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Has anyone done any serious research on the supposed advantages of DSD over PCM? I'm curious to know what you think at this point.

I've done a little bit of reading, including some highly suspicious sounding pronouncements from an executive at Sony. So far, I've found nothing empirical that points to PCM being definitively inferior. This whole DSD thing seems like a wash to me.

Does anyone out there have some hard science that tells me I'm dead wrong? I would love to see it.

Thanks in advance!
Eric \:\)

[ 01-28-2002: Message edited by: Curve Dominant ]


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Listen, no brickwall. I can FEEL the difference. A superior mastering format. From what I've seen and heard, all the processing obsticles have not been overcome. Distortions are much more subtle and it is reminiscent of the analog days and slew rate limitations. I'm ready for the NEXT generation.

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Quote:
digitaltoast writes:
Listen, no brickwall. I can FEEL the difference. A superior mastering format. From what I've seen and heard, all the processing obsticles have not been overcome. Distortions are much more subtle and it is reminiscent of the analog days and slew rate limitations. I'm ready for the NEXT generation.


digitaltoast,

Pardon, but it appears you misunderstood my question. What you described was an antecdotal account of...of what, exactly? An A/B/X test? Conducted by whom? Under what conditions? Was that difference you heard really DSD, or the platinum-tipped Monster Cables that the system was wired with?

I am also ready for the NEXT generation. I am NOT interested in the NEXT press release from Sony. That is why I am soliciting empirical evidence that suggests DSD is clearly an improvement over PCM, hopefully based on some sort of scientific analysis.

Sorry if I wasn't succinct in my original post.

Eric \:\)

[ 01-28-2002: Message edited by: Curve Dominant ]


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Has GM weighed in on this (perhaps in some other thread)?

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uhh... I think it's *empirical*.

Still, I applaud your request for some stronger information than another set of press releases designed to (yet again) open our wallets. Hope this thread takes off with some real observations based on actual listening!

ML

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Quote:
uhh... I think it's *empirical*.
Still, I applaud your request for some stronger information than another set of press releases designed to (yet again) open our wallets. Hope this thread takes off with some real observations based on actual listening!
ML


Thanks for the spell-check, Mark. A little embarrassing, but I'll get over it.

But, back to the topic at hand: can we take this lack of empirical observation as an indication of a succinct lack of legitamacy of the DSD phenomenon?

Empirically yours,
Eric \:\)


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Curve-

No sweat on the spell. Actually, I had to go to my dictionary (yes, I'm kind of a druid) and look for a bit before I found it myself. Odd word- but a really good one.

"can we take this lack of empirical observation as an indication of a succinct lack of legitamacy of the DSD phenomenon?"

Curve- probably not. I think it possible that the lack of empirical, first-hand observations on the DSD processes sound quality is because there are very few people here on these boards who have used it. This may be because:

A. It is a dismal failure, not a truly legitimate upgrade, fidelity-wise.

B. It is very difficult to purchase, used by few as of yet, so there are very few first-hand observations available as of yet.

My guess is that, on this board, possibly five people have used it.


Here's a question that relates very closely to your topic: Who manufactures this gear as of NOW? Where, if I had the $$$, could I get a DSD recording system? Anyone? GM? Micheal Bishop? Nika?

I'm looking forward to hearing this thing myself!

ML

[ 01-28-2002: Message edited by: Mark Lemaire ]

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Quote:
Originally posted by Mark Lemaire:


Here's a question that relates very closely to your topic: Who manufactures this gear as of NOW? Where, if I had the $$$, could I get a DSD recording system? Anyone? GM? Micheal Bishop? Nika?

I'm looking forward to hearing this thing myself!

ML

[ 01-28-2002: Message edited by: Mark Lemaire ]


The Pyramix DAW can handle DSD (www.merging.com) but converts into PCM so it cheats a little. SADiE can handle 2 track DSD.


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Tascam's got a variation on their DA98 tape 8 track recorder that does 2 channels of DSD. Supposed to be able to sync 16 of them. Only $60,000 for a twenty four track!
Ted


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Everyone,

One advantage is a sampling rate of 2.822 M. M
meaning Megahertz. The other is a low pass
filter cut off of 50K which is alot further
out frequency wise. This is the reason for
less audible artifacts due to filter influences.

"The Crafty One".

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Quote:
One advantage is a sampling rate of 2.822 M. M
meaning Megahertz. The other is a low pass
filter cut off of 50K which is alot further
out frequency wise. This is the reason for
less audible artifacts due to filter influences.
"The Crafty One"


Yeah, I read about that stuff too...in the public relations propaganda. Still no word on how any of that is an "advantage." Less audible artifacts? Which artifacts, and how much less audible?

Anyone else care to attempt to hit this one any farther than the pitcher's mound?

Eric \:\)


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As of now the only DSD recording product that exists is the Sony Sonoma. It will also be the only one that it will manufacture (according to my sources) as they will license all future DSD products to third parties. At the Sony booth at AES they had a variety of displays including Genex.
Some people feel that it will never be a multichannel recording medium, (DSP doesn't work on DSD....) but a mix down or mastering delivery medium. Many claim (engineers that I know that have worked with it, not sales guys) that upsampling to it, from and existing digital source is reason enough to justify it's cost as the final product simply sounds better.
Less fatiguing is the thing they claim most often as there is in reality, less processing involved....
I have a generation one SACD player, while i though it sounded pretty darn impressive, frankly I don't have the attention span to wait 20secs between a play command and hearing music.

I quess we'll have to wait and see though....


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Curve, I haven't touched an SACD/DSD system. I did some internet searches and found mostly hype & trype, as well.

I did find this one site that may indicate the *push* to DSD - "image Recordings" Talks about *features*

I am sure the sonics and 6-channels audio will eventually catch on. But, something tells me the "advanced copy protection" and "anti-piracy" features are a big reason companies like Sony want to chance from CD-DA. Don't get me wrong, the specs. look impressive...

Just for the curious, I found this link explaining SACD/DSD - "Heads Up" Explains the SACD/DSD Physical Media

[ 01-28-2002: Message edited by: AudioMaverick ]
OK, try the second link now... Copy/Paste error...

[ 01-28-2002: Message edited by: AudioMaverick ]



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Curve,

I don't know if I have the time to fully contribute here, but my answer to your question is "no".

Mark,

SADiE, Sonic, Genex, dCS, EMM Labs, Sony Sonoma, are all DSD capable devices. DSD cannot really be edited without first converting to PCM first. For more information you might look at a thread running right now over here:

http://duc.digidesign.com/cgi-bin/ubbcgi/ultimatebb.cgi?ubb=get_topic&f=2&t=005895

I might chime in later. It's a fairly sizeable topic. The bottom line is that we all already have DSD converters and devices, but the DSD signal gets downconverted and filtered for editing. The only difference with DSD in it's common form is that it gets stored at it's basic rate, and then downsampled and filtered for editing later, when you realize that you still have to work with it and add effects to it.

Nika.

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Curve D.,

Apparently you're nearsighted audio
wise, talking this "homebase" jibberish.
You must not be understanding
anything that you've read.(Including
the Sony info..) If you knew anything
about D/A conversion you'd know that
filters used in D/A conversion.
have an extreme influence upon
the sound quality of the converter.
The higher the frequency cut
off the better, & the less steep
the filter slope, the less artifacts
in the frequencies below the
filter cut off frequency. In
theory humans can hear out
to 20K. You need a digital
sampling rate of 40K to reproduce
20K. Filters in D/A's fall
somewhat above this frequency.
In doing audio tests frequencies
above the hearing threshold have
been known to affect those below
the threshold of hearing which
is in the bandwidth of normal
human hearing. Negative filter
artifacts can sometimes come in
the form of brightness in upper
frequencies which in essence can
be an inaccurate reproduction
of the original signal. In
establishing a frequency slope
you have to be very careful in
prescribingthe db/octave that
will produce the best results.
If you don't you'll have what are
called aliasing artifacts. These
aliasing artifacts come in the form of
smeared stereo imaging, & compromised
wave impulse responses. DSD has
helped in removing these by moving
the filter frequency higher,
& therefore fartheraway from the
human hearing bandwith
giving better percieved quality
in digital audio reproduction.
Study some info. about Nyquist
, & then come back, & talk to
us without complaining to those
who are tring to help = (biting
the hands that are trying to
feed you!)

P.S.:Everybody else. I wonder do
I have to include a glossary at the end
of this post for Curve D.?

Including: I.E.
D/A \:D igital to analogue conversion.

S/s, or fs:Sample rate.

etc. etc. etc.. Sheesh!

After coming off as a self superior
you'd think Curve D. would catch on!

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As Cat Stevens once said, "If you were right, I'd agree, but..."

[QUOTE]Originally posted by crafty1:
If you knew anything
about D/A conversion you'd know that
filters used in D/A conversion.
have an extreme influence upon
the sound quality of the converter.


If you knew anything about D/A conversion you'd understand that the whole process of "oversampling" at the D/A is to pull the filter way out of the audible range. A typical oversampling D/A converter of 8x has an FIR filter that is a bandreject filter that is in essence a true, vertical brick wall at the Nyquist frequency, extending upwards to 8x the sampling rate. Then a smooth analog filter is put in at the top frequency of the FIR filter. The low end of this filter is "in effect" a true brickwall because there is no valid audio information in a digital sampling system between 22.05kHz and 44.1kHz, so the filter is inherently a brickwall at 22.05kHz. The effects of an oversampling FIR filter are in no way audible. I think you're confusing this with the effects of an A/D converter's DOWNsampling filter, but an upsampling FIR has no audible effect on the program material if designed properly.

The higher the frequency cut
off the better, & the less steep
the filter slope, the less artifacts
in the frequencies below the
filter cut off frequency.


Which would make none of this true.

In theory humans can hear out
to 20K. You need a digital
sampling rate of 40K to reproduce
20K. Filters in D/A's fall
somewhat above this frequency.


And this is becomes moot.

In doing audio tests frequencies
above the hearing threshold have
been known to affect those below
the threshold of hearing which
is in the bandwidth of normal
human hearing.


And this is the biggest load of marketing B.S. I've seen you say so far. Please, do quote your sources on exactly which "audio tests" you're speaking of. Let me know where I can find a preprint somewhere. In wave mechanics classes in college I must have missed out the day that the professor talked about "frequencies
above the hearing threshold have been known to affect those below the threshold of hearing". Please do explain how this is possible? I've been around this field a while, but apparently not long enough if new revelations were made in this particular field.

Negative filter artifacts can sometimes come in
the form of brightness in upper
frequencies which in essence can
be an inaccurate reproduction
of the original signal.


Pardon me, but what converters are you using? I'd like to stay away from whatever brand that is if you're experiencing unnatural brightness in the high end due to filter artifacts in the D/A's. I'm VERY curious how they implement the filters in those D/A's. Then I'd be curious how aerodynamic they are, because a converter that demonstrates the properties you speak of sounds like it'd be better served measuring the height of a tall building than converting digital signals to analog.


In establishing a frequency slope
you have to be very careful in
prescribingthe db/octave that
will produce the best results.


Right. And with an oversampling converter you'd have to figure out exactly what the slope at 385kHz would have to be in order to mesh with the analog filter that you're adding.

If you don't you'll have what are
called aliasing artifacts.


Nope.

These aliasing artifacts come in the form of
smeared stereo imaging, & compromised
wave impulse responses.


And this would be wrong also.

DSD has helped in removing these by moving
the filter frequency higher,
& therefore fartheraway from the
human hearing bandwith
giving better percieved quality
in digital audio reproduction.


And this is wrong as well.

Study some info. about Nyquist
, & then come back, & talk to
us


That sounds like a great idea. You apparently have a lot of learning to do. I'd be happy to help explain further once you've done your appropriate reading. In the meantime, Eric's questions were valid, and your response was, while empirical, almost entirely false, and thus not what he was really looking for.

After coming off as a self superior
you'd think you would catch on!

Nika.

[ 01-28-2002: Message edited by: Nika ]

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Nika,
a)nice job shutting that one down. ;\)
b)this is a field i'm gradually learning. Can you tell me if this is right or wrong? If i remember correctly, aliasing distortion is a result of:
ie: 44.1khz doesn't deal with energy above 22.05kHz. Any energy that happens in the source signal above this range are then folded back to the frequency equal to the difference. So if said cymbals contained frequencies at say, 23.05kHz, then the 1 kHz that didn't get accepted would get place at 1kHz in the signal that was being recorded. This would cause the aliasing distortion because now you've got this rejected frequency that has been filtered placed on top of an otherwise natural 1kHz frequency from the cymbal. Does this make any sense at all? I'm just not sure if i've got that part correct or not. Forgive me if that was complete gibberish please \:D I'm still learning.
Cheers,
Shiver


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Quote:
Originally posted by Nika:
In wave mechanics classes in college I must have missed out the day that the professor talked about "frequencies
above the hearing threshold have been known to affect those below the threshold of hearing".


nika, i know you know way more technical stuff than i will ever know, but the above point has been brought up often in the last 15 years or so. i happen to believe it is true. do i have any hard data to back me up? no. however, a friend of mine used to have a loud punk rock band that had a regular gig at a school for the deaf. they "felt" the music and loved it.... so who knows?


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Quote:
posted by crafty1:
Study some info. about Nyquist
, & then come back, & talk to
us


Crafty1,

That's exactly what I did, actually. I read about DSD, did some research on Nyquist, FFT, read again about DSD, had some questions...and then, posted this thread.

I apologize, crafty1, if I came off as self-superior, although I must confess, I'm not exactly sure what that means. But, whatever it means, if I'm guilty of it, I apologize if it offended you.

In regards to your reply to my most recent post, please refer to Nika's breakdown of the finer technical details. I can trust in it's accuracy, based on my previous discourses with Nika on the finer points of digital audio technology. I hope you don't mind that deferral, but I am a composer of music foremost, and so I must trust in the opinions of those who's expertise is beyond my grasp.

So, anyhoo...

Tell me something good about DSD...
Eric \:\)


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Quote:
Originally posted by Shiver:
b)this is a field i'm gradually learning. Can you tell me if this is right or wrong? If i remember correctly, aliasing distortion is a result of:
ie: 44.1khz doesn't deal with energy above 22.05kHz. Any energy that happens in the source signal above this range are then folded back to the frequency equal to the difference. So if said cymbals contained frequencies at say, 23.05kHz, then the 1 kHz that didn't get accepted would get place at 1kHz in the signal that was being recorded. This would cause the aliasing distortion because now you've got this rejected frequency that has been filtered placed on top of an otherwise natural 1kHz frequency from the cymbal.


Close. Aliasing is a "mirroring" effect, so the 23.05kHz frequency actually comes out to 21.05kHz as the 1kHz difference is "mirrored" around the axis of the Nyquist frequency.

The best example of aliasing it that common one of the car in the movie. The movie is "sampling" at 24 frames per second. The car starts driving and the wheels appear to be moving forward, but as the car moves faster, the wheels appear to slow down and almost stop. Then the wheels start appear to move backward. This is an aliasing effect, a folding back of the speed of the wheels because the frames in the movie couldn't capture the wheels moving forward faster than the sampling rate of the film.

Cool?
Nika.

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[QUOTE]Originally posted by d gauss:
but the above point has been brought up often in the last 15 years or so.

The fact that it has been brought up many times does not make it inherently true.

If what you are talking about is "difference tones" then we need to qualify that that occurs A. only in non-linear systems, and B. the ear only hears a difference tone when all of the frequencies being played are within the ear's hearing range. So frequencies above 20kHz have no audible effect on your music whatsoever ESPECIALLY because the system we sample at now won't let them through!

If you're referring to something else then we'd have to rewind and figure out what it is and explain where it is relevant, but the statement unto itself is just not correct, and I don't understand well enough what he thinks he's implying in order to correct it.

...however, a friend of mine used to have a loud punk rock band that had a regular gig at a school for the deaf. they "felt" the music and loved it.... so who knows?

Yeah, this is pretty cool. I've read about "bone conduction" tests where they hold a metal rod up to a person's skull and transmit freqeuncies through it and their hearing improves. This was done with people that were hearing impaired - frequencies ABOVE20kHz were transmitted into their skull via metal rods on their bones, and their speech intelligibility went up. How 'bout that!?

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As I understand it, DSD removes the need for all the stuff that happens in order to turn audio into PCM. All digital audio starts off as a stream just like DSD but then gets decimation filtered. If we remove the need for this filtering doesn't it stand to reason that the quality will improve?

I don't know, I could be wrong. Nika, have you read this/

http://www.positive-feedback.com/0802/pappas.Meitner.rev.8n2.html

Here's an extract:

"Pappas: I understand why I think it’s a better format. Tell me why you guys think it’s a better format.

Meitner: To convert audio into PCM is a very alien thing, whereas if you look at the convert audio into one-bit format, it’s a very natural thing. In any form of conversion, you will lose something. You have to choose the format where you lose the least, which means the format that’s the friendliest to audio, which is definitely DSD over PCM.

Just look at one problem with PCM. Imagine what happens at your zero crossing. You have all those bits flipping. You have, you know, noise shock in the system coming off the power supply if all of a sudden 23 bits change from all zeros to all ones. You have that at every zero crossing. And you need really good error correction, because if a sign bit gets screwed up in the process, all of a sudden instead of your signal being positive, it thinks it’s negative."

Is the guy accurate in what he's saying here?


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[QUOTE]Originally posted by Rog:
As I understand it, DSD removes the need for all the stuff that happens in order to turn audio into PCM. All digital audio starts off as a stream just like DSD but then gets decimation filtered. If we remove the need for this filtering doesn't it stand to reason that the quality will improve?

What it is most respected for is the removal of the filter at the A/D side, not the D/A.

I don't know, I could be wrong. Nika, have you read this

No, but I'll read it later.

Here's an extract:

Meitner: To convert audio into PCM is a very alien thing, whereas if you look at the convert audio into one-bit format, it’s a very natural thing. In any form of conversion, you will lose something. You have to choose the format where you lose the least, which means the format that’s the friendliest to audio, which is definitely DSD over PCM.

This is an indication of what I don't care for with Ed Meitner. Everything he says sounds so "mystical" instead of substantiated with something other than terms like "alien", and "friendliest to audio". What in the world does that mean?

Just look at one problem with PCM. Imagine what happens at your zero crossing. You have all those bits flipping. You have, you know, noise shock in the system coming off the power supply if all of a sudden 23 bits change from all zeros to all ones. You have that at every zero crossing. And you need really good error correction, because if a sign bit gets screwed up in the process, all of a sudden instead of your signal being positive, it thinks it’s negative."

What he's referring to is that digital audio often works in what is called "two's complement" math. The short version of this is that:

3= 0011
2= 0010
1= 0001
0= 0000
-1= 1111
-2= 1110
-3= 1101
etc.

The reason for this is simple - it makes addition in the digital environment easy:
3+(-2)=-1
0011
1110
0001

etc.

Now the issue is that at the PCM conversion process when you go from 0 to minus 1, all 24 of your bits flip, which is quite a bit of a voltage change, drawing on the power supply, and effecting the rest of the conversion process. Since this crossing happens twice in every cycle, Ed is talking about a digital "shock" to the power supply and the voltages in the chip twice in every cycle. In DSD this does not happen. There is never a voltage swing of greater than 1 bit, which is very small.

What Ed speaks of is a valid concern, but something that has been dealt with by manufacturers. I have sent a couple of emails out to manufacturers to clarify exactly how, but I believe it involves not CONVERTING in two's complement, so that the voltage shock doesn't happen at the converter process, but then converting the resulting signal to two's complement through a simple math algorithm.

In any event, I'm pretty sure that the issue he's raising is not really a concern anymore, but it is rather an obstacle that PCM manufacturers had to deal with at one point.

Does this help?

Nika.

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Nika wrote:

Quote:
This is an indication of what I don't care for with Ed Meitner. Everything he says sounds so "mystical" instead of substantiated with something other than terms like "alien", and "friendliest to audio". What in the world does that mean?


Oh come on, Nika. What's so mystical about that? He's pretty to understand. He IS an engineer, after all.

Could it be that with Ed Meitner being heavily involved in the design of the new Digidesign 192 interface, you might be trying to find creative ways to discredit him because you see a lot of big ticket Apogee and Prism sales going out the window (in favor of people buying Digidesign 192's). ;\)

If you really want to cast aspersions on his integrity, please find something a little more substantial to criticize than the adjectives he uses. Just because he's a creative wordsmith is not a reason to not LIKE him.

Quote:
What Ed speaks of is a valid concern, but something that has been dealt with by manufacturers. I have sent a couple of emails out to manufacturers to clarify exactly how, but I believe it involves not CONVERTING in two's complement, so that the voltage shock doesn't happen at the converter process, but then converting the resulting signal to two's complement through a simple math algorithm.


I also wonder if the Ed Meitner idea of synchronizing the clock frequency of the switching power supply with the frequency of the converters (as implemented in the new Digi 192) might in some way also help this situation.

Lee Blaske

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Lee,

I've read a lot of stuff from Meitner, and I've talked to him as well. I was considering buying a set of his converters for use in PCM, but I found it so difficult to get anything from him tangible. It all came down to what I considered unsubstantiated claims about "biorhythms" or "natural sampling rates" or other "feel good", ethereal stuff, and very little actual information about where DSD shines from an analytical or empirical point of view.

I talk to any other converter manufacturer and I hear about their power supply, the IC's that they're using, the implementation of the filters, the analog section, yadda yadda, and I ask questions and get some pretty empirical answers. I've noticed that what I get from Mr. Meitner is more "feel good" verbiage like is probably adequate in the hi-fi industry, but not appropriate for those of us (or just me?) that actually want to understand where and if the differences occur. Bogus claims of "unnatural sampling rates" don't do me much good, and are also not akin to the type of "empirical" evidence that Eric was looking for when he started this thread.

Maybe it was in poor taste to publicly make my opinion of this known above? It's just kind of grated on me for about six months I suppose.

Also, do we know how much involvement Ed had in the Digi interface? I've seen this rumoured through many people except for Digidesign or Ed directly. All I got confirmation on is that they use the same conversion chip. If I missed a thread or some information elsewhere please let me know. That'd be great if it is the case, but I just haven't seen it really confirmed, yet.

