ANOTHER Example of Why I HATE DSD and Why Customers Who Bought Sony's Boloney Are So Annoying

[Miska]

Using resistor attenuation there will result in a minor degradation, and possibly why the lower output impedance of a good preamp makes a worthwhile improvement for many. That's also why you don't normally see the lower -3dB point of an amplifier set by the capacitance value of the input capacitor these days

You are more likely to see something like a 22uF Blackgate ( if any left!) or a quality electro of a similar or higher value..

 

It's still better than added distortion and noise of another entire device on the path with more cabling, connectors, etc.

 

Series input capacitors? No way, I have servos. Serves two purposes, negates any input offset plus doesn't require any internal offset adjustments either.

 

Why would pre-amp have any better stuff inside than a DAC? Or other way, why would DAC designer decide to have a bad design or worse than for a pre amp, especially when designing DAC to be connected straight to a power amp?

 

 

The series resistor is fitted to ensure stability of the opamp into the capacitive load of a cable.

 

50 ohm series resistor with special capacitance and output impedance compensation is better choice. I'm not at all sure that all pre-amps are properly designed in this respect either.

 

The LME49710 and LME49720 are particularly touchy as regards working into even a short cable load.

 

I have not used those yet anywhere. But selecting right opamp for right function is important part of the design. I feel sick when I read some people randomly swapping opamps into a DAC without understanding requirements for a particular function. In my designs it would even likely cause burned opamp (the two extra pins in a single op-amp have varying uses) and for some opamps it won't even fit, because not all use the standard pin layout (single op-amp in a 16-pin SOIC case).

[mav52]

It's still better than added distortion and noise of another entire device on the path with more cabling, connectors, etc.

 

Series input capacitors? No way, I have servos. Serves two purposes, negates any input offset plus doesn't require any internal offset adjustments either.

 

Why would pre-amp have any better stuff inside than a DAC? Or other way, why would DAC designer decide to have a bad design or worse than for a pre amp, especially when designing DAC to be connected straight to a power amp?

 

 

 

 

50 ohm series resistor with special capacitance and output impedance compensation is better choice. I'm not at all sure that all pre-amps are properly designed in this respect either.

 

 

 

I have not used those yet anywhere. But selecting right opamp for right function is important part of the design. I feel sick when I read some people randomly swapping opamps into a DAC without understanding requirements for a particular function. In my designs it would even likely cause burned opamp (the two extra pins in a single op-amp have varying uses) and for some opamps it won't even fit, because not all use the standard pin layout (single op-amp in a 16-pin SOIC case).

 

 

good point

[Alex66]

As I recall from the pre I made it used a 22 ohm on the output, it could drive the most ott cable lengths. There is a lot of negativity towards the humble OpAmp, but carefully chosen they can be exceptional devices. That is the important thing though the ideal amp for the I/V is not the same as one to drive half a kilometre of cable or for a phono input stage. Now a lot of them can be changed in many circuits with out catching fire or damage but unless you know why you are changing one you can cause a deterioration of quality of the output. The pre was in the days of vinyl so it was a required unit now Ill go with a good volume control either on the input of a stereo power amp or on the output of the DAC (buffered). Software volume controls lack a tactile quality that I like.

[Superdad]

Because it allows better performance for less money. For example the best multi-bit true PCM converter, PCM1704 costs around 60$ and has accuracy of around 18-19 PCM bits. Best delta-sigma converters have accuracy of 21-22 PCM bits and these converters traditionally do that kind of conversion DSP inside and cost around $10.

 

Most of the "PCM" DACs and ADCs are delta-sigma converters so they have to employ bunch of DSP inside to do the conversion. DSD on the other hand is 1-bit delta-sigma data, intended to bypass this back-and-forth conversion DSP at both ends, in ADC and DAC.

 

So essentially what I'm doing, is moving the format conversion DSP from the DAC to the computer. This allows making the DAC much simpler because it doesn't need to do any DSP. And also if you compare processing capabilities of a quad-core Core i7 CPU on a modern computer to capabilities of that $10 DAC chip...

 

You know I agree with your approach (I'm an NOS PCM1704 guy feeding it 384kHz), and while yes, modern S-D DAC chips are cheap, why do we have to settle for cheap? I wish there were more firms (like MSB and TotalDAC) producing discrete R2R ladder DACs with laser-trimmed resistors--and offering them as modules. MSB offered such to Hovland years ago, but even at OEM prices it was too expensive for what was going to be an elaborate product. Oh darn, I guess I just subverted my own wish...

