benchmarkmedia.com/dac1/DAC1-Manual.pdf).

 -Thanks for asking!!


----------



## A.Thorsen

Lurkers like me also appreciate Elias's time in threads like these. 
	

	
	
		
		

		
			





 It always helpful and appreciated when insiders and other folks like that participate with us 'round these parts.


----------



## gregeas

Thanks for the response, Elias. So in the real world the key is to output one audio stream at a time, through a software program like J River Media Center, correct? Will Windows system sounds playing at the same time (for example, if an email arrives in Outlook) result in Kmixer chicanery? My RME soundcard always went haywire in that situation, and I wonder if this is the cause. 

 Also, are there specific optimal settings for the media software when playing back FLAC files, for example? Is it ideal to upsample to the DAC1?


----------



## dw8083

Hi Elias,

 It would be great to see Benchmark come to the Head-fi show in San Jose in April!

 -David


----------



## The Monkey

Quote:


  Originally Posted by *dw8083* /img/forum/go_quote.gif 
_Hi Elias,

 It would be great to see Benchmark come to the Head-fi show in San Jose in April!

 -David_

 

It _will_ be great to see Benchmark at the International Meet.


----------



## EliasGwinn

Quote:


  Originally Posted by *gregeas* /img/forum/go_quote.gif 
_Thanks for the response, Elias. So in the real world the key is to output one audio stream at a time, through a software program like J River Media Center, correct? Will Windows system sounds playing at the same time (for example, if an email arrives in Outlook) result in Kmixer chicanery? My RME soundcard always went haywire in that situation, and I wonder if this is the cause. 

 Also, are there specific optimal settings for the media software when playing back FLAC files, for example? Is it ideal to upsample to the DAC1?_

 

One audio stream at a time is a good idea. Windows system sounds, though incredibly annoying, shouldn't cause any problems with the audio however. The sounds are 44/16, so as long as your audio is at least 44/1 kHz, your audio won't be converted. It will be invaded for the duration of the system sound only.

 As for recommended players, we will be implementing a section of the website outlining the popular players, their tested performance, and recommended configurations.


----------



## EliasGwinn

Quote:


  Originally Posted by *The Monkey* /img/forum/go_quote.gif 
_It will be great to see Benchmark at the International Meet._

 

yes, Benchmark will be there.


----------



## Andrew_WOT

Quote:


  Originally Posted by *EliasGwinn* /img/forum/go_quote.gif 
_USB protocol was not designed to stream data fluently and consistent; it was designed to transfer 'bursts' of data._

 

Isn't that the case for Bulk transfer type only, and not Isochronous?


----------



## greenleaf

Quote:


  Originally Posted by *EliasGwinn* /img/forum/go_quote.gif 
_As for recommended players, we will be implementing a section of the website outlining the popular players, their tested performance, and recommended configurations._

 

awsome 
	

	
	
		
		

		
		
	


	




 Foobar will be included, for sure?


----------



## smeggy

I wonder if these gems of information will finally put to rest all the debates about kmixer, asio and kernel streaming in regard to which sounds best.


----------



## gregeas

Yeah, after all the hoops I jumped through to avoid Kmixer, I find out its a myth...?


----------



## milkpowder

Quote:


  Originally Posted by *gregeas* /img/forum/go_quote.gif 
_Yeah, after all the hoops I jumped through to avoid Kmixer, I find out its a myth...?_

 

The myth has been _busted_! 
	

	
	
		
		

		
		
	


	




 I'm telling you, it's all placebo effect! Everyone can go back to using iTunes now


----------



## EliasGwinn

Yes, kmixer itself is relatively harmless. As for iTunes, you must be careful of a few things:

 1. iTunes volume control should always be set to 100%, as volume reduction in iTunes causes severe distortion. This is because volume calculations will result in 24-bit words, even if the audio was initially 16-bit (due to remainders after division). iTunes will then truncate to 16-bits, instead of dithering or simply passing 24-bits

 2. iTunes will convert the sample-rate of any audio with different sample-rates then that set in QuickTimes preferences (yes, QuickTime preferences affects iTunes!!) Be sure that sample-rate corresponds to the sample rate of the audio you are listening to.

 3. For the purest audio playback, do not use the "sound check" or "sound enhancer" features in iTunes. All of these DSP and/or audio plug-ins in all media players should be avoided to obtain faithful playback.

 All of this will be covered in our "Guide to Computer Audio", which will be available on our website soon. I'll let you know...

 edit: this refers to iTunes version 6. See our wiki for the most up-to-date information.


----------



## EliasGwinn

As for ASIO and/or kernel streaming, we haven't tested to prove or disprove specific performance of these technologies. We tested USB audio devices which used the ASIO protocol, and we were not satisfied with their performance. This could likely be the programmers, not the protocol (API). Nonetheless, as we discovered the potential that was inherent with native usbaudio.sys and kmixer, plus the added convenience of not having to deal with 3rd party drivers which may "argue" with other drivers on your computer, etc, we decided that "native" was the way to go.


----------



## EliasGwinn

Quote:


  Originally Posted by *Andrew_WOT* /img/forum/go_quote.gif 
_Isn't that the case for Bulk transfer type only, and not Isochronous?_

 

This is theoretically true, but USB wasn't really built for steady streaming, even in Isochronous. When you watch the stream on a bus analyzer, it becomes apparent. 

 I think what happens is...USB activity is sort of a queued process, and follows the queue with relative prorioty, etc. Therefore, when other activity takes priority, USB activity is compromised.

 I wish I knew with certainty, but even the most official publications on this will disagree with each other!! 

 Nonetheless, USB audio does suffer from 'ticks' and drop outs, which, for whatever reason, is due to insufficient streaming capabilities. When developing the DAC1 USB, we put several "checks" and buffers into place to monitor the stream and prevent these errors from occuring. We played audio through the DAC1 USB while taxing the processor of the computer with other apps, etc, and we couldn't get the DAC1 USB to tick or pop.


----------



## EliasGwinn

Quote:


  Originally Posted by *greenleaf* /img/forum/go_quote.gif 
_awsome 
	

	
	
		
		

		
			





 Foobar will be included, for sure?_

 

Yes, Foobar will be included. 

 In fact, we were very impressed with Foobar. It has a lot of really neat features that are specifically designed for faithful playback.


----------



## 5Kurt

Quote:


  Originally Posted by *EliasGwinn* /img/forum/go_quote.gif 
_This is theoretically true, but USB wasn't really built for steady streaming, even in Isochronous. When you watch the stream on a bus analyzer, it becomes apparent. _

 

Hi Elias, thanks for the very valuable information.

 What you have said for USB, can we assume the same for Firewire connections?


----------



## EliasGwinn

Quote:


  Originally Posted by *5Kurt* /img/forum/go_quote.gif 
_Hi Elias, thanks for the very valuable information.

 What you have said for USB, can we assume the same for Firewire connections?_

 

No, Firewire protocol is completely different. I don't know too much about Firewire yet, so I can't really tell you how they are different.


----------



## gregeas

Elias:

 Thanks for all of the valuable information here. 

 Do you have any observation on the quality of USB audio versus the quality of a good CD transport? I assume that as long as the bits are the identical, it all comes down to jitter. Can USB deliver a better signal to the DAC1 than the best transports?


----------



## 5Kurt

Elias :

 For people who already own a DAC-1 (not the USB version) and use PC as transport do you recommend SPDIF or Optical input?

 Thanks

 ps: sorry we ask you lot of things but we rarely find an engineer to ask such questions.


----------



## The Monkey

Also, Elias, what are your primary 'phones?


----------



## wang228

Quote:


  Originally Posted by *EliasGwinn* /img/forum/go_quote.gif 
_Here is a summary of what kmixer does to audio:

 CONDITION 1: One audio stream is sent through audio through kmixer at a time

 RESULT: kmixer streams the audio bit-transparently - that is, bit-for-bit, what goes in, also comes out. We have tested and proven this using a test function called 'Bittest' by Audio Precision.

 CONDITION 2: Two audio streams of same sample rate are sent through kmixer

 RESULT: kmixer streams both without problems, ASSUMING THE SUM OF THE AUDIO STREAMS DOES NOT ECLIPSE 0 dBFS!! Just like any digital mixer, if the sum of the audio eclipses 0 dBFS, digital clipping will occur, which is not popular among audio enthusiasts. However, if it does not eclipse 0 dBFS, there should be no problems. This was confirmed by playing a 'Bittest' stream with one app and a silence (all 0's) stream with another. The result was bit-transparency. NOTE: When multiple audio streams are summed in kmixer (even 16-bit audio streams), the result will be a 24-bit audio stream. THIS IS WHAT WE WANT, assuming we have a 24-bit device to recieve it.
_

 

Aah.. thank you. Finally things start to make sense for me why my dac indicates 44.1K sampling rate from USB even without bypassing kmixer. Somebody suggested it was the computer's unreliable clock, but i have been skeptical.


----------



## smeggy

Quote:


  Originally Posted by *The Monkey* /img/forum/go_quote.gif 
_Also, Elias, what are your primary 'phones?_

 

Hmmmm


----------



## EliasGwinn

I appreciate the questions, but I apologize in advance if I can't answer them all in a timely manner. 

 USB vs. TRANSPORT: Its a broad comparison, but it really depends on which transport, which USB audio device, and which DAC. If we're talking about the DAC1 USB, they will all perform equally well. A high quality transport will induce less jitter (notice I said 'high quality', not 'expensive'), but the DAC1 is truely immune to jitter. When it was designed, a test signal with TONS of jitter purposely added was used to really test the limits of the DAC1. And, quite literally, there was no performance degradation. With all that being said, a high quality transport will offer better error correction for disc errors. If the DAC1 is used, the only reason to spend money on a high quality transport is for better error correction. The USB solution, however, should be every bit (excuse the pun) as stable as a good transport, provided the harddrive isn't too fragmented, etc, etc.

 SPDIF vs. OPTICAL: This is also a broad comparison, and I would say it depends on which interface you are using. I wouldn't say that one is universally superior to to the other. However, with regards to the jitter with these interfaces, as mentioned before, it is not a concern with the DAC1.


----------



## EliasGwinn

Quote:


  Originally Posted by *The Monkey* /img/forum/go_quote.gif 
_Also, Elias, what are your primary 'phones?_

 

I do most (98%) of my listening through studio monitors (JBL 6332, Tannoy Reveal 6, and K&H 4-ways). I hope my lack of headphone experience doesn't disqualify me from this forum 
	

	
	
		
		

		
		
	


	




 .

 The headphones we have here at Benchmark are the Sennheiser 650 and 600's, as well as a few different Ultrasone's. At the recording studio where I work, we have the AKG K 240's and Sony MDR7509HD's.


----------



## 5Kurt

Quote:


  Originally Posted by *EliasGwinn* /img/forum/go_quote.gif 
_SPDIF vs. OPTICAL: This is also a broad comparison, and I would say it depends on which interface you are using. I wouldn't say that one is universally superior to to the other. However, with regards to the jitter with these interfaces, as mentioned before, it is not a concern with the DAC1._

 

Can you give us any recomendations about which USB interface we should use. (SPDIF and/or Optical)

 I think this is the key question and we end users get very limited reliable information.


----------



## EliasGwinn

Quote:


  Originally Posted by *5Kurt* /img/forum/go_quote.gif 
_Can you give us any recomendations about which USB interface we should use. (SPDIF and/or Optical)

 I think this is the key question and we end users get very limited reliable information._

 

As for a specific product, I don't feel comfortable recommending one because I'm not sure what our company policy is with that regard. It's a fair question, but its one I don't know how to answer fairly. Please forgive me.


----------



## 5Kurt

Quote:


  Originally Posted by *EliasGwinn* /img/forum/go_quote.gif 
_As for a specific product, I don't feel comfortable recommending one because I'm not sure what our company policy is with that regard. It's a fair question, but its one I don't know how to answer fairly. Please forgive me._

 

What I understood from your postings are, we should choose the interfaces that uses native usb drivers.

 Is it correct?


----------



## audioengr

Quote:


  Originally Posted by *EliasGwinn* /img/forum/go_quote.gif 
_This is theoretically true, but USB wasn't really built for steady streaming, even in Isochronous. When you watch the stream on a bus analyzer, it becomes apparent. 

 I think what happens is...USB activity is sort of a queued process, and follows the queue with relative prorioty, etc. Therefore, when other activity takes priority, USB activity is compromised.

 I wish I knew with certainty, but even the most official publications on this will disagree with each other!! 

 Nonetheless, USB audio does suffer from 'ticks' and drop outs, which, for whatever reason, is due to insufficient streaming capabilities. When developing the DAC1 USB, we put several "checks" and buffers into place to monitor the stream and prevent these errors from occuring. We played audio through the DAC1 USB while taxing the processor of the computer with other apps, etc, and we couldn't get the DAC1 USB to tick or pop._

 


 Does this mean that you are using asynchronous USB protocol for Win2000, XP and Vista?

 Steve N.


----------



## thomaspf

Hi Elias,

 It's always great to have a company stand behinds it products but some of what you posted here differs a bit from my experiences. So let me question some of the statements you made in order to avoid any confusions on this forum.

 1. If your sound card driver uses kmixer than the bits will get altered even if only a single stream is playing. What procedure did you use to test this otherwise? You'd be the first to come to a different conclusion.

 2. The Windows standard USB driver uses kmixer and any USB audio device using the standard driver will not work bit perfect unless you use kernel streaming. Vista which does not have kmixer anymore uses a different mixer but is still not bit perfect. Is Benchmark shipping with a different USB driver?

 3. I was also intrigued by your statement of a clock recovery system in the DAC1. Is that a new feature in the USB model. As far as I understood up to now, the DAC1 does not use any form of clock recovery at all but is using an AD1896 asynchronous sample rate converter running at a fixed frequency. While that reduces jitter is also changes all the samples in the process and therefore the DAC1 never actually plays bit perfect. Is that not correct?

 Cheers

 Thomas


----------



## Davesrose

Quote:


  Originally Posted by *EliasGwinn* /img/forum/go_quote.gif 
_I do most (98%) of my listening through studio monitors (JBL 6332, Tannoy Reveal 6, and K&H 4-ways). I hope my lack of headphone experience doesn't disqualify me from this forum 
	

	
	
		
		

		
			





 .

 The headphones we have here at Benchmark are the Sennheiser 650 and 600's, as well as a few different Ultrasone's. At the recording studio where I work, we have the AKG K 240's and Sony MDR7509HD's._

 









 Since you guys primarily use Senns, you obviously have a good ear and taste
	

	
	
		
		

		
		
	


	







 Must be why I like the DAC-1 with my Senns.
	

	
	
		
		

		
		
	


	




 Grados aren't so bad either
	

	
	
		
		

		
		
	


	




 As a Benchmark owner, this is proving a gread thread to read. You're certainly more then welcome here to provide insight on the specifications of Benchmark products. Welcome to the forum, Elias!

 Dave


----------



## greenleaf

Quote:


  Originally Posted by *EliasGwinn* /img/forum/go_quote.gif 
_With all that being said, a high quality transport will offer better error correction for disc errors. If the DAC1 is used, the only reason to spend money on a high quality transport is for better error correction. The USB solution, however, should be every bit (excuse the pun) as stable as a good transport, provided the harddrive isn't too fragmented, etc, etc.
_

 

an option is to play your music from the RAM rather than from the HDD. Foobar can do it: buffering option

 and of course rip your file with EAC in secure mode to a lossless format.

 if you do it this way I doubt a CD transport would be better.


----------



## greenleaf

Quote:


  Originally Posted by *thomaspf* /img/forum/go_quote.gif 
_Hi Elias,

 It's always great to have a company stand behinds it products but some of what you posted here differs a bit from my experiences. So let me question some of the statements you made in order to avoid any confusions on this forum.

 1. If your sound card driver uses kmixer than the bits will get altered even if only a single stream is playing. What procedure did you use to test this otherwise? You'd be the first to come to a different conclusion.

 2. The Windows standard USB driver uses kmixer and any USB audio device using the standard driver will not work bit perfect unless you use kernel streaming. Vista which does not have kmixer anymore uses a different mixer but is still not bit perfect. Is Benchmark shipping with a different USB driver?


 Thomas_

 

I'm pretty sure he said that they used an AP to test their dac.(http://ap.com)


----------



## milkpowder

Quote:


  Originally Posted by *greenleaf* /img/forum/go_quote.gif 
_an option is to play your music from the RAM rather than from the HDD. Foobar can do it: buffering option_

 

Didn't know you could do that; I'm going to set mine to 100MB and go from there. Thanks a lot!

 How is that different from a long output buffer length eg 8 seconds.


----------



## greenleaf

Quote:


  Originally Posted by *milkpowder* /img/forum/go_quote.gif 
_Didn't know you could do that; I'm going to set mine to 100MB and go from there. Thanks a lot!

 How is that different from a long output buffer length eg 8 seconds._

 

well, with this option you basically buffer the whole song 
	

	
	
		
		

		
		
	


	




 .
 and foobar will only buffer one song at a time (well, I think) so using a very high value is useless; unless you rip your music to image [one file per cd] + cue sheet; then I think it will basically buffer the whole cd but you'll need like 300mb RAM if you rip to lossless 
	

	
	
		
		

		
		
	


	




. I don't rip to image+cue, I rip to single flac files so I haven't tested this...


----------



## milkpowder

kk... I just set my buffer to the largest music file I'd potentially be playing, which is 135MB. How does the buffer work if I use upsampling (which I don't, but theoretically speaking)? Would the file be upsampled before being buffered or upsampled as the data is being read from the ram?

 BTW, I yearn for a Benchy DAC1... I have no money though and the UK dealer's prices are extortionate.


----------



## EliasGwinn

Quote:


  Originally Posted by *5Kurt* /img/forum/go_quote.gif 
_What I understood from your postings are, we should choose the interfaces that uses native usb drivers.

 Is it correct?_

 

5Kurt, 

 From our experiences, native drivers are bit transparent for 16-bit only (DAC1 USB not included...it is transparent at 24-bit). This is dangerous even when playing 16-bit files because volume adjustments in software, etc result in 24-bit words due to division. On the contrary, we rarely achieved bit-transparency with devices using custom drivers. But they were capable of 24-bit. The custom-driver devices, though not transparent, did not result in significant distortion. But, at the same time, all the bits are not making it through ok.

 The ideal solution would be a 24-bit native solution...hence our design goal.


----------



## asdf

Has anyone used the DAC1 (USB) while running linux? Any issues?

 --asdf


----------



## EliasGwinn

Quote:


  Originally Posted by *thomaspf* /img/forum/go_quote.gif 
_Hi Elias,

 It's always great to have a company stand behinds it products but some of what you posted here differs a bit from my experiences. So let me question some of the statements you made in order to avoid any confusions on this forum.

 1. If your sound card driver uses kmixer than the bits will get altered even if only a single stream is playing. What procedure did you use to test this otherwise? You'd be the first to come to a different conclusion.

 2. The Windows standard USB driver uses kmixer and any USB audio device using the standard driver will not work bit perfect unless you use kernel streaming. Vista which does not have kmixer anymore uses a different mixer but is still not bit perfect. Is Benchmark shipping with a different USB driver?

 3. I was also intrigued by your statement of a clock recovery system in the DAC1. Is that a new feature in the USB model. As far as I understood up to now, the DAC1 does not use any form of clock recovery at all but is using an AD1896 asynchronous sample rate converter running at a fixed frequency. While that reduces jitter is also changes all the samples in the process and therefore the DAC1 never actually plays bit perfect. Is that not correct?

 Cheers

 Thomas_

 

Thomas,

 I answered your first two questions in some detail in a previous post:

http://www.head-fi.org/forums/showpo...4&postcount=39

 But here's the testing method:

 "The testing consisted of the 'psuedo-random' bit-test that was mentioned in the press release. This is, quiet simply, testing "what-bits-go-in-and-what-bits-come-out". This is a standard test developed by Audio Precision, the leading audio electronics testing equipment manufacturer. When the Audio Precision (AP) sends a digital audio signal into a device, it checks to see if the exact same bits come out. So, for example, if the AP sends in 101100111000, a 'bit-transparent' data path will output the exact same bits: 101100111000. This was our testing proceedure."

 As for the question about the DAC1 clocking, it is true that we convert the sample-rate to a rate which the D-to-A chip is most efficient. Your assumption that the D-to-A is not getting a bit-perfect data stream is correct, but this is by design. A converter chip is going to perform best at a specific frequency, due simply to the real-life limitations of semiconductors. So from a distortion performance stand-point, it is best to convert to analog at the chips "favorite" sample-rate. 

 So, this raises the question of, "Why is it important to get bit-transparent audio from the computer if its not bit-transparent when it gets to the D-to-A chip?" 

 The answer, of course, is - who knows whats happening to audio when going through a computer system, etc. Its so hard to tell whats happening, and why, and what affect its having on the audio. All we know is, we want the audio to come out untouched, and there is no reason why it shouldn't be the case. Also, you can be sure that any signal processing happening behind the scenes in the computer is not done with the D-to-A's best interest in mind as it is with the DAC1. 

 Unlike in a computer, we know what is happening to the audio within the DAC1, every step of the way. And, everything that is happening is done with absolute care and specific design with the goal in mind to achieve the most accurate (least distortion) conversion possible. If the ideal D-to-A chip existed that performed equally well at all sample-rates, we would be using it, and so would every other D-to-A manufacturer. Unfortunately, real-life limitations must be taken into account. Well, perhaps I should say fortunately, because thats what makes an engineer's job special, and exciting, and challenging. And, I should say, secure 
	

	
	
		
		

		
		
	


	




 .


----------



## gregeas

So does this mean that Kmixer only alters the bits in non-USB soundcards (that are not using custom drivers), whereas in straight USB output from a PC, Kmixer doesn not alter the bits?

 Haven't we known all along that Kmixer didn't affect USB output?


----------



## A.Thorsen

Quote:


  Originally Posted by *audioengr* /img/forum/go_quote.gif 
_Does this mean that you are using asynchronous USB protocol for Win2000, XP and Vista?

 Steve N._

 

^^ Good question.


----------



## thomaspf

Quote:


  Originally Posted by *EliasGwinn* 
_"The testing consisted of the 'psuedo-random' bit-test that was mentioned in the press release. This is, quiet simply, testing "what-bits-go-in-and-what-bits-come-out". This is a standard test developed by Audio Precision, the leading audio electronics testing equipment manufacturer. When the Audio Precision (AP) sends a digital audio signal into a device, it checks to see if the exact same bits come out. So, for example, if the AP sends in 101100111000, a 'bit-transparent' data path will output the exact same bits: 101100111000. This was our testing proceedure."_

 

Hi Elias,


 thanks for the follow-up. I am a bit familiar with the AP but ours does not have a USB sound card attached to it and even if it would I am unclear how you wold check for bit transparency with the DAC1. Let's assume you feed it a stream of whatever bits, how do you feed those bits back to the AP for comparison.

 I have done this test by sending bits from one sound card to another and comparing the results for a couple of USB sound cards. My finding is that the bit stream is changed at the sender unless you use kernel streaming. Another quick test that is being used quite often is to send HDCD encoded data or DTS encoded PCM streams to a digital interface and check for the results via a HDCD capabale DAC or surround sound decoder. I think everyone agrees there is no good reason why the data is changed in the computer when only one stream is playing, it unfortunately is changed in most cases.

 Since the DAC1 does not have a digital passthrough, does not provide HDCD decoding, and is stereo only your claims on bit transparency are a bit hard to challenge but allow me to assume you might be in error unless you provide some new insights. Maybe if you or someone with a USB version of the DAC1 would disclose the USB chip being used we can close this issue.

 I believe on the clock recovery we basically say the same thing. The DAC1 does not have any clock recovery but relies on a standard off the shelf asynchronous sample rate conversion chip like many other designs as well.

 Cheers

 Thomas


----------



## 5Kurt

Quote:


  Originally Posted by *EliasGwinn* /img/forum/go_quote.gif 
_5Kurt, 
 From our experiences, native drivers are bit transparent for 16-bit only (DAC1 USB not included...it is transparent at 24-bit). This is dangerous even when playing 16-bit files because volume adjustments in software, etc result in 24-bit words due to division. On the contrary, we rarely achieved bit-transparency with devices using custom drivers. But they were capable of 24-bit. The custom-driver devices, though not transparent, did not result in significant distortion. But, at the same time, all the bits are not making it through ok.

 The ideal solution would be a 24-bit native solution...hence our design goal._

 

Elias, thank you very much for the answers.

 I got one last question. How is Vista compared to WinXp sonically with DAC-1. Is it worth the upgrade or shall we stick with XP?


----------



## EliasGwinn

The AP does not output USB. We play a "bit-test" audio file using a media player (Foobar, for example) through the stock USB port on the computer. Then we convert the I2S output from the USB chip (TAS1020B) to SPDIF and send that back to the AP. The AP compares the SPDIF signal to the bit-sequence it uses to create the "bit-test" audio file. It reports any errors - that is, descrepancies in digital data - to assure that it is the same data, bit-for-bit.


----------



## EliasGwinn

Quote:


  Originally Posted by *5Kurt* /img/forum/go_quote.gif 
_Elias, thank you very much for the answers.

 I got one last question. How is Vista compared to WinXp sonically with DAC-1. Is it worth the upgrade or shall we stick with XP?_

 

Vista handles native audio totally different then XP, but not necessarily better. Where XP will pass the audio at the original sample-rate (or the highest presented to kmixer), Vista has a setting to choose the output sample-rate. It then converts everything to that sample-rate. The sample-rate conversion is very well designed, however, and although it is not bit-transparent, it causes very little distortion.

 To answer your question about upgrading, I don't think its necessary for the sake of audio, but things may change in the near future, as fixes and updates come out.

 I should add that all the Vista testing we did was on Vista Beta 2, so there may be things different in the current release then what we had.


----------



## audioengr

Quote:


  Originally Posted by *EliasGwinn* /img/forum/go_quote.gif 
_The AP does not output USB. We play a "bit-test" audio file using a media player (Foobar, for example) through the stock USB port on the computer. Then we convert the I2S output from the USB chip (TAS1020B) to SPDIF and send that back to the AP. The AP compares the SPDIF signal to the bit-sequence it uses to create the "bit-test" audio file. It reports any errors - that is, descrepancies in digital data - to assure that it is the same data, bit-for-bit._

 

That's funny. I see how you made the measurement and it makes sense. I also use the TAS1020 in my products and I definitely hear a degrading difference when using the Windows drivers on XP rather than a custom driver. In fact, I also hear a difference when I bypass KMIXER. The driver makes a bigger difference in my case though.

 Steve N.


----------



## audioengr

Quote:


  Originally Posted by *EliasGwinn* /img/forum/go_quote.gif 
_Vista handles native audio totally different then XP, but not necessarily better. Where XP will pass the audio at the original sample-rate (or the highest presented to kmixer), Vista has a setting to choose the output sample-rate. It then converts everything to that sample-rate. The sample-rate conversion is very well designed, however, and although it is not bit-transparent, it causes very little distortion.

 To answer your question about upgrading, I don't think its necessary for the sake of audio, but things may change in the near future, as fixes and updates come out.

 I should add that all the testing we did was on Vista Beta 2, so there may be things different in the current release then what we had._

 


 You say that ALL the testing you did was on Vista. Does this include the bit-comparison testing of the driver?

 Did you not do the same bit comparison using XP?

 Steve N.


----------



## thomaspf

Quote:


 That's funny. I see how you made the measurement and it makes sense. I also use the TAS1020 in my products and I definitely hear a degrading difference when using the Windows drivers on XP rather than a custom driver. In fact, I also hear a difference when I bypass KMIXER. The driver makes a bigger difference in my case though.

 Steve N. 
 

Agreed!

 Unlesss you use kernel streaming to bypass kmixer on XP the bit stream is modified.

 If you use the directsound output on foobar you will go through kmixer if you select direct kernel streaming either directly or via an ASIO shim you will bypass it. On Vista the mixer has much better quality but it still changes the stream even if you have selected the matching sampling rate. On Vista you can use exclusive mode in addition to kernel streaming to bypass the mixer but I am unaware of any player app outside the pro music space.

 I don't see how a TAS1020 could be tweaked to work any other way with the USBaudio.sys driver.

 Cheers

 Thomas


----------



## EliasGwinn

Quote:


  Originally Posted by *audioengr* /img/forum/go_quote.gif 
_You say that ALL the testing you did was on Vista. Does this include the bit-comparison testing of the driver?

 Did you not do the same bit comparison using XP?

 Steve N._

 

Ahh...thank you for pointing out that mis-statement. What I meant to say is we did all Vista testing on Vista Beta 2. We did all of the same tests on all of the major operating systems:

 -Mac OSX
 -Windows Vista (Beta 2)
 -Windows XP (SP2)
 -Windows 2000 (SP4)

 Hope that answers your question!


----------



## asdf

Quote:


  Originally Posted by *EliasGwinn* /img/forum/go_quote.gif 
_Ahh...thank you for pointing out that mis-statement. What I meant to say is we did all Vista testing on Vista Beta 2. We did all of the same tests on all of the major operating systems:

 -Mac OSX
 -Windows Vista (Beta 2)
 -Windows XP (SP2)
 -Windows 2000 (SP4)

 Hope that answers your question!_

 

. . . linux? Any testing on linux???? Anything????


----------



## Jon L

Quote:


  Originally Posted by *thomaspf* /img/forum/go_quote.gif 
_Agreed!

 On Vista the mixer has much better quality but it still changes the stream even if you have selected the matching sampling rate. _

 

That's counterintuitive that if you set the Vista to 44.1kHz, it'll still "resample" to 44.1kHz, and I don't know why they didn't design it to just bypass Vista mixer in this scenario. 

 Same thing happens with SRC (Secret Rabbit Code) resampler in Foobar v.0.8.3. If you set it to 44.1kHz and activate it, it changes the 44.1kHz stream with audible difference. I don't know what to call it, but I kind of prefer Foobar with SRC "resampling" to 44.1kHz, which sounds a bit smoother and richer.


----------



## EliasGwinn

Quote:


  Originally Posted by *audioengr* /img/forum/go_quote.gif 
_That's funny. I see how you made the measurement and it makes sense. I also use the TAS1020 in my products and I definitely hear a degrading difference when using the Windows drivers on XP rather than a custom driver. In fact, I also hear a difference when I bypass KMIXER. The driver makes a bigger difference in my case though.

 Steve N._

 

Although I wouldn't dispute what your ears tell you, we found that digital audio data could be streamed bit-for-bit perfectly through kmixer when using usbaudio.sys. As mentioned before, this was proven using a digital audio bit stream with a known bit sequence, and it was found to be identical coming out as it was going in. Aside from the bit-test, we also measured the audio performance (Freq response, THD, IMD, etc) of this setup, and it was completely similar to the all other digital inputs of the DAC1 up to 96/24. The third test was listening. I am still conducting that test as we speak, as are several others here at Benchmark. With a CD transport feeding the Coax input, and the computer feeding the USB with the same music, no one has been able to differentiate the two inputs.

 With that being said, there are several things that can affect your audio within a computer besides simply going through kmixer. However, we have found that when everything is configured properly, the digital audio can be streamed transparently through kmixer.

 We will be publishing an online Wiki that will document everything we have found with various media players, OS's, etc. 

 Also, with all due respect, the Windows driver is only one aspect of the streaming technology. I am not familiar with your product, but the firmware for the TAS1020B is the main difference between our solution and others on the market. The majority of the development time of this product was spent programming the firmware for that chip. It is a completely custom solution designed to handle streaming USB audio gracefully.


----------



## EliasGwinn

Quote:


  Originally Posted by *asdf* /img/forum/go_quote.gif 
_. . . linux? Any testing on linux???? Anything???? 
	

	
	
		
		

		
		
	


	


_

 

Not yet. We don't have a linux machine set up, but I am considering doing that in the near future. I'll try to keep you all updated, but the new Benchmark Wiki should have an announcement when any new developments or tests happen.


----------



## thomaspf

While the ability to hear any differences between a bit perfect stream and a kmixer modified stream is one thing that is a bit different from the question of being bit perfect.

 Deleted the rest to work through the follow-up


----------



## The Monkey

Quote:


  Originally Posted by *EliasGwinn* /img/forum/go_quote.gif 
_I do most (98%) of my listening through studio monitors (JBL 6332, Tannoy Reveal 6, and K&H 4-ways). I hope my lack of headphone experience doesn't disqualify me from this forum 
	

	
	
		
		

		
		
	


	




 .
_

 

Not at all. Come to the NYC Meet tomorrow and we'll get you sorted out.


----------



## koob

Quote:


  Originally Posted by *thomaspf* /img/forum/go_quote.gif 
_there is no question that kmixer modifies 16/44.1 PCM streams and there is nothing you can do about it short of bypassing kmixer._

 

1) Microsoft documentation clearly states that kmixer will pass a single audio stream through unaltered if the destination device supports the incoming sample rate.
 2) Elias states that in Benchmark's testing, the stream recovered from the USB passthrough was bit-for-bit identical to the stream heading into the kmixer/usbaudio stack.

 In light of these 2 facts, how do you subtantiate your claim that "there is no question that kmixer modifies" the stream?

  Quote:


  Originally Posted by *thomaspf* /img/forum/go_quote.gif 
_Your test might not be perfect in that you don't catch these modifications but you can take my word for it that they are different._

 

Again, Benchmark's test compares the outgoing digital stream with the recovered digital stream and Elias states these were bit-for-bit identical. Are you suggesting the test is "not perfect" in that they can't tell the difference between a 1 and a 0 at a given stream position? Since the entire test is in the digital domain, it doesn't leave room for golden-ear testing/opinion. Are we supposed to take your word over direct binary comparison?


----------



## thomaspf

I have also spend a long time doing testing. Something does not add up, so we just need to get to the bottom of it but there is nothing to get emotional about. 

 Maybe I am wrong or the by pure accident the pseudo random test stream happens to get passed through unharmed. I am not doubting that the DAC1 is a great product and I always like computer focused audio devices or I would not be here. We will just have to find out.


 Cheers

 Thomas


----------



## smeggy

Well one thing's for sure, Kmixer cannot alter a stream and pass it through unchanged at the same time. If numerous tests show identical bit transfer then apparently Kmixer does not alter the signal unless something else is interfering, which is what Benchmark found and what MS is saying. 

 I tend to believe specific tests that prove a case. I don't know what tests you have performed to come to a different result but I'm interested to know as this might help us avoid compromised sound in the future, if it is indeed compromised.

 Sometimes I wonder about anecdotal results because some claim that different media players sound better than others when using the same streams and same output devices which seems like total nonsense to me. Same with claims that KS and ASIO sound different when they are outputting identical streams through identical devices.

 It would certainly be nice to put this all too often misunderstood subject to bed.


----------



## thomaspf

That is exactly what we are trying to do. I have performed a bunch of different tests all with the consistent results. For the USB specific tests I have been using an Audiotrak Optoplay which has been on the market for a couple of years and which happens to also use an TUS1020 chip that uses the standard USBaudio.sys driver up to 24/96. I don't believe there is any special firmware used on the converter.


 1. Recording a stream send out via the Audiotrak Optoplay. The resulting stream is different when using Directsound (kmixer) and is identical when using kernel streaming.

 2. Playback of PCM wav streams with an embedded DTS signal to a surround sound receiver. The receiver can not lock on to the surround encoding with Directsound but the same stream works just fine with kernel streaming.

 3. Playback of the udial test clip. I get completely black background with kernel streaming but distortions with Directsound.

 Cheers

 Thomas


----------



## lowmagnet

Quote:


  Originally Posted by *Herandu* /img/forum/go_quote.gif 
_Oh boy. More snake oil. The line output is voltage dependent, not current dependent. The input to your amplifier is specified in terms of Volts. Like 2V RMS. SO why do they need a high current going in t perhaps a 47K input on your amp? To blow your amp input up? 
 I pass._

 

They're referring to the balanced XLR outputs, not the line outputs:

  Quote:


  Originally Posted by *Elias Gwinn, Engineer, Benchmark Systems (on AudioAsylum)* 
_High-current output drivers. The balanced output drivers are designed to handle low impedance, high capacitance, and/or high-inductance loads without the THD+N suffering._

 

Source


----------



## wshtb

Guys, you don't need KernelStreaming or ASIO for bit-identical output. I had a Audigy NX usb sound card in the past and it plays 16bit 44.1k DTS CD fine using DirectSound and foobar. My receiver will show DTS when a DTS track comes on. If it is not bit-perfect the sound will be white noise.

 Coincidentally, I never had much luck with KernelStreaming or ASIO myself. They simply do not work at all for me.


----------



## thomaspf

Are you using the Microsoft USB driver for the Audigy? The Transit also works bit perfect but it also comes with it's own driver.

 Do you actually mean the Audigy 2NX? When I last tried that model it did not work bit perfect. If this is true that would be fantastic news, since that model is one of the very few that is using asynchronous mode for the USB audio. This means that the master clock of the playback is in the Audigy and the PC is slaved to it via a back channel that performs the rate control.

 Cheers

 Thomas


----------



## FRANKe

Quote:


  Originally Posted by *EliasGwinn* /img/forum/go_quote.gif 
_The short answer to the first question is: Kmixer will not affect the audio at all unless multiple audio streams are being played simultaneously. 

 Here is a summary of what kmixer does to audio:

 CONDITION 1: One audio stream is sent through audio through kmixer at a time

 RESULT: kmixer streams the audio bit-transparently - that is, bit-for-bit, what goes in, also comes out. We have tested and proven this using a test function called 'Bittest' by Audio Precision.

 CONDITION 2: Two audio streams of same sample rate are sent through kmixer

 RESULT: kmixer streams both without problems, ASSUMING THE SUM OF THE AUDIO STREAMS DOES NOT ECLIPSE 0 dBFS!! Just like any digital mixer, if the sum of the audio eclipses 0 dBFS, digital clipping will occur, which is not popular among audio enthusiasts. However, if it does not eclipse 0 dBFS, there should be no problems. This was confirmed by playing a 'Bittest' stream with one app and a silence (all 0's) stream with another. The result was bit-transparency. NOTE: When multiple audio streams are summed in kmixer (even 16-bit audio streams), the result will be a 24-bit audio stream. THIS IS WHAT WE WANT, assuming we have a 24-bit device to recieve it.

 CONDITION 3: Two audio streams with different sample rates are sent through kmixer

 RESULT: kmixer will up-sample the lower sample rate to the higher one. The higher one remains unaffected. This conversion is not very good though, and should be avoided. It is easily avoidable, however, as long as only one audio stream is playing at a time, or they are of equal sample-rates. But, who listens to more then one CD at a time anymore these days? (ok, Flaming Lips fans aside!!)_

 

 Quote:


 One audio stream at a time is a good idea. Windows system sounds, though incredibly annoying, shouldn't cause any problems with the audio however. The sounds are 44/16, so as long as your audio is at least 44/1 kHz, your audio won't be converted. It will be invaded for the duration of the system sound only. 
 

This is contrary to what I've read elsewhere. First of all, I've checked my Windows system sounds (C:\Windows\Media in XP) and they are mostly 16bit 22kHz and some are mono. And according to a snippet from a blog by Larry Osterman (an employee of MS working in the Windows Media and Devices Group, working on the core windows audio engine for Longhorn), all sounds will down-sample to the lowest common denominator (in Windows XP).
http://blogs.msdn.com/larryosterman/...20/471872.aspx
  Quote:


 Before Vista, the kernel audio stack set the output audio format to match the format of the audio being played. Normally, this isn't a problem, since it means that we do less DSP of the signals. Unfortunately, it can lead to some rather unanticipated consequences. For instance, if you're playing a system sound (usually stereo, 22kHz), at the same time you start playing your MP3 files, then the MP3 file rendering happens at 22kHz, which is a noticeable degradation of audio quality. Once the audio system goes quiet, the rendering format will reset to the format of the content being played, but that may be quite some time later. 
 

Elias, can you explain the discrepancy? Are your comments based on Vista only?

 -FRANKe


----------



## wshtb

Yes it was Audigy 2 NX. I did install the latest driver from Creative. To get bit-identical output for DTS CDs I need to manually set the SPDIF output frequency to 44.1k. It worked pretty well, even under linux. The only problem for Linux is that the SPDIF output frequency can only be set in Windows. Once set, I can reboot the machine into Linux and it will stay at 44.1k as long as the power is connected at all time. If it loose power, the SPDIF output revert back to 48khz.

 Unfortunately my unit died a year ago 

  Quote:


  Originally Posted by *thomaspf* /img/forum/go_quote.gif 
_Are you using the Microsoft USB driver for the Audigy? The Transit also works bit perfect but it also comes with it's own driver.

 Do you actually mean the Audigy 2NX? When I last tried that model it did not work bit perfect. If this is true that would be fantastic news, since that model is one of the very few that is using asynchronous mode for the USB audio. This means that the master clock of the playback is in the Audigy and the PC is slaved to it via a back channel that performs the rate control.

 Cheers

 Thomas_


----------



## thomaspf

Maybe this should go into a seperate thread but I wonder whether anyone else could reproduce this behavior with the 2NX.

 Cheers

 Thomas


----------



## Wavelength

Quote:


  Originally Posted by *thomaspf* /img/forum/go_quote.gif 
_I have also spend a long time doing testing. Something does not add up, so we just need to get to the bottom of it but there is nothing to get emotional about. 

 Maybe I am wrong or the by pure accident the pseudo random test stream happens to get passed through unharmed. I am not doubting that the DAC1 is a great product and I always like computer focused audio devices or I would not be here. We will just have to find out._

 

Thomas,

 FYI the test I think they are talking about works this way. At least this is what the AP test engineer told me when they were trying to sell me one of thier units.

 They give you 2 files a MP3 test file and a compressed WAV file of test data. As Elias said they can play these files via any player and test the SPDIF stream. Or you can test at the analog outputs for jitter and stuff.

 I chose the dScope III from Prism over the AP unit because it can kernel stream to any USB, Firewire or board level product and then test SPDIF or Analog.

 In my experience over the last few weeks, I agree with you that I don't see bit perfect signals coming out of the KMIXER. Actually sometimes I am at a loss with the KMIXER as some of my PC's seem to upsample and others don't. Guess that has to do with all the drivers and apps I have loaded as each one is different.

 Anyway... I do agree with Elias over the code development of the 1020. I have been coding on this for the last 9 months and it's a frustrating enviroment at best. But the good thing with the 1020 is the larger buffers it has over the PCM27xx parts and it's abilitiy to in software change 100%. I have been able to reduce jitter out of the 1020 just by changing some of the variables and some of the enumeration tables.

 I think more evaluation needs to be addressed in Windows as to what works, what doesn't and what works best.

 Thanks
 Gordon


----------



## EliasGwinn

Quote:


  Originally Posted by *FRANKe* /img/forum/go_quote.gif 
_This is contrary to what I've read elsewhere. First of all, I've checked my Windows system sounds (C:\Windows\Media in XP) and they are mostly 16bit 22kHz and some are mono. And according to a snippet from a blog by Larry Osterman (an employee of MS working in the Windows Media and Devices Group, working on the core windows audio engine for Longhorn), all sounds will down-sample to the lowest common denominator (in Windows XP).
http://blogs.msdn.com/larryosterman/...20/471872.aspx


 Elias, can you explain the discrepancy? Are your comments based on Vista only?

 -FRANKe_

 


 Frank,

 I will premise by saying that, although we have tested these scenarios using several different methods, our findings are based on testing, not on documentation. We spent significant amounts of time searching for official documents that could tell us exactly how kmixer, usbaudio.sys, etc, work. We read as much as we could find, but very little "official Microsoft" documentation was available. So, the the only way for us to find out for sure was to test, test, test. 

 We would never claim to be error-free engineers 
	

	
	
		
		

		
			





 , however, the tests we used were somewhat conclusive. We are always open to suggestions and discussion as to flaws in our testing. We want to know the truth as much as you all do!!

 With that all said, Frank, I have listed our testing methods several times in this thread, but I'll give you a brief re-cap. One test we used to determine SRC (sample-rate conversion) is simply by playing two audio files simultaneously and monitoring the output sample rate. By playing, for example, an audio track of silence at 44/16 with WinAmp, and 1k sine test tone at 48/16 with WMP, the output is 48/16 and the 1k is still 'pure' (FFT shows no sidebands and noise floor around -130 dBFS). Conversely, if we play silence at 48/16 with WinAmp, and 1k sine test tone at 44/16 with WMP, the output is still 48/16, but the 1k has severe distortion due to SRC. This is only an example, as these results were repeatable with various other combinations of media players, sample rates, etc.

 This was based on XP, not Vista. Our copy of Vista (beta 2) seemed to SRC all audio, regardless of what streams were or were not present. The SR in the 'sound properties' control panel seems to dictate this.

 I will read the link you posted. I'm very interested in this, as you all are. We have found what seems like valid test results, and we feel very confident with them...But I will never turn down more sources of information.


----------



## EliasGwinn

Quote:


  Originally Posted by *thomaspf* /img/forum/go_quote.gif 
_I have also spend a long time doing testing. Something does not add up, so we just need to get to the bottom of it but there is nothing to get emotional about. 

 Maybe I am wrong or the by pure accident the pseudo random test stream happens to get passed through unharmed. I am not doubting that the DAC1 is a great product and I always like computer focused audio devices or I would not be here. We will just have to find out.


 Cheers

 Thomas_

 

Thomas,

 I agree that claims made by audio companies about their products should be questioned, and I wish it happened more often!! That is why we include all test results in the manual for the DAC1 and DAC1 USB. Every manufacturer should be able to stand behind their design claims. 

 I appreciate your enthusiasm for truth in technology, and I assure you that we share that desire.


----------



## audioengr

Quote:


  Originally Posted by *EliasGwinn* /img/forum/go_quote.gif 
_Frank,

 I will premise by saying that, although we have tested these scenarios using several different methods, our findings are based on testing, not on documentation. We spent significant amounts of time searching for official documents that could tell us exactly how kmixer, usbaudio.sys, etc, work. We read as much as we could find, but very little "official Microsoft" documentation was available. So, the the only way for us to find out for sure was to test, test, test. 

 We would never claim to be error-free engineers 
	

	
	
		
		

		
		
	


	




 , however, the tests we used were somewhat conclusive. We are always open to suggestions and discussion as to flaws in our testing. We want to know the truth as much as you all do!!

 With that all said, Frank, I have listed our testing methods several times in this thread, but I'll give you a brief re-cap. One test we used to determine SRC (sample-rate conversion) is simply by playing two audio files simultaneously and monitoring the output sample rate. By playing, for example, an audio track of silence at 44/16 with WinAmp, and 1k sine test tone at 48/16 with WMP, the output is 48/16 and the 1k is still 'pure' (FFT shows no sidebands and noise floor around -130 dBFS). Conversely, if we play silence at 48/16 with WinAmp, and 1k sine test tone at 44/16 with WMP, the output is still 48/16, but the 1k has severe distortion due to SRC. This is only an example, as these results were repeatable with various other combinations of media players, sample rates, etc.

 This was based on XP, not Vista. Our copy of Vista (beta 2) seemed to SRC all audio, regardless of what streams were or were not present. The SR in the 'sound properties' control panel seems to dictate this.

 I will read the link you posted. I'm very interested in this, as you all are. We have found what seems like valid test results, and we feel very confident with them...But I will never turn down more sources of information._

 

Eliis - Based on the comments from the other contributors here and on other forums, there may be a dependency on the motherboard or the particular OS image on the computer. This begs the question of:

 Did you verify these findings on more than one PC or laptop, both AMD and Intel CPU's?

 Steve N.


----------



## EliasGwinn

Quote:


  Originally Posted by *audioengr* /img/forum/go_quote.gif 
_Eliis - Based on the comments from the other contributors here and on other forums, there may be a dependency on the motherboard or the particular OS image on the computer. This begs the question of:

 Did you verify these findings on more than one PC or laptop, both AMD and Intel CPU's?

 Steve N._

 

These findings were verified on both laptop and desktops. However, both were running XP SP2. Also, both were Intel's.

 Does anyone know any reason these results may differ based on processor?

 Thanks,


----------



## EliasGwinn

Quote:


  Originally Posted by *FRANKe* /img/forum/go_quote.gif 
_This is contrary to what I've read elsewhere. First of all, I've checked my Windows system sounds (C:\Windows\Media in XP) and they are mostly 16bit 22kHz and some are mono. And according to a snippet from a blog by Larry Osterman (an employee of MS working in the Windows Media and Devices Group, working on the core windows audio engine for Longhorn), all sounds will down-sample to the lowest common denominator (in Windows XP).
http://blogs.msdn.com/larryosterman/...20/471872.aspx


 Elias, can you explain the discrepancy? Are your comments based on Vista only?

 -FRANKe_

 

According to Larry's blog, it will only downsample if Windows sound is played simultaneously as the MP3. If a 44.1 audio file is being streamed, then a Windows sound is streamed, I am very certain that the Windows sound will be sample-rate converted.


----------



## Wavelength

Quote:


  Originally Posted by *EliasGwinn* /img/forum/go_quote.gif 
_These findings were verified on both laptop and desktops. However, both were running XP SP2. Also, both were Intel's.

 Does anyone know any reason these results may differ based on processor?

 Thanks,_

 

Elias,

 Well I will throw this out in regards to your question.

 I have an Apple MacBook with BootCamp and XP PRO SP2. I did some tests using my USB DAC (1020B) with Dr. Jordan Audio Suite. Pretty nice FFT analyzer and something I used before I got the dScope.

 I then got an update from Apple stating there was a new Bootcamp which came with upgraded drivers for XP. I loaded the results and could not believe how much better the same tests had.

 Some of my friends at Apple said the new Bootcamp included better streaming routines for audio.

 This is why I think there are too many variables here to just state to a defacto standard that this or that is true.

 Thanks
 Gordon


----------



## EliasGwinn

Quote:


  Originally Posted by *Wavelength* /img/forum/go_quote.gif 
_
 This is why I think there are too many variables here to just state to a defacto standard that this or that is true.

 Thanks
 Gordon_

 

Gordon,

 I couldn't say it better myself. A systems result is only as strong as its weakest link. This is true with any multi-part system, and especially true with computer systems. 

 However, one can say what one's component is capable of. That is, we can say that the DAC1 USB, when used with a typical Intel/XP/2000/OSX system, is capable of bit-transparent playback. 

 We cannot, however, say that it will ALWAYS be bit-transparent no matter what other hardware/software/settings implemented upstream from it. Many third-party plug-ins you can download for your media player will put your audio through the washing machine and dryer before it gets to your audio device.


----------



## FRANKe

Hi Elias - let me just say thank you for your diligence in responding to the myriad of questions and concerns here.

 And now, if you will, indulge me in another set of questions. You have been talking a lot about the goal of achieving "bit-transparent". And here is a little quote from your testing methods: 

  Quote:


  Originally Posted by *EliasGwinn* /img/forum/go_quote.gif 
_The testing consisted of the 'psuedo-random' bit-test that was mentioned in the press release. This is, quiet simply, testing "what-bits-go-in-and-what-bits-come-out". This is a standard test developed by Audio Precision, the leading audio electronics testing equipment manufacturer. When the Audio Precision (AP) sends a digital audio signal into a device, it checks to see if the exact same bits come out. So, for example, if the AP sends in 101100111000, a 'bit-transparent' data path will output the exact same bits: 101100111000. This was our testing proceedure. _

 

So my question to you then, is it possible to pass this test (with the same bits coming out as went in), but have varying degrees of jitter? In other words, I'm not an (audio) engineer so I'm just trying to make sense of this as an audio enthusiast. Does this AP test take jitter into account? And can two output methods (one through kmixer and one through ASIO) both be "bit-transparent", but yet have different amounts of jitter (and hence sound different)? And if so, are you saying that the amount of jitter between these two methods doesn't matter because your DAC is immune?

 Thanks.
 -FRANKe


----------



## Wavelength

Quote:


  Originally Posted by *FRANKe* /img/forum/go_quote.gif 
_So my question to you then, is it possible to pass this test (with the same bits coming out as went in), but have varying degrees of jitter? In other words, I'm not an (audio) engineer so I'm just trying to make sense of this as an audio enthusiast. Does this AP test take jitter into account? And can two output methods (one through kmixer and one through ASIO) both be "bit-transparent", but yet have different amounts of jitter (and hence sound different)? And if so, are you saying that the amount of jitter between these two methods doesn't matter because your DAC is immune?_

 

Elias, I think I can answer this.

 Franke,

 Fist off with USB you have no jitter on the interface itself because of the lack of clock. The only jitter you have in the system based on jitter is what we call intrinsic jitter. This is jitter that is a result of the recieving chips clocking maybe slow response on the pc and lack of acceptable buffer on the receiving side or whatever.

 Therefore the AP test is as I described a MP3 and WAV file that can be played on any available software.

 With most of these tests sets they do allow for the injection of jitter in the SPDIF realm and the rejection of said jitter can be tested in the analog or digital inputs of the test set.

 Here in USB land jitter can be tested in the analog domain using what is called sidebands. These are usually shown as spikes in the output surrounding a test tone, like the JTEST (1/4 Fs at -6dB and 1 bit usually around 229Hz). Though the JTEST is not really relevant for USB only SPDIF. It is better to test with common frequency and look for side bands in the FFT plot.

 The Miller Audio Suite is the most popular for this sort of test as it is very colorfull and used by Stereophile for testing dac's and jitter. Follow this link to see the Apple Airport Express on the Stereophile website:

http://www.stereophile.com/accessory...ple/index.html

 Presently Ap tests iPod devices and any Windows based system using the files so does Miller. The Prism dScope is the only test set that can stream directly to a device.

 But be forwarned on jitter testing and the numbers. The miller device with the 16 bit AD board Stereophile uses is only good to 130ps (I find this a little hard to believe for 16 bit board), the dScope at 24/192 is 100ps and the Ap standard ATS2 I think is 200ps and with the preformance option 150ps. I don't really think any of these test units can really test below 200ps. You really need a wavecrest and probe around internally to determine real jitter. One of these will set you back about the cost of BMW 7 series.

 Thanks
 Gordon


----------



## EliasGwinn

Quote:


  Originally Posted by *FRANKe* /img/forum/go_quote.gif 
_So my question to you then, is it possible to pass this test (with the same bits coming out as went in), but have varying degrees of jitter? In other words, I'm not an (audio) engineer so I'm just trying to make sense of this as an audio enthusiast. Does this AP test take jitter into account? And can two output methods (one through kmixer and one through ASIO) both be "bit-transparent", but yet have different amounts of jitter (and hence sound different)? And if so, are you saying that the amount of jitter between these two methods doesn't matter because your DAC is immune?

 Thanks.
 -FRANKe_

 

Bit-transparency is independent of jitter. The AP can measure jitter, but the bit-test is not dependent of jitter. Comparing ASIO vs. kmixer with regards to jitter is not a valid comparison. Comparing PCI sound cards vs. USB would be more appropriate. Even then, there are different methods of operation which would need to be included in the comparison.

 The Benchmark DAC1 and DAC1 USB are, in fact, immune to jitter. The DAC1 will perform identically regardless of how much jitter is present. I have posted an explanation of this earlier in the thread (maybe around page 3 or so...). You can also read about it on our website. Go to: 

http://www.benchmarkmedia.com/dac1/

 Then half-way down the overview section, there is a paragraph titled "Jitter-Immune UltraLock™".

 Thanks for the questions...
 Elias


----------



## EliasGwinn

USB interfaces will, in fact, have significant amounts of jitter if they are run in synchronous mode. Synchronous operation means that the host will dictate the sample-rate clock. 

 A lot of USB audio device are designed to run in asynchronous mode to eliminate the jitter, but the tradeoff is that, when in asynchronous mode, kmixer may have to sample-rate convert to conform to the clock of the device. This sample-rate conversion (as you all know) is extremely detrimental to the quality of the audio.

 The Benchmark DAC1 USB runs in synchronous mode. The reason for this is that it lets the host (kmixer) operate at the original sample-rate of the audio being played at all times. If the kmixer is not forced to do any sample-rate conversion, it can maintain bit-transparent operation. The tradeoff, of course, is significant amounts of jitter arriving at the DAC1. This is not a problem for the DAC1 however, because Benchmark's UltraLock clocking system makes it immune to jitter (as I have explained in detail several times before in this thread).

 Thanks for the questions...
 Elias


----------



## EliasGwinn

The last post actually brings a very important point to mind...something that should have been said much earlier in this thread.
*
 The behavior of Kmixer, etc is dependent on the device it is streaming to!! *

 In other words, kmixer may act differently for one USB audio device then it will act for another USB audio device. This may be the reason for a lot of misunderstanding about kmixer.

 Most importantly, it should be said that all statements made by Benchmark (specifically me) concerning kmixer being bit-transparent is assuming the audio device being used will _let_ kmixer operate bit-transparently.


----------



## FRANKe

Thank you Gordon and thank you Elias for the wealth of information.

 I think I understand everything you guys said. And forgive me if I drag this out a little longer. But Elias, you are bursting a lot of people's bubbles defending kmixer the way you are, and so I'm just trying to wrap my head around something.

 OK, the bottom line for me is that output from kmixer and ASIO "sound" different. I mean, heck, even different foobar ASIO plugins sound different. Elias, you are implying that kmixer is the one that is bit-transparent and, as you mused, some ASIO solutions are run through the washer and dryer. I thought ASIO "sounded" better and certain foobar ASIO plugins "sounded" even better because the audio stream and/or clock was more accurate and hence lower jitter. I mean even different computer configurations will sound different. Of course, this is all based on my personal comparisons, using a couple of different computers and laptops and with both a M-Audio USB Audiophile as well as a Wavelength Brick and Cosecant.

 So if the differences in sound are not jitter related, are you implying that with each configuration change, the bits are changing? And/or are you implying that the Benchmark USB DAC1 is immune to different computer "tweaks"?

 Thanks.
 -FRANKe

 Edit: sorry for the redundancy (I was typing the post at the same time as Elias' previous)


----------



## EliasGwinn

Franke, 

 In addition to my last few posts, I'll try to clarify a bit...

 First of all, I want to address my stance on ASIO. I do not have enough experience with ASIO as a whole (API) to condemn or support the API. We have done some testing with devices which use ASIO, but I am not an expert on all the different facets of ASIO. (the washer and dryer comment was not directed at ASIO).

 I believe that ASIO is probably a very good API, _if used right_. It has some limitations with regards to configuration, however, that make it less then appealing for an audio playback device. For example, only one audio app can access an ASIO driver at a time, unless the kmixer is placed in front of it. Also, my experiences with ASIO devices was that it did not follow the sample-rate dynamically. In other words, instead of simply streaming at the sample-rate of the audio app (synchronously), the sample-rate had to be specifically set within the ASIO driver (asynchronously). 

 With all that said, I will address your experiences with different audio configurations with your computer. The configurations within the computer will react dynamically to the audio device it is streaming to. In other words, it can not be universally stated that kmixer is better then ASIO, or vice versa. It depends on what device it is talking to.

 A solid example of this: If device XX had the exact same USB interface as the DAC1 USB, asynchronous mode might sound significantly better then synchronous because, even though kmixer will convert sample-rates, it won't pass high jitter levels to the D-to-A. However, with the DAC1 USB, synchronous will sound much better because the jitter levels don't affect the D-to-A within the DAC1, and the kmixer won't convert sample-rates.

 Different devices will react differently to different modes of operation within the computer. 

 Therefore, I cannot say that ASIO devices are inferior to native devices, or vice versa. All I can say with certainty is, the DAC1 USB will have identical D-to-A performance playing 96/24 audio through kmixer -> USB interface, as it will playing 96/24 audio straight from the digital signal generator of the AP -> SPDIF input of the DAC1. This is much I can confirm.


----------



## jpelg

Thanks for all the info, Elias. 

 I feel as if I've learned a lot from reading this thread, and do not have a headache as a result. Well done!


----------



## EliasGwinn

Thanks, jpelg!! That is what I strive for. You know, its not easy for us engineers to talk right. In fact, sometimes when people don't understand me, I instantly think its because they require different power supply voltages then what I'm supplying. Or we need a codec, one or the other.


----------



## Wavelength

Quote:


  Originally Posted by *EliasGwinn* /img/forum/go_quote.gif 
_USB interfaces will, in fact, have significant amounts of jitter if they are run in synchronous mode. Synchronous operation means that the host will dictate the sample-rate clock. 

 A lot of USB audio device are designed to run in asynchronous mode to eliminate the jitter, but the tradeoff is that, when in asynchronous mode, kmixer may have to sample-rate convert to conform to the clock of the device. This sample-rate conversion (as you all know) is extremely detrimental to the quality of the audio.

 The Benchmark DAC1 USB runs in synchronous mode. The reason for this is that it lets the host (kmixer) operate at the original sample-rate of the audio being played at all times. If the kmixer is not forced to do any sample-rate conversion, it can maintain bit-transparent operation. The tradeoff, of course, is significant amounts of jitter arriving at the DAC1. This is not a problem for the DAC1 however, because Benchmark's UltraLock clocking system makes it immune to jitter (as I have explained in detail several times before in this thread).

 Thanks for the questions...
 Elias_

 


 Elias,

 Actually USB does not have jitter in either mode. Jitter in SPDIF land basically is caused by the removing the clock from the data. In USB there is no clock.

 Though all parts have intrinsic jitter that have clocked audio, USB experiences more of this than say a SPDIF receiver for many reason's but the primary ones are:

 1) Lack of buffering
 2) Lack of audio clock source
 3) modulation from on board sources such as dac's and stuff

 USB gather's it's clock from the timing of packets of data. The longer the buffer the more the in coming clock is averaged out and therefore the locked in the audio clock becomes. Parts like the PCM270x series have very little buffers. The TUSB3200 and TAS1020 have configurable buffers and work much better. You can use USBVIEW.exe on the PC to look at the enumeration tables to see how large the incoming buffers are.

 USB runs at 12MHZ, wereas most audio does not run any were close to this frequency. Therefore the audio clock is being derived from a source that is not the best reference for audio. This is true in ISOSYNC mode only, in ASYNC mode we do have to give the clock to the controller and it paces the computer to that clock.

 Some of the combo chips like the PCM270x series have onboard class D headphone amplifiers and stuff that really stink up their preformance. Also their SpAct controller goes bizurt every 80-90 seconds.

 Anyway... Elias with XP ASYNC mode is not supported. You would have to have drivers for that.

 But in the TAS1020 (and TUSB3200) you can tune the intrinsic jitter very low. First the part is capable of Input and Output. If you are just using it as a dac then you can consume the resources for the ADC and make longer buffers. You can also tune the ACG (Adaptive Clock Generator) over the longer buffers and get the jitter down really low.

 To answer Franke question I think I would have to setup some tests. One that would be interesting would be to set the USB controller to ISO mode 16 bit and 44.1K only. Then send the test tone down via the KMIXER from a standard red book track. Then ASIO and KERNEL streaming.

 So far in the past 4 years of designing USB DACs this is what I can tell you. Not all ASIO's work the same way and the results vary from PC to DAC. Kernel streaming can result in some systems locking up. I have seen the same setup on one machine via the KMIXER go to 48K and another with the same track stay at 44.1.

 In all the MAC is more consistent than the PC in it's output.

 Well going to shut down and play some guitar.

 Later,
 Gordon


----------



## thomaspf

Very accurate with one minor variation. XP does actually support async audio in USBaudio.sys, however, the Vista implementation is more faithful to the standard.

 Cheers

 Thomas


----------



## Wavelength

Quote:


  Originally Posted by *thomaspf* /img/forum/go_quote.gif 
_Very accurate with one minor variation. XP does actually support async audio in USBaudio.sys, however, the Vista implementation is more faithful to the standard.

 Cheers

 Thomas_

 

Thomas,

 Sorry if this is the case but on the two PC's I use for testing they immediately reboot when I plug in the USB DAC with it set to ASYNC in the Enumeration. I recompile with ISO setting and it works fine.

 I talked to the company who supports the TI USB Audio solutions and they say the same thing. They say it requires a driver before it will work that way.

 If there is a way, please tell me who too talk to.

 Thanks
 Gordon


----------



## thomaspf

I have tried this with the Audigy 2NX. The Audigy works in async mode and if you do noy install their custom driver the system driver works just fine in stereo mode. The Audigy however does use a more general USB controller. There is also an older Emagic EMI 2|6 that works just fine in async mode.

 As I mentioned above the Vista implementation is more faithful to the standard.

 I would just report this as a bug and see whether it makes the bar. This is the first time I hear about an instant reboot when try to negotiate async. I am glad to hear you are working on this.

 Cheers

 Thomas


----------



## lowmagnet

My DAC1 arrived, and it's working great. Thanks Benchmark! 
	

	
	
		
		

		
		
	


	




 Added: After 3-4 hours listening to this, it's really just perfect. I'm not used to spending this much on sound hardware, but it's obviously money well-spent. It also matches my G5 and monitor in finish


----------



## EliasGwinn

That's great!!

 One thing to be carefull of... With Mac OSX, there is a utility called "Audio MIDI Setup". It can be found in Applications/Utilities/AudioMIDI Setup. Be sure to set the sample rate in this utility to match the sample-rate of the audio you are listening to. Otherwise, it will convert the sample-rate, causing significant distortion.

 Also, we recommend using the volume control on the DAC1, and keeping the volume in the software set to 100%. Software volume controls are rarely designed properly (iTunes is horrible), and will cause significant distortion.

 Enjoy your DAC1 USB!!
 -Elias


----------



## lowmagnet

Elias,

 Have you tested out on Leopard yet to see if the volume control errors are corrected? I do miss sound check a lot since I have some punk/SKA in my collection and it's almost universally LOUD and compressed.


----------



## EliasGwinn

I have not tested Leopard yet. 

 I know what you mean about "Sound Check"...it would be a great feature if it was designed well.


----------



## clarkc

Thankyou Elias (and others) for the really interesting input in this thread. 

 Elias, you mentioned that you will post a guide to PC audio sometime soon. For newbies like me to this stuff, this sounds very useful. When can we expect this?

 Thanks, Clark


----------



## gracky

Elias, with Macos X / Audio midi setup, output bits can be altered as well as sample rates. Won't it affect the quality of the digital out if the source file is in ordinary 16 bits?


----------



## lowmagnet

Quote:


  Originally Posted by *gracky* /img/forum/go_quote.gif 
_Elias, with Macos X / Audio midi setup, output bits can be altered as well as sample rates. Won't it affect the quality of the digital out if the source file is in ordinary 16 bits?_

 

The DAC1 is locked at [2ch-24bit] on my system, with no ability to alter it. And bit depth doesn't really affect sound quality like sampling rate does. Listening to 24-bit audio in 16-bit would be bad, but sample rate changes are worse.


----------



## EliasGwinn

Quote:


  Originally Posted by *gracky* /img/forum/go_quote.gif 
_Elias, with Macos X / Audio midi setup, output bits can be altered as well as sample rates. Won't it affect the quality of the digital out if the source file is in ordinary 16 bits?_

 

Great questions. It is ALWAYS recommended to use 24-bit settings when possible, EVEN WHEN THE AUDIO IS 16-BITS!!

 The reason is: when you operate at 24-bits, you don't actually add or modify any of the digital data, you simply give the data a "bigger path" to travel through. The reason this is important when playing 16-bit audio is because certain software operations can cause a 16-bit word to turn into a 24-bit word. 

 One easy example is applying volume control. If you change the volume to 90% for example, the software will multiply all the digital audio data by 0.9 (or divide by 10/9). The result is rarely an exactly 16-bit word, just like dividing regular numbers often results in non-integer numbers. It usually has a remainder...which will require more bits (digits, or decimal places in regular numbers). If the data path is limited to 16-bit, the remainders will be truncated. That's analogous to dividing 10 by 3, and having a result of 3. With extra digits available, a more accurate result of 3.333... can be achieved. 

 This will be further explained in an upcoming e-update that we will be sending out shortly. If you would like to receive this e-update, and future ones as well, you can sign up on our website:

http://www.benchmarkmedia.com/accoun...ribe_conf.html

 Thanks,
 Elias

 PS: The exception to this: If your audio device is a 16-bit device AND if your player offers dithering to 16-bits (such as Foobar) AND nothing else downstream will result in 24-bit words (Volume controls, DSP and Plug-ins, etc), then it may be better to use the dither-to-16-bit setting. But, this is still a bit risky...


----------



## EliasGwinn

Quote:


  Originally Posted by *lowmagnet* /img/forum/go_quote.gif 
_The DAC1 is locked at [2ch-24bit] on my system, with no ability to alter it. And bit depth doesn't really affect sound quality like sampling rate does. Listening to 24-bit audio in 16-bit would be bad, but sample rate changes are worse._

 

This is true... Although, truncating 24-bits to 16-bit is very bad, so that's subjective.


----------



## EliasGwinn

Quote:


  Originally Posted by *clarkc* /img/forum/go_quote.gif 
_Thankyou Elias (and others) for the really interesting input in this thread. 

 Elias, you mentioned that you will post a guide to PC audio sometime soon. For newbies like me to this stuff, this sounds very useful. When can we expect this?

 Thanks, Clark_

 

Hopefully we will have this available in the next week or so. We will be sending an e-update email announcing it. If you would like to receive this and other e-updates, you can sign up at:

http://www.benchmarkmedia.com/accoun...ribe_conf.html

 Thanks,
 Elias


----------



## lowmagnet

Quote:


  Originally Posted by *EliasGwinn* /img/forum/go_quote.gif 
_This is true... Although, truncating 24-bits to 16-bit is very bad, so that's subjective._

 

My mistake, I wrote the above incorrectly. Truncating 24 to 16 _is_ very bad. Running a higher bit-rate than your source data on Mac OS X is also bad. but not as bad as truncating data. Sorry.


----------



## Lord Chaos

I received my DAC1 USB yesterday. In a word: excellent.

 After I got the DAC1 for use with my Squeezebox I wanted better sound from the Powerbook because it's the computer I'm using most of the time and its headphone output is noisy. I started looking for USB devices... and that's about when the Email arrived from Benchmark. I ordered soon thereafter, and took home serial number 29.

 Pipe organ music is one of my standard tests because I've heard a lot of real pipe organs so I know what they should sound like. I've never found a satisfactory system... until now. The Powerbook/DAC1 USB/Shure E500 combination has none of the subtle raspiness that has afflicted organ music in other systems I've used. It is simply a delight to listen to.

 Thank you, Elais, for your comments here. They help me understand how this device works.


----------



## SamNOISE

EliasGwinn

 Wow the dude's been one patient cat in answering all of these audiogeek questions! I suggest you post an address that we can send a bottle of fine Vino & our best chocolate to!

 I'll attempt to diverge slightly from the topic that is being hammered to death:

 What is your take on using the BenchMark DAC1 as a simple preamplifier / DAC? In my case, I have been musing over a combo-DAC / Preamp such as Benchmark's product between my CD player or laptop and the power amplifier (PSAudio HCA-2).

 Many have suggested that this is not an idea situation as they feel that the preamp stage in products such as the DAC1 would be inferior to a 'classic', stand-alone preamplifier.

 Now, I'm not one to use bass / treble / balance etc controls - nor do I care about input switching etc. That said, do you feel that the built-in preamplifier in combo products such as the DAC1 are every bit as 'transparent' as any high-end preamplifier?

 Tnx. for your time man.

 Andrew D.
www.cdnav.com


----------



## Lord Chaos

The DAC1 manual specifically addresses this, and the USB version has internal jumpers for optimizing the output level. They highly recommend using the DAC1 in place of a pre-amp, which is what I'm going to do as soon as I get things arranged.

 I have the Powerbook connected to the DAC1 through USB. I'll use the S/PDIF from the PC, and then connect the DAC to my power amp. When I get the Mac Mini music server I'll use one of the other inputs for that. Then I'll get rid of the CD player, pre-amp, etc.


----------



## Scrith

Quick question about the new DAC1 USB:

 Other than the USB support, is there anything new (or improved) about the DAC section of the DAC1?

 I am a previous DAC1 owner (both a 2004 and an early 2006 model, which sounded different, by the way) that switched to another DAC that, in my opinion, sounded slightly better than the DAC1 (which I attributed to the resampling that was happening in the DAC1 in order to deal with jitter). I am using an external box to convert USB to S/PDIF (and wouldn't mind eliminating this USB box from the chain if I could replace my box+DAC with a new DAC1 USB and get sound that is at least as good).

 By the way, I always found that resampling up to 96K using a high-quality resampler (such as the Secret Rabbit resampler in Foobar2000) improved the sound with the DAC1, which I also attributed to the (relatively) poor job the DAC1 was doing in resampling music from 44K all the way to 117K (a guess...I don't remember the exact value of the data rate the DAC1 uses internally). Has the hardware that is being used to resample the data to the internal rate changed at all?


----------



## Audio_newb

Two questions, one for lowmagnet and one for Elias. Firstly, low, I was wondering how your new Benchmark setup compares to your Spitfire rig. Any thoughts would be great as I'm trying to justify the cost of the new Benchmark to myself.

 Elias, thanks for answering all our questions so far both here and over at the stereophile forums. I was wondering whether, now that there is a usb connection we might see future firmware updates and whether you guys were still working on anything like that. Thanks.


----------



## Shout It Out Lou

Hey Elias, to get terribly unsimple for one second and ask a direct question - are there any plans to ever release a remote controllable Dac1? This is the only thing stopping me from buying one now (used to own one but missed being lazy!).


----------



## lowmagnet

Quote:


  Originally Posted by *Audio_newb* /img/forum/go_quote.gif 
_Two questions, one for lowmagnet and one for Elias. Firstly, low, I was wondering how your new Benchmark setup compares to your Spitfire rig. Any thoughts would be great as I'm trying to justify the cost of the new Benchmark to myself._

 

It compares favorably. The little Spitfire and Cute Beyond (Class A) are nice, but they have their faults. 

 The Spitfire takes in coax or optical, and my upstairs system has a dead optical port on its main-board, so that was right out for when I'm at that desk. Also, it has signal locking problems on the downstairs system. It would play noise with no signal on.

 The Cute Beyond had serious problems with its volume control that couldn't be corrected by the ganging switch inside. Left channel was way off until about 9 O'Clock, which is painfully loud even in low-output mode.

 That said, on to the benchmark. It has a few features that are a big improvement over the Firestone devices:

It takes power from a standard 3-prong power connector to well-designed power supply instead of a wall-wart.
Both DAC and amp are in the same box, instead of stringing two boxes together.
Multi-source selection is possible here, but I haven't used it yet.
I can run my desktop speakers from the benchmark. When I pull my headphones, it plays over the desk speakers.
The volume control is detented and balanced between channels well.
The headphone amp gives me bass on Grado SR225 (which aren't well known for bass)

 I hope that covers it. I'm not going to say anything about soundstage or anything, because I'm not a fan of fuzzy terms to describe equipment. It sounds really great, and doesn't miss a thing. I can hear more background chatter in "Light Up" by Styx, which is one of my tests for clarity of sound.

 Plus, the thing tests well on the bench. I know you're going to read all of the above, and say "CURSE YOU HEAD-FI!!!", but you did ask for my honest opinion 
	

	
	
		
		

		
			





 (Unless you like tubes, warm sound and all of the other methods of listening to equipment instead of music. Some people roll that way.)


----------



## lowmagnet

Quote:


  Originally Posted by *Shout It Out Lou* /img/forum/go_quote.gif 
_Hey Elias, to get terribly unsimple for one second and ask a direct question - are there any plans to ever release a remote controllable Dac1? This is the only thing stopping me from buying one now (used to own one but missed being lazy!)._

 

What would you remote control? Volume, source and mute? 
	

	
	
		
		

		
		
	


	




 Hm, that could be useful 
	

	
	
		
		

		
		
	


	




 Then again, mine sits right to my left when I'm using it.


----------



## Audio_newb

Thanks, low. No curses from me. I knew what I was getting into when I went a lookin and I always have my credit card out of reach when I'm trawling the boards. Now I just need to wait for a birthday.


----------



## felicon

For those who are interested, the deeper technical discussion on USB with 24/96 versus Windows (XP, Vista) and MAC OS X, takes place here:

http://www.diyhifi.org/forums/viewto...?p=25218#25218


----------



## lowmagnet

This occurred on my Dell system at work with a different device, but it's important if you are hearing clicks and pops with any USB device. You won't hear the pops and clicks on the DAC1 since it should catch them, but it's good to check this out anyway, as it affects the performance of the USB port you're plugged into:

  Quote:


  Originally Posted by *Lowmagnet* 
_I was having clicks and pops so I brought up the hardware manager, looked through the tree to find my iMic (on my work machine) and wrote down the USB bus number (27CB). 

 I then switched to IRQ mode and found a USB bus without any shared IRQ (27CB was on the same IRQ as the onboard sound) and wrote the bus number down (27C9). 

 Then I went back into the connection view, and started plugging my iMic in to various ports until it came up on 27C9, and isolation was achieved._


----------



## lowmagnet

Quote:


  Originally Posted by *felicon* /img/forum/go_quote.gif 
_For those who are interested, the deeper technical discussion on USB with 24/96 versus Windows (XP, Vista) and MAC OS X, takes place here:

http://www.diyhifi.org/forums/viewto...?p=25218#25218




_

 

Enumeration is interesting stuff, and yep the chip does 24/96. Thanks for the info!


----------



## boggle

I have a question for Mr Gwinn. What do you (or others) think of the price doubling Empirical Audio Benchmark mod? I have read that the dac1 has 'empirically' (no pun intended) perfect specs - so where is the improvement? The psychology here is as interesting as anything, I would love to see some serious (decent sized n) double blind trials go on, perhaps at a meet (then posted as a separate thread). 

 Also (for Mr Gwinn), have you read the comments about the design of the dac1 on the lessloss website? It certainly sounds like they know what they are on about - I wonder if anyone could hear the difference there also. I think there is a very real phenomena related to money spent, research done/time spent, cool looks feeding into how we perceive the sound coming out of some of these boxes. 

 This has been a very enjoyable thread by the way. I am currently saving up for a dac1 and am sure i will love the sound, not just because of the time I have spent researching, thinking, talking about it


----------



## audioengr

Quote:


  Originally Posted by *boggle* /img/forum/go_quote.gif 
_I have a question for Mr Gwinn. What do you (or others) think of the price doubling Empirical Audio Benchmark mod? I have read that the dac1 has 'empirically' (no pun intended) perfect specs - so where is the improvement? The psychology here is as interesting as anything, I would love to see some serious (decent sized n) double blind trials go on, perhaps at a meet (then posted as a separate thread). 

 Also (for Mr Gwinn), have you read the comments about the design of the dac1 on the lessloss website? It certainly sounds like they know what they are on about - I wonder if anyone could hear the difference there also. I think there is a very real phenomena related to money spent, research done/time spent, cool looks feeding into how we perceive the sound coming out of some of these boxes. 

 This has been a very enjoyable thread by the way. I am currently saving up for a dac1 and am sure i will love the sound, not just because of the time I have spent researching, thinking, talking about it _

 


 If you come to the Head-Fi meet in San Jose you can hear for yourself. I can give you technical explanations about the improvements in person and you can listen to the modded DAC-1. You can hear my new headphone amp mod for it.

 Steve N.
 Empirical Audio


----------



## EliasGwinn

First of all, sorry for my absence on the thread lately...things have been pretty busy around here with our new products getting up on their own feet. So far, so good!

 Lord Chaos, Low Magnet (and anyone/everyone else), I'm glad to see your enjoying your new DAC1 USB's! 

 Now, to answer some questions:

  Quote:


  Originally Posted by *SamNOISE* /img/forum/go_quote.gif 
_EliasGwinn

 Wow the dude's been one patient cat in answering all of these audiogeek questions! I suggest you post an address that we can send a bottle of fine Vino & our best chocolate to!

 I'll attempt to diverge slightly from the topic that is being hammered to death:

 What is your take on using the BenchMark DAC1 as a simple preamplifier / DAC? In my case, I have been musing over a combo-DAC / Preamp such as Benchmark's product between my CD player or laptop and the power amplifier (PSAudio HCA-2).

 Many have suggested that this is not an idea situation as they feel that the preamp stage in products such as the DAC1 would be inferior to a 'classic', stand-alone preamplifier.

 Now, I'm not one to use bass / treble / balance etc controls - nor do I care about input switching etc. That said, do you feel that the built-in preamplifier in combo products such as the DAC1 are every bit as 'transparent' as any high-end preamplifier?

 Tnx. for your time man.

 Andrew D.
www.cdnav.com_

 

First of all, I'll premise by saying that not all D-to-A's have pre-amp quality output stages. However, the DAC1 is designed to operate as a pre-amplifier for the purpose of avoiding that extra stage (and extra cables, electronics, etc). We recommend using the DAC1 USB without a pre-amp simply because ANY device in the audio chain, no matter how well designed, will add some noise and distortion. Since a pre-amp is not needed with the DAC1, you will enjoy better performance by leaving it out of the signal path. With that said, there are people who particularly enjoy the sound shaping that a specific pre-amp achieves. For those people, using that pre-amp is a subjective decision. 

 I'd like to also add that the new DAC1 USB has specially designed high-current output drivers that will rival most any stand-alone pre-amp. These output drivers can drive almost any load impedance, capacitance, or inductance imaginable without any loss in THD+N performance.'

 Thanks,
 Elias


----------



## EliasGwinn

Quote:


  Originally Posted by *Scrith* /img/forum/go_quote.gif 
_Quick question about the new DAC1 USB:

 Other than the USB support, is there anything new (or improved) about the DAC section of the DAC1?

 I am a previous DAC1 owner (both a 2004 and an early 2006 model, which sounded different, by the way) that switched to another DAC that, in my opinion, sounded slightly better than the DAC1 (which I attributed to the resampling that was happening in the DAC1 in order to deal with jitter). I am using an external box to convert USB to S/PDIF (and wouldn't mind eliminating this USB box from the chain if I could replace my box+DAC with a new DAC1 USB and get sound that is at least as good).

 By the way, I always found that resampling up to 96K using a high-quality resampler (such as the Secret Rabbit resampler in Foobar2000) improved the sound with the DAC1, which I also attributed to the (relatively) poor job the DAC1 was doing in resampling music from 44K all the way to 117K (a guess...I don't remember the exact value of the data rate the DAC1 uses internally). Has the hardware that is being used to resample the data to the internal rate changed at all?_

 



 The DAC1 USB has several new features as compared to the DAC1. These are:

 * True native 24/96 bit-transparent USB audio interface
 * High-current output drivers (read end of previous post about this)
 * Two gain ranges for headphone amplifier for more/less sensitive headphones
 * Main outputs mute on headphone insertion (this feature is defeatable)

 Thanks,
 Elias


----------



## EliasGwinn

Quote:


  Originally Posted by *Audio_newb* /img/forum/go_quote.gif 
_Elias, thanks for answering all our questions so far both here and over at the stereophile forums. I was wondering whether, now that there is a usb connection we might see future firmware updates and whether you guys were still working on anything like that. Thanks._

 

Firmware updates will be addressed when it is apparent they are necessary. So far, the device seems to work flawlessly. 

 Thanks,
 Elias


----------



## EliasGwinn

Quote:


  Originally Posted by *Shout It Out Lou* /img/forum/go_quote.gif 
_Hey Elias, to get terribly unsimple for one second and ask a direct question - are there any plans to ever release a remote controllable Dac1? This is the only thing stopping me from buying one now (used to own one but missed being lazy!)._

 

We currently have no plans to incorporate a remote control into the DAC1 design, but we appreciate the suggestion and will keep it in consideration.

 Thanks,
 Elias


----------



## EliasGwinn

Quote:


  Originally Posted by *boggle* /img/forum/go_quote.gif 
_I have a question for Mr Gwinn. What do you (or others) think of the price doubling Empirical Audio Benchmark mod? I have read that the dac1 has 'empirically' (no pun intended) perfect specs - so where is the improvement?_

 

If you are considering having your DAC1 modified, please read Benchmark's official statement regarding modifications to Benchmark products (copied directly from Benchmark DAC1 USB manual, page 2 - http://www.benchmarkmedia.com/manual...USB_Manual.pdf ):

 Benchmark’s official statement regarding modifications:

 "CAUTION: DO NOT SUBSTITUTE PARTS OR MAKE ANY MODIFICATIONS WITHOUT THE WRITTEN APPROVAL OF BENCHMARK MEDIA SYSTEMS, INC. MODIFICATION MAY CREATE SAFETY HAZARDS AND VOID THE WARRANTY.

 NOTICE: CHANGES OR MODIFICATIONS NOT EXPRESSLY APPROVED BY BENCHMARK MEDIA SYSTEMS COULD VOID THE USER'S AUTHORITY TO OPERATE THE EQUIPMENT UNDER FCC REGULATIONS."

 John Siau, the director of engineering, and chief designer of the DAC1, has addressed this, so I will copy his response here. 

 John Siau: "I have not seen, measured, or listened to a modification that I would recommend. Modified units usually perform more poorly. We have measured modified units and have found the following problems:

 a) Distortion due to opamp substitutions
 b) Phase errors between channels due to capacitor changes
 c) UltraLock(tm) rendered non-functional due to IC change
 d) Frequency response problems due to substitution of incorrect capacitor values.

 We have fixed several modified units at customer expense after they failed. This has given us the opportunity to measure the performance and inspect the workmanship. In all cases the performance was degraded. In all of these RMA cases, the modifications caused failure of the product. I do not recommend the services of modifiers. We had good reasons for using the parts that we used. Our reasons had everything to do with performance and nothing to do with cost.

 OPA627 vs. NE5532:

 The OPA627 will offer no advantage over the NE5532 in this application. High-frequency THD+N may actually be slightly higher with the OPA627. We are using the new LM4562A in the DAC1 USB. The LM4562A is pin-compatible with the NE5532, but you can't just drop it in and get all of the advantages of the DAC1 USB. We changed all of the resistors in the XLR output pads to take advantage of the high drive capability of the LM4562A."

 Pay particular attention to items B and D in the list above. The modified units we have seen have incorrect capacitor values substituted for the original parts. Substituting incorrect capacitor values will result in serious deficiencies in sound field accuracy and also freq response accuracy.

 Thanks,
 Elias


----------



## kool bubba ice

Quote:


  Originally Posted by *EliasGwinn* /img/forum/go_quote.gif 
_If you are considering having your DAC1 modified, please read Benchmark's official statement regarding modifications to Benchmark products (copied directly from Benchmark DAC1 USB manual, page 2 - http://www.benchmarkmedia.com/manual...USB_Manual.pdf ):

 Benchmark’s official statement regarding modifications:

 "CAUTION: DO NOT SUBSTITUTE PARTS OR MAKE ANY MODIFICATIONS WITHOUT THE WRITTEN APPROVAL OF BENCHMARK MEDIA SYSTEMS, INC. MODIFICATION MAY CREATE SAFETY HAZARDS AND VOID THE WARRANTY.

 NOTICE: CHANGES OR MODIFICATIONS NOT EXPRESSLY APPROVED BY BENCHMARK MEDIA SYSTEMS COULD VOID THE USER'S AUTHORITY TO OPERATE THE EQUIPMENT UNDER FCC REGULATIONS."

 John Siau, the director of engineering, and chief designer of the DAC1, has addressed this, so I will copy his response here. 

 John Siau: "I have not seen, measured, or listened to a modification that I would recommend. Modified units usually perform more poorly. We have measured modified units and have found the following problems:

 a) Distortion due to opamp substitutions
 b) Phase errors between channels due to capacitor changes
 c) UltraLock(tm) rendered non-functional due to IC change
 d) Frequency response problems due to substitution of incorrect capacitor values.

 We have fixed several modified units at customer expense after they failed. This has given us the opportunity to measure the performance and inspect the workmanship. In all cases the performance was degraded. In all of these RMA cases, the modifications caused failure of the product. I do not recommend the services of modifiers. We had good reasons for using the parts that we used. Our reasons had everything to do with performance and nothing to do with cost.

 OPA627 vs. NE5532:

 The OPA627 will offer no advantage over the NE5532 in this application. High-frequency THD+N may actually be slightly higher with the OPA627. We are using the new LM4562A in the DAC1 USB. The LM4562A is pin-compatible with the NE5532, but you can't just drop it in and get all of the advantages of the DAC1 USB. We changed all of the resistors in the XLR output pads to take advantage of the high drive capability of the LM4562A."

 Pay particular attention to items B and D in the list above. The modified units we have seen have incorrect capacitor values substituted for the original parts. Substituting incorrect capacitor values will result in serious deficiencies in sound field accuracy and also freq response accuracy.

 Thanks,
 Elias_

 

6moons reviewed a modded DAC1, & the reviewer thought it was much improved over the stock DAC1. A lot of owners seem to love their Empirical Audio modded DAC1. I'm sure you frown on modding ingeneral.. I am a DAC1 owner by the way.. & think it's a fantastic DAC..


----------



## kool bubba ice

Quote:


  Originally Posted by *audioengr* /img/forum/go_quote.gif 
_If you come to the Head-Fi meet in San Jose you can hear for yourself. I can give you technical explanations about the improvements in person and you can listen to the modded DAC-1. You can hear my new headphone amp mod for it.

 Steve N.
 Empirical Audio_

 

sounds good..


----------



## boggle

I am too far away (other side of the world) to take up the Empirical audio offer of listening to the modded DAC1 - I would be interested in what people think though. Again, blind tests are a must. It is pretty easy to hear a qualitative difference when someone is standing there coaching you to hear it (by this I mean we tend to 'hear' differences that are not there). The other thing you need to do to convince is run the stats: get 10 to 20 people to do a double blind AB comparison with the same music and see if we get something significantly (in probability) different from a 50-50 split. I would almost bet a body part that this is not going to happen though.

 Suggested experimental design:

 Because we are in Headphones the 'double' part of the blind is easily cheated - just get the listener to face away from the source controller. 

 Switch between sources in some non-standard way. For example start with A for 3 minutes then reset the track and hit them with A again. Restart the track then hit em with B. Continue like this so that over 6 or eight changes there is 3 of each source: eg AABABB. The listener is simply asked to state whether the source he/she hears after the breaks (when we reset the track) is an improvement on the previous source. This will provide clear and convincing data. 

 If any doubt arises from the results it could be repeated with different music etc..


----------



## kool bubba ice

Quote:


  Originally Posted by *boggle* /img/forum/go_quote.gif 
_I am too far away (other side of the world) to take up the Empirical audio offer of listening to the modded DAC1 - I would be interested in what people think though. Again, blind tests are a must. It is pretty easy to hear a qualitative difference when someone is standing there coaching you to hear it (by this I mean we tend to 'hear' differences that are not there). The other thing you need to do to convince is run the stats: get 10 to 20 people to do a double blind AB comparison with the same music and see if we get something significantly (in probability) different from a 50-50 split. I would almost bet a body part that this is not going to happen though.

 Suggested experimental design:

 Because we are in Headphones the 'double' part of the blind is easily cheated - just get the listener to face away from the source controller. 

 Switch between sources in some non-standard way. For example start with A for 3 minutes then reset the track and hit them with A again. Restart the track then hit em with B. Continue like this so that over 6 or eight changes there is 3 of each source: eg AABABB. The listener is simply asked to state whether the source he/she hears after the breaks (when we reset the track) is an improvement on the previous source. This will provide clear and convincing data. 

 If any doubt arises from the results it could be repeated with different music etc.._

 

I'd like to compare the Stock/modded DAC1 with my DVD player & songs I know inside & out..Yes, I need a better CD player..


----------



## Davesrose

Quote:


  Originally Posted by *EliasGwinn* /img/forum/go_quote.gif 
_OPA627 vs. NE5532:

 The OPA627 will offer no advantage over the NE5532 in this application. High-frequency THD+N may actually be slightly higher with the OPA627. We are using the new LM4562A in the DAC1 USB. The LM4562A is pin-compatible with the NE5532, but you can't just drop it in and get all of the advantages of the DAC1 USB. We changed all of the resistors in the XLR output pads to take advantage of the high drive capability of the LM4562A."

 Pay particular attention to items B and D in the list above. The modified units we have seen have incorrect capacitor values substituted for the original parts. Substituting incorrect capacitor values will result in serious deficiencies in sound field accuracy and also freq response accuracy.

 Thanks,
 Elias_

 

Well shucks.....I've been completely satisfied with my stock DAC1 (it's great as it is, so why mess with perfection?). But now I'm wondering what the sonic differences are between the original DAC1 and DAC1 USB, if you guys changed op amps and redesigned the outputs. I don't use a computer as transport, so I didn't even think about considering the DAC1 USB. If it also offers improved audio for the digital audio inputs, I might just have to keep my eye on this


----------



## kool bubba ice

Quote:


  Originally Posted by *Davesrose* /img/forum/go_quote.gif 
_Well shucks.....I've been completely satisfied with my stock DAC1 (it's great as it is, so why mess with perfection?). But now I'm wondering what the sonic differences are between the original DAC1 and DAC1 USB, if you guys changed op amps and redesigned the outputs. I don't use a computer as transport, so I didn't even think about considering the DAC1 USB. If it also offers improved audio for the digital audio inputs, I might just have to keep my eye on this
	

	
	
		
		

		
		
	


	


_

 

x2..


----------



## Jon L

Quote:


  Originally Posted by *EliasGwinn* /img/forum/go_quote.gif 
_We are using the new LM4562A in the DAC1 USB. The LM4562A is pin-compatible with the NE5532, but you can't just drop it in and get all of the advantages of the DAC1 USB. We changed all of the resistors in the XLR output pads to take advantage of the high drive capability of the LM4562A."_

 

THAT IMO is huge news. LM4562 is capable of 45 mA current output and is National's newest breed of op-amps vs. NE5532, which is an ancient standard in comparison. 

 USB or not, it seems I need to do some comparisons between DAC-1 and DAC-1 USB


----------



## Iron_Dreamer

Quote:


  Originally Posted by *Jon L* /img/forum/go_quote.gif 
_THAT IMO is huge news. LM4562 is capable of 45 mA current output and is National's newest breed of op-amps vs. NE5532, which is an ancient standard in comparison. 

 USB or not, it seems I need to do some comparisons between DAC-1 and DAC-1 USB 
	

	
	
		
		

		
		
	


	


_

 

So it sounds like the DAC1 USB might perform better when used as a balanced headphone amp, as compared to the standard DAC1...


----------



## SamNOISE

[size=xx-small].[/size]
*kool bubba ice*

 First rule of life on Earth:

 Don't ever, and I mean _ever _take what 6-Moons says as anything even somewhat close to reality...

 Andrew D.
www.cdnav.com
[size=xx-small].[/size]


----------



## audioengr

Quote:


  Originally Posted by *EliasGwinn* /img/forum/go_quote.gif 
_If you are considering having your DAC1 modified, please read Benchmark's official statement regarding modifications to Benchmark products (copied directly from Benchmark DAC1 USB manual, page 2 - http://www.benchmarkmedia.com/manual...USB_Manual.pdf ):

 Benchmark’s official statement regarding modifications:

 "CAUTION: DO NOT SUBSTITUTE PARTS OR MAKE ANY MODIFICATIONS WITHOUT THE WRITTEN APPROVAL OF BENCHMARK MEDIA SYSTEMS, INC. MODIFICATION MAY CREATE SAFETY HAZARDS AND VOID THE WARRANTY.

 NOTICE: CHANGES OR MODIFICATIONS NOT EXPRESSLY APPROVED BY BENCHMARK MEDIA SYSTEMS COULD VOID THE USER'S AUTHORITY TO OPERATE THE EQUIPMENT UNDER FCC REGULATIONS."

 John Siau, the director of engineering, and chief designer of the DAC1, has addressed this, so I will copy his response here. 

 John Siau: "I have not seen, measured, or listened to a modification that I would recommend. Modified units usually perform more poorly. We have measured modified units and have found the following problems:

 a) Distortion due to opamp substitutions
 b) Phase errors between channels due to capacitor changes
 c) UltraLock(tm) rendered non-functional due to IC change
 d) Frequency response problems due to substitution of incorrect capacitor values.

 We have fixed several modified units at customer expense after they failed. This has given us the opportunity to measure the performance and inspect the workmanship. In all cases the performance was degraded. In all of these RMA cases, the modifications caused failure of the product. I do not recommend the services of modifiers. We had good reasons for using the parts that we used. Our reasons had everything to do with performance and nothing to do with cost.

 OPA627 vs. NE5532:

 The OPA627 will offer no advantage over the NE5532 in this application. High-frequency THD+N may actually be slightly higher with the OPA627. We are using the new LM4562A in the DAC1 USB. The LM4562A is pin-compatible with the NE5532, but you can't just drop it in and get all of the advantages of the DAC1 USB. We changed all of the resistors in the XLR output pads to take advantage of the high drive capability of the LM4562A."

 Pay particular attention to items B and D in the list above. The modified units we have seen have incorrect capacitor values substituted for the original parts. Substituting incorrect capacitor values will result in serious deficiencies in sound field accuracy and also freq response accuracy.

 Thanks,
 Elias_

 


 To my knowledge, an Empirical Audio modded DAC-1 has never been repaired by anyone but Empirical Audio. The only record I have of ANY repair is one where the customer twisted some heavy IC's on and off, breaking the ground connection to the circuit-board at the RCA jacks.

 The mods that we do here at Empirical Audio attempt to put the DAC-1 at the top of the price-performance curve knee, competing with DAC's in the $10-15K range. The stock DAC-1 is an excellent value IMO, and as I have heard the USB version in my reference system as well, I feel that it is definitely better than the non-USB version. It is one DAC that I can listen-to, even unmodded.

 Steve N.


----------



## audioengr

Quote:


  Originally Posted by *SamNOISE* /img/forum/go_quote.gif 
_[size=xx-small].[/size]
*kool bubba ice*

 First rule of life on Earth:

 Don't ever, and I mean ever take what 6-Moons says as anything even somewhat close to reality...

 Andrew D.
www.cdnav.com
[size=xx-small].[/size]_

 

I guess we will just have to wait for the feedback from Head-Fi meet then.....
	

	
	
		
		

		
		
	


	




 Steve N.


----------



## milkpowder

Steve, maybe you should send one of your modded units to Elias and John Siau
	

	
	
		
		

		
		
	


	




 I've been following this thread with great interest. Some of the more technical stuff is beyond me, but interesting to read nonetheless. I wouldn't mind having brushed metal silver DAC1 USB on my desk... Ah, the limitations of student life


----------



## kool bubba ice

Quote:


  Originally Posted by *SamNOISE* /img/forum/go_quote.gif 
_[size=xx-small].[/size]
*kool bubba ice*

 First rule of life on Earth:

 Don't ever, and I mean ever take what 6-Moons says as anything even somewhat close to reality...

 Andrew D.
www.cdnav.com
[size=xx-small].[/size]_

 

Well noted..


----------



## kool bubba ice

Quote:


  Originally Posted by *audioengr* /img/forum/go_quote.gif 
_I guess we will just have to wait for the feedback from Head-Fi meet then.....
	

	
	
		
		

		
		
	


	




 Steve N._

 

Yeah.. I'd love to see how far 500 could take me with your mods..


----------



## lowmagnet

Quote:


  Originally Posted by *kool bubba ice* /img/forum/go_quote.gif 
_Yeah.. I'd love to see how far 500 could take me with your mods..
	

	
	
		
		

		
		
	


	


_

 

I guarantee it'll make your wallet lighter by $500 dollars. If it ain't broke, don't break it. Just put it through an amp of choice and leave the DAC1 pristine for the day when you want to go back to hear what accuracy sounds like.


----------



## kool bubba ice

Quote:


  Originally Posted by *lowmagnet* /img/forum/go_quote.gif 
_I guarantee it'll make your wallet lighter by $500 dollars. If it ain't broke, don't break it. Just put it through an amp of choice and leave the DAC1 pristine for the day when you want to go back to hear what accuracy sounds like._

 

I feel you..I'm just never satisfied..I have yet to reach the point where my ears tell me, ok, this is when I stop..Ofcourse, I'm not made out of money either.. The search is on a budget.. I'd like to get as close as possible..


----------



## lowmagnet

Quote:


  Originally Posted by *kool bubba ice* /img/forum/go_quote.gif 
_I feel you..I'm just never satisfied..I have yet to reach the point where my ears tell me, ok, this is when I stop..Ofcourse, I'm not made out of money either.. The search is on a budget.. I'd like to get as close as possible.._

 

I really think that if you paired the DAC1 with something fuzzy it would make things better, but I don't know if I'd enjoy it. I'd just hate to see your costs sunk into something that you may hate...


----------



## kool bubba ice

Quote:


  Originally Posted by *lowmagnet* /img/forum/go_quote.gif 
_I really think that if you paired the DAC1 with something fuzzy it would make things better, but I don't know if I'd enjoy it. I'd just hate to see your costs sunk into something that you may hate..._

 

I agree.. At this point I'm just thinking about the mod.. It interests me.. Listening to it live might swing me in either direction.. Although, I don't like the idea of turning my DAC1 into a NOS dac.. I use my DAC1 for DVDS & gaming half the time..


----------



## OneStepCloser

I am planning on purchasing the USB Dac1 to pair with my Grado 325i's. Elias pointed out that for this benchmark the headphone output is supposed to be significantly better than previous versions of the Dac1. My questions is, would having a seperate headphone amp provide any benefit over just using the Dac1 headphone output/amp?


----------



## newguru

Quote:


  Originally Posted by *OneStepCloser* /img/forum/go_quote.gif 
_I am planning on purchasing the USB Dac1 to pair with my Grado 325i's. Elias pointed out that for this benchmark the headphone output is supposed to be significantly better than previous versions of the Dac1. My questions is, would having a seperate headphone amp provide any benefit over just using the Dac1 headphone output/amp?_

 

I have never used the DAC1 but just thinking out loud, if you wanted a tube sound to your amp, then the DAC1 wouldn't do it for you. I know alot of people here use the DAC1 and then output to their favorite tube amps.


----------



## OneStepCloser

Quote:


  Originally Posted by *newguru* /img/forum/go_quote.gif 
_I have never used the DAC1 but just thinking out loud, if you wanted a tube sound to your amp, then the DAC1 wouldn't do it for you. I know alot of people here use the DAC1 and then output to their favorite tube amps._

 

How about a Gilmore Lite? I have heard that they Gilmore Lite sounds really nice with the Grado line of headphones but I wonder if it would be noticable when compared to the Dac1 headphone out.


----------



## The Monkey

Quote:


  Originally Posted by *OneStepCloser* /img/forum/go_quote.gif 
_How about a Gilmore Lite? I have heard that they Gilmore Lite sounds really nice with the Grado line of headphones but I wonder if it would be noticable when compared to the Dac1 headphone out._

 

I'd go warmer than the G-lite for that combo. Plus, having heard both, I don't think the G-lite is a huge step up from the Benchmark HP out. I think the tube amp recommendation is a good one. I'm enjoying the combo of the DAC1 and the HeadRoom MH hybrid.


----------



## Davesrose

I have a SR325i. When I first got it, I wrote an impression of it compared to Senns.....just using the DAC1's hp out (the non USB version). IMO, the DAC1's amp is a good neutral "monitor" type output. Perfect for when you just want to analyze the tonality of a headphone. I'll second recommendations on a tube amp for the 325i. It seems a bit thin to me on a neutral SS amp IMO. The more tubes you give it, the more it gets bass and soundstage.


----------



## audioengr

Quote:


  Originally Posted by *newguru* /img/forum/go_quote.gif 
_I have never used the DAC1 but just thinking out loud, if you wanted a tube sound to your amp, then the DAC1 wouldn't do it for you. I know alot of people here use the DAC1 and then output to their favorite tube amps._

 

I'll be demonstrating a new mod for the DAC-1 headphone amp and comparing this to a modded Antique Audio Labs tube headphone amp at Head-Fest in April. They are closer than you imagine.....

 Steve N.
 Empirical Audio


----------



## OneStepCloser

Quote:


  Originally Posted by *The Monkey* /img/forum/go_quote.gif 
_I'd go warmer than the G-lite for that combo. Plus, having heard both, I don't think the G-lite is a huge step up from the Benchmark HP out. I think the tube amp recommendation is a good one. I'm enjoying the combo of the DAC1 and the HeadRoom MH hybrid._

 

Thanks for the input, the HeadRoom MH Hybrid is something I will consider buying later this year. I already have a Gilmore light but I had the feeling that I may not need to use it once the Dac1 arrives.


----------



## charlie1

Well now after following this thread for a while now, I just had to order one! What do some of you think on a scale of 1 to 10 will be the differance between my MICRO AMP and DAC with DESKTOP MODULE and the DAC1 USB. Lets say the MICRO is a 3. And do you think I should get a new amp to go along with it or will I be happy with it the way it is alone for now. It will be used with my RS-1s for now.


----------



## lowmagnet

Quote:


  Originally Posted by *charlie1* /img/forum/go_quote.gif 
_Well now after following this thread for a while now, I just had to order one! What do some of you think on a scale of 1 to 10 will be the differance between my MICRO AMP and DAC with DESKTOP MODULE and the DAC1 USB. Lets say the MICRO is a 3. And do you think I should get a new amp to go along with it or will I be happy with it the way it is alone for now. It will be used with my RS-1s for now._

 

Aren't you supposed to ask these questions before making a purchase? 
	

	
	
		
		

		
			





 I <3 my DAC1 with SR-225 so I'm sure the RS-1 will work out nicely. 

 I've got a pair of HD650 ordered so I can put to bed people's concerns of the DAC1 USB driving them. It's a safe bet since I've done my research and I know the DAC1 can drive them.


----------



## charlie1

Quote:


  Originally Posted by *lowmagnet* /img/forum/go_quote.gif 
_Aren't you supposed to ask these questions before making a purchase? 
	

	
	
		
		

		
		
	


	




 I <3 my DAC1 with SR-225 so I'm sure the RS-1 will work out nicely. 

 I've got a pair of HD650 ordered so I can put to bed people's concerns of the DAC1 USB driving them. It's a safe bet since I've done my research and I know the DAC1 can drive them._

 

Everyone has talked so highly about the DAC1,so with my AUDIO DISORDER I had to order one. I,m shure that I will like it for a while. BUY NOW -ASK QUESTIONS LATER.


----------



## JimP

x


----------



## schaqfu

Quote:


  Originally Posted by *lowmagnet* /img/forum/go_quote.gif 
_I've got a pair of HD650 ordered so I can put to bed people's concerns of the DAC1 USB driving them. It's a safe bet since I've done my research and I know the DAC1 can drive them._

 

Lowmagnet -- I will be very interested to hear your take on the HD 650s direct out of the DAC1 USB. I'm facing an imminent dilemma: I just ordered a HeadRoom Balanced Desktop with Home module/Home DAC and some balanced HD 650s, hoping that would be my big fat one-time, don't-bother-me-anymore, hifi purchase. The thing that's been bugging the hell out of me though is that they can't cram their Max DAC board in there, so I was having to make a tradeoff: giving up the AD1896 asynchronous chip (which my feeble mind hoped would reduce jitter) in favor of gaining a balanced headphone amp. 
	

	
	
		
		

		
		
	


	




 To my delight, the DAC1 USB looks to be everything I want in a DAC right here right now, including future ability to pair it with a balanced amplifier of my choice, perhaps such as the HeadRoom Desktop with Max module, which I think I will really like.

 My big fat dilemma question for all you geniuses on this thread now is: can I run balanced HD 650s directly out of the DAC1 USB now in balanced mode, without purchasing a separate balanced amp? I can't afford both right now, but I really like the preserved upgrade path for the future if I can stick with just the DAC for now. What do you all think? And alternatively, if I can't drive a pair of balanced cans directly out of the DAC1 USB, will driving them unbalanced for the time being be as pleasing overall as the order I'm now thinking of canceling with HeadRoom? Many thanks for the advice! 
	

	
	
		
		

		
		
	


	




 (This thread has been deliciously entertaining -- god bless that masked Elias man, whoever he was!)


----------



## schaqfu

Ooh, one more question... did we ever get an answer on whether the DAC1 USB will definitely be able to work with Vista as it does with XP? And by that, I mean can I expect in six months to be able to plug 'n play via USB direct to the DAC1 from a Vista machine and get bit-transparent, jitter-proof play with no modded drivers or difficulty using normal players like Foobar and iTunes and Winamp? Thanks!


----------



## lowmagnet

Quote:


  Originally Posted by *schaqfu* /img/forum/go_quote.gif 
_My big fat dilemma question for all you geniuses on this thread now is: can I run balanced HD 650s directly out of the DAC1 USB now in balanced mode, without purchasing a separate balanced amp?_

 

I won't be able to answer that question for you, since I'll be using the standard front outputs. If someone local has a pair of senn balanced cables, I could try a direct output out the back of the DAC1 to test that at least. I'm not really convinced of running balanced for short runs, especially since balanced headphones don't ground.

  Quote:


  Originally Posted by *schaqfu* /img/forum/go_quote.gif 
_Ooh, one more question... did we ever get an answer on whether the DAC1 USB will definitely be able to work with Vista as it does with XP?_

 

Elias stated that they tested on this platform, yes.


----------



## Davesrose

Quote:


  Originally Posted by *schaqfu* /img/forum/go_quote.gif 
_My big fat dilemma question for all you geniuses on this thread now is: can I run balanced HD 650s directly out of the DAC1 USB now in balanced mode, without purchasing a separate balanced amp? I can't afford both right now, but I really like the preserved upgrade path for the future if I can stick with just the DAC for now. What do you all think? And alternatively, if I can't drive a pair of balanced cans directly out of the DAC1 USB, will driving them unbalanced for the time being be as pleasing overall as the order I'm now thinking of canceling with HeadRoom? Many thanks for the advice! 
	

	
	
		
		

		
			



_

 

Yes, you could run the HD650 balanced off of the DAC1's XLR output. The DAC1 has a switch in back for variable output: so the volume knob will raise and lower the outputs. I'm not sure how the HD650 sounds balanced out of a DAC1 though. I just used it single ended with its amp for a few weeks before my SP SLAM came. But HeadRoom now is selling some cheaper balanced cables, so it's easier to decide if you want to try and see.


----------



## The Monkey

I had planned to balance my HD 650 cable this weekend, but didn't get to it. Will do so soon, however, and will post impressions re the balanced out from the DAC1.


----------



## schaqfu

Quote:


  Originally Posted by *Davesrose* /img/forum/go_quote.gif 
_Yes, you could run the HD650 balanced off of the DAC1's XLR output. The DAC1 has a switch in back for variable output: so the volume knob will raise and lower the outputs. I'm not sure how the HD650 sounds balanced out of a DAC1 though. I just used it single ended with its amp for a few weeks before my SP SLAM came. But HeadRoom now is selling some cheaper balanced cables, so it's easier to decide if you want to try and see._

 

Wow, I thought people were saying you could run them right out of the rear XLRs but I couldn't believe my eyes! So that switch that toggles between "variable" and "calibrated" is simply a toggle of the volume knob to control the XLR balanced output? Is that designed with the idea of hooking headphones directly up to it, or is that more because some professional grade equipment have different standards for current connecting to balanced outputs and it's important to be able to shift between different boxes? Translation: is this a hack to be plugging balanced HD650s directly into the rear XLR outputs or is this actually something contemplated by design? If this delivers a true "balanced can" experience this is an unbelievable value compared to HeadRoom's already *very* reasonably priced Balanced Desktop with integrated DAC.

 I'm dying to know if anyone has done this and can compare the experience. XLR balanced output direct to headphones from the DAC1 USB better or worse than traditional unbalanced front output from the DAC1 USB? Better or worse than first passing the balanced signal through a dedicated balanced amp like a HeadRoom Balanced Desktop? 
	

	
	
		
		

		
		
	


	










 Alert, Alert, Alert: VALUE OF THE CENTURY HERE if this works!!


----------



## jpelg

Quote:


  Originally Posted by *schaqfu* /img/forum/go_quote.gif 
_I'm dying to know if anyone has done this and can compare the experience. XLR balanced output direct to headphones from the DAC1 USB better or worse than traditional unbalanced front output from the DAC1 USB? Better or worse than first passing the balanced signal through a dedicated balanced amp like a HeadRoom Balanced Desktop? 
	

	
	
		
		

		
		
	


	










 Alert, Alert, Alert: VALUE OF THE CENTURY HERE if this works!!_

 

Check out Ferbose's famous thread here for a lot more info.

 Now as to whether you hear a significant difference, or if one exists you prefer it, is your call, and probably dependent on your headphone of choice. According to many here, Sennheisers seem to respond the greatest to balanced operation, but I'm not a proponent of either one (Senns or balanced setups).

 I owned a Benchmark DAC1 from the first production run ("ages" ago in audio years), an Apogee Mini-DAC, and now a 2006 Benchmark. The older Benchmark's headphone-out worked very well with a pair of Audio-Technica W2002's, and perked up Sennheisers a bit. If Grado's were still my headphone of choice, I'd go with the Apogee and use the headphone-out, which is much warmer than its XLR-outs.

 That said, I _love_ my current setup (DVD-A -> '06 Benchmark DAC1 -> Ultrasone Proline 750's)


----------



## EliasGwinn

The balanced XLR outputs on the DAC1 will not drive any headphones well, as is true for most all line level drivers. Headphones need to be driven with a very low impedance source for decent THD+N performance. The output impedance of the DAC1 headphone amp is nearly zero (<0.01 ohms), vs. 60 ohms for the XLR balanced line outputs. 

 The DAC1's headphone amp will drive the Senn 650's with no problem at all. The 650's are 600 ohm headphones which are a breeze to drive compared to 60 or 30 ohm headphones. In fact, with the 650's, you can't turn the DAC1 volume pot up very much without the volume being too loud!!
	

	
	
		
		

		
		
	


	




 Thats the reason the DAC1 USB has 10dB pads on the headphone outputs. This gives you the ability to really dig into the volume pot before it becomes too loud. The pads are also defeatable for lower impedance headphones.

 Thanks,
 Elias


----------



## EliasGwinn

EDIT: Please ignore...this was an accidentally repeated post (ie, same as previous post)


----------



## laxx

The HD650's are 300 ohm. =T


----------



## EliasGwinn

Quote:


  Originally Posted by *laxx* /img/forum/go_quote.gif 
_The HD650's are 300 ohm. =T_

 

You are absolutely right. I stand corrected.

 Nevertheless, the DAC1 USB can drive headphone loads as low as 30 ohms.

 Thanks,
 Elias


----------



## milkpowder

Just something I was wondering: Will the non-USB DAC1 get all the non-USB related tweaks that the DAC1 USB gets, if any? I'm seriously considering a DAC1, but probably won't be needing the USB function.

 Thanks in advance.


----------



## The Monkey

Quote:


  Originally Posted by *milkpowder* /img/forum/go_quote.gif 
_Just something I was wondering: Will the non-USB DAC1 get all the non-USB related tweaks that the DAC1 USB gets, if any? I'm seriously considering a DAC1, but probably won't be needing the USB function.

 Thanks in advance._

 

x2, and in addition: if yes, will those of us with the DAC1 be able to send in for an upgrade?


----------



## schaqfu

Quote:


  Originally Posted by *EliasGwinn* /img/forum/go_quote.gif 
_The balanced XLR outputs on the DAC1 will not drive any headphones well, as is true with most all line level outputs. Headphones require a very low impedance source driving them for decent THD+N performance. The output impedance of the DAC1 headphone amp is nearly zero (<0.01 ohms), vs. 60 ohms for the XLR balanced outputs. 

 The DAC1's headphone amp will drive the Senn 650's with no problem at all. The 650's are 600 ohm headphones which are a breeze to drive compared to 60 or 30 ohm headphones. In fact, with the 650's, you can't turn the DAC1 volume up very far without the volume being too loud!!
	

	
	
		
		

		
		
	


	




 Thats the reason the DAC1 USB has 10dB pads on the headphone outputs. This gives you the ability to really dig into the volume pot before it becomes too loud. The pads are also defeatable for lower impedance headphones.

 Thanks,
 Elias_

 

Thank you, Elias. That hits right to the heart of my question. But my motivation in asking is different -- I wasn't interested in connecting the HD650s out of the XLR outputs because I was concerned about achieving sufficient output. I am totally sold that the DAC1 USB's very fine headphone amp is more than enough. I was interested in the XLR outputs because they are balanced, and some among the Head-Fi community, most notably HeadRoom Corp., have burned oodles of hours evangelizing the benefits of driving headphones in balanced fashion out of a balanced amp. (See HeadRoom's description of their new budget-ended balanced amp here: http://www.headphone.com/products/he...lanced-amp.php) I would be purchasing Sennheiser HD650s with a modified balanced output XLR cable, totally eliminating the 1/4 TRS connector.

 The holy grail of bargain-finding I was hoping to pull off here was taking advantage of the DAC1 USB's powerful balanced analog output to completely skip the $1000+ separate balanced amplifier between the DAC1 USB and the balanced HD650s. This would be earth-shattering value as far as I'm concerned, since I had already decided to spend more on such an amp alone than on the DAC1 USB, as have many others.

 Given that we can attenuate the output of the DAC1 USB's balanced XLR outputs to such a profoundly huge degree -- up to and including the -30Db jumper pads and the entire range of the volume potentiometer -- don't you think it's possible that the DAC1 USB could be used to deliver a quality balanced output signal directly to high impedance headphones like the 300ohm HD650s? Oh pretty please? If, like me, you believe the Sennheisers benefit from a balanced versus unbalanced signal, this would save me the extra cost of a dedicated balanced amp which, after all, would essentially be taking the DAC1 USB's balanced XLR output and, in all likelihood, adding even more power to it before sending it to the headphones.

 Thanks so much for your thoughts on this. I honestly believe that if this is an even accidental benefit of your excellent engineering and marketing design, this would double your market among the headphone audiophile community in one fell swoop.


----------



## schaqfu

Yo ho, the innards start to bubble to the surface. It seems Benchmark's "Advanced USB" breakthrough was the work of a third party company called CEntrance. 

http://appleproaudio.com/index.php?n...rticle&sid=732

 Don't get me wrong: there is nothing wrong at all with hiring consultants and buying technology strategically and intelligently. I'd say Benchmark made a Class A business decision here. I'm glad they did.


----------



## milkpowder

Ah! Maybe that's why there's such a significant price premium
	

	
	
		
		

		
		
	


	




 Then again, as long as the product lives up to its expectations, price is not really an obstacle.


----------



## lowmagnet

Quote:


  Originally Posted by *schaqfu* /img/forum/go_quote.gif 
_Don't get me wrong: there is nothing wrong at all with hiring consultants and buying technology strategically and intelligently. I'd say Benchmark made a Class A business decision here. I'm glad they did._

 

CEntrance are also trying to wrap their collective brains around Firewire audio and make it a time-critical and viable platform for high-bandwidth audio. More power to 'em.


----------



## schaqfu

Quote:


  Originally Posted by *lowmagnet* /img/forum/go_quote.gif 
_CEntrance are also trying to wrap their collective brains around Firewire audio and make it a time-critical and viable platform for high-bandwidth audio. More power to 'em._

 

I never quite understood this -- are there some advantages to Firewire that people don't want to give up for USB 2.0? Macs support USB 2.0 now, don't they? Is it just that there's a lot of stranded investment in Firewire equipment? I guess Firewire 800 is faster, and would acquit well on raw data transfer stuff. Why all the defensive wars for Firewire?


----------



## lowmagnet

Quote:


  Originally Posted by *schaqfu* /img/forum/go_quote.gif 
_I never quite understood this -- are there some advantages to Firewire that people don't want to give up for USB 2.0? Macs support USB 2.0 now, don't they? Is it just that there's a lot of stranded investment in Firewire equipment? I guess Firewire 800 is faster, and would acquit well on raw data transfer stuff. Why all the defensive wars for Firewire? 
	

	
	
		
		

		
		
	


	


_

 

Firewire is designed more for real-time and streaming, which makes it proper for audio. But, of course, everyone fought over the fw standards for audio, and now you have consumer hardware that 'supports' but not fully like you can't run jvc with mitsu without issues. I've had nothing but trouble with the format mainly because of the connector shape and the fragility of a connection when bumped 
	

	
	
		
		

		
		
	


	




 800 has nice bandwidth, and usb2 has the same bandwidth as fw400. both good tech, and as an Apple user I couldn't be happier to have the choice of USB 1/2 and Fw400/800 out of the box.

 It's not a religious thing to Mac users, since it's been a while since our platform was USB1 only. I think Apple was trying to keep the FW400 platform viable and that's why we fell behind there. I also think they learned their lesson there, and embrace change a lot more.

 But this is a derail, and mods can move it out if they wish.

 I'm listening to Béla Fleck and the Flecktones "Left Of Cool" album on my DAC1 right now and WOW. ^_^;


----------



## schaqfu

Quote:


  Originally Posted by *lowmagnet* /img/forum/go_quote.gif 
_I'm listening to Béla Fleck and the Flecktones "Left Of Cool" album on my DAC1 right now and WOW. ^_^;_

 

No burning desire to experiment with balanced XLR output direct to your Grados, or, more understandably, other cans, eh? The native sound is good enough not to care? Do you know if the headphone amp components of the DAC1 USB have been upgraded since Iron_Dreamer did his seminal review a year ago putting it at the bottom of the group of DACs?


----------



## kool bubba ice

Quote:


  Originally Posted by *schaqfu* /img/forum/go_quote.gif 
_Wow, I thought people were saying you could run them right out of the rear XLRs but I couldn't believe my eyes! So that switch that toggles between "variable" and "calibrated" is simply a toggle of the volume knob to control the XLR balanced output?* Is that designed with the idea of hooking headphones directly up to it*, or is that more because some professional grade equipment have different standards for current connecting to balanced outputs and it's important to be able to shift between different boxes? Translation: is this a hack to be plugging balanced HD650s directly into the rear XLR outputs or is this actually something contemplated by design? If this delivers a true "balanced can" experience this is an unbelievable value compared to HeadRoom's already *very* reasonably priced Balanced Desktop with integrated DAC.

 I'm dying to know if anyone has done this and can compare the experience. XLR balanced output direct to headphones from the DAC1 USB better or worse than traditional unbalanced front output from the DAC1 USB? Better or worse than first passing the balanced signal through a dedicated balanced amp like a HeadRoom Balanced Desktop? 
	

	
	
		
		

		
		
	


	










 Alert, Alert, Alert: VALUE OF THE CENTURY HERE if this works!!_

 

No.. The XRL were not made for headphones but for loud speakers..But I hear all headphones benefit when balanced..especially higher OHM cans..I have yet to hear someone say they thought their cans single ended sounded better then when they were balanced..


----------



## schaqfu

Quote:


  Originally Posted by *kool bubba ice* /img/forum/go_quote.gif 
_No.. The XRL were not made for headphones but for loud speakers..But I hear all headphones benefit when balanced..especially higher OHM cans..I have yet to hear someone say they thought their cans single ended sounded better then when they were balanced.._

 

See http://www.head-fi.org/forums/showthread.php?t=89411

 In addition to Iron_Dreamer, who has put a great deal of thought into it, and made it work more successfully than the front panel headphone jacks, there appears to be a healthy community of 'amp-less' head-fiers who are going balanced directly out of their DACs into their HD-650s. The consensus seems to me to be that it doesn't take the place of a dedicated balanced headphone amp, but it's pretty darn good and the tradeoffs of tonal distortion are either inaudible or greatly outweighed by the increase in depth, soundstage, and especially articulation of bass. 

 All that and you don't even have to spring for a balanced amp?! That is too good to pass up trying. Another encouraging thing is, based on my readings of the threads, deriving the sonic benefits of balanced implementations is not terribly dependent upon the quality of cable used, unlike single-ended traditional stages. So I bought two pairs of replacement stock HD650 cable from Sennheiser for $12 apiece, and a couple of female XLR adapter for three bucks apiece which I will use to replace the TRS 1/4 plug on the cable, and hopefully by this time net week I'll be swimming in $3k waters for a mere $1250. 

 I'd still be very curious to hear Elias' considered thoughts on this, given we now know how widespread and loved a practice it is. You've shattered "general knowledge' before, in this very thread, so it would be great to know if you think there is any method most likely to approach optimal results.

 Thanks!


----------



## EliasGwinn

Quote:


  Originally Posted by *milkpowder* /img/forum/go_quote.gif 
_Just something I was wondering: Will the non-USB DAC1 get all the non-USB related tweaks that the DAC1 USB gets, if any? I'm seriously considering a DAC1, but probably won't be needing the USB function.

 Thanks in advance._

 

If I understand your question correctly, you are asking if there are additional features on the DAC1 USB that are not available on the DAC1 (non-USB)...correct? 

 There are several features on the DAC1 USB which are not available on the DAC1, and they are as follows:

 - Selectable gain range for headphone amp
 -- Lets you select the optimal range for your specific headphones so that the volume knob can be utilized more optimally

 - Main output mutes upon headphone insertion (defeatable)
 -- The analog outputs on the rear of the DAC1 USB will be muted when you insert the headphone plug if this feature is enabled.

 - High-Current output drivers
 -- The XLR and RCA outputs can now drive longer cables, low-impedance loads, high-capacitance loads, and/or high-inductance loads without any loss in THD+N performance

 - Advanced USB Audio for true native 96/24 bit-transparent playback
 -- No drivers or configuration necessary...plug it in and immediately get bit-transparency at rates up to and including 96/24

 Unfortunately, the DAC1's are not able to be upgraded to include these features.

 Thanks,
 Elias


----------



## lowmagnet

Quote:


  Originally Posted by *schaqfu* /img/forum/go_quote.gif 
_No burning desire to experiment with balanced XLR output direct to your Grados, or, more understandably, other cans, eh? The native sound is good enough not to care? Do you know if the headphone amp components of the DAC1 USB have been upgraded since Iron_Dreamer did his seminal review a year ago putting it at the bottom of the group of DACs?_

 

I'm not going to mod my Grados because they're hard wired. But since Sennheiser HD650s have those clip-in wires, I'm willing to try the output from the DAC1 on them. I'm not going to spend upwards of $100 on a cable to do so, however. If someone in the Raleigh area has a set of cables we can try them out.

 Since Iron_Dreamer's review of the previous version of the DAC1, the line stage has been improved, and the HPA-2 has been updated to handle 30-600 ohms instead of just 30-300 ohms. Of course the Senns were right on that border of the old amp design, which could have been a problem in the past.


----------



## EliasGwinn

Quote:


  Originally Posted by *schaqfu* /img/forum/go_quote.gif 
_Yo ho, the innards start to bubble to the surface. It seems Benchmark's "Advanced USB" breakthrough was the work of a third party company called CEntrance. 

http://appleproaudio.com/index.php?n...rticle&sid=732

 Don't get me wrong: there is nothing wrong at all with hiring consultants and buying technology strategically and intelligently. I'd say Benchmark made a Class A business decision here. I'm glad they did._

 

Benchmark hired Centrance as an independent programming contractor for the USB firmware. This press release is intended to celebrate their companies new technology. However, this programming design effort was nearly an equal collaboration between both parties. From Benchmark, specifically John Siau, the director of engineering, contributed huge amounts of troubleshooting programming code relating to issues of the typical dreaded USB audio "hiccups". The troubleshooting process took months of correspondence between our companies. The result, however, is very satisfactory, if I may say so!!
	

	
	
		
		

		
		
	


	




 Thanks,
 Elias


----------



## EliasGwinn

As I mentioned before, we do not recommend using the XLR outputs to drive headphones. However, with that being said, I would be interested to try it 
	

	
	
		
		

		
		
	


	




 !!

 Thanks,
 Elias


----------



## little-endian

Hi there Elias,

 it is great that there are employees like you who enjoy it to share detailed information that only 'insiders' know but which are of public interest.

 I'm a user of the 'classical' DAC1. Actually, I'm pretty satisfied with it, although I have to admid that I can't really distinguish between it and a low priced Yamaha DVD-S540 (so far, tested with 44,1 kHz / 16-Bit material only).

 However, I recognized terrifying differences when it comes to headphone usage. My Sennheiser HD-650 for example sounds very thin when directly connected to a notebook equipped with the Intel HDA Codec for example. The headphone output of my David Hafler preamp is much better but doesn't stand a chance compared to the HPA2 the DAC1 features.
 I also noticed the L/R-balance which tends to the right channel below the 3rd step of the volume control in my case.
 The gain reduction which is available on the DAC1 USB should of course attenuate this issue because the inaccuracy is spread over a wider range. But I wonder if this attenuation leads to an even more reduced THD or whatever. I mean, does it make any difference if the potentiometer or this jumper element reduces the volume except for the aspect of the loudness comfort and the improved L/R balance? At least when listening to HD 650 with their 300 Ohm. Since they were recommended by Benchmark in the first place (long before the USB version) I would suppose that the sonic results already reached the maximum. Am I correct? I hope so.  Except that, I've heard that electronical volume controls don't suffer from any gain asynchronism. Why does Benchmark then use potis instead of such a solution?

 From what I read in the manual, the source selector on the front doesn't have fixed positions anymore and goes back to its centered position instead, "scrolling" the inputs up and down. Is this true?

 When it comes to the much-lauded USB input which doesn't need 3rd party drivers, etc., I want to note here that one could still even beat this solution because it is limited to 96 kHz support. Assumed that there is any device from USB to S/PDIF which works up to 192 kHz (whether or not it used native drivers would be secondary for me as long as it is bitperfect). Not that I had heard about any such yet - just a thought. 
 At this point a further question arises for the first DAC1 editions (with the removable dustcap for the toslink receiver): In the US-manual they wrote that sample rates up to 192 kHz would be supported optically while the German translation claimes 96 kHz only. I asked the distributor who explained that the older versions had not always worked reliably above 96 kHz. It is open what means "not always". It still seems possible so far. Did Benchmark ever change something in regard to the S/PDIF receivers?

 Some people claim that they hear differences between DAC1s which were manufactured some years ago and more recently. Again - where there ever any changes which could have any impact to the sound?

 And a last question for now (although off-topic per se):

 How about the handling of "intersample peaks" of the DAC1 in general?

 Many thanks in advance!


----------



## schaqfu

Quote:


  Originally Posted by *EliasGwinn* /img/forum/go_quote.gif 
_As I mentioned before, we do not recommend using the XLR outputs to drive headphones. However, with that being said, I would be interested to try it 
	

	
	
		
		

		
		
	


	




 !!

 Thanks,
 Elias_

 

Sounds like a plan!! 
	

	
	
		
		

		
		
	


	




 Thanks, Elias, for all your superlative information!! Really, you've done more to enhance the customer experience by participating in this one online thread than almost any other company does even directly with their customers. And in my case (as, I'm sure, for others), you've heavily influenced my purchase decision.


----------



## schaqfu

Quote:


  Originally Posted by *EliasGwinn* /img/forum/go_quote.gif 
_There are several features on the DAC1 USB which are not available on the DAC1, and they are as follows:

 - High-Current output drivers
 -- The XLR and RCA outputs can now drive longer cables, *low-impedance loads*, high-capacitance loads, and/or high-inductance loads *without any loss in THD+N performance*_

 

Since the balanced XLR outputs have been substantially improved in the DAC1 USB, and "can now drive... low-impedance loads... without any loss in THD+D performance," this raises the question: will balanced headphones, HD 650s in particular, now play even better directly out of the balanced outputs? At 300 ohms they certainly qualify as low-impedance loads. Will the relatively minor distortion reported by current users playing their HD 650s out of the DAC1 balanced outputs be reduced? Any other benefits we can anticipate?


----------



## Gatticus

Quote:


  Originally Posted by *The Monkey* /img/forum/go_quote.gif 
_Are there any other balanced output USB DACs on the market?_

 

For 200 bucks you can get an EMU 0404 USB soundcard that has balanced outputs. 

http://www.emu.com/products/product....&product=15185


----------



## laxx

Quote:


  Originally Posted by *Gatticus* /img/forum/go_quote.gif 
_For 200 bucks you can get an EMU 0404 USB soundcard that has balanced outputs. 

http://www.emu.com/products/product....&product=15185_

 

There's also the Stello DA 220 MK2 and PS Audio Digital Link3 IIi.


----------



## EliasGwinn

Little-endian:

 I will try to answer all your questions...let me know if I miss any or misunderstand any.

 1. L/R issues below the 3rd step of the volume pot

 Any potentiometer (in real life, at least) will be inaccurate at this end of their range. The reason is because of real life mechanical limitations. Specifically, the potentiometer wiper approaches the end of the resistive element and approaches a drop off. This drop off causes inaccuracies. Unfortunately, there isn't a dual-pot (stereo pot) in existence that defeats this limitation.

 It is for this reason that we add the output pads (the DAC1 USB has selectable pads for the headphone output). These pads are intended to provide a listening level range so that the user's comfortable listening level is above the inaccurate portion of the volume pot. These pads do not change the THD+N performance, they simply attenuate to optimize the volume level range.

 2. Why doesn't the DAC1 use electronic volume control?

 We have not found, neither in our competitors products nor as available technology, any electronic volume controls that perform to our standard. All currently available electronic volume controls add significant amounts of distortion, so we have elected to maintain signal integrity rather then use these. It is unfortunate, however, as an electronic volume control would enable us to use a remote control, which has been requested by an innumerable amount of customers. However, we feel it is in the best interest of the goal of the product to maintain the signal integrity.

 3. Is the DAC1 USB input selector a scrolling selector?

 Yes. The switch is a momentary 3-position switch. You can scroll upwards through the inputs, or downwards through the inputs. Also, it is round-robin format - that is, the selections will go 3 -> 2 -> 1 -> 4 -> 3 -> 2 ....

 4. Why doesn't the USB input do 192 kHz?

 96kHz, 24-bit audio is the maximum bandwidth for the USB 1.1 protocol. 192 kHz is possible with USB 2.0. However, we elected to use the 1.1 protocol because there are A LOT (really, a whole bunch) of customers who do not have USB 2.0 ports. Conversely, we have a very small number of customers who are looking to stream 192 kHz audio via a computer. Hence, the decision was made to use the 96 kHz/ 24-bit, USB 1.1 interface.

 5. Do old DAC1's optical ports handle 192 kHz?

 DAC1's manufactured after 3/04 have optical ports that can handle 192 kHz.

 6. Are there sonic differences in old DAC1's vs. new DAC1's (non-USB)?

 There are no measurable differences between the older DAC1's and newer DAC1's. The main difference between the two is the RCA outputs are 30-ohms on DAC1's manufactured after 3/04. This will create a sonic difference DEPENDING ON THE EQUIPMENT DIRECTLY DOWNSTREAM OF THE DAC1. In other words, if you are driving a difficult load (an amplifier with a high capacitive input or low-impedance input, for example), the newer DAC1 will drive those better then the older. On properly designed loads, the older and newer DAC1's should perform identically.

 7. How does the DAC1 handle intersample peaks?

 Intersample peaks are true overs (greater then maximum possible value), thus will distort when played back. This is a very good question, however, as it is a topic the AES has recently been trying to tackle. Refer to AES R7-2006 for the latest discussion and details on standardizing measurement/metering practices for this.

 Thanks,
 Elias


----------



## Headphony

Quote:


  Originally Posted by *schaqfu* /img/forum/go_quote.gif 
_Thanks, Elias, for all your superlative information!! Really, you've done more to enhance the customer experience by participating in this one online thread than almost any other company does even directly with their customers. And in my case (as, I'm sure, for others), you've heavily influenced my purchase decision. 
	

	
	
		
		

		
			



_

 

I'll second that! What a great thread. I've had a DAC1 on my wish list long enough and couldn't resist the temptation anymore. Just pulled the trigger.


----------



## lowmagnet

I got the HD650 in today and the DAC1 is driving them really well! Also, I have my Grados plugged into the second plug and when I did, nothing happened to the volume of the main plug. Awesome design.


----------



## schaqfu

Quote:


  Originally Posted by *lowmagnet* /img/forum/go_quote.gif 
_I got the HD650 in today and the DAC1 is driving them really well! Also, I have my Grados plugged into the second plug and when I did, nothing happened to the volume of the main plug. Awesome design._

 

Will love to hear any further impressions you may have with extended listening, lowmagnet! I'm itching for mine but it won't arrive until next Tuesday at the earliest. Blast.

 Interesting question, since you've already tested it: what is the difference in volume between your Senns and Grados when they're plugged in simultaneously to the two headphone jacks? Is it correct that the higher impedance Senns are considerably lower volume, or is the difference not that major?

 Also, to anyone else, I'd still be very interested to hear thoughts on my question in my previous post: whether the upgraded XLR balanced outputs of the DAC1 USB, as compared to the original DAC1, are likely to do an even better job of driving balanced headphones directly? If so, that could provide a whole other benefit for future purchasers to consider in choosing between the two, and shrinks the $300 disparity some more.


----------



## lowmagnet

Quote:


  Originally Posted by *schaqfu* /img/forum/go_quote.gif 
_Interesting question, since you've already tested it: what is the difference in volume between your Senns and Grados when they're plugged in simultaneously to the two headphone jacks? Is it correct that the higher impedance Senns are considerably lower volume, or is the difference not that major?_

 

It's funny you mention that. After I plugged in the Grados, I put the same channel on both ears from the two headphones and listened to the music for a while. They're pretty close to each other, with the Grado at 98dB/V and the Sennheisers at 103dB/V.


----------



## BobTurbo

Hi Elias and head-fi members 
	

	
	
		
		

		
		
	


	




 Thanks for helping to debunk some of the garabge that fills the internet forums. I do not believe a single thing I read from internet forums or anywhere other than extremely trusted sources, so for that reason I want to ask whether EAC is necessary or a good idea, or with modern CD writers can I just use WMP11 to rip my audio CDs? Either way, I am going to use WMP for playback, it is just a question of whether I should do something like EAC->WAV->FLAC, EAC->WAV->WMA Lossless, EAC->WAV, WMP11->WAV, or WMP11->WMA Lossless. The reason I put in WAV is I was just wondering if the decoding of lossless formats such as FLAC, WMA Lossless, etc, adds any inaccuracy at any stage. I have read of some devices that rip things to linear PCM, so I am wondering why they chose that instead of a space-saving lossless format.

 I am thinking of having a setup as follows:

 Windows XP->Seagate HDD->WMP11->usb connection to benchmark dac1 usb->balanced output to ESI Near06 active monitors.

 Also I will have to read the dac's manual, but is it a good idea to disable any soundcards/onboard sound that may be present on the computer to limit any possible conflict?


----------



## little-endian

Hi BobTurbo,

 you said you don't believe a single thing you read in internet forums. This way it will be difficult now to tell you something.  Despite the fact that there is a lot of garbage out of the internet, I wouldn't see it so negative. Forums can be an ingenious source for information since you're able to meet people you probably wouldn't be aware of in "real life". Elias is propably the best example.

 When it comes to your backup intensions I may allay you: If the implementation of the used codec is flawless, then the whole process is lossless, hence its name.  Which lossless codec to choose depends here on compatibility, ressource usage, the question if it is open source, etc. Especially the last one is a real argument for future backups. How can you be sure that Microsoft won't change its WMA format one day, for example?

 After comaring some of them, my choice sticks on WavPack for now. It has great compression ratio and is - compared to this - very fast. On my PC it takes fewer ressources than FLAC while compressing even better. LA (lossless audio) files are a bit smaller, but it takes extremely long to convert to this format. Except that there is just a buggy plugin for playback in foobar.

 If you worry about the quality of the resulting lossless material you should take care about the process to retrieve the PCM ('WAV' files is most often just uncompressed PCM data including a 44 Byte header btw.). Here, EAC is the best program to do so. If the CD as well as the reader is flawless, every other tool would lead to the same result except the drive's read offset - but that doesn't have any impact on the sound quality. Neither has the drive itself as long as the data will be transferred without errors on the user data level (no E32 errors). Other info claiming that you need special optical pickups, belt-driven players, etc. to read an audio-cd perfectly may be put in the garbage section like the so-called professional journals (sadly one of the "trusted sources" for not few).

 Disabling other sound devices on your system won't harm but won't be absolutely necessary as long you can choose which devices should get the audio data in the program's setup.

 Kind regards,

 little-endian


----------



## schaqfu

Quote:


  Originally Posted by *lowmagnet* /img/forum/go_quote.gif 
_It's funny you mention that. After I plugged in the Grados, I put the same channel on both ears from the two headphones and listened to the music for a while. They're pretty close to each other, with the Grado at 98dB/V and the Sennheisers at 103dB/V._

 

Really?! Okay, who can explain to me the physics of how it's possible for 32 ohm cans and 300 ohm cans to have roughly the same volume when each is receiving an identical current.??? If this is the case in general, or at least sometimes, then what is the significance of higher impedance cans?


----------



## schaqfu

Quote:


  Originally Posted by *little-endian* /img/forum/go_quote.gif 
_Hi BobTurbo,

 you said you don't believe a single thing you read in internet forums. This way I will be difficult now to tell you something.  Despite the fact that there is a lot of garbage out of the internet, I wouldn't see it that negative. Forums can be an ingenious source for information since you're able to meet people you probably wouldn't be aware of in "real life". Elias is just the best example.

 When it comes to your backup intensions I may allay you: If the implementation of the used codec is flawless, then you whole process is flawless, hence its name.  Which lossless codec to choose depends here on compatibility, ressource usages, the question if it is open source, etc. Especially the last one is a real argument for future backups. How can you be sure that Microsoft won't change its WMA format some day, for example?

 After comaring some of them, my choice sticks on WavPack for now. It has great compression ratio and is - compared to this - very fast. On my PC it takes fewer ressources than FLAC while compressing even better. LA (lossless audio) files are a bit smaller, but it takes extremely long to convert to this format. Except that there is just a buggy plugin for playback in foobar.

 If you worry about the quality of the resulting lossless material you should take care about the process to retrieve the PCM ('WAV' files is most often just uncompressed PCM data including a 44 Byte header btw.). Here, EAC is the best program to do so. If the CD as well the reader is flawless, every other tool would lead to the same result except the drive's read offset - but that doesn't have any impact on the sound quality. Neither has the drive itself as long as the data will be transferred without errors on the user data level (no E32 errors). Other info claiming that you need special optical pickups, belt-driven players, etc. to read an audio-cd perfectly may be put in the garbage section like the so-called professional journals (sadly one of the "trusted sources" for not few).

 Disabling other sound devices on your system won't harm but won't be not absolutely necessary as long you can choose which devices should get the audio data in the program's setup.

 Kind regards,

 little-endian_

 

Great post, but I think we're running off topic. Two little guilty follow ups, and then let me see if I can artfully pull it back on topic...

 [guilty follow up #1] Just one note to BobTurbo... I believe when using lossless encoding (irrespective of compression), barring physical damage to the storage medium itself and the resulting loss of bits, there are only two points in the process that can introduce any inaccuracy to the digital audio -- reading of the source CD at the time of ripping, and timing-related data transmission problems in the actual delivery of bits to the DAC, kinda sorta broadly referred to as jitter, or bit loss due to underrun. If every bit is read correctly off the source CD -- which is quite likely with some basic form of error correction enabled -- then nothing you do in and around a losslessly encoded file will introduce any artifacts or any imperfections at all. As in, you can transcode from WMA lossless to ALAC to FLAC back to WMA lossless back to ALAC 100 times and the resulting binary output file will remain identical. Lossless compression is mathematical perfection.

 I guess I shouldn't say 100%, because there is always a miniscule chance that when moving any given bit of data from one sector of a hard disk to another the bit could be lost in the shuffle. But with checksum functions, and all manner of other error-correction algorithms, not to mention the very stable properties of a magneto-optical medium, it is extremely rare. Like, very. It's the same risk you take any time you move any kind of file at all on your hard drive. Like dragging a picture of mom from the "vacation" folder to the "family" folder. I'm sure you're not concerned about losing bits by doing that. Same with transcoding and manipulating lossless files -- no worries.

 [guilty follow up #2] little-endian... I normally completely agree with you about choosing open formats that won't lock you in to any single vendor. But in the case of just about all the lossless audio codecs, I think it's irrelevant. It's irrelevant because each of them, even mighty Apple's and Microsoft's, have publicly available decoders. It doesn't even matter if the encoders are closed, like with Apple. When talking about lossless files, if at any point in the future Microsoft decides to charge you a royalty for using WMA lossless, or Apple revokes your right to use ALAC, you can avoid lock-in by simply transcoding your files to another lossless format of your choice. That's a clear escape hatch that is 100% effective at preventing lock-in, and lock-in is the enemy as we both agree. It's a luxury that's peculiar to lossless audio (and other lossless data file formats) because they are true clones even when transcoded. That's not the case when, say, transcoding a Word document into a Star Office document, or even a lossy mp3 into a lossy AAC. In those situations the initial choice of file format is critical because each and every departure therefrom carries with it a departure tax -- loss of some data. Not so here. So my advice to BobTurbo would be to select the codec that most suits his needs right now, based on all the other criteria listed by little-endian, and rip away direct from CD to the resulting file rather than running it through the wav ringer first.

 [The return to topic] Ah, but here's how I hope this will come back on topic for this thread -- Elias mentioned several pages ago that Benchmark has not yet evaluated and come up with recommendations for media players in Windows that will function properly with the DAC1 USB's bit transparent playback over USB. He did indicate Foobar was extremely likely, but others were certain to join. Knowing this information would influence my choice of codec -- can we please ask Benchmark to give us a definitive list, even if temporarily partial, of media players that will work seamlessly for bit transparent playback in Windows on the DAC1 USB? For instance, I very much hope iTunes is on that list (with the volume slider turned all the way up and all sound enhancers/equalizers/volume levelers disabled, of course) because all my files are in ALAC and I just plain like it that way and don't want to have to move to FLAC and don't want to have to move to Foobar. I also certainly hope that Windows Media Player is on there, because it's the only program I'm interested in using for playback of 96 KHz / 24-bit files, in the form of WMA lossless high resolution tracks. When can we expect an answer and guidance from Benchmark? As with all your other answers, this one is much appreciated, Elias!!


----------



## lowmagnet

Quote:


  Originally Posted by *schaqfu* /img/forum/go_quote.gif 
_Really?! Okay, who can explain to me the physics of how it's possible for 32 ohm cans and 300 ohm cans to have roughly the same volume when each is receiving an identical current.??? If this is the case in general, or at least sometimes, then what is the significance of higher impedance cans?_

 

It's because they aren't receiving identical current. There are two HPA-2 circuits on the board of the DAC1, and they are drawing different current to power their respective loads.


----------



## EliasGwinn

BobTurbo,

 I haven't tested EAC or any other ripper, so I can't really answer your question. I have read a little bit about it, however, and the EAC technology makes sense. But I have no basis to give a qualified response. Sorry.

 Thanks,
 Elias


----------



## audioengr

Quote:


  Originally Posted by *lowmagnet* /img/forum/go_quote.gif 
_It's because they aren't receiving identical current. There are two HPA-2 circuits on the board of the DAC1, and they are drawing different current to power their respective loads._

 

The current is a function of the voltage and load impedance. The voltage output is a constant in both cases, so the change in load impedance causes a change in current.

 Steve N.


----------



## schaqfu

Quote:


  Originally Posted by *EliasGwinn* /img/forum/go_quote.gif 
_BobTurbo,

 I haven't tested EAC or any other ripper, so I can't really answer your question. I have read a little bit about it, however, and the EAC technology makes sense. But I have no basis to give a qualified response. Sorry.

 Thanks,
 Elias_

 

Elias, any update yet on what media players will work seamlessly, and bit-transparently, with the USB input of the DAC1 USB? See bottom of my previous post. I'm dying to confirm that iTunes, Windows Media Player and Foobar will all be supported. Thanks!


----------



## lowmagnet

Ok, I took them off my head and listened to them side by side, and the Grados seemed louder. Beats me.


----------



## EliasGwinn

Quote:


  Originally Posted by *schaqfu* /img/forum/go_quote.gif 
_Elias, any update yet on what media players will work seamlessly, and bit-transparently, with the USB input of the DAC1 USB? See bottom of my previous post. I'm dying to confirm that iTunes, Windows Media Player and Foobar will all be supported. Thanks!_

 

iTunes will play bit transparently under the following conditions:

 1. Volume control is set to 100% (VERY important, as this volume control will cause significant distortion at any other setting)

 2. Sample-rate set to the corresponding sample-rate of the audio being played. This is done in QuickTime, believe it or not. Go to QuickTime -> Edit -> Edit Preferences -> QuickTime Preferences -> Audio -> Sound Out -> Rate. Also, in this same window, change size to 24 bit, even if the audio is 16-bit.

 3. Disable "SoundCheck", "Sound Enhancer", or any other audio processing.

 You should be good to go then...

 Thanks,
 Elias


----------



## JimP

x


----------



## schaqfu

Quote:


  Originally Posted by *EliasGwinn* /img/forum/go_quote.gif 
_iTunes will play bit transparently under the following conditions:

 1. Volume control is set to 100% (VERY important, as this volume control will cause significant distortion at any other setting)

 2. Sample-rate set to the corresponding sample-rate of the audio being played. This is done in QuickTime, believe it or not. Go to QuickTime -> Edit -> Edit Preferences -> QuickTime Preferences -> Audio -> Sound Out -> Rate. Also, in this same window, change size to 24 bit, even if the audio is 16-bit.

 3. Disable "SoundCheck", "Sound Enhancer", or any other audio processing.

 You should be good to go then...

 Thanks,
 Elias_

 

Thanks again, Elias.

 Okay, I've made the adjustments, but now does anyone know how to import WMA lossless tracks (or any format for that matter) of 96 KHz / 24 bit into iTunes without iTunes automatically downconverting them to 44.1/16? That's maddening.


----------



## EliasGwinn

Quote:


  Originally Posted by *JimP* /img/forum/go_quote.gif 
_Any particular parameters to be set in using Foobar (FLAC files)?_

 

1. Volume at 100%

 2. 24-bit output

 3. Disable any audio DSP effects (by default, there are none, but just in case...)

 Foobar will automatically output at the correct sample-rate, so no need to worry about that.

 Thanks,
 Elias


----------



## schaqfu

Quote:


  Originally Posted by *EliasGwinn* /img/forum/go_quote.gif 
_1. Volume at 100%

 2. 24-bit output

 3. Disable any audio DSP effects (by default, there are none, but just in case...)

 Foobar will automatically output at the correct sample-rate, so no need to worry about that.

 Thanks,
 Elias_

 

Same with Windows Media Player? Including hi res files?


----------



## JimP

x


----------



## EliasGwinn

Quote:


  Originally Posted by *schaqfu* /img/forum/go_quote.gif 
_Same with Windows Media Player? Including hi res files?_

 

Windows Media Player needs a third party plug-in to play 24-bit files (at least last time I checked). The volume control on WMP is actually very well designed...it has very low distortion...so you don't have to worry about using it. However, for bit-transparency, it must be set to 100%.

 Thanks,
 Elias


----------



## shigzeo

Quote:


  Originally Posted by *lextek* /img/forum/go_quote.gif 
_I just got an email about this. Looks pretty, interesting. Lately I'e been consider a computer as a source._

 

how are you team old skool if you use computer as source? hmmm, i smell as covered up new skooler in old skool clothes. have fun if you get one though... you could just get a computer with optical out and not have to worry about the usb input; frees up your ports as well...


----------



## audioengr

Quote:


  Originally Posted by *Jon L* /img/forum/go_quote.gif 
_THAT IMO is huge news. LM4562 is capable of 45 mA current output and is National's newest breed of op-amps vs. NE5532, which is an ancient standard in comparison. 

 USB or not, it seems I need to do some comparisons between DAC-1 and DAC-1 USB 
	

	
	
		
		

		
			



_

 

To be completely fair, I should point out that the DAC-1 USB uses the LM4562 only in the 3 output drive locations. I know this because I just modded one. The other op-amps are still NE5532's. However, these are the newer TI version of 5532 which is used in all new DAC-1's and much better IMO than the older Philips version. Benchmark evidently made this change out of necessity, since the Philips version of NE5532 was obsolete. Turned out to be a very good thing though. The die layout or package frame of this part must be much improved because power delivery to the die is definitely better. They claimed no change in sound at the time, but the improvement was obvious to me, in midbass and overall "weight". This was the primary complaint of the older DAC-1's, they sounded "thin". Unfortunately, many audiophiles derived an opinion of the DAC-1 based on this early experience and never went back and revisited the newer DAC-1. It is definitely worth revisiting IMO.

 Steve N.


----------



## granodemostasa

Quote:


  Originally Posted by *audioengr* /img/forum/go_quote.gif 
_To be completely fair, I should point out that the DAC-1 USB uses the LM4562 only in the 3 output drive locations. I know this because I just modded one. The other op-amps are still NE5532's. However, these are the newer TI version of 5532 which is used in all new DAC-1's and much better IMO than the older Philips version. Benchmark evidently made this change out of necessity, since the Philips version of NE5532 was obsolete. Turned out to be a very good thing though. The die layout or package frame of this part must be much improved because power delivery to the die is definitely better. They claimed no change in sound at the time, but the improvement was obvious to me, in midbass and overall "weight". This was the primary complaint of the older DAC-1's, they sounded "thin". Unfortunately, many audiophiles derived an opinion of the DAC-1 based on this early experience and never went back and revisited the newer DAC-1. It is definitely worth revisiting IMO.

 Steve N._

 

Good thing you'll be at the national meet, I really look towards trying this dac out. Will i be able to try it out on my own rig while it's there... or are you guys going to have it on another rig?


----------



## little-endian

Hi EliasGwinn,

 sorry that it took some time for me to reply.

 Many thanks for your detailed answers. It is great to have you here!

 However I have additional questions about some of them (I'll use the numbering you have introduced):

 1. If I understand this correctly, besides the light L/R issue below 10 degrees I won't have any disadvantages in regard to THD, dynamic range, etc., because the volume pot and the selectable pad will operate exactly the same way then it comes to these values, am I right? Thus I wouldn't be at a complete loss with my "old" DAC1. 

 5. According to the distributor in Germany, Analog Audio, there are two different versions (leaving out the new USB model), one with the removable (and also losable) S/PDIF "dust cap" and one with some other cap design. The last one should support sample rates up to 192 kHz, the older one just *sometimes*. So, logically thought it isn't impossible even with the oldest models. Or do I miss something? Which type of S/PDIF receiver is used now in opposite to the older ones? Is it possible to determine the date of manufacturing on the basis of the serial number?

 6. If you refer to devices manufactured after 03/04 in general I suppose that the improvement isn't limited to the USB-version only.

 7. Yep, this document seems to cover quite exactly I was asking for. Thank you for this link. But intersample peaks don't necessarily have to cause distortion! In opposite to simple 0dBFS clipping which should cause distortion on all DACs, it depends on the converter's design what happens when intersample peaks occur. According to some forum discussions, given a DAC (mostly used in early CD-players) which doesn't use oversampling and assuming it has enough headroom in its analog output stages, the signal would be reproduced flawlessly (+3dBFS for example or even more). When oversampling is being used, it is claimed that things get even more complicated because overflows can occur not only within the analog stages but even before them, because the sample word length is limited also of course.

 Since the DAC1 changes the sample rate before converting like most other newer devices do, it would be very interesting how this was addressed at Benchmark. I mean, regardless to any AES papers who should know this better than you engineers? 

 Besides that maybe you could name the internal sample rate which is used by the DAC1. Because of the analog bandwidth of about 50 kHz and the famous nyquist theorem, it is guessed that a sample rate something around 100-110 kHz has to be used. But which value is it exactly (just for interest).

 Thanks again so far. Keep up the great support! 

 little-endian


----------



## little-endian

Quote:


  Originally Posted by *schaqfu* /img/forum/go_quote.gif 
_[guilty follow up #2] little-endian... I normally completely agree with you about choosing open formats that won't lock you in to any single vendor. But in the case of just about all the lossless audio codecs, I think it's irrelevant. It's irrelevant because each of them, even mighty Apple's and Microsoft's, have publicly available decoders. It doesn't even matter if the encoders are closed, like with Apple. When talking about lossless files, if at any point in the future Microsoft decides to charge you a royalty for using WMA lossless, or Apple revokes your right to use ALAC, you can avoid lock-in by simply transcoding your files to another lossless format of your choice. That's a clear escape hatch that is 100% effective at preventing lock-in, and lock-in is the enemy as we both agree. It's a luxury that's peculiar to lossless audio (and other lossless data file formats) because they are true clones even when transcoded._

 

Hi schaqfu,

 yes, closed stuff isn't that problem when it comes to lossless codecs, because of the reasons you already mentioned. At least as long as you are able to decode them, the OS still exists, etc. But this issue is a general and not specific to audio applications. From what I've heard, WMA lossless isn't even one of the most efficient. So why should I fiddle around with average algorithms which are closed into the bargain? Of course, this depends on the personal taste - I don't need colourful players, cover downloads and all these gimmicks so I stay with foobar.

 But hey, since it's lossless no one can be blaimed for bad sound quality (at most for cpu usage).


----------



## EliasGwinn

Little-endian,

 I'm not sure I understand your questions, but I'll try to clear this up...

 It is recommended to use the pad's to optimize the volume pot for your particular listening level. The pad's on the DAC1 will not cause an increase in THD+N, with one caveat: if the next component in the chain (preamp, amp, ...) has unusually low input impedance, or high input capacitance or inductance...or you are driving long cables (see table on page 8 of DAC1 manual: http://www.benchmarkmedia.com/dac1/DAC1-Manual.pdf ).

 The DAC1 USB's new output drivers eliminates that caveat. It has the ability to drive those difficult loads without any loss in THD+N. 

 As for improvements limited to the DAC1 USB (vs. the DAC1 standard), they are:

 - Selectable gain range for headphone amp
 -- Lets you select the optimal range for your specific headphones so that the volume knob can be utilized more optimally

 - Main output mutes upon headphone insertion (defeatable)
 -- The analog outputs on the rear of the DAC1 USB will be muted when you insert the headphone plug if this feature is enabled.

 - High-Current output drivers
 -- The XLR and RCA outputs can now drive longer cables, low-impedance loads, high-capacitance loads, and/or high-inductance loads without any loss in THD+N performance

 - Advanced USB Audio for true native 96/24 bit-transparent playback
 -- No drivers or configuration necessary...plug it in and immediately get bit-transparency at rates up to and including 96/24

 The type of optical jack has nothing to do with its ability to handle 192 kHz. The 'dust cap' optical jack and the 'hinged-door' optical jack perform the same. The only difference is the way they are mechanically closed. All of the DAC1's that are available from Analog Audio can handle 192 kHz optical. This is true of DAC1 (standard) and DAC1 USB's.

 As for intersample peaks, I'll have to get back to you on that.

 Thanks,
 Elias


----------



## Jetlag

Quote:


  Originally Posted by *EliasGwinn* /img/forum/go_quote.gif 
_All of the DAC1's ... can handle 192 kHz optical. This is true of DAC1 (standard) and DAC1 USB's_

 

This is probably a stupid question, but, the DAC 1 USB does 192/24 via the three other inputs (SPDIF/AES, AES, Toslink) and is only limited to 96/24 via USB, correct? This limit is due to the USB 1.1 specification, also correct?

 Thanks


----------



## audioengr

Quote:


  Originally Posted by *granodemostasa* /img/forum/go_quote.gif 
_Good thing you'll be at the national meet, I really look towards trying this dac out. Will i be able to try it out on my own rig while it's there... or are you guys going to have it on another rig?_

 

I'll be happy to use your amp and headphones with it for your audition.

 Steve N.


----------



## EliasGwinn

Sorry, please ignore this post. It is an errant repeat...


----------



## EliasGwinn

Sorry, please ignore this post. It is an errant repeat...


----------



## EliasGwinn

Sorry, please ignore this post. It is an errant repeat...


----------



## Lord Chaos

Elias, I've been using my DAC1-USB for a while now. It's great. I must be using really efficient headphones, though, because with the unit in its stock state and Itunes volume turned to max as recommended, I can only have the volume control to about the 4th detent, 3d detent with louder CDs. This is with Shure E500 'phones. When I use the DAC1 as a pre-amp for the stereo, I get the normal position of around 11 o'clock. Which makes me wonder what I'm doing to the E500s when I forget to unplug them...


----------



## Jetlag

Elias's computer must be in some sort of loop! A quadruple so far.


----------



## lowmagnet

Quote:


  Originally Posted by *Jetlag* /img/forum/go_quote.gif 
_Elias's computer must be in some sort of loop! A quadruple so far. 
	

	
	
		
		

		
		
	


	


_

 

The site's been rather slow lately.


----------



## gregeas

Elias,

 Well, this thread got me -- just ordered the USB DAC1 from Benchmark.

 Two questions. Can you suggest settings for J River Media Center? This is my playback software of choice.

 Also, what is the max safe distance for a USB cable run?


----------



## lowmagnet

Quote:


  Originally Posted by *gregeas* /img/forum/go_quote.gif 
_Also, what is the max safe distance for a USB cable run?_

 

USB 1.1 spec says 5 meters.


----------



## schaqfu

I just got my DAC1 USB delivered today. Sounds great -- I'm burning it in as we speak.

 I'd just like to say how silly I think it is that the DAC1 USB is limited to the USB 1.1 spec. What person connects a $1300 device to a computer that is so old it doesn't have USB 2.0? And even assuming a pre-2003 computer, it's cheap and easy to add USB 2.0 to it, and it's backwards compatible anyway, so they could always use it at 1.1 speeds which supports 96/24 processing. Who in the world processes 192/24 audio on a PC without USB 2.0 capability? Unless there is a technical reason I'm not aware of, this was a silly design decision -- fortunately one of extremely few on this miraculous device! The explanation on Benchmark's website simply makes reference to the wide adoption of USB 1.1 computers. This is not compelling. The DAC is capable of 192/24 resolution, yet we're only hampered when using the one connection that is unique to this updated -- and more expensive -- device. Not good.

 I'm very curious to know why.


----------



## Jetlag

Quote:


  Originally Posted by *schaqfu* 
_...how silly I think it is that the DAC1 USB is limited to the USB 1.1 spec. 

 What person connects a $1300 device to a computer that is so old it doesn't have USB 2.0? 

 The DAC is capable of 192/24 resolution, yet we're only hampered when using the one connection that is unique to this updated -- and more expensive -- device.

 I'm very curious to know why?_

 

This occurred to me as well. The beauty of this DAC is the ability to transparently stream our lossless music files via USB sans custom driver, one of a kind. I also can't imagine _anyone_ who would buy this device for this specific purpose would have only USB 1.1. Certainly the folks who will use this in the studio will not. Heck, my PC is from 2000 (yeah I know, but I'm on a budget) and yet I put a USB 2.0 PCI card into it. In fact, if I were to purchase this DAC, it's primary use would be via USB. It would be great (OK, *WAY* beyond great) if we could utilize the full 24/192 capabilities via USB (hint, hint, 
	

	
	
		
		

		
		
	


	




 )

<dream sequence starting>
Ahhh! Lossless FLAC files streaming via USB 2.0 to my DAC 1 USB and then out to my new active studio monitors (else insert favorite headphones +/- headphone amp here) in full 24/192! Nirvana found?
<dream sequence ending>

 1. Curious as to whether the PCB and other circuitry would allow for easy upgrade to 2.0?

 2. If the 2.0 upgrade is possible, would it still require no custom drivers?

 Thanks! Hope you feel well rested after the dream sequence, now BACK TO WORK!


----------



## EliasGwinn

Sorry, please ignore this post. It is an errant repeat...


----------



## Dave_M

A couple of things worth mentioning...

 1) The DAC1 resamples to 110 KHz. Even though it accepts 192 KHz music, it still resamples it to 110 KHz.

 2) There are cheap USB audio ICs available at the moment that just happen to support 24/96. One of these is probably being used for USB input in the new DAC1. Using one of these ICs (integrated circuits) takes a lot of work and hassle out of designing the USB audio input. It is possible to make it support 24/192 over USB 2.0 but this will require a lot more effort and the learning curve is steeper.

 If there was a USB 2.0 audio chip available that supported 24/192, then benchmark would probably have used that one instead. But I'm guessing that the benchmark engineers don't know how to implement a USB 2.0 solution. Unless they are just trying to save money which is ok.




 Information about the sample rate converter in the DAC1 is HERE.
  Quote:


 At the heart of the DAC1 are Analog Devices AD1853 DACs operating at a sampling rate of 110kHz. I talked to Allen H. Burdick, president of Benchmark Media, about the internal operation of the DAC1 and the decision to settle on the 110kHz sampling rate. According to Burdick, all incoming signals are brought to that sampling rate by using an Analog Devices AD1896 sample-rate converter. He says that operating at that rate offers a "20dB filter-performance improvement" with the AD1853 DAC chip over a higher sampling rate such as 192kHz. The tradeoff, he says, is reduced analog bandwidth -- down to 55kHz versus 96kHz. However, he feels that very little musical information resides that high anyway, and the improved filter performance achieved offsets that 41kHz loss in bandwidth. The DAC1 can accept digital input of signals up to 24 bits and 192kHz sampling rate.


----------



## EliasGwinn

Quote:


  Originally Posted by *Jetlag* /img/forum/go_quote.gif 
_This is probably a stupid question, but, the DAC 1 USB does 192/24 via the three other inputs (SPDIF/AES, AES, Toslink) and is only limited to 96/24 via USB, correct? This limit is due to the USB 1.1 specification, also correct?

 Thanks_

 

This is correct.

 All digital inputs can handle sample-rates up to 195 kHz, except the USB input, which can handle sample-rates up to 96 kHz. This is a limitation imposed by the USB 1.1 specification.

 Thanks,
 Elias


----------



## EliasGwinn

Quote:


  Originally Posted by *Lord Chaos* /img/forum/go_quote.gif 
_Elias, I've been using my DAC1-USB for a while now. It's great. I must be using really efficient headphones, though, because with the unit in its stock state and Itunes volume turned to max as recommended, I can only have the volume control to about the 4th detent, 3d detent with louder CDs. This is with Shure E500 'phones. When I use the DAC1 as a pre-amp for the stereo, I get the normal position of around 11 o'clock. Which makes me wonder what I'm doing to the E500s when I forget to unplug them..._

 

If you use the headphone input to the left (the one closest to the input switch and LED's; furthest from the volume knob), the rear outputs will be muted until you remove the headphone plug. This will remind you to remove the headphones when you're using the main outputs.

 There is also a 10 dB pad for the headphones so that you can turn the volume knob up further. It should have shipped with these pad's inserted, but you can open the DAC1 USB's chassis to check. If you want to do this, I can step through it with you.

 Thanks,
 Elias


----------



## EliasGwinn

Quote:


  Originally Posted by *gregeas* /img/forum/go_quote.gif 
_Elias,

 Well, this thread got me -- just ordered the USB DAC1 from Benchmark.

 Two questions. Can you suggest settings for J River Media Center? This is my playback software of choice.

 Also, what is the max safe distance for a USB cable run?_

 

Gregeas,

 I'm glad I could help you in your decision. Keep in touch...let me know what you think of it.

 As for J River, I have not tested it yet (or even used it at all). I will try to do it soon so that I can help you, but there are a few things that are generally a good idea with any media player.

 1. If there is a setting for sample-rate, be sure it matches the sample-rate of the audio you are playing.

 2. If there is a 'word-length' setting (aka 'bit-depth', aka 'size'), it should be set to 24-bit, regardless of the word-length of the audio you are playing

 3. Keep all audio processors off, where possible. This includes any EQ's, 'Sound Enhancers', Surround processors, Bass Boost, etc.

 4. Keep volume at 100%. Some players have better volume controls then others. As I haven't tested J River, I would recommend keeping the volume at 100%. In fact, the only player I've found to have a properly-designed volume control is Windows Media Player. Also, the Windows Volume Mixer (the system control, unrelated to a specific player) is a properly-designed volume control. Therefore, if you need PC control of volume, I recommend using this.

 As for USB cable length, Wikipedia says the max length is 5 meters.

 Thanks,
 Elias


----------



## EliasGwinn

Quote:


  Originally Posted by *schaqfu* /img/forum/go_quote.gif 
_I just got my DAC1 USB delivered today. Sounds great -- I'm burning it in as we speak.

 I'd just like to say how silly I think it is that the DAC1 USB is limited to the USB 1.1 spec. What person connects a $1300 device to a computer that is so old it doesn't have USB 2.0? And even assuming a pre-2003 computer, it's cheap and easy to add USB 2.0 to it, and it's backwards compatible anyway, so they could always use it at 1.1 speeds which supports 96/24 processing. Who in the world processes 192/24 audio on a PC without USB 2.0 capability? Unless there is a technical reason I'm not aware of, this was a silly design decision -- fortunately one of extremely few on this miraculous device! The explanation on Benchmark's website simply makes reference to the wide adoption of USB 1.1 computers. This is not compelling. The DAC is capable of 192/24 resolution, yet we're only hampered when using the one connection that is unique to this updated -- and more expensive -- device. Not good.

 I'm very curious to know why._

 

You're argument is valid. It is something we considered, and it was not an easy decision. We appreciate your input, however.

 Thanks,
 Elias


----------



## EliasGwinn

Quote:


  Originally Posted by *Dave_M* /img/forum/go_quote.gif 
_2) There are cheap USB audio ICs available at the moment that just happen to support 24/96. One of these is probably being used for USB input in the new DAC1. Using one of these ICs (integrated circuits) takes a lot of work and hassle out of designing the USB audio input. It is possible to make it support 24/192 over USB 2.0 but this will require a lot more effort and the learning curve is steeper.

 If there was a USB 2.0 audio chip available that supported 24/192, then benchmark would probably have used that one instead. But I'm guessing that the benchmark engineers don't know how to implement a USB 2.0 solution. Unless they are just trying to save money which is ok._

 

This is partly accurate. Let me clarify...

 The price of USB 2.0 chips was not a factor. However, there are no USB 2.0 chips available that support native USB audio. So, one of the trade-off's of using USB 2.0 is the necessity of a custom driver. As I've elaborated before, we tested several other (3rd party) USB audio devices that use custom drivers, and the results were not positive. In fact, throughout our testing, the only devices that achieved bit-transparency were native devices (devices which do not need custom drivers). 

 Also, custom drivers often cause conflicts with other drivers. Even if the driver is designed well, another driver may not 'get along' with it. Using the native drivers (which are inherently installed with the operating system) gives you the guarantee that, if a 3rd party driver ever worked on these computers, they will work with the DAC1 USB running.

 Another point you made is accurate, and partly affected our decision. That is, it would have taken longer to develop a USB 2.0 device. This wasn't an issue about saving money as much as getting the product to market as soon as possible. If the other factors weren't already weighing against a USB 2.0 solution, we may have decided it was worth waiting to release this product until a USB 2.0 solution was developed. 

 However, the decision was primarily based on the design goal of a native, bit-transparent USB audio solution.


----------



## Dave_M

Hi EliasGwinn, sorry if I have been repeating what others had said but I have not read through the whole thread.

 There is nothing wrong with using cheap off-the-shelf ICs to get the job done! After all, the cheaper the DAC is to produce, the more competitive you can be with pricing. And besides, an all in one integrated solution can often work better than a custom one with many ICs and discrete components.

 But I'm not sure I accept your argument of a native, bit-transparent USB audio solution. Since a "native solution" still needs drivers, it's just that those drivers were written by someone at Microsoft and come with the operating system. If a USB 2.0 one needs custom drivers, then it can be done just as well as "native solution", but just takes more time and money.


----------



## EliasGwinn

Quote:


  Originally Posted by *Dave_M* /img/forum/go_quote.gif 
_Hi EliasGwinn, sorry if I have been repeating what others had said but I have not read through the whole thread.

 There is nothing wrong with using cheap off-the-shelf ICs to get the job done! After all, the cheaper the DAC is to produce, the more competitive you can be with pricing. And besides, an all in one integrated solution can often work better than a custom one with many ICs and discrete components.

 But I'm not sure I accept your argument of a native, bit-transparent USB audio solution. Since a "native solution" still needs drivers, it's just that those drivers were written by someone at Microsoft and come with the operating system. If a USB 2.0 one needs custom drivers, then it can be done just as well as "native solution", but just takes more time and money._

 

You are correct. My use of the term 'native device' is defined as a device which uses native drivers. Theoretically, a custom driver may be designed as good as the native drivers. However, we have not seen any custom drivers currently available that are satisfactory. Meanwhile, the native drivers are very stable, very capable, and, most importantly, bit-transparent. This makes sence, as Microsoft's and/or Apple's programming team has a much broader and deeper familiarity of the system and environment in which they operate. 

 With that said, although all chips (both 1.1 and 2.0) are off-the-shelf chips, the firmware is not necessarily. There are some chips which have firmware included off-the-shelf; these are the chips used by most native USB audio devices. These off-the-shelf solutions are currently limited to 44/16 and 48/16 streams. Our solution uses a readily available USB chip (TI's TAS1020B) with custom-built firmware which enables it to stream bit-transparently at resolutions up to and including 96/24 using the native drivers. This firmware is what separates our USB solution from any others currently available. We are not aware of any USB audio devices which can stream 96/24 using native drivers.

 Thanks,
 Elias


----------



## Jetlag

Elias, does the DAC 1 USB upconvert the USB signal to 192KHz?


----------



## schaqfu

Quote:


  Originally Posted by *EliasGwinn* /img/forum/go_quote.gif 
_This is partly accurate. Let me clarify...

 The price of USB 2.0 chips was not a factor. However, there are no USB 2.0 chips available that support native USB audio. So, one of the trade-off's of using USB 2.0 is the necessity of a custom driver. As I've elaborated before, we tested several other (3rd party) USB audio devices that use custom drivers, and the results were not positive. In fact, throughout our testing, the only devices that achieved bit-transparency were native devices (devices which do not need custom drivers). 

 Also, custom drivers often cause conflicts with other drivers. Even if the driver is designed well, another driver may not 'get along' with it. Using the native drivers (which are inherently installed with the operating system) gives you the guarantee that, if a 3rd party driver ever worked on these computers, they will work with the DAC1 USB running.

 Another point you made is accurate, and partly affected our decision. That is, it would have taken longer to develop a USB 2.0 device. This wasn't an issue about saving money as much as getting the product to market as soon as possible. If the other factors weren't already weighing against a USB 2.0 solution, we may have decided it was worth waiting to release this product until a USB 2.0 solution was developed. 

 However, the decision was primarily based on the design goal of a native, bit-transparent USB audio solution._

 

This -- and Elias' following post as well -- is a totally legitimate explanation and rationalization of why they stuck with USB 1.1. This puts it into stark terms of why it was a market decision and not simply a lazy or errant one. I can't tell you how much I appreciate seeing this kind of response. It is very satisfying. I wish all manufacturers would do this. And, in fact, Elias, I highly recommend that you include something to this effect (perhaps more condensed) on the FAQ portion of your site. Right now the explanation posted sounds like a cop-out, seeming to say you chose USB 1.1 simply because it's widely available. The real answer is, this is the state of the art with respect to USB chips, while remaining within the (very well-received) design requirement of plug and play bit transparency.


----------



## Dave_M

Quote:


  Originally Posted by *schaqfu* /img/forum/go_quote.gif 
_This -- and Elias' following post as well -- is a totally legitimate explanation and rationalization of why they stuck with USB 1.1. This puts it into stark terms of why it was a market decision and not simply a lazy or errant one. I can't tell you how much I appreciate seeing this kind of response. It is very satisfying. I wish all manufacturers would do this. And, in fact, Elias, I highly recommend that you include something to this effect (perhaps more condensed) on the FAQ portion of your site. Right now the explanation posted sounds like a cop-out, seeming to say you chose USB 1.1 simply because it's widely available. The real answer is, this is the state of the art with respect to USB chips, while remaining within the (very well-received) design requirement of plug and play bit transparency._

 

I agree. Elias, you have retained all of your credibility when answering these questions. I think people can see through a lot of the BS you get from so many companies nowadays. Especially in this case anyone buying a DAC1 is not going to be your average schmuck.

 It usually the computer world that leads the way in technology and the hi-fi world that clings to old tech like valves. In this case it is benchmark who are waiting to the computer world to catch up. One day I'm sure there will be a DAC1 with 24/192 over USB 2.0, but until then I think they made the right decision to support 24/96 native. As having it work without custom drivers is a major advantage.




 Jetlag, I think all signals are re-sampled to 110 KHz. See post #265.


----------



## EliasGwinn

Quote:


  Originally Posted by *Jetlag* /img/forum/go_quote.gif 
_Elias, does the DAC 1 USB upconvert the USB signal to 192KHz?_

 

Jetlag,

 The short answer is... the USB signal is converted to 110 kHz, as Dave mentioned.

 The long answer is... (deep breath)

 The USB signal is first converted to I2S, maintaining its original sample-rate. I2S is a fundamental form of digital audio. When an AES/EBU or S/PDIF signal is streamed to the DAC1 via XLR/coax/optical, it is also first converted to I2S at its original sample-rate. The front panel switch chooses which of these I2S signals are sent to the next stage: the sample-rate converter (SRC) chip. This converts the sample-rate of the I2S signal to 110 kHz, regardless of the original sample rate.

 The reason it is converted to 110 kHz is because this is the sample-rate at which the D-to-A converter chip is most efficient. The trade-off of SRC far out-weighs the more significant distortion of the filter in the D-to-A chip operating at sample-rates other then its most efficient rate.

 Thanks,
 Elias


----------



## gregeas

Regarding 24/192 input, I don't see a practical reason why you would need it, unless you work in a mastering studio.

 For about five years I have done my best to enjoy hi-res audio (DVD-A and SACD), and just two months ago bought Arcam's new top-of-the-line univeral player. (I owned a variety of other universal players before that.) The problem is that after all these years there are still less than 25 hi-res disks that I care to own. And I don't see anything worthwhile on the horizon. (I'm not into classical.) Of course, there are some gems in my collection, and I hate to give them up, but how many times can I listen to Morph the Cat and Sea Change? Plus, the CDs of these albums sound damn good. 

 I did explore extracting the hi-res audio from DVD-A disks. This can be done but sounds very involved. So for now I'll live with 16/44.1 on my DAC1 and Transporter. 

 Does anyone else have a way to access hi-res audio files?


----------



## Riboge

Quote:


  Originally Posted by *EliasGwinn* /img/forum/go_quote.gif 
_... 

 With that said, although all chips (both 1.1 and 2.0) are off-the-shelf chips, the firmware is not necessarily. There are some chips which have firmware included off-the-shelf; these are the chips used by most native USB audio devices. These off-the-shelf solutions are currently limited to 44/16 and 48/16 streams. Our solution uses a readily available USB chip (TI's TAS1020B) with custom-built firmware which enables it to stream bit-transparently at resolutions up to and including 96/24 using the native drivers. This firmware is what separates our USB solution from any others currently available. We are not aware of any USB audio devices which can stream 96/24 using native drivers.

 Thanks,
 Elias_

 

E-mu and M-Audio both offer USB sound cards they claim use USB2 and can take in and put out 24/96. How do these reconcile with what you have said above? Do they not use native drivers?

 Also, my Stello 220 mkII has USB 1.1 input but is a 24/192 DAC. If I play recordings from the computer via USB that are 24/96, is that what reaches the DAC, and is it correct that the DAC can then upsample that to 192? How is this streaming or not streaming 24/96 via USB using a native driver?

 It is remarkable how hard it is to get consistent and reliable information like this which seems basic to understanding a given DAC and transfer mode.


----------



## EliasGwinn

Quote:


  Originally Posted by *Riboge* /img/forum/go_quote.gif 
_E-mu and M-Audio both offer USB sound cards they claim use USB2 and can take in and put out 24/96. How do these reconcile with what you have said above? Do they not use native drivers?

 Also, my Stello 220 mkII has USB 1.1 input but is a 24/192 DAC. If I play recordings from the computer via USB that are 24/96, is that what reaches the DAC, and is it correct that the DAC can then upsample that to 192? How is this streaming or not streaming 24/96 via USB using a native driver?

 It is remarkable how hard it is to get consistent and reliable information like this which seems basic to understanding a given DAC and transfer mode._

 

The E-mu and M-Audio both use custom drivers. They do not use native drivers.

 As for the Stello 220 mkII, I went to their website, but I couldn't find any spec's relating directly to the sample-rate capabilities of the USB input. You'll have to ask them about its USB capablities. The fact that it is a 24/192 DAC does not indicate anything about the USB sample-rate capabilities. For instance, the Benchmark DAC1 USB is also a 24/192 DAC, however the USB is limited to 96/24. If you play 24/192 audio on your computer with a USB audio device which cannot stream 24/192 via USB, the operating system will convert the sample-rate to one which the device is capable of streaming.

 As for consistent and reliable information, I couldn't agree with you more. I just went to the website for the Stello, and I couldn't find any worthwhile information. I hope our site is a little more straight-forward then that. 

 I also hope that if anyone feels that our site is not straight-forward or easy to navigate, that he/she will let me know so we can do something about it.

 Thanks,
 Elias


----------



## Riboge

Quote:


  Originally Posted by *EliasGwinn* /img/forum/go_quote.gif 
_The E-mu and M-Audio both use custom drivers. They do not use native drivers._

 

So does your comment about your estimation of custom drivers you have tried apply to these, ie, not good enough?


----------



## Graham Maynard

Hi Elias,

 Did Benchmark ever consider fitting an Alps motorised pot;- so that remote level control could be offered ?
 Could one be fitted by a competent technician ?

 Cheers ....... Graham.


----------



## EliasGwinn

Quote:


  Originally Posted by *Riboge* /img/forum/go_quote.gif 
_So does your comment about your estimation of custom drivers you have tried apply to these, ie, not good enough?_

 

Well, the question of "good enough" is relative to the application, and what you expect from the interface. If you expect bit-transparency, then they are not good enough. The units I tested had low-distortion, but not bit-transparent. Also, at 24/96, they 'choke' a lot. Specifically, the M-Audio Audiophile USB...when trying to record or playback at 24/96, it frequently has 'hiccups'. 24/96 is the extreme of its capabilities, so it can't quite handle it apparently.

 Thanks,
 Elias


----------



## EliasGwinn

Quote:


  Originally Posted by *Graham Maynard* /img/forum/go_quote.gif 
_Hi Elias,

 Did Benchmark ever consider fitting an Alps motorised pot;- so that remote level control could be offered ?
 Could one be fitted by a competent technician ?

 Cheers ....... Graham._

 

We have an Alps pot here which we are testing, but we have not drawn any conclusions on its performance yet. The form factor of the assembly is much too large for the DAC1, however, so it would not physically fit the DAC1.

 Thanks,
 Elias


----------



## Graham Maynard

Thanks Elias,

 It occured to me after posting that if necessary a DIYer could make up their own small "wall-wart" powered housing beside a racked DAC-1 if they needed a remotely controllable ALPS+buffer facility.

 Cheers ........ Graham.


----------



## choariwap

Quote:


  Originally Posted by *EliasGwinn* /img/forum/go_quote.gif 
_Well, the question of "good enough" is relative to the application, and what you expect from the interface. If you expect bit-transparency, then they are not good enough. The units I tested had low-distortion, but not bit-transparent. Also, at 24/96, they 'choke' a lot. Specifically, the M-Audio Audiophile USB...when trying to record or playback at 24/96, it frequently has 'hiccups'. 24/96 is the extreme of its capabilities, so it can't quite handle it apparently.

 Thanks,
 Elias_

 


 so maudio's drivers dont do bit perfect? even in ASIO? wow...

 i'm really itching to get a dac1 usb, but the closest i can get one is in hong kong. curses! soonest i can go is june-ish... sigh


----------



## TreAdidas

Quote:


  Originally Posted by *schaqfu* /img/forum/go_quote.gif 
_Since the balanced XLR outputs have been substantially improved in the DAC1 USB, and "can now drive... low-impedance loads... without any loss in THD+D performance," this raises the question: will balanced headphones, HD 650s in particular, now play even better directly out of the balanced outputs? At 300 ohms they certainly qualify as low-impedance loads. Will the relatively minor distortion reported by current users playing their HD 650s out of the DAC1 balanced outputs be reduced? Any other benefits we can anticipate?_

 


 I did not see an answer to this. I'm wildly interested in a response to this as well.


----------



## schaqfu

Quote:


  Originally Posted by *TreAdidas* /img/forum/go_quote.gif 
_I did not see an answer to this. I'm wildly interested in a response to this as well._

 

x2


----------



## audioengr

Quote:


  Originally Posted by *choariwap* /img/forum/go_quote.gif 
_so maudio's drivers dont do bit perfect? even in ASIO? wow...

 i'm really itching to get a dac1 usb, but the closest i can get one is in hong kong. curses! soonest i can go is june-ish... sigh_

 

They do, but it depends on the ASIO that you use. I actually prefer the sound of one that is probably not bit-perfect. Much more natural vocals.


----------



## EliasGwinn

Quote:


  Originally Posted by *TreAdidas* /img/forum/go_quote.gif 
_I did not see an answer to this. I'm wildly interested in a response to this as well._

 

The XLR outputs on the DAC1 USB are not designed for driving headphones because they have a 30 ohm resistor in series. For a headphone amp, ideally, you would want 0 ohms output impedance, which is how the HPA2 is designed (the HPA2 is Benchmark's signature headphone amplifier which is featured in the DAC1 and other products.)

 With that said, if one was to try driving balanced headphones from the XLR outputs, the DAC1 USB is much better suited for that then the DAC1. The DAC1 USB would be able to drive 300 ohm headphones to 28 dBu, but we have not tried nor measured the performance of such a setup. Therefore, we offer no claims or guarantees about the performance of this setup.

 Thanks,
 Elias


----------



## EliasGwinn

Please let me know if anyone tries the balanced headphones from the XLR outputs of the DAC1. I'm very curious as to how it performs.

 A question for you all: how are you building your balanced headphone cables? What type of wiring configuration are you using?

 Thanks,
 Elias


----------



## granodemostasa

Quote:


  Originally Posted by *EliasGwinn* /img/forum/go_quote.gif 
_Please let me know if anyone tries the balanced headphones from the XLR outputs of the DAC1. I'm very curious as to how it performs.

 A question for you all: how are you building your balanced headphone cables? What type of wiring configuration are you using?

 Thanks,
 Elias_

 

sadly, right now just about all headphone balenced cables are just standard cables with XLR termination. there has been talk of a cable that would have the ground separated but no one has built it yet.


----------



## kool bubba ice

Quote:


  Originally Posted by *EliasGwinn* /img/forum/go_quote.gif 
_Please let me know if anyone tries the balanced headphones from the XLR outputs of the DAC1. I'm very curious as to how it performs.

 A question for you all: how are you building your balanced headphone cables? What type of wiring configuration are you using?

 Thanks,
 Elias_

 

They sound great.. I'm using my Balanced SA5000/650's with my DAC1 from the XRL outputs.. My Sony 5000 are 70 ohm, & theres no distortion or hiss, even on the highest volume setting (Variable). I needed to Change the ATT to 0DB though.. Cause at -20DB the volume was set to low. My balanced cables were made by Alex.. apuresound.com. Not sure what type of wiring con fig was used..

 I think you would be impressed how headphones sound balanced through your DAC1x especially the 650s.. Awesome synergy.. In scale terms, If I rated the DAC1 headphone amp, I'd give it a 6.. Through balanced I'd rate it a 9...


----------



## kool bubba ice

I'm using the non USB version by the way..


----------



## TreAdidas

Quote:


  Originally Posted by *EliasGwinn* /img/forum/go_quote.gif 
_Please let me know if anyone tries the balanced headphones from the XLR outputs of the DAC1. I'm very curious as to how it performs.

 A question for you all: how are you building your balanced headphone cables? What type of wiring configuration are you using?

 Thanks,
 Elias_

 

I'm not really electronics savvy so, I was looking to buy some. I'm considering Zu Cable's Mobius V2, Moon Audio's Silver Dragon V2, and the Cardas Balanced headphone cable.

www.zucable.com (note the V2 is not yet on their website but it is being talked about here: http://www.head-fi.org/forums/showthread.php?t=233917 )

www.moon-audio.com

 I'm very interested in more information on this. Note the Sennheiser 650's nominal impedance is 300 OHM's


----------



## kool bubba ice

Gwinn,
 I have pics of my balanced cables in my sig, if that will give you any idea..


----------



## EliasGwinn

It seems (from the things I have read) that a balanced headphone configuration consists of driving the positive and negative terminal of the drivers with the positive and negative versions of the audio (respectively). In other words, the two terminals get the same active (with opposite polarity) signals.

 I've also heard of balanced headphones as being a positive signal lead and an isolated return path (sig ref). In other words, it would be similar to a regular headphone setup, but each headphone driver would have its own isolated return path (instead of sharing one).

 Thanks,
 Elias


----------



## choariwap

Quote:


  Originally Posted by *granodemostasa* /img/forum/go_quote.gif 
_sadly, right now just about all headphone balenced cables are just standard cables with XLR termination. there has been talk of a cable that would have the ground separated but no one has built it yet._

 

i think you're mistaken here.. balanced headphone cable terminate in TWO xlr plugs. the ground is separated in this case, actually there is no ground connection.

 the driver is hooked up to the + and - signal of one xlr, with the third xlr pin unused. same for the other driver/


----------



## choariwap

Quote:


  Originally Posted by *EliasGwinn* /img/forum/go_quote.gif 
_It seems (from the things I have read) that a balanced headphone configuration consists of driving the positive and negative terminal of the drivers with the positive and negative versions of the audio (respectively). In other words, the two terminals get the same active (with opposite polarity) signals._

 

thats correct.


----------



## EliasGwinn

I'd be interested in hearing the dual-active configuration, but I can't see how it would make a difference aside from having a signal with twice the magnitude. 

 Plus, they would have to be driven from a source with very low (near zero) output impedance. 

 The problem with using the DAC1 for such an application is the output drivers are designed to drive a line input device, hence the 60-ohm output impedance. With headphones, this output impedance will cause the damping factor to drop significantly, causing ringing, overshoot, and all sorts of distortion based on driver resonances. Let me elaborate on this further...

 Damping factor is a ratio of [load impedance vs. source impedance]. In the case of the HPA2, the output (source) impedance is less then 0.11 ohms. Therefore, the damping factor with 300-ohm headphones will be near 3000.

 The damping factor with a 30-ohm output impedance and 300-ohm headphones will be 10. The damping factor with 60-ohm outputs and 300-ohm headphones will be near 5.

 As you can see, we lose the tight control over the drivers which is essential for the reactive load which a speaker presents. 

 Another problem with a significant (>1-ohm) source impedance: the headphones have frequency-dependant resistance. Therefore, the voltage divider that is created between the source impedance and the load impedance is now a frequency-dependent voltage divider. This means the frequency response is going to be anything but flat.

 I would be hesitant to use this type of configuration. In fact, I don't recommend driving headphones with any source with an output impedance >1 ohm to drive headphones at all. 

 With that being said, it will not hurt the DAC1 to do this, so feel free to experiment with this setup without worrying about causing any damage. My philosophy is...if it sounds good to you...go for it! 

 However, don't evaluate the quality of the DAC1 based on driving headphones with the XLR outputs.

 Thanks,
 Elias


----------



## kool bubba ice

Quote:


  Originally Posted by *EliasGwinn* /img/forum/go_quote.gif 
_I'd be interested in hearing the dual-active configuration, but I can't see how it would make a difference aside from having a signal with twice the magnitude. 

 Plus, they would have to be driven from a source with very low (near zero) output impedance. 

 The problem with using the DAC1 for such an application is the output drivers are designed to drive a line input device, hence the 60-ohm output impedance. With headphones, this output impedance will cause the damping factor to drop significantly, causing ringing, overshoot, and all sorts of distortion based on driver resonances. Let me elaborate on this further...

 Damping factor is a ratio of [load impedance vs. source impedance]. In the case of the HPA2, the output (source) impedance is less then 0.11 ohms. Therefore, the damping factor with 300-ohm headphones will be near 3000.

 The damping factor with a 30-ohm output impedance and 300-ohm headphones will be 10. The damping factor with 60-ohm outputs and 300-ohm headphones will be near 5.

 As you can see, we lose the tight control over the drivers which is essential for the reactive load which a speaker presents. 

 Another problem with a significant (>1-ohm) source impedance: the headphones have frequency-dependant resistance. Therefore, the voltage divider that is created between the source impedance and the load impedance is now a frequency-dependent voltage divider. This means the frequency response is going to be anything but flat.

 I would be hesitant to use this type of configuration. In fact, I don't recommend driving headphones with any source with an output impedance >1 ohm to drive headphones at all. 

 With that being said, it will not hurt the DAC1 to do this, so feel free to experiment with this setup without worrying about causing any damage. My philosophy is...if it sounds good to you...go for it! 

 However, don't evaluate the quality of the DAC1 based on driving headphones with the XLR outputs.

 Thanks,
 Elias_

 

With all due respect, driving my headphones out of the DAC1 XLR output is a much more enjoyable listening experience then using the single ended HPA2. You should try it for yourself. I'm sure you will be surprised by the results. If going balanced with the DAC1 was a bad experience I doubt headfiers would be using the DAC1 as a balanced set up.. I'm sure amps that are made for balanced headphones will sound better cause they are tailor made for that purpose.. But the DAC1 does more then a decent job IMO..


----------



## audioengr

Quote:


  Originally Posted by *EliasGwinn* /img/forum/go_quote.gif 
_I'd be interested in hearing the dual-active configuration, but I can't see how it would make a difference aside from having a signal with twice the magnitude. 

 Plus, they would have to be driven from a source with very low (near zero) output impedance. 

 The problem with using the DAC1 for such an application is the output drivers are designed to drive a line input device, hence the 60-ohm output impedance. With headphones, this output impedance will cause the damping factor to drop significantly, causing ringing, overshoot, and all sorts of distortion based on driver resonances. Let me elaborate on this further...

 Damping factor is a ratio of [load impedance vs. source impedance]. In the case of the HPA2, the output (source) impedance is less then 0.11 ohms. Therefore, the damping factor with 300-ohm headphones will be near 3000.

 The damping factor with a 30-ohm output impedance and 300-ohm headphones will be 10. The damping factor with 60-ohm outputs and 300-ohm headphones will be near 5.

 As you can see, we lose the tight control over the drivers which is essential for the reactive load which a speaker presents. 

 Another problem with a significant (>1-ohm) source impedance: the headphones have frequency-dependant resistance. Therefore, the voltage divider that is created between the source impedance and the load impedance is now a frequency-dependent voltage divider. This means the frequency response is going to be anything but flat.

 I would be hesitant to use this type of configuration. In fact, I don't recommend driving headphones with any source with an output impedance >1 ohm to drive headphones at all. 

 With that being said, it will not hurt the DAC1 to do this, so feel free to experiment with this setup without worrying about causing any damage. My philosophy is...if it sounds good to you...go for it! 

 However, don't evaluate the quality of the DAC1 based on driving headphones with the XLR outputs.

 Thanks,
 Elias_

 

Elias - actually there are several advantages to balanced output drivers for headphones. Aside from the standard balanced advantage of common-mode noise rejection, the power supply di/dt delivery to the drivers will be balanced out and therefore power decoupling of these drivers will be easier. For instance, providing di/dt current for the BUF634 is particularly difficult and the bass usually suffers because of the lack of direct ground-return path for the output current. There are some modding tricks that can improve this a lot, but it would be much better to make it balanced IMO. the BUF634 is not a bad part, just difficult to feed.

 Steve N.

 Electrical Engineer/chief scientist
 Empirical Audio


----------



## schaqfu

I alternate between HD 650s and Shure E500 IEMs with my DAC1-USB, but the E500s are way too sensitive to be much use. Even with the -10Db jumpers on the headphone output enabled, it's way too loud past the 4th detent. I am wondering what is the preferred way to further attenuate volume output from the headphone output? Should I use an in-line hardware attenuator on the wire itself, or should I reduce the wave output volume in Windows? Any ideas? I know Elias has said if we're going to use any software volume reduction it should be the Windows volume slider because it's the best implemented without degrading the quality. But will I be degrading the quality more with that or with an in-line attenuator?

 Thanks for any ideas.


----------



## schaqfu

One more question... I note that there are two jumper pins that control the 10Db headphone output gain reduction, J8 and J9. I'm curious, does each one control a different headphone jack? And if so, is it possible to leave only one with the pin on, and one with the pin off, and have different gain settings on each of the headphone jacks? That would be really cool because it would allow me to plug in different resistance headphones into different jacks and get closer to the same volume with each.


----------



## JimP

X


----------



## lowmagnet

Quote:


  Originally Posted by *schaqfu* /img/forum/go_quote.gif 
_ I'm curious, does each one control a different headphone jack?_

 

Nope, I pulled just one last week. Got one -10dB channel and one -0dB channel.

 Having tried a pair of 16Ω ER-6i IEMs I feel your pain here. (literally, unfortunately)


----------



## lowmagnet

Quote:


  Originally Posted by *JimP* /img/forum/go_quote.gif 
_I think these are old tape transfers so some hiss in background,_

 

Yeah, they do sound nice and dynamic recordings, but the hiss was a big distraction. Also, changing system-wide sample rate for one or two source recordings is kinda a big pain in the butt. 
	

	
	
		
		

		
			





 I wonder if I can write an Applescript to change settings quickly. Maybe someone already wrote something.


----------



## audioengr

Quote:


  Originally Posted by *schaqfu* /img/forum/go_quote.gif 
_I alternate between HD 650s and Shure E500 IEMs with my DAC1-USB, but the E500s are way too sensitive to be much use. Even with the -10Db jumpers on the headphone output enabled, it's way too loud past the 4th detent. I am wondering what is the preferred way to further attenuate volume output from the headphone output? Should I use an in-line hardware attenuator on the wire itself, or should I reduce the wave output volume in Windows? Any ideas? I know Elias has said if we're going to use any software volume reduction it should be the Windows volume slider because it's the best implemented without degrading the quality. But will I be degrading the quality more with that or with an in-line attenuator?

 Thanks for any ideas._

 


 I tried digital volume control on the USB DAC-1 that I just modded. Not bad results at all. Other than this, the feedback resistors on the predriver op-amps must be changed to reduce gain.


----------



## schaqfu

Quote:


  Originally Posted by *lowmagnet* /img/forum/go_quote.gif 
_Yeah, they do sound nice and dynamic recordings, but the hiss was a big distraction. Also, changing system-wide sample rate for one or two source recordings is kinda a big pain in the butt. 
	

	
	
		
		

		
		
	


	




 I wonder if I can write an Applescript to change settings quickly. Maybe someone already wrote something._

 

Why do you have to change system-wide sample rate in order to play back different sample rate tracks? Doesn't the DAC1-USB take care of that automagically?


----------



## lowmagnet

Quote:


  Originally Posted by *schaqfu* /img/forum/go_quote.gif 
_Why do you have to change system-wide sample rate in order to play back different sample rate tracks? Doesn't the DAC1-USB take care of that automagically?_

 

It's my output sample rate I'm changing, generally kept at 44.1kHz, for obvious reasons. Switching isn't so much a pain as 99% unnecessary. I could set it at 96kHz but Elias mentioned issues with bit-perfect and Mac OS X when doing that. Leopard may fix this.


----------



## EliasGwinn

Quote:


  Originally Posted by *kool bubba ice* /img/forum/go_quote.gif 
_With all due respect, driving my headphones out of the DAC1 XLR output is a much more enjoyable listening experience then using the single ended HPA2. You should try it for yourself. I'm sure you will be surprised by the results. If going balanced with the DAC1 was a bad experience I doubt headfiers would be using the DAC1 as a balanced set up.. I'm sure amps that are made for balanced headphones will sound better cause they are tailor made for that purpose.. But the DAC1 does more then a decent job IMO.._

 

Kool bubba,

 I would like very much to hear it. I am very curious. And if you enjoy it, by all means, continue to do so!!

 Thanks,
 Elias


----------



## EliasGwinn

Quote:


  Originally Posted by *audioengr* /img/forum/go_quote.gif 
_Elias - actually there are several advantages to balanced output drivers for headphones. Aside from the standard balanced advantage of common-mode noise rejection, the power supply di/dt delivery to the drivers will be balanced out and therefore power decoupling of these drivers will be easier. For instance, providing di/dt current for the BUF634 is particularly difficult and the bass usually suffers because of the lack of direct ground-return path for the output current. There are some modding tricks that can improve this a lot, but it would be much better to make it balanced IMO. the BUF634 is not a bad part, just difficult to feed.

 Steve N.

 Electrical Engineer/chief scientist
 Empirical Audio_

 

Steve,

 I've read things like this on some websites. However, I can't say I agree with it. First of all, common-mode interference is not a problem with headphone drivers because there isn't a seperate ground path. In other words, if their was a seperate ground path from the headphones, then their may be noise issues due to the new reference. However, without this reference, it will not be manifested in the driver, mechanically or acoustically.

 As for the current rating and slew rate (di/dt) of the BUF634, we are not even approaching any limits that would require any additional driving devices. 

 Thanks,
 Elias


----------



## EliasGwinn

Quote:


  Originally Posted by *schaqfu* /img/forum/go_quote.gif 
_I alternate between HD 650s and Shure E500 IEMs with my DAC1-USB, but the E500s are way too sensitive to be much use. Even with the -10Db jumpers on the headphone output enabled, it's way too loud past the 4th detent. I am wondering what is the preferred way to further attenuate volume output from the headphone output? Should I use an in-line hardware attenuator on the wire itself, or should I reduce the wave output volume in Windows? Any ideas? I know Elias has said if we're going to use any software volume reduction it should be the Windows volume slider because it's the best implemented without degrading the quality. But will I be degrading the quality more with that or with an in-line attenuator?

 Thanks for any ideas._

 

Schaqfu,

 It is much, MUCH better to use the Windows Volume Mixer.

 The reason I stress this so much is because an in-line attenuator will raise the source impedance and ruin the frequency response of the headphones. It is very important to maintain a extremely low (<1 ohm) source impedance for the headphone driver. 

 Thanks!
 Elilas


----------



## EliasGwinn

Quote:


  Originally Posted by *schaqfu* /img/forum/go_quote.gif 
_One more question... I note that there are two jumper pins that control the 10Db headphone output gain reduction, J8 and J9. I'm curious, does each one control a different headphone jack? And if so, is it possible to leave only one with the pin on, and one with the pin off, and have different gain settings on each of the headphone jacks? That would be really cool because it would allow me to plug in different resistance headphones into different jacks and get closer to the same volume with each._

 

Schaqfu,

 The jumpers attenuate respective channels (right and left). The attenuation affects both headphone jacks.

 Thanks,
 Elias


----------



## EliasGwinn

Quote:


  Originally Posted by *schaqfu* /img/forum/go_quote.gif 
_Why do you have to change system-wide sample rate in order to play back different sample rate tracks? Doesn't the DAC1-USB take care of that automagically?_

 

Schaqfu,

 Mac's audio software does not dynamically change the sample-rate ever. This is not hardware or software specific (this even applies to Mac's built-in optical ports!!). It is very unfortunate, however, because many times, the user will not know that the OS is performing sample-rate conversion, and the audio suffers horribly because of it.

 It is very important to make sure the output sample-rate matches that of the audio when using Mac for audio. This is done in the "Audio MIDI Setup" page.

 Thanks,
 Elias


----------



## jsiau

Quote:


  Originally Posted by *audioengr* /img/forum/go_quote.gif 
_Aside from the standard balanced advantage of common-mode noise rejection, the power supply di/dt delivery to the drivers will be balanced out and therefore power decoupling of these drivers will be easier. For instance, providing di/dt current for the BUF634 is particularly difficult and the bass usually suffers because of the lack of direct ground-return path for the output current. There are some modding tricks that can improve this a lot, but it would be much better to make it balanced IMO. the BUF634 is not a bad part, just difficult to feed._

 


 Both of the above statements are incorrect. First common-mode rejection is not an issue. The headphone wiring usually provides a balanced connection to the 1/4" TRS plug. The TRS jack on the DAC1 is wired such that the common sleeve connection is wired to the analog ground reference at the opamp that drives the BUF634. There are no additional loads on this ground trace that would induce an interfering signal. In other words, the TRS jack is back-referenced to the analog ground at the BUF634 drivers. 

 Common-mode rejection is not an issue with either wiring scheme. Headphones have near perfect common-mode rejection as there is no return path for any common-mode signal. The drivers are isolated from the headphone frame (by many MegOhms). A voltage balanced drive offers no advantage, and the drivers have now way of "knowing" that they are being driven from a balanced drive. Headphones do not respond to common-mode voltage.

 There are two BUF634s in the DAC1. These are not difficult parts to drive when they are inside the feedback loop of an opamp. In this configuration, the BUF634/Opamp combination has excellent an PSRR (power supply rejection ratio) and is completely insensitive to audio-band noise on the +/- 18 volt power supply rails. In our lab tests we are able to inject a 3 volt AC waveform on top of either voltage rail without measuring or hearing a change in SNR. The BUF634's in the DAC1 have more than sufficient bypassing and do not suffer from any loss of low-frequency response even when driving 30-Ohm loads at the maximum rated output level of the HPA2.

 By the way, the BUF634 should never be used in an open-loop configuration. We have seen headphone amplifiers that use BUF634 buffers open-loop and they perform very poorly due to high distortion and poor PSRR.

 I can assure you that THD is higher when driving headphones from the XLR outputs of the DAC1. We have done this measurement. The difference is dramatic. Also, you will encounter some change in frequency response due to the fact that no headphone has perfectly constant input impedance over the audio band. There is also no guarantee that the left and right drivers will match each other. Low impedance drive solves these problems.

 Driving headphones from the DAC1 XLR outputs may create a warmer sound with some frequency-response and phase-response contouring. These changes may or may not be pleasing, but will definitely be less true to the original. At the end of the day, the increased THD may also add listening fatigue.

 Also, please remember that the coils in headphone drivers cannot tell if they are being driven from a single-ended amp or a balanced amp unless there is an additional leakage path back to the source. The voltage and current delivered to the drivers is identical in both cases, and the diaphragm movements will also be identical. Draw the circuit and trace the current paths if you have any doubts.


----------



## Jetlag

Excellent post John, thanks.


----------



## kool bubba ice

Quote:


  Originally Posted by *jsiau* /img/forum/go_quote.gif 
_Both of the above statements are incorrect. First common-mode rejection is not an issue. The headphone wiring usually provides a balanced connection to the 1/4" TRS plug. The TRS jack on the DAC1 is wired such that the common sleeve connection is wired to the analog ground reference at the opamp that drives the BUF634. There are no additional loads on this ground trace that would induce an interfering signal. In other words, the TRS jack is back-referenced to the analog ground at the BUF634 drivers. 

 Common-mode rejection is not an issue with either wiring scheme. Headphones have near perfect common-mode rejection as there is no return path for any common-mode signal. The drivers are isolated from the headphone frame (by many MegOhms). A voltage balanced drive offers no advantage, and the drivers have now way of "knowing" that they are being driven from a balanced drive. Headphones do not respond to common-mode voltage.

 There are two BUF634s in the DAC1. These are not difficult parts to drive when they are inside the feedback loop of an opamp. In this configuration, the BUF634/Opamp combination has excellent an PSRR (power supply rejection ratio) and is completely insensitive to audio-band noise on the +/- 18 volt power supply rails. In our lab tests we are able to inject a 3 volt AC waveform on top of either voltage rail without measuring or hearing a change in SNR. The BUF634's in the DAC1 have more than sufficient bypassing and do not suffer from any loss of low-frequency response even when driving 30-Ohm loads at the maximum rated output level of the HPA2.

 By the way, the BUF634 should never be used in an open-loop configuration. We have seen headphone amplifiers that use BUF634 buffers open-loop and they perform very poorly due to high distortion and poor PSRR.

*I can assure you that THD is higher when driving headphones from the XLR outputs of the DAC1*. We have done this measurement. The difference is dramatic. Also, you will encounter some change in frequency response due to the fact that no headphone has perfectly constant input impedance over the audio band. There is also no guarantee that the left and right drivers will match each other. Low impedance drive solves these problems.

 Driving headphones from the DAC1 XLR outputs may create a warmer sound with some frequency-response and phase-response contouring. These changes may or may not be pleasing, but will definitely be less true to the original. At the end of the day, the increased THD may also add listening fatigue.

 Also, please remember that the coils in headphone drivers cannot tell if they are being driven from a single-ended amp or a balanced amp unless there is an additional leakage path back to the source. The voltage and current delivered to the drivers is identical in both cases, and the diaphragm movements will also be identical. Draw the circuit and trace the current paths if you have any doubts._

 

That is true.. But the THD is still not a problem driving headphones balanced through the DAC1 XRL output.. I use both my SA5000 (70 ohm) & Sen 650 (300 ohm) balanced with the DAC1.. Music is much more enjoyable Via XRL outputs then through the DAC1's HP amp.. You should really try it for yourself..


----------



## EliasGwinn

Quote:


  Originally Posted by *kool bubba ice* /img/forum/go_quote.gif 
_That is true.. But the THD is still not a problem driving headphones balanced through the DAC1 XRL output.. I use both my SA5000 (70 ohm) & Sen 650 (300 ohm) balanced with the DAC1.. Music is much more enjoyable Via XRL outputs then through the DAC1's HP amp.. You should really try it for yourself.._

 

Kool bubba,

 Although we don't recommend using the DAC1's XLR outputs to drive your headphones, please feel free to continue doing so if you enjoy it. As I mentioned, doing so will not hurt anything, so there is no reason not to if that is your preference.

 I'm glad you enjoy the DAC1. Please keep up the feedback...its great to hear about user experiences!!

 Thanks,
 Elias


----------



## A.Thorsen

Please forgive me if this has already been discussed several times, but this seemed to sort of go with what's been discussed over the last page or so and I'd like to ask a question:

 This review says: 
http://www.positive-feedback.com/Iss...hmark_dac1.htm

  Quote:


 After reading the well written manual, it was obvious that the jumpers added to the output stage when using the balanced outputs were doing more than reducing the volume 20dBs. They were changing the textural cues and the bass response of the entire unit.

 I popped the top, found the very small jumpers [what a pain], *and moved them to the 0dB position on all four circuit board points as instructed. Dear Benchmark, the jumpers should be delivered in the zero position with the option to move them to the 10, 20, or 30 dB roll off positions if needed.*

 You have to assume minimal smarts on the part of the person who buys a separate DAC these days.

 Now we're cooking! The sound has transformed into a powerhouse of digital energy with charm and musicality galore. Using the Alesis or my Pioneer LD/CD/DVD Player, the audio approaches my super references. Moreover, the bass is room shattering with powerful, intense, focused bass going down to the center of the earth. It's now every bit as good as my venerable Theta 5a and equal to my ModWright Sony 999. Now you must make this change on the Benchmark DAC1 if you want this kind of solidity. Also, you will get added focus in the mids and highs with even more sweetness and smoothness on delicate passages.* Overall, the DAC1 in 0dB mode is 10% improved in the mids and highs and 25% improved on the bottom frequencies.* 
 

^^ Does the DAC1 now come with any of these suggested adjustments, like defaulting the jumpers at 0dB across the boards?

 I'm not sure 100 percent trust myself to do it if I feel the need. 
	

	
	
		
		

		
			





 I'd appreciate everyone's take on that.


----------



## kool bubba ice

Quote:


  Originally Posted by *EliasGwinn* /img/forum/go_quote.gif 
_Kool bubba,

 Although we don't recommend using the DAC1's XLR outputs to drive your headphones, please feel free to continue doing so if you enjoy it. As I mentioned, doing so will not hurt anything, so there is no reason not to if that is your preference.

 I'm glad you enjoy the DAC1. Please keep up the feedback...its great to hear about user experiences!!

 Thanks,
 Elias_

 

Thank you. It's the main reason I bought it.. Got great feedback from others.. & is a great bang for the buck balanced option..Even though it wasn't made for balanced headphones..


----------



## lowmagnet

Quote:


  Originally Posted by *A.Thorsen* /img/forum/go_quote.gif 
_^^ Does the DAC1 now come with any of these suggested adjustments, like defaulting the jumpers at 0dB across the boards?_

 

It defaults to -20dB.

  Quote:


  Originally Posted by *DAC1 USB Manual* 
_10, 20 and 30 dB pads are provided for interfacing directly to monitors and amplifiers that have too much input sensitivity to handle high-level (+29 dBu) signal levels._


----------



## schaqfu

Quote:


  Originally Posted by *EliasGwinn* /img/forum/go_quote.gif 
_Schaqfu,

 It is much, MUCH better to use the Windows Volume Mixer.

 The reason I stress this so much is because an in-line attenuator will raise the source impedance and ruin the frequency response of the headphones. It is very important to maintain a extremely low (<1 ohm) source impedance for the headphone driver. 

 Thanks!
 Elilas_

 

Perfect! Thanks for the superb feedback. As always.


----------



## schaqfu

Quote:


  Originally Posted by *EliasGwinn* /img/forum/go_quote.gif 
_Schaqfu,

 Mac's audio software does not dynamically change the sample-rate ever. This is not hardware or software specific (this even applies to Mac's built-in optical ports!!). It is very unfortunate, however, because many times, the user will not know that the OS is performing sample-rate conversion, and the audio suffers horribly because of it.

 It is very important to make sure the output sample-rate matches that of the audio when using Mac for audio. This is done in the "Audio MIDI Setup" page.

 Thanks,
 Elias_

 

But in Windows this is not an issue, right? I can just set iTunes to 24-bit output (via Quicktime, as you've explained) forever and forget about it irrespective of what resolution tracks I'm playing? Thanks.


----------



## schaqfu

Quote:


  Originally Posted by *schaqfu* /img/forum/go_quote.gif 
_But in Windows this is not an issue, right? I can just set iTunes to 24-bit output (via Quicktime, as you've explained) forever and forget about it irrespective of what resolution tracks I'm playing? Thanks._

 

While on that subject, (hopefully this is not off topic) does anyone know how to import 96/24 WMA Lossless files into iTunes and retain the full resolution? I know iTunes is capable of high output resolution, but when I drag in my hi res WMA tracks it converts them into 48/16 ALAC tracks. How do I get higher resolution ALAC tracks to result?

 Thanks!


----------



## audioengr

Quote:


  Originally Posted by *EliasGwinn* /img/forum/go_quote.gif 
_Steve,

 I've read things like this on some websites. However, I can't say I agree with it. First of all, common-mode interference is not a problem with headphone drivers because there isn't a seperate ground path. In other words, if their was a seperate ground path from the headphones, then their may be noise issues due to the new reference. However, without this reference, it will not be manifested in the driver, mechanically or acoustically.

 As for the current rating and slew rate (di/dt) of the BUF634, we are not even approaching any limits that would require any additional driving devices. 

 Thanks,
 Elias_

 


 Maybe we talk in person some time......

 Steve N.


----------



## audioengr

Quote:


  Originally Posted by *jsiau* /img/forum/go_quote.gif 
_Both of the above statements are incorrect. First common-mode rejection is not an issue. The headphone wiring usually provides a balanced connection to the 1/4" TRS plug. The TRS jack on the DAC1 is wired such that the common sleeve connection is wired to the analog ground reference at the opamp that drives the BUF634. There are no additional loads on this ground trace that would induce an interfering signal. In other words, the TRS jack is back-referenced to the analog ground at the BUF634 drivers. 

 Common-mode rejection is not an issue with either wiring scheme. Headphones have near perfect common-mode rejection as there is no return path for any common-mode signal. The drivers are isolated from the headphone frame (by many MegOhms). A voltage balanced drive offers no advantage, and the drivers have now way of "knowing" that they are being driven from a balanced drive. Headphones do not respond to common-mode voltage.

 There are two BUF634s in the DAC1. These are not difficult parts to drive when they are inside the feedback loop of an opamp. In this configuration, the BUF634/Opamp combination has excellent an PSRR (power supply rejection ratio) and is completely insensitive to audio-band noise on the +/- 18 volt power supply rails. In our lab tests we are able to inject a 3 volt AC waveform on top of either voltage rail without measuring or hearing a change in SNR. The BUF634's in the DAC1 have more than sufficient bypassing and do not suffer from any loss of low-frequency response even when driving 30-Ohm loads at the maximum rated output level of the HPA2.

 By the way, the BUF634 should never be used in an open-loop configuration. We have seen headphone amplifiers that use BUF634 buffers open-loop and they perform very poorly due to high distortion and poor PSRR.

 I can assure you that THD is higher when driving headphones from the XLR outputs of the DAC1. We have done this measurement. The difference is dramatic. Also, you will encounter some change in frequency response due to the fact that no headphone has perfectly constant input impedance over the audio band. There is also no guarantee that the left and right drivers will match each other. Low impedance drive solves these problems.

 Driving headphones from the DAC1 XLR outputs may create a warmer sound with some frequency-response and phase-response contouring. These changes may or may not be pleasing, but will definitely be less true to the original. At the end of the day, the increased THD may also add listening fatigue.

 Also, please remember that the coils in headphone drivers cannot tell if they are being driven from a single-ended amp or a balanced amp unless there is an additional leakage path back to the source. The voltage and current delivered to the drivers is identical in both cases, and the diaphragm movements will also be identical. Draw the circuit and trace the current paths if you have any doubts._

 


 Perhaps you will be at the Head-Fest this next weekend? We can discuss it more there and you can hear my modded DAC-1 headphone amp for yourself.

 I've been designing electronics for more than 30 years and I'm still learning new things. Perhaps we can learn from each other.

 Steve N.


----------



## EliasGwinn

Quote:


  Originally Posted by *schaqfu* /img/forum/go_quote.gif 
_But in Windows this is not an issue, right? I can just set iTunes to 24-bit output (via Quicktime, as you've explained) forever and forget about it irrespective of what resolution tracks I'm playing? Thanks._

 

This is part-true and part-not. Unlike OSX's Core Audio, Windows' Kmixer will dynamically change its output sample-rate with the stream it is being fed.

 However!!!!!!!.... iTunes will NOT change the sample rate dynamically, even when running in Windows. iTunes will stream everything at the sample-rate which is set in QuickTime. This is an issue with iTunes...most other media players for Windows will dynamically change the output sample rate.

 So, as long as the sample-rate set in QuickTime corresponds to that of the audio being played, iTunes will play bit-transparently. 

 Thanks,
 Elias


----------



## EliasGwinn

Quote:


  Originally Posted by *audioengr* /img/forum/go_quote.gif 
_Maybe we talk in person some time......

 Steve N._

 

Steve,

 I would enjoy meeting you sometime. I will not be at the Head-Fi meet this year, but I'm sure we'll meet at one of these shows sooner or later.

 Thanks,
 Elias


----------



## EliasGwinn

Ok Head-Fi-ers, I'm gonna give you guys a special sneak preview of something I've been working on for a while.

 This hasn't been announced yet, but I'll let you all know about it. I'd love to have some feedback on it if you have any.

http://www.BenchmarkMedia.com/wiki

 Thanks,
 Elias


----------



## A.Thorsen

Quote:


  Originally Posted by *EliasGwinn* /img/forum/go_quote.gif 
_Ok Head-Fi-ers, I'm gonna give you guys a special sneak preview of something I've been working on for a while.

 This hasn't been announced yet, but I'll let you all know about it. I'd love to have some feedback on it if you have any.

http://www.BenchmarkMedia.com/wiki

 Thanks,
 Elias_

 

Very informative! Definitely a great concept. I hope you keep going forward with that project. 
	

	
	
		
		

		
		
	


	






  Quote:


  Originally Posted by *A.Thorsen* 
_^^ Does the DAC1 now come with any of these suggested adjustments, like defaulting the jumpers at 0dB across the boards?_

 

^^

  Quote:


  Originally Posted by *lowmagnet* /img/forum/go_quote.gif 
_It defaults to -20dB._

 

^^

  Quote:


  Originally Posted by *DAC1 USB Manual* 
_10, 20 and 30 dB pads are provided for interfacing directly to monitors and amplifiers that have too much input sensitivity to handle high-level (+29 dBu) signal levels._

 


 Alright. If that's the case: That reviewer really seemed pretty enthusiastic about defaulting everything to 0dB for optimal results. Would it be better and/or should I make that request to have a DAC1 adjusted accordingly should I place an order for one?

 I'm just looking for some clarification/elaboration on that whole issue. Thanks!


----------



## EliasGwinn

Ok Head-Fi-ers, I'm gonna give you guys a special sneak preview of something I've been working on for a while.

 This hasn't been announced yet, but I'll let you all know about it. I'd love to have some feedback on it if you have any.

http://www.BenchmarkMedia.com/wiki

 Thanks,
 Elias


----------



## lowmagnet

Quote:


  Originally Posted by *A.Thorsen* /img/forum/go_quote.gif 
_Alright. If that's the case: That reviewer really seemed pretty enthusiastic about defaulting everything to 0dB for optimal results. Would it be better and/or should I make that request to have a DAC1 adjusted accordingly should I place an order for one?_

 

Don't let the article worry you. You can easily pull the jumpers and set them properly. The case has somewhat tricky (read: shallow) screws, so make sure you use a really flat-headed driver. If you have a pair of needle-nosed pliers or long forceps it's no problem. Please unplug before servicing, wouldn't want a fried head-fier.


----------



## fkclo

Quote:


  Originally Posted by *EliasGwinn* /img/forum/go_quote.gif 
_Ok Head-Fi-ers, I'm gonna give you guys a special sneak preview of something I've been working on for a while.

 This hasn't been announced yet, but I'll let you all know about it. I'd love to have some feedback on it if you have any.

http://www.BenchmarkMedia.com/wiki

 Thanks,
 Elias_

 


 Elias,

 This is simply GREAT! Not only is it informative, but also reflect how Benchmark has its customers in mind. I have been using OS X for years and do not know there is an Audio MIDI app. Thanks. 

 I am sure we fellow DAC1 USB owners will be grateful for this.

 Again, A BIG thank you.

 Francis
 Proud Owner of DAC1 USB from Hong Kong


----------



## A.Thorsen

Quote:


  Originally Posted by *lowmagnet* /img/forum/go_quote.gif 
_Don't let the article worry you. You can easily pull the jumpers and set them properly. The case has somewhat tricky (read: shallow) screws, so make sure you use a really flat-headed driver. If you have a pair of needle-nosed pliers or long forceps it's no problem. Please unplug before servicing, wouldn't want a fried head-fier._

 

^^ If I cross that bridge, I most certainly wouldn't have it plugged in.


----------



## JimP

x


----------



## smeggy

Quote:


  Originally Posted by *EliasGwinn* /img/forum/go_quote.gif 
_Ok Head-Fi-ers, I'm gonna give you guys a special sneak preview of something I've been working on for a while.

 This hasn't been announced yet, but I'll let you all know about it. I'd love to have some feedback on it if you have any.

http://www.BenchmarkMedia.com/wiki

 Thanks,
 Elias_

 

This is great! 
 You guys are doing a great service to the community by assisting in getting the best from digital systems regardless of make with this info and I thank you. Your in-depth discussions and explanations are really helping to lift the fog on the varied and confusing issues surrounding computer based output. I've learned a lot from this thread. This is the kind of thing that breeds brand loyalty.


----------



## kool bubba ice

Quote:


  Originally Posted by *A.Thorsen* /img/forum/go_quote.gif 
_Very informative! Definitely a great concept. I hope you keep going forward with that project. 
	

	
	
		
		

		
		
	


	








 ^^



 ^^




 Alright. If that's the case: That reviewer really seemed pretty enthusiastic about defaulting everything to 0dB for optimal results. Would it be better and/or should I make that request to have a DAC1 adjusted accordingly should I place an order for one?

 I'm just looking for some clarification/elaboration on that whole issue. Thanks! 
	

	
	
		
		

		
		
	


	


_

 

I'd let them do it..


----------



## A.Thorsen

Quote:


  Originally Posted by *kool bubba ice* /img/forum/go_quote.gif 
_I'd let them do it.._

 

So would I.


----------



## Mher6

Quote:


  Originally Posted by *EliasGwinn* /img/forum/go_quote.gif 
_Ok Head-Fi-ers, I'm gonna give you guys a special sneak preview of something I've been working on for a while.

 This hasn't been announced yet, but I'll let you all know about it. I'd love to have some feedback on it if you have any.

http://www.BenchmarkMedia.com/wiki

 Thanks,
 Elias_

 

In the guide it says to "This applies to digital volume controls in media players, Windows Volume Control, or any others." Is the Windows Volume Control you are refering to the volume control here? Start > Control panel > sounds and audio devices > device volume. Mine is set at 50%, should I change this to 100%?


----------



## sachu

Thanks for the info Elias.

 Regarding the setup for Foobar, do i need to disable the crossfeed plugin too in the DSP section.

 I kinda prefer the crossfeed for some albums, given the HD580s enormous soundstage.


----------



## Jon L

Quote:


  Originally Posted by *EliasGwinn* /img/forum/go_quote.gif 
_Ok Head-Fi-ers, I'm gonna give you guys a special sneak preview of something I've been working on for a while.

 This hasn't been announced yet, but I'll let you all know about it. I'd love to have some feedback on it if you have any.

http://www.BenchmarkMedia.com/wiki

 Thanks,
 Elias_

 

Interesting article. A couple of questions:

 1. Winamp is the only player where you specifically state it will play "bit-transparently." Does it mean foobar, iTunes, WMP will not play bit-perfectly? (even with ASIO plugins?)

 2. Do all the comments pertain to only when you use Benchmark DAC-1 USB via the USB input or all DAC's/soundcards w/ USB/spdif?

 3. The only player you mention as not having "high-distortion" digital volume control is WMP. Could you tell us what numbers are being measured and how, and especially at how much attenuation levels (-3 dB? -30 dB?). Were you able to subjectively correlate listening impressions of distortions caused by volume control by ear?

 TIA


----------



## lowmagnet

I guess some of you haven't used a wiki before. Create an account on the wiki and login. Go to the page that you want clarification on and hit the 'discussion' tab at the top, then the edit tab. Type in your question. Someone will eventually answer it there or on the main page.


----------



## EliasGwinn

Quote:


  Originally Posted by *A.Thorsen* /img/forum/go_quote.gif 
_Very informative! Definitely a great concept. I hope you keep going forward with that project. 
	

	
	
		
		

		
		
	


	




 Alright. If that's the case: That reviewer really seemed pretty enthusiastic about defaulting everything to 0dB for optimal results. Would it be better and/or should I make that request to have a DAC1 adjusted accordingly should I place an order for one?

 I'm just looking for some clarification/elaboration on that whole issue. Thanks! 
	

	
	
		
		

		
		
	


	


_

 

The output attenuators do not affect the performance of the output stage whatsoever (according to our testing and our ears!). I run my DAC1 in my recording studio at the -20 dB level because it allows me to run the volume pot into its most optimal range (>10 o'clock).

 Now, it should be said that certain equipment (of questionable design) being driven by the classic DAC1 may be affected by the increase in source impedance that the attenuators present. A well designed device (high input impedance, low input capacitance) will not have any performance comprise when using the attenuators. The attenuators would only affect the sound quality if the equipment that was being driven by the DAC1 had low input impedance or high input capacitance, or if the DAC1 was driving a long cable. 

 Unfortunately, some "audiophile" gear has poorly designed input stages with much too low impedance or large noise-draining capacitors which can affect frequency response drastically. In those cases, the source impedance should be as low as possible. I don't know what this reviewer was using as a pre and/or amplifier, so I can't comment on the validity of his experience. But if the device being driven is designed properly, the attenuators will have no affect on the audio whatsoever.

 The DAC1 USB, however, relieves this problem altogether with the new high-current output drivers. These new drivers will drive even the most difficult loads at any attenuator setting with out any performance comprimises.

 Thanks,
 Elias


----------



## EliasGwinn

Quote:


  Originally Posted by *Mher6* /img/forum/go_quote.gif 
_In the guide it says to "This applies to digital volume controls in media players, Windows Volume Control, or any others." Is the Windows Volume Control you are refering to the volume control here? Start > Control panel > sounds and audio devices > device volume. Mine is set at 50%, should I change this to 100%?_

 

This is a good question, and one that I get a lot. This is very important point:

 NOT ALL DIGITAL VOLUME CONTROLS ARE BUILT EQUALLY!!!

 Some (most) volume controls on media software induces A LOT of distortion (see the pictures on the iTunes page of the wiki). That is why I say, by default, assume a digital volume control is causing distortion.

 However, the Windows volume control mixer is actually built very well. It causes very, very little distortion, and the user should not be discouraged from using it.

 Personally, I leave it all the way up and use the DAC1's volume control pot. But, if you need computer-accessible volume control, the Windows volume mixer is the way to go (for Windows users). Also, if you are barely getting the DAC1 volume pot up before it is too loud, it is a good idea to use this to attenuate the signal before streaming to the DAC1 in order to allow you to crank the volume pot a bit more. Ideally, the volume pot should be set at or beyond 10 o'clock.

 Thanks,
 Elias


----------



## EliasGwinn

Quote:


  Originally Posted by *sachu* /img/forum/go_quote.gif 
_Thanks for the info Elias.

 Regarding the setup for Foobar, do i need to disable the crossfeed plugin too in the DSP section.

 I kinda prefer the crossfeed for some albums, given the HD580s enormous soundstage._

 

This brings up an interesting point.

 There are some settings which users prefer, and we wouldn't want to discourage anything which makes the listening experience more pleasurable (as long as its legal 
	

	
	
		
		

		
		
	


	




 !!)

 The recommendations on the Benchmark Wiki are those required for bit-transparent playback (or as close as can be possibly attained).

 Thanks,
 Elias


----------



## Lord Chaos

What should those of us using a Mac with a DAC1 USB do for volume control, Elias? The headphones I use with the DAC1 are efficient, and the control on the DAC1 is only up to the fourth detent most of the time. I tried an external attenuator but that caused sound problems, as the Wiki suggests.

 When I use the DAC1 as a pre-amp the volume is fine; I'm up around 10 o'clock or 11. How about putting a separate volume control on the unit for the headphone outputs? This would enable me to turn off the headphones when using speakers, too.


----------



## EliasGwinn

Quote:


  Originally Posted by *Jon L* /img/forum/go_quote.gif 
_Interesting article. A couple of questions:

 1. Winamp is the only player where you specifically state it will play "bit-transparently." Does it mean foobar, iTunes, WMP will not play bit-perfectly? (even with ASIO plugins?)
 TIA_

 

Thank you for pointing this out!! Foobar, iTunes, WMP will play bit-transparently. The settings I recommend are those necessary for achieving bit-transparency with these players.

  Quote:


  Originally Posted by *Jon L* /img/forum/go_quote.gif 
_2. Do all the comments pertain to only when you use Benchmark DAC-1 USB via the USB input or all DAC's/soundcards w/ USB/spdif?
 TIA_

 

No... This does not only apply to Benchmark's DAC1. However, not all soundcards and/or USB/spdif are capable of bit-transparent playback. 

  Quote:


  Originally Posted by *Jon L* /img/forum/go_quote.gif 
_3. The only player you mention as not having "high-distortion" digital volume control is WMP. Could you tell us what numbers are being measured and how, and especially at how much attenuation levels (-3 dB? -30 dB?). Were you able to subjectively correlate listening impressions of distortions caused by volume control by ear?

 TIA_

 

I can try to put some specific numbers together for you, but here is our testing method:

 We play a test-tone (10 kHz pure sine wave, -1 dBFS) via the media player in question. We set all settings necessary to achieve bit-transparency (sample rate, volume control, etc). We then monitor the FFT. When everything is set properly, the FFT looks like one tone at 10 kHz, with nothing but -135 dBFS noise floor around it (for 16-bit audio). 

 We then move the volume control, and monitor the FFT simultaneously. A well-built volume control will maintain the same FFT except the tone will be lower in amplitude.

 See this page of iTunes FFT measurements for FFT's of the effect of volume control, sample-rate conversion, word-length truncation, etc.

http://extra.benchmarkmedia.com/wiki...n_Measurements

 Thanks,
 Elias


----------



## EliasGwinn

Quote:


  Originally Posted by *Lord Chaos* /img/forum/go_quote.gif 
_What should those of us using a Mac with a DAC1 USB do for volume control, Elias? The headphones I use with the DAC1 are efficient, and the control on the DAC1 is only up to the fourth detent most of the time. I tried an external attenuator but that caused sound problems, as the Wiki suggests.

 When I use the DAC1 as a pre-amp the volume is fine; I'm up around 10 o'clock or 11. How about putting a separate volume control on the unit for the headphone outputs? This would enable me to turn off the headphones when using speakers, too._

 

Do you have the 10 dB headphone pads set in the DAC1? 

 Otherwise, you may want to use a media player that has well designed volume control. VLC is a Mac media player whose volume control causes very little distortion.

 Thanks,
 Elias


----------



## Lord Chaos

I haven't opened the DAC1 to look at the jumpers. The manual says that the pads are set for attenuation, but they may not be. I'll take a look.


----------



## audioengr

Quote:


  Originally Posted by *EliasGwinn* /img/forum/go_quote.gif 
_The output attenuators do not affect the performance of the output stage whatsoever (according to our testing and our ears!). I run my DAC1 in my recording studio at the -20 dB level because it allows me to run the volume pot into its most optimal range (>10 o'clock).

 Now, it should be said that certain equipment (of questionable design) being driven by the classic DAC1 may be affected by the increase in source impedance that the attenuators present. A well designed device (high input impedance, low input capacitance) will not have any performance comprise when using the attenuators. The attenuators would only affect the sound quality if the equipment that was being driven by the DAC1 had low input impedance or high input capacitance, or if the DAC1 was driving a long cable. 

 Unfortunately, some "audiophile" gear has poorly designed input stages with much too low impedance or large noise-draining capacitors which can affect frequency response drastically. In those cases, the source impedance should be as low as possible. I don't know what this reviewer was using as a pre and/or amplifier, so I can't comment on the validity of his experience. But if the device being driven is designed properly, the attenuators will have no affect on the audio whatsoever.

 The DAC1 USB, however, relieves this problem altogether with the new high-current output drivers. These new drivers will drive even the most difficult loads at any attenuator setting with out any performance comprimises.

 Thanks,
 Elias_

 


 I ALWAYS set the attenuators to minimum attenuation. IMO, any output impedance higher than 100 ohms tends to act as a low-pass R-C filter with most equipment and cables, not to mention the added noise of the series resistors themselves. Also, the earlier in the chain that you apply most of the gain, the less noise gets amplified by the downstream components, resulting in lower overall noise, so I prefer a much higher output voltage than the 2.25VRMS typical of most DAC's.

 BTW, regarding jumpers, I changed the three internal power supply voltage jumpers in the DAC-1 to low-resistance, high-current berillium-copper jumpers and it seems to have made an improvement. The stock jumpers are really designed for low-level signals, not power currents. This is a really cheap change.

 Steve N.


----------



## EliasGwinn

Quote:


  Originally Posted by *audioengr* /img/forum/go_quote.gif 
_I ALWAYS set the attenuators to minimum attenuation. IMO, any output impedance higher than 100 ohms tends to act as a low-pass R-C filter with most equipment and cables, not to mention the added noise of the series resistors themselves. Also, the earlier in the chain that you apply most of the gain, the less noise gets amplified by the downstream components, resulting in lower overall noise, so I prefer a much higher output voltage than the 2.25VRMS typical of most DAC's.

 BTW, regarding jumpers, I changed the three internal power supply voltage jumpers in the DAC-1 to low-resistance, high-current berillium-copper jumpers and it seems to have made an improvement. The stock jumpers are really designed for low-level signals, not power currents. This is a really cheap change.

 Steve N._

 

Steve,

 It is not just your opinion, it is a fact. Any output impedance WILL cause a RC low-pass filter (high-end roll off) with a capacitive load. 

 However, the exact roll-off quantity can be determined based on the specific numbers. The 100 ohm standard you have chosen will usually be safe, however, even 100 ohms will have audible roll-off effects with enough cable (> 400 ft @ 32 pf per foot) and/or load capacitance.

 If the attenuators in the DAC1 (classic) are set to 30 dB, the output impedance is 160 ohms. This means you can drive 255 ft of cable @ 32 pf per foot before you experience 0.1 dB attenuation at 20 kHz. In other words, driving 100 ft of cable into a well-built amplifier will be done just as well with the 30 dB attenuators as with the 0 dB attenuators. 

 Also, with the DAC1 USB, the 30 dB attenuation setting results in a lower output impedance (43 ohms) then with no attenuation (60 ohm).

 The manuals for the DAC1 and DAC1 USB illustrate their respective impedances when using various XLR attenuation settings, and also lists the length of cable that can be driven at these settings. Here are the links to these manuals:

 DAC1 (table is on page 8):
http://www.benchmarkmedia.com/dac1/DAC1-Manual.pdf

 DAC1 USB (table is on page 14):
http://www.benchmarkmedia.com/manual...USB_Manual.pdf



 Thanks,
 Elias


----------



## Jon L

Quote:


  Originally Posted by *EliasGwinn* /img/forum/go_quote.gif 
_I can try to put some specific numbers together for you, but here is our testing method:

 We play a test-tone (10 kHz pure sine wave, -1 dBFS) via the media player in question. We set all settings necessary to achieve bit-transparency (sample rate, volume control, etc). We then monitor the FFT. When everything is set properly, the FFT looks like one tone at 10 kHz, with nothing but -135 dBFS noise floor around it (for 16-bit audio). 
_

 

Oh, I only read the PC portion of the article. Anyway:

 "A 16-bit 10k sine wave played through iTunes on OSX 10.4.6 with the volume control set near 50% (-19dB). The resulting distortion is well above the -129 dBFS noise floor and, consequently, is detrimental to the quality of the audio."

 So can we assume the distortion decreases as volume control attenuates less? I say that b/c I have my Foobar volume usually around -3 dB, and I can't subjectively detect any audible sound quality difference. In fact, I don't think I hear any distortion at even -8 dB or so..


----------



## milkpowder

That's really all that matters, right? I admit I can't tell the difference between bit perfect, ASIO and whatever other technical jargon word there is for how my digital music is turning into analogue sound waves. I have also never heard any distortion from using software volume controls. This is not to say that there isn't _measurable_ distortion.


----------



## audioengr

Quote:


  Originally Posted by *EliasGwinn* /img/forum/go_quote.gif 
_Steve,

 It is not just your opinion, it is a fact. Any output impedance WILL cause a RC low-pass filter (high-end roll off) with a capacitive load. 

 However, the exact roll-off quantity can be determined based on the specific numbers. The 100 ohm standard you have chosen will usually be safe, however, even 100 ohms will have audible roll-off effects with enough cable (> 400 ft @ 32 pf per foot) and/or load capacitance.

 If the attenuators in the DAC1 (classic) are set to 30 dB, the output impedance is 160 ohms. This means you can drive 255 ft of cable @ 32 pf per foot before you experience 0.1 dB attenuation at 20 kHz. In other words, driving 100 ft of cable into a well-built amplifier will be done just as well with the 30 dB attenuators as with the 0 dB attenuators. 

 Also, with the DAC1 USB, the 30 dB attenuation setting results in a lower output impedance (43 ohms) then with no attenuation (60 ohm).

 The manuals for the DAC1 and DAC1 USB illustrate their respective impedances when using various XLR attenuation settings, and also lists the length of cable that can be driven at these settings. Here are the links to these manuals:

 DAC1 (table is on page 8):
http://www.benchmarkmedia.com/dac1/DAC1-Manual.pdf

 DAC1 USB (table is on page 14):
http://www.benchmarkmedia.com/manual...USB_Manual.pdf



 Thanks,
 Elias_

 



 Sounds like the 0dB setting is a good one, except for the early gain consideration. I'll give it a try.

 Steve N.


----------



## smeggy

Quote:


  Originally Posted by *milkpowder* /img/forum/go_quote.gif 
_That's really all that matters, right? I admit I can't tell the difference between bit perfect, ASIO and whatever other technical jargon word there is for how my digital music is turning into analogue sound waves. I have also never heard any distortion from using software volume controls. This is not to say that there isn't measurable distortion._

 


 Cloth ears 
	

	
	
		
		

		
			





 I'm the same really, I normally don't care what it's doing as long as it sounds good, but psychologically I feel better if I know it's doing what it should and can. Even if I can't tell


----------



## Scrith

Some positive feedback on the DAC1 USB:

 I received it a few days ago and immediately began comparing it to my existing setup (Lavry DA10 + Empirical Audio Off-Ramp Turbo 2 w/ Superclock 4). They sounded about the same to me. So I brought out my secret uber test: my wife's golden ears and her lack of interest for whatever strange setup I'm making her test. She listened to both setups without knowing which was which. She preferred the DAC1 USB by a slight amount, saying it seemed to have more detail and a broader soundstage (my interpretation of her hand gestures...close together for the DA10/Off-Ramp combo vs. far apart for the DAC1 USB).

 I've been living with the DAC1 USB for a couple of days now and I've noticed a few other things: Windows sounds (the clicks and pops and beeps in the UI) actually sound right (previously these were usually a bit distorted, or clipped at the beginning), the sound meter in Foobar2000 is not delayed by several seconds anymore, and my sound during games works a lot better even when playing music in the background with Foobar2000. I attribute all of these problems to M-Audio's lousy USB drivers. I understand that Empirical Audio will soon have a firmware update that makes the Off-Ramp work like the DAC1 USB (using Windows' USB driver), so maybe that will help there.

 Also, I'm able to listen to music resampled to 96K in Foobar2000 without any problems using the USB connection (something I have never been able to do with my other USB audio devices).

 Sound-wise the DAC1 USB really does seem to be improved. I changed to a Lavry DA10 in mid 2006 from a Benchmark DAC1 (built in late 2005) after my wife picked the Lavry in a blind comparison. And that late 2005 DAC1 sounded significantly better than my original (late 2003?) DAC1. Congratulations on the continued improvements to a wonderful DAC.

 By the way, I have had one wish for the DAC1 since I first started using them in late 2003: an on/off switch. Well, and I always wished for a USB or Firewire input, but that wish has finally been granted.


----------



## Jetlag

Quote:


  Originally Posted by *Scrith* /img/forum/go_quote.gif 
_ (my interpretation of her hand gestures...close together for the DA10/Off-Ramp combo vs. far apart for the DAC1 USB). 
	

	
	
		
		

		
		
	


	


_

 

It appears that women are now using the same system and scale to rate DACs as they do to rate men.


----------



## laxx

Quote:


  Originally Posted by *Jetlag* /img/forum/go_quote.gif 
_It appears that women are now using the same system and scale to rate DACs as they do to rate men. 
	

	
	
		
		

		
		
	


	


_

 

Hah! That's funny.


----------



## audioengr

Quote:


  Originally Posted by *Scrith* /img/forum/go_quote.gif 
_Some positive feedback on the DAC1 USB:

 I received it a few days ago and immediately began comparing it to my existing setup (Lavry DA10 + Empirical Audio Off-Ramp Turbo 2 w/ Superclock 4). They sounded about the same to me. So I brought out my secret uber test: my wife's golden ears and her lack of interest for whatever strange setup I'm making her test. She listened to both setups without knowing which was which. She preferred the DAC1 USB by a slight amount, saying it seemed to have more detail and a broader soundstage (my interpretation of her hand gestures...close together for the DA10/Off-Ramp combo vs. far apart for the DAC1 USB).

 I've been living with the DAC1 USB for a couple of days now and I've noticed a few other things: Windows sounds (the clicks and pops and beeps in the UI) actually sound right (previously these were usually a bit distorted, or clipped at the beginning), the sound meter in Foobar2000 is not delayed by several seconds anymore, and my sound during games works a lot better even when playing music in the background with Foobar2000. I attribute all of these problems to M-Audio's lousy USB drivers. I understand that Empirical Audio will soon have a firmware update that makes the Off-Ramp work like the DAC1 USB (using Windows' USB driver), so maybe that will help there.

 Also, I'm able to listen to music resampled to 96K in Foobar2000 without any problems using the USB connection (something I have never been able to do with my other USB audio devices).

 Sound-wise the DAC1 USB really does seem to be improved. I changed to a Lavry DA10 in mid 2006 from a Benchmark DAC1 (built in late 2005) after my wife picked the Lavry in a blind comparison. And that late 2005 DAC1 sounded significantly better than my original (late 2003?) DAC1. Congratulations on the continued improvements to a wonderful DAC.

 By the way, I have had one wish for the DAC1 since I first started using them in late 2003: an on/off switch. Well, and I always wished for a USB or Firewire input, but that wish has finally been granted. 
	

	
	
		
		

		
		
	


	


_

 

I recommend updating your playback setup for the Off-Ramp:
http://www.audiocircle.com/circles/i...?topic=40068.0

 This improves Off-Ramp clarity and focus significantly on a PC. I found it to be equivalent to the driverless firmware. I still recommmend Foobar 0.8.3 and SRC at 24/96.

 BTW - you should hear a modded DAC-1 USB with a Superclock clocking the upsampler. Really outstanding, but I still prefer the NOS I2S version with SRC upsampling by a small margin. Gives the customer more flexibility too.

 Steve N.


----------



## EliasGwinn

Quote:


  Originally Posted by *milkpowder* /img/forum/go_quote.gif 
_That's really all that matters, right? I admit I can't tell the difference between bit perfect, ASIO and whatever other technical jargon word there is for how my digital music is turning into analogue sound waves. I have also never heard any distortion from using software volume controls. This is not to say that there isn't measurable distortion._

 

The quality of any solution is limited by the perceived severity of the problem.


----------



## EliasGwinn

Quote:


  Originally Posted by *Scrith* /img/forum/go_quote.gif 
_Some positive feedback on the DAC1 USB:

 I received it a few days ago and immediately began comparing it to my existing setup (Lavry DA10 + Empirical Audio Off-Ramp Turbo 2 w/ Superclock 4). They sounded about the same to me. So I brought out my secret uber test: my wife's golden ears and her lack of interest for whatever strange setup I'm making her test. She listened to both setups without knowing which was which. She preferred the DAC1 USB by a slight amount, saying it seemed to have more detail and a broader soundstage (my interpretation of her hand gestures...close together for the DA10/Off-Ramp combo vs. far apart for the DAC1 USB).

 I've been living with the DAC1 USB for a couple of days now and I've noticed a few other things: Windows sounds (the clicks and pops and beeps in the UI) actually sound right (previously these were usually a bit distorted, or clipped at the beginning), the sound meter in Foobar2000 is not delayed by several seconds anymore, and my sound during games works a lot better even when playing music in the background with Foobar2000. I attribute all of these problems to M-Audio's lousy USB drivers. I understand that Empirical Audio will soon have a firmware update that makes the Off-Ramp work like the DAC1 USB (using Windows' USB driver), so maybe that will help there.

 Also, I'm able to listen to music resampled to 96K in Foobar2000 without any problems using the USB connection (something I have never been able to do with my other USB audio devices).

 Sound-wise the DAC1 USB really does seem to be improved. I changed to a Lavry DA10 in mid 2006 from a Benchmark DAC1 (built in late 2005) after my wife picked the Lavry in a blind comparison. And that late 2005 DAC1 sounded significantly better than my original (late 2003?) DAC1. Congratulations on the continued improvements to a wonderful DAC.

 By the way, I have had one wish for the DAC1 since I first started using them in late 2003: an on/off switch. Well, and I always wished for a USB or Firewire input, but that wish has finally been granted. 
	

	
	
		
		

		
			



_

 

Scrith,

 I'm glad you and your wife enjoy the DAC1 USB!! 
	

	
	
		
		

		
		
	


	







 We will keep in mind your suggestion for adding a power switch. We really appreciate feedback from our customers, so feel free to keep it coming!!

 Thanks,
 Elias


----------



## clar2391

This is a great thread with lots of useful information!

 Since reading about the various forms and levels of distortion created by the various media players, it occurred to me that there must be a player out there with no volume control, or at least one where the volume control could be disabled.

 I seem to remember a previous version of Foobar where the volume control was a plug-in that could be removed if not needed. The current version doesn't seem to offer this option.

 I've searched all over the internet for a way to disable the volume controls in Windows Media Player, iTunes, and Foobar, but no luck finding anything.

 I know that I can just check the volume controls in the various players to be sure they're at 100%, but it seems a better solution would be to just take the volume control out of the equation.

 Does anyone know if such a thing exists?


----------



## audioengr

Quote:


  Originally Posted by *clar2391* /img/forum/go_quote.gif 
_This is a great thread with lots of useful information!

 Since reading about the various forms and levels of distortion created by the various media players, it occurred to me that there must be a player out there with no volume control, or at least one where the volume control could be disabled.

 I seem to remember a previous version of Foobar where the volume control was a plug-in that could be removed if not needed. The current version doesn't seem to offer this option.

 I've searched all over the internet for a way to disable the volume controls in Windows Media Player, iTunes, and Foobar, but no luck finding anything.

 I know that I can just check the volume controls in the various players to be sure they're at 100%, but it seems a better solution would be to just take the volume control out of the equation.

 Does anyone know if such a thing exists?_

 


 If you use Foobar 0.8.3, then you just remove the volume control DSP tool from the "active" column. If you dont have Foobar 0.8.3, email me and I'll attach it in a return email. I will rename the extension so that your mail filters allow it. Just change it back to .exe . Foobar2000 version 0.8.3 is still the best sounding player IMO.

 Steve N.
 Empirical Audio


----------



## Jon L

Quote:


  Originally Posted by *Scrith* /img/forum/go_quote.gif 
_Some positive feedback on the DAC1 USB:

 Also, I'm able to listen to music resampled to 96K in Foobar2000 without any problems using the USB connection (something I have never been able to do with my other USB audio devices).
_

 

Thanks for your impressions. Since the new DAC-1 is fed by USB, do you have impressions of old DAC-1 vs. new DAC-1 when fed via spdif? 

 In addition, since Benchmark upsamples everything to 110kHz anyway, have you tried setting the SRC resampler to 110kHz instead of 96kHz? 

 Actually, I don't think Benchmark upsamples to exactly 110 kHz; Elias, what is the precise upsampling frequency of DAC-1?


----------



## EliasGwinn

Quote:


  Originally Posted by *Jon L* /img/forum/go_quote.gif 
_Thanks for your impressions. Since the new DAC-1 is fed by USB, do you have impressions of old DAC-1 vs. new DAC-1 when fed via spdif? 

 In addition, since Benchmark upsamples everything to 110kHz anyway, have you tried setting the SRC resampler to 110kHz instead of 96kHz? 

 Actually, I don't think Benchmark upsamples to exactly 110 kHz; Elias, what is the precise upsampling frequency of DAC-1?_

 

Jon,

 The exact SRC frequency is 110632.8125 Hz.

 Thanks,
 Elias


----------



## EliasGwinn

Quote:


  Originally Posted by *Jon L* /img/forum/go_quote.gif 
_Thanks for your impressions. Since the new DAC-1 is fed by USB, do you have impressions of old DAC-1 vs. new DAC-1 when fed via spdif? 

 In addition, since Benchmark upsamples everything to 110kHz anyway, have you tried setting the SRC resampler to 110kHz instead of 96kHz? 

 Actually, I don't think Benchmark upsamples to exactly 110 kHz; Elias, what is the precise upsampling frequency of DAC-1?_

 

After re-reading your post, I realized you are talking about setting the sample rate in foobar to match the SRC of the DAC1 USB. Unfortunately, this won't help because the USB input is limited to 96 kHz. So, it is going to get to the DAC1 USB post-software SRC down to 96 (probably).

 Thanks,
 Elias


----------



## Jetlag

I hope this is not considered off-topic, but since we have been speaking about native USB audio drivers...

 I have not done a clean re-install of XP in over 3 years (yeah, I know), and during this time I have installed and removed lots of HW and SW. I have a feeling that my "native" USB drivers may have either been replaced, overwritten, etc at least once along the way. In particular I've had at least 3 different audio cards from 2 manufacturers, a number (3-4) of different USB audio devices and who knows what I may have forgotten. Since my new USB DAC1 will be here by the end of the week I was wondering if there was a definitive list of 'native' MS USB drivers that I could compare to my INF folder?

 1. Can someone list the correct drivers (name, version, size) for XPSP2? 
 2. If not, can you point me to a list? 
 3. Finally, are there any drivers that I want to delete that may interfere with the bit-transparent audio that I am so looking forward to?

 Thanks!


----------



## EliasGwinn

Quote:


  Originally Posted by *Jetlag* /img/forum/go_quote.gif 
_I hope this is not considered off-topic, but since we have been speaking about native USB audio drivers...

 I have not done a clean re-install of XP in over 3 years (yeah, I know), and during this time I have installed and removed lots of HW and SW. I have a feeling that my "native" USB drivers may have either been replaced, overwritten, etc at least once along the way. In particular I've had at least 3 different audio cards from 2 manufacturers, a number (3-4) of different USB audio devices and who knows what I may have forgotten. Since my new USB DAC1 will be here by the end of the week I was wondering if there was a definitive list of 'native' MS USB drivers that I could compare to my INF folder?

 1. Can someone list the correct drivers (name, version, size) for XPSP2? 
 2. If not, can you point me to a list? 
 3. Finally, are there any drivers that I want to delete that may interfere with the bit-transparent audio that I am so looking forward to?

 Thanks!_

 

I will get a list of the drivers needed for the DAC1 USB, but you probably won't have any problems, especially if your connected to the internet. When you first plug in the DAC1 USB, if you don't have the necessary drivers, a window will pop up asking if Windows can search the web for the necessary drivers. You can choose "Yes, this time only", and it will find, download, and install the drivers you need.

 As a general rule, I recommend deleting all unused, non-native drivers and software. It may or may not make a difference, but I have found that the most unusual, unexpected, and difficult problems come about from conflicts in drivers/software. I make it a habit to remove all unnecessary software from my computer, especially since you can usually go to the manufacturer's website to download the drivers if you need them again in the future.

 Thanks,
 Elias


----------



## Jetlag

Quote:


  Originally Posted by *EliasGwinn* /img/forum/go_quote.gif 
_As a general rule, I recommend deleting all unused, non-native drivers and software. It may or may not make a difference, but I have found that the most unusual, unexpected, and difficult problems come about from conflicts in drivers/software. I make it a habit to remove all unnecessary software from my computer, especially since you can usually go to the manufacturer's website to download the drivers if you need them again in the future.
 Thanks,
 Elias_

 

I agree 100%, BUT, we all know that not all uninstall programs are created equally. Many tend to leave little software and driver 'dust bunnies' all over your HDD and sometimes it can be tricky if not downright impossible to ferret them all out.


----------



## EliasGwinn

Quote:


  Originally Posted by *Jetlag* /img/forum/go_quote.gif 
_I agree 100%, BUT, we all know that not all uninstall programs are created equally. Many tend to leave little software and driver 'dust bunnies' all over your HDD and sometimes it can be tricky if not downright impossible to ferret them all out._

 

This is very true. An occasional, good ol' complete format and re-install is a good idea for this reason.

 Thanks,
 Elias


----------



## little-endian

Hi Elias,

 thanks again for the details some postings ago. Wow, this is quite an active thread.

 You said the optical jack had nothing to do with the capability to handle 192 kHz. What does matter then instead? Although I purchased the DAC1 via Analog Audio, I got the one, Allan Burdick was using before. So I suppose it won't be one of the newest models. If can name you the serial number if that would be of any use for you.

 I'm not sure but I guess that the old version the DAC1 could be have one advantage over the USB version - when it comes to the display of errors, right? Just as a kind of "proof of concept", I hooked up some standard 5 1/4" CD-ROM to it. As long as the drive delivers the data correctly through its S/PDIF TTL connector, the sound will be exactly the same. Without much surprise, this is also the case. However, I discovered that several drives handled the output differently. Some activate their S/PDIF output until an audio track is actually being played, interrupting the signal while pause or stop. Also, the DAC1 loses sync after manually skipping the track. Only one of the drives I've tested, seems to provide a standard conform S/PDIF signal, which is all zero while reading mode 1 or 2 discs, stop or pause. Also, the signal is stable when skipping tracks. Strangly it seems to have problems when playing rewritables. Here, the DAC1 still syncs to the signal, but its "non pcm" led flashes up from time to time and hearable tickles occur.

 Elias, one question in general: For the case an uncorrectable errors occur while reading an audio-cd (E32), the S/PDIF standard states a flag to declare the sent data as "invalid", so a DAC, receiving the information knows that there were errors which couldn't be reconstructed by the CD-player's C2-decoder. How does the (old) DAC1 react in such a case? Which one (if any) of the diodes will light up (from my tests I tend to believe it is the 'non pcm' led)?Because the stream itself (the error led seems to be responsable for) is still conform to the standard, no matter of the amount of errors at the user data level.

 I've read the manual of the DAC1 USB. There it says that the error is displayed here by the ammount of blinking. I guess that in such experimental case, the old DAC1's error display will be more exact. Am I correct? Even if so, this "fault" of the new one will hardly be any issue for most users, since this is purely experimental.  I just wondered ...

 Another thing about the resampling to 110 kHz:

 So sources of 192 kHz won't give a DAC1's user very much in comparison to 96 kHz, right? Besides that it is commonly contested if higher sample rates than about the ~ 44.1 - 48 kHz are of any real advantage, at all.

 Some here thought about resampling audio to the sample rate which is internally used by the DAC1, before sending it via USB or S/PDIF. In the case of the USB connection, this intension will be useless already because of the limitation to 96 kHz, of course. But even when using other connections I wonder if the conversion of the DAC1 itself, which is done by the hardware from Analog Devices (as far as I know) wouldn't do this better. If this is true, every change of the sample rate would be then completely unnecessary, right?

 I've already heard several claims that different DAC1 builts (year of the manufaction) would sound differently. What I think to know from you, the sonic performance had never so far (neither older DAC1s nor the new USB version). Some posts ago here, another one made this claim. If Benchmark is right, then his expericene must be pure illusion. There is not other explanation. The oft-quoted topic "jitter" doesn't seem to be one more with the products of Benchmark.

 I'm keen on getting more information about all of this.

 Thanks,

 little-endian


 PS @ all: I think most of you quote too much. This makes it hard to read through the thread (especially via mobile phones 
	

	
	
		
		

		
		
	


	




). Remember: If everything is emphasized, actually nothing is emphasized.


----------



## laxx

Quote:


  Originally Posted by *Jetlag* /img/forum/go_quote.gif 
_I agree 100%, BUT, we all know that not all uninstall programs are created equally. Many tend to leave little software and driver 'dust bunnies' all over your HDD and sometimes it can be tricky if not downright impossible to ferret them all out._

 

Check out the program called Driver Cleaner. It's very nice.


----------



## audioengr

Quote:


  Originally Posted by *EliasGwinn* /img/forum/go_quote.gif 
_After re-reading your post, I realized you are talking about setting the sample rate in foobar to match the SRC of the DAC1 USB. Unfortunately, this won't help because the USB input is limited to 96 kHz. So, it is going to get to the DAC1 USB post-software SRC down to 96 (probably).

 Thanks,
 Elias_

 

I have found that upsampling to 24/96 prior to the hardware 24/192 upsampler can be beneficial, depending on the software upsampler.

 Steve N.


----------



## EliasGwinn

Little-Endian,

 There seems to be quite a few questions here, but I'll try to address them as best I can. 

 -----------Question about optical input--------------------

 We switched the optical jack to one that has a shutter door, replacing the one which has a removable plug/dust-cap (which was easy to lose when it wasn't used). The difference is purely mechanical, however; it does not determine the DAC1's ability to handle 192 kHz via the optical port. The determining factor is the date when it was manufactured because optical receivers were previously unable to accept 192 kHz. When receivers were available that were capable of accepting 192 kHz, we began to manufacture the DAC1 with them. This was appx. May 2004. I would be very interested to know the serial number and manufacturing date of your DAC1. Could you please PM them to me?

 ----------Question about error indicators-----------

 As for error indications on the DAC1 Classic and the DAC1 USB, the DAC1 USB actually has the ability to indicate more accurately what the problem is. The blinking scheme is more directly correlated to the nature of the problem.

 The experiment you discussed is interesting. However, I can't tell you exactly what error message would be displayed in that case because it depends on what the transport streamed. Different transports will react differently. Without investigating your experiment, if I were to guess, I would guess that the "Error" LED would light in that particular case. The "Non-PCM" LED should only occur when a digital signal is present which is not compatible with the DAC1. 

 ----------Question about Sample-rates and conversion---------

 About sample-rate conversion, the best results are achieved when the original sample-rate of the audio is left unchanged (ie, there is no sample rate conversion) before it is streamed to the DAC1. 

 Higher sample rates up to 96 kHz are generally agreed to be better then lower because of the added bandwidth and resolution for processing. However, most professional audio engineers (equipment manufacturers and studio engineers) believe that the technology for 192 kHz has not developed enough. This is not specific to a manufacturer; it is the state of the technology. The filter required in A-to-D conversion at 192 kHz has inherent issues, and the negatives are currently outweighing the positives. 

 --------Question about early DAC1's vs. later DAC1's---------

 The very first generation of DAC1's (manufactured between 10/02 and 5/03) had a few things that may have affected the sound. For example, the RCA outputs had a higher impedance which may affect the sound with certain amps/preamps. These were all relatively small factors that should not have made a difference in most cases. However, all DAC1's manufactured after 5/03 should sound the same. After all, they have the same IC's, OpAmps, circuit design, components, etc. 

 Hope I got all your questions...

 Thanks,
 Elias


----------



## little-endian

Hi Elias,

 thank you again for your interest and excellent support. Very unusual for a manufacturer's support staff ...

 Ah, I think now I understood what you mean in regard to the 192 kHz capability - the mechanical layout and the used transmitters were change independently, right?
 I'm glad to send you the date and serial number via PM. I will do this in about 9 hours when I come home.

 I agree with you that the new USB version is able to display the problem which occured more accurately, but this is just about the more on information, not the latency, right? The old DAC1 might be able to display the errors somewhat 'faster'. I mean, some errors occur so shortly in time that the error is over long ago before the indicators of the new DAC1 could start to blink, at all. That's what I thought. You see, since I have the old version and probably now chance to get the new version in exchange without a huge loss of money, I'm convulsively searching for any advantages mine could have, haha.

 Yeah, the benefits of higher and higher sample rates and bit deepths seems to become more and more of a theoretical nature. What is questionable is if the higher bandwidth does really change the sonic experience because it is far beyond of our ear's capabilities. But I won't follow this thoughts here because otherwise i fear we get too off-topic. Interesting that you confirm the pretty constant sonic performance over several 'generations' of the DAC1. So I suppose some people here (like it seems to be the case often when it cames to high-end) believe to hear thing that don't exist but what they want to hear. If nothing was changed, there is no other explanation but pure voodoo.

 Now to the experiment. I like it very much doing such tests because one gains interesting information, often not written in such detail anywhere and the good thing is that it is provable (in opposite to most sonic descriptions). Perhaps this would be worth an own thread but since I already started with it, I'll take the risk to get a bit off-topic:

 What you said about the different transports which will react differently should be examined. Thought strictly about that it is a real shame that this is the case! Why? Because everything is standardized. When it comes to S/PDIF, there are several flags to indicate the status. Whether or not the samples are valid, pcm or other data, etc. And what is? Almost every device handles this differently. There is resampling, the lack of error indication, data corruption and so on. Per se it is a fundamental requirement that a cd-rom drive or cd-player simply provides the data 1:1 which is stored on the medium (at the user's data level, the raw data will be different everytime, of course). What makes the digital technique so ingenious is wasted here. Many manufacturers seem to forget doing their homework. How could it otherwise be that some so-called 'high end' players aren't even able to use the deemphasis correctly? So, bitched enough about that for now.
 I better go on with the facts: I've checked the behavior of an ASUS CD-S500/A again. It is one of the few drives, I suppose that deliver a continues S/PDIF signal. Other ones enable this output only when an audio-cd is actually being played and interrupt the signal on manual track changes. Somehow the driver declares the data as "non pcm" when errors occur instead of simply use the intended 'valid'-flag for this. I pretty sure about this because the kmixer driver which I use for my (nowadays crappy) SBLive! value displays "AC3" for a short moment in that case. This is when the "non pcm"-led of the DAC1 ligths up for some ms. It seems to occur only on E32 errors - flawless (pressed) cds play without "non pcm" - errors while scratchy ones (or rewritables sometimes) produce this strange status. Amazingly, the rewritables (which this drive seems to have a problem with) can be copy via 'digital audio extraction' without any C2 errors (which EAC is able to display). The resulting files are free of errors, copied at > 10x. It is very strange that the same cd can't be read properly at 1x in burst access. Another open question. It is interesting also that almost every drive i tested provides a different sample rate. I think this is just some kind of nominal indication because otherwise the signal would have to be resampled (I hardly believe that a simple CD-ROM has even the ability to do this - which is very beneficial here) or the pitch would be lower which I don't believe either. 

 Please refer to one of my postings in another forum. It is German, but you can follow the links for the screenshots here.

http://www.hifi-forum.de/index.php?a...54&thread=6074

 When using a Plextor drive, no errors occur so far but besides that it is the kind of drive which enable the output not all of the time, it is suspected to be not bit-perfect. I can't test this yet because my ordered sound card with S/PDIF input and the VIA Envy24 chipset (said to process all data unchanged) didn't arrive so far. At one site the ASUS was tested to be bit-perfect (unfortunately the strange problem with rewriteble media) and that I can say about the sound quality - it is completely unnecessary to use any high quality transport as long as it is error free. Here a picture of the probably most puristic cd-player setup - just for a bit amusement. Forgive me the noisy quality - after purchasing the DAC1 there was no money left to afford a camera which is worth the name. 

 You're right, Elias. There are really a few questions here and it seems there are even more. Good to have you here ... 

 little-endian


----------



## EliasGwinn

Little-Endian,

 It seems perfectly obvious to me that the giraffe is causing the data read errors. I'm no zoologist or anything, but he looks quite suspect to me
	

	
	
		
		

		
			





 Elias


----------



## little-endian

@Elias

 Haha, you're funny man! 
	

	
	
		
		

		
		
	


	




 Oh yes, I learned that many things can change the integrity of a S/PDIF stream. Even a nice giraffe has to be checked twice for meaning no harm. Unfornuately the ears are not very suited for headphones - that a pity, their hearing capability is probably far beyond of ours. 
	

	
	
		
		

		
		
	


	




 Ah, I've sent you a PM for the serial number ...


----------



## music_man

mr. gwinn,

 i am intrested in what improvement the upgraded output drivers provide. if the non usb unit had an impedance mismatch with an amplifier what would i be hearing to let me know there was a problem?

 thanks,
 music_man


----------



## Crowbar

I have two questions.

  Quote:


  Originally Posted by *EliasGwinn* /img/forum/go_quote.gif 
_The Benchmark DAC1 USB runs in synchronous mode. The reason for this is that it lets the host (kmixer) operate at the original sample-rate of the audio being played at all times. If the kmixer is not forced to do any sample-rate conversion, it can maintain bit-transparent operation._

 

How is the time base for streaming determined in Windows? Is it from the PC's own frequency generator? Or is it like in some Linux drivers, where the software virtualizes the sound hardware buffer, and thus time base is derived from the sound hardware's own clock (the latter allows the hardware to be master)?

 My second question is regarding this statement from the manual:
  Quote:


 Instead the converter oversampling-ratio is varied with extremely high precision to achieve the proper phase relationship to the reference clock. 
 

How is that different from what an ASRC chip does?

 Thanks.

 PS
  Quote:


 This means you can drive 255 ft of cable @ 32 pf per foot 
 

My cables are 4 pF per foot.


----------



## EliasGwinn

Hey folks,

 We are in the middle of some serious office re-arrangements....moving this person's office here, and this person's office there, and engineering test benches over there, etc, etc etc. The best part of all of this is I am able to put together a brand new listening room!!

 Needless to say, I won't be able to answer your questions before the weekend, but I promise I'll be back on Monday or Tuesday and we'll pick it up then.

 Thanks!
 Elias


----------



## The Monkey

Elias,

 I was wondering if you could comment on Benchmark's decision not to include a power switch on the DAC1 and if it is indeed recommended to leave the unit on 24/7.

 Thanks!


----------



## kool bubba ice

Quote:


  Originally Posted by *The Monkey* /img/forum/go_quote.gif 
_Elias,

 I was wondering if you could comment on Benchmark's decision not to include a power switch on the DAC1 and if it is indeed recommended to leave the unit on 24/7.

 Thanks!_

 

I would assume that would be the case, since there is no power switch.. My VHP-1/XDAC3/X10-3 were the same way.. NP.


----------



## music_man

i hope mr. gwinn gets back soon. i cannot drive the krell with the standard dac1. then i saw it says it needs a low impedance xlr connection. i don't know about high current though. i don't want to fry anything. the standard dac1 is all distorted at the +4db at 0dbfs that the krell takes for input. that would be the 20db pad. i don't want to order the usb unit if i it won't work either.

 music_man


----------



## EliasGwinn

Hey folks, hope everyone had a good weekend!

 I will try to catch up on this thread throughout the day.

 Music_Man, can you please describe your problem in detail, including the entire signal chain in question and what problems you are noticing?

 Thanks,
 Elias


----------



## music_man

sony scd-1, into the benchmark with s/pdif, into krell kav400xi with xlr's. it is not actually distorted. the sound is much to "thin". it just lacks any "weight"(compared to the xlr's right out of the sony). plus hardly any bass. it is loud enough. also the left level is louder than the right level. i made another post about that but you can answer it here if you like. i know how to use a test tone and meter but i cannot do that at the moment. any other way to get the two sides even?

 thank you,
 music_man


----------



## EliasGwinn

Quote:


  Originally Posted by *music_man* /img/forum/go_quote.gif 
_mr. gwinn,

 i am intrested in what improvement the upgraded output drivers provide. if the non usb unit had an impedance mismatch with an amplifier what would i be hearing to let me know there was a problem?

 thanks,
 music_man_

 

Music_man,

 If the amplifier has low input impedance, the output drivers of the DAC1-Classic will suffer from distortion because their is not enough current limiting. Also, as with any balanced input, low-impedance will cause the common-mode rejection to suffer (EMI, ground loops, and other noise). If the amplifier has high input capacitance, the band-width will suffer (high-frequency roll-off). 

 Thanks,
 Elias


----------



## EliasGwinn

Quote:


  Originally Posted by *Crowbar* /img/forum/go_quote.gif 
_I have two questions.


 How is the time base for streaming determined in Windows? Is it from the PC's own frequency generator? Or is it like in some Linux drivers, where the software virtualizes the sound hardware buffer, and thus time base is derived from the sound hardware's own clock (the latter allows the hardware to be master)?
_

 

For the DAC1 USB, the clocking is done natively in the computer. This mode of operation is called Asynchronous. This mode is prone to significant amounts of jitter. Thankfully, the DAC1 is immune to jitter because it utilizes a proprietary clocking system called UltraLock.

  Quote:


  Originally Posted by *Crowbar* /img/forum/go_quote.gif 
_My second question is regarding this statement from the manual:

 How is that different from what an ASRC chip does?
_

 

It is, in fact, an ASRC chip that is used in the DAC1.

 Thanks,
 Elias


----------



## EliasGwinn

Quote:


  Originally Posted by *music_man* /img/forum/go_quote.gif 
_sony scd-1, into the benchmark with s/pdif, into krell kav400xi with xlr's. it is not actually distorted. the sound is much to "thin". it just lacks any "weight"(compared to the xlr's right out of the sony). plus hardly any bass. it is loud enough. also the left level is louder than the right level. i made another post about that but you can answer it here if you like. i know how to use a test tone and meter but i cannot do that at the moment. any other way to get the two sides even?

 thank you,
 music_man_

 

Music_man,

 I'm surprised by this. It could be that your DAC1 needs servicing, but its hard to say. Sometimes, when the DAC1 is A/B'd against other D/A sources (usually low quality), listeners sometimes say that the DAC1 seems to be "missing" low / mid range information. The reason for this is the absence of jitter artifacts. Converters that suffer from jitter distortion have an audible amount of extra low / mid-range information which is, in fact, jitter artifacts. 

 As for the issue with the balance between channels: Is the volume control low when you are experiencing this? The nature of a stereo pot is that they are inherently inaccurate in the first +/-25% of their range. Above that, the volume pot is extremely accurate between channels. If the volume gets too loud before you can turn the pot above the first 25%, the internal jumpers can adjust the range properly. 

 Please keep me informed about these issues. Also, let me know if you want us to take a look at it for you.

 Thanks,
 Elias


----------



## music_man

it was the calibrated pots that were off about 1db. someone else answered how to level them with a voltmeter in my other thread. it's fixed now.

 it does not seem "flat" simply in an a/b. if i listen to some other dacs they seem to have simply more "character". i know those are subjective terms.
 i think it is just because the dac1 is supposed to present what is on the source with out adding anything. that is why i said i prefer to listen to other dacs for enjoyment and use the dac1 for monitoring.

 that is not an insult, it is in fact a compliment. it is hard to make a product that does not add it's own sound to the signal chain. to be honest it seems the cheaper the dac the more color it adds. these are dacs with no jitter control. to some people this color is pleasent even though it means a technically inferior product. i know from a technical standpoint the dac1 is top of the line.

 music_man


----------



## EliasGwinn

Music_man,

 Thank you for the kind words. I'm glad you were able to correct the balance issue as well.

 It is true that, in an inexpensive design process, jitter usually is not addressed properly, if at all. The performance of these devices are usually very dependent on digital cables (length and quality), as well as source transmission. 

 I should say that this is not limited to inexpensive devices. We recently purchased another manufacturer's D-to-A device that has recently experienced a surge in popularity. (I won't mention the name of the manufacturer nor the device, as we refrain from publicly critiquing competitors. I will note that the device cost significantly more then the DAC1). We tested the device on the Audio Precision (AP) testing station, and we were shocked to find that this device had absolutely NO jitter attenuation!! In other words, if the AP induced the slightest amount of jitter on the digital signal, it rendered itself directly as distortion within the device. Here is a picture of the result of 0.25 UI of jitter at 44.1 kHz.






 The spike in the middle represents the audio - the only signal that was meant to be heard. The spikes all around it represent jitter artifacts at frequencies ranging from 100 Hz to 9 kHz, in steps of 300 Hz (ie, 100, 400, 700, etc.). The magnitude of the jitter is a very small 0.25 UI. 

 Thanks,
 Elias


----------



## dip16amp

Is the "UltraLock" more than just "asynchronous upsampling" which some other DACs also have or is there something more to it than a "marketing tool" as mentioned in post #3 and #5 in thread http://www.head-fi.org/forums/showthread.php?t=235830

 Just wanted to know if there is a difference.


----------



## The Monkey

Quote:


  Originally Posted by *The Monkey* /img/forum/go_quote.gif 
_Elias,

 I was wondering if you could comment on Benchmark's decision not to include a power switch on the DAC1 and if it is indeed recommended to leave the unit on 24/7.

 Thanks!_

 

Elias,

 Would you mind commenting on the above?


----------



## lowmagnet

Quote:


  Originally Posted by *The Monkey* /img/forum/go_quote.gif 
_Elias,

 Would you mind commenting on the above?_

 

He did


----------



## The Monkey

Quote:


  Originally Posted by *lowmagnet* /img/forum/go_quote.gif 
_He did_

 

Ah, but not whether it is recommended to leave on 24/7.


----------



## lowmagnet

Quote:


  Originally Posted by *The Monkey* /img/forum/go_quote.gif 
_Ah, but not whether it is recommended to leave on 24/7._

 

It's rack equipment, which can take 24/7 operation. Recommendation doesn't mean much to me since the alternative is to pull the power manually.

 I just wish the DAC1 would go into sleep mode when hooked up via USB. Unless you shut the host down, it's always up.


----------



## The Monkey

Quote:


  Originally Posted by *lowmagnet* /img/forum/go_quote.gif 
_It's rack equipment, which can take 24/7 operation. Recommendation doesn't mean much to me since the alternative is to pull the power manually.
_

 

Yes, but the recommendation means something to _me_, which is why I am asking Elias. But I agree, of course, that the inference must be that as rack equipment without a power switch it is built to withstand 24/7 uptime. It would be nice to see this confirmed by Benchmark.


----------



## chris_ah1

I was told in an audio store today that the PSU in the dac1 isn't the greatest - you should leave it to run for a bit before listening if you turn it off and don't want 24/7 operation.


----------



## Jetlag

This might be a stupid question, but was my new DAC1 USB supposed to come with a power cord? 

 There was not one in the box but the letter from Allem Burdick that came with it (see image) stated; "Enclosed in the package is a power cord and a BNC to RCA adapter for the digital coaxial input". Mine came with the adapter, a USB cable, spare fuses and a couple of pin jumpers. Just looking for clarification here, thanks.


----------



## music_man

i have the dac1 classic plugged into the krell with the xlr's. i am using the front panel variable gain. the internal pads are at the 0db setting. the volume knob is half way(50%). it sounds the best this way and i do not seem to hear any distortion.

 am i actually overloading the inputs on the amplifier this way and i am just not hearing it? or is this ok?

 thanks,
 music_man


----------



## twsmith

Quote:


  Originally Posted by *Jetlag* /img/forum/go_quote.gif 
_This might be a stupid question, but was my new DAC1 USB supposed to come with a power cord? 