Thanx!
Nika.

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Quote:
posted by Lee Blaske:
Could it be that with Ed Meitner being heavily involved in the design of the new Digidesign 192 interface, you might be trying to find creative ways to discredit him because you see a lot of big ticket Apogee and Prism sales going out the window (in favor of people buying Digidesign 192's).

If you really want to cast aspersions on his integrity, please find something a little more substantial to criticize than the adjectives he uses. Just because he's a creative wordsmith is not a reason to not LIKE him.


Lee,

That interview with Ed Meitner was one of the things I had read which lead me to suspect the validity of DSD, and post this thread. In my experience with human nature, I've found creative wordsmithing often to be camoflage for bullshit. To wit:
Quote:
Pappas: Everybody’s brain is so accustomed to having PCM represent an absolute value, and there are certainly a fair number of naysayers out there who say that you can’t do it.

Meitner: Yes, well, there always is. The thing is, there is an obsession with absolutes, totally forgetting that human sensory inputs are not so much sensitive to absolutes as they are to deltas, and our hearing is really no different. So, you know, this is also one of those philosophical things where PCM is this absolute machine, and DSD is this relative machine. And it’s just better for us as humans. And I firmly believe that general health would be better if PCM would not exist.

Pappas: And why, particularly, is that?

Meitner: Because there is a subliminal irritation about PCM that may just affect the psyches of people in a bad way, and certainly distracts from the pleasure of listening to music. And if listening to music was considered as relaxation and was supposed to be a way to relieve stress, then PCM, like CD playback, certainly doesn’t do it as well as some of the old analogue stuff did.

Pappas: So maybe that’s one of the reason the music industry sales have been down.

Meitner: Possible. Aside from the fact that, right now, it doesn’t seem to be the same scene as I remember from the ‘60s and ‘70s. This is a hard thing to say, but I hear from a lot of people that, with an LP, you used to sit down, close your eyes, and sort of float away with the music, relax and unstress. And with CD, it’s just not the same thing anymore. So even though you might not hear the problems glaring at you immediately, I’m sure they wear.

Pappas: Do you think the other thing might be the fact that when you ran an LP, it was generally about 22 minutes, and then it was time to get up and change it?

Meitner: Well, that could be too. Now with the CD you have the remote control. You can change it at all times. But I find a lot of people don’t even get to 20 minutes.


Please, give me a fucking break. Maybe the problem is we all don't drop the same quality of LSD as we used to! And did you catch the "dancing bits" reference earlier in the interview? What, am I at the fucking ballet now?

Meitner and Pappas both emphasize in that interview that Meitner's background is firmly in analog. Now, I don't have the same detailed grasp of the technical issues that Nika does, but reading that interview gave me the vague impression that perhaps Meitner A) didn't really know what he was talking about, or B) knew he was talking shit, because he was trying to make a sale.

And I had no idea he was in any way involved with Digi.

Eric \:\)


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I have to agree with Curve and Nika here. Like Curve, i'll admit i don't have near the technical understanding about this that Nika does, but i'm also a gear pimp part-time. If i fed a line like this to any of my customers, they'd think i was some sort of freak, and i live in the Land of Etherialism and "feeling" (Victoria, BC). The impression that i get from that interview is that basically, he's trying to convince us that our brains have become too lazy to understand and perceive "true" fidelity in audio. I could understand if he was doing a speech on music therapy and the possible benefits of higher fidelity on the brain, but he's done little or nothing to explain anything of any technical value to the professional audio community.
Just my "average joe consumer" .02 cents.
cheers,
Shiver


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Nika, & Everybody else,

Before you come to the conclusion
that anything "Nika" said has
validity (in Nika's return post
to me).

Read this \:\( Especially the section
detailing anti-aliasing filters)

http://www.dcsltd.co.uk/dacs.htm

And this is just info. that would
just be scraping the surface of a
long debate that would annul your
(Nika's) theories. Do some more
extensive study into DSD. It's not
your standard oversampling scheme.
Record, & playback are at the
same rate = 2.822 Mhz. Oversampling
is not involved. So
before you go blowing me off
get all your facts straight. DSD is not
like PCM. Oh, & by the way dCS makes
the best converters on the planet.
Don't believe me? Check the users
list out.

In closing:Rhetoric convinces people
sometimes more than facts. I think
you've all been duped by Nika's
rhetorical spiel vs. true facts
that exist in the real realm
of audio. Nika's spiel reminds me alot
of tapes I hear of Adolf Hitler.

"The Crafty One".

[ 01-29-2002: Message edited by: crafty1 ]

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crafty1 wrote:

Quote:
If you knew anything
about D/A conversion you'd know that
filters used in D/A conversion.
have an extreme influence upon
the sound quality of the converter.


crafty1,

In discussion of this stuff, can we please try to move away from using words like "extreme." Certainly, it's nice to see equipment improving, but there IS no "extreme" left anymore. Audio engineers have been working very hard over the years, and for most intents and purposes, they've pretty much got this stuff nailed. The difference between wax cylinders and CDs was extreme. The difference between high quality, high bit/sample rate PCM conversion and DSD is NOT extreme (especially if played back in real world listening environments).

If you stop speaking of these improvements in hugely hyperbolic terms, people might take you a little more seriously.

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Nika wrote:

Quote:
Also, do we know how much involvement Ed had in the Digi interface? I've seen this rumoured through many people except for Digidesign or Ed directly. All I got confirmation on is that they use the same conversion chip. If I missed a thread or some information elsewhere please let me know. That'd be great if it is the case, but I just haven't seen it really confirmed, yet.


Nika,

This info has been scattered through the DUC. I would think that at some point, Digidesign will make more of a big deal about this, but I'm sure at this point they're scrambling to get HD out the door.

Here's a quote that had the most info so far. (Excuse me for pasting it in again - I suppose I could just give a link to the thread, but you'd have to do a lot of scrolling to find it.) Ed Meitner is mentioned toward the top. My information is that Bruce Jackson, another person on the design team, was the founder of Apogee.

From Dave Clementson of Digidesign Engineering:

Quote:
The clock schemes are basically the same on the 192 and 96. Both boxes use the new LVDS-based DigiLink interconnect for tighter timing, longer cable runs, autoconfigure, and higher I/O density per card (thanks, John Weitz). They also use an all-new low phase noise, ECL PLL-based clocking scheme (thanks, Dave Hedberg and Bruce Jackson) to support the new multi-rate Loop Sync architecture. This new clock generator gives better jitter performance than any of the 3rd-party clock sources I've tested (which is all them except Lucent, which I haven't had a chance to measure yet). It also gives good jitter rejection from external clock sources.

Besides what you get with the 192's AES/TDIF/ADAT/SRC card (again, thanks John), the main differences between the 192 I/O and 96 I/O are in the analog signal paths. The 192 has

* ultra-wideband class A discrete bootstrap ADC driver (thanks, Ed Meitner)
* active power decoupling for every active stage (again, thanks Ed)
* fully symmetrical, double balanced signal path
* much better performing ADC chips (the DAC chips are the same as in the 96 I/O but the 192 has better analog circuitry)
* private low-noise shunt regulation for each converter rail
* only Vishay or Beyschlag MELF resistors and dual-biased Wima polypropylene caps in the signal path
* DC-coupled signal path with DC servo offset compensation
* laser-trimmed input network for optimum CMRR
* balanced soft-knee ADC clipper
* separate software-selectable +4 and -10 input jacks
* wideband class-A discrete balanced output driver
* low-noise, wideband current-feedback DAC buffer (thanks for the idea, Michael Grace)
* passive, pulse-optimized DAC anti-image filters
* all relay switching
* dual gain adjusts on each input and output channel for two separate headroom calibration presets
* individually shielded channels for low crosstalk

Oh yea, and it opeates at 192kHz as well, although that wasn't really the point of the design (although its "192" name obscures that fact). The point of the design was to make a truly fine converter box with absolutely no compromises (cost or otherwise) in the signal path, lots of useful features, and the flexibility and future-proofing a modular design affords. And it obviously wasn't an effort by just "anybody."


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Lee Blaskey,

If you judged differences in
audio equipment on the type
of equipment I'm exposed to
(Cello Audio, B & W, Krell
etc. etc. etc.= $100,000, &
up system costs) you might
use the word "extreme" as
well in your description
of differences. DSD is
not for the person
where MP3, & walkmans
are good enough.

"The Crafty One".

[ 01-29-2002: Message edited by: crafty1 ]

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posted 12-10-2001 02:46 PM
--------------------------------------------------------------------------------
Roger Nichols, & All,
1.)What is the dynamic range of SACD, & Direct Stream Digital(DSD) in theory?

2.)What is the sampling rate, & high frequency range of SACD, & DSD in theory?

3.)What are the typical noise floors for SACD/DSD products?

4.)How is (24bit 44.1/48/88.2/96K/192K material) = (PCM) converted to DSD.

5.)Are there any frequencies more optimal to work with as sources when considering the final format conversion from PCM to DSD.

6.)Are there any DSD recorders availble for multitrack to 2 track mixdown.

7.)Is DSD going to be available for multi-channel audio mixes including video?

"The Crafty 1".

Crafty, out of curiosity, how did you become such a "brilliant expert" in this subject in such a short period of time? I'm just curious of course, but it would seem to me that Nika's probably done a bit of study into this, as well as Curve i'm sure. Did you learn everything there is to know about it since December? ...amazing.
Cheers everyone,
Shiver


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Quote:
In closing:Rhetoric convinces people
sometimes more than facts. I think
you've all been duped by Nika's
rhetorical spiel vs. true facts
that exist in the real realm
of audio. Nika's spiel reminds me alot
of tapes I hear of Adolf Hitler.
"The Crafty One".


crafty1,

I know Nika, have spoken with him on the phone, and exhanged emails with him, as well as discussing digital audio on these threads. I just spoke with Nika on the phone today, albeit briefly, about this subject. I can assure you that Nika is about as Hitler-esque as Woody Allen. You've lost what little credibility you had with that gratuitously tasteless personal attack.

Back to the topic: I've been doing some more research, and have some more views to add in that regard, which I'll post later, because I have to go right now. I have a singer coming here to track vocals in about 20 minutes.

Eric \:\)


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Quote:
Originally posted by crafty1:
Nika, & Everybody else,

Before you come to the conclusion
that anything "Nika" said has
validity (in Nika's return post
to me).

Read this \:\( Especially the section
detailing anti-aliasing filters)

http://www.dcsltd.co.uk/dacs.htm

And this is just info. that would
just be scraping the surface of a
long debate that would annul your
(Nika's) theories. Do some more
extensive study into DSD. It's not
your standard oversampling scheme.
Record, & playback are at the
same rate = 2.822 Mhz.

(etc. etc. ad nauseum)
[ 01-29-2002: Message edited by: crafty1 ]


Look here bub...

A question was asked if anyone had evidence that DSD was better than PCM as a format. This would include PCM at higher sample rates (not just some oversampling scheme) where the anti aliasing filters are at a much higher cutoff (or did you forget to read up on that?)

96 kHz puts the filters @ 48 kHz
192 kHz puts the filters @ 96 kHz

So come up with something better than belittling bullshit and "I know more 'cause my gear is bigger than your gear". Try not to kick the cat while typing.
:rolleyes:

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OK, my singer called to say she'd be late, so I have a little time to add one thing...

EVEN IF DSD was some kind of superior high-resolution format, the whole logic behind using it as a consumer-end delivery system shoots itself in the foot...

1. Since one cannot edit DSD, it is rendered impractical and unusable for recording sessions.

2. Therefore, sessions will continue to be recorded in PCM.

3. So, all of those "evil" sounding artifacts allegedly produced by PCM will be faithfully rendered in high-resolution by DSD.

Correct me if I'm wrong, but if the signal hits PCM at any point in the path, the end listener is essentially hearing a PCM signal. Which would mean:

A. For a consumer to buy a DSD system, and DSD discs, would be a complete waste of money

B. For an engineer to master to DSD would be a complete waste of time and money

C. This whole concept of DSD as a practical option for the consumer is a complete hoax.

Where is the fundamental flaw in that equation?

Eric \:\)


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Quote:
If you judged differences in
audio equipment on the type
of equipment I'm exposed to
(Cello Audio, B & W, Krell
etc. etc. etc.= $100,000, &
up system costs) you might
use the word "extreme" as
well in your description
of differences. DSD is
not for the person
where MP3, & walkmans
are good enough.

"The Crafty One"


crafty1,

According to that post, DSD is not for the consumer. It is only for the golden ears using the $100K rigs such as yourself. If only you hear the difference, it must be anything but extreme, and a waste of resources at that, since when the consumer hears your recording, any benefits will have been completely wiped out by the time it hits the speakers.

You're just not making the sale, Sparky.

Eric \:\)


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Shiver,

It usually does'nt take me
long to pick up on a
subject especially
if I become really
interested in it.
After all I'm
"crafty".

"The Crafty One".

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Curve D.,

Who cares if you think I'm credible or not. As long as I'm right! And in this case I am. Additionally DSD can be edited. Sony has a product called a "Sonoma" that does this. On another note I do agree with you that the average consumer may not be able to hear the difference but why worry. Sony kicked out the $ to develope it not you. Save up for a high end system, & then enjoy. If no then keep what ever you listen on currently, & stick to buying CD's. Read the "dCS" documentation yet? It (the dCS info.) sure makes me look credible in the eyes of anyone else who really wants the facts, & does'nt post here just to be a part of the "Our Gang" club that seems to have formed here. Are you looking for facts, or a social club to belong to?

If your loyalty to Nika is based upon technical viability then your friendship just got washed away with the last high tide (Uh, rather my 1st post referencing the dCS stuff).

"The Crafty One".

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Oh, gawd...

First of all, the dCS literature is a bit confusing. What they have done is list all of their features and then listed that on both their ADC and DAC pages. You'll notice that there are quite a few things on that same page that are exclusive to their ADC's:

Quote:

dCS ADCs make extensive use of of FPGA technology...


In any event, what they are describing in the paragraph you are referring to is the set of filters that are used when they down convert the DSD signal for conversion.

The delta sigma modulator that is used in their A/D converters runs at the same rate as DSD - 2.2822MHz. They then downconvert and filter the material into PCM using different anti-aliasing filters. dCS is unique in that they provide up to 4 anti aliasing filters to choose from, and have worked out all of the math necessary to have four very complex FIR filter sets within their boxes.

Since they already did all of this work on the A/D side of things, and since DSD is akin to taking the signal straight off of the delta-sigma modulator in an A/D, they merely provided the same microchip with the same four algorithms for your filter choices in their DAC. This is no longer an "anti-aliasing" filter, as it's not preventing any aliasing on their DAC. I sent them an email today about this "misprint" on their website. It is instead a simple low pass FIR filter used to remove the harsh artifacts that DSD creates at around 70kHz. Since it is not preventing aliasing, and since there is no danger in allowing slightly higher than 22.05kHz material to pass through the system at the DAC side, there is no reason that someone wouldn't choose the highest of the four options they provide.

What is obvious, since there is not such a thing as an "anti-aliasing" D/A filter, is that they merely took the text that they'd written to describe the functionality of this chip as it more pertinently relates to an A/D, and pasted it into the D/A spec sheet, just like they did when describing their "FPGA" technology above.

This helps explain this quote from the same page:

Quote:

In DSD mode, the dCS 954 has 4 filter settings. This allows the output signal bandwidth to be adjusted to suit the capability of the monitoring chain.


It appears that dCS also allows the user up to four "reconstruction" filters on their DAC's. It appears that they may have had a misprint in regards to this as well, as there is not a specification listed in regards to their reconstruction filter settings. Instead there is only a reference back to the ADC "anti-aliasing" filter settings. Clearly this is a misprint. This is evidenced by the following quote off of the same page:

Quote:
A  Function Menu gives access to a wide range of extra functions such as reconstruction filter selection, self test, ...


If they really provide multiple reconstruction filter settings it'd be assuredly mentioned on the list of features at the top of the page. But it is not, an indication that they haphazardly threw the page together and accidentally copied too much of their pre-written text from their archives and other pages.


Crafty,

This is unfortunately what happens when you do very little homework and start to plagiarize other people's work, almost copying verbatim: when they are wrong about something you end up being wrong about it as well. This is unfortunately the situation you find yourself in now.

Anybody in here knows that there is not an anti-aliasing filter in a DAC. There is no question about this whatsoever. And in the attempts to substantiate this, claiming that with as little knowledge as you actually have in this field that you are "exposed to" $100,000 systems playback systems is frankly an embarassment to you I'm sure. If your only validation of dCS making the best converters is their list of clients then you might want to re-evaluate your ranking of sound quality.

dCS makes fine converters, and I researched buying a pair in my latest investment. About five years ago they were the real cat's meow of converters, borrowing mostly from their influence and experience in the audiophile markets. As of late, many classical and mastering engineers have shyed away from dCS in light of other companies that also make component level converters. Many of the clients on dCS' website also own brands such as EMM Labs, db Tech, Prism, Weiss, custom made ones and others.

I chose not to get dCS converters myself because of the same reason that many others are citing - poor command of inbound jitter and a sonic character that is just not as transparent nor as pleasing as another box I mentioned above. I chose db Tech instead. Feature wise dCS is very compelling, being the only off the shelf box that can do up to 192kHz sampling rates and DSD in one box with great versatility, but sonicly it is not what I needed. EMM Labs is another company making DSD/PCM boxes for conversion, and I expressed my reasons above for not purchasing their box.

It really appears to me that you have a lot more homework to do if you intend on being helpful in threads like this that are looking for factual information and not poor regurgitation and plagiarism of misprinted information on people's websites.

Frankly the fact that you are trying to explain to me how the conversion process works, and clarifying how oversampling "schemes" work in PCM converters as opposed to DSD converters I find a trifle amusing. It seems as though you've amused many others in this thread as well.

Nika.

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This has to be one of the strangest things I've read on this forum.

Quote:

Now the issue is that at the PCM conversion process when you go from 0 to minus 1, all 24 of your bits flip, which is quite a bit of a voltage change, drawing on the power supply, and effecting the rest of the conversion process. Since this crossing happens twice in every cycle, Ed is talking about a digital "shock" to the power supply and the voltages in the chip twice in every cycle. In DSD this does not happen. There is never a voltage swing of greater than 1 bit, which is very small.


Its not you, Nika, but what manufacturers (and that dcs guy) are saying. For the life of me, I can't think of any sane AD design that switching zeroes to all ones or vice-versa would cause a disruptive current suck on the power supply.

I was so astonished at reading this, that I just had to review several popular designs from parallel-encoding comparator banks (worst case), successive approximation, and single-chip converters and so on to see if it could possibly be true. I just don't see a reasonably designed digital output circuit that overloads its power supplies--even at high audio frequencies.

I suppose it could happen, but in order to get that effect, I think you'd have to try really hard or be very very unlucky. Voltage-shock, indeed.

-Dennis

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>> ATTENTION <<

Since we are now on page 2, I will re-post this link that Nika posted on page 1. Please read it thoroughly before you jump in with half-baked analyses...

http://duc.digidesign.com/cgi-bin/ubbcgi/ultimatebb.cgi?ubb=get_topic&f=2&t=005895

Nika,

So, how DOES one dither 1 bit, and where DOES the dither noise go?

Eric \:\)


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Curve D.,

Stop hypothesizing, & and then trying
to draw hard conclusions from your
unsolid hypothesis. I'd get a
full reply from dCS. Additionally
I did'nt plagarize because
dCS got credit. Did'nt I provide
a link to their web-site. How
much more credit do they deserve.
I'm not writing a scientific white
paper on this. This is not an AES
show. It's Music Player/Expert Forums
which at times is far from expert.
Honestly I was just trying to help
you out in an area I already have
some degree of understanding, & there
you go getting all sour on me, & the
rest of us posters/participants. I
notice neither you, or Nika can now deny
the presence(DSD wise), or importance of
filters in the digital domain.
Additionally you must have missed
info. in your Sony reseach because
they specifically state that they
apply a filter as a part of the
D/A function. Don't believe
me go to page 4 of this document:

http://imagtest.sel.sony.com/SEL/consumer/sacd/pdf/SACD.pdf

And while you're at it read this:
p.12, & then p. 13 which confirms
info. in my earlier statements, &
does apply to PCM playback.

http://imagtest.sel.sony.com/SEL/consumer/sacd/pdf/SCD1.pdf

To be honest I think you, & Nika
are about a month behind me on all
this. Possibly years regarding
PCM info..

Go figure to you both(Curve D., & Nika).
And don't be so sore. If you're wrong
you're wrong.

"The Crafty One".

Everyone else,

Ooops! I forgot my footnotes, &
bibliography sections. Paaaardon me!
(Joking:Hah ha!). Oh, & you can also
bet that Curve D. is going to be
in servere denial about the above
info. provided directy from Sony.
Curve D. might even try to
convince that DSD does'nt even
exist, and that it's PCM repackaged
with a new nameplate kind of like
a Cadillac/Escalade is just a
GMC/Denali with a Cadillac hood
ornament!

"The Crafty One".

[ 01-30-2002: Message edited by: crafty1 ]

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Dennis,

When it comes to understanding circuitry I'm sure you'll do much better than me, but this concept is something I've stumbled on a few times before. I think the first time I heard about it was through Ken Pohlmann's books. I did a quick search on the internet and I found the following paragraph. Perhaps this more closely describes not only the "glitch" and why it occurs, but how it is dealt with:

Quote:
The binary format used in digital audio transmission is called binary two’s
complement. This format allows for 32768 different signed integers in the
16-bit world.
Unfortunately this format requires that all of the bits must switch from all ones,
binary zero minus 1LSB, to all zeroes, binary zero, when the analog signal
transitions through zero. The resulting output from a R-2R DAC can contain a
glitch due to the fact that all of the current sources are switching at the same
time. This problem can be solved in a variety of manners. The cleanest
response is achieved by using two back to back DACs. This is the topology
that is referred to as “CoLinear”.


I hope this helps.

Nika.

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[QUOTE]Originally posted by crafty1:
...Honestly I was just trying to help
you out in an area I already have
some degree of understanding...


