John Siau: Benchmark Audio Guru
John Siau is one of the principals and Director Of Engineering at Benchmark Media, makers of both professional and consumer audio equipment. He has almost 40 years of experience designing high-end analog and digital circuits for use in a variety of applications. One of his recent projects was the upgrade to the Benchmark DAC1 HDR, which is now the DAC2 HGC, which includes DSD conversion. I spoke to him about DSD technology after reading a few of the white papers on the topic. The illustrations below were taken from a paper by Andreas Koch, a principal in Playback Designs (a hardware company that manufacturers DACs and advocates for the DSD format) from an article entitled, “DSD – the new Addiction”.They got me thinking about the DSD format.
Transcription of John Siau Interview April 3, 2013
MW: So let’s start with the basics of conversion technology shown in Figure 1 of the DSD document. What’s going on there?
JS: The block diagram shows a 1-bit oversampled ADC feeding PCM data to a 1-bit DAC. This topology was typical in the 1990’s but does not apply to most PCM converters manufactured after about the year 2000. Virtually all of today’s PCM converters use oversampled 4-bit conversion. Oversampled 1-bit converters are a relic from the past. The additional 3 bits improve the SNR of the overall conversion system while greatly reducing the amount of noise shaping that is required.
Figure 1 – A block diagram of the conversion from analog to digital in PCM and DSD (Click to enlarge)
MW: Explain the change to multi-bit oversampled converters
It was the move to 24-bit conversion that ultimately made the 1-bit delta-sigma converter obsolete.
JS: In the 90s the music industry began to mix and process audio in digital audio workstations (DAWs). The first DAWs used 16-bit processing, and recording engineers quickly recognized that this 16-bit processing introduced noise and distortion. Manufacturers responded with 18, 20, 22, and 24-bit systems. It was the move to 24-bit conversion that ultimately made the 1-bit delta-sigma converter obsolete. Benchmark’s first step beyond 1-bit conversion delta-sigma was the DAC2004, a 20-bit DAC that was introduced in 1997. It had two 1-bit delta-sigma DACs wired in parallel. This configuration produced a 3 dB improvement over existing systems. With 4-bit delta-sigma converters we can now achieve a 130 dB SNR. This is a full 10 dB better than the channel capacity of 64x DSD. A 1-bit system simply doesn’t have enough room in the format for both dither noise and the audio signal. DSD is limited to a 120 dB SNR over the audio band. You can pass an audio signal that’s partially dithered or an audio signal that has no dither but there’s not enough room to pass a fully dithered audio signal. You need more than 1-bit in order to be able to do that.
A 1-bit system simply doesn’t have enough room in the format for both dither noise and the audio signal.
MW: OK, so DSD may have a reduced SNR, but the simplicity of the data path must offer some sonic advantages.
JS: The dotted lines in figure 1 show how DSD can create a bypass path to eliminate several DSP blocks in the ADC and DAC (but this bypass assumes 1-bit conversion at each end). If DSD were designed today, we would probably consider using 4-bits instead of 1. The problem is that 1-bit DSD is nearly impossible to process. In the studio, DSD is processed as 8-bit data (or wider) at DSD sample rates, in a format known as DSD-wide. The additional DSD-wide data bits reduce the amount of noise shaping that must be applied in each processing step. Bottom line, the 1-bit DSD bypass shown in the diagram doesn’t really exist except in the very simplest direct-to-disk DSD recording.
MW: DSDs are not mixed in DSD?
JS: No, DSD signals are not 1-bit wide in the DAW. DSD becomes 4-bits wide or 8-bits wide or 16-bits wide depending on what word length the DAW can handle.
MW: Really. So is it true to say as a complement to what they say in here that every DSD file or project actually goes through a PCM stage as well?
JS: Shh! Please don’t use the PCM word! But OK yes, it is probably safe to say for about 99% of the DSD projects that have been done. There may be a few exceptions where somebody set up a microphone to a DSD direct to disc project and have done nothing with it. No processing, no level changes, no filtering, no mixing or have used that for archiving an analog tape…or archiving a vinyl disc.
MW: Those are rare.
JS: Yes. We’re going to do multichannel recording, we’re going to have to do mixing, we’re going to have to do EQ, we’re going to have to do various effects and all that has to be done in PCM. Now the PCM can all be done at DSD sample rates, and that’s fine. That’s well and good. The conversion from DSD to PCM is really an automatic thing, a very benign conversion. If you don’t change the sample rate, there is no loss of quality when expanding the word length. As soon as you do any mathematical operation on the DSD, the word length expands. And so DSD goes from being 1-bit to being multiple bits. The DSD DAW manufacturer chooses how many bits of precision they wish to preserve in the processing – the more the better.
MW: Is that an example of what Pyramix or the Sonoma system is doing? I mean productions that are actually happening in that domain, are they doing all these PCM steps and just keeping it under the covers?
