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Does DSD sound better than PCM?

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Kal Rubinson

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Sorry I don't understand the point you're making? Yes Morten Linberg records in DXD (he prefers it to DSD - please don't start...) into Pyramix and creates all kinds of versions because people want all kinds of versions of his recordings from the DXD multichannel original to Blu-ray 192/24, 96/24 down to CD. The point I was making was I am not aware of anyone recording in DXD just to convert and sell it in DSD.
Perhaps, I am not understanding your point. Morten and others who record in DXD also sell DSD files drawn from them.
 
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I appreciate the sentiment, but by that logic, why would one ever use DSD for audio production (or reproduction, for that matter)?
Seems like handicaps all the way down.
As explained, she prefers the sound and therefore will put up with the hassle.
 

M00ndancer

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As explained, she prefers the sound and therefore will put up with the hassle.
This is going to go in circles like a lot of other discussions here.
The consensus on ASR it that there "might" (as in "can perhaps find differences in audio way above 20 KHz and below -120 dB") be a difference between DSD and PCM due to the different tech. However, if you find a difference between a FLAC and a WAV, you're doing it wrong. FLAC is a non-destructive compression.
 

andreasmaaan

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As explained, she prefers the sound and therefore will put up with the hassle.
But neither PCM nor DSD have a “sound” in the first place o_O

Look, a recording engineer may not be the best person to ask about the technical aspects of a digital format, just as the engineers who design a digital format may not be the best people to use that format to make a recording.

I was at a highly-regarded and well-equipped recording studio in Berlin over the weekend. The senior mastering engineer there is convinced (as are the other engineers) that the new power cables he has installed in the main mastering room to feed their ATC SCM110ASL monitors and subs make an obvious, audible difference. I challenged them on this, asking whether they'd taken measurements to confirm what they'd heard. They of course hadn't done this, but they had "blind" tested the cables, with one person switching between cables while the other listened. In the first few minutes they'd got a couple of positive results, no doubt influenced by the knowledge of the tester, who was already convinced before they did the "test". They are now convinced of something that is frankly near-impossible.

I don't pretend to understand a lot more about electrical engineering than they do, but I am confident they are wrong and that I could design a test to show it.
 
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This is going to go in circles like a lot of other discussions here.
The consensus on ASR it that there "might" (as in "can perhaps find differences in audio way above 20 KHz and below -120 dB") be a difference between DSD and PCM due to the different tech. However, if you find a difference between a FLAC and a WAV, you're doing it wrong. FLAC is a non-destructive compression.
Look, sorry (heavy sigh...) I'm simply reporting a fact NOT an opinion. That fact being that Ms Cookie Marenco prefers the sound of DSD to PCM and that's why she works in that format for her own projects. I've become well aware since appearing on this site that everyone else considers this a mistaken belief. So be it.
 
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But neither PCM nor DSD have a “sound” in the first place o_O

Look, a recording engineer may not be the best person to ask about the technical aspects of a digital format, just as the engineers who design a digital format may not be the best people to use that format to make a recording.

I was at a highly-regarded and well-equipped recording studio in Berlin over the weekend. The senior mastering engineer there is convinced (as are the other engineers) that the new power cables he has installed in the main mastering room to feed their ATC SCM110ASL monitors and subs make an obvious, audible difference. I challenged them on this, asking whether they'd taken measurements to confirm what they'd heard. They of course hadn't done this, but they had "blind" tested the cables, with one person switching between cables while the other listened. In the first few minutes they'd got a couple of positive results, no doubt influenced by the knowledge of the tester, who was already convinced before they did the "test". They are now convinced of something that is frankly near-impossible.

I don't pretend to understand a lot more about electrical engineering than they do, but I am confident they are wrong and that I could design a test to show it.
First, see my reply to M00ndancer. My own opinion on the matter is and please believe me I'm happy to say "oh right I get it now" is I just don't understand how capturing sound in two different digital formats that, though they share similarities are also very different in the way they capture that signal and therefore on playback will 'build' that signal in exactly the same way therefore creating a difference (however small) in the sound reproduced at the other end of the chain is surely pretty likely isn't it?