I think "some" is an appropriate term.

To be honest I think you, & Nika
are about a month behind me on all
this. Possibly years regarding
PCM info..


You just started researching DSD a month ago and you think I'm a month behind you? You think I just heard the term this afternoon? You think you're years ahead on PCM yet somehow think that anti-aliasing filters go on the D/A side? Somehow you've determined that d/a conversion can allow aliasing through? And that this can make it sound "brighter"?


You're something else, kid.

Nika.

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Nika,
you know what? You know what i think is REALLY REALLY sad? .... that somebody's gone through their whole life with the name Stanley Lipshitz or whatever it was....that bummed me out.....really... what a drag. As IF he didn't get called Shitlips..... I thought having the last name Wang was tough, but boys oh boys....
Shiver
(Adam Wang...pronounced Vong...it's norwegian...no jokes, i've heard 'em already \:\)


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Thanks, Nika, now I see how it could happen in D -> A. Pohlmann's book has a pretty good section explaining it. I was looking at A -> D circuits. Funny that the inverse process using the R-2R method ( A -> D ), doesn't have this glitch. I see why.

BTW, how do you like your new db Techs? Have you gotten a chance to field test them yet?

-Dennis

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Dennis,

Someday you can explain to me in enough detail that next time I explain it properly.

I haven't gotten a chance to use the db Techs yet. I just moved them from their prized pedestal in my studio a few minutes ago. I'm in the process of purging and reaquiring right now, and the whole studio is essentially down. I have a few other purchases to make as well before it all goes back together. Next up is a Grace 901, I think. I also need to get some nice cases for everything pretty soon, a talkback circuit for classical work, a digital patchbay for home, a passive monitor attenuator for both work and home, etc. It'll be a little bit before it's all reassembled, but I expect it to be all right when it's all put back together. I'll get you some picks if you're interested, and once I've used the converters I'll send you a file. Did we ever hear back from Daniel on his Prism/db Tech shootout?

Nika.

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i believe if upsampled digitally to PCM, some of the limitations associated with the analog section in any PCM ADC could be bypassed. maybe my question is:

is it really artifacts we are concerned with here? though lacking the proper equipment to produce controlled tests myself, i've heard many claims of improved imaging, clarity and 'bottom' from higher sample rates. i've also heard people like Bob Katz comment on how they feel more relaxed when listening to DSD... as of yet i have yet to hear the word 'artifact' used in this context - are we maybe barking up the wrong tree?

in a conversation with a Tascram (yes, tascam of all people) rep a couple of months back, he had mentioned a conference in western canada where a mathematician proved to sony a 3bit format would be more efficient, and yield higher fidelity. i'm not sure how this relates to two's compliment (or if he even had his facts straight) - has anyone else heard this rumor?

also, in continuation - theoretically speaking (as an extension of the 96k thread?) we would not greatly benifet from increased sample depth (although i must admit the idea behind the multi-adc converter in neuman's D-01 is quite intreguing). also, theoretically, a greater fS will not genereate a difference either. how can resolution /actually/ be improved? only the analog section is left... any thoughts?

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Quote:
Originally posted by synesthesia:
i believe if upsampled digitally to PCM, some of the limitations associated with the analog section in any PCM ADC could be bypassed. maybe my question is:



i appologise for any redundancies in the above.... i failed to read any of page 2 before posting this, as a response to Curve's question at the end of 1... hope it wasn't too out of line.

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Nika, & Curve D.,

You're mixed up again. Your initial
spiel is regarding info. on DSD.
The info. you researched at
dCS is in relation to DSD.
DSD = Direct Stream Digital.
Direct meaning no filters
at the A/D conversion stage.
The filtering applied in
the DSD process takes place
at the D/A. This is
where filters(you, and
others sometimes might
term them "anti-aliasing"
for filtering normally used
when describing filtering
at the A/D in PCM) are
applied aspecifically
to DSD, (& normally
not applied in terms
when describing PCM
D/A conversion).
This is why you're confused
at the info. given on
the dCS site. Go figure
again. To get an understanding
of this go to the Sony documents
I've posted earlier on
this page, and via
the link at the end
of this post. If dCS does apply
a filter in the A/D process then
their DSD A/D converter is'nt
a DSD A/D converter in pure
form according to Sony's
description of the whole
DSD process. I'll have
to check into this myself.
Overall use the source
Sony documents I've listed
as your foundation in
acquiring knowledge of
DSD, & take other info.
from other manufacturers,
& engineers as additional
building blocks. Go to
the foundation apprentices.
Theoretically the filters
in DSD do what normally
would have been taken
care of at A/D in PCM.
In reality the filters
in D/A in DSD do
what the filters
at A/D, & D/A do
in PCM so they
from one
perspective can
be looked upon
as a type of
anti-aliasing
filter. Do
you get it
now theroretically
, & contextually?

"The Crafty One".


Curve D., & Everyone else,

I suggest you thoroughly review
these Sony documents to see that
principals you have locked into
your brains regarding PCM don't
even apply to DSD. I rest my
case. Re-learn it. Don't be
old dogs that can't learn
new tricks(new digital theory).

The above links can all
be sourced from this page:

http://www.sel.sony.com/SEL/consumer/sacd/static/download.html

"The Crafty One".

[ 01-30-2002: Message edited by: crafty1 ]

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Crafty what the hell are you doing that's making your posts come out five words wide and a billion lines long?


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Great thread ... apart from the moron.

Nika - can you tell me why DSD only allows basic editing and do DSP processing (without PCM conversion)?

Also - I think Genex do 192 and DSD conversion in one box?


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[QUOTE]Originally posted by synesthesia:
is it really artifacts we are concerned with here? though lacking the proper equipment to produce controlled tests myself, i've heard many claims of improved imaging, clarity and 'bottom' from higher sample rates. i've also heard people like Bob Katz comment on how they feel more relaxed when listening to DSD... as of yet i have yet to hear the word 'artifact' used in this context - are we maybe barking up the wrong tree?

It depends. Filters can have artifacts. Converting too low will cause HF distortion or attenuation. They're two different things, and it seems that DSD proponents seem to talk about both.

in a conversation with a Tascram (yes, tascam of all people) rep a couple of months back, he had mentioned a conference in western canada where a mathematician proved to sony a 3bit format would be more efficient, and yield higher fidelity. i'm not sure how this relates to two's compliment (or if he even had his facts straight) - has anyone else heard this rumor?

It has nothing to do with two's complement. This gets into the stuff that I'm not really straight on, but from what I understand, people are using multibit modulators now, so it would have been a while ago that this conversation would have happened? The mathemeticians and critics of DSD have all kinds of issues, including dithering issues and whatnot. Apparently a multibit modulator solves some of the issues, but I'm not exactly sure how. It would take a Paul Frindle to explain that one to us, and he's avoiding this thread like the plague.

also, in continuation - theoretically speaking (as an extension of the 96k thread?) we would not greatly benifet from increased sample depth (although i must admit the idea behind the multi-adc converter in neuman's D-01 is quite intreguing). also, theoretically, a greater fS will not genereate a difference either. how can resolution /actually/ be improved? only the analog section is left... any thoughts?

Well there's always the linearity of the converter and the slope of the filters and the analog section and I'm sure there are other areas I'm not privy to.

Of course there's always microphone placement, gain staging, engineering skills, yadda yadda yadda. I'm not totally convinced that we COULDN'T be fooled between reality and digital if we just used the top shizmo equipment.

Nika.

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Quote:
Originally posted by Rog:
Great thread ... apart from the moron.

Nika - can you tell me why DSD only allows basic editing and do DSP processing (without PCM conversion)?

Also - I think Genex do 192 and DSD conversion in one box?


Rog,

I knew that Genex had converters that did both, but I wasn't sure they were in the same box.

As for your other question, you understand that DSD doesn't have any location reference - it only tells you whether the next sample is higher or lower than the previous one. So the data stream is just a series of single bit 1's and 0's - either up or down. 1 0 1 0 0 1 0 0 1 0 1 1 0 0 0 1 0 etc.

Now in that little data stream I just pointed out, there are more zero's than one's. This indicates that the data stream is tending downward, right? It moved down more than up. But where is this in relation to the zero crossing? What is the amplitude of the waveform at this point? We have no idea! Thus we really can't edit this waveform, or cut and paste it without first reconciling at least some of this data to the location of this bit of data in relation to the zero crossing. We really don't have to know where EVERY sample is, but we should at least know where SOME of them are, no? If we knew the specific location of every 64th sample then we could actually edit with the precision of 44.1kPCM, as 44.1kPCM is 1/64th the sampling frequency of DSD.

Now we don't necessarily have to convert EVERY sample to PCM, but we have to have enough of a reference to know where, exactly to place this bit of data once it's moved.

That's the simplest answer for you. If you want to talk about processing it's a different story, but for editing that's a pretty good reason.

Does this help?
Nika.

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Quote:
Originally posted by Nika:


Rog,

I knew that Genex had converters that did both, but I wasn't sure they were in the same box.

As for your other question, you understand that DSD doesn't have any location reference - it only tells you whether the next sample is higher or lower than the previous one. So the data stream is just a series of single bit 1's and 0's - either up or down. 1 0 1 0 0 1 0 0 1 0 1 1 0 0 0 1 0 etc.

Now in that little data stream I just pointed out, there are more zero's than one's. This indicates that the data stream is tending downward, right? It moved down more than up. But where is this in relation to the zero crossing? What is the amplitude of the waveform at this point? We have no idea! Thus we really can't edit this waveform, or cut and paste it without first reconciling at least some of this data to the location of this bit of data in relation to the zero crossing. We really don't have to know where EVERY sample is, but we should at least know where SOME of them are, no? If we knew the specific location of every 64th sample then we could actually edit with the precision of 44.1kPCM, as 44.1kPCM is 1/64th the sampling frequency of DSD.

Now we don't necessarily have to convert EVERY sample to PCM, but we have to have enough of a reference to know where, exactly to place this bit of data once it's moved.

That's the simplest answer for you. If you want to talk about processing it's a different story, but for editing that's a pretty good reason.

Does this help?
Nika.


That's very helpful, thanks Nika \:\)

I can see why DSD is being aimed at the classical and jazz genres now. So, next question: if conversion to PCM is needed for processing, what kind of degredation (if any) will the audio suffer?

I'd be more than happy to use a multitrack DSD system with basic editing and use outbaord stuff for processing prior to mixdown onto a DSD 2 track. I guess this would be seen as a step backwards for many though (especially in PT based studios) Hell, artists might even have to try and get all the way through a song, use auto tune sparingly and play in time ... which kinda seems like a step forward to me ;\)


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Everybody,

Yeah, Nika is way off (acting like a moron), especially thinking that a filter is even applied in the A/D conversion of the DSD process. Sure filtering is applied at A/D in the PCM world but not in DSD. Once again people this is why they even call it DSD = Direct Stream Digital because processing normally implemented at the front end of digital in PCM is not applied until the back end D/A in DSD. And again check the Sony documents listed in my earlier posts. I'll notice when everyone wakes up to this because the dialogue will really turn around. "The Crafty One".

P.S.: Sorry about the short lines in previous posts. Maybe this double spacing will help. My AOL browser has been really acting up.

"The Crafty One".

[ 01-30-2002: Message edited by: crafty1 ]

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Quote:
Originally posted by Rog:

[...] So, next question: if conversion to PCM is needed for processing, what kind of degredation (if any) will the audio suffer?


Depends on the rate, and the resolution (and what doesn't in digital audio?) At the reasonably high rates of DSD-Wide (a lŕ Peter Eastty) for instance, it would seem that the biggest element of degradation is noise build-up (from successive stages of dither and noise shaping).


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Quote:
Originally posted by Nika:

[...] DSD doesn't have any location reference - it only tells you whether the next sample is higher or lower than the previous one.


In my opinion Nika is very, very far from being a "moron" as has been suggested elsewhere in this thread, but I'd like to point out that DSD does have a timelocation reference (the "other white meat", or axis).


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Hi GM,
what's the axis for the time location then? Thanks :-)

Crafty,
...where is Nika stating that there's a filter applied at the ADC in DSD? I believe he's mentioned nothing saying that it's filtered at the ADC, only filtered at the DAC using a simple low-pass filter and not an anti-aliasing one. Correct me if i'm wrong.

Cheers,
Shiver


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Crafty!

Quote:

Yeah, Nika is way off(acting like a moron),
especially thinking that a filter is even applied
in the A/D conversion of the DSD process.


Nika is really knowledgable, and in this instance, Nika is absolutely right. The Sigma in "delta-sigma" modulation IS A FILTER--a whole bunch of 'em, reall. Without them you'd simply have delta modulation. The integration block includes 5th order noise-shaping filters to get the inherent noise of sigma-delta out of the audio band.

For a high-level (marketing) explanation from one of the lead co-developers of this technology, Philips, check this out [pp. 7]. If you want more detail, check out the references on the back of said piece.

You are right in that downstream of sigma-delta ADC, there are no additional filters (or anything else except perhaps line-drivers) in a pure DSD path.

Almost all ADCs now have sigma-delta front ends. So the prevalent difference between DSD paths and PCM paths is that PCM output devices have decimators.

This is one of the more annoying things about how DSD is being sold. If you believe that simpler is better (as I do), it certainly has a simpler path than PCM devices [no decimators]. But somehow, the SP DSD alliance has managed to convince lots of folks that there is no filtering in the device--and that's just not so. If there weren't any filters, it wouldn't work well.

Does DSD sound better (really the topic of this thread)? I dunno. Haven't heard it.

-Dennis

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crafty1 wrote:

Quote:
P.S.: Sorry about the short lines in previous posts. Maybe this double spacing will help. My AOL browser has been really acting up.


We feel for you. Getting complicated, new-fangled, high-end technology to work correctly can be quite a challenge. You'll just have to endeavor to be even craftier to outsmart the wicked AOL beast. ;\)

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Quote:
Originally posted by crafty1:
If you don't you'll have what are
called aliasing artifacts. These
aliasing artifacts come in the form of
smeared stereo imaging, & compromised
wave impulse responses.


"This leads to an improvement in stereo imaging at the expense of a slight increase in aliasing."

the above is quoted from the dCS web page you linked us to. so which is it? the dCS source or your own opinion that is correct?

robb.

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Quote:
Originally posted by retreading:
This is one of the more annoying things about how DSD is being sold. If you believe that simpler is better (as I do), it certainly has a simpler path than PCM devices [no decimators]. But somehow, the SP DSD alliance has managed to convince lots of folks that there is no filtering in the device--and that's just not so. If there weren't any filters, it wouldn't work well.


Dennis,

Don't forget that the D/A's of a DSD system still have filters in them.

Nika.

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Nika,
That's the second such omission in 24 hours. I almost never think in that direction (D -> A). Shouldn't be so.
-Dennis

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Quote:
quote:
------------------------------------------------------------------------
Originally posted by Nika:

[...] DSD doesn't have any location reference - it only tells you whether the next sample is higher or lower than the previous one.
------------------------------------------------------------------------

In my opinion Nika is very, very far from being a "moron" as has been suggested elsewhere in this thread, but I'd like to point out that DSD does have a timelocation reference (the "other white meat", or axis).
--------------------
George Massenburg


Correct, which is why simple editing is possible in DSD (right?).

It's the EQ, FX, gain changes, etc. type of editing which become difficult in DSD (and therefore demanding conversion to PCM). Why is that? What's the missing reference point?

Eric \:\) The Original White Meat™


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Folks,

Let me start again then. DSD=Direct Stream Digital in theory has no filtering in the A/D section. The complete A/D-D/A process in DSD goes like this. Input to Delta Sigma Modulator(Noise Shaping) to (storage format = SACD) to Analog Low Pass Filter to Output. This is basically it in theory. No filter at the outset. I've used anti-aliasing as a "general" term in description of the D/A process in my earlier post because that is what is normally applied during normal PCM processing at the A/D but in essence is taken care of in the D/A process(as well as traditional filtering by one filter process) in DSD. This is the reason why Nika got so bent out of shape, & thought I did'nt know what I was talking about. I further tried to explain myself in later posts by pointing out that Nika was way off in critcizing me because Nika was judging my knowledge of this subject based upon Nika's knowledge of PCM A/D- D/A conversion, & not the "true" knowledge, & theory of DSD A/D-D/A conversion which was, & is the knowledge base from which I've posted from throughout this thread.

I admit I could have filled in alot more info. but I also like to keep things simple ,& sweet, & took for granted that Nika, & Curve D. knew where I was picking up from relating to DSD based upon their previous posts in this thread. I apparently was wrong, & should have started out with the elementary fundamentals of DSD, & built things up there. You'll also see that what I originally wrote on pertaining to filters, their cutoff frequencies, & slopes was dead on whether you're talking about DSD, or PCM excluding what I've just explained regarding the differences of filter placement between DSD, & PCM. Nika seems to post alot of info. that Nika is unknowledgable about. I noticed this in a post with Roger Nichols from last year:

http://www.musicplayer.com/ubb/ultimatebb.php?ubb=get_topic&f=2&t=001473

Oh well, some people can be duped by large words, technical terms, & long spiels but apparently not all including (gm), who quickly pointed out one of Nika's posts containing misinformation, & certainly not me. Once again I encourage you all to read all the Sony documents including the lit. on the Sony SCD-1, & 777 unit to grasp a good basic knowledge of DSD, & SACD accessed through the following link:

http://www.sel.sony.com/SEL/consumer/sacd/static/download.html

"The Crafty One".

[ 01-30-2002: Message edited by: crafty1 ]
[ 01-30-2002: Incredibly sloppy formatting corrected by GM]

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Crafty,
have you been reading the same document for a month? I'm not saying you know nothing about this, but if you're highly educated in this area then give us some more information, links, and an indication of how you've come about this. We're all here to learn, but you've got to come up with more information than one link over and over and over again. This is not the most detailed explanation of DSD in existence.. you've said yourself you have alot of knowledge in this area....Well, where's the info? Maybe you can skip the "general terms" (we can understand big words too you know) and don't worry about keeping it "sweet and simple"..it's a deep technical area..you're allowed to go there.
Shiver


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A completely subjective observation -

A partner and I had a demo of DSD and SACD done by Gus Skinas and Dr. Andrew Demery of Sony prior to the opening of AES.

For 30 minutes we listened to a variety of SACDs and the format sounds incredibly good to me. Not a lick of fatigue, a purely enjoyable listening experience that I've never experienced on digital before. The imaging and depth was unusually sharp and natural.

The DSD format will mosy likely be best used with high-end analog gear for a few years to come. The technology does not yet exist where real time processing in a DAW is possible. This will change as time passes.

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Tinder,

Quick question: I'm sure that the system you heard was most impressive - I've heard a similar system a couple of times before.

The converters were assuredly stellar - amps of top quality, speakers the same, etc. I'll bet that the room itself was even acceptable, though possibly not acoustically accurate.

What is the closest you have come to this layout of equipment and this experience with PCM?

Nika.

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Quote:
posted by crafty1:
Nika seems to post alot of info. that Nika
is unknowledgable about. I noticed this in
a post with Roger Nichols from last year:
http://www.musicplayer...
Oh well some people can be duped
by large words, technical terms, & long spiels
but apparently not all including (gm), who
quickly pointed out one of Nika's posts containing
misinformation, & certainly not me. Once
again I encourage you all to read all the
Sony documents including the lit.


crafty1,

As far as I could see, gm was simply putting a finer point on an issue Nika had posted on, not accusing him of spreading misinformation, or even saying he was wrong, as you claim. Same as on the thread you posted from Roger's forum.

Furthermore, these guys do not rely on sales brochures published by Sony to tell them how this technology works. Some of the people who post on this thread and this forum build this stuff. If Nika is wrong about something, THEY will tell us, NOT you. You, as you have admitted yourself, cannot even operate your AOL browser.

I beg of you to please leave us adults to our conversation. I, personally, am trying to learn something here, as are many others who are no doubt lurking on this thread to absorb professional wisdom. Your posts, although initially amusing, are now just static, disrupting the topic discussion, and wasting bandwidth. We all appreciate your willingness to contribute, but there is simply too much dialectric absorption in your attitude for you to make a meaningful contribution.

Please do not feel a need to reply to this post.
Eric \:\)


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Shiver,

I kept repetivly posting the same links because I consistently saw people on this post feeding off of the wrong information (Nika's). And some of the info. failed to acknowledge the rudiments of DSD theory. And my answer to your question is no relating to the subject of me reading the same source documents for a month. It's just been that I ran across these Sony documents about a month ago shortly after I inquired into DSD on another thread here. This is why I was stating in an earlier post to Nika, & Curve D. that they were at least a month behind me because I'd grasped the basics of DSD at least a month ago.

Actually I thought that Nika, & Curve D. had read these exact documents therefore thinking that they had a basic knowledge of DSD but apparently no. Once again this is my reason for urging everyone to get the foundation of DSD down then venture on from there. You should'nt quote rhetoric about PCM that does'nt apply to DSD, & especially when everyone else on the post is looking for new info on DSD. That's the mistake that Nika has been making, while speaking as if an authority on the subject but leaving gaping holes in what was supposed to be solid DSD theory but was'nt. I just was playing clean up man. Additionally it's not so much the case that Nika stated that there is a filter in the A/D of DSD but that so much of the theory that Nika has regarding DSD seems to be based upon PCM theory, & I was trying to point out to Nika that my post info. relating to filters in DSD were in regards to D/A in DSD. Nika went into along spiel about anti-aliasing filters in the A/D process, which does take place in PCM, as if that same PCM filtering process also took place during A/D in the DSD world. This obviously does'nt exist (filtering in the A/D realm of DSD) so therefore my statements reinforcing this.

"The Crafty One".

[ 01-30-2002: Message edited by: crafty1 ]
[ 01-31-2002: Egregiously inept formatting correct by GM]

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I'd like to repost this if yous don't mind, lest it got lost in the static...

Quote:
I'd like to point out that DSD does have a timelocation reference (the "other white meat", or axis).
--------------------
George Massenburg


Correct, which is why simple editing is possible in DSD (???).

