Dr. AIX's POSTS — 23 May 2015

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The discussion over at CA also included a great deal of push back on the DSD vs. PCM debate. One of the writers presented numerous lengthy comments about how modern ADC and DAC converters operate. Recall that the title of the post was “Mark Waldrep is claiming that PCM 24/96 is superior to DSD”.

I highlighted the comments made by this gentleman because he specifically called out both John Siau and myself.

“Let’s look at the recording and playback chain:

Virtually all ADCs take the analog signal through sigma-delta modulation into DSD format internally as a first step. It can either be left as DSD, or if the studio is ‘recording to PCM,’ internally the DSD format will be decimated to higher or lower resolution PCM. Note that this conversion from DSD format to PCM (and to lower rate PCM from higher rate) is a lossy one.

Then nearly all DACs (all delta-sigma DACs) take the bitstream, and if it is lower resolution PCM they first interpolate it to higher resolution PCM, then send it through a sigma-delta modulator to obtain a DSD bitstream. All the interpolation and modulation steps are lossy conversions. The DSD bitstream then goes through the final reconstruction filter for conversion back to analog.

That’s what happens with a ‘PCM’ recording. Everything you read where people complain about the deficiencies of DSD describes everything you have ever listened to through a DAC, the sole rare exception being if you listen only to material recorded before the late-80s/early-90s takeover of sigma-delta ADCs and do so only through an R2R DAC. That’s it. So when John Siau and Mark Waldrep talk about the sound of DSD, they are describing an unavoidable stage of every 24/96 recording Mark Waldrep ever made and everything John ever heard from his sigma-delta DACs. If they say they prefer the ‘sound of PCM,’ then what they must prefer is what is being lost during the lossy conversions to and from DSD format.”

So I sent this description of the process of AD and DA conversion to John for his thoughts. He replied below:

“Modern PCM sigma-delta converters produce much lower error signals than 1-bit sigma-delta DSD converters. The errors in the DSD system are due to the 1-bit quantization that occurs in 1-bit sigma-delta DSD converters. Multi-bit PCM sigma-delta converters can be fully dithered and do not suffer from this un-dithered truncation. Every added bit reduces the noise signal by 6 dB. A 4-bit sigma delta converter is 24 dB quieter than a 1-bit sigma-delta DSD converter. Right from the start, 1-bit DSD signals have higher losses than multi-bit PCM signals.

Conversion from 1-bit DSD to multi-bit PCM is a lossless process inside the audio band. The only thing that is removed is the out-of-band noise above the Nyquist limit of the PCM system. Nothing else is lost. Don’t believe the DSD marketing hype.

Conversion from multi-bit PCM to 1-bit DSD is always a lossy process. The loss is due to the 1-bit truncation. This truncation introduces a very large ultrasonic error signal that makes the ultrasonic region unusable for audio.

Processing a 1-bit signal to create a 1-bit signal is also always a lossy process. A volume control is one of the simplest processes in a multi-bit PCM system, but it creates a large error signal when applied in a 1-bit DSD system. The same is true for any other 1-bit to 1-bit DSP process. The lossy part of these DSP processes is the quantization back to 1-bit. Cascaded 1-bit truncation processes can rapidly degrade the audio quality. For this reason, DSD is always processed as multi-bit PCM.

Any DSP process applied to a 1-bit signal produces a multi-bit signal. No loss of information occurs until this is quantized back to a 1-bit signal. Why incur the loss by going back to a 1-bit signal after the processing chain?

All practical DSD systems require some sort of DSP processing (gain control, mixing, filtering, etc.) and all of these processes produce multi-bit PCM results. Taking these lossless multi-bit results and adding loss by truncating them back to a 1-bit DSD signal makes absolutely no sense. DSD complicates the processing and adds unnecessary losses to the signal path. DSD does not simplify the signal path. There is absolutely no truth to the marketing hype that claims that 1-bit DSD is a simpler system than multi-bit PCM. The exact opposite is true.

John Siau, Benchmark Digital”

I’ll parse through John’s response in a future post but the gist of the argument is that the CA comment conflates 1-bit DSD with PCM sigma delta conversion. Yes, modern converters make use of internal sigma delta conversion at very high sample rates BUT they are not 1-bit DSD processes, they use multiple bits making them very different than DSD.

