# Optical TOSLINK vs. USB: Which connection is better to connect a DAC?



## I3eyond

My options are, from my computer:
   
  X-Fi Fatalty Optical Out > DAC
   
  OR
   
  USB > DAC
   
  I would get the Transparent Performance USB Audio Cable if I go USB, or something from Audioquest if I run from my sound card's optical out.
   
  DAC I plan on using is the Music Fidelity V-DAC.
   
  Which will be better?


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## I3eyond

Anyone?  I KNOW there has to be some heavy opinions on this topic..


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## aamefford

You might try a repost (and search) in the computer audio thread.  I've tried both, I don't find much difference, if any.  I use optical from my airport express (arguably mid-fi) to my Nova and Headroom Ultra.  I use USB and Optical sort of interchangeably from my macbook pro to the Headroom ultra.  I can't discern a difference.


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## danielghofrani

I am not an expert at this but from what I heard, they each have their own strengths and weaknesses.
  but the prominent thing is the implementation of the USB or optical input in the DAC. 
  one DAC may work better with USB than optical or visa versa. 
  I hope that helps


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## barleyguy

One advantage of optical is that it electrically decouples your DAC from your source.  Sometimes this can be a huge advantage, such as cases where your DAC is being powered by the USB port of a computer and is getting switching noise.
   
  Also, USB is only stable to about 20 feet (if you're lucky), whereas optical should be stable to at least 330 feet (100 meters).  (It will actually work even farther in most cases, but I believe the spec is 100 meters.)   That's another big advantage.
   
  So for those reasons, I always choose optical over USB if it's an option.


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## palchiu

Toslink is direct digital signal from your soundcard, that's should be better.
   
  USB's signal should be: Soundcard-> PC's mainboard-> USB chip-> USB cable ->DAC's USB receiver.
   
  I've tried coxial vs. USB with my PC, coxial is more better. (I've mod. my card, removed Toslink to BNC connector)


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## vert

I've tried both and unfortunately, I didn't like either - compared to a Halide Bridge USB to SPDIF converter. I was only using a cheap USB printer cable. For toslink, I used a glass toslink cable. BTW, I'm not sure how much difference a high end toslink cable will make - there's quite a bit jitter via toslink, from what I understand. I would like to try an audiophile USB cable like the one you mentioned.


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## 00940

The V-DAC uses an asynchronous sample rate converter (hence the claim for ultra low jitter). The sonic differences in between digital inputs should be zero. 

The arguments for optical over USB would be isolation of the computer ground (can induce some noise in some designs) and the fact that your soundcard optical output is probably not as dependant as USB from the CPU. USB can be laggy when there is an heavy load on the CPU (not so much of a problem with modern CPU). The drivers for your soundcard might also be better optimized for gaming than the generic USB audio drivers.


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## UncleSkeebow

I prefer the sound of USB, (denser, warmer and more present) but if you have a noisy computer system, (interference) optical will isolate that noise and may in some cases give you better sound.


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## mikemalter

I have a digital and usb out, and the digital out sounds better.  I also bought a Monarchy DIP that goes between my DAC and input source.  So with the DIP in there, the difference between my computer and i170 is slight.
   
  Also to consider when looking at optical is the physical makup of the cable.  Is the media plastic or glass?  From my experience if you are using glass in your fiber cable the difference between coax and optical is very little.


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## myinitialsaredac

Quote: 





danielghofrani said:


> I am not an expert at this but from what I heard, they each have their own strengths and weaknesses.
> but the prominent thing is the implementation of the USB or optical input in the DAC.
> one DAC may work better with USB than optical or visa versa.
> I hope that helps


 


  This.
   
  If asynchronous and galvanically isolated, go USB.
   
  If asynchronous and handles high bit word length go USB
   
  If USB has line noise in it, go optical.
   
  If usb has no line noise but is isochronous go optical.
   
  Dave


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## 00940

myinitialsaredac said:


> This.
> 
> If asynchronous and galvanically isolated, go USB.
> 
> ...





 

Too much of a generalization imo... Implementation is at least as important as the technology by itself. Bad grounding/decoupling practices or HF noise riding on the PS lines for example can degrade the best theoretical design.

The funny thing with the V-DAC (which uses the SRC4392, based on the SRC4192) is that you can't indeed measure any jitter in USB mode (with a PCM2706 in adaptive mode), while the SPDIF inputs (hence the optical one) aren't that perfect. See : http://www.stereophile.com/content/musical-fidelity-v-dac-da-processor-measurements Looks like there might be problems with the implementation of the SPDIF receiver of the SRC4392...

Isochronous USB can btw be just as perfect as asynchronous. If you combine a decent isochronous USB receiver (such as PCM2706-7) with the SRC4192, the ASRC will function pretty much like a buffer and there is 0 jitter due to USB making it to the DAC (read here: http://www.diyaudio.com/forums/digital-source/46413-any-feedback-new-cs8421-high-res-asrc.html ).


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## myinitialsaredac

Quote: 





00940 said:


> Too much of a generalization imo... Implementation is at least as important as the technology by itself. Bad grounding/decoupling practices or HF noise riding on the PS lines for example can degrade the best theoretical design.The funny thing with the V-DAC (which uses the SRC4392, based on the SRC4192) is that you can't indeed measure any jitter in USB mode (with a PCM2706 in adaptive mode), while the SPDIF inputs (hence the optical one) aren't that perfect. See : http://www.stereophile.com/content/musical-fidelity-v-dac-da-processor-measurements Looks like there might be problems with the implementation of the SPDIF receiver of the SRC4392...Isochronous USB can btw be just as perfect as asynchronous. If you combine a decent isochronous USB receiver (such as PCM2706-7) with the SRC4192, the ASRC will function pretty much like a buffer and there is 0 jitter due to USB making it to the DAC (read here: http://www.diyaudio.com/forums/digital-source/46413-any-feedback-new-cs8421-high-res-asrc.html ).


 


  You are correct, I did generalize too much.
   
  Though I would say that my generalizations are correct most of the time. 
   
  Are there any DACs that currently use those chip lineups?
  I was under the impression there were no dacs that use a buffer currently.
  The issue I would argue still lies in using the computer clock. In that thread they specify that the chip if the jitter input is low will correct for theoretically all of it, however if it is high it will still cause phase modulation. 
   
  I think the theoretical ideal would to be using error-correction transfer into a buffer and clock it out on the DAC side.
  Matter of time eh?
   
  Dave


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## evulfuson

I'm currently using a udac with Alessandro MS1i's and AD700's. I thought about upgrading to the Maverick D1, would it be worth it to keep my udac as a usb to coaxial converter, or just connect the D1 via usb?
   
  I'm using a laptop and don't have any digital audio outputs.


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## vert

Quote: 





> I think the theoretical ideal would to be using error-correction transfer into a buffer and clock it out on the DAC side.
> 
> Matter of time eh?


 
   
  I believe the Chord QBD76 already does this. Doesn't matter though, I still get a different sound from USB, toslink, and Halide Bridge. The best sound sound is via Halide Bridge, but I'm sure if I spent beaucoup bucks on a USB cable, or connected the toslink to a nice sound card instead of my Macbook's digital output, I'd get much better results that might comparable to the Halide.
   
  What I'm trying to say is even with the most sophisticated jitter rejection technology DACs, the sound is still sensitive to high jitter.


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## 00940

myinitialsaredac said:


> The issue I would argue still lies in using the computer clock. In that thread they specify that the chip if the jitter input is low will correct for theoretically all of it, however if it is high it will still cause phase modulation.
> 
> I think the theoretical ideal would to be using error-correction transfer into a buffer and clock it out on the DAC side.




I actually contacted Bruno Putzeys a few years back. He didn't provide me with his detailled measurements for the SRC4192 but said that the jitter had to be at least in the "tens of ns" for the chip to revert to classical PLL operation. Gordon Rankin measured a PCM2706 at 3400ps (3.4ns).


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## myinitialsaredac

Quote: 





00940 said:


> I actually contacted Bruno Putzeys a few years back. He didn't provide me with his detailled measurements for the SRC4192 but said that the jitter had to be at least in the "tens of ns" for the chip to revert to classical PLL operation. Gordon Rankin measured a PCM2706 at 3400ps (3.4ns).


 

 Interesting  
	

	
	
		
		

		
			




   
  Dave


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## myinitialsaredac

Quote: 





vert said:


> Quote:
> 
> 
> 
> ...


 

 You are correct that the chord drops it into a RAM buffer, I wonder if it allows for two way comm over usb for error correction. If it does this and you experience difference in sound between halide and optical it means that:
  optical is flawed compared to the usb as the usb would be considered "perfect"
  Halide bridge introduces error
   
  If it is not two way comm over usb with error correction then it means there are differences between the three inputs in terms of bit correctness and possibly jitter. 
   
  Intriguing. 
   
  Dave


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## 00940

Can't sleep so I found this : http://www.stereophile.com/content/music-hall-dac252-da-processor-measurements For this DAC using the SRC4192, jitter is vanishing for both SPDIF and USB as it should... Which tends to point where a not so good implementation of the SRC4392 as SPDIF receiver. 

