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Posted: Mon Feb 09, 2004 4:51 am
by Kjoerup
So will the SH sheep now loudly proclaim that SACD is capable of reproducing a greater quantity of echo with more “lifelike” and “three-dimensional” qualities than poor, lowly redbook does? I suppose it does indeed, particularly if the self-proclaimed Tonmeister’s hand is pushing firmly on the echo knob or slider on the DSD chain. You know, the one he claims is identical yet somehow, uh, different from the redbook chain.

At least now we know that when Hoffman says “air” he really means “added echo”. And don’t forget that he recommends a $9000 SACD player as the only DAC out there that can adequately showcase said air/echo. Thus far he has yet to publicly comment on the benefits that the infamous hatrack may bring to SACD recordings, but time will tell. Or maybe I missed the official missive.

Hoffman has been ludicrously inconsistent on the subject of DSD/high resolution since day one. Well, actually, he began by saying that such resolutions were not necessary and could in fact be a hindrance to good (re)mastering. Nowadays, of course, he waves the DSD fanboy flag ever so high, all the while proving himself to be woefully incapable of answering the most basic of all questions. That is, given his decade-long chest-beating about his 16-bit/44.1 kHz-targeted masterings being “identical to the master tape”, why is that now, all of a sudden, DSD and SACD are deemed necessary to, uh, “sound identical to the master tape”? Audiophiles may claim differently, but basic physics tells me that tape does not age like fine wine and Father Time does not bestow magic new frequencies or airy spaciousness onto aging masters. But need I remind anyone here that science and logic are frowned upon at and that any such rationally based questions or inquiries are met with very much the same hostile zeal a medieval village mob brought to a witch burning?

Oh, yes, there is that ethereal “air” all cooped up in the SACD pits, of which Hoffman bitchily reminds us no sub-$9000 DAC will be capable of farting out. Bollocks!

I would like SH or his minions to tell me why a simple SACD analog-out recording to a Minidisc deck (COMPRESSED PCM 16-bit/44.1 kHz!) produces results that are virtually indistinguishable from the source in ABX testing. Putting aside the fact that the Hoffman gang doesn’t believe in ABX testing (wonder why!!!), doesn’t this tell you that SACD is a crock? If an MD recording can capture all frequencies and that elusive air that’s blowing in the wind, so too, a logical mind would surmise, could an uncompressed redbook CD!

SH is selling snake oil, pure and simple. And something tells me that if his check was signed by the Other Side, he’d be touting DVD-A rather than DSD. If he (or Sony or Philips, for that matter) was able to produce one iota of evidence, one logical argument or defense of the necessity and “superiority” of DSD and SACD, then perhaps a real dialogue would ensue. But I’m not holding my breath for that to happen.

But I do think that Hoffman's snidely bitchy responses – the only ones he seems capable of making with regard to this subject – speak volumes.

Posted: Mon Feb 09, 2004 6:34 am
by thomh
Question for the gang:

Who is able to hear the difference in the echo? When I play the wav file through my LynxTwo soundcard and listen through a pair of headphones, the last 4 snare hits has a deeper and more 3D echo. And when I a/b it against the SACD layer on the disc they are identical.

Gardo, I see you registered as a member a while back, and you were very vocal on the SH forum about the "big difference" between the two layers. Can you comment?

Posted: Mon Feb 09, 2004 8:58 am
by Rspaight
My SoundBlaster Audigy card doesn't give me a whole lot of difference apart from the higher level on the CD part. I haven't yet tried burning to CD-R...


Posted: Mon Feb 09, 2004 10:52 pm
by Kjoerup
I don't own the CCR SACD but I did burn the sample to CD-R this afternoon. What I did was put the CD player in repeat mode and let the sample play continuously. Yeah, the volume difference between the two bits is readily apparent, and I agree with Krabapple that these are really two different masterings and are thus hard to compare.

But what I DO hear is indeed added echo in the snares on the SACD sample. Or, conversely, LESS echo in the redbook sample. I played around a bit in Peak, using the real time VST plugin, adding a bit of extra echo to the redbook sample and was able to get a close approximation of the SACD sound. So, it seems a foregone conclusion that the vaunted "air" and "natural decay" that is supposedly the strength of DSD is, at least in this particular case, nothing more than echo added to the DSD signal during the mastering process. Or maybe the master really does sound like that and the redbook mastering is lacking. Take your pick. Regardless, it's clear that a 16-bit/44.1 KHz CD-R can retain and reproduce this echo.