 

 

True DSD-capability in a modern DAC kind of turns it into a "NOS" mode and allows bypass of the internal DSP (or part of it depending on chip). You still don't loose any of the traditional capabilities though. So you can even make comparisons!

 

Feeding the DAC at high one-bit rates you take s/w control of everything (a good thing) and turn the DAC chip into nothing more than a simple math processor. But to me that is like running the race and stopping just before the finish line: Don't give your lovely DSD512 (s/w modulated) signal to an ordinary DAC chip to have it poorly filtered and handed off to some compromised output stage. Just amplify it a bit with video amps and use a low-pass analog filter--to have a DAC-less DAC. See SONORE Rendu PureDSD DAC (it just does not take DSD512 and is DLNA-only at the moment).

[tailspn]

Just amplify it a bit with video amps and use a low-pass analog filter--to have a DAC-less DAC.

 

Ted Smith did a very similar thing with a DAC he designed several years ago. Gus Skinas of the Super Audio Center and Sonoma fame heard it, and thought it was the best DAC he'd ever experienced.

 

My new DAC (long, but with pictures) - Ted Smith - Computer Audio Asylum

[Miska]

You know I agree with your approach (I'm an NOS PCM1704 guy feeding it 384kHz), and while yes, modern S-D DAC chips are cheap, why do we have to settle for cheap? I wish there were more firms (like MSB and TotalDAC) producing discrete R2R ladder DACs with laser-trimmed resistors--and offering them as modules.

 

I'm fine with that too, but from technical perspective it goes this route...

1) You fine tune R2R ladders to max

2) You want to oversample at high rate to get best possible analog reconstruction with a simple analog filter

3) Settling time vs precision becomes problem and you hit the realization edges with gigaohm at one end and milliohm at the other

4) You realize that with noise shaping you don't need as many bits at such high oversampling rate anyway

5) You start dropping off bits and increasing frequency as you go, because you can, and performance keeps increasing

6) You end up with very high frequency and just one bit (practically PWM)

7) You realize most of the performance is tailorable as digital domain algorithms

8) DAC becomes almost pure software with very simple hardware

9) You can finally put all hardware effort into making very good analog design

 

But to me that is like running the race and stopping just before the finish line: Don't give your lovely DSD512 (s/w modulated) signal to an ordinary DAC chip to have it poorly filtered and handed off to some compromised output stage. Just amplify it a bit with video amps and use a low-pass analog filter--to have a DAC-less DAC. See SONORE Rendu PureDSD DAC (it just does not take DSD512 and is DLNA-only at the moment).

 

That's what I'm working on now. I've got my own CS4398 Direct DSD & analog filter for couple of years now, very good for DSD128. But now I'm going for discrete implementation and DSD512. But just passing bitstream to an analog filter doesn't give best possible performance, it needs more.

 

Since I'm constantly busy on software, it takes time to get something ready on hardware side. I don't mind if something suitable appears on the market, would save a lot of time & effort.

 

I just need to do some hardware stuff every now and then to stay somewhat sane...

[dallasjustice]

Miska,

I am sure that whatever you are working on will be great. I just don't like it when manufacturers disparage other approaches so as to promote their own design.

 

I can say from my experience that it is very much possible to get unbelievably great sound at very low speeds. That method works well in Vincent's design. Certainly, it's not your approach. As to your synopsis of all low speed DACs; II'll take that with a huge grain of salt. :-]

[Miska]

I am sure that whatever you are working on will be great. I just don't like it when manufacturers disparage other approaches so as to promote their own design.

 

I'm not a hardware manufacturer. I have no intent to sell any of the hardware I design or build. I design and build audio hardware for my own proof-of-concept purposes and because I've been DIY audio guy for 25+ years. (since there's always a chicken-egg problem for new technologies, something needs to be done to test software when hardware is not yet available on the market)

 

I've designed commercial hardware for non-hifi use (measurement, etc), which is completely out of context for this forum and I cannot even talk about that stuff anyway.

 

So count me as a DIY guy when I'm talking about hardware.