Thanks John. It’s really nice to have really smart friends.

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About Author

Dr. AIX

Mark Waldrep, aka Dr. AIX, has been producing and engineering music for over 40 years. He learned electronics as a teenager from his HAM radio father while learning to play the guitar. Mark received the first doctorate in music composition from UCLA in 1986 for a "binaural" electronic music composition. Other advanced degrees include an MS in computer science, an MFA/MA in music, BM in music and a BA in art. As an engineer and producer, Mark has worked on projects for the Rolling Stones, 311, Tool, KISS, Blink 182, Blues Traveler, Britney Spears, the San Francisco Symphony, The Dover Quartet, Willie Nelson, Paul Williams, The Allman Brothers, Bad Company and many more. Dr. Waldrep has been an innovator when it comes to multimedia and music. He created the first enhanced CDs in the 90s, the first DVD-Videos released in the U.S., the first web-connected DVD, the first DVD-Audio title, the first music Blu-ray disc and the first 3D Music Album. Additionally, he launched the first High Definition Music Download site in 2007 called iTrax.com. A frequency speaker at audio events, author of numerous articles, Dr. Waldrep is currently writing a book on the production and reproduction of high-end music called, "High-End Audio: A Practical Guide to Production and Playback". The book should be completed in the fall of 2013.

(13) Readers Comments

  1. Last month Peter Aczel of the Audio Critic published his “What I have learned after six decades in audio (call it my journalistic legacy)”. Although I may not agree with what he wrote 100%, there’s more truth here than what I read in 95% of what is written in all the hi end mags and websites. Here’s a bit of his thoughts on digital audio, read the rest here.
    http://www.theaudiocritic.com/audio-legacy-2/

    “We should all be grateful to the founding fathers of CD at Sony and Philips for their fight some 35 years ago on behalf of 16-bit, instead of 14-bit, word depth on CDs and 44.1 kHz sampling rate. Losing that fight would have retarded digital media by several decades. As it turned out, the 16-bit/44.1-kHz standard has stood the test of time; after all these years it still sounds subjectively equal to today’s HD techniques—if executed with the utmost precision. I am not saying that 24-bit/192-kHz technology is not a good thing, since it provides considerably more options, flexibility, and ease; I am saying that a SNR of 98.08 dB and a frequency response up to 22.05 kHz, if both are actually achieved, will be audibly equal to 146.24 dB and 96 kHz, which in the real world are never achieved, in any case. The same goes for 1-bit/2.8224 MHz DSD. If your ear is so sensitive, so fine, that you can hear the difference, go ahead and prove it with an ABX test, don’t just say it.”

    • Thanks Sal, I read the article and have a great deal of respect for Peter. A lot of what he states is interesting and has merit but he misses the mark on a number of important issues as well. Even the segment you quoted above contains some factual errors and and bad information. CDs don’t have a dynamic range of 98.08 unless you don’t count the dithering that is present all of the time…the number is actually 94 dB. Same problem with the 24-bit number. His evaluation of high-resolution audio equates high-res PCM with DSD 64…which are not remotely similar. Go figure.

      • I’d give Pete a break on his digital math, this stuff requires at least a masters in engineering to understand. Even you called on John Siau to pen the response to CA.
        I think the point Peter (and myself) are trying to make here is that a Red Book CD “if executed with the utmost precision” would be very hard to differentiate from a HD recording also done with the same utmost precision. Not impossible, but very difficult, requiring trained ears and excellent equipment.
        Also what he has done is call out for ABX trial of the different processes and including a DSD file would make for a very interesting outcome.
        I already proven to myself with your files that I can’t hear the deference. LOL

        • I agree with your assessment. CD, when done with great care, can produce absolutely stunning results. When a good CD is compared to a good HD-Audio recording, they are very difficult to differentiate.

  2. Thank you for this post, most enlightening.

  3. Mark,

    While I’m sure that John Siau’s response is correct, it leaves unsaid the most important rebuttal point – the assertion that converting analog to PCM involves going through DSD first. If that’s untrue, the rest of the DSD faithful’s argument fails from the outset, as it’s based on a fallacy.

