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Help me put this guy back in his right place, Another SACD vs DVD-A debate
Joe Bloggs
post Sep 29 2003, 04:15
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http://www4.head-fi.org/forums/showthread....20&pagenumber=6

This theaudiohobby guys is really bugging me.
I'm not trying to say that SACD must be superior to DVD-A (since actually they're probably both 'perfect'), but this guy's justifications for SACD are ridiculous, starting from the page before this:

QUOTE
Originally posted by theaudiohobby
Joe,

most of your post is simply not so, but rather than get into what is and what is not read this arcticle from  optical-disc systems journal, remember that is a partially a critique of DSD and you will see that the experts give more due to DSD that you want to accept. DVDA v Bitstream article also look at this article from this dcs paper and this paper and see that there are valid scientific reasons why folk might prefer DSD to DVDA happy reading.


As you can see his links are irrelevant to or even against his argument. I'm especially irked by him using the dcs paper to say that they find DSD to be superior. Yes, they pointed out flaws for 24/192 (does anyone here know more about this test? Was the ADC/DAC combination faulty or something??) but they hadn't even made any observations on the corresponding performance criteria for DSD yet! For all we know it could be even worse!

And bit resolution? Time resolution? I'd say 24/192 trumps DSD in all these areas. Sampling at 2.8MHz doesn't mean anything when you have practically zero S/N ratio at 100kHz and above.
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listen
post Sep 29 2003, 08:10
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What about this paper?

From what I understand of the format, it is presenting the signal in a similar manner to something that might happen inside a very cheap CD player. It (partly) makes up for it with sheer bandwidth, so it could be worse... but it just seems like an such an awful waste considering that PCM can do so much more with less.
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NumLOCK
post Sep 29 2003, 10:12
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QUOTE (Joe Bloggs @ Sep 29 2003, 04:15 AM)
As you can see his links are irrelevant to or even against his argument. I'm especially irked by him using the dcs paper to say that they find DSD to be superior. Yes, they pointed out flaws for 24/192 (does anyone here know more about this test? Was the ADC/DAC combination faulty or something??) but they hadn't even made any observations on the corresponding performance criteria for DSD yet! For all we know it could be even worse!

And bit resolution? Time resolution? I'd say 24/192 trumps DSD in all these areas. Sampling at 2.8MHz doesn't mean anything when you have practically zero S/N ratio at 100kHz and above.

I just love this dcs paper...

QUOTE
96kS/s, 24-bit: "some stereo image formation"


Oh damn, all my audio cd's are mono then ! biggrin.gif

QUOTE
192kS/s, 24-bit: "bass can appear light and slightly out of time"..    "stereo image can be strong but widened (1.5 times)"


I think these sentences are so ridiculous that they're not even worth arguing laugh.gif

In my opinion switching from PCM (96+ kHz) to DSD is a bit like going back from weighted number systems to Roman numbers smile.gif

A bit like saying MMMMMMCCXXXXIV instead of 6244... oh well, as long as it can simplify a $10 DAC design rolleyes.gif


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KikeG
post Sep 29 2003, 10:56
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I don't even buy what some of these dCS papers say. Again, they rely on non-rigorous (non-blind) anecdotal evidence, and on "possible" audible effects, again not confirmed by blind testing.

They look more as hi-res marketing than as anything else.

This post has been edited by KikeG: Sep 29 2003, 11:02
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2Bdecided
post Sep 29 2003, 11:07
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QUOTE (NumLOCK @ Sep 29 2003, 09:12 AM)
QUOTE
192kS/s, 24-bit: "bass can appear light and slightly out of time"..    "stereo image can be strong but widened (1.5 times)"


I think these sentences are so ridiculous that they're not even worth arguing laugh.gif

Before you laugh, remember one thing: DCS sell these things, and in this paper, they're basically saying that none of them sounds perfect.

I ask you to seriously consider the possibility that they are simply reporting what they hear, in the interests of scientific research.