 

I can say from my experience that it is very much possible to get unbelievably great sound at very low speeds. That method works well in Vincent's design. Certainly, it's not your approach. As to your synopsis of all low speed DACs; II'll take that with a huge grain of salt. :-]

 

What is low speed DAC here? For example MSB DAC is 1.5/3 MHz which is not low speed, it is very high speed for a multi-bit DAC. (for comparison, PCM1704 chip can do 768 kHz)

 

I don't get to subjective evaluations because there's no end on that discussion. But I have not seen a DAC that runs actual conversion stage at frequencies below 200 kHz and still offer competitive measured performance compared to delta-sigma designs.

 

HQPlayer supports PCM upsampling up to 1.536 MHz 32-bit with number of dithers and noise shapers. It also supports delta-sigma modulated output up to 24.576 MHz 1-bit. And all the processing is available for both, there is also specific support for older 18-, 20- and 22-bit DACs to fully utilize those without truncation. So I'm fully in both PCM and DSD camps. Base line is that I seek for ways to achieve best possible performance at most reasonable price. Currently I feel that I can get closest with DSD, but I reserve right to change my mind at any point in time...

[Miska]

By the way, that 9-step list is pretty much how current DAC chip evolution has gone. At least following companies seem to have concluded pretty much same way since they now produce delta-sigma DAC chips, even though many had multi-bit ladder designs in past:

- Burr-Brown/TI

- Analog Devices

- Philips/NXP

- Crystal/Cirrus Logic

- AKM

- Wolfson

- ESS

- Seiko NPC

- Niigata Seimitsu

 

Would all these be wrong, of course a possibility, but there is probably some commonly shared view behind their choices anyway?

 

P.S. And to be exact, BB/TI uses hybrid design in their best performing chips, 6-bit ladder combined with delta-sigma handling rest 18-bits (18 + 6 = 24) with actual analog stage operating with 66 levels.

[dallasjustice]

Jussi,

What are the universally accepted DAC measurements that 100% positively correlate with listening pleasure? If those measurements cannot be shown to be relevant to listening enjoyment, then why measure? A DAC manufacturer that measures AND listens may be one who has accepted that he can't know all of the best measurements that strongly correlate with listening fun.

 

DACs are not like speakers. Toole and Olive have done great work to show that speaker off axis linearity strongly correlates to listening preferences. There's no such study in the DAC world. There's some evidence that really high jitter can be objectionable and certain distortion spectra can be annoying, but there's little evidence that even DAC linearity strongly correlates to listening enjoyment. Why do folks seem to prefer high frequency rolloff and boosted bass? The B&K house curve has many cousins and they all have one thing in common: high frequency rolloff and boasted bass. I am not saying that everyone should start using target based DSP but when they do, the preferences are almost always non-flat targets. Why is that? I don't know.

 

Jussi, even discrete SDM DACs are cheap to build. You only need 1 resistor and the precision of that resistor is irrelevant to the sound. That's the reason you see so many SDM DACs these days. Sure they sound great but they are not categorically superior to other great multi bit designs in terms of listening enjoyment.

[barrows]

I am one of the ones that fights on this forum against the new "trend" of infatuation over DSD.

I am glad that Charles Hansen took the chance to tell the truth about this DSD crap, instead of playing it into marketing schemes like many other manufactures.

 

PS: I still want to see some technical specs regarding THD+N for their products.

 

Plenty of third party tests including full suites of measurements of Ayre QB-9 and DX-5 at stereophile.com. Every possible measurement you could desire is shown in those reviews. Of course the new version of the QB-9 will be a bit better, but that review may be a few months out. Independent measurements are certainly trustworthy, especially compared to some manufacturers claims.

[sandyk]

[series input capacitors? No way, I have servos. Serves two purposes, negates any input offset plus doesn't require any internal offset adjustments either./QUOTE]

 

Miska

Both my Class A preamp and 15W Ch. Class A P.A. are both fully DC coupled with servos. There are no capacitors in the signal path or the feedback networks of either. The front ends of both also use dual metal can , closely matched transistors,(LS313 and LS352) with front end balancing as posted in the DIY Audio Current Mirror thread of Nov.2008.

Regards

Alex

[Miska]

What are the universally accepted DAC measurements that 100% positively correlate with listening pleasure? If those measurements cannot be shown to be relevant to listening enjoyment, then why measure? A DAC manufacturer that measures AND listens may be one who has accepted that he can't know all of the best measurements that strongly correlate with listening fun.

