# Why would 24 bit / 192 khz flac sound any better than 16 bit / 44.1 khz flac if both are lossless (if at all)?



## thesuperguy

If both formats are lossless, what differentiates the 2 versions in terms of sound quality if at all?


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## KT66

It depends on the source, a straight transfer of 24/192 to 24/192 should sound better
 than a 24/192 converted to 16/44.1
  
 If the source is 16/44.1 both should sound the same


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## Kaffeemann

Both do sound the same. The additional information 24 bit/192 kHz files can store is inaudible.
 See here.


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## KT66

That's just opinion,
 I can hear the difference between 24/96 and 16/44 if the source is good enough.

 I can't hear much difference between 24/96 and 24/192 if any


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## mikeaj

"Lossless" just means that no data has been lost relative to some version somewhere along the line. In other words, the data is compressed to take less space, like in a .zip or .rar file. So what is the format of the original?
  
 24-bit / 192 kHz contains more data than 16-bit / 44.1 kHz, around 550% more data. With 192 kHz, you can represent more sounds that are too high for people to hear. With 24 bits, you can capture the noise floor of the recording setup and such with more resolution and detail, even though at playback that extra stuff is generally going to be below your ambient room noise level anyway and drowned out by that, not to mention by the actual intended sounds (music) itself.
  
 In terms of having enough data for playback purposes for human consumption and perceived sound quality, they are pretty much equivalent because the extra data is not really anything noticeable or useful for that purpose.
  
 In practice, it is possible for some playback gear to misbehave more with one sampling rate than another, and there are more technical constraints with 44.1 kHz and so on, but generally it still shouldn't make an audible difference. Similarly, you can construct some very artificial scenario where the extra bit depth is audible as lower noise. But usually under more controlled testing (perhaps not always) the differences people think they hear vanish.


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## kraken2109

kt66 said:


> That's just opinion,
> I can hear the difference between 24/96 and 16/44 if the source is good enough.
> 
> I can't hear much difference between 24/96 and 24/192 if any


 
 It's not opinion, it's basic digital audio theory.
 44.1kHz can perfectly store frequencies up to 22.05kHz. Unless you're telling me you can hear higher than that then it shouldn't sound different. There are some other complexities like anti-aliasing filters but those changes shouldn't be audible.
 Bit depth of 16 or 24 isn't going to make a difference when you're playing back recordings with a dynamic range of less than 20dB in 95% of music.


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## BlindInOneEar

kt66 said:


> That's just opinion,
> I can hear the difference between 24/96 and 16/44 if the source is good enough.
> 
> I can't hear much difference between 24/96 and 24/192 if any


 

 Can you name some examples of tracks where you can hear the differences?  Could you describe what the differences sound like?


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## thesuperguy

Gotta love headfi debates


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## RazorJack

thesuperguy said:


> If both formats are lossless, what differentiates the 2 versions in terms of sound quality if at all?


 
  
 It's quite simple really, one is unneccesarily more expensive than the other.


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## elmoe

kraken2109 said:


> It's not opinion, it's basic digital audio theory.
> 44.1kHz can perfectly store frequencies up to 22.05kHz. Unless you're telling me you can hear higher than that then it shouldn't sound different. There are some other complexities like anti-aliasing filters but those changes shouldn't be audible.
> Bit depth of 16 or 24 isn't going to make a difference when you're playing back recordings with a dynamic range of less than 20dB in 95% of music.


 
  
 Are you saying there is no audible difference between a CD and a SACD version of the same album?


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## RazorJack

If there are audible differences I'm sure it's just because of remastering.
  
 Identical source music on a CD and SACD will sound the same to humans, or I'd like to see some DBT results showing otherwise.


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## kraken2109

elmoe said:


> Are you saying there is no audible difference between a CD and a SACD version of the same album?


 
 Assuming they're the same master then yes.


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## elmoe

But the whole point of SACD is that they're remastered to sound better.


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## Digitalchkn

FYI,  check out this parallel thread if you want to get into more details:
http://www.head-fi.org/t/415361/24bit-vs-16bit-the-myth-exploded


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## ralphp@optonline

razorjack said:


> It's quite simple really, one is unneccesarily more expensive than the other.


 

 And the high resolution one may actually contain LESS information than the CD resolution one, in that many, many high resolution downloads, especially those purchased from HDTracks, do not come with full information booklets containing information such as recording data (time and place of the recording, the equipment used to make the recording, producer, recording engineer, mastering engineer) and musicains, etc. - i.e. LESS information.
  
 But hey it costs more and the high end audio press just love high resolution digital audio so high resolution just has to be BETTER


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## HPiper

Lets turn this around a bit, give me the name of a classical redbook cd that sounds as good as or better than the sacd of the same title. I am not saying anybody is wrong or right but I would like to hear for myself. I think in MOST cases I can hear a distinct improvement listening to the sacd layer vs the redbook layer of a hybrid sacd. It could be the mastering for sure but it strikes me as odd that they would devote so much more time and energy in mastering one over the other. I have read most of those papers saying there is no difference and while I admit the science looks solid, the result just confuses me because ( I think) I can hear a difference, a subtle one to be sure but still...


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## ralphp@optonline

hpiper said:


> Lets turn this around a bit, give me the name of a classical redbook cd that sounds as good as or better than the sacd of the same title. I am not saying anybody is wrong or right but I would like to hear for myself. I think in MOST cases I can hear a distinct improvement listening to the sacd layer vs the redbook layer of a hybrid sacd. It could be the mastering for sure but it strikes me as odd that they would devote so much more time and energy in mastering one over the other. I have read most of those papers saying there is no difference and while I admit the science looks solid, the result just confuses me because ( I think) I can hear a difference, a subtle one to be sure but still...


 
 All SACDs use DSD rather than PCM (which is what all CDs use) to encode the analog sound. So comparing the CD layer of an SACD to the SACD layer is not the same comparing a 16bit/44.1KHz audio file (PCM) to a 24bit/96khz or 192kHZ audio file (also PCM), which may well account for the subtle differences you are hearing.


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## joseph69

A couple of years ago I bought a Sony C-SACD 222ES and I found that when I went to purchase some SACD's, there weren't many available, and from what I understood at the time was that they really didn't take off like in sales as intended. I wound up finding Pink Floyds (Dark Side Of The Moon), and played it in SACD, then played it in my Sony CDP-C801ES, and the only difference that I could hear was maybe the SACD had a brighter sound…but I actually preferred the sound of it in the Cd format out of the 801ES. I wound up giving the SACD player to my father, i definitely liked the sound of the Cd better, I thought it was warmer and fuller. IMO.
 Also besides not being able to find many… at all, I wasn't going to replace 300+ Cd's for SACD's that didn't impress me anyway, so I also felt it was a little bit of a gimmick, to my ears and wallet.
 I don't mean to offend anyone that disagrees and loves SACD's, this is just what I heard and how I feel.


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## elmoe

joseph69 said:


> A couple of years ago I bought a Sony C-SACD 222ES and I found that when I went to purchase some SACD's, there weren't many available, and from what I understood at the time was that they really didn't take off like in sales as intended. I wound up finding Pink Floyds (Dark Side Of The Moon), and played it in SACD, then played it in my Sony CDP-C801ES, and the only difference that I could hear was maybe the SACD had a brighter sound…but I actually preferred the sound of it in the Cd format out of the 801ES. I wound up giving the SACD player to my father, i definitely liked the sound of the Cd better, I thought it was warmer and fuller. IMO.
> Also besides not being able to find many… at all, I wasn't going to replace 300+ Cd's for SACD's that didn't impress me anyway, so I also felt it was a little bit of a gimmick, to my ears and wallet.
> I don't mean to offend anyone that disagrees and loves SACD's, this is just what I heard and how I feel.


 
 I get where you're coming from but testing a single SACD is hardly what I call an objective test.


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## ralphp@optonline

elmoe said:


> I get where you're coming from but testing a single SACD is hardly what I call an objective test.


 

 While what you say is true, the SACD is for all practical proposes a dead format. Sure there still a few labels putting out SACDs but most of these are high priced reissues or obscure classical music and no one is releasing more current or popular music on SACD. Sure Sony is trying to breathe new life into the almost dead DSD format (the encoding scheme used for SACD) with the new big push for DSD downloads and DSD enabled DACs but it is still way too soon to jump on the DSD bandwagon, e.g. remember Betamax.


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## elmoe

ralphp@optonline said:


> While what you say is true, the SACD is for all practical proposes a dead format. Sure there still a few labels putting out SACDs but most of these are high priced reissues or obscure classical music and no one is releasing more current or popular music on SACD. Sure Sony is trying to breathe new life into the almost dead DSD format (the encoding scheme used for SACD) with the new big push for DSD downloads and DSD enabled DACs but it is still way too soon to jump on the DSD bandwagon, e.g. remember Betamax.


 
  
 I don't know about that - pretty much every DAC maker out there is making their product DSD compatible these days.


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## joseph69

elmoe said:


> I get where you're coming from but testing a single SACD is hardly what I call an objective test.


 
 Well like I said, it was hard enough too find this on SACD, and I'm very familiar with this Cd, and I wasn't doing a critically objective test to begin with…I just wanted to hear the difference. And like I said, I'm not out too offend anyone, and I was just checking out the difference, and this was enough for me.
  


ralphp@optonline said:


> While what you say is true, the SACD is for all practical proposes a dead format. Sure there still a few labels putting out SACDs but most of these are high priced reissues or obscure classical music and no one is releasing more current or popular music on SACD. Sure Sony is trying to breathe new life into the almost dead DSD format (the encoding scheme used for SACD) with the new big push for DSD downloads and DSD enabled DACs but it is still way too soon to jump on the DSD bandwagon, e.g. remember Betamax.


 
 Exactly what I felt at the time also about the limited choices too.
 And remember DAT also.


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## RazorJack

ralphp@optonline said:


> And the high resolution one may actually contain LESS information than the CD resolution one, in that many, many high resolution downloads, especially those purchased from HDTracks, do not come with full information booklets containing information such as recording data (time and place of the recording, the equipment used to make the recording, producer, recording engineer, mastering engineer) and musicains, etc. - i.e. LESS information.
> 
> But hey it costs more and the high end audio press just love high resolution digital audio so high resolution just has to be BETTER


 
  
 I had no idea what you're talking about (not staying up to date on "HD" audio or anything above 16 bit / 44.1 kHz as I really couldn't care less about this scam/fad) and at first I thought you were sarcastic, but then I checked that site and you're actually right!
  
 What a complete rip-off!


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## ralphp@optonline

elmoe said:


> I don't know about that - pretty much every DAC maker out there is making their product DSD compatible these days.


 
 The DAC makers are in business to sell DACs and if adding the latest fad, i.e. DSD, to their product helps to sell more product then why the hell not. But that does make DSD a worthwhile way to spend one's money.
  


razorjack said:


> I had no idea what you're talking about (not staying up to date on "HD" audio or anything above 16 bit / 44.1 kHz as I really couldn't care less about this scam/fad) and at first I thought you were sarcastic, but then I checked that site and you're actually right!
> 
> What a complete rip-off!


 
 Agree and thanks!
  


joseph69 said:


> Well like I said, it was hard enough too find this on SACD, and I'm very familiar with this Cd, and I wasn't doing a critically objective test to begin with…I just wanted to hear the difference. And like I said, I'm not out too offend anyone, and I was just checking out the difference, and this was enough for me.
> 
> Exactly what I felt at the time also about the limited choices too.
> And remember DAT also.


 
 Another thanks!


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## kraken2109

This thread was originally about flac which is PCM rather than SACD which is DSD.


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## ralphp@optonline

kraken2109 said:


> This thread was originally about flac which is PCM rather than SACD which is DSD.


 

 Exactly!


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## miceblue

kraken2109 said:


> kt66 said:
> 
> 
> > That's just opinion,
> ...



That's only 1 aspect of sampling though, the other aspect is filtering.

44.1 kHz / 2 may provide all the data humans can hear, but a sampling rate higher than that might be useful for designing filters, no? A filter capturing 20 kHz signals with a maximum limit of 22.05 kHz is pretty steep compared to capturing 20 khz signals with a maximum limit of say 48 kHz.


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## brhfl

It's worth pointing out that due to intermodulation distortion, 192kHz could quite possibly sound _worse_. Some explanations and tests/demos are about a third of the way down this page.


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## miceblue

brhfl said:


> It's worth pointing out that due to intermodulation distortion, 192kHz could quite possibly sound _worse_. Some explanations and tests/demos are about a third of the way down this page.



I've read that, and again, that's only one aspect of higher sampling rates. Aliasing from higher sampling rates is only a problem if it's not filtered.

Read: that ultra-frequency content is only a problem after the DAC. If the DAC filters out this content, then your amp and transducer is ultra-frequency content free.


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## brhfl

miceblue said:


> I've read that, and again, that's only one aspect of higher sampling rates. Aliasing from higher sampling rates is only a problem if it's not filtered.
> 
> Read: that ultra-frequency content is only a problem after the DAC. If the DAC filters out this content, then your amp and transducer is ultra-frequency content free.


 
 Hence 'possibly,' and 'test.'


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## JRG1990

It depends what filter the dac has a 192khz dac normally uses a 0.45/0.55 filter at all sample thats linear responce to 21.6k at 48khz , 43.2k at 96khz , 86.4k at 192khz 
 A dac using this filtering will upsample and then back downsample to avoid aliasing disortion
  
 This paper explains it better
  
  
 http://www.digitalaudio.dk/media/3Tradeoff_of_192_kHz.pdf


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## Cannikin

So, do many/most of you believe there is ZERO advantage to buying 24 bit HDtracks files if they are for playback on my AK100 + SRH1540 (or GR10) kit? Some albums are available ONLY as 24 bit files, but if I am not really gaining anything from the format, I will choose to buy and rip the CD to AIF instead. Opinions, please?


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## miceblue

cannikin said:


> So, do many/most of you believe there is ZERO advantage to buying 24 bit HDtracks files if they are for playback on my AK100 + SRH1540 (or GR10) kit? Some albums are available ONLY as 24 bit files, but if I am not really gaining anything from the format, I will choose to buy and rip the CD to AIF instead. Opinions, please?



I buy stuff from HD Tracks because the master sounds different from the normal CD pressing...sometimes.

Sometimes meaning, sometimes the master is terrible and you get junk HD files like this due to poor mastering and possible 16/44.1 upsampling.


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## JRG1990

Theres 0 audiable difference between 16 and 24 bit  16 bit has a dymanic range of 96db 24 bit is 144db, no dac/amp has a noise floor -144db , unbalanced cables can only manage -75db , no music has a dymanic range larger than 65db.
 0-144db is absolute silence to blown eardrums and if listening on speakers you'd need about 5000 rms to hit 144db.


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## Aggie-Luna

What about compresion?...my understanding is that one of the nice things of the 96/192 format is that the compresion is minimum, and the dynamic range is more close to the original masters


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## ralphp@optonline

cannikin said:


> So, do many/most of you believe there is ZERO advantage to buying 24 bit HDtracks files if they are for playback on my AK100 + SRH1540 (or GR10) kit? Some albums are available ONLY as 24 bit files, but if I am not really gaining anything from the format, I will choose to buy and rip the CD to AIF instead. Opinions, please?


 
  
 Stick with ripping CDs - it is cheaper and at least you get a booklet with the CD.


miceblue said:


> I buy stuff from HD Tracks because the master sounds different from the normal CD pressing...sometimes.
> 
> Sometimes meaning, sometimes the master is terrible and you get junk HD files like this due to poor mastering and possible 16/44.1 upsampling.


 
 Then why is HDTracks selling this garbage at such inflated prices?
  


aggie-luna said:


> What about compresion?...my understanding is that one of the nice things of the 96/192 format is that the compresion is minimum, and the dynamic range is more close to the original masters


 
 What are you talking about?


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## riverlethe

cannikin said:


> So, do many/most of you believe there is ZERO advantage to buying 24 bit HDtracks files if they are for playback on my AK100 + SRH1540 (or GR10) kit? Some albums are available ONLY as 24 bit files, but if I am not really gaining anything from the format, I will choose to buy and rip the CD to AIF instead. Opinions, please?


 
  
 There may be an advantage if the master is better on the 24-bit version.  Of course, it could just as easily be the opposite.  The bit and sampling rates have nothing to do with it.


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## esldude

jrg1990 said:


> Theres 0 audiable difference between 16 and 24 bit  16 bit has a dymanic range of 96db 24 bit is 144db, no dac/amp has a noise floor -144db , unbalanced cables can only manage -75db , no music has a dymanic range larger than 65db.
> 0-144db is absolute silence to blown eardrums and if listening on speakers you'd need about 5000 rms to hit 144db.


 

 Let us not play fast and loose with the facts now.  Unbalanced cables can and do exceed -75 db on the low end. By quite a margin in fact.  The connected equipment may or may not, but even unbalanced electronics are quite capable of exceeding this.  You do have a hard time getting below -120 db with active components, and not many quite reach that. 
  
 Music also can have greater dynamic range than 65 db though it isn't common.  Few recordings do.  24 bit if more than enough. 16 bit is likely all you really need for playback.  But no need to put out bad info on other aspects of the signal chain.


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## RazorJack

cannikin said:


> So, do many/most of you believe there is ZERO advantage to buying 24 bit HDtracks files if they are for playback on my AK100 + SRH1540 (or GR10) kit? Some albums are available ONLY as 24 bit files, but if I am not really gaining anything from the format, I will choose to buy and rip the CD to AIF instead. Opinions, please?


 
  
 Let your ears decide.


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## Cannikin

ralphp@optonline said:


> Stick with ripping CDs - it is cheaper and at least you get a booklet with the CD.


 
  
 Strictly speaking, that isn't always true. Case in point: some of the music I have bought from HDtracks is 44.1/16bit music (close to CD quality) and these albums are often priced around $12, which - in many cases - is less than the cost of purchase of a CD, especially if you factor in that my best option for CD purchases is Amazon, which costs extra to ship to me. (I live rural, and music shopping means a 90 mile drive)
  
 As for the booklet that accompanies most CDs: I don't care one whit about "liner notes" -- this material often gets a cursory glance and then is never viewed again. I mean, the internet is frequently a much richer source of artist information than what goes into these things. Ive yet to encounter liner notes/booklets that added significantly to the listening experience or my appreciation of the work. YMMV of course.


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## ralphp@optonline

razorjack said:


> Let your ears decide.


 

 Pray tell, exactly how does one do that without purchasing either the CD, the high resolution download or both?


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## Cannikin

ralphp@optonline said:


> Pray tell, exactly how does one do that without purchasing either the CD, the high resolution download or both?


 

 I assumed he meant that I ought to buy both and compare. Doing that *one time* for the sake of making a comparison is something I would think is worth the expense.....*once*, anyway.


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## RazorJack

cannikin said:


> I assumed he meant that I ought to buy both and compare. Doing that *one time* for the sake of making a comparison is something I would think is worth the expense.....*once*, anyway.


 
  
 Correct.
  
 But if you already know you're only gonna do it once, then why bother at all?


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## spook76

aggie-luna said:


> What about compresion?...my understanding is that one of the nice things of the 96/192 format is that the compresion is minimum, and the dynamic range is more close to the original masters




Unfortunately no. Dynamic compression happens to master tape and once it is done there is no way, short of having the music rerecorded, to recapture the uncompressed music.


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## Cannikin

razorjack said:


> Correct.
> 
> But if you already know you're only gonna do it once, then why bother at all?


 
 Ummm...because it would give me at least *some* data about the quality of my purchases and that information would provide *some* guidance for future purchases.  If the "high def" purchased version was no better (to MY ears, on MY equipment) than the lossless rip I did from my own CD, I would happily (although perhaps not unerringly) conclude that - in most instances - lossless CD rips would be perfectly adequate for my purposes.
  
 I found your_ "why bother?"_ question a bit nonsensical, to be honest; puzzling.


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## Digitalchkn

cannikin said:


> If the "high def" purchased version was no better (to MY ears, on MY equipment) than the lossless rip I did from my own CD, I would happily (although perhaps not unerringly) conclude that - in most instances - lossless CD rips would be perfectly adequate for my purposes.


 
  
 When doing this you have to be aware of different masters frequently often being used for CD vs hi-res version.  At that point it becomes more of a comparison of which master you prefer rather than the "24 vs 16 bit"


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## ralphp@optonline

cannikin said:


> I assumed he meant that I ought to buy both and compare. Doing that *one time* for the sake of making a comparison is something I would think is worth the expense.....*once*, anyway.


 

 As has been stated numerous times on this thread, whether the high resolution version sounds better than the standard CD version is more a function of how each was mastered and really has nothing to do with the purported benefits of the higher sampling rate and the increased bit depth of the high resolution version. In which case one just needs to find out, via forums, google, etc. which mastering has the best sound and buy the CD version of that mastering, if available.


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## RazorJack

cannikin said:


> Ummm...because it would give me at least *some* data about the quality of my purchases and that information would provide *some* guidance for future purchases.  If the "high def" purchased version was no better (to MY ears, on MY equipment) than the lossless rip I did from my own CD, I would happily (although perhaps not unerringly) conclude that - in most instances - lossless CD rips would be perfectly adequate for my purposes.
> 
> I found your_ "why bother?"_ question a bit nonsensical, to be honest; puzzling.


 
  
 Ok, understood.
  
 I find the hype around >16 bit, >44.1 kHz digital audio a bit nonsensical to be honest. But not puzzling


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## Digitalchkn

razorjack said:


> Ok, understood.
> 
> I find the hype around >16 bit, >44.1 kHz digital audio a bit nonsensical to be honest.


 
  
 Some hype may be there, but it does encourage development of higher quality gear!


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## miceblue

16+/44.1+ is almost always used in the studio though. People who manipulate the music find it useful.


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## riverlethe

razorjack said:


> Let your ears decide.




Can anyone think of advice worse than this? "Let your ____ decide."


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## thesuperguy

I posted a simple question and I expected a simple answer, which I got a few billion posts behind  , but I certainly did not expect such an active debate to spark up. I would have thought that this topic had been discussed into the ground already over the years but it seems like people can't get enough of it.
  
 Just for some background about my question - I was listening to "Time" from The Dark Side of the Moon which I ripped from an SACD in all its ultra high quality glory, and after getting a chance to listen to another typical lossless version of the song, I was astonished at the outstanding difference between both lossless formats. You would know this if you saw my previous thread somewhere over yonder, but turns out it's due to very noticeable mastering differences.


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## Aggie-Luna

interesting topic


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## KT66

Format is almost irrelevant, its all about the mastering.
To the OP you need to join and read stevehoffman.tv forum, it could save you a lot of money !


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## nanaholic

digitalchkn said:


> Some hype may be there, but it does encourage development of higher quality gear!


 
  
 It doesn't really help to have all the high quality gear in the world when the music industry goes back to being lazy/careless and making poor masters.  No amount of high-end gear can fix clipping for example, so all those songs that are mixed too hot is ruined forever and no file format or bitrate will fix it.


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## ralphp@optonline

nanaholic said:


> It doesn't really help to have all the high quality gear in the world when the music industry goes back to being lazy/careless and making poor masters.  No amount of high-end gear can fix clipping for example, so all those songs that are mixed too hot is ruined forever and no file format or bitrate will fix it.


 

 When it comes to popular music, i.e. any music that sells very well - be it country, rap, hip-hop, R&B, rock, pop, etc., the music industry just wants it to sound good on a smart phone with $5 ear buds, which of course means as an mp3 file. So unless one is listening to non-popular music, i.e. music that does NOT sell, such as jazz, world, classical, etc., all one really needs a smart phone and some really cheap ear buds, any else is just overkill and a waste of money.
  
 Luckily I listen to a lot of jazz and a lot of it is very well produced, recorded and mastered. Sounds great on good equipment and state of the art headphones.


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## Cannikin

ralphp@optonline said:


> As has been stated numerous times on this thread, whether the high resolution version sounds better than the standard CD version is more a function of how each was mastered and really has nothing to do with the purported benefits of the higher sampling rate and the increased bit depth of the high resolution version. In which case one just needs to find out, via forums, google, etc. which mastering has the best sound and buy the CD version of that mastering, if available.


 

 Is there a single, collated resource available to us that offers a rating/review of releases/remasters so we can at least have a clue as to which masters to avoid and which to seek out? Seems to me that such a thing would be incredibly helpful. (My apologies if the answer to this question resides within the realm of "common knowledge" - it isn't something I am aware of)
  
 If "googling" for the information, can you suggest ideal search terms for best results? I can imagine there is a *right way*, and a *wrong way* to find such data.


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## Digitalchkn

ralphp@optonline said:


> Luckily I listen to a lot of jazz and a lot of it is very well produced, recorded and mastered. Sounds great on good equipment and state of the art headphones.


 
  
 Exactly. A great master will be limited by poor equipment at the end of the chain - your setup. But a hi-res capable setup has enough overhead to preserve the full resolution of the source, even if it doesn't take advantage of 24 bits resolution.


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## ag8908

Partly because the bigger number fools your brain into thinking it sounds better. There is no substantive difference with respect to the vast majority of audiophile stuff, and many audiophile things sound worse. For example, the best headphone won't sound anywhere as good as the sweet spot of a basic 5.1 surround system, but that won't stop people from wasting money on the headphone.


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## ralphp@optonline

cannikin said:


> Is there a single, collated resource available to us that offers a rating/review of releases/remasters so we can at least have a clue as to which masters to avoid and which to seek out? Seems to me that such a thing would be incredibly helpful. (My apologies if the answer to this question resides within the realm of "common knowledge" - it isn't something I am aware of)
> 
> If "googling" for the information, can you suggest ideal search terms for best results? I can imagine there is a *right way*, and a *wrong way* to find such data.


 

 Not that I know of, however here are two things to keep in mind when looking for a reliable source of information.
  
 1) Avoid reviews and opinions from any source (print or online publications) that have any of the vendors of high resolutions downloads (e.g. HDTracks, Linn, Acoustic Sounds, etc.) as advertisers - their "opinions" will be more press release than actual opinions plus for these publications ANY and ALL high resolution downloads, especially DSD downloads, will sound great.
  
 2) Public or semi-public forums are a much better source of quality opinions although these too have their drawbacks (e.g. fan boys, people who need to justify their purchases) but by taking an average of the opinions one can draw some reasonable conclusions.
  
 So for example the sound quality of new Beck recording "Morning Phase" has been highly praised in the press and overwhelmingly trashed in online forums.


----------



## Digitalchkn

ag8908 said:


> Partly because the bigger number fools your brain into thinking it sounds better. There is no substantive difference with respect to the vast majority of audiophile stuff, and many audiophile things sound worse. For example, the best headphone won't sound anywhere as good as the sweet spot of a basic 5.1 surround system, but that won't stop people from wasting money on the headphone.


 
  
 There is general agreement that even not-so-high-end headphones have higher resolution than many of the better speaker systems.  So it depends on what you prefer.
  
 But why listen to others.
  
 I have done my own tests comparing 16/44 to even 24/48. I used same quality 24/96 source with large dynamic range to generate 16/44, 16/44 dithered, 24/48 and  16/44 upscaled to 24/48. I used quality conversion software with floating point resolution so the conversion error is well below the 24 bit noise floor.  On my 24 bit headphone chain I CAN TELL A DIFFERENCE between 16/44 and 24/48 versions.


----------



## Cannikin

ralphp@optonline said:


> Not that I know of, however here are two things to keep in mind when looking for a reliable source of information.
> 
> 1) Avoid reviews and opinions from any source (print or online publications) that have any of the vendors of high resolutions downloads (e.g. HDTracks, Linn, Acoustic Sounds, etc.) as advertisers - their "opinions" will be more press release than actual opinions plus for these publications ANY and ALL high resolution downloads, especially DSD downloads, will sound great.
> 
> ...




Thank you for this.

It's a shame there isn't a simple database resource that lists recordings, the various releases of each, and a way for listeners to give a rating and/or review, to serve as a way for others to isolate the best possible option for each recording. It's the sort of thing that is common in many other realms, and it seems odd to me that the "high res aficionado" community hasn't created something similar. Has this been suggested or attempted in the past? If so, what were the results?


----------



## Digitalchkn

cannikin said:


> Thank you for this.
> 
> It's a shame there isn't a simple database resource that lists recordings, the various releases of each, and a way for listeners to give a rating and/or review, to serve as a way for others to isolate the best possible option for each recording. It's the sort of thing that is common in many other realms, and it seems odd to me that the "high res aficionado" community hasn't created something similar. Has this been suggested or attempted in the past? If so, what were the results?


 
  
 SA-CD.net has user reviews forums.


----------



## ag8908

digitalchkn said:


> There is general agreement that even not-so-high-end headphones have higher resolution than many of the better speaker systems.  So it depends on what you prefer.
> 
> But why listen to others.
> 
> I have done my own tests comparing 16/44 to even 24/48. I used same quality 24/96 source with large dynamic range to generate 16/44, 16/44 dithered, 24/48 and  16/44 upscaled to 24/48. I used quality conversion software with floating point resolution so the conversion error is well below the 24 bit noise floor.  On my 24 bit headphone chain I CAN TELL A DIFFERENCE between 16/44 and 24/48 versions.




That's not true.I defy you to identify any part of a song on a headphone that you can't identify on the speaker system.

What happens with headphones and their lame 1 to 2 inch drivers is that they cause an instrument or sound that was intended to be subtle background noise to be pushed way forward and dominate the song in a way that the artist never intended. So you're listening to this distorted and disproportional version of the song and thinking it's better in just because that subtle sound is now dominating the track at the expense of stuff you were supposed to focus on. A large speaker driver can blow a headphone driver away in every objective technical aspect as you would expect based merely on the physical characteristics. The hd800 is the closest to replicating the speaker sound but even that fails woefully.

That's why beats and their heavily distorted sound dominated the market. If people want accurate natural sound they'll just listen to their speakers.


----------



## elmoe

ag8908 said:


> That's not true.I defy you to identify any part of a song on a headphone that you can't identify on the speaker system.
> 
> What happens with headphones and their lame 1 to 2 inch drivers is that they cause an instrument or sound that was intended to be subtle background noise to be pushed way forward and dominate the song in a way that the artist never intended. So you're listening to this distorted and disproportional version of the song and thinking it's better in just because that subtle sound is now dominating the track at the expense of stuff you were supposed to focus on. A large speaker driver can blow a headphone driver away in every objective technical aspect as you would expect based merely on the physical characteristics. The hd800 is the closest to replicating the speaker sound but even that fails woefully.
> 
> That's why beats and their heavily distorted sound dominated the market. If people want accurate natural sound they'll just listen to their speakers.


 
  
 This is just totally wrong on all levels. There are many headphones out there (given they're amplified properly) that rival speakers that cost much much more.
  
 Beats dominated the market because young people today listen to bass heavy music and like a subwoofer on their ear, but that doesn't mean hi-end headphones can't achieve a level a neutrality that cheaper speakers can - in fact it's usually the opposite. Go listen even to a Bose speaker setup and see if it's more neutral than the HD800


----------



## ag8908

elmoe said:


> This is just totally wrong on all levels. There are many headphones out there (given they're amplified properly) that rival speakers that cost much much more.
> 
> Beats dominated the market because young people today listen to bass heavy music and like a subwoofer on their ear, but that doesn't mean hi-end headphones can't achieve a level a neutrality that cheaper speakers can - in fact it's usually the opposite. Go listen even to a Bose speaker setup and see if it's more neutral than the HD800


 
  
 Sure, if you like to have tiny drivers of a size you find in clock radios pump a subtle background harp so loudly that it dominates the song. If you like distorted sound, whether it's extra bass, or exaggerated detail, then headphones are the way to go.
  
 But if you want a person to sound like they sound when singing in front of you, in real life, a headphone will never get close to a speaker.
  
 P.S. that line about how "this $1,500 headphone is so good that it'll sound like a $20,000 speaker system" is a misleading marketing spin that needs to stop. it won't even sound as good as a $600 speaker system.


----------



## elmoe

ag8908 said:


> Sure, if you like to have tiny drivers of a size you find in clock radios pump a subtle background harp so loudly that it dominates the song. If you like distorted sound, whether it's extra bass, or exaggerated detail, then headphones are the way to go.
> 
> But if you want a person to sound like they sound when singing in front of you, in real life, a headphone will never get close to a speaker.
> 
> P.S. that line about how "this $1,500 headphone is so good that it'll sound like a $20,000 speaker system" is a misleading marketing spin that needs to stop. it won't even sound as good as a $600 speaker system.


 
  
 Clearly you haven't found proper synergy in your headphone based system.
  
 I personally prefer speakers - no doubt about it. As far as details go however, my speakers do not beat any of the headphones in my signature. (I have Dynaudio Audience 62 floorstanding speakers). Don't get me wrong, my speakers are awesome and I love them, but if I want to listen to Nina Simone and feel like she's in front of me, I put on my SA5000s and close my eyes. My speakers can't achieve that at the same level.
  