JS: That’s exactly what they’re doing. Yeah. It’s PCM at the DSD sample rate. But, nothing bad happens up until that point. The loss of quality only comes when multi-bit PCM is dithered back down to 1-bit DSD. When you dither down to 1-bit, you’re adding huge amounts of quantization noise. Any 1-bit DSD signal has a 6 dB signal to noise ratio – at best (when the ultrasonic noise is included in the measurement). The noise situation gets worse when we have two cascaded multi-bit to 1-bit conversions (once in the ADC, and once in the DAW).
MW: Right.
JS: And so you have to apply very, very aggressive noise shaping to keep the in band noise down but this comes at the expense of a tremendous amount of out of band noise.
MW: Yeah, that’s the purple haze you see at the high end of the spectrum in DSD recordings.
JS: The spectrum analysis of DSD shows a huge amount of noise at high frequencies. You’ve got 6 dB of signal to noise ratio – at best.
…if you measure the signal to noise ratio of that whole wide band signal before you go through the analog low-pass filter, which they like to conveniently ignore, but it’s there in all the DSD systems. You’ve got 6 dB of signal to noise ratio.
MW: Back up on that a second. You mean the end result, if you include the ultra sonic noise maxes out at 6 dB signal to noise ratio?
JS: Yep, it can’t be any better than that.
MW: And so what a mastering engineer or an equipment manufacturer would do is simply would apply a low pass filter to remove the “purple haze” out of the equation.
JS: Unfortunately, the ultrasonic noise cannot be removed in mastering unless the DSD source is being transferred to PCM. The ultrasonic noise is always present in DSD signal, it cannot be removed until the DSD signal is converted to analog or to PCM. This means that the noise must be removed in the playback hardware. If the DSD DAC is equipped with a well-designed analog low-pass filter, we can achieve signal to noise ratios that start to rival some of the better PCM systems. DSD doesn’t approach the 144 dB SNR performance of a 24-bit system , but it certainly exceeds the -96 dB SNR performance of the CD format. With a well-designed filter, DSD can achieve a 120 dB signal to noise ratio, roughly equivalent to a 20-bit PCM system.
MW: They talk about making 20-20 kHz just stellar and they don’t really worry about anything higher than that because of this whole noise-shaping dilemma.
JS: Yeah. The problem is that that DSD marketing materials often show a nice, well-formed high frequency square wave. But, this waveform only exists before the analog low pass filter. It looks very different after the analog low pass filter. To his credit, Andreas Koch didn’t show the square wave in his paper but it’s something that does appear in many DSD marketing materials.
MW: Yeah. I’ve seen it in the standard Sony DSD white paper.
JS: So as far as that Figure 1 is concerned, the conversion from DSD to PCM is a very benign conversion. The conversion from PCM back to DSD is where all the problems occur. If you can avoid ever going back to 1-bit, you’re much better off. For this reason, all modern DACs avoid dithering all the way down to 1-bit. They usually stop at 4-bits. The modulators will modulate down to 4-bits and not to 1-bit, so the noise shaping doesn’t have to be nearly so aggressive. With 4-bits, there is also adequate space for the required dither.
MW: Why would they continue to make the claim that the bandwidth, just like in analog systems, goes up to 100 kHz?
JS: Well, it does before the analog low pass filter. Unfortunately the low pass filter is an absolute necessity.
MW: Because of all the noise that’s been shifted up there, right?
A DSD spectragraph without the low pass filter to remove the HF noise. Notice the butterfly plot on the right. Click to enlarge.
JS: They like to conveniently ignore the fact that a 50 kHz low pass filter is required in any practical DSD system and that it is, in fact, a requirement of the specification.
MW: Is it really?
JS: Yeah. The high-frequency noise is a disaster if it reaches power amplifiers and speakers. 128x DSD offers some improvements which allow expanding the usable bandwidth above the 50 kHz limit of 64x DSD.
MW: But then the files are going to get huge.
JS: Yes, but file size is less of an issue these days. In my opinion, DSD and PCM are both good distribution formats. They’re both perfectly adequate for distributing the final product to the consumer. PCM is a little bit easier for the consumer to work with and PCM simplifies the playback hardware. It’s a lot easier to do PCM volume control. It’s a lot easier to do soft fades, crossfades, or any other processing that is required for playback. All of these processing functions are far easier to do with a PCM source than with a DSD source. But, if you put the processing issues aside, DSD is adequate for conveying the entire signal to noise ratio and bandwidth captured in any of today’s best recordings.
MW: Is that inclusive of any of the things that I do? I’ve got Wallace Roney and the spectragraphs that I look at that and other things that we’ve done exceed 40-45 kHz. And my justification is that I don’t really care whether the speakers and the rest of the hardware can actually reproduce the increased fidelity. But if there was a musical sound in the room when they were performing, I want to be able to capture it and preserve it through the entire production chain. And because Blu-ray and DVD-Audio can deliver frequencies higher than the traditional human limits, I say let’s try to reproduce everything. Given the situation with SA-CD and DSD and this whole noise shaping thing, it doesn’t sound like that is an option for them.
JS: Right. These frequencies are above the playback capability of DSD. Remember, you’ve got a 50 kHz low pass filter that means you haven’t got a chance for accurately reproducing anything over about 47 kHz in DSD. The filter introduces phase distortion, amplitude errors, and ringing as we approach the 50 kHz cut-off frequency. In contrast, 96 kHz PCM will capture your ultrasonics just fine.