 

I've been measuring all my DIY hardware and also all the purchased hardware I've ever had. I believe I have some kind of idea of correlation between what I hear and what I see in measurements. And also kind of estimate of how certain type schematic will sound. Same for DSP algorithms too, I've been analyzing, simulating, tuning and listening DSP algorithms. Of course I still have a lot to learn, if I ever feel I've learned everything I will most likely go for something completely different and stop tweaking audio.

 

I always both measure and listen, but I'm sort of engineering perfectionist in a way that I demand both perfect measurements and sound (listening). I've probably told many times about my ranking system when I buy audio hardware... First I use measurements to reduce the potential hardware to a small group (thank you JA and PM!), on which I then perform my listening tests to select the one that sounds best. Same method is used for designing DSP algorithms, first an analysis check that the technical performance meets expectations and then listening tests to evaluate sound. And this cycle is repeated over and over again.

 

 

but there's little evidence that even DAC linearity strongly correlates to listening enjoyment.

 

It is possible learn to hear all kinds of errors and once you know those, it is very painful to listen because it bothers you all the time...

 

I have my quest to reach most detailed and accurate sound that is pleasing to listen to. Others can then evaluate how well I've got there regarding DSP stuff. I also tell the commercially available hardware I use myself.

 

The B&K house curve has many cousins and they all have one thing in common: high frequency rolloff and boasted bass. I am not saying that everyone should start using target based DSP but when they do, the preferences are almost always non-flat targets. Why is that? I don't know.

 

I covered this topic in my post here, on DSP area to the thread regarding house curves. Optimal house curves depend on speaker radiation pattern and acoustic properties of the room, and also listening axis vs the speaker.

 

Jussi, even discrete SDM DACs are cheap to build. You only need 1 resistor and the precision of that resistor is irrelevant to the sound. That's the reason you see so many SDM DACs now days. Sure they sound great but they are not categorically superior to other great multi bit designs in terms of listening enjoyment.

 

Well, it is more than one resistor. From objective evaluation point of view, if you can get 20 dB better performance for 1/10th of the price I don't see too many negative sides on this. For evaluating listening ejoyment, I prefer using standardized MOS style scoring methods.

 

I'm not trying to say multi-bit ladder DACs are bad. And performance of those can be also improved by upsampling and noise shaping. But so far it is a road where price increases exponentially and performance logarithmically. I'm trying to reach the inverse, logarithmic price increase and exponential performance increase.

 

You can find some interesting listening test results for lossy codec implementations here:

http://en.wikipedia.org/wiki/Codec_listening_test#Results

 

For example:

http://listening-tests.freetzi.com/html/AAC_at_128kbps_public_listening_test_results.htm

[barrows]

[series input capacitors? No way, I have servos. Serves two purposes, negates any input offset plus doesn't require any internal offset adjustments either./QUOTE]

 

Miska

Both my Class A preamp and 15W Ch. Class A P.A. are both fully DC coupled with servos. There are no capacitors in the signal path or the feedback networks of either. The front ends of both also use dual metal can , closely matched transistors,(LS313 and LS352) with front end balancing as posted in the DIY Audio Current Mirror thread of Nov.2008.

Regards

Alex

 

Uuuhh, Alex? A servo is a feedback network. Sure, it supposed to be for DC only, but do you really think only DC goes through it. I know, nitpicking, but I think it is a little misleading to call anything with a DC servo a no GBF design.

[Miska]

Both my Class A preamp and 15W Ch. Class A P.A. are both fully DC coupled with servos. There are no capacitors in the signal path or the feedback networks of either. The front ends of both also use dual metal can , closely matched transistors,(LS313 and LS352) with front end balancing as posted in the DIY Audio Current Mirror thread of Nov.2008.

 

Sounds good. There used to be plenty of factory matched bipolar transistors and J-FETs, but nowadays it seems to be hard to find those. SSM2212/SSM2220 is still available and couple of newer MATxx models, especially MAT12 looks promising.

[barrows]

 

Uuuhh, Alex? A servo is a feedback network. Sure, it supposed to be for DC only, but do you really think only DC goes through it. I know, nitpicking, but I think it is a little misleading to call anything with a DC servo a no GBF design.

 

OOOPs! please disregard, I am doing too many things at once here and got confused, my bad!

[Charles Hansen]

SORRY EVERYBODY!