At the point, I probably lost the OP... sorry for the OT.


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## twylight

imo go coax or toshlink
   
  Try both, I pick up a little noise on usb on a few systems and its not really any easier to work with.
   
  I think the "audio chips" behind coax and toshlink are a lot more mature than usb and its fair to say you can always get a good optical connection, but its possible a USB implementation (especially pre win7) might be slightly flawed.  Newer DACs and systems should have no issues...ive tried about 15-20.
   
  On my main rig and laptop I cannot tell the difference between the 3, so I use coax on the desktop and toshlink on the laptop.
   
  I am not a cable believer and I am not a good enough listener to hear jitter...so the math guys like to argue but I can NOT tell unless I am picking up noise.
   
  I have more trouble with tubes picking up room noise than USB anyways...
   
  Currently listening to my MHDT Havana -> speaker rig over coax from a win7 custom SSD based build...


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## VulgarDisplay

There should be no difference between USB and Optical because they are just sending a digital signal and not analog.  It's the same principal as HDMI cables.  A $3 hdmi cable is just as good or better than some $100 monster cable when it comes to digital signals.  The DAC turns 1's and 0's (digital data) into actual sound and it shouldn't get any interference of any sort because of this.  Perhaps being electrically connected through a USB cable could produce some noise, but I doubt it because the DAC is only looking for data and not any interfering signals.


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## mikemalter

Quote: 





vulgardisplay said:


> There should be no difference between USB and Optical because they are just sending a digital signal and not analog.  It's the same principal as HDMI cables.  A $3 hdmi cable is just as good or better than some $100 monster cable when it comes to digital signals.  The DAC turns 1's and 0's (digital data) into actual sound and it shouldn't get any interference of any sort because of this.  Perhaps being electrically connected through a USB cable could produce some noise, but I doubt it because the DAC is only looking for data and not any interfering signals.


 

 IMHO, It actally does matter, and it's not as simple as sending 1's and 0's because that is not all that there is.  Each cable has a completely different way to implement the signal, and it is nothing like computer networking.  Coax generally is best, followed by optical followed by USB.  Optical is divided by plastic and glass media where plastic is generally inferior to glass.  Also, the cable carries signals which get translated to those 1's and 0's and there are differences between cables too.


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## vert

Quote:


mikemalter said:


> IMHO, It actally does matter, and it's not as simple as sending 1's and 0's because that is not all that there is.


 

 For better or worse, in the real world, every little change makes a difference. Even with manufacturers who claim that their DAC rejects jitter to a negligible amount, you can easily test this by plugging in a toslink connection from a computer versus a high quality USB to spdif converter. The difference is quite large.


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## mikemalter

Quote: 





vert said:


> Quote:
> 
> 
> mikemalter said:
> ...


 

 'Zactly.


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## Slackboy72

Quote: 





uncleskeebow said:


> I prefer the sound of USB, (denser, warmer and more present) but if you have a noisy computer system, (interference) optical will isolate that noise and may in some cases give you better sound.


 


  You are taking the p*** aren't you? I can't detect sarcasm when it's typed.


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## ninjikiran

USB is universal, doesn't really change from system to system.  Toslink is ruled by implementation on PC end, and DAC end.  Different receivers, jitter rejection, ect ect ect.
   
  Whether the difference is audible its up to you.  I personally prefer good ol RCA/BNC(Spdif COAX) for audio, less in the signal path.
   
  Optical is more reliable over distance runs though~
   
  USB is the least reliable of the two since it is essentially a data cable rather than sending to a rather limited device(such as a dac).
   
  End of the day though.... if its audible to you stick with what you need.  If not go with what is that more elegant solution.


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## Hellrzr

Digital always sounds cleaner to me. With usb I somtimes get colloration or interference. Also, usb sounds really bad after 30ft or so.


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## Currawong

USB is digital, just FYI. 
   
  I think it very much depends on the DAC and implementation of the digital input.  With some I've noticed no different, but with others quite a bit where they were even sensitive to optical cables.


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## JRG1990

What about if the usb dac is mains powered, then the only signal the dac will want for the usb is the sound then how does that compare to optical,coax?.


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## grokit

I agree that there are pros and cons to USB and toslink-spdif when compared against each other. But coaxial-spdif is superior to both.


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## 00940

Quote: 





ninjikiran said:


> USB is universal, doesn't really change from system to system.  Toslink is ruled by implementation on PC end, and DAC end.  Different receivers, jitter rejection, ect ect ect.


 

 USB is not so universal. A USB DAC can implement two very different protocols (asynchronous or isochronous) and even two DAC using the same protocol can use receivers with very different performances. On the computer side, the quality of the USB signal can vary widely from a computer to another: some have very noisy ground lines, some have poor clocks, some don't have enough power and the USB connection suffer, and so on, and so on.
   
   
  Quote: 





			
				grokit said:
			
		

> /img/forum/go_quote.gif
> 
> I agree that there are pros and cons to USB and toslink-spdif when compared against each other. But coaxial-spdif is superior to both.


 
   
  There's no way to make such wide ranging statement, it's all a matter of implementation. In theory, USB in asynchronous mode +isolation would be much closer to technical perfection than coaxial spdif.


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## vert

Yeah, I think buying a DAC with a good USB implementation, like the M2Tech Evo, which I believe has the hiface module, would save money with not having to invest in a USB converter. I have experience with not so ideal DAC USB implementations, and the sound quality isn't that great.


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## grokit

Quote:


> Originally Posted by *grokit*
> 
> 
> 
> I agree that there are pros and cons to USB and toslink-spdif when compared against each other. But coaxial-spdif is superior to both.





 


  Quote: 





00940 said:


> There's no way to make such wide ranging statement, it's all a matter of implementation. In theory, USB in asynchronous mode +isolation would be much closer to technical perfection than coaxial spdif.


 
   

 I agree with you as far as USB's potential, but my experience in reality is where I am coming from. I have tried all *three *types of USB audio (synchronous, adaptive, and asynchronous) as well as coax and optical, these are my impressions, and I stand by them. I recognize there is much more out there than what I have experience with, but I try not to state opinions about gear or tech that I have not listened to personally.
   
  Also don't forget that these technologies are frequently combined, like when manufacturers add an asynchronous sample rate converter to an adaptive USB implementation, or use an adaptive USB chip that converts directly to S/PDIF inside the DAC. S/PDIF has been refined for many years in countless audio products, and this conversion technique can be a fairly good compromise between a simplistic adaptive implementation like the PCM270x chip from TI and a well done yet quite expensive asynchronous DAC design. There are inexpensive asynchronous DACs, but they usually require special software drivers which can bring about a whole new level of compromise.
   
  Please take a look at my signature for the appropriate qualifiers (IMO, IME, YMMV, etc.), they apply to everything I post and if you put my statement into that context, you will see that it's not so wide-ranging after all.


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## 00940

@Grokit: I know all that. The DAC I designed is using a pcm2707 followed by src4192. See my earlier posts in this thread for the rationale in picking those ICs and some links with interesting and somehow unexpected measurements.
   
   
  PS: I disabled signatures a long, long time ago. Not knowing what your experience is, not knowing what your opinion is based on, it's hard to say what I should make of your initial statement then.


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## grokit

Quote: 





00940 said:


> @Grokit: I know all that. The DAC I designed is using a pcm2707 followed by src4192. See my earlier posts in this thread for the rationale in picking those ICs and some links with interesting and somehow unexpected measurements.
> 
> 
> PS: I disabled signatures a long, long time ago. Not knowing what your experience is, not knowing what your opinion is based on, it's hard to say what I should make of your initial statement then.


 

 I C. Lol that you disabled other's signatures, yet you present your own. I love the Latin quotes btw, my current favorite is "Quis custodiet ipsos custodes?"


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## 00940

That signature must be 4 or 5 years old now... I disabled the signatures when the forum had layout problems and never went around the matter again. It's now fixed


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## Drake22

lol, so much BS in this thread.
   
  It's not an analog signal, so it doesn't have frequency response. It's digital, 1 or 0, and it doesn't matter how it's carried. 
  Noise? It's digital, there's no noise. The voltage usually used is either 2 or 5 volts. It's gotta be some hell of a distortion to turn 0 into 1, lol.
  Jitter? That is just latency and it doesn't have any effect on the character of the sound.
   
  There is no difference in sound between usb, optical, coaxial or hdmi. It's ridiculous to state otherwise.


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## scootermafia

Digital signals can be corrupted, despite being 1s and 0s.  There's all sorts of info here on head-fi if you know where to look.  Timing is important, too.


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## Prog Rock Man

There is a lot of suggestion of differences, but no proof any such is audible, apart from out and corruption of the signal, which is obvious.
   
  There is little to no evidence that jitter is audible except in extremes. In any case, jitter has nothing to do with the cable, except length which can corrupt the signal, which as above will be obvious.


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## mikemalter

Quote: 





drake22 said:


> lol, so much BS in this thread.
> 
> It's not an analog signal, so it doesn't have frequency response. It's digital, 1 or 0, and it doesn't matter how it's carried.
> Noise? It's digital, there's no noise. The voltage usually used is either 2 or 5 volts. It's gotta be some hell of a distortion to turn 0 into 1, lol.
> ...


