[theaudiohobby]

I have read Jazz comments and the paper is not valid because DSD uses 7th order noise shaping and the dissenting party have already answered that question themselves by saying this

Quote:

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.

The issue at stake is whether is you could implement a real-life system that did that whilst address the objections previously stated in another post. and philips say this

Quote:

Hence, it is clear that DSD as a carrier provides a signal bandwidth which exceeds 100 kHz by far. Obviously, the SNR decreases for high frequencies. As far as experimental evidence shows, the human auditory system is sensitive to ultra sound only if it is correlated with the signal in the 0-20 kHz band, hence, the increase in noise at high frequencies is not important. The excess in bandwidth can subsequently be used to allow for very slow low-pass filtering, which keeps the time response very accurate. Indeed, filtering must still be applied to prevent tweeters in sound reproduction systems from overheating. This can be achieved easily with second order filters, by which the level of high frequency noise is reduced by approximately 45 dB if the corner frequency is chosen as 100 kHz.

Those issues have been comprehensively addressed see the this dcs newsletter that says

Quote:

It is then also possible to choose a less rigid filter, since aliasing distortion then occurs at a frequency higher than the human hearing spectrum, and it can be argued that the distortions in this area have less audible effect. With those "mild" filters, impulse response is improved a lot though. In this higher sampling rate "trade off", DSD comes off particularly well, as with a sampling rate of 2.8MHz, the filter can be very gentle indeed.

Of course ,there are trade offs and you can read them in the paper.

Now read this statement

Quote:

if frequency response were the only issue, there would be no advantage in moving formats with higher sampling rates. However, the evidence is otherwise. Direct comparisons of the same source material, recorded and reproduced at 44.1 kS/s, 96 kS/s and 192 kS/s show that there is an advantage in going to the higher rates - it sounds better! The descriptions of those used to making such comparisons tend to involve such terms as “less cluttered”, “more air”, “better hf detail” and in particular “better spatial resolution”. We are left wondering - what mechanism can be at work? It seems unlikely that we have all suddenly developed ultrasonic hearing capabilities.... One can get oneself into a bit of a twist thinking about the energy in the ringing. After all, if it is in the audio band, allowing extra energy at higher frequencies through the system surely cannot cancel out some that is in the audio band? It does, though - so although we may not be able to hear energy above 20 kHz, its presence is mathematically necessary to localise the energy in signals below 20 kHz, and it is possible (and our contention) that we can hear its absence in signals with substantial high frequency content. A high sample rate system allows it through (fact) - and allows the high frequency signals to sound more natural (contention) but allowing better spatial energy localisation (fact).

In relation to DVDA, there some issues quoted that would probably be surmised as a digital, see this quote.

Quote:

The two highlighted effects with 192 kS/s are interesting. The bandwidth of 192 kS/s far exceeds the normal bandwidth attributed to human hearing, even using gentle roll off filters, so it is hard to believe it is related to frequency response. The stereo image widening is a very strong effect, observed by virtually all listeners, when a comparison with the analogue source is available. The bass problem is commented on by experienced listeners, and shows up particularly well on multi-mic’d rock music – it can enable 192 kS/s material to be identified in absence of the analogue source material for comparison.

That quote IMHO is alluding to digital sound because it says that the listeners could identify DVDA 192/24 even in the absence of the analogue source, of course I am sure you could apply some DSP to correct it, but it does not occur at all in DSD. Secondly, some of best turntables reports actual dynamic range at 80- 82db, given them bit depth of <15 bits and these are reference quality tables, why are they deemed to sound better than some digital systems, if you do some study the is issue clear, it is superior timing, to borrow a quote from Roksan 'Function of a record player is to measure the groove with respect to time.The more accurate is this measurement, the better is the performance of the record player.", Could it just be that poor time resolution has been the unravelling of CD?

 

[Czilla9000]

Joe, when I said "treat each other with respect" I meant between you and audiohobby. You guys were starting to take digs at each other.



Also, no sarcasm intented, I would like to see an unbiased report on the suppiority of DSD. I would have nothing against DSD if it were proven to be better than DVD-A or heck, even better than the CD.