 

I started this thread when it seemed like had reach a point at work where I didn't have work 100+ hour per week to stay on top of things. Of course right after that some more things hit the fan and made quite a mess for me to clean up. So I haven't had time to follow this thread, not even a tiny bit. And for some reason, I'm not subscribed to it (undoubtedly my fault, but I will rectify that shortly).

 

My whole point was this particular customer had read (or been told) AND BELIEVED IN HIS HEART the following things:

 

1) Digital volume controls are much better and "purer" and the best way to adjust volume (at least when starting with a digital source signal).

 

2) DSD is a much "purer" form of digital modulation and closer to the source material and is the best way to record and play back music.

 

3) He had been using J.River Media Center quite some time.

 

4) When he finally got a D/A converter that would decode DoP, he thought that he would be in "musical ecstacy".

 

5) When he found out that NOBODY can adjust the volume of a one-bit signal without first converting to a mutli-bit signal and then re-converting to a one-bit signal again, he completely flipped out.

 

6 He had had built up in his head a world where every part of his system would be as close to perfection as humanly possible. When reality hit him smack between the eyes, he literally lost it and didn't know who to blame.

 

7) My original point was that the blame should be pointed directly at Sony, as they are they ones who built up at least half of the myths that this guy was operating under.

 

The moral of the story is like that old US TV commercial for some kind of fake "butter" where "Mother Nature" came down from the heavens and scolded the people saying, "It's NOT NICE to foll Mother Nature!"

 

 

I thought it was pretty funny. Didn't mean to cause trouble, but it seems that everyone took it in the right spirit for the most part. Thanks for listening!

[sandyk]

Sorry to digress Charles

 

Miska

I have used SSM2220 in earlier preamp and RIAA phono preamp designs. Most LS313 appear to have very high HFE (>700 !) as well as being closely matched. Although the voltage ratings are low , they are fine in a typical Current Mirror as described in various Douglas Self books.

The LS352 is a modern replacement for the Motorola 2N3811A,which was used in many designs in the 90s by Erno Borbely.

You can still obtain 2N3811A on eBay.

 

Regards

Alex

[Charles Hansen]

Most LS313 appear to have very high HFE (>700 !) as well as being closely matched. Although the voltage ratings are low , they are fine in a typical Current Mirror as described in various Douglas Self books.

The LS352 is a modern replacement for the Motorola 2N3811A

 

 

Hello Alex,

 

For all you hard-core DIY guys, I find these types of parts absolutely useless, just like the THAT Corp parts. The MAX Ic is ONLY 10 mA, which means that you can only run them up to 5 mA peak if you want any reliability. And they don't show the curves, but typically the beta droop will start at only 1 or 2 milliamps. So unless you are building electrometers or some other non-audio product, the have ZERO application in the real world.

 

Sorry,

[One and a half]

 

Their AAVA volume control is also pretty good.

 

Can substantiate to that.

[ted_b]

Charles,

I am still somewhat confused by your attitude towards DSD and a growing base of users that your company clearly wants as customers. I assume you and your Company are quite willing to sell your new DSD DAC ( which goes for a premium over the earlier PCM version) and collect the monies, yet you are telling the world that those who love DSD are what? idiots? Seems disingenuous or maybe even hypocritical in more than one way. I may have missed something, I realize.

[sandyk]

Charles

I am fully aware of their specifications. I also suspect that Douglas Self and Erno Borbely know a little more than you do in this area. My own gear is very much based around the works of Douglas Self and Silicon Chip magazine. Distortion figures would likely be below the typical .0006% from 20Hz to 20kHz as in the Silicon Chip P.A. design, due to careful matching of devices and close tolerances of resistors in the front end especially.

I use these devices on the most linear parts of their curves, and mainly because I am able to completely balance the collector currents of both halves of the LTP. I bet that you don't even believe that anything more than a 1% match will give a further improvement ? BTW, Nelson Pass was also a participant in the Nov. 2008 DIY Audio Current Mirror thread where my claims were verified. Can YOU get quite good surround sound from just 2 stereo speakers on DTV with well recorded material, or room filling surround from just the stereo soundtrack of the BluRay of Ävatar., or from DVD-As such as "The Eagles-Hotel California" etc. as well as with many well recorded CDs and high res. material ? My preamp for example has been judged to outperform $14K retail Marantz preamp and Power Amp combo, as has a friend's 100W Ch/8 ohms using similar devices and similar front end balancing in his preamp and P.A. We also regularly compare our gear against loaned top stuff from Len Wallis Audio in Sydney. at our regular listening sessions with Infinity speakers plus Raal 100kHz tweeters, and now with B&W 802s.

