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DSD vs PCM resolution


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- I have a DIYINHK board, and as far as I can tell it only plays DSD as DoP. Therefore it is limited to DSD128 AFAIK. How did you get DSD256?

- I am now using a JLSounds USB card. With Linux (Ubuntu Studio 14) and HQ Player I can play DSD256. The only LPF is the capacitance in the Lundahl transformer.

 

DIYINHK and JLsounds are both supported on Linux and don't require use of DoP...

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I downloaded the Trial of HQ Player. Sorry, I have never encountered such an infuriatingly unusable piece of software before. Usability scores on one out of ten are minus infinity. I pass, thank you.

 

A great way to justify not trying what is possibly the highest SQ player currently available...

 

There is an initial learning curve, but once you're past it, it's really easy to use.

 

Use PCM if you can't use DSD.

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Bonjour,

 

Yep to all the above, Hazard, and great thread over at diyaudio which became much better than my older one.

 

The so-called 'birdies' most probably had to do with the implementation since the down-conversion to 88 PCM was OK, but some people conclude the format isn't good...

 

I agree, the birdies result from implementation, but non of the DAC le FlipFlop, but of the ADC modulator during SACD recording.

 

Both Doede Douma (the creator of DDDAC who inspired me to bypass the digital filter) John Swenson have reported similar problems. It may be we hearing the problems predicted by Lipshitz/Vanderkoy in their paper criticising DSD.

 

JS observed that the Burr Brown style analogue FIR Filter (as also found in several DIY DSD DAC's) do a good job of scrambling these tones out of existence.

 

I thin purely digital modulator (e.g. the one in the DSD converter from Foobar2k) can perform near the theoretical limits and avoid creating these tones. Real modulators in analogue hardware seem to have problems though.

 

Anyway, birdies were only observed playing SACD Rips, not with (CD Standard) PCM converted to DSD.

 

A great way to justify not trying what is possibly the highest SQ player currently available...

 

There is an initial learning curve, but once you're past it, it's really easy to use.

 

Use PCM if you can't use DSD.

 

Note, I use PCM only, normally.

 

My Pass D1 DAC handles up to 96kHz/24Bit, I use M2Tech Hi-Face to drive SPDIF from a spare Lenovo Thinkpad.

 

I have been keeping some DSD/SACD images around as they seem to have better overall mastering and dynamic range than CD copies in circulation. For these I use the Foobar2k DSD-PCM converter to 88.2kHz/24Bit. It sounds no worse than any notional DSD DAC I had around to play with.

 

I found no sound difference between J-River and HQ Player, non.

 

If it actually sounded better I might be willing to put up with the interface (read lack thereof), lack of usability and learning curve and all if it did anything for sound.

 

Salud M.I.

Magnum innominandum, signa stellarum nigrarum

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It may be we hearing the problems predicted by Lipshitz/Vanderkoy in their paper criticising DSD.

 

Wasn't that paper thoroughly debunked by another paper?

 

JS observed that the Burr Brown style analogue FIR Filter (as also found in several DIY DSD DAC's) do a good job of scrambling these tones out of existence.

 

I thin purely digital modulator (e.g. the one in the DSD converter from Foobar2k) can perform near the theoretical limits and avoid creating these tones. Real modulators in analogue hardware seem to have problems though.

 

There are excellent modulators and filters in HQ Player. Ideally, for further isolation, HQ Player should be used with an NAA (network-attached audio device) running a small HQ Player networkaudiodaemon but you can still get great results with a direct connection.

 

Anyway, birdies were only observed playing SACD Rips, not with (CD Standard) PCM converted to DSD.

 

Knowledge of the exact provenance could be helpful here: if the SACD was made from PCM tracks, this is going to give you vastly different results than if the SACD was made from pure DSD recordings as any conversion is lossy.

 

Besides, SACD is DSD64 only + DRM. My interest in DSD64 waned as soon as I heard DSD128, though I still look for DSD64 if I can get some (and not with DRM or SACD iso form, if I get the latter I'll extract the individual tracks).

 

For these I use the Foobar2k DSD-PCM converter to 88.2kHz/24Bit. It sounds no worse than any notional DSD DAC I had around to play with.

 

It's important here to note which DAC and how they work internally (any internal conversion and how).

 

I found no sound difference between J-River and HQ Player, non.

 

There are people with vastly better equipment than me who would agree to strongly disagree with you about what they perceive, but so long as you're happy, go with what you like.

 

If it actually sounded better I might be willing to put up with the interface (read lack thereof), lack of usability and learning curve and all if it did anything for sound.

 

Ensure you configured the software correctly (preferences and DSD settings, post a screenshot of these to get help or advice), and then play with the different modulator and filter settings, preferably with a well-implemented native DSD DAC (like the $190 iFi iDSD Nano giant-killer).