Look, I own a lot of DSD mastered CDs. I want to like SACD very badly, but I have to listen to reason first.

The Philips paper is interesting, but obviously there is a conflict of interest going on.


EDIT: The Philips dude said repetedy that SACD "is a consumer format". One of the things the Philips guy points out alot is that SACD can store more music than DVDA, making it a better consumer format.

 

[Joe Bloggs]

I insist the glass is half full because it is half full.

Quote:

Recording engineers, and many musicians – particularly in the classical area – are becoming aware that material recorded and edited using these higher sample rates has some attractive qualities. Current theory on how human hearing works has so far been unable to explain the basis for these qualities, but they are none the less easy to demonstrate.

Higher means 'higher than 44.1kHz', not 'the higher the better', especially not when at 24/192 you can capture all frequencies up to 96kHz with 144dB dynamic range and with the '2.8MHz' DSD signal you only have 120dB dynamic range starting from 0Hz down to next to nothing at 96kHz.

Quote:

blah blah blah

NO OBSERVATION means NO OBSERVATION. From the paper how do you know how long they listened to 24/96, 24/192 and DSD respectively?

I can assure you, the results from DVD-A are 'very good' too.

Re: impulse response of DVD-A vs DSD, here's someone's response from another forum:

Quote:

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.

You might also want to check out this thread to see how skeptical more technical people are regarding the claims in the dCS paper.

Now to rip the Philips paper apart...

Quote:

minimal frequency span needed to comply with the human auditory system appears to be roughly 0-350 kHz (roughly equal to 8.1 kHz).

What? So is it 350kHz or 8.1kHz? And how do they justify this claim when we're demonstrably unable to hear above 20kHz?

Quote:

It is important to note, however, that all artifacts which occur in the base band can be made almost arbitrarily small by increasing the order of the sigma delta modulator.

Not so.

"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."

"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."

This image, taken from the philips paer, proves the point that any signal processing for DSD must be done by converting to PCM first and then back to DSD later! Thus 'Direct Stream Digital' is an oxymoron--any SACD published through a mastering studio must have gone through more processing steps than an equivalent DVD-A, with more opportunities for quality degradation!

In case you were hoping that the transformations would be lossless:

Quote:

Q: and what is the process to transform delta-sigma into PCM? A: 1-bit > 16-bits. 20kHz low pass filter. Keep every 64th sample, throw away the rest. (there are more computationally efficient methods!!!). (note: this is just an example for converting into 16/44.1. Change numbers appropriately for different formats.) Q: Can it be lossless? A: Of course not. You lose everything above half the destination sample rate. Whether this is audible or not...

On another note: after one of these DSD->PCM->DSD, any debate about the questionable advantage of DSD in terms of impulse response will be moot, as the 'poor' impulse response and ringing 'problem' (at 96kHz

) of PCM will be inherited by the DSD it regenerates.

In this graph, although you can't see pre-ringing in the DSD plot, you CAN see the constant low level noise throughout the graph. The impulse may LOOK good but in fact it is only distinguished from the background noise by a bare 20dB, whereas the pulses for the PCM formats are distinguished from background noise by the full 144dB. The DSD plot looks different because DSD has the option of almost not doing any lowpass filtering before its Nyquist frequency of 1.4MHz to make things look good on a graph, whereas the Nyquist frequency for 24/192 is 96kHz. But it's neither a realistic nor a good option. So your amplifier can go up to 130kHz. The SACD player would have to filter the sound down to nothing at 130kHz, which is not all that much higher than 96kHz. And even this option of looking good on a graph will be gone for DSD in the event of going through a real sound studio, because for any post processing it has to be converted to PCM and back, and an impulse that went through those steps would look like one of the PCM graphs, even if its encoded in DSD--albeit with more background noise!

Quote:

Moreover, from 20 kHz and onwards the necessary signal-to-noise ratio becomes increasingly less important, which is in concordance which the natural behaviour of a SDM.

I.e. it is admitting that DSD is storing not much more than noise above 20kHz. By the same token you can have SACD quality by coupling a CD player with a high-frequency noise generator.

