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MattHooper

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Been said before. If there was a substantial difference in the sound of DSD and PCM it would have been confirmed a long time ago. If there are any differences they'll be small, fleeting, audible to only some people, on some particular material. In other words if there are real differences almost anything will cause a bigger difference than the sound quality of DSD vs PCM. And those differences simply won't be of large consequence. They'll be inconsequential. Considering all the inconvenience of DSD anything it simply represents a poor decision.

I don't listen to DSD so wouldn't know, but that diagnosis seems to tie up a lot of what I've read together.

Though, let's face it, even on an audio website like this an enormous amount of dialogue and measurements goes on evaluating components where sonic differences are at best "small, fleeting, audible to only some people, on some particular material."
 

pkane

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differences are at best "small, fleeting, audible to only some people, on some particular material."

It is the nature of the audiophile hobby to pursue smaller and smaller differences while exaggerating them to be huge, obvious, and not even subtle.

1599950356407.png
 

MattHooper

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It is the nature of the audiophile hobby to pursue smaller and smaller differences while exaggerating them to be huge, obvious, and not even subtle.

View attachment 82728

Absolutely.

But, to flip perspectives for just a moment, as an audiophile I also get the way we can blow up small differences. In a certain context, they are undoubtedly tiny differences. But in another, small differences can have subjectively big effects on one's enjoyment or perception of the sound.
For instance, the removal of a tiny peak in the upper frequencies, or a smoothing of the high frequencies of distortion, can make the difference between being slightly on edge and "aaaahhh." It's sort of like how a husband can notice the slightest change of tone in his wife's voice to catch when she's annoyed or whatever - something so minor others may miss, but which make a big difference to the domestic mood :)
 

pkane

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Absolutely.

But, to flip perspectives for just a moment, as an audiophile I also get the way we can blow up small differences. In a certain context, they are undoubtedly tiny differences. But in another, small differences can have subjectively big effects on one's enjoyment or perception of the sound.
For instance, the removal of a tiny peak in the upper frequencies, or a smoothing of the high frequencies of distortion, can make the difference between being slightly on edge and "aaaahhh." It's sort of like how a husband can notice the slightest change of tone in his wife's voice to catch when she's annoyed or whatever - something so minor others may miss, but which make a big difference to the domestic mood :)

Until someone actually pays attention, they may not ever notice these "imperfections" and think the reproduction is perfect. But as soon as you point it out to them, or they read someone else complaining about it, it starts to stick out like a sore thumb. Over time, it becomes nearly impossible to ignore these minor differences that don't bother anyone else in the world. And then, an audiophile is born ;)
 

KeithPhantom

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Different ADCs and DACs with associated analogue circuitry, for starters. Then there's whatever signal path additions are needed to match levels. On top of that, there's the possibility of downstream side effects of the residual ultrasonic noise after the DAC, which will differ. If you hear a difference, you have no way of knowing what actually caused it.


ADC chips, at least the good ones, use multi-level sigma-delta and convert to PCM or 1-bit DSD digitally. You can't escape that conversion.

If there is something audible that only one of the formats can capture, converting to the other and back has to reveal it. It's by far the simplest way to go about it, and if everything else is kept exactly the same, you can be sure any audible difference really does stem from the conversion. Of course, it could still be the result of a poorly executed conversion rather than an actual deficiency in the alternate format.
Interesting to know, thanks.
 

Esotechnik

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1. I generate in Adobe Audition white noise with peak levels -7 dB (48kHz/32bit, 30s length).
I convert it in Taskam HiRes editor to DSD 5.6M and back to PCM 48/32, w/o fadings.
Audio Diffmaker get results (wav2-wav1):
parameters: 307,4usec, -0,343dB (L), -0,343dB (R)..Corr Depth: 14,7 dB (L), 13,8 dB (R)

2. I download test track original2.wav from https://www.gearslutz.com/board/showpost.php?p=14767195&postcount=2023
I convert it to mp3 via lame 3.98.2 (--preset extreme)
and back (lame --decode) to WAV 44/16.
Audio Diffmaker says:
>> Parameters: 0sec, 0,004dB (L), 0,001dB (R)..Corr Depth: 69,1 dB (L), 83,6 dB (R)

3. original2.wav to DSD2.8M and back to PCM 44/24

>> ---Beginning difference extraction---
>> Files are stereophonic, at 44,1kHz sample rate, about 121,1 seconds long.
>> The Reference track is 24 bits; the Compared Track is 24 bits
>> Calculating Correlated null depth.
...
>> Complete: saving the "Difference" WAV file
>> Parameters: 378,7usec, 0,000dB (L), 0,000dB (R)..Corr Depth: 101,6 dB (L), 101,4 dB (R)

Conclusion: if records get from slow (milli littlebit) delta-sigma ADC, this is unsuitable for evaluation of time domain advantages of R2R DACs.
Shit Deltasigma in - DS out :)

4. Test DSD signals of 5th order (-9 dB peak) have 1% THD+N:
https://www.ap.com/technical-library/more-about-pdm/

Thus, 1bit is bad lossy format.
 