MW: Yeah. That’s what I use when I’m recording.
JS: And we’re not very good at capturing anything that’s much above 48 kHz at this point.
MW: Right.
…look at Figure 2, that is a very, very misleading figure cause it’s really…what you have is an FFT of the DSD and you have a straight line that’s drawn based on 6.02 dB per bit at approximately 144 dB representing the 24-bit PCM. That’s not what the 24-bit PCM will look like on an FFT.
JS: If you look at the paper here and look at Figure 2, that is a very, very misleading figure cause it’s really…what you have is an FFT of the DSD and you have a straight line that’s drawn based on 6.02 dB per bit at approximately 144 dB representing the 24-bit PCM. That’s not what the 24-bit PCM will look like on an FFT.
Figure 2 – A DSD 64 FFT plot vs. a PCM (96/192/384 kHz) line chart. Click to Enlarge.
MW: What would be different?
JS: Oh, it will be way quieter, way lower level than that DSD that you see there. You gotta be comparing FFTs to FFTs, not a line that’s drawn on there based on a calculation that’s not valid in this case. I have been rather outspoken about this from time to time. I have a product that supports DSD playback and I support DSD playback because…the theory being that if you have DSD material that you want to playback I want to give you a way to do it. It shouldn’t be absolutely necessary to convert it first. If you want to play it back directly we’ll give you a way to do that.
We do not recommend it at all for any kind of studio production work. It’s just completely unsuitable for professional applications…for any production work.
MW: Sure.
JS: We do not recommend it at all for any kind of studio production work. It’s just completely unsuitable for professional applications…for any production work. The only way it should exist, if it exists at all, should be as the final output from a mastering room, where for whatever reason we want to distribute this in a DSD format. Okay, let’s create a master in a DSD format that we can distribute.
MW: So what goes on earlier in the production process could be analog tape or anything else including PCM.
JS: Right. And PCM is a wonderful format to do all your production work. Do all your mixing, all your EQ, all your processing that you’re going to do, everything that you’re going to do in the mastering process and then the very final output can be DSD. There will be some loss in quality when you do the PCM to DSD conversion, but this loss is just because of the limitations of the DSD and not due to any limitations of the conversion process. You’ll have a better result doing that than trying to do all the processing in DSD.
MW: Exactly.
JS: We make what a lot of people have called the best D to A converter for DSD playback. They’re thrilled with the way our converter sounds for DSD playback. We worked hard to make sure that we did it all natively but DSD is not a format that I think is a great idea. It’s not.
MW: But it’s out there.
JS: It’s out there and we want to support it because it’s out there, but not because we want to encourage the proliferation of it.
MW: How do you handle volume control in that final output stage? Do you convert to analog and then turn it up and down.
JS: We actually don’t. We do process that at the high sample rate and we have multiple 1-bit converters that are available to us. So the increase in word length that we get as a function of that volume control makes use of the redundant 1-bit converters that we have running in parallel.
MW: I see.
JS: So we’re not converting it…in a way you could look at that as if it’s PCM because there’s multiple 1-bit converters summed together in the analog domain. But that’s what you have to do to get volume control to work. The good thing is we don’t take it from 1-bit to multi-bit and back to 1-bit before we convert it to analog.
MW: Yep, as you were saying before.
JS: Instead of sending identical DSD signals to sixteen balanced 1-bit converters that are wired in parallel, we start sending different DSD signals to reduce the signal amplitude. All summing occurs in the analog domain. It is very cool!
MW: There are a lot of varying positions on the validity of DSD and I appreciate your frank assessment and experienced input.
JS: And Stanley Lipshitz and John Vandercoy did a lot of work on this. They wrote a lot of papers and a lot of it fell on deaf ears.
MW: I’ve read those and actually met Stanley at an AES meeting in the UK some years back.
JS: I actually had a SONY engineer say to me one time and this is quite few years ago…he said, “we realized after we got a ways down the road that DSD was kind of a mistake but we had too much invested in it”.
I actually had a SONY engineer say to me one time and this is quite few years ago…he said, ‘we realized after we got a ways down the road that DSD was kind of a mistake but we had too much invested in it’.
MW: Wasn’t archiving their whole reason for coming up with it in the first place? It was going to be used to take their analog masters in their vault and putting in a format that they thought would preserve the most fidelity, right?
JS: Yeah. And conceptually it looked like a simple approach. And, DSD significantly outperformed the 16-bit PCM systems that were common at the time. As a distribution format, DSD is definitely a big step above 44/16 CDs, and we want to give people the best possible playback of the wonderful DSD recordings that already exist.
MW: And they tried to put in the successor to the CD and that’s where we got a format war.
JS: Yep. Moving forward, we should focus on 24/96, and 24/192 downloads as these formats offer the best quality available.
Moving forward, we should focus on 24/96, and 24/192 downloads as these formats offer the best quality available.