This is an observation and comment that's all - No-one recording in DSD is making some hugh financial gain by doing so, it's testy and time consuming to work with and is a very, very niche market, the only logical conclusion is they consider the format to give a sonic benefit to their projects.
 

DonH56

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I've not really been following this thread so apologies if this has already been said... Isn't it possible the differences heard have nothing (or very little if anything) to do with the actual format of the bits and more about how the analog output is treated? Perhaps the mixing engineer heard something she liked in the signal chain with DSD and stuck with it, though the actual cause of the difference might have been something other than DSD vs. PCM encoding. All that matters to her is that she heard a device and preferred the chain with DSD. It's possible the outcome would have been different with different components (e.g. DAC and output buffer).
 

andreasmaaan

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My own opinion on the matter is and please believe me I'm happy to say "oh right I get it now" is I just don't understand how capturing sound in two different digital formats that, though they share similarities are also very different in the way they capture that signal and therefore on playback will 'build' that signal in exactly the same way therefore creating a difference (however small) in the sound reproduced at the other end of the chain is surely pretty likely isn't it?
This is the crux of the disagreement I believe.

Psychoacoustic research has shown what humans are capable of hearing and what they are not. Both DSD and PCM are capable of keeping noise and all forms of distortion far below any demonstrated threshold of human hearing.

In this context, it would be frankly shocking if any human could actually hear any difference between the sound reproduced by either format (unless this were due to some extrinsic factor, for example like those Don mentioned in the previous post).
 
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This is the crux of the disagreement I believe.

Psychoacoustic research has shown what humans are capable of hearing and what they are not. Both DSD and PCM are capable of keeping noise and (all forms of distortion) far below any demonstrated threshold of human hearing.

In this context, it would be frankly shocking if any human could actually hear any difference between the sound reproduced by either format (unless this were due to some extrinsic factor, for example like those Don mentioned in the previous post).
Yeah I get that but - however small - surely because of their difference in signal capture each format may limit or emphasise a particular area more than the other on playback? Maybe the bonkers high (technical term) sample rate of DSD just manages to grab something more in the timing aspect - I don't know...
 
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I've not really been following this thread so apologies if this has already been said... Isn't it possible the differences heard have nothing (or very little if anything) to do with the actual format of the bits and more about how the analog output is treated? Perhaps the mixing engineer heard something she liked in the signal chain with DSD and stuck with it, though the actual cause of the difference might have been something other than DSD vs. PCM encoding. All that matters to her is that she heard a device and preferred the chain with DSD. It's possible the outcome would have been different with different components (e.g. DAC and output buffer).
Yep totally agree but I'm pretty sure she (Cookie Marenco) having been in the business for so long would have heard PCM through any number of pathways/DAC's and still favours DSD.
 

andreasmaaan

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Yeah I get that but - however small - surely because of their difference in signal capture each format may limit or emphasise a particular area more than the other on playback? Maybe the bonkers high (technical term) sample rate of DSD just manages to grab something more in the timing aspect - I don't know...
It doesn't though. If we assume humans cannot hear above 20KHz, properly filtered redbook PCM perfectly captures all timing information.

In other words, properly filtered PCM does not create transient distortion within the audio band. DSD can't improve upon perfection.

If we start looking at transient performance above 20KHz, differences can of course be found between PCM and DSD and between different sample rates of PCM and between PCM recordings to which different anti-aliasing filters have been applied. But these differences are not audible since humans cannot hear that high in frequency.
 
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It doesn't though. If we assume humans cannot hear above 20KHz, properly filtered redbook PCM perfectly captures all timing information.

In other words, properly filtered PCM does not create transient distortion within the audio band. DSD can't improve upon perfection.

If we start looking at transient performance above 20KHz, differences can of course be found between PCM and DSD and between different sample rates of PCM and between PCM recordings to which different anti-aliasing filters have been applied. But these differences are not audible since humans cannot hear that high in frequency.
I totally get the fact that 96/24 PCM can more than adequately grab all the frequency and dynamic range we could possibly need as humans, but I'm not convinced about timing. I can't find an online link to this paper beyond http://www.aes.org/e-lib/browse.cfm?elib=12372 - I had to buy it a while ago but its interesting reading: AES Convention Paper 5931 Physical and perceptual considerations for high-resolution audio. I can find scant other research in this area but if the data is correct then it suggests why DSD/PCM could have audible differences due to timing gaps between samples?
 