Thomh, if I understand you correctly, it's interesting that you seem to be saying that a downsampled SACD excerpt, when A/B'd against the actual SACD, sounds identical. So much for the need for higher resolutions then, no?

Posted: Sat Feb 14, 2004 6:05 pm
by thomh
Kjoerup wrote:Thomh, if I understand you correctly, it's interesting that you seem to be saying that a downsampled SACD excerpt, when A/B'd against the actual SACD, sounds identical. So much for the need for higher resolutions then, no?

Sorry for my late response. I wanted to wait until we had a chance to test some SACD -> Redbook conversions which we now have done.

Tests were done using my Sony DVP-NS905 (SACD) and Pioneer 656A (DVD-A) players through B&W Nautilus 805 speakers as well as a pair of Grado SR-80 headphones . Amplification was through various NAD amps. The analog out of the mentioned players were connected directly into the line inputs of my LynxTwo soundcard and some of the songs we recorded in 16/44.1kHz were:

Fortunate Son - Willie & The Poor Boys SACD
Down On The Corner - Willie & The Poor Boys SACD
Fashion - David Bowie Scary Monsters EMI SACD
Money - Pink Floyd DSOTM SACD
You're So Vain - Carly Simon No Secrets DVD-A
Hotel California - Eagles DVD-A
Slow Train Coming - title track - Bob Dylan SACD
Girl From The North Country (love that song) - Dob Dylan Nashville Skyline SACD
+ some classical stuff (which I won't bore you with)

Upon playback of each song, we synched and level matched the outputs of the players and the LynxTwo card. There were 3 of us rotating between listening and cueing up. Even though the test subject used the pre-amp remote for switching between sources, he did know what source was playing as the inputs on the preamp was switched around at random. Each listening session lasted about 1/2 hour. While this test was not scientifically rigorous in its nature, I think it gave us a pretty good idea of what to expect if indeed such a test was set up.

The conlusion that was unanimously reached was that there was no way we could reliably tell which source was playing and all we could do was to make random guesses. We tried to catch any subtle changes but none were obvious to us. This test showed that even the lowly Redbook format is pretty damn transparent and is able to capture any "magic" qualities that has been bestowed upon the hi-res formats (as it also does when recording vinyl).


Never mind what the audiophools will have you believe, Redbook is extremely high resolution to start with and if the source is of lower resolution (which is always the case with these analog master tapes) then there is *nothing* to gain from SACD or DVD-A. It is that simple. Anybody who tells you otherwise is either ignorant of the sampling theorem, *or* has some sort of an agenda, *or* is trying to sell you something, *or* is just too stupid to comprehend it.

Recording engineers very rarely use extra bits to extend the resolution but rather to give extra headroom during recording. Higher sampling rates increases bandwidth *only* and there is no more accuracy to be gained from it. This is not my opinion, it's how digital audio works. For mono/stereo playback, SACD/DVD-A is a crock, pure and simple. While I do believe that 20/88kHz would be ideal (bearing in mind the specs of todays recording and playback equipment), for mono/stereo playback 16/44.1kHz seems adequate.

Nevermind the 192 kHz sampling rate BS (as Dan Lavry so convincingly reminds us), what the hell're ya going to do with 24 replay bits? That is a dynamic range of 144 dB. Nowhere on Earth (except maybe in the minds of the audiophools) is it possible to achieve 24 bit resolution. At *best* maybe 20 bits. OK, audio signals can theoretically exist below the 20 bit level (-120dbfs) but the signals would be totally obscured by the electronic noise. You also have to take into account that even the best listening room has a 20-30 dB noise floor so to even utilize the 90 dB that lowly Redbook gives you, your peak SPL would need to be well over 110 dB. To get that 144 dB range, your peak SPL would need to be well above 160 dB. The sound pressure alone'll kill ya in seconds.

The "more points = more accuracy" bunk that SH and others are trying to sell to you shows complete ignorance or worse, simply because inorder for it to be so then they would first have to prove Nyquist, Shannon, science and math wrong. Do you think SH and his posse is up for that task? I doubt it, so instead they turn to the audiophools last resort by claiming, "Well, I can hear a big difference" simply because they know so well we cannot argue with that. SH seems to be too ignorant or agenda driven to argue the technical side of it so instead he resorts to comments like "Either his system isn't resolving properly, he can't hear properly..." yada, yada, yada. Why not just chalk it down to a difference in converters and be done with it. Pathetic.