 And yes, I would say 1500usd headphones (for the most part) can sound as good as a 20k speaker system. Headphones today are just that good, nothing to do with a marketing spin.


----------



## ralphp@optonline

ag8908 said:


> Sure, if you like to have tiny drivers of a size you find in clock radios pump a subtle background harp so loudly that it dominates the song. If you like distorted sound, whether it's extra bass, or exaggerated detail, then headphones are the way to go.
> 
> But if you want a person to sound like they sound when singing in front of you, in real life, a headphone will never get close to a speaker.
> 
> P.S. that line about how "this $1,500 headphone is so good that it'll sound like a $20,000 speaker system" is a misleading marketing spin that needs to stop. it won't even sound as good as a $600 speaker system.


 

 Just a brief reminder - this is a headphone forum and your posts are pretty much telling everyone that you are behaving just like a troll. You have been warned.


----------



## ag8908

ralphp@optonline said:


> Just a brief reminder - this is a headphone forum and your posts are pretty much telling everyone that you are behaving just like a troll. You have been warned.


 

 To be clear, stating that speakers sound better than headphones is a bannable offense on head-fi? I want to make sure I know the rule (I assume you are speaking as a moderator who represents this site's policies and not merely impersonating one.).


----------



## elmoe

I'll add to that that if you feel your HD800s aren't worth 200usd speakers I will be glad to trade you my Logitech computer speakers for it


----------



## ag8908

elmoe said:


> I'll add to that that if you feel your HD800s aren't worth 200usd speakers I will be glad to trade you my Logitech computer speakers for it


 

 The most expensive logitech computer speaker I'm aware of is about $100 MSRP new, and not on sale.  What model do you have?


----------



## elmoe

ag8908 said:


> The most expensive logitech computer speaker I'm aware of is about $100 MSRP new, and not on sale.  What model do you have?


 
  
 I don't know, but they're crappy I can tell you that. By your logic they should make a nice HD800 replacement though. 
	

	
	
		
		

		
		
	


	



  
 They're kinda like these:
  

  
 Check out the size of those drivers, and 4 of em too, double the fun of your HD800.


----------



## ag8908

elmoe said:


> I don't know, but they're crappy I can tell you that. By your logic they should make a nice HD800 replacement though.
> 
> 
> 
> ...


 
  
 This is actual trolling, in case anyone didn't understand the difference between trying to add accurate information to a discussion, and a post designed to upset another person in a childish manner (either by impersonating a moderator as ralph did or stuff like this).


----------



## elmoe

How am I trying to upset you? Sound quality is relative to size of driver, I am adding on to the accurate information of your post above. I don't blame you for not wanting my Logitech speakers though.


----------



## pockits

Speakers as loudspeakers, and little speakers as headphones sound different in all levels. There are crucial factors as the distance from our ear to the source of the sound ( membrane of the driver), for example. and how are this frequencies reproduce.
  
 As an engineer I can discuss all of this but as an audiophile I Can as well say this is all wrong.
  
 The basic genesis of the music is the recording of the instruments, as far as the theory goes, if you have enough information in the master tape you can, of course listen to everything your ears and body (psychoacoustics) let you.
  
 Regarding the theory it is, in fact , not only a theory but a scientific fact and is called the Nyquist rate, this is twice the bandwidth of a bandlimited function or a bandlimited channel, Ergo, if you hear 22 kHz, as a function you need twice the bandwidth or 44 kHz,.
  
 I am sure everyone here knows this in many ways and it is not an issue, but then again, if you master track has enough information, as an analogue support or digital support, you are going to have and maybe heard more information.
  
  
 In terms of electroacoustics not all the fancy speakers on the market can reproduce everything that's why we try and try until we find what we are looking for.


----------



## ag8908

elmoe said:


> How am I trying to upset you? Sound quality is relative to size of driver, I am adding on to the accurate information of your post above. I don't blame you for not wanting my Logitech speakers though.


 

 Copying another troll post that adds zero information other than a childish attempt to provoke another person.


----------



## ralphp@optonline

ag8908 said:


> To be clear, stating that speakers sound better than headphones is a bannable offense on head-fi? I want to understand this, because it would speak volumes about the credibility of this site and its operators (I assume you are speaking as a moderator who represents this site's policies and not speaking with authority you don't have).


 

 I am not a moderator nor do I know if your posts will get you banned (I seriously doubt it) but I do know that all you will end up doing is start a useless flame war.
  
 Just for the record I do most of listening via speakers not headphones but I do not agree with some of your statements regarding the sound of headphones (which of course means that I do agree with some of your statements). Nonetheless for the most part I keep my feelings about the superiority of speakers to myself since this is a HEADPHONE forum.
  
 What I'm trying to say is that even though one may absolutely hate heavy metal music it just is not polite to post one's opinions on heavy metal music on a thread about heavy metal. Sometimes the less said the better.


----------



## ag8908

ralphp@optonline said:


> I am not a moderator nor do I know if your posts will get you banned (I seriously doubt it) but I do know that all you will end up doing is start a useless flame war.
> 
> Just for the record I do most of listening via speakers not headphones but I do not agree with some of your statements regarding the sound of headphones (which of course means that I do agree with some of your statements). Nonetheless for the most part I keep my feelings about the superiority of speakers to myself since this is a HEADPHONE forum.
> 
> What I'm trying to say is that even though one may absolutely hate heavy metal music it just is not polite to post one's opinions on heavy metal music on a thread about heavy metal. Sometimes the less said the better.


 
  
 If you're not a moderator, why did you issue a "warning" to me? A warning is a communication designed to tell a person that they will be punished in some way if they don't comply with your wishes. If you don't want to start flame wars, then you may wish to start by refraining from dictating contrived and arbitrary rules about what people can and can't post, like you run this place.
  
 That's my last post on this two page distraction that you and elmoe (both of whom who did the same thing with me in another thread) started. Grow up already and please end your forum stalking. Thank you!


----------



## ag8908

pockits said:


> Speakers as loudspeakers, and little speakers as headphones sound different in all levels. There are crucial factors as the distance from our ear to the source of the sound ( membrane of the driver), for example. and how are this frequencies reproduce.


 
   
That's all true, but one of them sounds closer to the actual live sound than the other. That's the only point I'm making.


----------



## Digitalchkn

pockits said:


> As an engineer I can discuss all of this but as an audiophile I Can as well say this is all wrong.
> 
> ....
> 
> ...


 
 Greetings, from a fellow engineer.
  
 Nyquist is not wrong. But the idea that 22KHz BW is sufficient for human hearing is in question.  While we may not hear sinosoids at 22KHz, it may be that people do some perceive energy up at and beyond those frequencies.
  
 Besides, remember that the oddball sampling rate of 44.1KHz came from a practical need to sync up to video recorder frequencies of the time. At the time it was deemed sufficient to cover the 20KHz bandwidth that was common belief to be sufficient.  We ended up being stuck with it today.


----------



## ralphp@optonline

ag8908 said:


> If you're not a moderator, why did you issue a "warning" to me? A warning is a communication designed to tell a person that they will be punished in some way if they don't comply with your wishes. If you don't want to start flame wars, then you may wish to start by refraining from dictating contrived and arbitrary rules about what people can and can't post, like you run this place.
> 
> That's my last post on this two page distraction that you and elmoe (both of whom who did the same thing with me in another thread) started. Grow up already and please end your forum stalking. Thank you!


 

 I think that you are just a little confused about who is stalking and who is behaving like a child.
  
 Enough said. Now go find someone else to have a little spat with because I'm done.


----------



## pockits

digitalchkn said:


> Greetings, from a fellow engineer.
> 
> Nyquist is not wrong. But the idea that 22KHz BW is sufficient for human hearing is in question.  While we may not hear sinosoids at 22KHz, it may be that people do some perceive energy up at and beyond those frequencies.
> 
> Besides, remember that the oddball sampling rate of 44.1KHz came from a practical need to sync up to video recorder frequencies of the time. At the time it was deemed sufficient to cover the 20KHz bandwidth that was common belief to be sufficient.  We ended up being stuck with it today.


 

 Exactly... Red Book ... and all the NAB standards, but since not everyone here are engineers with electronics knowledge I try not to be a Pa*** in the A**.
  
 BTW Nyquist is clearly not wrong.


----------



## Cannikin

I find this forum a *challenging* place to get good, genuinely useful information because it seems that many discussions quickly degrade into _"I'm right and you're wrong!"_ arguments. It is a fact of life on most discussion forums, but it seems to me that is happens disproportionately often here. Why? Is it just because people are passionately committed to their listening experience, or is it a _giving in_ to the urge to indulge in "opinionated utterances"? I would think these discussions would be far more fruitful to the casual reader (such as myself) if I wasn't a required to weed out 75% of the comments for their excesses and confrontational nature.
  
 I now have to assume the above paragraph is going to be taken as an unnecessarily harsh criticism, and if it is, that is a shame. But please realize I am not being critical of the *people* who invest their time and energy to express an opinion/knowledge, I am being critical of the confrontational behavior of our species, and the nature of some of the content itself. I wish the signal-to-noise ratio was better, that's all.


----------



## ag8908

cannikin said:


> I find this forum a *challenging* place to get good, genuinely useful information because it seems that many discussions quickly degrade into _"I'm right and you're wrong!"_ arguments. It is a fact of life on most discussion forums, but it seems to me that is happens disproportionately often here. Why? Is it just because people are passionately committed to their listening experience, or is it a _giving in_ to the urge to indulge in "opinionated utterances"? I would think these discussions would be far more fruitful to the casual reader (such as myself) if I wasn't a required to weed out 75% of the comments for their excesses and confrontational nature.
> 
> I now have to assume the above paragraph is going to be taken as an unnecessarily harsh criticism, and if it is, that is a shame. But please realize I am not being critical of the *people* who invest their time and energy to express an opinion/knowledge, I am being critical of the confrontational behavior of our species, and the nature of some of the content itself. I wish the signal-to-noise ratio was better, that's all.


 

 In my opinion, there's nothing wrong with expressing your opinion and ending it. It's just that when someone who doesn't like what you say impersonates a moderator and threatens you with a "warning" for stating your opinion or makes childish taunts that have nothing to do with the discussion that it becomes a problem. It's quite sad if you think of it.


----------



## ralphp@optonline

digitalchkn said:


> Greetings, from a fellow engineer.
> 
> Nyquist is not wrong. But the idea that 22KHz BW is sufficient for human hearing is in question.  While we may not hear sinosoids at 22KHz, it may be that people do some perceive energy up at and beyond those frequencies.
> 
> Besides, remember that the oddball sampling rate of 44.1KHz came from a practical need to sync up to video recorder frequencies of the time. At the time it was deemed sufficient to cover the 20KHz bandwidth that was common belief to be sufficient.  We ended up being stuck with it today.




Question: if, as you state, it is possible that some people can perceive energy above the 20khz threshold has there ever been a test where a recording with only sounds above 20khz are played to see if some people can hear or sense that the recording is playing?

Such a test would put to rest the question of whether not these high frequencies matter or not.


----------



## Digitalchkn

ralphp@optonline said:


> Question: if, as you state, it is possible that some people can perceive energy above the 20khz threshold has there ever been a test where a recording with only sounds above 20khz are played to see if some people can hear or sense that the recording is playing?
> 
> Such a test would put to rest the question of whether not these high frequencies matter or not.


 
  
 What would be the value of that test?  I am not aware of a recording that has sounds only above 20KHz.


----------



## riverlethe

digitalchkn said:


> There is general agreement that even not-so-high-end headphones have higher resolution than many of the better speaker systems.  So it depends on what you prefer.
> 
> But why listen to others.
> 
> I have done my own tests comparing 16/44 to even 24/48. I used same quality 24/96 source with large dynamic range to generate 16/44, 16/44 dithered, 24/48 and  16/44 upscaled to 24/48. I used quality conversion software with floating point resolution so the conversion error is well below the 24 bit noise floor.  On my 24 bit headphone chain I CAN TELL A DIFFERENCE between 16/44 and 24/48 versions.




Nice. Can you demonstrate to anyone else that you CAN TELL A DIFFERENCE under proper testing conditions?


----------



## ralphp@optonline

digitalchkn said:


> What would be the value of that test?  I am not aware of a recording that has sounds only above 20KHz.


 
 I think that you misunderstood me. One of the reasons that people claim that recordings having a sampling rate above 44.1kHz are needed is that some people can perceive energy/sound above the 20khz threshold and by having a recording with only frequencies above 20kHz would show that this is either true or false. My guess is that no human will be able to hear these frequencies, which is not something the people pushing high resolution would like to have proved.
  
 Next up: the myth about higher sampling rates producing smoother, i.e. more accurate, sine waves at all frequencies but especially at or near 20kHz. Oh wait that one's probably been bandied about in the High-end Audio Forum section. I forgot that this is the Sound Science section.


----------



## Digitalchkn

riverlethe said:


> Nice. Can you demonstrate to anyone else that you CAN TELL A DIFFERENCE under proper testing conditions?


 
  
 I see no reason. If I can tell a difference then I am convinced. I don't have to convince anyone else


----------



## riverlethe

digitalchkn said:


> I see no reason. If I can tell a difference then I am convinced. I don't have to convince anyone else




Then why state it in a public forum? 

Seriously, what sort of test did you do?


----------



## Digitalchkn

ralphp@optonline said:


> I think that you misunderstood me. One of the reasons that people claim that recordings having a sampling rate above 44.1kHz are needed is that some people can perceive energy/sound above the 20khz threshold and by having a recording with only frequencies above 20kHz would show that this is either true or false. My guess is that no human will be able to hear these frequencies, which is not something the people pushing high resolution would like to have proved.
> 
> Next up: the myth about higher sampling rates producing smoother, i.e. more accurate, sine waves at all frequencies but especially at or near 20kHz. Oh wait that one's probably been bandied about in the High-end Audio Forum section. I forgot that this is the Sound Science section.


 

 My argument against this test is that it may not be representative of real-world acoustic content.  For instance you can you high frequency sine waves all you want, this testing may be limited in use. Basically, it's a similar type of argument that we are not sensitive to the specific phase of the sine wave, but it doesn't mean that phase response is not important.


----------



## Digitalchkn

riverlethe said:


> Then why state it in a public forum?
> 
> Seriously, what sort of test did you do?


 
  
 As I mentioned in my previous post I compared 16/44 to 24/48. I used same quality 24/96 source with large dynamic range to generate 16/44, 16/44 dithered, 24/48 and  16/44 upscaled to 24/48. I used decent quality conversion software with floating point resolution so the conversion noise is well below the 24 bit noise floor.   I did 24/48 so that I go through a similar conversion process from 24/96 as 16/44 (not identical, of course). The upscaling test is primarily to circumvent any issue that *may* arise with my DAC handling of different sampling rates (in other words, it also goes through similar conversion process rather than introduce one by the DAC that I can't quantify).


----------



## Brooko

digitalchkn said:


> As I mentioned in my previous post I compared 16/44 to 24/48. I used same quality 24/96 source with large dynamic range to generate 16/44, 16/44 dithered, 24/48 and  16/44 upscaled to 24/48. I used decent quality conversion software with floating point resolution so the conversion noise is well below the 24 bit noise floor.   I did 24/48 so that I go through a similar conversion process from 24/96 as 16/44 (not identical, of course). The upscaling test is primarily to circumvent any issue that *may* arise with my DAC handling of different sampling rates (in other words, it also goes through similar conversion process rather than introduce one by the DAC that I can't quantify).


 
  
 Was the test sighted, blind, ?  What software did you use?  How many iterations?  How did you volume match?
  
 Genuinely interested in your methodology - of the actual test itself.
  
 For the record - I have performed similar - using ABX software (so completely blind) - multiple iterations (usually 15-20 attempts at a time).  Tracks are volume matched to 0.1 dB.
  
 Test results show I can't tell the difference - however my ears are reasonably aged (47), and I know my hearing is a lot less than perfect (one of the drawbacks of age 
	

	
	
		
		

		
			





) 
  
 In the past when we've asked others to do this - the people who have actually performed proper blind abx tests haven't been able to tell differences either.  The ones that claim to be able to tell are usually the ones who will only do it sighted, and who won't produce abx logs sadly.
  
 Please note - I'm not trying to spark a war here.  I am genuinely interested.


----------



## dikkiedirk

nanaholic said:


> It doesn't really help to have all the high quality gear in the world when the music industry goes back to being lazy/careless and making poor masters.  No amount of high-end gear can fix clipping for example, so all those songs that are mixed too hot is ruined forever and no file format or bitrate will fix it.


 
 That happened/happens a lot. Many producers don't know or don't care that when recording at levels of 0dB or over it on digital equipment results in distortion (clipping). Analog equipment is more forgiving.


----------



## Digitalchkn

brooko said:


> Was the test sighted, blind, ?  What software did you use?  How many iterations?  How did you volume match?
> 
> Genuinely interested in your methodology - of the actual test itself.
> 
> ...


 
  
 I am a bit younger (35) but I can tell my hearing isn't what it used to be either.
  
 I used the ABX foobar2000 plug in to do double blind tests I think maybe 10 comparisons. The volumes were kept identical -- the difference is only related to the sampling rate/bit depth conversion.
  
 I don't recall exactly my source material but remember chosing some acoustic material with wide spectrum (harmonics in some passages close to 30KHz), large dynamic range, (read: little or no compression) and very pronounced 3D soundfield (in other words, not point sources with synthesized reverb but naturally captured acoustics).
  
 The effect was that more of 3D perception of the soundstage. In other words, the 16/44 slightly flattened the soundstage.  24/48 made it more 3-dimensional.  The difference is very subtle, of course.
  
 On the material I used, the difference between "truncated" 16/44 and 24/48 was more noticeable.The dithered version of 16/44 was much more of a challenge, just on the border of being barely noticeable.  Note that both versions were upscaled first.
  
 I didn't test 16/44 vs 24/96 in this case because I thought that to be an unfair comparison.


----------



## riverlethe

digitalchkn said:


> I am a bit younger (35) but I can tell my hearing isn't what it used to be either.
> 
> I used the ABX foobar2000 plug in to do double blind tests I think maybe 10 comparisons. The volumes were kept identical -- the difference is only related to the sampling rate/bit depth conversion.
> 
> ...




Ten comparisons of 16/44 "truncated" vs 24/48 and 16/44 dithered vs 24/48? And you got 7/10 right in each run?

Here's a study with dozens of participants and 554 trials: 

http://www.mixonline.com/recording/mixing/audio_emperors_new_sampling/


----------



## stv014

It would be most helpful to post some samples (with a maximum length of 30 seconds) of the actual files that were compared with the ABX plugin, and for converted files preferably also the original version. This way, others can try to reproduce the results with ABX testing, and/or find potential errors (like bad conversion, unmatched levels, etc.) easier.


----------



## Digitalchkn

riverlethe said:


> Ten comparisons of 16/44 "truncated" vs 24/48 and 16/44 dithered vs 24/48? And you got 7/10 right in each run?
> 
> Here's a study with dozens of participants and 554 trials:
> 
> http://www.mixonline.com/recording/mixing/audio_emperors_new_sampling/


 
  
 Good enough for me to be convinced. Not planning to release a report to AES anytime soon.


----------



## riverlethe

digitalchkn said:


> Good enough for me to be convinced. Not planning to release a report to AES anytime soon.




Well, perhaps you'll forgive me if I'm not the least bit impressed.


----------



## Brooko

riverlethe said:


> Ten comparisons of 16/44 "truncated" vs 24/48 and 16/44 dithered vs 24/48? And you got 7/10 right in each run?
> 
> Here's a study with dozens of participants and 554 trials:
> 
> http://www.mixonline.com/recording/mixing/audio_emperors_new_sampling/


 
  
 Thanks.  Really enjoyed reading that article.


----------



## Digitalchkn

riverlethe said:


> Well, perhaps you'll forgive me if I'm not the least bit impressed.


 
  
 You are most certainly forgiven.  It's a test for me, in my home setup, under reasonably know conditions, with best that I have available. It's not meant to be a be-all, end-all universal conclusion to the age-old question of "24 bits vs 16 bits".  The objective is not for me to convince the rest of the world... I get no finance gain from it.


----------



## Brooko

digitalchkn said:


> I am a bit younger (35) but I can tell my hearing isn't what it used to be either.
> 
> I used the ABX foobar2000 plug in to do double blind tests I think maybe 10 comparisons. The volumes were kept identical -- the difference is only related to the sampling rate/bit depth conversion.
> 
> ...


 
  
 Thanks.  I have to take you at your word - but as Steve requested, some of the samples you used would be nice to look at.  I'll remain skeptical (basically you appear to be one "anomaly" amongst the rest of us ordinary folk - and I don't mean that to sound derogatory).
  
 Can I suggest (in all seriousness) that you contact someone like Ethan Winer (who we see from time to time on these forums) to set up another test?  Assuming you can do this consistently and that there is nothing wrong with the files or testing method - this could be important actually establishing that there is an audible difference.  You would be helping a far wider community.  Plus I for one would be fascinated knowing that it can actually be done.
  
 It would also make Neil Young's current crusade a lot more believable.


----------



## Digitalchkn

brooko said:


> Thanks.  I have to take you at your word - but as Steve requested, some of the samples you used would be nice to look at.  I'll remain skeptical (basically you appear to be one "anomaly" amongst the rest of us ordinary folk - and I don't mean that to sound derogatory).
> 
> Can I suggest (in all seriousness) that you contact someone like Ethan Winer (who we see from time to time on these forums) to set up another test?  Assuming you can do this consistently and that there is nothing wrong with the files or testing method - this could be important actually establishing that there is an audible difference.  You would be helping a far wider community.  Plus I for one would be fascinated knowing that it can actually be done.
> 
> It would also make Neil Young's current crusade a lot more believable.


 
  
 I surprised this is generating so much interest.  Unfortunately I didn't keep a very good record of the testing when I did this few months back (I don't even remember exactly when I did it).
 It would be interesting to see this kind of test on a bigger scale.


----------



## riverlethe

digitalchkn said:


> I surprised this is generating so much interest.  Unfortunately I didn't keep a very good record of the testing when I did this few months back (I don't even remember exactly when I did it).
> It would be interesting to see this kind of test on a bigger scale.




Well, your claim probably wouldn't qualify for the $1 million JREF prize, but it certainly contradicts the conventional wisdom. The article I linked to talks about just such a test. I don't think any others have been done since then.


----------



## Brooko

digitalchkn said:


> I surprised this is generating so much interest.  Unfortunately I didn't keep a very good record of the testing when I did this few months back (I don't even remember exactly when I did it).
> It would be interesting to see this kind of test on a bigger scale.


 
  
 Check that link that Riverlethe left (http://www.mixonline.com/recording/mixing/audio_emperors_new_sampling/)
  
 This sort of testing has been done on a large scale.  That's why I was so interested in your result.  If you can consistently identify the differences using different samples - then your results are important.  But it would have to be redone with independent verification (which is why I suggested contacting Ethan).  At the very least - contact him.  I'd imagine he would be interested .......
  
 When was the last time you tested?


----------



## Digitalchkn

brooko said:


> Check that link that Riverlethe left (http://www.mixonline.com/recording/mixing/audio_emperors_new_sampling/)
> 
> This sort of testing has been done on a large scale.  That's why I was so interested in your result.  If you can consistently identify the differences using different samples - then your results are important.  But it would have to be redone with independent verification (which is why I suggested contacting Ethan).  At the very least - contact him.  I'd imagine he would be interested .......
> 
> When was the last time you tested?


 
  
 I see...
  
 I glanced at the testing method in the article.  There are bunch of test conditions that seem different.   For one,
 - Using headphone set up  (Centrance DACMini ->  Beyer T1)  rather than speakers
 - Exactly same analog chain in all cases rather than switching different D/As
  
 I must have done this around last October or so.
  
 By the way, I don't have hearing that of a dog.  These days my "bandwidth" rolls off just shy of 17.5KHz when subjected to sines, including this headphone setup.


----------



## Brooko

Premise is the same though - even if method is different.
  
 If you can still tell the difference consistently, and across different samples - then you should let someone independently test you.  What your claiming (to my knowledge) has never been shown before (ie consistently and across differing samples).  That makes this a special case.
  
*Question is will you do it?*  Like I suggested - contact Ethan (http://www.head-fi.org/u/160148/ethanwiner).  I'm sure he would be interested.


----------



## riverlethe

Why can't we confine the sample to what he mentioned?  I'd be impressed at any positive result.


----------



## Brooko

Sorry I'll clarify - not sample as in sample bit-rate.  But sample as in example.  If he can consistently show positive results using 2 or 3 examples (so it's not just one track) - then this is pretty impressive, and worth pursuing.  He'd have to have it independently tested though.  Then it's believable.


----------



## riverlethe

brooko said:


> Sorry I'll clarify - not sample as in sample bit-rate.  But sample as in example.  If he can consistently show positive results using 2 or 3 examples (so it's not just one track) - then this is pretty impressive, and worth pursuing.  He'd have to have it independently tested though.  Then it's believable.


 
  
 I know, but why not just one track?  Use his source and headphones with his choice of track(s).  Of course, make sure that the chosen track is carefully downsampled and level-matched, etc.  I'll put up $10.


----------



## RazorJack




----------



## Brooko

riverlethe said:


> I know, but why not just one track?  Use his source and headphones with his choice of track(s).  Of course, make sure that the chosen track is carefully downsampled and level-matched, etc.  I'll put up $10.


 
  
 Well whatever the test is - it makes naff all difference if he doesn't take it somewhere that it can be independently verified as being a valid test.  Again - the skeptic in me suggests that as much as I'd like to see it, Digitalchkn probably won't go through with it.  Love to be proven wrong though.  I'll bow out now.  Have made the suggestion.


----------



## miceblue

riverlethe said:


> digitalchkn said:
> 
> 
> > I am a bit younger (35) but I can tell my hearing isn't what it used to be either.
> ...



That's between DSD and down-sampled PCM though. It's not exactly the a fair comparison to say higher sampling rates provide no audible benefits. XD


----------



## Digitalchkn

brooko said:


> Well whatever the test is - it makes naff all difference if he doesn't take it somewhere that it can be independently verified as being a valid test.  Again - the skeptic in me suggests that as much as I'd like to see it, Digitalchkn probably won't go through with it.  Love to be proven wrong though.  I'll bow out now.  Have made the suggestion.


 
  
 Now I am intrigued. I'll try to recreate my test (too bad I sold my T1s) and post my files for others to review.


----------



## utmelidze

i can say only:

*When i hear my 24/192 vynil Rips , they sound just briliant
*when i compare them to actual mastered cd , vynil rip sounds to me lot better


but

*if i downconvert my own source to 16/44.1...i hear a difference
*it doesnt sound smooth enough for me and everything lacks some kind of extension
*when too much instruments get together on same time interval...more resolution makes it very clear sounding

actually i woul say, if i convert my recordings to 24/48Khz and compare them to 24/192, i would hear nearly no difference
but if i downconvert it to 16/44.1 then hell yeah i hear the difference


it depends on source, music art , the audio technic you use for that and hearing ability
i dont mean with hearing ability that somebody has bat ears 
what i mean, that some people have trained and sensitive ears, they hear separations and most of the people just dont hear it,m because they never gave their ears enough time to train for it

the first touch with good quality audio on good equipment feels strange
it needs some time to get on..but once you start sensing details and separations, you would never ever go back


so my answer is, it depends on....
and of course nobody can hear 96Khz 


dont mix apples and melons, guys

16bit and 24bit give resolutions
but 192Khz ist way too much...you just dont need that
i use that only because i have lot of HDD space and i really dont care about
i want that last 0.0001% quality ...een if i use extra 10GB for that


16 bit can be on its limits when you hear complex music with lots of instruments at the same time
24 bit indeen over covers all the needed possibilities
you would never ever need 32 bit audio
because 24 bit is far more then needed , but 16 bit ist just a lil bit on its limit when it comes to detalisation


----------



## Brooko

So much wrong here I don't really know where to start ^^^
  
 The maximum effective bitrate you're going to get from vinyl is really only 12-14 bits (note I said maximum effective, not maximum possible).  Actually - rather than me regurgitating what is available on the net already - here are a couple of topics from over at hydrogenaudio that deal with it .....
 http://www.hydrogenaudio.org/forums/index.php?showtopic=93998
 http://www.hydrogenaudio.org/forums/index.php?showtopic=35530
  
 I won't go into the blind testing between different formats.  Too much debate - but in all of the properly controlled blind tests that I know of (ie actually reported independently), no-one so far has been able to successfully abx 16/44.1 from 24/96 (from the same master) unless there was issues with the hardware or transoding errors.
  
 You can prefer vinyl - many do - but ripping at anything above 16/44.1 for personal listening (ie no post processing) is really just wasting space.
  
 If it keeps you happy though - go for it


----------



## utmelidze

As far analog signal has infinite possibilities
You can never code it effective with any amount of buts


----------



## utmelidze

It doesnt matter what dynamic range are you coding
More bits mean fine step ups and downs
And thats on xyz 

I know.what do u mewn with that
And that happens very often
You are right in most way
But......16 bit steps arent fine enough for some worst case scenarios
Its not accident they choose 16 bit
It is good in most situationts snd for nearly everybody

16 bit has worse steps as i said
It means you need lil bit more capacity infouence to clean the stepy wave to wavy signal
More bit means less step level so less infouence on compensating
Its not some audio tech but just digital basics


----------



## brhfl

utmelidze said:


> As far analog signal has infinite possibilities
> You can never code it effective with any amount of buts


 
 Absolutely not true. When we speak of the bits in the digital realm, we're really talking about dynamic range, signal-to-noise ratio - ~96dB for Red Book (16 bit). SNR on vinyl, due to inherent, physical limitations, is considerably lower, optimistically around 70dB. A 12 bit PCM signal would be ~72dB.


----------



## riverlethe

miceblue said:


> That's between DSD and down-sampled PCM though. It's not exactly the a fair comparison to say higher sampling rates provide no audible benefits. XD




The article I linked to talks about a comparison between SACD and DVD-a vs those same sources down sampled through an analog-digital-analog loop. The 554 trials they performed demonstrate that it's a perfectly fair comparison.


----------



## riverlethe

This is an earlier study showing no discernible differences between DVD-A (PCM) and SACD (DSD). 

http://issuu.com/manger-msw/docs/aes_paper_6086/1


----------



## miceblue

riverlethe said:


> miceblue said:
> 
> 
> > That's between DSD and down-sampled PCM though. It's not exactly the a fair comparison to say higher sampling rates provide no audible benefits. XD
> ...



DSD is 1-bit, some megahertz sampling. That's not even close to 24-bit, 192 kHz sampling rates.

DSD vs PCM is a completely different argument from PCM vs PCM or even DXD.


----------



## ralphp@optonline

miceblue said:


> DSD is 1-bit, some megahertz sampling. That's not even close to 24-bit, 192 kHz sampling rates.
> 
> DSD vs PCM is a completely different argument from PCM vs PCM or even DXD.


 

 Despite the big DSD push from several high end audio manufacturers and the high end audio press, DSD is NEVER going to catch on because of the limitations imposed by DSD when it comes to editing and production. In order to edit a DSD recording one must convert the DSD files to PCM, edit and then convert the PCM files back to DSD. So other than a pure, unedited DSD recording ALL DSD recordings are, in essence, PCM.
  
 So comparing DSD to PCM is totally fair.


----------



## riverlethe

miceblue said:


> DSD is 1-bit, some megahertz sampling. That's not even close to 24-bit, 192 kHz sampling rates.
> 
> DSD vs PCM is a completely different argument from PCM vs PCM or even DXD.




The differences are moot if they're inaudible.


----------



## Don Hills

riverlethe said:


> The differences are moot if they're inaudible.


 

 Finally some sanity.


----------



## Brooko

utmelidze said:


> As far analog signal has infinite possibilities
> You can never code it effective with any amount of buts


 
  
  


utmelidze said:


> It doesnt matter what dynamic range are you coding
> More bits mean fine step ups and downs
> And thats on xyz
> 
> ...


 
  
 Actually you couldn't be more wrong.  I'm guessing you also didn't follow that first set of links I left for you either 
	

	
	
		
		

		
			




  
 If you don't follow any of the other links - *just please follow this one* - it is a video but worth it.  It explains directly why what you posted is a common fallacy - http://xiph.org/video/vid2.shtml
  
 And here's some commentary on Neil Young's Pono which has direct commentary on HD audio (what we can and can't hear).  Again - you may find it enlightening ......
 http://news.cnet.com/8301-1023_3-57620489-93/sound-bite-despite-ponos-promise-experts-pan-hd-audio/


----------



## Digitalchkn

>


 


brooko said:


> Actually you couldn't be more wrong.