I would like to thank John Siau for sharing his expertise on this topic. Using DSD 64/128 for production work is clearly not a viable option for high-end music and it is doubtful that moving forward with DSD for downloading will have any benefit for music lovers. In fact, it may just confuse things all the more.
Where does Benchmark stand on DSD vs. PCM, and why?
1. Benchmark recognizes that DSD (64x and higher) has significant advantages over 44.1/16 PCM.
2. Benchmark recognizes that high-resolution PCM (96/24 and higher) has significant advantages over 44.1/16 PCM.
3. Benchmark’s measurements and calculations show that the performance of 64x DSD is almost identical to the performance of 20-bit 96 kHz PCM (the in-band SNR of 64x DSD is about 120 dB). 64x DSD does not have any time-domain, frequency domain, or linearity advantage over 96 kHz PCM. DSD marketing materials have been very misleading.
4. 24-bit 96 kHz PCM has a 24 dB noise advantage over 64x DSD (144 dB vs. 120 dB), but this 24 dB noise difference is completely masked by the noise produced by other components in our recording and playback systems, and by the noise limitations of our recording and listening spaces.
5. Benchmark recognizes that 64x DSD and 96/24 PCM formats outperform most of the recording and playback chain. Bandwidth of either digital transmission system meets or exceeds the bandwidth of our microphones, amplifiers, and speakers. Likewise, the SNR of either digital transmission system meets or exceeds the noise performance of microphones, microphone preamplifiers, and power amplifiers. In addition, these digital transmission systems both exceed the performance of most A/D and D/A converters. 64x DSD and 96/24 PCM are not the factors limiting the performance of our audio systems. Focusing on DSD vs. PCM will distract us from much bigger issues in the recording and playback chain. Any sonic advantage of one digital system over the other will be very small when compared to improvements that can be made in other parts of the signal chain.
6. 64x DSD and 96/24 PCM both offer excellent sonic performance as distribution formats. PCM is more compact, but DSD provides better copy protection (a frustration to those of us who use music servers, but an important consideration for copyright holders).
7. Every A/D and D/A converter that Benchmark has produced uses Sigma-Delta conversion with equally-weighted 1-bit conversion elements. Benchmark never used multi-level conversion because of the THD issues caused by the linearity errors that are common to all multi-level systems. Benchmark has always placed high priorities on THD and linearity, at the expense of SNR. Sigma-delta 1-bit DACs tend to produce more noise than multi-level systems (such as ladder DACs), but the 1-bit systems achieve near-perfect linearity, which in our opinion is much more important than SNR. Benchmark has overcome the noise limitations of 1-bit conversion through the use of parallel 1-bit conversion systems. Our DAC2 sums the outputs of four balanced converters. Each of these four converters has sixteen equally-weighted balanced 1-bit converters (for a total of 64) that are summed together to improve the SNR of the system. These 64 1-bit converters can be driven from a 32-bit PCM signal, or from a 1-bit DSD signal. Either way, the performance is nearly identical, and none of the multi-bit THD issues exist. In this sigma-delta configuration there is almost no difference between the in-band performance of PCM vs. DSD. The only measurable difference at the output of the DAC2 is that 64x DSD signals produce about 8 dB more noise in-band than 96/24 PCM (due to the SNR limitations of DSD). Ultrasonic noise is not an issue at the output of the DAC2 because we are careful to remove the ultrasonic noise produced by DSD noise shaping. These same filters also remove the ultrasonic images that are always produced by D/A conversion (DSD or PCM).
8. The ultrasonic noise produced by DSD noise shaping must be removed after D/A conversion. It cannot be removed from the DSD signal before D/A conversion. This noise is due to the 6-dB SNR of the 1-bit DSD transmission system. Aggressive noise-shaping must be used in the DSD A/D, and at least once more in the mastering process. This noise-shaping is used to achieve an excellent SNR in the audible band by moving most of the 1-bit quantization noise to ultrasonic frequencies. Each time this process is applied, the quality of the DSD audio degrades (noise and distortion both increase). For this reason, the quality of DSD degrades very quickly in the mixing and mastering process. DSD has produced impressive results when the mixing and mastering processes have been omitted from the signal chain. To date, most of the DSD vs. PCM listening tests have omitted these processing steps. Unfortunately very few recordings can be produced without some mixing, editing, and mastering. Cascaded DSD noise-shaping processes should be avoided. For this reason, Benchmark does not recommend recording and mixing in DSD.
9. The 24-dB noise advantage that 24/96 PCM has over 64x DSD begins to become significant in the mixing and mastering processes. In terms of in-band noise, each DSD noise-shaping process is equivalent to at least 16 cascaded 24-bit dither processes. In terms of distortion, there is no comparison; the DSD noise-shaping process adds distortion while the PCM dithering process is distortion-free.
10. If the ultrasonic noise of DSD is not removed after D/A conversion, it will usually cause distortion in the playback system. The slew-rate limitations of most power amplifiers will fold the ultrasonic noise into the audible band causing distortion that is not harmonically related to the music. If the power amplifier has sufficient slew rates to pass the ultrasonic frequencies, similar problems will occur in the speakers. For these reasons, the ultrasonic noise must be removed from a DSD source after D/A conversion or before amplification.