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Muso makes a good front end for HQPlayer and can be used with other streamers concurrently.

Bonjour,

 

 

 

I agree, the birdies result from implementation, but non of the DAC le FlipFlop, but of the ADC modulator during SACD recording.

 

Both Doede Douma (the creator of DDDAC who inspired me to bypass the digital filter) John Swenson have reported similar problems. It may be we hearing the problems predicted by Lipshitz/Vanderkoy in their paper criticising DSD.

 

JS observed that the Burr Brown style analogue FIR Filter (as also found in several DIY DSD DAC's) do a good job of scrambling these tones out of existence.

 

I thin purely digital modulator (e.g. the one in the DSD converter from Foobar2k) can perform near the theoretical limits and avoid creating these tones. Real modulators in analogue hardware seem to have problems though.

 

Anyway, birdies were only observed playing SACD Rips, not with (CD Standard) PCM converted to DSD.

 

 

 

Note, I use PCM only, normally.

 

My Pass D1 DAC handles up to 96kHz/24Bit, I use M2Tech Hi-Face to drive SPDIF from a spare Lenovo Thinkpad.

 

I have been keeping some DSD/SACD images around as they seem to have better overall mastering and dynamic range than CD copies in circulation. For these I use the Foobar2k DSD-PCM converter to 88.2kHz/24Bit. It sounds no worse than any notional DSD DAC I had around to play with.

 

I found no sound difference between J-River and HQ Player, non.

 

If it actually sounded better I might be willing to put up with the interface (read lack thereof), lack of usability and learning curve and all if it did anything for sound.

 

Salud M.I.

Forrest:

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Knowledge of the exact provenance could be helpful here: if the SACD was made from PCM tracks, this is going to give you vastly different results than if the SACD was made from pure DSD recordings as any conversion is lossy.

 

If I follow M.I's fallacious logic here, he seems to be suggesting that CD (PCM) files converted to DSD with a Sigma Delta Modulator weren't exhibiting any issues, only those made originally with a modulator of the same resolution did.

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We technically can't compare PCM and DSD as format. For any recording studio workflow.

 

We can compare full systems from capture device (ADC) to playback (DAC).

 

Quality here depend on implementation (quality of device realization and using) only. No difference DSD or PCM there used.

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Bonsoir mes Amis,

 

Wasn't that paper thoroughly debunked by another paper?

 

Debunked? L & V presented a further paper in replay on the same day that thoroughly "debunked" the papers of Philips and Angus criticising theirs, not just making claims (like the papers criticising theirs), but showing the full math.

 

There are excellent modulators and filters in HQ Player.

 

I only play PCM without digital filter, I fail to see relevance.

 

Knowledge of the exact provenance could be helpful here: if the SACD was made from PCM tracks, this is going to give you vastly different results than if the SACD was made from pure DSD recordings as any conversion is lossy.

 

I think the SACD's I like are made from analogue master tape.

 

It's important here to note which DAC and how they work internally (any internal conversion and how).

 

I had the iffy stuff here, both model (so-so sound, not much resolved, plus not balanced out which is a pain), several China made "Sabre" DAC's, a Weiss Sabre DAC (they all sounded rather similar - not bad not good) and a DIY Unit with discrete DSD DAC from Japan. They all did little to convince me of the merits of DSD or PCM -> DSD conversion. For the japan DAC I needed to add my old Tube Preamp back as well, no volume.

 

There are people with vastly better equipment than me who would agree to strongly disagree with you about what they perceive, but so long as you're happy, go with what you like.

 

My equipment is very modest. I follow the ideas of Arthur Salvatore in the system setup, due to trying his advise I removed my Tube Preamplif and got my speakers.

 

I use Thinkpad plus M2TECh hiface, Win8, Fidelizer and J-River (tried J-play, no difference). This replace CEC TL1 Beltdrive CD Transport.

 

DAC is Pass D1, refurbished in 2011 (nichicon Muse capacitors replacing old tired caps) and modified with no digital filter (on idea of Mr. Doede Douma - I much like the result) and J-Fets in the I/V-Converter which is said lowers distortion.

 

The DAC directly drive Amplif Pass Aleph 2, these I refurbished in 2011 with upgrades in Capacitors (Nichicon Muse KG Super Through and other Nichicon Muse and SCR Fast Cap) and Mosfets from Fairchild (more linear).

 

The speaker is A.R.S. Acoustique System Max, think Wilson Watt/Puppy but with good sound.

 

Cables are Kimber Argent for signal and Shunyata for mains.

 

Room treatment is Acoustic Sciences.

 

The Pass D1 DAC may accept 24bit/96kHz PCM, but its nonlinear distortions (THD) are at 74dB...

 

So? Speakers distort a lot more (I measured my ones while treating room).

 

Pass uses sans-contre reaction design, so distortion at maximum level is a little high. Lower level, distortion much lower. Music is most well below maximum (more like -20dB).