 There was not one in the box but the letter from Allem Burdick that came with it (see image) stated; "Enclosed in the package is a power cord and a BNC to RCA adapter for the digital coaxial input". Mine came with the adapter, a USB cable, spare fuses and a couple of pin jumpers. Just looking for clarification here, thanks.




_

 

Yes, you should have received a power cord. I received my DAC1 USB several weeks ago and it included the power cord plus everything else that you mentioned. Check your shipping box again - I seem to remember the power cord being stuffed down along one side near the bottom, but even if it's not there, I'm sure it was an oversight and Benchmark should be able to ship one out to you right away. Since it's a detachable cord, you should be able to substitute another power cord if you have one available for the interim.


----------



## EliasGwinn

Quote:


  Originally Posted by *The Monkey* /img/forum/go_quote.gif 
_Ah, but not whether it is recommended to leave on 24/7._

 

Monkey,

 The short answer to your question is: it doesn't matter either way. The DAC1's performance is not affected by the amount of time it has been powered. The DAC1 requires no "warm up" time. 

 Before I began working at Benchmark, I worked full-time as a recording engineer at a professional recording studio (Subcat Studios). I still work there for certain projects. We have a DAC1 from 2003, and, as long as I have worked there (+/- 3 years), I have never seen it powered-down. It sounds as good as new. 

 I own a very current DAC1 at my home, and I also never turn it off. However, when I first set up my home system (5-6 months ago), the DAC1 sounded just like the DAC1 - from the moment I powered it up. It hasn't changed a bit since.

 The reason the DAC1 doesn't have a power switch is to prevent accidental/inopportune power-down when it is in a critical audio path. Many professional studios and broadcast facilities use the DAC1 in critical audio paths. In these settings, the DAC1 is powered and in-use 24/7, and the accidental (catastrophic!) power-down is eliminated by the lack of a power switch. 

 Thanks,
 Elias


----------



## EliasGwinn

Quote:


  Originally Posted by *dip16amp* /img/forum/go_quote.gif 
_Is the "UltraLock" more than just "asynchronous upsampling" which some other DACs also have or is there something more to it than a "marketing tool" as mentioned in post #3 and #5 in thread http://www.head-fi.org/forums/showthread.php?t=235830

 Just wanted to know if there is a difference._

 

Dip16amp,

 There is, in fact, a lot of engineering that has gone into UltraLock - much more then adding a ASRC chip. 

 However, I'm going to let John Siau, the engineer who conceived and implemented this technology, to answer your question.

 Thanks,
 Elias


----------



## EliasGwinn

Quote:


  Originally Posted by *Jetlag* /img/forum/go_quote.gif 
_This might be a stupid question, but was my new DAC1 USB supposed to come with a power cord? 

 There was not one in the box but the letter from Allem Burdick that came with it (see image) stated; "Enclosed in the package is a power cord and a BNC to RCA adapter for the digital coaxial input". Mine came with the adapter, a USB cable, spare fuses and a couple of pin jumpers. Just looking for clarification here, thanks.




_

 

Jetlag, 

 The DAC1 USB absolutely should have came with a power cord. I apologize for this oversight. We can ship one to you immediately. Please PM your shipping info; also include the source where you ordered the DAC1 USB. 

 Fortunately and unfortunately, my co-workers in production are very human. So, if you don't mind, I'm gonna go jab them a bit about this and tell them that there are robots who are in line for their jobs who won't forget to add a power cord! 
	

	
	
		
		

		
		
	


	




 Thanks, and again, sorry about that!!
 Elias


----------



## EliasGwinn

Quote:


  Originally Posted by *music_man* /img/forum/go_quote.gif 
_i have the dac1 classic plugged into the krell with the xlr's. i am using the front panel variable gain. the internal pads are at the 0db setting. the volume knob is half way(50%). it sounds the best this way and i do not seem to hear any distortion.

 am i actually overloading the inputs on the amplifier this way and i am just not hearing it? or is this ok?

 thanks,
 music_man_

 

Music_man,

 I looked up the specs on your Krell. However, they do not list a Max. Input level. You will have to call or email them to find out.

 You'll need to know this: with 0 dBFS input, 0 dB output attenuation, and volume control at 12 o'clock, the DAC1 XLR outputs will be near 15 dBu. If you turn the volume control all the way up, the DAC1 will be near 28 dBu.

 Therefore, if the Krell can handle a 15 dBu input (probably very close to its limit, based on the other specs I found), you should be ok.

 Please let me know what you find...I'm curious...

 Thanks,
 Elias


----------



## The Monkey

Quote:


  Originally Posted by *EliasGwinn* /img/forum/go_quote.gif 
_The reason the DAC1 doesn't have a power switch is to prevent accidental/inopportune power-down when it is in a critical audio path. Many professional studios and broadcast facilities use the DAC1 in critical audio paths. In these settings, the DAC1 is powered and in-use 24/7, and the accidental (catastrophic!) power-down is eliminated by the lack of a power switch. 

 Thanks,
 Elias_

 


 Many thanks for your informative response Elias!


----------



## music_man

i will try to find out from krell.

 the strange thing is that at that setting the music is not louder than the output directly out of the sony cd player(balanced) with the krells volume in the same position.

 i can turn the dac1's volume to about 70% before i get distortion with the pads at 0. i think that krell has some type of overload input protection i will find out.

 thanks,
 music_man


----------



## jsiau

Quote:


  Originally Posted by *dip16amp* /img/forum/go_quote.gif 
_Is the "UltraLock" more than just "asynchronous upsampling" which some other DACs also have or is there something more to it than a "marketing tool" as mentioned in post #3 and #5 in thread http://www.head-fi.org/forums/showthread.php?t=235830

 Just wanted to know if there is a difference._

 

Great question!

 Benchmark's UltraLock(TM) system includes an ASRC, but the ASRC is just one component part of the total system. The entire UltraLock(TM) system includes:

 1) A low-jitter master clock
 2) A fully-shielded controlled-impedance clock transmission line
 3) 3-dimensional shielding surrounding all digital signals
 4) RF bypass caps connected using low-inductance RF layout techniques
 5) Upsampler with over 100dB jitter attenuation at 1 kHz, 160 dB at 10 kHz
 6) 3 Hz digital PLL corner frequency
 7) Data recovery system that can tolerate > 12 UI jitter
 8) Digital de-emphasis prior to upsampling (when required)
 9) Upsampling ratios selected based upon extensive performance measurements
 10) Soft mute for clean and rapid switching between inputs
 11) Careful management of power distribution and ground returns 
 12) More

 Swapping out the ASRC IC for a pin-compatible substitute will destroy the jitter attenuation of the DAC1. Swapping out the AES/EBU/SPDIF receiver will destroy the jitter tolerance of the DAC1. Changing the D/A conversion frequency will have significant digital filtering implications that have nothing to do with the performance of the ASRC. Replicating the DAC1 schematic without replicating the PCB layout techniques will significantly reduce jitter attenuation and degrade intrinsic jitter.

 We have seen modified DAC1 converters arrive at our facility for repair. These units showed dismal jitter performance due to the insertion of a pin-compatible ASRC. We are also aware of modifications that include replacing the AES/EBU/SPDIF receiver with another IC that lacks jitter tolerance. Other modifications replace multilayer ceramic RF capacitors with audio filter caps in RF portions of the DAC1! These inappropriate capacitor substitutions destroy the jitter performance of the DAC1.


----------



## music_man

mr. gwinn,

 i am playing a 0db test tone at 1khz. the pads are at 0db. i am reading 2.03vrms with the variable volume at 50%. i am reading 8.76vrms with the volume knob at 100%. 2.03v the krell should handle easily. should't it be reading much higher voltage? or does that seem right?

 i was also wondering if the dac1 usb sounds much better than the dac1 classic due to the new high current/low impedance drivers? i originally did not prefer the dac1 classic for enjoyment lsitening. now that i moved to the 0db pads i really do like it. apparently the lower impedance makes an audible difference.

 thanks,
 music_man


----------



## EliasGwinn

The DAC1 USB and the DAC1 Classic will "measure" identically on a test bench, but they _react _differently to different loads. What I mean by this is: for a properly designed load (high impedance, low capacitive load input and moderate length analog cables), they will perform essentially identically. However, with a compromised load, the DAC1 USB will be able to 'shoulder' the load better then the DAC1 Classic.

 For your setup specifically, the KAV400xi has sufficient input impedance (47k), but the manual, just like most (all?) manuals, does not state the capacitance that the source 'sees', so I can't say for sure. If there was a significant capacitance seen at the input, the result would be a high-end roll off.

 Hope that helps.

 Thanks,
 Elias


----------



## EliasGwinn

Music_Man,

 I noticed there was another part of your post which I didn't address.

 The 2.03 Vrms number seems right if you are measuring across pin 3 -> pin 1 or 2. If you measure across pin 2 -> 3, you should get double that amount: 4.06 Vrms.

 4.06 Vrms = 14.39 dBu

 The 8.76 Vrms at full output seems a bit low (~27 dBu). It should be reading closer to 9.5 Vrms (or 19 Vrms across pin 2 and 3) for +28 dBu. 

 Thanks,
 Elias


----------



## EliasGwinn

Quote:


  Originally Posted by *dip16amp* /img/forum/go_quote.gif 
_Is the "UltraLock" more than just "asynchronous upsampling" which some other DACs also have or is there something more to it than a "marketing tool" as mentioned in post #3 and #5 in thread http://www.head-fi.org/forums/showthread.php?t=235830

 Just wanted to know if there is a difference._

 

Music_man,

 I responded to the debate of AES vs. SPDIF, XLR vs. COAX on this thread...

http://www.head-fi.org/forums/showpo...99&postcount=9

 ...in case your interested.

 Thanks,
 Elias


----------



## music_man

thank you mr. gwinn. 

 i was measuring between 1 and 3. my multimeter could be a little off too. it has not been calibrated in a while.

 should i turn the volume down?
 there is no distortion and it is not even as loud as my cd players xlr output.
 i just do not want to go into the first 25% of the volume knob where the channels are uneven.

 the 0db pad does sound better to me, should it?
 when i had the 20 db pad on the volume did not get loud enough at 100%.

 thanks,
 music_man


----------



## EliasGwinn

Quote:


  Originally Posted by *music_man* /img/forum/go_quote.gif 
_thank you mr. gwinn. 

 i was measuring between 1 and 3. my multimeter could be a little off too. it has not been calibrated in a while.

 should i turn the volume down?
 there is no distortion and it is not even as loud as my cd players xlr output.
 i just do not want to go into the first 25% of the volume knob where the channels are uneven.

 the 0db pad does sound better to me, should it?
 when i had the 20 db pad on the volume did not get loud enough at 100%.

 thanks,
 music_man_

 

Music_man,

 I don't know the max. input level of the Krell, but I wouldn't think you need to turn it down. I'd say trust your ears, if you think it sounds ok, then it probably does. If you want to find out for sure, give Krell a call and find out their 'max. input level before clip'.

 Thanks,
 Elias


----------



## audioengr

Quote:


  Originally Posted by *jsiau* /img/forum/go_quote.gif 
_Great question!

 We have seen modified DAC1 converters arrive at our facility for repair. These units showed dismal jitter performance due to the insertion of a pin-compatible ASRC. We are also aware of modifications that include replacing the AES/EBU/SPDIF receiver with another IC that lacks jitter tolerance. Other modifications replace multilayer ceramic RF capacitors with audio filter caps in RF portions of the DAC1! These inappropriate capacitor substitutions destroy the jitter performance of the DAC1._

 


 These are certainly undesirable mods. There are a lot of hacks in the modding business unfortunately, just like there is a a lot of snake-oil in the cable business. They are mostly just trying to do what they believe is right and sounds good to them. I have gotten a dose of both from my competitors in the past.

 Since we are talking about all the positive attributes of your DAC design, then maybe you can answer me one question:

 Why did you ground the input winding of the pulse transformer, eliminating the potential galvanic isolation? Is this a pro-studio requirement?

 Steve N.
 Empirical Audio


----------



## EliasGwinn

Hey folks,

 Just spent the weekend in Philadelphia...what an amazing city!!

 Steve, I'm going to ask John Siau to reply to your question.

 Thanks,
 Elias


----------



## jsiau

Quote:


  Originally Posted by *audioengr* /img/forum/go_quote.gif 
_These are certainly undesirable mods. There are a lot of hacks in the modding business unfortunately, just like there is a a lot of snake-oil in the cable business. They are mostly just trying to do what they believe is right and sounds good to them. I have gotten a dose of both from my competitors in the past.

 Since we are talking about all the positive attributes of your DAC design, then maybe you can answer me one question:

 Why did you ground the input winding of the pulse transformer, eliminating the potential galvanic isolation? Is this a pro-studio requirement?

 Steve N.
 Empirical Audio_

 

Steve this is a great question.

 There is more to the DAC1 design than meets the eye. The input of the pulse transformer has a balanced controlled-impedance connection back to the input jack where it is grounded to the chassis. It is impossible to see this because these traces are fully surrounded by ground planes. All of the input currents are returned directly to the point at which the coax shield connects to the chassis. This chassis ground is not the analog ground for the internal electronics, and RF currents into this ground point cannot flow through the analog ground plane.

 The wiring of the pulse transformer secondary is as important as the primary. The secondary has a balanced connection to the digital audio receiver IC. This provides a well-defined path for the return currents.

 Floating the digital audio input will void the authority to operate the DAC1 under FCC and CE regulations (unless the modification has been subjected to full FCC and CE testing). It is also not legal to sell modified units unless they are tested. Do you have FCC and CE test results for modified units?

 Floating the coax shield usually increases RF emissions as well as susceptibility to RF interference. The digital inputs on the DAC1 are RF connections and need to be treated as such.


----------



## Dan the man

Really confused here. I thought you could simply use the optical Toslink audio out from the Mac to the Toslink input of the DAC 1 without a $300 USB option. WHat is the difference in the way the digital output from the computer is processed? What is the optical input on the DAC 1 for then??

 So does this mean I can't connect the Toslink out from my Mac computer to the presonus CS without another add-on and software to boot????


----------



## Lord Chaos

If your Mac has optical out, that's all you need. No USB option necessary; just connect the cable and go. I have my DAC1 USB connected to a PC using the S/PDIF, and to my Powerbook with USB because it has no optical out.


----------



## audioengr

Quote:


  Originally Posted by *jsiau* /img/forum/go_quote.gif 
_Steve this is a great question.

 There is more to the DAC1 design than meets the eye. The input of the pulse transformer has a balanced controlled-impedance connection back to the input jack where it is grounded to the chassis. It is impossible to see this because these traces are fully surrounded by ground planes. All of the input currents are returned directly to the point at which the coax shield connects to the chassis. This chassis ground is not the analog ground for the internal electronics, and RF currents into this ground point cannot flow through the analog ground plane.

 The wiring of the pulse transformer secondary is as important as the primary. The secondary has a balanced connection to the digital audio receiver IC. This provides a well-defined path for the return currents.

 Floating the digital audio input will void the authority to operate the DAC1 under FCC and CE regulations (unless the modification has been subjected to full FCC and CE testing). It is also not legal to sell modified units unless they are tested. Do you have FCC and CE test results for modified units?

 Floating the coax shield usually increases RF emissions as well as susceptibility to RF interference. The digital inputs on the DAC1 are RF connections and need to be treated as such._

 

Impedance-controlled connection? You mean the wires hanging out in the air soldered to the circuit-board?

 I can understand the need to pass FCC emission testing. This does break the Faraday shield. I did some work with this and some very high-frequency CPU's at Intel Corp in the past. It can be difficult. I suppose if I ever had a complaint about the emissions from my gear interfering with some other equipment that someone might lodge a complaint with the FCC. Very unlikely though. Neve had a complaint yet. I take a lot of care to insure that my transmission-lines are properly terminated and that the impedances and current paths are correct, so that the emissions are minimized. This is even better than shielding, grounding and other band-aids that are typically added after the fact. Even though the Farady shield is not perfect, the emissions are still low.

 I mod lots of different DAC's. Some of them do have the input winding floating, except for a .1uFd cap to ground on the low-side. Evidently some were able to pass CISPR, CE and FCC emissions testing or maybe they just dont care about emissions. This still does not maintain the Faraday shield, however it could. If the connector were isolated, but surrounded by a ring of small SMT capacitors that each connect to earth ground, then the FS would be intact, without actually DC-grounding the BNC connector. This would be the best of both worlds, galvanic isolation to minimize gorund-loop noise and shielding to minimize emissions.

 The problem with emissions testing is that most companies typically use the worst, cheapest digital cable they can find and do what they have to do in order to pass with this cable. All cable connections must be connected and running for the test.

 This is an unfair goal IMO. It causes companies to compromise their designs, usually adding jitter or potential ground-loops in order to accomodate a cheap, poor performing cable, or perhaps a bad impedance mismatch etc...

 My strategy is to force the customer to use my cable and my terminations. I have control over both ends of the transmission-line, the source and destination. This way I can truly control the reflections on the transmission-line and the current flows. It's the reflections and overshoots that are usually the emissions culprits.

 Here is another question for you, since you brought up impedance control and wrote "The digital inputs on the DAC1 are RF connections and need to be treated as such":

 Why did you use a 50 ohm characteristic impedance BNC connector for the S/PDIF input, which is specified at 75 ohms?

 Do you feel that the edge-rates are slow enough that the absolute impedance is not important?

 Steve N.


----------



## Rivendell61

Quote:


  Originally Posted by *audioengr* /img/forum/go_quote.gif 
_Here is another question for you, since you brought up impedance control and wrote "The digital inputs on the DAC1 are RF connections and need to be treated as such":

 Why did you use a 50 ohm characteristic impedance BNC connector for the S/PDIF input, which is specified at 75 ohms?

 Steve N._

 

Dan Lavry tried to explain that to you last year on his forum.
 Here are some Lavry quotes:

 "....RCA or just about anything out there....is somewhere between say 25 Ohms and 180 Ohms....that one inch discontinuity at the middle of a line with a signal of say 5 nsec rise time is virtually a non issue."

 And more Dan lavry: 
 "I agree that an RCA connector, (if it is far from 75 Ohms impedance), would be a disaster for 10GHz digital signal transmission. It is certainly NOT a problem for say audio signals because they are too slow. .....AES and SPDIF signals are still way too slow to care about the impedance of an RCA connector."

 "An SPDIF signal DOES NOT have spectra to GHZ."


 Also--Here on HeadFi Mr Siau has addressed the question in this post:
http://www.head-fi.org/forums/2414299-post3.html


 Mark


----------



## audioengr

Quote:


  Originally Posted by *Rivendell61* /img/forum/go_quote.gif 
_Dan Lavry tried to explain that to you last year on his forum.
 Here are some Lavry quotes:

 "....RCA or just about anything out there....is somewhere between say 25 Ohms and 180 Ohms....that one inch discontinuity at the middle of a line with a signal of say 5 nsec rise time is virtually a non issue."

 And more Dan lavry: 
 "I agree that an RCA connector, (if it is far from 75 Ohms impedance), would be a disaster for 10GHz digital signal transmission. It is certainly NOT a problem for say audio signals because they are too slow. .....AES and SPDIF signals are still way too slow to care about the impedance of an RCA connector."

 "An SPDIF signal DOES NOT have spectra to GHZ."


 Also--Here on HeadFi Mr Siau has addressed the question in this post:
http://www.head-fi.org/forums/showpo...68&postcount=3

 Mark_

 

Mark,
 I will have to differ with Mr. Lavry, who I have met and discussed many things with. I consider him a peer, as I do Mr. Siau. There is significant spectra approaching GHz in a proper S/PDIF signal (one that generates low-jitter at the receiver). Just display the spectra on a RF spectrum analyzer and you will see it.

 Many DAC's have significant filtering to slow the S/PDIF edges which does reduce this spectra. This is a bad idea IMO. Faster edges result in less jitter at the receiving device, so this is more desirable. You do have to take more care in terminating the transmission-line properly and impedance matching, but this is not rocket-science, at least for me. I have 30 years experience with high-speed digital interfaces, from IBM peripheral buses to massively parallel supercomputers. Rise-times of 2 nsec on cables are not unusual in these systems.

 What you must understand is that no one designer has infinite knowledge of all aspects of these designs. Some are strong in digital, some in analog. Few have much experience in power delivery, ESD, grounding and shielding and even less have much experience in HF digital transmission line effects and termination techniques. I am strong in certain areas and these designers may be strong in other areas. I would not try to compete with John Curl for instance to design a power amplifier circuit. However, I can improve the current paths and power delivery to his amp and make it sound even better. This is why I have found a niche in the modding business.

 I find all kinds of crazy things in some DAC's that I have modded. Some designers have limited experience and therefore just copy other designs which also may be non-ideal. I have found poor implementations of USB interfaces and S/PDIF interfaces in lots of products, even from large companies like Sony. I'm not your typical modder that swaps parts. I have extensive experience in design, so I reverse-engineer most designs to expose the weaknesses.

 The point is: there are no experts in everything. We are each experts in certain areas. This is why collaborations often lead to superior products.

 Steve N.


----------



## Crowbar

Jocko and others on the diyhifi forums actually performed measurements that show there is definitely a concern with proper connector impedance. Instead of arguing, let's see some properly conducted time-domain reflectometer experiments and settle this.

 Of course, the true solution is drop synchronous modes altogether. There's a reason the USB Audio standard supports asynchronous endpoints where the DAC provides flow control, and thus interface jitter becomes a non-issue without any ASRCing and its rate-estimation problem. The latter is explained by Bruno Putzeys in another forum:
  Quote:


 Any instabilities in the ratio signal will get encoded as phase modulation into the output. Such instabilities may stem from jitter. The whole jitter attenuation capability of an ASRC hinges on the low-pass filter. The jitter attenuation characteristic equals the frequency response of the ratio estimator's post filter. This filter fulfils the same function as the loop filter in an analogue clock recovery PLL. The advantage of the ASRC is then that you need only one crystal oscillator to cover all sampling rates.
 Of course the ratio estimator, being an implicit ADC, suffers from quantisation. The phase between the two clocks is quantised to a time span equal to 1 period of the highest frequency clock in the chip (sometimes a multiple of the output rate, sometimes a separate master clock signal). This error is added to the input jitter before being attenuated by the lowpass filter. 
 

Note that it is clear from the above that rate estimation is not a purely digital process.


----------



## Jetlag

A quick round of kudos to both Elias Gwinn and Rory Rall at Benchmark. After being informed that my DAC1 USB had arrived sans power cord they sent me one in just 4 business days! Excellent CS and much appreciated, thanks fellas!

 (BTW, it wasn't necessary guys but thanks for the Nordost Valhalla, I really was only expecting a stock cord. 
	

	
	
		
		

		
			





 )


----------



## EliasGwinn

For those following this discussion, I will quote John Siau's previous response to Steve's question (thanks for the link Rivendel61!!)

  Quote:


  Originally Posted by *jsiau* /img/forum/go_quote.gif 
_The connector is 50-Ohms. 50-Ohm connectors are far more durable than 75-Ohm connectors due to the extra dielectric material surrounding the center pin of the BNC. For this reason, it is common practice to use 50-Ohm BNC connectors in 75-Ohm systems when the signal bandwidth allows it. 

 The short interruption of the 75-Ohm transmission line is only significant for frequencies that are much higher than any contained in a digital audio signal. The 50-Ohm connector would be a factor for signals having a wavelength of 2 inches or less in coax (about 3 GHz). A 192 kHz digital audio signal transmits data using a clock that is 64 times the sampling frequency (192 kHz * 64 = 12.288 MHz). 3 GHz is the 244th harmonic of 12.288 MHz and does not exist in a 192 kHz digital audio signal. If it did, the box probably would not pass FCC and CE emissions tests.

 Changing the connector would reduce the durability of the product and would have absolutely no effect even at 192 kHz._


----------



## EliasGwinn

Quote:


  Originally Posted by *Jetlag* /img/forum/go_quote.gif 
_A quick round of kudos to both Elias Gwinn and Rory Rall at Benchmark. After being informed that my DAC1 USB had arrived sans power cord they sent me one in just 4 business days! Excellent CS and much appreciated, thanks fellas!

 (BTW, it wasn't necessary guys but thanks for the Nordost Valhalla, I really was only expecting a stock cord. 
	

	
	
		
		

		
		
	


	




 )_

 

Jetlag,

 Thank you for your graciousness and patience with our oversight. I'm glad we could straighten everything out.

 Thanks,
 Elias


----------



## audioengr

Quote:


  Originally Posted by *EliasGwinn* /img/forum/go_quote.gif 
_For those following this discussion, I will quote John Siau's previous response to Steve's question (thanks for the link Rivendel61!!)_

 

Thanks for the explanation. I have been putting 75 ohm BNC's on DAC's for years and never had any damage yet, however I too dont believe that the edge-rates are fast enough to warrant this precise impedance matching, particularly for stock consumer gear. 

 My modded gear is a different story. I get very high-speed edge-rates (not fundamental frequency) on my S/PDIF and so the spectra goes much higher. You can have a 100Hz digital signal with sub-1nsec edge-rate that will have significant GHz content. This is the difference between RF and digital transmission. Talking about the fundamental frequency without specifying the risetime is certainly incomplete. They are totally different animals due to the non-sinusoidal waveform of digital.

 I suppose this is like any other detail in audio. It is a matter of what is audible. There is certainly a lot of debate on this and I'm personally not convinced that even an RCA connector makes an audible difference in a typical stock piece of gear. There is some magnitude of reflection with all impedance discontinuities. The question is whether it is audible or not...... 

 I believe the answer is:
 It depends on a lot of other factors, the edge-rates, the cable impedance, the terminations etc...

 Steve N.


----------



## music_man

so once you modify the gear, material that normally only birds can hear suddenly become audible to humans? i want it!

 music_man


----------



## audioengr

Quote:


  Originally Posted by *music_man* /img/forum/go_quote.gif 
_so once you modify the gear, material that normally only birds can hear suddenly become audible to humans? i want it!

 music_man_

 

I'm not sure what this comment means, but yes, the HF extension, dynamics and clarity is improved dramatically. Improving power delivery, signal and power paths and reducing jitter accomplishes this. You can read the feedback from my customers on audiocircle, audiogon or asylum.

 Do birds actually hear HF better than humans? I know dogs do...

 Steve N.


----------



## Crowbar

Quote:


 The inner ear actually has nerve receptors for frequencies up to about 80 kilohertz. 
 

Source: US patent 5479168

 Peer reviewed study showing subconscious perception of post-20 kHz sound:
 Inaudible High-Frequency Sounds Affect Brain Activity: Hypersonic Effect
 The Journal of Neurophysiology Vol. 83 No. 6 June 2000, pp. 3548-3558

 Though we now have 96 kHz sampling rates so digitally 40 kHz sounds should be doable, I'd love to see, in light such information such as what I've referenced above, what excuse DAC makers have about analog filters starting attenuation pretty close to 20 kHz. The lack of higher bandwidth recordings is not an excuse, since it's the other way around: no one would make such recordings unless playback devices supported them.


----------



## music_man

as far as i know most adult humans can not even detect 20khz during the standard hearing test audiologists use.

 i am neutral on the subject of modding. i can see why the manufacturers would frown upon that though. for a lot of reasons. thats all i have to say on that subject.

 music_man


----------



## Jetlag

Is it just me or did anyone else notice that the title of this thread was "Benchmark DAC1 now available with USB" and not a sponsored promotional thread for advertising modding services?

 Elias has offered an exceptional _free_ tool for all of us computer audio fans by creating a WIKI dedicated to maximizing the performance and quality of our audio playback. John has also gone above and beyond by answering many of the more technical questions and IMOHO has done a very thorough and accurate job of doing so. Rory also helped me out by providing great customer service.

 I am most definitely an AV enthusiast but because of my tight budget there is no way I can spend over $1K on an item and then voluntarily terminate the manufacturers support for that product. I specifically chose the DAC1 USB not only because it is such a carefully designed, well supported and very highly regarded product, but also for the reasons outlined above. Personally I would never consider voiding the warranty but your own opinions may vary. 

 No offense intended, but I think if you want to promote your modding services there is an established 'members of the trade forum' just for that purpose. Just my $.02 of course and I hope this isn't considered inflammatory. Thus far this thread has been a cornucopia of useful information for DAC1 USB fans/owners with a rarely seen level of manufacturer participation. I'm not a Benchmark employee or rep, just a satisfied owner who would prefer to see this valuable thread stay on topic. My sincere apologies if I offended anyone by my comments.


----------



## lowmagnet

Quote:


  Originally Posted by *Jetlag* /img/forum/go_quote.gif 
_No offense intended, but I think if you want to promote your modding services there is an established 'members of the trade forum' just for that purpose._

 

x2


----------



## gregeas

I finally received my USB DAC1 this week. So far I've done a bit of listening in my main system, comparing the DAC1 to my Slimdevices Transporter (the Transporter is being used to send a digital signal via BNC coax to the DAC1). It's not obvious at this point that one or the other is lacking or superior, but more listening will follow this week. 

 I bought the USB DAC1 to be used primarily at work and also in my other systems when I want to add a high-end PC source. I love the fact that I can carry the DAC1 and my XPS 1210 laptop (with 300 lossless albums) in my small laptop bag. 

 I do have two questions for Elias. 

 First, the Wiki entry for setting up iTunes for XP states that "Word-lengths of longer then 16-bit are truncated to 16-bit, rather then being dithered. 
 Truncation will cause severe distortion." Does this mean that iTunes should be avoided in preference of another playback program? If so, which is optimum for XP? I have Jriver Media Center installed, but it doesn't look like this has been tested by Benchmark. 

 Second, and this may be stupid, but does the BNC cable twist into place onto the jack? This is how BNCs work for video, but I can't get my new cable twist into place, and I don't want to force things. 

 Looking forward to spending more time with the DAC1!

 PS: The Transporter was good enough to force a quick sale of my $2.5k Arcam CD player. In fact, for reasons I can't explain, the Transporter crushed the Arcam in my system (which is otherwise all Arcam components, with PMC speakers).


----------



## gregeas

EDIT: A restart solved the iTunes issue below, but I'm still curious if I'm sacrificing quality due to the bit truncation.

 * * * * * * *

 I have another question. For some reason iTunes won't output to the DAC1 and instead plays through my laptop speakers. This is despite setting the output to the DAC1 in the Sounds and Audio Devices control panel and in iTunes. Weird. 

 Anyhow, I'm trying to use Jriver Media Center as a temp solution before I try Foobar, which will take some time to set up. 

 My question is what output mode I should use: Media Center gives options for Asio, Wave Out, and Direct Sound. Asio is what I have used in the past to get (alleged) bit-perfect output. I am aware that I should set output depth to 24 bits and turn off all audio processing. Anything else?


----------



## Lord Chaos

I'm not Elias 
	

	
	
		
		

		
		
	


	




 but I do have two DAC1s. The BNC connector is twist-on bayonet. It helps if you push to compress the disc spring that's inside the BNC connector while you twist it onto the bayonet mount.

 The word-length issue is discussed earlier in this thread. It's not a problem unless you use Itunes' Equalizer, or set the volume for less than 100%. Windows, according to Elias, has a better volume control algorithm so you can use it as you want.


----------



## Dan the man

I am interested in a high end Apple Lossless MacPro computer to DAC to headphone amp to headphone setup with eventual use as a source for a high end two channel sound room setup with Wilson speakers and tube amps.

 There are four DACS that I am interested in regardless of price point.

 1. Personus Central Station
 2. Benchmark DAC 1 w/ or w/o USP
 3. Lavry da10
 4. Slim Devices Transporter

 I want to have the ability to use the computer via Toslink or Ethernet or Airport to get the highest and most stable output. Right now I have balanced AKG 701 and soon the Grado Statements. Can I use two tube amps such as the Doge 6210 for balanced tube headphone listening?

 How can this type of setup be achieved and what cables, software and settings on the Mac using itunes is needed?

 Do I need an external hard drive to store the applelossless files and how much RAM in the MAC is needed to get optimal buffering for playing entire albums/redbook CD data copies?

 I was also thinking of buying the OPO DigitalDV-970HD to be used as a transport for playing CD and SACD and feeding the digital output to one of the above DACS.

 My understanding is that the Transporter is the only one without built-in headphone amps but is capable of using its wireless interface to receive music from multiple computers on a LAN.

 Which DAC would give the best and most reliable SQ using the computer as the primary music source?


----------



## zheka

just finished reading the thread. 
 easily the most informative reading on the subject of computer based audio i ever came across. thanks to Benchmark folks for time and effort. I bookmarked the wiki pages since i am likely to visit them frequently


----------



## lowmagnet

Quote:


  Originally Posted by *Dan the man* /img/forum/go_quote.gif 
_I am interested in a high end Apple Lossless MacPro computer ..._

 

Could you start a new thread and replace your post here with a "."? This isn't really the place to start into these questions. I'd be happy to reply to the thread too, since I'm a Mac ALAC user with a DAC1


----------



## gregeas

Product suggestion for Benchmark:

 Make a version of the DAC1 with the XLR outputs optimized for balanced headphones. I imagine the form factor of the DAC1 could stay as is.

 I'm not a betting man, but if I were, I'd say that this product would be a huge hit with this crowd... I imagine that many could forego the XRL line outs...and use the RCAs for non-headphone listening. 

 I, for one, would buy this product tomorrow. I agree with others here who have said 2007 is the year of balanced headphones...


----------



## EliasGwinn

Quote:


  Originally Posted by *gregeas* /img/forum/go_quote.gif 
_First, the Wiki entry for setting up iTunes for XP states that "Word-lengths of longer then 16-bit are truncated to 16-bit, rather then being dithered. 
 Truncation will cause severe distortion." Does this mean that iTunes should be avoided in preference of another playback program? If so, which is optimum for XP? I have Jriver Media Center installed, but it doesn't look like this has been tested by Benchmark. _

 

iTunes can play 16-bit audio perfectly (bit-transparently) if the volume control is kept at 100%, and all DSP options are disabled (such as 'Sound Enhancer'). 

 If you want to control the volume through software, it is recommended to use the Windows Volume Control Mixer (the built-in, system-wide volume control).

  Quote:


  Originally Posted by *gregeas* /img/forum/go_quote.gif 
_Second, and this may be stupid, but does the BNC cable twist into place onto the jack? This is how BNCs work for video, but I can't get my new cable twist into place, and I don't want to force things. _

 

The BNC cable should be able to twist into place. I assume you are using an actual BNC connector, not a cable-TV-style coax connector, or a TNC (threaded) connector. The BNC connector will have a little slot that fits over the stud on outside of the jack. After it slips into place, twisting it clockwise should guide the stud along the slot to lock the connector in place. As Lord Chaos said, try applying some slight pressure along its axis if needed.

 Let me know if it is working for you. There is a possibility of a mis-machined jack, although we haven't seen that happen yet, to my knowledge.

 Thanks,
 Elias


----------



## EliasGwinn

Quote:


  Originally Posted by *gregeas* /img/forum/go_quote.gif 
_Product suggestion for Benchmark:

 Make a version of the DAC1 with the XLR outputs optimized for balanced headphones. I imagine the form factor of the DAC1 could stay as is.

 I'm not a betting man, but if I were, I'd say that this product would be a huge hit with this crowd... I imagine that many could forego the XRL line outs...and use the RCAs for non-headphone listening. 

 I, for one, would buy this product tomorrow. I agree with others here who have said 2007 is the year of balanced headphones..._

 

Gregeas,

 We appreciate suggestions and recommendations, and I will share it with the product development team.

 We have looked into the balanced headphone technology, and have engaged in discussions with other folks about it (see the following thread):

http://www.hydrogenaudio.org/forums/...howtopic=53801

 I agree that a DAC1 with balanced headphone outputs would be a good seller. However, if I can be honest, we aren't so sure that a balanced headphone configuration provides any performance advantages whatsoever. As much as we love to see new advancements in technology, no one has been able to articulate how balanced headphones provide a viable advantage. We are very open to be persuaded otherwise, but so far, we have not found any convincing explanations. If you know of any publications that may assist our research, please let us know.

 We may eventually build a test circuit to measure and listen to the differences....or perhaps purchase a balanced headphone amp from another company for the same purposes. 

 Thanks,
 Elias


----------



## EliasGwinn

Quote:


  Originally Posted by *Dan the man* /img/forum/go_quote.gif 
_I am interested in a high end Apple Lossless MacPro computer ..._

 

Dan,

 I see you have a lot to consider here. I could discuss this over the telephone if you were interested, but I don't think this thread can handle that much of a diversion. If this thread were Jenga, I'd be scared to breath next to it. 
	

	
	
		
		

		
		
	


	




 Thanks,
 Elias


----------



## gregeas

Quote:


  Originally Posted by *EliasGwinn* /img/forum/go_quote.gif 
_iTunes can play 16-bit audio perfectly (bit-transparently) if the volume control is kept at 100%, and all DSP options are disabled (such as 'Sound Enhancer'). 

 ... 

 Let me know if it is working for you. There is a possibility of a mis-machined jack, although we haven't seen that happen yet, to my knowledge._

 

Thanks for the info. Just to confirm, I do leave the bit depth set to 24 bits on iTunes, correct? (This is what the wiki advises.) So the 24-bit truncation in iTunes only has adverse effects on native 24-bit audio, such as a high-res master recording, right?

 As for the BNC, I think the terminations on the cable are to blame. They slip over the pins on the jack, but I have a tough time twisting them into place. I'll email the cable manufacturer to see if he has any ideas. 

 Regarding balanced headphones, I should state that I've never heard them. But judging from the legions of fans here, it's hard the imagine that the balanced improvement is a purely psycho-acoustic effect.


----------



## Jetlag

Quote:


  Originally Posted by *gregeas* /img/forum/go_quote.gif 
_...a purely psycho-acoustic effect._

 

Psycho-acoustic effect? In Audio/Video? No way, it _can't_ be!


----------



## Lord Chaos

Quote:


  Originally Posted by *EliasGwinn* /img/forum/go_quote.gif 
_If this thread were Jenga, I'd be scared to breath next to it. 
	

	
	
		
		

		
		
	


	


_

 

My co-workers are wondering why I'm laughing in my office... 
	

	
	
		
		

		
		
	


	




 And Elias, I have a question for you. I've checked my DAC1 USB and the jumpers are in place for the headphone output gain reduction. I still can only get to the fourth or fifth detent with the volume control when using my E500s. Is there a good way to control the volume in an Apple Powerbook/Itunes system?


----------



## music_man

hi mr gwinn,

 i am still having a litle issue with the krell. to get the dac1's output to be as loud as the sony cd players output i have to run the dac'1 at 5.66 volts. at 5.66 volts there is a small amount of distortion present in the tweeters. the sony is only putting out 4.63 volts. i measured them both. i don't understand this. how is this possible? shouldn't the same voltage from two diferent products into the same input cause the same db reading at the same distance? if i have to i could just back off on the volume. not a big problem. i would wish you could explain this though.

 also i was hoping you would comment on this. some people say the dac1's headphone amp is just an "add on" and it was just put there as an extra item. they say it is not so good sounding but it was not intended to be. i think it sounds worth $300 by itself. i know you sell it for about that by itself. i think people that do not like it simply are presenting the usual subjective opinions. it may not be everyones favorite but i certainly do not find it to be unworthy.

 i assume benchmark intended for the headphone amp to sound the best it could with in reason, and to compete with other comparably priced amps. is that correct?

 thanks,
 music_man


----------



## Lord Chaos

The headphone amps in my DAC1s don't sound at all like add-ons. I'm delighted with them every time I listen, and I've been using headphones since the early 1970s. This is the best I've ever had sound.


----------



## lowmagnet

Quote:


  Originally Posted by *Lord Chaos* /img/forum/go_quote.gif 
_And Elias, I have a question for you. I've checked my DAC1 USB and the jumpers are in place for the headphone output gain reduction. I still can only get to the fourth or fifth detent with the volume control when using my E500s. Is there a good way to control the volume in an Apple Powerbook/Itunes system?_

 

36Ω is really low impedance. 119dB SPL/mW is too ear-bleedingly loud, too fast. Putting those into your ears is proably considered a form of suicide in some states 
	

	
	
		
		

		
		
	


	




 You've really got no choice but to use software volume control, methinks.

 Believe me I feel your pain, my Grados are too loud too.


----------



## lowmagnet

Quote:


  Originally Posted by *music_man* /img/forum/go_quote.gif 
_also i was hoping you would comment on this. some people say the dac1's headphone amp is just an "add on" and it was just put there as an extra item. they say it is not so good sounding but it was not intended to be. i think it sounds worth $300 by itself. i know you sell it for about that by itself. i think people that do not like it simply are presenting the usual subjective opinions. it may not be everyones favorite but i certainly do not find it to be unworthy.

 i assume benchmark intended for the headphone amp to sound the best it could with in reason, and to compete with other comparably priced amps. is that correct?_

 

I'm confused by most people's statements on this subject myself. I just wrote it off as people not knowing what the heck they're talking about, but I'm willing to hear them explain why the DAC1's headphone jack is not good to them. Maybe in a separate threatd, though, Jenga that this one is.


----------



## Tassie Devil

Hi Everyone

 New here & just joined. Noticed this thread so thought to contribute. I have some really top new toys
	

	
	
		
		

		
		
	


	




 - here is a review 

 John

 Review of the Ultra Turbo Mod on the Benchmark DAC-1 by Steve Nugent at Empirical Audio + Esoteric VRDS transport

 This review has been edited quite a few times to correct preliminary observations which later proved to be untrue. Jumping to conclusions is a dangerous trap and one I fell into because of not allowing the newly installed gear long enough to bed in. I initially made the incorrect assumption that because the Esoteric UX1 LE http://www.teac.com/esoteric/UX-1LTD.html player was used, it was ready to judge after only a short warm up time. The Esoteric reviews stated how a minimum of 200 hours burn in time was required but my feeling now it should be changed to “warm up time” and I guess the same applies to the newly modded Benchmark DAC1 from Empirical Audio. http://www.empiricalaudio.com/ The quality of the musical sound coming from this combo has improved heaps over the last 2 weeks.

 Players discussed here are:
 1.Esoteric UX-1 Limited Edition with much admired VRDS transport – a universal player but the video circuitry can be turned off;
 2.Marantz SA-17 SACD player - highly modded with new clocks, black gate caps etc by James at Soundlabsgroup in Melbourne Australia.

 The DAC under discussion has been modded 2X by Steve Nugent at Empirical Audio, first with the turbomod available in 2006 and now with a further upgrade which involved a rewiring & removal of an opamp stage. I use a Meridian 861 for processing surround sound and for feeding synthesised surround from 2 CH into the non stereo channels.

 All L/R analog feeds discussed in this article go direct to a McIntosh C200 preamp -> Halcro DM68 amps -> Sound Labs “Majestic” full range electrostatic speakers. Connection between players and the DAC is via high quality RCA S/PDIF cabling (DH labs Silver sonic http://www.audioc.com/accessories1/dhlabs/d75.htm ) while the analog feed out from the Benchmark is via Kimber balanced. The stereo feed out from the players to the preamp (using their internal DACs) is via Siltec cabling. I feel the electronics coupled to those magnificent Sound Labs speakers [amps immediately behind them] are a first class test bed to demonstrate how good (or bad) the front end player/DAC gear is.

 The Benchmark DAC-1 has received high praise in its factory form by the guys at Stereophile but Steve Nugent’s modifications take it to a much higher level of excellence. I have lived happily with the earlier turbomod in the system here for some time. It was very good and was slightly preferred to the analog out of the Marantz but the latest mod takes digital to analog conversion (total cost of this mod around $USD2850) up to another level. 

 I should emphasise a number points about all this:

 1.The items under review must be left powered on long enough to settle down or false conclusions can be drawn;
 2.Preliminary thoughts were that observed differences were subtle and might not be so clearly observed on less resolving systems – the “weakest link” in the chain syndrome applying with a vengeance, but after the items had been left on for more than a week my mind has been changed and the differences involved with the Esoteric and newly modded DAC1 are very obvious and positive;
 3.The weakest link in the chain here is the software – huge differences in resolution etc are evident;
 4.Auditory memory is most unreliable and this makes subtle differences less easy to detect;
 5.Small differences in volume make comparisons difficult and can skew conclusions.

 The musical quality from the modded Marantz coupled to the earlier Benchmark turbomodded DAC-1 was very good, if a little clinical and that turbomodded DAC-1 was slightly preferred to the output from the internal DAC of the Marantz. Not so now, the latest mod by Steve Nugent blows his earlier effort away completely. The side to side, front to back and soundstage resolution is stunning and far exceeds anything I ever expected to hear from CDs. It has “life” to it. Yes, some of that can be thanked to the Esoteric player, but not all because reverti to the direct analog out (direct to preamp and speakers) from the player using its internal DAC and the sound is relatively flat and a lot less involving. It is good, but lacks the life and space around the music that is wrought with the latest Empirical Audio modded DAC-1. Both are streets ahead of the Meridian 861 processor with the sound using it as the4 DAC coming out as very woolly and ill defined.

 Now to try to get the Esoteric into perspective. Reading all the hype about the VRDS transport mechanism would have one believing it is the best thing invented since sliced bread. Well, one cannot help but admire the engineering of it but the bottom line is how it affects the audio quality and here my initial conclusion was that there was no dramatic chalk and cheese difference between it and the Marantz. I happen to have inadvertently purchased two identical CDs so, a day after getting the Esoteric both were played simultaneously in the Esoteric and the Marantz. Steve Nugent has rebuilt an Inday digital switcher (he was horrified at the original design) so it was very simple to switch between these two sources with both fed into the updated DAC-1. The beauty of this arrangement was that levels were identical and A/B comparison was a simple button switching affair. My wife and I listened very intently and she ultimately decided (without knowing which was playing) that the Esoteric was better, but not dramatically so. 

 However, a few days later, a friend with younger and more critical ears, compared the two players and his verdict was the sound from the Marantz was muddy and ill defined in comparison. He looked to see what major change had been made to the system and was surprised it was only the player which had been altered. He labelled the Marantz as “faulty” by comparison! In tandem with this I have altered my previous appraisal that the sound quality between the two players is similar. It obviously is not and the first comparison reported above reflects the mediocre quality of the CD used + there has been more time for the Esoteric to settle in to the system - it has been left permanently switched on since arrival. Traps for young and old players (pun intended)!!

 So yes, the improvement, on good software, is very evident. Digital sources are frequently and justifiably criticised as sounding edgy but the latest mod has removed that. The harpsichord is a particularly difficult instrument to reproduce well and often comes through with a raspy twang to it. Not so via the Esoteric/Empirical Audio modded Benchmark. Comparing that sound to Esoteric’s own internal DAC there is no twanginess there either but the sound was more recessed and less open and not as musically appealing as with the modded Benchmark. One CD my wife loves is that which has Neil Diamond singing “Song Sung Blue” (I guess Mozart lost any claim on copyright for the plagiarism involved). Previously his voice had a slight rasp to it but that has now gone. Similarly, the excellent “Cantus Let Your Voice be Heard” engineered by John Atkinson, previously had an edge to it. It is now gone and the illusion of the singers being in the room is thrilling. Every instrument and every voice is clearly and clearly defined. Having collected most of the standard classical repertoire (and repeated a lot of it) I like to purchase unknown (to me) works. In the last batch of acquisitions is an EMI classics “Panufnik conducts Panufnik” [0946 3 52289 2 2] and the audio quality on this is hair raising. When I looked at the sleeve notes it was amazing to see it was originally released in 1967 and 1975. So, I guess it was a simple recording without the forest of microphones going into 30+ channel mixers that have been the delight of engineers in more recent times. 

 In a word the audio is now more musical with quite incredible front to back and side to side soundstaging which makes multichannel SACD less appealing as it relies more on 6 speakers to give the same sort of life to the music. So, now both units have settled down I find 2CH CD via the modded DAC-1 exceeds the resolution and clarity of SACD on good software. And that is the rub with CDs. IMHO there is more care spent on engineering SACDs than on most CDs so it is unsurprising SACDS generally sound better in systems. But, this is no longer the case here. Maybe hard for people to believe, but the Steve Nugent modded Benchmark DAC-1 makes CDs sound better than SACDs!!!! 

 Bottom line – it is only after 20 years that the potential of the CD format is being realised. So folks, if you want to tap into this potential and have a big enough budget for it, purchase a Benchmark DAC-1 and send it to Steve Nugent for the ultimate massage. IMO, given a reasonable input (which could mean a clock etc upgrade on the player), the quality of the DAC is critical to the quality of the signal emerging in analog form, and the Empirical Audio modded Benchmark DAC-1 yields that quality in spades. I was starting to feel it was time to abandon CDs and only collect SACDs in future. With the caveat that the engineering on CDs varies a lot, I’ll continue to buy them as the library is so huge. But no system can turn an audio pigs ear piece of software into silk ear, on the contrary, a revealing system reveals the flaws just as much as it can reveal hidden glories. 

 I can forsee some reading this will mutter about me being just another kid, happy with his new toys and will not accept that (some) CDs sound better here than SACDs. Maybe the sound is “different” but that does not make it better etc etc. Well yes, I am happy with the latest upgrades and yes, it does sound different, but that difference is an improvement is by a wide and very obvious margin and is not the shift sideways that I will admit I have been guilty of in the past. Full marks to Steve Nugent at Empirical Audio for his latest upgrade and yes, hats off to TEAC for a very good player in the UX-1 LE. And my feeling is that SACDs could sound much, much better with better processing than I’m getting from the Esoteric (and I got from the Marantz) but with the paranoia about copy protection that is presently in place, modding of the digital SACD stream does not seem to have been attempted yet, or has it? That is a challenge for Steve Nugent!!!

 So, have I come to the end of the audio upgrade lunacy? I would like to think so, but then I proclaimed that last year .

 John

 .


----------



## Jetlag

Any chance you would be willing to send it to Benchmark for a few days so they could thoroughly bench test it and see how it compares to a stock DAC1 via a precise technical evaluation? I would love to see the two charted side by side.


----------



## slwiser

Quote:


  Originally Posted by *Jetlag* /img/forum/go_quote.gif 
_Any chance you would be willing to send it to Benchmark for a few days so they could thoroughly bench test it and see how it compares to a stock DAC1 via a precise technical evaluation? I would love to see the two charted side by side._

 

2x


----------



## kool bubba ice

Quote:


  Originally Posted by *Jetlag* /img/forum/go_quote.gif 
_Any chance you would be willing to send it to Benchmark for a few days so they could thoroughly bench test it and see how it compares to a stock DAC1 via a precise technical evaluation? I would love to see the two charted side by side._

 

Or they could just listen to the difference


----------



## Jetlag

If the audible differences are great it is something I would be interested in pursuing as long as I could justify it budget wise. I would also assume that a significant audible difference would be easily quantified via testing with laboratory grade test gear which Benchmark has plenty of.

 I spent lots of time getting the acoustics right in my last listeneing room using very good quality test gear, I would consider this doing the very same thing. I'm all for improved sound quality (well as long as I can afford it that is).


----------



## Tassie Devil

Hi Guys

 I can understand why you ask the questions you do but no, I'm not parting with it for Benchmark to test!!!

 But trust me, it is a remarkable break through for CD. I was a digital Luddite for 2 decades and loved my vinyl (now sold). However got I sucked into digital through laserdisks (long story I'll not bore you with).

 I tried out various processors from Theta through to Meridian 861 (which I still have). The first Benchmark purchased 12 months ago sounded ok but not spectacular. Nugent posts as "Audioengineer" on the digital Board of Audio Asylum (where I moderate the Music Board) and I followed up with a mod of the DAC 1 with him. Good but not spectacular. I then purchased a second DAC 1 and had it sent direct to him for a tubo mod. Result was much better. Both the unmodded and the turbo modded units were tested by some guys in the Melbourne Audio Club (I'm a member but was O/S at the time) and they voted the turbo as clearly better.

 Then a month ago Steve excitedly contacted me and said he had made a breakthrough. Result was the turbo modded unit was sent for a rebuild and that is the one I'm raving about.

 Trust me, I've been listening to recordings for 60 years (first was on a wind up Edison in the attic
	

	
	
		
		

		
		
	


	




 ) and this DAC, combined with the Esoteric is giving stunning results. I honestly never believed music could sound so good and, I'm not kidding when I say it sounds better than my SACDs.

 Now you might have guessed that I live in a very remote (but beautiful) part of the world so have mostly only experienced what I have purchased - there are no longer any genuine hi-fi stores in this State and pitifully few in Melbourne across the water. It is now all home theatre (which I dio not despise). And this very remoteness makes ideas like lending the modded DAC 1 to Benchmark rather impractical.

 But I'm seriously thinking about getting the other DAC 1 also modded for the system in the home office here. I might let the Melbourne Audio Club try that one out.

 Yes, I'm an incurable audio-video nut, but a happy one 
	

	
	
		
		

		
		
	


	




 John


----------



## music_man

this guy just shows up on this board. after we just got done complaining about this thread turning into a sales pitch for modders. i don't know about you guys but i am smelling some bs here. if i am wrong i do apologize. seems sketchy though.

 music_man


----------



## CanMad

Possibly it's a little bit off topic. But my guess would be that he is sincere, and just trying to offer some helpful experience. However the improvements would have to be pretty significant to justify spending more than double the cost of the original DAC1 IMO. 

 On topic, when I first bought my DAC1 (non-usb) I found a thread on a forum where John Siau was participating in the discussion. He suggested getting custom cables made to allow the balanced outputs to be hooked up to single ended inputs (which I did). From memory I believe this was to lower the impedance 'seen' by the pre-amp. However from the discussion on the new USB DAC1 it seems the impedance from the balanced output is 60 ohm i.e. double the single ended output. Is this a change due to the new output stage, or has this always been the case and John suggested the balanced to single ended connection for another reason?

 Also given the discussion on balanced drive headphones, Tyll places great importance on a true balanced source to obtain the noted benefits. With headrooms balanced DACs using two DACS per channel. How does the Benchmark produce it's balanced signal? It's possible that the Benchmark does not produce a 'true' balanced signal by Tyll's definition, in which case the touted benefits (of balanced headphone drive) may not be obtained by using the Benchmark as a balanced source.

 Maybe a possible new 'true' balanced model in the future?


----------



## The Monkey

Quote:


  Originally Posted by *lowmagnet* /img/forum/go_quote.gif 
_I'm confused by most people's statements on this subject myself. I just wrote it off as people not knowing what the heck they're talking about, but I'm willing to hear them explain why the DAC1's headphone jack is not good to them. Maybe in a separate threatd, though, Jenga that this one is._

 

Most of the complaints I have read complain that the HP out is too bright. In fact, I am beginning to suspect that this is what leads some to suggest that the DAC1 is a "brighter" DAC than, say, the DA-10. My own experience is that the HP out is fine, not great, but certainly not bad at all. Where others find it bright, I find it a bit thin. Maybe a difference without a distinction. I would prefer a little more body. But this subjective preference really is seeking a more euphonic sound, and that is definitely not what the DAC1 is about (correct me if I'm wrong, Elias). Bottom line, if you don't like the analytical HP out of the DAC1, find a nice warm amp to pair with it.


----------



## lowmagnet

Quote:


  Originally Posted by *The Monkey* /img/forum/go_quote.gif 
_Bottom line, if you don't like the analytical HP out of the DAC1, find a nice warm amp to pair with it._

 

Yeah, that pretty much sums up what I've said on the subject in response to those complaints. Essentially, the DAC1 is quite dead-on accurate when tested and the point of the design is accuracy. I don't like any solution that adds its own euphonics to the mix not called 'headphone' or 'speaker'. I'm a firm believer in Hi-Fi, and I happen to think that seeking out coloration is the opposite of fidelity. Just one man's opinion.


----------



## EliasGwinn

Has anyone else experienced this problem?

 That is, does anyone else use headphones that are too sensitive for the DAC1 such that the use of the volume control is limited?

 Thanks,
 Elias

 edit: this was meant to be posted waaaay back after Lord Chaos' post about the headphones being very loud


----------



## gregeas

Quote:


  Originally Posted by *EliasGwinn* /img/forum/go_quote.gif 
_Has anyone else experienced this problem?

 That is, does anyone else use headphones that are too sensitive for the DAC1 such that the use of the volume control is limited?
_

 

Yes, I have the exact same setup as Lord Chaos: I'm using Shure e500s out of the DAC1 headphone jack. It sounds fine, but I can't get past 9:00 o'clock without my ears bleeding. 

 I recently returned the Apogee MiniDAC because at normal listening levels, the e500s driven by the internal headphone amp had major, deal-breaking channel imbalance. I'm not hearing this on the DAC1, but I do wonder if sound quality would increase if I could get past 9:00.


----------



## EliasGwinn

Quote:


  Originally Posted by *music_man* /img/forum/go_quote.gif 
_hi mr gwinn,

 i am still having a litle issue with the krell. to get the dac1's output to be as loud as the sony cd players output i have to run the dac'1 at 5.66 volts. at 5.66 volts there is a small amount of distortion present in the tweeters. the sony is only putting out 4.63 volts. i measured them both. i don't understand this. how is this possible? shouldn't the same voltage from two diferent products into the same input cause the same db reading at the same distance? if i have to i could just back off on the volume. not a big problem. i would wish you could explain this though._

 

What audio source are you measuring the outputs with? What I mean is, are you measuring with music or a test tone? You should be using a constant level test tone to make sure the comparison is accurate.

 Assuming that you did use a test tone, and the voltage differences are as you say, the dB level will be different between the two, as it is directly proportional to the voltage. Loudness, however, has other factors involved - specifically, frequency content. If all things are equal, and the CD player sounds louder at identical voltage levels, it must be because there is additional frequency content in the analog signal of the CD player. This, most likely, is harmonic distortion. Without having the unit in front of me, I can't say for sure, but this is my best guesstimate.

  Quote:


  Originally Posted by *music_man* /img/forum/go_quote.gif 
_also i was hoping you would comment on this. some people say the dac1's headphone amp is just an "add on" and it was just put there as an extra item. they say it is not so good sounding but it was not intended to be. i think it sounds worth $300 by itself. i know you sell it for about that by itself. i think people that do not like it simply are presenting the usual subjective opinions. it may not be everyones favorite but i certainly do not find it to be unworthy.

 i assume benchmark intended for the headphone amp to sound the best it could with in reason, and to compete with other comparably priced amps. is that correct?

 thanks,
 music_man_

 

I'll try to answer this question without it sounding like an advertisement. At the risk of sounding less then humble, we feel the HPA2 (Benchmark's signature headphone amplifier, which is featured in the DAC1) is one of the most accurate and robust headphone amplifiers on the market (by robust, I mean it can handle an amazing array of loads without compromising the sound). 

 With that being said, there are users who simply don't want accuracy. WE HAVE NO PROBLEM WITH THAT!! I stress this because, at the end of the day, we are all music lovers, and understand that music is a very personal experience. And that personal experience cannot and should not be quantified. 

 However, our objective with the DAC1 (and the HPA2) is not to add certain colorations or artificial stereo width, because the product then becomes subjectively pleasing to some peoples tastes and ideas of what a device should sound like. 

 Our objective is to provide the most accurate reproduction and representation of the audio possible, and let the recording speak for itself. We believe, as designers of audio reproduction systems, that we owe it to the musicians/producers/recording engineers to represent the colors and images which they artistically derived. We believe it is not our place to add such impurities, even if it pleases our own subjectivities. If I want to hear such artifacts, I'll add it to my own hi-fi in my house.

 We use the most accurate measurement equipment and techniques available to achieve our objectives. We will strive and strive for precision and accuracy as we believe it is the crux of engineering fidelity. 

 I hope this doesn't come across as too much of a 'rant', or advertisement. If so, I apologize.

 Thanks,
 Elias


----------



## audioengr

Quote:


  Originally Posted by *music_man* /img/forum/go_quote.gif 
_this guy just shows up on this board. after we just got done complaining about this thread turning into a sales pitch for modders. i don't know about you guys but i am smelling some bs here. if i am wrong i do apologize. seems sketchy though.

 music_man_

 

I was as surprised to see this post as you. He generally posts only on Audio Asylum, where as he says, he is a moderator. However Audio Asylum website has been down for a day or two, so he found this thread....just a coincidence.

 He is an honorable gentleman and a really nice person, as most Aussies are.

 Steve N.


----------



## lowmagnet

Quote:


  Originally Posted by *EliasGwinn* /img/forum/go_quote.gif 
_That is, does anyone else use headphones that are too sensitive for the DAC1 such that the use of the volume control is limited?_

 

My Er-6i have that problem too, but they are also IEMs and they're 16ohm and 106dB/0.1V. the answer is less sensitive headphones. Especially since the 6i are designed for portable use.


----------



## EliasGwinn

Quote:


  Originally Posted by *CanMad* /img/forum/go_quote.gif 
_On topic, when I first bought my DAC1 (non-usb) I found a thread on a forum where John Siau was participating in the discussion. He suggested getting custom cables made to allow the balanced outputs to be hooked up to single ended inputs (which I did). From memory I believe this was to lower the impedance 'seen' by the pre-amp. However from the discussion on the new USB DAC1 it seems the impedance from the balanced output is 60 ohm i.e. double the single ended output. Is this a change due to the new output stage, or has this always been the case and John suggested the balanced to single ended connection for another reason?_

 

CanMad, this is a great question. The cables that John was referring to is not for impedance compensation, it is for proper signal routing and protection for the output drivers of the DAC1. Specifically, pin 3 of the DAC1 should be floating (ie, not connected), pin 2 should be wired to the center pin of the RCA cable, and pin 1 should be wired to the shield. This protects the inverted driver of the DAC1 from being shorted and damaged. The performance of this configuration will be near identical to the RCA outputs, but you will have the advantage of being able to use the attenuators if you need to. 

  Quote:


  Originally Posted by *CanMad* /img/forum/go_quote.gif 
_Also given the discussion on balanced drive headphones, Tyll places great importance on a true balanced source to obtain the noted benefits. With headrooms balanced DACs using two DACS per channel. How does the Benchmark produce it's balanced signal? It's possible that the Benchmark does not produce a 'true' balanced signal by Tyll's definition, in which case the touted benefits (of balanced headphone drive) may not be obtained by using the Benchmark as a balanced source.

 Maybe a possible new 'true' balanced model in the future?_

 

The DAC1 does produce a 'true' balanced signal. However, regardless of the source (two DAC's, balanced analog drivers, whatever...), we simply have no reason to believe balanced headphones provide any performance advantages. We have read the statements HeadRoom has made concerning this technology, and we have engaged in discussions regarding the validity of the claimed advantages. You can read about it on this post:

http://www.hydrogenaudio.org/forums/...pic=53801&st=0

 Not only is there no clear advantages, there is a MAJOR DISADVANTAGE in that it will double the source impedance seen by the headphones (which will halve the damping factor). So, from an engineering perspective, balanced headphones does not seem like a good idea. We are always open to engaging in analytical discussion on the matter. Also, if there is a real reason to believe this configuration is advantageous, we would absolutely be in favor of advancing the technology. Until then, we are considering it unfounded.

 Thanks,
 Elias


----------



## slwiser

Quote:


  Originally Posted by *EliasGwinn* /img/forum/go_quote.gif 
_
 The DAC1 does produce a 'true' balanced signal. However, regardless of the source (two DAC's, balanced analog drivers, whatever...), we simply have no reason to believe balanced headphones provide any performance advantages. We have read the statements HeadRoom has made concerning this technology, and we have engaged in discussions regarding the validity of the claimed advantages. You can read about it on this post:

http://www.hydrogenaudio.org/forums/...pic=53801&st=0

 Not only is there no clear advantages, there is a MAJOR DISADVANTAGE in that it will double the source impedance seen by the headphones (which will halve the damping factor). So, from an engineering perspective, balanced headphones does not seem like a good idea. We are always open to engaging in analytical discussion on the matter. Also, if there is a real reason to believe this configuration is advantageous, we would absolutely be in favor of advancing the technology. Until then, we are considering it unfounded.

 Thanks,
 Elias_

 

Elias 

 With this you are taking on one of the newest "religions" here on Head-Fi. Many are moving about as fast as they can to jump onto the train. I hope others engage you since I would like to see a vigorous debate from those who can actually set things up to test this tenet of the balanced system for headphones.


----------



## EliasGwinn

Quote:


  Originally Posted by *slwiser* /img/forum/go_quote.gif 
_Elias 

 With this you are taking on one of the newest "religions" here on Head-Fi. Many are moving about as fast as they can to jump onto the train. I hope others engage you since I would like to see a vigorous debate from those who can actually set things up to test this tenet of the balanced system for headphones._

 

I certainly want people to understand that we are not rejecting the idea of balanced headphones. We simply wish to determine the legitimacy of its claimed advantages, and we are open to discuss the idea with anyone who is interested in analyzing the technology.

 When analyzing new technologies to explore and develop, Benchmark does not simply follow consumer trends, but instead, strives to determine and develop technologies which truly progress the state of the art.

 And, just as a doctor doesn't guess when he is prescribing medicine (hopefully
	

	
	
		
		

		
			





 ), we will not resort to guessing when it comes to developing superior audio technology. 

 That should not imply that we only bench-test without listening. I, personally, listen to our products on a regular basis at home and at a recording studio where I moonlight. I listen along side, and in combination with, lots of other audio gear (pre's, amps, monitors, headphones, mics, compressors, eq's, etc.). I know from experience that specs and graphs don't tell the whole story, but designing intelligently does.

 Oh my, I just ranted again, didn't I? 
	

	
	
		
		

		
		
	


	




 Thanks,
 Elias


----------



## EliasGwinn

Oh and one more thing.....

 Have a great weekend everybody!

 Thanks,
 Elias


----------



## Tassie Devil

Quote:


  Originally Posted by *music_man* /img/forum/go_quote.gif 
_this guy just shows up on this board. after we just got done complaining about this thread turning into a sales pitch for modders. i don't know about you guys but i am smelling some bs here. if i am wrong i do apologize. seems sketchy though.

 music_man_

 

You are rather offensively suspicious Music Man. Why on earth would I be a salesperson for a modder who lives thousands of miles away? Come on, get a life and accept my findings as sincere from someone who only just found this site but who has been dabbling in high quality reproduction for 50 years (not 60 as it took the first 10 years to be able to afford good gear).

 And I think Nugent's prices for cabling is over the top and have bought elsewhere. I similarly remain unconvinced that he has found the Nirvana of transport using a laptop and his software. But I'll defend his brilliance at modding a Benchmark any day. I have the proof of the pudding here in a very high end system.

 And to reply to another poster, yes, those mods are expensive, but you get what you pay for. Pity Benchmark did not design it better in the first place. I'll get in touch with them to suggest they use Nugent as a consulant although I'd be surprised if they are open minded enough to do that. Such is life.

 John


----------



## Icarium

There are actually two Benchmark engineers (One their director) posting on this thread ;p The guy who posted right above you is one of them.


----------



## The Monkey

Quote:


  Originally Posted by *Tassie Devil* /img/forum/go_quote.gif 
_Pity Benchmark did not design it better in the first place. I'll get in touch with them to suggest they use Nugent as a consulant although I'd be surprised if they are open minded enough to do that. Such is life.

 John_

 

I think the Benchmark reps posting in this thread have shown they are plenty open-minded.


----------



## jpelg

Quote:


  Originally Posted by *slwiser* /img/forum/go_quote.gif 
_With this you are taking on one of the newest "religions" here on Head-Fi. Many are moving about as fast as they can to jump onto the train. I hope others engage you since I would like to see a vigorous debate from those who can actually set things up to test this tenet of the balanced system for headphones._

 

I severely doubt that such a discussion would lead to any conclusion other than that there are those people who hear a difference and like it, regardless of what the numbers say.


----------



## Tassie Devil

Quote:


  Originally Posted by *The Monkey* /img/forum/go_quote.gif 
_I think the Benchmark reps posting in this thread have shown they are plenty open-minded._

 

I hope you are correct Monkey but professional ego can sometrimes be disruptive to logical thought. 
	

	
	
		
		

		
		
	


	




 However the fact that such a basically good product is so well priced and that price has not risen as a result of such good press gives hope.
	

	
	
		
		

		
		
	


	




 I have just written the promised email and will let you guys here know the response.

 BTW the comment above about why I suddenly appeared here was spot on.
	

	
	
		
		

		
		
	


	




 Audio Asylum was down and I had been working on that essay/review for a week but could not post it there (although I did do so a hour or so ago). A Google search discovered you guys as interested in the Benchmark so I thought my findings would be of interest to you.

 But I'm not into headphones so am probably in the wrong slot. But I guess that is not important - it is how to achieve the best sounding music from those silver platters that is of most interest whether it be via headphones or speakers.
	

	
	
		
		

		
		
	


	




 John


----------



## Jetlag

Quote:


  Originally Posted by *EliasGwinn* 
_Our objective is to provide the most accurate reproduction and representation of the audio possible, and let the recording speak for itself. We believe, as designers of audio reproduction systems, that we owe it to the musicians/producers/recording engineers to represent the colors and images which they artistically derived. We believe it is not our place to add such impurities, even if it pleases our own subjectivities. If I want to hear such artifacts, I'll add it to my own hi-fi in my house.

 We use the most accurate measurement equipment and techniques available to achieve our objectives. We will strive and strive for precision and accuracy as we believe it is the crux of engineering fidelity._

 

I think I love you Elias, will you marry me?


----------



## CanMad

Thanks for your answers Elias.

 I often use the Benchmark hooked up to my PC, while working, using my RS-1s from the HP out. I think it is as intended and very accurate, but would agree that it sounds a little thin compared to my Sugden ( which doesn't seem to be very highly regarded around Head-fi these days). I can only get to around the 5th detent (depending on the recording) but I haven't ever messed with the output pads. I can't say I've noticed any channel imbalance though.

 My other source is a Townshend 565 (which is a hot rodded Pioneer), I have certainly noticed a wider soundstage using it compared to the Benchmark, and I do wonder if this is artificial? The main reason I got it was because once I got the Benchmark, it meant that my CDs sounded better than my SACDs in a number of ways (particularly the pin point placement of musicians). Also I couldn't pass my copy protected DVD-As to the Benchmark, and with a number of AIX recordings which sounded superb through the Benchmark I wanted that for all my DVD-As (If the recording was good of course, which some are definitely not).

 I for one would certainly be interested in hearing a hot rodded Benchmark, and comparing it to my Townshend. But would not be prepared to take the risk of losing it's accuracy for a more euphonic sound if this is what the result of the mods is actually doing, without hearing them for myself first. The benchmark still gets a lot of use in my main system hooked up to my squeezebox 3 playing apple lossless files, when I feel like a random mix of music.

 BTW, if anyone hasn't heard an AIX recording through their benchmark, it's really worth doing.

 PS John if you do bring the Turbo Benchmark to Melbourne, I'd love to hear it. I could give you a demo of my AKG K1000 rig, might convert you to a sometimes headphone listener!


----------



## jules650

Quote:


  Originally Posted by *EliasGwinn* /img/forum/go_quote.gif 
_And, just as a doctor doesn't guess when he is prescribing medicine (hopefully
	

	
	
		
		

		
		
	


	




 )_

 

You'd be surprised. 

 I can't claim to understand it all but thanks for your very informative posts.


----------



## kool bubba ice

Quote:


  Originally Posted by *Tassie Devil* /img/forum/go_quote.gif 
_You are rather offensively suspicious Music Man. Why on earth would I be a salesperson for a modder who lives thousands of miles away? Come on, get a life and accept my findings as sincere from someone who only just found this site but who has been dabbling in high quality reproduction for 50 years (not 60 as it took the first 10 years to be able to afford good gear).

 And I think Nugent's prices for cabling is over the top and have bought elsewhere. I similarly remain unconvinced that he has found the Nirvana of transport using a laptop and his software. But I'll defend his brilliance at modding a Benchmark any day. I have the proof of the pudding here in a very high end system.

 And to reply to another poster, yes, those mods are expensive, but you get what you pay for. Pity Benchmark did not design it better in the first place. I'll get in touch with them to suggest they use Nugent as a consulant although I'd be surprised if they are open minded enough to do that. Such is life.

 John_

 

I think you are out of line. The Benchmark was made at a reasonable price point in mind. Under a grand. I'm sure the engineers of the DAC1 could have made the DAC1 much better but would have made them raise the price. To get the maxed out mods from Steven, you are looking at 2,800..


----------



## vcoheda

Quote:


  Originally Posted by *EliasGwinn* /img/forum/go_quote.gif 
_The DAC1 does produce a 'true' balanced signal. However, regardless of the source (two DAC's, balanced analog drivers, whatever...), we simply have no reason to believe balanced headphones provide any performance advantages._

 

That is an interesting opinion and I'm sure you and others at Benchmark are experts when it comes to DACs, but I'm not sure how knowledgeable you are when it comes to headphone amps and the benefits of a balanced headphone amplifier. No offense, but in this area, I tend to trust the opinions of Ray Samuels, Justin Wilson over at HeadAmp, and Tyll Hertsens at HeadRoom, as well as others who have spent their lives making amps dedicated for headphone use.


----------



## kool bubba ice

* Quote:


  Originally Posted by EliasGwinn /img/forum/go_quote.gif 
I certainly want people to understand that we are not rejecting the idea of balanced headphones. We simply wish to determine the legitimacy of its claimed advantages, and we are open to discuss the idea with anyone who is interested in analyzing the technology

 

* Quote:


  Originally Posted by *EliasGwinn* /img/forum/go_quote.gif 
_.

 When analyzing new technologies to explore and develop, Benchmark does not simply follow consumer trends, but instead, strives to determine and develop technologies which truly progress the state of the art.

 And, just as a doctor doesn't guess when he is prescribing medicine (hopefully
	

	
	
		
		

		
		
	


	




 ), we will not resort to guessing when it comes to developing superior audio technology. 

 That should not imply that we only bench-test without listening. I, personally, listen to our products on a regular basis at home and at a recording studio where I moonlight. I listen along side, and in combination with, lots of other audio gear (pre's, amps, monitors, headphones, mics, compressors, eq's, etc.). I know from experience that specs and graphs don't tell the whole story, but designing intelligently does.

 Oh my, I just ranted again, didn't I? 
	

	
	
		
		

		
		
	


	




 Thanks,
 Elias_

 

Elias you should really try the 650's balanced through the DAC1 with your Hi Fi gear. I'd be surprised if you thought they sounded bad..


----------



## music_man

i'm sorry tassie devil. i should not have insulted you. i apologize. if you like it thats all that matters. i cannot see how mr. nugent can mod it to be more accurate. maybe add a "flavor" that some people would like. i primarily use it as a reference tool. it is very clean. i'd rather add the coloration elsewhere in the signal chain.

 i think benchmark did a great job within it's price point. i wouldn't go saying it is a shame they didn't do better. they did as they intended. many products are made just to sell a product. some will take it. some will leave it. that makes enough sales for such companies. benchmark made what they felt was best from the onset i think. it is not trying to sound good or pleasent. it is trying to sound true to the source. it measures as such. other products have their own sound and also measure likewise.

 mr. gwinn,
 i used a 0dbfs 1khz solid tone. the sony is louder at nearly 1 volt less. i scoped it. you are correct that sony is full of distortion! so much for sonys flagship sacd player. i guess that is why i am using the benchmark,eh?

 music_man


----------



## J-Pak

Quote:


  Originally Posted by *music_man* /img/forum/go_quote.gif 
_i'm sorry tassie devil. i should not have insulted you. i apologize. if you like it thats all that matters. i cannot see how mr. nugent can mod it to be more accurate. maybe add a "flavor" that some people would like. i primarily use it as a reference tool. it is very clean. i'd rather add the coloration elsewhere in the signal chain.

 i think benchmark did a great job within it's price point. i wouldn't go saying it is a shame they didn't do better. they did as they intended. many products are made just to sell a product. some will take it. some will leave it. that makes enough sales for such companies. benchmark made what they felt was best from the onset i think. it is not trying to sound good or pleasent. it is trying to sound true to the source. it measures as such. other products have their own sound and also measure likewise.

 mr. gwinn,
 i used a 0dbfs 1khz solid tone. the sony is louder at nearly 1 volt less. i scoped it. you are correct that sony is full of distortion! so much for sonys flagship sacd player. i guess that is why i am using the benchmark,eh?

 music_man_

 

The DAC1 could be more dynamic for one. If that's the only improvement the Empirical Audio mods offer, then it's still an improvement.


----------



## Jetlag

Quote:


  Originally Posted by *kool bubba ice* /img/forum/go_quote.gif 
_I'm sure the engineers of the DAC1 could have made the DAC1 much better but would have made them raise the price_

 

 KBI - Curious as to what you might mean by "better"?

 I am certainly not a paid Benchmark shill, but every technical review I have read about the DAC1 has essentially echoed the same thing that Elias mentioned in his earlier post. That is; this device converts digital audio to analog audio in the most precise and accurate way possible while completely blocking jitter and it's potential negative effects on this process. This is precisely what I want from a DAC (or really any audio gear for that matter).

 Another thing I like about their design philosophy is the apparent complete lack of typical nonsense fluff so common in many "high-end" audio products. We've all seen and read the marketing BS that many of these companies include in their promotional materials. One would need to be somewhat naïve to accept these claims at face value.

 I don't want to add or subtract anything from what the artist intended me to hear. I do believe some artists write and perform their music with a certain edginess for a reason. Am I to say I know better by trying to 'warm' the sound or attempt to make it sound 'smoother'? I wonder what the artist might have to say to me about that?

 Yeah, maybe they could have built the DAC1 with audiophile brand named wire or made the volume knob out of "Cardas wood", but quite frankly I bought it because they went for accuracy. As always just my $.02 USD (which really isn't worth much these days).


----------



## Crowbar

Quote:


 this device converts digital audio to analog audio in the most precise and accurate way possible 
 

There are always ways to improve quality given any equipment, such as using multiple DACs and dithering the LSBs, as modern DAC chips are not accurate to the full 24 bits (an examle of this is the Anagram module which several high end DACs use). But of course, such things increase the price. Another thing is that most solid state amplifiers have thermal memory distortion which doesn't affect THD and IMD measures. See this AES paper: http://www.aes.org/e-lib/browse.cfm?elib=7497 It's not new discovery; it's been known for long before by designers of vertical amplifiers in oscilloscopes. In the tube world, it exists but the time constant is far larger and below the audio band; it only affects DC there. It is worst in chip amplifiers such as op amps, where the input stages are thermally coupled to the output stages.

  Quote:


 while completely blocking jitter 
 

Asynchronous resampling attenuates jitter and embeds it in the data; it does not remove it.

  Quote:


 Another thing I like about their design philosophy is the apparent complete lack of typical nonsense fluff so common in many "high-end" audio products. 
 

Now this I agree with 
	

	
	
		
		

		
			





  Quote:


 As always just my $.02 USD (which really isn't worth much these days). 
 

Well, you got one out of three points correctly, so that's not too shabby ;P


----------



## Jetlag

I did not phrase my jitter statement very well, I simply meant to point out how the DAC1 deals with jitter. Your wording was much clearer.

 Yes, realistically it could have been made 'better', but I think the law of diminishing returns would have rapidly set in for each tiny incremental gain in performance. For it's price point, can we possibly agree that it is a very precise DAC?


----------



## Tassie Devil

Quote:


  Originally Posted by *music_man* /img/forum/go_quote.gif 
_i'm sorry tassie devil. i should not have insulted you. i apologize. if you like it thats all that matters. i cannot see how mr. nugent can mod it to be more accurate. maybe add a "flavor" that some people would like. i primarily use it as a reference tool. it is very clean. i'd rather add the coloration elsewhere in the signal chain.

 i think benchmark did a great job within it's price point. i wouldn't go saying it is a shame they didn't do better. they did as they intended. many products are made just to sell a product. some will take it. some will leave it. that makes enough sales for such companies. benchmark made what they felt was best from the onset i think. it is not trying to sound good or pleasent. it is trying to sound true to the source. it measures as such. other products have their own sound and also measure likewise.

 mr. gwinn,
 i used a 0dbfs 1khz solid tone. the sony is louder at nearly 1 volt less. i scoped it. you are correct that sony is full of distortion! so much for sonys flagship sacd player. i guess that is why i am using the benchmark,eh?

 music_man_

 

Thanks Music Man - the Devil can be friends with Music again 
	

	
	
		
		

		
		
	


	




 I understand your scepticism and I shared it for years arguing that if the product could be improved, then the manuifacturer would have done it so mods are a con. Well I finally moved on and dabbled with some player mods via a technician in Mebourne. The improvement with better timing clocks & Black Gate capacitors was obvious, and one could see from the extra cost of these items the manufacturers could not afford to put them in and expect to sell against similar products, particularly if Joe Blow would never appreciate any difference.

 Now the Benchmark is in a different league and have manufactured a class A product to begin with. But, in common with most other products it can be improved. Now you will have to trust me here - I hate colourations (which I believe tubes introduce heaps of) and am capable of detecting the mushiness they introduce (some refer to it as greater "musicality"). So NO, there are definitely not more colourations in the Nugent mod. What does come through is a dramatically cleaner soundstage and resolution AND a palatable front to back depth that I have not had here before. The bass is rock solid and everything falls into place on a well recorded CD. And of course that is a bit the problem (or a problem with the bits) with too many CDs - they are not well recorded. No player or DAC, modded or not, can turn mediocre input into glorious output. Garbage in -> garbage out applies with a vengeance.
	

	
	
		
		

		
		
	


	




 So, although I understand your scepticism, it is misplaced. I think I have played around with a variety of gear for long enough now to be able to recognise a really significant product. Mating two together, the Esoteric UIX-1 and the modded Benchmark are generating superb sounding music here and that input is now justifying the stupifying money that has been spent on the McIntosh preamp, those Halcro amps and the Sound Labs speakers.

 If you are interested in seeing pics of the listening room and flow charts of this system, have a look at my picture gallery on Audio Asylum at
http://gallery.audioasylum.com/ I'm there under the moniker John C. - Aussie Go down the right hand column and click on "anyone" 

 John


----------



## Crowbar

Quote:


  Originally Posted by *Jetlag* /img/forum/go_quote.gif 
_For it's price point, can we possibly agree that is a very precise DAC?_

 

Sure.


----------



## Jetlag

Quote:


  Originally Posted by *Crowbar* /img/forum/go_quote.gif 
_Sure._

 

I need to refrain from posting after a 14 hour work day and wait until I am rested.


----------



## Lord Chaos

Warmth... detail... I don't know, but what I get out of my DAC1s is simply music. It's highly dependent upon the quality of the original recording. There are a lot of recording engineers I wish could listen to this, so they could clean up their acts. But... a good recording is purely enchanting.


----------



## Tassie Devil

Quote:


  Originally Posted by *Lord Chaos* /img/forum/go_quote.gif 
_It's highly dependent upon the quality of the original recording._

 

No arguement about that. That is one curse of a highly resolving system - warts and all come through. Unfortunately with all this, the result can be no better than the weakest link. Sadly that is often the recording. Past that I have found the DAC or speakers to be the next weakest points and regard amplifiers as the least offensive. Please don't misinterpret that, I know there are badly designed amps out there which can really cause chaos with the music reproduction. And yes, cables can be a problem, but how much of a problem depends on how good the other items in the chain are.

 Putting together a nice sounding system is a bit of a black art 
	

	
	
		
		

		
		
	


	




 and the room acoustics often does not help - the reason a lot of you go for headphones.

  Quote:


 But... a good recording is purely enchanting. 
 

And that is the bottom line - if you are enjoying music on what you have, then end of story.
	

	
	
		
		

		
		
	


	




 John


----------



## yourmando

I've been watching this thread ever since the Benchmark Media Systems DAC1 USB was announced. I wanted to chime in to say "thank you" to Elias Gwinn, John Siau, and the other folks at Benchmark for providing so much useful, factual, and measurement backed information. 

 For example, I have been wondering for years about bit perfect playback using the computer as a source. Benchmark’s wiki is the only place anywhere that I have seen Audio Precision graphs with THD+N measurements for volume control in iTunes and other players on Mac OS X and Windows, plus "gotchas" such as static sample rate conversion settings on the Mac, etc.

 The DAC1 USB is unique in that Benchmark has really taken ownership of the stated measurable performance from the source all the way to to the preamp/gain control and amplification stage (with the headphone outs)--talk about a clean signal path! I think it is amazing that this small unit can replace three potentially very large boxes--the old, inconvenient cd player, the "audiophile" quality preamp, and the headphone amp--and have better measured performance and potentially lower cost than any one of these!

 Cheers,
 Armando
 P.S. You guys should consider blogging! I personally find myself waiting for any new nuggets that Elias and John post.


----------



## EliasGwinn

Hey Head-fi'ers, I hope everyone had a great weekend. I see there has been a flurry of activity since I last stopped by...always great to see...

 First off, I would like to say that this thread has been a lot of fun for me, and I really appreciate all of your positive (and otherwise) feedback. 

 I'd also like to say that I am excited by all the analytical discussions we have had, but I'd hate to see them turn angry or combative. As much as we all love audio, we should try to keep our heads cool. Unlike tube gear, opinions don't sound better when they're hot. 

 Thanks!!
 Elias


----------



## Jon L

I have to believe that a lot of the disagreement about the superiority of balanced headphones/amps comes from the Sennheiser HD6xx experience. 

 We have a very widely-used, well-known headphone that changes its sound character so drastically when driven balanced that most people who have heard it transform when balanced tend to become believers of balanced headphones in general. 

 I tend to believe that Senn's are a unique "anomaly" that has somewhat become representative of the superiority of balanced operation, which does not extend to many other headphones, at least not to any similar degree.


----------



## music_man

i have decided that the dac1 headphone amp is very good in it's own right. it is not fun or warm. it indeed allows one to hear things they will not with other headphone amps. it is very accurate and precise. the depth is what impresses me the most. of course that is if depth is present on the source. it only plays back what exists on the source. it adds or subtracts as little as possible. where other amps add their signature color/flavor. i think that is why people have mistakenly disliked it. it tells the truth. the truth is not always pretty. i see people said it is horrible with k701's. i can understand this statement. most people could not stand the brutal reality. the dac1+k701=elite surgeon's knife. this is no squishy teddy bear. no warm fuzzy feeling here. i mostly enjoy the warm fuzzy type of amps. i just discovered through critical listening the fine attributes of the dac1. try it. it is intresting to see that it is devoid of opamps also. 

 music_man


----------



## s.a.b.

Quote:


  Originally Posted by *music_man* /img/forum/go_quote.gif 
_ it tells the truth. the truth is not always pretty. i see people said it is horrible with k701's. i can understand this statement. most people could not stand the brutal reality. the dac1+k701=elite surgeon's knife. this is no squishy teddy bear. no warm fuzzy feeling here. 

 music_man_

 


 I have the DAC1 and k701 (as well as DT880 ('03) and DT990) and I wouldn't describe the reality as "brutal" at all.

 I do have the ago old questions for Elias - at least when listening thru headphones, I (and even non-audiophile folks at my house) do hear a difference between transports and even digital cables. Any thoughts on this?

 Thanks.


----------



## Crowbar

Quote:


  Originally Posted by *s.a.b.* /img/forum/go_quote.gif 
_hear a difference between transports and even digital cables._

 

First of all, unless you've done proper blind testing, your results are simply invalid. Psychological bias is far more powerful than your best intentions to evaluate objectively.

 Having said that, it may be possible there is a difference. I hope Wavelength won't mind me quoting from his emails here, as that will answer the question of how different USB cables and computers can plausibly produce different sound:
  Quote:


 First there is no error correction with ISO modes [isochronous is the USB Audio mode]. Only with bulk mode like they use in hard drives....In apple land many of the 2.0 complient controllers have ISO support built in. Windows does not take advantage of that but apple does. This means less over head and more consistent data in the pipe....I have a USB analyzer and depending on cables the error rate can get kinda high especially on lengths longer than 2m. 
 

So, no error correction in the standard, as in theory error rate should be very low anyway, but with Windows computers it apparently may not be that low. That's actual data corruption, not mere jitter, and if the error rate is truly significant, it would seriously compromise audio quality. Solution: blame Microsoft, then use a Mac and/or shorter USB cable; I don't know whether Linux takes advantage of the hardware isochronous mode support.

 SPDIF is unlikely to have actual data corruption, so then it comes down to the jitter rejection ability of the ASRC. The nature of processes such as rate estimation that the ASRC does are discussed in the third post here: http://recforums.prosoundweb.com/ind...t/17177/16575/


----------



## music_man

Quote:


  Originally Posted by *s.a.b.* /img/forum/go_quote.gif 
_I have the DAC1 and k701 (as well as DT880 ('03) and DT990) and I wouldn't describe the reality as "brutal" at all._

 

what do you mean by that statement? i didn't mean that the dac1 is "brutal". i meant to say that most people don't want to know how poor their recordings(in my case cd's) are.

 most audiophiles don't want the "truth". even though they say they do. ie, the truth being the recording as it exists with nothing added or subtracted. or as little as possible. most audiophiles want a pretty rendition of the truth pertaining to their particular tastes. regardless of what most audiophiles will admit.

 i admit it is tough to hear the blatant reality of most poor recordings. those that are supurb are tough to listen too for other reasons. they demand utmost attention of the listener or they become fatiguing. the dac1 provides this type of exacting performance.

 music_man


----------



## yourmando

Quote:


  Originally Posted by *Jetlag* /img/forum/go_quote.gif 
_I am certainly not a paid Benchmark shill, but every technical review I have read about the DAC1 has essentially echoed the same thing that Elias mentioned in his earlier post. That is; this device converts digital audio to analog audio in the most precise and accurate way possible while completely blocking jitter and it's potential negative effects on this process. This is precisely what I want from a DAC (or really any audio gear for that matter)._

 

 Quote:


  Originally Posted by *Crowbar* /img/forum/go_quote.gif 
_There are always ways to improve quality given any equipment, such as using multiple DACs and dithering the LSBs, as modern DAC chips are not accurate to the full 24 bits (an examle of this is the Anagram module which several high end DACs use). But of course, such things increase the price. Another thing is that most solid state amplifiers have thermal memory distortion which doesn't affect THD and IMD measures. See this AES paper: http://www.aes.org/e-lib/browse.cfm?elib=7497 It's not new discovery; it's been known for long before by designers of vertical amplifiers in oscilloscopes. In the tube world, it exists but the time constant is far larger and below the audio band; it only affects DC there. It is worst in chip amplifiers such as op amps, where the input stages are thermally coupled to the output stages.
 ...
 Asynchronous resampling attenuates jitter and embeds it in the data; it does not remove it.
 ...
 Well, you got one out of three points correctly, so that's not too shabby ;P_

 

You might be surprised to learn that the Benchmark DAC1 is actually entirely immune to jitter under all operating conditions. Their UltraLock design includes much more than an Asynchronous Sample Rate Converter to attenuate jitter. Check out the graphs in the manual:

http://www.benchmarkmedia.com/manual...USB_Manual.pdf

 We can see total jitter immunity to the limits of the Audio Precision System 2 Cascade test unit, up to 12.75 UI (2075 ns) of jitter. There is also a graph showing complete immunity to cable induced jitter by comparing a run of the mill 1000' (~305 m) unshielded cat 5e data cable to a direct digital input--no jitter induced sidebands. Another graph shows the eye pattern of multiple 1000' data cables and even the one with the tiny eye pattern sees no change in performance.

 As for improving the quality of the conversion--I have not seen any measured results that are as good as the dac1 under as wide a range of operating conditions. (Rant: often, only a few specs are provided, sometimes without stating measurement conditions, and it seems that some manufacturers simply list the specs of the dac chip they use, or misleading specs such as SNR that includes the output mute circuit.)

 In terms of measurable accuracy, the DAC1 is up there with any dac at any price, and performs as specified independent of sampling frequency, jitter, for very long digital and analog cable runs, through the full usable level range. And most importantly to us, the headphone outputs deliver the full rated performance of the dac1 (performance just as good balanced and unbalanced outputs), even at very high levels, and for headphones loads from 30 ohm to 600 ohm with no change in performance.

 I would personally give Jetlag a 3 out of 3 in his assessment! That's not to say we won't ever see effective number of bits approach 24 if we move past current technology. Given the alternatives, and that we are really limited by the best A/D converter used for the very best source material, and that we are probably already well past the audibility threshold for something at this stage in audio reproduction signal path, I would say we have hit the state of the art!

 Armando


----------



## music_man

i'd like to hear comments from people that are impressed with the headphone outputs and what level of stand alone amps they would compare it to.

 i am wondering if i should add an outboard amp to it and what is likely to considerably exceed the built in amps with k701's. if anything. i am very impressed with the accuracy and transparency of the dac1 both as a dac and headphone amp. i'd like to stick with that. i do not want to add an amp to it that has it's own sound.

 music_man


----------



## Lord Chaos

Music Man, I'm no expert on headphone amps but I have been using them since the days of the Shure Solo-Phone. I've also spent a lot of time recording. So, I can't say the HPA2 is the world's best headphone amp, but I can say it's the best sounding component I've ever heard. I look forward to putting on the headphones and disappearing into the music, which is what happens with a good recording.

 I think that anyone looking for high-quality sound would be pleased with the DAC1 and its headphone amp. I bought mine simply because it provided everything I needed in one box, and the reviews I read were positive. I bet $1000 that I'd like it too, and I do. So, I bought another one to use at my desk.


----------



## Crowbar

Quote:


  Originally Posted by *yourmando* /img/forum/go_quote.gif 
_We can see total jitter immunity to the limits of the Audio Precision System 2 Cascade test unit, up to 12.75 UI (2075 ns) of jitter._

 

I find it strange that the highest jitter frequency tested was 9 kHz, when jitter effects increase with frequency and is very significant even above 20 kHz. Take a look at this article by Lavry: http://www.lavryengineering.com/white_papers/jitter.pdf
 Of course, the nature of the effects going through an ASRC would be different, but is no reason not to consider such test cases.

 I'm also interested in the level of jitter after the ASRC, which is mostly dependent on the local clock. Dunn's paper from the 93rd AES convention derives a 20 ps maximum allowable: http://www.nanophon.com/audio/jitter92.pdf

  Quote:


 As for improving the quality of the conversion 
 

What do you mean? No current DAC chip manages actual 24 bits resolution; about 21 is the highest I've seen. More can be done with multiple DAC chips and scrambling. If you mean the quality of the IV conversion, I suggest you check out Hawksford's current-steering transimpedance amplifiers paper. In simulation, Fig.4-4 is the best performing IV I've seen, and from those that have built it I've not seen anything to make me doubt those results. Of course, tuning a discrete design where parts must be matched may not be economic for certain price ranges.

 Finally, you quoted me mentioning the thermal memory distortion, yet you did not address that point. This issue is most significant with chip amps (as even a skeptic like D.Self will point out), which is another reason to prefer discrete designs. This doesn't really show up in THD measurements. It would be interesting if someone would perform the memory measurement described in the paper I quoted before on various SS stages, including this one.

  Quote:


 (Rant: often, only a few specs are provided, sometimes without stating measurement conditions, and it seems that some manufacturers simply list the specs of the dac chip they use, or misleading specs such as SNR that includes the output mute circuit.) 
 

That is unfortunately the case. Summary metrics like THD/IMD have limited utility as they're not perceptually weighted. Some distortions are inaudible to a couple of percent (i.e. low order even harmonics), others in the parts per million (B/AB class crossover distortion). Some have audibility established but the ear's exact sensitivity to them not determined (such as the thermal memory issue I mentioned above).

  Quote:


 In terms of measurable accuracy, the DAC1 is up there with any dac at any price 
 

I've mentioned some measurements that I don't see there. I'm not singling out the DAC1, I'm talking about DACs in general, and I'm glad the DAC1 at least shows the amount of detailed measurements that it does; hopefully it will set a trend. But I would not say they are sufficient to justify your claim.

  Quote:


 and that we are really limited by the best A/D converter 
 

As I pointed out above, we are not. You can use a bunch of them to get an improvement.

  Quote:


 we are probably already well past the audibility threshold for something at this stage in audio reproduction signal path 
 

That's an unwarranted statement. The ITU recommends double-blind triple-stimulus with hidden reference testing; I'll believe it when I see the results.

 There are many other things wrong with sound reproduction besides the electronics, such as that all speakers and headphones have tremendous distortion (well, with the exception of Alan Hill's helium plasma device in the 1 kHz and up range). Even worse is the recording and playback geometry. Several things like binaural, Ambisonics, etc., but none of them work very well. The single thing that will make most of an improvement in audio is having one's personal HRTF. It's already well established that HRTF differences between individuals are very significant. I find it quite amusing when I read reviews about how some amp makes the imaging clearer and such nonsense, when it has nothing to do with that. HRTF measurement needs an anechoic chamber and expensive equipment, but in recent years the alternative of laser scanning and finite boundary method simulation allows one to compute the HRTF. I wonder how long until hi-end audio stores start carrying laser scanners so one can just go and get a scan, then they compute the HRTF, for a fee, and you can use it in your favorite DSP convolver to make those binaural recordings made with the pinna-less dummy head _really_ give the right positional sound. Anyway... getting too offtopic here with my rant


----------



## EliasGwinn

Quote:


  Originally Posted by *s.a.b.* /img/forum/go_quote.gif 
_I have the DAC1 and k701 (as well as DT880 ('03) and DT990) and I wouldn't describe the reality as "brutal" at all.

 I do have the ago old questions for Elias - at least when listening thru headphones, I (and even non-audiophile folks at my house) do hear a difference between transports and even digital cables. Any thoughts on this?

 Thanks._

 

s.a.b.,

 I'm not sure what is causing the differences you are hearing, but we have done extensive testing with different types of digital cables, and we have not been able to hear or measure any differences using various cables with the DAC1. Different analog cables can affect the output sound of the DAC1, but we have not been able to find any performance differences with different digital cables. 

 As Crowbar mentioned, sometimes its really difficult to hear differences between set-ups without a proper testing configuration. When conducting an A/B test, its very easy for even the most trained ear to lose their point of reference, making it very difficult to make accurate comparisons. 

 Thanks,
 Elias


----------



## StevieDvd

Elias,

 Thanks for the continued updates and persevering with the multiple and sometimes extensive questioning.

 Can I just ask for suggestions/tips on using the DAC1 balanced xlr outs feeding into a balanced headphone amp? Any optimum or suggested settings?

 Embarrasses me to admit I just left it at factory settings, all sounds well but it's be nice to know I'm not missing out for the sake of flipping a switch or jumper around.

 My amp (Rudistor NX-33) volume control goes to about the 12 o'clock position before it gets too loud to use.

 I read your position on using balanced headphones directly from the Dac1 but that's what I did initially before upgrading to balanced amp. 

 Have you opinions on balanced amps too?
	

	
	
		
		

		
		
	


	




 Steve


----------



## The Monkey

Elias,

 One design note for the good folks at Benchmark. The screws that need to be undone in order to get inside the unit could be improved. First, mine were screwed in so tight that it was almost impossible to remove them. Second, it is very easy to damage or "strip" the screw heads themselves. I'd love to see Benchmark go with hex socket head screws or thumb screws.


----------



## EliasGwinn

Steve,

 The factory settings on the DAC1 are suited to fit most set-ups. How far are you turning the volume pot of the DAC1? If you are not able to get the volume knob past 10 o'clock or so, then you may want to change the output attenuators.

 As for running balanced into an amp, that is the best way to interconnect the DAC1 and another device. Balanced headphones are a whole other subject...

 On the subject of balanced headphones, we have found no evidence, documentation, or explanation to support the claims of performance advantages. For those familiar with bridged-mode power amplifiers, balanced headphones are a very similar topology....

 Bridged-mode amplifying is used to increase power into a given load, while sacrificing performance (especially damping factor). This mode of operation is undesirable unless an increase in power is absolutely necessary.

 From everything we have seen, balanced headphones suffer from the same symptoms, while not having any benefits to offer. The benefit of bridged-amping (additional power) is not needed in this case (the DAC1's headphone output has more then enough power for any headphones we have encountered). Therefore, we simply have no reason to believe there are any advantages. 

 Some people may enjoy this particular configuration, and we encourage those people to continue to enjoy it. 

 As a company and equipment designer, we choose to follow routes of technology which are founded in some sort of explainable / provable / understandable manner. Otherwise, we would be chasing every whim and trend that floated through the audiophile community (and, trust me, that could get very exhaustive 
	

	
	
		
		

		
		
	


	




 ).

 Thanks,
 Elias


----------



## EliasGwinn

Quote:


  Originally Posted by *The Monkey* /img/forum/go_quote.gif 
_Elias,

 One design note for the good folks at Benchmark. The screws that need to be undone in order to get inside the unit could be improved. First, mine were screwed in so tight that it was almost impossible to remove them. Second, it is very easy to damage or "strip" the screw heads themselves. I'd love to see Benchmark go with hex socket head screws or thumb screws._

 

Monkey,

 Thank you very much for this very valuable feedback. I will bring this up with our production team.

 Thanks,
 Elias


----------



## music_man

mr gwinn,

 is the dac circuit connected to the headphone amp circuit balanced on the pcb? if not, why and how is it connected? edit: it appears that it is unbalanced?

 also, what is the headphone amp using in place of an op-amp as the amplifier? a buffer ic? edit: wasn't it discussed that the headphone amp was discrete? there is mention of op-amps here.

 edit: is there a pot on the pcb to adjust the headphone amps gain?

 thanks,
 music_man


----------



## yourmando

Quote:


  Originally Posted by *Crowbar* /img/forum/go_quote.gif 
_What do you mean? No current DAC chip manages actual 24 bits resolution; about 21 is the highest I've seen..._

 

I didn't mean to quibble with any specific points, but thought you might be interested to know that the DAC1 jitter immunity is more than just attenuation from an ASRC. Yes, I did mention that the effective number of bits of any DAC has not hit 24 bits (And this is probably not possible with current technology because of the thermal noise floor at room temp). My original point is that the DAC1 represents the current state of the art in terms of measurable performance over a wide range of real world operating conditions.

 For a third party look with Audio Precision measurements of the non-usb DAC1:
http://theaudiocritic.com/blog/index...Id=10&blogId=1

 I can't say it better than that! The USB version has more features (beyond USB) and a wider range of operating conditions.

 Armando


----------



## Crowbar

Well, specific points are what makes a discussion concrete, and I wouldn't call it quibbling. Looking at the linearity graph in your link, it's linear to about -115 dB. I don't know what DA chip is used in the DAC1, but AD1955 datasheet shows linearity down to -125 dB, and that's a 5 year old IC. The PCM1794A has similar performance. Both are cheap. My other two points are not discussed at the site you linked to. The dither test is nice, showing the ~18.5 bit effective resolution. Still, I'd still like to see the high frequency jitter test I mentioned above, and especially the thermal memory measurement. I would guess the DAC1 would do fine in the former, but I have my doubts about the latter due to the use of opamps.


----------



## yourmando

Quote:


  Originally Posted by *Crowbar* /img/forum/go_quote.gif 
_That's an unwarranted statement. The ITU recommends double-blind triple-stimulus with hidden reference testing; I'll believe it when I see the results.

 There are many other things wrong with sound reproduction besides the electronics, such as that all speakers and headphones have tremendous distortion..._

 

I would also love to see more double blind tests! In absence of this, I will pick the product with the best and most extensive measurements (and useful features), even if I feel it is probably overkill. That we are "probably" well past the audibility threshold is a sufficiently hedged comment. I, too, love to learn about the limits of perception, the design gear, etc. Like you, I believe that transducers--loudspeakers and their interaction with the room, headphones--are by far the limiting factor; The speakers are the sole determinant on the reproduction side of the width, height, and depth of the sound stage and size of virtual images (aside from intentional or unintentional distortion/processing in the electronic signal path).

 Let me put it this way, if a friend wanted a recommendation on the best near-field audio reproduction electronics components, I would recommend a computer or bit-perfect streamer as the source (not a cd player that I have to feed!), and a Benchmark DAC1 USB as the DAC/preamp/headphone amp with the volume control kept within arms length, because I know of no other combination with this level of end to end distortion and noise with real resistive loads. (This doesn’t include multiple channel and DRM restricted audio, which would require another setup.)

 Cheers,
 Armando


----------



## yourmando

Quote:


  Originally Posted by *Crowbar* /img/forum/go_quote.gif 
_Well, specific points are what makes a discussion concrete, and I wouldn't call it quibbling. Looking at the linearity graph in your link, it's linear to about -115 dB. I don't know what DA chip is used in the DAC1, but AD1955 datasheet shows linearity down to -125 dB, and that's a 5 year old IC. The PCM1794A has similar performance. Both are cheap. My other two points are not discussed at the site you linked to. The dither test is nice, showing the ~18.5 bit effective resolution. Still, I'd still like to see the high frequency jitter test I mentioned above, and especially the thermal memory measurement. I would guess the DAC1 would do fine in the former, but I have my doubts about the latter due to the use of opamps._

 

You are certainly the tough critic! “The” Audio Critic, champion of mesurement-driven and DBT evaluation was much “softer” than you:

  Quote:


 The cloud-cuckoo-land high-end DACs at ten times and fifteen times the Benchmark’s price are no better and in most cases not as good...

 Why can I confidently make that statement? Because I measured the DAC1 up and down and sideways with the Audio Precision SYS-2722, possibly the most sensitive and accurate audio-test instrument in the world, and found it to be as nearly perfect as a digital-to-analog converter can get at the present state of the art. Totally perfect 24-bit converters, with the theoretical noise floor of –146.24 dBFS and a perfect monotonicity “staircase” waveform at the ten lowest LSBs, do not yet exist, at any price, and probably never will. Still, the DAC1 yielded the best measurement figures that I have ever obtained out of a digital processor on my test bench, nor have I ever seen better measurements on other units in other publications...

 ...I’ll take electronic perfection, any day of the week, if it costs $975 instead of $17,500. 
 

We have about 20 pages in the DAC1 manual of Audio Precision graphs and specs, seven more independent graphs in the article linked above, and I’m sure there are plenty more out there. Your drive to know more amazes me. Good luck in your search, and please do let me know if you find some more data, or another unit with a comparable set of measurements showing better performance all the way to headphone amp stage with at least the same range of operating conditions! 
	

	
	
		
		

		
		
	


	




 Armando


----------



## Crowbar

Thanks yourmando, for again sidestepping my specific argument and going after stuff irrelevant for my argument, like a politician 
	

	
	
		
		

		
			




 Perfect example: I say such and such measurement is important, and you say oh, there are these tons of other measurements.
 Analogy: I go to the mechanic to have a detailed checkup on my car, and he checks everything but the brakes. I ask him to check the brakes, but he says, oh, but I checked all these dozens of other things, everything's perfect!
 I think you see the point. By asking him to check the brakes, it doesn't mean I think they are in poor condition, but that they are important for proper functioning.

 I think your defensiveness is due to a failure to see that my criticisms already presume that the DAC1 is doing a good job. I'm simply pointing out gaps that make your certainty of unrivaled superiority not fully justified, and your implication that any further improvements are beyond the audiblity, entirely unsupported. I'm sure virtually all commercial DACs suffer from issues I've mentioned, but that doesn't mean they don't exist. I'm not making a comparison here, as it's not something I'm really interested in.


----------



## Lord Chaos

I remember when a device with 60dB of dynamic range was considered good. What would it take to *hear* the difference between a noise/distortion floor of -115dB versus one at -125dB? I'm interested in this because the effects of using software volume control seem very slight to me after looking at the Benchmark Wiki graphs.


----------



## EliasGwinn

Crowbar,

 I want to let you know your points/questions are valid, and I am getting some information together to properly respond.

 Thanks,
 Elias


----------



## lowmagnet

Quote:


  Originally Posted by *EliasGwinn* /img/forum/go_quote.gif 
_Monkey,

 Thank you very much for this very valuable feedback. I will bring this up with our production team.

 Thanks,
 Elias_

 

Could you do us a favor and give us the thread type so we can replace screws on units in the field?


----------



## EliasGwinn

Quote:


  Originally Posted by *lowmagnet* /img/forum/go_quote.gif 
_Could you do us a favor and give us the thread type so we can replace screws on units in the field?_

 

4-40 thread, 1/4" length

 Thanks,
 Elias


----------



## EliasGwinn

Crowbar,

 I'll try to round up your questions and answer them here. Forgive me if I miss any...

 1. Thermal Memory Distortion

 This is, as you indicated, an often overlooked design consideration. I checked out the paper you linked (thank you for the references by the way...it makes for a much more constructive conversation when references are given). It seems they are referring to high-gain scenarios (specifically, power amps). 

 The DAC1 addresses Thermal Distortion conditions by maintaining low gain operating conditions and using stable resistors. The opamps used in the DAC1 are all operating in low-to-no gain, buffer-type applications. The gain is set with thin metal film resistors which are (pardon the pun) very resistant to changes due to current and temperature. If thermal conditions were to effect the open-loop gain of the opamp, the overall (closed-loop) gain will not be affected much because the metal film resistors in the feedback network are the dominant factor.

 Also, the I-to-V converter is external of the D/A chip. This configuration is much less prone to thermal distortion then voltage-output converters.

 2. Jitter measurements

 When you mentioned that we only tested the DAC1 with jitter frequencies up to 9 kHz, I was as confused as you were. In fact, we measure with jitter frequencies up to 100 kHz. I see where the miscommunication came about, though. On page 34 of the DAC1 USB manual:

http://www.benchmarkmedia.com/manual...USB_Manual.pdf

 ...we show a measurement of THD+N vs. Jitter Amp and Jitter Frequency. We show about 20 plots of THD+N vs. Jitter Amplitude, with each plot representing a different (constant) Jitter Freq. The consecutive Jitter Freq plots are increased by 500 Hz intervals from 2 Hz to 9 kHz. This plot demonstrates that the THD+N vs. Jitter Amp plots do not change based on Jitter Freq.

 The graph on the preceding page (33), however, is what you are looking for. It demonstrates the DAC1's Jitter Tolerance (Distortion vs. Jitter Freq) from 100 Hz to 100 kHz Jitter Freq's. 

 As can be seen from the two graphs, there is no change in performance with varying Jitter Amplitudes (up to 12.75 UI) or Frequencies (up to 100 kHz).

 3. Improvements to the DAC1

 There were several points you made about ways the DAC1 could be improved, so I'll try to address them all...

 The D/A converter used in the DAC1 is the AD1853. This chip, as it is used in the DAC1, actually achieves linearity down into the -130's dB (nearly -140 dB). I will try to find our measurement graph for this and post it here. 

 The DAC1 achieves 21-bits of signal-to-noise ratio. 

 The DAC1 can accurately resolve the 24th-bit (although it will be below the noise floor).

 4. Publishing performance plots of the DAC1

 I am very glad to see customers analyzing and appreciating the performance plots which are offered in the manual. We would have put even more plots in the manual, but the manual is already too thick. But, due to the feedback I am hearing from you all, we will publish all performance plots of the DAC1 on our website. I will keep you posted (pardon the pun).

 Thanks,
 Elias


----------



## CanMad

Hi Elias,
 Have you guy's at Benchmark looked at designing a DAC with a discrete output stage rather than op-amps? Possibly transformer based such as Townshend use?

 Cheers


----------



## little-endian

Hi Elias,

 finally I found the time to answer you. Because I thought, this topic could the of public interst, I answer you here. Maybe, it would be a good idea to create an own thread for the S/PDIF stuff at least, though.


 I've checked the speed of the crystal as you adviced. The clock is running at 28.322 MHz. However, I don't understand what you mean by "your's may or may not be upgraded". "Upgraded" sounds like changing devices after you already sold them instead of directly manufacturing new models. Except that, I thought that *all* DAC1s were capaple of 192 kHz - at least via the electrical input. So there were some even not accepting 192 kHz through the other inputs?

 After all, if the crystal (and the rest of the parts) are able to support 192 kHz, you would just have to exchange the optical receiver by a newer model, right?

 However, I doubt if the whole 192 kHz is worth the fuss since the DAC1 is going to convert everything down to ~ 110 kHz, anyway. The only (technical) advantage for 192 kHz material to be played back would be to avoid the conversation to 96 kHz in software (which could be worse) and the theoretical benefit of about 14 kHz Samplerate (the difference between the internal rate and the 96 kHz, resulting from the input's limitation).

 Since the DAC1 is able to recognize sample rates over a wide range (not only the common steps 44.1, 48, 96, etc.), I wonder if even the older optical inputs aren't able to receive 110 kHz SR. Thus, playing a file, sampled 192 kHz from the computer and resampling it to the internal sample rate of the DAC1 before sending it via Toslink could work and if the resampling is of the same quality like the one of the DAC1, no quality would be lost (in theory, I doubt that I could hear any difference, anyway).

 In regard to S/PDIF TTL from PC drives, I did additional tests and also tried the DAC1 in conjunction of a Yamaha CD-Player which is said to handle the "valid flag" in the S/PDIF-stream correctly.
 Although this conforms to the standard, I had to recognize that in the case of the DAC1, this is even a disadvantage to the (strange) usage of the "non pcm" flag, like pc drives seem to use it instead. Why? Because the DAC1 doesn't display invalid S/PDIF samples. 
 This is a real pity because this way I don't see anything. When using the PlexWriter as source (you could see it on the picture I had linked, hehe), on most (unfortunately not all) errors, the "non pcm" led lights up at least. This is unusual but still better than nothing.
 After thinking about it, this is clear. Even when (uncorrectable!) C2-errors occur, the S/PDIF stream itself stays valid, so of course the "error" led of the DAC1 doesn't show this. "Non pcm" isn't the case either (at least when using common CD-players as source). So the result is that the DAC1 simply doesn't have a led to show standard conform C2-errors. This would be a real feature for people who want to know in which conditions their CDs are (C2 errors can indicate a soon death, so one better makes a copy before it's too late).

 I wonder also how the DAC1 handles samples, flagged as invalid (VALID=true - the boolean logic is inverted here)? It doesn't show them, that's for sure. But is there a stage which performs interpolation like built into every CD-player or are they simply ignored? Because you have to distinguish between errors on the distance source --> DAC resulting in loss of sync ("error" led lights up) and erroneous content. The S/PDIF standard also mentions parity bits used for error detection. Does the DAC1 make use of them?
 Some time ago I had a soundcard, featuring two Toslink interfaces (in and out). When using a pretty long (and cheap) cable between it and the CD-Player, the sound from the sound card was distorted and noise. The amazing thing was that it even gaves a sound when holding the plug near the input. The more the distance, the more the distortion. The DAC1 however either gives a perfect sound signal or no at all. The is no in between. I wonder how it detects that the data is errorfree before converting it to analog. For the standard, here you can find more information:

http://www.epanorama.net/documents/audio/spdif.html

 It would be very interesting to know how the DAC1 handles and processes the whole S/PDIF data (interpolation, parity checks, etc.). If someone can organize such an overview, then YOU. 

 Thanks again, Elias!

 little-endian


----------



## Crowbar

Quote:


  Originally Posted by *CanMad* /img/forum/go_quote.gif 
_Possibly transformer based such as Ayre and Townshend use?_

 

LOL dude, all the tube guys try to build output-transformerless amps, and you want to add transformers... All magnetic core transformers have hysteresis problems and low frequency distortion, and usually phase problems. This is not a hi-fi solution.


----------



## Crowbar

Thanks for the comments.

  Quote:


  Originally Posted by *EliasGwinn* /img/forum/go_quote.gif 
_It seems they are referring to high-gain scenarios (specifically, power amps)....metal film resistors which are (pardon the pun) very resistant to changes due to current and temperature....the overall (closed-loop) gain will not be affected much because the metal film resistors in the feedback network are the dominant factor._

 

You're right about modern resistors being sufficiently stable. However, negative feedback is used for correction of static distortions, and its ability to correct for dynamic distortions of this type is limited. Couple this with an opamp's extremely high loop gain...
 I'm working on SPICE simulations based on thermal models for transistors, and I'm not sure if I can easily modify it for opamps, but I'm going to try this over the next month or so, as it's easier than the measurement setup described in that paper as I don't have the right equipment.

  Quote:


 Also, the I-to-V converter is external of the D/A chip. This configuration is much less prone to thermal distortion then voltage-output converters. 
 

Yes, that certainly helps. I've tried voltage-out DAC chips before and only one sounded good.

  Quote:


 I see where the miscommunication came about, though. On page 34 of the DAC1 USB manual 
 

Yes, that clear it up.

  Quote:


 This chip, as it is used in the DAC1, actually achieves linearity down into the -130's dB (nearly -140 dB). I will try to find our measurement graph for this and post it here. 
 

Here's the datasheet: http://www.analog.com/UploadedFiles/...ets/AD1853.pdf
 And the one for the AD1955: http://www.analog.com/UploadedFiles/...ets/AD1955.pdf
 Dynamic range and THD+N are about 5 dB better with the AD1955 (can be seen in the specs and the graphs as well). The linearity plots are in graphs 19 and 12, respectively. Though the AD1853's LSBs seem to go lower, note the major kink around 117 dB (preceded by a droop). Though it's not very large, the scale is logarithmic so the higher LSB linearity on this graph is less important than something with much larger energy.
 Overall though, the difference is not great. The digital filters the two chips use appear to be identical. I guess ADI saved money on redesigning that part hahaha


----------



## little-endian

Quote:


  Originally Posted by *EliasGwinn* /img/forum/go_quote.gif 
_The DAC1 achieves 21-bits of signal-to-noise ratio. 

 The DAC1 can accurately resolve the 24th-bit (although it will be below the noise floor)._

 

Interesting statement, Elias.
 I already asked myself why Benchmark doesn't list the dynamic range of the DAC1. Although this value is often used equally to the signal to noise ratio, some seem to distinguish between them. For example, according to the mastering engineer Bob Katz, one can hear details below the noise level, thus the dynamik range can be greater than the SNR, especially in conjunction with dithering (as far as I remember he gave 91 dB SNR and ~ 116 dB dynamik range for properly dithered 16 bit material).

 Now it would be interesting to know how great the dynamic range (!) of the DAC1 actually is. If it should be really able to resolve the 24th bit, it would have to exceed 140 dB. Is this the case?

 I'm confused also why more than 20 Bit of wordlength are used, at all if no converter is actually able to reach such a huge SNR and dynamic range. Many devices don't even match 20 bit performance (by pure math).

 I'm sure you can clarify this. Again this would be worth an own thread.

 little-endian


----------



## Crowbar

Quote:


  Originally Posted by *little-endian* /img/forum/go_quote.gif 
_I'm confused also why more than 20 Bit of wordlength are used, at all if no converter is actually able to reach such a huge SNR and dynamic range. Many devices don't even match 20 bit performance (by pure math)._

 

24 bits is more than the dynamic range of the human ear. The reason for long words is most types of DSP processing effectively reduces the resolution, even if the DSP internally uses more bits. You can easily see that with an image editing program. Take a look at the histogram of an image, then apply some global processing such as equalization, and take a look at the histogram again. It will no longer be smooth and continuous; the effective quantization is worse.
 192 kHz sampling rates also don't make sense from an audibility standpoint, but they could be useful to get increased effective resolution by dithering, and so it goes to the same point as the increased wordlength.


----------



## CanMad

My apologies to everyone, I didn't mean to thread cr@p. 

 It would seem however that a transformer based output stage might not be the most accurately measuring. Therefore I understand that this would be undesirable in a device such as the Benchmark which is striving to achieve the most accurate performance.

 However this is from the Townshend web page for peoples info. Make of it what you will :
*Audio Amplifier
 Normal practice is to use integrated circuit operational amplifiers in the audio signal path. Unfortunately, there are serious problems, even with the “Best” audio grade devices. The first problems are over-complication. These devices may contain up to 1000 transistors and resistors every one of which has the capacity to loose a minute amount of fidelity. Secondly, the myriad resistors are not perfectly linear at very low voltages. The result is a component which has slight veiling and “grunge” distortion. Our simple discrete component fully class A operational amplifier simply doesn’t suffer from this.
 Even the best mix of dual monolithic junction field effect transistors and bipolar junction transistors in a single ended pure class A configuration with optimum global feedback for the highest linearity and lowest distortion was still not good enough, so now the gain is provided by our unique EDCT wired step up transformer and unity gain buffer to eliminate all high order harmonics to bring the ultimate in fidelity. THD and IMD measured at better than -120dB.* 

 I'd still be interested if Benchmark considered a discrete output stage though?


----------



## Crowbar

Quote:


  Originally Posted by *CanMad* /img/forum/go_quote.gif 
_Not to mention all the VERY positive reviews?_

 

Reviews weren't based on blind tests, so they are irrelevant. Those people are hearing little but their psychological bias.

  Quote:


 Maybe the problems you mention are why they sound so good (I suspect probably not). 
 

Some people like the sound of coloration. Why do you think Grado headphones are popular?

  Quote:


 Anyway the question was more about a discrete output stage avoiding op-amps than specifically a transformer based one. 
 

I prefer discrete stages myself, and I have no problem with that. But the suggestion to add a distortion-producer a.k.a. transformer had to be addressed.

  Quote:


 I do realise that there are very good op-amps these days, but I don't think the Benchmark uses the latest and greatest (correct me if I'm wrong Elias). 
 

I think the new LM4562 opamp used in the latest version is very nice. It's already very linear before any feedback, which is not commonly seen in opamps.

  Quote:


 When are you starting your own hi-fi company 
 

Is that a challenge? One doesn't need to sell commercial equipment to exercise in electronics. I'm a long time DIYer and I've no doubt I and a number of other DIYers around this and other forums can build a DAC that beats even Lavry and MSB stuff for much less of the cost. As a DIYer I have none of the economic considerations of a company that has to maximize profit.

  Quote:


 as you obviously know more about engineering than Charles Hansen and Max Townshend? 
 

You're embarassing yourself with an _argumentum ad verecundiam_ here. I recommend taking a basic critical thinking course at your nearest college.

 But if you're going to go this route, I suggest you research the enormous engineering effort that has gone towards eliminating transformers from the audio signal path, with the numerous OTL tube amp designs from a variety of companies, and multiple improvements of the technology over time such as Rozenblit's Transcendent OTL, and culminating in Berning's ingenious ZOTL circuit. Speaking of Berning, he has a great visual demonstration of the evils of transformer coupling; take a look at Figure 3 to see the distortion and hysteresis caused by a transformer in the signal path: http://www.davidberning.com/Transfer%20Char.htm

  Quote:


 Thanks for focusing on the one part of the question that you thought was bad.....like a politician! 
 

Why focus on the other parts, if I didn't see anything wrong with them? Attention should be paid where there are improvements to be made.


----------



## lowmagnet

Guys, please cut it out. This is about the only thread I subscribe to because I'm tired of everything going downhill, and you two and your personal issues aren't helping any.


----------



## tszyn

Thanks to all for this very informative thread! I have posted a number of questions to the "computers-as-source components" forum in the hopes of taking care of a few loose ends in the whole KMixer discussion.

http://www.head-fi.org/forums/showthread.php?p=2962741

 I would greatly appreciate it if Elias or other experts who participated in this thread would provide their input.

 Thank you,

 Tomasz


----------



## little-endian

@Crowbar

 Of course I understand that it makes sense to user higher wordlengths and sample rates to have more reserves for processing audio material (rounding errors, headroom, etc.) but it doesn't explain its usage for pure digital to analog conversation for the endusers. I accept higher sample rates, used internally to simplify filtering, etc. but I don't see it for the wordlengths. Besides that, the "joke" is, that higher sample rates can be reproduced on the analog side, at least (although not hearable) but this is not the case for huge wordlengths. As far as I know, these 144 dB, 24 bit - sources theoretically provide, can't be really reproduced due to the noise of the signal path (resistance, etc.).

 Correct me if I'm wrong.


----------



## Crowbar

Yes. You'd have to use bulk metal foil resistors and very low noise transistors, maybe JFETs, etc. Well, JFETs are cheap, but the bulk foils... last time I checked Vishay was charging several dollars a piece 
	

	
	
		
		

		
			





 But 144 dB is not needed. That's more than the ratio of threshold of audibility to hearing damage. Even to accomodate large transients, you shouldn't need more than 15-20 dB less than that figure. More attention should be spent on the analog stages to preserve as much as possible of what the D/A gives you. Not to mention that a speaker unless electrostatic or plasma, will usually have distortion on the order of a percent.


----------



## little-endian

Of course, from the point of view in regard to the human ear's dynamic range, I want to claim that even 16 bit resolution is more than enough. No single track makes use of it thus the only benefit is the great SNR as far as I can see. One demo track (chinese drums) from a Burmester CD uses actually 40 dB of dynamic range which is already enormous! So I'm just interested if there is even one single device which reaches a signal to noise ratio of more than 144 dB. Otherwise the whole 24 bit fuss (for DACs) seems to be quite ridiculous to me.

 How low the dynamic range is allowed to be while still being able to deliver great music dynamic shows the LP - some people even prefer it to digital sources (which seems to be completely unfounded since no test I'm aware of compared the direct vinyl source versus one cascaded through a potent ADC and DAC, but just different masterings). I bet, no one would pass a blind test!


----------



## Crowbar

Quote:


  Originally Posted by *little-endian* /img/forum/go_quote.gif 
_No single track makes use of it_

 

That is false. I have a number of 24 bit 96 kHz recordings on DVD (not DVD-Audio) that have a dynamic range exceeding 16 bits. Despite the non-nil noise floor in some, as was already explained above, dynamic range beyond the noise floor can still make an audible difference.
 When you say a CD uses 40 dB, you must be looking at the power envelope variation between quietest and loudest passages, whereas the texture of an instrument is made up of the myriad harmonics that cover a much larger range. Even without this, just the power envelope in a classical orchestra typically does 100 dB, with potentially even larger transients.
 Just because most recordings are not of quality is not a reason not to support the ones that are, as few as they may be. Plus, if the playback gear support is there, the number of high resolution recordings will increase if people would bother to buy quality stuff.


----------



## little-endian

Interesting objection, Crowbar. Maybe I should have formulated it less absolute. I was referring to the fact that the level meter doesn't fall below -40dB within a track on any CD I tried, so far. Thinking twice, this would have to include any noise which is always present. Taking into account that one can resolve details below a noise what you've said makes sense of course. So is it all noise which prevents a level meter to go as low as the dynamic range actually is like you claim? I'm always willing to learn so how to you measure the real dynamic range of a given track (or parts of them)?

 However, one could question your statement that your recordings exceed 16 bits. If dithered 16 bit reaches over 116dB (don't remember the exact value right now) - then, do they really? 

 Cheers!


----------



## Crowbar

The effective dynamic range would be the difference between the tallest transients and the lowest signals which can be perceived, so ten or so dB below the noise floor. Use a wave editing program to see it visually; it may even have a function to estimate the dynamic range. The noise floor is easiest to see if there's a passage where no other sound exists, otherwise you sort of need to guess its level. Select that quietest passage which should be only the noise and compute the RMS in the program. Take off say 10 dB for sub-noise signals and compute the range.
 By the way, 16-bit DACs are generally not accurate to the LSB. And relying on dithering doesn't make sense with 44.1 or 48 kHz as all the bandwidth is needed to have sufficient sampling above Nyquist with margin for the filter; you don't have any spare (also of course the recording itself would have to be dithered, that information has to be encoded in there).

 Finally, I've mentioned research before allowing for the possibility that supersonic transients can influence perception. It's unfortunate that in general the analog filters start early on (both ADCs and DACs), when with oversampling and steep filters that is not necessary. It would be interesting to explore sound bandwidth higher than 20 kHz more than the few studies and experiments done that leave the issue inconclusive.


----------



## music_man

mr. gwinn,

 could you please try to answer some questions i asked in this thread if you have time? on page 26,post #506,5/15/2007. 

 thank you,
 music_man


----------



## EliasGwinn

Music Man,

 I'm sorry I missed this question. Thanks for following up, and please do the same if I miss another in the future.

  Quote:


  Originally Posted by *music_man* /img/forum/go_quote.gif 
_mr gwinn,

 is the dac circuit connected to the headphone amp circuit balanced on the pcb? if not, why and how is it connected? edit: it appears that it is unbalanced?_

 

The D/A chip provides a true balanced output for each channel. These balanced signals are sent through a differential amp just before headphone amplification. 

  Quote:


  Originally Posted by *music_man* /img/forum/go_quote.gif 
_also, what is the headphone amp using in place of an op-amp as the amplifier? a buffer ic? edit: wasn't it discussed that the headphone amp was discrete? there is mention of op-amps here._

 

The headphone amp is driven with the BUF634: http://www.ti.com/lit/gpn/buf634

 The BUF634 is an amazingly capable buffer with 2000V/us slew rate. It can drive an wide array of loads without suffering from performance loss.

  Quote:


  Originally Posted by *music_man* /img/forum/go_quote.gif 
_edit: is there a pot on the pcb to adjust the headphone amps gain?_

 

There is no pcb pot to adjust headphone gain. The DAC1 USB has two discrete gain settings: the DAC1 Classic gain, and a gain setting which is 10 dB less. The user can set this with jumpers on the pcb.

 Thanks,
 Elias


----------



## music_man

thank you for answering, mr. gwinn.

 music_man


----------



## audioengr

Quote:


  Originally Posted by *EliasGwinn* /img/forum/go_quote.gif 
_Music Man,

 I'm sorry I missed this question. Thanks for following up, and please do the same if I miss another in the future.



 The D/A chip provides a true balanced output for each channel. These balanced signals are sent through a differential amp just before headphone amplification. 


 The headphone amp is driven with the BUF634: http://www.ti.com/lit/gpn/buf634

 The BUF634 is an amazingly capable buffer with 2000V/us slew rate. It can drive an wide array of loads without suffering from performance loss.



 There is no pcb pot to adjust headphone gain. The DAC1 USB has two discrete gain settings: the DAC1 Classic gain, and a gain setting which is 10 dB less. The user can set this with jumpers on the pcb.

 Thanks,
 Elias_

 

The details: There are two BUF634's on the board for driving the headphone outputs. The op-amps that drive the BUF634's (inside the feedback loop) are SE and the op-amps that drive them are also SE, and the op-amps that drive them are also SE. From the headphone jacks you have to go back through 4 buffers before you get to the balanced op-amps that do the post-D/A filtering. IMO, the advantage of balanced outs for headphones is primarily at the output buffer, not in the previous stages, so I'm confused as to why this is revelant anyway.


----------



## music_man

so the headphone amp is connected to the dac ic in a single ended configuration?

 i also understand that the headamp circuit is not "discrete"? buffers and op-amps are ic's not transistors.

 mr. gwinn, please explain.

 music_man


----------



## audioengr

Quote:


  Originally Posted by *music_man* /img/forum/go_quote.gif 
_so the headphone amp is connected to the dac ic in a single ended configuration?

 i also understand that the headamp circuit is not "discrete"? buffers and op-amps are ic's not transistors.

 mr. gwinn, please explain.

 music_man_

 


 Music Man - I dont understand this line of questioning. It is a combination of both SE and balanced. You can have balanced op-amps in the D/A filtering/buffering and you can have balanced drivers for the headphone outputs, but from my experience they are independent things and each has their own advantages and disadvantages. Each can be created in almost any design. Either one can also be optimally designed or poorly designed. Just because there might be balanced configuration driving the SE headphone output drivers, this does not make it any better. The final result depends entirely on the implementation. It is possible to design a balanced headphone driver that sounds worse than a single-ended one. And therefore balanced is not the panacea that one would think it is. Likewise, good discrete design can be much more difficult than op-amp design, so it usually requires a lot of experience to get a good result.

 It is my experience that audiophiles that have limited technical knowledge often grab-onto concepts and devices and believe that if these are present in a circuit, it automatically makes the design superior. I can assure you that this is usually not the case. Again, it depends on the implementation as to whether the advantages of the concept or device are actually realized in the design. I can give you countless examples of designs that use great chips and sound lousy. This is where the experience of the designer comes in.

 All designers of course are experts in every aspect of the design process including PCB layout, parts selection,analog circuit design, digital circuit design, transmission-lines, grounding and shielding, ESD, EMI and power delivery.
	

	
	
		
		

		
		
	


	




 About 15-20 years ago, I finally realized that I did not know everything myself. I've been learning ever since.
	

	
	
		
		

		
		
	


	




 I'm not saying that balanced headphone output drivers is not a good idea, because I believe it is. What I am saying is that you either have it or you dont. There is no in-between that is somehow magically close. And a balanced driver can be added easily (given the real-estate) to a totally SE circuit and reap all of the benefits.

 Steve N.


----------



## music_man

yes, i understand all that. that is not what i was asking though. acording to you, the amp is neither balanced nor discrete. i was simply reiterating your statement. since the post above yours, by mr. gwinn says it "is" indeed a balanced configuration. you said it was not. i did not ask if one was better than the other 
	

	
	
		
		

		
		
	


	




 also, i seem to remember somewhere here, more than once it was mentioned that the headphone amp circuit was discrete. not by mr. gwinn. 
 apparently it is not discrete as it uses ic's both in it's input and output stages. again, i did not ask or state as to if this was a better design! i simply asked whether it was indeed discrete or not! 
 i will wait for mr. gwinn to answer.

 mr. nugent. i know benchmark is weary of this but i'd like to ask you. what do you do to (feel you have improved) the headphone amp in the dac1? to what level do you feel it can be improved(compare in dollar amount or to a comparable other amp please)? your site mentions all your mods but is vague about the amp mods.

 i don't think their head amp is as bad as some people say. please realise the board alone is $150. the amp that features it is $450. the dac1 has a much better power supply than benchmarks stand alone headphone amp(h1). i do not see how the dac1 is any different than a corda opera or grace m902(in function, not in specific sound). it is a bonafide one box combo. what one prefers in a sonic signature is up to themselves. i see the dac1 as square competition for any other popular one box combo. ymmv.

 music_man


----------



## audioengr

Quote:


  Originally Posted by *music_man* /img/forum/go_quote.gif 
_mr. nugent. i know benchmark is weary of this but i'd like to ask you. what do you do to (feel you have improved) the headphone amp in the dac1? to what level do you feel it can be improved(compare in dollar amount or to a comparable other amp please)? your site mentions all your mods but is vague about the amp mods.
 music_man_

 

I suppose it is vague. I like to keep as much as possible the mods I do a trade secret. As I mentioned before, I discovered that the BUF634 is a difficult device to feed in terms of power delivery, and the topology that is recommended by TI tends to aggravate this IMO. I actually had a mod for the volume/headphone for a long time, but I wasn't happy with the level of improvement and could not recommend it to my customers. I only had a couple of orders for this. Then, I revisited it a year later (mostly as a result of the Head-Fest) and developed some improvements that are significant enough that I now recommend this mod. The improvements consist of:

 1) rewiring of some traces
 2) some redesign of the circuitry around the volume pot
 3) replacing all of the op-amps
 4) power delivery improvements for the op-amps
 5) power delivery improvements for the output drivers

 There is barely enough space to accomplish all of this, which makes the rework difficult, and a bit expensive. I do not recommend this headphone section mod stand-alone. There is a subset of the Turbomod that should be combined with this.
 I demonstrated this mod at the Head-Fest in San Jose and Ray Samuels commented that it was quite good. It is not as good as a Raptor though. I want one of those.

 BTW, a really nice Head-fier brought his grace amp over to compare both the head-amp in it and the DAC to my modded DAC-1 about 2 weeks ago. I will see if he is willing to report back. He also brought me some awesome new music.

 Steve N.


----------



## music_man

in their stock form i like the graces headamp better and the benchmarks dac better. for enjoyment listening. this would go along with their intended purposes. the dac1 is sold as a dac and the grace a headamp.

 i do not mean any insult to benchmark. in this hobby we are all looking for a sonic signature each of us find pleasing. it is different for everyone. not everyone ever agrees that one specific amp sounds the best. in the studio the dac1 does exactly the job it is supposed to do. remember, the dac1 was originally designed for engineers. it just so happens that it became popular here as well. those that bash it in any way are looking for a different sonic signature. not finding fault with it as an engineering device. you can hear everything with it and i do mean everything. in that respect job acomplished. most people here consider exagerated and warm to be better. if you are looking for pure resolving accuracy you will spend over $15,000 to do better imo.

 music_man


----------



## EliasGwinn

Music_man,

 Just to clear up what you asked before, the DAC1 circuitry is not transistor based. 

 Also, the D-to-A chip has a (true) balanced output. This balanced output is then fed into a differential amp. The output is the difference of the + signal and the - signal. You now have a single-ended signal with 6 dB more amplitude and with the common-noise rejected. This single-ended signal is then sent on to the headphone circuitry.

 Hope that answers your questions...

 Thanks, 
 Elias


----------



## puntloos

Gents, (m/f)

 It's been an amazing topic so far, cool that actual 'savants' and designers are willing to put up with the (semi) laymen as well as pro discussion.

 One big question is this:

 Why use USB at all? I mean sure, if you're building a new computer or using a laptop, perhaps it is worth considering, but personally I have a M-Audio revolution 5.1 card, which I bought specifically for its ASIO/Kernel Streaming options paired with full 192Khz/24b SPDIF output. 

 Given that someone still has the choice, isnt the SPDIF/AES road the better one?

 Additionally, to the Benchmark people in here: 
 - Could you elaborate on your choice for sticking to upsampling (or downsampling!) to 100-ish Khz instead of going to 192Khz? Were the DAC chips back when you designed it still incapable of reaching 192K safely? By now, the Burr Brown PCM1796 sounds like the better chip. Additionally, modern media like HD-DVD and even DVD-A, SACD are capable of (and I believe sometimes using) the full 24/192/uncompressed, and downsampling hurts.. well somewhat. right? (Shameless plug, I've posted a related question regarding the impact of things like jitter here:  Applying science to audiophile voodoo?)
 - A lot of companies (unlike yours, it seems) justify the existence of their $23189471239487 DACs by claiming they use better technology, higher quality parts etc. Would you say that your DAC can take the pepsi challenge with 'any conceivable DAC out there' and win (or be equal) when it comes to exact reproduction)? Or are there DACs that outperform yours, if cost would not be a factor at all. 
 - Given that I don't care for USB (pending your answer on my previous question 
	

	
	
		
		

		
			





 ), I am planning to use a DAC to drive my system: a -passive- preamp (Adcom GFP-750 - which as I understand it is not much more than an attenuator('resistor') and a few relays) linked to a Bel Canto Evo 4 gen2 (input impedance: 200KOhm) powering Quad 989 speakers. Should this work, or is this not advisable (and if so, why not?)
 - One big question regarding Jitter in general.. something that has been burning in my mind for a while: Why don't DACs really buffer? When we are talking about CD players then I can see the need for a 'just in time' solution where instant reaction speeds are important, and error correction needs to make a best guess here-and-now. But for a DAC, why not just buffer 0.3 seconds before starting playback and run the playback off your own clock? If you're scared of noticable buffer sync issues you could allow for a mechanism to sliiightly tweak up or down your clock speed over time if things start to go wrong.. 

 Speaking of Pepsi Challenges, some claim their mods to the benchmark make it a better device. The request for this has been quietly ignored so here it is again: Audioengr: would you be willing to lend Benchmark one of your modded DACs, and Benchmark ppl: would you be willing to compare it to your stock model?


----------



## audioengr

Quote:


  Originally Posted by *puntloos* /img/forum/go_quote.gif 
_Gents, (m/f)Why use USB at all? I mean sure, if you're building a new computer or using a laptop, perhaps it is worth considering, but personally I have a M-Audio revolution 5.1 card, which I bought specifically for its ASIO/Kernel Streaming options paired with full 192Khz/24b SPDIF output. 

 Given that someone still has the choice, isnt the SPDIF/AES road the better one?

 Speaking of Pepsi Challenges, some claim their mods to the benchmark make it a better device. The request for this has been quietly ignored so here it is again: Audioengr: would you be willing to lend Benchmark one of your modded DACs, and Benchmark ppl: would you be willing to compare it to your stock model?_

 


 There are several reasons to use USB:

 1) The clock and power system of the computer is a noisy one in most cases, so moving the clock generation and power supply outside the computer is a big advantage to keeping the signals noise-free and the clock low-jitter

 2) there are a couple of excellent USB audio chips from TI that have very clever PLL design that can result in much lower jitter than that possible from a Transport. One of these is the TAS1020 that I use and Benchmark uses.

 3) S/PDIF is inherently jittery due to the encoding of clock and data into one signal and recovery of the clock and data at the receiver chip. There is usually another PLL involved in this decoding. If you can avoid this encoding/decoding process, it is a big win.

 4) S/PDIF inherently has cable length limitations because the cable contributes to the jitter significantly and directly. USB cable has less effect on the jitter due to the excellent PLL at the USB converter (assuming that a low-jitter clock is driving the chip).

 Here are some more things to read:

http://www.positive-feedback.com/Issue22/nugent.htm

http://www.positive-feedback.com/Issue14/spdif.htm

 As for the shoot-out, I'm all for it, however it should be performed by a 3rd party, not by Benchmark or myself. The 3rd party should have a sufficiently resolving full-range system (not a sub-satellite for instance).

 When I think about it, this has already been done by several of my customers that have multiple DAC-1's. They get one modded and then compare it to their stock unit:

http://www.audioasylum.com/audio/dig...es/129150.html

http://www.head-fi.org/forums/showpo...7&postcount=32

http://www.audioasylum.com/audio/dig...es/107696.html

http://www.audiocircle.com/circles/i...?topic=37963.0

 These of course are anecdotal only. Unless I actually hear the system that will be used for the shoot-out proposed, I do not trust it to be good enough. I have found that even many reviewers systems are sub-par and very colored.

 Scenerio 1
 As for buffering serial data in DAC's, this is not as easy as it might seem. If there is a master clock in the DAC and this clock can be driven back to the source, which is a slave device and uses this clock, then the design can be straightforward. Some expensive and professional DAC's do this, usually with a word-clock driven back to a transport. There is no PLL required to clock the FIFO output in this type of design, so it can be extremely low-jitter.

 Scenerio 2
 If the source clock cannot come from the DAC, such as a computer driving USB, then the receiving device must lock onto the arriving PC-generated clock using some type of PLL. The DAC cannot be the source of the clock (assuming no upsampling). If the DAC clock tracks the PC clock, then the FIFO memory in the DAC will not overrun or underrun. The PLL makes sure that the outgoing clock from the FIFO is closely matched to the PC clock. The problem is that the PLL adds it's own jitter, and can never be completely independent of the PC clock.

 One instance where Scenerio 1 can be implemented for computer audio is the network case. If ethernet protocol is used, either wirelessly or wired, then the clock at the receiving device becomes the source clock. This clock can be generated without PLL's and a FIFO data buffer can be used. This is something that I'm doing with the Squeezebox3 and my Pace-Car FIFO reclocker.

 Scenerio 1 might also be implemented with a PCI audio card with a master clock on it. If the master clock were driven from the DAC, then this would work well also. The master clock would have to be implemented in the DAC.

 Steve N.


----------



## puntloos

Hey Steve, thanks for the reply so far.

  Quote:


  Originally Posted by *audioengr* /img/forum/go_quote.gif 
_There are several reasons to use USB:

 1) The clock and power system of the computer is a noisy one in most cases, so moving the clock generation and power supply outside the computer is a big advantage to keeping the signals noise-free and the clock low-jitter

 2) there are a couple of excellent USB audio chips from TI that have very clever PLL design that can result in much lower jitter than that possible from a Transport. One of these is the TAS1020 that I use and Benchmark uses.

 3) S/PDIF is inherently jittery due to the encoding of clock and data into one signal and recovery of the clock and data at the receiver chip. There is usually another PLL involved in this decoding. If you can avoid this encoding/decoding process, it is a big win.

 4) S/PDIF inherently has cable length limitations because the cable contributes to the jitter significantly and directly. USB cable has less effect on the jitter due to the excellent PLL at the USB converter (assuming that a low-jitter clock is driving the chip).

 Here are some more things to read:

http://www.positive-feedback.com/Issue22/nugent.htm

http://www.positive-feedback.com/Issue14/spdif.htm
_

 

Curious, I was under the impression that the way USB was designed was mostly with data bursts in mind (i.e. lots of priority and scheduling stuff that was hard to control). Also, I think Im misunderstanding you cause you say it is bad that the clock is at the computer (noise by fans/hdds) but also that the USB system has the dac here. 

 Anyway, with for example the benchmark dac1, the effects of jitter should be 'solved' right?

  Quote:


 As for the shoot-out, I'm all for it, however it should be performed by a 3rd party, not by Benchmark or myself. 
 

Benchmark sounds like a fine party to me, since they will sell one DAC for each buy, modded or not. They have little to lose here (other than time) since you will have to buy a DAC from them first, before you can mod. 
	

	
	
		
		

		
		
	


	



  Quote:


 The 3rd party should have a sufficiently resolving full-range system (not a sub-satellite for instance).

 These of course are anecdotal only. Unless I actually hear the system that will be used for the shoot-out proposed, I do not trust it to be good enough. I have found that even many reviewers systems are sub-par and very colored.

 

Hey, send me your modded dac, I plan to audition multiple dacs soon since my old one - (Philips IS 5022) blew up 
	

	
	
		
		

		
		
	


	




 I've built my stereo with accurate reproduction in mind.

 Source: M-Audio Revolution 5.1 (spdif out, kernel streaming, 24bit upconvert)
 DAC: TBD
 Interconnects: Paul Speltz' Anti-ICs (balanced Neutrik) 
 Pre-amp: Adcom GFP-750 in Passive mode (so it's basically just an attenuator)
 Interconnects: Paul Speltz' Anti-ICs (balanced Neutrik) 
 Power Amp: Bel Canto Evo 4 gen II
 Speaker Wire: Paul Speltz' Anticables
 Speakers: Quad ESL 989's

 DACs currently on my list:

 * Benchmark DAC1 (not sure if I want USB or non-USB) ($1000)
 + Good price
 + Excellent anti jitter
 - Samples everything to 100ish Khz (even 192Khz DVD-A sources 
	

	
	
		
		

		
		
	


	




)
 - Upsampling can't be turned off

 * Bel Canto DAC3 ($2500)
 + Resamples to 192Khz
 + Has the same UltraLock anti jitter as the Benchmark
 + Better input stages than Benchmark
 +/- More modern DAC (BB 1792's).. Potentially better
 +/- Possibly handles 192Khz sources without resampling.. not sure, docs are fuzzy
 - Price ($2500)
 - Resampling can't be turned off

 * AQVox USB 2 D/A MKII ($1000) 
 + Resamples to 192Khz or 'bypass' (no resampling)
 + Handles 192Khz sources
 +/- BB 1796 DAC's, which are inferior to the Bel Canto's, but possibly better than Benchmark
 - Less documentation on jitter compensation, probably inferior to DAC1/DAC3
 - Possibly inferior output stage... might become an issue with passive pre-amp.

 As you can tell, this '192Khz source support' is an issue for me.. modern sources (DVD-A, SACD (in a way, it doesnt use PCM), HD-DVD and BlueRay) all support 192/24bit, so buying something that does less puts you a bit behind the times from the get go.

  Quote:


 Scenerio 1
 As for buffering serial data in DAC's, this is not as easy as it might seem. If there is a master clock in the DAC and this clock can be driven back to the source, which is a slave device and uses this clock, then the design can be straightforward. Some expensive and professional DAC's do this, usually with a word-clock driven back to a transport. There is no PLL required to clock the FIFO output in this type of design, so it can be extremely low-jitter.

 Scenerio 2
 If the source clock cannot come from the DAC, such as a computer driving USB, then the receiving device must lock onto the arriving PC-generated clock using some type of PLL. The DAC cannot be the source of the clock (assuming no upsampling). If the DAC clock tracks the PC clock, then the FIFO memory in the DAC will not overrun or underrun. The PLL makes sure that the outgoing clock from the FIFO is closely matched to the PC clock. The problem is that the PLL adds it's own jitter, and can never be completely independent of the PC clock.

 One instance where Scenerio 1 can be implemented for computer audio is the network case. If ethernet protocol is used, either wirelessly or wired, then the clock at the receiving device becomes the source clock. This clock can be generated without PLL's and a FIFO data buffer can be used. This is something that I'm doing with the Squeezebox3 and my Pace-Car FIFO reclocker.

 Scenerio 1 might also be implemented with a PCI audio card with a master clock on it. If the master clock were driven from the DAC, then this would work well also. The master clock would have to be implemented in the DAC.

 Steve N. 
 

Makes sense, mostly, although well.. Im sure it is still a touchy subject and benchmark/belcanto claim the issue of jitter is moot if you use their products.. Also, again you go for the 'clock' route, where I was suggesting 'better buffering by reading ahead at more than 1x speed'


----------



## audioengr

Quote:


  Originally Posted by *puntloos* /img/forum/go_quote.gif 
_Curious, I was under the impression that the way USB was designed was mostly with data bursts in mind (i.e. lots of priority and scheduling stuff that was hard to control). Also, I think Im misunderstanding you cause you say it is bad that the clock is at the computer (noise by fans/hdds) but also that the USB system has the dac here. 

 Anyway, with for example the benchmark dac1, the effects of jitter should be 'solved' right?_

 

USB has several modes of operation that have different protocols: Asynchronous, Block/Burst and Isochronous (AKA Synchronous Adaptive). Printers use a different protocol than streaming audio. Early development by TI set the stage for streaming audio, when they tried different protocols and decided that Synchronous Adaptive was the best performer with the least probability of drop-outs. Thier chips are designed around this protocol, although others can be implemented with some chips. Most USB adapters work this way.

 As for the jitter in a stock DAC-1, this is addressed in their design by using asynchronous upsampling like many other DAC's. It certainly does reduce the jitter of the incoming stream, probably better than many DAC's.

 However, this technique does not totally eliminate jitter IME. When I remove the upsampling chip and replace it with an I2S interface driven by my FIFO reclocker, the audible jitter is noticable lower.

 Steve N.


----------



## little-endian

Quote:


  Originally Posted by *puntloos* /img/forum/go_quote.gif 
_As you can tell, this '192Khz source support' is an issue for me.. modern sources (DVD-A, SACD (in a way, it doesnt use PCM), HD-DVD and BlueRay) all support 192/24bit, so buying something that does less puts you a bit behind the times from the get go._

 


 When cosidering the human's hearing capability, the only benefit of higher sample rates is to be able to user larger ranges for filtering cutoffs, preventing aliasing more easily and so on. The sampling itself is lossless in theory as long as frequency input is limited and thus complies with the theorem.

 So, upsampling to 192 kHz or oversampling in general maybe makes conversation easier to implement and increases the quality. Otherwise, Benchmark claimes that exactly this performance would be better when using about 110 kHz instead of 192 kHz, even if some analog bandwidth will be lost this way. But hey, we dont' hear it anyway and ultrasonic influences have never be proven so far. I doubt you would hear any diferrence, but perhaps you will feel better, knowing that the music arrives to you in its original sample rate.

 Cheers!


----------



## zheka

Quote:


  Originally Posted by *puntloos* /img/forum/go_quote.gif 
_As you can tell, this '192Khz source support' is an issue for me.. modern sources (DVD-A, SACD (in a way, it doesnt use PCM), HD-DVD and BlueRay) all support 192/24bit, so buying something that does less puts you a bit behind the times from the get go._

 

how can any of this formats be fed into stereo dac, w/ 192Khz support or not? am i missing something?


----------



## audioengr

Quote:


  Originally Posted by *little-endian* /img/forum/go_quote.gif 
_When considering the human's hearing capability, the only benefit of higher sample rates is to be able to user larger ranges for filtering cutoffs, preventing aliasing more easily and so on. The sampling itself is lossless in theory as long as frequency input is limited and thus complies with the theorem._

 

I do not agree. Higher sample rates do sound more like analog to me. Believe it or not, even vocalists sound much more smooth and natural with upsampling to 24/96. 24/192 is even a skosh better than that IMO. The upsampler must use a good algorithm though. Not all upsampling chips are equal. Likewise, computer upsampling codes all sound different, with a few exceptional ones. I find the original SRC code to 24/96 to be exceptionally good, and so do the reviewers that have reviewed my products. They all listen to the 16/44.1 and then the upsampled to 24/96 and never go back to the 16/44.1 again.

 Steve N.


----------



## Crowbar

Quote:


  Originally Posted by *audioengr* /img/forum/go_quote.gif 
_S/PDIF is inherently jittery due to the encoding of clock and data into one signal and recovery of the clock and data at the receiver chip. There is usually another PLL involved in this decoding._

 

In asynchronous isochronous mode USB Audio has no jitter issue whatsoever, since USB clocking has nothing to do with the audio clock. Jitter's only effect then could be data corruption, and that takes a lot of jitter, though it can be an issue since USB Audio has no error correction, and error rates on some Windows systems tend to be high.

  Quote:


 If the source clock cannot come from the DAC, such as a computer driving USB, then the receiving device must lock onto the arriving PC-generated clock using some type of PLL. The DAC cannot be the source of the clock (assuming no upsampling). 
 

Asynchronous isocrhonous USB Audio mode is computer driven, but the DAC provides flow control, thus the audio clocking is entirely up to the DAC. Therefore, your statement is false. Indeed, Wavelength who's another head-fi member builds and sells USB DACs that support the asynchronous USB mode and thus have no interface jitter without resampling, so it's just a matter of implementing the standard in the firmware of the USB controller being used.

  Quote:


 One instance where Scenerio 1 can be implemented for computer audio is the network case. 
 

Why bother, when you can do that with USB? The USB Audio standard has supported asynchronous isocrhonous mode from the beginning!

  Quote:


  Originally Posted by *puntloos* /img/forum/go_quote.gif 
_Hey Steve, thanks for the reply so far.
 Curious, I was under the impression that the way USB was designed was mostly with data bursts in mind (i.e. lots of priority and scheduling stuff that was hard to control)._

 

USB has other modes, such as bulk transfer, which support error correction, but lack streaming ability with guaranteed throughput to prevent underruns in cases of high system load.

  Quote:


 DACs currently on my list: 
 

I was going to recommend Wavelength's DACs because they eliminate jitter by using asynchronous modes, but then I noticed it's a non-oversampling DAC... Nooo... it's like getting one part perfect, and another one completely backwards... ;_;

  Quote:


  Originally Posted by *audioengr* /img/forum/go_quote.gif 
_USB has several modes of operation that have different protocols: Asynchronous, Block/Burst and Isochronous (AKA Synchronous Adaptive)._

 

Uh, you mixed them up here. Block and burst modes are standard data transfer modes for non-audio. USB Audio uses isochronous mode in one of three submodes, which are either synchronous, adaptive, or asynchronous.


----------



## Crowbar

Quote:


  Originally Posted by *audioengr* /img/forum/go_quote.gif 
_Believe it or not, even vocalists sound much more smooth and natural with upsampling to 24/96. 24/192 is even a skosh better than that IMO._

 

96 makes sense, but 192 doesn't and is just marketing. The only use from an engineering point of view is that it makes it easier to design the analog filter, but in practice DACs use the same analog filter for all modes. I suggest you do blind testing. You can easily do it with your computer using the abchr program. Take a 192 source and downsample it to 96 in a good sound program, then use these as the example files on the program. Run a set of tests, and I'm sure you won't be able to tell the difference unless the downsampling filter was poor quality.


----------



## little-endian

@zheka

 I'm not sure if I understand your question correctly. Are you referring to a specific DAC? For the DAC1, 192 kHz can be fed into via S/PDIF only (USB is limited to 96 kHz). 192 kHz transferred optically isn't supported by all devices, though. Many soundcards (and the older DAC1s) are limited to 96 kHz here as well.

 @audioengr

 Different sample rates may sound different due to different resampling or conversation performance at this given sample rate. However the common association of higher sample rates giving a more analog result after the reconstruction stage is actually wrong. A higher sample rate per se doesn't provide anything "more analog" but a higher niquist frequency thus a higher analog bandwidth. If one above 20 kHz is hearable or not is questionable. The error introduced by the sampling process lies in the limited bandwidth (which has to be limited to comply with the theorem) and the needed filtering but NOT it the actual missing information between the samples which are going to be reconstructed anyway.

 I just wanted to note that higher sample rates probably have practical benefits when it comes to the actual conversation, but that the actual information is already contained in 44,1 kHz for example (or even 40 kHz). At least from the ear's point of 'view'.


----------



## zheka

Quote:


  Originally Posted by *little-endian* /img/forum/go_quote.gif 
_@zheka
 I'm not sure if I understand your question correctly. Are you referring to a specific DAC? For the DAC1, 192 kHz can be fed into via S/PDIF only (USB is limited to 96 kHz). 192 kHz transferred optically isn't supported by all devices, though. Many soundcards (and the older DAC1s) are limited to 96 kHz here as well._

 

I understand that Benchmark DAC1 and many other DACs are capable of dealing with 24/192 PCM streams. however puntloos seems to imply that this somehow will allow playback of DVD-A, SACD, etc. material
 Correct me if i am wrong, but does not one need to use some sort of DSPs to work with multi-channel digital streams, and two channel DACs such as Benchmark are not meant for this kind of processing?
 in addition there are no transports that allow digital output of DVD-A and SACD data, i am not sure how that works with blue ray and hd-dvd.


----------



## lowmagnet

Quote:


  Originally Posted by *zheka* /img/forum/go_quote.gif 
_However puntloos seems to imply that this somehow will allow playback of DVD-A, SACD, etc. material
 Correct me if i am wrong, but does not one need to use some sort of DSPs to work with multi-channel digital streams, and two channel DACs such as Benchmark are not meant for this kind of processing?_

 

SACD uses DSD, which is incompatible and something like 2MHz sample rate. DVD-A can use LPCM or MLP so if you have a DVD-A with PCM data in 2 channel it _may_ work with the DAC1.


----------



## zheka

Quote:


  Originally Posted by *lowmagnet* /img/forum/go_quote.gif 
_ DVD-A can use LPCM or MLP so if you have a DVD-A with PCM data in 2 channel it may work with the DAC1._

 

AFAIK DVD-A players do not output hi-res digital signal due to some DRM regulations. 
 some might provide _downsampled_ digital signal which obviously defeats the purpose


----------



## gregeas

Good luck trying to get hi-res audio from a SACD or DVD-A. I've heard it can be done with DVD-A, but there are many, many steps involved... 

 The only real way you will have access to hi-res PCM files is if you have digitized LPs or come across master recordings from a studio. There are a few websites that offer hi-res downloads, but the options are extremely limited. Believe me, I've looked.

 Also, I don't see much content on the horizon for either SACD or DVD-A. I consider the formats to be virtually dead, except for classical music.


----------



## slwiser

Quote:


  Originally Posted by *gregeas* /img/forum/go_quote.gif 
_Good luck trying to get hi-res audio from a SACD or DVD-A. I've heard it can be done with DVD-A, but there are many, many steps involved... 

 The only real way you will have access to hi-res PCM files is if you have digitized LPs or come across master recordings from a studio. There are a few websites that offer hi-res downloads, but the options are extremely limited. Believe me, I've looked.

 Also, I don't see much content on the horizon for either SACD or DVD-A. I consider the formats to be virtually dead, except for classical music._

 

Check out DVDs by AIX records at www.aixrecords.com that provide DVD-A at 96/24 out of digital outputs into my Lavry DA10. Hi-Rez is their business.


----------



## slwiser

Quote:


  Originally Posted by *puntloos* /img/forum/go_quote.gif 
_

 - ...... Why don't DACs really buffer? When we are talking about CD players then I can see the need for a 'just in time' solution where instant reaction speeds are important, and error correction needs to make a best guess here-and-now. But for a DAC, why not just buffer 0.3 seconds before starting playback and run the playback off your own clock? If you're scared of noticable buffer sync issues you could allow for a mechanism to sliiightly tweak up or down your clock speed over time if things start to go wrong.. _

 

Some do buffer, some have a larger buffer than others while some don't have any buffer. The Chord64 has a 5 second buffer if I am not mistaken. My Lavry is built with some buffer but no one knows what size it may be.


----------



## puntloos

Quote:


  Originally Posted by *little-endian* /img/forum/go_quote.gif 
_When cosidering the human's hearing capability, the only benefit of higher sample rates is to be able to user larger ranges for filtering cutoffs, preventing aliasing more easily and so on. The sampling itself is lossless in theory as long as frequency input is limited and thus complies with the theorem._

 

Isn't it true that:

 1/ Tones in higher frequency ranges will result in harmonics in the audible range. Eliminate those tones, and you remove the audible resultant tones. 

 2/ At 'perfect' 44khz DAC will produce a block wave at 44khz. At 22khz (=an 11khz note) it will still be a very very blocky signal since it will only have 4 steps every cycle. Isn't a human ear able to distinguish this from a smooth sine? 

  Quote:


 So, upsampling to 192 kHz or oversampling in general maybe makes conversation easier to implement and increases the quality. Otherwise, Benchmark claimes that exactly this performance would be better when using about 110 kHz instead of 192 kHz, even if some analog bandwidth will be lost this way. But hey, we dont' hear it anyway and ultrasonic influences have never be proven so far. I doubt you would hear any diferrence, but perhaps you will feel better, knowing that the music arrives to you in its original sample rate. 
 

Well its as I said, all modern systems employ 192Khz now, which in my view is indeed probably enough for my ears until the end of time. Im very afraid that I would now commit to something fine-sounding but less-than-perfect and regret it later. (can you guys remember the time people called 128kbit mp3's CD quality?). 

 Putting it this way, considering say 'identical' DA convertors with the only difference being 192khz or 96khz, and if '192' has no actual -downsides- other than perhaps price, wouldn't you want the higher number? At a certain point, further 'improvements' go well beyond negligible into absurd, but until that time bigger is better! (you know what I mean)

  Quote:


  Originally Posted by *zheka* /img/forum/go_quote.gif 
_how can any of this formats be fed into stereo dac, w/ 192Khz support or not? am i missing something?_

 

Well first of all I just mentioned them as examples, trying to indicate that it seems to be a trend. Secondly most of those systems indeed do have stereo tracks, they are not 'by definition' multichannel.

 Interfacing DVD-A or SACD with a dac is often easy, they simply have spdif out too. If DRM is an issue, well.. where I live, if I buy a medium it is my right to play it the way I like it, including 'ripping to harddisk'.


  Quote:


  Originally Posted by *zheka* /img/forum/go_quote.gif 
_I understand that Benchmark DAC1 and many other DACs are capable of dealing with 24/192 PCM streams._

 

Thats kinda my point, -very few- can deal with 24/192, and the ones that do immediately downsample it (to somewhere near 24/96). Wether or not this is audible I can't tell, but it is at least theoretically inferior to the 24/192. 
  Quote:


 however puntloos seems to imply that this somehow will allow playback of DVD-A, SACD, etc. material
 Correct me if i am wrong, but does not one need to use some sort of DSPs to work with multi-channel digital streams, and two channel DACs such as Benchmark are not meant for this kind of processing?
 in addition there are no transports that allow digital output of DVD-A and SACD data, i am not sure how that works with blue ray and hd-dvd. 
 

SACD does use a different system than PCM, but that does not mean it can't be converted (mathematically) to its PCM equivalent. As for actually playing well let's not dive into DRM issues today and just state that legal or no it is definately possible to play a file, losslessly converted from SACD, in something like foobar2k.


----------



## EliasGwinn

WOW 
	

	
	
		
		

		
			





 !! The bursts of discussions on this forum amazes me!! I must say that I have enjoyed this thread very much and I appreciate all who have posted questions AND answers. These types of analytical discussions are essential for shared knowledge....this is what the internet was created for.

 I've got a whole lot of things to respond to...I'll try to do so in the most coherent way possible.

 I'll begin on a new post...and probably organize each post by subject.

 Thanks,
 Elias


----------



## zheka

Quote:


  Originally Posted by *slwiser* /img/forum/go_quote.gif 
_Check out DVDs by AIX records at www.aixrecords.com that provide DVD-A at 96/24 out of digital outputs into my Lavry DA10. Hi-Rez is their business._

 

very interesting. what are you using as transport?


----------



## slwiser

Quote:


  Originally Posted by *zheka* /img/forum/go_quote.gif 
_very interesting. what are you using as transport?_

 

I have a Oppo DV-981HD for my transport feeding my Lavry DA10. Using the AIX disks that I have the Lavry locks in at 96khz as it should.


----------



## audioengr

Quote:


  Originally Posted by *little-endian* /img/forum/go_quote.gif 
_@zheka
 @audioengr

 Different sample rates may sound different due to different resampling or conversation performance at this given sample rate. However the common association of higher sample rates giving a more analog result after the reconstruction stage is actually wrong. A higher sample rate per se doesn't provide anything "more analog" but a higher niquist frequency thus a higher analog bandwidth. If one above 20 kHz is hearable or not is questionable. The error introduced by the sampling process lies in the limited bandwidth (which has to be limited to comply with the theorem) and the needed filtering but NOT it the actual missing information between the samples which are going to be reconstructed anyway.

 I just wanted to note that higher sample rates probably have practical benefits when it comes to the actual conversation, but that the actual information is already contained in 44,1 kHz for example (or even 40 kHz). At least from the ear's point of 'view'._

 

Here has been my experience: When I play 44.1 through my Spoiler DAC, which does oversampling to a very high frequency, it still sounds more raspy than the same track upsampled using SRC to 24/96 through the same oversampler and analog filter. The 4-pole filter starts rolling off around 20kHz. I've measured it.

 Steve N.


----------



## zheka

@puntloos

 You basically arguing that there may be genuine 24/192 PCM material out there and down sampling to 110kHz will reduce quality. That's fine. Let me note that the real advantage of multi-channel hi-res audio lies in the fact that it's "multi-channel", not so much that it's "hi-res" . IMHO.

 On the side note, I did not know it's possible to rip SACDs. This of course is off topic here but would you please point me places where i can learn more about how that's done?


----------



## zheka

Quote:


  Originally Posted by *slwiser* /img/forum/go_quote.gif 
_I have a Oppo DV-981HD for my transport feeding my Lavry DA10. Using the AIX disks that I have the Lavry locks in at 96khz as it should._

 


 This probably has to do with the way AIX masters the disks.Good for them!

 I assume your player outputs other DVD-A disks only at 48kHz , correct?


----------



## slwiser

Quote:


  Originally Posted by *zheka* /img/forum/go_quote.gif 
_This probably has to do with the way AIX masters the disks.Good for them!

 I assume your player outputs other DVD-A disks only at 48kHz , correct?_

 

You are correct for protected DVD-As but again the AIX is open at 96kHz.


----------



## puntloos

Quote:


  Originally Posted by *zheka* /img/forum/go_quote.gif 
_@puntloos

 You basically arguing that there may be genuine 24/192 PCM material out there and down sampling to 110kHz will reduce quality. That's fine. Let me note that the real advantage of multi-channel hi-res audio lies in the fact that it's "multi-channel", not so much that it's "hi-res" . IMHO._

 

Many audiophiles would disagree 
	

	
	
		
		

		
		
	


	




 That said, perfect reproduction would probably involve recreating the soundwaves that were present (i.e. 3D!) when the band was there, and this might be easier with more speakers. Right now though, multichannel is for fake effects. 'Boom!' from side speakers does not accurate sound make.
  Quote:


 On the side note, I did not know it's possible to rip SACDs. This of course is off topic here but would you please point me places where i can learn more about how that's done? 
 

No, there are no SACD players for the PC. You could resample the analog output. Efforts are underway to crack the HDMI output of say a PS3.


----------



## zheka

Quote:


  Originally Posted by *slwiser* /img/forum/go_quote.gif 
_You are correct for protected DVD-As but again the AIX is open at 96kHz._

 

i like what AIX is doing, thank you for the link. I will probably get their sampler DVD. It looks like it can also be played on conventional dvd players as well as on computer DVD-ROMs 

 i wish i had a good 5.1 set up to for the playback.. well, maybe one day


----------



## slwiser

Quote:


  Originally Posted by *zheka* /img/forum/go_quote.gif 
_You basically arguing that there may be genuine 24/192 PCM material out there and down sampling to 110kHz will reduce quality. That's fine. Let me note that the real advantage of multi-channel hi-res audio lies in the fact that it's "multi-channel", not so much that it's "hi-res" . IMHO._

 

zheka

 I may be all wrong here but I think that Hi-Rez 5.1 is all sampled at the highest rate of 48 kHz (it may have 24 bits though), simply due to the amount of information that would be on the disk at 5.1 at 96 kHz/24. Stereo (two channels) can be supplied at 96/24 though. I have several DVD-As and they all show themselves being sampled at 48 kHz at 5.1. The best is the true stereo in Hi-Rez at 96/24.


----------



## zheka

Quote:


  Originally Posted by *slwiser* /img/forum/go_quote.gif 
_zheka

 I may be all wrong here but I think that Hi-Rez 5.1 is all sampled at the highest rate of 48 kHz (it may have 24 bits though), simply due to the amount of information that would be on the disk at 5.1 at 96 kHz/24. Stereo (two channels) can be supplied at 96/24 though. I have several DVD-As and they all show themselves being sampled at 48 kHz at 5.1. The best is the true stereo in Hi-Rez at 96/24._

 

AIX claim 5.1 at 24/96

 as far as size limitations i think if MLP is used dual layer DVD (9G) can hold more that 2 hours of 6 channel 24/96 data


----------



## zheka

Quote:


  Originally Posted by *puntloos* /img/forum/go_quote.gif 
_Many audiophiles would disagree 
	

	
	
		
		

		
			



_

 

no doubts about this. 
	

	
	
		
		

		
		
	


	




  Quote:


  Originally Posted by *puntloos* /img/forum/go_quote.gif 
_That said, perfect reproduction would probably involve recreating the soundwaves that were present (i.e. 3D!) when the band was there, and this might be easier with more speakers. Right now though, multichannel is for fake effects. 'Boom!' from side speakers does not accurate sound make._

 

I think multi-channel digital audio is long since past fake sound effects stage.
 well recorded 6 channel audio, even DTS/DDS, let alone uncompressed DVD-A, reproduced on good system is clearly superior to stereo , to my ears at least.


----------



## slwiser

Quote:


  Originally Posted by *zheka* /img/forum/go_quote.gif 
_AIX claim 5.1 at 24/96

 as far as size limitations i think if MLP is used dual layer DVD (9G) can hold more that 2 hours of 6 channel 24/96 data_

 

I guess I was all wrong there....but just maybe I was not wrong...read this from Dr. AIX;

http://www.audioasylum.com/scripts/t.pl?f=dvda&m=28265

 Check out what he says in one of his posts, this one in particular:

http://www.audioasylum.com/forums/dv...ges/28266.html

 It would appear that the words on the AIX website is open for some misunderstanding if what Dr. AIX is saying in this post is correct. 

 Note the liner page of an AIX disk says this for DVD-A. 5.1 "Stage" mix at 96 kHz/24 bits using MLP
 and 
 Stereo Mix at 96 kHz/24 bis using PCM

 The following is a discussion about MLP which appears to be a lossless format.

http://www.meridian-audio.com/p_mlp_mix.htm 

 This thread is a Benchmark DAC1 subject thread. I am sorry if I have push this to far off topic.


----------



## EliasGwinn

puntloos,

 You asked why someone would use USB at all vs. a sound card with a digital output. This is a very legitimate question, as each has inherent pros and cons.

 USB is obviously very convenient. Almost all computers (desktops and laptops) have an easily accessible USB port. It is true that it isolates the audio device from the high electro-magnetic (EM) field which exists inside of a computer. However, it doesn't necessarily cure the symptoms of this problem. As mentioned previously, the USB clock is just as prone to severe amounts of jitter. Often, this is intentional! Computer manufactures will add jitter to their clocks for the purpose of spreading the EM energy across a wider bandwidth (instead of a spike at a specific frequency) to measure better in FCC testing. 

 Some designers of USB audio devices alleviate this problem with asynchronous configurations, which makes the DAC clock the master, and the computer clock the slave. The problem with this is: when the computer is slave to an external clock, it will re-sample its data to the master. This re-sampling is often very poorly executed by the software, and results in severe distortion. The computer must be the master clock to ensure bit-transparency. 

 In our research and testing, we found the best configuration is isochronous/synchronous. This allows the data to be transmitted from the computer bit-transparently. This configuration, however, will have significant jitter. In the case of the DAC1, it is not affected by this jitter whatsoever, so this is a perfect solution all around. In fact, when we measure jitter-artifacts of the USB vs. the other digital inputs, there is no difference whatsoever. This is also evident in our listening room. All inputs perform identically. 

 Digital outputs on an internal sound card will probably be a little more subjected to jitter because of the environment, but otherwise should be about the same. In fact, an internal sound card is basically identical in operation with a few exceptions. A PCI card will have more direct access to data streams, depending on software configurations. Although USB is capable of handling data streams, it was not designed for it. In some specific systems with certain peripheral devices, drivers, etc, the USB performance may be compromised. 

 But, just like with any computer-related debate, it is entirely dependent on the specific system and configuration. There are very few absolutes with computer systems.

 Thanks,
 Elias


----------



## puntloos

Quote:


  Originally Posted by *zheka* /img/forum/go_quote.gif 
_I think multi-channel digital audio is long since past fake sound effects stage.
 well recorded 6 channel audio, even DTS/DDS, let alone uncompressed DVD-A, reproduced on good system is clearly superior to stereo , to my ears at least._

 

I do think this discussion is related to something we've previously touched upon: how do you separate the room from the speakers. Answer obviously is: you don't.

 A very philosophical question then is: should a recording contain room characteristics? Would the ideal recording of a choir be a hypothetical 100 singers, each in their own soundproof booth singing into a single microphone, then mixing that together across a soundstage and let the playback room acoustics work themselves out? 

 Or should a recording try to 'force' the room it was recorded in into the room it is played back at by introducing (for example) echo that was in the recording room but not in the playback room, or trying to cancel out echo that is in the playback room but wasnt in the recording room.

 If you want to 'transplant' a performance into your room, like the artist is standing there, then two speakers is all you need, and surround will just try to add distortedness into your room.

 The only situation I can conceive of when 'surround' is useful is when you have some futuristic digital processor that first measures a room, and then tries to calculate what every speaker should be producing to in effect mimic the original performance (including acoustics) by adding or substracting accents.

 I think windows vista comes with something resembling this technology but I doubt it will measure up to audiophile quality (yet), not to mention that a true surround system in my mind would have to consist of 4 or 8 identical speakers of sublime quality, which is very very very expensive. Some weak '2 decent fronts, 2 crappy rears, and some hidden away subwoofer' setup is something for people who also like many blinking lights and knobs on their audio components as well. 
	

	
	
		
		

		
		
	


	




 As for a recording where it sounds like you are one of the musicians i.e. 'singers are behind you, the choir in front of you.. well.. thats not how I experience audio when I go to a concert.


----------



## EliasGwinn

As for the DAC1 being capable of handling DVD-A, it is absolutely capable. DVD-A is nothing more then normal PCM digital audio at high resolutions. The DAC1 can handle resolutions above 192 kHz.

 The problem, as you all are noting, is many players do not actually stream the digital information at their true resolutions because of DRM.

 We've been testing many consumer-level DVD-A players to determine which, if any, play at full-rez. The only ones we found that do so are no longer in production. The DRM police have managed to ruin all the great benefits DVD-A was supposed to offer. 
	

	
	
		
		

		
		
	


	




 AIX is one exception. (If I'm not mistaken), they are not including the copyright protection status-bit that causes the DVD player to engage sample-rate conversion.

 Thanks,
 Elias


----------



## zheka

@ puntloos
 I agree that good quality multi-channel audio system is indeed very expensive. In addition it is more difficult to set up and probably allows for fewer compromises than stereo one. Plus there is shortage of well recorded multi-channel material. That’s why I am sticking with stereo for now, not because it sounds better but because it makes more sense.

 BTW, the “futuristic sound processors” are here already; take a look at DEQX product lineup http://www.deqx.com/index.html for example

 @ slwiser
 I am not sure I see any contradictions between what Dr. AIX is saying in that post and the liner page notes.

 Respectfully,

 gene


----------



## Crowbar

Re: high resolution recordings. There are many; theck out classicrecs.com or various Chesky recordings, there are 24/96 on DVD-Video (compatible with any player), and 24/192 on DVD-A, without DRM.

 Re: sound processors. Such sound processors are severely limited and cannot reproduce the sound because of how the recording was made in the first place.

 There are only two ways to really reproduce proper sound at the ears. The first is ambisonics, where an array of microphones is facing outward from a single point for the recording; then the soundfield is computed with spherical harmonics, and reproduced by a spherical arrangement of speakers around the listener. The more speakers are used, the more accurate the reproduction. Downside is you need more speakers for better results, and the sweet spot is pretty small.
 The second way is binaural recordings made with microphones in the ear canals of a dummy head, and played back by in-ear-canal headphones. Since the outer ear differs significantly between individuals and makes a big difference in positional perception, to get the full effect, it is necessary to convolve the signal with the specific listener's own HRTF (heat-related transfer function). These can be measured with an anechoic chamber, which is impractical, or a laser scan of the head can be made and then the HRTF computed by finite boundary method simulation. But even without that, good binaural recordings are far better than stereo. To play them over speakers, one needs to use DSP to compensate for the crosstalk (left speaker to right ear and the reverse), which is of only limited effectiveness.


----------



## Crowbar

Quote:


  Originally Posted by *EliasGwinn* /img/forum/go_quote.gif 
_The problem with this is: when the computer is slave to an external clock, it will re-sample its data to the master._

 

I don't know about various versions of Windows, but Apple and Linux programmers have confirmed for me this is not the case with their systems. They will not resample the data in asynchronous mode. If Windows is indeed resampling the data in asynchronous mode, that is not in the spirit of the USB Audio Class standard specification.


----------



## music_man

does the dac1 sound any different if you feed it a 24/192khz signal than if you feed it a 16/44.1. signal. i can't tell if i can really hear a difference. maybe because it just converts everything into 110khz anyways?

 music_man


----------



## Wavelength

Elias,

  Quote:


 Some designers of USB audio devices alleviate this problem with asynchronous configurations, which makes the DAC clock the master, and the computer clock the slave. The problem with this is: when the computer is slave to an external clock, it will re-sample its data to the master. This re-sampling is often very poorly executed by the software, and results in severe distortion. The computer must be the master clock to ensure bit-transparency. 
 

This is not how ASYNC works at all. First off the upper layers have no idea what the SYNCIN rate is. They merly know that the DAC's enumeration indicates that it supports rates like 44.1, 48, 88.2, 96 with a bit depth of let's say 24 bits. Therefore there is NO resampling done anywhere in this mode. Well other than what the KMIXER does but let's for this argument let's assume it's bypassed.

 The USB driver seeing that the enumeration indicates ASYNC mode realizes there is two pipes. The data pipe out to the dac and the SYNC IN pipe. The Sync IN pipe is like a flow control mechanism.

 The computer will send down data at the rate indicated by the application layer. Let's say 44.1... again for those of you that don't know this if the data is 16 bits say redbook then that is padded to 24 bits and piped out. In the ASYNC controller the buffer is measured and if the buffer is fine the ASYNC device tells the computer that it is on track by sending a stream to the computer over the Sync IN pipe that the rate is good. If the steam is too fast then the Sync IN pipe will tell the computer to slow down and if too slow to speed up.

 The good thing about ASYNC mode is now you can use a really clean ultra low jitter master clock, instead of the jittery derived clock inside the part. Remember gang USB in it self has no jitter because there is no clock riding on top of the data like there is in SPDIF. **** BUT there is intrinsic jitter most of which happens inside the USB controller. Some is generated between the USB controller and the dac because of power supplies, poor grounding yadayadayada...

 Anyways this external master clock can be very clean will then generate the BCLK, WCLK and clock out the data.

 I released code base two for my products on wed this took a couple of months where I ripped out all the bs code from the reference design to include only out going stuff. I tried SYNC, Adaptive and Async and found that Async was the best at least for the TAS1020. I rewrote the Software PLL used in Adaptive as they where updating the MCLK ever 4ms and that made Adaptive actually work about the same as SYNC mode in regards to jitter.

 In any mode you get bit perfect data. There is nothing special about Sync over any other mode. If the data is screwed up anywhere it will be between the application layer and the USB Driver.

 In many cases though with Elias product the jitter will be some what removed in the upsampling section of the dac.

 Since I don't use upsamplers ASYNC is the mode for me.

 Thanks
 Gordon


----------



## Crowbar

Thanks for clearing that up. That was my understanding from our email exchange, but after Elias' comment I started wondering whether there was some issue with the way Windows was handling things. My guess is he was referring to the kmixer resampling.

 What I don't understand is why you went NOS. Async removes the jitter issue, but what does that have to do with oversampling?
  Quote:


 Since I don't use upsamplers ASYNC is the mode for me. 
 

This statement seems to connect two unrelated things, NOS and asynch mode.

 There's no reason to make a DAC NOS whether one has low or high jitter or uses asynch or other modes. There are two cases with NOS: either there's no analog filter, which can be OK within the DAC with a non-feedback high bandwidth analog stage, but then is almost guaranteed to cause TIM and intermodulation related problems in the power amplifier or headphone amplifier, or have an analog filter which since it can't be infinitely steep to filter an image beginning immediately above the band, will alias image energy into the audio band. And even if the whole system has extremely high bandwidth into the MHz (including the speakers) so that images are not a problem, you end up relying on the very imperfect lowpass filter of the human ear, which is a problem given the references posted here and elsewhere that ultrasonic energy can influence perception.


----------



## EliasGwinn

Puntloos,

 On the question of: Why does the DAC1 re-sample to 110 kHz?

 Here is why: it is the highest frequency to maintain the full oversampling of the D-A chip. EVERY D-to-A chip on the market cuts the oversampling rate in half to accommodate 192 kHz. This will also implement a different type of digital low-pass filtering which is inferior to the filter used at and below 110kHz.

 This is also why most recording engineers don't use 192 kHz. The higher bandwidth seems appealing, but the stat-of-the-technology is such that 192 kHz conversion is actually inferior to 96 kHz.

 Also, the DAC1's oversampling ASRC and resulting 110 kHz sample rate reproduces 96 kHz signals much more faithfully then a D-A converting the original 96 kHz signal. This is because the Nyquist frequency is on the slope of the filter (attenuated, but not completely). This is undesirable for two reasons. The first reason is the Nyquist frequency is not faithfully converted to analog (ie, the analog bandwidth of 96 kHz conversion is actually less then 48 kHz). With the DAC1, the full bandwidth of a 96 kHz signal can be faithfully reproduced. The second problem with 96 kHz conversion is the frequencies at and above Nyquist (48 kHz and up) are not completely attenuated, so some aliasing and imaging will occur. With the 110 kHz upsampling and conversion in the DAC1, the frequencies below 55 kHz are not in danger of being aliased.

 Thanks,
 Elias


----------



## Crowbar

If I remember correctly, the DAC1 uses AD1853. That has internal oversampling (called interpolation filter in the datasheet). Are you bypassing that when feeding the 110 kHz in, or does it get 4x or whatever by the DAC chip's internal filter?


----------



## EliasGwinn

Puntloos,

 When you listed possible DAC's you might buy, you mentioned that the Bel Canto has the same UltraLock system as the DAC1. This is not true. They call their system UltraClock (which sounds strikingly similar to UltraLock...perhaps just coincidence
	

	
	
		
		

		
		
	


	




 ). More importantly, however, it is not the same circuitry. I am not familiar with the effectiveness of their jitter reduction technology, but I can assure you that the DAC1's jitter reduction technology is much more then a plug-and-play solution. Their are several design considerations that require meticulous engineering to achieve.

 You also mentioned that it has a better input stage then the DAC1. Can you elaborate on this a little bit?

 Also, please see my previous post concerning 192 kHz.

 Thanks,
 Elias

 ps. I realize that the thread has moved on to new topics, but I don't want to leave any questions unanswered.


----------



## EliasGwinn

Quote:


  Originally Posted by *audioengr* /img/forum/go_quote.gif 
_
 As for the jitter in a stock DAC-1, this is addressed in their design by using asynchronous upsampling like many other DAC's. It certainly does reduce the jitter of the incoming stream, probably better than many DAC's.

 However, this technique does not totally eliminate jitter IME. When I remove the upsampling chip and replace it with an I2S interface driven by my FIFO reclocker, the audible jitter is noticable lower.

 Steve N._

 

Steve,

 What method did you use to determine the jitter attenuation?

 Thanks,
 Elias


----------



## audioengr

Quote:


  Originally Posted by *EliasGwinn* /img/forum/go_quote.gif 
_Steve,

 What method did you use to determine the jitter attenuation?

 Thanks,
 Elias_

 


 In this case, I relied on my ears. Like I said "audible". It is not interesting to me to have a lot of measurements if they dont correlate to what I can hear. I let my ears be the final judge.

 You can throw stones at me all day on this, but after all the smoke clears, it is what is audible that matters IMO.

 Steve N.


----------



## EliasGwinn

Quote:


  Originally Posted by *audioengr* /img/forum/go_quote.gif 
_In this case, I relied on my ears. Like I said "audible". It is not interesting to me to have a lot of measurements if they dont correlate to what I can hear. I let my ears be the final judge.

 You can throw stones at me all day on this, but after all the smoke clears, it is what is audible that matters IMO.

 Steve N._

 

Steve,

 Don't worry, I don't enjoy stone-throwing matches either. As much as I am capable, I try to limit my discussion to constructive discussion, not destructive.

 I agree that no audio equipment can be judged on measurements alone. However, I do have doubts as to how reliably one can distinguish jitter performance well below -100 dBu just by listening. What source material were you listening to?

 Thanks,
 Elias


----------



## EliasGwinn

Quote:


  Originally Posted by *music_man* /img/forum/go_quote.gif 
_does the dac1 sound any different if you feed it a 24/192khz signal than if you feed it a 16/44.1. signal. i can't tell if i can really hear a difference. maybe because it just converts everything into 110khz anyways?

 music_man_

 

It depends on the source material. If something was recorded at 44.1 kHz/16-bits, then sample-rate converted to 192 kHz/24-bits, then there will be no difference at all. There are no bandwidth advantages because when the audio was recorded, it was low-passed at 22 kHz. Upsampling to 192 kHz will not add any more audio information above 22 kHz (except distortion). Also, the increase in word-length will not add any resolution because there will be no new information in the newly added 8 LSB's. 

 On the other hand, if something was recorded at 192 kHz/24-bits then down-sampled to 44.1 kHz/16-bit, then it will sound different. This is because the bandwidth will be reduced to 22 kHz and the noise floor will increase to 16-bit levels and resolution will decrease to 16-bit levels.

 I hope I answered your question. If not, please follow up.

 Thanks,
 Elias


----------



## EliasGwinn

Quote:


  Originally Posted by *Crowbar* /img/forum/go_quote.gif 
_If I remember correctly, the DAC1 uses AD1853. That has internal oversampling (called interpolation filter in the datasheet). Are you bypassing that when feeding the 110 kHz in, or does it get 4x or whatever by the DAC chip's internal filter?_

 

The filter in the D-A chip is frequency shifted to 55 kHz to not interfere with the Nyquist frequencies (see above post concerning why the DAC1 re-samples to 110 kHz).

 This, effectively, replaces the D-A filter with the SRC filter which is a much higher quality filter.

 Thanks,
 Elias


----------



## EliasGwinn

I would like to expound on why 192 kHz conversion is not a good idea.

 The filters in a D-A chip ideally pass all audio below the Nyquist frequency and block (filter) all audio above the Nyquist frequency. In reality, however, there will be some audio below the Nyquist that is being filtered some, and there will be some audio above the Nyquist which is not filtered enough. The latter is very dangerous because those frequencies will be aliased and cause distortion.

 Here's the problem with 192 kHz: the filter used for 192 kHz is of far less quality. This is true of ALL D/A chips on the market. What happens is this: the filter cut-off becomes less defined, causing audio below Nyquist to be attenuated. And, more importantly, AUDIO ABOVE NYQUIST IS NOT FULLY ATTENUATED!! The filter does not do its job as well at 192 kHz. 

 Another 'real-life' limitation to these filters is amplitude ripple. This means that the audio below Nyquists will have ripples in amplitude across the frequency spectrum. This is equivalent to inaccurate frequency response. This is also something that happens in all D-to-A chips, some more so then others.

 The problem with 192 kHz: the ripples in amplitude become much more exaggerated when filtering 192 kHz signals. Consequently, the frequency response is much less accurate and distortion goes up.

 This is why most all converter designers and recording engineers don't recommend 192 kHz. The chip technology has not provided the means to effectively convert 192 kHz without these problems. 

 So, the trade-off for the extra analog bandwidth is an increase in aliasing and frequency-response distortion. Although some people may 'enjoy' listening to 192 kHz more then lower rates, it is not as accurate, objectively speaking.

 Thanks,
 Elias


----------



## LAMark

Quote:


  Originally Posted by *audioengr* /img/forum/go_quote.gif 
_I suppose it is vague. I like to keep as much as possible the mods I do a trade secret. As I mentioned before, I discovered that the BUF634 is a difficult device to feed in terms of power delivery, and the topology that is recommended by TI tends to aggravate this IMO. I actually had a mod for the volume/headphone for a long time, but I wasn't happy with the level of improvement and could not recommend it to my customers. I only had a couple of orders for this. Then, I revisited it a year later (mostly as a result of the Head-Fest) and developed some improvements that are significant enough that I now recommend this mod. The improvements consist of:

 1) rewiring of some traces
 2) some redesign of the circuitry around the volume pot
 3) replacing all of the op-amps
 4) power delivery improvements for the op-amps
 5) power delivery improvements for the output drivers

 There is barely enough space to accomplish all of this, which makes the rework difficult, and a bit expensive. I do not recommend this headphone section mod stand-alone. There is a subset of the Turbomod that should be combined with this.
 I demonstrated this mod at the Head-Fest in San Jose and Ray Samuels commented that it was quite good. It is not as good as a Raptor though. I want one of those.

 BTW, a really nice Head-fier brought his grace amp over to compare both the head-amp in it and the DAC to my modded DAC-1 about 2 weeks ago. I will see if he is willing to report back. He also brought me some awesome new music.

 Steve N._

 

I am that Head-fier. I visited Steve in central Oregon, lugging along my Grace m902 head-amp and my Senn HD650's (w/Equinox cable). Actually I came to listen to Steve's other offerings, not really thinking about doing a direct comparison between the Grace and the modified DAC-1. But there it was...so what's an audiophile to do?-we made several comparisons of course. The configuration as I recall was using Steve's laptop as the source going out through USB (what else?) to an Offramp, then through I2S to the modded DAC-1. Then we took the output of the DAC-1 to the Grace so we could compare just the head-amp sections. We also had a configuration to compare the DAC-1 DAC+Headamp to the Grace DAC+Headamp, a comparison I have made before with a stock DAC-1 and smugly concluded the Grace to be superior. In this case, though there was a decided advantage to the modified DAC-1 in both comparisons, especially in the highs where the Grace had previously excelled, or so I had thought. Basically the configurations with the DAC-1 were superior by almost any measure-staging, resolution, you name it. None of the graininess I had noticed with the stock DAC-1. Even in the head-amp comparison, where I was pretty sure the Grace would hold its own, it didn't. Oh well. Now to save my pennies. Anybody wanna buy a nice headamp?


----------



## Crowbar

Seriously guys, I keep hearing claims of listening results contradicting measurements, but these are equally invalid unless you guys do those tests with proper blind methodology.

 As for the measurement people, I recommend more attention be paid to perceptually weighted metrics in order to have meaningful numbers. An excellent example is this paper:
 Hollier, M.P., Hawksford, M.O., and Guard, D.R., "Error Activity and Error Entropy as a Measure of Psychoacoustic Significance in the Perceptual Domain", Vision, Image and Signal Processing, IEE Proceedings, Vol.141 No.3, June 1994, pp:203-208.
 Also relevant is Hawksford, M.O., "System Measurement and Identification Using Pseudorandom Filtered Noise and Music Sequences", Journal of the AES, Vol.53 No.4, April 2005, pp. 275-296.


----------



## puntloos

Hey Elias,

 First off thanks for taking the time to thoroughly answer, again, it is very much appreciated.

  Quote:


  Originally Posted by *EliasGwinn* /img/forum/go_quote.gif 
_Puntloos,
 When you listed possible DAC's you might buy, you mentioned that the Bel Canto has the same UltraLock system as the DAC1. This is not true. They call their system UltraClock (which sounds strikingly similar to UltraLock...perhaps just coincidence
	

	
	
		
		

		
			





 ). More importantly, however, it is not the same circuitry. _

 

Hah, yeah I guess my brain just connected the dots the wrong way huh, basically both you and Bel Canto are 'extremely proud of the anti jitter' and both were called 'ultra....ock' .. Additionally, I seem to remember some review or some forum post that said the two used the same technology.
  Quote:


 I am not familiar with the effectiveness of their jitter reduction technology, but I can assure you that the DAC1's jitter reduction technology is much more then a plug-and-play solution. Their are several design considerations that require meticulous engineering to achieve.

 You also mentioned that it has a better input stage then the DAC1. Can you elaborate on this a little bit?

 

Well, admittedly this was my conclusion after reading what basically amounts to the 'marketing version' of the specs. Bel Canto claim complete galvanic isolation of the inputs and shielded input transformers. While I think these are 'good things' in general, maybe these are also totally trivial and you simply chose to not mention this in your specs since 'who wouldn't do it that way!'. 

  Quote:


 Also, please see my previous post concerning 192 kHz. 
 

Done.. thank you for your explaination.. I am afraid I am out of my depth though. While my 'cowpoopie detector' is not ringing with you, I really cant gauge how your design choices would rank against the choices the belcanto, aqvox or lavry engineers made. (aqvox and belcanto sample at 192khz with the Burr Brown 1796 and 1792 DACs respectively.). It would be interesting to hear you guys discuss this subject matter, although it'd probably be like an ant watching giants fight and trying not to get stepped on.

 I do have one additional question though. I am trying to arrange to get a Bel Canto, an Aqvox and a benchmark in one room. Do you have any suggestions on how, or what to test? For example is there some way you would reccommend someone with limited pro resources to create a bad jitter situation to test a dac's resilience?


----------



## Crowbar

I know these were meant for Elias but I'll chime in as well 
	

	
	
		
		

		
		
	


	




  Quote:


  Originally Posted by *puntloos* /img/forum/go_quote.gif 
_Bel Canto claim complete galvanic isolation of the inputs and shielded input transformers._

 

Input transformers are common on pro-audio DACs, and I see them mentioned in many receiver and ASRC chip datasheets. Nothing special about that. I found in my DIY that it helps if your source has a lot of noise (in my case the sound card S/PDIF out had lots of HF crap), and also can potentially prevent some ground loops. But in most cases I doubt it's necessary. You can always add an input transformer to any DAC that doesn't have one if you really feel the need, just make sure you pick the right one. scientificonversion.com make some nice S/PDIF transformers for example, but make sure you get a 1:1 and have proper termination.

  Quote:


 I am trying to arrange to get a Bel Canto, an Aqvox and a benchmark in one room. Do you have any suggestions on how, or what to test? For example is there some way you would reccommend someone with limited pro resources to create a bad jitter situation to test a dac's resilience? 
 

This is a great opportunity to get a friend and do some blind testing! I couldn't recommend this more! Another great test you can do is, if you have a high quality ADC, maybe one on a good musician's sound card, I suggest you email Hawksford referencing the papers from my previous post and ask him to send you his MATLAB code. This is definitely better than the stuff people would normally do at home such as with RMAA.


----------



## audioengr

Quote:


  Originally Posted by *EliasGwinn* /img/forum/go_quote.gif 
_Steve,

 Don't worry, I don't enjoy stone-throwing matches either. As much as I am capable, I try to limit my discussion to constructive discussion, not destructive.

 I agree that no audio equipment can be judged on measurements alone. However, I do have doubts as to how reliably one can distinguish jitter performance well below -100 dBu just by listening. What source material were you listening to?

 Thanks,
 Elias_

 


 Real music of course. I have a collection of excellent tracks that friends/colleagues turned me onto over the years who attend CES and drop in to see me every year. It's the best of the best. Very dynamic, extended and some superb vocals, piano, percussion, pretty much everything that pushes a system to be great. It requires many different tracks to do this type of evaluation. One or two is not sufficient. 

 I use a Toshiba laptop with Foobar 0.8.3 and SRC upsampling it to 24/96 for all tracks. Much more detailed and dynamic this way. DAC's are driven with my Off-Ramp I2S and Pace-Car reclocker or my Off-Ramp Turbo 2 (S/PDIF coax output). Both of these USB converters are clocked with the excellent Superclock4.

 Steve N.


----------



## puntloos

Quote:


  Originally Posted by *Crowbar* /img/forum/go_quote.gif 
_I know these were meant for Elias but I'll chime in as well 
	

	
	
		
		

		
		
	


	


_

 

Always welcome! Hehe as long as Elias will at least say he agrees or adds his own $0.02 - no offense of course but I surmise he knows his DAC better than you 
	

	
	
		
		

		
		
	


	




  Quote:


 Input transformers are common on pro-audio DACs, and I see them mentioned in many receiver and ASRC chip datasheets. Nothing special about that. I found in my DIY that it helps if your source has a lot of noise (in my case the sound card S/PDIF out had lots of HF crap), and also can potentially prevent some ground loops. But in most cases I doubt it's necessary. You can always add an input transformer to any DAC that doesn't have one if you really feel the need, just make sure you pick the right one. scientificonversion.com make some nice S/PDIF transformers for example, but make sure you get a 1:1 and have proper termination.

 

Well the point bel canto made was that their input stage is separated 'completely' from the output stage, more with each having their own power source etc. As mentioned, complete galvanic separation.

 Your point about adding your own transformer is, i suppose, valid, especially for me since Ive actually built some circuit boards etc (with etching etc) in my studies, but Ive always been hesitant about tampering with a $2.5K piece of gear. As many modders will agree - stuff can be improved on a vanilla device, but if it has certain features from the get-go, thats definately a plus.

  Quote:


 This is a great opportunity to get a friend and do some blind testing! I couldn't recommend this more! Another great test you can do is, if you have a high quality ADC, maybe one on a good musician's sound card, I suggest you email Hawksford referencing the papers from my previous post and ask him to send you his MATLAB code. This is definitely better than the stuff people would normally do at home such as with RMAA. 
 

I plan to, I have a 24/96 recording card and I have a coupla audiophile friends who plan to bring their own gear. One SACD player and one 'audiophile CD player with built-in DAC). My base set (Quad 989's with a Bel Canto Evo 4 gen2) should suffice as the fixed part of the experiment.


----------



## Crowbar

Quote:


  Originally Posted by *puntloos* /img/forum/go_quote.gif 
_no offense of course but I surmise he knows his DAC better than you 
	

	
	
		
		

		
		
	


	


_

 

No offense, but your questions weren't DAC1 specific. One was about galvanic isolation, and the other about listening tests.


----------



## puntloos

Quote:


  Originally Posted by *Crowbar* /img/forum/go_quote.gif 
_No offense, but your questions weren't DAC1 specific. One was about galvanic isolation, and the other about listening tests._

 

Even less offense 
	

	
	
		
		

		
		
	


	




 but we were actually comparing the input stages of DACs.. Elias asked why I thought the DAC3 had a better input stage, I listed a few features of the DAC3 (transformers, galvanic isolation, and separately powered stages) and you said that 'many' pro dacs contain such features. While Im certain you're right, we (or at least I) still don't know if this applies to the DAC1

 The open question still is if the DAC1's input stage has the features listed, and more generally is better or equal to the DAC3's .. which probably depends partially on input specs of the DAC3 none of us are privy too, but Elias might still be the best equipped to give a good guess..


----------



## Bootleg

.


----------



## tonygeno

I have hooked up the Dac1 USB to a Mac Mini via USB. The device shows up in the Sound System Preference. When listening to iTunes all is well and I can listen to the music through my stereo but when I listen to music over Safari (Rhapsody, for instance), the sound defaults to the computer speakers, even though I have chosen the Benchmark as my sound device. If I switch to the optical connection on the Benchmark, the music then plays on my stereo. Any ideas as to why?


----------



## lowmagnet

Quote:


  Originally Posted by *tonygeno* /img/forum/go_quote.gif 
_I have hooked up the Dac1 USB to a Mac Mini via USB. The device shows up in the Sound System Preference. When listening to iTunes all is well and I can listen to the music through my stereo but when I listen to music over Safari (Rhapsody, for instance), the sound defaults to the computer speakers, even though I have chosen the Benchmark as my sound device. If I switch to the optical connection on the Benchmark, the music then plays on my stereo. Any ideas as to why?_

 

Beside the settings in the System Preferences Audio pane, check your Audio Midi Setup program as well and confirm that audio is going out to the DAC1. I had a problem with web browsers going to neither internal nor the DAC1 and Audio Midi Setup fixed this issue.


----------



## tonygeno

Quote:


  Originally Posted by *lowmagnet* /img/forum/go_quote.gif 
_Beside the settings in the System Preferences Audio pane, check your Audio Midi Setup program as well and confirm that audio is going out to the DAC1. I had a problem with web browsers going to neither internal nor the DAC1 and Audio Midi Setup fixed this issue._

 

I'll give it a go. Thanks.


----------



## tonygeno

Unfortunately, that didn't fix it. It works in the digital input, where I get output, but not the USB input.


----------



## Audio_newb

Great thread so far. I think I've seen almost every aspect of the DAC1's design examined, but I don't think I've seen a discussion as to the choice of a volume pot. I'm not for or against, but just wondering as to the choice of what (I assume) is a single volume pot rather than a resistor ladder or some form of digitally controlled analog volume control.

 I don't know enough to question one method over another, other than the already mentioned mechanical limitations at the extremes of the volume pot, but was simply wondering if this design choice was made for cost reasons, or if the volume attenuation functions of the DAC1 are on par quality wise with its DAC functions.


----------



## music_man

lot's of people have found faults with the dac1. there are 31 pages in this thread so far. if it is so bad why is everyone giving it so much attention? hmm. i love it!

 music_man


----------



## Wavelength

Tonygeno,

 What os level are you on with your mac mini? There was a change in the usb subsystem that was fixed in 10.4.9. This was a kinda goofy thing in that it did not effect most audio usb controllers but did with the TAS1020 that is also used in the DAC1.

 You may simply have to upgrade the os. You can also look at the USB chain in the system profiler and see what the DAC1 is coming in as. Sometimes if there is a problem the info here will be garbled.

 Thanks
 Gordon


----------



## Lord Chaos

I figure the reason this thread has run 31 pages--so far--is that Elias takes no offense and keeps answering questions. In the years I've been involved with audio I've never known of of a company so willing to engage in the conversation.

 I still use my DAC1s every day and continue to be delighted with them. It's an excellent product. Simple, direct, and sounds great.

 I still wonder sometimes, though... if Benchmark wanted to make an ultimate, cost-no-object DAC, what would they change? How much of a difference would the changes make? I have no intention of going into the pursuit of modifications, as the sound quality of the stock unit is stellar, but in the design of any real-world product compromises must be made. I wonder what compromises were made with the DAC1... and how much midnight oil was burned in the discussion of those compromises.


----------



## music_man

yeah, i would like to really thank mr. gwinn. i know no other large company that has ever talked to their customers like this.

 i'd like everyone to check in that uses their dac1 as their headphone amp. or at least one of their headphone amps. how do you think it compares to your others? i don't mean saying this one or that one is better. i mean how does the "sound" compare. i don't like when people say something is just better. the dac1's amp is not insulted by anything. either you like hearing the truth or you don't.

 6moons compared the amp to the cia vhp-1 and found the dac1 "better". that is not so important though. it is what he said about the dac1's amp. exactly what i have been saying. i know of other amp that is as true to the source. that doesnt mean a more natural/neutral amp exists, i just don't know of it 
	

	
	
		
		

		
		
	


	




 music_man


----------



## EliasGwinn

Quote:


  Originally Posted by *puntloos* /img/forum/go_quote.gif 
_
 Hah, yeah I guess my brain just connected the dots the wrong way huh, basically both you and Bel Canto are 'extremely proud of the anti jitter' and both were called 'ultra....ock' .. Additionally, I seem to remember some review or some forum post that said the two used the same technology._

 

I'm not sure how their anti-jitter technology works, but, as I said before, there is no plug-and-play chip that creates jitter immunity. We use an ASRC, as do many other manufacturers. However, the entire anti-jitter technology in the DAC1 (UltraLock) is very much propriety and individual to the DAC1. This is why we named it...because it is a very specific topology. I'm not sure why Bel Canto chose a name so similar to ours (BC's UltraClock vs. Benchmark's UltraLock). It reminds me of the Walkmans I used to see on the streets in NYC with the brand-name Coby, written in a font and style suspiciously similar to Sony.

  Quote:


  Originally Posted by *puntloos* /img/forum/go_quote.gif 
_Well, admittedly this was my conclusion after reading what basically amounts to the 'marketing version' of the specs. Bel Canto claim complete galvanic isolation of the inputs and shielded input transformers. While I think these are 'good things' in general, maybe these are also totally trivial and you simply chose to not mention this in your specs since 'who wouldn't do it that way!'. _

 

An understandable interpretation.,.. Galvanic Isolation is just another way of saying transformer isolated. Both the BNC (coax) and XLR inputs on the DAC1 are transformer (galvanic) isolated. 

  Quote:


  Originally Posted by *puntloos* /img/forum/go_quote.gif 
_Done.. thank you for your explaination.. I am afraid I am out of my depth though. While my 'cowpoopie detector' is not ringing with you, I really cant gauge how your design choices would rank against the choices the belcanto, aqvox or lavry engineers made. (aqvox and belcanto sample at 192khz with the Burr Brown 1796 and 1792 DACs respectively.). It would be interesting to hear you guys discuss this subject matter, although it'd probably be like an ant watching giants fight and trying not to get stepped on._

 

We test most chips to determine if they are something we should use. Just like with any engineering, their are trade-offs (each has pros and cons, none out-shine in all respects). We have tested the 1792 (the 1796 is just a cheaper version of the 1792 with similar topology but less performance). It had advantages and disadvantages vs. the AD1853. 

 The difference between the BB 1796 vs. the DAC1's AD1853 are: the former has better filtering and signal-to-noise ratio. But it has serious linearity issues. Linearity means 1 dB increase in digital amplitude results in 1 dB increase in analog amplitude (input vs. output). The Burr-Brown chips have serious problems with linearity. Also, intermodulation distortion is an more of an issue with these chips. They are also very sensitive to temperature fluctuations, so that these issues are amplified as the temperature fluctuates.

 In other words, take your choice: more distortion and less noise (1796) or more noise and less distortion (1853). Since the S-to-N of the 1853 is -117 dB, we chose this over increased distortion.

  Quote:


  Originally Posted by *puntloos* /img/forum/go_quote.gif 
_I do have one additional question though. I am trying to arrange to get a Bel Canto, an Aqvox and a benchmark in one room. Do you have any suggestions on how, or what to test? For example is there some way you would reccommend someone with limited pro resources to create a bad jitter situation to test a dac's resilience?_

 

Run 500 feet of cable from the source to the DAC. Other then that, there is no way to definitely increase jitter. However, you could start with a cheap transport.

 Thanks,
 Elias


----------



## EliasGwinn

Quote:


  Originally Posted by *tonygeno* /img/forum/go_quote.gif 
_I have hooked up the Dac1 USB to a Mac Mini via USB. The device shows up in the Sound System Preference. When listening to iTunes all is well and I can listen to the music through my stereo but when I listen to music over Safari (Rhapsody, for instance), the sound defaults to the computer speakers, even though I have chosen the Benchmark as my sound device. If I switch to the optical connection on the Benchmark, the music then plays on my stereo. Any ideas as to why?_

 

I have been in contact (phone and email) with Tony since he posted this I have solved this problem. I will post the result for anyone else who encounters this.

 I don't know why, but Rhapsody simply refused to 'talk' to the DAC1 USB via the USB input. So, I tried other web-browser-based audio streams such as MySpace, and it was no problem. But not with Rhapsody. This was the case with both Firefox and Safari. This was also the case with other 3rd party USB devices.

 After messin' around for a while, I finally got it to work, and here's how: I changed the sample-rate of the Benchmark in the Mac Audio MIDI Setup. I changed it to 44.1 kHz (48 kHz worked also). After changing the sample-rate, I changed the "Default Ouput" to "Built-In Output". Then I immediately changed it back to "Benchmark 1.0". It worked.

 I don't know how Rhapsody works exactly (technically speaking), so I can't comment as to why it happened. But at least we know how to work around it.

 I will also add the side note which I offered to Tony, and is very important for anyone who uses Mac for audio:

 The sample-rate setting in Audio MIDI Setup should always be set to the sample-rate of the music file being played. In the case of Rhapsody, 44.1 kHz is most likely the original sample rate. This applies to iTunes and everything else, also. If you are playing 44.1 kHz audio, the sample-rate in Audio MIDI should be set as 44.1 kHz. If you're listening to 98 kHz audio, then you should set it to 98 kHz. The reason for this is that the Mac will convert the sample-rate from the original rate to the rate selected in Audio MIDI Setup (if the two rates are different). The sample-rate conversion is very poorly executed by the Mac (poor programming), and significant distortion will result. It is very important, from a fidelity standpoint, to always be sure your sample-rates are the same. 

 Thanks,
 Elias


----------



## EliasGwinn

Quote:


  Originally Posted by *Audio_newb* /img/forum/go_quote.gif 
_Great thread so far. I think I've seen almost every aspect of the DAC1's design examined, but I don't think I've seen a discussion as to the choice of a volume pot. I'm not for or against, but just wondering as to the choice of what (I assume) is a single volume pot rather than a resistor ladder or some form of digitally controlled analog volume control.

 I don't know enough to question one method over another, other than the already mentioned mechanical limitations at the extremes of the volume pot, but was simply wondering if this design choice was made for cost reasons, or if the volume attenuation functions of the DAC1 are on par quality wise with its DAC functions._

 

This is a great question. This is another example of engineering design trade-off. ie, each solution gives a little here, but takes away a little (or a lot!) there.

 Specifically, here are the common choices (not including digital volume control, which is pre-D-to-A):

 DIGITALLY CONTROLLED ANALOG VOLUME CONTROL:

 These are the poorest performing from a distortion point of view. These are simply not satisfactory with high-quality audio.

 POTENTIOMETER:

 This is good for the following: low noise, low distortion, low cost, tight volume resolution (ie, easy to achieve small changes in volume). Pots are bad for the following reason: the first 15-20% of rotation is not evenly matched from left to right. After this initial inaccuracy, however, it is VERY accurate. Very comparable to any other (analog) solution.

 RESISTOR LADDER:

 This is good for the following: low distortion, high channel accuracy, low noise (at rest). The last point indicates that it is noisy when switching from one volume setting to another (switching noise), but quiet once at rest. The resistor ladder is bad for the following reasons: expensive, low volume resolution. This last point is because a switching resistor ladder has 24-positions, where the detented-pot of the DAC1 has 41-detents (and can be settled between detents). This means you can make smaller changes in volume with a pot then you can with a resistor ladder.

 So, as you can see, there are trade-offs, and there are no right or wrong way. The reason we choose the potentiometer is because it works just as well as the resistor ladder after the first 20% of rotation, and the lower half of rotation should only be used to achieve full-off. So, as long as it is being operated in its upper 80%, it is as good as any other solution, while being significantly less expensive and higher volume resolution.

 Thanks,
 Elias


----------



## EliasGwinn

Quote:


  Originally Posted by *Lord Chaos* /img/forum/go_quote.gif 
_I figure the reason this thread has run 31 pages--so far--is that Elias takes no offense and keeps answering questions. In the years I've been involved with audio I've never known of of a company so willing to engage in the conversation.

 I still use my DAC1s every day and continue to be delighted with them. It's an excellent product. Simple, direct, and sounds great.

 I still wonder sometimes, though... if Benchmark wanted to make an ultimate, cost-no-object DAC, what would they change? How much of a difference would the changes make? I have no intention of going into the pursuit of modifications, as the sound quality of the stock unit is stellar, but in the design of any real-world product compromises must be made. I wonder what compromises were made with the DAC1... and how much midnight oil was burned in the discussion of those compromises. 
	

	
	
		
		

		
		
	


	


_

 

Well, I'd like to answer you, but I don't want to give any design secrets away 
	

	
	
		
		

		
		
	


	




 . 

 This question is hard to answer without sounding over-prideful, but I will say this, there are no parts in the DAC1 that can be improved with more expensive parts. The most expensive components of the DAC1 (and the only components which are chosen based on price) are the faceplate and chassis etc.; circuit components are not limited by cost. In other words, the sonic-performance of the DAC1 was not compromised based on price.

 The only way the DAC1's sonic performance could be improved is with a more elaborate topology - a whole new design technique. This would not cost a whole lot more from a component point of view, but it would from a R&D point of view.

 Thanks,
 Elias


----------



## Crowbar

The PCM1796 is not R2R, it's a hybrid multilevel sigma-delta DAC. The operation is the same as shown in Fig.30 in the PCM1794 datasheet (likewise for the PCM1792).

 Galvanic isolation is not necessarily transformer based; it can be optical as well


----------



## EliasGwinn

Thank you for catching that.

 The DAC1 is also optically isolated.

 Thanks,
 Elias


----------



## Lord Chaos

Quote:


  Originally Posted by *Elias* 
_...I will say this, there are no parts in the DAC1 that can be improved with more expensive parts._

 

Very interesting... and pretty much what I thought, given the way the DAC1 sounds. I know you caution people about the imbalance at low volume settings, but you must be using a really good pot in there because I've never noticed balance problems at the low settings required for my sensitive headphones and sensitive ears. Thank you for the response.


----------



## showflash

Quote:


  Originally Posted by *EliasGwinn* /img/forum/go_quote.gif 
_As for the DAC1 being capable of handling DVD-A, it is absolutely capable. DVD-A is nothing more then normal PCM digital audio at high resolutions. The DAC1 can handle resolutions above 192 kHz.

 The problem, as you all are noting, is many players do not actually stream the digital information at their true resolutions because of DRM.

 We've been testing many consumer-level DVD-A players to determine which, if any, play at full-rez. The only ones we found that do so are no longer in production. The DRM police have managed to ruin all the great benefits DVD-A was supposed to offer. 
	

	
	
		
		

		
			





 AIX is one exception. (If I'm not mistaken), they are not including the copyright protection status-bit that causes the DVD player to engage sample-rate conversion.

 Thanks,
 Elias_

 

Sorry that I didn't get my question in earlier. I've been reading every post and it took a couple of days.

 Elias you said that you hadn't found any DVD-A players that played full rez and then said that AIX was an exception. Did I misunderstand you and that AIX is an exception to the rule in so far as recorded material goes, and they play at the full resolution regardless of the player? I think that is what you meant. I am looking at the materials others directed me to regarding AIX. It should play in any DVD-A player at full resolution correct?

 This has been some thread and I am down to get a DAC from you as soon as I get out of school...Cisco networking related. My mind was pretty much made up on the Stereophile review when I read it. This has all been icing on the cake.

 Thanks for your replies!


----------



## puntloos

Quote:


  Originally Posted by *EliasGwinn* /img/forum/go_quote.gif 
_I'm not sure how their anti-jitter technology works, but, as I said before, there is no plug-and-play chip that creates jitter immunity. We use an ASRC, as do many other manufacturers. However, the entire anti-jitter technology in the DAC1 (UltraLock) is very much propriety and individual to the DAC1. This is why we named it...because it is a very specific topology. I'm not sure why Bel Canto chose a name so similar to ours (BC's UltraClock vs. Benchmark's UltraLock). It reminds me of the Walkmans I used to see on the streets in NYC with the brand-name Coby, written in a font and style suspiciously similar to Sony._

 

Weeeeeellll, given the device goal and situation of the device, I wouldn't call 'Ultralock' or 'Ultraclock' that crazy a name to derive that Bel Canto obviously nicked it on purpose. If you guys would've called it 'Zarfglub 2.0' and Bel Canto has 'Zarfglib 2.1' then Id call shenanigans. 
	

	
	
		
		

		
		
	


	




  Quote:


 An understandable interpretation.,.. Galvanic Isolation is just another way of saying transformer isolated. Both the BNC (coax) and XLR inputs on the DAC1 are transformer (galvanic) isolated. 
 

Ah, as I suspected. One last question about the Benchmark DAC1 inputs: how resilient are they? Short of putting 220V on the input leads, will they be able to resist some punishment in the way of static shock etc? (my home has quite fuzzy carpet, for example)

  Quote:


 The difference between the BB 1796 (24-bit ladder network D/A) vs. the DAC1's AD1853 (Delta-Sigma) are: the former has better filtering and signal-to-noise ratio. But it has serious linearity issues. Linearity means 1 dB increase in digital amplitude results in 1 dB increase in analog amplitude (input vs. output). The Burr-Brown chips have serious problems with linearity. Also, intermodulation distortion is an more of an issue with these chips. They are also very sensitive to temperature fluctuations, so that these issues are amplified as the temperature fluctuates.

 In other words, take your choice: more distortion and less noise (1796) or more noise and less distortion (1853). Since the S-to-N of the 1853 is -117 dB, we chose this over increased distortion.

 

As crowbar indicated, the BB actually also uses Delta-Sigma (both the 1796 and 1792). Did you base your comparison on the differences between 'ladder' DACs and 'DS' Dacs in general, or did you base your comparison on your other knowledge of the two devices?


----------



## EliasGwinn

Quote:


  Originally Posted by *showflash* /img/forum/go_quote.gif 
_Sorry that I didn't get my question in earlier. I've been reading every post and it took a couple of days.

 Elias you said that you hadn't found any DVD-A players that played full rez and then said that AIX was an exception. Did I misunderstand you and that AIX is an exception to the rule in so far as recorded material goes, and they play at the full resolution regardless of the player? I think that is what you meant. I am looking at the materials others directed me to regarding AIX. It should play in any DVD-A player at full resolution correct?

 This has been some thread and I am down to get a DAC from you as soon as I get out of school...Cisco networking related. My mind was pretty much made up on the Stereophile review when I read it. This has all been icing on the cake.

 Thanks for your replies!_

 

I don't really know too much about AIX's releases, or whether or not they will play on certain players. I know that the cheap (<$200) Panasonic, Pioneer, and Oppo players on the market won't put out full resolution, even with material that is not copy protected. AIX's releases my have some digital flag that manages to 'convince' the player to do otherwise, but I don't know. I have heard that they are trying to do things along those lines, but I really don't know.

 Thanks,
 Elias


----------



## EliasGwinn

Quote:


  Originally Posted by *puntloos* /img/forum/go_quote.gif 
_Weeeeeellll, given the device goal and situation of the device, I wouldn't call 'Ultralock' or 'Ultraclock' that crazy a name to derive that Bel Canto obviously nicked it on purpose. If you guys would've called it 'Zarfglub 2.0' and Bel Canto has 'Zarfglib 2.1' then Id call shenanigans. 
	

	
	
		
		

		
		
	


	


_

 

You're right, I have no idea how or why they chose that name. It just struck us as....interesting...

  Quote:


  Originally Posted by *puntloos* /img/forum/go_quote.gif 
_Ah, as I suspected. One last question about the Benchmark DAC1 inputs: how resilient are they? Short of putting 220V on the input leads, will they be able to resist some punishment in the way of static shock etc? (my home has quite fuzzy carpet, for example)_

 

They are very, VERY resilient. The transformer isolation is resilient to up to 1500V. Also, when we test for CE, we 'shoot' the DAC1 with a 'charge gun' that is equivalent to a VERY heavy static discharge. We shoot it at several points...the inputs, the outputs, the volume knob, the headphone jacks, etc... In no way was the DAC1 damaged from these static discharges.

  Quote:


  Originally Posted by *puntloos* /img/forum/go_quote.gif 
_As crowbar indicated, the BB actually also uses Delta-Sigma (both the 1796 and 1792). Did you base your comparison on the differences between 'ladder' DACs and 'DS' Dacs in general, or did you base your comparison on your other knowledge of the two devices?_

 

No, I just mis-spoke when I said it was a 'ladder-network' D/A (perhaps I should go back and edit that). We based our comparison on actual bench-testing with the evaluation boards from these companies. We tested with the factory-configured circuit, and then tested with tweaks for more in-depth charting of the characteristics. It is the only way you can know what a chip really does in use. The datasheets give a lot of information, but they won't necessarily tell you the worst case scenarios.

 If the BB chips would have tested better then the AD1853 in distortion and linearity, as well as filtering and noise, then we would have used the BB. But, the tests showed that the BB had serious distortion and linearity issues - enough so that we chose the AD1853 instead.

 Thanks,
 Elias


----------



## Scrith

This thread is great, but it's just crazy. I think people have started asking the same questions again because the thread is so huge that it has become impossible for anyone to fully appreciate.

 Please consider starting a manufacturer forum so we can have separate threads for all these great topics! I think this can be done here, at audioasylum.com, or at audiocircle.com for very little expense (monetary or time).


----------



## showflash

When/If you start a manufacturers forum area just move this and related discussions to that area.


----------



## puntloos

Elias: thanks again for your insightful and patient messages.

 One last and very important question:

 What are you guys @ Benchmark up to right now? Built a cool dac, now sitting on your behind raking in the cash? Chatting on forums? 
	

	
	
		
		

		
		
	


	




 Or can we await a DAC2 anytime soon? (two more and you've caught up with Bel Canto! *ducks*)


----------



## music_man

i hope mr. gwinn shares everyones "sense of humor". i am surprised he stuck around for all this. even though you are making funny i wouldn't push it. he has been very nice to us.

 music_man


----------



## Regnad

I have really enjoyed reading this thread, particularly since I just added a DAC1 to my Squeezebox and it sounds great! The music is in FLAC on a computer in another room and I have used both wired and wireless connections.

 My question:
 How would this setup compare to a local computer feeding a USB DAC-1? What would the differences be, both digitally and sonically? 

 I would really appreciate any information from those who have experience with both.

 Thanks a lot...


----------



## tonygeno

Quote:


  Originally Posted by *Wavelength* /img/forum/go_quote.gif 
_Tonygeno,

 What os level are you on with your mac mini? There was a change in the usb subsystem that was fixed in 10.4.9. This was a kinda goofy thing in that it did not effect most audio usb controllers but did with the TAS1020 that is also used in the DAC1.

 You may simply have to upgrade the os. You can also look at the USB chain in the system profiler and see what the DAC1 is coming in as. Sometimes if there is a problem the info here will be garbled.

 Thanks
 Gordon_

 

I am using 10.4.9. I spoke with Elias over at Benchmark and there was a setting in the Audio Midi Utility that I needed to change: the audio output to 44100 format (it was set a 96000) as Elias described above. After I did that, it worked.


----------



## eyeteeth

I'm throwing in my thanks to Elias Gwinn as well. 
*Thanks for taking the time Elias!
*
 My DAC1 is a few years old now and this is the thread I keep an eye on.
  Quote:


  Originally Posted by *Regnad* /img/forum/go_quote.gif 
_I have really enjoyed reading this thread, particularly since I just added a DAC1 to my Squeezebox and it sounds great! The music is in FLAC on a computer in another room and I have used both wired and wireless connections.

 My question:
 How would this setup compare to a local computer feeding a USB DAC-1? What would the differences be, both digitally and sonically? 

 I would really appreciate any information from those who have experience with both.

 Thanks a lot..._

 

Great question. I have the same set up (coaxial to DAC1) and as I'm well due for a computer upgrade I'm also wondering if I stay with the same configuration or get a laptop or other in the living room device and USB DAC1.


----------



## little-endian

@Regnad

 If this squeezeboxe doesn't change the data on its way to the DAC1, the sound will be the same.

 Within the claimed jitter performance by Benchmark, there is the simple formula "same data - same sound". Period.

 @Elias

 Hi there Elias, I answered your private message here in post #518 at Page 26. I suppose it was masked by the amount of postings here, just like a tiny little LSB while an explosion occurs.


----------



## EliasGwinn

Quote:


  Originally Posted by *puntloos* /img/forum/go_quote.gif 
_Elias: thanks again for your insightful and patient messages.

 One last and very important question:

 What are you guys @ Benchmark up to right now? Built a cool dac, now sitting on your behind raking in the cash? Chatting on forums? 
	

	
	
		
		

		
		
	


	




 Or can we await a DAC2 anytime soon? (two more and you've caught up with Bel Canto! *ducks*)_

 

The answer to this is highly classified. 
	

	
	
		
		

		
		
	


	




 Seriously, we don't talk about the products which we are currently developing for lots of reasons. But we have just released a 4-channel microphone pre-amp that has better performance then any mic-pre we can find. That is, it has the lowest distortion, noise, and RF susceptibility then any mic pre we know of.

 As I said, we can't talk about future products, but hopefully we will have another audiophile product release soon. 

 Thanks,
 Elias


----------



## EliasGwinn

Quote:


  Originally Posted by *Regnad* /img/forum/go_quote.gif 
_I have really enjoyed reading this thread, particularly since I just added a DAC1 to my Squeezebox and it sounds great! The music is in FLAC on a computer in another room and I have used both wired and wireless connections.

 My question:
 How would this setup compare to a local computer feeding a USB DAC-1? What would the differences be, both digitally and sonically? 

 I would really appreciate any information from those who have experience with both.

 Thanks a lot..._

 

I have not tested the Squeezebox, so I can't tell you whether or not it is bit-transparent. But, if we assume that it is, then the Squeezebox->DAC1 should have the same sonic performance as a local PC->DAC1 (save for any signal drop outs with either the Squeezebox or local PC.)

 Thanks,
 Elias


----------



## Lord Chaos

I bought my first DAC1 to use with a Squeezebox; this combination worked so well that, when I got the Email about the new USB version I promptly bought one to use when I'm in the living room. Listening to the same songs from the same computer that serves the Squeezebox sounds the same: a sonic delight.


----------



## Regnad

Is there any advantage to having the computer and drive in the living room, other than the user interface? 

 A bit OT, hope that's OK...


----------



## puntloos

Quote:


  Originally Posted by *Regnad* /img/forum/go_quote.gif 
_Is there any advantage to having the computer and drive in the living room, other than the user interface? 

 A bit OT, hope that's OK..._

 

The extra distance in cable (we assume) will increase the possibility of introducing rogue signals into your cable. Also, it dampens the signal a bit (not that much) and could introduce extra jitter. Additionally, if you are using grounded equipment you could create a bad earth loop which essentially means you introduce hum


----------



## EliasGwinn

Quote:


  Originally Posted by *Regnad* /img/forum/go_quote.gif 
_Is there any advantage to having the computer and drive in the living room, other than the user interface? 

 A bit OT, hope that's OK..._

 

Hello Regnad.

 I'm not sure I completely understand your question. Which user interface are you referring to?

 Thanks,
 Elias


----------



## Lord Chaos

Regnad, having the computer in the living room works for me as it's on my desk, and that's where I do my listening. My furniture and arrangement are atypical. 
	

	
	
		
		

		
			





 For those with a real living room, access to the computer would be different. Some PCs come with a remote control for music access. You could use Apple's Airtunes and their remote, or you could have a Squeezebox in your living room connected to a computer elsewhere. The Squeezebox remote would give you full control.


----------



## iceyo

dont think it worth $300 extra


----------



## Crowbar

Very long USB cable could cause increased error rate on some systems. Don't forget USB Audio doesn't use error correction.


----------



## EliasGwinn

Quote:


  Originally Posted by *little-endian* /img/forum/go_quote.gif 
_Interesting statement, Elias.
 I already asked myself why Benchmark doesn't list the dynamic range of the DAC1. Although this value is often used equally to the signal to noise ratio, some seem to distinguish between them. For example, according to the mastering engineer Bob Katz, one can hear details below the noise level, thus the dynamik range can be greater than the SNR, especially in conjunction with dithering (as far as I remember he gave 91 dB SNR and ~ 116 dB dynamik range for properly dithered 16 bit material).

 Now it would be interesting to know how great the dynamic range (!) of the DAC1 actually is. If it should be really able to resolve the 24th bit, it would have to exceed 140 dB. Is this the case?

 I'm confused also why more than 20 Bit of wordlength are used, at all if no converter is actually able to reach such a huge SNR and dynamic range. Many devices don't even match 20 bit performance (by pure math).

 I'm sure you can clarify this. Again this would be worth an own thread.

 little-endian_

 

Little-endian mentioned that I missed this question, so I'll try to address it here.

 The DAC1 has a S-to-N and Dynamic Range that is very close to the D-to-A chip used in the DAC1 - the AD1853. Therefore, the DAC1 has an S-to-N of ~ 117 dB and a Dynamic Range of ~ 116 dB. This, however, does not indicate that the 24th bit cannot be accurately realized. Conversely, the 24th bit can absolutely be realized, but it will be almost 30 dB below the noise floor. The human ear can discern tones more then 30 dB below the noise floor, so the 24 th bit is audible in the DAC1. We have confirmed this using an FFT and with listening tests.

 The human-hearing threshold is not dependent on bit-depth. If the 32nd bit of a 32-bit audio word is accurately reproduced, and the noise floor is sufficiently low (<160 dB), the tone will be audible (if there is enough clean amplification).

 Therefore, even if a system doesn't have a noise floor as low as the theoretical limit (-144 dB for 24-bit), the 24-th bit is still audible, and therefore valuable. (However, global-system limitations say if you have noisy analog electonics (including amplifier), then the 20-24 bits may be inaudible). 

 Thanks,
 Elias


----------



## Crowbar

Quote:


  Originally Posted by *EliasGwinn* /img/forum/go_quote.gif 
_The human-hearing threshold is not dependent on bit-depth._

 

I think this statement is unintentionally misleading, since if you have sufficient bit depth, then depending on levels, you will hit either hearing threshold, pain threshold (and above it, deafness threshold), or both. The human hearing dynamic range is what's at issue here. Levels should be matched to cover the full usable range, so that the lowest quantization code is at hearing threshold, and highest one below pain threshold for good safety margin (damage starts occuring below that threshold) -- that is, 120 dB is plenty. Within that, increasing bit depth beyond a certain limit will be inefficient since differences between nearest codes would be imperceptible (I'm simplifying here by assuming detectable differences are equal throughout the hearing range; they are not). If you have a 32 bit word as in your example, and the highest PCM code doesn't cause immediate hearing damage, then the lowest one will be far below the threshold of audibility. Hex 00000001 that is audible would mean an even momentary hex FFFFFFFF will cause instant hearing loss.


----------



## music_man

usb has a specified length limit. i think it is 15' or there about. you can increase this to about 45' with amplification. you will run into errors with an amplified signal. for data this is not a huge problem for casual computing. we would never use usb over 6' in pro audio. neither would users of mission critical bussiness applications. for music enjoyment it is up to you what level of error you can deal with. at the point of dropouts i would pass.

 if you need long runs i'd go with a bit perfect wireless solution or usb>cat5, coaxial etc.

 music_man


----------



## AndreYew

Quote:


  Originally Posted by *EliasGwinn* /img/forum/go_quote.gif 
_Conversely, the 24th bit can absolutely be realized, but it will be almost 30 dB below the noise floor. The human ear can discern tones more then 30 dB below the noise floor, so the 24 th bit is audible in the DAC1. We have confirmed this using an FFT and with listening tests._

 

This may be correct empirically, but the reasons stated are not. When you say noise floor, you mean the SNR over the entire measured bandwidth. When humans hear or when you use an FFT to pick up your 24th bit, that's done over the bandwidth of the ear's critical band at that frequency or the bin width of the FFT. If noise is white, and the signal fits within that bandwidth (bin or critical band), then the amount of noise will be reduced proportionally to the bin or critical bandwidth. For example, if your SNR measurement bandwidth is 20 kHz, and your bin width is 100 Hz, there will be 100/20000= 1/200 or 23 dB lower noise floor in your bin than the full 20 kHz SNR measurement.

 Bandwidth considerations aside, I think experiments have shown that humans will hear something like 10+ dB below the noise for the same bandwidth. But this all depends on the kind of signal you have. If you have a very pure tone (narrow spectral line), and if you measure long enough, you'll find it since you are essentially making a very narrow passband filter to reduce the noise floor to the level where you can detect the tone. But if you have a broadband signal, like a lot of music, then things get very complicated.

 --Andre


----------



## Crowbar

The AD1853 is simply not linear to 24 bits. There's significant offset at -110 dB as shown in the datasheet, and that makes the resolution of signals below that largely irrelevant since they're smaller in magnitude. Indeed, it would've been better for ADI to do what they did with the 1955 and shape it as noise, which is better than distortion.


----------



## puntloos

Quote:


  Originally Posted by *music_man* /img/forum/go_quote.gif 
_usb has a specified length limit. i think it is 15' or there about. you can increase this to about 45' with amplification. you will run into errors with an amplified signal. for data this is not a huge problem for casual computing. we would never use usb over 6' in pro audio. neither would users of mission critical bussiness applications. for music enjoyment it is up to you what level of error you can deal with. at the point of dropouts i would pass.

 if you need long runs i'd go with a bit perfect wireless solution or usb>cat5, coaxial etc._

 

The official max USB cable length is 5m.

 Wether or not a specific device can tolerate 10 or 15m depends on its design. 10m is usually safe but don't hurt me if you run into trouble.


----------



## EliasGwinn

I would like to share an experiment which we just conducted here. I took a classical piano recording and loaded it into Nuendo. Using a Lynx AES16, I piped the digital output to a DAC1, then sent the analog output of the DAC1 to an ADC1 (our A-to-D converter). I recorded the digital output of the ADC1 back into Nuendo. I then sent this 2nd generation recording through the same signal chain to record a 3rd generation. I did this 20 times, so that the final recording had been converted in and out of the analog domain 20 times.

 We then streamed the original recording and the 20th generation recording via AES XLR's to our listening room (acoustically treated 12'w x 14'l x 9'h with Klein and Hummel 4-way system). We put both the original recording and the 20th generation into an ABX switcher and conducted listening tests with the monitors and with Sennheiser HD600 headphones. After several tests by several people, the results averaged to 50% accuracy. In other words, no one could reliably discern the difference between the original and the 20th generation!! As hard as I tried to listen to differences in things such as noise floor, recorded room noise, mechanical (pedal, hammer, etc) noise, attacks, decays, reverb tails, etc.... I simply could not find any differences!!

 We are currently requesting the permission by the artist and label to post these recordings on our website to let users experience this test. I will keep you posted.

 Thanks,
 Elias


----------



## islewind

Quote:


  Originally Posted by *EliasGwinn* /img/forum/go_quote.gif 
_ In other words, no one could reliably discern the difference between the original and the 20th generation!!_

 

Very impressive. I appreciete the rigor with which you conducted your testing. It seems to me that if your test has been described in theorital terms as a "what-if", many many people would have contended that they could very easily hear the difference. 

 Very, very impressive! My friend's DAC1 is due to arrive any day now, I am eager to hear what it adds to his system.


----------



## Lord Chaos

I find this fascinating. Here we are discussing equipment that can actually give a usable signal at -116dB. If I'd had that when I was recording in the 1970s, I'd have considered it magic. I thought a 60dB dynamic range was pretty neat.

 I was listening last night to one of my LP transfers on my usual system (computer, squeezebox, DAC1, HD25) and the sound was just delightful. Better than I've heard, which reinforces my growing belief that the main difference I have heard between analog and digital sources comes from the final digital-to-analog conversion. I wouldn't stake my life's savings on a bet, bu this is the way my thoughts are running. Dynamic range is just part of the story.


----------



## audioengr

Only mildly interesting IMO. The system is too large a variable. Even the headphones are not very good. I've used these same headphones extensively and my own reference speaker system is much more live-sounding and extended with better detail and imaging etc...

 What would be interesting is the THD+N measurements and phase and group delay measurement, particularly with a transient signal, like an impulse or square-wave, not a steady-state sine-wave.

 Steve N.


----------



## Crowbar

I disagree with audioengr. Metrics that are not perceptually weighted are of limited utility. Blind listening tests are the answer, but if conducted properly.

 The most sensitive to distortion information the brain gets from music is spatial information, not tonal. Thus one should use a good binaural recording with very high quality headphones, since non-binaural recording and playback geometry guarantees that spatial information is ruined and the soundfield cannot be recreated properly at the ears.

 A couple of years ago I was playing around with opamps, and I noticed that chaining multiple opamps in series (unity gain) didn't seem to cause an audible effect, until I paid careful attention to imaging (it was a binaural recording)--there was a slight reduction of my ability to localize the sound sources.

 Conducting a blind test this way is difficult. One problem is that binaural recordings are made with different HRTFs than the listener's and even small variations affect auditory localization--a testament to the incredible sensitivity to distortion of spatial information. An alternative is to use a convolver plugin in the player application and HRTF data from the database; there are sets from many subjects there so one can usually find a set among those that works well for them.

 Unfortunately, I'd have to say most headphones are immediately disqualified when one looks at the terrible response curves. Electrostatics like Stax Omega 2 are low distortion but in bad need of equalization. The ultimate driver for such testing would be the Plasmasonic headphones; nothing I've seen in the speaker or headphone world comes close to these curves:


----------



## EliasGwinn

I will be repeating this test with different sources and D-to-A devices soon. I will be using the same A-to-D to maintain a single variable. In other words, everything will stay exactly the same except another D-to-A will be used instead of the DAC1.

 I'll let you know how it sounds.

 By the way, if anyone is in the neighborhood, we'd love to have you stop by and participate. You don't even have to call ahead, just stop by M-F, 9-5 EST. We can even use your favorite reference material and headphones (and/or headphone amp and/or D-to-A). 

 Thanks,
 Elias


----------



## EliasGwinn

Quote:


  Originally Posted by *audioengr* /img/forum/go_quote.gif 
_Only mildly interesting IMO. The system is too large a variable. Even the headphones are not very good. I've used these same headphones extensively and my own reference speaker system is much more live-sounding and extended with better detail and imaging etc...

 What would be interesting is the THD+N measurements and phase and group delay measurement, particularly with a transient signal, like an impulse or square-wave, not a steady-state sine-wave.

 Steve N._

 

Steve,

 No problem. We're planning to conduct these tests very soon. We'll post the results as soon as they are available.

 Thanks,
 Elias


----------



## Crowbar

Slightly off-topic, how does the best AD technology compare to the best DA? Which side needs catching up? Or are both technologies at the same level?


----------



## Crowbar

Quote:


  Originally Posted by *EliasGwinn* /img/forum/go_quote.gif 
_No problem. We're planning to conduct these tests very soon. We'll post the results as soon as they are available._

 

Since you have an ADC, it would be trivial for you to perform this measurement:
http://www.essex.ac.uk/ese/research/...%20testing.pdf
 No doubt H. would share his MATLAB code. I know I've suggested this before, and I still think it is a very good test.


----------



## zheka

Quote:


  Originally Posted by *EliasGwinn* /img/forum/go_quote.gif 
_I will be repeating this test with different sources and D-to-A devices soon. I will be using the same A-to-D to maintain a single variable. In other words, everything will stay exactly the same except another D-to-A will be used instead of the DAC1._

 

elias

 it may be interesting to compare originals to the resulting files with audio diffmaker 

http://www.libinst.com/Audio%20DiffMaker.htm

 what do you think?

 respectfully
 gd


----------



## clar2391

I've read some of your technical support documents at the Benchmark website referencing jitter after hundreds of feet of coax carrying a digital signal. I have a question along those lines regarding USB cable. I believe the limit to USB cable length is 15 meters. Is the DAC1 USB as immune to jitter on such long runs of USB cable in the same way that it is immune to jitter over long runs of coax cable?

 thanks in advance.


----------



## Crowbar

There is an issue that error rates go up with very long USB cables. This is separate from the DAC's jitter immunity, it's potential data corruption and loss depending on the specific system configuration.


----------



## EliasGwinn

Quote:


  Originally Posted by *clar2391* /img/forum/go_quote.gif 
_I've read some of your technical support documents at the Benchmark website referencing jitter after hundreds of feet of coax carrying a digital signal. I have a question along those lines regarding USB cable. I believe the limit to USB cable length is 15 meters. Is the DAC1 USB as immune to jitter on such long runs of USB cable in the same way that it is immune to jitter over long runs of coax cable?

 thanks in advance._

 

Clar,

 The limit to USB cable is 5 meters. We have not done testing on this; it is information from Wikipedia.

 As Crowbar mentioned, long USB cables won't suffer from jitter, they'll be more prone to signal drop-outs etc. 

 Thanks,
 Elias


----------



## Telynau

I have a short question concerning the proper Quicktime setting in iTunes. I do not have the Pro version of Quicktime on my MacBook. The non-Pro version doesn't have available the following sequence, which was provided to help users make sure the sample rate of the source material was properly reflected in Quicktime: QuickTime -> Edit -> Edit Preferences -> QuickTime Preferences -> Audio -> Sound Out -> Rate. Is there any setting we non-Pro users need to deal with in Quicktime? 

 Thanks -- James


----------



## tonygeno

You should be able to find those settings in the Audio Midi Setup which is in the Utilities folder.


----------



## EliasGwinn

Quote:


  Originally Posted by *Telynau* /img/forum/go_quote.gif 
_I have a short question concerning the proper Quicktime setting in iTunes. I do not have the Pro version of Quicktime on my MacBook. The non-Pro version doesn't have available the following sequence, which was provided to help users make sure the sample rate of the source material was properly reflected in Quicktime: QuickTime -> Edit -> Edit Preferences -> QuickTime Preferences -> Audio -> Sound Out -> Rate. Is there any setting we non-Pro users need to deal with in Quicktime? 

 Thanks -- James_

 

James,

 Check out this article about setting up iTunes on Mac:

http://extra.benchmarkmedia.com/wiki..._-_Setup_Guide

 Thanks,
 Elias


----------



## Telynau

Thanks, Elias. I have a Wavelength Audio Cosecant Version 2.0 USB DAC and on my MacBook, in Audio Midi Setup, once 44.1KHz is selected (that is all the Cosecant will play) the only bit length available is 16. So I guess that simplifies things for me. 

 Based on your posts in this thread I think I am going to get one of your USB DAC's to compare with the Cosecant. Very different animals, of course. Given the ease of setup of your USB DAC I may even build a dedicated PC-based playback system if I get energetic. I have an old laptop that might make a decent music player if I strip all the unneeded junk off it. 

 Regards, James


----------



## EliasGwinn

Yeah, old computers are good for that. Especially if you get an external firewire hard drive. Be careful of USB hard drives though...some old computers struggle when they try streaming data in and out simultaneously, and under the workload, it may break up the audio stream. The only way to know is by trying though. Perhaps you could borrow a friend's USB drive, or just buy one and return it quickly if it doesn't work.

 Thanks,
 Elias


----------



## nicke2323

Question to Elias:

 I just posted a link to several ABX tests comparing low-end to high-end CD players:
http://www.head-fi.org/forums/showthread.php?t=246048

 I have been considering an upgrade to the Benchmark DAC1, but as I mention in the above thread, ABX tests like these are starting to make me reconsider. Since your recent 20-generation D/A-A/D loop test shows that you at Benchmark are no strangers to ABX testing, perhaps you can shed some light on the sonic advantages of a high-end DAC like the DAC1?

 More specifically, do you think the DAC1 can be distinguished from modern entry-level CD-players in an ABX test? Have you done such tests at Benchmark?

 I have no doubt that the DAC1 measures very close to perfection, as all reviews seem to indicate. But I am not certain this translates to an audible difference. Does it?


----------



## Crowbar

ROFL, the Audio Note DAC3 is a 12 year old NOS DAC, of course others would be preferred.

 None of these tests were performed to ITU standards (see Recommendation Recommendation BS.1116-1), thus you can safely ignore them. Various of Peter Aczel and related (Douglas Self, Rod Elliott) 'debunking' claims have been themselves debunked in forums such as diyaudio and so on. Self's 'blameless' class B comes to mind (not so blameless after all heh). The Audio Engineering Society wouldn't be accepting papers on distortion any more if this was a solved issue as you're incorrectly implying. I have very rarely seen properly conducted blind tests as it is a huge undertaking. You can't just put people in a room for a few hours and perform a quality test; even the ITU recommendations are lacking if one is aiming for perfect guarantee. I've mentioned above the Pass comment so it's worth finding it in the thread.

 Also refer to the sticky thread in the DIY section here entitled "Subjective vs measurements in the perception of sound quality"; most of the points here have been touched upon there already.


----------



## audioengr

ABX tests are only as good as the system used and the listeners ears and training. The system is usually the culprit. If stereo salons cannot put together a decent sounding system, how can you expect reference systems to be used in these tests? It tends to be "professionals" that do these tests.

 Steve N.


----------



## mofonyx

Regarding the non-USB Benchmark DAC1, I have heard that the BNC digital coax input is not truly 75 ohm and this would cause some issues when connecting via coax from true RCA 75 ohm digital outs.

 Is this true and can anyone verify this?


----------



## dip16amp

Quote:


  Originally Posted by *mofonyx* /img/forum/go_quote.gif 
_Regarding the non-USB Benchmark DAC1, I have heard that the BNC digital coax input is not truly 75 ohm and this would cause some issues when connecting via coax from true RCA 75 ohm digital outs.

 Is this true and can anyone verify this?_

 


 This was asked in this thread on page 21 post 419 and answered on page 22 post 424.


----------



## EliasGwinn

Quote:


  Originally Posted by *nicke2323* /img/forum/go_quote.gif 
_Question to Elias:

 I just posted a link to several ABX tests comparing low-end to high-end CD players:
http://www.head-fi.org/forums/showthread.php?t=246048

 I have been considering an upgrade to the Benchmark DAC1, but as I mention in the above thread, ABX tests like these are starting to make me reconsider. Since your recent 20-generation D/A-A/D loop test shows that you at Benchmark are no strangers to ABX testing, perhaps you can shed some light on the sonic advantages of a high-end DAC like the DAC1?

 More specifically, do you think the DAC1 can be distinguished from modern entry-level CD-players in an ABX test? Have you done such tests at Benchmark?

 I have no doubt that the DAC1 measures very close to perfection, as all reviews seem to indicate. But I am not certain this translates to an audible difference. Does it?_

 

Nicke,

 I've read the thread you posted, but I choose to steer clear of such debates for a number of reasons (particularly because people get very angry in these types of debates 
	

	
	
		
		

		
			





) 

 In response to your question "What are the sonic advantages of the DAC1"...: To answer this, I'd like to address the debate of objectivity vs. subjectivity. If you can permit a metaphor for the sake of explanation, audio is like a picture. Some people prefer classic black and white pictures, some people like Sepia, some like classic film, some like crisp digital, etc. All of these preferences are absolutely valid...they should not be discredited as user inexperience / ignorance / etc. Subjectivity is as real and valid as objectivity....but very different.

 Subjectivity has the uncanny ability to change over a period of time. If you looked at the world through Sepia lenses 24/7, you would be in awe if you caught a glimpse of full, natural color, even if it is a blurry, out-of-focus image. You see my point?

 Our goal with the DAC1 is to offer 20/20 vision for D-to-A conversion. This is why measurements are very important. It is important to know exactly what is happening to the audio from a completely objective standpoint. Objectivity does not change over time. That is not to say that listening (subjectivity) is less important...its apples and oranges. 

 In response to your question "Do you think the DAC1 can be identified vs. a cheap CD player"...: This depends on the listener and the test setup. But, regardless of whether or not one can discern between the two in an ABX test, the inaccuracies of the CD player will be apparent more when you listen for long periods of time. 

 If we may use the picture analogy again: if someone put a film picture next to a digital picture, some people could tell the difference, and some would not be able to. However, if you looked at a film picture everyday for a few days/weeks/months, then saw the same image but captured digitally, the difference would jump right at you. Thats not to say one is better then the other, but the differences are accumulative and, therefore, more pronounced over time.

 This accumulative difference is why we conducted the 20-generation test. If there were severe artifacts in the D-to-A conversion process, they would be exposed in early generations. If the conversion is well done, the accumulative affect should be minimal.

 I apologize for all the metaphors...sometimes I feel its easier to make a point that way. I can elaborate on any point(s) if this isn't clear enough.

 Thanks,
 Elias

 ps. If the root of your question is, "Should I buy a DAC1 even if I don't hear a difference", that is something only you can answer. If you're happy with your CD player as-is, then don't buy a DAC1. Even if it is better, happiness is a state of mind, not a signal-to-noise ratio. However, if you want to hear every detail of the music without missing something thats hiding behind jitter-induced artifacts, then the DAC1 can and will achieve that.


----------



## EliasGwinn

Quote:


  Originally Posted by *mofonyx* /img/forum/go_quote.gif 
_Regarding the non-USB Benchmark DAC1, I have heard that the BNC digital coax input is not truly 75 ohm and this would cause some issues when connecting via coax from true RCA 75 ohm digital outs.

 Is this true and can anyone verify this?_

 

Mofonyx,

 I will quote John Siau, the engineer who designed the DAC1:

 "The connector is 50-Ohms. 50-Ohm connectors are far more durable than 75-Ohm connectors due to the extra dielectric material surrounding the center pin of the BNC. For this reason, it is common practice to use 50-Ohm BNC connectors in 75-Ohm systems when the signal bandwidth allows it.

 "The short interruption of the 75-Ohm transmission line is only significant for frequencies that are much higher than any contained in a digital audio signal. The 50-Ohm connector would be a factor for signals having a wavelength of 2 inches or less in coax (about 3 GHz). A 192 kHz digital audio signal transmits data using a clock that is 64 times the sampling frequency (192 kHz * 64 = 12.288 MHz). 3 GHz is the 244th harmonic of 12.288 MHz and does not exist in a 192 kHz digital audio signal. If it did, the box probably would not pass FCC and CE emissions tests.

 "Changing the connector would reduce the durability of the product and would have absolutely no effect even at 192 kHz." -John Siau

 Thanks,
 Elias


----------



## nicke2323

Thank you Elias for taking the time to reply to my questions! A few comments:

  Quote:


  Originally Posted by *EliasGwinn* /img/forum/go_quote.gif 
_I've read the thread you posted, but I choose to steer clear of such debates for a number of reasons (particularly because people get very angry in these types of debates 
	

	
	
		
		

		
		
	


	




)_

 

I'm pretty sure nobody here would take offense if you stated you had performed an ABX test and experienced listeners could easily tell the Benchmark from an entry-level CD player.

 That's all I want to know. Frankly, the fact that not even an engineering-oriented company like Benchmark will make a clear statement like this seems to imply that the difference may be too small to hear.

 Like I said before, I have no doubt that the DAC1 measures very close to perfect, and 20 generations of D/A-A/D looping introduce no audible artefacts. But would the result of the 20 generation ABX test be different if you used the DAC of a cheap CD player?

 Unfortunately I have no possibility of auditioning the DAC1 without ordering one for 1100 euros - quite a price to pay for something that may turn out to have no perceivable sonic improvement for me. (Incidentally, I would appreciate if you guys could tell the European distributors to stop selling the silver version of the DAC1 at a 100 euro markup, when the price is the same as the black rack version in the US. This is in addition to the large markup they have on the original version. US price $975 = 730€, cheapest Euro price 1100€. Taxes and import duties explain less than half of the difference.)

 In any case, it does inspire confidence that you are here, in active discussion with your potential customers. If I buy anything it will be a Benchmark.


----------



## The Monkey

Quote:


  Originally Posted by *nicke2323* /img/forum/go_quote.gif 
_I'm pretty sure nobody here would take offense if you stated you had performed an ABX test and experienced listeners could easily tell the Benchmark from an entry-level CD player._

 

Let's leave ABX tests out of an otherwise excellent thread.


----------



## little-endian

Quote:


  Originally Posted by *nicke2323* /img/forum/go_quote.gif 
_That's all I want to know. Frankly, the fact that not even an engineering-oriented company like Benchmark will make a clear statement like this seems to imply that the difference may be too small to hear._

 

Well, if you have a look into the manual of the DAC1, you can see that Benchmark is a little more brave with their promises here. There they state that they think, the reduced jitter will result in a clearly audible improvement. After listening to the DAC1 via Sennheiser HD-650 for the first time, I felt that I was able to follow that claim. It sounded somehow 'strange', like the midrange was reduced - exactly what is written in the manual. Interestingly, I made this experience *before* I read the text.

 However when I tried the DAC1 on my regular loudspeakers (a pair of Nubert nuWave 8, connected to a David Hafler 110 preamp and Kenwood M1D amplifier), I couldn't distinguish it from a Yamaha DVD-S540 (for about 80 EUR in Germany) which I felt pretty frustrating to be honest. I have to admit that the equipment is not really highend but nevertheless, I expected at least any difference (no matter if better or worse).

 I still think that the output of the Sennheiser headphones is 'reference' through a DAC1, probably due to its superb preamp function and match for the impedance, etc. I haven't compared 24-bit material yet though. Just plain 44.1 kHz/16-Bit.

 I'm not sure but I feel that somehow the conversion process from digital to analog seems to be quite exhausted.

 At the end, the DAC1 - as an external device - is more some kind of prove of concept for me. One can use the cheapest stuff as source as long as the data will be correct, the sound will be fine. Thanks to its great mute function, there are no crackling noises or pops if something goes wrong on its transfer.


----------



## EliasGwinn

Quote:


  Originally Posted by *nicke2323* /img/forum/go_quote.gif 
_T
 I'm pretty sure nobody here would take offense if you stated you had performed an ABX test and experienced listeners could easily tell the Benchmark from an entry-level CD player.

 That's all I want to know. Frankly, the fact that not even an engineering-oriented company like Benchmark will make a clear statement like this seems to imply that the difference may be too small to hear._

 

Nicke,

 I apologize for missing this question. We have not performed any such ABX tests with an inexpensive transports analog out vs. the DAC1. 

 Thanks,
 Elias


----------



## nicke2323

Quote:


  Originally Posted by *The Monkey* /img/forum/go_quote.gif 
_Let's leave ABX tests out of an otherwise excellent thread._

 

OK, I'm starting to see what you mean. I apologize for polluting. To anyone else who may be tempted to respond with ABX-related comments, please do so here instead:

http://www.head-fi.org/forums/showthread.php?t=246048

 And thanks again Elias for the reply.


----------



## mofonyx

Thank you for replying to an old question Elias and thanks dip16amp for referring properly to the post on the thread. I had some really informative reading. 
	

	
	
		
		

		
		
	


	





 I would like to add a question to my post because I can't seem to find these in the thread currently:

 How are the balanced XLR volume outputs controlled when the volume control is set to "Variable"? 
 Is the headphone amp on the DAC1 ultilised as a "preamp" to control the volume output on the XLR outputs on the rear?

 Thank you.


----------



## puntloos

Even with the level of candor Elias has been consistently showing, I doubt a he would ever say 'yeaaah well we tested until our ears started to bleed but the DAC that came with this bag of cornflakes was indistinguishable from our DAC1.'. Maybe his findings are in fact that their DAC1 is much better sounding, but even if it truely is so, his claim can't avoid sounding hollow either way, so it's best to stay away. 

 And yeah, please no ABXing in here, people have spent thousands of dollars on gamma radiation filters and will get very cranky if ever confronted with any solid facts proving it was wasted money.

 What ive been trying to find out myself is the impact of the choice of resampling frequencies. Benchmark (through Elias) claims that their choice of 100-ish kHz (so they downsample a 192K source) is the 'best choice' given the state of the art in DAC and sampler chips on the market. I would be interested in ABX comparisons of sources:

 192khz, nonresampled
 192khz, resampled to 100k
 96khz, nonresampled
 96khz, resampled to 100k

 Elias, a specific question if you will: The DAC1 resamples everything to 105 (or thereabouts) kHz. I get that there are arguments for downsampling from 192K to 105, and I get that there are good reasons for upsampling 44K to 105, but Ive always been taught that upsampling to neighboring frequencies (44->48 specifically, but I imagine it applies to 96->105 too) is a bad idea. Why do benchmark do this anyway? If the source is 96, why not pass it through 1:1? Why did you not provide a button to turn off resampling, especially in this case, but also in general?


----------



## EliasGwinn

Quote:


  Originally Posted by *mofonyx* /img/forum/go_quote.gif 
_
 How are the balanced XLR volume outputs controlled when the volume control is set to "Variable"? 
 Is the headphone amp on the DAC1 ultilised as a "preamp" to control the volume output on the XLR outputs on the rear?_

 

When the DAC1 is in variable mode, the balanced XLR outputs are controlled with the potentiometer on the front panel. 

 The headphone amp is isolated from the XLR outputs. Both the headphone amp and the XLR output drivers (when in variable mode) are 'down-stream' from the potentiometer.

 I can clarify further if you wish.

 Thanks,
 Elias


----------



## gregeas

Regarding digital sources, my experience is that they are very difficult to tell apart when switching after short listening sessions (a few minutes each). For example, I could not really distinguish between a Squeezebox 2 and my $3k Arcam CD player. My pre-amp allows me to level match the inputs and switch them remotely, so this kind of test is very easy. 

 All of this is an argument for _not _going for a source that is more exotic than the DAC1. Especially in light of the fact that you are getting a digital pre-amp and head-amp as part of the package. All for approximately $1k.

 If the DAC1 could be controlled by a remote, I'd ditch my dedicated pre-amp in a second. 

 Right now you can enter the realm of the high end with a laptop, DAC1, and nice powered speakers like the Genelec 8020As. This is my work right, along with the E500s.


----------



## EliasGwinn

Quote:


  Originally Posted by *puntloos* /img/forum/go_quote.gif 
_The DAC1 resamples everything to 105 (or thereabouts) kHz. I get that there are arguments for downsampling from 192K to 105, and I get that there are good reasons for upsampling 44K to 105, but Ive always been taught that upsampling to neighboring frequencies (44->48 specifically, but I imagine it applies to 96->105 too) is a bad idea. Why do benchmark do this anyway?_

 

With ASRC (Asynchronous Sample Rate Conversion), the input sample-rate is irrelevant to the output sample-rate. The ASRC takes the input data and interpolates 2^20 samples of it. In other words, the intermediate sample-rate is ~1.05 MHz. The ASRC then samples that data stream by the prescribed output sample-rate (110 kHz, in the case of the DAC1). Therefore, the ASRC achieves equal performance converting 44->110 kHz or 96->110 kHz.

  Quote:


  Originally Posted by *puntloos* /img/forum/go_quote.gif 
_If the source is 96, why not pass it through 1:1? Why did you not provide a button to turn off resampling, especially in this case, but also in general?_

 

I'll have to split these questions because they have two different answers. I'll answer the second question first, because it makes more sense that way.

  Quote:


  Originally Posted by *puntloos* /img/forum/go_quote.gif 
_"Why did you not provide a button to turn off resampling?"_

 

This is all about jitter. Jitter is the single worst enemy in digital conversion. It introduces tones which are completely unrelated to the audio...mainly in the mid-range frequencies (because of fold-back).

 The ASRC is the backbone of the UltraLock technology which makes the DAC1 immune to jitter. Even with the most expensive digital interconnect cables that money can buy, there will be transmission jitter. Even with the most exotic connectors on the source and/or DAC, there will be interface jitter. This is very well known among the audio engineering community. The question is, how are we going to deal with this jitter? Are we going to use a two-stage phase-lock-loop to attenuate it as much as possible (but not completely)? This is what most high-end DAC's use. 

 The ASRC gives us the ability to build a clocking system which will be independent of the clock of the incoming data stream, making it immune its jitter. So, regardless of the incoming sample-rate, the jitter on the signal will not affect the D-to-A conversion.

  Quote:


  Originally Posted by *puntloos* /img/forum/go_quote.gif 
_"If the source is 96, why not pass it through 1:1? "_

 

In case you mean, _"Why didn't you simply bypass sample-rate conversion?"_...the reason is the data would be laden with jitter. The UltraLock system needs to be engaged at all times to eliminate the artifacts of jitter.

 If, on the other hand, you meant _"Why didn't you simply convert the sample rate to its original sample rate?"_ We could have chosen to do exactly this, even with the ASRC. That is, we could have maintained the ASRC as well as the rest of the UltraLock system, while forcing it to output the same sample-rate as the incoming rate. The reason why we didn't do this is:

 1. As mentioned above, the output sample-rate of the ASRC is irrelavant to the input sample-rate. Therefore, we would not have gained any performance whatsoever by re-sampling at the original sample-rate.

 2. 110 kHz offers us the widest analog bandwidth without reducing the performance of our D-to-A converter. As I mentioned before, D-to-A chips that are currently available by IC manufacturers do not perform well above 115 kHz (or thereabouts). This is why many professionals state that 192 kHz is a bad idea. Its all about the current state-of-the-technology. Currently, the D-to-A's available can't perform well at 192 kHz. 

 110 kHz gives us the full bandwidth conversion without approaching the 'Achilles heel' of the D-to-A.

 Thanks,
 Elias


----------



## mofonyx

Quote:


  Originally Posted by *EliasGwinn* /img/forum/go_quote.gif 
_When the DAC1 is in variable mode, the balanced XLR outputs are controlled with the potentiometer on the front panel. 

 The headphone amp is isolated from the XLR outputs. Both the headphone amp and the XLR output drivers (when in variable mode) are 'down-stream' from the potentiometer.

 I can clarify further if you wish.

 Thanks,
 Elias_

 

Please do explain further. 

 To my very superficial understanding/discussion, the XLR outputs cannot be *directly* controlled by the potentiometer on the front panel because it isn't a 4 channel potentiometer, and hence it cannot be used for balanced volume attenuation.


----------



## EliasGwinn

Mofonyx,

 I will explain a very simplified version of the signal path. The analog signal is sent from the D-to-A chip to a set of differential amplifiers. Then, those signals (L and R) are sent simultaneously to the calibration pots (on the rear of the unit) and to the volume pot (on the front of the unit). The front-panel volume pot sends the signal to the headphone outputs and also to the rear-panel switch ('Variable'/'Mute'/'Calibrated'). When this switch is in 'Variable' mode, the signal from the front-panel volume pot is sent to the XLR output drivers and the RCA output drivers. The rear-panel calibration pots send the signal to another input of the rear-panel switch ('Variable'/'Mute'/'Calibrated'). When this switch is in 'Calibrated' mode, the signal from the rear-panel calibration pot is sent to the XLR output drivers and the RCA output drivers. 

 The front panel pot is not a 4-channel pot because the signal is unbalanced at that stage of the circuit. The signal from that pot are the source to the balanced output drivers when the unit is in 'Variable' mode.

 I hope that clears things up.

 Thanks,
 Elias


----------



## mofonyx

All right, does this mean that it makes no difference what mode of volume control is used, be it Variable or Calibrated, it should be the same quality of volume control (save for the potentiometer's initial 10-20% inaccuracy) because they both travel the same path eventually?

 Am I right to assume that signals from the front panel and rear panel pots both has to pass the balance output drivers at some stage?


----------



## puntloos

Quote:


  Originally Posted by *EliasGwinn* /img/forum/go_quote.gif 
_With ASRC (Asynchronous Sample Rate Conversion), the input sample-rate is irrelevant to the output sample-rate. The ASRC takes the input data and interpolates 2^20 samples of it. In other words, the intermediate sample-rate is ~1.05 MHz. The ASRC then samples that data stream by the prescribed output sample-rate (110 kHz, in the case of the DAC1). Therefore, the ASRC achieves equal performance converting 44->110 kHz or 96->110 kHz._

 

Curious. I understand what you are saying but I am confused why this device operates just fine at 1Mhz, while many DAC's do not perform well at even 192Khz, as you claim. Ah well, of course there is more to a DAC then up-down-resampling but I always thought that was the most fragile part.

 But thank you for the explaination there.

  Quote:


 snip, thanks for explaining 
 

 Quote:


 
 2. 110 kHz offers us the widest analog bandwidth without reducing the performance of our D-to-A converter. As I mentioned before, D-to-A chips that are currently available by IC manufacturers do not perform well above 115 kHz (or thereabouts). This is why many professionals state that 192 kHz is a bad idea. Its all about the current state-of-the-technology. Currently, the D-to-A's available can't perform well at 192 kHz. 

 110 kHz gives us the full bandwidth conversion without approaching the 'Achilles heel' of the D-to-A. 
 

Could you be a bit more specific about in what way 192Khz D-A's don't 'perform well? Phase issues? Non-linearity? Jitter? what?

 Im currently trying to set up a test day where I will audition your DAC1 (the non-USB version probably) and the aforementioned Bel Canto DAC3. These two dacs are about as diametrically opposite as you can imagine.. BC uses an ASRC too, and upsamples everything to 192/24. It uses the Burr-Brown 1792A. The Bel Canto sticks to 96/24, and uses an AD chipset. 

 UltraClock versus UltraLock. Round 1. Fight!

 If there are no discernible audible differences I'll probably go for the DAC1 since the BC is 50% more expensive.

 Thanks again for your anwers so far.


----------



## EliasGwinn

Quote:


  Originally Posted by *mofonyx* /img/forum/go_quote.gif 
_All right, does this mean that it makes no difference what mode of volume control is used, be it Variable or Calibrated, it should be the same quality of volume control (save for the potentiometer's initial 10-20% inaccuracy) because they both travel the same path eventually?

 Am I right to assume that signals from the front panel and rear panel pots both has to pass the balance output drivers at some stage?_

 

This is all correct.

 Thanks,
 Elias


----------



## EliasGwinn

Quote:


  Originally Posted by *puntloos* /img/forum/go_quote.gif 
_...I am confused why this device operates just fine at 1Mhz, while many DAC's do not perform well at even 192Khz, as you claim._

 

Puntloos,

 The up-sampling is an interpolation of the data, it is not actually the new sample-rate. The converter interpolates the data by 2^20, then selects every Xth sample, where X=# of interpolation samples/output-sample rate. 

 With 192 kHz, the chip can't interpolate as many samples, and therefore the resolution is compromised. 

  Quote:


  Originally Posted by *puntloos* /img/forum/go_quote.gif 
_Could you be a bit more specific about in what way 192Khz D-A's don't 'perform well? Phase issues? Non-linearity? Jitter? what?_

 

The problem with 192 kHz is all about the filtering. I will cut and paste a post I made before concerning this on page 30 of this thread (http://www.head-fi.org/forums/showpo...ostcount=590):

 The filters in a D-A chip ideally pass all audio below the Nyquist frequency and block (filter) all audio above the Nyquist frequency. In reality, however, there will be some audio below the Nyquist that is being filtered some, and there will be some audio above the Nyquist which is not filtered enough. The latter is very dangerous because those frequencies will be aliased and cause distortion.

 Here's the problem with 192 kHz: the filter used for 192 kHz is of far less quality. This is true of ALL D/A chips on the market. What happens is this: the filter cut-off becomes less defined, causing audio below Nyquist to be attenuated. And, more importantly, AUDIO ABOVE NYQUIST IS NOT FULLY ATTENUATED!! The filter does not do its job as well at 192 kHz.

 Another 'real-life' limitation to these filters is amplitude ripple. This means that the audio below Nyquists will have ripples in amplitude across the frequency spectrum. This is equivalent to inaccurate frequency response. This is also something that happens in all D-to-A chips, some more so then others.

 The problem with 192 kHz: the ripples in amplitude become much more exaggerated when filtering 192 kHz signals. Consequently, the frequency response is much less accurate and distortion goes up.

 This is why most all converter designers and recording engineers don't recommend 192 kHz. The chip technology has not provided the means to effectively convert 192 kHz without these problems.

 So, the trade-off for the extra analog bandwidth is an increase in aliasing and frequency-response distortion. Although some people may 'enjoy' listening to 192 kHz more then lower rates, it is not as accurate, objectively speaking.

 Thanks,
 Elias


----------



## puntloos

Quote:


  Originally Posted by *EliasGwinn* /img/forum/go_quote.gif 
_Puntloos,

 The up-sampling is an interpolation of the data, it is not actually the new sample-rate. The converter interpolates the data by 2^20, then selects every Xth sample, where X=# of interpolation samples/output-sample rate. 

 With 192 kHz, the chip can't interpolate as many samples, and therefore the resolution is compromised. 
_

 

Addition I think you would agree with:
_Assuming the same ASRC_
 i.e., if a DAC designer chooses to go for 192K, he/she might choose for a faster ASRC too, and mitigate this potential issue. Perhaps introducing new ones, yes.

  Quote:


 The problem with 192 kHz is all about the filtering. I will cut and paste a post I made before concerning this on page 30 of this thread (http://www.head-fi.org/forums/showpo...ostcount=590): 
 

I apologise for missing that post.. but carrying on:
  Quote:


 The filters in a D-A chip ideally pass all audio below the Nyquist frequency and block (filter) all audio above the Nyquist frequency. In reality, however, there will be some audio below the Nyquist that is being filtered some, and there will be some audio above the Nyquist which is not filtered enough. The latter is very dangerous because those frequencies will be aliased and cause distortion.

 Here's the problem with 192 kHz: the filter used for 192 kHz is of far less quality. This is true of ALL D/A chips on the market. What happens is this: the filter cut-off becomes less defined, causing audio below Nyquist to be attenuated. And, more importantly, AUDIO ABOVE NYQUIST IS NOT FULLY ATTENUATED!! The filter does not do its job as well at 192 kHz.

 Another 'real-life' limitation to these filters is amplitude ripple. This means that the audio below Nyquists will have ripples in amplitude across the frequency spectrum. This is equivalent to inaccurate frequency response. This is also something that happens in all D-to-A chips, some more so then others.

 The problem with 192 kHz: the ripples in amplitude become much more exaggerated when filtering 192 kHz signals. Consequently, the frequency response is much less accurate and distortion goes up.

 This is why most all converter designers and recording engineers don't recommend 192 kHz. The chip technology has not provided the means to effectively convert 192 kHz without these problems.

 So, the trade-off for the extra analog bandwidth is an increase in aliasing and frequency-response distortion. Although some people may 'enjoy' listening to 192 kHz more then lower rates, it is not as accurate, objectively speaking.

 

OK Im going to make a lot of assumptions now.. bear with me, or feel free to ignore me, you have been very accommodating already (thanks!).

 Anyway here goes.

 Assuming an 'identical quality' cutoff filter, am I correct in understanding that this '192Khz filter' in other DACs will have indeed achieved perfect cutoff at 110Khz? 

 You see, as I understand what you're saying, at 192, there is a bigger band to take care of, with the extra bandwidth being used at a less-than-perfect way.

 Or to put it another way: assume for argument's sake that there would be a perfect cutoff filter that cuts off at 110Hz, period, no ifs, buts and no any other annoying sideeffects. Additionally, assume an ASRC with higher bandwidth, (2*2^20?) i.e. no loss of resolution due to going for 192. Again: hypothetical.

 Would you agree that applying my hypothetical filter to the output of a 192Khz filtered dac will result in the same performance as a DAC that has a 110Khz cutoff filter?

 You see, maybe Ive already dropped the ball on the tech side of things (if so, just ignore my wild theories), but paying attention to ONLY the extra band, i.e. 111-192Khz, you could claim a lot of things about that band, bad frequency response, distortions.. but I wonder: at what point do these distortions negatively impact the overall signal quality compared to a signal with that band 100% gone? If that band is gone, how much distortion would that be?

 In my view *ducks* it does sound like having SOME signal there, even if it is only a bad approximation of the source, is better than having 0 signal there (which is an even worse approximation).

 Or is my ability to count so bad I should consider a job in mathematics?


----------



## Crowbar

No oversampling DAC's analog filter leaves significant signal in that range, which is as it should be. 111 kilosamples/s means 55.5 kHz is reproducible which is pointless for human ears. And you want more? 192 is useful in making the analog filter simpler, and allows things like more dithering to be used, especially if DSP is used where bits are lost then that can be beneficial.

  Quote:


 maybe Ive already dropped the ball on the tech side of things 
 

It's on the physiological side of things you've dropped the ball. Your concern is analogous to someone complaining that their non-quartz glass window is no good since by not being transparent to ultraviolet light it's limiting what they can see through it. Well, just like you can't see ultraviolet, you can't hear ultrasound.

  Quote:


 it does sound like having SOME signal there, even if it is only a bad approximation of the source, is better than having 0 signal there 
 

Absolutely false, since you're stressing more the analog stages without getting anything audible in return!


----------



## puntloos

First of all let's make one distinction here: there is quite a difference between upsampling a 44/16 source to 192/24 and having a source that is recorded in 192/24 (yes, currently they are rare, most 192K sources in fact are pre-upsampled from a 96/24 or even 44/16 - patooei. But 'soon' if not already, Im sure original 192K sources will be here)

  Quote:


  Originally Posted by *Crowbar* /img/forum/go_quote.gif 
_No oversampling DAC's analog filter leaves significant signal in that range, which is as it should be. 111 kilosamples/s means 55.5 kHz is reproducible which is pointless for human ears. And you want more? 192 is useful in making the analog filter simpler, and allows things like more dithering to be used, especially if DSP is used where bits are lost then that can be beneficial._

 

Weeee..eee..ee.ll.l...

 Obviously even dogs can't hear above (guessing) 44Khz, so 88Khz samplerate should suffice for even them. But, the reasons Ive gathered for raising frequency is: (again: for 192 native sources)

 1/ capturing and reproducing the source frequencies for sum frequencies. In theory, a 55K and a 60K sound will produce a difference frequency of 5Khz. A 110K DAC will not reproduce the 60K sound, losing the 5K freq too.

 2/ Sinewaves can't accurately be reproduced when there are too few sample points. I'm not sure what factors in a dac come into play when approximating a 'proper' sine, but at (say) 96K, a 44K signal can only be a square wave. To make it a sine, assume you need twice as many points.. 22K is then coming dangerously near.

 there are more reasons I believe but they escape me atm.. need more coffee.

  Quote:


 It's on the physiological side of things you've dropped the ball. Your concern is analogous to someone complaining that their non-quartz glass window is no good since by not being transparent to ultraviolet light it's limiting what they can see through it. Well, just like you can't see ultraviolet, you can't hear ultrasound. 
 

Perhaps quite a fair point.
  Quote:


 Absolutely false, since you're stressing more the analog stages without getting anything audible in return! 
 

I'm not quite sure if I agree with the 'stressing the analog' rationale. The quest for higher Khz/Bitdepth originates with the quest to approximate analog audio as closely as possible. What you are saying about analog stages should then also apply to 1950's LP players. Of course, wether or not the -rest- of my stereo is able to reproduce whatever I throw at it is another matter altogether, however I did indeed spec the rest of my system specifically around this purpose.

 One thing I can't help but wonder (Elias? 
	

	
	
		
		

		
			





 ) is how Benchmark arrived at their choice. One would imagine they simply TRIED all combinations their test boards could do, then pick the best few combinations of DACs, ASRCs and go from there. 

 I doubt they looked at the figures, pondered a while, and then never chose the '192' setting as a matter of principle. 

 Additionally, I don't believe that many 'high end' DAC builders would go for 192 because 'bigger is better'. Again, I would guess that most builders would get the best parts they can within their intended budget, and then see what combinations give the best measurements.


----------



## Crowbar

You're way off base here.
  Quote:


  Originally Posted by *puntloos* /img/forum/go_quote.gif 
_1/ capturing and reproducing the source frequencies for sum frequencies. In theory, a 55K and a 60K sound will produce a difference frequency of 5Khz. A 110K DAC will not reproduce the 60K sound, losing the 5K freq too._

 

There are two ways beat frequencies can be heard. It's known that the brain will form beat frequencies when you play one frequency in one ear and another in the other one. But if the frequencies are ultrasonic, that doesn't happen since they're not detected in the first place. Now, with ultrasound it's possible to create and audible beat frequency, and that's because of the way sound intermodulates in the ear. However, the same applies to bandlimited recording hardware (and all of it is). ADCs used for recording with the sort of filters they have would cause the same intermodulation to be present as actual 5K signals. During playback, that will play back simply because the intermodulation that your ear normally does would have happened at the microphone and ADC and was recorded as a real frequency.

  Quote:


 2/ Sinewaves can't accurately be reproduced when there are too few sample points. I'm not sure what factors in a dac come into play when approximating a 'proper' sine, but at (say) 96K, a 44K signal can only be a square wave. To make it a sine, assume you need twice as many points.. 22K is then coming dangerously near. 
 

Oh wow dude, *epic failure* in understanding basic sampling theory here! Shame... Nyquist limit's minimum 2x sampling frequency is *exactly* for sine waves; square waves are not bandlimited as it takes infinite sampling frequency to reproduce a true square wave, unless you change from a Fourier basis to say Haar wavelets (and not possible in the physical world since it requres infinite slew rate). You couldn't have possibly gotten this more backwards. Ouch.

  Quote:


 I'm not quite sure if I agree with the 'stressing the analog' rationale. The quest for higher Khz/Bitdepth originates with the quest to approximate analog audio as closely as possible. 
 

Wrong again. Digital and analog are just different ways to represent information. The 20 kHz bandlimit of human hearing directly translates to 40 kilosamples/s to encode the same information digitally, which needs to be raised for practical limits (space for the filter). Likewise for dynamic range and bit depth (20 bits is sufficient in optimal system = anechoic chamber). More generally, the discreteness limiting precision of digital systems doesn't make them inferior, since the physical world is also of limited precision (proof by contradiction: infinite precision analog implies infinite information density, which violates the Bekenstein bound).