They have no reason to lie, to invent this, or even to imagine it. I believe they sometimes (but not always?) use blind testing. Whichever this was, the placebo effect doesn't usually make the "better" thing sound worse!


btw - Joe Bloggs - you're wasting your time. The SACD vs DVD-A vs CD vs analogue thing will be scientifically sorted out, eventually. But it hasn't been yet, and for now, SACD is winning the marketing war with audiophiles. I think this is largely because SACDs are carefully mastered, and the ultra-sonic noise has an interesting effect on people's equipment. There are probably genuine audible differences between the formats, but the differences people hear in the home are probably mostly due to other things.


Cheers,
David.
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KikeG
post Sep 29 2003, 11:16
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QUOTE (2Bdecided @ Sep 29 2003, 11:07 AM)
remember one thing: DCS sell these things, and in this paper, they're basically saying that none of them sounds perfect.

Which is just not believable, in my opinion.

Also, from that reasoning, it's unavoidable to derive that the higher the format resolution, the better the sound. So they are clearly pushing high-res formats as better sounding, and those formats are what they are more interested in selling now.

This post has been edited by KikeG: Sep 29 2003, 11:24
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2Bdecided
post Sep 29 2003, 12:14
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QUOTE (KikeG @ Sep 29 2003, 10:16 AM)
QUOTE (2Bdecided @ Sep 29 2003, 11:07 AM)
remember one thing: DCS sell these things, and in this paper, they're basically saying that none of them sounds perfect.

Which is just not believable, in my opinion.

I thought you'd say that! I thought you'd pick me up on "There are probably genuine audible differences between the formats" too!


QUOTE
So they are clearly pushing high-res formats as better sounding, and those formats are what they are more interested in selling now.


Read the paper - better than CD, but not transparent. It's hardly a sale pitch! Which, to my logic, makes it all the more believable.

Cheers,
David.
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M
post Sep 29 2003, 12:18
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QUOTE (2Bdecided @ Sep 29 2003, 05:07 AM)
There are probably genuine audible differences between the formats, but the differences people hear in the home are probably mostly due to other things.

Amen, David... such as that little button (or buttons, on some models) that let you choose between the internally preset EQs! My wife and I spent a good chunk of the weekend comparing models, and she had all sorts of fun going from "Concert Hall" to "Dolby ProLogic II," and everything in between. And there I was, standing beside her, trying to explain to her that the "2-Channel Stereo" mix sounded more like the original CD - yes, that's right, the demo disc which so fascinated her was a standard 16bit/44.1KHz CD - than any of the others. Silly me. >_< (No sympathy... I love her more than life itself, and have plenty of opportunities to listen to my own music in the original mix, so it's not as harsh a sacrifice as it sounds. No pun intended. biggrin.gif )

- M.
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KikeG
post Sep 29 2003, 13:18
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QUOTE (2Bdecided @ Sep 29 2003, 12:14 PM)
Read the paper - better than CD, but not transparent. It's hardly a sale pitch!

??? It's clearly a sales pitch. According to them, 24/192 or SACD are the best, and CD the worst. They use a typical "audiophile-style" argument: there's always something better, no matter how good what you have now. So you can always buy something better. According to them, it's possible that 24/384 is not transparent either, which is just absurd.

Back to real world, if 24/192 is not transparent, then *nothing* is transparent. There's no analog audio playback format that is remotely close to 24/192. Not to say that there are no real-world mics or speakers that can cope with it adequately either.


Edit: moved addendum to next post.

This post has been edited by KikeG: Sep 29 2003, 14:36
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2Bdecided
post Sep 29 2003, 14:03
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QUOTE (KikeG @ Sep 29 2003, 12:18 PM)
QUOTE (2Bdecided @ Sep 29 2003, 12:14 PM)
Read the paper - better than CD, but not transparent. It's hardly a sale pitch!

??? It's clearly a sales pitch. According to them, 24/192 or SACD are the best, and CD the worst.

Not the 1997 aes97ny.pdf, but the 1998 effects.pdf.



QUOTE
Back to real world, if 24/192 is not transparent, then *nothing* is transparent.


Nothing is more likely. Things can be perceptually transparent for many people and many signals. But in engineering, nothing is perfect. Things can, at best, be "good enough".

I know you know that, and that you meant "perceptually transparent". I was just being picky. It's just worth remembering: even at 24/192, it would be trivial to show that the signal coming out of the A>D>A process was different from the one going in. But probably not using your ears!

QUOTE
There's no analog audio playback format that is remotely close to 24/192. Not to say that there are no real-world mics or speakers that can cope with it adequately either.