I have no problem with SACD or DVD-A for their multichannel capabilities (of which I am not yet set up for), but there is absolutely no solid and reliable evidence to back up the claim that these formats improve resolution in the lower octaves. NONE. SH's lame evidence proof positive is ridiculous and I would like to see him try to introduce this outside his own sanctuary. If he is so sure that he has proved the sampling theorem wrong then let him or any of his cohorts write up a paper pointing to what part of the theorem is flawed, set up listening tests and present it at the next AES meeting. $ony has had ample time to set up such controlled listening tests. They have NOT. You can betcha that they have done so in-house and if these tests showed these "day and night" differences that audiophools are masturbating over, you can be damn sure that they would have presented their results with all the fanfare that their money could buy. Again, they have NOT. Instead they set out to convince gullible or agenda-driven audiophools. Until somebody does provide us with proof positive, all we hear again and again are the same ol' audiophool buzzwords, and since it comes from those that claim to hear big differences between cables and interconnects and shout the technological supremecy of analog and vinyl from the rooftops, I will continue to remain faithful to math, science and good ol' commonsense and trust my ears (without the aid of my eyes).

RANT OFF and goodnight from Oslo.

Posted: Sun Feb 15, 2004 10:05 am
by Kjoerup
Well said, Thom! :!:

Posted: Wed Apr 07, 2004 12:33 pm
by krabapple
Nevermind the 192 kHz sampling rate BS (as Dan Lavry so convincingly reminds us), what the hell're ya going to do with 24 replay bits? That is a dynamic range of 144 dB.

Just a followup to this -- Mr. Lavry has recently completed and posted a white paper on sampling theory and the fallacy of 'more samples are better'

[url=]Sampling Theory for Digital Audio

Posted: Sun Apr 11, 2004 5:59 pm
by Grant
Kjoerup wrote:
But what I DO hear is indeed added echo in the snares on the SACD sample. Or, conversely, LESS echo in the redbook sample. I played around a bit in Peak, using the real time VST plugin, adding a bit of extra echo to the redbook sample and was able to get a close approximation of the SACD sound. So, it seems a foregone conclusion that the vaunted "air" and "natural decay" that is supposedly the strength of DSD is, at least in this particular case, nothing more than echo added to the DSD signal during the mastering process. Or maybe the master really does sound like that and the redbook mastering is lacking. Take your pick. Regardless, it's clear that a 16-bit/44.1 KHz CD-R can retain and reproduce this echo.

According to the guy who mastered the CCR SACD/CD hybrid, the natural decay on the snare heard on the SACD is on the master tape. NO echo was added.

Posted: Sun Apr 11, 2004 6:38 pm
by lukpac
So then there was apparently some problem with the 16/44 transfer...

Posted: Tue Apr 27, 2004 3:02 pm
by krabapple
I revisited the 'SACD is fundamentally flawed' thread on SHTv today for the first time in ages..and can report that the madness continues (e.g., LeeS is still being alowed to release torrents of bullshit at whim; the cluesless are still thanking the clueless for all the 'interesting information' being presented).

Thomh, where are you these days?

Posted: Wed Apr 28, 2004 8:42 am
by Rob P
I've been reading it, too. It's ludicrous. I've now learned that SACD has 64 times more resolution than CD, thanks to LeeS. I've also learned that Yesman, who is so insecure that he makes postings on the board about his great new projects that he is working on, also thinks SACD is better. People not following the party line there (SACD is better than CD) are being shoved to the side behind the scenes.

I think the only reason the thread is still open is because it would make them look bad to lock it down. It's too late for me; they already have lost most, if not all, of their credibility in my book.

Posted: Mon Jun 07, 2004 12:25 am
by krabapple
Those looking for a corrective to SACD enthusiast bull (and aren't we all?) are hereby directed to this HT forum thread for a stellar example:

Another Lee Scoggins Smackdown

Posted: Mon Jun 07, 2004 4:28 am
by thomh
krabapple wrote:Those looking for a corrective to SACD enthusiast bull (and aren't we all?) are hereby directed to this HT forum thread for a stellar example:

Another Lee Scoggins Smackdown

That LeeS is quite a character. At least on that forum they do let him get away with it. He reminds of Reagan's comment that "facts are stupid things".