  Quote:


 What you are saying about analog stages should then also apply to 1950's LP players. 
 

Incredibly limited dynamic range and distorted frequency response requiring RIAA equalization. Oh please.


----------



## EliasGwinn

Quote:


  Originally Posted by *puntloos* /img/forum/go_quote.gif 
_Addition I think you would agree with:
Assuming the same ASRC
 i.e., if a DAC designer chooses to go for 192K, he/she might choose for a faster ASRC too, and mitigate this potential issue. Perhaps introducing new ones, yes._

 

I would be very happy if Analog Devices, Burr Brown, et al, were able to come up with a D-to-A chip that was able to maintain performance at 192 kHz. If and when they do, be sure that we will have a proto-type on our bench and we will be eagerly testing (and possibly drooling
	

	
	
		
		

		
		
	


	




 ). But, as it stands, there is no such chip.

  Quote:


  Originally Posted by *puntloos* /img/forum/go_quote.gif 
_Would you agree that applying my hypothetical filter to the output of a 192Khz filtered dac will result in the same performance as a DAC that has a 110Khz cutoff filter?

 You see, maybe Ive already dropped the ball on the tech side of things (if so, just ignore my wild theories), but paying attention to ONLY the extra band, i.e. 111-192Khz, you could claim a lot of things about that band, bad frequency response, distortions.. but I wonder: at what point do these distortions negatively impact the overall signal quality compared to a signal with that band 100% gone? If that band is gone, how much distortion would that be?

 In my view *ducks* it does sound like having SOME signal there, even if it is only a bad approximation of the source, is better than having 0 signal there (which is an even worse approximation)._

 

I think the basic concept is being mis-understood. Here's the heart of the problem:

 With 192 kHz sample-rates, there is more distortion across the ENTIRE bandwidth. So, not only will the signals in the added extra bandwidth be distorted, but also everything within the very audible bandwidth 20 Hz-20 kHz. 

 If I'm not mistaken, you're assumption is the extra bandwidth is better present and distorted then not present at all. The problem with this theory is that ALL the bandwidth will be distorted. So, adding the extra analog bandwidth 48 kHz-96 kHz (which is the bandwidth added when you sample at 192 kHz instead of 96 kHz) comes at a cost of distorting the ENTIRE bandwidth. For example, 500 Hz may be 4 dB higher then it should be, and 1 kHz may be 3 dB lower then it should be...all because of the ripple effect in the cut-off filter.

 In addition, the cut-off filter at 192 kHz will not reliably filter out-of-band signals. Consequently, the audio information over 96 kHz will be aliased into the audible region causing even more distortion on top of all the other distortion.

 So, it is our recommendation (as well as the recommendation of many professional engineers) that a less distorted analog bandwidth of 1 Hz - 48 kHz is much more advantageous then a more distorted analog bandwidth of 1 Hz - 96 kHz.

 Thanks,
 Elias


----------



## EliasGwinn

EDIT:...deleted because it was a repeat of the previous post...

 Thanks,
 Elias


----------



## EliasGwinn

EDIT: ...deleted because it was a repeat of the previous post...

 Thanks,
 Elias


----------



## Crowbar

Quote:


  Originally Posted by *EliasGwinn* /img/forum/go_quote.gif 
_I would be very happy..._

 

 Quote:


  Originally Posted by *EliasGwinn* /img/forum/go_quote.gif 
_I would be very happy..._

 

 Quote:


  Originally Posted by *EliasGwinn* /img/forum/go_quote.gif 
_I would be very happy..._

 

Huh? I'm seeing three identical posts here, #688-690.


----------



## EliasGwinn

yeah, i don't know why it repeated my post like that...???

 yeah, i don't know why it repeated my post like that...???

 yeah, i don't know why it repeated my post like that...???

 ok, just messin with ya this time... 
	

	
	
		
		

		
			





 Thanks,
 Elias


----------



## EliasGwinn

Generally speaking, there are three arguments for higher sample-rates. I will state them here for the sake of this discussion, but I am not implying or denying their validity. Here they are:

 1. The cutoff filter will affect the 20-20k range less as it is moved up the frequency spectrum. The higher the sample rate, the higher you can move your filter, the less you attenuate 20k.

 2. DSP (EQ, gain calculations, reverbs, etc) all benefit from more samples to work with

 3. The ultrasonic frequencies, although not detectable by the human ear when heard by themselves, affect how we perceive the audio to which it is related when it is included.

 The third point is a heavily debated point, and I cannot claim any authority in the matter, so I will not state an opinion. 

 The first point is true, but follows the law of diminishing returns. That is, when you move the filter just past 20kHz, you get better high freq linearity as you move up in freq. But, after you move it far enough away, you don't gain much more linearity. For example, if the sample-rate is 96k, which puts the filter at 48k, you are more then an octave above 20k. Depending on the filter design, this should be more then enough bandwidth to accurately reproduce 20k. 

 However, if you keep moving your cutoff frequency up, the chip's filter performance becomes compromised and you start to introduce non-linearity, inaccurate frequency response, and inefficient stop-band performance, which will introduce aliasing. The point where you start compromising the filter's performance is above 115 kHz...which is why we chose 110 kHz as our re-sample freq. It keeps us in a safe area of the filter, yet is more then competent for 96k (actually, more competent then a 96k conversion because the filter cutoff freq is moved up and away from the recorded Nyquist by 7kHz (48->55kHz), which means that it will not attenuate 47kHz as much).

 By the way, none of this is dependent on how much money you spend on a chip. The chips used in the DAC1 are nearly the same price as those used in the most expensive converters in the world. The chips are not very expensive. You can't buy a chip which performs better at 192kHz then at 96 kHz...they don't exist. The parts that are expensive, and therefore limited by cost, are the faceplate, chassis, knobs, multi-channel (8) designs, etc. 

 Thanks,
 Elias


----------



## Crowbar

The filter issue is taken care of by oversampling. I think the real reason for 192's existence is "more is better" type marketing.


----------



## Telynau

I went ahead and bought a DAC1 USB and have been running it for about eight hours now as a USB DAC (not a headphone amp). I need quite a bit of time with a component before I can assess its true nature, so all I can do at this point is give my first impressions.

 In short, this is an unusually good component, regardless of price. Going a bit out on a limb, I would say world class. It seems to be neutral and uncolored; the underlying recording generates the sound, not the DAC. It is very detailed and clean, but not "forward" and absolutely not bright, harsh or any of the bad things I have come to associate with most digital playback. It is not "laid back" in the sense of a recessed sound. 

 A lot of what I am hearing seems to be the result of a very, very low noise floor -- there is a lot of silence (some people call it blackness) in the background of the playback that "sounds" like the absence of noise of one type or another. It means that you can resolve micro detail at low volumes and can be happy listening at lower volumes (though high volumes sound good too). 

 If you haven't heard the effect of a very low noise floor it is a little hard to describe. It comes across as greater clarity and purity of sound. Depending on the recording, it makes us think "that sounds good" or "that sounds real." 

 I have the DAC1 USB running on the following setup.

 Apple Pismo (500Mhz, 512MB memory) running Tiger, which I just bought used on eBay. I have used all of the settings recommended in this thread to play music back on Apple. I am using this Pismo to test how modest of a PC I can use to generate great sound. As of now, the Pismo is passing the test. Keep in mind that all it does is play music using iTunes.

 All music has been burned using a high quality Plextor drive, with iTunes error correction on and using Apple Lossless Compression.

 LaCie D2 Extreme 250GB triple interface drive, which I am running into the Pismo using the Firewire 400 port.

 I am running the DAC1 USB directly into a McIntosh 2102 tube amp (100 watts per channel via a total of 8 KT-88's) which is powering some fine vintage Bozak Concert Grand speakers (actually, the pair that were reviewed for Stereophile).

 As a listener I am most sensitive to tone and I really hate digital artifacts. I especially hate the distortions that suck the life out of music, leaving behind, zombie like, only the cold, lifeless structure and form. 

 Anyway, the DAC1 USB doesn't do any of the many bad things I hate and appears to do unusually good things such as 1) neutrally present the music as it was recorded (subject to the limitations of the source recording on CD -- most of which, unfortunately, still aren't as good as they should be) and 2) present the music with a low, low noise floor that will encourage music nuts to listen to their favorite recordings late into the night and early morning. 

 So congratulations to Benchmark. While I will know much more in the coming weeks, I think Benchmark deserves credit for introducing a really noteworthy component -- one that I suspect a lot of people are going to want as the heart of their PC-based playback systems.

 And thanks to Elias, whose many explanatory posts (and unfailing patience and politeness) persuaded me to give the DAC1 USB a try for my second system. For the record, I have no association with Benchmark or any other part of the audio or music business. I bought the DAC1 USB off the Benchmark web site at its list price of $1275 plus shipping. I own and have spent many happy hours listening to another unusually good USB DAC, the Wavelength Audio Cosecant V2.0. 

 Regards, James


----------



## puntloos

Quote:


  Originally Posted by *Crowbar* /img/forum/go_quote.gif 
_You're way off base here.

 There are two ways beat frequencies can be heard. It's known that the brain will form beat frequencies when you play one frequency in one ear and another in the other one. But if the frequencies are ultrasonic, that doesn't happen since they're not detected in the first place. Now, with ultrasound it's possible to create and audible beat frequency, and that's because of the way sound intermodulates in the ear. However, the same applies to bandlimited recording hardware (and all of it is). ADCs used for recording with the sort of filters they have would cause the same intermodulation to be present as actual 5K signals. During playback, that will play back simply because the intermodulation that your ear normally does would have happened at the microphone and ADC and was recorded as a real frequency._

 

There has been a lot of discussion about this, and as far as I know the conclusion simply hasn't been reached yet. If a certain system (or room!) has an imperfect transfer function, then these difference frequencies occur, and indeed a 50K and a 60K 'source' sound will result in a 10K target sound. The question is wether or not recording gear should and will record this resultant frequency, or should (also) store the two source frequencies that created it.

  Quote:


 Oh wow dude, *epic failure* in understanding basic sampling theory here! Shame... Nyquist limit's minimum 2x sampling frequency is *exactly* for sine waves; square waves are not bandlimited as it takes infinite sampling frequency to reproduce a true square wave, unless you change from a Fourier basis to say Haar wavelets (and not possible in the physical world since it requres infinite slew rate). You couldn't have possibly gotten this more backwards. Ouch. 
 

Yeah I got it twisted around, my bad. The argument still holds though, as you graciously confirm. 
	

	
	
		
		

		
		
	


	




 - If you need 'unlimited' sampling frequency to reproduce a true square wave, then the higher samplefrequency reproduces a square wave better. 

  Quote:


 Wrong again. Digital and analog are just different ways to represent information. The 20 kHz bandlimit of human hearing directly translates to 40 kilosamples/s to encode the same information digitally, which needs to be raised for practical limits (space for the filter). Likewise for dynamic range and bit depth (20 bits is sufficient in optimal system = anechoic chamber). More generally, the discreteness limiting precision of digital systems doesn't make them inferior, since the physical world is also of limited precision (proof by contradiction: infinite precision analog implies infinite information density, which violates the Bekenstein bound). 
 

That is simply one way of looking at things. I hold that the 'perfect' digital audio would be at infinite bits, and infinite khz. Naturally there are all sorts of more or less subtle differences, but this point I was making wasn't intended to be the end-all of all arguments.

  Quote:


 Incredibly limited dynamic range and distorted frequency response requiring RIAA equalization. Oh please. 
 

Well only cause you asked so nicely.

  Quote:


  Originally Posted by *Crowbar* /img/forum/go_quote.gif 
_The filter issue is taken care of by oversampling. I think the real reason for 192's existence is "more is better" type marketing._

 

For $100 DACs, Im sure that might be the case. But specifically with the target audience (that is definately not impressed with things like 89578923475234 PMPO WATT POWER!@#) I dont think that hardware designers in the (semi) audiophile segment will be -that- influenced by the pure '192 is larger than 96 so it must be better!' argument.

 My personal reasoning for considering 192K:
 - 192 is the limit of DVD-A, and seems to at least be a number quite often used to indicate the top end of the spectrum. Leaving aside for a moment wether or not DACs can do it -now-, I do not think that in the coming 10 years (and perhaps even our lifetimes!) there will be a need to up bitdepth and samplerate, simply cause the sources don't even hypothetically support it yet. (even BluRay and HD-DVD 'only' specify a max of 192/24
 - High end recording outfits commonly record single sources ('microphones') at 96, after which one would sensibly upconvert to 192 inside the computer to start doing the 'DSP work' i.e. mixing/mastering, and this 192-mastered stereo track has only relevant data. It's not just upconverting-for-upconverting's sake.
 - The '192' manufacturers I am looking at do (of course!) claim that their gear does work satisfactorily at 192. While they need to live off their work and might even lie about their choices after mass production started, the company I chose have been working with custom digital gear since 1999, so at least they probably won't make beginners mistakes. It would be interesting to invite someone from that company in here to debate the issue with Benchmark, but I suppose this topic is 'Benchmarks' and competition would not be welcome (at least not in this specific topic)


----------



## Crowbar

Quote:


  Originally Posted by *puntloos* /img/forum/go_quote.gif 
_The question is wether or not recording gear should and will record this resultant frequency, or should (also) store the two source frequencies that created it._

 

That's been already answered.

  Quote:


 If you need 'unlimited' sampling frequency to reproduce a true square wave, then the higher samplefrequency reproduces a square wave better. 
 

You don't seem to get it. True square waves are not physically possible. They don't exist in this physical universe.

  Quote:


 I hold that the 'perfect' digital audio would be at infinite bits, and infinite khz. 
 

You cannot hold things if you cannot back them up. The physical universe doesn't have infinite resolution. Infinite kHz cannot exist since it's infinite energy. Any actual signal in the physical universe itself has a limiting resolution, and thus it only takes a finite number of bits to match it--any more bits cannot match it better, since there is no higher resolution to match it to. It is exactly because spacetime is not infinitely differentiable that infinite digital precision is useless--after a point, adding more bits doesn't do anything since the analog signal itself has lower resolution. I'm not even talking about the limit of hearing's resolving power, which is orders of magnitude from that.

  Quote:


 Naturally there are all sorts of more or less subtle differences, 
 

This is not about subtle differences; you are making a claim that goes against laws of physics such as the Bekenstein bound.

  Quote:


 but this point I was making wasn't intended to be the end-all of all arguments. 
 

No, but it should at least resemble reality in a small way, whereas you are headed into dreamland.


----------



## puntloos

Quote:


  Originally Posted by *Crowbar* /img/forum/go_quote.gif 
_-snip-_

 

You're missing the point, but I hereby give up trying. Please look up inductive reasoning and consider that when someone mentions that apples fall to the ground, you cannot just shoot that down because you have accellerated a tree to near light speed and the apples fell up.

 In any case, again I thank Elias for his patience and clear explainations, I also apologise for getting some things wrong and getting some things right here and there too, but I think I've learned a thing or two, which was the whole point, wasn't it?

 I'll be back if/when I have more questions or when I have some results of my ABX testing with the benchmark.


----------



## dmk005

This thread reads like DAC Primer! I have been considering moving on a DAC purchase and I cannot thank you all enough. I am glad you are all willing to share/debate the salient points with such passion/thoroughness/clarity. I was considering the Bel Canto DAC3 but now will purchase the Benchmark DAC instead. I plan on using it in between my MAC mini and an RSA Apache which is on the way. I had considering going for a high quality Meridian CD player but hate the workload it would require.

 Have I truly entered geek-dom having enjoyed this thread so much?


----------



## EliasGwinn

Quote:


  Originally Posted by *puntloos* /img/forum/go_quote.gif 
_The '192' manufacturers I am looking at do (of course!) claim that their gear does work satisfactorily at 192. While they need to live off their work and might even lie about their choices after mass production started, the company I chose have been working with custom digital gear since 1999, so at least they probably won't make beginners mistakes._

 

I think I may have been unclear with some of my earlier statements. The statements I made earlier do not and should not imply that the DAC1's performance is compromised at 192 kHz. The DAC1 performs equally at 192 kHz as with any other sample-rate. This is because we re-sample to 110 kHz. 

 The point I was trying to make was that the D-to-A chips used by all DAC manufactures do not perform well at 192 kHz (YET!). We prevent that situation by re-sampling to 110 kHz, which is the sample-rate which the D-to-A chip will maximize performance.

 Also, on a more general note, the A-to-D chips used by manufacturers also do not perform will at 192 kHz (again, YET!). This is another part of the debate...possibly more important then the D-to-A side. 

 Whenever IC manufacturers solve this limitation, Benchmark and every other digital converter manufacturer on the market will begin using those chips. But that time has not come yet, so we will continue recommending recording at 96 kHz for now.

 Also, to address your final point, Puntloos....

  Quote:


  Originally Posted by *puntloos* /img/forum/go_quote.gif 
_It would be interesting to invite someone from that company in here to debate the issue with Benchmark, but I suppose this topic is 'Benchmarks' and competition would not be welcome (at least not in this specific topic)_

 

I would be very much in favor of any sort of constructive discussion with any designer, manufacturer, user, etc. 

 This thread is not 'Benchmark's' thread, it is open to anyone and everyone. We may attempt to maintain specific discussion topics, but it is not limited to Benchmark products by any means.

 Please pass an invitation to this thread to the manufacturer you are referring to.

 Thanks,
 Elias

 PS. At the risk of getting into semantics, I'd like to point out that Benchmark began shipping our first D-to-A's and A-to-D's in early 1996.


----------



## EliasGwinn

Quote:


  Originally Posted by *dmk005* /img/forum/go_quote.gif 
_Have I truly entered geek-dom having enjoyed this thread so much? 
	

	
	
		
		

		
		
	


	


_

 

Welcome to geek-dom, my friend!! I accepted that lot when I graduated with a degree in EE (because, as we all know, you can't spell 'GEEK' without 'EE'!!).

 Also, I'm glad to hear that you have been enjoying this thread and have decided to acquire a DAC1. I'll anxiously be awaiting your impressions!!

 Thanks,
 Elias


----------



## Headphony

I've been greatly enjoying my DAC1 USB. Superb product! My question is: what is the best way to connect the DAC1 to 2 amps with only RCA inputs? (for headphones and speakers ampage) I've tried using Cardas XLR to RCA adapters but the sound quality is compromised compared to the direct RCA out. Would some kind of RCA y-splitter be a better option? Or maybe I'm best off just using the one set of RCA outputs and switching the cable from amp to amp.
 Many thanks, Elias, for your excellent contributions to this thread thus far!


----------



## EliasGwinn

The RCA splitter will work fine *IF* the amps have appropriately high input impedance. If you can find out the input impedance of the RCA inputs of your amps, let me know what they are and I can confirm whether it will be ok to do this.

 Thanks,
 Elias


----------



## Dan the man

Hi Elias: I am confused about the differences between the DAC 1/USB implimentation of I2S and the USB solution offered by Steve Nugent at Empiracle audio with the total cost for the Off-Ramp I²S/minimum mods Benchmark DAC-1 combination package at $2,645 which Steve Nugent feels is his "most affordable computer solution" that he believes makes world-class sound. But at twice the cost of the DAC1/USB.

 He claims that his version is better than the DAC1/USB because "the I2S Benchmark does not have the upsampler chip in it. This allows you to control the sound, from bit-perfect 16/44.1 to 24/96 upsampled on the computer. This is the main difference." What does this mean in laymans terms.

 This combination and description was reviewed http://www.6moons.com/audioreviews/e...l/offramp.html


 He offers other mods as you must know but how they impact SQ is hard to imagine...see below from his website.

 $44452) Off-Ramp I2S driving Benchmark DAC-1 with I2S, Turbomod, OPA-627 op-amp upgrade - Very detailed and dynamic
 $37453) Off-Ramp I2S driving Benchmark DAC-1 with I2S, Turbomod, dual op-amp upgrade
 $45454) Off-Ramp Turbo 2 S/PDIF driving Benchmark DAC-1 with Turbomod, SuperTurboclock, OPA627 op-amp upgrade, BNC/Trans. upgrade
 $38455) Off-Ramp Turbo 2 S/PDIF driving Benchmark DAC-1 with Turbomod, SuperTurboclock, dual op-amp upgrade, BNC/Trans. upgrade
 $26456) Off-Ramp I2S driving Benchmark DAC-1 with I2S, minimum mods $$$


----------



## Dan the man

Hi Elias: I am confused about the differences between the DAC 1/USB implimentation of I2S and the USB solution offered by Steve Nugent at Empiracle audio with the total cost for the Off-Ramp I²S/minimum mods Benchmark DAC-1 combination package at $2,645 which Steve Nugent feels is his "most affordable computer solution" that he believes makes world-class sound. But at twice the cost of the DAC1/USB.

 He claims that his version is better than the DAC1/USB because "the I2S Benchmark does not have the upsampler chip in it. This allows you to control the sound, from bit-perfect 16/44.1 to 24/96 upsampled on the computer. This is the main difference." What does this mean in laymans terms.

 This combination and description was reviewed http://www.6moons.com/audioreviews/e...l/offramp.html


 He offers other mods as you must know but how they impact SQ is hard to imagine...see below from his website.

 $44452) Off-Ramp I2S driving Benchmark DAC-1 with I2S, Turbomod, OPA-627 op-amp upgrade - Very detailed and dynamic
 $37453) Off-Ramp I2S driving Benchmark DAC-1 with I2S, Turbomod, dual op-amp upgrade
 $45454) Off-Ramp Turbo 2 S/PDIF driving Benchmark DAC-1 with Turbomod, SuperTurboclock, OPA627 op-amp upgrade, BNC/Trans. upgrade
 $38455) Off-Ramp Turbo 2 S/PDIF driving Benchmark DAC-1 with Turbomod, SuperTurboclock, dual op-amp upgrade, BNC/Trans. upgrade
 $26456) Off-Ramp I2S driving Benchmark DAC-1 with I2S, minimum mods $$$


----------



## puntloos

Quote:


  Originally Posted by *Dan the man* /img/forum/go_quote.gif 
_Hi Elias: I am confused about the differences between the DAC 1/USB implimentation of I2S and the USB solution offered by Steve Nugent at Empiracle audio with the total cost for the Off-Ramp I²S/minimum mods Benchmark DAC-1 combination package at $2,645 which Steve Nugent feels is his "most affordable computer solution" that he believes makes world-class sound. But at twice the cost of the DAC1/USB.

 He claims that his version is better than the DAC1/USB because "the I2S Benchmark does not have the upsampler chip in it. This allows you to control the sound, from bit-perfect 16/44.1 to 24/96 upsampled on the computer. This is the main difference." What does this mean in laymans terms.

 This combination and description was reviewed http://www.6moons.com/audioreviews/e...l/offramp.html


 He offers other mods as you must know but how they impact SQ is hard to imagine...see below from his website.

 $44452) Off-Ramp I2S driving Benchmark DAC-1 with I2S, Turbomod, OPA-627 op-amp upgrade - Very detailed and dynamic
 $37453) Off-Ramp I2S driving Benchmark DAC-1 with I2S, Turbomod, dual op-amp upgrade
 $45454) Off-Ramp Turbo 2 S/PDIF driving Benchmark DAC-1 with Turbomod, SuperTurboclock, OPA627 op-amp upgrade, BNC/Trans. upgrade
 $38455) Off-Ramp Turbo 2 S/PDIF driving Benchmark DAC-1 with Turbomod, SuperTurboclock, dual op-amp upgrade, BNC/Trans. upgrade
 $26456) Off-Ramp I2S driving Benchmark DAC-1 with I2S, minimum mods $$$_

 

So confused you've posted it twice, huh? 
	

	
	
		
		

		
			





 Anyway I know this a huge topic with what, 30+ pages now, but Im pretty sure that around page 15, this discussion was had, specifically between elias and audioengr if I recall correctly, or at least a guy offering comparable mods.

 Summarizing what I remember from this topic is that Elias has not checked/tested these mods himself but he could not imagine such mods improving the DAC1, and even thought it would degrade performance. But, just go back to around page 15 and look for discussions between Elias and audioengr.


----------



## slwiser

Quote:


  Originally Posted by *puntloos* /img/forum/go_quote.gif 
_ But, just go back to around page 15 and look for discussions between Elias and audioengr._

 

It is interesting to have two "members of the trade" duking it out in the same thread concerning a commercial product which one of them produces. Especially about how relatively "bad" the commercial product sounds out of the box and how much better it can be made by the other for only a few thousand dollars which is only several times what initial commercial product costs.


----------



## puntloos

Quote:


  Originally Posted by *slwiser* /img/forum/go_quote.gif 
_It is interesting to have two "members of the trade" duking it out in the same thread concerning a commercial product which one of them produces. Especially about how relatively "bad" the commercial product sounds out of the box and how much better it can be made by the other for only a few thousand dollars which is only several times what initial commercial product costs._

 

Well, I did have an interesting discussion about some design choices with a DAC designer recently. One of the things that 'irked' me was that they made a lot of choices for me (kinda like OS-X! *ducks*) leaving me to have to go with those. 

 Their DAC has no controls to 'turn off resampling' to 'switch resampling frequency' to 'change filters' etc, even though all those functions are easily accessible by just sending some simple control signals here and there.

 The engineer's argument was that they were confident that their choices were the best for the design, and that lots of knobs, dials and whatnot will in general simply degrade the quality from 'optimal'.

 Still, I would've liked to have found out myself. One specific one is that their DAC indeed runs at 192K resampling, and Benchmark claims 110Khz is 'better'. Im not 100% sure but I would not be surprised that switching resampling frequency would be mostly trivial to do. 

 A mod that would allow me to do so would be welcome.

 On the other hand, I think I'm with Elias when it comes to switching parts inside the DAC because the switched-in parts supposedly have 'better specs'. Even when the part in question is purely an upgrade made with more modern quality components, I would expect the rest of the device suddenly to fall 'out of tune' with this config, even if the core part would be better 'in principle'.


----------



## dmk005

Quote:


  Originally Posted by *puntloos* /img/forum/go_quote.gif 
_On the other hand, I think I'm with Elias when it comes to switching parts inside the DAC because the switched-in parts supposedly have 'better specs'. Even when the part in question is purely an upgrade made with more modern quality components, I would expect the rest of the device suddenly to fall 'out of tune' with this config, even if the core part would be better 'in principle'._

 

iMod owners/lovers would clearly disagree with this. I am not an advocate of custom modifications but I like the market influence these efforts bring. FWIW.


----------



## EliasGwinn

(boring but necessary) DISCLAIMER: Benchmark recommends that no parts are added, removed, and/or substituted within our products. Any modifications will void all warranties.

 Dan the man: regarding your specific question about Empirical Audio's mods, I don't know what the mods consist of, nor have a I heard or tested them, so I can't comment specifically regarding them.

 To answer the looming question, we are not upset by mods performed by Empirical Audio and others. We have no desire to 'duke it out' with anyone. However, I will describe some anecdotes about modded DAC1's. The following examples are not specific to, nor implied towards, Empirical Audio or any other specific modifier. 

 We have had many customers return modded DAC1's for repair because the product failed or simply did not sound right. These repairs were not covered by the warranty, of course, because they were modded. After we repair the DAC1 back to its original design, many customers hear it again for the first time and wish they had not modded the DAC1 in the first place.

 We also take the opportunity to measure the performance of the modded DAC1's. Our experience has been that, at best, the mod's did absolutely nothing to the performance (because the 'mod' was something as pointless as adding 'audiophile-grade' capacitors to the power supply). At worst, however, we have seen DAC1's whose performance was significantly reduced because of mods. Often, the inter-channel phase is altered because it seems to (artificially) spread the sound-stage wider. Then, when it is A/B'd against the original DAC1, people say "Wow, your mod seems to make the sound-stage just spread out!!" This is one type of inaccuracy we carefully avoided during design, but typically show up in mods.

 NO parts in the DAC1 were chosen or limited based on cost. The parts in the DAC1 are chosen very specifically for optimized performance. The circuit within the DAC1 is very dependent on the components which are part of the design. If these components are removed or substituted, or if foreign components are added, the circuit is no longer operating in the manner in which it was designed. And, since none of the modders have a schematic for the DAC1, they are making un-informed decisions about what role the components are playing. Thats why we see mods with $50 'audiophile' caps in the power supply...where they don't touch the audio (but the modder will still charge $1500!!).

 If, hypothetically speaking, the mod's were actually improving the performance of the DAC1, we would be happy to recommend them to our customers so that they could enjoy the DAC1 even more. But, as I mentioned before, every mod we have heard and tested demonstrate NO improvement in performance whatsoever, and, more often, reduced the quality of the sound.

 Thanks,
 Elias


----------



## gjwaudio

Hi Elias

 Let me begin with a huge THANK YOU for your continued _heroic_ participation in this forum. Your willingness to reveal and explain so many facets of the DAC1 - with patience and professionalism - is singular in my experience, and raises Benchmark up from the crowd.

 As an owner of both the ADC1 and DAC1 (Classic, with 192 upgrade), I can attest from personal experience that the "Benchmark Sound" is neutral and honest to its input. I was stunned at the fidelity of the ADC1 when used to transfer LPs at 24/96... but that is a topic for another thread. My point is, I believe your products tell The Truth, without embellishment. Well done... and as it should be in professional use.

 Now, with the greatest respect for you, and everyone involved in creating the DAC1, I must say my experience is at odds with this statement:
  Quote:


  Originally Posted by *EliasGwinn* /img/forum/go_quote.gif 
_...NO parts in the DAC1 were chosen or limited based on cost. The parts in the DAC1 are chosen very specifically for optimized performance...

 If, hypothetically speaking, the mod's were actually improving the performance of the DAC1, we would be happy to recommend them to our customers so that they could enjoy the DAC1 even more. But, as I mentioned before, every mod we have heard and tested demonstrate NO improvement in performance whatsoever, and, more often, reduced the quality of the sound..._

 

The issue is with the unbalanced outputs - specifically the nasty RCA sockets Benchmark has chosen to implement. You do great disservice to the unbalanced user (sorry about the awful term...) by limiting their sound with this "weak link". For a modest increase in parts cost, you could give these people access to all the sound quality they've paid for.

 One need only inspect the connection side of this assembly, to see how appallingly small is the contact area for the hot pin. Improving this interface yields sonic improvement - no question. It doesn't even require extreme jewellery like WBTs and the like... I have replaced my stock RCAs with three dollar (Teflon dielectric) chassis-mounts, and found the improvement immediately perceivable - and well worth the effort. Snug, but satisfying.






 My question: How has this aspect escaped the very thorough and thoughtful design/ development process ?

 I raised this point with Rory a year ago - who passed it on to Engineering. No response... apparently it fell on deaf ears (...ouch !). Can you offer your take - and possibly experience - on this issue ?

 Thanks again Elias, for making the time to keep on top of the discussion we have going on here.
 Cheers,
 Grant


----------



## slwiser

Quote:


  Originally Posted by *gjwaudio* /img/forum/go_quote.gif 
_






 My question: How has this aspect escaped the very thorough and thoughtful design/ development process ?
 Grant_

 

From my initial inspection of your modification it would appear that in order to install you new RCA inputs the lower one has to be bend down. Now given no other changes to the Benchmark cause additional changes will involve more cost. Would you have purchased a Benchmark with that bend RCA plug in the first place? What would this have suggested to you concerning their quality control? Or is this a illusion to me.


----------



## audioengr

Quote:


  Originally Posted by *EliasGwinn* /img/forum/go_quote.gif 
_(boring but necessary) We have had many customers return modded DAC1's for repair because the product failed or simply did not sound right. These repairs were not covered by the warranty, of course, because they were modded. After we repair the DAC1 back to its original design, many customers hear it again for the first time and wish they had not modded the DAC1 in the first place.Thanks,
 Elias_

 

To my knowledge, none of Empirical Audio modded DAC's has ever been returned to Benchmark for repair. We repair all DAC's that fail in the field ourselves. We offer a 1 year warranty on the DAC-1 with our mods. We have never had even one returned for warranty repairs. They are very reliable.

 There are other modders that mod the DAC-1 and I am not familiar with their rework quality.

 Again, I will lend a modded DAC-1 to a third party with a resolving system that I can verify sound quality to do a head-to-head comparison of the stock and modified DAC-1 or DAC-1 USB, providing that they post the results on three forums: audioasylum, audiogon and head-fi. Alternatively if someone wants to use my reference system for this purpose, this is possible as well. You provide transportation and I will provide housing for one night.

 Steve N.
 Empirical Audio


----------



## audioengr

Quote:


  Originally Posted by *gjwaudio* /img/forum/go_quote.gif 
_Hi Elias

 Let me begin with a huge THANK YOU for your continued heroic participation in this forum. Your willingness to reveal and explain so many facets of the DAC1 - with patience and professionalism - is singular in my experience, and raises Benchmark up from the crowd.

 As an owner of both the ADC1 and DAC1 (Classic, with 192 upgrade), I can attest from personal experience that the "Benchmark Sound" is neutral and honest to its input. I was stunned at the fidelity of the ADC1 when used to transfer LPs at 24/96... but that is a topic for another thread. My point is, I believe your products tell The Truth, without embellishment. Well done... and as it should be in professional use.

 Now, with the greatest respect for you, and everyone involved in creating the DAC1, I must say my experience is at odds with this statement:


 The issue is with the unbalanced outputs - specifically the nasty RCA sockets Benchmark has chosen to implement. You do great disservice to the unbalanced user (sorry about the awful term...) by limiting their sound with this "weak link". For a modest increase in parts cost, you could give these people access to all the sound quality they've paid for.

 One need only inspect the connection side of this assembly, to see how appallingly small is the contact area for the hot pin. Improving this interface yields sonic improvement - no question. It doesn't even require extreme jewellery like WBTs and the like... I have replaced my stock RCAs with three dollar (Teflon dielectric) chassis-mounts, and found the improvement immediately perceivable - and well worth the effort. Snug, but satisfying.






 My question: How has this aspect escaped the very thorough and thoughtful design/ development process ?

 I raised this point with Rory a year ago - who passed it on to Engineering. No response... apparently it fell on deaf ears (...ouch !). Can you offer your take - and possibly experience - on this issue ?

 Thanks again Elias, for making the time to keep on top of the discussion we have going on here.
 Cheers,
 Grant_

 

I guess you dont understand volume manufacturing process. These connectors would have to be hand-wired and hand-soldered. I believe except for the transformer/AC inlet and the S/PDIF jack, everything in the DAC-1 is machine assembled. If it were possible to machine-assemble these and maintain quality, they would probably be machine assembled too. Once you start adding manual interventions like this, the price goes up dramatically.

 Steve N.


----------



## gjwaudio

@slwiser

 You are correct in thinking the "bent jack" is a distortion of the photo. These particular sockets fit in the existing holes, and the lower one just clears the PCB. Both pieces are mounted at 90 degrees to the back plate. See in the photo where the insulating tape safeguards the solder points on the PCB.






 @Steve
  Quote:


  Originally Posted by *audioengr* /img/forum/go_quote.gif 
_I guess you dont understand volume manufacturing process. These connectors would have to be hand-wired and hand-soldered. I believe except for the transformer/AC inlet and the S/PDIF jack, everything in the DAC-1 is machine assembled. If it were possible to machine-assemble these and maintain quality, they would probably be machine assembled too. Once you start adding manual interventions like this, the price goes up dramatically._

 

I appreciate your point - yes, there is unavoidable handiwork required to install higher quality RCA connectors - and that takes time, and time is money etc. etc. This may have been a deliberate decision by Benchmark.

 We can see however, that they saw fit to use a chassis-mounted BNC connector, requiring manual assembly & soldering. Somebody thought it was important enough to use this connector (I'm no expert in volume manufacturing - maybe a PCB-mount version does not exist). What is the incremental cost of adding chassis-mounted RCAs, given that a human must be involved on behalf of the BNC ?

 What I _do_ know is the excellent sound within the stock DAC1 is more clearly perceived when you improve the RCA connectors. (All you balanced-mode guys can stop laughing now...).

 My original question challenges the assertion by Elias that NOTHING - regardless of parts cost - can be changed to improve the DAC1, without a more elaborate topology (pardon my paraphrase Elias). This is what I conclude from reading post #612 & #710.

 That position is different from stating something like "We made it as good as possible, within a price point" - where a justification could be made for sockets so... "economical" (again... my phrasing). However, this is a very high-quality/ high-value boutique product - and make no mistake - I LOVE the sound my Benchmarks give me... and that sound is throttled (in my personal opinion) by the choice of RCA sockets. 

 Given that Benchmark is courting the home user with the DAC1 (...you don't see many RCAs on serious pro gear), I wonder why they missed on a feature that so many audiophiles value highly ?

 I'm curious to read Elias' response.
 Cheers,
 Grant


----------



## Kizza

Does seem a little overpriced


----------



## EliasGwinn

Grant,

 You make a good point. I guess I should specify that no circuit components were limited by cost. 

 The DAC1 was originally designed for professional applications where balanced interconnection is prominent. The heavy-duty BNC connector is chosen specifically for its durability, as coaxial digital transmission is very prominent in the pro-industry. We did not use heavy-duty RCA connectors because they are not often used in the pro-industry, and thus would have added a (seemingly unneeded) increase in the assembly labor of the DAC1. For unbalanced users, a more durable RCA connector would understandably be more desirable. 

 We appreciate your feedback about the connectors, and we respect your opinion about the results attained with your mod. However, it is our belief that the RCA connectors used in the DAC1 do not limit or degrade the audio quality of the RCA outputs. 

 We take these criticisms very seriously, and appreciate the opportunity to learn what our customers want. 

 Thanks,
 Elias


----------



## audioengr

Quote:


  Originally Posted by *gjwaudio* /img/forum/go_quote.gif 
_@slwiser

 You are correct in thinking the "bent jack" is a distortion of the photo. These particular sockets fit in the existing holes, and the lower one just clears the PCB. Both pieces are mounted at 90 degrees to the back plate. See in the photo where the insulating tape safeguards the solder points on the PCB.






 @Steve


 I appreciate your point - yes, there is unavoidable handiwork required to install higher quality RCA connectors - and that takes time, and time is money etc. etc. This may have been a deliberate decision by Benchmark.

 We can see however, that they saw fit to use a chassis-mounted BNC connector, requiring manual assembly & soldering. Somebody thought it was important enough to use this connector (I'm no expert in volume manufacturing - maybe a PCB-mount version does not exist). What is the incremental cost of adding chassis-mounted RCAs, given that a human must be involved on behalf of the BNC ?

 What I do know is the excellent sound within the stock DAC1 is more clearly perceived when you improve the RCA connectors. (All you balanced-mode guys can stop laughing now...).

 My original question challenges the assertion by Elias that NOTHING - regardless of parts cost - can be changed to improve the DAC1, without a more elaborate topology (pardon my paraphrase Elias). This is what I conclude from reading post #612 & #710.

 That position is different from stating something like "We made it as good as possible, within a price point" - where a justification could be made for sockets so... "economical" (again... my phrasing). However, this is a very high-quality/ high-value boutique product - and make no mistake - I LOVE the sound my Benchmarks give me... and that sound is throttled (in my personal opinion) by the choice of RCA sockets. 

 Given that Benchmark is courting the home user with the DAC1 (...you don't see many RCAs on serious pro gear), I wonder why they missed on a feature that so many audiophiles value highly ?

 I'm curious to read Elias' response.
 Cheers,
 Grant_

 


 I too have found that steel conductors, such as those used in the PC mount connectors are detrimental to the sound. I avoid steel, even in coax cables. Copper is best with brass next best. A resolving system with excellent imaging is required to hear the difference though.

 Just for grins, I searched some of my catalogues and found a PC-mount 50 ohm BNC jack for $1.98 retail in qty 1 from Kobiconn. Less than $1 in qty 1000.

 Bomar has even better BNC PC-mount jacks for around $1 in 1000 qty in both 50 and 75 ohms. Amphenol is more like $5 in qty.

 Steve N.


----------



## gjwaudio

Hi Elias, thanks for you reply. In the spirit of Debate and Inquiry, can we look at the yin-yang nature of your answer ?
  Quote:


  Originally Posted by *EliasGwinn* /img/forum/go_quote.gif 
_...The DAC1 was originally designed for professional applications where balanced interconnection is prominent... We did not use heavy-duty RCA connectors because they are not often used in the pro-industry..._

 

Absolutely agree... By Professionals - For Professionals. The DAC1-Classic has been well received by the target audience (pro-audio users), whose ravings have spurred sales in both the Pro and Home markets. _("If it's good enough for Bob Katz, I guess it'll do for me !")_ That's one reason I bought mine !

 With the introduction of the DAC1-USB, Benchmark is clearly reaching out to the Home user - where unbalanced connections are in the majority. (I suspect precious few USB ports are being used in professional DAW installations, since digital I/O is usually provided by a dedicated card [and/or breakout box]. Yes... there are USB-based recording interfaces, but integrating the DAC1-USB via it's USB input in that setting, is not possible).

 If we can agree that Home users are the primary target for this model - and RCAs are their preferred connection - consider this assertion:
  Quote:


  Originally Posted by *EliasGwinn* /img/forum/go_quote.gif 
_...For unbalanced users, a more durable RCA connector would understandably be more desirable._

 

Does it follow then, that the DAC1-USB would better serve it's primary user group with a "more durable RCA connector" ? It's already differentiated from the DAC1-Classic by price and feature set, so why not build it as Audiophile-Ready ? I ask you, what is the incremental cost of adding decent chassis-mounted RCAs, given that human assembly is already required for the BNC ? Why not remove that from the list of considerations a prospective buyer must weigh ?
  Quote:


  Originally Posted by *EliasGwinn* /img/forum/go_quote.gif 
_...it is our belief that the RCA connectors used in the DAC1 do not limit or degrade the audio quality of the RCA outputs._

 

Hmmm... somewhat in contradiction to the previous assertion, but OK... Does that mean you (Benchmark) actually installed and evaluated higher quality RCA sockets during DAC1 development - and measured (and more importantly HEARD !) no difference in audio quality ? OR was the decision based solely on previously held beliefs, without resort to the physical world ?
  Quote:


  Originally Posted by *EliasGwinn* /img/forum/go_quote.gif 
_We take these criticisms very seriously, and appreciate the opportunity to learn what our customers want._

 

I accept this as true - and not a glib tagline to close with, when there's nothing left to say. It's for that reason I'm banging the drum on this issue today - not for my benefit (Hey... mine's already "fixed" !) - but for those who haven't yet pulled their money out of their pockets. These guys & girls should be able to experience all the quality that's built into the DAC1.

*Let The Music Out Of The BOX !!*

 Well... enough of that. Let's all have a Great Weekend !
 Cheers,
 Grant


----------



## EliasGwinn

Quote:


  Originally Posted by *gjwaudio* /img/forum/go_quote.gif 
_With the introduction of the DAC1-USB, Benchmark is clearly reaching out to the Home user - where unbalanced connections are in the majority. (I suspect precious few USB ports are being used in professional DAW installations, since digital I/O is usually provided by a dedicated card [and/or breakout box]. Yes... there are USB-based recording interfaces, but integrating the DAC1-USB via it's USB input in that setting, is not possible)._

 

This is possible. I have done exactly this using Nuendo.

  Quote:


  Originally Posted by *gjwaudio* /img/forum/go_quote.gif 
_Does it follow then, that the DAC1-USB would better serve it's primary user group with a "more durable RCA connector" ? It's already differentiated from the DAC1-Classic by price and feature set, so why not build it as Audiophile-Ready ? I ask you, what is the incremental cost of adding decent chassis-mounted RCAs, given that human assembly is already required for the BNC ? Why not remove that from the list of considerations a prospective buyer must weigh ?_

 

This is a valid point, and it is something we will consider when designing products in the future.

  Quote:


  Originally Posted by *gjwaudio* /img/forum/go_quote.gif 
_Hmmm... somewhat in contradiction to the previous assertion, but OK... Does that mean you (Benchmark) actually installed and evaluated higher quality RCA sockets during DAC1 development - and measured (and more importantly HEARD !) no difference in audio quality ? OR was the decision based solely on previously held beliefs, without resort to the physical world ?_

 

We did not do A/B comparisons with various RCA connectors. We have, however, listened to and tested the DAC1 via the RCA connectors and compared the performance to the XLR outputs and found them to be very comparable (the XLR's will have 6 dB more gain and will be more immune to common-mode noise due to balanced circuitry).

 I'll clarify the statements that appeared contradictory to you. The RCA connectors on the DAC1 may not be as physically robust as those you suggest, but we believe they are more then capable electrically (and also physically, in most settings).

  Quote:


  Originally Posted by *gjwaudio* /img/forum/go_quote.gif 
_I accept this as true - and not a glib tagline to close with, when there's nothing left to say. It's for that reason I'm banging the drum on this issue today - not for my benefit (Hey... mine's already "fixed" !) - but for those who haven't yet pulled their money out of their pockets. These guys & girls should be able to experience all the quality that's built into the DAC1._

 

It is very true for future product development. We take all of our user feedback to the drawing board with us when its time to develop new products.

 Thanks,
 Elias


----------



## Crowbar

Quote:


  Originally Posted by *audioengr* /img/forum/go_quote.gif 
_I too have found that steel conductors, such as those used in the PC mount connectors are detrimental to the sound._

 

Could be because it's magnetic--that could introduce some hysteresis. The question is is it no less than -100 dB from signal to be audible, or is this more placebo.


----------



## cansman

Hi Elias,

 I just bought the DAC1 USB. It has astonishing resolution and is dead quiet! 
	

	
	
		
		

		
			





 I have a few questions, please:

 1. What is the recommended operating temperature range? My unit is running really warm ( I should say hot) in tropical weather 31 degrees C.

 2. Related to the above, are any of the circuitry running in class A bias?

 Thanks!
 cansman


----------



## twsmith

Quote:


  Originally Posted by *cansman* /img/forum/go_quote.gif 
_What is the recommended operating temperature range? My unit is running really warm ( I should say hot) in tropical weather 31 degrees C._

 

I've also noticed my DAC1 runs somewhat warm almost all the time even in standby mode (no signal), although it seems to get much warmer when in active mode. I've always assumed this is normal but is this something I need to be concerned about?


----------



## EliasGwinn

Quote:


  Originally Posted by *cansman* /img/forum/go_quote.gif 
_Hi Elias,

 1. What is the recommended operating temperature range? My unit is running really warm ( I should say hot) in tropical weather 31 degrees C.

 2. Related to the above, are any of the circuitry running in class A bias?

 Thanks!
 cansman_

 

The DAC1 operates near body temperature (~100 degree F). 

 The DAC1 uses a high-current, low-impedance design which optimizes audio performance and uses significant power. This is a design trade-off. When designing audio equipment, there exists a trade-off between audio performance and power consumption. The same things that make an audio circuit superior also induces power consumption - specifically, high voltage rails, high currents and low impedance. These are important for two main reasons: increased signal-to-noise performance and higher bandwidth.

 The opamps in the DAC1 (NE5532 and, in the DAC1 USB, LM4562) are both internally class AB. The have near non-existent cross-over distortion (internally biased) and very linear slew-rates. 

 If your DAC1 seems to be warmer then 100 degrees F, make sure nothing is placed on top of the DAC1.

 Thanks,
 Elias


----------



## twsmith

Quote:


  Originally Posted by *EliasGwinn* /img/forum/go_quote.gif 
_If your DAC1 seems to be warmer then 100 degrees F, make sure nothing is placed on top of the DAC1._

 

Elias,

 What about placing the DAC1-USB on top of another component? Space is at a premium in my set-up, so I've had to stack mine on an amp that also can get fairly warm. There is nothing on top of the DAC.

 Thanks


----------



## Herandu

Quote:


  Originally Posted by *EliasGwinn* /img/forum/go_quote.gif 
_The DAC1 operates near body temperature (~100 degree F). 

 The DAC1 uses a high-current, low-impedance design which optimizes audio performance and uses significant power. This is a design trade-off. When designing audio equipment, there exists a trade-off between audio performance and power consumption. The same things that make an audio circuit superior also induces power consumption - specifically, high voltage rails, high currents and low impedance. These are important for two main reasons: increased signal-to-noise performance and higher bandwidth.

 The opamps in the DAC1 (NE5532 and, in the DAC1 USB, LM4562) are both internally class AB. The have near non-existent cross-over distortion (internally biased) and very linear slew-rates. 


 Thanks,
 Elias_

 

Do these NE5532 need high current with a low impedance, or is that for the DAC chips? 
 Also, are you sure that when designing audio equipment there is a trade off between audio performance and power consumption? Care to point us to the source of this information?

 I have nothing against the NE5532. Very good set of chips for about U$1.00 a piece down my end. But I use those in cheap to build circuits. Would the AN627 etc not be more the kind of stuff we should be expecting in a high-end DAC? Or is the quality of the sound not down to the cost of the components, but to the high current, low impedance power supply


----------



## Herandu

Quote:


  Originally Posted by *audioengr* /img/forum/go_quote.gif 
_I guess you dont understand volume manufacturing process. These connectors would have to be hand-wired and hand-soldered. I believe except for the transformer/AC inlet and the S/PDIF jack, everything in the DAC-1 is machine assembled. If it were possible to machine-assemble these and maintain quality, they would probably be machine assembled too. Once you start adding manual interventions like this, the price goes up dramatically.

 Steve N._

 

I don't believe that for one moment. It might be true in a Western factory and Japan, but not so in China, and less so in Taiwan and South Korea. I have seen the rows of people on the assembly lines assembling and soldering up together quite complex items that were not subject to additional labour cost. Factories I have toured are the likes of Jesmay, Tonic, Archief etc. All low cost solution providers that use a lot of hand labours in stead of just machinery.


----------



## cansman

Thanks Elias for all your feedback. I actually decided to buy the DAC1 USB after reading through this thread.

 Cheers,
 cansman


----------



## EliasGwinn

Quote:


  Originally Posted by *twsmith* /img/forum/go_quote.gif 
_Elias,

 What about placing the DAC1-USB on top of another component? Space is at a premium in my set-up, so I've had to stack mine on an amp that also can get fairly warm. There is nothing on top of the DAC.

 Thanks_

 

This shouldn't be a problem. It will increase the temp of the DAC1, but it should not affect its performance at all.

 Thanks,
 Elias


----------



## EliasGwinn

Quote:


  Originally Posted by *Herandu* /img/forum/go_quote.gif 
_Do these NE5532 need high current with a low impedance, or is that for the DAC chips? 
 Also, are you sure that when designing audio equipment there is a trade off between audio performance and power consumption? Care to point us to the source of this information?

 I have nothing against the NE5532. Very good set of chips for about U$1.00 a piece down my end. But I use those in cheap to build circuits. Would the AN627 etc not be more the kind of stuff we should be expecting in a high-end DAC? Or is the quality of the sound not down to the cost of the components, but to the high current, low impedance power supply_

 

The 5532 doesn't need high-current...it provides high-current. The 4562 provides even more current....hence the 'High-Current Output Drivers' in the DAC1 USB. 

 The reason one would want a high-current chip is because a low output impedance requires a lot of current. The chip must provide high current-output before distortion. A low-impedance design is desirable because it offers significant improvements in bandwidth because the effect of the RC network will be minimized. 

 Also, we run these chips on 18V rails which provides an excellent signal-to-noise ratio. This consumes a lot of power as well. For these reasons, it is at the expense of power consumption that we achieve superior sonic performance.

 Thanks,
 Elias


----------



## cansman

Elias,

 I have set iTunes volume to maximum. The Audio MIDI Default Output is set to Benchmark 1.0 @ 44.1kHz and 24-bits

 I am probably just hearing things but I find that toggling the System Output between Benchmark 1.0 and Built-in Audio (Properties for Built-in Audio at various volume settings and mute) is varying very slightly the sound!

 Shouldn't the choice of System Output (Benchmark 1.0 or Built-in Audio) not matter? I believe that the System Output is concerned with alert and interface sounds rather than the streaming audio output.

 Would you be able to clarify? As I said again, I am probably hearing things in my head!

 Thanks so much!

 cansman


----------



## EliasGwinn

That's interesting... I'll have to look into that. I tried it for a moment and couldn't hear a difference, but I only listened very briefly. It could be that there is some mixing happening...similar to kmixer. And when the mixing happens, there may be some DSP (sample-rate conversion or something) that affects the quality. I'll try to find some time to put it on the bench and determine what is happening, if anything. For right now, I would suggest to leave 'System Output' to 'Built-in'.

 Thanks,
 Elias


----------



## cansman

Thanks Elias. For your info, it appears to my ears that the best setting for System Output is Benchmark 1.0. But at the same time, Properties for Built-in Audio be set to maximum volume, unmuted.

 Cheers!
 cansman


----------



## cansman

BTW, I'm using the USB input, if that matters.

 Thanks!


----------



## Wiza_Gab

Can it be a good DAC for electronic music or rap ? (I listen to alot of different kind of music too)

 And cansman, why do you have a DAC1 and some headamps, I tought it was a stand-alone DAC, it doesn't sound good without headamp ?


----------



## lowmagnet

Quote:


  Originally Posted by *Wiza_Gab* /img/forum/go_quote.gif 
_Can it be a good DAC for electronic music or rap ? (I listen to alot of different kind of music too)_

 

I listen to everything on the DAC1 (EVERYTHING) including electronic and rap. DAC1's accuracy lends to it the ability to play any format back without 'color'.


----------



## Jetlag

I was actually hoping someone would invent a device that could automatically filter out rap, country or anything by Barbara Streisand or Fleetwood Mac.


----------



## cansman

Hi Elias,

 Do you have any updates on the Mac OS testing regarding System Output? Thanks!


----------



## puntloos

Quote:


  Originally Posted by *lowmagnet* /img/forum/go_quote.gif 
_I listen to everything on the DAC1 (EVERYTHING) including electronic and rap. DAC1's accuracy lends to it the ability to play any format back without 'color'._

 

Rap without color? Is that Eminem and stuff?


----------



## EliasGwinn

Quote:


  Originally Posted by *Wiza_Gab* /img/forum/go_quote.gif 
_Can it be a good DAC for electronic music or rap ? (I listen to alot of different kind of music too)

 And cansman, why do you have a DAC1 and some headamps, I tought it was a stand-alone DAC, it doesn't sound good without headamp ?_

 

Wiza_Gab,

 The DAC1 is designed to playback music exactly as the artists/producers/engineers wanted the music to sound. When the people were in the studio making the music, they were listening through a pro-quality D-to-A converter, such as the DAC1. (and the coolest part for bass-heavy music is the DAC1's bandwidth extends below 1 Hz, and is incredibly phase accurate for really tight bass...not floppy or muddy at all).

 The headphone amplifier in the DAC1 is similar in respect to its pinpoint accuracy. Some folks prefer to drive other headphone amps with the DAC1 because they have a particular fondness for certain headphone amps. I work in a professional studio and I have a listening room at my house, and I use nothing but the DAC1 in both locations.

 Thanks,
 Elias


----------



## Wiza_Gab

I'm the one who thanks you Elias. 
	

	
	
		
		

		
		
	


	




 I'm probably going to buy it soon, it looks fantastic because currently I feel that the Total Bithead hasn't enough power to drive my HD650 at their best.


----------



## cansman

Sorry Wisa_Gab,

 I missed your post to me. I have several amps for different use - like the Hornet and Tomahawk for portable duties. My Naim amp is used with my main sound system.

 The Benchmark is used on my home office desk. As Elias said, I feel the headphone section of the DAC is very good and the bass, really tight and timing, quite superb!

 Regards,
 cansman


----------



## Lord Chaos

Bass from the Benchmark is exactly what goes into it. If the recording engineer used a good mic on the drums, you will feel the thump of the attack.

 Mine (I have one in the bedroom and one on my desk) work with everything I play. Other than the Revox A77 tape deck I had, the DAC1 is the best component I've ever bought.


----------



## dmk005

Does anyone use the DAC 1 USB as a preamp? I received my DAC1 USB this week and am comparing it to a Meier Corda Opera both as a preamp with speakers and as a headphone amp. I notice the bass with the DAC 1 USB as a preamp is a little leaner than the Opera although not at all thin, just less warm, less punchy. I am comparing relative to the Opera here. I am not comparing the DAC1 USB preamp to it's headamp but comparing the DAC1 USB headamp to the Opera headamp and then comparing the DAC1 USB preamp to Opera as a preamp. By the way, the highs on the DAC1 USB extend beautifully and the mid range is slightly more transparent than the Opera and very well defined. I found a surprising source song with lots of attack dynamics...believe it or not the song Buenos Aires on the Madonna "Evita" motion picture soundtrack.

 If you use your DAC1 USB as a preamp, do you leave the -20db jumper in place? I am considering removing it since my monitors don't seem to be too sensitive and with some source material, I need the extra gain (classical).

 I have found the DAC1 USB to be a solid performer as a DAC. It is very stable with zero hiccups in my limited use.


----------



## Lord Chaos

I use the DAC1 USB, which is on my desk and connnected to two computers, as a pre-amp. It drives the Sony ES-line power amp, and sounds good. I haven't done anything with the settings, so the 20dB pad is wherever it was set at the factory. The volume control is at a good place, about 10 or 11 o'clock, for normal listening.

 I also use this one with headphones (E500) when I want isolation.

 One computer is a PC, connected with S/PDIF. The other is a Mac Powerbook, using the USB.


----------



## dmk005

Quote:


  Originally Posted by *Lord Chaos* /img/forum/go_quote.gif 
_I use the DAC1 USB, which is on my desk and connnected to two computers, as a pre-amp. It drives the Sony ES-line power amp, and sounds good. I haven't done anything with the settings, so the 20dB pad is wherever it was set at the factory. The volume control is at a good place, about 10 or 11 o'clock, for normal listening.

 I also use this one with headphones (E500) when I want isolation.

 One computer is a PC, connected with S/PDIF. The other is a Mac Powerbook, using the USB._

 

Mine is connected via USB to a Mac Mini. My Denon 5801 is the power amp being driven by the DAC 1 USB and the Opera. I do not have any sources of variation that could be throwing this A/B test off except perhaps the difference in gain between the preamps and the fact that the Opera has a low/high gain setting. I use a radio shack SPL meter upon set up to make sure I am listening with the same gain settings. My speakers are the terrific Legacy Studio's in gorgeous rosewood. This is my office set up and really both DAC/AMP combos sound good but....I cannot avoid pointing out the differences I observe.

 Question for Elias or others who may know; Does the DAC1 USB use it's headamp as part of the chain for the preamp? I appreciate the base when comparing these components via the headamp but less so when listening via the DAC1 USB as the headamp. Not sure how relevant that point is when doing the comparison but I need to make sure my chosen source is locked in before adding to the chain.


----------



## EliasGwinn

Hi dmk, thanks for your questions...

  Quote:


  Originally Posted by *dmk005* /img/forum/go_quote.gif 
_I do not have any sources of variation that could be throwing this A/B test off except perhaps the difference in gain between the preamps and the fact that the Opera has a low/high gain setting. I use a radio shack SPL meter upon set up to make sure I am listening with the same gain settings._

 

This is the biggest factor in comparing products, so it is very good that you recognized and calibrated it. (Well, the ubiquitous 'placebo effect' is also a big factor, but we'll leave that for some other folks to discuss in another thread.)

  Quote:


  Originally Posted by *dmk005* /img/forum/go_quote.gif 
_Does the DAC1 USB use it's headamp as part of the chain for the preamp? I appreciate the base when comparing these components via the headamp but less so when listening via the DAC1 USB as the headamp._

 

The headamp and the preamp portion of the DAC1 are independant, but both are being driven from the same source. Regarding your observations, we have heard and tested many, many audiophile devices which intentionally boost the low-end to give an impression of an 'impressive bottom'. I have not heard or tested the pre-amp that you are using, so I do not wish to imply that it is doing that. However, the DAC1 was designed to reproduce the bottom, mids, and tops exactly as they are presented in the source material.


----------



## tosehee

I have read your article already and have set the settings accordingly with iTune. Audio MIDI, Sound Check, Sound Enhancers, and etc.

 So, what you are saying is, if I get the older model, I would get the same result coming from Optical. is this correct? I just want to double check because someone told me that if the sound is coming from optical, it's already processed by the interval sound card, and if I pass that to DAC, I am basically re-processing it and the sound would be worst since the data is already altered unlike direct USB connection.

 Is this true statement or false? I am currently using HD 600 and planning to get other high impedance cans. I doubt that I would buy any low impedance cans in the future. So, the switch and a new driver may not be a killing option for me.

 Thanks again for your prompt reply.

 Regards.


----------



## EliasGwinn

Tosehee,

 The DAC1 Classic will serve your purposes well.

 Thanks,
 Elias


----------



## tosehee

Quote:


  Originally Posted by *EliasGwinn* /img/forum/go_quote.gif 
_Tosehee,

 The DAC1 Classic will serve your purposes well.

 Thanks,
 Elias_

 

Just to re-iterate, what basis does this person come up with the idea that internal sound card processes the sound already if using the optical line, but a direct raw feed if using USB?


----------



## EliasGwinn

Tosehee,

 I'm not sure what basis their conclusion was made upon, but we have tested the optical output of a MacBook and have concluded that it is, in fact, bit-transparent. We proved this by playing an audio file which was actually a pre-known, pseudo-random bit sequence, and using an Audio Precision to monitor the digital output and compare the bits to the original sequence. If the audio was unprocessed, the sequences should match perfectly. The optical output was providing digital output which was bit-for-bit exact to the original audio file.

 Thanks,
 Elias


----------



## tosehee

Thanks again. 

  Quote:


  Originally Posted by *EliasGwinn* /img/forum/go_quote.gif 
_Tosehee,

 I'm not sure what basis their conclusion was made upon, but we have tested the optical output of a MacBook and have concluded that it is, in fact, bit-transparent. We proved this by playing an audio file which was actually a pre-known, pseudo-random bit sequence, and using an Audio Precision to monitor the digital output and compare the bits to the original sequence. If the audio was unprocessed, the sequences should match perfectly. The optical output was providing digital output which was bit-for-bit exact to the original audio file.

 Thanks,
 Elias_


----------



## cansman

Dear Elias,

 I re-read the Benchmark Wiki on Mac OSX. I missed this earlier but ONLY under the Audio MIDI Setup page does it actually state to put ALL volume controls on devices, input and output, to 100%:

 "It also has volume controls for each device and signal path (that is, input and output). It is recommended, however, to keep these volume controls at 100%."

 Also, in the above screenshot where Built-in Output is set to Digital Out, notice BOTH System Output and Default Output are set to Built-in Output (naturally, this is the default if there are no other devices connected to the Mac). This implies however that both System and Default Outputs are optically streamed out of the Mac to the Benchmark. I presume this somehow ensures bit-perfect streaming.

 Consequently, since I'm using the USB mode, the System and Default Outputs should also be set to Benchmark 1.0 to mimic the above setting to ensure bit-perfect streaming. This might explain why I was hearing verying sound qualities in the different settings, as stated in my earlier post.

 Unfortunately, I have only my ears to guide me (I completely agree with Benchmark stance on the subjectivity of human listening perception!). I do not have the techinical equipment nor the expertise to verify the above statements. That's where I need your clarification.

 If my observations are accurate, my recommendations are as follows (I'm only trying to help and clarify 
	

	
	
		
		

		
			





):

 1. Could you please expand the Benchmark Wiki section on Mac OSX to include USB use (in particular to recommend users to set System and Default Outputs to Benchmark 1.0).

 2. Also, this is not stated in the web-site or elsewhere but under system preferences>sound>sound effects, Alert volume should also be set to 100%, with 'user interface sound effects' and 'play feedback when volume is changed' both checked (once again this needs to be verified).

 3. Would the instructions to put Mac OSX volume control to the maximum on the web and instruction manual be expanded to include the above observations.

 Thank you for reading this post. This is my last initiated post regarding Mac OSX sound settings.

 Warm regards,
 cansman


----------



## EliasGwinn

Cansman,

 I really appreciate your feedback on these articles. I figured that, as computer technology is dynamic and evolutionary, so must my articles. Therefore, I am relying heavily on people such as yourself to 'keep it in check'. Thank you.

 I would like to perform the tests on these settings soon, but I'm not sure when I will be able to get around to it. I am currently digging my way through a buried desk...but once I get some priority tasks taken care of, I will try this and report back (and update the article on the website).

 Thanks again, cansman, and everyone else who keeps the flow of information alive and exciting.

 Elias


----------



## cansman

Hi Elias,

 Thank you for your reply - I look forward to the test results! This will indeed help Mac OS X users like myself have the confidence of the best setting for the DAC1 USB.

 Cheers,
 cansman


----------



## Soundproof

To all who have participated in this thread - A Big THANKS!
 Lots of useful information, lots of myths examined, quite a few discounted.

 As one in the process of moving my music to hard disk storage, at full resolution, this has made things a lot clearer.

 A question on the potential difference of quality between the optical connection and the USB connection.
 Stereophile has examined the Grace m902. Wes Philips was dissatisfied with the sound he heard through the USB connection, and was given the advice by John Atkinson that it should be ditched. This due to fears of ground loop induced hum. http://stereophile.com/headphones/406grace/index1.html

 In other comments, including re. the Benchmark, they've made it clear that they prefer the optical connection.
 Meanwhile - and true to audiophilia indeterminacy - accepted belief at AudioAsylum is that optical TosLink is inferior to USB.

 Any thought here, Elias? What are your own experiences with the two connection methods? Any differences?

 I use, and will be using, Macs to port out my music to both a Benchmark DAC1 USB I have connected to one sound system, and a Grace m902 I'm also using, and would like to hear your opinion on this particular topic: USB vs. TosLink optical - advantages/disadvantages.


----------



## Crowbar

Comments like this on Toslink are meaningless without context!
 As has been said a million times around head-fi, Toslink transmitter and receiver components have high intrinsic jitter that exceeds that of S/PDIF and USB. Measurements by someone at an RF lab posted at the diyhifi forums confirm this. The only advantage of Toslink is ground loop prevention--but you shouldn't have ground loops on properly constructed and used equipment in the first place (and with S/PDIF transformer isolation also gets rid of ground loops). If your source is a PC, then you may also get some electrical noise over the USB connection.
 In the case of a DAC like this one, you probably can't hear any difference in regards to the jitter, since the ASRC attenuates it by a large amount.
 All of this is stuff that has been covered on head-fi already.
 By the way, most stuff posted on the AA forums is fairy tales, so I suggest you go to more respectable places, or shortly you'll be buying gold conductor cables and cryo-treated connectors.


----------



## Soundproof

Are you referring to the Stereophile reviewers' opinion?

 I'm just asking yours. Or actually that of Elias at Benchmark.

 Between the points of attitude I'm able to cull the facts that:

 1. Toslink has high intrinsic jitter, exceeding that of S/PDIF and USB,
 2. That ground loop may be prevented.
 3. And that some electrical noise may be transmitted down the USB wire.

 Hey - that's pretty helpful - though somewhat at variance with JA and WS at Stereophile.

 I agree with your opinion on Audio Asylum. It's a myth factory. And no - you won't see me buying gold conductor cables.


----------



## EliasGwinn

Soundproof,

 We have done several listening and bench tests comparing the various digital inputs of the DAC1 USB. All tests indicate that all four digital inputs perform equally with the DAC1 USB.

 This isn't generally the case, for instance, with all D-to-A converters and/or other digital audio receiving devices. It usually depends on galvanic isolation, PLL design, etc.

 Luckily, you won't have to worry about any of that with the DAC1. For example, I am using a Lynx AES16 PCI interface which offers 16 channels of AES digital audio I/O to and from my computer. It is a very well built card...well respected in the industry. I use it to stream digitally to my DAC1 USB for doing audio editing with my digital audio workstation. However, when I listen to iTunes, etc, I simply use the USB port -> DAC1 USB so that I don't have to reconfigure the AES16. The USB port works and sounds just as well!! Which makes me 
	

	
	
		
		

		
		
	


	




 !!

 Thanks,
 Elias


----------



## rmh1

Any chance that there will be a DAC1-USB released without the headamp? That would save a bit of cash for those of us with amps already. Probably not the original intent of the DAC1, but the USB version is obviously a hit around here and I (and probably others) might be more inclined to buy it if I could save the cash for balanced cables, etc. Just an idea.

 Cheers


----------



## EliasGwinn

Rmh1,

 We currently do not have any plans to make a DAC1 USB without the headphone amp. Sorry...

 Thanks,
 Elias


----------



## choariwap

sir,

 i just bought a dac1 usb and i'm very pleased with it 
	

	
	
		
		

		
		
	


	




 one thing though, the one i got is an ROHS compliant version. there is a page in the manual that states that ROHS compliant versions are less reliable in the long term, so much so that extended warrantee only goes to 2 years instead of 5.

 1. does this mean that the life expectancy of my dac1 is only 2 years? thats a pretty short time for a 1375 usd dac.
 2. have you been getting a lot of RMAs for ROHS compliant products?
 3. what sort of failures are there?
 4. what should i do to preserve the life of my dac1? i've read that lead free solder doesnt do well if thermal cycled.. does the auto standby feature drop temps?
 5. what was the lead free solder and manufacturing treatment used for the dac1 usb?

 i'm more than a little nervous about this.

 thanks!

 ps: i'm not a heavy user, just a home user. listen to it for around 3-4 hours twice a week on weekdays and 8-16 hours total over the weekend.


----------



## EliasGwinn

We have no reason to believe that the life-expectancy of the ROHS compliant DAC1's are any shorter then the non-ROHS versions (which we are no longer manufacturing). 

 The reason for the shortened warranty is because the ROHS regulation standards includes the use of lead-free solder. This has been known to cause 'tin whiskers', which could lead to short-circuits, and potentially damage the unit. The only known method of preventing 'tin whiskers' is adding lead - hence the dilemma. Every company which manufacturers electronics to be sold in the EU and other ROHS-compliant country is experiencing the same dilemma.

 So far, since the transition to ROHS-only manufacturing, we have not seen any increase in RMA's or failed units. However, it would be catastrophic to any small company if the lead-free solder failed and they had to recall every single unit made - especially if the majority of the products were sold abroad. 

 Thermal cycling may increase the chances of failure. Therefore, it is recommended to keep the unit powered up continuously. The auto-standby feature of the DAC1 USB will not cause the unit to cool, so it does not present a problem.

 Thanks,
 Elias

 ps. If you feel that the lead-free solder regulation is a bad idea, please write to the appropriate government representatives of your country and tell them to modify the ROHS standards to allow lead-based solder.


----------



## EliasGwinn

Cansman,

 I got a chance to put the MacBook on the bench. It seems to be bit-transparent independent of what 'System Output' is set to.

 Also, I noticed that iTunes was able to achieve bit-transparency at 44/24, which is new to this version (7.3.1). It is still NOT bit-transparent at 24-bits with other sample-rates. (I don't have any idea why it would be like this, but
	

	
	
		
		

		
		
	


	




 ...thats a question for Apple).

 Thanks,
 Elias


----------



## gregeas

Elias, 

 Since you're still here after all these months, I wanted to run a new product idea by you. 

 I can't be the only one who has the need for a new type of digital two-channel pre-amp. Essentially what I'm envisioning is a full set of digital inputs, a high-quality converter like the one in the DAC1, and remote controlled input switching and volume (with a nice analog pot). While you're at it make it full rack width... 

 Right now I'm using my Slimdevices Transporter for this; it converts digital signals from my cable box and PS3 and of course handles PC music playback with an excellent UI. It does the job pretty well -- conversion is very nice --but doesn't have an analog volume control, and the inputs can't easily be switched remotely (the process requires multiple button presses). So I've had to keep my pre-amp in the rack to make the system user friendly. 

 So really the only differences between what I'm proposing and the DAC1 are a remote control and perhaps more a few more inputs for consumer audio. If this product existed I could get rid of my pre-amp altogether and pair my digital pre-amp with my amps directly... Sweet. 

 I thought about getting a nice audiophile pre-pro, but I'd be paying for extra channels of conversion and video features I don't need.


----------



## cansman

Quote:


  Originally Posted by *EliasGwinn* /img/forum/go_quote.gif 
_Cansman,

 I got a chance to put the MacBook on the bench. It seems to be bit-transparent independent of what 'System Output' is set to.

 Also, I noticed that iTunes was able to achieve bit-transparency at 44/24, which is new to this version (7.3.1). It is still NOT bit-transparent at 24-bits with other sample-rates. (I don't have any idea why it would be like this, but
	

	
	
		
		

		
			





 ...thats a question for Apple).

 Thanks,
 Elias_

 

Hi Elias,

 Thanks so much for testing the Mac OS X! It's really good to know that I can divert system sounds away from the DAC1 - it gets annoying with the system sounds occasionally breaking through!

 Based on your testing, is bit-transparency 44/24 achieved regardless of volume settings of the Built-in Audio (input and output)? This helps in being able to independently set system sounds to a reasonable level without it being too loud through the computer speakers.

 Related to the above and as my previous post, if the above is true, I presume the alert volume setting under system preferences>sound>sound effects can be set at whatever level and the choice to engage or disengage the "play user interface sound effects" and "play feedback when volume is changed" settings does not affect bit-transparency?

 Finally, thanks for discovering this - it is really baffling that iTunes is not bit-perfect at other sampling rates other than 44/24! Fortunately for me, most of my music is encoded with apple lossless.

 Thanks once again Elias for being so helpful.

 Warm regards,
 cansman


----------



## EliasGwinn

Cansman,

 If you are sending the system output to the 'built-in output', I am fairly certain that the volumes and other settings for system output will not affect the 'default output', which is the DAC1 USB in this case. 

 iTunes is bit-transparent at other sample-rates with 16-bit audio, but only 44.1kHz with 24-bit audio. It is bit-transparent at the following resolutions: 44/16, 44/24, 48/16, 96/16.

 Thanks,
 Elias


----------



## cansman

Elias,

 You have answered all my questions 
	

	
	
		
		

		
		
	


	




. Thanks once again for putting the DAC1 on the test bench and clarifying bit transparent sampling rates.

 Very much appreciated!

 Cheers,
 cansman


----------



## EliasGwinn

Quote:


  Originally Posted by *gregeas* /img/forum/go_quote.gif 
_Elias, 

 ...a new type of digital two-channel pre-amp. Essentially what I'm envisioning is a full set of digital inputs, a high-quality converter like the one in the DAC1, and remote controlled input switching and volume (with a nice analog pot). While you're at it make it full rack width... 

 So really the only differences between what I'm proposing and the DAC1 are a remote control and perhaps more a few more inputs for consumer audio. If this product existed I could get rid of my pre-amp altogether and pair my digital pre-amp with my amps directly... Sweet._

 

Gregeas,

 We don't currently have any plans to build such a device, but I am interested in your suggestion. Could you provide a more detailed list of what features you are specifically requesting?

 Thanks,
 Elias


----------



## gregeas

Quote:


  Originally Posted by *EliasGwinn* /img/forum/go_quote.gif 
_Gregeas,

 We don't currently have any plans to build such a device, but I am interested in your suggestion. Could you provide a more detailed list of what features you are specifically requesting?

 Thanks,
 Elias_

 


 The idea would to make a pro studio/audiophile grade "digital pre-amp" that would combine the features of the DAC1 and a traditional pre-amp into a single chassis. Another way to think of this is as a simplified pre-amp/processor focused on two channel audio. 

 It would have these features:

 * Full rack width size
 * High quality analog volume control
 * Switchable digital inputs including coax x 3, optical x 2, XLR, USB (similar to the array on the DAC1 plus a few extra consumer inputs)
 * RCA x 2 and balanced output (an extra set of RCA out would be excellent for external headamps)
 * Top-quality D to A conversion
 * Head amp a plus
 * Analog inputs a plus
 * Inputs and volume controlled remotely
 * Some kind of visual display for the volume level for non-desktop use

 I got this idea from using a pre-amp and a Slimdevices Transporter, which has multiple digital inputs, in my main system. Many of the features above overlap in these two boxes, but I can't combine them without making sacrifices. For example, the Transporter doesn't have an analog volume control, just a digital one. 

 The idea is to use the digital pre-pro as a high-quality hub for various devices that have digital output: cable box, DVD player, CD player, computer, game console, etc. 

 The digital pre-pro would allow me to switch inputs and control volume remotely, from my listening position. Combine it with a nice amp/speakers and BAM, you've got an amazing minimalist system. 

 The drawback of using a DAC1 in this type of non-desktop setting is that there is no remote... 

 I don't think this would require much engineering in addition to what Benchmark has already done, and I imagine a product like this would find its way into home audiophile systems and pro studios, much as the DAC1 has done. 

 I'd buy this product in a second. It might be a bit ahead of its time, but demand for something like this will be there, I believe. Witness the growing audiophile acceptance of computer audio... I for one have ditched all of my disk spinner at this point.


----------



## choariwap

Quote:


  Originally Posted by *EliasGwinn* /img/forum/go_quote.gif 
_We have no reason to believe that the life-expectancy of the ROHS compliant DAC1's are any shorter then the non-ROHS versions (which we are no longer manufacturing)._

 

thanks for the quick reply elias, your posts are always very helpful 
	

	
	
		
		

		
		
	


	




 what lead free alternative is benchmark using? what solder alloy and manufacturing treatment? i've read that there were initially many solder options but now they have been narrowed down to just a few with each having different weaknesses and strengths. 

 thanks!


----------



## xp9433

Quote:


  Originally Posted by *gregeas* /img/forum/go_quote.gif 
_The idea would to make a pro studio/audiophile grade "digital pre-amp" that would combine the features of the DAC1 and a traditional pre-amp into a single chassis. Another way to think of this is as a simplified pre-amp/processor focused on two channel audio. 

 ........I'd buy this product in a second. It might be a bit ahead of its time, but demand for something like this will be there, I believe. Witness the growing audiophile acceptance of computer audio... I for one have ditched all of my disk spinner at this point._

 

Elias

 I think gregeas is spot on. I'd buy one in heart beat as well. It would solve all my media requirements.

 Frank


----------



## EliasGwinn

Although I appreciate the feedback about new products, it should be said that there are no plans for such a product. If we were to make such a product, it wouldn't be available for at least 6-9 months, and probably a lot longer. But, as I said, we have no plans for such a product yet.

 But we do appreciate hearing from our customers about what they want. We take this feedback very seriously, and we factor it in to our product development plans. However, with a company as small as ours, as well as creating such intensely engineered products, product development is a slow beast. If we were Sony, and pushing out off-the-shelf solutions in a shiny plastic box, we could probably have something available in the next few weeks/months. But, thankfully, we aren't Sony!! 
	

	
	
		
		

		
		
	


	




 Thanks,
 Elias


----------



## EliasGwinn

Dear Head-Fi-ers,

 I am adding new articles to Benchmark's Audio-Wiki. I would like to extend an offer to everyone to make suggestions for topics to be written about. If there are any ambiguous audio-related subjects, or anything you would like clarified, please feel free to suggest them for coverage in Benchmark's Audio-Wiki articles.

 Thanks,
 Elias


----------



## Bootleg

Quote:


  Originally Posted by *gregeas* /img/forum/go_quote.gif 
_>The new DAC1 USB, which begins shipping worldwide on March 1, 2007 
 >at a price of $1275 USD, also includes high-current output drivers that 
 >can be configured to mute upon headphone insertion. 

 What the heck are high-current output drivers?_

 

hey greg,

 your PM's are full...send me an email


----------



## Bootleg

Quote:


  Originally Posted by *EliasGwinn* /img/forum/go_quote.gif 
_We have no reason to believe that the life-expectancy of the ROHS compliant DAC1's are any shorter then the non-ROHS versions (which we are no longer manufacturing). 

 The reason for the shortened warranty is because the ROHS regulation standards includes the use of lead-free solder. This has been known to cause 'tin whiskers', which could lead to short-circuits, and potentially damage the unit. The only known method of preventing 'tin whiskers' is adding lead - hence the dilemma. Every company which manufacturers electronics to be sold in the EU and other ROHS-compliant country is experiencing the same dilemma.

 So far, since the transition to ROHS-only manufacturing, we have not seen any increase in RMA's or failed units. However, it would be catastrophic to any small company if the lead-free solder failed and they had to recall every single unit made - especially if the majority of the products were sold abroad. 

 Thermal cycling may increase the chances of failure. Therefore, it is recommended to keep the unit powered up continuously. The auto-standby feature of the DAC1 USB will not cause the unit to cool, so it does not present a problem.

 Thanks,
 Elias

 ps. If you feel that the lead-free solder regulation is a bad idea, please write to the appropriate government representatives of your country and tell them to modify the ROHS standards to allow lead-based solder._

 


 HOLD THE PHONE!

 Am I smoking the wrong stuff, or am I hearing that ALL (meaning 100%) of all Benchmark DAC1 USB's are only covered by a TWO year warranty as opposed to FIVE??? (Including those sold in the US???)


----------



## riceboy

Quote:


  Originally Posted by *Bootleg* /img/forum/go_quote.gif 
_HOLD THE PHONE!

 Am I smoking the wrong stuff, or am I hearing that ALL (meaning 100%) of all Benchmark DAC1 USB's are only covered by a TWO year warranty as opposed to FIVE??? (Including those sold in the US???)_

 

I would like to know as well. Could someone from Benchmark give us confirmation on this? 2 to 5 years is a huge difference. Thanks in advance.


----------



## gregeas

I have another idea for a Benchmark product. First, I understand the chances of these coming to market are slim...but this one has potential.

 The idea is to make a miniature version of the DAC1 with internal batteries that could be charged via the USB connection, much like an iPod or Blackberry phone. I would also include a jack for an external Elpac-type power supply for long-term use.

 Keep the head amp, but lose the XLR inputs and outputs.

 I'd call this a "transportable" product that could work on a desktop, at home, or on the road. The new Lisa III head amp fits into this category. 

 If the output quality of the DAC1 could be maintained, this could be a killer product, as this plus a laptop would make a portable audiophile source. I have to imagine there's a market for this type of thing, given its versatility. 

 I've been thinking about this since often toss my DAC1 in my laptop bag. The DAC1 isn't a crippling burden, but it would be nice if it were a bit smaller, and you didn't have to carry a power cable. 

 What do you think?


----------



## EliasGwinn

Benchmark still offers a 5 year warranty to US customers who register their product.

 Thanks,
 Elias


----------



## wakeride74

Is there any sonic benefit in using XLR to RCA vs. just RCA out to a dedicated headphone amp?


----------



## EliasGwinn

If you are driving RCA inputs, there are no benefits for using the XLR outputs. In fact, if the device you are driving has low input impedance, the DAC1's RCA outputs are designed with 30 ohm output impedance to accommodate for that. So, in that case, it is advantageous to use the RCA outputs.

 Thanks,
 Elias


----------



## puntloos

A new question that came up for me:

 When connecting the DAC1 to a 'faraway' (10M) computer, how afraid should I be of leak currents? Is it safe enough to just yank the (BNC) connector off the benchmark while both devices are up and running, or should you always turn off/disconnect from mains either or both devices?


----------



## EliasGwinn

Holy smokes!! Can you believe this thread has gone on for 40 pages?!? 
	

	
	
		
		

		
			





 wow....

 Thanks to you all for this thread...it really has been a lot of fun. I don't mean this to sound like a 'good-bye' speech or anything, but it just dawned on me... Lets see if we can hit 80 pages...
	

	
	
		
		

		
		
	


	




 Thanks again!!
 Elias


----------



## EliasGwinn

Quote:


  Originally Posted by *puntloos* /img/forum/go_quote.gif 
_A new question that came up for me:

 When connecting the DAC1 to a 'faraway' (10M) computer, how afraid should I be of leak currents? Is it safe enough to just yank the (BNC) connector off the benchmark while both devices are up and running, or should you always turn off/disconnect from mains either or both devices?_

 

puntloos, 

 Are you referring to leak currents on the digital connection between the DAC1 and the computer? If so, you really don't need to worry about that at all. There are only two operating states with the DAC1. Either it is working at its optimal performance specs, or else there is no sound output and the error indicators are lit. The way UltraLock works is, the DAC1 will lock to the digital signal as long as it is legible at all to the DAC1, and the conversion quality will be of the same high performance level as expected from the DAC1. In other words, if there is audio coming from DAC1, and there are no error indicators lit on the front panel, the sound quality is as good it would be from the best (ideal) digital signal/source. However, if the digital signal/noise is too erroneous such that UltraLock cannot lock to it, the DAC1 will simply not output sound, and the error indicators on the front will tell you so. Then, there will be no question that something is wrong with the connection.

 As for disconnecting the BNC cable, you do not need to turn-off or mute anything when you connect or disconnect any digital inputs. The DAC1 has a soft-mute function which prevents large 'pops' whenever a digital signal is connected and/or removed.

 Thanks,
 Elias


----------



## Lord Chaos

40 pages... and the "Jenga Thread" rolls on. As do my delightful DAC1s.


----------



## gjwaudio

Hi All... Yes 40 pages and barely a Flame ! Let's hear it for intelligent discourse.

 To continue on then, I have a story to share, and a few questions for Elias. Although general in nature, they can be specific here, since the gear involved is made by Benchmark (ADC1 & DAC1-Classic).

 The whole tale is rather long, so let me first ask:

 Q1: Why does 192 KHz SR sound best (better than 96 KHz, which sounds better than 44.1 KHz) ?

 Q2: Why does the DAC1-CL produce a different sound quality when different digital cables connect it to the same source ?

 Hmmm... Lots of Debate lurking in those two questions.

 Background: I believe official Benchmark position (please correct me if I'm wrong) is the DAC1 is insensitive to:
 a) interface jitter
 b) transmission protocol (AES/EBU, optical, USB or S/PDIF)
 c) cable type/ construction (meaning different brands within each protocol)

 Further, as stated a few posts ago, the DAC1 either works - AT FULL POTENTIAL - or not at all (muting when input is too weak or noisy to read). You may hear variations in sound quality due to the properties of your ANALOGUE connection downstream from DAC1 (impedance mis-match, cable length, RF interference, etc), but the signal present at the outputs will always be "the same" for a given incoming datastream.

 OK so far. And just to be clear - I LOVE MY BENCHMARKS !! - and this is NOT a challenge to Elias or Benchmark, as much as a quest for enlightenment... to explain the following experience and observations.


 Last week (away on holiday) I paid a visit to a professional recording studio in Montreal. My friend is the owner, and I've been bugging him for years about getting a DAC1 into the monitoring system ("How else do you REALLY know what you're doing?"). This time I hauled in both my DAC1, and the newly-acquired ADC1 (BTW... EVERYBODY archiving from analog to digital storage needs an ADC1 !!!)

 There was a mixdown booked in the big room - so my timing couldn't be better. The engineer - we'll call him "Bob" - had been working all day on "touch ups" to various songs (changes requested by the client based on prior rough mixes). Bob is a seasoned studio pro (20+ years) with many fine albums to his credit - and has VERY good ears. After the session was over (and the client safely out of the building) I offered to show him "something interesting".

 For all the gear-heads, here's the relevant equipment Bob was using...
 - Custom-built audio PC (2 x UAD1 effects cards/ RME 9652 Hammefall PCI audio interface)
 - Apogee AD-8000 converters with D2A option (lightpipe for recording, S/PDIF for mixing) 
 - Blue Sky BMC (bass management controller)
 - KRK Systems E8 powered monitors (L/R only) + Gale 10 subwoofer

 Recorded audio file format: 24-bit/44.1 KHz SR broadcast wave.
 DAW: Steinberg Nuendo v3.x
 Music: 1m30s section looped for continuous playback of female vocal, piano, bass & drums accompaniment. Think of Sade's first album - Smokey, Sumptous and Seductive.

 To this setup I introduced the DAC1 & ADC1, with a selection of digital cables:
 - Apogee Wyde Eye 110 ohm AES/EBU (Neutrik XLRs)
 - Apogee Wyde Eye 75 ohm S/PDIF coaxial (solder-type BNCs)
 - Mogami 75 ohm S/PDIF coaxial (Canare 75 ohm RCAs + Benchmark BNC adapters)
 - Petra 75 ohm "video cable" (solder-type BNCs)

 Analogue connections via Mogami quad-star balanced mic cable.

 == The Demos ==

 Test1 (digital to analog converter):
 Substituted AD-8000 DAC section with DAC1 (using existing S/PDIF cable, and feeding BMC from Benchmark XLRs (variable O/P set, and kept constant for all tests).

 PlayBack01 - 44.1 KHz SR (original rate of project files)

 Test2 (increasing the sample rate): 
 Reconfigure. Put the AD-8000 DAC back in service, but fed it's analogue output into the ADC1. ADC1 -> DAC1 via Wyde Eye AES/EBU. DAC1 balanced into BMC -> monitor system. (eg: analogue sound from Apogee converter is encoded by ADC1 and decoded into monitor system by DAC1)

 PlayBack02 - 44.1 KHz SR
 PlayBack03 - 48 KHz SR
 PlayBack04 - 96 KHz SR
 PlayBack05 - 192 KHz SR


 Test3 (digital cable substitution):
 Signal path remains as in Test2, ADC1 kept at 192 KHz SR. Digital cable linking the two Benchmarks is changed for this test.

 PlayBack06 - 192 KHz SR- AES/EBU - Wyde Eye 110
 PlayBack07 - 192 KHz SR - S/PDIF - Mogami digital coax
 PlayBack08 - 192 KHz SR - S/PDIF - Wyde Eye digital coax
 PlayBack09 - 192 KHz SR - S/PDIF - Petra "video cable" coax

 PlayBack10 - 192 KHz SR - S/PDIF - Mogami digital coax
 PlayBack11 - 192 KHz SR - S/PDIF - Wyde Eye digital coax
 PlayBack12 - 192 KHz SR - S/PDIF - Petra "video cable" coax

 == Results & Reactions ==

 PB01: I wish I had a photo of the look on Bob's face !! Twenty seconds into it he muttered a few choice words... at the end went on about bass extension and mid-bass definition. (remember he'd just spent a few hours tweaking the mix - so he was intimately familiar with "the sound" of his tracks... OR WAS HE !!)

 The DAC1 allowed him to quickly fine-tune the EQ of the bass drum - get it out of the way for the bass - and bring clarity to the foundation of the mix.

 Test2 was interesting for the fact that although an additional encode/decode stage was in the chain, the Benchmarks enhanced the listening experience (remember the analogue source is now the (lesser) Apogee unit). PB03 sounded no better than PB02, as expected. PB04 improved overall spacial definition and PB05 added a little more of the same. It just got better as the SR rose.

 From Q1 at the top... how can it be Elias, that the 192 KHz (when resampled to DAC1's internal 110) sounds "better" than at 96 KHz (and this in light of your explanation that 192 filters perform worse than those at 96 KHz... Can Benchmark indeed perform miracles ?).

 As interesting as the increasing-SR demo was... Test3 was even more curious ! Bob declared 192 KHz did sound the best, thus we kept it constant for the next round.

 PB07 had a slight advantage over PB06 - could it be protocol related ? Didn't have more AES/EBU flavours to test. The Wyde Eye is the most costly cable in the group, and was judged to perform worse than either the Mogami or the Petra. PB07 to PB12 presented us with clear, indisputable evidence that swapping cables between the Benchmarks produced different results in the monitors - all other things being equal.

 Bob described the Wyde Eye as "bumping the bass up an octave"... since the extreme lows were missing compared to the other two cables. Of course it didn't shift the frequencies, but that was the impression when the bottom fell out of the bass range. The Mogami extended a touch deeper than the Petra... but the Petra had better textural definition in the lows & mid-bass, and the greatest clarity in the higher range (reverb tails, spacial positioning, etc). No doubt there is some insanely expensive cable that can do greater magic between the ADC1 & DAC1... but the punchline is... the Petra is cheap cheap cheap to buy (see this page -> PETRA C2207/NY/G/25FT Digital Component Video Cable.

 In the audio production world, even small improvements are highly prized - and Bob was impressed - and perturbed - with the implications of this little demonstration. And he's SURE he needs a DAC1... soon !

 I'm now at a loss to understand how this experience squares with the technical position of Benchmark engineering (remember Q2 ?). 

 If you're still reading... congratulations for your tenacity, but this post has gone on long enough !! I eagerly await comments and explanations.

 Cheers...
 Grant


----------



## puntloos

Quote:


  Originally Posted by *EliasGwinn* /img/forum/go_quote.gif 
_puntloos, 

 Are you referring to leak currents on the digital connection between the DAC1 and the computer? If so, you really don't need to worry about that at all. There are only two operating states with the DAC1. Either it is working at its optimal performance specs, or else there is no sound output and the error indicators are lit. The way UltraLock works is, the DAC1 will lock to the digital signal as long as it is legible at all to the DAC1, and the conversion quality will be of the same high performance level as expected from the DAC1. In other words, if there is audio coming from DAC1, and there are no error indicators lit on the front panel, the sound quality is as good it would be from the best (ideal) digital signal/source. However, if the digital signal/noise is too erroneous such that UltraLock cannot lock to it, the DAC1 will simply not output sound, and the error indicators on the front will tell you so. Then, there will be no question that something is wrong with the connection.

 As for disconnecting the BNC cable, you do not need to turn-off or mute anything when you connect or disconnect any digital inputs. The DAC1 has a soft-mute function which prevents large 'pops' whenever a digital signal is connected and/or removed._

 

Let me describe my current setup then:

 1/ I've now bought a Benchmark DAC1. (but the store knows it is 'on trial'.)
 2/ Ive got one PC, in a closet somewhere (controlled by my pocketPC over wireless) which has one long high-end coax cable (RCA) going to my stereo.
 3/ At the stereo, I have a RCA splitter, and I'm sending the signal to both the DAC1 and the Bel Canto DAC3 
 4/ The DAC1 connects to the RCA-in of my preamp, the DAC3 is using XLRs

 Notes:
 #3 is BLASPHEMY, yes I know, but it does work flawlessly so far. Also it means I can actually truely ABX both devices under at least 'similar' source quality.
 #4 is a touch unfair to the benchmark DAC1, balanced XLR is simply superior. I do plan to switch them around at some point during testing.

 Tests so far leave me hard pressed to hear any differences. I thought my DAC sounded better, then I turned down its volume a little bit and then they sounded equal. 

 My worry and my question revolve not around sound quality/jitter but basically that I know computers are often wired very badly, and you could literally get a -powerful- shock from cables (like the RCA) connected to them. 

 This is especially true for situations where Im holding the 'scary' cable coming from my PC in my hand, and I know that if i touch its leads it will hurt a bit.. wildly guessing (based on PC powersupply specs) Im probably feeling 12 volts at 1.5 amps.. and then plugging it into expensive gear like that.. or will something that hurts me not hurt a (decently constructed) electrical device?


----------



## puntloos

Hi gjwaudio, nice post

 Let me address a few points
  Quote:


  Originally Posted by *gjwaudio* /img/forum/go_quote.gif 
_Q1: Why does 192 KHz SR sound best (better than 96 KHz, which sounds better than 44.1 KHz) ?_

 

In theory this of course is a no-brainer, but in practice a lot more comes into view. One of the things that is true though is that upsampling 44 to 96 is less accurate than both 44-to-88 (doubling is easy) and 44-to-192 (more room for approximation/aliasing/dithering). Also, the DAC1 internally works at even higher frequencies than 100/192, but the ASRC is a device that tries to make an as accurate as possible 'analog' approximation of its input signal, and THEN lets a sampler (at 96K) take samples. With 192-in, the ASRC approximation would be more accurate. (assuming that 192 doesn't introduce more unwanted artifacts due to device limitations)

 At least, that is my theory.
  Quote:


 Q2: Why does the DAC1-CL produce a different sound quality when different digital cables connect it to the same source ? 
 

That is quite curious. Are you ABSOLUTELY POSITIVE? The only way would be to have 2 DAC1s and do a blind ABX test. Just 'switching stuff around' in my experience has more to do with what you think you hear than what you actually hear.

  Quote:


 -snip- 
 

Am I hearing you correctly that what you did was use the benchmark ADC to upconvert the output of a 44K DA convertor? If so then it wouldn't make much sense, weird.

 One thing that puzzles me: (in general, not related to gjwaudio's post)

 If sources (microphones connected to an ADC's input) are recorded at 96/24, is it not true that MIXING these sources at 192/24 would lose you less quality then mixing at 96/24? 
 - Since currently mixing is mostly math inside computers, why not?
 - Even if your target is 96/24 again, attenuation/mixing digitally is a well known source of quality loss, so why not go as high as you can and downmix to target at the very last stage
 - And why settle for 96/24 if your media supports 192/24 as well.. DVD-A?


 Am I missing something?


----------



## EliasGwinn

These are excellent posts and questions...I hope to respond to all of them today. 

 Thanks,
 Elias


----------



## gjwaudio

Hi puntloos... glad you enjoyed The Tale, and it provoked some thought.

  Quote:


  Originally Posted by *puntloos* /img/forum/go_quote.gif 
_That is quite curious. Are you ABSOLUTELY POSITIVE? The only way would be to have 2 DAC1s and do a blind ABX test. Just 'switching stuff around' in my experience has more to do with what you think you hear than what you actually hear._

 

Yes... *ABSOLUTELY* POSITIVE. And Bob - who earns his living based on his ability to discern and create tiny differences in soundscapes - concurs. He found the differences in presentation (by switching digital links) a little disturbing, because it throws into doubt all the mixing decisions made whilst listening through "lesser" playback chains. ("OMG... what have I done that the mastering guy must UNDO ?"... or something like that).

  Quote:


  Originally Posted by *puntloos* /img/forum/go_quote.gif 
_Am I hearing you correctly that what you did was use the benchmark ADC to upconvert the output of a 44K DA convertor? If so then it wouldn't make much sense, weird._

 

Yes, again... correct. Consider that no matter what the digital source characteristics (44.1/ 96/ 192 KHz) once the sound exits the DAC - in this case the AD-8000 - it's analogue and the sampling game begins anew. We were all astonished to hear the polish and clarity of the track after it passed through the ADC1/DAC1 process, compared to the more direct route from the AD-8000 to the monitors.

 Expressed another way:
 A) audio PC -> AD-8000 -> monitors
 B) audio PC -> DAC1 -> monitors
 C) audio PC -> AD-8000 -> ADC1 -> DAC1 -> monitors

 The B-chain sounded better than the A-chain (as expected)... BUT the C-chain was superior to the B-chain.

 Q3: Given the AD-8000 made analogue that was inferior to the DAC1 sound, how is it that the C-chain was judged the BEST ? What Silver Bullet is hiding inside the ADC1 ?

 In fact, Bob said the C-chain makes it sound like it's mastered and ready to press ! Hands down, no contest. Now there's a weird processing loop for the mastering suite !

 OK - let's chew on that for a bit...
 Grant


----------



## EliasGwinn

Grant,

 Since I wasn't involved in your listening test, I wouldn't deny what you heard. I can only tell you about the experiences of my own ABX listening tests, and measurement tests we have performed. We have done tests similar to yours where we used high-precision, short-length Mogami cables; long, generic, impedance mismatched cables; and everything in-between. Although the digital signal looks VERY different at the end of the cable, the DAC1 does not seem to react to those differences. As I said, we have performed ABX listening tests and high-precision measurement tests, and we have never been able to hear or see any differences. 

 Thanks,
 Elias


----------



## EliasGwinn

Quote:


  Originally Posted by *puntloos* /img/forum/go_quote.gif 
_This is especially true for situations where Im holding the 'scary' cable coming from my PC in my hand, and I know that if i touch its leads it will hurt a bit.. wildly guessing (based on PC powersupply specs) Im probably feeling 12 volts at 1.5 amps.. and then plugging it into expensive gear like that.. or will something that hurts me not hurt a (decently constructed) electrical device?_

 

I'm not sure what you are asking, but your coaxial digital audio cable should not be shocking you at all. It seems you have a defective sound card. What sound card are you using?

 Thanks,
 Elias


----------



## fc911c

Hello

 I jst received my DAC1 usb a few days ago and so far so good, no complaints at all. I was wondering if using a longer than provided USB cable will effect SQ. I have the provided cable plugged into a USB Hub which then goes to the PC.

 Are there any other setting's in Foobar I need to make besides changing the out put to 24-Bit?

 Thanks
 Frank


----------



## EliasGwinn

Frank,

 We haven't tested for length limitations for USB cables. I've read that 15 ft (5m) is the limit, but I can't say that with any authority. However, what I can say is that if it sounds good, its probably fine. There will not be any jitter-related distortion due to the UltraLock system. There may be some issues with signal drop-outs, but, as I said, I haven't tried yet, so I don't know.

 For setting up foobar, you'll want to read this article from our Audio Wiki:

http://extra.benchmarkmedia.com/wiki..._-_Setup_Guide

 Let me know if you have any more questions.

 Thanks,
 Elias


----------



## Bootleg

Elias,

 If two people are going to be using the DAC1 USB's headphone section at the same time, say 15 or so feet away while watching a movie, is it better to run TWO 15 foot extensions (one from each jack), or ONE extension and then a "Y" at the end?

 What I'm trying to figure out is, does each headphone jack have its own amp driving it, or do the jacks share and amp?

 If each headphone jack has its own independent amplification, it would seem to me that 2 15' extensions would be the best.

 Thoughts?


----------



## fc911c

Quote:


  Originally Posted by *EliasGwinn* /img/forum/go_quote.gif 
_Frank,

 We haven't tested for length limitations for USB cables. I've read that 15 ft (5m) is the limit, but I can't say that with any authority. However, what I can say is that if it sounds good, its probably fine. There will not be any jitter-related distortion due to the UltraLock system. There may be some issues with signal drop-outs, but, as I said, I haven't tried yet, so I don't know.

 For setting up foobar, you'll want to read this article from our Audio Wiki:

http://extra.benchmarkmedia.com/wiki..._-_Setup_Guide

 Let me know if you have any more questions.

 Thanks,
 Elias_

 

Wow now that's service. I will check out the wiki

 Thanks Elias

 Frank


----------



## puntloos

Quote:


  Originally Posted by *EliasGwinn* /img/forum/go_quote.gif 
_I'm not sure what you are asking, but your coaxial digital audio cable should not be shocking you at all. It seems you have a defective sound card. What sound card are you using?

 Thanks,
 Elias_

 

The soundcard is OK, its a m-audio revolution 5.1.. and actually I didnt test it here, I just know sometimes such things happen, with sound, but also with a monitor cable.. bottomline usually is that computers with non-perfect powersupplies seem to sometimes have a knack for lifting what-supposed-to-be-ground off it.

 Did you see my question about mixing 96/24 sources @ 192? Would you concur that is better practice?


----------



## EliasGwinn

Quote:


  Originally Posted by *puntloos* /img/forum/go_quote.gif 
_If sources (microphones connected to an ADC's input) are recorded at 96/24, is it not true that MIXING these sources at 192/24 would lose you less quality then mixing at 96/24? 
 - Since currently mixing is mostly math inside computers, why not?
 - Even if your target is 96/24 again, attenuation/mixing digitally is a well known source of quality loss, so why not go as high as you can and downmix to target at the very last stage
 - And why settle for 96/24 if your media supports 192/24 as well.. DVD-A?


 Am I missing something?_

 

This point is debated regularly among professional engineers, and each has their own opinion. A BIG factor is how well the software was designed. Specifically, does it perform SRC (sample-rate conversion) without too much distortion? What algorithms are used to perform gain changes? 

 Generally speaking, the foundation of your argument is true. That is, the more samples available, the better the software will perform functions. However, does the advantages of mixing at 96k vs. 192k outweigh the side-effects of SRC? Its impossible to say except on a case by case basis.

 As for settling for 96/24 vs. 192/24, it is widely agreed upon among audio engineers that the state-of-the-technology yields more negative artifacts vs. the benefits gained at 192 kHz. This is only an issue during A-to-D and D-to-A conversion, so it does not affect the debate of SRC to 192 for mixing purposes.

 Thanks,
 Elias


----------



## gjwaudio

Hi Elias
 Forgive my "War and Peace" posting... the questions may be obscured by all the yak. I intended this Q3 query specifically for you (...recall that our listening session concluded with passing analogue through the ADC1/DAC1 combo as shown in the C-chain below)
  Quote:


  Originally Posted by *gjwaudio* /img/forum/go_quote.gif 
_...no matter what the digital source characteristics (44.1/ 96/ 192 KHz) once the sound exits the DAC - in this case the AD-8000 - it's analogue and the sampling game begins anew. We were all astonished to hear the polish and clarity of the track after it passed through the ADC1/DAC1 process, compared to the more direct route from the AD-8000 to the monitors.

 Expressed another way:
 A) audio PC -> AD-8000 -> monitors
 B) audio PC -> DAC1 -> monitors
 C) audio PC -> AD-8000 -> ADC1 -> DAC1 -> monitors

 The B-chain sounded better than the A-chain (as expected)... BUT the C-chain was superior to the B-chain.

 Q3: Given the AD-8000 made analogue that was inferior to the DAC1 sound, how is it that the C-chain was judged the BEST ? What Silver Bullet is hiding inside the ADC1 ?_

 

The heart of the question is: What's happening inside the ADC1 that makes it sound so good ? Listening through the DAC1 (...and resampled to 110 KHz internally), why does the sound change "for the better" as the SR is increased on the ADC1 (...from 96 KHz up to 192 KHz) ?

 Thanks again Elias for taking the time to keep up with the activity on this thread !

 Cheers...
 Grant


----------



## EliasGwinn

Quote:


  Originally Posted by *Bootleg* /img/forum/go_quote.gif 
_Elias,

 If two people are going to be using the DAC1 USB's headphone section at the same time, say 15 or so feet away while watching a movie, is it better to run TWO 15 foot extensions (one from each jack), or ONE extension and then a "Y" at the end?

 What I'm trying to figure out is, does each headphone jack have its own amp driving it, or do the jacks share and amp?

 If each headphone jack has its own independent amplification, it would seem to me that 2 15' extensions would be the best.

 Thoughts?_

 

Bootleg,

 The headphone jacks on the front of the DAC1 are driven with the same amp. Therefore, you will get the same performance using a single extender + Y-adapter as you would using two extenders connected to seperate headphone outputs. 

 However, using a headphone extension presents a potential problem. The negative conductor of each headphone is usually kept isolated from each other until the plug to prevent crosstalk. When the plug is connected to the DAC1, it grounds both conductors to chassis, nearly eliminating the potential for crosstalk. Using the extender puts a marginal amount of impedance between the negative terminals at the plug to ground. Therefore, it will subject your headphones to a small amount of cross-talk. Now, there is reason to be OK with this. That is because headphones already present an unnatural amount of isolation between channels. That is, your ears are used to hearing a little bit of the same thing. Therfore, you may find that you don't mind a little bit of crosstalk. Let me know what you find...

 Thanks,
 Elias


----------



## Hadden

Quote:


  Originally Posted by *fc911c* /img/forum/go_quote.gif 
_
 Are there any other setting's in Foobar I need to make besides changing the out put to 24-Bit?

 Thanks
 Frank_

 

Yes, other than 24bit I don't know which output method I should select -- Asio/DirectSound1or 2, Waveout etc. Should I use the Dither option as well? Playback seems improved when I disabled the DSP I was using, as recommended. Don't know what alltogether gives unaltered sound.


----------



## Bootleg

Quote:


  Originally Posted by *EliasGwinn* /img/forum/go_quote.gif 
_Bootleg,

 The headphone jacks on the front of the DAC1 are driven with the same amp. Therefore, you will get the same performance using a single extender + Y-adapter as you would using two extenders connected to seperate headphone outputs. 

 However, using a headphone extension presents a potential problem. The negative conductor of each headphone is usually kept isolated from each other until the plug to prevent crosstalk. When the plug is connected to the DAC1, it grounds both conductors to chassis, nearly eliminating the potential for crosstalk. Using the extender puts a marginal amount of impedance between the negative terminals at the plug to ground. Therefore, it will subject your headphones to a small amount of cross-talk. Now, there is reason to be OK with this. That is because headphones already present an unnatural amount of isolation between channels. That is, your ears are used to hearing a little bit of the same thing. Therfore, you may find that you don't mind a little bit of crosstalk. Let me know what you find...

 Thanks,
 Elias_

 

Thanks Elias!

 I'll give it a go this weekend and report back.


----------



## puntloos

One more quick question for Elias (but others are welcome too 
	

	
	
		
		

		
			





 ):

 - Given a pristine (or at least not badly damaged) CD, would you prefer a 'high end CD transport' with SPDIF out, or a 'standard' computer (PC + normal CDRom player with proper ripping software and SPDIF out), or no preference at all?

 Small elaboration:
 Many audiophiles seem to think that some hugely expensive CD transport can more accurately read CDs than a normal PC CDrom player can, even though the PC uses all ripping techniques like C1/C2 correction, accuraterip etc etc. Any truth in this? 

 Mind you when a CD is (badly?) damaged I'm at least a bit more open to believe that a top class CD player might be able to approximate the original data better than a CDRom player. Still, the CDRom player isn't in a hurry and could take as many tries as it wants to read the real data..

 Thoughts?


----------



## EliasGwinn

Quote:


  Originally Posted by *Hadden* /img/forum/go_quote.gif 
_Yes, other than 24bit I don't know which output method I should select -- Asio/DirectSound1or 2, Waveout etc. Should I use the Dither option as well? Playback seems improved when I disabled the DSP I was using, as recommended. Don't know what alltogether gives unaltered sound._

 

Hadden,

 If you are using the DAC1 USB, you'll see "Benchmark 1.0" in the "Output Device" menu. That is the output you should select. 

 The dither option should be used if you are playing 24-bit audio with a device which does not support 24-bit playback. In that case, you should set the "Output Format" to the maximum bit depth which your hardware is capable, and apply dither. The DAC1 USB is capable of 24-bit playback, so dither is not necessary.

 These settings, along with those documented in the Foobar page on Benchmark's 'Audio Information Center' (http://extra.benchmarkmedia.com/wiki...php/Main_Page) should ensure bit-transparent playback.

 Thanks,
 Elias


----------



## EliasGwinn

Quote:


  Originally Posted by *puntloos* /img/forum/go_quote.gif 
_One more quick question for Elias (but others are welcome too 
	

	
	
		
		

		
		
	


	




 ):

 - Given a pristine (or at least not badly damaged) CD, would you prefer a 'high end CD transport' with SPDIF out, or a 'standard' computer (PC + normal CDRom player with proper ripping software and SPDIF out), or no preference at all?

 Small elaboration:
 Many audiophiles seem to think that some hugely expensive CD transport can more accurately read CDs than a normal PC CDrom player can, even though the PC uses all ripping techniques like C1/C2 correction, accuraterip etc etc. Any truth in this? 

 Mind you when a CD is (badly?) damaged I'm at least a bit more open to believe that a top class CD player might be able to approximate the original data better than a CDRom player. Still, the CDRom player isn't in a hurry and could take as many tries as it wants to read the real data..

 Thoughts?_

 

Puntloos,

 This is a very good question...and, unfortunately, I don't have an answer for you. We've been asked this question many times, but there are so many factors which would make it incredibly difficult to test. Specifically, it would be difficult to say that one type is better because every drive uses different lasers/readers/decoders/drivers... In other words, its completely possible that CD player 'X' is better then CD-ROM 'Y', and CD-ROM 'Y' is better then CD player 'Z'....even if 'Z' is $10,000!! (Please don't ask me what CD player 'X' is...
	

	
	
		
		

		
		
	


	




 )

 Unofficially, (...yes, that is a disclaimer 
	

	
	
		
		

		
		
	


	




 ), I would trust a CD-ROM with good software before I would a CD player...specifically because of what you mentioned. That is, the CD-ROM is not playing in real-time, so it can go back and re-read until it feels that it has read it properly. Also, if I was very concerned that everything be read perfectly, I would find out what CD-ROM's are used by mastering houses and other institutions which would require high-precision CD reading.

 Thanks,
 Elias


----------



## gjwaudio

Hi Elias

 Any chance you've given post #799 some thought ?

 Thanks...
 Grant


----------



## EliasGwinn

Quote:


  Originally Posted by *gjwaudio* /img/forum/go_quote.gif 
_Hi Elias
 Forgive my "War and Peace" posting... the questions may be obscured by all the yak. I intended this Q3 query specifically for you (...recall that our listening session concluded with passing analogue through the ADC1/DAC1 combo as shown in the C-chain below)

 The heart of the question is: What's happening inside the ADC1 that makes it sound so good ? Listening through the DAC1 (...and resampled to 110 KHz internally), why does the sound change "for the better" as the SR is increased on the ADC1 (...from 96 KHz up to 192 KHz) ?

 Thanks again Elias for taking the time to keep up with the activity on this thread !

 Cheers...
 Grant_

 

Grant,

 I'm sorry I haven't gotten to this question sooner.

 I'd tell you that the reason it sounded so good is because of the 'snake oil' we use, but I have a feeling there are some folks on this thread who may call me out on that. 
	

	
	
		
		

		
		
	


	




 Honestly, I wish I could say that our converters could somehow remove the distortion, noise, and phase inaccuracies added by the previous converter, but the truth is our converters are designed to make everything sound 'as good as they really are'. In other words, if poor equipment or processes are used to create an audio signal, no device can remove or 're-construct' the distorted artifacts from that audio signal. That would be one special device that every studio and audiophile would have to own!!

 Earlier in this thread, I wrote about an experiment we conducted. We took a digital recording of a solo piano, converted it to analog via the DAC1, then converted to digital via the ADC1 and re-recorded it. We then took that 2nd generation, and sent it back through the chain and re-recorded. We repeated with the 3rd generation, and the 4th, and on and on, until we had a recording that went through the DAC1->ADC1 chain 20 times. We then performed double-blind ABX tests between the original digital recording (monitored through a DAC1) and the 20th generation recording (monitored through another, gain-matched DAC1). We performed this ABX test with several engineers, INCLUDING THE ORIGINAL RECORDING ENGINEER who recorded the solo piano piece used in the test. No one who took the test could accurately and consistently determine which recording was the original and which was converted 20 times!!

 I would hypothesize that the reason your experiment yielded better sound quality coming from the ADC1->DAC1 chain is because the gain of that chain was slightly higher then unity. In other words, it was louder coming from that chain versus coming straight from the other converter. Even a slight increase in gain (<1dB) is hard to perceive as louder, but will influence your opinion of the sound quality. But, if I may say so, it is great to hear that it didn't degrade the sound quality even with the extra A-D-A conversion cycle! 
	

	
	
		
		

		
		
	


	




 Thanks,
 Elias


----------



## aljordan

Hi,

 Thanks to all for a very interesting thread to read through. 

 I currently use a PC as the main source for my two channel audio system. I currently use the onboard DACs of a Lynx Two B PCI card which directly feed my amplifier (or amplifiers when using the subs). 

 I have been tempted to try a Benchmark DAC for a while now. I am curious if there are any theoretical or practical advantages / disadvantages of using a Benchmark over the DACs / analog outputs of the Lynx Two B.

 Thank you,
 Alan


----------



## euclid

Quote:


  Originally Posted by *aljordan* /img/forum/go_quote.gif 
_Hi,

 Thanks to all for a very interesting thread to read through. 

 I currently use a PC as the main source for my two channel audio system. I currently use the onboard DACs of a Lynx Two B PCI card which directly feed my amplifier (or amplifiers when using the subs). 

 I have been tempted to try a Benchmark DAC for a while now. I am curious if there are any theoretical or practical advantages / disadvantages of using a Benchmark over the DACs / analog outputs of the Lynx Two B.

 Thank you,
 Alan_

 

the advantages are numerous provided you are able to send bit perfect digital stream from your particular PCI card to the DAC1. this is the exact topic which has been explored deeply within this thread, it needs to be read to be fully understood and then only you can decide if you will benefit.

 off hand the advantage would be isolating the analogue conversion away from the PC to decrese electical interference(as with any external DAC), then the benefit comes from the implementation of Benchmarks DAC design which by all accounts and Elias' explainations is quite good, esp regarding jitter immunity and 110hz sample rate conversion for optimal performance of the DAC chip. plus headphone amp built-in.



 i just felt inclined to post to formally thank Elias for his continued involvment in this thread. my understanding of not only the DAC1, but digital audio in general has increased exponentially. based in what ive read here ive decided to buy a DAC1 as im sure many others have, more manufactures should use this thread as an example of how far good customer service can go to generate customers.

 i just need to decide if i really need the USB functionality. i currently have a Chaintech AV710 which i am using as DAC and analogue output for my PC, but it is capable of sending bit perfect output over SPDIF. would my CPU be taxed less by sending audio through the PCI soundcard or by using the USB interface direct to the DAC1? also is there any way to allow a USB equiped DAP(digital audio player) to send its digital signal to an external USB DAC such as the Benchmark? that would be amazing.


----------



## EliasGwinn

Quote:


  Originally Posted by *aljordan* /img/forum/go_quote.gif 
_Hi,

 Thanks to all for a very interesting thread to read through. 

 I currently use a PC as the main source for my two channel audio system. I currently use the onboard DACs of a Lynx Two B PCI card which directly feed my amplifier (or amplifiers when using the subs). 

 I have been tempted to try a Benchmark DAC for a while now. I am curious if there are any theoretical or practical advantages / disadvantages of using a Benchmark over the DACs / analog outputs of the Lynx Two B.

 Thank you,
 Alan_

 

Hello Alan, and welcome to the forum. 

 As Euclid stated, the biggest advantage of using an external DAC vs. a PCI card is isolating the analog audio from the intensive EMI environment of a computer chassis. Also, it is very difficult to maintain clean digital signals in a computer as well, as the level of high-frequency EMI will introduce significant amounts of jitter-based noise to the signal. Even with an on-board converter clock source, the amount of noise may be detrimental to the quality of the clock, causing severe distortion during conversion.

 Also, an external DAC will have a dedicated power supply which will be able to provide optimal voltage rails to maximize signal-to-noise ratio. 

 There are practical, feature-related advantages as well. For example, the HPA2 is a ultra-clean, un-colored headphone amp which can handle the most diverse range of headphone impedances and designs. Also, the front-panel volume control is incredibly convenient, and, unlike a digital volume control, the DAC1's passive, analog design maintains the full signal-to-noise resolution of the audio.

 Many, many more advantages that are specific to a given system. If you have any more questions, especially relating to your particular system, please feel free to ask!

 Thanks,
 Elias


----------



## EliasGwinn

Quote:


  Originally Posted by *euclid* /img/forum/go_quote.gif 
_i just need to decide if i really need the USB functionality. i currently have a Chaintech AV710 which i am using as DAC and analogue output for my PC, but it is capable of sending bit perfect output over SPDIF. would my CPU be taxed less by sending audio through the PCI soundcard or by using the USB interface direct to the DAC1? also is there any way to allow a USB equiped DAP(digital audio player) to send its digital signal to an external USB DAC such as the Benchmark? that would be amazing._

 

Hi Euclid!

 Your question - whether the computer will be more taxed with PCI-based SPDIF or USB - is a very difficult one to answer, as it is a case-by-case scenario. There is no consistent relationship between the two because it depends on two main factors: the 'weight' of the driver that the PCI card is using; and the 'weight' of other USB devices in use. If the driver for the PCI card is intensive, then it will tax the computer more. If there are several USB devices being used, then the computer may stumble from time to time when trying to stream audio via USB. It is very much a case-by-case basis.

 Your second question - whether a digital audio player can stream digital audio to a USB DAC - is even more difficult to answer. I can tell you that the DAC1 USB is built for native drivers - i.e., drivers which are built-in to the operating system. So, the DAP would need to have native USB audio drivers for it to work with the DAC1. I don't know of any that have such driver, but they may exist. If you find one, please let me know!

 Thanks,
 Elias


----------



## Scrith

Quote:


  Originally Posted by *EliasGwinn* /img/forum/go_quote.gif 
_If you are using the DAC1 USB, you'll see "Benchmark 1.0" in the "Output Device" menu. That is the output you should select._

 

I've been a satisfied DAC1 USB customer for several months now and wanted to share a discovery I made about using it with Windows Vista (which I've shared here and elsewhere, but not in this thread).

 If you use the Kernel Streaming output option in Foobar2000, the Windows USB driver that the DAC1 uses seems to enter into an "exclusive" mode that prevents other applications from creating sound. This is a good thing, in my opinion (no mixing going on), but has another benefit...the Windows mixer in Vista (which is called something like the audio graph, I believe) seems to have a low process priority. You can hear the effect of this if you try to play audio in Foobar with the DirectSound output method while your CPU usage is near 100% (as sometimes happens to me while ripping a CD into FLAC format using EAC with multiple compression threads, or playing a CPU-intensive game). It sounds like lots of clicking and popping (and dropouts).

 However, if you switch to Kernel Streaming output, the problem disappears. I assume this is because the low-priority mixing process is no longer being used.

 As usual, I'll close by asking for a Benchmark internet forum! This 41-page thread is a real pain when you're trying to hunt for information or follow a thread that comes-and-goes over a period of several days or weeks.


----------



## euclid

Quote:


  Originally Posted by *EliasGwinn* /img/forum/go_quote.gif 
_Hi Euclid!

 If there are several USB devices being used, then the computer may stumble from time to time when trying to stream audio via USB. It is very much a case-by-case basis.


 Thanks,
 Elias_

 

thanks for responding, this is the exact situation i am concerned about. my computer is not dedicated for audio use i also run apps and games, i am concerned i could overload my USB ports with a keyboard/mouse/card reader/DAP all requesting bandwidth at any given time... if i am also trying to continuously stream audio data to the DAC1 i am worried about lag. 

 the DAP question still beckons me. i imagine that since DAPs have firmware then some reflash hack can be devised to handshake with the DAC1 since it is designed to use native USB drivers. i'll create a dedicated thread and hopefully dig up some more info on this possibility. 

 thanks again

 edit: http://www.head-fi.org/forums/showthread.php?t=256327

 Elias does the DAC1-USB specifically use the USB drivers designed for Microsoft/Apple/Linux OS, or can a new third-party driver be used as a handshake even now after the R&D has been completed on the DAC1-USB?


----------



## EliasGwinn

Quote:


  Originally Posted by *Scrith* /img/forum/go_quote.gif 
_As usual, I'll close by asking for a Benchmark internet forum! This 41-page thread is a real pain when you're trying to hunt for information or follow a thread that comes-and-goes over a period of several days or weeks._

 

Scrith, 

 Thanks for the suggestion. We've recently started an Audio Information Center  (http://extra.benchmarkmedia.com/wiki....php/Main_Page) to document all sorts of setup configurations, technology FAQ's, and other valuable information related to audiophiles, recording studios, and other environments where the DAC1 is used. 

 We may consider starting a Benchmark Forum, but we'll probably wait until after we get the Audio Information Center healthy and full first.

 Thanks,
 Elias


----------



## EliasGwinn

Quote:


  Originally Posted by *euclid* /img/forum/go_quote.gif 
_thanks for responding, this is the exact situation i am concerned about. my computer is not dedicated for audio use i also run apps and games, i am concerned i could overload my USB ports with a keyboard/mouse/card reader/DAP all requesting bandwidth at any given time... if i am also trying to continuously stream audio data to the DAC1 i am worried about lag._

 

Euclid, 

 We've had many customers with heavy USB activity who had no problems with the DAC1 whatsoever. We've had very few customers who actually had any issues with streaming to USB because of interfering devices, etc. And, as always, you'll have 30 days to try the USB interface to see if it will work properly. If there are any problems, you'll know right away. Then you can either return the unit or use another digital input. 

 HEY HEAD-FIER'S!! Does anyone here use the USB interface of the DAC1 USB along with many USB devices simultaneously? Have you experienced any problems? Can anyone share any experiences with Euclid? 

  Quote:


  Originally Posted by *euclid* /img/forum/go_quote.gif 
_Elias does the DAC1-USB specifically use the USB drivers designed for Microsoft/Apple/Linux OS, or can a new third-party driver be used as a handshake even now after the R&D has been completed on the DAC1-USB?_

 

It uses the USB drivers from Windows and OSX, but there is no reason to believe it is limited to these. The DAC1 USB 'says' to the host "I am a USB Audio device capable of 2-channels of input at resolutions up to 96/24." If a driver is designed such that it can 'hear' that message and react to it properly, I would think any OS could work with the DAC1. I'm not an expert on OS's though, so don't quote me on that.

 Thanks,
 Elias


----------



## mofonyx

Hi Elias,

 I couldn't find exact information about the Benchmark DAC1 regarding this on your official website so I'll ask it here.

 What is the maximum voltage swing and maximum gain on the Benchmark DAC1 (on both balanced XLR and single ended RCA) using the Variable volume control option?

 Thank you.


----------



## EliasGwinn

mofonyx,

 Here's what the max voltages are for the analog outputs of the DAC1 with no attenuation and full scall (0 dBFS) digital input:

 XLR in calibrated mode = configurable up to 23 V (29 dBu); factory calibrated for 12.3 V (24 dBu)
 XLR in variable mode = 17.5 V (27 dBu)

 RCA in calibrated mode = configurable up to 3.7 V (13.5 dBu); factory calibrated for 2 V (8 dBu)
 RCA in variable mode = 2.7 V (11 dBu)

 Thanks,
 Elias

 ps. Here's a great tool for converting dBu to voltage, and vice-versa: http://www.sengpielaudio.com/calculator-db-volt.htm


----------



## dmk005

Quote:


  Originally Posted by *EliasGwinn* /img/forum/go_quote.gif 
_mofonyx,

 Here's what the max voltages are for the analog outputs of the DAC1 with no attenuation and full scall (0 dBFS) digital input:

 XLR in calibrated mode = configurable up to 23 V (29 dBu); factory calibrated for 12.3 V (24 dBu)
 XLR in variable mode = 17.5 V (27 dBu)

 RCA in calibrated mode = configurable up to 3.7 V (13.5 dBu); factory calibrated for 2 V (8 dBu)
 RCA in variable mode = 2.7 V (11 dBu)

 Thanks,
 Elias

 ps. Here's a great tool for converting dBu to voltage, and vice-versa: http://www.sengpielaudio.com/calculator-db-volt.htm_

 

I would love to know how the above information is useful as I continue my n nOOb learning curve.


----------



## Lord Chaos

In response to Elias' request for real-world USB stories:

 I use my DAC1-USB with a Mac Powerbook G4 (17", 1.5GHz, 512MB RAM). It has two USB 2.0 ports. One of these is dedicated to the DAC1, while the other has a four-port powered hub connected. The hub hosts the hard disk with my music, my USB adapter for the headset, and the Griffin control knob.

 Occasionally, when the computer is very busy, I get very brief interruptions in the audio from the DAC1. Note that the breaks are clean: music, then maybe a tenth of a second of silence, and then music. No static. The interruptions are rare enough that I don't think about them.


----------



## Scrith

There's a thread on Slashdot about Windows Vista networking problems when playing audio. I haven't noticed it myself (I'll be testing it carefully tonight, however), but perhaps this is also related to the Vista "audio graph" process, which (apparently) processes/mixes all DirectSound audio sent to Windows by various applications. Perhaps the Kernel Streaming output option in Foobar2000 that I mentioned a few posts ago is another solution to this problem (using Kernel Streaming seems to bypass this Windows system process).


----------



## EliasGwinn

We will be getting our new copy of Vista this week. All the testing we have done so far is on the RC2 version of Vista. When the software arrives, we will install and begin testing. I will let you know about any discoveries immediately.

 Thanks,
 Elias


----------



## euclid

Quote:


  Originally Posted by *EliasGwinn* /img/forum/go_quote.gif 
_mofonyx,

 Here's what the max voltages are for the analog outputs of the DAC1 with no attenuation and full scall (0 dBFS) digital input:

 XLR in calibrated mode = configurable up to 23 V (29 dBu); factory calibrated for 12.3 V (24 dBu)
 XLR in variable mode = 17.5 V (27 dBu)

 RCA in calibrated mode = configurable up to 3.7 V (13.5 dBu); factory calibrated for 2 V (8 dBu)
 RCA in variable mode = 2.7 V (11 dBu)

 Thanks,
 Elias


 ps. Here's a great tool for converting dBu to voltage, and vice-versa: http://www.sengpielaudio.com/calculator-db-volt.htm_

 

holy crap! could 12.3v XLR out be dangerous high input into a preamp? i only have 15ft cable run.

 is there any way to get the voltage under 6v for the XLR in calibrated mode, < 3.0v for each phase 0 and 180?

 ps: i think i know why people have been happy with the balanced HD650 directly from the XLR output


----------



## puntloos

Hi guys,

 OK my test between the Benchmark DAC1, The Bel Canto DAC3 and the (built-in) DAC inside a Bow Technologies ZZ-Eight CD player is done.

 Well, there's good news and bad news for Elias. (& co)

 The good news is that the DAC1's audio quality beats the $7500 CD player's easily. The bad news is that the Bel Canto DAC3 sounds just as good as the Benchmark to me and one of my friends. The second friend had a mild preference for the DAC3 - so I returned the DAC1 (not just because of the minute possible audio quality difference, I was also allowed to still return the DAC1, and the DAC3 would have to be sold secondhand if the DAC1 had the way stronger vote.)

 Some more specifics:

 Setup Used: 
 - Bel Canto Evo 4 gen2
 - Quad 989 speakers.
 - PC playing various audio sources (winamp -> kernel streaming -> M-Audio Revolution 5.1 -> 75ohm coax cable)

 The test:
 Essentially two people sat behind eachother (in the center of the room), while a third person would secretly switch between the 3 devices as source. Devices were all matched to the Bow's audio output. I know, this isn't exactly fair since at least the DAC3 uses digital attenuation. Oh well.

 The ABX tests were then done with multiple sources :

 - Always a use - Madeleine Peyroux (HDCD and normal CD)
 - November '99 - Manu Katzche
 - One without the Other - Dorian Michael - 192/24 DVDA
 - Bad Condition - Steve Pierson - 96/24 DVDA

 Sony Sound Forge 9 was used to downsample sources when needed. (best quality)

 Of course for the CD player, some sources had to be downsampled. I have no idea how to make HDCD sources so the CD player had the advantage with Madeleine, but the disadvantage with the DVDA sources.
 The DAC3 was not able to handle 192Khz sources, so we downsampled.
 The results basically were as follows:

 The statements below were verified by doing the actual ABXing, i.e. if the listener was able to pick out the source they claimed had their preference 3 out of 4 times when been given samples without telling which was which. Also a statement is only mentioned below when 2 out of 3 of our listeners agrees.

 - We were able to distinguish between 96/24 sources and a 48/16 downsample, also played with the same DACs. (downsampling 96 to 44 did seem less fair). The differences were subtle, usually described as more defined highs, but also the stereo positioning seemed more defined.

 - The CD player lost in every situation except when playing the HDCD version while the DAC's had to upsample the normal CD data. The conclusion seems to be that if you have a choice, get the HDCD.

 - Lagavulin Whisky tastes very nice.

 - The sound the CD player made was, in general, much thinner and sounded anemic. Also the depth of the soundstage just seemed less deep, all instruments sounded like they were on one line right between the speakers, no 3rd dimension.

 - Nobody could distinguish between XLR and RCA('tulip') connectors anywhere in the signal line.

 Finally, we tried introducing jitter by (instead of using 75ohm coax cable) we used two long (10m) single strands of standard lamp wire. No luck, both DACs played just fine.

 In the end I guess the conclusion must be that sound-quality wise the DAC3 and the DAC1 are very close rivals, and your choice will be mainly based on things like:
 - Aesthetics, few people will disagree that the DAC3 is built 'prettier', while the DAC1 is an industrial looking piece of equipment.
 - Headphone outs. The DAC3 has none.
 - New Price: DAC1-USB was about $1300, DAC3 is $2500 (also USB)
 - Remote: DAC3 has one, DAC1 doesnt.

 In my case, I bought a brand-new, in-warranty DAC3 for $1500, which perfectly matches my Bel Canto amplifier, so the choice wasn't hard. Sound quality wise, either will do just fine for most people, including me.


----------



## euclid

thanks puntloos, this looks pretty good for the DAC1, the Bel Canto is fairly highly regarded as a "musical" DAC, at least the Bel Canto 2 was.


----------



## EliasGwinn

PLEASE IGNORE THIS POST. It is a repeat of my previous post.

 ps. does anyone know why it does that?


----------



## EliasGwinn

Quote:


  Originally Posted by *euclid* /img/forum/go_quote.gif 
_holy crap! could 12.3v XLR out be dangerous high input into a preamp? i only have 15ft cable run.

 is there any way to get the voltage under 6v for the XLR in calibrated mode, < 3.0v for each phase 0 and 180?

 ps: i think i know why people have been happy with the balanced HD650 directly from the XLR output 
	

	
	
		
		

		
			



_

 

12.3Vrms will not be 'dangerous' for your pre-amp - that is, it will not damage your pre-amp. However, it may cause distortion if the pre-amp is not designed to accept +24 dBu input (which is the case for most 'audiophile' pre-amps and amps). Ideally, it would accept +24 or even +28 dBu to maintain the highest signal to noise ratio. That is, the S-to-N ration suffers when the signal has to be attenuated just to get into the (pre)amp, then the signal is boosted again within the (pre)amp. But....such is the case with most 'audiophile' (pre)amps. The attenuators in the DAC1 make it compatible with this type of gear, but also compatible with professional gear that will accept 'full-swing' signals (+24 dBu or greater).

 If you set the output attenuators of the DAC1 to -10 dB, the max Vrms (in factory-set calibrated mode) will be 3.88V. The DAC1 is factory-set with -20 dB attenuation, which will yield a max Vrms of 1.23V.

 Thanks,
 Elias


----------



## euclid

Quote:


  Originally Posted by *EliasGwinn* /img/forum/go_quote.gif 
_12.3Vrms will not be 'dangerous' for your pre-amp - that is, it will not damage your pre-amp. However, it may cause distortion if the pre-amp is not designed to accept +24 dBu input (which is the case for most 'audiophile' pre-amps and amps). Ideally, it would accept +24 or even +28 dBu to maintain the highest signal to noise ratio. That is, the S-to-N ration suffers when the signal has to be attenuated just to get into the (pre)amp, then the signal is boosted again within the (pre)amp. But....such is the case with most 'audiophile' (pre)amps. The attenuators in the DAC1 make it compatible with this type of gear, but also compatible with professional gear that will accept 'full-swing' signals (+24 dBu or greater).

 If you set the output attenuators of the DAC1 to -10 dB, the max Vrms (in factory-set calibrated mode) will be 3.88V. The DAC1 is factory-set with -20 dB attenuation, which will yield a max Vrms of 1.23V.

 Thanks,
 Elias_

 

thanks very much for the explaination. from my expereince most sources that output 2.5v are still too powerful to plug in directly to power amp, therefore the preamp is actually attenuating the source signal. i think in my system a 12.3v input signal would only allow me to use the first two or three volume steps of my preamp stepped attenuator even in +0db unity gain, major attenuation would be in order.

 its nice to know that by setting the jumpers a more civil voltage can be output from the DAC1, i see how a professional application would differ though.


----------



## EliasGwinn

Quote:


  Originally Posted by *euclid* /img/forum/go_quote.gif 
_... a more civil voltage ..._


----------



## dmk005

Quote:


  Originally Posted by *puntloos* /img/forum/go_quote.gif 
_Hi guys,

 OK my test between the Benchmark DAC1, The Bel Canto DAC3 and the (built-in) DAC inside a Bow Technologies ZZ-Eight CD player is done.

 Well, there's good news and bad news for Elias. (& co)

 The good news is that the DAC1's audio quality beats the $7500 CD player's easily. The bad news is that the Bel Canto DAC3 sounds just as good as the Benchmark to me and one of my friends. The second friend had a mild preference for the DAC3 - so I returned the DAC1 (not just because of the minute possible audio quality difference, I was also allowed to still return the DAC1, and the DAC3 would have to be sold secondhand if the DAC1 had the way stronger vote.)

 Did you compare the DAC1 via USB over the DAC 3 over USB using a 24/96 source? This may be important for those who see USB as their only acceptable alternative to computer-based audio and the DAC3 does not support 24/96 over USB. They claim, however, that it sounds as good if not better...just wondering if you tried this particular set up in your A/B testing.

 Some more specifics:

 Setup Used: 
 - Bel Canto Evo 4 gen2
 - Quad 989 speakers.
 - PC playing various audio sources (winamp -> kernel streaming -> M-Audio Revolution 5.1 -> 75ohm coax cable)

 The test:
 Essentially two people sat behind eachother (in the center of the room), while a third person would secretly switch between the 3 devices as source. Devices were all matched to the Bow's audio output. I know, this isn't exactly fair since at least the DAC3 uses digital attenuation. Oh well.

 The ABX tests were then done with multiple sources :

 - Always a use - Madeleine Peyroux (HDCD and normal CD)
 - November '99 - Manu Katzche
 - One without the Other - Dorian Michael - 192/24 DVDA
 - Bad Condition - Steve Pierson - 96/24 DVDA

 Sony Sound Forge 9 was used to downsample sources when needed. (best quality)

 Of course for the CD player, some sources had to be downsampled. I have no idea how to make HDCD sources so the CD player had the advantage with Madeleine, but the disadvantage with the DVDA sources.
 The DAC3 was not able to handle 192Khz sources, so we downsampled.
 The results basically were as follows:

 The statements below were verified by doing the actual ABXing, i.e. if the listener was able to pick out the source they claimed had their preference 3 out of 4 times when been given samples without telling which was which. Also a statement is only mentioned below when 2 out of 3 of our listeners agrees.

 - We were able to distinguish between 96/24 sources and a 48/16 downsample, also played with the same DACs. (downsampling 96 to 44 did seem less fair). The differences were subtle, usually described as more defined highs, but also the stereo positioning seemed more defined.

 - The CD player lost in every situation except when playing the HDCD version while the DAC's had to upsample the normal CD data. The conclusion seems to be that if you have a choice, get the HDCD.

 - Lagavulin Whisky tastes very nice.

 - The sound the CD player made was, in general, much thinner and sounded anemic. Also the depth of the soundstage just seemed less deep, all instruments sounded like they were on one line right between the speakers, no 3rd dimension.

 - Nobody could distinguish between XLR and RCA('tulip') connectors anywhere in the signal line.

 Finally, we tried introducing jitter by (instead of using 75ohm coax cable) we used two long (10m) single strands of standard lamp wire. No luck, both DACs played just fine.

 In the end I guess the conclusion must be that sound-quality wise the DAC3 and the DAC1 are very close rivals, and your choice will be mainly based on things like:
 - Aesthetics, few people will disagree that the DAC3 is built 'prettier', while the DAC1 is an industrial looking piece of equipment.
 - Headphone outs. The DAC3 has none.
 - New Price: DAC1-USB was about $1300, DAC3 is $2500 (also USB)
 - Remote: DAC3 has one, DAC1 doesnt.

 In my case, I bought a brand-new, in-warranty DAC3 for $1500, which perfectly matches my Bel Canto amplifier, so the choice wasn't hard. Sound quality wise, either will do just fine for most people, including me._

 

Did you get a chance to compare a 24/96 source with the DAC1 over USB with the DAC3 over USB? I obviously ask because the DAC3 does not support 24/96 over USB but they still claim their implementation sounds just as good or better than the DAC1 24/96 over USB. If you use a PC as a source, USB is a must and I wonder what your tests may have turned up.


----------



## puntloos

Quote:


  Originally Posted by *dmk005* /img/forum/go_quote.gif 
_Did you get a chance to compare a 24/96 source with the DAC1 over USB with the DAC3 over USB? I obviously ask because the DAC3 does not support 24/96 over USB but they still claim their implementation sounds just as good or better than the DAC1 24/96 over USB. If you use a PC as a source, USB is a must and I wonder what your tests may have turned up._

 

Hmm, nope, both DACs claim that the source is really irrelevant (be it USB/SPDIF/TOSLINK, all simply get 'dejittered' and the perfect input signal gets fed to the DA section) so comparing 24/96 and 24/48 (I think thats the current max of the DAC3 on USB) is a bit unfair.. I don't have a long USB cable anyway. But why do you say that using the PC as a source, USB is a must? My M-Audio card nicely supplies 192/24 from its SPDIF, which I've tested with some DVD-audio tracks at that rate.

 btw Bel Canto is planning to release an update to their USB in Q3/Q4 2007 which should get them in line with the DAC1. But again, USB really Isn't my cup of tea.


----------



## little-endian

Quote:


  Originally Posted by *puntloos* /img/forum/go_quote.gif 
_One more quick question for Elias (but others are welcome too 
	

	
	
		
		

		
		
	


	




 ):

 - Given a pristine (or at least not badly damaged) CD, would you prefer a 'high end CD transport' with SPDIF out, or a 'standard' computer (PC + normal CDRom player with proper ripping software and SPDIF out), or no preference at all?

 Small elaboration:
 Many audiophiles seem to think that some hugely expensive CD transport can more accurately read CDs than a normal PC CDrom player can, even though the PC uses all ripping techniques like C1/C2 correction, accuraterip etc etc. Any truth in this? 

 Mind you when a CD is (badly?) damaged I'm at least a bit more open to believe that a top class CD player might be able to approximate the original data better than a CDRom player. Still, the CDRom player isn't in a hurry and could take as many tries as it wants to read the real data..

 Thoughts?_

 

Oh yes, when it comes to me, i have thoughts indeed about that topic. Thoughts of huff, to be honest. Why? Because there are so much rumours and absolute nonesense about this that it is unbelievable. One could really doubt the human's common sense.

 Claims that a stand alone player or a cd-rom drive will be able to read the data better or worse in general are false as well. A cd-rom drive has - per se - a better chance to read the required data more than once - when they are reading faster than at single speed (for real time playback). However, in regular playback mode (let us call it CDDA-reading in opposite to DAE), it performs the same as a stand alone player. Some differences exist when it comes to interpolation between wrong or missing samples. Stand alone players are often in advantage here. But this don't has to be this way necessariliy. Some drives were tested in this regard by Pio2001 here for example:

http://pageperso.aol.fr/lyonpio2001/dae/dae.htm

 Puntloos, please always keep in mind that - regardless of any introduces jitter at the output - *every* transport will provide you exactly the same data as far as it is possible for it to reconstruct the data without errors. Same data and a DAC which doesn't care for jitter --> same sound. Period.

  Quote:


  Originally Posted by *EliasGwinn* /img/forum/go_quote.gif 
_Puntloos,

 This is a very good question...and, unfortunately, I don't have an answer for you. We've been asked this question many times, but there are so many factors which would make it incredibly difficult to test. Specifically, it would be difficult to say that one type is better because every drive uses different lasers/readers/decoders/drivers... In other words, its completely possible that CD player 'X' is better then CD-ROM 'Y', and CD-ROM 'Y' is better then CD player 'Z'....even if 'Z' is $10,000!! (Please don't ask me what CD player 'X' is...
	

	
	
		
		

		
		
	


	




 )_

 

Elias, i have to admit that i was a bit surprised about this answer. The question is actually easier than you possibly think. Benchmark itself claims that their DAC1 produces the same sound when fed with the same data, no matter how much jitter the stream contains. The question if different transports provide the same data (after the C1/C2-stages) can be easily proven. Every PC will reveal this. So any other answers to certain scenarios can be derived from this perception.


----------



## dmk005

Quote:


  Originally Posted by *puntloos* /img/forum/go_quote.gif 
_Hmm, nope, both DACs claim that the source is really irrelevant (be it USB/SPDIF/TOSLINK, all simply get 'dejittered' and the perfect input signal gets fed to the DA section) so comparing 24/96 and 24/48 (I think thats the current max of the DAC3 on USB) is a bit unfair.. I don't have a long USB cable anyway. But why do you say that using the PC as a source, USB is a must? My M-Audio card nicely supplies 192/24 from its SPDIF, which I've tested with some DVD-audio tracks at that rate.

 btw Bel Canto is planning to release an update to their USB in Q3/Q4 2007 which should get them in line with the DAC1. But again, USB really Isn't my cup of tea._

 

Actually, I would argue it is not unfair. Bel Canto told me they think their 24/48 over USB sounds better or at least as good compared to what they have heard from any current DAC who claims to support 24/96 over USB. The only one I know of is the DAC1.

 As for USB being a must, that may be an overstatement since there are other options but from a practical pov, I prefer USB because it is universally available with every PC and MAC currently sold and I enjoy the flexibility computer as a source component brings to me. I know SPDIF supports 192/24 and therefore the DAC3 would be comparable to the DAC over SPDIF but I wonder about Bel Canto's claim.


----------



## puntloos

Quote:


  Originally Posted by *dmk005* /img/forum/go_quote.gif 
_Actually, I would argue it is not unfair. Bel Canto told me they think their 24/48 over USB sounds better or at least as good compared to what they have heard from any current DAC who claims to support 24/96 over USB. The only one I know of is the DAC1._

 

Perhaps they were indeed thinking of the Benchmark when making the claim, perhaps not. Fact remains that I couldn't even distinguish the DAC3 from the Benchmark when the DAC3 was given 'even more' advantage - (24/96 - not just 24/48) while the DAC1 got 24/96 too. (and as most people here agree, which connection type, USB, SPDIF/etc, don't matter).

 As mentioned, one of my listeners had a 'slight preference' for the DAC3, which was reproducable, so appearantly he had the better ears, but it was a minor preference.

 And all that's is even leaving the difference between 'perfect reproduction' and 'sweet sounding' alone.

 One thing I can say is that Im fairly sure that Bel Canto, like Benchmark, have not committed the sin of trying to 'pretty up' the sound just to sway uncritical people with 'earpopping' effects.

  Quote:


 As for USB being a must, that may be an overstatement since there are other options but from a practical pov, I prefer USB because it is universally available with every PC and MAC currently sold and I enjoy the flexibility computer as a source component brings to me. I know SPDIF supports 192/24 and therefore the DAC3 would be comparable to the DAC over SPDIF but I wonder about Bel Canto's claim. 
 

Agreed I guess, it might be very convenient at certain moments. For someone who is 'serious' about their home setup though, it's simply not much of an issue since with a $50 pci card you get more range (both cable-length and khz/bitdepth-wise).


----------



## EliasGwinn

Quote:


  Originally Posted by *puntloos* /img/forum/go_quote.gif 
_ (and as most people here agree, which connection type, USB, SPDIF/etc, don't matter)._

 

This is true to an extent. But, to avoid confusion, it should be pointed out that there are major differences in the USB capabilities between the Benchmark DAC1 and the Bel Canto DAC3. Specifically, the Bel Canto is only capable of up to 48/16...it cannot stream 24-bit audio via USB. This means that any 24-bit source or any 16-bit source that was increased to 24-bit will be truncated.

 The DAC1 can stream 96/24 'bit-transparently'. Even the M-Audio PCI cards are not capable of bit-transparency. The DAC1 is the only device (including PCI, USB, etc) we have found to be proven 'bit-transparent' when streaming digital audio from a computer at resolutions up to 96/24.

 Thanks,
 Elias


----------



## Bootleg

Quote:


  Originally Posted by *EliasGwinn* /img/forum/go_quote.gif 
_This is true to an extent. But, to avoid confusion, it should be pointed out that there are major differences in the USB capabilities between the Benchmark DAC1 and the Bel Canto DAC3. Specifically, the Bel Canto is only capable of up to 48/16...it cannot stream 24-bit audio via USB. This means that any 24-bit source or any 16-bit source that was increased to 24-bit will be truncated.

 The DAC1 can stream 96/24 'bit-transparently'. Even the M-Audio PCI cards are not capable of bit-transparency. The DAC1 is the only device (including PCI, USB, etc) we have found to be proven 'bit-transparent' when streaming digital audio from a computer at resolutions up to 96/24.

 Thanks,
 Elias_

 

This is what got me firmly into the Benchmark world. I woke up and realized that it was so much easier for me to manage my music on my PC...and along came a device that made my computer sound GOOD.


----------



## JimP

My Benchmark DAC1USB is the cornerstone of my system(s), extremely happy with this product. I would like to run this out to multiple amps. What is better solution:

 1. run RCA out to amp 1, and use an XLR to RCA converter simultaneously out to amp 2 (can this be done, any issues?)

 2. get an input/output switcher, like one of those Mapletree Audio boxes (adding another box and more interconnects degrades sound?)

 thanks in advance for any input
 __________________


----------



## integrale

Hi All,

 First I want to thank all of you for a terrific thread: it's rare to see a truly useful discussion anywhere online, and I don't believe I've ever seen someone in Elias' position contribute to such a degree. Wonderful resource.

 After reading all (42!) pages of this thread, I bought a DAC1 USB. I've been using it is my main headphone amp (Senn 650s, AKG 271s, and Beyer 990s) and it drives my main system (rotel driving B&Ws). Here are my thoughts after two weeks of use, in no particular order:

 -Outstanding sound, especially the low noise floor, unassailable detail, faithful reproduction, and bulletproof reliability (even when switching out of "standby" mode, though still an odd choice not to have a power button!) I've never heard an artifact or noticed any jitter. It's just transparent.

 -Great build quality. Chunky, detented volume control feels nice, faceplate is well-milled, connectors (esp. headphone jacks) are solid. The only part I was surprised at was the RCA outs - I thought I was going to crash through the rear faceplate when I was attaching my interconnect. The aesthetics could use a little work, but I'm not picky, and there's come cache to having a component that looks like it belongs in a studio (I have the black, non-rackmount).

 -My motherboard has S/PDIF RCA out, and I can hear absolutely no difference between this and the USB input. EXCEPT: when I plug my iPod into my computer and it is doing a heavy sync via USB, the PC (brand new top-shelf HP media center) struggles to stream consistently and the sound gets choppy and I can't listen to it. I imagine this is true of using an external USB HDD as well. So I'm currently only using the digital output from my motherboard (anyone want an X-Fi XtremeMusic?)

 -Preamp section. I would love to use this unit as a preamp, but the lack of a remote to select inputs is a deal killer. I echo the suggestion made earlier that Benchmark start working on a preamp or integrated amp with the DAC1 technology (and headphone amp).

 -Installation. The unit worked perfectly as soon as I plugged it in. The Wiki on Benchmark's website and this thread helped me get Vista and iTunes/foobar up to snuff. I'm interested to hear what Elias finds when testing the final release of Vista.

 So, after all that, my new favorite toy is going back to Benchmark...so I can get a DAC1. I don't need the USB input, and for $300, I can sacrifice the functionality. So a simple question: is everything the same on the DAC1 as it is on the DAC1 USB? Have there been any changes to the DAC1 in the last few years, or would even a used unit provide the same performance?

 Outstanding product, and I recommend it to everyone who asks.


----------



## Bootleg

The headphone amp has been tweaked a little bit in the USB version...check your owner's manual for more info (i don't have it in front of me).

 I don't think its an earthshaking change, but it did sound like they made some adjustments based upon their experience with the standard DAC1.


----------



## Rivendell61

Quote:


  Originally Posted by *integrale* /img/forum/go_quote.gif 
_So a simple question: is everything the same on the DAC1 as it is on the DAC1 USB? Have there been any changes to the DAC1 in the last few years, or would even a used unit provide the same performance?_

 

integrale, see this post:
http://www.head-fi.org/forums/2815332-post216.html

 Mark


----------



## Bootleg

Quote:


  Originally Posted by *Rivendell61* /img/forum/go_quote.gif 
_integrale, see this post:
http://www.head-fi.org/forums/showpo...&postcount=216

 Mark_

 

BOOOOM!

 Rivendell, that is one heck of a second post!

 Thanks!


----------



## JimP

double posting, sorry


----------



## fkclo

Quote:


  Originally Posted by *JimP* /img/forum/go_quote.gif 
_My Benchmark DAC1USB is the cornerstone of my system(s), extremely happy with this product. I would like to run this out to multiple amps. What is better solution:

 1. run RCA out to amp 1, and use an XLR to RCA converter simultaneously out to amp 2 (can this be done, any issues?)

 2. get an input/output switcher, like one of those Mapletree Audio boxes (adding another box and more interconnects degrades sound?)

 thanks in advance for any input
 ___________________

 

Hi Jim,

 I am actually using both ! I use the XLR out to provide the balanced input to my RPX-100 balanced headphone amp, and use the RCA output to go into a RWA Signature 3S switch boxes, through which I can get connected to 4 different headphone amps (with one going back as SE input to RPX-100).

 Through my RPX-100, I try to listen to the difference between the XLR direct input and the RCA SE input through the switch box, and honestly, I cannot tell. I am using some of the best headphones available in the market already, and doing the A/B using the balanced out driving a pair of Equinox cabled HD650.

 So, from my experience, the Benchmark DAC1USB is very versatile and you can be happy with either choice. If you have only 2 amps to connect to, I would suggest the first one is more direct and cost-effective. But if you have more amps, that a high quality switch box is not a bad idea. BTW, the RWA switch is a great - it use a DACT 4 position selector, and has a mute (non-selected) feature. Very good and robust build.

 Regards,
 F. Lo


----------



## integrale

Quote:


  Originally Posted by *Bootleg* /img/forum/go_quote.gif 
_BOOOOM!

 Rivendell, that is one heck of a second post!

 Thanks!_

 

Makes my first post pale in comparison 
	

	
	
		
		

		
			





 Thanks, Rivendell.

 I think, in the end, I'll keep the DAC1 USB...even though I could save $600+ by buying a used DAC1. The USB input is useful in that it frees up the BNC input, and I like the security of having a new product. And $100 of the cost was towards sales tax, which helps support those troubled NY schools...

 This one's for the kids.


----------



## puntloos

Quote:


  Originally Posted by *integrale* /img/forum/go_quote.gif 
_Hi All,

 -My motherboard has S/PDIF RCA out, and I can hear absolutely no difference between this and the USB input. EXCEPT: when I plug my iPod into my computer and it is doing a heavy sync via USB, the PC (brand new top-shelf HP media center) struggles to stream consistently and the sound gets choppy and I can't listen to it. I imagine this is true of using an external USB HDD as well. So I'm currently only using the digital output from my motherboard (anyone want an X-Fi XtremeMusic?)_

 

Very true, it amazes me how many modern mainboards STILL haven't worked out this type of problems. Just using an USB HDD and indeed suddenly my (USB) mouse feels sluggish. Sigh. 
  Quote:


 Outstanding product, and I recommend it to everyone who asks. 
 

Agreed. (get the Bel Canto if you want the same stuff, but prettier looking and a remote 
	

	
	
		
		

		
		
	


	




 )


----------



## EliasGwinn

Quote:


  Originally Posted by *JimP* /img/forum/go_quote.gif 
_1. run RCA out to amp 1, and use an XLR to RCA converter simultaneously out to amp 2 (can this be done, any issues?)_

 

This works fine - just as good as two RCA outputs. BUT, it is important to leave pin 3 'floating'. In other words, when you buy or make an XLR to RCA, it is VERY important to have pin 3 unconnected. Some adapters will connect pin 3 to pin 1 (or ground/shield), and this is VERY WRONG for the DAC1 (this is for impedance-balanced outputs, not 'active-balanced' outputs). So, just make sure pin 3 is NOT connected to anything, and you will be fine.

 The output levels will be different between the XLR outputs and the RCA outputs, so don't be surprised when one is louder then the other (the difference between the two depends on where the output attenuators are set). 

 Thanks,
 Elias


----------



## wakeride74

Has anyone tried the DAC1 with some Grado PS-1's and a tube amp like a SP or RSA Raptor, Woo, etc.? It seems like the detail of the DAC1 would mate well with the warmth of the PS-1's.


----------



## fc911c

Quote:


  Originally Posted by *wakeride74* /img/forum/go_quote.gif 
_Has anyone tried the DAC1 with some Grado PS-1's and a tube amp like a SP or RSA Raptor, Woo, etc.? It seems like the detail of the DAC1 would mate well with the warmth of the PS-1's._

 

I don't have PS-1's wish I did, but I have at the monent a SP Extreme Platinum and MS2. In a few days my RS1's and RS2's will be hear if that would be any help to you?

 Frank


----------



## wakeride74

Quote:


  Originally Posted by *fc911c* /img/forum/go_quote.gif 
_I don't have PS-1's wish I did, but I have at the monent a SP Extreme Platinum and MS2. In a few days my RS1's and RS2's will be hear if that would be any help to you?

 Frank_

 

Probably not but thank you for the response. The PS-1's are a very different breed and do not sound like any other Grado. The GS1000 would probably be the closest and they are still quite far. The Denon D5000's are somewhat similar but again the PS-1's are just a very unique animal.

 I've found that the PS-1's just hit a sweet spot for me so I'm probably going to offload my Stello and a couple other things to find a pair. Based on what I've read about the DAC1 it seems that the two would make a good match but I'd really like to hear from someone that has heard them together... even if only in a meet condition.


----------



## Bootleg

Thanks Elias...I have been thinking about doing the same thing to feed a Zone 2 amp...

 So does that mean, if pinned out correctly and levels were matched, that there should be no sonic differences between the two?

 How about driving both at the same time?

 How about the long term reliability effects on the Benchmark?

 Mahalo!


----------



## EliasGwinn

Quote:


  Originally Posted by *Bootleg* /img/forum/go_quote.gif 
_Thanks Elias...I have been thinking about doing the same thing to feed a Zone 2 amp...

 So does that mean, if pinned out correctly and levels were matched, that there should be no sonic differences between the two?

 How about driving both at the same time?

 How about the long term reliability effects on the Benchmark?

 Mahalo!_

 


 There will be no sonic differences whatsoever. They use the exact same circuit, only the balanced outputs have more gain and dual active lines (for true balanced output).

 There are no problems driving them both at the same time, and there will be no difference in the long-term performance and/or expected lifetime.

 In other words....go for it!!

 Thanks,
 Elias


----------



## Bootleg

Thanks Elias!

 Still can't help but be insecure over a "floating" pin, but if you say its good, I'll go for it!


----------



## EliasGwinn

Quote:


  Originally Posted by *Bootleg* /img/forum/go_quote.gif 
_Thanks Elias!

 Still can't help but be insecure over a "floating" pin, but if you say its good, I'll go for it!_

 

Fear not, my friend. It is no more of a problem then running the DAC1 with no XLR's connected at all. 

 The real danger is when pin 3 is connected to ground, as is often the case with XLR-to-RCA connectors. This can significantly shorten the life of an output driver because it is shorting the output directly to ground; that is, it is driving a 0-ohm load.

 Thanks,
 Elias


----------



## wakeride74

I've read that the XLR outs on the DAC1 sound better than using the RCA outs, is there any truth to this? Also if there is a SQ improvement using the XLR's would I need to do any mods to use these XLR to RCA adaptors when connecting it to a separate headphone amp or home receiver?


----------



## EliasGwinn

Wakeride74,

 If your driving single-ended (unbalanced), the XLR output and RCA output will perform identically (except for differences in gain / output level). They are driven using identical circuits (again, except for gain), and have identical output impedances (30 ohm for RCA; 30 ohm for each 'pin' of XLR when no attenuation is used). If your amp requires the use of attenuation on the XLR outputs (which most do), then the output impedance of the XLR will be increased, and the RCA will have a slight advantage at that point (although the performance of the two will remain the same, the slightly higher output impedance may interact with cable capacitiance, etc). 

 As for the Cardas adapters, they don't specify the internal connections, so I can't say if they're wired correctly or not. Not even the Cardas webpage talks about the wiring (its strange what important details some manufacturers fail to list on their product webpages and manuals!!) 

 The important thing to know about XLR-to-RCA adapters is that pin-3 must NOT be connected to ground (nor should the be connected to anything else for that matter). The XLR output of the DAC1 is a true 'active balanced' output - that is, each leg (+ and -) are actively driven with independent opamps. If pin-3 were connected to ground, this short-circuit can shorten the life of the output driver. 

 XLR-to-RCA adapters that have pin-3 connected to ground are for 'impedance balanced' outputs, which have no signal on pin-3. 

 Thanks,
 Elias


----------



## wakeride74

Good to know, thanks for the info! Now I don't have to buy those XLR to RCA's.


----------



## wakeride74

Ok, I have another question re: using the DAC1 with my stereo/speaker rig. I know it will improve SQ for music and probably movies too but if I run a optical out of my Toshiba HD-A2 to the DAC1 and then RCA's out of the DAC1 to my Denon receiver (which is usually set to auto select highest SQ from the source, so in the case of HD-DVD's that would be DTS or Dolby 5.1 for most SD) will the receive still play the 5.1 sound from the Toshiba or will having the DAC1 in the chain put everything into 2 channel?


----------



## EliasGwinn

Wakeride74,

 I'm not familiar with the operations of this receiver. Which model is it?

 Thanks,
 Elias


----------



## wakeride74

Quote:


  Originally Posted by *EliasGwinn* /img/forum/go_quote.gif 
_Wakeride74,

 I'm not familiar with the operations of this receiver. Which model is it?

 Thanks,
 Elias_

 

AVR-3801

 I'm assuming that the DVD player is sending a sound over as 5.1 but I'm not sure if having the DAC1 in the chain will put things back into 2 channel, in which case the receiver may not recognize the signal as a 5.1 feed???


----------



## EliasGwinn

So... You're routing the digital optical signal from the Toshiba to the DAC1 only? If so, the Denon receive will never know if the signal was meant to be 5.1 or not. In other words, the Denon looks for digital signal, and if none are found, it will look for analog signals. If the digital out from the Toshiba is going to the DAC1 only, the 5.1 digital audio will never make it to the Denon. 

 I hope that helps.

 Thanks,
 Elias


----------



## wakeride74

Quote:


  Originally Posted by *EliasGwinn* /img/forum/go_quote.gif 
_So... You're routing the digital optical signal from the Toshiba to the DAC1 only? If so, the Denon receive will never know if the signal was meant to be 5.1 or not. In other words, the Denon looks for digital signal, and if none are found, it will look for analog signals. If the digital out from the Toshiba is going to the DAC1 only, the 5.1 digital audio will never make it to the Denon. 

 I hope that helps.

 Thanks,
 Elias_

 

I actually don't have a DAC1 yet but the Toshiba would go optical out to the DAC1 and then the DAC1 would go RCA out to the Denon (since the DAC1 does not have a digital out).


----------



## EliasGwinn

You will need to use an optical switcher and/or 1-to-2 adapter (or just manually switch the optical cable to the Denon) when you want to do 5.1 playback. 

 Or, you can buy another cheap DVD player for 2-channel playback through the DAC1 and use the Toshiba for video playback and 5.1.

 Thanks,
 Elias


----------



## sangel

Dear Elias,

 For the last 3 years I am enjoyning a DAC1, with another power amp, with very good results.
 After upgrading the power amp due to its driving capability, 
 now, I have introduced another power amp within my current audio setup.
 I drive the DAC1 directly to the power amp in a differential mode (XLR).
 My DAC1 current settings are:
 1. Attenuation pad at ''0''
 2. Use of the variable output and 60 Ohm output impedance.
 3. XLR I/O cable 1 m length. (braided shielding of silver plated copper and housed in Teflon
 double insulated)

 Now, the question is that my power amp's input impedance is very low, ie 2.2 kohms
 and I am not sure if the Classic DAC1 drivers are suitable to drive this very low load.
 To be honest, up to moderate listening level - just loud enough - the sound is very good, (pot between 0 - 12)
 althougth, I am afraid to raise the volume due to any distortion / overload might occur as
 a result to damage my speakers (ie- magnepan 3.6.....tweeter is very sensitive).

 Here are the specs I have found for my Amp (word.doc. document)

 Finally, using the usb Dac1 with the new drivers, is it worth it, or not. 

 Also, another issue regarding the variable operation of the DAC1 is that when I turn the pot clockwise
 to increase the gain, then for the first 2 turns (klicks), the gain of one channel is somehow lower than the other.
 (Remark: To be confirmed on the above, I have swiched the amp channels, and the issue remains the same).
 Thanks in advance for your kind response

 My very best from Athens Greece


 Sangel 

 PS. I've choosen the amp based of its performance with my speakers.
 Introducing the DAC1 = ECXELLENCE

 Here are the specs I have found for my Amp 

 PBN Denali 
 Stereo Amplifier 
 Technical Spécifications:

 STEREOMONO
 Power Output into 8 Ohms:2 x 160 W1 x 600 W
 Power Output into 4 Ohms:2 x 320 W1 x 1200 W
 Power Output into 2 Ohms: 1 x 1900 W
 Input Impedance: RCAInput Impedance: XLR100 KOhms 2.2 KOhms
 Input Voltage to drive the amplifier to clipping:1.5 Vrms
 Voltage Gain:26 db
 Power Supply Capacitance:170,000 µF
 Power Transformer(s):1 x 2260 VA 
 Dynamic Headroom db:0.5 db
 Total Harmonic Distortion:<0.001%
 Slew Rate70 volts / µ second
 Power consumption:400 W Idle, 2kW Max
 Dimentions: 11"H, 19"W, 16"D
 Weight: 115 lbs


----------



## EliasGwinn

Sangel,

 In the spec's for this amp, notice this line:

 Input Voltage to drive the amplifier to clipping: 1.5 Vrms

 This is equivilent to 5.74 dBu. The design of this amp is a perfect example of why we include output attenuators in the DAC1. If a signal over 5.8 dBu will clip this amp, it has far too much gain on the front end. 

 An ideal signal path would keep the signal amplitude as high as possible from source device to receiving device. The reason for this is to keep the signal-to-noise ratio maximized. The DAC1 is built to drive up to +29 dBu analog signal levels (professional levels). This keeps the signal well above the noise floor. However, this would obviously clip this amp in question, so it must be attenuated.

 If the 10 dB attenuators are used in the DAC1, we can avoid clipping this amp through the first 20 pot detents (12 o'clock). The 20 dB attenuators will allow us the entire pot rotation without clipping this power amp (the max analog output of the DAC1 with the 20 dB attenuators is near 5.5 dBu).

 The input impedance of this amp (2.2k) is much lower then it should be. This is where the advantages of the high-current output drivers of the DAC1 USB are realized. With the attenuators activated, the output impedance of the DAC1 USB will be almost 75% less then that of the DAC1. This is especially important when driving a low impedance input stage. Even without the attenuators engaged, the DAC1 USB output drivers will be much less susceptible to distortion because of their low-impedance drive capability.

 The potentiometer of the DAC1, just as with any 2-channel potentiometer that goes to full-off, experiences inter-channel level inaccuracies just above the 'full-off' position. This is because it is so close to 'full-off', the end of the range where the wiper disconnects from the resistive element. This part of the range is not meant to be used during listening, it is meant to achieve 'full-off' volume only. Over the rest of the rotation range of the pot (the listening portion), the channels are matched within 0.1 dB of each other. 

 If your amp is fixed gain, you will want to adjust the DAC1 output attenuators so that a comfortable listening volume range is achieved in the later portion of the pot rotation.

 Thanks,
 Elias


----------



## sangel

Hello Elias,

 Thank you very much for your kind and proffesional description.
 Athought, I have 2 remarks, noticed after very careful listening.
 1. DAC1 setting at ''0'' attenuation sounds, to me, better (more detail) than with the attenuators ''on''.
 2. Also, at ''0'' attenuation setting, and setting the pot at almost 2'oclock,
 the amp is still very stable - no clipping - which seems that the signal is over 5.8 dBu (Have not tried to raise the volume more than that). I think input sensitivity of the amp and clipping are frequency dependent..
 Intrducing the attenuators, then the output impedance of the dac1 raises as a result for a mismatch with the amp's input load (2.2K).

 Based of the above, do you recommend to to give a try for the DAC1 usb, according to your previous description!! (Better drive + better impedance matching with att's) OR its not woth it.

 Thank you again.

 Best

 Sangel


----------



## EliasGwinn

Quote:


  Originally Posted by *sangel* /img/forum/go_quote.gif 
_...do you recommend to to give a try for the DAC1 usb, according to your previous description!! (Better drive + better impedance matching with att's) OR its not woth it._

 

It probably will make a big difference, but I really can't say for sure because it depends on how the input stage is designed (specifically the level of input capacitance). What is the make and model of your amp?

 If there is an issue (high input capacitance) with the input stage, the output drivers of the DAC1 USB will be much more suitable in driving it properly. And you wouldn't believe how many 'audiophile' amplifiers are built with high input capacitance (to keep RF noise out). So, I wouldn't be surprised if the DAC1 USB was much better suited for your amp. And, if you're in the U.S. and get the DAC1 USB directly from us, you can try it for 30 days to see if it makes a difference. If there are no differences, just return before 30 days for a refund.

 Thanks,
 Elias


----------



## sangel

Hello Elias,

 My amp is a Sierra Audio Denali - true differential(Now PBN / Olympia Amps + Montana Speakers).
 To be honest the DAC1 is an exchellent match with this amp - having tried out quite expensive preamps..- this minimum system setup comes to be great, competitive and cost effective.
 I do not know if the pot is quite good or is worth of replace it with someting else.
 Elias, does the output impedance of the dac1 30+30 Ohms (xlr) through all the steps of the pot!! ie remains unchanged!!
 About the dac1 usb I mifght ask the local dealer here in Athens to borrow one for A/B comparison.

 Best


 Sangel


----------



## EliasGwinn

Sangel,

 PBN is mailing me a manual for the amp. I'm very curious to see their recommended gain configuration...

 The volume pot on the DAC1 does not affect the output impedance whatsoever. With 0dB attenuation, it is 60 ohms (30 + 30) output impedance at all positions of the volume knob.

 With regards to the quality of the DAC1's volume pot, it does not compromise the sound quality whatsoever. As we have discussed, the balance between channels is better then 0.1 dB, except when it is near 'full-off' (below 9 o'clock). Also, when it is above 9 o'clock, the performance is equal to when the volume control is bypassed (calibrated mode). In other words, the volume pot does not affect the sound quality at all. 

 Thanks,
 Elias


----------



## euclid

hi again Elias, 

 i am happy to say i received the DAC1 and so far am very happy with the sound quality. one problem im having is that i have almost no useable range on the volume pot when using the headphone amplifer. even with 300ohm Sennheiser HD650 i can barley get past 9:00 on the pot. i know i could upgrade to the DAC1 USB and have the gain jumper but it seems overkill to spend an additional $300 for this feature alone. 

 so im wondering if the DAC1 USB stronger XLR output drivers would be beneficial in my setup too. my current DAC1 is feeding a Headamp GS-X headphone/preamp in another room. i have the attenuation set to 20db in the DAC1 which means 500ohm output. the input impeadance of the GS-X preamp is labeled 50k ohms, would i see any benefit in lower distortion by using the DAC1 USB if i am attenuating that by 20db too? thanks


----------



## EliasGwinn

Euclid,

 Its difficult to say whether the DAC1 USB's output drivers will affect the quality of sound, because it depends on the input capacitance of the GS-X, as well as the type and design of the input stage (i.e., discrete, transformer, etc). They don't offer any of this information on their website (actually, I couldn't find any information at all...only pictures...???).

 It is safe to assume that the DAC1 USB is capable of handling any 'adverse' driving requirements. But, I can't tell you with certainty that this particular device would benefit. (But I will say that you might be suprised to learn how many 'audiophile' devices are designed with extremely large input capacitance to ward off any RF noise at the input. Although it may be effective at filtering RF, it can cause serious high-end distortion).

 Hope that helps!

 Thanks,
 Elias


----------



## zkn

Quote:


  Originally Posted by *euclid* /img/forum/go_quote.gif 
_i am happy to say i received the DAC1 and so far am very happy with the sound quality. one problem im having is that i have almost no useable range on the volume pot when using the headphone amplifer. even with 300ohm Sennheiser HD650 i can barley get past 9:00 on the pot. i know i could upgrade to the DAC1 USB and have the gain jumper but it seems overkill to spend an additional $300 for this feature alone._

 

really? i was hoping upgrading to 300 ohm hd650s would fix this issue for me but apparently not? how come there's such a design fault with the older model?

 my genelec monitors seem to be extremely hot as well from this preamp: gain set to half on speakers, attenuators to -30dB and the max volume is reached at 12:00

 does the attenuator settings affect the headphone out?


----------



## EliasGwinn

zkn,

 The gain level of the headphone outputs were purposely designed to accommodate professional applications where analytical monitoring is extremely important. 

 The output attenuators do not affect the headphone outputs. 

 Which Genelec monitors are you using?

 Thanks,
 Elias


----------



## zkn

8030a+7050b

 I'm forced to run the sub through speakers because of this. If run through the subwoofer first, its volume is fixed to the maximum and the gains on the speakers don't have affect it. It's no problem though but the headphone out volume pot issue is kind of a bummer. 

 Can anyone else confirm the position of the pot with stock HD650 unbalanced through headphone out?


----------



## EliasGwinn

quoted from Genelec 8030a manual:

  Quote:


 The input sensitivity of the loudspeakers can
 be matched to the output of the mixing console
 or other source by adjusting the volume
 control on the front panel 
 

The volume control (gain) of the 8030's should be turned all the way down to accommodate the maximum signal amplitude from the DAC1. It is highly recommended to configure in this manner because maintaining maximum signal amplitude from the DAC1 to the amplifier/speaker will maximize the signal-to-noise ratio and minimize distortion in the amp. This is true from any source to any load. This is standard practice in professional recording and broadcasting facilities for this reason.

 Thanks,
 Elias


----------



## Lord Chaos

The headphone output of the DAC1 is very strong. My DAC1 USB drives Shure E500s, and by the time I get to 9:00 on the volume pot my ears are being turned inside out. In the bedroom I have a standard DAC1 driving Sennheiser HD25s, and the same applies. Fortunately the pot is even better balanced than Elias has said, so it's not a big problem. The most frustrating aspect is that the steps in volume are large at that point, so finding the volume I want is a challenge. Sometimes I end up having the music louder than I'd really prefer.

 I'd like to see the Benchmark folks look into this, and perhaps offer a front-panel output attenuator switch that could be set to -10dB, -20dB and -30dB. This would accommodate high-efficiency headphones.

 Off-topic: a headphone board, and its spell-check program doesn't know Sennheiser?


----------



## euclid

Quote:


  Originally Posted by *EliasGwinn* /img/forum/go_quote.gif 
_Its difficult to say whether the DAC1 USB's output drivers will affect the quality of sound, because it depends on the input capacitance of the GS-X, as well as the type and design of the input stage (i.e., discrete, transformer, etc). They don't offer any of this information on their website (actually, I couldn't find any information at all...only pictures...???).
_

 


 the GS-X info is not posted on the Headamp website but it is a balanced version if the GS-1 which is the 50k OHM impedance spec i quoted.

http://www.headamp.com/home_amps/gs1/index.htm

 i am going to contact Headamp to find out the exact specs on the GS-X and hopefully get that capacitance figure for you.

 regarding the DAC1 headphone out, i am using my computer media software as Benchmark suggests, Winamp and Foobar set to 100% and 100% on Windows volume. whith this setting my 42ohm Audio Technica W5000 and AD2000 can get to about step 4 or 5 which is just past the point of the right channel kicking in after the initial pot imbalance, so i have virtually no volume control. the Senn HD650 which are 300 OHM will get to about 9:00 or step 10, i cant detect any imbalance at that setting but as pointed out the volume steps are fairly high. 

 i could always lower the volume in software but i dont want to introduce digital distortion as pointed out by Elias earlier.

 how many additional volume steps can i expect from the 10db down setting on the DAC1 USB?


----------



## gregeas

Quote:


  Originally Posted by *EliasGwinn* /img/forum/go_quote.gif 
_quoted from Genelec 8030a manual:

 The volume control (gain) of the 8030's should be turned all the way down to accommodate the maximum signal amplitude from the DAC1. It is highly recommended to configure in this manner because maintaining maximum signal amplitude from the DAC1 to the amplifier/speaker will maximize the signal-to-noise ratio and minimize distortion in the amp. This is true from any source to any load. This is standard practice in professional recording and broadcasting facilities for this reason._

 

Interesting. At work I have Genelec 8020As driven by a USB DAC1, and I dropped the internal XLR jumpers way down so I could keep the volume on the 8020As at the max level. The reason I did this is that the volume pots on the speakers are not stepped, so it's a hassle to balance them each time I power on the Genelecs. Is it a mistake to drive them with their volume at max? 

 I also use my USB DAC1 with e500s, and like others can't get past 9:00... But I've not noticed any sort of channel imbalance. It would be nice to have more usable range, however. 

 Finally, when outputting from the USB DAC1's RCAs to my headphone amp (Triad Audio Lisa III XP), I can't get the volume control past 9:00 when driving my Senn HD650s. My e500s don't even stand a chance. There is no way to attenuate the RCA output, correct? I suppose I could have the gain on the head amp dropped down.


----------



## wakeride74

So the the Output Level switch on the rear has three positions; down is variable, up is calibrated and middle is ???

 What should this be switched to when:

 1. Using the headphone out of the DAC1?

 2. Using the RCA outs to a dedicated headphone amp or receiver (so as to bypass the volume control on the DAC1)?


----------



## zkn

Quote:


  Originally Posted by *EliasGwinn* /img/forum/go_quote.gif 
_The volume control (gain) of the 8030's should be turned all the way down to accommodate the maximum signal amplitude from the DAC1._

 

turned all the way down? there would be no sound then

 do you mean maxed out?


----------



## EliasGwinn

Quote:


  Originally Posted by *Lord Chaos* /img/forum/go_quote.gif 
_I'd like to see the Benchmark folks look into this, and perhaps offer a front-panel output attenuator switch that could be set to -10dB, -20dB and -30dB. This would accommodate high-efficiency headphones._

 

Thanks for the feedback...it is extremely valuable to us. We have discussed this and will definitely take it into consideration when planning future products and services. 

 And, please don't think that I'm just saying some generic PR b.s. I'm very serious when I say this because you all (head-fi community) have contributed so much to our understanding of what the user wants and needs. For example, we put the 10 dB gain reduction in the headphone amp of the DAC1 because of user feedback. 

 And now we are finding out that the gain needs to be reduced even further. I'm not sure when we can provide a product or service to remedy this situation, but we will look into it with complete earnestness.

 Thanks,
 Elias


----------



## EliasGwinn

Quote:


  Originally Posted by *euclid* /img/forum/go_quote.gif 
_i could always lower the volume in software but i dont want to introduce digital distortion as pointed out by Elias earlier._

 

With Windows, you can use the system volume control mixer (the gray mixer with the faders). It causes very little (almost no) distortion. However, as with any digital volume control, you will lose word-length as you decrease the volume. That is, every 6 dB of digital attenuation results in 1-bit decrease. This is why we see our analog volume control as a big advantage over D/A's with digital volume control.

  Quote:


  Originally Posted by *euclid* /img/forum/go_quote.gif 
_how many additional volume steps can i expect from the 10db down setting on the DAC1 USB?_

 

The HPA2 (headphone amp) volume of the DAC1 USB w/ 10 db gain reduction is approximately 8 steps below that of the DAC1 HPA2 w/ no gain reduction. In other words, with the 10 dB HPA2 gain reduction, you can turn up the volume knob +/- 8 more steps before you reach the same volume as with no gain reduction.

 Thanks,
 Elias


----------



## EliasGwinn

Quote:


  Originally Posted by *gregeas* /img/forum/go_quote.gif 
_Interesting. At work I have Genelec 8020As driven by a USB DAC1, and I dropped the internal XLR jumpers way down so I could keep the volume on the 8020As at the max level. The reason I did this is that the volume pots on the speakers are not stepped, so it's a hassle to balance them each time I power on the Genelecs. Is it a mistake to drive them with their volume at max?_

 

Well, good question... This, along with this point made by zkn:

  Quote:


  Originally Posted by *zkn* /img/forum/go_quote.gif 
_turned all the way down? there would be no sound then

 do you mean maxed out?_

 

Let me ask you both... Do these Genelec's go to 'full-off' when you turn the volume all the way down? 

 The reason I ask: if they go all the way to 'full-off', chances are that they are attenuator controls, not gain controls. In other words, the gain may be fixed, and the front panel knobs may only attenuate the incoming signals. 

 I'm curious to find out...

 Thanks,
 Elias


----------



## EliasGwinn

Quote:


  Originally Posted by *gregeas* /img/forum/go_quote.gif 
_...when outputting from the USB DAC1's RCAs to my headphone amp (Triad Audio Lisa III XP), I can't get the volume control past 9:00 when driving my Senn HD650s. My e500s don't even stand a chance. There is no way to attenuate the RCA output, correct? I suppose I could have the gain on the head amp dropped down._

 

The RCA outputs of the DAC1 (USB) are configured to match that of most common consumer audio devices, such as CD players, etc. The best thing to do, in your case and in general, is to minimize the gain on the downstream device to allow maximum signal amplitude from the source.

 Another thing you can do, if you want, is run the DAC1 in calibrated mode and adjust the calibration potentiometers in the back of the DAC1 to suit your headphone amp. To do this, you will want to use a digital test tone (I can post one to use via the computer, or you can buy a test CD) and a multimeter to make sure the balance between channels is accurate.

 Thanks,
 Elias


----------



## EliasGwinn

Quote:


  Originally Posted by *wakeride74* /img/forum/go_quote.gif 
_So the the Output Level switch on the rear has three positions; down is variable, up is calibrated and middle is ???_

 

The middle position will mute all analog outputs except the HPA2 (headphone) outputs. This is to allow you to mute your speaker system when you want to listen solely via headphones. The DAC1 USB will allow you to do that simply by plugging the headphones in the left output of the HPA2 - the other analog outputs will mute automatically.

  Quote:


  Originally Posted by *wakeride74* /img/forum/go_quote.gif 
_What should this be switched to when:

 1. Using the headphone out of the DAC1?_

 

See above.

  Quote:


  Originally Posted by *wakeride74* /img/forum/go_quote.gif 
_2. Using the RCA outs to a dedicated headphone amp or receiver (so as to bypass the volume control on the DAC1)?_

 

The 'Calibrated' position of the switch will bypass the volume control of the DAC1.

 Thanks,
 Elias


----------



## gregeas

Quote:


  Originally Posted by *EliasGwinn* /img/forum/go_quote.gif 
_Well, good question... This, along with this point made by zkn:

 Let me ask you both... Do these Genelec's go to 'full-off' when you turn the volume all the way down? _

 

Yes, I just tested my Genelecs. With their volume set at 0, at the DAC1 maxed, I hear nothing out of the speakers.


----------



## zkn

I believe the 8020 have the on/off integrated on the gain control pot and my 8030 have on/off switch separated.

 Either way, there's absolutely no sound with 8030 either when power on but gain at zero.


----------



## EliasGwinn

Ok, that confirms that they are passive attenuators, not gain adjustment pots, which also indicates invariable gain....too bad. 

 It is still better to keep the inter-component signal amplitude high, but to do achieve inter-channel balance, you'll need a SPL meter (or measurement mic and analysis hardware/software) to adjust the Genelec volume pots. Otherwise, you can turn them all the way up, as Lord Chaos mentioned, and keep the DAC1 at a minimum. 

 This is a perfect example of why amps should not have too much gain (if any at all) built into the front end - or at least have it adjustable. With the -30 dB attenuators in the DAC1, the peak output amplitude is well below the amplitude of almost all consumer level devices. Yet, it is still to hot for these Genelecs? 
	

	
	
		
		

		
			





 Thanks,
 Elias


----------



## zkn

Oh well. At least when the monitor pots are maxed out the DAC1 pot remains about at the same level (9:00) as it will likely be with the HD650s so there's no need to adjust when changing from one to another.

 Now only if it automuted like the new revision.


----------



## sangel

Hello Elias,

 Thanks again.
 Please, let me know if you can send or fax to to me a copy of the PBN Denali Amp. My fax no is 0030 210 6178910. (This is because I have been gotten the amp from a second hand owner who didn't had the manual).

 Keep in touch.

 My very best

 sangel


----------



## Terje

Hello Elias,

 Is there any good idea to connect an active subwoofer to the rca outputs, if you have active monitors 
http://www.quad-hifi.co.uk/model.php...=5&model_id=23 connected to xlr out ? 
 Will DAC1 filter the right waves to sub and speakers ?
 If not, do you have any suggestion on how to set up an active subwoofer where the monitors also are active(xlr out to xlr in amp), in use with DAC1 ?

 Can you recommend any good analog xlr - xlrm cables ?

 What amplifier do you use to drive JBL LSR6332 ? 

 Regards
 Terje


----------



## EliasGwinn

Quote:


  Originally Posted by *sangel* /img/forum/go_quote.gif 
_Hello Elias,

 Thanks again.
 Please, let me know if you can send or fax to to me a copy of the PBN Denali Amp. My fax no is 0030 210 6178910. (This is because I have been gotten the amp from a second hand owner who didn't had the manual).

 Keep in touch.

 My very best

 sangel_

 

Send an email to me (elias(at)benchmarkmedia.com) and i'll reply with the manual attached. The 'manual' that they sent me is only a two-sided sheet....its more like a cut sheet. It doesn't offer too much info, especially what I really wanted to know, which is 'input sensitivity'. 

 Thanks,
 Elias


----------



## zkn

Quote:


  Originally Posted by *zkn* /img/forum/go_quote.gif 
_Can anyone else confirm the position of the pot with stock HD650 unbalanced through headphone out?_

 

To answer myself now that I have received the sennheisers:

 Unfortunately even at 300 ohm, the max listening volume with HD650 is reached before 09:00 with regular DAC1 and the steps with volume pot are just too huge for finetuning.

 Very disappointed by this as I don't want a separate amplifier.

 How's the volume adjustment when driving HD650 through balanced out?
 Not that I want to do this, since that's reserved for monitors, just curious.


----------



## EliasGwinn

Quote:


  Originally Posted by *Terje* /img/forum/go_quote.gif 
_Hello Elias,

 Is there any good idea to connect an active subwoofer to the rca outputs, if you have active monitors 
http://www.quad-hifi.co.uk/model.php...=5&model_id=23 connected to xlr out ? 
 Will DAC1 filter the right waves to sub and speakers ?
 If not, do you have any suggestion on how to set up an active subwoofer where the monitors also are active(xlr out to xlr in amp), in use with DAC1 ?

 Can you recommend any good analog xlr - xlrm cables ?

 What amplifier do you use to drive JBL LSR6332 ? 

 Regards
 Terje_

 

If the subwoofer has a high-passed output, you can drive your active monitors directly from that. Otherwise, you will have to use a line-level crossover to split the low signal for the sub and the high signal for the monitors. 

 The DAC1 does not have crossover filtering.

 We recommend (and use) Canare StarQuad analog balanced cables with Neutrik connectors. They are both very high quality, sonically and mechanically, and not over-priced. We have our cables built by Have, Inc (www.haveinc.com).

 I drive my JBL LSR32's with a Hafler PRO1200. It is perhaps slightly underpowered, but very capable.

 Thanks,
 Elias


----------



## euclid

Quote:


  Originally Posted by *zkn* /img/forum/go_quote.gif 
_To answer myself now that I have received the sennheisers:

 Unfortunately even at 300 ohm, the max listening volume with HD650 is reached before 09:00 with regular DAC1 and the steps with volume pot are just too huge for finetuning.

 Very disappointed by this as I don't want a separate amplifier.

 How's the volume adjustment when driving HD650 through balanced out?
 Not that I want to do this, since that's reserved for monitors, just curious._

 

i am ordering a DAC1 USB this week to see how much usable range is gained on the headphone out. $300 is a substantial upgrade cost for such a minor feature but i really dont want to run a seperate headphone amp for this setup. ill let you know how it compares to the standard DAC1


----------



## EliasGwinn

Head-Fi folks,

 Since some of you have mentioned that the headphone volume is higher then you wish it was, we're looking into offering a mod for the DAC1 USB to decrease the gain of the headphone circuit. Unfortunately, we cannot mod the DAC1 Classic because there is no circuit to adjust the gain range, as there is with the DAC1 USB.

 We are trying to determine how much lower the gain should be. So, if anyone could provide feedback about their experience with the DAC1 (Classic or USB) and the headphone volume control, it would be greatly appreciated. 

 Specifically, we would like to know what the typical usable range is. What is the maximum, and what is the average placement of the volume control? 

 We will also need to figure out a mod price that will be no more then it absolutely needs to be to cover operating costs (the receive/repair/ship process requires a time investment from a few employees). We will hopefully be able to offer Head-Fi members a special discounted price since your feedback has been invaluable to us. I will let you know as we proceed.

 Thanks,
 Elias


----------



## euclid

Quote:


  Originally Posted by *EliasGwinn* /img/forum/go_quote.gif 
_Head-Fi folks,

 Since some of you have mentioned that the headphone volume is higher then you wish it was, we're looking into offering a mod for the DAC1 USB to decrease the gain of the headphone circuit. Unfortunately, we cannot mod the DAC1 Classic because there is no circuit to adjust the gain range, as there is with the DAC1 USB.

 We are trying to determine how much lower the gain should be. So, if anyone could provide feedback about their experience with the DAC1 (Classic or USB) and the headphone volume control, it would be greatly appreciated. 

 Specifically, we would like to know what the typical usable range is. What is the maximum, and what is the average placement of the volume control? 

 We will also need to figure out a mod price that will be no more then it absolutely needs to be to cover operating costs (the receive/repair/ship process requires a time investment from a few employees). We will hopefully be able to offer Head-Fi members a special discounted price since your feedback has been invaluable to us. I will let you know as we proceed.

 Thanks,
 Elias_

 

that sounds great,

 Elias i sent you Private Message.


----------



## fishski13

Elias,
 i'm looking to utilize both RCA and XLR outputs from my DAC1. is there an XLR to RCA adapter that you would recommend? thanks!

 I also appreciate the generous time you have invested in this thread. by the tone of your voice when i spoke to you on the phone, i can tell you're a pretty passionate dude. 

 a happy DAC1 owner,
 PACE


----------



## sangel

Thanks Elias. OK.

 have a nice day.

 Sangel


----------



## EliasGwinn

Quote:


  Originally Posted by *fishski13* /img/forum/go_quote.gif 
_Elias,
 i'm looking to utilize both RCA and XLR outputs from my DAC1. is there an XLR to RCA adapter that you would recommend? thanks!

 I also appreciate the generous time you have invested in this thread. by the tone of your voice when i spoke to you on the phone, i can tell you're a pretty passionate dude. 

 a happy DAC1 owner,
 PACE_

 

Thanks PACE!

 For a high quality XLR-to-RCA cable, call HAVE, Inc and tell them you want: 

 "2 unbalanced cable assemblies made with Canare L-4E6S cable, each terminated with
 - NEUNC3FXB with pin-3 floating (not connected)
 - NEUNYS373 on the other end."

 If they have any questions, you can have them call me (our phone number is in my signature below).

 Thanks,
 Elias


----------



## fishski13

Quote:


  Originally Posted by *EliasGwinn* /img/forum/go_quote.gif 
_Thanks PACE!

 For a high quality XLR-to-RCA cable, call HAVE, Inc and tell them you want: 

 "2 unbalanced cable assemblies made with Canare L-4E6S cable, each terminated with
 - NEUNC3FXB with pin-3 floating (not connected)
 - NEUNYS373 on the other end."

 If they have any questions, you can have them call me (our phone number is in my signature below).

 Thanks,
 Elias_

 

thanks, but i have DIN only inputs on my Naim Audio equipment. i was hoping to try an XLR-RCA adapter so that i don't need to change out my cabling. i'll try contacting HAVE, Inc to see what they can do for me.

 PACE


----------



## euclid

Cardas, Tara Labs, and Neutrik make Female XLR to Female RCA adapters, Neutrik is least expensive by a long shot.

http://www.dedicatedaudio.com/inc/sdetail/542
http://tara-direct.com/results.asp?ID=8
http://www.markertek.com/Product.asp...&search=0&off=


----------



## fishski13

Quote:


  Originally Posted by *euclid* /img/forum/go_quote.gif 
_






 Cardas, Tara Labs, and Neutrik make Female XLR to Female RCA adapters, Neutrik is least expensive by a long shot.

http://www.dedicatedaudio.com/inc/sdetail/542
http://tara-direct.com/results.asp?ID=8
http://www.markertek.com/Product.asp...&search=0&off=_

 

thanks! i know that an adapter isn't ideal but i'll suck it and see for myself. currently, i am using only the RCA outs from the DAC1 into my Naim integrated amp with my headphone amp connected to the line out of the line stage. i would like to utilize the XLR out on the DAC1 to feed the signal directly to my headphone amp, bypassing the active line stage, hopefully yielding sonic gains.

 PACE


----------



## EliasGwinn

Quote:


  Originally Posted by *euclid* /img/forum/go_quote.gif 
_





 Cardas, Tara Labs, and Neutrik make Female XLR to Female RCA adapters, Neutrik is least expensive by a long shot.

http://www.dedicatedaudio.com/inc/sdetail/542
http://tara-direct.com/results.asp?ID=8
http://www.markertek.com/Product.asp...&search=0&off=_

 

It is important to make sure these adapters have pin 3 floating (not connected). If pin 3 is connected to ground (as is the case with many of these adapters), you can shorten the life of the output driver, possibly ruining it.

 Thanks,
 Elias


----------



## Lord Chaos

Elias, I'm starting to quantize what level settings work on my DAC1s. So far, I've checked the "classic" model, and for a fairly loud song with Sennheiser HD25s I'm up four detents from off. I plan to check the same song with the same cans on the USB version, but I'm wondering if the level comparison will be accurate because the classic DAC1 is being driven by a Squeezebox, with its own level control. I have that turned up all the way, but don't know how that matches the USB output.


----------



## ezkcdude

Quote:


  Originally Posted by *Lord Chaos* /img/forum/go_quote.gif 
_Elias, I'm starting to quantize what level settings work on my DAC1s. So far, I've checked the "classic" model, and for a fairly loud song with Sennheiser HD25s I'm up four detents from off. I plan to check the same song with the same cans on the USB version, but I'm wondering if the level comparison will be accurate because the classic DAC1 is being driven by a Squeezebox, with its own level control. I have that turned up all the way, but don't know how that matches the USB output._

 

It's SPDIF, right? As long as there is no funny stuff going no (such as ReplayGain), the digital signal from a SqueezeBox or any soundcard is nominally the same (of course, jitter may be different). The levels will be the same, so don't worry.


----------



## EliasGwinn

Quote:


  Originally Posted by *Lord Chaos* /img/forum/go_quote.gif 
_Elias, I'm starting to quantize what level settings work on my DAC1s. So far, I've checked the "classic" model, and for a fairly loud song with Sennheiser HD25s I'm up four detents from off. I plan to check the same song with the same cans on the USB version, but I'm wondering if the level comparison will be accurate because the classic DAC1 is being driven by a Squeezebox, with its own level control. I have that turned up all the way, but don't know how that matches the USB output._

 

They should be matched, but I can't say for sure. But, go ahead and test like this, and we'll go with it.

 Thanks,
 Elias


----------



## euclid

Quote:


  Originally Posted by *EliasGwinn* /img/forum/go_quote.gif 
_They should be matched, but I can't say for sure. But, go ahead and test like this, and we'll go with it.

 Thanks,
 Elias_

 

Elias is there a standard for sources to send the same level digital signal? can i assume my soundcard is outputting optical at the same level my CD player sends Coaxial? 

 also regarding the adapters, i have only owned the Tara Labs and i know they can be disassembled and unsoldered to float pin 3, the Cardas will probibly have to be cut open which will damage the special Cardas heatshink and i have no idea about the Neutrik.


----------



## ezkcdude

Quote:


  Originally Posted by *euclid* /img/forum/go_quote.gif 
_Elias is there a standard for sources to send the same level digital signal? can i assume my soundcard is outputting optical at the same level my CD player sends Coaxial? 

 also regarding the adapters, i have only owned the Tara Labs and i know they can be disassembled and unsoldered to float pin 3, the Cardas will probibly have to be cut open which will damage the special Cardas heatshink and i have no idea about the Neutrik._

 

I guess you didn't read my message. SPDIF is a standard for sending digital audio data. It should be the same no matter your source. The signal can be sent optically (toslink) or electrically (coaxial). Unless you have applied gain to your original source files, there will be no difference. The Benchmark guy should know this stuff already.


----------



## euclid

Quote:


  Originally Posted by *ezkcdude* /img/forum/go_quote.gif 
_I guess you didn't read my message. SPDIF is a standard for sending digital audio data. It should be the same no matter your source. The signal can be sent optically (toslink) or electrically (coaxial). Unless you have applied gain to your original source files, there will be no difference. The Benchmark guy should know this stuff already._

 

cool i did overlook your post, i knew S/PDIF is the standard but i wasnt sure what is actually standarized and dont understand how light pulse is measured in volts to be compared and matched to coax output level.


----------



## EliasGwinn

Quote:


  Originally Posted by *euclid* /img/forum/go_quote.gif 
_cool i did overlook your post, i knew S/PDIF is the standard but i wasnt sure what is actually standarized and dont understand how light pulse is measured in volts to be compared and matched to coax output level._

 

The voltage level and optical intensities of the different transmission mediums may be different from one another, but this doesn't relate to a difference in audio amplitude. If all the 1's and 0's are the same, a 5V digital signal will produce the same audio as a 3.3V digital signal.

 Specifically, S/PDIF is standardized to transmit at 0.5 Vp-p, AES3id (AES over coaxial) is standardized to transmit at 1.0 Vp-p, and AES3 (AES over balanced cable) is standardized to transmit at 2-7 Vp-p. 

 But all three will produce the same audio qualities if all the 1's and 0's are the same (assuming proper reception and conversion).

 Thanks,
 Elias


----------



## pianomav

I just got my dac1 usb this morning and it sounds amazing. I tried using the optical input and now i'm playing around with the usb. One thing I noticed is the popping sound when playing flac in foobar. 

 Here's my setting in foobar (under preference):
 Playback volume: max
 source mode: none
 processing: none
 preamp: 0 db
 Output: DS: Benchmark 1.0 (selected)
 Buffer length: 1500 ms
 Output data format: 24bit

 Is the optimal settings in foobar? How do I fix the 'popping' sounds while playing audio via usb? Thanks in advance and appreacite any help on this.


----------



## chesebert

46 pages not a real comparison between old DAC1, new DAC1, Stello 220, Lavry, Transporter, or any other well-known CD player.

 is DAC 1 so bad that no one wants to talk about 'how it sounds'?


----------



## Hadden

Quote:


  Originally Posted by *chesebert* /img/forum/go_quote.gif 
_46 pages not a real comparison between old DAC1, new DAC1, Stello 220, Lavry, Transporter, or any other well-known CD player.

 is DAC 1 so bad that no one wants to talk about 'how it sounds'?_

 

Don't interrupt the infomercial.


----------



## poo

I agree - it would be great to get some impressions of the sound. Many of us don't have the ability to hear the Dac1 ourselves (without buying one first), so any descriptions are helpful, especially comparative descriptions that offer context.


----------



## euclid

listening to FLAC or CD on my computer Chaintech AV710 outputting optical to DAC1 the headphone amp is 200% better than i expected based on impressions that i have read, i will not be buying another headphone amp for this i will upgrade to the USB instead to get the added volume range and hopefully a mod in the future. its not harsh or thin or sucked out or basically anything, its clean, decay lasts forever its really digging detail deep in the recording. if you want euphoric look elsewhere. 

 when it comes down to the form factor and function the Benchmark does what i need. i am using the volume controlled headphone amp for my computer and i am running the RCA out to tripath amp to drive bookshelves for my computer. then using calibrated XLR with 20db attenutation to connect balanced to Headamp GS-X in another room which preamps my main sytem, no splitters or adapters all i need to do it disconnect the balanced cables from my Meridian G08 and plug in the balanced runs from the DAC1, if the GS-X had 2 balanced inputs like the Apache i wouldnt need to disconnect anything just change inputs. i bought the black DAC1 with newer non-rackmount faceplate, it looks 4x nicer than that the Lavry "cablebox" and is 4x smaller than the Stello DA220, with more functionality than both. 
	

	
	
		
		

		
		
	


	




 this "infomerical" is called manufacturer support and i am surprised Benchmark has been so open about their engineering. go back and reread the first few pages badmouthing the USB untill Elias registered and defended the design, it didnt start as a nice thread but eventually the haters shut up, IMHO there is nothing not to like about this unit except the high volume level on the headphone amp.


----------



## tonygeno

It sounds very good. Plus this isn't a comparison thread, now is it? Those interested in comparisons can do a search as there are several threads comparing the Benchmark to other dacs. And if that doesn't sufficiently float your boat, there's always the option of starting a new thread requesting the Benchmark vs the dac of your choice.


----------



## zkn

Elias

 What kind of monster load headphones am I supposed to drive with the non-USB DAC1 headphone outs again?

 They're so powerful that nothing seems to handle it or am I missing some pro audio world point here?


----------



## s.a.b.

I drive the "standard" 250 ohm Beyer dt990, 880 ('03 version), as well as the AKG 701, with the non-USB DAC1 and have no problem with the gain- the volume control is always between 10 and 12 o'clock. 



  Quote:


  Originally Posted by *zkn* /img/forum/go_quote.gif 
_
 What kind of monster load headphones am I supposed to drive with the non-USB DAC1 headphone outs again?

 They're so powerful that nothing seems to handle it or am I missing some pro audio world point here?_


----------



## zkn

How is that possible?

 I can't get even to 09:00 with the 300 Ohm HD650 from digital source with full digital volume.


----------



## poo

Quote:


  Originally Posted by *tonygeno* /img/forum/go_quote.gif 
_It sounds very good. Plus this isn't a comparison thread, now is it? Those interested in comparisons can do a search as there are several threads comparing the Benchmark to other dacs. And if that doesn't sufficiently float your boat, there's always the option of starting a new thread requesting the Benchmark vs the dac of your choice._

 

Thanks for that revelation...


----------



## poo

Quote:


  Originally Posted by *euclid* /img/forum/go_quote.gif 
_listening to FLAC or CD on my computer Chaintech AV710 outputting optical to DAC1 the headphone amp is 200% better than i expected based on impressions that i have read, i will not be buying another headphone amp for this i will upgrade to the USB instead to get the added volume range and hopefully a mod in the future. its not harsh or thin or sucked out or basically anything, its clean, decay lasts forever its really digging detail deep in the recording. if you want euphoric look elsewhere. 

 when it comes down to the form factor and function the Benchmark does what i need. i am using the volume controlled headphone amp for my computer and i am running the RCA out to tripath amp to drive bookshelves for my computer. then using calibrated XLR with 20db attenutation to connect balanced to Headamp GS-X in another room which preamps my main sytem, no splitters or adapters all i need to do it disconnect the balanced cables from my Meridian G08 and plug in the balanced runs from the DAC1, if the GS-X had 2 balanced inputs like the Apache i wouldnt need to disconnect anything just change inputs. i bought the black DAC1 with newer non-rackmount faceplate, it looks 4x nicer than that the Lavry "cablebox" and is 4x smaller than the Stello DA220, with more functionality than both. 
	

	
	
		
		

		
		
	


	




 this "infomerical" is called manufacturer support and i am surprised Benchmark has been so open about their engineering. go back and reread the first few pages badmouthing the USB untill Elias registered and defended the design, it didnt start as a nice thread but eventually the haters shut up, IMHO there is nothing not to like about this unit except the high volume level on the headphone amp._

 

Excellent information - much appreciated. By 'high volume level on the headphone amp' that you mention, do you mean you need to turn volume up more than you expected to achieve a suitable level? You make it sound as though this is better on the USB version - am I reading this correctly?

 Thanks.


----------



## zkn

No he means what I've been on about all the time

 you reach the max listening levels way too early with the original DAC1

 USBDAC1 has a solution for this but IMO they increased the price way too much. On the other hand that also makes me not to regret buying the old one just before USB version rolled out.


----------



## thomaspf

Quote:


  Originally Posted by *euclid* 
_this "infomerical" is called manufacturer support and i am surprised Benchmark has been so open about their engineering. go back and reread the first few pages badmouthing the USB untill Elias registered and defended the design_

 

I don't think anyone was badmouthing USB. And the registering of Elias does not change the facts about the Windows USB driver in any way. 

 There is just no point in arguing with sales people.

 Cheers

 Thomas


----------



## euclid

Quote:


  Originally Posted by *zkn* /img/forum/go_quote.gif 
_Elias

 What kind of monster load headphones am I supposed to drive with the non-USB DAC1 headphone outs again?

 They're so powerful that nothing seems to handle it or am I missing some pro audio world point here?_

 

i asked Elias this same question becuase it didnt make sense to me either, what headphones could possibly need that much gain? since its Saturday and he wont likely respond until Monday i will paraphase our conversion. 

 he explained that in the pro audio application of recording a live event the mics are hooked to an ADC then back through a DAC for monitoring, during the initial setup process (in silence) the ambient noise pollution is measured by cranking the headphone volume on the DAC1 and listening for interference or stray sound that may degrade the live recording. so basically for this application the louder the better and he explained that a Benchmark mic preamp uses the same headphone amp as the DAC1 and pro users are complaining it is not loud enough. 

 we are simply using it as a home headphone amp for casual listening, which was not its intended purpose.

 also regarding the Beyers reaching 12:00, i get the HD650 up to about 10:00 before its too loud, even though the Beyers are less impedance than the Senns when i owned the DT880 i needed to raise the volume of my GS-X higher to reach the same listening level as the Senn HD650, the Beyers are probibly not as efficient as the Senns.

 if i was just using the Senns i would probibly not upgrade to the USB to get the 10db attenuation, but i want to hopfullly get my Audio Technicas to the 10:00-11:00 volume level and Senns past 12:00. right now my ATs are about 4 or 5 clicks from the bottom, no usable volume range on the standard DAC1 with low impedance headphones, i imagine Grado would be the same.


----------



## tonygeno

Quote:


  Originally Posted by *poo* /img/forum/go_quote.gif 
_Thanks for that revelation... 
	

	
	
		
		

		
		
	


	


_

 

You're welcome. Certain posters want to turn this into a this versus that thread and this thread is anything but. Forty-six good pages evidently pisses certain people off.


----------



## euclid

Quote:


  Originally Posted by *thomaspf* /img/forum/go_quote.gif 
_I don't think anyone was badmouthing USB. And the registering of Elias does not change the facts about the Windows USB driver in any way. 

 There is just no point in arguing with sales people.

 Cheers

 Thomas_

 

sorry but i disagree with you, this is more than marketing BS, it is good engineering and Benchmark has been explaining and defending their DAC1 design.

 for sure there was at least 5 people within the first 2 pages who instantly dismissed the USB upgrade as an overpriced afterthought to cash in on redundant USB input. after Benchmark came forward and set the facts straight and even shared some seemingly proprietary development info there was simply no denying it was more than just hype. 46 pages later it seems everyones questions have been answered and the only ones left posting are now owners asking for tech support for our own setups. 

 i know that by going optical out of a PC soundcard i am sending insane levels of jitter in the name reducing RF interference of coax in such a noisy environment, Benchmark is the only manufacturer that has openly discussed and adequately explained how they handle jitter, and by totally reclocking the DAC1 should be completely immune by design. since recieving it myself i realise the sound quality claims are real. 

 there are a ton of DAC comparison threads floating around, why does this thread need to turn into more SQ comparisons? i wouldnt trust other headfiers saying they like the Stello better becuse it sounds "fuller", 99% chance they are just hearing high levels of jitter and interpretting it as a warmer midrange, likewise it has no internal headphone amp so any Stello DA220 sound impression is also based on external interconnects and seperate headphone amp.

 edit: one more thing, i have been reading DAC1 impressions on Headfi since before i registered, especially from meet impressions, and each seems to regurgitate the same info about the headphone amp being nothing special but not once can i recall reading about the high gain setting. is it possible that alot of these computer setups have been compensating for the high headphone gain by lowering the volume digitally in software?... and as explained by Benchmark also introducing high levels of distortion. in my experience you need to go VERY high up in the amp chain to maintain this level of upper detail, bass and overall resolution without associated grain and distortion. i am looking forward to receiving the USB version so i can give the headphone amp some more critical listening and will post some actual impressions in a new thread when i can.


----------



## poo

Quote:


  Originally Posted by *euclid* /img/forum/go_quote.gif 
_there are a ton of DAC comparison threads floating around, why does this thread need to turn into more SQ comparisons?_

 

Because SQ is what it's all about... at least for me...

 Comparisons are all many of us have to measure so called 'real world' situations. This thread is about (or at least originally was about) the fact that the DAC1 is now available with USB, and that in itself begs for comparison.

 If we didn't compare components - there wouldn't be much point to the site in general - everyone would be happy with what they had. Discussing DACs (or any components for that matter) in a comparative context gives a discussion meaning. In fact most comments _are_ comparative, wether intentionally or not, because peoples impression of something is guided by what previous experience they have to compare it too.


----------



## zkn

Quote:


  Originally Posted by *euclid* /img/forum/go_quote.gif 
_also regarding the Beyers reaching 12:00, i get the HD650 up to about 10:00 before its too loud, even though the Beyers are less impedance than the Senns when i owned the DT880 i needed to raise the volume of my GS-X higher to reach the same listening level as the Senn HD650, the Beyers are probibly not as efficient as the Senns._

 

I own the DT770/80 which are almost the same as 300 Ohm sennheisers so maybe.. but I don't want to get 400 EUR 600 Ohm manufacture beyers.


----------



## infinitesymphony

I have a question about DAC1 differences... I've heard from several people that during the course of production, the DAC1 has received some significant updates without version designations. It appears that the USB version has the most changes with regard to the original design. Can anyone comment on any or all of the changes that have taken place?

 I apologize if this has been answered earlier in the thread.


----------



## thomaspf

Quote:


 sorry but i disagree with you, this is more than marketing BS, it is good engineering and Benchmark has been explaining and defending their DAC1 design. 
 

I appreciate your enthusiasm but I am unaware of any engineering Benchmark has done on the USB part except for interfacing to the digital output. The chip is a standard TI chip and the firmware comes from Centrance. Maybe Centrance has done some engineering on Benchmark's behalf.

 Some of the claims made on this thread about this solution are to quote you again 'marketing BS'. The DAC1 is a very good DAC but how can they claim that their device changes the behavior of how Windows works. 

 I looked at a bunch of USB audio devices in detail including the DAC1 USB and the only detail in how their USB audio device looks different to the system is that it annouces itself as only supporting 24bit PCM. That does not make the system suddenly send bit perfect data to it. 

 Cheers

 Thomas


----------



## sangel

Hello Elias,

 Here, http://www.benchmarkmedia.com/dac1/dac1-usb_par.pdf i.e. a bench test review,
 I have found an issue releted to DAC1 usb, output impedance on XLR's.
 The reviewer found - measured - an output impedance - on XLR - of 133 Ohms.
 Now, considering that the output impedance - from the specs - is 60 Ohms, (i.e. 30 + 30 Ohms per +/- leg ), then what is the real status for this subject... This is very important...as you may realize.. 

 Thanks in advance.

 Best

 Sangel

 ps. As far as I now the DAC1 output impedance remains stable - unchanged (for fix and variable output modes on BOTH RCA & XLR)


----------



## EliasGwinn

Hey folks,

 To those who question the integrity of this thread, I'd like to say, "Great!" Questioning manufacturers is a great thing to do, and I encourage you to do so, not only to myself and Benchmark, but all the other manufacturers as well. Audio equipment costs too much to accept claims without analyzing, scrutinizing, and questioning them. And I believe it is the designers responsibility to explain themselves in clear, concise, _objective_ terms. 

 I do not, and will not, make subjective claims about the sonic nature of our products, because it would be undeniably biased. It would turn this thread into 'marketing B.S.', and I don't want that. I won't let that happen. The sonic nature of the DAC1 will speak for itself, either with trail-based listening, or through the current owners. We offer a 30-day trial period because we know that a perspective customer needs to hear the product to know for sure if it will suit them.

 The information I present on this thread is strictly based on objective, measurable facts. I am not a 'sales person'. I am an electrical engineer (mainly R&D), and I double as technical support. 

 There are three main reasons we encourage this type of disussion: 

 1- Customers need to know how to make the most of their audio setup. Without doing hundreds of hours of research and testing, it is impossible to know the best settings for all the devices in the audio chain. We have done that testing, and we can provide that information to you. 

 2 - It provides feedback to us, which is immensely important for designing (or correcting) products to better suit the users. The headphone volume is a perfect example. Without hearing the users tell us that the volume is too loud, we would not have known any different. However, with this information, we can take action and re-adjust the design for a lower volume. The result is a better product, which is better for the users as well as us.

 3 - There is way too much 'snake oil' in this industry. People are being taken advantage of by manufacturers who make outrages claims, exploiting customers simply because they can't know any better without a background in electronic design and thousands of dollars of testing equipment. We pride ourselves in making products that are very 'real'. They have proven results, and can stand up to the scrutiny of the most stringent of tests. They are built to be transparent, faithful, and completely accurate. They are not built to an aesthetic. This forum provides us the opportunity to objectively explain how every component works to make the product exactly what it is. We want to debunk any mysteries or unknowns related to audio electronics.

 Hope that helps explain things!!

 Thanks,
 Elias


----------



## chesebert

IIRC, DAC1 uses a switching PS. could you do me a favor and run a FFT on the DC output of that PS. I would like to know the harmonic content of the DC noise and also their respective amplitudes. 

 My next question is why switching vs linear, which is easier to design.







 while you are doing the FFT for DC output, would you also run a FFT on the Vo of the analog stage? I don't remember seeing this graph published by stereophile.


----------



## EliasGwinn

Quote:


  Originally Posted by *chesebert* /img/forum/go_quote.gif 
_IIRC, DAC1 uses a switching PS. could you do me a favor and run a FFT on the DC output of that PS. I would like to know the harmonic content of the DC noise and also their respective amplitudes. 

 My next question is why switching vs linear, which is easier to design.






 while you are doing the FFT for DC output, would you also run a FFT on the Vo of the analog stage? I don't remember seeing this graph published by stereophile. 
	

	
	
		
		

		
		
	


	


_

 

Chesebert,

 Thanks for the questions.

 The DAC1 does not have a switching power supply, it has a linear power supply. We could run an FFT of the DC output if you like. 

 Another interesting fact about the power supply is, when the DAC1 was in development, we wanted to test how immune the audio circuit was to noise on the power supply. So we injected high levels of audio-band signal onto the DC rails to determine if there was any crosstalk. The maximum power supply noise in the audio circuit was less then -126 dB (as posted in the specs), even with noise artificially imposed onto the supply rails.

 As for running "a FFT on the Vo of the analog stage?", do you mean a Frequency Response plot, or an idle-channel noise FFT? These are both available in our manual, and I would be glad to post them here independently. If I'm misunderstanding what you meant, could you please clarify?

 Thanks,
 Elias


----------



## chesebert

Quote:


  Originally Posted by *EliasGwinn* /img/forum/go_quote.gif 
_Chesebert,

 Thanks for the questions.

 The DAC1 does not have a switching power supply, it has a linear power supply. We could run an FFT of the DC output if you like. 

 Another interesting fact about the power supply is, when the DAC1 was in development, we wanted to test how immune the audio circuit was to noise on the power supply. So we injected high levels of audio-band signal onto the DC rails to determine if there was any crosstalk. The maximum power supply noise in the audio circuit was less then -126 dB (as posted in the specs), even with noise artificially imposed onto the supply rails.

 As for running "a FFT on the Vo of the analog stage?", do you mean a Frequency Response plot, or an idle-channel noise FFT? These are both available in our manual, and I would be glad to post them here independently. If I'm misunderstanding what you meant, could you please clarify?

 Thanks,
 Elias_

 

I am sorry, I should have been more clear. I mean a FFT of the 1Khz on the Vo. I am curious to see the harmonic content and their relative amplitudes. 

 Thanks


----------



## Jetlag

Quote:


  Originally Posted by *EliasGwinn* /img/forum/go_quote.gif 
_We recommend (and use) Canare StarQuad analog balanced cables with Neutrik connectors. They are both very high quality, sonically and mechanically, and not over-priced. We have our cables built by Have, Inc (www.haveinc.com)_

 

Exactly what I am using:






 Although I intended to listen to music using USB from my newly built computer via FLAC + Foobar, I changed my mind in the end. My chain is FLAC (CDs ripped using EAC + AccurateRip plugin) stored on my Infrant NV running Slimserver > gigabit ethernet > SB3 (digital out at fixed volume) > SPDIF > DAC1 USB > XLR cables above > Paradigm Active Ref Studio 20s.

 To keep computer sounds separate from my audio chain I use my sound card connected to my LC-32GP1U via 1/8" stereo cable. This way the Active 20s play music, and the LCD speakers produce all sound from my PC.

 I am a former DAC1 classic owner and love the sound due to it being as neutral as anything I have heard. I don't want coloration, warming, softening, etc., and none of my CDs had any layers of grunge or veils that required removing. IMOHO the DAC1 is very accurate and effectively converts the digital signal into analog without inflicting any harm or its own personality onto the music. What goes in is what comes out which is exactly what I want, I want to hear what the musician(s) & their chosen recording engineer intended me to hear. I squeezed my setup together for this photo, but it gives you an idea of how much I enjoy working in my home office now.


----------



## Lord Chaos

The reason I mentioned the volume characteristics of the Squeezebox is that I was very surprised to find that it varied the digital output to my DAC1. I thought digital was always full-honk. Turning down the Squeezebox volume reduces both the unit's headphone amp (wimpy and noisy) and the S/PDIF level.

 The DAC1 USB has internal jumpers you can set to reduce the gain in the headphone amp by 10dB. The "regular" DAC1 doesn't have these.

 Both units have excellent sound quality, better than any other digital component I've heard in many years of using audio equipment. Pipe organ music, which I find unlistenable on most systems due to the graininess, is smooth and musical from the DAC1. Everything else simply sounds like music, not machinery. The quality of the listening experience is entirely dependent upon the recording quality: whatever is there, is what I get from the DAC1.

 I'm not the ultimate tweaker. I'm looking for "good enough." In this case, good enough is outstanding. My standard is live unamplified music.


----------



## Jetlag

Quote:


  Originally Posted by *Lord Chaos* /img/forum/go_quote.gif 
_The reason I mentioned the volume characteristics of the Squeezebox is that I was very surprised to find that it varied the digital output to my DAC1. I thought digital was always full-honk. Turning down the Squeezebox volume reduces both the unit's headphone amp (wimpy and noisy) and the S/PDIF level._

 

To keep this from happening open SS, click on "player settings" and then use the drop down menu at the top and select "audio". 2/3 down the page you'll see a selection for "Digital Volume Control". Use the drop down menu and select "digital output volume is fixed" and click the "change" button. Done!

 (Not trying to hijack this great thread BTW)


----------



## pianomav

Quote:


  Originally Posted by *Jetlag* /img/forum/go_quote.gif 
_Exactly what I am using:






 Although I intended to listen to music using USB from my newly built computer via FLAC + Foobar, I changed my mind in the end. My chain is FLAC (CDs ripped using EAC + AccurateRip plugin) stored on my Infrant NV running Slimserver > gigabit ethernet > SB3 (digital out at fixed volume) > SPDIF > DAC1 USB > XLR cables above > Paradigm Active Ref Studio 20s.

 To keep computer sounds separate from my audio chain I use my sound card connected to my LC-32GP1U via 1/8" stereo cable. This way the Active 20s play music, and the LCD speakers produce all sound from my PC.

 I am a former DAC1 classic owner and love the sound due to it being as neutral as anything I have heard. I don't want coloration, warming, softening, etc., and none of my CDs had any layers of grunge or veils that required removing. IMOHO the DAC1 is very accurate and effectively converts the digital signal into analog without inflicting any harm or its own personality onto the music. What goes in is what comes out which is exactly what I want, I want to hear what the musician(s) & their chosen recording engineer intended me to hear. I squeezed my setup together for this photo, but it gives you an idea of how much I enjoy working in my home office now.




_

 

Jetlag, 

 What's the part ID for those xlr cable? Is this the HAVEFLEX STARQUAD CBL XLRM-XLRF ? I'm trying to get a pair myself... thanks...


----------



## MusicFirst

Very nice Aquos, Jetlag. I have the 32" 1080p. What size is yours?


----------



## Jetlag

Mine are a slightly older version of the HAVEFLEX STARQUAD CBL XLRM-XLRF 6' BLACK, Part ID: 201000-06BLA. Now they are made with nickel shells and silver contact pins vs my black shells and gold pins. You can still get the black ones at B&H Photo, part number CAXMXF6 but I prefer the look and lower cost of the newer ones. (or you could always get a 1 meter pair of Stealth Indras for just $7475 for balanced XLR.) 
	

	
	
		
		

		
		
	


	




  Quote:


  Originally Posted by *MusicFirst* /img/forum/go_quote.gif 
_Very nice Aquos, Jetlag. I have the 32" 1080p. What size is yours?_

 

LC-32GP1U (32" 1080P)

 OK, now back to discussing the DAC1 USB (and please pardon the interruption)


----------



## pianomav

thnx Jetlag...


----------



## EliasGwinn

Quote:


  Originally Posted by *infinitesymphony* /img/forum/go_quote.gif 
_I have a question about DAC1 differences... I've heard from several people that during the course of production, the DAC1 has received some significant updates without version designations. It appears that the USB version has the most changes with regard to the original design. Can anyone comment on any or all of the changes that have taken place?

 I apologize if this has been answered earlier in the thread._

 

The early (pre May 2004) DAC1's had a few things that we've since upgraded. Those things are:

 1 - Maximum input sample rate: 96 kHz. Upgraded to >192 kHz 

 2 - Unbalanced outputs had output impedance of 1.25k. This was to prevent distortion when the R & L were "Y'd" together for forced mono. We've since lowered the output impedance to 30 ohms.

 3 - Continuous volume pot was replaced with detented volume pot

 4 - Trim resistor added to eliminate DC offset at volume pot

 These changes (except the 192kHz) were made based on field experience and user feedback. The circuit board revision number was changed upon each revision. 

 Thanks,
 Elias


----------



## EliasGwinn

Quote:


  Originally Posted by *thomaspf* /img/forum/go_quote.gif 
_I appreciate your enthusiasm but I am unaware of any engineering Benchmark has done on the USB part except for interfacing to the digital output. The chip is a standard TI chip and the firmware comes from Centrance. Maybe Centrance has done some engineering on Benchmark's behalf.

 Some of the claims made on this thread about this solution are to quote you again 'marketing BS'. The DAC1 is a very good DAC but how can they claim that their device changes the behavior of how Windows works. 

 I looked at a bunch of USB audio devices in detail including the DAC1 USB and the only detail in how their USB audio device looks different to the system is that it annouces itself as only supporting 24bit PCM. That does not make the system suddenly send bit perfect data to it. 

 Cheers

 Thomas_

 

Thomas,

 The firmware for the DAC1 USB was developed jointly between Benchmark and Centrance. The majority of the code was initially written by Centrance, and the majority of the trouble-shooting of the code was done by Benchmark.

 The DAC1 USB firmware does not change how Windows operates. The firmware simply provides a non-obstructive path to stream bit-transparent data. 

 I have personally ran tests to verify bit-transparency. These tests were developed by Audio Precision, the leader in audio testing equipment and software. The Audio Precision has a predetermined bit-stream. We made a .wav file of that bit stream and played it through various common media players. The audio was streamed through the USB port (using USBaudio.sys), converted to PCM, and sent back into the Audio Precision. The Audio Precision compared this returned bit-stream with the initial predetermined-bit stream. If any bits had changed, the Audio Precision would report an error. If there are no bits changed, the Audio Precision reports no errors. 

 This is not a distortion test, or an averaging function, or an FFT. This test checks the bit-stream, bit for bit.

 Thanks,
 Elias


----------



## EliasGwinn

Quote:


  Originally Posted by *sangel* /img/forum/go_quote.gif 
_Hello Elias,

 Here, http://www.benchmarkmedia.com/dac1/dac1-usb_par.pdf i.e. a bench test review,
 I have found an issue releted to DAC1 usb, output impedance on XLR's.
 The reviewer found - measured - an output impedance - on XLR - of 133 Ohms.
 Now, considering that the output impedance - from the specs - is 60 Ohms, (i.e. 30 + 30 Ohms per +/- leg ), then what is the real status for this subject... This is very important...as you may realize.. 

 Thanks in advance.

 Best

 Sangel

 ps. As far as I now the DAC1 output impedance remains stable - unchanged (for fix and variable output modes on BOTH RCA & XLR)_

 

Sangel

 The impedance mentioned in the review was measured with the 20 dB attenuators activated. The output impedance of the DAC1 without output attenuation is 60 ohms.

 Thanks,
 Elias


----------



## infinitesymphony

Quote:


  Originally Posted by *EliasGwinn* /img/forum/go_quote.gif 
_The early (pre May 2004) DAC1's had a few things that we've since upgraded. Those things are..._

 

Thanks! Those all sound like great updates. I've been craving a DAC1 for use as a DAC/preamplifier with balanced studio monitors, and reading this thread hasn't made it any easier on my wallet. 
	

	
	
		
		

		
		
	


	




 Now, to decide between the regular or the USB version...


----------



## audioengr

Quote:


  Originally Posted by *EliasGwinn* /img/forum/go_quote.gif 
_The early (pre May 2004) DAC1's had a few things that we've since upgraded. Those things are:

 1 - Maximum input sample rate: 96 kHz. Upgraded to >192 kHz 

 2 - Unbalanced outputs had output impedance of 1.25k. This was to prevent distortion when the R & L were "Y'd" together for forced mono. We've since lowered the output impedance to 30 ohms.

 3 - Continuous volume pot was replaced with detented volume pot

 4 - Trim resistor added to eliminate DC offset at volume pot

 These changes (except the 192kHz) were made based on field experience and user feedback. The circuit board revision number was changed upon each revision. 

 Thanks,
 Elias_

 


 I think you forgot a few other upgrades:

 1) added a buffer to the unbalanced outs and moved the voltage divider, so that it is independent of the balanced outs.

 2) added film caps to the AC-coupling in the signal paths.

 3) adjusted the LED brightness

 4) volume control has detents

 5) changed the op-amps from Philips to TI

 #5 probably made the biggest sound improvement, along with the unbalance out improvements


----------



## Lord Chaos

Speaking of LED brightness: I found a good use for the HD25 instruction booklet. It fits nicely over the LED on my DAC1 to keep the thing from waking me up. 
	

	
	
		
		

		
			





 The DAC1 does double-duty as a flashlight...


----------



## chesebert

Where are the FFT graphs I asked


----------



## thomaspf

Quote:


  Originally Posted by *EliasGwinn* 
_I have personally ran tests to verify bit-transparency. These tests were developed by Audio Precision, the leader in audio testing equipment and software. The Audio Precision has a predetermined bit-stream. We made a .wav file of that bit stream and played it through various common media players. The audio was streamed through the USB port (using USBaudio.sys), converted to PCM, and sent back into the Audio Precision. The Audio Precision compared this returned bit-stream with the initial predetermined-bit stream. If any bits had changed, the Audio Precision would report an error. If there are no bits changed, the Audio Precision reports no errors. _

 

Well, you can not have tested tis with the DAC1 since you can not record the digital output. So I assume you used some test setup.

 The problem is that Kmixer in Windows is actually multiplying all values by a number smaller than 1.0 at max volume so there is actually no way this can be bit perfect. I seem to recall I spoke to you on the phone some time around the beginning of this thread.

 You chose to ignore this little technical detail waving some audio precision test around. Since there are now USB->S/PDIF adapters around using the same Centrance code I was offered the opportunity to test one. Obviously this can not work bit perfect with wav or Directsound since you can't get the bits from Windows. 

 It works bit transparently just fine with kernel streaming like most of the other USB adapters using the USBaudio.sys driver. Your device is basically not better or worse than anyone else. The one custom driver that can work bit perfect with wav and DirectSound on XP that I know of is the M-audio Transit.

 On Vista you can get the stack to pass 16 bits through unmangled at 44.1 Khz if you select 24bit / 44.1 Khz in the audio control panel and output the 16bit stream as 24bit data with the data residing in the 16 most significant bits.

 None of this changes the fact that the DAC1 is a nice DAC and that having a USB interface is great. You might want to change your Wiki but other than that let's just move on.


 Cheers

 Thomas


----------



## sangel

Thanks a lot Elias.

 Best Regards

 Sangel


----------



## EliasGwinn

Quote:


  Originally Posted by *thomaspf* /img/forum/go_quote.gif 
_I seem to recall I spoke to you on the phone some time around the beginning of this thread.

 You chose to ignore this little technical detail waving some audio precision test around. Since there are now USB->S/PDIF adapters around using the same Centrance code I was offered the opportunity to test one. Obviously this can not work bit perfect with wav or Directsound since you can't get the bits from Windows. _

 

Thomas,

 I remember speaking to you on the phone. The last thing I remember speaking about was that you wanted to view our test files. So, we posted them for you to download. I hadn't heard back from you until now.

 I respect your stance on this, and I respect your credentials. If you are correct, if DirectSound is not capable of bit-transparency, then the Audio Precision testing device is flawed. This is not entirely impossible. However, it is a very well respected testing platform, perhaps the most respected in the audio community. Perhaps we should bring this discussion to them if we really want to determine where the inconsistencies are.

  Quote:


  Originally Posted by *thomaspf* /img/forum/go_quote.gif 
_The one custom driver that can work bit perfect with wav and DirectSound on XP that I know of is the M-audio Transit._

 

We tested this device as well, and found it to NOT be bit-transparent. Apparently, one of our testing mechanisms is flawed (or perhaps both).

 Again, I respect your aggressiveness in defending your test results. I apologize if you felt that I did not sufficiently follow up with you earlier. It was my impression that you were taking the test files we gave you, and now you were conducting new tests. 

 Thanks,
 Elias


----------



## EliasGwinn

Quote:


  Originally Posted by *chesebert* /img/forum/go_quote.gif 
_Where are the FFT graphs I asked 
	

	
	
		
		

		
		
	


	


_

 

Chesebert, 

 I'm sorry, we've been crazy around here lately trying to prepare for the AES show next weekend. 

 I'll try to make the plot and post it today.

 Thanks,
 Elias


----------



## EliasGwinn

Chesebert,

 Here is the 1k FFT you requested. 

 To Note: 

 * The first harmonic is on the order of -110 dBA.
 * All tones below -115 dBA are questionable because this approaches the limit of the testing equipment. This is because the testing equipment captures the signal utilizing its own set of gain stages, filters, A-to-D's, etc. Anything below -115 dBA is below the limits of the accuracy of the testing equipment. That is to say, these may be artifacts of the DAC1 or artifacts of the testing equipment or both.

 Thanks,
 Elias


----------



## EliasGwinn

Quote:


  Originally Posted by *audioengr* /img/forum/go_quote.gif 
_I think you forgot a few other upgrades:

 1) added a buffer to the unbalanced outs and moved the voltage divider, so that it is independent of the balanced outs._

 

Not exactly - we added a buffer after the voltage divider but did not
 change the position of the voltage divider in the circuit. We also
 changed the attenuator slightly so that the RCA outputs produce 2 Vrms
 when the XLR outputs produce +24 dBu (XLR attenuators set to 0 dB).


  Quote:


  Originally Posted by *audioengr* /img/forum/go_quote.gif 
_2) added film caps to the AC-coupling in the signal paths._

 

No, the DAC1 has always had film caps at the AC-coupling point in the
 signal path. There are actually three capacitors in parallel that are
 used for AC-coupling. We have a 1000uF cap in parallel with a 0.47uF
 cap in parallel with a 0.1uf cap. These characteristics of these caps
 compliment each other and combine to create a coupling cap that exceeds
 the performance of any single capacitor.

  Quote:


  Originally Posted by *audioengr* /img/forum/go_quote.gif 
_3) adjusted the LED brightness_

 

This is true.

  Quote:


  Originally Posted by *audioengr* /img/forum/go_quote.gif 
_4) volume control has detents_

 

This is also true.

  Quote:


  Originally Posted by *audioengr* /img/forum/go_quote.gif 
_5) changed the op-amps from Philips to TI

 #5 probably made the biggest sound improvement, along with the unbalance out improvements_

 

Until somewhere around the year 2000 or 2001 the Philips part was
 significantly superior to both the TI and Fairchild versions of the
 5532. All of the manufacturers have done die shrinks on the 5532 and
 the performance has either stayed the same or improved. We have
 conducted very detailed tests of these parts, and we continue to do so
 whenever a manufacturer makes a change.

 The end result of the 5532 die shrinks is that the TI part is now
 equivalent or better than the best Philips parts ever produced.
 Phillips ended production abruptly in 2003 after a fire at their plant.
 Fortunately we had already qualified the TI parts in 2002 and simply
 made the switch when the Phillips parts dried up. The 2003 Philips
 parts are very similar to the 2002 to 2007 TI parts. The pre-2002 TI
 parts should be avoided and should never be used to repair a DAC1.

 Thanks,
 Elias


----------



## audioengr

Quote:


  Originally Posted by *EliasGwinn* /img/forum/go_quote.gif 
_Not exactly - we added a buffer after the voltage divider but did not
 change the position of the voltage divider in the circuit. We also
 changed the attenuator slightly so that the RCA outputs produce 2 Vrms
 when the XLR outputs produce +24 dBu (XLR attenuators set to 0 dB)._

 

I have early versions of the DAC-1 to mod and they dont have the buffer at the RCA's. The voltage divider was right at the output. This is what killed the RCA output sound quality. the new design puts the voltage divider in-between two op-amps, where it should be. I dont really care, because I remove it anyway. Every resistor in the signal path adds noise.




  Quote:


 No, the DAC1 has always had film caps at the AC-coupling point in the
 signal path. There are actually three capacitors in parallel that are
 used for AC-coupling. We have a 1000uF cap in parallel with a 0.47uF
 cap in parallel with a 0.1uf cap. These characteristics of these caps
 compliment each other and combine to create a coupling cap that exceeds
 the performance of any single capacitor. 
 

This was not in the original DAC-1. I have modded plenty of these.

  Quote:


 Until somewhere around the year 2000 or 2001 the Philips part was
 significantly superior to both the TI and Fairchild versions of the
 5532. All of the manufacturers have done die shrinks on the 5532 and
 the performance has either stayed the same or improved. We have
 conducted very detailed tests of these parts, and we continue to do so
 whenever a manufacturer makes a change.

 The end result of the 5532 die shrinks is that the TI part is now
 equivalent or better than the best Philips parts ever produced.
 Phillips ended production abruptly in 2003 after a fire at their plant.
 Fortunately we had already qualified the TI parts in 2002 and simply
 made the switch when the Phillips parts dried up. The 2003 Philips
 parts are very similar to the 2002 to 2007 TI parts. The pre-2002 TI
 parts should be avoided and should never be used to repair a DAC1. 
 

Every DAC-1 with Philips op-amps sounded poor to me. The TI change was a big improvement IMO. The bass was just non-existent with the Philips parts, and this is dynamic bass I'm talking about, not steady-state. Steady-state measurements have limited usefulness for music. Music has a transient nature.

 Steve N.


----------



## Patu

Shouldn't it be possible to change the fuse of DAC1 so that I can use it here in Finland (220-230V power grid)? If so then what kind of fuse should I use and are they easily available?


----------



## audioengr

Quote:


  Originally Posted by *Patu* /img/forum/go_quote.gif 
_Shouldn't it be possible to change the fuse of DAC1 so that I can use it here in Finland (220-230V power grid)? If so then what kind of fuse should I use and are they easily available?_

 

It's a simple matter to reverse the fuse holder and change it to 240VAC operation. I personally replace the fuse with a 3 amp fast blo. Slow-blow fuses are the worst for adding inductance and killing the dynamics. I am only concerned about protecting the transformer wiring and not burning the board or starting a fire etc... I dont care about killing IC's, so I usually go with a higher rated fuse. Do this at your own risk. Any Radio Shack will have these.

 Steve N.
 Empirical Audio


----------



## EliasGwinn

Quote:


  Originally Posted by *Patu* /img/forum/go_quote.gif 
_Shouldn't it be possible to change the fuse of DAC1 so that I can use it here in Finland (220-230V power grid)? If so then what kind of fuse should I use and are they easily available?_

 

Patu,

 The power entry module (where you connect the power cord) has a fuse drawer, which you can remove by squeezing the two plastic tabs inward. To setup the DAC1 for 220V operation:

 1. Remove fuse drawer from power entry module

 2. Remove the gray plastic fuse holder from the fuse drawer

 3. Rotate gray plastic fuse holder 180 degrees

 4. Re-insert gray plastic fuse holder into the fuse drawer. The final result will leave the metal prongs on the bottom of the fuse drawer, and the outside of the fuse drawer should read "220".

 5. Re-insert the fuse drawer into the power entry module.

 That is all! After doing this, your DAC1 can be operated with 220V power.

 CAUTION: FOR CONTINUED FIRE HAZARD PROTECTION ALWAYS REPLACE THE FUSES WITH THE CORRECT SIZE AND TYPE (0.5A 250 V SLO-BLO® 5 X 20 MM – LITTELFUSE® HXP218.500 OR EQUIVALENT).

 Thanks,
 Elias


----------



## Patu

Good to know that it's so easy. Thanks for the help guys.


----------



## puntloos

Quote:


  Originally Posted by *chesebert* /img/forum/go_quote.gif 
_46 pages not a real comparison between old DAC1, new DAC1, Stello 220, Lavry, Transporter, or any other well-known CD player.

 is DAC 1 so bad that no one wants to talk about 'how it sounds'?_

 

Bah, what about my totally awesome comparison between the Benchmark DAC1, The Bel Canto DAC3 and the (built-in) DAC inside a Bow Technologies ZZ-Eight CD player on page 42 of this thread.


----------



## chesebert

Quote:


  Originally Posted by *puntloos* /img/forum/go_quote.gif 
_Bah, what about my totally awesome comparison between the Benchmark DAC1, The Bel Canto DAC3 and the (built-in) DAC inside a Bow Technologies ZZ-Eight CD player on page 42 of this thread._

 

oh my, the shameless self-promotion 
	

	
	
		
		

		
			





 your CDP is probably having problem (power issue? imp mismatch? cable synergy?) driving your integrated as my Central Station can do fairly convincing 3D soundstage even with its opamp output, not unlike the ones employed in DAC1.


----------



## euclid

i received the DAC1 USB last week. in comparison to the standad DAC1 the usable range on the headphone amp has improved enough for me. now the Senn HD650 can take the volume dial to around 1:00-2:00, the Audio Technica W5000 now have a usable volume range at 10:00-11:00 on the volume knob. 

 with the standard DAC1 the W5000 were really maxed by 9:00 and there was noticable inconsistency in the volume steps, sometimes i would take it one notch further and the volume would raise too much, then i would back it off, raise it agian and the volume was differet. the USB doesnt have that problem with the -10db jumper installed the dial goes high enough to overcome the tracking problems of the pot at low level.


----------



## olivier

hello there, 

 i am very impressed by the discussion here... 49 page of informative posts!

 i would like to get the benchmark dac but don't know if i should go for the usb version or not. the two major options for me are

 1) squeezebox, airport or other, <digital IC> benchmark non usb
 2) pc or mac <usb connect> benchmark usb.

 i would prefer 1) because of the simplicity and less cables. but from what i read here 2) would reduce jitter if done properly, and i am not sure that i could get the same sound quality from 1) 

 can someone guide me here!!?

 thanks!!, cheers, 
	

	
	
		
		

		
		
	


	




 olivier

 Ps: and byw, i have read this http://www.head-fi.org/forums/showpo...&postcount=216 but it's not answering my question. all i really care about is output to a preamp 
	

	
	
		
		

		
		
	


	




 so the gains for headphones don't really matter to me.


----------



## little-endian

A DAC1 without USB would be more financial because the extra charge for the USB-version is much higher than a comparable usb-device which delivers a s/pdif signal. Also keep in mind that the USB-connection will be limited to a sample rate of 96 kHz due to the (actually unnecessary) USB 1.1 compatibility.

 In regard to jitter (s/pdif vs. USB), you may tick this off as pure voodoo since Benchmark itself stated the immunity already several times and besides that it is doubt if one can hear this at all, anyway.


----------



## olivier

thanks little endian. 
	

	
	
		
		

		
		
	


	




 i was wondering what is a pc or a mac is actually sending out as a signal to either an airport (or equivalent) or a usb dac like the benchmark. 

 is it the actual mp3, aac, flac ... file?

 or

 is it the decompressed file which is an audio file with the corresponding sampling rate?

 i used to think it would be the first, hence that the dac should be able to read all the different formats, but reading this forum and others, i seem to understand now that it is the second instead, which actually makes more sense, as the comp can easily do the decompression part ...


----------



## little-endian

You're welcome, olivier.

 When we're talking about DACs, they always expect PCM data (except DSD which is used by SACDs, but there are only proprietary and ridiculous expensive solutions for that).

 The "special cases" where AC3, DTS and so on comes in place, are actually combinations of decoders and DACs, as they are often found in home cinema amplifiers. DACs by itself never heard about such formats like MP3, Vorbis, etc. - same for every sound card in a PC. It is simply not their task.


----------



## olivier

thanks for clarifying this little endian!

 i have soo much to learn and it's always a pleasure to do so!

 one more question i have in mind is basically the opposition of dac with dac/preamp. let me explain this.

 some users use the benchmark dac as a preamp too, and they seem satisfied with this. nevertheless, i don't think the benchmark, as a preamp, can match the quality of high-end audio equipment. 

 so, perhaps i should use the benchmark in conjunction with a good preamp, to take advantage of the audio quality of my preamp. my concern is that, by doing this, the benchmark preamplifies the signal sent to the preamp, and the signal spit out by the benchmark will be distorted in any case. there is no way the preamp can correct for that. 

 hence my (temporary) conclusion: get a dac that is a pure dac, that does not preamplify the signal, and use a good preamp after the dac. 






 i am getting anything wrong here?


----------



## infinitesymphony

Quote:


  Originally Posted by *olivier* /img/forum/go_quote.gif 
_nevertheless, i don't think the benchmark, as a preamp, can match the quality of high-end audio equipment._

 

Why not? It shouldn't be a difficult task...

  Quote:


  Originally Posted by *olivier* 
_so, perhaps i should use the benchmark in conjunction with a good preamp, to take advantage of the audio quality of my preamp. my concern is that, by doing this, the benchmark preamplifies the signal sent to the preamp, and the signal spit out by the benchmark will be distorted in any case. there is no way the preamp can correct for that._

 

All DACs must have some form of amplification, otherwise where does the analog sound come from? The Benchmark has a fixed output that bypasses the volume control, if that's a concern.

  Quote:


  Originally Posted by *olivier* 
_i am getting anything wrong here?_

 

More components = more cables = more potential for problems. Keeping everything inside one tightly-integrated box should be better than endlessly splitting it into different pieces of gear with huge differences in power section construction, parts, grounding, etc.


----------



## olivier

thanks for your input, infinitesymphony.

 let me first check if i'm following you here: are you saying that 5-10k$+ preamps can easily be beaten by a device such as the benchmark when it comes to preamplification? that would be a neat, but surprising, discovery!

 i understand that there must be _some_ type of amplification built in the dac, simply to output something... but, if this is the same as the output from a good preamp, why can't we say the same of the output of any device? i.e., why don't we plug our cd players/radios/etc into amps directly? a preamp should be a little bit more than a source selector, or not?

 perhaps someone had a chance to compare the benchmark with and without a preamp, associated with high end equipment to tell us the result?

 cheers, olivier


----------



## milkpowder

Quote:


  Originally Posted by *olivier* /img/forum/go_quote.gif 
_let me first check if i'm following you here: are you saying that 5-10k$+ preamps can easily be beaten by a device such as the benchmark when it comes to preamplification? that would be a neat, but surprising, discovery!_

 

That would be a very bold claim to make. Which "high end" pre-amps are we talking about? I would have a hard time believing the pre-amp section of the Benchmark is "better" than a dedicated LAMM or BAT pre-amp. Of course, price isn't the most reliable indicator of performance, but still...


----------



## Scrith

Quote:


  Originally Posted by *olivier* /img/forum/go_quote.gif 
_let me first check if i'm following you here: are you saying that 5-10k$+ preamps can easily be beaten by a device such as the benchmark when it comes to preamplification?_

 

I, for one, certainly think this is possible. The reason is that pre-amplifiers are supposed to have, ideally, ZERO affect on sound quality (i.e. they do not modify the quality of the sound that started in the source and is traveling to the amplifier through the pre-amp). I believe it is much easier to achieve this result with a simple device that has circuitry optimized for a very specific situation (DAC1 feeding an amp) than a complex over-engineered device that is trying to work in a very general situation ($10K preamp that has to work with any source, analog or digital, outputting at any signal level).


----------



## olivier

hello scritch! 

 that, for one, is an interesting opinion. 
	

	
	
		
		

		
		
	


	




 i don't personally know well what preamps are for, and why one should or not use one, but let me make a wild guess here.

 perhaps the amp needs a certain type of input (power, impedance, etc) to be at its optimal level, and who knows if any source can do that?

 namely, if the output of the benchmark (or any other) is really different from the output of my cd player, dvd player, radio etc, perhaps not all of these should be plugged directly into my amp? so ... would it be the case that the benchmark has the perfect output for my amp, but other sources don't!


----------



## tubaman

Quick question: 
 I just got a DAC1 USB. Is it ok to leave it on all the time? It does get quite warm. I just wanted to know if leaving it on will change the life expectancy of the unit , everything else being equal. It's not a real issue (I am the original owner, so I have the 5-yr warranty), just wanted to check.. (in the manual of my previous CD player, it was suggested that the unit be on all the time so to maintain operating temperature to extend the life of the unit)


----------



## pianomav

Shouldn't be an issue... Although I have mine plugged in to a power-strip, so when i'm not using it i just turn off the power strip..


----------



## olivier

i just read there...

http://www.hometheaterhifi.com/volum...03-part-1.html

 that the DAC-1 can be used either as a stand-alone DAC or as a all-in-one DAC/Preamp. according to these reviews, with a high-end equipment, one can hear noticeable differences, in favor of the stand-alone use.

 hence my question: how can one switch from stand-alone to all-in-one use? it that related to the `calibrated' vs 'variable' switch on the back of the unit that infinitesymphony mentioned earlier?

 thanks! olivier


----------



## little-endian

Quote:


  Originally Posted by *olivier* /img/forum/go_quote.gif 
_namely, if the output of the benchmark (or any other) is really different from the output of my cd player, dvd player, radio etc, perhaps not all of these should be plugged directly into my amp? so ... would it be the case that the benchmark has the perfect output for my amp, but other sources don't! 
	

	
	
		
		

		
			



_

 

Hi olivier,

 i don't think the DAC1's output is that different (except for the measurable better signal probably).

 From the practical point of view, today, the only use of a preamplifier is to select the source, to control the volume and maybe to adjust loudness/bass/treble. What needs actually preamplification is just the phono source because all other levels are high enough in the first place. Thus many "preamplifiers" even reduce (!) the signal level before it reaches the power amplifier - it depends on the volume setting. In the case your power amp allows you to control the volume or input sensitivity, there is nothing against connecting a cd-player for instance directly to it. In the case of the DAC1, the problem of the volume control is none because of its selectable variable setting.

 Your point in regard to match the source's signal to the power amp might be an argument but for me it sounds like a bit voodoo since not even the best pre amp would be able to add any detail which isn't present in the source, leaving it intact at best.

 @tubaman

 Yeah, it's okay to leave it on all the time. Elias from Benchmark already confirmed this here.


----------



## poo

Quote:


  Originally Posted by *little-endian* /img/forum/go_quote.gif 
_From the practical point of view, today, the only use of a preamplifier is to select the source, to control the volume and maybe to adjust loudness/bass/treble. What needs actually preamplification is just the phono source..._

 

I think you are confusing the use of a pre amp into power amp with integrated amplifiers and receivers.

 Pre amps have more uses than phono amplification...


----------



## poo

Quote:


  Originally Posted by *olivier* /img/forum/go_quote.gif 
_...i don't personally know well what preamps are for, and why one should or not use one, but let me make a wild guess here._

 

I think you need to research that in another thread before you ask questions about how good the DAC1 is at performing as a pre. Why bother if you don't know its application?


----------



## olivier

Quote:


  Originally Posted by *little-endian* /img/forum/go_quote.gif 
_Hi olivier,

 From the practical point of view, today, the only use of a preamplifier is to select the source, to control the volume and maybe to adjust loudness/bass/treble. What needs actually preamplification is just the phono source because all other levels are high enough in the first place. Thus many "preamplifiers" even reduce (!) the signal level before it reaches the power amplifier - it depends on the volume setting. In the case your power amp allows you to control the volume or input sensitivity, there is nothing against connecting a cd-player for instance directly to it. In the case of the DAC1, the problem of the volume control is none because of its selectable variable setting.
_

 

Thanks for the info little-endian. What you say about preamplification is consistent with what I read on other places. Still, some DAC's offer to bypass the preamp stage, so my guess is that the preamplification stage is never neutral, and there may be some benefit to leave this job to a specialized unit. 

 This being said, I think that my money can be more wisely spent looking for a great DAC/Preamp instead of looking for two separate units. The main reason for this is that most DACs at a reasonable price range will preamp anyway, and the distortions it introduces cannot be corrected for, no matter how good the preamp it feeds is.

 Cheers, Olivier


----------



## EliasGwinn

Quote:


  Originally Posted by *olivier* /img/forum/go_quote.gif 
_hello scritch! 

 that, for one, is an interesting opinion. 
	

	
	
		
		

		
		
	


	




 i don't personally know well what preamps are for, and why one should or not use one..._

 

Olivier,

 A pre-amp is not much more then what the name indicates: pre (before) amp (amplifier). A pre-amp can serve several functions, include (but not limited to):

 1. Amplitude adjustment/volume control (boost OR attenuation)

 2. Driving capability (certain pre-amps drive certain amplifiers better or worse then others)

 3. Tone shaping (this can include an EQ section, or adding tube-tone, etc)

 4. Source selecting and multiple-outputs

 5. Phono-pre (I mention this specifically because phono players require a special pre-amp circuit to operate properly)

 If you need these features, then a pre-amp is necessary. If you do not need these features, then adding a pre-amp will do nothing more then add noise, distortion, and cost to your system. Even the best pre-amp in the world will add some small amount of noise and distortion. If its not needed, then you may want to consider omitting it.

 The DAC1 has the 1st, 2nd and 4th features on this list. Therefore, if you don't have analog sources (tape, phono), and if you want the audio to be as transparent and uncolored as possible (i.e., no EQ or tubes), then the DAC1 will serve as the pre-amp, and it will perform this function as good or better then any other pre-amp. However, if you want to use another pre-amp to utilize those other features, the calibrated RCA outputs of the DAC1 will output the same levels as a CD transport. The DAC1 sound quality will remain, but the signal amplitude will be consistent with other typical devices.

 Thanks,
 Elias


----------



## olivier

Elias,

 Thank you so much for your thorough explanation. I guess the only point that I still need to understand is ... is there any more distortion from the benchmark in dac/preamp mode as compared to dac mode? My first guess would be that raising the signal to a higher gain can introduce more noise, but I can't be sure of that.

 Of course, the same question applies to other brands of DAC/preamps as well, although I don't necessarily expect an answer from you on them.

 Cheers, and thanks again!

 Olivier


----------



## EliasGwinn

Quote:


  Originally Posted by *olivier* /img/forum/go_quote.gif 
_ ... is there any more distortion from the benchmark in dac/preamp mode as compared to dac mode?_

 

Olivier,

 The distortion does not increase with higher output levels with the DAC1. The DAC1 is designed to operate completely in its linear region at all output levels. In fact, if your amp was able to accept +29dBu input (most can't), it would be advantageous to keep the DAC1's output level as high as possible to obtain optimal signal-to-noise ratio. 

 Also, amplifiers are more subject to distortion vs. gain, so minimizing the amplifiers gain would be an added benefit of a high output level from the DAC1.

 Thanks,
 Elias


----------



## EliasGwinn

Quote:


  Originally Posted by *euclid* /img/forum/go_quote.gif 
_i received the DAC1 USB last week. in comparison to the standad DAC1 the usable range on the headphone amp has improved enough for me. now the Senn HD650 can take the volume dial to around 1:00-2:00, the Audio Technica W5000 now have a usable volume range at 10:00-11:00 on the volume knob. 

 with the standard DAC1 the W5000 were really maxed by 9:00 and there was noticable inconsistency in the volume steps, sometimes i would take it one notch further and the volume would raise too much, then i would back it off, raise it agian and the volume was differet. the USB doesnt have that problem with the -10db jumper installed the dial goes high enough to overcome the tracking problems of the pot at low level._

 


 Euclid,

 I just noticed this post (sorry, I've been in NYC for the past 5 days). 

 I'm glad to hear that the -10dB worked for you. I know that the HD650's are high-Z, low sensitivity headphones, but what about the W5000? Are they particularly sensitive? How do they compare with other common headphones?

 Thanks,
 Elias


----------



## euclid

Quote:


  Originally Posted by *EliasGwinn* /img/forum/go_quote.gif 
_Euclid,

 I just noticed this post (sorry, I've been in NYC for the past 5 days). 

 I'm glad to hear that the -10dB worked for you. I know that the HD650's are high-Z, low sensitivity headphones, but what about the W5000? Are they particularly sensitive? How do they compare with other common headphones?

 Thanks,
 Elias_

 

no problem i wasnt expecting a response anyway 
	

	
	
		
		

		
		
	


	




 the W5000 are 42ohm 102db/w which is comparable in sensitivity to the Senn HD-650 at 103db/w the difference there is impedance. 

 suprisingly i have about the same usable range on my Shure E4c IEM as well which are 29ohm 109db/w, but IEMs use balanced armature drivers which have natural high frequency rolloff, the w5000 is a fairly bright sounding headphone with exceptional high frequency extension, all other things equal i think the Shures(or any IEM) could be driven to a higher volume without becoming uncomfortable just based on their sound signature.

 i also wanted to mention that attenuating the headphone amp on the DAC1 USB any further might affect the HD-650, they get loud enough now with the -10db jumper but past that 2:00 point i can basically take the volume all the way up without them really screaming. the standard DAC1 gain can [over]drive the HD-650 to ridiculous levels but the USB doesnt have that much headroom. 

 thanks again for the help


----------



## EliasGwinn

Quote:


  Originally Posted by *euclid* /img/forum/go_quote.gif 
_the W5000 are 42ohm 102db/w which is comparable in sensitivity to the Senn HD-650 at 103db/w the difference there is impedance. 

 suprisingly i have about the same usable range on my Shure E4c IEM as well which are 29ohm 109db/w, but IEMs use balanced armature drivers which have natural high frequency rolloff, the w5000 is a fairly bright sounding headphone with exceptional high frequency extension, all other things equal i think the Shures(or any IEM) could be driven to a higher volume without becoming uncomfortable just based on their sound signature.

 i also wanted to mention that attenuating the headphone amp on the DAC1 USB any further might affect the HD-650, they get loud enough now with the -10db jumper but past that 2:00 point i can basically take the volume all the way up without them really screaming. the standard DAC1 gain can [over]drive the HD-650 to ridiculous levels but the USB doesnt have that much headroom. 
_

 

Thanks so much for this feedback. So, perhaps we shouldn't decrease the gain of the headphone amp beyond that of the DAC1 USB?

 I can't stress enough how much this info helps us. You all are my insiders into the 'real-world'. I just noticed that this thread has gone on for 50 pages!! And it is still an incredible resource of communication between you all and us. I can't thank you all enough for the invaluable experience this has been. I hope it goes on for another 50, and beyond!!

 Thanks a million!!
 -Elias


----------



## tubaman

This seems to be the appropriate thread for my question:

 When using the DAC1 (USB) to feed a headphone amp, how should the volume control on DAC1 be set?

 I have two choices on this: calibrated or variable (adjustable in the back). According to the manual calibrated is near full volume. I tried near full volume (at about 5 o'clock) and I can barely turn the Prehead's volume control past 6 o'clock. 

 I have been using the variable setting and the volume control on the DAC1 is at about 11 o'clock, the Prehead at about 9-10 o'clock. 

 I wonder if there is a correct way (e.g. general principle) to do set the volume control on the units, anyone?


----------



## xp9433

""When using the DAC1 (USB) to feed a headphone amp - Corda Prehead""

 Tubaman interesting you are using the Prehead rather than the DAC1 headphone out. 

 Is this because of the cross feed circuit in the prehead or just overall better sound quality? Please enlighten me.

 Frank


----------



## EliasGwinn

Quote:


  Originally Posted by *tubaman* /img/forum/go_quote.gif 
_When using the DAC1 (USB) to feed a headphone amp, how should the volume control on DAC1 be set?

 I have two choices on this: calibrated or variable (adjustable in the back). According to the manual calibrated is near full volume. I tried near full volume (at about 5 o'clock) and I can barely turn the Prehead's volume control past 6 o'clock. 

 I have been using the variable setting and the volume control on the DAC1 is at about 11 o'clock, the Prehead at about 9-10 o'clock. 

 I wonder if there is a correct way (e.g. general principle) to do set the volume control on the units, anyone?_

 

Tubaman,

 Are you using the XLR outputs or the RCA outputs? If you are using the XLR outputs, there are output attenuators you can adjust to accommodate the input sensitivity of your headphone amp. If you are using the RCA outputs, it is perfectly fine to use the volume control to achieve the proper signal level for your headphone amp. In fact, it is fine to use the volume control for this purpose with the XLR outputs as well.

 Thanks,
 Elias


----------



## tubaman

Quote:


  Originally Posted by *EliasGwinn* /img/forum/go_quote.gif 
_Tubaman,

 Are you using the XLR outputs or the RCA outputs? If you are using the XLR outputs, there are output attenuators you can adjust to accommodate the input sensitivity of your headphone amp. If you are using the RCA outputs, it is perfectly fine to use the volume control to achieve the proper signal level for your headphone amp. In fact, it is fine to use the volume control for this purpose with the XLR outputs as well.

 Thanks,
 Elias_

 

Hi Elias, thanks for the reply. I am using RCA outputs now. What do you mean by "the appropriate signal level" (to be achieved by the volume control)? How do I know when it's appropriate? (E.g. minimum of ___ detents)


----------



## infinitesymphony

Quote:


  Originally Posted by *tubaman* /img/forum/go_quote.gif 
_Hi Elias, thanks for the reply. I am using RCA outputs now. What do you mean by "the appropriate signal level" (to be achieved by the volume control)? How do I know when it's appropriate? (E.g. minimum of ___ detents)_

 

Something that specific might be hard to determine, since it depends on what signal level your amplifier expects. If you have testing equipment, you could measure the variable output of the DAC1 until it matches the level listed in your amplifier's manual (if it's in there).


----------



## euclid

i would think the volume pot has to be over that 9:00-10:00 hump because there are tracking problems below that. you can change internal jumpers to set attenuation of the XLR outputs so you can have enough headroom to use the variable setting on the DAC1. other than that limitation it should be linear response no matter how far up the volume on the DAC1 is set.


----------



## infinitesymphony

Quote:


  Originally Posted by *euclid* /img/forum/go_quote.gif 
_i would think the volume pot has to be over that 9:00-10:00 hump because there are tracking problems below that._

 

Is this true for both the continuous and detented volume knobs?


----------



## EliasGwinn

Quote:


  Originally Posted by *tubaman* /img/forum/go_quote.gif 
_Hi Elias, thanks for the reply. I am using RCA outputs now. What do you mean by "the appropriate signal level" (to be achieved by the volume control)? How do I know when it's appropriate? (E.g. minimum of ___ detents)_

 

If you can find out what the input sensitivity (max input before clip) on your (pre)amp is, I'll let you know where to set the volume control.

 The RCA outputs are configured to the -10 dBu standard level, which is common to most RCA outputs on consumer devices. So, it should be ok as it is. However, if there are gross inconsistencies between the volume of the DAC1 vs. other components, you can use the front panel volume control for accommodating. 

 You could also re-calibrate the analog outputs using the calibration trim-pots in the rear, but this requires using a test tone and a measuring device such as a multimeter. There is info in the manual about this type of procedure, and I could answer any additional questions.

 Thanks,
 Elias


----------



## olivier

Quote:


  Originally Posted by *EliasGwinn* /img/forum/go_quote.gif 
_Olivier,

 The distortion does not increase with higher output levels with the DAC1. The DAC1 is designed to operate completely in its linear region at all output levels. In fact, if your amp was able to accept +29dBu input (most can't), it would be advantageous to keep the DAC1's output level as high as possible to obtain optimal signal-to-noise ratio. 
_

 

Elias,

 Thank you so much for your answers so much to the point. I am sure that your contributions are much appreciated by all readers of this forum. I am now quite convinced that the benchmark is the way to go for me, the quality of the support has completed the job of my online research. 

 I am thus going to save on a preamp, and pair the benchmark with an amp tube (probably CJ LP70S). My idea is that the tube amp may add some smoothness to the rather analytical DAC. I'll let everyone here know how that's doing!

 Thanks again, cheers,

 Olivier


----------



## poo

Quote:


  Originally Posted by *olivier* /img/forum/go_quote.gif 
_Elias... I am sure that your contributions are much appreciated by all readers of this forum. I am now quite convinced that the benchmark is the way to go for me, the quality of the support has completed the job of my online research._

 

x2


----------



## riverlethe

Wow, looking at the specs and reviews on this thing... Extremely low distortion, ample power and a flat frequency response. I don't expect Elias to comment, for obvious reasons, and I don't want to turn this into a flame-fest, but what more could one possibly need that couldn't be accomplished with a computer (adding nothing to the analog signal chain) EQ or DSP of some kind? 
 I hypothesize that all this "soundstage" talk with different amps and tubes, etc., is simply a matter of frequency-response manipulation with varying psychoacoustic effects. 
 I'm not an electrical engineer or even an experienced audiophile, and I may very well be wrong, so feel free to ignore me.


----------



## poo

Quote:


  Originally Posted by *riverlethe* /img/forum/go_quote.gif 
_I hypothesize that all this "soundstage" talk with different amps and tubes, etc., is simply a matter of frequency-response manipulation with varying psychoacoustic effects. 
 I'm not an electrical engineer or even an experienced audiophile, and I may very well be wrong, so feel free to ignore me._

 

I agree that it is probably the case from time to time, but in this particular thread, there is very little talk of acoustic impressions. Most of the information presented on these pages is fact and figures based. I don't own a DAC1 yet, but do plan to buy one based on my research - which has taken a few months.

 If you are serious about answering your queries, read through the 50 odd pages here as a good start to understanding the thinking and development behind the DAC1. I'm not saying it _will_ disprove what you are questioning, but it will present you with some of the slightly different techniques used by the engineers involved in developing the product, which I believe make all the difference (such as the way audio is 'reclocked' internally).

 The only way to test the validity of any product is to use it yourself, which I look forward to doing...


----------



## riverlethe

Quote:


  Originally Posted by *poo* /img/forum/go_quote.gif 
_I agree that it is probably the case from time to time, but in this particular thread, there is very little talk of acoustic impressions. Most of the information presented on these pages is fact and figures based. I don't own a DAC1 yet, but do plan to buy one based on my research - which has taken a few months.

 If you are serious about answering your queries, read through the 50 odd pages here as a good start to understanding the thinking and development behind the DAC1. I'm not saying it will disprove what you are questioning, but it will present you with some of the slightly different techniques used by the engineers involved in developing the product, which I believe make all the difference (such as the way audio is 'reclocked' internally).

 The only way to test the validity of any product is to use it yourself, which I look forward to doing..._

 

I'm not questioning the DAC1 here. I'm questioning people who feel they need MORE than the DAC1's built-in headphone amplifier (assuming the HPA2 itself isn't overkill). I'd guess they're simply altering the FR in some way that they find enjoyable, if it isn't placebo effect. I don't think that's necessarily "wrong," but I think it could be done a lot more cheaply. 
 Again, I don't have much knowledge of the subject, and I'd like someone to explain if I'm wrong.


----------



## poo

I see. I'm no more qualified than you, but would tend to agree with your last comments for the most part. Again, I have not tried the DAC1 for myself so cannot provide a personal impression, but specification and information provided by Benchmark supports your point of view.


----------



## Lord Chaos

People have different perceptions, different sensitivity, different expectations. Where you stop on the path to perfection is a function of these and other factors, with finances being a big one.

 I've been using DAC1s for about 9 months now, and am still delighted with them. I've heard a lot of audio equipment over the years and the DAC1--either the plain or USB subspecies--is solidly at the top of the heap... for my perceptual systems.


----------



## olivier

Hello again folks,

 I am wondering what are the differences between the Benchmark DAC1 and the DAC1 USB, besides the obvious USB connection and the price tag. I read Elias mentioning several upgrades in the most recent versions of the DAC1 but I don't know if these modifications apply to both DAC1 and DAC1 or to the DAC1 USB only.

 Cheers! O.


----------



## tubaman

Quote:


  Originally Posted by *riverlethe* /img/forum/go_quote.gif 
_I'm not questioning the DAC1 here. I'm questioning people who feel they need MORE than the DAC1's built-in headphone amplifier (assuming the HPA2 itself isn't overkill). I'd guess they're simply altering the FR in some way that they find enjoyable, if it isn't placebo effect. I don't think that's necessarily "wrong," but I think it could be done a lot more cheaply. 
 Again, I don't have much knowledge of the subject, and I'd like someone to explain if I'm wrong._

 

You realize where you are? This is HEAD-FI ..the need is the the question..it's the want!.


----------



## riverlethe

Quote:


  Originally Posted by *tubaman* /img/forum/go_quote.gif 
_You realize where you are? This is HEAD-FI ..the need is the the question..it's the want!._

 

Haha, yes... I didn't mean "need" in the general sense, but "need for the purpose of accurate audio reproduction," which may or may not even be a "want."


----------



## infinitesymphony

Quote:


  Originally Posted by *olivier* /img/forum/go_quote.gif 
_I am wondering what are the differences between the Benchmark DAC1 and the DAC1 USB, besides the obvious USB connection and the price tag. I read Elias mentioning several upgrades in the most recent versions of the DAC1 but I don't know if these modifications apply to both DAC1 and DAC1 or to the DAC1 USB only._

 

Same here... I've been waiting for a comparison since little-endian's post (#964) about S/PDIF versus USB; the claim is that since the DAC1 reclocks, the type and quality of the digital input do not matter.

 It's my understanding that the majority of "upgrades" took place around mid-2004, with some parts improvements here and there at later dates. But, the USB apparently has changeable gain, which allows you to tweak it for a certain pair of headphones (what euclid was talking about). If the default gain is too high, it could mean that the volume change between adjacent detents on the volume knob would be too large. With a lower gain, the knob could be turned further and with more resolution; smaller volume changes between detents.


----------



## EliasGwinn

Hey all,

 Sorry I haven't been able to keep up with you...I've been pretty loaded-down with other work lately. Its good to have an escape to Head-Fi every now and then 
	

	
	
		
		

		
			





 .

 There are several features specifically on the DAC1 USB which are not available on the DAC1, and they are as follows:

 - Selectable gain range for headphone amp
 -- Lets you select the optimal range for your specific headphones so that the volume knob can be utilized more optimally

 - Main output mutes upon headphone insertion (defeatable)
 -- The analog outputs on the rear of the DAC1 USB will be muted when you insert the headphone plug if this feature is enabled. This lets you switch to headphones without having to manually shut-down your loudspeaker system.

 - High-Current output drivers
 -- These are new OpAmps that have just come out in the last year+/-. They sound identical to the DAC1 classic output drivers, but the difference is the XLR and RCA outputs can now drive longer cables and/or low-impedance inputs and/or high-capacitance inputs without suffering loss in THD+N performance. 

 - Advanced USB Audio for true native, 96/24, bit-transparent playback
 -- No drivers or configuration necessary...plug it in and immediately get bit-transparency at rates up to and including 96/24

 - Auto-standby mode
 -- When the selected digital input no longer sees digital signal (or non-compatible digital signal), the DAC1 will begin 'Standby' mode.

 I hope that clears it up a bit.

 Thanks,
 Elias
 __________


----------



## olivier

Hi Elias, that clarifies a lot, thanks!

  Quote:


  Originally Posted by *EliasGwinn* /img/forum/go_quote.gif 
_- Auto-standby mode
 -- When the selected digital input no longer sees digital signal (or non-compatible digital signal), the DAC1 will begin 'Standby' mode.
 ___________

 

as I see it, this is the only extra feature of the USB version that I would benefit from, using the DAC1 directly plugged in a power amplifier (no long cables involved), and being fed by a SB3. Am I missing something here?

 O.


----------



## EliasGwinn

If you don't use headphones, the first two won't mean much to you. And if you don't use a computer as a source, then you won't benefit from the USB interface.

 Depending on input impedance and capacitance of your amp, you may benefit from the high-current output drivers. 

 Thanks,
 Elias


----------



## olivier

Quote:


  Originally Posted by *EliasGwinn* /img/forum/go_quote.gif 
_Depending on input impedance and capacitance of your amp, you may benefit from the high-current output drivers. _

 

Are high-current output drivers at least as good as low-current for ALL power amps? Or does which is best depend on the amp? Thanks! 
	

	
	
		
		

		
			




 you're doing such a great educative job here 
	

	
	
		
		

		
		
	


	




 Olivier


----------



## EliasGwinn

Quote:


  Originally Posted by *olivier* /img/forum/go_quote.gif 
_Are high-current output drivers at least as good as low-current for ALL power amps? Or does which is best depend on the amp? Thanks! 
	

	
	
		
		

		
		
	


	



 you're doing such a great educative job here 
	

	
	
		
		

		
		
	


	




 Olivier_

 

The first thing you said is right.

 Basically, the difference is that the output drivers of the DAC1 USB will perform exactly the same as the DAC1 drivers, except when the load is challenging. Examples of a challenging load is high capacitance or low impedance. These require circuit configurations which draw a lot of current from the output drivers. When a lot of current is being drawn from drivers, it risks an increase of distortion. The output drivers on the DAC1 USB will handle these high-current situations very gracefully without suffering significant distortion.

 Thanks,
 Elias


----------



## riverlethe

Elias, I have a few questions about my new DAC1. If they've already been answered, please refer me to the page number. 
	

	
	
		
		

		
		
	


	




 1. Do all the outputs measure as flat in frequency response as the DAC itself? I'm specifically interested in the HPA2, of course.

 2. Concerning 24-bit playback from the DAC1 manual: "The reason is that digital volume controls and digital mixers increase the word-length of the audio." Does this still apply if I have the volume at 100% and no EQ or effects applied, using Windows Vista? Are there no other advantages to 24-bit upsampling with the DAC1, or is this just referring to settings within the operating system?


----------



## EliasGwinn

Quote:


  Originally Posted by *riverlethe* /img/forum/go_quote.gif 
_1. Do all the outputs measure as flat in frequency response as the DAC itself? I'm specifically interested in the HPA2, of course._

 

The frequency response on page 24-25 of the DAC1 USB manual apply to all analog outputs, including the HPA2. 

  Quote:


  Originally Posted by *riverlethe* /img/forum/go_quote.gif 
_2. Concerning 24-bit playback from the DAC1 manual: "The reason is that digital volume controls and digital mixers increase the word-length of the audio." Does this still apply if I have the volume at 100% and no EQ or effects applied, using Windows Vista? Are there no other advantages to 24-bit upsampling with the DAC1, or is this just referring to settings within the operating system?_

 

Perhaps the premise of this situation is being mis-understood. Using the digital volume control or mixer will increase the digital 'words' to 24-bits, but that will not increase the quality to that of audio which was _recorded_ at 24-bits. In other words, it will not increase the quality of 16-bit audio to 24-bit audio. The extra eight bits are simply being used as numeric place-holders during volume manipulation, which is a mathematical operation. In the quote above, we are referring to the situation when the last 8 'place-holder' bits are cut-off. This will cause major distortion. Therefore, it is important to maintain a 24-bit 'path', even for 16-bit audio.

 If you are keeping volume at 100% and are not using DSP, then this may not be an issue for you. However, there may be other processes resulting in 24-bit digital data that could be subject to truncation. With the myriad of drivers, software, and processes happening on any given computer, one can never be sure. That is another reason why a 24-bit path is a good, safe decision.

 Thanks,
 Elias


----------



## infinitesymphony

Are there any plans to bring the improvements of the DAC1 USB to the regular DAC1? Apologies if this has been asked before.


----------



## EliasGwinn

Quote:


  Originally Posted by *infinitesymphony* /img/forum/go_quote.gif 
_Are there any plans to bring the improvements of the DAC1 USB to the regular DAC1? Apologies if this has been asked before. 
	

	
	
		
		

		
		
	


	


_

 

No. The features of the DAC1 USB will not be added to the DAC1 Classic.

 Thanks,
 Elias


----------



## riverlethe

Here's an obsessive question for you, Elias. How long is the input selector switch supposed to last?


----------



## EliasGwinn

Quote:


  Originally Posted by *riverlethe* /img/forum/go_quote.gif 
_Here's an obsessive question for you, Elias. How long is the input selector switch supposed to last?_

 

Infinitely.

 I'm kidding.

 Seriously, for a really, really, really long time.

 For example, at the studio I work at, we've got one of the original production DAC1's. We reference (A/B) our mixes against specific recordings which we are particular impressed with. We have our recording software feeding the XLR digital input and the reference CD feeding the optical input. We are constantly switching back and forth on the front of the DAC1. After 3+ years of constant switching, the switch has yet to fail.

 Hope that helps...

 Thanks,
 Elias


----------



## Lord Chaos

Now, if I could just consistently remember that "UP" goes from Mac to PC and "DOWN" goes the other way, I wouldn't get lost in the other choices.


----------



## riverlethe

Excellent, it should last long enough for my next major audiophile upgrade to be invented... Hearing implants!

 Elias, have you ever tested the X-fi on Vista64 for bit-transparency?


----------



## gjwaudio

Hi Elias
_(sorry to rattle the bones of Skeletons In The Closet 
	

	
	
		
		

		
		
	


	




, but...)_

 Did you ever come to a firm conclusion over the issue Thomas raised HERE ? If indeed _"Kmixer in Windows is actually multiplying all values by a number smaller than 1.0 at max volume"_ (as he states), how could the true bitstream get through to the DAC1 USB ?
  Quote:


  Originally Posted by *EliasGwinn* /img/forum/go_quote.gif 
_...If you are correct, if DirectSound is not capable of bit-transparency, then the Audio Precision testing device is flawed. This is not entirely impossible. However, it is a very well respected testing platform, perhaps the most respected in the audio community. Perhaps we should bring this discussion to them if we really want to determine where the inconsistencies are._

 

Have you had any communication with Audio Precision on this question ?

 What implications are there for using the just-announced ADC1 USB, via the USB connection ? If the real bits can't get through, it sure reduces the value of that feature.

 I hope I'm very wrong here, and you can show us tests that incontrovertibly prove USB to be a reliable channel for bit-perfect IN & OUT datastreams (using Windows XP PCs).

 Thanks again Elias for your continued enthusiasm in this heroic thread !
 Grant


----------



## EliasGwinn

Grant,

 We believe the Audio Precision testing equipment is accurate. We have not contacted Audio Precision on account of Thomas' claims. Audio Precision is the accepted standard in audio testing by most all manufacturers, reviewers, and other professionals.

 Aside from the bit-transparency test, we also test the sonic performance of the USB interface as confirmation. The USB interfaces perform to the same spec as the other digital interfaces.

 Thanks,
 Elias


----------



## StevieDvd

This thread has been going for so long I'll need to add it to my Christmas card list!

 Well done Elias for your endurance & patience with us
	

	
	
		
		

		
		
	


	




 Steve


----------



## dty

I have a question about the benchmark dac1, If I buy the $975 version and plug it in my soundcard in my computer will I lose sound quality vses the USB version?


----------



## EliasGwinn

dty,

 Without actually testing your specific soundcard, it's hard to know about the sound quality from the digital output of your soundcard. It really depends on the software associated with the soundcard. That is, some soundcards are not bit-transparent because of their drivers and other related software. Sometimes, a lack of bit-transparency is due to sample-rate conversion, truncation, dithering, and/or other DSP. Sometimes the result of this DSP is relatively benign, but sometimes it results in significant distortion.

 I know that doesn't answer your question completely, but, as I said, I have no way of knowing unless I actually had the card to test.

 Thanks,
 Elias


----------



## xp9433

Quote:


  Originally Posted by *gjwaudio* /img/forum/go_quote.gif 
_Hi Elias

 Did you ever come to a firm conclusion over the issue Thomas raised HERE ?_

 

gjwaudio

 I was of the belief and clear interpretation that the ball was very much in Thomas's court! Elias answered politely (in the extreme) as always. Elias had a very firm conclusion!

 It is/was very much up to Thomas to support/prove his assertions - not the other way around.

 I am sure if Thomas comes up with valid (tested/proven) technical support for his arguments we will all listen. Until then, I will trust Elias's tests and explanations thanks very much.

 Frank


----------



## schugh

Just wanted to add myself to the list of people having pops/click/dropouts through USB.

 I recently acquired a Stello DA220 MKII on a good used price and I just had to try it out and was going to sell my DAC1. But I like the DAC1 so much that instead of selling it, I upgraded to the USB version (thanks to the local dealer for letting me do that).