But there's no analogue equipment with brick-wall filters, or that kind of time-domain ringing either. I'm not convinced that it's audible, but simple intuition says that it's not a good thing to have it flying around an audio system either.


We're having this discussion again, aren't we?

And no, I couldn't hear anything in the ringing test!

Cheers,
David.
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KikeG
post Sep 29 2003, 14:30
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QUOTE (2Bdecided @ Sep 29 2003, 02:03 PM)
But there's no analogue equipment with brick-wall filters, or that kind of time-domain ringing either.

It depends. Some analog parts of most record/playback chains will have glentle (or not so gentle) rolloffs below the ringing frequency of a 96 KHz or 192 KHz sampling system. What importance does some subtle, occasional, ringing at 40 KHz (on a high-res system) have, if your speakers or mics roll off quite below 30 KHz?

And of course, all discussion is related to audibility. If not, we wouldn't talking about audio.


Edit:
More about dCS papers: if dCS people were truly interested in scientific research, they would at least have presented some DBTs that supported their claims. Those DBT results shouldn't be difficult to obtain, if the sonic differences were so clear, as they suggest. Also, those results would make a world of difference in the credibility of their claims. But still, they have not presented them. The same old story we've heard so many times.

AFAIK, so far nobody has been able to prove such claims by means of a DBT, not even to prove that well-implemented, plain-old cd-audio is sonically distinguishable form high-res formats, when playing regular music.

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Joe Bloggs
post Sep 29 2003, 15:03
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David: I may be wasting my time, but the way this guy fudges technical terms to support his POV is ridiculous and he must be eradicated mad.gif

More from that guy

QUOTE
Joe and czilla9000,

read the papers before we discuss further at the rate you guys are going my response will probably be good enough for an AES paper. 


suffice to say that the dcs paper does say this



quote:
--------------------------------------------------------------------------------

detailed comparisons not yet performed on enough systems, but well
liked by (classical) artists after sessions
no busy signal break up
very good separation of reverberation and room acoustics
no observations on bass so far
strong stereo image formation, no observations on width so far

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



The philips response also says this


quote:
--------------------------------------------------------------------------------

10. The impulse responses of 4 different systems in a multichannel configuration are depicted: a 48 kHz system, with a bandwidth of 20 kHz (that is, 8 kHz transition bandwidth is allowed for anti-aliasing filtering), a 96 kHz system with 35 kHz bandwidth (26 kHz transition bandwidth), a 192 kHz system with 75 kHz bandwidth (42 kHz transition bandwidth) and an SACD system with 95 kHz bandwidth. Though none of the systems reproduces the input exactly, the DSD systems shows the least artifacts. Clearly, the 48 kHz system has great difficulty in reproducing the click; due to the steep filtering it starts ringing at a -30 dB level approximately 1 ms before the click, which is very audible. Also at the higher sampling frequencies, the ringing phenomenon cannot be removed, though it is reduced significantly. Only the DSD system is very effective in suppressing the ringing effect, due to very slow filtering above 95 kHz. The price to pay for this is the increase in noise floor with respect to the other systems; however, as the noise floor contains only high frequency components which are uncorrelated with the audio, they are not perceptible.

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



and that correlates exactly with the main decenting paper I gave you that says


quote:
--------------------------------------------------------------------------------
An alternative bitstream converter has been reported that uses a combination of a linear quantizer with dither to guarantee linearity, together with 4th-order noise shaping. Conversion to a 1-bit code (typically from a 4-bit code) is then performed by an open-loop, optimal code conversion table which minimises spectral modulation. Potentially the system produces high resolution with no low-level correlated distortion. However, the bit rate is again very much greater than PCM, and although solving the problems of correlated and idle-channel distortion it is too bit inefficient.
--------------------------------------------------------------------------------



and philips answer to their objection is this


quote:
--------------------------------------------------------------------------------