It's the EQ, FX, gain changes, etc. type of editing which become difficult in DSD. Why is that? What's the missing reference point?

Eric \:\)


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Curve D.,

You're just upset that I've pointed

out several areas where you, & Nika

(partners in misinformation) have been

grossly in error. And while you're

talking about being an adult why don't

you be one(an adult), and admit your

wrongs on this thread starting with

your off color comment to me about

not hitting past the pitchers mound,

or something like that. Actually

I think I've hit several home runs

by setting things straight, & helping

to get people back on the right

information track. But don't worry

I forgive you, & Nika whether you

fess up to your ignorance or not.

Oh, & from what I've seen of the

inaccuracies you people(yourself, &

Nika)have been quoting you need

to look to somebodys literature

or source white papers because

you both got lost on a tangent

far far away from the land

of "real world" DSD.


"The Crafty One".

[ 01-30-2002: Message edited by: crafty1 ]

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SCENARIO 1 - Imagine this: a new digital professional recording format is introduced and is touted as a way of replacing or augmenting the current pro format. Little is known about this new format outside of the group of early designers of the first converters and some early adopters at studios or labels. The biggest problems with this new format are that editing may be very difficult or impossible and that mixing within this format without conversion to the 'old' format is absolutely impossible. No company or individual is making any means of mixing in the new format. There are no outboard tools for this format in its native domain. Almost no one knows how the new format might be mastered or authored for consumer release.

SCENARIO 2 - Now imagine this: suppose while this new recording format is just getting started - and before the professional recording community at-large gets to even start using the new format - a number of vocal opponents ( many of whom have NEVER actually worked in the new format ) manage to kill off further development of the new format. No editing system gets further development because there are not enough end-users interested. No mixing desk for the format is made. No recording or mixing tools get introduced because many potential users couldn't see any reason for even considering what benefits the new format may offer if given the chance to fully mature.

If scenario 2 had actually happened we would have:
NO Pro Tools
NO Oxfords
NO digital mixers of any kind
NO Sonic Solutions
NO SADiE
NO Nuendo
NO whatever your favorite digi workstation is
etc, etc, etc...

Scenario 1 actually DID happen. In the early 1970s, PCM was viewed with suspicion that it would be too difficult to work with; that it would require unbelievably powerful and expensive computers to edit; that digital mixing would be difficult and very expensive to implement; that simple tools like compressors or limiters would be difficult to make work in the digital format that could EVER compete with analog equivalents; pcm EQ was beyond imagining; and there would no consumer format that could carry the recording. When a consumer format finally was introduced a few years later (CD), the first players were somewhat clumsy and quite expensive. The first converters didn't sound all that good - and they were EXPENSIVE and almost impossible for average studios to purchase. The best-sounding pcm system then, Dr. Tom Stockham's 50 kHz Soundstream system, couldn't be bought at ANY price. You had to rent it on a per-project basis, and a Soundstream specialist or two had to come along with the system to keep it working. The format could only be edited at Soundstream's Salt Lake City facility on an editing system that would take 30 minutes to process 1 minute of audio. Digital mixing in the format eventually became possible, but the mix parameters had to be written in computer language and you had to come back the next day to hear what the results were. All post-production work was limited to 2-track stereo.

The early adopters of pcm were crazy individual engineers and mostly small classical and jazz labels.

SOUND FAMILIAR???

If you are not familiar with DSD recording tools you owe it to yourself as an audio professional to find a way to actually work with them in some regard. Get out of your usual "comfort zone" and try something new! Forget the company literature and specs. The stuff IS available for use, either on-loan for projects or for rent through the pro rental companies. Most manufacturers will give extended demo sessions if asked. Hell - if any of you are in the Cleveland area (and I have the time and am not out on-session) I'd be glad to schedule a side-by-side comparison at our facilities. We can do very carefully set-up A/B testing, but we are not a lab. We don't have A/B/X testing available. We DO NOT sell DSD equipment. We sell way more pcm CDs than DSD SACDs.

I spent the first half of my career working exclusively in analog. The second half has been spent working with pcm. During the last four years I have moved to working almost exclusively with DSD, even though it has been exasperating and quite difficult at times. Personally, I see something there worth pursuing. A handful of like-minded engineers and producers are doing the same for the same reasons. It sure isn't because they're making any money by working and releasing in this (currently) somewhat cumbersome format. There is development in DSD DSP taking place where any pcm intersteps may be eliminated. The tools are being developed slowly, but no where near as slowly as the pcm tools were! The DSD R&D work is quite expensive and the funding for it is constantly being threatened. The companies providing the funding are unsure just what interest there is in the pro market. Only a small handful of dedicated individuals are doing the R&D on DSD tools at present - many of them are the very same people who developed the pcm tools we take for granted today.

now I'll climb down off my soapbox...

[ 01-30-2002: Message edited by: mbishopsfx@aol.com ]


Best Regards;
Michael Bishop
Telarc International
http://www.telarc.com
SACD, DSD & DVD-A Editing and Mastering available now at:
Wired 4 Music
http://www.wired4music.com
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gm,

Thanks for the clean up of my previous post,

and adding some sobriety to a disillusioned

poster(Nika).

"The Crafty One".

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Quote:
SCENARIO 1 - Imagine this: a new digital professional recording format is introduced and is touted as a way of replacing or augmenting the current pro format. Little is known about this new format outside of the group of early designers of the first converters and some early adopters at studios or labels. The biggest problems with this new format are that editing may be very difficult or impossible and that mixing within this format without conversion to the 'old' format is absolutely impossible. No company or individual is making any means of mixing in the new format. There are no outboard tools for this format in its native domain. Almost no one knows how the new format might be mastered or authored for consumer release.

SCENARIO 2 - Now imagine this: suppose while this new recording format is just getting started - and before the professional recording community at-large gets to even start using the new format - a number of vocal opponents ( many of whom have NEVER actually worked in the new format ) manage to kill off further development of the new format. No editing system gets further development because there are not enough end-users interested. No mixing desk for the format is made. No recording or mixing tools get introduced because many potential users couldn't see any reason for even considering what benefits the new format may offer if given the chance to fully mature.

If scenario 2 had actually happened we would have:
NO Pro Tools
NO Oxfords
NO digital mixers of any kind
NO Sonic Solutions
NO SADiE
NO Nuendo
NO whatever your favorite digi workstation is
etc, etc, etc...

Scenario 1 actually DID happen. In the early 1970s, PCM was viewed with suspicion that it would be too difficult to work with; that it would require unbelievably powerful and expensive computers to edit; that digital mixing would be difficult and very expensive to implement; that simple tools like compressors or limiters would be difficult to make work in the digital format that could EVER compete with analog equivalents; pcm EQ was beyond imagining; and there would no consumer format that could carry the recording. When a consumer format finally was introduced a few years later (CD), the first players were somewhat clumsy and quite expensive. The first converters didn't sound all that good - and they were EXPENSIVE and almost impossible for average studios to purchase. The best-sounding pcm system then, Dr. Tom Stockham's 50 kHz Soundstream system, couldn't be bought at ANY price. You had to rent it on a per-project basis, and a Soundstream specialist or two had to come along with the system to keep it working. The format could only be edited at Soundstream's Salt Lake City facility on an editing system that would take 30 minutes to process 1 minute of audio. Digital mixing in the format eventually became possible, but the mix parameters had to be written in computer language and you had to come back the next day to hear what the results were. All post-production work was limited to 2-track stereo.
The early adopters of pcm were crazy individual engineers and mostly small classical and jazz labels.

SOUND FAMILIAR???

If you are not familiar with DSD recording tools you owe it to yourself as an audio professional to find a way to actually work with them in some regard. Get out of your usual "comfort zone" and try something new! Forget the company literature and specs. The stuff IS available for use, either on-loan for projects or for rent through the pro rental companies. Most manufacturers will give extended demo sessions if asked. Hell - if any of you are in the Cleveland area (and I have the time and am not out on-session) I'd be glad to schedule a side-by-side comparison at our facilities. We can do very carefully set-up A/B testing, but we are not a lab. We don't have A/B/X testing available. We DO NOT sell DSD equipment. We sell way more pcm CDs than DSD SACDs.

I spent the first half of my career working exclusively in analog. The second half has been spent working with pcm. During the last four years I have moved to working almost exclusively with DSD, even though it has been exasperating and quite difficult at times. Personally, I see something there worth pursuing. A handful of like-minded engineers and producers are doing the same for the same reasons. It sure isn't because they're making any money by working and releasing in this (currently) somewhat cumbersome format. There is development in DSD DSP taking place where any pcm intersteps may be eliminated. The tools are being developed slowly, but no where near as slowly as the pcm tools were! The DSD R&D work is quite expensive and the funding for it is constantly being threatened. The companies providing the funding are unsure just what interest there is in the pro market. Only a small handful of dedicated individuals are doing the R&D on DSD tools at present - many of them are the very same people who developed the pcm tools we take for granted today.

now I'll climb down off my soapbox...

[ 01-30-2002: Message edited by: mbishopsfx@aol.com ]
--------------------
Best Regards;
Michael Bishop
Telarc International


Michael,

That was quite a long post, and an extensive endorsement of DSD. Yet, nowhere in that post, did you once mention any bit of empirical observation as to WHY I should cut Sony a very large check out of my already stretched budget to invest in this technology.

You sell SACDs. That's great. I'm happy for you that business is good for Telarc.

But, we, who are producing the music that you sell, would like to understand the underlying principles of this technology, before we reach into our pockets; at least I THOUGHT that was what this discussion was about. SInce you seem to have access to this technology, it would seem you might have something to offer this discussion in that regard, besides a sales pitch. Not taking anything away from your sales pitch: it was a good pitch, except for one small thing...

As far as "getting out of my usual comfort zone and trying something new" is concerned: I can assure you that living the life of a struggling composer/producer of new original music is anything BUT a "comfort zone." What makes it even more uncomfortable at times is when resource-draining technologies are thrust upon us. And so, we will make efforts to determine if those technologies are a contribution to our craft, or if they are simply redundant upgrades filling the coffers of the likes of Sony and Phillips Electronics at our expense.

Therefore, criticism inevitably WILL be leveled at emerging technologies such as DSD. It would be highly naive and disingeuous to expect otherwise. If said technology cannot weather that criticism, it will fail, and for good reasons. You cannot blame this constructive debate by our small group of engineer-types in this obscure chatroom for the failure of a technology that will contribute to mankind. THAT is where you blew your pitch.

If this discussion "kills off" DSD, it deserves to be killed. If you would like to credibly take issue with that, I would suggest you make a contribution to our debate that goes a little farther than, "Try it! You'll like it!" I'm sure I'll like it. I like french fries, too. But I know better than to attempt to subsist on a diet of french fries.

Thanks!
Eric \:\)

[ 01-30-2002: Message edited by: Curve Dominant ]


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mbishopsfx@aol.com ,

Don't mind Curve D.. Curve D. is

a "hand biter": def. = one that bites the

hand that feeds him/her. I found this

out early on.

Hah!

"The Crafty One".

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Hi Mr. Bishop,
Just so we're clear from the getgo here, i'm not in a position to be buying one of these systems, nor do i have near the technical knowledge that people like Nika, Curve, GM, Roger etc have. I'm simply here learning about the pros and cons of this new format.

I think what alot of this boils down to is this is not an effort to kill off DSD by any stretch, but more trying to gain some REAL evidence, other than "it sounds good" as to why this should be regarded as a format that's not going to go the way Beta did. yes, Beta performed better, but it didn't last. Why? I have no clue. ;-) But unless there's some sort of tangible evidence that it's not marketing hype like a huge chunk of the audio industry already is, it's not likely to go without scrutiny. I look at 96k and the higher sample rates and i'm still very confused as to why theyr'e there in the first place when it's been fairly clearly stated and proven in some circles that 44.1 should be completely sufficient and that what we need to work on is on the converter end.
anyways, that's my .02 cents regarding this issue for now. got a client so i gotta scoot.
cheers everyone and thanks for all the info.
Shiver


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Quote:
Originally posted by crafty1:
Curve D.,


Everyone else,

Ooops! I forgot my footnotes, &
bibliography sections. Paaaardon me!
(Joking:Hah ha!). Oh, & you can also
bet that Curve D. is going to be
in servere denial about the above
info. provided directy from Sony.
Curve D. might even try to
convince that DSD does'nt even
exist, and that it's PCM repackaged
with a new nameplate kind of like
a Cadillac/Escalade is just a
GMC/Denali with a Cadillac hood
ornament!

"The Crafty One".

[ 01-30-2002: Message edited by: crafty1 ]



Oh Boy! This para has just got to be bait LOL!

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LMAO!!!

E \:\)


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It comes down to what is better voltage or current. Chicken or egg. Turns out both are rather important, and you cannot have one without the other. PCM has the historical advantage of financing. SACD has the advantage of simplicity. Put disc in - disc plays...caveman style. Who the fuck knows what will become of DVD-A, submenus, mixdowns, LFE, etc. 90% of audio salespeople don't even realize they sell 2 types of DVD players, let alone the consumer. It shouldn't be difficult to listen to music. This should be the rationale behind professionals' support of the medium. Admittedly, I don't know much on the copy protection schemes on each format - however CD's held up pretty well, now DVD-R's are already shipping, practically before the media. That can't be good. Philosophy aside.

At AES there was a bit of talk on advantages concerning perceptible noise floor and distortion. I think the engineer was from TC, not sure.

The advantageous nearly balance out the recent barrage of disadvantages levied on this forum.

When, early on, I spoke of the "NEXT" format. I was speaking of what lies beyond PCM and DSD.

Regrettably all the critical listening I have done was at the convenience of someone who had something to sell me. My convertors just aren't up to par yet.

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I did not write that the discussion on this chat list would "kill DSD". What I was referring to are the public negative opinions and statements made by well-known audio professionals on how a DSD system works, or does not work, or how it sounds, only to find out they haven't even used or tested such a system! The discussion here is all fine and good - discuss away - but I haven't read anything here that would kill a format off. I'm not sure why any audio "professional" would dismiss a different use of digital technology out-of-hand without actually trying it or listening to a demonstration! All some people see is an evil Sony/Philips 'monster' wanting all your hard-earned money - all the while listening to CDs (Sony/Philips) on a Walkman (Sony) CD player, watching the Super Bowl on your (Philips) TV, and tapping out responses to emails on your (Sony) PC. You may as well substitute the name of any other electronics giant in the above examples. Let's face it, it does take an electronics "giant" to effectively develop and promote a different way of recording and releasing digital media in an improved way in today's tough economy. Even the giants are having a rough go of it right now. Unless you really ARE satisfied with how CDs sound, or worse: MP3 performance, I would think any audio pro would want the possibility of something better. I have never been happy with CD's performance - I know from my own experience that much more quality is possible than what CD's compromised 16-bit wordlength and 44.1 kHz sampling rate can do. I don't want to "settle" for that quality in MY recordings if at all possible.

As far as adding my own meaningful empirical experiences to this discussion, I have done so on many other occasions elsewhere on this site. I have been interviewed numerous times on DSD in domestic and foriegn trade and hi-fi mags. I wrote an article for The Absolute Sound on the subject. I feel little need to repeat myself here. Poke around - you'll find record of my comments around.

I certainly would not encourage any person reading this thread or otherwise go out and buy a DSD system of any kind at this point. It's still expensive and there are hardly any tools to work in it on. There's a lot of development to go yet. DSD technology is NOT for the struggling composer/producer/engineer. I have stated that here in other threads. Only the brave - or foolish (!) - jump into a new technology that does not have a real market for it yet. I did not say we sell alot of SACDs - we certainly don't. I wrote that we sell way more pcm CDs than SACDs. That will be the case for at least three more years. The label I work for (Telarc) is working toward SACD - or DVD-A - or other SURROUND format to hopefully be accepted in place of stereo CDs in the home. Surround music playback in the home is what we are REALLY pushing for - not DSD. DSD just happens to be OUR preferred method of recording in surround and stereo. SACD just happens to be OUR preferred means of releasing those recordings because, by our own experience and extensive listening tests, we feel the format represents what we recorded in session best. Believe me, it would be ALOT EASIER to be working in 24/96 pcm instead of DSD. We have all the equipment we would need to record, mix, and edit in 24/96 pcm and it sure is alot easier to work with than DSD. We record, edit and mix in DSD and release our "premium" product on SACD. What does that tell YOU? Maybe it tells you we're crazy. We may be, but draw your own conclusions...

What I request of any of you is to take a clear, unbiased look at ANY emerging technology and consider what it might become if developed fully. It is up to you to get past the company literature and claims, and past the opinions of those who are too close-minded to consider other possibilities. It's up to you to get your hands on new equipment and try it. Determine for yourself what it can and cannot do. Try to determine if it will help sell one more copy of your recording. Try a little crystal ball gazing and see if the technology has any future to it. It helps to carefully look at the track record of those currently working with the new technology. See what they may have done with past new technologies.

When it comes down to it, any of this equipment is just a set of tools. Any good tools will get the job done in good form. The tools come in quite a wide variety. Use the tools that enable you to get your product out the way YOU want it. The hardest part is coming up with music people want to buy. At that point, the tools don't mean a thing.

With Best Regards,

Michael Bishop
Telarc International


Best Regards;
Michael Bishop
Telarc International
http://www.telarc.com
SACD, DSD & DVD-A Editing and Mastering available now at:
Wired 4 Music
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Quote:
Originally posted by mbishopsfx@aol.com:


[...]

If you are not familiar with DSD recording tools you owe it to yourself as an audio professional to find a way to actually work with them in some regard. Get out of your usual "comfort zone" and try something new!

[...]

I spent the first half of my career working exclusively in analog. The second half has been spent working with pcm. During the last four years I have moved to working almost exclusively with DSD, even though it has been exasperating and quite difficult at times. Personally, I see something there worth pursuing. A handful of like-minded engineers and producers are doing the same for the same reasons.


Thanks Michael. Couldn't agree more.

I, too, spent my first years on analog -- somewhere in the range of 20, I think. Over time, I built up a HUGE sensitivity to the problems -- noise, distortion, inflexibility. Not to mention the huge expense of this klunky technology.

Starting in 1982 I started doing all of my work in PCM. It was not very good, but it was a fact that we who pioneered found so many things wrong with the technology - and were unabashedly vocal about the shortcomings- that improvements came, albeit at first somewhat slowly.

Somewhere around 1993 I had a conversation with Ayataka Nishio and Kyoshi Hasama (head of engineering for Sony Classical at that time) at an InterBEE show. Nishio (creator of SuperBitMapping) was thinking of a format that utilized/stored the single-bit-stream from the modulator of a 5390 A/D before the decimatorthereby avoiding the need for a brick-wall filter (relying simply on a single-pole rolloff). The point I remember best from the conversation was our saying, "Someone's got to invent the future." They went off, working pretty much as a "black" project. Until, that is, such time that Sony, desparately seeking a new technology to confront the DVD-A format as well as the other Japanese manufacturers who wished no more than to teach Sony a lesson (and smarting from 20 years of Sony/Philips collecting HUGE sums in license fees for the CD), looked at Nishio and said, "You're it".

I have mixed feelings about DSD.

On one hand the development has been hampered by the extraordinarily narrow vision of Sony managers. For instance, for a very long time managers at the time had this thing about "single-bit" technology, even though the 64FS single-bit a/d is inherently flawed, and wouldn't allow the process to mature. Engineers had to work pretty much in secret to improve it.

On the other hand, I've heard DSD at it's best, and it's pretty impressive...and, what's more, really capable of moving wellbeyond artifacts inherent in 44.1-48k PCM. I heard a playback at Seigan Ono's mastering studio one night of a terrific live recording that he had made at Clinton, NY...and I was transformed.

I would only add that the most obvious features of pioneers of any stripe are numerous arrows in the back.


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Michael,

I have nothing against progress, moving forward, etc. And though I don't have much sympathy for Sony, it certainly doesn't make sense to belligerantly avoid SACD for fear that they could end up dominating the world of audio, for if we stay stagnant then we stay with CD's - which are Sony just the same!

What my issue is with DSD is that noone to date has convinced me of what is wrong with the CURRENT technology. What in the world is inherently wrong with PCM such that we require something new? What is wrong with the sampling frequencies that we have?

Now I know that the typical response is "hey, Nika, if you can't hear the difference the more power to ya!", but it's not that simple. Conceptually PCM fits our needs. Conceptually the filters can be made to be gentle enough, the sampling rate is high enough, phase and amplitude is captured, we can't hear above 20k anyway. Conceptually everything about 44.1kS/s PCM is fine. So where is the problem?

I have read a lot of what you've had to say about DSD, including a pretty comprehensive description of your listening test in Belgium (?) in 1996 where you set up a DSD system, 96kS/s PCM system and a 192kS/s PCM system simoultaneously and the consensus was that DSD was the way to go. But is it not possible that you were actually comparing the particular IMPLEMENTATION of a certain type of converters? I mean, didn't you have to choose a particular converter to do your test with? And doesn't that converter sound different than every other converter out there? So isn't it possible that another change in converters could have indeed actually changed your response away from the DSD and more towards one of the other formats? Isn't the particular box chosen for the testing itself such a critical component of the results of the test? It's something akin to the Heisenberg uncertainty principle - that we can't actually just compare formats without chosing a particular box to compare with, and the unique characteristics of that particular box will in some capacity dictate the results.

So since we can't merely resort to testing in order to determine the difference between formats for this reason, we have to address it from a scientific and mathematical point of view. We actually have to look at it and figure out the REASONS that one format could be better than another. Then we can COMBINE this information with some valid listening tests and yield a valid conclusion, but neither the science on it's own, nor listening on it's own can really be valid tests of the two FORMATS. They can only be valid tests of two different boxes' DERIVATION of those formats.

To me, this is what Eric was addressing - some sort of scientific or mathematical explanation of the difference so that he can give more credit to a listening test. He has still not succeeded in getting any valid information in this regard. The only contribution has been provided by what appears to be a fourth grader, and in correcting his statements I believe that we are perceived to be presenting a negative cast on DSD, as though I/we have never heard it before and are dismissing it out of hand.