 

.Regards

Alex

 

Current Mirror Discussion - Page 15 - diyAudio

 

P.S.

I find your recent aggressive and arrogant dismissal of other peoples viewpoints, on almost every subject, quite interesting.

Even the very title of this thread makes one wonder.

Are you under the affluence of incahol ? (grin)

[rennq]

Charles,

I am still somewhat confused by your attitude towards DSD and a growing base of users that your company clearly wants as customers. I assume you and your Company are quite willing to sell your new DSD DAC ( which goes for a premium over the earlier PCM version) and collect the monies, yet you are telling the world that those who love DSD are what? idiots? Seems disingenuous or maybe even hypocritical in more than one way. I may have missed something, I realize.

 

You really don't know?

 

Same reason we have 3D TV. Industry giants create new buzz word, people think they need it, customers won't buy without it, manufacturers then have to include it, a little time passes it becomes irrelevant, move on to the next buzz… 4K!

 

And my guess is the price premium is for reasons other than DSD capability.

[Hiro]

And yes, HQPlayer supports all digital processing for both PCM and DSD, including volume control, convolution engine (for DRC) and speaker distance & level adjustments. There's also DSD rate conversion and I recommend upsampling DSD to higher rate if possible when using digital volume control to gain extra dynamic range for output. And of course more to come.

 

The Ayre is limited to 64x.

[Charles Hansen]

Charles,

I am still somewhat confused by your attitude towards DSD and a growing base of users that your company clearly wants as customers. I assume you and your Company are quite willing to sell your new DSD DAC ( which goes for a premium over the earlier PCM version) and collect the monies, yet you are telling the world that those who love DSD are what? idiots? Seems disingenuous or maybe even hypocritical in more than one way. I may have missed something, I realize.

 

No Ted, not at all. I will elucidate my position in one place so that people won't have to hunt through pages and pages of threads.

 

1) The only reason the DSD exists at all is because the patents for CD were expiring. This had been a HUGE bonanza for Sony and Phiips as the royalty per CD was onl $0.07, yet it added up to $1 BILLION per year. The costs of inventing the scheme had been written off decades earlier so this represented a massive profit center for Sony. They did NOT want to lose this profit.

 

2) At the same time, every other manufacturer was SICK of Sony/Philip making so much money while they received nothing. When the DVD standard was finalized, almost all of the major manufacturers owned some patents that were critical to the DVD concept, so they all wanted the format that replaced CD to be based on DVD, as they felt that the entire distribution of royalties was much more equitable.

 

3) As it turned out, the only viable physical format for a CD replacement was based on the DVD technology, using red laser of much shorter wavelengths than the IR lasers used in CD. So both SACD and DVD-Audio were based on the physical format of DVD. There wasn't enough time to develop anything else.

 

4) So now the two factions lined up against each other. Sony and Philips had the advantage of only having to align two companies, where as the DVD-Audio Working group was a GIANT committee with representatives from over two dozen companies having input.

 

5) Both groups felt that the key selling point for both formats would be multi-channel capability (surround sound) which would easily piggy back on to the booming home theater craze where everyone was purchasing mutli-channel systems.

 

6) There were a few voices (notably Pioneer, pushing for a higher fidelity standard of 96/24, which would give over double the bandwidth of CD.

 

7) Sony played one of their typical dirty tricks when they announced that DSD would play UP TO 100 kHz with a dynamic range of UP TO 120 dB. This is complete and utter bullshit, as the S/N ratio of DSD at 100 kHz is negative. If you look at a reasonable bandwidth of say 20 kHz to 100 kHz, there is MORE NOISE in the DSD system than there is total signal.

 

8) Once Sony raised that red herring, it was difficult to fight against. The DVD committee screwed up royally and decided to fight fake fire with real fire So they made Lossless compression a mandatory feature of DVD-Audio, and in the process lost ALL backwards compatibility with the hundreds of millions of DVD players that were already in use.

 

9) So DVD-Audio was now an orphan that required the customer to purchase a brand-new player for which no software existed. That was one of the turning points where DVD- Audio started to lose the war.