Other issues which often appear to be confusing, are data rates in connection to the bandwidth claimed by SACD. The SACD format comprises (apart from its red-book conforming CD layer) two different music streams: a stereo 2 channel stream, and a surround 6 channel stream. Hence, an SACD contains 8 channels of high-quality audio. Because all channels are 2.8 MHz sample rate, 1-bit signals, the total data rate equals 2.8 Mbyte/s (or 22.6 Mbit/s). On these signals, lossless coding is applied. This lossless coding scheme is specifically developed for coding 1-bit signals. From experience of over 100 recordings, the average coding gain is roughly 2.4 - 2.5 for pop recordings, and 2.6 - 2.7 for classical recordings. This corresponds to a data rate per channel of about 1.1-1.2 Mbit/s. This indicates that on average 70 minutes of a DSD signal can be recorded on an SACD in the 8-channel format. For 6 channels, this amounts to roughly 95 minutes. Also, the high sampling rate of DSD allows for the use of filters with slow roll-off. We can compare this to DVD-A. The DVD-A format that gets closest to the SACD characteristics is DVD-A at 192
kHz, 20 bit, which reaches the same dynamic range, but is either of lower bandwidth than SACD if sloppy anti-aliasing filters are used, or has the same bandwidth using steep filters. Using a compression factor of 2 the data rate amounts to 1.9 Mbit/s, which is almost twice as much as the data rate for DSD. Hence, even if only six channels are used on the optical disk (compared to 6+2 on SACD), only 55 minutes of music can be stored - much less then the 74 minutes that we are accustomed to from CD.

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



I will return with some detailed answer to objections when the need arises. but as it is stands, it is easy to see here that SACD has a few aces up its sleeve that DVDA as it stands simply cannot match.


The responses from philips are interesting, although I have no idea where he pulled them from (!) So SACD has a higher compression rate than DVD-A now? ohmy.gif sad.gif Better impulse response?
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KikeG
post Sep 29 2003, 15:36
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Tell him some uncontrovertible facts:

- 24/192 DVD-A has much better dynamic range than SACD. DVD-A has 24-bit performance (144 dB SNR) up to 96 KHz, whilst SACD has the equivalent of just 20-bit performance (120 dB SNR) just up to 20 KHz, and quite worse performance over this frequency. Over 50 KHz the signal is very noisy, in fact Sony recommended to filter SACD output over 50 KHz.

- It's impossible to totally avoid quantization distortion in SACD. Due to its 1-bit nature, it's not possible to adequately dither it.

It's true that SACD may have better impulse response due to its higher sampling rate. However, instead of having higher ringing just in the proximities of high-frequency impulsive signals (as DVD-A has), it will have relatively high levels of a mixture of distortion and constant noise at these same frequencies where DVD-A ringing occasionally appears.
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Joe Bloggs
post Sep 29 2003, 16:45
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Is there an exact dynamic range profile for DSD?

Can't totally avoid quantization distortion? Can't you just make the dither range fullscale? tongue.gif

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2Bdecided
post Sep 29 2003, 16:47
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There's a sense in which they're trying to have their cake and eat it.

They state that ringing around 96kHz will be audible, but noise above 95kHz will not be. Now look, either audio signals above the generally accepted hearing limit are audible in some way, or they're not. If they're not, let's stick with CD. If they are, then we'd better have a much better understanding of what's involved before we start making such bold claims!



"Clearly, the 48 kHz system has great difficulty in reproducing the click"

Yes, and the human ear will have great difficulty in hearing most of it, too!


Be very careful with these comparisons. It's easy to do one or more of the following:

1. average the data many times. The "ringing" on DVD-A adds up, the noise on SACD cancels.
2. Use a gentle filter on the SACD signal which makes the noise just below the width of one pixel on a linear time-domain plot. This noise is about 20dB down.


But there's no point waging a holy war over this. For all we understand at the moment, either SACD or DVD-A should sound fine. The reasons quoted why SACD isn't good enough ("imperfectible") are very true, but the problems can be minimised to the point where they don't really matter. Also, this time domain smearing in DVD-A (and, more so, CD) has not been shown to be a cause of audible problems.

No matter how many times they try to write "time domain ringing/smearing" in the same sentence as the word "audible", this is just conjecture.


At the end of the day, the technical arguments will be irrelevant. They're not what will decide the format war. I too find it very annoying that there is so much false and misleading science used to sell SACD, but until more work is done, you can rebut it by saying why it's wrong (or at least, unproven/unjustified), but can't provide anything better.