This is not the case. I have heard DSD. I have not "worked" with DSD, but I have heard it. And in every situation I've heard it in it sounded good. It better have. Sony sure made certain that it did by providing a $5,000 player, a $10,000 amplifier, a $17,000 pair of speakers and a finely tuned room to listen to it in. Yeah, it sounded good, but there was no possible way to compare it to anything to see if it sounded BETTER. There wasn't even a normal CD around to plug in to the same system to compare. This demonstrates an extreme example of the fact that listening test on SACD/DSD are for the most part invalid, as the listening tests themselves are so far from A/B/X that I have to immediately dismiss any results I came to. Imagine if all listening tests were A......and then three days later, in a different room, with different speakers, with different amps, with a different couch, with different ambiance we play B. How can I really get anything valid out of that? This isn't to mention that even if we COULD compare the two in the same room, once again, which converters were chosen for the test and why?

So I'm afraid that I won't be jumping on this bandwagon until one of the two following things happens:

1. The scientific or mathematical community explains to us why higher frequency sampling is beneficial (DSD really is just another derivation of higher frequency sampling)

and

2. I can hear the difference in a controlled enough environment, and the difference reflects the reasons given in #1.

I appreciate your invitation to stop by next time I'm in Cleveland. I may make an appointment with you to do that.

But if neither of the above happens then I would just as soon we continue to perfect a technology which SHOULD be sufficient, and which is well under development. Part of my rationale is that I feel it unnecessary to use so much more storage space if it is unnecessary in the quest for better audio. If we can make the existing paradigm work to what science tells us it's capability should be then it seems so much more efficient that I'm prone to support that.

Having said all of this, I do have a fairly open mind. The only reason for my skepticism now is that my mind was open enough to not simply buy into the marketing and feel good spiels of Ed Meitner about biorhythyms. It would have been much easier to merely buy into the notion of higher frequencies and removing filters and all that rather than having an open enough mind to do some research into the validity of the cause.

If you have something to feed into my open mind (including my potential visit to Cleveland) please, do share.

Thank you for putting up with this dissertation. Your opinions are certainly respected, though cautiously not taken for granted.

Sincerely,
Nika.

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Quote:
posted by Michael Bishop:
What I request of any of you is to take a clear, unbiased look at ANY emerging technology and consider what it might become if developed fully. It is up to you to get past the company literature and claims, and past the opinions of those who are too close-minded to consider other possibilities. It's up to you to get your hands on new equipment and try it. Determine for yourself what it can and cannot do. Try to determine if it will help sell one more copy of your recording. Try a little crystal ball gazing and see if the technology has any future to it. It helps to carefully look at the track record of those currently working with the new technology. See what they may have done with past new technologies.


Michael:

Yes, but I would add one thing: try to get under the hood of the product that leaves the assembly line, and see if it's all that the manufacturer cracks it up to be. Put down your crystal ball for a moment, and pick up a magnifying glass, unroll some blueprints, and show me the math.

If you read carefully through this thread, you'll notice that a lot of the skepticism comes from the obvious gaps between what Sony says DSD is, and what we will in reality be getting. It looks great, in theory. I looked at the diagrams of the Delta Sigma Modulator - very cool, a quite elegant design which, I believe, dates back to 1946, and the name delta-sigma modulation was coined by Inose and Yasuda at the University of Tokyo in 1962. But gm brought up an interesting point:
Quote:
the development has been hampered by the extraordinarily narrow vision of Sony managers. For instance, for a very long time managers at the time had this thing about "single-bit" technology, even though the 64FS single-bit a/d is inherently flawed, and wouldn't allow the process to mature. Engineers had to work pretty much in secret to improve it.


Which leads me to wonder: what exactly is it that Sony has ultimately rolled out? Is it "the future," or will it be, as crafty1 so eloquently postulated,
"PCM repackaged
with a new nameplate kind of like
a Cadillac/Escalade is just a
GMC/Denali with a Cadillac hood
ornament!"


I'm sure that the folks Sony hired to help them stave off DVD-A (Ayataka Nishio, Ed Meitner, et al) are perfectly capable of inventing the future. But there are also some very huge $$$ for Sony if they can get everyone in the world to throw away their CDs and CD players in exchange for a new format. If Sony is in a hurry to beat DVD-A to market, what's stopping them from throwing Nishio's and Meitner's resumes in your face just to get you gung-ho over a format which is a step forward in theory, but which in reality, the box itself is nothing but a watered-down version of what potentially could be an advancement? A lame repackaging of PCM? The "New Coke," if you will? Wouldn't you want to know the truth behind the marketing? So they paid Ed some big $$$ to talk shit. So what? Look under the muthafuckin' hood, my brutha. What do you see?

Quote:
posted by Michael Bishop:
As far as adding my own meaningful empirical experiences to this discussion, I have done so on many other occasions elsewhere on this site. I have been interviewed numerous times on DSD in domestic and foriegn trade and hi-fi mags. I wrote an article for The Absolute Sound on the subject. I feel little need to repeat myself here. Poke around - you'll find record of my comments around.


Which makes me even more suspicious of DSD. I have been poking around, and have found nothing beyond the antecdotal. You have now posted two very lengthy posts in defence of DSD, yet offered not one word of hard evidence that it is clearly superior to PCM. You have a record label, and record sales are off. If we all switch to DSD, your sales go up. Convince me, Michael, that I am merely being cynical, PLEASE. I'm a composer, and I would LOVE to hear my music in higher fidelity than it sounds now. Feel a little need to repeat yourself, if you have something worth repeating to say.

Thanks!
Eric \:\)


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What my issue is with DSD is that noone to date has convinced me of what is wrong with the CURRENT technology. What in the world is inherently wrong with PCM such that we require something new? What is wrong with the sampling frequencies that we have?

---------------------------------------


Nika,

My thoughts exactly. Guys, suppose I have a reasonable understanding of how the technology works, but not at an electrical engineering level. Let's also assume SACD (Sony Audio CD) technology currently sounds better than anything else at any price.

Why can't PCM technology be made to sound better than DSD next year? Every high end converter uses a motorola or sharc DSP these days. Why can't a good engineer use the PCM architecture (flaws and warts included) and build a converter with a clever algorithm better than DSD?

To me it looks like this: Suppose all speakers currently uses dome tweeters for high-frequencies (PCM). Sony invents this new high frequency driver called a ribbon (DSD). It is very expensive and every audiophile engineer that has heard it says it awesome, a must have. Sony has the patent on it and tells the world why dome tweeters (PCM) are flawed. (limited hifrequency response and inferior transient response) And in some ways tries to smear and bad mouth companies that make dome tweeters.

My question would be why can't a dome tweeter be made to have the same or better frequency response and transient as the Sony patented "ribbon"?

For DSD VS PCM this question has not clearly been answered. It is not good enough to change the way we do things just because DSD is the best right now. What about in a few years? Can't the PCM camp catch up?

I have my own person views on this, but since I haven't had the chance to take DSD home and demo it, I guess my opinion is meaningless.


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I have to say that I am disappointed as well with this topic - the only DSD information posted was a couple of links to Sony Marketing brochures - pullleeeaaase! That is so bloody lame. I went thru a zillion Google search results also with no better luck.

For now it seems a moot point for me and only something to keep an eye on - there are not tools to produce the products I produce (even if I could afford them) that would work in that format that no one I know can listen to. Something to keep an eye on, but nothing to get all bent over.


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I haven't posted on this thread because I haven't heard DSD yet either, and I certainly haven't the knowledge to discuss either PCM or DSD on a scientific level.

I will say this, though: While the science of it is interesting to read, I don't need the science to hear a difference if any reasonable difference is there. As it stands, the DSD thing is too much $$$ for me and my clients. If it survives, though, it will get cheaper and eventually the less monied of the engineering community will be able to choose the format that pleases their ears. While careful testing/ comparison procedures are hard to make happen, they are not THAT hard to create. After doing my own tests on gear I'm considering (which DO involve me not knowing what I'm hearing until after I've written down my impressions of it), I usually feel pretty comfortable voting with my wallet. If the quality differences are so subtle that I can't get a consistent set of reactions from my own brain when I'm CONCENTRATING on hearing the differences, then I do the only sane thing:

I buy another pair of microphones. They ALWAYS sound different!

ML

PS: I really appreciate posts from folks who HAVE heard DSD. I also really like it when folks act like they would act in PERSON if we were all, you know, together. Like polite, or at least civil. Many other boards in the web are full of wankers and wanna-bees, with hardly anyone posting that is more than a senior wanker or 1st-class wannabee. Some of the posters on THIS board are very senior, knowledgable people. I appreciate their taking the time to type in characters that spell words that deliver their thoughts to us. Bitch-slapping guys like this around just encourages them to post less.

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Call me stupid but SACD players can be had for as little as Ł200 - and they play 44.1 discs too. Go and buy one - get one SACD disc and a CD of the same recording. Have a listen.

If that's not objective enough then maybe some kind soul would use a pyramix or similar to convert DSD to 44.1 and provide a CD version of a mastered DSD disc?

I think, given the state of the technology at this stage, this is the nearest most people will get to judging the quality of DSD. Empirical evidence is scarce or non-existant and endorsements from GM certainly help but I think I'll just go and buy a player and see what the fuss is about. If I like it I'll wait for flexible, affordable multitrack recorders to surface.


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I would love to hear DSD. Furthermore, I would love to understand it more fully. I'm currently studying something, ironically, that may be put aside by it (PCM DSP). But check this out:

It's fun to discuss and argue the ins and outs and accuse each other of not being able to hear anymore, but there's another side to it. The concept of DSD is a step in the direction of progress. Even if DSD flopped miserably, it wouldn't be the first format to do so (remember Beta video? And it was supposed to be better...).

Progress is all about trying new things. There should be no reason to question our current formats because the moment we become comfortable in what we have, we'll surely stop improving and reaching for what we could have.

DSD is one small example of progress. It may turn out that it is more useful in some industrial application. Who knows? That's still no reason to question it's existence. It may lead to something more advanced and more exciting. I would like to be a part of that. Wouldn't you?

Just a thought.
Please continue.

adam - "it's all progressive."


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Quote:
Originally posted by Tony Mah:

To me it looks like this: Suppose all speakers currently uses dome tweeters for high-frequencies (PCM). Sony invents this new high frequency driver called a ribbon (DSD). It is very expensive and every audiophile engineer that has heard it says it awesome, a must have. Sony has the patent on it and tells the world why dome tweeters (PCM) are flawed. (limited hifrequency response and inferior transient response) And in some ways tries to smear and bad mouth companies that make dome tweeters.

My question would be why can't a dome tweeter be made to have the same or better frequency response and transient as the Sony patented "ribbon"?

Tony W Mah


uh-oh... i know alphajerk's going to have something so say about THAT annalogy ;\)

Quote:
ML

PS: I really appreciate posts from folks who HAVE heard DSD. I also really like it when folks act like they would act in PERSON if we were all, you know, together. Like polite, or at least civil. Many other boards in the web are full of wankers and wanna-bees, with hardly anyone posting that is more than a senior wanker or 1st-class wannabee.


\:\) thanx for reminding some of us to stay out of this one ;\)

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Quote:
Originally posted by Mark Lemaire:

If [DSD] survives, though, it will get cheaper and eventually the less monied of the engineering community will be able to choose the format that pleases their ears.


Amen. ;\)

It is for this reason that I hope labels like Telarc continue to do good work with DSD, that it does "survive" and that it continues to evolve and improve in parallel with PCM. With all reasonable cynicism aside, it's nice to see one of the monoliths coming out with a delivery format that's excessively good (remember DCC?).

-dg


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Since three or four years I read a lot about DSD. The only disadvantage compared to PCM is maybe the higher noisefloor (in theorie) and certainly the difficulties to work with (mixing or EQing isn´t as easy as with PCM).

But what are the advantages?
What we are looking for is an absolutely transparent system. Until now I heard DSD three times. The demonstration were done by SONY with very very expensive equipment (amps, cables, loudspeakers). I really was impressed. And yes there was also something I would say "emotionally strong" (excuse my bad english, can´t find better words).

On the other hand I also heard 100.000$ amps, cables, speakers where the source was 24/96 PCM - and I also was impressed. But I never heard DSD and PCM in comparison.

I agree with Nika, I don´t now from any scientific comparison between DSD and PCM.

Apart from that I would like to read about facts. What are the system inherent problems with PCM. What is better with DSD - noise, all kinds of distortion, behaviour with transients?
Is there a way to get around the problems of PCM (decimation?) and DSD (again, noise, EQing, mixing?).

In the end, is a signal coming from a microphone or an analog mixer and then digitally recorded better representet with DSD or with PCM? And Why?

Are we able to describe facts here, or can we only talk about emotions?

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Guy, in all honesty...

I did not post this thread with the intention of starting a big fracus (sp?). Although it has been fun and interesting debate, I really was just trying to find out if anyone had any meaningful analysis of DSD vs PCM.

A friend of mine bought an SACD system for his home recently, and it got me curious, that's all. It sounds good, but once again, the whole system down to the cables is all-pro, very expensive stuff. He gets home from a business trip tomorrow, so I'm gonna hit him up for a visit next week, and see if we can arrange some sort of A/B/X test. Can't guarantee a high degree of scientific integrity, though: it's in the basement of a rowhome in South Philly. I'll see what I can do.

In the meantime, thanks to everybody who participated in this discussion, especially crafty1, who's comedic relief was much welcome since I stopped watching Seinfeld reruns some months ago.

Eric \:\)
----------------
"Dancing bits...what, am I at the f*cking ballet now?!"


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\:\) well curve, thanks for starting the damn thing in the first place! \:\) Rarely have i gotten so wrapped up in lurking around. I wish i had more info to contribute but i sure as hell learned alot. But i WAS very serious in my feeling bad about the fellow Nika mentioned in one of the threads that was linked to....Stanley Lipshitz...my my my. Brilliant he was/is (?) i'm sure, but definately unlucky. Like i said, i'm not too worried about offending anyone with the joke because my last name's Wang...didn't serve me too well in junior high... \:D
Cheers,
Shiver


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Shiver Wang? Sounds like a porn name...

E \:\)


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ROTF! Ha! Well, I DO star in alot of adult films...let's see:
"Tight Anal Rampage"(1 through 7)
"Knockin' on Heather's Door"
"The Only One Left For Debbie to Do"
"Late Night with David Leatherman"
Those are my most recent. \:D
Cheers,
Shiver Wang
(Porn Star Extraordinaire)

[ 01-31-2002: Message edited by: Shiver ]


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Quote:
Originally posted by Curve Dominant:

EVEN IF DSD was some kind of superior high-resolution format, the whole logic behind using it as a consumer-end delivery system shoots itself in the foot...

1. Since one cannot edit DSD, it is rendered impractical and unusable for recording sessions.

2. Therefore, sessions will continue to be recorded in PCM.

3. So, all of those "evil" sounding artifacts allegedly produced by PCM will be faithfully rendered in high-resolution by DSD.

Correct me if I'm wrong, but if the signal hits PCM at any point in the path, the end listener is essentially hearing a PCM signal. Which would mean:

A. For a consumer to buy a DSD system, and DSD discs, would be a complete waste of money

B. For an engineer to master to DSD would be a complete waste of time and money

C. This whole concept of DSD as a practical option for the consumer is a complete hoax.

Where is the fundamental flaw in that equation?

Eric \:\) [/QB]


Well for one, you CAN edit DSD. We've been doing it for 3 years - first on a prototype Sonic USP system, then on the Sony Sonoma. Sadie is set to release a great stereo DSD workstation and Sonic Solutions has just announced that they will develop DSD editing for the HD system.

You're assuming that unless your source material is the highest possible resolution, then it's not worth putting on SACD, which is absurd. With that logic, we shouldn't be reissuing 78s on CD.

If your circustances require that you work in PCM, be satisfied that once it reaches the SACD master, all of the resolution is preserved, and in case you haven't noticed, PCM - even 44 or 48k - can sound very very good, especially if you can keep it 24 bit.

And if you've never heard a well-engineered pure DSD recording, you're in for a treat.


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Quote:
quote:
------------------------------------------------------------------------
Originally posted by Curve Dominant:
EVEN IF DSD was some kind of superior high-resolution format, the whole logic behind using it as a consumer-end delivery system shoots itself in the foot...
1. Since one cannot edit DSD, it is rendered impractical and unusable for recording sessions.
2. Therefore, sessions will continue to be recorded in PCM.
3. So, all of those "evil" sounding artifacts allegedly produced by PCM will be faithfully rendered in high-resolution by DSD.
Correct me if I'm wrong, but if the signal hits PCM at any point in the path, the end listener is essentially hearing a PCM signal. Which would mean:
A. For a consumer to buy a DSD system, and DSD discs, would be a complete waste of money
B. For an engineer to master to DSD would be a complete waste of time and money
C. This whole concept of DSD as a practical option for the consumer is a complete hoax.
Where is the fundamental flaw in that equation?
Eric

------------------------------------------------------------------------
Well for one, you CAN edit DSD. We've been doing it for 3 years - first on a prototype Sonic USP system, then on the Sony Sonoma. Sadie is set to release a great stereo DSD workstation and Sonic Solutions has just announced that they will develop DSD editing for the HD system.

You're assuming that unless your source material is the highest possible resolution, then it's not worth putting on SACD, which is absurd. With that logic, we shouldn't be reissuing 78s on CD.

If your circustances require that you work in PCM, be satisfied that once it reaches the SACD master, all of the resolution is preserved, and in case you haven't noticed, PCM - even 44 or 48k - can sound very very good, especially if you can keep it 24 bit.

And if you've never heard a well-engineered pure DSD recording, you're in for a treat.
--------------------
David Glasser
Airshow Mastering
Boulder, CO


David,

I'll tell you what's absurd: out of all of the posts on this thread, you chose that one to take entirely out of context to respond to.

If you had to dig so far into this thread to find that post, then surely you would have noticed a clarification I made of that, which I actually posted TWICE, which read:
Quote:
I'd like to repost this if yous don't mind, lest it got lost in the static...

quote:

------------------------------------------------------------------------
I'd like to point out that DSD does have a timelocation reference (the "other white meat", or axis).
--------------------
George Massenburg
------------------------------------------------------------------------

Correct, which is why simple editing is possible in DSD (???).
It's the EQ, FX, gain changes, etc. type of editing which become difficult in DSD. Why is that? What's the missing reference point?
Eric


So, yes, and I know you had to have noticed, that I am aware that DSD is capable of some level of editing.

So, David, since you have brought up the subject of editing DSD, and since you have obviously enjoyed the benefit of experience of working with DSD, could you at least contribute to our discussion in a way that transcends yet another sales pitch? For example:
1. How superior, scientifically speaking, is DSD over PCM?
2. To what extent do you edit DSD? Cut and paste? EQ? FX? Real time gain changes? Can you be the least bit specific?

Please enlighten us! Three years of experience with DSD cannot have left you entirely tongue-tied in regards to the finer technical details, eh? Do you know why you are using DSD? Can you at least explain that tiny bit of information to us? This is a tech board, after all.

Quote:
David Glasser writes:
You're assuming that unless your source material is the highest possible resolution, then it's not worth putting on SACD, which is absurd. With that logic, we shouldn't be reissuing 78s on CD.


You're assuming that my source material should be worth putting on SACD, without your even explaining why. Which is absurd. With that logic, we should be reissuing everything ever recorded on every sham format that gets shoved down our throats by the likes of the Sony Corporation. Nice try, but you strike out with that pitch.

Quote:
If your circustances require that you work in PCM, be satisfied that once it reaches the SACD master, all of the resolution is preserved, and in case you haven't noticed, PCM - even 44 or 48k - can sound very very good, especially if you can keep it 24 bit.


Thanks for the reassurance. I'm really sold on DSD now, knowing that all of my resolution will be preserved on it. Whew!

Quote:
And if you've never heard a well-engineered pure DSD recording, you're in for a treat.


A treat! Oh, goody! I like treats! Gee, thanks for the treat, David! But, since you can afford the time to post your sales pitch here, can you at least take a moment to explain why DSD will be such a treat, compared to PCM, in mathematical or scientific terms? That is what this discussion is about, after all. I mean, your website is cute, and all. But, you haven't made the sale yet, and we are the types who like to look under the hood before we buy.

A few more questions, David, since you've decided to grace us with your presence:
Quote:
you CAN edit DSD. We've been doing it for 3 years - first on a prototype Sonic USP system


Hmm...you have access to a prototype. Interesting. What exactly is your business affiliation with companies producing and promoting DSD-related products?
Quote:
Sadie is set to release a great stereo DSD workstation


How great is it?
Quote:
Sonic Solutions has just announced that they will develop DSD editing for the HD system.


Really? Can you be a little bit specific? How certain are you that they will do what they have "announced," and why?

You have to understand something, David: some of us are not sold on DSD just yet. We have a sneaking suspicion that it is PCM with a fancy faceplate. You come around here with a sales pitch, but nothing empirical to back it up. We get even more suspicious. The bullshit is piling up so high and so fast with DSD, that we start to feel like need wings to stay above it. Since you and your folks have so much invested in DSD, it may behoove you to cut the crap, unroll some blueprints, and explain to us exactly why, in scientific terms, DSD is where we want to be.

Thanks!
Eric \:\)


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Hi all,
Don't have anything to contribute in regards why DSD is better and how much better it is over PCM, however I would just like to suggest that the answer won't be found unless you direct the search into the source of the origin(which is the inventer/creater of DSD).

I am like everyone else, is very curious about this technology and would be very interested to grasp an understanding of the whole concept of DSD.

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Editing DSD is done in "DSD-wide" which is a 4-bit word. So one step of decimation is already done (some people say PCM wide is almost the same as PCM narrow). I don´t know from any possibility to edit or mix the origin DSD 1-bit signal.

If someone produces completely in analogue and only at the final end converts the analogue signal for release on SACD in DSD, than DSD makes sence.

It also makrs sence to use DSD for cheap consumer players because the DA conversion is done easier - and cheaper. Maybe that´s why SONY decided to release the expensive DSD player first on the High End market, to distract poeple from the fact that DSD is the ideal format for cheap players?