 

  
  The issue is not so much in the 1 and 0 world.  It's the part before and after that.  Remember that all of this is happening in real time.  There is the conversion of the mechanical process of reading and interperting pits and then converting that information to a signal. Then the signal hits the wire and then it gets recieved and then converted.  So really a lot of this process is analog in nature.  Yes, in code it is all 1's and 0's, but there is a lot of non 1 and 0 stuff going on underneith the covers to get it there.
   
  I understand your perspective regarding differences in sound between different digital connections.  From my perspective, I have heard the difference clearly.


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## grokit

Quote: 





drake22 said:


> lol, so much BS in this thread.
> 
> *It's not an analog signal, so it doesn't have frequency response*. It's digital, 1 or 0, and it doesn't matter how it's carried.
> Noise? It's digital, there's no noise. The voltage usually used is either 2 or 5 volts. It's gotta be some hell of a distortion to turn 0 into 1, lol.
> ...


 

 Read and learn:
   
  http://en.wikipedia.org/wiki/Latency_%28audio%29
   
  http://en.wikipedia.org/wiki/Digital-to-analog_converter#Practical_operation


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## Drake22

Quote: 





grokit said:


> Read and learn:
> 
> http://en.wikipedia.org/wiki/Latency_%28audio%29
> 
> http://en.wikipedia.org/wiki/Digital-to-analog_converter#Practical_operation


 

 lol, 5k posts you have, are they all worthless like this?
   
  What about latency? It's just a delay, the sound itself doesn't change.
  What does this have to do with DAC when we are talking about cables, and how the data is carried INTO the dacs 
   

   Quote: 





mikemalter said:


> I understand your perspective regarding differences in sound between different digital connections.  From my perspective, I have heard the difference clearly.




  Seriously? ... If there are no corrupted cables or ports, and everything is connected correctly, there can't be any difference... Because it's just data that is carried into the dac, in where it get's converted into analog.
  DAC itself doesn't care in which way he receives the data. it's digital and that's it. No noise, no losses in digital transmission, so cable used doesn't matter. The only thing different between cables is the latency which just means you ll receive the signal sooner or later, but that doesn't change the sound itself.


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## 00940

A little knowledge is a dangerous thing.
   
  The DAC do care in which way he receives the data. In SPDIF (and in most USB protocols), the receiver chip recovers the system clock for the DAC from the timing of the incoming digital signal, through a PLL (a highly analog process). There is no dedicated clock line. The accuracy of this recovery affects the D/A process. That's where jitter enters the figure. Jitter is not delay, jitter is variation from the ideal timing. Some jitter can be traced back to the sending clock, some can be linked to the transfer interface (which is very much an analog thing). How much jitter is audible, that is debatable. However, the fact that jitter in spdif system can and do distort the analog output signal is well established. Some reading :
   
  http://www.scalatech.co.uk/papers/aes93.pdf
  http://www.wolfsonmicro.com/documents/uploads/misc/en/Jitter_performance_of_spdif_digital_interface_transceivers.pdf
   
  I think you misunderstand the point about noise too. By linking the ground of the computer and the ground of the device, noise can couple through (of course it depends a lot on how your DAC is designed and layed out and this is especially true for non isolated, usb powered devices). It doesn't affect the "transfer of the data". It can however affect the proper operation of the chips inside your dac and reduce their performances. Spdif receivers and DAC are mixed-signal ICs, they rely on a very clean supply and ground to fully meet their specifications. That's why most datasheets include suggested layouts and the like.


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## Drake22

Quote: 





00940 said:


> A little knowledge is a dangerous thing.
> 
> The DAC do care in which way he receives the data. In SPDIF (and in most USB protocols), the receiver chip recovers the system clock for the DAC from the timing of the incoming digital signal, through a PLL (a highly analog process). There is no dedicated clock line. The accuracy of this recovery affects the D/A process. That's where jitter enters the figure. Jitter is not delay, jitter is variation from the ideal timing. Some jitter can be traced back to the sending clock, some can be linked to the transfer interface (which is very much an analog thing). How much jitter is audible, that is debatable. However, the fact that jitter in spdif system can and do distort the analog output signal is well established. Some reading :
> 
> ...


 

 Okay.
  About jitter. So are you implying that if I send a same digital sinewave, one through spdif, and one through coaxial, I will get different analog output? No I won't. It will be same analog output, hence the sound will also be the same. It's gotta be an insane jitter to corrupt the signal. Perhaps usb 1.0 or 1.1 dac could cause it but that is obsolete anyway. If you receive different analog outputs from coax and spdif that would just mean that the DAC is defective.
   
  Again noise. No effect on the sound you confirmed that, thank you. Performance? Your sata bus of your harddrive gets the same noise. What kind of noise it has to be to corrupt the data?


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## 00940

Quote:


drake22 said:


> Okay.
> About jitter. So are you implying that if I send a same digital sinewave, one through spdif, and one through coaxial, I will get different analog output? No I won't. It will be same analog output, hence the sound will also be the same. It's gotta be an insane jitter to corrupt the signal. Perhaps usb 1.0 or 1.1 dac could cause it but that is obsolete anyway. If you receive different analog outputs from coax and spdif that would just mean that the DAC is defective.
> 
> Again noise. No effect on the sound you confirmed that, thank you. Performance? Your sata bus of your harddrive gets the same noise. What kind of noise it has to be to corrupt the data?


 
   
  - Coaxial is a particular methode of transmitting spdif, so is optical. What do you want to say exactly ?
   
  - It is mathematically proven that jitter, as small as it is, will corrupt the analog signal. Did you even read the first paper I linked ? Jitter as low as 150ps can be reliably measured in practice by analysing the analog output of a DAC. The question is how much jitter (and of which type, signal correlated or not) is needed to have an audible effect. To get an accurate analog signal, you need the correct samples values at the DAC input AND those values must be fed at the correct timing.
   
   
  - You realize that most USB DAC on the market are still using USB 1.1 receivers from TI, don't you ?
   
  - Don't put words in my mouth. Noise will not corrupt data but it will surimpose itself on the output signal and corrupt the analog signal. Example: depending on which PC I connect my usb DAC to (desktop, laptop on batteries, etc), I can get the noise floor (at the analog output) varying by as much as 6dB. With sensitive headphones, that can get pretty audible.
   
  - SATA has error correction and works only with purely digital devices. It doesn't care much about noise. However, you have to realize that USB receivers, SPDIF receivers and DAC are real time, mixed-signal devices. They are nothing like SATA. You put noise on the spdif receiver supply and the (analog) PLL performance is reduced which in turn reduce the accuracy of the system clock it's producing. You put noise on the DAC supply pins and you'll find part of it on the analog output. It's not a matter of corrupted data.
   
   
  You could read those too:
  http://www.tnt-audio.com/clinica/diginterf1_e.html
  http://www.stereophile.com/reference/1093jitter


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## Drake22

Quote: 





00940 said:


> Quote:
> 
> - Coaxial is a particular methode of transmitting spdif, so is optical. What do you want to say exactly ?
> 
> ...


 

 It can't be real-time...
   
  DACs don't do bit by bit real-time conversion, they use buffer. Look at this Julia screenshot:

   
  For example our buffer is 32 samples. When it gets all 32 samples it converts all of those into analog wave. And it doesn't care how much jitter there was, the only thing that matters is that all 32 samples are received in one tact.
  So it doesn't convert bit by bit in real time, but in those pieces of samples. And it will wait for all samples before converting them into analog wave.
  Let's say jitter is a random delay on each bit. But buffer waits until all bits of given sample are completed. It doesn't record the actual delay, it just waits until all 32 are ready and the converts that. Just one bit comes sooner, one later.
   
  You are talking about jitter in general, I am talking how it works with DACs.
   
  And USB 1.1 sucks big time, that's why you can't go higher than 44khz/16bit in usb 1.1.


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## 00940

Yeah, sure, right, whatever. You're mixing up latency buffers and what happens at the DAC side, don't know what spdif is, have no clue about I2S timing at a DAC's inputs. And yet you put your nonsense forward as if it were the Gospel.
   
  That latency buffer has nothing to do with jitter. It's there to avoid missing samples.
   
  To put it simply... Your data (0 and 1) is stored on your HDD. It must be transferred towards a digital transmission interface (and sometimes manipulated along the way). This interface will push data out at every tick of a clock. The DAC, hardware side, is expecting that data at every tick. See the problem now ? How can we guarantee that the digital transmission interface will always have something to send, at every tick ? What happens if the HDD slows down, if your computer is suddenly loaded down and the software manipulation of the data takes more time than expected ? That's where the latency buffer comes into play. We don't start the transmission immediately but we wait to have some samples on hand on a fast buffer next to the digital transmission interface so that we can preserve the real time flow needed for the spdif (or isochronous usb) connection to work.
   