As for the 'SACD is fundamentally flawed' thread on the SH forum, I noticed that Seagoat started ruffling some feathers and was quickly silenced by SH. I wonder what he wrote that was so bad that SH had to delete it?

For a more technical discussion on Dan Lavry's excellent Sampling Theory paper, go to George Massenburg's forum

What I find amusing is that nobody can touch Dan on the science and math, so instead we are faced with the usual subjectivist "but, what if...." or "I can hear it" mumbo-jumbo.

As for myself, between being knee-deep in work and trying to spend some quality time with my family, I have further investigated and tested for any sign of that SACD magic.

Over the past months I have had a total of 24 audiophiles in and out of my home with their various SACD players inorder to record them to my DAW at 4416 for a/b listening tests. The major players tested were:

Sony DVP-NS900V
Denon 2900
Pioneer DV-868 (59 in the U.S.)
Philips DV-963SA
Linn Unidisk 1.1

Vinyl has also been tested with the following TTs:

VPI Scout w/ Shure V15VxMR
Thorens TD-850 w/Ortofon 540 MK. II
both through a Gram Amp 2 Special Edition phono preamp

Most of the discs that we have used for these a/b tests have been good old classics from the analog era remastered again for SACD, DVD-A or vinyl.

At the most there were 4 participants listening and discussing at one time. I let them play with the remote so they could switch between the sources at will. We also concentrated on different sections of the music by playing one 15 second sequence and immeditely switch to the other source for comparison.

The conclusion drawn from these each of these tests have been consistent:

When time-synched and level-matched, *no one* has yet been able to tell the original from the copy.

Also, there is also an AES paper entitled:

"DVD-Audio versus SACD - Perceptual Discrimination of Digital Audio
Coding Formats" by Dominik Blech and Min-Chi Yang, AES Convention Paper
6086, May 2004.

The abstract reads as follows

To study perceptual discrimination between two digital audio coding
formats, "Direct Stream Digital" and high resolution (24-bit, 176.4
kHz) PCM, subjective listening comparison tests were conducted with
specially recorded sound stimuli in stereo and surround. To guarantee
their reliability, validity and objectivity, the double-blind ABX
tests followed three main principles: The signal chain should be based
on identical audio components as far as possible; these components
should be able to convey very high audio frequencies; and the test
population should consist of various groups of subjects with different
listening expectations and perspectives. The results showed that
hardly any of the subjects could make a reproducible distinction
between the two encoding systems. Hence it may be concluded that no
significant differences are audible.

A couple of extracts

Figure 9a shows quite clearly how the score distribution for each
music example hovers around the 50% (chance) level for the stereo
examples. The surround examples (Figure 9b) show this even more
clearly. This observation is reinforced if the total of all correct
answers is compared with the total of all incorrect answers: Of a
total 2,900 choices (145 test sequences 20 choices per test sequence)
there were 1,454 correct choices and 1,446 incorrect ones (see Figure

Though less readily formulated with mathematical equations, the high
level of frustration felt by many subjects during their tests left
quite a strong impression. These people, for the most part, were well
accustomed to critical listening on a professional level, but they
found that they could not even begin to recognize any sonic differences.

The conclusion drawn by the authors were that you cannot hear a difference between DSD and PCM (same music, same converters, same DAW). So somebody tell Hoffman that it is not only *my* ears that have a hard time telling the difference between the formats.

Posted: Mon Jun 07, 2004 4:31 am
by thomh
Since that thread on the George Massenburg forum is rather long and it contains a lot of noise, read this post from Mr. Lavry carefully as it pretty much says it all.

I first stated my objections to 192KHz sampling for audio publicaly, while chairing the seminar "All about AD converters" at the NY AES last year.

My white paper "Sampling Theory" is at
under "suport". I posted it a few weeks ago, and have since been involved in conversations on the proaudio NG. I am pleased with the response. A number of highly respected engineers are of the opinion that it is about manufacturers making money, not about scientific and engineering issues.