 
  
  
  
  


brooko said:


> Actually you couldn't be more wrong.  I'm guessing you also didn't follow that first set of links I left for you either
> 
> 
> 
> ...


 
 I encourage folks arguing either side of the "analog vs digital" to refer to chapters 2 and 3  of Digital Audio Signal Processing by Udo Zolzer  for an excellent description of the sampling, reconstruction, quantization, and noise shaping theory, before accepting watered-down versions suggested on various youtube videos as the gospel. As usual, there is more than meets the eye.


----------



## Brooko

digitalchkn said:


> I encourage folks arguing either side of the "analog vs digital" to refer to chapters 2 and 3  of Digital Audio Signal Processing by Udo Zolzer  for an excellent description of the sampling, reconstruction, quantization, and noise shaping theory, before accepting watered-down versions suggested on various youtube videos as the gospel. As usual, there is more than meets the eye.




Funnily enough I downloaded and started reading the book. Basically got lost within the first few pages. You realise it's basically an advanced text book right? Not exactly user friendly to us hobbyists. Personally I'll stick to Montgomery's watered down version. At least I could understand that one. Thanks for the reference though.


----------



## Digitalchkn

brooko said:


> Funnily enough I downloaded and started reading the book. Basically got lost within the first few pages. You realise it's basically an advanced text book right? Not exactly user friendly to us hobbyists. Personally I'll stick to Montgomery's watered down version. At least I could understand that one. Thanks for the reference though.


 
 Well, I never said it is light reading 
	

	
	
		
		

		
		
	


	



  
 If you do go through it, one day, you'll find lots of fascinating things as a result. Some examples of this are:
 - Your ability to get a *perfect* copy of analog signal is to have a theoretically ideal reconstruction filter .... that is basically impossible to create in a real "PCM" DAC.  Fortunately, these days there are tricks to overcome this, but unfortunately it sometimes makes the direct A/B comparisons kinda to hard to make (particularly when using different dacs).
 - SNR is actually not a single number (e.g. 96dB for 16 bits) but highly depends on the "randomness" of the signal you are trying to sample
 - We can apply a mathematical "trick", referred to as Dithering, that takes advantage of limitations of our hearing to make it seem like we are using more bits. The effectiveness of this depends on the type of dithering that is being applied and, again, the input signal itself.
 - Analog tape noise (for instance) is not same *type* of noise as sampling noise, making things a bit more difficult to make apples-to-apples comparisons. In fact, digital noise changes depending on the input signal.
  
 That's why the watered-down version should be treated with care!


----------



## Brooko

Again though - in the "watered down version" and in his blog, Monty does list exactly those sort of things that we need to take care with (ie use of dither, background noise, theoretical noise floor etc, etc).  But what I like about his video and blog is that they are easier for a 'layman' (translate to idiot 
	

	
	
		
		

		
		
	


	




) like me to understand + he also applies it to what is real-world (ie audible).
  
 I think that Monty and (the little I understood of) Zolzer were in agreement though - it is possible to digitise a wave form so that any difference is inaudible.  That's where the rubber really hits the road (so to speak).
  
 Curious DC - what field are you in?  You seem very knowledgeable on the topic.  I'm enjoying your insights.


----------



## Digitalchkn

brooko said:


> Again though - in the "watered down version" and in his blog, Monty does list exactly those sort of things that we need to take care with (ie use of dither, background noise, theoretical noise floor etc, etc).  But what I like about his video and blog is that they are easier for a 'layman' (translate to idiot
> 
> 
> 
> ...


 
  
 I am neither in the music business nor in hi-fi equipment.  I am an electrical engineer in a field completely unrelated to audio. So this is purely self-interest. Been interested in sound recording, reproduction since I was a teenager.  Amateur guitarist (for fun). Occasionally make recordings in spare time in a home environment. Hence, the rate conversion software.
  
 Don't feel like you are left out. I don't think it's light reading for anyone, really.  It does help to have an electronics engineering degree with some signal processing focus to help dive deeper into nuts and bolts of it  (I don't use it much of it these days so am a bit rusty on the math).


----------



## miceblue

digitalchkn said:


> I encourage folks arguing either side of the "analog vs digital" to refer to chapters 2 and 3  of Digital Audio Signal Processing by Udo Zolzer  for an excellent description of the sampling, reconstruction, quantization, and noise shaping theory, before accepting watered-down versions suggested on various youtube videos as the gospel. As usual, there is more than meets the eye.



Interesting. I'll have to read through this now that I'm on spring break. 

I'm studying bioengineering, though far away from the tech stuff. I'm more into the biological side of it with tissue engineering and whatnot. I did take a few courses regarding signal processing though since 1) this hobby, 2) it might be useful for understanding cochlear implants (though again, I would be more interested in a tissue engineering approach to restore hearing).


----------



## stv014

brooko said:


> Funnily enough I downloaded and started reading the book. Basically got lost within the first few pages.


 
  
 Which is probably why it was posted. It is a common tactic by audiophiles to try to confuse people with walls of information, and use the "audio is infinitely complex and you do not understand it" argument to justify believing in whatever they want.


----------



## stv014

> Originally Posted by *Digitalchkn* /img/forum/go_quote.gif
> 
> In fact, digital noise changes depending on the input signal.


 
  
 Well, with dithering (the simple +/-1 LSB TPDF type), I can subtract the quantized signal from its original version, and the difference (the quantization error) sounds and looks (in a spectrum analyzer) like white noise at a constant RMS level no matter what the input signal is. This is easy to verify in practice.


----------



## stv014

> Originally Posted by *Digitalchkn* /img/forum/go_quote.gif
> 
> - We can apply a mathematical "trick", referred to as Dithering, that takes advantage of limitations of our hearing to make it seem like we are using more bits. The effectiveness of this depends on the type of dithering that is being applied and, again, the input signal itself.


 
  
 It is actually noise shaping that takes advantage of the limitations of hearing (notably the hearing threshold increasing significantly towards 20 kHz) to improve the perceived dynamic range of quantized audio. Simple dithering sounds like white noise, or just slightly colored to reduce the A-weighted noise level by 1-2 dB. Also, for correctly implemented dithering, the input signal should not matter, other than the theoretical possibility of it correlating with the PRNG used for dithering, which in practice should be negligible. Of course, a louder input will perceptually mask the noise more, but that is the same for analog noise as well.


----------



## Digitalchkn

>


 


stv014 said:


> Well, with dithering (the simple +/-1 LSB TPDF type), I can subtract the quantized signal from its original version, and the difference (the quantization error) sounds and looks (in a spectrum analyzer) like white noise at a constant RMS level no matter what the input signal is. This is easy to verify in practice


 
  
 If the signal has sufficiently large variance, then quantization noise looses correlation to the input signal, including the non-dithered quantization. Are you referring to non-shaped TPDF generating white noise?


----------



## Digitalchkn

stv014 said:


> Which is probably why it was posted. It is a common tactic by audiophiles to try to confuse people with walls of information, and use the "audio is infinitely complex and you do not understand it" argument to justify believing in whatever they want.


 
  
 It was posted for educational purposes. And yes, audio is pretty complex. That's why we are still finding new ways to make improvements in this field.


----------



## Illmatiic

Can anyone explain why my 24bit Vinyl downloads have a much lower play volume compared to my 16bit?
  
 The two files I'm comparing are the exact same song with the following specifications:
  
 24bit:
 Sample rate: 96khz
 Avg. Bitrate: 2761 kbps
  
 VS.
  
 16bit:
 Sample rate: 44.1khz
 Avg. Bitrate:  925kbps
  
 The problem persists when played through Windows Media Player, VLC, and Winamp.


----------



## miceblue

illmatiic said:


> Can anyone explain why my 24bit Vinyl downloads have a much lower play volume compared to my 16bit?
> 
> The two files I'm comparing are the exact same song with the following specifications:
> 
> ...



Is your 16-bit one a vinyl rip too?


----------



## Illmatiic

miceblue said:


> Is your 16-bit one a vinyl rip too?


 
  
 No, the 16-bit is CD rip.


----------



## miceblue

illmatiic said:


> miceblue said:
> 
> 
> > Is your 16-bit one a vinyl rip too?
> ...



Well n you're comparing 2 different masters. 

Vinyl rips are dependent on the rip settings and whatnot, so they're usually not too reliable.


----------



## pockits

This is very easy to uderstand in practice. Because the DR is higher for a 24bit depth sample. Since you have much more headroom available.


----------



## Illmatiic

miceblue said:


> Well n you're comparing 2 different masters.
> 
> 
> 
> ...


 
  
 Gotcha, I figured you were hinting that when you asked the question.


----------



## miceblue

Interesting:
http://www.aes.org/e-lib/browse.cfm?elib=9902
http://www.merging.com/uploads/assets//Merging_pdfs/dxd_Resolution_v3.5.pdf


----------



## Samuel777

thesuperguy said:


> If both formats are lossless, what differentiates the 2 versions in terms of sound quality if at all?


 
Why does 24 bit / 192 khz flac sound any better than 16 bit / 44.1 khz flac if both are lossless ? Difference come from the quality of headphones, and the quality of music player (source).
  
 sam


----------



## riverlethe

digitalchkn said:


> It was posted for educational purposes. And yes, audio is pretty complex. That's why we are still finding new ways to make improvements in this field.




Except, they're not really improvements if they're inaudible. The threshold of audibility in digital to analog conversion was probably passed 30+ years ago. Now, AAC might be a little better than mp3 at the same bitrate, but they're both pretty transparent at 256kbps+. Is anyone aware of a large scale study comparing mp3/AAC to lossless or CD quality?


----------



## DogMeat

samuel777 said:


> Why does 24 bit / 192 khz flac sound any better than 16 bit / 44.1 khz flac if both are lossless ? Difference come from the quality of headphones, and the quality of music player (source).
> 
> sam


 
 which TOTALLY makes sense if he's using the same gear for them both.




  
  
 The way I get my head around thinking of this is as being rather similar to video in the way moving images are better at higher FPS than at lower frame rates.
 There's no loss of DATA in each single frame, but faster rates make the moving image more coherent.
  
 Simplistic, but it kind of makes it more understandable to me.


----------



## kraken2109

dogmeat said:


> which TOTALLY makes sense if he's using the same gear for them both.
> 
> 
> 
> ...


 

 But this analogy fails because our eyes can actually process the extra frames in video, whereas our ears can't hear the higher frequencies that a higher sampling rate can provide.


----------



## elmoe

kraken2109 said:


> But this analogy fails because our eyes can actually process the extra frames in video, whereas our ears can't hear the higher frequencies that a higher sampling rate can provide.




Yet supposedly we shouldn't be able to see a difference between 60hz and 120hz screens but the difference is clear.


----------



## DogMeat

yes. 
which is why I said "simplistic".

I don't think it's just about pitch, though, is it?

Isn't it about richness too?

Help us out.
Can you give amore apropos way of thinking about it?
Not being challenging, just trying to grip this issue myself.

I take those tests...and I can generally tell the diff. about 90%, most days.

probly 'cuz I live with dogs. 
they've taught me all the tricks.


----------



## kraken2109

elmoe said:


> Yet supposedly we shouldn't be able to see a difference between 60hz and 120hz screens but the difference is clear.


 
 Where is the science that says we can't see the difference?
  


dogmeat said:


> yes.
> which is why I said "simplistic".
> 
> I don't think it's just about pitch, though, is it?
> ...


 
 Research the Nyquist-Shannon sampling theorem.


----------



## elmoe

kraken2109 said:


> Where is the science that says we can't see the difference?
> 
> Research the Nyquist-Shannon sampling theorem.




Its there, google for it if youre interested.


----------



## riverlethe

elmoe said:


> Its there, google for it if youre interested.




What is where?


----------



## riverlethe

dogmeat said:


> yes.
> which is why I said "simplistic".
> 
> I don't think it's just about pitch, though, is it?
> ...




Are you aware of the difference between "simplistic" and "completely invalid?"

What on earth is "richness?"


----------



## elmoe

Articles about the limit of what the human eye can see vs the reality, that at higher framerates things get smoother, more fluid, sharper and noticeably so.


----------



## riverlethe

elmoe said:


> Articles about the limit of what the human eye can see vs the reality, that at higher framerates things get smoother, more fluid, sharper and noticeably so.




Scientific studies, or just "articles?" Supposedly, fighter pilots have been shown to perceive 255 fps. Nerves can fire 1000 times per second, so that would be the hard limit. I'm not aware of anyone hearing above 22khz.


----------



## elmoe

riverlethe said:


> Scientific studies, or just "articles?" Supposedly, fighter pilots have been shown to perceive 255 fps. Nerves can fire 1000 times per second, so that would be the hard limit. I'm not aware of anyone hearing above 22khz.




Scientific studies yes, and now we have 1000hz screens that are also a step up from the 100/200/400hz screens (I saw one just today actually).

This kind of thing tends to make me think that the way we test many things aren't necessarily as accurate as we believe them to be. Take ABX DBTs for example, if its in fact true that our audible memory is very short then doesnt that make DBTs incredibly difficult to give accurate results? Im of the opinion that a long 30+ mins test before switching is much more plausible to accurately test differences in audio gear rather than a few minutes or even seconds as are most DBTs done.

Of course this is strictly based on my own experiences and not scientific studies but honestly a DBT with ABX where things are switched every few seconds seems to me like an innacurate way to properly test things.


----------



## riverlethe

elmoe said:


> Scientific studies yes, and now we have 1000hz screens that are also a step up from the 100/200/400hz screens (I saw one just today actually).
> 
> This kind of thing tends to make me think that the way we test many things aren't necessarily as accurate as we believe them to be. Take ABX DBTs for example, if its in fact true that our audible memory is very short then doesnt that make DBTs incredibly difficult to give accurate results? Im of the opinion that a long 30+ mins test before switching is much more plausible to accurately test differences in audio gear rather than a few minutes or even seconds as are most DBTs done.
> 
> Of course this is strictly based on my own experiences and not scientific studies but honestly a DBT with ABX where things are switched every few seconds seems to me like an innacurate way to properly test things.




I'd like to see a link to such a study about framerate. 

This still says nothing about people hearing greater than 22khz or 96dB of dynamic range. It simply isn't analogous to frames per second.


----------



## elmoe

Well as said above fighter pilots see a max of 254 fpa supposedly, try a video game at 255fps then up the limit to 600 and I guarantee youll see a difference - I do anyway. Alternatively google for studies.

Youre right this isnt analogous to 24 bit 192khz, what it is analogous to is that our ways to measure thinga scientifically arent always as accurate as wed like to think they are. That being said, its rare when I hear a difference between 16/44 and 24/192 but it does happen sometimes on good recordings of the same mastering, or so I hear. However ir is a very faint difference so much in fact that IF there is indeed one, it really doesnt matter.


----------



## riverlethe

elmoe said:


> Well as said above fighter pilots see a max of 254 fpa supposedly, try a video game at 255fps then up the limit to 600 and I guarantee youll see a difference - I do anyway. Alternatively google for studies.
> 
> Youre right this isnt analogous to 24 bit 192khz, what it is analogous to is that our ways to measure thinga scientifically arent always as accurate as wed like to think they are. That being said, its rare when I hear a difference between 16/44 and 24/192 but it does happen sometimes on good recordings of the same mastering, or so I hear. However ir is a very faint difference so much in fact that IF there is indeed one, it really doesnt matter.




What game are you running at 600fps? Aren't they usually CPU-limited before 200fps?

Anyway, the only test I found reference to WAS the fighter pilots...


----------



## elmoe

Theres plenty you can try if you have a decent recent gaming rig, cod4 for example, just lower settings and resolution if needed to get higher fps and theres a console command to draw fpa on screen or you can use a capture app like fraps to draw fps as well.


----------



## Don Hills

elmoe said:


> ...  a DBT with ABX where things are switched every few seconds seems to me like an innacurate way to properly test things.


 
  
 There is nothing in the design of an ABX test that requires you to switch "every few seconds". You control when the switching occurs, not the test. You can listen for a week at a time to A, B, or X if you think it will help you to hear a difference.


----------



## elmoe

don hills said:


> There is nothing in the design of an ABX test that requires you to switch "every few seconds". You control when the switching occurs, not the test. You can listen for a week at a time to A, B, or X if you think it will help you to hear a difference.


 
  
 But how many of those tests actually do listen for more than say, 15 minutes before switching? The reports I've read for every single ABX test so far have always been significantly less than that. If you have contradicting reports, please share them, I'm interested.


----------



## stv014

> The reports I've read for every single ABX test so far have always been significantly less than that.


 
  
 That is because the testers quickly learn that they get better scores with faster switching. Auditory memory fades quickly, and longer switching times have been proven to make the results less reliable. In a proper ABX/DBT, it makes a negative result more likely, while in casual sighted testing, it is easier to get a false positive (i.e. imaginary difference) if there is a longer time to forget what the other source sounded like. Which is why audiophiles believe longer switching time is better, because it "improves" their usual (flawed) testing method. It is also why "modding" audio equipment (like op amp, cable, etc. upgrades which inherently take some time to perform) is popular, and large differences are often heard with no objective evidence.
  
 Note that fast switching ability does not mean that the sample length or the overall time spent on testing also has to be short; with a thorough comparison of a longer sample that has more different sounding parts, there may be a better chance of finding an audible artifact.


----------



## elmoe

stv014 said:


> That is because the testers quickly learn that they get better scores with faster switching. Auditory memory fades quickly, and longer switching times have been proven to make the results less reliable. In a proper ABX/DBT, it makes a negative result more likely, while in casual sighted testing, it is easier to get a false positive (i.e. imaginary difference) if there is a longer time to forget what the other source sounded like. Which is why audiophiles believe longer switching time is better, because it "improves" their usual (flawed) testing method.


 
  
 I don't agree here and you've clearly misunderstood me. You can take a 30s long recording, play it for 15 minutes, and then switch during your DBT. I don't see how your auditory memory will fade any quicker than if you gave it a single 30s listen. Auditory memory might fade quickly from a single listen but if you loop a short recording for 15 minutes, you will remember things a lot better than if you heard it only a single time. I don't see how it is easier to get a "false positive" this way, given that you take the time to do plenty of switching. It's an ABX DBT, it shouldn't matter how long you take, it's a blind test anyhow. If audiophile results get better when doing this it goes to show that I'm not entirely wrong about the methodology. You say testers know they get better scores switching quickly but you say they get more false positive switching slowly - which is it?


----------



## stv014

elmoe said:


> I don't agree here and you've clearly misunderstood me. You can take a 30s long recording, play it for 15 minutes, and then switch during your DBT.


 
  
 As I already explained, switching time (the delay between listening to the same part of two samples from A, B, X, and Y, for the purpose of making the decision for a trial), and the time spent on discovering possible differences in whatever way you like are different issues. Making the latter long and the former short is fine, and is probably a common approach to ABX testing.


----------



## elmoe

stv014 said:


> As I already explained, switching time (the delay between listening to the same part of two samples from A, B, X, and Y, for the purpose of making the decision for a trial), and the time spent on discovering possible differences in whatever way you like are different issues. Making the latter long and the former short is fine, and is probably a common approach to ABX testing.


 
  
 And a flawed one, in my opinion. You still haven't offered anything other than your personal opinion as far as that goes, so it remains an unsubstantiated claim.


----------



## stv014

> You say testers know they get better scores switching quickly but you say they get more false positive switching slowly - which is it?


 
  
 I am not sure if you misunderstood what I said, or are intentionally trying to create confusion. I meant false positives become even more likely in *sighted* testing. Shorter switching delay (which does *not* equal short time spent on testing overall) improves scores in *blind* testing. In either case, lacking the ability of fast switching makes an incorrect (be it false positive or negative) result more likely.


----------



## elmoe

stv014 said:


> I am not sure if you misunderstood what I said, or are intentionally trying to create confusion. I meant false positives become even more likely in *sighted* testing. Shorter switching delay (which does *not* equal short time spent on testing overall) improves scores in *blind* testing. In either case, lacking the ability of fast switching makes an incorrect (be it false positive or negative) result more likely.


 
  
 Let's not have a discussion based on the premise that either one of us is trying to intentionally confuse the other please. We'll waste less time and show more respect this way.
  
 Sighted testing isn't testing at all, as the tester can lie to get a perfect score, so let that not even enter the discussion. Where is the evidence that shorter switching delay improves scores in blind testing? I think if anything, the contrary is more logical and seems to go along with my personal experience. It's a lot easier (when there actually IS a difference, mind you) to get accurate results (ie: to hear the difference) when I take the time to listen to the recording repetitively before switching than it is when I listen for a short amount of time before switching. My ears (brain?) need some time to adapt, if I don't take that time then the results are ultimately flawed, especially if the differences are subtle.


----------



## stv014

elmoe said:


> And a flawed one, in my opinion. You still haven't offered anything other than your personal opinion as far as that goes, so it remains an unsubstantiated claim.


 
  
 As opposed to yours ? There has definitely been research done on fast vs. slow switching times in blind tests, as well as on the recommended maximum delay, maybe someone who has more time to spend (waste ?) on arguing on internet forums will bother to look them up, and post the links. Also, your assumption that ABX testers never spend significant time on finding differences is not based on any real data (and is false in my case, for example).


----------



## stv014

> Where is the evidence that shorter switching delay improves scores in blind testing?


 
  
 As mentioned above, it exists, but I do not find it worth spending the time to look it up (knowing that so far I have never encountered a case when it was not a waste of time in similar arguments, they just kept going around in circles). Even if I posted a link to a paper, you would predictably just question its validity. I do not post frequently on Head-Fi these days, and prefer not to spend my time on pointless arguments, so I just add you to my block list.


----------



## elmoe

stv014 said:


> As opposed to yours ? There has definitely been research done on fast vs. slow switching times in blind tests, as well as on the recommended maximum delay, maybe someone who has more time to spend (waste ?) on arguing on internet forums will bother to look them up, and post the links. Also, your assumption that ABX testers never spend significant time on finding differences is not based on any real data (and is false in my case, for example).


 
  
 My assumption is that it doesn't matter how much time you spent if you're only listening to a short time before switching.
  
 Mine is an opinion that varies greatly from the "usual procedure", so obviously, aside from my personal experience, there isn't going to be much in the way of evidence. You're free to test it out for yourself if you're interested though. I'll be waiting for those links, as for wasting your time, you don't have to reply or even read my posts, if you feel this discussion is a waste of time then why do you bother replying?
  
 You're starting to sound much like those stubborn audiophiles who aren't interested in any discussion and believe whatever they hear is right. 
  
  


stv014 said:


> As mentioned above, it exists, but I do not find it worth spending the time to look it up (knowing that so far I have never encountered a case when it was not a waste of time in similar arguments, they just kept going around in circles). Even if I posted a link to a paper, you would predictably just question its validity. I do not post frequently on Head-Fi these days, and prefer not to spend my time on pointless arguments, so I just add you to my block list.


 
  
 Well, then add me to your block list if that's what you want 
	

	
	
		
		

		
			





 It's hilarious that your arguments here are the very same ones audiophiles use to disprove your own arguments - "whatever I say you're going to question the validity", "I'll block you and ignore you", etc etc.
  
 So you're allowed to question the validity of my opinion but when I question yours and ask for proof, you don't want to waste your time showing me that proof, you'd rather just block me?
  

  
 Thanks, this goes a long way to make me realize that people like you in the sound "science" forum aren't all that scientific to begin with.


----------



## limpidglitch

When in doubt, Wikipedia.
  
 I leave it to elmoe to check the references.


----------



## elmoe

limpidglitch said:


> When in doubt, Wikipedia.
> 
> I leave it to elmoe to check the references.


 
  
 Thanks, but this is not what we were talking about. The point that I made was that I need time to adapt to a particular sound signature before switching to another for the DBT to be worthwhile, it's different than echoic memory.


----------



## limpidglitch

elmoe said:


> …I need time to adapt to a particular sound signature before switching to another for the DBT to be worthwhile…


 

 No worry, with DBT you can do that.


----------



## elmoe

limpidglitch said:


> No worry, with DBT you can do that.


 
  
 I know, my point was I cannot trust a DBT test without this, because without a time to adapt nobody knows what anything really sounds like.


----------



## limpidglitch

elmoe said:


> I know, my point was I cannot trust a DBT test without this, because without a time to adapt nobody knows what anything really sounds like.


 
  
 You'll just have to trust that the subjects are doing their best.
  
 Have you ever been tested?


----------



## kraken2109

I think this thread may have gone slightly off topic...


----------



## castleofargh

elmoe said:


> limpidglitch said:
> 
> 
> > When in doubt, Wikipedia.
> ...


 
  
 pretty much the opposite of what I experience. I could tell you a list of sonic differences from listening to the same gear with the same music file if you just give enough time to my brain to start making stuff up. and I would believe I'm right.
 I wrote something just yesterday about some 3seconds lag that was already enough for me to feel insecure about my analysis (the time needed for me to unplug one source and replug the other one). I didn't know about that echoic memory thing(super interesting), but I guess it just explains what I always felt in practice.


----------



## esldude

elmoe said:


> My assumption is that it doesn't matter how much time you spent if you're only listening to a short time before switching.
> 
> Mine is an opinion that varies greatly from the "usual procedure", so obviously, aside from my personal experience, there isn't going to be much in the way of evidence. You're free to test it out for yourself if you're interested though. I'll be waiting for those links, as for wasting your time, you don't have to reply or even read my posts, if you feel this discussion is a waste of time then why do you bother replying?
> 
> ...


 

 Elmoe,
  
 Listen to what folks are telling you.  It has indeed been tested rigorously.  Subjects detect differences at lower levels with fast switching rather than with slower switching.  Yes, the common argument you are making when tested doesn't work out to be true.  You do less well if the switching is very long.  Past 10 seconds and things fall off quite a bit. In speech intelligibility testing somewhere around 200 msec switching was needed for the most discriminating results.  You probably feel much better and more confident with longer times between comparisons, but despite your feelings of confidence your accurate discrimination will be worse.
  
 http://www.nousaine.com/nousaine_tech_articles.html
  
 On this page, down in the middle of it read the PDF from a magazine article on_ Flying Blind Long Term Listening.   _There are more scholarly works on the subject.  But this gets to the gist of it.  And we'll see if you believe it or just retrench as stv104 and I think you will.  We've been here, and done that about a million times.  It gets old.  You can supply all the credible info you want to most people, and they just refuse to believe it because they don't want to.  Maybe you are different, I hope so.  So far your postings follow right along with someone who we are wasting our time to converse with. Not trying to be lacking in respect for you, just being honest.


----------



## bigshot

elmoe said:


> Thanks, but this is not what we were talking about. The point that I made was that I need time to adapt to a particular sound signature before switching to another for the DBT to be worthwhile, it's different than echoic memory.




I find that for myself, quick switching back and forth between samples is the best way to determine whether a difference exists between two very similar samples. In order to figure out exactly what the difference is, it takes a little longer on each sample. But after a minute or two on any sample, my ears adjust and I'm not getting anything out of it any more.


----------



## elmoe

limpidglitch said:


> You'll just have to trust that the subjects are doing their best.
> 
> Have you ever been tested?


 
  
 I've done the test myself, with someone to do the switching for me, which is how I found out that taking my time between switches helped. 


kraken2109 said:


> I think this thread may have gone slightly off topic...


 
  
 Yes sorry about that, but it seems most of the responses above cover what I wanted to know so I won't hijack it much further.
  
  


castleofargh said:


> pretty much the opposite of what I experience. I could tell you a list of sonic differences from listening to the same gear with the same music file if you just give enough time to my brain to start making stuff up. and I would believe I'm right.
> I wrote something just yesterday about some 3seconds lag that was already enough for me to feel insecure about my analysis (the time needed for me to unplug one source and replug the other one). I didn't know about that echoic memory thing(super interesting), but I guess it just explains what I always felt in practice.


 
  
 I think you've misunderstood, the time in between the switch (the 3 seconds lag as you called it) is not what I'm referring to. That lag should be as minimal as possible so you get to hear the other source or whatever it is you're testing, right away. I was saying that given a small recording (say 30s), I need to listen to that recording for 5-15 minutes before switching between source (or whatever I'm testing) so that my ears can adapt to its sound signature before I can pick out any differences.
  
  


esldude said:


> Elmoe,
> 
> Listen to what folks are telling you.  It has indeed been tested rigorously.  Subjects detect differences at lower levels with fast switching rather than with slower switching.  Yes, the common argument you are making when tested doesn't work out to be true.  You do less well if the switching is very long.  Past 10 seconds and things fall off quite a bit. In speech intelligibility testing somewhere around 200 msec switching was needed for the most discriminating results.  You probably feel much better and more confident with longer times between comparisons, but despite your feelings of confidence your accurate discrimination will be worse.
> 
> ...


 
  
 Ok I've finished reading that article, and what I'm saying is very different from what's argued in it, so let me be clearer. In the article, it's explained that long term listening (we're not talking about 15 minutes, then switch here, but about many many hours/days/weeks of listening without any switching) is not going to accurately help you find differences as opposed to a DBT ABX with switching.
  
 I couldn't agree more with that, but that is not what I'm saying. I'm saying that WITHIN AN ABX DBT, instead of listening for only a FEW SECONDS before switching, we should be listening for 5-15 minutes and THEN switch, so our ears can adapt somewhat to the sound signature before we switch, thus making the difference more obvious.
  
 That being said, I do think that quick switching can be the better choice for some things (such as this thread for example, determining whether or not 2 samples, one at 116/44 and the other at 24/192, are different or not). However, when comparing gear, for example 2 different DACs, fast switching is more likely to confuse the brain than to help pick out differences. So if you're going to pick out one aspect (such as in the tests from the article, distortion), then quick switching is no doubt most effective and that's logical because you're listening for it during the test, so you don't need prolonged exposure to hear it - you either do or you don't. But comparing 2 different DACs, you're not listening just for distortion or any single aspect, there are many many things to compare and without taking your time with each, there is no way to accurately compare their sound.
  
 So yes, I believe you and this article absolutely, but that's beside the point and it isn't what's ultimately interesting to me: knowing if different gear brings different sound and how. So you're right, I was too hasty before and only remembered back to the few ABX DBT tests I did myself comparing gear, without thinking about the usefulness of these tests to compare smaller aspects such as "is there or is there not distortion". For these kinds of tests, I completely agree, quick switching is the prerogative. My problem is that the same procedure is used when comparing gear and that seems flawed to me. I'm not trying to be stubborn, it just seems illogical.
  
  


bigshot said:


> I find that for myself, quick switching back and forth between samples is the best way to determine whether a difference exists between two very similar samples. In order to figure out exactly what the difference is, it takes a little longer on each sample. But after a minute or two on any sample, my ears adjust and I'm not getting anything out of it any more.


 
  
 Yes I can see how that would be the case when comparing 2 short samples. But like I said above, when comparing 2 different DACs or amps or any other piece of gear, wouldn't you prefer to take the time to a) try a variety of recordings and b) listen long enough to register all the different ways it sounds?


----------



## limpidglitch

elmoe said:


> I've done the test myself, with someone to do the switching for me, which is how I found out that taking my time between switches helped.


 
  
 I've had much the same experience as bigshot, but then I don't have much experience with comparing DACs and amplifiers in this way.

 If you've had more success in blind tests with longer switching times, there's really not much to argue about.


----------



## elmoe

limpidglitch said:


> I've had much the same experience as bigshot, but then I don't have much experience with comparing DACs and amplifiers in this way.
> 
> If you've had more success in blind tests with longer switching times, there's really not much to argue about.


 
  
 I agree, but it's nice to see what others have experienced doing the same kind of testing.


----------



## bigshot

elmoe said:


> Yes I can see how that would be the case when comparing 2 short samples. But like I said above, when comparing 2 different DACs or amps or any other piece of gear, wouldn't you prefer to take the time to a) try a variety of recordings and b) listen long enough to register all the different ways it sounds?


 
  
 A yes, B not more than a couple of minutes between switches.
  
 I generally switch back and forth a lot in different parts of the music and with different music. But if I stay on any one sample longer than a couple of minutes, I end up listening to the music, not listening for any difference. My ears and brain just can't hold very similar comparisons that long. If I listen for a long time, I often hear things that I didn't hear before, but when I take that section and switch A/B over it, I realize it was just my faulty memory or not paying attention. The difference is in my memory, not in the test.


----------



## elmoe

bigshot said:


> A yes, B not more than a couple of minutes between switches.
> 
> I generally switch back and forth a lot in different parts of the music and with different music. But if I stay on any one sample longer than a couple of minutes, I end up listening to the music, not listening for any difference. My ears and brain just can't hold very similar comparisons that long. If I listen for a long time, I often hear things that I didn't hear before, but when I take that section and switch A/B over it, I realize it was just my faulty memory or not paying attention. The difference is in my memory, not in the test.


 
  
 That makes sense actually. I'll have to try another DBT just to confirm when I get another DAC.