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March Audio

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I totally get the fact that 96/24 PCM can more than adequately grab all the frequency and dynamic range we could possibly need as humans, but I'm not convinced about timing. I can't find an online link to this paper beyond http://www.aes.org/e-lib/browse.cfm?elib=12372 - I had to buy it a while ago but its interesting reading: AES Convention Paper 5931 Physical and perceptual considerations for high-resolution audio. I can find scant other research in this area but if the data is correct then it suggests why DSD/PCM could have audible differences due to timing gaps between samples?
PCM Timing is not an issue. See this video. Specifically at 21 mins

 

M00ndancer

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Yep totally agree but I'm pretty sure she (Cookie Marenco) having been in the business for so long would have heard PCM through any number of pathways/DAC's and still favours DSD.
"long time in the business" has no real value here. If you subjectively perceive the sound to be "better", without measuring it, it's only hearsay. Does she produce great music? Do you enjoy listening to it? Good, keep listening. But do not buy into that "I can feel the music", "It sounds better", "There is a difference between A and B"
If she find that she prefers the workflow that her DSD recording/mastering gives her, fine. Have no issue with that. But don't expect anyone here to take her statements at face value.
 

M00ndancer

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PCM Timing is not an issue. See this video. Specifically at 21 mins

Thanks, yep seen it along time ago and it's a great explanation but sorry I just don't get the timing explanation he gives and cannot relate this in any way to music capture with the complexities of varying waveform. The paper I referred to states microseconds between samples that in the authors view will lead to loss of very(!) fast transients in the music?
 

andreasmaaan

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I totally get the fact that 96/24 PCM can more than adequately grab all the frequency and dynamic range we could possibly need as humans, but I'm not convinced about timing. I can't find an online link to this paper beyond http://www.aes.org/e-lib/browse.cfm?elib=12372 - I had to buy it a while ago but its interesting reading: AES Convention Paper 5931 Physical and perceptual considerations for high-resolution audio. I can find scant other research in this area but if the data is correct then it suggests why DSD/PCM could have audible differences due to timing gaps between samples?
The paper is here.

The relevant passage is:

"In some musical instruments, acoustic pressure builds up extremely fast during onset transients reaching tens of dBs within a few microseconds. For example, transient onset of xylophone shows waveforms with rise time of less than 10 μs with instantaneous peak output reaching 126dB SPL. Trumpet playing fortecan register 120-130dB peak SPL with steep rise of the waveform within only 10μs to full signal level. Snare drum reaches 130dB and cymbals 136dB peak SPL within microseconds [9].Rogowski ‘s values were measured during a short musical selection captured using 1/4 inch, Brüel&Kjaer 4135 microphone and a 192kHz, 12bit A/D conversion. Can CD-rate sampling of audio every 22.7μs register the full waveform detail of the sound of these instruments? Based on these onset requirements, to achieve a transparent recording medium, one should sample audio with less than 1μs between samples to accurately capture steep waveform changes. One revealing transient test of recording system is to use the sound of dangling keys or striking wine glasses as a source. Our familiarity with these sounds is frequently refreshed and can be used to inform us of problems with the transparency of the recording system. A brutal case of poor transparency occurred in early consumer CD players that used single D/A converter working with both channels in multiplex. Noticeable directional shifts were sometimes heard on transients when left and right channel converted the same onset in a sequence of samples."

There's some confused information in that. Most importantly, the author seems to forget the relationship between risetime and frequency.

If an acoustic source's output has a risetime of under 22.7us, that is simply because it contains frequency content above 22.05KHz (ie the maximum frequency that redbook can accurately capture, and a bit above the maximum frequency that most humans can hear).

So the higher frequency content is present, and measurable (both in the frequency domain and the time domain, which is the domain that the above passage focuses on) - but it is not audible. The highest audible content that is present simply can't have a risetime of less than around 23 or more, depending on the hearing ability of the person listening of course.

In other words, redbook cannot capture this instrument in its fullness, but it can capture every component of the instrument that is audible to humans.
 
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