 

At diyaudio publish modification with J-Fet 2SK170 for lower HD, I have applied it, sound is better I think, distortion is lower.

 

If I follow M.I's fallacious logic here, he seems to be suggesting that CD (PCM) files converted to DSD with a Sigma Delta Modulator weren't exhibiting any issues, only those made originally with a modulator of the same resolution did.

 

Observation:

 

1) Play CD-File converted to DSD64/128/256 via Le DAC FlipFlop, no birdies, okay sound, but not as good as Pass D1

2) Play SACD Rip file directly via Le DAC FlipFlop, birdies, strange sounds

 

Theory after observation:

 

Digital conversion PCM->DSD is different to using DSD ADC. DSD ADC seems less perfect than digital conversion.

 

The logic is not fallacious, but other theories are possible.

 

Salud M.I.

Magnum innominandum, signa stellarum nigrarum

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Debunked? L & V presented a further paper in replay on the same day that thoroughly "debunked" the papers of Philips and Angus criticising theirs, not just making claims (like the papers criticising theirs), but showing the full math.

 

Not just debunked, thoroughly debunked. Not sure if it was 'Philips and Angus' though and wish I could recall the other paper or the authors.

 

 

I only play PCM without digital filter, I fail to see relevance.

 

The relevance is obvious if you really want a good way to listen to DSD...

 

 

 

I think the SACD's I like are made from analogue master tape.

 

Maybe, but then you would prefer DSD128 or Quad DSD.

 

I had the iffy stuff here, both model (so-so sound, not much resolved, plus not balanced out which is a pain), several China made "Sabre" DAC's, a Weiss Sabre DAC (they all sounded rather similar - not bad not good) and a DIY Unit with discrete DSD DAC from Japan.

 

So you make your opinion of a format with iffy equipment? How about trying better ones?

 

My equipment is very modest. I follow the ideas of Arthur Salvatore in the system setup, due to trying his advise I removed my Tube Preamplif and got my speakers.

 

There's nothing wrong with modest equipment and I think one can make modest equipment sound really good with effort. I do like what I read a few years ago on Salvatore's site, especially when it comes to getting new equipment or improving on manufacturers' commercial items.

 

I use Thinkpad plus M2TECh hiface, Win8, Fidelizer and J-River (tried J-play, no difference). This replace CEC TL1 Beltdrive CD Transport.

 

I'd recommend HQ Player over JRiver any day.

 

1) Play CD-File converted to DSD64/128/256 via Le DAC FlipFlop, no birdies, okay sound, but not as good as Pass D1

2) Play SACD Rip file directly via Le DAC FlipFlop, birdies, strange sounds

 

Theory after observation:

 

Before embarking on any theory, what's the precise SACD title, how was it ripped, in what format, what's the Le DAC FlipFlop doing internally, can you record and share a segment of the 'birdies', etc...?

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2) Play SACD Rip file directly via Le DAC FlipFlop, birdies, strange sounds

 

I have never heard such form a DAC made to DSD/SACD requirements. And not from my discrete DSC1 design either (which is compliant).

 

Digital conversion PCM->DSD is different to using DSD ADC. DSD ADC seems less perfect than digital conversion.

 

Get for example some of the Opus 3 DSD128 recordings, made using ADC built around PCM4202 true 1-bit ADC.

Signalyst - Developer of HQPlayer

Pulse & Fidelity - Software Defined Amplifiers

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So? Speakers distort a lot more (I measured my ones while treating room).

 

So, that's not quite 24bit performance, not even close to it (74dB).

 

Theory after observation:

 

Digital conversion PCM->DSD is different to using DSD ADC. DSD ADC seems less perfect than digital conversion.

 

The logic is not fallacious, but other theories are possible.

 

Salud M.I.

 

Your theory is manifestly flawed. You're talking about the same resolution of a modulator, only in the latter case (direct capture from the modulator) you're bypassing several processing steps that introduce ringing and aliasing. You'd be surprised, btw, how many PCM recordings started out as DSD, and no one ever heard any frickin birdies on them - ditto for original DSD recordings from Channel and others that receive highest marks for sound quality, Grammys, Stereophile's Recording of the Month Awards, etc

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Bruce had a DSD ADC comparison here, but unfortunately the files don't seem to be online anymore:

DSD Battle Royale!!

 

Those files are nice for comparison and I also ran some analysis on the files too.

Very cool. I asked Bruce if he could point me to the files.

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Very cool. I asked Bruce if he could point me to the files.

 

I think there were Ella - 'Black Coffee' and Crosby, Stills, Nash (perhaps Young) and a third one.

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Very cool. I asked Bruce if he could point me to the files.

 

It was absolutely wonderful, and I really hope he can put them back up. I was *very* surprised at the degree of variation in sound between ADCs. It's something I'd always thought of as having variations, but small variations in a fairly cut-and-dried process. Unh-uh.