Cheers,
David.
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2Bdecided
post Sep 29 2003, 16:49
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QUOTE (Joe Bloggs @ Sep 29 2003, 03:45 PM)
Is there an exact dynamic range profile for DSD?

Can't totally avoid quantization distortion? Can't you just make the dither range fullscale? tongue.gif

No, it depends on the DSD modulator. You can make it equal to almost anything you like. It depends on how aggressively you want to push the noise into the ultra-sonic region.

20-bit (i.e. 120dB) to 20kHz is possible (but not always used, I might add). But there's no useable hard fixed digital full scale in DSD either - it just goes unstable above a certain input, so you stay at least 6dB below the theoretical 0dB FS.

Cheers,
David.
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Joe Bloggs
post Sep 29 2003, 17:03
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QUOTE
No, it depends on the DSD modulator. You can make it equal to almost anything you like. It depends on how aggressively you want to push the noise into the ultra-sonic region.


How does it alter that? I'd like to learn more about noise shaping. You've also left behind a very confused crowd in my last thread about DSD. http://www.hydrogenaudio.org/forums/index....howtopic=13450&. Help us. tongue.gif

Found this from here: http://www.hometheatermag.com/hirezaudio/

QUOTE
All digital signals begin as 1-bit representations of the analog waveform after they're converted by 64x-oversampling Delta-Sigma modulation. Whereas the PCM system then sends the signal through a series of filters, which can cause audible problems during recording and playback, DSD records the 1-bit signal directly and eliminates several steps of the record/playback process.


True? False? If that's indeed the first recording medium I know it can't be only 64x oversampling biggrin.gif And what is the process to transform delta-sigma into PCM? Can it be lossless?

(I guess probably not, otherwise there wouldn't be all that fuzz about mastering in DSD. Just master in PCM and then release as SACD.)

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2Bdecided
post Sep 29 2003, 17:47
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Noise shaping is like negative feedback. It can be modelled as negative feedback. You look at the output, look at what you don't want, and subtract this from the input - hopefully correcting the output "next time around" as it were.

The reason I didn't come back and explain it is because it doesn't make that much intuitive sense to me, but the result falls out of the equations nicely. I can't type the equations, and I can't scan the diagrams, so I thought you'd get on better with a web search. Unfortunately, you've found a marketing based site, rather than an engineering based site.

I've tried google, and it hasn't helped me either. I think you've been here:
http://www.cs.tut.fi/~rosti/1-bit/
and I don't think it's too helpful.


Assume you have a signal, and you quantise it. Quantise = round it to some value. 8=bit audio has 255 possible values. 1-bit audio has only 2: 1 or 0.

The rounding introduces an error - noise or distortion. When the signal is big, and the error is small, then the error sounds like noise, and fills the audio band equally. When the signal is small and the error is (relatively) large, then it sounds like distortion, at harmonics of the signal frequency.

If you put negative feedback (i.e. noise shaping) around the quantisation process, then it will attempt to correct it's own faults. Of course, it can't correct them - you'll always have exactly the same amount of noise or distortion, because you have to round. However, if you put a filter in the feedback loop, then what gets through the filter is removed from the error. So, you let the audible error through, subtract it from the input, and you now have less error in the audible band, but twice as much (say) above the audible band.

In DSD, it's almost all error. The signal is much smaller than the error. But you have a tiny signal band (0-20kHz), and a huge ultrasonic band (20kHz-1.4MHz), so you can push the (huge) error down very severely in the audible band - there's plenty of room to push it into the ultrasonic region.

QUOTE
How does it alter that?


The filter in the feedback loop is altered. If it lets through ten times as much signal in the audio band as in the ultrasonic band, then there will be 1/10th as much nosie in the audio band as in the ultrasonic band (remember - what gets through the filter is subtracted, but the equivalent amount of energy is moved into other frequency regions). The limit is where the ultrasonic region is saturated, and/or the whole thing goes unstable. Because it's a loop, if you get it wrong, it can make a sound even when there's no input. Like when you put a microphone in front of a speaker which is playing the output of that microphone.


(but remember - In DSD, it's classic negative feedback, but similar - I'm struggling to explain this, and I'm hoping you might have looked at op-amps, negative feedback, and filters at some time in physics at school or something to help with this - if not, I'm probably not helping at all!)