  This explanation isn't too bad: http://support.apple.com/kb/ht1314


----------



## Drake22

Quote: 





00940 said:


> Yeah, sure, right, whatever. You're mixing up latency buffers and what happens at the DAC side, don't know what spdif is, have no clue about I2S timing at a DAC's inputs. And yet you put your nonsense forward as if it were the Gospel.
> 
> That latency buffer has nothing to do with jitter. It's there to avoid missing samples.
> 
> ...


 
   

 [size=medium]You've just answered yourself in your own statement. What about jitter when data goes into buffer with jitter already. And then inside the DAC you'll have the same jitter regardless if it's received from usb or optical if we have buffer in the end after transmission. So jitter won't affect the processing in any case.
   
  I know what an spdif is, in that sentence I meant toslink vs coaxial.
   
  Nonsense is saying that you can hear the difference between different digital carriers.
  I'll see my way out ​[/size]


----------



## 00940

Quote: 





> Originally Posted by *Drake22* /img/forum/go_quote.gif
> 
> What about jitter when data goes into buffer with jitter already.


 

 ??? Once the data goes into a big enough buffer, there is no jitter to speak about anymore. The sample simply wait for their turn to leave the buffer, at the tick of the clock. Once they leave the buffer... well go back reading TNT's article, it's decent enough.
   
  Quote: 





> Originally Posted by *Drake22* /img/forum/go_quote.gif
> 
> And then inside the DAC you'll have the same jitter regardless if it's received from usb or optical if we have buffer in the end after transmission.


 
   
  Who told you had a buffer in the end after the transmission ? Because you don't in most cases or not big enough to remove the jitter. And of course, there's still the question of the system clock DAC side. How do you generate it properly, knowing that it must be somehow linked to the source time domain ? Because, even if you have a buffer on the receiving side, the data still has to leave it 1/on proper timing so that the D/A conversion is done properly, 2/ in some link with the source clock so that you don't empty or fill in too quickly your buffer.
   
   
  edit:
  
  Quote: 





> Originally Posted by *Drake22* /img/forum/go_quote.gif
> 
> Nonsense is saying that you can hear the difference between different digital carriers.


   
  I'm not making the case you can hear it... I'm making the case that you can objectively measure it and that there are sound engineering reasons for that. Differences in between digital carriers are cold, hard facts as far as SPDIF and USB audio protocols are concerned.
   
  Audibility is another thing.
   
   
   
  This whole discussion remind me of this :
   
  http://www.aoselectronics.com/jitter_article.html


----------



## Drake22

Quote: 





00940 said:


> ??? Once the data goes into a big enough buffer, there is no jitter to speak about anymore. The sample simply wait for their turn to leave the buffer, at the tick of the clock. Once they leave the buffer... well go back reading TNT's article, it's decent enough.
> 
> Who told you had a buffer in the end after the transmission ? Because you don't in most cases or not big enough to remove the jitter. And of course, there's still the question of the system clock DAC side. How do you generate it properly, knowing that it must be somehow linked to the source time domain ? Because, even if you have a buffer on the receiving side, the data still has to leave it 1/on proper timing so that the D/A conversion is done properly, 2/ in some link with the source clock so that you don't empty or fill in too quickly your buffer.


 

 *on the way out*
   
_"??? Once the data goes into a big enough buffer, there is no jitter to speak about anymore."_
   - that's my point you've been disagreeing about. And that's gotta be some insane jitter to miss the buffer, hence why there are different buffer settings to prevent that.
   
_"Who told you had a buffer in the end after the transmission ?"_
   - every DAC has a buffer.. find me a DAC with no buffer and 0 latency that does bit by bit real time conversion
   
  Mate, it seems that you have such a superDAC that has no latency buffer. In that case, you are awesome.
  Seriously speaking, you are the one with no clue.


----------



## 00940

Quote: 





drake22 said:


> _"??? Once the data goes into a big enough buffer, there is no jitter to speak about anymore."_
> - that's my point you've been disagreeing about. And that's gotta be some insane jitter to miss the buffer, hence why there are different buffer settings to prevent that.
> 
> _"Who told you had a buffer in the end after the transmission ?"_
> ...


 


 Here is a simplified typical digital chain:
   
  HDD/CD/DVD - latency buffer - spdif emitter - SPDIF cable - spdif receiver (dir 9001 for ex.) - I2S lines - DAC chip (PCM1794 for ex.).
   
   
  Jitter is a problem that starts with the spdif emitter. Explain me how the latency buffer will change anything.


----------



## Drake22

Quote: 





00940 said:


> Here is a simplified typical digital chain:
> 
> HDD/CD/DVD - latency buffer - spdif emitter - SPDIF cable - spdif receiver (dir 9001 for ex.) - I2S lines - DAC chip (PCM1794 for ex.).
> 
> ...


 

 Your DAC has a buffer.
  After "spdif receiver (dir 9001 for ex.)" there is buffer. Your receiver collects the data and packs it into the buffer. It then passes it through i2s bus to the DAC who converts that buffer into analog wave on the way out.
  This makes any jitter that happens during spdif transmission worthless, simply because of the buffer after that, as it waits for bits and doesn't care if they come with jitter or not.
   
  Taking a DAC that has for example a toslink input and a coax input, and sending the same signal through them will result in the same graph on the output. So if everything working correctly - no sound difference.
  If people say they can hear difference between them it's either:
    a) they lie for whatever reasons (they bought an expensive cable, self-suggestion or whatever)
    b) their device is defective


----------



## 00940

Quote:


drake22 said:


> Your DAC has a buffer.
> After "spdif receiver (dir 9001 for ex.)" there is buffer. Your receiver collects the data and packs it into the buffer. It then passes it through i2s bus to the DAC who converts that buffer into analog wave on the way out.
> This makes any jitter that happens during spdif transmission worthless, simply because of the buffer after that, as it waits for bits and doesn't care if they come with jitter or not.


 

 It has been fun but now it's time to wake up. Take the blue pill and open your eyes. There is no such buffer. Read TI's datasheets. Read this http://www.tnt-audio.com/clinica/diginterf1_e.html (obviously, you still haven't done it). Everything you're saying is wishful thinking.
   
  You know why ? Because consummers don't want delays in playback devices and because IC manufacturers find big buffers too costly. Read this carefully: http://www.eetimes.com/design/audio-design/4009467/The-D-A-diaries-A-personal-memoir-of-engineering-heartache-and-triumph (there are three parts, don't skip the 2nd).
   
  If you want, you can read this: http://www.wolfsonmicro.com/documents/uploads/misc/en/A_high_performance_SPDIF_receiver_Oct_2006.pdf   It's maybe the closest thing to what you're describing. But you have to know that even the wolfson current spdif receivers don't use that technology according to Wolfson's representatives.
   
  Tell me... why do engineers at Wolfson bother trying to improve their SPDIF receivers if everything is already perfect ?
   
  Tell me... why can we measure, through the analog outputs of a DAC, differences in jitter if connected through different methods ? Logically speaking, it's only because there are differences showing up in the analog waveform, isn't it ? Or are the guys from MSB wrong ? http://www.msbtech.com/support/JitterPaper.pdf
   
   
  You might be right that differences in sound are non-audible in between digital cables (the noise question set apart obviously). You are completly wrong in the reasons why it might be so.


----------



## Prog Rock Man

Interesting debate, but surely the OPs question is which sounds better USB or optical, not which one works better? My impression is that when they work, they work as well as each other and that without the audibility testing that is banned from much of the forum, you cannot actually tell which, if any sounds better.


----------



## 00940

For those questions, I stand by my post #8


----------



## Prog Rock Man

Quote: 





00940 said:


> The V-DAC uses an asynchronous sample rate converter (hence the claim for ultra low jitter). The sonic differences in between digital inputs should be zero.
> 
> The arguments for optical over USB would be isolation of the computer ground (can induce some noise in some designs) and the fact that your soundcard optical output is probably not as dependant as USB from the CPU. USB can be laggy when there is an heavy load on the CPU (not so much of a problem with modern CPU). The drivers for your soundcard might also be better optimized for gaming than the generic USB audio drivers.


 

 Post Non 8. It contains suggestions of why there may be a difference and instances where there a fault could be present. That is not meant to be a criticism. I am just making the point that suggestion is used a lot to explain supposed differences in sound, that then vanish when subjected to an audibility test. Then, if there is a fault, such as a ground loop, then that is not so much a reason as why one type of cable is better than another, it is just an issue than can arise with certain types of connection.


----------



## Justin Uthadude

My small experience goes with the people who've said it depends upon the individual dac's implementation of the two interfaces. I just bought a v-link usb-s/pdif converter. The sound I'm getting out of my same dac through its coax input is noticeably better than the sound I was getting out of its usb input. Admittedly, the converter is using asynch mode to get the usb signal, so that may play into the equation.


----------



## 00940

@Prog Rock Man: I see your point. However, as I was rather making the point that, in this case, there should theoretically be no differences in sound, I would be very pleased if an audibility test proved me right on this. 
	

	
	
		
		

		
		
	


	



   
  Of the three potential problems, two (CPU load and drivers) should be easily apparent in games by just checking the FPS. The last (ground isolation) is more tricky.