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Bruce had a DSD ADC comparison here, but unfortunately the files don't seem to be online anymore:

DSD Battle Royale!!

 

Those files are nice for comparison and I also ran some analysis on the files too.

 

They were terrific for comparison, until I bollixed up an OS X upgrade and lost them a while ago. I'd of course be very interested in hearing more about your analysis and anything you were able to draw from it.

One never knows, do one? - Fats Waller

The fairest thing we can experience is the mysterious. It is the fundamental emotion which stands at the cradle of true art and true science. - Einstein

Computer, Audirvana -> optical to EtherREGEN -> microRendu -> ISO Regen -> Pro-Ject Pre Box S2 DAC -> Apollon Audio 1ET400A Mini (Purifi based) -> Vandersteen 3A Signature.

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Bruce had a DSD ADC comparison here, but unfortunately the files don't seem to be online anymore:

DSD Battle Royale!!

 

Those files are nice for comparison and I also ran some analysis on the files too.

 

I wish Bruce could also test the Sony DSD128 ADC that was used on this Cat Stevens release.

 

Cat Stevens-Tea For The Tillerman-DSD Double Rate 56MHz128fs Download|Acoustic Sounds

 

Gus Skinas apparently favored the converter over the EMM Labs ADC8 MkIV.

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Just in case there are still some questions about the possibility of accurately doing sample rate conversions from PCM to PCM with accuracy down to the dither noise, here is what I did.

 

1. I took a music file (88/24) that had lots of high frequency content and downsampled it to 32/24

2. I resampled the 32/24 to 44/24. The result was an apodized file that had essentially zero energy above 18 kHz. This is the reference file for future comparisons.

3. I upsampled the reference file to 192/24 format

4. I downsampled the 192/24 back to 44/24 producing the output file.

5. I nulled the output file against the reference file to create a difference file.

 

Except for a brief start-up transient, the two files differs by no more than +-2 LSB (maximum level of dither noise). Boosting the difference file 100 dB made it easily audible as white noise. A spectral plot of the boosted file showed that the 65K Blackman Harris FFT was at the -75 dB level, corresponding to the unboosted difference file having noise at -175 dB in the FFT bins.

 

Software used to do the resamplings was iZotope RX4 advanced with SRC sampling slope 32, offset shift 1.0, preringing 1.0. If you try this with an earlier version of the iZotope 64 bit SRC you will not get a good null, because of the sub-sample offset used by earlier versions of this software. This is a pretty good practical demonstration of the Sampling Theorem at work using commercially available software on real music. I deliberately chose the two sampling rates as non-integer multiples because some people have questioned the possibility of upsampling and downsampling with non-integer formats.

 

I suspect it would be possible to get even deeper nulls by using 32 bit integer format and I don't believe it would be necessary to use such brutal apodizing of the original signal if I had used a sharper filter. If anyone is curious about other possibilities I suggest they do similar experiments.

 

Why is this relevant to DSD? It's because I do not believe it is possible to pass similar tests using DSD64 and probably also DSD128. It's also relevant because it shows that a 23 bit null shows up in spectral plots at -175 dB level, basically showing that comparisons at -130 dB indicate junk performance.

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Just in case there are still some questions about the possibility of accurately doing sample rate conversions from PCM to PCM with accuracy down to the dither noise, here is what I did.

 

1. I took a music file (88/24) that had lots of high frequency content and downsampled it to 32/24

2. I resampled the 32/24 to 44/24. The result was an apodized file that had essentially zero energy above 18 kHz. This is the reference file for future comparisons.

3. I upsampled the reference file to 192/24 format

4. I downsampled the 192/24 back to 44/24 producing the output file.

5. I nulled the output file against the reference file to create a difference file.

 

Except for a brief start-up transient, the two files differs by no more than +-2 LSB (maximum level of dither noise).

 

I wonder what happens with an impulse response during these sample rate conversions...

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1. I took a music file (88/24) that had lots of high frequency content and downsampled it to 32/24

2. I resampled the 32/24 to 44/24. The result was an apodized file that had essentially zero energy above 18 kHz. This is the reference file for future comparisons.

 

Tony,

 

Does you mean 16 kHz?

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Tony,

 

Does you mean 16 kHz?

 

The file that I used as the reference file was in the 44/24 format with energy extending up to about 18 kHz on the spectrum plot. Downsampling to a 32 kHz sampling rate was a convenient way to cut off audio above 16 kHz and the filter that I used was not terribly steep, hence there was still detectable energy up to 18 kHz. This left a 4 kHz gap between the energy in the test signal and Nyquist frequency for the 44/24 sample rate, making subsequent filters non-critical.

 

(16 kHz is the Nyquist frequency for a sample rate of 32 kHz.)

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