 However, at the moment I am little frustrated with the dropouts etc when connected to the USB. I have a dual athlon 64 running Vista ultimate with 2 GB of RAM. Granted I have several USB devices connected to the computer so maybe all that load might be the problem. (USB Mouse, three external USB HDs etc). Hopefully I can figure out what the problem is but there are times if I am doing anything at all such as burning a DVD that the DAC1 is unusable through the usb connection because of too many dropouts or whatever.

 Also I don't know if anyone else has suggested it but if not here is a suggestion. Why did Benchmark decide to use numbers for the input selector? There is enough room to put abbreviated names such as opt./coax/xlr/usb. I will eventually remember what the numbers correspond to, but I am not sure that I should have to.

 Thanks,

 -- Sanjay


----------



## poo

Very interested to know why these dropouts are occurring. USB devices can add load to any power supply, so it would be good to know if you have the same dropout issue with the majority of your other USB devices disconnected.


----------



## infinitesymphony

Quote:


  Originally Posted by *schugh* /img/forum/go_quote.gif 
_Just wanted to add myself to the list of people having pops/click/dropouts through USB._

 

Here's something you may want to try, from another thread:
  Quote:


  Originally Posted by *germanium* 
_Try going to the control panel/power options & changing the profile to high performance. This stops the processor from changing multipliers all the time. This stops the stuttering of sound in Windows Vista. Windows Vista over rides the BIOS settings in this regard so even if disable this feature in the BIOS Windows Vista will still change the multiplier if left in any other mode than high performance.

 I Have a Core 2 Duo E6600 overclocked & this helped tremendously & stopped all audio stuttering._


----------



## dmk005

I use a Mac and have never heard a pop/dropout/click I didn't instigate 
	

	
	
		
		

		
			





. 

 So perhaps this is a W-intel only issue?

 david


----------



## EliasGwinn

Quote:


  Originally Posted by *schugh* /img/forum/go_quote.gif 
_... at the moment I am little frustrated with the dropouts etc when connected to the USB. I have a dual athlon 64 running Vista ultimate with 2 GB of RAM. Granted I have several USB devices connected to the computer so maybe all that load might be the problem. (USB Mouse, three external USB HDs etc). Hopefully I can figure out what the problem is but there are times if I am doing anything at all such as burning a DVD that the DAC1 is unusable through the usb connection because of too many dropouts or whatever._

 

Sanjay,

 Having several USB devices connected will often cause this problem. Please try the DAC1 USB without the other devices connected (or just the mouse) to see if the problem persists.

 Thanks,
 Elias


----------



## schugh

Hi Everyone, thank for the replies. I do plan to try it with everything disconnected other then the mouse and the DAC1. Just a bit busy with work these days, but as soon as I get a chance I will report back my findings. I also have a couple of notebooks and will try the DAC1 with them. Then I also have the Stello with USB and my Headroom Micro DAC and will try them under the same circumstances and see if the same problem happens with them.

 infinitesymphony: I did try that trick but it does not seem to have helped me.

 -- Sanjay


----------



## mofonyx

I'm sure this has been asked before, but I can't seem to find it in this massive thread.

 I realise that there isn't an on/off switch on the DAC1, so is it perfectly fine to leave it on all the time? Would this cause increased wear and tear of the equipment?

 Thanks.

 edit: Oops, I found it immediately after posting this. Accordingly it should be fine to leave it on all the time.


----------



## Dead Ghost

I'm glad that this thread wasn't lost. Very informative thread.


----------



## yourmando

I'm so happy this thread survived. Now, some questions for Elias:

I notice that several dvd transports were tested for bit transparency, and the results were posted in the Benchmark Wiki. This is awesome, and I haven't seen systematic tests like this anywhere.

DVD Transports - Benchmark

 It looks like most tested did not fare so well. My questions here are--could you shed some light on the testing methodology? For example, were these results only for dvd-audio sources with the copy protection flag set to off?

 Also, do you plan on testing the Oppo DV-980H? This the newest Oppo universal player, and they claim that this is their best audio effort to date. There is a forthcoming Oppo DV-983H player which will have the exact same audio section as the 980, so testing the 980 will kill 2 birds with one stone (the 983 is essentially a 980 with high end ABT based deinterlacing and scaling).

Have you had a chance to test Mac OS X Leopard? I have recently upgraded, and it looks like everything is the same, including the Audio Midi Setup app that one must manually change to match the sample rate. Again, the Benchmark wiki has the best info I have seen anywhere with regard to actual measurements of bit transparency of different player software and OS's.
Thank you!


----------



## EliasGwinn

Hey Everyone!! Good to see we're back up and running!!

 Now, getting to your questions...

  Quote:


  Originally Posted by *yourmando* /img/forum/go_quote.gif 
_[*]I notice that several dvd transports were tested for bit transparency, and the results were posted in the Benchmark Wiki. This is awesome, and I haven't seen systematic tests like this anywhere.

DVD Transports - Benchmark

 It looks like most tested did not fare so well. My questions here are--could you shed some light on the testing methodology? For example, were these results only for dvd-audio sources with the copy protection flag set to off?

 Also, do you plan on testing the Oppo DV-980H? This the newest Oppo universal player, and they claim that this is their best audio effort to date. There is a forthcoming Oppo DV-983H player which will have the exact same audio section as the 980, so testing the 980 will kill 2 birds with one stone (the 983 is essentially a 980 with high end ABT based deinterlacing and scaling).
_

 

The testing was done with a DVD-A that I burned using Minnetonka DiscWelder. I honestly do not recall seeing an option with regards to the copy-protection flag. I may go back and investigate, however. That seems promising.

 We have no plans to test DVD players, but we may in the future. We are up to our eyeballs with projects, so it may be a few weeks/months.


  Quote:


  Originally Posted by *yourmando* /img/forum/go_quote.gif 
_[*]Have you had a chance to test Mac OS X Leopard? I have recently upgraded, and it looks like everything is the same, including the Audio Midi Setup app that one must manually change to match the sample rate. Again, the Benchmark wiki has the best info I have seen anywhere with regard to actual measurements of bit transparency of different player software and OS's.
_

 

We do not have a copy of Leopard yet, so we haven't had an opportunity to see its capabilities. 

 However, you may note that there are some updates to the "iTunes on Mac" page of the Benchmark Audio Wiki. We have tested iTunes 7.5, and *it does not look good!!!* In fact, even 44/16 had horrible distortion (with AudioMIDI set to 44100). Oddly enough, only 96/24 was bit-transparent with iTunes 7.5 (with AudioMIDI set to 96000). Also, the volume control on iTunes 7.5 seems to be much better then the volume control on the previously tested version (6.2, I believe...??? It says on the Benchmark Wiki). So, the sample-rate issue is worse, but the volume control is better...double-edged sword. Soon, they'll get it right...I hope!! Maybe with Leopard...???

 For all you Mac users, there is a player that plays absolutely perfectly. It is called VLC. It lacks the GUI that iTunes is very good at, but it is the best performing player, sonically. Everything works great, including the volume control. However, you still need to change the sample-rate in AudioMIDI setup. This is an issue with OSX (maybe fixed with Leopard...??? Can anyone confirm this by playing audio files with different sample rates and see if the sample-rate changes in AudioMIDI...???)

 Thanks,
 Elias


----------



## Wavelength

Elias,

 Well first off you really don't need to fool with the Midi settings. It's best to leave these alone.

 I have checked your DAC1 USB with all the settings and it does work fine on all the settings and from what I can tell on my Prism dScope III the use of Lepoard and iTunes 7.5 results in the best measurements.

 I know the USB drivers in Lepoard have been extensively changed and that resulted in me delaying my ASYNC USB release. As of today the Async code for my TAS1020 dac is released and seems to work great in all os's but best in Lepoard and Vista.

 I would suggest users of 7.5 iTunes to upgrade to Lepoard for the best results.

 One of the best ways to test your dac would be using the Faber Acoustics testing software. There you can make test tones very easily at any Fs rate.

 You can also use MAX if you want to try different Fs rate music and more easily see problem areas.

Max from sbooth.org

 I did send my DAC1 USB to Microsoft concerning the bit perfect and windows. There were able to make it bit perfect in Vista out of the box. They did have to set the device configuration for 24 option for it too be bit perfect with 16 bit data for some reason. There testing for XP was what I expected and they did say any PCM USB device would have issues with the KMIXER. Though not really a problem as there are ways around that. Though we have seen that several of the ASIO drivers seem to be corrupting data for some reason.

 As a suggestion to users out there from my experience Vista may not be your cup of tea but in the audio area this is really were great strides have been made. If you have a dedicated computer I would upgrade to Vista for the best and easiest sonic value.

 Thanks
 Gordon


----------



## pwfletcher

I recently put together the following system and have a few technical questions re the Benchmark DAC1 USB:

 System:
 Mac Mini (running Leopard) as the source component 
 -> plugged in to a two week old Benchmark DAC1 via USB
 -> plugged in to a Simaudio I-7 via Nordost Quattro Fil Balanced XLR
 -> driving Dynaudio Confidence C1s via Nordost SPM Speaker Wire.

 Also, my CDs have been ripped using Apple Lossless.

 The system sounds amazing, but I want to use the DAC1 to its fullest potential (as any good equipment tweaking obsessed audiophile would). So my question is, will there be any sonic benefit if I set the XLR attenuator jumpers to -0db as opposed to the current setting of -20db or should I leave it alone?

 Thanks


----------



## Kiep

I have tried using 0db setting with my old dac1 and Sim P-5 preamp. It worked and I think the sound was a little less restrained but the volume adjustment range was much smaller. Try it and see what happens.


----------



## poo

Quote:


  Originally Posted by *EliasGwinn* /img/forum/go_quote.gif 
_We have tested iTunes 7.5, and *it does not look good!!!*_

 

So based on your findings, what is the latest version of iTunes should we be using?


----------



## jlingo

Quote:


  Originally Posted by *audioengr* /img/forum/go_quote.gif 
_
 The point is: there are no experts in everything. We are each experts in certain areas. This is why collaborations often lead to superior products.

 Steve N._

 

I agree entirely, and there is no need to be embarrassed about our weaknesses.


----------



## ken36

The optical input till is OK with me. Audiogon is an excellent place to buy a near new DAC1. I think I bought mine for $750 shipped from a nice guy in NYC. Of course, there was a modicum of negotiation.


----------



## gregeas

Quote:


  Originally Posted by *Wavelength* /img/forum/go_quote.gif 
_There testing for XP was what I expected and they did say any PCM USB device would have issues with the KMIXER. Though not really a problem as there are ways around that. Though we have seen that several of the ASIO drivers seem to be corrupting data for some reason._

 

What about using XP with a program like J River Media Center, which permits ASIO output? This would still be bit perfect to the USB DAC1, correct?

 I actually like Media Center much better than iTunes, so I'm using it in any case. 

 I have taken to using my laptop and DAC1 in my surround system, and the sound quality is outstanding with Media Center. If there are any bits being altered, I certainly can't hear the problem.


----------



## Wavelength

Quote:


  Originally Posted by *gregeas* /img/forum/go_quote.gif 
_What about using XP with a program like J River Media Center, which permits ASIO output? This would still be bit perfect to the USB DAC1, correct?

 I actually like Media Center much better than iTunes, so I'm using it in any case. 

 I have taken to using my laptop and DAC1 in my surround system, and the sound quality is outstanding with Media Center. If there are any bits being altered, I certainly can't hear the problem._

 

Greg,

 Yes that would be bit perfect if the ASIO does not corrupt the data. We have actually found some versions and setups of the generic ASIO drivers to not be bit perfect. But at this time I cannot recall all the situations with which ones or how and why.

 But in general I found that Vista worked better because of the lack of the KMIXER and also because of the updated USB drivers (Good Job DJ!).

 Thanks
 Gordon


----------



## sangel

Hello Elias,

 Back again.....reinforced, offcourse..

 I have just recieved my new DAC1 usb and I am ready to set it up.
 But before doing so, I would be very grateful if you let me know the MAX usb cable length, for best results, between the PC's usb and the DAC1 usb input.
 Note, my PC to DAC1 usb distance is about 5 meters.
 Alternatively, do you suggest any of the market's usb - bridge type extension cables or usb - electroptical fiber type cables!!

 Your input would be very much appreciated.

 Best

 sangel


----------



## poo

Quote:


  Originally Posted by *sangel* /img/forum/go_quote.gif 
_But before doing so, I would be very grateful if you let me know the MAX usb cable length, for best results, between the PC's usb and the DAC1 usb input.
 Note, my PC to DAC1 usb distance is about 5 meters._

 

The first few points here will answer your question, and there is plenty more info too...

USB.org - FAQ: Cables, Connectors, and Networking

 In short, you should be fine.


----------



## tuffgong

This is the best thread i've ever read ever, bar none. I just wanted to thank Benchmark for participating. My question to Elias is:
 Some of the reviews mention you are always upgrading your pcb board layout. It appears from this thread that anything after may '04 has the same parts, but from what I understand the pcb layout is always being perfected, this being one of the places you engineers shine the most. Audiogon.com has an extremely strong presence in the second hand market for the dac-1, it is a steal at $750 used, and very telling that most of the dac-1's are being sold on audiogon, not ebay, probably because ebay is too inexperienced for your product. I just purchased one that is being delivered next week, the previous owner said it was made two years ago. If at all possible, could you list serial number ranges for models made before the '04 change? Also, could you detail your changes made to the pcb layout, if they were significant, and what serial number ranges have which pcb layout? I bet a lot of audiogoners would love to know what they are getting into with a used model, knowledge is power. Perhaps this is what people refer to when they say some dac-1's sound different. The parts could be the same, perhaps the specs so close to be negligible, but I believe you folks are always improving on the layout of the board, the one place to improve upon. Perhaps if you did indulge us with this info, you might sell alot of new units and see a rash of listings on audiogon, laf j/k. This thread seems to get all the facts straight, my apologies if this was adressed earlier but after page 65 I just had to fast forward... it took me two days to get to 65. Most enjoyable though. Besides the pcb info, i'd love to hear Elias' thoughts on upgrading everything outside the dac-1, does a power conditioner help? Silver Signiature blahblah IEC cord help? Putting a weight with sorbathane feet on top of the unit help? Is this all mental masturbation? I plan on buying some PMC AML-1's (active monitors) for the benchmark to run into for the purest, more revealing sound possible. If that's what they used to master it with (probably a benchmark as well), why try to improve it. I noticed Rick Rubin uses the benchmark and Stevie Wonder likes the aml's, so i'm going that direction and retiring from the upgrade game. Anyone want some platinum audio duo's and solo's for sale? Selling my amp next... 
	

	
	
		
		

		
			




 Thank you Benchmark.
 I love the review on audioreview of the benchmark that says you should have made this unit twice as big and twice as heavy and called it the merlot signiature dac and charged 3 times as much, so funny.


----------



## Telynau

Per Elias's suggestion, I downloaded VLC. Bottom line, if you want to play your albums off a hard drive (which is all I ever do), it works. 

 On one of my systems I use a Mac PowerBook Pismo (500MHz, 512MB). It won't run Leopard, so Gordon Rankin's upgrade suggestion doesn't help (I will try the Leopard upgrade on my MacBook after Leopard settles down some more).

 VLC is noticeably more clear to my ears than iTunes 7.5. 

 Two quirks of VLC really stumped me, so I will repeat the resolutions here for those who might want to try it on their Mac's. 

 First, in the Benchmark Wiki Elias says to keep the VLC volume control at 100%. OK, that is the consistent advice we hear (e.g, from Gordon, too). The quirk is that in VLC, under "VLC -- Preferences -- Audio" you will see that the volume can be set anywhere from zero to 1024. This volume control should be set at 256 because 256 = 100% or "0 db". If you set it higher you will probably add distortion because you are amplifying the signal. I made the mistake of setting it at 1024 (thinking that was 100%) and spent several hours figuring out why at that level the music was clearly distorted on my system.

 The slide volume controller on the little VLC menu that pops up when you start VLC is a different matter entirely. It varies the volume level you have set in VLC Preferences. So after you set the Preferences -- Audio level to 256, you want to set the slider on the little menu all the way to the right, which gives you the full 256.

 With these settings in place you will then use your preamp or other volume control to actually control the volume of the music.

 Second, I suspect the computer guys would have gotten this in about 10 seconds, but if you select "mute" from the Audio pull down menu (or otherwise find that you have muted the sound somehow -- which I did several times) in order to unmute you just right click on the "mute" selection. Left click puts mute on; right click unmutes.

 The big downside to VLC is that the documentation is very, very sparse. But I got it up and running -- and it is a solution to play Apple lossless files clearly, with low or no distortion, on a Mac that can't be upgraded to Leopard. If all you want to do is play your albums (and probably a lot more if you can figure out how to use the more advanced functions), it will do the job. 

 Thanks for the tip, Elias! Regards, James


----------



## thomaspf

Quote:


  Originally Posted by *xp9433* /img/forum/go_quote.gif 
_gjwaudio

 I was of the belief and clear interpretation that the ball was very much in Thomas's court! Elias answered politely (in the extreme) as always. Elias had a very firm conclusion!

 It is/was very much up to Thomas to support/prove his assertions - not the other way around.

 I am sure if Thomas comes up with valid (tested/proven) technical support for his arguments we will all listen. Until then, I will trust Elias's tests and explanations thanks very much.

 Frank_

 

What more can I do? I have tested the firmware that Benchmark is using and it does not trick the Windows driver into transparency for 16/44.1 data since that is just not possible. The firmware registers the USB device as only supporting 24 bit streams but kmixer is still slightly changing the bits due the way volume processing is implemented.

 You can easily get around this by using kernel streaming but that solution does of course work for every USB dac that uses the Windows supplied USB audio driver. So there is nothing special about the Benchmark USB code. Vista does not have kmixer but its pipeline is also not bit transparent not for the Benchmark or any other USB device. Again using kernel streaming for devices using the standard Microsoft USB driver solves this problem.

 Besides all this, the Benchmark is still a nice DAC. I thought we had closed this the last time round.


 Cheers

 Thomas

 P.S.: I am not sure I posted this on this thread but if you send a 16bit stream in the upper bits of a 24bit PCM stream and set the format in the Vista sound control panel to 24/44.1 then a 16bit stream survives in there unmodified.


----------



## fradoca

thanks thomaspf for the info!
 I live in italy and i want to buy a dac1 usb in the next few days.
 But i'd like to know if benchmark will release soon a new
 version of the dac1 ( a dac 2).The price here is 1100 euro
 and for me is a money effort.I've tried to ask to Elias
 but got no reply about a new dac.I would not buy something
 that becomes obsolete in the next 6-8 months due to the
 release of a new product
 thanks

 cheers

 Francesco


----------



## Wavelength

Thomas,

 Maybe a better way to say this is that there is not a way for any USB DAC to fool the KMIXER into bit perfect accuracy.

 Hey guys this is no big deal. Bypass the KMIXER as it is told everywhere or go to Vista for better results.

 I do agree with others though and heck I would not have bought the DAC1 if they would not have said it was bit perfect out of the box in all os's. I think they should change the marketing on that one.

 Thanks
 Gordon


----------



## Scrith

Quote:


  Originally Posted by *thomaspf* /img/forum/go_quote.gif 
_P.S.: I am not sure I posted this on this thread but if you send a 16bit stream in the upper bits of a 24bit PCM stream and set the format in the Vista sound control panel to 24/44.1 then a 16bit stream survives in there unmodified._

 

This seems like an important caveat to your argument that bit-perfect playback in Vista is not possible with USB devices. If I am reading it correctly, a USB DAC owner using Vista simply needs to set their device to 24/44.1 in the Vista Sound Control Panel (and use 24-bit output in a program such as Foobar2000) to get bit-perfect 16/44.1 sound (which is what most users care about, since CDs are 16/44.1).

 I don't think this is new information (I seem to remember doing this on the day I installed Vista for use with my DAC1 USB), but it's interesting to be reminded of how important it is to set this up properly in Vista.


----------



## Scrith

On another note, Gordon's efforts with USB Audio at Wavelength Audio have apparently yielded important results: he has asynchronous USB support working in one of his DACs.

 I am somewhat skeptical that any difference in sound quality can actually be heard between async and standard USB audio, but I haven't tested it myself so this is just speculation.

 But it is an important milestone, because in theory this feature finally makes an external DAC completely independent of the computer it is attached to for the timing of the data for playback (assuming the computer can keep up with a perfect 44.1 playback speed).

 The downside of this breakthrough is that it is only available on ridiculously expensive hardware (sorry Gordon). Maybe a company like Benchmark could license this technology and make it available at a more appropriate cost so a much greater number of people could one day appreciate it (assuming it actually sounds better)?


----------



## EliasGwinn

Hello Head-Fiers...

 I'm sorry I haven't responded lately, but my email notification doesn't seem to be working
	

	
	
		
		

		
			





 ??

 Anyway, I'll try to answer all of your questions as soon as possible.

 Thanks!
 Elias


----------



## fradoca

hi,
 sorry for the question..maybe it's a stupid question.
 I'd like to buy a dac 1 usb.I mainly work with wavelab and soudforge.
 If i set Asio Kernel Streaming driver in wavelab to bypass kmixer and i have my
 usb dac 1 connected to the pc will i be able to listen to the music/wav files
 played in wavelab and soundforge??
 Sorry if this has been already asked but this issue of kmixer and
 usb devices is driving me mad!
 thanks to everybody

 cheers

 Francesco


----------



## EliasGwinn

Quote:


  Originally Posted by *Wavelength* /img/forum/go_quote.gif 
_Elias,

 Well first off you really don't need to fool with the Midi settings. It's best to leave these alone._

 

I'm not sure what you're referring to, but I'm assuming it has to do with AudioMIDI Setup. No, there are no MIDI settings that are applicable. With regards to the DAC1 USB, all applicable settings in AudioMIDI setup are under the 'Audio' tab. 

  Quote:


  Originally Posted by *Wavelength* /img/forum/go_quote.gif 
_I have checked your DAC1 USB with all the settings and it does work fine on all the settings and from what I can tell on my Prism dScope III the use of Lepoard and iTunes 7.5 results in the best measurements....

 ...I would suggest users of 7.5 iTunes to upgrade to Lepoard for the best results._

 

Can you please elaborate? I'm very curious about Leopard. What did you test for? What were your results, specifically?

  Quote:


  Originally Posted by *Wavelength* /img/forum/go_quote.gif 
_One of the best ways to test your dac would be using the Faber Acoustics testing software. There you can make test tones very easily at any Fs rate.

 You can also use MAX if you want to try different Fs rate music and more easily see problem areas.

Max from sbooth.org_

 

We use Audio Precision for all waveform generation and testing. I'll look into these, however.

  Quote:


  Originally Posted by *Wavelength* /img/forum/go_quote.gif 
_I did send my DAC1 USB to Microsoft concerning the bit perfect and windows. There were able to make it bit perfect in Vista out of the box. They did have to set the device configuration for 24 option for it too be bit perfect with 16 bit data for some reason. There testing for XP was what I expected and they did say any PCM USB device would have issues with the KMIXER. Though not really a problem as there are ways around that. Though we have seen that several of the ASIO drivers seem to be corrupting data for some reason._

 

You sent a Benchmark DAC1 USB to Microsoft? They already have one...we gave them one during development so that we could simultaneously analyze the results as the firmware development progressed. However, before development began with the DAC1 USB, the engineers at Microsoft assured us that bit-transparency was, in fact, possible with XP at resolutions up to 192 kHz, 24-bit, through Kmixer. In fact, this was the basis for our quest to develop firmware that achieved bit-transparency at high-resolution through kmixer. If they had believed it was not possible, we would have never attempted it. But, alas, they said it was possible, and we have found it to be accurate. We also found similar problems with ASIO, as you mentioned.

  Quote:


  Originally Posted by *Wavelength* /img/forum/go_quote.gif 
_As a suggestion to users out there from my experience Vista may not be your cup of tea but in the audio area this is really were great strides have been made. If you have a dedicated computer I would upgrade to Vista for the best and easiest sonic value._

 

My Beta version of Vista had a weird behavioral quirk where it would sample-rate convert _everything_ to the highest possible sample rate, even if the appropriate sample rate was selected. Microsoft engineers indicated that this was intentional (although I can't remember why...I'll have to go back and check my emails...) However, we haven't tested the official release of Vista yet, so I'll keep you posted as to what we find.

 Thanks,
 Elias


----------



## EliasGwinn

Quote:


  Originally Posted by *pwfletcher* /img/forum/go_quote.gif 
_I recently put together the following system and have a few technical questions re the Benchmark DAC1 USB:

 System:
 Mac Mini (running Leopard) as the source component 
 -> plugged in to a two week old Benchmark DAC1 via USB
 -> plugged in to a Simaudio I-7 via Nordost Quattro Fil Balanced XLR
 -> driving Dynaudio Confidence C1s via Nordost SPM Speaker Wire.

 Also, my CDs have been ripped using Apple Lossless.

 The system sounds amazing, but I want to use the DAC1 to its fullest potential (as any good equipment tweaking obsessed audiophile would). So my question is, will there be any sonic benefit if I set the XLR attenuator jumpers to -0db as opposed to the current setting of -20db or should I leave it alone?

 Thanks _

 

pwfletcher, 

 We recently wrote about this in our newsletter, "Feedback". You can read it here:

Benchmark Media Systems -- Feedback Newsletter

 The monthly "Feedback" newsletter contains two tech articles that answer common questions and concerns, or addresses critical issues with audio technology (there is no advertising-type garbage in the newsletter...it is purely informational). We will be sending the next edition of the newsletter very soon. You can sign up to receive the newsletter at the bottom-left of our homepage:

Benchmark Media Systems -- Precision Audio Elecronics

 Thanks,
 Elias


----------



## EliasGwinn

Quote:


  Originally Posted by *poo* /img/forum/go_quote.gif 
_So based on your findings, what is the latest version of iTunes should we be using?_

 

Poo,

 I can't answer that for sure. I can tell you that version 6.0 for Mac was not perfect (by a long shot), but it was at least capable of bit-transparent playback for 16-bit audio (as long as the sample-rate was set appropriately).

 However, the current Mac/Tiger version (7.5) only seems to play 96/24 bit-transparently now. It is really quite odd. 

 However, we haven't completed testing yet. We will be updating our Audio Wiki as more information becomes available.

 Does anyone know where old versions of iTunes can be found? I'd like to test different versions, but Mac only offers the latest.

 Thanks,
 Elias


----------



## EliasGwinn

Quote:


  Originally Posted by *gregeas* /img/forum/go_quote.gif 
_What about using XP with a program like J River Media Center, which permits ASIO output? This would still be bit perfect to the USB DAC1, correct?

 I actually like Media Center much better than iTunes, so I'm using it in any case. 

 I have taken to using my laptop and DAC1 in my surround system, and the sound quality is outstanding with Media Center. If there are any bits being altered, I certainly can't hear the problem._

 

We haven't tested J River yet. However, an option for ASIO output will not necessarily cause bit-errors. In fact, it is usually the ASIO driver for an audio device that causes the errors, not the media player. I will try to test J River and let you know what I find.

 Thanks,
 Elias


----------



## EliasGwinn

Quote:


  Originally Posted by *sangel* /img/forum/go_quote.gif 
_Hello Elias,

 Back again.....reinforced, offcourse..

 I have just recieved my new DAC1 usb and I am ready to set it up.
 But before doing so, I would be very grateful if you let me know the MAX usb cable length, for best results, between the PC's usb and the DAC1 usb input.
 Note, my PC to DAC1 usb distance is about 5 meters.
 Alternatively, do you suggest any of the market's usb - bridge type extension cables or usb - electroptical fiber type cables!!

 Your input would be very much appreciated.

 Best

 sangel_

 


 Sangel,

 5 meters is specified as the maximum USB cable length. However, we have not tested this, nor have we tested bridge extention cables or electrooptical fiber cables. If you try any of these, please let us know what you find! (It will either work or not work...and you'll know if it doesn't work because the DAC1 USB will indicate an error and no audio will come out). 

 Thanks,
 Elias


----------



## thomaspf

Quote:


  Originally Posted by *EliasGwinn* 
_ However, before development began with the DAC1 USB, the engineers at Microsoft assured us that bit-transparency was, in fact, possible with XP at resolutions up to 192 kHz, 24-bit, through Kmixer. In fact, this was the basis for our quest to develop firmware that achieved bit-transparency at high-resolution through kmixer. If they had believed it was not possible, we would have never attempted it. But, alas, they said it was possible, and we have found it to be accurate. We also found similar problems with ASIO, as you mentioned.
_

 

Now that I a bold statement I seriously doubt! Would you be in a positon to share the name of the person who told you that in a PM or is that a secret?

 I'd just want to check whether that is an official Microsoft position ...

 As I stated before in my tests the firmware in the DAC1 can't seem to impact how Windows works internally and therefore I can't seem to reproduce bit perfect playback other than using kernel streaming.

 Cheers

 Thomas

 P.S.: Or just send me your phone number and we can chat a bit more about this


----------



## EliasGwinn

Quote:


  Originally Posted by *tuffgong* /img/forum/go_quote.gif 
_
 Some of the reviews mention you are always upgrading your pcb board layout. It appears from this thread that anything after may '04 has the same parts, but from what I understand the pcb layout is always being perfected, this being one of the places you engineers shine the most. ... I just purchased one that is being delivered next week, the previous owner said it was made two years ago. If at all possible, could you list serial number ranges for models made before the '04 change? Also, could you detail your changes made to the pcb layout, if they were significant, and what serial number ranges have which pcb layout? _

 

Tuffgong,

 All information pertaining to the evolution of the PCB can be found at the beginning of the manual for the DAC1.


  Quote:


  Originally Posted by *tuffgong* /img/forum/go_quote.gif 
_ Besides the pcb info, i'd love to hear Elias' thoughts on upgrading everything outside the dac-1, does a power conditioner help? Silver Signiature blahblah IEC cord help? Putting a weight with sorbathane feet on top of the unit help? Is this all mental masturbation? I plan on buying some PMC AML-1's (active monitors) for the benchmark to run into for the purest, more revealing sound possible. If that's what they used to master it with (probably a benchmark as well), why try to improve it. I noticed Rick Rubin uses the benchmark and Stevie Wonder likes the aml's, so i'm going that direction and retiring from the upgrade game. Anyone want some platinum audio duo's and solo's for sale? Selling my amp next... 
	

	
	
		
		

		
		
	


	



 Thank you Benchmark.
 I love the review on audioreview of the benchmark that says you should have made this unit twice as big and twice as heavy and called it the merlot signiature dac and charged 3 times as much, so funny._

 

Well, the short answer to this is that I think you are going in the right directly with professional equipment. It will usually provide the most accurate representation of the audio, as that is the name of the game in pro audio. 

 As for the fancy power cables, it may make a difference with some products but it will not affect the power that the components in the DAC1 receive. In other words, it won't make a difference with our products. I definitely discourage power conditioners. Not only will they not help, but will often hinder the delivery of power. As for the feet you mentioned, no comment.

 As for your post-script, it is often a marketing strategy for some companies to boost the price of their products to imply superiority. The kicker is that it often works. People often assume quality comes with cost...because it makes sense. What they don't realize is that it is easy to increase the price, but it is not easy to increase quality. That is why you should know who is designing the circuits, not the advertisements. 

 The DAC1 will stand up to most any D/A, spec for spec. I say "most" because there are hundreds of DAC's I've never tested. However, we have tested 5-digit DAC's that could not hang with the DAC1. The DAC1 achieves near the current theoretical limits of performance _even in the most adverse of conditions_ - and that is what sets us apart.

 Benchmark products are always priced based on two factors: the amount of development time and the cost of the components. We will not inflate our prices as a marketing device.

 Thanks,
 Elias


----------



## EliasGwinn

Quote:


  Originally Posted by *Telynau* /img/forum/go_quote.gif 
_Per Elias's suggestion, I downloaded VLC. Bottom line, if you want to play your albums off a hard drive (which is all I ever do), it works. 

 On one of my systems I use a Mac PowerBook Pismo (500MHz, 512MB). It won't run Leopard, so Gordon Rankin's upgrade suggestion doesn't help (I will try the Leopard upgrade on my MacBook after Leopard settles down some more).

 VLC is noticeably more clear to my ears than iTunes 7.5. 

 Two quirks of VLC really stumped me, so I will repeat the resolutions here for those who might want to try it on their Mac's. 

 First, in the Benchmark Wiki Elias says to keep the VLC volume control at 100%. OK, that is the consistent advice we hear (e.g, from Gordon, too). The quirk is that in VLC, under "VLC -- Preferences -- Audio" you will see that the volume can be set anywhere from zero to 1024. This volume control should be set at 256 because 256 = 100% or "0 db". If you set it higher you will probably add distortion because you are amplifying the signal. I made the mistake of setting it at 1024 (thinking that was 100%) and spent several hours figuring out why at that level the music was clearly distorted on my system.

 The slide volume controller on the little VLC menu that pops up when you start VLC is a different matter entirely. It varies the volume level you have set in VLC Preferences. So after you set the Preferences -- Audio level to 256, you want to set the slider on the little menu all the way to the right, which gives you the full 256.

 With these settings in place you will then use your preamp or other volume control to actually control the volume of the music.

 Second, I suspect the computer guys would have gotten this in about 10 seconds, but if you select "mute" from the Audio pull down menu (or otherwise find that you have muted the sound somehow -- which I did several times) in order to unmute you just right click on the "mute" selection. Left click puts mute on; right click unmutes.

 The big downside to VLC is that the documentation is very, very sparse. But I got it up and running -- and it is a solution to play Apple lossless files clearly, with low or no distortion, on a Mac that can't be upgraded to Leopard. If all you want to do is play your albums (and probably a lot more if you can figure out how to use the more advanced functions), it will do the job. 

 Thanks for the tip, Elias! Regards, James_

 

Thank you for this valuable input, James!! 

 I will investigate this and add it to our wiki (I'll give you a nod, as well, if you don't mind!!)

 Thanks,
 Elias


----------



## EliasGwinn

Quote:


  Originally Posted by *thomaspf* /img/forum/go_quote.gif 
_...I have tested the firmware that Benchmark is using and it does not trick the Windows driver into transparency for 16/44.1 data since that is just not possible. The firmware registers the USB device as only supporting 24 bit streams but kmixer is still slightly changing the bits due the way volume processing is implemented._

 

Thomas,

 I would like to know more about your tests. Could you please explain the method you used to achieve your conclusion?

 Thanks,
 Elias


----------



## EliasGwinn

Quote:


  Originally Posted by *fradoca* /img/forum/go_quote.gif 
_thanks thomaspf for the info!
 I live in italy and i want to buy a dac1 usb in the next few days.
 But i'd like to know if benchmark will release soon a new
 version of the dac1 ( a dac 2).The price here is 1100 euro
 and for me is a money effort.I've tried to ask to Elias
 but got no reply about a new dac.I would not buy something
 that becomes obsolete in the next 6-8 months due to the
 release of a new product
 thanks

 cheers

 Francesco_

 

I replied to your PM, but I would also like to state this publicly.

 It is our company policy that we do not discuss future products. I don't want to seem coy or aloof about this, but we learned this lesson the hard way.

 Our default reply to this question is, "We currently have no announcements regarding new products." Again, please don't be put off by this. It is very important that we follow this protocol.

 Thanks,
 Elias


----------



## fradoca

Quote:


  Originally Posted by *EliasGwinn* /img/forum/go_quote.gif 
_I replied to your PM, but I would also like to state this publicly.

 It is our company policy that we do not discuss future products. I don't want to seem coy or aloof about this, but we learned this lesson the hard way.

 Our default reply to this question is, "We currently have no announcements regarding new products." Again, please don't be put off by this. It is very important that we follow this protocol.

 Thanks,
 Elias_

 


 Ok Elias thanks for the reply.Mine it was just curiosity before purchasing
 such a fantastic product as yours...no intention to mess with 
 your company policy...really


----------



## EliasGwinn

Quote:


  Originally Posted by *Wavelength* /img/forum/go_quote.gif 
_Thomas,

 Maybe a better way to say this is that there is not a way for any USB DAC to fool the KMIXER into bit perfect accuracy.

 Hey guys this is no big deal. Bypass the KMIXER as it is told everywhere or go to Vista for better results.

 I do agree with others though and heck I would not have bought the DAC1 if they would not have said it was bit perfect out of the box in all os's. I think they should change the marketing on that one.

 Thanks
 Gordon_

 

Gordon, 

 I think you are misunderstanding the premise. We are not claiming to 'fool' kmixer. Kmixer does not need to be fooled. (Thomas seems to contend otherwise, and I plan to work with him to determine where our differences originate). Our tests indicate (and Microsoft has said) that bit-transparency is completely possible through kmixer. This has nothing to do with marketing, just like listing a THD+N spec or other performance specs. Please don't confuse this.

 Thanks,
 Elias


----------



## EliasGwinn

Quote:


  Originally Posted by *Scrith* /img/forum/go_quote.gif 
_On another note, Gordon's efforts with USB Audio at Wavelength Audio have apparently yielded important results: he has asynchronous USB support working in one of his DACs.

 I am somewhat skeptical that any difference in sound quality can actually be heard between async and standard USB audio, but I haven't tested it myself so this is just speculation.

 But it is an important milestone, because in theory this feature finally makes an external DAC completely independent of the computer it is attached to for the timing of the data for playback (assuming the computer can keep up with a perfect 44.1 playback speed).

 The downside of this breakthrough is that it is only available on ridiculously expensive hardware (sorry Gordon). Maybe a company like Benchmark could license this technology and make it available at a more appropriate cost so a much greater number of people could one day appreciate it (assuming it actually sounds better)?_

 

My understanding of ASYNC USB is that the USB-audio device dictates the sample rate, and the OS must sync to the USB-audio device's clock. Because, if so, this would put the computer into sample-rate conversion mode to accommodate the exact frequency of USB device's clock. As that correct, Gordon, or am I misunderstanding something? 

 Whereas SYNC USB would allow the computer to operate at the native sample-rate of the recording, and the USB device would simply accept the data at the sample-rate it is received. This mode (sync) will cause the digital audio to have more jitter, and this is a problem for DAC's that are affected by jitter. Luckily, the DAC1 USB is not affected by jitter, so it can run in this mode flawlessly. Also, sample-rate conversion is not invoked, so there is less DSP distortion.

 ...but perhaps I'm misunderstanding the ASYNC mode.

 Thanks,
 Elias


----------



## EliasGwinn

Quote:


  Originally Posted by *thomaspf* /img/forum/go_quote.gif 
_Now that I a bold statement I seriously doubt! Would you be in a positon to share the name of the person who told you that in a PM or is that a secret?

 I'd just want to check whether that is an official Microsoft position ...

 As I stated before in my tests the firmware in the DAC1 can't seem to impact how Windows works internally and therefore I can't seem to reproduce bit perfect playback other than using kernel streaming.

 Cheers

 Thomas

 P.S.: Or just send me your phone number and we can chat a bit more about this_

 

Thomas,

 I'll have to go back through my old emails and find out. In the mean time, please check my last post to you, re: your testing methods. I am very interested in your findings. 

 Thanks,
 Elias


----------



## EliasGwinn

I'm still not getting email notifications when someone posts to this thread. I didn't have this problem before. Is anyone else having this problem?

 Thanks,
 Elias


----------



## Wavelength

Quote:


 I'm not sure what you're referring to, but I'm assuming it has to do with AudioMIDI Setup. No, there are no MIDI settings that are applicable. With regards to the DAC1 USB, all applicable settings in AudioMIDI setup are under the 'Audio' tab. 
 

Elias, yes that's what I mean don't mess with any of the options in the AudioMidi setup Audio tab.

  Quote:


 Can you please elaborate? I'm very curious about Leopard. What did you test for? What were your results, specifically? 
 

G5 Dual OSX10.5.1=iTunes 7.5=>DAC 1 USB=>Prism dScope III

 I have 16 and 24 bit digital created oscillator files in iTunes. I can output these and then test for drop outs, thd and other things. I also have 16 and 24 bit JTEST files I can send over and test.

 I personally use Faber to test most of my stuff. It's oscillator and combined testing ability are really worth the $. On the PC I use Dr. Jordan Audio Test Suite the same way.

  Quote:


 We use Audio Precision for all waveform generation and testing. I'll look into these, however. 
 

The problem with the Ap is that you cannot do any native real USB Audio testing. They require you to play files in an application like iTunes, Media Player. The problem with that is that these can corrupt the data before it actually gets to the USB driver and therefore the data is not responsible.

 In the Prism software it can kernel stream to any device on the computer and therefore can fully test natively any USB Audio product.

  Quote:


 You sent a Benchmark DAC1 USB to Microsoft? They already have one...we gave them one during development so that we could simultaneously analyze the results as the firmware development progressed. However, before development began with the DAC1 USB, the engineers at Microsoft assured us that bit-transparency was, in fact, possible with XP at resolutions up to 192 kHz, 24-bit, through Kmixer. In fact, this was the basis for our quest to develop firmware that achieved bit-transparency at high-resolution through kmixer. If they had believed it was not possible, we would have never attempted it. But, alas, they said it was possible, and we have found it to be accurate. We also found similar problems with ASIO, as you mentioned. 
 

Yes basically because of your claim of bit perfection. Hakon Strande who is in charge of the Windows Sound Group told me the following statement.

  Quote:


 There is no way in XP to get bit perfect data through a USB PCM device without bypassing the KMIXER. 
 

I asked if he wanted to check out the Benchmark and he said he didn't need too. So I sent it too Thomas who did the testing in the Audio Test lab and concluded that Hakon statement was true in XP.

  Quote:


 I think you are misunderstanding the premise. We are not claiming to 'fool' kmixer. Kmixer does not need to be fooled. (Thomas seems to contend otherwise, and I plan to work with him to determine where our differences originate). Our tests indicate (and Microsoft has said) that bit-transparency is completely possible through kmixer. This has nothing to do with marketing, just like listing a THD+N spec or other performance specs. Please don't confuse this. 
 

Not possible as per Hakon statement.

 BUT in Vista if you had the 24 bit option checked and sent 16 bit padded data to 24 bits then Vista did do bit perfect data without any drivers, bypassing or other BS that we have to do in XP.

  Quote:


 My Beta version of Vista had a weird behavioral quirk where it would sample-rate convert _everything_ to the highest possible sample rate, even if the appropriate sample rate was selected. Microsoft engineers indicated that this was intentional (although I can't remember why...I'll have to go back and check my emails...) However, we haven't tested the official release of Vista yet, so I'll keep you posted as to what we find. 
 

Why don't you have Vista? Vista and Lepoard are extensively better at handling audio than their predisessors. I would say it's a requirement to have these in your testing suite.

 Thanks
 Gordon


----------



## Wavelength

Quote:


  Originally Posted by *EliasGwinn* /img/forum/go_quote.gif 
_My understanding of ASYNC USB is that the USB-audio device dictates the sample rate, and the OS must sync to the USB-audio device's clock. Because, if so, this would put the computer into sample-rate conversion mode to accommodate the exact frequency of USB device's clock. As that correct, Gordon, or am I misunderstanding something?_

 

Elias,

 No there is no sample rate conversion. Basically the feedback pipe dictates a speed control of the amout of data to send. The CPORT of the TAS1020 therefore does not change and can be rock solid. Or as I am doing with my stuff is piping a 0.5ps low jitter MASTER clock into MCLKin. Thereby bypassing the internally developed clocks (which are now stopped to keep the system quite) and derive the Bit Clock and Word Clock from them.

 ~~~~~~~~~~~

  Quote:


 Originally Posted by Scrith View Post
 On another note, Gordon's efforts with USB Audio at Wavelength Audio have apparently yielded important results: he has asynchronous USB support working in one of his DACs.

 I am somewhat skeptical that any difference in sound quality can actually be heard between async and standard USB audio, but I haven't tested it myself so this is just speculation.

 But it is an important milestone, because in theory this feature finally makes an external DAC completely independent of the computer it is attached to for the timing of the data for playback (assuming the computer can keep up with a perfect 44.1 playback speed).

 The downside of this breakthrough is that it is only available on ridiculously expensive hardware (sorry Gordon). Maybe a company like Benchmark could license this technology and make it available at a more appropriate cost so a much greater number of people could one day appreciate it (assuming it actually sounds better)? 
 

Scrith,

 Actually you can cruise over to AudioAsylum there are some really great comments on ASYNC and how much better it sounds under the PC Audio section of their forums.

 I can't really talk about what is going to happen with ASYNC but I can tell you I have four new ASYNC products that are going to be released at CES and some maybe better suited to your price point.

 Thanks
 Gordon


----------



## infinitesymphony

Sorry to briefly interrupt the ASYNC discussion, since it's a potentially interesting development. 
	

	
	
		
		

		
			





  Quote:


  Originally Posted by *EliasGwinn* /img/forum/go_quote.gif 
_We recently wrote about this in our newsletter, "Feedback". You can read it here:

Benchmark Media Systems -- Feedback Newsletter_

 

This is a pretty good article about adjusting gain staging using the DAC1's output attenuator. It makes great sense to adjust the gains this way, especially with the DAC1 as the preamplifier in an active monitor setup (which is how I would / will be using the DAC1). It will also prevent me from blowing out my ears with a bump of the volume knob. 
	

	
	
		
		

		
		
	


	




 However, I remember reading in the manual for Mackie HR824s that turning down the input sensitivity knob on the monitors could affect the frequency response and noise floor, and that it should be left up... Two conflicting opinions, maybe?


----------



## poo

Quote:


  Originally Posted by *EliasGwinn* /img/forum/go_quote.gif 
_Does anyone know where old versions of iTunes can be found? I'd like to test different versions, but Mac only offers the latest.

 Thanks,
 Elias_

 

Hey Elias - many thanks for your response! I'm the Australian guy that emailed recently regarding the international sales restrictions on Benchmark product. I really appreciate your efforts in supporting your product and the end users that purchase it.

 The following will provide you older versions of iTunes - _please_ let me know your findings - had confirmation my DAC1 shipped yesterday and can't wait!

 Mac versions:
Old Version of iTunes for Mac Download

 Windows versions:
Old Version of iTunes Download

 Look forward to the answer... fingers crossed Apple haven't screwed this up!


----------



## EliasGwinn

Quote:


  Originally Posted by *Wavelength* /img/forum/go_quote.gif 
_Elias, yes that's what I mean don't mess with any of the options in the AudioMidi setup Audio tab._

 

It is very important to set the sample-rate to correspond to the sample-rate of the audio you are playing. This is found in the Audio tab of AudioMIDI setup. If you don't set this properly, the audio will be sample-rate converted. I would not discourage anyone from utilizing this option.

  Quote:


  Originally Posted by *Wavelength* /img/forum/go_quote.gif 
_G5 Dual OSX10.5.1=iTunes 7.5=>DAC 1 USB=>Prism dScope III
_

 

What were your results from testing Leopard?

  Quote:


  Originally Posted by *Wavelength* /img/forum/go_quote.gif 
_
 Yes basically because of your claim of bit perfection. Hakon Strande who is in charge of the Windows Sound Group told me the following statement....

 ...I asked if he wanted to check out the Benchmark and he said he didn't need too. So I sent it too Thomas who did the testing in the Audio Test lab and concluded that Hakon statement was true in XP.

 ...Not possible as per Hakon statement.
_

 

That is interesting. I will try to determine who from Microsoft told us differently. I'm very curious why there are conflicting statements from within Microsoft.

  Quote:


  Originally Posted by *Wavelength* /img/forum/go_quote.gif 
_
 Why don't you have Vista? Vista and Lepoard are extensively better at handling audio than their predisessors. I would say it's a requirement to have these in your testing suite._

 

I do have Vista. I haven't tested its audio capabilities yet. Its hard to keep up with continuously testing other companies product performance because we've been busy working on our own products, specifically the ADC1 USB.

 Thanks,
 Elias


----------



## EliasGwinn

Quote:


  Originally Posted by *infinitesymphony* /img/forum/go_quote.gif 
_However, I remember reading in the manual for Mackie HR824s that turning down the input sensitivity knob on the monitors could affect the frequency response and noise floor, and that it should be left up... Two conflicting opinions, maybe?_

 

No, the input sensitivity that they speak of is an input attenuation circuit. This does not affect the gain. The gain is fixed on this, and most, amplifier input stages, and the volume knob simply attenuates. Ideally, the input section would have variable or low-fixed gain, and the source can provide the gain to maximize signal-to-noise between components. However, amplifiers need to accommodate sources that don't have high-outputs, so they build in tons of gain. 

 Thanks,
 Elias


----------



## EliasGwinn

Quote:


  Originally Posted by *poo* /img/forum/go_quote.gif 
_Hey Elias - many thanks for your response! I'm the Australian guy that emailed recently regarding the international sales restrictions on Benchmark product. I really appreciate your efforts in supporting your product and the end users that purchase it.

 The following will provide you older versions of iTunes - please let me know your findings - had confirmation my DAC1 shipped yesterday and can't wait!

 Mac versions:
Old Version of iTunes for Mac Download

 Windows versions:
Old Version of iTunes Download

 Look forward to the answer... fingers crossed Apple haven't screwed this up! 
	

	
	
		
		

		
		
	


	


_

 

Awesome!! Great work, Poo!! I'll test them when I get an opportunity. Hopefully it will be this week.

 As a temp solution, I know 6.0 will do 44/16 bit-transparently AS LONG AS YOU HAVE 41000 SELECTED IN AudioMIDI Setup. This is very important, because the SRC (sample-rate conversion) is horrible.

 Thanks,
 Elias


----------



## Wavelength

Quote:


  Originally Posted by *EliasGwinn* /img/forum/go_quote.gif 
_It is very important to set the sample-rate to correspond to the sample-rate of the audio you are playing. This is found in the Audio tab of AudioMIDI setup. If you don't set this properly, the audio will be sample-rate converted. I would not discourage anyone from utilizing this option._

 

Elias,

 Actually that is not true unless you are playing 44.1 and try and force it to another sample rate. Otherwise the USB interface will follow how it is opened.

 This is the way it works on a mac which is different from a PC. When you hit play the interface is opened to the USB driver which is told the FS rate. If you hit pause the interface is closed. You can watch this easily on a USB monitor or if you want have one of the GPIO programmed to set an led or something. If your songs continue and do not change Fs then the interface is held open (or active).

 If you change Fs rates because of track or what ever the the interface will be closed it will do a change Fs request to the dac and then open the interface again making it active at the new Fs rate.

 ~~~~~~~~~

 In regards to the KMIXER here is something to think about. I have listened to your dac with and without the KMIXER and it sounds much better with the KMIXER bypassed. This is an easy test anyone can do.

 If it sounds better with the KMIXER bypassed then how can it be bit perfect when it's not?

 ~~~~~~~~~

 Vista handles audio much better than XP does. It's also very different and therefore should be evaluated to assure your customers are not running into problems.

 Thanks
 Gordon


----------



## korben_dallas

I have a question about async usb..

 How do you handle underrun condition? Can you always guarantee the PC will keep up with the DAC and feed it enough data?


----------



## thomaspf

That question applies to anything that requires real time response from a PC. If you hog down the CPU completely so your application can not render enough data to send an audio stream at say 44.1Khz you will get drop outs.

 However, this also holds true for a PCI sound card where the crystal on the sound card determines the rate of playback.


 In async mode the backchannel from the USB device tells the PC how many samples per transmission to send to the device. This is equivalent to a sound card requesting another audio buffer from the app. Either you have enough cycles to keep real time conditions or not.

 Cheers

 Thomas


----------



## Telynau

Elias, here is where I got the info on the VLC volume control setting (VLC -- Preferences -- Audio). The post with the information is the fourth one down, by the Moderator, DJ. 

The VideoLAN Forums &bull; View topic - VLC vs. Winamp or MPC (sound quality issue)

 Regards, James


----------



## Wavelength

Quote:


  Originally Posted by *korben_dallas* /img/forum/go_quote.gif 
_I have a question about async usb..

 How do you handle underrun condition? Can you always guarantee the PC will keep up with the DAC and feed it enough data?_

 

In ASYNC especially with the TAS1020 it's actually much easier to assure that you are not underrun or overrun because instead of the PC controlling your buffering like ADAPTIVE does. The DAC is actually controlling the flow control.

 In my new code for the beta I kinda left it like ADAPTIVE did which basically has like 1 entire frame in reserve. This caused a little poping when the system was overloaded.

 So I optimized the code to assure there is always much more than needed data so the computer can be off and the internal buffer can make up for it.

 The TAS1020 has 1304 bytes of data. You can't use all that because it has some issues with DMA. But you can use most of it and that is were ASYNC really makes it's case.

 Thanks
 Gordon


----------



## EliasGwinn

I've just received some very interesting information from an Apple engineer about the operation of iTunes / Core Audio. I'm going to do some testing, and post the conclusions soon. 

 For now, I just want to let you all know this:

 If you are using iTunes 7.0 or higher, it is important to set the sample-rate in AudioMIDI *BEFORE OPENING ITUNES*. If you have iTunes open, close it, then set the sample-rate in AudioMIDI. Apparently iTunes locks to the sample-rate that is set in AudioMIDI when iTunes is launched, and the sample-rate can not be changed until iTunes is closed and re-launched.

 I'll let you know as I find out more.

 Thanks,
 Elias


----------



## poo

Quote:


  Originally Posted by *EliasGwinn* /img/forum/go_quote.gif 
_I've just received some very interesting information from an Apple engineer about the operation of iTunes / Core Audio. I'm going to do some testing, and post the conclusions soon. 

 For now, I just want to let you all know this:

 If you are using iTunes 7.0 or higher, it is important to set the sample-rate in AudioMIDI *BEFORE OPENING ITUNES*. If you have iTunes open, close it, then set the sample-rate in AudioMIDI. Apparently iTunes locks to the sample-rate that is set in AudioMIDI when iTunes is launched, and the sample-rate can not be changed until iTunes is closed and re-launched.

 I'll let you know as I find out more.

 Thanks,
 Elias_

 

Great update Elias - keep 'em coming! 
	

	
	
		
		

		
			





 Would also be very helpful to have the same questions answered for running iTunes under a windows environment (XP in my case) as I run an OSX rig and an XP rig. I presume AudioMIDI is OSX only - is there an XP equivalent or other setting in XP that needs similar attention?

 Thanks again Elias - I know you have a myriad of other things to do besides answering the many questions that Head-fi throws at you!


----------



## tuffgong

Hello, I just got my Dac-1 in the mail today, I just want to say it sounds terrific, it completely blackens the background. Much louder/lower bass, the windows rattle with my little duo's now. I switched the jumper on the inside to do coax only, only to my chagrin realizing I was not properly grounded while doing it. I'm curious what if anything would be damaged, is there a simpler cure rather than buying a grounding wristband before moving the jumper (nitrile gloves?). What would it cost to have your team fix a unit that was affected by this? I thought I was doing the right thing with the jumper, who knows maybe it's all buggered now, or maybe things go wrong rarely with this issue. I certainly think it sounds fabulous, but what do I know. Thanks again, even my silly mp3's sound great.


----------



## sangel

Hello Elias,

 Thanks for your response.
 By the way, if you remember, what was the max length of your usb cable under testing with the DAC1 usb!!

 Thanks

 sangel


----------



## Wavelength

Quote:


  Originally Posted by *EliasGwinn* /img/forum/go_quote.gif 
_I've just received some very interesting information from an Apple engineer about the operation of iTunes / Core Audio. I'm going to do some testing, and post the conclusions soon. 

 For now, I just want to let you all know this:

 If you are using iTunes 7.0 or higher, it is important to set the sample-rate in AudioMIDI *BEFORE OPENING ITUNES*. If you have iTunes open, close it, then set the sample-rate in AudioMIDI. Apparently iTunes locks to the sample-rate that is set in AudioMIDI when iTunes is launched, and the sample-rate can not be changed until iTunes is closed and re-launched._

 

Elias, others;

 Ok here is how it works as I just tested this again.

 If you set the Fs rate in the Audio pane of the Audio Midi Setup to say 96k. Then you start iTunes and play then CoreAudio is told to upsample from 44.1 to 96k.

 If you go into Audio Midi Setup during your iTunes session and reset this to something (like 44.1) else then it will follow. You don't have to leave iTunes to change the sampling rate.

 Therefore say if you are playing Red book stuff but then want to play high rez stuff, keep iTunes and Audio Midi Setup up and change the rate according to the needs of the track.

 As for quality of the upsampling got me, I am not one for mucking with the data.

 Test Setup: G5 Dual ---->USB Analyzer-->DAC--->Test System

 I used for the dac both my Crimson and the Benchmark both had the same results.

 With the USB Analyzer I can easily tell what the Fs rate is by the amount of data sent per frame @ 1ms.

 Thanks
 Gordon


----------



## EliasGwinn

Quote:


  Originally Posted by *tuffgong* /img/forum/go_quote.gif 
_Hello, I just got my Dac-1 in the mail today, I just want to say it sounds terrific, it completely blackens the background. Much louder/lower bass, the windows rattle with my little duo's now. I switched the jumper on the inside to do coax only, only to my chagrin realizing I was not properly grounded while doing it. I'm curious what if anything would be damaged, is there a simpler cure rather than buying a grounding wristband before moving the jumper (nitrile gloves?). What would it cost to have your team fix a unit that was affected by this?_

 

Your DAC1 probably did not suffer from any ESD damage as long as you didn't touch any IC's. If there was any damage, you would hear it. The only precaution necessary before changing the jumpers is to touch the chassis wall first to discharge any major ESD build up. This will NOT prevent ESD altogether, but it will release the majority of any build up so that there are no large arc's.

 If the unit has suffered any ESD damage, we would repair the unit without charge since it is under warranty. 

 Thanks,
 Elias


----------



## EliasGwinn

Quote:


  Originally Posted by *sangel* /img/forum/go_quote.gif 
_Hello Elias,

 Thanks for your response.
 By the way, if you remember, what was the max length of your usb cable under testing with the DAC1 usb!!

 Thanks

 sangel_

 

The longest USB cable that we have tested the DAC1 USB with is approximately 3 meters.

 Thanks,
 Elias


----------



## tuffgong

Hi Elias, thanks for all your help, when you say touch ic's, do you mean the interconnects on the back of the unit, or any connecting wires within the unit? For any non-professionals out there, I tried to boost the rca output with the trim pots on the back, but it induces distortion on my stereo immediately. On the flip side, I can turn my stereo up higher instead without adding noise, so I just let the amp give me more volume instead of trying to give it a higher signal. Cheers.


----------



## EliasGwinn

Quote:


  Originally Posted by *Wavelength* /img/forum/go_quote.gif 
_If you set the Fs rate in the Audio pane of the Audio Midi Setup to say 96k. Then you start iTunes and play then CoreAudio is told to upsample from 44.1 to 96k.

 If you go into Audio Midi Setup during your iTunes session and reset this to something (like 44.1) else then it will follow. You don't have to leave iTunes to change the sampling rate.

 Therefore say if you are playing Red book stuff but then want to play high rez stuff, keep iTunes and Audio Midi Setup up and change the rate according to the needs of the track._

 

I'm afraid I disagree. According to an Apple engineer and my testing, iTunes locks to the sample-rate of CoreAudio (AudioMIDI) upon launching. After iTunes is launched, you can change the Fs of CoreAudio via AudioMIDI, and the newly set Fs will apply to the output of CoreAudio. However, iTunes _itself_ will still be locked at the initial rate. The audio will be SRC'd by CoreAudio to the newly set rate. That is why it _looks_ like the Fs is changing....CoreAudio's Fs is changing, but iTunes is not.

 For example, if CoreAudio Fs is set to 96k in AudioMIDI when iTunes is launched, iTunes will be locked at 96 kHz output. If you play a 44.1 kHz iTunes will SRC it to 96 kHz before sending it to CoreAudio. If you then change AudioMIDI (CoreAudio) to 44.1, CoreAudio will SRC the 96k that it is receiving from iTunes to 44.1. The resulting output will be 44.1 kHz, but it is 96k in-between iTunes and CoreAudio. This is why you see the sample-rate following AudioMIDI...it will!! But iTunes output sample-rate is not changing, and CoreAudio is SRC'ing it to the set rate.

 iTunes output Fs will not change until you re-launch iTunes. At that time, it will lock to the Fs currently set in AudioMIDI.

 (In case anyone is confused about the difference between AudioMIDI and CoreAudio: CoreAudio is the audio engine in OS X. AudioMIDI is simply the user interface to CoreAudio - that is, the window where you can change the settings of CoreAudio.).

 I should say that this is all tested with OSX 10.4.5 (Tiger). Gordon may be getting different results because of Leopard. I have ordered Leopard, and it will be here early next week. 

 However, the engineer from Apple said this is a decision they made when designing iTunes 7.0 and up, so it shouldn't be affected by the OS.

  Quote:


  Originally Posted by *Wavelength* /img/forum/go_quote.gif 
_As for quality of the upsampling got me, I am not one for mucking with the data._

 

Ideally, we wouldn't _have_ to change anything to avoid SRC. Unfortunately, OS X is set up in such a fashion that it is a task to avoid SRC. But they seem willing to work with us re: bugs, so maybe there are brighter days ahead...maybe.

 Thanks,
 Elias


----------



## EliasGwinn

Quote:


  Originally Posted by *tuffgong* /img/forum/go_quote.gif 
_Hi Elias, thanks for all your help, when you say touch ic's, do you mean the interconnects on the back of the unit, or any connecting wires within the unit? For any non-professionals out there, I tried to boost the rca output with the trim pots on the back, but it induces distortion on my stereo immediately. On the flip side, I can turn my stereo up higher instead without adding noise, so I just let the amp give me more volume instead of trying to give it a higher signal. Cheers._

 

IC = integrated circuit. Aka, the 'chips'. The 'chips' in the DAC1 are the only things that are (realisically) damageable from ESD. If you touch chassis before going inside the unit, and avoid touching the 'chips', you should be fine.

 Thanks,
 Elias


----------



## EliasGwinn

Quote:


  Originally Posted by *tuffgong* /img/forum/go_quote.gif 
_ For any non-professionals out there, I tried to boost the rca output with the trim pots on the back, but it induces distortion on my stereo immediately. On the flip side, I can turn my stereo up higher instead without adding noise, so I just let the amp give me more volume instead of trying to give it a higher signal. Cheers._

 

I forgot to say in my last post...

 The thing you did with the RCA's is the right thing to do. That is, set the output to the upper limit before distortion. *BUT*, unless you have proper testing equipment, it is better to error on the too-low side vs. the too-high side.

 Thanks,
 Elias


----------



## restock

Quote:


 If you are using iTunes 7.0 or higher, it is important to set the sample-rate in AudioMIDI BEFORE OPENING ITUNES. If you have iTunes open, close it, then set the sample-rate in AudioMIDI. 
 

Elias/Gordon, what should the word length be set to in AudioMidi when primarily playing 16/44.1 files - 2ch16bit or 2ch24bit. Does the word length setting matter at all?

 Also, there is a volume slider below the sample rate setting in AudioMidi that got me confused. What should that one be set to? What does this volume slider do?


----------



## poo

Got my DAC1 yesterday. In short - I'm not totally sure what I was expecting, but I can say that I was not expecting the quality and general awesomeness I can hear. I have three audio systems that have just been totally transformed!

 That's it - back to listening!


----------



## EliasGwinn

Quote:


  Originally Posted by *restock* /img/forum/go_quote.gif 
_Elias/Gordon, what should the word length be set to in AudioMidi when primarily playing 16/44.1 files - 2ch16bit or 2ch24bit. Does the word length setting matter at all?_

 


 If you are using an audio device that interfaces to the Mac @ 24-bit, then you should always use 24-bit when possible. I can explain why in detail, if you want. But, just know that 24-bit is best if your device is capable.

  Quote:


  Originally Posted by *restock* /img/forum/go_quote.gif 
_Also, there is a volume slider below the sample rate setting in AudioMidi that got me confused. What should that one be set to? What does this volume slider do?_

 

Keep this volume control at 100%. Software volume controls are usually damaging to the audio. The volume control on the newest iTunes is the exception. It works exceptionally well.

 Thanks,
 Elias


----------



## restock

Elias, thanks very for the quick response. 

 I do have Benchmark DAC1USB as well as a Wavelength Brick in my two systems 
	

	
	
		
		

		
		
	


	




 I assume for the DAC1USB you would certainly recommend the 24bit setting.

 I did not realize the volume control in AudioMIDI was simply connected to the main volume control. I usually make sure that one is set to max.

 I am looking forward to hearing more about your tests of iTunes on Leopard when you get to it.


----------



## Wavelength

Restock, Elias;

 FYI I was making a Brick today and checked the thing out again with the Audio Midi Setup and found that the problem only happens when you put a 24/96 dac on.

 I talked to Apple and they say it has something to do with the USB init to the Audio Midi Setup that is causing the problem and for 24/96 dacs it pushes the Fs (sample rate) to the max and for 16 bit it leaves it at 44.1.

 I think Lepoard and 7.5 iTunes sounds much better than earlier versions.

 Well it could be worse things in life.

 Thanks
 Gordon


----------



## korben_dallas

Gordon,

 Have you achieved bit-perfect playback with Vista? If so, what was your setup?


----------



## sangel

Hello Elias,

 Thank you very much.

 Best regards

 sangel


----------



## Wavelength

Elias,

 I talked to Apple yesterday and this morning had a conference call with the engineers.

 They were not aware of this problem and it had not been seen in their problem reports.

 They are going to try and reproduce it and if not I can send them out my DAC1 for them to see what the problem is.

 I did send them your enumeration as they said it maybe linked to the way you handle something.

 Thanks
 Gordon


----------



## Wavelength

Quote:


  Originally Posted by *korben_dallas* /img/forum/go_quote.gif 
_Gordon,

 Have you achieved bit-perfect playback with Vista? If so, what was your setup?_

 

KD,

 Actually as Thomas has indicated in several posts now this is easy to do. Much easier and more assured than XP ever would and also applications where kernel streaming and ASIO were not supported via Direct Sound (iTunes, WMP, etc) can achieve this.

 It does require a 24 bit dac and setting that option in the device panel. At that point 16 bit audio will cruise through uninterrupted.

 I tried this out with J River, iTunes and Media Player. Results were much better.

 Thanks
 Gordon


----------



## Wavelength

Gang, Elias;

 I had a long banter with Apple yesterday in regards to the 7.5 stuff.

 Ok here it is and this has been the same for a long while:

  Quote:


 I don't believe there have been any changes in this regard, but the driver will try to select formats in the following order the first time a device is attached to a particular port on a system:

 1. 2-channel 16-bit @ 44.1kHz
 2. 1-channel 16-bit @ 44.1kHz
 3. 2-channel 16-bit @ highest supported sample rate
 4. 1-channel 16-bit @ highest supported sample rate
 5. alternate setting 1 @ highest supported sample rate

 If the device doesn't do 16 bit, the correct behavior is probably 24-bit at the highest supported sample rate. After changing the sample rate, channel count or bit depth, the Mac should remember the last setting if you plug it back into the same USB port.

 One last thing: 10.4.11 and Leopard have the same version of the USB audio driver. 
 

So really nothing has changed here and why this came up seems to be a question.

 Elias, the guys in the sound group want to know who you are talking too as it may not be the correct people.

 Thanks
 Gordon


----------



## thomaspf

Quote:


  Originally Posted by *Wavelength* /img/forum/go_quote.gif 
_KD,

 Actually as Thomas has indicated in several posts now this is easy to do. Much easier and more assured than XP ever would and also applications where kernel streaming and ASIO were not supported via Direct Sound (iTunes, WMP, etc) can achieve this.

 It does require a 24 bit dac and setting that option in the device panel. At that point 16 bit audio will cruise through uninterrupted.

 I tried this out with J River, iTunes and Media Player. Results were much better.

 Thanks
 Gordon_

 

And just to be extra clear about this to avoid any more drawn out discussions with the marketing folks from Benchmark. This will give you a 16 bit stream embedded in the top 16 bits of a 24 bit stream. If you use a true 24 bit stream the lower bits still get modified. Vista is not truly bit transparent except that you can pass through a 16 bit stream inside a 24 bit stream as described. 

 Cheers

 Thomas


----------



## FRANKe

Quote:


  Originally Posted by *thomaspf* /img/forum/go_quote.gif 
_Vista is not truly bit transparent except that you can pass through a 16 bit stream inside a 24 bit stream as described. 

 Thomas_

 

Thomas, is this a general "blanket" statement, or are you specifically referring to the Benchmark DAC and the use of "shared mode" (with iTunes & WMP)?

 With Vista's "exclusive mode" is it not "truly bit transparent"? 

 FWIW, I believe XXHighEnd is the only media player right now utilizing "exclusive mode".
XXHighEnd - Index

 It would be interesting to see the results of these bit transparent tests done with XXHighEnd. And BTW, it has been reported that XXHighEnd will not lock onto "exclusive mode" with the Benchmark DAC presumably because the DAC is "reporting" as 24bit and at the moment XXHighEnd does not output 24bit.
Benchmark DAC1 USB


----------



## tuffgong

Hello again, just some questions for the jury before I go. I am considering buying a dac1 usb as well for dj purposes, I love the dac-1 classic alot. The DJ program that seems to be the best for me is 'disco xt'. It automixes flac files, and the company emailed me back and said the program uses directsound, so I think that means with vista, running at 24 bit/48hz, with volume at 100% on the dj program, I should have the best quality possible considering the automix feature? I also think i'm going to use my old platinum audio solo's to take on the road, and I was going to bi-amp each speaker with a pair of bryston powerpac 120 sst's, kind of a poor man's aml1 
	

	
	
		
		

		
			





 My amateur question is how can I split the xlr signal coming out the back of the dac1 so I can get four signals to amplify without messing up the line levels like a normal guitar center splitter. Thanks in advance.


----------



## thomaspf

Quote:


  Originally Posted by *FRANKe* /img/forum/go_quote.gif 
_Thomas, is this a general "blanket" statement, or are you specifically referring to the Benchmark DAC and the use of "shared mode" (with iTunes & WMP)?

 With Vista's "exclusive mode" is it not "truly bit transparent"? 

 FWIW, I believe XXHighEnd is the only media player right now utilizing "exclusive mode".
XXHighEnd - Index

 It would be interesting to see the results of these bit transparent tests done with XXHighEnd. And BTW, it has been reported that XXHighEnd will not lock onto "exclusive mode" with the Benchmark DAC presumably because the DAC is "reporting" as 24bit and at the moment XXHighEnd does not output 24bit.
Benchmark DAC1 USB_

 


 No blanket statements from me in general...

 Benchmark has been claiming for a while now that by changing the USB firmware in their DAC, they can get the USB audio driver in Windows to produce bit perfect results for applications like foobar as described on their WEB site. I have tried to correct that statement from the beginning since there is nothing any USB device can do to achieve that with applications using WAV or DirectSound APIs using the usbaudio driver in Windows.

 This has always worked with kernel streaming. The wasapi in Vista is a new option for a direct sound API and the first commercial application I have seen a while back that supported exclusive wasapi was ntrack but that is not a conventional software player. I am pretty sure if your application supports wasapi in exclusive mode then this should work but I have not tested it.


 Cheers

 Thomas


----------



## EliasGwinn

Quote:


  Originally Posted by *Wavelength* /img/forum/go_quote.gif 
_Gang, Elias;

 I had a long banter with Apple yesterday in regards to the 7.5 stuff...

 "I don't believe there have been any changes in this regard, but the driver will try to select formats in the following order the first time a device is attached to a particular port on a system:

 1. 2-channel 16-bit @ 44.1kHz
 2. 1-channel 16-bit @ 44.1kHz
 3. 2-channel 16-bit @ highest supported sample rate
 4. 1-channel 16-bit @ highest supported sample rate
 5. alternate setting 1 @ highest supported sample rate

 If the device doesn't do 16 bit, the correct behavior is probably 24-bit at the highest supported sample rate. After changing the sample rate, channel count or bit depth, the Mac should remember the last setting if you plug it back into the same USB port.

 One last thing: 10.4.11 and Leopard have the same version of the USB audio driver."

 ...So really nothing has changed here and why this came up seems to be a question._

 

Gordon,

 This information correlates with what I understand about OS X 10.4. However, this information does not pertain to iTunes. It is iTunes 7 that has changed, not OS X. 

 iTunes 7 sets a default output sample rate, when it is launched, to match that set in AudioMIDI. After iTunes is launched, if the user changes the sample-rate in AudioMIDI, the sample rate of the final digital output will correspond with that change. In other words, if you monitor the USB activity, or the built-in digital audio output, the sample-rate will change according to the settings in AudioMIDI. 

 However, the important detail to understand is that the sample-rate of the (internal) digital audio, leaving iTunes and going to CoreAudio, will remain at the sample-rate that iTunes defaulted to upon launch. *You will not be able to monitor this sample rate with a USB analyzer or any other external hardware.* This is a difference between the two software ports _inside the computer_.

 Here is a quote from my correspondence with Apple:

  Quote:


 On the Mac, the OS (i.e. CoreAudio) mixes all of the audio into one stream which it then hands to the hardware. In order for this to work, all of the audio streams must be mixed at the current sample rate of the hw. So, *somebody* has to do SRC and it can either be the app or CoreAudio doing it for you. Pre iTunes 7.0, we let CoreAudio do it. Post-7.0, we do it. 

 One caveat is that iTunes doesn't notice if the user changes the hw sample rate *after* it launches -- It captures it at launch time and leaves it that way. It's certainly possible that these people didn't figure that out, so changing the hw rate after we launch means we'll sometimes be doing SRC *and* CoreAudio might be doing SRC at the same time. For example:

 1) At launch, the hw sample rate is 48kHz. We set up our processing chain to run at 48kHz.

 2) You play a 44.1kHz file. We do SRC to 48kHz and hand the data to CoreAudio who send it along to the device.

 3) You change the hw sample rate to 44.1kHz while we're running.

 4) You play the same 44.1kHz file. We now do SRC to 48kHz and give the data to CoreAudio who does SRC back to 44.1kHz!

 

.

 Thanks,
 Elias


----------



## EliasGwinn

Quote:


  Originally Posted by *thomaspf* /img/forum/go_quote.gif 
_And just to be extra clear about this to avoid any more drawn out discussions with the marketing folks from Benchmark...

 Cheers

 Thomas_

 

 Quote:


  Originally Posted by *thomaspf* 
_Benchmark has been claiming for a while now that by changing the USB firmware in their DAC, they can get the USB audio driver in Windows to produce bit perfect results for applications like foobar as described on their WEB site. I have tried to correct that statement from the beginning since there is nothing any USB device can do to achieve that with applications using WAV or DirectSound APIs using the usbaudio driver in Windows._

 


 Hello Thomas,

 I would like to recognize your contributions to this thread; they are appreciated. However, I would like to clarify a few things before we continue forward.

 First, our marketing department (which consists of 1 person, a graphic designer) has no input and makes no claims pertaining to the performance or capabilities of our products. Those claims are solely made by the two engineers in the company: myself and John Siau, the designer of the DAC1 and most other current Benchmark products. 

 If you are confused and thought that I was involved in marketing, I would like to clarify that my title at Benchmark is "Applications Engineer", and I hold a degree in Electrical Engineering. 

 Benchmark has always had a mission to maintain objective, scientific truth and honest claims about our products' performance, and audio technology in general, in an effort to achieve within the audio community a clear understanding of audio technology. In fact, this forum is one example of our dedication to maintaining an honest discourse about our products and technology in general. The 'Audio Wiki', the 'Feedback - Informational Newsletter', and the white papers on our website are other examples. 

 We refuse to engage in the "Black Magic" or "Snake Oil" claims that linger like a bad smell among the audiophile community. We refuse to let marketing-spin interrupt objective fact. We pride ourselves in being as honest and transparent as possible. I know many folks in both the pro and audiophile community will attest to that, as it has become a large part of our reputation. 

 Now, as for XP transparency and the DAC1 USB...

 I would like to clarify our claims: *WE DO NOT CLAIM THAT OUR FIRMWARE MANIPULATES THE BEHAVIOR OF KMIXER or USBaudio.sys*. Please do not propagate that misinformation. We claim that Windows XP can inherently operate bit transparently *as long as the relevant software (app, driver and/or firmware) doesn't prevent transparent operation*.

 I understand that you and Gordon disagree with this claim. I respect your objections, and I would like to resolve this issue. *Please understand: if Kmixer is not capable of transparent operation - we would like to know just as much as you!* However, according to our tests, and communications with Microsoft, we have concluded that it is possible. 

 Please understand: we are not 'closing the case' on this issue. If you have proof that shows otherwise, I would like to see it and take it into consideration. I know that Thomas has claimed that he has developed his own software to check for bit-transparency. He claims that this software proves that XP is not bit-transparent. This intrigues me, and I am very interested in obtaining a copy of this software so that I can also see the results. 

 I also understand that Thomas and/or Gordon have also spoken with representatives at Microsoft. Perhaps the discrepancies in statements from Microsoft (and/or Apple) are occurring because the statements are represented here as 2nd hand communication. In other words, I'm sure if these representatives were speaking to each other, the discrepancies would be cleared up relatively quickly and succinctly. 

 To conclude, the main point I would like to stress is that I (and I'm sure many other Head-Fi folks) would not like to see slanted statements and spin in this thread. I represent Benchmark as a qualified engineer, and I do not work in marketing. All claims that I and/or Benchmark make are true to the extent of our knowledge and testing. We will respect and will try to resolve any discrepancies. 

 Thank you for your participation.

 -Elias


----------



## EliasGwinn

Quote:


  Originally Posted by *thomaspf* /img/forum/go_quote.gif 
_No blanket statements from me in general..._

 

...isn't _that statement_ a 'blanket statement?











 Just kidding...trying to lighten the mode a bit after my last saga of a post


----------



## EliasGwinn

Quote:


  Originally Posted by *tuffgong* /img/forum/go_quote.gif 
_...how can I split the xlr signal coming out the back of the dac1 so I can get four signals to amplify without messing up the line levels like a normal guitar center splitter. Thanks in advance._

 

You can use simple 'Y' splitters, as long as the input impedances on the loads (active crossovers, I believe...) are sufficiently high (>1.5k). Also, you should be aware that if one of the devices are powered down, it may present an awkward load and distort the output. It won't damage the DAC1 USB, but the signal may become distorted.

 Thanks,
 Elias


----------



## Wavelength

Elias,

 ThomasPF is with Microsoft! He tested your DAC1 USB at Microsoft.

 ~~~

 Look neither of us are your enemies... I don't see you as a competitors and have been trying to help you out here as Thomas has.

 The only thing we are saying as is most of the rest of the Windows team is that it is impossible to get bit perfect audio *without* bypassing the KMIXER.

 That is an easy thing to do and I am sure your insrtuctions for doing that will make it capable of bit perfect resolution.

 If you follow Thomas's instructions for Vista there too you will get audio nirvana.

 ~~~~~~~~~

 You may want to pick up a Prism dScope III for testing then you can easily see what is and is not working. Since it has native mode it can stream kernel, DirectSound and via drivers and you can quickly see what's bit perfect or not.

 ~~~~~~~~~

 Heck next time we are in your area I say let's have some dinner and drinks at my fav restaurant the Dinosuar (my wife is from Manilus).

 Thanks
 Gordon


----------



## EliasGwinn

Quote:


  Originally Posted by *Wavelength* /img/forum/go_quote.gif 
_Elias,

 ThomasPF is with Microsoft! He tested your DAC1 USB at Microsoft.

 ~~~

 Look neither of us are your enemies... I don't see you as a competitors and have been trying to help you out here as Thomas has.

 The only thing we are saying as is most of the rest of the Windows team is that it is impossible to get bit perfect audio *without* bypassing the KMIXER.

 That is an easy thing to do and I am sure your insrtuctions for doing that will make it capable of bit perfect resolution.

 If you follow Thomas's instructions for Vista there too you will get audio nirvana.

 ~~~~~~~~~

 You may want to pick up a Prism dScope III for testing then you can easily see what is and is not working. Since it has native mode it can stream kernel, DirectSound and via drivers and you can quickly see what's bit perfect or not.

 ~~~~~~~~~

 Heck next time we are in your area I say let's have some dinner and drinks at my fav restaurant the Dinosuar (my wife is from Manilus).

 Thanks
 Gordon_

 

Gordon,

 Thank you for this kind note. And don't worry, I was not feeling attacked or insulted. I simply wanted to make sure this thread did not stray from the facts. Sometimes mis-quoting or mis-stating someone, even if unintentionally, can drive the conversation to undesirable places. But I feel we are all working towards the same goals, and I really do appreciate your correspondence.

 Thomas and I are corresponding with Microsoft engineers to come to a consensus on this topic of bit-transparency through kmixer. Hopefully, we will be able to provide a resolution to the audio community once and for all.

 I would be happy to join you for an evening at the Dino. Please let me know when you are in town next.

 Thanks,
 Elias


----------



## poo

Quote:


  Originally Posted by *EliasGwinn* /img/forum/go_quote.gif 
_Thomas and I are corresponding with Microsoft engineers to come to a consensus on this topic of bit-transparency through kmixer. Hopefully, we will be able to provide a resolution to the audio community once and for all._

 

Many thanks for your efforts here guys - this is exactly the sort of thing that makes this thread so valuable! Wonderful to see that those who disagree are open to being wrong in an effort to better our understanding and enjoyment.

 Looking forward to your findings!


----------



## thomaspf

You are welcome. And just to clarify one more thing. I am not posting here under any capacity other than being an interested head-fi member. I do not work in the audio team and do not speak for them. 

 Microsoft also does not test devices. I do that as a personal hobby in my spare time.

 And I repeat what I have said before. This is not a big issue since you can always use kernel streaming to go direct.

 Cheers

 Thomas


----------



## cansman

Hi Elias,

 I would like to confirm bit perfect usage for iTunes 7.5 & OSX 10.5:

 1. Set Audio MIDI to 44.1kHz & 24-Bits BEFORE opening iTunes 7.5

 2. If iTunes is opened, & settings for Audio MIDI changed (e.g., to 48kHz), there will be SRC, even if Audio MIDI settings are changed back to 44.1kHz

 3. To undo SRC, close iTunes & Audio MIDI and repeat step 1.

 Also, Elias, I believe the Front Row program actually does SRC. Therefore, it might be best to play music via iTunes rather than Front Row - this is my observation; verification needed.

 Thanks Elias for always helping Mac users to get bit perfect output for our Benchmark DAC1s!

 Cheers,
 cansman


----------



## korben_dallas

Elias,

 Ok I have read through this entire thread (can you believe it???!!!).

 First, I have to compliment you greatly for supporting your product in this fashion. I am greatly considering the DAC1 USB, even though the price is somewhat out of my range.

 Second, I want to thank you Elias, as well as Gordon, Thomas, and Steve N. for your contributions to all things digital & analog discussed here. This thread will go down in the anals of internet history as the "benchmark" 
	

	
	
		
		

		
			





 for technical discourse.

 So on to my questions, which might seem quite random but are a product of your many nuggets of interest in your posts here..

 1. You stated surround sound, such as 5.1 / 7.1 is an expensive proposition. Can you elaborate? After you answer, I'd like to inform/remind you that decoding (dolby, dts, etc) in Vista is no longer at the driver level, it is at the application and/or filter level, such that a hardware manufacturer like Benchmark need not be concerned about licensing of those technologies, since they are the responsibility of the playback application. A 5.1 USB DAC would be AWESOME, and I believe would be a godsend to HTPC enthusiasts like myself. There are very few USB audio devices capable of 5.1 output, and none what I would consider high quality.
 2. Some AVR manufacturers have the option to disable the front panel / LED's. Two reasons for this: to reduce the potential for interference/distortion, and to reduce the light in a dark room suitable for movie playback. Does the DAC1 USB have this option? If not, could an official mod be constructed to accomplish this?
 3. I am very interested in hearing the results of your discussions with Microsoft on bit-transparency, particularly when it comes to Vista.
 4. Is there a difference between "bit-perfect" and "bit-transparent"? I realize that your ASRC is resampling to 110kHz, but up to that point is the interest of delivering the bits as unchanged as possible.
 5. You have said to avoid power conditioners with the DAC1 USB. Does this also include surge protectors? I'll be honest, I'd be wary of plugging in a $1300 piece of equipment directly to the wall, especially with the lightning storms and electrical weirdness in my city.
 6. In general, is it best to have a solid state amplifier (I have a Crest 450 2-channel) with output pots at 100%?
 7. Do you have an opinion about tube power amplifiers? My own experience (with musical instruments) has been that the preamp is much more important wrt tubes than the power amp. And how about tube power wrt to DAC's?


----------



## EliasGwinn

Quote:


  Originally Posted by *cansman* /img/forum/go_quote.gif 
_Hi Elias,

 I would like to confirm bit perfect usage for iTunes 7.5 & OSX 10.5:

 1. Set Audio MIDI to 44.1kHz & 24-Bits BEFORE opening iTunes 7.5

 2. If iTunes is opened, & settings for Audio MIDI changed (e.g., to 48kHz), there will be SRC, even if Audio MIDI settings are changed back to 44.1kHz

 3. To undo SRC, close iTunes & Audio MIDI and repeat step 1._

 

You are correct until half-way through #2. Let me modify your notes...

 1. Set AudioMIDI to sample-rate *X* ( = to that of the audio you will be playing) BEFORE opening iTunes 7.5

 2. If iTunes is opened, & settings for Audio MIDI changed (e.g., to a sample rate other then X), there will be SRC, *but the SRC will be disabled as soon as*Audio MIDI settings are changed back to *X*

 3. To *avoid SRC for sample rates other then X*, close iTunes & Audio MIDI and repeat step 1.

 Thanks,
 Elias


----------



## Wavelength

Quote:


  Originally Posted by *cansman* /img/forum/go_quote.gif 
_Also, Elias, I believe the Front Row program actually does SRC. Therefore, it might be best to play music via iTunes rather than Front Row - this is my observation; verification needed._

 

Cansman,

 No Front row does not do the SRC it is as it say it is only a front end program to access iTunes and other applications.

 On the MAC CoreAudio does all the SRC and only follows what iTunes tells it too do.

 On the PC much of Core Audio I would presume is located in QuickTime.

 In general iTunes really does not really deal a whole lot with hardware that is preserved to the CoreAudio and QuickTime on the PC.

  Quote:


 1. You stated surround sound, such as 5.1 / 7.1 is an expensive proposition. Can you elaborate? After you answer, I'd like to inform/remind you that decoding (dolby, dts, etc) in Vista is no longer at the driver level, it is at the application and/or filter level, such that a hardware manufacturer like Benchmark need not be concerned about licensing of those technologies, since they are the responsibility of the playback application. A 5.1 USB DAC would be AWESOME, and I believe would be a godsend to HTPC enthusiasts like myself. There are very few USB audio devices capable of 5.1 output, and none what I would consider high quality. 
 

Korban, 

 Part of the problem with this is the fact that most of the USB Audio chips only work to 12MHZ or high speed mode. If you are a huge company then you take USB 2.0 capable chips or create your own that will support the needed bandwidth. Oxford Semiconductor made 3, Firewire chips that did 7.1 and I did a project for another company using that and 4 dual triode tubes for the output. It was ok, high jitter but ok for what they wanted to sell it at.

 My take is this and I have about ton of these things laying around. The 5.1 and 7.1 units are all pretty cheap. That's really the markets fault as people don't want to spend more than a couple hundred bucks on these.

 Steve Guttenburg did a pretty big study on the content of multi channel dvd ounce and found that most mixing engineers didn't even include the information.

 I have switched hundreds of customers over the years who have asked for costly custom amp and digital solutions to just concentrate on two channel stereo. If you do this really well then it doesn't matter about the 5.1 or 7.1 stuff it sounds great no matter if you are playing stereo or watching a movie.

 Thanks
 Gordon


----------



## EliasGwinn

Quote:


  Originally Posted by *korben_dallas* /img/forum/go_quote.gif 
_
 1. You stated surround sound, such as 5.1 / 7.1 is an expensive proposition. Can you elaborate? After you answer, I'd like to inform/remind you that decoding (dolby, dts, etc) in Vista is no longer at the driver level, it is at the application and/or filter level, such that a hardware manufacturer like Benchmark need not be concerned about licensing of those technologies, since they are the responsibility of the playback application. A 5.1 USB DAC would be AWESOME, and I believe would be a godsend to HTPC enthusiasts like myself. There are very few USB audio devices capable of 5.1 output, and none what I would consider high quality.

 2. Some AVR manufacturers have the option to disable the front panel / LED's. Two reasons for this: to reduce the potential for interference/distortion, and to reduce the light in a dark room suitable for movie playback. Does the DAC1 USB have this option? If not, could an official mod be constructed to accomplish this?

 3. I am very interested in hearing the results of your discussions with Microsoft on bit-transparency, particularly when it comes to Vista.

 4. Is there a difference between "bit-perfect" and "bit-transparent"? I realize that your ASRC is resampling to 110kHz, but up to that point is the interest of delivering the bits as unchanged as possible.

 5. You have said to avoid power conditioners with the DAC1 USB. Does this also include surge protectors? I'll be honest, I'd be wary of plugging in a $1300 piece of equipment directly to the wall, especially with the lightning storms and electrical weirdness in my city.

 6. In general, is it best to have a solid state amplifier (I have a Crest 450 2-channel) with output pots at 100%?

 7. Do you have an opinion about tube power amplifiers? My own experience (with musical instruments) has been that the preamp is much more important wrt tubes than the power amp. And how about tube power wrt to DAC's?_

 

I will address the questions only in the capacity which I feel qualified to answer.

 1. This is a very interesting point, and we will take it into consideration for future products.

 2. The DAC1 does not have a feature to disable the LED's. We may be able to modify your DAC1 for you, but I'll have to check for sure.

 3. As am I!! 
	

	
	
		
		

		
		
	


	




 4. In my mind, there is no difference between 'bit-perfect' and 'bit-transparent.' Perhaps someone else has different interpretations...

 5. Actually, power conditioners will not affect the performance of the DAC1, for better or worse. However, you may want to think twice about using them for other devices, especially power amplifiers. Most power conditioners will increase the source impedance of the AC line. The result will be a voltage lag (lower voltage peaks) which will lower the voltage available for the rectifiers in the power supply. Lots of gear needs power conditioning to filter AC line noise. You should contact the manufacturer of the equipment in question to determine if it benefits from a power conditioner. The DAC1 has sufficient filtering so that AC line noise will never affect its performance.

 As for surge suppressors, you should also be careful with these. Specifically, never use a M.O.V. (Metal Oxide Varistor) suppressors on an ungrounded outlet. The MOV 's threshold voltage will be lowered on every surge, eventually approaching the normal line voltage. When its threshold reaches the normal line voltage, hazardous voltages may begin to appear on ground bus. If the ground is floating (not earthed), the chassis could be very dangerous to touch!! Don't worry about protecting the DAC1 from a surge, the built-in filters will protect it. It has been CE tested for surge handling and it had no problems. Furthermore, we have never gotten a call about a damaged DAC1 because of a surge.

 6. I'm not familiar with the topology of that specific amp, but those pots are usually input attenuators. We recommend setting these between 25-50%, and attenuating the DAC1 as necessary to accommodate the volume. This will usually result in the best signal-to-noise ratio.

 7. I can't comment on tube gear. I have nothing against it, but I will leave it to the folks who have more experience with it.

 Thanks!
 Elias


----------



## restock

Quote:


 No Front row does not do the SRC it is as it say it is only a front end program to access iTunes and other applications. 
 

Actually something seems to have changed with OS X Leopard - Front Row used to start iTunes (when starting Front Row or playing songs in Front Row); now it doesn't... Thus it doesn't seem to be just an interface anymore.

  Quote:


 On the MAC CoreAudio does all the SRC and only follows what iTunes tells it too do. 
 

As long as iTunes/Front Row don't pass on wrong instructions...


----------



## EliasGwinn

Quote:


  Originally Posted by *Wavelength* /img/forum/go_quote.gif 
_
 On the MAC CoreAudio does all the SRC and only follows what iTunes tells it too do._

 

Gordon,

 This is incorrect. iTunes does not control CoreAudio's sample-rate. That is set by the user in AudioMIDI. iTunes has its own output sample-rate that is set and locked when iTunes is launched. The sample-rate that iTunes locks to is the sample-rate AudioMIDI Setup is set at upon iTunes' launching.

 CoreAudio will do SRC only if it is set to a sample-rate different from that which iTunes is locked to. The only way this can happen is if the user changes the sample-rate in AudioMIDI Setup _after_ iTunes is launched.

 iTunes will SRC any audio it is playing that does not have the same sample-rate that iTunes is locked to. For example, if iTunes locks to 44.1kHz upon launching, any audio that is not 44.1k will be SRC'd to 44.1. At that point, iTunes will send 44.1k audio to CoreAudio. If CoreAudio has been changed to a different sample-rate since iTunes was opened, CoreAudio will *also* SRC the audio. It is very possible for audio to be SRC'd twice before leaving the Mac!!

 Thanks,
 Elias


----------



## cansman

Thanks Elias and Gordon for your responses. Appreciate it!


----------



## Wavelength

Quote:


  Originally Posted by *EliasGwinn* /img/forum/go_quote.gif 
_Gordon,

 This is incorrect. iTunes does not control CoreAudio's sample-rate. That is set by the user in AudioMIDI. iTunes has its own output sample-rate that is set and locked when iTunes is launched. The sample-rate that iTunes locks to is the sample-rate AudioMIDI Setup is set at upon iTunes' launching.

 CoreAudio will do SRC only if it is set to a sample-rate different from that which iTunes is locked to. The only way this can happen is if the user changes the sample-rate in AudioMIDI Setup after iTunes is launched.

 iTunes will SRC any audio it is playing that does not have the same sample-rate that iTunes is locked to. For example, if iTunes locks to 44.1kHz upon launching, any audio that is not 44.1k will be SRC'd to 44.1. At that point, iTunes will send 44.1k audio to CoreAudio. If CoreAudio has been changed to a different sample-rate since iTunes was opened, CoreAudio will *also* SRC the audio. It is very possible for audio to be SRC'd twice before leaving the Mac!!

 Thanks,
 Elias_

 

Elias,

 You are missunderstanding what I was saying. Yes iTunes does get the sample rate from the Audio Midi Setup.

 But iTunes does not have any SRC in the code. Look at Core Audio and what it can do. I wrote a little player in less than an hour because the damn Core Audio does everything. It's so easy to use and has a ton of options.

 Yes I do see your point about SRC twice as you maybe correct about that. Only when iTunes=Audio Midi Setup=Fs of the song when you start iTunes will it be direct and without SRC from what you say.

 I still have not gotten confirmation about that from Apple yet.

 Thanks
 Gordon


----------



## EliasGwinn

Quote:


  Originally Posted by *Wavelength* /img/forum/go_quote.gif 
_But iTunes does not have any SRC in the code._

 

Gordon,

 This is the only point that we are disagreeing on. iTunes v7 *only* (not iTunes 6) has SRC in the code. And, I might add, the SRC is of very high quality. It has made iTunes for Mac a very viable high-quality playback system.

 By the way, here is what the engineers from Apple told us in an email:

  Quote:


 On the Mac, the OS (i.e. CoreAudio) mixes all of the audio into one stream which it then hands to the hardware. In order for this to work, all of the audio streams must be mixed at the current sample rate of the hw. So, *somebody* has to do SRC and it can either be the app or CoreAudio doing it for you. Pre iTunes 7.0, we let CoreAudio do it. Post-7.0, we do it. 

 One caveat is that iTunes doesn't notice if the user changes the hw sample rate *after* it launches -- It captures it at launch time and leaves it that way. So changing the hw rate after we launch means we'll sometimes be doing SRC *and* CoreAudio might be doing SRC at the same time. For example:

 1) At launch, the hw sample rate is 48kHz. We set up our processing chain to run at 48kHz.

 2) You play a 44.1kHz file. We do SRC to 48kHz and hand the data to CoreAudio who send it along to the device.

 3) You change the hw sample rate to 44.1kHz while we're running.

 4) You play the same 44.1kHz file. We now do SRC to 48kHz and give the data to CoreAudio who does SRC back to 44.1kHz!

 

Thanks,
 Elias


----------



## joijwall

I'm lost, Elias and Gordon! Could you repeat what settings I should use to get bit-transparency using Mac OS X Leopard and iTunes 7.5? And should I use AIFF or WAV when ripping my cd:s? And is VLC still perfect in Leopard? Just got a DAC1 today! Thanks! Joachim


----------



## EliasGwinn

Quote:


  Originally Posted by *joijwall* /img/forum/go_quote.gif 
_I'm lost, Elias and Gordon! Could you repeat what settings I should use to get bit-transparency using Mac OS X Leopard and iTunes 7.5? And should I use AIFF or WAV when ripping my cd:s? And is VLC still perfect in Leopard? Just got a DAC1 today! Thanks! Joachim_

 

Hello Joachim,

 Without getting into the nuts and bolts (as they are addressed in the past few pages), here are two different recommended solutions we are offering to our customers who are using iTunes 7.x on Mac OS X. The first is the easiest solution for high-quality playback, and the second is the solution for bit-transparency at all sample-rates. The second should be used with great caution, as it can easily be set wrong which will result in serious distortion.

 1. *The ‘Set It And Forget It’ solution for iTunes 7.x*: Before opening iTunes, set the sample rate in of CoreAudio (in AudioMIDI Setup) to 96 kHz. Do not change the sample rate of CoreAudio unless iTunes is restarted after the change is made. This solution will prevent CoreAudio from applying SRC (the quality of CoreAudio’s SRC is horrible). Also, by having iTunes locked at 96 kHz, all audio with sample rates below 96 kHz will be up-sampled to 96 kHz. As the quality of iTunes’ SRC is very good (virtually inaudible), this will cause virtually no loss in sonic quality. Also, by avoiding down-sampling by iTunes, this setting will never result in a loss of bandwidth. 

 2. *The ‘Bit-Transparency For Each Sample Rate’ solution*: *NOTE: _This solution is rather cumbersome, offers virtually no quality improvement over the first solution, and can easily be mis-configured which will cause severe distortion. _Before opening iTunes, set the sample rate of CoreAudio (in AudioMIDI Setup) to that of the audio you will be playing. Do not change the sample rate of CoreAudio unless iTunes is restarted after the change is made. This solution will prevent CoreAudio from applying SRC, and avoid SRC by iTunes for all audio with the same sample rate that iTunes is locked to. 

 Thanks,
 Elias


----------



## poo

Quote:


  Originally Posted by *EliasGwinn* /img/forum/go_quote.gif 
_Hello Joachim,

 Without getting into the nuts and bolts (as they are addressed in the past few pages), here are two different recommended solutions we are offering to our customers who are using iTunes 7.x on Mac OS X. The first is the easiest solution for high-quality playback, and the second is the solution for bit-transparency at all sample-rates. The second should be used with great caution, as it can easily be set wrong which will result in serious distortion.

 1. *The ‘Set It And Forget It’ solution for iTunes 7.x*: Before opening iTunes, set the sample rate in of CoreAudio (in AudioMIDI Setup) to 96 kHz. Do not change the sample rate of CoreAudio unless iTunes is restarted after the change is made. This solution will prevent CoreAudio from applying SRC (the quality of CoreAudio’s SRC is horrible). Also, by having iTunes locked at 96 kHz, all audio with sample rates below 96 kHz will be up-sampled to 96 kHz. As the quality of iTunes’ SRC is very good (virtually inaudible), this will cause virtually no loss in sonic quality. Also, by avoiding down-sampling by iTunes, this setting will never result in a loss of bandwidth. 

 2. *The ‘Bit-Transparency For Each Sample Rate’ solution*: *NOTE: This solution is rather cumbersome, offers virtually no quality improvement over the first solution, and can easily be mis-configured which will cause severe distortion. Before opening iTunes, set the sample rate of CoreAudio (in AudioMIDI Setup) to that of the audio you will be playing. Do not change the sample rate of CoreAudio unless iTunes is restarted after the change is made. This solution will prevent CoreAudio from applying SRC, and avoid SRC by iTunes for all audio with the same sample rate that iTunes is locked to. 

 Thanks,
 Elias_

 


 Perfect! Just what I needed!!

 Now on to the next bit I need... can you provide the same info for my other system - Win XP running iTunes on a PC?

 Thanks again Elias!

 Should have mentioned (in case of relevance) outputting to airport express (optical) or USB into DAC1.


----------



## poo

Another question... sorry... 
	

	
	
		
		

		
			





 I am using XLR outputs (unbalanced floating pin 3) connected to a Musical Fidelity A3.5 integrated Amp. 

 I currently have the 'Output Level Switch' on the factory 'variable' setting, so the volume control is active (DAC1 working as a pre-amp as I understand it).

 I don't want to use the DAC1 as a preamp - I think in my case it is better to use the integrated MF amp to do both pre and power, so I presume I have to switch to calibrated (which I haven't tried yet because of a warning in the manual referring to the 'calibrated' setting when driving power amps). I also don't want to have to remember to readjust the volume control after headphone use before switching back to the integrated amp.

 I don't have any tools or experience in altering the settings of calibration trimmers, so am not sure calibrated is the way to go either. 

 After reading the manual a few times - I get the impression that the DAC1 USB is factory set in such a way that all I really have to do is switch to calibrated and it will be fine as is, but I wanted to check my assumption is correct before destroying something 
	

	
	
		
		

		
		
	


	




 Advice on how to proceed would be much appreciated...


----------



## joijwall

Hello again! Notes from my first day of listening. 
 I've compared MacBook running Mac OS X Leopard (10.5.1) and iTunes 7.5 or VLC 0.8.6d with a very good cd player. No measuring, listening only, both from cds and from AIFF-files made from cds in iTunes. 
 Settings in MacOSX AudioMIDI was always 44,1/24, and both Mac's and iTunes's volume were set to max. VLC had the standard value 256. 
 The first comparison was between iTunes and the cd. The cd had a rich sound, while iTunes sounded detailed, but also thin and a bit strange. This was disappointing. Switching to VLC though raised new hope for DAC1. Now the cd sound in comparison felt a bit overfed and crowded in the midrange, while VLC filled the room with rich but crystal clear sounds. But the difference was was not overwhelming. In fact, the presentations seemed rather similar.
 More tests were done. Sometimes the records seemed to fit the cd better, but I usually preferred the effortless detail from VLC. I don't know why iTunes was thin in the first test. Later on I rather compared it to the slightly messy but rich sound from the cd, while VLC continued open, loving and generous. 
 This difference between iTunes and VLC makes me suspect that VLC has the bit-transparancy, and something needs to be done to make iTunes deliver truly. 
 But I've only my ears to trust, so I'm really looking forward to your Leopard installation, Elias. Hopefully your testing skills, as well as Gordon's and perhaps also your contacts with Apple engineers, will get iTunes to the top.
 Tell me all about it! 
 Joachim


----------



## joijwall

Me again. This time I bought a cheap Toslink optical cable and connected it to my harddisk DVD. To get any sound at all from a dvd I needed to set Convert Dolby Digital to PCM in the DVD. I'm not sure this was needed for CD though. Playing cd's, I actually preferred the DVD+DAC1 before the very good CD-player used earlier. Very nice indeed! /Joachim


----------



## joijwall

Another question: 
 I get the impression direct cd playback on my Mac sound better than ripping to AIFF via iTunes and then play the file - even in VLC. Is there a better (EAC-like) rip program for Mac? 
 (An observation: using TOAST and export to AIFF makes a file playable in iTunes, but not in VLC! I tried it because Toast had different rendering settings, and I thought Best looked more interesting than Normal or Fastest.) /Joachim


----------



## poo

Quote:


  Originally Posted by *joijwall* /img/forum/go_quote.gif 
_Another question: 
 I get the impression direct cd playback on my Mac sound better than ripping to AIFF via iTunes and then play the file - even in VLC. Is there a better (EAC-like) rip program for Mac? 
 /Joachim_

 

There are other apps, but not really 'better' ones IMO. iTunes should do a fine job - it's a lossless format after all, and not even compressed! Did you hear a difference or is it an 'impression' as you suggest?


----------



## joijwall

It's an impression. 
 But I actually found a ripping program called Max, that has some modes that reads the information several times before creating the AIFF-file (it's using CoreAudio for AIFF). There was a small difference (in bytesize) between the Max and the iTunes files. Only 5kB I think, but still.


----------



## EliasGwinn

Quote:


  Originally Posted by *poo* /img/forum/go_quote.gif 
_... can you provide the same info for my other system - Win XP running iTunes on a PC?_

 

Actually, the same directions apply (just replace "CoreAudio/AudioMIDI" with "QuickTime").

 In other words...

 1. *The ‘Set It And Forget It’ solution for iTunes 7.x*: Before opening iTunes, set the sample rate in of QuickTime to 96 kHz. That is all. By having iTunes locked at 96 kHz, all audio with sample rates below 96 kHz will be up-sampled to 96 kHz. As the quality of iTunes’ 7.x SRC is very good (virtually inaudible), this will cause virtually no loss in sonic quality. Also, by avoiding down-sampling by iTunes, this setting will never result in a loss of bandwidth.

 2. *The ‘Bit-Transparency For Each Sample Rate’ solution*: Before opening iTunes, set the sample rate of QuickTime to that of the audio you will be playing. When you change the sample rate of QuickTime, iTunes will need to be restarted to operate at the new sample rate. This solution will avoid SRC by iTunes for all audio with the same sample rate that iTunes is locked to.

 Thanks,
 Elias


----------



## EliasGwinn

Quote:


  Originally Posted by *poo* /img/forum/go_quote.gif 
_I am using XLR outputs (unbalanced floating pin 3) connected to a Musical Fidelity A3.5 integrated Amp. 

 I currently have the 'Output Level Switch' on the factory 'variable' setting, so the volume control is active (DAC1 working as a pre-amp as I understand it).

 I don't want to use the DAC1 as a preamp - I think in my case it is better to use the integrated MF amp to do both pre and power, so I presume I have to switch to calibrated (which I haven't tried yet because of a warning in the manual referring to the 'calibrated' setting when driving power amps). I also don't want to have to remember to readjust the volume control after headphone use before switching back to the integrated amp.

 I don't have any tools or experience in altering the settings of calibration trimmers, so am not sure calibrated is the way to go either. 

 After reading the manual a few times - I get the impression that the DAC1 USB is factory set in such a way that all I really have to do is switch to calibrated and it will be fine as is, but I wanted to check my assumption is correct before destroying something 
	

	
	
		
		

		
		
	


	




 Advice on how to proceed would be much appreciated..._

 

As long as you keep the volume down on your Integrated Amp when you first hook up the DAC1, you won't destroy anything. The only thing at risk of damage is your speakers *IF *a high line-level signal input causes your amp to push too much output to your speakers. 

 But, in your scenario, I'm assuming that you haven't adjusted the DAC1's output attenuation jumpers from their factory setting. The factory setting is -20dB, as this signal level almost definitely will not cause amp-to-speaker damage. Also, in your case, you are only using one leg of the balanced output, which will be 6 dB less then full-balanced. Therefore, the highest level possible from the DAC1 (w/ -20 dB attn., floating pin-3) would be -2 dBu (which is equivalent to a -22 dBu average signal). In other words, you'll be fine.

 If you're still concerned, start with the DAC1 in variable mode and slowly bring up the volume. If you get the volume all the way up with no problems, you can switch to 'calibrated' with no problems.

 Thanks,
 Elias


----------



## EliasGwinn

Quote:


  Originally Posted by *joijwall* /img/forum/go_quote.gif 
_Another question: 
 I get the impression direct cd playback on my Mac sound better than ripping to AIFF via iTunes and then play the file - even in VLC. Is there a better (EAC-like) rip program for Mac? 
 (An observation: using TOAST and export to AIFF makes a file playable in iTunes, but not in VLC! I tried it because Toast had different rendering settings, and I thought Best looked more interesting than Normal or Fastest.) /Joachim_

 

Sorry, Joachim, I don't know the answer to this.

 -Elias


----------



## poo

Quote:


  Originally Posted by *EliasGwinn* /img/forum/go_quote.gif 
_Actually, the same directions apply (just replace "CoreAudio/AudioMIDI" with "QuickTime")._

 

Thanks - pretty obvious really, I always forget QT is back there behind iTunes... 
	

	
	
		
		

		
		
	


	




 Thanks for the info re: calibrated vs variable - figured it would be fine, but prefer to check first when there are that many components involved 
	

	
	
		
		

		
		
	


	




 Looking forward to getting home 
	

	
	
		
		

		
		
	


	





 Home now... and happy... oh so very happy...


----------



## little-endian

Quote:


  Originally Posted by *joijwall* /img/forum/go_quote.gif 
_Another question: 
 I get the impression direct cd playback on my Mac sound better than ripping to AIFF via iTunes and then play the file - even in VLC. Is there a better (EAC-like) rip program for Mac?_

 

If your copy which resides in the AIFF-Container is free of errors, then differences in audio quality are impossible despite of jitter since one transfers the message, not the messenger (like Bob Katz tends to say).

 You should check for this.


----------



## Wavelength

Quote:


  Originally Posted by *joijwall* /img/forum/go_quote.gif 
_Another question: 
 I get the impression direct cd playback on my Mac sound better than ripping to AIFF via iTunes and then play the file - even in VLC. Is there a better (EAC-like) rip program for Mac? 
 (An observation: using TOAST and export to AIFF makes a file playable in iTunes, but not in VLC! I tried it because Toast had different rendering settings, and I thought Best looked more interesting than Normal or Fastest.) /Joachim_

 

Joijwall,

 One of the inherent benefits of computer audio is that any error free ripping will result in a secure file on the computer that will play the same way all the time.

 Most error free ripping programs will read a track several times create a 32 bit bcc and if most of the rips are the same assume the track is correct.

 When you play a cd on a computer or a transport for that matter it does not have the time to re-read if it get's an error, therefore it will never sound as good as a ripped song.

 Also CD's don't last forever... they degrade over time. This is not the case with a file which can be copied over to new drive etc and will be the same image until altered or deleted.

 ~~~~

 Sorry that I have not been around too much. CES in 2 weeks and have several amplifiers and preamps to finish up.

 Thanks
 Gordon


----------



## joijwall

Thanks Gordon,
 I guess I'm a bit sound-paranoid at the moment, wanting my very new DAC to perform from the best only. My hopes and goals was and is to go for iTunes and Apple Lossless from an external harddrive. But there were some comments about iTunes 7.5 not having bit-transparent playback on Mac (VLC's playback was said to bit bit-transparent though, and I believe I hear a difference.), which also made me shaky regarding it's ripping skills. Perhaps I'm being over cautious and should relax and only worry about cleaning the disks. Still, I'm looking forward to more test results from you technical people. 
	

	
	
		
		

		
		
	


	




 Regards, Joachim


----------



## EliasGwinn

Quote:


  Originally Posted by *joijwall* /img/forum/go_quote.gif 
_Thanks Gordon,
 I guess I'm a bit sound-paranoid at the moment, wanting my very new DAC to perform from the best only. My hopes and goals was and is to go for iTunes and Apple Lossless from an external harddrive. But there were some comments about iTunes 7.5 not having bit-transparent playback on Mac (VLC's playback was said to bit bit-transparent though, and I believe I hear a difference.), which also made me shaky regarding it's ripping skills. Perhaps I'm being over cautious and should relax and only worry about cleaning the disks. Still, I'm looking forward to more test results from you technical people. 
	

	
	
		
		

		
		
	


	




 Regards, Joachim_

 

Joachim,

 Don't worry, your paranoia is justified. Just don't let it get the better of ya! 
	

	
	
		
		

		
		
	


	




 Plus, as long as you keep in touch with us, we should be able to get past most bugs together...

 iTunes should work fine as long as you follow the directions I posted a few pages back. 

 Enjoy your Holidays!
 -Elias


----------



## Wavelength

Gang,

 I agree with Elias on this... don't get hung up on preception it will drive you crazy.

 I have more than 1500 cd's ripped by iTunes and have not found any issues with it.

 What ever you decide it's mostly in the setup. Make sure it has error correction. Make sure the vol slider is were it needs to be and most of all turn off all the bs like sound enhancers or dsp stuff.

 That is when it will sound the best.

 FYI I loaded VLC and I don't find the application very well written. Two users of mine cannot even figure out how to work it. If I get time I will put it on the CES system today and see how well it works.

 Thanks
 Gordon


----------



## EliasGwinn

Hey folks,

 We will be releasing a pair of tech-notes that deal specifically with computer playback systems. If you care to receive them, go to this link:

Feedback Newsletter

 Don't worry, there is no marketing B.S. that comes with it. No product announcements, no promo material... Nothing but cold, hard, objective tech papers (okay, there is some colored fonts and backgrounds on the pages, but don't let it fool you - the facts are cold and hard!! 
	

	
	
		
		

		
			





).

 Thanks,
 Elias


----------



## joijwall

I'm all ears about VLC! Do you use the 0.8.6d Janus version? If VLC performs better than iTunes in playback I'll stick to it. Sometimes sound is more important even than usability, looks, Apple-devotion etc. 
	

	
	
		
		

		
		
	


	



 Great idea with news letter, Elias! 
 /Joachim


----------



## cansman

Quote:


  Originally Posted by *EliasGwinn* /img/forum/go_quote.gif 
_Hey folks,

 We will be releasing a pair of tech-notes that deal specifically with computer playback systems. If you care to receive them, go to this link:

Feedback Newsletter

 Don't worry, there is no marketing B.S. that comes with it. No product announcements, no promo material... Nothing but cold, hard, objective tech papers (okay, there is some colored fonts and backgrounds on the pages, but don't let it fool you - the facts are cold and hard!! 
	

	
	
		
		

		
		
	


	




).

 Thanks,
 Elias_

 

Thanks Elias. Looking forward to receiving the tech notes! Don't know whether you celebrate Christmas; either ways, season greetings!

 cansman


----------



## mofonyx

EliasGwinn,

 Is there a measured DC offset on the Benchmark DAC-1's analogue outputs (XLR or RCA) if any?

 Thank you.


----------



## little-endian

If the topic is about the specs right now - Elias, what's the dynamic range of the DAC1? In the manual, only the SNR is mentioned.


----------



## Terje

Hello,

 Is it possible to listen to ordinary dvd cd`s(movies) when connected to dac1 usb?

 What about dvd audio?

 BlueRay and hddvd movies, when connected to dac1 usb?

 If it is not possible to get sound with these formats, can you explain why?


----------



## Terje

Sorry, double posting here.

 Hello, 

 Is it possible to hear the music
 and sound on ordinary dvd cd`s(movies) when connected to dac1 usb?

 DvdAudio?

 BlueRay and hddv?

 If not, can you explain why?


----------



## infinitesymphony

Quote:


  Originally Posted by *Terje* /img/forum/go_quote.gif 
_Is it possible to hear the music
 and sound on ordinary dvd cd`s(movies) when connected to dac1 usb?

 DvdAudio?

 BlueRay and hddv?

 If not, can you explain why?_

 

Like all stereo DACs, it's only possible to hear the audio if it is downmixed to 2.0 stereo before being sent to the DAC. It's the job of external DSP to downmix surround content (like in a HT receiver or in software on a computer), so most stereo-only DACs don't handle it.


----------



## little-endian

In general, your DAC1 wants Stereo PCM with a pretty wide range of supported sample rates and word lengths up to 24 Bit. When feeding it via USB, the SR is limited to 96 kHz, though.

 By the way, if you think a soundcard in a PC makes this better - they also always receive PCM, no matter which format you're going to play. DACs aren't aware of lossy compression schemes like AC3, MP1/2/3, Vorbis, etc. but (L)PCM only. Other constructions have built-in decoders.


----------



## EliasGwinn

Hey Head-Fiers!

 Just got back from a very relaxing Holiday vacation. I hope your holidays were enjoyed as much as possible!!

 I'm sorry to leave your questions hanging. I'll answer them as soon as possible.

 Thanks,
 Elias


----------



## ataraxia

This will be off-topic, but I'm glad you enjoyed your holidays Elias.


----------



## EliasGwinn

Quote:


  Originally Posted by *mofonyx* /img/forum/go_quote.gif 
_EliasGwinn,

 Is there a measured DC offset on the Benchmark DAC-1's analogue outputs (XLR or RCA) if any?

 Thank you._

 

Hello Mofonyx...sorry I haven't gotten to this question sooner.

 The DC offset on the DAC1's output is +/- 5 mV, depending on gain setting.

 Thanks,
 Elias


----------



## EliasGwinn

Quote:


  Originally Posted by *little-endian* /img/forum/go_quote.gif 
_If the topic is about the specs right now - Elias, what's the dynamic range of the DAC1? In the manual, only the SNR is mentioned._

 

little-endian,

 The dynamic range of the DAC1 is 116 dB.

 Thanks,
 Elias


----------



## EliasGwinn

Quote:


  Originally Posted by *Terje* /img/forum/go_quote.gif 
_Hello,

 Is it possible to listen to ordinary dvd cd`s(movies) when connected to dac1 usb?

 What about dvd audio?

 BlueRay and hddvd movies, when connected to dac1 usb?

 If it is not possible to get sound with these formats, can you explain why?_

 

Terje,

 The DAC1 is compatible with any 2-channel PCM data. Therefore, if a DVD video (or DVD audio or BlueRay or HD-DVD) is formated with 2-channel PCM audio, the DAC1 will convert it properly. Also, even if the audio on a disc is not PCM format, many DVD players will convert the audio to PCM.

 Thanks,
 Elias


----------



## EliasGwinn

Quote:


  Originally Posted by *ataraxia* /img/forum/go_quote.gif 
_This will be off-topic, but I'm glad you enjoyed your holidays Elias. 
	

	
	
		
		

		
		
	


	


_

 

Thank you Ataraxia. I hope you enjoyed yours as well. 

 How could I not be happy after I got some real, old-fashion meals to stick to my bones. 
	

	
	
		
		

		
		
	


	




 God bless the mothers of the world


----------



## Ooztuncer

Hello,
 I am the new member to DAC1 family and i have two quick questions:

 1) Is there a difference between January 2003 built DAC1 and the current version, except the USB input? I bought my DAC1 this morning from one of the recording studios and the owner remembers that it was updated at least once by Benchmark. Is there a way to learn what has changed/serviced? He thinks that the variable-level-pot was updated but he is not sure. I have the serial number but no build date sticker.

 2) I am waiting to go home and put this puppy into action. I read somewhere in this thread that some people prefers to set XLR output attenuation jumper to 0db (rather than -20db by default). According to my manual it is already at 0db by factory default - am I wrong?

 Thanks!
 Onur


----------



## EliasGwinn

Quote:


  Originally Posted by *Ooztuncer* /img/forum/go_quote.gif 
_Hello,
 I am the new member to DAC1 family and i have two quick questions:
_

 

Hello Onur, welcome to the DAC1 Family! 
	

	
	
		
		

		
			





  Quote:


  Originally Posted by *Ooztuncer* /img/forum/go_quote.gif 
_Hello,
 1) Is there a difference between January 2003 built DAC1 and the current version, except the USB input? I bought my DAC1 this morning from one of the recording studios and the owner remembers that it was updated at least once by Benchmark. Is there a way to learn what has changed/serviced? He thinks that the variable-level-pot was updated but he is not sure. I have the serial number but no built date sticker._

 

Since January 2003, there have been several revisions to the DAC1. 

 In May 2003, the XLR output attenuators were added. 

 In November 2004:
 - the RCA output impedance was changed from 1.5 k-ohm to 30 ohms
 - the optical input was upgraded to operate at 192 kHz
 - the DC offset was reduced, and a detented volume control was used
 - a trim-pot was added to properly balance between channels.

  Quote:


  Originally Posted by *Ooztuncer* /img/forum/go_quote.gif 
_2) I am waiting to go home and put this puppy into action. I read somewhere in this thread that some people prefers to set XLR output attenuation jumper to 0db (rather than -20db by default). According to my manual it is already at 0db by factory default - am I wrong?_

 

I'm not totally sure if your DAC1 has output attenuators. Also, when you buy a DAC1 used, there is no way to know what the attenuators are set to without opening the chassis and looking (unless you want to measure the output voltage with a known digital input). 

 However, you can determine whether the attenuators are set properly for your system without opening the chassis. Here's how: 

*Before *you turn your amplifier on, make sure the volume control of the DAC1 is all the way down, and make sure the DAC1 is in "Variable" mode (see the switch on the back panel). After you do this, turn on the amplifier and begin streaming digital music to the DAC1. Slowly bring up the volume until you reach a comfortable level. If this level is not achieved within the volume control range of 10 and 5 o'clock, the output attenuators should be adjusted. 

 To determine the how to correctly set your attenuators for optimum sonic quality, I suggest reading this article about setting the output attenuators. 

 Thanks,
 Elias


----------



## mofonyx

I haved asked this before, but I would just like to make sure.

 I know it's all right to leave the Benchmark DAC-1 turned on (because there is no "off" switch). 

 But is it okay to leave it on with the volume knob turned on to my listening level and with my headphones plugged in all the time? Or would it be better to turn the volume knob to minimum when I put my headphones down?

 Also, 
  Quote:


  Originally Posted by *EliasGwinn* 
_The DC offset on the DAC1's output is +/- 5 mV, depending on gain setting._

 

What do you mean by "gain setting" - Is it Variable, Calibrated and Default? and how much will the DC offset vary with different gain settings?

 Thanks again.


----------



## EliasGwinn

Quote:


  Originally Posted by *mofonyx* /img/forum/go_quote.gif 
_I know it's all right to leave the Benchmark DAC-1 turned on (because there is no "off" switch). 

 But is it okay to leave it on with the volume knob turned on to my listening level and with my headphones plugged in all the time? Or would it be better to turn the volume knob to minimum when I put my headphones down?_

 

It will not hurt the DAC1 at all. You may want to contact your headphone manufacturer if there are concerns about having constant audio through your headphones.

  Quote:


  Originally Posted by *mofonyx* /img/forum/go_quote.gif 
_What do you mean by "gain setting" - Is it Variable, Calibrated and Default? and how much will the DC offset vary with different gain settings?_

 

In other words, the DC offset will vary based on the gain settings of the volume control, trim-pots and/or attenuators. The higher the gain setting, the higher the DC offset. It will vary no more then 8mV.

 Thanks,
 Elias


----------



## mofonyx

Thanks for the quick reply Elias..


----------



## little-endian

Quote:


  Originally Posted by *EliasGwinn* /img/forum/go_quote.gif 
_The dynamic range of the DAC1 is 116 dB._

 

Interesting. So the dynamic range equals the signal to noise ratio. Because in most other systems, these two values often differ a bit.

 Actually, I always thought that the noise would limit the dynamic range per se but according to Bob Katz, the dynamic range can be even higher than the signal to noise ratio in dithered systems (~ 115 dB dynamic range while < 91 dB SNR on a properly dithered audio-cd for instance).

 At least, there seems to be no DAC available at all which would reach the theoretical limit of 144 dB for 24-bit-audio. I'm very sure that even the dynamic range of 16-bit-audio is enough in most listening situations but a bit funny if one recognizes all the "24 bit" here and there advertisement if even the finest equipment seems to not be able to catch up.


----------



## mofonyx

Quote:


  Originally Posted by *little-endian* /img/forum/go_quote.gif 
_Interesting. So the dynamic range equals the signal to noise ratio. Because in most other systems, these two values often differ a bit._

 

I believe that's just a coincidence.


----------



## xenithon

Hi all. I have a quick question regarding the XLR output attenuation jumpers...

 ...the DAC1 will be used with output set to "calibrated" and directly feeding an amplifier. I read that the default/factory setting is -20dB for the XLR output; adjustable via the jumpers. I have read though that the performance/quality is much better when these jumpers are set to 0dB (in the manual it says _0dB = attenuator disabled_).

 Has anyone tried this? Would you agree? Is there anything wrong or potentially harmful about using it like this? Everything else would be left at stock/factory settings.

 EDIT: just to add, it would be feeding a Stax SRM-717 energizer. Not sure if, or which specs would affect this, but what I know off hand is
 - Gain :60dB 
 - Rated Input Level :100mV (with 100V output) 
 - Input Impedance :50Kohm (RCA), 50kohm x 2 (XLR)


----------



## EliasGwinn

Quote:


  Originally Posted by *little-endian* /img/forum/go_quote.gif 
_Interesting. So the dynamic range equals the signal to noise ratio. Because in most other systems, these two values often differ a bit....

 ...
_

 

The dynamic range and signal-to-noise ratio of the DAC1 are both determined by the D/A chip used (AD1853).

 Thanks,
 Elias


----------



## EliasGwinn

Quote:


  Originally Posted by *xenithon* /img/forum/go_quote.gif 
_Hi all. I have a quick question regarding the XLR output attenuation jumpers...

 ...the DAC1 will be used with output set to "calibrated" and directly feeding an amplifier. I read that the default/factory setting is -20dB for the XLR output; adjustable via the jumpers. I have read though that the performance/quality is much better when these jumpers are set to 0dB (in the manual it says 0dB = attenuator disabled).

 Has anyone tried this? Would you agree? Is there anything wrong or potentially harmful about using it like this? Everything else would be left at stock/factory settings.

 EDIT: just to add, it would be feeding a Stax SRM-717 energizer. Not sure if, or which specs would affect this, but what I know off hand is
 - Gain :60dB 
 - Rated Input Level :100mV (with 100V output) 
 - Input Impedance :50Kohm (RCA), 50kohm x 2 (XLR)_

 

xenithon,

 The October edition of our "Feedback Newsletter" had a detailed article about the optimal attenuator settings for a system.

 From the spec's that you've listed, you will get significant clipping if you operate in Calibrated mode with 0 dB attenuation. The average level from the DAC1 with those settings is 1.22 Vrms, or about 12x the rated input level of the Stax.

 If you read the article in that link, you'll know everything you need to know to properly configure your system. And if you have any questions, please feel free to ask!! 
	

	
	
		
		

		
		
	


	




 Thanks,
 Elias


----------



## xenithon

Hi Elias. Thanks for the link and the info about the average output level. I will certainly follow that procedure in the newsletter to try and find the optimal attenuator settings. 

 What I am a little confused about though is that my previous DAC, the Musical Fidelity X-DAC V3, has a rated output of 2.2VRMS, and worked fine with the SRM-717 with no clipping.

 Admittedly though, perhaps all the exact specs would explain why this was so: 
 - Output at 0dB level: 2.2VRMS at 1kHz nominal 
 - Output impedance: 50 Ohms 

 Looking purely at the specs as per the newsletter, is there an attenuation setting which appears to be - at least on paper - ideal to match the Stax's specs?

 Thanks and have a great new year Elias!
 X


----------



## kool bubba ice

Quote:


  Originally Posted by *EliasGwinn* /img/forum/go_quote.gif 
_Terje,

 The DAC1 is compatible with any 2-channel PCM data. Therefore, if a DVD video (or DVD audio or BlueRay or HD-DVD) is formated with 2-channel PCM audio, the DAC1 will convert it properly. Also, even if the audio on a disc is not PCM format, many DVD players will convert the audio to PCM.

 Thanks,
 Elias_

 

So The DAC1 can support 2 channel Uncompressed PCM to take advantage of Blu Rays audio? Or Dolby HD? Or would I need a HDMI device, as I been told, next gen audio HD/BR needs more bandwidth that optical/coaxial doesn't have..


----------



## infinitesymphony

Quote:


  Originally Posted by *kool bubba ice* /img/forum/go_quote.gif 
_So The DAC1 can support 2 channel Uncompressed PCM to take advantage of Blu Rays audio? Or Dolby HD? Or would I need a HDMI device, as I been told, next gen audio HD/BR needs more bandwidth that optical/coaxial doesn't have.._

 

Coaxial can support up to 24-bit/192 kHz two-channel audio (and some people have found optical to work at that rate, too). However, transmitting 5.1+ channels of uncompressed 24/192 content requires a higher bandwidth than S/PDIF. But, as long as the player can downmix to two-channel, or the media you're using has a stereo track, you can use it with a stereo DAC like the DAC1.

 Edit: Except with the new HD formats, there are restrictions that prevent the signal from being digitally output over S/PDIF. So, the DAC1 can play 24/192 uncompressed PCM, but it will be almost impossible to find a player capable of outputting that information.


----------



## riverlethe

Elias, 
 I noticed the new DAC1/HD650 package and I'm curious as to why the HD650 was chosen. Have you performed any frequency response tests with various headphones?


----------



## EliasGwinn

Quote:


  Originally Posted by *xenithon* /img/forum/go_quote.gif 
_Hi Elias. Thanks for the link and the info about the average output level. I will certainly follow that procedure in the newsletter to try and find the optimal attenuator settings. 

 What I am a little confused about though is that my previous DAC, the Musical Fidelity X-DAC V3, has a rated output of 2.2VRMS, and worked fine with the SRM-717 with no clipping.

 Admittedly though, perhaps all the exact specs would explain why this was so: 
 - Output at 0dB level: 2.2VRMS at 1kHz nominal 
 - Output impedance: 50 Ohms 

 Looking purely at the specs as per the newsletter, is there an attenuation setting which appears to be - at least on paper - ideal to match the Stax's specs?

 Thanks and have a great new year Elias!
 X_

 

Hello X,

 What makes this assessment difficult is that different manufacturers use different methods for listing specs. When the amp says "Rated Input Level", does that mean it is the input level at which the THD+N spec was measured? Or is it the highest input level before the THD+N rises above a certain threshold? 

 You will probably want to contact the manufacturers to find out what the maximum input level is. But, at the end of the day, you will want to feed your amp only as much as needed to create the volume output desired for comfortable listening levels. That's what we are talking about in the article from the newsletter. The important thing is to make sure the volume control of the DAC1 is in its optimal range, and then you will get the best results.

 Thanks,
 Elias

 ps. sorry it has taken so long to respond....we're preparing for T.H.E. and NAMM and its C.R.A.Z.Y. around here!!


----------



## xenithon

No problem - and thanks for the info. I just got some balanced cables and have been doing some listening. I generally need to have the volume of the Stax amp at around 12 o'clock. On quieter albums it can go up to 2 o'clock, rarely higher. The DAC1 volume control is not being used (in calibrated mode).


----------



## EliasGwinn

Quote:


  Originally Posted by *kool bubba ice* /img/forum/go_quote.gif 
_So The DAC1 can support 2 channel Uncompressed PCM to take advantage of Blu Rays audio? Or Dolby HD? Or would I need a HDMI device, as I been told, next gen audio HD/BR needs more bandwidth that optical/coaxial doesn't have.._

 

Kool,

 If a Blu-Ray or HD-DVD player can output 2-channel uncompressed PCM via coax, optical, or XLR, then the DAC1 will support it.

 Will you need an HDMI device? This question is difficult to answer because the protocols and regulations change so often, often without official declaration or documentation. 

 Thanks,
 Elias


----------



## EliasGwinn

Quote:


  Originally Posted by *riverlethe* /img/forum/go_quote.gif 
_Elias, 
 I noticed the new DAC1/HD650 package and I'm curious as to why the HD650 was chosen. Have you performed any frequency response tests with various headphones?_

 

Great question. We did not choose the HD650's because of frequency response or distortion tests, but because we are very familiar with the HD650's sonic performance and find them very capable. We've had the HD650's for a few years and are very happy with them.

 The DAC1/HD650 package was created as a response to a lot of feedback from our customers who wanted a high-performance, quality headphone system but don't have the time or resources to try every combination. Many of our customers simply call us and say, "I love the DAC1, and I want headphones, what should I buy?" This package makes it easier, and saves them a little bit of money.

 Thanks,
 Elias


----------



## riverlethe

Quote:


  Originally Posted by *EliasGwinn* /img/forum/go_quote.gif 
_Great question. We did not choose the HD650's because of frequency response or distortion tests, but because we are very familiar with the HD650's sonic performance and find them very capable. We've had the HD650's for a few years and are very happy with them.

 The DAC1/HD650 package was created as a response to a lot of feedback from our customers who wanted a high-performance, quality headphone system but don't have the time or resources to try every combination. Many of our customers simply call us and say, "I love the DAC1, and I want headphones, what should I buy?" This package makes it easier, and saves them a little bit of money.

 Thanks,
 Elias_

 

Ermm, actually $1275 + $350 = $1625.


----------



## vcoheda

not sure if this was mentioned. but in the recent issue of stereophile there is an article on the DAC1 and the article mentions this thread and provides the url.


----------



## xp9433

Quote:


  Originally Posted by *gregeas* /img/forum/go_quote.gif 
_Elias, 

 Since you're still here after all these months, I wanted to run a new product idea by you. 

 I can't be the only one who has the need for a new type of digital two-channel pre-amp. Essentially what I'm envisioning is a full set of digital inputs, a high-quality converter like the one in the DAC1, and remote controlled input switching and volume (with a nice analog pot). While you're at it make it full rack width... 

 Right now I'm using my Slimdevices Transporter for this; it converts digital signals from my cable box and PS3 and of course handles PC music playback with an excellent UI. It does the job pretty well -- conversion is very nice --but doesn't have an analog volume control, and the inputs can't easily be switched remotely (the process requires multiple button presses). So I've had to keep my pre-amp in the rack to make the system user friendly. 

 So really the only differences between what I'm proposing and the DAC1 are a remote control and perhaps more a few more inputs for consumer audio. If this product existed I could get rid of my pre-amp altogether and pair my digital pre-amp with my amps directly... Sweet. 

 I thought about getting a nice audiophile pre-pro, but I'd be paying for extra channels of conversion and video features I don't need._

 

Elias

 Full credit for againing showing your sensitivity to members on this site.
 Back in August the above suggestion was made, and Benchmark have responded (whether already on the drawing board or not).

 NEW PREAMP DAC FROM BENCHMARK: New Benchmark DAC1 PRE Stereo Playback Pre-amplifier | Computer Audiophile 

 Congratulations and thanks!

 Frank


----------



## infinitesymphony

Quote:


  Originally Posted by *xp9433* /img/forum/go_quote.gif 
_NEW PREAMP DAC FROM BENCHMARK: New Benchmark DAC1 PRE Stereo Playback Pre-amplifier | Computer Audiophile_

 

Looks good.

 We're just one step away from the DAC1 Surround Preamp.


----------



## poo

Why do I always time my purchases so poorly...


----------



## gregeas

Whoa! This is cool. 

 One question: no remote control, right? This could be great paired with a small amp like the Parasound Zamp or the PS Audio Trio.

 I love the way high-end equipment keeps getting smaller and smaller...


----------



## xp9433

Quote:


  Originally Posted by *poo* /img/forum/go_quote.gif 
_Why do I always time my purchases so poorly..._

 

Can't be too hard to sell or trade what you have?

 I was waiting to see what was new at CES before finalising my PC Audio development plan. The Benchmark DAC Pre is now a certainty for one of two system paths I am trying to decide on.

 Frank


----------



## poo

Quote:


  Originally Posted by *xp9433* /img/forum/go_quote.gif 
_Can't be too hard to sell or trade what you have?_

 

Three weeks after receiving it... 
	

	
	
		
		

		
		
	


	




 Yeah I could but the DAC1 USB is fine - just that I would have bought the PRE if I new about it.


----------



## EliasGwinn

Quote:


  Originally Posted by *poo* /img/forum/go_quote.gif 
_Three weeks after receiving it... 
	

	
	
		
		

		
		
	


	




 Yeah I could but the DAC1 USB is fine - just that I would have bought the PRE if I new about it. 
	

	
	
		
		

		
		
	


	


_

 

Poo,

 If you bought your DAC1 USB within the last 30 days (and if it is in like-new condition), you can exchange it and get full value towards a DAC1 PRE.

 Thanks,
 Elias


----------



## poo

Quote:


  Originally Posted by *EliasGwinn* /img/forum/go_quote.gif 
_Poo,

 If you bought your DAC1 USB within the last 30 days (and if it is in like-new condition), you can exchange it and get full value towards a DAC1 PRE.

 Thanks,
 Elias_

 

Sweet 
	

	
	
		
		

		
		
	


	










 Ooops... got all excited and forgot to ask how I go about doing that...


----------



## tuffgong

The Pre looks like a serious giant killer, very nice. Some basic questions Elias, are the analog in's rca jacks? Did you all upgrade the jacks slightly like some people had asked in this thread? Can I use the rca in's to use as an input into the computer to create wav's of all my old cassettes/lp's, as oppossed to my very old fashioned hhb 800? To be able to hook all these sources up in one small box for 1.5k is amazing. Thanks.


----------



## Terje

Elias

 You have offered a very good support-service in this thread. Tremendous man-ours worked. 

 Can you devalue a comparism between Dac1 Usb and Dac Pre ?

 What is new, supported fileformats, different sound, is it possible to connect a turntable to analog in, etc ?

 Different range of use between Dac1 Usb and Dac Pre ?

 Everything you can bring forward that has something to do with differences between Dac1 Usb and Dac Pre, please. I am courious.

 Then you can copy it and paste it in this thread when you need.

 Thank you.


----------



## EliasGwinn

Quote:


  Originally Posted by *poo* /img/forum/go_quote.gif 
_Sweet 
	

	
	
		
		

		
			











 Ooops... got all excited and forgot to ask how I go about doing that..._

 

We're getting the details together for an exchange policy. I'll let you know as soon as we have it worked out. Until then, enjoy the DAC1 USB!

 Thanks,
 Elias


----------



## EliasGwinn

Quote:


  Originally Posted by *tuffgong* /img/forum/go_quote.gif 
_The Pre looks like a serious giant killer, very nice. Some basic questions Elias, are the analog in's rca jacks? Did you all upgrade the jacks slightly like some people had asked in this thread? Can I use the rca in's to use as an input into the computer to create wav's of all my old cassettes/lp's, as oppossed to my very old fashioned hhb 800? To be able to hook all these sources up in one small box for 1.5k is amazing. Thanks._

 

Tuffgong,

 The RCA and coaxial jacks have been ugraded to high-quality, chassis-mounted connectors.

 The DAC1 PRE does not convert analog signals to digital signals, so it can't be used for archiving analog sources. The analog inputs are for interfacing an analog source to the built-in HPA2 headphone amplifier or to the device that the DAC1 PRE is driving, such as an power amplifier or powered speakers. 

 The ADC1 USB can be used to archive. It will deliver high-quality, high-resolution wav's of cassettes, LP's, radio programs, etc. directly to the USB port on your computer or the digital input to your CD burner.

 Thanks,
 Elias


----------



## EliasGwinn

Quote:


  Originally Posted by *Terje* /img/forum/go_quote.gif 
_
 Can you devalue a comparism between Dac1 Usb and Dac Pre ?_

 

The DAC1 PRE is a whole new beast. It is Benchmark's first stereo system pre-amplifier. It can anchor an entire media system consisting of analog and digital sources, and deliver a pristine signal to your amplifier. It has 6 inputs, including analog, USB, optical and 3 coaxials! And, to officially bring it into the world of pre-amplifiers, it has a power button! 
	

	
	
		
		

		
		
	


	




 It also features the HPA2 0-ohm headphone amplifier. The analog and coaxial inputs feature high-quality chassis-mounted RCA type jacks. The analog circuit uses LM4562 throughout. The 4562 is the newest, state-of-the-tech, high-current, low-distortion opamps from National Instruments. The analog inputs are compatible with most analog sources including iPod, cassette, phono pre, tuner, etc. It is not a phono input, so it cannot connect directly with a turntable. A phono pre is required to interface a turntable to the DAC1 PRE.

 The sound of the DAC1 PRE is the familiar "Benchmark Sound" of clarity, definition, and accuracy.

 I hope that helps!!

 Thanks,
 Elias


----------



## Terje

Quote:


 I hope that helps!! 
 

Elias

 That helps. I am just an old "hobby wannabe computeraudio nerd" that needs some time to figure out things. I was almost on my way taking the subway to a pro audio shop. Phurschasing the DAC1 USB, but something else(not computer audio-hifi related) came up and took the money. 

 A couple of more questions, please.

 Is the difference between DAC1 USB and DAC1 PRE-anolog in, good analog circuits, uppgraded rca jacks, 1 more digital in, power button? 

 It is a difference. Is the analog XLR-Output connectors and circuits/drivers the same as in DAC1 USB?

 If you ONLY need the digital inputs that is already on the DAC1 USB, is there any difference in use, in stereo audio listening?

 Thank you.


----------



## poo

Quote:


  Originally Posted by *EliasGwinn* /img/forum/go_quote.gif 
_And, to officially bring it into the world of pre-amplifiers, it has a power button! 
	

	
	
		
		

		
		
	


	


_

 

Good lord! A what?!?!?!???


----------



## EliasGwinn

Another feature of the DAC1 PRE which I forgot to mention, and will be especially interesting to the Head-Fi community...:

 Another headphone gain range!! The DAC1 PRE offers a 0 dB, a -10 dB AND a -20 dB gain range for even the most sensitive headphones. The DAC1 USB offers only a 0 dB and a -10 dB gain range.

 Thanks,
 Elias


----------



## gregeas

Is there IR control for the Pre? This is key for making it useful in the living room IMO.


----------



## WindowsX

Could someone make some comparison between grace m902 and dac1 pre's specifications in terms of pre-amplifier?


----------



## Scrith

Given that it is designed to be used as a pre-amp, is the variable output circuitry (i.e. the volume control) in the DAC1 PRE significantly different/better than the DAC1 USB? I'm asking because I use the variable output from my DAC1 USB to control an amplifier directly (rather than going through a pre-amp with the calibrated output setting).


----------



## Mik

Quote:


  Originally Posted by *gregeas* /img/forum/go_quote.gif 
_Is there IR control for the Pre? This is key for making it useful in the living room IMO._

 

I'm interested in this question as well. If it has a remote, this would be the ideal 2ch dac/pre for me.


----------



## Drumonron

I'm interested in purchasing a used Benchmark DAC1 USB soo...
 Update @ 1/14/08- never mind, I pulled the trigger and a new DAC1 USB is 
 on the way.


----------



## EliasGwinn

Quote:


  Originally Posted by *gregeas* /img/forum/go_quote.gif 
_Is there IR control for the Pre? This is key for making it useful in the living room IMO._

 

Gregeas,

 There is no IR control for the DAC1 PRE. 

 Thanks,
 Elias


----------



## EliasGwinn

Quote:


  Originally Posted by *WindowsX* /img/forum/go_quote.gif 
_Could someone make some comparison between grace m902 and dac1 pre's specifications in terms of pre-amplifier?_

 

Would you like a comparison between performance spec's or product features?


----------



## Terje

Sorry, double posting.


----------



## Terje

Elias

 On the DAC1 PRE are the analog audio input be passed out through balanced outputs?

 Is the analog XLR-Output connectors and circuits/drivers the same as in DAC1 USB?

 If you ONLY need the digital inputs that is already on the DAC1 USB, is there any difference in use, in stereo audio listening?


----------



## EliasGwinn

Terje,

 The analog inputs of the DAC1 PRE are routed to both the unbalanced RCA outputs and the balanced XLR outputs.

 The main differences between the DAC1 PRE and the DAC1 USB:

 -DAC1 PRE has 5 digital inputs (USB, optical, coax(x3))
 -DAC1 USB has 4 digital inputs (USB, optical, coax(x1), and XLR)
 -DAC1 PRE uses teflon chassis-mounted RCA analog and digital connectors
 -DAC1 USB uses PCB mounted RCA connectors
 -DAC1 PRE has 3 gain ranges for the HPA2 headphone amplifier
 -DAC1 USB has 2 gain ranges for the HPA2 headphone amplifier
 -DAC1 PRE uses National Instrument LM4562 opamps throughout the analog circuits
 -DAC1 USB uses 5532's every except the output drivers, which uses the 4562's
 -DAC1 PRE has power button
 -DAC1 USB does not have power button

 Thanks,
 Elias


----------



## EliasGwinn

Quote:


  Originally Posted by *Scrith* /img/forum/go_quote.gif 
_Given that it is designed to be used as a pre-amp, is the variable output circuitry (i.e. the volume control) in the DAC1 PRE significantly different/better than the DAC1 USB? I'm asking because I use the variable output from my DAC1 USB to control an amplifier directly (rather than going through a pre-amp with the calibrated output setting)._

 

The DAC1 PRE utilizes the same volume control circuitry as the DAC1 USB.

 Thanks,
 Elias


----------



## rmh1

Elias,

 A couple of questions for you:

 1. When will the Pre be released?
 2. Will you be offering this in the black as well?
 3. Can you set the headphone outs independantly so one can be set for a high impedance phone and the other a lower impedance?
 4. Are these changes made internally with jumpers?
 5. Did you find that nobody was using the AES/EBU input on the DAC1 - hence the decision to offer the three coaxial inputs on the pre?

 Looks like a very nice product.

 Robb


----------



## EliasGwinn

Quote:


  Originally Posted by *rmh1* /img/forum/go_quote.gif 
_Elias,

 A couple of questions for you:

 1. When will the Pre be released?
 2. Will you be offering this in the black as well?
 3. Can you set the headphone outs independantly so one can be set for a high impedance phone and the other a lower impedance?
 4. Are these changes made internally with jumpers?
 5. Did you find that nobody was using the AES/EBU input on the DAC1 - hence the decision to offer the three coaxial inputs on the pre?

 Looks like a very nice product.

 Robb_

 

1. We are taking orders now. The first unit will ship mid-February
 2. We currently do not have plans to offer a black face plate for the DAC1 PRE, but this may change
 3. Both headphone jacks are driven by the same amplifier, so they cannot be driven independently
 4. The headphone gain range selection is made internally with jumpers
 5. The DAC1 PRE features 3 coax inputs because it is designed to anchor a media system with devices like digital cable, digital TV, satellite digital radio, DVD players, etc. These types of devices devices rarely have XLR digital outputs. This was the basis for this decision.

 Thanks,
 Elias


----------



## rmh1

Great, thanks for the quick response. This looks like the product I need. A suggestion though for future products: An external gain control for headphone output. With people here using such a wide array of headphones, from very low impedance to the sugggested Senns in your package, to even the 600 ohm Beyers, it would be nice to not have to move the jumpers to get the full volume control range for each headphone. I realize this would require a whole new layout of the board, but a nice feature most amp producers here incorporate. Other than that, and the black option, looks like you have made another winner. Congrats.

 Robb


----------



## Matias

I have an ESI Juli@ sound card and I use the ASIO plug-in for Winamp. I have great improvements in soundstage by enabling 176.4kHz in Ultra mode on the plug-in.

 I'm just about to buy a DAC1 USB, and my question is: *is it worth doing 176.4kHz upsampling and dithering before feeding the DAC1?*

 I already read ~80 pages of the thread, I know it upsamples everything to 110kHz, the optimum DAC frequency, but anyways, subjective testing is also important. Of course I'll test it when I get mine, but I would like to hear from you guys, and from Elias as well.

 Thanks.


----------



## Terje

Quote:


 Great, thanks for the quick response. This looks like the product I need. A suggestion though for future products: An external gain control for headphone output. With people here using such a wide array of headphones, from very low impedance to the sugggested Senns in your package, to even the 600 ohm Beyers, it would be nice to not have to move the jumpers to get the full volume control range for each headphone. I realize this would require a whole new layout of the board, but a nice feature most amp producers here incorporate. Other than that, and the black option, looks like you have made another winner. Congrats. 
 

I agree.

 One more thing. In a living room when you are sitting in the sofa or in an deep chair it is nice with a remote control.

 Thanks

 Terje


----------



## Terje

Elias,

 One last question from me, or more correctly from another member in the slimdevices forum.

  Quote:


 Would be good if it had the option of HT-Bypass on the analogue input (for integration with HT setup), I also wonder if it retains the auto turn on/off feature of the DAC1 USB model.

 I already own an original DAC1 and have been very happy with it even at its higher cost to other DACs like the Citypulse etc 
 

.

 Thanks for the response.

 Terje


----------



## happybob

seems like each generation of the benchmark media dac1 is a little more interesting. but i agree with most of the posters above, a remote would make this product much more desirable. without a remote, even with all the extra inputs, the new dac1 pre still seems better suited for desktop use. maybe benchmark is going after the growing number of people who use their computers as the multimedia, tv, dvd, blu-ray center? a dac without a remote is okay since you can still use the remote that comes with the pre-amp or the integrated amp, but as a pre-amp, if it wants to make the way into the living room, a remote is a must!


----------



## gregeas

Quote:


  Originally Posted by *happybob* /img/forum/go_quote.gif 
_seems like each generation of the benchmark media dac1 is a little more interesting. but i agree with most of the posters above, a remote would make this product much more desirable. without a remote, even with all the extra inputs, the new dac1 pre still seems better suited for desktop use. maybe benchmark is going after the growing number of people who use their computers as the multimedia, tv, dvd, blu-ray center? a dac without a remote is okay since you can still use the remote that comes with the pre-amp or the integrated amp, but as a pre-amp, if it wants to make the way into the living room, a remote is a must!_

 

I agree with this. In an AV rack -- especially one with a TV -- a pre must have a remote. On a desktop, it doesn't matter. 

 Incidentally, in recent months I did get rid of my analog pre-amp and now use a Slimdevices Transporter as a "digital pre-amp" to control my cable box, game consoles, and, of course, music. 

 There are a few drawbacks to using the Transporter -- switching inputs isn't as easy as it should be -- but all in all it does well. We're probably one generation away from what I imagine is the perfect device.


----------



## EliasGwinn

Ah, the remote control. 
	

	
	
		
		

		
			





 I know, I know... the DAC1 needs a remote. 
	

	
	
		
		

		
		
	


	




 We have not yet found an independent volume controller of high enough quality to suit the DAC1 product line. Many preamps use a digitally controlled volume control IC, which has significant issues with regards to audio quality. We don't want to go that route because we feel it would compromise all the things the DAC1 PRE was created for...ultra-low distortion playback controller.

 I really want to say, however, that your suggestions and other feedback are very much appreciated. We will take all of this into account during future product development. 

 Thanks,
 Elias


----------



## gregeas

Well, I should add that I still have and love my USB DAC1, which I'm using as a souce (not pre-) in my other systems.


----------



## EliasGwinn

Ok, friends...I'm off to Anaheim, CA for the NAMM convention. I'll be back Tuesday...I'll talk to you all then!! 
	

	
	
		
		

		
		
	


	




 Thanks,
 Elias


----------



## Drumonron

Have a safe trip and a good time Elias. I am looking forward to my Benchmark DAC1 USB and it should arrive on 1/16/08-I'll post my comments on it. I appreciate, as always, your tireless responses...your informational nuggets are like gold. Your an incredible asset to Benchmark and Head-fi.

 Thanks


----------



## Wavelength

Elias,

 Have fun at NAMM... was going to go but caught the flu at CES. Wanted to stop by and chat but the room was too busy for that.

 ~~~~~

 I got the latest Stereophile with John's remarks from the review of your unit and I was wondering if you could clarify the statement. iTunes 7.5 introduces truncation and DSP attifacts as high as -80dB only when used at 44.1K and that 96K works fine. (detail #1).

 Allot of times the Manufactuer's Comments are written quickly because they give you freaken 24 hours to respond.

 Was this due to the double sampling problem associated with setting the Audio Midi sample rate or some other problem.

 Anyways have fun at the show.

 Thanks
 Gordon


----------



## mcbiff

I just bought a DAC1 (non-USB) from a fellow member here at head-fi. It will be used between my laptop and my speaker system which consists of a Densen B110 amplifier and a pair of Monitor Audio RS6 speakers. Now, I have read most of this thread, and the issue has been touched upon to some extent, but I feel I need to ask once and for all:

 Should I set the output to variable or calibrated if connecting the DAC1 to my Densen using RCA (i.e. unbalanced) cables?

 If I did miss an answer to this specific question I do apologize, and if someone could point me in the right direction I'd be grateful. Otherwise I hope one of the Benchmark representatives has the time to give me a straight answer. 

 My understanding from what I've gathered here is that I won't damage anything as long as I make sure to turn down the volume of my amplifier when switching the DAC1 to calibrated. I just feel (perhaps incorrectly) that if I don't need the pre-amp functionality of the DAC1 I should cut that part out of the equation so to speak.

 Thanks


----------



## infinitesymphony

Quote:


  Originally Posted by *mcbiff* /img/forum/go_quote.gif 
_Should I set the output to variable or calibrated if connecting the DAC1 to my Densen using RCA (i.e. unbalanced) cables?

 My understanding from what I've gathered here is that I won't damage anything as long as I make sure to turn down the volume of my amplifier when switching the DAC1 to calibrated. I just feel (perhaps incorrectly) that if I don't need the pre-amp functionality of the DAC1 I should cut that part out of the equation so to speak._

 

As you have hinted, the real reason for variable output is to use the DAC1 as a preamplifier to feed a power amplifier (either in active monitors or a standalone amp). Calibrated mode essentially turns the DAC1 into a normal source, with an output no higher than that which you'd see from a typical CD player or DAC.

 The only situation where you'd use variable output to feed a preamplifier (like the one in your Densen integrated amplifier) is if the preamp can't handle the output level of the DAC1. This would result in distortion due to overload. However, most amps are designed with a fair amount of headroom to prevent overloading the inputs.

 So, calibrated.


----------



## happybob

Quote:


  Originally Posted by *EliasGwinn* /img/forum/go_quote.gif 
_Ah, the remote control. 
	

	
	
		
		

		
		
	


	




 I know, I know... the DAC1 needs a remote. 
	

	
	
		
		

		
		
	


	




 We have not yet found an independent volume controller of high enough quality to suit the DAC1 product line. Many preamps use a digitally controlled volume control IC, which has significant issues with regards to audio quality. We don't want to go that route because we feel it would compromise all the things the DAC1 PRE was created for...ultra-low distortion playback controller.

 I really want to say, however, that your suggestions and other feedback are very much appreciated. We will take all of this into account during future product development. 

 Thanks,
 Elias_

 

what about as a compromise have a remote that can at least allow source switching? would that also impact the audio quality?


----------



## mcbiff

Quote:


  Originally Posted by *infinitesymphony* /img/forum/go_quote.gif 
_As you have hinted, the real reason for variable output is to use the DAC1 as a preamplifier to feed a power amplifier (either in active monitors or a standalone amp). Calibrated mode essentially turns the DAC1 into a normal source, with an output no higher than that which you'd see from a typical CD player or DAC.

 The only situation where you'd use variable output to feed a preamplifier (like the one in your Densen integrated amplifier) is if the preamp can't handle the output level of the DAC1. This would result in distortion due to overload. However, most amps are designed with a fair amount of headroom to prevent overloading the inputs.

 So, calibrated. 
	

	
	
		
		

		
		
	


	


_

 

Awesome. Thanks so much for the help!


----------



## poo

Quote:


  Originally Posted by *infinitesymphony* /img/forum/go_quote.gif 
_As you have hinted, the real reason for variable output is to use the DAC1 as a preamplifier to feed a power amplifier (either in active monitors or a standalone amp). Calibrated mode essentially turns the DAC1 into a normal source, with an output no higher than that which you'd see from a typical CD player or DAC.

 The only situation where you'd use variable output to feed a preamplifier (like the one in your Densen integrated amplifier) is if the preamp can't handle the output level of the DAC1. This would result in distortion due to overload. However, most amps are designed with a fair amount of headroom to prevent overloading the inputs.

 So, calibrated. 
	

	
	
		
		

		
		
	


	


_

 

Agreed - has been a much better way to go for me...


----------



## Wavelength

Gang look it has been assumed that Benchmark has some inherent problem with apple iTunes 7.5 that mysteriously occured. They state in their Detail#1 in the latest Stereophille MC page 135 that when using 441 and iTunes 7.5 that a DSP errors to -80dB and truncation occurs that severly limits the quality of sound from any device. They also state that using VLC resolves this issue.

 I did the following test: PowerMac G5---->USB--->Benchmark====> Prism dScope III Audio Analyzer

 iTunes 7.5 using a 10K test tone showing the tone and the FFT analysis of the waveform:






 VLC using the same 10K test tone:






 As you can see there is very little difference between these two and there really should not be. As both of them use Core Audio and if they are both running at 44.1 with the correct setting then this should not happen.

 The test file is a digitally created 10KHz wav file no compression. That is indicated by the FFT spike at 10KHZ.

 Guys let's face it... I don't see the problem they are stating even with their equipment.

 NOW if it is because of sample rate adjustment somewhere then ok. But this is the way it has worked on an Apple since day 1. You have to set the correct rate before you enter the iTunes application.

 Let's face it really the only problem with iTunes is that Lossless appears to work in 16 bit only files. If you want to use 24 then AIFF, WAV or other file is required.

 If I am missing something please advise.

 Thanks, back to work have tons of new products to get out. Thanks all for those that stopped by at CES. We had a great deal of fun!
 Gordon


----------



## EliasGwinn

Gordon,

 Your findings correspond to our final conclusions (both here and here ), also stated several times on this thread as well. The statement you refer to in this post was an initial observations only...not final conclusions. 

 Please refer to our final assessments of iTunes 7 when quoting us in future posts.

 Thanks,
 Elias


----------



## shed

Can the DAC1 accept a S/PDIF signal over the AES XLR jack?


----------



## EliasGwinn

Quote:


  Originally Posted by *shed* /img/forum/go_quote.gif 
_Can the DAC1 accept a S/PDIF signal over the AES XLR jack?_

 

Yes.

 Thanks,
 Elias


----------



## clarke68

Hey Elias,

 As others have repeatedly mentioned, this is a great thread! Your presence here is helpful, and says a lot about Benchmark's respect for their customer base.

 Question for you about the future: is Benchmark looking into supporting HDMI? High-res players (like the Oppo 980) are coming on the market that support SACD over HDMI, and it would be nice to be able to drive them into something other than a home theater receiver.

 Thanks! Hope 'ya had a good time at NAMM.


----------



## EliasGwinn

clarke,

 We currently do not have any plans to integrate an HDMI interface in our products. Also, we don't support the SACD format, as it has a lot of unresolved issues relating to audio quality.

 Thanks,
 Elias


----------



## joijwall

Reading Gordons Post I couldn't avoid raising the eyebrow on "the only problem with iTunes is that Lossless appears to work in 16 bit only files". I use iTunes to rip my cds to Apple Lossless and listen from my hdd. Is this good strategy, or should I avoid Apple Lossless? Thanks! /Joachim


----------



## Wavelength

Elias,

 It might be a good idea for you and John to write another Comment for Stereophile to print. The one in the recent Stereophile makes it look like iTunes is broken.

 When in fact the only real problem has been there all along. That being it does not change sample rates on the fly.

 ~~~~~~~~~

 Joachim,

 The nice thing about lossless is just that it will be original content. You can convert this to other formats without losing information.

 One thing we found that on slower machines Apple Lossless does not sound as spacious as AIFF/WAV files do. At CES we showed users that had slower computers to convert their files to AIFF and the space was recovered.

 So using Lossless now and changing later will not result in loss or errors of the files you have.

 Thanks
 Gordon


----------



## Angsila

Gordon,

 What do you define to be "slower machines" ?


----------



## jdh500

I'd be interested in the answer to the question on whether the DAC1 pre can be configured with an *HT bypass *feature on the Analogue input ? If not currently available, is it something that can be considered as an option in any future revision.

 JDH.


  Quote:


  Originally Posted by *Terje* /img/forum/go_quote.gif 
_Elias,

 One last question from me, or more correctly from another member in the slimdevices forum.

 Thanks for the response.

 Terje_


----------



## EliasGwinn

Quote:


  Originally Posted by *jdh500* /img/forum/go_quote.gif 
_I'd be interested in the answer to the question on whether the DAC1 pre can be configured with an *HT bypass *feature on the Analogue input ? If not currently available, is it something that can be considered as an option in any future revision.

 JDH._

 

JDH,

 The DAC1 PRE doesn't need an HT Bypass feature because it does not have a HT-mode to bypass.

 Hope that helps...

 Thanks,
 Elias


----------



## Wavelength

Angsila,

 Well the best way to understand this is with some insite on how we tested this.

 First in content with all computers we tested the results were identical. It was only with spatial content that we could tell the difference in computer speed and file types.

 We could not tell say the difference between a WAV and AIFF file on any of the systems. For the test since in previous tests we concluded that Apple Lossless was truely lossless that we converted 4 tracks into AIFF and WAV format.

 Computers: iBook 800mhz G4, MacBook Core 2 duo 2.2Ghz, Mac Mini 1.6GHZ Core Duo, MacBook Pro 2.2Ghz and iMac Core 2 Extreme 2.8GHZ

 My personal system: Cain & Cain WWWS speakers, Corona Cobalt version VT52 amps, Royal preamplifier and Crimson Silver DAC.

 With the iBook it was most apparent that the AIFF/WAV files retrieved more air and more depth and height. With the MacBook and MacMini it was a little more air and mainly depth.

 System 2: Wilson Watt Puppy 8, Levinson amp & pre, Crimson Silver. The MacBook Pro seemed really good and revealing in this system but then we upgraded to the new iMac with the Extreme processor and the space increased immensly.

 CES System: Vaughn Zinfandel (97dB effecient 25-40KHz), Silver Cardinal 300B Amplifiers, Royal Preamplifier both Cosecant V3 and Crimson Silver. Using the Mac Mini the depth on AIFF/WAV files over Lossless was mainly in depth. It seem to blow back past the back wall. Using the MacBook there was just a little less difference.

 ~~~~~

 Gang remember these are revealing systems retailing for more than most of the cars we are driving. Therefore it's easier to hear the difference.

 The test is easy one to try...

 In iTunes basically set your importing to AIFF or WAV:

 Preferences->Advanced->Importing: Import Using

 Then select say 3-4 songs that you know well and do:

 Advanced: Convert Songs to AIFF/WAV

 Play the lossless then play the converted and see what you think.

 Thanks
 Gordon


----------



## Scrith

Gordon, your results indicate at least one of two problems that I can think of off the top of my head:

 1) iTunes needs to use a bigger buffer for playback (and start playback once the buffer has more data in it), because the buffer must be running out of data at some point during playback due to iTunes not being able to decompress fast enough. I don't mean to imply that the computer is too slow (although I suppose that could also be the case, I highly doubt a modern computer isn't capable of decompressing Apple Lossless at 44.1K/sec), but rather than there could be CPU usage "hiccups" (by background processes, for example) during playback that causes iTunes to decompress more slowly, resulting in the buffer becoming empty at certain times (which would be audible, and ideally it would generate an error message for the user, though I highly doubt a program like iTunes would do such a thing).

 2) Heavy CPU usage causes the part of the Mac OS that transfers data to the USB bus (or the part that schedules these transfers) to act erratically. This would indicate that the Mac OS is a poor choice for music playback with USB devices (since it should be capable of something like scheduling time-sensitive data transfers).

 The options seem to be 1) get a faster Mac (although this may just make the problem happen less often, since faster computers can still become very busy with background processes), 2) avoid iTunes, 3) don't run other programs when using iTunes, 4) don't use audio formats that require decompression during playback, or 5) stop using Macs.

 The implied solution seems to be #4 (to not use Apple Lossless), but this is a poor conclusion (in my opinion) because the Apple Lossless data format (or any other lossless format, e.g. FLAC) will have zero effect on the sound one hears if the software and hardware that supports it are functioning properly. The cause of this problem is iTunes or the Mac OS, not the lossless data format.

 I mention this because I am sick of reading a post with someone saying something like "WAV sounds better than FLAC on my system" and then reading responses from a bunch of lemmings that echo that statement throughout the internet, causing countless people to begin believing that FLAC files must somehow not be as good as WAV files. Your post reminded me of the type of post that ends up generating such a reaction, even if that reaction was not the one that was intended.


----------



## Wavelength

Scrith,

 First these tests were done on mac not a pc. On a pc there are other issues that make people go crazy over formats and they have reason for this. From my testing here is a basic problem with the PC.

 There are no commom dll's that do encoding, decoding of files. Therefore there are 4 compiler versions and no common code base so you rip in one and play in another and the code base, compilers etc... can change the character of sound.

 In my exprerience Vista is much better base for audio than XP. The usb drivers are much better written and the consistency of sound is much easier to accomplish. I personally think it is an annoying operating system constantly asking me if I want to do this or that. I double clicked the damn application, why would you ask me if I want to run it?

 Anyways, remember these are suddle sound differences on $50k or more systems. Most people would not be able to hear the differences in their own systems.

 I have never had any problems with pops or clicks.

 I may do some FFT plots of different applications and file types but I am pretty busy right now. I think on the PC you would be able to see the differnces.

 But like the posts I did for VLC and iTunes on the MAC, I don't think you are going to see any differnce as CoreAudio is doing all the work there.

 Thanks
 Gordon


----------



## qlee

Hi Elias,

 This is a great thread, a ton of information! Thanks much.

 After playing with my DAC1 USB for three months, I am very happy with it. Now I have a little question on "bit perfect", again

 I used a USB sniffer(15 day demo of USB Monitor Pro) to capture the traffic on the USB port connecting to DAC1. Test file was created in Audition, 1 second of silence, 1 second of 440Hz Sine wave. I created three versions of each, 8 bit, 16 bit, 24 bit, all in 44.1KHz. Player is foobar 0.9.5 on WinXP/SP2. Here is the finding on the USB port,

 1) The word length is always 24 bit, no matter the source word length, or foorbar output setting. Foobar will round to its 'output data format' first, then either kmixer or usbaudio will pad it to 24 bit.

 2) The last bit of audio is not bit-perfect, it's -1 at 40%, +1 rarely, no change for the rest.

 16bit sample, one channel shown, Sine
 Original Captured
 000000 FFFFFF -1
 080500 080500
 100200 100200
 17ED00 17ECFF -1

 24bit sample, one channel shown, Sine
 Original Captured
 000000 FFFFFF -1
 0804CC 0804CC
 100187 100186 -1
 17EE2A 17EE29 -1

 This last bit change is assumed to be caused by kmixer, it should not affect SQ, as it's way below the -120db noise floor. But technically, it's not bit perfect

 3) Tried WMP, same thing on 16bit, it doesn't play 24bit file.

 4) This is very obivious to see on 1 second silence file, as 40% 000000 changed to FFFFFF, a few changed to 000001.

 5) Tried asio4all, this one works with dac1, and it IS bit perfect!

 So I was wondering if I missed anything here, or DAC1 is just 23 bits perfect due to kmixer mangling the last bit. I did follow all the instructions. My test setup can be easily reproduced on any PC, any one interested to try on his/her setup?

 Thanks,
 Qlee


----------



## Wavelength

Qlee,

 Basically the way I understand it is that Foobar will output via DirectSound (& KMIXER) the size deemed by the source material (i.e. 8 bits, 16bits, 24 bits).

 Any USB device can declare themselves as either fixed or variable size. In most cases with base code from TI for the TAS1020 the device is fixed as shown here in the benchmark enumeration:

 Audio Class Specific Audio Data Format 
 Audio Stream Format Type Desc. 
 Format Type: 1 PCM
 Number Of Channels: 2 STEREO
 Sub Frame Size: 3
 Bit Resolution: 24
 Sample Frequency Type: 0x04 (Discrete)
 Sample Frequency: 44100 Hz
 Sample Frequency: 48000 Hz
 Sample Frequency: 88200 Hz
 Sample Frequency: 96000 Hz

 Therefore the usb device driver will pad all data to the appropriate size before outputing the data stream.

 The idea of bit perfect is easy as long as you play by the rules and bypass the KMIXER. Any device can therefore be bit perfect.

 But really your test is a little flawed as it merly shows the data from the PC to the DAC. Therefore this test could be true of any dac and not only that of the Benchmark.

 Thanks
 Gordon


----------



## EliasGwinn

qlee,

 We are currently discussing this topic with a Microsoft engineer who helped build Kmixer. We had a lengthy conference call the other day, and we learned a lot about the inner-operation of kmixer. I hope to have some conclusive information for you all soon. I won't post anything specific until we have a clear understanding.

 However, in your particular setup, you may be seeing dither applied to the audio in Audition. In fact, I'd be surprised if Audition _wasn't_ applying dither. This dither will cause random changes in the LSB of the audio.

 This is not to say that kmixer is bit-perfect...we are looking into this. (And I might add that Thomas may be vindicated 

 Thanks,
 Elias


----------



## qlee

Thank you, Gordon and Elias, for the reply.

 DAC1 only takes 24bit audio, there is nothing wrong here.

 I used Audition only to generate the test file, especially the 1sec silence is easy to spot if anything changes. Dither or not doesn't really matter here. I compared the USB capture to the .wav file with a hex editor(xvi32). There is no audition involved here during playback.

 When using Audition to play back, the result is the same, 24th bit got changed.

 If I understand right, the audio flow is like, 
 .wav -> foobar -> kmixer -> usb driver
 And the difference is between .wav and usb capture.

 Again .wav -> foobar -> asio4all -> usb driver is bit perfect.
 Somehow Kernel Streaming in foobar doesn't work with foobar on my setup.

 The whole point is DAC1 is 23bit perfect, which is good enough for audio play by all means. And kmixer is just a blackbox with no documented behavior, which is the pain in the b..t and we have spent all the time around it. I tend to believe ms did it on purpose

 The other thing is my 16bit Plantronic usb headset is bit-perfect when foobar is set to 8 or 16bit. It got +/-1 on 16th bit when foobar is set to 24 or 32 bits. Interesting.

 I mean guys, get a USB sniffer, and try it on yours.

 Thanks,
 qlee


----------



## thomaspf

I can't say I have been happy with the way kmixer or even Vista work. I just spend uncountable hours getting to the bottom of the details.

 As you found out yourself getting any crisp answer is very difficult but I am glad I could finally introduce you to the right person. I am very interested what you finally conclude in your tests.

 qlees test is completely identical to what I see. However, the bit modifications can spread at least to the second lowest bit as well.

 Cheers

 Thomas


----------



## riverlethe

Mr. Gwinn, I am curious as to why the DAC1 PRE uses LM4562's exclusively. Does Benchmark now believe that op-amps may create an audible difference, or is Benchmark simply caving to market demands?


----------



## EliasGwinn

Quote:


  Originally Posted by *riverlethe* /img/forum/go_quote.gif 
_Mr. Gwinn, I am curious as to why the DAC1 PRE uses LM4562's exclusively. Does Benchmark now believe that op-amps may create an audible difference, or is Benchmark simply caving to market demands?_

 

Riverlethe,

 The short answer is... BOTH!!

 The long answer is... Benchmark has always known that op-amps create audible differences. The 5532 has a bad reputation because it is old technology and it has been mis-used in ill-designed circuits with inevitably poor results. However, if the 5532 circuit is designed right, it is a very capable, low-noise, low-distortion op-amp. 

 The 4562 is sonically equivalent to the 5532. However, it output-performs the 5532 in its ability to deliver higher currents before distortion. This gives it the distinct advantage of setting up low output-impedance gain circuits, which is a great advantage. 

 Also, with regards to "caving to market demands", the 4562 is quickly becoming recognized for its great performance and sonic purity. Since this 'hype' has actual merit, we have no problem boasting about our new favorite op-amp 
	

	
	
		
		

		
			





 .

 Thanks,
 Elias


----------



## riverlethe

Quote:


  Originally Posted by *EliasGwinn* /img/forum/go_quote.gif 
_Riverlethe,

 The short answer is... BOTH!!

 The long answer is... Benchmark has always known that op-amps create audible differences. The 5532 has a bad reputation because it is old technology and it has been mis-used in ill-designed circuits with inevitably poor results. However, if the 5532 circuit is designed right, it is a very capable, low-noise, low-distortion op-amp. 

 The 4562 is sonically equivalent to the 5532. However, it output-performs the 5532 in its ability to deliver higher currents before distortion. This gives it the distinct advantage of setting up low output-impedance gain circuits, which is a great advantage. 

 Also, with regards to "caving to market demands", the 4562 is quickly becoming recognized for its great performance and sonic purity. Since this 'hype' has actual merit, we have no problem boasting about our new favorite op-amp 
	

	
	
		
		

		
		
	


	




 .

 Thanks,
 Elias_

 

I have no doubt it's a great piece of equipment.

 Do you believe the DAC1 PRE sounds better than the DAC1 USB or DAC1?


----------



## Wavelength

Qlee,

 What you are seeing is just the way it works. Thomas has pointed this out in previous posts as to what is and what is not bit perfect.

 The great thing is this is not set in stone it's easy at some point to fix these problems and they exist for all products not just the Benchmark.

 I have a USB Analyzer here that I use all the time. It is hooked between the development system (USB DAC) and the computer under test. It's a valuable tool for this type of product.

 Kernel streaming problems may indicate that your device driver for the USB hardware is not current or slightly mismatched. You can check for the most current from the maker of your computer or mother board.

 ~~~~~~

 Thomas, please keep us all informed!

 ~~~~~~

 Elias, yea go with the new National parts they are much better than the NE553x series!

 Thanks
 Gordon


----------



## poo

Quote:


  Originally Posted by *EliasGwinn* /img/forum/go_quote.gif 
_Poo,

 If you bought your DAC1 USB within the last 30 days (and if it is in like-new condition), you can exchange it and get full value towards a DAC1 PRE.

 Thanks,
 Elias_

 

Hey Elias... any news on the procedures involved to get this underway? Obviously been far longer than 30 days now...


----------



## EliasGwinn

Quote:


  Originally Posted by *poo* /img/forum/go_quote.gif 
_Hey Elias... any news on the procedures involved to get this underway? Obviously been far longer than 30 days now..._

 

Poo,

 I'm sorry we haven't finalized this yet... we've been crazy busy with NAMM and THE.

 We will still honor this policy, and I promise to get the details together very soon.

 Thanks for your patience,
 Elias


----------



## furball

Hi Elias,

 Even though I have just registered, I have been a long time lurker of this thread. Thank you for putting all your time to answering our questions. 
	

	
	
		
		

		
			





 With the DAC1 PRE coming out, I just couldn't control myself any longer. Just placed an order for a DAC1 PRE yesterday. Hoping to get it by mid February, when it ships.
	

	
	
		
		

		
		
	


	




 Regarding the new DAC1 PRE, I have a couple of questions. I am not too technically proficient, so please bear it with me.

 1) What is the DAC1 PRE's factory default gain setting for the headphone out? Currently I have a Sony MDR-SA5000 headphone. I think the impedance is somewhere in the 70 Ohms range. Do you think the default setting is optimal for use with this headphone? If not, what setting should I change it to?

 2) I also have an Etymotic ER4S. I think the impedance on that one is around 100 Ohms. What gain setting should I use for this headphone?

 3) If I change the default gain setting on the headphone out using the internal jumpers, would that change the gain setting on the audio out (those RCA jacks on the back) at the same time? Or would that not affect the gain setting of the audio out?

 4) I looked at the PDF manual for the DAC1 USB version, and tried to understand how to work those jumpers. But I am really technically deficient.
	

	
	
		
		

		
		
	


	




 I can't make out what is what. Is there a more detailed illustrated step by step instruction on how to set those jumpers?

 5) Is there still a constant level audio out on the DAC1 PRE? I know on the previous versions there are both constant and variable level audio out, you just need to toggle that switch on the back. The picture of the back panel on your website is a bit fuzzy, the same toggle is still there, but I can't make out the markings.




 And just a couple of suggestions for future improvements, if I may.

 1) Can you include more than one set of analog in? This will make it more like a real preamp. Because I have a tape deck, and a tuner deck. I really don't want to constantly plug and unplug cables all the time. 
	

	
	
		
		

		
		
	


	




 2) About the face place... I am a fan of matched colors. I prefer the color of the volume knob match that of the color of the faceplate. So everything is either all black (or shades of black), or all silver (or shadese of silver). The DAC1 PRE has a black knob with a silver faceplate. I don't know, it's kind of ugly, if I may say so. 
	

	
	
		
		

		
		
	


	




 Anyways, just a suggestion for future improvements. I decided to purchase one despite the color scheme, because I want the sound quality. But I would love it more if the color scheme were more consistent.


----------



## poo

Quote:


  Originally Posted by *EliasGwinn* /img/forum/go_quote.gif 
_Poo,

 I'm sorry we haven't finalized this yet... we've been crazy busy with NAMM and THE.

 We will still honor this policy, and I promise to get the details together very soon.

 Thanks for your patience,
 Elias_

 

No problem at all - still loving the DAC1 USB 
	

	
	
		
		

		
		
	


	




 Just wanted to be sure I didn't miss something!


----------



## EliasGwinn

Regarding the DAC1 PRE exchange program, we will be directly contacting customers who purchased a DAC1 and/or DAC1 USB between 12/04/07 and 1/04/07 AND registered the product with us. Please monitor your email in the next few days for correspondence regarding this. Please PM me with any questions or concerns directly relating to this.

 Thanks,
 Elias


----------



## EliasGwinn

Quote:


  Originally Posted by *furball* /img/forum/go_quote.gif 
_Hi Elias,

 Even though I have just registered, I have been a long time lurker of this thread. Thank you for putting all your time to answering our questions. 
	

	
	
		
		

		
		
	


	




 With the DAC1 PRE coming out, I just couldn't control myself any longer. Just placed an order for a DAC1 PRE yesterday. Hoping to get it by mid February, when it ships.
	

	
	
		
		

		
		
	


	




 Regarding the new DAC1 PRE, I have a couple of questions. I am not too technically proficient, so please bear it with me.

 1) What is the DAC1 PRE's factory default gain setting for the headphone out? Currently I have a Sony MDR-SA5000 headphone. I think the impedance is somewhere in the 70 Ohms range. Do you think the default setting is optimal for use with this headphone? If not, what setting should I change it to?

 2) I also have an Etymotic ER4S. I think the impedance on that one is around 100 Ohms. What gain setting should I use for this headphone?

 3) If I change the default gain setting on the headphone out using the internal jumpers, would that change the gain setting on the audio out (those RCA jacks on the back) at the same time? Or would that not affect the gain setting of the audio out?

 4) I looked at the PDF manual for the DAC1 USB version, and tried to understand how to work those jumpers. But I am really technically deficient.
	

	
	
		
		

		
		
	


	




 I can't make out what is what. Is there a more detailed illustrated step by step instruction on how to set those jumpers?

 5) Is there still a constant level audio out on the DAC1 PRE? I know on the previous versions there are both constant and variable level audio out, you just need to toggle that switch on the back. The picture of the back panel on your website is a bit fuzzy, the same toggle is still there, but I can't make out the markings.




 And just a couple of suggestions for future improvements, if I may.

 1) Can you include more than one set of analog in? This will make it more like a real preamp. Because I have a tape deck, and a tuner deck. I really don't want to constantly plug and unplug cables all the time. 
	

	
	
		
		

		
		
	


	




 2) About the face place... I am a fan of matched colors. I prefer the color of the volume knob match that of the color of the faceplate. So everything is either all black (or shades of black), or all silver (or shadese of silver). The DAC1 PRE has a black knob with a silver faceplate. I don't know, it's kind of ugly, if I may say so. 
	

	
	
		
		

		
		
	


	




 Anyways, just a suggestion for future improvements. I decided to purchase one despite the color scheme, because I want the sound quality. But I would love it more if the color scheme were more consistent._

 

Furball,

 I haven't forgotten about you, but I don't have answers to all of your questions. I promise to let you know as soon as I have any information with regards to your questions.

 Thanks,
 Elias


----------



## KarateKid

Sorry, just a quick question. What does the latest version of the DAC1 USB uses for opamp? Does the opamp affect the XLR output connection?

 Thanks!

 One more question, when should one expect to see the DAC1 Pre available in stores locally here in Canada? I'd like to try it out as it looks like a mini-upgrade from the already praised DAC1 USB.


----------



## EliasGwinn

Quote:


  Originally Posted by *furball* /img/forum/go_quote.gif 
_1) What is the DAC1 PRE's factory default gain setting for the headphone out? Currently I have a Sony MDR-SA5000 headphone. I think the impedance is somewhere in the 70 Ohms range. Do you think the default setting is optimal for use with this headphone? If not, what setting should I change it to?

 2) I also have an Etymotic ER4S. I think the impedance on that one is around 100 Ohms. What gain setting should I use for this headphone?_

 

The factory default gain setting for the HPA2 headphone amp in the DAC1 PRE will be -20dB, or 20 dB below that of the HPA2 gain range of the classic DAC1. 
 The -20dB setting offers plenty of level, even for the Sennheiser 650's, even when listening to quiet passages in classical recordings. The Sony's you mentioned have a sensitivity of 102dB @ 1mW. The Etymotic ER4S have a sensitivity of 100 dB @ 1mW. ...both significantly more sensitive then the Sennheiser 650's (low 90's...give or take).



  Quote:


  Originally Posted by *furball* /img/forum/go_quote.gif 
_3) If I change the default gain setting on the headphone out using the internal jumpers, would that change the gain setting on the audio out (those RCA jacks on the back) at the same time? Or would that not affect the gain setting of the audio out?_

 

The gain jumpers for the headphones do not affect the XLR or RCA outputs. Also, the attenuation jumpers for the XLR outputs do not affect the RCA outputs.



  Quote:


  Originally Posted by *furball* /img/forum/go_quote.gif 
_4) I looked at the PDF manual for the DAC1 USB version, and tried to understand how to work those jumpers. But I am really technically deficient.
	

	
	
		
		

		
		
	


	




 I can't make out what is what. Is there a more detailed illustrated step by step instruction on how to set those jumpers?_

 

We currently do not have more illustrations, but when you have the unit in front of you and take a look at the jumpers, hopefully it will be more obvious. If you have need any more assistance, simply call or email me....or contact me through this forum.



  Quote:


  Originally Posted by *furball* /img/forum/go_quote.gif 
_5) Is there still a constant level audio out on the DAC1 PRE? I know on the previous versions there are both constant and variable level audio out, you just need to toggle that switch on the back. The picture of the back panel on your website is a bit fuzzy, the same toggle is still there, but I can't make out the markings._

 

Yes, the DAC1 PRE also has a "Calibrated" level output setting to maintain a constant gain setting.


  Quote:


  Originally Posted by *furball* /img/forum/go_quote.gif 
_And just a couple of suggestions for future improvements, if I may.

 1) Can you include more than one set of analog in? This will make it more like a real preamp. Because I have a tape deck, and a tuner deck. I really don't want to constantly plug and unplug cables all the time. 
	

	
	
		
		

		
		
	


	


_

 

This would be a whole new product (....in a much bigger box too!! 
	

	
	
		
		

		
		
	


	




). When we design a product, we usually use customer suggestions to determine feature sets. The feedback from most of our customers was that more digital inputs were desired over more analog inputs to accommodate the constantly-growing digital media systems (Digital cable, digital satellite radio, digital TV, etc). Most of our customers don't have a need for more then one analog input. But, we appreciate your feedback, and will consider it when designing new products.



  Quote:


  Originally Posted by *furball* /img/forum/go_quote.gif 
_2) About the face place... I am a fan of matched colors. I prefer the color of the volume knob match that of the color of the faceplate. So everything is either all black (or shades of black), or all silver (or shadese of silver). The DAC1 PRE has a black knob with a silver faceplate. I don't know, it's kind of ugly, if I may say so. 
	

	
	
		
		

		
		
	


	




 Anyways, just a suggestion for future improvements. I decided to purchase one despite the color scheme, because I want the sound quality. But I would love it more if the color scheme were more consistent._

 

Haha.... Fair enough. I actually like the black-on-silver look...but its one of those things where you can't please everyone. Again, thanks for your suggestions. We will keep them in mind in the future...

 Thanks,
 Elias


----------



## EliasGwinn

Quote:


  Originally Posted by *KarateKid* /img/forum/go_quote.gif 
_Sorry, just a quick question. What does the latest version of the DAC1 USB uses for opamp? Does the opamp affect the XLR output connection?

 Thanks!

 One more question, when should one expect to see the DAC1 Pre available in stores locally here in Canada? I'd like to try it out as it looks like a mini-upgrade from the already praised DAC1 USB._

 

The DAC1 USB uses the LM4562's for output drivers and 5532's everywhere else. They will have lower distortion and better high-freq response when the DAC1 is driving high capacitance and/or low impedance loads. 

 The DAC1 PRE will begin shipping mid-Febuary. The first batch is quickly selling out, so you may want to tell your local audio dealer to order soon.

 Thanks,
 Elias


----------



## furball

Thank you Elias! You are the best!

 Now, if I can just get my hands on that DAC PRE right now... 
	

	
	
		
		

		
		
	


	




 I can't wait to play with it. 
	

	
	
		
		

		
		
	


	





 About the headphone out gain setting. If the -20dB setting is more than enough to drive the HD650's, isn't that a little too high for the Sony and the Etymotic?

 I had a Benchmark DAC1 from many moons ago (must be 3 years ago now), the headphone out was really loud. Using the Etymotic ER4S, I couldn't turn the volume past the first 2 clicks before getting painfully loud.

 An even lower setting for more sensitive headphones would be really appreciated for future products.


----------



## EliasGwinn

Quote:


  Originally Posted by *furball* /img/forum/go_quote.gif 
_About the headphone out gain setting. If the -20dB setting is more than enough to drive the HD650's, isn't that a little too high for the Sony and the Etymotic?_

 

No, it should be fine. My point was that, even though the -20dB is the lowest setting, it was still enough for even the least sensitive. However, it didn't get too loud until almost at full volume. You should be able to get to 10-12 o'clock on the volume control with no problem, depending on the type of music and listening preference. 

 When you get your DAC1 PRE, please let me know how well the volume control works with your headphones. I'd be very interested...

 Thanks,
 Elias


----------



## furball

I will definitely let you know. And thank you Elias for answering all my questions! 
	

	
	
		
		

		
		
	


	





  Quote:


  Originally Posted by *EliasGwinn* /img/forum/go_quote.gif 
_When you get your DAC1 PRE, please let me know how well the volume control works with your headphones. I'd be very interested..._


----------



## KarateKid

Just a question for all of you, in case anyone knowledgeable could answer it:

 Elias said that the "driver outputs" are the only spots where the oamps in the dac1 usb are lm4562, the rest are TI5532s.... what does "driver output" mean (what headphone output it applies to)?

 Thanks for all the help


----------



## jdh500

Sorry Elias it doesn't really help, I think I have to assume you don't understand what a HT bypass feature actually is? To make it clearer, most hi-end 2ch pre-amp manufactures incorporate a feature that directs the output 2ch L & R from an AV processor though the 2ch pre-amp input which has zero gain (ie. HT bypass), and allows the AV pre-amp to do volume control etc rather than the 2ch pre-amp. Basically a handy feature to have if your looking at a easy way to integrate a surround sound system with a dedicated 2ch system.

 At this stage I assume it does not have the HT bypass feature on the analogue input which is a bit of a negative in terms of system integration. Looks like I will have to stick with another brand of dedicated 2ch pre-amp and use the benchmark mainly as a DAC and headphone amp.

 ARE you able to answer my other question regarding the auto turn on feature or if it maintains its last state either on/off when power is applied to the Benchmark DAC 1 pre. this is also an important feature for system integration.

 Are you going to have the manual up for the DAC 1 pre on your web site soon ?

 Regards,

 JDH.


  Quote:


  Originally Posted by *EliasGwinn* /img/forum/go_quote.gif 
_JDH,

 The DAC1 PRE doesn't need an HT Bypass feature because it does not have a HT-mode to bypass.

 Hope that helps...

 Thanks,
 Elias_


----------



## riverlethe

Quote:


  Originally Posted by *jdh500* /img/forum/go_quote.gif 
_Sorry Elias it doesn't really help, I think I have to assume you don't understand what a HT bypass feature actually is? To make it clearer, most hi-end 2ch pre-amp manufactures incorporate a feature that directs the output 2ch L & R from an AV processor though the 2ch pre-amp input which has zero gain (ie. HT bypass), and allows the AV pre-amp to do volume control etc rather than the 2ch pre-amp. Basically a handy feature to have if your looking at a easy way to integrate a surround sound system with a dedicated 2ch system.

 At this stage I assume it does not have the HT bypass feature on the analogue input which is a bit of a negative in terms of system integration. Looks like I will have to stick with another brand of dedicated 2ch pre-amp and use the benchmark mainly as a DAC and headphone amp.

 ARE you able to answer my other question regarding the auto turn on feature or if it maintains its last state either on/off when power is applied to the Benchmark DAC 1 pre. this is also an important feature for system integration.

 Are you going to have the manual up for the DAC 1 pre on your web site soon ?

 Regards,

 JDH._

 

Can't you just use calibrated volume control?


----------



## furball

If you want the AV receiver to do the volume control, then what is the point of hooking up that AV receiver to a preamp?

 Your post is confusing.

 What is the signal path?
 1) From receiver to preamp to amp?
 2) From preamp to receiver to amp?


 If it is signal path #1 you are talking about, use the receiver's pre-out to connect to the preamp.

 If it is signal path #2 you are talking about, DAC1 has a calibrated (fixed volume) audio out, which you can use to connect to the receiver's audio in.



  Quote:


  Originally Posted by *jdh500* /img/forum/go_quote.gif 
_To make it clearer, most hi-end 2ch pre-amp manufactures incorporate a feature that directs the output 2ch L & R from an AV processor though the 2ch pre-amp input which has zero gain (ie. HT bypass), and allows the AV pre-amp to do volume control etc rather than the 2ch pre-amp. Basically a handy feature to have if your looking at a easy way to integrate a surround sound system with a dedicated 2ch system.
_


----------



## poo

Quote:


  Originally Posted by *furball* /img/forum/go_quote.gif 
_If you want the AV receiver to do the volume control, then what is the point of hooking up that AV receiver to a preamp?

 Your post is confusing.

 What is the signal path?
 1) From receiver to preamp to amp?
 2) From preamp to receiver to amp?


 If it is signal path #1 you are talking about, use the receiver's pre-out to connect to the preamp.

 If it is signal path #2 you are talking about, DAC1 has a calibrated (fixed volume) audio out, which you can use to connect to the receiver's audio in._

 

Guys look into what HT bypass actually is first - it is a great feature used in many 2ch amps (Musical Fidelity A5 or A35 integrated for example).

 The basic advantage is that the 2ch amp can drive the front speakers (as it usually would in a 2ch setup), meaning that the AV receiver doesn't have to, thereby better driving the center and surrounds.

 I'm not sure of how much benefit you would gain by adding HT bypass to a device like the DAC1 PRE over a good quality HT receiver though...


----------



## infinitesymphony

But the DAC1 PRE isn't an amp. The DAC1 PRE isn't even a typical preamp/processor. It's just a DAC with a lot of inputs...

 IMO, HT bypass would not make sense on the DAC1 PRE. The DAC1 PRE is not capable of decoding a surround signal, so there's no way to get it to handle only the front two channels and bypass the rest.

 HT bypass doesn't make a lot of sense in general. It's the equivalent of a tape loop...


----------



## poo

Quote:


  Originally Posted by *infinitesymphony* /img/forum/go_quote.gif 
_The DAC1 PRE is not capable of decoding a surround signal_

 

It doesn't need to, the AVR does that... 

 HT bypass means you send the pre-out of your mains from the AVR into your 2 channel gear. There is no decoding and no volume control from the 2ch amp used.

 As I have explained (and will try to do a better job now 
	

	
	
		
		

		
			





), it means that you can use a better quality version of what you already have in your AVR, by bypassing the AVR's internal preamp and passing it on to a 2 channel integrated amp for the front speakers.

 Not only are you now using a better quality preamp, you are taking that job away from your AVR so it can do a better job of driving the center and surround channels. You still only use the volume control of the AVR to control the whole system.

 Hope this helps explain HT Bypass a little better.


----------



## poo

Actually, you would be able to test the effectiveness of including HT Bypass in the DAC1 PRE now anyway since it can already be used as a pre amp.

 All you need to do is...

 Simply run an RCA from the mains/front pre out from your AVR to an input on the DAC1 PRE. You set the level for the fronts via the level settings on the AVR and by keeping the volume knob on the DAC1 PRE at a set point - say 10 to or at 12 O'clock position.

 To listen to HT you switch to the input on the DAC1 PRE connected to the AVR (ensuring that the volume is in the correct position used for leveling earlier of course).

 All your doing is routing the line level signal through your pre-amp - as you would do with any other line level signal, and as would otherwise happen if using HT bypass...


 Having said all of this, I doubt the DAC1 PRE used in this way will offer a significant (if any) improvement over a decent HT receiver...


----------



## infinitesymphony

Quote:


  Originally Posted by *poo* /img/forum/go_quote.gif 
_Having said all of this, I doubt the DAC1 PRE used in this way will offer a significant (if any) improvement over a decent HT receiver..._

 

It seems like that path could only hurt sound quality... The signal has to run through the receiver's DAC and preamp output sections, then out over cables, through the preamp section of the DAC1, back over cables, and through another amp's input section.

 Why not just connect the receiver's preamp outputs to a front channel power amplifier? What's the point of running the signal through the DAC1 PRE if its best capabilities aren't being used?


----------



## poo

Quote:


  Originally Posted by *infinitesymphony* /img/forum/go_quote.gif 
_What's the point of running the signal through the DAC1 PRE if its best capabilities aren't being used?_

 

The point would be to _potentially_ have a better PRE stage working for you... again - I'm not suggesting anyone would get any benefit from doing this - just that it is possible, and in effect very similar to the HT bypass feature that jdh500 was asking about.


----------



## infinitesymphony

Quote:


  Originally Posted by *poo* /img/forum/go_quote.gif 
_The point would be to potentially have a better PRE stage working for you... again - I'm not suggesting anyone would get any benefit from doing this - just that it is possible, and in effect very similar to the HT bypass feature that jdh500 was asking about._

 

But the signal is already running through the receiver's preamp... 
	

	
	
		
		

		
		
	


	




 The only way to avoid the receiver's built-in preamp would be to plug directly into the DAC1 PRE, which is the whole point of having it...

 *shrug* In any case, you have to admit that this feature is not essential, especially since the DAC1 has a calibrated output.


----------



## furball

I think you are confused with terminologies here.

 A receiver's preout only outputs a fixed volume signal - meaning the signal strength from the preout does not change when you rotate that volume knob on the receiver.

 Every decent receiver has a preouts.


 So you want to connect your receiver's preout to DAC1 PRE's audio in, and essentially use the DAC1 PRE for volume control. In essence, there is no such feature as an "HT Bypass feature," it's just a fancy name referring to the preout on your receiver.


 If you do intend to connect the preout from your receiver to the audio in on the DAC1 PRE, you are really just using the DAC1 PRE for volume control. Is the potentiometer in the DAC1 PRE that much better than the potentiometer inside your receiver?





  Quote:


  Originally Posted by *poo* /img/forum/go_quote.gif 
_Actually, you would be able to test the effectiveness of including HT Bypass in the DAC1 PRE now anyway since it can already be used as a pre amp.

 All you need to do is...

 Simply run an RCA from the mains/front pre out from your AVR to an input on the DAC1 PRE. You set the level for the fronts via the level settings on the AVR and by keeping the volume knob on the DAC1 PRE at a set point - say 10 to or at 12 O'clock position.

 To listen to HT you switch to the input on the DAC1 PRE connected to the AVR (ensuring that the volume is in the correct position used for leveling earlier of course).

 All your doing is routing the line level signal through your pre-amp - as you would do with any other line level signal, and as would otherwise happen if using HT bypass...


 Having said all of this, I doubt the DAC1 PRE used in this way will offer a significant (if any) improvement over a decent HT receiver..._


----------



## infinitesymphony

Quote:


  Originally Posted by *furball* /img/forum/go_quote.gif 
_A receiver's preout only outputs a fixed volume signal - meaning the signal strength from the preout does not change when you rotate that volume knob on the receiver._

 

That's not true for any receiver I've used. Preamp outputs generally are for connecting to a power amp. If they were fixed output, the system would be at full volume all the time.

 I'm going to start another thread about HT bypass, just because it's so confusing to me, and because this discussion isn't really about the DAC1. 
	

	
	
		
		

		
		
	


	




 Edit: Here's the thread: What is the purpose of the HT Bypass feature?


----------



## CanMad

HT bypass is unity gain i.e. the signal passed into the pre-amp is passed out at exactly the same level. The idea of it on a pre-amp is to allow the integration of a HT set-up with a two channel set-up. i.e. the main left and right speakers and the poweramp(s) driving them are used for both HT and 2 channel listening.

 HT bypass on the DAC1 pre, would mean that the volume control on the DAC1 pre does not affect the level of the signal input on the analogue inputs. When in HT bypass the volume control on the HT avr would be controlling the volume.

 Basically the DAC1 pre has a HT bypass i.e. set the input on the DAC1 pre to the analogue input, which would have the L and R pre-outs from the HT avr connected to them, move the switch on the back of the DAC1 pre to the calibrated setting, and then whatever level of signal is input by the AVR to the analogue inputs will be output at the same level on the analogue outs which would be connected to your poweramp(s). When you want to go back to two channel listening using the DAC1 pre as your preamp, move the switch on the back of the unit back to the variable setting.

 A bit of a hassle maybe having to use the switch on the back but probably more convenient than dis-connecting and re-connecting L and R interconnects from the DAC1 pre and HT avr to the poweramp(s) driving your front L and R mains.


----------



## Wavelength

Quote:


  Originally Posted by *infinitesymphony* /img/forum/go_quote.gif 
_That's not true for any receiver I've used. Preamp outputs generally are for connecting to a power amp. If they were fixed output, the system would be at full volume all the time.

 I'm going to start another thread about HT bypass, just because it's so confusing to me, and because this discussion isn't really about the DAC1. 
	

	
	
		
		

		
		
	


	




 Edit: Here's the thread: What is the purpose of the HT Bypass feature?_

 

Infinite,

 Another solution is dump the AV unit all together and use a computer for your AV functions.

 I have tons of users doing this... basically take like a MAC Mini and connect the DVI up to your Flat Panel and the Audio can go out your USB DAC. If you want 3.1, 5.1 or 7.1 you can concatinate other devices for those outputs. Then simply use Front Row and play your DVD's through your system or your music what ever you are doing.

 Thanks
 Gordon


----------



## EliasGwinn

Quote:


  Originally Posted by *CanMad* /img/forum/go_quote.gif 
_
 Basically the DAC1 pre has a HT bypass i.e. set the input on the DAC1 pre to the analogue input, which would have the L and R pre-outs from the HT avr connected to them, move the switch on the back of the DAC1 pre to the calibrated setting, and then whatever level of signal is input by the AVR to the analogue inputs will be output at the same level on the analogue outs which would be connected to your poweramp(s). When you want to go back to two channel listening using the DAC1 pre as your preamp, move the switch on the back of the unit back to the variable setting._

 

This is my understanding of it as well.

 Ideally, the AVR would have a digital HT bypass which would strip the digital front L and R channels from X.1 encoded digital inputs and send a digital 2-channel to a high-end DAC for driving the mains. ...well, one can only wish 
	

	
	
		
		

		
		
	


	




 Thanks,
 Elias


----------



## EliasGwinn

Quote:


  Originally Posted by *KarateKid* /img/forum/go_quote.gif 
_Just a question for all of you, in case anyone knowledgeable could answer it:

 Elias said that the "driver outputs" are the only spots where the oamps in the dac1 usb are lm4562, the rest are TI5532s.... what does "driver output" mean (what headphone output it applies to)?

 Thanks for all the help_

 

karateKid,

 In this case, the "output drivers" refer to the final stage that drives the XLR and RCA outputs. They do not apply to the HPA2 headphone amp, in this case. 

 Thanks,
 Elias


----------



## Kiep

I have a question regarding attenuation pads on DAC1usb. 

 Normally I use it at -20db setting with a consumer level balanced preamp.
 Recently I have tried it at 0db and my preamp handles it ok however the frequency extremes loose on definition and after few hours I detect a channel imbalance. I assume this has to do with the input level being too high.

 Has anyone tried 0db setting with the calibrated pots turned down to -10dbu? I am curious if there would be any benefit to it vs jumpered attenuators.


----------



## EliasGwinn

Quote:


  Originally Posted by *Kiep* /img/forum/go_quote.gif 
_I have a question regarding attenuation pads on DAC1usb. 

 Normally I use it at -20db setting with a consumer level balanced preamp.
 Recently I have tried it at 0db and my preamp handles it ok however the frequency extremes loose on definition and after few hours I detect a channel imbalance. I assume this has to do with the input level being too high.

 Has anyone tried 0db setting with the calibrated pots turned down to -10dbu? I am curious if there would be any benefit to it vs jumpered attenuators._

 

If you are using the calibrated setting, there will be very little difference between these -10dB attenuators and -10dB on the trim-pots. The only difference will be a decreased output impedance, which will be noticeable if you're driving long cables or devices with high capacitance or low impedance.

 If you choose to use the calibrated trim-pots, you will need a test tone to properly calibrate the two channels. Otherwise, it will be difficult to achieve accurate channel balance.

 Let me know if you have any more questions with this...

 Thanks,
 Elias


----------



## Kiep

Elias:
  Quote:


 ...If you are using the calibrated setting, there will be very little difference between these -10dB attenuators and -10dB on the trim-pots. The only difference will be a decreased output impedance, which will be noticeable if you're driving long cables or devices with high capacitance or low impedance.... 
 

Thats what I'm wondering about. I don't have long cables but the output impedance always makes a difference imo hence I want to try it.

  Quote:


 ...If you choose to use the calibrated trim-pots, you will need a test tone to properly calibrate the two channels. Otherwise, it will be difficult to achieve accurate channel balance... 
 

I can create suitable waveforms for the input and I can use Behringer DEQ2496 to measure the output signal. What frequency/level would you recommend? I will measure the current output as reference before I change anything.

 Thank you in advance.


----------



## EliasGwinn

Quote:


  Originally Posted by *Kiep* /img/forum/go_quote.gif 
_I can create suitable waveforms for the input and I can use Behringer DEQ2496 to measure the output signal. What frequency/level would you recommend? I will measure the current output as reference before I change anything.

 Thank you in advance._

 

1kHz sine wave @ -1 dBFS would be a suitable test signal.

 Thanks,
 Elias


----------



## cansman

Hi Elias,

 I read John's response in Stereophile that iTunes 7.5 is a lemon! 

 iTunes 7.6 has just been released - it is also included in the 10.5.2 massive upgrade. Would you know whether the bug has been fixed?

 Bit accuracy...sigh! 
	

	
	
		
		

		
			





 Thanks and regards,
 cansman


----------



## EliasGwinn

Quote:


  Originally Posted by *cansman* /img/forum/go_quote.gif 
_Hi Elias,

 I read John's response in Stereophile that iTunes 7.5 is a lemon! 

 iTunes 7.6 has just been released - it is also included in the 10.5.2 massive upgrade. Would you know whether the bug has been fixed?

 Bit accuracy...sigh! 
	

	
	
		
		

		
		
	


	




 Thanks and regards,
 cansman_

 

Cansman,

 Actually, since that response was published, we had some extended correspondence with the engineers at Apple, and we also conducted further testing on iTunes 7.5. It turns out that CoreAudio, NOT iTunes, was causing the problems. 

 Here is a follow-up response that I wrote for Stereophile (its also published in the "Manufacturer's Comments" section of the March '08 magazine):

Stereophile: Third Time Lucky?

 As noted in this response, iTunes 7.5 actually is capable of high-quality playback (even bit-transparency), but it requires a thorough understanding of how iTunes and CoreAudio work together.

 Let me know if anything is unclear in that article...

 Thanks,
 Elias


----------



## riverlethe

How should software EQ be used with digital output? Should bands only be reduced rather than increased, or doesn't it matter? Do you know if the graphical equalizer in Windows Media Player 11 causes distortion?


----------



## EliasGwinn

Quote:


  Originally Posted by *riverlethe* /img/forum/go_quote.gif 
_How should software EQ be used with digital output? Should bands only be reduced rather than increased, or doesn't it matter? Do you know if the graphical equalizer in Windows Media Player 11 causes distortion?_

 

Riverlethe,

 All EQ's cause distortion - intentional distortion. In other words, they distort the frequency response. 

 Since our company's philosophy is to establish the most transparent playback system, we do not recommend using EQ on playback.

 However, if you decide you want to use EQ, I would recommend using subtractive EQ'ing (that is, reducing frequencies) as much as possible, and always more then additive EQ'ing. The reason for this is because additive EQ'ing may increase the overall level of the signal to a degree which causes clipping ('digital overs'). Digital-overs are never pretty, so you'd be wise to cut at least as much as you boost.

 Thanks,
 Elias


----------



## cansman

Quote:


  Originally Posted by *EliasGwinn* /img/forum/go_quote.gif 
_Cansman,

 Actually, since that response was published, we had some extended correspondence with the engineers at Apple, and we also conducted further testing on iTunes 7.5. It turns out that CoreAudio, NOT iTunes, was causing the problems. 

 Here is a follow-up response that I wrote for Stereophile (its also published in the "Manufacturer's Comments" section of the March '08 magazine):

Stereophile: Third Time Lucky?

 As noted in this response, iTunes 7.5 actually is capable of high-quality playback (even bit-transparency), but it requires a thorough understanding of how iTunes and CoreAudio work together.

 Let me know if anything is unclear in that article...

 Thanks,
 Elias_

 

Thanks Elias for clarifying. I believe everything you said in the article is the same as previous advice for achieving bit-transparent playback with OSX and iTunes. Thanks. That's good to know!

 Kind regards,
 cansman


----------



## dspargo

Hello Elias,

 How do I connect my Benchmark DAC1 USB to my Wooaudio 6 which has RCA inputs. I want to use the balanced outputs on the Benchmark. I am already using the RCA outputs from the Benchmark and they go to the RCA inputs on my Meier Opera.

 Thanks,
 Don


----------



## poo

Quote:


  Originally Posted by *dspargo* /img/forum/go_quote.gif 
_Hello Elias,

 How do I connect my Benchmark DAC1 USB to my Wooaudio 6 which has RCA inputs. I want to use the balanced outputs on the Benchmark. I am already using the RCA outputs from the Benchmark and they go to the RCA inputs on my Meier Opera.

 Thanks,
 Don_

 

You need a cable with XLR connections on one end for the DAC1, and rca on the other. Remember to ensure that the XLR end is connected properly as per instructions in the manual - which explains that you must leave pin 3 floating.

 I am using my DAC1 USB in the same way - sounds great!


----------



## joijwall

Elias,
 First of all congratulations to the DAC1 Pre, I hope it will be a great success!

 I'm still a bit shaky about the correct use of my DAC1 USB and I would appreciate your advice. I'm on a MacBook with the recent 10.5.2 and iTunes 7.6/QuickTime 7.4.1. As backup player I use VLC 0.86d, since you pointed out VLC earlier as very good.

 1) Is there any difference for DAC1 if use the USB or the optical output from my MacBook? In the shop where I bought DAC1 they said USB is far better. 

 2) Will the sound from DAC1 change after some playing? I have the feeling that playback has become a bit "softer" compared to the beginning?

 3) iTunes is my main player. But my impression is that VLC sounds "better". That is clearer, airer, more at ease. Is there perhaps some settings I missed in iTunes?
 a) I use 24 bit and 44100 in CoreAudio, which I understand is the best setting for AppleLossless for both VLC and iTunes. I haven't seen in the wiki how VLC performs if I set it CoreAudio to 96000. Volume settings is disabled in my CoreAudio.
 b) I've found the Equalizer Window in iTunes and turned it off (preset set to "Flat" just in case)
 c) In iTunes settings for Playback I've unchecked Sound Enhancer and Sound Control.
 d) iTunes volume is set to max
 e) I've also found the picture/sound setting window in QuickTime, which has full volume, and balance, treble and bass in the middle. I can't find a way to turn it off, and I can't find the Equalizer although a setting is called "Show Equalizer".
 f) VLC has volume set to 100% (256), equalizer and others off.

 4) I've read in the wiki that iTunes uses QuickTime. Will QuickTime Player give the same bits to DAC1 as iTunes?

 5) Are Apple interested in bit transparancy, or is that of minor interest for them?

 6) When do you think your Leopard testing is ready for the wiki?

 By the way, many of my hifi friends has suddenly become interested in a different approach back home after visiting me. 
	

	
	
		
		

		
		
	


	




 Joachim


----------



## Terje

Quote:


  Originally Posted by *poo* /img/forum/go_quote.gif 
_You need a cable with XLR connections on one end for the DAC1, and rca on the other. Remember to ensure that the XLR end is connected properly as per instructions in the manual - which explains that you must leave pin 3 floating.

 I am using my DAC1 USB in the same way - sounds great!_

 

I assume you have done a comparision between rca-rca and xlr-rca?
 My thougts is that it is best to use xlr-xlr. If not possible, rca-rca is better. The same connection in both ends.


----------



## roadtonowhere08

Hi all,

 I just got a DAC1 (non-USB), and so far I love it. I have a question though. I am used to an E-MU 1212M and an ESI Juli@ driving my Face Audio F500TS balanced. I have noticed that at stock and with the volume set to calibrated, the signal is much lower than the two aforementioned sound cards, and I have to turn up the volume on foobar higher. Before I could get a good level at around -40dB on foobar, and now I have to go to around -25dB to get about the same sound level. In addition, this makes my sub much louder than the mains due to their (I believe) lowered input level into the amp. I am pretty sure that this is due to the XLR jumpers, so I am wondering if I can set them to -10 or even 0. Am I correct in thinking that setting them to 0 would be the same as passing an unaltered balanced signal to my amp with identical gain as the sound cards?

 Just in case I am wrong, the input sensitivity of my amp is .775V if that helps at all. Thanks in advance for any help


----------



## sejarzo

Quote:


  Originally Posted by *roadtonowhere08* /img/forum/go_quote.gif 
_I just got a DAC1 (non-USB), and so far I love it. I have a question though. I am used to an E-MU 1212M and an ESI Juli@ driving my Face Audio F500TS balanced. I have noticed that at stock and with the volume set to calibrated, the signal is much lower than the two aforementioned sound cards, and I have to turn up the volume on foobar higher. Before I could get a good level at around -40dB on foobar, and now I have to go to around -25dB to get about the same sound level._

 

You shouldn't be using any digital attenuation in Foobar if you want the best quality......it reduces volume by reducing resolution (hacking off bits to reduced the digital signal level.)

 Run all volume controls in any software at 100%, control volume via the pot on your DAC-1, and rebalance your sub.


----------



## roadtonowhere08

Quote:


  Originally Posted by *sejarzo* /img/forum/go_quote.gif 
_You shouldn't be using any digital attenuation in Foobar if you want the best quality......it reduces volume by reducing resolution (hacking off bits to reduced the digital signal level.)

 Run all volume controls in any software at 100%, control volume via the pot on your DAC-1, and rebalance your sub._

 

I read that foobar uses 64 bit volume attenuation, but I could be wrong. Honestly, I cannot hear any difference when I use foobar's volume control, so I want to keep using that. 

 I am just wondering if I move the jumper to 0 instead of -20dB, will I be getting a regular +4dB balanced signal, or is it a higher rated signal? If it is a regular balanced signal, I read that a higher signal running from the DAC to the amp will increase SNR. However, if someone is sure that running a jumper at 0 will clip my amp, I will not even bother opening it up. 

 It is odd though that my amp's pots are at the 11 position in a 7 to 5 range, and it is a rather beefy power amp. The pots used to be at the 9 position when using both of my balanced sound cards. I just want a confirmation that changing the jumpers will not clip the amp's inputs.


----------



## poo

Quote:


  Originally Posted by *Terje* /img/forum/go_quote.gif 
_I assume you have done a comparision between rca-rca and xlr-rca?
 My thougts is that it is best to use xlr-xlr. If not possible, rca-rca is better. The same connection in both ends._

 

Using both RCA-RCA and XLR-RCA. I don't hear any difference, nor should I - both are terminated the same way for the DAC1 - you leave pin 3 floating for the XLR-RCA cable.


----------



## sejarzo

Quote:


  Originally Posted by *roadtonowhere08* /img/forum/go_quote.gif 
_I read that foobar uses 64 bit volume attenuation, but I could be wrong. Honestly, I cannot hear any difference when I use foobar's volume control, so I want to keep using that._

 

As far as I know, regardless of how accurately Foobar calculates the attenuated output by increasing bit depth in the calculation, the reality is that it ends up sending a lower resolution/manipulated signal to the DAC1.

 Someone correct me if I'm wrong, but if you require 25 dB of attenutation, I'm fairly certain that's virtually reducing 16-bit recordings to 12-bit.


----------



## KarateKid

A newbie's question here: How do I hook this DAC1 through XLR to a hd650? I have the cables but I've read that you need to set it to "variable" mode? Does this allow the front pot to control the volume once I'm using XLR? Also I've heard that you have to open the DAC1 up to play with the db setting for optimal sound? Is this true and necessary?


----------



## Kiep

Quote:


  Originally Posted by *KarateKid* /img/forum/go_quote.gif 
_A newbie's question here: How do I hook this DAC1 through XLR to a hd650? I have the cables but I've read that you need to set it to "variable" mode? Does this allow the front pot to control the volume once I'm using XLR? Also I've heard that you have to open the DAC1 up to play with the db setting for optimal sound? Is this true and necessary?_

 

Yes, you have to set it to variable mode in order to enable volume pot adjustment of the xlr outputs. You can try it with just a normal connection to a preamp, the loudness should vary with pot setting. Once you confirm that , turn the pot to zero and plug you headphones in.
 The attenuation jumpers inside also affect the XLR volume, start with -20dB setting and see if the pot regulation is comfortable. If it is too quiet change it to -10dB or to 0 for no attenuation. Let us know if you like the sound better than from the normal headphone output. I am curious myself.


----------



## EliasGwinn

Quote:


  Originally Posted by *dspargo* /img/forum/go_quote.gif 
_Hello Elias,

 How do I connect my Benchmark DAC1 USB to my Wooaudio 6 which has RCA inputs. I want to use the balanced outputs on the Benchmark. I am already using the RCA outputs from the Benchmark and they go to the RCA inputs on my Meier Opera.

 Thanks,
 Don_

 

Hello dspargo,

 It's interesting that you ask this, as it is the topic of our next "Feedback" newsletter. 

 Poo's instructions are correct. To connect the XLR outputs to RCA inputs, you must have a cable that has a female XLR connector with pin-3 floating (not connected) and a male RCA on the other end. 

 If you buy this cable, double check to make sure pin-3 is floating...MAKE SURE IT IS NOT GROUNDED!! There is a possibility of damaging the DAC1 if you use an XLR output with pin-3 grounded for extended periods of time. 

 There are a few places you can buy these cables. We sell them (part#: 500-06902-000 and 500-06906-000 are 2' and 6' versions, respectively). But wherever you buy them, check and double-check to make sure pin-3 is floating.

 Thanks,
 Elias


----------



## EliasGwinn

Quote:


  Originally Posted by *joijwall* /img/forum/go_quote.gif 
_1) Is there any difference for DAC1 if use the USB or the optical output from my MacBook? In the shop where I bought DAC1 they said USB is far better. _

 

There are no differences between the optical output and USB output, provided that all settings are correct.

  Quote:


  Originally Posted by *joijwall* /img/forum/go_quote.gif 
_2) Will the sound from DAC1 change after some playing? I have the feeling that playback has become a bit "softer" compared to the beginning?_

 

This is an interesting observation that a lot of our customers have. The sound of the DAC1 does not change, but instead your perception changes. Many feel the DAC1 is bright when they first listen, but this is because it doesn't have all the distortion (jitter, etc) that typically 'beef-up' the mids and low-mids...things that usually cause long-term listening fatigue. Consequently, once accustomed to the DAC1's transparency, the listener deeply appreciates the clarity of the sound. The true musical nature of the recording can finally speak for itself, unimpeded. 

 That was the long answer to your question, but the "softer" perception is very common. The entire playback system (the DAC1 + you're ears) have achieved their 'break-in' period. 


  Quote:


  Originally Posted by *joijwall* /img/forum/go_quote.gif 
_3) iTunes is my main player. But my impression is that VLC sounds "better". That is clearer, airer, more at ease. Is there perhaps some settings I missed in iTunes?
 a) I use 24 bit and 44100 in CoreAudio, which I understand is the best setting for AppleLossless for both VLC and iTunes. I haven't seen in the wiki how VLC performs if I set it CoreAudio to 96000. Volume settings is disabled in my CoreAudio.
 b) I've found the Equalizer Window in iTunes and turned it off (preset set to "Flat" just in case)
 c) In iTunes settings for Playback I've unchecked Sound Enhancer and Sound Control.
 d) iTunes volume is set to max
 e) I've also found the picture/sound setting window in QuickTime, which has full volume, and balance, treble and bass in the middle. I can't find a way to turn it off, and I can't find the Equalizer although a setting is called "Show Equalizer".
 f) VLC has volume set to 100% (256), equalizer and others off._

 

It seems you have all the correct settings. With these settings, iTunes and VLC should perform equally. 



  Quote:


  Originally Posted by *joijwall* /img/forum/go_quote.gif 
_4) I've read in the wiki that iTunes uses QuickTime. Will QuickTime Player give the same bits to DAC1 as iTunes?_

 

Yes.

  Quote:


  Originally Posted by *joijwall* /img/forum/go_quote.gif 
_5) Are Apple interested in bit transparancy, or is that of minor interest for them?_

 

Yes, but it is one of many things they are interested in. Overall system performance is the main thing. Therefore, the biggest problem in CoreAudio (sample-rate conversion), may not see an upgrade for some time, because higher quality conversion will take more system resources.

  Quote:


  Originally Posted by *joijwall* /img/forum/go_quote.gif 
_6) When do you think your Leopard testing is ready for the wiki?_

 

Soon, hopefully. 

  Quote:


  Originally Posted by *joijwall* /img/forum/go_quote.gif 
_By the way, many of my hifi friends has suddenly become interested in a different approach back home after visiting me. 
	

	
	
		
		

		
			





 Joachim_

 

Great!! And as always, feel free to ask as many questions as needed...

 Thanks,
 Elias


----------



## EliasGwinn

Quote:


  Originally Posted by *roadtonowhere08* /img/forum/go_quote.gif 
_Hi all,

 I just got a DAC1 (non-USB), and so far I love it. I have a question though. I am used to an E-MU 1212M and an ESI Juli@ driving my Face Audio F500TS balanced. I have noticed that at stock and with the volume set to calibrated, the signal is much lower than the two aforementioned sound cards, and I have to turn up the volume on foobar higher. Before I could get a good level at around -40dB on foobar, and now I have to go to around -25dB to get about the same sound level. In addition, this makes my sub much louder than the mains due to their (I believe) lowered input level into the amp. I am pretty sure that this is due to the XLR jumpers, so I am wondering if I can set them to -10 or even 0. Am I correct in thinking that setting them to 0 would be the same as passing an unaltered balanced signal to my amp with identical gain as the sound cards?

 Just in case I am wrong, the input sensitivity of my amp is .775V if that helps at all. Thanks in advance for any help 
	

	
	
		
		

		
		
	


	


_

 

Hello roadtonowhere08,

 Without being familiar with these other sound cards, I would guess that you are correct in your assumption. That is, the other cards' outputs are configured for the typical +4 dBu output, which is how the DAC1 is with the attenuators set at 0 dB.

 However, if you change the attenuators from -20 to 0 dB, and also turn foobar's volume down by 20 dB, you will have the exact same signal level from the DAC1's outputs as you did before you changed those things. In other words, your adding 20 and subtracting 20 all at the same time. 

 If you were not clipping your amp before the changes, you will not be clipping after the changes...you'll be right where you started. However, you will have a slightly compromised signal-to-noise ratio because the digital signal has been attenuated so much. In other words, you will reducing the signal level, keeping the same amount of noise, and then amplifying all of it. This may or may not make a difference in your system, depending on the s-to-n ratio of the other gear etc. 

 The ideal configuration has the digital volumes as close to 0 dB (unity gain) as possible, and the attenuation is done in the analog domain. This way, the noise gets attenuated as much as the signal.

 I hope this long-winded explanation makes sense...please let me know if you'd like me to clarify any of this...

 Thanks,
 Elias


----------



## EliasGwinn

Quote:


  Originally Posted by *KarateKid* /img/forum/go_quote.gif 
_A newbie's question here: How do I hook this DAC1 through XLR to a hd650? I have the cables but I've read that you need to set it to "variable" mode? Does this allow the front pot to control the volume once I'm using XLR? Also I've heard that you have to open the DAC1 up to play with the db setting for optimal sound? Is this true and necessary?_

 

Hello KarateKid,

 First of all, we recommend not changing the attenuation jumpers unless you determine that the output level needs to be adjusted. 

 Second, you should definitely use the 'Variable' mode to control the volume. Start with the volume all the way down, and slowly increase to determine the ideal volume. The goal is to have comfortable listening levels when the volume control is between an 11 o'clock and 3 o'clock position. If comfortable listening levels are outside of this region, then the attenuation jumpers should be adjusted.

 Also, we do not recommend driving headphones directly from the the XLR outputs. You won't damage anything, but there will be significant distortion (non-linear frequency response). There are several reasons why it is advantageous to avoid this mode of operation (and we can discuss these, if you want...there are several pages within this thread based on this exact topic). At first, you may enjoy the sound of XLR-driven headphones, but you also may experience listening fatigue after listening in this mode for extended periods.

 The HPA2 headphone amp that is built into the DAC1 is ideal for driving headphones. This is the same headphone-amp used by recording and mastering engineers in some of the biggest studios in the world (not to mention the broadcast engineers at ABC, CBS, NPR, etc....).

 Thanks,
 Elias


----------



## Wavelength

Quote:


 Quote:
 Originally Posted by joijwall View Post
 1) Is there any difference for DAC1 if use the USB or the optical output from my MacBook? In the shop where I bought DAC1 they said USB is far better.
 There are no differences between the optical output and USB output, provided that all settings are correct. 
 

Joijwall, Elias;

 Actually USB would be much better than the Toslink output on the Macbook. The toslink is not really that well done unless you want to physically isolate the DAC1 from the computer.

 We did some testing with the Prism dScope III a couple of months ago with the Macbook and jitter was really high.

 As Elias has pointed out much of that will be filtered out by the DAC1 but in most cases I have found that the more there is at the beginning the more that is going to get through in the end.

 Better off to use the USB I would think in this case.

 Thanks
 Gordon


----------



## little-endian

Quote:


  Originally Posted by *Wavelength* /img/forum/go_quote.gif 
_Actually USB would be much better than the Toslink output on the Macbook. The toslink is not really that well done unless you want to physically isolate the DAC1 from the computer._

 

Assuming one has an appropriate D/A-converter, the DC-isolation is actually the big privileg of the Toslink connection - and a bit more facinating as well, of course. 
	

	
	
		
		

		
		
	


	




  Quote:


  Originally Posted by *Wavelength* /img/forum/go_quote.gif 
_We did some testing with the Prism dScope III a couple of months ago with the Macbook and jitter was really high._

 

That's quite possible. 
	

	
	
		
		

		
		
	


	




  Quote:


  Originally Posted by *Wavelength* /img/forum/go_quote.gif 
_As Elias has pointed out much of that will be filtered out by the DAC1 but in most cases I have found that the more there is at the beginning the more that is going to get through in the end._

 

From what I'm aware of, the jitter actually isn't filtered out at all but the oversampling ratio is varied instead to prevent buffer underruns or overflows (Elias may correct me here if I'm wrong since he has to know the concrete implementation).

 If the datasheets from Benchmark are correct - and I want to believe they are - then they have measured the effects of jitter by far more detailed than *anyone* is able to hear hence if they are telling the truth, there is nothing but the data which can change the sound quality.

 @Elias: Since the output of the DAC1 is muted incredibly fast when no S/PDIF-signal is present anymore, I suppose that the used buffer has to be comparatively small. I'd be interested to know the exact size of it.


----------



## roadtonowhere08

Quote:


  Originally Posted by *EliasGwinn* /img/forum/go_quote.gif 
_Hello roadtonowhere08,

 Without being familiar with these other sound cards, I would guess that you are correct in your assumption. That is, the other cards' outputs are configured for the typical +4 dBu output, which is how the DAC1 is with the attenuators set at 0 dB.

 However, if you change the attenuators from -20 to 0 dB, and also turn foobar's volume down by 20 dB, you will have the exact same signal level from the DAC1's outputs as you did before you changed those things. In other words, your adding 20 and subtracting 20 all at the same time. 

 If you were not clipping your amp before the changes, you will not be clipping after the changes...you'll be right where you started. However, you will have a slightly compromised signal-to-noise ratio because the digital signal has been attenuated so much. In other words, you will reducing the signal level, keeping the same amount of noise, and then amplifying all of it. This may or may not make a difference in your system, depending on the s-to-n ratio of the other gear etc. 

 The ideal configuration has the digital volumes as close to 0 dB (unity gain) as possible, and the attenuation is done in the analog domain. This way, the noise gets attenuated as much as the signal.

 I hope this long-winded explanation makes sense...please let me know if you'd like me to clarify any of this...

 Thanks,
 Elias_

 

Makes perfect sense. I will keep everything as is, as what you said makes sense. The digital signal is less attenuated, from what I have read the -20dB jumpers make no difference in sound, and the pots on the amp are almost at the noon position which improves tracking from what I have read. Since the run of cable from the DAC to the amp is only 2 feet, the decrease in signal coming from the DAC will have little to no EMI interference. Does this check out to you?


----------



## KarateKid

Quote:


  Originally Posted by *EliasGwinn* /img/forum/go_quote.gif 
_Hello KarateKid,

 First of all, we recommend not changing the attenuation jumpers unless you determine that the output level needs to be adjusted. 

 Second, you should definitely use the 'Variable' mode to control the volume. Start with the volume all the way down, and slowly increase to determine the ideal volume. The goal is to have comfortable listening levels when the volume control is between an 11 o'clock and 3 o'clock position. If comfortable listening levels are outside of this region, then the attenuation jumpers should be adjusted.

 Also, we do not recommend driving headphones directly from the the XLR outputs. You won't damage anything, but there will be significant distortion (non-linear frequency response). There are several reasons why it is advantageous to avoid this mode of operation (and we can discuss these, if you want...there are several pages within this thread based on this exact topic). At first, you may enjoy the sound of XLR-driven headphones, but you also may experience listening fatigue after listening in this mode for extended periods.

 The HPA2 headphone amp that is built into the DAC1 is ideal for driving headphones. This is the same headphone-amp used by recording and mastering engineers in some of the biggest studios in the world (not to mention the broadcast engineers at ABC, CBS, NPR, etc....).

 Thanks,
 Elias_

 

Yeah well I just thought I could see what the fuss regarding XLR is all about. I mean isn't XLR directly off the DAC1 USB better than the HP-2 amp (the national opamp in the XLR vs the TI in the hp2)? Shouldn't it be better or good enough to use until I can get a balanced amp?


----------



## infinitesymphony

Edit: Wrong info.


----------



## KarateKid

Quote:


  Originally Posted by *EliasGwinn* /img/forum/go_quote.gif 
_karateKid,

 In this case, the "output drivers" refer to the final stage that drives the XLR and RCA outputs. They do not apply to the HPA2 headphone amp, in this case. 

 Thanks,
 Elias_

 

infinitesymphony,

 Doesn't Elias mean that XLR output in the dac1 USB is 4562?


----------



## EliasGwinn

Quote:


  Originally Posted by *Wavelength* /img/forum/go_quote.gif 
_Actually USB would be much better than the Toslink output on the Macbook. The toslink is not really that well done unless you want to physically isolate the DAC1 from the computer.

 We did some testing with the Prism dScope III a couple of months ago with the Macbook and jitter was really high.

 As Elias has pointed out much of that will be filtered out by the DAC1 but in most cases I have found that the more there is at the beginning the more that is going to get through in the end._

 

Gordon,

 This may be the case with some DAC's, but it really will not make a difference to the DAC1 / USB / PRE. Even with >12 UI of jitter (a ridiculous, astronomical amount), the DAC1 does not see any change in performance. 

 In fact, we created an interesting test where we connected the XLR/AES input of the DAC1 via 1000 feet of CAT-5 cable wired to XLR connectors. Even with this (insane) scenario, the DAC1 suffered no performance loss whatsoever.

 Thanks,
 Elias


----------



## EliasGwinn

Quote:


  Originally Posted by *little-endian* /img/forum/go_quote.gif 
_@Elias: Since the output of the DAC1 is muted incredibly fast when no S/PDIF-signal is present anymore, I suppose that the used buffer has to be comparatively small. I'd be interested to know the exact size of it. 
	

	
	
		
		

		
		
	


	


_

 

I will get back to you on this (I want to make sure I only post correct information)....

 Thanks,
 Elias


----------



## infinitesymphony

Quote:


  Originally Posted by *KarateKid* /img/forum/go_quote.gif 
_infinitesymphony,

 Doesn't Elias mean that XLR output in the dac1 USB is 4562?_

 

I looked back through the thread, and I think you're right. Sorry about that. 
	

	
	
		
		

		
		
	


	




 But, I'm assuming that LM4562 is also used for the headphone amplifier portion, since it was mentioned that LM4562 would be superior to NE5532 for low-impedance headphones.


----------



## EliasGwinn

Quote:


  Originally Posted by *roadtonowhere08* /img/forum/go_quote.gif 
_Makes perfect sense. I will keep everything as is, as what you said makes sense. The digital signal is less attenuated, from what I have read the -20dB jumpers make no difference in sound, and the pots on the amp are almost at the noon position which improves tracking from what I have read. Since the run of cable from the DAC to the amp is only 2 feet, the decrease in signal coming from the DAC will have little to no EMI interference. Does this check out to you?_

 

This sounds like a good setup. You also have the option of utilizing the DAC1's volume control and decreasing digital attenuation, but I doubt that will improve SNR much, if at all. I'd say you're good as you are...

 Thanks,
 Elias


----------



## KarateKid

Quote:


  Originally Posted by *EliasGwinn* /img/forum/go_quote.gif 
_karateKid,

 In this case, the "output drivers" refer to the final stage that drives the XLR and RCA outputs. They do not apply to the HPA2 headphone amp, in this case. 

 Thanks,
 Elias_

 

 Quote:


  Originally Posted by *infinitesymphony* /img/forum/go_quote.gif 
_I looked back through the thread, and I think you're right. Sorry about that. 
	

	
	
		
		

		
			





 But, I'm assuming that LM4562 is also used for the headphone amplifier portion, since it was mentioned that LM4562 would be superior to NE5532 for low-impedance headphones._

 

Is the 4562 any good for the hd650s? Aren't the 650s "high impedance" headphones?


----------



## infinitesymphony

Okay, since Elias is around, he can correct me if I'm wrong again. 
	

	
	
		
		

		
		
	


	




 The DAC1 PRE uses LM4562 everywhere.
 The DAC1 USB uses LM4562 for the output drivers (the question is, which physical outputs are these?), NE5532 for everything else.
 The DAC1 regular uses NE5532 everywhere.


----------



## EliasGwinn

Quote:


  Originally Posted by *KarateKid* /img/forum/go_quote.gif 
_Yeah well I just thought I could see what the fuss regarding XLR is all about. I mean isn't XLR directly off the DAC1 USB better than the HP-2 amp (the national opamp in the XLR vs the TI in the hp2)? Shouldn't it be better or good enough to use until I can get a balanced amp?_

 

The XLR outputs on the DAC1 USB are not designed to drive headphones. The output impedance is too high (anything over 0.5 ohms is too high for driving headphones). The HPA2 headphone amp has <0.01 ohm output impedance. This is one reason it achieves ultra-low-distortion headphone amplification.

 The 4562's are great line amps for the line outputs (XLR and RCA), but they are not meant to be headphone amplifiers. They will do better then the 5532's at that job, but still not ideal. The HPA2 headphone output driver is not a 5532...it is a BUF634, which is extremely well suited for driving the awkward load that headphones present.

 There's a lot of discussion about balanced headphones earlier in this thread, but I'll give a brief description of the problems with balanced headphones...

 ALL headphones have non-linear mechanical impedances (that is, the mass and shape of a speaker will resonate more at certain frequencies and much less at other frequencies). This means the physical build of the headphones (as well as other physical impedances, like your head and ears!) will try to override the electrical system (amplifier and speaker coil). 

 To create low-distortion headphone response, one must consider 'damping factor'. A high damping factor will control the response of the speaker, thus preventing the physical impedances from dictating frequency response. Damping factor is the ratio of speaker (load) impedance to amplifier (source) impedance. In other words, the best damping factor will result from a low source impedance. Again, the source impedance from the HPA2 is less then 0.01 ohms...as low as gets!!

 Balanced headphone amps will double the source impedance of an unbalanced headphone amp. No matter how low the impedance of a balanced headphone amp, it could be half that much if it was unbalanced. This is one reason balanced headphone amps are not a good idea. (It should also be noted that the balanced output of the DAC1 / USB / PRE is 60 ohms or greater, depending on the attenuator settings).

 Not only will the source impedance double with balanced headphone amplifiers, but the total distortion and noise of the amplifier will double as well!! Every output device (opamp, transistor, tube) creates some distortion and some noise. If there are two opamps or transistors or tubes driving each headphone speaker, twice as much distortion and noise will be added!!

 The result of balanced headphones is less damping factor, more distortion, and more noise. Also, balanced headphones configurations offer no real benefits, to boot.

 Feel free to use the XLR outputs of the DAC1 / USB / PRE for balanced headphone outputs (as mentioned above, the DAC1 USB and DAC1 PRE will do better then the DAC1 at this task, because of the 4562's). It won't damage anything to operate in this configuration. But, for the reasons above, I don't recommend it.

 Thanks,
 Elias


----------



## KarateKid

Quote:


  Originally Posted by *EliasGwinn* /img/forum/go_quote.gif 
_The XLR outputs on the DAC1 USB are not designed to drive headphones. The output impedance is too high (anything over 0.5 ohms is too high for driving headphones). The HPA2 headphone amp has <0.01 ohm output impedance. This is one reason it achieves ultra-low-distortion headphone amplification.

 The 4562's are great line amps for the line outputs (XLR and RCA), but they are not meant to be headphone amplifiers. They will do better then the 5532's at that job, but still not ideal. The HPA2 headphone output driver is not a 5532...it is a BUF634, which is extremely well suited for driving the awkward load that headphones present.

 There's a lot of discussion about balanced headphones earlier in this thread, but I'll give a brief description of the problems with balanced headphones...

 ALL headphones have non-linear mechanical impedances (that is, the mass and shape of a speaker will resonate more at certain frequencies and much less at other frequencies). This means the physical build of the headphones (as well as other physical impedances, like your head and ears!) will try to override the electrical system (amplifier and speaker coil). 

 To create low-distortion headphone response, one must consider 'damping factor'. A high damping factor will control the response of the speaker, thus preventing the physical impedances from dictating frequency response. Damping factor is the ratio of speaker (load) impedance to amplifier (source) impedance. In other words, the best damping factor will result from a low source impedance. Again, the source impedance from the HPA2 is less then 0.01 ohms...as low as gets!!

 Balanced headphone amps will double the source impedance of an unbalanced headphone amp. No matter how low the impedance of a balanced headphone amp, it could be half that much if it was unbalanced. This is one reason balanced headphone amps are not a good idea. (It should also be noted that the balanced output of the DAC1 / USB / PRE is 60 ohms or greater, depending on the attenuator settings).

 Not only will the source impedance double with balanced headphone amplifiers, but the total distortion and noise of the amplifier will double as well!! Every output device (opamp, transistor, tube) creates some distortion and some noise. If there are two opamps or transistors or tubes driving each headphone speaker, twice as much distortion and noise will be added!!

 The result of balanced headphones is less damping factor, more distortion, and more noise. Also, balanced headphones configurations offer no real benefits, to boot.

 Feel free to use the XLR outputs of the DAC1 / USB / PRE for balanced headphone outputs (as mentioned above, the DAC1 USB and DAC1 PRE will do better then the DAC1 at this task, because of the 4562's). It won't damage anything to operate in this configuration. But, for the reasons above, I don't recommend it.

 Thanks,
 Elias_

 

I see, thank you very much for that very detailed explanation. 

 In this case, if a person doesn't have a dedicated amp to go with a DAC1, they might as well take the DAC1 Pre instead since it offers the better clearer less distorted 4562 opamp?


----------



## CanMad

The LM4562 is not recommended for loads less than 600 ohms.


----------



## infinitesymphony

Quote:


  Originally Posted by *KarateKid* /img/forum/go_quote.gif 
_In this case, if a person doesn't have a dedicated amp to go with a DAC1, they might as well take the DAC1 Pre instead since it offers the better clearer less distorted 4562 opamp?_

 

From Elias's message, it seems that the headphone output does not use the LM4562, it uses BUF634, which is even better for headphones than the LM4562. So, in that respect, all models of the DAC1 are equally good through the headphone output.


----------



## joijwall

Thank you very much, Elias!
 Only remains to see then, if the Leopard behaves like the Tiger... 
	

	
	
		
		

		
		
	


	




 And like the brake-in of DAC1 really is in my brain and not in the machine, my preference of VLC was perhaps broken-in by the posting you made last year. Presumptions are good reality-creators!

 I've thought about one more thing: Are there different versions of DAC1 USB regarding firmware or software? And is it possible to upgrade DAC1 by downloading and installing new firmware/software?

 Thanks! Joachim


----------



## dspargo

Quote:


  Originally Posted by *Terje* /img/forum/go_quote.gif 
_I assume you have done a comparision between rca-rca and xlr-rca?
 My thougts is that it is best to use xlr-xlr. If not possible, rca-rca is better. The same connection in both ends._

 

What do you lose by going XLR to RCA? I don't have many options besides reaching around and moving the cables when I want to switch from the Opera to the Wooaudio 6. Both amps do not have XLR input.


----------



## dspargo

Quote:


  Originally Posted by *EliasGwinn* /img/forum/go_quote.gif 
_Hello dspargo,

 It's interesting that you ask this, as it is the topic of our next "Feedback" newsletter. 

 Poo's instructions are correct. To connect the XLR outputs to RCA inputs, you must have a cable that has a female XLR connector with pin-3 floating (not connected) and a male RCA on the other end. 

 If you buy this cable, double check to make sure pin-3 is floating...MAKE SURE IT IS NOT GROUNDED!! There is a possibility of damaging the DAC1 if you use an XLR output with pin-3 grounded for extended periods of time. 

 There are a few places you can buy these cables. We sell them (part#: 500-06902-000 and 500-06906-000 are 2' and 6' versions, respectively). But wherever you buy them, check and double-check to make sure pin-3 is floating.

 Thanks,
 Elias_

 

Thanks Elias!


----------



## poo

Quote:


  Originally Posted by *dspargo* /img/forum/go_quote.gif 
_What do you lose by going XLR to RCA?_

 

Nothing.

 XLR - RCA will sound the same as RCA - RCA if cabled correctly.

 It is obviously different to an XLR - XLR connection, but that isn't what you are asking.


----------



## EliasGwinn

Quote:


  Originally Posted by *dspargo* /img/forum/go_quote.gif 
_What do you lose by going XLR to RCA? I don't have many options besides reaching around and moving the cables when I want to switch from the Opera to the Wooaudio 6. Both amps do not have XLR input._

 

XLR-to-RCA will be equivalent to RCA-to-RCA. Just remember to make sure the cable has pin-3 floating!

 Thanks,
 Elias


----------



## EliasGwinn

Quote:


  Originally Posted by *little-endian* /img/forum/go_quote.gif 
_ Quote:


  Originally Posted by *Wavelength* /img/forum/go_quote.gif 
As Elias has pointed out much of that will be filtered out by the DAC1 but in most cases I have found that the more there is at the beginning the more that is going to get through in the end.

 Better off to use the USB I would think in this case.

 Thanks
 Gordon

 

From what I'm aware of, the jitter actually isn't filtered out at all but the oversampling ratio is varied instead to prevent buffer underruns or overflows 

 If the datasheets from Benchmark are correct - and I want to believe they are - then they have measured the effects of jitter by far more detailed than *anyone* is able to hear hence if they are telling the truth, there is nothing but the data which can change the sound quality.

 @Elias: Since the output of the DAC1 is muted incredibly fast when no S/PDIF-signal is present anymore, I suppose that the used buffer has to be comparatively small. I'd be interested to know the exact size of it. 
	

	
	
		
		

		
		
	


	


_

 

L-e, Gordon, et. al., 

 With the AD1896 (the ASRC used in the UltraLock clocking system of Benchmark converters), the jitter of the incoming signal will not affect the conversion, it will merely (and inconsequentially) affect the amount of data in the input buffer between samples (as l-e correctly stated). 

 The speed of the Fin/Fout ratio updater of the ASRC will dictate the amount of jitter seen by the converter. The AD1896 has a fast start-up (higher jitter), but settles very quickly (~500ms) to a slower mode (low jitter). In the slower mode, the jitter is attenuated VERY effectively. Specifically, it begins rolling off at 1 Hz, and is lower then -100 dB at 1kHz. (All of this can be seen in the performance charts of the AD1896 ASRC: http://www.analog.com/UploadedFiles/...ets/AD1896.pdf)

 The bottom line is that the effects of jitter is a constant, known quantity that is completely independent of the amount of incoming jitter. In other words, the incoming clock could have 0.05 UI of 15 kHz jitter or 12 UI of broadband jitter, and the amount of distortion will not change AT ALL!!! Pretty amazing... This is an important performance characteristic because it eliminates the differences between cable quality, cable lengths, environmental EMI, AC-induced signal noise, transmission methods (optical vs. electrical vs. USB), etc, etc, etc... 

 It should also be noted that the amount of jitter from the local on-board clock also affects the amount of jitter-induced distortion during conversion. This is why intelligent clock circuitry and board-layout is extremely important for low-distortion performance. All-in-all, for all levels of incoming jitter, the amplitude of jitter-induced artifacts of the DAC1 is ~-133 dB...well below the threshold of hearing. (btw, this performance spec has been verified by many reviewers...see John Atkinson's tests in Stereophile, for example).

 L-e, the RAM in the AD1896 is 512 words per channel.

 Thanks,
 Elias


----------



## Wavelength

Quote:


  Originally Posted by *poo* /img/forum/go_quote.gif 
_Nothing.

 XLR - RCA will sound the same as RCA - RCA if cabled correctly.

 It is obviously different to an XLR - XLR connection, but that isn't what you are asking._

 

Gang,

 One thing you can do to get the benefits of balanced connection into an RCA is use the Jensen Transformer series ISO-MAX.

WELCOME TO JENSEN TRANSFORMERS, INC.

 I have used these on a number of large setups where you want one connection but have another.

 In this case you could use an XLR cable to the ISO-MAX and then from there RCA into your pre/integrated setup.

 Customers who want to use my USB dacs but want the computer as far away from their system have gone to this approach or in the case of the Cosecant (transformer output) is lift the ground.

 Thanks
 Gordon


----------



## Wavelength

Elias,

  Quote:


 The bottom line is that the effects of jitter is a constant, known quantity that is completely independent of the amount of incoming jitter. In other words, the incoming clock could have 0.05 UI of 15 kHz jitter or 12 UI of broadband jitter, and the amount of distortion will not change AT ALL!!! Pretty amazing... This is an important performance characteristic because it eliminates the differences between cable quality, cable lengths, environmental EMI, AC-induced signal noise, transmission methods (optical vs. electrical vs. USB), etc, etc, etc... 
 

Several people as well as myself have looked at the AD1896 and found that actually it acts more like a filter to jitter than a brick wall.

 In your case the intrinsic jitter of the TAS1020 is really pretty low and all that get's filtered out.

 I am not familar with the AKM receiver that is in the Benchmark.

 In out tests we used the Cirrus/Crystal receiver the CS8416. We would send it X amount and test the dac output for sidebands (PCM1794A). We found the output jitter from the AD1896 to filter in a pretty linear fashion. In all of our tests we used a 0.5ps jitter OSC at Fs*512 (incoming Fs so it's a direct multiple of 2, 4 upsampling) as the OSC into the AS1896. We also found that running very hs oscillators into the AD1896 to get 192K upsampling caused a pretty significant amount of intrinsic jitter. But again external gates are required for these speeds and the complexity of the design increase 5 fold. 

 ~~~~~~~

 But guys don't dwell on this, in 99% of the cases the Benchmark will remove the crap you guys are feeding it. You better off asking about system integration of the product, cable types, length, headphone questions and such.

 Thanks
 Gordon


----------



## roadtonowhere08

Some great information here. I have another question regarding pre-emphasis flags from older CDs. I use my computer for playback, and I noticed in the DAC1 literature that it applies de-emphasis if the marker was set. I have a set of Black Sabbath CDs from 1986 that have pre-emphasis, and I want to play them without having to use a program to correct the change in frequency. If they are .flac images with .cue sheets with the pre-emphasis flag set, and I play the .cue sheets from foobar, would the DAC1 know to apply de-emphasis? Thanks in advance for any help.


----------



## EliasGwinn

Quote:


  Originally Posted by *roadtonowhere08* /img/forum/go_quote.gif 
_Some great information here. I have another question regarding pre-emphasis flags from older CDs. I use my computer for playback, and I noticed in the DAC1 literature that it applies de-emphasis if the marker was set. I have a set of Black Sabbath CDs from 1986 that have pre-emphasis, and I want to play them without having to use a program to correct the change in frequency. If they are .flac images with .cue sheets with the pre-emphasis flag set, and I play the .cue sheets from foobar, would the DAC1 know to apply de-emphasis? Thanks in advance for any help. 
	

	
	
		
		

		
		
	


	


_

 

This is an excellent question. I'll have to test for this. It would depend on the ripping software. The software would need to recognize pre-emphasis and encode that within the subsequently created audio file. I don't know if they take that into account or not. I'll try to find out...

 Thanks,
 Elias


----------



## doctorcilantro

Ellias,

 Quick question: Are the fuses in the DAC1 in the signal path?

 thanks
 DC


----------



## music_man

this may come very late in the game but i wanted to mention it.

 some time ago some folks mentioned that the front pot on the dac1 "sounded" better than the rear pots. this is somewhat true. however there is a big problem with using the front pot. the levels or voltage difference between the channels can vary as much as 1 volt. especially at the lower gain settings that are normally used. it is not untill the pot is at 1 o'clock that the differences become negligeble. even then i find that with a test tone the pot fluctuates under load. the rear pots are much more stable. the rear pots can be calibrated to the exact same voltage and stay that way for a great length of time. i think that is something to consider. i do not hear vastly different levels between the channels as being "better" for most purposes.

 i would suggest that if you have the dac1 classic to put it on the 0db pads then set the rear pots as high as your preamp can tolerate before saturation.
 with the 0db pads in place the impedance is more similar to the usb version. the impedance on the padded outputs was an issue for me on the classic model. this mostly overcomes that i think. my pots are set at 3.43 volts and they could go higher but that sounds best. it took me many hours to arrive at that setting.

 music_man


----------



## EliasGwinn

Quote:


  Originally Posted by *doctorcilantro* /img/forum/go_quote.gif 
_Ellias,

 Quick question: Are the fuses in the DAC1 in the signal path?

 thanks
 DC_

 

DC,

 No, the fuses are not in the audio signal path. 

 Thanks,
 Elias


----------



## doctorcilantro

Music_Man,

 Interesting points. I use the DAC1 XLR outs (3rd pin floated) on Calibrated to connect to a Sunfire Vacuum Tube Preamp. 

 Since I use Replay Gain w/ J. River Media Center, all volumes get reduced (not changed via compression/normalization) to a traget -89db IIRC. This Calibrated output combined with the 0db jumper setting is a great option for me to compensate for this often excessive reduction in volume which is needed when attempting to playback 40000K+ files at similar volume levels. 

 Anyway, I kind of set this all up by ear comparing output levels from my 1212M, and matching the L/F output singal with a Wavelab generated test tone as I watched the VU meters on my Tascam MKIII. I did not pull out my multi-meter as I have limited knowledge as to how I could tell what level of voltage would oversaturate the Sunfire preamp. I have been told this tube preamp can take a lot of abuse in terms of a hot input signal. I think Sunfire told me that when I asked about running +4 out of the EMU1212M into it.

 If you have some insight you could offer an newb like me, I'd be grateful. Things sound good, albeit, I was recently thinking that the volume was a tad loud for the Sunfire to be only at 9 o'clock.

 DC


----------



## EliasGwinn

Quote:


  Originally Posted by *doctorcilantro* /img/forum/go_quote.gif 
_If you have some insight you could offer an newb like me, I'd be grateful. Things sound good, albeit, I was recently thinking that the volume was a tad loud for the Sunfire to be only at 9 o'clock.

 DC_

 

The fixed (calibrated) output levels can be set by, first, playing music through the DAC1 and changing one of the calibration trim pots to a comfortable, appropriate volume. Then, play a constant level digital tone though the DAC1 and adjust the other output to match the voltage of the appropriately set output. 

 Let me know if you have any more questions.

 Thanks,
 Elias


----------



## doctorcilantro

Thanks Elias. I guess I have done what you suggest, except I did not use a multi-meter; I used the VU meter. 

 The XLR go to the pre-amp and the RCA outs go to the VU meters on the Tascam.

 I guess I'll check it out again tonight.

 thanks
 DC


----------



## music_man

only a volt-meter will tell the true story here. i used vu meters as well when i began. once i pulled out the volt meter i found that the vu meters were way off. if you really want to be sure they are the same you need a true rms calibrated and certified multimeter. that is an expensive tool though. still, a $30 voltmeter is going to be better than almost any vu meters i think.

 mr. gwinn told how to set it up with a volt meter much earlier in this thread.

 you really need to know what your amp/preamp can take input wise. you don't want distortion present.

 music_man


----------



## doctorcilantro

Yep, mine were quite a bit off L 6.01 / R 6.23.

 As I said earlier since I'm getting quite a bit of volume reduction via software (Replay Gain). I set both to 4v and I'm going to test output listening levels through speakers. 

 I have to call Sunfire on Monday and ask how much they advise as max input voltage.

 Is the pro +4db setting equivalent to this: 4db(v) equates to a voltage ratio of 1.6:1 (actually 1.58:1)? 

 So when I use 4v it's a 4 x 1.58db increase? This would sort of make sense in that my average reduction, that I'm compensating for, is about -12db and I was using +10db (now about +6db) if the above calculation applies in this manner.

 OR....is 4dbv = 14.26dbu?

dB dBu dBV dBm to volts conversion calculator volt - volts to dBu and dBV dB - convert dB volt calculation online attenuation loss gain ratio reference - sengpielaudio

 I need to do a lot more reading, but that's why I ask questions.

 DC


----------



## Terje

Elias or others.

 Is it right to say: When a analog wave, from for example a microphone is discovered by the ADC(analog to digital converter) the wave is first sampled/discovered as a 1bit audio. Then up sampled to for example 24bit?

 Asking this because the DSD format is 1bit/2.8224MHz. 
 The word length is 1bit? 
 Do sound engineer’s uppsample to 24bit primary to be able to use more tracks, equalisation etc when recording?


----------



## Crowbar

One can trade quantization resolution for sampling rate and the reverse, but in practice the trade is not perfect (and that's one of the reasons that DSD sucks).


----------



## Kiep

Quote:


  Originally Posted by *EliasGwinn* /img/forum/go_quote.gif 
_If you are using the calibrated setting, there will be very little difference between these -10dB attenuators and -10dB on the trim-pots. ...

 Thanks,
 Elias_

 

I have tried setting the attenuators to 0dB and turning down the trim pots at the back to about 3V output. I have to say that indeed the sound is different, in my case I perceive it as more coherent, "rounder" tonally and musical (yes, the M word). However YMMV, as usual. 

 I will try 3.43V as suggested here and see if this sounds better.


----------



## EliasGwinn

Quote:


  Originally Posted by *doctorcilantro* /img/forum/go_quote.gif 
_Is the pro +4db setting equivalent to this: 4db(v) equates to a voltage ratio of 1.6:1 (actually 1.58:1)? 

 OR....is 4dbv = 14.26dbu?_

 

Doctor Cilantro,

 +4 dBu = 1.23 Vrms.

 Be careful to distinguish between dBu, dBv and dBV. dBv (not commonly used) is actually the same as dBu (dBu is the current, standard term). dBV is a seperate measurement, and it is always 2.2 dB lower then the equivalent dBu. dBFS is the reference for digital levels, and its maximum value is 0 dBFS.

 +4dBu is the professional standard that you are referring to. It is a calibration standard that sets the 'average' level of a 'full scale signal' at +4 dBu. This is determined by assuming the average level is 20 dB below the maximum value. Therefore, +4 dBu assume +24 dBu maximum peaks. 

 One important note about dB's. When comparing different dB scales, they will all add or subtract linearly. That is, assume we have a 10 dBu signal, which is equivalent 7.8 dBV. If we reduce the signal by 2 dB, the signal is now 8 dBu AND ALSO 5.8 dBV. As you can see, each dB scale is equally reduced (by 2 dB, in this example). This is also true for dBFS. If you reduce a signal by 2 dBFS, its analog equivalents will also be reduced by 2 dB (assuming everything else remains consistent). 

 For digital to analog converters, the +4 dBu standard is set by calibrating the output to +4 dBu when using a -20 dBFS signal, or calibrating the output to +24 dBu with a 0 dBFS signal. This calibration dictates that the maximum peak output of the converter will be +24 dBu. 

 Hope that helps...

 Thanks,
 Elias


----------



## EliasGwinn

Quote:


  Originally Posted by *Terje* /img/forum/go_quote.gif 
_Elias or others.

 Is it right to say: When a analog wave, from for example a microphone is discovered by the ADC(analog to digital converter) the wave is first sampled/discovered as a 1bit audio. Then up sampled to for example 24bit?

 Asking this because the DSD format is 1bit/2.8224MHz. 
 The word length is 1bit? 
 Do sound engineer’s uppsample to 24bit primary to be able to use more tracks, equalisation etc when recording?_

 

Terje,

 The 24-bit format (PCM, or Pulse Code Modulation) is a completely different format from the 1-bit format (DSD, or Direct Stream Digital). The two formats are not compatible. In other words, they require completely different ADC's and DAC's. 

 24-bit ADC's that use PCM encode the audio with 24-bits immediately. In other words, the audio is not increased to 24-bits after the fact. Likewise, 1-bit DSD converters encode the audio with 1-bit immediately, and do not increase bits after the fact.

 The word lengths of these formats are fundamental to their nature.

 Thanks,
 Elias


----------



## Terje

Elias,

 Thank you for clarify that. 
 I was confused


----------



## doctorcilantro

Thanks Elias. I have more reading to do but your post was very informative. 

Replay Gain - Calibration

 RG is utilized by J. River, and I have it set to playback files at this target 83db so I'm trying to compensate for that reduction in gain with the trim pots:
  Quote:


 Finding a standard

 Having calculated a representative RMS energy value for the audio file, we now need to reference this to a real world sound pressure level. The audio industry doesn't have any standard for listening level, but the movie industry has worked to an 83dB standard for years.

 What the standard actually states is that a single channel pink noise signal, with an RMS energy level of -20 dB relative to a full scale sinusoid should be reproduced at 83 dB SPL (measured using a C-weighted, slow averaging SPL meter). In simple terms, this means that everyone can set their volume control to the same (known, calibrated) gain.

 ASIDE: This number (83dB SPL) wasn't picked at random. It represents a comfortable average listening level, determined by professionals from years of listening. That reference level of -20dB pink noise isn't random either. It causes the calibrated average level to be 20dB less than the peak level. In other words, it leaves 20dB of headroom for louder than average signals. So, if CDs were mastered this way, the average level would be around -20dB FS, leaving lots of room for the dramatic peaks which make music exciting.


----------



## slowth

wow.. that's helpful for me too! thanks!


----------



## Tarkovsky

I just read through this entire thread and it's been very informing. Particularly the discussion on the problems with 192khz audio. I was looking at a number of DAC upgrades, but I think the benchmark might well be the one for me - jitter rejection is useful for someone who uses cheapo toslinks out of a computer and I really like to see manufacturers listen to what the end consumer thinks of their product. 
 Unfortunately for the moment at least my student loan says otherwise... Until I muster the pennies from busking (or find one on ebay) I'm trying to make the best of what I've got - I'm currently running the monitor outs from an mbox 2 into my headphone amp (a GS green solo), is this 'bad' - as in will this mismatch impedances? If so will I be getting a response shift or worse damaging anything? I'm running the output on the mbox cranked, in theory for a low S/N ratio (I figure the pot may just be an attenuator), but the mbox draws power off usb (eek). Fortunately nothing else does on my imac, and its usb power seems to be comparatively sturdy. Have I got this set up 'best'?


----------



## schugh

Without getting overly complicated, all of a sudden on my Vista desktop computer I am getting the error "USB Device not recognized".
 The DAC has been working fine for the last few months. I was using kernel streaming with foobar2000. Over the weekend I changed my heatsink and fan.
 Plugged everything back in. When I went to listen today, I was getting dropouts etc. I though maybe the DAC1 needs to be in a different USB port where it was working flawlessly before. I tried a different port and I got the error.
 I put it back into the same port and same error.

 I even plugged the DAC1 on my XP laptop and again same error.

 I have no idea why this might be the case and find myself suddenly frustrated.

 If anyone else has experience this problem or knows what might be going on, please let me know. 

 Also, I should add, that I have the headroom Desktop Portable with USB DAC functionality and it is still working fine on both the desktop and my notebook.
 When I connect it I get the "USB Audio Device"

 When the DAC1 was working, I would see something like "installed benchmark 1.0" or something after I plugged in the usb cable. Now all of a sudden it is always "USB device not recognized".

 Thanks

 -- Sanjay


----------



## EliasGwinn

Quote:


  Originally Posted by *schugh* /img/forum/go_quote.gif 
_Without getting overly complicated, all of a sudden on my Vista desktop computer I am getting the error "USB Device not recognized".
 The DAC has been working fine for the last few months. I was using kernel streaming with foobar2000. Over the weekend I changed my heatsink and fan.
 Plugged everything back in. When I went to listen today, I was getting dropouts etc. I though maybe the DAC1 needs to be in a different USB port where it was working flawlessly before. I tried a different port and I got the error.
 I put it back into the same port and same error.

 I even plugged the DAC1 on my XP laptop and again same error.

 I have no idea why this might be the case and find myself suddenly frustrated.

 If anyone else has experience this problem or knows what might be going on, please let me know. 

 Also, I should add, that I have the headroom Desktop Portable with USB DAC functionality and it is still working fine on both the desktop and my notebook.
 When I connect it I get the "USB Audio Device"

 When the DAC1 was working, I would see something like "installed benchmark 1.0" or something after I plugged in the usb cable. Now all of a sudden it is always "USB device not recognized".

 Thanks

 -- Sanjay_

 

Sanjay,

 This is a very interesting phenomenon. It sounds like the DAC1's USB output chip may have gotten a power surge from the USB line and fried it. 

 I'll PM you and we can discuss this further...

 Thanks,
 Elias


----------



## HeadphoneAddict

Great. Surge protectors for AC lines, then were had to start protecting telephone lines before entering the computer, now USB?


----------



## doctorcilantro

Never used the DAC1 USB but I'm guessing it has its own power supply and like any other piece of expensive a/v equipment should be protected; not scolding but trying to point out the difference between USB and AC. Did you possibly leave the power supply switch on (on your PC) and when you closed up your PC after working on it, inserted the power cord, and sent a surge into through the open switch? even though the DAC1 may have been on at the time, maybe the surge traveled down the usb cable. I fail to see how you could protect against other than protecting the PC itself or associated eqp. from surges. 



 DC


----------



## EliasGwinn

For the record, I don't know if there was a power surge on the USB port. It was more of a thought-out-loud (...which has a sense of permanence when written on the internet 
	

	
	
		
		

		
		
	


	




).

 I have not heard of power surges damaging equipment on the USB. I don't know if it is actually the case. 

 I have arranged a warranty repair with the poster, and we will find out what the issue is.

 Thanks,
 Elias


----------



## Tarkovsky

What's the opinion on using the DAC 1 running straight into a power amp (quad 306) - would I be better off with the higher current of the DAC1 USB? I'd be using the RCAs/unbalanced.


----------



## schugh

Quote:


  Originally Posted by *doctorcilantro* /img/forum/go_quote.gif 
_Never used the DAC1 USB but I'm guessing it has its own power supply and like any other piece of expensive a/v equipment should be protected; not scolding but trying to point out the difference between USB and AC. Did you possibly leave the power supply switch on (on your PC) and when you closed up your PC after working on it, inserted the power cord, and sent a surge into through the open switch? even though the DAC1 may have been on at the time, maybe the surge traveled down the usb cable. I fail to see how you could protect against other than protecting the PC itself or associated eqp. from surges. 
 DC_

 

When I changed the heat-sink and fan on my pc, everything was unplugged and disconnected off course and I took the pc to the kitchen table where I made the change.

 Afterwards, I put it back in place under my desk, plugged everything in and powered it back on (no different from as if I had shut down the computer and powered it on). This was last Friday. Last night I went to listen to some music using headphones via the DAC and it was playing fine except for some dropouts type of thing. I just tried to change the port the DAC was plugged into and that's when the problem started.

 My PC and a couple of devices are plugged into a UPS and the DAC1 and other items are plugged into an APC surge suppressor.

 I've tried 3 different usb cables, 3 computers, 2 AC outlets and always the same problem.

 I've also ruled out a software issue as my other two USB DACs: Headroom desktop portable and Stello DA220 MKII are recognized and work fine.

 -- Sanjay


----------



## Crowbar

Surge protectors on USB? Why not.

 My old computer died when the PSU shorted between the high and low voltage sides. EVERY card inside, along with the motherboard, and all drives, got fried.


----------



## Jacobie423

Hey,

 I am new to these boards, and new to high end audio in general. Anyway, hopefully someone can help me out.

 I am looking to buy an external DAC, and have been looking at a Benchmark DAC1 non-USB model. I am also interested in a Stello DA100 usb dac. 

 Now, the only reason I would favor the Stello over the DAC1 is because of the USB input, but I am not sure which would deliver better quality: SPDIF or USB?

 I would be running SPDIF out from an M-Audio Audiophile 192 internal sound card. Would this deliver better quality than connecting a Stello DA100 via usb?

 Thanks, in advance, for the advice.


----------



## ted betley

I have recently bought a Benchmark usb dac and have been listening to it with my WMP11 for a week and I am overwhelmed at it's sonic excellence. Ripped cd's playback from either hard drive or flash drive as wav files are very very good. I am a satisfied customer, thank you Benchmark. However I am not a pc/IT type person. I get by but just barely. I am an audiophile. 

 When I tried to listen to some 24/96 files off of Linn website or other sources I get an error message from my pc indicating I don't have 24/96 capability. Question what's the easiest way to get/download 24/96 codecs that are compatible with WMP11? I really don't want foobar or other sophisticated programs because I love the user friendliness of WMP11 and it's Library functionality.


----------



## Tarkovsky

Quote:


  Originally Posted by *Jacobie423* /img/forum/go_quote.gif 
_Hey,

 I am new to these boards, and new to high end audio in general. Anyway, hopefully someone can help me out.

 I am looking to buy an external DAC, and have been looking at a Benchmark DAC1 non-USB model. I am also interested in a Stello DA100 usb dac. 

 Now, the only reason I would favor the Stello over the DAC1 is because of the USB input, but I am not sure which would deliver better quality: SPDIF or USB?

 I would be running SPDIF out from an M-Audio Audiophile 192 internal sound card. Would this deliver better quality than connecting a Stello DA100 via usb?

 Thanks, in advance, for the advice._

 

The benchmark dac1 is not affected by the common problems with spdif as it totally resists jitter with some clever proprietary technology. For that reason you can use any kind of connection you like and have no problems. Probably the number one reason I'm thinking of skimping on food for a little while to save up for one...


----------



## EliasGwinn

Quote:


  Originally Posted by *Tarkovsky* /img/forum/go_quote.gif 
_... Probably the number one reason I'm thinking of skimping on food for a little while to save up for one..._

 







 We reserve the right to deny DAC1's to customers who starve themselves to buy them.


----------



## EliasGwinn

Quote:


  Originally Posted by *Tarkovsky* /img/forum/go_quote.gif 
_What's the opinion on using the DAC 1 running straight into a power amp (quad 306) - would I be better off with the higher current of the DAC1 USB? I'd be using the RCAs/unbalanced._

 

Its hard to answer this without having the amp here to test. The output drivers of the DAC1 USB are much more 'invincible'...in other words, they can handle almost any load. The DAC1 can handle most loads, but may experience distortion and/or high frequency roll-off if driving a low-impedance or high-capacitance load (or long cables).

 Thanks,
 Elias


----------



## EliasGwinn

Quote:


  Originally Posted by *Crowbar* /img/forum/go_quote.gif 
_Surge protectors on USB? Why not.

 My old computer died when the PSU shorted between the high and low voltage sides. EVERY card inside, along with the motherboard, and all drives, got fried._

 

Exactly. I have also seen examples of equipment being damaged because of power surges/shorts/etc within a computer. 

 The dangerous thing about a USB port is that it is also designed to be a power supply in some cases. If a high voltage spike was pushed out the USB port, I would not be surprised if it would fry the USB chip of the DAC1 USB. We haven't seen this happen yet, but I wouldn't be shocked.

 Thanks,
 Elias
 Thanks,
 Elias


----------



## EliasGwinn

Quote:


  Originally Posted by *ted betley* /img/forum/go_quote.gif 
_I have recently bought a Benchmark usb dac and have been listening to it with my WMP11 for a week and I am overwhelmed at it's sonic excellence. Ripped cd's playback from either hard drive or flash drive as wav files are very very good. I am a satisfied customer, thank you Benchmark. However I am not a pc/IT type person. I get by but just barely. I am an audiophile. 

 When I tried to listen to some 24/96 files off of Linn website or other sources I get an error message from my pc indicating I don't have 24/96 capability. Question what's the easiest way to get/download 24/96 codecs that are compatible with WMP11? I really don't want foobar or other sophisticated programs because I love the user friendliness of WMP11 and it's Library functionality._

 

There is one solution that we have found, but its not exactly 'elegant'. Here is the link:

Combined Community Codec Pack

 If you want to try it, I can offer some help setting it up. But, I can't guarantee anything about its performance, reliability, etc. 

 Thanks,
 Elias


----------



## Tarkovsky

Are you suggesting I send you one? Also can I order the dac 1 from your site if I live in the uk? Even with VAT and import tax there's a £60 markup. That could keep me in pot noodle for a month or so
	

	
	
		
		

		
		
	


	




.


----------



## Tarkovsky

Hi ted. Even though it's not so user friendly I think you should move to VLC. It's free and open source and handles all files, formats etc. Also there really is no need to use wavs when there are lossless formats available. These produce exactly the same bitstream and have no quality loss whatsoever. I'd also recommend ripping using a really good ripping program like EAC (which is also free) as this will really make a difference and iron out some of the problems of using a computer cd drive for audio, in fact newer cd players are using some of same technology (along with high quality drives) because it works so well.
 Hope that didn't baffle you. If you have any questions on computer music/audio feel free to pm me. I doubt I have the technical knowledge of elias but I know a thing or two about using a computer as source.


----------



## EliasGwinn

Quote:


  Originally Posted by *Tarkovsky* /img/forum/go_quote.gif 
_Are you suggesting I send you one? Also can I order the dac 1 from your site if I live in the uk? Even with VAT and import tax there's a £60 markup. That could keep me in pot noodle for a month or so
	

	
	
		
		

		
		
	


	




._

 

Ahh...you're in the UK. You will have to order from a UK dealer.

 Thanks,
 Elias


----------



## Wavelength

Quote:


  Originally Posted by *EliasGwinn* /img/forum/go_quote.gif 
_Exactly. I have also seen examples of equipment being damaged because of power surges/shorts/etc within a computer. 

 The dangerous thing about a USB port is that it is also designed to be a power supply in some cases. If a high voltage spike was pushed out the USB port, I would not be surprised if it would fry the USB chip of the DAC1 USB. We haven't seen this happen yet, but I wouldn't be shocked.

 Thanks,
 Elias
 Thanks,
 Elias_

 

Elias,

 Unlike other USB controllers the TAS1020B does not require VBUS to operate. I did not check your PCB out but in most cases VBUS is not even connected. Only the VGND (Ground reference) is connected.

 The TAS1020 operates under the assumption of getting SOF packets when active and waiting for init's after the computer has been turned off or put to sleep/stanby to determine if the link is active or not.

 Therefore the only way to fry the TAS1020 would if the computer's USB controller (in the PC) arced over internally and took out D+/D- which would fry the input resistors to the DAC on the USB BUS.

 But really the possibility of that happening would mean the PC had gone first.

 There's always the possibility that something happened in unison with the initial incident.

 Thanks
 Gordon


----------



## sejarzo

Quote:


  Originally Posted by *EliasGwinn* /img/forum/go_quote.gif 
_There is one solution that we have found, but its not exactly 'elegant'. Here is the link:

Combined Community Codec Pack

 If you want to try it, I can offer some help setting it up. But, I can't guarantee anything about its performance, reliability, etc. 

 Thanks,
 Elias_

 

Elias, I've seen this recommended before--and I'm curious to see if there is some trick in the CCCP setup, because for me it's been a total bust!

 I could easily play any 24/96 .wav or lossless WMA file I had in WMP10, but when I "upgraded" to WMP11, 24/96 .wav capability was thoroughly broken. I could play a couple of 24/96 lossless WMA samples, but that's it. I got an error message that suggested a codec file was corrupt, so I uninstalled CCCP, rolled back to WMP10.....everything worked fine......"re-upgraded" to WMP11, and it didn't work. I have WMP11 set to auto download codecs, and I saw a message in the lower left that says the codec was downloaded, but also an error dialog box and the file didn't play.

 I then re-installed CCCP, and it still doesn't work correctly. WMP11 doesn't report any codec problems, but 24/96 .wav's from iTrax simply start off with a loud "snap" and then low static, while some other 24/96 samples are nothing but white noise!

 What I find very odd about all this is that I can find no info for WMP11 on the MS site that addresses 24/96 capability.....and several users who have the same issues I do have posted questions on the MS help forums, but received no help at all from the MVP's.

 I can play my 24/96 files in other players (Foobar, Media Player Classic, et al) just fine......only WMP11 has issues.


----------



## doctorcilantro

VLC doesn't support lossless .ape files (which support 24/96 iirc)

Media Jukebox: Free J. River Media Jukebox software

 DC


----------



## EliasGwinn

Quote:


  Originally Posted by *sejarzo* /img/forum/go_quote.gif 
_Elias, I've seen this recommended before--and I'm curious to see if there is some trick in the CCCP setup, because for me it's been a total bust!

 I could easily play any 24/96 .wav or lossless WMA file I had in WMP10, but when I "upgraded" to WMP11, 24/96 .wav capability was thoroughly broken. I could play a couple of 24/96 lossless WMA samples, but that's it. I got an error message that suggested a codec file was corrupt, so I uninstalled CCCP, rolled back to WMP10.....everything worked fine......"re-upgraded" to WMP11, and it didn't work. I have WMP11 set to auto download codecs, and I saw a message in the lower left that says the codec was downloaded, but also an error dialog box and the file didn't play.

 I then re-installed CCCP, and it still doesn't work correctly. WMP11 doesn't report any codec problems, but 24/96 .wav's from iTrax simply start off with a loud "snap" and then low static, while some other 24/96 samples are nothing but white noise!

 What I find very odd about all this is that I can find no info for WMP11 on the MS site that addresses 24/96 capability.....and several users who have the same issues I do have posted questions on the MS help forums, but received no help at all from the MVP's.

 I can play my 24/96 files in other players (Foobar, Media Player Classic, et al) just fine......only WMP11 has issues._

 

Sejarzo,

 I completely empathize with you. We've heard mixed results of 24-bit playback with WMP and/or CCCP. I, personally, have never seen any version of WMP play 24-bit files without CCCP. However, the last time I downloaded and installed CCCP it still did not enable 24-bit playback! It may be that CCCP only works with certain versions of WMP...I can't say for sure.

 Needless to say, I've become incredibly frustrated with WMP for this reason and other reasons, including 16-bit volume control.

 Good luck.

 Thanks,
 Elias


----------



## sejarzo

Quote:


  Originally Posted by *EliasGwinn* /img/forum/go_quote.gif 
_ .... I completely empathize with you. We've heard mixed results of 24-bit playback with WMP and/or CCCP. I, personally, have never seen any version of WMP play 24-bit files without CCCP. However, the last time I downloaded and installed CCCP it still did not enable 24-bit playback! It may be that CCCP only works with certain versions of WMP...I can't say for sure._

 

Thank goodness, I was starting to think I was a total doofus for not being able to figure things out and make it work.

 What I find particularly aggravating is that there are no error messages shown, but playback consists of nothing but the loud snaps or white noise.....I would sort of prefer that WMP11 simply displayed a meaningful error message, you know??? 

 I'm still a diehard Foobar 0.8.3 user myself, and I suppose I will continue to be one for quite a while.


----------



## ted betley

Elias I tried using foobar (in lieu of wmp11) with my Benchmark usb dac and contrary to popular opinion I could not get foobar to sound as good as wmp11. Part of the reason may be that I couldn't get asio to work with foobar. I use foobar 0.9.5.1 and asio 1.2.6. Asio loads into my foobar component file but does not show in my preferences. Should I continue to try to get asio to work or is there some other way to optimize foobar to exceed sq of wmp11?


----------



## doctorcilantro

Why would one need CCCP to playback wma at 24bit? Directshow, yes, CCCP is great and now incorporates a newer (seemingly stable) version of FFDshow to boot. Microsoft never ceases to amaze with the sub-par offerings they shove via their OS.

 I recommended J. River Media Center to Benchmark a year ago or so when on the phone; you guys still need to check out it. Foobar or J. River Media Center is all one needs for high quality playback of multiple filetypes upon install. 

 Besides impeccable sound quality, a high quality asio interface, multiple zone output, advanced sorting & tagging functions, support for VERY LARGE libraries, iPod support, 32bit internal processing, native flac & ape support, a Library server (listen to your library over WAN), ...well I could go on and on. Mc is foobar on steroids; for me there is nothing that comes close to this app. So if your banging your head against the wall with WMP, take a chance and check out a trial. I think J. river even has an audio-only version named Media Jukebox which is 100% free and has many of the aformentioned functions.

 DC


----------



## Terje

Quote:


 Mc is foobar on steroids 
 

What is Mc ?


----------



## Wavelength

Gang,

 I recomend J River also. This is by far one of the best all around software packages for the PC. You can rip in secure mode (their version of error correcting). It can be used really well in Vista as well as XP. Tons of options and plugin's for all kinds of stuff.

 Many of you are still using XP and it works really well. But did you know that in Vista the USB and Audio routines were really overhauled and too me work better and have less issues. Not to mention it's allot easier to setup.

 Yes Vista does get a little annoying but it does have better results.

 Thanks
 Gordon


----------



## EliasGwinn

Quote:


  Originally Posted by *ted betley* /img/forum/go_quote.gif 
_Elias I tried using foobar (in lieu of wmp11) with my Benchmark usb dac and contrary to popular opinion I could not get foobar to sound as good as wmp11. Part of the reason may be that I couldn't get asio to work with foobar. I use foobar 0.9.5.1 and asio 1.2.6. Asio loads into my foobar component file but does not show in my preferences. Should I continue to try to get asio to work or is there some other way to optimize foobar to exceed sq of wmp11?_

 

Ted,

 We've put together a "Audio Info Page" on our website that has our recommendation for setting up various players and O.S.'s. Here is the link to the foobar page on there:

Foobar2000 for Windows - Setup Guide - Benchmark

 If you have any questions about anything on that or another page, just let me know.

 Thanks,
 Elias

 I


----------



## EliasGwinn

Quote:


  Originally Posted by *doctorcilantro* /img/forum/go_quote.gif 
_I rec. J. River to Benchmark a year ago or so when on the phone; you guys still need to check out it._

 

I know, I know...
	

	
	
		
		

		
			





...I've got to test J. River. I have it high on the 'to-do' list.

 Speaking of testing 3rd party products, I just tested this USB extender:

Amazon.com: USB Extender, RJ45 Cable Extends To 150ft: Electronics

 At lengths up to 175 feet, it works great with the DAC1 USB and/or DAC1 PRE.

 I didn't measure jitter since jitter doesn't effect the DAC1. I'm guessing its pretty severe at such lengths. 

 Thanks,
 Elias


----------



## ted betley

Thanks Elias. So then I don't have to worry about Kernel Streaming or using Asio to optimize sonics?


----------



## EliasGwinn

Quote:


  Originally Posted by *ted betley* /img/forum/go_quote.gif 
_Thanks Elias. So then I don't have to worry about Kernel Streaming or using Asio to optimize sonics?_

 

edit: Kernel Streaming and ASIO are good for avoiding kmixer. However....

 ...if you're using the DAC1 USB or DAC1 PRE, there will be no processing to the audio by Kmixer, with one exception - if you play audio from more then one application at the same time. If you play audio from two different applications simultaneously, and if the audio streams have different sample rates, then kmixer will sample-rate convert the 2nd stream to match the sample rate of the 1st. This sample rate conversion is very poor, and should be avoided.