 

10) There were two other factors that greatly contributed to Sony gaining the PERCEPTION that SACD sounded better that DVD Audio:

 

a) They hired Andreas Koch and Ed Meitner to build ALL of the hardware required by the record companies required to make discs with a completely new modulation scheme. In contrast, DVD-Audio was like the old Wild West - no rules whatsoever and anything goes An SACD could not be made without a Sony trained and approved engineer on board the engineering team for making the recording. (99% were simply transfers of old analog classics.) Anybody could make DVD-Audio discs with easily available hardware and extremely low cost (eg, $39) software packages.

 

b) Completely by accident, SACD had a huge sonic advantage because it required no brickwall filter during recording and relatively gentle 3rd order analog filters during playback. At the time, very few people (if any) realized the degree of sonic degradation created by steep brickwall filters.

 

11) So when people started comparing the SACDs against either CD's or DVD-Audio discs. the combination of FAR BETTER hardware, along with FAR BETTER MASTERING, along with the (virtual) elimination of brickwall filters, most people said, "Aha! Sony is right! DSD is a better format than PCM!"

 

12) In the end, both formats flopped for different reasons. DVD-Audios generally required a video display just to navigate the menus. People didn't want to add a video display to their audio system. This was proven when we started shipping our C-5xe "universal" player. It was two channels only, so we lost the ENTIRE multi-channel crowd, but that turned out to be a drop in the bucket. The C-5xe was BY FAR the most successful product we had built up to that time. A consumer could buy it, not care about the "format wars" or who won or who lost -- they could just buy any software they wanted BECAUSE THEY LIKED THE MUSIC and didn't have to pay any attention to how it was recorded. If it was an audio disc, it would play and a video monitor was NOT required. We sold thousands of those things!

 

13) Eventually both formats died off for a very simple reason. Releasing a disc in EITHER format was a money losing proposition. If you release a disc and it makes money, you say, "Hey we just made $XXX,000! Let's do that again!" But that NEVER happened with either format. So after Sony' bribery money ran out, that was pretty much the end of SACD, and since the DVD-Audio committee was SO STUPID as to require a completely different player than a regular DVD player, it was also doomed to failure.

 

The sad part is that if they had just made DVD-Audio the same as DVD without a video section, we all could have been enjoying 2 channels of 96/24, or 5.1 multi channel all the way up to 48/24. And they would have played in ANY DVD player with (at the very most) a firmware update. But that's what happens when things are run by a committee.

 

14) The story would have ended there except for Gordon Rankin. He had been on parts of the USB committees over the years, and he made sure that USB had the ability to make a good audio data transfer link. So when he released his "Streamlength" asynchronous isochronous USB firmware, all of a sudden any computer made in the last ten years could serve as a transport of higher performance than ANY conventional S/PDIF transport EVER MADE! Ayre was the first licensee, and more people wanted solid state equipment than tubed equipment so we ended up setting a NEW sales record with that product. For 2-1/2 years, the least expensive product we had ever made (excluding accessories) made up over 40% of our revenue stream!

 

15) Now we have to split the story somewhat. First we will look at multi-channel PCM. Whilt it is TRIVIAL to record and make multi-channel PCM files for USB DACs, there is virtually no software available, very few software players that will handle it, and very few USB DACs are made with more than two channels. Essentially it is a solution waiting for a problem to solve. So we will leave that alone, except to say that if one thought it were a viable market, it would be trivially easy to make a 16-channel 192/24 USB playback system. But nobody really cares except a handfull of people that generally just purchase Blu-ray discs of live audio concerts.

 

16) Everything changed for DSD, when in 2006 Sony announced the "DSD Disc". This was essentially an SACD but without the copy protection that made it impossible to play on a computer. I first heard about it in 2008 (I think, I can't remember any more). I was seated next to Gus Skinas, who had been part of the Sony SACD team. His role was to be the liaison between the recording studios and the technical people at Sony who would loan out the hardware required to make an SACD. When he told me about it, I talked to Gordon Ranking about it and we said that it would be trivial to packetize the DSD stream so that it looked like PCM. But then we realized it was a fool's errand because the only source of software was to (illegally in this country) rip an SACD with one of the rare specific models of PlayStation 3s. So we said, "Screw it."