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mansr

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Audio Diffmaker is worse than useless. If it were merely useless, it wouldn't have led you to those erroneous conclusions.

The misunderstanding of the distortion figure quoted from the AP article is all on you, though.
 

Esotechnik

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I didn't realise there was anybody still believing this bollox.
Datasheet for DAC AD1862 - Settling Time 350 ns
DAC PCM1704:
Settling Time ±0.0003% of FSR, ±1.2mA Step 200 ns

Op-amp AD797 (used in Audio Precision 2722):
"800 ns Settling Time to 16 Bits (10 V Step)"

SAR ADC LTC2380-24:
Transient Response Full–Scale Step: 95 ns

Any other delta-sigma converters "for audio" - time is unspecified or for old ADC PCM1760:
"Output Data Delay fs = 48kHz 1.5 ms"
This is only 666 Hz in frequency equivalent.

Another delta-sigma 1 bit vs. multibit (SAR) ADC comparison, see fig.5-11:
https://www.researchgate.net/public...o_measurements_using_Maximum_Length_Sequences
 

mansr

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Datasheet for DAC AD1862 - Settling Time 350 ns
DAC PCM1704:
Settling Time ±0.0003% of FSR, ±1.2mA Step 200 ns

Op-amp AD797 (used in Audio Precision 2722):
"800 ns Settling Time to 16 Bits (10 V Step)"

SAR ADC LTC2380-24:
Transient Response Full–Scale Step: 95 ns

Any other delta-sigma converters "for audio" - time is unspecified or for old ADC PCM1760:
"Output Data Delay fs = 48kHz 1.5 ms"
This is only 666 Hz in frequency equivalent.

Another delta-sigma 1 bit vs. multibit (SAR) ADC comparison, see fig.5-11:
https://www.researchgate.net/public...o_measurements_using_Maximum_Length_Sequences
You are comparing apples to orangutans.
 

Thomas savage

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There's something about the way DSD has been marketed , iv always ' felt ' it's more authentic and more ' analogue ' . It's a bit like how they have gone about marketing MQA imo . When listening however that impression always disappears.

If there's some effort made to preserve dynamic range and push either MQA or DSD towards folks who listen with speakers and a hifi rather than earbuds or radio etc then by dint of that one might be better served . So nothing strictly to do with its technical merit . Having said that I'm rarely impressed with ' audiophile' recordings , they all to often seem sanitised and dull to me , bit like audiophiles themselves lol.
 

Jimbob54

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IKEA has high dynamic rage , as dose Evey carpark in the southwest of England right now.

Its hard to know where the puns stop and the typos and spelling errors begin. Bravo!
 

andymok

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Although interesting and quite informative, this thread about DSD is yet (n+1)th attempt at showing just how useless it is supposed to be - compared to the PCM.

The reason for the above statment ? Not a single member - at least in this thread - has not even hinted at the fact that she/has ever had access to the live microphone feed and consequent monitoring of both PCM and DSD recording.
Like in ACTUALLY BEING THERE - not just spreading hearsay by other people, who did compare the three.

I am doing it - constantly. Because I record music - mostly acoustical, any genre for which musicians themselves must have no clue what electricity is. Vocal, choir, classical, jazz, ethno - as long it is unplugged. Musicians are allowed LED lamps so that they can see their sheet music - everything else is off limits.
There are cases I have to do with Nikola Tesla's "Call From The Grave" - 50/60Hz hum ( and all its harmonics ... ) - plus any ultrasonic noise from the lighting. I will try to suppres the lighting to the max ... - but sometimes, it is unfortunately impossible to do that.

The importance of the recording - and consequent mastering, IF and WHEN required - plays far more important role than does the recording medium.
A properly recorded analog cassette master will always trounce a poorly recorded hirez - regardless if it is PCM, DSD or whatever.
That means only relatively simple miking techniques that can preserve tiny time differences in original sound are really suitable. Any multimiking is bound to blur these tiny differences - or nullify them completely.