11. Benchmark introduced 64X DSD on the new Benchmark DAC2 converter family. This gives our customers the ability to play DSD recordings in native format. Existing DSD recordings should not need to be converted to PCM to be enjoyed on a Benchmark converter.
12. Currently there is no practical way to play SACD disks through a high-quality outboard converter. SACD copy protection holds most existing DSD recordings captive to the limited quality of the low-cost conversion systems built into SACD players. It is our hope that many of the fine recordings that exist on SACD disks will be released for purchase as DSD downloads.
You conclude:
“Using DSD 64/128 for production work is clearly not a viable option for high-end music and it is doubtful that moving forward with DSD for downloading will have any benefit for music lovers. In fact, it may just confuse things all the more.”
I certainly don’t have the knowledge or expertise to agree or disagree. But, I believe your use of the word “clearly” ventures into hyperbole. Wouldn’t you like to know what Andreas Koch and other engineers who support DSD have to say in rebuttal before going so far? As I am sure you know, Mr. Koch and others believe strongly that DSD offers the best format for the future of high end digital playback. Mr. Koch, of course, is a principal of Playback Designs, a manufacturer of high end DSD capable DACs. But he was also one of the prime developers of DSD/SACD while at Sony and was a sought after design consultant to manufacturers of high end DSD capable DACs before forming his own company.
Contrary to your conclusion, virtually every reviewer of recently introduced quality DSD capable DACs prefers the sound of DSD playback to PCM. Common among these reviewers is the view that DSD is superior to PCM in addressing the traditional shortcomings of digital music reproduction when compared to vinyl. Do you just simply dismiss their opinions?
I don’t doubt that 24/96 or 24/192 PCM can sound excellent, but is it not possible that DSD64 or DSD128 simply sounds better? And isn’t that the crux of the matter – the sound quality, not the technology!
Robert, thanks for your comment and I do appreciate the issues brought up in your comment. However, there are two completely separate things in play. The first is the statement that I made referring to the viability of the DSD (at any multiple) for doing production work. By production work, I mean going to a session and actually capturing a performance (or intercuts for editing, which is very common), bringing the raw materials back to the studio for editing, equalizing, processing (reverb for example) and mixing and finally to mastering for final tweaks. The only phase of this sequence that can be accomplished in native DSD is the first one! That’s why people that choose to record in DSD from Channel Classics, Design w Sound, EMI and others move their native DSD files into the PCM format to handle the rest of the process before transcoding back to DSD at the final output stage.
Here’s a tech note that I downloaded this morning that describes one of Kent Poon’s productions (I have edited out parts that refer to mics etc.):
This Jazz Trio tune was recorded on a Genex 9048 recorder in multi-channel DSD, which is a research project by producer Kent Poon. DSD is a 1bit format, which cannot be mixed, level changed or processed. We use Switzerland Weiss SARACON software to convert multi-track DSD to PCM 24bit/192kHz for post production. The final DSD 5.6MHz DIFF file is created by Weiss SARACON software too.
The second issue that you bring up is about whether the final output using DSD is aesthetically pleasing to Andreas or any others…including reviewers. That’s a personal taste issue and one that I cannot really speak to. Do I think they are making this up? No.
However, when a prominent reviewer states flat out that a DSD 64 or DSD 128 file can completely capture the “magic of an analog tape at 15 ips”, I have to wonder. If that’s his baseline for the ultimate audio fidelity that we can technically achieve, then he needs to search out more recordings to listen to.
The point is there are major shortcomings associated with DSD just as there are engineering challenges with PCM. After carefully evaluating DSD files, talking to my engineering friends (John Siau, Robert Stuart, Jon Reichbach and others) and knowing something about the audiophile publishing world, I’ve concluded that DSD doesn’t measure up.
However, you are absolutely right in stating that the sound experience is what ultimately counts. Unfortunately for me, I need to have great sound AND great technical specifications. My own recordings have certainly been able to reach listeners and reviewers in ways that I couldn’t ever have imagined so I keep making more.
Once, you’ve heard them on a great system…you’ll know what I mean. Have I heard recordings in DSD that sounded great? Yes…but never any that sounded better than what I already have.
I am aware that DSD must be moved to the PCM format for processing purposes. Obviously, those producers who choose to use the format must believe that the end product is worth the trouble. Of course, as you know, the most important stage is the quality of the capture of the original performance, irrespective of format. This involves other matters such as microphone placement, etc., etc. David Chesky and others emphasized this point at a seminar on hi-res digital at RMAF 2012 in Denver. Hi-res format files are no guarantee of better sound.
Your experience has no doubt led you to believe in the superiority of PCM. It is interesting that others, such as Cookie Marenco of Blue Coast Records, have chosen to go the DSD route. I’m not sure what to make of your comment about 15 ips analog tape. Surely, you don’t take issue with the generally accepted view among the audiophile community that, until recent developments in hi-res digital, analog playback provided superior sound quality to digital. And that is the reason why reviewers use analog playback or “vinyl” as the standard of comparison.