The few that argue on behalf of 192KHz talk about "things we do not know" and about listening tests. Such listening tests, taking a device designed to do 192 and switching in a final X2 decimation to 96KHz is NOT VALID, because it tests the last stage decimator and nothing else. The real test should be based on the comparing a 192KHz device against a 96KHz device. By the time you do 192KHz, way too far from what is optimal for audio, the damage is already done.

I am not saying that all distortions are bad. There are those that like tube distortions and more. I am saying that whatever you hear is NOT outside the audio hearing range, thus can be contained within 20KHz or so, certainly within 44-48KHz (88-96KHz sampling). Moreover, in audio, the lowest bandwidth device is the bottleneck (week link in the chain). Lets see:
1. Mics? Who is using 96KHz mics? Most Mics drop at about 20KHz...
2. Speakers? Who is using 96KHz mics? Most Mics drop at about 20KHz...
3. How much energy do musical instrument put out above 40KHz?
4. Ear - my dog does not hear 48KHz

So all of a sudden we are presented with that gizmo- 192KHz?

It is true that there is a lot we can not explain. I have been catering to the ear for many years, and do not know all the answers. But that does not mean that everything is fuzzy. There are some things we know, and know well. We know that sampling must exceed twice the bandwidth of interest. Not by much, just a tiny bit.

Only in audio there is such a disconnect between content bandwidth and sampling. No one else (medical, instrumentation, telecom, video) goes nuts with faster sampling because it yields no positives, only negatives:
1. Less accuracy (there is always a tradeoff between speed and accuracy)
2. More data (storage, data transfer...)
3. More processing required (often traded with lower quality processing)

Again, for those that like a certain distortion associated with 192KH sampling, it is all contained under 20KHz or so. One should not take a whole industry into having to lower transparency, twice the storage requirement and so on. Why not instead, manufacture that distortion (if you like it) with a 96KH device? (96KHz is already an overkill, 60KHz would have been an optimal rate, taking care of ALL the issues including filters, pre ringing and there is nothing else to worry about!).

My paper may be of interest to some, while other "ear types" may find it "too much". I kept the math to a minimum, so it is about graphs, plots and text. I put some energy to explaining what Nyquist finding is all about (he was a major contributor to modern technology).
Common sense may be misleading folks to think that "the more the better". What is true for say pixels and video or computer screen, is not true for sampling limited bandwidth signal. Making an analogy here is wrong! You need 2 points to draw a straight line. No need for more. You need 3 point for a circle. Well, the bandwidth restriction ends up with: you need only to exceed twice the highest frequency you deal with, thus 88.2KHz accommodates 44.1KH of audio.

I have heard numerous folks that understand math, engineering and science talk about the 192KHz being all about selling new gear to make money. In fact, there are some well respected engineers making such statements on the PRO Audio NG right now. I talked to a lot of folks that admitted privately that they are afraid, uncomfortable, consider it unwise... to raise objections to their employer. I see a strong correlation between the who promotes 192 and who sells such gear. I get ZERO scientific arguments suggesting that my comments are not on solid grounds. I get some folks that want to promote 192 to IGNORE all the science and engineering and math. All I hear is "maybe it is something we do not understand", and “golden folks hear this” (a 192KHz design, against a 192KHz design plus a less than perfect decimator to 96KHz, which is how you listen to a final stage decimator). Folks, if that is what you like, lets do a real good X2 up-sampler than use the imperfect X2 down sampler (decimator), and there is your 192KHz sound. This is the ONLY VARIABLE in most of the listening test. Many folks reported that they like the sound of that X2 last stage decimator. They were confused into thinking it is a test comparing 192KHz to 96KHz.

I am sad to see audio going in the wrong direction. I am not going to be a part of it, though it does impact my economics. The upside: I can look at myself in the morning. I did not sell out.

I few preemptive comments:
1. Nyquist is immuned to real life imperfections. All the noise, be it analog or quantization or any linearity causing distortions follows Nyquist. It is solid and does not need verification for every (or any) piece of gear.
2. Again, I do not tell folks what they hear and what they do not. The argument is not about what is good or bad. It is about signals distortions and what not. The standard for “people with ears” is to go for a listening test. My points are valid independently of listening test, because my argument does not require such tests, other than reference to what is the maximum possible hearing bandwidth. Therefore I built my case staying away from subjectivity such as good vs bad sound.
It is possible to figure some things with math. I stop where math and science and real world engineering do, and it still lets me tell a story full of facts.


