PCM to DSD...the new PS Audio DAC

[Alrainbow]

I have the upgrade on order too. I am hoping if is in that realm as well. As for 2 DSD after I heard Myles , system it sounds really close not sure it beats his tape though . As I guess the tape beats vinyl as he showed me , but his vinyl is something that is just fantastic too. Only when he went back and forth could I tell the tape was better. I cannot imagine the direct stream getting close to his analog system.

Al

 

[Orb]

I do not readily see why that should be. Understand, that DSD and current production PCM application audio converters are all based on sigma-delta technology, with the following three primary differences between the two. Pure DSD converters have 1-bit of native resolution, while PCM application converters have 5-bits or 6-bits of native resolution. This means that PCM application converters have much higher native dynamic range than do DSD converters. The second main difference is that DSD utilizes much stronger noise-shaping to effectively increase audio band dynamic range than do the PCM application converters. DSD must include strong noise-shaping, simply because it's native resolution is so low. However, since both converters are sigma-delta based, both utilize some degree of noise-shaping. The third main difference is oversampling bandwidth, the spectrum of which, sigma-delta converters utilize to relocate the quantization noise stemming from their low native resolutions. DSD sigma-delta needs greater oversampling bandwidth than do PCM application sigma-delta converters, because DSD has much lower native resolution.

This situation is complicated, however, because DSD is necessarily recorded at a much higher native sample rate than PCM for the reasons discussed above. This fact has key implications for the signal reconstruction filter design, which, in my opinion, is what primarily gives DSD it's particular sound character. Taken in total, it's not readily apparent to me why converting PCM to analog via 1-bit sigma-delta (DSD) should sound better than converting PCM to analog via 5-bit sigma-delta. However, subjective performance is usually determined by the implementation details, and so, perhaps, an answer is yet to be found therein.

Regarding filter design, yeah I would say this provides sound character for both DSD and also PCM.
Just to clarify though; not all DACs utilise SDM and also part of such discussions (after all the context should be whole chain including how recorded) worth considering also the ADCs and if used digital microphone (although these comes back to PDM and round about way DSD/SDM).
Not disagreeing BTW, just adding to the discussion.

Cheers
Orb

 

[tailspn]

My 2 cents.
Sigma Delta - E d. Ok I cannot write the mathermatical symbols here. What this means mathameticall is to add up all (Sigma = summation) of all the small changes (delta).

DSD is just adding those changes digitally to the prior voltage level and then smoothing it out using a low pass filter. Not complicated. What makes it complicated is the noise and the prior voltage level.

PCM is a absolute sample of the full voltage level at a particular interval. 24bit/192khz is the 24 bit absolute digital representation of the signal taken every 192,000 times a second. The PCM signal does not rely on the last voltage level to decide on the current level. A low pass filter is also used to clean out noise although a low pass is critical in the beginning.

May I suggest another way of looking at DSD. People get confused by looking at it as a digital format. It isn't, it's analog. While it's storeable digitally, since it's restricted to two levels, it has no values associated with it. It's actually an analog modulation process of a simple squarewave bitstream, running at 2.82MHz for 64fs DSD.

The process is very simple, and is explained here:

http://www.embedded.com/design/debu...-of-sigma-delta-analog-to-digital-converters-

It's as analog as FM radio, where a RF carriers's frequency is modulated by the instantaneous audio signal amplitude. In DSD, which is actually a coined marketing term for 1-bit two level Pulse Density Modulation (PDM), a bit stream's density is modulated by the instantaneous analog signal amplitude. The higher the instantaneous analog level, the greater the bit density. The Scarlet Book spec chose 0dB to equal 50% modulation, leaving +6dB headroom.

An important point is that, unlike PCM (which many confuse with multi-bit two level PDM), there is no value represented. To obtain a value, a multi-bit PDM word has to be generated, which then has a mathematical absolute value/weight relationship between the bit streams, and therefore operable in a digital computer. But it has no greater data resolution. To get more resolution, you increase the bitstream rate.

The point of all this is recognizing all acoustic to electric transformation (A/D conversion) begins with a variant of Delta-Sigma Modulation, which operates at least 8X any PCM sample rate. To yield PCM (a series of 2's compliment binary word values, like frames in a motion picture film strip) the PDM bit stream(s) must be filtered (decimated) to remove all energy above the half frequency (Nyquist) of the new lower PCM word rate, then be converted to a value/word based system. None of that comes without some cost to sound quality.