----------



## Makiah S

http://people.xiph.org/~xiphmont/demo/neil-young.html
  
 Here read through this, I felt it makes a lot of sense, and further supports my opinion that 24/192 formatting is uncessiarly terrible for play back


----------



## gevorg

What if you're adding DSPs to your playback chain, such as TB Isone. Wouldn't high resolution recordings (in particular, the 24-bit side of them) will give you more "headroom" for DSP processing, the same headroom that pro audio guys use when they record/master in 24/96 and then down-convert to redbook when they're done.


----------



## kraken2109

gevorg said:


> What if you're adding DSPs to your playback chain, such as TB Isone. Wouldn't high resolution recordings (in particular, the 24-bit side of them) will give you more "headroom" for DSP processing, the same headroom that pro audio guys use when they record/master in 24/96 and then down-convert to redbook when they're done.


 
 It's possible that 24bit would help, but chances are you wouldn't even be using half of what 16bit is capable of.


----------



## bigshot

I would think that if a DSP required that kind of headroom, it would automatically upsample/process/downsample on the fly. It wouldn't require you to have high bitrate music.


----------



## castleofargh

gevorg said:


> What if you're adding DSPs to your playback chain, such as TB Isone. Wouldn't high resolution recordings (in particular, the 24-bit side of them) will give you more "headroom" for DSP processing, the same headroom that pro audio guys use when they record/master in 24/96 and then down-convert to redbook when they're done.


 

 that's a very good point. I guess it would be hard to make a general statement because each dsp does its own thing. I only get how it works for pictures, where the resulting quality depends a lot on how much data we have before the effect. but I guess it would translate more into sample rate than into bits in our situation right? and as long as the sample is big enough to get the right wave, should it matter?
 argghhhh I wish I would know more about all this.


----------



## ralphp@optonline

castleofargh said:


> that's a very good point. I guess it would be hard to make a general statement because each dsp does its own thing. I only get how it works for pictures, where the resulting quality depends a lot on how much data we have before the effect. but I guess it would translate more into sample rate than into bits in our situation right? and as long as the sample is big enough to get the right wave, should it matter?
> argghhhh I wish I would know more about all this.


 

 There have been links to several very good articles and videos about digital audio posted throughout this thread so if you go back and read/watch some of them then you will "know more about all this". One thing you will learn the difference between sampling rate and bit depth. Another would be that a higher sampling rate does not mean a "smoother" sine wave, which is a common audiophile myth, often passed on by the clowns in the audio press. And yet another would be that a greater bit depth does not add any dynamic range to the recording, greater bit depth just means that more dynamic range is available, however it the actual music on the recording that provides the dynamic range. For example a recording of an acoustic guitar will have less dynamic range than a recording of a full symphony orchestra regardless of whether it is 16 or 24 bit, again a common myth that is not dispelled by the audio press.
  
 Lately I've been thinking about how the whole shift from disc based playback, be they black plastic LPs or shiny silver CDs or SACDS, to computer based playback has given the high end audio world a whole new area to exploit with misinformation and lies, very similar to the early days of the high end cable craze. More myths and misinformation basically means more completely useless but highly profitable products to sell to all the kool-aid drinking audiophiles.


----------



## castleofargh

ralphp@optonline said:


> castleofargh said:
> 
> 
> > that's a very good point. I guess it would be hard to make a general statement because each dsp does its own thing. I only get how it works for pictures, where the resulting quality depends a lot on how much data we have before the effect. but I guess it would translate more into sample rate than into bits in our situation right? and as long as the sample is big enough to get the right wave, should it matter?
> ...


 
 yup, that's why I said "and as long as the sample is big enough to get the right wave, should it matter?" having a few less "dots" on the wave wouldn't change the wave unless there was really too
 few reference values. but I don't really know how many is enough as music is not juste 1 sine wave.
  
 I hungrily eat anything you guys through at me, but unless it's basic and noob oriented (like the videos from xiph.org), I usually don't grasp most of it.


----------



## ralphp@optonline

castleofargh said:


> yup, that's why I said "and as long as the sample is big enough to get the right wave, should it matter?" having a few less "dots" on the wave wouldn't change the wave unless there was really too
> few reference values. but I don't really know how many is enough as music is not juste 1 sine wave.
> 
> I hungrily eat anything you guys through at me, but unless it's basic and noob oriented (like the videos from xiph.org), I usually don't grasp most of it.


 

 Unfortunately the xiph.org videos, while fairly simple and, as you say, aimed at noobs, could go a long way in saving audiophiles some money if they would only watch them instead of ust believing the lies and misinformation that come out of the high end audio press (and Mr. Neil Young, although I think he's just being misled by others looking to use his good name and money for ill gotten gains).


----------



## esldude

ralphp@optonline said:


> Lately I've been thinking about how the whole shift from disc based playback, be they black plastic LPs or shiny silver CDs or SACDS, to computer based playback has given the high end audio world a whole new area to exploit with misinformation and lies, very similar to the early days of the high end cable craze. More myths and misinformation basically means more completely useless but highly profitable products to sell to all the kool-aid drinking audiophiles.


 
 Amen to this.  I had hopes that simple good quality playback from a computer would fix all perceived and real ills for digital audio.  That finally much of the high end BS would be out.  Instead, it appears it may become the most corrupted area of music playback ever.  In just the last 5 years stunningly ridiculous products have been put forward on equally stunningly untrue ideas to solve completely non-issues.  Many of them are really crazy, and are already accepted by the mainstream computer based audiophile in the high end realm as known and certain issues.  Which of course can only be reduced by terribly expensive products.  Though the problems they reputedly solve are never quite solved.  Leaving another level of refinement and expense available for the next go around.


----------



## Mambosenior

Kudos to you esisude. Can I get a "Alleluia!"


----------



## Don Hills

castleofargh said:


> yup, that's why I said "and as long as the sample is big enough to get the right wave, should it matter?" having a few less "dots" on the wave wouldn't change the wave unless there was really too few reference values. but I don't really know how many is enough as music is not juste 1 sine wave.
> ...


 
   
So long as there are at least two "dots" for the highest frequency contained in the music, you have enough.

  
 (To be pedantic, that should be "*significant *frequency". Some instruments produce frequencies (harmonics) that can be higher than you can hear. Some people argue that you can hear their presence or absence and the sampling rate should be chosen to capture them all. Others argue that since you can't hear them, there is no need to capture them.)


----------



## castleofargh

thank you.
 so in the end back to the "if it can't be heard, some will hear it" situation. ^_^


----------



## ralphp@optonline

castleofargh said:


> thank you.
> so in the end back to the "if it can't be heard, some will hear it" situation. ^_^


 
 The ability to hear frequencies, sounds, distortions, etc. that the all humans have been conclusively proven not to be able to hear is directly proportional to the amount of money which can be made by claiming to hear such frequencies, sounds, distortions, etc.. So for example, if you one claims to be able the pico-second length jitter distortions present in non-asynchronous USB (which has been claimed) then as a solution to this very serious "jitter problem" one can sell asynchronous USB based DACs to replace all those horrible sounding and jitter filled non-asynchronous USB DACs.
  
 Perhaps the best example of the FUD principle (FUD = fear, uncertainty and doubt) is the Beatles catalog - first released as remastered standard resolution CDs (16bit/44.1kHz), then released as 24bit/44.1kHz "high resolution" files with the as yet to be released "super high resolution" (24bit/96kHz) files waiting in the wings.
  
 Seems to me the software side as the edge in the FUD war since they get the poor audiophile to buy everything over and over and over again.


----------



## castleofargh

don hills said:


> castleofargh said:
> 
> 
> > yup, that's why I said "and as long as the sample is big enough to get the right wave, should it matter?" having a few less "dots" on the wave wouldn't change the wave unless there was really too few reference values. but I don't really know how many is enough as music is not just 1 sine wave.
> ...


 

  I've looked at http://people.xiph.org/~xiphmont/demo/neil-young.html not getting half of it, that lead me to this http://en.wikipedia.org/wiki/Sampling_theorem and then in-between all the stuff I really don't get, I've seen that sentence "Nyquist's result that equi-spaced data, with two or more points per cycle of highest frequency, allows reconstruction of band-limited functions." and I remembered you telling me exactly that, but at the time it didn't compute for some reason.
  
 so if I'm not lost in space, the reason why we have 44khz (or 48khz) isn't random luck but "simply" to have at least 2 sample values of the highest audible frequency, for us 20khz. \o/ because yeah I'm a genius 2*20=40 . I'm self amazed by my math talent.
 for lower frequencies we obviously get more reference points as the wave is bigger in time, so it's of no concern at all, at least precision wise.
  
 and restricting the frequency range (cutting, filtering.. whatever) is just a condition to make that rule stay true?
 sorry for the ranting, but such a simple thing is making me understand tonnes of stuff I've seen and accepted, but without knowing why.
  
 so if I got it all right, increasing sample rate cannot add any precision unless we use a recording that has cues above 20khz, on a player that will not cut them out, for the benefit of friendly animals able to ear those sounds?


----------



## bigshot

Yes. The amount of samples used to reproduce the audible spectrum in regular CD quality sound is the same as the number used for the same spectrum in high sampling rates. The additional samples in high sampling rates just extend frequencies beyond the range of human hearing.


----------



## ralphp@optonline

castleofargh said:


> so if I got it all right, increasing sample rate cannot add any precision unless we use a recording that has cues above 20khz, on a player that will not cut them out, for the benefit of friendly animals able to ear those sounds?


 
 And the benefit of the many different people who profit by selling a high resolution versions of a recordings.


----------



## limpidglitch

castleofargh said:


> and restricting the frequency range (cutting, filtering.. whatever) is just a condition to make that rule stay true?


 

 The rule stays true no matter what. You cut the top frequencies to avoid aliasing.
 The maths here can seem more complicated than it really is, so put more simply:
  
 Imagine you write a computer program that records outside temperatures through the year, but by chance you neglect to allow for the possibility of negative temperatures.
 So when the temperature gradually creeps down below zero degrees, what does the program record? It might just get completely confused and return error messages, or it might faithfully record the data the only way it knows how. As far as the program knows negatives do not exist, so it just discards the minus signs and returns 3, 2, 1, 0, 1, 2,… mirroring the negative numbers back up as positive, making a right mess of your data.

 Much the same happens when you try to sample a signal that is more than half the sampling frequency. Using a sampling frequency of 48kHz, a 24kHz signal remains as 24kHz, but 26 becomes 22, 28 becomes 20 and 48 becomes 0. Everything above 24kHz gets folded symmetrically back down, just like the temperature recorder folded data below zero back up.

 In both cases you can mend things by either increasing the sample space, or if you want to keep things as they are, just decide that it's better to return nulls than non-sense, and filter out the impossible values before you feed them on.


----------



## Don Hills

castleofargh said:


> ... so if I'm not lost in space, the reason why we have 44khz (or 48khz) isn't random luck but "simply" to have at least 2 sample values of the highest audible frequency, for us 20khz. \o/ because yeah I'm a genius 2*20=40 . I'm self amazed by my math talent.  ... and restricting the frequency range (cutting, filtering.. whatever) is just a condition to make that rule stay true?
> ...so if I got it all right, increasing sample rate cannot add any precision unless we use a recording that has cues above 20khz, on a player that will not cut them out, for the benefit of friendly animals able to ear those sounds?


 
  
 As others have pointed out, yes, that's basically it.
 There are those that argue that the small gap between 20 KHz and 22.05 KHz is too small to apply an effective enough filter to make sure aliasing doesn't occur, and that the filter can cause audible side effects. 48 KHz is better, and of course even higher sampling rates allow using filters less likely to cause audible effects. 
 There are also those that argue that the frequencies above 20 KHz matter. I suggest doing the equivalent of sitting back with some popcorn and watch the fight.


----------



## limpidglitch

don hills said:


> As others have pointed out, yes, that's basically it.
> There are those that argue that the small gap between 20 KHz and 22.05 KHz is too small to apply an effective enough filter to make sure aliasing doesn't occur, and that the filter can cause audible side effects.


 
   
 It's often been repeated that a too steep cut-off will cause ringing, but I've never really understood why this is (though I suspect some parallels to ringing in band limited square waves). 
 Anyone have a simple explanation, maybe including how to decide on an optimal cut-off rate?
  
  
 Quote:


don hills said:


> There are also those that argue that the frequencies above 20 KHz matter. I suggest doing the equivalent of sitting back with some popcorn and watch the fight.





  
 There are also those that argue that super sonic frequencies will do more harm than good (oscillations, IM distortion and whatnot)
 I think I'll join you with that popcorn.


----------



## bigshot

don hills said:


> As others have pointed out, yes, that's basically it.
> There are those that argue that the small gap between 20 KHz and 22.05 KHz is too small to apply an effective enough filter to make sure aliasing doesn't occur, and that the filter can cause audible side effects. 48 KHz is better, and of course even higher sampling rates allow using filters less likely to cause audible effects.
> There are also those that argue that the frequencies above 20 KHz matter. I suggest doing the equivalent of sitting back with some popcorn and watch the fight.




Modern DACs upsample to apply the rolloff clleanly.

Controlled testing has shown that frequencies above the range of human hearing add nothing to music. In fact, above 14kHz, there isn't much of anything to hear.


----------



## Mambosenior

bigshot said:


> In fact, above 14kHz, there isn't much of anything to hear.




That makes me feel so kHz deprived. (Please, say it ain't so!)


----------



## Chesterfield

In response to the original set of questions, I can say, anecdotally, that I hear a very distinct difference between 16/44.1 and HDTracks titles that I convert to 24/48. Sometimes it's a small improvement, sometimes quite significant--but so far, with every title I've downloaded, there's been an improvement. I'm not going to speculate on the cause. But, considering I was, for a time, a professional classical musician and have a highly trained ear (thanks to well over a decade of private lessons, teaching, etc.) I think it's fair to rule out a placebo effect. I can't hear above 16.5 kHz (hell, why would I want to?), nor do I believe frequencies higher than that have an unconscious or subconscious effect (it'd be like saying pinching a paraplegic in the leg produces subconscious pain. It doesn't). For whatever reason, 24/48 always sounds better to me than 16/44.1. But, frankly, I think the difference--while often significant to me--would be non-existent to an untrained ear. This is not an ad for HDTracks: I think their customer service is consistently wretched--far, far worse than any other company I buy from--and I can't wait for competing companies to emerge.


----------



## esldude

chesterfield said:


> In response to the original set of questions, I can say, anecdotally, that I hear a very distinct difference between 16/44.1 and HDTracks titles that I convert to 24/48. Sometimes it's a small improvement, sometimes quite significant--but so far, with every title I've downloaded, there's been an improvement. I'm not going to speculate on the cause. But, considering I was, for a time, a professional classical musician and have a highly trained ear (thanks to well over a decade of private lessons, teaching, etc.) I think it's fair to rule out a placebo effect. I can't hear above 16.5 kHz (hell, why would I want to?), nor do I believe frequencies higher than that have an unconscious or subconscious effect (it'd be like saying pinching a paraplegic in the leg produces subconscious pain. It doesn't). For whatever reason, 24/48 always sounds better to me than 16/44.1. But, frankly, I think the difference--while often significant to me--would be non-existent to an untrained ear. This is not an ad for HDTracks: I think their customer service is consistently wretched--far, far worse than any other company I buy from--and I can't wait for competing companies to emerge.


 

 It is never fair to rule out placebo.  It is something that effects all humans.  Even with your background you would be susceptible. 
  
 Unless they are the same master (and they won't be from HDtracks), then they can sound different to varying degrees for reasons having nothing to do with sample rate or bit depth.  Convert the HDtracks to 44.1 khz, and back to original rate (96, 192khz whatever).  Then put the two files in Foobar and see if you can successfully ABX them.


----------



## Chesterfield

I'm well aware of the power of suggestion, and of the placebo effect. I have a Ph.D. in a field affiliated with psychology. I wasn't trying to suggest that I'm immune to the placebo effect, but I do resist having all my training and expertise in music so casually dismissed, which is why I mentioned it.

Yes, it's entirely possible that of the 40+ titles I've bought from HDTracks, every one uses a better master than their CD counterparts.


----------



## ab initio

esldude said:


> It is never fair to rule out placebo.  It is something that effects all humans.  Even with your background you would be susceptible.
> 
> Unless they are the same master (and they won't be from HDtracks), then they can sound different to varying degrees for reasons having nothing to do with sample rate or bit depth. * Convert the HDtracks to 44.1 khz, and back to original rate (96, 192khz whatever).  Then put the two files in Foobar and see if you can successfully ABX them. *


 
  
 This is a great test. @Chesterfield, if you do this test, will you be kind enough to snip out 30 second segments from the tracks you test and share them here so others can try it too? We can all compare results at the end! Then whoever did the best buys a everyone a Narragansett.
  
 Cheers


----------



## bigshot

The way to know for sure is to take one of your HD Tracks, bounce it down to 16/44.1 and back up to "HD" again, then do a blind test. Five bucks says if you do this, the difference will go away.


----------



## elmoe

bigshot said:


> The way to know for sure is to take one of your HD Tracks, bounce it down to 16/44.1 and back up to "HD" again, then do a blind test. Five bucks says if you do this, the difference will go away.


 
  
 Well obviously, if you do it that way the difference will go away. A 16/44.1 track "bounced up" to 24/96 is still essentially 16/44.1, there is absolutely no point in doing things this way.
  
 The only way to test this out is to take a genuine 24/96 (or 192) track and downsample it to 44/96 then compare both.
  
 If you take a FLAC file and convert it to 24kbps then transcode it to FLAC again, it's going to sound like the 24kbps file. If you compare that FLAC 24kbps transcode to the original FLAC, you will hear a clear difference...


----------



## Chesterfield

I'll be happy to do the test, once a methodology is resolved, provided it doesn't take an unreasonably long time. But I should say, ab initio, there's just no way you'd hear the difference on your equipment.




ab initio said:


> This is a great test. @Chesterfield, if you do this test, will you be kind enough to snip out 30 second segments from the tracks you test and share them here so others can try it too? We can all compare results at the end! Then whoever did the best buys a everyone a Narragansett.
> 
> Cheers


----------



## esldude

elmoe said:


> Well obviously, if you do it that way the difference will go away. A 16/44.1 track "bounced up" to 24/96 is still essentially 16/44.1, there is absolutely no point in doing things this way.
> 
> The only way to test this out is to take a genuine 24/96 (or 192) track and downsample it to 44/96 then compare both.
> 
> If you take a FLAC file and convert it to 24kbps then transcode it to FLAC again, it's going to sound like the 24kbps file. If you compare that FLAC 24kbps transcode to the original FLAC, you will hear a clear difference...


 

 Well actually no the difference would not go away.  The most straightforward approach is down convert to 16/44 and compare to the original file.  And bouncing down to 16/44 then back should be the same.  The main advantage to going back to the higher rate is for automated comparison.  When software like Foobar or hardware DACs switch from one rate to another there are often artifacts that would corrupt the test.  So having two files at the same sample rate prevent that.  If you have another way to switch without artifacts from switching then you can use two different sample rate files.


----------



## blackwolf1006

It depends on the source. I downloaded a CD loss-less 16/44 rip and a SACD 24/96 loss-less rip of the same album and I can hear the difference in the diffrence. you will notice this a lot more on Classical and Jazz. They're some instances on a SACD rip where I hear the artist breathing patterns on air instruments. The jump from 16 to 24 is Noticeable if you have the gear that can play the files correctly.
  
 Very new to this, so forgive my lack on knowledge.


----------



## riverlethe

blackwolf1006 said:


> It depends on the source. I downloaded a CD loss-less 16/44 rip and a SACD 24/96 loss-less rip of the same album and I can hear the difference in the diffrence. you will notice this a lot more on Classical and Jazz. They're some instances on a SACD rip where I hear the artist breathing patterns on air instruments. The jump from 16 to 24 is Noticeable if you have the gear that can play the files correctly.
> 
> Very new to this, so forgive my lack on knowledge.


 
  
 Are you sure the 16/44 version was from the same master or "re-master?"


----------



## blackwolf1006

riverlethe said:


> Are you sure the 16/44 version was from the same master or "re-master?"


 
  
 Same album and recording ripped from different source.
  
 http://www.amazon.com/Night-Sessions-Chris-Botti/dp/B00005RGNE


----------



## bigshot

esldude said:


> Well actually no the difference would not go away.  The most straightforward approach is down convert to 16/44 and compare to the original file.  And bouncing down to 16/44 then back should be the same.  The main advantage to going back to the higher rate is for automated comparison.  When software like Foobar or hardware DACs switch from one rate to another there are often artifacts that would corrupt the test.  So having two files at the same sample rate prevent that.  If you have another way to switch without artifacts from switching then you can use two different sample rate files.


 
  
 Exactly, and you also run the risk that your system plays high rates differently than redbook.


----------



## castleofargh

elmoe said:


> bigshot said:
> 
> 
> > The way to know for sure is to take one of your HD Tracks, bounce it down to 16/44.1 and back up to "HD" again, then do a blind test. Five bucks says if you do this, the difference will go away.
> ...


 
  taking the 16/44 back to 24/96 shouldn't bring much new problems and will sound like a 16/44. but the advantage is ease of use for the test, and also making sure that we're testing the file resolution, and not the way our system handles them. on some systems there are some very measurable differences depending on the resolution it's reading. if all it takes for that system to be great is to feed him 24/96, then reencoding 16/44 into 24/96 would bring that same benefit. and instead of showing the superiority of 24/96 as a resolution, it would have shown that this particular gear is bad at handling 16/44. another system could bring some bad IMD in the audible range with 24/96 and give the same false idea that 16/44 is superior. to test gears it's ok, to test file resolution it's better to try and bypass that possibility.
  
  


blackwolf1006 said:


> riverlethe said:
> 
> 
> > Are you sure the 16/44 version was from the same master or "re-master?"
> ...


 
 that helas isn't enough to know. just like the same album on vinyl will not have received the same treatment, chances are that the hires one is also remastered.
 just take your 24/96, reencode it in 16/44 and listen to both. they probably will still sound different.


----------



## sonitus mirus

bigshot said:


> Exactly, and you also run the risk that your system plays high rates differently than redbook.


 
  
 Playing the Devil's advocate here.  Does this mean that the difference in quality is not the same, or are you suggesting that subtle clues can be picked up by our brains to create bias where no legitimate difference exists?  I mean, if many systems realistically sound different when playing back at a higher rate, then for those systems there could be a real improvement.


----------



## riverlethe

I actually wonder if this debate matters at a time when you can buy a 4TB hard drive for $150. It might be better to buy high res versions just to show producers that customers are interested in better sound quality than what they've been offering...


----------



## Chesterfield

I don't have a pony in this race: frankly, I couldn't care less if 24/192 is proven to be better than lossless 16/44.1 (that's why, in my initial post, when I said my HDTracks purchases sound better than CD lossless counterparts, I added "for whatever reason", leaving open the possibility of better masters). Musical memory is a funny thing--for those of us who don't have perfect pitch, it's all too brief. By "musical memory," I don't, of course, just mean remembering melody or drum rhythm: I mean remembering pitch, dynamics, timbre, soundstage, etc. For most songs, unless they're remarkably minimalist, it's a mass of information all at once. Moving on--I think it's perfectly valid to administer these tests to Joe Schmo, so long as your goal is to determine whether such file differences matter to Joe Schmo. But if your goal is to guage whether or not 24/192 is better than 16/44.1, then you need a far more rigorous methodology than any I've seen: for me, you'd need, say, 50 professional musicians--preferably with perfect pitch. I'm familiar with all the arguments--some very well informed--that say anything over 16/44.1 is pointless. I understand the scientific claims--and yet, my ears tell me there's a difference. I've long believed that, for me, bit depth was key. So I decided to put that hypothesis to the test (of course, this is hardly conclusive--just anecdotal). For me, high res music invariably sounds more polished, more mellifluous (even 16/44.1, by comparison, sounds the tiniest bit grainy). It's subtle, but, for me, definitely there. For an informal test (I use a Mac, so can't use Foobar), I scaled an HDTracks 24/44.1 file back to 16/44.1 (Coldplay's "Ink"--my wife's unusual choice). My wife played the versions back to back, switching at my request, and I noted which track I thought was which. We were in separate rooms. I spent 30 mins before the test familiarizing myself with the song. She played the song in pairs, one version 16/44.1, the other 24/44.1, ten pairs total. I "guessed" the high res version 100% of the time. I'm not offering this as proof of any kind. Maybe my DAC manages 24 bit better than 16. Who knows. As I said, I'm not going to believe that 24/192 is definitively better than 16/44.1 until a test is devised with a methodology I can respect (I don't care what Joe Schmo hears). But until then, I'll trust my own ears.


----------



## Chesterfield

Nicely said.



riverlethe said:


> I actually wonder if this debate matters at a time when you can buy a 4TB hard drive for $150. It might be better to buy high res versions just to show producers that customers are interested in better sound quality than what they've been offering...


----------



## esldude

sonitus mirus said:


> Playing the Devil's advocate here.  Does this mean that the difference in quality is not the same, or are you suggesting that subtle clues can be picked up by our brains to create bias where no legitimate difference exists?  I mean, if many systems realistically sound different when playing back at a higher rate, then for those systems there could be a real improvement.


 

 Well depends on the difference.  I have a 192khz capable sound card in a desktop.  The 176 and 192 rates have extra noise.  As in  SNR of only 60 db.  And not ultrasonic noise, but noise spread evenly over the spectrum.  So with headphones that is audible, and different.  Not better. 
  
 I think bigshot was referring to such or that some DACs have a different roll off rate near the upper octave at different sample rates.  Either way a sample rate could cause a genuine difference whether an improvement or not.  Now in good properly operating gear it shouldn't be a problem, and in the great majority of gear not faced with extreme loads it is the case.


----------



## bigshot

riverlethe said:


> I actually wonder if this debate matters at a time when you can buy a 4TB hard drive for $150. It might be better to buy high res versions just to show producers that customers are interested in better sound quality than what they've been offering...


 
  
 LPs sounded pretty good. Then CDs were introduced and LP sound quality took a nose dive. For a while CDs sounded good. Then they introduced SACD and CDs started getting hot mastered. Now they are introducing HD tracks and blu-ray audio which supposedly has even bigger numbers (and copy protection). How many copies of Dark Side of the Moon can the record companies convince you to buy?
  
 Filesizes do matter. We are heading towards a world that replaces physical objects with streaming, and home stereo components with portable DAPs and phones. How many different file sizes for music that sounds the same do you really want? I only want one... the smallest size that meets and exceeds my ability to hear. That is AAC 256 VBR. I don't really have any use for CDs any more. It all gets ripped and plopped on a media server to stream to me.


----------



## riverlethe

bigshot said:


> LPs sounded pretty good. Then CDs were introduced and LP sound quality took a nose dive. For a while CDs sounded good. Then they introduced SACD and CDs started getting hot mastered. Now they are introducing HD tracks and blu-ray audio which supposedly has even bigger numbers (and copy protection). How many copies of Dark Side of the Moon can the record companies convince you to buy?
> 
> Filesizes do matter. We are heading towards a world that replaces physical objects with streaming, and home stereo components with portable DAPs and phones. How many different file sizes for music that sounds the same do you really want? I only want one... the smallest size that meets and exceeds my ability to hear. That is AAC 256 VBR. I don't really have any use for CDs any more. It all gets ripped and plopped on a media server to stream to me.




24/96kHz lossless is what, 3 megabits per second? I'm paying $40/month for 50 Mbps. It's trivial.


----------



## castleofargh

-because the hires version usually costs more.
  
 -because when I try an ABx I do not hear a difference.
  
 -because I don't know if my system will stay stable with 96khz samples. there are some choices that are done by the guys creating dacs and amps. having to control a wider frequency range does come at a price in electronic(and I'm not just talking money, I'm talking conceding something to gain something else). some manufacturers already have a hard time keeping a flat clean response from 20hz to 20khz, imagine when they have to stay flat and clean from 20hz to 30 or 40khz. that's double the frequency range, not a small feat. and if you care for those stuff. do you ask the manufacturer where the low pass filter is? at 20khz bringing not much the the actual sound(probably the best choice) or at 30khz? requiring to keep a good response up to that level?  and impedance and jitter will start having a real effect at those frequencies. even a cable might start to have a real impact. I think you take the problem of hires for something a little too simple and optimistic.
 just like DSD is in fact impractical and most manufacturers go with multi bit dacs and cut the high frequencies just like a pcm file. making something that is not really better in any way.
 and that for drivers that mostly won't be able to output those frequencies, into ears that don't register them, what a waste. we just end up praying that no IMD will jump back into audible range so that at least we don't make the music worst than with 16/44.
  sure it can go the other way around too and actually give a better signal with hires files, but as long as the doubt is here, I'd rather use smaller files that cost less.
  
 - I use mp3 on my daps, so if I chose size over quality when it's a proved fact, you can guess why I don't care much for hypothetical improvements with hires.
  
 here you have my own reasons. others might have other reasons for other choices. but if you want the last "bits" of real sound, just get a headphone or speaker with less distortion and less roll off. that's where the improvement can be of a meaningful magnitude.


----------



## bigshot

riverlethe said:


> 24/96kHz lossless is what, 3 megabits per second? I'm paying $40/month for 50 Mbps. It's trivial.


 
  
 Try to get that on your phone on the road with Starbucks wifi or 3g!


----------



## elmoe

bigshot said:


> Try to get that on your phone on the road with Starbucks wifi or 3g!


 
  
 4G is 10 times faster than 3G at the minimum. Not that its a problem because nowadays smartphones can handle USB OTG allowing you to hook up large USB keys to your phone as removable media, essentially letting you carry your music (even large filesizes) in your pocket.
  
 Ah wait you're still using Apple products. Nevermind then.


----------



## haloxt

http://en.wikipedia.org/wiki/Sound_from_ultrasound
  
 Inaudible sounds still have to be produced, causing distortion in equipment like instruments, amps, and speakers, and when the sound waves are created and bounce around will cause interference with each other. Humans may not be able to hear inaudible frequencies directly, but it should be possible for people to at least unconsciously hear the resultant interference caused by certain natural inaudible sounds. But given professional violinists' inability to consciously differentiate between super expensive Stradivarius violins and other violins, I don't expect to see anytime soon people prove that people can consciously be aware of natural inaudible frequencies causing sound wave interference.


----------



## stv014

haloxt said:


> http://en.wikipedia.org/wiki/Sound_from_ultrasound


 
  
 Good thing people usually do not listen to "music" with ultrasonic content at 100-110 dB SPL. And even when such level of ultrasound is actually present, the intermodulation products from the non-linearity of air could be captured by the microphone used for recording, since they are already in the audio band.


----------



## bigshot

There was a test at the AES that concluded that at high volume, people can perceive ultrasonic frequencies as sound pressure, but the presence or absence of ultrasonic content made absolutely no difference to the perception of sound quality in music.


----------



## Don Hills

stv014 said:


> Good thing people usually do not listen to "music" with ultrasonic content at 100-110 dB SPL. And even when such level of ultrasound is actually present, the intermodulation products from the non-linearity of air could be captured by the microphone used for recording, since they are already in the audio band.


 

 There's been some work on reproducing sound "in thin air", by focusing beams of ultrasonic sound to cross at the desired point. The non-linearity of the air mixes the two signals and makes their difference frequencies audible. It's not very practical because of the high ultrasonic SPL needed to reach useful levels of air non-linearity.


----------



## hogger129

I really can't tell the difference in anything above CD quality.


----------



## bigshot

Neither can other humans.


----------



## blades

bigshot said:


> Neither can other humans.


 
  
 In fact you can remove up to 80% of the data from a red book CD and still have no audible change.


----------



## dikkiedirk

To answer the question of the TS:
  
 Because higher numbers, either bits or kHz or $$ mean better sound, IT HAS TO!!
  
 Better listen to great MUSIC then to a great system. Choice of great music is strictly personal.


----------



## blades

dikkiedirk said:


> To answer the question of the TS:
> 
> Because higher numbers, either bits or kHz or $$ mean better sound, IT HAS TO!!
> 
> Better listen to great MUSIC then to a great system. Choice of great music is strictly personal.


 
  
 Why does it have to?


----------



## dikkiedirk

blades said:


> Why does it have to?


 
 Aren't bigger nummers always better? Pun intended.


----------



## blades

I guess that settles it.


----------



## sandab

The main difference is in the frequency cutoff for the analog low-pass filter on the DAC's output.  The more gradual an analog filter can cut off, the less impact it will have on phase.  So higher output rate and a digital low-pass will sound better than relying on a very steep (brick) analog filter.  Of course, you can do this in the DAC, assuming it has a DSP.  This will add cost and complexity though since the DSP itself is a source of HF ground noise so will need its own separate power source.  Or you can do it in the transport or computer and feed a higher rate to the DAC over optical (which makes for nice isolation) or USB (which may potentially be noisier, but not as bad as a DSP).  Or put it in the source material itself.