 In other words... play one audio file at a time, and you will have top-notch digital audio streaming to the DAC1 USB/PRE.

 Thanks,
 Elias


----------



## ted betley

thanks!!


----------



## kool bubba ice

Elias, what is the failure rate with the DAC1? Having a 5 yr warranty is impressive & I'm sure the failure rate is very low.


----------



## EliasGwinn

Quote:


  Originally Posted by *kool bubba ice* /img/forum/go_quote.gif 
_Elias, what is the failure rate with the DAC1? Having a 5 yr warranty is impressive & I'm sure the failure rate is very low._

 

I don't know the answer to this, but it is very low. I don't know if we have a running, calculated rate. I can try to find out if you really want to know...

 Thanks,
 Elias


----------



## Wavelength

Quote:


  Originally Posted by *EliasGwinn* /img/forum/go_quote.gif 
_...if you're using the DAC1 USB or DAC1 PRE, there will be no processing to the audio by Kmixer, with one exception - if you play audio from more then one application at the same time. If you play audio from two different applications simultaneously, and if the audio streams have different sample rates, then kmixer will sample-rate convert the 2nd stream to match the sample rate of the 1st. This sample rate conversion is very poor, and should be avoided.

 In other words... play one audio file at a time, and you will have top-notch digital audio streaming to the DAC1 USB/PRE.

 Thanks,
 Elias_

 

Elias,

 Actually if you are using DirectSound the audio will always go through the KMIXER regardless of other applications. Thomas can share his experience with that.

 Of course with Vista the KMIXER is gone and results there have been much better with DirectSound applications.

 We found that ASIO can sound different between applications. There are a few out there. ASIO4ALL.com , ASIO2KS.de are both free and usb-audio.com cost some amount of money, I can't remember. 

 On some PC's Kernel Streaming will cause lock ups. This is usually due too the wrong USB driver being installed. If you can determine the USB host interface controller. You can usually go too their site and download the most recent USB driver and that will fix the problem.

  Quote:


 Speaking of testing 3rd party products, I just tested this USB extender:

 Amazon.com: USB Extender, RJ45 Cable Extends To 150ft: Electronics

 At lengths up to 175 feet, it works great with the DAC1 USB and/or DAC1 PRE.

 I didn't measure jitter since jitter doesn't effect the DAC1. I'm guessing its pretty severe at such lengths. 
 

Elias, problem with these is they cause data errors at the TAS1020. If you have the emulator on the TAS1020 you can read the error rate received. I tested a number of cables and the only one that really worked was the Opticis cable:

:::OPTICIS:::

 It's a little pricey and comes with a sub standard power supply but anyone with a little sense could pick up a good linear regulated 5V supply (Jameco.com) and hack in the small I think 1.3mm power cable.

 Thanks
 Gordon


----------



## gjwaudio

Hi Gordon

 I'm curious about your observations, can you please elaborate on this testing...

  Quote:


  Originally Posted by *Wavelength* /img/forum/go_quote.gif 
_We found that ASIO can sound different between applications._

 

Was the Benchmark DAC1 involved ? Was this USB-audio or a dedicated soundcard ? "CD-quality" or "Hi-Rez" music ? Playback through which applications ? Listening via headphones or speakers ?

 My interest is piqued because I had a similar experience listening to 24/96 files through my DAC1 Classic (fed via S/PDIF coax from M-Audio Delta 1010, using M-Audio's ASIO driver).

 Any info would be appreciated.

 Cheers,
 Grant


----------



## roadtonowhere08

Quote:


  Originally Posted by *EliasGwinn* /img/forum/go_quote.gif 
_This is an excellent question. I'll have to test for this. It would depend on the ripping software. The software would need to recognize pre-emphasis and encode that within the subsequently created audio file. I don't know if they take that into account or not. I'll try to find out...

 Thanks,
 Elias_

 

Any update on playing pre-emphasis CDs via the computer?


----------



## doctorcilantro

Alex B discussed pre-emphasis quite a bit a few years ago at Interact (J. River forum); I couldn't find anything in a search but he responded to the post I made which addressed the questions put forth here.

 DC
  Quote:


 I don't' see how it could be possible on PC playback. The audio playback system does not have a mechanism for embedding that kind of information in the played PCM stream.

 I guess that a standard for adding this info in the PCM stream exists. Otherwise the de-emphasis feature in DAC1 would be useless. I have assumed that stand-alone CD players with integrated DACs read this flag and switch the de-emphasis correction on using an external method outside the actual audio stream, but maybe this is incorrect.

 I started a thread about pre-emphasis three years ago: Pre-emphasis.

 Here is how I use EQdb for de-emphasis correction: EQdb 2.0.2 Released


----------



## Wavelength

Quote:


  Originally Posted by *gjwaudio* /img/forum/go_quote.gif 
_Hi Gordon

 I'm curious about your observations, can you please elaborate on this testing...



 Was the Benchmark DAC1 involved ? Was this USB-audio or a dedicated soundcard ? "CD-quality" or "Hi-Rez" music ? Playback through which applications ? Listening via headphones or speakers ?

 My interest is piqued because I had a similar experience listening to 24/96 files through my DAC1 Classic (fed via S/PDIF coax from M-Audio Delta 1010, using M-Audio's ASIO driver).

 Any info would be appreciated.

 Cheers,
 Grant_

 

Grant,

 We do have the USB1 here but I have not really tested ASIO drivers on it. I think in general all ASIO drivers will treat the interface the same way so differing sound will effect all products.

 What we really found was that for different front end software different ASIO drivers sounded different and this was when we investigated the 3 types, like:

 ASIO2KS: Best for Foobar and J River
 ASIO4ALL: Winamp Pro

 I do remember looking at the ASIO stack years ago but cannot in my mind determine what or why they are different sounding.

 I can say this for Vista. In all respects using DirectSound over ASIO in Vista always proved to sound better than XP and ASIO no matter what the front end program was.

 Thanks
 Gordon


----------



## EliasGwinn

Quote:


  Originally Posted by *Wavelength* /img/forum/go_quote.gif 
_Elias,

 Actually if you are using DirectSound the audio will always go through the KMIXER regardless of other applications. Thomas can share his experience with that._

 

You may be misunderstanding my point. Streaming audio from multiple app's may cause sample-rate conversion by kmixer. However, streaming from one app only will have no affect on audio quality.

 I am speaking about kmixers interaction with the USB interface on DAC1 products. This may not hold true with products from other manufacturers.

  Quote:


  Originally Posted by *Wavelength* /img/forum/go_quote.gif 
_We found that ASIO can sound different between applications. 

 Elias, problem with these is they cause data errors at the TAS1020. If you have the emulator on the TAS1020 you can read the error rate received._

 

Please understand that the USB firmware in the DAC1 USB and DAC1 PRE are completely different from that which is used by other manufacturers. The firmware in DAC1 products was custom built collaboratively between Centrance and Benchmark. It is not an off-the-shelf solution. USB interfaces in other products may have varying results that do not correspond to that achieved with the DAC1 products.

 Thanks,
 Elias


----------



## EliasGwinn

Quote:


  Originally Posted by *roadtonowhere08* /img/forum/go_quote.gif 
_Any update on playing pre-emphasis CDs via the computer?_

 

Unfortunately, I don't have any information on this yet. It is on my to-do list and I will try to get an answer for you very soon.

 Thanks,
 Elias


----------



## thomaspf

Quote:


  Originally Posted by *EliasGwinn* /img/forum/go_quote.gif 
_You may be misunderstanding my point. Streaming audio from multiple app's may cause sample-rate conversion by kmixer. However, streaming from one app only will have no affect on audio quality.

 I am speaking about kmixers interaction with the USB interface on DAC1 products. This may not hold true with products from other manufacturers.



 Please understand that the USB firmware in the DAC1 USB and DAC1 PRE are completely different from that which is used by other manufacturers. The firmware in DAC1 products was custom built collaboratively between Centrance and Benchmark. It is not an off-the-shelf solution. USB interfaces in other products may have varying results that do not correspond to that achieved with the DAC1 products.

 Thanks,
 Elias_

 

Well, here we go again. So, you are claiming now that your adapter does not undergo the standard volume processing in kmixer?

 Last time I looked at the DAC1 it still used the standard USBaudio driver in Windows and that always applies volume processing on 16/44.1 streams no matter which firmware is connected. Have you now concluded otherwise?

 Cheers

 Thomas


----------



## EliasGwinn

Quote:


  Originally Posted by *thomaspf* /img/forum/go_quote.gif 
_Well, here we go again. so, you are claiming now that your adapter does not undergo the standard volume processing in kmixer?

 Last time I looked at the DAC1 it still used the standard USBaudio driver in Windows and that always applies volume processing no matter which firmware is connected. Have you now concluded otherwise?

 Cheers

 Thomas_

 

No, the DAC1 does, in fact, use kmixer (unless specifically bypassed). However, the DAC1 USB may interact with kmixer differently then other USB interfaces (with regard to clock control and sample rate conversion).

 The debate over whether kmixer is bit-transparent is something entirely different.

 Thanks,
 Elias


----------



## thomaspf

Well to start with the DAC1 does not interact with kmixer at all. The DAC interacts with USBaudio.sys and that driver interacts with kmixer. In fact is does that the same way it interacts for any other device that uses the standard Windows driver.

 The only difference I have found looking at the way the firmware behaves is that it registers itself as a device that only supports 24 bit data and if my memory serves me correctly all the relevant bit rates 44.1, 48, and 96Khz. So the main difference is that firmware does not support 16bit mode which is natively supported by the USB chipset. I also understand that Centrance has implemented improved buffering which will reduce drop outs which is a great thing but orthogonal to the kmixer issue.


  Quote:


  Originally Posted by *EliasGwinn* 
_No, the DAC1 does, in fact, use kmixer (unless specifically bypassed). However, the DAC1 USB may interact with kmixer differently then other USB interfaces (with regard to clock control and sample rate conversion).

 The debate over whether kmixer is bit-transparent is something entirely different._

 

If I interpret what you are saying correctly then you only clarify how kmixer works. No sample rate conversion if the sound card supports the bit rate that is requested by the app. No mixing or resampling if there is only a single app.

 That is indeed correct and of course applies not only to the DAC1 but to any sound card or USB adapter that supports the standard sample rates. The DAC1 works well in that regard and so do many other USB adapters using the TAS1020 chip.

 While we are talking about this. Have you now concluded that kmixer is not bit transparent unless explicitely bypassed? 

 Cheers

 Thomas


----------



## Wavelength

Quote:


  Originally Posted by *EliasGwinn* /img/forum/go_quote.gif 
_I am speaking about kmixers interaction with the USB interface on DAC1 products. This may not hold true with products from other manufacturers._

 

I think Thomas replied to this, but really common the USB driver cannot mystically control the upper layers of the audio stack.

  Quote:


 Please understand that the USB firmware in the DAC1 USB and DAC1 PRE are completely different from that which is used by other manufacturers. The firmware in DAC1 products was custom built collaboratively between Centrance and Benchmark. It is not an off-the-shelf solution. USB interfaces in other products may have varying results that do not correspond to that achieved with the DAC1 products.

 Thanks,
 Elias 
 

Elias,

 Actually I understand this very well. I bought your unit because of the claims you made. I put it under some pretty heavy testing. Since I wrote my code for Wavelength and code for other companies and have the emulator here and a USB analyzer, I can tell you a few things.

 First the Centrance code is based on the TI sample code called "REFERNCE". Basically what they did was add support for 48. 88.2 and 96 which is pretty easy to do. They actually could have changed the SoftPLL code a little more to add longer delays and less of an incident to the clock changes of the host.

 But ok... here is what you need to know about the TAS1020 that really has nothing to do with coding at all.

 The TAS1020 has firmware and software (Centrance in your case). The firmware is solid in ROM code and cannot be altered. The firmware handles what is called the Serial Interface Engine (SIE). The SIE works with the firmware to gather and send USB packets to the HOST. The SIE will compare the CRC of a packet and strip the header info and present only audio data to the circular buffer used to fuel the DAC or ADC. If the CRC is incorrect an error is indicated and the error count is incremented. The packet is still applied to the circular buffer. The source code for the firmware ROM is included in the REFERNCE code because the firmware and software are linked together.

 Actually the software really has no interaction in receiving a packet of ISO DATA. Sure it sets up the DMA pointers lengths and such but other than that it has nothing to do with it.

 Therefore when a CRC error happens the Application would not even know that is happening. Just like if the transport feeding the SPDIF too your dac get's an error your dac has no idea. As well as if the SPDIF receiver in your DAC get's and error it doesn't really do anything as it can't. What's it going to do FIX it.

 Allot of these USB repeaters cause CRC errors because they cannot retain what is called the Differential EYE that is required for good USB communications. This especially true when you are talking larger packets of data like 576 bytes per 1ms that you would need for 24/96.

 I would suggest you get a USB Analyzer and see for yourself. Though that won't tell you if the TAS1020 got and error but if the Analyzer does then you know for sure the TAS1020 would.

 Elias, the idea of a forum like this is to exchange information. One of the biggest problems with Computer Audio is that there is so much misinformation out there that people have been lead down a path of poor sound and feel the technology is not there yet.

 Let's not add to that problem.

 Thanks
 Gordon


----------



## doctorcilantro

I have a question regarding the output of a -1db 1khz sine wave measured on the XLR outputs of the DAC1.

 I set the Calibrate output to 4v and then applied the Replay Gain in J. River which, after analysing the file, used a -19db reduction to achieve the target -89db. Accordingly, the voltage dropped significantyly on the DAC1. IIRC, it was .5mv. 

 I figured I could simply crank up the pots to 4v, with the volume reduction applied, and achieve a new reference level that incorporated the volume leveling that is applied by J. River.

 But I can't seem get the output up over 1.2v with the -19db (as shown in J. River) reduction. 

 I ended up setting the DAC1 with the unadultered -1db 1khz sine wave to 7.7v and then used the volume leveling. I see peaks of 4v now while playing music out of the DAC1 while volume leveling is active.

 I'm obvously confused as to why the signal was unable to be increased to a 4v; was the -19db reduction crushing the signal? I'm a bit confused on how to find a "correct" calibrated output while using such volume leveling.

 DC


----------



## eweitzman

Gordon, Thomas, Elias,

 The crux of this perennial kmixer/transparency/firmware/yada-yada-yada debate is whether the bits make it from the app to the USB port without being changed. It seems that Gordon's USB analyzer can detect errors but cannot compare the data received to the data that was sent. Plus it's probably not something your average computer user would have sitting in a box somewhere.

 I'm no driver/firmware developer, but looking at some of the architecture diagrams for the windows sound system (on XP), I see that filters can be inserted between kmixer and the driver. I've pondered looking into writing a filter that would grab the stream, dump it into a file, then allow post-playback comparison with the source. This seems like an obvious way to figure out what's going on, so there must be difficulties with this approach that I'm not aware of that have prevented others from doing it. Things like timing and/or latency.

 Do any of you have experience with this and can maybe advise me about what the pitfalls would be?

 Thanks,
 - Eric


----------



## thomaspf

I have done simple straight forward recordings of the stream from a digital output. 

 Steve is now shipping the Centrance or I should probably say a variant of the Centrance USB firmware on his adapters and they register the same way the DAC1 does.

 The result just confirmed what you do expect from kmixer in the middle. It does ever so slightly change the bits.

 It is also straight forward to get around this. Just use direct kernel streaming and you are bit transparent. I believe I suggested this some time back very early on this thread.

 The DAC1 is a great product and it is a good choice that it uses the stable standard driver. However, in order to get bit transparent playback you need to bypass kmixer like you have to do with any other USB adapter using that driver.

 Cheers

 Thomas


----------



## EliasGwinn

Quote:


  Originally Posted by *Wavelength* /img/forum/go_quote.gif 
_Elias, the idea of a forum like this is to exchange information. One of the biggest problems with Computer Audio is that there is so much misinformation out there that people have been lead down a path of poor sound and feel the technology is not there yet.

 Let's not add to that problem.

 Thanks
 Gordon_

 

I hope you don't think that I have intentions to mis-inform. We may have different ideas, but I try to speak for myself and my own knowledge. I don't claim them to be absolutes.

 We appreciate your contribution to this thread, but please don't instigate accusations and finger pointing. We all have the same objectives...to understand all things audio as much as possible.

 Thanks,
 Elias


----------



## EliasGwinn

I will not be active on this forum until next week. Have a great holiday!

 Thanks,
 Elias


----------



## Wavelength

Quote:


  Originally Posted by *eweitzman* /img/forum/go_quote.gif 
_Gordon, Thomas, Elias,

 The crux of this perennial kmixer/transparency/firmware/yada-yada-yada debate is whether the bits make it from the app to the USB port without being changed. It seems that Gordon's USB analyzer can detect errors but cannot compare the data received to the data that was sent. Plus it's probably not something your average computer user would have sitting in a box somewhere._

 

Eric,

 I have many testing devices including the Prism dScope III which is the only audio test gear that can stream out USB natively with test data and compare the analog signal.

 I use the USB Analyzer to test the USB line itself. That being errors and data and yes I can capture data with it and compare it too the source.

 But your method maybe better or an even better method would be too write a fake audio output device driver which would you could store the streamed data.

 All of these would be helpful in determining what is going on in the audio stack.

 Thanks
 Gordon


----------



## sejarzo

Quote:


  Originally Posted by *doctorcilantro* /img/forum/go_quote.gif 
_ ... I'm obvously confused as to why the signal was unable to be increased to a 4v; was the -19db reduction crushing the signal? I'm a bit confused on how to find a "correct" calibrated output while using such volume leveling.

 DC_

 

I'm more than a bit confused by this post........

 If a 0 dBFS digital signal should produce a 4 Vrms analog signal at the XLR's on the DAC1, a -20 dBFS signal should only produce 0.4 Vrms (20 dB reduction is equal to a 10x reduction in voltage.)

 If you adjusted a -20 dBFS signal to 4 Vrms at the outputs of the DAC1, a 0 dBFS digital signal would drive the output way beyond clipping. Seems to me that any output stage wouldn't be designed with such massive gain, anyway.

 Why would you be need such a high reduction in level in JRiver?

*EDIT:* After looking at the specs on the Benchmark site, I remain confused.....

 According to Benchmark, with the attenutation jumpers set for no attenuation, the output range on the XLR's for a 0 dBFS digital input is +9 to +29 dBu, which I calculate to be 2.18 Vrms to 21.8 Vrms (0 dBu = 0.775 Vrms.)

 If the maximum digital signal that you will send to the Benchmark after replay gain adjustment is roughly -19 dBFS, the max output from the balanced connections with the adjustment pots cranked to the max should be +10 dBU (the stated max output level of +29 dBu, less 19 dB.) +10 dBu is equivalent to 2.45 Vrms....or about double what you measured based on this:

  Quote:


  Originally Posted by *doctorcilantro* /img/forum/go_quote.gif 
_But I can't seem get the output up over 1.2v with the -19db (as shown in J. River) reduction._

 

Are you consistently measuring the output level between pins 2 and 3 of the XLR's (between the "hot" and "cold" signals) or did you possibly measure between pin 2 and pin 1 (hot to ground) or pin 3 and pin 1 (cold to ground)? If you measured from either signal pin to ground, you should get 1/2 the voltage measured between the signal pins.

 What are you striving for when you refer to a " 'correct' calibrated output"?


----------



## doctorcilantro

I've had a long day , and plan to post more details tomorrow but here goes. Thanks for you interest.

 Only the -1db 1khz sine wave results in -19db in J. River with Replay Gain enabled; I rarely see reductions this extreme. Usually the average is -8db to -12db. Since the sine wave is close to clipping, it accordingly needs a great amount of reduction to hit the target -89db that RG uses as the standard.

 I have the 3rd pin floated on the XLR output cables. I was measuring the RCA hot pin and the RCA body.

 DC


----------



## vcoheda

elias:

 could you please comment on the difference in application between the coaxial digital input and the XLR balanced digital input on the DAC1. for example, if i intend to use the DAC1 with a balanced setup (amp and headphones), and use the coaxial instead of the XLR digital input from my music server to the DAC1, does this mean the setup is NOT balanced?

 basically, is there a difference between these two setups.

 1. MusicServer (_digital coaxial cable_) > DAC1 (XLR analog cable) > Bal Amp > Bal HP

 2. MusicServer (_digital XLR cable_) > DAC1 (XLR analog cable) > Bal Amp > Bal HP

 thanks


----------



## sejarzo

Quote:


  Originally Posted by *doctorcilantro* /img/forum/go_quote.gif 
_I have the 3rd pin floated on the XLR output cables. I was measuring the RCA hot pin and the RCA body._

 

That explains it...you 1.2 Vrms measurement was what you should have seen for a -20 dBFS digital input. What I don't get is why you want 4 Vrms peaks, because that's way beyond the level that clips your monoblocks (which are spec'ed for an input sensitivity of only 1 volt!) 

 Why do you bother with XLR-to-RCA cables when the DAC1 has RCA outputs?


----------



## emmodad

Quote:


  Originally Posted by *vcoheda* /img/forum/go_quote.gif 
_elias:

 could you please comment on the difference in application between the coaxial digital input and the XLR balanced digital input on the DAC1. for example, if i intend to use the DAC1 with a balanced setup (amp and headphones), and use the coaxial instead of the XLR digital input from my music server to the DAC1, does this mean the setup is NOT balanced?

 basically, is there a difference between these two setups.

 1. MusicServer (digital coaxial cable) > DAC1 (XLR analog cable) > Bal Amp > Bal HP

 2. MusicServer (digital XLR cable) > DAC1 (XLR analog cable) > Bal Amp > Bal HP

 thanks_

 


 (In that which follows, Wikipedia is actually helpful in riding to the rescue... take a look at AES/EBU - Wikipedia, the free encyclopedia , as well as the "Hardware Specifications" section of S/PDIF - Wikipedia, the free encyclopedia )



 As Elias recently posted innocuously that he'd be away for a few days (which may mean he's off to some recording / partying bacchanal or somesuch 
	

	
	
		
		

		
			





 )... and as I'm terribly awake....to hopefully help to clarify for you, let's address the two "Balanced" signal paths in your system... a priori, I have no association with Benchmark except owning a DAC1, among several DACs.



 Short version:


> In simplest terms, no, there is no significant difference (caveats not to be beaten to death here: cable quality, environmental noise, standard topics of debate wrt cable design, digital data transmission, noise isolation, yadayada).


Digital audio data is transferred to a DAC on optical fibre, balanced cable or unbalanced cable. In a well-designed system, the bits of digital audio data "arriving and greeted by the DAC" will be identical no matter which of those "digital cables" is used.

 Without getting into arcane stuff about digital data transmission: it shouldn't matter if you connect to the DAC1 from a S/PDIF source using S/PDIF coax cable (RCA connector) or from an AES3 aka AES/EBU source using AES/EBU balanced cable (XLR connector); although between these two there may be a preference for AES/EBU connection (ie in an electrically-noisy environment). And as the marketing makes clear, Benchmark are particularly proud of their digital receiver designs and jitter tolerance...

 The DAC receives, understands, decodes and unpacks the bits of digital audio data which arrived from the digital source (ie your MusicServer).

 The DAC then performs Digital-to-Analog conversion of the audio samples, creating an analog output signal. This analog output signal is buffered into a robust form which can drive cables and attached equipment (and, with some DACs, apparently also some peoples' balanced heaphones 
	

	
	
		
		

		
		
	


	




 ), and is provided in balanced and unbalanced analog output signal versions on respective XLR and RCA output jacks.

 Hence, _balanced electrical transmission of digital audio data into the DAC1_ is a function which is separated (thru the process of Digital-to-Analog conversion) (and a bunch o' electrical isolation stuff) from _balanced electrical transmission of analog audio signals_ out of the DAC1 to your amp...

 Your two "balanced" electrical transmission functions are separate; for your concern, which is the analog side o' things, you are "balanced"* no matter which flavor of digital input you choose to utilize. [* comment here is wrt analog output signal connection, not your frame of mind, disposition, Head-Fi wallet utilization profile, etc. 
	

	
	
		
		

		
		
	


	




 ]



 Less-short 
	

	
	
		
		

		
		
	


	




 version:

 You are looking at two instances of signal transmission:


> 1/ (Digital source = MusicServer) > (balanced or unbalanced digital audio signal) > DAC1


and then


> 2/ DAC1 > (balanced or unbalanced analog audio signal) > (analog playback = amp & HP)


in your particular case, 2/ defaults to


> DAC1 > (balanced analog audio signal) > balanced (amp & HP)


 
 Digital Audio path into DAC

 The digital audio information which is output from your MusicServer can (assuming availability of appropriate hardware output connections) be transmitted onward in one of several data formats, over different possible physical transport links:


> data format: either professional (AES/EBU) or consumer (S/PDIF); "S/PDIF is essentially a minor modification of the original AES/EBU standard for consumer use, providing small differences in the protocol and requiring less expensive hardware" (credit Wikipedia)
> 
> physical transport (ie the "digital cable"): consumer S/PDIF format data is generally carried on either TOSlink optical or RCA-terminated 75-ohm unbalanced coax cable; professional AES/EBU format data is generally carried on either XLR-terminated 110-ohm balanced coax cable or BNC-terminated 75-ohm unbalanced coax cable. You will often see a small RCA-to-BNC connecter/adapter widget allowing you to use a "typically-available" RCA-RCA 75-ohm coax cable with a BNC input connector. Less commonly, you may also see ST (aka "glass" or "glass fibre") as another type of high-end optical connector. (IIRC, Benchmark's DAC1 manuals have a useful section addressing the different connectors.)


There are many posts in many fora about the merits of coax vs TOS, or TOS vs coax; jitter tradeoffs; possible implementation-dependent TOS performance limitations for 192 Khz; etc, etc...you can decide for yourself..... also some interesting observations and opinions about cable length considerations on SPDIF coax given quality of the particular receiver circuitry used. Will leave it to Benchmark to express their own opinion wrt any of those.

 (Cable cognesenti and high-speed data transmission folk, please don't bash the oversimplified following): without getting into big cable shootout stuff, simply recall that even though the signal from your MusicServer is digital audio, it is still a transmission of several channels of audio data, along with associated subcode, headers, etc., as a stream of packaged packets of "bits," via electrical signals (which optimally represent those bits by looking something like "square wave signals" with high-speed transitions between two different voltage levels), traveling in analog electrical fashion over good ol' wires..... so there can be benefits in "balanced" electrical connection, similar to the case with what you would consider a more-traditional audio-frequency analog signal.

 There are those (myself included) who opine that, if you have the appropriate AES/EBU gozouttas and gozintas (or, for the non-EE folk, that translates to output connectors and input connectors), with bang-for-the-buck consideration AES/EBU can be a preferable hardware interconnect for digital signals. However, S/PDIF coax can certainly be just fine. Both are generally considered preferable to TOS optical.

 Input receive, decode, unpack; and data conversion processes follow as noted previously.



Analog Audio path out of DAC

 You've obviously studied, understood and come to your conclusions, and are using balanced connections in the analog audio side of the system. So, no explanation necessary. 
	

	
	
		
		

		
		
	


	






 So: your MusicServer's digital audio data can be transmitted on balanced or unbalanced cables to the DAC1; whichever is used, the digital audio signal data which is received, decoded, and disassembled-into-two-channels-of-audio-samples by the DAC1 will* be the same. (* this is where the high-speed data transmission geeks, experts, and opinionators will have itchy keyboard fingers, but for sake of Head-Fi, if the equipment manufacturers have done an expected good job, the correct bits will arrive at the DAC1, and will be correctly received, understood and decoded for the next step: conversion to analog).

 The DAC1 will then perform Digital to Analog conversion on those two channels, and the D/A chip output will be provided at the DAC1 output in both balanced and unbalanced voltage-drive versions. You are using the balanced analog outputs on XLR connectors.

 So as noted earlier in the non-War-and-Peace Short Version, _balanced electrical transmission of digital audio data into the DAC1_ is a function which is separated (thru the process of Digital-to-Analog conversion) (and a bunch o' electrical isolation stuff) from _balanced electrical transmission of analog audio signals_ out of the DAC1 to your amp...

 Your concern is the analog part of the equation following the DAC, and is "balanced" no matter which flavor of digital input you choose to utilize.



 HTH, we aim to entertain

 emmo = asleep
 dad = still awake


----------



## doctorcilantro

Quote:


 What are you striving for when you refer to a " 'correct' calibrated output"? 
 

I meant that I want to correctly compensate with hardware (DAC1), the reduction in volume implemeneted by Replay Gain.

  Quote:


  Originally Posted by *sejarzo* /img/forum/go_quote.gif 
_That explains it...you 1.2 Vrms measurement was what you should have seen for a -20 dBFS digital input. What I don't get is why you want 4 Vrms peaks, because that's way beyond the level that clips your monoblocks (which are spec'ed for an input sensitivity of only 1 volt!) 

 Why do you bother with XLR-to-RCA cables when the DAC1 has RCA outputs?_

 

I'm using the RCA outputs on the DAC1 to connect a Tascam tapedeck. I use the XLR (w/ floated pin) to the unbalanced Sunfire. I read here that 3.? was recommended as the output voltage on the DAC1. 

  Quote:


 If a 0 dBFS digital signal should produce a 4 Vrms analog signal at the XLR's on the DAC1, a -20 dBFS signal should only produce 0.4 Vrms (20 dB reduction is equal to a 10x reduction in voltage.) 
 

This is what I observed, about .5 with -19db reduction applied via Replay Gain. This is not a user setting; Replay Gain on auto calculates peak and intensity, the formula utilized results in the target 83db.

  Quote:


 Having calculated a representative RMS energy value for the audio file, we now need to reference this to a real world sound pressure level. The _audio industry_ doesn't have any standard for listening level, but the _movie industry_ has worked to an 83dB standard for years. 
 What the standard actually states is that a single channel pink noise signal, with an RMS energy level of -20 dB relative to a full scale sinusoid should be reproduced at 83 dB SPL (measured using a C-weighted, slow averaging SPL meter). In simple terms, this means that everyone can set their volume control to the same (known, calibrated) gain. 
ASIDE: This number (83dB SPL) wasn't picked at random. It represents a comfortable average listening level, determined by professionals from years of listening. That reference level of -20dB pink noise isn't random either. It causes the calibrated average level to be 20dB less than the peak level. In other words, it leaves 20dB of headroom for louder than average signals. So, if CDs were mastered this way, the average level would be around -20dB FS, leaving lots of room for the dramatic peaks which make music exciting. 
 

I'm really a noob with all this, and I appreciate any input you have. I'm just trying to compensate for the 83db which I believe is too low. So my thinking was to measure the voltage with RG applied and crank it up a bit; obviously it's a little more complicated than that.

 EDIT - Okay so the pin to pin voltage 1.2 x 2 equaling 2.4 makes sense. If the pots are cranked and RG is applying -19dbfs and I'm using the 0db attenuation, I'd be running a little hot at 2.4vrms as you stated the standrad is around 2vrms for RCA?

 Sunfire Classic manual: 
http://www.sunfire.com/pdf/Classic%2...p%20Manual.pdf

 DC


----------



## Wavelength

Quote:


  Originally Posted by *vcoheda* /img/forum/go_quote.gif 
_elias:

 could you please comment on the difference in application between the coaxial digital input and the XLR balanced digital input on the DAC1. for example, if i intend to use the DAC1 with a balanced setup (amp and headphones), and use the coaxial instead of the XLR digital input from my music server to the DAC1, does this mean the setup is NOT balanced?

 basically, is there a difference between these two setups.

 1. MusicServer (digital coaxial cable) > DAC1 (XLR analog cable) > Bal Amp > Bal HP

 2. MusicServer (digital XLR cable) > DAC1 (XLR analog cable) > Bal Amp > Bal HP

 thanks_

 

Gang,

 In general the best results for SPDIF are via BNC to BNC. This is the only true 75 ohm connector. Using an RCA SPDIF cable is the worst because an RCA cable can never be 75 ohms. Most of the impedance is due to the connector size and what is called DOD or diameter over diameter. Look at a BNC cable and see the pin size center conductor then look at the outer conducter it's about the same size as the out side of an RCA. If we were to take the RCA center conductor and calculate the outer diameter it would I think have to be like 4".

 BTW if you unit's output has an RCA connector there is a great little device available at Radio Shack that is a RCA 75 ohm BNC convertor. Use this with a BNC-BNC cable for best results into the DAC1's BNC connector.

 Best digital SPDIF cable I have tried and used is the Nirvana T2. Killer!

 XLR SPDIF have a terrible problem with reflections. If you look at the waveform it always has some asymetrical problems because of the length the drivers and receiver circuit. Also XLR is really hard to keep the impedance 110 as there are 2 active conductors and ground and as we saw above the connector becomes more of an issue with 3 conductors and the diameter sizes.

 Thanks
 Gordon


----------



## sejarzo

Quote:


  Originally Posted by *Wavelength* /img/forum/go_quote.gif 
_.... In general the best results for SPDIF are via BNC to BNC. This is the only true 75 ohm connector. Using an RCA SPDIF cable is the worst because an RCA cable can never be 75 ohms._

 

So I take it that you dispute the notion that the Canare RCAP connector cannot present a true 75 ohm characteristic impedance as is claimed?

Specs for the Canare RCAP-C Series Connector

 As you indicate, I thought that this was strictly a geometry issue and there is no getting around the diameter of the RCA pin--but my understanding of characteristic impedance is quite limited.


----------



## doctorcilantro

Blue Jeans Cable--"True 75 ohm" RCA Plugs


Zaolla Products

 "A superior High-End Digital interconnect.... We achieve true 75 Ohm by creating precision uniform cable geometry, using FPE, a special foam dielectric material. The result is a purer signal transmission with no jitter, and with dramatic reductions in phase-related high frequency signal anomalies."

 DC


----------



## sejarzo

Quote:


  Originally Posted by *doctorcilantro* /img/forum/go_quote.gif 
_EDIT - Okay so the pin to pin voltage 1.2 x 2 equaling 2.4 makes sense. If the pots are cranked and RG is applying -19dbfs and I'm using the 0db attenuation, I'd be running a little hot at 2.4vrms as you stated the standrad is around 2vrms for RCA?_

 

Most single-ended analog outputs on CD players, DAC's, etc. are designed to put out 2.0 Vrms for a full-scale, 0 dBFS digital signal. Some put out as little as 1.2 Vrms, but my Marantz SA8001 puts out 2.4 Vrms per the specs and my DMM.

 The difference between 2.4 and 2.0 Vrms amounts to only 1.6 dB, so it's not a lot--about one click on a typical detented volume control or stepped attenuator.

 I'm not a JRiver user, but if use of replay gain/volume leveling is important for you, can you set the replay gain target 4 or 5 dB higher? 

 [dB difference = 20 * log(Vb/Va)...so if you desire Vb to be 2.0 Vrms for the output at a 0 dBFS digital signal, and your current Va is 1.2 Vrms, that calculates out to a 4.44 dB increase.]


----------



## doctorcilantro

For some reason I always thought that using a fixed compensation with Replay Gain (+10db) would cause clipping, but I guess I'd just be setting it to 93db as opposed to 83db. Quiet songs have +db adjustments and loud songs have -db adjustments in Replay Gain, which can be confusing, but you can't think in terms of clipping when you see +db values in Replay Gain when it's shooting for 83db. I could have sworn, trying this in th e past, that I was seeing clipping the the EMU Patchmix.

 I guess I'll try the RG adjustment in J. River and then reset the pots to 2vrms which is what Sunfire just advised over the phone.



 DC


----------



## vcoheda

Quote:


  Originally Posted by *Wavelength* /img/forum/go_quote.gif 
_Gang,

 In general the best results for SPDIF are via BNC to BNC. This is the only true 75 ohm connector. Using an RCA SPDIF cable is the worst because an RCA cable can never be 75 ohms. Most of the impedance is due to the connector size and what is called DOD or diameter over diameter. Look at a BNC cable and see the pin size center conductor then look at the outer conducter it's about the same size as the out side of an RCA. If we were to take the RCA center conductor and calculate the outer diameter it would I think have to be like 4".

 BTW if you unit's output has an RCA connector there is a great little device available at Radio Shack that is a RCA 75 ohm BNC convertor. Use this with a BNC-BNC cable for best results into the DAC1's BNC connector.

 Best digital SPDIF cable I have tried and used is the Nirvana T2. Killer!

 XLR SPDIF have a terrible problem with reflections. If you look at the waveform it always has some asymetrical problems because of the length the drivers and receiver circuit. Also XLR is really hard to keep the impedance 110 as there are 2 active conductors and ground and as we saw above the connector becomes more of an issue with 3 conductors and the diameter sizes.

 Thanks
 Gordon_

 

this is good information. thanks.


----------



## doctorcilantro

Interesting thread regarding J. River, Replay Gain, and +0dbfs:

Did you know about 0 dBFS+?

 DC


----------



## Lord Chaos

I use the BNC cable that came with my Benchmark, and a BNC/RCA adapter to connect to the RCA/mini phone plug that plugs into the sound card. Theoretical nightmare, but it sounds great.


----------



## doctorcilantro

Here are some measurements:

 DAC to Pre (0dbfs+ sine wave at -20db applied by Replay Gain in J. River): .850v

 Pre to monoblock:

 Pre @ 9o'clock = 40mv
 Pre @ 12o'clock = 297mv
 Pre @ 2o'clock = .562v
 Pre @ 3o'clock = .935v

 When you say the standard is 2vrms output on a lot of cd players, I still don't understand if that is 1vrms per channel or 2vrms per channel. Given the amount of gain at the above voltages I'm inclined to think the former (e.g. LOUD).

 My speakers are 98db/1w sensitive so I don't/can't really crank the pots up to 1vrms; listening gets loud at 10 o'clock.

  Quote:


 i would suggest that if you have the dac1 classic to put it on the 0db pads then set the rear pots as high as your preamp can tolerate before saturation.
 with the 0db pads in place the impedance is more similar to the usb version. the impedance on the padded outputs was an issue for me on the classic model. this mostly overcomes that i think. my pots are set at 3.43 volts and they could go higher but that sounds best. it took me many hours to arrive at that setting. 
 

Am I missing something with the 3.43vrms suggested by MusicMan above? After the measurements I made, cranking it up this much would blow apart my system. As I said my system is super loud due to 25 tube watts into 98db sensitive RF7s.

 DC


----------



## sejarzo

The typical output level out of the RCA's on CD players or DAC's is 2Vrms for a 0 dBFS digital signal on either left or right.....however, I don't really understand what you mean by "per channel". 

 You are seeing evidence of my concern that in the digital era, preamps are either unnecessary or, at the very least, their gain is way too high, because the standard output of digital sources is usually 4 dB or more above clipping for modern power amps with input sensitivities of around 1.2 Vrms. 

 In the "old days" increasing the source output voltage would improve signal to noise ratio in a meaningful way, but as most output stages have a lower noise floor now, setting your sources to a lower level to make volume control less touchy/more precise isn't likely to result in noise levels that intrude on the music.


----------



## Jaw007

Quote:


  Originally Posted by *majid* /img/forum/go_quote.gif 
_I got an email from them announcing an updated DAC1 with 24/96 USB input:

Benchmark Media Systems -- Professional Audio Products






 They claim no special drivers (e.g. ASIO) are required for it. You pay dearly for the privilege of USB, though, an extra $300..._

 

Thanks for the information.Very nice piece of equipment.


----------



## doctorcilantro

Sorry I meant left or right output (e.g. per channel).

 I'm starting to feel dense: a -20db 0dbfs sine wave, with 0db attenuator setting, can't be turned up to 2vrms. 

 If want to recreate a 0dbfs 2vrms output with an initial -20db reduction in J. River, it seems that's not possible with the DAC1. At this point, I don't think that would even be warranted or desired given how sensitive the pre/amps are.

  Quote:


 According to Benchmark, with the attenutation jumpers set for no attenuation, the output range on the XLR's for a 0 dBFS digital input is +9 to +29 dBu, which I calculate to be 2.18 Vrms to 21.8 Vrms (0 dBu = 0.775 Vrms.)

 If the maximum digital signal that you will send to the Benchmark after replay gain adjustment is roughly -19 dBFS, the max output from the balanced connections with the adjustment pots cranked to the max should be +10 dBU (the stated max output level of +29 dBu, less 19 dB.) +10 dBu is equivalent to 2.45 Vrms....or about double what you measured... 
 

Since I have effectively turned the XLR outputs into RCA by floating the 3rd pin, does the +9 to +29 still apply? Or does the XLR output now conform to the unbalanced RCA specs:

  Quote:


 UNBALANCED ANALOG OUTPUTS: 
 Number of Balanced Analog Outputs: 2 
 Output Connector: RCA 
 Output Impedance: 2.5 kΩ 
 Output Level Calibration Controls: Shared with Balanced Outputs 
 Output Level Range (at 0 dBFS) In “Calibrated” Mode: -3 dBu to +17 dBu 
 Output Level Range (at 0 dBFS) In “Variable” Mode: Off to +17 dBu 
 Calibration Adjustability: 2 dB / turn 
 Output Level Variation with Sample Rate (44.1 kHz vs. 96 kHz): < +/- 0.006 dB 
 



  Quote:


 [dB difference = 20 * log(Vb/Va)...so if you desire Vb to be 2.0 Vrms for the output at a 0 dBFS digital signal, and your current Va is 1.2 Vrms, that calculates out to a 4.44 dB increase.] 
 

If I add about 5db in Replay Gain I should see about 2vrms. I'll check this out. 

 DC


----------



## sejarzo

I don't believe that using "half" of the balanced output necessarily makes it equivalent to the unbalanced spec. The fact that the output level calibration control is shared with the balanced outputs seems to imply that it's located ahead of separate balanced and unbalanced driver stages, but that's only my guess.

 The +9 to +29 dBu/2.18 to 21.8 Vrms range for the balanced output with no attenuation jumper should refer to the level between pins 2 and 3 on the XLR's, so if you are taking a signal from pin 2 to signal ground via your XLR-to-RCA cable, the range of adjustment should be half of that, or 1.09 to 10.9 Vrms at a 0 dBFS digital input.

 If you invoke a 20 dB level reduction (equivalent to a 10x reduction in voltage) in JRiver, that means you should only get a 1.1 Vrms max between pin and sleeve of the RCA. Inexpensive DMM's probably aren't super accurate in that range--mine reads 0.08 or 0.09 Vrms high--so the measured 1.2 Vrms that you previously noted seems very reasonable.

 So yes, in your case with monoblocks that clip at 1 Vrms input and very sensitive speakers, it seems that 1.1 to 1.2 Vrms would be a logical setting for the output of the DAC1, doesn't it?


----------



## sejarzo

Quote:


  Originally Posted by *doctorcilantro* /img/forum/go_quote.gif 
_For some reason I always thought that using a fixed compensation with Replay Gain (+10db) would cause clipping, but I guess I'd just be setting it to 93db as opposed to 83db. Quiet songs have +db adjustments and loud songs have -db adjustments in Replay Gain, which can be confusing, but you can't think in terms of clipping when you see +db values in Replay Gain when it's shooting for 83db._

 

I just re-read that, and saw that you find quiet tracks actually have boost applied.

 What bothers me about using replay gain in any case is that it seems to deal with average signal levels. I primarily listen to classical, and many movements in symphonies have a huge dynamic range--I know there are sections that are very quiet, but EAC reports that there are peak levels of 98-99% in those same cuts.

 That means that even if the average level of the cut is low, there really should not be any digital gain applied for volume normalization because the peaks would clip....unless the algorithm would simultaneously apply some sort of compression, and I don't want to go there!


----------



## EliasGwinn

edit: duplicate post...


----------



## EliasGwinn

Quote:


  Originally Posted by *doctorcilantro* /img/forum/go_quote.gif 
_
 Since I have effectively turned the XLR outputs into RCA by floating the 3rd pin, does the +9 to +29 still apply? Or does the XLR output now conform to the unbalanced RCA specs:_

 

I'm sorry I've been M.I.A. for the last week...but I'll can help you with this setup.

 By floating pin-3, you are lowering the output by 6 dB (or 1/2 the voltage). So, the output in that case would range from +3 to +23 dBu for 0 dBFS inputs. Any attenuation in the software should be calculated from those numbers.

 As Sejarzo mentioned, if you are using Replay Gain, it is wise to configure it so that no boosts are applied. 

 Thanks,
 Elias


----------



## EliasGwinn

Quote:


  Originally Posted by *vcoheda* /img/forum/go_quote.gif 
_elias:

 could you please comment on the difference in application between the coaxial digital input and the XLR balanced digital input on the DAC1. for example, if i intend to use the DAC1 with a balanced setup (amp and headphones), and use the coaxial instead of the XLR digital input from my music server to the DAC1, does this mean the setup is NOT balanced?

 basically, is there a difference between these two setups.

 1. MusicServer (digital coaxial cable) > DAC1 (XLR analog cable) > Bal Amp > Bal HP

 2. MusicServer (digital XLR cable) > DAC1 (XLR analog cable) > Bal Amp > Bal HP

 thanks_

 

vcoheda,

 The balanced digital input does not affect the balanced/unbalanced output. There will be no difference whatsoever between the two scenarios you've outlined.

 Balanced digital transmission is simply a type of signal transmission mechanism that was originally thought to be superior to coaxial with regards to noise rejection and transmission distance capabilities. It was developed very early in the life of digital music. However, it has since been tested and generally agreed upon that well-shielded coaxial cable with tight impedance characteristics and impedance matching is more noise-immune and can successfully travel further distances then balanced digital transmission.

 The bottom line is...coaxial is technically superior to balanced digital transmission. There will be no differences in performance between coaxial and balanced input to the DAC1, as the DAC1 is built to be immune to any quality differences in digital signals. So, unless you're sending a digital signal over 1000 feet, there will be absolutely no difference with the DAC1.

 Thanks,
 Elias


----------



## ted betley

Elias I hate to bother you but I tried j river and loved the sound through my Benchmark usb dac. However based on all the research I have done on the various sites it seems that folks prefer asio vs direct sound as an interface. I can get direct sound to work but not asio4all. Any advice on how to do that? I downloaded asio4all, followed the 9 page documantation, set audio output options to asio, chose asio4all and still no success. I get a 'direct show filters not available...'or something similar diagnostic and then my pc unselects the Benchmark as the ouput device (in the Asio4all panel) and it reverts back to direct sound.


----------



## joijwall

I follow the bit-transparancy discussions with manic interest, although some/most of it is over my head. 
 My listening experience using MacBook 10.5.2/iTunes 7.6.1/DAC1 USB compared with a very good Linn Majik CD is that I hear no or minimal difference. This is disturbing. Connecting DAC1 USB directly to my Majik CD makes the sound clearer and better in my ears. 
 I'd like my Mac/DAC to perform better! Why? How? News?


----------



## infinitesymphony

Quote:


  Originally Posted by *joijwall* /img/forum/go_quote.gif 
_I follow the bit-transparancy discussions with manic interest, although some/most of it is over my head. 
 My listening experience using MacBook 10.5.2/iTunes 7.6.1/DAC1 USB compared with a very good Linn Majik CD is that I hear no or minimal difference. This is disturbing. Connecting DAC1 USB directly to my Majik CD makes the sound clearer and better in my ears. 
 I'd like my Mac/DAC to perform better! Why? How? News?_

 

Are you saying:

 MacBook -> DAC1 analog output sounds the same as Majik analog output
 or
 MacBook -> DAC1 sounds the same as Majik -> DAC1
 or
 Majik's -> DAC1 sounds better than MacBook -> DAC1

 Either way, they should all be bit-perfect.


----------



## riverlethe

Quote:


  Originally Posted by *joijwall* /img/forum/go_quote.gif 
_I follow the bit-transparancy discussions with manic interest, although some/most of it is over my head. 
 My listening experience using MacBook 10.5.2/iTunes 7.6.1/DAC1 USB compared with a very good Linn Majik CD is that I hear no or minimal difference. This is disturbing. Connecting DAC1 USB directly to my Majik CD makes the sound clearer and better in my ears. 
 I'd like my Mac/DAC to perform better! Why? How? News?_

 

Did you volume-match for this comparison?


----------



## Crowbar

Quote:


  Originally Posted by *riverlethe* /img/forum/go_quote.gif 
_Did you volume-match for this comparison?_

 

high five


----------



## Terje

I think this is an interesting test 

Odyssey Audio HK Forum - Mac Notebook + Benchmark DAC1 Pre Audition


----------



## Wavelength

Quote:


  Originally Posted by *joijwall* /img/forum/go_quote.gif 
_I follow the bit-transparancy discussions with manic interest, although some/most of it is over my head. 
 My listening experience using MacBook 10.5.2/iTunes 7.6.1/DAC1 USB compared with a very good Linn Majik CD is that I hear no or minimal difference. This is disturbing. Connecting DAC1 USB directly to my Majik CD makes the sound clearer and better in my ears. 
 I'd like my Mac/DAC to perform better! Why? How? News?_

 

Joijwall,

 Do make sure the equalizer, sound check and sound enhancer are off on iTunes. These are default on...

 We found on some MacBooks that they just can't keep up with the demands of Apple Lossless files. We found using uncompressed files like AIFF and WAV to sound better:

 One thing you can do to recover space on slower machines is to expand the audio files to AIFF (or WAV). It appears that allot of processing takes place to decode the files on the fly and therefore chaning the file types to straight PCM (AIFF/WAV) that the processor has less to do.

 Since the files are lossless they are fine to convert and you do not need to re-rip your cd's. Simply convert the files from Apple Lossless to either AIFF or WAV. But do note that WAV files do not imbede the cue information and if your library is damaged it will take some time reloading the cue information for the songs.

 Preferences->Advanced->Importing:

 Select AIFF or WAV as your output type and hit <OK>

 Select a couple of songs and then pull down Advanced and hit Convert songs to AIFF/WAV (which ever you seleted above).

 This will make copies of the songs you just selected. You can see the file types by adding KIND to your view options (View: View options...).

 Thanks
 Gordon


----------



## little-endian

Quote:


  Originally Posted by *EliasGwinn* /img/forum/go_quote.gif 
_L-e, Gordon, et. al., 

 With the AD1896 (the ASRC used in the UltraLock clocking system of Benchmark converters), the jitter of the incoming signal will not affect the conversion, it will merely (and inconsequentially) affect the amount of data in the input buffer between samples (as l-e correctly stated)._

 

Hi Elias,

 sorry for my pretty late reply here.

 Yeah, the buffer architecture of the DAC1 seems to be really impressive. What I like about your company Benchmark in general is the lack of voodoo. For instance it seems totally odd to me that some people try to reduce jitter at the transmission stage by changing cables, etc. while the problem arises during conversion and hence has to be fixed there as well.

  Quote:


  Originally Posted by *EliasGwinn* /img/forum/go_quote.gif 
_L-e, the RAM in the AD1896 is 512 words per channel._

 

Does the term "words" equal to samples in this case, hence about 11ms @ 44.1 kHz or am I missing something?

 Besides that I've a question in regard to "bad data" sent to D/A-converters like the Benchmark DAC1:

 Can this do any harm? I mean especially in the case of the USB-version which is served by a PC where the data integrity is more prone to errors due to the source (file sharing, etc.). Don't get me wrong, I'm not referring to transmission errors between the USB-port and the DAC1 for instance but "bad data" by itself. In the worst case it could be white noise or audio with intersample peaks sent to the converter. Do the analog parts care about outputting totally distorted and clipping sound at all?

 That would be really interesting to know since this can happen faster than one thinks (last time I forgot to tickle off the "play" option in Cool Edit while browsing through archives and suddenly "iiiiiiiiiiiiiiiiiiiiik" @ fullscale; however it was played through a cheap Transit USB, not the "baby" hehe).

 Cheers,

 little-endian


----------



## EliasGwinn

Quote:


  Originally Posted by *ted betley* /img/forum/go_quote.gif 
_Elias I hate to bother you but I tried j river and loved the sound through my Benchmark usb dac. However based on all the research I have done on the various sites it seems that folks prefer asio vs direct sound as an interface. I can get direct sound to work but not asio4all. Any advice on how to do that? I downloaded asio4all, followed the 9 page documantation, set audio output options to asio, chose asio4all and still no success. I get a 'direct show filters not available...'or something similar diagnostic and then my pc unselects the Benchmark as the ouput device (in the Asio4all panel) and it reverts back to direct sound._

 

Ted,

 My sincere apologies for not responding to this sooner. I just noticed it now... 
	

	
	
		
		

		
		
	


	




 Asio4All works with the DAC1, but it may be having an issue with J.River. I haven't worked with J River yet, so I cannot say for sure. I'll (hopefully) be testing J River soon, and I will post any and all results here as well as on the Benchmark Audio-Wiki.

 Thanks,
 Elias


----------



## EliasGwinn

Quote:


  Originally Posted by *joijwall* /img/forum/go_quote.gif 
_I follow the bit-transparancy discussions with manic interest, although some/most of it is over my head. 
 My listening experience using MacBook 10.5.2/iTunes 7.6.1/DAC1 USB compared with a very good Linn Majik CD is that I hear no or minimal difference. This is disturbing. Connecting DAC1 USB directly to my Majik CD makes the sound clearer and better in my ears. 
 I'd like my Mac/DAC to perform better! Why? How? News?_

 

As the other posters mentioned, the two should be exactly the same as long as the volumes are matched and the DSP functions in iTunes/CoreAudio are set properly.

 If you have any questions about how to set these parameters, please don't hesitate to ask.

 Thanks,
 Elias


----------



## EliasGwinn

Quote:


  Originally Posted by *little-endian* /img/forum/go_quote.gif 
_For instance it seems totally odd to me that some people try to reduce jitter at the transmission stage by changing cables, etc. while the problem arises during conversion and hence has to be fixed there as well._

 

How true...

  Quote:


  Originally Posted by *little-endian* /img/forum/go_quote.gif 
_Does the term "words" equal to samples in this case, hence about 11ms @ 44.1 kHz or am I missing something?_

 

You got it...

  Quote:


  Originally Posted by *little-endian* /img/forum/go_quote.gif 
_Besides that I've a question in regard to "bad data" sent to D/A-converters like the Benchmark DAC1:

 Can this do any harm? I mean especially in the case of the USB-version which is served by a PC where the data integrity is more prone to errors due to the source (file sharing, etc.). Don't get me wrong, I'm not referring to transmission errors between the USB-port and the DAC1 for instance but "bad data" by itself. In the worst case it could be white noise or audio with intersample peaks sent to the converter. Do the analog parts care about outputting totally distorted and clipping sound at all?_

 

No, this will not damage the DAC1. However, be careful not to damage your speakers with full scale noise. 

 Thanks,
 Elias


----------



## doctorcilantro

I'd join & start a thread here at J. River; some VERY knowledgeable folks that will help you out:

Media Center 12

 DC


----------



## little-endian

Elias, thanks again for your quick and professional answers.

 Since some use regular CD-players to playback dts via S/PDIF for instance, I almost expected that there shouldn't be any damage done. In this operation, the player's built-in DAC will output more or less white noise all the time and nobody really cares as long as the speakers and hence the listener's ears are fed by the analog output.


----------



## joijwall

After your replies, folks, I've been listening more. 
 My iTunes settings are correct, Sound/MIDI is set to 24/44,1, I've tried by ear to volume match. 
 But listening is difficult. I tried Gordon's AIFF-suggestion and had the feeling AIFF was better than Apple Lossless, but by so little. And I thought suddenly VLC didn't perform Cosi Fan Tutte as well as iTunes (used to be other way round), and Linn vs LinnDAC was dashing over the finish line today as one.

 Still, what I hope for is to get my MacBook/DAC1 USB to sound better than my Linn CD. I thought reading from the harddrive was superior to reading a CD, and I was pretty sure DAC1 was a better DAC than the Linn one. But I can't hear the great difference.

 So where did my hopes go wrong? Have I missed something?
 Thanks! Joachim


----------



## little-endian

Hi joijwall,

 actually getting an answer to your question is straightforward if one excludes any voodoo and proceeds rationally instead:

 Since the logic is much more powerful than our hearing abilities, a good question for the beginning is "what can affect the analog quality which is "reconstructed" on a fixed digital basis?".

 The right answer which can be proven is "data integrity and time".

 The first can be examined by recording the S/DPIF-output of your different sources and comparing the result with the original file or data from the CD. If there are no differences (besides the offsets of course), then there is only the variation in time left to explain sonic differences.

 And since the Benchmark DAC1 doesn't care of the last parameter due to its ultralock system (as advertised), pushing jitter artefacts far below the hearing threshold, you may lay back and relax becoming aware that imagination is something great but really no impartial counsel.


----------



## helian

Hi Elias,

 I just got a DAC1 USB. I'm using it with Linux, so I'd like to know if there is any plans on adding a linux section to Benchmark Audio-Wiki.



 Thanks


----------



## joijwall

"The first can be examined by recording the S/DPIF-output of your different sources and comparing the result with the original file or data from the CD. If there are no differences (besides the offsets of course), then there is only the variation in time left to explain sonic differences."
 I'm with you this far, little-endian. How do I record and compare S/DPIF-output and CD-source? What do you say about my presumptions that HDD reads data better than a CD player, and that DAC1 should convert extremely well? It's easy to hear difference between a Linn Majik CD and Linn Akurate CD. Shouldn't I hear some difference between Linn Majik CD and MacBook/DAC1? Thanks! Joachim


----------



## little-endian

Hi joijwall,

  Quote:


  Originally Posted by *joijwall* /img/forum/go_quote.gif 
_How do I record and compare S/DPIF-output and CD-source?_

 

Well, the effort actually depends on the data rates you're trying to compare. For instance to do that for 96kHz/24Bit or even 192kHz material, one will face mainly two problems: The equipment capable of recording that rates over S/PDIF, source capable of deliver such rates over S/PDIF and of course the old annoying topic copy protection.

 But let's stick with 44.1kHz/16Bit for now like it comes from regular audio-cds. The deal would be to record the player's output with a sound card featuring a S/PDIF input (~ 20 EUR) and compare that result with the files you get via DAE (Digital Audio Extraction) from your CD.

 If the output of your player is bitperfect (unfortunately not all are, sad but true) and the transmission is fine, both files (the recorded one and the "ripped" one) should match. There might be missing / repeated samples, also sporadical within the track (I encounter this with my Transit USB, still have to investigate the reason) but the proofe is still valid.

  Quote:


  Originally Posted by *joijwall* /img/forum/go_quote.gif 
_What do you say about my presumptions that HDD reads data better than a CD player, and that DAC1 should convert extremely well?_

 

It depends on what you mean by "reading better". On physical layer, every readout or storage is analog, hence varies in quality (for instance, when reading a CD a high-frequency signal is digitized and processed further) but at the end, what counts is if the "user data" (the raw data before the error correction stages may contain errors) is free of errors or not. There are debates about different amounts of jitter though. However, this doesn't affect the data integrity when within the threshold levels.

  Quote:


  Originally Posted by *joijwall* /img/forum/go_quote.gif 
_It's easy to hear difference between a Linn Majik CD and Linn Akurate CD. Shouldn't I hear some difference between Linn Majik CD and MacBook/DAC1? Thanks! Joachim_

 

If you use different players or DAC configurations, the sound may be different although they are fed by the same data, yes.

 Even if it might be bad for your stomach one hint at last: The undoubtful exclusive and well produced Linn players - and even if their drives are of highest quality materials - won't deliver you other data than the cheapest, rattly PC CD-ROM for ~ 15 EUR.

 Amazing, huh?


----------



## infinitesymphony

The Linn Majik CD costs $3,500 while the DAC1 USB costs $1,275. If they sound the same, doesn't that speak for the value of the DAC1?


----------



## Wavelength

Quote:


  Originally Posted by *little-endian* /img/forum/go_quote.gif 
_Hi joijwall,



 Well, the effort actually depends on the data rates you're trying to compare. For instance to do that for 96kHz/24Bit or even 192kHz material, one will face mainly two problems: The equipment capable of recording that rates over S/PDIF, source capable of deliver such rates over S/PDIF and of course the old annoying topic copy protection._

 

Guys,

 Really bits are not bits as we have found out. There is all kinds of things that make up sound. In the case of the computer why SHOULD the AIFF sound different from the Losless? They are identical files, heck the AIFF was created from the Lossless.

 Joijwall,

 Maybe you better give us the run down of your system in complete detail, including the ac and audio layout and maybe I can see something.

 We have actually found several user's trying to isolate computer power supplies from the audio. This is fine if the computer section floats the earth connection (center one on a 3 prong). But if it doesn't it causes a small ground loop that can steer the output to sound poor.

 Heck even to say that the USB cables can sound different. The best inexpensive cable is the Belkin Gold 2.0 USB. We actually found the Kimber with the ferrites on both ends to sound poor compared to the Belkin. But these are easy to remove and Ray has been made aware of this.

 Anyways spending hours looking at bits is basically a waste of time.

 Thanks
 Gordon


----------



## EliasGwinn

Gordon,

 I have to respectfully disagree with every point in this post. 

  Quote:


  Originally Posted by *Wavelength* /img/forum/go_quote.gif 
_Guys,

 Really bits are not bits as we have found out._

 

Bits _ARE_ bits. Data is data. Data sets are finite, discrete values that can be directly compared with certainty. Its just math.

  Quote:


  Originally Posted by *Wavelength* /img/forum/go_quote.gif 
_In the case of the computer why SHOULD the AIFF sound different from the Losless? They are identical files, heck the AIFF was created from the Lossless._

 

AIFF's are not usually created from Lossless-coded files. They are usually created from the raw data. If they sound different, its because the data (bits) have been changed.

  Quote:


  Originally Posted by *Wavelength* /img/forum/go_quote.gif 
_We have actually found several user's trying to isolate computer power supplies from the audio. This is fine if the computer section floats the earth connection (center one on a 3 prong). But if it doesn't it causes a small ground loop that can steer the output to sound poor._

 

It is a BAD idea to float the ground on a computer power cord. If the DAC1 and computer are plugged into the same power outlet, there will be little chance of a ground loop. BTW, if ground loops are a problem, it will cause a hum. If you don't hear a hum, you don't have to think about this. BTW, we've only had one case of a DAC1 USB with a hum, and this customer was using a laptop whose power cord had no ground plug.

  Quote:


  Originally Posted by *Wavelength* /img/forum/go_quote.gif 
_Heck even to say that the USB cables can sound different. The best inexpensive cable is the Belkin Gold 2.0 USB. We actually found the Kimber with the ferrites on both ends to sound poor compared to the Belkin. But these are easy to remove and Ray has been made aware of this._

 

If a USB audio device is sensitive to different USB cables, such that the sound of the device will change with each successive cable, perhaps the USB device is not built properly.

  Quote:


  Originally Posted by *Wavelength* /img/forum/go_quote.gif 
_Anyways spending hours looking at bits is basically a waste of time.

 Thanks
 Gordon_

 

Analyzing and understanding the effect that various sources have on the digital data is very important. Several sources will modify the digital data (often to horrible ends). This includes software and hardware. In these cases, one could spend hours trying to fix the problem by changing cables, power cords, internal components, auxiliary components, etc, etc, etc (ad nauseum), when the problem was simply an improper setting in a menu somewhere.

 Thanks,
 Elias


----------



## little-endian

Well, Elias was faster than me. I totally agree with his disagreement. 
	

	
	
		
		

		
		
	


	




 The digital technique may be one of the most ingenious one ever invented but is unfortunately also one of the most poorly unterstood. 
	

	
	
		
		

		
		
	


	




 I just want to add something here:

 If we speak about digital information, it even doesn't have to consist of bits actually. Every number system or alphabet is per se digital because it has a well defined, discret character set, where nothing can exist in between. Thus digital <> bits but digital --> bits for instance (*one* possible char set of infinite).

 When you read a book, you're indeed retrieving digital information since simple text is purely digital although it is always represented in an analog form (for instance the paper and ink whose quality may vary).

 Hence, digital data exists only on a logical level beyond physical inadequacy.

*That* is the big deal.

 Never forget that.

 End of philosophy.


----------



## Wavelength

Elias, Little-Endian;

 Guys go back and read the post!

 I suggested that he take a Lossless file and convert it to AIFF. The instructions were clear and easy. We have done extensive tests on this and yes the Lossless files on both PC and MAC are pure and the same and are a bit match to their uncompressed formats of WAV/AIFF.

 The question to you is... if the bits are bits then why don't they sound the same?

 Guys I have been draining my brain trying to find out why on the PC the applications all seem to sound different. If bits are bits then why? They are all the same files going to the same place.

 I don't think it's bits are bits not yet at least.

 Elias,

 Ground loops often occur because of the computer. If you take really high end advice then all the components would be floated except the preamplifier. That (I am not kidding here... story to follow) is ground to a 1/2" steel post in the ground at least 2 feet.

 Jonathan Valin got sited by Cincinnati Gas & Electric for this very setup. They came out and disconnected his entire power to his house.

 Most of the noise caused by ground loops in computer systems stem from the fact that all grounded devices need to be plugged into the same outlet. But most of these audio guys want to plug xyz into power conditioner A and then the computer into the wall. If there is a ground potential between the units then ground loop noise occurs.

 We found that if you used the 3 prong plugs on some early MacBooks and Intel Core Duo Mini's that ground loop would occur and the user needed to use a cheater plug to remove the ground.

 This is no big deal as the common ground will still exist between the USB dac, computer and the rest of the system. Unless you use like an opticis cable to isolate the computer from the dac.

 Thanks
 Gordon


----------



## Scrith

Quote:


  Originally Posted by *Wavelength* /img/forum/go_quote.gif 
_if the bits are bits then why don't they sound the same?_

 

They sound the same to me. With regular tests and with blind tests. Same with my wife. Have you tried blind testing or are you reiterating the opinions of many people that tend to echo the same opinion they read somewhere over and over again here on the internet?


----------



## little-endian

Hey Gordon,

 no need to get upset here.

 I have read your post - in opposite to you when it comes to mine I'm afraid. 
	

	
	
		
		

		
		
	


	




 You're invited to read my post #1428 to find the answer to your question.

 You should allow yourself to trust a professional engineer from a company which manufactures these devices and one who is indeed layman in comparison but has understood the most important basics pretty well (me) a bit more.

 Thanks,

 little-endian


----------



## Crowbar

Isn't defeating the ground connection on a computer illegal? A computer is a significant RF radiator.

 little-endian: he _is_ an engineer, and he _does_ manufacture DACs.


----------



## little-endian

Quote:


  Originally Posted by *Crowbar* /img/forum/go_quote.gif 
_little-endian: he is an engineer, and he does manufacture DACs._

 

Yes, of course I know. That's why I mentioned it.

 And without flattering too much, Elias' support is one of the best I've ever seen from a company - really excellent. Many, many thanks at this point again!


----------



## Crowbar

I was referring to Wavelength.


----------



## Wavelength

Quote:


  Originally Posted by *little-endian* /img/forum/go_quote.gif 
_Hey Gordon,

 no need to get upset here.

 I have read your post - in opposite to you when it comes to mine I'm afraid. 
	

	
	
		
		

		
			





 You're invited to read my post #1428 to find the answer to your question.

 You should allow yourself to trust a professional engineer from a company which manufactures these devices and one who is indeed layman in comparison but has understood the most important basics pretty well (me) a bit more.

 Thanks,

 little-endian_

 

LE,

 I did read that post and do agree with your statement. I also want you to realize that we have characterized and held listening test's at CES with dealers, media and other even competing companies.

 It was plain to hear that when converting a lossless file to another format (WAV/AIFF) which was uncompressed that there was a differential in sound that was easily heard. It more true in slower machines but we did the test on both OSX and Windows as both the MAC mini we had setup and the MacBook both had bootcamp installed with Vista Ultimate.

 The thing you further have to understand is this. With a product like the Benchmark that puts the data through a upsampler to remove the jitter there should be LESS of difference in sound.

 As I am sure from your alias (don't you hate these things, just say who you are) that you have some programming and therefore math background then I would suggest re-reading LaPlace and Forier work in math. This math was conceived in the early 1820's (why???) and is the basis for much of how digital audio works.

 But in the end Joijwall did hear a difference from the same file. Which means as bits are not bits as they would be identical in nature.

 Guys for years we have struggled with SPDIF and originally people would say bits are bits why does this digital cable sound different than that one. Now we know allot more of why that is.

 Missinformation is the biggest crime in Computer Audio. The idea that the KMIXER is bit perfect. Common.... There is not a person alive who has heard the KMIXER then bypassed it and heard the difference and said they where the same.

 If the KMIXER is truely bit perfect then... bits are bits correct?

 I have heard the truth, what about you?

 Thanks
 Gordon


----------



## sejarzo

Someone please explain to this dumb chemical engineer why we cannot have a definitive answer to all of this via a simple objective test that doesn't involve listening.....please?!?!?

 Why is it so hard to prove or disprove that KMIXER, or any other playback app or combination thereof, is bit perfect?

 Does no device exist that can take an S/PDIF input from a computer source and convert the bit stream back to a file to be compared to the original?


----------



## EliasGwinn

Quote:


  Originally Posted by *Wavelength* /img/forum/go_quote.gif 
_LE,
 I also want you to realize that we have characterized and held listening test's at CES with dealers, media and other even competing companies._

 

Can you please explain in detail the configuration for this listening test?

  Quote:


  Originally Posted by *Wavelength* /img/forum/go_quote.gif 
_It was plain to hear that when converting a lossless file to another format (WAV/AIFF) which was uncompressed that there was a differential in sound that was easily heard. It more true in slower machines but we did the test on both OSX and Windows as both the MAC mini we had setup and the MacBook both had bootcamp installed with Vista Ultimate._

 

This issue deals with the quality of format conversion within the media player. I make no claims in that regard.

  Quote:


  Originally Posted by *Wavelength* /img/forum/go_quote.gif 
_The thing you further have to understand is this. With a product like the Benchmark that puts the data through a upsampler to remove the jitter there should be LESS of difference in sound._

 

Can you explain this in more detail?

  Quote:


  Originally Posted by *Wavelength* /img/forum/go_quote.gif 
_But in the end Joijwall did hear a difference from the same file. Which means as bits are not bits as they would be identical in nature._

 

I don't want to speak for Joijwall, but I believe he was comparing the difference between his transport's internal DAC vs. the DAC1. However, I could be very wrong about that. 

  Quote:


  Originally Posted by *Wavelength* /img/forum/go_quote.gif 
_Guys for years we have struggled with SPDIF and originally people would say bits are bits why does this digital cable sound different than that one. Now we know allot more of why that is._

 

It is important to make a distinction between data integrity and transmission integrity.

  Quote:


  Originally Posted by *Wavelength* /img/forum/go_quote.gif 
_
 Missinformation is the biggest crime in Computer Audio. The idea that the KMIXER is bit perfect. Common.... There is not a person alive who has heard the KMIXER then bypassed it and heard the difference and said they where the same.

 If the KMIXER is truely bit perfect then... bits are bits correct?

 I have heard the truth, what about you?
_

 

I don't hear any difference with the DAC1 USB when I bypass kmixer. And I am alive. 

 There may be some confusion about 'cause and effect' in some of your conclusions. However, I am not discrediting your opinions. We may have to look at the empirical methods a little bit closer.

 Thanks,
 Elias


----------



## EliasGwinn

Quote:


  Originally Posted by *sejarzo* /img/forum/go_quote.gif 
_Why is it so hard to prove or disprove that KMIXER, or any other playback app or combination thereof, is bit perfect?

 Does no device exist that can take an S/PDIF input from a computer source and convert the bit stream back to a file to be compared to the original?_

 

A test exists with Audio Precision (AP) equipment. However, our results with these tests conflict with the testimony of one of Kmixer's programmers. The AP test indicated that the playback through Kmixer is truly bit-for-bit perfect, exact, transparent. The programmer says this is not possible because Kmixer performs a floating-point operation on all audio. 

 There are some people who have done data comparisons, with conflicting results. Some have determined that kmixer is bit-transparent, and some have determined that it is not.

 There may be (and probably are) several variables that are not being taken into account. We may never know all the variables, simply because of the many layers and generations of computer code that makes the whole system.

 So, we've performed FFT analysis' of the signal pre and post kmixer as a way to determine its affect. There is no discernible affect that we've noticed. And, please understand that we would shout loud and often if we did notice a problem. We would be telling all DAC1 USB/PRE owners to avoid kmixer so that the system sounded as good as possible. However, we have no reason to believe that there is a problem.

 Thanks,
 Elias


----------



## sejarzo

So...I thus take it that a similar test using the AP equipment could be also used to determine whether a playback chain all the way from a lossless file through a playback app, drivers, OS, and some external device is bit perfect or not.

 And it also seems to me that if a given configuration of hardware/drivers/OS is bit-perfect with one (or more?) playback apps on .wav files, and then is found to be not bit-perfect on lossless compressed files, the errors are introduced in the playback app/plug-ins--but that could always be the case, right? There sure aren't any guarantees that freeware code is error-free.

 This seems to be ripe for a truly independent test at a university.


----------



## KarateKid

I'm very interested in spending some money on a balanced amp setup. I already have balanced hd650 cables. However there are just so many choices out there in terms of balanced amps, none of which are accessable locally. The popular balanced amps out there have been out for quite some time so I'm not sure if they have the latest things in there like the lm4562s. There's so many variables like opamps vs discrete designs that is needs to be considered. 

 Do you think one needs to spend a lot of xlr interconnects? I know Benchmark aim to be transparent as possible when it comes to audio, I want to preserve that signature. Anyone got any tips or advice on xlr IC or a good balanced amp and whether I should go with opamps based or a discrete based?


----------



## sejarzo

Sure, build your own balanced IC's out of Mogami balanced mic cable and Neutrik XLR connectors. The recorded signal has passed through that already, and if it didn't degrade the signal between the mic and the preamp, or the preamp and the A/D, well????


----------



## Ross MacGregor

Quote:


  Originally Posted by *Wavelength* /img/forum/go_quote.gif 
_LE,
 It was plain to hear that when converting a lossless file to another format (WAV/AIFF) which was uncompressed that there was a differential in sound that was easily heard. It more true in slower machines but we did the test on both OSX and Windows as both the MAC mini we had setup and the MacBook both had bootcamp installed with Vista Ultimate._

 

Hi Gordon,
 There are two variables that come to mind that may be affecting the test:
 A) Is the player properly transforming the lossless formated data into the original data? Is it making approximations or taking shortcuts?
 B) Is decoding the lossless format taxing the computer and causing transmission problems?


----------



## EliasGwinn

Quote:


  Originally Posted by *KarateKid* /img/forum/go_quote.gif 
_Do you think one needs to spend a lot of xlr interconnects? I know Benchmark aim to be transparent as possible when it comes to audio, I want to preserve that signature. Anyone got any tips or advice on xlr IC or a good balanced amp and whether I should go with opamps based or a discrete based?_

 

I don't think you need to spend more then $100 per cable. If you are using balanced cabling, I recommend Starquad analog cabling. Several manufacturers make starquad cable - Canare, Mogami, etc. I also recommend Neutrik connectors.

 There are many reasons why we recommend not employing balanced headphone drive systems, including significant distortion. So I have no suggestions except to avoid the topology all together. 

 Thanks,
 Elias


----------



## KarateKid

Quote:


  Originally Posted by *EliasGwinn* /img/forum/go_quote.gif 
_I don't think you need to spend more then $100 per cable. If you are using balanced cabling, I recommend Starquad analog cabling. Several manufacturers make starquad cable - Canare, Mogami, etc. I also recommend Neutrik connectors.

 There are many reasons why we recommend not employing balanced headphone drive systems, including significant distortion. So I have no suggestions except to avoid the topology all together. 

 Thanks,
 Elias_

 

Do you mean I should not bother getting a balanced headphone amp with balance outs? Or one shouldn't use the back of the DAC1's xlr.


----------



## EliasGwinn

Quote:


  Originally Posted by *KarateKid* /img/forum/go_quote.gif 
_Do you mean I should not bother getting a balanced headphone amp with balance outs? Or one shouldn't use the back of the DAC1's xlr._

 

It is not a good idea to drive _headphones _with balanced cables, no matter what source. It causes significant distortion. I can explain this in detail, or you can find my posts on the subject earlier in this thread (good luck).

 Don't confuse with driving equipment with balanced cables. That is not a problem. But the idea of balanced headphones has major problems.

 I'll be back here on Head-Fi Wednesday. Have a great weekend!!

 Thanks,
 Elias


----------



## KarateKid

Quote:


  Originally Posted by *EliasGwinn* /img/forum/go_quote.gif 
_It is not a good idea to drive headphones with balanced cables, no matter what source. It causes significant distortion. I can explain this in detail, or you can find my posts on the subject earlier in this thread (good luck).

 Don't confuse with driving equipment with balanced cables. That is not a problem. But the idea of balanced headphones has major problems.

 I'll be back here on Head-Fi Wednesday. Have a great weekend!!

 Thanks,
 Elias_

 

I see.

 But wasn't one of the main attractions of driving headphones with balanced cable being one cable going to one cup thus eliminating any sort of cross talk? Do you think there's a better connection than single ended or is it just a myth?


----------



## KarateKid

Quote:


  Originally Posted by *EliasGwinn* /img/forum/go_quote.gif 
_It is not a good idea to drive headphones with balanced cables, no matter what source. It causes significant distortion. I can explain this in detail, or you can find my posts on the subject earlier in this thread (good luck).

 Don't confuse with driving equipment with balanced cables. That is not a problem. But the idea of balanced headphones has major problems.

 I'll be back here on Head-Fi Wednesday. Have a great weekend!!

 Thanks,
 Elias_

 

Sorry to keep you. I hope you have a good weekend. However, you did not recommend using balanced headphones straight out of the DAC1, is it better to use a balanced headphone amp, even if it's still not ideal as you've explained?


----------



## emmodad

Quote:


  Originally Posted by *KarateKid* /img/forum/go_quote.gif 
_Sorry to keep you. I hope you have a good weekend. However, you did not recommend using balanced headphones straight out of the DAC1, is it better to use a balanced headphone amp, even if it's still not ideal as you've explained?_

 

1/ read the thread

 2/ use the search function


----------



## little-endian

Quote:


  Originally Posted by *Wavelength* /img/forum/go_quote.gif 
_LE,

 I did read that post and do agree with your statement. I also want you to realize that we have characterized and held listening test's at CES with dealers, media and other even competing companies._

 

If you recognized and agree, it should be clear that listening tests are not even necessary or helpful to follow up what I referred to. Logic's nature is to lead to the same result if one is following the same steps no matter *who* follows them.

  Quote:


  Originally Posted by *Wavelength* /img/forum/go_quote.gif 
_It was plain to hear that when converting a lossless file to another format (WAV/AIFF) which was uncompressed that there was a differential in sound that was easily heard. It more true in slower machines but we did the test on both OSX and Windows as both the MAC mini we had setup and the MacBook both had bootcamp installed with Vista Ultimate._

 

There are two possible explanations while the first one is by far more likely: Either something is really going wrong during the realtime conversion from the compressed data to PCM (restored data doesn't match the original one) or the conversion process affects the output's jitter the player device isn't taking care of good enough (catchword: "time is the enemy"). 

  Quote:


  Originally Posted by *Wavelength* /img/forum/go_quote.gif 
_The thing you further have to understand is this. With a product like the Benchmark that puts the data through a upsampler to remove the jitter there should be LESS of difference in sound._

 

If Benchmark's claims are correct, any jitter won't be *less* audible but at -130dBFS definately *not* audibly at all.

  Quote:


  Originally Posted by *Wavelength* /img/forum/go_quote.gif 
_As I am sure from your alias (don't you hate these things, just say who you are) that you have some programming and therefore math background then I would suggest re-reading LaPlace and Forier work in math. This math was conceived in the early 1820's (why???) and is the basis for much of how digital audio works._

 

Well from my alias, I think I'd have to vote for WAV instead of AIFF. 
	

	
	
		
		

		
			





 When it comes to digital audio, I rather think about Nyquist / Shannon. However, to understand the fact that characters are just characters (and thus bits just bits of course), none of these principles are needed but pure logic.

  Quote:


  Originally Posted by *Wavelength* /img/forum/go_quote.gif 
_But in the end Joijwall did hear a difference from the same file. Which means as bits are not bits as they would be identical in nature._

 

Wrong. It means that it was either imagination or the data didn't arrive on time, in fact resulting in different sounding audio.

  Quote:


  Originally Posted by *Wavelength* /img/forum/go_quote.gif 
_Guys for years we have struggled with SPDIF and originally people would say bits are bits why does this digital cable sound different than that one. Now we know allot more of why that is._

 

Well, I have my doubts that the "mass" really knows what's going on. It can be proven that a cable doesn't matter at all when it comes to transmissions as long as the jitter and attenuation is low enough to allow the receiver to properly detect the signal. In the case of real-time applications like D/A conversion, it depends on the specific device. It's not the cable's job to keep the jitter as low as possible for unjustified amounts of money if a simple buffer for a few cents could have the same effect.

  Quote:


  Originally Posted by *Wavelength* /img/forum/go_quote.gif 
_Missinformation is the biggest crime in Computer Audio._

 

That's why I'm taking the time to reply here. However I feel the battle has been already lost long ago.

  Quote:


  Originally Posted by *Wavelength* /img/forum/go_quote.gif 
_The idea that the KMIXER is bit perfect. Common.... There is not a person alive who has heard the KMIXER then bypassed it and heard the difference and said they where the same.

 If the KMIXER is truely bit perfect then... bits are bits correct?_

 

I don't know much about Windows' kmixer.sys, but I know if my data from the S/PDIF-output matches the DAE-results or not and that's enough for me. If one connects the Benchmark DAC1 USB directly to the PC, it's of course by far more difficult to check if the data will really remain intact. You'd either simply have to believe Benchmark's tests or perform your own by sniffing the data between the USB-port and the DAC.

 Thus, the S/PDIF connection has one big advantage here: one can record the output from the PC soundcard or a stand-alone player with a cheap recording device and compare. If the data matches, one can be sure that the DAC1 or any other DAC will get the same data as well (assuming the receivers are of the same "recognition quality") and satisfy the paranoia.


----------



## KarateKid

Quote:


  Originally Posted by *emmodad* /img/forum/go_quote.gif 
_1/ read the thread

 2/ use the search function_

 

Okay buddy, why do you care? He said he does not recommend balanced headphone cables out of the dac1 or in general however I wanted to know if the sound is better out of a balanced AMP instead of directly out of the back of a DAC1. So cool it.


----------



## emmodad

Quote:


  Originally Posted by *KarateKid* /img/forum/go_quote.gif 
_Okay buddy, why do you care? He said he does not recommend balanced headphone cables out of the dac1 or in general however I wanted to know if the sound is better out of a balanced AMP instead of directly out of the back of a DAC1. So cool it._

 

"so cool it"?!?! my, how droll.

 poor Elias has responded directly to the topic of balanced cans on DAC1 XLR outs before, IIRC several times. and many, many others have posted, expressed opinions, discussed technical pros and cons....

 the subject of your query has been addressed to death a/ in this thread and b/ in many, many others discussing impressions / appropriateness / technical issues / subjective impressions wrt use of DAC1 balanced outs to directly drive balanced cans.

 so the suggestions wrt reading and searching do stand. BION, all you have to do is invest a small amount of your own energy and time.


----------



## joijwall

Back today in the forum and A LOT to think about. Thanks all! Please let me clarify: 
 a) I took a Lossless file (made in iTunes from CD), converted it to AIFF, and compared the two files played from iTunes and think I heard a difference.
 b) Using my Linn CD as transport with DAC1 was better than the Linn CD. At least at first, but after my last try I don't know anymore.
 c) DAC1 has of course great value, no question about it! But I actually expected MacBook/iTunes/DAC1 to beat the Linn Majik CD clearly.
 These are my thoughts:
 1) Benchmark has tested earlier MacOS Tiger/iTunes with bitperfect result, but the Leopard tests are not yet completed. I cannot be sure my setup is bitperfect.
 2) There could be a bit-difference between Lossless and AIFF/WAVE, depending on the player used. In my case MacBook/iTunes might not be able to decode perfectly on the fly. You seem all to agree on that. 
 3) If everything is indeed bit-perfect from the source all the way to DAC1, the DAC1 analog output quality is what could effect the sound. What I understand, Benchmark suggests a removal of the headphone-mute-function for best quality? 
 I'm pretty sure though the DAC1 outputs is at least as good as the ones in Linn Majik CD.
 4) I don't know much about ground loops, but I could try optical or run on batteries and listen. I've also moved my music files to a separate HDD not long ago, maybe that has some influence also. 
 5) Regarding CD-players, are there other important things involved than reading from the disc, D/A-converting and the electronic components (including analog outputs)? I thought perfecting these three gave better sound (and higher price). How does DAC1 perform in these areas? I listened to Linn Klimax DS ($30000 HDD/Streamer/DAC-combo) and it sounded fantastic. Which led me in on the DAC1 path by the way.
 6) Listening comparison is really difficult and we're easily affected by expectations.
 So, in the end, I'm interested in three things: 
 - Give my DAC1 the original source bits
 - Make sure DAC1's analog output is well taken care of
 - Relax and let the music speak
 /Joachim


----------



## Wavelength

Quote:


  Originally Posted by *EliasGwinn* /img/forum/go_quote.gif 
_Can you please explain in detail the configuration for this listening test?

 This issue deals with the quality of format conversion within the media player. I make no claims in that regard.

 Can you explain this in more detail?_

 

Elias, this was covered earlier in the thread but sure. We had 2 idential Cosecant v3 dacs, both MACs (2.2GHZ/2GB/250gb MacBook Core 2 Duo, MacMini 1.6ghz Core Duo 2gb/250gb) connected via Opticis cables and then using idential lengths of Nirvana Audio SX cable to my Royal Preamplifier and Cardinal Ag amplifiers.

 We took a group of 5 songs on each platform (Vista Ultimate J River and iTunes OSX 10.5.1) that were in lossless format and converted these to WAV (PC) and AIFF (iTunes). So in a sense they are identical. Same way I told Joijwall to do this.

 We played the 5 songs one in lossless then using the AIFF/WAV in eight fomations. MacBook to Mini and then MacBook to MacBook, Mini to Mini both with Vista and iTunes.

 It was clear that more spatial information was apparent in the non-lossless files. It was more apparent on the slower machine (mini).

 After the show we took a 8 core 3GHZ MacPro with lossless and AIFF files and found there to be no difference. So it is assumed by this that slower machines have problem decoding the files on the fly. Someone told me that the original lossless format on the Mac was done with the Altivec process and that the Intel base machines don't fair as well.

 Maybe why my G4 still sounds very good to me.

  Quote:


 I don't want to speak for Joijwall, but I believe he was comparing the difference between his transport's internal DAC vs. the DAC1. However, I could be very wrong about that. 
 

Well that is what he was asked to do and that is how he responded. 

  Quote:


 It is important to make a distinction between data integrity and transmission integrity. 
 

Granted and that is why I don't feel recording a stream will give you total detail. If for example as I have said that XP does not preform as well as vista does because the USB drivers are not as good. This does not mean the recording of the stream would be any different. There is TIME and DATA too consider, especially in your case with Adaptive mode.

  Quote:


 I don't hear any difference with the DAC1 USB when I bypass kmixer. And I am alive. 
 

Common be serious... If that was the case then why even tell people how to bypass it? 

 As for the KMIXER being bit perfect, I have run tests on the Prism and we can truely tell that the LSB is changing which causes changes in the THD and the Spectral response.

 This is the same results Microsoft came up with and they use the AP and spent years figuring out how to get rid of the KMIXER so all's I can say is who you going to believe?

 Thanks
 Gordon


----------



## little-endian

Quote:


  Originally Posted by *Wavelength* /img/forum/go_quote.gif 
_After the show we took a 8 core 3GHZ MacPro with lossless and AIFF files and found there to be no difference. So it is assumed by this that slower machines have problem decoding the files on the fly._

 

Sorry, but if such systems shouldn't be able to decode a "simple" lossless file in realtime, I would be seriously worried about them.

 Take a old, dusty P1 @ 133 MHz and a Transit USB for instance and you'll have your data 100% restored in realtime, I'll bet you. Today's processors do this almost in standby.


----------



## Crowbar

Wavelength, are you sure you're getting identical bits? If you are, and you use asynch so there's no interface jitter at all, then I also can't see why it makes a difference.

 There's a simple test that can be used to check. Run a pseudorandom number generator on the DAC controller and send the same data from the PC, then compare the streams for match in the DAC (obviously pseudorandom number generators are deterministic so the sequence will match if you use the same seed).

 I've found that sometimes sending between two computers with the second emulating a USB Audio sink will give errors (though the first one is a pretty old computer so it could be just its lousy USB hardware). The omission of error correction from the standard definitely was a bad idea.


----------



## joijwall

Sorry Gordon, here's my setup:
 Linn Majik CD, Linn Exotik Pre, Linn C2200 Amp (all grounded from same outlet), Inoaudio Pi60 speakers. Linn Silver interconnects, Lejonklou powerchords.
 I want to beat the CD using a MacBook 2,16GHz/2 GB/120 GB MacOSX 10.5.2/iTunes 7.6.2 (all latest)+DAC1 USB using the Benchmarks USB cable and Linn silver interconnects. 
 Seems like a fair competition to me, and I'd like DAC1 to win big.
 /Joachim


----------



## joijwall

By the way, how is it possible to test the bit-transparancy at all on the Mac? When directing the output to DAC1 in Audio/Midi setup, I can choose sample rate, but I cannot set wordlength, it is always 2ch-24bit. 
 Doesn't this mean bits sent to USB is changed? I mean, the original CD is 44,1/16, but the USB output is 44,1/24, thus we cannot compare the bits unless we convert it back after the USB?
 If I understand Benchmarks wiki correctly, the 16->24 conversion is done by iTunes. Will a 16->24 conversion change the "music" information in any way, resulting in DAC1 presenting a different analog output than if the original 44,1/16 was feeding it?
 In my case, I guess DAC1 gets a 44,1/24 from my MacBook, but a 44,1/16 from the optical output on my Linn Majik CD. 
 In short: Since DAC1 wants the 24 bits, and setup forces iTunes to convert 16->24, don't we need to check for bit-transparancy somewhere inside DAC1?
 /joijwall


----------



## Tarkovsky

No.

 Resampling and word length change are entirely different. You can store the exact same number in 4 digits as you can two. For example. 99 and 99.00.

 Using 24bit is always a benefit.


----------



## Wavelength

Quote:


  Originally Posted by *little-endian* /img/forum/go_quote.gif 
_Sorry, but if such systems shouldn't be able to decode a "simple" lossless file in realtime, I would be seriously worried about them.

 Take a old, dusty P1 @ 133 MHz and a Transit USB for instance and you'll have your data 100% restored in realtime, I'll bet you. Today's processors do this almost in standby. 
	

	
	
		
		

		
		
	


	


_

 

LE,

 We did the samething on a PC, read the post I listed above. The same thing happened with WAV and Lossless files there.

 What I am saying is this. You can capture the data from each of these compare them and they will be the same as they should because when compared as files they are.

 BUT... when listening it's easy to tell difference on a good system. This system was voted by SoundStage as the best demo room at CES. Believe me I can't afford this stuff and I am building it. The equipment in that room retailed for like $85k not including cables, racks, power conditioner and computer stuff.

SoundStage! Network Las Vegas 2008 Special Show Site - www.AudioVideoShows.com

 If you have a clear system and have the skills to listen you can hear a difference.

 I have a ton of degrees but as I said before... my more than 40 years of being a musician decide more than anything else how something is going to sound.

 Thanks
 Gordon


----------



## Wavelength

Quote:


  Originally Posted by *joijwall* /img/forum/go_quote.gif 
_By the way, how is it possible to test the bit-transparancy at all on the Mac? When directing the output to DAC1 in Audio/Midi setup, I can choose sample rate, but I cannot set wordlength, it is always 2ch-24bit. 
 Doesn't this mean bits sent to USB is changed? I mean, the original CD is 44,1/16, but the USB output is 44,1/24, thus we cannot compare the bits unless we convert it back after the USB?
 If I understand Benchmarks wiki correctly, the 16->24 conversion is done by iTunes. Will a 16->24 conversion change the "music" information in any way, resulting in DAC1 presenting a different analog output than if the original 44,1/16 was feeding it?
 In my case, I guess DAC1 gets a 44,1/24 from my MacBook, but a 44,1/16 from the optical output on my Linn Majik CD. 
 In short: Since DAC1 wants the 24 bits, and setup forces iTunes to convert 16->24, don't we need to check for bit-transparancy somewhere inside DAC1?
 /joijwall_

 

Joijwall,

 In the enumeration (USB setup between the computer and the endpoint device) of the USB DAC1 it declarles all data to be 24 bits. Actually the padding can happen anywhere. What padding means is that 8 bits of 0 is added to each sample to make it 24. There is no upsampling preformed here in this case.

 Elias can answer this... but I think what they did was take the output of the TAS1020 (the USB Audio controller) in I2S format to an SPDIF transmitter. They then could feed the USB data and take the SPDIF input into their Ap analyzer and see if it was bit perferct.

 So in a sense it would be hard to check bit perfect data with 16 going in and 24 out. You would have to write a program to strip the padding before comparing it.

 Thanks
 Gordon


----------



## Wavelength

Quote:


  Originally Posted by *joijwall* /img/forum/go_quote.gif 
_Sorry Gordon, here's my setup:
 Linn Majik CD, Linn Exotik Pre, Linn C2200 Amp (all grounded from same outlet), Inoaudio Pi60 speakers. Linn Silver interconnects, Lejonklou powerchords.
 I want to beat the CD using a MacBook 2,16GHz/2 GB/120 GB MacOSX 10.5.2/iTunes 7.6.2 (all latest)+DAC1 USB using the Benchmarks USB cable and Linn silver interconnects. 
 Seems like a fair competition to me, and I'd like DAC1 to win big.
 /Joachim_

 

Joachim,

 One thing to try first is to disconnect the MagSafe connector off the MacBook and play it and then play it with it connected.

 I suggest 2 cable changes... I would get a Belkin Gold USB 2.0 cable. These run about 10$ and are allot better than the stock which I think is still sitting in my box. I would not use silver cables between the bechnmark and the Linn. Linn prides themselves in system concept. In the case of the Benchmark I would use a good copper cable from Kimber or better yet Nirvana Audio.

 I would set the Audio Midi settings to 24/44.1 don't let the upsamplers work.

 Thanks
 Gordon


----------



## Wavelength

Quote:


  Originally Posted by *Crowbar* /img/forum/go_quote.gif 
_Wavelength, are you sure you're getting identical bits? If you are, and you use asynch so there's no interface jitter at all, then I also can't see why it makes a difference.

 There's a simple test that can be used to check. Run a pseudorandom number generator on the DAC controller and send the same data from the PC, then compare the streams for match in the DAC (obviously pseudorandom number generators are deterministic so the sequence will match if you use the same seed).

 I've found that sometimes sending between two computers with the second emulating a USB Audio sink will give errors (though the first one is a pretty old computer so it could be just its lousy USB hardware). The omission of error correction from the standard definitely was a bad idea._

 

Crowbar,

 First there is no golden dagger for getting rid of all the jitter. Sure it is much easier with ASYNC but I don't think that has anything to do with our conclusion.

 I did have a USB Analyzer there and we did verify no USB errors occured. While this is true at the line level it is possible at the controller (TAS1020) that an error occured. I have tested this in the lab with the emulator attached with break points and counters on errors. But I think it's still has too do with speed since it happens in both PC and MAC land.

 I have been asked to design an ASYNC USB to SPDIF converter for another company I may start with Neil Sinclair (Theta). This may shed more light on the subject as soon as I can get to it.

 Thanks
 Gordon


----------



## Crowbar

I'm just trying to understand what can cause a difference. An asynch interface is no different conceptually than say transferring the audio file over the Internet. Yet I'm sure no one will claim if it goes over servers ABC it will sound different than if it goes over servers XYZ. If it's not the bits, then what? Perhaps somehow the input timing jitter is cross-coupled in a non-obvious way to other circuits. Or just RF crap coming in over the USB cable is not perfectly filtered.

 What's the point of asynch USB to S/PDIF? The whole benefit of USB is that it avoids S/PDIF's problem. This only makes sense if there's a clock line from the DAC to the converter driving its timing.


----------



## Ross MacGregor

While I have seen just about every aspect of the DAC1 discussed here. I haven't seen any discussion of the USB error rate. The DAC1 and most other USB DACs are not performing any error correction (please correct me if I'm wrong Elias) so USB errors can be detrimental to your DACs performance.

 I've found claims that the USB error rate is very low - meaning you can expect to see one error over several days of continuous use.

 I am wondering if Elias and Wavelength can share with us what error rates they have been able to measure with their devices in the lab and what factors can effect the error rate (cable quality, cable length, overloaded CPU, USB drivers, etc). I made note that Wavelength has measured significant errors when using some USB extension cables.

 Thanks,
 Ross

  Quote:


  Originally Posted by *Wavelength* /img/forum/go_quote.gif 
_I did have a USB Analyzer there and we did verify no USB errors occured. While this is true at the line level it is possible at the controller (TAS1020) that an error occured._


----------



## sckvee

"kmixer truth" if one notice that it is using floating-point operation might assume that kmixer can never be bit trasparent, Incorrect kmixer is bit-trasparent when used properly Elias Gwinn tests are correct, setting Volume control on kmixer Fully open will mathematically not change the sample values you just need to disable Microsoft software synth not to affect the results.

 I Know, i know but why direct kernel streaming sounds better, Playing files on WMP or I Tunes or whatever software that is using Direct Sound Interface will sound bad.
 Direct X is designed for use in low latency situation like video playback, video games and similar needs it can skip, alter predict samples and that has the similar coloration signature of a jitter induced distorsion even it is not showed on a test geat like it is still there.
 Cure ? use Multimedia Extensions Interface "MME" on your player it is absolute bit-transparent on the majority of the softwares it skips internal volume control and ties the control to kmixer volume control.

 You cannot evaluate USB DAC type of gear performance on Windows Vista system, Vista is unable to disable internal resampling so you get 2 antialias filters in a row, one from OS one from DAC off course best setting in Vista is maximum sample rate that dac can support (for the non computer geaks the setup dialog is in control panel -sounds)

 About the duel between Computer and Disk Transport sorry to disrupt your myth most of the cd trasports even the cheap ones makes no mistakes and you can be sure that computers dont make mistakes (1+1 will always be 2 never 3) , everybody screams now but different cables sound different, different transports sound different even different computers will sound different that is a FACT.
 This has been said a million times im gonna say it again it is all about trasmission timing problem and high frequency noise shifting the clock frequency, take a relative cheap cd tansport and expensive one (it has to be native cd transport not a dvd/sacd modern device) record spdif in to a pc with high end professional grade sound card and record the same track from the same disk you will get 2 bit to bit correct tracks why there is a difference offcourse timing errors now connect the high end studio pc to the dac it is gonna sound even worse and the high end sound card has a verry stable clock and low jitter spdif output why is it sound worse ?
 Many of the bad sound quality of a computer is tied with the high frequency noise that millions of switches kickin in and out at a few Gigahertz creating noise that outputs at every cable connected to the computer disturbing the dac clock even if it is completly isolated it still receive the signal remember this is in Ghz range noise.
 This comes to the question why is lossless sounding bad when beign played at realtime and sound the same when decoded first then played as wav file compared to the original wav rip previosly mention by J. Gordon Rankin from 
 Wavelength Audio, offcourse J. Gordon is not imagining things deference is audible it has also been said that 2Ghz system playing lossless sound worst than playing the decoded wav/aiff file from the lossless file that is playing at realtime and as the processing power is climbing the sound gets closer to the original file, this phenomenon is caused by the cpu power supply (not the main power suppy that you connect to the AC) that lies on the motherboard that stabiles and delevers cpu current, this is very efficient switchin power supply that is build for economy not high end so when cpu puts pressure on the power supply it spits out VHF noise excessively and the weaker ones even emit high pitch audible whistle that can be heard in a quiet room, when you use a system with higher frequency cpu the overhead on the cpu is less and the need for less power drain is releaving the strain on the cpu power supply plus usualy the faster cpu's are paired with better motherboards and the radiation effect is less apparent.

 Recomendation to J. Gordon Rankin is to test a machine with uncompressed file wile cpu is loaded with mathematical operation to see if there is an effect and to try to test a diffieren types of motherboards (2 phase 3 phase 8 phase types) and try to shield the dac form this emitions i think that the problem comes from the the cpu power supply not from the cpu itself.

 Greetings from Europe cheers.



 Note: Again, the cpu power supply is the one that lies on the motherboard not the main power supply in the metal case.


----------



## Terje

Quote:


 Recomendation to J. Gordon Rankin is to test a machine with uncompressed file wile cpu is loaded with mathematical operation to see if there is an effect and to try to test a diffieren types of motherboards (2 phase 3 phase 8 phase types) and try to shield the dac form this emitions i think that the problem comes from the the cpu power supply not from the cpu itself. 
 


 Do you think your theory is correct when using a RJ45 connection-ethernet cable-RJ45-DAC(SB3/Transporter)?
 Maybe DAC1 should have an RJ45 connection?


----------



## sckvee

Squeezebox is not a DAC its a whole system, think of it like 

 Transporter to be iPod on steroid's.

 Receiver/Duet to be low power Notebook with a nice soundcard.


 The problem that J. Gordon was talking about when playing lossless files, on Slim Devices gear using internal dac will be constant and it is gonna be dismissed as existing because you can't hear the effect that it has on the sound, you just exept it as a part of it.
 Using SB digital output to dac supposed to be theoretically better than pc to dac considering the fact that ARM processors draw less power and create less noise in the system then Intel cpu's never really try it my self, people that i have spoke that try it using Benchmark Media USB dac have not notice any difference i'll borow one and try it myself. I think its not logical to power up a PC and a SB to listen to music when i can just power up a PC ?

 Cheers .


----------



## Crowbar

If this is really the case, then galvanic isolation together with good shielding and proper filtering at tracks crossing the shield should take care of it.


----------



## music_man

opinions please,

 if i go envy24hts(toslink)>dac1>amp will that sound markedly better than envy24hts(line)>amp? all considered? the wolfson is a good dac and i don't really see the dac1 making much use of the noisy computer.

 music_man


----------



## Wavelength

Quote:


  Originally Posted by *Crowbar* /img/forum/go_quote.gif 
_I'm just trying to understand what can cause a difference. An asynch interface is no different conceptually than say transferring the audio file over the Internet. Yet I'm sure no one will claim if it goes over servers ABC it will sound different than if it goes over servers XYZ. If it's not the bits, then what? Perhaps somehow the input timing jitter is cross-coupled in a non-obvious way to other circuits. Or just RF crap coming in over the USB cable is not perfectly filtered.

 What's the point of asynch USB to S/PDIF? The whole benefit of USB is that it avoids S/PDIF's problem. This only makes sense if there's a clock line from the DAC to the converter driving its timing._

 

Crowbar,

 Really at this point I am not sure where the problem lies. The good news is hard drives are cheap so save or convert your lossless to WAV/AIFF and avoid the problem.

 But remember with USB there is no jitter because there is no clock. Well at least there is no jitter in the interface itself like there is with SPDIF. Jitter in SPDIF happens because the data is wrapped inside the clock. In USB and especially ASYNC mode the jitter becomes only that which is "intrinsic" to the controller, clock and dac.

 Actually running a clock from the dac to the transport does not solve the jitter problems. The effect of slew rate and cable distance of clocks exceeding 10MHz will not resolve this. Anyways the SPDIF still wraps the data around a clock. Guido Tent of TentLabs has a noval approach to the problem. Since sending the clock is a bad idea he puts a VCXO (variable clock) inside the transport and puts the PLL in the DAC and only sends a voltage.

 But really if you want to fix SPDIF you have to get rid of the idea all together and ATAPI the data off the disk like a computer does then work out something like USB, Ethernet or Firewire to the dac which is basically back to the computer in the first place.

 Thanks
 Gordon


----------



## Wavelength

Quote:


  Originally Posted by *Ross MacGregor* /img/forum/go_quote.gif 
_While I have seen just about every aspect of the DAC1 discussed here. I haven't seen any discussion of the USB error rate. The DAC1 and most other USB DACs are not performing any error correction (please correct me if I'm wrong Elias) so USB errors can be detrimental to your DACs performance.

 I've found claims that the USB error rate is very low - meaning you can expect to see one error over several days of continuous use.

 I am wondering if Elias and Wavelength can share with us what error rates they have been able to measure with their devices in the lab and what factors can effect the error rate (cable quality, cable length, overloaded CPU, USB drivers, etc). I made note that Wavelength has measured significant errors when using some USB extension cables.

 Thanks,
 Ross_

 

Ross,

 Here are some of the things that I would consider variables for errors on the USB link:

 1) Cable length, best to use a good USB 2.0 cable at less than 3m. The Belkin gold 2M is what I ship with most of my dacs. It is very good cable for the money.

 2) USB port: On PC's run USBView and see where the port is connected. On the MAC run the System Profiler. Ports connected directly to the USB Host controller are your best bets. NEVER use a USB port on the front of your computer. They are wired and have the worst results. Most host controllers have 2, 4 or 7 ports. Remember your keyboard, mouse and blue tooth are running off the USB host controller. We found on new iMac that one of the ports is shared by all internal stuff and does not sound as good as the other three.

 3) Computer Accounting: I have designed 7 motherboards for PC's over the years. Believe me... there is some Accountant next to each engineer who designed and laid out (pcb layout) a motherboard. The use of cheap parts and poor execution can have a drastic effect on how things work.

 Take for instance I have a HUSH Multimedia PC that I use for testing and some development. Understand this was developed for audio video use. There are no fans which makes it quiet. BUTTTTTTT it uses a single 64V switching power supply. This comes into a board that has eight DCDC power convertors for the +/-12 drives, +5v drives, +5v logic, +3.3v, +1.8v core and +/-12v for PCI. All this makes it not the greatest unit for audio video use. Only the lack of fans and it's cool shell make it worthy of that.

 4) MBPS: The higher you run the interface in Bytes or Bits per second the higher the rate is. I disgree with the idea of Upsampling on the Computer side to the highest rate. It causes more errors than just leaving it red book.

 ~~~~~~

 Anyways... I ran some tests using error flagging counter and my USB analyzer and found that typically running red book & 2m cable on a direct port into the HOST usb controller that I received no errors in a 24 hour period of sending a 1KHZ sine wave into my TAS1020 emulated system. I used this as the basis because I could also count internal packet errors that the analyzer did not see.

 I have run 24/96 on the same interface and setup and have not seen errors either.

 But I have on occasion plugged my dac into the front of my G5 quad and seen errors right off the bat at 24/96.

 The good thing is this... you follow these rules and really you may never have a problem. I did have a customer who actually had a bad motherboard and would never have even known it. The dac sounded bad he went to the AppleStore they looked at it, plugged in an iPhone and it complained about port speeds.

 Later,
 Gordon


----------



## Crowbar

Quote:


  Originally Posted by *Wavelength* /img/forum/go_quote.gif 
_Actually running a clock from the dac to the transport does not solve the jitter problems. The effect of slew rate and cable distance of clocks exceeding 10MHz will not resolve this._

 

It does if you synchronously reclock the signal at the DAC--which would be done as a matter of fact in any system that sends the clock to the transport.


----------



## EliasGwinn

Quote:


  Originally Posted by *KarateKid* /img/forum/go_quote.gif 
_... you did not recommend using balanced headphones straight out of the DAC1, is it better to use a balanced headphone amp, even if it's still not ideal as you've explained?_

 

A properly designed balanced headphone amp will drive balanced headphones better then directly driving them from the XLR outputs of the DAC1. This is because the output impedance for a headphone amp should be as close to 0 (zero) ohms as possible. The output impedance of XLR outs on the DAC1 is at least 60 ohms. The output impedance of the headphone output of the HPA2 headphone amps, which are built into each DAC1, is less then 0.1 ohms. Since balanced headphones have two outputs per channel (+ and -), the output impedance will NECESSARILY be twice (2x) as much as it would be if it were unbalanced. This is one of several problems with balanced headphone topologies.

 As far as eliminating cross-talk, most high-quality headphones will not have common return conductors and, consequently, will not have problems with crosstalk. The HD650's, for example, have separate return conductors for each driver. Balanced headphone lines will not provide any additional separation.

 Thanks,
 Elias


----------



## EliasGwinn

Quote:


  Originally Posted by *joijwall* /img/forum/go_quote.gif 
_3) If everything is indeed bit-perfect from the source all the way to DAC1, the DAC1 analog output quality is what could effect the sound. What I understand, Benchmark suggests a removal of the headphone-mute-function for best quality? _

 

First of all, we do not suggest removing the headphone-mute-function. Where did you hear that? 

 The quality of the sound will be a result of many things:
 - The information encoded within the data 
 - The quality of the time information accompanying the data
 - The competency of the DAC in handling any timing errors
 - The quality of the circuit board layout in minimizing EMI interference that my interfere with the converter clock (i.e., optimized shielding, traces, and filtering)
 - The ability of the power supply to filter noise and regulate a consistent power delivery
 - The quality of the D/A chip, and the topology which it is implemented within the DAC
 - The quality of the analog circuit design, especially by employing a thorough understanding the strengths and weaknesses of various components and designing sympathetically according to these strengths and weaknesses.
 - Spending time and resources on performance rather then aesthetics and marketing

  Quote:


  Originally Posted by *joijwall* /img/forum/go_quote.gif 
_4) I don't know much about ground loops, but I could try optical or run on batteries and listen. I've also moved my music files to a separate HDD not long ago, maybe that has some influence also. _

 

I don't recommend running audio equipment on batteries. Don't worry about ground loops unless you hear a 'hum' or 'buzz'. 

 The external HDD should not be a problem, unless it is connected via USB simultaneously with a USB audio device. This scenario may lead to drop-outs, etc.

  Quote:


  Originally Posted by *joijwall* /img/forum/go_quote.gif 
_5) Regarding CD-players, are there other important things involved than reading from the disc, D/A-converting and the electronic components (including analog outputs)? I thought perfecting these three gave better sound (and higher price). How does DAC1 perform in these areas? I listened to Linn Klimax DS ($30000 HDD/Streamer/DAC-combo) and it sounded fantastic. Which led me in on the DAC1 path by the way._

 

All of the things mentioned above will influence a transports analog outputs as much as a DAC.

 Thanks,
 Elias


----------



## Lord Chaos

I'd like to know what happens when a USB device, such as a DAC1 USB, encounters an error from the USB. Is there any error correction in the data stream that allows the device to recover the word? Does the device interpolate a new word?


----------



## EliasGwinn

Quote:


  Originally Posted by *joijwall* /img/forum/go_quote.gif 
_By the way, how is it possible to test the bit-transparancy at all on the Mac? When directing the output to DAC1 in Audio/Midi setup, I can choose sample rate, but I cannot set wordlength, it is always 2ch-24bit. 
 Doesn't this mean bits sent to USB is changed? I mean, the original CD is 44,1/16, but the USB output is 44,1/24, thus we cannot compare the bits unless we convert it back after the USB?
 If I understand Benchmarks wiki correctly, the 16->24 conversion is done by iTunes. Will a 16->24 conversion change the "music" information in any way, resulting in DAC1 presenting a different analog output than if the original 44,1/16 was feeding it?
 In my case, I guess DAC1 gets a 44,1/24 from my MacBook, but a 44,1/16 from the optical output on my Linn Majik CD. 
 In short: Since DAC1 wants the 24 bits, and setup forces iTunes to convert 16->24, don't we need to check for bit-transparancy somewhere inside DAC1?
 /joijwall_

 

Establishing a 24-bit connection will not affect 16-bit audio. Think of it like lanes on a highway. If we want the ability for 24 cars to travel side-by-side, we need 24 lanes. 16 cars will travel down that 24-lane highway just as they would if it were 16-lanes. However, a 16-lane will not allow 24 cars to travel side-by-side. This is why a 24-bit connection is always encouraged, even in 16-bit applications.

 This is different from over-sampling the audio. Over-sampling, or any re-sampling for that matter, involves major DSP (*D*igital *S*ignal *P*rocessing). This DSP _will_ change the audio information. Sometimes it will result in major sonic degradation, sometimes it will be unnoticeable.

 However, a 24-bit path will never change 16-bit signal. 

 Thanks,
 Elias


----------



## EliasGwinn

Quote:


  Originally Posted by *Lord Chaos* /img/forum/go_quote.gif 
_I'd like to know what happens when a USB device, such as a DAC1 USB, encounters an error from the USB. Is there any error correction in the data stream that allows the device to recover the word? Does the device interpolate a new word?_

 

No, there is no error-correction in the DAC1 USB. The DAC1 USB will convert all audio data as it sees it. However, we have seen very few errors (usually none) in our testing.

 Thanks,
 Elias


----------



## eweitzman

Quote:


  Originally Posted by *EliasGwinn* /img/forum/go_quote.gif 
_A properly designed balanced headphone amp will drive balanced headphones better then directly driving them from the XLR outputs of the DAC1. This is because the output impedance for a headphone amp should be as close to 0 (zero) ohms as possible. The output impedance of XLR outs on the DAC1 is at least 60 ohms. The output impedance of the headphone output of the HPA2 headphone amps, which are built into each DAC1, is less then 0.1 ohms. Since balanced headphones have two outputs per channel (+ and -), the output impedance will NECESSARILY be twice (2x) as much as it would be if it were unbalanced. This is one of several problems with balanced headphone topologies._

 

To elaborate (and not to be argumentative)... Some people were talking about driving balanced phones from the DAC1 in  this thread. I explained why the frequency response would be boosted 3db boost in the bass and 1db in the treble. This post explains why, and this this post gives some numbers.

 In my opinion, the frequency distortion from the 60 ohm output impedance will swamp any extra noise or distortion or other problems due to two opamps per channel. The bass boost at around 80Hz will make the phones sound more rhythmical while the treble boost will make them sound more detailed.

 - Eric


----------



## Lord Chaos

Thank you, Elias. That corroborates my experience, but I was curious.


----------



## joijwall

Thanks, Elias!
  Quote:


  Originally Posted by *EliasGwinn* /img/forum/go_quote.gif 
_First of all, we do not suggest removing the headphone-mute-function. Where did you hear that? _

 

Page 13 Rev D of the Manual: 
 "TIP: If the DAC1 USB is being used in a critical signal chain (...) the headphone mute switch should be defeated using the internal jumpers."
 Maybe I shouldn't have used "removing"? Or am I wrong to understand the passage to refer to some kind of bypassing?
 /Joachim


----------



## Ross MacGregor

Quote:


  Originally Posted by *joijwall* /img/forum/go_quote.gif 
_Page 13 Rev D of the Manual: 
 "TIP: If the DAC1 USB is being used in a critical signal chain"_

 

That's a really odd tip. I wouldn't expect the headphone out to be part of a critical signal chain in the first place. For example, you are recording or broadcasting a live event. IE: we don't want Aslee Simpsons "mic" to go dead!


----------



## Ross MacGregor

Quote:


  Originally Posted by *Wavelength* /img/forum/go_quote.gif 
_Here are some of the things that I would consider variables for errors on the USB link:_

 

Gordon, 
 Thank you for the wealth of information on the subject. This is exactly what I was looking for. It's great to hear from someone who has actually tested the hardware under real operating conditions. 

 Great USB tips for DAC owners, this should be on a Web page or stickied somewhere more permanent. I might have used my front panel USB ports without thinking they would be more prone to error.

  Quote:


  Originally Posted by *Wavelength* /img/forum/go_quote.gif 
_The good thing is this... you follow these rules and really you may never have a problem. _


----------



## Wavelength

Quote:


  Originally Posted by *Ross MacGregor* /img/forum/go_quote.gif 
_Gordon, 
 Thank you for the wealth of information on the subject. This is exactly what I was looking for. It's great to hear from someone who has actually tested the hardware under real operating conditions. 

 Great USB tips for DAC owners, this should be on a Web page or stickied somewhere more permanent. I might have used my front panel USB ports without thinking they would be more prone to error._

 

Ross,

 I talked to my web guy yesterday about this. We are going to rework our concept page into a white paper kind of format. I have written a ton of these little things for my dealers and I will make up seperate pages for these.

 You can keep in touch on usbdacs.com. None of this stuff is dac specific. all of the setup information can be used by most products.

 Thanks
 Gordon


----------



## EliasGwinn

Quote:


  Originally Posted by *joijwall* /img/forum/go_quote.gif 
_Page 13 Rev D of the Manual: 
 "TIP: If the DAC1 USB is being used in a critical signal chain (...) the headphone mute switch should be defeated using the internal jumpers."_

 

Ahh, I see where your concern comes from. This is only referring to situations where the DAC1 USB should never be muted. This tip is for critical signal chains that cannot be disrupted. In other words, if the DAC1 USB is being used in a television or radio broadcast, the audio cannot be disrupted under any circumstances.

 The left-most headphone jack has an internal switch that mutes the main outputs when the headphone plug is inserted. However, this has no affect on the sonic performance of the unit. It will, however, be a bit "OOPS!!" if a radio DJ plugged in his headphones and the radio station was muted!!


----------



## vergefio

Hello all 
	

	
	
		
		

		
		
	


	




 this is my first post in this forum...

 I've recently bought a dac1 usb to connect with my laptop (just amazing quality); i'm using foobar 0.9.5.1 and i wanted to check if i'm going bit-perfect playing some track downloaded from:

Multichannel Sound 5.1 - sr.se

 I can hear oly static from output:benchmark1, so i made some research and i tried asio4all and kernel streaming to see if i could go bit-perfect (even if as far as i read from Elias foobar can play bit-perfect just through DS). Choosing asio4all under output tab (setted up like in the first page of the guide in this forum) turned back in static again. I tried then with KS but i can't choose dac1 from the menu so i assume it doensn't work.

 Now, what's going on here? What does i have to check in my system and what to do?

 Thx in advance
 Simone


----------



## furball

Those are DTS or Dolby Digital signals right?

 Your software player has to decode the DTS or Dolby Digital signals, and then transfer the decoded PCM signal to an external DAC, only then can you hear the music.

 Sounds like your software player is either not decoding the DTS/Dolby Digital signals, or your software player is outputting the DTS/Dolby Digital signals directly out to the external DAC.

 Benchmark DAC1 cannot decode DTS/Dolby digital signals.


----------



## vergefio

Thx furball for your reply, those files are .WAV file 44Khz/16bit encoded from a .DTS source and they should be readable by the dac1 only in bit-perfect scenario and they should prove the bit-perfect chain; in the case we hear a static it means the playback it's somewhat managed by kmixer or whatever and it's not bit-perfect.






 (This is what i've understood reading through the forum though, i'd be happy if someone can shed some light on it)


----------



## furball

Benchmark DAC1 does not decode DTS/Dolby Digital signals. And Benchmark DAC1 does not have digital output. It only has analog output. In order to test to see Benchmark DAC1 outputs a bit perfect digital output, you need to open up the case and do some mucking around with the internal components.

 How is your setup hooked up?


 That wave file is still a DTS encoded file. You need a DTS decoder to decode that file. This can only be done in the following ways,
 1) Your software player
 2) Your software player outputs the digital signal to an external DTS decoder (such as your receiver)


 Because the Benchmark DAC1 does not have a DTS decoder, all you hear is pink noise.

 And because Benchmark DAC1 does not have a digital output (pass through), you cannot hook up your DTS decoder to the Benchmark DAC1 to test to see if the bit perfect digital output is extracted from the USB port.


----------



## infinitesymphony

But more importantly, the DAC1 does not accept multi-channel signals because it's a two-channel DAC. You'll have to decode the DTS stream in software, then downmix it to 2.0 stereo, and by that point, it won't be bit-perfect.


----------



## eweitzman

Quote:


  Originally Posted by *infinitesymphony* /img/forum/go_quote.gif 
_But more importantly, the DAC1 does not accept multi-channel signals because it's a two-channel DAC. You'll have to decode the DTS stream in software, then downmix it to 2.0 stereo, and by that point, it won't be bit-perfect._

 

Or use three DAC1s 
	

	
	
		
		

		
		
	


	




 - Eric


----------



## furball

No, this still would not work.

 DTS, Dolby Digital, and their variants are a compression scheme designed to compress multichannel audio signals into one encoded digital signal.

 A decoder, such as the one that exists in your receiver, decodes the DTS, Dolby Digital and other such formats, decodes the encoded digital signal into into 5 or 7 channels of analog signals.

 There is only one encoded digital stream that carries the encoded DTS/Dolby Digital signal. The DTS/Dolby Digital decoder decodes that one encoded digital stream into the respective 5 or 7 channels of analog signal.


 Benchmark DAC1 only decodes PCM style (2 channel) digital signals. It cannot decode DTS, Dolby Digital signals. You need your receiver to do that.


  Quote:


  Originally Posted by *eweitzman* /img/forum/go_quote.gif 
_Or use three DAC1s 
	

	
	
		
		

		
		
	


	




 - Eric_


----------



## eweitzman

I don't have a multichannel setup (computer or otherwise) so it's just academic for me....

 I just figured the single six channel digital stream (DTS or DD) would be decoded/demuxed into six independent, uncompressed digital channels before D to A conversion in a PC. This is how mp3 is played back. The multiple frequency-based and scaled channels are reconstructed and merged into plain old PCM by software before hitting the DAC chip.

 - Eric


----------



## furball

Really? I always thought that DTS/Dolby Digital signals are outputted in one digital signal.

 The DTS/Dolby Digital decoder inside your receiver only outputs 5 or 7 channel analog outputs, it does not output 5 or 7 channel PCM style digital signals.

 As far as I know this also applies to computer soundcards. That one coaxial cable or optical cable carries the entire multichannel digital stream. Those 5 or 7 channel outputs are analog outputs.


----------



## eweitzman

You're correct in what you're saying, but there's more.

 Just think about mp3 playback for a minute. Two channels of stereo are encoded into 32 or so low bitrate, frequency-limited channels. Possibly with joint stereo processing that collapses some channel pairs into single channels. Or flac for that matter: two channels of PCM are encoded as polynomial coefficients and correction bits. How do you play these through your DAC1? Your media player (foobar, winamp, whatever) converts these mishmashes into two channel PCM which the DAC1 then converts to analog.

 Now take a six channel format. Different encoding, more channels in the end, but the same deal in theory. Hardware support is another matter, but I don't see any reason why a multichannel pro audio interface couldn't spit out six discrete (digital) PCM channels after the DTS/DD data has been decoded, leaving the D-to-A conversion to something else, like a triplet of DAC1s.

 Maybe this isn't done for commercial or licensing reasons. Can somebody who knows chime in?

 - Eric


----------



## infinitesymphony

Quote:


  Originally Posted by *eweitzman* /img/forum/go_quote.gif 
_Maybe this isn't done for commercial or licensing reasons. Can somebody who knows chime in?_

 

I don't know for sure, but my guess is that there isn't a market for it. The best solution would be a multi-channel DAC, but those are practically non-existent. The only current multi-channel solutions are home theater preamplifiers and receivers.

 If Benchmark released a multi-channel / home theater preamp version of the DAC1 PRE, I'd be there. Add IR remote on/off and volume support and I'd _be there_.


----------



## furball

PCM digital signal only carries 2 channels.


 DTS/Dolby Digital signals are encrypted. The encryption is there for copyright protection. Because of the encryption, the DTS/Dolby Digital decoder is both a decoder and a multichannel DAC, you cannot separate out the two. The multichannel digital to analog conversion is done at the decoder level, and the decoder only outputs analog outputs. The decoder will not output PCM style digital output for each of the 5 or 7 channels.

 I am sure there are ways around this. But you have to dig around on google to see what you can turn up.


 If you want better sound quality for DTS/Dolby digital materials, get a USB DAC card that has an optical or coaxial digital output, hook it up to your receiver, let your receiver decode the DTS/Dolby Digital signals.


----------



## EliasGwinn

Quote:


  Originally Posted by *eweitzman* /img/forum/go_quote.gif 
_Maybe this isn't done for commercial or licensing reasons._

 

I believe this is true. However, ...:

Oppo DV-980H w/3x S/PDIF digital outputs pre-installed


----------



## furball

There are a mind boggling array of multi-channel compression formats, that's why you see new receivers come on the market so quickly. Receiver manufacturers cannot even keep up with the developments of these new formats. The receiver you purchased last Christmas is already obsolete by this time now.

 I say Benchmark should leave all this multichannel stuff and the constant product cycling to the likes of Yamaha, Denon, Sony, etc., and just concentrate on doing an outstanding job with PCM decoding. Otherwise we will see a new Benchmark once every quarter, and the resale value of old Benchmarks would plummet...
	

	
	
		
		

		
		
	


	





  Quote:


  Originally Posted by *infinitesymphony* /img/forum/go_quote.gif 
_If Benchmark released a multi-channel / home theater preamp version of the DAC1 PRE, I'd be there. Add IR remote on/off and volume support and I'd be there. 
	

	
	
		
		

		
		
	


	


_


----------



## infinitesymphony

Quote:


  Originally Posted by *EliasGwinn* /img/forum/go_quote.gif 
_I believe this is true. However, ...:

Oppo DV-980H w/3x S/PDIF digital outputs pre-installed_

 

Whoa... So, only $3,485 for the ultimate 5.1-channel DAC1 experience.


----------



## furball

Isn't stripping out and outputting un-encrypted digital signal from SACD's/DVD-A's kind of...illegal?
	

	
	
		
		

		
		
	


	




 Anyways, I have a question, suppose you do have the 2 channel stereo PCM stream from SACD's/DVD-A's, can Benchmark DAC1 decode those signals?


  Quote:


  Originally Posted by *EliasGwinn* /img/forum/go_quote.gif 
_I believe this is true. However, ...:

Oppo DV-980H w/3x S/PDIF digital outputs pre-installed_


----------



## EliasGwinn

Quote:


  Originally Posted by *furball* /img/forum/go_quote.gif 
_Isn't stripping out and outputting un-encrypted digital signal from SACD's/DVD-A's kind of...illegal?
	

	
	
		
		

		
		
	


	


_

 

I don't know.

  Quote:


  Originally Posted by *furball* /img/forum/go_quote.gif 
_Anyways, I have a question, suppose you do have the 2 channel stereo PCM stream from SACD's/DVD-A's, can Benchmark DAC1 decode those signals?_

 

Yes.

 Thanks,
 Elias


----------



## eweitzman

furball,

 PCM refers to how data is encoded/represented: as 16 bit (or 24 bit, etc) periodic samples. The S/PDIF and AES3 transmission standards carry two interleaved channels of PCM and some control info, but their two channel nature is not a limitation of PCM. ADAT carries 8 channels of PCM for example.

 Within a playback machine, you might be able to get at the individual data channels after decoding from format XYZ to PCM. Looks like switch-box.com has done that. I read sometime ago that one of the Oppo machines was capable of outputting SACD digital data via the HDMI connection. As it turns out, the DSD data is converted to PCM (24/88 IIRC) and that's what you get. Not DSD, but not reconverted analog either. One of the later HDMI standards (1.3 IIRC) allows protected/DRM data to be passed through the interface with a protection flag.

 The legal issues with SACD and DVD-A are different.

 Because they are both encrypted digital data on discs, it is illegal to rip the data from the media in the US. But once the data has been converted to PCM signals inside a properly-licensed and manufactured machine, the data can be transported outside to other devices such as our DAC1s. There's a further issue with SACD in that the technology is licensed from Sony who stipulate (via licensing) that the digital DSD data stream NOT be made available outside the box. DVD-A might have similar licensing issues, I don't know about that. I'm pretty sure you can backup or play the decoded "LPCM" hi-rez streams from "enhanced" DVD-V like DualDisc, DAD, and so on. Copying is still verboten because it's copyrighted material, but not because of the encryption sanctions in the DMCA act.

 In the case of the Oppo, they must be using one of the DAC chips that accepts DSD data, converts it to PCM, and then converts the PCM to analog. These DAC chips are the basis of the cheap universal players. I think these chips also function as sample rate converters so the PCM version of the DSD data can be grabbed as an output from the DAC chip. That's probably how SACD gets to go down the HDMI pipe in the vanilla Oppo, and also probably what the modifier grabs and sends to the three S/PDIF outputs that he adds to the back of the case.

 I was thinking that something like the RME Fireface 400 could output the six PCM channels that could be derived from DTS/DD. It has an ADAT optical port that provides 8 channels of PCM. Again, it comes down to hardware: how would you connect three DAC1s to the ADAT output? Is there a breakout box of some kind that can sit in the middle?

 - Eric


----------



## eweitzman

Quote:


  Originally Posted by *EliasGwinn* /img/forum/go_quote.gif 
_I believe this is true. However, ...:

Oppo DV-980H w/3x S/PDIF digital outputs pre-installed_

 

That's too cool!

 - Eric


----------



## furball

Very well written. And thank you for pointing out my misuse of some technical terms.

 Well, like you said, it seems there is no hardware that can do what you want to do.

 Would a software solution work? Of course then you have to make the software decoder output to 3 S/PDIF output channels. These have to be custom built.




  Quote:


  Originally Posted by *eweitzman* /img/forum/go_quote.gif 
_furball,

 PCM refers to how data is encoded/represented: as 16 bit (or 24 bit, etc) periodic samples. The S/PDIF and AES3 transmission standards carry two interleaved channels of PCM and some control info, but their two channel nature is not a limitation of PCM. ADAT carries 8 channels of PCM for example.

 Within a playback machine, you might be able to get at the individual data channels after decoding from format XYZ to PCM. Looks like switch-box.com has done that. I read sometime ago that one of the Oppo machines was capable of outputting SACD digital data via the HDMI connection. As it turns out, the DSD data is converted to PCM (24/88 IIRC) and that's what you get. Not DSD, but not reconverted analog either. One of the later HDMI standards (1.3 IIRC) allows protected/DRM data to be passed through the interface with a protection flag.

 The legal issues with SACD and DVD-A are different.

 Because they are both encrypted digital data on discs, it is illegal to rip the data from the media in the US. But once the data has been converted to PCM signals inside a properly-licensed and manufactured machine, the data can be transported outside to other devices such as our DAC1s. There's a further issue with SACD in that the technology is licensed from Sony who stipulate (via licensing) that the digital DSD data stream NOT be made available outside the box. DVD-A might have similar licensing issues, I don't know about that. I'm pretty sure you can backup or play the decoded "LPCM" hi-rez streams from "enhanced" DVD-V like DualDisc, DAD, and so on. Copying is still verboten because it's copyrighted material, but not because of the encryption sanctions in the DMCA act.

 In the case of the Oppo, they must be using one of the DAC chips that accepts DSD data, converts it to PCM, and then converts the PCM to analog. These DAC chips are the basis of the cheap universal players. I think these chips also function as sample rate converters so the PCM version of the DSD data can be grabbed as an output from the DAC chip. That's probably how SACD gets to go down the HDMI pipe in the vanilla Oppo, and also probably what the modifier grabs and sends to the three S/PDIF outputs that he adds to the back of the case.

 I was thinking that something like the RME Fireface 400 could output the six PCM channels that could be derived from DTS/DD. It has an ADAT optical port that provides 8 channels of PCM. Again, it comes down to hardware: how would you connect three DAC1s to the ADAT output? Is there a breakout box of some kind that can sit in the middle?

 - Eric_


----------



## EliasGwinn

Quote:


  Originally Posted by *eweitzman* /img/forum/go_quote.gif 
_how would you connect three DAC1s to the ADAT output? Is there a breakout box of some kind that can sit in the middle?_

 

RME TDIF-1

 I doubt that the Fireface will convert 5.1 encodes to multi-2-channel PCM's, but I could be wrong. Also, there may be some freeware out there that can do it. If anyone finds anything, let the rest of us know!!

 Thanks,
 Elias


----------



## joijwall

Does all this mean DAC1 understands PCM only, or does it know other digital to analog conversions also?
 I'm also curious about the different players, both PC based and regular CD-players (at least the ones that plays MP3, I guess a CD is PCM?). 
 Are all players equally perfect when converting WAV, AIFF, AAC, FLAC etc into PCM? That is, will all send the same PCM-bits? Or do I need to choose player carefully?
 /Joachim


----------



## Tarkovsky

Hi there joijwall, Elias was kind enough to provide us with a guide on the different players out there and how to get bit-perfect playback. You can find it over at the benchmark website but it's also linked to further back in this thread.


----------



## gjwaudio

Quote:


  Originally Posted by *infinitesymphony* /img/forum/go_quote.gif 
_Whoa... So, only $3,485 for the ultimate 5.1-channel DAC1 experience. 
	

	
	
		
		

		
		
	


	


_

 

Well... maybe, but how are you going to change the volume ?

 For ULTIMATE quality from such a 3xDAC1 setup, you can't use software to change the levels or you'll be tossing out bits (and some of your "resolution"). How can you implement the analog-domain output control, and have all six channels track together - without introducing another six-ganged volume pot between the DAC1 analog outs and your amplifiers ?

 Can you imagine the grief using some Rube-Goldberg-esque arrangement to turn 3 DAC1 knobs in sync ? Sure... a custom _Benchmark DAC-Racque_ to triangulate the controls just right, so the flywheel buffered MasterKnob (actuated by remote controlled step motor) adjusts each sprocket-modified DAC1 volume control via precisely tensioned, non-magnetic carbon fibre chain-link belts.

 Laughable indeed... straight out of Charles Rodriguez... but just in case, I'm off to the Patent Office !

 Cheers,
 Grant


----------



## EliasGwinn

Quote:


  Originally Posted by *joijwall* /img/forum/go_quote.gif 
_Does all this mean DAC1 understands PCM only, or does it know other digital to analog conversions also?
 I'm also curious about the different players, both PC based and regular CD-players (at least the ones that plays MP3, I guess a CD is PCM?). 
 Are all players equally perfect when converting WAV, AIFF, AAC, FLAC etc into PCM? That is, will all send the same PCM-bits? Or do I need to choose player carefully?
 /Joachim_

 

Here is the link to the "Audio Wiki" on our website. 

Main Page - Benchmark

 That page is filled with all the information you will need about any player.

 There are too many variables to tell you that one player is the best. However, iTunes and Foobar lead the pack.

 The DAC1 only works with PCM, which includes MP3's, all varieties of Lossless files, WAV, AIFF, etc. It does not include DSD (SACD). However, Dolby Digital, DTS, and ADAT are all encoded versions of PCM, which can be decoded and converted to the DAC1.

 Thanks,
 Elias


----------



## eweitzman

Quote:


  Originally Posted by *gjwaudio* /img/forum/go_quote.gif 
_Well... maybe, but how are you going to change the volume ?
 ...
 Laughable indeed... straight out of Charles Rodriguez... but just in case, I'm off to the Patent Office !_

 

Sorry, Grant, but I beat you to it. 2001 prototype of the two channel version shown below.

 I'm considering adapting this to the DAC1PRE. (not)

 - Eric


----------



## Wavelength

Eric,

 In most cases the dac chips that have both DSD and PCM inputs do not convert DSD to PCM but actually the reverse. The PCM is converted to a bit stream and outputed as a sigma delta dac. Basically the DSD bypasses the conversion and goes directly to the output.

 Many of the original SONY SACD's units had jumpers and other things that when cut or switched the DSD information would come out as SPDIF information and some people had recorded the info into their computer and such. I cannot remember what the deal was but I don't think it was as good of quality as the DSD output was.

 Most engineers I talked too that did SACD said the contract the the MONEY was hard to swallow. In general this is what killed the idea. The DVD-A stuff just fell by the wayside because people figured the same thing.

 It doesn't really matter at this point. If you are in this forum you all ready realize hard formats will be dead soon and there are more and more hirez download sites poping up all the time.

 My DSL line went from 5MBS to 10MBPS last week for no extra charge and I hope too look at some of the new sites and download some 24 bit content.

 All my buddies who are musicians are asking about master tapes on line and that's when things will get real interesting. The idea of much more dynamic masters is staggering compared to most of the overlly compressed disks you get now.

 Anyway.. have fun back to work,
 Gordon
  Quote:


  Originally Posted by *eweitzman* /img/forum/go_quote.gif 
_Within a playback machine, you might be able to get at the individual data channels after decoding from format XYZ to PCM. Looks like switch-box.com has done that. I read sometime ago that one of the Oppo machines was capable of outputting SACD digital data via the HDMI connection. As it turns out, the DSD data is converted to PCM (24/88 IIRC) and that's what you get. Not DSD, but not reconverted analog either. One of the later HDMI standards (1.3 IIRC) allows protected/DRM data to be passed through the interface with a protection flag.

 The legal issues with SACD and DVD-A are different.

 Because they are both encrypted digital data on discs, it is illegal to rip the data from the media in the US. But once the data has been converted to PCM signals inside a properly-licensed and manufactured machine, the data can be transported outside to other devices such as our DAC1s. There's a further issue with SACD in that the technology is licensed from Sony who stipulate (via licensing) that the digital DSD data stream NOT be made available outside the box. DVD-A might have similar licensing issues, I don't know about that. I'm pretty sure you can backup or play the decoded "LPCM" hi-rez streams from "enhanced" DVD-V like DualDisc, DAD, and so on. Copying is still verboten because it's copyrighted material, but not because of the encryption sanctions in the DMCA act.

 In the case of the Oppo, they must be using one of the DAC chips that accepts DSD data, converts it to PCM, and then converts the PCM to analog. These DAC chips are the basis of the cheap universal players. I think these chips also function as sample rate converters so the PCM version of the DSD data can be grabbed as an output from the DAC chip. That's probably how SACD gets to go down the HDMI pipe in the vanilla Oppo, and also probably what the modifier grabs and sends to the three S/PDIF outputs that he adds to the back of the case.

 - Eric_


----------



## sfogg

Quote:


  Originally Posted by *eweitzman* /img/forum/go_quote.gif 
_In the case of the Oppo, they must be using one of the DAC chips that accepts DSD data, converts it to PCM, and then converts the PCM to analog. These DAC chips are the basis of the cheap universal players. I think these chips also function as sample rate converters so the PCM version of the DSD data can be grabbed as an output from the DAC chip. That's probably how SACD gets to go down the HDMI pipe in the vanilla Oppo, and also probably what the modifier grabs and sends to the three S/PDIF outputs that he adds to the back of the case._

 

Eric,

 The DAC(s) in the Oppo are actually PCM only. The all in one chip (audio decoder, post processing, video processing and even HDMI transmitter) converts DSD to PCM for the Oppo's DACs. The 980 does have the ability to output native DSD over its HDMI interface, or converted PCM, but it can never feed DSD to its DACs natively.

 Because of the all in one nature of the decoding chips the limitations of audio over HDMI actually apply to its own DACs as well. Audio on HDMI is carried in the VBI of the video. The video bandwidth effects the available audio bandwidth. 480i/p does not have enough bandwidth to carry 6x96/24 so when you have the Oppo set at 480i/p output and play a DVD-A the DACs are fed at 48kHz. If you switch video output to 720p or 1080i/p then the DACs are fed at 96kHz.

 Shawn


----------



## sfogg

Quote:


  Originally Posted by *furball* /img/forum/go_quote.gif 
_DTS/Dolby Digital signals are encrypted. The encryption is there for copyright protection. Because of the encryption, the DTS/Dolby Digital decoder is both a decoder and a multichannel DAC, you cannot separate out the two. The multichannel digital to analog conversion is done at the decoder level, and the decoder only outputs analog outputs. The decoder will not output PCM style digital output for each of the 5 or 7 channels._

 

There are many DD/DTS decoders that output PCM to separate DACs within the player/pre-pro or pass the PCM on to additional DSPs for post processing then eventually on the DACs. For example the Oppo DV-980H does this. 

 There are even pre-pros that will output decoded DD/DTS as PCM over multiple SPDIF connections. Theta Casablanca with the digital outputs card option will do this, pretty much any Meridian from the 565 on will do this too to be able to feed Meridian's DSP speakers.

 Shawn


----------



## ted betley

Elias: I finally figured out how to engage asio with j river and it works nicely. But I got an Opticis SUB extension cable today and what a fantastic cable it is! Before Opticis I was running a short usb into my dac which in turn was driving about 40 feet of unbalanced interconnect (Guitar Center generic cable). So when I began using the Opticis I could now run a half meter of silver KK (1120 or something) interconnect or a meter of mogami balanced cable. I could not believe it but my KK unbalanced sounds better than the mogami balanced. I had the opposite effect on an scd1. I have only about 25 hours of breakin on the Benchmark balanced analogue outs. Could this still be an issue or should I change my xlr pad down from -20 db to 0 db? Or is it possible that the unbalanced outs of the Benchmark are better than the balanced outs? Either way I could not be happier with my Benchmark usb dac.


----------



## gjwaudio

Hi Ted
  Quote:


  Originally Posted by *ted betley* /img/forum/go_quote.gif 
_...I could not believe it but my KK unbalanced sounds better than the mogami balanced ... Or is it possible that the unbalanced outs of the Benchmark are better than the balanced outs? Either way I could not be happier with my Benchmark usb dac._

 

You will be surprised at how the sound from the unbalanced connection improves when you take care of RCA sockets (see here). 

 It is telling that Benchmark have addressed this issue in the DAC1 PRE. Good for all you new buyers. Too bad they dropped the BNCs though - my experience indicates it's a better connection for S/PDIF. (...maybe in the DAC1-PRE Mk2 !).

 Cheers
 Grant


----------



## illkemist

Quote:


  Originally Posted by *gjwaudio* /img/forum/go_quote.gif 
_Hi Ted


 You will be surprised at how the sound from the unbalanced connection improves when you take care of RCA sockets (see here). 

 It is telling that Benchmark have addressed this issue in the DAC1 PRE. Good for all you new buyers. Too bad they dropped the BNCs though - my experience indicates it's a better connection for S/PDIF. (...maybe in the DAC1-PRE Mk2 !).

 Cheers
 Grant_

 

Hmm...I didn't notice this on the DAC1 USB that I returned. But I just took a look at my Pre, and chuckled. Thanks for the link.


----------



## SCOTTY1

I'm currently trying to decide whether to get a B'mark Pre or a Stello DA220 Mk2 & would be grateful for any input from those that have heard these units.


----------



## poo

Quote:


  Originally Posted by *SCOTTY1* /img/forum/go_quote.gif 
_I'm currently trying to decide whether to get a B'mark Pre or a Stello DA220 Mk2 & would be grateful for any input from those that have heard these units._

 

I think the most helpful input I can offer is to read through the previous 150something pages here, then compare that info to the thread on the Stello DA220.


----------



## SCOTTY1

Yeah, I know but I supppose what I can't find is a post along the lines of, "I have lived with B'Mark Pre for x weeks now and my impressions are ......"


----------



## EliasGwinn

Quote:


  Originally Posted by *ted betley* /img/forum/go_quote.gif 
_I could not believe it but my KK unbalanced sounds better than the mogami balanced. I had the opposite effect on an scd1. I have only about 25 hours of breakin on the Benchmark balanced analogue outs. Could this still be an issue or should I change my xlr pad down from -20 db to 0 db? Or is it possible that the unbalanced outs of the Benchmark are better than the balanced outs? Either way I could not be happier with my Benchmark usb dac._

 

Ted,

 The balanced and unbalanced outputs use the same exact same circuitry and have identical performance. The difference you are experiencing may be a result of the differences in output level. With the -20 dB pads, the XLR outputs will be 4.5 dB below that of the RCA outputs. This difference in amplitude will not affect the quality of the sound, but it will affect the perception of quality.

 Thanks,
 Elias


----------



## EliasGwinn

Quote:


  Originally Posted by *SCOTTY1* /img/forum/go_quote.gif 
_I'm currently trying to decide whether to get a B'mark Pre or a Stello DA220 Mk2 & would be grateful for any input from those that have heard these units._

 

If you have any questions that I can answer, don't hesitate to ask.

 Thanks,
 Elias


----------



## poo

Quote:


  Originally Posted by *SCOTTY1* /img/forum/go_quote.gif 
_Yeah, I know but I supppose what I can't find is a post along the lines of, "I have lived with B'Mark Pre for x weeks now and my impressions are ......"_

 

In your shoes (and assuming it has all the features you are after), I would buy the Benchmark, and if it doesn't perform as you'd hoped, sell it here in the for sale forum. Nothing like trying it yourself, and you won't lose much reselling it here if you decide to. Surely the ability to try it yourself outweighs the small financial loss you may incur?


----------



## illkemist

Quote:


  Originally Posted by *SCOTTY1* /img/forum/go_quote.gif 
_Yeah, I know but I supppose what I can't find is a post along the lines of, "I have lived with B'Mark Pre for x weeks now and my impressions are ......"_

 

Well, I've had it for about 5 days. It's too early to draw any conclusions. However, I did have the DAC1 USB for about two prior to upgrading to this unit. So far, I have been very happy. To be honest, aside from features, I can't really tell any difference in performance between the two products. Is there a particular reason why you're looking at the Pre and not the DAC1 USB?


----------



## pcf

Quote:


  Originally Posted by *SCOTTY1* /img/forum/go_quote.gif 
_Yeah, I know but I supppose what I can't find is a post along the lines of, "I have lived with B'Mark Pre for x weeks now and my impressions are ......"_

 

Hi,

 I received my DAC1 pre three days ago. Before that I had a DAC1USB for twe weeks before I returned it and got the pre instead. So far I could not hear the difference in the two units. 

 I went for the DAC1 pre because of the extra inputs and I do want to try 
 to use it as a pre amp.

 I was using the DAC1 pre for two days purely as a DAC and today I plugged it directly into my power amps and it seemed to clean up the sound- I don't want to jump into any conclusion yet after only using it for two days.

 My set up is like this:

 Meridian 500 transport
 Oppo 970 (for SACD)

 DAC1pre

 Garrard 401 with SME4 and Goldring Elite

 EAR 834p phono preamp

 Tube Technology Prophet pre amp
 Tube Technology Genesis mono blocks x2

 Proac D25

 I was using DAC1 USB purely as DAC for twe weeks and was very happy with the result. The music was clean and detailed. I also listened to some music (FLAC) from my laptop via the USA input and it sounds good too.

 I replaced the DAC1 USBwith the DAC1 pre two days ago and everything sounded the same.

 Today I bypassed the pre amp and ran the DAC1pre directly into my mono blocks. The sound seems to gain more clarity and openness. I only briefly listened to the vinyl but it also sounds good.

 Like I said before- I don't want to make any judgement after only a day's listening but it is promising.

 The only thing is that if I do use the DAC1 pre as a pre amp, I would only have a analog input which I will use for my turntable. If I do want to listen to SACD I can't. Also the source selector can be a pain to use. I wish I could just go from one input to another instead of rotating the knob. Also the source status display doesn't light up when the source is off which means that you can't easily tell which input you are on. The red mark on the volume knob is also very hard to see in low light.

 I am keeping the DAC pre because of its sound quality. As a pre amp, it is not as easy to use as it should be. I am someone who is used to turn on six different things before any sound is heard- turn on the pre amp; wait for a while before turning on the power amps; turn on the phono amp; turn on the power supply of the turntable; turn on the turntable. So if I find DAC1 pre tedious to use, can you imagine someone who is used to a remote control?

 It seems the "pre" part of the DAC1 is more like an after thought. it is a shame because it sounds great as a pre amp and yet it is not user friendly.(Not enough analog input; source select..etc..)

 -Paul


----------



## SCOTTY1

Were I based in the US I would take advantage of the 30 day trial period without hesitation. However, here in the UK, there is no such facility & in addition the Pre costs £1,300 vs. $1,575 which is a rather brutal exchange rate. Leaving such prosaic details aside, I need a new pre as my ancient MF Pre 3a is dead. Whereas the virtues of the DAC function of the B'Mark have been well aired, there is understandably little on the net about the beefed up pre-amplifier. I know all this stuff is hopelessly subjective but how does it compare to a stand alone pre-amplifier ?


----------



## SCOTTY1

PCF, thanks for your post. I think the fact that you feel the B'mark is even fit to stand in the same locker-room as your Tube Tech. unit is a fairly hefty endorsement.


----------



## pcf

Hi Scotty,

 I am very positive about DAC1's sound quality (both as a dac and a pre amp) but l don't think it is user friendly enough as a pre amp. I do like the multiple digital inputs though. If you want to play music from you computer; squeeze box; ipod and cd transport etc.. It is great! (That's the main reason I changed from the USB version to the Pre). But if you have more than one analog source( vinyl and sacd for example)or a remote control, you are stuck. The balanced input is also gone if you do care about that. 

 1300 pounds is a lot more than $1575.00! Benchmark also only has one dealer in the UK (SCV London) which seems to sell professional gears only. Are you located near London? You could give them a call and see if they can let you listen to it.

 Everything has always been a lot more expensive in the UK. I move from London to California 9 years ago and found that everything was so much cheaper here!

 Do you need an analog input? If you don't you can save a lot of money by going for the DAC1 USB or the regular version even. I really couldn't tell them apart in term of sound quality.

 Good luck!

 -Paul


----------



## roberto

Elias,

 I bought the DAC1 Pre and got it hooked up with my computer through the Toslink connection. I do like this DAC, but in order to enjoy music without background noise I'll need to place my computer in another room. I haven't heard a computer to be quiet enough if you listen to classical music. So, I'm thinking of buying a 24 feet DVI cable for my LCD, getting longer USB cable and Hub for other peripherals, and using the DAC1 Pre with a longer -at least 15 feet- Toslink cable. This way the noisy device of my system (computer) could be placed in a separate room, and still control everything from the listening room. 
 My only concern is the Toslink cable length between the computer and the DAC1 Pre. I've read that Toslink distances are very limited and signal attenuation occur. This would mean MUCH higher jitter... I guess. I also read that the DAC1 is immune to jitter. In other words Toslink cable length would not affect its performance. Can you reassure me regarding this issue? Is it worth investing money in these cables? 
 And also... Toslink cable prices vary so much and the so called "hi-performance" cables may be beyond my budget. Do you think I'll be safe getting a 15 foot Toslink cable within the two digit price range?? 

 Sorry if I repeated someone's questions or if these concerns were already addressed. 

 Thank you very much for all your support, it's great you do this!!


----------



## EliasGwinn

Roberto,

 You have nothing to worry about. The sonic performance of the DAC1 PRE will not be comprimised with a long, affordable Toslink cable.

 If the signal attenuation in the optical cable is too great, the DAC1 PRE will simply go into auto-standby mode. In other words, the result will either be full quality or no sound at all. If signal loss is too great and the DAC1 PRE goes into auto-standby, you may need a shorter or higher-quality cable. However, 15 foot should not be a problem with an inexpensive cable.

 Thanks,
 Elias


----------



## roberto

Elias, 

 Thank you very much!!


----------



## furball

Get a Slim Devices Squeezebox, you don't have to mess with long cable runs.

 oops


----------



## roberto

Furball,

 Thank you for the advice. I thought about Squeezebox, and it is probably the best solution if you only want music. But with the setup I imagined, I could control the computer. HUGE difference for me. And if Elias is right, the cables are still cheaper then the new device. 

 Thanks anyway,
 roberto

 PS: Nice link


----------



## furball

Oops


----------



## little-endian

Hi Elias,

 yesterday, I've just recognized that the DAC1 responds to the 217 Hz burst used by GSM-compatible mobile phones as well and wondered why.

 Is it so difficult to shield this and prevent interspersing the audibly buzzing? And how about the ADC1 - would an audio engineer have this sound in his recording when he forgets his mobile in the near which might acknowledge a short message arriving by chance? 
	

	
	
		
		

		
		
	


	




 For your information: currently I'm using the DAC1 in conjunction with the HD-650 only.


----------



## infinitesymphony

Motorola have some sort of solution for the GSM buzz, but it doesn't sound like anything desirable for the audio chain.


----------



## riverlethe

Even if they shielded the DAC1, you would still get noise in any other cable or audio device in proximity. The signal is strong enough to turn my monitors on from standby.


----------



## little-endian

Well, actually it's not a big problem at all (at least I presume that no harm to the hardware is done by this interference). I was just interested if the DAC1 would "react" to this. I always thought that just radio receivers were especially prone to this by nature.


----------



## G-U-E-S-T

Hi Elias,
 I am interested in trying the new DAC1 Pre with some active Quad 12L speakers. I would also like to use my subwoofer. Is there any problem using both the XLR outs and the RCA outs simultaneously on the DAC1 Pre? I've been told that the DAC1 Pre's output impedance, and the driving current, would suffer (due to the power supply stress of all outputs simultaneously being used) - and also that the balanced operation of the DAC1 Pre's XLR outs would also be compromised when driving both balanced and unbalanced outputs simultaneously. But I would like to confirm all this with you. Could you please comment on these concerns, and also give any other advice you may have for my application? Thanks!


----------



## Covenant

I'm sorry if this question has been asked already, i read through the first 25pgs before i couldnt absorb information any longer 
	

	
	
		
		

		
			





 I am considering purchasing a DAC1 for use in a balanced system, running the XLR outputs to a balanced B22 amplifier. One thing that AMB specifically cautions against with the B22 is to ensure that there is no DC offset at the source, as the amp design does not handle this well.

 Mr Elias, does the DAC1 (USB or non-USB) measure any DC offset at its outputs, either unbalanced or balanced?

 Thank you again for your time and patience. Its awesome to see a product in the (often snake-oil laden) audiophile industry so well backed up by the engineers who designed it, kudos to you


----------



## riverlethe

Quote:


  Originally Posted by *little-endian* /img/forum/go_quote.gif 
_Well, actually it's not a big problem at all (at least I presume that no harm to the hardware is done by this interference). I was just interested if the DAC1 would "react" to this. I always thought that just radio receivers were especially prone to this by nature._

 

I've heard cell phone noises if I had it near my headphone cable. There was even a study recently saying that sleeping too close to your cell phone can disrupt your sleep cycle...


----------



## Covenant

Or if Elias isnt available to post, have any of the DIYers checked the DAC1's outputs for DC offset?


----------



## little-endian

Quote:


  Originally Posted by *riverlethe* /img/forum/go_quote.gif 
_I've heard cell phone noises if I had it near my headphone cable._

 

Hmm, maybe it has even nothing to do with the DAC1 in this case but the noise is inducted directly into the cable. 
	

	
	
		
		

		
		
	


	




  Quote:


  Originally Posted by *riverlethe* /img/forum/go_quote.gif 
_There was even a study recently saying that sleeping too close to your cell phone can disrupt your sleep cycle..._

 

I've heard that, too. However depending on the network's "Periodical Location Update (PLU)", I wouldn't give too much about this. In the case of the German provider T-Mobile for instance, the mobile will send an update every 6 hours while just listening the other period. Hence the exposure should be almost zero when in standby.


----------



## Terje

A DAC1 computer audio tweak
Odyssey Audio HK Forum - eeepc + Benchmark DAC1 Pre = great sound!


----------



## infinitesymphony

Quote:


  Originally Posted by *Terje* /img/forum/go_quote.gif 
_A DAC1 computer audio tweak
Odyssey Audio HK Forum - eeepc + Benchmark DAC1 Pre = great sound!_

 

That looks more like an advertisement to me... The head administrator of Odyssey Audio posts in his own forum that his equipment magically makes the DAC1 sound better, without any kind of objective testing or results.

 "We again proved that there are science and technique behind hifi instead of myth."

 No, not really... No proof.


----------



## furball

Hi Elias,

 I have 2 questions regarding the SPDIF/AES BNC input. The manual says that the DAC1 comes with a BNC-to-RCA adapter. It also says,

 "TIP: Shielded 75-Ohm coaxial cable is highly recommended for stable performance. Do not use 50-Ohm cables."



 My questions are,

 1) do consumer grade coaxial cables have 75 ohm impedance of 50 ohm impedance?

 2) With the supplied BNC-to-RCA adapter, does this mean that I can use a consumer grade coaxial cable to connect the coaxial out from my source to the coaxial in of the DAC1?


----------



## little-endian

Hi furball,

 the following from my side in advance until Elias replies:

  Quote:


  Originally Posted by *furball* /img/forum/go_quote.gif 
_2) With the supplied BNC-to-RCA adapter, does this mean that I can use a consumer grade coaxial cable to connect the coaxial out from my source to the coaxial in of the DAC1?_

 

Actually, it is a RCA-to-BNC converter since the input is the BNC-interface (at this point by the way, I really wonder why Benchmark decided to use RCA for the analog output instead of BNC as well).

 You may use any coaxial consumer source to feed the DAC1. It won't matter if the consumer or the prosessional format is used, it recognizes both. At least I had no problem with my DVD-Player and a cheap cinch-cable to the DAC1, at all. However, I prefer the optical way.


----------



## Covenant

/pines away for his answer

 I'm in the planning stages for my B22 amp at the moment, so i need to know about that DC offset issue to make sure the DAC1 is the amp i want to purchase


----------



## furball

Thanks for the answers!

 I am going to use the coaxial connection because when I was placing my order with monoprice, I was too cheap to order that premium optical cable, I ordered a coaxial cable instead.
	

	
	
		
		

		
		
	


	





  Quote:


  Originally Posted by *little-endian* /img/forum/go_quote.gif 
_Hi furball,

 the following from my side in advance until Elias replies:



 Actually, it is a RCA-to-BNC converter since the input is the BNC-interface (at this point by the way, I really wonder why Benchmark decided to use RCA for the analog output instead of BNC as well).

 You may use any coaxial consumer source to feed the DAC1. It won't matter if the consumer or the prosessional format is used, it recognizes both. At least I had no problem with my DVD-Player and a cheap cinch-cable to the DAC1, at all. However, I prefer the optical way. _


----------



## Wavelength

Quote:


  Originally Posted by *furball* /img/forum/go_quote.gif 
_Hi Elias,

 I have 2 questions regarding the SPDIF/AES BNC input. The manual says that the DAC1 comes with a BNC-to-RCA adapter. It also says,

 "TIP: Shielded 75-Ohm coaxial cable is highly recommended for stable performance. Do not use 50-Ohm cables."
 My questions are,

 1) do consumer grade coaxial cables have 75 ohm impedance of 50 ohm impedance?

 2) With the supplied BNC-to-RCA adapter, does this mean that I can use a consumer grade coaxial cable to connect the coaxial out from my source to the coaxial in of the DAC1?_

 

Furball,

 SPDIF is always better when used with BNC 75 ohm cables. There really is no way to make an RCA cable 75 ohm's though many claim to have done this the problem is the connector is the termination and impedance is mostly declared by what is called diameter over diameter. This is the ratio between the outside ground connector and the inside signal connector. You can see on a BNC that the center conductor is very small. Someone ounce said that for an RCA connector with it's fat center to be 75 ohms the outside ground connection needed to be something like 5" in diameter.

 Radio Shack does have a real nice RCA->BNC gold connector if your souce does not have the BNC. But it is best to use a BNC cable if you can find a good one.

 My fav and one that has been tested by a number of magazines is the Nirvana Audio T2 cable. I did tests using this and a number of other cables and the results were pretty staggering.

 ~~~~~~~~

 LE--- RCA's are great for the audio connection mainly because they have existing cables that work well with them. Using 75 ohm BNC's would not make any sense for audio use.

 Thanks
 Gordon


----------



## Crowbar

Faux-75 ohm RCA:
Canare Corp. - Quality Cables and Connectors: RCA Crimp Plugs


----------



## furball

Thank you Wavelength and Crowbar!


----------



## Scrith

Quote:


  Originally Posted by *Wavelength* /img/forum/go_quote.gif 
_My fav and one that has been tested by a number of magazines is the Nirvana Audio T2 cable. I did tests using this and a number of other cables and the results were pretty staggering._

 

Hmm, I did a Google search to find out more about these cables and the only thing I found was that Wavelength and Nirvana Audio shared booths at a couple of CES shows (what a strange coincidence!).

 What exactly are these tests you performed, and what was so staggering about the results?


----------



## Wavelength

Quote:


  Originally Posted by *Scrith* /img/forum/go_quote.gif 
_Hmm, I did a Google search to find out more about these cables and the only thing I found was that Wavelength and Nirvana Audio shared booths at a couple of CES shows (what a strange coincidence!).

 What exactly are these tests you performed, and what was so staggering about the results?_

 

Scrith,

 You can go here to see more info on Steve's cables:

Nirvana Audio - Products

 The basic test I do on any SPDIF cable is three fold. I measure it's ability to deliver the Master Clock output into a known dac. I then determine it's ability to recover data checking for errors and finnally the amount of jitter attenuation it has.

 I have the Prism dScope III Audio Test set it allows me to measure percisely all parameters of SPDIF cables, dacs, usb/firewire/ethernet dacs.

 The FFT analyer on this puppy is good to better than -175dB and the only analyzer available that can test natively for computer audio.

 The Nirvana T2 uses a patent that assures the 75 ohms impedance into the RF range and also down into the audio range.

 This was one of the reason's why we shared suites at CES. I found Steve's cables to be worthy of the price. Which is not something I can say for most of the cable makers out there.

 Why in the world should speaker cables be $18K for a pair in copper. Common....

 Thanks
 Gordon


----------



## Matias

I have my DAC1 USB for quite some time now. I'm using WinXP SP2 + Winamp + ASIO output + ESI Juli@ (soundcard) + SPDIF + DAC1.

 I've found that it's clearly better then using the DAC1 driver + DirectSound. A friend of mine listened to both modes too and, without me saying anything, agreed. Sure, I've read the wiki, disabled EQ and volume, but still.

*[size=medium]The native ASIO path is simply better than DirectSound.[/size]* 

 Any thoughts?


----------



## Crowbar

[size=medium]*Was this a blind test?*[/size]


----------



## Matias

Yes, it was a blind test. The difference is quite obvious.


----------



## Crowbar

I don't know what the smiley implies. My best guess is that you assumed the difference you heard in a sighted test is so obvious that you're certain you'd hear it in a blind test, and didn't actually bother to do it.
 But I don't know you so I'll give you the benefit of the doubt.
 How did you make the test blind? Who was switching the software audio path?


----------



## Matias

Do you want me to shoot a video to prove it to you or what? 
	

	
	
		
		

		
		
	


	



 Let's focus on the problem itself, shall we? 
	

	
	
		
		

		
		
	


	




 ASIO bypasses Windows' kmixer. DirectSound does not. Knowing that kmixer messes up with the sound, how on earth is the Benchmark USB driver supposed to sound the same as the native ASIO path?

 Don't get me wrong, I payed 300 bucks more to have this feature, I'd really WANT it to be as clear as my ASIO output... But it simply is not.


----------



## Wavelength

Matias,

 Maybe a better test would be to do this and it's merly software:

 WinAmp->DirectSount->USB input DAC1
 WinAmp->ASIO->USB input DAC1

 Beacuse if the sound card has a driver it may in it self discard the kmixer for it's own code.

 The above example would prove if you hear the difference and considering how much rederact there is on the KMIXER... I am sure you will.

 Thanks
 Gordon


----------



## EliasGwinn

Hello Head-Fi people!!

 I'm pleased to announce that Benchmark Media Systems has moved to a brand new facility (and I mean brand new...my office window still has the sticker on it!! 
	

	
	
		
		

		
		
	


	




). We are still in Syracuse...in fact, we are less then a mile from a previous facility. 

 I'm sorry I haven't been participating in the discussions here lately. The whole moving process was quite consuming. But, alas, we are up and running, and now I look forward to continuing these discussions with you all.

 Thanks!!
 Elias


----------



## EliasGwinn

Quote:


  Originally Posted by *G-U-E-S-T* /img/forum/go_quote.gif 
_Hi Elias,
 I am interested in trying the new DAC1 Pre with some active Quad 12L speakers. I would also like to use my subwoofer. Is there any problem using both the XLR outs and the RCA outs simultaneously on the DAC1 Pre? I've been told that the DAC1 Pre's output impedance, and the driving current, would suffer (due to the power supply stress of all outputs simultaneously being used) - and also that the balanced operation of the DAC1 Pre's XLR outs would also be compromised when driving both balanced and unbalanced outputs simultaneously. But I would like to confirm all this with you. Could you please comment on these concerns, and also give any other advice you may have for my application? Thanks!_

 

G-U-E-S-T,

 The DAC1 PRE is able to drive all 4 outputs (RCA L; RCA R; XLR L; XLR R)simultaneously without any loss in performance whatsoever.

 The only thing that you will need to take into account is that the RCA outputs have different output levels then the XLR outputs. Therefore, you may need to calibrate the input sensitivity of your subwoofer amp to get the appropriate output level. 

 Let me know if you need help with this.

 Thanks,
 Elias


----------



## Covenant

Welcome back Elias.

 I'd still like to know if the DAC1 has any DC offset at the outputs, both balanced and unbalanced, when you finish moving into your new office


----------



## Matias

Quote:


  Originally Posted by *Wavelength* /img/forum/go_quote.gif 
_Matias,

 Maybe a better test would be to do this and it's merly software:

 WinAmp->DirectSount->USB input DAC1
 WinAmp->ASIO->USB input DAC1

 Beacuse if the sound card has a driver it may in it self discard the kmixer for it's own code.

 The above example would prove if you hear the difference and considering how much rederact there is on the KMIXER... I am sure you will.

 Thanks
 Gordon_

 

Gordon,

 Thanks for your answer. 
 The ASIO plug-in, or any other ASIO application, works only with ASIO-compatible drivers. Benchmark DAC1's is not an ASIO-compatible driver, therefore it doesn't show as an option for me to select.





 If Benchmark could release an ASIO-compatible driver, it would solve the problem. 
 But then they couldn't advertise it as being plug-n-play anymore...


----------



## EliasGwinn

Quote:


  Originally Posted by *Covenant* /img/forum/go_quote.gif 
_I'm sorry if this question has been asked already, i read through the first 25pgs before i couldnt absorb information any longer 
	

	
	
		
		

		
		
	


	




 I am considering purchasing a DAC1 for use in a balanced system, running the XLR outputs to a balanced B22 amplifier. One thing that AMB specifically cautions against with the B22 is to ensure that there is no DC offset at the source, as the amp design does not handle this well.

 Mr Elias, does the DAC1 (USB or non-USB) measure any DC offset at its outputs, either unbalanced or balanced?

 Thank you again for your time and patience. Its awesome to see a product in the (often snake-oil laden) audiophile industry so well backed up by the engineers who designed it, kudos to you 
	

	
	
		
		

		
		
	


	


_

 

The DC offset on any active output pin is on the order of 0.0005 V, or half a millivolt. When the attenuators are engaged on the XLR outputs, the DC offset is attenuated like-wise. In other words, if the 20 dB attenuators are engaged, the DC offset will be reduced by 20 dB or 90%. Therefore, the 0.5 millivolts will become 0.05 millivolts.

 Thanks,
 Elias


----------



## EliasGwinn

Quote:


  Originally Posted by *Matias* /img/forum/go_quote.gif
_ Gordon,

 Thanks for your answer.
 The ASIO plug-in, or any other ASIO application, works only with ASIO-compatible drivers. Benchmark DAC1's is not an ASIO-compatible driver, therefore it doesn't show as an option for me to select.

 If Benchmark could release an ASIO-compatible driver, it would solve the problem.
 But then they couldn't advertise it as being plug-n-play anymore..._

 

  Matias,

 There is a free ASIO "wrap-around" that will enable ASIO-operation with the DAC1 USB. It is called ASIO4ALL. It allows you to interface the DAC1 USB to ASIO applications.

 Thanks,
 Elias


----------



## EliasGwinn

Quote:


  Originally Posted by *furball* /img/forum/go_quote.gif 
_Hi Elias,

 I have 2 questions regarding the SPDIF/AES BNC input. The manual says that the DAC1 comes with a BNC-to-RCA adapter. It also says,

 "TIP: Shielded 75-Ohm coaxial cable is highly recommended for stable performance. Do not use 50-Ohm cables."



 My questions are,

 1) do consumer grade coaxial cables have 75 ohm impedance of 50 ohm impedance?

 2) With the supplied BNC-to-RCA adapter, does this mean that I can use a consumer grade coaxial cable to connect the coaxial out from my source to the coaxial in of the DAC1?_

 

There are both 50-ohm and 75-ohm consumer-grade cables. Cheap cables will have wider tolerances. In other words, they may be anywhere from 65-85 ohm. Professional cables usually have tighter precision. We use (and sell) Canare and Mogami cable and connectors, as they are manufactured very well.

 However, don't hesitate to use consumer grade 75-ohm RCA cables. They will introduce more jitter, but the UltraLock system will make any jitter a moot point.

 Thanks,
 Elias


----------



## Scrith

Quote:


  Originally Posted by *EliasGwinn* /img/forum/go_quote.gif 
_There is a free ASIO "wrap-around" that will enable ASIO-operation with the DAC1 USB. It is called ASIO4ALL. It will bypass kmixer._

 

It should be noted that there is nothing magical or special about the ASIO protocol for sending data to an audio device's driver...using it simply enables users to bypass the Windows kmixer when using an audio application. *One could also use KernalStreaming to achieve the same result as using ASIO.* The advantage to KernalStreaming (assuming it is supported by the sound device's driver) is that it does not require some Windows add-on like ASIO4ALL.

 Please note than in Windows Vista kmixer has changed dramatically (and has been renamed to something else as a result...sorry I'm not sure what the new name is) and is no longer an audio quality compromise when it is part of the audio data chain. In other words, using ASIO or KernalStreaming in Windows Vista does not seem to provide any improvement over just using DirectSound (unlike in Windows XP, where using DirectSound output seems to have an audible effect on sound quality).


----------



## Wavelength

Quote:


  Originally Posted by *Matias* /img/forum/go_quote.gif 
_Gordon,

 Thanks for your answer. 
 The ASIO plug-in, or any other ASIO application, works only with ASIO-compatible drivers. Benchmark DAC1's is not an ASIO-compatible driver, therefore it doesn't show as an option for me to select.





 If Benchmark could release an ASIO-compatible driver, it would solve the problem. 
 But then they couldn't advertise it as being plug-n-play anymore..._

 


 Matias,

 You can use generic ASIO drivers from:

 ASIO4ALL: ASIO4ALL - Universal ASIO Driver
 ASIO2KS: ASIO2KS - Generic ASIO driver for WDM soundcards.

 These will let you use the DAC1 USB with ASIO and determine your output better.

 Thanks
 Gordon


----------



## Matias

Elias, Scrith and Gordon, thank you for you answers. I already knew about ASIO4All, always thought it was made for onboard soundchips, didn't occur to me it'd work with Benchmark's driver. It worked indeed! 
	

	
	
		
		

		
			










 Well, after a couple of initial tests (1 hour or so) with several songs, I noticed ASIO4All sounded somehow differente from DirectSound, not sure if it was better or worse. But I am sure that BOTH sound worse than Juli@'s SPDIF. The latter just has more dynamics (transients), clearer mids and is overall more realistic.

 I give up now. Even using an ASIO path (not native, btw) and bypassing kmixer, Benchmark's USB still sounds a little worse than a true native ASIO+SPDIF output. Is it perhaps the USB circuitry? Or the cheap USB cable that came with it? 
	

	
	
		
		

		
		
	


	




 PS: Did anyone else compare the DAC1's attenuator (variable output) agains a high-end DACT CT2 attenuator? I did. It was a massacre. The DACT is a LOT cleaner and detailed, wider soundstage. Of course it costs half the DAC1, but still.


----------



## Wavelength

Quote:


  Originally Posted by *Matias* /img/forum/go_quote.gif 
_Elias, Scrith and Gordon, thank you for you answers. I already knew about ASIO4All, always thought it was made for onboard soundchips, didn't occur to me it'd work with Benchmark's driver. It worked indeed! 
	

	
	
		
		

		
		
	


	




 Well, after a couple of initial tests (1 hour or so) with several songs, I noticed ASIO4All sounded somehow differente from DirectSound, not sure if it was better or worse. But I am sure that BOTH sound worse than Juli@'s SPDIF. The latter just has more dynamics (transients), clearer mids and is overall more realistic.

 I give up now. Even using an ASIO path (not native, btw) and bypassing kmixer, Benchmark's USB still sounds a little worse than a true native ASIO+SPDIF output. Is it perhaps the USB circuitry? Or the cheap USB cable that came with it? 
	

	
	
		
		

		
		
	


	




 PS: Did anyone else compare the DAC1's attenuator (variable output) agains a high-end DACT CT2 attenuator? I did. It was a massacre. The DACT is a LOT cleaner and detailed, wider soundstage. Of course it costs half the DAC1, but still. 
	

	
	
		
		

		
		
	


	


_

 

Matias,

 There are allot of ways to set this all up. I am not sure why you have the output set to 88.2 instead of 44.1 as this will envoke upsampling unless you are truely using 24/88.2 files. I am not a fan of upsamplers on the PC or the MAC.

 There is as others suggested Kernel Streaming which may fare better in your situation for USB.

 It's up too you as you are the only one you have to satisfy!

 Thanks
 Gordon


----------



## Matias

Gordon,

 Notive that even though there's an option to resample, it is disabled. The DAC already does it's internal resampling to 110kHz. 
	

	
	
		
		

		
		
	


	




 Cheers,
 Matias


----------



## tarobyte

Elias,
 Thanks to your contributions to this thread and other positive feeback, I'm determined to acquire the DAC1 USB, but unfortunately in Japan the only place I can find that carries the unit, fujiya avic, is selling it for about a $300 premium (about 160,000JPY) and they don't even have the product in stock. The Benchmark site doesn't seem to allow direct purchase from JP customers either. There should be alot of latent demand for this product in JP but there is much to be desired around the distribution model.

 This is obviously not an engineering question, but I figure that your presence in this forum is with the end goal of increasing sales I thought it might be okay to inquire.


----------



## Matias

tarobyte, I'm from Brazil and I ordered mine through B&H Photo. No problems!


----------



## poo

Quote:


  Originally Posted by *tarobyte* /img/forum/go_quote.gif 
_Elias,
 Thanks to your contributions to this thread and other positive feeback, I'm determined to acquire the DAC1 USB, but unfortunately in Japan the only place I can find that carries the unit, fujiya avic, is selling it for about a $300 premium (about 160,000JPY) and they don't even have the product in stock_

 

Same deal in Australia unfortunatley - premiumn price - no stock - no expertise or help/advice.

 I also purchased from B&H - very helpful, personalised service and support (which was the last thing I expected from them TBH). I hope Benchmark are wise enough to understand why they should 'turn a blind eye' to international trade in these circumstances. If resellers in those countries were providing appropriate support (or at least supporting the brand with stock), it would be a different matter.

 Nice product though


----------



## riverlethe

Elias, 
 Recently, when adjusting the volume on the DAC1 the first one or two times, I've started hearing some kind of brief static noise in the left channel. This happens in both speaker and headphone outputs. Is this normal?


----------



## jdh500

I paid $1895 AUD for the DAC 1 pre from a hifi store in Sydney last week even theough the RRP is $2295 AUD in Australia. I also prevously purchased my original DAC 1 from the same store, pretty happy with overall service and price even though it is substantially higher in Australia compared to the USA etc.

 JDH.


  Quote:


  Originally Posted by *poo* /img/forum/go_quote.gif 
_Same deal in Australia unfortunatley - premiumn price - no stock - no expertise or help/advice.

 I also purchased from B&H - very helpful, personalised service and support (which was the last thing I expected from them TBH). I hope Benchmark are wise enough to understand why they should 'turn a blind eye' to international trade in these circumstances. If resellers in those countries were providing appropriate support (or at least supporting the brand with stock), it would be a different matter.

 Nice product though 
	

	
	
		
		

		
		
	


	


_


----------



## poo

Quote:


  Originally Posted by *jdh500* /img/forum/go_quote.gif 
_I paid $1895 AUD for the DAC 1 pre from a hifi store in Sydney last week even theough the RRP is $2295 AUD in Australia. I also prevously purchased my original DAC 1 from the same store, pretty happy with overall service and price even though it is substantially higher in Australia compared to the USA etc.

 JDH._

 

I called a few 'specialist' stores in Sydney (and two interstate) before making my decision to import. None were able to offer any sercive, advice, expertise or movement on price.

 Might be worth mentioning the store you are referring to so that others can enjoy the same benefits you speak of.


----------



## jdh500

For those interested, I purchased my DAC1 and DAC1pre from Instyle Home Theatre & HiFi in Sydney, Aust. Speak to Mario the store owner for details, this HiFi dealer was actually very helpful and went out their way to set up a new account with the Australian distributor to order them in for me. They said they were likely to order another one in as a shop demo, so more than likely you should be able to end up having a demo of it too.

 JDH.


  Quote:


  Originally Posted by *poo* /img/forum/go_quote.gif 
_I called a few 'specialist' stores in Sydney (and two interstate) before making my decision to import. None were able to offer any sercive, advice, expertise or movement on price.

 Might be worth mentioning the store you are referring to so that others can enjoy the same benefits you speak of. 
	

	
	
		
		

		
		
	


	


_


----------



## EliasGwinn

Quote:


  Originally Posted by *tarobyte* /img/forum/go_quote.gif 
_Elias,
 Thanks to your contributions to this thread and other positive feeback, I'm determined to acquire the DAC1 USB, but unfortunately in Japan the only place I can find that carries the unit, fujiya avic, is selling it for about a $300 premium (about 160,000JPY) and they don't even have the product in stock. The Benchmark site doesn't seem to allow direct purchase from JP customers either. There should be alot of latent demand for this product in JP but there is much to be desired around the distribution model.

 This is obviously not an engineering question, but I figure that your presence in this forum is with the end goal of increasing sales I thought it might be okay to inquire._

 

tarobyte and other folks in Europe and Asia,

 Sorry... 
	

	
	
		
		

		
		
	


	




 but the extra cost associated with foriegn distribution is unavoidable due to import/export tarrifs, customs fees, and other taxes and fees. 

 Also, I apologize for limited stock. We've been busier then we ever have been...which is a good problem to have, I guess 
	

	
	
		
		

		
		
	


	




 ....but we don't want to make it difficult for you to find them. We are aware of these issues, and we appreciate when you let us know. Hopefully, we can get stock back up to normal levels soon.

 Thanks,
 Elias


----------



## luciyuspax

EliasGwinn, If it is okay, I would like to ask you a question about Benchmark DAC1. Why was Benchmark DAC1 engineered as an upsampling DAC? Its DAC chip allows for non-upsampling processing as well(I mean only its DAC chip,not Benchmark DAC1 as whole),right?

 Although we can argue that external sound cards like E-MU 0404usb or m-audio 2496 usb are not bit perfect with their special usb drivers,they seem to be non upsampling DACs.

 Nowadays I notice that dedicated DACs mostly are upsampling and external lower price DACs like e-mu 0404 usb or m-audio external models (described as sound cards)& internal sound cards are non upsampling. Does upsampling process provide better sound quality over non upsampling?

 I would really appreciate it if you could provide me links to technical documents about it. I am considering buying Benchmark DAC1 sometime in 2009 and I really wanna understand this whole upsampling issue. Don't be afraid to go too much technical as I am a physics student specializing in electronics.

 Have a nice day all head-fi members!


----------



## infinitesymphony

The external cards you listed are capable of bit-perfect output. Regarding upsampling, it's really a case-by-case basis in terms of better/worse because the quality of the implementation varies.

 The Benchmark DAC1 upsamples to a very specific rate (~110 kHz) to maximize the performance of the Analog Devices AD1853 DAC chip inside. See this earlier post for more info.


----------



## EliasGwinn

luciyuspax,

 The process of upsampling does not inherently improve sound quality during D/A conversion. However, Benchmark converters re-sample for a very specific reason: jitter immunity.

 Benchmark converters use a proprietary clocking system (we refer to it as UltraLock). It works like this... 

 The incoming digital signal is immediately re-sampled by an ASRC (asyncronous sample rate converter). The ASRC, as the name implies, is not syncronized to the clock of the incoming digital signal. 

 Therefore, its performance is independant of the quality of that clock. In other words, it doesn't matter if the signal came from a cheap transport with cheap cables, or from a $10,000 signal chain. The large amount of jitter caused by the cheap transport and cheap cable will be moot with respect to the ASRC process. The output of the ASRC is then clocked to an on-board clock with extremely low jitter and strategic sheilding and board traces.

 The output of the ASRC can be configured to any sample rate that we choose, including the original sample rate. However, we dictated the re-sample rate as 110 kHz because it is the highest sample-rate at which the digital interpolation filter of the D/A chip will operate optimally. 

 The ill-effects of the digital interpolation filter at higher-then-110 kHz include pass-band ripple (non-linearities in frequency response) and inferior attenuation of stop-band frequencies (which results in aliasing). Therefore, the D/A performance is optimized by maintaing 110 kHz.

 Many converter designers have since employed similar topologies, but use lower re-sampling frequencies, such as 96 kHz. By resampling to 110 kHz, the low-pass filter of the ASRC and D/A are moved as far up as possible as to not infringe on the analog bandwidth of the audio.

 I hope I explained this clearly... 
	

	
	
		
		

		
		
	


	




 Thanks,
 Elias


----------



## luciyuspax

Yeah,your explanation was very clear. Thank you very much for the answer. Infinitesymphony, It is nice to know that my e-mu 0404 usb has bitperfect output.

 I remember reading another post of EliasGwinn,in which he was explaining why they chose to go with default windows usb drivers for Benchmark DAC1. He had said that they had noticed that many of external usb sound cards using special usb drivers were not actually bit-perfect( he hadn't specified any brand of usb audio about this) and in most cases that windows' own usb audio drivers were bit perfect.


----------



## Wavelength

Elias,

 Why use a fixed clock for the ARCS? Most people who have tested ARSC's feel they sound better when used on a fixed multiple like 2, 4... instead of fractional. 

 So say for 44.1 upsample to 88.2 or 176.4 and so forth.

 In several of the recording studios I work with here (Sound Images, Ashley Shephard's Audio Grotto) feel that when recording that the use of multiples are the key to better results when down sampling. Most of the recordings we have done recently are done at 24/88.2 or 24/176.4 so the output to 16/44.1 sounds better.

 On the dCS gear many have talked about there ability to change the upsampling on the fly and many feel that multiple over fractionals always sound better.

 Your thoughts?
 Thanks
 Gordon


----------



## thomaspf

Quote:


  Originally Posted by *luciyuspax* /img/forum/go_quote.gif 
_Yeah,your explanation was very clear. Thank you very much for the answer. Infinitesymphony, It is nice to know that my e-mu 0404 usb has bitperfect output.

 I remember reading another post of EliasGwinn,in which he was explaining why they chose to go with default windows usb drivers for Benchmark DAC1. He had said that they had noticed that many of external usb sound cards using special usb drivers were not actually bit-perfect( he hadn't specified any brand of usb audio about this) and in most cases that windows' own usb audio drivers were bit perfect._

 


 I think you got this the wrong way round. The Windows USB driver is known not to be bit perfect when used with DirectSound since it will always render kmixer. You can of course get around this by sending the audio data via Direct Kernel Streaming.

 Custom drivers might be able to get around this and you can use any arbitrary DirectSound application. 

 The default Windows drivers have gotten a lot of testing over the years and are very stable.

 Cheers

 Thomas


----------



## 03lab

Hi everyone, first post from a long time lurker and audiophile.

 I'm looking to buy a DAC1 PRE but unfortunately the distributor in Germany and/or Benchmark must have forgotten to do the conversion from USD to EUR. The PRE goes for EUR 1600 around here, which means a hefty $1000 premium on what it sells for in the US, surely something must be wrong here. 
	

	
	
		
		

		
			





 For $1000 I could easily fly to the US and pick one up myself!

 Please advise as to where I can get one at reasonable cost. Thanks!


----------



## spyderx

Quick question:
 I'm using a DAC1 USB. MacMini USB --> DAC1 Balanced --> BAT VK-300xSE --> Totem Hawk speakers

 I'm using the balanced outs from the DAC1, into the BAT. I have it set for "calibrated" outs and I have not touched the -20db jumper inside the DAC1 or the calibration pots yet.

 I seem to have pretty good range on the volume, with normal listening position around 90 (scale goes to 140 on the BAT). 

 Am I going to benefit SQ by dropping the to 0db attenuation on the DAC1? The BAT has an amazingly low noise floor, so I have no noise at all at these levels.

 Thanks


----------



## G-U-E-S-T

Hi Elias,

 I have a question about the ASRC function and the "jitter-immune" claims of the DAC1 products please.

 As you said, the ASRC function of the DAC1 utilizes its own internal clock (instead of the recovered S/PDIF clock) when internally feeding samples to the DA chip. But before this happens, surely the recovered clock is being utilized by the ASRC algorithm, yes? [Otherwise the ordering of the samples as sent, would be unknown, when trying to reconstruct and zip them into the new sampling rate.]

 And if so, would not any "primary" (pre-ASRC) jitter artifacts from the S/PDIF transfer, just be getting remapped to a higher sample rate - and then presented with less "secondary" (post-ASRC) jitter artifacts to the DAC chip?

 Meaning, the pre-ASRC jitter artifacts from the transport-to-DAC1 transfer would still remain - even if the internal ASRC-chip next presented no jitter or artifacts of its own to the DAC-chip.

 For an analogy: Like if a song is recorded with jittery digital equipment in the recording studio, and thus that original jitter gets onto the burned CD and is permanent - this jitter can never be later removed, no matter how jitter-free any later CD playback machine may be. (In this analogy, the ASRC-process represents the later CD playback machine.)

 Is this what Benchmark means by marketing the DAC1 as "jitter-immune", with "no jitter artifacts presented to the DAC chip"? I'm just trying to better understand the details...


----------



## Crowbar

Quote:


  Originally Posted by *G-U-E-S-T* /img/forum/go_quote.gif 
_But before this happens, surely the recovered clock is being utilized by the ASRC algorithm, yes? [Otherwise the ordering of the samples as sent, would be unknown, when trying to reconstruct and zip them into the new sampling rate. 

 And if so, would not any "primary" (pre-ASRC) jitter artifacts from the S/PDIF transfer, just be getting remapped to a higher sample rate - and then presented with less "secondary" (post-ASRC) jitter artifacts to the DAC chip?_

 

Though this is not directly how it's implemented, in mathematical terms the rate estimation of the incoming clock is in effect low-pass filtered. Depending on where the corner of the filter is, you get a good amount of jitter attenuation above that frequency. The attenuation can be made very large. For example, the new ESS Sabre DAC chip has 0.1 Hz corner on its integrated ASRC's rate estimation, which is way below audio band. In theory about any modern ASRC including the one in the DAC1 should have any artifacts below audible levels, but I can pick out an audible difference difference between stand-alone CS and AD ASRC chips, the test board ESS sent me, and non-ASRC with the transport slaved to the DAC clock (all with AD797 I/V and buffer) (true, the DAC chips were different too, but those also are specced with artifacts below what should be audible level; in my experience ASRC/digital filter makes more difference and sigma delta DACs with the same external filter sound pretty much the same to me).


----------



## infinitesymphony

Quote:


  Originally Posted by *G-U-E-S-T* /img/forum/go_quote.gif 
_For an analogy: Like if a song is recorded with jittery digital equipment in the recording studio, and thus that original jitter gets onto the burned CD and is permanent - this jitter can never be later removed, no matter how jitter-free any later CD playback machine may be. (In this analogy, the ASRC-process represents the later CD playback machine.)_

 

But that's not true. For example, a pressed CD with a high jitter rate can be re-burned to have a lower jitter rate.

 The timing of the incoming signal doesn't matter as long as all of the samples are sent correctly; this is the one part where the quality of the transport matters. If the stream is 44,100 Hz, that should be how many samples the reclocking system takes before sending out one second's worth of digital audio. The only point of reclocking is to reduce jitter--the resampling portion is separate. Or at least, that's my understanding of it.


----------



## Crowbar

[edit:nvm]


----------



## thomaspf

Quote:


  Originally Posted by *infinitesymphony* /img/forum/go_quote.gif 
_But that's not true. For example, a pressed CD with a high jitter rate can be re-burned to have a lower jitter rate.

 The timing of the incoming signal doesn't matter as long as all of the samples are sent correctly; this is the one part where the quality of the transport matters. If the stream is 44,100 Hz, that should be how many samples the reclocking system takes before sending out one second's worth of digital audio. The only point of reclocking is to reduce jitter--the resampling portion is separate. Or at least, that's my understanding of it. 
	

	
	
		
		

		
		
	


	


_

 


 You have to be careful with this. The original poster spoke about jitter in the recording chain. That is very different from "pit" jitter on a CD.

 The jitter that is permanently added to the signal is created by the analog to digital converter and comes from the fact that the samples are not actually takes at equi distant intervalls but at times in the jitter intervall. This jitter can not be removed.

 Pit jitter on CDs is a non issue with todays equipment. 20 years ago CD players operated synchronosly and created their output clock from the bit stream they read from the spinning disc. When CD players got added to cars a long time ago this technology did not work and the problem was solved correctly. The ouput clock is generated from a crystal and the rotational speed of the dics is adjusted asynchronously. As long as you can read the bits correctly reburning a disc is not going to make any difference. 

 PC drives which can for example be found in the highest end Meridian CD player read CD asynchronously in any case and even do rereads if they encounter any read errors.

 Cheers

 Thomas


----------



## EliasGwinn

Quote:


  Originally Posted by *luciyuspax* /img/forum/go_quote.gif 
_Yeah,your explanation was very clear. Thank you very much for the answer. Infinitesymphony, It is nice to know that my e-mu 0404 usb has bitperfect output.

 I remember reading another post of EliasGwinn,in which he was explaining why they chose to go with default windows usb drivers for Benchmark DAC1. He had said that they had noticed that many of external usb sound cards using special usb drivers were not actually bit-perfect( he hadn't specified any brand of usb audio about this) and in most cases that windows' own usb audio drivers were bit perfect._

 

Luciyuspax,

 The bit-perfect issue is currently unresolved. The main issue is that our test equipment tells us one thing and an engineer at Microsoft has told us something else. 

 We use Audio Precision (AP) hardware and software, which is the leading digital and analog audio testing platform. However, our AP interface does not have USB input, so it can only test Kmixer's bit-transparency by proxy via a USB audio device. 

 We have tested several external USB audio interfaces, and the results we found is that the non-native devices all failed AP's bit-transparency test. The native devices that used Windows USB drivers and kmixer, etc, passed the bit-transparent test. 

 The engineer from Microsoft says that Kmixer uses floating point which will never have absolute bit-perfect transmission. We are speaking with them as well as AP to determine where the differences lie.

 Thanks,
 Elias


----------



## Wavelength

Gang,

 Jitter is an in depth conversation that could barry the length of this tread by a factor of 2.

 I totally agree with Thomas on the pit jitter issue. Many transport companies have gone to ATAPI interface instead of SPDIF off the raw transports. This way they can read the data off the CD the same way a computer can. But I have yet to see anyone's transport re-read on an error. Still many companies still rely on the poor SPDIF coming off the raw transport as defacto standard for the data they use. Some are goo enough to reclock this but in many cases this is the cause for poor jitter on the transport end.

 But in declaring low jitter, the power supply has more to do with jitter than almost anything else. Jitter is really dominated by really low frequency noise in the area below 10Hz. Most semiconductor companies that specialize in regulator technology don't even spec below 10Hz because the noise stats sky rocket. But these are exactly the frequencies that effect clocks and audio is full of clocks.

 If we look at the basic preface of digital audio there is in each piece what is called "intrinsic jitter". This is the jitter that a part has no matter how the out side world effects it. This intrinsic jitter can then be amplified by the power supply. You can think of this increase in jitter like a carrier wave. The power supplies noise will basically modulate this jitter to a higher level.

 Let's look at a standard oscillator used for say running an ARSC. The intrinsic jitter for these range in the sub 1ps area to as much as 150ps. Heck if you are making a microwave and need an oscillator 150ps is no big deal. But in audio we need to look to the sub 1ps units. If you talk to the designers of these clocks and ask about power supplies to drive them most of them will talk about elaborate discrete supplies. Guido Tent has some good information up on TentLabs.com. Guido work at Phillips for years in their R&D for noise reduction.

 Standard linear regulators have improved in the noise area but they are still very high in the uV area. Most of them below 100uV but this in it self can cause the intrinsic jitter to amplify by 20-30dB.

 The oscillator companies that spec sub 1ps jitter are using supplies lower than 10nV (below 1Hz) of noise to justify their measurements.

 ~~~~~~~ ARSC's

 It is true that ARSC's can remove allot of the jitter coming in by their simply reclocking at an async rate. But most of the testing on this shows that ARSC's act like low pass filters to the incoming jitter.

 So we have to look at the jitter equation for ARSC's kind of like this:

 Jitter Out = (Jitter in/LP filter) + intrinsic jitter * power supply noise + intrinisc jitter oscillator * power supply noise

 This kind of basic but shows that there is intrinsic jitter in the ARSC as well. Becuase there is tons of clocks all over the place here in both the filtering area as well as the input and output clocking areas. This jitter added to both the oscillator jitter and the attenuated jitter input is going to be applied to the next device in the chain. That being mostly the dac.

 Dac's also have intrinsic jitter... in the same vein.

 As Crowbar mentioned the new ESS part is getting allot of press. Dustin did a killer job on this. I have his earlier ARSC which was very good. The design uses allot of isolating techniques to assure the ARSC section does not invade the dac section and add jitter. Dustin also was awarded a patent for thier jitter reduction which is basically as Crowbar mentioned an ARSC with very low cutoff point.

 Well these are just some things to think about. Sorry if it's too technical, too much coffee on my part.

 Thanks
 Gordon


----------



## EliasGwinn

Quote:


  Originally Posted by *Wavelength* /img/forum/go_quote.gif 
_Elias,

 Why use a fixed clock for the ARCS? Most people who have tested ARSC's feel they sound better when used on a fixed multiple like 2, 4... instead of fractional. 

 So say for 44.1 upsample to 88.2 or 176.4 and so forth.

 In several of the recording studios I work with here (Sound Images, Ashley Shephard's Audio Grotto) feel that when recording that the use of multiples are the key to better results when down sampling. Most of the recordings we have done recently are done at 24/88.2 or 24/176.4 so the output to 16/44.1 sounds better.

 On the dCS gear many have talked about there ability to change the upsampling on the fly and many feel that multiple over fractionals always sound better.

 Your thoughts?
 Thanks
 Gordon_

 

Gordon,

 The fact that the ASRC is asynchronous means that the original sample-rate is completely irrelevent to the SRC process. The data simply registers into a buffer waits there until it is ready to be processed. 

 Thanks,
 Elias


----------



## Wavelength

Quote:


  Originally Posted by *EliasGwinn* /img/forum/go_quote.gif 
_We use Audio Precision (AP) hardware and software, which is the leading digital and analog audio testing platform. However, our AP interface does not have USB input, so it can only test Kmixer's bit-transparency by proxy via a USB audio device._

 

Elias,

 I use the Prism dScope III and it has native output and input. It shows the KMIXER fluctuating error of at least one bit.

 But this subject should be dead for two reason's:

 1) Everyone who has every bypassed the KMIXER has heard better results. So there must be something changed in the KMIXER.

 2) Since you can bypass it, then what's the big deal.

 Well 3 reason's... If Microsoft says so why even try and prove them wrong. They must know a little more about it than you do since they wrote it.

 Thanks
 Gordon


----------



## EliasGwinn

Quote:


  Originally Posted by *03lab* /img/forum/go_quote.gif 
_Hi everyone, first post from a long time lurker and audiophile.

 I'm looking to buy a DAC1 PRE but unfortunately the distributor in Germany and/or Benchmark must have forgotten to do the conversion from USD to EUR. The PRE goes for EUR 1600 around here, which means a hefty $1000 premium on what it sells for in the US, surely something must be wrong here. 
	

	
	
		
		

		
		
	


	




 For $1000 I could easily fly to the US and pick one up myself!

 Please advise as to where I can get one at reasonable cost. Thanks!_

 

03lab,

 We don't control pricing for foriegn dealers, and that has good and bad effects. The bad effect is that some dealers charge too much. The good effect is that you can shop around to different dealers to find a better price. 

 Let them know that, if they want you as a customer, they've got to provide competitive pricing.

 Thanks,
 Elias

 ps. You can begin your search for dealers/distributors here: 

Benchmark Media -- Dealers

 Feel free to investigate dealers outside of your country, as well. For example, EU dealers can sell to anywhere in the EU.


----------



## EliasGwinn

Quote:


  Originally Posted by *spyderx* /img/forum/go_quote.gif 
_Quick question:
 I'm using a DAC1 USB. MacMini USB --> DAC1 Balanced --> BAT VK-300xSE --> Totem Hawk speakers

 I'm using the balanced outs from the DAC1, into the BAT. I have it set for "calibrated" outs and I have not touched the -20db jumper inside the DAC1 or the calibration pots yet.

 I seem to have pretty good range on the volume, with normal listening position around 90 (scale goes to 140 on the BAT). 

 Am I going to benefit SQ by dropping the to 0db attenuation on the DAC1? The BAT has an amazingly low noise floor, so I have no noise at all at these levels.

 Thanks_

 

SpyderX, 

 From your description, it seems your current configuration is setup optimally. The only advantage you would gain by dropping the attenuation of the DAC1 would be a lower noise floor, but that does not seem to be a problem in your case. 

 You can find read more about this here:

Benchmark Media: Feedback Newsletter

 Thanks,
 Elias


----------



## EliasGwinn

Quote:


  Originally Posted by *G-U-E-S-T* /img/forum/go_quote.gif 
_Hi Elias,

 I have a question about the ASRC function and the "jitter-immune" claims of the DAC1 products please.

 As you said, the ASRC function of the DAC1 utilizes its own internal clock (instead of the recovered S/PDIF clock) when internally feeding samples to the DA chip. But before this happens, surely the recovered clock is being utilized by the ASRC algorithm, yes? [Otherwise the ordering of the samples as sent, would be unknown, when trying to reconstruct and zip them into the new sampling rate.]_

 

G-U-E-S-T,

 The ASRC averages the incoming clock to manage the data buffer and determine the SRC ratio. This topology allows this particular chip to achieve jitter attenuation with a corner frequency of 0.1 Hz with a very steep roll-off - effectively attenuating all jitter to levels well below the threshold of hearing (greater then -130 dB).

 Thanks,
 Elias


----------



## EliasGwinn

Quote:


  Originally Posted by *thomaspf* /img/forum/go_quote.gif 
_You have to be careful with this. The original poster spoke about jitter in the recording chain. That is very different from "pit" jitter on a CD.

 The jitter that is permanently added to the signal is created by the analog to digital converter and comes from the fact that the samples are not actually takes at equi distant intervalls but at times in the jitter intervall. This jitter can not be removed._

 

RIGHT! 

 More specifically, the jitter that is present during A-to-D does not get carried on as jitter...it becomes audible distortion. At that point, it is a part of the audio...it is no longer in the clock.

 Thanks,
 Elias


----------



## Crowbar

Quote:


  Originally Posted by *EliasGwinn* /img/forum/go_quote.gif 
_The fact that the ASRC is asynchronous means that the original sample-rate is completely irrelevent to the SRC process._

 

ASRCs vary their performance depending on the exact ratio. There's an explanation by Bruno Putzeys at PSW Recording Forums: Bruno Putzeys => The pros and cons of SRC for jitter reduction. (post #239898)
  Quote:


 Of course the ratio estimator, being an implicit ADC, suffers from quantisation. The phase between the two clocks is quantised to a time span equal to 1 period of the highest frequency clock in the chip....This error is added to the input jitter before being attenuated by the lowpass filter. Whether this effect is detectable at all depends on the spectral distribution of the quantisation error *which in turn depends on the ratio of the input and output clocks.*
 ...if you're using an ASRC as a DAC front end, use an odd ball output frequency to minimise the odds of this happening. 
 

There's a nice AP graph there illustrating it. Additionally, ASRCs using polyphase filters (most of them) have more constraints on the sample rate ratio and performance (though the 130 dB mentioned should be sufficient in theory, plenty of people claim to hear a difference with the 175 dB of Dustin's ASRC).


----------



## EliasGwinn

Quote:


  Originally Posted by *Wavelength* /img/forum/go_quote.gif 
_But in declaring low jitter, the power supply has more to do with jitter than almost anything else. Jitter is really dominated by really low frequency noise in the area below 10Hz. Most semiconductor companies that specialize in regulator technology don't even spec below 10Hz because the noise stats sky rocket. But these are exactly the frequencies that effect clocks and audio is full of clocks._

 

This isn't exactly true. Poorly designed power supplies matched with sensitive components will result in heavy intrinsic jitter, but that is only one face of the beast.

 There are a lot of converters on the market that don't attenuate any jitter below 5 kHz. That means transmission jitter (usually around 2kHz and 400 Hz), interference jitter (@ freq.'s below the cutoff), and intrinsic jitter (@ freq.'s below the cutoff) will be of equal concern with these devices.

 In fact, one could argue that distortion from higher frequency jitter is more detrimental then that of low frequency jitter. This is because low freq jitter has side bands that will be very near the fundamental, and will be masked, to a various degrees, by the fundamental. High freq jitter, on the other hand, will result in side bands that are far enough away that they sound like seperate, independant, yet obnoxious spurious tones. 

 That being said, low frequency jitter should be dealt with properly as well, especially in professional quality converters.

  Quote:


  Originally Posted by *Wavelength* /img/forum/go_quote.gif 
_If we look at the basic preface of digital audio there is in each piece what is called "intrinsic jitter". This is the jitter that a part has no matter how the out side world effects it. This intrinsic jitter can then be amplified by the power supply. You can think of this increase in jitter like a carrier wave. The power supplies noise will basically modulate this jitter to a higher level._

 

I'm not sure if we share the same definition of "intrinsic jitter". In my mind, intrinsic jitter encompasses the jitter from power supply fluxuations and noise, as well as jitter from thermal noise, etc. Not to get into semantics, but its important to make sure we're talking about the same thing.

  Quote:


  Originally Posted by *Wavelength* /img/forum/go_quote.gif 
_~~~~~~~ ARSC's

 It is true that ARSC's can remove allot of the jitter coming in by their simply reclocking at an async rate. But most of the testing on this shows that ARSC's act like low pass filters to the incoming jitter.

 So we have to look at the jitter equation for ARSC's kind of like this:

 Jitter Out = (Jitter in/LP filter) + intrinsic jitter * power supply noise + intrinisc jitter oscillator * power supply noise

 This kind of basic but shows that there is intrinsic jitter in the ARSC as well. Becuase there is tons of clocks all over the place here in both the filtering area as well as the input and output clocking areas. This jitter added to both the oscillator jitter and the attenuated jitter input is going to be applied to the next device in the chain. That being mostly the dac.

 Dac's also have intrinsic jitter... in the same vein._

 

I agree with all of this, except the equation. Poorly designed power supplies can induce and contribute to the intrinsic jitter within clocks, PLL's, transmitters, ASRC's, DAC's, etc. However, they will not amplifiy intrinsic jitter. Modulate, yes...amplify, no.

 But the rest of it is very true. Simply recoverying and re-clocking data is not going to eliminate all jitter artifacts from the conversion. This is why circuit board layout and proper component selection is equally important for a high-resolution conversion system. Cookbook products will give you cookbook results. 

 Thanks,
 Elias


----------



## EliasGwinn

Quote:


  Originally Posted by *Wavelength* /img/forum/go_quote.gif 
_But this subject should be dead for two reason's:

 1) Everyone who has every bypassed the KMIXER has heard better results. So there must be something changed in the KMIXER._

 

Be careful with this. Not only is this not true, we don't want to toss the baby with the bathwater. Subjective, biased listening test should never render scientific testing obsolete (and vice versa).

  Quote:


  Originally Posted by *Wavelength* /img/forum/go_quote.gif 
_2) Since you can bypass it, then what's the big deal._

 

I don't think anyone is making a big deal of it. When someone asks, I will answer.

 But, I agree. This isn't a big deal for two reasons. 

 1. Like you said, if you prefer to bypass kmixer, then you should bypass it.

 2. Bit-perfect or not, kmixer is doing little-to-no damage to the audio. The programmer at Microsoft said basically what you said...the floating point math will cause an error at the 24-th bit. If this error results in any distortion at all, it will be below the threshold of hearing.

  Quote:


  Originally Posted by *Wavelength* /img/forum/go_quote.gif 
_Well 3 reason's... If Microsoft says so why even try and prove them wrong. They must know a little more about it than you do since they wrote it._

 

We're not out to prove Microsoft, or anyone else, wrong. We investigate to learn the truth, no matter which side the truth resides.

 The issue, as I see it, is that either AP needs to readjust their bit-transparency test, or this engineer at Microsoft is overlooking something. I wouldn't bet money on which one is at fault. But, this is a matter of resolving a technical disparity. After all, what kind of engineer can witness such a disparity and just let it be. 
	

	
	
		
		

		
			





 Thanks,
 Elias


----------



## thomaspf

Quote:


  Originally Posted by *EliasGwinn* /img/forum/go_quote.gif 
_RIGHT! 

 More specifically, the jitter that is present during A-to-D does not get carried on as jitter...it becomes audible distortion. At that point, it is a part of the audio...it is no longer in the clock.

 Thanks,
 Elias_

 

Almost 
	

	
	
		
		

		
		
	


	




 If you could playback these samples at exactly the same intervals that you recorded them at bascially recreating te recording jitter at playback time there would be no distortion.

 With a perfect jitter free playback you would be able to isolate the jitter distortions from the recording chain.

 In the real world you have the recording jitter that is permanently encoded in the samples overlaid by the output jitter of the conversion process and the distortions are a complex function of both.

 Cheers

 Thomas


----------



## Wavelength

Quote:


  Originally Posted by *EliasGwinn* /img/forum/go_quote.gif 
_I'm not sure if we share the same definition of "intrinsic jitter". In my mind, intrinsic jitter encompasses the jitter from power supply fluxuations and noise, as well as jitter from thermal noise, etc. Not to get into semantics, but its important to make sure we're talking about the same thing._

 

Elias,

 By definition intrinsic means within. Intrinsic jitter in oscillators is basically the best attainable jitter. The power supply simply agrevates that jitter.

 It can be said that a system has certain intrinsic jitter and therefore would contain the basic components, capacitors, semiconductors and power supply.

 But for example let's look at simple building blocks. The simple 74xx157 quad 2 input multiplexor for years was used as a selction device for many a hardware. Designers would multiplex Master Clock, Bit Clock, Word Clock and DATA from two seperate devices. Later designers found that using 74xx125 (quad low enable buffers) instead of the 74xx157 resulted in a much lower jitter. In further testing it became apparent that using PICOgates (single gates each with their own power and ground) resulted in further lowering of the jitter. This intrinsic jitter has been written about in many text and websites. With the PICOgate investigation it was said that the intrinsic jitter of each gate was multiplied by the association of the other gates and also the routing and supply of the power supply lines.

 This was written about years ago. The semiconductor companies even at one point started making multiple gate technology with the power and ground pins in the center of the part instead of at the opposing corners.

 ~~~~~~

 To me it's much more apparent with power supplies for the low frequency noise. Above a certain point regulators are no even functioning and the impedance and noise is merly controlled by the capacitor. Yes capacitors can have a big influence in jitter and noise. Even the kind of capacitor can make a big difference.

 Of course there is PCB layout issues and stuff but we are getting off target.

 The point is that ARSC's do in fact reclock the data and therefore can remove jitter found on it's inputs.

 All ARSC's will have intrinsic jitter as well as the supplied oscillator. If they didn't the result would be 0 and we all know that's impossible.

 To me using an ARSC as an SRSC, that being an even non fractional multiple of the incoming sample rate (i.e. 2x44.1=88.2 or 4x44.1=176.4) will always sound better than using a fractional.

 Take any dCS unit and play with the sampling on a track and you will agree with me here.

 Heck Elias you do a bunch of recording what's your preface on recording speeds? Again here we always do 88.2/24 or 176.4/24 as it sounds better when we drop it out at 44.1/16.

 Well back to work!
 Thanks
 Gordon


----------



## EliasGwinn

Quote:


  Originally Posted by *thomaspf* /img/forum/go_quote.gif 
_Almost 
	

	
	
		
		

		
		
	


	




 If you could playback these samples at exactly the same intervals that you recorded them at bascially recreating te recording jitter at playback time there would be no distortion.

 With a perfect jitter free playback you would be able to isolate the jitter distortions from the recording chain._

 

Hmmm... 

 ....jitter de-modulation...

 COOL!!


----------



## EliasGwinn

Quote:


  Originally Posted by *Wavelength* /img/forum/go_quote.gif 
_Elias,

 By definition intrinsic means within. Intrinsic jitter in oscillators is basically the best attainable jitter. The power supply simply agrevates that jitter.

 It can be said that a system has certain intrinsic jitter and therefore would contain the basic components, capacitors, semiconductors and power supply._

 

Ok...I see what you mean. I wouldn't use that definition of "intrinsic jitter", but there is no need to debate. I understand what you mean.

  Quote:


  Originally Posted by *Wavelength* /img/forum/go_quote.gif 
_To me using an ARSC as an SRSC, that being an even non fractional multiple of the incoming sample rate (i.e. 2x44.1=88.2 or 4x44.1=176.4) will always sound better than using a fractional.

 Take any dCS unit and play with the sampling on a track and you will agree with me here.

 Heck Elias you do a bunch of recording what's your preface on recording speeds? Again here we always do 88.2/24 or 176.4/24 as it sounds better when we drop it out at 44.1/16._

 

The problem with the "non-fractional" theory is that, with ASYNC sample rate conversion, it is never truly non-fractional. Even if we want to convert 88.2 to 44.1, it is never exactly 88.2 or 44.1. More realistic numbers are 88.20114 kHz and 44.09907 kHz, etc. In order to do exactly-non-fractional, real-time SRC, it would need to be synchronous.

 I record at 96/24. 

 Thanks,
 Elias


----------



## Crowbar

Quote:


  Originally Posted by *EliasGwinn* /img/forum/go_quote.gif 
_The problem with the "non-fractional" theory is that, with ASYNC sample rate conversion, it is never truly non-fractional. Even if we want to convert 88.2 to 44.1, it is never exactly 88.2 or 44.1. More realistic numbers are 88.20114 kHz and 44.09907 kHz, etc._

 

This goes back to the Putzeys post I referenced on the last page, as it's with some of the close in frequencies that the quantization error of the phase between the input and output clocks is largest and so it's best to use a frequency that isn't so close. So I can't agree with Wavelength on using ASRC to approximate a non-fractional SRC with such frequencies.


----------



## ted betley

Ok Elias I have a problem with pops/crackles. I'm using dell xp sp2 cicsplayer -> asio 4all-> opticsis cable-> generic usb cable-> benchmark dac and the sound is fabulous. However I get these pesky pops every now and then (it went away for 3 days after I plugged my mouse/keyboard usbs into the front of the dell leaving my audio usb output all alone except for non used periferalls). I set my latency high in asio4all. Don't know what else to try. Any ideas?


----------



## infinitesymphony

Quote:


  Originally Posted by *ted betley* /img/forum/go_quote.gif 
_opticsis cable-> generic usb cable->_

 

Isn't the Opticis an extension cable? How many feet of USB cable are you using?


----------



## EliasGwinn

Quote:


  Originally Posted by *ted betley* /img/forum/go_quote.gif 
_Ok Elias I have a problem with pops/crackles. I'm using dell xp sp2 cicsplayer -> asio 4all-> opticsis cable-> generic usb cable-> benchmark dac and the sound is fabulous. However I get these pesky pops every now and then (it went away for 3 days after I plugged my mouse/keyboard usbs into the front of the dell leaving my audio usb output all alone except for non used periferalls). I set my latency high in asio4all. Don't know what else to try. Any ideas?_

 

Ted,

 After 3 days, the crackles came back? Did anything change after the 3rd day? Have you tried disconnecting all USB devices?

 Have you tried other audio apps and/or API's?

 If you want, you can call me and we can try to narrow the problem down a little bit more - 315-579-3019.

 Thanks,
 Elias


----------



## ted betley

when is a good time to call


----------



## G-U-E-S-T

Quote:


  Originally Posted by *EliasGwinn* /img/forum/go_quote.gif 
_G-U-E-S-T,

 The ASRC averages the incoming clock to manage the data buffer and determine the SRC ratio...Elias_

 


 Thanks Elias - so if I am understanding correctly, the raw incoming transmission jitter is not actually being eliminated - but rather, the timing accuracy of all the original samples is being re-processed according to the average (mean) accuracy of the incoming clock. So the incoming stream is being fed into an ASRC process which interpolates and creates almost three times as many new samples, has a continuously varying SRC ratio (according to the variable incoming clock average), and also has its own intrinsic process jitter. Which I believe means, that the entire incoming stream is being ultimately converted into something entirely different, digitally, from what was originally recorded and on the CD/DVD. So while it may or may not sound like what was originally on the CD/DVD, nevertheless the trade-off is that whatever this new post-processed digital stream actually is, at least it can be internally re-clocked and further sent in relatively jitter-free fashion to behave well with the DAC chip - and hopefully the DAC1's analog output stage will then also perform better than the original source. Am I getting this correctly?


----------



## thomaspf

That sounds about right. A shorter way of saying this is that an ASRC is not removing jitter but translating it into very low level broadband noise.


----------



## Crowbar

But ASRC allows the filtering of jitter.

 Of course, if you recover a clock by high quality PLL you still filter it.


----------



## chiamingen

Edited


----------



## Wavelength

Quote:


  Originally Posted by *ted betley* /img/forum/go_quote.gif 
_Ok Elias I have a problem with pops/crackles. I'm using dell xp sp2 cicsplayer -> asio 4all-> opticsis cable-> generic usb cable-> benchmark dac and the sound is fabulous. However I get these pesky pops every now and then (it went away for 3 days after I plugged my mouse/keyboard usbs into the front of the dell leaving my audio usb output all alone except for non used periferalls). I set my latency high in asio4all. Don't know what else to try. Any ideas?_

 

Ted,

 A couple of things come to mind. First on your dell all the USB ports are not created equally. Depending on which one you have that could be the whole problem. If you have a desktop machine NEVER use one of the front USB ports. If you are using a laptop and one from the same side as the DVD drive then go to the other side. Also many times there are internal hubs and you could be sharing with other devices. There is a crude program called USBView you can google it and find what the tree looks like on your computer or go into Control Panels and look at the device tree and make sure you are directly connected to one of the HOST ports.

 The other problem is in the Opticis power supply. That thing is like $1.73 power supply and reaks havoc on anything attached to it. What I tell customers is buy a linear 5v regulated (must be regulated). Hack off the 1.7mm cable from the opticis supply and hack off the (usually 2.5mm) connector on the 5V supply and wind them together to get a better supply.

 Guys... Dell computers are cheap for a reason. Don't spend a large chunk of change on the dac end and expect the cheap Dell to do it's job.

 You may also want to make sure you have the latest USB drivers. These are available from the company who makes the chips. Primarily Intel, look up the chip type in the Control Panel Devices and then go to the companies website and download the latest drivers.

 Also make sure at the end of the opticis cable that you are using a good USB 2.0 cable. Believe me it will make a difference.

 Thanks
 Gordon


----------



## ted betley

thanks gordon. I am using a backside usb port and a battery power supply for the opticsis. I just got rid of all the anti virus stuff. We'll see how that works.


----------



## Wavelength

Quote:


  Originally Posted by *Crowbar* /img/forum/go_quote.gif 
_But ASRC allows the filtering of jitter.

 Of course, if you recover a clock by high quality PLL you still filter it._

 

Crowbar,

 With a PLL it is true you are filtering the input control voltage to the VCXO, but the PLL output is not necessarily filtering the jitter like an ARSC would. The PLL output is usually based on some relative input clock. For SPDIF that is usually the Frame Sync. The output PLL of the SPDIF therefore gives you MCLK which is then divided down to Bit Clock and Word Clock.

 In Adapative mode USB the reference becomes the SOF frame which comes at what every the rate of the enumeration is applied (typically 1ms).

 With an ARSC the data is pumped in serially to a parallel register array. The math is done then pumped out using a completely different clock.

 In the past it was thought that only jitter on the Frame Sync (or SOF for Adaptive USB) caused jitter in the system. While true this is where a large precentage of the jitter. But now there is evidence that jitter can develop in many areas of the Digital Audio system. It can be between the DATA and Bit Clock, between Word Clock and Bit Clock.

 There are many ways to filter this. Some designers re-clock the Bit Clock, Word Clock and Data right before the DAC. Sometimes a secondary PLL/VCXO is used to re-clock thereby reducing the clock jitter. Others use secondary PLL/VCXO and use fifo's to buffer the Word Clock and Data into the DAC which has similar results to an ARSC.

 Again Crowbar as you mentioned the ARSC and the fifo method both have the problem of jitter into their systems, some of which cannot be removed.

 It's been kind of interesting to follow some of the Stereophile tests of USB dacs that also support SPDIF and utilize ARSC's. If what most of the designers say were true then the jitter specs for all inputs would be the same. But as we see that is not really the case. Albeit there have been some errors in measurement done there. But still it is very interesting to look at the results.

 Thanks
 Gordon


----------



## Wavelength

Quote:


  Originally Posted by *ted betley* /img/forum/go_quote.gif 
_thanks gordon. I am using a backside usb port and a battery power supply for the opticsis. I just got rid of all the anti virus stuff. We'll see how that works._

 

Ted,

 Dave Clark of Positive Feedback could not get his Dell to work. It always had the same problem you are plagued with. I don't recall what dac he was using but he finnally went to another computer and all's fine.

 Yea part of the problem here is there is little emphasis on quality parts these days.

 You may still want to find a copy of USBView and make sure nothing else is sharing that port (like bluetooth).

 Thanks
 Gordon


----------



## poo

Quote:


  Originally Posted by *chiamingen* /img/forum/go_quote.gif 
_I just got my dac1 yesterday. It sound great like everyone had said
	

	
	
		
		

		
			




 I notice there are no way to switch off the dac, only the standby mode. 
 So whenever i wanted to switch it if i just plugged out the power socket. Is that the right way to switch it off? Will it damage my dac in long run?_

 

Listening to mine now... smooth hey!

 It's designed to be left plugged in, and will go in to standby (as you know) when not in use. No need to turn it off or unplug it (though Greenpeace may suggest otherwise...)


----------



## ted betley

Gordon if I have to I will buy another computer. What did Dave buy to fix his problem? What do you recommend?


----------



## EliasGwinn

Quote:


  Originally Posted by *ted betley* /img/forum/go_quote.gif 
_when is a good time to call_

 

I'm here Mon-Thur, 9a-5p EST. Rory (our sales person) is here M-F, 9-5. 

 1-800-BNCHMRK (800-262-4675). 

 We'll get this figured out or we'll fix it for you.

 Thanks,
 Elias


----------



## EliasGwinn

Quote:


  Originally Posted by *G-U-E-S-T* /img/forum/go_quote.gif 
_Thanks Elias - so if I am understanding correctly, the raw incoming transmission jitter is not actually being eliminated - but rather, the timing accuracy of all the original samples is being re-processed according to the average (mean) accuracy of the incoming clock._

 

You're very close... Think of the digital audio samples as cars, and the ASRC as a toll booth on a pay highway. The cars will be coming up to the booth at all sorts of different speeds, but they will only leave the toll booth when the gate rises and lets them through. 

 The booth operator must let the cars through at a very precise, consistent rate (he has a very fancy stop watch). To acheive a consistent (very low jitter) rate, the booth operator averages the rate at which the cars arrived (60 cars arrived during a 600 second period = 1 car per 10 seconds). Even if the first 10 cars arrived within 50 seconds of each other, they will pass through once every 10 seconds - no more, no less.

 As you can see, this system is immune to micro fluctuations in arrival time up to a certain threshold. This system can be adjusted to a definitive threshold for macro fluctuations. This is why there is a limit for low-frequency jitter rejection. The limit of the UltraLock system in Benchmark converters is 0.1 Hz. The trade-off for such a low cutoff frequency is longer latency (the amount of time between when the sample arrives and when it is converted).

 In other words, the booth operator can increase the amount of time over which his average is calculated so that heavier or lighter times of traffic don't cause fluctuations in his calculations. If he averages over an entire day, rush-hour will not affect his precision, but weekends may throw him off. If he averages over a week, holidays or seasonal travel may throw him off. The larger the time of averaging, the longer the cars have to wait to pass through the booth. Thus, accuracy = latency.

 Latency is only a problem for people (typically musicians) who are trying to monitor real-time input. In other words, if a guitar player is recording guitar through an A-to-D converter, then into the computer, then out through the DAC1, the sound of his guitar may be several milliseconds later then when he actually played it (due to the samples filling up buffers before conversion). This may be distracting for the artist performing.

 For playback, latency is a non-issue. The latency of the buffers in your computer / transport / etc will far out-weigh the latency of the DAC1. If you can't wait an extra 10 milliseconds to hear jitter-free audio, then perhaps you should replace your espresso with herbal tea. 
	

	
	
		
		

		
		
	


	




  Quote:


  Originally Posted by *G-U-E-S-T* /img/forum/go_quote.gif 
_So the incoming stream is being fed into an ASRC process which interpolates and creates almost three times as many new samples, has a continuously varying SRC ratio (according to the variable incoming clock average), and also has its own intrinsic process jitter. Which I believe means, that the entire incoming stream is being ultimately converted into something entirely different, digitally, from what was originally recorded and on the CD/DVD. So while it may or may not sound like what was originally on the CD/DVD, nevertheless the trade-off is that whatever this new post-processed digital stream actually is, at least it can be internally re-clocked and further sent in relatively jitter-free fashion to behave well with the DAC chip - and hopefully the DAC1's analog output stage will then also perform better than the original source. Am I getting this correctly?_

 

EXACTLY!! 

 To acheive jitter-immunity, the signal must be re-sampled. Therefore, the trade-off is whatever noise and/or distortion is caused by the ASRC. The ASRC used in the DAC1 is the AD1896, so we are talking about -133 dB distortion and 142 dB dynamic range!!! In other words, beyond the threshold of hearing.

 The DAC1's analog stage performs one function: condition the analog output of the D-to-A converter to the appropriate voltage and impedance for driving the next device. It is designed to do this with the lowest amount of noise and the least amount of sonic modification as possible.

 Thanks,
 Elias


----------



## EliasGwinn

Quote:


  Originally Posted by *chiamingen* /img/forum/go_quote.gif 
_I just got my dac1 yesterday. It sound great like everyone had said
	

	
	
		
		

		
		
	


	



 I notice there are no way to switch off the dac, only the standby mode. 
 So whenever i wanted to switch it if i just plugged out the power socket. Is that the right way to switch it off? Will it damage my dac in long run?_

 

Hello Chaimingen,

 Glad to hear that you're enjoying the DAC1. 

 Removing the power cable will not damage the DAC1, but it is unnecessary to power it down. It is designed to withstand 24-hr/365-day operation.

 Thanks,
 Elias


----------



## ted betley

This is additional/clarifying data on my problem (ref voice mail to you at 9:00am this am). The problem was very intense last night -- pops and crackles very frequent and very intense. This was good for switching things. Things I could switch easily were players and driver interfaces (cicsplay/ j river and asio4all vs direct sound). Problem was bad on cicsplay/asio4all bad but not as bad with j river/asio4all; problem went away with j river/direct sound. So eureka I thought I found it. This am when I turned on my audio sys there was no signal into the Benchmark at all. It was lost sometime overnight while not playing anything. So I rebooted this am, re inserted usb cables at both ends and got my signal back into the Benchmark (two light came back on). I tried cicsplay/asio4all got pops, tried j river / ds and got one pop. I rebooted again and got j river/ds to play ok for one hour straight. I'm thinking now that because of the overnight phenomena the problem might be the opticis. So later today I will bypass the opticis and go straight from dell -> Benchmark with usb cable (the one that came with the Benchmark, by the way is this a belden?). We'll see what happens. Any comments would be appreciated.


----------



## EliasGwinn

Quote:


  Originally Posted by *ted betley* /img/forum/go_quote.gif 
_This is additional/clarifying data on my problem (ref voice mail to you at 9:00am this am). The problem was very intense last night -- pops and crackles very frequent and very intense. This was good for switching things. Things I could switch easily were players and driver interfaces (cicsplay/ j river and asio4all vs direct sound). Problem was bad on cicsplay/asio4all bad but not as bad with j river/asio4all; problem went away with j river/direct sound. So eureka I thought I found it. This am when I turned on my audio sys there was no signal into the Benchmark at all. It was lost sometime overnight while not playing anything. So I rebooted this am, re inserted usb cables at both ends and got my signal back into the Benchmark (two light came back on). I tried cicsplay/asio4all got pops, tried j river / ds and got one pop. I rebooted again and got j river/ds to play ok for one hour straight. I'm thinking now that because of the overnight phenomena the problem might be the opticis. So later today I will bypass the opticis and go straight from dell -> Benchmark with usb cable (the one that came with the Benchmark, by the way is this a belden?). We'll see what happens. Any comments would be appreciated._

 

I would guess that the USB extender is not the culprit. If you are making changes in your software, and it is affecting the performance, then I am very certain that the problem lies internally. There is some software conflict happening with the media app(s) and/or asio4all and/or hardware management software (drivers, etc). 

 Thanks,
 Elias


----------



## ted betley

...and other than the hum it's not clicking. I'll let it run a few days to see/hear what happens.

 Other suggestions are to defrag/ get a pci sound card with dedicated usb / toslink drivers and see if that solves it.

 Comments?


----------



## EliasGwinn

ted betley;4288107 said:
			
		

> ...and other than the hum ...QUOTE]
> 
> Now there's a hum??


----------



## ted betley

I fixed the hum with a cheater plug---I'm on the same ac circuit as my pc now.


----------



## EliasGwinn

Quote:


  Originally Posted by *ted betley* /img/forum/go_quote.gif 
_I fixed the hum with a cheater plug---I'm on the same ac circuit as my pc now._

 

This is never a good idea. I would recommend re-instating the ground pin and using the same outlet that the amplifier is using. You should avoid any ground loops that way.

 Thanks,
 Elias


----------



## eweitzman

Quote:


  Originally Posted by *ted betley* /img/forum/go_quote.gif 
_Ok Elias I have a problem with pops/crackles. I'm using dell xp sp2 cicsplayer -> asio 4all-> opticsis cable-> generic usb cable-> benchmark dac and the sound is fabulous. However I get these pesky pops every now and then (it went away for 3 days after I plugged my mouse/keyboard usbs into the front of the dell leaving my audio usb output all alone except for non used periferalls). I set my latency high in asio4all. Don't know what else to try. Any ideas?_

 

Ted,

 I had a similar problem. Yours might be related.

 I was planning on using my DAC1PRE with an IBM X22 laptop made in 2003 and running xpsp2. Unfortunately, the system was plagued with pops/crackles and, at high bandwidths, dropouts for several seconds. After going through a litany of driver/port/software/hardware elimination with Elias, no solution was forthcoming and I nearly gave up. After doing some digging around into USB problems, I found software tools to monitor the timing and duration of the computer's interrupts and deferred procedure calls (DPCs, interrupt handlers that get queued to run later when hardware interrupts occur). This way, I was able to isolate the problem to a few devices that were making long-running calls into the kernel. It turned out that ACPI.SYS was the culprit, hanging the machine for unacceptably long time periods -- in interrupt-handling timeframes, which must be very very short -- and (presumably) causing USB packets to be dropped.

 ACPI.SYS isn't your garden variety driver, though: It's part of the hardware abstraction layer (HAL) that is installed at a very early phase during a clean OS install. ACPI is the power management system that allows for, among other things, system hibernation and waking upon hardware events like LAN wakeup signals and pressing the power button. There is a technique to turn it off beneath an XP installation, but this is not recommended and the OS should be reinstalled clean. There's a magic keypress to make during the installation of XP that lets you choose the standard HAL instead of accepting the ACPI HAL. I created a new partition on my laptop, then installed XP without the ACPI HAL, and all USB problems have vanished. The power management functions that we're all used to are no longer available, however.

 The weird thing about this is that my IBM T23 laptop manufactured exactly one month after the X22 uses the same Intel USB chipset and it works fine with the DAC1PRE!

 I should mention that I went through this several months ago. If anyone needs more details, speak up and I'll dig out the nitty gritty and post it here later.

 - Eric


----------



## EliasGwinn

Quote:


  Originally Posted by *eweitzman* /img/forum/go_quote.gif 
_The weird thing about this is that my IBM T23 laptop manufactured exactly one month after the X22 uses the same Intel USB chipset and it works fine with the DAC1PRE!
_

 


 Apple is going through a spell of this right now. Almost all of their new MacBook and MacBook Pro's are not compatible with most Firewire audio interfaces. No one knows why, and Apple isn't talking about it much.

 But the real kick-in-the-pants is that every now and then, one of these computers will work perfectly fine with all firewire audio interfaces!! There doesn't seem to be any reason why the one would work, and so many don't, but... 
	

	
	
		
		

		
			





 It just goes to show that no two computers are alike, even if 99.9% of the hardware and software are similar. 

 BTW, Eric, great job figuring out that power management bug. That was such a deeply burried bug, but you widdled it out! Big thumbs up!!

 Also, thanks for offering to help others sort through their computer's issues. Thats what makes the online community so powerful. Hopefully HAL doesn't catch on to Head-Fi!! 
	

	
	
		
		

		
		
	


	




 Thanks,
 Elias


----------



## ted betley

To eweitzman: yes I would like to know more, specifically what I can try to rid my sys of these nasty pops. I did take out my opticis cable and am running pop free but I won't declare victory until I get 3 days of no pops.


----------



## Wavelength

Ted,

 One thing that has been a problem for some PC's is that the USB driver is generic instead of specific.

 If you go to Control Panels: Device Manager : Universal Serial Bus controllers

 You can then click on the Host Controller and look under the Driver and make sure it was written by the company and not the Microsoft generic one.

 Also you can check to see if it is the latest one. Here you can also kinda tell were your dac is plugged into.

 On a side note with the Opticis. I use these at shows all the time. I have never really had a problem with these. I use a custom linear regulated supply on mine. But several users are using the battery supply like yourself.

 Elias,

 HUH... on the Firewire brand new MacBook Pro into Metric Halo ULN2 worked fine yesterday after we updated the DSP and firmware in the ULN2.

 As for the cheater plug thing it should not be a problem to remove this. The DAC1 does not constitute allot of current. The floating ground will then permit the single point ground at the preamp/integrated amplifier.

 Shun Mook who have a pretty radical idea of grounding float everything to one central point. Then they run a 18ga wire to a post in the ground. They also put these ferric sleves around all power lines. Funny story Jonathan Valin of TAS had them do his house. The power company terminated service when they saw the ferric sleeve (bright purple and red) around the incoming service.

 Gang I have seen allot of Computers causing ground loops. Some have 2 wire plugs that are polarized and some 3. In general all the items hooked together will require the same power from the same outlet. Even common power on seperate outlets can cause hum. It's best to get a good power stripe and plug everything into it.

 Thanks
 Gordon


----------



## ted betley

Gordon:
 One thing that has been a problem for some PC's is that the USB driver is generic instead of specific.

 If you go to Control Panels: Device Manager : Universal Serial Bus controllers

 You can then click on the Host Controller and look under the Driver and make sure it was written by the company and not the Microsoft generic one.
 [under control panel ... it says Generic usb audio; driver provider microsoft]

 Also you can check to see if it is the latest one. Here you can also kinda tell were your dac is plugged into.
 [How do I tell if it is the latest one?]

 On a side note with the Opticis. I use these at shows all the time. I have never really had a problem with these. I use a custom linear regulated supply on mine. But several users are using the battery supply like yourself

 [Once while I was using the opticis I demagged a cd while being roughly 3 feet from a portion of the opticis and I heard an induced pop directly correlated with my demagging action. Yes the demag device was powered by the same a/c circuit as my entire front end--dac,opticis battery charger, dbx crossover]


----------



## eweitzman

Quote:


  Originally Posted by *ted betley* /img/forum/go_quote.gif 
_To eweitzman: yes I would like to know more, specifically what I can try to rid my sys of these nasty pops. I did take out my opticis cable and am running pop free but I won't declare victory until I get 3 days of no pops._

 

Ted,

 I'd recommend *against* removing ACPI unless you're sure it's the problem because removing it requires major surgery on your operating system. I'd recommend diagnosing the problem in detail by running RATTV3. Google for it and you'll find it at Microsoft's website. RATTV3 is not a simple system to run, but once you get the hang of it, it will reveal any latency problems in your system.

 A better understanding of latency issues can be gained by reading about and then running a program called "DPC Latency Checker". You can get it here. It's not as good a tool for pinpointing the problem but will reveal if your machine has latency problems.

 - Eric


----------



## eweitzman

Here's my email to Elias from March after I solved the latency/pop mystery on my five year old IBM X22 laptop. The procedures involved apply to Windows XP only AFAIK.

 - Eric

  Code:


```
[left]Elias, I have great progress to report. Complicated, but great. I can use the DAC1PRE with my old IBM X22 laptop without dropouts. I created a new partition on my X22 laptop's hard drive and installed a clean XPSP2. Still had dropouts. Turned off services, etc, still had dropouts. Seems the "reduce load and complexity" approach doesn't work with this machine. So I turned on the DPC/latency monitor program RATTV3 for a very short time while playing a 24/96 file that normally drops out for 5-10 seconds at a time and got a snapshot of just a few seconds of the machine when a long dropout occurred. The log shows very long times spent handling interrupts in ACPI.sys. This is the driver for the Advanced Configuration and Power Interface. ACPI allows the OS to control the power management functions of the hardware. Searching around the net, I found that disabling ACPI was a hot topic about five years ago. It is not a simple driver that can be disabled: it's part of the hardware abstraction layer (HAL) that the OS runs on top of. When installing XP, it detects the machine type and installs the HAL before installing anything else. HALs can be selected during setup with a judiciously pressed F5 when setup is prompting you to press F6 to install additional device drivers. This is the clean way to not have ACPI: select "Standard PC" at this point and no ACPI support will be installed. Alternatively, after the OS is installed with the ACPI HAL, you can expand the "Computer" node in Device Manager to the ACPI computer "device", then open it's properties, update (or remove?) the driver, select the "Standard PC" HAL and reboot. XP will then redetect much of the hardware after starting, reassign IRQs and so on, then it needs another reboot. After that, there's no ACPI layer, and probably a mangled hardware hive in the registry. In any case, without ACPI (clean install or removed), the DAC1 runs trouble-free with XPSP2 on this machine. But there's a price: no XP power management functionality, no hibernate and standby support. The older APM power management functions can be turned on (at least on my laptop) without ACPI so there's some power management when running on battery, but still no hibernate/standby. These are hardware functions availalbe via ACPI only. Google for "disable ACPI XP" and you'll find detailed procedures for removing it. Here's Microsoft's note on forcing install to use the Standard PC HAL instead of the ACPI HAL: http://support.microsoft.com/kb/299340/en-us - Eric[/left]
```


----------



## ted betley

cool I'll try it


----------



## ted betley

I d/l ed dpc latency checker and already noticed a spike that took me right up to the red line (max latency of 1932 usec). I use cicsplay which is a ram playback system and since I have 1 gig of memory I usually analyze windows task manager and end processes not critical to playback while during playback. So when I got my 1932 spike was when I ended msmgs.exe (I never the use message manager how do I disable completely?)/ Maybe my process ending process is spike inducing?


----------



## ted betley

To eweitzman: I got dpc latency checker and yes something is amiss somewhere. Some dpc's > 4000usec. I d/l ed ratvv3 and it is apparently open and running but how do I get the report? Does it just come or must I query for it? Where is the report deposited?


----------



## eweitzman

Ted,

 IIRC, the RATTV3 report file is re-created automatically after each run cycle, about every 3 minutes. The report is created in the %SYSTEM32%\logfiles\rattv3 directory. Check the README file in the RATTV3 group in the start menu to be able to interpret the report. For example, the report file on my machine is C:\WINDOWS\system32\LogFiles\RATTV3\MTW.cswa-accumulator-report.txt.

 Another "IIRC"-qualified answer is how to turn off msmsgs.exe, aka Windows Messenger. Even if you shut it off using the Services management console, XP will turn it back on next reboot. If you never use it, just uninstall it. Go to Control Panel > Add or Remove Programs > Add or Remove Windows Components. Uncheck "Windows Messenger," press "Next" and follow along You might need to have your XP installation CD handy.

 - Eric


----------



## ted betley

Got the file accumulator report but the format/text is not decipherable. Any hints on how to open? When it asked me what program made the report I answered notepad. I guess that was not right.


----------



## eweitzman

Quote:


  Originally Posted by *ted betley* /img/forum/go_quote.gif 
_Got the file accumulator report but the format/text is not decipherable. Any hints on how to open? When it asked me what program made the report I answered notepad. I guess that was not right._

 

Never noticed that since I use emacs. It's a text file alright, but it has unix line endings instead of DOS line endings. I would have thought Notepad would be smart enough... but MS Word can open it.

 - Eric


----------



## ted betley

ok thanks I'll stop restart and try again


----------



## ted betley

It turns out when you print the cumulative report it comes out ok . So what was your criteria for identifying a problem, ocurrences > 1000000 usec or high frequency of lower latencies?


----------



## Audio-Omega

Does it improve the sound quality of a CD player by much ? I know that depends on the quality of the CD player for a start. Or do you guys prefer the sound quality straight from a CD player ? Sorry if this has been raised.


----------



## ted betley

I have been monitoring for 1 day and max latency I get is 4-5000 usec which to me is .05 sec. I have not heard any glitches yet maybe the opticis--out of sys for 3 days now--was the problem. I do get a lot of activity on one usb driver but none exceeding 4000 u sec. Eweitzman was your problem id'd with latencies over 1 second?


----------



## ted betley

What happens to the opticis when you have bends with radii less than 3 inches? Maybe that was my problem?


----------



## stax#1

The Benchmark USB DAC is a very disappointed piece of audio gear. I am not going to explain about all the negative things about this DAC. You are better off spending a little more money on a Hagerman USB DAC. I have the Hagerman Chime DAC, but Jim Hagerman is no longer producing this wonderful DAC. Jim has a new USB DAC (DA-10) which will probably out perform the Chime DAC. The only issue I have with the new DAC is it doesn't have a rectifier tube. If you are interested in a Chime DAC, once in a blue moon, you will see one in audiogon. So, before you commence in purchasing a benchmark, you might want to consider a Hagerman USB dac. This will save you time and money when you decide the benchmark is not all that it talks up to be. Oh yeah, I forgot to mention that I replace all my tubes with mullards and this made the Chime DAC out perform all the DAC's I have ever owned. Good Day! David


----------



## infinitesymphony

Quote:


  Originally Posted by *stax#1* /img/forum/go_quote.gif 
_The Benchmark USB DAC is a very disappointed piece of audio gear. I am not going to explain about all the negative things about this DAC. You are better off spending a little more money on a Hagerman USB DAC. I have the Hagerman Chime DAC, but Jim Hagerman is no longer producing this wonderful DAC. Jim has a new USB DAC (DA-10) which will probably out perform the Chime DAC. The only issue I have with the new DAC is it doesn't have a rectifier tube._

 

What was wrong with the DAC1? It sounds like you just prefer tube gear...

 Doesn't the assembled version of the Hagerman DA-10 cost around $2,300? That's almost double the cost of the DAC1 USB.


----------



## ted betley

I love the Benchmark; I don't have a problem with it I have a problem with the opticis fiber optic cable.


----------



## ted betley

Well I believe problem is solved --either the opticis is defective or my dell can't push the usb/opticis/usb cable chain (good news--thanks elias this was your suggestion) but the bad news is I cant use the opticis. I had the Benchie parked in my system such that I could use 1 meter balanced ic's to my front end electronics. Also I could place my oppo (modded by shawn fogg) close to the Benchie while plugging the oppo pc into my shunyata. Now I can't. 

 Mr eweitzman I did get very good latencies nothing over 5-6000 usec even though the dpc latency checker did say this might be a problem, it never was and I did learn a lot so thank you too. Even though the latency checker said that I was pushing things with 4-5000 usec ocurences I never got a pop or crackle. I have been pop free for 3 days. While I'm happy I solved my problem I really need a good usb extender but I tried 2 now (one the opticis recommended by gordon) and one a pure hub extender recommended by elias)and neither worked for me. I'll have to figure out how refit my components to get max sound quality. I know I'll need at least a 10 meter pair of double cross mogamis. Oh well but while driving my front end with garden variety unbalanced ic's the benchie still sounds great. I hope somebody else can learn from my experience.

 Live learn and spend a little $.


----------



## LIVE_EVIL

Is there a black faceplate DAC1 PRE or they are all silver?


----------



## EliasGwinn

Quote:


  Originally Posted by *LIVE_EVIL* /img/forum/go_quote.gif 
_Is there a black faceplate DAC1 PRE or they are all silver?_

 

Hello Live Evil,

 We only make silver DAC1 PRE's. 

 Thanks,
 Elias

 ps. Your username is such a great album...


----------



## LIVE_EVIL

Hi Elias,
 Thanks a lot for the answer. 

 Yeah, this live album is really great. Black Sabbath started for me from this record. "Heaven and Hell" and "The sign of the Southern Cross" contribute to such an epic performance.


----------



## I-Love-Music

Never tried it, I will though.


----------



## erotisches

If DAC1 comes with remoter control for source switching, that would be great.


----------



## 03lab

Finally managed to get a DAC1 Pre for under $2500 (
	

	
	
		
		

		
			





) and it does sound great, but I was wondering if it's normal for the DAC to get very warm even if it is just plugged in but turned off?


----------



## Matias

I've just bought HiFi-Tuning ceramic fuses and 2 Noise Destroyers.
 I strongly suggest that DAC1 users try at least the fuses. They did wonders with it! Sub-bass is a lot deeper, as well as the soundstage and focus/pin-point.

 Definitively worth the 13 euros!


----------



## EliasGwinn

Quote:


  Originally Posted by *erotisches* /img/forum/go_quote.gif 
_If DAC1 comes with remoter control for source switching, that would be great._

 

Thanks for the suggestions, erotisches!


----------



## EliasGwinn

Quote:


  Originally Posted by *03lab* /img/forum/go_quote.gif 
_Finally managed to get a DAC1 Pre for under $2500 (
	

	
	
		
		

		
		
	


	




) and it does sound great, but I was wondering if it's normal for the DAC to get very warm even if it is just plugged in but turned off?_

 

Hey 03lab, 

 I'm glad to hear you were able to find a modestly priced unit. 

 Putting your hand on the DAC1 PRE should feel like the temperature of a mug of coffee. This will be the case even when it is in standby mode, as this mode does not actually disengage the power to the unit. It simply puts the unit in 'standby'.

 Thanks,
 Elias


----------



## Bostonears

Quote:


  Originally Posted by *Matias* /img/forum/go_quote.gif 
_I've just bought HiFi-Tuning ceramic fuses and 2 Noise Destroyers.
 I strongly suggest that DAC1 users try at least the fuses. They did wonders with it! Sub-bass is a lot deeper, as well as the soundstage and focus/pin-point.

 Definitively worth the 13 euros! 
	

	
	
		
		

		
		
	


	


_

 

If, as EliasGwinn has stated, the fuses are not in the signal path for the DAC1 series, how would different fuses possibly affect the sound? If the original fuses were limiting current draw, wouldn't that have blown the fuses?


----------



## Crowbar

Fuses are always in the signal path, since the output signal of any audio equipment is power supply current modulated by the input signal. Filtering input power is different from saying power supply-side stuff is not in the signal path--technically it is. The electrical signal that comes out of audio equipment first came into it over the power cable.


----------



## 03lab

Quote:


  Originally Posted by *EliasGwinn* /img/forum/go_quote.gif 
_Putting your hand on the DAC1 PRE should feel like the temperature of a mug of coffee. This will be the case even when it is in standby mode, as this mode does not actually disengage the power to the unit. It simply puts the unit in 'standby'._

 

Phew, glad to hear this is normal, I was afraid I'd have to send the unit back (getting this particular DAC1 was quite an ordeal!). 

 So far I absolutely love my DAC1 PRE, which is now driving my active ADAM's and my HD600 wirelessly from my MacBook Pro via an Airport Express. It's the perfect minimalist setup I've always wanted!


----------



## Bostonears

Quote:


  Originally Posted by *Crowbar* /img/forum/go_quote.gif 
_Fuses are always in the signal path, since the output signal of any audio equipment is power supply current modulated by the input signal. Filtering input power is different from saying power supply-side stuff is not in the signal path--technically it is. The electrical signal that comes out of audio equipment first came into it over the power cable._

 

Have you upgraded the fuses in any of your audio gear and heard a difference?


----------



## Crowbar

Of course not. The power supply should have a high enough rejection ratio to filter any contribution from the fuses. That doesn't take away from what I wrote, however--I worded myself very carefully.


----------



## Matias

Crowbar, you should try them for youself. They are somewhat cheap, for such a difference it makes, defitively worth trying.


----------



## Matias

Quote:


  Originally Posted by *Bostonears* /img/forum/go_quote.gif 
_If, as EliasGwinn has stated, the fuses are not in the signal path for the DAC1 series, how would different fuses possibly affect the sound? If the original fuses were limiting current draw, wouldn't that have blown the fuses?_

 

They are not in the signal path, ok, but they are directly feeding all the analog section's power. Don't ask me the technical stuff. Use your ears instead.


----------



## Bostonears

Quote:


  Originally Posted by *Matias* /img/forum/go_quote.gif 
_They are not in the signal path, ok, but they are directly feeding all the analog section's power. Don't ask me the technical stuff. Use your ears instead. 
	

	
	
		
		

		
		
	


	


_

 

And the difference you hear with these fuses, is it only through the analog outputs (RCA and/or XLR), or is it also through the headphone jacks?


----------



## bsckwan

Is it possible that the 2 input could sound different with coaxial sounding a little bit louder or better?


----------



## Matias

Bostonears, analog outs and headphone jacks both.

 bsckwan, I said that too some time ago. Correct bit-transparent SPDIF sound better/cleaner then the USB implementation.


----------



## ted betley

Matias: question I have a usb benchmark with asio4all which limits me to 24/48 going to the dac. How do you find juli--any problems or difficulties? Is the digital out good (what are you using-toslink or spdif)? Have you tried some of the 24/96 and 24/192 downloads?


----------



## ted betley

Elias can the benchmark be used with asio that has 24/192 capabilities? I currently use with asio4all which was free and works well but only up to 24/48. I would like to try to use some 24/96 and 24/192 downloads but cannot with asio4all


----------



## EliasGwinn

Quote:


  Originally Posted by *Matias* /img/forum/go_quote.gif 
_They are not in the signal path, ok, but they are directly feeding all the analog section's power. Don't ask me the technical stuff. Use your ears instead. 
	

	
	
		
		

		
			



_

 

Actually, the fuses aren't directly feeding the analog circuitry either. After the AC is delivered through the fuses, it is heavily filtered and regulated with our power supply circuitry. 

 Thanks,
 Elias


----------



## Dreadhead

Quote:


  Originally Posted by *ted betley* /img/forum/go_quote.gif 
_Elias can the benchmark be used with asio that has 24/192 capabilities? I currently use with asio4all which was free and works well but only up to 24/48. I would like to try to use some 24/96 and 24/192 downloads but cannot with asio4all_

 


 I will let Elias deal with the Benchmark question but asio4all supports 24/96 if the device it's attached to supports it. I listen to 24/96 all the time on my through optical out of a m-audio transit (up to 24/96). My stand alone digital equalizer even confirms that the data is 24/96.

 Cheers,
 Chris


----------



## EliasGwinn

Quote:


  Originally Posted by *bsckwan* /img/forum/go_quote.gif 
_Is it possible that the 2 input could sound different with coaxial sounding a little bit louder or better?_

 

Have you checked all the volume controls in your computer? What operating system are you using? Windows, Mac?

 Thanks,
 Elias


----------



## EliasGwinn

Quote:


  Originally Posted by *ted betley* /img/forum/go_quote.gif 
_Matias: question I have a usb benchmark with asio4all which limits me to 24/48 going to the dac. How do you find juli--any problems or difficulties? Is the digital out good (what are you using-toslink or spdif)? Have you tried some of the 24/96 and 24/192 downloads?_

 

The DAC1 USB is capable of up to 24/96. Is ASIO4ALL limited to 24/48?

 Thanks,
 Elias


----------



## ted betley

yes I believe so. I can play 24/96 flac but not wav files


----------



## EliasGwinn

Quote:


  Originally Posted by *ted betley* /img/forum/go_quote.gif 
_Elias can the benchmark be used with asio that has 24/192 capabilities? I currently use with asio4all which was free and works well but only up to 24/48. I would like to try to use some 24/96 and 24/192 downloads but cannot with asio4all_

 

Well, this answers my previous post 
	

	
	
		
		

		
		
	


	




 The DAC1 USB is capable of 24/192 with all digital inputs EXCEPT USB. USB is limited to 24/96.

 If you have an ASIO-based computer interface that transmits digitally at 24/192, then the DAC1 will work great with it. (assuming that the hardware and/or the software associated with it does not mangle the digital data).

 Thanks,
 Elias


----------



## Matias

ted betley, indeed, 24/192 through SPDIF works great here. Maybe I'll try those new HRx later. 
	

	
	
		
		

		
		
	


	





 EliasGwinn, even though the analog section is heavily filtered, as well as the USB implementation is supposed to be bit-identical and jitter-free, my ears tell me otherwise. 
	

	
	
		
		

		
		
	


	




 I would be great if you could measure those differences and see it for yourself.


----------



## Matias

03lab,

 My DAC1 USB stays warm after the 15s power-off too, burning away electricity and raising my power bill, definitively not an eco-friendly solution. The only way to completely shut if off, that is, not staying warm afterwards, is disconnecting the AC cable! 
	

	
	
		
		

		
		
	


	




 Doing it everyday is sure to wear off the connection, and so you have to decide between energy saving versus equipment life-time. No comments...

  Quote:


  Originally Posted by *03lab* /img/forum/go_quote.gif 
_Finally managed to get a DAC1 Pre for under $2500 (
	

	
	
		
		

		
		
	


	




) and it does sound great, but I was wondering if it's normal for the DAC to get very warm even if it is just plugged in but turned off?_


----------



## Crowbar

Quote:


  Originally Posted by *Matias* /img/forum/go_quote.gif 
_03lab,

 My DAC1 USB stays warm after the 15s power-off too, burning away electricity and raising my power bill, definitively not an eco-friendly solution. The only way to completely shut if off, that is, not staying warm afterwards, is disconnecting the AC cable! 
	

	
	
		
		

		
		
	


	




 Doing it everyday is sure to wear off the connection, and so you have to decide between energy saving versus equipment life-time. No comments..._

 

Do you drive or take the bus or another gasoline or diesel using vehicle? A liter of gasoline is 9,000 watt-hours, 11,000 for diesel. Using a vehicle to go a single kilometer pales in comparison to electronic equipment's standby power for a whole day.


----------



## yourmando

A question for Elias:

 I've been meaning switch from using the rca outputs to using the XLR so that I can pad down the output volume. Unfortunately, most of time I find the volume know at around 9 o'clock at critical listening levels, sometimes even less (and more so when I have music at a 'work' volume). I'd rather have the volume closer to 12 o'clock during critical listening. I know I can use a special XLR to RCA cable with a floating pin. There's also the option of using a normal XLR cable and a Jensen xlr to rca transformer box, which also brings the "Pro" +4dBu signals to "Consumer" -10dBV signals. Can you help me decide which route to take?

 Ordinarily, I would just go with a short length of the XLR cable with the floating pin and connect it to my existing RCA cables. But the other day I noticed some loud hum from the listening position. I use the dac1 usb as my preamp, and it's the only thing connected to my speakers (no cable box or other sources). I finally narrowed down the problem--if I plug the dac1 usb into a different wall socket from the other components, the hum mostly goes away. (Still audible from near the speakers, but I don't mind that). The 2 different wall socket situation is not ideal because I have 2 power strips coming in across different sides of the room. A bit messy. And my options are limited because I need the dac1 in a certain position to be near the volume control and headphone out.

 So I thought the Jensen solution might kill 2 birds with 1 stone: 1) eliminate hum in cases where I need to plug into the same power strip from the same wall socket (I'm not sure if this is true), 2) allow me to convert to rca w/o a special cable, and bonus 3) give me more room to pad down, since it pads down to bring signals to a consumer level, and then I can pad down again using the internal xlr jumper if I need to. The drawbacks are cost, added distortion (not sure if it would ever be audible), and the fact that it might not have the intended effect on hum.

 The setup: DAC1 USB to Linkwitz Lab Orion+ (dac1 rca interconnects to active crossover, and from there multiple rca interconnects to the amp channels...). (And of course the headphone out, but I've never had hum or other problems with that.)

 The Jensen transformer:

JENSEN TRANSFORMERS, INC. - ISO-MAX® PC-2XR Stereo "Pro" to "Consumer" Converter / Isolator

 Thank you,
 Armando


----------



## poo

Quote:


  Originally Posted by *EliasGwinn* /img/forum/go_quote.gif 
_The DAC1 USB is capable of 24/192 with all digital inputs EXCEPT USB. USB is limited to 24/96._

 

DAMN! I didn't realise that (or at least forgot). Is that a limitation of USB within the DAC1, or USB in general? Can I ask for a simple explanation as to why this is the case? Something to do with bandwith?


----------



## erotisches

One question, if my PC is also a source for home theater, can dac1 be used simultaneously with the sound card? I also need digital out for the surround channel whereas ,if all output is fed through dac1, I will only have 2ch output. Digital output would be nice too 
	

	
	
		
		

		
		
	


	




 ahhh I'm so greedy.
 Sorry if this is answered in FAQ, I admit that I haven't read it.


----------



## Bostonears

Quote:


  Originally Posted by *Matias* /img/forum/go_quote.gif 
_My DAC1 USB stays warm after the 15s power-off too, burning away electricity and raising my power bill, definitively not an eco-friendly solution. The only way to completely shut if off, that is, not staying warm afterwards, is disconnecting the AC cable! 
	

	
	
		
		

		
		
	


	




 Doing it everyday is sure to wear off the connection, and so you have to decide between energy saving versus equipment life-time. No comments..._

 

According to the specs, the DAC1 uses about 8 watts of energy in standby. That works out to about one kilowatt hour every 5 days, which in most places would cost less than $1 a month on an electric bill.

 Nevertheless, if you want to disconnect the DAC1 when not in use, a better way than actually pulling the plug would be via a switched outlet on a power strip or AC conditioner. Personally, I would like to see Benchmark put a real power switch on the DAC1, or a 12 volt trigger that would permit lower standby power consumption (possibly under 1 watt).


----------



## bsckwan

Quote:


  Originally Posted by *Matias* /img/forum/go_quote.gif 
_Bostonears, analog outs and headphone jacks both.

 bsckwan, I said that too some time ago. Correct bit-transparent SPDIF sound better/cleaner then the USB implementation._

 

I've been comparing the following 2 configuration:

 1.) Mac OS X 10.5.3 with iTunes 7.6.2 playing AIFF files --> (via USB with 24 bit / 44.1 kHz setting in Audio Midi and standard USB cable) Benchmark DAC1 PRE
 2.) Sony DVP-S9000ES --> (via coaxial) Benchmark DAC1 PRE

 I wanted 1. to sound better but 2. always end up sounding cleaner. 

 I thought 1. is suppose to be bit transparent.

 A question for Elias, 

 Is there anyway to improve on 1 and whether this is usually the case with DAC1 - SPDIF better then USB?


----------



## Matias

bsckwan, I removed the player variable: my tests were done with an ASIO soundcard's SPDIF against DAC1 USB in WinXP. Same result though.


----------



## 03lab

Quote:


  Originally Posted by *Crowbar* /img/forum/go_quote.gif 
_Do you drive or take the bus or another gasoline or diesel using vehicle? A liter of gasoline is 9,000 watt-hours, 11,000 for diesel. Using a vehicle to go a single kilometer pales in comparison to electronic equipment's standby power for a whole day._

 

Yes ... and at night, I even turn on the light sometimes. 
	

	
	
		
		

		
			





 I like to safe energy nonetheless though and on a global scale it does make a difference IMHO. 

 Anyway, I was more concerned about sending the DAC1 back for repair than power consumption in standby, as I always disconnect the power cord to my equipment when not in use.


----------



## Wavelength

Quote:


  Originally Posted by *poo* /img/forum/go_quote.gif 
_DAMN! I didn't realise that (or at least forgot). Is that a limitation of USB within the DAC1, or USB in general? Can I ask for a simple explanation as to why this is the case? Something to do with bandwith?_

 

Poo,

 Both Windows and MAC OSX have capabilities over 200Kbps with USB 1.1 protocols. The problem is with the TAS1020. It really does not have the buffer space to pull off 176.4/192k. For 24/96 operation the buffer needs to be at least 829.4 bytes (don't ask) for the TAS1020. No single ISO link in the TAS1020 can exceed 1024 bytes. It would require 1658 bytes for 192 which is more than the part has access to.

 I am looking at some ARM controllers and other processors that handle both USB 2.0 speeds and have a lot larger buffer space.

 I think ounce you get above 96k that you are pushing the host 1.1 ports as you doing about 5mbps. It would of course require 10mbps for 192 and considering how much is spent on the ports looking for a part that handles 480mbps might be the forward thinking way too go.

 Thanks
 Gordon


----------



## EliasGwinn

Quote:


  Originally Posted by *yourmando* /img/forum/go_quote.gif 
_the other day I noticed some loud hum from the listening position. I use the dac1 usb as my preamp, and it's the only thing connected to my speakers (no cable box or other sources). I finally narrowed down the problem--if I plug the dac1 usb into a different wall socket from the other components, the hum mostly goes away. (Still audible from near the speakers, but I don't mind that). The 2 different wall socket situation is not ideal because I have 2 power strips coming in across different sides of the room._

 

Hello Armando,

 First of all, I would recommend avoiding the transformer at all cost. It will add significant distortion, which will absolutely be audible (assuming your system is high-resolution). I highly recomment the XLR connector with the unconnected pin-3.

 The hum problem is strange, because usually ground-loops are worse when the components are plugged into different outlets. It sounds like there is an AC leak into the signal-ground in your amplifier. Are you good with electronics? It would be interesting to see what would happen if you connected the RCA shield to the chassis in your amplifier.

 Let me know...
 Thanks,
 Elias


----------



## yourmando

Hi Elias,

 Very interesting. No, I'm not handy with electronics. I'm a software engineer who has never soldered a thing. How would I connect the RCA shield to the chassis?

 I'll definitely stay away from the transformer. I'd love to find the root cause anyway. Yup, the system Linkwitz definitely created a highly resolving system, combining a very interesting controlled directivity open baffle dipole design (including dipole woofers) and active crossover/eq...

 Thanks again,
 Armando


----------



## EliasGwinn

Quote:


  Originally Posted by *poo* /img/forum/go_quote.gif 
_DAMN! I didn't realise that (or at least forgot). Is that a limitation of USB within the DAC1, or USB in general? Can I ask for a simple explanation as to why this is the case? Something to do with bandwith?_

 

96/24 is the upper limit of audio over USB 1.1, due to bandwidth limitations.

 We could have went with USB 2.0, but there are no audio-specific USB 2.0 chips. Therefore, we would have to build custom drivers, and that would defeat our goal of making this device a completely 'native' solution. 

 Thanks,
 Elias


----------



## EliasGwinn

Quote:


  Originally Posted by *erotisches* /img/forum/go_quote.gif 
_One question, if my PC is also a source for home theater, can dac1 be used simultaneously with the sound card? I also need digital out for the surround channel whereas ,if all output is fed through dac1, I will only have 2ch output. Digital output would be nice too 
	

	
	
		
		

		
		
	


	




 ahhh I'm so greedy.
 Sorry if this is answered in FAQ, I admit that I haven't read it._

 

Can you go into more detail about what you're trying to do?

 Thanks,
 Elias


----------



## EliasGwinn

Quote:


  Originally Posted by *bsckwan* /img/forum/go_quote.gif 
_1.) Mac OS X 10.5.3 with iTunes 7.6.2 playing AIFF files --> (via USB with 24 bit / 44.1 kHz setting in Audio Midi and standard USB cable) Benchmark DAC1 PRE

 Is there anyway to improve on 1 and whether this is usually the case with DAC1 - SPDIF better then USB?_

 

If you are playing 44.1 kHz files, and the volume control is set to maximum, and you've followed the other 'Audio Wiki' recommended settings (no sound-check, no sound-enhancer, etc), then you've got bit-transparent playback. 

 If you've got bit-transparent playback on BOTH the CD transport and computer, they should sound identical. Don't assume that the CD transport is bit-transparent though...we've seen the opposite quite often.

 Thanks,
 Elias


----------



## EliasGwinn

Quote:


  Originally Posted by *03lab* /img/forum/go_quote.gif 
_Anyway, I was more concerned about sending the DAC1 back for repair than power consumption in standby, as I always disconnect the power cord to my equipment when not in use._

 

Don't worry about this. You won't damage the DAC1 like this.

 Thanks,
 Elias


----------



## EliasGwinn

Quote:


  Originally Posted by *Wavelength* /img/forum/go_quote.gif 
_Both Windows and MAC OSX have capabilities over 200Kbps with USB 1.1 protocols._

 

This isn't true. I'm assuming you meant 200 *M*bps, but even still, its not true. The highest data rate of USB 1.1 (Full Speed) is 12 Mbps.

 Thanks,
 Elias


----------



## EliasGwinn

Quote:


  Originally Posted by *yourmando* /img/forum/go_quote.gif 
_Very interesting. No, I'm not handy with electronics. I'm a software engineer who has never soldered a thing. How would I connect the RCA shield to the chassis?_

 

Well, you could test to see if this eliminates the hum by using a scrap of wire and just touching the rca sheild to a bare part of the chassis. If so, then you could try something more permenant.

 Thanks,
 Elias


----------



## G-U-E-S-T

Quote:


  Originally Posted by *EliasGwinn* /img/forum/go_quote.gif 
_Hello Armando,

 First of all, I would recommend avoiding the transformer at all cost. It will add significant distortion, which will absolutely be audible (assuming your system is high-resolution)...Elias_

 

Hi Elias,

 Armando was asking about the Jensen transformer product, PC-2XR. These are not your ordinary transformers! Did you check-out its specifications? If not, could you please take just a moment to see them with graphs at this link: http://www.jensentransformers.com/datashts/pc2xr.pdf

 These particular Jensen input-transformers are designed specifically for the use Armando is describing, and they are very well known for being transparent. They might be the best of their kind in the industry. I've also personally used them in hi-resolution systems and can vouch for their quality. So please forgive my double-questioning of your information, as I mean no disrespect - but are you absolutely sure about the accuracy of the advice you've given Armando here, about this particular Jensen transformer product?


----------



## Crowbar

Looking at the transformer specs, the phase distortion is not too bad, but the THD is unacceptable, especially at low frequencies. This is nothing close to transparent. Note also that they don't present a measurement of the hysteresis, which is a form of distortion specific to cored inductors. No surprise--why show even more of the distortion your equipment does in your marketing...


----------



## G-U-E-S-T

Quote:


  Originally Posted by *Crowbar* /img/forum/go_quote.gif 
_Looking at the transformer specs, the phase distortion is not too bad, but the THD is unacceptable, especially at low frequencies. This is nothing close to transparent. Note also that they don't present a measurement of the hysteresis, which is a form of distortion specific to cored inductors..._

 

Typical THD is 0.015% at 20Hz and less than 0.001% at 1kHz (+4dBu levels). Can this possibly be audible? Plus the transformers actually decouple and remove existing ground-noise distortion otherwise riding on the audio signal through an unbalanced interconnect cable.

 Also I've never heard of Jensen transformers suffering any problems whatsoever with saturation (hysteresis). Have you? They are used industry-wide in critical applications (recording, live, etc) and have a solidly established reputation for their quality.

 Sorry if these are dumb questions/statements - I'm definitely not trying to be argumentative, just wanting to hash out this information for everybody's sake (mine included). What am I missing here? I'm ready to learn - thanks in advance for any replies...


----------



## Wavelength

Quote:


  Originally Posted by *EliasGwinn* /img/forum/go_quote.gif 
_This isn't true. I'm assuming you meant 200 *M*bps, but even still, its not true. The highest data rate of USB 1.1 (Full Speed) is 12 Mbps.

 Thanks,
 Elias_

 

Elias,

 Sorry I meant >200K like above 192k the sampling rate. The USB 1.1 spec does not declare a maximum sampling speed. The only problem here is that the TAS1020 cannot do more than 24/96k as the buffer size for the endpoint cannot exceed 1024. Since the endpoint data in the TAS1020 is only 1304 of which much of that is used for the ROM code leaving only 1176 available for ISO data it does not make it capable of either 176.4/192k.

 There are many capable Codec oriented ARM controllers capable of the higher rates. Centrance as did I decided that allot of the code was available with the Reference Code give by TI for the TAS1020. This gives you a boost up on getting your feet wet.

 Many companies are now going to offer source code software to support the higher rates with much larger buffers.

 Thanks
 Gordon


----------



## yourmando

Quote:


  Originally Posted by *EliasGwinn* /img/forum/go_quote.gif 
_Well, you could test to see if this eliminates the hum by using a scrap of wire and just touching the rca sheild to a bare part of the chassis. If so, then you could try something more permenant.

 Thanks,
 Elias_

 

Unfortunately, this did not work. I've been troubleshooting more with the help of the Linkwitz Orion user forum. The DAC1, ATI amplifier, and active crossover/asp all have properly designed chassis ground (I'm not ruling out that one of the pieces is defective). This is unusual in that if only those 3 components are connected to the same power strip/socket, I get ground loop hum. And there are others with identical components with no hum.

 The only solution I could find was to float the ground in either the DAC1 or the ASP/crossover (I did not attempt to float the amp). This breaks the loop and kills the hum. So I am leaving the ASP floated with a two-prong plug. This solves the problem, but I do feel like I'm "cheating." 
	

	
	
		
		

		
			





 Everything is still grounded through the shield of the interconnect, so I should be ok (I hope 
	

	
	
		
		

		
		
	


	




)

 Thanks again,
 Armando
 P.S. I'd also be interested to see some more Audio Precision type measurements on a Jensen, just out of curiosity. I love how the DAC1 has tons of performance measurements.


----------



## Bostonears

Quote:


  Originally Posted by *yourmando* /img/forum/go_quote.gif 
_The only solution I could find was to float the ground in either the DAC1 or the ASP/crossover (I did not attempt to float the amp). This breaks the loop and kills the hum. So I am leaving the ASP floated with a two-prong plug. This solves the problem, but I do feel like I'm "cheating." 
	

	
	
		
		

		
		
	


	




 Everything is still grounded through the shield of the interconnect, so I should be ok (I hope 
	

	
	
		
		

		
		
	


	




)_

 

It is unsafe to use a cheater plug to float the ground. If a short were to develop inside the unit, unimpeded AC voltage could zap anyone who touches the chassis.

 If floating the ground removes the hum, a better solution is to use a Hum-X from Ebtech, which is designed for the specific purpose of breaking an AC ground loop while still leaving the ground intact. It's available for about $60 at many online stores. I used one quite successfully when I had a nasty ground loop between a DAC and a tube amp. (I put the Hum-X on the DAC's AC line, and I didn't notice any audible degradation of signal besides killing the hum.)


----------



## Crowbar

Why pay $60 when there's an $2 way to defeat ground loops? Leave earth connected to the chassis, but between earth and the power supply ground use a heavy 10 ohm resistor, paralleled with two antiparallel diodes (opposite directions, in parallel) of a current rating higher than the equipment fuse to provide protection if the resistor burns, and a 100 nF cap to bypass RF.


----------



## Bostonears

Quote:


  Originally Posted by *Crowbar* /img/forum/go_quote.gif 
_Why pay $60 when there's an $2 way to defeat ground loops? Leave earth connected to the chassis, but between earth and the power supply ground use a heavy 10 ohm resistor, paralleled with two antiparallel diodes (opposite directions, in parallel) of a current rating higher than the equipment fuse to provide protection if the resistor burns, and a 100 nF cap to bypass RF._

 

That may in fact be what the Hum-X does, although your DIY solution would be beyond the skills of many users. (Not everyone is a DIYer.) Either way, the point is that a cheater plug is unsafe for devices intended to be grounded. Any solution that works without compromising the safety of the ground is fine.


----------



## Bysheon

Excellent thread. I've read most of it and the discussions in this thread was one of the reasons I decided to buy the dac1. I'm very satisfied with my dac1. Kudos to Benchmark for this great product.

 A quick question - can changing the USB polling rate, for example changing the frequency to 1000Hz, affect the performance of dac1? I want to to this to improve my gaming experience, but my tech knowledge is kinda zero. Is it safe to do this?


 Edit: Spelling

 Thanks


----------



## EliasGwinn

Quote:


  Originally Posted by *Bysheon* /img/forum/go_quote.gif 
_Excellent thread. I've read most of it and the discussions in this thread was one of the reasons I decided to buy the dac1. I'm very satisfied with my dac1. Kudos to Benchmark for this great product.

 A quick question - can changing the USB polling rate, for example changing the frequency to 1000Hz, affect the performance of dac1? I want to to this to improve my gaming experience, but my tech knowledge is kinda zero. Is it safe to do this?


 Edit: Spelling

 Thanks_

 

Hello Bysheon,

 The DAC1 USB is not affected by the USB polling rate. Polling rate only affects items that use 'interrupt transfer mode', such as a mouse. It determines how often the computer 'asks' the device if there is a request for action.

 The DAC1 USB uses 'isochronous transfer mode'. This means that it establishes a data-flow connection of a certain bandwidth with the computer, and then continues at delivering data as it becomes available. In this case, the data rate is similar to the polling rate, but it is determined by the sample-rate of the audio.

 Thanks,
 Elias


----------



## Bysheon

Quote:


  Originally Posted by *EliasGwinn* /img/forum/go_quote.gif 
_Hello Bysheon,

 The DAC1 USB is not affected by the USB polling rate. Polling rate only affects items that use 'interrupt transfer mode', such as a mouse. It determines how often the computer 'asks' the device if there is a request for action.

 The DAC1 USB uses 'isochronous transfer mode'. This means that it establishes a data-flow connection of a certain bandwidth with the computer, and then continues at delivering data as it becomes available. In this case, the data rate is similar to the polling rate, but it is determined by the sample-rate of the audio.

 Thanks,
 Elias_

 

Hi, Elias
 Thanks for the guick answer! Good news.


----------



## Terje

If you type youtube.com in your url and Benchmark DAC1 USB in the search field....you can see an interview with director of technical engineering at Benchmark Jhon Siou...and a blip of DAC1.


----------



## John Reeves

Hi,

 I am the owner of a DAC-1, and am very pleased with it. I did get eyestrain reading through all the posts although this was a little academic since I had already got the DAC-1.

 I am contemplating new headphones for use with the headphone output. I do not want to use my UE10's, and want to improve on my Ultrasones. 

 I noted that the headphone out was designed for Senns HD650 and wished to know if anyone had tried these headphones and compared the sound with a dedicated headphone amp. It does seem that to get the best out of the Senns you need a Cardas or similar cable and a carefully matched head amp. Ditto for the HD600's.

 Any comments would be much appreciated.

 Regards

 John


----------



## EliasGwinn

Quote:


  Originally Posted by *John Reeves* /img/forum/go_quote.gif 
_I noted that the headphone out was designed for Senns HD650 and wished to know if anyone had tried these headphones and compared the sound with a dedicated headphone amp. It does seem that to get the best out of the Senns you need a Cardas or similar cable and a carefully matched head amp. Ditto for the HD600's._

 

Hello John,

 Thanks for joining our thread. 

 The headphone amp built into the DAC1 is not specifically designed for the Sennheiser HD650's. We simply recommend the 650's as a high-quality companion to our products.

 The DAC1 contains a headphone amp design that was initially a stand-alone module that has become very highly regarded amond audio professionals. The amp built into the DAC1 is the HPA-2, which has been integrated into audio consoles used by television, radio, recording, and mastering studios around the world. They've replaced the console's stock headphone amp with the HPA-2 module to be used as a reference headphone amp. We put the exact same HPA-2 circuit in the DAC1 / USB / PRE. 

 Thanks,
 Elias


----------



## poo

Hey Elias,

 There is much debate regarding how to best drive/power the HD650s. For a long time, my HD650s have been my primary headphones, driven directly from the HPA-2 of my DAC1 USB. Sounds fantastic, but I often wonder if there is more to appreciate by adding an external headphone amp to the mix.

 Most comments regarding this issue seem to relate to voltage, suggesting that (summarised) 'for the HD650, the more voltage the better'.

 Can you comment on this? Does your experience suggest that I will or won't benifit from adding an amp in this case?

 Thanks again for your help.


----------



## EliasGwinn

Quote:


  Originally Posted by *poo* /img/forum/go_quote.gif 
_Hey Elias,

 There is much debate regarding how to best drive/power the HD650s. For a long time, my HD650s have been my primary headphones, driven directly from the HPA-2 of my DAC1 USB. Sounds fantastic, but I often wonder if there is more to appreciate by adding an external headphone amp to the mix.

 Most comments regarding this issue seem to relate to voltage, suggesting that (summarised) 'for the HD650, the more voltage the better'.

 Can you comment on this? Does your experience suggest that I will or won't benifit from adding an amp in this case?

 Thanks again for your help._

 

I'm not sure I understand the idea about 'more voltage = better'. Higher voltage will only increase volume. Unless they are referring to voltage rails. In that case, higher voltage rails will get you more headroom...i.e., more output before clipping. The HPA-2 operates with 18V rails...about as much as you'll find anywher.

 I don't think there is anything to gain from adding an external headphone amplifier. The HPA-2 in the DAC1 is our top-of-the-line, studio reference quality headphone amp. We didn't skimp at all when building into the DAC1. Its got 0-ohm output impedance, which will tame even the lowest-Z headphones on the market. It has enough headroom to drive a load at the end of 1000' of cable without struggling, and operates gracefully at lower gain settings as well.

 I don't think there is another headphone amp on the market that can do what the HPA-2 does, as cleanly as it does it. 

 Now, with all that said, if you're looking for a headphone amp that will color the sound, add warmth, etc, then you may want to look for another headphone amp. The HPA-2 is designed for maximum transparency.

 Thanks,
 Elias


----------



## poo

Thanks Elias - fully answers my speculation... colour isn't my thing at this stage


----------



## infinitesymphony

I see a Benchmark DAC1 USB in my future. 
	

	
	
		
		

		
		
	


	




 Elias, you'd said earlier that Benchmark is constantly evaluating the performance of new DAC chips. Have there been any contenders to the Analog Devices AD1853, and is it still the best for this purpose/implementation? I'm always interested in reading more about the objective differences between DACs.


----------



## nettokung

I use it with my system (W2002+HA2002). the sound's great and better than my old soundcard.


----------



## G-U-E-S-T

Elias, did you miss my last post to you, about the Jensen transformer? I would sure appreciate your feedback. If I am incorrect about any of it, don't worry about being "tactful" - I am here to learn! Thanks in advance.


----------



## John Reeves

Hi,

 I have been utilizing a Benchmark DAC to run into a Naim pre-amp and power amplifier. I have had this equipment from new and it is about 22 years old. The improvement from running out of the Juli@ was a definite step forward.

 However, I decided that since I was not using any analogue sources, then I would run the Benchmark straight into the Naim 250 amp.

 This was a big mistake. The improvement was unbelievable, and I do not exaggerate. I don't believe in snake oil, so I can assure you that this is correct. I had friends around and the improvement was so obvious it would have been pointless to carry out blind listening tests. So it was obvious that over the years of good service the Naim equipment was probably a bit tired and needed capacitors etc replaced.

 So, the mistake was showing up the deficiencies of my existing system, which had degraded gradually over the course of years, so giving me the dreadful disease UPGRADITUS.

 OK the lots being changed. I'm selling the Naim stuff and replacing it with simply a new power amp. I'm going to keep the Isobariks in the short term, but will be looking to possibly replace them- short term, who am I kidding, even my wife doesn't believe me since she has recognized the symptoms of UPGRADITUS.

 So, fellow Headfiers beware, buying a Benchmark can easily lead to you catching this dreadful disease whose side effects include an empty wallet and phone calls from the bank manager.

 Cheers

 John


----------



## EliasGwinn

Quote:


  Originally Posted by *infinitesymphony* /img/forum/go_quote.gif 
_Have there been any contenders to the Analog Devices AD1853_

 

Hey! Sorry I've been MIA...our sales person, Rory, has been out of the office this week, so I've been on full-time phone and email tech-support and sales duty. And, I'll tell you what, I have a new level of respect for Rory after this week!!

 RE: Dac chips... The AD1853 is still our chip of choice. AD has another chip (AD1955) that has 3 dB less noise then the 1853, but it has functionality issues that make its use questionable.

 As for other dac makers, we've found that Cirrus chips have issues with their low-pass filters, Burr Brown dacs have linearity issues, and AKM dacs have poor stop-band limitations which will cause aliasing.