----------



## vert

Optical from my Macbook is the lowest performing connection method to my DAC. Your performance from your sound card might be better. I'm willing to bet a V-link is going to be much better, as in no comparison, than the optical from your PC. It'll also prob be better than going straight USB into the DAC.


----------



## grokit

Quote: 





drake22 said:


> lol, 5k posts you have, are they all worthless like this?
> 
> What about latency? It's just a delay, *the sound itself doesn't change*.


 
   
  Evidently you are incapable of learning and have to make things personal. To quote from one of the links from which you evidently know better than:
   
"Instead of impulses, usually the sequence of numbers update the analogue voltage at uniform sampling intervals.
These numbers are written to the DAC, typically with a clock signal that causes each number to belatched in sequence, at which time the DAC output voltage changes rapidly from the previous value to the value represented by the currently latched number. The effect of this is that the output voltage is _held _in time at the current value until the next input number is latched resulting in a piecewise constant or 'staircase' shaped output. This is equivalent to a zero-order hold operation and *has an effect on the frequency response* of the reconstructed signal."


----------



## Drake22

Quote: 





grokit said:


> Wow, what a Richard's nicknamish post. Evidently you are incapable of learning and have to make things personal. To quote from one of the links from which you evidently know better than:
> 
> "Instead of impulses, usually the sequence of numbers update the analogue voltage at uniform sampling intervals.
> These numbers are written to the DAC, typically with a clock signal that causes each number to belatched in sequence, at which time the DAC output voltage changes rapidly from the previous value to the value represented by the currently latched number. The effect of this is that the output voltage is _held _in time at the current value until the next input number is latched resulting in a piecewise constant or 'staircase' shaped output. This is equivalent to a zero-order hold operation and *has an effect on the frequency response* of the reconstructed signal."


 
  Your posts are worthless. Probably all of them are it seems.
  
  What does your paragraph have to do with latency? Your paragraph talks about general quality loss from converting squarewave signal into sinewaves. Of course, it has an effect on FR, and what about it? 
   
  Mate, you need to get a referencing lesson from 00904, and stop linking worthless wiki posts that are off topic anyway.


----------



## Currawong

From a practical standpoint, the issues with USB, such as noisy power from a computer, can be fixed by using an isolated power supply. The limitations of optical can not, at least by the end user.  The Lampizator site has some images from a scope of an optical S/PDIF signal and it is far from being a square wave.


----------



## MacedonianHero

Quote: 





currawong said:


> From a practical standpoint, the issues with USB, such as noisy power from a computer, can be fixed by using an isolated power supply. The limitations of optical can not, at least by the end user.


 


  Good point...as well a USB DAC that can strip out the time domain and re-clocks it back in with minimal jitter can then take USB sound to the next level (as experienced when I moved up from my PS Audio DLIII to my Cary XCiter). The biggest difference in SQ was via USB input.


----------



## grokit

Quote: 





drake22 said:


> Your posts are worthless. Probably all of them are it seems.
> 
> What does your paragraph have to do with latency? Your paragraph talks about general quality loss from converting squarewave signal into sinewaves. Of course, it has an effect on FR, and what about it?
> 
> Mate, you need to get a referencing lesson from *00904*, and stop linking worthless wiki posts that are off topic anyway.


 

 At least I cite my sources, and I'm not attacking you personally 
	

	
	
		
		

		
			




   
  I'm just trying to point out misinformation, like when you posted this gem:
   
  "Jitter? *That is just latency and it doesn't have any effect on the character of the sound.*
  There is no difference in sound between usb, optical, coaxial or hdmi. It's ridiculous to state otherwise."

   

  What my previous post did was describe the part of the audio clocking dynamics that can create jitter, and cause errors in the resulting sound. I also think that optical s/pdif carries more jitter than coaxial s/pdif, and that jitter can absolutely have an effect on the character of the sound. These are my opinions, and I can cite multiple references them. But somehow I have a feeling that would be a waste of time, as they would be lost on you. 

   

  I don't know who "00904" is but I haven't found anything wrong with what *00940* is saying. From the Apple page he referenced, it says that we need to increase buffer size to decrease latency, because *latency can have a negative effect on the sound*:

   

  "Lower buffer sizes and higher sample rates may result in less monitoring latency, but these settings require more computational power. If you set the I/O buffer size to a value too low for your computer to handle smoothly, you may hear dropouts, clicks, pops or other artifacts in the audio."

   

  Can you please cite a factual reference for your opinions, without the unnecessary hostility?

   

  Peace Out


----------



## leeperry

grokit said:


> unnecessary hostility?


 
   
  Yeah, ppl act like fourth grade kids when it comes to audio...I don't think I'll ever understand why.
   
  Anyway, going fully async does seem to changes things: http://www.lavryengineering.com/lavry_forum/viewtopic.php?f=1&t=979
   
_"the measurements are virtually identical with any of the inputs- USB, XLR, RCA, or Optical"_
   
  A friend of mine has ripped SACD's digitally like this:
   
_1) DSD internally converted to a 24bit/88.2KHz PCM stream by the Oppo DV-980H player
 2) The PCM stream is conveyed into a high-quality HDMI 1.3 cable
 3) The HDMI is connected to an Octava 1x2 HDMI Distribution Amp with Toslink Out
 4) The PCM stream is splitted into a toslink cable
 5) The toslink cable is connected to a M-Audio Transit USB adapter
 6) The PCM stream is captured by Cockos Reaper 3.1x using the M-Audio ASIO drivers.
 7) Final track splitting (no other editing is involved) is done in Reaper._
   
  Yeah, it did go through POF toslink...and yeah, it sounds amazing! I could even provide a 30" sample.
   
  Our empirical experience is based on cheapo DAC's with poor jitter tolerance, and isochronous USB transports. Or maybe they didn't measure the right stuff, but if there's someone I trust in the digital audio world that's Mr Lavry.


----------



## Prog Rock Man

I think that the whole of Leeperry's link to Lavry Engineering is worth reproducing....
   
  ".I consult your expertise in this area as this place are backed by solid scientific papers. Having read the skin effect on cables as well as comments by Mr Lavry regarding over-priced digital cable, I would like to know whether this applies the same for USB Cable.

 Does a reasonably well built usb cable perform as well as it's much more expensive counter-parts? There have been lots of review stating 3 and even 4digits priced usb cables transforming a sound system. However, it's my understanding that async code place the re-clocking task on the converter and thus it does not depend on the quality of the usb cable...

 We have seen no scientific evidence to this effect. When the DA11 is tested, the measurements are virtually identical with any of the inputs- USB, XLR, RCA, or Optical. And we do not use an expensive esoteric cable for the testing. I cannot comment on other designs.

 There are other factors when using a USB source that are more likely to have an audible effect on the audio; like software interactions or other high-bandwidth devices sharing the same USB port.

 There is definitely one way in which these expensive USB cables do transform the sound systems- by making the system more expensive.

 Brad Johnson
 Lavry Engineering Technical Support"
   
  So, according to Lavry, if you buy a DA11 it makes no difference, to answer the OPs question. I suspect that is true of other DACs as well.


----------



## grokit

Quote: 





leeperry said:


> Yeah, ppl act like fourth grade kids when it comes to audio...I don't think I'll ever understand why.


 

  It seems to happen mainly in three areas: Sound science/DBTs (especially cables), digital/jitter/1's and 0's, and "to Apple or not to Apple"
   
  What I have learned so far is if you have an asynchronous master audio clock outside the computer you can eliminate many problems related to jitter, including whatever your method of transmission is; USB, spdif or what have you. If you are using adaptive or even isochronous protocols where the computer controls the master audio clock you are asking for trouble, and USB/toslink can expose the most flaws compared to other methods of digital transmission. But the DA11, like the Streamlength codec and other asynchronous master clocking/re-clocking methods are providing a remotely-based master audio clock and are therefore reducing or eliminating the possibility of timing errors.
   
  I love it when the manufacturer/designers weigh in. I am very interested in what Mr. Johnson/Lavry has to say about these matters, and am a big fan of Gordon Rankin as well. But I also realize that these guys have something to sell, and they (most likely) truly believe that their own product(s) has solved these issues in the very best way. I believe what they are saying for the most part but at the same time I also realize that every "problem" in this area has more than one possible solution.
   
  As far as cables go, there are so many people that report clear auditory differences with a "better" USB cable that I want to try one sometime (within reason of course). Maybe with better DACs it doesn't matter as much; I would like to conduct my own experiments regarding this sometime, on my own setup(s) because that is really all that matters to me. When I replaced my cheap toslink cable with a glass one it improved the sound out of my DAC2 more than adding an external clock did, but on my Mini-i the results were pretty much the opposite.
   
  I know I have come around to the benefits of upgraded headphone audio cables (against my better judgement), so it makes sense to me to give the digital cable an upgrade at some point to see if I can hear the difference for myself. It would be a final tweak though, after everything else is optimized. If it's placebo so be it, as long as I get some imagined benefit that works for me it's all good!