Might even be true--I suspect the reason why people sometimes prefer vinyl to CD is the high frequency noise.

 

[Jasper994]

Quote:

Originally posted by Czilla9000 EDIT: The Philips dude said repetedy that SACD "is a consumer format". One of the things the Philips guy points out alot is that SACD can store more music than DVDA, making it a better consumer format.

Do we really need more filler material that we already have on an album? I can just see it now... The average SACD comes with 25 songs... to bad there's still only 1 or 2 good songs per disc...

 

[theaudiohobby]

Quote:

Higher means 'higher than 44.1kHz', not 'the higher the better', especially not when at 24/192 you can capture all frequencies up to 96kHz with 144dB dynamic range and with the '2.8MHz' DSD signal you only have 120dB dynamic range starting from 0Hz down to next to nothing at 96kHz.

Excuse me, I have followed with a more recent dcs newsletter and it is definitely making comparisions with DVDA and the comparisions are very favourable and validate philips assertions of better transient response than DVDA at its best. now read this

Quote:

It is then also possible to choose a less rigid filter, since aliasing distortion then occurs at a frequency higher than the human hearing spectrum, and it can be argued that the distortions in this area have less audible effect. With those "mild" filters, impulse response is improved a lot though. In this higher sampling rate "trade off", DSD comes off particularly well, as with a sampling rate of 2.8MHz, the filter can be very gentle indeed. However this sampling rate allows only for 1 bit in the SACD standard, so the signal to noise ratio is far from optimal. This problem is solved by introducing so-called "noise-shaping", whereby superfluous noise energy is literally moved to regions of the audio spectrum above the bandwidth of human hearing. Filtering from about 70kHz is consequently necessary because monitoring equipment can often not handle these high levels of noise energy in the upper bands. So again, every system has it's Pro's and Con's!

In other words, DSD has better transient response however the trade off in high noise floor. Now let's go one better and see read this

Quote:

With DSD, noise is always present at a very high level (see newsletter # 4) and here lies a big difference with PCM. ‘Silence never really occurs in DSD’. Since the signal is always predominantly noise-like, the effects of signal related jitter will also be noise-like and will tend not to be objectionable. So, in spite of the fact that DSD should be quite a bit more sensitive to jitter than PCM, the products are likely to be noise-like and benign, with none of the harmonic distortion effects that would occur in PCM. If error products are present, they will show up as a rise in noise floor, so they are easy to detect.

In other words, the presence of that noise actually has some beneficial effects that are otherwise unavailable to DVDA that is that jitter is easier deal with in DSD. how ironic...

Quote:

NO OBSERVATION means NO OBSERVATION. From the paper how do you know how long they listened to 24/96, 24/192 and DSD respectively?

Have you read the subsequent papers, newsletter 6 and newsletter 4 before saying this, there is no doubt that DSD performs better in the time domain than DVDA and unfortunately for DVDA, its architecture PCM is the key to its waterloo.

Quote:

Not so. "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."

I am sorry you are flat wrong here, that question was already resolved as far back as 2000, read David Rich report and Sony's response in AES 2000 on Stereophile. The fact that DCS and other companies are already implementing studio products that give textbook DSD performance in itself is sufficient empirical evidence that you are flat wrong, the greatest irony though it that the increased noise in DSD has beneficial effects wrt to jitter handling because pulls the rug from right under the feet of the dissenters and I suspect is the reason why companies like Teac, Yamaha, Denon etc has reluctantly albeit belately started supporting DSD.

Finally, YOU DO NOT NEED AN INTERMEDIATE PCM STAGE and there are quite a few products on the market that preserve your signal in the DSD domain. DCS already has one, check their site As for that thread on hydrogen audio, if the guys only spent a little time doing a little careful study, it will some become clear there is no substance to most of their objections. Being skeptical is not equivalent to being correct.



As an aside, I am still persuaded that the proponents of DSD/SACD hit upon this goldmine accidentally, because looking at the technical merits at face value, it looks as if DVDA is superior, it is only as you begin to investigate the science that it becomes clear DSD/SACD has superior real world performance and is cheaper to implement in consumer systems.