 Another nice thing about the 1853 is that it has a current output, which allows us to build our own I-V converter with much lower distortion then most built-in IV converters.

 Thanks,
 Elias


----------



## EliasGwinn

Quote:


  Originally Posted by *G-U-E-S-T* /img/forum/go_quote.gif 
_...about the Jensen transformer..._

 

G-U-E-S-T,

 I haven't really looked at the spec sheet for that Jensen transformer, but the spec you mentioned (0.015% @ 20Hz) is rather detrimental. That equates to -76.5 dB, or about 30 dB more then the DAC1. Try one and hear it for yourself...it won't be subtle.

 Transformers are used in pro audio for a few reasons: 1. they are a great tool for providing common-mode rejection, when that is an issue 2. sometimes they are needed to provide electrical isolation and/or impedance conversion 3. some people actually enjoy the distortion of transformers on certain instruments, just like some people enjoy tube distortion.

 As Crowbar mentioned, using a transformer would really reduce the transparency of the audio. Sometimes they're necessary, sometimes they are desired, but they are never transparent.

 Thanks,
 Elias


----------



## ted betley

Congrats to all the staff @ Benchmark for the 'shoot the lights out' review at Absolute Sound.


----------



## Matias

Quote:


  Originally Posted by *EliasGwinn* /img/forum/go_quote.gif 
_The HPA-2 in the DAC1 is our top-of-the-line, studio reference quality headphone amp. We didn't skimp at all when building into the DAC1._

 

Hi Elias,

 If it could only have a better attenuator, say Alps or DACT, then it would be really top. If it only had more room, I would mod it myself...

 Thanks,
 Matias


----------



## Covenant

Quote:


  Originally Posted by *EliasGwinn* /img/forum/go_quote.gif 
_I don't think there is anything to gain from adding an external headphone amplifier. The HPA-2 in the DAC1 is our top-of-the-line, studio reference quality headphone amp._

 

Hi Elias,

 I'm curious, does the HPA-2 drive the balanced outputs of the DAC1 as well? I've been hearing for ages that the HD650 scales extremely well with balanced operation, and having a reference-grade USB DAC with true balanced outputs and a high quality inbuilt balanced amplifier all in one unit would be very attractive.

 Also, is the HPA-2 a discrete circuit, or opamp based? Just asking because I'm not familiar with the HPA-2 at all.


----------



## NewMexiCat

+1 for Covenant's fine questions.

  Quote:


  Originally Posted by *Covenant* /img/forum/go_quote.gif 
_Hi Elias,

 I'm curious, does the HPA-2 drive the balanced outputs of the DAC1 as well? I've been hearing for ages that the HD650 scales extremely well with balanced operation, and having a reference-grade USB DAC with true balanced outputs and a high quality inbuilt balanced amplifier all in one unit would be very attractive.

 Also, is the HPA-2 a discrete circuit, or opamp based? Just asking because I'm not familiar with the HPA-2 at all._


----------



## infinitesymphony

Those questions were answered earlier in the thread. The headphone and line output sections are different, the DAC1 is not designed to drive headphones out of the balanced outputs (and Elias recommends against it), and the headphone section uses either a NE5532 (regular DAC1) or a LM4562 (DAC1 USB and DAC1 PRE) op-amp for the headphone section. At least, that's what I remember.


----------



## EliasGwinn

Quote:


  Originally Posted by *Covenant* /img/forum/go_quote.gif 
_...does the HPA-2 drive the balanced outputs of the DAC1 as well?_

 

No, the HPA-2 drives the headphone output only.

  Quote:


  Originally Posted by *Covenant* /img/forum/go_quote.gif 
_I've been hearing for ages that the HD650 scales extremely well with balanced operation, and having a reference-grade USB DAC with true balanced outputs and a high quality inbuilt balanced amplifier all in one unit would be very attractive._

 

The DAC1/USB/PRE does, in fact, have true-balanced outputs. These outputs can drive balanced headphones. However, I strongly advise against this configuration using any balanced headphone amplifier. 

 I've written about this here and many other places, but I'll quickly summerize for you.

 Balanced headphone have many inherent drawbacks, and very little substantial benefits. 

 Inherent drawbacks include: 
 - Double output impedance
 - 50% reduction of damping factor
 - 100% increase in noise
 - +/-200% increase in distortion. 

 Benefits include:
 - Increase gain 
 - Increase slew rate

 These 'benefits' (increase in gain and slew rate) are hardly worth the major drawbacks. The gain and slew rate of the Benchmark's HPA-2 are far more then they need to be for the application, so increasing them won't gain any performance increase.

  Quote:


  Originally Posted by *Covenant* /img/forum/go_quote.gif 
_Also, is the HPA-2 a discrete circuit, or opamp based? Just asking because I'm not familiar with the HPA-2 at all._

 

I've addressed this here, but I'll recap for you. The output driver of the HPA-2 is the BUF634, a high-speed buffer amplifier capable of 250 mA output current and 2000 V/uS slew rate. 

 Thanks,
 Elias


----------



## Covenant

Thank you very much for your prompt response, Mr Elias. Its great to see you're still active here and so knowledgeable about your product, I wish more manufacturers would aspire to such support. You're a gentleman and a scholar 
	

	
	
		
		

		
		
	


	




 I wont pretend to fully understand the link you provided listing the technical reasoning against balanced operation, however this seems to fly in the face of conventional wisdom on these boards. Its been drilled into us like a religious sermon that balanced operation will always exceed unbalanced operation in terms of extrating the last ounces of sound quality from a headphone, especially with the Sennheisers.

 The reason touted is normally the elimination of crosstalk and the doubled voltage swing provided to each driver - the Senns are reputedly very voltage-swing dependant, and subjective listening tests seem to have verified this.

 I'm not disputing your information, however i'm just curious as to why oppinions are differing so much on this. Even if the balanced line-out stage of the DAC1 USB was somewhat inferior to the headphone amp for the purpose of driving headphones, I would have wagered that a balanced HD650 would still far outperform an unbalanced HD650 with the DAC1 USB as source/amp.


----------



## katalyst^

Is there any difference between the DAC1 and DAC1 USB in terms of the quality of the headphone amp?


----------



## applebook

Quote:


  Originally Posted by *katalyst^* /img/forum/go_quote.gif 
_Is there any difference between the DAC1 and DAC1 USB in terms of the quality of the headphone amp?_

 

The headphone amp is not affected by the DAC output


----------



## luciyuspax

Hi guys! I am planning to buy benchmark DAC1 USB. I have sometimes the hum problem when I am listening to music with my headphones through headphone output of my onkyo integrated amplifier. I guess I have a ground problem.

 Does Benchmark DAC1 suffer from hum problems as well? I will use DAC1 only plugged to my laptop purely for headphone listening. Only DAC1 and laptop will be plugged into the same strip. Also about those surge protectors with power filters(not the big powercenter models in a box), how do they behave with DAC1? I will either buy a monster HTS1000 or belkin pureav surge protector. I heard that they generally improve video quality of plasma TVs etc but that some amplifier really do not like them. Will they improve or decrease sound quality of DAC1 USB?

 Have a nice day all!


----------



## katalyst^

*blank stare* I am confused 
	

	
	
		
		

		
		
	


	




 Someone mentioned above that they use different opamps - I thought the head amp might have been upgraded a little at the same time as the addition of the USB.

 Does that mean that if I don't need USB, I can buy an ordinary DAC1 and it'll be otherwise identical?


----------



## anadin

I've just ordered a Benchmark DAC1 PRE and can't wait for it to arrive.

 .


----------



## eweitzman

Quote:


  Originally Posted by *EliasGwinn* /img/forum/go_quote.gif 
_Inherent drawbacks include: 
 - Double output impedance
 - 50% reduction of damping factor
 - 100% increase in noise
 - +/-200% increase in distortion. 

 Benefits include:
 - Increase gain 
 - Increase slew rate_

 

The damping factor decrease is just an effect of the increase in output impedance. What Elias doesn't tell you is how this will effect what you hear: the frequency response of the phones will change, as may the quality of the bass. What you will hear will sound different and be less accurate, but not necessarily better. If you want tone controls, use an equalizer so you know what the changes are.

 - Eric


----------



## Matias

Elias,

 A friend was about to buy a DAC1 and 2 retailers (B&H and Audio Revelation) said they are no longer authorized to sell outside the USA! What's going on??


----------



## katalyst^

Quote:


  Originally Posted by *Matias* /img/forum/go_quote.gif 
_Elias,

 A friend was about to buy a DAC1 and 2 retailers (B&H and Audio Revelation) said they are no longer authorized to sell outside the USA! What's going on?? 
	

	
	
		
		

		
		
	


	


_

 

Great 
	

	
	
		
		

		
		
	


	




 Now I have to either buy a second-hand one or pay a ridiculous 60-90% RRP premium to Australian resellers. There are reseller profit margins and then there is '****ing your customers in the ass'.


----------



## poo

Quote:


  Originally Posted by *katalyst^* /img/forum/go_quote.gif 
_There are reseller profit margins and then there is '****ing your customers in the ass'._

 

Couldn't agree more. I imported mine from B&H not quite a year ago for the same reason. I guess Benchmark don't feel they can 'turn a blind eye' for the benifit of their end users... shame.


----------



## anadin

I am awaiting delivery of my DAC-1 Pre and have a couple of questions.

 Is it worth the extra cost of upgrading the power cord and the USB cable, are the supplied cables of good quality that an upgrade isn't needed.

 Many thanks.


----------



## poo

anadin, that's a real can of worms question that you will most likely get a range of answers to. Personally, I would suggest that the cables supplied are fine and your money would be much better spent on other things (like a bottle of scotch to enjoy when your DAC arrives).

 I guess it depends whether you are into cables or not really - if better cables are going to give you warm fuzzies, go for it...


----------



## dspargo

Quote:


  Originally Posted by *EliasGwinn* /img/forum/go_quote.gif 
_No, the HPA-2 drives the headphone output only.



 The DAC1/USB/PRE does, in fact, have true-balanced outputs. These outputs can drive balanced headphones. However, I strongly advise against this configuration using any balanced headphone amplifier. 

 I've written about this here and many other places, but I'll quickly summerize for you.

 Balanced headphone have many inherent drawbacks, and very little substantial benefits. 

 Inherent drawbacks include: 
 - Double output impedance
 - 50% reduction of damping factor
 - 100% increase in noise
 - +/-200% increase in distortion. 

 Benefits include:
 - Increase gain 
 - Increase slew rate

 These 'benefits' (increase in gain and slew rate) are hardly worth the major drawbacks. The gain and slew rate of the Benchmark's HPA-2 are far more then they need to be for the application, so increasing them won't gain any performance increase.



 I've addressed this here, but I'll recap for you. The output driver of the HPA-2 is the BUF634, a high-speed buffer amplifier capable of 250 mA output current and 2000 V/uS slew rate. 

 Thanks,
 Elias_

 

Elias, 

 What are the balanced outputs supposed to be used for?

 I bought a HeadRoom Balanced Desktop Amp for my setup and this is a serious bummer. What do you recommend I do?

 Thank you,
 Don


----------



## Mazz

Quote:


  Originally Posted by *EliasGwinn* /img/forum/go_quote.gif 
_Balanced headphone have many inherent drawbacks, and very little substantial benefits. 

 Inherent drawbacks include: 
 - Double output impedance
 - 50% reduction of damping factor
 - 100% increase in noise
 - +/-200% increase in distortion. 

 Benefits include:
 - Increase gain 
 - Increase slew rate

 These 'benefits' (increase in gain and slew rate) are hardly worth the major drawbacks. The gain and slew rate of the Benchmark's HPA-2 are far more then they need to be for the application, so increasing them won't gain any performance increase._

 

I don't have any opinion on the matter, I don't recall too much of my electrical engineering courses, and I haven't had a serious listen to balanced headphones. But what I would like to know is whether these comparisons are on an apples-for-apples basis - i.e. driving the same pair of headphones to the same volume level, which presumably means to the same voltage differential (barring minor transient differences). It seems you'd get the double the slew rate from having two "identical" amplification circuits driving the signal in opposite directions. You'd have the same gain (by definition). You'd presumably have double some components of noise (IIRC some components are largely gain-independent), but maybe the same level of others (e.g. the component of noise present on the input gets amplified to the same level either way), and less of others (e.g. lower levels of noise induced on parts of the headphone cables - same principle as running long microphone cables to a mixer balanced rather than single-ended, but over a much shorter distance). I'm not sure why you get 200% increase (i.e. 3 times as much) increase in distortion, but perhaps it's in part due to the fact that it's really hard to get two absolutely identical amplification circuits.

 And I'd also like to know how it sounds, because perhaps some or all of these effects are below the audible threshold and many people think it sounds better in various ways...but that takes a bunch of careful listening, and can be quite subjective, so I don't think this is the thread for it.


----------



## pcf

Ken Kessler gave the DAC1pre a glowing review in this month's Hifi news.
 It gets a "Hi-finews outstanding product" tag. Here's how the artictle ends:

 HI FI NEWS VERDICT
 Not just good, but stupefyingly good. Benchmark's DAC1 pre provides you with
 three highend components in one-DAC, headphone amplifier and preamplifier. In every mode, it excels. It sounds natural and authoritative, almost valve-like in the mid and treble. sublimely well made. It's an absolute joy to use. It does what it should, without drama. It is turly the Swiss Army Knige of hi fi.


----------



## sonq

Quote:


  Originally Posted by *pcf* /img/forum/go_quote.gif 
_Ken Kessler gave the DAC1pre a glowing review in this month's Hifi news.
 It gets a "Hi-finews outstanding product" tag. Here's how the artictle ends:

 HI FI NEWS VERDICT
 Not just good, but stupefyingly good. Benchmark's DAC1 pre provides you with
 three highend components in one-DAC, headphone amplifier and preamplifier. In every mode, it excels. It sounds natural and authoritative, almost valve-like in the mid and treble. sublimely well made. It's an absolute joy to use. It does what it should, without drama. It is turly the Swiss Army Knige of hi fi._

 

Is the Dac1 pre just the same as Dac1 but comes with addition of USB and preamp function? Used through the HP jack, is there any difference in sound quality between the 3 variants?


----------



## Wavelength

Elias,

 Several companies offer balanced options for amplifiers and it basically means there is a seperate return (ground) for each phone.

 The benefit of this can be seen in a current loop. If you have seperate grounds for each the amplifier the loop will consist of the amplifier it's coupling cap and series resistance the headphones and the resistance of the headphone cables.

 If you use a common ground like most headphone amplifiers do then the grounds get summed and the effect both the amplifier circuits. Also many heaphone cables use the same geometry for each of the 3 conductors making the ground connection the weakest link.

 Better headphones typically sum the ground connections at the jack but even then it does not sound as good as a totally sperate connection back to the amplifier ground.

 Having worked with Arye for the last year and talking to Charlie Hansen about their amplifier technology I would have to say he would not agree with these statements:

  Quote:


 Inherent drawbacks include: 
 - Double output impedance
 - 50% reduction of damping factor
 - 100% increase in noise
 - ±200% increase in distortion. 

 Benefits include:
 - Increase gain 
 - Increase slew rate 
 

All of their power amplifiers (Product of the year for Stereophile) are balanced amplifiers.

 In most cases wouldn't you say that if it is truely Balanced and the noise is the same for both channels, then wouldn't the noise be cancelled to only the inherent noise of the amplifier?

 The same can be said for distortion as the summation of most of the second, third and so on would be removed.

 But in general I guess it would be completely silly to say anything of the differences in specifications because the unit would be designed completely different from just putting two amplifiers together.

 Thanks
 Gordon


----------



## EliasGwinn

Quote:


  Originally Posted by *katalyst^* /img/forum/go_quote.gif 
_Is there any difference between the DAC1 and DAC1 USB in terms of the quality of the headphone amp?_

 

No, all DAC1's are built with the same HPA-2, and they all have the same quality. 

 However, the DAC1 USB and DAC1 PRE have additional features related to the headphone amp. 

 Specifically, the auto-mute function, which mutes the main outs when headphones are plugged in for seamless switching from loudspeakers to headphones (this feature can be disabled by the user.) 

 Another feature is the selectable gain range, which allows the user to customize the gain of the HPA-2 to suit their particular headphones' sensitivities. The DAC1 USB has two ranges, and the DAC1 PRE has 3 ranges.

 Thanks,
 Elias


----------



## EliasGwinn

Quote:


  Originally Posted by *luciyuspax* /img/forum/go_quote.gif 
_Only DAC1 and laptop will be plugged into the same strip._

 

You shouldn't have any hum problems with this setup at all.

  Quote:


  Originally Posted by *luciyuspax* /img/forum/go_quote.gif 
_Also about those surge protectors with power filters(not the big powercenter models in a box), how do they behave with DAC1? I will either buy a monster HTS1000 or belkin pureav surge protector. I heard that they generally improve video quality of plasma TVs etc but that some amplifier really do not like them. Will they improve or decrease sound quality of DAC1 USB?_

 

They won't affect the performance of the DAC1 whatsoever. The power supply is built to be impervious to fluctuations in power.

 Thanks,
 Elias


----------



## Matias

Quote:


  Originally Posted by *Matias* /img/forum/go_quote.gif 
_Elias,

 A friend was about to buy a DAC1 and 2 retailers (B&H and Audio Revelation) said they are no longer authorized to sell outside the USA! What's going on?? 
	

	
	
		
		

		
		
	


	


_

 

Elias,

 Would you please confirm this? My friend is just about to order one, in other words, you are losing a customer because of this.


----------



## EliasGwinn

Quote:


  Originally Posted by *eweitzman* /img/forum/go_quote.gif 
_The damping factor decrease is just an effect of the increase in output impedance._

 

Yes, this is correct. And a detrimental affect, at that...

  Quote:


  Originally Posted by *eweitzman* /img/forum/go_quote.gif 
_What Elias doesn't tell you is how this will effect what you hear: the frequency response of the phones will change, as may the quality of the bass. What you will hear will sound different and be less accurate, but not necessarily better. If you want tone controls, use an equalizer so you know what the changes are._

 

Exactly. A high damping factor basically allows the amp to keep the driver's non-linearities in check. Speakers, as mechanical moving piston-like entities, are subject to distortion from physical impedance, such as air, and your head. The drivers in headphones can become very unpredictable if the amplifier does not maintain a high damping factor.

 All things being equal, a balanced headphone amp will reduce the damping factor in half.

 Thanks,
 Elias


----------



## EliasGwinn

Quote:


  Originally Posted by *Matias* /img/forum/go_quote.gif 
_Elias,

 A friend was about to buy a DAC1 and 2 retailers (B&H and Audio Revelation) said they are no longer authorized to sell outside the USA! What's going on?? 
	

	
	
		
		

		
			



_

 

Dealers are only permitted to sell in designated geographical areas. This is common to many manufacturer/distributor/dealer relationships. 

 If you are trying to order from a country which is not listed on our "Dealers and Distributors" page, you may be able order directly from us via the webstore or telephone.

 Thanks,
 Elias


----------



## EliasGwinn

Quote:


  Originally Posted by *katalyst^* /img/forum/go_quote.gif 
_Great 
	

	
	
		
		

		
		
	


	




 Now I have to either buy a second-hand one or pay a ridiculous 60-90% RRP premium to Australian resellers. There are reseller profit margins and then there is '****ing your customers in the ass'._

 

It may seem that non-U.S. dealers are adding extra profit margin to the product price, but the price increase is usually a direct result of the costs of import/exporting, shipping, duties, VAT's, customs, etc. 

 If you feel a particular dealer is charging more then the normal price + import fees, call them out on it. Ask them to break down the reasoning behind the price, item by item. You will usually find that it is, in fact, justifiable. But, in the case when it is not justifiable, simply tell them that they are losing business because of inflatted cost. Tell them that you will be telling everyone on the web that they are over priced. They may change their tune.

 However, I'd like to reiterate, you will typically find that the higher price is only a direct result of the cost of importing the products. In that case, you may want to contact your government and tell them that the import taxes are too high.

 Please, keep in touch and let me know what you find.... I'm very curious as welll...

 Thanks,
 Elias


----------



## EliasGwinn

Quote:


  Originally Posted by *anadin* /img/forum/go_quote.gif 
_I am awaiting delivery of my DAC-1 Pre and have a couple of questions.

 Is it worth the extra cost of upgrading the power cord and the USB cable, are the supplied cables of good quality that an upgrade isn't needed.

 Many thanks._

 

Hello Anadin,

 Welcome to the Benchmark family. Glad to have you aboard.

 In short, it is not worth buying an aftermarket power cord or USB cable. You will not notice any improvement in performance. 

 Benchmark products are designed to minimize or eliminate the effects of external variables, such as AC line quality, jitter, etc. The cables that come with the unit are very capable, and the DAC1 PRE will acheive top performance with them.

 Thanks,
 Elias


----------



## EliasGwinn

Quote:


  Originally Posted by *dspargo* /img/forum/go_quote.gif 
_Elias, 

 What are the balanced outputs supposed to be used for?

 I bought a HeadRoom Balanced Desktop Amp for my setup and this is a serious bummer. What do you recommend I do?

 Thank you,
 Don_

 


 The balanced outputs on the DAC1 are for connecting to high-impedance inputs on pre-amps, amplifiers, etc. They are not designed for driving headphones.

 The balanced outputs on your HeadRoom amp are, apparently, supposed to be used for balanced headphones. However, it is my view point that balanced headphone amplifiers acheive lower performance then their unbalanced counterpart.

 I would recommend you switch to an unbalanced headphone configuration.

 Thanks,
 Elias


----------



## EliasGwinn

Quote:


  Originally Posted by *Mazz* /img/forum/go_quote.gif 
_But what I would like to know is whether these comparisons are on an apples-for-apples basis - i.e. driving the same pair of headphones to the same volume level, which presumably means to the same voltage differential (barring minor transient differences)._

 

Absolutely. All things remaining equal, an unbalanced amplifier driving a set of headphones will perform better then the same amplifier in a balanced configuration driving the same set of headphones.

 CAVEAT: The arguement for isolated ground returns may be valid for some headphones, but most high-end headphones already have isolated ground returns in unbalanced configuration.

  Quote:


  Originally Posted by *Mazz* /img/forum/go_quote.gif 
_It seems you'd get the double the slew rate from having two "identical" amplification circuits driving the signal in opposite directions. You'd have the same gain (by definition). You'd presumably have double some components of noise (IIRC some components are largely gain-independent), but maybe the same level of others (e.g. the component of noise present on the input gets amplified to the same level either way), and less of others (e.g. lower levels of noise induced on parts of the headphone cables - same principle as running long microphone cables to a mixer balanced rather than single-ended, but over a much shorter distance). I'm not sure why you get 200% increase (i.e. 3 times as much) increase in distortion, but perhaps it's in part due to the fact that it's really hard to get two absolutely identical amplification circuits._

 

You would double the slew rate and double the gain (6 dB increase), but a decent headphone amp should not be needing more of those anyways. And increase in slew-rate and/or gain would not increase the performance of the HPA-2, for instance.

 Balanced amps will double the noise, also (amp1 noise + amp2 noise). However, balanced cables don't eliminate noise. Balanced input stages do. HEADPHONES DON'T HAVE BALANCED INPUT STAGES!! Well, more accurately, they are no more balanced using two active signals vs. an active and a return line. 

 200% increase in distortion is due to adding the distortion from two amps (amp1 THD + amp2 THD), plus another huge increase in distortion from the doubling of output impedance (50% damping factor). 

  Quote:


  Originally Posted by *Mazz* /img/forum/go_quote.gif 
_And I'd also like to know how it sounds, because perhaps some or all of these effects are below the audible threshold and many people think it sounds better in various ways...but that takes a bunch of careful listening, and can be quite subjective, so I don't think this is the thread for it._

 

A very well built balanced headphone amplifier will, at best, be indistinguishable from its unbalanced counterpart (except louder). At worst, there will be plenty of audible distortion. Whether someone prefers that distortion is subjective, and is at the user's discretion. My point is that the user should be informed that balanced headphones are not better by default. In fact, they are inferior with regards to performance spec's. 

 Thanks,
 Elias


----------



## EliasGwinn

Quote:


  Originally Posted by *sonq* /img/forum/go_quote.gif 
_Is the Dac1 pre just the same as Dac1 but comes with addition of USB and preamp function? Used through the HP jack, is there any difference in sound quality between the 3 variants?_

 

The DAC1 PRE differs from the DAC1 in these regards:
 - Stereo Analog inputs
 - 5 digital inputs (USB, TOSlink, 3x coax RCA)
 - Chassis-mounted Teflon RCA connectors
 - 3 gain-ranges for the headphone amp to suit different headphone sensitivities
 - Auto-mute feature which mutes the main outputs when headphones are plugged in
 - Standby On/Off switch
 - High-current opamps throughout analog circuitry that can drive difficult loads without loss of performance


----------



## EliasGwinn

Quote:


  Originally Posted by *Wavelength* /img/forum/go_quote.gif 
_Better headphones typically sum the ground connections at the jack but even then it does not sound as good as a totally sperate connection back to the amplifier ground._

 

Summing ground connections at the jack is the same as remaining seperate until the circuit ground. That has been my arguement against the isolated ground arguement. The Sennheiser HD650's, for example, have isolated grounds that remain isolated until they are at the circuit board.

 There may be an advantage for headphones that don't use isolated grounds (i.e., use common ground conductors from both drivers to the circuit). However, that's usually only the case on low-quality headphones...and in those cases, the common ground should be the least of your concerns. 

  Quote:


  Originally Posted by *Wavelength* /img/forum/go_quote.gif 
_Having worked with Arye for the last year and talking to Charlie Hansen about their amplifier technology I would have to say he would not agree with these statements:

 All of their power amplifiers (Product of the year for Stereophile) are balanced amplifiers._

 

Are you referring to amplifiers with balanced inputs, or amplifiers with balanced headphone outputs? Because these are very different.

  Quote:


  Originally Posted by *Wavelength* /img/forum/go_quote.gif 
_In most cases wouldn't you say that if it is truely Balanced and the noise is the same for both channels, then wouldn't the noise be cancelled to only the inherent noise of the amplifier?

 The same can be said for distortion as the summation of most of the second, third and so on would be removed._

 

First of all, common signals across a speaker will be cancelled even in an unbalanced configuration. Think about it, a speaker only reacts to _differences_ in voltage...just like how a resistor will only conduct current if there is a voltage differential across it. Any common signal across a speaker will be moot. In other words, the common-mode rejection of a headphone driver does not change depending on the type of amp you use.

 Second, the inherent noise of the amplifier is what I had been referring to all along. But, in a balanced amp, you have 2 amps, so you have 2x noise.

 Third, the distortion (2nd, 3rd, etc. harmonics) won't be common, they will be of opposite polarity. If they were common, that would mean the real signal (the fundamental) would also be common and would cancel. In fact, they will double, just like the fundamental, and just like the noise.

 Thanks,
 Elias


----------



## anadin

Quote:


  Originally Posted by *EliasGwinn* /img/forum/go_quote.gif 
_Hello Anadin,

 Welcome to the Benchmark family. Glad to have you aboard.

 In short, it is not worth buying an aftermarket power cord or USB cable. You will not notice any improvement in performance. 

 Benchmark products are designed to minimize or eliminate the effects of external variables, such as AC line quality, jitter, etc. The cables that come with the unit are very capable, and the DAC1 PRE will acheive top performance with them.

 Thanks,
 Elias_

 



 Many thanks for the reply.

 A couple of more questions I hope you dont mind.

 I plan on using the DAC1 PRE via the USB connection using either Wasapi or Kernel streaming with Foobar which one would you recommend.

 Also I want to use my stock Denon AH-D2000's headphones via the headphone out they are rated at 25 ohms, would the factory settings be the best gain or would you suggest me changing the gain to best accomodate the headphones.

 Many thanks again.


----------



## EliasGwinn

Quote:


  Originally Posted by *anadin* /img/forum/go_quote.gif 
_Many thanks for the reply.

 A couple of more questions I hope you dont mind.

 I plan on using the DAC1 PRE via the USB connection using either Wasapi or Kernel streaming with Foobar which one would you recommend.

 Also I want to use my stock Denon AH-D2000's headphones via the headphone out they are rated at 25 ohms, would the factory settings be the best gain or would you suggest me changing the gain to best accomodate the headphones.

 Many thanks again._

 

Apparently you are using Vista. In that case, a lot of users prefer Kernal streaming, and I would agree with that.

 For 25-ohm headphones, you may want to use the lowest HPA-2 gain setting. Are you getting the DAC1 PRE or the DAC1 USB?

 Thanks,
 Elias


----------



## Matias

Quote:


  Originally Posted by *EliasGwinn* /img/forum/go_quote.gif 
_In short, it is not worth buying an aftermarket power cord or USB cable. You will not notice any improvement in performance. 

 Benchmark products are designed to minimize or eliminate the effects of external variables, such as AC line quality, jitter, etc. The cables that come with the unit are very capable, and the DAC1 PRE will acheive top performance with them._

 

There is controversy. 
	

	
	
		
		

		
		
	


	




 Just yesterday we tested my DAC1 USB in a system based on Dynaudio Focus 220 speakers and Krell 400ix integrated amplifier, with an Arcam transport. Switching the stock fuses to the HiFi-Tuning.com ones I've said a couple os pages before, and switching power cables from a good one to a Cardas Golden Reference, both gave clear results: wider soundstage, heavier and faster bass transients, clearer mid range (voices).

 But then again, there are those who don't "believe" in cables... even though there's nothing to believe, but to be heard. 
	

	
	
		
		

		
		
	


	




 Cheers,
 Matias


----------



## Mazz

Quote:


  Originally Posted by *EliasGwinn* /img/forum/go_quote.gif 
_It may seem that non-U.S. dealers are adding extra profit margin to the product price, but the price increase is usually a direct result of the costs of import/exporting, shipping, duties, VAT's, customs, etc. _

 

That's not likely to be the case in Australia.

 If I import a DAC1 USB/PRE myself, I pay a retailer in the US (not a distributor). I incur currency translation fees.

 I then pay for shipping and insurance.

 I then pay government fees. The DAC1 is (helpfully) emblazoned on the back "Made in USA", which means (barring exceptions buried in the Australian Customs department documents) that it falls under the Australia-US Free Trade Agreement and no duty is charged. I presume (not having done this) that GST (10%), a customs processing fee (on the order of $35 if handled electronically) and any fees charged by a customs broker if the carrier doesn't do it (maybe $50-ish) are charged.

 So, we're not talking about particularly high import costs here...

 Now if an authorized Australian retailer imports a DAC1, they may be able to get a better price (depending on whether Benchmark expects Australian retailers to buy from a US distributor, or has appointed an Australian distributor). Chalk one advantage up to the retailer.

 They presumably trade regularly in the US, and thus are likely to be able to get a better deal on overseas payments. Chalk up number two.

 They pay the same fee structure, except retailers of any size would likely import several (or many) at a time and amortize the fees over all of those units. They'd also be likely to get cheaper per-unit shipping rates for larger shipments. Chalk up number three.

 They (retailer or distributor) may incur the cost of capital if they don't have an immediate firm order (say 5% for 6 months shelf life). A local distributor may charge extra for providing local service (but I don't know if there is local service for Benchmarks here or not). The retailer may charge extra for local service (i.e. expected cost of warranty returns, perhaps carrying extra units for warranty replacements whilst the originals are shipped back for repair).

 So if the retailer charged (very roughly) something like 5-15% more than it costs me to import myself, that might be fair enough given that they're providing extra value. If they are actually charging 60% more than my landed cost, that seems like a rip-off.


----------



## Mazz

Quote:


  Originally Posted by *EliasGwinn* /img/forum/go_quote.gif 
_ high damping factor basically allows the amp to keep the driver's non-linearities in check. [...] The drivers in headphones can become very unpredictable if the amplifier does not maintain a high damping factor.

 All things being equal, a balanced headphone amp will reduce the damping factor in half._

 

What constitutes "high"? To my mind it's not a question of "this parameter is twice as good so it must be a preferable design", it's "this design has a set of different parameters - do those differences meet the threshold of detectability when I'm listening, or is it better than this other design with a different set of parameters"? At what level damping factor does a further reduction to half the current level make a difference I can hear (all other things being relatively equal)?

 And if I read the definition of damping factor correctly (no guarantees!), different headphones with the same amp will have damping factor variations of 10 or more (because their impedances vary by 10 or more). Doesn't this then have a much bigger effect then balanced vs single-ended? Should I throw away my D5000s because their impedance is only 25ohms and get HD650s (which IIRC are in the region of 300 or 600 ohms)?


----------



## Mazz

Quote:


  Originally Posted by *EliasGwinn* /img/forum/go_quote.gif 
_A very well built balanced headphone amplifier will, at best, be indistinguishable from its unbalanced counterpart (except louder)._

 

I'm interested in comparisons at the same loudness, which presumably means a balanced amp runs each internal amp at half the gain that a single-ended amp would. Not sure whether that makes much of a difference...

  Quote:


  Originally Posted by *EliasGwinn* /img/forum/go_quote.gif 
_At worst, there will be plenty of audible distortion. _

 

Arguing by reference to worst cases isn't that helpful as we don't tend to buy (or keep) the worst case implementations.

  Quote:


  Originally Posted by *EliasGwinn* /img/forum/go_quote.gif 
_My point is that the user should be informed that balanced headphones are not better by default. In fact, they are inferior with regards to performance spec's. _

 

Good point, although I'd say "with regards to _some _of the performance specs".


----------



## dspargo

Quote:


  Originally Posted by *EliasGwinn* /img/forum/go_quote.gif 
_The balanced outputs on the DAC1 are for connecting to high-impedance inputs on pre-amps, amplifiers, etc. They are not designed for driving headphones.

 The balanced outputs on your HeadRoom amp are, apparently, supposed to be used for balanced headphones. However, it is my view point that balanced headphone amplifiers acheive lower performance then their unbalanced counterpart.

 I would recommend you switch to an unbalanced headphone configuration.

 Thanks,
 Elias_

 

Thank you for your response Elias. I would like to take the headphone amp in the DAC1 USB completely out of my configuration.

 I would like to use the balanced outputs on my DAC1 USB and connect to the balanced inputs on my HeadRoom Balanced Desktop Amp.

 How would you connect to a HeadRoom Balanced Desktop Amp that has both balanced and unbalanced inputs?


----------



## anadin

Quote:


  Originally Posted by *EliasGwinn* /img/forum/go_quote.gif 
_Apparently you are using Vista. In that case, a lot of users prefer Kernal streaming, and I would agree with that.

 For 25-ohm headphones, you may want to use the lowest HPA-2 gain setting. Are you getting the DAC1 PRE or the DAC1 USB?

 Thanks,
 Elias_

 


 I will be getting the DAC1 PRE.


----------



## Bostonears

Quote:


  Originally Posted by *Matias* /img/forum/go_quote.gif 
_There is controversy. 
	

	
	
		
		

		
			





 Just yesterday we tested my DAC1 USB in a system based on Dynaudio Focus 220 speakers and Krell 400ix integrated amplifier, with an Arcam transport. Switching the stock fuses to the HiFi-Tuning.com ones I've said a couple os pages before, and switching power cables from a good one to a Cardas Golden Reference, both gave clear results: wider soundstage, heavier and faster bass transients, clearer mid range (voices).

 But then again, there are those who don't "believe" in cables... even though there's nothing to believe, but to be heard. 
	

	
	
		
		

		
		
	


	




 Cheers,
 Matias_

 

Matias,
 Did you independently test the fuses and the power cable, and find that each of them made the noticeable improvement on its own? Or, did you swap them both at once (which would make it impossible to distinguish which part was responsible for the sonic changes)?


----------



## poo

Quote:


  Originally Posted by *EliasGwinn* /img/forum/go_quote.gif 
_It may seem that non-U.S. dealers are adding extra profit margin to the product price, but the price increase is usually a direct result of the costs of import/exporting, shipping, duties, VAT's, customs, etc. 

 If you feel a particular dealer is charging more then the normal price + import fees, call them out on it. Ask them to break down the reasoning behind the price, item by item. You will usually find that it is, in fact, justifiable. But, in the case when it is not justifiable, simply tell them that they are losing business because of inflatted cost. Tell them that you will be telling everyone on the web that they are over priced. They may change their tune.

 However, I'd like to reiterate, you will typically find that the higher price is only a direct result of the cost of importing the products. In that case, you may want to contact your government and tell them that the import taxes are too high.

 Please, keep in touch and let me know what you find.... I'm very curious as welll..._

 

Elias, I appreciate your response, but (for Australia at least), much of it is misguided IMO.

 First, it isn't up to _*Benchmark's*_ end users to tell their Government, or _*Benchmark's*_ resellers how to sell *Benchmark's* products.

 Second, At this point, most (if not all) Australian resellers of your products are selling them ilegally (unless I missed one when I was shopping around). Any reseller selling electrical products that are manufactured outside of Australia for foreign markets must _apply for and have issued_ a C-tick authorisation specific to that retailer _and_ product. 

 For Benchmark to specify that an Australian end user has no choice other than to purchase within Australia from resellers who are selling that product illegally, is possibly not in the companies' best interests...

 Your only listed Australian dealer is 'Sound and Music' in Melbourne. I presume they are your designated distributor. None of the stores I approached purchase Benchmark product from them, so why should I pay those resellers a premium?

 Regardless of where I purchased my DAC1 USB, Benchmark turns the same profit.


----------



## tyler69

hi there. i'm interested in trading the dac1usb for my dac1. what i'd like to know is:

 does the dac1 usb wake up from standby as soon as "any" source is recognized or does it only wake up when the source is not only recognized but also selected on the front?

 thanks.

 p.s.: does anybody know the power consumption of the dac1usb while in standby mode?


----------



## EliasGwinn

Mazz, Poo...

 Regarding the price of Benchmark products in Australia, I must admit that I have very limited knowledge about the sales-side of my company. My work here focuses on the technology and the end user. But, I will print a copy of your posts and bring it to Rory Rall, our sales manager, and to John Siau, the vice-president. I'll try to have an answer for you soon...

 Regarding my advice about telling the retailer and/or government to lower their margins, I didn't mean to imply that it was your responsibility to do that. However, I am a big advocate of consumer and citizen power. I always like to remind people that their voice and opinions should be heard, because you are their bottom line. 

 Thanks,
 Elias


----------



## katalyst^

Quote:


  Originally Posted by *poo* /img/forum/go_quote.gif 
_Your only listed Australian dealer is 'Sound and Music' in Melbourne. I presume they are your designated distributor. None of the stores I approached purchase Benchmark product from them, so why should I pay those resellers a premium?_

 

A quick comparison:

 The American RRP is USD 975. An individual could import the DAC1 for, say, $1050 USD. In all probability the item would be listed as having a value under $1000 AUD, and no GST would be paid. However, including GST, the total is around $1150. At prevailing exchange rates, the total cost for an individual is around $1200 AUD. AUD 1795 is listed as the RRP for the DAC1 at Sound and Music; it is a hefty 50 per cent premium. 

 That premium might substantially represent profit, in which case Benchmark is doing a great disservice to its true customers, for the benefit of a company the function of which is a pretentious intermediary and shopfront operator. The margin would be even greater given that Sound and Music presumably buys the units from Benchmark at below the American RRP. 

 I wonder whether the margin that Benchmark earns is less or more. If less, it is mildly amusing to think that Benchmark allows Sound and Music to earn a far greater margin than it does itself. If more, then this will fall on deaf ears because Benchmark and Sound and Music are happily fleecing audiophools of their money.

 If the margin is substantially similar between Sound and Music and American resellers, then there is no better reason for Australian customers to pay $1795 when it could be more efficiently had for $1200 over the Internet. Australian customers are paying to cover the gross inefficiencies of Sound and Music.


----------



## katalyst^

Quote:


  Originally Posted by *EliasGwinn* /img/forum/go_quote.gif 
_Regarding my advice about telling the retailer and/or government to lower their margins, I didn't mean to imply that it was your responsibility to do that. However, I am a big advocate of consumer and citizen power. I always like to remind people that their voice and opinions should be heard, because you are their bottom line. 

 Thanks,
 Elias_

 

To be honest, I don't think that Sound and Music, or any other Australian reseller, would care in the slightest. That they are able to price their units from $1795-$2095 shows how effectively they are insulated from competition: there is no need or incentive to bring prices closer to the US RRP, and because US retailers cannot sell internationally, we cannot bring competitive pressure to bear by shopping elsewhere. It is a case of accept the price or find another DAC.


----------



## Wavelength

Quote:


  Originally Posted by *EliasGwinn* /img/forum/go_quote.gif 
_Summing ground connections at the jack is the same as remaining seperate until the circuit ground. That has been my arguement against the isolated ground arguement. The Sennheiser HD650's, for example, have isolated grounds that remain isolated until they are at the circuit board._

 

Actually not true... remember you can't think of electronics in terms of voltage. You have to think of electronics in current and voltage. For any path their will be a voltage across and also that loop will carry the current signal. look at some of the app notes for power supplies or opamps most of them will comment on both aspects and will have the return (ground) reference to the point that yields the best results.

  Quote:


 Are you referring to amplifiers with balanced inputs, or amplifiers with balanced headphone outputs? Because these are very different. 
 

Elias, the Arye are totally balanced on the input and the output. They have no reference to ground. With that comes some problems especially with sub woofers and such that will reference the NEGATIVE signal to earth ground.

 Take a look at their site:

ayre home

 The thing to think about here is if you do make a balanced output amplifier it is in it's own sense a totally different concept than a single ended amplifier. Therefore the specifications cannot be looked at as just a double single ended amplifier but on their own merits.

 Thanks
 Gordon


----------



## infinitesymphony

I think Elias was only talking about balanced headphones; Ayre does not make a balanced headphone amplifier. His point was pretty clear in this statement:

  Quote:


  Originally Posted by *EliasGwinn* 
_Balanced amps will double the noise, also (amp1 noise + amp2 noise). However, balanced cables don't eliminate noise. Balanced input stages do. HEADPHONES DON'T HAVE BALANCED INPUT STAGES!!_

 

Maybe there would be a benefit to having balanced headphones if they had dedicated input stages. Otherwise, if we're just talking about regular balanced gear (ex. active studio monitors), then I'm sure he would recommend balanced over unbalanced output.


----------



## EliasGwinn

Quote:


  Originally Posted by *Mazz* /img/forum/go_quote.gif 
_What constitutes "high"? To my mind it's not a question of "this parameter is twice as good so it must be a preferable design", it's "this design has a set of different parameters - do those differences meet the threshold of detectability when I'm listening, or is it better than this other design with a different set of parameters"? At what level damping factor does a further reduction to half the current level make a difference I can hear (all other things being relatively equal)?_

 

I agree!! Just because a certain parameter is twice as good, does not mean that design is better. For example, twice as much slew-rate and gain does not acheive a better design (unless the design was suffering to begin with). 

 However, I feel that damping factor is the most critical of parameters. A headphone driver is a mechanical device. Based on its shape, construction, and external impedances (air, head, etc), it will tend to vibrate much easier in some frequencies versus others. With mechanical loads, damping factor becomes very critical for linear frequency response. 

 You can see this when measuring the distortion performance of a headphone amp. If you take distortion measurements when the amp is driving a linear load (i.e., a purely resistive load), the results will not be indicative of what the amp does when driving a 'real' load (i.e., headphones). A real measurement of a headphone amp's performance is how well it measures while driving low-impedance headphones. With most headphone amps, the results will change dramatically.

 The wikipedia article on damping factor is pretty simple, yet informing.

  Quote:


  Originally Posted by *Mazz* /img/forum/go_quote.gif 
_And if I read the definition of damping factor correctly (no guarantees!), different headphones with the same amp will have damping factor variations of 10 or more (because their impedances vary by 10 or more). Doesn't this then have a much bigger effect then balanced vs single-ended? Should I throw away my D5000s because their impedance is only 25ohms and get HD650s (which IIRC are in the region of 300 or 600 ohms)?_

 

YES!! You are absolutely correct. The impedance of the headphones will have as much of an affect on the damping factor as the amp. The tradeoff is usually a lower sensitivity (less efficiency). But, again, a good headphone amp won't have problems delivering enough gain.

 Thanks,
 Elias


----------



## EliasGwinn

Quote:


  Originally Posted by *dspargo* /img/forum/go_quote.gif 
_How would you connect to a HeadRoom Balanced Desktop Amp that has both balanced and unbalanced inputs?_

 

Hey dspargo,

 Simply connect the balanced outputs of the DAC1 to the balanced inputs of the HeadRoom (using balanced cables, of course).

 Let me know if you need any more assistance...I'll be glad to help.

 Thanks,
 Elias


----------



## EliasGwinn

Quote:


  Originally Posted by *anadin* /img/forum/go_quote.gif 
_I will be getting the DAC1 PRE._

 

Great. That means you'll have all three gain ranges available. You will probably want to use the lowest gain range, considering the low impedance of your headphones. 

 If you need assistance setting up the gain on your DAC1 PRE, just let me know. (It's pretty simple, but I'll be here just in case.)

 Thanks,
 Elias


----------



## EliasGwinn

Quote:


  Originally Posted by *tyler69* /img/forum/go_quote.gif 
_does the dac1 usb wake up from standby as soon as "any" source is recognized or does it only wake up when the source is not only recognized but also selected on the front?_

 

The DAC1 USB will wake up when the last selected source is recognized. In other words, the second part of your question is correct. If a signal is present on an input that is not currently selected, simply change the input selector to that input and you will be up and running.

  Quote:


  Originally Posted by *tyler69* /img/forum/go_quote.gif 
_p.s.: does anybody know the power consumption of the dac1usb while in standby mode?_

 

8 Watts.

 Thanks,
 Elias


----------



## EliasGwinn

Mazz, Poo, katalyst^

 First of all, thank you for alerting us to the price discrepancy issue of Australian dealer pricing. I've discussed this with our sales manager and vice-president, and we are looking into it. We don't want any dealers charging unreasonable amounts for our products. We'll do our best to determine what is causing this price increase.

 Something thing to keep in mind: the local dealer takes responsible for returns and warranty repairs. If you bought directly from us, it would cost $200 to ship it to us for a warranty repair. Although we enjoy a very low rate of product defects, having a local dealer can be thought of as an insurance plan.

 However, we will discuss your concerns with our Australian distributor, and determine the best route to take. I'll keep you all updated as we make progress...

 Thanks again for your invaluable feedback. 
 -Elias


----------



## EliasGwinn

Quote:


  Originally Posted by *Wavelength* /img/forum/go_quote.gif 
_Actually not true... remember you can't think of electronics in terms of voltage. You have to think of electronics in current and voltage. For any path their will be a voltage across and also that loop will carry the current signal. look at some of the app notes for power supplies or opamps most of them will comment on both aspects and will have the return (ground) reference to the point that yields the best results._

 

I'm not sure what your point is here. The Sennheiser HD650's use a seperate conductor for the return path of each driver. They are not tied together _until they are at the circuit board._ They are isolated w/ regards to voltage _and_ current...if that answers your question.

  Quote:


  Originally Posted by *Wavelength* /img/forum/go_quote.gif 
_Elias, the Arye are totally balanced on the input and the output. They have no reference to ground. With that comes some problems especially with sub woofers and such that will reference the NEGATIVE signal to earth ground.

 Take a look at their site:

ayre home

 The thing to think about here is if you do make a balanced output amplifier it is in it's own sense a totally different concept than a single ended amplifier. Therefore the specifications cannot be looked at as just a double single ended amplifier but on their own merits._

 

Their amplifiers are basically operating in bridged-mode. Nothing new there. In fact, most audio engineers will tell you to avoid bridged-mode unless you're desperate for power, since it will double the output impedance, distortion, noise, etc. 

 Please explain how balanced output amplifiers are "totally different concepts" from double single-ended amplifiers.

 Thanks,
 Elias


----------



## Matias

Quote:


  Originally Posted by *Bostonears* /img/forum/go_quote.gif 
_Matias,
 Did you independently test the fuses and the power cable, and find that each of them made the noticeable improvement on its own? Or, did you swap them both at once (which would make it impossible to distinguish which part was responsible for the sonic changes)?_

 

Yes, we tested them independently in that system (Krell 400xi, Dynaudio Focus 220, Arcam transport).
 Actually there were 6 variables independently tested:

 - 2006 DAC1 vs 2008 DAC1 USB (both on coax SPDIF): USB version is slightly better, softer sounding.

 - power cables: more defined low-bass dynamics (kick drum, plucked string bass), deeper soundstage and more natural voices.

 - HiFi-Tuning.com's fuses: same as power cables, but a lot cheaper, making it a "must buy". 
	

	
	
		
		

		
			





 - HiFi-Tuning.com's Noise Destroyers (2 units): slightly softer and airy cymbals, slightly deeper soundstage, more natural voices. Differences are small, however.

 - IC: big differences between those tested. Focus, soundstage, extension on both bass and treble.

 - digital coax: none to be heard, or at least extremely small so that we couldn't hear switching them. Note: both were good coax, not a good one vs a cheapo one.


----------



## Mazz

Quote:


  Originally Posted by *EliasGwinn* /img/forum/go_quote.gif 
_Something thing to keep in mind: the local dealer takes responsible for returns and warranty repairs. If you bought directly from us, it would cost $200 to ship it to us for a warranty repair. Although we enjoy a very low rate of product defects, having a local dealer can be thought of as an insurance plan._

 

Thanks for taking up this issue, Elias - much appreciated. I should point out that I don't own a DAC1, but it's an option I'm considering. However the Australian pricing leads to one of four choices:

 1. Bend over and pay the local prices, swearing all the way.
 2. Import one from overseas. This is not hard to achieve, despite the territorial exclusivity that Benchmark would like to enforce. There are reputable services that will help for a small fee, let alone friends in the US...
 3. Buy one second hand.
 4. Buy something else.

 Of these, (3) & (4) presumably lower Benchmark's ongoing sales, and you'll rarely hear about it directly from non-customers. I doubt (1) will occur at current prices as even if (2) DOES require a $200 repair shipment, I would still be several hundred dollars ahead of (1). Local dealers really need to appreciate that calculation as more and more consumers are capable of making it. If dealers lower their prices to the point where the cost of the "insurance" looks more reasonable, they'd have a much better chance of getting business - that they otherwise don't even know they're missing out on.

 I should also point out Benchmark's local prices are bad but could be even worse. I recently bought some Denon D5000s which retail in Australia for *three times* the going price in the US. Guess which option I didn't pursue?


----------



## Bostonears

Quote:


  Originally Posted by *Matias* /img/forum/go_quote.gif 
_Yes, we tested them independently in that system (Krell 400xi, Dynaudio Focus 220, Arcam transport).
 Actually there were 6 variables independently tested:

 - 2006 DAC1 vs 2008 DAC1 USB (both on coax SPDIF): USB version is slightly better, softer sounding.

 - power cables: more defined low-bass dynamics (kick drum, plucked string bass), deeper soundstage and more natural voices.

 - HiFi-Tuning.com's fuses: same as power cables, but a lot cheaper, making it a "must buy". 
	

	
	
		
		

		
		
	


	




 - HiFi-Tuning.com's Noise Destroyers (2 units): slightly softer and airy cymbals, slightly deeper soundstage, more natural voices. Differences are small, however.

 - IC: big differences between those tested. Focus, soundstage, extension on both bass and treble.

 - digital coax: none to be heard, or at least extremely small so that we couldn't hear switching them. Note: both were good coax, not a good one vs a cheapo one._

 

Very interesting comparison. Thanks.


----------



## tyler69

Thank you Elias.


----------



## poo

Quote:


  Originally Posted by *EliasGwinn* /img/forum/go_quote.gif 
_Something thing to keep in mind: the local dealer takes responsible for returns and warranty repairs. If you bought directly from us, it would cost $200 to ship it to us for a warranty repair. Although we enjoy a very low rate of product defects, having a local dealer can be thought of as an insurance plan._

 

Thanks for your reply to this issue Elias. There are fewer companies around that offer the level of service and genuine assistance that Benchmark seem to.

 In relation to the warranty issues you have raised, there are differences in Australian laws and trade organisations (like the ACCC) which may make the issue irrelevant. I'm no expert in the area, but through the course of my work have had to investigate at great length the way in which warranty is honoured by manufacturers of products sold in Australia. The ACCC's view (abridged) is that the warranty is an agreement between the manufacturer of a product and the end user. Essentially, regardless of where the product is purchased, the manufacturer has the responsibility of meeting the terms of the products' warranty (it is the manufacturer's product and warranty conditions after all, not the retailer's).

 There may be US laws or trade agreements that contraviene the ACCC's views, but considering that the body exists to support consumers, I would be surprised.


----------



## EliasGwinn

Quote:


  Originally Posted by *poo* /img/forum/go_quote.gif 
_Essentially, regardless of where the product is purchased, the manufacturer has the responsibility of meeting the terms of the products' warranty (it is the manufacturer's product and warranty conditions after all, not the retailer's)._

 

This is still the case. The warranty is always between the end user and Benchmark, and Benchmark always takes responsibility for all warranty issues. My point is that, in the event of a product return or warranty repair/replacement, the end user needs only to drop off the unit at the dealer from which it was bought. The dealer has the capacity to immediately take care of the situation. It makes it easier, quicker, and cheaper for the end user.

 Thanks,
 Elias


----------



## Wavelength

Quote:


  Originally Posted by *EliasGwinn* /img/forum/go_quote.gif 
_Their amplifiers are basically operating in bridged-mode. Nothing new there. In fact, most audio engineers will tell you to avoid bridged-mode unless you're desperate for power, since it will double the output impedance, distortion, noise, etc. 

 Please explain how balanced output amplifiers are "totally different concepts" from double single-ended amplifiers.

 Thanks,
 Elias_

 

Elias,

 Actually they are not in bridged mode and my point was that if you designed a well made balanced output system that it would preform better than simple putting two single ended amps together with one out of phase of another.

 A perfect example of the best differential driver would be a transformer. Truely the only real differential driver capable of true balanced operation.

 Thanks
 Gordon


----------



## dspargo

Quote:


  Originally Posted by *EliasGwinn* /img/forum/go_quote.gif 
_Hey dspargo,

 Simply connect the balanced outputs of the DAC1 to the balanced inputs of the HeadRoom (using balanced cables, of course).

 Let me know if you need any more assistance...I'll be glad to help.

 Thanks,
 Elias_

 

Do you recommend I go balanced to balanced or unbalanced to unbalanced?


----------



## EliasGwinn

Quote:


  Originally Posted by *dspargo* /img/forum/go_quote.gif 
_Do you recommend I go balanced to balanced or unbalanced to unbalanced?_

 

Balanced-to-Balanced.

 Thanks,
 Elias


----------



## Audio-Omega

Is Benchmark DAC better than those dacs found in $1000 to $3000 cd players ?


----------



## EliasGwinn

Quote:


  Originally Posted by *Audio-Omega* /img/forum/go_quote.gif 
_Is Benchmark DAC better than those dacs found in $1000 to $3000 cd players ?_

 

Without putting one of these cd players side-by-side next to the DAC1, I would not be able to answer this.

 I will say, we have not seen too many devices whose d-to-a performance matches the DAC1.

 Perhaps some of the other head-fiers can share specific experiences they've had with the DAC1 and cd transports...

 Thanks,
 Elias


----------



## Audio-Omega

It's hard to find a good 5-disc cd player, so I have been thinking about getting a Benchmark DAC1 to give my old cd player a new lease of life. Most $1000 plus cd players are single disc.


----------



## 8thdwarf

I would like the Dac-1 Pre to feed two unbalanced input stereo amplifiers without 'Y' adapters. Can the levels of the two outputs be adjusted close to each other once a balanced to unbalanced adapter is used? I hope this question is not stupid


----------



## Matias

Here's what is inside my Benchmark DAC1 USB. 
	

	
	
		
		

		
		
	


	




 The whole board.





 The heart: DAC and opamp.





 Outputs:





 Power:





 Phone amp and atenuator:


----------



## EliasGwinn

Quote:


  Originally Posted by *Audio-Omega* /img/forum/go_quote.gif 
_It's hard to find a good 5-disc cd player, so I have been thinking about getting a Benchmark DAC1 to give my old cd player a new lease of life. Most $1000 plus cd players are single disc._

 

You have the right idea. Let me know if you have any questions...

 Thanks,
 Elias


----------



## EliasGwinn

Quote:


  Originally Posted by *8thdwarf* /img/forum/go_quote.gif 
_I would like the Dac-1 Pre to feed two unbalanced input stereo amplifiers without 'Y' adapters. Can the levels of the two outputs be adjusted close to each other once a balanced to unbalanced adapter is used? I hope this question is not stupid 
	

	
	
		
		

		
			



_

 

This is a great question. The balanced and unbalanced outputs can be matched in calibrated mode using the calibration trim-pots, but if you are driving an amp directly, you will want to be in variable mode. The levels can be matched in variable mode. 

 However, the high-current output drivers on the DAC1 PRE are more then capable of driving two amp inputs via a Y-adapter. You will not lose any performance in this configuration. I would use this configuration if I were you. 

 Thanks,
 Elias


----------



## Scrith

Quote:


  Originally Posted by *EliasGwinn* /img/forum/go_quote.gif 
_However, the high-current output drivers on the DAC1 PRE are more then capable of driving two amp inputs via a Y-adapter. You will not lose any performance in this configuration. I would use this configuration if I were you._

 

I can verify this statement. I used balanced Y-adapters to connect my DAC1 (2006 version) to two amps at the same time via the calibrated output (one was a Dynamight headphone amp, the other was a PS Audio GCC-250, both of which have volume controls, which made the usage of the calibrated mode possible).

 I was concerned that I would lose some audio quality by sending the DAC1's output to two amps via splitters, so I conducted various careful tests (vs. a very expensive Cardas Golden Reference XLR cable!) and was able to enthusiastically conclude that there was no audible degradation of sound quality whatsoever (the test systems were DAC1 to SFT Dynamight to Qualia 010 headphones w/ Black Dragon cable and DAC1 to PS Audio GCC-250 to Totem Model 1 Signature speakers, with fancy cables throughout).

 I'm not using this setup anymore, so my balanced splitter cables are available for a reasonable price if you are interested (these were custom made for me by DIYCable.com, I believe the type was known there as "Exodus Quadfield").

Here is a related thread here at Head-Fi.


----------



## Matias

Alright, you will hate me for this, but after an amp upgrade, I clearly hear that the focus on SPDIF is a lot better then optical, at least using a cheap Monster Lightspeed 100. 

 Any suggestions on a high-end optic cable?


----------



## Bostonears

Quote:


  Originally Posted by *Matias* /img/forum/go_quote.gif 
_Alright, you will hate me for this, but after an amp upgrade, I clearly hear that the focus on SPDIF is a lot better then optical, at least using a cheap Monster Lightspeed 100. 

 Any suggestions on a high-end optic cable?_

 

If you're going to use TOSLINK, I highly recommend an inexpensive _glass _fiber optic cable, such as this one
Amazon.com: Amphenol Ultra Series - Premium Optical Glass TOSLINK Cable - 6': Electronics


----------



## indianbraker

i would love to own this unit. im stuck with a zero dac for now though so ill be thankful for what i got.


----------



## syfjhz22

DAC-1 is a great DAC, I love to listen it.


----------



## Matias

Quote:


  Originally Posted by *Bostonears* /img/forum/go_quote.gif 
_If you're going to use TOSLINK, I highly recommend an inexpensive glass fiber optic cable, such as this one
Amazon.com: Amphenol Ultra Series - Premium Optical Glass TOSLINK Cable - 6': Electronics_

 

Very interesting.
 Do you own one of these? Do they sound anything near SPDIF?


----------



## Bostonears

Quote:


  Originally Posted by *Matias* /img/forum/go_quote.gif 
_ Do you own one of these?_

 

Some factory (in China) churns out these low priced glass fiber optic cables for various brand names. I own three of them. I purchased mine at Sound Professionals, and they are often available on eBay (just search for "glass Toslink").

  Quote:


  Originally Posted by *Matias* /img/forum/go_quote.gif 
_Do they sound anything near SPDIF?_

 

SPDIF refers to the data protocol, not the cable itself, but I assume you want to know if they sound like coax digital cables. I have simultaneously hooked up my best coax digital cable (Kimber) and one of these glass fiber cables to a Squeezebox, fired up some tunes, and just switched the DAC back and forth between the coax and optical inputs. Over the years, I have tried this test with three different high quality DACs (Grace m902, Dodson DAC 263, and Benchmark DAC1 Pre) with dozens of songs, using both speakers and headphones, and in no case could I distinguish any difference between the coax and the glass fiber. With the Dodson DAC, the input switching was so seamless (absolutely no gaps, clicks, or other noises) that I wasn't sure the DAC was even switching inputs. I had to physically disconnect the cables during playback to be sure the DAC was actually using the corresponding input.

 In short, I'm sold on these cheap glass optical cables.


----------



## Matias

Thanks, Bostonears. You are right about that SPDIF: it's the same protocol on both cables, be it coax or optic. My mistake.

 Very interesting. I will search some of those, let's see. 
 Thank you very much.

 Did you try the fuses on the DAC1? Now those did wonders to it.


----------



## 8thdwarf

Elias, thanks for the help.


 Scrith thanks for the help, offer and link. I've some decisions 1st so I'll save your info.


----------



## Wavelength

Quote:


  Originally Posted by *Bostonears* /img/forum/go_quote.gif 
_Some factory (in China) churns out these low priced glass fiber optic cables for various brand names. I own three of them. I purchased mine at Sound Professionals, and they are often available on eBay (just search for "glass Toslink").


 SPDIF refers to the data protocol, not the cable itself, but I assume you want to know if they sound like coax digital cables. I have simultaneously hooked up my best coax digital cable (Kimber) and one of these glass fiber cables to a Squeezebox, fired up some tunes, and just switched the DAC back and forth between the coax and optical inputs. Over the years, I have tried this test with three different high quality DACs (Grace m902, Dodson DAC 263, and Benchmark DAC1 Pre) with dozens of songs, using both speakers and headphones, and in no case could I distinguish any difference between the coax and the glass fiber. With the Dodson DAC, the input switching was so seamless (absolutely no gaps, clicks, or other noises) that I wasn't sure the DAC was even switching inputs. I had to physically disconnect the cables during playback to be sure the DAC was actually using the corresponding input.

 In short, I'm sold on these cheap glass optical cables._

 

Boston,

 Remember that with SPDIF you are not only looking at data but the clock riding on the back of the data(Biphase). The best way to test the spdif interface is too monitor the PLL on the receiver side as well as the error indicators to determine which interface, cable etc... works best. I have a widget around here somewhere that has an AKM, Cirrus/Crystal and one of those TI receivers and allot of bond out testing stuff that I made years ago too test this.

 By looking at the PLL you can see how well the receiver is working at realigning the output clock Master Clock to the input stream. Basically the receivers look at the framing information to determine the Master Clock output. The Framing output of the pll is multiplied by 256 and the Master Clock is then outputed. From there the Bit Clock and Word Clock are derived. You can also put a spectral analyzer on the Master Clock and look for jitter another indication of how well the cable and interface are working.

 In my experience glass/plastic has never really done SPDIF justice, nor has AES/EBU.

 The best interface I found is BNC on both sides and using a truely 75 ohm cable. Even there the best I have found is the Nirvana Audio T2.

 Guys an RCA connector can never be 75 ohms. Anyone telling you they have rewritten the physics books and created a truely 75 ohm rca cable is only kidding themselves. Not too say these don't work... they just don't work as well.

 If you want to run longer SPDIF cables then I do agree that glass/plastic or toslink may be your best bet. The galvanic isolation will help the receiver side a ton.

 Thanks
 Gordon


----------



## iNiGFx

yeah, i wanna to sell my DA10 to get the DAC1U


----------



## EliasGwinn

Quote:


  Originally Posted by *iNiGFx* /img/forum/go_quote.gif 
_yeah, i wanna to sell my DA10 to get the DAC1U_

 

Welcome...


----------



## DigiPhx

Quote:


  Originally Posted by *EliasGwinn* /img/forum/go_quote.gif 
_Welcome... 
	

	
	
		
		

		
		
	


	


_

 

Welcome too,
 I'm planning to get DAC1U in this month, it will connect to my Dyna Air6 for my reference. hehe


----------



## doctorcilantro

When the XLR output is used but as unbalanced (w/ 3rd pin floated), does the DAC1 still output +4DBv ? I'm thinking of using the RCA unbalanced out as well and it seems that with the attenuator at 0 for XLR output the RCA should be at -10dbu when the XLR is calibrated to +4dbv at 0dbfs.

 Here are some measurements:

 DAC to tube preamp (0dbfs+ sine wave at -20db applied by Replay Gain in J. River playback software): .850v

 Pre to monoblock:

 Pre @ 9o'clock = 40mv
 Pre @ 12o'clock = 297mv
 Pre @ 2o'clock = .562v
 Pre @ 3o'clock = .935v

 I guess I'm trying to figure out what is happening at my RCA outs right now with the above measurements (given that I am using unbalanced XLR).

 thanks!
 DC


----------



## EliasGwinn

Quote:


  Originally Posted by *doctorcilantro* /img/forum/go_quote.gif 
_When the XLR output is used but as unbalanced (w/ 3rd pin floated), does the DAC1 still output +4DBv ?_

 

If you are using an unbalanced interconnect from the XLR output, the output will be 6 dB less then if it were a fully balanced connection. 

 With the output attenuators set at '0dB', the balanced outputs can reach +29 dBu peak (+9 dBu RMS). If use an unbalanced interconnect, it will only be able to reach +23 dBu max peak (+3 dBu RMS).

 I can clarify further if you'd like.

 Thanks,
 Elias


----------



## Quaddy

elias, would appreciate your input on this;

 i recently got my dac1*pre*, am still waiting on my balanced xlr pair of IC's which will run from the dac1's balanced outputs to the balanced inputs of my headroom balanced desktop! lot of balanced there 
	

	
	
		
		

		
			





 my question is, in light of that setup is there an optimal configuration that should be set to provide my balanced out with the best quality feed, i have got everything as stock ATM, and havent touched any switches, knobs or alike.

 its predominantly going to be used to feed the balanced amp with balanced phones, what do you recommend i tweak if anything(_pads, jumpers etc_)

 thanks!


----------



## EliasGwinn

Quote:


  Originally Posted by *Quaddy* /img/forum/go_quote.gif 
_what do you recommend i tweak if anything(pads, jumpers etc)_

 

As soon as you take the DAC1 PRE out of the box, set it so that the face-plate is facing northeast until the first sign of dawn. ...or, wait, is it southwest....??? 

 KIDDING!! 
	

	
	
		
		

		
		
	


	




 No need to tweak anything unless the output level is too hot/not hot enough. The only reason to adjust the attenuators is to accommodate for the sensitivity of the device being driven by the DAC1 PRE. Since sensitivities are all over the map, from piece to piece, the DAC1 PRE's attenuators will make this a non-issue.

 There's a more detailed explanation of this on our "Feedback" newsletter page: Benchmark Media Systems - Feedback Newsletter

 If you have any more questions about this, I'd be glad to help.

 Thanks,
 Elias


----------



## doctorcilantro

MusicMan mentioned here that he thinks XLR sounds better with zero attentuation; 0db attenuator setting via jumper. That's how I run my DAC1 now (assuming the USB is same with jumpers etc.).

 dC


----------



## EliasGwinn

Quote:


  Originally Posted by *doctorcilantro* /img/forum/go_quote.gif 
_MusicMan mentioned here that he thinks XLR sounds better with zero attentuation; 0db attenuator setting via jumper._

 

It will definitely sound louder! 

 And a hotter DAC1 level driving a pre will make the pre sound different, depending on how it reacts to hot signals.

 Thanks,
 Elias


----------



## Quaddy

hi elias

 great linkage

 its all in theory at the moment you see, making it slightly hard to envisage without my interconnects and getting hands on

 but i will definitely flick the gain switch on the headroom to low then as a good starting point and will work my way through the rest of the material you just provided - thanks

 like peter falk might say, _just one more thing_: for the usage mentioned before, am i best to leave it on variable output as opposed to calibrated? i am confused about the XLR trim-pots / variable / calibrated options for xlr out TBH 
	

	
	
		
		

		
		
	


	




 thanxion

  Quote:


  Originally Posted by *EliasGwinn* /img/forum/go_quote.gif 
_As soon as you take the DAC1 PRE out of the box, set it so that the face-plate is facing northeast until the first sign of dawn. ...or, wait, is it southwest....??? 

 KIDDING!! 
	

	
	
		
		

		
		
	


	




 No need to tweak anything unless the output level is too hot/not hot enough. The only reason to adjust the attenuators is to accommodate for the sensitivity of the device being driven by the DAC1 PRE. Since sensitivities are all over the map, from piece to piece, the DAC1 PRE's attenuators will make this a non-issue.

 There's a more detailed explanation of this on our "Feedback" newsletter page: Benchmark Media Systems - Feedback Newsletter

 If you have any more questions about this, I'd be glad to help.

 Thanks,
 Elias_


----------



## EliasGwinn

Quote:


  Originally Posted by *Quaddy* /img/forum/go_quote.gif 
_like peter falk might say, just one more thing: for the usage mentioned before, am i best to leave it on variable output as opposed to calibrated? i am confused about the XLR trim-pots / variable / calibrated options for xlr out TBH 
	

	
	
		
		

		
		
	


	




 thanxion_

 

Variable = front-panel volume control active
 Calibrated = front-panel volume control inactive; constant level output (similar to a transport with no volume control)

 Don't worry about the trim-pots...they're only there in case you want to calibrate the output to a standard level that your system is designed for...more of a professional audio thing.

 For your setup, you can keep the DAC1 PRE in calibrated mode and use the volume control on the Headroom only. OR, you can keep the DAC1 PRE in variable mode if you want to control the volume using the volume knob on the DAC1 PRE. OR, you can keep the DAC1 PRE in variable mode if you want to drive the headroom with a specific output level from the DAC1 PRE. In that case, you would simply turn up the volume knob until you've reached the appropriate output level, and then keep it set there.

 Options, options, options... 
	

	
	
		
		

		
		
	


	




 Thanks,
 Elias


----------



## doctorcilantro

Quote:


  Originally Posted by *EliasGwinn* /img/forum/go_quote.gif 
_If you are using an unbalanced interconnect from the XLR output, the output will be 6 dB less then if it were a fully balanced connection. 

 With the output attenuators set at '0dB', the balanced outputs can reach +29 dBu peak (+9 dBu RMS). If use an unbalanced interconnect, it will only be able to reach +23 dBu max peak (+3 dBu RMS).

 I can clarify further if you'd like.

 Thanks,
 Elias_

 

I thought the RCA outs were like a Tape Out and unaffected by the calibration pots on the back? Are they always at -10dbu or just 6db less than whatever is set for the XLR out?

 thanks,
 dc


----------



## EliasGwinn

Quote:


  Originally Posted by *doctorcilantro* /img/forum/go_quote.gif 
_I thought the RCA outs were like a Tape Out and unaffected by the calibration pots on the back? Are they always at -10dbu or just 6db less than whatever is set for the XLR out?

 thanks,
 dc_

 

The RCA outs are also calibrated by the trim-pots. However, the attenuators only affect the XLR outputs, not the RCA outputs.

 What I was referring to in the quote is using the XLR outputs with unbalanced cables (XLR connectors with pin-3 floating). This will have a total output of 6 dB less then if balanced cables were used.

 With 0 dB XLR attenuation, the RCA outputs are always 19 dB lower then the full balanced outputs. With 10 dB XLR attenuation, the RCA outputs are 9 dB less. And so on...

 I hope this answers your question.

 Thanks,
 Elias


----------



## Quaddy

thanks elias. you have helped me out there. cheers!!


----------



## nd4speed

If your pre is anything like the USB, listen to it on the default 20 db pad first; I found this ended up working best (I started with the 30db pad but it was too soft). You really don't want to touch the trim pots, as they are calibrated at the factory (unless you're prepared to measure the outputs).

  Quote:


  Originally Posted by *Quaddy* /img/forum/go_quote.gif 
_hi elias

 great linkage

 its all in theory at the moment you see, making it slightly hard to envisage without my interconnects and getting hands on

 but i will definitely flick the gain switch on the headroom to low then as a good starting point and will work my way through the rest of the material you just provided - thanks

 like peter falk might say, just one more thing: for the usage mentioned before, am i best to leave it on variable output as opposed to calibrated? i am confused about the XLR trim-pots / variable / calibrated options for xlr out TBH 
	

	
	
		
		

		
		
	


	




 thanxion_


----------



## doctorcilantro

del


----------



## doctorcilantro

Thanks. So would it further important to match the Contour tube to the other two?

 On another note, I just adjusted the trim pots on my DAC1 output. Previously, going to my mono-blocks with an input sensitivity of 1v, I had .880v at about 2 o'clock on the Sunfire preamp master gain. I didn't have enough headroom and never could use the volume above 10 o'clock.Carver discusses a few disadvantages of this situation in the manual, so I re-adjusted the system to have:

 DAC1 outputs 265.5mv per channel into Sunfire
 Sunfire outputs to Quicksilver mini-mites:
 9 20.5mv

 12 122mv

 3 323

 MAX .995v

 Things sound sweeter already and I really enjoy the greater control of the gain (more steps?). The mini-mites are 25w per channel and the RF-7 speakers are 98db/w sensitive so I still have high gain but with much better low volume range control. The overall volume is still suprisingly loud at even 11 playing something chill like Jack Johnson. So far so good...


 So......Serjazo told me:


  Quote:


 "A max input signal of 1.2 Vrms shouldn't clip the inputs on your Sunfire....I didn't look at the manual, but it should handle that signal level easily.

 Where do you run your volume control, though? My measurements of several preamps and the pre-out of an NAD integrated showed that the "zero gain" point is usually around 1 to 2 o'clock.....so if you run the knob lower than that, *the preamp is actually attenuating the input signal, sending a weaker one to the power amp!*

 (The specs on the Sunfire show the line input gain to be 12 dB, that's a voltage gain factor of 4, so it would drive the outputs to 8 Vrms with the knob cranked to the max with a typical CD player input....and that's what my Musical Fidelity CDPre24 was spec'ed at, within a couple of tenths Vrms.)

 For the life of me, I fail to understand the decisions about the relative gain of devices in the chain these days. It does make sense to increase the voltage out of a CD player or DAC.....thus placing the signal higher above the noise floor.....versus what typical old phono preamps put out, which was usually less than 1 Vrms on a test tone, or tape decks that were designed to put out 0.775 Vrms when the VU meters were at 0 dB VU. But those were not hard upper limits as is 0 dBFS in the digital realm...there was headroom above the 0 dB references in those days, because 0 dB wasn't referenced to a constrained maximum level.

 I really wish I knew why makers construct power amps that clip at 1 or 1.2 Vrms when preamps can put out 8 to 10 Vrms. The current generation Quad SS amps clip at the old 0.775 Vrms standard that was typical in European gear for many years....that's ~ 8 dB less than a CD player puts out! Many users end up having to add fixed attenuators between non-Quad preamps and the 606 or 909 to avoid hair-trigger volume control.

 Power amps could be designed for 8 to 10 dB less gain in most cases and probably be even more usable as a result--finer volume control over the range of CD's that are mastered all over the place these days.

 Maybe it's to give the user the perception that if the system is loud with the volume control at 9:30, it's great because that means there is a lot of headroom available.......when in reality, it just means you better not go above 1 o'clock or you will damage your tweeters in a hurry!"


----------



## doctorcilantro

I was wondering if the signal ground is connected the chassis ground on the DAC1 (pin 1 of the XLR out)?

 thanks
 DC


----------



## EliasGwinn

Quote:


  Originally Posted by *doctorcilantro* /img/forum/go_quote.gif 
_I was wondering if the signal ground is connected the chassis ground on the DAC1 (pin 1 of the XLR out)?

 thanks
 DC_

 

If by "signal ground", you're referring to pin 1, then YES. Pin 1, which is connected to the sheild of the cable is tied to the chassis of the DAC1, creating a 'complete' signal sheild.

 Thanks,
 Elias


----------



## EliasGwinn

To our Friends in Australian:

 As per our discussion, we've spoken with our Australian distributor. Together, we worked through the pricing structure, and we've managed to reduce the price by $216 AUD!! 

 Please see the following website for more details: 

Sound & Music Distribution - Product Information for Benchmark BENCHMARKDAC1

 Thanks again for your feedback!
 -Elias


----------



## Scrith

Quote:


  Originally Posted by *nd4speed* /img/forum/go_quote.gif 
_If your pre is anything like the USB, listen to it on the default 20 db pad first; I found this ended up working best (I started with the 30db pad but it was too soft)._

 

Any thoughts on 20db attenuation in the DAC1 USB (or Pre) sounding better than 30db, Elias? I set mine to 30db upon acquiring the DAC1 USB also (and find that I am usually listening to music with the volume on the DAC1 USB at around 12 o'clock) and am now wondering if I should give 20db a try (although I'm guessing that will result in the volume knob being closer to 9 o'clock for my typical listening situation). My amps are Jeff Rowland Model 201 monoblocks.


----------



## EliasGwinn

Quote:


  Originally Posted by *Scrith* /img/forum/go_quote.gif 
_Any thoughts on 20db attenuation in the DAC1 USB (or Pre) sounding better than 30db, Elias? I set mine to 30db upon acquiring the DAC1 USB also (and find that I am usually listening to music with the volume on the DAC1 USB at around 12 o'clock) and am now wondering if I should give 20db a try (although I'm guessing that will result in the volume knob being closer to 9 o'clock for my typical listening situation). My amps are Jeff Rowland Model 201 monoblocks._

 

It sounds like 30 dB will be perfect for you. Here is an article from our "Feedback" newsletter discussing the differences in attenuation settings: Benchmark Media: Feedback Newsletter

 Thanks,
 Elias


----------



## nd4speed

Hi Elias,
 Just wondering if you guys do all your DAC-1 in-house testing and measurements with the same USB cable that comes bundled with the Benchmark to consumers, or are you guys using aftermarket/"special" USB cables. If so which one? Thanks.


----------



## akwok

Quote:


  Originally Posted by *nd4speed* /img/forum/go_quote.gif 
_Hi Elias,
 Just wondering if you guys do all your DAC-1 in-house testing and measurements with the same USB cable that comes bundled with the Benchmark to consumers, or are you guys using aftermarket/"special" USB cables. If so which one? Thanks._

 

It wouldn't matter.


----------



## Matias

'Team "I don't believe in cables"' => we forgive you.


----------



## cansman

Quote:


  Originally Posted by *EliasGwinn* /img/forum/go_quote.gif 
_To our Friends in Australian:

 As per our discussion, we've spoken with our Australian distributor. Together, we worked through the pricing structure, and we've managed to reduce the price by $216 AUD!!_

 

Thanks Elias and Benchmark for your willingness to be customer-driven in contrast to an increasingly profit-driven world! Appreciate it!

 cansman


----------



## anadin

Anychance of doing the same for "rip-off Britain".


----------



## poo

Quote:


  Originally Posted by *EliasGwinn* /img/forum/go_quote.gif 
_To our Friends in Australian:

 As per our discussion, we've spoken with our Australian distributor. Together, we worked through the pricing structure, and we've managed to reduce the price by $216 AUD!! 

 Please see the following website for more details: 

Sound & Music Distribution - Product Information for Benchmark BENCHMARKDAC1

 Thanks again for your feedback!
 -Elias_

 

Great to hear - it's a significant reduction that should help all stakeholders, and a much more realistic reflection of the US price set by Benchmark. 

 Thanks for understanding the issues presented, taking them seriously and providing a solution - couldn't ask for more, and certainly wouldn't expect as much from other suppliers...


----------



## EliasGwinn

Quote:


  Originally Posted by *nd4speed* /img/forum/go_quote.gif 
_Hi Elias,
 Just wondering if you guys do all your DAC-1 in-house testing and measurements with the same USB cable that comes bundled with the Benchmark to consumers, or are you guys using aftermarket/"special" USB cables. If so which one? Thanks._

 

We use the same USB cable that we bundle-package with our DAC1 USB/PRE's.

 Thanks,
 Elias


----------



## Matias

Quote:


  Originally Posted by *Bostonears* /img/forum/go_quote.gif 
_If you're going to use TOSLINK, I highly recommend an inexpensive glass fiber optic cable, such as this one
Amazon.com: Amphenol Ultra Series - Premium Optical Glass TOSLINK Cable - 6': Electronics



_

 

This Amphenol Premium Optical TOSLINK arrived here yesterday. I listened to it in the late hours and it is just as *bad* as the cheap Monster LightSpeed 100 I was using... 
	

	
	
		
		

		
		
	


	




 Coax still rules the SPDIF world. 

 Now I need to sell both...


----------



## Quaddy

Quote:


  Originally Posted by *Matias* /img/forum/go_quote.gif 
_This Amphenol Premium Optical TOSLINK arrived here yesterday. I listened to it in the late hours and it is just as *bad* as the cheap Monster LightSpeed 100 I was using... 
	

	
	
		
		

		
		
	


	




 Coax still rules the SPDIF world. 

 Now I need to sell both..._

 

yup, me and lisnalee have been discussing this and think we both are preferring coax over the toslink, and lisnalee is even using nice qual VDH too


----------



## poo

Quote:


  Originally Posted by *Quaddy* /img/forum/go_quote.gif 
_yup, me and lisnalee have been discussing this and think we both are preferring coax over the toslink, and lisnalee is even using nice qual VDH too 
	

	
	
		
		

		
		
	


	


_

 

I would expect that if both cables are sending a bit-perfect signal, the sound should be identical... but then I'm not using Nordost Valhalla, so perhaps I'm missing half the bits...


----------



## Matias

Bits are OK, timing is not. That is called jitter.
 Square waves can be deformed and be misconverted. Proof is that the SPDIF code has error correction algorythms.
 There is more than 0s and 1s.

 But more important, the ear senses the sound focusing more or less, and at the end of the day, that is what matters.


----------



## Dougr33

Hey Elias,
 I'm a poor audiophile on a budget; can't really afford my music addiction, so ...

 1. Is there any reason (besides improved OpAmps) not to forgo the USB, and feed a normal DAC1 with the digital out of a good audio card (M-Audio Audiophile 24/96) from a program like Itunes or Foobar2000? Any 'sonic' downsides vs. the USB input on the more expensive model?

 2. Can I really get the "same" quality output if my transport is a $100 panasonic dvd or an excellent (but 8 yr. old) Rotel CD player (rca digital outs)? I plan on directly driving a good amp like the NAD C272 or maybe even an S200 into my Von Schweikert VR2s.

 3. If I go the simpler route of a Roku or Squeezebox, does Benchmark find one better than other (strictly audiowise)? Do they really need an improved power supply to feed a clean digital stream to the DAC1.

 4. Lastly, as you know, some power amps have "balanced" inputs, and some have "truly balanced" inputs. Figuring a low-noise home environment, for a 12' run, would it really matter (decent shielded RCAs vs. balanced), and is it true that if the amp isn't "truly" balanced, the results are usually better with unbalanced RCAs. Is there a distance (will never be longer than 25') where balanced becomes de rigour?

 Thank you so much for your time and your professionalism


----------



## EliasGwinn

Hi Doug!

  Quote:


  Originally Posted by *Dougr33* /img/forum/go_quote.gif 
_1. Is there any reason (besides improved OpAmps) not to forgo the USB, and feed a normal DAC1 with the digital out of a good audio card (M-Audio Audiophile 24/96) from a program like Itunes or Foobar2000? Any 'sonic' downsides vs. the USB input on the more expensive model?_

 

Using an original DAC1 with a card like the M-Audio Audiophile is not a bad way to go. That system will sound very good. However, you should keep in mind that the Audiophile card is not bit-transparent. That means that the digital audio data that is present in the recording is somehow modified by this card and/or its drivers. I don't know how or why it is being modified, I only know that the tests show that the data going in to the card does not match the data coming out. 

 The USB input of the DAC1 is completely bit-transparent, so there are no doubts about the integrity of the audio data.

  Quote:


  Originally Posted by *Dougr33* /img/forum/go_quote.gif 
_2. Can I really get the "same" quality output if my transport is a $100 panasonic dvd or an excellent (but 8 yr. old) Rotel CD player (rca digital outs)? I plan on directly driving a good amp like the NAD C272 or maybe even an S200 into my Von Schweikert VR2s._

 

Yes.

  Quote:


  Originally Posted by *Dougr33* /img/forum/go_quote.gif 
_3. If I go the simpler route of a Roku or Squeezebox, does Benchmark find one better than other (strictly audiowise)? Do they really need an improved power supply to feed a clean digital stream to the DAC1._

 

I haven't tested these devices, so I can't comment on them. 

  Quote:


  Originally Posted by *Dougr33* /img/forum/go_quote.gif 
_4. Lastly, as you know, some power amps have "balanced" inputs, and some have "truly balanced" inputs. Figuring a low-noise home environment, for a 12' run, would it really matter (decent shielded RCAs vs. balanced), and is it true that if the amp isn't "truly" balanced, the results are usually better with unbalanced RCAs. Is there a distance (will never be longer than 25') where balanced becomes de rigour?_

 

If the amp does not have a differential amplifier on the input, then you would do better using 75-ohm coaxial cable (RG-59 or better), simply because the sheilding will be better. 25' is a pretty long run for unbalanced...but it all depends on your EMI environment.

 Thanks,
 Elias


----------



## Covenant

Quote:


  Originally Posted by *EliasGwinn* /img/forum/go_quote.gif 
_To our Friends in Australian:

 As per our discussion, we've spoken with our Australian distributor. Together, we worked through the pricing structure, and we've managed to reduce the price by $216 AUD!!_

 

Are the DAC1/USB/PRE purchasable directly from the US supplier, or is the local distributor the only channel to purchase one?

 Checking Sound & Music's pricing page on the DAC1 USB they list it for $2005 AUD, which is more than $500 more expensive than buying direct from the US, even after the price reduction.


----------



## G-U-E-S-T

Quote:


  Originally Posted by *Matias* /img/forum/go_quote.gif 
_Bits are OK, timing is not. That is called jitter.
 Square waves can be deformed and be misconverted. Proof is that the SPDIF code has error correction algorythms.
 There is more than 0s and 1s..._

 

OK, dumb question... Why are users hearing different things using coax vs. optical? Since the users are not hearing bit-errors (i.e. clicks and pops), it has been proposed to be jitter that accounts for the difference here. But the DAC1 Pre is supposed to be jitter-immune, right? Elias what are your thoughts on this please, as far as what might account for such audible coax/optical input differences with the DAC1 Pre?


----------



## doctorcilantro

Quote:


 OK, dumb question... Why are users hearing different things using coax vs. optical? Since the users are not hearing bit-errors (i.e. clicks and pops), it has been proposed to be jitter that accounts for the difference here. But the DAC1 Pre is supposed to be jitter-immune, right? Elias what are your thoughts on this please, as far as what might account for such audible coax/optical input differences with the DAC1 Pre? 
 

Toslink is immune to RFI/EMI.....

 Also, you can't simply point at the cable. What is is attached to? I've read that even digital outs can impart distortion. I wish I had the article handy; I'll look for it. 

 DC


----------



## EliasGwinn

Quote:


  Originally Posted by *Covenant* /img/forum/go_quote.gif 
_Are the DAC1/USB/PRE purchasable directly from the US supplier, or is the local distributor the only channel to purchase one?

 Checking Sound & Music's pricing page on the DAC1 USB they list it for $2005 AUD, which is more than $500 more expensive than buying direct from the US, even after the price reduction._

 

Covenant,

 We cannot sell (ship) directly to Australia. If you have a shipping address outside of Australia that we can deliver to, we could possibly arrange that.

 However, even if we did sell directly to Australia, after shipping costs and import taxes, you would be paying close to $2000 AUD anyway. So, considering the convenience of having the local shop to deal with, its not too far off.

 Thanks,
 Elias


----------



## anadin

I would really appreciate it if anyone could tell me or even EliasGwinn himself if its ok to drive a pair of Balanced HD650's directly from the Benchmark DAC 1 Pre's balanced outputs.

 Many thanks.


----------



## EliasGwinn

Quote:


  Originally Posted by *G-U-E-S-T* /img/forum/go_quote.gif 
_OK, dumb question... Why are users hearing different things using coax vs. optical? Since the users are not hearing bit-errors (i.e. clicks and pops), it has been proposed to be jitter that accounts for the difference here. But the DAC1 Pre is supposed to be jitter-immune, right? Elias what are your thoughts on this please, as far as what might account for such audible coax/optical input differences with the DAC1 Pre?_

 

GUEST,

 We have done numerous listening tests and measurement tests with a variety of cables, and we haven't been able to detect any differences. We've done tests using 1000 feet of Cat-5 cable vs. 3-foot high quality 75-ohm coax cable vs. the skinny cheap optical cable that comes free with some devices vs. 20' glass optical cable, etc. Based on our tests, there are no audible differences between different types of cable. This is especially evident when using an 'ABX box', which enables the listener to switch between DAC1's using different types of cables, but never knowing which one they are listening to.

 I know some people have said that they hear differences, and I can't explain their experiences without actually knowing all the variables involved. (Perhaps the data coming from optical was different vs. coax...???). 

 Thanks,
 Elias


----------



## EliasGwinn

Quote:


  Originally Posted by *anadin* /img/forum/go_quote.gif 
_I would really appreciate it if anyone could tell me or even EliasGwinn himself if its ok to drive a pair of Balanced HD650's directly from the Benchmark DAC 1 Pre's balanced outputs.

 Many thanks._

 

Anadin,

 You can use the balanced outputs to drive balanced headphones without worrying about damaging the unit. However, we strongly suggest not using balanced headphones at all, as they expose the weaknesses of both headphones and the headphone amplifier (this goes for all cans and amps, as it is an inferior topology). 

 We've discussed this many times on this thread...including just a few weeks ago. However, I can explain it again, if you like.

 Thanks,
 Elias


----------



## sk007

Elias,

 I noticed that the silver units now has black knobs. so the question is: are there any other changes inside the DAC1 USB to make it a new production run different than the ones starting 2007?

 how do you guys think about the new black knobs? cosmetic-wise of course


----------



## Headphony

Quote:


  Originally Posted by *sk007* /img/forum/go_quote.gif 
_...how do you guys think about the new black knobs? cosmetic-wise of course 
	

	
	
		
		

		
		
	


	


_

 

I like the silver knobs better.


----------



## EliasGwinn

Quote:


  Originally Posted by *sk007* /img/forum/go_quote.gif 
_I noticed that the silver units now has black knobs. so the question is: are there any other changes inside the DAC1 USB to make it a new production run different than the ones starting 2007?_

 

No, the black knob / silver faceplate scheme is simply a new look that we are cultivating. It began with our PRE420 (a professional microphone pre-amplifier). We were very happy with that look, and our customers we as well. Next was the DAC1 PRE. It has also been very successful (for more reasons then its color scheme, however 
	

	
	
		
		

		
		
	


	




). 

 I think it gives us a very distinguishable look...I like it.

  Quote:


  Originally Posted by *sk007* /img/forum/go_quote.gif 
_how do you guys think about the new black knobs? cosmetic-wise of course 
	

	
	
		
		

		
		
	


	


_

 

Great question for the head-fi crew!! I'm curious as well.

 I've already said it, but I'll register my official opinion again:

 I LIKE THE SILVER/BLACK COMBO!! 
	

	
	
		
		

		
		
	


	




 Thanks,
 Elias


----------



## stvn758

Why only one Toslink connector?

 Cheaper units have several more inputs.


----------



## EliasGwinn

Quote:


  Originally Posted by *stvn758* /img/forum/go_quote.gif 
_Why only one Toslink connector?_

 

There is no more room on the rear panel. To add more TOSLink connectors, we would have to lose other inputs and/or outputs or make the box bigger. We didn't want to lose any of the other I/O's, and our customers love the size of the unit. Also, almost no one else has requested additional TOSlink. So...that's the reasons.

  Quote:


  Originally Posted by *stvn758* /img/forum/go_quote.gif 
_Cheaper units have several more inputs.
	

	
	
		
		

		
		
	


	


_

 

Additional TOSLink inputs don't cost more money. It is simply a design decision. For the reasons above, we decided to have only one TOSlink.

 Thanks,
 Elias


----------



## stvn758

M AUDIO CO2 - COAXIAL / OPTICAL BI-DIRECTIONAL DIGITAL CONVERTER- EACH

 Both my CD players have Toslink. If I used one of these adaptors am I right in thinking your Jitter Lock would see no serious harm came to the music?


----------



## Matias

Quote:


  Originally Posted by *EliasGwinn* /img/forum/go_quote.gif 
_GUEST,

 We have done numerous listening tests and measurement tests with a variety of cables, and we haven't been able to detect any differences. We've done tests using 1000 feet of Cat-5 cable vs. 3-foot high quality 75-ohm coax cable vs. the skinny cheap optical cable that comes free with some devices vs. 20' glass optical cable, etc. Based on our tests, there are no audible differences between different types of cable. This is especially evident when using an 'ABX box', which enables the listener to switch between DAC1's using different types of cables, but never knowing which one they are listening to.

 I know some people have said that they hear differences, and I can't explain their experiences without actually knowing all the variables involved. (Perhaps the data coming from optical was different vs. coax...???). 

 Thanks,
 Elias_

 

I also believe the optical out of the ESI Juli@ may be the cause of this result.
 But still, others have also noted in completely different systems that coax sound more focused then optical. Maybe optical converters are worse? Who knows!


----------



## EliasGwinn

Quote:


  Originally Posted by *stvn758* /img/forum/go_quote.gif 
_M AUDIO CO2 - COAXIAL / OPTICAL BI-DIRECTIONAL DIGITAL CONVERTER- EACH

 Both my CD players have Toslink. If I used one of these adaptors am I right in thinking your Jitter Lock would see no serious harm came to the music?_

 

Stvn, 

 As far as jitter is concerned, you are correct. You don't have to worry about the jitter that this device may cause. 

 However, there's a lot about how the M-Audio box works that I don't know. Since its active (powered), it makes me a little nervous that the data may be changed. Maybe its fine...I don't know. 

 An safer and less expensive solution is an optical switcher, like the one here: Amazon.com: Nyrius SW100 Digital Audio Optical Toslink 3-Way Selector Switch for Fiber Optic Home Theater Connections (Silver): Electronics

 Thanks,
 Elias


----------



## pbarach

Quote:


  Originally Posted by *EliasGwinn* /img/forum/go_quote.gif 
_The USB input of the DAC1 is completely bit-transparent, so there are no doubts about the integrity of the audio data._

 

So if I have a 24-bit audio file on my Windows XP PC, then this will be transmitted at 24 bits (and bit-perfectly) to the DAC1 via the USB input?

 An additional question: If I decide I don't need the USB input, and I decide instead to purchase the DAC1 model that comes without the USB input, are there any other differences in the circuit design or components of these two DAC1 models, other than not having that USB input in the less expensive model?


----------



## stvn758

Quote:


  Originally Posted by *EliasGwinn* /img/forum/go_quote.gif 
_Stvn, 

 As far as jitter is concerned, you are correct. You don't have to worry about the jitter that this device may cause. 

 However, there's a lot about how the M-Audio box works that I don't know. Since its active (powered), it makes me a little nervous that the data may be changed. Maybe its fine...I don't know. 

 An safer and less expensive solution is an optical switcher, like the one here: Amazon.com: Nyrius SW100 Digital Audio Optical Toslink 3-Way Selector Switch for Fiber Optic Home Theater Connections (Silver): Electronics

 Thanks,
 Elias_

 

Kikatek.com | (F8V3152AEA) Belkin Toslink Splitter ODT F/ 2 x ODT F (Blister)

 How about this, looks perfect. No messing about, no power source - obviously don't play both CD players at once.


----------



## emmodad

Quote:


  Originally Posted by *pbarach* /img/forum/go_quote.gif 
_An additional question: If I decide I don't need the USB input, and I decide instead to purchase the DAC1 model that comes without the USB input, are there any other differences in the circuit design or components of these two DAC1 models, other than not having that USB input in the less expensive model?_

 

perhaps you could invest a few moments in reading/research?:


> 1/ The DAC1 - Benchmark Media Systems, Inc.
> 
> 2/ asked and answered countless times in this thread.


poor Elias is overloaded enough, shouldn't have to address this again.


----------



## pbarach

Quote:


  Originally Posted by *emmodad* /img/forum/go_quote.gif 
_perhaps you could invest a few moments in reading/research?:



			1/ The DAC1 - Benchmark Media Systems, Inc.

 2/ asked and answered countless times in this thread.
		
Click to expand...

poor Elias is overloaded enough, shouldn't have to address this again._

 

Your sarcasm isn't appreciated, and I would not have asked the questions if my research (yes, I did some) had located an answer. All you had to do was send the link and be helpful. The link does not, however, answer my questions about whether the DAC1 USB and the non-USB models are otherwise the same in design and components.

 I hope you don't work for Benchmark, because I would never purchase anything from a manufacturer who responded to a potential customer with the imperious tone that your post displayed.


----------



## emmodad

Quote:


  Originally Posted by *pbarach* /img/forum/go_quote.gif 
_Your sarcasm isn't appreciated, and I would not have asked the questions if my research (yes, I did some) had located an answer. All you had to do was send the link and be helpful. The link does not, however, answer my questions about whether the DAC1 USB and the non-USB models are otherwise the same in design and components.

 I hope you don't work for Benchmark, because I would never purchase anything from a manufacturer who responded to a potential customer with the imperious tone that your post displayed._

 

no I don't work for Benchmark.

 yes, the Benchmark site has info addressing your question.

 your precise question has been discussed numerous times in this thread.

 read. then post.


----------



## emmodad

oh and btw pbarach, i decided to see how hard it would be to find the info you're looking for. i set a limit of five whole minutes, and used the amazing capability of "search this thread" function. spent a few minutes searching on various combos of terms, and inward from each end of the thread (begin, ie after product discussion started; and most-recent)

 here are a few simple results in under five minutes search / reading time (hint -- instead of posting to elias, do a logically-obvious search: his posts in this thread): 161, 216, 1252, 1753, 1762.....

 just imagine: you invest 10-15 minutes.. or (shudder) half an hour.... you'll see that elias has addressed your questions a few times.


----------



## infinitesymphony

Since so many people seem to want to know the technical differences between years and models of the DAC1, maybe those changes could be posted on the Benchmark Audio Wiki to avoid redundancy.


----------



## pbarach

Flame wars are a stupid waste of everyone's time, so I'll refrain for continuing the one that someone seems to be igniting in response to my question. If I happened not to find the information and asked my questions here, all that was needed was a pointer or a hyperlink to the right place to look. Clearly, since these two questions have been asked here repeatedly, the answers aren't easy enough to find. 

 The Benchmark WIKI, which I had read before posting here, says nothing about various in the components or design of the various DAC1 models, which was my second question. I also found nothing about this on the Benchmark website. I want to go through the 94 pages of this thread and MAYBE find what I needed to know, when perhaps someone who posts here (Elias or anyone else) would be helping to help someone who's new to this area of audio components.


----------



## Quaddy

maybe this can help highlight some of the differences between dac1 pre and usb - sorry if this wasnt your question

Benchmark DAC1 now available with USB - Page 121 - Head-Fi: Covering Headphones, Earphones and Portable Audio


----------



## emmodad

Quote:


  Originally Posted by *pbarach* /img/forum/go_quote.gif 
_Flame wars ..._

 

flame wars? hardly. more an admonishment that you could perhaps put forth some effort. 

  Quote:


  Originally Posted by *pbarach* /img/forum/go_quote.gif 
_The Benchmark WIKI, which I had read before posting here, says nothing about various in the components or design of the various DAC1 models, which was my second question. I also found nothing about this on the Benchmark website._

 

ummmm.... RTFM? you know, the documentation for this USD1000 product expenditure you're considering?


> open new tab in browser
> go to benchmark website
> select DAC1 USB product page
> select link to view manual
> ...


 hmmm. if the audio path circuit topology is unchanged from DAC1, what are the chips used in the DAC1 and this thing?


> open new tab
> google "benchmark DAC1 components"
> scan first few pages of links, opening several which are reviews including discussion of components (dogpile search yields even more). also find and open link with detail discussion of component selection by DAC1 product designer [here pbarach, as it appears to be such a burden: "An inside look at the Benchmark DAC1 - from the designer" Digital Drive ]
> 
> total time expended: 3 mins. reading the John Siau linked page: 1.5 mins.


  Quote:


  Originally Posted by *pbarach* /img/forum/go_quote.gif 
_I want to go through the 94 pages of this thread and MAYBE find what I needed to know...._

 

hmmm, yes, perhaps do see one of your points. 94 pages at ~1 minute per page to scan for relevant posts (picking up plenty of useful info on the way)... maybe add 30 minutes to read few posts, follow some links, use the search function...

 two hours or so of time, why yes, can see how that burden could be a significant terrible personal inconvenience, on top of ten minutes or so of other searching.

 oh btw, the page noted by Quaddy, as well as many more with useful info answering your questions, also comes up in the search results of elias' posts in this thread from which i listed a few post numbers. terribly inconvenient, though, requires reading thru several pages listing his posts....


----------



## TreAdidas

Quote:


  Originally Posted by *EliasGwinn* /img/forum/go_quote.gif 
_Great question for the head-fi crew!! I'm curious as well.

 I've already said it, but I'll register my official opinion again:

 I LIKE THE SILVER/BLACK COMBO!! 
	

	
	
		
		

		
		
	


	




 Thanks,
 Elias_

 

You know it is funny you ask that because when I first saw it I thought wow that is good combo but I wished it were the inverse! A black faceplate with silver knobs would look soooo trick.


----------



## Dougr33

Quote:


  Originally Posted by *emmodad* /img/forum/go_quote.gif 
_flame wars? hardly. more an admonishment that you could perhaps put forth some effort. 

 ETC, ETC, ad nauseum...._

 

Seriously, are you okay? You could have taken all this energy (and typing) and helped the next 20 people trying to find answers but lacking your search savvy. Wow.


----------



## Dougr33

Elias...

 Design-wise, I like the black knob on the Pre.. it's balanced by the black selector knob. Not as much on the smaller units.. I think the all-silver is quite elegant. But it's not offensive either!


----------



## Quaddy

the PRE is beautiful in the flesh - have no fear - i have no cosmetic concerns here! - the black source selector knobs and the volume knob go very well with the silver unit, the knobs have a machine knurled finish apart from on the very front of the button and are classy and have a nice heavyweight touch to their action

 aesthetics aside, the unit over the last few weeks has really impressed me and seems to be able to accomodate me with whatever configuration i wish to try input and output wize.

 i am actually warming to the detail and clean clinical representation of the inbuilt amp also - worlds apart from the warmer and fuller sound of my HR balanced amp


----------



## emmodad

Quote:


  Originally Posted by *Dougr33* /img/forum/go_quote.gif 
_Seriously, are you okay? You could have taken all this energy (and typing) and helped the next 20 people trying to find answers but lacking your search savvy. Wow._

 

just fine, thanks for asking.

 somewhat dismayed with increasing number of folk who default to loading Elias' inbox without doing very simple homework. the product documentation is (one would think) an obvious starting point; and although i appreciate the compliment 
	

	
	
		
		

		
			





 , a google search on "benchmark DAC1 components" doesn't seem like rocket science.

 Elias's involvement in this thread and his contribution to head-fi are not only unique and to be appreciated, but he's involved in several other boards too. Respectfully, just hoping that folk could keep same in mind, do basic research and not load him with stuff which has been asked & answered.


----------



## EliasGwinn

Quote:


  Originally Posted by *pbarach* /img/forum/go_quote.gif 
_So if I have a 24-bit audio file on my Windows XP PC, then this will be transmitted at 24 bits (and bit-perfectly) to the DAC1 via the USB input?_

 

Exactly. That is the major difference between our USB interface and most other computer audio interfaces. 

 But this is not only important for 24-bit audio files, but for 16-bit audio files also. The reason is that audio software and other internal processes may result in 24-bit audio _streams_, even when playing 16-bit audio files. A simple example is when a digital volume control is used. When a process like this occurs, the operation usually results in 24-bit data because of the math (e.g., remainders). If the audio interface is not 24-bits and/or bit-transparent, the result can be a severely distorted version of the audio.

  Quote:


  Originally Posted by *pbarach* /img/forum/go_quote.gif 
_An additional question: If I decide I don't need the USB input, and I decide instead to purchase the DAC1 model that comes without the USB input, are there any other differences in the circuit design or components of these two DAC1 models, other than not having that USB input in the less expensive model?_

 

Yes, there are several feature differences and a very important difference in the circuit. 

 The new features include: 
 1) Auto-standby
 2) An auto-mute function w/ headphone usage
 3) multiple gain rages for the headphone amp 
 4) and, of coarse, the USB input

 The new circuit change is the output drivers are built around the new National Semiconductor LM4562 opamps. These came out a few months before the DAC1 USB was developed, and we jumped at the opportunity to use them. 

 The LM4562's are exceptional drivers, capable of delivering high-currents and driving low-impedance or high-capacitance loads without distortion. They have since been implemented in high-end audiophile gear across the board, and are quickly becoming recognized as the best part for the job. 

 The DAC1 PRE uses the LM4562's all throughout the analog circuitry, whereas the DAC1 USB uses them in the output section only.

 Thanks,
 Elias


----------



## MarkyMark

Hi Elias,

 I have the DAC1 classic (March '07 build) and have read that the HPA2 was designed to drive headphone of 60 ohm or higher (I understand this has been changed to 30 or higher for the USB model). What problems would arise from attempting to drive my Denon headphones, which are rated 25 ohms?

 Also, a suggestion for improving the product would be the addition to the DAC1 to be able to indicate the bit depth and sample rate of the incoming digital feed. 

 Cheers


----------



## EliasGwinn

Quote:


  Originally Posted by *stvn758* /img/forum/go_quote.gif 
_Kikatek.com | (F8V3152AEA) Belkin Toslink Splitter ODT F/ 2 x ODT F (Blister)

 How about this, looks perfect. No messing about, no power source - obviously don't play both CD players at once.
	

	
	
		
		

		
		
	


	


_

 

I'm not sure. There isn't much information about it. I don't know how it works, so I'm a little suspicious.

 Thanks,
 Elias


----------



## EliasGwinn

Quote:


  Originally Posted by *MarkyMark* /img/forum/go_quote.gif 
_I have the DAC1 classic (March '07 build) and have read that the HPA2 was designed to drive headphone of 60 ohm or higher (I understand this has been changed to 30 or higher for the USB model). What problems would arise from attempting to drive my Denon headphones, which are rated 25 ohms?_

 

The HPA2 that is built into your DAC1 is the same as that built into the DAC1 USB (with the exception of the mutliple gain ranges, which is only available on the DAC1 USB, and more-so on the DAC1 PRE).

 The only problems that arise from a low impedance headphone is usually ultra-high sensitivity and a reduction in damping factor.

 The high sensitivity may simply make the headphones difficult to use because of the high output levels.

 The low damping factor will cause more distortion (resonance) then would occur with higher impedance headphones. Fortunately, the HPA2 has an ouput impedance of 0.01 ohms, so it is probably the best suited for low impedance headphones.

 When we quote a minimum impedance, we say that we only guarentee our performance specs to that minimum. It doesn't mean you the unit will actually be damaged by using low impedance cans (unless they're below 5-ohms!!).

 Thanks,
 Elias


----------



## luciyuspax

Which gain setting should be used for properly driving sennheiser hd600 to loud levels? I will be buying dac1 usb in a few months. Will I have to open dac1's case or will it sound perfect for hd600 right out of the box?


----------



## feoteng2003

Hello, EliasGwinn 

 I am currently using DAC-1 through XLR to my Dynaudio BM5A
 Just wondering what would be the best set up for both?

 Balanced Analog Outputs:10db or 20db?


 -4db or 0db on BM5A?


----------



## poo

Quote:


  Originally Posted by *luciyuspax* /img/forum/go_quote.gif 
_Which gain setting should be used for properly driving sennheiser hd600 to loud levels? I will be buying dac1 usb in a few months. Will I have to open dac1's case or will it sound perfect for hd600 right out of the box?_

 

That really depends on your ears... out of the box with my HD650s I rarely get past half way on the volume knob - which for me is really loud. Usually 'comfortable' listening for me is around 1/4 of the full volume I guess.


----------



## EliasGwinn

Quote:


  Originally Posted by *TreAdidas* /img/forum/go_quote.gif 
_You know it is funny you ask that because when I first saw it I thought wow that is good combo but I wished it were the inverse! A black faceplate with silver knobs would look soooo trick.



_

 

Interesting!!


----------



## EliasGwinn

Quote:


  Originally Posted by *luciyuspax* /img/forum/go_quote.gif 
_Which gain setting should be used for properly driving sennheiser hd600 to loud levels? I will be buying dac1 usb in a few months. Will I have to open dac1's case or will it sound perfect for hd600 right out of the box?_

 

Hello Luciyuspax.

 As Poo mentioned, it really depends on you listening preference. The DAC1 USB only has one alternate (10 dB lower) gain setting, and it is shipped in that setting. If I had to guess, I'd say that you'll be happy with that setting.

 If you find you need more volume, you can change the jumpers to the higher gain setting.

 If you find you want a lower gain range, you'll need the DAC1 PRE. It has a three gain ranges: original, 10 dB lower, and 20 dB lower.

 Thanks,
 Elias


----------



## EliasGwinn

Quote:


  Originally Posted by *feoteng2003* /img/forum/go_quote.gif 
_Hello, EliasGwinn 

 I am currently using DAC-1 through XLR to my Dynaudio BM5A
 Just wondering what would be the best set up for both?

 Balanced Analog Outputs:10db or 20db?


 -4db or 0db on BM5A?_

 

Hello Feoteng.

 The best gain setting for your BM 5A's is '-10'. This will maximize your signal to noise ratio.

 After you've set that, follow this article's instructions to determine the best attenuation setting in the DAC1: Benchmark Media: Feedback Newsletter

 Let me know if you have any questions about this...

 Thanks,
 Elias


----------



## rmh1

How about a Black on Black Pre, like all your other equiptment? I still think thats the best look.


----------



## poo

^ Nice avatar...


----------



## EliasGwinn

Quote:


  Originally Posted by *rmh1* /img/forum/go_quote.gif 
_How about a Black on Black Pre, like all your other equiptment? I still think thats the best look._

 

We'll probably keep the black face/black knob option available indefinately. Aside from the asthetic quality, a low-profile face is essential in some setups.

 Regarding other color schemes...

 I enjoy hearing about everyone's opinions and suggestions for color schemes! They are all taken to heart, and we will keep them in mind for the future.

 If we heard an OVERWHELMING number of responses for a specific variation, we would absolutely consider it. However, we are really excited about our new color scheme (silver faces with black knobs), and apparently a lot of end users are as well! So, we plan to ride it out for a while. 

 Thanks,
 Elias


----------



## rmh1

@ Poo - See you in Brisbane Mate! For the Bledisloe Cup!

 @ Elias - Didn't see the option of a Black FacePlate anywhere for the Pre. Is this an option?


----------



## Dougr33

Quote:


  Originally Posted by *EliasGwinn* /img/forum/go_quote.gif 
_... However, we are really excited about our new color scheme (silver faces with black knobs), and apparently a lot of end users are as well! So, we plan to ride it out for a while. 

 Thanks,
 Elias_

 

Maybe a knob-swap program for new purchases, or just sell the knobs online. Anodize in some cool colors for true custom look. There are plenty of ways to complicate your market model!!


----------



## DigiPhx

Quote:


  Originally Posted by *EliasGwinn* /img/forum/go_quote.gif 
_Hello Feoteng.

 The best gain setting for your BM 5A's is '-10'. This will maximize your signal to noise ratio.

 After you've set that, follow this article's instructions to determine the best attenuation setting in the DAC1: Benchmark Media: Feedback Newsletter

 Let me know if you have any questions about this...

 Thanks,
 Elias_

 


 Same problem for my Dynaudio Air 6 Speaker here,
 I've set the gain to '-10', the knob is near to the mid-rotation, but the DSP still detected the input level is too low, is it possible to set the jumper to '0'??

 Thanks


----------



## EliasGwinn

Quote:


  Originally Posted by *rmh1* /img/forum/go_quote.gif 
_@ Elias - Didn't see the option of a Black FacePlate anywhere for the Pre. Is this an option?_

 

No, the DAC1 PRE is only available with the Silver faceplate.

 Thanks,
 Elias


----------



## EliasGwinn

Quote:


  Originally Posted by *Dougr33* /img/forum/go_quote.gif 
_ There are plenty of ways to complicate your market model!!_


----------



## EliasGwinn

Quote:


  Originally Posted by *DigiPhx* /img/forum/go_quote.gif 
_Same problem for my Dynaudio Air 6 Speaker here,
 I've set the gain to '-10', the knob is near to the mid-rotation, but the DSP still detected the input level is too low, is it possible to set the jumper to '0'??

 Thanks_

 

DigiPhx,

 Yes, you can set your attenuators to 0 dB. This will give you maximum output level. Be careful though...start with the volume control very low.

 Thanks,
 Elias


----------



## rmh1

Quote:


  Originally Posted by *EliasGwinn* /img/forum/go_quote.gif 
_We'll probably keep the black face/black knob option available indefinately. Aside from the asthetic quality, a low-profile face is essential in some setups._

 

Was hoping this could apply to the Pre as well, Elias. Hope you guys will change your mind and offer this option like you do on your other products. May have to sell my CD-3000 for this one.


----------



## milkweg

$300.00 just for USB? Not in my lifetime. I've read some people say that this DAC sounds no better than EMU 0404 anyway and that has USB and costs less than $200.00 total.


----------



## poo

^ Sure sounds different to me... But if the 0404 sounds as good to you, then I guess you win! All that extra cash could go towards wax removal!


----------



## emmodad

Quote:


  Originally Posted by *milkweg* /img/forum/go_quote.gif 
_$300.00 just for USB? Not in my lifetime. I've read some people say that this DAC sounds no better than EMU 0404 anyway and that has USB and costs less than $200.00 total._

 

1/ (product price differential) "just for USB" is certainly incorrect. you may wish to read the product specs, or merely a few posts earlier in this thread (ie http://www.head-fi.org/forums/4682665-post1877.html)

 2/ "I've read some people say": hmmmm, have you actually listened to those two products...... in any event, having compared precisely the two (among many, many others), I would think these particular "some people" might also be ones who say that Britney Spears sounds like Natalie Merchant....

 ymmv, imho, fwiw, yada yada


----------



## illkemist

Quote:


  Originally Posted by *milkweg* /img/forum/go_quote.gif 
_$300.00 just for USB? Not in my lifetime. I've read some people say that this DAC sounds no better than EMU 0404 anyway and that has USB and costs less than $200.00 total._

 

I've _HEARD_ people say that eating PopRocks and drinking Coke at the same time will make your stomach explode.


----------



## emmodad

Quote:


  Originally Posted by *illkemist* /img/forum/go_quote.gif 
_I've HEARD people say that eating PopRocks and drinking Coke at the same time will make your stomach explode._

 

heh. check out "the Death of Little Mikey" on snopes:

snopes.com: Pop Rocks Death


----------



## milkweg

Quote:


  Originally Posted by *poo* /img/forum/go_quote.gif 
_^ Sure sounds different to me... But if the 0404 sounds as good to you, then I guess you win! All that extra cash could go towards wax removal!_

 

"Sounds different" doesn't necessarily mean better. There's a post buried from someone in these forums that did an extensive search for a good DAC to mate with his HD650s and in the end he preferred the EMU0404. And he compared to DACs costing twice as much as the Benchmark DAC1.

 Yea, it costs more and has a brick inside it so it must be better doesn't fool me. Enjoy your expensive placebo though.


----------



## EliasGwinn

Quote:


  Originally Posted by *milkweg* /img/forum/go_quote.gif 
_"Sounds different" doesn't necessarily mean better. There's a post buried from someone in these forums that did an extensive search for a good DAC to mate with his HD650s and in the end he preferred the EMU0404. And he compared to DACs costing twice as much as the Benchmark DAC1.

 Yea, it costs more and has a brick inside it so it must be better doesn't fool me. Enjoy your expensive placebo though. 
	

	
	
		
		

		
		
	


	


_

 

Hello Milkweg! 

 Welcome to the thread...we appreciate your input as a proponent of EMU.

 We actually have the 0404 here at Benchmark from when we did a survey of existing technologies. You're right, it isn't heavy at all! Maybe 100g tops... That must keep shipping costs low, especially coming from...where is that made?

 Can you give us your reasons for preferring the EMU? Is it solely based on the fact that it is cheap?

 Thanks,
 Elias


----------



## nd4speed

Quote:


  Originally Posted by *EliasGwinn* /img/forum/go_quote.gif 
_Hello Milkweg! 
 Is it solely based on the fact that it is cheap?_

 

LOL

 I think Milkweg found the one guy on this entire forum that prefers an EMU to a Benchmark and decided to believe him over the countless that hear otherwise.

 Personally, I'd love to see "giant-killers" of this scale (I think there's a little bit of that in all of us), but I'm a little more pragmatic.

 There's only one way to be sure Milkweg; rather than make broad, sweeping generalizations, buy both and return the one you like less. Benchmark offers a very liberal 30 day return policy


----------



## Audio-Omega

Where is the best place to buy a DAC1 in Australia ? I don't think I can buy one directly from the manufacturer.


----------



## joijwall

I use DAC1 USB with my MacBook and have three questions:
 1) I experience differences soundwise between iTunes 8.0 and VLC 0.8.6i but want to make sure I'm really correct. How should I proceed to make a good comparison in a pure listening test?
 2) Has anyone on this forum gone beyond placebo ears and done some highly technical measures on these two apps bit performance lately?
 3) I've read in he wiki that iTunes and QuickTime is working together. Is QuickTime also involved when using VLC? How do I make sure QuickTime has no sound enhancers or equalizers on?
 / Joachim


----------



## poo

Quote:


  Originally Posted by *milkweg* /img/forum/go_quote.gif 
_Enjoy your expensive placebo though. 
	

	
	
		
		

		
			



_

 

I've got two 'expensive placebos' and both sound great.

 If the 404 sounded better I would have two of those instead, but then I've actually owned and spent significant time with DAC1 and 404 before talking about my preferences, whereas you are regurgitating dribble about something you haven't heard now aren't you?


----------



## Gatsu

Something that may be useful to people who run off OSX. I hacked up an Applescript to enter all the desired settings into Audio MIDI Setup.

  Code:


```
[left]tell application "Audio MIDI Setup" to launch tell application "System Events" tell application process "Audio MIDI Setup" tell window "Audio MIDI Setup" tell tab group 1 tell radio button "Audio Devices" delay 0.3 click end tell tell pop up button 8 -- "Properties for" click click menu item "Benchmark 1.0" of menu 1 end tell delay 0.3 tell pop up button 5 -- "Default Output" click click menu item "Benchmark 1.0" of menu 1 end tell delay 0.3 tell pop up button 6 -- "System Output" click click menu item "Benchmark 1.0" of menu 1 end tell delay 0.3 tell pop up button 1 -- "Format" click click menu item "2ch-24bit" of menu 1 end tell delay 0.3 tell combo box 1 tell button 1 tell application "Audio MIDI Setup" to activate click keystroke (ASCII character 31) keystroke (ASCII character 31) keystroke (ASCII character 31) keystroke return delay 0.3 tell application "Audio MIDI Setup" to quit end tell end tell end tell end tell end tell end tell[/left]
```

Basically it just opens Audio MIDI Setup and clicks the appropriate menus. Not much to it really and it can easily be changed to apply the settings for some other device. But its nice to not have to do it manually every time I change output devices.


----------



## EliasGwinn

That is very cool!! 
	

	
	
		
		

		
		
	


	









  Quote:


  Originally Posted by *Gatsu* /img/forum/go_quote.gif 
_Something that may be useful to people who run off OSX. I hacked up an Applescript to enter all the desired settings into Audio MIDI Setup.

  Code:



		Code:
	

[left]tell application "Audio MIDI Setup" to launch tell application "System Events" tell application process "Audio MIDI Setup" tell window "Audio MIDI Setup" tell tab group 1 tell radio button "Audio Devices" delay 0.3 click end tell tell pop up button 8 -- "Properties for" click click menu item "Benchmark 1.0" of menu 1 end tell delay 0.3 tell pop up button 5 -- "Default Output" click click menu item "Benchmark 1.0" of menu 1 end tell delay 0.3 tell pop up button 6 -- "System Output" click click menu item "Benchmark 1.0" of menu 1 end tell delay 0.3 tell pop up button 1 -- "Format" click click menu item "2ch-24bit" of menu 1 end tell delay 0.3 tell combo box 1 tell button 1 tell application "Audio MIDI Setup" to activate click keystroke (ASCII character 31) keystroke (ASCII character 31) keystroke (ASCII character 31) keystroke return delay 0.3 tell application "Audio MIDI Setup" to quit end tell end tell end tell end tell end tell end tell[/left]


Basically it just opens Audio MIDI Setup and clicks the appropriate menus. Not much to it really and it can easily be changed to apply the settings for some other device. But its nice to not have to do it manually every time I change output devices._


----------



## G-U-E-S-T

Quote:


  Originally Posted by *EliasGwinn* /img/forum/go_quote.gif 
_...the output drivers are built around the new National Semiconductor LM4562 opamps.... The LM4562's are exceptional drivers, capable of delivering high-currents and driving low-impedance or high-capacitance loads without distortion.... The DAC1 PRE uses the LM4562's all throughout the analog circuitry..._

 

Hi Elias,

 Just wondering if Benchmark had considered using discrete opamps rather than chip opamps (just thinking of sound quality here, not space considerations). For example, the newest DACs from Bryston and PS-Audio etc. are heavily marketing their choice of using 100% class A discrete output stages, and users seem to be raving over the significant difference this seems to bring to the resulting sonics. What are your thoughts on this please? Thanks in advance...


----------



## oxophone

Has anyone got their Benchmark DAC1 Pre modified by Asi-tek?
Benchmark DAC1 & DAC1 USB Premium Modifications!!!

 If yes, would it be possible to provide some feedback on the 'improvement' in the sound quality? 

 or, would someone with technical knowledge - may be Elias Gwinn - can enlighten me if Asi-tek's claimed modification would improve the sound quality?


----------



## Quaddy

Quote:


  Originally Posted by *oxophone* /img/forum/go_quote.gif 
_may be Elias Gwinn - can enlighten me if Asi-tek's claimed modification would improve the sound quality?_

 


 isn't this akin to asking apple how the iMod sounds?

 not really the done thing.


----------



## oxophone

"isn't this akin to asking apple how the iMod sounds?"

 what's wrong in asking about the benefit of Benchamark mods?


----------



## zenjazz

1. how good is DAC1 pre's headphone amp output? is it comparable to any stand alone high audiophile grade headphone amplifier?

 2. Is Grado GS1000 a good match for Benchmark DAC1 pre?

 Users, please comment.


----------



## HeadLover

Quote:


  Originally Posted by *oxophone* /img/forum/go_quote.gif 
_Has anyone got their Benchmark DAC1 Pre modified by Asi-tek?
Benchmark DAC1 & DAC1 USB Premium Modifications!!!

 If yes, would it be possible to provide some feedback on the 'improvement' in the sound quality? 

 or, would someone with technical knowledge - may be Elias Gwinn - can enlighten me if Asi-tek's claimed modification would improve the sound quality?_

 

I also wonder the same thing!
 And
 Why Benchmark won't put a more "better" version with this improvements?


----------



## EliasGwinn

Quote:


  Originally Posted by *G-U-E-S-T* /img/forum/go_quote.gif 
_Hi Elias,

 Just wondering if Benchmark had considered using discrete opamps rather than chip opamps (just thinking of sound quality here, not space considerations). For example, the newest DACs from Bryston and PS-Audio etc. are heavily marketing their choice of using 100% class A discrete output stages, and users seem to be raving over the significant difference this seems to bring to the resulting sonics. What are your thoughts on this please? Thanks in advance..._

 

G-U-E-S-T, 

 We use opamps because they acheive much lower distortion then discrete transistors. Transistors have a more narrow linear region, and suffer from significant distortion. 

 We use discrete transistors in our microphone pre-amplifiers because, in conjunction with servo opamps, they build a system that can acheive much higher gain bandwidth. In a mic-pre, you need up to 70 dB of gain. In a DAC, you don't need that much gain, so you don't have to comprimise the clean performance of low-gain opamps.

 Thanks,
 Elias


----------



## EliasGwinn

Quote:


  Originally Posted by *oxophone* /img/forum/go_quote.gif 
_Has anyone got their Benchmark DAC1 Pre modified by Asi-tek?
Benchmark DAC1 & DAC1 USB Premium Modifications!!!

 If yes, would it be possible to provide some feedback on the 'improvement' in the sound quality? 

 or, would someone with technical knowledge - may be Elias Gwinn - can enlighten me if Asi-tek's claimed modification would improve the sound quality?_

 

We strongly recommend that you do NOT have your DAC1 modified. Although you are certainly welcome to do whatever you please with your DAC1, there are many reasons to not have it modified. For example:

 1. WARRANTY WILL BE VOIDED

 If the DAC1 is modified, the warranty will be voided. As many mod's cause irratic behavior, you may likely face hefty repair charges.

 2. THE MOD'S DEGRADE PERFORMANCE

 ...or, at best, do nothing at all. We've seen almost every mod out there, and most severely degrade the performance. Some changed parts that had no affect on the audio at all!

 3. THE MODDERS DON'T KNOW THE CIRCUIT

 The modder's do not have a schematic, and don't know the circuit. They are replacing parts based on the assumption that their parts are compatible with the circuit, when they are often not!

 So, as I said, you are free to do whatever you want to the DAC1. But, for the sake of the sonic performance (and your wallet), you would be well-advised to avoid these modifications.

 Good luck!

 Thanks,
 Elias


----------



## HeadLover

Why won't Benchmark do that mods on their own ?
 I mean, offering a 500$ + stock price (or more) and putting better things for those who want it ?


----------



## EliasGwinn

Quote:


  Originally Posted by *HeadLover* /img/forum/go_quote.gif 
_And Why Benchmark won't put a more "better" version with this improvements?

 ....

 Why won't Benchmark do that mods on their own ?
 I mean, offering a 500$ + stock price (or more) and putting better things for those who want it ?_

 

This is a great question that, inadvertantly, explains why folks should not have their DAC1 modified...

 The DAC1 circuit is specifically designed and built with optimal components for optimal performance. No components were comprimised for cost reasons. They were all selected because they created the best possible performance. 

 If someone's mod actually improved the performance of the DAC1 by changing some components, we would certainly switch and use those components. The price difference among components are so small, that it would have no affect on the cost (which should make you ask why the modder's charge so much...???).

 However, the modified components do not improve the performance, and actually degrade the performance most times. 

 The DAC1 cannot be improved by changing parts. The components were chosen when the circuit was optimized during the design phase. Any other parts will comprimise that optimization.

 Thanks,
 Elias


----------



## HeadLover

Ok I guess you are right (didn't test it for weeks and months to know better)
 But
 why not offering even something with better stuff?
 I am sure that things like maybe a better stronger PSU or better inputs, or opmaps (maybe going discarte) or what ever, can improve it
 Not?

 Just my thoughts, I mean we all looking for getting better every time, not ?


----------



## EliasGwinn

Quote:


  Originally Posted by *HeadLover* /img/forum/go_quote.gif 
_But
 why not offering even something with better stuff?
 I am sure that things like maybe a better stronger PSU or better inputs, or opmaps (maybe going discarte)
 Just my thoughts, I mean we all looking for getting better every time, not ?




_

 

HeadLover,

 We are always looking for the same thing too!! We are always looking for ways to improve our product. Our latest products always contain the best components and design techniques available. The DAC1 PRE has the best of everything, and, cannot be improved upon by upgrading any parts.

 When the 4562 opamps came out in late 2006, we built a new circuit optimized for those OPAMPS because they provided better performance then the 5532 which the DAC1 uses. That new circuit design became the DAC1 USB. 

 Now, many mod-ers are putting the 4562 into the original DAC1 which uses the 5532's. The problem is that the circuit was not built for that - that circuit was optimized for the 5532. The 5532 gain-stages in the DAC1 circuit are not compatible with the 4562. The circuit had to be redesigned to use the 4562 - hence the new product, the DAC1 USB. 

 Discrete transistor stages are not better then OPAMP stages. "Discrete=better" is another example of marketing mis-leading the consumers at the expense of sonic performance. 

 We have poured our sweat and blood into our product designs to make them as good as possible. If there was anything we could have done differently to improve performance, we would have done it. Just compare our performance to anything else out there and you'll hear why we say that it can't get much better.

 Thanks,
 Elias


----------



## HeadLover

amm
 So I need to put something like 1500$ to test it 
	

	
	
		
		

		
		
	


	




 I just think (just like you), that a user want to see improvments all the time!
 Like bigger PSU, or even extrnal one or what ever, I mean, I would want a 2009 (soon to come) product to be better than a 2005 one


----------



## EliasGwinn

Another word on discrete transistors...

 They are extremely susceptible to degredation from imperfect power sources. Any fluctuation from perfect DC rails causes a lose in performance. This is indicated by the specification: 'Power Supply Rejection Ratio'. Most products with discrete components won't list this spec...guess why?

 Our op-amp stages can maintain full performance even when being driven with noisy power. Our power supply is extremely capable of preventing that scenario, but its good to know that you don't need a $500 power cord just to get your gain-stages to sound good!! 
	

	
	
		
		

		
		
	


	




 Thanks,
 Elias


----------



## oxophone

Here's a reply I got from Elias and would like to share with others:


 1. How good is DAC1 Pre's headphone output compared to a good quality pro-audiophile grade stand alone headphone amp?

 The HPA2 (the headphone amplifier built into the DAC1 PRE) is one of the most reverred headphone amp on the market. Major recording, mastering, and broadcast studios use our headphone amplifier for optimal results.

 The HPA2 is incredibly accurate thanks to its 0.01 ohm output impedance. Unlike most of its peers, it is extremely resilient to variations in loads, even being able to drive two low-impedance headphones simultaneously without a loss in quality.


----------



## oxophone

Is there a way to use an Ipod [6th gen or may be touch] as a source for DAC1 pre?

 or, do I need something like Wadia 170i transport?


----------



## Quaddy

Quote:


  Originally Posted by *oxophone* /img/forum/go_quote.gif 
_Is there a way to use an Ipod [6th gen or may be touch] as a source for DAC1 pre?

 or, do I need something like Wadia 170i transport?_

 

yes i sometimes run my ipod into it, the pre has RCA in on the back, so you could run a LOD (3.5mm)(captive or rocket dock style with own IC) to RCA in to the DAC - but for optimal results, ultimately a wadia or alike would be preferable for digital to digital


----------



## EliasGwinn

Quote:


  Originally Posted by *oxophone* /img/forum/go_quote.gif 
_Is there a way to use an Ipod [6th gen or may be touch] as a source for DAC1 pre?

 or, do I need something like Wadia 170i transport?_

 

Oxophone,

 The iPod can be used as a source to the DAC1 PRE by feeding its analog signal to the analog input of the DAC1 PRE.

 The iPod will only transmit digitally to the DAC1 PRE with a device like the Wadia 170i. Apple has made the iPod very secure so that digital transmission can only be accessed proprietarily. It is somewhat frustrating, but the Wadia makes it possible.

 Thanks,
 Elias


----------



## oxophone

Does Red Wine modified Ipod [IMod] make things better in this regard? 
 [I am not sure whether Imod sends out digital or analogue sound]


----------



## oxophone

Quote:


  Originally Posted by *Quaddy* /img/forum/go_quote.gif 
_ultimately a wadia or alike would be preferable_

 

Is there any 'alike' product?


----------



## Quaddy

@oxophone

 i have pm'ed you, as this is off topic to the thread


----------



## G-U-E-S-T

Hi Elias,

 Thank you so much for your extremely thorough answers and help. One more question has occurred to me, regarding the DAC1 PRE. Does it have much headroom beyond 0dbfs? Since some post-mastering DSP's can actually process a digital signal beyond 0dbfs (especially with the recording industry's "loudness wars" these days), I'm wondering if the DAC1 PRE is designed to have any headroom past 0dbfs, so as to avoid audible digital clipping artifacts in such circumstances. Please advise? Thanks again in advance.


----------



## hpatel

Hello Elias,

 I just bought a Dac 1 Pre from a fellow head-fier. I've had it for two days or so and am enjoying it immensely.

 I have discovered that my lifelong preference for playing music with eq applied to make it sound better has completely vanished! 

 I read this thread over a week or so and recall you saying that it is normal for the case to be slightly warm, when in operation. I'm envisioning placing it under my LCD monitor which has a circular base that just perfectly fits the top of the Dac Pre, with a mouse pad in between, to prevent scratches.

 Should I be concerned about the unit becoming too hot if I do this? In other words does the top of the unit function as a heat sink that shouldn't be interfered with?

 Thanks. 

 Harry.


----------



## EliasGwinn

Quote:


  Originally Posted by *G-U-E-S-T* /img/forum/go_quote.gif 
_Hi Elias,

 Thank you so much for your extremely thorough answers and help. One more question has occurred to me, regarding the DAC1 PRE. Does it have much headroom beyond 0dbfs? Since some post-mastering DSP's can actually process a digital signal beyond 0dbfs (especially with the recording industry's "loudness wars" these days), I'm wondering if the DAC1 PRE is designed to have any headroom past 0dbfs, so as to avoid audible digital clipping artifacts in such circumstances. Please advise? Thanks again in advance._

 

G-U-E-S-T,

 I think you're referring to "inter-sample overs", which are signals that went above the maximum threshold of the A-to-D converter between consecutive samples. 

 Headroom is usually not the limiting factor in these cases. The DAC1 PRE has plenty of available analog headroom. However, the problem is that digital circuitry does not go above 0 dBFS. Therefore, almost any DSP that is applied to a signal with inter-sample overs will render the overloads as digital distortion. This means any d-to-a converters with oversampling filters (such as the ones used in the DAC1 and most other high-quality DAC's) will clip when inter-sample overs occur. 

 Thanks,
 Elias


----------



## EliasGwinn

Quote:


  Originally Posted by *hpatel* /img/forum/go_quote.gif 
_Hello Elias,

 I just bought a Dac 1 Pre from a fellow head-fier. I've had it for two days or so and am enjoying it immensely.

 I have discovered that my lifelong preference for playing music with eq applied to make it sound better has completely vanished! 

 I read this thread over a week or so and recall you saying that it is normal for the case to be slightly warm, when in operation. I'm envisioning placing it under my LCD monitor which has a circular base that just perfectly fits the top of the Dac Pre, with a mouse pad in between, to prevent scratches.

 Should I be concerned about the unit becoming too hot if I do this? In other words does the top of the unit function as a heat sink that shouldn't be interfered with?

 Thanks. 

 Harry._

 

Harry,

 The DAC1 may become too hot if you cover it with a mouse pad. The top of the chassis is a large part of the heat-dispersion of the unit, so covering it with an insulating material is not in the best interest of the DAC1.

 However, you should be fine placing the LCD directly on the top of the chassis. Most LCD bases have little rubber thingies for that exact reason. Plus, as long as you're not twisting and turning it often, it shouldn't hurt it to just sit on top even without the rubber pads.

 Thanks,
 Elias


----------



## G-U-E-S-T

Quote:


  Originally Posted by *EliasGwinn* /img/forum/go_quote.gif 
_G-U-E-S-T,

 I think you're referring to "inter-sample overs", which are signals that went above the maximum threshold of the A-to-D converter between consecutive samples. 

 Headroom is usually not the limiting factor in these cases. The DAC1 PRE has plenty of available analog headroom. However, the problem is that digital circuitry does not go above 0 dBFS. Therefore, almost any DSP that is applied to a signal with inter-sample overs will render the overloads as digital distortion. This means any d-to-a converters with oversampling filters (such as the ones used in the DAC1 and most other high-quality DAC's) will clip when inter-sample overs occur. 

 Thanks,
 Elias_

 



 Thanks Elias, so how does the DAC1 PRE handle a clipped digital signal when it encounters one? Some DAC's (like Lynx I believe) are designed to make a "best guess" rounded interpolation when encountering the clipped section of a digital waveform (and also give extra headroom for it) - which should result in nicer sonics in such situations rather than just allowing the sharp clipped section of the waveform to process through the D-A directly. Does the DAC1 PRE have any provisions like this?


----------



## HeadLover

BTW
 Why don't the DAC-1 have some kind of inside buffer, so it can fix a not so good synch data from a PC ?


----------



## G-U-E-S-T

While I am awaiting Elias' reply to my above question, I have also ordered and received a Benchmark DAC1 PRE. I am still auditioning it in my system, and if anybody is interested (including Elias), I will be glad to share my impressions here in this thread. Shall I do so? Please post here to let me know.


----------



## poo

^ go nuts...


----------



## HeadLover

Quote:


  Originally Posted by *G-U-E-S-T* /img/forum/go_quote.gif 
_While I am awaiting Elias' reply to my above question, I have also ordered and received a Benchmark DAC1 PRE. I am still auditioning it in my system, and if anybody is interested (including Elias), I will be glad to share my impressions here in this thread. Shall I do so? Please post here to let me know._

 

Yep!
 Post some nice review


----------



## EliasGwinn

Quote:


  Originally Posted by *G-U-E-S-T* /img/forum/go_quote.gif 
_Thanks Elias, so how does the DAC1 PRE handle a clipped digital signal when it encounters one? Some DAC's (like Lynx I believe) are designed to make a "best guess" rounded interpolation when encountering the clipped section of a digital waveform (and also give extra headroom for it) - which should result in nicer sonics in such situations rather than just allowing the sharp clipped section of the waveform to process through the D-A directly. Does the DAC1 PRE have any provisions like this?_

 

G-U-E-S-T, et. al.,

 I'm sorry I've been absent - I've been in San Francisco for the past week attending the AES conference. 

 As for inter-sample overs, they will clip any over-sampling filter, including ours. The way an over-sample filter works is, there are 256 samples interpolated between each 'real' sample. So, if there are overs between the 
 'real' samples, the interpolated samples will max out at full signal (all 1's) for the amount of time the over lasts. This will result in a clipped sound. 

 Thanks,
 Elias


----------



## EliasGwinn

Quote:


  Originally Posted by *HeadLover* /img/forum/go_quote.gif 
_BTW
 Why don't the DAC-1 have some kind of inside buffer, so it can fix a not so good synch data from a PC ?_

 

HeadLover,

 The DAC1 has a buffer on all inputs, and an additional one on the USB input. However, it isn't the quality of sync that causes errors. It is the PC allowing other processes to interrupt, or trump, the USB stream. This is because USB was never built to be 'streaming'. It was meant to be a 'data burst' transmission system. In other words, it would 'burst' packets of data to and from peripheral devices. The USB protocol was never intended for conitnously streaming anything, like video or audio. Buffers can take care of it in almost all cases, but if a computer simply overrides USB activity for an extended period, there isn't much that can be done (except determine what is the conflicting process that is overriding USB, and disabling it).

 But it is only the rare computer that has any difficulties at all with streaming even high-res audio to the DAC1 USB/PRE.

 Thanks,
 Elias


----------



## HeadLover

Quote:


  Originally Posted by *EliasGwinn* /img/forum/go_quote.gif 
_HeadLover,

 The DAC1 has a buffer on all inputs, and an additional one on the USB input. However, it isn't the quality of sync that causes errors. It is the PC allowing other processes to interrupt, or trump, the USB stream. This is because USB was never built to be 'streaming'. It was meant to be a 'data burst' transmission system. In other words, it would 'burst' packets of data to and from peripheral devices. The USB protocol was never intended for conitnously streaming anything, like video or audio. Buffers can take care of it in almost all cases, but if a computer simply overrides USB activity for an extended period, there isn't much that can be done (except determine what is the conflicting process that is overriding USB, and disabling it).

 But it is only the rare computer that has any difficulties at all with streaming even high-res audio to the DAC1 USB/PRE.

 Thanks,
 Elias_

 

So I don't understand, I know of USB and USB 2 (and even 3 to be soon)
 They all meant to send a lot of data and do it fast
 If it isn't that good for audio
 How can you make sure in the Benchmark DAC1 that it will be more than just "OK"
 ?
 I am sure anyone buying it for using it with a PC, won't nothing less than really good SQ
 Not?


----------



## EliasGwinn

Quote:


  Originally Posted by *HeadLover* /img/forum/go_quote.gif 
_So I don't understand, I know of USB and USB 2 (and even 3 to be soon)
 They all meant to send a lot of data and do it fast
 If it isn't that good for audio
 How can you make sure in the Benchmark DAC1 that it will be more than just "OK"
 ?
 I am sure anyone buying it for using it with a PC, won't nothing less than really good SQ
 Not?_

 

HeadLover,

 This does not affect sound quality in general, it only affects drop-outs. In other words, in 98% of the cases where a PC is streaming audio to the DAC1 via USB, the audio will sound equally good as when a high-end transport is feeding the DAC1 with coax or XLR digital. For the other 2%, the computer will have some process that interrupts the USB stream that causes a drop-out where there will be no audio for +/- 0.5 sec at a time. In these cases, it is necessary to determine which process is interrupting (usually an anti-virus program or a power-management program or another USB device). Once these processes are found, they can be disabled to acheive top-level quality USB playback.

 Thanks,
 Elias


----------



## dd051

Elias, are there any plans to do usb to i2s with a next gen dac?


----------



## warnsey

G'day guys,

 Well my Benchmark DAC1 pre should arrive here on Monday (I believe I am one of the lucky ones in Aus to actually get one in). Anyway, I have a quick question with connections.

 I am wondering what the best way to hook it up to my set-up will be.

 Option 1: XLR out from Benchmark to XLR in of my mono-blocks and RCA out of Benchmark to RCA in of Subwoofer.

 Option 2: RCA out of Benchmark to Subwoofer RCA in> RCA out of Subwoofer to RCA in of mono-blocks.

 Anyone have any thoughts on what will give me the best audio quality?

 Cheers


----------



## G-U-E-S-T

Quote:


 ...Anyone have any thoughts on what will give me the best audio quality? 
 

Do you need the crossover in your subwoofer? If not, option one would be my recommendation...


----------



## infinitesymphony

Quote:


  Originally Posted by *dd051* /img/forum/go_quote.gif 
_Elias, are there any plans to do usb to i2s with a next gen dac?_

 

What advantage would that have over the Benchmark DAC1 USB when the signal begins as USB no matter what? If you wanted I2S for the input of some other DAC, that would be understandable, but a USB to I2S converter would make more sense for that application. Just curious.


----------



## EliasGwinn

Quote:


  Originally Posted by *dd051* /img/forum/go_quote.gif 
_Elias, are there any plans to do usb to i2s with a next gen dac?_

 

Actually, the USB receiver in the DAC1 USB/PRE converts the incoming USB signal to I2S before passing to the converter section. So, it's already here!

 Thanks,
 Elias


----------



## EliasGwinn

Quote:


  Originally Posted by *warnsey* /img/forum/go_quote.gif 
_G'day guys,

 Well my Benchmark DAC1 pre should arrive here on Monday (I believe I am one of the lucky ones in Aus to actually get one in). Anyway, I have a quick question with connections.

 I am wondering what the best way to hook it up to my set-up will be.

 Option 1: XLR out from Benchmark to XLR in of my mono-blocks and RCA out of Benchmark to RCA in of Subwoofer.

 Option 2: RCA out of Benchmark to Subwoofer RCA in> RCA out of Subwoofer to RCA in of mono-blocks.

 Anyone have any thoughts on what will give me the best audio quality?

 Cheers_

 

Hello Warnsey,

 Congrats on your new DAC1 PRE! I'm excited to hear about your experience with it...

 With regards to your wiring configuration, I agree with what G-U-E-S-T says. You can operate in both of these configurations, but if the sub has a crossover, then the two options presents a tradeoff. 'Option 1' gives you a direct, balanced connection to your main amplifiers, which will result in lower THD+N for the mains. 'Option 2' seperates the frequency bands which will result in lower inter-modulation distortion for your mains. I would suggest listening to both configurations to determine which sounds better to you.

 If the sub does not have a crossover (high-pass) output, then 'Option 1' is by far the better option.

 Thanks,
 Elias


----------



## ted betley

Elias does the Benchmark when receiving a signal on bnc also convert to I2S? I'm using an Audio Alchemy cable which is by definition I2S. Does it still convert whatever comes over a bnc input or does it pass through?


----------



## EliasGwinn

Quote:


  Originally Posted by *ted betley* /img/forum/go_quote.gif 
_Elias does the Benchmark when receiving a signal on bnc also convert to I2S? I'm using an Audio Alchemy cable which is by definition I2S. Does it still convert whatever comes over a bnc input or does it pass through?_

 

Ted,

 The cable itself is not I2S, although it may have been for the purpose of I2S transmission. I2S is a data transmission protocol. It differs from normal AES/EBU and SPDIF by transmitting the audio data and clock/control signals on independent streams. 

 Your cable may have been designed/manufactured to optimize that type of transmission, but if you are using it to connect a SPDIF output to a SPDIF input, then it is not using I2S. In other words, the data transmission protocol is defined by the source, not the cable.

 Thanks,
 Elias


----------



## ted betley

Thank you. Does Benchmark recommend any type of bnc cable to use with it's product?


----------



## EliasGwinn

Quote:


  Originally Posted by *ted betley* /img/forum/go_quote.gif 
_Thank you. Does Benchmark recommend any type of bnc cable to use with it's product?_

 

Any 75-ohm cable will work fine. You're Audio Alchemy cable is probably fine. 

 With UltraLock, the DAC1 will work at top performance regardless of cable. The only exception to that is if you don't use a cable with the proper characteristic impedance (75-ohm for coax, 110-ohm for XLR). Even in those cases, you risk, at worst, signal drop out. In other words, if the connection is improper, you will get nothing at all. If the connection is working, its working at top quality, due to the UltraLock.

 An example: we compared the performance of the DAC1 using 1000' of CAT5 vs. a 3' high-quality coaxial. The performance between the two were identical!! It would remain identical until we increased the length far enough to cause the signal to out-right drop out.

 Belden 1694A is great 75-ohm cable. Many outlets sell this cable, including us.

 Thanks,
 Elias


----------



## warnsey

Quote:


  Originally Posted by *EliasGwinn* /img/forum/go_quote.gif 
_Hello Warnsey,

 Congrats on your new DAC1 PRE! I'm excited to hear about your experience with it...

 With regards to your wiring configuration, I agree with what G-U-E-S-T says. You can operate in both of these configurations, but if the sub has a crossover, then the two options presents a tradeoff. 'Option 1' gives you a direct, balanced connection to your main amplifiers, which will result in lower THD+N for the mains. 'Option 2' seperates the frequency bands which will result in lower inter-modulation distortion for your mains. I would suggest listening to both configurations to determine which sounds better to you.

 If the sub does not have a crossover (high-pass) output, then 'Option 1' is by far the better option.

 Thanks,
 Elias_

 

Thanks for the reply 

 cheers


----------



## ted betley

thanks Elias


----------



## Audioscore

Hi Elias,

 I’m planning on buying the Dac1 Pre or Dac1 USB, I only need one digital input on the Dac1, but I would buy the Dac 1 Pre, if it is slightly better sounding than the Dac1 USB.

 I’ve just been reading the Tone Audio Review of Dac1 Pre, which says “Doing a head to head comparison with the DAC1 USB that I own, I could swear that the new Pre version was a bit more open, neutral and dimensional than the older model, but you know how these things can fool you”.

 I’d be very grateful if you could you please confirm that the Dac1 Pre is a bit more open, neutral and dimensional sounding than the Dac1 USB.

 Regards,

 Owen.


----------



## EliasGwinn

Quote:


  Originally Posted by *Audioscore* /img/forum/go_quote.gif 
_Hi Elias,

 I’m planning on buying the Dac1 Pre or Dac1 USB, I only need one digital input on the Dac1, but I would buy the Dac 1 Pre, if it is slightly better sounding than the Dac1 USB.

 I’ve just been reading the Tone Audio Review of Dac1 Pre, which says “Doing a head to head comparison with the DAC1 USB that I own, I could swear that the new Pre version was a bit more open, neutral and dimensional than the older model, but you know how these things can fool you”.

 I’d be very grateful if you could you please confirm that the Dac1 Pre is a bit more open, neutral and dimensional sounding than the Dac1 USB.

 Regards,

 Owen._

 

Hello Owen...

 The difference between the DAC1 PRE circuit and DAC1 USB circuit is that the DAC1 PRE uses the new LM4562 opamps throughout the analog section, whereas the DAC1 USB uses them solely as the output amplifiers. The rest of the DAC1 USB analog circuitry uses 5532's. 

 Based on our testing, these circuits should have no sonic differences. However, we have had a number of reviewers and users mention that the DAC1 PRE sounds slightly better then the DAC1 USB. I, personally, find them to be very, very close sounding. 

 Does anyone on the board here have any comments no this?

 Thanks,
 Elias


----------



## Audioscore

Thanks Elias,

 Greatly appreciated.

 Regards,

 Owen.


  Quote:


  Originally Posted by *EliasGwinn* /img/forum/go_quote.gif 
_Hello Owen...

 The difference between the DAC1 PRE circuit and DAC1 USB circuit is that the DAC1 PRE uses the new LM4562 opamps throughout the analog section, whereas the DAC1 USB uses them solely as the output amplifiers. The rest of the DAC1 USB analog circuitry uses 5532's. 

 Based on our testing, these circuits should have no sonic differences. However, we have had a number of reviewers and users mention that the DAC1 PRE sounds slightly better then the DAC1 USB. I, personally, find them to be very, very close sounding. 

 Does anyone on the board here have any comments no this?

 Thanks,
 Elias_


----------



## poo

Thank you EveSucher... fantastic news!


----------



## G-U-E-S-T

Well so far our new Benchmark DAC1 PRE sounds *fantastic* in our system - much MUCH better than any of my previous DACs. 

 I interface the DAC1 PRE variable output to my Musical Fidelity A5cr amplifier, which has only unbalanced inputs. When I connected directly with good RCA cables, the sound was great, but my wife and I thought it sounded noticeably damped in the high frequencies, like the "air" around the sound was missing and less lifelike and enjoyable. I own a Jensen PC-2XR "input transformer" (mentioned in a previous post) which I then put between the DAC1 PRE's balanced outputs (internally jumpered to -30db) and my amplifier's unbalanced inputs. This nicely improved all aspects of the reproduced sound frequencies, from the very deepest bass on up - and also fully restored the air and lifelike aspect to the music, and then some! There is an exceptional and sweet high frequency reproduction now present - very significantly better than I've ever heard before from my system. 

 For the record, the DAC1 PRE also generates phenomenally good bass reproduction and midrange tone as well. Just excellent, I can't imagine a better sounding DAC (and I've heard a great many). Our ears are finally spoiled now! We couldn't be happier with our purchase. 
	

	
	
		
		

		
			





 So to recap, in our opinion the DAC1 PRE is a genuinely outstanding product. I am selling all my other DACs and also my $2900 preamplifier, and from now on, we will just simply run everything from the DAC1 PRE's volume control (which by the way is also excellent - completely transparent as far as we can tell). In my system I did need to put the Jensen transformer between the DAC1 PRE and amplifier to make it jump from sounding great to sounding GREAT - and when I later obtain an amp with balanced inputs (I plan to be auditioning a Bryston 4B-SST amp against my Musical Fidelity A5cr amp soon), I will see what difference (if any) a direct balanced connection might make as well.

 Thanks to you Elias for all your advice and help here - it made the difference in our purchase decision, and our system sounds better now than we previously thought possible.

 Nitpicky ideas for future DAC1 versions would include (if chassis space allows) a second optical input, an AES/EBU input, and a remote control. A trigger input/output would be nice too, so just hitting the source control / power button of the DAC1 PRE could also conveniently activate an entire system. But even without these, I would most definitely buy a DAC1 PRE again.


----------



## Quaddy

agree that AES is missed greatly, i am having to do all sorts of impedance matching and coupling to take an AES feed down to coax

 i miss an optical out actually! to be able to run optically out what i am getting in

 i am using the Pre's balanced outs to a balanced in amplifier, and having just re-evaluated the sound, i can tell you i am now much happier without the extra element in the chain, preferring to come straight out of the DAC1 rather than balanced daisychaining into another amp! it seemed to deaden the sound and make it lifeless and withdrawn and that was through valhalla balanced!!!!!! it may have needed some tampering with the internal jumpers actually


----------



## parrot5

Listening to it for the first time, with the AKG K701. I really need to write my impression down, as I'm sure if I wait until I get accustomed to the sound, I'd take the quality as granted.... 

*My last DAC and Amp combo was a Headroom MicroDAC 2007 connected to Meier Corda Arietta or Original Master.* (Before then I had M-Audio soundcards/recording devices and portable amps) I remember when I first heard my last combo, I thought "oh wow there are a lot more details to be heard" and there are better separation between musical instruments.

 With the upgrade to the DAC1 USB, it's NOT just the same old "oh wow there are a lot more details to be heard" or "there are better separation between musical instruments". *Of course they're still true* (as expected with a DAC upgrade), *but in addition*, now I finally experience what's "high dynamic range" and "soundstage/positioning".

 Dynamic Range:
 The DAC1 USB strikes me first with its truly broad dynamic range. With my previous DACs and Amp combos, yeah you hear more details than the last combos, but details were the in-your-face type, almost as if it's afraid you'll miss it. With the DAC1 USB, you hear more details, but you hear the nuances too. Each note actually varies in tone and dynamics. Each note is then followed by proper decay that gradually fades into the darkness, and during the decay you don't lose the details.
 I can almost guarantee that you will hear an instrument that's in the background and say, "hey, I thought this line has fewer notes".

 Soundstage / positioning:
 I've always read the the AKG K701 has a wide soundstage, with a lot details, and that the instrument positioning is so precise. With my previous combos, I experience a bit of that, but never led me to euthiastically praise them. Now with the DAC1, I can appreciate how the mixing engineer place each instrument in their space. It's simply unbelievable how much I've been missing from the same recordings!

 An interesting note about the 701:
 With my other amps, the 701 is not that enjoying at low volume. With the DAC1, it's still enjoyable at low volume, with less loss in details. Don't know what to give the credit to, as I don't know much about electronic, but it's a nice surprise.


 Conclusion:
 In head-fi, people always say garbage in, garbage out. (Bad recordings will sound bad in good systems.) I have a different take on that now. Sometimes you think a certain recording is garbage, but that maybe bacause you haven't heard it with a good enough DAC and amp. (It may not have been "properly" mastered to sound good in all players, from 10K CD players to $50 iPods.)
 With my new DAC1 USB, I'm eager now to listen to all of my music collection again, even those that I thought are garbage.


----------



## milkweg

Quote:


  Originally Posted by *EliasGwinn* /img/forum/go_quote.gif 
_Hello Milkweg! 

 Welcome to the thread...we appreciate your input as a proponent of EMU.

 We actually have the 0404 here at Benchmark from when we did a survey of existing technologies. You're right, it isn't heavy at all! Maybe 100g tops... That must keep shipping costs low, especially coming from...where is that made?

 Can you give us your reasons for preferring the EMU? Is it solely based on the fact that it is cheap?

 Thanks,
 Elias_

 

 No, not solely, mostly because of all the resounding reviews and user comments it has received and how versatile a bit of kit it is. It is far more than just a DAC. But yes, I don't pay $1,000.00 for a DAC when a $200.00 DAC sounds 98% as good. And according to one person on these forums it sounds better than a $2,000.00 DAC. I've never even heard the Benchmark DAC1 so you will have to send me a review copy if you want me to believe it is $800.00 better sounding than the EMU 0404. Where it was built has nothing to do with sound quality either. Nice try though.


----------



## Bostonears

Quote:


  Originally Posted by *milkweg* /img/forum/go_quote.gif 
_I've never even heard the Benchmark DAC1 so you will have to send me a review copy if you want me to believe it is $800.00 better sounding than the EMU 0404. Where it was built has nothing to do with sound quality either. Nice try though._

 

As far as I can see, Elias didn't say anything about where the EMU0404 was built. Nice try pinning that on him though.


----------



## EliasGwinn

Thanks for the feedback and suggestions, G-U-E-S-T, Quaddy, Parrot5!


----------



## EliasGwinn

Quote:


  Originally Posted by *milkweg* /img/forum/go_quote.gif 
_No, not solely, mostly because of all the resounding reviews and user comments it has received and how versatile a bit of kit it is. It is far more than just a DAC. But yes, I don't pay $1,000.00 for a DAC when a $200.00 DAC sounds 98% as good. And according to one person on these forums it sounds better than a $2,000.00 DAC. I've never even heard the Benchmark DAC1 so you will have to send me a review copy if you want me to believe it is $800.00 better sounding than the EMU 0404. Where it was built has nothing to do with sound quality either. Nice try though._

 

Milkweg,

 I would certainly agree that if the 0404 makes you happy, you shouldn't feel the need to buy a more expensive one.

 Enjoy!

 Thanks,
 Elias


----------



## G-U-E-S-T

parrot5 you said it all very well above, I definitely agree with your descriptions!


----------



## G-U-E-S-T

Quote:


  Originally Posted by *EliasGwinn* /img/forum/go_quote.gif 
_Thanks for the feedback and suggestions, G-U-E-S-T,... !_

 

You're welcome Elias! Hey just one more thing that would be really nice on the DAC1 PRE, that I forgot to mention above: Having a 12V trigger (in/out) would be great. That way, I could power on my amp etc by just pushing the source-select/power button on the DAC1 PRE, which would be very cool!

 Outstanding product - enjoying it more each day.


----------



## MarkyMark

Quote:


  Originally Posted by *EliasGwinn* /img/forum/go_quote.gif 
_Thanks for the feedback and suggestions, G-U-E-S-T, Quaddy, Parrot5!_

 

Hi Elias,

 I would also suggest you consider a means of displaying bit depth and sampling frequency (I have seen this mentioned in other forums...)

 For example, I use hi-rez quite a bit on my DAC1 and I like to know whether it might be getting downsampled or not due to the copy protection flag being set.

 Thanks.
 Mark


----------



## milkweg

Quote:


  Originally Posted by *Bostonears* /img/forum/go_quote.gif 
_As far as I can see, Elias didn't say anything about where the EMU0404 was built. Nice try pinning that on him though._

 

Yes, he did. Go back and read what he implied. What part of the below quote do you not understand? Actually, I was wrong about the $800.00 difference in price. That is for an older model. Their latest bit of kit is $1300.00 difference in price. There is no way in hell it sounds $1300.00 better than the EMU0404 and I don't even need to hear it to know that as fact.

 "That must keep shipping costs low, especially coming from...where is that made?"


----------



## poo

^ Thanks again for your uninformed & inexperienced impressions. As mentioned, I'm glad you're happy with what you're using... 

 Don't feel pressured to hang around and help us all 'understand' the error of our purchasing decision. 

 Your comments regarding price are largely irrelevant. It is the same argument I could make of the EMU0404 vs the much cheaper anything else. I guess it depends on your financial situation, what you consider 'audio quality' to be, and what your ears can hear (again, not that yours have heard the difference...)


----------



## HeadLover

I am still waiting to see an even "premium" version of the DAC1 (maybe DAC2 ?!) that will have a better clock like the ultraclock or clock4 and some other improvements (maybe better PSU, cups, and so on) - every thing they can put to the MAX.
 Why not?
 So every one can find something nice for his budget


----------



## EliasGwinn

Quote:


  Originally Posted by *HeadLover* /img/forum/go_quote.gif 
_I am still waiting to see an even "premium" version of the DAC1 (maybe DAC2 ?!) that will have a better clock like the ultraclock or clock4 and some other improvements (maybe better PSU, cups, and so on) - every thing they can put to the MAX.
 Why not?
 So every one can find something nice for his budget 
	

	
	
		
		

		
		
	


	


_

 

HeadLover,

 It is a common (and financially lucrative) marketing strategy to develop tiers of quality to address multiple markets simultaneously. However, we don't follow that trend. We build the best product that we can build from the starting gate. The DAC1, especially the DAC1 USB and PRE, are composed of the best of everything we could build using the parts that are available. 

 The various sections of the DAC1 (PSU, D/A, analog circuitry) are all approaching theoretical limits of noise and distortion performance, again, considering the parts that are available on the market. Almost all of the noise and distortion artifacts of the DAC1 are inaudible under normal conditions.

 Thanks,
 Elias

 Thanks,
 Elias


----------



## HeadLover

Quote:


  Originally Posted by *EliasGwinn* /img/forum/go_quote.gif 
_HeadLover,

 It is a common (and financially lucrative) marketing strategy to develop tiers of quality to address multiple markets simultaneously. However, we don't follow that trend. We build the best product that we can build from the starting gate. The DAC1, especially the DAC1 USB and PRE, are composed of the best of everything we could build using the parts that are available. 

 The various sections of the DAC1 (PSU, D/A, analog circuitry) are all approaching theoretical limits of noise and distortion performance, again, considering the parts that are available on the market. Almost all of the noise and distortion artifacts of the DAC1 are inaudible under normal conditions.

 Thanks,
 Elias

 Thanks,
 Elias_

 

amm yes but still
 There are better (and much more expensive things) like black gates
 opmaps
 different DAC chip
 Bigger PSU
 and so on

 Why not putting them one if someone willing to pay more ?


----------



## Audioscore

Elias,

 Would I be right in thinking that the Dac1 Pre would outperform any of the latest high end AV Receivers from the top AV Receiver Manafacturers on sound quality.

 I know that I'll only get downsampled two channel PCM Audio through spdif from my Blu-Ray DVD Player.

 Regards,

 Owen.


----------



## bizkid

Hello Elias,

 I think my Benchmark DAC 1 USB isnt properly calibrated. Could you give instructions how to measure and calibrate the output? Others might find this interesting too.

 As far i could find out you'll need a multi-meter and generate a sine waveform with 50 hz to measure what comes out of the DAC.
 I've read that it's better to use 50hz so a (cheap) multi meter will be able to read the voltage, is this true?

 Thanks alot & Best Regards, Christian


----------



## EliasGwinn

Quote:


  Originally Posted by *Audioscore* /img/forum/go_quote.gif 
_Elias,

 Would I be right in thinking that the Dac1 Pre would outperform any of the latest high end AV Receivers from the top AV Receiver Manafacturers on sound quality.

 I know that I'll only get downsampled two channel PCM Audio through spdif from my Blu-Ray DVD Player.

 Regards,

 Owen._

 

Audioscore,

 I don't want to make any blanket statements regarding the performance of all AV receivers, simply because I haven't heard or tested them. However, of all D/A converters we've heard and tested, we haven't found any that out-performs the DAC1. 

 There may be one out there that does, but we haven't heard of it...

 Thanks,
 Elias


----------



## Audioscore

Thanks Elias,

 I really appreciate your help.

 Regards,

 Owen.


  Quote:


  Originally Posted by *EliasGwinn* /img/forum/go_quote.gif 
_Audioscore,

 I don't want to make any blanket statements regarding the performance of all AV receivers, simply because I haven't heard or tested them. However, of all D/A converters we've heard and tested, we haven't found any that out-performs the DAC1. 

 There may be one out there that does, but we haven't heard of it...

 Thanks,
 Elias_


----------



## EliasGwinn

Quote:


  Originally Posted by *bizkid* /img/forum/go_quote.gif 
_Hello Elias,

 I think my Benchmark DAC 1 USB isnt properly calibrated. Could you give instructions how to measure and calibrate the output? Others might find this interesting too.

 As far i could find out you'll need a multi-meter and generate a sine waveform with 50 hz to measure what comes out of the DAC.
 I've read that it's better to use 50hz so a (cheap) multi meter will be able to read the voltage, is this true?

 Thanks alot & Best Regards, Christian_

 

Christian,

 That should work fine. You will need to play a digital test tone at a constant output level. 50 Hz will optimize the readings, as cheap multi-meters are built for AC outlet testing, primarily.

 As the constant-level test-tone is playing, adjust the potentiometer for one of the two channels to a level that seems suitable for your particular system. 

 After the first channel is set, read the (AC) voltage between any two pins (I recommend 2-3) on that channel. 

 Then, while the test-tone is still playing, measure the output voltage between the same two pins on the other channel and adjust the corresponding potentiometer to match it to the first channel as best as possible.

 Thanks,
 Elias


----------



## milkweg

Quote:


  Originally Posted by *poo* /img/forum/go_quote.gif 
_^ Thanks again for your uninformed & inexperienced impressions. As mentioned, I'm glad you're happy with what you're using... 

 Don't feel pressured to hang around and help us all 'understand' the error of our purchasing decision. 
_

 

A fool and his money are soon parted. It's not the overall cost I am balking at, is it is when I read it is $300.00 just to add USB that I balked. Adding USB to a device costs no where near $300.00.


----------



## 1UP

Quote:


  Originally Posted by *EliasGwinn* /img/forum/go_quote.gif 
_The DAC1, especially the DAC1 USB and PRE, are composed of the best of everything we could build using the parts that are available. The various sections of the DAC1 (PSU, D/A, analog circuitry) are all approaching theoretical limits of noise and distortion performance, again, considering the parts that are available on the market._

 

Hi Elias, did you guys evaluate the AD1955 over the AD1853?

 AD1955 SNR 120db, AD1853 SNR 117db

 I thought the AD1955 was AD's flagship?


----------



## EliasGwinn

Quote:


  Originally Posted by *milkweg* /img/forum/go_quote.gif 
_A fool and his money are soon parted. It's not the overall cost I am balking at, is it is when I read it is $300.00 just to add USB that I balked. Adding USB to a device costs no where near $300.00._

 

I understand your reservations. Would you like me to explain the $300 price difference?

 Thanks,
 Elias


----------



## milkweg

No, it would be just a waste of your time because I don not plan to buy one. If the EMU0404 is crap then it is as they say, "ignorance is bliss", and my wallet loves me for it.


----------



## Elk

I would. I want to know if I can get the same sound out of a PC from the coax as I could from USB for an added $300. Cant really afford either DAC but may just get one anyway.


----------



## Scrith

Quote:


  Originally Posted by *HeadLover* /img/forum/go_quote.gif 
_amm yes but still
 There are better (and much more expensive things) like black gates
 opmaps
 different DAC chip
 Bigger PSU
 and so on

 Why not putting them one if someone willing to pay more ?_

 

What makes you think these are better? I'm guessing you read the silly claims of someone who would very much like to give you these items in exchange for your hard earned money. If you believe these claims, I have some very nice swampland in Florida you might be interested in.


----------



## milkweg

Quote:


  Originally Posted by *Elk* /img/forum/go_quote.gif 
_I would. I want to know if I can get the same sound out of a PC from the coax as I could from USB for an added $300. Cant really afford either DAC but may just get one anyway. 
	

	
	
		
		

		
		
	


	


_

 

On my EMU 0404 I have compared coax, optical and USB and do not hear a difference. Some peoples mb's have USB issues and that can cause USB DAC issues, sound drop outs, scratchy sound etc. I've never had any such issue but still prefer to just use coax or optical from my soundcard to EMU 0404 because it requires less cpu resources. It is less driver hassle too. If using internal soundcard and USB external DAC/soundcard I have to switch between which soundcard to use in driver properties and that gets annoying. Using coax or optical from soundcard to EMU I never have to change any settings once the soundcard is set to send digital. It sends both digtial and analog signal at the same time.


----------



## tps

Here's a question/suggestion: Would it be possible to add some kind of Test indicator to the DAC1 which indicates a bit-perfect signal path? You'd play some sort of test file at the bit-depth and sample rate you wanted to test and if the DAC1 (or DAC1 USB or DAC1 PRE) received the signal OK, it would give a "bit-perfect" indication. Maybe the test signal could be something along the lines of the HDCD signal, where it contains a psuedo-random code in the LSB. I realize this doesn't test the high bits, but my guess is that it would catch 99% of the problems which normally occur.


----------



## bizkid

Quote:


  Originally Posted by *EliasGwinn* /img/forum/go_quote.gif 
_Christian,

 That should work fine. You will need to play a digital test tone at a constant output level. 50 Hz will optimize the readings, as cheap multi-meters are built for AC outlet testing, primarily.

 As the constant-level test-tone is playing, adjust the potentiometer for one of the two channels to a level that seems suitable for your particular system. 

 After the first channel is set, read the (AC) voltage between any two pins (I recommend 2-3) on that channel. 

 Then, while the test-tone is still playing, measure the output voltage between the same two pins on the other channel and adjust the corresponding potentiometer to match it to the first channel as best as possible.

 Thanks,
 Elias_

 

Hi Elias, thanks for the instructions. I measured the XLR outputs and they are perfectly calibrated. However i found the root of my problem:

 The 2 headphone outputs on the DAC1 are slightly de-calibrated and i used these for reference.
 The right side measures 2.8V, the left 3.1V with volume raised. The difference varies throughout the volume range but stays at a minimum of 0.3V.

 I noticed that there's a small poti on the PCB which says something like L/R balance close to the headphone section. Is that actually used to calibrate it?

 Best, Christian


----------



## EliasGwinn

Quote:


  Originally Posted by *Elk* /img/forum/go_quote.gif 
_I would. I want to know if I can get the same sound out of a PC from the coax as I could from USB for an added $300. Cant really afford either DAC but may just get one anyway. 
	

	
	
		
		

		
		
	


	


_

 

The quality of the digital audio coming from your soundcard's coaxial output depends on the soundcard. More specifically, it depends on the spec's of the digital signal path and the quality of any DSP that the soundcard may perform on the audio. 

 If it is a 24-bit digital signal path with no DSP, it will perform identically to the USB input of the DAC1 USB/PRE. 

 If there is poor-quality sample-rate conversion, bit-depth truncation, or any other distructive DSP, the quality of the audio may suffer to an extent that rivals low bit-rate MP3's.

 The problem is, these destructive processes are not always user de-selectable. In other words, you may not know why or what is happening, and you may not be able to change/correct it with the soundcards software.

 Before we came out with our USB solution, we looked at a selection of popular soundcards to find a high-quality solution that our customers could pair with the original DAC1. We found that most soundcards do all sorts of unexplained DSP. Some were worse then others, but we never found a suitable interface. This is why we developed our USB solution.

 Thanks,
 Elias


----------



## HeadLover

HI

 I wanted to ask a question about sampling rate
 There is a litte of argument about it on the PICO thread

 Can a USB do 96/24 nativ?
 or what?

 And how is it implemented in the DAC1 ?

 What about 192/24 ??
 And, when getting sample rate like 24/96, is it a "true" sample ?
 Means I get it on the output? with out any down sampling or what ever?

 And is there any problem with USB and 24/96 ?

 Also, does the DAC1 use the USB 5v power? because if so, it can cause some problems, like on my PC or portable pc (laptop) where jumps on the 5V and so on, can cause downgrade to SQ

 I hope you can give me (and the other guys here) some goos explanation about it


----------



## EliasGwinn

Quote:


  Originally Posted by *Elk* /img/forum/go_quote.gif 
_I would. I want to know if I can get the same sound out of a PC from the coax as I could from USB for an added $300. Cant really afford either DAC but may just get one anyway. 
	

	
	
		
		

		
		
	


	


_

 

I should also add that there are other features that account for the $300 increase...

 1 - The most obvious is the USB solution. Unlike most other dac's with USB input, our USB solution is NOT an off-the-shelf solution. Our solution required a significant co-development period with a contracted software developer (Centrance). Our USB solution was the only _native_ 24-bit / 96-kHz USB solution at the time of its release, and may still be the only one. It is bit-transparent at all sample rates, automatically adjusting to whatever sample rate the audio program delivers.

 2 - A hidden feature of the DAC1 USB/PRE is the redesigned analog output stage. The output drivers of the DAC1 USB/PRE are the latest opamps from National Semiconductor, the LM4562's. They are capable of delivering high amounts of current before distortion. This allows the performance to be maintained even in low impedance situations. Furthermore, it allowed us to lower the output impedance of the DAC1 so that amp's and pre's with significant input capacitance won't roll-off the high frequencies.

 3 - Multiple gain ranges for the headphone amplifier to accommodate a wider range of headphone sensitivities. 

 4 - An auto-mute switch was built into one of the two headphone jacks. This feature automatically mutes the main outputs (XLR and RCA) whenever headphones are plugged in. This makes switching from headphones to loudspeakers seamless.

 5 - An automatic standby mode was designed to detect when the DAC1 USB is no longer being sent a signal. When it senses the lack of signal, it will automatically engage the standby mode. When a digital signal is once again present, it will automatically re-awaken and begin converting.

 Thanks,
 Elias


----------



## EliasGwinn

Quote:


  Originally Posted by *tps* /img/forum/go_quote.gif 
_Here's a question/suggestion: Would it be possible to add some kind of Test indicator to the DAC1 which indicates a bit-perfect signal path? You'd play some sort of test file at the bit-depth and sample rate you wanted to test and if the DAC1 (or DAC1 USB or DAC1 PRE) received the signal OK, it would give a "bit-perfect" indication. Maybe the test signal could be something along the lines of the HDCD signal, where it contains a psuedo-random code in the LSB. I realize this doesn't test the high bits, but my guess is that it would catch 99% of the problems which normally occur._

 

Yes, this would be possible, but it would be a very expensive feature. It would require serious software development and more built-in 'intelligence' chips.

 But it is something we will keep in mind when we develop new products. Thanks for the idea!

 -Elias


----------



## EliasGwinn

Quote:


  Originally Posted by *bizkid* /img/forum/go_quote.gif 
_Hi Elias, thanks for the instructions. I measured the XLR outputs and they are perfectly calibrated. However i found the root of my problem:

 The 2 headphone outputs on the DAC1 are slightly de-calibrated and i used these for reference.
 The right side measures 2.8V, the left 3.1V with volume raised. The difference varies throughout the volume range but stays at a minimum of 0.3V.

 I noticed that there's a small poti on the PCB which says something like L/R balance close to the headphone section. Is that actually used to calibrate it?

 Best, Christian_

 

The L/R balance potentiometer is for balancing the all variable outputs, including XLR, RCA, and headphone.

 Measure the XLR or RCA outputs in variable mode (w/ volume around 1 o'clock). Determine if the is balance is offset similar to the headphones. If so, the L/R balance pot is where you should adjust this.

 Thanks,
 Elias


----------



## skamp

Quote:


  Originally Posted by *EliasGwinn* /img/forum/go_quote.gif 
_Yes, this would be possible, but it would be a very expensive feature. It would require serious software development and more built-in 'intelligence' chips._

 

Personally I just check for bit-perfect output myself: record (digitally) what comes in, and compare input and output PCM byte for byte. Once I've established bit-perfection, I don't really need a LED to tell me that (although that would be a neat gizmo).


----------



## EliasGwinn

Quote:


  Originally Posted by *HeadLover* /img/forum/go_quote.gif 
_HI

 I wanted to ask a question about sampling rate
 There is a litte of argument about it on the PICO thread

 Can a USB do 96/24 nativ?
 or what?

 And how is it implemented in the DAC1 ?

 What about 192/24 ??
 And, when getting sample rate like 24/96, is it a "true" sample ?
 Means I get it on the output? with out any down sampling or what ever?

 And is there any problem with USB and 24/96 ?

 Also, does the DAC1 use the USB 5v power? because if so, it can cause some problems, like on my PC or portable pc (laptop) where jumps on the 5V and so on, can cause downgrade to SQ

 I hope you can give me (and the other guys here) some goos explanation about it 
	

	
	
		
		

		
		
	


	


_

 

The DAC1 USB is absolutely capable of doing true, bit-transparent, 2-channel 96/24 natively. 

 Thereotically, USB1.1 is capable of 2-channel 192/24, but it would be extremely difficult to implement and maintain because it is pushing bandwidth limitations of a connection that is already fragile. 

 The DAC1 USB does not use the USB power supply. It uses its own internal torodial power supply.

 Thanks,
 Elias


----------



## HeadLover

amm, and what about USB2 ??
 Can it have 24/196KHZ ??

 Also,
 What about "unregualer" sample rates like 24/48 and 24/88 and 24/176 ?
 Can it handle it ?


----------



## HeadLover

Oh and BTW
 When using XLR with an amp that have XLR inputs
 Will it be better than using RCA ?
 What will give better SQ ?


----------



## EliasGwinn

Quote:


  Originally Posted by *HeadLover* /img/forum/go_quote.gif 
_amm, and what about USB2 ??
 Can it have 24/196KHZ ??

 Also,
 What about "unregualer" sample rates like 24/48 and 24/88 and 24/176 ?
 Can it handle it ?_

 

USB 2.0 can theoretically do 75 channels of 192/24, but there are no native drivers to support USB 2.0.

 Benchmark's USB solution will do 44.1, 48, 88.2, and 96 kHz, all at 24-bit. Even when playing 16-bit audio at any of these sample rates, the DAC1 USB/PRE establishes a 24-bit connection to ensure that no bits will be truncated.

 Thanks,
 Elias


----------



## HeadLover

I am really looking into buying it!!
 Seem very nice
 BUT
 I am just afraid of you getting something better soon or something that will be much more better
 So it is a hard choice if I should wait or pick it up now
 I guess the one with the PRE is the best one right now, right?
 Will all kind of tweaks and so on to provide best SQ ?


----------



## EliasGwinn

Quote:


  Originally Posted by *HeadLover* /img/forum/go_quote.gif 
_Oh and BTW
 When using XLR with an amp that have XLR inputs
 Will it be better than using RCA ?
 What will give better SQ ?_

 

Well, it depends on the source and amp, but, assuming everything was designed properly, a truly balanced (XLR) connection will only be better then an unbalanced connection (RCA) with regards to noise performance. Truly balanced connections are more immune to noise then unbalanced connections.

 Note: not all XLR connectors are truly balanced!

 Thanks,
 Elias


----------



## HeadLover

I think that someone told it here or other places, that the DAC1 isn't really fully balanced,
 Is it true or what ?
 Can you put some more light about it please ?


----------



## parrot5

Quote:


  Originally Posted by *EliasGwinn* /img/forum/go_quote.gif 
_........ *snip*
 5 - An automatic standby mode was designed to detect when the DAC1 USB is no longer being sent a signal. When it senses the lack of signal, it will automatically engage the standby mode. When a digital signal is once again present, it will automatically re-awaken and begin converting.

 Thanks,
 Elias_

 

Now that you mention it should go to standby when there is no signal being transmitted, I find it strange that mine doesn't do it.. I am using the USB (4) and the Coaxial (1) connection, but even with no program utlizing the soundcards (and no Windows sound), the connection is still on. The input indicator lights stay on (when I've chosen 1 or 4). Any ideas?


----------



## EliasGwinn

Quote:


  Originally Posted by *HeadLover* /img/forum/go_quote.gif 
_I am really looking into buying it!!
 Seem very nice
 BUT
 I am just afraid of you getting something better soon or something that will be much more better
 So it is a hard choice if I should wait or pick it up now
 I guess the one with the PRE is the best one right now, right?
 Will all kind of tweaks and so on to provide best SQ ?_

 

In most cases, there aren't any tweaks needed. If any are needed, they are few, and they are all explained in the manual. 

 Thanks,
 Elias

 ps. I just noticed that my last post was the 2000th of this thread!


----------



## EliasGwinn

Quote:


  Originally Posted by *HeadLover* /img/forum/go_quote.gif 
_I think that someone told it here or other places, that the DAC1 isn't really fully balanced,
 Is it true or what ?
 Can you put some more light about it please ?_

 

Absolutely not true. 

 Each leg of the balanced outputs are driven by independent amplifiers. Truly active balanced outputs.

 Thanks,
 Elias


----------



## EliasGwinn

Quote:


  Originally Posted by *parrot5* /img/forum/go_quote.gif 
_Now that you mention it should go to standby when there is no signal being transmitted, I find it strange that mine doesn't do it.. I am using the USB (4) and the Coaxial (1) connection, but even with no program utlizing the soundcards (and no Windows sound), the connection is still on. The input indicator lights stay on (when I've chosen 1 or 4). Any ideas?_

 

Your computer/soundcard is sending blank packets of data. Usually a device doesn't stop sending packets until it is shut down.

 Disconnect the cable and see if the DAC1 goes into auto-standby mode.

 Thanks,
 Elias


----------



## HeadLover

So
 Will there be a new version soon or something?
 I just HATE to drop down that amount of $ and see after few weeks or months a brand new version 
	

	
	
		
		

		
		
	


	




 What do you say? what is the RD team working on now ?
 And when will it be out?
 (if you can tell)
 It is just, that I have DAC, but I want one with USB, so I am not in a rush.


----------



## EliasGwinn

Quote:


  Originally Posted by *HeadLover* /img/forum/go_quote.gif 
_So
 Will there be a new version soon or something?
 I just HATE to drop down that amount of $ and see after few weeks or months a brand new version 
	

	
	
		
		

		
		
	


	




 What do you say? what is the RD team working on now ?
 And when will it be out?
 (if you can tell)
 It is just, that I have DAC, but I want one with USB, so I am not in a rush._

 

We have no announcements concerning new poducts.

 Thanks,
 Elias


----------



## parrot5

Quote:


  Originally Posted by *EliasGwinn* /img/forum/go_quote.gif 
_Your computer/soundcard is sending blank packets of data. Usually a device doesn't stop sending packets until it is shut down.

 Disconnect the cable and see if the DAC1 goes into auto-standby mode.

 Thanks,
 Elias_

 

Oh when disconnected of course it goes to standby. I was just wondering if that's how it should be (not going to standby when connected to computer). Thanks for the clarification.


----------



## G-U-E-S-T

Hi Elias,

 I do have another question for you please. Is it damaging to the DAC1 PRE if the power is completely cut off to the unit when not in use? 

 You see, I realize that the DAC1 has a great standby function - but the power distribution device that we currently use in our stereo system, completely cuts off power to the DAC1 when the stereo system is shut off. The DAC1 gets nice and warm during use, and of course gets completely cool when the system is off. I usually have the DAC1 PRE on for several hours during each evening, and completely powered off otherwise.

 I will eventually (soon!) get a new power distribution solution that will keep power to the DAC1 even when the rest of the stereo system is turned off.

 But in the meantime I am now getting very worried, that the current on-off power cycling we have innocently been doing so far (and the hot-cold temperatures resulting), might accidentally have been doing damage to the electronics of the DAC1 PRE(?). So please Elias, is the DAC1 PRE designed to be durable enough, to definitely handle this on-off power cycling that we have been doing each day so far?

 Thank you in advance for your reply, and again for this great product.


----------



## HeadLover

I want to add to G-U-E-S-T question,
 How stable and reliable is the PSU on the DAC1 ?
 Will a power cond improve it? or can it just "plug and play" and that all?
 The same goes for getting other than stock power cable unit.
 (Does it matter on it ??)


----------



## EliasGwinn

Quote:


  Originally Posted by *G-U-E-S-T* /img/forum/go_quote.gif 
_ Is it damaging to the DAC1 PRE if the power is completely cut off to the unit when not in use?_

 

Power-cycling the DAC1 PRE will not affect its performance or lifespan whatsoever. You can feel free to turn it off with the rest of your stereo equipment...it won't hurt at all.

 My own personal DAC1 setup is very simlar to yours. Its on a switched power distribution unit along with my powered monitors. My DAC1 is only turned on when I'm using it, and immediately powered down when I'm done. 

 Conversely, the recording studio where I work (Subcat Studios) has left their DAC1 powered-on for the last 4+ years straight (less a day here and there)!! And it sounds as good as a new one coming off the assembly line!

 Thanks,
 Elias


----------



## EliasGwinn

Quote:


  Originally Posted by *HeadLover* /img/forum/go_quote.gif 
_I want to add to G-U-E-S-T question,
 How stable and reliable is the PSU on the DAC1 ?
 Will a power cond improve it? or can it just "plug and play" and that all?
 The same goes for getting other than stock power cable unit.
 (Does it matter on it ??)_

 

The PSU on the DAC1 can tolerate extreme variations in the AC supply without affecting sonic performance. 

 The PSU has more then 20% tolerance with respect to nominal AC supply voltage (110 or 220). In other words, if the a 110V supply sags to 90V, the performance will not be affected at all.

 Also, it is remarkably immune to impurities of the AC supply. We can inject the power rails with significant amounts of noise without affecting the sonic performance.

 A power conditioner and after-market power cable will not affect the perfomance of the DAC1. Save your money to buy more CD's!

 Thanks,
 Elias


----------



## HeadLover

Really?
 Did you test?
 Just wonder cause many tell me that power cond and power lines and so on, really effect DAC and PRE
 How is it with the DAC1 ?
 Have you done tests with it ?


----------



## EliasGwinn

Quote:


  Originally Posted by *HeadLover* /img/forum/go_quote.gif 
_Really?
 Did you test?
 Just wonder cause many tell me that power cond and power lines and so on, really effect DAC and PRE
 How is it with the DAC1 ?
 Have you done tests with it ?_

 

Yes, we have. We've applied extreme disruptions to the power supply to test the performance and limitations of the PSU and audio circuit. 

 We've injected several volts of AC, starting at 60 Hz and sweeping up to 20 kHz, on the power supply rails. During CE testing, we've injected RF into the hot and neutral AC lines. During all of these tests, there was no detectable change in performance.

 Also, we can drop the AC voltage below the threshold of operation for the regulators, such that the regulators will shut off. In this case, several volts of AC are present on the rails. Still, the performance is not affected.

 The circuits in the DAC1/USB/PRE are extremely immune to even extreme variations in quality of AC power.

 If, as you mentioned, many people told you that power affects their dac or pre, then they need to question the manufacturers of those devices. We believe that a well-built device will be immune to such variations, as much as possible.

 Thanks,
 Elias


----------



## HeadLover

Cool !

 I have one more question, I am thinking of getting myself a power amp - digital one, from NUFORCE (like the 8.5 vs or even the mono 9 SE v2)

 How well will the DAC1 be with it?
 The NuForce 9 SE v2 is sure high end, but some claim it is to much "digital" so they use tubes or what ever as a pre to soften things up
 Can I use it with the DAC1 working as both a PRE and a DAC (connected from my PC as a source)

 BTW
 Thanks for all the great answers !


----------



## HeadLover

BTW
 I have one more question
 Why Benchmark won't make a POWER amp to accomplish the DAC1 PRE ?
 I think that a nice two channel one, with something like 70W to 100W to each side will be just sweet!
 So I can buy both the DAC1 and the POWER1 (just a name I have made up now) and than just connect it to a speakers and that all!

 That can sure be nice!


----------



## EliasGwinn

Quote:


  Originally Posted by *HeadLover* /img/forum/go_quote.gif 
_Cool !

 I have one more question, I am thinking of getting myself a power amp - digital one, from NUFORCE (like the 8.5 vs or even the mono 9 SE v2)

 How well will the DAC1 be with it?
 The NuForce 9 SE v2 is sure high end, but some claim it is to much "digital" so they use tubes or what ever as a pre to soften things up
 Can I use it with the DAC1 working as both a PRE and a DAC (connected from my PC as a source)

 BTW
 Thanks for all the great answers !_

 

The DAC1 PRE will work properly with NUFORCE amps, but it will be up to you to determin if you enjoy how it sounds.

 Thanks,
 Elias


----------



## G-U-E-S-T

Elias, you're the greatest - thank you again for all your excellent help. You are very much appreciated!


----------



## Quaddy

hi elias, regarding dac1pre.

 i am trying to discover a little about the DAC1 pre RCA *input* impedance, as have a selectable impedance interconnect and am just trying to get the chain in harmony as much as possible

 thanks for any info


----------



## EliasGwinn

Quote:


  Originally Posted by *Quaddy* /img/forum/go_quote.gif 
_hi elias, regarding dac1pre.

 i am trying to discover a little about the DAC1 pre RCA *input* impedance, as have a selectable impedance interconnect and am just trying to get the chain in harmony as much as possible

 thanks for any info_

 

The input impedance of the analog RCA inputs is 20k. 

 Tell me more about these interconnects...selectable impedance?

 Thanks,
 Elias


----------



## Quaddy

many thanks elias, thats helped a lot 
	

	
	
		
		

		
		
	


	




 - i will select the LOW ssetting then on the cables (5-50k)

 as per my signature really. the MIT shotgun S1 rca with switchable impendance boxes. see here


----------



## gjwaudio

Hi Elias

 I'm wondering about the DAC1 Classic and absolute polarity.

 I presume it preserves the polarity of the incoming signal (...oooh, nice alliteration there 
	

	
	
		
		

		
		
	


	




).

 That is, for a digital input with a positive-going pulse, both the RCA and XLR outputs will present a positive-going pulse. In the case where the XLR connection is wired to an unbalanced input (as specified in the User Manual - with pin 3 left floating) the pulse will still be positive-going.

 Why do I ask ? Just confirming the Benchmark is non-inverting... I suspect somewhere in my PB chain the signal is being inverted, and I want to rule out the DAC1 as the culprit.

 Thanks for your help.
 Grant


----------



## StanZigo

Quote:


  Originally Posted by *EliasGwinn* /img/forum/go_quote.gif 
_Yes, we have. We've applied extreme disruptions to the power supply to test the performance and limitations of the PSU and audio circuit. 

 We've injected several volts of AC, starting at 60 Hz and sweeping up to 20 kHz, on the power supply rails. During CE testing, we've injected RF into the hot and neutral AC lines. During all of these tests, there was no detectable change in performance.

 Also, we can drop the AC voltage below the threshold of operation for the regulators, such that the regulators will shut off. In this case, several volts of AC are present on the rails. Still, the performance is not affected.

 The circuits in the DAC1/USB/PRE are extremely immune to even extreme variations in quality of AC power.
 Elias_

 

I can testify to this.

 A while back I had a malfunctioning fuse in the main distribution cabinet of my building. It was arcing like mad, all the lights in my apartment were flickering. It was so bad, it ended up breaking my TV and the power supply of my monitor.

 But I never heard the slightest problem with my DAC1. Nothing. EVER.

 Power cables don't matter, power supplies do.


----------



## EliasGwinn

Quote:


  Originally Posted by *gjwaudio* /img/forum/go_quote.gif 
_Hi Elias

 I'm wondering about the DAC1 Classic and absolute polarity.

 I presume it preserves the polarity of the incoming signal (...oooh, nice alliteration there 
	

	
	
		
		

		
		
	


	




).

 That is, for a digital input with a positive-going pulse, both the RCA and XLR outputs will present a positive-going pulse. In the case where the XLR connection is wired to an unbalanced input (as specified in the User Manual - with pin 3 left floating) the pulse will still be positive-going._

 

Absolutely. (pun intended!)

 We've done polarity tests with pulse waves, such as the one you mentioned. 

 Rest assured, the DAC1 is not the inverting culprit in your system.

 Thanks,
 Elias


----------



## EliasGwinn

Quote:


  Originally Posted by *StanZigo* /img/forum/go_quote.gif 
_I can testify to this.

 A while back I had a malfunctioning fuse in the main distribution cabinet of my building. It was arcing like mad, all the lights in my apartment were flickering. It was so bad, it ended up breaking my TV and the power supply of my monitor.

 But I never heard the slightest problem with my DAC1. Nothing. EVER.

 Power cables don't matter, power supplies do._

 

Wow... Now that is a powerful testimony (ok, ok, I'll stop with the puns).

 Your final statement is spot-on:

  Quote:


 Power cables don't matter, power supplies do. 
 

...truth.

 Thanks,
 Elias


----------



## Headphony

From page 77:
  Quote:


  Originally Posted by *EliasGwinn* /img/forum/go_quote.gif 
_...Thermal cycling may increase the chances of failure. Therefore, it is recommended to keep the unit powered up continuously. The auto-standby feature of the DAC1 USB will not cause the unit to cool, so it does not present a problem..._

 

From page 202:
  Quote:


  Originally Posted by *EliasGwinn* /img/forum/go_quote.gif 
_...Power-cycling the DAC1 PRE will not affect its performance or lifespan whatsoever. You can feel free to turn it off with the rest of your stereo equipment...it won't hurt at all..._

 

I don't quite understand that. Is there a difference between the USB and PRE versions, with respect to the advisability of keeping the unit powered on?


----------



## tps

Well, I ordered a DAC1 PRE yesterday to use with Magnepan MMG speakers and Outlaw 2200 amps. Both Magnepan and Outlaw seem to be "no nonsense" companies who let the quality of their products speak for themselves. And from everything I can tell from this forum and the online manuals, Benchmark also takes a no nonsense approach. I'm looking forward to some great listening sessions in a few days when the DAC1 PRE arrives!


----------



## HeadLover

Quote:


  Originally Posted by *tps* /img/forum/go_quote.gif 
_Well, I ordered a DAC1 PRE yesterday to use with Magnepan MMG speakers and Outlaw 2200 amps. Both Magnepan and Outlaw seem to be "no nonsense" companies who let the quality of their products speak for themselves. And from everything I can tell from this forum and the online manuals, Benchmark also takes a no nonsense approach. I'm looking forward to some great listening sessions in a few days when the DAC1 PRE arrives!_

 

looking for a full review of it when you get it


----------



## paradoe

I'm preparing to get a DAC1u


----------



## ert

New DAC1 PRE owner here. I've been listening to this unit for the past few days, and the sound is absolutely wonderful. I have never been able to listen to music for 6+ hours on headphones (low volume thankyou) without suffering fatigue. This is by far the most transparent audio device I have come across. I've been reading this thread in its entirety (phew!) for the last few days and I'd like to thank Elias for his patience and positive attitude towards the community. As a fellow engineer who deals with similar subjects (DSP), I can't say I'd have the guts/patience to discuss techical details at this level on an open forum.

 My objective in purchasing a DAC1 PRE was to provide a quality listening setup at my desk while using headphones. My reference system contains Marantz SACD, Magnum Dynalab FM as sources with Classe integrated and B&W CDM9NT speakers, so my requirements are pretty high wrt sound quality. Previously I have been using a Headroom Supreme with BS1 P/S feeding Gradi 325i and Etymotic ER4S phones.

 My desktop digital source is either going to be a DAP or music-server connected to the DAC1 PRE via coax/glass or native USB*. I've recently begun ripping my CD collection to flac and the prospect of having a true bit-transparent audio server is pretty exciting. I have to say it will beat walking around with armfuls of CDs! There are several nice FM tuners and CD players with similar form factors that I'll probably pick up to fill out the system.

 The build quality of the unit is superb, inside and out. The inside of the unit smells like candy for some reason, I'm guessing due to the PCB wash used. I refrained from licking the DAC1 PRE.

 The manual is clear and well presented. The included cables are of nice quality and the spare fuses are appreciated.

 The sound quality from the onboard headphone amp is outstanding. I was planning to use my Headroom amp with the DAC1 PRE, but so far I am leaning towards just using the included amp. It's true that the onboard amp is very transparent and neutral, although I'm not sure that always sounds "better" to me. I'm a fan of Headroom's crossfeed and having used it for so many years, it takes a bit of getting used to listening with out it. The DAC1 has made that pretty easy though! I'll need to do a lot more listening before I make up my mind. It does seem that using an outboard amp with a warmer tone can help thin recordings. This is not a poke at the DAC1, merely saying that sometimes you may need to occasionally color the sound to account for deficiencies in the recording. It's nice that the DAC1 doesn't force you into that position!

 Using uncompressed classical audio samples and A/B testing I can't discern any difference between a DAP via optical, my PC via USB and my CD transport via coax. I've pulled all sorts of cables from my junk cable collection and the DAC1 does not seem to be affected by cable quality or construction. 

 The additional 20dB attenuation on the headphone out is greatly appreciated. My Grados and Etys were too loud past 9 o'clock with the factory setting. Now I can comfortably listen between 10 and 2.

 I do have a few very minor suggestions:

 1. Provide output pads on the unbalanced outputs. I know that the level can be set with trimmers, but having say a -20dB jumper on the RCAs would be nice. I always need to use a line level attenuator on my digital sources, and the DAC1 is no exception. I don't know if thats common on DACs or perhaps my amps are just way to sensitive. Currently I have a Harrison Labs 6dB attenuator between the DAC1 PRE and my Headroom amp. It would be nice not to have additional components sticking out. I guess another alternative would be to use XLR->RCA cables and just use the XLR pads to reduce the signal level.

 2. Provide front panel indicators for the incoming data rate. Ideally, I'd say move the DAC1 PRE logo elsewhere and stick some LEDs in its place. However, I understand that re-doing the faceplate and machining costs would be prohibitively expensive for such a small change. Any way to use the existing LEDs to achieve this? Of course, the front panel is an elegant design as is, and cramming in more blinking lights may detract from usability.

 Again, thank you Elias for the great help and providing a no-nonsense DAC and preamp. It's so refreshing to discuss audio on a true scientific technical level when so much of the community is basically engaging in modern day alchemy.

 *So far my DAC1 PRE USB experience has not been good. [EDIT - Revised - no problems. see two posts down for resolution.] To be fair and to the point, this is not a fault of the DAC1, but is entirely OS dependent. My laptop seems to suffer the ubiquitous USB sound device pop/clicks problem. When my laptop is idle, the sound is perfect. However, if I do something as simple as use Firefox (or IE) I get pops/clicks galore. This problem of course is related to the genius decision by Microsoft to allocate device IRQs somewhat randomly and not according to common sense. In my case, my video card, USB root host, and Wireless card all share the same IRQ. Anytime there is wireless or video acitivity (ie all the time), the sound is choppy.

 I'm using a Lenovo Thinkpad T60 with Windows XP and foobar 2000 to play flac files. The CPU is not under any load whatsoever. This problem is pretty surprising to me given that I've NEVER had a technical problem like this with Thinkpads. The pops/clicks seem to be independent of disk and cpu activity. I've run disk and cpu stress tests while simultaneously playing foobar and don't get the clicks/pops. I'm going to try and debug this for awhile longer, but I may end up just using a standalone DAP, which is a bit disappointing since I'd prefer a PC interface.

 As an alternative, can anyone recommend a simple CardBus SPDIF interface? I'm a bit hesitant to introduce a PC sound device in the chain, due to all the previous reasons that have been stated wrt unknown DSP going on.

 EDIT - it seems I can get a mini dock from Lenovo with SPDIF so I may end up using that despite the fact that I hate using Frankenstein-ish laptop add ons.


----------



## nd4speed

Quote:


  Originally Posted by *ert* /img/forum/go_quote.gif 
_*So far my DAC1 PRE USB experience has not been good. To be fair and to the point, this is not a fault of the DAC1, but is entirely OS dependent. My laptop seems to suffer the ubiquitous USB sound device pop/clicks problem. When my laptop is idle, the sound is perfect. However, if I do something as simple as use Firefox (or IE) I get pops/clicks galore. This problem of course is related to the genius decision by Microsoft to allocate device IRQs somewhat randomly and not according to common sense. In my case, my video card, USB root host, and Wireless card all share the same IRQ. Anytime there is wireless or video acitivity (ie all the time), the sound is choppy.

 I'm using a Lenovo Thinkpad T60 with Windows XP and foobar 2000 to play flac files. The CPU is not under any load whatsoever. This problem is pretty surprising to me given that I've NEVER had a technical problem like this with Thinkpads. The pops/clicks seem to be independent of disk and cpu activity. I've run disk and cpu stress tests while simultaneously playing foobar and don't get the clicks/pops. I'm going to try and debug this for awhile longer, but I may end up just using a standalone DAP, which is a bit disappointing since I'd prefer a PC interface._

 

Regarding dropouts, try using a computer with nothing else installed on it (a single purpose machine you won't use while playing music) before changing interfaces.

 I had the same problem on my dual core laptop w/ 2 gigs of ram (very infrequently however). It didn't appear to be a problem with resources because I would watch resource monitor while it was going on and results were inconclusive...in short don't trust it. Computers were never meant to be single purpose music servers so it can often be frustrating troubleshooting problems of this nature because there is so much going on in the background (services and processes kicking on and off, memory being allocated and deallocated, reading and writing from disk cache, memory, HD, etc.)

 I then grabbed my much less powerful laptop with only the OS and foobar installed and I've never had the problem since.

 Also make sure you turn off sleep of the HD, monitor, NIC etc. and any screensavers.


----------



## ert

Quote:


  Originally Posted by *nd4speed* /img/forum/go_quote.gif 
_Regarding dropouts, try using a computer with nothing else installed on it (a single purpose machine you won't use while playing music) before changing interfaces._

 

Even if I had another spare machine, it kind of defeats the purpose in my case because I want a minimal desktop setup. I can definitely appreciate using a dedicated media server in a different situation though.

 I was able to correct the issue - it turned out to be the PCI video card, which is a common problem with PCI audio and usb devices according to some sites I've been reading. Decreasing the video hardware acceleration by one step was all it took.

 The problem manifested as pops/crackles when pages with graphics were rendered in a browser. For example, if I load news.yahoo.com or rapidly scroll the site, it sounds like someone is scraping the RCA inputs over a cheese grater (that bad). No other application caused this problem. Going to the Troubleshoot tab in the Advanced section of the display adapter, I reduced the hardware acceleration setting by one, which has the effect of disabling bitmap accelerations. 

 That's a fine tradeoff for me. Most of my PC use is coding in various IDEs and there's no graphic overhead there anyway.


----------



## nd4speed

Quote:


  Originally Posted by *ert* /img/forum/go_quote.gif 
_For example, if I load news.yahoo.com or rapidly scroll the site, it sounds like someone is scraping the RCA inputs over a cheese grater (that bad)._

 

LOL, I'm glad you were able to solve it, that could get annoying after a while.


----------



## Quaddy

Elias, one more question if i may...

 the DAC1 pre's fuse holder has two fuses in.
 for my own amusement i am wanting to replace them with hifituning ceramic fuses (exactly the same spec: 250v,5x20mm,slow-blow,500mA)

 i wouldnt need to replace both would i? as isnt the only one active the fuse in the grey plastic holder at the bottom?

 if so i will just buy one fuse you see. and leave the one in the top as stock

 thanks for any info


----------



## Bostonears

Quote:


  Originally Posted by *ert* /img/forum/go_quote.gif 
_There are several nice FM tuners and CD players with similar form factors that I'll probably pick up to fill out the system._

 

Other than the Parasound Z series, what FM tuners have you found in this narrow component form factor? For CD players, I have seen the Shanling PCD300/3000. Any others you've found? How about an AC power conditioner?


----------



## EliasGwinn

Quote:


  Originally Posted by *Headphony* /img/forum/go_quote.gif 
_Is there a difference between the USB and PRE versions, with respect to the advisability of keeping the unit powered on?_

 

Very observant, Headphony! Those two opposing quotes side-by-side make me sound very presidential, eh?

 Actually, the first quote is addressing thermal cycling of lead-free solder. It may cause or contribute to 'tin whiskers' forming and protruding from solder connections. There isn't enough evidence to verify this, but it is a concern when considering the expected lifespan of a product using a new form of solder. This is the same for both the USB and PRE and any other device using lead-free solder (because of the ROHS requirement). 

 However, there is also a subtle advantage of power cycling: increasing the lifespan of the power caps. At normal operating temperatures, the caps in the DAC1/USB/PRE will work for +/-10 years of operation. So, if you won't be using the DAC1 for a few weeks/months/years, it would be a good idea to turn it off. This won't affect the actual product's lifespan because it isn't difficult to re-cap.

 As I mentioned in a previous post, I power-down my DAC1 at all times that it is not in use. I don't give it any 'warm-up' period; I go straight into listening.

 Conversely, the recording studio I work at has had three (non-ROHS) DAC1's powered almost consistently for 5 years.

 Neither the studio's DAC1's or my DAC1 have given us any problems at all.

 So... I guess the final verdict is, you'll be fine leaving it on, and you'll also be fine turning it off. 

 Thanks,
 Elias


----------



## EliasGwinn

Quote:


  Originally Posted by *ert* /img/forum/go_quote.gif 
_I was able to correct the issue - it turned out to be the PCI video card, which is a common problem with PCI audio and usb devices according to some sites I've been reading. Decreasing the video hardware acceleration by one step was all it took.

 The problem manifested as pops/crackles when pages with graphics were rendered in a browser. For example, if I load news.yahoo.com or rapidly scroll the site, it sounds like someone is scraping the RCA inputs over a cheese grater (that bad). No other application caused this problem. Going to the Troubleshoot tab in the Advanced section of the display adapter, I reduced the hardware acceleration setting by one, which has the effect of disabling bitmap accelerations. _

 

Very interesting! I'm glad you figured it out. 

 Let me know how things are working out...

 Thanks,
 Elias


----------



## EliasGwinn

Quote:


  Originally Posted by *Quaddy* /img/forum/go_quote.gif 
_Elias, one more question if i may...

 the DAC1 pre's fuse holder has two fuses in.
 for my own amusement i am wanting to replace them with hifituning ceramic fuses (exactly the same spec: 250v,5x20mm,slow-blow,500mA)

 i wouldnt need to replace both would i? as isnt the only one active the fuse in the grey plastic holder at the bottom?

 if so i will just buy one fuse you see. and leave the one in the top as stock

 thanks for any info_

 

Both fuses are in the circuit at all times, for both 110 and 220V. 

 One is on the hot leg of the plug, and the other is on neutral.

 Thanks,
 Elias


----------



## Quaddy

thanks elias.


----------



## ert

Quote:


  Originally Posted by *Bostonears* /img/forum/go_quote.gif 
_Other than the Parasound Z series, what FM tuners have you found in this narrow component form factor? For CD players, I have seen the Shanling PCD300/3000. Any others you've found? How about an AC power conditioner?_

 

I have compiled a short list (below). I'm actually leaning towards the Parasound and the Shanling myself 
	

	
	
		
		

		
		
	


	




. There are some more expensive CD players, but with the DAC1, they seem unnecessary. I'm holding out for a Naim tuner, but they're hard to find. There were also a lot of shelf systems in the late 80s and 90s by some high end companies that might provide some good used components.

 Tuners
 Parasound Z
 Naim NAT-02 (discontinued)
 Tivoli Audio Model One

 CD players
 Shanling PCD-300A
 Musical Fidelity X-Ray v3 [ce]
 TEAC PD-H300 (discontinued?)
 Cyrus 8x


----------



## zenjazz

A stupid question: can Benchmark Pre be used directly with Cowon O2? is Cowon O2's AV out of any help?


----------



## HeadLover

BTW, I have also a little stupid question
 Can the DAC be used with speakers that don't need an amp and so on, like Audioengine A5 ?


----------



## Quaddy

Quote:


  Originally Posted by *zenjazz* /img/forum/go_quote.gif 
_A stupid question: can Benchmark Pre be used directly with Cowon O2? is Cowon O2's AV out of any help?_

 

hmmm, i use a cowon Q5 directly interfaced with the DAC1 pre (turn spdif on in menu), via the cowons proprietory AV-out digital coax into coax in on the DAC1

 but am not sure if the O2 has digital out, via a similar flylead.


----------



## EliasGwinn

Quote:


  Originally Posted by *zenjazz* /img/forum/go_quote.gif 
_A stupid question: can Benchmark Pre be used directly with Cowon O2? is Cowon O2's AV out of any help?_

 

It's hard to tell. I took a look at their website, and from the (very little) information that I could find, it seems that the 02 does not have a digital audio output. You may want to contact the manufacturer to find out.

 You could send the analog output into the analog input of the DAC1 PRE, but you are then relying the D-to-A in the 02...a scary prospect.

 Thanks,
 Elias


----------



## EliasGwinn

Quote:


  Originally Posted by *HeadLover* /img/forum/go_quote.gif 
_BTW, I have also a little stupid question
 Can the DAC be used with speakers that don't need an amp and so on, like Audioengine A5 ?_

 

The DAC1 works great with powered speakers. 

 Thanks,
 Elias


----------



## EliasGwinn

TO WHOM IT MAY CONCERN:

 An interesting read about the 'aging' of IC's, etc:

Analog Devices: The Long Term Stability of Precision Analog ICs, or How to Age Gracefully and Avoid Sudden Death :: Analog Microcontrollers

 This is FYI, so that you can question the next person who sells you something that doesn't sound good but says it's because the IC's didn't 'break-in' yet.

 Thanks,
 Elias


----------



## Bostonears

Quote:


  Originally Posted by *EliasGwinn* /img/forum/go_quote.gif 
_TO WHOM IT MAY CONCERN:

 An interesting read about the 'aging' of IC's, etc:

Analog Devices: The Long Term Stability of Precision Analog ICs, or How to Age Gracefully and Avoid Sudden Death :: Analog Microcontrollers

 This is FYI, so that you can question the next person who sells you something that doesn't sound good but says it's because the IC's didn't 'break-in' yet.

 Thanks,
 Elias_

 

I've never heard anybody claim that ICs need to break in. My understanding is that it's the passive components, especially capacitors, that need to break in.


----------



## riverlethe

Quote:


  Originally Posted by *EliasGwinn* /img/forum/go_quote.gif 
_
 However, there is also a subtle advantage of power cycling: increasing the lifespan of the power caps. At normal operating temperatures, the caps in the DAC1/USB/PRE will work for +/-10 years of operation. So, if you won't be using the DAC1 for a few weeks/months/years, it would be a good idea to turn it off. This won't affect the actual product's lifespan because it isn't difficult to re-cap._

 

Will having the DAC1 in standby increase the lifespan of the caps?


----------



## poo

Quote:


  Originally Posted by *Bostonears* /img/forum/go_quote.gif 
_I've never heard anybody claim that ICs need to break in..._

 

You do actually read other threads on Head-fi right? I'm frequently baffled by how many threads here discuss the need to burn in ICs


----------



## EliasGwinn

Quote:


  Originally Posted by *riverlethe* /img/forum/go_quote.gif 
_Will having the DAC1 in standby increase the lifespan of the caps?_

 

No, standby does not de-activate the power supply.

 But, again, I wouldn't worry about this too much. Just power-down the unit when it won't be used for several days/weeks/months. 

 Thanks,
 Elias


----------



## Bostonears

Quote:


  Originally Posted by *poo* /img/forum/go_quote.gif 
_You do actually read other threads on Head-fi right? I'm frequently baffled by how many threads here discuss the need to burn in ICs 
	

	
	
		
		

		
		
	


	


_

 

I'm sure plenty of end users make such claims, but I should have stated it more clearly: I've never heard an electronics manufacturer claim that ICs need break in. I have heard electronics manufacturers, including at least one capacitor maker, say that capacitors go through an internal chemical transformation during break in, and that seems reasonable considering the composition of capacitors. Also I have no doubt that speaker and headphone drivers can change significantly with break in, as their mechanical components loosen up. However, I have yet to hear anyone provide a coherent explanation for how ICs (or cables for that matter) would change during break in.


----------



## HeadLover

WHY doesn't the DAC1 PRE have more than one output RCA ?
 I think it will be cook to have at least two sets
 One with variable output and one with fixed
 So I can use both headphones amp, and a power amp with it.


----------



## Quaddy

Quote:


  Originally Posted by *HeadLover* /img/forum/go_quote.gif 
_WHY doesn't the DAC1 PRE have more than one output RCA ?
 I think it will be cook to have at least two sets
 One with variable output and one with fixed
 So I can use both headphones amp, and a power amp with it._

 

just use the XLR out as well as the RCA out, buy a few neutrik XLR to rca gender benders and then you have dual RCA outputs


----------



## Quaddy

Quote:


  Originally Posted by *Bostonears* /img/forum/go_quote.gif 
_I'm sure plenty of end users make such claims, but I should have stated it more clearly: I've never heard an electronics manufacturer claim that ICs need break in. I have heard electronics manufacturers, including at least one capacitor maker, say that capacitors go through an internal chemical transformation during break in, and that seems reasonable considering the composition of capacitors. Also I have no doubt that speaker and headphone drivers can change significantly with break in, as their mechanical components loosen up. However, I have yet to hear anyone provide a coherent explanation for how ICs (or cables for that matter) would change during break in._

 

MIT audio state burn in is required for their cables

 my MIT cables included a booklet detailing the required burn in for their IC's

 they state the 2/2 rule;

 75% of performance in two days
 100% in two weeks

 i am not arguing about whether you believe it or not, i am just providing facts that a cable manufacturer quoted burn in as you said you hadnt seen any manufacturer say


----------



## EliasGwinn

Quote:


  Originally Posted by *HeadLover* /img/forum/go_quote.gif 
_WHY doesn't the DAC1 PRE have more than one output RCA ?
 I think it will be cook to have at least two sets
 One with variable output and one with fixed
 So I can use both headphones amp, and a power amp with it._

 

Tell me more about what you are trying to do.


----------



## HeadLover

Quote:


  Originally Posted by *EliasGwinn* /img/forum/go_quote.gif 
_Tell me more about what you are trying to do._

 

I want to connect it both to a heaphones amp (need fixed output ones) and at the same time connect it to a streo power amp

 So, missing two outputs


----------



## EliasGwinn

Quote:


  Originally Posted by *HeadLover* /img/forum/go_quote.gif 
_I want to connect it both to a heaphones amp (need fixed output ones) and at the same time connect it to a streo power amp

 So, missing two outputs 
	

	
	
		
		

		
		
	


	


_

 

Two possible solutions:

 1. Use the RCA outputs in 'Calibrated' (fixed) mode to feed the headphone amp, and use the HPA2 (headphone output) with a phono-to-RCA cable to feed the amplifier. The HPA2 may be a bit hot, so if you have the DAC1 USB / PRE, you would want to lower the gain range of the HPA2 as much as possible.

 2. Use the HPA2 for your headphones. 
	

	
	
		
		

		
		
	


	




 Thanks,
 Elias


----------



## HeadLover

Why not adding RCA out to the unit?
 I mean, it will be nice to have both variable and not, will even make it easier to use and so on


----------



## EliasGwinn

Quote:


  Originally Posted by *HeadLover* /img/forum/go_quote.gif 
_Why not adding RCA out to the unit?
 I mean, it will be nice to have both variable and not, will even make it easier to use and so on



_

 

Maybe on the DAC2. Stay tuned!


----------



## HeadLover

Quote:


  Originally Posted by *EliasGwinn* /img/forum/go_quote.gif 
_Maybe on the DAC2. Stay tuned! 
	

	
	
		
		

		
		
	


	


_

 

I am always tuned in, hope it will come soon


----------



## linuxworks

Quote:


  Originally Posted by *tps* /img/forum/go_quote.gif 
_Well, I ordered a DAC1 PRE yesterday to use with Magnepan MMG speakers and Outlaw 2200 amps._

 

good choice on the maggies. I started with that pair and I love them. a bit inefficient but not a huge problem given their good flat sound (heh) 
	

	
	
		
		

		
		
	


	




 I rarely run into folks who have the maggies and it seems that 'boxy tower' speakers are so much the rage these days (sigh).


----------



## jotos

I have a brand new DAC1 USB. 
 My unit behaves a little strange with standby in USB mode. If I pull out the USB cable from the active Benchmark standby mode is never engaged. If I shutdown my computer with the usb cable attached standby mode is working.

 The only way to make standby work is to reset the unit or start my computer and thereafter shut it down.

 I´m a little puzzled here because I mailed Benchmark support and got the answer that it´s normal that both LED´s is active even in standby.


----------



## EliasGwinn

Quote:


  Originally Posted by *jotos* /img/forum/go_quote.gif 
_I have a brand new DAC1 USB. 
 My unit behaves a little strange with standby in USB mode. If I pull out the USB cable from the active Benchmark standby mode is never engaged. If I shutdown my computer with the usb cable attached standby mode is working.

 The only way to make standby work is to reset the unit or start my computer and thereafter shut it down.

 I´m a little puzzled here because I mailed Benchmark support and got the answer that it´s normal that both LED´s is active even in standby._

 

Hello Jotos,

 What they meant by "normal" is that this happens with a lot of computers. The DAC1's USB input is always locked to the computer in slave mode. Certain computers lock to the DAC1 differently then others. The DAC1's USB input has a "handshake" with the computer. For the USB input to go into standby, certain computers must 'sign off'. If the USB cable is removed from the DAC1 without those computers signing off, the DAC1 won't know that it is disconnected, and it will stay in 'active' mode. 

 Thanks,
 Elias


----------



## jotos

Quote:


  Originally Posted by *EliasGwinn* /img/forum/go_quote.gif 
_Hello Jotos,

 What they meant by "normal" is that this happens with a lot of computers. The DAC1's USB input is always locked to the computer in slave mode. Certain computers lock to the DAC1 differently then others. The DAC1's USB input has a "handshake" with the computer. For the USB input to go into standby, certain computers must 'sign off'. If the USB cable is removed from the DAC1 without those computers signing off, the DAC1 won't know that it is disconnected, and it will stay in 'active' mode. 

 Thanks,
 Elias_

 

Ahh. Thanks for the explanation. Now I can focus on listening 
	

	
	
		
		

		
		
	


	




 Update: I just moved my DAC1 from my Thinkpad (Dac1 standby when shut down) to my HTPC. With my HTPC DAC1 never goes to standby, not even when HTPC is shut down. Are there PC:s that lacks the ability to force standby?


----------



## Bostonears

Quote:


  Originally Posted by *Quaddy* /img/forum/go_quote.gif 
_MIT audio state burn in is required for their cables

 my MIT cables included a booklet detailing the required burn in for their IC's

 they state the 2/2 rule;

 75% of performance in two days
 100% in two weeks

 i am not arguing about whether you believe it or not, i am just providing facts that a cable manufacturer quoted burn in as you said you hadnt seen any manufacturer say_

 

I know MIT cables have "networks", but those boxes have ICs in them?!? What?! Don't ICs require continuous voltage to power them? How the heck could you do that in an interconnect or speaker cable, where the whole idea is to transmit a signal by varying the voltage. Do the cables require an external power source in addition to the signal?


----------



## G-U-E-S-T

Quote:


  Originally Posted by *EliasGwinn* /img/forum/go_quote.gif 
_Maybe on the DAC2. Stay tuned! 
	

	
	
		
		

		
		
	


	


_

 

Elias, is such an updated product release imminent? Did I buy a DAC1 PRE right before the "next greatest" thing comes out?


----------



## EliasGwinn

Quote:


  Originally Posted by *jotos* /img/forum/go_quote.gif 
_I just moved my DAC1 from my Thinkpad (Dac1 standby when shut down) to my HTPC. With my HTPC DAC1 never goes to standby, not even when HTPC is shut down. Are there PC:s that lacks the ability to force standby?_

 

That's right. Some computers simply don't 'sign-off' on their USB port.

 Thanks,
 Elias


----------



## EliasGwinn

Quote:


  Originally Posted by *G-U-E-S-T* /img/forum/go_quote.gif 
_Elias, is such an updated product release imminent? Did I buy a DAC1 PRE right before the "next greatest" thing comes out? 
	

	
	
		
		

		
		
	


	


_

 

No, it was merely a way of telling 'HeadLover' that we will take his suggestions into account when we develop new products. 

 Thanks,
 Elias


----------



## Eric2

Quote:


  Originally Posted by *EliasGwinn* /img/forum/go_quote.gif 
_No, it was merely a way of telling 'HeadLover' that we will take his suggestions into account when we develop new products. 

 Thanks,
 Elias_

 

That’s a shame I will have to cross it of my Xmas list


----------



## tps

Quote:


  Originally Posted by *linuxworks* /img/forum/go_quote.gif 
_good choice on the maggies. I started with that pair and I love them. a bit inefficient but not a huge problem given their good flat sound (heh) 
	

	
	
		
		

		
		
	


	




 I rarely run into folks who have the maggies and it seems that 'boxy tower' speakers are so much the rage these days (sigh)._

 

I've had the DAC1 PRE for about a week now. An EXCELLENT sounding piece of equipment. The MMGs are ruthlessly revealing speakers, and the DAC1 also seems to be very revealing. On a good recording, such as a Linn or Classic Records 24/96 file, the DAC1 / 2200 / MMG combo blows me away. On average recordings, well, it sounds average, but that's the way it was recorded...


----------



## oxophone

Cowon O2 has a USB/ TV out port. Does it mean it is also a digital out port, and hence, can be used with the DAC1 pre?


----------



## Quaddy

Quote:


  Originally Posted by *oxophone* /img/forum/go_quote.gif 
_Cowon O2 has a USB/ TV out port. Does it mean it is also a digital out port, and hence, can be used with the DAC1 pre?_

 

i just had a look at the O2 specs for you, and the only audio output it has is a 3.5mm headphone jack, there is no digital out.

 but you could in theory link it to your dac1 pre with a mini to RCA analogue cable, but the sound may not be too impressive!


----------



## EliasGwinn

Quote:


  Originally Posted by *oxophone* /img/forum/go_quote.gif 
_Cowon O2 has a USB/ TV out port. Does it mean it is also a digital out port, and hence, can be used with the DAC1 pre?_

 

I looked at their website, but I couldn't find anything indicating a digital audio output.

 As Quaddy mentioned, you could connect the analog output to the analog input of the DAC1 PRE, but you will be at the mercy of the internal D/A converter of the Cowon 02.

 Thanks,
 Elias


----------



## oxophone

Thanks. 

 Found that Cowon Q5 and Q5W have digital out: SPDIF. 
 Do you think it will be of any help? 

 I am a techo-illiterate. So, would it be possible for you to kindly browse Q5W's manual and check the compatibility? 
 []http://www.cowonamerica.com/download...w_manual.html]

 Otherwise, have you (or any one else) checked the sound quality of Q5W and DAC1 (pre) combo?


----------



## Quaddy

lol - yes, i was saying this some posts back that i use the Q5 via spdif through the DAC1 pre, and its fantastic sound, the quality of the Pre turns the Q5 into a high quality transport and fed with lossless its very good IMHO - *its compatible!*


----------



## EliasGwinn

Quote:


  Originally Posted by *oxophone* /img/forum/go_quote.gif 
_Found that Cowon Q5 and Q5W have digital out: SPDIF. Do you think it will be of any help? 

 Otherwise, have you (or any one else) checked the sound quality of Q5W and DAC1 (pre) combo?_

 

Oxophone,

 Any device with SPDIF digital output will be compatible with the digital inputs of the DAC1 PRE.

 There can be no better answer for you then the testimony of a user of these products. Apparently Quaddy has used the Q5 and DAC1 PRE with great success. 

 Thanks,
 Elias


----------



## Nikita

Nice DAC. Impressive sound, i have heard it for few minutes and... OMG


----------



## John Reeves

Hi Elias,

 Thank you for all the help and advice you give on this forum. I wonder if you can answer this query for me.

 I have a meridian power amplifier connected directly to my DAC1. My speakers allow for 2 stereo power amplifiers to be used i.e Bi-amped, no pre-amp needed. I am considering adding a second stereo power amp. Can I split the feed from the DAC1 to the 2 stereo power amplifiers. I am using the unbalanced RCA outputs.

 Many thanks

 John


----------



## poo

^ John you can use the XLR connections on the back of your DAC1 and terminate the other end of the cable with RCA for your second amp. Just make sure you leave pin 3 of the XLR connection floating as explained in the manual. This way you can leave your first amp as is, and feed the second with an XLR to RCA cable.


----------



## EliasGwinn

Quote:


  Originally Posted by *John Reeves* /img/forum/go_quote.gif 
_Hi Elias,

 Thank you for all the help and advice you give on this forum. I wonder if you can answer this query for me.

 I have a meridian power amplifier connected directly to my DAC1. My speakers allow for 2 stereo power amplifiers to be used i.e Bi-amped, no pre-amp needed. I am considering adding a second stereo power amp. Can I split the feed from the DAC1 to the 2 stereo power amplifiers. I am using the unbalanced RCA outputs.

 Many thanks

 John_

 

John,

 You can use the XLR outputs (as 'Poo' indicated above), or you can use a splitter on the RCA outputs. The benefit of using the splitter on the RCA's is that the two amplifiers will be receiving identical input signal levels, whereas the XLR outputs will be at a significantly different level. 

 The RCA outputs of the DAC1 should have no problems being double-loaded, depending on the input impedance and capacitance of the amplifiers (the high-current output drivers of the DAC1 USB and DAC1 PRE would be better suited for this). 

 I recommend testing, though. Connect splitters to the RCA and connect one pair of RCA outputs to your current single-amp setup. Connect the other RCA pair to the second amp without connecting that amp to the speakers. Then, simply compare the sound of the single-amp setup with the RCA's connected to the second amp versus without the RCA's connected to the second amp. The single-amp setup should sound no different with or without the second amp being connected to the RCA's.

 Let us know what you find! 
	

	
	
		
		

		
		
	


	




 Thanks,
 Elias


----------



## ted betley

elias: can one use an rca input output from a digital source into an rca cable through an adapter on the other end into an xlr into the the aes/ebu input on th back of the dac1? will this work ok? I have two spdif inputs both w/o toslink and only one spdif input on the dac1


----------



## EliasGwinn

Yes, you can connect an RCA digital output to the XLR digital input of the DAC1. Here are the adaptors you will need:

Canare BNC Female -to- XLR Male Impedance Transformer

Canare BNC Male -to- RCA Male 75-Ohm Coaxial Cable - 1m

 ...or, instead of this cable, you could use your preference of digital 75-ohm RCA Male-to-RCA Male cable with this adaptor:

BNC Male -to- RCA Female Adapter

 Thanks,
 Elias


----------



## ted betley

thanks works like a charm


----------



## omegared

Hi Elias, I have some synchronization issue with my DAC1 USB to my HTPC. Every now and then, my HTPC is having trouble synchronizing with my DAC1 USB. On my DAC1, I can see the LED blinking for awhile and later the lights are off. 
 I need to unplug and replug in the USB cable in order to make it work again. Is there anything I missed here? Any XP settings needed to be done? Does anyone have the same problem?


----------



## bralk

Yes - I have the same problem, feeding the dac1 usb from a Macbook G4 running Leopard and iTunes 8.

 cheers

 Tom


----------



## EliasGwinn

Quote:


  Originally Posted by *omegared* /img/forum/go_quote.gif 
_Hi Elias, I have some synchronization issue with my DAC1 USB to my HTPC. Every now and then, my HTPC is having trouble synchronizing with my DAC1 USB. On my DAC1, I can see the LED blinking for awhile and later the lights are off. 
 I need to unplug and replug in the USB cable in order to make it work again. Is there anything I missed here? Any XP settings needed to be done? Does anyone have the same problem?_

 

Thats odd...does it happen while you're playing audio?

 If I was to wager a guess, I would say that your power management is shutting down the USB ports. 

 How often does it happen?

 Thanks,
 Elias


----------



## EliasGwinn

Quote:


  Originally Posted by *bralk* /img/forum/go_quote.gif 
_Yes - I have the same problem, feeding the dac1 usb from a Macbook G4 running Leopard and iTunes 8.

 cheers

 Tom_

 

Tom,

 I'm also curious about your experience. When does this problem happen? Whil playing audio? While idle? After its been on for a long time? Not long after power-up?

 Thanks,
 Elias


----------



## omegared

Quote:


  Originally Posted by *EliasGwinn* /img/forum/go_quote.gif 
_Thats odd...does it happen while you're playing audio?

 If I was to wager a guess, I would say that your power management is shutting down the USB ports. 

 How often does it happen?

 Thanks,
 Elias_

 

It happens at random interval. Sometimes during playing audio, sometimes leaving my pc idle. I have ensure that the checkbox is unchecked. That means it "should not" power down when idle.


----------



## omegared

Quote:


  Originally Posted by *bralk* /img/forum/go_quote.gif 
_Yes - I have the same problem, feeding the dac1 usb from a Macbook G4 running Leopard and iTunes 8.

 cheers

 Tom_

 

Anyone else encounter the same problem?


----------



## bralk

Quote:


  Originally Posted by *EliasGwinn* /img/forum/go_quote.gif 
_Tom,

 I'm also curious about your experience. When does this problem happen? Whil playing audio? While idle? After its been on for a long time? Not long after power-up?

 Thanks,
 Elias_

 

Hi Elias

 The problem i not occurring regularly. If I play without changing playlists/tracks (using iPod Touch as a remote) there is no problem. But sometimes after having made some changing around between playlists the usb connection dies. When I check the audio/midi panel the Benchmark is still listed as active - but after this check the usb connection is reestablished. Just opening and closing the audio/midi panel solves the problem.

 The problem occurs perhaps once every second day.

 I will change to my macmini to see if it is a problem with my G4.

 I will aso try using a shorter usb cable ( now using 15 feet which is perhaps 
 too long ?)

 cheers

 Tom


----------



## Quaddy

this used to happen to me with my dac1 pre fairly frequently via USB when i was using a dell latitude x300 as source, and a G3 imac

 now i use a newer imac and a trick usb cable and it hasnt happened once, rock solid connection

 so rest assured its not the doing of the DAC1 pre, but the randomness and wide ranging computer environments we all have.

 ensure ALL usb hubs in device manager (view by connection off the dac1) are not allowed to power save as elias stated.

 and if you have vista, dig deep to disable the selective usb suspend settings.

 i also seem to remember running my wav files off an external USB hard drive wasnt helping matters, i switched to firewire external as well, and not had it since.


----------



## EliasGwinn

Quote:


  Originally Posted by *omegared* /img/forum/go_quote.gif 
_It happens at random interval. Sometimes during playing audio, sometimes leaving my pc idle. I have ensure that the checkbox is unchecked. That means it "should not" power down when idle._

 

Do you have another computer that you could try this on? I'm curious to see if your DAC1 works well with other computers.

 Thanks,
 Elias


----------



## omegared

Quote:


  Originally Posted by *EliasGwinn* /img/forum/go_quote.gif 
_Do you have another computer that you could try this on? I'm curious to see if your DAC1 works well with other computers.

 Thanks,
 Elias_

 

Unfortunately no. Let me try what Quaddy suggested first to ensure ALL usb root hub do not have power save enabled. Will update you again.


----------



## BitPerfect

Greetings DAC1ers and head-fiers,

 Before I ask my question, I'd like to add my voice to those praising Elias Gwinn's remarkable contributions throughout this looong thread. Like others, I've read through the whole thing and I've never seen such a high light/heat ratio in any prolonged forum discussion. Just exemplary. 
	

	
	
		
		

		
			





 OK, so I think I have too much gain in my system and I'm wondering what to do about it.

 I recently replaced a DAC1 with a DAC1 Pre, allowing me to eliminate both an M-Audio S/PDIF to USB box and a preamp (for vinyl listening) and get back to balanced connections between DAC1 Pre and my ATI 3002 power amp. Most of my listening is sourced from my music server (Apple Lossless files). 

 In my listening room, I typically source through a Squeezebox Duet through digital coax into the DAC1 or via USB from my Mac laptop when listening to 24/96 content.

 I have the DAC1 built-in pads at 30dB. In this setup, I find that if I turn off digital attenuation in the Duet and in iTunes on the laptop, I can't set the DAC1 volume above 9 or 10 detents for comfortable listening. My power amp has 28dB gain and my speakers (Spendor S8e) have 89dB sensitivity.

 Looking at the volume control graphs in the DAC1 manual, I'd much prefer to see the DAC1 volume control at the 12 o'clock position rather than 9:45. I could turn it up and digitally attenuate the Duet and/or iTunes, but I don't know how much dynamic range I might be losing that way.

 Does it make sense to consider adding a passive attenuator pads of some kind between the DAC1 and power amp?

 Thanks!


----------



## EliasGwinn

Quote:


  Originally Posted by *BitPerfect* /img/forum/go_quote.gif 
_...

 OK, so I think I have too much gain in my system and I'm wondering what to do about it.

 ...

 I have the DAC1 built-in pads at 30dB. In this setup, I find that if I turn off digital attenuation in the Duet and in iTunes on the laptop, I can't set the DAC1 volume above 9 or 10 detents for comfortable listening. My ATI 3002power amp has 28dB gain and my speakers (Spendor S8e) have 89dB sensitivity.

 ...

 Does it make sense to consider adding a passive attenuator pads of some kind between the DAC1 and power amp?_

 

Hello BitPerfect! Welcome to "The Thread"!

 If the volume control of your DAC1 PRE is set at the 10th detent (-15 dB gain), and the attenuators are set at 30 dB, there is a total of 45 dB of attenuation before your power amp. That means your amp is seeing a maximum input of -20 dBu (give or take). 

 If anything over -20 dBu is too loud for your amp, then your amp has too much gain / power. 

 Are you absolutely sure that you have all four XLR jumpers set at the -30 dB postion? The -30 dB postions in the DAC1 PRE are the jumpers closest to the XLR connectors.

 Thanks,
 Elias


----------



## BitPerfect

Quote:


  Originally Posted by *EliasGwinn* /img/forum/go_quote.gif 
_Are you absolutely sure that you have all four XLR jumpers set at the -30 dB postion? The -30 dB postions in the DAC1 PRE are the jumpers closest to the XLR connectors._

 

Yes, I'm sure. As a sanity check I popped the top and looked again the other day. But I appreciate the confirmation of the position. On my original DAC1 "classic" I seem to remember those jumper positions being marked, but not on the Pre, unless my eyesight is failing.

 I can live with it. The power amp delivers 300wpc into 8 ohms. That may be contributing to the issue. But I really like that effortless power when the time comes to crank it up a bit. 
	

	
	
		
		

		
		
	


	




 Most likely, I'll just brave the balance uncertainties down the curve of your volume control, because my ears tell me that the Duet's digital control is messing with the sound. I do wish that more consumer power amps (or any) offered input attenuation like some of the pro stuff.

 Thanks again for being such a help, Elias.


----------



## Scrith

Also on an gain-related note, I switched my DAC1 USB's attenuation to -30db (from the default -20db) awhile ago because I found that most of my listening was happening at around the 9 o'clock position on the DAC1's volume control. I now listen to it at around the 12 o'clock position, but something seems to be lacking (and I fully appreciate that this is merely a subjective test result). Could one hear a difference between -20db and -30db settings when adjusting for the same volume?

 I am using Jeff Rowland Model 201 monoblock amps (set to the default 26db gain).


----------



## EliasGwinn

Quote:


  Originally Posted by *BitPerfect* /img/forum/go_quote.gif 
_On my original DAC1 "classic" I seem to remember those jumper positions being marked, but not on the Pre, unless my eyesight is failing._

 

The marking is actually below the jumpers, just beside the XLR connector. It's tucked away...thats why I thought I'd ask you to double check...but it apparently you already had!

  Quote:


  Originally Posted by *BitPerfect* /img/forum/go_quote.gif 
_Most likely, I'll just brave the balance uncertainties down the curve of your volume control, because my ears tell me that the Duet's digital control is messing with the sound. I do wish that more consumer power amps (or any) offered input attenuation like some of the pro stuff._

 

Yeah, there is no reason an amp should be built with 28 dB of gain at the input stage. It's detrimental to signal-to-noise ratio. Proper gain-staging means sending the hottest signal possible from one device to the next, and using as little gain as possible. 

 I would suggest using a passive attenuator in your case. Try to use one that has as little output impedance as possible. 

 Thanks,
 Elias


----------



## Scrith

Well-known high-end audio manufacturer Ayre will soon be releasing a DAC supporting USB input that supports an asynchronous USB mode (via USB firmware licensed from Wavelength Audio, I believe). This asynchronous USB mode allows clock(s) on the DAC to determine the transfer rate of data from the computer, rather than relying on the computer's (inaccurate and jitter-prone) clock(s) to determine the rate at which data is received (and thereby necessitating resampling on the DAC side to overcome this poor timing). This new firmware is particularly interesting because it works with the existing Windows and Mac USB drivers.

 I mentioned this a few months ago and haven't heard much about it since then, so I'm wondering if there is any update. Does Benchmark have any plans for a new DAC (or an update to the firmware for existing DAC(s)) that supports this new variation on USB audio data transfer? This evolutionary step seems very important, because it finally eliminates all computer clock related jitter from the source (without providing a band-aid solution like resampling or intermediate buffering).


----------



## BitPerfect

Quote:


  Originally Posted by *EliasGwinn* /img/forum/go_quote.gif 
_I would suggest using a passive attenuator in your case. Try to use one that has as little output impedance as possible._

 

Cool, I'll give that a try then. I see Hosa and others have inline balanced attenuators that are switchable to 20, 30, and 40 dB. Means I'd need to buy two of them. That feeds perfectly into my finely tuned neurotic-audio-obsessive personality. I can lose at least two nights sleep wondering if they're perfectly matched. 

 No mention of output impedance on most of them, but the amp has 28 kOhm nominal input impedance, and the cables are short, so I should be alright. I'll probably only lose one night's sleep over that one. 

 Thanks!


----------



## EliasGwinn

Scrith,

 Regarding the -30 dB attenuation setting, the sound should not change if the final output level is the same. The attenuators are purely passive, just before the output. The only change is the output impedance (lower output impedance is better), but the -30 dB is the lowest output impedance setting (43 ohms) of them all. So, I don't know why you are hearing something going missing when you go from -20 to -30 dB.

  Quote:


  Originally Posted by *Scrith* /img/forum/go_quote.gif 
_Well-known high-end audio manufacturer Ayre will soon be releasing a DAC supporting USB input that supports an asynchronous USB mode (via USB firmware licensed from Wavelength Audio, I believe). This asynchronous USB mode allows clock(s) on the DAC to determine the transfer rate of data from the computer, rather than relying on the computer's (inaccurate and jitter-prone) clock(s) to determine the rate at which data is received (and thereby necessitating resampling on the DAC side to overcome this poor timing). This new firmware is particularly interesting because it works with the existing Windows and Mac USB drivers._

 

The problem with this implementation is that it requires the operating system to re-sample the audio to the clock of the converter. This is a problem because the re-sampling algorithm that Windows and Mac uses is absolute garbage. 

 Even if the master clock from the outboard DAC tells the computer to play at the same sample rate (44.1k for a CD, for example), the computer still must re-sample the data to be synchronized to the DAC's master clock. 

 So, it is true that the audio is transferred using the DAC's clock, but it is at a very costly tradeoff. For outboard DAC's that struggle with jitter, it may be a worthwhile tradeoff.

  Quote:


  Originally Posted by *Scrith* /img/forum/go_quote.gif 
_I mentioned this a few months ago and haven't heard much about it since then, so I'm wondering if there is any update. Does Benchmark have any plans for a new DAC (or an update to the firmware for existing DAC(s)) that supports this new variation on USB audio data transfer? This evolutionary step seems very important, because it finally eliminates all computer clock related jitter from the source (without providing a band-aid solution like resampling or intermediate buffering)._

 

This USB mode (asynchronous) is not new. When we designed our USB interface, we made the decision to not use this mode. This is because, as I mentioned, it requires the computer to re-sample the data. Therefore, with our method, the computer can simply pass the data to the DAC1 bit-transparently. 

 Is there a lot of jitter on the data coming from the computer using our solution? There sure is! But, luckily, the DAC1 can handle any amount of jitter with no loss in sonic quality. In fact, the performance of the DAC1 using the USB from a computer is equivalent to that of the DAC1 using a high-end transport with a high-quality 75-ohm coaxial cable.

 The DAC1's USB solution is the best of both worlds...it eliminates the horrible OS re-sampling and it is immune to the computer's jitter!

 Thanks,
 Elias


----------



## EliasGwinn

Quote:


  Originally Posted by *BitPerfect* /img/forum/go_quote.gif 
_Cool, I'll give that a try then. I see Hosa and others have inline balanced attenuators that are switchable to 20, 30, and 40 dB. Means I'd need to buy two of them. That feeds perfectly into my finely tuned neurotic-audio-obsessive personality. I can lose at least two nights sleep wondering if they're perfectly matched. 

 No mention of output impedance on most of them, but the amp has 28 kOhm nominal input impedance, and the cables are short, so I should be alright. I'll probably only lose one night's sleep over that one. 

 Thanks!_

 

If you have that dream about the interconnects tying your feet and hands together, and the optical cable starts shining its light at you, DON'T LOOK INTO THE LIGHT!!


----------



## emmodad

Quote:


  Originally Posted by *BitPerfect* /img/forum/go_quote.gif 
_Cool, I'll give that a try then. I see Hosa and others have inline balanced attenuators that are switchable to 20, 30, and 40 dB. Means I'd need to buy two of them. That feeds perfectly into my finely tuned neurotic-audio-obsessive personality. I can lose at least two nights sleep wondering if they're perfectly matched. 

 No mention of output impedance on most of them, but the amp has 28 kOhm nominal input impedance, and the cables are short, so I should be alright. I'll probably only lose one night's sleep over that one. 

 Thanks!_

 

small caution: most all fixed or adjustible XLR-style "in-line attenuators" you find (such as the Hosa, IIRC) are designed for mic lines ie they are designed to be used in low-impedance signal chains. Only fixed ones I've seen that are designed for audio line-level are from rothwell in the UK (google on "rothwell attenuator"). Elias may know of something else.

 Other options would be to follow the DAC-1 by a passive stereo attenuator of reasonable quality and good tracking, so you can set it and forget it; or a pair of high-quality stepped mono adjustibles.

 For the passive stereo, it gets expensive in the "audiophile" realm (Placette, Goldpoint level controls, other similar designs from Europe); or a Presonus Central Station is a good bang-for-the-buck solution for simple, clean high-bandwidth passive level control, using the TRS main inputs (the RCA inputs go through some active circuitry to do -10 to +4 level shifting). You can find them in good seasonal sales from folk like Musician's Friend for say USD 300-400 or so. Of course, YMMV concerning your connections single-ended, balanced, etc. Another inexpensive solution from the pro space is an a-designs ATTY passive stereo attenuator, IIRC USD 100 or so.

 With any of those, you can either use DAC-1 in fixed output mode and then control volume with the inserted atteniuator; or use the DAC-1 vol control with the attenuator following in set&forget. Elias may have comments about gain staging, depening on the gain of your particular amps....

 tweakaudio Ultimate Attenuators: a small audiophile and recording business in Boulder Creek, CA builds amongst other things these excellent (and not cheap, $350?/pair, but very clean) mono attenuators that are essentially high-quality resistors and stepped pots soldered directly together with RCA or XLR male and female connectors. they're tiny, beautifully made, and install directly at the input jacks of your amps. A pair of these would quite possibly appeal to a finely tuned neurotic-audio-obsessive personality...


----------



## BitPerfect

Quote:


  Originally Posted by *EliasGwinn* /img/forum/go_quote.gif 
_If you have that dream about the interconnects tying your feet and hands together, and the optical cable starts shining its light at you, DON'T LOOK INTO THE LIGHT!!_

 

Dude! You have that dream too?


----------



## BitPerfect

Quote:


  Originally Posted by *emmodad* /img/forum/go_quote.gif 
_small caution: most all fixed or adjustible XLR-style "in-line attenuators" you find (such as the Hosa, IIRC) are designed for mic lines ie they are designed to be used in low-impedance signal chains. Only fixed ones I've seen that are designed for audio line-level are from rothwell in the UK (google on "rothwell attenuator"). Elias may know of something else.

 Other options would be to follow the DAC-1 by a passive stereo attenuator of reasonable quality and good tracking, so you can set it and forget it; or a pair of high-quality stepped mono adjustibles.
 ..._

 

Thank you for your time and generosity in detailing this. You've undoubtedly saved me from making a mistake.

 Your mention of the the tweakaudio nude ultimate attenuator reminded me that I had sent something about that to an Audiogoner friend a couple of years ago. I had completely forgotten about it. An email search shows he subsequently bought an NHT PVC Pro for his small office system, which at around $90 might be another candidate for balanced VC. They also have an unbalanced unit with Jensen transformers.

 I will do some research on all your suggestions. Again thanks!


----------



## G-U-E-S-T

Regarding passive attenuators that attach to your amp, please consider Scott Endler's Shotgun Attenuators. They are absolutely the best passive solution I've heard to date. I strongly recommend avoiding the rothwell attenuators - they are the worst I've ever heard and they sound really bad in comparison, although I can't explain why.


----------



## BitPerfect

Quote:


  Originally Posted by *G-U-E-S-T* /img/forum/go_quote.gif 
_Regarding passive attenuators that attach to your amp, please consider Scott Endler's Shotgun Attenuators. They are absolutely the best passive solution I've heard to date. I strongly recommend avoiding the rothwell attenuators - they are the worst I've ever heard and they sound really bad in comparison, although I can't explain why._

 

Helpful! I had not seen these. I like that there is an XLR version. I'm in favor of first trying the simplest thing that could possibly work. This is pretty darn simple (as are the rothwells for that matter). Appreciate the feedback.


----------



## emmodad

you're welcome. funny, I had thought of the NHT but forgot to mention it, also seems to get good comments.

 do google on the Ultimate Attenuators, they may not be the correct solution for you but the pictures are quite something to see....

 glad to be of assistance to someone from the home playground of MythBusters 
	

	
	
		
		

		
			





 and those shotguns look potentially cost-effective too..


----------



## Mazz

Quote:


  Originally Posted by *EliasGwinn* /img/forum/go_quote.gif 
_The problem with this implementation is that it requires the operating system to re-sample the audio to the clock of the converter. This is a problem because the re-sampling algorithm that Windows and Mac uses is absolute garbage. 

 Even if the master clock from the outboard DAC tells the computer to play at the same sample rate (44.1k for a CD, for example), the computer still must re-sample the data to be synchronized to the DAC's master clock. _

 

Elias, could you clarify this surprising statement? Are you implying that it's limitation of the USB audio protocol spec, or of the default USB audio drivers in Windows and the Mac that forces resampling by the OS, or something else entirely? 

 I had a (quick) look at the USB 2.0 Audio spec (although didn't look at the 1.x version) and didn't see any obvious indications that you can't have the computer specify a sample rate that should be used by the DAC's clock in async mode. If that's correct, that suggests you _should _be able to have the computer stream a bit-perfect stream to the DAC _provided _the DAC can provide a master clock that matches the sampling rate for the audio stream. (Building the hardware implementation is another matter - I remember reading no standard USB chipsets handled it in the past...)

 So if it's true that the USB spec can handle it, that suggests that the default USB audio drivers don't do it. And that means you can only do this with custom drivers (or proper resampling code in your audio player), both of which are certainly feasible, although at the cost of complicating the development - and use of - the product.


----------



## omegared

Quote:


  Originally Posted by *omegared* /img/forum/go_quote.gif 
_Unfortunately no. Let me try what Quaddy suggested first to ensure ALL usb root hub do not have power save enabled. Will update you again._

 

Hi Elias. My sychronisation problem is resolved! I left the system playing over several nite and allowing it to idle during this period. I'm still able to play music thru DAC1 without any problem! Apparently, like Quaddy adviced, you need to unplug the usb device before going to system devices to disable usb power down when idle. If not, the settings will reset itself upon system shutdown / reboot.


----------



## Scrith

Regarding asynchronous USB support, here is a quote from the Wavelength Audio page:

 "The Firmware that runs the USB controller (TAS1020B) of the product is located on the DAC module and therefore changes the way the DAC and the computer talk to each other. Settings like bit size (i.e. 16 or 24 bits) and sample rate (i.e. 44.1K, 48K, 88.2K and 96K) will be communicated to the computer. All the DAC's on this page use Asynchronous USB mode."

 This seems to indicate that the DAC is telling the computer which format to use for the data it sends the DAC (16/24 bits, 44/48/88/96K), as Elias said such a system might.

 I am a bit confused by Elias' statement that the computer will always resample to the requested rate, even if the data is already at that rate. Are you sure about this, Elias? If this is not the case (and it certainly seems unlikely that a computer would resample in this case), then the asynchronous method would appear to have a major advantage (the DAC is controlling the rate at which it receives data).

 I appreciate that the DAC1 is "the best of both worlds" but I think the best option for a DAC would be to have no resampling whatsoever, with a DAC that is controlling the rate at which it is receiving data from the source and using some internal buffering to overcome any latency in the arrival of the requested data between the source and the DAC.


----------



## EliasGwinn

Quote:


  Originally Posted by *Mazz* /img/forum/go_quote.gif 
_Elias, could you clarify this surprising statement? Are you implying that it's limitation of the USB audio protocol spec, or of the default USB audio drivers in Windows and the Mac that forces resampling by the OS, or something else entirely? 

 I had a (quick) look at the USB 2.0 Audio spec (although didn't look at the 1.x version) and didn't see any obvious indications that you can't have the computer specify a sample rate that should be used by the DAC's clock in async mode. If that's correct, that suggests you should be able to have the computer stream a bit-perfect stream to the DAC provided the DAC can provide a master clock that matches the sampling rate for the audio stream. (Building the hardware implementation is another matter - I remember reading no standard USB chipsets handled it in the past...)

 So if it's true that the USB spec can handle it, that suggests that the default USB audio drivers don't do it. And that means you can only do this with custom drivers (or proper resampling code in your audio player), both of which are certainly feasible, although at the cost of complicating the development - and use of - the product._

 

The problem stems from the fact that no two clocks are ever equal. Even if they are both said to be at 44.1 kHz, one may actually be 44100.01 Hz and the other may be at 44099.99 Hz. 

 This is why recording studios use a master clock. Even if they set all their A-to-D's to 44.1 kHz, the samples would not be truly synchronized. This results in pops and clicks.

 In a USB playback system, unless the driver and the media software are directly linked, the software operates using an internal clock. This data stream is then sent to the audio stack in the operating system. 

 At this point, the OS determines how to stream the data to the peripheral device (the USB audio device). If the OS is receiving instructions to stream it at a specific sample rate (e.g., the DAC's master clock), then the OS must re-sample it to match that rate. On the other hand, if the OS receives instructions to stream it using the original clock, then no re-samplign occurs.

 Thanks,
 Elias


----------



## EliasGwinn

Quote:


  Originally Posted by *omegared* /img/forum/go_quote.gif 
_Hi Elias. My sychronisation problem is resolved! I left the system playing over several nite and allowing it to idle during this period. I'm still able to play music thru DAC1 without any problem! Apparently, like Quaddy adviced, you need to unplug the usb device before going to system devices to disable usb power down when idle. If not, the settings will reset itself upon system shutdown / reboot. 
	

	
	
		
		

		
		
	


	


_

 

Great to hear! I'm sure this will also prove valuable to other people who's computers are cutting off the USB connection.

 Thanks,
 Elias


----------



## EliasGwinn

Quote:


  Originally Posted by *Scrith* /img/forum/go_quote.gif 
_I am a bit confused by Elias' statement that the computer will always resample to the requested rate, even if the data is already at that rate. Are you sure about this, Elias?_

 

Please read the post I wrote a few minutes ago concerning this. Basically, its because the sample rates will never be the same, and there must be a master clock, to which all others are sampled to.

  Quote:


  Originally Posted by *Scrith* /img/forum/go_quote.gif 
_If this is not the case (and it certainly seems unlikely that a computer would resample in this case), then the asynchronous method would appear to have a major advantage (the DAC is controlling the rate at which it receives data)._

 

This is questionable. I can understand why one might think that it would be advantageous to have the DAC control the stream rate. But, keep in mind, the computer is still the device that is pushing the data out. In other words, the DAC isn't "pulling" the data out by itself. It is merely specifying the rate. The computer still has to push the data (at the DAC's rate), and so we don't really bypass the computer's imperfections, let alone the USB cable, and the EMI environment that surrounds a computer.

 So, even if the computer did not re-sample, and the device was receiving bit-transparent (or otherwise high-precision) audio, the asynchronous method still may not have less jitter.

  Quote:


  Originally Posted by *Scrith* /img/forum/go_quote.gif 
_I appreciate that the DAC1 is "the best of both worlds" but I think the best option for a DAC would be to have no resampling whatsoever, with a DAC that is controlling the rate at which it is receiving data from the source and using some internal buffering to overcome any latency in the arrival of the requested data between the source and the DAC._

 

Ideally, a digital stream would never need to be resampled before conversion. However, in reality, jitter will always be present when data is being transmitted between devices, and so there is a need to address jitter. Benchmark's jitter immunity is a result of the data being re-sampled asynchronously (not based on the incoming clock). This re-sampling is done on-board using a high-resolution ASRC chip (AD1896) millimeters away from the D/A chip.

 Our philosophy is to avoid any processing in the computer, do the hard work in our box where we can control the quality of the processing.

 Thanks,
 Elias


----------



## G-U-E-S-T

Hi Elias,

 I have a question please about the analog inputs on my DAC1 PRE. Do they have any provision at all for interrupting any potential ground loop current or other EMI/RFI flowing through the interconnect cable shield? 

 I like very much how the digital inputs are isolated, but I worry about the possible lack of decoupling/isolation on the analog rca inputs. As you know, this can be a problem when interfacing some single-ended equipment - and I would guess (as a layman) that any possible inlet for EMI/RFI (even the analog ins) could possibly also affect the internal DA circuitry as well.

 Thanks in advance for your reply!


----------



## Mazz

Quote:


  Originally Posted by *EliasGwinn* /img/forum/go_quote.gif 
_In a USB playback system, unless the driver and the media software are directly linked, the software operates using an internal clock. This data stream is then sent to the audio stack in the operating system. _

 

Ah, yes, I see what you're saying. If I could rephrase it, in order to have only one clock in the playback data path the audio player also needs to respond to the DAC's requests to speed up/slow down data transmission, otherwise you have two clocks and risk buffer under-runs/over-runs somewhere in the system. And we don't know if any media players support that mode (and the default OS drivers should provide the right APIs for the media players to use if you also want to avoid custom drivers and I don't know if they do either).

  Quote:


  Originally Posted by *EliasGwinn* 
_So, even if the computer did not re-sample, and the device was receiving bit-transparent (or otherwise high-precision) audio, the asynchronous method still may not have less jitter._

 

I don't understand why this might be the case. Under those circumstances there is only one data clock _that affects audio output_ _timing_, and thus the jitter due to computer generation and data transmission in other methods disappears. Yes, you still have to arrange your software and communications stacks to avoid buffer under-runs and over-runs, but you're not getting jitter _due to data transmission between computer and DAC, nor due to the computer's clock_. True, you still have other sources of jitter - e.g. due to the DAC's clock - but you have that under all possible circumstances anyway. 

 Are you saying this method might have more jitter than others because it's DAC clock implementation has more than an alternative method? If so, this is an attribute of the implementation, not of the method. Or do you have some attribute of the method in mind?

 I think it's also worth pointing out that - were there to be truly only one clock that affected data output - that on the computer _sample rate conversion _would be required rather than resampling. The sample rate conversion algorithm (presuming a Nyquist limited original sampling process and good quality ADC clock, as for any good recording) is unique, entirely deterministic and independent of the quality of the computer's clock, so if implemented properly wouldn't introduce any errors. But it sounds like we don't have a truly one clock world today so it doesn't really matter...


----------



## Scrith

Hmm, this is definitely a thought-provoking topic (the downfalls of asynchronous USB support in a DAC). I wonder if the Wavelength and Ayre people have really considered what this means on the computer end.

 Meanwhile, as a long-time programmer, I think this problem is being constrained by the hardware that is available to the DAC. I think a much easier approach to this jitter problem (rather than resampling to work around it) would be to create a buffer (perhaps 1 second worth of audio at the current playback rate) that contains not only the data but information (for each 16-bit audio data word) about the time since the previous piece of data was received. The playback code could then sum all these times and base the current playback rate on the average time between each piece of data. This would reduce jitter to ridiculously low levels (if the buffer had 1000 words, the jitter would be reduced to 1/1000 of the level at which it was originally received). This approach would prevent buffer over- and under-runs as well, since you are using data at the "average" rate at which it arrived.


----------



## EliasGwinn

Quote:


  Originally Posted by *G-U-E-S-T* /img/forum/go_quote.gif 
_Hi Elias,

 I have a question please about the analog inputs on my DAC1 PRE. Do they have any provision at all for interrupting any potential ground loop current or other EMI/RFI flowing through the interconnect cable shield? 

 I like very much how the digital inputs are isolated, but I worry about the possible lack of decoupling/isolation on the analog rca inputs. As you know, this can be a problem when interfacing some single-ended equipment - and I would guess (as a layman) that any possible inlet for EMI/RFI (even the analog ins) could possibly also affect the internal DA circuitry as well.

 Thanks in advance for your reply!_

 

G-U-E-S-T,

 The shield of the RCA connector is bonded to the chassis (earth), which is isolated from the signal grounds, so it can't get into the analog OR digital signal ground. 

 Thanks,
 Elias


----------



## clarke68

Quote:


  Originally Posted by *Scrith* /img/forum/go_quote.gif 
_I think a much easier approach to this jitter problem (rather than resampling to work around it) would be to create a buffer (perhaps 1 second worth of audio at the current playback rate) that contains not only the data but information (for each 16-bit audio data word) about the time since the previous piece of data was received._

 

Isn't that how the Genesis Digital Lens worked?


----------



## nae45ro

Hey Elias,

 I've just ordered the Benchmark and I have several questions if you have the time 

 The DAC will be connected as follows :

 1) PC > DAC-1 > Bada ph-12 amp > Beyerdynamic DT-880 headphones
 2) Cambridge Audio 640C > DAC-1 > Vincent SV-226 amp > Chario Academy millenium 2 speakers

 I will therefore use 2 inputs and 2 outputs. The questions would be :

 1) What inputs/outputs would you use in the 2 configs ? (ex : config 1 > input 4 with output RCA, config 2 ...)
 2) What cables would you use (how dependent on the quality of the cables is each input/output) ?
 3) Any other recommendation regarding positions of pins inside the unit or anything else ?
 4) Recommended players for PC and their config ? (I now use Foobar 2000)

 PS : I will only use DAC-1 for listening to music in wav, flac and ape !

 Thanks !


----------



## EliasGwinn

Quote:


  Originally Posted by *Mazz* /img/forum/go_quote.gif 
_Ah, yes, I see what you're saying. If I could rephrase it, in order to have only one clock in the playback data path the audio player also needs to respond to the DAC's requests to speed up/slow down data transmission, otherwise you have two clocks and risk buffer under-runs/over-runs somewhere in the system. And we don't know if any media players support that mode (and the default OS drivers should provide the right APIs for the media players to use if you also want to avoid custom drivers and I don't know if they do either)._

 

Right. In other words, the media player and all software processes in-between are operating at a rate dictated by the computer. A USB audio device is not going to dictate that rate for that whole stack without custom drivers and agreeable software. And even then, the only way it would be driven directly from the DAC's master clock is if there were a copper trace from the DAC's clock to the CPU clock input....not likely.

 I suppose this doesn't mean that sample-rate conversion is inevitable. Thereotically, it could fill buffers at the USB port, and transmit the data using a different clock. But, to avoid sample-rate conversion, the control clock has to know what the original sample-rate of the audio was and control the flow at the original sample rate.

  Quote:


  Originally Posted by *Mazz* /img/forum/go_quote.gif 
_Under those circumstances there is only one data clock that affects audio output timing, and thus the jitter due to computer generation and data transmission in other methods disappears. Yes, you still have to arrange your software and communications stacks to avoid buffer under-runs and over-runs, but you're not getting jitter due to data transmission between computer and DAC, nor due to the computer's clock. True, you still have other sources of jitter - e.g. due to the DAC's clock - but you have that under all possible circumstances anyway. 

 Are you saying this method might have more jitter than others because it's DAC clock implementation has more than an alternative method? If so, this is an attribute of the implementation, not of the method. Or do you have some attribute of the method in mind?_

 

There will be many sources of jitter in asynchronous mode. 

 1. First of all, the DAC's master clock is not the clock that controls the host. The DAC's master clock is sent to the TAS1020B, which uses a PLL to estimate and regenerate the master clock. The TAS1020B is the _real _master clock coming down the USB cable, and it is not a low-jitter crystal oscillator. I haven't measured the jitter on that clock, but it isn't designed to be a low jitter source. It doesn't even specify a jitter-spec on the data sheet.

 2. As the TAS1020B sends a master (L/R) clock 'tick' (which tells the USB port to send a frame), the impedance and capacitance of the USB cable/connections will add jitter to this master clock signal.

 3. When the host (computer) receives the clock signal, it must apply another PLL-driven clock to estimate the received clock. The new PLL-driven clock is what pushes the data out the USB port.

 4. Then, this data must travel along the impedance/capacitance-ridden USB cable/connectors to get to the DAC. 

 5. Immediately inside the DAC, the data goes through the TAS1020B before it is delivered to the D/A chip.

 As you can see, it isn't as simple as the DAC reaching into the computer and grabbing samples at will. On the surface, it may seem that the DAC's master clock would be best, but it doesn't really get you anywhere. The real reason that asynchronous mode exists is so that a device can serve as master clock for multiple synchronized devices (ADC's, etc).

  Quote:


  Originally Posted by *Mazz* /img/forum/go_quote.gif 
_I think it's also worth pointing out that - were there to be truly only one clock that affected data output - that on the computer sample rate conversion would be required rather than resampling. The sample rate conversion algorithm (presuming a Nyquist limited original sampling process and good quality ADC clock, as for any good recording) is unique, entirely deterministic and independent of the quality of the computer's clock, so if implemented properly wouldn't introduce any errors. But it sounds like we don't have a truly one clock world today so it doesn't really matter..._

 

I would say that "sample-rate conversion" is a type of re-sampling. It is true that this process could be done 'neatly' in a computer...in fact, Vista does it very neatly. However, XP and OSX do it very poorly.

 Thanks,
 Elias


----------



## gevorg

Hi Elias,

 For the DAC1 USB, what is the V and mA output of the HPA2 headamp?

 I'm considering to use low impedance and high current headphones with DAC1 USB, so just wanted to compare HPA2's specs to standalone amps.

 Thank you!


----------



## Mazz

Quote:


  Originally Posted by *Scrith* /img/forum/go_quote.gif 
_ The playback code could then sum all these times and base the current playback rate on the average time between each piece of data. _

 

I assume you're referring to the playback code in the DAC, not on the computer? If so, you're basically saying the DAC has to do clock estimation, which all of them do, and that it can use some form of averaging in doing so in order to improve the clock frequency estimate (whilst still tracking slow-moving variations in the incoming clock frequency), which I think almost all of them do already. They just tend to talk about it in different terms - PLLs, low pass filters, etc.


----------



## Mazz

Quote:


  Originally Posted by *EliasGwinn* /img/forum/go_quote.gif 
_I suppose this doesn't mean that sample-rate conversion is inevitable. Thereotically, it could fill buffers at the USB port, and transmit the data using a different clock. But, to avoid sample-rate conversion, the control clock has to know what the original sample-rate of the audio was and control the flow at the original sample rate._

 

Yes, that was what I was getting at. But as you pointed out, the software stack today doesn't seem to support that method.

  Quote:


  Originally Posted by *EliasGwinn* /img/forum/go_quote.gif 
_In other words, the media player and all software processes in-between are operating at a rate dictated by the computer. A USB audio device is not going to dictate that rate for that whole stack without custom drivers and agreeable software._

 

Agreed.

  Quote:


  Originally Posted by *EliasGwinn* /img/forum/go_quote.gif 
_And even then, the only way it would be driven directly from the DAC's master clock is if there were a copper trace from the DAC's clock to the CPU clock input....not likely._

 

This is sufficient, but not necessary (nor likely, as you point out 
	

	
	
		
		

		
			





). Instead you merely need the DAC to be the rate controller for data transmission, and you need the computer software stack to be sufficiently responsive, which isn't too hard to arrange under most circumstances these days. Remember the old dial-up modem flow control protocols which solved a remarkably similar problem? You can either have the DAC do high-watermark/low-watermark flow control (e.g. telling the computer "send a little bit faster than you are" or "a little bit slower" or even "stop sending until I tell you to start again"), or you can have it say something like ("send me the next chunk of audio in the next X milliseconds please")...

  Quote:


  Originally Posted by *EliasGwinn* /img/forum/go_quote.gif 
_There will be many sources of jitter in asynchronous mode. 

 1. First of all, the DAC's master clock is not the clock that controls the host. The DAC's master clock is sent to the TAS1020B, which uses a PLL to estimate and regenerate the master clock. The TAS1020B is the real master clock coming down the USB cable, and it is not a low-jitter crystal oscillator. I haven't measured the jitter on that clock, but it isn't designed to be a low jitter source. It doesn't even specify a jitter-spec on the data sheet.
 [...]_

 

Ah, thanks for this elaboration, but I see we misunderstood each other - which puts your original comments in a different light and probably answers the question I posed in response. I assumed you were talking about (possibly hypothetical) scenarios where the external DAC incorporates the master clock (plus some flow control protocol, as suggested further above). You're writing about cases where the master clock is derived instead from the USB transmission. 

 Thanks for a very interesting discussion!


----------



## EliasGwinn

Quote:


  Originally Posted by *Scrith* /img/forum/go_quote.gif 
_Meanwhile, as a long-time programmer, I think this problem is being constrained by the hardware that is available to the DAC. I think a much easier approach to this jitter problem (rather than resampling to work around it) would be to create a buffer (perhaps 1 second worth of audio at the current playback rate) that contains not only the data but information (for each 16-bit audio data word) about the time since the previous piece of data was received. The playback code could then sum all these times and base the current playback rate on the average time between each piece of data. This would reduce jitter to ridiculously low levels (if the buffer had 1000 words, the jitter would be reduced to 1/1000 of the level at which it was originally received). This approach would prevent buffer over- and under-runs as well, since you are using data at the "average" rate at which it arrived._

 

Hey Scrith,

 We could employ this method...the technology does exist. You'd be interested in reading this paper.

 However, there is more to consider here...there are _major_ benefits of employing an asynchronous sample-rate converter (ASRC). Specifically, it optimizes filter performance. 

 The most critical element of digital conversion (aside from jitter attenuation) is low-pass (stop-band) filtering. Without a well designed filter system, the audio will have aliasing, non-linear frequency response, and high-frequency attenuation. 

 Here's what the ASRC does for us:

 1. The digital filters in DAC-chips suffer from the fact that they are sharing a die with analog circuitry. It is difficult to optimize both the digital and analog performance of a single chip. The ASRC (asynchronous sample rate converter) in the DAC1 is a dedicated digital DSP environment, where all resources are optimized towards the heavy-math required to build a high quality filter.

 2. By converting the sample rate to 110 kHz, we are optimizing the filter of the DAC chip. As I mentioned before, DAC filters are inherently compromised because they are hybrid digital/analog chips. It requires a lot of DSP horse-power to create a digital filter that works well. So, by pushing the new nyquist frequency out to 110 kHz, we reduce the risk of the DAC's filter causing aliasing of ultra-sonic audio and/or attenuating high-freq audio. Also, any inaccuracies in the filter are far removed from the audio band. 

 We could have implemented a jitter-reduction system that did not convert the sample rate. However, the ASRC method creates harmony amongst all the components.

 When we design a product, we look at the strengths and weaknesses of all components available and design the most complimentary combination possible. From the outside, it may seem convoluted, but the results are what matter in the end.

 Thanks,
 Elias


----------



## EliasGwinn

Quote:


  Originally Posted by *nae45ro* /img/forum/go_quote.gif 
_Hey Elias,

 I've just ordered the Benchmark and I have several questions if you have the time 

 The DAC will be connected as follows :

 1) PC > DAC-1 > Bada ph-12 amp > Beyerdynamic DT-880 headphones
 2) Cambridge Audio 640C > DAC-1 > Vincent SV-226 amp > Chario Academy millenium 2 speakers

 I will therefore use 2 inputs and 2 outputs. The questions would be :

 1) What inputs/outputs would you use in the 2 configs ? (ex : config 1 > input 4 with output RCA, config 2 ...)
 2) What cables would you use (how dependent on the quality of the cables is each input/output) ?
 3) Any other recommendation regarding positions of pins inside the unit or anything else ?
 4) Recommended players for PC and their config ? (I now use Foobar 2000)

 PS : I will only use DAC-1 for listening to music in wav, flac and ape !

 Thanks !_

 

Nea45ro,

 I'm sorry its taken so long to respond. I've been buried in phone calls, emails, etc. 

 As far as digital inputs and digital cables, you don't have to worry about it too much. The performance of the DAC1 is consistent regardless of connection type. I see the 640C has both optical and coaxial. Feel free to use either one.

 Analog cables are a different story. A good quality starquad XLR cable is a great interconnect.

 I suggest using the balanced XLR cables for the setup that will require the longest cable run. I would also suggest using XLR Y-adaptors so that you can connect both devices with XLR balanced cables. Don't worry about over-loading the output of the DAC1. It can handle both devices simultaneously.

 Regarding pins inside the unit, I would suggest not changing anything until it is apparent that you need to. The only thing you may want to change is the output attenuators, but only if the output is too high or too low for your system.

 As for media player, foobar is great. If you enjoy using foobar, don't worry about changing a thing. Here is an article I wrote about setting up foobar...

Foobar2000 for Windows - Setup Guide - Benchmark

 Thanks,
 Elias


----------



## EliasGwinn

Quote:


  Originally Posted by *gevorg* /img/forum/go_quote.gif 
_Hi Elias,

 For the DAC1 USB, what is the V and mA output of the HPA2 headamp?

 I'm considering to use low impedance and high current headphones with DAC1 USB, so just wanted to compare HPA2's specs to standalone amps.

 Thank you!_

 

gevorg,

 Max Vrms = 8.7 Vrms
 Max Vrms w/ 10 dB gain reduction jumper = 2.7 Vrms
 Max I = 250 mA

 More info on page 44: http://www.benchmarkmedia.com/manual...nual_Rev_C.pdf

 Thanks,
 Elias


----------



## nae45ro

Cool, thanks for the answers. Distance between DAC and both amp (speaker amp and headphone amp) is about 0.5m !


----------



## Scrith

Quote:


  Originally Posted by *EliasGwinn* /img/forum/go_quote.gif 
_We could employ this method...the technology does exist. You'd be interested in reading this paper._

 

Very interesting. Yes, this is more technical description of what I was describing...use a buffer to generate an average clock rate for a large sample of data, then base the output rate on that, thereby reducing outgoing jitter to a very small fraction of the incoming jitter without having to worry about buffer over- and under-runs. And no resampling is required.

 This paper is from late 2006...where is the Benchmark DAC2 that will be based on it?!?


----------



## HeadLover

Quote:


  Originally Posted by *Scrith* /img/forum/go_quote.gif 
_Very interesting. Yes, this is more technical description of what I was describing...use a buffer to generate an average clock rate for a large sample of data, then base the output rate on that, thereby reducing outgoing jitter to a very small fraction of the incoming jitter without having to worry about buffer over- and under-runs. And no resampling is required.

 This paper is from late 2006...where is the Benchmark DAC2 that will be based on it?!? 
	

	
	
		
		

		
		
	


	


_

 

I am really thinking of getting the DAC1, but if there is going to be DAC2, I guess it will be worth waiting, not?


----------



## john11f

Hi Elias, I'm not into the technical aspect of these equipment but I own a Benchmark DAC1 Pre and was wondering which headphones did you (or the co.) think would be a good match for the headphone amp?


----------



## EliasGwinn

Quote:


  Originally Posted by *nae45ro* /img/forum/go_quote.gif 
_Cool, thanks for the answers. Distance between DAC and both amp (speaker amp and headphone amp) is about 0.5m !_

 

No problem! 

 You should be fine using XLR or RCA with your setup. XLR is ideal for noise performance, but for 0.5m, you shouldn't have much problem with RCA.

 Thanks,
 Elias


----------



## EliasGwinn

Quote:


  Originally Posted by *Scrith* /img/forum/go_quote.gif 
_Very interesting. Yes, this is more technical description of what I was describing...use a buffer to generate an average clock rate for a large sample of data, then base the output rate on that, thereby reducing outgoing jitter to a very small fraction of the incoming jitter without having to worry about buffer over- and under-runs. And no resampling is required.

 This paper is from late 2006...where is the Benchmark DAC2 that will be based on it?!? 
	

	
	
		
		

		
			



_

 

Scrith,

 We already use this topology, except that we intentionally choose to convert the sample-rate. As I mentioned in a previous post, it allows the ASRC chip to perform the critical filtering, and creates optimal conditions for the filter of the DAC chip.

 In other words, we accept the -133 dB THD+N artifacts of the ASRC versus the much, much worse artifacts from filter aliasing and non-linearity.

 We have no plans to modify our approach until we find a system that results in better performance. That won't change soon because the Benchmark engineers (John Siau and myself) can easily beat up the Benchmark marketing person. 
	

	
	
		
		

		
		
	


	




 Thanks,
 Elias


----------



## EliasGwinn

Quote:


  Originally Posted by *HeadLover* /img/forum/go_quote.gif 
_I am really thinking of getting the DAC1, but if there is going to be DAC2, I guess it will be worth waiting, not?



_

 

There are no plans for a DAC2. Sorry...

 Thanks,
 Elias


----------



## EliasGwinn

Quote:


  Originally Posted by *john11f* /img/forum/go_quote.gif 
_Hi Elias, I'm not into the technical aspect of these equipment but I own a Benchmark DAC1 Pre and was wondering which headphones did you (or the co.) think would be a good match for the headphone amp?_

 

Hello John,

 Finding headphones that you like is very much a personal, subjective choice. 

 However, John Siau and I use the Sennheiser HD650's, and we love them. The HD600's and AKG 701's are also very good. 

 Let me know what you find!

 Thanks,
 Elias


----------



## Lord Chaos

I just bought a pair of Denon AH-D7000 to use with my DAC1 USB. 
	

	
	
		
		

		
		
	


	




 I'm looking forward to this... they should arrive tomorrow.


----------



## emmodad

Quote:


  Originally Posted by *EliasGwinn* /img/forum/go_quote.gif 
_We have no plans to modify our approach until we find a system that results in better performance. That won't change soon because the Benchmark engineers (John Siau and myself) can easily beat up the Benchmark marketing person. 
	

	
	
		
		

		
		
	


	


_

 

hmmm... i seem to recall hearing mention of Rory taking up karate lessons...?


----------



## Bostonears

Quote:


  Originally Posted by *Lord Chaos* /img/forum/go_quote.gif 
_I just bought a pair of Denon AH-D7000 to use with my DAC1 USB. 
	

	
	
		
		

		
		
	


	




 I'm looking forward to this... they should arrive tomorrow._

 

I use a Markl modded Denon AH-D2000 with my DAC1 Pre. The amp seems to be a really good match for the low 25 ohm impedance of the Denons.

 By the way, I leave my phones plugged into the right jack of the DAC1 Pre all the time. When I want to listen to speakers, I just turn on the power amp. With that configuration, the phones are playing (and burning in) anytime I'm listening to music.


----------



## simplystax

Hello Elias 
 I have just bought the Benchmark DAC1 PRE and am using the XLR outputs to my Stax energiser and Krell amp. The Stax/Krell use true balanced designed circuits so my question is this: is the Benchmark DAC a proper balanced designed dac and what is the output level on the XLR outputs when the internal XLR jumpers are set to 0db and the rear switch is set to Calibrated mode? Hope you or anyone else can help. Thanks.


----------



## EliasGwinn

Quote:


  Originally Posted by *simplystax* /img/forum/go_quote.gif 
_Hello Elias 
 I have just bought the Benchmark DAC1 PRE and am using the XLR outputs to my Stax energiser and Krell amp. The Stax/Krell use true balanced designed circuits so my question is this: is the Benchmark DAC a proper balanced designed dac and what is the output level on the XLR outputs when the internal XLR jumpers are set to 0db and the rear switch is set to Calibrated mode? Hope you or anyone else can help. Thanks._

 

Hello Simplystax,

 Welcome to "The Tread" (sic)!

 The DAC1 PRE has a truly balanced, dual-active XLR output. The XLR outputs will have a maximum value of +24 dBu when the output mode is set to calibrated, the XLR attenuators are set to 0 dB, and the calibration potentiometers are set to the factory default. '

 Feel free to ask any more questions that may help you with your system.

 Thanks,
 Elias


----------



## HeadLover

Hi,
 I have a question as for the phones amp in the DAC1, how is it with HD650 ?
 And, how balanced it?
 I had amps (with DACS) that the left side was stronger than the right side 
	

	
	
		
		

		
		
	


	




 (really not nice, on low volumes in preticuler)

 So, how is the amp and volume control?
 And is the volume control allow a good control of it? so even at the low volume or high, I can "play" with it till I get the exact volume I want, and not to much high or low ??


----------



## Quaddy

Quote:


  Originally Posted by *HeadLover* /img/forum/go_quote.gif 
_Hi,
 I have a question as for the phones amp in the DAC1, how is it with HD650 ?
 And, how balanced it?
 I had amps (with DACS) that the left side was stronger than the right side 
	

	
	
		
		

		
		
	


	




 (really not nice, on low volumes in preticuler)

 So, how is the amp and volume control?
 And is the volume control allow a good control of it? so even at the low volume or high, I can "play" with it till I get the exact volume I want, and not to much high or low ??_

 


 regarding 650's, see a few pages back, straight from the horses mouth so to speak, from elias.

 i have used sen 600's with the HPA2 (the amp in the DAC1 pre) and it drove them commandingly, as you would expect.

 balanced its great, true balanced performance that i can ascertain, i used to run xlr out to my headroom balanced rig

 great, non-stepped volume control, allowing very precise control over knob and volume levels. 

 its a very configurable item in terms of outputs and levels, anyway these are just a few words from an end user, i am sure elias can help out in a more technical way if required

_this is all relating to dac1 pre not dac1_


----------



## HeadLover

Yes I am talking about the PRE DAC1
 But for balanced I meant that some times I had devices where the left channel of the sound was higher than the right and so on.
 How is the DAC1 ?
 And volume control, do I need to add gain for the HD650 or the default is good enough ?