    I don’t work in the field, so I don’t know whether it’s true or not, but I seriously doubt it could be true, and that going from analog through DSD to get to PCM is ever done except in the rare case of making a PCM version of something archived in DSD.

    As to the bit yesterday about “multi-channel mono,” that would be true if only one mike was used per instrument; using two enables positioning a stereo image of each instrument in the overall mix. Maybe you should use five for locating it in the oveall surround mixes!

    • Phil, the central issue that John makes very clear is that “converting analog to PCM” does not involve a path through DSD encoding, unless you make the error of equating multibit SDM (sigma delta modulation) with 1-bit DSD. It is extremely rare to find a production that was created in DSD and them converted to PCM. I believe that Blue Coast does it and maybe a couple of others but it makes no sense. Conversions the other way…from PCM or DXD…to DSD does happen. It’s also a conversion that lowers the fidelity of the original recording but some see greater value in DSD and thus do these conversions and charge more for them.

      The level of misinformation regarding DSD is amazing!

  4. John’s response is also a bit of a cop-out. The poster on CA was using “DSD” as a catchall to refer to SDM as “DSD” is a more well known acronym. So the poster on CA is correct that nearly all modern DACs convert PCM to SDM (in this case it could be called multi-bit DSD in the vernacular). This is clear if you follow the whole discussion and know the previous posts made by this poster.

    John’s response is a literalist one ( as he is reacting only to your quote), and interprets the post as seeming to say that modern DACs convert all PCM to 1 bit DSD. But that isn’t what the post was saying.

    Almost all modern DACs DO convert PCM to SDM, so the point of the CA poster is well taken (convert DSD here to SDM or “multi-bit DSD” as it is sometimes called for convenience):

    That’s what happens with a ‘PCM’ recording. Everything you read where people complain about the deficiencies of DSD describes everything you have ever listened to through a DAC, the sole rare exception being if you listen only to material recorded before the late-80s/early-90s takeover of sigma-delta ADCs and do so only through an R2R DAC. That’s it. So when John Siau and Mark Waldrep talk about the sound of DSD, they are describing an unavoidable stage of every 24/96 recording Mark Waldrep ever made and everything John ever heard from his sigma-delta DACs. If they say they prefer the ‘sound of PCM,’ then what they must prefer is what is being lost during the lossy conversions to and from DSD format.”/

    So the poster on CA, taken in context, is correct.

    • Sorry, dyslexia strikes. Meant DSM (Delta Sigma Modulation), not SDM.

    • Sorry Danny, I did read a great deal of the poster’s comments. Equating DSD (which is universally marketed a “1-bit” SDM) is not the same a multibit SDM. There is not such thing as multibit DSD. When you move past 1-bit, it becomes PCM and dithering becomes a possible. The thought that John’s equipment and my recordings have passed through a DSD (1-bit SDM) stage is wrong.

    • The real path inside the modern DAC’s:
      PCM stream –> INTERPOLATION FILTER (decimation) –> one or multibit DSM conversion –> DSD stream –> SC DAC –> ANALOG signal

      If the DAC DSD capable the path is sorter: DSD stream –> SC DAC –> ANALOG signal
      The DSM conversion can be one or multi bit, the multi bit DS modulator give better quality DSD stream on the output.
      The AD conversion is the same in the reverse way. The DSD –> PCM or PCM –> DSD conversion is always lossy.

      • thanks for clarifying , but as someone pointed out, are all modern recordings done in DSD (1 bit stream) ? I know Sony did/maybe still does on their redbook CDs in the 90s, but they don’t even have SACDs (at least not common)
        I thought they are all made driectly as 24 bit recording (at least as stated on the CDs), in other words,
        PCM straight away without all those conversion they talk about.
        So are all these so called 24 bit recordings are really fake ones but are 1 bit that turn to 24 bits and then you can down load them as 24 bit Hi RES FLAC or they may have SACD versions but that is also really from DSD to PCM and back to DSD process ???! I am confused, the more I read!

  5. Brilliant, educational article. Thank you.

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