So again, what´s the advantage of DSD? If it is only that we don´t need the decimation to PCM, than I think it´s not a big advantage compared to the disadvantages in production. And there will be better decimation algorithms in the future for PCM.

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Jeez, Curve! Could you give these guys a break? You're acting as paranoid as if this were an X-Files episode! While these guys who have been using DSD are probably on an inside track on SOME level, (or else they wouldn't have access to beta testers, yadda yadda), try to act just a little less snide when someone who uses this shit posts!

(but on the other hand)... I appreciate you trying to seperate the wheat from the chaff in terms of who that's posting here has a vested interest in seeing DSD succeed (ie: secretly pimping their own gear)-- or fail (protecting their own ass, etc).

I'm just asking you to cool your jets here a little so your requests regarding more specific information are taken seriously. Remember, Mr crafty hasn't posted here for a while- these are OTHER guys, who may be OK. Anyhow, they may know stuff you want to find out. OK? (grovel, grovel....)

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Agreed ... here Curve, smoke this ==~ \:\)

One question, if the playback of an SACD disk is interupted, does that mean that the rest of the track is up shit creek? Doesn't it need a continous flow of 1s or 0s to work and so a scratched SACD is far worse than a scratched CD?


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Quote:
Originally posted by Curve Dominant:



You have to understand something, David: some of us are not sold on DSD just yet.



I don't think that any amount of chatter on this forum will 'sell' you on anything. If you want the knowledge you seek, you'll have to get it the old-fashioned way; it won't come pouring out of your internet connection.

The information and equipment are out there and easy to come by. If you are curious about DSD, then listen to some SACDs at the local hi-fi emporium, go to your pro-audio dealer and listen to DSD converters (from Genex & dCs), rent some DSD equipment for your next gig, check out the wealth of technical info from Sony & Phillips, attend the AES and Surround conventions, try to arrange a visit w/ a studio that is doing DSD recording.

You are located in Philadelphia, just a stone's throw from many folks who are working w/ DSD. You should take advantage of that.

Regards,


David Glasser
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Curve - maybe a switch to Sanka would help and maybe you should take up David's Pepsi challenge in & around Philly instead of flogging every person that uses DSD and tries to state some thoughts about it ......... I mean they're out there on the edge of technology, trying something new. Breath in ~ breath out. Ahhhh.

Tony

[ 02-01-2002: Message edited by: td ]

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I don't think theoretical considerations mean a hill of beans in audio unless you are speaking of obvious degradation. Implementation has always been the main thing that counts. The real issue about DSD is how well it will be implemented vs. how well future PCM products will be implemented. Hopefully both will soon reach the point that conversion to analog or conversion from one to the other causes no perceptible degradation.

Myself, I'm really happy to see some real competition in audio quality for the first time in years. We all will benefit from this.


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Everybody,

My bet is on the Sadie system
versus the Sony. I have disliked
Sony editing products going
back to the early days of 1630.
I've been a believer in their(Sadie)
product ever since Glen Meadows
of 'Masterfonics' fame turned
me on to Sadie back in 95'.
I'm still sticking with the dCS
DSD converters, & then possibly
Apogee DSD conversion(if they
come up with them in the future).


"The Crafty One".

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Quote:
You are located in Philadelphia, just a stone's throw from many folks who are working w/ DSD. You should take advantage of that.
Regards,
--------------------
David Glasser
Airshow Mastering
Boulder, CO


Well, not exactly, David.

Every studio owner I've talked to so far has got their sights set on upgrading to 96K, so they can go with DVD-A. I called one of the biggest commercial mastering houses here, and they do not master with DSD. I didn't speak with the head engineer yet, but the house manager told me they had a SACD player in to demo about a year ago. They did some A/B testing, and...a year later, still feel no need to jump in. I asked the manager what he thought of the sound, and he said, "Yeah, it sounded great." I asked him how much better than the Redbook CD, and he said, "A little bit." They do master with DVD-A, however, and are very happy with that format.

As far as the local hi-fi emporiums go, those $5.50/hour salesman aren't going to tell me anything more in-depth than what Sony wants them to tell me about the nuts and bolts of DSD. You refuse to answer any of my questions as well, about the nuts and bolts of DSD, even though you've been working with it for three years, and have time to post here.

The following attitude is indicative of the highest level of constructive input we've recieved so far:
Quote:
poated by Tony:
Curve - maybe a switch to Sanka would help and maybe you should take up David's Pepsi challenge in & around Philly instead of flogging every person that uses DSD and tries to state some thoughts about it ......... I mean they're out there on the edge of technology, trying something new. Breath in ~ breath out. Ahhhh.
Tony


That's it? That's the most specific anyone who advocates DSD can be? "Try it, you'll like it!" followed by a snarky taunt. Very helpful. You're a real pillar of the scientific community, Tony! \:D

How is it "on the edge"? What makes it so "new"? If you have to flog someone in order to get a straight answer out of them, is that such a bad thing? I'll stick to real coffee, thank you very much. So far, the thoughts posted by people who do use DSD have been utterly useless. To me, that says something about them, and format.

Once again: Does anyone out there have some hard science that tells me I'm dead wrong? I would love to see it.

Thanks in advance!
Eric \:\)


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Curve D.,

Did it ever come to mind that the DSD info could be so proprietary in nature that Sony has a lock on it, & that's why people are not giving up the inside story on DSD, especially to early nay sayers as yourself. And not for the reason of trying to push a new product just to enhance their earings ratios but to in fact properly promote the new medium by influencing people to go and actually listen to it before killing it at the drawing board. Some of the most revered high-end equipment has only 'good' technical spec's but when you actually listen to it 'it's a whole new world'. Let the ears be the judge sometime.

"The Crafty One".

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Point well taken, crafty one.

My ears haven't decided yet.

E \:\)


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Eric-

I don't think that the folks who want you to switch over to decaf are all "pro DSD" and trying to get you to back down. I, for one, just want to assume that MOST of these guys are using the thing because (in descending order of liklihood):

1. it sounds better to them and it's worth the effort/expense

AND/OR:

2. The snob value of using this new technology will sell them records (they hope)

3. They all secretly will get their bank accounts stuffed by Sony if they can get this stupid thing respected by other engineers.

4. They secretly work for a paramilitary wing of Sony that will achieve world domination if it can brainwash enough gullible engineers from Philadelphia into purchasing this expensive box without checking it out before they buy. So far, their evil plan seems not to be working.


I, for one, don't really need the science of it explained to me before I can decide if it sounds good-- anymore than I need to understand the science behind how my TV works before I turn it on and watch 'The Prisoner'. Now THERE was a man justified in feeling paranoid!

"I am not a number- I am a free man!"

HAAAAA hAhAHAHAHAHAHAHAHAHAH!!!!!!

sincerely,

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DSD has the potential of sounding better than CDs today. And as has been said a few times it may already be self-evident for those who've heard it. Specwise, it ought to compete very well with DVD-A. But that's just one aspect isn't it? After all, can't you just remaster to DSD/DVD-A or whatever?

Often enough, you'll hear folks right on this form yearn for that good old analog sound--in a production context. In other words, recording, mixing, & mastering. Though I'm also certain that many of these same folks wouldn't give up the flexibility and myriad conveniences of digital production.

So my question is, if DSD/DVD-A are better consumer formats (the marketplace will judge soon enough) what are the expected production advantages of DSD? Will it be easier to edit, process, route, mix, master, ...

Are there any other advantages or is it just supposed to sound better than 48+/24?

Anyone with experience or guesses, wanna chime in on this one?
I haven't heard any so far.

-Dennis

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I'm in the middle of (DSD) post-production right now, so this is brief:

1. DSD is not pcm in disguise. The pcm part of co-processing DSD data during mixing, editing, etc. is 4-bit at 64x, also known as "wide DSD". The effective bandwidth exceeds that of oversampled LPCM. There is R&D underway to eliminate the pcm co-processing.
2. SACDs play the same as a CD - interrupting it does nothing unusual. Hybrid SACDs play in ALL CD-compatible players.
3. DSD is being edited, mixed, EQd, compressed, etc everyday in our facility on the Sonoma DSD editor. The Sonoma is NOT a Sony pro audio product. It is under development.
4. Sony does not pay me or Telarc to say good things about DSD. Actually, no one at Sony has ever even suggested what we say about DSD, good or bad.
5. I use DSD, (Meitner converters, not dCS) because it sounds much closer to the board output than any pcm converter at any rate has...so far. And because I can.
6. Many, many projects will not benefit from release on SACD at this time. Not all projects are maintained at high-resolution all the way through the process to be worth the extra time, expense, etc. Not all projects should be heard in high-resolution. Transferring 16-bit 44.1 pcm recordings to DSD does not get you anything more than what you started with. An SACD of that pcm recording will not sound any different than a CD of the same recording.
7. An outstanding DSD recording from SACD will sound better than the same recording as played from a CD regardless of the system. So will an outstanding analog recording transferred to DSD.
8. The quality of the music is MUCH more important than the means of recording or reproducing it.


Instead of going 'round and 'round with speculation here, take advantage of the wealth of knowledge within the Audio Engineering Society. The nitty-gritty on DSD and delta-sigma conversion can be found through the the AES preprint order page:

http://www.aes.org/publications/preprints/search.html

and search for these publications:
AES preprint 5377:
DSD-Wide. A Practical Implementation for Professional Audio

AES preprint 5392:
Investigation of Practical 1-Bit Delta-Sigma Conversion for Professional Audio Applications

AES preprint 5393:
Acheiving Effective Dither in Delta-Sigma Modulation Sustems

AES preprint 5394:
The Practical Performance Limits of Multi-Bit Sigma-Delta Modulation

AES preprint 5395 (the Ying...)
Why 1-Bit Sigma-Delta Conversion is Unsuitable for High-Quality Applications

AES preprint 5396 (and the Yang...)
Why Direct Stream Digital (DSD) is the best choice as a digital audio format

AES preprint 5397:
SDM versus LPCM: the debate continues

AES preprint 5398:
Towards a Better Understanding of 1-Bit Sigma-Delta Modulation

AES preprint 5399:
Editing and Switching in 1-Bit Audio Systems

Happy Reading!

With Best Regards,
Michael Bishop
Telarc International


Best Regards;
Michael Bishop
Telarc International
http://www.telarc.com
SACD, DSD & DVD-A Editing and Mastering available now at:
Wired 4 Music
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Quote:
I'm in the middle of (DSD) post-production right now, so this is brief:
1. DSD is not pcm in disguise. The pcm part of co-processing DSD data during mixing, editing, etc. is 4-bit at 64x, also known as "wide DSD". The effective bandwidth exceeds that of oversampled LPCM. There is R&D underway to eliminate the pcm co-processing.
2. SACDs play the same as a CD - interrupting it does nothing unusual. Hybrid SACDs play in ALL CD-compatible players.
3. DSD is being edited, mixed, EQd, compressed, etc everyday in our facility on the Sonoma DSD editor. The Sonoma is NOT a Sony pro audio product. It is under development.
4. Sony does not pay me or Telarc to say good things about DSD. Actually, no one at Sony has ever even suggested what we say about DSD, good or bad.
5. I use DSD, (Meitner converters, not dCS) because it sounds much closer to the board output than any pcm converter at any rate has...so far. And because I can.
6. Many, many projects will not benefit from release on SACD at this time. Not all projects are maintained at high-resolution all the way through the process to be worth the extra time, expense, etc. Not all projects should be heard in high-resolution. Transferring 16-bit 44.1 pcm recordings to DSD does not get you anything more than what you started with. An SACD of that pcm recording will not sound any different than a CD of the same recording.
7. An outstanding DSD recording from SACD will sound better than the same recording as played from a CD regardless of the system. So will an outstanding analog recording transferred to DSD.
8. The quality of the music is MUCH more important than the means of recording or reproducing it.

Instead of going 'round and 'round with speculation here, take advantage of the wealth of knowledge within the Audio Engineering Society. The nitty-gritty on DSD and delta-sigma conversion can be found through the the AES preprint order page:
http://www.aes.org/publications/preprints/search.html
and search for these publications:
AES preprint 5377:
DSD-Wide. A Practical Implementation for Professional Audio
AES preprint 5392:
Investigation of Practical 1-Bit Delta-Sigma Conversion for Professional Audio Applications
AES preprint 5393:
Acheiving Effective Dither in Delta-Sigma Modulation Sustems
AES preprint 5394:
The Practical Performance Limits of Multi-Bit Sigma-Delta Modulation
AES preprint 5395 (the Ying...)
Why 1-Bit Sigma-Delta Conversion is Unsuitable for High-Quality Applications
AES preprint 5396 (and the Yang...)
Why Direct Stream Digital (DSD) is the best choice as a digital audio format
AES preprint 5397:
SDM versus LPCM: the debate continues
AES preprint 5398:
Towards a Better Understanding of 1-Bit Sigma-Delta Modulation
AES preprint 5399:
Editing and Switching in 1-Bit Audio Systems
Happy Reading!
With Best Regards,
Michael Bishop
Telarc International


\:\)


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Thanks for all the links, Michael! I wonder if this is some of the scientific reading material that some of us have been lusting for.

thinking this over a bit more, I came up with these thoughts:

Even though it has been perhaps the most important recurring theme on this thread, I wonder sometimes if connecting the *science of a process* to the *sensation of hearing that process in action* is really so key to understanding whether it is the right thing to use in one's music.

One might as well study the taste buds to decide why a certain food appeals, or our auditory system to understand why a certain piece of music makes us feel nostalgic...


I expect that a lot of these guys that are using DSD are doing so because they like it's sound. They don't have to to justify their choice with science- though that may frustrate the hell out of folks that want hard science answers- they may have just heard it and listened hard and made their decisions.

BUT anyone who is invested in DVD at this early stage is hoping their investment works out, I assume. Some may (do) have ties to Sony- I suppose some of them are insiders on some level (after just reading your post above, Michael, I guess many users DON'T have any connection to Sony either). So in some cases it might be hard to get straight answers from a given user for that reason too.

Anyhow, I'd love to see 'why' DSD should sound better scientifically- but if it really does sound better to ME when I finally have a chance to test it and use it- then knowing WHY may be as pointless and someone explaining WHY a certain joke is funny or WHY a certain food tastes good to me. I'll just 'know'.

Mark 'I don't understand art but I know what I like' Lemaire

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I think we need to consider the issue of DSD as a production format seperately from DSD as a distribution format.

As a distribution format, DSD is far, far better than CD. I've had the privilage of listening to it in David Glasser's mastering rooms, and he's made me convert, at least as concerns DSD's sonic merits. As for its technical necessity (or even its long-term sufficiency), I still have my doubts. But if SACD beats out DVD-A in the consumer marketplace, I won't be crushed. I just hope one or the other wins, because I'd love to see conventional CDs go the way of the analog cassette.

As a production format, DSD has the practical difficulties mentioned by others. Most of the important audio processing algorithms that we use with PCM have yet to be invented for DSD. I think that DSD-oriented engineers will eventually learn that its one-bit form is best reserved for distribution. My guess is that in a few years, Michael Bishop will think the idea of actually _recording_ in a one-bit format was rather quaint. If we have to record in DSD, I hope it turns out to use multibit modulators which can be properly dithered. I hope we don't have to settle for the current DSD bitrate which, from an information theoretic viewpoint, is already inadequate to store the output of the latest PCM audio converters. Our production format ought to be better than the delivery format.

I don't see a whole lot of difference between high-rate PCM data format, and this "DSD-wide" format. You can do the processing slowly at 24-bit resolution, or quickly at 4-bit resolution. Multiply sample-rate by word width and you get the channel capacity. If the channel capacity is the same, then you're basically handling the same amount of information. That means we're down to arguing about who can implement the better processing algorithm. Same as it ever was.

DSP guys who code for PCM are wise to use oversampling and then decimate back down to their preferred resolution. DSP guys who burn Xilinx chips for DSD probably don't have to oversample, and instead of putting a decimation filter at the back end, they need to build another modulator. I personally think its more difficult to design a good modulator than a good decimator, but maybe I just don't have true religion. Both camps had better keep track of their accumulator widths, or all bets are off.

I'm still at a loss to find anything about DSD that's THEORETICALLY superior to PCM. I'll admit that there's still a lot wrong with current IMPLEMENTATIONS of PCM. The economics associated with the personal studio revolution (of which I am a beneficiary, BTW) have made for some pretty damn sloppy DSP coding in the PCM world. If PCM is to reach its true potential, that has got to stop. When PCM algorithms start being routinely coded with oversampling and high-order noise shaping, I predict that noone (not even Michael Bishop) will be able to tell it from DSD.

Best regards to all,

David L. Rick
Hach Company (the day job)
Seventh String Recording (sleepless nights)
drick@hach.com

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My sentiments, David, exactly. Now back to Michael's reading list.

These papers are expensive at $10 apiece. AES policies for access are a bit unusual for such organizations these days. Most research docs are free. But I'm sure there's something about AES finances I don't understand.

-Dennis

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Quote:
posted by David L. Rick:
My guess is that in a few years, Michael Bishop will think the idea of actually _recording_ in a one-bit format was rather quaint. If we have to record in DSD, I hope it turns out to use multibit modulators which can be properly dithered. I hope we don't have to settle for the current DSD bitrate which, from an information theoretic viewpoint, is already inadequate to store the output of the latest PCM audio converters.


This reminds me of a question someone once asked me: how does one dither one bit, and where does the dithering noise go?

Quote:
Multiply sample-rate by word width and you get the channel capacity. If the channel capacity is the same, then you're basically handling the same amount of information. That means we're down to arguing about who can implement the better processing algorithm. Same as it ever was.


This is why I always suspected DSD was a "6 of one, half-dozen of the other" sort of marketing gimmick. Higher sampling rate+lower bit depth, as opposed to vice-versa...it just didn't seem like "The New Frontier" that some peeps were making it out to be.With all of Ed Meitner's flowery analogies, there was no hard math that I could see poking it's head through, and saying, "This is a shorter distance from point A to point B!" And that is ultimately what we are looking for, right? A clearer signal path. I'm all for clear and pro-active progress. Juggling numbers is not progress.

Quote:
I'm still at a loss to find anything about DSD that's THEORETICALLY superior to PCM.


Well, we've gone to 4 pages of debate on gm's board over this issue, with no hard math posted which would contradict that statement.

Quote:
I'll admit that there's still a lot wrong with current IMPLEMENTATIONS of PCM. The economics associated with the personal studio revolution (of which I am a beneficiary, BTW) have made for some pretty damn sloppy DSP coding in the PCM world. If PCM is to reach its true potential, that has got to stop. When PCM algorithms start being routinely coded with oversampling and high-order noise shaping, I predict that noone (not even Michael Bishop) will be able to tell it from DSD.


\:\)

Eric Vincent
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OK thanks! I'll try not to get too excited until we're hearing 24 bits or more at 2+ MHz! Maybe by then I'll be able to afford it, and people will routinely listen to 16 bit 2+Mhz at home. My prediction- like the hula hoop, great home stereos will become hip again, maybe by then.

"I don't think theoretical considerations mean a hill of beans in audio unless you are speaking of obvious degradation. Implementation has always been the main thing that counts. The real issue about DSD is how well it will be implemented vs. how well future PCM products will be implemented. Hopefully both will soon reach the point that conversion to analog or conversion from one to the other causes no perceptible degradation.

"Myself, I'm really happy to see some real competition in audio quality for the first time in years. We all will benefit from this."

This is the essence- it's junk science if it doesn't deliver a classic recording medium, and if it does, it's progress!

Ted


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Quote:
Originally posted by David L. Rick:
[...] You can do the processing slowly at 24-bit resolution, or quickly at 4-bit resolution. [...]

...or, as Ed Meitner suggests, you can do single-bit at 256FS.

In fact, it may matter how you get there. What if there were sample-rate related artifacts (as distinct from the bandwidth limitations, including filtering problems, that have everyone's attention) in PCM? Then, only a higher sample rate is going to improve things? And, if that's true, what's the 'next frontier' in sample rates? 96kHz doesn't seem like a big step. Maybe 192 (which sounds pretty good to me)?


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George,

If we keep pushing for ever larger amounts of information content in our digital signals, I would guess that the conversion process will start moving faster in the direction of higher sampling rates. As David suggested above, the information content of a signal is the product of the length of the code words (bit depth) and sampling rate.

With today's technology, we're much better at building super fast clocks than we are at resolving very small voltages. DSD's ~3MHz sampling rate has a long way to go before it bumps up against the ~100GHz bandwidth offered by high-end oscilloscopes. This isn't exotic equipment in the larger (non audio) world. The ADC's of these beasts are 12-bits wide and can resolve single microvolts using off-the-shelf instrument grade preamps. Today's best audio converters are skimming the same voltage range and there's not much farther to push since Johnson Noise Land lives just a few dB down from there (at least at room temperature). In other words, there's lots of land to explore bandwidth wise, and not much more amplitude wise.

So if the industry keeps pushing for more bits-per-second, we will all see higher conversion rates. But is that important for production? I recall from last year's monster 96k thread, that we really don't have all the kinks sorted out in real-world converters--at least not to the point where we can take advantage of all those bits flying by really fast.

-Dennis

As an aside ...

Indeed you can view the triumph of digital computing over analog computing as a victory of the precision clock-makers over precision voltage measurers. Digital has two levels of interest but takes time (you need a fast clock). Analog computers work almost instantaneously, but in order to get high-quality answers you need to measure really precisely. Well who won?

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I'd made the observation that the choice between 24-bit, 96ksps and 4-bit, 576ksps was an irrelevancy from an information-theoretic standpoint.

Quote:
Originally posted by George Massenburg:

...or, as Ed Meitner suggests, you can do single-bit at 256FS.


George, I think that's the one thing we don't want to do! Using a one-bit as an intermediate storage format to which we return after each processing step is a really, really bad idea. That's because to get back there, we have to go through another single-bit modulator, and those are inherently and irretrievably flawed (c.f. Lipshitz and Vanderkooy). If the marketplace dictates that we have to put our audio through a one-bit modulator in order to deliver it to the consumer, well, so be it. But let's do that once, and only once, at the end of the production chain.