----------



## poo

^ Default is fine, if anything (specific to the 650s) it is too loud, but I'm sure that is to allow headroom for other cans and possibilities.

 I have not experienced balanced from the DAC1 in a environment quiet enough to be definitive about it (only at meets), but L & R seemed to be well balanced to my ears.


----------



## GUINNE55

From the website, it looks like the same headphone amp is in all models of the DAC1


----------



## BitPerfect

Quote:


  Originally Posted by *GUINNE55* /img/forum/go_quote.gif 
_From the website, it looks like the same headphone amp is in all models of the DAC1_

 

The DAC1 Pre differs from previous models in offering three gain settings (0 dB, -10dB, -20dB) for headphone listening.

 That might make a difference for the OP, since volume pots (nearly?) always exhibit balance quirks at low volume settings. Benchmark recommends listening with the volume control set above the 10th detent position.

 Previous posters have mentioned that the default setting for the DAC1 USB is a bit loud with the HD650 in the "12 o'clock" position, so the additional level of attenuation might be desirable.

http://www.head-fi.org/forums/f46/be...ml#post4685125


----------



## GUINNE55

I don't know if thats worth a 600 dollar premium over the original DAC1.


----------



## BitPerfect

Quote:


  Originally Posted by *GUINNE55* /img/forum/go_quote.gif 
_I don't know if thats worth a 600 dollar premium over the original DAC1._

 

If you don't have a need for USB digital input, or the analog input, or the other additional digital inputs, and don't desire the interconnect driving capability of the new LM4562 op-amps, then the "classic" model might be just right for you and you can happily save the additional $300 or $600. 
	

	
	
		
		

		
			





 To quote Elias in an earlier posting (http://www.head-fi.org/forums/f46/be...ml#post3653999):

  Quote:


 -DAC1 PRE has 5 digital inputs (USB, optical, coax(x3))
 -DAC1 USB has 4 digital inputs (USB, optical, coax(x1), and XLR)
 -DAC1 PRE uses teflon chassis-mounted RCA analog and digital connectors
 -DAC1 USB uses PCB mounted RCA connectors
 -DAC1 PRE has 3 gain ranges for the HPA2 headphone amplifier
 -DAC1 USB has 2 gain ranges for the HPA2 headphone amplifier
 -DAC1 PRE uses National Instrument LM4562 opamps throughout the analog circuits
 -DAC1 USB uses 5532's every except the output drivers, which uses the 4562's
 -DAC1 PRE has power button
 -DAC1 USB does not have power button


----------



## GUINNE55

"interconnect driving capability of the new LM4562 op-amps"

 Could you explain this for me?

 Btw 600 for a power button?


----------



## BitPerfect

Quote:


  Originally Posted by *GUINNE55* /img/forum/go_quote.gif 
_"interconnect driving capability of the new LM4562 op-amps"

 Could you explain this for me?_

 

I'll try, although I'm sure Elias can do a much better job. The LM4562s in the output stage have lower distortion and better high-frequency response than the NE5532s when the DAC1 is driving high capacitance and/or low impedance loads.

  Quote:


 Btw 600 for a power button? 
	

	
	
		
		

		
		
	


	



 

Now you're just trolling.


----------



## athenaesword

Quote:


  Originally Posted by *BitPerfect* /img/forum/go_quote.gif 
_If you don't have a need for USB digital input, or the analog input, or the other additional digital inputs, and don't desire the interconnect driving capability of the new LM4562 op-amps, then the "classic" model might be just right for you and you can happily save the additional $300 or $600. 
	

	
	
		
		

		
		
	


	




 To quote Elias in an earlier posting (http://www.head-fi.org/forums/f46/be...ml#post3653999):_

 

so the pre has no balanced connections?


----------



## BitPerfect

Quote:


  Originally Posted by *athenaesword* /img/forum/go_quote.gif 
_so the pre has no balanced connections?_

 

It has no balanced AES/EBU XLR input connection like the previous two models. It has both balanced and unbalanced analog outputs.

 See: DAC1 PRE Photo Gallery


----------



## Lord Chaos

Balance between left and right on both of my DAC1s is just fine, and due to having efficient phones I'm rarely listening at anything like Benchmark's recommended position. The Denon AH-D7000 I just got require more power than the others, so on a wide-dynamic-range recording I'm around the 10th detent. With the others I'm at around 5th or 6th.


----------



## sadhill

Hi Elias, Hi all,

 I discover this technically very interesting thread today. Owing 2 DAC's (EMU 0404USB and Zero Chinese DAC), I am considering to upgrade to a Benchmark DAC 1 (opportunity to buy a 2 years old DAC1, or maybe straight away a new DAC1 USB).

 Referring to previous posts, and namely #2119, Elias, you write that DAC 1 anyway resamples digital input. Does that mean that a non-jitter free USB to SPDIF transport (say a rather low priced one, but able to be bit-perfect with ASIO drivers bypassing KMIXER) would possibly yield good results and produce a jitter-free signal with the help of the DAC 1 ?


----------



## EliasGwinn

Hello Head-Fiers!

 I've been away for the holidays...so I apologize for my M.I.A. status.

 Hope all is well between the cans 
	

	
	
		
		

		
		
	


	




 Thanks,
 Elias


----------



## EliasGwinn

Quote:


  Originally Posted by *HeadLover* /img/forum/go_quote.gif 
_Hi,
 I have a question as for the phones amp in the DAC1, how is it with HD650 ?
 And, how balanced it?
 I had amps (with DACS) that the left side was stronger than the right side 
	

	
	
		
		

		
		
	


	




 (really not nice, on low volumes in preticuler)

 So, how is the amp and volume control?
 And is the volume control allow a good control of it? so even at the low volume or high, I can "play" with it till I get the exact volume I want, and not to much high or low ??_

 

HeadLover,

 The HD650's are a particularly good choice to use with the DAC1. Many folks on this forum (myself included) use the HD650's w/ their DAC1 with great results (as they have mentioned). 

 The balance between the left and right channel is very tight when the volume control is above the first 5-10% of its rotation. The DAC1 USB and the DAC1 PRE have gain-range adjustments so that the volume control can be used in its optimum position even at low volumes.

 Thanks,
 Elias


----------



## davidw89

Can't find the Dac1 USB on their website, can someone link me. Pricing btw?


----------



## EliasGwinn

Quote:


  Originally Posted by *davidw89* /img/forum/go_quote.gif 
_Can't find the Dac1 USB on their website, can someone link me. Pricing btw?_

 

Here it is: DAC1 USB

 Thanks,
 Elias


----------



## EliasGwinn

Quote:


  Originally Posted by *sadhill* /img/forum/go_quote.gif 
_Hi Elias, Hi all,

 I discover this technically very interesting thread today. Owing 2 DAC's (EMU 0404USB and Zero Chinese DAC), I am considering to upgrade to a Benchmark DAC 1 (opportunity to buy a 2 years old DAC1, or maybe straight away a new DAC1 USB).

 Referring to previous posts, and namely #2119, Elias, you write that DAC 1 anyway resamples digital input. Does that mean that a non-jitter free USB to SPDIF transport (say a rather low priced one, but able to be bit-perfect with ASIO drivers bypassing KMIXER) would possibly yield good results and produce a jitter-free signal with the help of the DAC 1 ?_

 

Hello sadhill,

 A USB -> SPDIF converter can yield good results, but the problem with most of the those converters is that we haven't found any that are both bit-transparent and capable of 24-bit operation. A 24-bit connection is very important, even when listening to 16-bit audio, because certain processes in the computer can cause the data stream to increase to 24-bits, and those bits will be truncated with a 16-bit device. 

 If you knew absolutely that the USB-SPDIF interface was 24-bits and bit-transparent, then you could expect to get good results. Remember, just because a device is ASIO and bypasses kmixer doesn't mean that its bit-transparent. Many times the devices driver will cause alterations to the data. FYI...

 Thanks,
 Elias


----------



## HeadLover

Quote:


  Originally Posted by *EliasGwinn* /img/forum/go_quote.gif 
_Hello sadhill,

 A USB -> SPDIF converter can yield good results, but the problem with most of the those converters is that we haven't found any that are both bit-transparent and capable of 24-bit operation. A 24-bit connection is very important, even when listening to 16-bit audio, because certain processes in the computer can cause the data stream to increase to 24-bits, and those bits will be truncated with a 16-bit device. 

 If you knew absolutely that the USB-SPDIF interface was 24-bits and bit-transparent, then you could expect to get good results. Remember, just because a device is ASIO and bypasses kmixer doesn't mean that its bit-transparent. Many times the devices driver will cause alterations to the data. FYI...

 Thanks,
 Elias_

 

I don't understand
 If I am using 24 bit (on my foobar2000) with WASAPI, why won't it be bit perfect ?


----------



## HeadLover

OH and I have one more question
 How is the combo of the AudioEngine A5N with the DAC1 ?
 I am having them (soon to have) as a speakers for my PC, how is this combo ?

 What do you think ?


----------



## Quaddy

i use the DAC1 and A5's.

 sound great to me as '_budget_' speakers.

 although you may get peeved off with the A5 auto-sleep feature, which *may *be worse with that exact setup, i am not sure.


----------



## HeadLover

What do you mean?
 I mean, if I play music all time (when I hear music I mean), should I have a problem?
 And I mean how is the combo, means also that isn't the DAC1 to "bright" or what ever to this kind of speakers or what ever?


----------



## Quaddy

well the dac1 is bright and the A5's arent bright. i find they are warmish, so evens out there.

 but yes on my A5's the auto-sleep kicks in and disrupts audio playing, whih is very annoying

 i could do with feeding my A5's from another source other than DAC1 to see if i still get the same problem

 IMHO, audioengine should have made that feature defeatable!!!!

 but dac1 and A5 sound is fine to me, nearfield.


----------



## HeadLover

What do mean auto sleep? what just when you play it goes to sleep?
 How come?
 Can you explain a little more? and why is it with the DAC1 ? with something like a nice CDP is it sill the same ?


----------



## dmashta

Quote:


  Originally Posted by *Quaddy* /img/forum/go_quote.gif 
_...
 but yes on my A5's the auto-sleep kicks in and disrupts audio playing, whih is very annoying
_

 

 Quote:


  Originally Posted by *HeadLover* /img/forum/go_quote.gif 
_What do mean auto sleep? what just when you play it goes to sleep?
 How come?
 Can you explain a little more? and why is it with the DAC1 ? with something like a nice CDP is it sill the same ?_

 

the A5 goes into standby, power-saving mode after about 15 mins of inactivity (no audio input). it doesn't go into standby while there's music playing so it does not disrupt audio playback. nothing to do with DAC1 either. and when in standby, play a song and it automatically "wakes up". you will hear a slight pop when it goes into standby.


----------



## HeadLover

Oh so it isn't that of a big deal I guess


----------



## dmashta

Quote:


  Originally Posted by *HeadLover* /img/forum/go_quote.gif 
_Oh so it isn't that of a big deal I guess 
	

	
	
		
		

		
		
	


	


_

 

no, not at all. sounds like quaddy might be having problem with his pair.

 i also agree with quaddy. the a5s are warmish and balances out the dac1. no harshness with that pairing.


----------



## Quaddy

my A5's cut off when there is music playing, and takes a severe rising of the volume knob to bring them back on, which is why i suspect it may be a disruption from the dac1 signal somehow, with its own cutout technology, i will feed it with something else, see if that makes a dif.

 i use dac1 rca out to mini-in on A5, in calibrated/fixed mode. maybe i need to alter the levels inside the DAC1 again

 sorry for some non dac1 talk in this


----------



## dmashta

Quote:


  Originally Posted by *Quaddy* /img/forum/go_quote.gif 
_my A5's cut off when there is music playing, and takes a severe rising of the volume knob to bring them back on, which is why i suspect it may be a disruption from the dac1 signal somehow, with its own cutout technology, i will feed it with something else, see if that makes a dif.

 i use dac1 rca out to mini-in on A5, in calibrated/fixed mode. maybe i need to alter the levels inside the DAC1 again

 sorry for some non dac1 talk in this_

 

no problem here with the dac1 nor any other source so if yours still cutting off with other sources, you definitely have a defective pair. you don't need to adjust the volume either to bring it back on.


----------



## HeadLover

I have a question please
 (Hope you can help me out)

 Say, if I am using the DAC1 as my main thing(mean I am not have a sound card)

 Will using the PC be slower? like when I am gaming or other stuff?
 For example, when using foobar2000 and I try to seek on the song with the PICO connected to the USB, I need to wait a second or two each time (kind of a delay), while doing the same with the SoundCard isn't a problem

 so, how slow is it ?
 And I will like to know more about it if can, if someone can elaborate on it please.


----------



## Quaddy

i use the dac1 as my soundcard.

 no delay here apart from normal seek delay, dac1 is not adding anything on my system here.

 some claim that usb at a low level introduces processing power, which i am sure it does, but internal cards running off the pci-bus also have a payload.

 your usb mouse will be polling more cpu cycles i bet.

 i am using directsound v2.0 and bypassing windows mixer in vista

 no slowness at all.


----------



## ted betley

Elias is there a way to input an I2S signal into the Benchmark USB dac?


----------



## EliasGwinn

Quote:


  Originally Posted by *ted betley* /img/forum/go_quote.gif 
_Elias is there a way to input an I2S signal into the Benchmark USB dac?_

 

No, the DAC1 is compatible with AES or SPDIF input signals only.


----------



## slwiser

Quote:


  Originally Posted by *ted betley* /img/forum/go_quote.gif 
_Elias is there a way to input an I2S signal into the Benchmark USB dac?_

 

Yes there is but it will cost you....see here: Interconnects and Cables by Empirical Audio and click on component modes on the left side links.


----------



## HeadLover

I guess no one knows yet, but still
 Wonder how good will it be with the new HD800, maybe will need a little update to the gain to have a little bit more?


----------



## HeadLover

And one more thing please, can I have it pre-factory with a little more gain? like 12 instead of the default 10 for my HD650 ?
 I love having a lot of volume when I want and can.


----------



## EliasGwinn

Quote:


  Originally Posted by *HeadLover* /img/forum/go_quote.gif 
_And one more thing please, can I have it pre-factory with a little more gain? like 12 instead of the default 10 for my HD650 ?
 I love having a lot of volume when I want and can._

 

We can't change the gain structure on your unit, but the gain range jumpers will make it so that you will have plenty of room for customization.

 Thanks,
 Elias


----------



## HeadLover

Quote:


  Originally Posted by *EliasGwinn* /img/forum/go_quote.gif 
_We can't change the gain structure on your unit, but the gain range jumpers will make it so that you will have plenty of room for customization.

 Thanks,
 Elias_

 

Amm ok
 And one more thing, is there any and I mean ANY kind of optimization that can be done to the unit by you? something more or some kind of improvement before I buy it ?

 And
 One more thing (I don't know if asked before), How does the unit work with LINUX ?
 Will it still be good with the USB ? and will it still be 24bit and bit perfect ?
 And can I do firmware updates to the unit? in any some kind of way? do you have this kind of thing for the unit ?


----------



## HeadLover

Oh and something more,
 As for using the DAC1 PRE with my PC
 Will it be better using USB vs COAX ?
 Did you tested it even with good sound card and good cables and so on?
 And still is USB is better ?


----------



## EliasGwinn

Quote:


  Originally Posted by *HeadLover* /img/forum/go_quote.gif 
_Amm ok
 And one more thing, is there any and I mean ANY kind of optimization that can be done to the unit by you? something more or some kind of improvement before I buy it ?_

 

Yes, all parameters are optimized on all units before they are shipped.

  Quote:


  Originally Posted by *HeadLover* /img/forum/go_quote.gif 
_And
 One more thing (I don't know if asked before), How does the unit work with LINUX ?
 Will it still be good with the USB ? and will it still be 24bit and bit perfect ?
 And can I do firmware updates to the unit? in any some kind of way? do you have this kind of thing for the unit ?_

 

We have several customers who use the USB with Linux, but we haven't tested it ourselves and can't guarantee its performance. There are too many variables with Linux...each setup is different...we couldn't gaurantee across the board.

 Thanks,
 Elias


----------



## HeadLover

Ok thank you!
 So, with windows VISTA 64 bit (or upcoming 7) using foobar2000, I should get the best results, right?
 All I need to do is choose it as my output, than use 24 bit, and set the volume to max on foobar2000 and disable all the DSP and gain stuff, right?

 Than I will get PERFECT Bit for sure? no way for improve it here?

 Are you sure it will get me really BIT perfect?
 I heard you have say before, that you have tested with some sound cards using COAX, and they weren't bit perfect at all,
 So using USB like this will gruntee me bit perfect?


----------



## EliasGwinn

**repeat post


----------



## EliasGwinn

Quote:


  Originally Posted by *HeadLover* /img/forum/go_quote.gif 
_Oh and something more,
 As for using the DAC1 PRE with my PC
 Will it be better using USB vs COAX ?
 Did you tested it even with good sound card and good cables and so on?
 And still is USB is better ?_

 

I recommend USB of the DAC1 PRE. We've looked at a lot of sound cards, and we haven't found a sound card that does what its supposed to (transfer the digital data bit-transparently at 24-bits). The USB of the DAC1 does exactly what its supposed to, so if you have the DAC1 PRE, use USB.

 Thanks,
 Elias


----------



## HeadLover

Need to get money!!!




 Soon I will have something like 1300$ on my PAYPAL, than only 300 more or something, and I buy to me a new DAC1 PRE





 I hope it is as good as you say!
 I plan on "clean" my desktop, so I will have only the DAC1 serving me both as a DAC, USB input, PRE, AMP for HD650.

 What do you say?
 Also, I have some kind of power conditioner, Do I need it with the DAC1? or does it have so good PSU that it doesn't even matter?


----------



## Lil' Knight

Sorry for my noob question, but exactly, what's the biggest difference between the DAC1 usb and DAC1 pre? I'm thinking of upgrading my source in the near future and USB input, XLR output are a must.


----------



## compuryan

Quote:


  Originally Posted by *HeadLover* /img/forum/go_quote.gif 
_Need to get money!!!




 Soon I will have something like 1300$ on my PAYPAL, than only 300 more or something, and I buy to me a new DAC1 PRE





 I hope it is as good as you say!
 I plan on "clean" my desktop, so I will have only the DAC1 serving me both as a DAC, USB input, PRE, AMP for HD650.

 What do you say?
 Also, I have some kind of power conditioner, Do I need it with the DAC1? or does it have so good PSU that it doesn't even matter?_

 

A power conditioner could certainly help. Why not wait until you get your DAC1 PRE and run it through the power conditioner and then without. If it sounds better with the conditioner, then keep it.


----------



## HeadLover

Quote:


  Originally Posted by *compuryan* /img/forum/go_quote.gif 
_A power conditioner could certainly help. Why not wait until you get your DAC1 PRE and run it through the power conditioner and then without. If it sounds better with the conditioner, then keep it._

 

The problem is that it take a lot of space on my desktop 
	

	
	
		
		

		
		
	


	



 I want to "clean" things up


----------



## Quaddy

i recall elias saying that the nature and robustness of the DAC1's psu, that a conditioner or trick power cable wasnt required.


----------



## BitPerfect

Quote:


  Originally Posted by *Lil' Knight* /img/forum/go_quote.gif 
_Sorry for my noob question, but exactly, what's the biggest difference between the DAC1 usb and DAC1 pre? I'm thinking of upgrading my source in the near future and USB input, XLR output are a must._

 

The biggest difference? The DAC1 Pre has one analog input. The DAC1 USB has none. Have you looked at Benchmark's website? It answers such questions and more.


----------



## HeadLover

EliasGwinn, I have a question for you
 I have the money, I want to buy it 
	

	
	
		
		

		
			





 But,
 Will there be any kind of update or upgrade to the unit soon? I am afraid to buy and than to find out there will be one month from now or so
 Also, can I update the firmware of the unit ?
 And, if like one year or so you will put a new unit, will there be a way for a trade in or something? so I can return the unit, pay X money, and get the new one ??


----------



## poo

^ Your question is not going to be answered in the way you hope. Look a the history of this thread and of Benchmark product releases for you answer.


----------



## HeadLover

Ok
 BTW, now that Windows 7 Beta is out, anyone tried DAC1 USB or PRE with it?
 I wonder how is and how is working with it (I assume most of us will use 7 and not Vista like one year from now)


----------



## HeadLover

OH BTW
 [size=large]*I have just put an order for my brand new 220V DAC1 PRE*[/size] 
	

	
	
		
		

		
		
	


	




 So
 I just hope it will be AMAZING as I want!
 Both the DAC, USB, PRE, and amp for my HD650





 Now I am "true" Benchmark fan


----------



## EliasGwinn

Quote:


  Originally Posted by *Lil' Knight* /img/forum/go_quote.gif 
_Sorry for my noob question, but exactly, what's the biggest difference between the DAC1 usb and DAC1 pre? I'm thinking of upgrading my source in the near future and USB input, XLR output are a must._

 

The DAC1 PRE has following features which the DAC1 USB does not have: 
 - a stereo analog input
 - 5 digital inputs (USB, optical, 3x coaxial)
 - upgraded opamps
 - upgraded connectors
 - a wider gain range selection for the headphone amplifier
 - and an on/off button. 

 Thanks,
 Elias


----------



## EliasGwinn

Quote:


  Originally Posted by *HeadLover* /img/forum/go_quote.gif 
_Also, I have some kind of power conditioner, Do I need it with the DAC1? or does it have so good PSU that it doesn't even matter?_

 

No need for a power conditioner. The performance of the DAC1 is consistent regardless of the quality of the power.

 Thanks,
 Elias


----------



## HeadLover

Well a little off topic, but I MUST ask
 What is that thing on you photo here?!


----------



## EliasGwinn

Quote:


  Originally Posted by *HeadLover* /img/forum/go_quote.gif 
_Well a little off topic, but I MUST ask
 What is that thing on you photo here?!



_

 

That's my dog, Frank!

 Thanks for asking!


----------



## HeadLover

Haa ok
 You need to take a better photo, really hard to tell what it is from the photo 
	

	
	
		
		

		
		
	


	




 And I am really exited from the new DAC1 PRE

 Hope to write only good things soon here

 I hope that using FOOBAR2000 with WASPI (24 bit) and DAC1 PRE and HD650 (with moon audio blue dragon V3) and with an amazing FLAC files of KISS ALIVE! will sound amazing!





 This will be one of the first thing I will be hearing with it.

 BTW,
 What do I need to set up with it? any thing speical? after I connect it to my PC with the USB and turn it on, what then? need what player? what will give me BEST result and BIT Perfect ?
 (I am using Vista 64 BIT)


----------



## poo

Quote:


  Originally Posted by *HeadLover* /img/forum/go_quote.gif 
_Now I am "true" Benchmark fan 
	

	
	
		
		

		
		
	


	


_

 

Calm down mate... you haven't even heard one yet!

 I'm sure you'll enjoy it!


----------



## HeadLover

Haa, I meant, that now I bought one, I hope to enjoy it very much and that it will serve me for many years to come 
	

	
	
		
		

		
		
	


	




 BTW
 Do I need to do any thing speical for using it?
 After I connect it?
 How do I assure that playing FLAC files from my PC (VISTA 64BIT) will be true bit perfect and with excellent quality?


----------



## bizkid

I've owned the DAC1 USB for some months now and basically i'm happy with it. However there are 2 things where i see the need for improvement.

 First the easy one, the volume pot. Atleast on my DAC1 the L/R Balance varies more than i'd like throughout the range. That means the headphone out is unusable if you don't attentuate the volume in the digital domain to reach the (very small!) sweetspot of the poti. That's just not good enough for 1300 Euro, sorry.

 The second one goes a little deeper, i'm not happy with the resampling algo that is used, especially since EVERYTHING get's resampled to the DACs optimal rate (which makes sence) and there's no way turning that off thus limiting the DACs capability. The problem is that the resampling algo in the DAC1 doesn't appear to be a good one, atleast sonically.

 You can see measured comparisons of various resampling soft/hardware here:
SRC Comparisons

 Yesterday i ABX'd a propper resampled 96khz file and the original 44khz file (thus the DAC1 has more resampling to do). Out of 10tries i scored 100%.
 I'd really love to hear how the DAC1 sounds without any of it's own subpar resampling.

 One more note, Elias i know that you guys are using a pair of O300 at Benchmark to test the DAC1. I also happen to own a pair and let me tell you it's almost impossible to ABX the differences in resampling through that. With my headphone rig it's a breeze. So while this probably doesn't affect most of your users using speakers/monitors i hope there will be some improvement on those aspects. I love the rest the DAC1 has to offer!


----------



## HeadLover

Amm, what do you mean volume balance?


----------



## bizkid

I mean the channel imbalance between the left/right channels while using the volume pot.


----------



## infinitesymphony

Quote:


  Originally Posted by *bizkid* /img/forum/go_quote.gif 
_First the easy one, the volume pot. Atleast on my DAC1 the L/R Balance varies more than i'd like throughout the range. That means the headphone out is unusable if you don't attentuate the volume in the digital domain to reach the (very small!) sweetspot of the poti. That's just not good enough for 1300 Euro, sorry._

 

Doesn't the DAC1 USB have internal jumpers to change the attenuation? You may want to check the manual.

  Quote:


  Originally Posted by *bizkid* 
_You can see measured comparisons of various resampling soft/hardware here:
SRC Comparisons_

 

What relevance do software resampler measurements have if you have no way of measuring the quality of the DAC1's resampling?

  Quote:


  Originally Posted by *bizkid* 
_Yesterday i ABX'd a propper resampled 96khz file and the original 44khz file (thus the DAC1 has more resampling to do). Out of 10tries i scored 100%.
 I'd really love to hear how the DAC1 sounds without any of it's own subpar resampling._

 

There are a couple of problems with your test. The DAC1 does not resample to 96 kHz, it resamples to ~110 kHz, so both of your files will be resampled by the DAC1 on playback. Second, you used a software resampler to perform the 44.1 kHz -> 96 kHz conversion; the 96 kHz file will have been resampled twice (by two different resamplers) by the time it reaches your ears. If you hear a difference, it probably lies there.


----------



## bizkid

The internal attenuation of 10dB isn't even enough for the HD650 or 600 Ohm Beyers to reach the sweet spot. And imho the attentuation is just a cheap fix to a whole other problem, the poti itself. Come on i've seen better pots in sub 250$ china gear.

 Please do a research on measuring resampling and see yourself what results you get by upsampling from 44khz to 110khz (or higher ie 192khz) straight with a so-so algorithm vs 44khz -> good resampler to 96khz -> to 110khz (via same or worse algo). Its much less of a problem to go from 96khz to any higher rate even with an average resampler.


 The website also offers comparisons of hardware implenentations, very expensive ones.


----------



## HeadLover

I wonder, did you check it as a pre?
 or do you have the version with the PRE ?
 Also, maybe only your unit isn't ok?


----------



## bizkid

I have the USB variant, not pre but it doesn't matter. Potentiometers vary, even the better ones (APLS & co). There's definately a chance to catch one that is worse or better. Many manufactures select their potentiometers. I also use the DAC1 for my O300 Studio Monitors and setting the xlr output to calibrated is necessary, i control the volume in my DAW (usually not recommended).
 My SPL Phonitor has no measurable offset between the WHOLE potentiometer range.


----------



## HeadLover

Lets wait to offical repley for this 
	

	
	
		
		

		
			




 I hope my new DAC1 PRE won't have such problems 
	

	
	
		
		

		
		
	


	




 BTW
 What did you like about it ?


----------



## bizkid

The no1 thing i like about it is the sound ofcourse, with a 2nd place going to the form factor/looks and i/o options (specially usb). The headphone amps are not the last word in headphone amplification but very clean and absolutely useable, 3rd place 
	

	
	
		
		

		
		
	


	



 And i like to buy gear not made in china


----------



## HeadLover

I wonder, did you try it with HD600 or HD650?
 Most people claim very good results there


----------



## G-U-E-S-T

Quote:


  Originally Posted by *bizkid* /img/forum/go_quote.gif 
_...the resampling algo in the DAC1 doesn't appear to be a good one, atleast sonically..._

 

Could you please explain what you mean by "doesn't appear to be a good one"?

 I ask because I have done similar comparisons with various files (including files already quality algo-upsampled to 96kHz versus 44.1kHz) and to my ears the entire processing scheme of the DAC1 PRE (which is my version), including its upsampling, sounds just superb. The volume control is very acceptable as well.

 Normally the more rounds of resampling that occur, the more resulting digital distortion. So two upsamplings (i.e. first to 96kHz, then to 110kHz) are usually not going to be better than just one.

 I wonder, since your listening tests were not double blind, if the placebo effect is very likely affecting you here as well?

 Either way: Although the results of my own listening tests do not agree with yours, your opinion is respected regardless!


----------



## Lil' Knight

After reading about 50 pages of this thread, I decide to pull the trigger on the DAC1 USB. Hopefully, it'll be a big improvement from my DacMagic.
 Just some small questions:
 + Since it already has a native USB driver, I don't have to use ASIO4ALL with foobar2k, right?
 + How do the headphones out on the DAC1 drive low impedance phones, like the ESW10JPN and PK1? I'll sell my portable amp if I buy the DAC1.
 + Can I use both the headphones out and XLR outputs at the same time?
 + And lastly, anyone tried it with some tube amps? How do they sound together?


----------



## HeadLover

Yep, I still wonder how to use mine with foobar2000
 Please advice.


----------



## Bojamijams

You will have to use ASIO4ALL.


----------



## HeadLover

Can you explain more?
 Also, can I use WASAPI instead?


----------



## bizkid

Quote:


 Normally the more rounds of resampling that occur, the more resulting digital distortion. So two upsamplings (i.e. first to 96kHz, then to 110kHz) are usually not going to be better than just one. 
 

I'm curious what kind of "quality" algo did you use? 
 2x resampling with a resampler that leaves almost no measurable artifacts vs 1x with a so-so one is a BIG difference which you can easily measure and see with your eyes. This is not about opinions/ears but measurable facts.

 I suggest you to read into that http://src.infinitewave.ca/help.html

 I'm not sure what you mean by double blind test but loading 2 files into foobar and doing an abx comparison is pretty much as blind as it get's if i didnt miss anything there.



 Quote from a SOS review: http://www.soundonsound.com/sos/jul0.../benchmark.htm

 "The aspect that initially troubled me with this unusual approach to D-A conversion was the fact that the input is always passing through an SRC using complex non-integer ratios. The received wisdom is that non-integer SRC processes produce potentially audible artefacts"



 But as the whole DAC1 design relies on the use of the AD1896 and it's SRC we probably won't see a change. The fact remains that the actual SRC algo used is a weak spot in the DAC1 design.
 There are other manufactures who offer their own AD1896 like chips but these have problems too, worse ones than the AD1896 has. It's the best currently avaible for this application until analog (or somebody else) will release a superior version.


----------



## HeadLover

EliasGwinn, can you please comment on some of the lated stuff that bizkid told here please ??


----------



## bizkid

Quote:


  Originally Posted by *HeadLover* /img/forum/go_quote.gif 
_EliasGwinn, can you please comment on some of the lated stuff that bizkid told here please ??_

 

Give him a break it's weekend 
	

	
	
		
		

		
		
	


	




 Also you won't hear anything i wrote about with a HD650. Don't worry. These are very subtle differences.


----------



## Quaddy

Quote:


  Originally Posted by *Lil' Knight* /img/forum/go_quote.gif 
_After reading about 50 pages of this thread, I decide to pull the trigger on the DAC1 USB. Hopefully, it'll be a big improvement from my DacMagic.
 Just some small questions:
 [size=medium]*2 Since it already has a native USB driver, I don't have to use ASIO4ALL with foobar2k, right?*[/size]
 + How do the headphones out on the DAC1 drive low impedance phones, like the ESW10JPN and PK1? I'll sell my portable amp if I buy the DAC1.
 [size=medium]*2 Can I use both the headphones out and XLR outputs at the same time?*[/size]
 + And lastly, anyone tried it with some tube amps? How do they sound together?_

 

1 you can still pick your preferred method of bypassing windows mixer, i use direct sound v2 via my off ramp then into dac1 pre, this is a different thing to usb driver functionality

 2 yep sure can, independent.


----------



## HeadLover

Ok I still don't understand.
 Will there be any different from ASIO4ALL to WASAPI to Direct Sound to what ever?
 I mean, will they all put bit perfect into it?
 Also, must I choose 24 bit on my foobar2000 as output? or will it do it auto or what ever?


----------



## Lil' Knight

Quote:


  Originally Posted by *HeadLover* /img/forum/go_quote.gif 
_Also, must I choose 24 bit on my foobar2000 as output? or will it do it auto or what ever?_

 

24-bit output
 100% volume
 all DSP turned off


----------



## HeadLover

Ok Thank you!
 And what about the device? does it matter what I choose?


----------



## Bostonears

Quote:


  Originally Posted by *bizkid* /img/forum/go_quote.gif 
_I have the USB variant, not pre but it doesn't matter._

 

Yes it does matter. The DAC1 USB only has 0dB and -10dB attenuation jumper settings for the headphone output. The DAC1 Pre has 0dB, -10dB and -20dB settings, so it's more likely to have one that will hit the sweet spot on the volume knob for any given headphones.


----------



## bizkid

Quote:


  Originally Posted by *Bostonears* /img/forum/go_quote.gif 
_Yes it does matter. The DAC1 USB only has 0dB and -10dB attenuation jumper settings for the headphone output. The DAC1 Pre has 0dB, -10dB and -20dB settings, so it's more likely to have one that will hit the sweet spot on the volume knob for any given headphones._

 

I refered to the quality of the volume potentiometer. But you're right the -20dB jumper makes it alot more useable compared to -10dB. Too bad Benchmark didn't introduce it into the other DACs.


----------



## HeadLover

Why does the -20 or -10 matter?
 What phones are you using?
 I think that for the HD650, the default should be good enough, not ?


----------



## Lil' Knight

So you will use the DAC1 to drive your HD650 directly?


----------



## HeadLover

I will be using the DAC1 PRE with its headphones amp to drive mine HD650
 Yes
 Why not?


----------



## G-U-E-S-T

Quote:


  Originally Posted by *bizkid* /img/forum/go_quote.gif 
_...2x resampling with a resampler that leaves almost no measurable artifacts vs 1x with a so-so one is a BIG difference which you can easily measure and see with your eyes...

 Quote from a SOS review: Benchmark DAC1

 "The aspect that initially troubled me with this unusual approach to D-A conversion was the fact that the input is always passing through an SRC using complex non-integer ratios. The received wisdom is that non-integer SRC processes produce potentially audible artefacts"
 ..._

 

But the resampling scheme in the DAC1 PRE is not a "so-so" one at all, but is in fact an excellent implementation which rather obviously sounds fantastic when listening.

 I hear no audible difference whatsoever between straight 44.1kHz files versus those first resampled by Lynx hardware/software combo in DAW to 96kHz, when put through the DAC1 PRE.

 Regarding your linked quote above, you somewhat conspicuously left out the rest of the reviewer's quoted sentence. Out of honesty and for the benefit of you and all the other readers here, I will give the complete sentence with emphasis on the part of real importance:

_"The aspect that initially troubled me with this unusual approach to D-A conversion was the fact that the input is always passing through an SRC using complex non-integer ratios. The received wisdom is that non-integer SRC processes produce potentially audible artefacts, *but Benchmark claim that the design of the AD1896 SRC chip is such that it does not require (or benefit from) integer ratios between input and output, and its performance exceeds that of the D-A converter, so it is not a quality bottleneck.* With this in mind, I put it to a listening test.."_

 I might note that the reviewer (Hugh Robjohns) had no complaints, and in fact was so impressed with the DAC1 that he bought the review sample.


----------



## bizkid

Dude....

 "but Benchmark *claim*"

 The sentence before the "," are facts. The part you bolded are "claims".
 Hope you know the difference.

 Most DAWs have a so-so SRC resampler as you might have seen on the website i mentioned, if you even bothered looking.


 I didnt intend to post this to start a debate, i just wanted send some feedback over to benchmark and i know elias prefers public posting vs PM


----------



## emmodad

Quote:


  Originally Posted by *bizkid* /img/forum/go_quote.gif 
_Dude....

 "but Benchmark *claim*"

 The sentence before the "," are facts. The part you bolded are "claims".
 Hope you know the difference.

 Most DAWs have a so-so SRC resampler as you might have seen on the website i mentioned, if you even bothered looking. [snip])_

 






 do continue, bizkid. highly entertaining.....


----------



## HeadLover

I am still waiting for EliasGwinn to comment on those things!


----------



## Bostonears

Quote:


  Originally Posted by *HeadLover* /img/forum/go_quote.gif 
_Why does the -20 or -10 matter?
 What phones are you using?
 I think that for the HD650, the default should be good enough, not ?_

 

The response of the volume pot is non-linear, and it behaves best in the middle of its range. Depending on how loudly you like to listen and the type of music you listen to (for example, rock is usually mixed more loudly than classical), the -20dB setting may get you the better part of the pot's range.


----------



## HeadLover

Oh, and what is the default when I have it? the DAC1 PRE ?
 And I want the HD650 to get LOUD with it


----------



## dmashta

Quote:


  Originally Posted by *Bostonears* /img/forum/go_quote.gif 
_The response of the volume pot is non-linear, and it behaves best in the middle of its range. Depending on how loudly you like to listen and the type of music you listen to (for example, rock is usually mixed more loudly than classical), the -20dB setting may get you the better part of the pot's range._

 

i have always been curious about this but never quite sure what it means. on the dac1 pre, i've adjusted the volume pot through its range but never noticed a 'sweet spot'. sure it sounds best within a certain range but that's just my loudness preference, no? so the inner workings of the pot notwithstanding, what exactly is this sweet spot we talking about?


----------



## EliasGwinn

Hello Bizkid...

  Quote:


  Originally Posted by *bizkid* /img/forum/go_quote.gif 
_I've owned the DAC1 USB for some months now and basically i'm happy with it. However there are 2 things where i see the need for improvement.

 First the easy one, the volume pot. Atleast on my DAC1 the L/R Balance varies more than i'd like throughout the range. That means the headphone out is unusable if you don't attentuate the volume in the digital domain to reach the (very small!) sweetspot of the poti. That's just not good enough for 1300 Euro, sorry._

 

Any potentiometer that has a 'full-off' position will have a short range at the beginning where the channels don't track well. This is because the wipers are completely off the elements at 'off', and the first part of the range is the steep part of the logarithmic curve. 

 The potentiometers we use track very tightly after the first 10-15% of their range. If your's isn't tracking well beyond the first 10-15%, your unit may have a defective pot.

  Quote:


  Originally Posted by *bizkid* /img/forum/go_quote.gif 
_The second one goes a little deeper, i'm not happy with the resampling algo that is used, especially since EVERYTHING get's resampled to the DACs optimal rate (which makes sence) and there's no way turning that off thus limiting the DACs capability. The problem is that the resampling algo in the DAC1 doesn't appear to be a good one, atleast sonically.

 You can see measured comparisons of various resampling soft/hardware here:
SRC Comparisons

 Yesterday i ABX'd a propper resampled 96khz file and the original 44khz file (thus the DAC1 has more resampling to do). Out of 10tries i scored 100%._

 

I'm not sure I understand your testing procedure, but it sounds like you were ABX'ing the SRC performance of the software. The DAC1 will resample everything to 110 kHz, and the performance of that SRC won't vary based on the incoming sample rate. 

 Can you describe in more detail how you determined your opinion of the DAC1's SRC?

 Thanks,
 Elias


----------



## HeadLover

BTW, why did it upsample from the USB only to 110MH and not the 192KHZ it can ?
 And if it so, isn't it better try and use it with COAX or what?


----------



## EliasGwinn

Quote:


  Originally Posted by *Lil' Knight* /img/forum/go_quote.gif 
_After reading about 50 pages of this thread, I decide to pull the trigger on the DAC1 USB. Hopefully, it'll be a big improvement from my DacMagic.
 Just some small questions:
 + Since it already has a native USB driver, I don't have to use ASIO4ALL with foobar2k, right?
 + How do the headphones out on the DAC1 drive low impedance phones, like the ESW10JPN and PK1? I'll sell my portable amp if I buy the DAC1.
 + Can I use both the headphones out and XLR outputs at the same time?
 + And lastly, anyone tried it with some tube amps? How do they sound together?_

 

Hello Lil' Knight,

 - No, you don't have to use ASIO4ALL. Btw, are you using XP or Vista? ...shouldn't make a difference, just curious.

 - The HPA2 (the DAC1's headphone amp) maintains its performance even with 30 ohm headphone loads, mainly because of the 0-ohm output impedance.

 - Yes, you can use both headphones and XLR at the same time. However, the left headphone jack has a feature that automatically mutes the XLR and RCA outputs whenever headphones are plugged in. This feature makes it easy to go from loudspeakers to headphones without having to turn off your amplifier, etc. This feature is defeatable if you don't want it. Also, the right-hand headphone jack doesn't have this feature.

 - I don't have much experience with the DAC1 driving a tube amp, but a lot of DAC1 users do that.

 Thanks,
 Elias


----------



## EliasGwinn

Quote:


  Originally Posted by *dmashta* /img/forum/go_quote.gif 
_i have always been curious about this but never quite sure what it means. on the dac1 pre, i've adjusted the volume pot through its range but never noticed a 'sweet spot'. sure it sounds best within a certain range but that's just my loudness preference, no? so the inner workings of the pot notwithstanding, what exactly is this sweet spot we talking about?_

 

Dmashta,

 The sound of the pot doesn't vary, but the L/R track balance is optimized after the first 10-15% of the full rotation. In other words, it is ideal to operate the volume pot beyond the 9-10 o'clock position.

 As I mentioned earlier, this is simply due to the nature of 2-channel, full-off, logarithmic pots. The first 10% above 'full-off' is a huge jump in value...too much to expect an inter-channel accuracy less then 0.1 dB, which is required for good L/R balance.

 Thanks,
 Elias


----------



## dmashta

Quote:


  Originally Posted by *EliasGwinn* /img/forum/go_quote.gif 
_Dmashta,

 The sound of the pot doesn't vary, but the L/R track balance is optimized after the first 10-15% of the full rotation. In other words, it is ideal to operate the volume pot beyond the 9-10 o'clock position.

 As I mentioned earlier, this is simply due to the nature of 2-channel, full-off, logarithmic pots. The first 10% above 'full-off' is a huge jump in value...too much to expect an inter-channel accuracy less then 0.1 dB, which is required for good L/R balance.

 Thanks,
 Elias_

 

Okay, thanks. I've had no issue with channel imbalance (not that i listen to music that low) so i went looking for a sweet spot of another kind.


----------



## bizkid

Quote:


  Originally Posted by *EliasGwinn* /img/forum/go_quote.gif 
_Dmashta,

 The sound of the pot doesn't vary, but the L/R track balance is optimized after the first 10-15% of the full rotation. In other words, it is ideal to operate the volume pot beyond the 9-10 o'clock position.

 As I mentioned earlier, this is simply due to the nature of 2-channel, full-off, logarithmic pots. The first 10% above 'full-off' is a huge jump in value...too much to expect an inter-channel accuracy less then 0.1 dB, which is required for good L/R balance.

 Thanks,
 Elias_

 

Hi Elias, then i just have bad luck with my pot, the L/R balance varies until i reach the sweetspot (around 1clock) and then again from 2oclock on.

 I'll make some measurements and upload them here to make my point a little clearer concering the SRC. Anyway i'm curious if the AD1896 can be used without SRC but still make use of it's re-clocking capabilities? The chip has a "bypass" pin but it's not really clear to me if it just bypasses the SRC or the whole chip alltogether.


----------



## EliasGwinn

Quote:


  Originally Posted by *HeadLover* /img/forum/go_quote.gif 
_BTW, why did it upsample from the USB only to 110MH and not the 192KHZ it can ?
 And if it so, isn't it better try and use it with COAX or what?_

 

I can explain it to you in full detail if you like, but I've explained it earlier in this thread (here, for example). 

 The short answer is this: all digital inputs (coax, USB, optical, XLR) are re-sampled to 110 kHz because this frequency optimizes the performance of the digital filters, even better then 192 kHz.

 Thanks,
 Elias


----------



## HeadLover

Quote:


  Originally Posted by *EliasGwinn* /img/forum/go_quote.gif 
_I can explain it to you in full detail if you like, but I've explained it earlier in this thread (here, for example). 

 The short answer is this: all digital inputs (coax, USB, optical, XLR) are re-sampled to 110 kHz because this frequency optimizes the performance of the digital filters, even better then 192 kHz.

 Thanks,
 Elias_

 

Ok thanks!


----------



## EliasGwinn

Quote:


  Originally Posted by *bizkid* /img/forum/go_quote.gif 
_Hi Elias, then i just have bad luck with my pot, the L/R balance varies until i reach the sweetspot (around 1clock) and then again from 2oclock on._

 

Who did you buy the unit from?

  Quote:


  Originally Posted by *bizkid* /img/forum/go_quote.gif 
_I'll make some measurements and upload them here to make my point a little clearer concering the SRC. Anyway i'm curious if the AD1896 can be used without SRC but still make use of it's re-clocking capabilities? The chip has a "bypass" pin but it's not really clear to me if it just bypasses the SRC or the whole chip alltogether._

 

Well, re-clocking is actually SRC. Even if we are talking about re-clocking to the 'same' sample rate (i.e., 96k -> 96k), the algorithm used will be the same. No two clocks are exactly the same (96.0001 kHz -> 95.9999 kHz). So, re-clocking will always convert the sample rate, and the choice of final sample rate won't affect the performance, even if its the 'same' sample rate.

 The thing about your ABX test is that both 'A' and 'B' are going through the exact same SRC/algorithm, and so any differences between the two are inherent in the audio data.

 Thanks,
 Elias


----------



## poo

Quote:


  Originally Posted by *EliasGwinn* /img/forum/go_quote.gif 
_I can explain it to you in full detail if you like, but I've explained it earlier in this thread (here, for example)._

 

To be honest, I would find it really helpful if you could explain it in full detail; but also in lay terms if possible. The post that you linked to was helpful in regards to a general understanding, but for someone like myself with no technical understanding of what it _actually_ means, your explanation just resulted in me having faith that you know what your talking about. I'm not suggesting you don't - just that _I'm_ really non the wiser 
	

	
	
		
		

		
		
	


	




 So when you say:  Quote:


  Originally Posted by *EliasGwinn* /img/forum/go_quote.gif 
_all digital inputs... are re-sampled to 110 kHz because this frequency optimizes the performance of the digital filters, even better then 192 kHz._

 

 it sounds like (for example) there could have been filters to better handle the 192kHz signal, but we didn't use them for "other reasons".


----------



## EliasGwinn

Quote:


  Originally Posted by *poo* /img/forum/go_quote.gif 
_To be honest, I would find it really helpful if you could explain it in full detail; but also in lay terms if possible. The post that you linked to was helpful in regards to a general understanding, but for someone like myself with no technical understanding of what it actually means, your explanation just resulted in me having faith that you know what your talking about. I'm not suggesting you don't - just that I'm really non the wiser 
	

	
	
		
		

		
		
	


	




 So when you say: "all digital inputs... are re-sampled to 110 kHz because this frequency optimizes the performance of the digital filters, even better then 192 kHz." 
 ...it sounds like (for example) there could have been filters to better handle the 192kHz signal, but we didn't use them for "other reasons"._

 

I'll try to explain it in layman's terms...feel free to ask me to clarify anything.

 The filtering is done by performing math operations on the digital data. However, only so many operations can be performed per second (thats why your computer slows down whenever there are too many things happening). With such an increase in data (192 kHz is 2x the amount of data as 96k and 4x as much as 48 kHz), the filter must reduce the amount of math operations to keep up with the data. With fewer math operations, the precision of the filter is compromised.

 The highest amount of data the filter can process before reducing the math is about 117 kHz. We choose 110 kHz to be absolutely certain that the filter is always in the full-performance mode.

 Thanks,
 Elias


----------



## HeadLover

amm, so why not just put a better and faster processor?
 How does some DAC's out there goes even up to 768MHZ ?
 It sure be nice if all data that goes into the DAC can be upsampled to 192KHZ


----------



## EliasGwinn

Quote:


  Originally Posted by *HeadLover* /img/forum/go_quote.gif 
_amm, so why not just put a better and faster processor?
 How does some DAC's out there goes even up to 768MHZ ?
 It sure be nice if all data that goes into the DAC can be upsampled to 192KHZ_

 

Keep in mind, just because a DAC goes up to a certain sample rate doesn't mean that the filters perform well. In fact, it probably means the opposite.

 When a circuit is designed, one must consider all the tradeoffs...the strengths and weaknesses of the available technology. Then, the design engineer must decide which tradeoffs to take based on what factors are most important.

 The tradeoff in this case is *system bandwidth* versus *sonic performance*. 

*System bandwidth* is, more or less, the highest analog frequency (aka Nyquist Frequency) of a digital converter, which is always one-half of the sample rate. The system bandwidth of the DAC1's digital filter is 55 kHz, whereas the system bandwidth of a 192 kHz conversion is 96 kHz. 

*Sonic performance*, in this case, is defined by the pass-band ripple and stop-band attenuation. Pass-band ripple is another way of saying frequency-response distortion. Stop-band attenuation is necessary to prevent aliasing, which is a type of digital distortion that is very unflattering. Stop-band is the audio at frequencies above the Nyquist Frequency. If these frequencies are not properly filtered out, they will be 'folded' back into the audio band as a tone with a frequency equal to f_audio - f_Nyq. If you're unfamiliar with what aliasing sounds like, listen to a 96 kbps MP3.

 We decided that we would rather have a very high performing circuit with an analog bandwidth that is limited to 55 kHz then a less accurate circuit whose bandwidth goes to 96 kHz.

 Thanks,
 Elias


----------



## HeadLover

I see
 Well I agree with you here, I mean I (and I guess others) will prefer a much better Sound Quality, than just numbers


----------



## bizkid

Grrr now i found the culprit of my resampling troubles. 
	

	
	
		
		

		
		
	


	




 The problem with my ABX test was... windows. When i set it up some time ago I set the audio options of the DAC1 connected via USB to 24bit 96khz in windows, and using direct sound in foobar.

 That means when i compared the 44khz file to the 96khz one i actually ABXd windows resampling algo which resampled the 44khz stream to 96khz. OUCH! Sorry for the missunderstanding to Elias and the other people i were in disagreement.

 One note for windows users: If you listen to 44khz most of the time, it's the best to use that value in the windows audio setup. You DON'T want to resample twice, once in software (be it windows itself or foobar), one time in the DAC 1. I measured the results of 2x software resampling and it's a big NO NO. If anyone is curious i can post the graphs.

 Atleast we brought this thread back to live a bit


----------



## HeadLover

Quote:


  Originally Posted by *bizkid* /img/forum/go_quote.gif 
_Grrr now i found the culprit of my resampling troubles. 
	

	
	
		
		

		
			





 The problem with my ABX test was... windows. When i set it up some time ago I set the audio options of the DAC1 connected via USB to 24bit 96khz in windows, and using direct sound in foobar.

 That means when i compared the 44khz file to the 96khz one i actually ABXd windows resampling algo which resampled the 44khz stream to 96khz. OUCH! Sorry for the missunderstanding to Elias and the other people i were in disagreement.

 One note for windows users: If you listen to 44khz most of the time, it's the best to use that value in the windows audio setup. You DON'T want to resample twice, once in software, one time in the DAC 1. I measured the results of 2x software resampling and it's a big NO NO. If anyone is curious i can post the graphs.

 Atleast we brought this thread back to live a bit 
	

	
	
		
		

		
		
	


	


_

 

You should use VISTA and WASAPI
 Than it doesn't matter, WASAPI will take control and set the right stuff (it you put the allow control on your volume control panel)


----------



## bralk

When can we expect the new pre with wireless remote?


----------



## HeadLover

Are you sure you want a remote?
 Putting more stuff into the unit, and the remote, might reduce the sound quality, at least I think so.

 Bue, I am sure will like to hear if there any kind of updates to the DAC1 soon or in the future


----------



## poo

Cool thanks for the help understanding Elias!

 So what is in the way of increasing System Bandwidth? What steps need to be taken to increase system bandwidth and maintain quality? There are a few (admittedly much more expensive) DACs out there (like the Berkeley Audio Design Alpha DAC) which seem to do both successfully. I'm not asking you to compare or comment on this or any other DAC, just giving an example.

 Is it fair to say that both high system bandwidth and high sonic performance are currently achievable in a device if the budget allows?


----------



## Scrith

Quote:


  Originally Posted by *EliasGwinn* /img/forum/go_quote.gif 
_Can you describe in more detail how you determined your opinion of the DAC1's SRC?_

 

One thing I've tried is doing resampling on the computer end (using various resamplers) and letting the DAC1 playback this already-resampled data. It definitely sounds different, and many would argue that it sounds better to use a high-quality resampler on the computer-end to get the data to 24/96 and then send it to the DAC1. This leads to the conclusion that perhaps the resampling that is going on in the DAC1 isn't as good as it could be because the DAC1 seems to sound better when it is fed data that has already been resampled to a very high rate with a high-quality resampler (giving the DAC1's resampler more data to work with when it then converts this 96K input to 110K).

 As a test, try using the Secret Rabbit resampler in Foobar2000 to resample some audio data to 96K and compare how that sounds with un-resampled playback of the same data on a DAC1.


----------



## HeadLover

amm maybe because you give him 24 bit and 96KHZ, it is much better just plain 16/44
 not?

 I wonder what Elias will say

 Also, maybe the more upsample, you just lose details and it seem to be more "smooth" while it is worse ?


----------



## bizkid

Scrith if you read my last few posts you will see that this is exactly what i thought, and i was wrong. Doublecheck if your OS isnt resampling the audio stream.

 I will show you my SRC measurements to proof my point:






 This is a sine sweep resampled from 44khz to 110khz via izoptope rx (demo is a free download, everybody can try this out themself). The RX SRC algo is one of the best avaible in software form.






 This is a sine sweep from the same 44khz file resampled to 110khz in an old version of soundforge. Not as good as the RX one but there's also much worse.






 Here i simulated using a good SRC algo (RX) to go from 44khz to 96khz, and then with an average one (soundforge) to 110khz. Compare it to the graphs above and it's more than obvious that additional resampling, even with a good algo, is always worse.

 And this is exactly what happens if you resample in foobar and then again in the DAC1. It's a no-no. Ofcourse this doesn't mean that there's no audible difference but if you want a "clean" signal, don't do additional resampling.


----------



## ted betley

can you elaborate on what the plots are showing? what is good/bad?


----------



## bizkid

Quote:


  Originally Posted by *ted betley* /img/forum/go_quote.gif 
_can you elaborate on what the plots are showing? what is good/bad?_

 

Please visit this website i mentioned just a few posts ago: SRC Comparisons and click on "help", this explains everything.
 The plots are just 1 part of many measurements but enough to show my point.


----------



## Scrith

Quote:


  Originally Posted by *bizkid* /img/forum/go_quote.gif 
_And this is exactly what happens if you resample in foobar and then again in the DAC1. It's a no-no. Ofcourse this doesn't mean that there's no audible difference but if you want a "clean" signal, don't do additional resampling._

 

Thanks for the enlightening information. I personally have given up on resampling in Foobar2000 prior to sending it to the DAC1 USB because, for some reason, listening to music that I was resampling myself (using the Secret Rabbit resampler that some people around here seem to worship) seemed to get on my nerves after a few minutes.

 That being said...you seem to be basing your conclusion on the excellent job that the izoptope rx resampler does when going from 44K to 110K (as shown in your first graph). How do you know the one in the DAC1 is doing that good of a job? Do you have a graph that shows what 44K data sent to a DAC1 ends up looking like after it gets resampled there?


----------



## bizkid

Quote:


  Originally Posted by *Scrith* /img/forum/go_quote.gif 
_Thanks for the enlightening information. I personally have given up on resampling in Foobar2000 prior to sending it to the DAC1 USB because, for some reason, listening to music that I was resampling myself (using the Secret Rabbit resampler that some people around here seem to worship) seemed to get on my nerves after a few minutes.

 That being said...you seem to be basing your conclusion on the excellent job that the izoptope rx resampler does when going from 44K to 110K (as shown in your first graph). How do you know the one in the DAC1 is doing that good of a job? Do you have a graph that shows what 44K data sent to a DAC1 ends up looking like after it gets resampled there?_

 


 Hi Scrith, my point is not that the RX is doing such a good job. The DAC1 SRC itself will probably not look anything like that. Not even Weiss has a hardware based SRC that good (but software).
 The thing is, even if you use RX to go to 96khz (it will look just as clean as the first graph), adding the 2nd SRC to 110khz seems to multiply the additional SRCs artifacts (3rd pic). It will always look much worse than going directly to 110khz no matter what SRC algo is used, i tried lot's of different SRCs. So it doesn't really matter how good or bad the SRC in the DAC1s AD1896 is, if you SRC before it, it's artifacts will be multiplied by an unknown factor.


----------



## HeadLover

http://www.head-fi.org/forums/f46/go...review-402263/

 Hope you like it guys 
	

	
	
		
		

		
		
	


	




 Like I have said there, I wished there was a new USB with 192/24 input, and upsampling and filters even up to 768MHZ/32BIT with out a loss of quality.

 And an LCD screen for seeing the input I get (bits and rate)

 SQ is great in that unit!


----------



## EliasGwinn

Hey folks!

 I just got back from Anaheim, CA, at the NAMM show. I'll try to catch up with your questions as quickly as possible.

  Quote:


  Originally Posted by *poo* /img/forum/go_quote.gif 
_Cool thanks for the help understanding Elias!

 So what is in the way of increasing System Bandwidth? What steps need to be taken to increase system bandwidth and maintain quality? There are a few (admittedly much more expensive) DACs out there (like the Berkeley Audio Design Alpha DAC) which seem to do both successfully. I'm not asking you to compare or comment on this or any other DAC, just giving an example.

 Is it fair to say that both high system bandwidth and high sonic performance are currently achievable in a device if the budget allows?_

 

The problem is this: we can't buy a D/A chip that performs well at 192 kHz. No matter how much money we wanted to spend, they simply don't exist.

 The only way we could increase bandwidth without compromising sonic quality is to manufacture our own D/A chip. Until we have a few hundred million dollars to invest in a silicon wafer production facility, we will have to buy D/A chips from the current companies. Unfortunately, they don't make any that perform well at 192kHz without compromising sonic quality.

 As for the other DAC's on the market that convert at higher sample rates, you would see filter-related short-comings if you did an apples-to-apples measurement. People may say that one sounds better at 192 kHz...and they are entitled to their opinion...but the fact is that the performance of current technology is compromised at those high sample rates. Even the manufacturer of the D/A chip says so...

 The Berkely Alpha DAC doesn't convert at 192k either. I'm not sure what sample-rate they use because they don't specify, but the website says it converts at "almost 176.4kHz". I wouldn't be surprised if they designed it using the same idea as the DAC1. They wouldn't be the first to learn from our design...

 Remember, friends... we could take the DAC1 circuit and put it in a big, fancy chassis with extra features and charge ten times the cost of the DAC1. It would sound the same, but it would be a $10,000 DAC. We don't do that because we are engineers, not marketing or business people. And, honestly, we don't believe that is a sustainable way to do business. An honest company making a great product at an honest price will be better off in the long run.

 Thanks,
 Elias


----------



## nae45ro

Finally my DAc-1 is here. Sound quality is as expected excellent. Tons of micro-level details and the agrssivity of my M-Audio firewire soundboard is gone. Only one problem. You recommend keeping the volume at about 10 o'clock for normal listening. With the volume set to 12 o'clock on my headphone amplifier (Bada Ph-12), I have to keep the volume knob on DAC-1 close to the maximum for normal listening volume. I switched the jumpers from -20 (default)to -10 and then 0 but something weird happened > The volume was different on left and right headphone with these 2 settings !


----------



## Bostonears

Quote:


  Originally Posted by *EliasGwinn* /img/forum/go_quote.gif 
_The only way we could increase bandwidth without compromising sonic quality is to manufacture our own D/A chip. Until we have a few hundred million dollars to invest in a silicon wafer production facility, we will have to buy D/A chips from the current companies._

 

You don't need to own your own wafer fab to have your own chips. There are many so-called "fabless semiconductor" companies that have IC chips custom made by one of the existing wafer fabs. (I used to work for a company that had its own very high performance DSP chips custom made.) You do, however, need to have some very skilled and experienced chip designers.


----------



## EliasGwinn

Quote:


  Originally Posted by *nae45ro* /img/forum/go_quote.gif 
_Finally my DAc-1 is here. Sound quality is as expected excellent. Tons of micro-level details and the agrssivity of my M-Audio firewire soundboard is gone. Only one problem. You recommend keeping the volume at about 10 o'clock for normal listening. With the volume set to 12 o'clock on my headphone amplifier (Bada Ph-12), I have to keep the volume knob on DAC-1 close to the maximum for normal listening volume. I switched the jumpers from -20 (default)to -10 and then 0 but something weird happened > The volume was different on left and right headphone with these 2 settings !_

 

Make sure all 4 jumpers were set equally. If any of the jumpers aren't set equally, there will be a channel imbalance.

 Also, we recommend having the volume control _above_ 10 o'clock, not _at_ 10 o'clock. In other words, we recommend having the volume anywhere between 10 o'clock and maximum. It is perfectly fine to have the volume control near maximum. 

 Thanks,
 Elias


----------



## EliasGwinn

Quote:


  Originally Posted by *Scrith* /img/forum/go_quote.gif 
_One thing I've tried is doing resampling on the computer end (using various resamplers) and letting the DAC1 playback this already-resampled data. It definitely sounds different, and many would argue that it sounds better to use a high-quality resampler on the computer-end to get the data to 24/96 and then send it to the DAC1. This leads to the conclusion that perhaps the resampling that is going on in the DAC1 isn't as good as it could be because the DAC1 seems to sound better when it is fed data that has already been resampled to a very high rate with a high-quality resampler (giving the DAC1's resampler more data to work with when it then converts this 96K input to 110K).

 As a test, try using the Secret Rabbit resampler in Foobar2000 to resample some audio data to 96K and compare how that sounds with un-resampled playback of the same data on a DAC1._

 

Scrith,

 In this expirament, you are changing one variable: software SRC on versus off. Any differences you are hearing are due to that variation.

 To test the SRC of the DAC1, you would need to keep all of the other parameters constant and have the on/off of the DAC1's SRC be the only variable. 

 Thanks,
 Elias


----------



## EliasGwinn

Quote:


  Originally Posted by *Bostonears* /img/forum/go_quote.gif 
_You don't need to own your own wafer fab to have your own chips. There are many so-called "fabless semiconductor" companies that have IC chips custom made by one of the existing wafer fabs. (I used to work for a company that had its own very high performance DSP chips custom made.) You do, however, need to have some very skilled and experienced chip designers._

 

True...very true


----------



## nae45ro

Thanks Elias, I'll keep it this way then. problem number 2 :

 Each time I close the PC, the DAC stays on. After reopening the PC, I have no sound, no matter the audio or video software used (foobar, vlc player...). Only after pulling out the power cord or the usb cable and repluging them the sound is on again. What can be the cause ?

 PS : When I have no sound, foobar plays the songs like it knows the is a soundboard attached but there's no sound (when there is no soundboard, the songs don't play at all)


----------



## EliasGwinn

Quote:


  Originally Posted by *nae45ro* /img/forum/go_quote.gif 
_Thanks Elias, I'll keep it this way then. problem number 2 :

 Each time I close the PC, the DAC stays on. After reopening the PC, I have no sound, no matter the audio or video software used (foobar, vlc player...). Only after pulling out the power cord or the usb cable and repluging them the sound is on again. What can be the cause ?

 PS : When I have no sound, foobar plays the songs like it knows the is a soundboard attached but there's no sound (when there is no soundboard, the songs don't play at all)_

 

When you re-open the PC, are the DAC1 LED's illuminated? You will need to see the top and bottom blue LED's of the DAC1 for the PC audio to play. If you don't see them, switch the input switch until the top (3) and bottom (1) LED's are illuminated.

 Thanks,
 Elias


----------



## nae45ro

Yep, tehy are illuminated all the time no matter if computer is on or off !


----------



## EliasGwinn

Are you shutting down the computer, or simply hibranating?


----------



## nae45ro

Shutting down


----------



## EliasGwinn

So, let me try to understand:

 You have Foobar open, and you're playing audio through the DAC1 USB (via the USB input). 

 Then, you shut-down the computer. 

 Then you start the computer. 

 Then you open Foobar. However, when you press play, there is no sound from the DAC1 USB?

 Is that all correct?


----------



## nae45ro

Yep, all correct. Foobar is playing the file but there's no sound. I have to unplug the DAc from the computer and replug it again to work !


----------



## EliasGwinn

after you start your computer, will WMP or iTunes play through the DAC1?


----------



## nae45ro

Hmm, weird. Today it worked normally !


----------



## nae45ro

Back again. Same problem. It doesn't work in WMP or iTunes either !


----------



## infinitesymphony

It sounds like your computer isn't seeing/remembering the DAC1. With these kinds of sporadic connection issues, it might be the computer (Windows, the motherboard, etc.). Have you tried plugging into a different USB port?


----------



## EliasGwinn

Quote:


  Originally Posted by *nae45ro* /img/forum/go_quote.gif 
_Back again. Same problem. It doesn't work in WMP or iTunes either !_

 

Nae45ro,

 What is the current status of this issue?

 Thanks,
 Elias


----------



## Lil' Knight

Just got my DAC1 usb this morning. Definitely a big improvement from my last DAC 
	

	
	
		
		

		
			





 I have one question, is it normal to leave the volume knob on the DAC1 at 100% and adjust to main volume on my headamp?


----------



## EliasGwinn

Quote:


  Originally Posted by *Lil' Knight* /img/forum/go_quote.gif 
_Just got my DAC1 usb this morning. Definitely a big improvement from my last DAC 
	

	
	
		
		

		
		
	


	




 I have one question, is it normal to leave the volume knob on the DAC1 at 100% and adjust to main volume on my headamp?_

 

You can do it that way if you want. The sound quality will be the same.

 Thanks,
 Elias


----------



## Lil' Knight

Quote:


  Originally Posted by *EliasGwinn* /img/forum/go_quote.gif 
_You can do it that way if you want. The sound quality will be the same.
_

 

Thanks for the response, Elias.
 It looks like that it's not normal, right? It's the best way to connect my balanced amp to the DAC1 usb? I've already leave 100% volume in foobar.


----------



## EliasGwinn

Quote:


  Originally Posted by *Lil' Knight* /img/forum/go_quote.gif 
_Thanks for the response, Elias.
 It looks like that it's not normal, right? It's the best way to connect my balanced amp to the DAC1 usb? I've already leave 100% volume in foobar._

 

It isn't the most common way to connect, but it is normal. The alternative would be to use "Calibrated" (i.e., fixed) mode. At factory setting, this is equivalent to having the volume control pinned at +/- 95%. This is no better or worse then the method you are using.

 Thanks,
 Elias


----------



## Lil' Knight

That's great! Thanks for your big help.


----------



## Lil' Knight

Back with another question lol.
 So I followed the instruction to setup the DAC1 with foobar on Benchmark's website.
 In the Setting, I chose the output WASAPI: Speaker(Benchmark 1.0) and Buffer Length is 1120ms.
 I didn't add anything to the ASIO Virtual Device.

 Is that the best way to configure? What's the best Buffer Length?
 And, I've noticed a big drop in memory when using the WASAPI. The memory keeps falling down after 3 or 4 hours using constantly. If I choose DS: Primary Sound Driver, everything is fine and there's no drop in the memory. I still don't know what happens


----------



## Lil' Knight

Back with another question lol.
 So I followed the instruction to setup the DAC1 with foobar on Benchmark's website.
 In the Setting, I chose the output WASAPI: Speaker(Benchmark 1.0) and Buffer Length is 1120ms.
 I didn't add anything to the ASIO Virtual Device.

 Is that the best way to configure? What's the best Buffer Length?
 And, I've noticed a big drop in memory when using the WASAPI. The memory keeps falling down after 3 or 4 hours using constantly. If I choose DS: Primary Sound Driver, everything is fine and there's no drop in the memory. I still don't know what happens


----------



## Quaddy

under playback-->output try 650ms as the buffer, this is the sweet spot for me, after hours of experimentation, in fact anything above that figure gave me choppy playback, and now the memory imprint is a lot less, strange but true....


----------



## EliasGwinn

Lil Knight,

 The buffer setting will be dependent on your specific system. I would suggest adjusting it until everything works smoothly.

 Let me know if you need more assistance.

 Thanks,
 Elias


----------



## Bojamijams

I have my buffer set to largest possible with ASIO. The buffer is only for recording where you want the lowest latency possible.. for playback, latency is not an issue, so set it to large and you won't get any chops 

 I don't understand how a large buffer can give you choppy playback unless you have very little free RAM.


----------



## Quaddy

Quote:


  Originally Posted by *Bojamijams* /img/forum/go_quote.gif 
_I have my buffer set to largest possible with ASIO. *The buffer is only for recording* where you want the lowest latency possible.. for playback, latency is not an issue, so set it to large and you won't get any chops 

*I don't understand* how a large buffer can give you choppy playback unless you have very little free RAM._

 


 me neither! 
	

	
	
		
		

		
		
	


	




 oh and the buffer i am talking about is playback buffer, nothing to do with recording.

 i am using wasapi on vista with foobar

 2GB ram

 choppy at anything over 650, i think it only crops up on very few setups thats 650ms trick, as there is ony a few out there with the same issue.

 as elias, said you need to experiment with your setup, find that sweet spot - good luck!


----------



## Lil' Knight

Thanks all for the advice.
 I still haven't figured out why there's a constant drop in memory in a long run 
	

	
	
		
		

		
		
	


	



 And, do you guys add anything to the ASIO Virtual Devices?


----------



## tubaman

I'm getting a DAC1 PRE and I have a question regarding the analog input. With the analog input as the selected source and the output set at "calibrated," is it equal (volume) as turning the vol. all the way up (or so) using "variable?" 

 I plan to connect the tuner to my DAC1 PRE and then output to the headphone amp.


----------



## Bostonears

Quote:


  Originally Posted by *tubaman* /img/forum/go_quote.gif 
_I'm getting a DAC1 PRE and I have a question regarding the analog input. With the analog input as the selected source and the output set at "calibrated," is it equal (volume) as turning the vol. all the way up (or so) using "variable?" 

 I plan to connect the tuner to my DAC1 PRE and then output to the headphone amp._

 

If you're only planning to use the headphone amp in the DAC1 Pre, the "calibrated" setting won't make any difference, because it only affects the analog (preamp) outputs on the rear.

 Nevertheless, to answer your question relative to the analog _outputs_ when using analog inputs, the output volume at the "calibrated" setting is lower than the variable volume turned all the way up, at least on my unit with the jumpers in the default positions.


----------



## tubaman

Quote:


  Originally Posted by *Bostonears* /img/forum/go_quote.gif 
_If you're only planning to use the headphone amp in the DAC1 Pre, the "calibrated" setting won't make any difference, because it only affects the analog (preamp) outputs on the rear.

 Nevertheless, to answer your question relative to the analog outputs when using analog inputs, the output volume at the "calibrated" setting is lower than the variable volume turned all the way up, at least on my unit with the jumpers in the default positions._

 

Thanks for the reply. Is the volume much lower, or just a little lower? I think in an earlier post it was mentioned that vol. at calibrated is 95% of that with volume nob. all the way up in variable mode; maybe that explains it. 

 BTW I didn't make myself clear, I meant to connect the analog output (unbalanced) to an external headphone amp.


----------



## Scrith

Here is a link to a very interesting question-and-answer round with several computer-based audio experts on the state of current computer audio technology.

 Some of the most interesting answers were from Charles Hansen of Ayre Acoustics. Ayre is going to be releasing a new USB DAC in March that uses the asynchronous approach to USB audio (as I told Elias about several months ago). Here are Mr. Hansen's thoughts on a question about USB and other computer interfaces to DACs:

  Quote:


 USB - There are a ton of USB DACs out there that suck. The problem is that Burr-Brown released a bunch of cheap chips that had both a DAC and USB support. (Some also have a digital out, as used in the Wadia dock.) They are super easy to use and require no programming. But they have two problems. The first is that they are limited to a max resolution of 48/16. The second is that the jitter performance is HORRIBLE. We are talking literally 100x worse than a good one-box CD player.

 There is another chip from Burr-Brown called the TAS1020B. It is programmable, but it is super hard to do so. Only two people have done it in audio to my knowledge. There is a third- party developer called Centrance that has written code for the device that allows it to go up to 96/24. This is used by some pro companies like Benchmark and Lavry. Some high end companies like PS Audio and Bel Canto also use it. But they have done nothing to address the HORRIBLE jitter performance. About the only good thing about this approach is that the DAC is not built in. Therefore, many of these guys use jitter reduction devices, typically a sample-rate converter (SRC), before the DAC chip.

 But the best way to do USB is what is called "asynchronous" mode. This means that the master audio clock is in the DAC box instead of the computer. This is the only way to get jitter performance as low as a good one-box player. Gordon Rankin of Wavelength wrote the code for the TAS1020B to do this, and we are licensing it from him. dCS also has some new add-on boxes for their big stacks of boxes that uses "asynchronous" mode. Everything else is seriously flawed by comparison.

 Using an SRC chip to reduce jitter is somewhat controversial from a sonic standpoint. (They tend to measure well enough.) Basically, what they do is throw away all of the original data and calculate new data that is their best guess as to what the data would have been if there hadn't been any jitter. The "best guess" refers to the algorithm used. The latest ones measure extremely well, but still haven't convinced everyone from a sonic standpoint. 
 

Any thoughts on those comments, Elias?


----------



## Quaddy

good linkage, am gonna have a good read now. thanks.


----------



## Lord Chaos

Scene: Inside the Benchmark Media Systems boardroom

 President: "Folks, we've been busted."
 Engineer 1: "I told you we shouldn't have used that cheap chip."
 Engineer 2: "But it sounds really good!"
 President: "I'm afraid that's no longer good enough. We have to have the words to back it up now."
 Engineer 1: "What do you want us to do?"
 President: "I need you to scrap all of your DAC designs and start over. Oh... we'll need the new design by next Thursday."
 Engineer 1: "Yes, Sir!"
 Engineer 2, as they walk out the door: "But... they sound so good!"


----------



## tubaman

Let's not forget the DAC1 USB is only $300 more than the DAC1, which is $1,000.


----------



## EliasGwinn

Quote:


  Originally Posted by *tubaman* /img/forum/go_quote.gif 
_I'm getting a DAC1 PRE and I have a question regarding the analog input. With the analog input as the selected source and the output set at "calibrated," is it equal (volume) as turning the vol. all the way up (or so) using "variable?" 

 I plan to connect the tuner to my DAC1 PRE and then output to the headphone amp._

 

Hello Tubaman,

 At factory setting, the 'Calibrated' output will about 3 dB below the maximum variable setting (assuming the input level isn't too high to limit the output).

 Thanks,
 Elias


----------



## EliasGwinn

Quote:


  Originally Posted by *Scrith* /img/forum/go_quote.gif 
_Here is a link to a very interesting question-and-answer round with several computer-based audio experts on the state of current computer audio technology.

 Some of the most interesting answers were from Charles Hansen of Ayre Acoustics. Ayre is going to be releasing a new USB DAC in March that uses the asynchronous approach to USB audio (as I told Elias about several months ago). Here are Mr. Hansen's thoughts on a question about USB and other computer interfaces to DACs:



 Any thoughts on those comments, Elias?_

 

Mr. Hansen may be an eloquent speaker ("There are a ton of USB DACs out there that suck"), but I must disagree with him on a few points:

 1. Centrance didn't "program" the TAS1020B. Centrance sells a firmware solution that was co-developed between them and Benchmark.

 2. Lavry doesn't have a USB product, as far as I know. Could be wrong...

 3. As I mentioned in previous posts (somewhere around #2100), asynchronous USB does not reduce jitter to levels low enough to eliminate the need for a jitter mechanism in the DAC. 

 4. As I also mentioned in previous posts (somewhere around #2100), asynchronous necessitates SRC by the computer, which is WAY worse then the ASRC in our box.

 5. "Everything else is seriously flawed." Coming from a DAC manufacturer, I question the objectivity of this statement.

 6. "Basically, what they do is throw away all of the original data and calculate new data that is their best guess as to what the data would have been if there hadn't been any jitter. The "best guess" refers to the algorithm used. The latest ones measure extremely well, but still haven't convinced everyone from a sonic standpoint." With the exception of the statement "measure extremely well", nothing in this quote is true. "still haven't convinced everyone from a sonic standpoint."...does this mean that no one thinks the DAC1 sounds good???

 Thanks,
 Elias


----------



## Scrith

Thanks for the reply Elias.

 You're definitely right about there being some bias (and misinformation) in Mr. Hansen's replies (the DAC1 sounds and measures remarkably well, by almost all accounts, including mine!). But it could probably be even better, couldn't it? 
	

	
	
		
		

		
			





 On point 3: data that is sent asynchronously does not contain jitter because there is no timing associated with it (it is just data that has been sent to an external device to use at the rate at which it sees fit). One example of something like this is a printer, which is sent data (to be printed) at a rate at which the printer determines (the computer cannot possibly know how quickly the printer can consume the data, so it must wait for a signal from the printer before sending additional data). USB audio can work this way also (I'm not sure whose bright idea it was to have the computer determine the rate at which the data is being sent...probably the same people who figured out S/PDIF), and that is what people mean when they talk about asynchronous USB. If the device in question is buffering this data and asking (and receiving) it such that buffer over- and under-runs do not occur (not a difficult task...this is what USB printers and hard drives are doing) the device can play back this data at a rate that is entirely independent of the rate at which the data arrived (much like a printer can print at its own pace).

 On point 4: why would I need to use SRC on my computer if I am sending data to an external device? If the external DAC is like an audio printer (as described above) it will take the data at whatever rate it arrives and play it back at the expected rate based on its local clock. The data does not need to be resampled to something other than 16/44 in order for the DAC to play it (or am I missing something about DAC chips not being able to use data in standard formats like 16/44 and 24/96?).


----------



## EliasGwinn

Quote:


  Originally Posted by *Lord Chaos* /img/forum/go_quote.gif 
_Scene: Inside the Benchmark Media Systems boardroom

 President: "Folks, we've been busted."
 Engineer 1: "I told you we shouldn't have used that cheap chip."
 Engineer 2: "But it sounds really good!"
 President: "I'm afraid that's no longer good enough. We have to have the words to back it up now."
 Engineer 1: "What do you want us to do?"
 President: "I need you to scrap all of your DAC designs and start over. Oh... we'll need the new design by next Thursday."
 Engineer 1: "Yes, Sir!"
 Engineer 2, as they walk out the door: "But... they sound so good!"_

 

It's actually the same chip, just different implementations. Also, the president is also Engineer #1...John Siao.

 Thanks,
 Elias


----------



## EliasGwinn

Quote:


  Originally Posted by *tubaman* /img/forum/go_quote.gif 
_Let's not forget the DAC1 USB is only $300 more than the DAC1, which is $1,000._

 

Its not a question of cost, its just a different implementation of the same USB chip. We looked at the Asynchronous mode in Fall '07, but when we learned that it would induce SRC by the computer, we quickly realized it was a really bad idea.

 Thanks,
 Elias


----------



## EliasGwinn

Quote:


  Originally Posted by *Scrith* /img/forum/go_quote.gif 
_You're definitely right about there being some bias (and misinformation) in Mr. Hansen's replies (the DAC1 sounds and measures remarkably well, by almost all accounts, including mine!). But it could probably be even better, couldn't it? 
	

	
	
		
		

		
		
	


	


_

 

Perhaps, but I don't think this would help at all. Even if this method as perfect as they claim it is, it wouldn't affect the performance of the DAC1 because jitter is a non-issue. 

 You might say, "You could eliminate the ASRC if there was no jitter!" Sure, but then the digital filters wouldn't perform as well. There are major sonic benefits to converting the sample-rate to 110 kHz, and Mr. Hansen isn't considering this at all. He thinks the ASRC is strictly a method to deal with jitter. It does that and much more.

  Quote:


  Originally Posted by *Scrith* /img/forum/go_quote.gif 
_On point 3: data that is sent asynchronously does not contain jitter because there is no timing associated with it (it is just data that has been sent to an external device to use at the rate at which it sees fit). One example of something like this is a printer, which is sent data (to be printed) at a rate at which the printer determines (the computer cannot possibly know how quickly the printer can consume the data, so it must wait for a signal from the printer before sending additional data). (I'm not sure whose bright idea it was to have the computer determine the rate at which the data is being sent...probably the same people who figured out S/PDIF), and that is what people mean when they talk about asynchronous USB._

 

The isn't exactly how asynchronous USB Audio works. What your describing is bulk transfer, where data is sent in packets when needed. USB audio is isochronous transfer. Isochronous means that the data transfer is happening at a regular (periodic) rate. 'Asynchronous', in the case of isochronous USB audio transfer mode, refers to 'not synchronized to the computer.' Instead, it is synchronized to an external clock...the DAC in this case.

  Quote:


  Originally Posted by *Scrith* /img/forum/go_quote.gif 
_If the device in question is buffering this data and asking (and receiving) it such that buffer over- and under-runs do not occur (not a difficult task...this is what USB printers and hard drives are doing) the device can play back this data at a rate that is entirely independent of the rate at which the data arrived (much like a printer can print at its own pace)._

 

Not really...remember, USB audio is 'isochronous', not 'bulk' transfer mode.

  Quote:


  Originally Posted by *Scrith* /img/forum/go_quote.gif 
_On point 4: why would I need to use SRC on my computer if I am sending data to an external device? If the external DAC is like an audio printer (as described above) it will take the data at whatever rate it arrives and play it back at the expected rate based on its local clock. The data does not need to be resampled to something other than 16/44 in order for the DAC to play it (or am I missing something about DAC chips not being able to use data in standard formats like 16/44 and 24/96?)._

 

It's not that YOU need to use SRC on your computer... Its that the computer will have to re-sample the data to correspond to the new clock...the master clock....the clock from the USB audio device. I posted about this previously (somewhere around +/- post# 2100). 

 Example: What if Mr. Hansen's USB audio device says, "Send me all audio data at my sample-rate: 44.1001 kHz (or 48k, or 96k)." USBaudio.sys looks at the audio stack, which is sending audio at 44.1002 from iTunes. How does it reconcile that difference? It must re-sample.

 When we designed the Benchmark/Centrance solution, we said, "Why don't we allow the computer to be master so the sample-rate can stay consistent until it enters the DAC1? Jitter isn't an issue...only bit-transparency. Just send us the proper data, and we'll take care of the rest!!"

 Whereas the asynchronous mode causes SRC...offers a non-transparent data path...and doesn't do anything for jitter... We are quite happy of our solution!






 Thanks,
 Elias


----------



## Scrith

Sorry for the distraction Elias and everyone, but I think this is an interesting technical path to venture down. And so I continue:

 Well I admit that I simplified it a bit in my description of how the asynchronous mode for USB audio works (I didn't mean to imply it was using a bulk transfer mode, like hard drives and printers, but merely to describe the asynchronous mode that it is using in those terms, which I think is viable because it is controlling the outgoing data rate, like those devices are doing). In the Asynchronous mode of USB audio, the endpoint (e.g. DAC) cannot tell the source (e.g. computer) to stop and start, but it can tell the computer to slow down (when the DAC's buffer, which is being consumed based entirely upon a clock local to the DAC, is getting full) or speed up (when its buffer is getting low). The computer does not decide to send data at a certain rate (as your iTunes at 44.10002 example states), it sends data at a rate controlled by the DAC. What you are describing is the "adaptive" mode of USB audio (the DAC regularly adjusts its clock/PLL, which is used to resample the incoming data to the desired rate, based on the average incoming data rate).

 This is a very important distinction...the DAC is controlling the rate the data is being sent by the computer. This eliminates the need for any resampling (the data will be consumed by the DAC from its buffer at precisely the rate its local clock requires...so it does NOT need to resample from 44.10002 to 44.10000...it can simply use the data at 44.1000 and tell the computer to slow down if it is trying to send the data at 44.10002).

 Now, the part about the DAC chip *wanting* the data at 110K...that I do not know anything about. If there is some inherent advantage to that (and perhaps you have measured the performance of the chip the DAC1 is using and determined that it does indeed do a better job with data resampled to 110K than with data at 44.1K that has not been resampled) then we may be getting to the crux of the matter (why the DAC1 is resampling). But, even so, wouldn't it be better to resample data from a known rate (exactly 44.10000) than a variable rate (whatever the recent average incoming data rate is...say the iTunes 44.10002 that you used as an example)?

Here is a link to a post by John Swenson (from a question I asked at Computer Audio Asylum about this very topic in 2005) that explains the differences between the isochronous USB audio modes a bit better than I can.


----------



## tubaman

From my prior experience with the DAC1 USB I know the unit can run pretty hot, but how hot is too hot? Is overheating ever a concern? 

 I put a HRS Damping Plate, which is 5.5 x 4.5 x 0.7 (inch) and basically made of plastic, on top of my DAC1 PRE. Plastic surface supposedly traps heat, should I be concerned about this? 

 HRS Damping Plate: 
Damping Plate reduces vibrations on audio or video performance


----------



## bigfishybob

Hi guys

 I'm interested in buying a Benchmark DAC1 USB. Does anyone know if this is Vista 64 compatible?

 Apologies if this has been covered already.

 Thanks


----------



## HeadLover

I am using it with Vista 64bit
 Works great!


----------



## bigfishybob

Quote:


  Originally Posted by *HeadLover* /img/forum/go_quote.gif 
_I am using it with Vista 64bit
 Works great!_

 

Great. Thanks for the quick reply.


----------



## Lil' Knight

Yup, I'm also using it with Vista 64bit and have zero problem. It's such an awesome equipment. I almost load it at 100% 
	

	
	
		
		

		
			





 XLR and RCA outputs to my amps and headphones jack for my portable phones.


----------



## Scrith

I'm using Vista64 Ultimate and the DAC1 is working great for me (whether I use DirectSound, KernalStreaming, or WASAPI output from Foobar2000).

 All my DAC1s run a bit warm, but not certainly not hot to the touch.


----------



## Mazz

Quote:


  Originally Posted by *EliasGwinn* /img/forum/go_quote.gif 
_You might say, "You could eliminate the ASRC if there was no jitter!" Sure, but then the digital filters wouldn't perform as well. There are major sonic benefits to converting the sample-rate to 110 kHz, and Mr. Hansen isn't considering this at all. _

 

Well, to be precise (as in the joke about the statistician, applied mathematician and pure mathematician) ... you COULD eliminate the ASRC if there was no jitter AND STILL convert the sample-rate to 110kHz - using an SRC. 

 As I pointed out earlier in the thread, SRC has major benefits. The correct solution is a deterministic algorithm - there's no guesswork or implementation variations involved - and thus any competent programmer should be able to write a perfect implementation. (The fact that many implementations are broken doesn't mean a correct implementation is impossible.) It's also immune to jitter and it doesn't add any jitter. This is because inputs to the algorithm are not clocked - they are merely sequences of samples taken (implicitly) at a fixed known clock rate - as are the outputs.

  Quote:


  Originally Posted by *EliasGwinn* /img/forum/go_quote.gif 
_Isochronous means that the data transfer is happening at a regular (periodic) rate. _

 

Well, to be precise...isochronous means that the bandwidth is reserved on the USB bus for the particular USB endpoint. Whilst it's true that the USB bus operates at a fixed data rate, there may or may not be any data sent during any particular timeslot reserved for that device. (I haven't looked at the specs to see what the bandwidth allocation granularity is, but I'd bet good money it isn't fine enough to distinguish between some of your example rates such as 44.1001kHz and 44.1002kHz.) If data is sent during some timeslots and not during others, most people wouldn't consider this to be "data transfer happening at a regular (periodic) rate". 

 Furthermore, some readers might be lead to believe by this that audio travels over the USB bus at (say) 44.1 kHz (within error bounds), which I'm pretty sure isn't true. The audio bits travel in packets at the USB bus data rate which IIRC is measured in megabits/second...and then the DAC has to infer what the computer's playback frequency really is somehow.

  Quote:


  Originally Posted by *EliasGwinn* /img/forum/go_quote.gif 
_It's not that YOU need to use SRC on your computer... Its that the computer will have to re-sample the data to correspond to the new clock...the master clock....the clock from the USB audio device. I posted about this previously (somewhere around +/- post# 2100). _

 

That is one way to deal with the problem, but as we have discussed earlier you can deal with it in other ways. For example, much like Scrith was saying, you can have the computer speed up/slow down its average transfer rate in response to the DAC. But as you pointed out in response, we (probably) don't have audio drivers built into OSes and media players that can handle these types of control requests (yet). It's not hard to see how to modify existing media players & drivers to do so, but I don't yet see the market demand that will drive such changes. If I had to guess, I'd say an Open Source media player/driver combination might be the first to do it. I suspect this is the biggest issue with asynchronous mode today. It would be interesting to obtain an async mode DAC and see if one can observe these types of issues in practice with today's software stacks.

  Quote:


  Originally Posted by *EliasGwinn* /img/forum/go_quote.gif 
_When we designed the Benchmark/Centrance solution, we said, "Why don't we allow the computer to be master so the sample-rate can stay consistent until it enters the DAC1? Jitter isn't an issue...only bit-transparency. Just send us the proper data, and we'll take care of the rest!!"_

 

The only problem is that IIRC the sample rate doesn't stay that consistent in a computer - the clocks tend to be a bit loose. And you have the media player's playback sample rate, embedded in the USB bus data transfer rate, so it's not quite as simple as you made it seem.

 In fact "Just send us the proper data [and desired sample rate], and we'll take care of the rest!!" seems to be a great description of the guiding philosophy for asynchronous mode USB. If the software stacks ever support it properly, it seems clear that it would be a technically superior solution. Whether you can hear the difference in practice or not is another question...


----------



## EliasGwinn

Quote:


  Originally Posted by *tubaman* /img/forum/go_quote.gif 
_From my prior experience with the DAC1 USB I know the unit can run pretty hot, but how hot is too hot? Is overheating ever a concern? 

 I put a HRS Damping Plate, which is 5.5 x 4.5 x 0.7 (inch) and basically made of plastic, on top of my DAC1 PRE. Plastic surface supposedly traps heat, should I be concerned about this? 

 HRS Damping Plate: 
Damping Plate reduces vibrations on audio or video performance_

 

Tubaman,

 I would recommend not using this damping plate. It will not affect the performance of the DAC1 USB, and it may cause the components to have a shorter life-span because of the increased heat.

 Thanks,
 Elias


----------



## dvse

Quote:


  Originally Posted by *EliasGwinn* /img/forum/go_quote.gif 
_Example: What if Mr. Hansen's USB audio device says, "Send me all audio data at my sample-rate: 44.1001 kHz (or 48k, or 96k)." USBaudio.sys looks at the audio stack, which is sending audio at 44.1002 from iTunes. How does it reconcile that difference? It must re-sample._

 


 Elias,

 Surely this is some misunderstanding? The standard way audio playback is implemented from the application side is that the application writes PCM to a buffer and either polls in a loop or gets an event when the buffer needs to be refilled. Actual sampling rate doesn't come into this at all, it's just a setting on the playback device, transparent to the application. There is no need to resample anything. This is the same flow control idea as implemented by asynchronous USB receivers (and just about everywhere else except digital audio, it seems).


----------



## lyricalmoments

I recently purchased the DAC USB, tested it at the shop running windows media player and it was fine...

 Brought the unit back home and hooked it up to my mac pro and realised that the bass and mids were very unbalanced, culprit happened to be Itune's DSP features (the Sound Enhancer was ON, dur).

 I'm just wondering if there are any recommended software player for Mac OS 10.5.6 that's better than Itunes?

 Thanks!


----------



## lyricalmoments

Nonetheless, I've got the Benchmark DAC USB hooked up to my Yamamoto HA-02 headphone amp and AKG 701 and I'm loving it!


----------



## EliasGwinn

Quote:


  Originally Posted by *dvse* /img/forum/go_quote.gif 
_Elias,

 Surely this is some misunderstanding? The standard way audio playback is implemented from the application side is that the application writes PCM to a buffer and either polls in a loop or gets an event when the buffer needs to be refilled. Actual sampling rate doesn't come into this at all, it's just a setting on the playback device, transparent to the application. There is no need to resample anything. This is the same flow control idea as implemented by asynchronous USB receivers (and just about everywhere else except digital audio, it seems)._

 

Sample rate is not just a setting. Even if the data is not being transferred at the specified period, the sample rate of the audio is a very important, very specific quantity.

 The question is, what happens when the sample-rate of this audio is different from the sample-rate of the USB device? If the playback sample-rate is based on the USB device instead of the audio data, any discrepencies in sample-rate will result in a combination of three possible outcomes: 1. sample-rate conversion; 2. pitch shifting; 3. phase shifting because of repeated or skipped samples.

 The way to avoide this is by allowing the comptuter to keep its clocking system in sync, so that the data arrives at the DAC just like with any other digital source. Yes, there will be jitter, but there is jitter from every digital audio source, through every digital interconnect, and at every digital audio receiver. Dealing with jitter is, and always will be, essential. Re-inventing the wheel to shed a little bit of jitter at the risk of other sonic pit-falls is not an approach that we are interested in.

 Thanks,
 Elias


----------



## EliasGwinn

Quote:


  Originally Posted by *lyricalmoments* /img/forum/go_quote.gif 
_I recently purchased the DAC USB, tested it at the shop running windows media player and it was fine...

 Brought the unit back home and hooked it up to my mac pro and realised that the bass and mids were very unbalanced, culprit happened to be Itune's DSP features (the Sound Enhancer was ON, dur).

 I'm just wondering if there are any recommended software player for Mac OS 10.5.6 that's better than Itunes?

 Thanks!_

 

Try this:

ITunes-QuickTime for Mac - Setup Guide - Benchmark

 Let me know if you have any other questions.

 Thanks,
 Elias


----------



## Mazz

Quote:


  Originally Posted by *EliasGwinn* /img/forum/go_quote.gif 
_The question is, what happens when the sample-rate of this audio is different from the sample-rate of the USB device? If the playback sample-rate is based on the USB device instead of the audio data, any discrepencies in sample-rate will result in a combination of three possible outcomes: 1. sample-rate conversion; 2. pitch shifting; 3. phase shifting because of repeated or skipped samples.

 The way to avoide this is by allowing the comptuter to keep its clocking system in sync, so that the data arrives at the DAC just like with any other digital source._

 

More precisely, that should read "*One *way to avoid this is..." rather than "*The *way...". A different way is by having the USB device *CHANGE* its playback sample rate to conform to the sample rate setting of the audio stream. 

  Quote:


  Originally Posted by *EliasGwinn* /img/forum/go_quote.gif 
_Yes, there will be jitter, but there is jitter from every digital audio source, through every digital interconnect, and at every digital audio receiver._

 

More precisely, there is jitter at every digital audio receiver ONLY where the data clock is implied or transmitted with the audio sample data. If you can provide an interconnect that sends the data without transmitting data clock signals (actual or implied), you eliminate one entire source of jitter.

  Quote:


  Originally Posted by *EliasGwinn* /img/forum/go_quote.gif 
_Re-inventing the wheel to shed a little bit of jitter at the risk of other sonic pit-falls is not an approach that we are interested in._

 

Firstly there's no re-invention here. My several-year-old Rotel AV receiver already changes its playback sample rate for SPDIF inputs (and I bet it was far from the first to do this). When I view digital TV the receiver runs at 48kHz; when I play ripped CD audio it runs at 44.1kHz. There are even protocol bits in the SPDIF headers to specify the desired playback sample rate - and that protocol has been around for many years. If Rotel were to add a USB interface then playback rate switching could also be done very easily for USB. 

 And secondly, as an engineering tradeoff, I will always take "eliminate the problem with no downside" solution over other alternatives. I'd lay good money that practically EVERY asynchronous USB DAC will switch sample playback rates to match the audio stream settings - because it's an old well-known mechanism, it is very easy to do, and (if the software stack works the way dvse says it does) then there is no risk of "sonic pit-falls" in comparison to clocking data on the computer.


----------



## dvse

Quote:


  Originally Posted by *EliasGwinn* /img/forum/go_quote.gif 
_Sample rate is not just a setting. Even if the data is not being transferred at the specified period, the sample rate of the audio is a very important, very specific quantity.

 The question is, what happens when the sample-rate of this audio is different from the sample-rate of the USB device? If the playback sample-rate is based on the USB device instead of the audio data, any discrepencies in sample-rate will result in a combination of three possible outcomes: 1. sample-rate conversion; 2. pitch shifting; 3. phase shifting because of repeated or skipped samples.

 Thanks,
 Elias_

 

Elias,

 The way asynchronous USB works is as follows:

 1.) Application fills a memory buffer with PCM and sets the sample rate on the playback device (assuming that it is supported)
 2.) USB driver sends packets to the device which are stored in the local buffer; playback starts, going by the local clock
 3.) Once a certain part of the local buffer is consumed, a control message is sent to the driver, which sends more packets from its buffer or if it's running low, notifies the application to refill.

 The entire chain works by the clock local to the DAC and any synchronisation issues are avoided. This works because the channel bandwidth is much greater than what is actually needed to feed the DAC.

 No resampling occurs anywhere in the chain unless the original PCM uses some sampling rate not supported by the device (and it's easy to support all the standard ones).

 This requires different behaviour from the driver, but is completely transparent to the application.


----------



## EliasGwinn

Quote:


  Originally Posted by *Mazz* /img/forum/go_quote.gif 
_More precisely, that should read "*One *way to avoid this is..." rather than "*The *way..."._

 

I'll give you this. I try to avoide speaking in 'absolutes'...I let this one slip. Good call...

  Quote:


  Originally Posted by *Mazz* /img/forum/go_quote.gif 
_A different way is by having the USB device *CHANGE* its playback sample rate to conform to the sample rate setting of the audio stream._

 

This is how synchronous USB works. The TAS1020B takes its cue from the incoming sample rate, adjusts its clock accordingly, and sends the data out at the native sample rate of the audio data.

  Quote:


  Originally Posted by *Mazz* /img/forum/go_quote.gif 
_More precisely, there is jitter at every digital audio receiver ONLY where the data clock is implied or transmitted with the audio sample data. If you can provide an interconnect that sends the data without transmitting data clock signals (actual or implied), you eliminate one entire source of jitter._

 

You're talking about using I2S-type transmission...data with no clock. This involves using large buffers, buffer management, and some sort of sample-rate information that could be usually obtained by reading SOF.

 This isn't far from the way UltraLock works. The clock-portion of the incoming data is completely thrown out. The data is buffered, the ASRC determines the sample by averaging the incoming rate over a long period of time (64 samples), affectively eliminating any affect of jitter at or above 1 Hz.

 The sample-rate conversion (to 110kHz) is there to optimize the filters in the DAC chip.

  Quote:


  Originally Posted by *Mazz* /img/forum/go_quote.gif 
_Firstly there's no re-invention here. My several-year-old Rotel AV receiver already changes its playback sample rate for SPDIF inputs (and I bet it was far from the first to do this). When I view digital TV the receiver runs at 48kHz; when I play ripped CD audio it runs at 44.1kHz. There are even protocol bits in the SPDIF headers to specify the desired playback sample rate - and that protocol has been around for many years. If Rotel were to add a USB interface then playback rate switching could also be done very easily for USB._

 

I'm not familiar with your receiver, but it is probably just sensing the sample rate and displaying the result. It probably uses the clock on the SPDIF line for the conversion with a PLL for jitter attenuation, as most DAC's do.

  Quote:


  Originally Posted by *Mazz* /img/forum/go_quote.gif 
_And secondly, as an engineering tradeoff, I will always take "eliminate the problem with no downside" solution over other alternatives. I'd lay good money that practically EVERY asynchronous USB DAC will switch sample playback rates to match the audio stream settings - because it's an old well-known mechanism, it is very easy to do, and (if the software stack works the way dvse says it does) then there is no risk of "sonic pit-falls" in comparison to clocking data on the computer._

 

Ideally, every engineer would love to eliminate a problem without a downside. However, that scenario isn't really a 'tradeoff'. When a real tradeoff situation comes about, you must determine the best solution that acheives the most important objectives.

 I will admit, I haven't seen or heard _everybody's_asynchronous solutions. I'm speaking from what we had learned when we developed our USB solution 2 years ago, and what I've learned since from reading and talking with other engineers. So, it is possible that there is an asynchronous solution out there that has no sonic pit-falls. But I'll have to see it to believe it. 

 Thanks,
 Elias


----------



## EliasGwinn

Quote:


  Originally Posted by *dvse* /img/forum/go_quote.gif 
_Elias,

 The way asynchronous USB works is as follows:

 1.) Application fills a memory buffer with PCM and sets the sample rate on the playback device (assuming that it is supported)
 2.) USB driver sends packets to the device which are stored in the local buffer; playback starts, going by the local clock
 3.) Once a certain part of the local buffer is consumed, a control message is sent to the driver, which sends more packets from its buffer or if it's running low, notifies the application to refill.

 The entire chain works by the clock local to the DAC and any synchronisation issues are avoided. This works because the channel bandwidth is much greater than what is actually needed to feed the DAC.

 No resampling occurs anywhere in the chain unless the original PCM uses some sampling rate not supported by the device (and it's easy to support all the standard ones).

 This requires different behaviour from the driver, but is completely transparent to the application._

 

When you say "local buffer", do you mean the buffer in the USB receiver chip in the USB device?


----------



## Quaddy

Quote:


  Originally Posted by *Mazz* /img/forum/go_quote.gif 
_More precisely, there is jitter at every digital audio receiver ONLY where the data clock is implied or transmitted with the audio sample data. *If you can provide an interconnect that sends the data without transmitting data clock signals (actual or implied), you eliminate one entire source of jitter.*_

 

as far as i know, a USB interface/interconnect does just this, it doesnt send down the interleaved data along with the signal, but am not so sure its as clear cut as to _*completely*_ eliminate jitter because of the lack of data in the equation, this must reduce jitter substantially though, besides AFAIK a realtime stream with interleaving(eg. toslink and SPDIF) can increase the likelyhood of transmission errors resulting in harmonic distortion, SNR and gain discrepancies/anomalies compared with non-interleaved carrier...

 or does the USB interface & interconnect still send down data clock signals even when it omits to send down sample rate data?






_bring back the manchester code_





 j/k


----------



## Mazz

Quote:


  Originally Posted by *EliasGwinn* /img/forum/go_quote.gif 
_This isn't far from the way UltraLock works. The clock-portion of the incoming data is completely thrown out. The data is buffered, the ASRC determines the sample by averaging the incoming rate over a long period of time (64 samples), affectively eliminating any affect of jitter at or above 1 Hz._

 

Right, and that's really great for a whole lot of use cases - especially for any source/input connection that has incoming jitter, which is most of them. (However it's not true that the "clock-portion of the incoming data is completely thrown out" because you MUST "average the incoming rate over a long period of time" in order to infer a smoothed sample rate for use by the ASRC.)

 But as clever and effective as that is, it is a totally unnecessary mechanism for asynch USB - where there is _no incoming playback clock_ signal and hence _no input jitter_, period. *Neither should there be.* Computer audio systems should merely be data delivery systems. Basically they need to make sure the data arrives at the transceiver (the DAC) in a timely fashion and present a UI to the user for control...

  Quote:


  Originally Posted by *EliasGwinn* /img/forum/go_quote.gif 
_You're talking about using I2S-type transmission...data with no clock._

 

Unless I'm mistaken I2S sends an explicit clock signal. But you're right - I am talking about data with no clock. On the other hand, the data stream _has semantics_, including an attribute specifying the playback sample rate the receiving device must use. As you pointed out this is done in electronic devices all the time:

  Quote:


  Originally Posted by *EliasGwinn* /img/forum/go_quote.gif 
_The TAS1020B takes its cue from the incoming sample rate, adjusts its clock accordingly, and sends the data out at the native sample rate of the audio data._

 

 Quote:


  Originally Posted by *EliasGwinn* /img/forum/go_quote.gif 
_This involves using large buffers, buffer management, and some sort of sample-rate information that could be usually obtained by reading SOF._

 

I'm not familiar with the acroynm "SOF", but buffer management is needed, although "large buffers" may not be for some value of large 
	

	
	
		
		

		
		
	


	




 (or more precisely, "large" depends on the requirements of your application). As dvse asserts, buffer management already takes place on the playback computer today. This is merely extending it to be ultimately controlled by the external DAC.

  Quote:


  Originally Posted by *EliasGwinn* /img/forum/go_quote.gif 
_Ideally, every engineer would love to eliminate a problem without a downside. However, that scenario isn't really a 'tradeoff'. When a real tradeoff situation comes about, you must determine the best solution that acheives the most important objectives._

 

True, it's not a "tradeoff" as I stated it - apologies for abusing the term to make a point.

 I think there may be actual tradeoffs here because the DAC1 has a broader set of use cases than a straight "asynch USB DAC", but the tradeoff is *NOT* about ASRC vs asynch mode USB. It's likely one or more of the following:

 (1) You have built a product which does a really great job on input signals that have jitter. Why go to the expense of building and maintaining a second mechanism (for what might turn out to be a negligible or even entirely inaudible improvement) for one interface method?

 (2) Asynch USB ideally relies on USB audio drivers built in to major OSes. Those (IIRC) don't support the highest sample rates and/or bit depths. You want to sell a product that does as far as possible...but don't really want to ship custom USB audio drivers.

 (3) You need to support really low latency playback, and you can't achieve it with asynch USB because of the distributed buffer management.

 I can understand any of these tradeoffs leading you to avoid asynch mode USB (although the second is pretty weak because it (presumably) applies to your existing USB support). But if you're only addressing use cases that can be supported by asynch USB mode, I've not seen any reason to use ASRC instead.

 Cheers,
 Mazz.


----------



## dvse

Quote:


  Originally Posted by *EliasGwinn* /img/forum/go_quote.gif 
_When you say "local buffer", do you mean the buffer in the USB receiver chip in the USB device?_

 

Yes, buffer in the USB receiver.


----------



## Dougr33

Is anyone feeding their Dac1 with the usb from a 'netbook'.. I'm thinking of getting a Dell Mini9, as they're dead quiet (fanless with SSD), run Foobar2000 or such, and having a NAS stuck in the closet. Wondering if the netbook would be up to delivering bit perfect that way (would a usb drive attached to it be too much work for the processor)? Thanks for any thoughts.


----------



## Scrith

Ayre has posted a white paper on their new asynchronous DAC.


----------



## Quaddy

Quote:


  Originally Posted by *Dougr33* /img/forum/go_quote.gif 
_Is anyone feeding their Dac1 with the usb from a 'netbook'.. I'm thinking of getting a Dell Mini9, as they're dead quiet (fanless with SSD), run Foobar2000 or such, and having a NAS stuck in the closet. Wondering if the netbook would be up to delivering bit perfect that way (would a usb drive attached to it be too much work for the processor)? Thanks for any thoughts._

 

be fine. i had the dac1 pre hooked up to an asus ssd netbook - nothing different about it really.


----------



## lyricalmoments

Quote:


  Originally Posted by *EliasGwinn* /img/forum/go_quote.gif 
_Try this:

ITunes-QuickTime for Mac - Setup Guide - Benchmark

 Let me know if you have any other questions.

 Thanks,
 Elias_

 

Thanks Elias for the link


----------



## Techbadger

As a new owner of a Dac1 Pre I have only 2 questions:

 1. Are there any plans for a black version (or at least black front plates that I can swap out to), all the rest of my gear is black, just want a match. I mean I chose function over form but would like both.

 2. On my system, I am planning to output my HTPC through Digital 75ohm coax because I wish to play Reference Recordings 176.4k Wav files and if I read this thread correctly I can only get as high as 96k though the USB. Is there a better way?

 Beyond that, astounding good player!

 Thanks all!


----------



## infinitesymphony

Quote:


  Originally Posted by *Techbadger* /img/forum/go_quote.gif 
_2. On my system, I am planning to output my HTPC through Digital 75ohm coax because I wish to play Reference Recordings 176.4k Wav files and if I read this thread correctly I can only get as high as 96k though the USB. Is there a better way?_

 

Any cheap sound card that has bit-perfect digital output via coaxial/optical and supports 24-bit/192kHz will work. Your onboard audio might even work; many current motherboards support Intel's HD Audio spec and have coaxial digital outputs.


----------



## Techbadger

They Symph,

 My sound card is an HT Omega Claro+ based on a C-Media 8788 chipset.

 There doesn't seem to be a "bit perfect" driver for the card yet but I suspect there will be since its lesser chipset antecedents have had open driver with "bit perfect" capabilities.

 I suspect my new Dac1 pre will decode the wav files music better than my sound cards DACs ever would, which is why I would prefer using it.

 Time will tell I guess...


----------



## EliasGwinn

Quote:


  Originally Posted by *Techbadger* /img/forum/go_quote.gif 
_They Symph,

 My sound card is an HT Omega Claro+ based on a C-Media 8788 chipset.

 There doesn't seem to be a "bit perfect" driver for the card yet but I suspect there will be since its lesser chipset antecedents have had open driver with "bit perfect" capabilities.

 I suspect my new Dac1 pre will decode the wav files music better than my sound cards DACs ever would, which is why I would prefer using it.

 Time will tell I guess..._

 

Hello Techbadger!

 With regards to the DAC1 PRE faceplate, we don't have any new announcements regarding new features or products. Sorry...

 Regarding 174 and/or 192 kHz playback from an HTPC, you will need a sound card that has bit-transparent digital output at these resolutions. The only one we know of that does bit-transparent digital output is the Lynx AES16. Its not cheap...$600 or so. 

 Keep in touch...let me know how it goes.

 Thanks,
 Elias


----------



## EliasGwinn

Mazz,

 With regards to the ASRC in Benchmark converters, you mentioned that you weren't sure about the merits of our solution, beyond jitter attenuation.

 The short answer is this: the filter in the ASRC chip is much better then the filter in the DAC chip, and the filter in the DAC chip works best at 110 kHz. So, we filter with the ASRC chip and re-sample at 110 kHz to relieve the DAC chip.

 The loooonnnggg answer:

 DAC chips have built-in low-pass filters to filter all info above Nyquist. But DAC chips are asked to perform too many tasks at once. It is unreasonable to expect a single chip to be both an excellent DAC with an excellent analog output and ALSO have an excellent digital filter. Manufacturing IC's is a task full of tradeoffs. Analog performance vs. digital performance in a chip is a tradeoff is the major tradeoff in a DAC chip.

 Consequently, the digital low-pass filters in DAC chips have serious limitations. They have the filter cutoff frequency exactly at Nyquist. The reason the filters are built this way is because it requires significantly less DSP to filter at Nyquist. Its a trade-off for higher d-to-a performance. Having the filter exactly at Nyquist poses some major problems: it filters some of the audio info below Nyquist and, more critically, doesn't filter all of the audio above Nyquist. This latter condition will result in digital aliasing. 

 So, with the ASRC, we have excellent filtering via the DSP-powerhouse AD1896 and get rid of jitter, aliasing, and relieve the low-quality filtering of the DAC chip. The sample-rate conversion (ASRC) has THD artifacts well below the threshold of hearing (less then -133 dBFS). This is an easy tradeoff! 

 Thanks,
 Elias


----------



## Scrith

Here's an interesting white paper on Ayre's experience with digital filters.


----------



## Techbadger

Quote:


  Originally Posted by *EliasGwinn* /img/forum/go_quote.gif 
_Hello Techbadger!

 Regarding 174 and/or 192 kHz playback from an HTPC, you will need a sound card that has bit-transparent digital output at these resolutions. The only one we know of that does bit-transparent digital output is the Lynx AES16. Its not cheap...$600 or so. 

 Thanks,
 Elias_

 

Actually you can get bit perfect down to 25 bucks. I found a bit perfect path using Foobar2000 and a couple plug-ins. About $175 total cost.

 Sounds fantastic!


----------



## daio78

.


----------



## daio78

Hello everyone !
 I ve just received my dac1 pre 3 days ago and I have to say it sounds rather nice 
 I'm still waiting for him to warm up a bit more before doing test against my current configuration !

 I was just wondering what could be the best transport that can be used ?
 I have right now an HTPC that is dedicated to HC with an emu1616m connected directly to a power amp (rotel 1077) and I'm in the process of switching from a 5 channels system to a 2 channels one if improvement is really worth. 

 And I was wondering if changing the transport could improve the sounds ? I was thinking about a SB3, an EEEPC (when they ll release one with a touch screen) , Airport express, AppleTVor any other slimdevice without DD....

 Have people here tried those configurations ? PC, EEEPC , SB3, Airport, AppleTV ? USB or Toslink ? 

 Furthermore I've read the best configuration is to avoid resampling in the player (foobar or other) and let the DAC do the resampling ? Can anyone confirm that there is no need to resample in Foobar ? (Even at 110 Khz ?)

 Thanks in advance for the replies !

 PS : It s nice to see staff from Benchmark so active on the forums


----------



## EliasGwinn

daio78,

 I would recommend using the USB port of any computer or the optical output of a Mac computer. I would avoid the Airport express or Apple TV, just because they are limited to 48 kHz, 16-bits. I would also avoid the digital outputs from soundcards (except the built-in digital output on Macs), as some of them will affect the audio.

 With regards to sample-rate conversion, it is unnecessary, and it will not provide any sonic benefits to you.

 Thanks,
 Elias


----------



## Techbadger

But if trying to decode a 176.4k wav I would think the Coax would be the best source since the USB is limited to 96k, right?


----------



## EliasGwinn

Quote:


  Originally Posted by *Techbadger* /img/forum/go_quote.gif 
_But if trying to decode a 176.4k wav I would think the Coax would be the best source since the USB is limited to 96k, right?_

 

If the soundcard is bit-transparent, then it would be best. However, many soundcards are not, so its a little bit difficult to answer this one.

 If you are confident in the quality of the soundcard's performance, then go for it. Otherwise, as a safe alternative, I would suggest using iTunes with the sample-rate set at 96 kHz. We have tested the sample-rate converter in iTunes, and it performs very well.

 Thanks,
 Elias


----------



## HeadLover

I am using the DAC1 PRE with USB
 Works great !!!


----------



## Techbadger

USB does sound fantastic but in the case of 176.4k Wav files, Coax sounds better I have tried it with MediaMonkey and Foobar2000 setup in "bit perfect" configurations and they appear to sound fantastic. I say appear because I am relying on my ears and have no idea if they are truly "bit perfect".

 BTW I won my DAC1 Pre at CES but as amazing as this unit is sounding, I would have spent the money without looking back anytime!

 Only thing I would chance is a remote control for source would be nice and the option of a black faceplate, neither are sonic improvement because sonically this is a truely amazing little (can't believe how small this is!) package!

 Excellent work Benchmark!


----------



## HeadLover

I think the BEST will be and "upgrade" so we can have a "true" 24/196 from the USB.
 I know there are some ways to do it, and I hope Benchmark will come with a unit that can do it.
 Than USB will be truly perfect


----------



## Scrith

I am utterly confused by all these requests for DACs that can play music sampled at rates higher than the already ridiculously high rate of 96K. Where is this music you guys want coming from? I know that, based on what I listen to on a regular basis, approximately 0.01% of it is available is resolutions higher than 16/44. Yes, I know there are a few websites out there selling "music" that is sampled at higher rates...I've checked them out, and most of it is stuff that I wouldn't listen to unless those companies were paying me.


----------



## 8thdwarf

Future proofing ...maybe the future's already  here. 
 Roll your own hi-Q music in hi-rez.


----------



## yipchunyu

I am considering gettng a DAC1 / DAC1 Pre.
 It's easy to get a second hand DAC1 with about half the price of a DAC1 Pre.
 For the price, I am quite sure it's the best buy.
 However, with extra money to get a analog input (and usb). I can't sure now. For me, it's better to have an analog input and so I can input with AV Amp's pre out (and so the AV system and stereo system share the same active monitors).
 However, by doing so, the signal already pass via my cheap av amp and I can't sure whether I can get good sound from DAC1 Pre.
 Anyone can share your idea for me?


----------



## infinitesymphony

Quote:


  Originally Posted by *HeadLover* /img/forum/go_quote.gif 
_I think the BEST will be and "upgrade" so we can have a "true" 24/196 from the USB.
 I know there are some ways to do it, and I hope Benchmark will come with a unit that can do it.
 Than USB will be truly perfect 
	

	
	
		
		

		
		
	


	


_

 

Gear like the E-MU 0404 USB supports 24-bit/192kHz over USB, but it requires USB 2.0 support and a driver install, so it's not as seamless as the DAC1 would be from one computer to another.

  Quote:


  Originally Posted by *Scrith* /img/forum/go_quote.gif 
_I am utterly confused by all these requests for DACs that can play music sampled at rates higher than the already ridiculously high rate of 96K. Where is this music you guys want coming from? I know that, based on what I listen to on a regular basis, approximately 0.01% of it is available is resolutions higher than 16/44. Yes, I know there are a few websites out there selling "music" that is sampled at higher rates...I've checked them out, and most of it is stuff that I wouldn't listen to unless those companies were paying me._

 

My guess is that Techbadger is looking at the Reference Recordings HRx series, which is being released at 24-bit/176.4kHz. RR have had some very good-sounding releases in the past.


----------



## yipchunyu

But does it restrict DAC1 to sound great in XP but not the vista? How about the windows 7? I heard someone use ASIO4ALL to drive DAC1. Is it better?


----------



## HeadLover

WASAPI with Foobar2000 and Vista, and your DONE.
 Amazing quality, and bit perfect.


----------



## Quaddy

Quote:


  Originally Posted by *HeadLover* /img/forum/go_quote.gif 
_WASAPI with Foobar2000 and Vista, and your DONE.
 Amazing quality, and bit perfect._

 

x2 

 i cant be doing with asio after using wasapi.


----------



## nauxolo (Feb 19, 2018)

The unfinished front can't arrive the dish.


----------



## Scrith

Quote:


  Originally Posted by *8thdwarf* /img/forum/go_quote.gif 
_Future proofing ...maybe the future's already  here. 
 Roll your own hi-Q music in hi-rez._

 

I have a few "ultra high quality" vinyl rips at 24/96, most from high-end vinyl (e.g. MFSL or some expensive Japanese pressing) made with high-end gear, some even from "virgin" vinyl records. The vinyl lovers rave about how much better these sound vs. CD rips...but I can't stand them because the surface noise is clearly audible and highly annoying. I'll take a 16/44 CD rip over that stuff any day (and do...I rarely play the vinyl versions).


----------



## HeadLover

Yep but usually it seems like the vinyl rips at 24/96 has much better dynamics and less "loudness war", and the sound seem to be more "live"
 And yes, the poping and cracking isn't that nice


----------



## infinitesymphony

Quote:


  Originally Posted by *nauxolo* /img/forum/go_quote.gif 
_I was looking into the different models, and I was wondering if I needed the pre-amp function that costs an extra 300 dollars? What is wrong with just doing unbalanced line-out to a power-amp or XLR to speakers? I know there must be, but not sure what it is_

 

Nothing is wrong with it--this is what all of the DAC1s do. The only reason the DAC1 Pre is named that way is because it has multiple digital inputs and a set of analog inputs. If you don't need that functionality, look at the DAC1 USB or regular DAC1, though there are some sonic considerations since the Pre contains some minor upgrades.


----------



## EliasGwinn

Quote:


  Originally Posted by *yipchunyu* /img/forum/go_quote.gif 
_I am considering gettng a DAC1 / DAC1 Pre.
 It's easy to get a second hand DAC1 with about half the price of a DAC1 Pre.
 For the price, I am quite sure it's the best buy.
 However, with extra money to get a analog input (and usb). I can't sure now. For me, it's better to have an analog input and so I can input with AV Amp's pre out (and so the AV system and stereo system share the same active monitors).
 However, by doing so, the signal already pass via my cheap av amp and I can't sure whether I can get good sound from DAC1 Pre.
 Anyone can share your idea for me?_

 

Do you need to keep the AV pre in the system? Does it serve a necessary function? If so, you'd still be well advised to connect the 'pre out' to the input of the DAC1 PRE so that the DAC1 PRE can connect directly to your amplifiers.

 Thanks,
 Elias


----------



## EliasGwinn

Quote:


  Originally Posted by *yipchunyu* /img/forum/go_quote.gif 
_But does it restrict DAC1 to sound great in XP but not the vista? How about the windows 7? I heard someone use ASIO4ALL to drive DAC1. Is it better?_

 

The DAC1 works very well with XP and Vista operating systems. We have not tried it with Windows 7 yet. 

 A lot of DAC1 users prefer to use ASIO, but it isn't necessary.

 Thanks,
 Elias


----------



## Scrith

Quote:


  Originally Posted by *HeadLover* /img/forum/go_quote.gif 
_Yep but usually it seems like the vinyl rips at 24/96 has much better dynamics and less "loudness war", and the sound seem to be more "live"
 And yes, the poping and cracking isn't that nice 
	

	
	
		
		

		
			



_

 

You're right, for some recordings the vinyl version is properly mastered when compared to the CD version (a good example being the Red Hot Chili Peppers albums). This, of course, has nothing to do with vinyl (other than vinyl probably not doing so well if it were mastered like certain CDs). Luckily outside of the pop- and heavy-metal music world this seems to be the exception rather than the rule.

 I know some people are probably used to hearing popping, cracking, scratching, and other surface noise from vinyl and have learned to ignore it, but I personally find it very distracting (and I grew up listening to vinyl). What we really need is digital transfers from the original studio tapes (or original digital masters that haven't been downsampled for more recent music, of course).


----------



## lyricalmoments

Quote:


  Originally Posted by *EliasGwinn* /img/forum/go_quote.gif 
_Hello Simplystax,

 Welcome to "The Tread" (sic)!

 The DAC1 PRE has a truly balanced, dual-active XLR output. The XLR outputs will have a maximum value of +24 dBu when the output mode is set to calibrated, the XLR attenuators are set to 0 dB, and the calibration potentiometers are set to the factory default. '

 Feel free to ask any more questions that may help you with your system.

 Thanks,
 Elias_

 

Hi Elias,

 Just to clarify too, if my source is USB from the computer, and I'm to connect to a balanced rig like the Stax via XLR, would it still be considered as true balanced? (As compared to Toslink and Optical-in since the source is already unbalanced)

 Or is true balance only possible if I'm to use the AES input? 

 Thanks!


----------



## EliasGwinn

Quote:


  Originally Posted by *lyricalmoments* /img/forum/go_quote.gif 
_Hi Elias,

 Just to clarify too, if my source is USB from the computer, and I'm to connect to a balanced rig like the Stax via XLR, would it still be considered as true balanced? (As compared to Toslink and Optical-in since the source is already unbalanced)

 Or is true balance only possible if I'm to use the AES input? 

 Thanks!_

 

The digital signal doesn't determine whether something is balanced or not. Balanced interconnection is purely in the analog domain, as it should be (AES over XLR is balanced during transmission, but then goes through a transfomer and becomes unbalanced as it enters the D/A circuit). 

 The reason interconnects are balanced is because it allows the down-stream device to 'unbalance' (take the difference between the normal and inverted signals), which doubles the signal and cancels any common noise. Keeping something balanced defeats the purpose of balanced interconnection. 

 Thanks,
 Elias


----------



## EliasGwinn

Quote:


  Originally Posted by *nauxolo* /img/forum/go_quote.gif 
_I was looking into the different models, and I was wondering if I needed the pre-amp function that costs an extra 300 dollars? What is wrong with just doing unbalanced line-out to a power-amp or XLR to speakers? I know there must be, but not sure what it is

 I tried reading through all 157 pages, but is there anything that I need to be wary about before I purchase the DAC1 PRE? I was hoping not have to upgrade this anymore. Is this future proof? Is it versatile?_

 

Hello Nauxolo,

 Let me know if I can answer any specific questions for you.

 Thanks,
 Elias


----------



## yipchunyu

Quote:


  Originally Posted by *EliasGwinn* /img/forum/go_quote.gif 
_Hello Nauxolo,

 Let me know if I can answer any specific questions for you.

 Thanks,
 Elias_

 

EliasGwinn, I am interested on your DAC 1 PRE.
 However, for me, it seems the analog input is "after thought". Any rationale for your company to build it? How does it compare to other pre-amp?


----------



## Dougr33

Google for some reviews, save Elias some time!


  Quote:


  Originally Posted by *yipchunyu* /img/forum/go_quote.gif 
_EliasGwinn, I am interested on your DAC 1 PRE.
 However, for me, it seems the analog input is "after thought". Any rationale for your company to build it? How does it compare to other pre-amp?_


----------



## G-U-E-S-T

Another benefit of balanced interconnects is the avoidance of ground loops.

 Elias, may I ask a few questions please:

 1) When using the analog inputs of the DAC1 PRE, does the volume control run strictly passively, or does it apply gain at some point on the dial (if so, where on the dial)?

 2) Same question as #1, but this time for the digital inputs;

 3) Is it ok to plug the DAC1 PRE into a power device that delivers balanced power?

 Thanks in advance for your replies! P.S. I'm really enjoying this DAC...


----------



## EliasGwinn

Quote:


  Originally Posted by *yipchunyu* /img/forum/go_quote.gif 
_EliasGwinn, I am interested on your DAC 1 PRE.
 However, for me, it seems the analog input is "after thought". Any rationale for your company to build it? How does it compare to other pre-amp?_

 

Yipchunyu,

 The rationale for building the DAC1 PRE with the analog inputs was to give users a pure path from both analog and digital sources, to the DAC1 PRE, and directly to amplifier. I cannot comment about other pre-amplifiers...I will let the other head-fi members offer their thoughts on that. 

 Thanks,
 Elias


----------



## EliasGwinn

Quote:


  Originally Posted by *G-U-E-S-T* /img/forum/go_quote.gif 
_Another benefit of balanced interconnects is the avoidance of ground loops.

 Elias, may I ask a few questions please:

 1) When using the analog inputs of the DAC1 PRE, does the volume control run strictly passively, or does it apply gain at some point on the dial (if so, where on the dial)?

 2) Same question as #1, but this time for the digital inputs;

 3) Is it ok to plug the DAC1 PRE into a power device that delivers balanced power?

 Thanks in advance for your replies! P.S. I'm really enjoying this DAC..._

 

The volume control and calibration potentiometers both control active gain circuits. They are never passive. This applies to both the digital and analog inputs.

 With regards to question #3, what power device are you referring to?

 Thanks,
 Elias


----------



## G-U-E-S-T

Quote:


  Originally Posted by *EliasGwinn* /img/forum/go_quote.gif 
_The volume control and calibration potentiometers both control active gain circuits. They are never passive. This applies to both the digital and analog inputs.

 With regards to question #3, what power device are you referring to?

 Thanks,
 Elias_

 

I was considering plugging the DAC1 PRE into one of the balanced outlets of a Monster Power HTPS 7000 Signature MKII, which delivers balanced power from three of its six outlet banks.

 However I also remember your earlier advice that the DAC1 PRE does not need any power conditioner, but will perform optimally when plugged straight into the wall. I will continue doing exactly that, if you advise it - the only reason I am considering plugging it into the Monster Power device at our entertainment center, would be for potential convenience only.

 Thanks in advance for your advice.


----------



## lamikeith

Greetings Head-Fi'ers!

 It has taken me about three weeks to read this thread. 2300 posts is not a small number! I am particularly impressed that the signal to noise ratio has remained excellent. Thank you all for your pertinent questions and informative replies, as I have learned a great deal about the current state of digital audio. Special thanks must go to Elias Gwinn for his tireless patience in answering the hundreds of questions asked. I could not have asked for more information transparency of a product or company. Therefore, I am happy to say that a DAC1 Pre with my name on the box is in the capable hands of UPS, delivery expected today. Oh, happy day!

 -Mike


----------



## HeadLover

I have the DAC1 PRE also!
 It is a GREAT GREAT device, you are sure going to enjoy it!
 I have never heard anyone say bad things about it 
	

	
	
		
		

		
		
	


	



 (sure some love it more and some less)


----------



## EliasGwinn

Quote:


  Originally Posted by *G-U-E-S-T* /img/forum/go_quote.gif 
_I was considering plugging the DAC1 PRE into one of the balanced outlets of a Monster Power HTPS 7000 Signature MKII, which delivers balanced power from three of its six outlet banks.

 However I also remember your earlier advice that the DAC1 PRE does not need any power conditioner, but will perform optimally when plugged straight into the wall. I will continue doing exactly that, if you advise it - the only reason I am considering plugging it into the Monster Power device at our entertainment center, would be for potential convenience only.

 Thanks in advance for your advice._

 

If it is more convenient for you, you can use the balanced power. It won't effect the DAC1 PRE's performance at all.

 Thanks,
 Elias


----------



## G-U-E-S-T

Thank you Elias, I really appreciate your advice.

 P.S. I am going to have to send you some tech support money soon, for all your nice help!


----------



## lork

Hello, Eilias-

 Thanks you for your many contributions to this forum! I am an engineer (though not an audio one) and I appreciate the clear engineering you folks have put into your product, and your willingness to explain it. All of that predisposes me favorably toward the DAC1.

 I don’t understand the ‘ultralock’, though, and would appreciate your comments / explaining more how it works. As far as I can tell, it's something like this:

 The DAC1 uses synchronous USB transfer, and I think those (let’s say 44.1 ks/s audio) packets come spaced 1 ms or so apart. I believe this means that the DAC1 output must somehow be adjusted to a clock rate determined by the packet arrival times so the input buffer neither under- or over-runs, since no request to slow down or speed up transmission can be made. There is some buffer on the chip (TAS1020? If so that’s a few (say 3) kb per channel, = about 70 packets at 44.1 ks/s), and so the best one can do re: figuring out the “true” incoming clock rate is some sort of running average on this (via PLL or software). Averaging over 70 packets = 70 ms, this gives about 14 Hz (= 1/.07 s) of resolution (and rejection of faster variations) on the actual incoming frequency. I have assumed that the reason the (non USB, but a similar argument applies to SPDIF) Chord DAC64 buffers for 1-3 seconds is to improve this resolution/rejection to the ~ 1 Hz level.

 As far as I can tell from your posts, the DAC1 does not change the sample output clock at all (so no real jitter there), but matches the incoming data rate to the outgoing rate by varying the upsampling ratio. I assume this allows for better output clock performance (less jitter on that clock, no wandering PLL, etc), but it doesn’t seem to me it would prevent a wandering rate of input packet arrivals from appearing (effectively, via upsampling adjustments) at the output, with a 3dB level of ~ 14 Hz. This 14 Hz width wandering would convolve (smear out) in some weird way with the whole audio spectrum, and would not manifest as a 14 Hz (inaudible) signal.

 If I am right about all/some/enough of that, then there _is_ variation in the effective output rate of the DAC1- not a fast one (so maybe it’s not called “jitter” anymore) but quite possibly a slow one that would still (at up to ~ 14 Hz) potentially affect the music. The more consistent the packet arrival times, the less of an issue this might be (depending on how the lock to that stream is done, e.g. if the lock wanders even for a fixed input frequency), since the average wouldn’t be changing, but it’s definitely not total immunity from input data arrival variations.

 So my first question is: am I even remotely understanding this?

 My second question is: if that’s roughly accurate, then assuming both a USB and a well-clocked SPDIF stream (e.g. from a Squeezebox) are available, would the DAC1 perform better with the Squeezebox, since any jitter in that SPDIF stream would be at far higher frequency than 14 Hz, and so be better rejected, while the USB is a bit more of an unknown and might have occasional wanderings below ~ 14 Hz?

 My last is: if the above is generally right, then was Benchmark’s choice to devote resources to a subtle fix for output clock rate jitter rather than implementing a big input buffer imply that output clock jitter is a more audible problem (perhaps it spreads across the audio band worse, increasing noise level more problematically?) than the lower frequency wandering?

 Many thanks, for any comments and for all your patient & informative posts so far!


----------



## lamikeith

I found the web page for the AD1896 ASRC interesting...
Analog Devices: AD1896: 192 kHz Stereo Asynchronous Sample Rate Converter :: Sample Rate Converters :: Audio/Video Products

 Direct link to the data sheet. The "Theory of Operation" starts on page 18.
http://www.analog.com/static/importe...ets/AD1896.pdf

 Hope that helps.

 -Mike


----------



## HeadLover

Hi

 I have a question, how come this small unit get so hot ?!
 And why is it? and is it ok?


----------



## lork

Thanks, Mike/lamikeith-

 That is a very interesting note. Much more so than the TAS1020 one. Will take me more than the quick read I just did to digest it. Maybe the interpolation filter explains how they get below 14 Hz, though I need to think a while on that.

 Much appreciated!

 PS: I also found "Measurement Techniques for Digital Audio" by Dunn (available from the audio precision website) to be pretty helpful in trying to understand things. Ch 1 is all about jitter. Maybe you might find it useful, if you don't already know about it.


----------



## yipchunyu

Hi EliasGwinn,
 I just got the DAC 1 Pre.
 I found that the analog input's sound input is very low. (i input from pre-out of my av amp).
 When I connect via digital input, i set the volume close to the min.
 But when I use the analog input, I need to set the volume to near to the max.
 Any tips on this?


----------



## EliasGwinn

Quote:


  Originally Posted by *G-U-E-S-T* /img/forum/go_quote.gif 
_Thank you Elias, I really appreciate your advice.

 P.S. I am going to have to send you some tech support money soon, for all your nice help! 
	

	
	
		
		

		
		
	


	


_

 

PM me for cash-drop-off location 
	

	
	
		
		

		
		
	


	




 Actually, I'd prefer if you re-appropriated that money to buy music from musicians. After all, without them, none of us would be talking about any of this.

 Thanks,
 Elias


----------



## EliasGwinn

Great questions and analysis, Lork!!

  Quote:


  Originally Posted by *lork* /img/forum/go_quote.gif 
_So my first question is: am I even remotely understanding this?_

 

A lot of what you said is on the right track, but you're misunderstanding how UltraLock works. After you read the AD1896 paper, you'll understand it a lot better. And I'll be happy to clear up any questions that remain.

  Quote:


  Originally Posted by *lork* /img/forum/go_quote.gif 
_My second question is: if that’s roughly accurate, then assuming both a USB and a well-clocked SPDIF stream (e.g. from a Squeezebox) are available, would the DAC1 perform better with the Squeezebox, since any jitter in that SPDIF stream would be at far higher frequency than 14 Hz, and so be better rejected, while the USB is a bit more of an unknown and might have occasional wanderings below ~ 14 Hz?_

 

Actually, the point your making about low-frequency jitter is very important, as it will be very detrimental to audio quality if not properly addressed. Many converter manufacturers don't address low-frequency jitter issues, and consequently their converters have no jitter attenuation below 1 kHz. You'd be surprised at how many 'high-end' converters have this problem. 

 Benchmark's UltraLock will reject jitter below 1 Hz. Consequently, Benchmark's DAC1 USB/PRE will perform equally well using USB or a well-clocked SPDIF stream.

  Quote:


  Originally Posted by *lork* /img/forum/go_quote.gif 
_My last is: if the above is generally right, then was Benchmark’s choice to devote resources to a subtle fix for output clock rate jitter rather than implementing a big input buffer imply that output clock jitter is a more audible problem (perhaps it spreads across the audio band worse, increasing noise level more problematically?) than the lower frequency wandering?_

 

Not true. Low frequency jitter (it is still called jitter at low frequencies) is a very important issue that must be addressed to achieve high quality conversion. UltraLock addresses this issue.

 Thanks,
 Elias


----------



## EliasGwinn

Quote:


  Originally Posted by *HeadLover* /img/forum/go_quote.gif 
_Hi

 I have a question, how come this small unit get so hot ?!
 And why is it? and is it ok?_

 

The normal operating temperature of the DAC1 is 85-95 degrees F. In other words, if you rest your hand on the top, it should feel like a mug of hot coffee.

 It runs hot because it is a low-impedance circuit with internal linear power supply. 

 This temperatur is ok...it will not affect the performance of the unit.

 Thanks,
 Elias


----------



## EliasGwinn

Quote:


  Originally Posted by *lork* /img/forum/go_quote.gif 
_Thanks, Mike/lamikeith-

 That is a very interesting note. Much more so than the TAS1020 one. Will take me more than the quick read I just did to digest it. Maybe the interpolation filter explains how they get below 14 Hz, though I need to think a while on that.

 Much appreciated!

 PS: I also found "Measurement Techniques for Digital Audio" by Dunn (available from the audio precision website) to be pretty helpful in trying to understand things. Ch 1 is all about jitter. Maybe you might find it useful, if you don't already know about it._

 

Yes! Juliann Dunn is the 'Godfather of Digital Audio Measurement' (I just made up that title, but I stand behind it). 

 I look forward to discussing all of this with you, Lork!

 Thanks,
 Elias


----------



## EliasGwinn

Quote:


  Originally Posted by *yipchunyu* /img/forum/go_quote.gif 
_Hi EliasGwinn,
 I just got the DAC 1 Pre.
 I found that the analog input's sound input is very low. (i input from pre-out of my av amp).
 When I connect via digital input, i set the volume close to the min.
 But when I use the analog input, I need to set the volume to near to the max.
 Any tips on this?_

 

When using the analog input, the DAC1 PRE's volume is relative to the amplitude of the analog signal coming from your AV amp. Is there anything (volume control, etc) limiting the output level of your pre-out from the AV amp? Can you get the output level any hotter?

 The analog gain range of the DAC1 PRE is as follows:

 Off to +3.5 dB (RCA in to RCA out)
 Off to +19 dB (RCA in to XLR out)
 Off to +13 dB (RCA in to Headphone)

 Factory-Set Analog-Input Gain In ‘Calibrated’ Mode:

 +0.5 dB (RCA in to RCA out)
 +16 dB (RCA in to XLR out)

 Maximum Analog Input Level:

 +15 dBu

 Maximum Analog Input @ Factory-set Calibration Levels:

 +13 dBu

 Let me know if this helps.

 Thanks,
 Elias


----------



## Bostonears

Quote:


  Originally Posted by *yipchunyu* /img/forum/go_quote.gif 
_Hi EliasGwinn,
 I just got the DAC 1 Pre.
 I found that the analog input's sound input is very low. (i input from pre-out of my av amp).
 When I connect via digital input, i set the volume close to the min.
 But when I use the analog input, I need to set the volume to near to the max.
 Any tips on this?_

 

You might try your AV amp's Tape Out (or Fixed Level Output) rather than the Pre-out? The Tape Out or its equivalent wouldn't be attenuated.


----------



## lork

Thanks for the info/comments, Elias. I will have some follow-on questions, but am still parsing through the ASRC info. I'll post my remaining questions once done.

 Separately, I do understand that low frequency jitter matters- my question was whether there was an issue of relative importance compared to HF jitter.

 In the meantime, once I had the ASRC keyword from lamikeith, I started to find a lot of good info, Listing links here in case anyone else is curious and (like me) hasn't already found them.

 Particularly, the diyaudio forum on ASRC (link below) is quite something.

 What is ASRC (theory)?
diyAudio Forums - Asynchronous Sample Rate Conversion - Page 1 

 Is Ultralock just ASRC?
http://www.head-fi.org/forums/2911683-post407.html

 What is Ultralock?
 Start with the AD1896 sheet theory of operation (p 18) as lamikeith very helpfully suggested

 Jitter effects:
 Ch 1 "Measurement Techniques for Digital Audio" by Dunn (AP High Performance Audio Analyzer & Audio Test Instruments : Library )


----------



## EliasGwinn

Quote:


  Originally Posted by *lork* /img/forum/go_quote.gif 
_Separately, I do understand that low frequency jitter matters- my question was whether there was an issue of relative importance compared to HF jitter._

 

Low-frequency jitter is arguably _more _problematic then high-frequency jitter. The reason is that the jitter-induced sidebands will be very near the fundamental, causing a bluring of the space and tones. And very low frequency jitter (<10 Hz) will make things sound out-of-tune. For example, if a recording of a piano playing an A440 is being played with a 5 Hz jitter problem, the sidebands will come up at 435 Hz and 445 Hz. This will cause beat-frequencies, making the A440 sound out-of-tune.

 High-frequency jitter, on the other hand, will produce tones at (seemingly) random frequencies. This is also detrimental to the audio quality, but since it is seemingly random, it just causes a 'muddled' sound. It also results in a burring of the image, and masks low-amplitude sounds.

 They are both bad, really. Jitter is an ugly beast.

 Thanks,
 Elias


----------



## Lucabeer

Elias, would there be a chance in future of something like a DAC 2 USB with all the qualities of DAC 1 USB but _without_ headphone amplification?

 I might be interested in such a product if priced reasonably (800 US$) because I already own an excellent headphone amp and really wouldn't need another one but I'd love to have a Benchmark DAC (although I can't justify the expense of the DAC 1 USB)...


----------



## tubaman

Hi Elias, 
 With this particular amp (Sugden Headmaster; spec. at Sugden Audio - Bijou Series
 Do you suggest setting the output at "calibrated" or somewhere else? 
 Thanks!


----------



## holdendebeans

Quote:


  Originally Posted by *HeadLover* /img/forum/go_quote.gif 
_WASAPI with Foobar2000 and Vista, and your DONE.
 Amazing quality, and bit perfect._

 

Please elaborate for the very ignorant. How do I use/adjust WASAPI? Does it matter, SQ-wise, if I use Monkeymedia or Media Player? Can I stream S/PDIF toslink?

 Many appologies for my ignorance but thanks in advance.


----------



## HeadLover

There is a complete guide here (can't remmber the link right now)
 All you do is this, download FOOBAR2000 and download the WASAPI plugin.
 Than you put the plugin to the foobar components dir.
 Than start up foobar, go to the setting, choose WASAPI as the output (to the DAC1), set the buffer size to be low, but no to low (can cause some no so good sound glitches), and set the volume to the max, disable any DSP or and PRE amp and gain and so on, and you have BIT PERFECT (using VISTA)

 Foobar2000 is a great player and almost anyone here at the forum (who uses Vista or XP) use it, very "Audiophile" player.


----------



## holdendebeans

Quote:


  Originally Posted by *HeadLover* /img/forum/go_quote.gif 
_There is a complete guide here (can't remmber the link right now)
 All you do is this, download FOOBAR2000 and download the WASAPI plugin.
 Than you put the plugin to the foobar components dir.
 Than start up foobar, go to the setting, choose WASAPI as the output (to the DAC1), set the buffer size to be low, but no to low (can cause some no so good sound glitches), and set the volume to the max, disable any DSP or and PRE amp and gain and so on, and you have BIT PERFECT (using VISTA)

 Foobar2000 is a great player and almost anyone here at the forum (who uses Vista or XP) use it, very "Audiophile" player._

 


 I downloaded Foobar2000 and the WASAPI plug-in. How do I get the plug-in into Foobar? It is not an option on the component selection page.


----------



## holdendebeans

Nevermind. I cut/pasted the plug-in file to the component directory in the foobar file. No go. Re-install, repeat, all good. (but 16/48 only).


----------



## EliasGwinn

Quote:


  Originally Posted by *tubaman* /img/forum/go_quote.gif 
_Hi Elias, 
 With this particular amp (Sugden Headmaster; spec. at Sugden Audio - Bijou Series
 Do you suggest setting the output at "calibrated" or somewhere else? 
 Thanks!_

 

Tubaman,

 Ask the manufacturer what the maximum input level is for this amp. Let me know what they say...

 Thanks,
 Elias


----------



## G-U-E-S-T

Hi Elias,

 We continue to really enjoy our DAC1 PRE - it has tremendously increased our enjoyment of music in our home. I keep forgetting to ask you this next question, and it is a weird one - so get ready! 
	

	
	
		
		

		
			





 How robust/durable is the DAC1 PRE? I ask this because I previously had a DAC unit where the manufacturer warned that it needed to always be handled gently, that it should never be bumped or set down hard, etc. We of course always try to handle our equipment extremely carefully and gently - but I wonder please, really how "delicate" is the DAC1 PRE? Are there any components or electronics within it that could be made to misfunction or go out of spec, etc, if it was set down with anything less than a gentle "bump" on a shelf, or anything else like that? I really have no idea how delicate the inner electronics of such things actually are... what do you think and advise please? Thank you again for all your nice help.


----------



## EliasGwinn

Quote:


  Originally Posted by *G-U-E-S-T* /img/forum/go_quote.gif 
_...how "delicate" is the DAC1 PRE? Are there any components or electronics within it that could be made to misfunction or go out of spec, etc, if it was set down with anything less than a gentle "bump" on a shelf, or anything else like that? I really have no idea how delicate the inner electronics of such things actually are... what do you think and advise please? Thank you again for all your nice help._

 

The DAC1 PRE is robust enough that you don't need to worry about it. I'd recommend not dropping it down the stairs or running over it with your car, but it may even survive that. 

 ...the wonders of surface mount soldering...
	

	
	
		
		

		
		
	


	




 Thanks,
 Elias


----------



## tubaman

Quote:


  Originally Posted by *EliasGwinn* /img/forum/go_quote.gif 
_Tubaman,

 Ask the manufacturer what the maximum input level is for this amp. Let me know what they say...

 Thanks,
 Elias_

 

Hi Elias, 
 I got a reply from Sugden today, regarding input level that said, "The input has an infinite overload facility recommended normal input for CD."
 I suppose this means "calibrated" for the amp? 

 BTW I solved the clipping problem by upgrading the RAM from 2G to 3G.


----------



## G-U-E-S-T

Quote:


  Originally Posted by *EliasGwinn* /img/forum/go_quote.gif 
_The DAC1 PRE is robust enough that you don't need to worry about it. I'd recommend not dropping it down the stairs or running over it with your car, but it may even survive that. 

 ...the wonders of surface mount soldering...
	

	
	
		
		

		
		
	


	




 Thanks,
 Elias_

 

Thank you Sir!


----------



## EliasGwinn

Quote:


  Originally Posted by *tubaman* /img/forum/go_quote.gif 
_Hi Elias, 
 With this particular amp (Sugden Headmaster; spec. at Sugden Audio - Bijou Series
 Do you suggest setting the output at "calibrated" or somewhere else? 
 Thanks!_

 

Tubaman,

 The Headmaster will begin to clip with an input of -2.2 dBu (!). This means that the input of the Headmaster will be overdriven by the DAC1 PRE in the factory-set calibrated mode with any digital input over -13 dBFS. Therefore, you'll have to either re-calibrate, which requires a digital voltmeter and a digital test tone, or operate in variable-mode and keep the volume control near 11 o'clock.

 Thanks,
 Elias


----------



## HeadLover

EliasGwinn

 I have a little problem.
 I have the Audioengine A5 speakers connected to my DAC1 PRE.
 but it seem like the right channel is stronger than the left (or the other way)
 Is there a way to change it a little bit?


----------



## HeadLover

Also,
 It seem like the DAC1 PRE is giving way to much gain to those speakers, cause I am almost at 0% volume on the speaker, and yet the volume is more like it should be where it is 25%
 Not nice at all, I can't hear at low volume.

 What can be done?


----------



## tubaman

Quote:


  Originally Posted by *EliasGwinn* /img/forum/go_quote.gif 
_Tubaman,

 The Headmaster will begin to clip with an input of -2.2 dBu (!). This means that the input of the Headmaster will be overdriven by the DAC1 PRE in the factory-set calibrated mode with any digital input over -13 dBFS. Therefore, you'll have to either re-calibrate, which requires a digital voltmeter and a digital test tone, or operate in variable-mode and keep the volume control near 11 o'clock.

 Thanks,
 Elias_

 

Thanks Elias. Do you mind explaining how the thing basically works? It seems strange to have to adjust the output for every component used with the PRE. 
 The vol. produced at calibrated (Sony DVP7700's coax) is not much different from the Magnum Dynalab tuner I feed Input 2 of Headmaster and is the same as using the DVP7700 to feed Input 3 directly. 
 BTW clarification on my end, the clipping I was referring should actually be "drop-out" - as in music stops while Foobar appeared to be still playing. I upgraded the RAM of my laptop from 2G to 3G and no problem anymore.


----------



## HeadLover

So, is there anything I can do to make the volume (the fixed one) out of my DAC1 PRE be more "usuable" for my A5 speakers?
 I know many have them here, what can be done?


----------



## EliasGwinn

Quote:


  Originally Posted by *HeadLover* /img/forum/go_quote.gif 
_EliasGwinn

 I have a little problem.
 I have the Audioengine A5 speakers connected to my DAC1 PRE.
 but it seem like the right channel is stronger than the left (or the other way)
 Is there a way to change it a little bit?_

 

 Quote:


  Originally Posted by *HeadLover* /img/forum/go_quote.gif 
_Also,
 It seem like the DAC1 PRE is giving way to much gain to those speakers, cause I am almost at 0% volume on the speaker, and yet the volume is more like it should be where it is 25%
 Not nice at all, I can't hear at low volume.

 What can be done?_

 

 Quote:


  Originally Posted by *HeadLover* /img/forum/go_quote.gif 
_So, is there anything I can do to make the volume (the fixed one) out of my DAC1 PRE be more "usuable" for my A5 speakers?
 I know many have them here, what can be done?_

 

You mentioned the 'fixed' volume control of the DAC1 PRE, so I assume you are using the volume control on the front of the speakers. 

 The channel imbalance is probably due to the fact that the volume control on the front of the speakers is being used near the bottom of its range. 

 To reduce the amount of signal that you are sending to the speakers, you will need to recalibrate the 'fixed' levels (which requires a voltmeter, a constant-amplitude test tone, and a small, flat screw driver) or use the 'variable' mode and set the volume control to the desired level. The variable mode will not reduce quality, so don't hesitate to go that route.

 Let me know how it goes.

 Thanks,
 Elias


----------



## HeadLover

Oh I have tried using the variable mode and it sure his great, I am putting in on a step than I can use the speakers with like 25% volume for my music like I love (I don't love hearing loud)
 Will it reduce quality?
 Also, what is better, using the volume on FOOBAR2000/windows or the volume on the DAC1 PRE for it ? (as for SQ)

 And,
 How come that with other speakers I had (Some creative) it was really fine using the 'fixed', but here with the A5 is it way to loud?
 And is it ok?

 I don't have all those tools, so I guess using the variable will be the best for me?


----------



## HeadLover

Oh and one more thing, can't I use the DAC1 PRE as a PRE with the A5, mean to put the A5 on the MAX volume and than use the DAC1 for tune the volume? or will it be bad?
 I just want to know what will be the BEST COMBO for giving me both SQ and a good volume control and not need to "blow my ears out"


----------



## HeadLover

So, what will be the best combo for giving me sound?


----------



## infinitesymphony

HeadLover, slow down... Yes, you can set the monitors to maximum or near-maximum and use them like a typical preamp + active monitors setup. Use the knob on the DAC1 Pre to control volume; it's analog and won't attenuate the digital signal like a software mixer might.


----------



## Quaddy

elias, a question if i may regarding dac1 pre output power

 can you tell me what output power the HPA2 is able to provide? in watts

 i would be running an AKG K1000 into the pre via single ended TRS on the headphone section, which i realize wont drive them with much authority

 as the likes of the AKG K1000(120ohm) require a lot of power, maybe a very bare minimum of 1 watt, ideally 8 watts and above, as they are ear speakers, not headphones

 so i was wondering what the figures are for the pre

 thanks very much for any info


----------



## EliasGwinn

Quote:


  Originally Posted by *HeadLover* /img/forum/go_quote.gif 
_Oh I have tried using the variable mode and it sure his great, I am putting in on a step than I can use the speakers with like 25% volume for my music like I love (I don't love hearing loud)
 Will it reduce quality?
 Also, what is better, using the volume on FOOBAR2000/windows or the volume on the DAC1 PRE for it ? (as for SQ)

 And,
 How come that with other speakers I had (Some creative) it was really fine using the 'fixed', but here with the A5 is it way to loud?
 And is it ok?

 I don't have all those tools, so I guess using the variable will be the best for me?_

 


  Quote:


  Originally Posted by *HeadLover* /img/forum/go_quote.gif 
_Oh and one more thing, can't I use the DAC1 PRE as a PRE with the A5, mean to put the A5 on the MAX volume and than use the DAC1 for tune the volume? or will it be bad?
 I just want to know what will be the BEST COMBO for giving me both SQ and a good volume control and not need to "blow my ears out" 
	

	
	
		
		

		
		
	


	


_

 

I recommend using the DAC1 PRE as the VC (volume control) as much as possible. Also, Foobar's VC is very high quality, so you can use it as well. 

 For your system, I recommend the following steps to setup your system:
 1. Set the DAC1 PRE's VC at 10 o'clock (or about 30%)
 2. Play one of the loudest tracks that you listen to
 3. Adjust the VC of the A5's to achieve a volume that is the most quiet that you would want to listen to your music
 4. If this results in the A5's VC being below 30%, turn down Foobar 30%.
 5. Repeat steps 3 and 4 until the VC of the A5 is above 30% at the most quiet listening position
 6. Once you've set the VC of the A5 and Foobar, use the DAC1 PRE to control the volume. Since the quietest setting will be 30% of the DAC1 PRE's VC, it will only be used at or above the 30% setting for all listening levels.

 This method will give you the best sound quality.

 The reason the A5's are louder then the Creative speakers is because the built-in amplifiers are more powerful.

 Thanks,
 Elias


----------



## EliasGwinn

Quote:


  Originally Posted by *Quaddy* /img/forum/go_quote.gif 
_elias, a question if i may regarding dac1 pre output power

 can you tell me what output power the HPA2 is able to provide? in watts

 i would be running an AKG K1000 into the pre via single ended TRS on the headphone section, which i realize wont drive them with much authority

 as the likes of the AKG K1000(120ohm) require a lot of power, maybe a very bare minimum of 1 watt, ideally 8 watts and above, as they are ear speakers, not headphones

 so i was wondering what the figures are for the pre

 thanks very much for any info_

 

Hey Quaddy,

 The HPA2 amp can drive up to 250 mA. With a 120 ohm load, you will be able to deliver up to 7.5 watts.

 Thanks,
 Elias


----------



## Quaddy

many thanks for the prompt and helpful reply!


----------



## HeadLover

Thank you!
 So
 right now I am doing like this, 100% volume on foobar2000 using Vista and WASAPI, so the digital is bit perfect (no DSP and 24 bit output)
 I have set the DAC1 as for something like between the 9 to 10 on the volume knob.
 And I am using my A5 for volume, where the lowest I get is like 8 and a little.
 How is it?
 I really don't want to touch in the digital domain so it will be bit perfect


----------



## EliasGwinn

Hey Head-Fi friends,

 Benchmark just started using Twitter to send quick tips about using our products and other general audio info. Feel free to join us:

Twitter / benchmarkmedia

 Thanks,
 Elias


----------



## tuffgong

Hi Headlover, I understand what you are getting at, so here is a reasoned guess for an answer on my part. Ideally, if you want the most bitperfect sound possible, then your comp settings are correct, but you wish to avoid the volume pot on the benchmark. You can avoid this by going calibrated, and either a) adjusting jumpers inside as everyone helped me do way back when in this thread to appropriate db padding, or b) adjusting the little pots on the back with a tiny screwdriver, as elias suggested, but just adjust -as best you can to get them equal-, realizing that your ears will eventually tell which speaker is a little more biased, if at all, and adjust that one speaker accordingly. The reason Elias wanted you to use a test signal and take measurements is because he is working with recording engineer standards, and they need perfect levels, which I am guessing you might not, as neither do I. At lot of people have reported improved results by having the damping pads/jumpers at 0db, which I changed them to just to be safe, but i had to turn down the voltage on the back to compensate, so perhaps voltage is the best route. By going calibrated and attenuating volume this way, and just using your a5 volume, you avoid any dac volume pot interference, which is minimal but you wanted the cleanest route. You are affecting voltage/damping the other route, which I surmise is more bit perfect. Then your a5 doesn't have to be at 8, ideally you'd want it at 5 for your sweet spot, reason being there's no need to drive them that hard with the other adjustments, and allows more variation for mood, longevity of speakers etc. Hopefully some/all of my previous advice is correct.
 On a side note, a link to some headphone art I think people might dig:
sts9 Store - Merch Detail . Cheers.


----------



## Dougr33

I'm not getting much with searches...

 Is there any consensus on the best linux media player (hopefully Ubuntu compatible) to output the Very Best sound quality via USB?

 Tricks in settings for best quality? Thanks folks!


----------



## BitPerfect

I doubt you'll find consensus on *anything* in the Linux community, but I'd suggest you at least take a look at:

 * Songbird
 * Amarok
 * mplayer (minimalist)
 * Banshee


----------



## Dougr33

Thanks, I'll give them a try.


----------



## ted betley

Elias have you seen my pm? Also sent one to Benchmark.


----------



## MarkyMark

Quote:


  Originally Posted by *EliasGwinn* /img/forum/go_quote.gif 
_Tubaman,

 The Headmaster will begin to clip with an input of -2.2 dBu (!). This means that the input of the Headmaster will be overdriven by the DAC1 PRE in the factory-set calibrated mode with any digital input over -13 dBFS. Therefore, you'll have to either re-calibrate, which requires a digital voltmeter and a digital test tone, or operate in variable-mode and keep the volume control near 11 o'clock.

 Thanks,
 Elias_

 

Hi Elias,

 I also have the DAC1 and Sugden Headmaster connected using RCA's. However, my DAC1 is the "Classic" version. Does the same advice apply in my case???

 Cheers.


----------



## EliasGwinn

Quote:


  Originally Posted by *MarkyMark* /img/forum/go_quote.gif 
_Hi Elias,

 I also have the DAC1 and Sugden Headmaster connected using RCA's. However, my DAC1 is the "Classic" version. Does the same advice apply in my case???

 Cheers._

 

Yes, all DAC1 models have the same output levels.

 Thanks,
 Elias


----------



## MarkyMark

Elias,

 Is there any difference in sound quality of the headphone amp in the DAC1 classic versus the DAC1 Pre?

 Thanks,
 Mark


----------



## EliasGwinn

Quote:


  Originally Posted by *MarkyMark* /img/forum/go_quote.gif 
_Elias,

 Is there any difference in sound quality of the headphone amp in the DAC1 classic versus the DAC1 Pre?

 Thanks,
 Mark_

 

The amp is exactly the same, but it has two additional features that you should consider:

 1. The DAC1 PRE's headphone amp has three gain ranges: the first is exactly as the DAC1 Classic, the second is 10 dB lower, and the third is 20 dB lower. This is essential for sensitive headphones, as the original DAC1 can get very loud very quickly. The lower gain ranges of the DAC1 PRE allow you to use the volume range in the sweet spot without blowing your head off!!

 2. The DAC1 PRE has an auto-mute switch that will mute the main outputs (XLR and RCA) whenever a headphone plug is inserted into the left jack. This is great for bouncing between loudspeakers and heaphones...you don't have to turn off the amps everytime you want to listen to headphones.

 Let me know if I can answer anything else for you...

 Thanks,
 Elias


----------



## G-U-E-S-T

Elias, just wanted to say thanks again to you and all at Benchmard Media for making this great DAC1 PRE. 
 My family and I (and our guests) all just love it. We currently use it with a Logitech Squeezebox 3 (optical connection), with great results.
 Best Regards!


----------



## athenaesword

Quote:


  Originally Posted by *infinitesymphony* /img/forum/go_quote.gif 
_HeadLover, slow down... Yes, you can set the monitors to maximum or near-maximum and use them like a typical preamp + active monitors setup. Use the knob on the DAC1 Pre to control volume; it's analog and won't attenuate the digital signal like a software mixer might._

 

is this pre-amp stage available in all DAC1s or only in the DAC1 Pre? I'm using a pair of Adam A7s as well and would like to use the benchmark like you suggested.


----------



## EliasGwinn

Quote:


  Originally Posted by *G-U-E-S-T* /img/forum/go_quote.gif 
_Elias, just wanted to say thanks again to you and all at Benchmard Media for making this great DAC1 PRE. 
 My family and I (and our guests) all just love it. We currently use it with a Logitech Squeezebox 3 (optical connection), with great results.
 Best Regards!_

 

G-U-E-S-T,

 I'm glad to hear it! I will pass the word to the team here, especially the designer, John Siau. 

 All the best,
 Elias


----------



## EliasGwinn

Quote:


  Originally Posted by *athenaesword* /img/forum/go_quote.gif 
_is this pre-amp stage available in all DAC1s or only in the DAC1 Pre? I'm using a pair of Adam A7s as well and would like to use the benchmark like you suggested._

 

Athenaesword,

 All DAC1's can be used as pre-amps. The later models have upgraded op-amps, but they all have the same pre-amp architecture. 

 Thanks,
 Elias


----------



## athenaesword

Quote:


  Originally Posted by *EliasGwinn* /img/forum/go_quote.gif 
_Athenaesword,

 All DAC1's can be used as pre-amps. The later models have upgraded op-amps, but they all have the same pre-amp architecture. 

 Thanks,
 Elias_

 

so what's the primary difference between teh basic model and the usb model, just the usb input?


----------



## EliasGwinn

Quote:


  Originally Posted by *athenaesword* /img/forum/go_quote.gif 
_so what's the primary difference between teh basic model and the usb model, just the usb input?_

 

There are four major differences between the DAC1 and DAC1 USB:

 1. USB input capable of 96 kHz/24 bit
 2. Multiple headphone gain ranges to suit different headphone sensitivities.
 3. Auto-mute switch that automatically mutes the main analog outputs whenever a headphone plug is inserted. This allows you to seamlessly switch between headphones and loudspeakers without having to turn off the amps/speakers.
 4. Auto-standby feature that will put the unit to 'sleep' when the selected digital input is inactive. The original DAC1 has no standby functions.

 All the best,
 Elias


----------



## EliasGwinn

Quote:


  Originally Posted by *athenaesword* /img/forum/go_quote.gif 
_so what's the primary difference between teh basic model and the usb model, just the usb input?_

 

OOPS! I forgot one of the major differences between the DAC1 and DAC1 USB. 

 The output drivers were upgraded using superior op-amps from National Semiconductor (LM4562's). The new output drivers are capable of delivering higher amounts of current, acheive lower output impedances, and are less susceptible to distortion from high load capacitance and/or low load impedances.

 Thanks!
 Elias


----------



## EliasGwinn

Here is a DAC1 comparison chart:

DAC1 Comparison Chart | Benchmark Media Systems, Inc.


----------



## EliasGwinn

Hey Head-Fi Friends!

 There is a *REMOTE* possibility that Benchmark has a new product on the way. 

Lookie 
	


 shhh....!! This is the first public word of this!!


----------



## HeadLover

you can't do it to us!
 We want more information!!!


----------



## s.a.b.

Quote:


  Originally Posted by *EliasGwinn* /img/forum/go_quote.gif 
_OOPS! I forgot one of the major differences between the DAC1 and DAC1 USB. 

 The output drivers were upgraded using superior op-amps from National Semiconductor (LM4562's). The new output drivers are capable of delivering higher amounts of current, acheive lower output impedances, and are less susceptible to distortion from high load capacitance and/or low load impedances.

 Thanks!
 Elias_

 

Assuming that what is being driven is relatively "easy", would the sonic qualities between the different opamps be the same or is there still likely to be at least some minimal difference?


----------



## Quaddy

Quote:


  Originally Posted by *EliasGwinn* /img/forum/go_quote.gif 
_Hey Head-Fi Friends!

 There is a *REMOTE* possibility that Benchmark has a new product on the way. 

Lookie 
	


 shhh....!! This is the first public word of this!!_

 

elias, please provide any additional info on this when you are permitted to do so

 i would take a wild stab in the dark that this unit would feature a infra red device that allowed wireless control of this device 
	

	
	
		
		

		
		
	


	




 i am curious as to what *HDR* would stand for...


----------



## Scrith

My guesses on the HDR:

 - there's a remote control (could this be the R in HDR?)

 - Firewire (or USB 2.0) support for 24/192 data rates?!? (could this be the HD in HDR?)

 - asynchronous USB input support? (I guess this depends more on CEntrance supporting it?)


----------



## HeadLover

I think it will have both a remote control, and a USB 2.0 with 24/192 and maybe some new upgrades and add-ons like better opmaps, chips and so on


----------



## gregeas

Nice -- if this is correct, the HDR is exactly what I've been waiting for. With inputs and volume controllable remotely, this will work as an excellent "digital pre-amp." 

 My Transporter can do some of this, but switching inputs is a hassle. 

 Any idea about the release date?


----------



## Scrith

Elias, we need some info on this! Several people I know are considering jumping ship for the new Ayre DAC (with asynchronous USB support), which is being released within the next week or so...


----------



## HeadLover

I hope this new HDR will give Ayre DAC a run for the money 
	

	
	
		
		

		
		
	


	




*We need some more information !!!*


----------



## lamikeith

Quote:


  Originally Posted by *EliasGwinn* /img/forum/go_quote.gif 
_Hey Head-Fi Friends!

 There is a *REMOTE* possibility that Benchmark has a new product on the way. 

Lookie 
	


 shhh....!! This is the first public word of this!!_

 

Ooooo, a DAC1 with remote control!? If the product is released soon, then I hope that those of us who recently purchased a DAC1-PRE will be offered the option to trade-up.


----------



## Quaddy

Quote:


  Originally Posted by *lamikeith* /img/forum/go_quote.gif 
_Ooooo, a DAC1 with remote control!? If the product is released soon, then I hope that those of us who recently purchased a DAC1-PRE will be offered the option to trade-up._

 

now theres an optimist at work 
	

	
	
		
		

		
		
	


	



_
 but yes, i agree, yes please. ahh but bah & meh. i got mine from a retailer, they wouldnt entertain it, maybe those who bought directly from benchmark..._


----------



## HeadLover

I have bought direct from Benchmark, will there be a trade-in ???
 I wish so!!!

 It will be cool to have a remote and maybe even USB 2.0 with 24/192 input, and other nice improvements, like a better USB, better headphones amp, better opamps, DAC chip and so on


----------



## lamikeith

HDR is an acronym for...

 Home Digital Receiver
 High Data Rate
 High Definition Remote
 Heavy Dollar Requirement

 Let's hope for one of the former and not the latter.


----------



## HeadLover

I am hoping for High Data Rate
 High Definition Remote



 And like I have said, much more updates and upgrades.

 I want something to TOP even high end DAC's like ones that cost even 5000$, I mean, why not?
 The DAC1 PRE I have is a really good unit, and I hope they will make it even better


----------



## lamikeith

Apparently the leak page is a work in progress. The selector LEDs now have labels (thank you!) and the unit will be in black as well as silver.

Benchmark DAC1 HDR

 " DAC1 HDR is a...
 (DAC1 HDR available in Black)"

 I would *love* to see a picture of the unit's back panel.


----------



## Headphony

Quote:


  Originally Posted by *EliasGwinn* /img/forum/go_quote.gif 
_Hey Head-Fi Friends!

 There is a *REMOTE* possibility that Benchmark has a new product on the way. 

Lookie 
	


 shhh....!! This is the first public word of this!!_

 

Cool, finally remote control. This could be hard to resist.


----------



## EliasGwinn

Hey, sorry that last link I posted was wrong. Here is the real link:

DAC1 HDR | Benchmark Media Systems, Inc.

 Have a great Holiday!

 -Elias


----------



## Quaddy

interesting, looks the same rear end as the pre, and still 24/96

 but at least it has remote for folks that need it


----------



## s.a.b.

I've tried connecting (using the digital coax out) my Motorola HD DVR to my DAC1 but the red "non PCM" signal on front of the DAC1 lights up and I get no sound.

 Am I doing something wrong?

 Thanks in advance for your help.


----------



## gregeas

Quote:


  Originally Posted by *s.a.b.* /img/forum/go_quote.gif 
_I've tried connecting (using the digital coax out) my Motorola HD DVR to my DAC1 but the red "non PCM" signal on front of the DAC1 lights up and I get no sound.

 Am I doing something wrong?

 Thanks in advance for your help._

 

Make sure your cable box is set to output PCM and not Bitstream. Then it should work fine.


----------



## G-U-E-S-T

Hi Elias,

 I see the website description says "The DAC1 HDR’s (High Dynamic Range) motor-driven (Alps potentiometer) volume control maintains the dynamic range of the converter and audio output. In contrast, digital volume controls reduce dynamic range, and analog volume IC’s introduce distortion and noise."

 Aside from the remote control aspect: Is the new Alps volume control pot on the upcoming DAC1 HDR, a better sonic performer than the one used in the DAC1 PRE?


----------



## s.a.b.

Quote:


  Originally Posted by *gregeas* /img/forum/go_quote.gif 
_Make sure your cable box is set to output PCM and not Bitstream. Then it should work fine._

 

Thanks.

 I read the manual for the Motorola DCT 6400 (which I have thru TW cable) and went to the settings in the menu for both audio set-up and cable box set-up, but can't find how to change the SPDIF's output to PCM (Curiously under the HDMI connection, it can be set to 2 channel PCM which is what it is set at).

 I tried the DAC1 again but it's still getting the "non PCM" indication.

 BTW, I have no problem with the audio when I run the DVR's analogue outs to my preamp- I'm just curious as to how the sound would differ using the DAC1. 

 Any suggestions?


----------



## Bostonears

Quote:


  Originally Posted by *s.a.b.* /img/forum/go_quote.gif 
_I read the manual for the Motorola DCT 6400 (which I have thru TW cable) and went to the settings in the menu for both audio set-up and cable box set-up, but can't find how to change the SPDIF's output to PCM (Curiously under the HDMI connection, it can be set to 2 channel PCM which is what it is set at).

 I tried the DAC1 again but it's still getting the "non PCM" indication._

 

Most cable TV service providers (Time Warner, Comcast, etc.) have not implemented firmware for the Motorola 6400 or 3400 series boxes that would covert audio to 2 channel PCM output via Toslink or SPDIF. The boxes just output whatever compressed digital audio signal comes along with the TV signal, which is usually Dolby Digital 2.0 or 5.1 (depending on the program), neither of which is compatible with the Benchmark DAC1 devices.


----------



## HeadLover

So will there be some kind of a trade in or something for guys who have the DAC1 and want the new HDR one ???


----------



## EliasGwinn

Quote:


  Originally Posted by *G-U-E-S-T* /img/forum/go_quote.gif 
_Hi Elias,

 I see the website description says "The DAC1 HDR’s (High Dynamic Range) motor-driven (Alps potentiometer) volume control maintains the dynamic range of the converter and audio output. In contrast, digital volume controls reduce dynamic range, and analog volume IC’s introduce distortion and noise."

 Aside from the remote control aspect: Is the new Alps volume control pot on the upcoming DAC1 HDR, a better sonic performer than the one used in the DAC1 PRE?_

 

The custom Alps pot in the DAC1 HDR is a higher quality pot then the one used in our other products. It has better channel matching, tighter tolerance, a larger geometry, and a longer life expectency. 

 The new pot won't change the 'tone' of the audio because the volume circuit is designed such that neither the old nor the new pot will contribute distortion or noise to the audio. However, the level-balance between channels will be tighter and the pot itself is much more durable.

 All the best,
 Elias


----------



## HeadLover

*EliasGwinn

*Is there a way that someone like me who has the DAC1 PRE can upgrade it to the new HDR one ???
 (with adding money of curse)


----------



## EliasGwinn

Quote:


  Originally Posted by *HeadLover* /img/forum/go_quote.gif 
_*EliasGwinn

*Is there a way that someone like me who has the DAC1 PRE can upgrade it to the new HDR one ???
 (with adding money of curse)_

 

I'm not sure. Let me look into this, and I'll let you know.

 Thanks,
 Elias


----------



## HeadLover

Quote:


  Originally Posted by *EliasGwinn* /img/forum/go_quote.gif 
_I'm not sure. Let me look into this, and I'll let you know.

 Thanks,
 Elias_

 

Ok thanks!
 I really hope so! I have no problems paying what needed so you can make the upgrade and make my unit the HDR one


----------



## rms

Why is there not an AES input on the HDR? Some of the high end sound cards (Lynx) do not have spdif.


----------



## EliasGwinn

Quote:


  Originally Posted by *rms* /img/forum/go_quote.gif 
_Why is there not an AES input on the HDR? Some of the high end sound cards (Lynx) do not have spdif._

 

The coaxial inputs on the DAC1 HDR are compatible with AES inputs. Just use an impedance transformer (Canare Impedance Transformer Xlrf-Bnc Jack 10Db) and a BNC-RCA cable (Benchmark Media).

 This will not affect the quality whatsoever because the UltraLock will isolate the data from the incoming clock and reclock it with an internal low-jitter clock.

 Thanks,
 Elias


----------



## HeadLover

So, what about the update from PRE1 to HDR ?


----------



## roker

Gentlemen, I'll buy your "obsolete" DAC1s

 just make the offer good


----------



## hh83917

I am upgrading my DAC1 Pre to the HDR because I only got it one month ago. I'm glad they just come out with it because there is a 30 days return policy. I was a few days off but I emailed Benchmark and was allowed to exchange for the HDR with the price difference. I'm now waiting for my HDR to arrive before I ship the DAC1 Pre back. Their customer service is great and I'm a happy customer.


----------



## HeadLover

I have bought it like 2 or 3 months ago, what about me? is there a way I can exchange it?


----------



## hh83917

Question about the DAC1s. Since the DAC1 have 2 headphone outputs, does that mean I can plug in 2 headphones at the same time? What my real question is will DAC1 be able to drive 2 high end headphones (high impedance) without loss in quality?


----------



## EliasGwinn

Quote:


  Originally Posted by *HeadLover* /img/forum/go_quote.gif 
_I have bought it like 2 or 3 months ago, what about me? is there a way I can exchange it?_

 

I'll have an official policy to post on this subject within the next few days. 

 Thank you for your patience.

 All the best,
 -Elias


----------



## EliasGwinn

Quote:


  Originally Posted by *hh83917* /img/forum/go_quote.gif 
_Question about the DAC1s. Since the DAC1 have 2 headphone outputs, does that mean I can plug in 2 headphones at the same time? What my real question is will DAC1 be able to drive 2 high end headphones (high impedance) without loss in quality?_

 

Yes, the HPA2 (the headphone amplifier in all DAC1 products) can drive 2 sets of headphones simultaneously without loss in quality. In fact, it can drive two 60-ohm headphones at +18 dBu while still maintaining full specified performance. 

 High-impedance headphones are definately not a problem, as low-impedance headphones are the most trying on a heapdhone amp. High-impedance headphones require large voltage rails to achieve adaquate volume, but low-impedance headphones will draw significant amounts of current from the amp. A headphone amp's performance is limited by the amount of current it can drive before causing significant distortion. The HPA2 can drive up to 250 mA of current.

 Thanks,
 Elias


----------



## Bojamijams

250ma per channel or together?


----------



## EliasGwinn

Quote:


  Originally Posted by *Bojamijams* /img/forum/go_quote.gif 
_250ma per channel or together?_

 

Per channel.


----------



## Bojamijams

Ooooh goody. 
	

	
	
		
		

		
		
	


	




 Thanks.


----------



## pcf

A question for Elias:

 I have four digital sources going into my DAC1 pre. Which one do you think will give me the best sound, or are they supposed to be the same?
 1) Ipod classic (lossless files)- Wadia iTransport-DAC1pre
 2) cd transport (Meridian500)-DAC1pre
 3) laptop with window vista(iTunes/Alec)-DAC1pre
 4) SqueezeBox Duet-DAC1pre

 To my ears they all sound very similar. Maybe the cd transport and iTransport come out on top sometimes. I don't know...

 Sorry if similar topics have already been covered.


----------



## EliasGwinn

Quote:


  Originally Posted by *pcf* /img/forum/go_quote.gif 
_A question for Elias:

 I have four digital sources going into my DAC1 pre. Which one do you think will give me the best sound, or are they supposed to be the same?
 1) Ipod classic (lossless files)- Wadia iTransport-DAC1pre
 2) cd transport (Meridian500)-DAC1pre
 3) laptop with window vista(iTunes/Alec)-DAC1pre
 4) SqueezeBox Duet-DAC1pre

 To my ears they all sound very similar. Maybe the cd transport and iTransport come out on top sometimes. I don't know...

 Sorry if similar topics have already been covered._

 

Assuming that they all are 44/16, and there is no digital volume control or other DSP engaged, they will sound identical (although Vista re-samples the audio, it is very transparent and should not be audible). 

 If it is above 44/16, the transports may truncate and/or down-sample, usually with horrific sonic repercussions.

 Thanks,
 Elias


----------



## hh83917

Quote:


  Originally Posted by *EliasGwinn* /img/forum/go_quote.gif 
_Yes, the HPA2 (the headphone amplifier in all DAC1 products) can drive 2 sets of headphones simultaneously without loss in quality. In fact, it can drive two 60-ohm headphones at +18 dBu while still maintaining full specified performance. 

 High-impedance headphones are definately not a problem, as low-impedance headphones are the most trying on a heapdhone amp. High-impedance headphones require large voltage rails to achieve adaquate volume, but low-impedance headphones will draw significant amounts of current from the amp. A headphone amp's performance is limited by the amount of current it can drive before causing significant distortion. The HPA2 can drive up to 250 mA of current.

 Thanks,
 Elias_

 

Thank you Elias for answering. Sounds like I can have a guest HD650 listening to the same tunes with me.


 Howard.


----------



## pcf

Quote:


  Originally Posted by *EliasGwinn* /img/forum/go_quote.gif 
_Assuming that they all are 44/16, and there is no digital volume control or other DSP engaged, they will sound identical (although Vista re-samples the audio, it is very transparent and should not be audible). 

 If it is above 44/16, the transports may truncate and/or down-sample, usually with horrific sonic repercussions.

 Thanks,
 Elias_

 

Thanks Elias! Now I can trust my ears a bit more!

 -Paul


----------



## EliasGwinn

Quote:


  Originally Posted by *hh83917* /img/forum/go_quote.gif 
_Thank you Elias for answering. Sounds like I can have a guest HD650 listening to the same tunes with me.


 Howard._

 

Absolutely. In fact, when you have your headphones on and your friend plugs their headphones in, you'll be amazed at how consistent the sound remains. If you have you're eyes closed, you won't even know when they plug in.

 Let me know how it goes.

 Thanks,
 Elias


----------



## EliasGwinn

Quote:


  Originally Posted by *pcf* /img/forum/go_quote.gif 
_Thanks Elias! Now I can trust my ears a bit more!

 -Paul_

 

As a post-script to my previous answer...

 Any two sources that send the exact same digital audio _data _to the DAC1 will sound the same. In other words, the performance will not be affected by clock quality, digital cable quality, jitter, etc. 

 With the DAC1, you only need to be concerned that the source is sending the appropriate data. We have seen corrupted data from several DVD transports, computer media players, networked music players, etc. Usually, it is a result of volume adjustment, DSP, sample-rate conversion, or word-length truncation. 

 Provided that the sources are bit-transparent, they will all sound identical with the DAC1.

 All the best,
 Elias


----------



## aCuria

Does any one know what advantages does the 4562 bring to the DAC1 USB/PRE?

 And what other advantages do the USB and PRE versions have over the DAC1, other than the input/output flexibility?

 Another thing:

 If i use the DAC1 with active monitors like the BM5A, how will the volume control work out?


----------



## EliasGwinn

Quote:


  Originally Posted by *aCuria* /img/forum/go_quote.gif 
_Does any one know what advantages does the 4562 bring to the DAC1 USB/PRE?

 And what other advantages do the USB and PRE versions have over the DAC1, other than the input/output flexibility?

 Another thing:

 If i use the DAC1 with active monitors like the BM5A, how will the volume control work out?_

 

The 4562 can push higher current levels without sacrificing performance. This allows us to lower the output impedance, which is good for a few reasons. It is less likely to attenuate higher frequencies due to load capacitance. Also, it optimizes common-mode rejection ratio. Also, if the load has low impedance, the increase current draw won't induce distortion from the 4562.

 Here is a comparison list between the three currently available DAC1 models (the new DAC1 HDR is not on the list, but it has everything the DAC1 PRE has plus a remote control and custom Alps volume pot):

DAC1 Comparison Chart | Benchmark Media Systems, Inc.

 The DAC1's volume control works great with the BM5A's. In fact, they are such a good combo, that we sell them as a package. 

 All the best,
 Elias


----------



## 2danes

Hello EliasGwinn, I have been reading and reading through this thread, and there is some great information here. I have a DAC1 USB and am very pleased with the results. I am feeding my Apple tv, which is loaded with apple lossless, not wireless, direct to the DAC1, then to the Mcintosh C-46 pre, then to some monoblocks. I am really enjoying the sound. I know that the Apple tv is limited to 44.1 output, and my question is, will I benefit if I change over the Apple tv to a mac mini to stream 96/24, or will I not see an audible difference ?? Any info would be greatly appreciated...... thanks


----------



## lowmagnet

Elias:

 Maybe you could petition the moderators to rename this thread to "The official DAC1 thread" since the DAC1 USB came out two product revisions ago?


----------



## aCuria

Quote:


  Originally Posted by *EliasGwinn* /img/forum/go_quote.gif 
_The 4562 can push higher current levels without sacrificing performance. This allows us to lower the output impedance, which is good for a few reasons. It is less likely to attenuate higher frequencies due to load capacitance. Also, it optimizes common-mode rejection ratio. Also, if the load has low impedance, the increase current draw won't induce distortion from the 4562._

 

I dont quite get it 
	

	
	
		
		

		
			





 1) The 4562 can push higher current levels without sacrificing performance. This allows us to lower the output impedance, which is good for a few reasons. 

 It is less likely to attenuate higher frequencies due to load capacitance. 
 -meaning on the DAC-1 the high frequencies tend to drop off more as compared to the DAC-1-USB?
 -Does this occur at the unbalanced out, balanced out and the headphone amp?
 -what kind of components have high load capacitances?



 It optimizes common-mode rejection ratio.
 -"The common-mode rejection ratio (CMRR) of a differential amplifier (or other device) measures the tendency of the device to reject input signals common to both input leads."
 -How does this affect the sound in the end?


 Also, if the load has low impedance, the increase current draw won't induce distortion from the 4562.
 -Am i right to say the 4562s have a lower output impedance than those used in the vanilla DAC-1, which helps to suppress resonances and free oscillations at the speakers?


 Comparing the usb to the pre, does the second 4562 help drive the analogue out as well? or is it linked to the analogue in of the pre (ie: if i connect the usb and the pre using coaxial, will there be any difference in the output?)


 Once again, thanks for the help. I am deciding between the vanilla DAC1, the DAC1 usb, and the DA-10. I dont really need the usb, although its nice to have. It will be nice if you could point out the advantages of the DAC1 over the DA-10 as well =)


----------



## G-U-E-S-T

Hi Elias,

 Just wanted to say that if Benchmark does decide to offer a trade-up program (DAC1-PRE to DAC1-HDR), I would likely take advantage of it. As you know, my family and I absolutely love our DAC1-PRE (received in 10/08), it has radically improved our music listening experience and made it fun again - but the new features of the HDR version would just be so great for us to have. Either way though, thanks again for such great products (and support)! You're the best.


----------



## HeadLover

I don't think they have a trade-in program
 I have sent them mail, and no they won't do it, what a bummer


----------



## aCuria

Quote:


  Originally Posted by *HeadLover* /img/forum/go_quote.gif 
_I don't think they have a trade-in program
 I have sent them mail, and no they won't do it, what a bummer 
	

	
	
		
		

		
		
	


	


_

 

Perhaps you could try to sell it? The DAC-1 seems to hold its value well compared to the others


----------



## HeadLover

Amm yes I could, but the new HDR only offer a remote and maybe a little better pot, I don't think it is worth it, not ??


----------



## Quaddy

i think i will have to sell my pre for the HDR, i now need the remote capability, i didnt initially, plus its an excuse to try a bit of new kit 
	

	
	
		
		

		
		
	


	



*
 edit:* _elias, any photos doing the rounds of the remote that is included?_


----------



## aCuria

Quote:


  Originally Posted by *HeadLover* /img/forum/go_quote.gif 
_Amm yes I could, but the new HDR only offer a remote and maybe a little better pot, I don't think it is worth it, not ??_

 

Probably not, but if benchmark had a trade up program, i think you will be facing similar losses no?


----------



## HeadLover

No
 A trade in will be better, at least for me

 I really don't understand why brands don't have such a thing.

 I prefer buying from brands that DO care of their costumers and know that tech is changing and we can't sell and buy each time, so if there something new and I am a return costumer, at least do me a *discount *or something!


----------



## roker

Quote:


  Originally Posted by *HeadLover* /img/forum/go_quote.gif 
_No
 A trade in will be better, at least for me

 I really don't understand why brands don't have such a thing.

 I prefer buying from brands that DO care of their costumers and know that tech is changing and we can't sell and buy each time, so if there something new and I am a return costumer, at least do me a *discount *or something!_

 

buyer remorse is a pain

 but that's the way it is.

 who knows, that elias guy might hook you up or something.


----------



## infinitesymphony

Quote:


  Originally Posted by *HeadLover* /img/forum/go_quote.gif 
_No
 A trade in will be better, at least for me

 I really don't understand why brands don't have such a thing._

 

Because then they would be faced with the burden of selling an unknown amount of used equipment in various conditions. Either they take the hit and give you a discount, which requires them to devote time and resources to collecting and repurposing old units, or you sell it and give yourself the discount.


----------



## pcf

Hi Elias,

 Some questions about the Benchmark HDR. Is the remote only for volume control? Can it switch sources? I couldn't find any details pictures on your website. Also, Are we still stuck with 1 analog source? People might want to know these before they make their decision.

 I enjoy the DAC1pre's sound quality very much but there are one or two things that I thought it could improve- one of them is the need of a remote control which has been addressed in the upcoming HDR. The other problem I have with the pre is the lack of anlog inputs. Anyone who listens to vinyl and SACD will find the pre wanting. If I want to use it as a preamp, I have to use a switch box which is not ideal. 

 Are you offering any exhange deals? People who bought the pre will not likely spend more on the HDR unless they sell first.

 Keep up the good work!


----------



## aCuria

Quote:


  Originally Posted by *pcf* /img/forum/go_quote.gif 
_Hi Elias,

 Some questions about the Benchmark HDR. Is the remote only for volume control? Can it switch sources? I couldn't find any details pictures on your website. Also, Are we still stuck with 1 analog source? People might want to know these before they make their decision._

 

from here, 

Benchmark Media Unveils Their Flagship DAC1 HDR with Remote Control - Audio Server & MP3 Player News, News, Stereo Amplifier News, Stereo Preamplifier News - HomeTheaterReview.com

 The remote control of the DAC1 HDR controls on/off, input selection, volume, soft-mute, and adjustable dim level. The user can adjust the "dim" volume setting and the "normal" volume settings independently, and the DAC1 HDR will remember those volume settings.

 and here

DAC1 HDR Rear View | Benchmark Media Systems, Inc.

 Theres only one set of analogue inputs


----------



## The Monkey

Quote:


  Originally Posted by *HeadLover* /img/forum/go_quote.gif 
_No
 A trade in will be better, at least for me

 I really don't understand why brands don't have such a thing.

 I prefer buying from brands that DO care of their costumers and know that tech is changing and we can't sell and buy each time, so if there something new and I am a return costumer, at least do me a *discount *or something!_

 

My guess is that the reason Benchmark does not behave in the way you suggest is because it wants to stay in business.


----------



## pcf

Quote:


  Originally Posted by *aCuria* /img/forum/go_quote.gif 
_from here, 

Benchmark Media Unveils Their Flagship DAC1 HDR with Remote Control - Audio Server & MP3 Player News, News, Stereo Amplifier News, Stereo Preamplifier News - HomeTheaterReview.com

 The remote control of the DAC1 HDR controls on/off, input selection, volume, soft-mute, and adjustable dim level. The user can adjust the "dim" volume setting and the "normal" volume settings independently, and the DAC1 HDR will remember those volume settings.

 and here

DAC1 HDR Rear View | Benchmark Media Systems, Inc.

 Theres only one set of analogue inputs 
	

	
	
		
		

		
		
	


	


_

 

Thanks a lot for the info and the link. The remote will help a lot. I used the DAC1pre as a preamp for a shot time along with a switchbox from Mapletree for the analog inputs. It was such a pain and I eventually gave up and put my preamp back in the chain.


----------



## pcf

Quote:


  Originally Posted by *The Monkey* /img/forum/go_quote.gif 
_My guess is that the reason Benchmark does not behave in the way you suggest is because it wants to stay in business._

 

Hi Monkey,
 I can totally understand if Benchmark doesn't offer any discount or trade but I also share people's frustration. DAC1pre was out only a short time ago advertised as a fully functioned preamp. A lot of people already voiced their opinions about the lack of remote controt. Like everyone else, I got the impression that it was Benchmark's design decision to omit the remote. Had we know there was a new version coming round the corner the decision might have been different. Don't forget people in Europe are paying a lot more than US customers for these products. I can understand why they would feel hard done by. 
 In an ideal world I would have rather seen Benchmark bypassed the released of DAC1pre completely and just go straight to the HDR a little later.


----------



## Quaddy

elias, can you provide any info on when and how much the HDR will be here in the UK from official distributors?

 thanks.


----------



## EliasGwinn

Quote:


  Originally Posted by *2danes* /img/forum/go_quote.gif 
_Hello EliasGwinn, I have been reading and reading through this thread, and there is some great information here. I have a DAC1 USB and am very pleased with the results. I am feeding my Apple tv, which is loaded with apple lossless, not wireless, direct to the DAC1, then to the Mcintosh C-46 pre, then to some monoblocks. I am really enjoying the sound. I know that the Apple tv is limited to 44.1 output, and my question is, will I benefit if I change over the Apple tv to a mac mini to stream 96/24, or will I not see an audible difference ?? Any info would be greatly appreciated...... thanks_

 

If you are playing redbook (44.1 kHz / 16-bit) audio without volume control or any other DSP, then upgrading won't make a difference in the sound. However, it is sometimes difficult to know if there is any DSP occuring. If there is DSP of any sort (re-sampling, volume control, etc), the 16-bit limitation will cause distortion due to truncation. 

 In other words, the 96/24 interface doesn't necessarily sound better, but its more capable of handling the data stream without negative effects. 

 Thanks,
 Elias


----------



## EliasGwinn

Quote:


  Originally Posted by *lowmagnet* /img/forum/go_quote.gif 
_Elias:

 Maybe you could petition the moderators to rename this thread to "The official DAC1 thread" since the DAC1 USB came out two product revisions ago?_

 

I wouldn't want to assume ownership of this thread. This is the user's thread, and I just participate.

 All the best,
 Elias


----------



## EliasGwinn

Quote:


  Originally Posted by *aCuria* /img/forum/go_quote.gif 
_I dont quite get it 
	

	
	
		
		

		
		
	


	




 1) The 4562 can push higher current levels without sacrificing performance. This allows us to lower the output impedance, which is good for a few reasons. 

 It is less likely to attenuate higher frequencies due to load capacitance. 
 -meaning on the DAC-1 the high frequencies tend to drop off more as compared to the DAC-1-USB?
 -Does this occur at the unbalanced out, balanced out and the headphone amp?
 -what kind of components have high load capacitances?_

 

All types of components can have high input capacitance: amps, pre-amps, etc. Manufacturers often add a capacitor on the input to filter RF and EMI noise. Unfortunately they will also filter the high-frequency of the audio if the source impedance isn't low enough. The DAC1 USB acheives lower output impedances then the DAC1, so it will be less likely to allow the load to filter the high-frequencies. 

 This doesn't apply to the headphone amp, unless you used the headphone output to drive a pre-amp/amp.



  Quote:


  Originally Posted by *aCuria* /img/forum/go_quote.gif 
_It optimizes common-mode rejection ratio.
 -"The common-mode rejection ratio (CMRR) of a differential amplifier (or other device) measures the tendency of the device to reject input signals common to both input leads."
 -How does this affect the sound in the end?_

 

Common-mode rejection is important for eliminating noise. Also, if a load's differential amp isn't designed properly, it could lead to distortion. Lower source impedance helps prevent this also.


  Quote:


  Originally Posted by *aCuria* /img/forum/go_quote.gif 
_Also, if the load has low impedance, the increase current draw won't induce distortion from the 4562.
 -Am i right to say the 4562s have a lower output impedance than those used in the vanilla DAC-1, which helps to suppress resonances and free oscillations at the speakers?_

 

The 4562 allows the DAC1 USB/PRE/HDR's output stage to have a lower output impedance. However, this won't affect the speakers. It only affects the interaction between the DAC1 and the next device down the path (amplifier or pre-amp, usually).


  Quote:


  Originally Posted by *aCuria* /img/forum/go_quote.gif 
_Comparing the usb to the pre, does the second 4562 help drive the analogue out as well? or is it linked to the analogue in of the pre (ie: if i connect the usb and the pre using coaxial, will there be any difference in the output?)_

 

The DAC1 USB has the 4562's on the output driver section only, while the DAC1 PRE has 4562's all throughout the analog circuit. There are no measurable sonic differences between the two, but many users and reviewers have mentioned preferring the sound of the DAC1 PRE.


  Quote:


  Originally Posted by *aCuria* /img/forum/go_quote.gif 
_Once again, thanks for the help. I am deciding between the vanilla DAC1, the DAC1 usb, and the DA-10. I dont really need the usb, although its nice to have. It will be nice if you could point out the advantages of the DAC1 over the DA-10 as well =)_

 

Sorry, but I can't comment on any competitors product. Best of luck on your purchase decision! 
	

	
	
		
		

		
		
	


	




 All the best,
 Elias


----------



## artears

Hi Elias,

 I am interested in pre or the hdr model and did not make up my mind yet. I had the chance to audition DAC1 pre before and I did not find it to be too analytical as some will claim here about other versions of DAC1 or DAC1 in general. Maybe that engaging feeling was coming from the synergy between my ACS T2 iems and the pre, who knows. I was planning on getting the pre version until I saw the hdr as well. It seems like the difference between those two are the motorized potentiometer and the remote. There is no upgrade to any other parts regarding dac section on hdr over pre version, am I right? Considering that I mostly will use it with a computer setup, I might as well choose not to get the hdr and save another $300. Are there any sonic differences? Which one do you recommend? 

 I have another question: Regarding the outputs in the back of the unit, does the unit control the volume no matter what is given as input? I am curious if a digital input also enables the use of the volume or the volume control is only limited to whenever an anolog input is connected to it?

 I appreciate your time and efforts here. Thank you very much.


----------



## EliasGwinn

Ok folks, here is the official DAC1 HDR trade-in policy:

 Any DAC1, DAC1 USB, and/or DAC1 PRE that was purchased directly from Benchmark on or after March 1, 2009 may be returned for a full refund of the purchase price towards a DAC1 HDR. The returned unit(s) must be in like-new condition or they will be subject to a restocking fee.

 This offer ends May 1, 2009.

 All the best,
 Elias


----------



## pcf

Quote:


  Originally Posted by *EliasGwinn* /img/forum/go_quote.gif 
_Ok folks, here is the official DAC1 HDR trade-in policy:

 Any DAC1, DAC1 USB, and/or DAC1 PRE that was purchased directly from Benchmark on or after March 1, 2009 may be returned for a full refund of the purchase price towards a DAC1 HDR. The returned unit(s) must be in like-new condition or they will be subject to a restocking fee.

 All the best,
 Elias_

 

Thanks Elias! That's a very good gesture! Now people who just bought a DAC1pre will not feel hard done by!


----------



## EliasGwinn

Quote:


  Originally Posted by *pcf* /img/forum/go_quote.gif 
_Hi Elias,

 Some questions about the Benchmark HDR. Is the remote only for volume control? Can it switch sources? I couldn't find any details pictures on your website. Also, Are we still stuck with 1 analog source? People might want to know these before they make their decision.

 I enjoy the DAC1pre's sound quality very much but there are one or two things that I thought it could improve- one of them is the need of a remote control which has been addressed in the upcoming HDR. The other problem I have with the pre is the lack of anlog inputs. Anyone who listens to vinyl and SACD will find the pre wanting. If I want to use it as a preamp, I have to use a switch box which is not ideal. 

 Are you offering any exhange deals? People who bought the pre will not likely spend more on the HDR unless they sell first.

 Keep up the good work!_

 

If any of these questions remain, please feel free to ask.

 All the best,
 Elias


----------



## EliasGwinn

Quote:


  Originally Posted by *pcf* /img/forum/go_quote.gif 
_... Like everyone else, I got the impression that it was Benchmark's design decision to omit the remote. 

 ...

 In an ideal world I would have rather seen Benchmark bypassed the released of DAC1pre completely and just go straight to the HDR a little later._

 

The reason a remote control was not included in any previous products was because we were not happy with the solutions available. Just like with any technology, there may be plenty of solutions, but they all carry trade-off's. We couldn't find any remote-control solution that didn't trade the quality of the audio for convenience, which is fundamentally against our design philosophy.

 With the custom-made, motor-driven Alps volume pot, the audio quality isn't sacrificed at all. Once we had that, we were finally satisfied enough to move forward with implementing remote control. 

 We would have liked to include a remote control with the original DAC1 PRE, but we wouldn't do it unless it was as sonically pure as the rest of the product.

 All the best,
 Elias


----------



## EliasGwinn

Quote:


  Originally Posted by *EliasGwinn* /img/forum/go_quote.gif 
_Ok folks, here is the official DAC1 HDR trade-in policy:

 Any DAC1, DAC1 USB, and/or DAC1 PRE that was purchased directly from Benchmark on or after March 1, 2009 may be returned for a full refund of the purchase price towards a DAC1 HDR. The returned unit(s) must be in like-new condition or they will be subject to a restocking fee.

 This offer ends May 1, 2009.

 All the best,
 Elias_

 

Please note that I edited my previous post to include the following addendum:


 "This offer ends May 1, 2009."


 Thanks,
 Elias


----------



## EliasGwinn

Quote:


  Originally Posted by *Quaddy* /img/forum/go_quote.gif 
_elias, can you provide any info on when and how much the HDR will be here in the UK from official distributors?

 thanks._

 

Unfortunately I don't have this information. Please contact your dealer and/or the local distributor (SCV in the UK).

 Thanks,
 Elias


----------



## pcf

Quote:


  Originally Posted by *EliasGwinn* /img/forum/go_quote.gif 
_If you are playing redbook (44.1 kHz / 16-bit) audio without volume control or any other DSP, then upgrading won't make a difference in the sound. However, it is sometimes difficult to know if there is any DSP occuring. If there is DSP of any sort (re-sampling, volume control, etc), the 16-bit limitation will cause distortion due to truncation. 

 In other words, the 96/24 interface doesn't necessarily sound better, but its more capable of handling the data stream without negative effects. 

 Thanks,
 Elias_

 

Hi Elias,

 aCuria answered some of those questions about the remote for me. I still couldn't find a detail list of what the remote control can do on your DAC1 HDR web page. It would also be nice to have a photo of the remote and the black faceplate.
 Another question I have would be the same old one about the analog source. I'm sure you have your reason for keeping just one analog input. But do you have any suggestion for someone like me who needs more than one analog inputs( vinyl; SACD; etc..)? I did try to use the DAC1pre as a preamp but gave up at the end because it's not practical to have to switch inputs in a different switch box (I have one from Mapletree.)all the time. Any advice here?

 Don't know about others- but that would be the major factor for me to decide whether I want to buy the HDR or not. I would like to use it as a preamp rather than just a DAC and headphone amp. I guess in the back of my mind I am worried that Benchmark would come up with a preamp with more than one input after I have bought the HDR.
	

	
	
		
		

		
			





 I love the sound of the DAC1 though!



 Thanks again!

 Paul


----------



## EliasGwinn

Quote:


  Originally Posted by *artears* /img/forum/go_quote.gif 
_Hi Elias,

 I am interested in pre or the hdr model and did not make up my mind yet. I had the chance to audition DAC1 pre before and I did not find it to be too analytical as some will claim here about other versions of DAC1 or DAC1 in general. Maybe that engaging feeling was coming from the synergy between my ACS T2 iems and the pre, who knows. I was planning on getting the pre version until I saw the hdr as well. It seems like the difference between those two are the motorized potentiometer and the remote. There is no upgrade to any other parts regarding dac section on hdr over pre version, am I right? Considering that I mostly will use it with a computer setup, I might as well choose not to get the hdr and save another $300. Are there any sonic differences? Which one do you recommend? 

 I have another question: Regarding the outputs in the back of the unit, does the unit control the volume no matter what is given as input? I am curious if a digital input also enables the use of the volume or the volume control is only limited to whenever an anolog input is connected to it?

 I appreciate your time and efforts here. Thank you very much._

 

Hello Artears,

 The remote control and the new Alps volume and are the only differences between the DAC1 HDR and DAC1 PRE. The new pot will offer some sonic differences. It won't change the 'tone', but it has better inter-channel tracking and longer life. Most pots will suffer from crackling, etc after a few years. This pot will last significantly longer then most.

 The volume control affects all inputs equally.

 Thanks,
 Elias


----------



## EliasGwinn

Quote:


  Originally Posted by *pcf* /img/forum/go_quote.gif 
_...I still couldn't find a detail list of what the remote control can do on your DAC1 HDR web page. It would also be nice to have a photo of the remote and the black faceplate..._

 

The remote will control 'On/Off', Input selection, volume up/down, mute, and dim. 

 We included an 'intelligent dim' setting. The user can set the dim level with the remote, and the DAC1 HDR will remember it. The user can change the 'normal' volume setting, but it won't affect the 'dim' setting stored in memory. Also, adjusting the 'dim' setting won't affect the 'normal' setting. 

 The 'soft mute' is also linked to the user-set 'dim' setting. That is, the soft-mute engages a volume fade from the normal setting to the 'dim' setting before the mute is engaged. After mute is disengaged, the volume ramps from the 'dim' setting back to the 'normal' setting.

 No pictures of the remote control or black faceplate are available yet.

 All the best,
 Elias


----------



## Bostonears

Quote:


  Originally Posted by *EliasGwinn* /img/forum/go_quote.gif 
_Ok folks, here is the official DAC1 HDR trade-in policy:

 Any DAC1, DAC1 USB, and/or DAC1 PRE that was purchased directly from Benchmark on or after March 1, 2009 may be returned for a full refund of the purchase price towards a DAC1 HDR. The returned unit(s) must be in like-new condition or they will be subject to a restocking fee.

 This offer ends May 1, 2009.

 All the best,
 Elias_

 

Shouldn't Benchmark be willing to offer the same return/upgrade policy to anyone who purchased a unit from a Benchmark authorized dealer? 

 My own DAC1 Pre is too old to qualify, and I'm not looking to trade it, but it was purchased from an authorized dealer in the U.S.. If I were the dealer, I would be unhappy about any policy which favors purchases directly from the manufacturer. That makes for a company competing against its own distribution channels on an unlevel playing field.


----------



## aCuria

Quote:


  Originally Posted by *EliasGwinn* /img/forum/go_quote.gif 
_Ok folks, here is the official DAC1 HDR trade-in policy:

 Any DAC1, DAC1 USB, and/or DAC1 PRE that was purchased directly from Benchmark on or after March 1, 2009 may be returned for a full refund of the purchase price towards a DAC1 HDR. The returned unit(s) must be in like-new condition or they will be subject to a restocking fee.

 This offer ends May 1, 2009.

 All the best,
 Elias_

 

Wow! I am really impressed by this. When this topic was broached earlier i did not expect it to happen.
	

	
	
		
		

		
		
	


	




 Now if you have a student deal too...


----------



## 2danes

Thanks Elias for the reply. I am quite pleased with the sound, so I will just leave it as is for now.


----------



## aCuria

Quote:


  Originally Posted by *EliasGwinn* /img/forum/go_quote.gif 
_Common-mode rejection is important for eliminating noise. Also, if a load's differential amp isn't designed properly, it could lead to distortion. Lower source impedance helps prevent this also.

 The DAC1 USB has the 4562's on the output driver section only, while the DAC1 PRE has 4562's all throughout the analog circuit. There are no measurable sonic differences between the two, but many users and reviewers have mentioned preferring the sound of the DAC1 PRE.

 Elias_

 

Is the common-mode rejection something that either happens or does not happen? or does it always occur to some varying degree

 by "all throughout the analog circuit" you mean one at the output driver section and the other... at the analog input?

 "no measurable sonic differences" implies that there is a measurable sonic difference between the vanilla DAC-1 and the pre? If so, what is this difference?

 Lastly, what are your personal views on the sonic differences between the DAC-1 and the USB version?

 Once again thanks for the help. It is much appreciated.


----------



## EliasGwinn

Quote:


  Originally Posted by *Bostonears* /img/forum/go_quote.gif 
_Shouldn't Benchmark be willing to offer the same return/upgrade policy to anyone who purchased a unit from a Benchmark authorized dealer? 

 My own DAC1 Pre is too old to qualify, and I'm not looking to trade it, but it was purchased from an authorized dealer in the U.S.. If I were the dealer, I would be unhappy about any policy which favors purchases directly from the manufacturer. That makes for a company competing against its own distribution channels on an unlevel playing field._

 

Each dealer is free to make their own policy regarding trade-ins. 

 Besides, our trade-in policy is merely a two-week extension of our normal 30-day trial policy. 

 All the best,
 Elias


----------



## EliasGwinn

Quote:


  Originally Posted by *aCuria* /img/forum/go_quote.gif 
_Wow! I am really impressed by this. When this topic was broached earlier i did not expect it to happen.
	

	
	
		
		

		
		
	


	




 Now if you have a student deal too... 
	

	
	
		
		

		
		
	


	


_

 

We don't offer student discounts.

 All the best,
 Elias


----------



## EliasGwinn

Quote:


  Originally Posted by *aCuria* /img/forum/go_quote.gif 
_Is the common-mode rejection something that either happens or does not happen? or does it always occur to some varying degree_

 

It happens to varying degrees. Specifically, it is measured and specified as CMRR (Common Mode Rejection Ratio). 


  Quote:


  Originally Posted by *aCuria* /img/forum/go_quote.gif 
_by "all throughout the analog circuit" you mean one at the output driver section and the other... at the analog input?_

 

There are several gain stages, which use opamps, throughout the analog portion of the circuit. These are all upgraded in the DAC1 PRE and DAC1 HDR.



  Quote:


  Originally Posted by *aCuria* /img/forum/go_quote.gif 
_"no measurable sonic differences" implies that there is a measurable sonic difference between the vanilla DAC-1 and the pre? If so, what is this difference?_

 

There are no measurable sonic differences between the DAC1 USB and DAC1 PRE/HDR. However, the original DAC1, which has the 5532 opamps, may have significant sonic differences in the presence of high load capacitance or low load impedance. In those cases, the audio may experience high-frequency roll-off and/or distortion.

 All the best,
 Elias


----------



## hh83917

I'm glad to heard about the trade-in policy Benchmark is offering to recent DAC1 PRE customers toward the new DAC1 HDR. It is certainly a nice gesture from Benchmark and also signifies the great customer service they provide.

 I'm now waiting for my DAC1 HDR to arrive, hopefully by this week.


----------



## Quaddy




----------



## hh83917

Thanks posting the new picture. I was worried if Benchmark is gonna go with a cheap looking remote. Fortunately the remote in the picture doesn't look too bad.


----------



## G-U-E-S-T

Hi Elias,

 Just wanted to ask, if you could please consider adding those little 12v in/out triggers on the next iteration of the DAC1? 

 It would be so great if we could turn on/off an entire system, via the DAC1 remote control - and would also make it even more "family friendly" for home systems. 
	

	
	
		
		

		
		
	


	




 Thanks again, very much, for all your nice help and support!


----------



## EliasGwinn

Quote:


  Originally Posted by *G-U-E-S-T* /img/forum/go_quote.gif 
_Hi Elias,

 Just wanted to ask, if you could please consider adding those little 12v in/out triggers on the next iteration of the DAC1? 

 It would be so great if we could turn on/off an entire system, via the DAC1 remote control - and would also make it even more "family friendly" for home systems. 
	

	
	
		
		

		
		
	


	




 Thanks again, very much, for all your nice help and support!_

 

Thanks for the suggestion! We will certainly take this into consideration. 

 All the best,
 Elias


----------



## synfreak

Hi Elias!

 Just a thought of how to get things running later on in the EU, because of the new "energy save" laws over here.

 They will force to have any product (at least "entertainmant electronics") must be very restrictive with their power-draw in stanby modes.
 So, if the product doesn't have a dedicated "hard off" switch, the stand-by consumption must be limited to a very low draw (lets say ~1 Watt).

 I don't have the correct numbers/values at hand, but maybe it would be a good idea to get in contact with your european distributers to sort that one out.

 Greetings
 Harald

 P.S.:
 Take a look at: https://www.zvei.org/fileadmin/user_...18_L339-45.pdf

 From page 6:
  Quote:


 (a) Power consumption in ‘off mode’:
 Power consumption of equipment in any off-mode condition shall not exceed 1,00 W.

 (b) Power consumption in ‘standby mode(s)’:
 The power consumption of equipment in any condition providing only a reactivation function, or providing only a
 reactivation function and a mere indication of enabled reactivation function, shall not exceed 1,00 W.

 The power consumption of equipment in any condition providing only information or status display, or providing
 only a combination of reactivation function and information or status display, shall not exceed 2,00 W. 
 

Puuuhh, I hate this kind of "technocratic" language ...


----------



## EliasGwinn

Thanks Synfreak! I hadn't heard about this.


----------



## synfreak

Quote:


  Originally Posted by *EliasGwinn* /img/forum/go_quote.gif 
_Thanks Synfreak! I hadn't heard about this._

 

Hi Elias!

 Yeah, ...
 I think MOST manufacturers from outside, and even the majority from inside the EU, haven't got that one on their business-plans ...

 I'm not shure, how fast these plans will be forced, but if they kick in as estimated for the start of the next year, a lot of currently produced hardware will get some serious problems.





 Harald


----------



## Lord Chaos

This post is only an attempt to get the last page of the thread to load, which it won't do using normal buttons.

 Edit: for some reason the list shows a page 169, but it apparently doesn't really exist.


----------



## Quaddy

Quote:


  Originally Posted by *Lord Chaos* /img/forum/go_quote.gif 
_This post is only an attempt to get the last page of the thread to load, which it won't do using normal buttons.

 Edit: for some reason the list shows a page 169, but it apparently doesn't really exist._

 

_
 this is a known bug, where a phantom +1 page is created, but it doesnt exist, you are on the most current page._

 OT: cant wait to see the dac1 HDR in black, any renders/photos yet benchmark?


----------



## hh83917

I've received my DAC1 HRD today and it's working great. The volume adjustment is much more precise now and having a remote is very convenient. 
	

	
	
		
		

		
		
	


	




 Elias, quick question, is there a reason for changing the optical out port from the retractable cover on the DAC1 PRE to the small detachable cover like the one on the original DAC1? 

 Not like it matters a lot, but I personally found these detachable optical port cover inconvenient because they are small and I can loose it easily.

 Example:

 DAC1 USB, DAC1 PRE





 DAC1, DAC1 HDR





 Thanks,
 Howard.


----------



## Quaddy

/\ congrats on getting the HDR!!!

 please post Photos of the unit and the remote, what colour did you choose?


----------



## EliasGwinn

Quote:


  Originally Posted by *hh83917* /img/forum/go_quote.gif 
_I've received my DAC1 HRD today and it's working great. The volume adjustment is much more precise now and having a remote is very convenient. 
	

	
	
		
		

		
		
	


	




 Elias, quick question, is there a reason for changing the optical out port from the retractable cover on the DAC1 PRE to the small detachable cover like the one on the original DAC1? 

 Not like it matters a lot, but I personally found these detachable optical port cover inconvenient because they are small and I can loose it easily.

 Example:

 DAC1 USB, DAC1 PRE





 DAC1, DAC1 HDR





 Thanks,
 Howard._

 

The 'garage-door' optical ports were susceptible to breaking. The hinged-door broke on too many units, so we are currently using the original style on all units.

 All the best,
 Elias


----------



## hh83917

Quote:


  Originally Posted by *EliasGwinn* /img/forum/go_quote.gif 
_The 'garage-door' optical ports were susceptible to breaking. The hinged-door broke on too many units, so we are currently using the original style on all units.

 All the best,
 Elias_

 

I see, that sound like a good reason to use the traditional optical port. I never had the 'garage-door' port break on me, but it's good to know that Benchmark chooses long-lasting over convenience, because I would have done the same too. Thank you for the quick explanation and I am very happy with the new volume control. 
	

	
	
		
		

		
		
	


	





 Best regards,
 Howard.


----------



## hh83917

Quote:


  Originally Posted by *Quaddy* /img/forum/go_quote.gif 
_/\ congrats on getting the HDR!!!

 please post Photos of the unit and the remote, what colour did you choose?_

 

Thanks, I got the silver one because it matches the iMac better and less apparent when it gets dusty. 
	

	
	
		
		

		
		
	


	




 Few photos of the HDR as requested:
 Front





 Back





 Accessories (basically the same as the other DAC1s other than the extra remote)





 Remote (not the prettiest but gets the job done)





 Remote closer shot (there's on/off, input switch, volume up/down, and mute. I get the control I've wanted for being a couch potato 
	

	
	
		
		

		
		
	


	




)





 Thanks,
 Howard.


----------



## Quaddy

hey thanks howard, much appreciate the eye candy!! good job

 enjoy 
	

	
	
		
		

		
		
	


	




_p.s. i see what you mean, with the sticking out optical plug bung_


----------



## Quaddy

_sry, dble post_


----------



## EliasGwinn

Quote:


  Originally Posted by *hh83917* /img/forum/go_quote.gif 
_I see, that sound like a good reason to use the traditional optical port. I never had the 'garage-door' port break on me, but it's good to know that Benchmark chooses long-lasting over convenience, because I would have done the same too. Thank you for the quick explanation and I am very happy with the new volume control. 
	

	
	
		
		

		
		
	


	





 Best regards,
 Howard._

 

Glad to hear it!

 Let me know if you have any questions, especially regarding the remote. Have you played with the independenet dim control yet? The DAC1 HDR will remember its position even if you change the normal volume. Also, the soft-mute fades to and ramps from the dim setting. Cool, eh?

 All the best,
 Elias


----------



## lamikeith

Those wondering what the DAC1 HDR looks like in black can wonder no more:

DAC1 HDR | Benchmark Media Systems, Inc.
DAC1 HDR Photo Gallery | Benchmark Media Systems, Inc.

 Enjoy!


----------



## hh83917

@Quaddy: Thanks, I'm glad you like it.

 @Elias: Yes, I've just learned about the mute buttons. They are indeed very cool. It will remember the last volume position, which is nice. Also, I found the dim setting pretty useful. I can now press a button and it will go to the preset dim level I've set it as and answer a quick incoming call.

 Another difference I found on the PRE and the HDR is that the HDR does not have auto standby after 15 seconds.

 Elias, can you explain a little about the "Phase-Accurate Multi-Track and 5.1" and how do one hooks up a 5.1 for listening (for future references)? Thanks.


 Regards,
 Howard.


----------



## EliasGwinn

Quote:


  Originally Posted by *hh83917* /img/forum/go_quote.gif 
_@Elias: Yes, I've just learned about the mute buttons. They are indeed very cool. It will remember the last volume position, which is nice. Also, I found the dim setting pretty useful. I can now press a button and it will go to the preset dim level I've set it as and answer a quick incoming call._

 

Great! Yeah, I love the way it works.

  Quote:


  Originally Posted by *hh83917* /img/forum/go_quote.gif 
_Another difference I found on the PRE and the HDR is that the HDR does not have auto standby after 15 seconds._

 

Thats true. We figured it was unnecessary with the remote. 

  Quote:


  Originally Posted by *hh83917* /img/forum/go_quote.gif 
_Elias, can you explain a little about the "Phase-Accurate Multi-Track and 5.1" and how do one hooks up a 5.1 for listening (for future references)? Thanks._

 

Phase-accuracy is a very important consideration when it comes to multi-track audio. Many devices have phase differentials between the various channels of the audio. This will cause inaccuracy in the space perception of the sounds, and can cancel frequencies if the channels are ever summed.

 To take advantage of this feature, you need a 5.1 source with 3 stereo digital outputs. These are _very_ hard to find for consumer media playback systems. However, there is a gentleman who is modding Oppo DVD players to send 5.1 from three spdif outputs. 

 The people that take advantage of the phase accurate 5.1 are studio engineers who have digital interfaces with multiple stereo digital outputs.

 All the best,
 Elias


----------



## lamikeith

I traded in a DAC1 PRE that I had for a few weeks for a DAC1 HDR soon after it was announced. The units are functionally identical except for the remote control.

 The remote volume performs better than expected. With this remote, there are three ways to "mute" the volume: dim, soft mute, and full mute.

 dim: Mechanically lowers the volume POT to a lower "dim" level. The dim level is user adjustable and moves to stay within full volume level rather than remaining fixed. I like it.

 soft mute: Mechanically lowers the volume POT to the dim level, then mutes. 

 mute/off: Pressing the remote "off" button once causes the unit to mute and indicate mute by lighting all LEDs. Pressing "off" again puts the unit in standby. 

 on: Pressing "on" returns the unit to an active, full volume state, no matter which mute/standby state the unit was in previously.

 I've been playing with all of the remote buttons and mute modes, exercising every possible combination, and the unit consistently performs in a logical manner.

 The sound is identical to the DAC1 PRE: excellent.

 This is a nicely engineered, well thought out product. Thank you, Benchmark!


----------



## HeadLover

Is there any SQ different from the PRE to the HDR ???

 BTW
EliasGwinn

 I have heard of a new DAC chip that is capable of doing 32 bit

*ESS 32 Bit Sabre DAC Chip*


http://www.audiophilia.com/wp/?p=1268

 Will there be a new DAC from Benchmark one day using this new chip? seem to be a better quality that what we have right now, not ?


----------



## hh83917

@Elias: Thank you for the explanations.

 @HeadLover: The SQ should be the same as the DAC1 PRE. Main difference for me is that the volume controls on the HDR is more precise now (does not have the dials like the original volume knob) thanks to motorized knob. 
 The new 32bit chip looks interesting, I wonder when will manufacturers start to implement them.

 @lamikeith: I felt the same way too. Other than that, I found myself pressing the remote and watch the volume goes up and down just for fun yesterday...


----------



## emmodad

.... except in the minds of ESS marketing people, that is.


 remember that a DAC of this type, in very simplified terms, looks like:


> input data interface > oversampling digital filter > DAC (modulator) > output conditioning


ESS adds claimed special jitter reduction stage before the modulator


 before anyone's blood pressure rises, this chip is not a 32-bit DAC in terms of performance. this is a DAC chip which:


> - accepts PCM input data in data word sizes up to 32 bits. "32" is simply the possible width of a data word (ie 16, 18, 20, 24 etc resolution data packed in a 32-bit word, which could be a typical data word size coming from ie a digital audio workstation during intermediate processing)
> 
> - contains internal "32-bit" digital filters. this "32" is unspecified, but from public info most probably refers to size of data word accepted at filter input; older public info for the recent/previous generation of this chip indicates that digital filters have internal 48-bit calculation precision, and that the digital filters contribute less than -170 dB of noise as they process signal before it is sent to the modulator. this is performance rather below 32-bit equivalent
> 
> ...


use the search button and google to find numerous scientific and academic articles explaining why reasonable real-world limitations constrain achieveable performance to this range of 21-23 bit-equivalent.


 bottom line: although initial (very uncontrolled) listening tests are indeed reporting positive impressions (which I have no reason to doubt and on which I have no opinion), the ESS 9018 accepts input data words in format up to 32-bit wide. calling this a "32-bit DAC" is simply disingenuous marketing nonsense from ESS. they're not the only ones, AKM is doing something similar in current marketing.


----------



## Quaddy

/\ sure you have got the right thread here?


----------



## The Monkey

Quote:


  Originally Posted by *hh83917* /img/forum/go_quote.gif 
_0000 
 The new 32bit chip looks interesting, I wonder when will manufacturers start to implement them.
_

 

Here's one.


----------



## The Monkey

Quote:


  Originally Posted by *EliasGwinn* /img/forum/go_quote.gif 
_The 'garage-door' optical ports were susceptible to breaking. The hinged-door broke on too many units, so we are currently using the original style on all units.

 All the best,
 Elias_

 

A very smart move. I just had a "garage door" port break on me (on a different brand unit) and it pisses me off to no end.


----------



## emmodad

well, info responding to posts 2532 & 2533.....


----------



## hh83917

@The Monkey: I think the new Headroom one uses the 9008 chip instead of the 9018 that claims 32 bit.

 @emmodad: Thanks for the long explanation. Unfortunately, that's what marketing people do most of the time. Anyways, I'm happy to have my DAC1 now and probably will stick with it for years as I gradually upgrade other components, such as speakers, in the future.


----------



## EliasGwinn

Quote:


  Originally Posted by *emmodad* /img/forum/go_quote.gif 
_before anyone's blood pressure rises, this chip is not a 32-bit DAC in terms of performance. _

 

Exactly. This chip does have some interesting potential characteristics, but it does not acheive 32-bit performance. Bit-depth only affects the dynamic range (aka signal-to-noise ratio). In other words, it won't affect the tone of the DAC, merely the noise floor. 

 The aspect of DAC chips that could use a real breakthrough is filter performance. Currently, there is not enought horsepower (DSP) in a DAC chip to effectively eliminate all frequencies above Nyquist while remaining linear for all frequencies below Nyquist. 

 All the best,
 Elias


----------



## emmodad

Quote:


  Originally Posted by *EliasGwinn* /img/forum/go_quote.gif 
_Exactly. This chip does have some interesting potential characteristics, but it does not acheive 32-bit performance. Bit-depth only affects the dynamic range (aka signal-to-noise ratio). In other words, it won't affect the tone of the DAC, merely the noise floor._

 

yes, although this chip provides no better dynamic range ability than several other high-end devices as mentioned (TI, Wolfson,). It provides only an input data format ability to accept 32-bit wide data words, and (apparently) to pass those 32-bit wide data words to input of the digital filters. Some other high-quality DAC chips have data input formats only up to and including 24-bit wide words.


  Quote:


  Originally Posted by *EliasGwinn* /img/forum/go_quote.gif 
_The aspect of DAC chips that could use a real breakthrough is filter performance. Currently, there is not enought horsepower (DSP) in a DAC chip to effectively eliminate all frequencies above Nyquist while remaining linear for all frequencies below Nyquist._

 

Integrated Circuit architectures are always subject to tradeoffs of cost / performance / etc. DAC chips (such as those used in the DAC1) designed in the early 1990s were subject to constraints of manufacturing technology and costs of that era, hence the amount of silicon area dedicated to the digital filtering engine was limited (and generally hardwired, ie not a "software code on DSP" implementation)

 this is why, ie, the architecture of the Berkeley Audio Design alpha DAC is an interesting example (digital filtering is performed in an ADI SHARC DSP external to and prior to the actual DAC chip); and one of the interesting points of the new flagship DAC chips from Wolfson (a product designer has the choice of using Wolfson's on-chip digital filtering engine, for very good performance at lowest parts cost, or putting a DSP (as digital filtering engine) in front of the DAC chip, and use the Wolfson with its on-chip digital filtering engine bypassed).

 so, I'm waiting to see what John does with a block of dsp horsepower in front of a DAC chip with excellent performing modulator (hint Wolfson or ADI). Oh: and remind him to do HDCD decoding in that dsp; price the product at a premium to DAC1 but well below alpha; and the resulting "DAC2HDCD" will sell like wildfire.

 wishful thinking ps: give it a native FireWire interface...... 24/192 through a modern John Siau design...... mmmmmmmmmmm.


----------



## gregeas

Agreed with emmodad we are entering an era of new potential for high-end DACs. For the last five years or so it has struck me how little evolution there was in DAC design. But now we have new, interesting products from Ayre, Weiss, Berkeley, PS Audio, etc. The ESS Sabre chip is going into some interesting products as well. The acceptance of PC audio by audiophiles is opening up the market. 

 In the future DAC designers should consider adding a HDMI input. My laptop has an HDMI port, and I'm experimenting with this connection to my Cary pre-pro. I was surprised that everything worked, and the quality was quite good. Blu-ray players like the new Oppo are universal and can output hi-res PCM and DSD via HDMI. 

 BTW, I'm back to using a Benchmark (Pre) in my main rig. I have a PS3, cable box, and laptop connected. Works great connected to my speaker amp!


----------



## HeadLover

I hope some really new stuff be out soon, something really "revolutionary" kind of.

 And who knows, maybe there will be a DAC2 some day


----------



## EliasGwinn

Quote:


  Originally Posted by *emmodad* /img/forum/go_quote.gif 
_Integrated Circuit architectures are always subject to tradeoffs of cost / performance / etc. DAC chips (such as those used in the DAC1) designed in the early 1990s were subject to constraints of manufacturing technology and costs of that era, hence the amount of silicon area dedicated to the digital filtering engine was limited (and generally hardwired, ie not a "software code on DSP" implementation)

 this is why, ie, the architecture of the Berkeley Audio Design alpha DAC is an interesting example (digital filtering is performed in an ADI SHARC DSP external to and prior to the actual DAC chip); and one of the interesting points of the new flagship DAC chips from Wolfson (a product designer has the choice of using Wolfson's on-chip digital filtering engine, for very good performance at lowest parts cost, or putting a DSP (as digital filtering engine) in front of the DAC chip, and use the Wolfson with its on-chip digital filtering engine bypassed)._

 

Exactly. This is what we do with the ASRC, as well. The bulk of our filtering is done in an Analog Devices AD1896 ASRC chip. The AD1896 is a digital-only chip, so it doesn't have to be 'room-mates' with analog circuitry. This free's the chip designer to build a DSP work-horse that can execute thoroughly.

  Quote:


  Originally Posted by *emmodad* /img/forum/go_quote.gif 
_so, I'm waiting to see what John does with a block of dsp horsepower in front of a DAC chip with excellent performing modulator (hint Wolfson or ADI). Oh: and remind him to do HDCD decoding in that dsp; price the product at a premium to DAC1 but well below alpha; and the resulting "DAC2HDCD" will sell like wildfire.

 wishful thinking ps: give it a native FireWire interface...... 24/192 through a modern John Siau design...... mmmmmmmmmmm. 
	

	
	
		
		

		
		
	


	


_

 

Great feedback. I will forward this to the man, himself.

 All the best,
 Elias


----------



## Lord Chaos

Add me to the list of folks who'd like to have a DAC1 Firewire.


----------



## EliasGwinn

Quote:


  Originally Posted by *hh83917* /img/forum/go_quote.gif 
_Remote (not the prettiest but gets the job done)_

 

The remote was selected for its simplicity, long range, and resistance to interference from other remotes. We developed our own hardware IR decoder to eliminate the need to place a noisy microprocessor in the DAC1 chassis. The exceptional timing accuracy of our hardware IR decoder gives the DAC1 HDR exceptional immunity to false triggers caused by other IR remotes, Video sceens, and high-efficiency flourescent lighting. A metal remote would have added significant costs without improving audio quality. We invested money in the audio path (custom Alps pot) rather than on the remote control housing. Other manufacturers have implemented cheap IC-based analog or digital volume controls but increase price because of a fancy remote. 

 We expect many of our users will control the DAC1 HDR from a universal remote such as one from the Logitch Harmony series. Harmony files supporting the HDR will be available soon. Until then, the Harmony remotes can learn the HDR command set.

 All the best,
 Elias


----------



## hh83917

Quote:


  Originally Posted by *EliasGwinn* /img/forum/go_quote.gif 
_The remote was selected for its simplicity, long range, and resistance to interference from other remotes. We developed our own hardware IR decoder to eliminate the need to place a noisy microprocessor in the DAC1 chassis. The exceptional timing accuracy of our hardware IR decoder...etc._

 

Elias,

 Don't worry, I wasn't complaining. I like the remote a lot and its key placements. Frankly, I quite enjoy the simplicity of it. The remote works very well with the IR decoder in the HDR and, thanks to Benchmark's hard work, is extremely responsive to the remote. I only wonder if its durability will match the DAC1. But, if I can always order a replacement if it ever break or get lost, I don't see a problem with it.

 I currently use the HDR mainly on my iMac, but talking about the Harmony remotes, I do own two of them, the old 880 and the newer Harmony One. I use the Harmony One on my home theater system, but although it drops the learning curve and complexities for the rest of the family, I found it to be a bit laggy and unresponsive even after tweaking (personal opinion). The relationship is something like: "I like it, but it can be much better...", but I guess that's Logitech's problem. Anyways, it is nice to see Benchmark submitting the remote codes to the Harmony database. It will make many home theater enthusiasts very happy. 
	

	
	
		
		

		
			





 Best regards,
 Howard.


----------



## G-U-E-S-T

Hi Elias,

 Naive question please: With the new remote-motorized volume control, is it better to always use the remote for volume adjustment, instead of turning the volume control manually? I.E. Is it now discouraged (or perhaps risky to the operation of the motorized pot etc), to make a habit of just adjusting the volume control by hand?


----------



## little-endian

Hi Elias,

 after quite a while, I'd have two questions about the DAC1's volume control, important for me in particular at present:

 After having not used the DAC1 (the classic version) for about a month, I encountered slight crackling noises when adjusting the volume. Since it is for the time of adjustment only, it is definitely due to the poti, what makes be wonder if that's such a good sign. It disappears again after some movements and otherwise, the DAC1 is in best shape, though.

 Benchmarks claims that their decision to use a poti instead of an electronic volume control would be that the last one wouldn't archive the same quality - also in terms of dynamic range.

 However, that seems somehow odd to me. Aren't there nowadays suitable DSPs, able to handle 32-bit or more floating point precision to control the volume by math? If it is not implemented correctly, one would get a higher noise floor of course but is there really no way for Benchmark to implement this? Somehow, this analog poti seems like the only real "weak point" of the DAC1 since prone to wear...


----------



## EliasGwinn

Quote:


  Originally Posted by *G-U-E-S-T* /img/forum/go_quote.gif 
_Hi Elias,

 Naive question please: With the new remote-motorized volume control, is it better to always use the remote for volume adjustment, instead of turning the volume control manually? I.E. Is it now discouraged (or perhaps risky to the operation of the motorized pot etc), to make a habit of just adjusting the volume control by hand?_

 

Not at all. The motor is built with a clutch so that manual adjustments (including overriding the motor while it is being driven) will not fatigue the motor or potentiometer.

 Feel free to adjust the volume by hand or remote or both (even at the same time!).

 All the best,
 Elias


----------



## EliasGwinn

Quote:


  Originally Posted by *little-endian* /img/forum/go_quote.gif 
_Hi Elias,

 after quite a while, I'd have two questions about the DAC1's volume control, important for me in particular at present:

 After having not used the DAC1 (the classic version) for about a month, I encountered slight crackling noises when adjusting the volume. Since it is for the time of adjustment only, it is definitely due to the poti, what makes be wonder if that's such a good sign. It disappears again after some movements and otherwise, the DAC1 is in best shape, though.

 Benchmarks claims that their decision to use a poti instead of an electronic volume control would be that the last one wouldn't archive the same quality - also in terms of dynamic range.

 However, that seems somehow odd to me. Aren't there nowadays suitable DSPs, able to handle 32-bit or more floating point precision to control the volume by math? If it is not implemented correctly, one would get a higher noise floor of course but is there really no way for Benchmark to implement this? Somehow, this analog poti seems like the only real "weak point" of the DAC1 since prone to wear... 
	

	
	
		
		

		
		
	


	


_

 

We can replace your potentiometer. Call us to set up an RMA (1-800-BNCHMRK / 1-800-262-4675).

 The DAC1 HDR's potentiometer is much higher quality, and is less prone to this issue.

 With regards to a DSP volume control, digital attenuation will always reduce the dynamic range of a converter, no matter how well it is built. I'm actually writing an article for our Feedback Newsletter outlining the various volume control mechanisms. 

 All the best,
 Elias


----------



## ert

Speaking of knobs, the source knob on my DAC1 PRE is loose. Is there a way to tighten the knob on the shaft? Sometimes there's a set screw, but I can't see one.


----------



## little-endian

Hi Elias,

 thanks for the quick reply, also if it is not what I wanted to hear. 
	

	
	
		
		

		
		
	


	




 Hmm, so I interprete that as a "not so good sign". What causes that behavour actually? Why does it disappear again after a while of usage? Seems like a contact issue. Besides the slight noise occuring during the adjustment, are there any other degrades in sound quality to expect (in theory)?

 I called the number you had given, however I seem to have missed office hours for today already.

 Some more questions in advance:

 1. If I do nothing, will it get worse or remain more or less the same?

 2. I purchased the device 2007 from the official seller in Germany at that time, Analog Audio. How would such a RMA take place in detail? What about shipping / customs fuss?

 3. If I decide to get it replaced, would it be possible to get such a potentiometer used for the DAC1 HDR or even replace the whole DAC1 to a HDR (paying the difference)?

 I'm keen on reading your article about volume controls since I'm still confused why volume controls, offered by some software like iTunes and others are said to have gained quite a nice performance meanwhile while implementing this into a DAC shall still be that big problem.


----------



## EliasGwinn

Quote:


  Originally Posted by *ert* /img/forum/go_quote.gif 
_Speaking of knobs, the source knob on my DAC1 PRE is loose. Is there a way to tighten the knob on the shaft? Sometimes there's a set screw, but I can't see one._

 

The source knob is a friction fit...i.e, there is no set screw. I've never seen one come loose. Hmmm...??? 

 We can replace the knob and/or source switch for you. Call us (9a-5p EST)to set up an RMA (1-800-BNCHMRK / 1-800-262-4675).

 All the best,
 Elias


----------



## EliasGwinn

Quote:


  Originally Posted by *little-endian* /img/forum/go_quote.gif 
_Hmm, so I interprete that as a "not so good sign". What causes that behavour actually? Why does it disappear again after a while of usage? Seems like a contact issue. Besides the slight noise occuring during the adjustment, are there any other degrades in sound quality to expect (in theory)?

 1. If I do nothing, will it get worse or remain more or less the same?

 2. I purchased the device 2007 from the official seller in Germany at that time, Analog Audio. How would such a RMA take place in detail? What about shipping / customs fuss?

 3. If I decide to get it replaced, would it be possible to get such a potentiometer used for the DAC1 HDR or even replace the whole DAC1 to a HDR (paying the difference)?

 I'm keen on reading your article about volume controls since I'm still confused why volume controls, offered by some software like iTunes and others are said to have gained quite a nice performance meanwhile while implementing this into a DAC shall still be that big problem._

 

Contact Analog Audio to have your DAC1 PRE repaired. They are an authorized repair station for us. 

 The pot noise doesn't affect the overall performance of the DAC1 PRE, it simply makes a noise during rotation. It is an easy repair, and if your warranty is valid, it will be covered.

 Regarding upgrading, the DAC1 PRE cannot be retro-fitted with the new custom Alps pot. Ask Analog Audio about their policies for trade-ins, but I think that they do not allow it.

 DSP volume controls, such as that in iTunes or one built into a DAC, can perform wonderfully. However, ALL pre-DAC digital attenuation will lower the dynamic range. In other words, they will reduce the signal-to-noise ratio. Here's why: 

 Imagine a room with 10 foot ceilngs. In this example, the 'ceiling' is the 'maximum output' from the converter. The 'floor' is the 'noise floor' of the converter. The 'height' of the room is the 'dynamic range'. The noise floor of the converter is constant, regardless of the level of the input signal. If the ceiling (volume) is too high, you can install a suspended ceiling (digitally attenuate the volume). However, the floor will remain at the same level. By using digital attenuation, the ceiling is lowered, but the 'noise floor' stays the same. Therefore, you have less height (dynamic range).

 With post-DAC attenuation, you attenuate everything coming from the DAC: the signal (ceiling) AND the noise (floor). Therefore, the height (dynamic range) remains the same (assuming you are using an attenuator that doesn't contribute too much noise).

 Does that make sense?

 Thanks,
 Elias


----------



## little-endian

Hello Elias,

 thanks again.

 Well, actually it is the "old" DAC1 without USB - the one Allen Burdick owned before to be exact. 
	

	
	
		
		

		
		
	


	




 Hmm, right now the slight noise is gone. It is not really an issue (right now), more a psychological one (a weak point of the 'perfect' DAC1 since other parts like the chipset have no wear at all, right?)

 Actually, that topic deserves an own thread in my opinion but I'll risk it to post it here:

 Yeah, nice example. However, maybe you forgot to mention the floor's basement below. 
	

	
	
		
		

		
		
	


	




 Feel free to correct me if I should be wrong; the following according to my current knowledge:

 1. Dynamic range and signal to noise ratio are not exactly the same, although normally used synonymously, right? At least, the "mastering guru" Bob Katz explained that with usage of dithering, one may archive a higher dynamic range than the SNR (for 16 bit a DR of about 120 dB and SNR of 91 dB for instance). It seems that information can be retrieved even at negative SNRs, where the noise is louder than the actual signal.

 2. In the case of the DAC1, the SNR is about ~ 116 dB according to the data sheet. No word about the dynamic range, though. As far as remember, how already stated somewhere here that the DAC1 would be able to perform at true 24-bit accuracy, however would be limited in terms of SNR by its own noise @ -116dBFS.

 3. If we adjust the volume before converting it to analog, there are two causes of quality degradation in theory: Due to the performed math and rounding errors and the fact that the maximum values reached by the audio signal would be always lower than 0dBFS, reducing the dynamic range by definition - right?

 So I understand your point, however I have a "crucial question" as a 'counterstrike':

 Of course, our ear's dynamic range (and signal to noise ratio as well I think) is limited also of course. So when using a poti (assuming it won't introduce any noise by its own), the signal as well as the noise would be lowered the same, keeping the dynamic range the same by definition (DR = max / min). Assuming some music that reaches 90 dB in its loudest parts at a given playback volume and calm periods at 20 dB. Now I lower the volume with your poti. In theory, the dynamic range stays the same, right? But now the loudest parts may not louder than 60 dB for instance. What happens with the low parts then? Aren't they lost anyway because they would below 0dBSPL? As far as I know, the maximum hearable dynamic range is limited by the playback volume as well - if one wants to hear something on a lower volume (than intended), one has to choose wether to lose the lower parts or to compress the dynamic range (like Dolby's midnight modes offer it).

 So far my confusion.


----------



## EliasGwinn

Quote:


  Originally Posted by *little-endian* /img/forum/go_quote.gif 
_1. Dynamic range and signal to noise ratio are not exactly the same, although normally used synonymously, right? At least, the "mastering guru" Bob Katz explained that with usage of dithering, one may archive a higher dynamic range than the SNR (for 16 bit a DR of about 120 dB and SNR of 91 dB for instance). It seems that information can be retrieved even at negative SNRs, where the noise is louder than the actual signal._

 

Yes and no...there are a lot of conflicting opinions on this. Dynamic range (DR) and signal-to-noise ratio (SNR) spec's are effectively telling the same story for an audio component. However, they are often calculated and stated differently.

 Here's the story behind DR and SNR measurements of digital audio gear: SNR is calculated by measuring the output level of a device with a known input signal, then measuring the noise floor with no input signal. The difference between those measurements (the max output and noise floor) is the SNR. 

 However, many manufacturers of DAC's, CD players, etc, install auto-mute circuits to mute the DAC's output when there was no digital audio input. This caused false SNR measurements because the noise measurements were artificially low. DR measurements were introduced to defeat this. DR is calculated by measureing the noise floor while sending a very low input signal (-60 dBFS) into the device. This low-level signal prevents the auto-mute circuit from engaging. (An example of a suspicious SNR spec is the Bryston BDA-1, which uses the Cirrus CS4398 DAC chip. The Cirrus chip has a dynamic range of 120 dB. However, the Bryston BDA-1 claims to have a "signal-to-noise ratio of -140 dB". Even if they paralelled 8 of the Cirrus chips, they couldn't reduce the noise level 20 dB. The Bryston needs to be tested with a low-level signal to determine the dynamic range _without_ muting.)



http://www.cirrus.com/en/pubs/proDat.../CS4398_F1.pdf

 To answer your other question... it is true that a properly-dithered digital recording can capture details well below the noise floor. It is also true that human's can hear sounds below the noise floor. There have been many demonstrations of this. 

 Dynamic range, as Bob Katz was referring to it, can also describe the difference between the highest and lowest levels of sound. For example, the dynamic range of a recording describes the difference between the loudest passages and the quietest passages. In this case, dynamic range has nothing to do with the noise floor (unless the quietest passage is REALLY quiet).

  Quote:


  Originally Posted by *little-endian* /img/forum/go_quote.gif 
_2. In the case of the DAC1, the SNR is about ~ 116 dB according to the data sheet. No word about the dynamic range, though. As far as remember, how already stated somewhere here that the DAC1 would be able to perform at true 24-bit accuracy, however would be limited in terms of SNR by its own noise @ -116dBFS._

 

Correct. Specifically, the intrinsic noise of the D/A chip is the dominating noise contributor. This is the case with most DAC's (although some don't like to admit it...see comments above)

  Quote:


  Originally Posted by *little-endian* /img/forum/go_quote.gif 
_3. If we adjust the volume before converting it to analog, there are two causes of quality degradation in theory: Due to the performed math and rounding errors and the fact that the maximum values reached by the audio signal would be always lower than 0dBFS, reducing the dynamic range by definition - right?_

 

A properly dithered, high-quality digital volume control won't degrade the signal. It will simply lower the dynamic range of the DAC. If the digital audio is dithered to 16-bits (as in the case with 16-bit USB devices), it will also increase the noise level of the recording.

  Quote:


  Originally Posted by *little-endian* /img/forum/go_quote.gif 
_Of course, our ear's dynamic range (and signal to noise ratio as well I think) is limited also of course. So when using a poti (assuming it won't introduce any noise by its own), the signal as well as the noise would be lowered the same, keeping the dynamic range the same by definition (DR = max / min). Assuming some music that reaches 90 dB in its loudest parts at a given playback volume and calm periods at 20 dB. Now I lower the volume with your poti. In theory, the dynamic range stays the same, right? But now the loudest parts may not louder than 60 dB for instance. What happens with the low parts then? Aren't they lost anyway because they would below 0dBSPL? As far as I know, the maximum hearable dynamic range is limited by the playback volume as well - if one wants to hear something on a lower volume (than intended), one has to choose wether to lose the lower parts or to compress the dynamic range (like Dolby's midnight modes offer it)._

 

Not really. Supposedly the dynamic range of human hearing is near 100 dB. Recorded music rarely has 40 dB of dynamic range, let alone 70 dB as in your example. 

 Most importantly, your example outlines the reason for proper gain staging. If your amplifier is the correct size (maximum output is barely louder then the loudest you would ever listen) and gain stages of your devices are properly configured (keeping signals as hot as possible between devices and attenuate for volume control only near the end of the signal path), the dynamic range will be maintained. In other words, full-digital amplitude into the DAC1 -> analog volume control -> proper-size amplifier -> speakers.

 All the best,
 Elias


----------



## geremy

Has anyone compared the DAC1 *USB or PRE* with the Lavry DA10 or DA11? I am having a hard time deciding between these two units. I appreciate Benchmark's openness concerning the design and performance of their unit. Lavry does not operate in the same manner, so it is hard to make direct comparisons. 

 I have found plenty of reviews pitting the DAC1 vs. the DA10, but this was before Benchmark upgraded the op-amps in the analog path. 

 Thanks.


----------



## aCuria

Quote:


  Originally Posted by *geremy* /img/forum/go_quote.gif 
_Has anyone compared the DAC1 *USB or PRE* with the Lavry DA10 or DA11? I am having a hard time deciding between these two units. I appreciate Benchmark's openness concerning the design and performance of their unit. Lavry does not operate in the same manner, so it is hard to make direct comparisons. 

 I have found plenty of reviews pitting the DAC1 vs. the DA10, but this was before Benchmark upgraded the op-amps in the analog path. 

 Thanks._

 

Theres a thread here.
http://www.head-fi.org/forums/f7/ben...6/#post5686982

 The USB and PRE are relatively new and there are few direct comparisons.


----------



## geremy

Thank you for the link!


----------



## ert

Quote:


  Originally Posted by *EliasGwinn* /img/forum/go_quote.gif 
_The source knob is a friction fit...i.e, there is no set screw. I've never seen one come loose. Hmmm...??? 

 We can replace the knob and/or source switch for you. Call us (9a-5p EST)to set up an RMA (1-800-BNCHMRK / 1-800-262-4675).

 All the best,
 Elias_

 

Thanks for the offer, but it's hardly loose, really, I was just looking for an excuse to take it apart 
	

	
	
		
		

		
			





 . It's in no danger of coming off, just every so slightly wobbly side to side with no apparent loss in functionality.


----------



## Wiza_Gab

1 : What is better as for connecting it to a computer (sound quality wise) ? USB, optical audio, etc ?

 2 : Does it bypass the PC soundcard no matter how you connect it ?

 3 : The DAC alone, does it compare to Stello DA100 or Havana ? Both are in the same price range but have no headphones amp, are their DACs better ?

 4 : how do the DAC-1 couple with a tubed amp like RSA Raptor, is it a good match ?

 Thanks a lot !


----------



## EliasGwinn

Quote:


  Originally Posted by *Wiza_Gab* /img/forum/go_quote.gif 
_1 : What is better as for connecting it to a computer (sound quality wise) ? USB, optical audio, etc ?_

 

This all depends on which USB and optical interface you're talking about.

 With the USB interface on DAC1 products, the USB is always the safest bet. However, with Mac's, the optical output will provide equal performance.

 With other USB and optical interfaces, all bets are off. It all depends on the implementation. One optical interface may be better then another USB interface, but a different USB interface may be better then both!

  Quote:


  Originally Posted by *Wiza_Gab* /img/forum/go_quote.gif 
_2 : Does it bypass the PC soundcard no matter how you connect it ?_

 

USB and optical interfaces _usually_ bypass the D/A of the internal soundcard.

  Quote:


  Originally Posted by *Wiza_Gab* /img/forum/go_quote.gif 
_3 : The DAC alone, does it compare to Stello DA100 or Havana ? Both are in the same price range but have no headphones amp, are their DACs better ?_

 

I refrain from making subjective statements about competitors' products. But I'll say these two things:

 1. Our DAC and headphone amplifier are used by some of the biggest recording studios, mastering studios, film studios, television production facilities, and radio studios in the world. Our clients include Skywalker Sound, ABC, NPR, Boston Symphony Orchestra, Lurssen Mastering, Michael Wagner, etc.

 2. We offer a 30-day trail on all of our products. You can perform head-to-head comparisons of our product with any others, and if you decide you don’t want to keep our product, simply return it within 30 days, and you’ll be fully refunded (provided that it is returned in ‘like-new’ condition). 

  Quote:


  Originally Posted by *Wiza_Gab* /img/forum/go_quote.gif 
_4 : how do the DAC-1 couple with a tubed amp like RSA Raptor, is it a good match ?_

 

Again, I'll refrain from making subjective statements. I'm sure other members of the head-fi community or the audio community at-large will be happy to comment on this.

  Quote:


  Originally Posted by *Wiza_Gab* /img/forum/go_quote.gif 
_Thanks a lot !_

 

No Worries! 
	

	
	
		
		

		
		
	


	




 All the best,
 Elias


----------



## G-U-E-S-T

Quote:


  Originally Posted by *EliasGwinn* /img/forum/go_quote.gif 
_...The DAC1 HDR's potentiometer is much higher quality, and is less prone to this issue..._

 

Hi Elias,

 I noticed you mentioned "less" prone - does this imply that the new pot is probably still susceptible (to some lesser degree) however, to this 'crackling' phenomenon, over time?

 Also please, what specifically is responsible for this crackling sound in a worn volume pot, and what is the particular difference in the new pot that makes it less susceptible to the issue over time? That is, do these volume pots use "brushes" (or some other culprit component) internally, and what is the specific mechanical difference between the old/new pots with respect to such a specific component, that should make the new pot better in this respect?

 Thanks again in advance, as always!


----------



## peter73

Hi Elias,

 Thank you for taking so much of your time to answer all our questions about your marvellous DAC.

 I have an observation on mine that I would like to share. I have just recently (and totally by chance) found out that the two headphone outs actually sound slightly different. The one on the right (closer to the volume pot) sounds more transparent, opened and easier on the ears - as if I am into the music, while when listening through the other one the sound is flatter, shut in and a bit distant and it also causes slight fatigue after having listened for some time. Please consider that the difference, although clearly distinguishable, is not significant, but yet big enough to make me accept having a headphone jack plugged close enough to the volume control to cause inconvenience when operating it.

 Did you have similar experience with the DAC1's you've heard and what might be the explanation?

 Thank you,
 Peter


----------



## mopps

Quote:


  Originally Posted by *peter73* /img/forum/go_quote.gif 
_I have an observation on mine that I would like to share. I have just recently (and totally by chance) found out that the two headphone outs actually sound slightly different. The one on the right (closer to the volume pot) sounds more transparent, opened and easier on the ears - as if I am into the music, while when listening through the other one the sound is flatter, shut in and a bit distant and it also causes slight fatigue after having listened for some time. Please consider that the difference, although clearly distinguishable, is not significant, but yet big enough to make me accept having a headphone jack plugged close enough to the volume control to cause inconvenience when operating it.

 Did you have similar experience with the DAC1's you've heard and what might be the explanation?_

 

Hi Peter,

 I'm not a Lavry or Benchmark user (not yet), but had similar observations with my Lake People G99/2, a German professional reference headphone amp (the newer models are the G100, V100, V200). I also heard a slight difference between its two headphone outputs, using my Ultrasone Edition 9 and a borrowed Edition 9 (both well burnt-in and both sounding identical) by plugging them back and forth.

 The manufacturer explained that the two outputs are built absolutely similar and in parallel, but admitted that, after a longer period of using and plugging the headphones in and out, there might occur some (hardly) measurable small electrical differences in comparison to an unused or barely used jack, due to the very quick (even a few tens of plugging procedures can be sufficient) abrasion of the gold (worst) or silver (slightly better) coating of the big plugs and jacks, even with the built in nearly perfect Neutrik's. In my device the almost exlusively used jack was the one farther away from the "dangerous" on/off switch, and this was the worse sounding one.

 The manufacturer elaborated on the special sensitivity of modern low-impedance headphones to this electrical phenomenon and its potential audibility. According to him, alone for that reason XLR-connections (for unbalanced operation one 3-pin XLR) make sense and should be the new industry standard, instead of the old TRS-connections which originate from the area of the 600 - 2000 ohms monsters. Or in reverse: low impedance headphones with TRS-connectors seem to be nonsense, if TRS then better go for tradional high impedance headphones.


----------



## peter73

Thank you, Mopps,
 It does make sense.


----------



## BitPerfect

Nice review 
	

	
	
		
		

		
		
	


	




Benchmark DAC1 HDR Review | Computer Audiophile


----------



## G-U-E-S-T

It seems we've lost contact with Elias - hey Elias, are you still there? I hope all our ongoing questions haven't totally worn you out!


----------



## EliasGwinn

Quote:


  Originally Posted by *G-U-E-S-T* /img/forum/go_quote.gif 
_It seems we've lost contact with Elias - hey Elias, are you still there? I hope all our ongoing questions haven't totally worn you out! 
	

	
	
		
		

		
		
	


	


_

 

Hi GUEST,

 I'm still here. I am still catching up from the last 1.5 weeks spent in L.A. for the Head-Fi meet (obviously) as well as other business engagements.

 I hope to post full replies to the open-ended posts as soon as possible.

 All the best,
 Elias


----------



## doggiehowser

I am surprised that Elias hasn't run out of steam after so many years on this thread 
	

	
	
		
		

		
		
	


	




 I am only at page 100+

 In any case, I have been using the DAC1 Pre for a couple of months now. I recall reading that some people have commented there are very few "satisfied customers" testimony.. so here's mine.

 Like many here, I started off my journey in headfi. But the iMod 80GB was getting severely short on storage space 

 It was about that time that 
 a. I noticed my left ear wasn't as sensitive as the right (not sure if it is cos of old age or too much loud music) and I began to consider going the speaker/hifi route
 b. I was helping my father in law assemble a new hifi gear to replace his old Quad electrostatics and got interested in hifi in general
 c. I heard about the Wadia iTransport and figured it'd be a good partner for the newer 6G iPods with the 160GB storage (but with the not so hot DAC).

 But for the Wadia to work, I needed a DAC. I figured I could stretch my budget to the DAC1. Of all the DACs in the market, one of the things that made the DAC1 so appealing to me was that it was relatively immune to jitter. I liked that one reviewer even used a "coat hanger" and an old PC CDROM writer to double up as a source. I knew the jitter on the Wadia was bad. And I was also a cheapskate so I didn't think I would spend that much on cables (but that's going to change)

 My local dealer had run out of stock of the DAC1 (classic) and the DAC1 USB, so I had to spring for the DAC1 Pre which meant really going out of my original budget. But in hindsight (looking at the improvements from USB to Pre as posted by the members here and by Elias), I am quite glad I made that choice (chassis mounted RCA etc).

 I initially used the DAC1 Pre as a DAC/pre-amp, replacing my iMod/cap dock/Axiom passive preamp with the Wadia/iPod/DAC1 Pre, hooking up to my father-in-law's old Quad606 power amp and B&W601S2 that I was using for home theater duties.

 Since then, I have upgraded the rest of the components. And at each stage, the DAC1's clarity has been revealed even more, after being hidden by less competent gear before. Initially, I still prefered my headfi gear. They were far more detailed and more musical. But in its current stage, that is no longer the case.

 At the moment, the 3 coax inputs have been used (Squeezebox Duet, Wadia iTransport and the coax from the Marantz SA8003 for redbook audio). No optical. Ironically, I still have not used the USB interface 
	

	
	
		
		

		
		
	


	




 which is the source of a lot of controversy here.

 DAC1 Pre is no longer working as a pre-amp and is set to calibrated output.

 XLR output is sent to the Bel Canto Pre3. I got it primarily for the Home Theater (HT bypass) feature but have noticed there was an improvement in soundstage. 

 As I began ripping my old CD collections into iTunes, I also noticed that I have some really badly mastered CDs 
	

	
	
		
		

		
		
	


	




 And the Benchmark with all its clarity revealed just how bad they were. So I have begun to feed the RCA output to a Yaqin CD2 tube buffer which mellows out the sound a little before feeding it to the Pre3. 

 So I have the best of both worlds. The XLR for the clean Benchmark sound.. and RCA (buffered with a tube) when the CDs are not so hot. 

 I suspect that the arguments for/against modding are actually looking at it from different angles.

 I think Elias and Benchmark are right in that in its current state, the Benchmark DACs are at their most neutral/accurate settings. That's what they strove to achieve for a studio environment and that's what they delivered. And modding the DAC may yield a scientifically less "accurate" device.

 But it is the deviation from perfection that may make the sound appear fuller or warmer or fuller, which some users may like. (not unlike what the tube buffer is doing)


----------



## doggiehowser

Quote:


  Originally Posted by *yipchunyu* /img/forum/go_quote.gif 
_Hi EliasGwinn,
 I just got the DAC 1 Pre.
 I found that the analog input's sound input is very low. (i input from pre-out of my av amp).
 When I connect via digital input, i set the volume close to the min.
 But when I use the analog input, I need to set the volume to near to the max.
 Any tips on this?_

 

Not sure if you are still in this forum since this was from months ago 
	

	
	
		
		

		
		
	


	




 But I am "sharing" my hifi setup (power amp/speakers) with my AVR home theater setup as well.

 Initially, I was using the DAC1 Pre as the pre-amp. So the AVR pre-out goes to the DAC1 Pre's analog input.

 The AVR's Pre-out is already attenuated by its own volume controller. So it is going to be much lower in level than a full blown line input (like from a CD player). That explains the difference in loudness you are experiencing.

 Switching to the AVR's tape out is NOT a solution, since I expect you would want the AVR's volume knob to control ALL the channels as well accordingly (ie center/surround/sub/surround back as well as the front L/R). If you did that, the AVR's volume knob would control all the channels EXCEPT the front L/R (which is controlled by the DAC1 Pre). And my ears aren't golden enough to do this kind of level matching 
	

	
	
		
		

		
		
	


	




 The solution I used... may or may not work for you.

 I dialed the DAC1 Pre volume knob abt halfway (this is too loud for listening from a normal digital source BTW on my setup).

 I sent some test tones from the AVR and used my RadioShack SPL to adjust the amp's level controls so I could get the same levels on all the channels. If your AVR has an AutoEQ/AutoSetup wizard with its own mic, you can use it too. I used it to EQ the speakers first, but prefer to finetune with the SPL.

 For my digital source... well, I was using only one at that time (my Squeezebox Duet coax output) and it had its own digital volume control. So this way, I could also attenuate the digital signal from the Duet and was able to use the DAC1 to switch between the AVR and the Duet without killing my speakers.

 The downside is that digital mixers/volume controls are pretty bad (if you have read through all 170+ pages of this thread 
	

	
	
		
		

		
		
	


	




). When I did an A:B comparison of the original CD with the Duet later on, I felt the Duet digital volume control removed some of the transients and dynamics. Turning off the digital volume control on the Duet restored the detail.

 When I didn't want to compromise on the SQ on the Duet... the only way I could fix THAT was to introduce a separate pre-amp that was able to handle Home Theater bypass (ie the AVR pre-out goes direct to the power amp when it was selected, without going through the pre-amp's volume control). A number of new pre-amps have this feature. I wished the DAC1 did as well, but them's the breaks.


----------



## EliasGwinn

Quote:


  Originally Posted by *G-U-E-S-T* /img/forum/go_quote.gif 
_Hi Elias,

 I noticed you mentioned "less" prone - does this imply that the new pot is probably still susceptible (to some lesser degree) however, to this 'crackling' phenomenon, over time?_

 

Sure. All components have a limited lifetime. All resistors, capacitors, opamps. tubes, IC's, will eventually fail.

  Quote:


  Originally Posted by *G-U-E-S-T* /img/forum/go_quote.gif 
_Also please, what specifically is responsible for this crackling sound in a worn volume pot, and what is the particular difference in the new pot that makes it less susceptible to the issue over time? That is, do these volume pots use "brushes" (or some other culprit component) internally, and what is the specific mechanical difference between the old/new pots with respect to such a specific component, that should make the new pot better in this respect?

 Thanks again in advance, as always! 
	

	
	
		
		

		
		
	


	


_

 

I will aggregate this information for you and post it a.s.a.p.

 All the best,
 Elias


----------



## EliasGwinn

Quote:


  Originally Posted by *peter73* /img/forum/go_quote.gif 
_Hi Elias,

 Thank you for taking so much of your time to answer all our questions about your marvellous DAC.

 I have an observation on mine that I would like to share. I have just recently (and totally by chance) found out that the two headphone outs actually sound slightly different. The one on the right (closer to the volume pot) sounds more transparent, opened and easier on the ears - as if I am into the music, while when listening through the other one the sound is flatter, shut in and a bit distant and it also causes slight fatigue after having listened for some time. Please consider that the difference, although clearly distinguishable, is not significant, but yet big enough to make me accept having a headphone jack plugged close enough to the volume control to cause inconvenience when operating it.

 Did you have similar experience with the DAC1's you've heard and what might be the explanation?

 Thank you,
 Peter_

 

Hello Peter,

 I can't say that I've shared your experience, and I've listened closely to 100's of DAC1's. The two headphone jacks are literally connected in parallel, so there shouldn't be a difference. 

 I'll recreate this exact experiment, though. I'll see if I can hear a difference between the two HPA2 outputs.

 I'd recommend trying your experiment again, this time having someone switching the output jacks without you knowing which one is selected. Have them mark your answers, doing it 10 times without telling you whether you were right or wrong. This will eliminate any sub-conscience effects.

 Let me know!

 Thanks,
 Elias


----------



## EliasGwinn

Quote:


  Originally Posted by *doggiehowser* /img/forum/go_quote.gif 
_I suspect that the arguments for/against modding are actually looking at it from different angles.

 I think Elias and Benchmark are right in that in its current state, the Benchmark DACs are at their most neutral/accurate settings. That's what they strove to achieve for a studio environment and that's what they delivered. And modding the DAC may yield a scientifically less "accurate" device.

 But it is the deviation from perfection that may make the sound appear fuller or warmer or fuller, which some users may like. (not unlike what the tube buffer is doing) 
	

	
	
		
		

		
			



_

 

Hello Doggie,

 We strongly discourage anyone from modding the DAC1. It will void the warranty. Also, the modders don't have a schematic for the DAC1 and do not know how the circuit is designed. In other words, they are GUESSING! We've seen many mods to the DAC1, and they are usually severely misguided. 

 By all means, we encourage you to strive to create a system to your liking, even if it is not 'accurate'. However, if you want to make your system fuller or warmer, it cannot be acheived by modding the DAC1. You should consider adding a tube pre-amp or tube amp. 

 All the best!
 -Elias


----------



## doggiehowser

Quote:


  Originally Posted by *EliasGwinn* /img/forum/go_quote.gif 
_Hello Doggie,
 By all means, we encourage you to strive to create a system to your liking, even if it is not 'accurate'. However, if you want to make your system fuller or warmer, it cannot be acheived by modding the DAC1. You should consider adding a tube pre-amp or tube amp._

 

That is what the tube buffer is for 
	

	
	
		
		

		
		
	


	




 it's not a mod to the DAC1 at all. It just takes a normal line out from any device (CDplayer/DAC/tuner/etc) passes it through a tube stage and then outputs over normal analog line out. No opening up of the DAC1 
	

	
	
		
		

		
		
	


	




 Musical Fidelity made a couple of those but they are out of stock.






 Unlike a tube pre-amp which is ALWAYS on, with the tube buffer (being fed by the Benchmark's RCA outputs) and then feeding into my pre-amp's line input, I can switch to a tubed sound when I want to.

 Most of the time, the Benchmark's XLR output goes straight to the pre-amp's XLR input which gives me the clean Benchmark sound signature


----------



## doggiehowser

Quote:


  Originally Posted by *emmodad* /img/forum/go_quote.gif 
_and remind him to do HDCD decoding in that dsp; price the product at a premium to DAC1 but well below alpha; and the resulting "DAC2HDCD" will sell like wildfire.

 wishful thinking ps: give it a native FireWire interface...... 24/192 through a modern John Siau design...... mmmmmmmmmmm. 
	

	
	
		
		

		
		
	


	


_

 

 Quote:


  Originally Posted by *Lord Chaos* /img/forum/go_quote.gif 
_Add me to the list of folks who'd like to have a DAC1 Firewire._

 

Add me to the list but if there's Firewire/iLink, I'd like the ability to work with the Denon/iLink/Firewire from SACD players in DSD mode. I don't need multichannel DSD. Just a good external stereo DSD DAC that works just as well as the current Benchmark for Redbook audio.

 I think it'd be an amazing audio upgrade from most "cheapo" universal DVD/DVDA/SACD players.


----------



## Phileas1

I think I've found most of the answer to my own question (below) elsewhere on this thread.

 I have to use a test tone & voltmeter to get the trim pots to the same level.

 Hi

 I've just been 'playing' with the calibration trimmers with a view to using the fixed xlr output of the DAC1 into my power amp and using the volume control on my SqueezeBox to adjust for louder tracks.

 I'm a bit confused about the trimmers. There seems to be no 'bottoming-out' point. I'm guessing if I want to set them to equal levels (I've already un-set them from their factory settings) I need to turn them both until they're definitely at one end e.g. max attenuation (is this clockwise?) and then count the turns in the opposite direction until I reach the level I want.


----------



## EliasGwinn

Quote:


  Originally Posted by *doggiehowser* /img/forum/go_quote.gif 
_Add me to the list but if there's Firewire/iLink, I'd like the ability to work with the Denon/iLink/Firewire from SACD players in DSD mode. I don't need multichannel DSD. Just a good external stereo DSD DAC that works just as well as the current Benchmark for Redbook audio.

 I think it'd be an amazing audio upgrade from most "cheapo" universal DVD/DVDA/SACD players._

 

As always, thanks for the suggestions!

 All the best,
 Elias


----------



## EliasGwinn

Quote:


  Originally Posted by *Phileas1* /img/forum/go_quote.gif 
_I think I've found most of the answer to my own question (below) elsewhere on this thread.

 I have to use a test tone & voltmeter to get the trim pots to the same level.

 Hi

 I've just been 'playing' with the calibration trimmers with a view to using the fixed xlr output of the DAC1 into my power amp and using the volume control on my SqueezeBox to adjust for louder tracks.

 I'm a bit confused about the trimmers. There seems to be no 'bottoming-out' point. I'm guessing if I want to set them to equal levels (I've already un-set them from their factory settings) I need to turn them both until they're definitely at one end e.g. max attenuation (is this clockwise?) and then count the turns in the opposite direction until I reach the level I want._

 

Let me know if you need any help with this.

 All the best,
 Elias


----------



## Laptopboy

What could be the cause of the S/PDIF port sounding better than the USB port?

 I've tried a number of different USB cables (cheap, but a coupld different Belkins were used).

 The difference in sound is that the USB has all the attributes of a high jitter feed:

 1) Diminished Soundstage
 2) Reduced depth
 3) Reduced low level ambient information
 4) Increased brightness.

 The CD was copied via EAC and ( IIRC )by WINAMP to an IBM T30 LAPTOP. There is no question that when using the USB port against the original CD, both USB versions sound the same, yet not as good as the S/PDIF input.

 I tried an ACER mini, but that was even worse.....

 I'm very confused.....

 BTW: the best sounding USB cable was the cable shipped from Benchmark.


----------



## EliasGwinn

Quote:


  Originally Posted by *Laptopboy* /img/forum/go_quote.gif 
_What could be the cause of the S/PDIF port sounding better than the USB port?

 I've tried a number of different USB cables (cheap, but a coupld different Belkins were used).

 The difference in sound is that the USB has all the attributes of a high jitter feed:

 1) Diminished Soundstage
 2) Reduced depth
 3) Reduced low level ambient information
 4) Increased brightness.

 The CD was copied via EAC and ( IIRC )by WINAMP to an IBM T30 LAPTOP. There is no question that when using the USB port against the original CD, both USB versions sound the same, yet not as good as the S/PDIF input.

 I tried an ACER mini, but that was even worse.....

 I'm very confused.....

 BTW: the best sounding USB cable was the cable shipped from Benchmark._

 


 What operating system and media player are you using?

 Thanks,
 Elias


----------



## gevorg

Hi Elias,

 What do you think about NOS (non-oversampling) DACs? Particularly, ones that use Philips TDA1543 chip in an array of 8 or so, feeded by I2S stream from some re-clocking TCXO (crystal oscillator). A common perception by users who like NOS DACs is that this way the sound is "more musical" and "less fatiguing". Can oversampling DACs like DAC1 sound like this too?


 Thank you!!


----------



## Laptopboy

Thanks for responding Eliase,

 I am using Foobar 2000 on windows XP.

 I believe it is configured correctly -- I'm not home at the moment, so I can't read out the configs....

 From memory, I have selected the Benchmark as the output device. Outputs are buffered for a second or two, the output is direct at 24/96 ( I do have some 24/96 downloads -- they do sound better than redbook.) The windows volume control is at max -- that should bypass kmixer -- there are no DSP's selected from within foobar.

 As far as I can tell, it's a bit perfect path -- unless I am missing something. I've gone through the relevant documentation. I will check later tonight for precise foobar details as well as the control panel/sound settings and re-post. 

 One huge reason for selecting the Benchmark was for it's USB capability. Shouldn't it actually sound BETTER than S/PDIF? Or at least equal? BTW, I've been using the Benchmark since last summer/fall and it's my only source -- it is "broken in". 

 The system itself has only one ground point, that is at the Benchmark. 

 The ground on the preamp has been lifted. 

 The laptop's third prong has not been lifted -- maybe lift this?


----------



## Bach-Fan

Hi Elias,

 I bookmarked this thread long ago and check it every week for updates. I've learned a lot from you here. And the goodwill of the Benchmark brand could not be higher with me due to your honest and understated approach to answering these questions. I have the strong sense that you are shooting straight with us. Rare for a vendor to do that. And it makes me confident that the facts favor buying a Benchmark. So, if you are ever in doubt as to whether the hours you've spent with us are worth it, do not doubt it. They are. And when I have the bucks together, I will be buying a DAC from you, sooner or later.

 Three questions:

 1. That Benchmark has chosen the Dynaudio BM-5A self-powered speakers is a strong recommendation for them. Do you have a recommendation for somewhat larger, "bigger-sounding" speakers that complement the DAC1? (Classical music is my thing. Am concerned BM-5A's may be too small to reproduce orchestral optimally.)

 2. USB 3.0 is just around the corner, with its much greater bandwidth and throughput and other goodies. What do you see as the significance of USB 3.0 for DACs generally and computer audio in the next 3 years or so? 

 3. What advice would you give to a cheapskate like me, who is reluctant to plunk down two grand for a DAC now, fearing that it will be obsolete long before it reaches the end of its useful life (as designed). 

 Thanks.


----------



## Laptopboy

This is and interesting link comparing laptop jitter on USB while running on battery vs switching wall wart:

http://http://www.head-fi.org/forums...rticle-429634/


----------



## HeadLover

I really hope for a new model with USB 2 or even 3.0 and a maybe better chip (32 bit maybe?!)
 And so on


----------



## EliasGwinn

Quote:


  Originally Posted by *Laptopboy* /img/forum/go_quote.gif 
_Thanks for responding Eliase,

 I am using Foobar 2000 on windows XP.

 I believe it is configured correctly -- I'm not home at the moment, so I can't read out the configs....

 From memory, I have selected the Benchmark as the output device. Outputs are buffered for a second or two, the output is direct at 24/96 ( I do have some 24/96 downloads -- they do sound better than redbook.) The windows volume control is at max -- that should bypass kmixer -- there are no DSP's selected from within foobar.

 As far as I can tell, it's a bit perfect path -- unless I am missing something. I've gone through the relevant documentation. I will check later tonight for precise foobar details as well as the control panel/sound settings and re-post. 

 One huge reason for selecting the Benchmark was for it's USB capability. Shouldn't it actually sound BETTER than S/PDIF? Or at least equal? BTW, I've been using the Benchmark since last summer/fall and it's my only source -- it is "broken in". 

 The system itself has only one ground point, that is at the Benchmark. 

 The ground on the preamp has been lifted. 

 The laptop's third prong has not been lifted -- maybe lift this?_

 

Do you have any volume attenuation in Foobar?

 If you have a truly bit-transparent data path, the USB and SPDIF inputs should sound identical. We've done many tests, both listening and measurement, and they've all resulted in a conclusion that there are no differences between the two.

 In fact, I had this test set up at the Head-Fi meet in LA last month. I had the same CD playing from a DVD player (coax AND optical), a Wadia iTransport (coax), and a MacBook (USB). The listeners were able to scroll through the different inputs and try to determine if there were any differences. No one could hear a difference!

 So, I don't know what to tell you. If you're hearing a difference, there must be some incorrect setting in Foobar. Take screen shots of the Foobar preferences and email them to me (elias_AT_benchmarkmedia.com).

 All the best,
 Elias


----------



## Laptopboy

For what it is worth, everyone that has heard my system can hear the difference. Two close friends have experience recording live music; to them the difference is obvious and immediate -- I can't fool them, and I have tried. So, it's not just me....

 I will take the screen shots, but tonight is booked. I can do that tomorrow night at the earliest.

 Thanks Elias.

 EDIT: Also, the sound can change DRAMATICALLY depending on the USB cable being used. It is especially poor with cheap fully sheilded USB 2.0 cables. I tried two different Belkin cables, one with a clear jacket, silver connectors, the other with a black jacket and gold connectors. They were awfully bright, compressed and the soundstage depth completely vanished.


----------



## EliasGwinn

Quote:


  Originally Posted by *gevorg* /img/forum/go_quote.gif 
_Hi Elias,

 What do you think about NOS (non-oversampling) DACs? Particularly, ones that use Philips TDA1543 chip in an array of 8 or so, feeded by I2S stream from some re-clocking TCXO (crystal oscillator)._

 

A non-oversampling DAC is basically a DAC with analog anti-aliasing filters. These analog filters suffer from high-frequency distortion and phase distortion. They aren't steep enough to block frequencies higher then Nyquist and leave the pass-band in tact. If the steepness is increased, major phase and distortion issues come about.

 Also, since these filters are built using analog components on a wafer, they are severely temperature unstable. They will drift depending on the operating scenario.

  Quote:


  Originally Posted by *gevorg* /img/forum/go_quote.gif 
_A common perception by users who like NOS DACs is that this way the sound is "more musical" and "less fatiguing". Can oversampling DACs like DAC1 sound like this too?

 Thank you!!_

 

I can't comment on the subjective impressions of these people, but I find the DAC1 far from fatiguing. In fact, I listen to it all day, everyday. I listen at work here at Benchmark, then I go home and work in my studio while listening through it.

 However, subjective impressions are impossible to quantify, so you'll have to listen for yourself to determine the answers to your questions.

 All the best,
 Elias


----------



## G-U-E-S-T

Quote:


  Originally Posted by *G-U-E-S-T* /img/forum/go_quote.gif 
_...please, what specifically is responsible for this crackling sound in a worn volume pot, and what is the particular difference in the new pot that makes it less susceptible to the issue over time? That is, do these volume pots use "brushes" (or some other culprit component) internally, and what is the specific mechanical difference between the old/new pots with respect to such a specific component, that should make the new pot better in this respect?

 Thanks again in advance, as always! 
	

	
	
		
		

		
			



_

 


  Quote:


  Originally Posted by *EliasGwinn* /img/forum/go_quote.gif 
_...I will aggregate this information for you and post it a.s.a.p. ..._

 

Hi Elias, just following up with you regarding the above. Hope you are doing well! 
	

	
	
		
		

		
		
	


	




 P.S. I concur with Elias, the DAC1 is definitely not fatiguing in any way.


----------



## georgiholebrook

I am a newbie to this forum site. 
 I hope you guys help me always if I fall in trouble. And I'm really happy to join with you. 
 Pls share your valuable tips and tricks with me.


----------



## minidiscs

I been running my Benchmark Dac1 Pre since a week after Can Jam ended. I think its the best piece of gear I have bought. I just need to find some better recordings to put this thing through. It brought back life to my HD600's and its nice to finally have a volume knob on my Dynaudio BM5A + BM9S setup. 

 In comparison to my Moodlab Dice this thing whoops its butt on the mid and high separation in the vocals. I been sampling 'Endless Love' duet by Mariah Carey and Luther Vandross and you can tell that there is a major improvement in the detail inflections of the voices. 

 Only thing I noticed is that the unit is kinda warm.


----------



## ert

Elias,

 Is there any problem with connecting both the RCA and balanced outputs to two different amplifiers simultaneously?


----------



## Laptopboy

I don't know how to do a screen capture, so:

 Playback settings:

 Volume Control on MAX
 Replay gain, 
 source mode none
 Processing none

 Preamp, with RG info +/-0db(89 db), without +/- 0db. Both sliders are in the mid position, but both source and processing of replay gain is set to none anyway.

 DSP manager has NO active DSPs configured.

 Output Device is set to Benchmark
 output format is 24 bits. Dithering is NOT checked.

 Tools/convertor FLAC settings are <N/A> for avg bit rate and level 5 for settings. Preferred bit depth is 16, but the Keep lossless source at original bit depth is checked. Dither is set to never. (I don't use this convertor anyway.)

 I hope that's enough info. 

 The windows volume control is at max, all advanced controls are disabled.


----------



## HeadLover

Just press the "print screen" on your pc keyboard, than post it to a paint program, and save it and upload it.


----------



## Laptopboy

I think I may have fixed the problem

 There was some sort of program installed called "SoundMAX". After I uninstalled it, the sound quality seemed to improve. Hopefully it's not a placebo effect. 

 Initial impressions are that he bass response is not quite as deep, but the difference is very subtle. In some ways, the USB sounds better: Bigger, and better inner detail. The CD may have a slight edge in ambient information, but it's hard to say. 

 Last night was extremely humid, so I can't say if the humidity levelled the playing field or not. I'll know more after a few days.