 

When Sony (who is facing HUGE financial troubles and has been for the last ten years) finally gave up on SACD, they gave the rights and all of the designs for the Sonoma Audio Workstation, which is basically the only way to make a modern SACD recording. The Sonoma turns the 64x 1-bit DSD signal into a 64x 8-bit signal, which has 256x the resolution of an SACD. So by turning the DSD signal into a very high speed PCM signal, they can now do things like adjust the volume, fade in or out, add EQ or reverb or compression or any number of things that have become mandatory in this day and age of multi-track recording.

 

The problem is that now Sony has to talk out of both sides of their mouth at once. They painted themselves into a LITTLE TEENSY CORNER because they said that one of the prime advantages of a one-bit system was that was always inherently linear. But when people found out the trught that probably less that 0.001% of all DSD recordings actually were transferred into PCM and then back to DSD, they look pretty damn stupid.

 

17) But there is the inescapable fact that DSD (in general) DOES NOT USE ANY BRICKWALL FILTERS and therefore it is MUCH easleri to get good sounding results from a DSD than from normal PCM. So there aer still a WHOLE bunch of people that (incorrectly) believe that DSD sounds "better" than PCM.

 

Then SOME of these people started record companies selling downloads of DSD recordings because now there is a way to play them on a computer and you DON'T have to worry about the laser burning out or the SPECIAL IRREPLACEABLE CHIP crapping out or any of that traditional problems with SACD.

 

18) But the truth is that there are still some HUGE problems with DSD, especially with regards to out of ban noise, the need to replace ALL of the studio's recording systems, DAW's (Digital Audio Workstation), and everything down the line. Then at the other end, the consumer has to find a disc player or computer DAC AND computer playback software that will handle this completely different modulations scheme.

 

Of course the question becomes WHY?

 

And the typical answer is that "DSD sounds better than PCM". Well, Ive got news for you. It can sound better. And it can sound worse. IT ALL DEPENDS ON THE IMPLEMENTATION.

 

19) We have recently introduced a two channel A/D converter that will output both PCM and DSD. But this converter has a few tricks up its sleeves. Specifically we have taken a page from the "What is so great about DSD" manual and applied it to PCM. And it turns out that with quad rate PCM you can get ALL OF THE SONIC ADVANTAGES OF DSD WITH NONE OF THE PRACTICAL DISADVANTAGES.

 

So we are literally on the verger of a whole new era of good sounding recordings. (YOU READ IT HERE FIRST!)

 

Go read John Atkinson's review of the QA-9 on the Stereophile website:

 

Ayre Acoustics QA-9 USB A/D converter | Stereophile.com

 

In it, he says that he ripped his best sounding LPs with the QA-9, and when he compared the copy with the original, he tried "until his ears bled" but could hear NO DIFFERENCE BETWEEN THE ANALOG ORIGINAL AND THE DIGITAL COPY.

 

Now this is something of a breakthrough. When the digital is SO GOOD that it is INDISTINGUISHABLE from the analog, we have transcended a barrier.

 

And this is ALL DONE with PURE PCM, which ANY STUDIO IN THE WORLD CAN HANDLE. No new hardware, no new software, no new nothing. Just better sounding music on PCM. And downloading quad rate PCM is trivial as HD Tracks has shown us all.

 

You can hear another original recording of an organ recording at:

 

John Marks Records - Jul Downloads

 

In another month or so we will have 4 downloads available of a purist recording using the QA-9 in quad-rate PCM mode with just two microphones in a nice hall in Berlin of a piano. These will be preludes by Debussy, played absolutely, stunningly beautifully by a soon-to be-famous pianist named Katie Mahan Gorgeous playing and gorgeous sound.

 

20) So the bottom line is NOT that you are an idiot if you like DSD. DSD can sound wonderful. But what I am saying is THERE IS NOTHING MAGIC ABOUT DSD. WE CAN GET ALL OF THE GOOD THINGS ABOUT DSD IN A HIGH SAMPLE RATE PCM RECORDING ALSO!

 

So its more like the difference between tubes and transistors It's hard to make a bad sounding tube amp or preamp. And is DAMNED hard to make a great sounding solid state amp or preamp.

 

Well, we have cracked the nut. There is NO LONGER ANY REASON to get all worked up over DSD. Every time we send out a A/D converter for trial, It never comes back. In another year or two, MANY releases will be made on the QA-9. And many releases will be made on ADCs where the engineers have stolen our ideas and applied them to their products. (Of course the copy is never as good as the original, but that is a different story altogether

 

I hope that clarifies things a bit

 

Best,