----------



## SilverEars

Interesting, didn't expect this to come off in a sampling rate thread.    
  
 I have read that Wolfson has low pass built in, possibly analog?  Actually, the WM8741 differentiates fomr WM8740 in that the filter can be switched via software.   ESS implementation I've seen is with low pass outside the DAC and people have said the LPF is not implemented in the chip like some.  
  
  

 http://www.mezzohifi.com/private_folder/608e348712076d73f11a929240fa3d78.jpg


----------



## castleofargh

sandab said:


> The main difference is in the frequency cutoff for the analog low-pass filter on the DAC's output.  The more gradual an analog filter can cut off, the less impact it will have on phase.  So higher output rate and a digital low-pass will sound better than relying on a very steep (brick) analog filter.


 
 I think that is exactly why NOS DACs have(should have) disappeared. most DACs already use the best sample rate at the best moment, whatever the original sample rate of that track is(prononce dothraki). so I can hardly see that as a reason to use hirez.
  
 people who truly believe 20 or 25khz are part of the music should go for hirez(or learn a little more about how human hearing works). if we admit hearing those, it does make sense to go hirez for several technical reasons if you're positive you headphone/speakers can actually deal with those frequencies.
 now if you are a human being older than 15, chances are hirez is a waste of space and money for you.
  
 all I know is that if 22 or 30khz were necessary to hear the "real" music, that would mean I'm ****ed, because at my age I can't hear above 17khz.
 the fact that I still enjoy music very much is to me a clear proof that I don't mind missing 30khz sounds and in fact I would live very well with music cut off at 14khz. my headphones showed me that long ago. but everybody's free to believe what he wants and buy what he likes. the human race is not in danger from more hirez files, so who cares?


----------



## Mambosenior

In the hi-rez version of "Let It Be," the message "Yoko broke up the Beatles" can be heard continuously at around 23khz. Playing the album through my Audio-gd DAC, the message is heard in Chinese.

And I was ready to ditch hi-rez!


----------



## esldude

mambosenior said:


> In the hi-rez version of "Let It Be," the message "Yoko broke up the Beatles" can be heard continuously at around 23khz. Playing the album through my Audio-gd DAC, the message is heard in Chinese.
> 
> And I was ready to ditch hi-rez!


 

 I don't see why it is so hard to see how hirez has more info to work with.  All anyone has to do is play back hirez files at half the sample rate to hear a difference.  Depending on your hearing you gain an additional 12-20 khz of bandwidth that way.


----------



## nick_charles

esldude said:


> I don't see why it is so hard to see how hirez has more info to work with.  All anyone has to do is play back hirez files at half the sample rate to hear a difference.  Depending on your hearing you gain an additional 12-20 khz of bandwidth that way.


 
  
 The issue is not whether high res has more data but whether the difference is audible.
  
 To date when the audible difference between high res and non high res (from the same source)  has been tested with material with ultra high frequencies that has been nobbled by downsampling or low pass filters the difference is not normally audible. This has been supported by several published papers going back to the late 70s including from professional broadcasting bodies and has not been contradicted by *any* of the industry proponents of high res with carefully controlled listening tests. Yes there are a bunch of anecdotes out there but at present there is only one paper which _might_ possibly support audibility of high sampling rates (Pras and Gustavino) but it has somewhat questionable stats and method and a small sample, the largest scale test the Meyer and Moran study found nobody who could reliably detect the presence of a secondary A/D/A loop inserted after a high res player output.
  
 Humble red book competently captures above 20K and even for those lucky few who can hear above 20K anything above 20K filtered out from musical content is not missed


----------



## esldude

nick_charles said:


> The issue is not whether high res has more data but whether the difference is audible.
> 
> To date when the audible difference between high res and non high res (from the same source)  has been tested with material with ultra high frequencies that has been nobbled by downsampling or low pass filters the difference is not normally audible. This has been supported by several published papers going back to the late 70s including from professional broadcasting bodies and has not been contradicted by *any* of the industry proponents of high res with carefully controlled listening tests. Yes there are a bunch of anecdotes out there but at present there is only one paper which _might_ possibly support audibility of high sampling rates (Pras and Gustavino) but it has somewhat questionable stats and method and a small sample, the largest scale test the Meyer and Moran study found nobody who could reliably detect the presence of a secondary A/D/A loop inserted after a high res player output.
> 
> Humble red book competently captures above 20K and even for those lucky few who can hear above 20K anything above 20K filtered out from musical content is not missed


 

 I knew I should have used the facetious smiley face. 
  
 You misunderstood me.  The last sentence should have been the tipoff.
  
 By playing back at half the sample rate I didn't mean resample say 96khz to 48 khz.. I meant taking the 96 khz data and playing it back at a 48 khz rate.  Which cuts all frequencies by half.  So 40 khz content becomes 20 khz.  Then ultrasonics become truly audible.  Everything else about it is messed up.  But the crazy claims about ultrasonic content can at least be heard in a fashion.  Then compared to 48 khz played back at 24 khz you really would hear a "hirez" difference.


----------



## nick_charles

esldude said:


> I knew I should have used the facetious smiley face.
> 
> You misunderstood me.  The last sentence should have been the tipoff.
> 
> By playing back at half the sample rate I didn't mean resample say 96khz to 48 khz.. I meant taking the 96 khz data and playing it back at a 48 khz rate.  Which cuts all frequencies by half.  So 40 khz content becomes 20 khz.  Then ultrasonics become truly audible.  Everything else about it is messed up.  But the crazy claims about ultrasonic content can at least be heard in a fashion.  Then compared to 48 khz played back at 24 khz you really would hear a "hirez" difference.


 
  
 Sorry, I'm obviously not paying enough attention !


----------



## bigshot

Comedy is serious business. It should be left to professionals... like me!


----------



## ab initio

esldude said:


> I knew I should have used the facetious smiley face.
> 
> You misunderstood me.  The last sentence should have been the tipoff.
> 
> By playing back at half the sample rate I didn't mean resample say 96khz to 48 khz.. I meant taking the 96 khz data and playing it back at a 48 khz rate.  Which cuts all frequencies by half.  So 40 khz content becomes 20 khz.  Then ultrasonics become truly audible.  Everything else about it is messed up.  But the crazy claims about ultrasonic content can at least be heard in a fashion.  Then compared to 48 khz played back at 24 khz you really would hear a "hirez" difference.


 
  
 I thought the original statement was so good that I couldn't think of a reply worthy enough to follow it!
  
 Cheers


----------



## BirdManOfCT

ralphp@optonline said:


> And the high resolution one may actually contain LESS information than the CD resolution one, in that many, many high resolution downloads, especially those purchased from HDTracks, do not come with full information booklets containing information such as recording data (time and place of the recording, the equipment used to make the recording, producer, recording engineer, mastering engineer) and musicains, etc. - i.e. LESS information.
> 
> But hey it costs more and the high end audio press just love high resolution digital audio so high resolution just has to be BETTER


 

 Some of the re-masterings are welcome. Provided it's not tweaking to make it into a different animal. Some of the songs before re-mastering have cymbals that sound like glass shattering, while the re-mastered version sounds much more like cymbals. If someone can't tell the difference, even with cheap gear, they should give up listening critically and ONLY enjoy the music. I like to do both.


----------



## BirdManOfCT

chesterfield said:


> I don't have a pony in this race: frankly, I couldn't care less if 24/192 is proven to be better than lossless 16/44.1 (that's why, in my initial post, when I said my HDTracks purchases sound better than CD lossless counterparts, I added "for whatever reason", leaving open the possibility of better masters). Musical memory is a funny thing--for those of us who don't have perfect pitch, it's all too brief. By "musical memory," I don't, of course, just mean remembering melody or drum rhythm: I mean remembering pitch, dynamics, timbre, soundstage, etc. For most songs, unless they're remarkably minimalist, it's a mass of information all at once. Moving on--I think it's perfectly valid to administer these tests to Joe Schmo, so long as your goal is to determine whether such file differences matter to Joe Schmo. But if your goal is to guage whether or not 24/192 is better than 16/44.1, then you need a far more rigorous methodology than any I've seen: for me, you'd need, say, 50 professional musicians--preferably with perfect pitch. I'm familiar with all the arguments--some very well informed--that say anything over 16/44.1 is pointless. I understand the scientific claims--and yet, my ears tell me there's a difference. I've long believed that, for me, bit depth was key. So I decided to put that hypothesis to the test (of course, this is hardly conclusive--just anecdotal). For me, high res music invariably sounds more polished, more mellifluous (even 16/44.1, by comparison, sounds the tiniest bit grainy). It's subtle, but, for me, definitely there. For an informal test (I use a Mac, so can't use Foobar), I scaled an HDTracks 24/44.1 file back to 16/44.1 (Coldplay's "Ink"--my wife's unusual choice). My wife played the versions back to back, switching at my request, and I noted which track I thought was which. We were in separate rooms. I spent 30 mins before the test familiarizing myself with the song. She played the song in pairs, one version 16/44.1, the other 24/44.1, ten pairs total. I "guessed" the high res version 100% of the time. I'm not offering this as proof of any kind. Maybe my DAC manages 24 bit better than 16. Who knows. As I said, I'm not going to believe that 24/192 is definitively better than 16/44.1 until a test is devised with a methodology I can respect (I don't care what Joe Schmo hears). But until then, I'll trust my own ears.


 

 Good stuff. I'm aware of placebo effect, too, but it's amazing how over time the there-is-no-difference starts giving ground when things improve. What was "impossible" 10 or 20 years ago is now "common knowledge" since the science caught up and "fixed" it.


----------



## tarnumf

Not sure if this was mentioned in this topic before.. But theoretically, couldn't frequencies in inaudible range still distort audible frequencies through interference effect, producing beats?


----------



## Head Injury

tarnumf said:


> Not sure if this was mentioned in this topic before.. But theoretically, couldn't frequencies in inaudible range still distort audible frequencies through interference effect, producing beats?


 
 It's called intermodulation distortion (IMD) and yes it can. However, if it's present from how the recording was made (i.e. what we'd theoretically want to produce being all into hi-fi and crap) then the parts that distort the audible range will still be present with a 44.1 kHz sampling rate. Anything higher sampling rates and frequencies can _add_ will be unwanted distortion not present on the recording, just flaws in the playback chain. This is why higher sampling rates can actually be harmful, as opposed to the standard.


----------



## Morphious

While doing research on Lossless,  I downloaded 2 dozen albums and MQS files from LINN and others. I also have a program called Similarity App. In this application you can run an Analysis of a files to see lots of information about the audio track. Most of the files barely had any peaks in the frequency Khz range that reached above 88Khz.  I did a little checking and found that ultrasonically, Not many instruments can go over 100Khz no matter what, Symbols can hit up to 102Khz.   My style of music had a high peaks of 55Khz on a 2014 master hi rez file. Everything above this if encoded into a 192 Khz files is useless and empty space ( I.E. a 0 in digital language) Most space above 100Khz is set to 0. This means it is just wasted space that cannot affect the soundstage, or anything else in the audio spectrum as there is nothing there to play or affect anything as it is a 0 ( or row of zeros )   But zeros do take up physical space in a file, and makes the file larger. This would lead me to believe that 96Khz would be near the best any file can be heard ( if we could hear that high), Maybe with clipping on an a properly recorded symbol, but even the reference material out there are not recorded to this level.   
  
 Bit depth is a different story. the more / wider the bit depth the more it affects the whole spectrum of sound. It's more 1's and 0's that be crammed into the full frequency spectrum as the file is being played, and this has to have an effect on the sound in every way.   Similar to color bit depth, the more bit depth you have, the more colors you have. Even if you cannot see all of the colors above 24bit, it does affect the image clarity and depth. 
  
 Being Hi-Fi type people, we all want to listen to the best quality audio we can, even if we cannot hear it  
  
 Someone please correct me if I have went off on a misinformed tangent...


----------



## Head Injury

morphious said:


> Someone please correct me if I have went off on a misinformed tangent...


 
 To keep it simple:
  
 Anything above what we can hear (or anything above 20 kHz for virtually all humans) won't be audible and won't affect the audible frequencies except in unwanted ways like IMD, so it's all wasted space. We can't hear what we can't hear, seems pretty straightforward right?
  
 Bit depth only determines maximum signal-to-noise ratio (6 dB per bit) of the file, and has no other effect on sound quality. It doesn't improve image clarity or depth or other meaningless terms. Just how loud things can get above the noise floor. Your color depth analogy is flawed because color and sound are both stored in very different ways; color bit depth determines the number of possible colors a pixel can display while audio bit depth determines the number of volume steps (but not the _size_ of the steps). Color depth is actually more analogous to the sampling rate in audio, but still not quite the same so don't go claiming it means higher sampling rate improves the sound.
  
 16 bits sampled at 44.1 kHz can perfectly capture all sounds up to 96 dB above the noise floor within the range of human hearing. The only thing increasing the depth to 24 bits does is allow the perfect capture of all sounds up to 144 dB above the noise floor. There's no increase in quality up to 96 dB, just an increase in range above it.


----------



## SilverEars

head injury said:


> Your color depth analogy is flawed because color and sound are both stored in very different ways; color bit depth determines the number of possible colors a pixel can display while audio bit depth determines the number of volume steps (but not the _size_ of the steps). Color depth is actually more analogous to the sampling rate in audio, but still not quite the same so don't go claiming it means higher sampling rate improves the sound.


 
 I agree that they totally different.  People that fall for HD audio marketing are likely thinking it's the same as panel resolution and bit depth which are different.


----------



## sandab

silverears said:


> I agree that they totally different.  People that fall for HD audio marketing are likely thinking it's the same as panel resolution and bit depth which are different.


 
 No, 88x16 has _exactly_ the same entropy limit (maximum information content) as 44x32.  There is absolutely no difference.  People who tell you otherwise are clueless.


----------



## SilverEars

sandab said:


> No, 88x16 has _exactly_ the same entropy limit (maximum information content) as 44x32.  There is absolutely no difference.  People who tell you otherwise are clueless.


 
 Can you explain this further?  I'm not familiar with entropy limit, so I don't understand what you are referring to.


----------



## castleofargh

morphious said:


> While doing research on Lossless,  I downloaded 2 dozen albums and MQS files from LINN and others. I also have a program called Similarity App. In this application you can run an Analysis of a files to see lots of information about the audio track. Most of the files barely had any peaks in the frequency Khz range that reached above 88Khz.  I did a little checking and found that ultrasonically, Not many instruments can go over 100Khz no matter what, Symbols can hit up to 102Khz.   My style of music had a high peaks of 55Khz on a 2014 master hi rez file. Everything above this if encoded into a 192 Khz files is useless and empty space ( I.E. a 0 in digital language) Most space above 100Khz is set to 0. This means it is just wasted space that cannot affect the soundstage, or anything else in the audio spectrum as there is nothing there to play or affect anything as it is a 0 ( or row of zeros )   But zeros do take up physical space in a file, and makes the file larger. This would lead me to believe that 96Khz would be near the best any file can be heard ( if we could hear that high), Maybe with clipping on an a properly recorded symbol, but even the reference material out there are not recorded to this level.
> 
> Bit depth is a different story. the more / wider the bit depth the more it affects the whole spectrum of sound. It's more 1's and 0's that be crammed into the full frequency spectrum as the file is being played, and this has to have an effect on the sound in every way.   Similar to color bit depth, the more bit depth you have, the more colors you have. Even if you cannot see all of the colors above 24bit, it does affect the image clarity and depth.
> 
> ...


 


  ultrasonics... ^_^
 they could replay my music with the star spangled banner sung by the chipmunk from 40khz to 60khz it wouldn't change a thing for me. make yourself a favor, go to your audiologist and ask him to test you, to see where you actually hear something. it should clear things up like A LOT.
  
  
 now bits wouldn't work with an analogy for colors, but it would for brightness in a way. every time you add a bit you can lower the brightness a little more. the music close to 0DB(loudest) would be a spot in your face changing intensity. and adding bit depth to 16bit would be like adding a little spot next to it that would do brightness dimmer than the minimum the first spot ever does. what are the odds for you to ever see the second spot? not much.
 and to that you have to add the fact that the dimmer level for the first spot was already lower than what you can normally see. meaning that the second spot will have variations between "is it off?" and "I think it's off".
 that's what we do when we go from 16bit to 24bit. we don't add inter values for the first spot, that's some misunderstood marketing stuff. all the sound between 0db and -96db will stay the same on 16 and 24bit(given we actually have 16bit).
 personally I like to output 24bit(but don't use 24bit tracks) so that I can set my volume with the computer. but that's a gadget, it's not really for music and I could move my butt and go change the volume on the amp.


----------



## sandab

silverears said:


> Can you explain this further?  I'm not familiar with entropy limit, so I don't understand what you are referring to.


 
 Entropy is the amount of information.  The limit is the upper bound on the information content.  Shannon's Sampling Theorem underlies much of modern information theory.  When the entropy has reached its limit we call it a random sequence.


----------



## sandab

I took a look at the magnitude of the quantization error over 25 cycles.
  
 Here's 44.1/16 with a 0dB 20k sinusoidal:
  

 Top is the samples (just over 2.2 per cycle) in blue.  Overlaid in green is the numerical error, scaled so 0 to 1.0 is 0.1%.  The bottom in red is the real part of a cooley-tukey DFFT (approximation of frequency spectrum).  Phase is ignored here.
  
 Here's the same in 44.1/24:

 For comparison, here's 44.1/16 at -12dB (2^(-12/3) = 1/16), which is equivalent to 44.1/12:

 44.1/24 at -12dB:
  

 A good spectrum should spike to 1.0, like 44.1/16 at 1kHz 0 dB:
  

 By comparison, here is 96/24 at -12dB, 20kHz:
  

 Hard to say how precise the frequency spectrum is for 44.1, but needless to say there's quite a bit of distortion going on.  That energy is going to go someplace else on the spectrum, and with 44.1 it's almost guaranteed to be "someplace audible".
  
 Even 48/16 looks better:


----------



## sandab

I also added an FFT of the error, which shows an even spread across the spectrum:

 The pink line is the error (QE) spectrum... clearly it's across the board.  Its Y axis is 0.1%, so it's about 0.01% evenly distributed across the spectrum.
  
 At -12dB:
  

 So around 0.1% at -12dB.  By comparison 44.1/24 @ -12dB doesn't rise from the X axis:


----------



## castleofargh

and what about a more usefull frequency? like 14 or 15khz, or just something in the medium, where it matters. because I have no record with loud 20khz signal, and in fact no headphone that doesn't roll off long before that. given the improvement from 44.1 to 48khz, I would suspect the differences become marginal very fast at lower frequencies.
  
 but thanks for taking the time to do all this.


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## esldude

And of course there is the change adding dither would make.


----------



## sandab

Dither would just randomize the noise - flatten the spectrum.  But it's already pretty flat, so I'd have to question the usefulness of dither except to flatten the error spectrum in conjunction with resampling or applying gain, or some other numeric operation, to reduce rounding error bias.


----------



## RRod

But our ears would have to be able the hear the lower-frequency content, which is several dB below the 20kHz tone, in a time-span of 0.00125 seconds.


----------



## sandab

castleofargh said:


> and what about a more usefull frequency? like 14 or 15khz, or just something in the medium, where it matters. because I have no record with loud 20khz signal, and in fact no headphone that doesn't roll off long before that. given the improvement from 44.1 to 48khz, I would suspect the differences become marginal very fast at lower frequencies.
> 
> but thanks for taking the time to do all this.


 
 I think that depends on the recording and exactly how the highs were rolled off during the CD-rate cut of the master.  If you put a mic to an instrument, you'll find it has supersonic content; so at some point that needs to be filtered.  Filters don't have perfect cutoffs but a wide attenuation ramp - and the wider, the better they tend to sound due to reduced ringing and phase distortion.  Except for digital sinc-based pass/stop filters, which have flat phase response, meaning the phase has no complex component - so represents a simple propagation delay.  But they still ring; that's inevitable when parts of the band is removed from transients.  I think you'll find most if not all 44.1 content has some signal at 20k.
  
 Since the error spectrum is so flat, it just tells us that treble content reduces the noise floor, potentially beating the noise floor for any repeating waveform.
  
 And since you mentioned it ...  15.5kHz @ 44.1/16 and 12 bits (0dB and -12dB)
  

  

 Edit: the second error spectrum might benefit from dithering, as it has an odd low-frequency bias.  Not sure if that's simply an artifact of the limited signal window (FFT assumes an infinite signal).  I'll see what adding a dither instead of mathematically correct rounding might do to the error for this.  It could also be a beat in which case nothing can be done about it other than redithering to add enough additional white noise to swamp it.


----------



## doraymon

Interesting article here.


----------



## BlackstoneJD

I am a pretty hardcore audiophile and tend to agree that the format itself is not as important as I thought. This is evidenced by the fact that a lot of the material that is being released on PONO and HDTracks is just better than what was released on CD. The differences between the Crosby Stills and Nash CDs and the 192KHz files is significant in my opinion, but they are also so different that it is not an apples to apples comparison. But almost without exception, the material I get from HD Tracks is superior to the prior releases, is less fatiguing to the ear and has a certain "quietness" or blackness between the notes. It is just more musical and effortless sounding. You do not necessarily loose that, however, in my experience, by playing those same files back at 44.1KHz. In fact much of the improvement is still there.
  
 What I suspect is actually happening is that the content itself is better which makes up the lion share of the perceived improvement. But it does not really matter at the end of the day, because the product that is being put out is consistently superior, in my experience, than the CD releases that preceded them.
  
 In some cases, music I like that I never considered "audiophile grade" in the sense that it sounded harsh or unpleasant when played loud now makes the cut. Led Zeppelin is one of the best examples of this. Some very old jazz albums that were thin and crackly now have real weight and body. For example, Joe Henderson's "Page One" was a terrible CD. It is an excellent high resolution download. It is also my favorite jazz record of all time.
  
 Generally speaking, when I play something from my CD library, I am reaching for the volume control to turn it down. When I play something from my high resolution library, I am reaching to turn it way up.
  
  
 Frankly, overall I am astonished at how good some of the downloads are. It could be that new techniques are being used in creating them. Regardless, I think it is worth the price of admission to subsidize what is clearly an improvement in many old, historic recordings.


----------



## MatsP

kaffeemann said:


> Both do sound the same. The additional information 24 bit/192 kHz files can store is inaudible.
> See here.


 

 That's complete rubbish. A child would be able to understand that using more bits/higher frequency makes for a more accurately reproduced waveform, particularly in lower frequencies, and at lower levels.


----------



## castleofargh

matsp said:


> kaffeemann said:
> 
> 
> > Both do sound the same. The additional information 24 bit/192 kHz files can store is inaudible.
> ...


 

 a child maybe, but someone who understands digital audio would find what you say even stranger than what you quoted.
 because 1: https://en.wikipedia.org/wiki/Nyquist%E2%80%93Shannon_sampling_theorem
 and 2: the delicate part to reconstruct perfectly is in fact the upper limit of the frequency range recorded, not the low freqs.
  
 I have no idea what low levels are supposed to change. you have enough dynamic range to record a signal and avoid hearing noise, or you don't.


----------



## gregorio

blackstonejd said:


> Frankly, overall I am astonished at how good some of the downloads are. It could be that new techniques are being used in creating them. Regardless, I think it is worth the price of admission to subsidize what is clearly an improvement in many old, historic recordings.


 
  
 Your suspicions are correct, it's not the format, it's the content itself. And, it's not due to "new techniques" either, it's essentially due to techniques which date back to the 1940's. In other words, whether a song is released in 24/192 or 16/44.1 should make absolutely no audible difference whatsoever, the fact that there often is a difference is because you are listening to two different songs or more precisely, to two different versions of the same song which have been specifically designed to sound different. You could take that 24/192 version and providing it's converted properly to 16/44.1, you would be unable to hear a difference between them.
  
 There is no technical or fidelity reason why HDTracks (or anyone else) couldn't still release two different versions, say an "audiophile grade" version and a "standard" version both in 16/44.1. The reason they don't, is marketing. It's trivially easy to convince people who know little or nothing about how digital audio works that bigger data numbers (a higher sample rate and more bits) means higher resolution/fidelity and then you've got the relatively easy marketing task of justifying a far higher price for an "audiophile grade" version because it's physically different. It would be far more difficult, maybe bordering on the impossible, to justify a far higher price between two different versions which are physically the same (both 16/44.1). This marketing strategy is obviously very effective for some/many consumers. MatsP for example has obviously swallowed it all, hook, line and sinker, even to the point of effectively saying that those who don't believe this marketing BS has a level of understanding lower than a child. But then, as with all the very best scams, they are so effective that the victims continue to believe the BS and never realise they've been scammed.
  
 None of this really helps you in many cases though because often, if you want an "audiophile grade" version you don't have a choice, as they only sell that version in one of the "bigger data numbers" formats, not in 16/44.1. While you could just convert it to 16/44.1, with no loss of fidelity, you've still obviously got to pay for that "bigger data number" format in the first place. I'm not sure if this is ever going to change, as long as bigger data numbers can be converted (via marketing) into higher prices consumers can be charged/will pay, the data numbers will continue to get bigger, regardless of the actual fact that there are no resolution/fidelity differences. The only way this knowledge can help you in practice, is in knowing not to automatically dismiss 16/44.1 recordings just because they are 16/44.1, especially if there isn't a bigger data number version it's competing against.
  
 G


----------



## axle_69

matsp said:


> That's complete rubbish. A child would be able to understand that using more bits/higher frequency makes for a more accurately reproduced waveform, particularly in lower frequencies, and at lower levels.


 

 Not nice... Actually you don't need 24 bits, if you want to be picky 20 bits is enough dynamic range for music, and 16 bits is better than vinyl. 88.2 or 96 kHz is useful to take the band pass filter during DAC away from our hearing limit because filters are not perfect, but most of us don't hear above 16-17 kHz and have some loss above 10 kHz, so having a filter at 22.05 kHz (for 44.1 kHz) isn't that bad. 192 kHz? Does your system play 96 kHz signals? They may be present in the original sound, some instruments vibrate that high, but it may be close to the limit of what the amplifier can handle.


----------



## Baxide

matsp said:


> That's complete rubbish. A child would be able to understand that using more bits/higher frequency makes for a more accurately reproduced waveform, particularly in lower frequencies, and at lower levels.


 

 Unfortunately a grown person can't tell the difference between an mp3 file and a 24/192 file in most cases.
 The high bit rate and high sampling rate files come into their own when you have a wide audio spectrum and wide dynamic range file. But if you are listening to an audio track that has been subject to the audio loudness war then the mp3 file is as good as identical in the listening test.


----------



## jcx

actually I believe some have pointed to Loudness War dynamic level compression equivalent to clipping does add harmonics and IMD at high levels that can challenge mp3 codecs - perceptual lossy data compression codec algorithms often perform better on more "musical" source
  
 if you are indirectly referencing mp3 18 kHz low pass filtering then even few youngsters are liable to notice or prefer music with an extra 2 kHz extension to 20kHz


----------



## Baxide

The problem I have with most of this is the fact that the word better is heavily misused in its intended understanding. Let's look at what we really mean when we are talking about being better. You'll first need a DAC that can decode the two types of audio files into an analogue audio sound where a difference can be detected in the first place. I have heard a couple of lovely sounding DACs that made mp3 sound amazing. They were mostly valve based. With such a DAC the argument for or against 16 or 24 bit sounding different would be next to impossible to resolve. So that brings me to the question as to exactly why we would want an even higher bit and sample rate if we are actually after a quality of playback that is enjoyable to our individual taste? I had a recent taste of this with a friend that I would like to share with you. For some reason he has a DAC and loads of 24/96kHz files that sound out of this world. He also has some of the files in CD format and 24/192. But they cannot match the same level of enjoyment and foot tapping experience in terms of bass solidness, soundstage, and cleanness of the highs. His set up is in total harmony with 24/96. As far as I am concern his other music collection in the other formats are just a waste of time and money. But we don't know what will sound best in our system until we have acquired them and tried them out. And that is what makes this kind of discussion controversial. It's not so much as what is better, but what makes your own system set up sound better, or at least just as good for a lower expenditure on music formats.


----------



## castleofargh

baxide said:


> The problem I have with most of this is the fact that the word better is heavily misused in its intended understanding. Let's look at what we really mean when we are talking about being better. You'll first need a DAC that can decode the two types of audio files into an analogue audio sound where a difference can be detected in the first place. I have heard a couple of lovely sounding DACs that made mp3 sound amazing. They were mostly valve based. With such a DAC the argument for or against 16 or 24 bit sounding different would be next to impossible to resolve. So that brings me to the question as to exactly why we would want an even higher bit and sample rate if we are actually after a quality of playback that is enjoyable to our individual taste? I had a recent taste of this with a friend that I would like to share with you. For some reason he has a DAC and loads of 24/96kHz files that sound out of this world. He also has some of the files in CD format and 24/192. But they cannot match the same level of enjoyment and foot tapping experience in terms of bass solidness, soundstage, and cleanness of the highs. His set up is in total harmony with 24/96. As far as I am concern his other music collection in the other formats are just a waste of time and money. But we don't know what will sound best in our system until we have acquired them and tried them out. And that is what makes this kind of discussion controversial. It's not so much as what is better, but what makes your own system set up sound better, or at least just as good for a lower expenditure on music formats.


 

 it's not hard to get one or 2 highres albums and convert them down to lower resolutions for a test. if we're testing the resolution, it's way more pertinent than listening to a CD version and a highres version where chances are the mastering is different.
 now some mastering are better than others. and just like we will have our own preferences for one interpretation of a classical piece by some maestro, we will have a preference for one mastering over another. that's obvious and fine as long as it's not mistaken for a resolution sounding better.


----------



## old tech

blackstonejd said:


> I am a pretty hardcore audiophile and tend to agree that the format itself is not as important as I thought. This is evidenced by the fact that a lot of the material that is being released on PONO and HDTracks is just better than what was released on CD. The differences between the Crosby Stills and Nash CDs and the 192KHz files is significant in my opinion, but they are also so different that it is not an apples to apples comparison. But almost without exception, the material I get from HD Tracks is superior to the prior releases, is less fatiguing to the ear and has a certain "quietness" or blackness between the notes. It is just more musical and effortless sounding. You do not necessarily loose that, however, in my experience, by playing those same files back at 44.1KHz. In fact much of the improvement is still there.
> 
> What I suspect is actually happening is that the content itself is better which makes up the lion share of the perceived improvement. But it does not really matter at the end of the day, because the product that is being put out is consistently superior, in my experience, than the CD releases that preceded them.
> 
> ...


 
 Funnily enough, I'm having the opposite experience, ie usually turning the volume up with CDs and down with Hi Res.  Probably because most of the CDs I've hung on to are either older CDs or the better mastered later ones - I can't stand over compressed/brickwalled music.  What this demonstrates to me at least, is that even most of the hi res remasters have not escaped the general trend towards loudness.


----------



## RRod

old tech said:


> Funnily enough, I'm having the opposite experience, ie usually turning the volume up with CDs and down with Hi Res.  Probably because most of the CDs I've hung on to are either older CDs or the better mastered later ones - I can't stand over compressed/brickwalled music.  What this demonstrates to me at least, is that even most of the hi res remasters have not escaped the general trend towards loudness.


 
  
 It's because the hi-res push is, at its core, not about sound quality. They'll happily charge you $30 for a 60-year-old album that still sounds bad just because the format is more specced.
  
 I have classical CDs that have me both turning the volume up and turning it down, which means that already the medium can present the material with enough dynamic range as to be annoying in my listening environment (a home, not even a bus or anything). The audiophile classical labels were putting out amazing sounding, dynamic stuff as soon as digital was even a thing back around 1980. By 1990 the engineers who knew their shinola had consistency down, I wager due to the leeway allowed by high bit depths and better DAWs. But still, it was all coming out on the lowly CD because the end product still didn't exceed the format's abilities. SACD and DVD-A brought a little bit of the hi-res shenanigans to the classical world, but the latter died and the former only hung on due to embracing the hybrid format (but unfortunately kept multichannel mixes in verdammten DSD) and was thus really a gussied-up CD. So basically I'm sitting here with 35 years or so of tremendous sounding classical CDs, that all fit on my rockboxed ipod classic thanks to the Opus codec, and wondering "What's the big deal?"


----------



## gregorio

axle_69 said:


> Actually you don't need 24 bits, if you want to be picky 20 bits is enough dynamic range for music, and 16 bits is better than vinyl.


 
  
 If you want to be picky, around 10bits is enough dynamic range for music and most music genres require even less.
  


baxide said:


> His set up is in total harmony with 24/96. As far as I am concern his other music collection in the other formats are just a waste of time and money. But we don't know what will sound best in our system until we have acquired them and tried them out. And that is what makes this kind of discussion controversial. It's not so much as what is better, but what makes your own system set up sound better, or at least just as good for a lower expenditure on music formats.