----------



## EliasGwinn

Quote:


  Originally Posted by *G-U-E-S-T* /img/forum/go_quote.gif 
_Quote:
 Originally Posted by G-U-E-S-T 
 ...please, what specifically is responsible for this crackling sound in a worn volume pot, and what is the particular difference in the new pot that makes it less susceptible to the issue over time? That is, do these volume pots use "brushes" (or some other culprit component) internally, and what is the specific mechanical difference between the old/new pots with respect to such a specific component, that should make the new pot better in this respect?

 Hi Elias, just following up with you regarding the above. Hope you are doing well! 
	

	
	
		
		

		
		
	


	




 Thanks again in advance, as always! 


 P.S. I concur with Elias, the DAC1 is definitely not fatiguing in any way._

 

Hello G-U-E-S-T,

 I'm sorry it has taken so long to respond to you. 

 The pots in the DAC1/USB/PRE are great pots, but the new Alps pots in the DAC1 HDR are simply better. The pots in the DAC1/USB/PRE became noisy for two reasons. The first wasn't the fault of the pot; it was a result of a subtle DC offset in the gain circuit. That was corrected with a circuit revision in 2004. The other reason for the crackle-y volume control was because of leaky pot shafts. Lubrication would leak onto the resistive element. This didn't affect the tone/performance of the DAC1, but it did cause the noise when the volume-control is adjusted. Otherwise, it is a benign defect. We have replaced most of these pots already.

 The new Alps pot in the DAC1 HDR has a larger resistive element and wiper element, and a better build-construction, generally speaking. They are more expensive, but part of what you're paying for is increased quality control. They are solid, well-built pots. The larger elements decrease variances, making for more consistent performance. Also, the larger elements are less susceptible to heat (expansion and contraction), also contributing to more consistent performance. 

 I hope that answers your question. Please let me know if you'd like me to clarify anything...

 All the best,
 Elias


----------



## EliasGwinn

Quote:


  Originally Posted by *minidiscs* /img/forum/go_quote.gif 
_I been running my Benchmark Dac1 Pre since a week after Can Jam ended. I think its the best piece of gear I have bought. I just need to find some better recordings to put this thing through. It brought back life to my HD600's and its nice to finally have a volume knob on my Dynaudio BM5A + BM9S setup. 

 In comparison to my Moodlab Dice this thing whoops its butt on the mid and high separation in the vocals. I been sampling 'Endless Love' duet by Mariah Carey and Luther Vandross and you can tell that there is a major improvement in the detail inflections of the voices. 

 Only thing I noticed is that the unit is kinda warm._

 

Hey minidiscs,

 I'm glad to hear you're enjoying the DAC1 PRE!

 The DAC1 PRE gets warm because it's circuits are low-impedance. This acheives optimal performance, but draws more current then high-impedance circuitry. 

 All the best!
 -Elias


----------



## EliasGwinn

Quote:


  Originally Posted by *ert* /img/forum/go_quote.gif 
_Elias,

 Is there any problem with connecting both the RCA and balanced outputs to two different amplifiers simultaneously?_

 

No problems at all! That is, it won't hurt the DAC1 or comprimise its performance. However, you should understand that the output levels will be different.

 All the best,
 Elias


----------



## EliasGwinn

Quote:


  Originally Posted by *Laptopboy* /img/forum/go_quote.gif 
_I think I may have fixed the problem

 There was some sort of program installed called "SoundMAX". After I uninstalled it, the sound quality seemed to improve. Hopefully it's not a placebo effect. 

 Initial impressions are that he bass response is not quite as deep, but the difference is very subtle. In some ways, the USB sounds better: Bigger, and better inner detail. The CD may have a slight edge in ambient information, but it's hard to say. 

 Last night was extremely humid, so I can't say if the humidity levelled the playing field or not. I'll know more after a few days._

 

This doesn't surprise me... Equipment companies will throw their DSP plug-in's in-line without your awareness to make their speakers sound better. Unfortunately, these DSP plug-in's are the last thing you'd want for a true high-resolution system.

 Keep in touch...let me know if everything continues to work well.

 All the best,
 Elias


----------



## Roseval

A simple check is to open the properties of the playback device in the sound panel.
 Check the 'Enhancement' tab, disabling all DSP might be an enhancement.
 Also check the 'Configure' button and see if the speakers are set to full range.


----------



## ert

Quote:


  Originally Posted by *EliasGwinn* /img/forum/go_quote.gif 
_No problems at all! That is, it won't hurt the DAC1 or comprimise its performance. However, you should understand that the output levels will be different.

 All the best,
 Elias_

 

That's fantastic. Thanks.


----------



## Xxtest83

Hello Elias,

 the Benchmark DAC-1 HDR gets very warm even when shut down. Can i prolong its life by always disconnecting it from the power (with a switch on the power socket)?

 Greetings,
 Stefan


----------



## nomar

Hi,
 Just got Benchmark DAC 1 HDR literally hours ago but my happiness was marred somewhat by missing power cord ! . I assume it should be supplied together since on pg 46 manual shows weight with power cord inc. I asked the distributor but his reply that Benchmark doesn't provide it, can Elias confirm this ?
 Anyway few pictures.






the IR receiver does not have any light right ?





 with ATH W1000 + Promitheus Audio pre-amp ( won't be using this anymore after a while.




 My Harbeth C7 ES3 driven amazingly by 10W Trends Audio 10.1(as power amp ) but I've got Hypex Ucd180 Hxr on order and once it arrives both Promitheus and Trends will be on sale.


----------



## EliasGwinn

Quote:


  Originally Posted by *Roseval* /img/forum/go_quote.gif 
_A simple check is to open the properties of the playback device in the sound panel.
 Check the 'Enhancement' tab, disabling all DSP might be an enhancement.
 Also check the 'Configure' button and see if the speakers are set to full range._

 

Thanks Roseval!

 All the best,
 -elias


----------



## EliasGwinn

Quote:


  Originally Posted by *Xxtest83* /img/forum/go_quote.gif 
_Hello Elias,

 the Benchmark DAC-1 HDR gets very warm even when shut down. Can i prolong its life by always disconnecting it from the power (with a switch on the power socket)?

 Greetings,
 Stefan_

 

Hello Stefan,

 The DAC1 HDR will get about as warm as the outside of a mug of hot coffee or tea (or cider, sake, etc). If it is getting significantly hotter then that, it may need to be serviced.

 The life of the DAC1 HDR will not vary significantly based on power periods. In other words, feel free to switch it off or leave it on. It won't change the life-expectancy much either way.

 All the best,
 Elias


----------



## EliasGwinn

Quote:


  Originally Posted by *nomar* /img/forum/go_quote.gif 
_Hi,
 Just got Benchmark DAC 1 HDR literally hours ago but my happiness was marred somewhat by missing power cord ! . I assume it should be supplied together since on pg 46 manual shows weight with power cord inc. I asked the distributor but his reply that Benchmark doesn't provide it, can Elias confirm this ?
 Anyway few pictures.





the IR receiver does not have any light right ?





 with ATH W1000 + Promitheus Audio pre-amp ( won't be using this anymore after a while.




 My Harbeth C7 ES3 driven amazingly by 10W Trends Audio 10.1(as power amp ) but I've got Hypex Ucd180 Hxr on order and once it arrives both Promitheus and Trends will be on sale._

 


 Hello Nomar,

 Thank you for contacting me about this. All of our products come with power cords. I'm very sorry yours did not. We will send you one immediately. 

 Please send me a PM (private message) with the name of your dealer and/or distributor, as well as the serial number of your product.

 Thanks again, Nomar.

 All the best,
 Elias


----------



## EliasGwinn

Quote:


  Originally Posted by *nomar* /img/forum/go_quote.gif 
_Hi,
 Just got Benchmark DAC 1 HDR literally hours ago but my happiness was marred somewhat by missing power cord ! . I assume it should be supplied together since on pg 46 manual shows weight with power cord inc. I asked the distributor but his reply that Benchmark doesn't provide it, can Elias confirm this ?_

 

I'm sorry I spoke too soon. We do not ship 220V versions of our product with power cables. This is because there are far too many different types of plugs and sockets, and we can't cover them all.

 Some of our distributors and international dealers supply cables with the products, but it is left to their discretion. Some choose not to supply them because they have found that their customers rarely use them, choosing to buy after-market power cables instead.

 If this is an issue for anyone, let me know and we can try to have a cable provided. This is not an official offer for cables, but something that I can try to do in special circumstances.

 All the best,
 Elias


----------



## Bach-Fan

My question from my June 17 post seem to have been posted in the midst of an ongoing conversation, and been lost in the shuffle. So I am reposting.

 1. That Benchmark has chosen the Dynaudio BM-5A self-powered speakers is a strong recommendation for them. Do you have a recommendation for somewhat larger, "bigger-sounding" speakers that complement the DAC1? (Classical music is my thing. I am concerned BM-5A's may be too small to reproduce big symponic music.)

 2. USB 3.0 is just around the corner, with its much greater bandwidth and throughput and other goodies. What do you see as the significance of USB 3.0 for DACs generally and Benchmark in particular? 

 3. Same question as above for Windows 7, which will go on sale in a couple of months. Any relevant changes in how the new OS will handle audio?

 Thanks for all the reliable and honest info you provide. I've followed this thread closely for over a year. You have sold me on Benchmark as perhaps the finest DAC maker on the planet.


----------



## EliasGwinn

Quote:


  Originally Posted by *Bach-Fan* /img/forum/go_quote.gif 
_My question from my June 17 post seem to have been posted in the midst of an ongoing conversation, and been lost in the shuffle. So I am reposting._

 

I apologize for not addressing your questions earlier. I wish I had seen this post, because these are great questions...

  Quote:


  Originally Posted by *Bach-Fan* /img/forum/go_quote.gif 
_1. That Benchmark has chosen the Dynaudio BM-5A self-powered speakers is a strong recommendation for them. Do you have a recommendation for somewhat larger, "bigger-sounding" speakers that complement the DAC1? (Classical music is my thing. I am concerned BM-5A's may be too small to reproduce big symponic music.)_

 

I would suggest Klein and Hummel active 3-way speakers, specifically the O400's, w/ an active O800 subwoofer. These aren't cheap, but they are incredibly faithful speakers. They won't add color, etc, so if you're looking for tube-type warmth, these won't do it. But you won't be disappointed when using these to reproduce a symphony orchestra.

  Quote:


  Originally Posted by *Bach-Fan* /img/forum/go_quote.gif 
_2. USB 3.0 is just around the corner, with its much greater bandwidth and throughput and other goodies. What do you see as the significance of USB 3.0 for DACs generally and Benchmark in particular?_

 

The real question is whether chip manufacturers are going to make a USB 3.0 chip that works with native USB audio drivers. We are convinced that native is the way to go because they work flawlessly since the drivers are built into the operating system. When you implement a custom driver, you risk cross-platform compatibility issues, as well as cross-driver issues (i.e, one driver causing problems with the other, or arguing about resources).

  Quote:


  Originally Posted by *Bach-Fan* /img/forum/go_quote.gif 
_3. Same question as above for Windows 7, which will go on sale in a couple of months. Any relevant changes in how the new OS will handle audio?_

 

I can't answer this question, as I don't have much inside info on the audio architecture of Windows 7. My guess is that it will be very similar to Vista's audio architecture.

  Quote:


  Originally Posted by *Bach-Fan* /img/forum/go_quote.gif 
_Thanks for all the reliable and honest info you provide. I've followed this thread closely for over a year. You have sold me on Benchmark as perhaps the finest DAC maker on the planet._

 

Well thank you for making this thread alive! It wouldn't be any fun if it was just me 
	

	
	
		
		

		
		
	


	




 (in fact, it wouldn't exist if it wasn't for your questions and contributions).

 All the best,
 Elias


----------



## Laptopboy

Quote:


  Originally Posted by *EliasGwinn* /img/forum/go_quote.gif 
_This doesn't surprise me... Equipment companies will throw their DSP plug-in's in-line without your awareness to make their speakers sound better. Unfortunately, these DSP plug-in's are the last thing you'd want for a true high-resolution system.

 Keep in touch...let me know if everything continues to work well.

 All the best,
 Elias_

 

Well, since the discovery of bogus drivers, I upgraded the playback to the latest foobar2k and ASIO4ALL V2.

 The difference is stunning. The sound quality via the USB port is far better than the CD transport via S/PDIF. 

 Thanks for listening to my griping.....it made me dig alot deeper into the configuration of the laptop.

 The sound quality at this point is simply excellent. I never thought that redbook could ever sound this good. 24/96 sounds wonderful.

 FYI: Laptop -> BM USB -> calibrated RCA outputs -> Melos 110B Linestage -> Phase Linear 400 -> Watson Lab Model 7. Foundation Research LC-1 on the Melos. DH labs AC Power on BM. Kimber Hero throughout. Yes, it's old stuff, but the results are what matters: Incredibly dynamic sound with excellent soundstaging/imaging and tonality.


----------



## Bach-Fan

The review below is mostly favorable, but there's a paragraph criticizing the preamp section that strikes me as very odd. Here is what the review says:
  Quote:


 ... Listening to the HDR through its analog inputs was a little different story. I heard a difference in sound quality when i listened to the same tracks via the USB input compared to the analog inputs. Testing this further I connected my DAC1 PRE to my Mac via USB and connected the PRE's analog outputs to the DAC1 HDR's analog input via single ended RCA interconnects. Since the HDR and the PRE sound identical via the USB input and analog output this configuration enabled me to single out the analog input of the HDR for listening sessions. Right away after clicking play through iTunes I could hear the difference in sonic quality between the USB input and the analog input of the HDR. The analog input constricted the sound stage and dynamics a bit and made the highs a little thin sounding. The bottom end had less definition through the analog inputs as opposed to very tightly defined bass via the USB input. ... Needless to say I was less impressed with the sound of the HDR trough the analog inputs, but that does not mean I was unimpressed overall. In fact nothing could be further from the truth. ... 
 

I'm no audio engineer, but this seems like an incompetent test. Instead of feeding the DAC1 HDR a line-level source, he fed it a source that was already amplified - and therefore beyond what the preamp circuitry was designed to handle. Elias am I right about this? 

 If, in fact, the reviewer did perform this "test" as described - and no editor called him on it before it was published - it seems to be a flashing red-light to treat this blog with a healthy dose of skepticism overall. 

 Here's the link to the review. It is more or less favorable overall:

Benchmark DAC1 HDR Review | Computer Audiophile


----------



## Fred Flintstone

Quote:


  Originally Posted by *EliasGwinn* /img/forum/go_quote.gif 
_No problems at all! That is, it won't hurt the DAC1 or comprimise its performance. 
 All the best,
 Elias_

 

I noticed that PS audio doesn't recommend connecting the balanced XLR and unbalanced RCA audio outputs at the same time for their Digilink III so am I to assume that Benchmark is doing something different in their design to allow this?


----------



## BitPerfect

The Audio Critic, long an advocate of measurement-based audio reviews, has just reviewed the HDR. I don't think you could ask for a more favorable review. 
	

	
	
		
		

		
		
	


	




http://theaudiocritic.com/plog/index...Id=40&blogId=1


----------



## bralk

Quote:


  Originally Posted by *Bach-Fan* /img/forum/go_quote.gif 
_1. That Benchmark has chosen the Dynaudio BM-5A self-powered speakers is a strong recommendation for them. Do you have a recommendation for somewhat larger, "bigger-sounding" speakers that complement the DAC1? (Classical music is my thing. I am concerned BM-5A's may be too small to reproduce big symponic music.)_

 

For large scale reproduction you need large speakers. I can recommend 
 ATC SCM 100 ASL or perhaps the SCM 50 ASL depending on your listening environment. 

 cheers

 Tom


----------



## Runkby

I have a question. I have been using the benchmark dac1 hdr for over a month now and I currently own two pairs of headphones. I have the sennheiser hd 650 with an equinox cable as well as stock denon ah-d7000 headphones. For the source I often use my computer with the USB, which is great, however I was messing around today with the hdr connected to my cd player. I am using a krell kav300cd and I tried using both rca cables to connect them as well as the coax optical out on the cd player with an apogee cable into one of the digital imputs on the hdr. If i have the cd player on, but paused, if i raise the volume to max I hear a static noise when using the rca connections, but when I connect the cd player with the ditital coax connection the static seems to go away. What could be the cause of this? I know my cd player had a problem in the past unlreated to this, but I just recieved it back from krell and they claim it's in top conditon (they just replaced the laser reader and some other thing). I also, by the way, tried it with two different rca cables and with both cables the noise was very prominent at a high volume level, whereas it was not with the digital input. Any ideas as to why?


----------



## Runkby

I'll throw another question out there while I'm here... As I previously stated I have both the Sennheiser HD 650 as well as the Denon AH-D7000 headphones. When I use the hdr with the sennheisers it seems like there may not be enough power to fully drive them? I understand that the 650's are hard to drive in addition to being 300ohms but is that just how the amp is or is there something I can do to adjust the settings to get more output on the volume. Also it would be nice if in the future you offered both balanced inputs as well as outputs on the back; balanced headphone jacks would also be nice


----------



## all300b

Dear Elias,

 I am yet another new Benchmark DAC1-pre owner as of last night. I am currently running the DAC into an outboard dedicated pre-amp and then to a power amp that happens to have a lot of gain. I have needed to keep the output on my source gear low in the past. As I expected, the standard -20db factory setting on the Benchmark is not quite enough. Which do you recommend would sound better:

 - the -30 db settings run in fixed output 
 - the variable output with the pot turned just a few clicks

 If you think they sound very close the variable output is more convenient as a general volume control because the DAC is easy to reach right on top of my rack while the pre-amp is inside.

 Thanks very much. Sincerely,

 Bryan


----------



## Bach-Fan

I have read of people using the DAC1 in a home theatre setup. Given that the DAC1 is two-channel and most home theatres are 5.1 channels (or more), how in the world could this possibly work? And how would one go about setting this up using the DAC1?


----------



## Quaddy

i use the dac1 pre in my 'home theatre', well its a small bedroom rig, where my only speakers are audioengine stereo (2.0) A5's

 i dont require 5.1 in my room, although the main living room has 5.1 speakers in.

 i am feeding my freesat box, my Archos 7, my samsung LCD and blu ray player all to my dac1 via stereo analogue inputs. as all of those devices are HDMI, i simply output one analogue RCA connection from the TV which gets its feed from all the other devices, into my analogue in on my DAC1 which gets fed out to my headphones and speakers.

 also feeding it my music devices: rega jupiter, SPL phonitor, off ramp turbo 2, sony D-VE7000S

 no idea how people that need 5.1 are utilizing it, m,y guess is they are using it as an input/output 'switch' for multiple devices

 but if you are a fan of stereo, then its great


----------



## doggiehowser

Quote:


  Originally Posted by *Bach-Fan* /img/forum/go_quote.gif 
_I have read of people using the DAC1 in a home theatre setup. Given that the DAC1 is two-channel and most home theatres are 5.1 channels (or more), how in the world could this possibly work? And how would one go about setting this up using the DAC1?_

 

I use a hifi pre-amp with a HT bypass input.

 For hifi,

 Duet/Wadia iTransport feeds DAC1
 DAC1 feeds Bel Canto Pre3 pre amp
 Pre3 feeds Bel Canto REF1000 power amps
 REF1000 drives the Thiel CS2.4 SE speakers

 For HT
 Oppo/S550/XA2 feeds Onkyo AVR
 Onkyo AVR drives center/side/surround
 AVR front pre-out feeds the Pre3 HT bypass input (Pre3's pre-amp functions disabled so in effect goes direct to REF1000s and CS2.4)


----------



## Xxtest83

I have a Lehmann BCL Pro (with XLR inputs) connected to the XLR outputs from my DAC1 HDR, but since i plan to buy active speakers soon (probably K+H o300), the XLR outputs have to get free.
 Do you think it might be possible to hear a difference when i feed the Lehmann amp through a Cinch-XLR cable from the Cinch-output compared to the XLR-XLR connection?


----------



## Roseval

A RCA to XLR plug won’t break the bank so you can try this at marginal cost.

 A lot of components are single ended internally. To make them balanced you need a balun to create balanced output. Maybe removing this component from the chain by going RCA yields an audible effect.


----------



## Xxtest83

Would it hurt the amp (or even the DAC1) if i connected it from RCA to XLR? The amp has not a balanced out but a single ended. My guess is that this version has XLR inputs just because it is more common in studios (pro version).

 What about a XLR Splitter? It is galvanic splitted between in- and outputs, whatever that means.


----------



## Roseval

<<The amp has not a balanced out >>??
 If you mean DAC 1 RCA out to amp XLR in, this in general works good.
 As it is a DAC 1 you might use the volume control starting at zero for safety.


----------



## EliasGwinn

Quote:


  Originally Posted by *Bach-Fan* /img/forum/go_quote.gif 
_I'm no audio engineer, but this seems like an incompetent test. Instead of feeding the DAC1 HDR a line-level source, he fed it a source that was already amplified - and therefore beyond what the preamp circuitry was designed to handle. Elias am I right about this? _

 

Its hard to say without knowing his signal levels. Calibrating signal levels is an essential part of assessing the performance of a piece of audio gear. He doesn't mention his process for setting levels, so one can only speculate.

 All the best,
 Elias


----------



## EliasGwinn

Quote:


  Originally Posted by *Fred Flintstone* /img/forum/go_quote.gif 
_I noticed that PS audio doesn't recommend connecting the balanced XLR and unbalanced RCA audio outputs at the same time for their Digilink III so am I to assume that Benchmark is doing something different in their design to allow this?_

 

Hmmm...How much do they charge for that thing?!?

 I don't know enough about their product to say for sure, but it's surprising that they wouldn't design it with autonomous outputs.

 Atb,
 e


----------



## EliasGwinn

Quote:


  Originally Posted by *BitPerfect* /img/forum/go_quote.gif 
_The Audio Critic, long an advocate of measurement-based audio reviews, has just reviewed the HDR. I don't think you could ask for a more favorable review. 
	

	
	
		
		

		
		
	


	




http://theaudiocritic.com/plog/index...Id=40&blogId=1_

 

Now this is an example of a knowledgeable reviewer...and NOT just because he likes the HDR 
	

	
	
		
		

		
		
	


	




. Peter Aczel is a no-B.S. 'engineer' in the truest sense of the word...someone who's not afraid to dig into the nuts-and-bolts to understand the gear under review. 

 Atb,
 e


----------



## EliasGwinn

Quote:


  Originally Posted by *bralk* /img/forum/go_quote.gif 
_For large scale reproduction you need large speakers. I can recommend 
 ATC SCM 100 ASL or perhaps the SCM 50 ASL depending on your listening environment. 

 cheers

 Tom_

 

ATC makes incredible speakers, but be prepared to write five numbers on your check! (worth it if you've got it)

 Atb
 -e


----------



## EliasGwinn

Quote:


  Originally Posted by *Runkby* /img/forum/go_quote.gif 
_If i have the cd player on, but paused, if i raise the volume to max I hear a static noise when using the rca connections, but when I connect the cd player with the ditital coax connection the static seems to go away. What could be the cause of this?_

 

Noisy analog electronics in the CD player. If its only noise with the analog connection, its a tell-tale sign that the analog outputs of the CD player are noisy.

 All the best,
 Elias


----------



## EliasGwinn

Quote:


  Originally Posted by *Runkby* /img/forum/go_quote.gif 
_I have both the Sennheiser HD 650 as well as the Denon AH-D7000 headphones. When I use the hdr with the sennheisers it seems like there may not be enough power to fully drive them? I understand that the 650's are hard to drive in addition to being 300ohms but is that just how the amp is or is there something I can do to adjust the settings to get more output on the volume._

 

The headphone amplifier in the DAC1 HDR has a customizable gain-range. The factory setting is -10 dB (+14 dBu max output). In other words, you have 10 dB of additional gain available. 

 The manual has instructions for changing this, and I can help you as much as you need.

 All the best,
 Elias


----------



## EliasGwinn

Quote:


  Originally Posted by *all300b* /img/forum/go_quote.gif 
_Dear Elias,

 I am yet another new Benchmark DAC1-pre owner as of last night. I am currently running the DAC into an outboard dedicated pre-amp and then to a power amp that happens to have a lot of gain. I have needed to keep the output on my source gear low in the past. As I expected, the standard -20db factory setting on the Benchmark is not quite enough. Which do you recommend would sound better:

 - the -30 db settings run in fixed output 
 - the variable output with the pot turned just a few clicks

 If you think they sound very close the variable output is more convenient as a general volume control because the DAC is easy to reach right on top of my rack while the pre-amp is inside.

 Thanks very much. Sincerely,

 Bryan_

 

Hello Bryan,

 You will get the best performance from the variable volume control when it is positioned between 11:00 o'clock and 3:00 o'clock. If the output is too high in the current configuration to operate at those positions, I strongly recommend moving the output attenuators of the DAC1 PRE to 30 dB.

 Let me know if I can do any more to assist you.

 All the best,
 Elias


----------



## EliasGwinn

Quote:


  Originally Posted by *Bach-Fan* /img/forum/go_quote.gif 
_I have read of people using the DAC1 in a home theatre setup. Given that the DAC1 is two-channel and most home theatres are 5.1 channels (or more), how in the world could this possibly work? And how would one go about setting this up using the DAC1?_

 

For a 5.1 DAC1 setup, you would need a source that sends three seperate digital signals to three seperate DAC1's (e.g., one for L/R, one for RL/RR, and one for C/Sub). 

 Or, as someone else pointed out, you can use another A/V pre-pro for your 5.1, then bypass it's internal D/A for 2-channel (stereo) and pass the digital signal to the DAC1 HDR. When in 5.1 mode, the front L/R analog outputs will go into the analog inputs of the DAC1 HDR, which is driving the amplifiers. OR, the output of the DAC1 can feed analog inputs of the pre-pro. 

 We have customers who have all of these types of setups.

 All the best,
 Elias


----------



## EliasGwinn

Quote:


  Originally Posted by *Xxtest83* /img/forum/go_quote.gif 
_I have a Lehmann BCL Pro (with XLR inputs) connected to the XLR outputs from my DAC1 HDR, but since i plan to buy active speakers soon (probably K+H o300), the XLR outputs have to get free.
 Do you think it might be possible to hear a difference when i feed the Lehmann amp through a Cinch-XLR cable from the Cinch-output compared to the XLR-XLR connection?_

 

I tried to look-up the specs on the Lehmann BCL Pro headphone amplifier, but I couldn't find any. Can you provide a link?

 Thanks,
 -Elias


----------



## EliasGwinn

Quote:


  Originally Posted by *Xxtest83* /img/forum/go_quote.gif 
_Would it hurt the amp (or even the DAC1) if i connected it from RCA to XLR? The amp has not a balanced out but a single ended. My guess is that this version has XLR inputs just because it is more common in studios (pro version).

 What about a XLR Splitter? It is galvanic splitted between in- and outputs, whatever that means._

 

It will not hurt the DAC1 to use the RCA's to drive the BCL while simultaneously using the XLR's to drive the K+H's. Also, it will not hurt the DAC1 to use XLR splitters. However, if the input impedance of the BCL is too low, it may cause distortion w/ XLR splitters.

 Do you know the input impedance of the BCL pro?

 Thanks,
 Elias


----------



## EliasGwinn

Quote:


  Originally Posted by *Roseval* /img/forum/go_quote.gif 
_<<The amp has not a balanced out >>??
 If you mean DAC 1 RCA out to amp XLR in, this in general works good.
 As it is a DAC 1 you might use the volume control starting at zero for safety._

 

Both sentences are true.

 Thanks,
 Elias


----------



## Xxtest83

Thanks for your answer Elias.

 The specs of the Lehmann Audio Black Cube Linear Pro are:

 input impedance: 47 k Ohm
 maximum gain: switchable 0 + 10 + 18 + 20 dB
 frequency response: 10 - 35.000 Hz
 S/N ratio: > 95 dB at gain 0 dB
 output power: 300 Ohm/200 mW, 60 Ohm/400 mW
 output impedance: 5 Ohm
 power consumption: ca. 10 W
 connections: 2x XLR symmetric
 channel seperation: > 70dB/10kHz

Lehmannaudio :: HighEnd vom Toningenieur :: Pro-Audio

 My main concern is if a difference to the worse in audioquality is audible when i change to RCA->XLR connection. I guess i will have to buy some cables and try it out.

 Regards,
 Stefan


----------



## EliasGwinn

Quote:


  Originally Posted by *Xxtest83* /img/forum/go_quote.gif 
_My main concern is if a difference to the worse in audioquality is audible when i change to RCA->XLR connection. I guess i will have to buy some cables and try it out.

 Regards,
 Stefan_

 

Stefan,

 You will have no problem driving both of these from the XLR output using splitters. But if you want to try the RCA solution, you should use a cable that has pin 3 of the XLR tied to the shield of the RCA (like this one).

 Let me know how it works out.

 All the best,
 Elias


----------



## Xxtest83

-deleted-


----------



## Xxtest83

I found out that my HDR Version has an irregularity in left-right channel balance with its volume pot. The middle of the stage gets shifted in the left area when the pot is at 9 to 11 o'clock, and slight to the right at around 3 o'clock.

 The problem does not occur when i avoid the volume pot by setting the XLR-out to calibrated. It is a problem with the volume pot, and since this is the main part which got worked on in the HDR Version, i wanted to ask if this might be a standard problem of this verion, or if i just got a defective unit.


----------



## EliasGwinn

Quote:


  Originally Posted by *Xxtest83* /img/forum/go_quote.gif 
_I found out that my HDR Version has an irregularity in left-right channel balance with its volume pot. The middle of the stage gets shifted in the left area when the pot is at 9 to 11 o'clock, and slight to the right at around 3 o'clock.

 The problem does not occur when i avoid the volume pot by setting the XLR-out to calibrated. It is a problem with the volume pot, and since this is the main part which got worked on in the HDR Version, i wanted to ask if this might be a standard problem of this verion, or if i just got a defective unit._

 

The new volume pot in the DAC1 HDR is better then the pot in the other DAC1 products. However, subtle channel differences will always occur when using a stereo pot. 

 Stereo pots are ideal for maintaining dynamic range, low distortion, and linear frequency response. Slight channel inbalances are the trade-off (but definately worth it compared to the loss of dynamic range with digital volume controls, distortion with analog volume IC's, or non-linear response of passive attenuators).

 If there are gross errors between the channels, perhaps you have received a defective pot. How bad are the differences?

 Thanks,
 Elias


----------



## Xxtest83

For me the effect is significant. But since you asked i just tried to quantify the effect by reducing the left or right channel volume at the PC.
 When the Volume pot is at 10 o'clock i have to reduce the left channel by 2 percent and at 3 o'clock to increase left channel by 1 percent to get the Stage where it is with other Sources (where it should be). Those 2 Percent sound significant to me, especially because the balance-change happens through a small degree (9 to 11 o'clock).

 I wonder if it is only my unit having this problem, or only me having a problem with this effect.


----------



## EliasGwinn

I'll contact you via PM to arrange a solution...

 -Elias


----------



## little-endian

Hello Elias,

 as I recently noted, the "interior view" of the DAC1 picture gallery have been removed.

 One of them was available under:

http://www.benchmarkmedia.com/dac1/p...-rm_inside.jpg

 Any special reason for that?


----------



## EliasGwinn

Good question...?? They must have been forgotten when the new website was built. 

 I've sent an email to our webmaster...we should see them by this afternoon.







 ATB,
 E


----------



## hh83917

I was just reading about some DAC1 HDR reviews few minutes ago and it mentioned that it uses National Semiconductor LM4562 op-amp. Then I remembered I've saw that National Semiconductor somewhere... and it turns out that the apartment I live in now was right across the National Semiconductor manufacturing plant. I thought that was pretty funny.

 It also just happens that my younger goes to Syracuse University, where Benchmark is located. And I just happened to be a DAC1 HDR owner. Small world indeed. 

 P.S: Been using the DAC1 HDR for few months now and still enjoying it, thanks Benchmark.


----------



## denonfan

I just purchased a Benchmark DAC1 USB. However, I am having major problems with the USB connection. I am running Vista Ultimate, 32bit, computer is a Thinkpad, about 2 years old, music player is foobar2000 v0.9.6.1. I am playing APE files. The Benchmark is plugged into a dedicated USB port on the laptop, no USB hub involved. I have restarted the laptop, plugged the laptop into a differet USB port on the laptop, all to no avail.

 Whenever foobar is running by itself, the USB connection works fine. No problems.

 However, whenever I am browsing the web using either Firefox or IE, the Benchmark seems to lose the USB connection. Those 2 blue LED lights start flashing, I get some nasty clicks, then no sound. Sometimes the sound comes back, sometimes the sound doesn't. And the problem doesn't go away until I close either the Firefox or the IE, and restart foobar2000 and just let foobar2000 play on its own.

 The buffer length in foobar2000 is set to max, 8000ms, output device set to Benchmark 1.0. I even fiddled around with the process priority setting in the Task Manager, setting the foobar2000 process to realtime. No help.

 This is most frustrating. Can anyone help? I don't know what else to do. Did I get a defective unit?

 Thank you




 Update 07/23/09 --

 -- Just got off the phone with Elias from Benchmark media. Elias spent a considerable amount of time going over this problem with me. In the end, it looks like the problem is with my laptop, not the Benchmark DAC1. Oh well, I have been thinking about picking up a new laptop for a while now. I guess now I have that extra incentive. 

 -- Two thumbs up to Elias for outstanding customer service, and to Benchmark Media for manufacturing an outstanding product.


----------



## little-endian

@Elias

 Hehe, thanks for investigating. Not that I'd miss them that much, just wanted to prevent wild speculations on another forum about their absence in the first place. 
	

	
	
		
		

		
		
	


	




 @denonfan

 If the connection works before starting any browser, it's quite unlikely you got a defective unit. 
	

	
	
		
		

		
		
	


	




 Does the browsing actually work flawlessly? Any suspect high CPU usage?

 How does it work on a different PC? There are different chipsets and some are unfortunately just too lame ...

 Puh, if you got "nasty clicks" over USB, there seems to be an advantage of S/PDIF .. never got one no matter how crappy the connection was (just to settle myself as an owner of the "good old one" 
	

	
	
		
		

		
		
	


	




).


----------



## gevorg

Hi Elias,

 I have a general question about how jitter originates in the computer, since there is some confusion/debate about it in some audio forums.

 Can the jitter in computer originate outside of the digital output device (like the soundcard)? Just in theory, regardless whether its audible or very small.

 For example, can the software or operating system affect jitter if it processes audio in bitperfect mode (no volume/resampling/DSP changes)? What about non-audio hardware components like hard drive, motherboard, power supply, etc, can they affect jitter directly (not by increasing EMI/RFI on the audio output device)?

 Thank you.


----------



## Runkby

will my dac 1 hdr work with windows 7 or the mac os x snow leopard coming out soon?


----------



## EliasGwinn

Quote:


  Originally Posted by *gevorg* /img/forum/go_quote.gif 
_Hi Elias,

 I have a general question about how jitter originates in the computer, since there is some confusion/debate about it in some audio forums.

 Can the jitter in computer originate outside of the digital output device (like the soundcard)? Just in theory, regardless whether its audible or very small.

 For example, can the software or operating system affect jitter if it processes audio in bitperfect mode (no volume/resampling/DSP changes)? What about non-audio hardware components like hard drive, motherboard, power supply, etc, can they affect jitter directly (not by increasing EMI/RFI on the audio output device)?

 Thank you._

 

Jitter is caused by transmission devices (transmitter chip on a sound card, USB port, etc), transmission mediums (cable), and receiving devices. Other devices may affect the jitter levels, but not directly. For example, the power supply can affect jitter by affecting the performance of the transmission device.

 If the digital data is going into a buffer, this jitter is irrelevent. If the data is going to a converter (D/A, sync-sample rate converter, etc), this jitter can be devestating. 

 Bit-perfect mode is irrelevent in this matter. Jitter is independent of the the data.

 All the best,
 Elias


----------



## EliasGwinn

Quote:


  Originally Posted by *Runkby* /img/forum/go_quote.gif 
_will my dac 1 hdr work with windows 7 or the mac os x snow leopard coming out soon?_

 

Since the Benchmark AdvancedUSB interface uses the native drivers, we expect it will be fully compatible with Windows 7 and Mac OS X snow leopard. However, we can't guarantee it.

 All the best,
 Elias


----------



## EliasGwinn

We just received the following email from a customer who is using his DAC1 USB with Linux (posted here with his permission):

 "As far as I know there were some doubts about Linux support of DAC1 USB in several discussion forums. Although researching hard, I found no reasonable information on the web.

 I've recently tested the device and it works absolutely fine with Linux (the kernel I've been running at the time of the test was 2.6.28). I don't expect your company to provide any support to Linux users, in fact, no support is necessary 
	

	
	
		
		

		
			





, but I believe status of Linux support may be noted in the product description on the web (e.g., with a disclaimer stating "third party driver, no official support")." - David A.

 Has anybody else had any experience with Linux using Benchmark's AdvancedUSB interface?

 All the best,
 Elias

 ps. we're also discussing this on the new Benchmark Audio Forum:
AdvancedUSB (DAC1 USB, PRE, HDR; ADC1 USB) experiences with Linux | Benchmark Interaction


----------



## Roseval

Quote:


 Has anybody else had any experience with Linux using Benchmark's AdvancedUSB interface? 
 

Had great troubles getting it to work on Ubuntu (8?)/ALSA.
 In the end it turned out to be the combination of Pulse and ALSA causing the problems.
 More details can be found here: USB Audio


----------



## Xxtest83

Hello,

 is it possible to set either RCA or XLR out to "calibrated" and the other one to "variable" ?


----------



## Runkby

Quote:


  Originally Posted by *EliasGwinn* /img/forum/go_quote.gif 
_Since the Benchmark AdvancedUSB interface uses the native drivers, we expect it will be fully compatible with Windows 7 and Mac OS X snow leopard. However, we can't guarantee it.

 All the best,
 Elias_

 

Elias,

 I just installed windows 7 on my computer now and i cant seem to figure out how to get the benchmark device to be recognized... it's not working and i need your help... any suggestions?


----------



## HeadLover

I am having Windows 7 RTM - everything seem to work fine out of the BOX

 The DAC1 PRE with USB works even better than in Vista


----------



## ztsen

Quote:


  Originally Posted by *Runkby* /img/forum/go_quote.gif 
_will my dac 1 hdr work with windows 7 or the mac os x snow leopard coming out soon?_

 

Windows 7 doesnt seem work well with USB Audio. Especially the desktop composition feature is really killing the SQ. I always get distort or glithes here even after I turn off that feature. I posted at windows 7 forum but no help. I think they have put the product into manufacturing version, not sure if final release will solve this issue.

 I am using the windows 7 RC version. (same issue to both 32/64-bits)


----------



## HeadLover

I have win 7 RTM - like I have said before - the DAC1 PRE works GREAT !!!


----------



## thisbenjamin

If you're having issues with the DAC1's USB input, I would look at the driver for your USB controller, or the physical hardware tolerances. If you're running on a system that previously supported the DAC1 using a different OS, I would blame driver support and not the OS, my T60 supported the DAC1 usb interface without issues on the x64 arch (didn't try x86) of win7.


----------



## Runkby

yeah well im going to reformat my comp now and try to get as many drivers as i can get to see if that would make any difference... if that doesnt work i suppose ill just partition the hd and install xp then try to install 7 on vmware so i can still play with it before the full release comes out... .


----------



## Runkby

i got it!... all you had to do was run all of the updates then when i plugged the benchmark in, windows 7 figured the rest out and it's working just as good as it did with windows xp


----------



## helian

Elias,

 I've been using the dac1 usb in Linux for more than a year now. It worked out of the box using the defaul alsa drivers and with 4front-tech's OSS driver.
 I've tested it with 16/44.1, 24/48, 24/88.2 and 24/96 files and they all played fine.



 Regards,

 -vicente


----------



## G-U-E-S-T

Hi Elias,

 I read your online post regarding the use of the DAC1 directly connected to the amp, where you advise the following:

_Quote from Elias Gwinn: "Avoid using amplifiers that are too powerful for your system! You'll get best results when using 75-95% of the amps total power."_

 I always thought it was better to have more clean amp power on tap to properly handle musical transient peaks, similar to what Musical Fidelity advises at this link (also note their interesting "System Diagnostic" link on that page). I have been using a powerful (255W @ 8-ohm) Musical Fidelity stereo amplifier with some very nice 7-ohm minimum bookshelf speakers. After reading your quote above, I'm now wondering if my amp is "too powerful" and whether optimally I should change this. 

 Please advise? Thanks again, in advance.


----------



## roadtonowhere08

Quote:


  Originally Posted by *G-U-E-S-T* /img/forum/go_quote.gif 
_Hi Elias,

 I read your online post regarding the use of the DAC1 directly connected to the amp, where you advise the following:

Quote from Elias Gwinn: "Avoid using amplifiers that are too powerful for your system! You'll get best results when using 75-95% of the amps total power."

 I always thought it was better to have more clean amp power on tap to properly handle musical transient peaks, similar to what Musical Fidelity advises at this link (also note their interesting "System Diagnostic" link on that page). I have been using a powerful (255W @ 8-ohm) Musical Fidelity stereo amplifier with some very nice 7-ohm minimum bookshelf speakers. After reading your quote above, I'm now wondering if my amp is "too powerful" and whether optimally I should change this. 

 Please advise? Thanks again, in advance._

 

It is better to have too much beef than to run of of it and damage you speakers with clipped beef. That is what the volume knob is for. Never go over a certain point, and you will be fine. 

 All speakers have a dynamic wattage rating that is higher than the continuous rating. This is for large transients. The beefier amps are good for dynamics.


----------



## mrarroyo

roadtonowhere08 you are correct, at one time power amps and the better receivers had a "headroom" of 3 dB over the continuous output. This meant that a 200 wpc amp that had a 3 dB headroom could have peaks at 400 wpc.

 I also believe that more speakers are damaged by clipping caused by an anemic amp for the application than too much power.


----------



## Bach-Fan

Elias, if I buy a DAC1 HDR today, will it work seamlessly with a computer running the 64-bit version  of Windows 7 - due out in 3 months?


----------



## HeadLover

I have mine working with win 7 RTM 64 bit

 Working just great!!!


----------



## EliasGwinn

Quote:


  Originally Posted by *Xxtest83* /img/forum/go_quote.gif 
_Hello,

 is it possible to set either RCA or XLR out to "calibrated" and the other one to "variable" ?_

 

No, the output mode applies to both the RCA and XLR.

 All the best,
 Elias


----------



## pcf

Quote:


  Originally Posted by *EliasGwinn* /img/forum/go_quote.gif 
_No, the output mode applies to both the RCA and XLR.

 All the best,
 Elias_

 

Hi Elias,

 Can you advice on how to use an XLR-RCA cable to gain an extra output on The Benchmark? I seem to remember it was mentioned before but couldn't find it through all these pages.

 Thanks!

 Paul


----------



## EliasGwinn

Quote:


  Originally Posted by *G-U-E-S-T* /img/forum/go_quote.gif 
_Hi Elias,

 I read your online post regarding the use of the DAC1 directly connected to the amp, where you advise the following:

Quote from Elias Gwinn: "Avoid using amplifiers that are too powerful for your system! You'll get best results when using 75-95% of the amps total power."

 I always thought it was better to have more clean amp power on tap to properly handle musical transient peaks, similar to what Musical Fidelity advises at this link (also note their interesting "System Diagnostic" link on that page). I have been using a powerful (255W @ 8-ohm) Musical Fidelity stereo amplifier with some very nice 7-ohm minimum bookshelf speakers. After reading your quote above, I'm now wondering if my amp is "too powerful" and whether optimally I should change this. 

 Please advise? Thanks again, in advance._

 

I agree with them, for the most part. I'll clarify my statement:

 It is best to have an amplifier whose input peaks at the same level as the maximum output level of the source (e.g., DAC, PRE), and this amplifier/speaker combination should only be "loud" enough so that maximum comfortable listening levels are acheived near peak input (transient peaks reach full-scale). In other words, it is ideal to use ALL of your headroom without going over.

 The reason for this is to acheive the best possible dynamic range. If you must attenuate the source to prevent the amplifier/speakers from becoming too loud for comfort, then you are losing dynamic range. The amount of dynamic range you are losing is equivelant to the amount of attenuation you are employing. 

 The reason is that the amplifier has a specified noise floor. This noise doesn't attenuate when the source is attenuated. So, when the source is attenuated, the result is lower signal-to-noise ratio.

 In a high-precision playback system, amplifiers are typically the limiting factor in dynamic range. If your amp is too powerful for your typical listening levels, you will attenuate your source and decrease your dynamic range even further.

 Does this make sense to you?

 All the best,
 Elias


----------



## EliasGwinn

Quote:


  Originally Posted by *roadtonowhere08* /img/forum/go_quote.gif 
_It is better to have too much beef than to run of of it and damage you speakers with clipped beef._

 

It is true that clipping your amp can damage your tweeters because of the increase in high-frequency content. You certainly don't want to operate a power amp into clipping.

 I would also agree that it is better to error on the side of too much headroom rather then not enough.

 However, the idea of "the more headroom the better" is not true.

 All the best,
 Elias


----------



## EliasGwinn

Quote:


  Originally Posted by *Bach-Fan* /img/forum/go_quote.gif 
_Elias, if I buy a DAC1 HDR today, will it work seamlessly with a computer running the 64-bit version  of Windows 7 - due out in 3 months?_

 

We don't have Windows 7 yet, so I haven't tested it yet. But several customers have done it successfully (including one responding to your post).

 All the best,
 Elias


----------



## EliasGwinn

Quote:


  Originally Posted by *pcf* /img/forum/go_quote.gif 
_Hi Elias,

 Can you advice on how to use an XLR-RCA cable to gain an extra output on The Benchmark? I seem to remember it was mentioned before but couldn't find it through all these pages.

 Thanks!

 Paul_

 

The XLR connector must have a floating pin-3 (nothing connected to it). Pin 2 is connected the center pin of the RCA, and Pin 1 is connected to the outer-casing of the RCA. Again, pin-3 is not connected to anything.

 Here is the cable:

Benchmark Media

 All the best,
 Elias


----------



## denp

Quote:


  Originally Posted by *bralk* /img/forum/go_quote.gif 
_For large scale reproduction you need large speakers. I can recommend 
 ATC SCM 100 ASL or perhaps the SCM 50 ASL depending on your listening environment. 

 cheers

 Tom_

 

I also happen to use SCM100A connected directly to HDR DAC1 / PRE - I can testify it's an incredible match! 

 What's also recommended is to shift the attenuation to -30dB; First this improves the usable Volume range (in my case), second it lowers the Output impedance of the DAC1/PRE to 43 Ohms, which is greatly desired by the ATCs! 

 Having used the ATC with a bunch of other DAC & PRE combos, I can say DAC1/PRE fits the bill very well. Overall creates a more insightful (musically) combo than any other one I tried with the ATCs..


----------



## pcf

Quote:


  Originally Posted by *EliasGwinn* /img/forum/go_quote.gif 
_The XLR connector must have a floating pin-3 (nothing connected to it). Pin 2 is connected the center pin of the RCA, and Pin 1 is connected to the outer-casing of the RCA. Again, pin-3 is not connected to anything.

 Here is the cable:
Benchmark Media

 All the best,
 Elias_

 

Thanks Elias!

 Paul


----------



## gevorg

Quote:


  Originally Posted by *EliasGwinn* /img/forum/go_quote.gif 
_I agree with them, for the most part. I'll clarify my statement:

 It is best to have an amplifier whose input peaks at the same level as the maximum output level of the source (e.g., DAC, PRE), and this amplifier/speaker combination should only be "loud" enough so that maximum comfortable listening levels are acheived near peak input (transient peaks reach full-scale). In other words, it is ideal to use ALL of your headroom without going over.

 The reason for this is to acheive the best possible dynamic range. If you must attenuate the source to prevent the amplifier/speakers from becoming too loud for comfort, then you are losing dynamic range. The amount of dynamic range you are losing is equivelant to the amount of attenuation you are employing. 

 The reason is that the amplifier has a specified noise floor. This noise doesn't attenuate when the source is attenuated. So, when the source is attenuated, the result is lower signal-to-noise ratio.

 In a high-precision playback system, amplifiers are typically the limiting factor in dynamic range. If your amp is too powerful for your typical listening levels, you will attenuate your source and decrease your dynamic range even further.

 Does this make sense to you?

 All the best,
 Elias_

 

What if your amplifier has gain controls, will that help to improve dynamic range? Or its the same as attenuating at the source? For example, say a low-impedance and high sensitivity headphones are too loud at 20% of maximum volume of a given powerful amplifier. Then you reduce the gain inside amplifier so that the maximum listenable volume increases to around 50% of maximum volume. At the same volume level, will the dynamic range improve?


----------



## Shadorne

Quote:


  Originally Posted by *denp* /img/forum/go_quote.gif 
_I also happen to use SCM100A connected directly to HDR DAC1 / PRE - I can testify it's an incredible match! 

 What's also recommended is to shift the attenuation to -30dB; First this improves the usable Volume range (in my case), second it lowers the Output impedance of the DAC1/PRE to 43 Ohms, which is greatly desired by the ATCs! 

 Having used the ATC with a bunch of other DAC & PRE combos, I can say DAC1/PRE fits the bill very well. Overall creates a more insightful (musically) combo than any other one I tried with the ATCs.._

 

DITTO Here. We must start a "club"
	

	
	
		
		

		
			





 I also use the DAC1 with Active ATC 100's. I read somewhere that Doug Sax installed DAC1 with ATC's in his new Mastering Lab in Ojai, Califoria. So it sure seems like a great combo to a few folks. I did some tests myself and the improvement with the DAC1 versus other sources was substantial. I suspect you really hear the phase accuracy of this DAC when using phase compensated Active speakers (speakers which are about as good as it gets in transient response).


----------



## Shadorne

Quote:


  Originally Posted by *G-U-E-S-T* /img/forum/go_quote.gif 
_Hi Elias,

 I read your online post regarding the use of the DAC1 directly connected to the amp, where you advise the following:

Quote from Elias Gwinn: "Avoid using amplifiers that are too powerful for your system! You'll get best results when using 75-95% of the amps total power."

 I always thought it was better to have more clean amp power on tap to properly handle musical transient peaks, similar to what Musical Fidelity advises at this link (also note their interesting "System Diagnostic" link on that page). I have been using a powerful (255W @ 8-ohm) Musical Fidelity stereo amplifier with some very nice 7-ohm minimum bookshelf speakers. After reading your quote above, I'm now wondering if my amp is "too powerful" and whether optimally I should change this. 

 Please advise? Thanks again, in advance._

 

Actually I think there is not a single simple answer here. Ratings of speakers are very subjective anyway. Many speakers with high ratings are simply designed to compress (long voice coil in short gap) - sure they can handle huge amounts of power without breaking but they quickly sound dull as the voice coil heats up and often these designs have a smaller Xmax - so you are just listening to distortion anyway.

 Practically speaking you generally need a coupe of hundred watts for each large high quality woofer (12" +). A midrange can often use no more than 50 to 100 watts. While a tweeter you may be fine with a mere 10 watts. Practically speaking, for "best" you might want to go Class D amps with several large woofers for the bass whilst it would be "best" to run your tweeter and mids on a good Class A - or for those who like warmth you could simply add tubes.

 This is why Active speakers are so attractive - you get the Hanna Montana experience - the "best of both worlds" - great bass that even when over-driven does so without adding distortion to the mids and highs...
	

	
	
		
		

		
			





 (We can all put up with some distortion in the bass but distortion in the mids and highs is AWFUL, nails on blackboard)


----------



## EliasGwinn

Quote:


  Originally Posted by *gevorg* /img/forum/go_quote.gif 
_What if your amplifier has gain controls, will that help to improve dynamic range? Or its the same as attenuating at the source? For example, say a low-impedance and high sensitivity headphones are too loud at 20% of maximum volume of a given powerful amplifier. Then you reduce the gain inside amplifier so that the maximum listenable volume increases to around 50% of maximum volume. At the same volume level, will the dynamic range improve?_

 

It's hard to say because amplifiers have very different implementations of 'gain control'. Sometimes its simply input attenuators.

 Can you describe your proposed setup more specifically, including model numbers?

 All the best,
 Elias


----------



## svetlanafoster

I am Svetlana, I am new to this wonderful community. Thank you for having me here and I hope I can also contribute in some small way. I want every man, woman and child on this 

 beautiful earth to have a copy of the Holy Bible. Did you know, there are many countries where the Bible is illegal ?!! I only found that out recently. I found this glorious 

 site that has a sweet sweet beautiful computer Bible BIBLE VERSE I really love the name of the site ! I also love rabbits, I have five (yes!) pet rabbits 

 of my own, they run around in my backyard; my neighbours are not very pleased with them, but they love them too.
multiplication tables 
vocabulary words 
bible verses


----------



## yilmaz196

Hi Elias, i just bought usb dac1. I have a question. Even when i am not listening to it, it is still hot. There is no on-off button so i am switching to another input(i use usb input for listening) to standby mode. But like i said it is still hot. No lights but always hot. Is this normal, this dac never off i think.


----------



## EliasGwinn

Quote:


  Originally Posted by *yilmaz196* /img/forum/go_quote.gif 
_Hi Elias, i just bought usb dac1. I have a question. Even when i am not listening to it, it is still hot. There is no on-off button so i am switching to another input(i use usb input for listening) to standby mode. But like i said it is still hot. No lights but always hot. Is this normal, this dac never off i think._

 

This is normal. When the DAC1 USB is in 'Standby' mode, all electronics are still powered. Therefore, the temperature of the unit will not change. This won't affect the performance of the DAC1 USB, nor will it shorten the life-span. It is not a problem.

 All the best,
 Elias


----------



## Gerykatss

I am looking for some sites which are interested in some
 link or banner exchange with my Istockvanities.com: Bathroom Vanities - Bathroom Vanity on Sale
 I have already recommended urls of site to a couple of friends 
Istockvanities.com: Discount Bathroom Vanity - Discount Bathroom Vanities
 and Istockvanities.com: Glass Vanity - Glass Vanities Omline Sale
 It's easy and we can start today!


----------



## G-U-E-S-T

Quote:


  Originally Posted by *EliasGwinn* /img/forum/go_quote.gif 
_I agree with them, for the most part. I'll clarify my statement:

 It is best to have an amplifier whose input peaks at the same level as the maximum output level of the source (e.g., DAC, PRE), and this amplifier/speaker combination should only be "loud" enough so that maximum comfortable listening levels are acheived near peak input (transient peaks reach full-scale). In other words, it is ideal to use ALL of your headroom without going over.

 The reason for this is to acheive the best possible dynamic range. If you must attenuate the source to prevent the amplifier/speakers from becoming too loud for comfort, then you are losing dynamic range. The amount of dynamic range you are losing is equivelant to the amount of attenuation you are employing. 

 The reason is that the amplifier has a specified noise floor. This noise doesn't attenuate when the source is attenuated. So, when the source is attenuated, the result is lower signal-to-noise ratio.

 In a high-precision playback system, amplifiers are typically the limiting factor in dynamic range. If your amp is too powerful for your typical listening levels, you will attenuate your source and decrease your dynamic range even further.

 Does this make sense to you?

 All the best,
 Elias_

 

Hi Elias,

 Thank you very much for your reply, and sorry about the delay in responding. I *think* that I understand what you are explaining - but I just don't personally know how this could, practically speaking, be achieved (?)

 For instance, my bookshelf speakers are 7-ohm minimum, and I never drive them to really high levels. My current stereo amp gives 255 watts (way more than I probably need) into 8 ohms, from 1.5v input, with <0.01% THD (20Hz - 20KHz) and >109dB SNR.

 But it is typical of my sources to provide what I believe is very significantly more than 1.5 volts peak output! For example:

 1) My DAC1-Pre "variable" RCA output reaches up to 11dBu (which I think is about 2.75v rms, 7.8v peak-to-peak), with <0.00056% THD and 116dB SNR.

 2) Another of my sources (a SqueezeBox Classic) on its RCA outputs gives about 2.1v rms (6.0v peak-to-peak), with <0.002% THD and >100dB SNR.

 So if I didn't attenuate these sources, I would probably go deaf in a matter of moments! It seems if I understand your advice correctly, that I would need to find an amp that has an input sensitivity of almost 3v -- and even then, only outputting a small number (maybe 50 or 75?) of watts, to power my bookshelf speakers. Is such an amp out there?

 Also, if I do recall correctly: I think you previously recommended getting things set up to where the DAC1 volume level is comfortable within around the 10 o'clock to 2 o'clock range. So since this range is (obviously) far from full volume, wouldn't this also be less than ideal as well, as per your advice above?

 Very sorry if these are "dumb" questions, but I really am trying to understand! 
	

	
	
		
		

		
		
	


	




 [Edit: Typo - changed the "116dB THD" to "116dB SNR"]


----------



## Quaddy

elias, would appreciate some backup here, have the dac1 pre and am using one set of headphones with it only, the sennheiser hd800 which has a impedance rating of 300ohm

 with regard to headphone gain adjustment would the unit be best at position A, B, or removed

 to be honest i cant remember what position they are at now, or if they are removed, i will look soon

 i appreciate this all depends on setup, but on paper is there anything which shouts to you that would maybe be better matched on paper with the gain range for 300ohm

 thanks!

*edit:* _well i went through the options and had a good listen at each change, to my ears, the hd800 sounds a lot more driven and powered correctly with the headphone gain jumpers removed, not at -10db or -20db - in fact i am suprised at how anemic they sounded with anything other than removed for max gain. guess i answered my own question there anyway 
	

	
	
		
		

		
		
	


	


_


----------



## EliasGwinn

Quote:


  Originally Posted by *G-U-E-S-T* /img/forum/go_quote.gif 
_Hi Elias,

 Thank you very much for your reply, and sorry about the delay in responding. I *think* that I understand what you are explaining - but I just don't personally know how this could, practically speaking, be achieved (?)

 For instance, my bookshelf speakers are 7-ohm minimum, and I never drive them to really high levels. My current stereo amp gives 255 watts (way more than I probably need) into 8 ohms, from 1.5v input, with <0.01% THD (20Hz - 20KHz) and >109dB SNR.

 But it is typical of my sources to provide what I believe is very significantly more than 1.5 volts peak output! For example:

 1) My DAC1-Pre "variable" RCA output reaches up to 11dBu (which I think is about 2.75v rms, 7.8v peak-to-peak), with <0.00056% THD and 116dB THD.

 2) Another of my sources (a SqueezeBox Classic) on its RCA outputs gives about 2.1v rms (6.0v peak-to-peak), with <0.002% THD and >100dB SNR.

 So if I didn't attenuate these sources, I would probably go deaf in a matter of moments! It seems if I understand your advice correctly, that I would need to find an amp that has an input sensitivity of almost 3v -- and even then, only outputting a small number (maybe 50 or 75?) of watts, to power my bookshelf speakers. Is such an amp out there?

 Also, if I do recall correctly: I think you previously recommended getting things set up to where the DAC1 volume level is comfortable within around the 10 o'clock to 2 o'clock range. So since this range is (obviously) far from full volume, wouldn't this also be less than ideal as well, as per your advice above?

 Very sorry if these are "dumb" questions, but I really am trying to understand! 
	

	
	
		
		

		
		
	


	


_

 


 They're not dumb questions! I'm happy to explain.

 Tell me this... You use the RCA outputs of the DAC1 to directly drive your amplifier, correct? Where do you normally have the volume control set?


----------



## EliasGwinn

Quote:


  Originally Posted by *Quaddy* /img/forum/go_quote.gif 
_elias, would appreciate some backup here, have the dac1 pre and am using one set of headphones with it only, the sennheiser hd800 which has a impedance rating of 300ohm

 with regard to headphone gain adjustment would the unit be best at position A, B, or removed

 to be honest i cant remember what position they are at now, or if they are removed, i will look soon

 i appreciate this all depends on setup, but on paper is there anything which shouts to you that would maybe be better matched on paper with the gain range for 300ohm

 thanks!

*edit:* well i went through the options and had a good listen at each change, to my ears, the hd800 sounds a lot more driven and powered correctly with the headphone gain jumpers removed, not at -10db or -20db - in fact i am suprised at how anemic they sounded with anything other than removed for max gain. guess i answered my own question there anyway 
	

	
	
		
		

		
		
	


	


_

 

I use the HD650's, which are also 300 ohm. I keep the HPA2 gain range in the -10 dB position (the 'A' position).

 The optimal setting is whatever allows you to use the volume control in the 'sweet spot' (10 o'clock - 3 o'clock). If this range is too loud, decrease the gain range jumper setting. If this range is not loud enough, reduce the gain range setting.

 All the best,
 Elias


----------



## G-U-E-S-T

Quote:


  Originally Posted by *EliasGwinn* /img/forum/go_quote.gif 
_They're not dumb questions! I'm happy to explain.

 Tell me this... You use the RCA outputs of the DAC1 to directly drive your amplifier, correct? Where do you normally have the volume control set?_

 

When using the RCA outputs, and for our "regular" daily listening: I think we probably keep the volume dial between around 8 o'clock and 11 o'clock on average. We hardly ever try to rock the house down! 
	

	
	
		
		

		
		
	


	




 P.S. I also just noted from the manual, that in variable mode, the DAC1 volume control attenuates the signal until the dial reaches around 3 o'clock (the 30th detent or so), at which point 0dB attenuation is reached (per the manual graph).


----------



## EliasGwinn

Quote:


  Originally Posted by *G-U-E-S-T* /img/forum/go_quote.gif 
_When using the RCA outputs, and for our "regular" daily listening: I think we probably keep the volume dial between around 8 o'clock and 11 o'clock on average. We hardly ever try to rock the house down! 
	

	
	
		
		

		
		
	


	




 P.S. I also just noted from the manual, that in variable mode, the DAC1 volume control attenuates the signal until the dial reaches around 3 o'clock (the 30th detent or so), at which point 0dB attenuation is reached (per the manual graph)._

 

So, in your case, the output from the DAC1 is around 0.25 V. This is a very small percentage of your amplifier's operating range. 

 Your amplifier's input circuitry will have a certain noise level, which is also amplified along with the input signal. The more powerful the amp, the louder the noise AND signal gets amplified. 

 So, in your case, you've got a small signal and a (relatively) large noise. Consequently, your signal-to-noise ratio is suffering. 

 If your amplifier had a lower gain ratio (less power), you could increase the signal and decrease the noise. Consequently, you would improve your signal to noise ratio.

 Now, with that said, it is true that you want to have ample headroom, but as you can see here, too much headroom is detrimental as well. Ideally, the output of the DAC1 would peak just below the maximum input level of the amplifier without it becoming too loud for your listening comfort.

 All the best,
 Elias


----------



## G-U-E-S-T

Quote:


  Originally Posted by *EliasGwinn* /img/forum/go_quote.gif 
_So, in your case, the output from the DAC1 is around 0.25 V. This is a very small percentage of your amplifier's operating range. 

 Your amplifier's input circuitry will have a certain noise level, which is also amplified along with the input signal. The more powerful the amp, the louder the noise AND signal gets amplified. 

 So, in your case, you've got a small signal and a (relatively) large noise. Consequently, your signal-to-noise ratio is suffering. 

 If your amplifier had a lower gain ratio (less power), you could increase the signal and decrease the noise. Consequently, you would improve your signal to noise ratio.

 Now, with that said, it is true that you want to have ample headroom, but as you can see here, too much headroom is detrimental as well. Ideally, the output of the DAC1 would peak just below the maximum input level of the amplifier without it becoming too loud for your listening comfort.

 All the best,
 Elias_

 

Thanks very much for this very interesting SNR information Elias. As you probably know, this is not an aspect usually mentioned, as the gospel-dogma for most audiophile-oriented consumer buyers (and dealers!) is to always buy as many extremely-clean watts as you can afford - and that is in fact how my dealer sold me a (apparently WAY over-powered) $3000 stereo amp to accompany our really nice bookshelf speakers. And honestly, it really does sound extremely good just as it is! On your advice though, I will now begin looking for a new (and much lower powered) stereo amp that is more appropriate for my system. If I understand you correctly, for us this should ideally be a stereo amp with only approx. 50 watts of power, and requiring approx. 3-volts of input signal to actually reach that full 50 watts. Is that about correct?

 I'm embarassed to admit this, but so far in my rather careful initial search for such an amp, I cannot seem to find anything that properly fits those parameters as I believe you've described for me. Can you perhaps please, suggest any stereo amps that would be good choices for us to try?

 Also, is there a general formula available that you could please share, for calculating both the nominal voltage and the peak voltage that the DAC1 outputs at various detents (or perhaps at the approx. clock-hour positions) on it's volume dial? That would be *so* helpful to have, so that DAC1 users could try to find amps with input sensitivities that properly match their usual source signal voltages. I'm also not certain if the 0.25-volt value you mentioned in your last post, represents nominal output, or peak output (?)

 Thanks again in advance Elias - I think this info will help a lot of people!


----------



## wavoman

Help with the trim pots, pretty please!

 Well I see by searching this thread -- a little too late, I'm afriad -- that I should not have screwed with the trimmers.

 Before I opened up the box and moved the jumper to 0, solving my balanced output volume problem, I used a small screwdriver to turn the trimmers.

 But the trimmers don't seem to have ANY mechanical stops on either side!

 I see I am supposed to have test tones and meters -- but I don't.

 All I want to do now is get L and R at the same level -- but how do I know where the trimmers are? I turn and turn and turn and never hit a stop in either direction ... did WAY MORE than 10 turns!

 Any help would be appreciated. For now I have moved the toggle to VARIABLE to take the trimmers out of play, but I don't want to leave it there ... I wanted fixed (calibrated) output to my amp for speakers, and wanted to use the volume control only for headphones!

 Elias, Help !! ?? Am I crazy -- is there a mechanical stop and I just gave up?

 THANKS!


----------



## synfreak

Hi Wavoman!

 Do you own a (good quality) "soundcard/audiointerface" and some dedicated software (i.e. audacity)?

 you then might be able to use the meters in the software to adjust the analogue levels of your DAC1.

 This might not be 100% perfect (as it would be with a multimeter and the replay of a sinus signal), but it should be within about 0,5db (depends on the precission of the software meter).

 Cheers
 Harald


----------



## wavoman

Quote:


  Originally Posted by *synfreak* /img/forum/go_quote.gif 
_...you then might be able to use the meters in the software to adjust the analogue levels of your DAC1..._

 

What a great idea, I never thought to do that! Thanks!

 Maybe the Benchmark folks will weight in, too. I am really interested to understand why there are no physical stops (or if there are, and I'm just all thumbs).


----------



## EliasGwinn

Quote:


  Originally Posted by *G-U-E-S-T* /img/forum/go_quote.gif 
_Thanks very much for this very interesting SNR information Elias. As you probably know, this is not an aspect usually mentioned, as the gospel-dogma for most audiophile-oriented consumer buyers (and dealers!) is to always buy as many extremely-clean watts as you can afford - and that is in fact how my dealer sold me a (apparently WAY over-powered) $3000 stereo amp to accompany our really nice bookshelf speakers. And honestly, it really does sound extremely good just as it is! On your advice though, I will now begin looking for a new (and much lower powered) stereo amp that is more appropriate for my system. If I understand you correctly, for us this should ideally be a stereo amp with only approx. 50 watts of power, and requiring approx. 3-volts of input signal to actually reach that full 50 watts. Is that about correct?

 I'm embarassed to admit this, but so far in my rather careful initial search for such an amp, I cannot seem to find anything that properly fits those parameters as I believe you've described for me. Can you perhaps please, suggest any stereo amps that would be good choices for us to try?

 Also, is there a general formula available that you could please share, for calculating both the nominal voltage and the peak voltage that the DAC1 outputs at various detents (or perhaps at the approx. clock-hour positions) on it's volume dial? That would be *so* helpful to have, so that DAC1 users could try to find amps with input sensitivities that properly match their usual source signal voltages. I'm also not certain if the 0.25-volt value you mentioned in your last post, represents nominal output, or peak output (?)

 Thanks again in advance Elias - I think this info will help a lot of people! 
	

	
	
		
		

		
			



_

 

If you're going to be changing your amplifier, remember that it IS important to have ample headroom. In other words, don't go TOO small. 

 Ideally, the unbalanced input of the amp would be capable of the full 2.75 Vrms (7.75 Vp-p) output of the DAC1, at minimum. Then, the ideal amp will drive YOUR speakers to YOUR comfortable listening level when the DAC1's volume control is between 11 o'clock and 3 o'clock while listening to YOUR type of music. (I emphasis "YOUR" because these quantities are not objective, they are subjective. In other words, use this guide to suit YOUR specific listening situation). 

 Regarding calculating voltage-to-volume control position, use the calculator on this website, along with the 'Volume Control Curve' in the DAC1 manual, will enable you to convert between the two. If you need any assistance with this, let me know and I will help.

 All the best,
 Elias


----------



## Roseval

Quote:


 I'm embarassed to admit this, but so far in my rather careful initial search for such an amp, I cannot seem to find anything that properly fits those parameters 
 


 You might try to find an amp matching your speakers but a good alternative might be to look for a pair of active speakers.
 This are speakers with a build in amplifier.
 The good ones have each driver powered by it’s own matched amplifier.
 The crossover is active so done before the signal enters the poweramp
 This in general allows for more exact and steeper filters then possible with passive components.
 In the pro-world, active speakers are common, in home audio they are rare.

 A bit more details: The Well Tempered Computer


----------



## wavoman

Quote:


  Originally Posted by *wavoman* /img/forum/go_quote.gif 
_Maybe the Benchmark folks will weight in, too. I am really interested to understand why there are no physical stops (or if there are, and I'm just all thumbs)._

 

Elias -- any thoughts on how to get the trims all the way one way or the other ...are there really no physical stops??

 Thanks!


----------



## EliasGwinn

Quote:


  Originally Posted by *wavoman* /img/forum/go_quote.gif 
_Help with the trim pots, pretty please!

 Well I see by searching this thread -- a little too late, I'm afriad -- that I should not have screwed with the trimmers.

 Before I opened up the box and moved the jumper to 0, solving my balanced output volume problem, I used a small screwdriver to turn the trimmers.

 But the trimmers don't seem to have ANY mechanical stops on either side!

 I see I am supposed to have test tones and meters -- but I don't.

 All I want to do now is get L and R at the same level -- but how do I know where the trimmers are? I turn and turn and turn and never hit a stop in either direction ... did WAY MORE than 10 turns!

 Any help would be appreciated. For now I have moved the toggle to VARIABLE to take the trimmers out of play, but I don't want to leave it there ... I wanted fixed (calibrated) output to my amp for speakers, and wanted to use the volume control only for headphones!

 Elias, Help !! ?? Am I crazy -- is there a mechanical stop and I just gave up?

 THANKS!_

 

Hello Wavoman,

 The trim-potentiometers do not have physical stops, as is common for these types. Without a volt-meter, it is difficult to acheive a precise balance. 

 Your options are:

 -Send the unit to us ($50 for re-calibration) 
 -Take it to any audio technician to have it properly balanced
 -Buy a volt-meter from radio-shack for $10 and do it yourself. If you do this, I'll help you via telephone.

 All the best,
 Elias


----------



## EliasGwinn

Quote:


  Originally Posted by *Roseval* /img/forum/go_quote.gif 
_You might try to find an amp matching your speakers but a good alternative might be to look for a pair of active speakers.
 This are speakers with a build in amplifier.
 The good ones have each driver powered by it’s own matched amplifier.
 The crossover is active so done before the signal enters the poweramp
 This in general allows for more exact and steeper filters then possible with passive components.
 In the pro-world, active speakers are common, in home audio they are rare.

 A bit more details: The Well Tempered Computer_

 

I couldn't agree more! We strongly recommend active speakers for these exact reasons. 

 All the best,
 Elias


----------



## wavoman

Quote:


  Originally Posted by *EliasGwinn* /img/forum/go_quote.gif 
_Buy a volt-meter from radio-shack for $10 and do it yourself. If you do this, I'll help you via telephone._

 

Sounds like fun. More so for me than for you, but I appreciate it. I gather you would prefer a business day, so it might be next week. I'll hit ratshack 2nite on the way home, then PM you to pick the best time for you.

 Actually maybe I'll buy a slightly better unit off the web.

 THANKS!


----------



## EliasGwinn

Quote:


  Originally Posted by *wavoman* /img/forum/go_quote.gif 
_Sounds like fun. More so for me than for you, but I appreciate it. I gather you would prefer a business day, so it might be next week. I'll hit ratshack 2nite on the way home, then PM you to pick the best time for you.

 Actually maybe I'll buy a slightly better unit off the web.

 THANKS!_

 

Don't forget, you will also need a test tone. Is your DAC1 equiped with a USB interface? Or do you have other means of playing audio from your computer digitally into the DAC1? If so, I can upload a test tone that you can download for this...

 atb,
 e


----------



## wavoman

Quote:


  Originally Posted by *EliasGwinn* /img/forum/go_quote.gif 
_Don't forget, you will also need a test tone. Is your DAC1 equiped with a USB interface? Or do you have other means of playing audio from your computer digitally into the DAC1? If so, I can upload a test tone that you can download for this..._

 

Thanks for all your help! No USB on my Benchmark, but I have a S/PDIF output from my PC via my MAUDIO 192 card, no problem. Tested it with the Benchmark, it works fine.

 So if you would upload the file that would be GREAT!

 An auction was just ending on eBay, so I scored a fully professional Fluke meter for $50, about one-third of list. I'm reading the pdf manual downloaded from Fluke now. Yea I know, it's too good a tool for me! Whenever I pull a DeWalt out of the closet my brother-in-law says to me "hey that's not for you, you are not in the league that equipment demands". Same here ... what is it about yellow?

 Will I need the cover off? Grounding wrist-strap? I assume so.

 I am totally out of my depth here but more than willing to try. Hey, I changed the jumper!


----------



## Shadorne

Quote:


  Originally Posted by *EliasGwinn* /img/forum/go_quote.gif 
_I couldn't agree more! We strongly recommend active speakers for these exact reasons. 

 All the best,
 Elias_

 

ATC has been making active speakers for more than twenty years. 

  Quote:


 Active loudspeakers are today used by virtually every recording company, every major recording studio, and every major film studio. 
 

No sh^t Sherlock. There is a good reason for this - it really is a no brainer for anyone with an engineering degree and who understand amplifiers and transducers...of course, for those without the maths and physics background, all you have to do is have a good long listen to a good active speaker and you'll find there is no going back.


----------



## G-U-E-S-T

Quote:


  Originally Posted by *EliasGwinn* /img/forum/go_quote.gif 
_If you're going to be changing your amplifier, remember that it IS important to have ample headroom. In other words, don't go TOO small. 

 Ideally, the unbalanced input of the amp would be capable of the full 2.75 Vrms (7.75 Vp-p) output of the DAC1, at minimum. Then, the ideal amp will drive YOUR speakers to YOUR comfortable listening level when the DAC1's volume control is between 11 o'clock and 3 o'clock while listening to YOUR type of music. (I emphasis "YOUR" because these quantities are not objective, they are subjective. In other words, use this guide to suit YOUR specific listening situation). 

 Regarding calculating voltage-to-volume control position, use the calculator on this website, along with the 'Volume Control Curve' in the DAC1 manual, will enable you to convert between the two. If you need any assistance with this, let me know and I will help.

 All the best,
 Elias_

 






 Thanks Elias! Learning a lot here. Here's what comes to my mind right away:

 1) As long as you clearly stay within your amp's range of clean output levels: Why would one really want very much (ample) extra headroom, beyond the levels that you ever actually listen at?

 2) Have you (or anyone else here) ever found or known of, any quality solid-state amp that delivers on those types of specs discussed above (input sensitivity approx 2.75v or above, output wattage not more than 100 watts)?

 3) Do amp manufacturers ever build-in resistors (or sometimes even step-down input transformers) at their inputs to achieve the desired input sensitivity - as this seems like it would entail all the same mentioned drawbacks as other pre-amp attenuation schemes do?

 4) If using active speakers, does the same advice about the DAC1 volume dial still apply (i.e. best to keep it between 11:00 and 3:00 on the dial)?

 [P.S. - Also, when it comes to actives (and I have tried a few), they all seem to be either very ugly, very directional, very near-field, or *very* expensive - and I've yet to hear any (so far up to about the $2200 mark), that sound even half as good as our current separates do! 
	

	
	
		
		

		
		
	


	




]


----------



## Shadorne

Quote:


  Originally Posted by *G-U-E-S-T* /img/forum/go_quote.gif 
_Have you (or anyone else here) ever found or known of, any quality solid-state amp that delivers on those types of specs discussed above (input sensitivity approx 2.75v or above, output wattage not more than 100 watts)?

 3) Do amp manufacturers ever build-in resistors (or sometimes even step-down input transformers) at their inputs to achieve the desired input sensitivity - as this seems like it would entail all the same mentioned drawbacks as other pre-amp attenuation schemes do?

 4) If using active speakers, does the same advice about the DAC1 volume dial still apply (i.e. best to keep it between 11:00 and 3:00 on the dial)?

 [P.S. - Also, when it comes to actives (and I have tried a few), they all seem to be either very ugly, very directional, very near-field, or *very* expensive - and I've yet to hear any (so far up to about the $2200 mark), that sound even half as good as our current separates do! 
	

	
	
		
		

		
		
	


	




]_

 

All the 200+ watt power amps I have owned came with potentiometers for attenuation of inputs and channel matching. Although on my Bryston 4B it is an upgrade to have the trim pots added.

 You can find very good active speakers starting around $4,000 however, in general, they are indeed ugly and being built mostly for professionals they often lack the kind of coloration that consumers prefer and may be judged as sounding thin and harsh (a midrange suckout or politely called the BBC dip coupled with enhanced bass response is the most popular sound that consumers all seem to gravitate towards and you can easily find this on any normal hi-fi shop floor but this pleasing sound is less common with Active designs)


----------



## Roseval

Quote:


 You can find very good active speakers starting around $4,000 however, in general, they are indeed ugly and being built mostly for professionals they often lack the kind of coloration that consumers prefer and may be judged as sounding thin and harsh (a midrange suckout or politely called the BBC dip coupled with enhanced bass response is the most popular sound that consumers all seem to gravitate towards and you can easily find this on any hi-fi shop floor) 
 

You sums it up well.
 Indeed a lot of these consumer systems are modest versions of car stereo with their brilliant treble and muscle bass.
 Actives targeting the home are rare, the few I know can be find here (including some pro models as well) The Well Tempered Computer


----------



## Ssnake51

Quote:


  Originally Posted by *Shadorne* /img/forum/go_quote.gif 
_All the 200+ watt power amps I have owned came with potentiometers for attenuation of inputs and channel matching. Although on my Bryston 4B it is an upgrade to have the trim pots added.

 You can find very good active speakers starting around $4,000 however, in general, they are indeed ugly and being built mostly for professionals they often lack the kind of coloration that consumers prefer and may be judged as sounding thin and harsh (a midrange suckout or politely called the BBC dip coupled with enhanced bass response is the most popular sound that consumers all seem to gravitate towards and you can easily find this on any normal hi-fi shop floor but this pleasing sound is less common with Active designs)_

 

I'm a big fan of active monitors. I know they don't come close to the $4000 price you mentioned, but my Dynaudio BM6A's sound fantastic to me when paired with the Benchmark.


----------



## Shadorne

Quote:


  Originally Posted by *Ssnake51* /img/forum/go_quote.gif 
_I'm a big fan of active monitors. I know they don't come close to the $4000 price you mentioned, but my Dynaudio BM6A's sound fantastic to me when paired with the Benchmark.
	

	
	
		
		

		
		
	


	


_

 


 Dynaudio make their own drivers. They make awesome speakers both passive and active. BM6A's are very popular. 

 I know that most people on these forums must prefer headphones and therefore a DAC1 makes a lot of sense (includes a good headphone amp), however, for folks who listen mostly to music through passive speakers then I would assume that if they have invested so much on a source and also own a big power amp or high powered integrated then surely they would be looking to spend at LEAST as much on speakers if not a lot more - that is kind of where I came up with $4,000 (new retail). FWIW - a setup without a large majority of the spend on music and speakers is ass backwards in my book. 

 I'd add that Beolab 5 might be the kind of active speaker that would turn the crank of most consumers. It looks simply great (conversation piece), sounds excellent with awesome natural sounding wide even dispersion and has a powerful yet articulate bass with a sweet midrange that is just a wee bit turned down - so it won't sound hot or harsh when cranked (that little Vifa 3 inch dome can get a little congested sounding at higher SPL's but it never sounds harsh)


----------



## EliasGwinn

Quote:


  Originally Posted by *wavoman* /img/forum/go_quote.gif 
_Will I need the cover off? Grounding wrist-strap?_

 

No and no. Not for this. Just a small screwdriver to turn the pots, a test-tone, and a volt meter.

 I'll post the test-tone today.

 atb,
 e


----------



## EliasGwinn

Regarding active speakers, there are a number of different "decor-friendly" models available at different price points. I haven't listened to them, so I can't say which ones I recommend. But here are a few brands to check out:

 Dynaudio, Quad, ATC, Salagar, Meredian

 All the best,
 Elias


----------



## denonfan

In case no one remembers me, I was the one a couple of months ago had some issues with crackling noise with my USB DAC1.

 After almost TWO months of trying to figure out what was causing this crackling noise with my USB DAC1, I finally nailed the culprit. 

 Guess what, with Vista, I had to enable Vista aero interface, in order to get crackling free sound from my USB DAC1. If I choose any other theme other than Vista aero, I get the crackling noise. Simple as that. And I found out about this AFTER replacing the entire motherboard, graphics card, hard drive, RAM, LCD inverter, etc.

 Man, some combination of driver, my computer hardware, and Vista must be the cause of this conflict.

 But I am glad it's all sorted out. I am really liking the sound.


----------



## EliasGwinn

Quote:


  Originally Posted by *denonfan* /img/forum/go_quote.gif 
_In case no one remembers me, I was the one a couple of months ago had some issues with crackling noise with my USB DAC1.

 After almost TWO months of trying to figure out what was causing this crackling noise with my USB DAC1, I finally nailed the culprit. 

 Guess what, with Vista, I had to enable Vista aero interface, in order to get crackling free sound from my USB DAC1. If I choose any other theme other than Vista aero, I get the crackling noise. Simple as that. And I found out about this AFTER replacing the entire motherboard, graphics card, hard drive, RAM, LCD inverter, etc.

 Man, some combination of driver, my computer hardware, and Vista must be the cause of this conflict.

 But I am glad it's all sorted out. I am really liking the sound._

 

Bizarre...but I'm glad you found the culprit!! Enjoy!

 All the best,
 Elias


----------



## EliasGwinn

Another manufacturer of AMAZING active monitors that are also asthetically pleasing...

 Focal

 (Focal JMlab enceintes acoustiques haute fidélité haut-parleurs voiture audiomobile Car audio tuning, Focal Professional studio monitor loudspeakers)

 Especially the Twin6 (<$3200/pair), but I'd guess their less expensive models are great too.


----------



## Shadorne

Quote:


  Originally Posted by *EliasGwinn* /img/forum/go_quote.gif 
_Another manufacturer of AMAZING active monitors that are also asthetically pleasing...

 Focal Especially the Twin6 (<$3200/pair), but I'd guess their less expensive models are great too._

 

Good one! They are a keeper for sure. Unbeatable value - where else can you get a passive speaker as well as 400 WATTS of amplification for that combined price new! 
	

	
	
		
		

		
			





 (Not to mention Active designs are superior with less IMD, better crossover design and better phase/impulse response)

 Personally, I find the metal tweets a bit splashy or in your face but that is just me, and at that price point, the Twin6 is a Home Run!


----------



## G-U-E-S-T

Hey Elias (and gang),

 I will definitely have to try those Focals - thanks for the reference! May I ask again please: With active speakers, does the previous advice that one's regular listening levels should reach up to around 3:00 on the DAC1 volume dial, still hold?

 Also, is anybody aware of ANY standalone solidstate stereo amp such as Elias describes for me - i.e. has anybody ever seen one that (for SNR reasons) doesn't utilize resistors/pots/input-transformers at its inputs, has an input sensitivity just under 3 volts (rms), and outputs only up to about 100 watts? All that I've found, have input sensitivities *much* lower than that. 

 I would really like to try Elias' advice - if this is actually possible!


----------



## Shadorne

Quote:


  Originally Posted by *G-U-E-S-T* /img/forum/go_quote.gif 
_Hey Elias (and gang),

 I will definitely have to try those Focals - thanks for the reference! May I ask again please: With active speakers, does the previous advice that one's regular listening levels should reach up to around 3:00 on the DAC1 volume dial, still hold?_

 

The Focal Twin 6 Active have an adjustment for input sensitivity so you can connect up pro audio levels +4 dbU (1.2 volts) versus consumer RCA which uses as reference -10 dbV (.3 volts). If you set the Focals at pro levels you will naturally have the volume setting higher on whatever preamp you use.

 Frankly I would not worry too much about amp input attenuation resistors that are switched in - a potentiometer is more of a worry but on a good quality product all it means is that L and R level matching might not be perfect unless you measure with a precision microphone. (pros will care about this but consumers need not worry)

 Noise from a resistor attenuator in a good product will be several orders of magnitude lower than electronics. I think the point is to keep the signal level reasonably high in electronics and interconnect cabling.

 FWIW: You can agonize over this but unless you have a speaker capable of 120 db SPL then you are NEVER going to get full dynamic range from your system - the noise floor in most quiet places is around 30 db spl so if peaks are 120 db SPL (limit of human hearing) then that gives you 90 db dynamic range or basically as much as is on the handful of near perfectly recorded CD's that might exist (although the room reverb may limit your practical dynamic range on certain busy kinds of music as it high spl's will overload the room). If you play soft music at max 70 db spl peaks then your dynamic range is limited to 40 db SPL in a great space (although your brain/ear can work out some sustained notes at mid range frequencies below the noise floor). In any case you will be in luck because most modern music is compressed heavily and you are lucky to get more than even 10 db dynamic range on many pop CD's...google loudness wars if you don't believe the recording industry would be so insane as to develop a media (redbook CD) with 96 db dynamic range and then squash all the music they offer into a few bits (10 db)...it is maddening but that is what artists demand and recording engineers do.


----------



## DeusEx

I'm considering the Stello DA100, Benchmark DAC1, and Bel Canto DAC3 for a new DAC. The Stellos~$700, DAC1's ~$1K, and Bel Canto~$2K. I know it's a wide range, but I want something that has a good price-vs-performance ratio, as well as something I can use as a stepping stone to high-end gear (which pretty much means it'll last me a couple of yrs..)

 Any advice? It's going to be used with Energy RC-10 bookshelves (and a possible Dynaudio upgrade), as well as with high-end headphones (either D7000 or HD800, in the future)


----------



## Quaddy

the dac1 is the more interchangeable system imo, i mean in the fact that it has far more ins and outs and analogue connections than most other dacs on the market no matter what their pricepoint!

 from that point of view, the dac1 has been the one remaining, original bit of kit i have not sold or traded - for me that speaks volumes, no pun intended.

 i also feel its the most customisable unit, numerous times, have i been able to change jumpers and gain to match with whatever headphone or equipment was passsing through, and as you mention it, if its price performance ratio and a keeper you are after, i cant recommend the dac1 range enough, i have the pre model to be specific.


----------



## Quaddy

elias, kind of a trivial question...

 after the dac1 pre has been powered off (unpluged from mains) and then with reconnection to the power it defaults to the analog input (shown via the blue led on front panel)

 is there any way to change this default?, is there a jumper or similar, ideally would prefer it to come onto my main source which is USB

 thanks


----------



## MarkyMark

Anyone impressions on how the HD800 is driven by the HPA2 circuit?


----------



## Quaddy

drives them fine for my liking, nice and down the middle, no peak or trough colouring just accurate and plain just how i like it

 in terms of juice, the HPA2 is powering them heartily, at 10-11 oclock on the dial thats as much as i want to go thank you.

 at the end of the day people either like or dislike the HPA-2 and thats how it will be with HD800 jacked in

  Quote:


  Originally Posted by *MarkyMark* /img/forum/go_quote.gif 
_Anyone impressions on how the HD800 is driven by the HPA2 circuit?_


----------



## MarkyMark

Cheers Quaddy!


----------



## hh83917

Quote:


  Originally Posted by *Quaddy* /img/forum/go_quote.gif 
_drives them fine for my liking, nice and down the middle, no peak or trough colouring just accurate and plain just how i like it

 in terms of juice, the HPA2 is powering them heartily, at 10-11 oclock on the dial thats as much as i want to go thank you.

 at the end of the day people either like or dislike the HPA-2 and thats how it will be with HD800 jacked in_

 

I agree with Quaddy as well. I recently got my HD800 and have been enjoying them very much through the DAC1 HDR. 

 I used to have HD650 with Zu Mobius cable on balanced output. Now I preferred the HD800 over the HD650 even it is still currently running unbalanced. The HD800 stock cable is much much better than the HD650's. DAC1 provided HD800 with a very clean and detailed sound that is very much to my liking as well. The soundstage is amazing on the HD800 and the DAC1's clean background re-enforces that spaciousness. 

 The HD650s sounds like you sit at the first row of the concert with heavy bass and somewhat small soundstage, which is a bit close for me. The HD800s sounds like you are in the middle of the concert hall with music surrounding you due to the new large vibrating elements. 

 Just like Quaddy mentioned, people here either like or dislike the HPA2. But, I personally felt it's a pretty powerful headphone amp and drives almost everything well. Kudos to Benchmark for providing such a well-rounded and flexible DAC. 
	

	
	
		
		

		
		
	


	




 Howard.


----------



## peanuthead

Sorry if this question has been asked before, but how do you connect the DAC1 to a headphone amp - through calibrated or variable output? Is one better than the other?
 Also, do you use the volume pot on the amp or the DAC1 for adjusting headphone volume? Thanks!


----------



## EliasGwinn

Hi all, I'm sorry I haven't posted in a while.

 I want to reply to all of these today...hopefully I'll have a chance to.

 All the best,
 Elias


----------



## EliasGwinn

Quote:


  Originally Posted by *peanuthead* /img/forum/go_quote.gif 
_Sorry if this question has been asked before, but how do you connect the DAC1 to a headphone amp - through calibrated or variable output? Is one better than the other?
 Also, do you use the volume pot on the amp or the DAC1 for adjusting headphone volume? Thanks!_

 

Typically you would want to send the highest amplitude signal that the next device can handle. In other words, if the headphone amp can take a maximum of 2 volt input, you would want to configure the source (the DAC1 USB) to send a signal up to, but no higher then 2 volts.

 However, there are exceptions, such as if the other volume control is passive. Passive volume controls present a major pit-fall because their impedance is always changing, which will change the response of the system based on volume control. In this case, you'd be better off leaving the passive volume control all the way open and use the DAC1's volume control.

 You'll want to speak with the manufacturer of the headphone amp to determine the maximum input level and volume control implementation. Let me know what they say.

 Thanks,
 Elias


----------



## peanuthead

So how would I go about setting the DAC1 to output at 2 volts? And would this be through the calibrated (fixed?) or variable mode?

 If I was going to connect the DAC1 to my Headamp GS-1 amp, how would you suggest I do it? Thanks.


  Quote:


  Originally Posted by *EliasGwinn* /img/forum/go_quote.gif 
_Typically you would want to send the highest amplitude signal that the next device can handle. In other words, if the headphone amp can take a maximum of 2 volt input, you would want to configure the source (the DAC1 USB) to send a signal up to, but no higher then 2 volts.

 However, there are exceptions, such as if the other volume control is passive. Passive volume controls present a major pit-fall because their impedance is always changing, which will change the response of the system based on volume control. In this case, you'd be better off leaving the passive volume control all the way open and use the DAC1's volume control.

 You'll want to speak with the manufacturer of the headphone amp to determine the maximum input level and volume control implementation. Let me know what they say.

 Thanks,
 Elias_


----------



## EliasGwinn

Quote:


  Originally Posted by *peanuthead* /img/forum/go_quote.gif 
_So how would I go about setting the DAC1 to output at 2 volts? And would this be through the calibrated (fixed?) or variable mode?

 If I was going to connect the DAC1 to my Headamp GS-1 amp, how would you suggest I do it? Thanks._

 

Contact the manufacturer and find out the maximum input level and volume control implementation (passive? IC? digital? gain circuit?). When you know these things, we can determine the proper way to connect everything. 

 All the best,
 Elias


----------



## EliasGwinn

Quote:


  Originally Posted by *G-U-E-S-T* /img/forum/go_quote.gif 
_I will definitely have to try those Focals - thanks for the reference! May I ask again please: With active speakers, does the previous advice that one's regular listening levels should reach up to around 3:00 on the DAC1 volume dial, still hold?_

 

Yes...between 11 o'clock and 3 o'clock is the ideal.

 All the best,
 Elias


----------



## EliasGwinn

Quote:


  Originally Posted by *Quaddy* /img/forum/go_quote.gif 
_elias, kind of a trivial question...

 after the dac1 pre has been powered off (unpluged from mains) and then with reconnection to the power it defaults to the analog input (shown via the blue led on front panel)

 is there any way to change this default?, is there a jumper or similar, ideally would prefer it to come onto my main source which is USB

 thanks_

 

The DAC1 PRE remembers the input that was being used when it was powered down for 10-20 minutes. After that, it will default to the analog input.

 Best,
 Elias


----------



## hpz

Hi Elias,

 I have just purchased a DAC1 USB and i am using them with IEMs via the headphone outputs. Is there anyway to lower the gain even further than level -10db? 

 I'm only using about 4-5 ticks up (8 o clock) and its already loud enough, but the sound is imbalanced and unclear.

 Anyway to reduce the gain even more?

 Any help would be appreciated
 Thanks


----------



## little-endian

Hi Elias,

 after some time again I want to revive two questions I had before and add another one:


*1. DAC1 inside view*

 Weren't you going to investigate why the inside views disappeared on Benchmark's web site?


*2. Volume control*

 Here and in the feedback newsletter you explained how and why DSPs reduce the SNR aka dynamic range compared to volume pots.

 I don't doubt that a second however it is still not totally clear to me why this gives any drawback in real-life practice:

 You said a DSP controls the values before the D/A conversion. Lower values mean a lower SNR by definition. A volume pot might lower the noise as well, keeping the SNR by definition, isn't it? But if one listens at lower levels for comfortable listening, reaching ~ 70 dB(A) at maximum for instance, don't my ears limit the SNR / dynamic range as well? Aren't the ears comparable to the DAC - lower "values" (or sound pressures) increase the "wasted" headroom? From hearing threshold level to maximum there might be just 40-50 dB left (due to environment noise). Lowering the volume, wouldn't the dynamic range decrease due to hearing capability anyway - regardless of the technique used to lower the volume?

 Asked another way: Assuming I have music playing with an approximate dynamic range of 30 dB (this should be realistic for recordings beyond the loudness war; early 80s for instance) - one time with the DAC1, the other with an AVR using a DSP for volume control for instance.

 In both cases I set the volume so the loudest parts reach 80 dB(A) (and the quietest 50 dB (A) respectively). What exactly will I gain in regard to SNR or dynamic range when listening to the DAC1 compared to the AVR?

 Of course - by definition, the noise will be slightly higher with the AVR since the DAC's full scale isn't entirely used anymore but wouldn't that introduced error below the ~ 50 dB of the quietest parts in the music anyway - being masked so to speak?


*3. AVRs*

 I know you avoid taking position when it comes to competitive products for good reason but what do you think in general about most AV-receivers. As far as I know almost all use DSPs to adjust the volume, aren't they? Are 32-bit DSPs common or something else?


 Maybe it would be no mistake to create an own thread "SNR / dynamic range" to discuss that for all in more detail.


----------



## Shadorne

Quote:


  Originally Posted by *little-endian* /img/forum/go_quote.gif 
_*3. AVRs*

 I know you avoid taking position when it comes to competitive products for good reason but what do you think in general about most AV-receivers. As far as I know almost all use DSPs to adjust the volume, aren't they? Are 32-bit DSPs common or something else?


 Maybe it would be no mistake to create an own thread "SNR / dynamic range" to discuss that for all in more detail._

 

My Anthem AVM20 uses a crystal 3310 chip (matched resistor array on a chip) - these are pretty good at volume control ( I am not saying they are by any means perfect but the specs ain't bad) 
	

	
	
		
		

		
			





 Complete Digital Volume Control
 — 2 Independent Channels
 — Serial Control
 — 0.5 dB Step Size
 Wide Adjustable Range
 — -95.5 dB Attenuation
 — +31.5 dB Gain
 Low Distortion & Noise
 — 0.001% THD+N
 — 116 dB Dynamic Range
 Noise Free Level Transitions
 Channel-to-Channel Crosstalk Better Than
 110 dB

 So I guess the answer is it all depends....what kind of DSP or AVR...


----------



## wavoman

We have discussed here from time-to-time Benchmark's "UltraLock", and posters have asked if it is more than just re-clocking on the DAC side. The Benchmark man, EG, says "yes, much more".

 I now have my own proof. Through some poor re-cabling and SPDIF signal splitting, I have (without meaning to) introduced a high degree of jitter into my SPDIF chain.

 One of my DACs, which does re-clock for sure, can no longer reliably sync with the SPDIF signal. The manufacturer says it is the jitter I have introduced. Even though his unit re-clocks, the SPDIF receiver circuit itself cannot "lock on", the SPDIF jitter is so bad. He said "Benchmark's UltraLock can do this" ... and indeed it can. I plugged my Benchmark DAC-1 in instead, and it sync'd right away. Sounded great!

 There is no mistaking this -- I plugged in one, then the other -- back and forth ... the non-Benchmark would lock-on sometimes and then suffer audio drop-outs, other times just display "No Data". But the Benchmark DAC-1 was rock solid every time!!


----------



## Quaddy

/\ good stuff, i always had the feeling and with experience of their product that the dac1 and benchmark in general really dont dabble in snake oil and fluffy vague talk.

 i always have seen there products to be solid, reliable, highly practical and more heeled in the realm of professional type audio where everything needs to be as it says on the tin, not like these boutique type products where the language can get very fancy and very meaningless real quick


----------



## JamieGreen

First I have to say that I like the Benchmark DAC1 pre with my IEM (Wum3x) very much with 20 dB gain reduction. The question is if I can set one headphone jack -20dB and second left with -10dB for my another 300Ohm headphones?

 Thanks


----------



## wavoman

Quote:


  Originally Posted by *Quaddy* /img/forum/go_quote.gif 
_...heeled in the realm of professional type audio ..._

 

Yea that's it exactly. It is both pro gear, _and_ well suited for home use. Not a lot of manufacturers fit so well in to both worlds. The other DAC is extremely high-end audiophile, but simply can't tolerate the cable length and patch panels I use, at least as far as its SPDIF input module is concerned. 

 If I used AES/EBU everywhere, with tight-tolerance 110 ohm connectors for the SPDIF I would be fine, but like everyone I am lazy and sloppy -- I am re-using 75-ohm composite video wiring. Still, the Benchmark DAC-1 locks on no problem.


----------



## little-endian

Quote:


  Originally Posted by *wavoman* /img/forum/go_quote.gif 
_One of my DACs, which does re-clock for sure, can no longer reliably sync with the SPDIF signal. The manufacturer says it is the jitter I have introduced. Even though his unit re-clocks, the SPDIF receiver circuit itself cannot "lock on", the SPDIF jitter is so bad. He said "Benchmark's UltraLock can do this" ... and indeed it can. I plugged my Benchmark DAC-1 in instead, and it sync'd right away. Sounded great!_

 

Yeah, I think that would be my forth point for Elias. 
	

	
	
		
		

		
		
	


	




 It is interesting - I recognized that my Harman/Kardon AVR for example outputs a distored audio signal if the S/PDIF is too bad while the Benchmark DAC1 plays the same source flawlessly and never distorts but mutes when in doubt.

 The question is if the jitter really increases with cable length or just the attenuation.

 It would be interesting to know what causes the different input sesitivity of different devices.


----------



## Shadorne

Quote:


  Originally Posted by *little-endian* /img/forum/go_quote.gif 
_The question is if the jitter really increases with cable length or just the attenuation._

 

Yes. Cable length is a factor.


----------



## wavoman

I did a bunch of experiments tonight, and the Benchmark UltraLock is amazing.

 My SPDIF source was a Wadia iTransport. I ran the SPDIF to a 75-ohm RCA patch panel, and then into either a Wadia DAC or the Benchmark. Both locked on at 44.1 flawlessly. 

 Then I patched other units in the middle of the chain. First a Sonifex (pro gear) Redbox SPDIF repeater. I only plugged one thing into it, not multiple. The Benchmark locked on, no problem. The DAC would lock on, then lose it, etc. -- lots of drop outs.

 Then I took the Sonifex out, and put in a SPDIF upsampler, the Monarch DIP. This is also supposed to be a jitter-reduction device. Again, Benchmark had no problem. But the Wadia would not sync. I am testing these one-at-a-time, not splitting the signal.

 Very impressive performance for the Benchmark. Cable length, jitter injection, etc. are all real-world problems if you have a complex set-up. I am now a huge UltraLock fan!

 I came out of the Benchmark balanced in to a Stax 717, and listened with Lambda Sigs, and also Baby Orpheus with a moon-audio adapter. Wonderful wonderful SQ (the iPod in the Wadia iTransport had all uncompressed WAV files).


----------



## EliasGwinn

Quote:


  Originally Posted by *hpz* /img/forum/go_quote.gif 
_Hi Elias,

 I have just purchased a DAC1 USB and i am using them with IEMs via the headphone outputs. Is there anyway to lower the gain even further than level -10db? 

 I'm only using about 4-5 ticks up (8 o clock) and its already loud enough, but the sound is imbalanced and unclear.

 Anyway to reduce the gain even more?

 Any help would be appreciated
 Thanks_

 


 hpz,

 The HPA2 headphone amplifier in the DAC1 USB has two gain ranges...normal and '-10 dB'. In the DAC1 PRE and DAC1 HDR, the HPA2 has an additional lower gain range (-20 dB). 

 IEM's often have very low impedance and very high sensitivity. The HPA2 is a very powerful headphone amplifier, so it doesn't surprise me that your headphones are too sensitive for the HPA2. You may find the additional gain ranges of the DAC1 PRE and DAC1 HDR more suitable for your headphones.

 All the best,
 Elias


----------



## G-U-E-S-T

Well, it wasn't easy - but I finally located a great solid-state stereo amp (not class-D or tube) with excellent specifications, and that produces less than 30 watts with an input sensitivity greater than 1.5 volts.

 As always, Elias is correct - this has in fact made a very noticeable and quite nice improvement to our overall sound. Since we only listen at low/medium levels through terrific bookshelf speakers with a powered subwoofer, we might not even normally be using half the power of this much less powerful amp.

 Thanks Elias for your advice about maintaining a good signal-to-noise ratio, by not using an amp with "too much" power for normal listening levels. I'm thinking that like myself, many people probably buy a way-overpowered amp for their needs (which seems to be the common sales pitch these days), thus actually reducing the quality of their sound.

 Also, since it's not that easy to actually locate such an amp as described above (class A/AB, great specs, less than 30w, input sensitivity > 1.5v), perhaps Benchmark would consider actually adding such an item to their lineup?


----------



## lamikeith

Quote:


  Originally Posted by *G-U-E-S-T* /img/forum/go_quote.gif 
_Well, it wasn't easy - but I finally located a great solid-state stereo amp (not class-D or tube) with excellent specifications, and that produces less than 30 watts with an input sensitivity greater than 1.5 volts._

 

Please, tell us the brand and model of the amp so that we all may benefit from your research.


----------



## Shadorne

Quote:


  Originally Posted by *G-U-E-S-T* /img/forum/go_quote.gif 
_Well, it wasn't easy - but I finally located a great solid-state stereo amp (not class-D or tube) with excellent specifications, and that produces less than 30 watts with an input sensitivity greater than 1.5 volts.

 As always, Elias is correct - this has in fact made a very noticeable and quite nice improvement to our overall sound. Since we only listen at low/medium levels through terrific bookshelf speakers with a powered subwoofer, we might not even normally be using half the power of this much less powerful amp.

 Thanks Elias for your advice about maintaining a good signal-to-noise ratio, by not using an amp with "too much" power for normal listening levels. I'm thinking that like myself, many people probably buy a way-overpowered amp for their needs (which seems to be the common sales pitch these days), thus actually reducing the quality of their sound.

 Also, since it's not that easy to actually locate such an amp as described above (class A/AB, great specs, less than 30w, input sensitivity > 1.5v), perhaps Benchmark would consider actually adding such an item to their lineup? 
	

	
	
		
		

		
		
	


	


_

 

FANTASTIC - you must be delighted.

 What "terrific bookshelves" and what amplifier?

 Also "good signal to noise" is not that hard to achieve - you make it sound like it was some nigh impossible quest. Use good balanced XLR equipment with ordinary "Mogami" interconnects and plug everything into the same outlet with no cheater plugs and you should be done in most cases. I wonder if you did not have a ground loop issue (all too common with RCA where negative signal wire and ground are one and the same)? Or perhaps the simple answer might be that something else was wrong with your previous amplifier? All I am suggesting is that there might be more to it before jumping to conclusions (other than this fixed your problem which is great news).

 For example, do not confuse "way-overpowered" amplifier (and assume that most people are making a mistake by having a powerful amplifier) with input sensitivity and gain - the difference between 30 watts and 300 watts is not that significant....it will only be perceived as twice as loud - we are talking a mere 10 decibels here (it is the sensitivities/gains that count not the overall power capability). The most "common" error in amplifier selection is a poor match to speakers: either impedance match issues or the amp is underpowered or overpowered for the speaker. I simply don't think you can jump to the conclusion that you have stumbled on something that is a problem for most people with powerful amplifiers...


----------



## EliasGwinn

Quote:


  Originally Posted by *little-endian* /img/forum/go_quote.gif 
_*1. DAC1 inside view*

 Weren't you going to investigate why the inside views disappeared on Benchmark's web site?_

 

Here is the internal view of the DAC1 USB:

http://www.benchmarkmedia.com/system...-rm_inside.jpg

 Here is the internal view of the DAC1 HDR:

http://www.benchmarkmedia.com/system...hdr-inside.jpg

  Quote:


  Originally Posted by *little-endian* /img/forum/go_quote.gif 
_*2. Volume control*

 Here and in the feedback newsletter you explained how and why DSPs reduce the SNR aka dynamic range compared to volume pots.

 I don't doubt that a second however it is still not totally clear to me why this gives any drawback in real-life practice:

 You said a DSP controls the values before the D/A conversion. Lower values mean a lower SNR by definition. A volume pot might lower the noise as well, keeping the SNR by definition, isn't it? But if one listens at lower levels for comfortable listening, reaching ~ 70 dB(A) at maximum for instance, don't my ears limit the SNR / dynamic range as well? Aren't the ears comparable to the DAC - lower "values" (or sound pressures) increase the "wasted" headroom? From hearing threshold level to maximum there might be just 40-50 dB left (due to environment noise). Lowering the volume, wouldn't the dynamic range decrease due to hearing capability anyway - regardless of the technique used to lower the volume?

 Asked another way: Assuming I have music playing with an approximate dynamic range of 30 dB (this should be realistic for recordings beyond the loudness war; early 80s for instance) - one time with the DAC1, the other with an AVR using a DSP for volume control for instance.

 In both cases I set the volume so the loudest parts reach 80 dB(A) (and the quietest 50 dB (A) respectively). What exactly will I gain in regard to SNR or dynamic range when listening to the DAC1 compared to the AVR?

 Of course - by definition, the noise will be slightly higher with the AVR since the DAC's full scale isn't entirely used anymore but wouldn't that introduced error below the ~ 50 dB of the quietest parts in the music anyway - being masked so to speak?_

 

Don't confuse dynamic range of the music to the dynamic range of the audio electronics. Even if the dynamic range of the music is 10 dB, you can still hear noise well below that. 

 Noise can only be masked by more noise, not a collection of tones. If there is enough tones in enough different frequencies at sufficient amplitudes, it may approach noise and create a masking affect. But a violin recording at -2 dBFS will not mask noise at -80 dBFS. 

 With that being said, other factors may limit the signal-to-noise of your listening experience (HVAC noise, your ears ringing, etc). However, many people have very quiet listening environments that benefit from a maximized electronic signal-to-noise ratio.

  Quote:


  Originally Posted by *little-endian* /img/forum/go_quote.gif 
_*3. AVRs*

 I know you avoid taking position when it comes to competitive products for good reason but what do you think in general about most AV-receivers. As far as I know almost all use DSPs to adjust the volume, aren't they? Are 32-bit DSPs common or something else?
_

 

I don't have enough information to say with authority what DSP chips most AVR's use. However, it doesn't change the fact that the D/A chip will still have a fixed noise floor that will be 'closer' to the signal when a lower digital signal is presented. 

 All the best,
 Elias


----------



## EliasGwinn

Quote:


  Originally Posted by *JamieGreen* /img/forum/go_quote.gif 
_First I have to say that I like the Benchmark DAC1 pre with my IEM (Wum3x) very much with 20 dB gain reduction. The question is if I can set one headphone jack -20dB and second left with -10dB for my another 300Ohm headphones?

 Thanks_

 

Hello JamieGreen,

 I'm sorry I haven't addressed your question sooner. 

 The short answer is... no. Both headphone jacks use the same HPA2 headphone amplifier. The gain setting on that HPA2 affects both jacks equally.

 All the best,
 Elias


----------



## EliasGwinn

Quote:


  Originally Posted by *little-endian* /img/forum/go_quote.gif 
_Yeah, I think that would be my forth point for Elias. 
	

	
	
		
		

		
		
	


	




 It is interesting - I recognized that my Harman/Kardon AVR for example outputs a distored audio signal if the S/PDIF is too bad while the Benchmark DAC1 plays the same source flawlessly and never distorts but mutes when in doubt.

 The question is if the jitter really increases with cable length or just the attenuation.

 It would be interesting to know what causes the different input sesitivity of different devices._

 

Cable lenght absolutely affects jitter levels.

 The thing that affects input jitter sensitivity is the clock-recovery mechanism. We have demonstrated that the DAC1 can maintain its fully-rated performance even with severe levels of jitter on the line.

 All the best,
 Elias


----------



## rmh1

Hey Elias,

 Looks like the manual for the HDR needs an update. The section on changing the jumpers is the old one from the PRE manual, pages 15 and 16. You need to change the header descriptions and photos on pages 18 and 19 of the HDR manual to indicate the new header numbers. 

 Here is the internal view of the DAC1 HDR:

http://www.benchmarkmedia.com/system...hdr-inside.jpg


----------



## mahesh

After reading 180 pages,i orderd dac 1 usb and got it yesterday,
 its still burning,i am using adam a5 and sennheiser hd 25 ii. Till now iam satisfied but I will try to give my impression later,thank you Elias and everyone for helping to choose dac1.
 Sorry for my bad english


----------



## mahesh




----------



## EliasGwinn

@ mahesh:


----------



## pekingduck

Hi Elias,

 Any chance of changing the HPA2's gain jumpers to a front panel switch in the next generation DAC1?

 As a potential buyer I find it quite troublesome having to open the chassis everytime to change the gain, especially for people having headphones of widely varying sensitivities.

 Thanks


----------



## little-endian

Quote:


  Originally Posted by *EliasGwinn* /img/forum/go_quote.gif 
_Here is the internal view of the DAC1 USB:

http://www.benchmarkmedia.com/system...-rm_inside.jpg

 Here is the internal view of the DAC1 HDR:

http://www.benchmarkmedia.com/system...hdr-inside.jpg_

 

Thanks for the update Elias. I knew everything was fine, hehe.

  Quote:


  Originally Posted by *EliasGwinn* /img/forum/go_quote.gif 
_Don't confuse dynamic range of the music to the dynamic range of the audio electronics. Even if the dynamic range of the music is 10 dB, you can still hear noise well below that.

 Noise can only be masked by more noise, not a collection of tones. If there is enough tones in enough different frequencies at sufficient amplitudes, it may approach noise and create a masking affect. But a violin recording at -2 dBFS will not mask noise at -80 dBFS._

 

Yes I understand what you mean. This is one part, lossy codecs make use of, isn't it? Similarly, dithering still allows details to be resolved despite its introduced noise floor, right? 

  Quote:


  Originally Posted by *EliasGwinn* /img/forum/go_quote.gif 
_With that being said, other factors may limit the signal-to-noise of your listening experience (HVAC noise, your ears ringing, etc). However, many people have very quiet listening environments that benefit from a maximized electronic signal-to-noise ratio._

 

Yeah, okay let's assume one has optimal listening conditions - no background noise, a high SNR of the electronic devices, etc ...

  Quote:


  Originally Posted by *EliasGwinn* /img/forum/go_quote.gif 
_I don't have enough information to say with authority what DSP chips most AVR's use. However, it doesn't change the fact that the D/A chip will still have a fixed noise floor that will be 'closer' to the signal when a lower digital signal is presented._

 

... won't my hearing capability limit the SNR / dynamic range as well I can enjoy when listening to music at low levels? So lower values given to a DAC (either because the music contains quite passages or the overall volume is lowered by a DSP) mean a lower SNR because the DAC's maximum output will be lower as well while its noise floor will be unchanged. That's clear so far. But a level of 0dBSPL is the absolute minimum to hear something, correct? When I set my volume so that a maximum of let's say 60dBSPL is archived - wouldn't be my SNR / dynamic range 60dB at best anyway due to my ears?

 So I think I understand your explanation why the poti solution the DAC1 uses is superior - I just wonder if the benefit isn't consumed by the limitations of hearing per se. 
	

	
	
		
		

		
		
	


	





  Quote:


  Originally Posted by *EliasGwinn* /img/forum/go_quote.gif 
_The thing that affects input jitter sensitivity is the clock-recovery mechanism. We have demonstrated that the DAC1 can maintain its fully-rated performance even with severe levels of jitter on the line._

 

Yeah, indeed. The performance of the DAC1 on that regard is undisputable and just great. I little hope arises: Does Benchmark plan to develop a device for multichannel PCM by chance? You mentioned that some use several DAC1s to built a multichannel environment but since it's quite a fumble and still involves electronics (the power amplifier), it would be great to have a solution including a DAC and the amplifier with all the Benchmark philosophy.


----------



## Shadorne

Quote:


  Originally Posted by *little-endian* /img/forum/go_quote.gif 
_


 ... won't my hearing capability limit the SNR / dynamic range as well I can enjoy when listening to music at low levels? So lower values given to a DAC (either because the music contains quite passages or the overall volume is lowered by a DSP) mean a lower SNR because the DAC's maximum output will be lower as well while its noise floor will be unchanged. That's clear so far. But a level of 0dBSPL is the absolute minimum to hear something, correct? When I set my volume so that a maximum of let's say 60dBSPL is archived - wouldn't be my SNR / dynamic range 60dB at best anyway due to my ears?_

 

Probably less than 60 db SPL - it is rare to get such a perfectly quiet environment (a whisper is at 20 decibels and a library may be at 30 decibels)

 You need very good speakers to compete with electronics (even ordinary electronics) - few speakers can play cleanly (less than a third of a percent distortion) at 120 db SPL over the full range and therefore give you near real dynamic range of a perfectly recorded highly dynamic CD: 120 db SPL minus 30 db SPL (of quiet environment) = 90 db dynamic range or pretty much the Full Monty as far as your ears and listening environment will allow (a CD can carry 96 db in theory).


----------



## little-endian

Hello Shadorne,

 thanks for your comment. I'm aware that real-life listening-conditions will be much worse in most cases - recordings nowadays often feature a lousy dynamic range (shame on *everyone* following this due to marketing b/s) not to mention the environmental background noise. But since I try to understand Elias' explanation in theory, I want to take that aside for the moment.

 I think I got his point - a DSP will lower the dynamic range by definition because the noise floor is always the same (taking dither aspects aside again) while the maximum output of the DAC will be lower; hence closer to the noise. A solution based on a poti in comparison will lower *both* - the signal as well the noise, maintaing the dynamic range (or SNR) by definition. I hope no mistake from my side so far.

 The point where my doubts and confusion kick in is when it comes to our hearing cabapility:

 Let's say I have music playing with a dynamic range of 30 dB and I want the maximum to played at 80dB(SPL). Let's further say that my DAC (may it be 24-bit) features a SNR of 110dB. To simplify things, I just assume that an input signal of 0dBFS would lead to a sound pressure of 110dB(SPL) if not lowered in volume. Now I could use a DSP (as many AVRs seem to make use of) to lower the values so 0dBFS becomes -30dBFS (to reach an output of 80dB(SPL). Since the noise floor will be almost unchanged (assuming a sufficient dithering is used), the SNR remains at 80dB, meaning a loss of 30dB.

 When a potentiometer after the DAC is used to lower the volume, the input signal to the DAC remains unchanged; resulting to 110dB lowered to 80dB. However, since the poti reduces the noise floor also (then to -30dBSPL), the full SNR of 110dB is maintained, hence no loss. *But* since my hearing is limited to 0dBSPL, the gained SNR of 30dB in this case (from -30dBSPL to 0dbSPL) is lost anyway.

 That's my thought. Now you're welcome to correct me.


----------



## EliasGwinn

Quote:


  Originally Posted by *pekingduck* /img/forum/go_quote.gif 
_Hi Elias,

 Any chance of changing the HPA2's gain jumpers to a front panel switch in the next generation DAC1?

 As a potential buyer I find it quite troublesome having to open the chassis everytime to change the gain, especially for people having headphones of widely varying sensitivities.

 Thanks_

 

Hello Pekingduck,

 Thanks for your suggestion...we will definately take it into consideration.

 For your scenario, I would recommend indefinately removing the chassis screws. You can keep the chassis cover in place without the screws, and simply lift it up when you want to change the HPA2 gain range.

 All the best,
 Elias


----------



## hoibe

Im getting an iPhone for x-mas and I cant wait! How much do you know about the iPhone? Please tell.



 ________________
unlock iphone 3g


----------



## Shadorne

Quote:


  Originally Posted by *little-endian* /img/forum/go_quote.gif 
_Hello Shadorne,

 Let's say I have music playing with a dynamic range of 30 dB and I want the maximum to played at 80dB(SPL). Let's further say that my DAC (may it be 24-bit) features a SNR of 110dB. To simplify things, I just assume that an input signal of 0dBFS would lead to a sound pressure of 110dB(SPL) if not lowered in volume. Now I could use a DSP (as many AVRs seem to make use of) to lower the values so 0dBFS becomes -30dBFS (to reach an output of 80dB(SPL). Since the noise floor will be almost unchanged (assuming a sufficient dithering is used), the SNR remains at 80dB, meaning a loss of 30dB.

 When a potentiometer after the DAC is used to lower the volume, the input signal to the DAC remains unchanged; resulting to 110dB lowered to 80dB. However, since the poti reduces the noise floor also (then to -30dBSPL), the full SNR of 110dB is maintained, hence no loss. *But* since my hearing is limited to 0dBSPL, the gained SNR of 30dB in this case (from -30dBSPL to 0dbSPL) is lost anyway.

 That's my thought. Now you're welcome to correct me._

 

There is a noise floor in all equipment - when you use a pot to attenuate a very small signal you still end up with the inherent noise of the equipment itself. If 110 db SPL is full scale and your S/N for that equipment (typical of extremely high end stuff) then as you lower the signal level with a potentiometer you will still have noise at a level 0 db SPL. You cannot attenuate noise a further 30 db SPL because there is no amplifier that has better than roughly 110 db SPL S/N (which is measured at close to full output).

 Keeping signal levels as high as possible (without clipping) is a much more critical aspect of recording in a studio - this is because sounds that are recorded much too softly may lose some detail below the equipment noise floor and this detail will be lost permanently (details cannot be boosted or easily recovered if they fall far below the noise level or quantization of the ADC).


----------



## rmh1

Elias,

 Page 18 got edited correctly and the titles and pictures on page 19 are now correct. The descriptions on page 19 for changing the jumpers are still wrong though and are copied from the Pre manual with the wrong jumpers described. Sorry to be a pain but...

 Also, I second the request for an external gain control on the next revision. 

  Quote:


  Originally Posted by *rmh1* /img/forum/go_quote.gif 
_..... A suggestion though for future products: An external gain control for headphone output. With people here using such a wide array of headphones, from very low impedance to the sugggested Senns in your package, to even the 600 ohm Beyers, it would be nice to not have to move the jumpers to get the full volume control range for each headphone...._

 

Actually a third, as this is a suggestion I made from January of 2008. Close to the perfect DAC. Nice work.


----------



## little-endian

Hi Shadorne,

 thanks again. I hope, *Elias* will catch up with this as well since he mentioned, DSPs are inferior in gerneral compared to potis when it comes to volume control (reference implementation assumed of course).

 You say one can't attenuate the noise further. Why this? If a poti is claimed to maintain the full dynamic range / SNR, doesn't it *have* to lower the noise floor below 0dBSPL by definition?

 In regard to clipping, I totally agree. But I still prefer a recording 10 dB below full scale instead of one single clipping. It is unbelievable how crappy most of modern productions are being made today. The sad truth is that the user's equipment isn't necessarily the limiting factor anymore but the source. Even a DAC1 is of not much use if the source material is full of clipping and a lousy dynamic range.


----------



## nae45ro

Hi guys. A while ago I was reporting a problem with my Benchamrk as follows > each time I opened the computer I had to unplu and replug the USB cable from the DAC to the computer, otherwise I had no sound. In the meantime, I upgrade d from XP to Windows 7 and now I have another problem. I do have sound when the computer starts but while being in an application (foobar, any game, VLC...), the DAC randomly stops working (the 2 leds are clipping) and sometimes it recovers by itself, other times I have to unplug and the replug the USB cable. It's really annoying since it's happening so often I can't enjoy my music anymore. Anyone knows what the problem can be ? Elias ?


----------



## EliasGwinn

Quote:


  Originally Posted by *Shadorne* /img/forum/go_quote.gif 
_There is a noise floor in all equipment - when you use a pot to attenuate a very small signal you still end up with the inherent noise of the equipment itself. If 110 db SPL is full scale and your S/N for that equipment (typical of extremely high end stuff) then as you lower the signal level with a potentiometer you will still have noise at a level 0 db SPL. You cannot attenuate noise a further 30 db SPL because there is no amplifier that has better than roughly 110 db SPL S/N (which is measured at close to full output).

 Keeping signal levels as high as possible (without clipping) is a much more critical aspect of recording in a studio - this is because sounds that are recorded much too softly may lose some detail below the equipment noise floor and this detail will be lost permanently (details cannot be boosted or easily recovered if they fall far below the noise level or quantization of the ADC)._

 

It is true that the amplifier's noise floor will not go down when you attenuate the DAC1. This is why I suggest not using an amplifier more powerful then you need. For two amplifiers with the same S-N ratio, the less powerful will be quieter.

 However, if your D/A has reduced S-N and your amp applies enough gain to raise that noise above its own noise floor, then the volume control will make a difference. 

 All the best,
 Elias


----------



## EliasGwinn

Quote:


  Originally Posted by *nae45ro* /img/forum/go_quote.gif 
_Hi guys. A while ago I was reporting a problem with my Benchamrk as follows > each time I opened the computer I had to unplu and replug the USB cable from the DAC to the computer, otherwise I had no sound. In the meantime, I upgrade d from XP to Windows 7 and now I have another problem. I do have sound when the computer starts but while being in an application (foobar, any game, VLC...), the DAC randomly stops working (the 2 leds are clipping) and sometimes it recovers by itself, other times I have to unplug and the replug the USB cable. It's really annoying since it's happening so often I can't enjoy my music anymore. Anyone knows what the problem can be ? Elias ?_

 

Are you using a laptop? 

 Do you have any other USB devices connected to the computer?

 Are there any things that cause the problem (when you are running certain programs, etc)?

 Do you have any background software running (anti-virus, etc)? If so, what brand?

 Typically, this happens because the computer is trying to manage resources. In other words, the computer will mandate power and bandwidth limitations. It will automatically shut off certain connections to manage resources.

 Best,
 Elias


----------



## nae45ro

1) I am using a desktop
 2) Other devices connected through USB would be mouse, keyboard and printer (99% of the time turned off) and memory sticks (also rarely used)
 3) The problems generally occurs when a program that has sound is running like foobar or VLC (movies) or different games. If I stop the respective program, sometimes the 2 leds stop clipping and the DAC works again, sometimes not.
 4) Antivirus would be Nod 32

 Computer is a quadcore with 4GB of Ram so I think it would be pretty hard for it to be run out of resources !


----------



## little-endian

Quote:


  Originally Posted by *EliasGwinn* /img/forum/go_quote.gif 
_It is true that the amplifier's noise floor will not go down when you attenuate the DAC1._

 

Why not?

  Quote:


  Originally Posted by *EliasGwinn* /img/forum/go_quote.gif 
_However, if your D/A has reduced S-N and your amp applies enough gain to raise that noise above its own noise floor, then the volume control will make a difference._

 

I would understand that if if have a low level signal (far below full scale) and raise that afterwards with the amplifier. However I still don't get it when it comes to listening at low levels in general (see my postings above).


----------



## EliasGwinn

Quote:


  Originally Posted by *nae45ro* /img/forum/go_quote.gif 
_1) I am using a desktop
 2) Other devices connected through USB would be mouse, keyboard and printer (99% of the time turned off) and memory sticks (also rarely used)
 3) The problems generally occurs when a program that has sound is running like foobar or VLC (movies) or different games. If I stop the respective program, sometimes the 2 leds stop clipping and the DAC works again, sometimes not.
 4) Antivirus would be Nod 32

 Computer is a quadcore with 4GB of Ram so I think it would be pretty hard for it to be run out of resources !_

 

Its not that the computer is running out of resources, but even 'powerful' computers are always 'managing' resources. 

 When you upgraded from XP to Win7, did you keep the same computer? 

 What brand computer is it?

 Have you tried the DAC1 USB on another computer to determine if the same thing happens?

 Thanks,
 Elias


----------



## nae45ro

It's the same computer and it's not a certain brand (it has been made with top components). The problem seems to solved today when I played with the cables. I replaced the interconnects and haven't had the problem today (pretty weird as I've studied the old interconnect and it's in very good shape ) !


----------



## Shadorne

Quote:


  Originally Posted by *EliasGwinn* /img/forum/go_quote.gif 
_ For two amplifiers with the same S-N ratio, the less powerful will be quieter.

 All the best,
 Elias_

 

In theory yes (isolated on a lab bench), however, the noise floor also depends as much on the gain (or input sensitivity as anything else). A more powerful amplifier operating with low input sensitivity (needs a high signal to drive it) may be quieter than a less powerful amplifier operating with high input sensitivity. This is because the high input sensitivity can compromise the S/N or dynamic range from the pre-amp and you end up amplifying the inherent noise floor of the output of the preamp.

 My point is that only 10 db separate normal amplifiers from powerful ones. One cannot assume that absolute maximum "power output rating" will necessarily determine the noise floor. For instance, the design of the input stage (balanced versus RCA) may play a bigger role in the "real world" than whether the amp maximum power is rated 100 watts or 200 watts in to an 8 ohm load. (Ground loops being a significant source of hum or noise floor in the real world)


----------



## hoibe

Is there any difference between a apple iphone and a 3G one? Which one is better? And what's so good about iPhone?



 ________________
unlock iphone 3gs 3.1


----------



## hoibe

I want to buy an iphone converter for mac, which one will support most formats to iphone with good quality. Where can I go to search one? Anyone could help me? Thanks in advance!



 ________________
unlock iphone 3gs


----------



## hoibe

I am planning on getting an iphone, and I want to know how much i should be expecting to pay monthly. I hear that its around $120.



 ________________
unlock iphone 3g


----------



## john11f

I know this forum covers mostly head-fi gear but i also connectan active speakers to my DAC1 Pre.

 i'm on the verg to upgrade my interconnects to my Dynaudio BM6A MKII and is choosing between a Nordost Baldur and the Furutech Evolution.

 Elias, do you have any expert opinion on these cables?

 Thanks in advance!


----------



## EliasGwinn

Quote:


  Originally Posted by *john11f* /img/forum/go_quote.gif 
_I know this forum covers mostly head-fi gear but i also connectan active speakers to my DAC1 Pre.

 i'm on the verg to upgrade my interconnects to my Dynaudio BM6A MKII and is choosing between a Nordost Baldur and the Furutech Evolution.

 Elias, do you have any expert opinion on these cables?

 Thanks in advance!_

 

I'm sorry, but I don't know enough about those models of cables to advise you.

 All the best,
 Elias


----------



## EliasGwinn

Quote:


  Originally Posted by *Shadorne* /img/forum/go_quote.gif 
_In theory yes (isolated on a lab bench), however, the noise floor also depends as much on the gain (or input sensitivity as anything else). A more powerful amplifier operating with low input sensitivity (needs a high signal to drive it) may be quieter than a less powerful amplifier operating with high input sensitivity. This is because the high input sensitivity can compromise the S/N or dynamic range from the pre-amp and you end up amplifying the inherent noise floor of the output of the preamp.

 My point is that only 10 db separate normal amplifiers from powerful ones. One cannot assume that absolute maximum "power output rating" will necessarily determine the noise floor. For instance, the design of the input stage (balanced versus RCA) may play a bigger role in the "real world" than whether the amp maximum power is rated 100 watts or 200 watts in to an 8 ohm load. (Ground loops being a significant source of hum or noise floor in the real world) 
	

	
	
		
		

		
			



_

 

I agree 100%.

 Best,
 Elias


----------



## G-U-E-S-T

Quote:


  Originally Posted by *EliasGwinn* 
_... the ideal amp will drive YOUR speakers to YOUR comfortable listening level when the DAC1's volume control is between 11 o'clock and 3 o'clock..._

 


 Sorry to not check-in for a while - over here we've been enjoying the music! 
	

	
	
		
		

		
		
	


	




 Elias just for the record, I understood what you were saying - for example, I took the above comment of yours to be very clear indeed.

 Elias, your excellent advice has audibly paid off in spades for us - thanks again for explaining how/why finding an amplifier that ideally meets your above-quoted description (which was not easy by the way), is much desireable. Our previous nice amp was very powerful but way too loud for us at the DAC1's optimal volume control positions, e.g. most especially when approaching 3 o'clock on the DAC1 volume dial. I'm absolutely certain that some others in similar positions could also benefit from your advice here, about getting the right-performing amp as you've described, if they get a chance to try it. Either way, for my family, your advice (as always) has really helped - and we keep enjoying it each day. You're the best!


----------



## EliasGwinn

Thanks for the kind words, GUEST. Its always great to know your work is being appreciated.

 All the very best,
 Elias


----------



## urbo73

Hi Elias,

 I just received my DAC1 HDR and will be trying out some K702s in a week. What are the best jumper settings for the headphone amp gain for the K702/K701? I know you said you've used the K701, which is the same headphone. Should I go ahead and set it to A or B or OFF (i.e -10, -20, or 0)?


----------



## EliasGwinn

Quote:


  Originally Posted by *urbo73* /img/forum/go_quote.gif 
_Hi Elias,

 I just received my DAC1 HDR and will be trying out some K702s in a week. What are the best jumper settings for the headphone amp gain for the K702/K701? I know you said you've used the K701, which is the same headphone. Should I go ahead and set it to A or B or OFF (i.e -10, -20, or 0)?_

 

It depends on the type of music you will be playing and your preferred listening levels. The best jumper settings will allow you to achieve a mild-yet-adaquate listening level at 11 o'clock on the volume knob while playing your loudest music. If it is too loud at that position, move from 'A' to 'B'. If you can't get your 'quiet' music loud enough, move from 'A' to 'OFF'. 

 If you have additional questions, please feel free to ask!

 All the best,
 Elias


----------



## urbo73

Thanks Elias. I will see, though I thought it was not source dependant, but more how much mW it would output and what's the correct power for each headphone.

 BTW, I have the DAC1 HDR sitting on a shelf (i.e. not in a rack with open air underneath) and it's very warm even when off! Is that normal?? Why is that? Also when on, how does it ventilate and cool? Weird..


----------



## JMS

Hi Elias,

 I have an old Benchmark DAC1 Rev H. I'm using it to drive a Bryston 4B-SST power amp, and recently I discovered that the XLR connection sounds different from the RCA connection to the power amp (after volume matching). The XLR output sounds distinctly higher, as if passed through a high-pass filter, and to me the RCA output sounds more correct. I don't think it's a cable issue because I've tried different cables of both types. I'm not sure if it's DAC1 or the amp, but I don't have another amp with balanced inputs to test.

 Again, I'm not sure if the issue is with the DAC1, but do you have any intuition as to why there may be a difference?

 Thanks.


----------



## JMS

dupe


----------



## EliasGwinn

Quote:


  Originally Posted by *urbo73* /img/forum/go_quote.gif 
_Thanks Elias. I will see, though I thought it was not source dependant, but more how much mW it would output and what's the correct power for each headphone._

 

It does that to a certain extent, but there is a lot of overlap between the three selections. Its really meant to be a 'volume' adjustment - something that allows each user to set the overall output-level range to optimize the volume potentiometer performance. 

  Quote:


  Originally Posted by *urbo73* /img/forum/go_quote.gif 
_BTW, I have the DAC1 HDR sitting on a shelf (i.e. not in a rack with open air underneath) and it's very warm even when off! Is that normal?? Why is that? Also when on, how does it ventilate and cool? Weird.._

 

It is normal...the voltage regulators are strapped to the chassis, so the entire chassis acts as a heat sink. That is why it feels warm.

 It should feel like a mug of coffee...that is normal. This is true when it is 'On' or in 'Standby'. The internal electronics are never shut-down unless the power to the unit is disconnected.

 All the best,
 Elias


----------



## EliasGwinn

Quote:


  Originally Posted by *JMS* /img/forum/go_quote.gif 
_Hi Elias,

 I have an old Benchmark DAC1 Rev H. I'm using it to drive a Bryston 4B-SST power amp, and recently I discovered that the XLR connection sounds different from the RCA connection to the power amp (after volume matching). The XLR output sounds distinctly higher, as if passed through a high-pass filter, and to me the RCA output sounds more correct. I don't think it's a cable issue because I've tried different cables of both types. I'm not sure if it's DAC1 or the amp, but I don't have another amp with balanced inputs to test.

 Again, I'm not sure if the issue is with the DAC1, but do you have any intuition as to why there may be a difference?

 Thanks._

 

I can only guess, as I've never heard of this before. My guess is that it has something to do with the differences between the designs of the RCA and XLR input stages in the amplifier. Possibly the DC coupling or input transformer of the XLR stage is causing significant low-frequency attenuation. 

 The XLR and RCA outputs on the DAC1 use the exact same signal source and components.

 All the best,
 Elias


----------



## urbo73

Quote:


  Originally Posted by *EliasGwinn* /img/forum/go_quote.gif 
_It is normal...the voltage regulators are strapped to the chassis, so the entire chassis acts as a heat sink. That is why it feels warm.

 It should feel like a mug of coffee...that is normal. This is true when it is 'On' or in 'Standby'. The internal electronics are never shut-down unless the power to the unit is disconnected.

 All the best,
 Elias_

 

So it's ok to just leave it on standby when not using it? No need to disconnect in order to have it cool down once in a while? Built to last?


----------



## john11f

Hi Elias,
 you recommend the Senn HD650 for the DAC1 pre. How about the HD800?


----------



## Quaddy

i am not voiding elias answering, but just adding my take.... my 800s are fine here with the pre, same impedance as the 650's, its fairs pretty well and the headphone stage of the pre handles the power requirements in its stride, its a lean sound with the pre pairing, how i like it, but if are overtly bass hungry, it will not please


----------



## EliasGwinn

Quote:


  Originally Posted by *urbo73* /img/forum/go_quote.gif 
_So it's ok to just leave it on standby when not using it? No need to disconnect in order to have it cool down once in a while? Built to last? 
	

	
	
		
		

		
		
	


	


_

 

Correct! 

 All the best,
 Elias


----------



## EliasGwinn

Quote:


  Originally Posted by *john11f* /img/forum/go_quote.gif 
_Hi Elias,
 you recommend the Senn HD650 for the DAC1 pre. How about the HD800?_

 

I wish I could answer, but I haven't heard the HD800's. Sorry...

 Best,
 Elias


----------



## urbo73

Quote:


  Originally Posted by *EliasGwinn* /img/forum/go_quote.gif 
_It depends on the type of music you will be playing and your preferred listening levels. The best jumper settings will allow you to achieve a mild-yet-adaquate listening level at 11 o'clock on the volume knob while playing your loudest music. If it is too loud at that position, move from 'A' to 'B'. If you can't get your 'quiet' music loud enough, move from 'A' to 'OFF'. 

 If you have additional questions, please feel free to ask!

 All the best,
 Elias_

 

OK, perhaps this is too simple, or lost on me a bit. Bear with me 
	

	
	
		
		

		
		
	


	




 I just got my 702s today, and I'm using the DAC1 HDR as is - i.e. assuming it's at A when shipped, since I have not opened the box up to look.

 I listened to a wide variety of music (via a CD transport) and all sounds really great. Fantastic product so far! More on the DAC later, but all systems check so far 
	

	
	
		
		

		
		
	


	




 On some CDs, a good *loud* volume was around 12 o'clock - these are the more pop/rock loud-produced CDs. On others (older/progressive rock, classical, & jazz) 2-3 o'clock was needed for a good *loud* volume. Also these have more dynamic range, so it makes sense. OK, so that's all fine and dandy! That's why there's a volume control 
	

	
	
		
		

		
		
	


	




 But my confusion is, how far do I want to (or don't want to) go with the volume control? In other words, all these positions are confusing me, when I bet it's simple. What is the ideal range on the "clock"? In what range is it most efficient? I don't want to under or over-drive it. Is this making sense? In other words, is it bad to go past 3 o'clock? Does it even matter? That sort of thing. I guess the 3 settings have me all confused on this simple matter. Is it better to have volume in the lower range of the control or the higher?


----------



## EliasGwinn

Quote:


  Originally Posted by *urbo73* /img/forum/go_quote.gif 
_On some CDs, a good *loud* volume was around 12 o'clock - these are the more pop/rock loud-produced CDs. On others (older/progressive rock, classical, & jazz) 2-3 o'clock was needed for a good *loud* volume. Also these have more dynamic range, so it makes sense. OK, so that's all fine and dandy! That's why there's a volume control 
	

	
	
		
		

		
		
	


	


_

 

This sounds like your settings are perfect! The ideal range of the volume control is 11 o'clock to 3 o'clock. If you are always in that range, I wouldn't change a thing!

 All the best,
 Elias


----------



## jpelg

Been meaning to post this question for a while now, and thought this thread is as good a place to elicit a response as any. 

 In the October 2009 issue of The Absolute Sound, John Siau responds to Alan Taffel's column reviewing the DAC1-PRE. One of Taffel's points was his observation of the DAC1 having different results with different transports, despite a relatively long history from both Benchmark & many users touting that such differences are greatly minimized (if not _completely_ eradicated) with the DAC1.

 Mr. Siau describes some recent findings at Benchmark claiming that both CD & DVD transports not only "produce non-recoverable read errors", but that some DVD players actually insert some of their own data "not true to the original data" into the digital stream. Siau states that these errors, both soley & possibly in combination, is what adversely affected Taffel's final listening results. Benchmark's complete writeup is found on their website in their July 2009 Feedback Newsletter.

 Unfortunately, Benchmark does not go as far to list those CD & DVD transports they actually tested, and their specific findings. I'd love to see that information posted somewhere, Elias!


----------



## urbo73

It would be nice to see a list of the transports tested and which did what, but I don't know that Benchmark will do this. I know there are some players that do NOT modify the digital content over their coaxial/optical S/PDIF outputs, but that is illegal in terms of copyright reasons (I believe the only legal way to output is over HDMI). So they will neither confirm nor deny what they do to not get in trouble. If Benchmark were to test such a player, would they publish the results, possibly creating trouble for the manufacturer?

 On the other hand, it would be nice to see how badly some fare over the others, though I don't know how much it would matter. I'm not a believer that high-rez audio really means anything anyway on playback. More bits are good for recording/engineering purposes (i.e. more headroom) and higher sampling rates don't reproduce a more accurate waveform than the Nyquist theorem proves. I firmly believe that, and have not seen any scientific fact to prove that wrong. It's all marketing hype to me - 24/96 24/192, etc. 

 I would ask Benchmark if they plan to add an HDMI input however to their DAC.


----------



## music_man

i don't know if this has been asked but i was hoping mr. gwinn could answer it. 

 it there any difference in sound on the dac1 when using optical,coaxial(rca) or aes/ebu? i am assuming it is designed so they are all equal quality connections?

 edit: is see you already answered this question! "no difference between cables".

 thank you,
 music_man


----------



## yilmaz196

I am using dac1 usb only as dac through balanced outs to my wa22. Should I keep the volume knob at max?


----------



## EliasGwinn

Quote:


  Originally Posted by *jpelg* /img/forum/go_quote.gif 
_Been meaning to post this question for a while now, and thought this thread is as good a place to elicit a response as any. 

 In the October 2009 issue of The Absolute Sound, John Siau responds to Alan Taffel's column reviewing the DAC1-PRE. One of Taffel's points was his observation of the DAC1 having different results with different transports, despite a relatively long history from both Benchmark & many users touting that such differences are greatly minimized (if not completely eradicated) with the DAC1.

 Mr. Siau describes some recent findings at Benchmark claiming that both CD & DVD transports not only "produce non-recoverable read errors", but that some DVD players actually insert some of their own data "not true to the original data" into the digital stream. Siau states that these errors, both soley & possibly in combination, is what adversely affected Taffel's final listening results. Benchmark's complete writeup is found on their website in their July 2009 Feedback Newsletter.

 Unfortunately, Benchmark does not go as far to list those CD & DVD transports they actually tested, and their specific findings. I'd love to see that information posted somewhere, Elias!_

 

Here is a list of the transports we tested:

DVD Transports - Benchmark

 However, please note:

 - This is not meant to represent a complete survey of all transports but merely demonstrate that transports will often corrupt data
 - John Siau suggested that transport errors *MAY *have caused the reviewer to arrive at his conclusions, but it can't be known for sure without testing the devices. There were more questions then answers with this review setup.
 - We've conducted several tests that prove that the performance of the DAC1 conversion is ONLY dependent on the data. In other words, if a given set of 1's and 0's arrive at the DAC1, they will ALWAYS produce the same result regardless of transport and/or digital cable. If there are differences in sound, it must be due to inaccuracies in the data.

 Best,
 Elias


----------



## EliasGwinn

Quote:


  Originally Posted by *yilmaz196* /img/forum/go_quote.gif 
_I am using dac1 usb only as dac through balanced outs to my wa22. Should I keep the volume knob at max?_

 

It depends on the input capabilities of the WA22. Ask the manufacturer what the maximum input level is for the WA22, and I'll tell you where to put the volume knob.

 All the best,
 Elias


----------



## pcf

Hi Elias,

 I have also been under the impression that the performance of DAC1 was less dependent on the transports used than others. That is based on our phone conversations and info from this thread. I also own two of the DVD players on your transport test list. (Oppo and Pioneer). I did my own listening tests comparing those two budget players as transport to some cd transports that costs ten times more. They seemed to sound the same to my ears when they were hooked up to the DAC1.
 But are you telling us now that different transports affect the performance of DAC1 more than we were led to believe? I thought that was one of the strong selling point of DAC1.

 I am still very happy with my DAC1pre though.

 Keep up the good work.

 Paul


  Quote:


  Originally Posted by *EliasGwinn* /img/forum/go_quote.gif 
_Here is a list of the transports we tested:

DVD Transports - Benchmark

 However, please note:

 - This is not meant to represent a complete survey of all transports but merely demonstrate that transports will often corrupt data
 - John Siau suggested that transport errors *MAY *have caused the reviewer to arrive at his conclusions, but it can't be known for sure without testing the devices. There were more questions then answers with this review setup.
 - We've conducted several tests that prove that the performance of the DAC1 conversion is ONLY dependent on the data. In other words, if a given set of 1's and 0's arrive at the DAC1, they will ALWAYS produce the same result regardless of transport and/or digital cable. If there are differences in sound, it must be due to inaccuracies in the data.

 Best,
 Elias_


----------



## jpelg

Quote:


  Originally Posted by *EliasGwinn* /img/forum/go_quote.gif 
_Here is a list of the transports we tested:
DVD Transports - Benchmark_

 

Thanks so much for the information, and the clarification Elias. I apprears that the Panasonic DVD-S47 continues to be one of the few transports that outputs unadulterated hi-rez digital.


----------



## EliasGwinn

Quote:


  Originally Posted by *pcf* /img/forum/go_quote.gif 
_Hi Elias,

 I have also been under the impression that the performance of DAC1 was less dependent on the transports used than others. That is based on our phone conversations and info from this thread. I also own two of the DVD players on your transport test list. (Oppo and Pioneer). I did my own listening tests comparing those two budget players as transport to some cd transports that costs ten times more. They seemed to sound the same to my ears when they were hooked up to the DAC1.
 But are you telling us now that different transports affect the performance of DAC1 more than we were led to believe? I thought that was one of the strong selling point of DAC1.

 I am still very happy with my DAC1pre though.

 Keep up the good work.

 Paul_

 

No, what I'm saying is that different transports may or may not mangle the data. Almost all transports work perfectly at redbook (44/16), although errors may result because of the condition of the CD itself and any dust particles that are in the optical system. Each transport has its own method of dealing with a read-errors, and the data may be modified as a result.

 Best,
 Elias


----------



## pcf

Quote:


  Originally Posted by *EliasGwinn* /img/forum/go_quote.gif 
_No, what I'm saying is that different transports may or may not mangle the data. Almost all transports work perfectly at redbook (44/16), although errors may result because of the condition of the CD itself and any dust particles that are in the optical system. Each transport has its own method of dealing with a read-errors, and the data may be modified as a result.

 Best,
 Elias_

 

Thanks for the quick reply and clarification!

 Paul


----------



## G-U-E-S-T

[That's weird - an earlier post I made above, disappeared overnight! So I will post it again.]

 Elias, I'm curious please: You've mentioned earlier that the ideal volume-dial position on the DAC1 is between approximately 12 and 3 o'clock, and that attenuating below this can compromise system SNR. But, so long as it doesn't overload the input of an amplifier of course: What about going beyond 3 o'clock on the DAC1 dial (i.e. into the DAC1's up to 10db "positive-gain" region) - why please is doing that, considered less than ideal?


----------



## urbo73

I would like to also understand this better. Why exactly is the 11-3 or 12-3 range the most ideal, or best? What happens if I'm under 11 or over 3 let's say?


----------



## EliasGwinn

Quote:


  Originally Posted by *G-U-E-S-T* /img/forum/go_quote.gif 
_Elias, I'm curious please: You've mentioned earlier that the ideal volume-dial position on the DAC1 is between approximately 12 and 3 o'clock, and that attenuating below this can compromise system SNR. But, so long as it doesn't overload the input of an amplifier of course: What about going beyond 3 o'clock on the DAC1 dial (i.e. into the DAC1's up to 10db "positive-gain" region) - why please is doing that, considered less than ideal?_

 

 Quote:


  Originally Posted by *urbo73* /img/forum/go_quote.gif 
_I would like to also understand this better. Why exactly is the 11-3 or 12-3 range the most ideal, or best? What happens if I'm under 11 or over 3 let's say?_

 

The ideal range of the DAC1 volume control (10:30-ish to 3ish) is where you will acheive the most accurate left-right balance. When you adjust a stereo potentiometer, you are actually moving two wipers across two elements (one for left and one for right). The 'middle' of the elements acheive pin-point precision, whereas the extreme ends of the elements are less likely to match each other quite so exactly.

 G-U-E-S-T,

 The SNR won't be affected by the pot position, which is one of the great benefits of using an analog volume control. Whatever noise is going into the pot will be attenuated or amplified equally w/ respect to the signal. Therefore, the signal-to-noise ratio will always remain consistent through that circuit.

 Perhaps you are thinking about our conversation with high-power amps, which will limit the amount of signal you can send from the DAC1 before it becomes too loud. This will affect SNR. Although, someone made a great point (I believe in this thread) that its more a function of input gain/sensitivity. The ideal amp will allow you to send high levels from the DAC1 without it becoming too loud for your listening preference. 

 All the best,
 Elias


----------



## G-U-E-S-T

Quote:


  Originally Posted by *EliasGwinn* /img/forum/go_quote.gif 
_...The SNR won't be affected by the pot position, which is one of the great benefits of using an analog volume control. Whatever noise is going into the pot will be attenuated or amplified equally w/ respect to the signal. Therefore, the signal-to-noise ratio will always remain consistent through that circuit.

 Perhaps you are thinking about our conversation with high-power amps, which will limit the amount of signal you can send from the DAC1 before it becomes too loud. This will affect SNR. Although, someone made a great point (I believe in this thread) that its more a function of input gain/sensitivity. The ideal amp will allow you to send high levels from the DAC1 without it becoming too loud for your listening preference. 

 All the best,
 Elias_

 

Hi Elias,

 I apologize, you are right - I was in fact thinking of the high-power amp conversation. Actually I think I've (hopefully) learned several things along the way here:

 1) For best channel-balance performance, stay between approx. 11 and 3 o'clock on the volume-dial;

 2) The DAC1 output's signal-to-noise ratio (SNR) is always constant, regardless of its volume-dial position and/or internal attenuator-pad selection;

 3) The amplifier's input-stage has its own inherent noise that gets amplified along with the input signal. So generally speaking, and obviously assuming no ground-loops or other external issues: The optimum SNR through the amp's input stage is by definition achieved by delivering the highest-level (i.e. greatest voltage, least attenuation) input signal from DAC1 to amp as possible, without ever exceeding the amps maximum input voltage (since exceeding this would overload the amp's input stage, and cause the amp to clip and distort). Thus the _ideal_ amp for the DAC1 will have a low input sensitivity, being able to at least handle the maximum peak voltage output of the DAC1;

 4) Well-built amps perform best (i.e. with lowest distortion & noise) at the top-end (i.e. between 75% and 95%) of their power-delivery capability. So the _ideal_ amp, when receiving the highest possible DAC1 signal as described in (3) above, will also simultaneously be driving the user's speakers to their most comfortable listening levels with their preferred music type, without also having too much more additional headroom. Thus the ideal amp will have a maximum input voltage at least equal to (but at most only slightly greater than) the maximum peak voltage output of the DAC1, and will also be producing comfortable listening levels when receiving from the DAC1 these optimally highest-possible input signal levels (which also might mean that in a system with proper gain-staging, the ideal home-use amps for most users are likely of much lower power than the marketing strategies of many high-end amp manufacturers would have us believe).

 Elias, please advise: Do these above points look correct to you?


----------



## music_man

Quote:


  Originally Posted by *Matias* /img/forum/go_quote.gif 
_There is controversy. 
	

	
	
		
		

		
			





 Just yesterday we tested my DAC1 USB in a system based on Dynaudio Focus 220 speakers and Krell 400ix integrated amplifier, with an Arcam transport. Switching the stock fuses to the HiFi-Tuning.com ones I've said a couple os pages before, and switching power cables from a good one to a Cardas Golden Reference, both gave clear results: wider soundstage, heavier and faster bass transients, clearer mid range (voices).

 But then again, there are those who don't "believe" in cables... even though there's nothing to believe, but to be heard. 
	

	
	
		
		

		
		
	


	




 Cheers,
 Matias_

 

i know this is an old post. forget dbt. with pro equipment a/b is easy. if i have 5 dac1's in a rack, 4 have supplied power cables and one has any power cable you choose. i challenge you to "guess" which dac1 has the better power cable 5 times in a row. of course you cannot look behind the rack. you can listen to anything you choose for as long as you wish.

 i do feel power cords make a difference in some equipment but not on the dac1. neither do cables.

 i would filter my electric if it had noise in it otherwise i would simply plug the dac1 into a distribution strip as is done in the biggest most respected studios in the world.

 music_man


----------



## music_man

i am assuming most people here have one dac1 which is on a table or on top of another component. with it's stock feet how do you even use an aftermarket powercord? the housing of the iec connector will lift up the back of the dac1! unless you are using isolation or some platform as well. the cardas above would certainly lift up the dac1 i'd think. anyone tried it?

 edit: i see many reviews on the net that state the dac1 is very power cable dependent. i like anything that might elevate my audio experience. i have a whole drawer of super expensive power cables and i have done the test i stated above. the only thing i have noticed is that some cables messed up the sound a little. this is because cable companies use tricks in the geometry,shielding,dielectric etc to change the electron flow. nothing actually made it better. than if you get a power cable why not mod the dac1 altogether? mr. gwinn has been through that with us as well. hey remember "pet rocks"? showing my age.

 i noticed another thing while i was playing with cables. the iec socket on all my dac1's are a little larger than the cord end. i can wiggle them around quite a bit. anyone else notice this? 

 music_man


----------



## doctorcilantro

@ ELIAS

 Need your opinion on how the new ADC1 USB will play with WASAPI and Sampitude. I posted this in the Samplitude forum; my concern is that ASIO protocol may be overall more elegant (all i/o handled in Patchmix, and less-cpu intensive):

  Quote:


 Hi all,

 I have a question about recording with my 1616M.

 I currently use the EMU ASIO driver to record and monitor a wet signal simultaneously; input balanced into the 1616M ADC converters and push back out via coax S/PDIF to my monitors.

 I may have some hardware changes soon and would like to be able to do the same thing but using two separate usb devices capable of 24/96 (& 192kHz).

 I'm not sure of the Benchmark ADC1 USB uses async usb but my playback devices will also be USB which would utilize WASAPI. I'm not real familar with the ADC1 USB but maybe some of you have an opinion on whether or not this will work as well as the all-in-one 1616M method.

 I don't need low latency as I am recording vinyl with digital RIAA implemented in Sampitude, so I am listening to vinyl this way, but also recording it simultaneously.

 thanks & happy holidays!
 DC


----------



## music_man

could someone please tell me. is balanced mic wire(xlr) the appropriate cable to use with the dac1 on the analog outs and 110ohm for the aes/ebu? like vtg,belden,mogami? 

 thanks,
 music_man


----------



## JMS

Quote:


  Originally Posted by *EliasGwinn* /img/forum/go_quote.gif 
_I can only guess, as I've never heard of this before. My guess is that it has something to do with the differences between the designs of the RCA and XLR input stages in the amplifier. Possibly the DC coupling or input transformer of the XLR stage is causing significant low-frequency attenuation. 

 The XLR and RCA outputs on the DAC1 use the exact same signal source and components.

 All the best,
 Elias_

 

Hi Elias, revisiting this issue from about a month ago, I now think it's probably an issue with the DAC1 rather than the amp. I just tried comparing the noise floors of the DAC1's balanced output and the unbalanced (with digital input set to silence). With unbalanced, I hear no static/white noise at all through the speakers, while switching to balanced, I hear a very faint amount. It doesn't seem to be the amp's own noise -- when I disconnect the XLR input from the amp altogether, I hear no noise.

 Not sure if this is necessarily causing the sonic differences I hear, but it does suggest that the two outputs are not identical, at least in terms of noise floor.


----------



## EliasGwinn

Quote:


  Originally Posted by *G-U-E-S-T* /img/forum/go_quote.gif 
_Hi Elias,

 I apologize, you are right - I was in fact thinking of the high-power amp conversation. Actually I think I've (hopefully) learned several things along the way here:

 1) For best channel-balance performance, stay between approx. 11 and 3 o'clock on the volume-dial;

 2) The DAC1 output's signal-to-noise ratio (SNR) is always constant, regardless of its volume-dial position and/or internal attenuator-pad selection;

 3) The amplifier's input-stage has its own inherent noise that gets amplified along with the input signal. So generally speaking, and obviously assuming no ground-loops or other external issues: The optimum SNR through the amp's input stage is by definition achieved by delivering the highest-level (i.e. greatest voltage, least attenuation) input signal from DAC1 to amp as possible, without ever exceeding the amps maximum input voltage (since exceeding this would overload the amp's input stage, and cause the amp to clip and distort). Thus the ideal amp for the DAC1 will have a low input sensitivity, being able to at least handle the maximum peak voltage output of the DAC1;

 4) Well-built amps perform best (i.e. with lowest distortion & noise) at the top-end (i.e. between 75% and 95%) of their power-delivery capability. So the ideal amp, when receiving the highest possible DAC1 signal as described in (3) above, will also simultaneously be driving the user's speakers to their most comfortable listening levels with their preferred music type, without also having too much more additional headroom. Thus the ideal amp will have a maximum input voltage at least equal to (but at most only slightly greater than) the maximum peak voltage output of the DAC1, and will also be producing comfortable listening levels when receiving from the DAC1 these optimally highest-possible input signal levels (which also might mean that in a system with proper gain-staging, the ideal home-use amps for most users are likely of much lower power than the marketing strategies of many high-end amp manufacturers would have us believe).

 Elias, please advise: Do these above points look correct to you?_

 

This is 99% correct!

 The 1% that I would change is that the ideal amp will deliver a comfortable volume range when the DAC1's volume control is in its best range (11 - 3 o'clock), with 3 o'clock being the loudest the listener would ever want. Also, I would suggest erring on the side of being *slightly *too powerful, versus not powerful enough.

 All the best,
 Elias


----------



## john11f

Hi Elias,

 I currently use a Dynaudio BM6A MKII w/ my DAC1 Pre connected to an iMac w/c is similar to the BM5A recommended use on the Benchmark website. I find that the bass is a bit lacking and would like to improve on the sound. Have you had any experience connecting a BM9S sub?


----------



## EliasGwinn

Quote:


  Originally Posted by *music_man* /img/forum/go_quote.gif 
_i know this is an old post. forget dbt. with pro equipment a/b is easy. if i have 5 dac1's in a rack, 4 have supplied power cables and one has any power cable you choose. i challenge you to "guess" which dac1 has the better power cable 5 times in a row. of course you cannot look behind the rack. you can listen to anything you choose for as long as you wish._

 

This is a good test, but you can do something like this without 5 DAC1's. Have your spouse, child, or friend switch the object in question (power cable, digital cable, transport, etc) without you seeing. Obviously, you need to control your own testing (make sure you can't tell which cable/device is connected), but you owe it to yourself to challenge the supposed improvements with expensive aftermarket add-ons.

 Best,
 Elias


----------



## EliasGwinn

Quote:


  Originally Posted by *doctorcilantro* /img/forum/go_quote.gif 
_@ ELIAS

 Need your opinion on how the new ADC1 USB will play with WASAPI and Sampitude. I posted this in the Samplitude forum; my concern is that ASIO protocol may be overall more elegant (all i/o handled in Patchmix, and less-cpu intensive):_

 

I'm currently testing this. Keep in touch...

 Best,
 Elias


----------



## EliasGwinn

Quote:


  Originally Posted by *music_man* /img/forum/go_quote.gif 
_could someone please tell me. is balanced mic wire(xlr) the appropriate cable to use with the dac1 on the analog outs and 110ohm for the aes/ebu? like vtg,belden,mogami? 

 thanks,
 music_man_

 

This is correct.

 Best,
 Elias


----------



## EliasGwinn

Quote:


  Originally Posted by *JMS* /img/forum/go_quote.gif 
_Hi Elias, revisiting this issue from about a month ago, I now think it's probably an issue with the DAC1 rather than the amp. I just tried comparing the noise floors of the DAC1's balanced output and the unbalanced (with digital input set to silence). With unbalanced, I hear no static/white noise at all through the speakers, while switching to balanced, I hear a very faint amount. It doesn't seem to be the amp's own noise -- when I disconnect the XLR input from the amp altogether, I hear no noise.

 Not sure if this is necessarily causing the sonic differences I hear, but it does suggest that the two outputs are not identical, at least in terms of noise floor._

 

Here's something you can try to test this theory: Use an XLR to RCA cable (or adaptor) to connect the XLR outputs from the DAC1 to the RCA inputs of your amp. The level will be a little lower (or higher, depending on the attenuator settings), so keep this in mind.

 The XLR to RCA cable must have only pin-2 and pin-1 connected (pin-3 needs to be disconnected or "floating"), which creates an unbalanced connection similar to the RCA outs. Here is a cable like this: Benchmark Media

 Best,
 Elias


----------



## EliasGwinn

Quote:


  Originally Posted by *john11f* /img/forum/go_quote.gif 
_Hi Elias,

 I currently use a Dynaudio BM6A MKII w/ my DAC1 Pre connected to an iMac w/c is similar to the BM5A recommended use on the Benchmark website. I find that the bass is a bit lacking and would like to improve on the sound. Have you had any experience connecting a BM9S sub?_

 

I haven't heard the BM9S, but the DAC1 PRE can easily drive a sub and powered monitors simultaneously. We have MANY customers doing exactly that.

 All the best,
 Elias


----------



## music_man

thank you for replying mr. gwinn. i don't have many questions about the dac1. it does everything i ever wanted it to. including sounding great!

 music_man


----------



## john11f

Hi Elias,

 What's your take on the USB to SPDIF conversion to the Benchmark? Does it improve sound?

 The dealer told me it improves soundstage even for ordinary MP3 or ALAC (but specially for high-res stuff). Given I use active speakers and listen nearfield, would this make a difference?


----------



## peter73

Dear Elias,

 I own the DAC1 and I am planning to buy a Sennheiser HD650. I have read above in this tread as well as elsewhere in the net that the headphone out of the DAC might not be sufficient to drive these headphones. On the other hand I also read that the headphone out of the DAC has been designed with these particular phones in mind.

 Could you, please share a bit more information on this matter, especially if you have tried this combination yourself.

 Thank you,
 Peter


----------



## music_man

mr. gwinn, could the xlr outputs of the dac1 classic be split with 'y' cables to feed two balanced devices? will this degrade sound quality? will it even work?

 thank you,
 music_man


----------



## MarkyMark

Quote:


  Originally Posted by *jpelg* /img/forum/go_quote.gif 
_Thanks so much for the information, and the clarification Elias. I apprears that the Panasonic DVD-S47 continues to be one of the few transports that outputs unadulterated hi-rez digital._

 

So do the DVD-S77 and DVD-S97. Can be had cheap on ebay.

 The thing to bear in mind is that you will only get the unadulterated digital output when there is no copy protection set. For DVD-A, that means no copy flag and a disc without a watermark. For DVD-V's, no CSS protection must be set.

 For 24/96 stereo DVD-A's that are protected, you can consider getting around it by making a DAD (DVD-V) disc. Use DVDA-Explorer to get the data then use something like DVD Audio Creator to make the disc. Even if the data has a watermark, it won't matter because no checking is done for a DVD-V's.


----------



## EliasGwinn

Quote:


  Originally Posted by *john11f* /img/forum/go_quote.gif 
_Hi Elias,

 What's your take on the USB to SPDIF conversion to the Benchmark? Does it improve sound?

 The dealer told me it improves soundstage even for ordinary MP3 or ALAC (but specially for high-res stuff). Given I use active speakers and listen nearfield, would this make a difference?_

 

I'm sorry I didn't respond to this sooner. Usually Head-Fi emails me when new posts are made, but I didn't get one this time.

 The Bel Canto USB to SPDIF conversion will not improve the sound over the USB input of the DAC1 USB/PRE/HDR. The Bel Canto device simply licensed the same firmware that we developed for our USB interface. In other words, it is the exact same thing.

 All the best,
 Elias


----------



## EliasGwinn

Quote:


  Originally Posted by *peter73* /img/forum/go_quote.gif 
_Dear Elias,

 I own the DAC1 and I am planning to buy a Sennheiser HD650. I have read above in this tread as well as elsewhere in the net that the headphone out of the DAC might not be sufficient to drive these headphones. On the other hand I also read that the headphone out of the DAC has been designed with these particular phones in mind.

 Could you, please share a bit more information on this matter, especially if you have tried this combination yourself.

 Thank you,
 Peter_

 

Peter,

 EDIT: I just noticed that you are asking about the 650's, not the 800's! We have the 650's here for personal use, and we sell the 650's as a companion to our DAC1 models. They work very well with the DAC1. We have MANY customers who use the 650's with their DAC1 and are very satisfied. 

 I can give you two more things to keep in mind:

 1. The HPA2 (Benchmark's signature headphone amp that is built into all DAC1 models) has more then enough power to sufficiently drive any pair of headphones. 

 2. The engineers at Sennheiser use the DAC1 to listen to their headphones

 I hope that helps,
 Elias


----------



## EliasGwinn

Quote:


  Originally Posted by *music_man* /img/forum/go_quote.gif 
_mr. gwinn, could the xlr outputs of the dac1 classic be split with 'y' cables to feed two balanced devices? will this degrade sound quality? will it even work?

 thank you,
 music_man_

 

There should be no problem with this setup.

 If you have any issues, let me know. 

 All the best,
 Elias


----------



## EliasGwinn

Quote:


  Originally Posted by *MarkyMark* /img/forum/go_quote.gif 
_So do the DVD-S77 and DVD-S97. Can be had cheap on ebay.

 The thing to bear in mind is that you will only get the unadulterated digital output when there is no copy protection set. For DVD-A, that means no copy flag and a disc without a watermark. For DVD-V's, no CSS protection must be set.

 For 24/96 stereo DVD-A's that are protected, you can consider getting around it by making a DAD (DVD-V) disc. Use DVDA-Explorer to get the data then use something like DVD Audio Creator to make the disc. Even if the data has a watermark, it won't matter because no checking is done for a DVD-V's._

 

Very true.

 Great post.

 -Elias


----------



## peter73

Quote:


 I can give you two more things to keep in mind:

 1. The HPA2 (Benchmark's signature headphone amp that is built into all DAC1 models) has more then enough power to sufficiently drive any pair of headphones. 

 2. The engineers at Sennheiser use the DAC1 to listen to their headphones

 I hope that helps,
 Elias 
 

Thank you, Elias - prompt and to the point as always.
 Have a nice day,
 Peter


----------



## john11f

quite an invaluable thread for Benchmark owners.

 Thanks again, Elias!


----------



## music_man

i'd like to thank you once again as well, mr. gwinn! 

 music_man


----------



## Pepsi

Quote:


  Originally Posted by *EliasGwinn* /img/forum/go_quote.gif 
_Peter,

 EDIT: I just noticed that you are asking about the 650's, not the 800's! We have the 650's here for personal use, and we sell the 650's as a companion to our DAC1 models. They work very well with the DAC1. We have MANY customers who use the 650's with their DAC1 and are very satisfied. 

 I can give you two more things to keep in mind:

 1. The HPA2 (Benchmark's signature headphone amp that is built into all DAC1 models) has more then enough power to sufficiently drive any pair of headphones. 

 2. The engineers at Sennheiser use the DAC1 to listen to their headphones

 I hope that helps,
 Elias_

 

Yes the amp does have the sufficient energy to drive any pair of cans, but there is always better, and that built in amp will never sound as good as my WA5, but nonetheless this is truly an AMAZING dac, I love this thing.


----------



## EliasGwinn

Thanks for the kind words, friends.

 Btw, I am working on a very cool project right now. I can't tell you much about it, except that it is NOT a Benchmark product, but you all will LOVE it. 

 I wanted to give the head-fi crew an early heads-up because this thread has been very fun and lively. I haven't mentioned the project publicly anywhere else...yet...

 All the best,
 Elias


----------



## FallenAngel

Tease!


----------



## peter73

Quote:


 that built in amp will never sound as good as my WA5 
 

Maybe because it was not meant to sound *like* your WA5. But "better" or "worse" is a higly subjective statement. Check this article from the bibile of audio subjectivism.


----------



## Mark DeB

Quote:


  Originally Posted by *MarkyMark* /img/forum/go_quote.gif 
_So do the DVD-S77 and DVD-S97. Can be had cheap on ebay.

 The thing to bear in mind is that you will only get the unadulterated digital output when there is no copy protection set. For DVD-A, that means no copy flag and a disc without a watermark. For DVD-V's, no CSS protection must be set.

 For 24/96 stereo DVD-A's that are protected, you can consider getting around it by making a DAD (DVD-V) disc. Use DVDA-Explorer to get the data then use something like DVD Audio Creator to make the disc. Even if the data has a watermark, it won't matter because no checking is done for a DVD-V's._

 

Thank you for that information. I don't know DVD Audio Creator, but LPLEX can be used to create the files for a DVD-V, and Nero can be used to burn it. (For more info, see Tool to put 96/24 PCM in a DVD - iTrax HD Audio Forums, which explains how a conversion from WAV to WAV in Foobar--a kludge--may be required to get LPLEX to work.)

 Thanks again!


----------



## islewind

Quote:


  Originally Posted by *Pepsi* /img/forum/go_quote.gif 
_Yes the amp does have the sufficient energy to drive any pair of cans, but there is always better, and that built in amp will never sound as good as my WA5_

 

 Quote:


  Originally Posted by *peter73* /img/forum/go_quote.gif 
_Maybe because it was not meant to sound *like* your WA5. But "better" or "worse" is a higly subjective statement._

 

Well said. Consider that when Sennheiser debuted the HD800 at CES last year, they chose a $6000 Meitner CDP and the Benchmark DAC-1 for the associated gear.


----------



## asmagus

Quote:


  Originally Posted by *EliasGwinn* /img/forum/go_quote.gif 
_Peter,

 (...)

 1. The HPA2 (Benchmark's signature headphone amp that is built into all DAC1 models) has more then enough power to sufficiently drive any pair of headphones. 
 (..)_

 

So quantity is good but what about quality of HPA2? Will it fit 24-32 Ohm headphones?

 I can't find info about used in B. DAC op-amps, about power etc. Can you provide more info?


----------



## Pepsi

Quote:


  Originally Posted by *peter73* /img/forum/go_quote.gif 
_Maybe because it was not meant to sound *like* your WA5. But "better" or "worse" is a higly subjective statement. Check this article from the bibile of audio subjectivism._

 

Of course, some may enjoy the sound of the built in amp, no ears are the same. I can see where you misunderstood me, and those are my apologies for not being clear, i just meant to say i preferred the sound of my amp rather than the Dac1's built in, therefore the sound is better to me. To others? It could be complete rubbish, so to each his own, i was only sharing my opinion.


----------



## anetode

Quote:


  Originally Posted by *asmagus* /img/forum/go_quote.gif 
_So quantity is good but what about quality of HPA2? Will it fit 24-32 Ohm headphones?

 I can't find info about used in B. DAC op-amps, about power etc. Can you provide more info?_

 

The HP amp's output resistance is .01ohms, so it shouldn't have any trouble


----------



## music_man

i think the dac1's headphone amp is the best at being neutral. super dry. crystal clear. when people want "better" i think they are looking to an amp that does something to the sound. the dac1 tells the truth. to be honest it is the reference in much of the high end mastering industry. as well as the dt880 pro. people that don't like this stuff are "audiophiles". this is engineer approved! there is nothing wrong with that either. i am simply pointing out the difference in sound signature. it is not meant to wow you. please do not take this as an insult as i do not mean any.

 not to mention what is really amzing is that the dac1 is first and foremost a dac. the headamp is a bonus. one of the best bonus' you will ever be getting imo. i listen to a different amp at home because it pleases my ears. i work as an engineer, i play as an audiophile. for reference material i would look to the dac1.

 music_man


----------



## Phelonious Ponk

The DAC1/pre is grossly over-engineered. They took a no-limits approach to the design and built a unit that has noise, distortion and jitter levels well below what is audible. What comes out of the thing is as transparent as it gets. You can do different, if you have that odd approach to hifi that uses electronic components like eq to tweak sound to taste, but you can't do better, you can only prefer other. And if you do prefer "other," I'd suggest picking your color in transducers, where color really can't be avoided anyway. Or a good eq.

 P


----------



## anetode

You can find products with better jitter measurements, with better snr, with more extravagant engineering, etc. You just won't _hear_ any improvement. Benchmark has been testing the market and edging the prices up with incremental DAC1 upgrades, so even they can throw more money and more features at the device.


----------



## music_man

phelonious, does that apply to the current spec base model dac1 as well? i mean not pre,hdr,usb etc. you put a "/" in your statement so i am assuming it does?

 well i couldn't be more pleased with it. i switch sometimes between a dac1 and a da924 in my own systems just because i have two stereo's and preffered to get two different products. they are most certainly "competing" products to me. how much does the dac1 cost again? yeah, thats what i thought. to say it is a bargain is like saying albert einstein was smart lol. 

 no offense to lavry, it is one heck of a dac as well. it is just designed in a manner that makes it very expensive. the thing about the lavry is it will most likely last a lot longer because of the "oven". if that is important to some people. even when it is newer the oven probably keeps things in better spec. especially if you constantly power it down. so they are both great products imo.

 music_man


----------



## himiagg

I have the same problem posted by Xxtest83 on 07-19-2009, #2647 (page 177).
 I bought my DAC1 HDR in November 2009, and noticed very soon that the balance is shifted to the left at lower volumes (9 to 11 o'clock).
 I confirmed this by measurements using SPL meter first, and then by measuring the output at the XLR outputs, at 1 kHz, on 100K resistive load.
 I played a 1kHz sine wave from a CD, using coax digital connection, and measured Vrms with Tektronix TDS2002B oscilloscope. Maybe this is not the best setup to measure the channel tracking, but this is what I got:

 Position| L-R
 (o'clock)|(dB)

 5|-0.42
 4|-0.48
 3|-0.55
 2|-0.44
 1|-0.12
 12|0.27
 11|0.73
 10|1.32
 9|2.44

 So my question is the same - is it something typical for this design or the pot is defective? 

 At night, I would listen to my system at about 9 o'clock potentiometer position. That's when channel imbalance is getting quite annoying to my ears. At day time though, sometimes, I would go to as high as 2 o'clock (which is about -3 dBFS, accordingly to my measurements), with some recordings. 

 My gain jumper setting is default (-20dB). I tried to set jumpers to -30 dB attenuation, to shift the working range of the pot, but it is clearly too quiet, especially for some recorded material which is recorded at somewhat lower levels (for example, Donald Fagen's "The Nightfly", or even quieter, "Rachmaninoff performs his solo piano works"/"A Window in Time").
 Basically, at -30dB setting, I'm loosing about 7 dB of usable range at the top.

 Here I have to say that besides this channel tracking issue with my unit, the DAC1 HDR is absolutely marvelous piece of audio equipment, I love the sound (which I guess is no sound of its own)!

 But what can be done to remedy this channel tracking problem?


----------



## music_man

it is a known fact that almost any analog pot has channel differences at the lower end of it's range.if you then have to crank it at the 30db i don't know what to tell you.

 yours still seems a little odd. i am not getting that difference out of the multiple dac1's(standard) we own. not that much at least.

 i'll just wait for mr. gwinn to answer. 

 music_man


----------



## EliasGwinn

Quote:


  Originally Posted by *himiagg* /img/forum/go_quote.gif 
_I bought my DAC1 HDR in November 2009, and noticed very soon that the balance is shifted to the left at lower volumes (9 to 11 o'clock).

 ...

 So my question is the same - is it something typical for this design or the pot is defective? 
_

 

The differences should not be this large. We calibrate the balance to within 0.005 dB at about the 12:45 position (9 dBu on the XLR outputs w/ 0 dB attenuation using a -10 dBFS input signal). In the 'prime range' (11-3 o'clock), the balance should be near or below 0.1 dB.

 Either your test setup is wrong, or your DAC1 needs to be calibrated. If you would like to send your DAC1 in to be re-calibrated, email "sales at benchmarkmedia dot com".

 All the best,
 Elias


----------



## EliasGwinn

Quote:


  Originally Posted by *asmagus* /img/forum/go_quote.gif 
_So quantity is good but what about quality of HPA2? Will it fit 24-32 Ohm headphones?

 I can't find info about used in B. DAC op-amps, about power etc. Can you provide more info?_

 

Low-impedance headphones is where the HPA2 will out-perform most other headphone amps. The low output impedance (<0.01 ohms) will ensure that it maintains a decent damping ratio, whereas most headphone amplifiers will suffer with low-impedance headphones.

 The HPA2 is the same headphone amplifier that is used in major recording, mastering, and broadcast studios around the world. We used to sell it as a module that could replace the headphone amp in large-format audio consoles. 

 It is a reference-grade headphone amplifier...which is to say that its success is based in transparency, purity, and truthful performance. It is not hyped, colored, or otherwise sonically sculpted. But most importantly, it not only achieves as good or better performance then any headphone amp available, it will maintain that performance level in an incredibly large range of headphone impedances.

 All the best,
 Elias


----------



## himiagg

Quote:


  Originally Posted by *EliasGwinn* /img/forum/go_quote.gif 
_The differences should not be this large. We calibrate the balance to within 0.005 dB at about the 12:45 position (9 dBu on the XLR outputs w/ 0 dB attenuation using a -10 dBFS input signal). In the 'prime range' (11-3 o'clock), the balance should be near or below 0.1 dB.

 Either your test setup is wrong, or your DAC1 needs to be calibrated. If you would like to send your DAC1 in to be re-calibrated, email "sales at benchmarkmedia dot com".

 All the best,
 Elias_

 

Music man and Elias, thank you both for such a prompt reply! I'll send my DAC1 for recalibration... I'm going to miss it a lot while being without it!


----------



## Bach-Fan

Elias,

 You mentioned a few days ago that you are working on an exciting new project. The suspense is killing me. What more can you tell us now about this project? And when will we know full details?


----------



## Bach-Fan

(removes inadvertent double post. Sorry.)


----------



## EliasGwinn

I will have more details soon, but I can tell you this...its all about making recordings the way they used to be made...artists giving real performances, collisions of talent and spontaneity.


----------



## music_man

mr. gwinn, excuse me if this has been answered.
 using the dac1's spdif. is the dac1's output affected by what is fed into into it? ie, $50 cdp,$5000 cdp,computer? or does it output the same quality sound no matter what you feed into it's spdif input?

 thank you,
 music_man


----------



## EliasGwinn

Quote:


  Originally Posted by *music_man* /img/forum/go_quote.gif 
_mr. gwinn, excuse me if this has been answered.
 using the dac1's spdif. is the dac1's output affected by what is fed into into it? ie, $50 cdp,$5000 cdp,computer? or does it output the same quality sound no matter what you feed into it's spdif input?

 thank you,
 music_man_

 

There are two factors that generally affect the sonic quality of a D/A process: audio data (1's and 0's) and clock quality (precision). The sonic quality of DAC1 is only limited by the data that it is given; it is not affected by the clock quality. If the data are the same, then the quality will be the same. 

 Most CD transports will give the same data, but the clock quality may differ. The DAC1 will perform equally between any CD transport that delivers identical data. This also holds true for any other digital sources, such as computers. 

 All the best,
 Elias


----------



## music_man

thank you. you answered before i could edit my post. i was going to edit to ask you if it matters if you feed the dac1 audio from a dvd clock. which is at a different frequency than a standard redbook clock.

 i think you already answered the question above!

 thanks,
 music_man


----------



## Prog Rock Man

I have gone throught this brilliant thread, but I may have missed the answer, so how do cables cause jitter?

 EDIT - could you also clear something up for me. Is the Benchmark solution to USB jitter synchronous, asynchronous or adaptive?

 Thanks.


----------



## Bach-Fan

Elias,

 I have been using Airport Express's RCA outputs to drive an integrated tube amp, Jolida JD707A. There is no easy way to bypass the Jolida's preamp. Three questions about using the DAC1 USB in this setup:

 (1) If I were to buy a Benchmark DAC1 USB and use the Airport Express toslink-out to connect to the DAC1 - how would I go about adjusting the volume (or other settings) on these devices to maximize audio fidelity, minimize distortion, etc. 

 (2) Have your customers had mostly good luck using AE's toslink with DAC1 units?

 (3) I will be using Apple's iPhone remote application - which controls the iTunes software remotely over wifi - to choose tracks and also to adjust volume. You have pointed out that the downside to this approach is reduced dynamic range. But I don't understand why this would be a bad thing. Some symphonic music - Tchaicovski's 1812 Overture, for example - has a dynamic range that outstrips the ability of my system to reproduce. Also, this is a bedroom system, so I want the louds to be reduced while still being able to hear the soft parts. So I'm having trouble understanding why the iTunes volume control is not actually an advantage over the DAC1's analog volume control, rather than a disadvantage. I must be oversimplifying or something. 

 Thanks.


----------



## himiagg

Quote:


  Originally Posted by *EliasGwinn* /img/forum/go_quote.gif 
_The differences should not be this large. We calibrate the balance to within 0.005 dB at about the 12:45 position (9 dBu on the XLR outputs w/ 0 dB attenuation using a -10 dBFS input signal). In the 'prime range' (11-3 o'clock), the balance should be near or below 0.1 dB.

 Either your test setup is wrong, or your DAC1 needs to be calibrated. If you would like to send your DAC1 in to be re-calibrated, email "sales at benchmarkmedia dot com".

 All the best,
 Elias_

 

Well, I got my DAC1 HDR back from service last week.

 I have to say, the Benchmark's customer service is top-notch! I spoke to Rory at Benchmark on Monday morning, he reviewed my e-mail and quickly issued the RMA number; I sent the unit out on Monday, they got it on Wednesday morning, fixed it and sent it back the same day, so on Friday I got it back already! Wow! Now I know that this company is a true benchmark in customer service as well...

 Here is what I measured this time (calculating a difference in dB from measured V (RMS) at 100K load between "+" and the ground terminals of each XLR output, playing 1kHz sine wave from a CD player via digital connection, measuring with Tektronix TDS2002B oscilloscope):

 Position| L-R
 (o'clock)|(dB)

 5|-0.69
 4|-0.68
 3|-0.62
 2|-0.51
 1|-0.5
 12|-0.4
 11|-0.3
 10|0
 9|-0.24

 So for all my purposes, now this is virtually perfect. Moreover, although I could not reliably measure the signal levels below 9 o'clock position, I could not hear any noticeable shift in channel balance all the way down to my hearing threshold... This is excellent! Interestingly though, this time the device seems to be calibrated for a perfect balance point at about 10 o'clock position.
 It's not quite within 0.1 dB in the 'prime range', but it probably has something to do with my measurement technique. Anyway, after I got the unit back, I don't hear the kind of channel imbalance which I definitely could hear before.

 Thank you guys again for helping me to sort this issue out!

 On a side note, I think it would be useful to have some sort of channel balance control available - it may not matter in studio applications, where there would be other devices in the signal chain to take care of the channel balance.
 But for home audio applications, when it might be desirable to make some balance correction (to compensate for other component's imbalance, or speaker placement with respect to walls and/or listening position), it can become a problem when the DAC1 is the only device in between the digital source and the power amplifier(s) with no other available balance or channel gain adjustment.

 So here is my question: I noticed that there is a trim pot on the PCB which is marked "L/R balance", if I remember correctly. Does it have the same function as a conventional channel balance control on any commercial stereo equipment? I could not find any mention of it in the manual.

 It's not that I would dare to modify the device in any way, but I'd be glad to see some form of more easily accessible L/R balance control in the future models of it. If the trim pot which I mentioned above is doing what I think it's doing, the feature is almost there...


----------



## music_man

i am assuming the dac1 is your preamp? 
 my personal dac1 is the classic model so it is feeding a preamp. i use the rear fixed pots. i was able to calibrate the rear fixed outputs with a high end calibrated meter to "0" difference! it took a steady hand and a long time but i did it. since my preamp has a very high end relay based attenuator, the balance is far better than what any human ear can hear.

 high end equipment does not usually have balance controls for the user. so being able to use the rear outputs is more control than one would usually get with good equipment.

 i must say that with good cables through the balanced outputs the dac1 is among the best of the best there is. many people say this or that beats it. i don't agree. some dac's may sound different but not better. then factor in the price and this is just amazing. the headphone output is my "reference" as well. both in the studio and at home.

 music_man


----------



## EliasGwinn

Quote:


  Originally Posted by *Prog Rock Man* /img/forum/go_quote.gif 
_I have gone throught this brilliant thread, but I may have missed the answer, so how do cables cause jitter?_

 

Jitter is timing error. The digital signal carries a clock, which is like a metronome for the converter. Jitter results in the clock signal to be delayed, which causes the converter to fire at the wrong time.

 Different cables have different impedances (resistance, capacitance, and inductance), which forms a filter that attenuates high-frequencies. When a digital clock changes states (from a '1' to a '0', for example), it ideally changes instantaneously, but a lack of high-frequency capability causes the transition to be slower. 

 Also, noise creates inaccuracies during the transition. The most common culprit of noise is signal reflection. Signal reflection is minimized by proper impedance termination and the proper characteristic impedance of the cable. However, poor cable sheilding will also allow EMI noise to impurify the signal. 

 For more information about jitter and its affects on digital audio conversion, see this app note by John Siau:

Benchmark Media -- Jitter and Its Effects

  Quote:


  Originally Posted by *Prog Rock Man* /img/forum/go_quote.gif 
_ could you also clear something up for me. Is the Benchmark solution to USB jitter synchronous, asynchronous or adaptive?_

 

Benchmark's USB interface is synchronous, but the jitter attenuation technology is asynchronous.

 All the best,
 Elias


----------



## EliasGwinn

Quote:


  Originally Posted by *Bach-Fan* /img/forum/go_quote.gif 
_(1) If I were to buy a Benchmark DAC1 USB and use the Airport Express toslink-out to connect to the DAC1 - how would I go about adjusting the volume (or other settings) on these devices to maximize audio fidelity, minimize distortion, etc._

 

Use the analog volume control on the front of the DAC1 USB.

  Quote:


  Originally Posted by *Bach-Fan* /img/forum/go_quote.gif 
_(2) Have your customers had mostly good luck using AE's toslink with DAC1 units?_

 

Yes, but keep in mind that it is only capable of streaming 16-bits. This means that you don't want any processing happening, including volume contol. Also, this means that you shouldn't play 24-bit audio files.

  Quote:


  Originally Posted by *Bach-Fan* /img/forum/go_quote.gif 
_(3) I will be using Apple's iPhone remote application - which controls the iTunes software remotely over wifi - to choose tracks and also to adjust volume. You have pointed out that the downside to this approach is reduced dynamic range. But I don't understand why this would be a bad thing. Some symphonic music - Tchaicovski's 1812 Overture, for example - has a dynamic range that outstrips the ability of my system to reproduce. Also, this is a bedroom system, so I want the louds to be reduced while still being able to hear the soft parts. So I'm having trouble understanding why the iTunes volume control is not actually an advantage over the DAC1's analog volume control, rather than a disadvantage. I must be oversimplifying or something. _

 

You are thinking of "dynamic range" as it applies program-material, which is defined by the difference between the RMS and Peak signal levels within the program material.

 The "dynamic range" that is affected by digital volume control is synonomous with signal-to-noise ratio. In other words, decreasing dynamic range means increasing the noise floor relative to the overall system level.

 Best,
 Elias


----------



## music_man

i know this probably does not matter with the dac1. in general is it true that the best coaxial cable length for avoiding jitter is 3 meters?

 thanks,
 music_man


----------



## Prog Rock Man

Much obliged for the reply Elias. Thanks.


----------



## Bmac

Well, I've had my DAC-1 PRE for a few months now and have slowly been wading through this thread. I'm finally finished. I feel like I've just earned my undergrad in DAC-1. 
	

	
	
		
		

		
			





 I bought it before I found this thread, and once I saw this I thought it would be worthwhile to read through. Little did I know at the time that I had stumbled upon one of the most informative and interesting threads in all of the internets. What Elias has done here is real testament to the customer service of Banchmark and his presence and dedication has really served to reinforce my satisfaction with the DAC-1 purchase.

 I'm using it with some Dynaudio BM5A's, an HSU sub and headphones (AKG K701, K270 Playback, Grado SR60, Shure E4G). The transparency of the DAC-1 is awesome. I've had some other decent quality DAC's in this system, the Neko D100, the Music Hall 25.2 and an Essence STX. There were things I liked about all of them, but prefer the DAC-1 PRE to the others by a good margin. It's really an awesome piece of equipment.

 After having gone through here, I do have a few questions I was hoping Elias could address.

 The first is in regards to J. River Media Centre. Fairly early on in this thread some people had asked if you had ever done any testing with it, and you hadn't at the time, and there is still nothing in the Benchmark Wiki with regard to it. I was just wondering if you ever had a chance to play with it and if there were any settings that you could recommend for bit-perfect playback. I only use J. River with FLAC files and just want to make sure that I am getting the best possible quality from the pc side.

 I know for other programs that you've recommended using a 24-bit word length so that if digital volume control is used or the bits are otherwise somehow altered that the 24-bit word won't be truncated to 16-bits. In J. River, you can choose 16-bit, 24-bit, or use the native word length of the file. I have all volume controls set to 100%, so I have been using the native word length. J. River seems to recommend setting it to 16-bit, which doesn't seem like a great idea to me, so this is why I'm wondering if you have tested it, to see if there is a preferred setting.

 The second is just a general question with regard to Windows 7 audio. I know you did testing with XP, but didn't see much information with regards to Vista and Windows 7 audio. Have you done much testing with these OS's? I believe I read somewhere in the thread that there was no way to bypass SRC in Vista/7 unless using asio/kernel streaming, so even if playing a 16/44.1 file and having the sounds in the control panel set to 24/44.1, Windows will still resample, but the sample rate conversion is of such high quality that it won't degrade the audio. Is this correct? Or is there a way to bypass the Windows SRC with the proper settings?

 I've tried WASAPI, but occasionally when I do other things while listening the audio will get buggy, skip and not sound right. I can't say I noticed a difference in sound quality whether using kernel streaming or not, so my settings right now in J. River are to playback at the source sample rate and bit-depth, all volume set to 100%, with sounds in the OS set to 24/44.1. I'm extremely happy with the sound from the Benchmark DAC-1 PRE as it stands, but would really appreciate knowing if you think there is anything I could do to improve these settings.

 Thanks.


----------



## Roseval

It is very easy to bypass the mixer in Vista/Win7
 This is called WASAPI (MS own ASIO) and J River has an interface for it
 As WASAPI does ‘nothing’, it talks straight to the sound card, what the player send to WASAPI must match the capabilities of the audio device.
 I got the DAC 1 to work both over SPDIF and USB
 The screenshots with the J River settings can be found here: The Well Tempered Computer


----------



## Bmac

I've used WASAPI with J. River with the exact settings that were in the link. My previous post where I said kernel streaming was a typo and I've edited it. My problem with WASAPI is that if I do other things besides just music it can cause problems with the audio; it starts getting really choppy and just sounds weird, so for the time being I am just using DirectSound.


----------



## music_man

i have a cd that has some very minor "skipping". the cd looks fine. obviously it is a laser tracking error for some reason. isn't the dac1 supposed to "fix" an issue like this so the disc plays properly? or can nothing fix a read error? i thought that is what "error correction" or oversampling is for?

 thanks,
 music_man


----------



## EliasGwinn

Quote:


  Originally Posted by *music_man* /img/forum/go_quote.gif 
_i have a cd that has some very minor "skipping". the cd looks fine. obviously it is a laser tracking error for some reason. isn't the dac1 supposed to "fix" an issue like this so the disc plays properly? or can nothing fix a read error? i thought that is what "error correction" or oversampling is for?

 thanks,
 music_man_

 

The DAC1 can't fix data errors, only clock errors. If the data is not received at the DAC1, it cannot 'generate' that data. It can only alleviate deviations from ideal clock cycles.

 All the best,
 Elias


----------



## wgb113

I read about 12% into the thread before my head started spinning and I'll continue to read but I need the help of Elias and all of you DAC1 users because I've got my finger on the trigger:

 1) How have the DAC1 and DAC1 USB changed over the years in terms of hardware and performance? I am looking for a pre-owned unit but will go new if any changes are significant.

 2) My plan is to connect my iMac to the DAC1 and use it with a pair of AKG K701 and Dynaudio BM5A or BM6AmkII. Any experience you all might have as to how they match up with the DAC1 is appreciated.

 A used DAC1 USB for the price of a new DAC1 seems to be my sweet spot at the moment.

 Bill


----------



## wgb113

Alright, in fear of flames I searched the thread for Dynaudio and AKG and read the correspondence...Bmac has a nearly identical system as what I'm after.

 Still looking for answers to the first question.

 Will the DAC1 USB's high current output op-amps on the output stage only be noticeable with my proposed setup?

 Will only 2 selectable gain ranges on the headphone amp suffice?

 The auto mute and standby functions are nice-to-haves but not deal breakers for me.

 Bill


----------



## music_man

thank you once again mr. gwinn.

 i understand that no device can "create" missing data. i do not know what made me think that last night. i do not know why this disc skips twice always in the same exact place. there are no visable scratches. it must be a bad recording. pretty minor issue though.

 music_man


----------



## wgb113

So from reading the owner's manual for the DAC1 USB it appears that the ability to adjust the gain on the headphone outputs would benefit someone with a set of 62 ohm AKG 701s as well as a pair of 300 ohm Sennheiser 580s. Where would you place the jumper for each and would it constantly have to be changed???

 Bill


----------



## revenge

Although a low impedance headphone, K701 is quite inefficient so it takes a lot of power to make it sing. I'm pretty sure that whatever setting works for HD580, will work for K701 as well (I used to set the gain on my Prehead to high for both). Ideally you should use the headphone amp with the volume knob around 12. Unfortunately for my Ultrasone Edition 8, even the lowest gain on my DAC1 HDR is still way too lound for anything above 10 but for your headphones the high gain should be perfect. Mind you, I doubt K701 will be a great performer with DAC1.


----------



## wgb113

Thanks for the input.

 Why do you feel the AKG 701 and DAC1 won't perform well together? I am looking for a highly analytical sound.

 Bill


----------



## revenge

Well, you know it's not all about power and resolution. Sinergy is also a very important factor and generally K701 is better matched with tubes or a warmer SS with enough power and plenty of bass. 
 I haven't tested my K701 straight out of the Benchmark yet, but I have tried it with my Lavry DA10 and I haven't been impressed at all. Considering the Lavry is a bit more musical (or let's say warmer) than the Benchmark, I think it's fair to expect a highly analytical sound indeed, but leaning toward the bright side of things, with a lean bass and mediocre soundstage.


----------



## wgb113

Quote:


  Originally Posted by *revenge* /img/forum/go_quote.gif 
_Well, you know it's not all about power and resolution. Sinergy is also a very important factor and generally K701 is better matched with tubes or a warmer SS with enough power and plenty of bass. 
 I haven't tested my K701 straight out of the Benchmark yet, but I have tried it with my Lavry DA10 and I haven't been impressed at all. Considering the Lavry is a bit more musical (or let's say warmer) than the Benchmark, I think it's fair to expect a highly analytical sound indeed, but leaning toward the bright side of things, with a lean bass and mediocre soundstage._

 

I can appreciate that viewpoint but those "nasty" descriptives of clinical/analytical are exactly what I'm after with this system. If I want warm distortion that I also find appealing for a change I'll just listen on my other system. Neither is right or wrong, but what YOU want. Anyhow, that's another thread.

 So back to Benchmark, I think I'm going to focus my search on a DAC1 USB.

 Bill


----------



## EliasGwinn

Hello all...

 I'm sorry I've been MIA for a few days/weeks. I've been spear-heading some big projects that have been VERY time consuming.

 Bill,

 Do you feel sufficiently informed about your gear decisions? Are there any remaining questions I can answer for you?

 Best,
 Elias


----------



## MarkyMark

Hi,

 Apologies if this has been asked b4.

 I have a DAC1 Classic, March 2007 build and have just started using the XLR outputs connected to an SPL Auditor headphone amp. It is currently configured with factory default of -20db attenuation. Is it worthwhile changing this to something else?

 Thanks.


----------



## EliasGwinn

Quote:


  Originally Posted by *MarkyMark* /img/forum/go_quote.gif 
_Hi,

 Apologies if this has been asked b4.

 I have a DAC1 Classic, March 2007 build and have just started using the XLR outputs connected to an SPL Auditor headphone amp. It is currently configured with factory default of -20db attenuation. Is it worthwhile changing this to something else?

 Thanks._

 

What is the maximum input level that the SPL is capable of receiving via XLR?

 Best,
 Elias


----------



## wgb113

Quote:


  Originally Posted by *EliasGwinn* /img/forum/go_quote.gif 
_Hello all...


 Bill,

 Do you feel sufficiently informed about your gear decisions? Are there any remaining questions I can answer for you?

 Best,
 Elias_

 

Elias,

 I'm all set for now!

 Thanks,
 Bill


----------



## MarkyMark

Quote:


  Originally Posted by *EliasGwinn* /img/forum/go_quote.gif 
_What is the maximum input level that the SPL is capable of receiving via XLR?

 Best,
 Elias_

 

Hi Elias,

 This is what I found in the user manual:

 "Inputs (XLR sockets, electronically balanced):

 Impedance: ca. 20 kΩ balanced
 ca. 10 kΩ unbalanced
 Maximum Input Level: +24 dBu"

 Would I be correct in assuming that no attenuation is required in calibrated mode (with factory settings) because output at 0dBFS is 24dBu?

 BTW, are the 8 screws for the lid intended to be removed with a regular screwdriver? The ones on my unit seem to be very tight. 

 Cheers,
 Mark


----------



## EliasGwinn

MarkyMark,

 Yes, based on what you have listed, it seems that the 0 dB attenuator setting is appropriate for your system.

 However, I would be careful to listen closely for any clipping, etc. It is better to be slightly under clipping then slightly over.

 Let me know how it goes...

 All the best,
 Elias


----------



## MarkyMark

Hi Elias,

 I changed the pad settings on my unit to 0db attenuation initially but found the signal to hot so have settled on -10db as a happy compromise. There is no hiss or hum even at maximum volume so all seems well.

 Thanks a lot for your input 

 Cheers,
 Mark


----------



## EliasGwinn

Great!

 Thus, the beauty of the passive attenuators.

 [note on report card] "DAC1 plays well with others!"

 Best,
 Elias


----------



## HugoFreire

Elias,

 Could you please advice us on the best buffer length configuration on foobar?
 What about WASAPI vs. Kernel for Windows 7 64-bit?

 Thank you!


----------



## EliasGwinn

In both cases, the sonic performance won't be affected (ie., distortion, noise, etc). However, I would suggest setting these configurations in a trial-and-error based on what is the most stable. 

 Windows 7 seems to have stability issues w/ regards to streaming audio. This isn't specific to USB or ASIO or anything...it seems to be across the board. 

 So, if you reach a stable configuration, keep those settings! If you don't have any stability issues, then don't worry about these settings.

 All the best,
 Elias


----------



## HugoFreire

Nice, thanks for the answer!

  Quote:


  Originally Posted by *EliasGwinn* /img/forum/go_quote.gif 
_In both cases, the sonic performance won't be affected (ie., distortion, noise, etc). However, I would suggest setting these configurations in a trial-and-error based on what is the most stable. 

 Windows 7 seems to have stability issues w/ regards to streaming audio. This isn't specific to USB or ASIO or anything...it seems to be across the board. 

 So, if you reach a stable configuration, keep those settings! If you don't have any stability issues, then don't worry about these settings.

 All the best,
 Elias_


----------



## masterchoi

Is there any way to turn dac1usb off other than take plug off? I use it thru usb from iMac and even when I turn off iMac, dac1 stays on all day. Have anyone successfully turned off their dac1usb using mac?


----------



## poo

^ When my Mac sleeps, DAC1 USB goes into standby.


----------



## masterchoi

Are you also using usb as connection?


----------



## EliasGwinn

Masterchoi,

 Some computers don't send a 'termination announcement' to the USB device. If the DAC1 USB doesn't receive the 'termination announcement', it will stay on awaiting more audio.

 You can engage the 'Standby' mode by changing the input of your DAC1 USB to one that is not active.

 However, the DAC1 USB remains fully powered even in 'Standby' mode. If you want to completely un-power the DAC1 USB, you will need to remove the power cord (or use a switched outlet like a power-strip). 

 All the best,
 Elias


----------



## masterchoi

Thank you EliasGwinn,

 I'll just turn power strip off when necessary.

 I have one more question regarding dac1usb performance. I heard from few members here states that "premium power cord" and "premium power conditioner" will improve dac1s performance noticeably and will benefit more than other dacs. Is that statement true? or will it at least change characteristic?


----------



## EliasGwinn

Quote:


  Originally Posted by *masterchoi* /img/forum/go_quote.gif 
_I heard from few members here states that "premium power cord" and "premium power conditioner" will improve dac1s performance noticeably and will benefit more than other dacs. Is that statement true? or will it at least change characteristic?_

 

This is not true. The power supply is designed such that it will maintain optimal performance even in drastically poor power conditions (i.e, large noise injected on power rails, 120V line voltage varying between 90V and 150V, etc). We have done extensive testing, both listening and measuring, and we have found that the performance will not vary based on external power conditions. 

 All the best,
 Elias


----------



## Prog Rock Man

Would Benchmark consider making a USB only DAC? I am hoping for something smaller and cheaper!!!


----------



## EliasGwinn

Quote:


  Originally Posted by *Prog Rock Man* /img/forum/go_quote.gif 
_Would Benchmark consider making a USB only DAC? I am hoping for something smaller and cheaper!!!_

 

Hey Prog Rock Man,

 I feel ya...but, unfortunately, it wouldn't be much cheaper or smaller. 

 The biggest expenses are the engineering/designing time, the power supply, the chassis, the 3-D circuit board, the chips, and the 5-year warranty. These would all be included in any product we make. 

 Anything saved by removing the non-USB inputs would be spent to design a new circuit board without those connections. ...i.e., stepping over a dime to pick up a nickel...

 Also, the size wouldn't change either, without significantly increasing the cost to re-design the chassis.

 Now, we could make something that was smaller and cheaper, but it wouldn't perform as well. We could put an off-the-shelf power supply, USB solution, DAC system, and volume control...but we'll leave the cheap stuff for someone else to make.

 All the best,
 Elias


----------



## poo

Hi Elias,

 I posted this elsewhere in the forum, (thread here) but would love your feedback when you can please:

 I've been using a pair of Unique Melody custom Shure SE530 IEMs with a Benchmark DAC1. Sounds great, but the Shures are far too sensitive for the HPA2 attenuation steps.

 I presume the cheapest way around this is to make a cable between the Benchmark and my Shures with a resistor in it? Is this likely to impact on sound quality in any way? I find the HPA2 as it is about 3 or 4 times louder than I would like it to be for the SE530. What resistor(s) should I be using to make such a cable?

 Is there a better way to achieve what I'm trying to do?

 Thanks


----------



## EliasGwinn

Quote:


  Originally Posted by *poo* /img/forum/go_quote.gif 
_Hi Elias,

 I posted this elsewhere in the forum, (thread here) but would love your feedback when you can please:

 I've been using a pair of Unique Melody custom Shure SE530 IEMs with a Benchmark DAC1. Sounds great, but the Shures are far too sensitive for the HPA2 attenuation steps.

 I presume the cheapest way around this is to make a cable between the Benchmark and my Shures with a resistor in it? Is this likely to impact on sound quality in any way? I find the HPA2 as it is about 3 or 4 times louder than I would like it to be for the SE530. What resistor(s) should I be using to make such a cable?

 Is there a better way to achieve what I'm trying to do?

 Thanks_

 

Hey there, poo...

 I highly recommend NOT putting resistors between a set of headphones and any headphone amp. The reason is that it will increase the source impedance, which will diminish the damping ratio - a major factor in the quality of a headphone system.

 The HPA2 has less then 0.1 ohm output impedance. This optimizes the damping ratio. The proper place in the circuit to attenuate is before the amplifier. Because of user feedback such as yours, we included a 10 dB attenuation option in the DAC1 USB and both a 10 dB and 20 dB attenuation option in the DAC1 PRE and HDR.

 Best,
 Elias


----------



## EliasGwinn

Hey there, 'Head-Fi' friends!

 A few weeks ago, I told you that I was working on something really cool, and I was really excited to tell you about it. Well, I can finally tell you...

 I am producing a new video series about making recordings!! Each video documents a set of talented collaborators (musical artists, recording engineers, recording studio, Benchmark) as they attempt to completely produce a recording in one session. This harks back to the days of the early masters at Atlantic, Stax, Sun, Decca, Parlaphone, etc, who made several immortal recordings, each one completed in a few hours. 

 Each video begins with a glimpse into the recording session and ends with the mastered recording. The series is called "Masters From Their Day", which is a double entandre: the mastered recording from their one day in the studio; and also an homage to the artists, engineers, and producers from those seminal years..the 'masters from their day'.

 The videos are available at Masters From Their Day | Home, and there is a free lossless download available from each session.

 Please take a moment to watch the first episode and leave a comment. The project will only continue if people are showing interest, so simply watching and discussing would be a major show of support.

 Thanks so much!!

 All the best,
 Elias


----------



## wgb113

Sounds cool Elias! I'll have to check it out when I get home.

 Bill


----------



## masterchoi

Hello Mr. Elias

 It was excellent performance and great project!


----------



## EliasGwinn

Thanks masterchoi!

 wgb113, I'm excited to see what you think!

 Best,
 -Elias


----------



## Bmac

I watched the Masters From Their Day video Elias. Pretty cool. I thought the audio quality (and the video quality for that matter) was quite good. I enjoyed the performance and the music as well.

 I'm not sure how closely this type of session correlates with what it was like to make a recording back in the day, but I enjoyed it nonetheless and am looking forward to the next installment.

 Plus, when the trombone player had his dreads up while playing (about 6:22 in) he kind of reminded me of a young, bearded Bette Midler for some reason which made me laugh.

 Good job!


----------



## Bmac

Just a side note Elias, and certainly not a big deal, but it would be great if the downloads were available in other more popular formats like flac as well.


----------



## masterchoi

Also when I try to download, it open within quicktime plugin in the safari web browser. Is there any way to download it as file?


----------



## EliasGwinn

Quote:


  Originally Posted by *Bmac* /img/forum/go_quote.gif 
_I watched the Masters From Their Day video Elias. Pretty cool. I thought the audio quality (and the video quality for that matter) was quite good. I enjoyed the performance and the music as well._

 

Thanks, Bmac!

  Quote:


  Originally Posted by *Bmac* /img/forum/go_quote.gif 
_I'm not sure how closely this type of session correlates with what it was like to make a recording back in the day, but I enjoyed it nonetheless and am looking forward to the next installment._

 

Yeah, we aren't trying to duplicate the classic recording methods...just revisit a few of the parameters.

  Quote:


  Originally Posted by *Bmac* /img/forum/go_quote.gif 
_Plus, when the trombone player had his dreads up while playing (about 6:22 in) he kind of reminded me of a young, bearded Bette Midler for some reason which made me laugh._

 

HA!! Thats funny...


----------



## EliasGwinn

Quote:


  Originally Posted by *Bmac* /img/forum/go_quote.gif 
_Just a side note Elias, and certainly not a big deal, but it would be great if the downloads were available in other more popular formats like flac as well._

 

We plan to offer FLAC, ALAC, and WMA-Lossless soon. So many things to coordinate, this has taken a back seat. But anyone should be able to play the AIF files that are available and convert them to their favorite formats.

 Thanks again, Bmac.
 -Elias


----------



## EliasGwinn

Quote:


  Originally Posted by *masterchoi* /img/forum/go_quote.gif 
_Also when I try to download, it open within quicktime plugin in the safari web browser. Is there any way to download it as file?_

 

I'll post the links here in a moment.

 EDIT: I'm not doing the upload right, and our IT person isn't in the office, so this will have to wait until tomorrow. Sorry!

 Best,
 Elias


----------



## 5aces

Thank you for putting this material up.

 Came Out Of A Lady-what a fun song!

 I'm in love with the lady vocalist,uh oh.

 Keep up the Hi-Rez work.


----------



## EliasGwinn

Quote:


  Originally Posted by *5aces* /img/forum/go_quote.gif 
_Thank you for putting this material up.

 Came Out Of A Lady-what a fun song!

 I'm in love with the lady vocalist,uh oh.

 Keep up the Hi-Rez work._

 

Thanks 5aces! 

 We already shot two more episodes. Make sure you sign-up on the website to be notified when the new episodes are posted.

 Best,
 Elias


----------



## 5aces

http://www.snapdrive.net/files/38791...utOfALady.flac

 Get that HD file up please,really good song!


----------



## music_man

mr gwinn, is it ok to very often operate the input selector switch? or will it fail much quicker if used often?

 thank you
 music_man


----------



## EliasGwinn

Quote:


  Originally Posted by *music_man* /img/forum/go_quote.gif 
_mr gwinn, is it ok to very often operate the input selector switch? or will it fail much quicker if used often?

 thank you
 music_man_

 

Music_man,

 That is a heavy duty switch that should not fail even with significant usage.

 In other words, it should be fine.

 Best,
 Elias


----------



## music_man

thank you mr gwinn! that is good news!

 music_man


----------



## Soundfan

Quote:


  Originally Posted by *EliasGwinn* /img/forum/go_quote.gif 
_I know, I know...
	

	
	
		
		

		
			





...I've got to test J. River. I have it high on the 'to-do' list.

 Speaking of testing 3rd party products, I just tested this USB extender:

Amazon.com: USB Extender, RJ45 Cable Extends To 150ft: Electronics

 At lengths up to 175 feet, it works great with the DAC1 USB and/or DAC1 PRE.

 I didn't measure jitter since jitter doesn't effect the DAC1. I'm guessing its pretty severe at such lengths. 

 Thanks,
 Elias_

 


 Elias,

 If I connect my main Desktop PC to this USB Extender you mention above, through 30 meters of CAT6 Ethernet Cable to my DAC1 Pre which is in the other room.

 Would this USB Extender through 30 meters of CAT6 Ethernet Cable give me the exact same data and audio quality in my DAC1 Pre, as it would by connecting a Laptop directly by a 1 meter USB Cable to the DAC1 Pre through the Laptops USB Port. 

 Thanks Elias.


----------



## EliasGwinn

Yes, this adaptor is passive and does not change the data whatsoever. I can't guarantee that it will work flawlessly since we don't manufacture it and therefore can't claim its quality. But, if you try it and you aren't getting any audible drop-outs, then it is working as good as it needs to be to not affect the audio quality.

 All the best,
 Elias


----------



## Soundfan

Thanks Elias,

 That will save me the cost of buying a laptop.


----------



## music_man

Quote:


  Originally Posted by *EliasGwinn* /img/forum/go_quote.gif 
_Yes, this adaptor is passive and does not change the data whatsoever. I can't guarantee that it will work flawlessly since we don't manufacture it and therefore can't claim its quality. But, if you try it and you aren't getting any audible drop-outs, then it is working as good as it needs to be to not affect the audio quality.

 All the best,
 Elias_

 

this is the bottom line on any digital cable and should be written in gospel imo. said by someone who designs the equipment that accepts the cables!

 thank you again mr gwinn!
 music_man


----------



## wgb113

On the Benchmark video...very cool! There's something about the "simplicity" of a group of musicians playing together in the same room for the majority of the recording. You just know it when you hear it and this video shows that.

 I signed up for the future installments. Looking forward to them!

 Bill


----------



## wgb113

On the Benchmark video...very cool! There's something about the "simplicity" of a group of musicians playing together in the same room for the majority of the recording. You just know it when you hear it and this video shows that.

 I signed up for the future installments. Looking forward to them!

 Bill


----------



## EliasGwinn

wgb113, I couldn't agree more. This band can play that song equally as well
 at a concert as you hear it on the recording.

 Thanks for watching!
 -Elias


----------



## music_man

my older dac1 is accepting 192khz input and working on the toslink. does that mean it is the revision that accpets 192khz? probably a dumb question.

 edit: it sounds a lot better when i let the dac1 do the upsampling. instead of the $25 soundcard. i'd still like to know the answer to the above question though please. i guess i can't be sure what revision it is unless i open it?

 thank you,
 music_man


----------



## EliasGwinn

If it is working with 192 kHz going into the Toslink, then you have the 192 kHz update.

 Best,
 Elias


----------



## music_man

thank you mr gwinn.

 music_man


----------



## techenvy

has anyone compared the benchmark dac 1 usb to the hiface usb?
   
  thanks


----------



## G-U-E-S-T

Quote: 





techenvy said:


> has anyone compared the benchmark dac 1 usb to the hiface usb?
> 
> thanks


 


 Guess not, techenvy


----------



## Bmac

Hi Elias,
   
  I was reading through John Siau's interviews in the Benchmark feedback newsletter and found the answer to the following question very interesting:
   
  "EAN: The audio world is often filled with tweaks and gizmos that claim to improve the sound of audio; many of those improvements don’t show up in test measurements? Are there some audible differences that just can’t be measured? Or do you believe that the claimed improvements are not really there?
   
  Siau: I firmly believe that we can measure many defects that are not audible. I am sure that there are a few audible defects that we are not detecting with our test equipment, but I believe these are rare. It may surprise you that I have measured significant differences in speaker cables. People have claimed audible differences in cables, and it turns out that differences can be measured! BTW, zip-cord isn’t half bad! But maybe this is a topic for another day."
   
  I was just wondering if you could elaborate on how and what differences John was able to measure in different cables and the subjective impressions of the measured differences if any. It would be great to hear what an independent third party like yourselves was able to find.
   
  Thanks.
  is a topic for another day.EAN: The audio world is often filled with tweaks and gizmos that claim to
  improve the sound of audio; many of those improvements don’t show up in test
  measurements? Are there some audible differences that just can’t be measured?
  Or do you believe that the claimed improvements are not really there?
  Siau: I firmly believe that we can measure many defects that are not audible. I am sure
  that there are a few audible defects that we are not detecting with our test equipment,
  but I believe these are rare. It may surprise you that I have measured significant
  differences in speaker cables. People have claimed audible differences in cables, and it
  turns out that differences can be measured! BTW, zip-cord isn’t half bad! But maybe this
  is a topic for another day.EAN: The audio world is often filled with tweaks and gizmos that claim to
  improve the sound of audio; many of those improvements don’t show up in test
  measurements? Are there some audible differences that just can’t be measured?
  Or do you believe that the claimed improvements are not really there?
  Siau: I firmly believe that we can measure many defects that are not audible. I am sure
  that there are a few audible defects that we are not detecting with our test equipment,
  but I believe these are rare. It may surprise you that I have measured significant
  differences in speaker cables. People have claimed audible differences in cables, and it
  turns out that differences can be measured! BTW, zip-cord isn’t half bad! But maybe this
  is a topic for another day.EAN: The audio world is often filled with tweaks and gizmos that claim to
  improve the sound of audio; many of those improvements don’t show up in test
  measurements? Are there some audible differences that just can’t be measured?
  Or do you believe that the claimed improvements are not really there?
  Siau: I firmly believe that we can measure many defects that are not audible. I am sure
  that there are a few audible defects that we are not detecting with our test equipment,
  but I believe these are rare. It may surprise you that I have measured significant
  differences in speaker cables. People have claimed audible differences in cables, and it
  turns out that differences can be measured! BTW, zip-cord isn’t half bad! But maybe this
  is a topic for another day.


----------



## Roseval

http://www.hydrogenaudio.org/forums/index.php?showtopic=75627&st=25


----------



## Bmac

Thanks for the link!


----------



## EliasGwinn

Hi Bmac,
   
  John did a test w/ a variety of different 'types' of cables driving passive speakers. 
   
  To give a brief summary, the tests proved that the major defining characteristic of a speaker cable is its inductance.  Inductance causes much more sonic variation then 'skin effect', the commonly quoted culprit.  Especially with long speaker cables (> 20 ft), very audible changes in frequency response begin to occur because of the LCR filter created w/ the speaker. 
   
  In other words, a speaker has a complex impedance that varies radically w/ frequency and coil temperature.  Passive crossovers create even more radical variations.  The higher the inductance of the cable, the more the frequency response of the system will be affected by these variations.
   
  The output impedance of the amplifier will further exaggerate the variations.  For this reason, a low output impedance will give the most linear results.
   
  Best,
  Elias
  
  Quote: 





> Originally Posted by *Bmac* /img/forum/go_quote.gif
> 
> I was just wondering if you could elaborate on how and what differences John was able to measure in different cables and the subjective impressions of the measured differences if any. It would be great to hear what an independent third party like yourselves was able to find.


----------



## Bmac

Thanks Elias!


----------



## axellone

But you never tried to use an AKG K1000 with dac1?
  It' was a wonderful surprise!


----------



## ANagorny

Hi Elias!
   
  First of all, thank you so much for the great products and for your patience in answering hundreds of questions here!
   
  I recently purchased a DAC1 HDR, and have a bit of an issue with the channel imbalance - it seems to be shifted to the left channel a bit. As you suggested earlier in this thread, I measured the AC voltage between the 2 and 3 pin of the XLR outputs while playing 60Hz sine signal.
   
  The results I have got are below:

   
  Could you please let me know if this is within the specification, and if there is any way for me to correct this imbalance (maybe by some internal calibration controls inside the DAC1)?
   
  Really appreciate your help!!!
   
  Regards,
  Anatoliy


----------



## EliasGwinn

Hi ANagorny,
   
  That looks about right.  You will want to keep the volume control at or above the 11 o'clock position. 
   
  Let me know if I can answer any more questions for you.
   
  All the best,
  -Elias


----------



## ANagorny

Thanks for answering so promptly, Elias!
   
  I can still hear the imbalance though - and the problem goes away when I switch to Calibrated setting. Is there any way to individually adjust channel balance in the DAC1 HDR and reduce the level of left channel?
   
  If I am not mistaken, I remember you mentioning somehting about L/R calibration controls in DAC1 USB...
   
  Thanks again!


----------



## EliasGwinn

There is a L/R calibration pot.  It is factory calibrated to achieve a balance within 0.01 dB when the volume knob is near 2 o'clock.
   
  Where do you usually have the volume knob while listening?
   
  Thanks,
  Elias


----------



## ANagorny

My listening position is around 12 o'clock, and 2-3 o'clock when I really crank it up 
   
  I have actually got some comments from John Siau on the Benchmark board, he kindly recalculated my voltage measurements into dB:
   

 [size=small]Pot Rotation[/size] [size=small]L (Volt)[/size] [size=small]R (Volt)[/size] [size=small]Imbalance (dB)[/size] [size=small]9 o'clock[/size] [size=small]0.024[/size] [size=small]0.021[/size] [size=small]1.160[/size] [size=small]10 o'clock[/size] [size=small]0.108[/size] [size=small]0.1[/size] [size=small]0.668[/size] [size=small]11 o'clock[/size] [size=small]0.272[/size] [size=small]0.26[/size] [size=small]0.392[/size] [size=small]12 o'clock[/size] [size=small]0.497[/size] [size=small]0.488[/size] [size=small]0.159[/size] [size=small]13 o'clock[/size] [size=small]0.825[/size] [size=small]0.817[/size] [size=small]0.085[/size] [size=small]14 o'clock[/size] [size=small]1.303[/size] [size=small]1.289[/size] [size=small]0.094[/size] [size=small]15 o'clock[/size] [size=small]1.735[/size] [size=small]1.71[/size] [size=small]0.126[/size] [size=small]Maximum[/size] [size=small]1.899[/size] [size=small]1.871[/size] [size=small]0.129[/size]         [size=small]Calibrated[/size] [size=small]1.271[/size] [size=small]1.272[/size] [size=small]-0.007[/size]
   
  I am quoting his reply as well, I think it will be useful to the community here:
   
  Quote: 





> Balance is adjustable, but is factory preset to high accuracy. It is unlikely that the setting has been lost, so please check the following before adjusting the balance control:
> 
> 1)      Make sure that all 4 XLR output attenuator jumpers are set identically. This is this most common cause of an imbalance. Factory preset is 20 dB.
> 2)      If you are using the XLR outputs, insure that a normal listening level is achieved near a 12 o’clock volume control position. If normal listening levels are not obtained near 12 o’clock, change the XLR output attenuator settings. XLR outputs can be adjusted over a 30dB range to optimize the gain-staging between the DAC1 and your power amplifier.
> ...


 
   
  If the factory-set precision is 0.01dB, then my unit seems to be a bit off... I am not exactly clear about how the L/R pot actually works and how much it changes the balance, so would be great to hear some comments about it. Thanks!


----------



## EliasGwinn

We calibrate it to within 0.01 dB at a particular spot near 1 o'clock, but it is normal for the pot to have a 0.1 dB variance at any given spot, even around the balanced spot. 
   
  But you can feel free to balance it using the L/R pot, which is a small trim-pot that you adjust with a small flat screwdriver.  Just BARELY turn it...it is very close as is.
   
  Best,
  Elias


----------



## ANagorny

Thanks a lot Elias! Do I assume correctly that turning it clock-wise will shift the balance to the right?


----------



## Theiresias

Dear Elias,
   
  I am one step away from purchasing the DAC1 USB, however, I have one question left:
   
  With my current USB solution, the EMU 0202, I sometimes get interruptions or drop-outs due to high "dpc latency" caused by my wifi connection, a widely known problem.
   
  Will this happen with the DAC1 USB, too? Or has Benchmark found a way to fix that problem (which is of course not a problem of the DAC itself, but rather, in my case, caused by the wireless network adapter)?
   
  This has nothing to do with "jitter", does it?
   
  Thank you so much for your patience in answering all these questions!


----------



## EliasGwinn

Quote: 





anagorny said:


> Thanks a lot Elias! Do I assume correctly that turning it clock-wise will shift the balance to the right?


 

 Hi ANagony,
   
  I apologize that it has taken so long to reply to your question.
   
  Turning the balance pot clock-wise will make the right channel louder.
   
  Best,
  Elias


----------



## EliasGwinn

Hi Theiresias,
   
  The DAC1 USB has a buffer that minimizes drop-outs, etc., but it cannot eliminate interruptions from a faulty internet connection.  This may affect jitter, but it is beyond the scope of normal jitter.
   
  All the best,
  Elias
  
  Quote: 





theiresias said:


> Dear Elias,
> 
> I am one step away from purchasing the DAC1 USB, however, I have one question left:
> 
> ...


----------



## Theiresias

Hi Elias,
   
  thank you for your reply. It is not, however, a faulty internet connection I was talking about, rather a wireless adapter that spends to much time in its DPC routine due to "egoistical" drivers. Here is a picture:
   

   
  Regards - Martin


----------



## Dalamar

$300 for a usb jack? I'm glad I'm logical enough to realize there's no reason to spend over $200 TOTAL on a dac.


----------



## EliasGwinn

Oh, wow...yeah, I can imagine that causing havoc in a system.
   
  As I mentioned, the DAC1 USB has a buffer to minimize data underflows, but in severe situations such as this one, the system interruption may be too much for the buffer to overcome.
   
  Let me know if you discover anything more about your situation.
   
  All the best,
  Elias
  
  Quote: 





theiresias said:


> Hi Elias,
> 
> thank you for your reply. It is not, however, a faulty internet connection I was talking about, rather a wireless adapter that spends to much time in its DPC routine due to "egoistical" drivers. Here is a picture:
> 
> ...


----------



## sw98

I hope I am not repeating postings on this thread, but I have a quick question about the USB vs. the PRE lineup. I will be only using the USB digital input and using the integrated headphone amp for playback on my computer. Does audio coming from the digital USB input go through the LM4562 op-amps on the PRE as well as on the DAC1 USB version? Will the LM4562's improve my sound quality on the integrated headphone amp?


----------



## EliasGwinn

Yes, the audio does go through LM4562's on its way to the headphone amp circuit, but the headphone amp circuit uses the 5532 because it performs its function in that circuit much better then any other component available.
   
  The performance of the headphone amp in the DAC1 USB will be identical to the headphone amp in the DAC1 PRE.  The 4562's become beneficial when you are using the other analog outputs.
   
  All the best,
  Elias
  
  Quote: 





sw98 said:


> I hope I am not repeating postings on this thread, but I have a quick question about the USB vs. the PRE lineup. I will be only using the USB digital input and using the integrated headphone amp for playback on my computer. Does audio coming from the digital USB input go through the LM4562 op-amps on the PRE as well as on the DAC1 USB version? Will the LM4562's improve my sound quality on the integrated headphone amp?


----------



## sw98

Hi Elias,
   
  Will the DAC1 USB drive passive speakers via RCA?


----------



## Dalamar

Quote: 





sw98 said:


> Hi Elias,
> 
> Will the DAC1 USB drive passive speakers via RCA?


 

 No. You need an amp.


----------



## sw98

Quote: 





dalamar said:


> No. You need an amp.


 

  
  If I get active that have a built-in amp, should I have the volume on speakers at 100% and control the volume on the DAC1 USB?


----------



## Dalamar

Up to you. If the volume control is digital I'd be inclined to say don't (but you probably wont hear a difference), but if it's analog attenuation then sure.


----------



## sw98

Quote: 





dalamar said:


> Up to you. If the volume control is digital I'd be inclined to say don't (but you probably wont hear a difference), but if it's analog attenuation then sure.


 

  
   
  Cool, thanks.


----------



## EliasGwinn

Quote: 





sw98 said:


> If I get active that have a built-in amp, should I have the volume on speakers at 100% and control the volume on the DAC1 USB?


 
   
   

 I would suggest using the DAC1's volume control as long as you are in the optimal range of the volume pot (11 - 3 o'clock).  If this range is too loud or not loud enough, then I would adjust the volume controls on the speakers to achieve this position.
   
  All the best,
  Elias


----------



## 5aces

Thanks for another HD song release on the  Masters From Their Day program.
   
  Glad I signed up for the video and song updates.
   
  88.2 kHz,24 bit "In The Water" song sounds so good on the JH-13's!
   
  Enjoyable to watch the live sound session,read about the recording details,then have a listen to a new song.
   
  Much appreciated.


----------



## EliasGwinn

Thanks for watching and listening!!
   
  If you would like to see this project continue, please consider sharing the link with friends, family, other members of this and other forums, Facebook, Twitter, email, blog, etc.  This project will only be able to continue if there is sufficient interest and viewership, so please spread the word!!
   
  Thanks so much!!
   
  -Elias


----------



## peterbudden

post removed by author


----------



## cansman

Hi Elias,
   
  I have been using my DAC1 USB for around 2 1/2 years now - it is both reliable and sounds wonderful! As you have mentioned in a recent post, the optimal volume pot range is 11 to 3 O'clock. To use this range with my headphones, I have had to use the iTunes digital volume control. Fortunately, I believe that iTunes forms a 24-bit dithered connection with the DAC1. Am I right to say this?
   
  Also, how much of the digital volume control of iTunes can I use before there is too much of a compromise in sound quality, e.g., not less than the halfway mark? I'm sure there is a limit as to how much I can use the iTunes volume control before the overall resolution of my 16-bit music files drops below 16-bit, isn't that right?
   
  Thanks in advance for answering my questions.
   
  Cheers,
  cansman


----------



## cansman

Hi Elias,
   
  Anyone there...
	

	
	
		
		

		
			





?
   
  Cheers,
  cansman


----------



## EliasGwinn

I apologize for the delayed response.  We're in the middle of trade-show season (AES, Rocky Mountain Audio), so the prep for that has been taking up almost all my time.  The rest has been divided between the MastersFromTheirDay.com video series and preparing to release Benchmark's newest product that we will be announcing next week (I can't give it away yet...but lets just say it is several ADC1's in one box).
   
  Its been very crazy-busy...
   
  Quote: 





cansman said:


> Hi Elias,
> 
> I have been using my DAC1 USB for around 2 1/2 years now - it is both reliable and sounds wonderful! As you have mentioned in a recent post, the optimal volume pot range is 11 to 3 O'clock. To use this range with my headphones, I have had to use the iTunes digital volume control. Fortunately, I believe that iTunes forms a 24-bit dithered connection with the DAC1. Am I right to say this?
> 
> ...


 


  Yes, iTunes volume control is 24-bits, and works wonderfully.  I've done tests on the most recent iTunes releases (v10) and it is still behaving as it should.
   
  To determine how many decibels you can reduce the volume before you are comprimising, you must look at the S/N of the D/A you are using.  The lower the noise of the D/A, the lower you can reduce the digital audio volume before the D/A's noise floor becomes audible.  In the case of the is the DAC1, you can reduce the volume by more then 20 dB before it begins to compromise the noise performance of the DAC1.  In terms of the iTunes volume control, this means you can reduce the volume to about 10% of the volume slider (90% down from full-volume).
   
  Best,
  Elias


----------



## cansman

Hey Elias,
   
  Thanks so much for responding to my queries despite your busy schedule. Congratulations on Benchmark's imminent launch of a new product!
   
  In regard to the iTunes digital volume slider, from what you have said and assuming that the volume control is subjectively linear, i.e., a 50% reduction of the slider will result in 1/2 perceived loudness (-10dB), this means that the D/A converter in the DAC1 has a real world S/N ratio of around 126dB, is that right?
   
  How did I come out with that figure? Essentially, a 100% to 50% volume slider reduction will result in -10dB volume reduction; then from 50% to 25% reduction, another -10dB. Finally from 25% to 12.5% reduction - which is around the 10% lower threshold you recommended - another -10dB reduction. This adds up to a total of -30dB reduction. Now, the DAC1 must at least meet the theoretical best S/N ratio of CD reply (when playing 16-bit music), and that is 96dB (is that right?) in order not to compromise sound quality. Consequently, 96dB + 30dB = 126dB!
   
  Is my explanation correct or am I totally way off? I am no EE or Sound Engineer, just a music and hi-fi enthusiast so I might be totally wrong! Please feel free to correct me!
   
  Finally, I am relieved to know that I can use a significant portion of the iTunes volume control (without compromising sound quality) because I have been contemplating buying a DAC1 HDR. With some of my cans, the volume level is actually too loud when the DAC1 volume knob is at the 11th indent - around 11 o'clock (as suggested by you and the manual). From my subjective experience, I actually do find that using too little of the volume knob of the DAC1 results in the music sounding somewhat off balance - it is like there are some frequency shifts - no longer a linear frequency response. The music also sounds a little more grittier and less transparent. I guess that is why Benchmark recommends an optimal operating range for the use of the volume pot in the DAC1.
   
  Elias, I appreciate your response only when you are available to do so. Thanks again for your help!
   
  Cheers,
  cansman


----------



## cevinuubf

Jon L said:
			
		

> /img/forum/go_quote.gif
> 
> I don't know what they're smoking at Benchmark, but clearly their DAC-1 success has gone to their head. Such baseless, ridiculous hyperbole is sad to witness. It might help if they revealed what exactly is so stupendous about their USB implementation.
> 
> ...






Where is its documentation? Thanks for your answer! I still wonder why though I already got it solved.


----------



## lhan

Hi Elias,
   
  I just got a DAC1 recently (non USB or other, REV.C) and I want to use the Calibrated mode to output.
  But the prior user tuned the trim-pot, I want to turn it back to the factory setting.
  Sorry I can't wait your reply and I've done the tests below:
   
  I used a multimeter to measure the RCA and XLR balance.
  I took the 1kHz Square 0dBFS test tone for testing, from here: http://binkster.net/extras.shtml#cd .
  Attenuation of XLR is set to 0dB.
  The output result I got with RCA is 1.92V and XLR is 7.92V (tuned the same of both Left and Right using the trimmer).
  (BTW my EMU 1212M soundcard's XLR output I measued is Left 7.95V and Right 7.99V, with the same tone tested.)
   
  I want to ask that are these values I got OK?
  Or how can I have the proper way to get the factory setting as the manual says RCA 2vRMS (8.2 dBu) or XLR +4 dBu@-0 dBFS or something?
   
  Thank you!
   
  Regards,
  lhan


----------



## EliasGwinn

[size=small][size=11pt]Hi Ihan,[/size][/size]
   
  [size=small][size=11pt]To calibrate the DAC1, you will need a constant-level digital test-tone, a volt-meter, and a small flat-head screw-driver that can turn the trimmers.  [/size][/size]
   
   
  [size=small][size=11pt]1. Determine the proper output level you are trying to acheive.  If you are sending the DAC1's output into a pre-amp, you will want to ask the manufacturer of the pre-amp for the ideal signal-level for their pre-amp's input.  If you are just 'going by ear', play music through the system and adjust one calibration trimmer of the DAC1.[/size][/size]
   
  [size=small][size=11pt]2. Calculate the voltage that corresponds to the manufacturers suggestions using this dB-Voltage converter: http://www.sengpielaudio.com/calculator-db-volt.htm[/size][/size]
   
  [size=small][size=11pt]3. Play the constant-level digital test-tone into the DAC1 [/size][/size]
   
  [size=small][size=11pt]4. Measure the voltage from the inner contact to the outside (for RCA) or from pin-2 to pin-3 (for XLR)[/size][/size]
   
  [size=small][size=11pt]5. Adjust each channels calibration trimmers appropriately.[/size][/size]
   
  [size=small][size=11pt]Best,[/size][/size]
  [size=small][size=11pt]Elias[/size][/size]


----------



## cansman

Hey Elias,
   
  Would you be able to comment on my above post #2963? Thanks so much!
   
  cansman


----------



## EliasGwinn

Rich text editor, ContentEditor_1, press ALT 0 for help.    Quote:


			
				cansman said:
			
		

> In regard to the iTunes digital volume slider, from what you have said and assuming that the volume control is subjectively linear, i.e., a 50% reduction of the slider will result in 1/2 perceived loudness (-10dB), this means that the D/A converter in the DAC1 has a real world S/N ratio of around 126dB, is that right?
> 
> How did I come out with that figure? Essentially, a 100% to 50% volume slider reduction will result in -10dB volume reduction; then from 50% to 25% reduction, another -10dB. Finally from 25% to 12.5% reduction - which is around the 10% lower threshold you recommended - another -10dB reduction. This adds up to a total of -30dB reduction. Now, the DAC1 must at least meet the theoretical best S/N ratio of CD reply (when playing 16-bit music), and that is 96dB (is that right?) in order not to compromise sound quality. Consequently, 96dB + 30dB = 126dB!


 

 You had the right idea, but your numbers are off.  First of all, a 50% decrease in amplitude corresponds to a 6 dB drop.  So, at 12.5% volume, it's actually -18 dB.  I'm not sure iTunes volume control is this precise (in other words, I'm not sure that the graphic representation of 50% is equal to an audible 50%), but its pretty close.
   
  In order to add no more then 1 dB of noise, the playback system must have a noise-floor at least 6 dB lower then the music.  This is because when you add noises together - one 6 dB lower then the other - the result is 1 dB above the higher noise floor.  Likewise, when you add noises together of the same amplitude, you have a 3 dB increase in noise.
   
  For example, if my playback system has a SNR (signal-to-noise ratio) of 102 dB and the music has a SNR of 96 dB (16-bit), the overall SNR will be 95 dB (a 1 dB increase).  If my playback system has a SNR of 96 dB and the music's SNR is 96 dB, the result will be a SNR of 93 dB (a 3 dB increase).
   
  So, your complete playback system should have a SNR at least 6 dB below the music to have a minimal noise contribution.
  
  The DAC1 has a SNR of 114 dB (worst case) - measurable both at the built-in HPA2 headphone outputs and the main analog outputs.  This means that you can apply 12 dB of digital attenuation before you add more then 1 dB of noise to a 16-bit recording.
   
  If you are using any additional equipment beyond the DAC1 (e.g., a 3rd party headphone amp, a pre-amp, a power amp), you need to add their noises into that equation also.
   
   
  Quote: 





> Finally, I am relieved to know that I can use a significant portion of the iTunes volume control (without compromising sound quality) because I have been contemplating buying a DAC1 HDR. With some of my cans, the volume level is actually too loud when the DAC1 volume knob is at the 11th indent - around 11 o'clock (as suggested by you and the manual).


 
   
  Yes, as I mentioned above, if you are using the HPA2 headphone amp that is built into the DAC1, then you can apply up to 12 dB of attenuation in iTunes before adding significant noise to a 16-bit recording.
   
  The higher models of the DAC1 series have selectable gain-ranges that allow you to reduce the headphone output below the standard DAC1 level without diminishing SNR (DAC1 USB has one range 10 dB below the standard DAC1 level, and the DAC1 PRE and DAC1 HDR have two ranges below: 10 dB and 20 dB).  This allows you to customize the headphone amp to suit your specific headphones' sensitivities and listening preferences.
   
  Best,
  Elias


----------



## lhan

Quote: 





eliasgwinn said:


> [size=small][size=11pt]Hi Ihan,[/size][/size]
> 
> [size=small][size=11pt]To calibrate the DAC1, you will need a constant-level digital test-tone, a volt-meter, and a small flat-head screw-driver that can turn the trimmers.  [/size][/size]
> 
> ...


 
   
  Hi Elias,
   
  Thanks for your reply. 
  But I'm still curious that could I get more close/precise value as the factory setting with multimeter?
   
  Regards,
  lhan


----------



## cansman

Thanks Elias for clarifying my understanding - appreciate your patience and help, as always!


----------



## EliasGwinn

Quote: 





lhan said:


> Hi Elias,
> 
> Thanks for your reply.
> But I'm still curious that could I get more close/precise value as the factory setting with multimeter?
> ...


 


  The factory setting for calibrated output is:
  - output attenuators in the -20 dB position
  - +4 dB when a 0 dBFS signal is used (using the calculator on the site I mentioned above, this equates to 1.228 Vrms)
   
  If you have the output attenuators in the 0 dB position, the factory calibrated setting would yield 12.28 Vrms using a 0 dBFS signal.
   
  Best,
  Elias


----------



## Phelonious Ponk

Elias - 
   
  Sorry if this question has already been asked somewhere in this 200 pages, but do you hear a difference between usb and optical implementations on the DAC1?
   
  P


----------



## EliasGwinn

Quote: 





phelonious ponk said:


> Elias -
> 
> Sorry if this question has already been asked somewhere in this 200 pages, but do you hear a difference between usb and optical implementations on the DAC1?
> 
> P


 
   
  Regarding USB vs. Optical:

 Provided that the audio data is the same, all digital inputs of the DAC1 sound identical.  The only way there would be a difference in sound between the USB and optical input of the DAC1 USB/PRE/HDR is if the two sources were providing different data.  This may seem obvious, but it shouldn't be overlooked.  Many 3rd-party USB-to-SPDIF converters actually change the data (they are NOT bit-transparent).  In this case, it will probably sound different.

 The USB input is usually the safest bet because you know that the data is getting from the hard-drive to the DAC1 in as short of a path as possible.

 Best,
 Elias


----------



## spagetka

I would like to know if it is possible to add remote to my DAC1 PRE by any chance. Thank you in advance.
   
  PF'11


----------



## EliasGwinn

Quote:


spagetka said:


> I would like to know if it is possible to add remote to my DAC1 PRE by any chance. Thank you in advance.
> 
> PF'11


 

  Unfortunately this is not possible.
   
  Best,
  Elias


----------



## JMS

I have Squeezebox Touch -> Benchmark DAC1, and I'm hearing a click/pop at my speakers/headphones about 30% of the time when I pause/unpause or manually skip to the next track. I don't know if it damages my speakers but it sure is annoying to hear.The Squeezebox people are saying it's the DAC's problem because their digital outputs are always on. This has been an issue since I got my DAC1 5 years ago.
   
  Any way to fix this?


----------



## bralk

JMS:
   
  Of course they are saying that !
   
  I am using DAC1 and DAC1- HDR with  Stax phones and other gear and I have never heard clicks/pops or other anomalies.
  Sources are MacMini/Powerbook G4/iTunes with TIFF files.
   
   
  Cheers
   
  Tom


----------



## EliasGwinn

Quote: 





jms said:


> I have Squeezebox Touch -> Benchmark DAC1, and I'm hearing a click/pop at my speakers/headphones about 30% of the time when I pause/unpause or manually skip to the next track. I don't know if it damages my speakers but it sure is annoying to hear.The Squeezebox people are saying it's the DAC's problem because their digital outputs are always on. This has been an issue since I got my DAC1 5 years ago.
> 
> Any way to fix this?


 


 Hi JMS,
   
  This is very strange...but I'll try my best to help you.
   
  Please verify: Are you using the base-level DAC1, as opposed to the DAC1 USB/PRE/HDR?
   
  When the click/pop is happening, do the lights on the front of the DAC1 change?
   
  Do you have another digital source that can use the same connection as the Squeezebox (for example, a CD player)?
   
  Best,
  Elias


----------



## TwoTrack

I am certainly happy with the non-USB DAC1 I have.  It has superb resolution and superb bass.


----------



## JMS

Quote: 





eliasgwinn said:


> Quote:
> 
> 
> 
> ...


 

 I'm using the original base-level DAC1, revision H from 5 years ago. The lights in front do not change, and this happens regardless of whether I'm using optical or coaxial input. I don't have any issues when pausing tracks using my PS3's digital output. I do think it's something specific with Squeezebox players.


----------



## EliasGwinn

Quote: 





jms said:


> I'm using the original base-level DAC1, revision H from 5 years ago. The lights in front do not change, and this happens regardless of whether I'm using optical or coaxial input. I don't have any issues when pausing tracks using my PS3's digital output. I do think it's something specific with Squeezebox players.


 


  Yeah, I think you're right.  Did you tell Squeezebox this?  What did they say?


----------



## EliasGwinn

Quote: 





twotrack said:


> I am certainly happy with the non-USB DAC1 I have.  It has superb resolution and superb bass.


 


 Glad to hear it!!  8D


----------



## mahesh

Hi Elias hi all
I have benchmark dac 1 usb connected to my mac pro via usb and benchmark to powered monitors Adam s1x, monitors volume is set to minimum,i set dac volume attunation to -30db, but still sound is too loud, even dac volume is at 9o'clock
So Please help me,if there is something i can do.
I heard about rothwell In-Line Attenuators,and read some reviews,and seems it degrade sound quality.
Thx in advance


----------



## EliasGwinn

Hi mahesh,
   
  Unfortunately, based on the information you've given me, I think the Adam S1X may be too powerful for your listening preferences.  The peak levels coming from your DAC1 (with the -30dB attenuation and volume knob at 9 o'clock) would be about -25 dBu, give or take.  If that is too loud, then your amp/speakers are too loud. 
   
  Yes, you can use in-line attenuators, but the more you reduce the signal that is driving your powered monitors, the more you will reduce your dynamic range (increasing noise amplification).
   
  Does that make sense?
   
  Best,
  Elias


----------



## khaos974

Found something interesting.



More here: iPad Streams High-Resolution Audio to DAC1


----------



## cansman

Hi Elias,
   
  Does Benchmark have any closed-headphone recommendation to go with the DAC1? I am currently enjoying your open-headphone recommendation of the HD650s.
   
  Thanks!
  cansman


----------



## EliasGwinn

Hi Cansman,
   
  Unfortunately, I can't recommend any closed-back headphones.  I haven't tried more then 10-15 different models, but I have yet to find any that I PERSONALLY enjoy as much as my HD650's.  But, as I said, I haven't tried most models. There may be something out there for me, but I'm so happy w/ my 650's that I really don't look too much further.  
	

	
	
		
		

		
			




   
  Best,
  Elias


----------



## good-vibrations

Hi Elias
   
  I'm in the UK and have recently bought a Benchmark DAC1 USB.
   
  Delighted with it so far and would appreciate your advice.
   
  My OS is WinXP and I use Foobar2000 (v 1.1.1) to playback FLAC files using output device setting DS:Benchmark 1.0, with output format set to 24-bit.
   
  I know that USB playback is 'limited' to 24/96, but when I playback 24/192 it still works but I'm not clear on how/where the down sampling is being processed, so would appreciate your help.
   
  Thanks
  good-vibrations


----------



## EliasGwinn

Hi good-vibrations (btw, I love that song!!)
   
  In the scenario you've described above, I can't say for sure where the sample-rate conversion is happening. However, I am certain that it is within the computer, and most likely within Foobar or Kmixer.
   
  Either way, I would recommend not setting it above 96 kHz.
   
  Best,
  Elias


----------



## vrln

Just got a Benchmark DAC-1 PRE for my Genelec 8040 monitors. The combination is just fantastic... Wonderful little box  Doesn´t seem to matter what source I use, all sounds the same to me. Even highly jittery optical via my old LCD TV. Before I had the PRE, I had the normal DAC-1 as a loaner from a local distributor and quickly fell in love with the highly transparent sound. Tried quite a few DACs with the Genelecs and this is by far the best synergy I´ve heard out of them. Almost all preamps have issues with how sensitive the Genelecs are when driven via XLR. On my previous system, I could only use 1-8 out of 75 volume settings... Now with the -30dB attenuators engaged it´s perfect, which the volume usually at 11-12. 
   
  If Elias is here, I have a few questions though...
   
  1.) About the jumpers to configure XLR attenuation (from 0 to -30dB): in the pictures in the manual, the jumpers have this small metal plate behind the pins inside the jumper itself. These all point downwards in the official pictures, but in my PRE P3 and P4 have the plate inside the jumper pointing upwards. Does this affect anything? I corrected the jumpers to look like in the manual picture while I set it to -30 dB. There´s no mention about it in the manual.

   

  2.) I have a CD transport that sends digital via BNC. I have two cables: BNC-BNC and RCA-RCA, both 75 ohm. I also have to adapters: BNC-RCA and RCA-BNC. Which one should I use? Is there any difference? So two options are:

   

  a.) plug in an RCA-BNB adapter into the RCA #3 digital in of the PRE, and then use a BNC-BNC cable between the two units.

   

  b.) plug in an BNC-RCA adapter into the BNC out of my digital transport, and then use my normal RCA-RCA cable between the two units.

   

  PS: bought a Squeezebox Touch today... I guess I won´t be using my CD transport much anymore... It´s also nice with Spotify premium streaming 320kbs ogg vorbis...


----------



## good-vibrations

Quote: 





eliasgwinn said:


> Hi good-vibrations (btw, I love that song!!)
> 
> In the scenario you've described above, I can't say for sure where the sample-rate conversion is happening. However, I am certain that it is within the computer, and most likely within Foobar or Kmixer.
> 
> ...


 
   
  Hi Elias
   
  Many thanks for your reply.
   
  What I should have said was 'when I playback 24/192 *files* it still works'.
   
  I'm trying to find out how the Benchmark Dac1 USB is designed to act when it receives files higher than 24/96.
   
  I've set up a Resampler in Foobar (SoX mod2) to Downsample X2 but ONLY frequencies 176400 and 192000.
   
  It sounds good and could be left in place all the time (as it only Downsamples 176400 and 192000 files) but, just in case it has same detrimental affect on other files, I only activate it when needed.
   
  Is this the best approach for highest SQ when playing files higher than 24/96?
   
  Appreciate you advice.
   
  Thanks.
   
  GV


----------



## EliasGwinn

Quote:


vrln said:


> 1.) About the jumpers to configure XLR attenuation (from 0 to -30dB): in the pictures in the manual, the jumpers have this small metal plate behind the pins inside the jumper itself. These all point downwards in the official pictures, but in my PRE P3 and P4 have the plate inside the jumper pointing upwards. Does this affect anything? I corrected the jumpers to look like in the manual picture while I set it to -30 dB. There´s no mention about it in the manual.
> 
> 
> 
> ...


----------



## EliasGwinn

If you are trying to play sample-rates above 96 kHz while using the USB interface of the DAC1 USB, then your setup seems to be solid.  iTunes is also very good at converting sample-rates, but I don't think you need to change anything.
   
  Best,
  Elias

  
  Quote: 





good-vibrations said:


> Hi Elias
> 
> Many thanks for your reply.
> 
> ...


----------



## good-vibrations

Quote: 





eliasgwinn said:


> If you are trying to play sample-rates above 96 kHz while using the USB interface of the DAC1 USB, then your setup seems to be solid.  iTunes is also very good at converting sample-rates, but I don't think you need to change anything.
> 
> Best,
> Elias


 

 Hi Elias
   
  Thanks for your time and advice.
   
  GV


----------



## cansman

Hi Elias,
   
  Is it essential that the DAC1 be earthed? I have been using it for a while with an unearthed socket - everything appears alright.
   
  Thanks,
  cansman


----------



## tvrboy

Question for Mr. Gwinn:
  I was wondering if you could discuss specifically how and why there are audible differences between competently designed DACs. In the Sound Science forum here we often argue whether or not there is a difference between DACs. Quite a few people believe that "digital is digital" and the digital signal from an ipod will be the same as a DAC1 or any other high-end DAC. I've read in a few places that although the digital conversion process may be quite similar, the power supply and analog output stage are two things that make DACs sound different from each other. If this is true, could you give some examples and measurements proving how these things can make an audible difference in the sound of a DAC? Thanks.


----------



## EliasGwinn

Hi tvrboy,
   
  This is a great question.  I would like to answer this thoroughly, so please excuse me while I prepare a complete answer.
   
  Best,
  Elias


----------



## wgb113

Elias,
   
  Any updates with the DAC1 and OSX Lion + iTunes 10.4?  I see in Audio MIDI they state "2ch-24bit integer".  Just curious as to why they added that last bit? 
   
  Bill


----------



## good-vibrations

Hi Elias
   
  When/will the Centrance universal driver be made available for benchmark dac1 versions with a USB connection?
   
http://centrance.com/downloads/ud/
   
  Thanks for your help.
   
  GV


----------



## EliasGwinn

Quote: 





tvrboy said:


> Question for Mr. Gwinn:
> I was wondering if you could discuss specifically how and why there are audible differences between competently designed DACs. I've read in a few places that although the digital conversion process may be quite similar, the power supply and analog output stage are two things that make DACs sound different from each other. If this is true, could you give some examples and measurements proving how these things can make an audible difference in the sound of a DAC? Thanks.


 


 I'm sorry I haven't posted an answer on this yet.  As I'm preparing my reply, I'm realizing that to answer requires a complete explanation of electronic design!!  Your questions are what every engineer must learn and conquer to make a great product.
   
  But I will finish and post my reply soon.
   
  Best,
  Elias


----------



## EliasGwinn

Quote: 





wgb113 said:


> Elias,
> 
> Any updates with the DAC1 and OSX Lion + iTunes 10.4?  I see in Audio MIDI they state "2ch-24bit integer".  Just curious as to why they added that last bit?
> 
> Bill


 


  No updates yet, but there will be soon.  I'll keep you posted.
   
  Best,
  Elias


----------



## EliasGwinn

Quote: 





good-vibrations said:


> Hi Elias
> 
> When/will the Centrance universal driver be made available for benchmark dac1 versions with a USB connection?
> 
> ...


 

 Hi GV,
  
  We have a company policy not to discuss future plans for product development, so unfortunately, I can't answer this question.
   
  Do you have any suggestions for us?  Would you like to be able to use this driver with the DAC1? 
   
  Best,
  Elias


----------



## daisangen

I'm looking for an all-in-one source solution (balanced output, volume control, as transparent as possible) to feed my Genelecs. The Benchmark DAC1 has piqued my interest.
   
  My question is, are there any sonic differences between USB and PRE models when using balanced XLR output? From what I gathered, the USB model uses both NE and LM opamps, while the PRE is all LM.
   
  The PRE is tempting because it is more versatile but as of now, I don't need the additional inputs and the price jump is quite significant, but I'd hate if I skimped on some and ended up with the inferior product.


----------



## vrln

Quote: 





daisangen said:


> I'm looking for an all-in-one source solution (balanced output, volume control, as transparent as possible) to feed my Genelecs. The Benchmark DAC1 has piqued my interest.
> 
> My question is, are there any sonic differences between USB and PRE models when using balanced XLR output? From what I gathered, the USB model uses both NE and LM opamps, while the PRE is all LM.
> 
> The PRE is tempting because it is more versatile but as of now, I don't need the additional inputs and the price jump is quite significant, but I'd hate if I skimped on some and ended up with the inferior product.


 

 I´m running a pair of Genelec 8040´s with a Benchmark PRE... Before buying the PRE I auditioned the USB model for a few days. To be honest with you, I´d be lying if I said I could differentiate them in a blind test. Do you have a local Benchmark dealer? If yes, I´m sure they can get you a home audition. Or order from a store that allowes returns like Thomann.
   
  I wanted the analog inputs for old analog-output retro consoles, so I went with the PRE in the end. The Benchmark is a very good pairing with the Genelec - most Genelec dealers agree with that too. Many feel it´s the best pairing for them actually. All in all I´m very happy with the combo. It not only does well recorded music well (Diana Krall, Leonard Cohen, Guns and Roses latest album etc really well), but it´s great for movies and gaming too. What it doesn´t like is overly compressed pop music, but that´s just accuracy for you 
   
  PS: If you are using them in a nearfield setup, you might want to enable the highest internal attenuation settings in the Benchmark (very easy to change, just a few jumpers inside - it´s covered in the manual). The Genelecs are very sensitive, so they really like the added attenuation.


----------



## vrln

Quote: 





cansman said:


> Hi Elias,
> 
> Does Benchmark have any closed-headphone recommendation to go with the DAC1? I am currently enjoying your open-headphone recommendation of the HD650s.
> 
> ...


 
   
  I´ll +1 the HD 650 suggestion. Best dynamic headphone there is on the market in my experience. It´s the best allrounder if you ask me. It might not have the "wow"-factor of the HD 800, but it doesn´t have its flaws either. Last but not least, it´s the most comfortable headphone out there, even more comfortable than the Stax headbands


----------



## good-vibrations

Quote: 





eliasgwinn said:


> Hi GV,
> 
> We have a company policy not to discuss future plans for product development, so unfortunately, I can't answer this question.
> 
> ...


 

 Hi Elias
   
  Still really happy with my DAC1 USB as it is but, would like to play FLAC files up to 24/192 via USB if possible.
   
  So, just looking for your advice as to whether or not the CEntrance Universal Driver would enable this.
   
  If Benchmark offered this (or a similar) driver as a free/low cost downloadable 'upgrade' that would. imo, bring the DAC1 USB  up to date and extend the 'life' of the range.
   
  It would not do your already very good PR any harm either. 
   
  Look forward to hearing your thoughts.
   
  Thanks for your help.
   
  GV


----------



## Willakan

Quote: 





good-vibrations said:


> Hi Elias
> 
> Still really happy with my DAC1 USB as it is but, would like to play FLAC files up to 24/192 via USB if possible.
> 
> ...


 

 For various reasons, processing of 192khz files actually results in lower fidelity audio - this applies to prettymuch any DAC, not just the Benchmark. You are actually getting better sound (probably not audibly better, but still) with 24/96.
  To demonstrate I'm not insane, here is a statement from Benchmark backing me up: http://api.viglink.com/api/click?format=go&key=04fea777994d26cd84e01a5e54f4c01d&loc=http%3A%2F%2Fwww.head-fi.org%2Ft%2F571259%2Fhi-rez-another-myth-exploded&v=1&libid=1315846866259&out=http%3A%2F%2Fwww.soundstage.com%2Findex.php%3Foption%3Dcom_content%26view%3Darticle%26id%3D126%3A96khz-vs-192khz%26catid%3D57%3Areader-feedback%26Itemid%3D24&ref=http%3A%2F%2Fwww.head-fi.org%2Ff%2F133%2Fsound-science&title=Hi-Rez%20-%20Another%20Myth%20Exploded!&txt=Benchmark%20Statement&jsonp=vglnk_jsonp_13158468654591


----------



## Cpluse

[size=8pt]Ok I trying  to read all these post to learn more about the Dac1 and the more I read the pre is sounding better to add  for future use. Plus reading about adding an ipad to the mix makes it more exciting. But what I have a question. Will I be able to use something like a SM Pro [/size][size=medium]MPatch 4M or Knob? Will I have the same quilty or I be losing something? [/size]


----------



## good-vibrations

Quote: 





willakan said:


> For various reasons, processing of 192khz files actually results in lower fidelity audio - this applies to prettymuch any DAC, not just the Benchmark. You are actually getting better sound (probably not audibly better, but still) with 24/96.
> To demonstrate I'm not insane, here is a statement from Benchmark backing me up: http://api.viglink.com/api/click?format=go&key=04fea777994d26cd84e01a5e54f4c01d&loc=http%3A%2F%2Fwww.head-fi.org%2Ft%2F571259%2Fhi-rez-another-myth-exploded&v=1&libid=1315846866259&out=http%3A%2F%2Fwww.soundstage.com%2Findex.php%3Foption%3Dcom_content%26view%3Darticle%26id%3D126%3A96khz-vs-192khz%26catid%3D57%3Areader-feedback%26Itemid%3D24&ref=http%3A%2F%2Fwww.head-fi.org%2Ff%2F133%2Fsound-science&title=Hi-Rez%20-%20Another%20Myth%20Exploded!&txt=Benchmark%20Statement&jsonp=vglnk_jsonp_13158468654591


 
   
  Hi Willakan
   
  Thanks for the link/info.
   
  It does ring a bell now that you have pointed it out and iirc is at least one of the reasons Benchmark internally re-sample at around 110khz rather than 192khz or higher.
   
  I'll continue to re-sample 24/192 (and 24/176.4) as described in a previous post (#2991) and choose 24/96 files for any future downloads.
   
  Thanks for your input and help.
   
  GV


----------



## daisangen

Thanks for your thoughts.
   
  I think I will go with the USB then, due to these reasons:
   
  - It's cheaper and available in black unlike the PRE
  - DAC1 looks better without the source selector knob
  - USB, coax, and optical inputs are more than enough for now and foreseeable future
  - I can't ever see myself using analog sources
  - Not having a remote makes the PRE much less desirable (I'll pass on HDR because I have no need remote or additional inputs as of now; those would be solely for future-proofing)
  
  How much can you use the volume control with the max attenuation? Talking about normal levels, not party levels. Though those vary greatly too...
   
  Quote: 





vrln said:


> I´m running a pair of Genelec 8040´s with a Benchmark PRE... Before buying the PRE I auditioned the USB model for a few days. To be honest with you, I´d be lying if I said I could differentiate them in a blind test. Do you have a local Benchmark dealer? If yes, I´m sure they can get you a home audition. Or order from a store that allowes returns like Thomann.
> 
> I wanted the analog inputs for old analog-output retro consoles, so I went with the PRE in the end. The Benchmark is a very good pairing with the Genelec - most Genelec dealers agree with that too. Many feel it´s the best pairing for them actually. All in all I´m very happy with the combo. It not only does well recorded music well (Diana Krall, Leonard Cohen, Guns and Roses latest album etc really well), but it´s great for movies and gaming too. What it doesn´t like is overly compressed pop music, but that´s just accuracy for you
> 
> PS: If you are using them in a nearfield setup, you might want to enable the highest internal attenuation settings in the Benchmark (very easy to change, just a few jumpers inside - it´s covered in the manual). The Genelecs are very sensitive, so they really like the added attenuation.


----------



## EliasGwinn

Hi everyone!  I'm sorry I haven't been responding.  Somehow these latest posts escaped my radar.
   
  I'll try to answer everyone's questions, but please let me know if I've left any unanswered.
   
  Best,
  Elias


----------



## EliasGwinn

Quote:


cpluse said:


> [size=8pt]Ok I trying  to read all these post to learn more about the Dac1 and the more I read the pre is sounding better to add  for future use. Plus reading about adding an ipad to the mix makes it more exciting. But what I have a question. Will I be able to use something like a SM Pro [/size][size=medium]MPatch 4M or Knob? Will I have the same quilty or I be losing something? [/size]


 


 Hi Cpluse,
   
  You can use monitor controllers with the DAC1, although without testing these particular models, I can't tell you if they will degrade the quality.  Perhaps you could acquire one of them temporarily for evaulation?
   
  Best,
  Elias


----------



## EliasGwinn

Quote:


daisangen said:


> How much can you use the volume control with the max attenuation? Talking about normal levels, not party levels. Though those vary greatly too...


 


 The ideal scenario is when you acheive comfortable listening levels with the volume control between the 10 o'clock and 3 o'clock positions.  So, I would suggest trying the DAC1 USB in the factory configured setting (-20 dB).  If you find that you cannot turn the volume up above 10 o'clock without it becoming too loud, then you will want to move the jumpers into the -30 dB position.
   
  Best,
  Elias


----------



## Lord Chaos

Long-term report on Benchmark DAC1-USB:
   
  I bought one of these as soon as it was announced.(serial number 29) to use with my Mac Powerbook-based music system. I used it nearly every day to listen to music with headphones or through an amp and speakers. Its sound still delighted me until the day I took it out of service and replaced it with a DAC1 HDR. Why the replacement? Benchmark offered a good trade-in on the USB (for a limited time), and I wanted the remote control.
   
  I've used the HDR for a couple of weeks now, and its sound is just as transparently delightful as the USB. What's on the recording comes out of the unit. It all depends upon the mastering.
   
  It's also a civilized unit, fading the audio gracefully when desired, and bringing it back just as nicely. It informs me of what it's doing by changing the pattern of the display LEDs. It's all the pre-amp I'll ever need, and I am thinking about building a tri-amped speaker system to go with it. Genelec offers ready-made solutions, expensively. I have lots of speakers lying around...


----------



## EliasGwinn

Glad to hear you're enjoying your HDR, Lord Chaos!


----------



## s.a.b.

I replaced my basic DAC1 with the HDR version.
   
  However, the HDR version does not have a BNC connection so I must use an adapter to use my same Stereovox digital cable.
   
  Should there be a sonic difference using an adapter?  Any particular adapter recommended?  Should I bother to have Stereovox cable reterminated so an adapter can be avoided?  (Using a Radio Shack adapter it seems there is a very slight alteration in pitch, but I confess this could be psychological.)
   
  Thanks.


----------



## Dougr33

Hi Elias,
  I've finally saved up for, and got a great price on a used DAC1 USB. I'm LOVING it!  I'm using it to directly drive my NAD C275bee amp and psb Image T6 speakers, as well as AKG K702 phones.
   
  Maybe these questions have been answered directly, but I've not found them, so hope you don't mind!
   

 Most of my music are flac files thru J. Rivers Media Center. I removed the HPA jumper so it goes full bore, which allows me to get a nice level on the 702s while using Replay Gain (I bounce around a lot while listening, so this is invaluable to me).  But I'm curious: it outputs 24 bits to the DAC1, using Wasabi event mode(so volume controlled by the dac after the replay gain adjustment, of course). Am I losing much sound quality using the replay gain? (it sounds great, but I can't tell a difference with it off.. hard to measure because of course the volume level is hard to match for a/b tests).
 Is there any way I can get use out of the AES input? I'm using the usb for the computer, spdif for a cd deck, and toslink for the tv.  Are there adapters that would allow another spdif coax input, or are voltages/resistance too different?
   
  Thanks for your time sir!!


----------



## EliasGwinn

Quote: 





s.a.b. said:


> I replaced my basic DAC1 with the HDR version.
> 
> However, the HDR version does not have a BNC connection so I must use an adapter to use my same Stereovox digital cable.
> 
> ...


 
   
  There should be no sonic difference whatsoever.  Benchmark converters are immune to differences in differences between interfaces.  As long as the data remains in tact (which, in your case, should not be a problem), the sound will not be affected.
  
  Best,
  Elias


----------



## EliasGwinn

Quote: 





dougr33 said:


> Hi Elias,
> I've finally saved up for, and got a great price on a used DAC1 USB. I'm LOVING it!  I'm using it to directly drive my NAD C275bee amp and psb Image T6 speakers, as well as AKG K702 phones.
> 
> Maybe these questions have been answered directly, but I've not found them, so hope you don't mind!
> ...


 
   
  Hi Doug,
   
  Just to address your question here, I'll paste Rory's reply from the Benchmark forum:
   
   
   
 [size=0.8em] 1. We have not tested the Replay Gain feature of J River, but based on its handling of the signal (64bit), I'm not surprised that you cannot hear a difference between on/off. What does happen is that the noise floor will come up by the gain reduction amount of the program, but its affect sonically will be inaudible at even the most extreme listening levels.[/size]

 [size=0.8em] 2. The AES input on the DAC1USB and DAC1 will accept a 75 ohm coaxial cable by the addition of an Impedance Transformer. There are a few companies that offer these. We sell one manufactured by Canare. It can be found here: [/size]

 [size=0.8em] http://www.benchmarkmedia.com/catalog/female-male-impedance-transformer-p-51.html?osCsid=c42d6e31210df4114ec13ea11fa732ad [/size]

 [size=0.8em] Simply plug it into the DAC's rear XLR digital input, and connect the coax cable from the source. A cable with a BNC connector on one end will be required:[/size]

 [size=0.8em] http://www.benchmarkmedia.com/catalog/75ohm-canare-lv61s-p-75.html?osCsid=c42d6e31210df4114ec13ea11fa732ad[/size]

  Best,
  Elias


----------



## aivar1988

hi fellow head-fi'ers and dac1 owners.
1 question - is there any firmware updates for dac1 usb or theres no such thing?


----------



## ceyser

[size=0.8em] Hi ,[/size]

 [size=0.8em] I have 3 years old Benchmark Dac-Pre , I am very happy with the product.I am using XLR outputs with Calibrated mode tru Preamp.[/size]

 [size=0.8em] I tried to lower trimmer values but unfortunately i broke left channel trimmer screw ..[/size]

 [size=0.8em] Can i find the competible trimmers for replacement or i would like to replace trimmers by fixed value resisitors .
 Could you help me the what is the trimmer value.. or what would be the fix resistor value to achive max +9 dBU output level.[/size]

 [size=0.8em] My current setup Attenuator jumpers are off position.
 Thanks for your kind support..Best regards,[/size]


----------



## fallsroad

Quote: 





ceyser said:


> [size=0.8em] Hi ,[/size]
> 
> [size=0.8em] I have 3 years old Benchmark Dac-Pre , I am very happy with the product.I am using XLR outputs with Calibrated mode tru Preamp.[/size]
> 
> ...


 
   
  Best place to start is here:
   
  service@benchmarkmedia.com
   
  Good folks, very helpful.


----------



## 340519

Hi all, first post on head-fi. I have the Dac 1 (1.5 years old) and am considering the Dac Pre, but only if there is a sonic benefit. Any comments? I have a yba transport, foobar windows 7 running through an M2Tech Evo usb to spdif converter, dac 1 feeding a Bryston 4bsst2, and Philharmonic speakers with AKG over ear headphones. Thanks for the input.


----------



## 340519

Quote: 





dmdm said:


> Hi all, first post on head-fi. I have the Dac 1 (1.5 years old) and am considering the Dac Pre, but only if there is a sonic benefit. Any comments? I have a yba transport, foobar windows 7 running through an M2Tech Evo usb to spdif converter, dac 1 feeding a Bryston 4bsst2, and Philharmonic speakers with AKG over ear headphones. Thanks for the input.


 
  Anyone?  Thanks.


----------



## mtkversion

Why not take advantage of ordering directly from Benchmark and doing their 30 day audition offer?
   
  Test it for yourself and let your ears do the deciding. 
   
  All you'd be out is the $20 or so in return shipping if you don't like it.


----------



## 340519

Quote: 





mtkversion said:


> Why not take advantage of ordering directly from Benchmark and doing their 30 day audition offer?
> 
> Test it for yourself and let your ears do the deciding.
> 
> All you'd be out is the $20 or so in return shipping if you don't like it.


 
  Thanks.  I just ordered the hdr.  I'll sell the dac 1 on ebay and be done with it.


----------



## guildenstern

Quote: 





dmdm said:


> Thanks.  I just ordered the hdr.  I'll sell the dac 1 on ebay and be done with it.


 

 Why sell on ebay? Be kind to head-fi-ers and sell on the "For Sale" forum on this site!


----------



## stakatch

Hi
  I am very happy with my  benchmarks dac1 USB. Now I have very well combo with difficult to drive Hifiman he-4.
  I have a question.Which usb cable should I Use?usb cable ferrite or non ferrite? 
  Also I want to buy Hifiman _[size=small]He[/size]_[size=small]-[/size]_[size=small]6[/size]_[size=small] which are known for being [/size]_[size=small]power hungry[/size]_[size=small].,,,,,,[/size] Which head amp do you recomend?
  Thanks


----------



## tuffgong

HI Elias, haven't posted here in a few years, very glad to see you still kickin butt, I have a new question for you. I'd like to run -two- benchmark usb dacs at once, one to run a pair of speakers, one to run a sub. It appears both Vista and Windows 7 will not allow you to run two default output sources at once. You have to pick, or the computer picks for you. The only way i've seen around this is some software hack called Virtual Audio Cable, with no help on how to set it up, or to go all the way back to windows XP, back before the RIAA held everyone hostage with DRM crap. Do you, or does anyone here, have any suggestions on how to play a simple flac file with VLC, and have that sound go to -two- usb dacs at once? Or do I have to install XP on an Ultrabook, the height of irony.


----------



## Bmac

Why do you even need two DAC's? Just use the RCA outputs for the speakers and the XLR outputs for the sub or vice versa. You can buy XLR->RCA adapters or cables if need be.
   
  If you really did need 2 DAC's though you could always just buy a USB->S/PDIF converter and use 2 digital outs from the converter to the DAC (ie. use both digital coax and optical outs).


----------



## tuffgong

Following your idea I found the halide design usb/spdif converter, I never knew they had designed these, although this one doesn't split into two channels, still, very useful, thanks.


----------



## tuffgong

The V-link also looks good, and splits into two outs. Thanks.


----------



## Bmac

You could also just buy y splitters if you don't want to use RCA and XLR outputs. I have used these in the past, though I don;t recall them costing this much:
   
http://www.takefiveaudio.com/mall/shopexd.asp?id=287


----------



## BleaK

Hi guys, I may have a deal on a old Benchmark DAC1 (from 2006). I have optical out from my computer and if I buy it I plan to use that. It currently lies around 500usd, is this a good deal?


----------



## kawaivpc1

Hello guys,

I'm trying to buy a used DAC1 USB. 
Does it still worth buying after all these years?
Its used price is now only $450. 
Should I buy iDSD Micro instead?
I'm interested in DAC1 mainly because I love the sound of DAC2 HGC. Some people have mentioned that DAC1 is slightly worse than DAC2 but it's still amazing. 


I'm looking for the best sound quality at affordable price. 
Please tell me your opinion. I use Shure's SRH1840. 
I'm using Geek Out 1000 as a main DAC.


----------



## James-uk

Dac1 is audibly transparent just like the dac2 so it will sound the same. the only difference is the specs , obviously the dac2 is better on paper but you will not hear the difference .
Edit : the headphone amp section is identical on both .


----------



## kawaivpc1

james-uk said:


> Dac1 is audibly transparent just like the dac2 so it will sound the same. the only difference is the specs , obviously the dac2 is better on paper but you will not hear the difference .
> Edit : the headphone amp section is identical on both .




Thanks for confirming this, I will get a used unit soon. 
Can you play DSD files through DAC1?
I think DoP may work?


----------



## kawaivpc1

Also, does it come with a stable USB ASIO driver that you can use on DAWs such as Pro Tools and Logic Pro?


----------



## James-uk

kawaivpc1 said:


> Thanks for confirming this, I will get a used unit soon.
> Can you play DSD files through DAC1?
> I think DoP may work?




No DSD. Not sure about the usb driver I only use mine for playback of redbook via MacBook air/iPad.


----------



## James-uk

I suppose this is where the dac1 will fall short. It's features are 'limited' compared to today's dacs ie 'hi res' (dac1 usb *only* accepts up to 24/96)and DSD playback. Neither of which are useful for transparent playback so are just gimmicks really.


----------



## kawaivpc1

Yeah, I read that it supports up to 24 bit 192kHz. Isn't it up to 24/192? 
I think there can be a way. For instance, Meridian Explorer's web page doesn't say it will support DSD but you can play DSD files through Explorer. It's possible if you tweak JRiver or Foobar's setting. 

For sound quality, does it sound better than iDSD Micro, Geek Out 1000, Sony PHA-3, etc? 
If it sounds the same as DAC2, its used price $450 is a bargain for me.


----------



## James-uk

That is a bargain. it only accepts upto 24/96 on the usb input.


----------



## kawaivpc1

james-uk said:


> That is a bargain. it only accepts upto 24/96 on the usb input.






What happens when you play 24bit 192kHz files?
Does it mute itself?
or automatically down sample to 96kHz?


----------



## kawaivpc1

Is there any way around 24bit 192kHz files? I think you can try to use USB to analog adapter cable. According to DAC1's manual, DAC1 supports 24bit 192kHz analog input.


----------



## Roseval

kawaivpc1 said:


> Is there any way around 24bit 192kHz files? I think you can try to use USB to analog adapter cable. According to DAC1's manual, DAC1 supports 24bit 192kHz analog input.


 
  
 24/192 is by design a digital signal, nothing analog about 24 bits 
	

	
	
		
		

		
			




  
 It accepts 24/192 max over the SPDIF- AES/EBU
 The USB accepts 24/96 max. It is a bit dated design (adaptive mode) and of course it won't recognize DoP.
  
 To play 24/192 you need a sound card with SPDIF out
 If you send a signal > 96 kHz straight over the USB, it won't play.
 Alternative, a USB to SPDIF converter supporting 24/192.
 If you want to play DSD you need a media player converting it on the fly to PCM


----------



## kawaivpc1

roseval said:


> 24/192 is by design a digital signal, nothing analog about 24 bits
> 
> It accepts 24/192 max over the SPDIF- AES/EBU
> The USB accepts 24/96 max. It is a bit dated design (adaptive mode) and of course it won't recognize DoP.
> ...




Thanks! That sounds really good. 
Would you give me a few store link to that USB to SPDIF converter or cable? 
I don't know if it makes sense to buy two converters for this purpose. Is there any simple USB to SPDIF cable solution?
What do you use to play digitally stored 24bit 192 files theough DAC1?


----------



## Roseval

There is a (incomplete) list of USB to SPDIF converters on my website http://thewelltemperedcomputer.com/HW/USB_SPDIF.htm
  
As many today DACs do have a USB input capable of 24/192 or even higher I wouldn't go the additional converter way.
  
 I use Toslink most of the time. The standard says it is limited to 24/96 but in practice it plays higher


----------



## kawaivpc1

http://www.pcm63.com/files/usb_to_spdif_cable.JPG


Do you think this will work?
also, if I use a cheap soundcard for spidif instead, will It decrease overall sound quality of DAC1?


----------



## kawaivpc1

M2Tech hiFace 2 makes the most sense to my budget. 

Then, how do you use Toslink on PC to DAC1 setup?

Many thanks,


----------



## kawaivpc1

http://www.amazon.com/gp/aw/d/B0093KZTEA/ref=mp_s_a_1_11?qid=1424295163&sr=8-11&pi=AC_SX110_SY165_QL70&keywords=usb+to+spdif



If I'm going to use something like this, it's like using two converters. Do yiu think there's going to be loss in sound quality in this setup? Is pure USB connection better?


----------



## Roseval

kawaivpc1 said:


> M2Tech hiFace 2 makes the most sense to my budget.
> 
> Then, how do you use Toslink on PC to DAC1 setup?
> 
> Many thanks,


 

 My PC happens to have a Toslink out.
  
 The "problem" with SPDIF is that the sample rate is generated by the sender.
 If the sender has a cheap clock, this will generate tons of input jitter at the DAC.
 I'm not familiar with all the products you listed but the Hiface2 does have an excellent reputation.


----------



## kawaivpc1

roseval said:


> My PC happens to have a Toslink out.
> 
> The "problem" with SPDIF is that the sample rate is generated by the sender.
> If the sender has a cheap clock, this will generate tons of input jitter at the DAC.
> I'm not familiar with all the products you listed but the Hiface2 does have an excellent reputation.




That's what I'm afraid of. If I'm using two converters at a time, their quality should match. 

I found that my mobo has a toslink out as well. Can you play 24bit 192kHz FLAC files through toslink - dac1?


----------



## kawaivpc1

https://geizhals.at/asus-rog-rampage-iv-black-edition-90mb0gx1-m0eay0-a1024617.html#rdat


wait! I found that my computer has an SPDIF out !
In this case, can I just connect DAC1 with an SPDIF cable? and play 24bit 192kHz files? If this works, I don't have to worry about anything.


----------



## Roseval

kawaivpc1 said:


> Can you play 24bit 192kHz FLAC files through toslink - dac1?


 
 Maybe. It is far in excess of the standard but he hardware often performs better


----------



## Roseval

kawaivpc1 said:


> https://geizhals.at/asus-rog-rampage-iv-black-edition-90mb0gx1-m0eay0-a1024617.html#rdat
> 
> 
> wait! I found that my computer has an SPDIF out !
> In this case, can I just connect DAC1 with an SPDIF cable?


 

 Sure


----------



## kawaivpc1

roseval said:


> Sure




Sweet. Now, problem's solved. Thanks!!

I can get full 24bit 192kHz from DAC1 easily by using SPDIF out of my motherboard!


----------



## djcarpentier

Just thought i'd chime in as i just received my dac1. The th600 sounds superb out of the headphone jack. Dead black background. Running it via spdif to my studio monitors, also sounds superb. Excellent sound and value considering the price these can be had for right now.


----------



## hifimckinney

majid said:


> I got an email from them announcing an updated DAC1 with 24/96 USB input:
> 
> http://www.benchmarkmedia.com/eupdate
> 
> ...


 
  
 Does anyone know if this will work with iPhone?


----------