1. People can not hear above 40KHz (yes/no)

2. Nyquist points out that we can retrieve ALL the information, without distortions of signals under 40KHz by sampling at 88.2 or 96KHz (yes/no)

3. Time domain and frequency domain are not a separate issue. A lower bandwidth yields wider impulse response and visa versa. For example you can not have a 1usec impulse (first zero crossings) out of a 20KHz system. Time domain and frequency domain are tied to each other. (yes/no)

4. Most microphone do not pick up signals much higher than 20KHz. 40KHz of audio provide plenty of bandwidth margin (yes/no)

5. Most speakers do not respond to signals much higher than 20KHz. 40KHz of audio provide plenty bandwidth margin (yes/no)

6. In audio, the lowest bandwidth device dictates the overall bandwidth thus with real musical instruments and mics and speakers under 40KHz we can capture ALL the signal information with 88.2KHz sampling and do not need to use 192KHz sampling. (yes/no)

7. Musical instrument generate very little energy above 40KHz. (yes/no)

8. Even dogs do not hear 96KHz, so 192KHz sampling is not needed for dogs either (yes/no)

9. Moors law does not apply to the ear. Audio bandwidth does not double every few years. (yes/no)

10. There is a tradeoff between speed and accuracy. Very high speed converters yield less bits, lower speed converters yield more bits. Therefore there is an optimum rate for AD which depends on the application. Too slow cause loss of bandwidth, too fast causes loss of accuracy, too much data and increased processing power requirement. (yes/no)

11. Given a fixed set of conditions, such as modulator order, over sampling ration and modulator bits, a design of a noise shaper with a goal of accommodating 48KHz audio provides more accuracy for signals under 40KHz, than a modulator aiming at 96KHz audio. In fact choosing the faster rate it is equivalent to an X2 reduction in over sampling ratio. The over sampling ratio is a major factor in defining the design quality. (yes/no)

12. Comparing a device designed for 192KHz to 96KHz sampling by switching on and off one more stage of X2 decimator yields erroneous results. The design of the modulator is already optimized for 192KHz, thus all the tradeoffs between speed and accuracy are already done. A real comparison would require using the same technology with different noise shaping, one optimized for 96KH and the other for 48KHz of audio. (yes/no)

13. “Localized” sampling rate of the AD front end (over sampling) and DA (up sampling) help ease analog filter design. Therefore the high pole filters issues are a thing of the past. Phase linearity issues are now easy to avoid. The up sampling and over sampling should not to be confused with the basic sampling rate of digital audio. (yes/no)

14. When people hear things while utilizing a 192KHz system, it does not mean they hear energy above 20KHz (or 40KHz to have a safe margin). In fact what they hear is energy under 40KHz. (yes/no)

15. If they hear “only” the energy under 40KHz, than 88.2KHz is fast enough rate to include all the audio, at sample times and between samples. It may not be intuitive, but connecting the “sample dots” will be done correctly and the original signal is retrieved. (yes/no)

16. Once we agree with question 1 (40KHz is enough bandwidth for audio), the answers to questions 2-15 should not be based on hearing. It may require some learning, but it does not involve the ear. (yes/no)

17. Let us define “perfect reproduction” as total transparency. If one could captures the audio waveform with a mic (after the musical instrument generated it), and were able to generate the same waveform out of a speaker (before the ear), than we have transparency. Accepting of such a definition does require knowledge of the ear or knowledge regarding musical instrument. (yes/no)

18. Some gear (that alters the waveform) such as eq, compressors and more, requires evaluation by ear. The goal of transparency will be met by keeping the wave shape intact. We are dealing with wave shapes before they arrive at the ear, thus such definition provide freedom from dependency on the ear. (yes/no)

19. The ear is not a scope it is not a meter. The ear tells you what it hears, but not why it hears it. (yes/no)

20. Too many of the audio EE type of engineers become an extension of the sales and marketing, denying what they know to be fundamentally true. It may be wiser (economically) to go with the flow, but it is less honest to do so. (yes/no)

Hint: the answer to all the questions is yes.

Dan Lavry

So how did everybody score on the test?

Posted: Mon Jun 07, 2004 7:14 am
by lukpac
"There is a big reason: native DSD to many of us engineers sounds much better."