I’m sure well recorded music sounds excellent in both DSD and PCM hi-res formats. For most listeners, technical specifications per se mean little. It’s what they hear that counts. I strongly believe that anyone looking to spend well over $1000 for a new DAC would be foolish to purchase one that did not provide DSD playback capability in addition to PCM.
@Robert Allan, quote, “Surely, you don’t take issue with the generally accepted view among the audiophile community that, until recent developments in hi-res digital, analog playback provided superior sound quality to digital. ”
When I see a sentence start with ‘Surely,’ I take it as a sign to read what follows with my critical faculties fully operational. This sentence is no exception. The only people who generally accept that view are the least informed audiophiles on the planet. There are a lot of them, unfortunately.
Every time anyone has attempted to test these claims, such as the one you stated above, Robert, in a way that prevents the audiophiles from being ‘led by the eye’, they have failed the test. Dismally. There is only one sane conclusion: the general views of audiophiles almost always arise from the widespread use of a hopeless, incompetent test method. And they don’t understand this and attribute the results of such testing as being entirely due to the sound waves in the air at the time.
The truth, about the relative sonic merits of technologies in the music recording chain, is almost like a secret, esoteric knowledge that dare only be whispered between sworn members in cloistered halls at night, for fear of being overheard by the general populace and dragged into the town square and burned at the stake. I think it is an absolute tragedy that genuine progress in audio quality is so difficult, unpleasant and uncomfortable.
The Real-HD website and articles are a source of audio truth, and I treasure it.
Grant…thanks for the post and compliment on this site presenting the “truth”. In reality, as the readers of my posts can readily appreciate I have very strong opinions on audiophile issues and especially the DSD issue. I do honestly try to present factual information as best I can discover it. I do my research, I respect the fact that there are others that know more about engineering issues etc…and when I get something twisted or wrong, I will acknowledge it.
I present my view of the truth and the views of others that are knowledgeable. I respect John Siau and I believe him when he spells out the misrepresentations put out by the DSD camp.
I feel most comfortable when I know that the methodology used in a particular recordings or employed by a specific format is integrated, simply and elegant. For me HD PCM is exactly that. That’s why I spend so much time looking at spectragraphs…not because they are the prime determinant of how a track will sound. I choose to look at them because they illuminate the veracity of the methods behind a track. I can see whether the engineer(s) took care to maintain the ultrasonics or chose or had to roll them off. It matters to me.
And I believe it makes a difference in the end result. We’re all looking for a great music experience. We will all find it in our own unique and personal ways. Vinyl LPs or analog tape may deliver it for others and CDs or MP3s might be enough for the average music fan. My plea is that music lovers keep an open mind to the world of HD-Audio especially in aggressive surround mixes…this is my audio nirvana.
David Chesky and others including Jim Merod of Blu Port Jazz are absolutely right that the skilled and experience of the engineering/production team that are actually capturing the original tracks are responsible for 90% of the ultimate sound of the release. The formats matter very little. The recordings that I produce sound different than Chesky’s and Cookie’s and work for some and not for others.
As a long time engineer and the owner of multiple analog tape machines over the years (including my cherished Nagra IV-S with QGB adaptor), I know the world of analog recording. I’ve made literally hundreds of albums and recordings using analog tape. But I do take issue with those that try to elevate analog tape to “holy grail” status. It is merely a format that has problems of its own…and for those who love it…a sound of its own (that’s why Cookie records here projects on analog tape).
Using vinyl or analog tape as the standard for comparison means accepting a particular color of sound as a pure reference. I prefer to have my reference be as close to the microphone feed as possible…in terms of color, dynamics, frequency response and clarity. It’s all about the marketing opportunities when it comes to DSD, SA-CD and now HiFi Blu-ray.
I have no argument with adding DSD to an expensive DAC…everyone should get the most capable and best sounding piece of gear that meets their needs. But the fact remains that most of the DSD releases have been through the PCM world along the production pathway. I would prefer to stay in the PCM world.
“Using vinyl or analog tape as the standard for comparison means accepting a particular color of sound as a pure reference.”
I agree with that statement and was exactly what I thought about DSD after reading the article. It explains why when I compare some DSD recordings to 24/192 file, the DSD sounded fuller (maybe some parts louder) and the 24/192 file sounded cold and clinical. It is now clear to me that the DSD file conversion, even though they are technically monitored, may still have added certain coloration that sounded similar to analog. It is like in photography, some prefer the clean and low noise photos of a high-end DSLR camera, while others like the look of the grainy old photo look of an analog camera. No one can say which is better, because they both are good in their own aspects and ones personal preference.
Grant, you wrote:
“When I see a sentence start with ‘Surely,’ I take it as a sign to read what follows with my critical faculties fully operational. This sentence is no exception. The only people who generally accept that view are the least informed audiophiles on the planet. There are a lot of them, unfortunately.
Every time anyone has attempted to test these claims, such as the one you stated above, Robert, in a way that prevents the audiophiles from being ‘led by the eye’, they have failed the test. Dismally. There is only one sane conclusion: the general views of audiophiles almost always arise from the widespread use of a hopeless, incompetent test method. And they don’t understand this and attribute the results of such testing as being entirely due to the sound waves in the air at the time.”