----------



## mikemalter

Quote: 





grokit said:


> It seems to happen mainly in three areas: Sound science/DBTs (especially cables), digital/jitter/1's and 0's, and "to Apple or not to Apple"
> 
> What I have learned so far is if you have an asynchronous master audio clock outside the computer you can eliminate many problems related to jitter, including whatever your method of transmission is; USB, spdif or what have you. If you are using adaptive protocols where the computer controls the master audio clock you are asking for trouble, and USB/toslink will expose the most flaws compared to other methods of digital transmission to the DAC. But the DA11, like the Streamlength codec and other asynchronous master clocking/re-clocking methods are providing a remotely-based master audio clock and are therefore reducing or eliminating the possibility of timing errors.
> 
> ...


 

 Grokit, another issue is the quality of the equipment the person has.  If it is higher quality equipment, than things like the differences in digital sources will be more apperent as will differences in cables themselves.  It seems to me that the people making the more childish type of arguments don't have access to equipment that will perform at the level necessary to detect changes in sources or cables.


----------



## grokit

Quote: 





mikemalter said:


> Grokit, another issue is the quality of the equipment the person has.  If it is higher quality equipment, than things like the differences in digital sources will be more apperent as will differences in cables themselves.  It seems to me that the people making the more childish type of arguments don't have access to equipment that will perform at the level necessary to detect changes in sources or cables.


 

  Totally agreed, I was just editing my post to reflect that better components can either accentuate these differences or even sometimes negate them, depending upon what the tweak is. All of this is very inter-dependent with the quality of the component(s), and the rest of the setup as a whole. You're to quick for me!


----------



## ROBSCIX

For both USB and Optical there is another aspect that definitely influences both and that is the Tx/Rx modules in both the source and destination devices.
  If the units are poor quality they can affect the signals in negative ways.  I think many times people do not consider these units although they do have a significant affect on the signals.


----------



## leeperry

grokit said:


> it makes sense to me to give the digital cable an upgrade at some point to see if I can hear the difference for myself. It would be a final tweak though, after everything else is optimized.


 

 All cables color the sound drastically IME....but I run an adaptive USB transport, so you know.
  Anyway, this thread is useless w/o samples 
	

	
	
		
		

		
		
	


	



   
04 - Thriller.flac
06 - Billie Jean.flac
   
  These are 30" 88.2/24 samples of the Thriller SACD, and I've got no problem believing this quote: http://www.avsforum.com/avs-vb/showpost.php?p=19187517&postcount=6
   
_"The Thriller SACD is the best version ever released of the album. Steve Hoffman has said it sounds the closest he has heard to the master tapes."_
   
  Toslink sounds horrid on my PCM1793/DIR9001/CS8414/WM8804 gear...the trebles are mushy, the sound is audibly smeared(poor clock extraction?). Nothing works better than an uber-short coax wiring IME. The 2 samples I just posted did go through a 1m POF toslink cable, would you say that they sound "_bad_"™?


----------



## mikemalter

Quote: 





robscix said:


> For both USB and Optical there is another aspect that definitely influences both and that is the Tx/Rx modules in both the source and destination devices.
> If the units are poor quality they can affect the signals in negative ways.  I think many times people do not consider these units although they do have a significant affect on the signals.


 

 For optical cables, glass is better than plastic and it puts it in the range of coax.  Get glass fiber, it's a little more expensive, but there are some deals out there.


----------



## ROBSCIX

Quote: 





mikemalter said:


> For optical cables, glass is better than plastic and it puts it in the range of coax.  Get glass fiber, it's a little more expensive, but there are some deals out there.


 
  I have quite a few optical cables, some glass and some plastic.


----------



## mikemalter

Quote: 





robscix said:


> I have quite a few optical cables, some glass and some plastic.


 

 Rob, what are your experiences with optical vs. plastic?


----------



## googleborg

Quote: 





mikemalter said:


> Grokit, another issue is the quality of the equipment the person has.  If it is higher quality equipment, than things like the differences in digital sources will be more apperent as will differences in cables themselves.  It seems to me that the people making the more childish type of arguments don't have access to equipment that will perform at the level necessary to detect changes in sources or cables.


 


  what, like, _dedicated measuring equipment_?
   
  oh! you meant a fancy hi-fi! 
	

	
	
		
		

		
		
	


	



   
  stop embarrassing yourself.


----------



## grokit

< Sounds like jealousy to me 
	

	
	
		
		

		
			




   
  I've seen $200+ plastic, I mean _polymer _cables, and $35 glass ones. I went from a decent quality plastic fiber toslink cable to a similarly-priced glass one, and I noticed a difference. But that was with my highest-quality DAC, the NOS Bel Canto DAC2, which has its own master clock that is optimized for s/pdif: "A local crystal oscillator reference clock drives the DAC directly for minimal jitter."
   
  Where I found the Firestone Bravo re-clocker to make a profound difference when used with my balanced Mini-i DAC, on the Bel Canto it had no effect. But what did have a notable effect was switching to the glass cable, that really tightened things up with the BC. So I tried the glass cable with the Mini-i, and it did not make a difference when compared to the cheaper plastic one.
   
  The difference in the two results, I believe is the superior "local crystal oscillator reference clock" in the Bel Canto, making it sensitive enough to resolve a difference in the heightened transmission quality of the glass. I've never tried a high-dollar "advanced polymer" toslink cable though, they are likely very nice as well and probably more durable. 
   
  Another difference would be that the Mini-i just sounds better when using its coaxial input, rather than its toslink or USB input. So the Bravo is fully utilized with it, providing a re-clocking function along with an s/pdif coaxial converter. The Bel Canto sounded equally good with coaxial and toslink, but better with the glass toslink.
   
  These results are all highly system-interdependent; a tweak that improves one component won't automatically have the same result on another one. We have to look at the DAC conversion as a collection of processes to discover where the deficiency may lie.
   
  Like Robsix said, the Tx/Rx modules are part of the story as well, just like every other link in the digital-analog conversion chain; they all matter in regards to the final product, the sound that hits our ears.


----------



## 00940

Quote: 





grokit said:


> < SBut that was with my highest-quality DAC, the NOS Bel Canto DAC2, which has its own master clock that is optimized for s/pdif: "A local crystal oscillator reference clock drives the DAC directly for minimal jitter."
> 
> Where I found the Firestone Bravo re-clocker to make a profound difference when used with my balanced Mini-i DAC, on the Bel Canto it had no effect.


 

 Not very surprising... the Bel Canto DAC2 uses an AD1896 asynchronous sample rate converter (just like the Benchmark DAC1). Any jitter at the input is highly attenuated and transformed into noise (as explained here : http://www.diyaudio.com/forums/digital-source/28814-asynchronous-sample-rate-conversion.html , by an engineer behind such chip's design). A same clock drives the ASRC output and the DAC input.
   
  I really like that DAC design btw and I used it as a a starting point for my own DAC. The analog I/V stage is using the very nice sounding ths4130 fully-balanced opamp and a fully DC coupled output stage.


----------



## leeperry

ASRC is just upsampling, and I sure as hell don't want my music to pass through a mandatory upsampling pass...all it'll do is increase THD+N and make the sound brighter. It's just a dirty way to get rid of jitter, the poor man's solution..ghetto style. The best Sabre implementations are said to disable its internal ASRC, which many ppl call colored and clinical sounding to the utmost.
   
  Glass toslink very much matters if you use cheap gear in realtime, it's been thoroughly discussed in this thread: http://www.head-fi.org/forum/thread/459752/which-is-the-best-optical-cable-under-300/135#post_6599838
   
_"I swapped out the 1M Monster cable with the 1M Dayton GOC-3 Glass Toslink cable and viola!

 The cymbal smashes became three dimensional with a better sense of space, they were better located in the sound stage and no longer too bright and came to life. The bass surprisingly tightened up even further with better definition."_
   
  The best sounding cable I found is this one: http://www.parts-express.com/pe/showdetl.cfm?Partnumber=180-951
   
  But it doesn't hold a chance against a short coax IME. There are physical reasons as to why glass sounds better than POF:
  -lower constringence/optical distortion, much higher OTF/MTF sharpness than plastic
  -much higher bandwidth
   
  All this allows for a much smoother clock extraction in the end.


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## 00940

The Bel Canto DAC2 bright ? Come on... 
	

	
	
		
		

		
		
	


	




  http://www.soundstage.com/revequip/belcanto_dac2.htm (and I globally agree with that review). The Benchmark DAC1 uses the same chip and is more clinical, but it's pretty much down to its analog stage, nothing to do with upsampling.
   
  As for an ASRC being "just upsampling", sure, did I ever said the contrary ? Actually, asynchronous sample rate conversion is a more accurate term than the much marketing-abused "upsampling" which has different meanings according to whom you're asking to define it.
   
  What are the "best Sabre implementations" which are "said" to disable upsampling so that I don't buy them ?


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## leeperry

Well, the holy Benchmark DAC-1 is full of horrid sounding 5532's and 4562's, together w/ ASRC many ppl call it bright and shrill: http://ravenda.wordpress.com/2009/08/14/audiogddac19/
   
_"Benchmark DAC1 ($995 ~ harsh, flat out boring)"_

 That's what crappy dual opamps and ASRC do...just like in the DacMagic(that's fed off a noisy SMPS too), miracles do not occur in the audio world.
   