QUOTE
All digital signals begin as 1-bit representations of the analog waveform after they're converted by 64x-oversampling Delta-Sigma modulation. Whereas the PCM system then sends the signal through a series of filters, which can cause audible problems during recording and playback, DSD records the 1-bit signal directly and eliminates several steps of the record/playback process.


Marketing BS. SACD fans want you to think that all A>D and D>A convertors are basically 1-bit convertors. Well, some are, some aren't. The best aren't. (The best were when SACD was invented, which is why we have it, but this is no longer true!).

Interestingly, before SACD, 256x oversampling wasn't uncommon. They use 64x oversampling on SACD, because 256 gives too high a data rate. But people used 256x, even for D>A conversion of CD quality material because it was measurably better than 64x. Draw your own conclusions. Hint: SACD is compromised, though probably still good enough. 256x 3 or 4 bits would be better, if they're guesses and theory is correct. If they're not, then it's overkill anyway.


QUOTE
And what is the process to transform delta-sigma into PCM?

1-bit > 16-bits. 20kHz low pass filter. Keep every 64th sample, throw away the rest. (there are more computationally efficient methods!!!).

QUOTE
Can it be lossless?


Of course not. You lose everything above half the destination sample rate. Whether this is audible or not...


QUOTE
(I guess probably not, otherwise there wouldn't be all that fuzz about mastering in DSD. Just master in PCM and then release as SACD.)


There are a lot of SACDs sourced from 24/96 masters. And some SACD fans claim they sound better than the 24/96 versions. Draw your own conclusions. Hint: SACD can't add anything useful, but there's always distortion, placebo, poor DVD-A players etc etc etc.


I hope this is some help.

I still think you're wasting your time trying to convince him!

Cheers,
David.
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Joe Bloggs
post Sep 29 2003, 18:06
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Re: noise shaping, I have this page. There's a section on noise shaping. I've found the rest of the article quite useful, so I might find this useful too. Unfortunately I don't know what the terms in the equation stand for. E.g. x(n): function varying with frequency? Or varying with sample number?

Edit: definitely sample number. Give me a few days, I may be able to sort this out yet... tongue.gif Especially with you help in giving the equations some intuitive background to place it on smile.gif

Edit: u ( n ) = x ( n ) + ( y ( n ) - u ( n ) ) * h ( m ), huh? How can you have u(n) on both sides of the equation? (And I've done my share of programming-- i=i+1 wink.gif but I still can't figure out exactly what this stands for ohmy.gif )

Edit: Is it that the whole noise shaping process is being treated as an offline process here, where u(n), y(n) etc. represent the entire audio clip; so this step takes the quantized output y(n), puts in on the RHS, calculates the difference between the quantized waveform y(n) and the unquantized waveform u(n), filters the difference waveform by convolving with h(m), and then adding it back on x(n) to obtain the *new* u(n), and try quantizing again with the new waveform? And 2nd order noise shaping would be taking the quantized version of that new y(n), calculating y(n)-u(n) again, filtering, etc.?

That sounds like a plausible enough way to shape the noise. Now to figure out how to do this in realtime. laugh.gif -> wacko.gif

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KikeG
post Sep 29 2003, 19:24
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QUOTE (Joe Bloggs @ Sep 29 2003, 04:45 PM)
Can't totally avoid quantization distortion? Can't you just make the dither range fullscale? tongue.gif

No. In fact, the amount of dither applicable before unstability is quite sub-optimal. It's explained at the link at the 2nd. post of the thread.
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Pio2001
post Sep 29 2003, 23:00
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QUOTE (Joe Bloggs @ Sep 29 2003, 05:03 PM)
quote:
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10. The impulse responses of 4 different systems in a multichannel configuration are depicted: a 48 kHz system, with a bandwidth of 20 kHz (that is, 8 kHz transition bandwidth is allowed for anti-aliasing filtering), a 96 kHz system with 35 kHz bandwidth (26 kHz transition bandwidth), a 192 kHz system with 75 kHz bandwidth (42 kHz transition bandwidth) and an SACD system with 95 kHz bandwidth. Though none of the systems reproduces the input exactly, the DSD systems shows the least artifacts.