Let me say it up front - I prefer DSD over PCM. To my ears, it sounds closer to the live mic feed than does PCM. Still, this holds true only if and when DSD is of sufficiently high oversampling frequency. DSD64 - or what is more known as SACD - is just not good enough. There is far too much ultrasonic noise above 20 kHz - which can and does affect the performance in nominally audible band ( 20 Hz-20kHz ). TBH, in that case, I am likely to prefer any 24bit depth PCM with sampling at least 88.2kHz. It takes DSD128 and above for the DSD to really start to deliver.

It means that the data required for the about comparable - not SAME - audio quality, PCM will require smaller file than DSD. But once one accepts that "waste", there is a sonic reward.

DSD ( if sampling is high enough...) can preserve spatial cues better than "equivalent" PCM. For that, recording venue has to be above certain limit in size - a typical jazz club is simply too small for this to be readily audible. But once great enough concert hall or church is in question, one can hear/sense/percept how the sound "travels"... - giving even on reproduction the proper scale of size of the venue, even on 2 channel only.
For those insisting Redbook CD is all that it takes ... - bounce the master in DSD to RBCD - and you get the typical flat soundstage with next to no depth.

DSD - even the lowly DSD64 - has frequency response extended above 20 kHz. By the time present standard in DSD recording - DSD256 - is achieved, it can cover response up to 100 kHz with about 100dB SNR even at that frequency - or something in the general vicinity of these two figures.
There is music WELL above 20 kHz - even well above 100 kHz. But present definition of HiRez Audio, which requires frequency response to about 40 kHz, is a good starter and a reasonable value for the interim period, before the full 100 kHz bandwidth becomes the standard.

You might ask - why ?
https://www.cco.caltech.edu/~boyk/spectra/spectra.htm

The instrument that is on average the most alive above 20 kHz is harpsichord. In a good recording, there are overtones to at least 50 kHz. Although relatively low in level, they ARE there - and are not from lighting, CRTs, digital artefacts or any other source than the instrument itself.
The second instrument with a little less broad, but for that matter, far more precise defined content above 20 kHz, are chimes.


I agree DSD is difficult to work with, that it has to be converted into some form of PCM for editing - and , besides that, NOTHING else can be done to it in digital domain. In that, it is only slightly more forgiving tha recording direct to analog disk - multiple takes are possible, which can be later edited/spliced.
But there is no EQ, pitch autotune, or any other of untold amount of plugins available in PCM.
There are far fewer musicians capable of recording without the usual "safety net of PCM - we'll fix it in the mix" approch.
And even fewer willing to leave an error here or there in the finished product - despite the performance "in one go" containing that inexplicable but perceptible spirit a note by note perfect studio edited recordings almost always lack.

I agree that old recordings are NOT the proper representation of what DSD can do - either from analog tapes, but even more so from the early digital recording era.
I see there are now some attempts to standardize the way old recordings should be converted to DSD in order to be "legit" new DSD recordings. Although a move in right direction, the audio quality can never equal a well made DSD recording of today.

DSD will forever remain a niche within a niche - but, for those who understand it, know what it takes and are willing to go an extra mile, it will perhaps continue to represent the last safe heaven of honest music recording.

I've always wanted to see how the waveform (ie time domain) differs between 44.1/16 and DSD, on a thing that has a lot and huge micro dynamics (eg drums, percussions)
 

Frank Dernie

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The reason for the above statment ? Not a single member - at least in this thread - has not even hinted at the fact that she/has ever had access to the live microphone feed and consequent monitoring of both PCM and DSD recording.
I haven't myself, though I do have both PCM and DSD recorders.
OTOH I have compared the live feed from the microphone to the output of my DAT recorder (48/16) about 25 or 30 years ago and the output of the recorder sounded the same as the microphone feed to me.
You can't get better than that, whatever other recorder or technology you try.
I suppose it may be possible that on other sorts of music there may be a minuscule difference, but I doubt it.

Edit, in case anybody was wondering anout my cloth ears, I clearly hear the difference between microphone feed and recorder output on all the reel-to-reel tape recorders I used before. I used to set levels to get a satisfactory (to me) balance between tape overload and tape noise.
 

Esotechnik

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Why in the world would you use Diffmaker once you've used Deltawave?
Deltawave require windows x64, but I prefer x86 due to drivers problem with some hardware (windows 2012R2 vs. windows7 x86 with 8GB RAM and fix128 PAE patch).

With 48k/24bit 30s -7dB white noise wav files and Korg Audio Gate 3.0.2 converter (in hi-quality mode, to DSD64 and back) results is better.
>> Parameters: 808,1nsec, 0,000dB (L), 0,000dB (R)..Corr Depth: 89,0 dB (L), 89,0 dB (R)

"Hi-res" lossy 1-bit format has ENOB=14 :facepalm:
 
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