  Apparently, Wavelength's Sabre implementations do disable its internal ASRC: http://www.usbdacs.eu/urzadzenia/Denominator/6.html/
   
  Well, it's great that you enjoy upsampling...it's still a dirty way to fix the jitter problem. You can easily measure the THD/THD+N drastic increases when upsampling, using WaveSpectra. Upsampling does sound clearer, but it drastically colors the sound too...and what sounds great for a day/week can become dead boring after a few weeks/months IME. There are smarter ways to fix the jitter problem, that don't rape the waveforms at that.


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## 00940

The Bel Canto DAC2 has ASRC and is warm sounding. The Benchmark DAC1 has ASRC and is rather cold. From where do you draw the conclusion that the coldness is coming from the ASRC ?
   
  "Many people" hear many things. I'm not forced to agreed with them. I've heard the Benchmark and it certainly wasn't harsh. Clean sounding, yes. Harsh, no.
   
  Drastic increase in THD ? That's nonsense. Could you please check the THD+N of the Benchmark you were just dismissing ?


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## leeperry

nonsense? that upsampling increases THD+N? you're kidding, right? I did some experiments on the Reclock forum a while ago: http://forum.slysoft.com/showpost.php?p=227530&postcount=4000
   
  You can do it too, make a 1kHz/10kHz 44.1kHz test tone wav file using SineGen, upsample it to 192kHz and look at the drastic increase in THD+N in WaveSpectra.
   
  Upsampling doesn't come for free...it can somehow "smooth" the waveforms(even though it's highly debated), but it DOES increase distortion and colors the sound. Anyway, we're rather OT at this point and I've got no interest in discussing those matters tbh. We all seem to be happy w/ the gear we own, it's all that matters in the end


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## ROBSCIX

Quote: 





mikemalter said:


> Rob, what are your experiences with optical vs. plastic?


 
  I have never really sat down and tried to compare the two in any serious way.  I find myself reaching for coaxial cables more just out of habit.
  I have many associate that prefer glass if that is what you are asking.


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## Justin Uthadude

Quote: 





leeperry said:


> ...the drastic increase in THD+N....


 

 three one-thousandths of one percent?
  drastic?
   
  Add to that mind-boggling amount of distortion the fact that I'm listening to it with 60 year old tubes.
  There's no hope for guys like me


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## 00940

If you open the SRC4192 datasheet, you''ll find a nice summary of why I don't loose sleep over ASRC induced distortion. As a remainder, -140db is 0.00001%
   

   
  Even the best DAC chips will have more distortion, especially as THD goes up with frequency. However, I'll gladly trade the extra distortion against the easier analog filtering I get by upsampling to 96khz. See for example the PCM1794:


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## leeperry

justin uthadude said:


> three one-thousandths of one percent?
> drastic?
> 
> Add to that mind-boggling amount of distortion the fact that I'm listening to it with 60 year old tubes.
> There's no hope for guys like me


 

  Well, yeah...w/ your 1% THD amp, the audiophile fever will quickly end up in inaudible differences IMHO. POF toslink might not be your biggest problem right now 
	

	
	
		
		

		
		
	


	



    


00940 said:


> Even the best DAC chips will have more distortion, especially as THD goes up with frequency. However, I'll gladly trade the extra distortion against the easier analog filtering I get by upsampling to 96khz.


 
   
  Well, who wants to add noise to the incoming signal on purpose? once it's dirty, you can't clean it...
   
  Yep, the whole idea behind upsampling is that the oversampling curves will provide a more gentle filtering...each to his own, as usual. Each time I've heard upsampling, it was brighter(due to the increased THD+N) and sounded utterly colored to my ears. Hopefully, in a $1.5K DAC it'll sound amazing \o/


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## 00940

@leeperry :You know what's funny ? Somehow we should stay at 44.1Khz and throw DVD-A and SACD to the bin. Distortion figures would be lower than with the high-resolution formats. Higher THD doesn't come from upsampling per se but from the inability of the DACs to cope with high frequency signals as well as with the good old 44.1Khz.
   
  @Justin Uthadude: I somehow agree. Chasing the last digits of THD isn't a goal by itself. My favorite amp to this day is still the Audiovalve RKV, not the most neutral ever, and my headphones of choice are the AKG K340, which, I'll admit it, have their coloration. However, I still want a clean source. I've enough trouble with amps and headphones to voice my system as pleases me. I don't want another bottleneck or link to care about.


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## leeperry

00940 said:


> You know what's funny ? Somehow we should stay at 44.1Khz and throw DVD-A and SACD to the bin. Distortion figures would be lower than with the high-resolution formats. Higher THD doesn't come from upsampling per se but from the inability of the DACs to cope with high frequency signals as well as with the good old 44.1Khz.


 

 What's funny is that you compare interpolated upscaled audio data and genuine hi-res files...they have nothing in common whatsoever. One has more genuine informations, the other is filled w/ bogus interpolated data. But hey, you can upsample 44.1@384kHz if you like...I'm sure the experts on the amazing computeraudiophile.com website do exactly that.
   
  Many TI DAC's lower their upsampling rate >95.9kHz, but this is not the case w/ AKM chips. Anyway, I'm from Team Oversampling, you're from Team Upsampling...let's just agree to disagree. This is OT discusson anyway


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## 00940

We're OT since a long time already and the OP is long gone.
   
  You mean the website where they discuss the merits of respective usb cables ? 
	

	
	
		
		

		
		
	


	



   
  It was a jest, because 16 to 24bits saves the day as far as high-resolution formats and THD+N are concerned. However, it is also quite serious. Many professionals, including Dan Lawry (in here: http://www.lavryengineering.com/documents/Sampling_Theory.pdf ), have pointed out that 192khz is completly useless. 24bits/96khz is already overkill. The DAC IC don't care if they receive a file sampled "natively" at 192khz or upsampled. In both cases, they will add more distortion of their own than if they had received files sampled at 96 or 44.1 khz, just have a look at the datasheets.
   
  Bogus data is perfectly fine, when the differences in the analog waveform end up under -140db (Benchmark measured distortion due to their use of an ASRC at -135db, http://www.hydrogenaudio.org/forums/lofiversion/index.php/t75505.html ).


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## Justin Uthadude

OK, back OT
  I think it still boils down to implementation. Good USB dac beats a dac with a poor s/pdif interface, and vice versa.
  Beware, using the better (for you) interface could be another one of those Head-Fi 'sorry about your wallet's as the following excerpt suggests:
   

  [size=10pt]Another interesting thing about audibility of jitter is it's ability to mask other sibilance in a system. Sometimes, when the jitter is reduced in a system, other component sibilance is now obvious and even more objectionable than the original jitter was. Removing the jitter is the right thing to do however, and then replace the objectionable component. The end result will be much more enjoyable.[/size]
[size=10pt]Jitter can even be euphonic in nature if it has the right frequency content. Some audiophiles like the effect of even-order harmonics in tubes, and like tubes, jitter distortion can in some systems "smooth" vocals. Again, the right thing to do is reduce the jitter and replace the objectionable components. It is fairly easy to become convinced that reducing jitter is not necessarily a positive step, however this is definitely going down the garden path and will ultimately limit your pursuit of audio nirvana.[/size]
[size=10pt]Sibilance in a system caused by preamp, amps and other components and cables can also be so high that changes in jitter are not very audible. This is why there is such contention on the web forums about jitter and its importance. What matters in the end is if you are happy with the sound of your system, and whether or not you can hear this distortion.[/size]


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## mikemalter

Quote: 





00940 said:


> We're OT since a long time already and the OP is long gone.
> 
> You mean the website where they discuss the merits of respective usb cables ?
> 
> ...


 

 I have some 24bit 192kHz music I downloaded from HD Tracks.  I also have music server software on my laptop that has drivers for 24/192 along with a digital output.  The music played 24/192 is much better sounding by many orders of magnitude with more detail than anything else I have in my library.  The music recording business is not my business or area of expertise, however I do have an experience of its results.  And I am shifting my entire music playback system architecture to feed my DAC from a music server which I intend to load with 24/192 music because of superior results.  I don't understand the metrics you are using, but (not to be sniping at you, please don't take it that way) I don't listen to metrics, I listen to music and I will move toward what sounds best and give me the most enjoyment.
   
  What is your experience with higher resolution music?  Have you experienced it to be just about the same as "regular" music?


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## 00940

I've been exposed to some. When I was listening to HD music, it was as a rule better than usual... because the recordings had been remastered with greater attention. This by itself might be a very good reason to get into high-rez formats.
   
  Keep in mind that it's very hard to fully isolate those 4 variables:
   
  - quality of the recording/mastering
  - 16 vs 24bits
  - 44.1khz vs 96khz or 192khz
  - quality of the sound system
   
  However, it is a mathematical certainty that, by itself, 192khz cannot sound better than 96khz.
   
   
  edit: grammar


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## mikemalter

Quote: 





00940 said:


> I've been exposed to some. When I had it was as a rule better than usual... because the recordings had been remastered with greater attention. This by itself might be a very good reason to get into high-rez formats.
> 
> Keep in mind that it's very hard to fully isolate those 4 variables:
> 
> ...