I see lots of rhetoric in the above but nothing in the way of evidence or substance to support these views. If I understand you correctly, virtually all reviewers of the lead audiophile publications have been hopelessly misled to be counted among “the least informed audiophiles on the planet”. As are most of the equipment manufacturers as well.
What incredible fortune you must have to be endowed with the wisdom and insight to recognize the widespread hopeless incompetence of an entire section of the industry. It is a shame that you chose not to share the brilliant methodology that you employ to arrive at your conclusions.
Hello Robert, I shan’t reply extensively to your sarcasm-drenched questions directed to me. It seems my first post on this thread has already come true, where I am dragged out into the town square for a bit of burning-stake admonishment for daring to expose the flawed writings of the audio press. Such is life. At least I am not alone.
The absolute best analogue master tape I give a sonic score of 4/10, where 5 is the maximum score possible within the crippling limitations of 2-channel playback. That is an excellent score and I congratulate it for getting so good. The absolute best vinyl LP I give 3/10 sonically, for despite being clearly audibly distorted it still manages to sound thoroughly enjoyable. However, vinyl LP is subject to so much production variability that a score of 2/10 would be more reasonable.
I remember J Gordon Holt, founder and key visionary of Stereophile, started leading the publication into multi-channel audio in the early 90’s. Those were exciting times! But it was soon realised that the customer base was not ready for real progress, so for commercial reasons the publication reversed course and started sailing backwards into the warm and lucrative embrace of audiophiles by looking at expensive tube amplifiers and vinyl players, and opening columns dedicated to the worship of the golden days and technologies to suit. Holt retired and has ever since been bitter and disillusioned abut the whole audiophile arena. Such is the fate of visionaries. He’s lucky he wasn’t burned at the stake! 😉
I would certainly prefer that all comments remain polite and respectful. This is a new site and I recognize that the topics under discussion are hotly debated among audiophiles and other interested readers. We all have our own opinions and all are free to explain and debate here without apprehension.
There are lots of different formats and production techniques in the world of music production. I gravitate towards those that reproduce music with the most accuracy rather than those the just feel good. That’s my left brain and its need to make sense out of an otherwise right brain creative experience.
Thanks for expressing your thoughts. All views are welcome…but let’s make sure that we maintain a level or decorum and style that encourages reasoned discussion and avoid sarcasm and attacks. Thanks to all.
Grant,
My sarcasm was obviously directed at the testy tone of your post. Of course you are entitled to your strong opinions, but let there be no confusion about the fact that they are just that, i.e. your opinions. For example, you write about “daring to expose the flawed writings of the audio press”. You may disagree with the writings of the audio press and you may believe those writings to be flawed. But, it is simply illogical to postulate a priori that that those writings are flawed and therefore require exposure, especially where the views of the audio press are generally ad idem.
Perhaps you should have mentioned that J. Gordon Holt’s greatest contribution was to review audio equipment based on how it sounds and not upon its technical specifications or its price. From the writings of colleagues, his bitterness was apparently part of his personality.
Oh, come on. Anyone who has listened to a DSD recording knows that DSD’s audible SNR is far beyond 6 decibels. In fact, even nasty old Redbook CD audio has a SNR that is excessive for even very quiet listening environments. I have a few 16/44 classical CDs that demand my constant knob-twiddling attention to remain listenable, and I live out in the country in a very quiet house. So I really don’t see how a supposedly-deficient SNR is a valid argument against DSD.
And the protective filtering required by DSD at 50,000 Hz may cause distortion way up in the ultrasonics, but this isn’t consciously audible to people.
But if PCM is easier to work with, them I’m happy with PCM recordings. I have heard straight hi-rez PCM recordings via Blu-Ray and downloads, PCM to DSD recordings on SACD, and supposedly pure-DSD recordings on SACD, and they can all sound amazing, or not. And I cannot, just by listening, hear any difference between PCM and DSD playback, and I’ve done a lot of critical blind A/B testing. I know some folks may gain ontological solace when viewing a frequency curve that’s smooth out to infinity, but I only care about what I hear.
The 6dB SNR would apply if there were no filtering, it is the wideband delta sigma noise up to the nyquist frequency (1.4mHz for DSD x64). While it might seem at first glance that DSD has wider bandwidth than any PCM, because of the astronomical sampling rate, that turns out not to be true, because of the huge need for filtering the high frequency noise, which doubles in each higher octave for delta sigma modulation. I applaud you for doing critical blind A/B testing and honestly reporting what you have not heard. More people should do that before claiming huge benefits to technologies like DSD, and therefore saddling us with cumbersome and restrictive formats that do not allow user digital signal processing, something I do in all my systems, and the digital signal processing has immediately obvious audible benefits (I do dsp in 24/96), unlike these differences between digital formats, which I feel are overhyped.