 

[Orb]

Although tailpn,
this also may be a bit confusing because of MEMs microphone (focusing on digital models) using PDM and also compatible with the DSD stream/interface of chip-controllers; not all due to some specs but quite a few are.
In that context I would say it is perceived as digital.

More semantics than anything, and I appreciate all 3 of us are not disagreeing.
Cheers
Orb

 

[tailspn]

Yes, PDM, either 1-bit two level (DSD) or multi-bit two level can be interfaced with a digital system/controller, given that there's only two levels. But there's still no absolute values represented, so that has to be generated in the receiving element.

My objective was to shed some light over the confusion of multi-bit PDM and PCM being used interchangeably. They're very different. Also, PCM, in all instances at the A/D level is derived from a PDM Delta-Sigma Modulator, so can hardly be more accurate.

Best,

Tom

 

[adyc]

May I suggest another way of looking at DSD. People get confused by looking at it as a digital format. It isn't, it's analog. While it's storeable digitally, since it's restricted to two levels, it has no values associated with it. It's actually an analog modulation process of a simple squarewave bitstream, running at 2.82MHz for 64fs DSD.

The process is very simple, and is explained here:

http://www.embedded.com/design/debu...-of-sigma-delta-analog-to-digital-converters-

It's as analog as FM radio, where a RF carriers's frequency is modulated by the instantaneous audio signal amplitude. In DSD, which is actually a coined marketing term for 1-bit two level Pulse Density Modulation (PDM), a bit stream's density is modulated by the instantaneous analog signal amplitude. The higher the instantaneous analog level, the greater the bit density. The Scarlet Book spec chose 0dB to equal 50% modulation, leaving +6dB headroom.

An important point is that, unlike PCM (which many confuse with multi-bit two level PDM), there is no value represented. To obtain a value, a multi-bit PDM word has to be generated, which then has a mathematical absolute value/weight relationship between the bit streams, and therefore operable in a digital computer. But it has no greater data resolution. To get more resolution, you increase the bitstream rate.

The point of all this is recognizing all acoustic to electric transformation (A/D conversion) begins with a variant of Delta-Sigma Modulation, which operates at least 8X any PCM sample rate. To yield PCM (a series of 2's compliment binary word values, like frames in a motion picture film strip) the PDM bit stream(s) must be filtered (decimated) to remove all energy above the half frequency (Nyquist) of the new lower PCM word rate, then be converted to a value/word based system. None of that comes without some cost to sound quality.

Is multi-bit DSD/SDM digital or analog?

 

[tailspn]

Is multi-bit DSD/SDM digital or analog?

Confusing terms. DSD is only 1-bit two level PDM. Since it's purely a bit density modulation scheme with no associated value, it's analog. Multi-bit two level PDM is also a bit density modulation scheme, but there's a weight relationship between the parallel bit streams. The output of the first multi-bit DS Modulator still has no absolute values, and is usually where the decimation filtering begins. At some further point it can be a series of values, but up to that point, it's analog. Now you're over my head, for it's individual A/D Converter design dependent. Here's the hot kid on the block converter that shows the process:

http://www.ardatech.com/1201.shtml

 

[mep]

May I suggest another way of looking at DSD. People get confused by looking at it as a digital format. It isn't, it's analog. While it's storeable digitally, since it's restricted to two levels, it has no values associated with it. It's actually an analog modulation process of a simple squarewave bitstream, running at 2.82MHz for 64fs DSD.

That may explain why so many people (including me) say that DSD sounds like analog and explains my preference for DSD over PCM.

 

[awsmone]

I think was a wonderful explanation by Tom

This issue was covered in great detail in computer audiophile

I think the whole schema from production points out the issue is really "moot"

Other than a few very rare 1 bit converters, most A/D is done in DSM and then turned into DSD or PCM as formats

Also 1 bit DSD cannot be I understand manipulated so any editing mixing etc is usually done in DSM or pcm or even analog ( i think there is a way to splice)

Anyway my point is we start generally with DSM so saying one format is better is like saying which cordial tastes better in dirty water lol

Also the subsequent production and implementation has more of an effect that the format I believe

People talk about DAC which actually often covers a lot of components in the conversion process all of which effect sound

http://www.computeraudiophile.com/f.../multibit-direct-stream-digital-debate-18437/

 

[Orb]

Yes, PDM, either 1-bit two level (DSD) or multi-bit two level can be interfaced with a digital system/controller, given that there's only two levels. But there's still no absolute values represented, so that has to be generated in the receiving element.