 

 Totally get your bullet points.  Can you elaborate a bit, or possibly point me to a link that fleshes out the thought that from a mathematical perspective 192 cannot sound better than 96?  It would help me understand the issue better.  Thanks.


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## monoethylene

If I am not completely wrong it has sth to do with the Nyquist theorem and in summary there are no more useful information included in a 192 kHz sample as in a 96 kHz sample.


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## mikemalter

Quote: 





monoethylene said:


> If I am not completely wrong it has sth to do with the Nyquist theorem and in summary there are no more useful information included in a 192 kHz sample as in a 96 kHz sample.


 

 Can you help me understand in simple terms how doubling sample size does not provide any more useful information?  If that were the case, why are labels recording at 192?


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## leeperry

192kHz is like 4K for video, all it does is capture more noise...and it also forces most DAC chips to lower their internal oversampling rate(check the last two pages): http://www.lavryengineering.com/documents/Sampling_Theory.pdf
   
  I've got a DVD-A w/ 24/96 and 24/192(Grover Washington Jr), good luck DBT'ing them.
   
  That friend of mine who rips SACD's digitally said that SACD is more less capped at 40kHz: http://www.google.com/search?hl=en&source=hp&q=sacd+40khz&aq=f&aqi=&aql=&oq=
   
_"SACD format plays 20hz to 40khz, and has like and extra 30db of dynamic range"_


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## monoethylene

When I am going back to Nyquist, this theorem says that an audio wave has to be sampled with twice the maximum frequency to include all information to reconstruct this wave in a "economical" way. 44.1 kHz is the CD standard and the double of 22 kHz (normally the human limit of hearing). The same for DVD with 48 kHz as twice of 24 kHz (I dont know exactly why there is 48 kHz chosen. Maybe it has sth to do with the format itself?? ). So, 96 kHz is already a two time oversampling of the 48 kHz and in terms of Nyquist useless because there are no more significant information. 192 kHz is four time 48 kHz and in this meaning even more useless, because there are also no more new information. I, for myself compare it with a periodic sine wave and after 2*pi the same sine wave starts again.. 
   
  A time ago I have read the paper of Lavry. There it is also explained. Concerning your question of 192 kHz samplings my personal opinion is to get more consumers with not so much background knowledge. But this is my personal opinion and I am not an expert in digital signal processing. I only want to give myself logical answers and of course I want to understand it 
   
  ou, I see that I am very slow in writing (((


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## mikemalter

Quote: 





leeperry said:


> 192kHz is like 4K for video, all it does is capture more noise...and it also forces most DAC chips to lower their internal oversampling rate(check the last two pages): http://www.lavryengineering.com/documents/Sampling_Theory.pdf
> 
> I've got a DVD-A w/ 24/96 and 24/192(Grover Washington Jr), good luck DBT'ing them.
> 
> ...


 


   


  Quote: 





monoethylene said:


> When I am going back to Nyquist, this theorem says that an audio wave has to be sampled with twice the maximum frequency to include all information to reconstruct this wave in a "economical" way. 44.1 kHz is the CD standard and the double of 22 kHz (normally the human limit of hearing). The same for DVD with 48 kHz as twice of 24 kHz (I dont know exactly why there is 48 kHz chosen. Maybe it has sth to do with the format itself?? ). So, 96 kHz is already a two time oversampling of the 48 kHz and in terms of Nyquist useless because there are no more significant information. 192 kHz is four time 48 kHz and in this meaning even more useless, because there are also no more new information. I, for myself compare it with a periodic sine wave and after 2*pi the same sine wave starts again..
> 
> A time ago I have read the paper of Lavry. There it is also explained. Concerning your question of 192 kHz samplings my personal opinion is to get more consumers with not so much background knowledge. But this is my personal opinion and I am not an expert in digital signal processing. I only want to give myself logical answers and of course I want to understand it
> 
> ou, I see that I am very slow in writing (((


 


 Thanks for the responses, this is very interesting to me as I have no technical background in this area.  Personally, I like to eliminate superstition from my understanding wherever possible, and it seems like 192 is better may be superstitious thinking.


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## MacedonianHero

Quote: 





leeperry said:


> Glass toslink very much matters if you use cheap gear in realtime, it's been thoroughly discussed in this thread: http://www.head-fi.org/forum/thread/459752/which-is-the-best-optical-cable-under-300/135#post_6599838
> 
> _"I swapped out the 1M Monster cable with the 1M Dayton GOC-3 Glass Toslink cable and viola!
> 
> The cymbal smashes became three dimensional with a better sense of space, they were better located in the sound stage and no longer too bright and came to life. The bass surprisingly tightened up even further with better definition."_


 






 I can't believe someone actually quoted the junk that I write.


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## leeperry

Well, when you hear stuff on your own you're delusional and a poor victim of teh 3vil placebo...when you're two or more, you start to sound less "out there".
   
  What are your thoughts on coax/toslink/USB after all this time?


----------



## MacedonianHero

Quote: 





leeperry said:


> Well, when you hear stuff on your own you're delusional and a poor victim of teh 3vil placebo...when you're two or more, you start to sound less "out there".
> 
> What are your thoughts on coax/toslink/USB after all this time?


 

 LoL...then call me dilusional. 
	

	
	
		
		

		
		
	


	



   
  COAX > TOSLINK GLASS > TOSLINK PLASTIC
   
  USB is hard to place in there as I find it is so DAC dependent. With my Cary Xciter it is actually about equal to toslink (with a cheapo Monster plastic cable), but with my old PS Audio DLIII, USB was the worst by a good margin.
   
  At least that's how I hears it.


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## grokit

Just thought I would check in on the 192kHz thing, as I have read from various sources that agree with the above posts that it is useless for listening at that bandwidth and is a gimmick in that respect. But it is evidently very useful for recording engineers, as it gives them more headroom for non-destructive edits when mixing multiple audio tracks together and what not. So for professional recording/mixing/mastering applications in the digital domain it is very useful, but that is the only place IIRC. Sorry I don't have time to offer a more detailed technical explanation, or cite a source for this ATM, but it's not a big secret or anything in professional audio circles. For audiophile listening however, 192kHz is totally unnecessary as pointed out above.


----------



## monoethylene

Quote: 





> Hi *Mike*
> 
> "Thanks for the responses, this is very interesting to me as I have no technical background in this area.  Personally, I like to eliminate superstition from my understanding wherever possible, and it seems like 192 is better may be superstitious thinking."


 
  If you are interested in this, maybe this will help:
   
  http://www.dspguide.com/pdfbook.htm
   
  It is a little bit more as only Nyquist and so on but I like it


----------



## Mangeurdpommes

from dspguides.com :
   
   


> Consider an analog signal composed of frequencies between DC and 3 kHz. To properly digitize this signal it must be sampled at 6,000 samples/sec (6 kHz) *or higher*.


 
   
  So we might say, we can digitize an analog signal with a double sampling frequency. Mathematically this would give : For a given analog signal AS of frequency F, its "lossless" digital signal DS can be fully constructed with a sampling rate N in [2*F : +infinity[. So yes, 2*F is enough but this is in an ideal physical world, but electronic components are not theoritically perfect.
   
   
    
  Quote:


> The key point to remember is that a digital signal _cannot_ contain frequencies above one-half the sampling rate (i.e., the Nyquist frequency/rate).


 
   
   


> The analog filter used to convert the zeroth-order hold signal, (c), into the reconstructed signal, (f), needs to do two things: (1) remove all frequencies above one-half of the sampling rate, and (2) boost the frequencies by the reciprocal of the zeroth-order hold's effect, i.e., _1/sinc(x)_. This amounts to *an amplification of about 36% at one-half of the sampling frequency.* Figure (e) shows the ideal frequency response of this analog filter.


 
   
  Several sampling techniques exists. Shanon and Nyquist set the basis and formalized the theory. But today, in practice, more complex problematics appears due to some component imprecision. I think CD and SACD are technologically well advanced and the main problem still the source, how to record ! Our media are under-exploited. 
   
  This is just my opinion. I can only feel the difference between a Master 24b/192KHz plugged to a cambridge DAC (Magic Plus) when its plugged to a real amplifier. But from the headphone (which still a hifi ones), just on classical tracks.
   
  Concerning USB or Toslink, it depends on the soft, that's true, listening the same hi quality track under the same reader but on two different computer (VAIO and MacBook Pro) : USB didn't change anything but toslink did not sounds that good on the older Macbook pro.


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## starstern

usb 3.1 vs thunderbolt vs metal halide bridge ;                 ~~  just head phone jack out for audio  to dac  ???
  
http://www.audiophilleo.com/audiophilleo2.aspx
   *can one   elaborate   on how best to connect a dac to a system and total 0f gadgets to get best sound out ";*
http://www.audiophilleo.com/purepower.aspx
  
  
 http://www.empiricalaudio.com/products/off-ramp-converter
  
  
  
  
  
 ~~MCRU linear power supply
 ~ pwx power supply ?


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