They mean this ? http://forum.cdfreaks.com/showthread.php?s...1864#post376461
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Joe Bloggs
post Oct 1 2003, 06:39
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2BDecided: so how is my understanding of noise shaping going? huh.gif

To everybody else: what is linearity error and what has it got to do with jitter and choosing between SACD and DVD-A, if anything? There's this dCS Elgar SACD DAC thingo and it has this great measurement in the linearity error category and theaudiohobby is using this to push the superiority of SACD.
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KikeG
post Oct 1 2003, 10:03
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QUOTE (Joe Bloggs @ Oct 1 2003, 06:39 AM)
To everybody else: what is linearity error and what has it got to do with jitter

It has nothing or very little to do with jitter.

QUOTE
and choosing between SACD and DVD-A, if anything?


Nothing either. there's no reason why a good DVD-A DAC can't achieve good linearity too.

QUOTE
There's this dCS Elgar SACD DAC thingo and it has this great measurement in the linearity error category and theaudiohobby is using this to push the superiority of SACD.


It doesn't say at anywhere that a good 24/192 DAC can't achieve same or better performance.

This post has been edited by KikeG: Oct 1 2003, 10:06
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2Bdecided
post Oct 1 2003, 11:46
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QUOTE (Joe Bloggs @ Oct 1 2003, 05:39 AM)
2BDecided: so how is my understanding of noise shaping going? huh.gif

To everybody else: what is linearity error and what has it got to do with jitter and choosing between SACD and DVD-A, if anything? There's this dCS Elgar SACD DAC thingo and it has this great measurement in the linearity error category and theaudiohobby is using this to push the superiority of SACD.

The dCS Elgar plus is a multibit DAC!

5-bits, 64x oversampling, to be precise.
http://www.dcsltd.co.uk/Elgar.htm

Anyone who uses the (excellent) performance of the DCS Elgar as some kind of justification for the superiority of SACD doesn't know what they are talking about!


DVD-A, correctly dithered, in infinitely linear. The dCS Elgar manages linearity down to the equivalent of the 27-th bit level with a DVD-A source.

SACD, as a format, is not linear, as proven in the Lipshitz and Vanderkooy papers at the start of this (or the other?) thread. I haven't seen results which conclusively show the level of the non linearities of a real SACD disc, played through the elgar Plus. I suspect they'd be worse than DVD-A. The Stereophile result looks like it is (linear down to the equivalent of the ~22nd bit), but it's swamped by noise, so this isn't a fair comparison.


Did I mention that you're wasting your time? :-) When you've dismissed all "technical" arguments, he'll just say "it sounds better to me", and you can't argue with that!


btw - noise shaping is an active (feedback!) process. You can't do it in the way you suggest - that's not feedback, that's subtraction. In a feedback loop, the output affects the next input, which affects the next output, which affects the next input, which... ! You don't get that with a single subtraction because the result of the subtraction on this sample can't affect the next sample.

Cheers,
David.
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Joe Bloggs
post Oct 1 2003, 15:20
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QUOTE
btw - noise shaping is an active (feedback!) process. You can't do it in the way you suggest - that's not feedback, that's subtraction. In a feedback loop, the output affects the next input, which affects the next output, which affects the next input, which...


Then I'm back to this question sad.gif

Edit: u ( n ) = x ( n ) + ( y ( n ) - u ( n ) ) * h ( m ), huh? How can you have u(n) on both sides of the equation? (And I've done my share of programming-- i=i+1 but I still can't figure out exactly what this stands for)

Especially as u(n), x(n) etc. really seem to stand for a whole waveform instead of just one sample?

And my interpretation does have 'feedback'--the output waveform affects the next input waveform, which affects the next output waveform, which affects the next input, which... biggrin.gif

How can you convolve a single sample with a filter? wacko.gif

I know my interpretation won't work as a real-time process as it is, but is it workable for an offline process? :x

QUOTE
Did I mention that you're wasting your time? :-) When you've dismissed all "technical" arguments, he'll just say "it sounds better to me", and you can't argue with that!


True ph34r.gif, but I feel like there is more at stake than what his final beliefs are; this thread is also going to sway the beliefs of the rest of the population in the forum. >_<

This post has been edited by Joe Bloggs: Oct 1 2003, 15:28
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