My objective was to shed some light over the confusion of multi-bit PDM and PCM being used interchangeably. They're very different. Also, PCM, in all instances at the A/D level is derived from a PDM Delta-Sigma Modulator, so can hardly be more accurate.

Best,

Tom

Totally agree Tom; and why I ended up in a stupid technical argument awhile ago in a different thread where their narrative is to try and say they are the same sigh; why I like to use these days MEMs PDM microphone as an example as it is clearly PDM not PCM and also integrates-interfaces in most cases into DSD chips-solutions showing the relationship between PDM and DSD.
Since that discussion though given up discussing technical aspects most of the time now.
Thanks
Orb

 

[jeremya]

Thank you for your response, Ken, although I must say it confirms my confusion more than clarifies!

Check out this presentation series by Andreas Koch:

 

[wyan]

Has anyone got the chance to compare the this PS Audio DAC to lampizator's? Any thoughts? Thanks.

 

[wisnon]

Yes, my pal AL has both and contact me or him for details offline.

 

[bmoura]

Yes, my pal AL has both and contact me or him for details offline.

You can read Al's comparison of the MSB, DirectStream and Lampizator "Big 7" on the Head-Fi forum as well (see page 5, comment # 68)

A key section there "I have not done a direct comparison as yet, but clearly the Lampizator is way above my DirectStream. I do feel the Lampizator is as good or even better than my MSB. But in different types of presentation. When I get back from my vacation in NC. I will, do a A/B to map out better the difference in them. "

http://www.head-fi.org/t/715155/lampizator-thread-comments-reviews-tuning/60#post_10681863

 

[wyan]

Thanks, Wisnon and Bmoura, I am looking for digital source that could get close to tape/LP system. I guess this type of DSD dac's could be the answer.

 

[bmoura]

Thanks, Wisnon and Bmoura, I am looking for digital source that could get close to tape/LP system. I guess this type of DSD dac's could be the answer.

Some of the DSD downloads now available from sites like Blue Coast/Downloads Now, DSD File, NativeDSD plus the Analog Tape transfers and DSD recordings on eOnkyo Music and Super HiRez are just excellent musically and sonically.
And in some cases, like Blue Coast/Downloads Now and the "Just Listen" section of NativeDSD, they even offer free DSD music files to try! Definitely worth listening to.

 

[j.phelan]

It should be known - after all these years - that DSD is a multi-bit recording process. Which means it decimates and oversamples - just like good old PCM.

On playback, all DSD DACs transcode to PCM. I suspect most didn't know this -

Finally, the sample rate (on SACD playback) is 25-30kHz - depending on the microphone used. Mikes have limits - but they would truncate anyway, to remove the large amounts of noise above 30kHz.

So - DSD is PCM - not the other way around....

 

[tailspn]

So - DSD is PCM - not the other way around....

Actually, DSD is analog. It's a 1-bit two level pulse density modulation (PDM) process where the bit density is proportional to the modulating signal level. There's no digital value(s) associated with the bit stream like PCM. Only the varying density of the clock bits. PCM is like frames in a movie film, where each frame is a stand alone digital value (2's complement binary digital word) of the signal level at the time of the sample. The two modulation processes (PDM amd PCM) couldn't be more different.

Multi-bit Pulse Density Modulation (PDM), as found in the majority of A/D converter front ends are also not PCM, for there still no value associated with the bit rate. There is however a binary relationship between the multi bit streams (usually 5 to 8), which makes the multi-bit streams process-able.

None of the recordings I've been associated with are decimated PCM, although that's a very common feature in most A/D converters to output PCM as an on-chip extra process. As long as you only want to change levels and do edits, there no reason to preform a DSD to PCM decimation conversion. It can be done at the bit rate in multi-bit PDM.

Also, all DAC's do not "transcode to PCM".

I do agree that microphones used in acoustic recordings have virtually no sensitivity above 25KHz.

 

[Andre Marc]

It should be known - after all these years - that DSD is a multi-bit recording process. Which means it decimates and oversamples - just like good old PCM.

On playback, all DSD DACs transcode to PCM. I suspect most didn't know this -

Finally, the sample rate (on SACD playback) is 25-30kHz - depending on the microphone used. Mikes have limits - but they would truncate anyway, to remove the large amounts of noise above 30kHz.

So - DSD is PCM - not the other way around....

Gee, lots of misinformation here. As Tailspn noted

 

[rockitman]

Actually, DSD is analog.

R u kidding me ? Please state your case. You complete post fails to convince me.