Quote:
GM again:

In fact, it may matter how you get there. What if there were sample-rate related artifacts (as distinct from the bandwidth limitations, including filtering problems, that have everyone's attention) in PCM? Then, only a higher sample rate is going to improve things? And, if that's true, what's the 'next frontier' in sample rates? 96kHz doesn't seem like a big step. Maybe 192 (which sounds pretty good to me)?


You may be correct. Some people think that PCM (as implemented) doesn't start to compete sonically with DSD until 192ksps. If that's so, it's not because of "channel capacity", but because of what we're putting into the channel. Whether it's the time response of the filters, a crappy quantized-coefficient implementation of them, or aliasing artifacts inadvertantly modulated down to baseband, something is messing up the conversion process and the semiconductor manufacturers are too focused on price-point to fix it.

Sony's solution was to avoid having to decide what's the best decimation filter at record time. (Unfortunately, they've effectively left the decision in the hands of the folks who make $200 playback equipment.) I'm rather inclined to agree with them, at least in the following way: If I can't buy a 96ksps PCM converter with a decimation filter that I can trust, then maybe I should use a 192ksps converter, followed by a software down- (or up-) conversion algorithm whose provenance and performance is known to me. In fact, I can do all my intermediate production at 192ksps if I decide that sounds any better (and I don't have to mortgage my house to buy the tools). Finally, if it takes a one-bit modulator to get my work to consumers I can still do that at the end.

Rereading your comment, I realize that you what you meant by "sample-rate related artifacts" was not problems in the decimation filter, but subsequent processing artifacts which alias into the audio band. I agree that oversampling will reduce these. But how much oversampling is actually needed? If we process our audio with something as egregious as a squaring circuit, it will only double the bandwidth. So if we perform the processing at 192ksps, being careful to filter to 96kHz bandwidth before each processing step, we should be in good shape.

Best regards,

David L. Rick
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I just wanted to throw in that I'm agreeing with one of the opinions above.

There is nothing wrong with SACD as a delivery format, and I believe that there are probably advantages: it will be much more practical for manufacturers of home hi-fi equipment to design that equipment so that they can meet their demands for affordability while providing better sound quality. SACD is a fairly practical way of allowing the mastering studio to do the D/A filtering, standardizing this one aspect of the playback system. At least we engineers will be able to ensure the quality of the D/A filter, while there are still many areas where we will have no control.

I am still not in agreement that DSD converters can be made to sound better than PCM converters can be made to sound, and as this is the case, I still support working at lower rates, conserving harddrive space, and not simply subsribing to higher rates of everything for unsubstantiated everything. Therefore, I do not see any reason to be in support of recording DSD, and processing at DSD is practically impossible without first converting to PCM of some form anyway. But delilvering at DSD is something I haven't seen a valid reason to avoid, and I do see benefits of supporting.

Nika.

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Gentlemen,
I respectfully suggest that Mike Bishops posts be re read and that he be prompted for more input.
Several pages back I think his posts got shouted down a bit.

As a former Telarc artist I can clearly state that I have absolutely nothing positive to say about the label. I hated being there.
BUT to their credit, Telarc relentlessly pursues Audio Excellence.
They do this without being a multinational conglomerate that also manufactures gear (ie. sony)
Telarc has always been on the very bleeding edge and it is no surprise to see them out there now. Somehow Telarc spends studip amounts of money on the highest calibre gear and still stay in business!
And again, I clearly state that I was treated like shit as an artist on there roster. My respect for their quest for audio excellence remains in spite.
Mike mastered my first two records. Not for a minute did I sense the slightest air of audio superiority (or bullshit) about him.
He is all about SOUND.
He was always available to solve tech issues (and I had a "first batch" 02R that needed much help!) and helped me get the most out of the gear that I had. Not once did he put down my studio gear. (02R-DA88's at that time, far below Telarcs usual stuff)
Is it possible to be a cutting edge audio elitest and a humble, practical, down to earth engineer?!
Yes.
I have never seen evidence of Mike having an agenda of any sort. His picture is seldom in MIX mag standing next to the latest Digital Mixer and Phil Ramone.
His car was like all of ours (late late late model - he aint getting rich endorsing anything).
SO,
If HE hears a difference, well shit then, I have to believe that there is a great deal to it.
There are only a very few people whose opinion would carry that much weight with me.
Mr. Bishop is one of them

Also,
Crafty, Please post some bio info on yourself and perhaps you real name. Almost every one in here puts there entire reputation on the line when posting. Please be kind enough to do the same.

as a side note:
I am sure that when digidesign comes out with PT DSD - the upgrade path will suck my bank account dry, as the one to HD just did!

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Ed-

Thanks for your post.

I also hear nothing but good things about Michael Bishop (know nothing of Telarc and how they treat their artists) and was sorry to see him 'shouted down' as well.

Hoping to see him and other DSD users post here with not only their scientific arguments, but also their personal opinions.

ML

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3 µs is 333 kHz ! Far away outside the audible range.

All signals looks perfectly clean to me : same energy (surface under the curve), but more or less lowpassed in frequency.

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PS : not the same energy, of course (silly me), the highs are removed.

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Quote:
Originally posted by Rog:
http://www.merging.com/2002/images/dsdresponseneon.gif

Any thoughts?


Notice the extreme difference in ringing around the main impulse between the various pcm and DSD samples. I had never before seen such an analysis and comparison of the various formats, but this very well illustrates the difference that I ***hear*** between them. In all versions of pcm recordings - at any sample rate - I hear "clatter" around the detail of the signal. It was one of those things I got used to in pcm over the past 22 years, but now find really grating after living with DSD the past 4 years. That "clatter" is also how I identify that a track is of pcm origin in comparison to a DSD track. Once you know the "clatter" is there it's easy to find on a repeatable basis, as in blind testing - and you don't have to have "golden ears" to find the artifacts.

Thanks for posting the link, Rog!

With Best Regards,
Michael Bishop
Telarc International


Best Regards;
Michael Bishop
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I am actually confused by the chart. How is it possible that a 333kHz impulse could get into a 44.1kS/s PCM system in ANY capacity?

Anyone? What am I missing. I must be missing something, here...

Thanx!
Nika.

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Quote:
Originally posted by Nika:


Dennis,

Don't forget that the D/A's of a DSD system still have filters in them.

Nika.


I think it might be possible to move these filters all the way to the speaker driver, given enough time and R&D monies.

Thinking ahead,
Dan

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Quote:
Notice the extreme difference in ringing around the main impulse between the various pcm and DSD samples. I had never before seen such an analysis and comparison of the various formats, but this very well illustrates the difference that I ***hear*** between them. In all versions of pcm recordings - at any sample rate - I hear "clatter" around the detail of the signal. It was one of those things I got used to in pcm over the past 22 years, but now find really grating after living with DSD the past 4 years. That "clatter" is also how I identify that a track is of pcm origin in comparison to a DSD track. Once you know the "clatter" is there it's easy to find on a repeatable basis, as in blind testing - and you don't have to have "golden ears" to find the artifacts.
Thanks for posting the link, Rog!
With Best Regards,
Michael Bishop
Telarc International


Michael,

Do you hear the same amount of "clatter" at 10K or 20K as the amount you hear at 333K? What about the 20hz to 1K range? Are there illustrations of the amount of "clatter" in that range floating around that we could look at?

Thanks!

Eric \:\)


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Quote:
Originally posted by td:
Curve - maybe a switch to Sanka would help and maybe you should take up David's Pepsi challenge in & around Philly instead of flogging every person that uses DSD and tries to state some thoughts about it ......... I mean they're out there on the edge of technology, trying something new. Breath in ~ breath out. Ahhhh.

Tony

[ 02-01-2002: Message edited by: td ]


Speaking of Pepsi...what do they use for sweetening up there in Philly. When I moved from Chicago to Miami, I was a bit disappointed that my daytime drink of choice (Pepsi in a can) wasn't the same. It seems that down in Florida there is a greater concentration of sugar to corn syrup AND a greater concentration of syrup to the carbonated water - I suppose this is because they expect you to have to use ice, and sugar is more plentiful down here than corn.

Perhaps a bit too sensitive,
toast

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There might be some clatter in PCM that is not in DSD, but it's not what we can see in this picture.

Look at what happens to a pulse when it's mathematically lowpassed : http://pageperso.aol.fr/Lyonpio2001/pictures/lowpass/lowpass.htm

In the picture we're discussing about, PCM versions of the pulse are perfectly lowpassed versions of the analog one, with no distortion at all exept the removal of ultrasounds.

Therefore they should sound exactly the same to any human, exept maybe some asthmatic children, that are said to be able to hear up to 30 kHz. For them, the 48 kHz version will sound different from the others.
That, is valid only if the samples are actually played as we see them displayed on the picture, of course. That must be impossible since the air itself doesn't transmit the sound well, if at all, at such frequencies as 333 kHz !

For me, this picture is aimed at fooling people into believing that DSD is superior to PCM. What MAY actually be, independently of this picture, but this advertising is just deceiving of the masses.

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Quote:
posted by digitaltoast:
Speaking of Pepsi...what do they use for sweetening up there in Philly.


I generally don't consume refined sugar or anything that contains it, so I couldn't tell you.

Rog,

Where did you find that jpeg you posted?

Any idea who created that demonstration?

E \:\)


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Quote:
Originally posted by Nika:
How is it possible that a 333kHz impulse could get into a 44.1kS/s PCM system in ANY capacity?


It depends on the DAC circuitry itself, i.e. how it measures the voltage. If it measures the electric charge accumulated by the current, then the same final accumulated charge in a 1/44100 of a second can be produced by bursts of current as short as you want (3 µs) inside the 1/44100 s window.
After that, the oscillations come either from the playback filters (DAC), either from the display used.
Cool Edit actually displays pure digital pulses exactly like that, with an oscillating line between the samples : http://pageperso.aol.fr/Lyonpio2001/pictures/lowpass/cool.gif
The squares are the samples themselves. This digital file presents one sample at full scale.
The curve displayed is the regular display of Cool Edit, that draws the perfect mathematical curve fitting the points. Another proof that the oscillating pulses are not distorded.

Yet another proof : The Fourier transform of a Dirac distribution is a constant function. To lowpass it is to cut it so as it becomes a "step" function, and the inverse Fourier transform of a step function has a Cardinal Cosinus shape (Cos X/X).

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Quote:
posted by Pio2001:
There might be some clatter in PCM that is not in DSD


If there was "clatter" in the source PCM, of course you wouldn't hear it in DSD, because all of the 3rd harmonic distortion caused by the cyclic non-random HF noise from the Delta Sigma Modulator would smother it...right?

Reference Stan Lipshitz:
Quote:
At the Audio Engineering Symposium in September, 2000, Stan Lipshitz showed that a 1-bit system is inherently non-perfectible, and he is right. You cannot properly dither without overloading the converter, because the correct dither (white noise with a triangular pdf and a width of 2 LSBs) covers the full range of the converter! Thus, as soon as you put in any input, the converter overloads. Of course the manufacturers don't put in 2 LSBs of dither, but then the converter is not correctly dithered and suffers from both distortion and noise modulation (and idle tones). This is why the state-of-the-art converters do not use 1-bit internally but rather a low number of bits, such as 3. The trend for professional audio equipment is 8-bit, 64 times oversampling, noise-shaped.

Mike Bishop? Glasser? Is this why you think DSD "sounds" like analog? Yeah, it's "less fatigueing..." on a bad mix, perhaps. I could buy a BBE Aural Exciter off of Ebay for cheap if I want to mix harmonic distortion into my tracks, or buy an "Analog Simulator" plugin - but at least I can disable a plugin. DSD-generated distortion cannot be disabled, as far as I can see.

Milk and cookies have kept The Curve up late thinking about this. I just don't see how it's a "cleaner" path as Phillips is suggesting. There are a lot of folks more knowledgable than myself who participate on this board, and this thread has been up for awhile now, but still, no one has yet explained how you dither a 1-bit signal, and where the dithering noise goes.

What am I missing? Thanks in advance.

Eric \:\)

(Edited for spelling errors)

[ 02-17-2002: Message edited by: Curve Dominant ]


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Quote:
delilvering at DSD is something I haven't seen a valid reason to avoid, and I do see benefits of supporting.
Nika.


Nika,

Read my previous post, and then re-iterate your view.

Thanks in advance.

Eric \:\)


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Quote:
Originally posted by Curve Dominant:


Nika,

Read my previous post, and then re-iterate your view.

Thanks in advance.

Eric \:\)


Eric,

OK, no problem. Let's review how a D/A works for a moment:

A signal at 44.1kS/s has absolutely no data in it beyond 22.05kS/s, as we know from Nyquist. But all it is is a series of points with no data between them. Depending on how that data is turned into analog signals additional information CAN get into the system.

For instance, if a D/A uses "sample and hold" then any particular value will be held until the next value comes along. This takes our series of dots and turns them into a bunch of miniature stair-steps, like square waves. This, obviously, is not desireable. What we want is a smooth line connecting all of the dots in sinusoidal fashion.

What happens is we "oversample" the material, creating a bunch of dots between all of the dots, and using filters to insure that they conform to the waveform that we actually want there. If we use an "8x oversampling" D/A it will add 8 additional dots between each pair of dots in our timeline, adding a filter that is a brickwall at 22.05kHz so that the resultant stairsteps now contain no material that is less than 529.2kHz (3rd harmonic of 176kHz, which is 8x 22.05kHz). The bottom line is that we still have stair-steps, but they are much smaller steps, and the minimum frequency over 22.05kHz that is contained in their material is up at around .5MHz. At this point the signal is actually turned into a 1 bit datastream, akin to DSD. Then it is converted to analog using "sample and hold". Then an analog low pass filter is put in at just below .5MHz that essentially "rounds" this small series of stair steps into the sinusoidal waveforms that we were looking for. This entire process is done in the D/A converter, and the quality of these filters is one of the differences between "adequate" and "great" converters. Quality is important because the digital filter that is used for the oversampling process can suffer from errors that will negatively affect your audio. This process is just as imperfect as the A/D conversion process.

The problem here is that home hi-fi equipment generally has very poor filters in it - sometimes none at all. This means that the way you hear the material in your studio does not translate as well to home hi-fi equipment.

Let's discuss the paradigm of tracking and mixing and recording PCM, and then turning it into DSD for delivery. The process for turning PCM into DSD is essentially the same process as oversampling in a D/A. The material is oversampled at 64x with a filter put in and the result is then "encoded" in a 1 bit datastream. This process would obviously happen after any dithering that would applied, and after all other processing and mastering. Basically the mastering engineer would run your 44.1kS/s datastream into a little black box that would oversample and filter it into DSD, just like your D/A's do. Except that the mastering engineer has the tools to do this process at a level that we can't afford in the studios and most of our listening audience can't afford either. Then your material is put on an SACD disk for duplication and distribution.

We're not saying here that our listening audience does not have the ability to purchase D/A's that can perform to the level that a mastering engineer's can. We're not saying that SACD will inherently sound better than we can make PCM sound, because it's really the same thing, SACD as a delivery format just being one step beyond PCM. Instead what we're conceeding here is that by taking the filters and the oversampling technology out of the consumers hands and putting them into the mastering engineer's hands we have helped to standardize what our consumers are listening to. This way:

Record, edit, mix, master -> storage (delivery format) -> {consumer playback} oversampling D/A with filtering turning it into a 1 bit DSD datastream -> conversion to analog -> to amplification -> to speakers

turns into:

Record, edit, mix, master, oversampling D/A with filtering turning it into a 1 bit DSD datastream -> storage (delivery format) -> {consumer playback} conversion to analog -> to amplification -> to speakers

Now if we could only standardize what speakers our consumers listen to!?

So again, SACD is not inherently going to make the music sound better, but by using it as a delivery format it helps to remove one potential for sound degradation from the hi-fi componentry, further standardizing the way that our consumers hear what we intended for them to hear.

Does this help?

Nika.

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Quote:
posted by Nika:
The process for turning PCM into DSD is essentially the same process as oversampling in a D/A. The material is oversampled at 64x with a filter put in and the result is then "encoded" in a 1 bit datastream. This process would obviously happen after any dithering that would applied, and after all other processing and mastering. Basically the mastering engineer would run your 44.1kS/s datastream into a little black box that would oversample and filter it into DSD, just like your D/A's do. Except that the mastering engineer has the tools to do this process at a level that we can't afford in the studios and most of our listening audience can't afford either. Then your material is put on an SACD disk for duplication and distribution.


Yes, but don't you still end up with all of the flawed audio inherent in putting it through a 1-bit Delta Sigma Modulator? That's one of the points I seem to be stuck on. Isn't the 3rd harmonic distortion from statistical noise modulation inherent from the DSM still there? And what about all of the HF noise generated, because we have a signal that has full scale peak to peak noise up to 1.4Mhz, correct? Won't that burn some tweeters if we don't filter it - which brings us to: what type of filters? Analog? And won't that roll-off audio as well...that can't possibly compete with DVD-A standards.

Quote:
posted by Nika:
what we're conceeding here is that by taking the filters and the oversampling technology out of the consumers hands and putting them into the mastering engineer's hands we have helped to standardize what our consumers are listening to.


This is something else I don't get: What type of D/A filter? If it's 1-bit data going through a FIR filter: doesn't that just turn it back into multi-bit PCM, but with all of the flawed audio from the DSM in tact?

The only logical reason I can see for Sony/Phillips to opt for 1-bit, is because they can manufacture cheaper consumer gear because there's less information that needs to be processed at the end.

I still don't see how DSD is not PCM, except that it has to really bend over backwards and jump through hoops in order to maintain integrity as a 1-bit data stream. Which it can't anyway if it needs to be processed - "DSD Wide" is multi-bit PCM as far as I can see.

Thanks, Nika, for you patience. Sorry if I sound like a retard sometimes.

E \:\)


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Nika,

I apologize if there seems to be a delayed reaction in my posts, but as you know I'm a little new to this field, and sometimes I have to think for awile to make sure I have my facts straight (and of course I'm still not sure but anyway...).

You posted:
Quote:
What happens is we "oversample" the material, creating a bunch of dots between all of the dots, and using filters to insure that they conform to the waveform that we actually want there. If we use an "8x oversampling" D/A it will add 8 additional dots between each pair of dots in our timeline, adding a filter that is a brickwall at 22.05kHz so that the resultant stairsteps now contain no material that is less than 529.2kHz (3rd harmonic of 176kHz, which is 8x 22.05kHz). The bottom line is that we still have stair-steps, but they are much smaller steps, and the minimum frequency over 22.05kHz that is contained in their material is up at around .5MHz. At this point the signal is actually turned into a 1 bit datastream, akin to DSD. Then it is converted to analog using "sample and hold". Then an analog low pass filter is put in at just below .5MHz that essentially "rounds" this small series of stair steps into the sinusoidal waveforms that we were looking for. This entire process is done in the D/A converter, and the quality of these filters is one of the differences between "adequate" and "great" converters. Quality is important because the digital filter that is used for the oversampling process can suffer from errors that will negatively affect your audio. This process is just as imperfect as the A/D conversion process.


This seems to contradict what you've posted on your legendary "96K..." thread, as well as on the DUC. As I understand (and based somewhat on your guidance), the so-called "staircase" waveform you referred to does not exist at the output of the DAC, because the reconstruction filter removes all frequencies above the passband (20K). Isn't the fundamental frequency of the staircase at the sample rate, which is 44.1?

Also, one of the things I seem to remember you stressing constantly on the "96K..." thread, was that it was impossible to create information that has not passed via the input. This can't be what you're referring to when you say, "...we 'oversample' the material, creating a bunch of dots between all of the dots..." because it would imply that you are creating new information. Am I simply mis-interpreting your analogy?

And I don't understand how the signal could end up on a 1-bit DAC, which with its 1-bit signal path restriction must be vastly inferior to a 24-bit PCM DAC.

Can you clarify? Thanks!

E \:\)


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Quote:
posted by synesthesia:
in a conversation with a Tascram (yes, tascam of all people) rep a couple of months back, he had mentioned a conference in western canada where a mathematician proved to sony a 3bit format would be more efficient, and yield higher fidelity. i'm not sure how this relates to two's compliment...


It doesn't. It relates to dither.

Again, Stan Lipshitz, Professor of Applied Mathematics at Waterloo University:
Quote:
A 1-bit system is inherently non-perfectible...You cannot properly dither without overloading the converter, because the correct dither (white noise with a triangular pdf and a width of 2 LSBs) covers the full range of the converter! Thus, as soon as you put in any input, the converter overloads. Of course the manufacturers don't put in 2 LSBs of dither, but then the converter is not correctly dithered and suffers from both distortion and noise modulation (and idle tones). This is why the state-of-the-art converters do not use 1-bit internally but rather a low number of bits, such as 3.


Eric \:\)


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Quote:
posted by Mike Bishop:
Transferring 16-bit 44.1 pcm recordings to DSD does not get you anything more than what you started with.


That is correct, yet deceitful: it will get you less than what you started with. 1-bit conversion of a 16/44.1 pcm recording will produce artifacts inherent of the limitations of 1-bit conversion. The 1-bit data path will introduce errors in the signal that cannot be removed, by any means, ever.

Mike, unless you can show me hard math that contradicts this, you stand corrected. Show me the math.

Eric \:\)


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[QUOTE]Originally posted by Curve Dominant:
Yes, but don't you still end up with all of the flawed audio inherent in putting it through a 1-bit Delta Sigma Modulator? That's one of the points I seem to be stuck on. Isn't the 3rd harmonic distortion from statistical noise modulation inherent from the DSM still there? And what about all of the HF noise generated, because we have a signal that has full scale peak to peak noise up to 1.4Mhz, correct? Won't that burn some tweeters if we don't filter it - which brings us to: what type of filters? Analog? And won't that roll-off audio as well...that can't possibly compete with DVD-A standards.

The point here is to put it through exactly the same signal path and exactly the same D/A conversion as PCM does once PCM is upsampled and reduced to a DSD type of data stream.

You have to remember that a D/A converter has (2) filters in it that function together to perform all of the filtering. One is digital. The other is analog. The digital filter is added during upsampling and reduction from PCM to DSD. The analog filter is added after the D/A conversion. both, however, are said to happen "in the D/A converter", and both are an integral part of the D/A conversion process.