As for myself, I have not done so much blind testing, but it is my belief, from JAES published research, and casual listening, that supersonic frequencies ARE at least somewhat important. My best systems use super tweeters with response to 40kHz. So I do care about extended frequency response and low noise to at least that point. Furthermore high frequency loss and noise in an audio system is cumulative, so it has been reasonably argued that any given component should have response well above 20kHz, and low noise out there as well. I think it is wonderful to have audio files in 24/96 and 24/192 for this reason, among others.
This is an area where subjective assessment has to be guided by a capacity for matahematical analysis. I have had a long interest in delta modulation, initially from a communications perspective. Some of the basic engineering tenets gained then apply equally well in the epanded bandwidth of audio.
Firstly I have always thought that delta modulation (which is essentially a slope-following or differential coding scheme) is well suited to the ‘triangular’ spectrum of speech and music. The early efforts of dBx to produced their delta-modulation based recording system confirnmed this to me in some of their early demonstrations. Unfortunately, unless you are prepared to sample at 68.719476736GHz to meet Stuart Boothroyd’s proposed criteria of 26kHz flat bandwidth and equivalent 20-bit resolution with a uniform step size (true primitive delta modulation), you must use adaptive delta modulation with variable step size, introduced by two or more levels of integration in the loop (in primitive delta modulation, this is the single order low-pass filter 6db/octave starting at 0Hz – or thereabouts!). That way be dragons!
Secondly, there seems to be a lack of under standing that quantization noise is usually characterised as occupying the full (Nyquist) spectrum. The bit that you can hear is the fraction 20kHz/Nyquist bandwidth. The remainder is the bit that we try to get rid of with the reconstruction filter at the end of the chain, or in the process of higher order integration. When Sony proposed the DSD standard, they did so in the basis of a particular higher-order integration process. Subsequently, Stanley Lipschitz showed this process was not unconditionally linear, and not a fantastic idea for professional mastering (AES Convention, 2000, Los Angels, IIRC). Multi bit delta modulation (and therefore not pure delta mod, more differential PCM) has emerged to deal with this. Elvis has left the building …
This discussion has already covered some of the issues in post production. So I suspect that we need to call in Lincoln Majorca to re-fashion his production technique of ‘direct to disc’ to ‘direct to DSD’ to demonstrate the true capabilities of this capture and distribution format. Thelma Houston, where are you when we need you?
John…this detailed explanation and information is much appreciated.
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When do we do PCM at DSD sample rates? An excerpt from the article:
JS: Yes. We’re going to do multichannel recording, we’re going to have to do mixing, we’re going to have to do EQ, we’re going to have to do various effects and all that has to be done in PCM. Now the PCM can all be done at DSD sample rates, and that’s fine. That’s well and good. The conversion from DSD to PCM is really an automatic thing, a very benign conversion. If you don’t change the sample rate, there is no loss of quality when expanding the word length. As soon as you do any mathematical operation on the DSD, the word length expands. And so DSD goes from being 1-bit to being multiple bits. The DSD DAW manufacturer chooses how many bits of precision they wish to preserve in the processing – the more the better.
-reub
That’s what DXD is all about.
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Any truly knowledgeable person knows that moving to DSD 128 or DSD 256 or PCM 384 is a complete waste of data. The Benchmark DAC2 HGC produces the finest audio playback I’ve ever heard. It’s the pinnacle of both analog and digital signal processing. Why would the design team at Benchmark add a capability for DXD (aka PCM) or DSD 128 when those format don’t result in “better” audio? The DAC2 is a far better piece than the inexpensive DACs that support every flavor under the sun.
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We all know DSD is inferior, that it needs filters and stuff. The advantage of DSD as a format is that is so niche and hipster that there’s much more care in the mastering stage of DSD recordings. Usually they’re lower volume and less compressed. There’s a lot of recordings from the pre-90s that have amazing sources yet the remasters post 90s usually decrease in Dynamic Range for the CD format. When you compare to the SACD counterparts you get at least the same dynamic range if not more. Then there’s the 192/24 scams. Most 192/24 remasters are either upsampled to waste more space and bandwidth or compressed remasters that are less dynamic than the 16 bit counterparts. 192 Khz is a waste and no one should be promoting it for final releases, 48/24 should be the goal standard. So even if SACD is inferior in every way and a pain to work with, 24 bit PCM is 90% dead already because the industry decided to scam on it and SACD can survive happily despite its downfalls just because it’s weird and hipster and people dedicate more time and care to dying unpractical obscure formats for some reason.
There is much more care in the mastering stage of DSD recordings? I don’t know where you get this idea. The audiophile labels that work in high-resolution PCM are equally concerned with maintaining fidelity and crafting the best sound possible. They benefit from not having to worry about the crazy amount of noise inherent in DSD. CDs produced from older analog master can be dreadfully over processed and mastered but if transferred to high-resolution (96 kHz/24-bit PCM) without any equalizing or processing will reproduce exactly the fidelity of the analog original. At 48 kHz, there will be loss of potential high frequency information. This is why the major labels transfers are done using PCM.
DSD has no future in the commercial recording industry. Every new major project is recorded at 24-bits in studios all around the world. Whether it is necessary to deliver 24-bits to consumers is a point to consider since the dynamic range is squashed below that threshold.