 
  
 That's not what makes this kind of discussion controversial. There are two reasons why it's controversial: 1. Compared to many audiophiles and to audio professionals, this approach is backwards! It's not about finding music which suits your system setup but about creating a system setup which suits the music/audio. 2. There is no sound system ever invented which can actually play 24bits, no commercial recordings ever released which use more than about 10bits or so and no human can hear audio frequencies up to 48kHz. It's therefore not possible to setup a system which "is in total harmony" with 24/96 without it being just as much in "total harmony" with 16/44.1, unless of course the system is defective/won't physically play 16/44.1. As others have said, it's the mastering which makes the difference, not the audio container (format) the master is contained in.
  


rrod said:


> By 1990 the engineers who knew their shinola had consistency down, I wager due to the leeway allowed by high bit depths and better DAWs.


 
  
 Not so much. DAWs didn't really become practical until the mid '90's and even then, only for basic editing tasks (rather than recording, processing or mixing) and higher bit depths were not so relevant either. It was a combination of factors; improving ADCs, higher channel counts, the fact that digital was maturing and the engineers were really learning the tricks of how to get the best from digital and of course the pro audio manufacturers were coming out with improved mixing consoles and related equipment, with more functionality and signal paths more suited to the workflows and fidelity requirements of these engineers who knew their digital "shinola".
  
 G


----------



## dmbr

My understanding is that there are reasons to record in "HD" (>44hz/>16bit), so what makes "HD" audio potentially better is that it eliminates the possibility of the conversion to 16/44 degrading sound quality.

That is to say, 24/192 files can sound better than 16/44 not because the sampling rate and bit depth are higher, but because the sampling rate and bit depth have not been altered.


----------



## RRod

dmbr said:


> My understanding is that there are reasons to record in "HD" (>44hz/>16bit), so what makes "HD" audio potentially better is that it eliminates the possibility of the conversion to 16/44 degrading sound quality.
> 
> That is to say, 24/192 files can sound better than 16/44 not because the sampling rate and bit depth are higher, but because the sampling rate and bit depth have not been altered.


 
  
 Try for yourself. Take a 24/192 file, convert it to 16/44.1, then convert it back to 24/192 and compare it to the original. A good resampler doesn't do anything that my ears can detect.


----------



## gregorio

dmbr said:


> My understanding is that there are reasons to record in "HD" (>44hz/>16bit), so what makes "HD" audio potentially better is that it eliminates the possibility of the conversion to 16/44 degrading sound quality.
> 
> That is to say, 24/192 files can sound better than 16/44 not because the sampling rate and bit depth are higher, but because the sampling rate and bit depth have not been altered.


 
  
 There are reasons to record using 24bit rather than 16bit but not because of higher fidelity and there's no longer much reason to record at higher than 44.1k, except obviously for marketing purposes. Your statement about conversion degrading sound quality used to be true, maybe 15 or so years ago. As RRod said, decent resamplers are transparent these days. In fact, sample and bit rate conversions are routine within mixing processors these days and a commercial mix is likely to contain multiple, possibly even dozens of sample and/or bit rate conversions before it even arrives at final mastering. The whole "keep it at 192k" (or whatever) to legitimately call it "Hi-rez" is just marketing BS which is largely divorced from the actual reality of creating a commercial recording these days.
  
 G


----------



## ralphp@optonline

Perhaps simplest way to address this issue is just to ask exactly what part of the Nyquist–Shannon sampling theorem is it that one disagrees with and to please provide mathematical proof of the error in the original theorem. Everything else is just pointless back and forth with no chance of resolution.
  
 So for example, if the minimum sampling rate, which Nyquist states needs to be a minimum of twice the highest frequency, i.e. 44.1kHz sample rate for frequencies up to 22kHz, needs to be higher than 2X then state the minimum required sample rate and provide mathematical proof.


----------



## RRod

Those who refuse to do blind listening tests don't often seem convinced by appeals to signal processing theory.


----------



## gregorio

ralphp@optonline said:


> Perhaps simplest way to address this issue is just to ask exactly what part of the Nyquist–Shannon sampling theorem is it that one disagrees with and to please provide mathematical proof of the error in the original theorem.


 
  
 Seems like a good idea in theory but this approach still has problems:
  
 1. Many don't understand what a theorem actually is, or how this particular one applies to digital audio. Many theories exist to explain some observed natural phenomena and may or may not be correct or entirely correct. This doesn't apply to digital audio though, because digital audio is a technology invented from the theory, not something which existed naturally and then a theory was invented to try to explain it. A fairly subtle but important difference which many audiophiles can't or don't want to grasp.
  
 2. Physical audio equipment (ADCs and DACs) cannot perfectly apply the Nyquist-Shannon sampling theorem, there are real world engineering constraints which make it effectively impossible, a fact which the better educated audiophiles and audiophile equipment marketing departments are happy to exploit. In practise, real world engineering constraints still allow us to get very close to a perfect implementation of Nyquist-Shannon and any imperfections which do exist are (or should be) many times below audibility.
  
 3. You assume, incorrectly, that Nyquist-Shannon automatically applies to all digital audio but this isn't always the case in the audiophile world. There was a trend at one point (I don't know if it still exists) by a few audiophile DAC manufacturers to market "filterless" DACs. This deliberately breaks the Nyquist-Shannon theorem, which specifically requires a band-limited signal (a signal which does not exceed half the sampling rate).
  
 4. To some/many audiophiles their perception is sacrosanct, hearing is believing and they "trust" their hearing. Because this belief is sacrosanct and therefore unquestionable, anything which does question (or worse, disagrees with) it, MUST be incorrect. No matter how proven Nyquist-Shannon is, or how impossible to disprove, it just doesn't make any difference, it MUST somehow be wrong or it (and any other part of accepted science) MUST have missed something. Unquestionable belief in a perception is an irrational position to take and even more frustratingly, rational arguments are typically ineffective against irrational people. In fact, the more rational/proven/accepted the facts presented, the more irrational/ludicrous the responses have to become in order to preserve the sanctity of their belief, and this inevitably results in accusations of a complete lack of basic education or of actual insanity. As personal attacks are forbidden here, and this site largely exists to cater to audiophiles, the only option is to effectively ban science and rationality from any discussion. We end up with a hobby/passion entirely based on science but where the discussion of that science is effectively forbidden (except in this sub-forum), a thoroughly bizarre state of affairs!!
  
 I'm usually all for letting sleeping dogs lie and allowing people to accept any old nonsense they want to support whatever beliefs they want. The problem is that many of these dogs aren't sleeping! Many/Most who come to this site come here for information and immediately become targets for indoctrination by these (very awake) dogs and the manufacturers/retailers who make a living from them (and their shills). All of which is incredibly frustrating for those of us who actually love audio. [/rant]! 
  
 G


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## castleofargh

maybe this topic should end, or be renamed for what it has turned into? as the flac question was really answered on first page.
 I was tempted to just move posts to some cd vs highres topic, but it hard to find one that wasn't locked for turning into a troll fest or a boxing match.


----------



## ghostnote

gregorio said:


> Seems like a good idea in theory but this approach still has problems:
> 
> 1. Many don't understand what a theorem actually is, or how this particular one applies to digital audio. Many theories exist to explain some observed natural phenomena and may or may not be correct or entirely correct. This doesn't apply to digital audio though, because digital audio is a technology invented from the theory, not something which existed naturally and then a theory was invented to try to explain it. A fairly subtle but important difference which many audiophiles can't or don't want to grasp.
> 
> ...


 
 /thread


----------



## 1c3d0g

So before it gets locked, as I still am quite confused by all this, is there a benefit to setting my PC to 24-bit/192 kHz as opposed to 16-bit/44.1 kHz?


----------



## axle_69

Playing two different original files (192/24 vs 44.1/16, HiRes vs CD) or choosing between playing an original file at 44.1/16 or upsampling it in the computer  to 192/24? If it is the latter you should be upsampling to a multiple, in this case 176.4. If it should be done in the computer/software or in the DAC/firmware depends on which does a better job, or if it should be done at all... To add a bit to the confusion in delta-sigma demodulators there is also oversampling that take the signal to the MHz region. Google a bit about binary-weighted DACs (and R-2R) and delta-sigma DACs, and search in other forums.
 I don't upsample original files above 88.2 or 96 kHz, in the computer or in the DAC (some DACs don't allow turning off upsampling or choosing the oversampling of the delta-sigma) and since the music player does a good job (upsamples before it starts transmitting and allocates enough resources so the pace through SPDIF is kept constant) I prefer to upsample ripped CDs in the computer (but when using iTunes not upsampling at all sounded better). But I guess it will depend a lot on the player/DAC combination, SPDIF/USB async., etc. Google, google, google... and do whatever sounds better to you (if you notice any difference at all).


----------



## Deftone

Iv got question, if we could all hear the difference would a 24/96 track that is brick walled to something like DR5-6 does it make high resolution aspect of it pointless considering is cutting off the top and bottom?


----------



## RazorJack

1c3d0g said:


> So before it gets locked, as I still am quite confused by all this, is there a benefit to setting my PC to 24-bit/192 kHz as opposed to 16-bit/44.1 kHz?


 
  
 No.
  
 Nein.
  
 Non.
  
 Nee.
  
 It's really not that confusing, is it?  Just use 16 bit / 44.1 kHz, it's more than sufficient for our human ears.


----------



## RRod

deftone said:


> Iv got question, if we could all hear the difference would a 24/96 track that is brick walled to something like DR5-6 does it make high resolution aspect of it pointless considering is cutting off the top and bottom?


 
  
 It would certainly make the 24-bits part pointless, but delivery at 24-bits is already pointless even for material with much higher DR ratings.


----------



## spruce music

deftone said:


> Iv got question, if we could all hear the difference would a 24/96 track that is brick walled to something like DR5-6 does it make high resolution aspect of it pointless considering is cutting off the top and bottom?


 

 Depends.  If we can all hear the difference maybe.  We can't of course.  Somewhere near 20 khz is the limit.  So called higher rez is merely higher bandwidth.  Much of it we can't hear.
  
 The higher sample rate gives you higher bandwidth.  The extra bits don't do a better or finer job of encoding material below 20 khz.  That is the mind picture used to market higher rez.  But it is not so.


----------



## noplsestar

Hi,
  
 I´m new here and sorry if the questions were asked in another post:
  
 Is it better to download a track - 16 bit 44.1 - directly from let´s say Qobuz/HD Tracks etc. in FLAC / ALAC version OR is it better to buy a CD and then convert it with iTunes in lossless ALAC?
 OR is it even better to download it in the original WAV version. Do you hear a difference between FLAC/ ALAC and WAV? Some different questions, I know. Just curious what you think/hear!
  
 Thanks for any replies!


----------



## reginalb

noplsestar said:


> Hi,
> 
> I´m new here and sorry if the questions were asked in another post:
> 
> ...


 
  
 That is a hard question to answer. Sometimes Quobuz/HDtracks/any similar store might have different masters, sometimes they dont. 
  
 Case in point: The newest RHCP album from the dynamic range database
  
CD, iTunes, HDtracks
  
 As you can see, they all appear to have identical DR. Now, I am sure it could be purely by coincidence, but I doubt it. That was likely one master, sampled 3 ways. They probably sound audibly identical. Take the cheapest of the 3 (which will probably be iTunes or CD, or go with 7Digital, Amazon, or Google, probably all identical)
  
 However, Norah Jones Feels Like Home:
CD, SACD
  
 Those are different masters, without a doubt. And the DSD download fro Acoustic Sounds is also from that master. 
  
 But even the HD music stores aren't the same. Here is a quote from me in another post


> Michael Jackson Thriller provides an illuminating example. I currently own this, but I have the 30th Anniversary Edition, with an overall DR score of 09 (http://dr.loudness-war.info/album/view/110165) I uploaded that one, so I know its the score that my own copy got. The best example are the vinyls and the first pressing Japanese CD at 15. That one would be harder to find, presumably, but the DSD download from Acoustic Sounds is easily available to download (for $25) just behind those masters with an overall score of 14. I assume 14 and 15 are pretty indistinguishable, and you in fact find that one track has more DR in the DSD version. I assume more is not always better (I am not an engineer so that is just a lay person's assumption) - and I pretty much consider anything on there in the green to be just fine.
> 
> So, go to Acoustic Sounds and get the DSD, but you could also go to HD Tracks and get the 24-bit/88.2KHz version which scores just behind the AS DSD, and is $7 cheaper. You could also get a 24/96 version for the same price as the HDTracks version from Acoustic Sounds. But wait, it's the _lowest _scoring example (tied with some others). It's actually slightly worse than my MP3 download.


 
  
 As a recap of that

Original Japanese CD
DR 15 (Price would vary, expensive on eBay from what I've seen)

25th Anniversary CD
DR 9 (more tracks, though)

Acoustic Sounds DSD              
DR 14 ($25)

HDtracks 24/88.2                      
DR 13 ($18)

Acoustic Sounds 24/96
DR 7 ($18)

  
 To answer your question, it's a minefield.


----------



## noplsestar

reginalb said:


> That is a hard question to answer. Sometimes Quobuz/HDtracks/any similar store might have different masters, sometimes they dont.
> 
> Case in point: The newest RHCP album from the dynamic range database
> 
> ...


 
  
 Thanks!
  
 I already got the "thriller" in DSD from HD Tracks ... and now seeing that it has a very good DR score I´m kind of glad I made the right decision when buying it 
  
 What I didn´t get so far is the DR thingy. It means dynamic range, am I right? So when there´s more dynamic in the recording (master), it´s "better".
  
 The strange thing is: I can hear that the new Red Hot Chili Peppers album is really compressed (like many rock albums), but despite this fact (and the bad DR data), I really like the mix of their new album (maybe because Nigel Godrich was the engineer, I don´t know). So does this mean, not to be tooo picky about the DR data?
  
 And about my other questions: Do you know if it´s better to listen to FLAC, ALAC or WAVs? Or are they all the same (if I have a good DAC)?


----------



## Deftone

As long as they are using the same master all lossless file are the same sound quality


----------



## reginalb

noplsestar said:


> Thanks!
> 
> I already got the "thriller" in DSD from HD Tracks ... and now seeing that it has a very good DR score I´m kind of glad I made the right decision when buying it
> 
> ...


 
  
 With regards to RHCP, you just can't know the counterfactual. I had an argument in the thread I quoted myself from. Some people who are (allegedly) involved in the recording industry argue that compression is good, most people like it. I don't think they're correct, and their reasoning contains that very flaw: You don't know the counterfactual.
  
 I don't just not listen to music because it's compressed, I like that RHCP album. I acutally provided the HDtracks data in the DRdb, because I bought it hoping it would be a better master than the one that I was streaming from Play Music. It's not different, would I like a more dynamic one more? I think so. That doesn't mean that this one is terrible. Would it be better if it were more dynamic? You just can't answer that. I also like Californication. But I really like the rough mix of Californication. 
  
 Regarding lossless vs. lossy: I personally think that you're OK with relatively high bit-rate, VBR lossy encodes. I have two copies of my library, one that's in whatever the original format I obtained was, and one where the whole thing is re-encoded with the Opus codec. They sound the same to me.


----------



## RRod

noplsestar said:


> What I didn´t get so far is the DR thingy. It means dynamic range, am I right? So when there´s more dynamic in the recording (master), it´s "better".


 
  
 It's more of a compression detector than a true dynamic range measurement, in that the DR for an album can be lower than an individual track, but certainly the album is at least as dynamic as its most dynamic track…


----------



## noplsestar

Thanks for your posts! Now I'm a but more wiser on that topic (and won't buy an album without checking on that DR website first)


----------



## noplsestar

Did you read that?

http://www.innerfidelity.com/content/great-paper-released-meta-analysis-high-resolution-audio-perceptual-evaluation#SDpMGqCMjc1TAVkt.97


----------



## reginalb

noplsestar said:


> Did you read that?
> 
> http://www.innerfidelity.com/content/great-paper-released-meta-analysis-high-resolution-audio-perceptual-evaluation#SDpMGqCMjc1TAVkt.97


 
  
 Yes, there is a thread about it, here you are: http://www.head-fi.org/t/812565/a-meta-analysis-of-high-resolution-audio-perceptual-evaluation-or-how-we-learned-to-stop-worrying-and-love-hi-res
  
 Here's an alternate discussion on it: https://www.reddit.com/r/headphones/comments/4qoe3y/great_paper_released_a_metaanalysis_of_high/
  
 and the first comment here: https://www.reddit.com/r/audiophile/comments/4rjzoa/what_to_listen_for_in_high_resolution_audio/


----------



## willyvlyminck

elmoe said:


> But the whole point of SACD is that they're remastered to sound better.


 That is marketing ,if they don't come up with so called new systems, sales lowering because a good cd player can last a lifetime


----------



## gregorio

willyvlyminck said:


> That is marketing ,if they don't come up with so called new systems, sales lowering because a good cd player can last a lifetime


 
  
 I think you are maybe getting confused. Some/Most SACDs are in fact mastered or remastered to sound better! Assuming we mean "has a larger dynamic range" and therefore "sounds better" in certain listening situations (and worse in some others). The reason SACDs are often mastered/remastered to "sound better" is because listeners were effectively constrained to a limited set of relatively high quality listening senarios, SACD players were not portable and SACDs were specifically designed not to be ripped. Where the marketing comes in is that it's easy to create exactly the same "better sounding" master/remaster (for the same listening scenarios) using the 44.1/16 (CD) format but of course the labels/distributors don't want to do that because it would undermine the cost difference they can charge for the HD/DSD format versions.
  
 G


----------



## Hi-Fi'er

Truth about Audio: https://www.youtube.com/watch?v=Z5S_DI99wd8&feature=youtu.be


----------



## castleofargh

I do like Mr Waldrep, since long ago. we share, among other things, a strong love for DSD sarcasm. but next time I'd like it if you could avoid spamming the link on several topics. spam is never cool.
  
 just a little argument about the video, pono actually can sound audibly different at 16/44 if you have a headphone that extends high enough, and if the listener can still hear high freqs. because the low pass on 16/44 start way sooner in the audible range on the pono so there is an EQ difference within the audible range. so his example is less than perfect IMO even if I tend to agree with the conclusion.


----------



## Asuhra

I create the most advanced Audio Subliminals on Earth and the thing I found long ago is that there is an insane difference between 44.1Khz and 48Khz, let alone more Khz. 44.1Khz subliminals sound very harsh, sounds are too congested also, sometimes even the volume has weird spikes after converting to 44Khz which are not present at 48Khz. 48Khz is night and day difference, much smoother sounds, more info can be put on the audio, etc. Now I'm about to venture into the 96Khz world, hoping it will take my subliminals to the next level, although they are already able to change me completely in one day at only 44.1Khz and 48Khz.
  
 Also, most DACs and DAPs only go from 20hz to 20Khz Freq Range but I can tell you right now the subconscious mind can hear at least up to 30Khz and it will act on that info put above the 20Khz range. 
  
 It pisses me off to no end when I see every damn DAC and DAP limited to 20Khz!!!!! So freaking limiting for my work...Arghhhh


----------



## gregorio

asuhra said:


> I create the most advanced Audio Subliminals on Earth ...


 
  
 Are you saying you're a sound designer/re-recording mixer for feature films?
  


asuhra said:


> 44.1Khz subliminals sound very harsh ...


 
  
 That's a contradiction! Either the sounds are subliminal, in which case one is not consciously aware of them and therefore they don't sound harsh (or any other adjective) or, they sound harsh and are therefore not subliminal, which is it?
  


asuhra said:


> 48Khz is night and day difference...


 
  
 Of course it's not!
  


asuhra said:


> Now I'm about to venture into the 96Khz world, hoping it will take my subliminals to the next level, although they are already able to change me completely in one day at only 44.1Khz and 48Khz.


 
  
 It won't make any difference. I think maybe you are confusing the word "subliminal" with the word "inaudible"?
  


asuhra said:


> Also, most DACs and DAPs only go from 20hz to 20Khz Freq Range but I can tell you right now the subconscious mind can hear at least up to 30Khz and it will act on that info put above the 20Khz range.


 
  
 You can "tell us right now" any old rubbish, for example "I can tell you right now" that the moon is made of cheese. This is the science forum however and unless you can substantiate what you are telling us, then it's of little worth and, if you're going to try to tell us something which flies in the face of science, then how you substantiate what you're telling us had better be quite extraordinary or it will just be viewed as some nutter spouting rubbish. For example, can you explain how the "mind", subconscious or otherwise, can "hear" up to 30kHz when there are no physiological structures in the ear capable of responding to frequencies that high?
  


asuhra said:


> It pisses me off to no end when I see every damn DAC and DAP limited to 20Khz!!!!! So freaking limiting for my work...Arghhhh


 
  
 Why would that piss you off? I'm the opposite, it would piss me off no end to see a DAC/DAP go beyond 20kHz. What work is it that you do which would be limited/affected?
  
 G


----------



## castleofargh

I was lost at "most advanced Audio Subliminals on Earth".
 aside from wondering excitedly if I could use it for world domination, I'm really unclear on what that is.


----------



## Ruben123

asuhra said:


> I create the most advanced Audio Subliminals on Earth and the thing I found long ago is that there is an insane difference between 44.1Khz and 48Khz, let alone more Khz. 44.1Khz subliminals sound very harsh, sounds are too congested also, sometimes even the volume has weird spikes after converting to 44Khz which are not present at 48Khz. 48Khz is night and day difference, much smoother sounds, more info can be put on the audio, etc. Now I'm about to venture into the 96Khz world, hoping it will take my subliminals to the next level, although they are already able to change me completely in one day at only 44.1Khz and 48Khz.
> 
> Also, most DACs and DAPs only go from 20hz to 20Khz Freq Range but I can tell you right now the subconscious mind can hear at least up to 30Khz and it will act on that info put above the 20Khz range.
> 
> It pisses me off to no end when I see every damn DAC and DAP limited to 20Khz!!!!! So freaking limiting for my work...Arghhhh




Just to be sure, youre kidding right?


----------



## fuzun

Well, yes beyond 20khz is inaudible but they are still "energy" and they go through ear canal as well. Maybe even though we don't hear, can they make difference at the audio we perceive as stressing or making resonances some parts of brain or head?


----------



## Mr Rick

fuzun said:


> Well, yes beyond 20khz is inaudible but they are still "energy" and they go through ear canal as well. Maybe even though we don't hear, can they make difference at the audio we perceive as stressing or making resonances some parts of brain or head?


 
  
 Can you give me a scientific study that backs up your 'facts'?


----------



## fuzun

mr rick said:


> Can you give me a scientific study that backs up your 'facts'?


 
 It is only a question. Do you want facts about being sound waves are energy?


----------



## castleofargh

fuzun said:


> Well, yes beyond 20khz is inaudible but they are still "energy" and they go through ear canal as well. Maybe even though we don't hear, can they make difference at the audio we perceive as stressing or making resonances some parts of brain or head?


 

 but if it's not audible, is it audio? ^_^
 there are plenty of energy sources hitting us all day long, but if we lack the receptors they go unnoticed. so how significant is the ultrasonic content of a song? I'd guess it's not unless it starts affecting the playback system in a negative way.
 when you look at albums, you usually have very little loudness in the ultrasounds compared to the rest of the music. and ultrasounds as physical pressure consciously felt on the skin requires a really loud signal (I think it was above 100db but you'd have to find the paper to get the value they found).
 last but not least, the guy mastering the album also fails to hear ultrasounds, so how do you think he masters that area?


----------



## fuzun

castleofargh said:


> but if it's not audible, is it audio? ^_^
> there are plenty of energy sources hitting us all day long, but if we lack the receptors they go unnoticed. so how significant is the ultrasonic content of a song? I'd guess it's not unless it starts affecting the playback system in a negative way.
> when you look at albums, you usually have very little loudness in the ultrasounds compared to the rest of the music. and ultrasounds as physical pressure consciously felt on the skin requires a really loud signal (I think it was above 100db but you'd have to find the paper to get the value they found).
> last but not least, the guy mastering the album also fails to hear ultrasounds, so how do you think he masters that area?


 
 I agree with you. I mean they are inaudible in a direct way.
  
 Also there are a lot of energy sources but how much are them mechanic? Due to natural selection, our body is very capable of perceiving mechanical energy only if it has enough power. Unlike electromagnetic energy, we really do not need any specific receptors to perceive mechanical energy because almost our entire body is somehow capable of interpreting mechanical stress (if they have enough power). The thing is, if you have specific receptors, you get proper feedback according to characteristics of the wave. However, like you said I don't think this should be considered "audio".
  
 But I am not sure if these indirect effects would make sq any better. Beyond this point, scientific works needed. My personal opinion is that they may effect in a positive way because people who make music get that indirect effect too (if that statement is true). And we would like to listen music how it was intended to sound. (Mastering, ... not involved. This is only in theory.)
  
 Considering low levels of ultrasound found in music files due to cropping, bad(?) mastering(?), (I think) It would be impossible to perceive them even indirect form.
  


> and ultrasounds as physical pressure consciously felt on the skin requires a really loud signal (I think it was above 100db but you'd have to find the paper to get the value they found).


 
  
 If you take hand skin (where energy reflects to epidermal hand cells), yes it needs big magnitude. But ear is very complex organ and there are different type of cells at where stereocilia located. Maybe these cells need less power than hand cells to notice it to brain via neurons?
 If we can make them notice the brain, we still wont be getting that as "audio" or "sound". It is a thing that I cant describe because of lack of knowledge and english. But I think it like this: When you listen extreme bassy musics, you think like your brain is shaking (maybe not correct term but you got the point). That "shaking feeling" is not audio but it is a intended thing to experience while listening music. Because music makers got that feeling too.
  
  
 Sorry for bad English. I hope you get the point.


----------



## Asuhra

Argh!!!
 Just wrote a giant reply responding to everyone about the subl1minals and re-sampling. Went to dinner, clicked preview before posting and everything disappeared. Just f.... great...
 Restarting, this time summarized.
 Subl1minal Audios are done in the frequencies range between 15.5Khz and 20Khz because of the DAPs and DACs limits but that means you can only put so much info on the 44.1Khz samples because there is no more headroom. At 48Khz you have much more headroom to put info between the 15.5 and 20Khz Freq range when doing subliminals (it's a mathematical thing). I am guessing that's why 44.1Khz samples sound so much worst.
 It has been proven by dog trainers that you can hear at least up to 30Khz because they got affected by the subl1minals meant for their dogs @ 30Khz range. I have proven that on myself time and time again.
 As for the resolution, that has also been proven that some people notice differences up till 96Khz. More than that is placebo, at least for the conscious mind. 192Khz resolution seems really overkill, let alone more.
 The subconscious mind is really the one in control 90% of the time you're awake and every decision you make is made at the subconscious level first. The conscious mind is only really there to obey the subconscious unless you have tremendous will strength and can overpower it.
 That said, you can be listening to the most stunning track of all time but if that track has something above the 15.5Khz range that your subconscious mind doesn't like, like noise, distortion or even bad subliminal messages you will never like the track and you will have no clue why. For your conscious mind the music sounds amazing and wants more of it but since the subconscious is the one in control and he doesn't like it he will make you not like the song and your conscious mind will come up with all sorts of nonsense reasons for why you don't like the song. 
 I could add one of my subliminals to any song you want and reprogram your subconscious mind to hear the sound more 3D like, crystal clear, etc and it will happen sooner or later, depends how long the subconscious resists the suggestions, if he resists it at all.
 That's the reason why I would like DAPs and DACs to go at least till the 30Khz range, not to listen to music but to create out of this world subliminals. I could put several layers of messages at different Frequency ranges instead of only being able to put them between the 15.5 and 20Khz range.
 Btw, to listen to music tracks I completely cut the EQ above the 15Khz range because God knows what crap can be in the inaudible range and how the subconscious might interpret it. I don't consciously hear anything at above 14Khz anyway so I can cut it all.
 What I do is I have 2 audio players, one playing music with freq cut above ~15khz and another player playing my subliminals @ freq above 15.5Khz. This way I'm having fun while also working on myself. 2 in 1 solution.  Until someone develops a DAP that can do this I can only use my android with rocket player playing the subl1minals and N7Player playing my music. 
 I only work with professionals and corporations because my subl1minals are at a point where I could get someone to kill himself in just hours so they have to be highly controlled like class A substances but you can still get nice subliminals for the general public and get more info about subl1minals from 1ndigo M1nd Labs which I think is the second best producer in the World right now. He as a long ass way to go and has some atrocious subl1minals but he also has some nice pearls between all the crap, including the Free Self Confidence subliminal that I recommend. Anyway, he has the best forum on the subject so if you want to know more about this topic go take a look.
 I just find it amazing how we go through our entire lives without knowing a thing about the subconscious mind and how deeply it affect us.
  
 [edit] Replaced some of the i's with 1's so this info doesn't show up in search engines. Don't want publicity for myself and not interested in promoting others for free.


----------



## Asuhra

What the hell is wrong with my text, doesn't fit the screen??
  
 edit: nevermind, it's fixed but lost the formating.


----------



## spruce music

asuhra said:


> Argh!!!
> Just wrote a giant reply responding to everyone about the subliminals and re-sampling. Went to dinner, clicked preview before posting and everything disappeared. Just f.... great...
> Restarting, this time summarized.
> Subliminal Audios are done in the frequencies range between 15.5Khz and 20Khz because of the DAPs and DACs limits but that means you can only put so much info on the 44.1Khz samples because there is no more headroom. At 48Khz you have much more headroom to put info between the 15.5 and 20Khz Freq range when doing subliminals (it's a mathematical thing). I am guessing that's why 44.1Khz samples sound so much worst.
> ...


 

 Okay, go to 48 khz and you get pretty close to nothing extra.  The transition between flat and no response is now 4 khz wide instead of 2050 hz wide.  Both are flat to 20 khz (usually depending on correct reconstruction filtering).   So both 48 and 44.1 khz will give you the same result between 15.5 khz and 20 khz.  If you go to 88 or 96 khz you have flat response to 40 khz.  Plenty more room for your sub-liminals now.  Though I don't belief any of that stuff you wrote.  If your dog trainers are hearing 30 khz, it most likely is IMD reflecting back below the 20 khz range.


----------



## old tech

reginalb said:


> Yes, there is a thread about it, here you are: http://www.head-fi.org/t/812565/a-meta-analysis-of-high-resolution-audio-perceptual-evaluation-or-how-we-learned-to-stop-worrying-and-love-hi-res
> 
> Here's an alternate discussion on it: https://www.reddit.com/r/headphones/comments/4qoe3y/great_paper_released_a_metaanalysis_of_high/
> 
> and the first comment here: https://www.reddit.com/r/audiophile/comments/4rjzoa/what_to_listen_for_in_high_resolution_audio/


 

 For a more science/methodology based discussion...
https://hydrogenaud.io/index.php/topic,112204.0.html


----------



## old tech

asuhra said:


> I create the most advanced Audio Subliminals on Earth and the thing I found long ago is that there is an insane difference between 44.1Khz and 48Khz, let alone more Khz. 44.1Khz subliminals sound very harsh, sounds are too congested also, sometimes even the volume has weird spikes after converting to 44Khz which are not present at 48Khz. 48Khz is night and day difference, much smoother sounds, more info can be put on the audio, etc. Now I'm about to venture into the 96Khz world, hoping it will take my subliminals to the next level, although they are already able to change me completely in one day at only 44.1Khz and 48Khz.
> 
> Also, most DACs and DAPs only go from 20hz to 20Khz Freq Range but I can tell you right now the subconscious mind can hear at least up to 30Khz and it will act on that info put above the 20Khz range.
> 
> It pisses me off to no end when I see every damn DAC and DAP limited to 20Khz!!!!! So freaking limiting for my work...Arghhhh


 
  
 The search for credible evidence that sound outside the 20-20khz range can be perceived has been the subject at the fringe of audio science for decades.  To date, there still is not any credible evidence.  The closest any experiment has got was the Oohashi study which was not able to be replicated and later rejected.  In contrast there are many well designed experiments over the past decades which found no evidence that humans can perceive these frequencies, at least at sound pressure levels which wouldn't fry the listener.
  
 Just because you believe you can perceive them does not make it a fact.  Astrologists also believe that the position of the planets affect personality types and human behaviour but again all the available evidence contradicts it, even if common sense doesn't come into play before hand.
  
 The other thing which works against your speculation is that even if it was possible that these out of range frequencies affect the sound within the range, it would only be relevant to the live acoustic event and not a recording of it.  That is because the effect is by definition in the human range of hearing so providing the mike and recording/playback chain can handle 20-20khz those effects will be recorded and played back.  There is no logical reason why in addition the out of range frequencies would also need to be captured in addition to the sound which has been changed by those frequencies.