The idea behind DSD as a delivery format is to take half of this filtering process out of the consumer's hands. While in the 96k thread we discussed the theoretical aspects of whether or not it was possible to design a perfect filter, in here we're discussing the real world applications, that we can't leave it up to hi-fi companies to design perfect filters, so we might be better off removing as much of that from their control as we can.

This is something else I don't get: What type of D/A filter? If it's 1-bit data going through a FIR filter: doesn't that just turn it back into multi-bit PCM, but with all of the flawed audio from the DSM in tact?

I don't think that this is how it works. It'd be nice if Mr. Frindle was around. We take PCM data and filter/upsample it in one fell swoop, then modulating it to a one bit datastream? So it's not 1 bit going through an FIR. It's 24 bits going through an FIR.

I still don't see how DSD is not PCM, except that it has to really bend over backwards and jump through hoops in order to maintain integrity as a 1-bit data stream. Which it can't anyway if it needs to be processed - "DSD Wide" is multi-bit PCM as far as I can see.

Well it kind of really IS just a really fast PCM data stream where we leave off a bunch of the bits - kind of.

Notice that I'm not addressing doing ANY processing in DSD. All I'm addressing is taking a 24 bit signal and upsampling and filtering it to the same rate as DSD, then turning it into DSD 1 bit and putting that on a disk for delivery. Then all the consumer's gear has to do is have a very gentle analog filter to roll off all of the high frequency stuff that we haven't gotten rid of yet - basically the last filter in a typical PCM D/A, but in this case it will be the only filter, because the first half of that filtering process was done at the mastering house.

Nika.

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Quote:
Originally posted by Curve Dominant:


That is correct, yet deceitful: it will get you less than what you started with. 1-bit conversion of a 16/44.1 pcm recording will produce artifacts inherent of the limitations of 1-bit conversion. The 1-bit data path will introduce errors in the signal that cannot be removed, by any means, ever.

Mike, unless you can show me hard math that contradicts this, you stand corrected. Show me the math.

Eric \:\)


Eric,

Allright, here's the math:

The process that the data goes through to become a 1 bit datastream is exactly the same process that the data goes through to become converted to analog from normal PCM anyway, namely: upsampling and filtering, then turned into a 1 bit signal. That's the same process as any PCM converter you have on your shelf.

So if there's going to be inherent loss because the upsampling/filtering algorithms aren't perfect, it happens to both your PCM datastream at the conversion AND EQUALLY to the PCM datastream that gets converted to DSD, only to be further converted to analog. This is because DSD is the halfway step of the conversion process.

Now, in the land of theoreticals, those filters can be made to be perfect, right? But in a practical world we may have issues with that. This is just a way of taking SOME of the conversion process out of the hands of the end users and putting it into the hands of mastering engineers who have VERY GOOD, or rather VIRTUALLY TRANSPARENT upsampling algorithms.

Nika.

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[QUOTE]Originally posted by Curve Dominant:

This seems to contradict what you've posted on your legendary "96K..." thread, as well as on the DUC. As I understand (and based somewhat on your guidance), the so-called "staircase" waveform you referred to does not exist at the output of the DAC, because the reconstruction filter removes all frequencies above the passband (20K). Isn't the fundamental frequency of the staircase at the sample rate, which is 44.1?


Yes, but the "reconstruction filter" is actually two filters - the digital and the analog working in combination with each other - one before conversion to analog and one after. That's all one big "reconstruction filter". I have attempted in this thread to dig into more of how that "filter" or "filters" work, describing each step of that filtering process and what the signal is at each step along the way.

Also, one of the things I seem to remember you stressing constantly on the "96K..." thread, was that it was impossible to create information that has not passed via the input. This can't be what you're referring to when you say, "...we 'oversample' the material, creating a bunch of dots between all of the dots..." because it would imply that you are creating new information. Am I simply mis-interpreting your analogy?

Hmm. No. We're not "creating" that information. The two dots rather "inherently imply" a waveform passing through them. We are simply further defining where that waveform passes by adding additional dots in it's path. That's different than "creating new information". Do you follow?

And I don't understand how the signal could end up on a 1-bit DAC, which with its 1-bit signal path restriction must be vastly inferior to a 24-bit PCM DAC.

Not at all. The 1 bit DAC is actually a part of the 24 bit PCM DAC.

I hope this helps. I'm going to bed now.

Nika.

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I follow the gist of what you're saying, Nika, and I do think SACD as a delivery format sounds appealing in this regard. I have a profound distrust for super-cheap digital. So much better to have a simple cheap analog only for the "consumer" piece. Even a very expensive reproduction system would be hard pressed to have the quality of D/A conversion a good mastering house has.
Ted


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Quote:
Originally posted by Curve Dominant:


I generally don't consume refined sugar or anything that contains it, so I couldn't tell you.

Rog,

Where did you find that jpeg you posted?

Any idea who created that demonstration?

E \:\)


Sorry Eric, I forget - it may have been from the Nuendo forum. Maybe you could find out by mailing someone at Merging?


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Quote:
posted by Nika:
The 1 bit DAC is actually a part of the 24 bit PCM DAC.


What??!!!

The 1-bit DAC is a part of WHOSE 24-bit PCM DAC exactly? Crystal? AKM? Burr Brown?

It is possible that there are some old, badly sounding, inefficient convertors floating around out there that used the 1-bit format. Most manufacturers are steering clear of 1-bit these days, as I understand, because of its inherent flaws.

C'mon, Nika, I thought you were the "go-to" guy on this stuff. Don't almost all of the latest generation of high quality convertors use multibit internal delta/sigma modulation techniques? As I understand, some very high quality ones actually use oversampled 24bit direct conversion.

Nika, at any rate, please let us know which convertor manufacturers are still using 1-bit DSM components, so we can all steer clear of that manufacturer and its convertors.

E \:\)


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Eric,

From my understanding (which is admittedly a little thin at this point in the conversation) there is an inherent benefit in multibit at the same sample rate, but the one bit DSD is at a much faster rate than multibit oversampled PCM, to the point that I think that the drawbacks don't compare.

8x oversampled, but multibit vs. 64x oversampled but one bit. Is there an advantage? I don't know. I was led to believe that they could be effectively the same.

Nika.

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Quote:
Eric,
From my understanding (which is admittedly a little thin at this point in the conversation) there is an inherent benefit in multibit at the same sample rate, but the one bit DSD is at a much faster rate than multibit oversampled PCM, to the point that I think that the drawbacks don't compare.
8x oversampled, but multibit vs. 64x oversampled but one bit. Is there an advantage? I don't know. I was led to believe that they could be effectively the same.
Nika.


Nika,

That trade-off only works until you get down to 2 bits, which is the number of bits you need to properly dither, and that's where you get into trouble in other areas. Yes, you can still juggle the numbers at 2-bit or 1-bit if you like, but if the correct dither (white noise with a triangular pdf and a width of 2 LSBs) covers the full range of the converter, the convertor overloads upon input. That's why, normally, when you hear of low-bit convertors, you hear of as low as 3-bit convertors, but usually not 2 or 1.

So, yes, the 1-bit convertor may have a higher sampling rate, but that's not going to help the other problems you get with a 1-bit DSM, such as the non-random cyclic HF noise, the resultant 3rd harmonic distortion, and the excessive HF pressure which, although is above the range of human hearing, will no less produce enough energy to burn the shit out of the tweeters in your monitors - all of these factors are beyond fixing with high sampling rates, and must be dealt with in other ways. And they are, no doubt, but in ways that significantly degrade the audio, and that is why the 1-bit convertor is considered obsolete today.

I hope this helps.

Eric \:\)


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Quote:
Originally posted by Curve Dominant:


Nika,

That trade-off only works until you get down to 2 bits, which is the number of bits you need to properly dither, and that's where you get into trouble in other areas. Yes, you can still juggle the numbers at 2-bit or 1-bit if you like, but if the correct dither (white noise with a triangular pdf and a width of 2 LSBs) covers the full range of the converter, the convertor overloads upon input. That's why, normally, when you hear of low-bit convertors, you hear of as low as 3-bit convertors, but usually not 2 or 1.

So, yes, the 1-bit convertor may have a higher sampling rate, but that's not going to help the other problems you get with a 1-bit DSM, such as the non-random cyclic HF noise, the resultant 3rd harmonic distortion, and the excessive HF pressure which, although is above the range of human hearing, will no less produce enough energy to burn the shit out of the tweeters in your monitors - all of these factors are beyond fixing with high sampling rates, and must be dealt with in other ways. And they are, no doubt, but in ways that significantly degrade the audio, and that is why the 1-bit convertor is considered obsolete today.

I hope this helps.

Eric \:\)


Eric,

This is not correct. Dither is not applied at this stage. Once the data is at this point dither has already been applied, so having enough extra bits at the tail end of the converter to dither prior to conversion to analog is unnecessary. I think you might be confusing DSD as a form of recording/editing where it falls weak specifically because of the dither problems within 1 bit signals, and DSD as a delivery format where the signal has already been dithered.

As for the other issues you mention with mostly HF problems in DSD signals, this is being dealt with with analog filters in playback equipment that are well above the hearing range such that they should have no effect on hearing, but low enough to remove the problems you're speaking of. Hutch from Manley addressed manufacturing concerns with equipment that is intended to play back HF DSD data, and I believe they design their equipment specifically to handle those issues accordingly.

I'm not here defending 1 bit converters - you don't seem in any way interested in engaging in conversation on the matter anyway. Your manner in this thread has become suspiciously hostile and agressive, mocking and attacking people who have attempted to answer your questions, and then faulting them for not taking more time to engage the conversation with you.

I'll remind you that you asked me a very simple question: "what is the supposed benefit of using DSD as a delivery format". I guess I'll just restate my answer and leave it at that. I'm not DEFENDING why it is valid. I'm merely presenting a possibility to you for the sake of your exploration. You did at least APPEAR to be asking these questions for the sake of exploring. I'm not convinced anymore that that was your motive.

The answer I was presenting to your question was "to remove the digital part of the reconstruction filter from the consumer's equipment, essentially giving them the data one stage later in the digital audio conversion process."

I'm going to leave this thread alone for now. I see no reason to be asked what appears to be a genuine question and give a very long and comprehensive answer to the nature that I believed I was being asked, only to be mocked and essentially "reprimanded" for giving such an answer. It seems you have also alienated and insulted others who were attempting to contribute as well. I hope you find the answers you're looking for at some point, if indeed you're looking for answers. I believe you're actually looking for something else.

I'll see you on the next thread,
Nika.

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Quote:
I'm going to leave this thread alone for now. I see no reason to be asked what appears to be a genuine question and give a very long and comprehensive answer to the nature that I believed I was being asked, only to be mocked and essentially "reprimanded" for giving such an answer. It seems you have also alienated and insulted others who were attempting to contribute as well. I hope you find the answers you're looking for at some point, if indeed you're looking for answers. I believe you're actually looking for something else.
I'll see you on the next thread,
Nika.


Before you go, do you mind explaining what is it, exactly, do you believe I'm looking for? I don't work for (or sell products of) any rival to Sony or Phillips, if that's what you're implying; I'm a freelance composer, curious about emerging formats. That's it.

Trying to get empirical data out of the pro-SACD camp can be a frustrating experience, as you have noted often yourself, Nika. What we've clearly established from that exercise is: "Super Audio" it's not. If I've been looking for an answer, and that's the answer, then yes, we've reached the point of redundancy.

I'm not here defending 1 bit converters - you don't seem in any way interested in engaging in conversation on the matter anyway. Your manner in this thread has become suspiciously hostile and agressive, mocking and attacking people who have attempted to answer your questions, and then faulting them for not taking more time to engage the conversation with you.

That is incorrect. What I faulted was a lack of specifics, amidst an over-abundance of rhetoric, but you know that, because you complained about that yourself earlier in this thread. Do I have to get crafty1 back here as a reminder?

As for the other issues you mention with mostly HF problems in DSD signals, this is being dealt with with analog filters in playback equipment that are well above the hearing range such that they should have no effect on hearing, but low enough to remove the problems you're speaking of. Hutch from Manley addressed manufacturing concerns with equipment that is intended to play back HF DSD data, and I believe they design their equipment specifically to handle those issues accordingly.

I don't remember seeing Hutch posting anything to that effect. Manley makes front-end processors, not consumer gear. And I don't see how analog filters will make it a competing product to PCM products. If the high frequency problems inherent in 1-bit signal are not inherent in well-designed PCM, why switch?

I'm not trying to "mock" you, Nika, I'm simply trying to understand how "super" the "Super Audio CD" really is. If you're saying that, theoretically, it could possibly work just as well as today's CD, fine! I'll take your word for it, and I apologize if I offended you, but maybe you shouldn't have taken this so personally.

On that note, if I suddenly start seeing SACD players in the Sweetwater catalogs, I'm going to get awfully suspicious! \:D

Eric \:\)


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A post-script to my previous post:

Nika called me here at the Curve Lab to remind me that, yes, there are some lucky souls who count Manley gear as components in their home audio systems. I have to point out, however, that if your home-system is not an all-Manley system (which it can't be because Manley doesn/t manufacture stereo speakers), any non-linearity in your system can reproduce aliasing error created by 1-bit conversion, even if those artifacts originally occurred outside of the passband.

Nika also asked me to tone it down a bit, which I will. \:\) Things appear harsher in print than they are originally intended; I have to improve my dithering ratio in that regard.

E \:\)


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George asked me to clarify something here:

Tom Jung was playing back some SACD material with the player's internal LP filters disengaged and ran into some "chirping" sounds coming through his Manley 50w monoblocks from all this unfiltered HF energy. This was because these amplifiers are basically flat to 200KHz and will play back the spurious high frequency noise that rings down. So we made some different external passive LP filters for Tom to play with and also had him try a 400pf cap in the amplifier's feedback circuit to achieve the high frequency roll off required in two different ways. That all worked. And worked well.

I find it goofy that we, as a playback gear manufacturer, have worked hard over the years to get better HF response out of our tube gear, and specifically our output transformers, but now we are being asked to roll all that achievement off because of some inherent problem with SACD and the players' manufacturers inability to make nice sounding filters...

So, ok. If someone needs one of our amplifiers to be 20dB down at 50KHz, we can comply with the request... or if someone needs some pretty inaudible external LP passive filters made, we have done that too.


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Thank you EveAnna for that cool bit, although I have to apologize for chuckling at this part (because I know it's not really funny, it kinda sucks actually):
Quote:
I find it goofy that we, as a playback gear manufacturer, have worked hard over the years to get better HF response out of our tube gear, and specifically our output transformers, but now we are being asked to roll all that achievement off because of some inherent problem with SACD and the players' manufacturers inability to make nice sounding filters...


Well, I just returned from my very first reliable experience A/B'ing SACD with Redbook CD, at a big name hi-fi store here in downtown Philadelphia which I will refrain from naming, in order to spare the store manager further humiliation...LOL!

I started out playing dumb: "So, why's SACD different?" First off, he couldn't properly pronounce "pulse code modulation," and mangled it a completely different way each time he attempted the phrase - that was funny enough. Then he told me that DSD is "pure digital: all ones and zeroes, PCM is not." So I asked him, "Pcm's not ones and zeroes?" He shook his head no, and said, "It's agalithms." I said, "You mean algorithms?"...this is the store manager, mind you, and it went on like this for some time, and my face was in excruciating pain trying to suppress an imminent explosion of laughter.

Finally a younger dude, who at least knew what a sampling rate was, walked over to save his boss from a complete massacre. Young dude had Miles "Kind Of Blue" on a SACD and Redbook (on one disc, that is), so into the isolated and treated room we went.

The player was a Sony (duh), I didn't catch the model number, but it was going for $1700. Speakers were Klipsch, $1500. Young dude puts on the Redbook first..."So What" opens (oh, what a gorgeous intro to a song), and we let it play through Miles' first solo. Then, same disc, same system, no settings changed, young dude made the switcheroo to SACD.

I could tell the difference, almost immediately, before Miles' solo.

The Redbook sounded much better and clearer, in my opinion. The SACD sounded mildly muffled and distorted in comparison, very subtly, mind you, but noticable. It was really obvious in the hi hat cymbles. The Redbook had a nice "tss tss" where the SACD was more "csh csh" and at a lower pitch (??).

The SACD had a noticably lower overall level; I thought young dude turned the volume down at first, but he swears he didn't touch anything. I could swear I heard the harmonic distortion characteristic of 1-bit, not just around the edges, but all the way through the mix. It was in the high range, however, where the difference was most obvious. The Redbook had details and nuances that were dampened in SACD.

The SACD had a little bit less dimension than the Redbook.

The Klipsch speakers sounded vry nice. I might get myself a set for the fouton room.

That's my report. The Curve will prepare himself some dinner now, and watch The News Hour with Jim Lehrer. Bye!

E \:\)


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In the magazine TAS (The Absolute Sound) the editor wrote in an article with the headline: The Truth about SACD ".... it is probably worth noting, at this point, that the DVD-A disc has the support of most of the major labels save Sony, Telarc, and fistful a minor labels, but so far there has been anything but unanimity (thatis, a standard) for the sounds coming of those disc.
And that leaves us with SACD, a differnetial PCM system (called, by the technofroids, delta-sigma), using one bit (more in actual practice) and an out-of -this-world sampling rate (more than 2 megahertz). To make this beastie work, there must be all sorts of noise-shaping applied to the signal, virtually all of it above the range of contemporary electronics.
What I´m saying, in short, is that the ultra-sonic, and quite noisy, SACD algorithms can (and to my ears, do) create perturbations further down, within the range of human hearing. (I think this is the source of the noisy artifacts I hear on the forte massed strings and brass in Telarc´s recent recordings of "Scheherazade" and Berlioz´"Symphonie Fantastique". Knowing well that there might be first-generation problems with SACD technology, and wanting to see SACD at its best, I asked to audition the two-channel Accuphase DP-100 digital deck and its DC-101 processor, which received a rave review in these pages and which costs some $ 28,000 dollars.
..... Beyond that, in direct A/B comparisons with the Burmester CD gear on Telarc´s dual layer CD-compatible SACD discs, the Accuphase came out a distant (read: Tibetan) second (Burmester 969, the playback deck, comes in at $ 27,930, and the 970, its upsamplig DAC, at $ 30,670).
The two-piece Accuphase set was colored in the same way one would hear if he crossed an old Conrad-Johnson tubed unit with something from Sonic Frontiers; that is, it sounded warm, a bit lush - especially mid-bass - and veiled, as if a gauzy scrim had been droped between the reproduced sound and the listener. The one area of superiority lay in the Accuphase´s reproduction of depth and space, characterisics we might have attributed to the fact that we were listening to two-channel versions of surround-sound originals. There was none of the Burmester´s exquisitely detailed high frequencies, none of its tonal neutrality, little of its wide dynamics, and none of its rock-em, sock-em deep bass. In other words, playing regular old 16/44 CDs, the Burmester shamed the Accuphase to a degree I would not have thought possible."

This quotation is from TAS issue 134, page 29.
I think we don´t need DSD for high resolution audio, maybe for cheap consumer players with cheap DA converters.

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Great thread. I don't understand a word of it, because I have no clue what DSD, dCS, and FPGA mean. For all I know -

- DSD is a popular new hallucinogen

- dCS is a recently-approved abortion procedure

- FPGA is a golf league for fat people

Judging from some of the posts I've read, I don't think I'm the only one who's confused. Could someone post a concise primer - PLEASE?


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Quote:
Originally posted by Dan South:
Great thread. I don't understand a word of it, because I have no clue what DSD, dCS, and FPGA mean.


Apparently it would mean nothing unless you have a stereo made by Manley that costs as much as the house I live in.

I think the primary thing you need to know here is the concept of a clash in the philosophy of increased signal resolution at a limited sample frequency versus an extreamly limited signal resolution at a ridiculous sample frequency - Direct Stream Digital. Right?

Personally, I don't know anyone who even has a DVD-A player, and only a few could explain what one is, much less a DSD setup....


Hmm.. Thinking... Isn't all of this out of context relative to the way speaker drivers work?

One would presume a high frequency driver is a non-perfect device, in that it's going to be "missing" harmonic detail in certain events captured by a certain sample rate, played back at a certain sample rate even in a perfect electrical amplification system. In other words, if one sample goes to digital nothing for 1 clock cycle and then back to full level, at 96K it's probably not going to "catch"/reproduce that accurately - whether it's periodic or not. In which case the driver would probably be more influenced by a cumulative "mechanical rounding off" of the signal, based on perhaps the average of what happens across 3 samples instead of 1.

Hmm. Really, drivers sort of act as a mechanical "frequency dither"; they're going to round off events that happen at ultra high sampling playback frequencies. They don't reproduce them, they just miss them or turn them into harmonic distortion if they can't react fast enough, right? They *can* respond with an infinite analog resolution within their operating bandwidth.

In which case the more accurate the sample is across that error range, the more accurate the reproduction will be. Increase clock frequency and I think you're getting diminishing returns mechanically since a driver isn't going to be able to keep up-ballistics wise, right? On the other hand, increased resolution of bit depth may yield a finer interaction with the mechanical dynamic of the driver since a minute averaged but more accurate fluctuation over a longer time span may yield a
better mechanical reproduction instead of less bit depth at a faster rate - 10 "blurry" samples in the same time span as 3 "detailed" samples. The 3 detailed samples would translate better than a whole bunch of less detailed samples that the driver can't discern routinely anyhow.

So ultra-high/higher sampling frequency without taking the mechanical means of reproduction into account doesn't make sense, while higher bit depth could translate more accurately since mechanical drivers are capable of "infinite" resolution within their operating bandwidth.

Right?

Man, I love making posts half asleep, always humbling when awake...


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Quote:
What I´m saying, in short, is that the ultra-sonic, and quite noisy, SACD algorithms can (and to my ears, do) create perturbations further down, within the range of human hearing. (I think this is the source of the noisy artifacts I hear on the forte massed strings and brass in Telarc´s recent recordings of "Scheherazade" and Berlioz´"Symphonie Fantastique". Knowing well that there might be first-generation problems with SACD technology, and wanting to see SACD at its best, I asked