----------



## RRod

castleofargh said:


> when you look at albums, you usually have very little loudness in the ultrasounds compared to the rest of the music. and ultrasounds as physical pressure consciously felt on the skin requires a really loud signal (I think it was above 100db but you'd have to find the paper to get the value they found).


 
  
 Yeah, the math never really works out to anything logical. I just took a 192k album and found the ultrasonic content is 50dB RMS below the stuff <20k. So I guess if people are willing to listen to this at 150dBSPL then perhaps they'll "feel" the ultrasounds, in addition to their skin peeling off…


----------



## spruce music

There is this.  Young adults 18-20 years old with good hearing can hear to 24 khz.  Of course thresholds for that are over 100 db. 
  
 http://www.isa-audiology.org/periodicals/1971-2001_Audiology/1984,%20%20Audiology,%20%20Vol.%20%2023/No.%205%20%20(441-524)/Henry%20%20Fast,%20%20Audiology,%201984.pdf
  
 You also will notice a flatter response around 14-16 khz with rapidly rising thresholds as you go higher.  The reason for that is the hair-like cells in the ear really respond to 15 khz.  At very high levels they respond weakly a bit above that point.  There are no cells sized to respond to 20 khz actually. 
  
 You almost could say human hearing is good to 15 khz with a transition band to 20 khz before no response.  Other research I have read show by age 28-30 only a very small percentage could respond to anything at any level above 20 khz.
  
 Music pretty much never has enough energy in those higher frequencies to be heard.  And these are test tones.  If mixed with other sounds such minor response will be fully masked by lower frequencies.   If it worries you, go to 96 khz and you have it covered until humans evolve better hearing at high frequencies.


----------



## reginalb

asuhra said:


> Argh!!!
> Just wrote a giant reply responding to everyone about the subliminals and re-sampling. Went to dinner, clicked preview before posting and everything disappeared. Just f.... great...
> Restarting, this time summarized.
> Subliminal Audios are done in the frequencies range between 15.5Khz and 20Khz because of the DAPs and DACs limits but that means you can only put so much info on the 44.1Khz samples because there is no more headroom. At 48Khz you have much more headroom to put info between the 15.5 and 20Khz Freq range when doing subliminals (it's a mathematical thing). I am guessing that's why 44.1Khz samples sound so much worst.
> ...


 
  
 Let's say that this is all true, just for the fun of it. 
  
 What's the point? A female Soprano tops out at like 1,200-1,300 Hz, I think Mariah's whistle register squeals are at like 3Khz. 
  
 You've got cymbals and piccolos that can produce some pretty high frequencies, but nowhere near 20Khz.
  
 So it might be that there are applications for capturing higher frequencies (and there are - some scientific research), but it's not music. If music isn't produced up there, why would you bother creating a device for playing back sound up there for sale to the general public? I don't listen to bats as a leisure activity, personally.


----------



## castleofargh

rrod said:


> castleofargh said:
> 
> 
> > when you look at albums, you usually have very little loudness in the ultrasounds compared to the rest of the music. and ultrasounds as physical pressure consciously felt on the skin requires a really loud signal (I think it was above 100db but you'd have to find the paper to get the value they found).
> ...


 

 you made me think of a new angle. ultrasounds work nicely to disinfect tools, maybe your room is cleaner when you listen to highres music? maybe adding a machine to create some mist would improve the process? is there a correlation between humidity levels and people thinking highres sounds better?


----------



## Roseval

reginalb said:


> You've got cymbals and piccolos that can produce some pretty high frequencies, but nowhere near 20Khz.


 
 Crash cymbals can go far in excess of 20 kHz
 http://www.drummerworld.com/forums/showthread.php?t=66957


----------



## old tech

roseval said:


> Crash cymbals can go far in excess of 20 kHz
> http://www.drummerworld.com/forums/showthread.php?t=66957


 
 And a dog whistle is around 21khz.  Either is moot is far as human hearing is concerned.


----------



## Roseval

old tech said:


> And a dog whistle is around 21khz.  Either is moot is far as human hearing is concerned.


 
 The post I responded to suggested that no musical instrument produces frequencies even near 20 kHz.
 That is simple not true.
  
 Frequencies above the upper treshold of our hearing we cannot hear by definition.
 But content above this threshold can generate IMD that maps into out audible range.


----------



## Ruben123

Idk about those imd etc but you sure need a microphone that records those ultrasounds and well many microphones don't actually do that AFAIK


----------



## spruce music

ruben123 said:


> Idk about those imd etc but you sure need a microphone that records those ultrasounds and well many microphones don't actually do that AFAIK


 

 Actually most condenser mics do.  They may have drooping response.  I have several that are only spec'd to 20 khz, but have substantial response to 30 khz or more before dying way off.
  
 You can do the old jangling key test.
  
 Get two or three metal keys on a keyring.  Jangle them in front of the mic and record at a high sample rate.  They don't sound very loud, but most of the output is ultrasonic.  You get something to at least the high 30 khz or circa 40 khz range that way.


----------



## sonitus mirus

ruben123 said:


> Idk about those imd etc but you sure need a microphone that records those ultrasounds and well many microphones don't actually do that AFAIK


 
  
 I think there is more to it than simply having a capable microphone.  It would require the instrument to produce ultrasounds, a microphone to record them, a media format capable of playing back these ultrasounds, transducers that could reproduce them with some kind of accuracy, and ears that could benefit from all of this.  If anything in this chain is a limiting factor, we are most likely left with a typical 20-20kHz frequency range.  If the recorded instrument is creating IMD that impacts the audible range, it is already being captured with a typical recording microphone.  The only thing to consider is if the actual ultrasounds are making any kind of difference for the listener, because any audible influence should already be present in the recording.  
  
 I would think that system/equipment generated noise would be more likely to occur on a 24/192 audio file than any sounds created from instruments with regards to ultrasounds.  I mean, Led Zeppelin probably wasn't using microphones that recorded ultrasounds, so any HD files probably only include recording equipment noise above 20kHz and not crash cymbal sounds from Bonham.


----------



## old tech

roseval said:


> The post I responded to suggested that no musical instrument produces frequencies even near 20 kHz.
> That is simple not true.
> 
> Frequencies above the upper treshold of our hearing we cannot hear by definition.
> But content above this threshold can generate IMD that maps into out audible range.


 
 Yes there are instruments that produce sound frequencies above 20khz, just as there are many consumer electronics and natural phenomena producing such frequencies around us all the time.
  
 But does it affect the frequencies within the audible range? Possibly yes but probably, maybe not.
  
 I can understand how supersonic or subsonic recorded content can introduce IMD, through interaction with system electronics and speakers.  But, using the dog whistle example, is this relevant to the live acoustic event?  If someone blow a dog whistle next to you while you are chatting with someone else, does it affect the tonality of the conversation or the background noise?


----------



## Roseval

> But does it affect the frequencies within the audible range? Possibly yes but probably, maybe not.


 
 Good point.
 Indeed IMD does exist but is it loud enough to become audible?
 I have my doubts.
 Scientific evidence is rare but this survey by Pras indicates that people are able to discriminate between 44 and 88. This is only the case with an orchestral recording.
 Maybe because of the IMD?
  
http://www.academia.edu/441305/Sampling_Rate_Discrimination_44.1_KHz_Vs._88.2_KHz
 (scroll down a little)


----------



## gregorio

fuzun said:


> [1] Also there are a lot of energy sources but how much are them mechanic? Due to natural selection, our body is very capable of perceiving mechanical energy only if it has enough power. Unlike electromagnetic energy, we really do not need any specific receptors to perceive mechanical energy because almost our entire body is somehow capable of interpreting mechanical stress (if they have enough power).
> [2] Beyond this point, scientific works needed.
> [3] Considering low levels of ultrasound found in music files due to cropping, bad(?) mastering(?), (I think) It would be impossible to perceive them even indirect form.
> [4] If you take hand skin (where energy reflects to epidermal hand cells), yes it needs big magnitude. But ear is very complex organ and there are different type of cells at where stereocilia located. Maybe these cells need less power than hand cells to notice it to brain via neurons?
> [5] But I think it like this: When you listen extreme bassy musics, you think like your brain is shaking (maybe not correct term but you got the point). That "shaking feeling" is not audio but it is a intended thing to experience while listening music. Because music makers got that feeling too.


 
  
 1. What do you mean by "mechanic" energy as being different to electromagnetic energy? Mechanic energy IS electromagnetic energy (if we're talking about sound waves).
 2. Why? There's no point as far as you are concerned, if you're going to ignore the science (which has already been done) and invent your own theories. And, there's no point as far as science is concerned because science has already done the work!
 3. No, it's not due to mastering (good or bad), it's due to the fact that's there's little there in the first place.
 4. Science has already done this work, in some cases many decades ago! Science knows the physiological structure of the ear and even what amounts of energy are needed to cause a response in those structures at various frequency ranges.
 5. You're free to think whatever you like but this is the science forum and science exists specifically to separate what people may think from the actual facts.
  


> Originally Posted by *Asuhra* /img/forum/go_quote.gif
> 
> [1] At 48Khz you have much more headroom to put info between the 15.5 and 20Khz Freq range when doing subliminals (it's a mathematical thing).
> [2] It has been proven by dog trainers that you can hear at least up to 30Khz because they got affected by the subliminals meant for their dogs @ 30Khz range. [2a] I have proven that on myself time and time again.
> [3] As for the resolution, that has also been proven that some people notice differences up till 96Khz.


 
  
 1. A "mathematical thing"? Don't you mean an anti-mathematical thing? The math/science actually demonstrates the exact opposite of what you're stating!
 2. Maybe you're confused? This is the science forum, NOT the dog trainers forum. 
 2a. There's only two options: 1. You've proven that you are not a homo sapien or 2. You've somehow managed to convince yourself but haven't proven anything. If you're going to dispute the accepted science you're going to need exceptional actual proof, not just your personal conviction.
 3. Again, science/math proves there is no more resolution, only the same resolution in a wider band of frequencies. Please provide a reference to the proof that people can tell any resolution difference between 96kHz and 44.1kHz with music recordings.
  
 Quote:


spruce music said:


> [1] Crash cymbals can go far in excess of 20 kHz ...
> [2] Actually most condenser mics do.  They may have drooping response.


 
  
 1. Yes but how much ultrasonic energy depends on where we record them, both in terms of the acoustic properties of the room and the distance the mic is from the cymbal. Concert halls, for example, are designed to absorb high frequencies and boost lower frequencies. Additionally, more high frequency energy is lost (absorbed by air molecules) than lower frequency energy, so the greater the distance from the sound source, the less high freq content is present. These two factors are cumulative.
 2. Yes but as you say, their response is not flat, they roll off significantly beyond 20kHz (except of course those condensers specifically designed to have a response higher than 20kHz). Additionally of course we have the common use of dynamic (rather than condenser) mics, which roll off earlier than 20kHz.
  


sonitus mirus said:


> But does it affect the frequencies within the audible range? Possibly yes but probably, maybe not.
> 
> If the recorded instrument is creating IMD that impacts the audible range, it is already being captured with a typical recording microphone. The only thing to consider is if the actual ultrasounds are making any kind of difference for the listener, because any audible influence should already be present in the recording.


 
  
 A. For the result of a frequency modulation to be audible, the original frequencies must also fall within the audible band. This is simple enough to test for yourself. With a 96kHz sample rate, use a tone generator to compare say a 1kHz sine wave and a 1kHz square wave, the tonal difference should be obvious. Try the same again but at 12kHz and now you can't hear the 12k square wave, it just sounds like two 12kHz sine waves! This is because an actual square wave cannot exist as an acoustic sound wave. A square wave has to be approximated by frequency modulating sine waves (a fundamental + odd-harmonics) but at 12kHz those odd-harmonics are beyond the audible frequency spectrum and therefore all you hear is the 12kHz fundamental (sine wave).
  
 B. This isn't the case with IMD however, which only occurs within electronic systems (rather than with acoustic instruments), when over-driven circuits can generate tones. The energy which caused the circuit to generate the IMD does not have to be within the hearing band but to be audible, the tones generated obviously have to fall within the hearing band (which, as you say, would then be recorded by a mic anyway). If that IMD product is going to modulate with the existing frequency content then to be audible it must fall within the audible band, as per point "A".
  
 G


----------



## gregorio

roseval said:


> Scientific evidence is rare but this survey by Pras indicates that people are able to discriminate between 44 and 88. This is only the case with an orchestral recording.
> Maybe because of the IMD? http://www.academia.edu/441305/Sampling_Rate_Discrimination_44.1_KHz_Vs._88.2_KHz
> (scroll down a little)


 
  
 I haven't had time to go through it in fine detail but there are a few things I noticed:
  
 1. Only 3 of the 16 participants demonstrated any ability to differentiate with a significance better than chance/guessing.
 2. If I read correctly, although there was significance in the results of those 3, they were actually wrong. They thought the 44.1kHz recording was the higher res version!
 3. To be sure one is actually testing what one is aiming to test, all variables other than the variable being tested need to be eliminated. This is considerably more difficult that it usually appears! In this study, the ADC's internal clock was used for the 44.1kHz recording while an external clock was used for the 88.1kHz recording. I'm not sure why they did this, external clocking usually adds jitter to the system. This could explain why the 44.1kHz recording was thought by the 3 to be the higher res version, although with most modern, high quality ADCs the level of additional jitter from external clocking should not normally be audible. This is still a variable which should have been eliminated though, as was using such different recorders.
 4. I didn't see where they explained their methodology for down-sampling. Certainly it was fairly common a few years ago that re-sampling software would reduce amplitude of the signal (to eliminate illegal intersample peaks at the high sample rates used for conversion). Generally the reduction was only about 0.2dB or so and most would not detect any difference, a trained listener, with certain types of audio material and certain listening environments might be able to. Volume matching is another variable which is more difficult in practice to eliminate than it may seem to be.
 5. Yes, I agree, IMD is another potential variable which I'm not sure was eliminated.
  
 G


----------



## fuzun

Who is ignoring science? There is no thing as ignoring science. If you don't believe gravity there is no gravity? The statements I have posted are *my opinions (some of them are incorrect)* and *questions *that I posted to *wide the discussion*.
  
 Can you post scientific materials or your opinions instead of arguing?
  
 "You're free to think whatever you like but this is the science forum and science exists specifically to separate what people may think from the actual facts." Whats the point of writing this? It was written to give an example in a indirect way.


----------



## sonitus mirus

gregorio said:


> I haven't had time to go through it in fine detail but there are a few things I noticed:
> 
> 1. Only 3 of the 16 participants demonstrated any ability to differentiate with a significance better than chance/guessing.
> 2. If I read correctly, although there was significance in the results of those 3, they were actually wrong. They thought the 44.1kHz recording was the higher res version!
> ...


 
  
 What I find most strange, provided I'm reading the data correctly, is that no difference was shown between 88.2kHz downsampled to 44.1kHz.  No difference was found between the downsampled 44.1kHz and the native 44.1kHz files.  But somehow a few individuals consistently managed to select the native 44.1kHz as the better sounding version to the 88.2kHz, or they heard a difference, statistically speaking, but were clearly unsure which file was the 88.2kHz.
  
 88.2kHz = file A
 88.2kHz downsampled to 44.1kHz = file B
 Native 44.1kHz = file C
  
 No difference found between files A and B or between files B and C, but there was a statistical difference found between files A and C. (with file C being chosen as the HD version more often)
  
 I don't get it.  Seems like a bad test or something wrong with the process somewhere. 
  
 Edit:  After typing it out, I suppose one could assume that perhaps file A is too close to the same as file B, and file B is too close to the same as file C, but there is somehow just enough of a difference to be identified between file A and C.  The results still seems a bit odd.


----------



## gregorio

sonitus mirus said:


> ... provided I'm reading the data correctly ...


 
  
 I've used this caveat as well and I presume for the same reason, I found the paper quite confusing. For example: "_This finding provides support for theories that high-resolution formats better reproduce the details of transients and room acoustics_" - Hang on a minute, as the statistically significant results were actually wrong, doesn't that suggest the exact opposite, that 44.1 better reproduces the details? Then there's the apparently illogicality of the results which you highlighted, which could conceivably just be a quirk/fluke of statistics, although by definition, probably not!  Then there's the weaknesses in methodology, the failure to eliminate other potential variables, some of which I mentioned. And additionally, some fairly obvious factual errors in the introduction.
  


fuzun said:


> Who is ignoring science? There is no thing as ignoring science. If you don't believe gravity there is no gravity?


 
  
 Of course one can ignore the science. If you don't believe in gravity, there is of course still gravity, however one could attribute the effects of gravity to say a magic spell cast by a witch, rather than to the scientific law of gravity. Theorising about the witch and then posing questions about those theories IS ignoring the science!
  


fuzun said:


> Can you post scientific materials or your opinions instead of arguing?


 
  
 There are no scientific materials regarding the witch and my opinion is that she doesn't exist, so no, apart from just not responding, I see no alternative to disputing/arguing. Let's get away from silly metaphors and take one of your actual examples, "_And we would like to listen music how it was intended to sound. (Mastering, ... not involved. This is only in theory.)_" - That's a contradiction which makes no sense. The whole point of mastering is to affect/change the studio mix so that consumers can "listen to the music how it was intended to sound". Without mastering, the only way of achieving that aim would be to listen to the music in the studio in which it was mixed, which obviously is not practical and why mastering was invented in the first place!
  
 G


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## akg fanboy

Hey everyone, I don't know if this is the right thread for this, but I do think higher quality (16.flac/24.flac) music do have a purpose for normal people who listen to their music for recreational purposes, *even though* most people cannot hear the difference through normal means.
 I think higher quality music works better with equalization and has less distortion, at least according to the distortion my ears heard. My 256/320 mp3s generally exhibit more noise and distortion than my lossless 16 bit flac music which sound much cleaner. Basically, I normally cannot tell the difference very well between my 320 mp3s and flacs unless I EQ them. Is this just a coincidence with better recording/mastering or can my findings actually be confirmed to be true?


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## castleofargh

it could depend on the EQ you're applying. admittedly if you end up with some freqs real close to 0db or even slightly clipped, then mp3 could have some/more clipping than the flac file. and of course that can be more audible. that due to intersample clipping. if the signal is consistently at a good -3db, then you would avoid that problem IMO.
  
 some nice peak meter could show that(and they will by oversampling, so that the peaks are closer to the real peaks). if you can get your problem/difference while no peak meter can show potential clipping, then I'd be curious to see a short sample of the 2 versions.
  
 otherwise, usually DSPs that really benefit from oversampling will just do so and then downsample back to whatever you originally used on the fly.


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## Warmuser

Hi everybody, can you explain me If this album is a true fake? The frequences looks like the .mp3 type. So I want to understand if these album is a fake or I'm wrong. Thanks
  
 Here I made the screens of any track: http://imgbox.com/g/0AnaQljpwf


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## akg fanboy

castleofargh said:


> it could depend on the EQ you're applying. admittedly if you end up with some freqs real close to 0db or even slightly clipped, then mp3 could have some/more clipping than the flac file. and of course that can be more audible. that due to intersample clipping. if the signal is consistently at a good -3db, then you would avoid that problem IMO.
> 
> some nice peak meter could show that(and they will by oversampling, so that the peaks are closer to the real peaks). if you can get your problem/difference while no peak meter can show potential clipping, then I'd be curious to see a short sample of the 2 versions.
> 
> otherwise, usually DSPs that really benefit from oversampling will just do so and then downsample back to whatever you originally used on the fly.




So you are saying the frequencies should be at 0dB on the frequency response chart? My music player does show frequencies in real time.

Since DSPs benefit from oversampling, does that mean a 24/192khz file that is already "oversampled" compared to a 16/44 file work better for the DSP? I feel like a true 24 bit music file would have more headroom for equalization with less quality degredation but I would have to find one


----------



## castleofargh

I mean 0db as in digital value, the maximum loudness you can record on a digital PCM file. the music is made of sample points with values all below 0db. a 16bit file goes from 0 to -96db, and a 24bit file goes from 0 to -144db. 0 is always the upper limit you can record on the digital support. and when you reach 0 or try to get over it, a digital system can only register up to 0, so it replaces everything above 0 with 0. that's clipping.
 because the samples registered won't always fall exactly at the peak of the signal, you can end up with properly recorded digital values(below 0db), but actual peaks that get above it. look up "intersample clipping" online, it's really not a new problem, but one that can be solved soooo easily.
  
  
  if something works better at a higher sample rate, then as I said above, it will just convert the file on the fly to that sample rate. do you know what sample rate that is for each DSP you own? I don't, so I don't get how I could improve anything by oversampling in advance.
  
 whatever you try, check some peak meter(hopefully one that does oversample a good deal to show intersample peaks at a reliable value), and verify first that you're not clipping anything with or without your EQ on the songs that feel wrong to you. if that is really out of the way, then we can look for other reasons why you get what you describe.


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## Roseval

> Since DSPs benefit from oversampling, does that mean a 24/192khz file that is already "oversampled" compared to a 16/44 file work better for the DSP? I feel like a true 24 bit music file would have more headroom for equalization with less quality degredation but I would have to find one


 
 I wonder if you are mixing up sample rate and bit depth.
  
 A true 24/192 file is not oversampled but simply recorded with a word length of 24 bits and a sample rate of 192. This means it has a dynamic range of 144 dB and the highest possible frequency is 96 kHz (1/2 Fs) 
 CD players use oversampling (8 times)
 If you don't, the first alias starts right after 1/2 Fs (22 kHz)
 You need a very sharp (brick-wall) filter to get rid of it.
 If you over-sample 8 times (353 kHz), the first alias is at 176 kHz so you can use a very smooth filter.
  
 To the best of my knowledge DSP doesn't profit by increasing the sample rate.
 In fact a lot of DSP chips are limited to 48 kHz and the more expensive to 96 kHz simply because the higher the sample rate the shorter the time one have to perform the calculations.
 In case of 44.1 you have 0.00002 of a second to do all the calculations.
  
 DSP does profit by increasing the precision. Doing all calculations with 32 or better 64 bit keeps down the quantization error. 
  
 As DACs only accept integers, as DSP uses a precision much higher that 16 or 24 integer, in the end any DSP action has to be dithered.
 Here a 24 bit recording has the advantage as with  16 bits the dither is at -96 dBFS but with a 24 at -144 dBFS
 You have to play FFF loud to make something at -144 audible if possible at all.
  
 To sum up, DSP is about doing calculations
 This has nothing to do with sample rate except that sample rate might be a limiting factor (time)
 This has all to do with precision, the more bits the better.


----------



## drtechno

thesuperguy said:


> If both formats are lossless, what differentiates the 2 versions in terms of sound quality if at all?


 
 well the 24 bit has better signal to noise when recording.  I wish it was 40 bit like the digital audio workstation programs (but for a different reason).


----------



## akg fanboy

Ah okay, I am not the most knowledgeable in this subject. So I am assuming we are talking about how 24 bit has a higher dynamic range so therefore it should theoretically clip less when EQing if I were to exceed that 144dB threshold? I don't listen to my music very loud, usually just 8 or 9 o'clock low gain on my matrix mstage hpa2 classic (the dial starts at 6 o' clock). Most metal songs I listen to had an average dynamic range of 15-20dB on my music analyzer plugin with around 30dB peaks. The highest dynamic range I've gotten out of a track was around 52dB peak for a classical piece. Neither of them are exceeding the threshold of a 16 bit music file if I am not mistaken. I think maybe compression has something to do with the clipping and a song that uses compression will have a greater chance of clipping even if it does not surpass 96dB. And where do I find a peak meter? My music player is jriver which allows plugins


----------



## drtechno

akg fanboy said:


> Ah okay, I am not the most knowledgeable in this subject. So I am assuming we are talking about how 24 bit has a higher dynamic range so therefore it should theoretically clip less when EQing if I were to exceed that 144dB threshold? I don't listen to my music very loud, usually just 8 or 9 o'clock low gain on my matrix mstage hpa2 classic (the dial starts at 6 o' clock). Most metal songs I listen to had an average dynamic range of 15-20dB on my music analyzer plugin with around 30dB peaks. The highest dynamic range I've gotten out of a track was around 52dB peak for a classical piece. Neither of them are exceeding the threshold of a 16 bit music file if I am not mistaken. I think maybe compression has something to do with the clipping and a song that uses compression will have a greater chance of clipping even if it does not surpass 96dB. And where do I find a peak meter? My music player is jriver which allows plugins


 
 well it isn't clipless. back when protools and nuendo came out The mixing enviroment was 24 bit. it didn't take long for then to change the mixing environment to 32 bit because things clip too easily when mixing inside the program. A few years ago the environment went to 40 bit.
  
 when someone likes me mixes, we create the enviroment. Nothing really has changed here compared to the days of an analog desk and tape drives.
 Then it goes to the master gain shredder (mastering engineer).
  
 Sound card using consumer audio drivers have about 6 db of headroom on the master. On asio drivers there is no extra headroom.
  
 btw when the audio music is transferred to cd media,  they bring it out to analog and clip the input of a modified converter. The signal is more treated as a percentage in a deflection scale than a audio signal. 
  
 oops I need to add some more:
  
  
 ok I don't know where you are measuring, but any readings you are getting after conversion will vary depending on your converter.
 btw those analyser plugins are not intended for measurements. They are not accurate. maybe you should try a piece of pro audio software. This one was originally written by Sony: http://www.magix-audio.com/us/spectralayers-pro/?utm_source=sonycreativesoftware&utm_medium=referral&utm_campaign=redirect&lang=en&prdt=spectralayerspro


----------



## gregorio

roseval said:


> [1] To the best of my knowledge DSP doesn't profit by increasing the sample rate.
> [2] DSP does profit by increasing the precision. Doing all calculations with 32 or better 64 bit keeps down the quantization error.


 
  
 1. Generally that's true, although there are some DSPs out there which do. Some non-linear DSPs such as some compressors and modelling plugins for example. Also, it very much depends on the programming of the DSP and any design considerations/limitations. I have come across DAW plugin DSPs which sounded better operated at 96kHz than at 44.1kHz, simply because that was the sample rate which seems to have attracted the most programming effort. That doesn't appear to be quite as common today as it was a few years ago and probably doesn't affect many consumer DSPs.
  
 2. True but only up to a point. In DAWs, the cumulative quantisation error of successive DSPs can theoretically be an issue but we'd have to use quite a few dozen DSPs to get to that stage with 24bit.
  


drtechno said:


> [1] well the 24 bit has better signal to noise when recording.  [2] I wish it was 40 bit like the digital audio workstation programs (but for a different reason).
> 
> [3] well it isn't clipless. back when protools and nuendo came out The mixing enviroment was 24 bit. it didn't take long for then to change the mixing environment to 32 bit because things clip too easily when mixing inside the program. A few years ago the environment went to 40 bit.
> 
> [4] btw when the audio music is transferred to cd media,  they bring it out to analog and clip the input of a modified converter.


 
  
 1. 24bit doesn't have a signal to noise ratio, it just has a lower noise floor than 16bit. This lower noise floor does not directly affect the SNR when recording, as the SNR is defined by the mics, mic-preamps and the SNR of what's being recorded and where.
  
 2. You could make it 512bit if you want, it still won't affect the SNR though.
  
 3. Why are you coming here and just making stuff up? ProTools was 16bit when it came out, then it's mixing environment changed to 56bit about 20 years ago (Protools TDM), the cheaper LE versions of Protools started at 32bit float and all today's versions are 64bit float. Nuendo started at 32bit float and is now 64bit float. No current DAWs I'm aware of have a 40bit mix environment.
  
 4. No they don't! I'm sure the odd person might have done what you're suggesting, years ago, but it's extremely atypical today!
  


akg fanboy said:


> [1] So I am assuming we are talking about how 24 bit has a higher dynamic range so therefore it should theoretically clip less when EQing if I were to exceed that 144dB threshold? [2] The highest dynamic range I've gotten out of a track was around 52dB peak for a classical piece. Neither of them are exceeding the threshold of a 16 bit music file if I am not mistaken.


 
  
 1. No, you're looking at this backwards. 0dBFS is the maximum level a digital audio file can have and this is exactly the same maximum with 24bit as with it is with 16bit. The added dynamic range of 24bit is at the quietest end of the scale, not the loudest. So you are just as likely to clip at 24bit as at 16bit when EQ'ing. Commercial music releases are optimised to 0dBFS (or just fractionally below), so you will need to lower the (digital) level of the file before adding EQ, otherwise you are likely to exceed 0dBFS (clip).
  
 2. No, you're not mistaken and those figures look pretty representative. CD has a theoretical maximum dynamic range of 96dB (-96dBFS to 0dBFS), so your classical recording is actually using about 9 of the available 16bits, meaning you could lower the digital level of your 16bit file by about 40dB before you start hitting the digital noise floor.
  
 G


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## drtechno

"
  
 1. 24bit doesn't have a signal to noise ratio, it just has a lower noise floor than 16bit. This lower noise floor does not directly affect the SNR when recording, as the SNR is defined by the mics, mic-preamps and the SNR of what's being recorded and where.
 "
  
 Obviously you don't realize ADCs still fallow the rules of analog up till the threshold of the digital domain.
 So the ADC converter chips have a signal to noise product depending on signal level.
 Of course you can look at most ADC chips and they have them stated in their data sheets.
  
   
 

 3. Why are you coming here and just making stuff up? ProTools was 16bit when it came out, then it's mixing environment changed to 56bit about 20 years ago (Protools TDM), the cheaper LE versions of Protools started at 32bit float and all today's versions are 64bit float. Nuendo started at 32bit float and is now 64bit float. No current DAWs I'm aware of have a 40bit mix environment.
  
 the audio is in 40 bit. the program itself might be 64 bit but the digital audio is 40 bit.
 Obviously, you don't realize this.
  
 btw I'm still out there in the industry.  
  
  
 I just don't work at that publisher anymore.


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## gregorio

drtechno said:


> [1] Obviously you don't realize ADCs still fallow the rules of analog up till the threshold of the digital domain.
> [2] the audio is in 40 bit. the program itself might be 64 bit but the digital audio is 40 bit.
> Obviously, you don't realize this.
> 
> [3] btw I'm still out there in the industry.


 
  
 1. Why "obviously"? ADCs must follow the laws of physics and due to thermal noise, this resultant noise floor creates a theoretical limit of about 21bits of dynamic range. In practice most pro ADCs achieve considerably less dynamic range than 21bits. However, this is irrelevant because that's still roughly 100 times more dynamic range than can practically be achieved in real world acoustics (noise floors of recording environments) or in the analogue chain prior to the ADC (mics/preamps). This is where the SNR is defined, not by the ADC, as already mentioned!
  
 2. You're right, I obviously don't realise that, just as I don't realise that the earth is flat or that the moon is made of cheese! Just repeating a figure you have made up does not make it more convincing! The digital audio file/s themselves are usually 24bit or 16bit (or occasionally 32bit float) and is independent of the mix environment within the DAW, the bit depths of which I have already given you for the two DAWs you mentioned. 40bit is a number you've just made up! If you wish to prove you're not just making it up it's simple, just provide the information from Stienberg or Avid/Digidesign.
  
 3. There is no way that even an apprentice audio engineer would not know points #1 and #2 above, even after just a few months in the industry, let alone 25 years! Which means either you're lying/trolling or, as mentioned, you are deliberately trying to mislead and have some other role in the industry which is not that of an audio engineer, of which you obviously have little/no actual knowledge.
  
 Lying about (or at least misrepresenting) your position in the industry and therefore your level of knowledge/understanding is troubling. More troubling still, is that you're just making stuff up and passing it off as fact. This is troubling because this is the science forum, do you honestly believe we're all dumb enough to just swallow whatever BS you fancy making up? Even if you do believe we're all dumb enough, what do you hope to actually achieve? Is this just some rather sad attempt at self-aggrandisement or are you leading up to a dodgy marketing strategy of some product you're trying to sell?
  
 G


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## fuzun

Maybe this discussed a while ago but I could not find.
  
 We know that our devices which is in audio reproduction chain are not perfect. Any of them. So some sound gets distorted.
  
 This is a straight (maybe a bad) example but I am sure you will get this:

  
 24 bits for not in perfect system may be redundant for us for listening purposes but in real world?
  
 Bits are just example. Consider the amount of information stored in files. According to Nyquist theorem we dont need more than 40khz for listening but does not that applies for a perfect situation with zero entropy?


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## spruce music

I agree most of Dr. Techno's posts seem either ill informed or out of date.  I also wonder if English is not his native language and we have a language barrier.  If so this is more a case of miscommunication instead of something nefarious.
  
 Digital mixers (as in hardware mixers) would commonly run 40 bit float.  Especially those that used SHARC processors for various DSP effects. Perhaps that is what he has in mind. 
  
 I am all for calling out bad info in the Sound Science forum, but maybe we should try a bit less confrontational approach to find out if we are hearing what Dr. Techno is actually saying first.
  
 A few examples of current hardware digital mixers that do the DSP in 40 bit float:
  
 http://ams-neve.com/88d/
  
 http://www.fullcompass.com/prod/254390-Midas-M32
  
 https://www.bhphotovideo.com/c/product/791831-REG/Behringer_X32_X32_32_Channel_16_Bus_Total.html
  
 http://www.stagetec.com/en/audio-mixing-consoles/crescendo/specifications.html
  
 http://www.adorama.com/phacapela16.html
  
 http://www.soundcraft.com/products/vi7000


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## drtechno

spruce music said:


> I agree most of Dr. Techno's posts seem either ill informed or out of date.  I also wonder if English is not his native language and we have a language barrier.  If so this is more a case of miscommunication instead of something nefarious.
> 
> Digital mixers (as in hardware mixers) would commonly run 40 bit float.  Especially those that used SHARC processors for various DSP effects. Perhaps that is what he has in mind.
> 
> ...


 

 you forgot all those semi-pro interfaces (firewire,usb,thunderbolt) that has mixer built in them too (granted they are mostly used during tracking for direct monitoring).
  
 I can't upload a datasheet. but here is one  of the converter chips that is out there and its "analog characteristics"
  
 pg 7 of AK5394AVS which is here:  https://www.akm.com/akm/en/file/datasheet/*AK5394AVS*.pdf


----------



## drtechno

btw you guys assume too much


----------



## gregorio

spruce music said:


> Digital mixers (as in hardware mixers) would commonly run 40 bit float.  Especially those that used SHARC processors for various DSP effects. Perhaps that is what he has in mind.


 
  
 We don't need to guess what he has in mind, he was very specific. He not only talked about DAWs (rather than hardware mixers) but two specific DAWs, ProTools and Nuendo:
  
 "_back when protools and nuendo came out The mixing enviroment was 24 bit. it didn't take long for then to change the mixing environment to 32 bit because things clip too easily when mixing inside the program. A few years ago the environment went to 40 bit._" - None of this is correct!
  
 G


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## drtechno

gregorio said:


> We don't need to guess what he has in mind, he was very specific. He not only talked about DAWs (rather than hardware mixers) but two specific DAWs, ProTools and Nuendo:
> 
> "_back when protools and nuendo came out The mixing enviroment was 24 bit. it didn't take long for then to change the mixing environment to 32 bit because things clip too easily when mixing inside the program. A few years ago the environment went to 40 bit._" - None of this is correct!
> 
> G


 
  
  
 If you got to harp on things I am recalling from memory, then you need to prove I'm wrong, or be quiet. Not complain or make statements you can't back up.


----------



## drtechno

gregorio said:


> 1. Why "obviously"? ADCs must follow the laws of physics and due to thermal noise, this resultant noise floor creates a theoretical limit of about 21bits of dynamic range. In practice most pro ADCs achieve considerably less dynamic range than 21bits. However, this is irrelevant because that's still roughly 100 times more dynamic range than can practically be achieved in real world acoustics (noise floors of recording environments) or in the analogue chain prior to the ADC (mics/preamps). This is where the SNR is defined, not by the ADC, as already mentioned!
> 
> 2. You're right, I obviously don't realise that, just as I don't realise that the earth is flat or that the moon is made of cheese! Just repeating a figure you have made up does not make it more convincing! The digital audio file/s themselves are usually 24bit or 16bit (or occasionally 32bit float) and is independent of the mix environment within the DAW, the bit depths of which I have already given you for the two DAWs you mentioned. 40bit is a number you've just made up! If you wish to prove you're not just making it up it's simple, just provide the information from Stienberg or Avid/Digidesign.
> 
> ...


 
 well you've got lots to learn then


----------



## gregorio

drtechno said:


> If you got to harp on things I am recalling from memory, then you need to prove I'm wrong, or be quiet. Not complain or make statements you can't back up.


 
  
 "_The Pro Tools mix engine has traditionally employed *48-bit fixed point arithmetic*, but floating point is also used in some cases, such as with Pro Tools HD Native. The new HDX hardware uses 64-bit floating point summing._" - Wiki
  
 "_Mixing level scaling stores 48-bit results using a *56-bit* accumulator for maximum precision._" - 1999. Pro Tools 5.0.1 Reference Guide (page 283 explaining the TDM mix architecture, introduced in 1994).
  
 "_Steinberg has labeled Nuendo its "Media Production System" and with good reason. Open the program, click "New Project," and there's a list of various application templates ranging from Pro Logic Video Mixdown and 24/96 DVD 5.1 Authoring to *32-bit* Stereo Master and Audio/MIDI Music Production._" - Mix magazine review of initial Nuendo release.
  
 OK, I've provided some back up for my statements, now it's your turn! Provide corroborating info that Nuendo ever had a 24bit mix environment, that Pro Tools or Nuendo ever had a 40bit mix environment. Of course, you won't be able to do that! Even a relative newbie recording, mix or mastering engineer knows that Nuendo is a host based platform (and can therefore can't be anything other 32 or 64 bit mix environment), let alone someone who's been in the industry for 25 years! It's also inconceivable you wouldn't know the basic architecture of the industry standard DAW software (Pro Tools). Likewise, one might have to explain to a 1st year music technology student that at least 3 of the 24bits cannot be anything other than thermal (Johnson) noise and therefore that a 40bit file format with an additional 16bits of thermal noise is ridiculous but one would be shocked to have to explain that to practising professional, let alone a 25 year veteran!
  
 For this (and several other reasons), your pre-emptive excuse: "I am recalling from memory" is hogwash. There's only one way that someone professing to be a highly experienced professional audio engineer could "recall from memory" something which so obviously flies in the face of basic digital audio theory, and that's if they have no understanding of basic digital audio theory and must therefore be lying about being a highly experienced audio engineer! What I find baffling is that even after your lie was exposed, you continue to defend your "facts" as not just made up and challenge those of us who have in fact been pro audio engineers for 25 years (or more). Regardless of why you decided to post made up "facts" in the science forum and regardless of whether  you're willing to publicly admit it, you (and many/most of us) know that you made them up and you must surely realise that continuing to defend those "facts" will achieve nothing besides digging a deeper hole for yourself!
  
 My apologies to others if my challenging of this member seems overly harsh but there's already way too much made up BS from marketers/retailers in the consumer audio world, without having to waste time dealing with someone just making up BS for self-aggrandisement and/or the fun of misleading others!
  
 G


----------



## drtechno

gregorio said:


> "_The Pro Tools mix engine has traditionally employed *48-bit fixed point arithmetic*, but floating point is also used in some cases, such as with Pro Tools HD Native. The new HDX hardware uses 64-bit floating point summing._" - Wiki
> 
> "_Mixing level scaling stores 48-bit results using a *56-bit* accumulator for maximum precision._" - 1999. Pro Tools 5.0.1 Reference Guide (page 283 explaining the TDM mix architecture, introduced in 1994).
> 
> ...


 
 basic audio theory don't cover programs. and the theory of 16 extra bits of thermal noise is flawed. but the 40 bit is unpublished info I got from a protools daw programmer that programmed may daws besides protools.
  
 that part doesn't matter. but if you really want to know about value spacifics , A 64-bit engine is technically not better than a 32-bit engine *for summing*.

 A 32-bit engine does all calculations on the fpu, thereby using registers that can be as large as 80 bit. (The accumulator even extends this)
 A 64-bit engine has no other choice than using SSE instructions, of which the registers are limited to 64 bit strings. Some people will argue that SSE strings are 128 bit, but that's incorrect. The 128 bit string is split up in two 64 bit strings.

 So, if well written, a 32-bit audio engine can be more precise than a 64 bit engine, since it can hold larger strings of numbers.
 That is when the numbers are kept as long as possibly can in the registers and are brought back to 32-bit as less as possible.
 On the other hand, 64-bit is more precise than 32-bit thereby avoiding any rounding errors. But these rounding errors are neglectable when it comes to adding and subtracting numbers, which is what a summing engine basically does.
 Rounding and additive (cumulative) errors are important for any process where DSP is concerned, but all DAW and plugin manufacturers are already using double precision for most -if not all- of their critical calculations.

 In practice 32 bits vs. 64 bit will make no difference at all, unless you are really abusing the audio engine and are burning every gain stage by dozens of dB's and/or in scientific tests set up to expose the problem. In every day use (read: proper use) there shouldn't be any difference.

 In other words, if you can hear (and measure) the difference between a 32-but engine and a 64-bit engine, then something is/was wrong (badly designed) with the 32-bit engine to start with.
 In a properly build DAW, the numbers are not stored in 32-bit strings, but kept in the FPU registers as long as possibly can before they are dumped (and rounded) to a 32-bit string. Thereby the intermediate roundings (80 bits =>32 bits) are much less frequent, than when using SSE strings which need to be dumped as soon as the 64bit strings are "full".
 Of course, depending on the internal structure of each an every DAW -and the way things are tested- there are arguments in favor of both techniques. All I said is that in normal, proper, non-abusive use, the (rounding) error in a 32-bit mixer will remain in the LSB's, exactly as in a 64bit mixer.
  
  
  
 some other fun facts:
  
 8-bit = 256 Dec.
 100000000 Bin.
 Dynamic Range = 48.16dB
 16-bit = 65536 Dec.
 10000000000000000 Bin.
 Dynamic Range = 96.32dB
 20-Bit = 1048576 Dec.
 100000000000000000000 Bin.
 Dynamic Range = 120.4dB
 24-Bit = 16777216 Dec.
 1000000000000000000000000 Bin.
 Dynamic Range = 144.48dB
 32-Bit = 4294967296 Dec.
 100000000000000000000000000000000 Bin.
 Dynamic Range = 192.64dB
 64-Bit = 18446744073709551616 Dec.
 10000000000000000000000000000000000000000000000000000000000000000 Bin.
 Dynamic Range = 385.28dB
 ********************************


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## gregorio

drtechno said:


> basic audio theory don't cover programs. and the theory of 16 extra bits of thermal noise is flawed.


 
  
 You asked for and received evidence that your statements were false. I've asked for you to provide evidence to support your statements but your only response is just to make up more ridiculous "facts", more ridiculous even than the "facts" you've already made up! If digging a deeper hole for yourself is in fact what you're trying to achieve, then you're doing an excellent job.
  
 Back up your statements with some corroborating evidence. Put up or shut up!!
  
 G


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## drtechno

gregorio said:


> You asked for and received evidence that your statements were false. I've asked for you to provide evidence to support your statements but your only response is just to make up more ridiculous "facts", more ridiculous even than the "facts" you've already made up! If digging a deeper hole for yourself is in fact what you're trying to achieve, then you're doing an excellent job.
> 
> Back up your statements with some corroborating evidence. Put up or shut up!!
> 
> G


 
 interesting. since I copied and pasted that reply from a different forum.
 because I couldn't said it better myself.
 https://www.gearslutz.com/board/steinberg-cubase-nuendo/360368-cubase-5-summing-why-not-64-bit-floating.html
 In my opinion,  the 16 bit extra noise theory you have is flawed.
 do you have some findings to show to back up this theory?


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## gregorio

drtechno said:


> [1] interesting. since I copied and pasted that reply from a different forum.
> because I couldn't said it better myself.
> https://www.gearslutz.com/board/steinberg-cubase-nuendo/360368-cubase-5-summing-why-not-64-bit-floating.html
> [2] In my opinion,  the 16 bit extra noise theory you have is flawed.
> [3] do you have some findings to show to back up this theory?


 
  
 1. I didn't ask for a link to one side of another argument on another forum from unattributed amateurs, students or newbies, I asked for supporting evidence, which you have still not supplied! do you know what "evidence" is? That hole is just getting deeper!
 2. You've already demonstrated a willingness to just make up "facts", so why should your opinion be anything other than worthless, unless you can convincingly corroborate it? That's going to be a very tall order though as it flies in the face of the known science. You do realise this is the science forum?!
 3. You can look up Johnson Noise for yourself. You refuse to back-up any of your laughable theories (#2) but you're asking for even more evidence from me, to back up knowledge which any experienced audio engineer should already know anyway! Surely that hole is deep enough already?
  
 BTW, your added/edited bits:
  
 1. "_but the 40 bit is unpublished info I got from a protools daw programmer that programmed may daws besides protools._" - So now you're progressing beyond just making stuff up to making up attributions for the stuff you're making up! I've personally known a number of Pro Tools programmers, including two heads of program development. Pro Tools has never been 40bit and has never even planned a 40bit architecture, prove otherwise!
 2. "_In practice 32 bits vs. 64 bit will make no difference at all, unless you are really abusing the audio engine and are burning every gain stage by dozens of dB's and/or in scientific tests set up to expose the problem. In every day use (read: proper use) there shouldn't be any difference._" - That's not necessarily true, which you'd know if you were an audio engineer! There are applications where it could make a difference, film mixing for example, where up to 1,000 or so channels of audio are being mixed.
 3. Even if we accept that 32bit is indistinguishable from 64bit when mixing (which it is in music), how does that support your argument? What has mixing environments got to do with distribution formats and what benefit would there be to a 40bit distribution format? When I asked for supporting evidence, I meant evidence to support your argument, not mine!
 4. What has the dynamic range of 32bit or 64bit fixed (integer) got to do with anything? It's not connected to your assertions on mix environments and it's certainly not recordable or reproducible. Maybe you're confusing float and integer bit depths? Which is not something an experienced audio engineer would do!
  
 Again, some actual evidence please. I'll change the choice to something more applicable: Put up, shut up or appear to be an idiot!
  
 G


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## Arpiben

For those interested and not to bring more agitation or thermal noise to the actual debate.

Thermal noise or Johnson - Nyquist noise as a value of: - 131 dBm at room temperature & with 20kHz bandwidth.
Something in between 21/22 bits in digital coding.
Rgds.


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## gregorio

arpiben said:


> Thermal noise or Johnson - Nyquist noise as a value of: - 131 dBm at room temperature & with 20kHz bandwidth.
> Something in between 21/22 bits in digital coding.
> Rgds.


 
  
 I think the exact value of the noise would depend on the exact construction/components of the individual ADC, although your figure does appear quite reasonable/representative. In practical implementation, I can't think of any pro ADCs off the top of my head which actually achieve 21bits, although several achieve between 20 and 21bits.
  
 G


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## castleofargh

I see that as a best case scenario, where a great many other noises can come over it. it's significant of what can be achieved as a limit. I don't imagine some 100% superconductor ADC in liquid nitrogen to be sold anytime soon. ^_^


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## Arpiben

gregorio said:


> I think the exact value of the noise would depend on the exact construction/components of the individual ADC, although your figure does appear quite reasonable/representative. In practical implementation, I can't think of any pro ADCs off the top of my head which actually achieve 21bits, although several achieve between 20 and 21bits.
> 
> G


 
  
 As far as I remember, you are right;

there are other noises` contribution: Shot Noise, Flicker Noise, etc...
the theoritical value depends on ADC`s types/components.
  
 Regarding the Effective Number Of Bits it can be aproximated by:
  
  

  
  
 where *SINAD*: Signal to (Noise Adding Distorsion) Ratio
 when dealing with sine wave.
  
  
 Forgetting maths,my purpose when I posted a typical thermal noise value was more for giving an idea of one of the *analog domain*`s limitation.
 In *digital domain*, some DAC manufacturers are providing  Noise Floor values around -300 dB or FFT Jitter around -180 dB.
 Therefore, I admit that it is sometimes easy to mix both domains and forget the limitations of analog one


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## drtechno

well the thermal noise is not really a big player in the noise generated by the ADC its the noise generated by the shift register mixing the resistor network value and the clock signal. This noise is a major variable in the overall quantization noise, and it has a name call spious noise. Granted most of this noise is supposed to be suppressed by the input circuit. But like all filter circuits. they are never perfect. That is why certification authorities measure the  amount of this type of rf leakage coming out from the input of a converter.


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## castleofargh

I removed the last few posts, TOS doesn't allow personal attacks.


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## gregorio

castleofargh said:


> I removed the last few posts, TOS doesn't allow personal attacks.


 
  
 Fair enough, it was getting a bit over heated. Lets do this properly then:
  


drtechno said:


> well the thermal noise is not really a big player in the noise generated by the ADC its the noise generated by the shift register mixing the resistor network value and the clock signal. This noise is a major variable in the overall quantization noise, and it has a name call spious noise.


 
  
 Clock signal error and the noise it generates is not called spurious noise, it's called "jitter noise". A good professional ADC will have jitter in the few tens of pico second range and produce jitter noise down around the LSB or in some cases even lower. Below is the jitter spectrum of the roughly decade old Prism Orpheus, showing it's jitter noise (with a 1kHz input signal) is below -140sBFS, which is several times lower in level than the thermal noise and therefore NOT a major variable or "player" relative to thermal noise!
  

 (Hugh Robjohns, 2010. "_Does Your Studio Need A Digital Master Clock?_", SoundonSound.)
  
 Being the science forum, if you are going to make statements of fact or refute the statements of others, you MUST back up your arguments with some convincing evidence, otherwise it will be viewed as nothing more than the made-up nonsense of a shill or troll. Posting deliberately made-up nonsense is an insult to this forum, it's guests and members. In order to receive politeness and respect here, you must obviously treat this forum and it's members with politeness and respect, NOT with contempt!!!
  
 G


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## Arpiben

Dealing with *Jitter* and its different aspects in digital audio I warmly recommend some Julian Dunn's documents, such as:
  
http://www.audiophilleo.com/zh_hk/docs/Dunn-AP-tn23.pdf.
http://www2.electron.frba.utn.edu.ar/~jcecconi/Bibliografia/13%20-%20Medicion%20de%20Amplificadores/Documentos/AudioPrecision_AN5_DigitalAudioMeasurement.pdf
  
 Despite the fact they were wriiten around 2000, they are providing a very comprehensive way of jitter issues.


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## drtechno

Thanks for the post Arpiben. Yes clocking  is very important as that was drilled into my head by Mr. Ludwig and Mr. Williams about 20 years ago to use an external clock.


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## nick_charles

arpiben said:


> Dealing with *Jitter* and its different aspects in digital audio I warmly recommend some Julian Dunn's documents, such as:
> 
> http://www.audiophilleo.com/zh_hk/docs/Dunn-AP-tn23.pdf.
> http://www2.electron.frba.utn.edu.ar/~jcecconi/Bibliografia/13%20-%20Medicion%20de%20Amplificadores/Documentos/AudioPrecision_AN5_DigitalAudioMeasurement.pdf
> ...


 
  
 Dunn's papers are very informative and definitely worth reading, the only thing that his and Hawksford's (also highly readable) papers lack is that, they do not address audibility of jitter except as a mathematical model based on rather extreme sound pressure levels. To date we have little (well none really)  empirical evidence that jitter as found in semi-competent digital components is actually audible, of course that does not stop a cottage industry in jitter reducers....


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## spruce music

drtechno said:


> Thanks for the post Arpiben. Yes clocking  is very important as that was drilled into my head by Mr. Ludwig and Mr. Williams about 20 years ago to use an external clock.


 

 I'll repeat the link from gregario's post.
  
 http://www.soundonsound.com/techniques/does-your-studio-need-digital-master-clock

 Rethink that external clock idea.


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## old tech

drtechno said:


> Thanks for the post Arpiben. Yes clocking  is very important as that was drilled into my head by Mr. Ludwig and Mr. Williams about 20 years ago to use an external clock.


 
 And I always believed that clocking inaccuracies is more likely with external clocks...
  
 As for the old jitter chestnut, is that really an issue today?  Was it ever an issue outside the dawn of digital audio for most well implemented converters?
  
 I am always amused by those claiming to hear jitter as they usually are the same people espousing the virtues of analog equipment, but seem deaf to the audible issues of wow, flutter, rumble etc.


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## Arpiben

nick_charles said:


> Dunn's papers are very informative and definitely worth reading, the only thing that his and Hawksford's (also highly readable) papers lack is that, they do not address audibility of jitter except as a mathematical model based on rather extreme sound pressure levels. To date we have little (well none really)  empirical evidence that jitter as found in semi-competent digital components is actually audible, of course that does not stop a cottage industry in jitter reducers....




I have to agree with you in the sense that I didn't find papers addressing the jitter's audibility thresholds.
As a starting point I have J.Dunn's value of 100ns for below 100Hz....
Thanks for your point.


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## Arpiben

[@] [/@]les[/@]





spruce music said:


> I'll repeat the link from gregario's post.
> 
> http://www.soundonsound.com/techniques/does-your-studio-need-digital-master-clock
> 
> ...




Gregorio's post and link are showing some drawbacks of external reference clocks as well as their utilities when dealing with big chains/numbers of ADCs, DACs,etc...

Summarising, I am assuming when dealing with one or a small number of ADCs you will have no benefits in using Cesium/Rubidium clocks.
When dealing with a certain amount of clock recovered equipments there is need if not must for such external references.

In Telecom networks the recovered clock need to be regenerated roughly after every 20 equipments.
Otherwise you are degrading the Wander (Jitter below 10Hz)

I am assuming a kind of similar behavior when dealing with Audio.

Now remains the question of audibility of those issues as per nick_Charles's post.


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## gregorio

arpiben said:


> [1] Summarising, I am assuming when dealing with one or a small number of ADCs you will have no benefits in using Cesium/Rubidium clocks.
> When dealing with a certain amount of clock recovered equipments there is need if not must for such external references.





> [2] Now remains the question of audibility of those issues as per nick_Charles's post.


 
  
 1. Starting about 20 years ago, there was a big thing in the pro audio community about external clocking, largely due to the misleading marketing claims by one of the most respected high-end manufacturers of pro ADCs at the time. The whole thing culminated into probably the most infamous public argument in the history of the pro audio community, as it was extremely acrimonious and involved a considerable number of the biggest/most influential names in the business (Bob Katz, Bob Ohlsson, Dan Lavry, Nika Aldrich and a whole slew of others). That aside, here are the basic rules of studio/pro clocking:
  
 Regardless of how good/expensive, an external clock will not improve the performance of a pro ADC. An external clock will degrade the performance of an ADC or in the very best case scenario not make any difference. When linking digital audio equipment together (say more than one ADC, a digital mixer, etc.), a master-clock source is absolutely required, otherwise the system simply won't work. In general, the best source for this distributed master-clock would again be the internal clock of the ADC. There are some exceptions however, some complex studio topologies and most scenarios where audio and video require synchronisation for example. In these cases an external master-clock maybe beneficial or even unavoidable but even so, there is no advantage with those ridiculously expensive clocks over far cheaper alternatives because the clock signals they produce are never directly used by ADCs. Even in this scenario, the ADC is still using it's internal clock, which in effect is regenerating the external clock source. The determining factor is the quality of the clock recovery/regeneration topology of the ADC not the accuracy of the external clock. A decent $200 external master-clock will end up with a regenerated clock signal in the ADC which is no less (or more) accurate than if the external master-clock were a $10k+ atomic clock!
  
 2. The only published study figures I've seen, indicates a threshold of audibility (with music material rather than test signals) of 20ns. I can't remember where I saw it though and I also believe it's been disputed (claiming the figure should be significantly higher). Even accepting this 20ns figure as the audibility threshold, that's still several hundred times more jitter than modern pro ADCs produce!
  


drtechno said:


> Yes clocking  is very important as that was drilled into my head by Mr. Ludwig and Mr. Williams about 20 years ago to use an external clock.


 
  
 Oh dear! Are you really claiming to have been taught by the legendary mastering engineer Bob Ludwig? If so, I'm going to call you out on that lie too! Bob Ludwig knows his stuff and while I can't quote him directly, I can quote him indirectly (from another legendary mastering engineer, Bob Ohlsson) from over a dozen years ago:
  
 "_Bob Ludwig made it very clear during his workshop the other day at AES that what sounds best to him in his room is using the internal clock of an A to D and, for playback only, clocking the entire system off the internal clock of the D to A converter. He spoke very highly of using the Big Ben for locking to video but never suggested he uses it in place of an internal clock when that's possible._"
  
 It's not plausible that you even have a college level education in digital audio, let alone be personally "drilled" by one of the industry's greats. Why do you persist in these lies, especially such obvious lies? I can't see how being exposed as a persistent liar is of any benefit to you and your lies are certainly of no benefit to anyone here, so for you own (and everyone else's) sake, please STOP!
  
 G


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## castleofargh

my very limited understanding about external clocks aligns with gregorio. to me it's a tool to match different devices when timing is important, and that's it. not a way to "upgrade" the ADC clock.
  
 about jitter, there are all sorts of values that came out from more or less serious studies, and it's to be expected as jitter isn't only one constant thing with one single cause. most would manufacture the jitter and/or the test signal; making it a special case in a special case. with musical content, people have a lot less sensitivity to jitter. well they have a lot less sensitivity to everything ^_^.
  
  
 now the modo part.
 drtechno's post didn't claim that he worked or learned directly under those guys. so as long as he doesn't explicitly say so, I don't appreciate to see him treated as a liar. at least not over something unclear like this. again, argue the claims, not the people.
 about the external clock, trying to play devil's advocate here, is it possible that Ludwig changed his mind over the years on the matter? only fools never change their mind. or that doc got his information from a bad source? or maybe he mixed up his memories(I have a few personal anecdotes on the matter that nobody cares about).


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## spruce music

https://www.researchgate.net/publication/242508896_Detection_threshold_for_distortions_due_to_jitter_on_digital_audio
  
 You can get the entire article done in 2005 here.  This one using music, and 2AFC blind testing.  Some people reliably detected 500 nseconds of random jitter.  Most couldn't do that.  No one could detect 250 nanoseconds of random jitter.
  
 I believe Eric Benjamin and Benjamin Gannon working for Dolby Labs got different results with a different methodology.  They used non-random jitter at something like. 1500 hz and 1800 hz. Non-random jitter would of course be more audible.  Using high level test tones at 10 nanoseconds on 17 khz  tones some people heard it (jitter makes more difference on higher frequencies).
  
 Jitter of 121 picoseconds of a 20khz tone at maximum level could alter the LSB for a redbook format.  Doesn't mean it would be audible, but it changes the sample value.  Lower frequencies are proportionally less effected.  Lower level tones also would need more jitter to alter the LSB. So lower frequency lower level sounds needs lots of jitter to even change the sample values much less change them enough to be heard.
  
 Very inexpensive consumer gear typically has jitter levels below 500 picoseconds.  Jitter just isn't a problem.


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## gregorio

> Originally Posted by *spruce music* /img/forum/go_quote.gif
> 
> [1] (jitter makes more difference on higher frequencies).
> 
> [2] Very inexpensive consumer gear typically has jitter levels below 500 picoseconds.  Jitter just isn't a problem.


 
  
 1. That's my understanding. More importantly though, it's the understanding of the industry. For at least 15 years or so, most/all pro ADC manufacturers have been attenuating jitter noise in the higher frequency band.
  
 2. I don't know much about the jitter specs of inexpensive consumer gear. Pro audio ADCs typically produce less than 100ps of jitter and figures near half of that are not at all uncommon.
  


castleofargh said:


> [1] drtechno's post didn't claim that he worked or learned directly under those guys. so as long as he doesn't explicitly say so, I don't appreciate to see him treated as a liar.
> 
> [2] about the external clock, trying to play devil's advocate here, is it possible that Ludwig changed his mind over the years on the matter? only fools never change their mind. [a] or that doc got his information from a bad source? * or maybe he mixed up his memories(I have a few personal anecdotes on the matter that nobody cares about).
> *


*

  
 1. He did explicitly say so: "Yes clocking is very important as that was drilled into my head by Mr. Ludwig and Mr. Williams about 20 years ago to use an external clock." - "drilled into my head" means to repeat the information numerous times until understanding/acceptance is certain. Drilling something into someone's head takes time and persistence and he is clear that this "drilling" was done to him by Ludwig (and Williams). drtechno's statement was about as unambiguous as I can imagine.
  
 2. It's certainly possible that Ludwig changed his mind on the issue. However, if that is the case it would have to have been in response to some change which occurred in the manufacture of pro ADCs. It is possible that an external clock could improve the performance of an ADC but the ADC would need to have a very poor quality internal clock, a high quality clock recovery mechanism and both internal and external clock sources would need to be routed through that recovery mechanism. I know of no pro ADCs which fulfil these requirements since I've been in the business, although maybe some did before then, which could account for a change of mind (if there ever were one) but that would be outside the time-frame described by drtechno! It is possible, though very unlikely, that Ludwig was at one time simply mistaken but I find that very hard to believe.
 2b. No. It's information posted by Bob Olhsson personally (another of the most respected/influential mastering engineers) in 2004.
 2c. Bob Ohlsson was posting about an AES event he was present at, just a few days prior to his post, so the memories were pretty fresh.
  
 G
*


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## Arpiben

*1. Jitter makes more difference on high frequencies *

True by definition of jitter when dealing with ADC/DAC....
Jitter is dealing with f>10Hz.
Therefore Jitter is characterizing short time variations t<0.1s.
By reducing Jitter at high frequencies you are improving sampling rate accuracy.
ADC/DAC need short term stability from clocks.

On the other hand, Wander (Jitter f<10Hz) is characterizing long term stability.
In this case Wander low frequencies make more difference.

Atomic clocks have long term stability, here window time is counted in days ( around 200ns/3days)
Short term jitter is around 1ns/1s window.

Edited: 
As mentioned by spruce music and by gregorio , with proper ADC's clocks, Jitter falls around picoseconds values.
On the contrary, even with low phase noise (Jitter) external Atomic clocks you destroy the benefits by using cables and PLLs down to your sampling frequencies.


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## castleofargh

/!\ this is casual audio talk from a distant future /!\
  
 I wish I could listen to music the way the artist intended, but atomic clocks have so much jitter, the trebles just feel artificial.
 portable devices will never sound as good as a home system, it's pretty obvious that when I'm altering the flow of time around me by walking down the street, my DAP and my head don't follow the same movement, that's sure to create jitter.
 real audiophiles put the playback system at the same level as their head. I couldn't stand the pitch error from the variation of altitude between my head and my desk.


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## drtechno

castleofargh said:


> /!\ this is casual audio talk from a distant future /!\
> 
> I wish I could listen to music the way the artist intended, but atomic clocks have so much jitter, the trebles just feel artificial.
> portable devices will never sound as good as a home system, it's pretty obvious that when I'm altering the flow of time around me by walking down the street, my DAP and my head don't follow the same movement, that's sure to create jitter.
> real audiophiles put the playback system at the same level as their head. I couldn't stand the pitch error from the variation of altitude between my head and my desk.




  
 The only issue I ever had with an atomic clock and a trinity clock divider was an install issue that I solved was adding a clock distribution amp. The signal out of the trinity is 3V p-p and the converters in that installation was 5v clock and it had the jitters until I put the distro amp in service. Jitter is not really a big issue on newer converters as there has been advancements in that part of the technology. However, I'm talking about the $2200-$16000 type that have advanced (high end tracking and mastering class).


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## Arpiben

When dealing with clock distribution (Atomic, OCXO or whatever), you need to avoid:

any loop possibilities,
any excessive cable length ( around 5m with 75Ohms coaxial)
  
 I have no experience at all with clocking in Audio industry. But I am quite familiar with it in Telecommunication domain: Microwave radio, Satellites & IP networks.   
  
 Thanks to @spruce music & @gregorio I received my missing clues about audible jitter & atomic clocking.
 Thanks to @drtechno I had the curiosity to dig inside pro audio industry technical specifications of such equipments.
  
 I was quite surprised not to find any relevant informartion regarding what really matters:

output user clocks: 44.1kHz / 48 kHz / 192 kHz etc...
  
 Instead I found, IMHO, lots of non useful data or not understood ones at f=10 MHz:

phase noise / low phase noise
short term stability without mention of  Allan Deviation or variant ones ( TDEV,MDEV,etc)
long term stability without mention of MTIE ( Max Time Interval Error)
aging, drift, etc...
  
 As a reminder Phase Noise (db/Hz) and Jitter (db/s) are equivalent, one is measured in frequential domain when Jitter is expressed in time domain.
  
 Anyway,relevant information is also missing in lots of other audio(or non audio) fields: amplifiers, etc...
	

	
	
		
		

		
			




  
 Thanks.


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## kutya

Hello dear community 

The other day i got a couple of 24/96 and 24/192 flac files, with my favorite music. 
And I was amazed that a 24/96 and 24/192 sounds significantly quieter than 16 / 44.1 
What makes this possible?
Thanks for help


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## bigshot

Most likely because it's at a lower volume level. It might be different mastering too.


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## gregorio

kutya said:


> And I was amazed that a 24/96 and 24/192 sounds significantly quieter than 16 / 44.1
> What makes this possible?



Audio Compression. Almost certainly, the 16/44.1 version has had more compression/limiting applied.

G


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## redrol

16/44.1 for life..
Actually, lets look for better recordings and mixing.. lets stop brickwall mastering.  Redbook has plenty of dynamic range.


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## Whitigir

What is the dynamic range and limitations of Microphones and it technologies again ?


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