 

[Joe Bloggs]

Quote:

Excuse me, I have followed with a more recent dcs newsletter and it is definitely making comparisions with DVDA and the comparisions are very favourable and validate philips assertions of better transient response than DVDA at its best ... In other words, DSD has better transient response however the trade off in high noise floor.

So DSD, instead of having ringing in *one* inaudibly high frequency, has a lot of noise in *all* inaudibly high frequencies.

1. They are both inaudible, so why are you saing DSD is superior???
2. Let's say we pretend for a second that you can hear those high frequencies. Why is noise over the full spectrum preferable to noise in one group of frequencie, the transition band frequencies? The comment about wanting to have the cake and eat it too definitely applies here.

3. Do you realize how much higher the noise floor is?

In order to get acceptable performance in the audible band, DSD has to push more noise into these high frequencies. With such high amounts of noise are you still willing to have such a gentle filter that it leaves all the noise in? As I said before, if your amp can go only go up to 130kHz (even that is probably marketing crap) you'd have to cut at 130kHz, and the impulse response for that is not going to look much different from cutting at 96kHz.

Also, let's consider 3 possibilities:

1. We can't hear above 20kHz
2. We can hear a bit above 20kHz but not much higher
3. We can hear all the way to 96kHz and beyond

Now according to real research, the 1st option is the most likely, with the supporting evidence of your everyday hearing test and also ABX tests between CD and higher-quality formats--i.e. no test has proven that people can distinguish between CD and a higher-resolution format.

If you are going to pursue high-res formats, you must discard (1) and assume either 2 or 3 is true. Given that (1) is most probably the actual truth, (2) at least is much more likely to be true than (3).

Now here's the kicker: if you can hear above 20kHz but can't hear anywhere close to 96kHz, you can never hear the ringing at 96kHz in a 24/192 recording. However, you CAN hear the extra noise from the DSD system.

Even if you could hear all the way up to 96kHz and beyond, it'd still be debatable whether ringing at a single frequency or broadband noise throughout all high frequencies is to be preferred.

Quote:

Quote: Not so. "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." I am sorry you are flat wrong here...

You FOOL, are you trying to say that there is no limit to the amount of noise shaping that can be done? If that were the case you could have infinite dynamic range and zero noise for DSD for all frequencies up to 1.4MHz, with ALL the quantization noise pushed up to the maximum frequency, 1.4MHz! Is that the case now? NO! Once again you demonstrate your ignorance of digital signal basics. If you don't have a stack of DSD propaganda on your hard drive you'd be naked!

Quote:

benefits w.r.t. jitter handling...

You might want to look at some jitter measurements of real SACD players before you say this.

From here: http://www.stereophile.com/showarchives.cgi?814:6

No, actually this is a dedicated DAC, which should have better performance than most standalone SACD players.

In contrast here is a multiformat one-box player playing a *CD*, of all things:

The lower black trace is the jitter measurement for CD, the gray trace *above* for DSD.

Whether the theoretical advantage is real or not, I don't see any real world advantages.

Not to mention that we are discussing the relative merits of the formats--where you must be comparing the best playback, not the jitter from some $100 player

Quote:

YOU DO NOT NEED AN INTERMEDIATE PCM STAGE

That's funny, Philips doesn't seem to think so, and Philips should know

Just because you have dedicated systems for processing DSD doesn't mean their innards can't all be coded in PCM.

Although the Philips paper could possibly be outdated, and 8-bit DSD could possibly be directly mixed / filtered. (not 1-bit) But as we all know these DSD mastering boards are insanely expensive, which was the biggest point at the start of this debate--that if SACD takes over the market indie musicians will be driven out of business.

Quote:

As for that thread on hydrogen audio, if the guys only spent a little time doing a little careful study, it will some become clear there is no substance to most of their objections. Being skeptical is not equivalent to being correct.

If you are so sure of yourself why don't you start posting there.

Quote:

it is only as you begin to investigate the science that it becomes clear DSD/SACD has superior real world performance and is cheaper to implement in consumer systems.

And you have not once demonstrated that you have any knowledge of the science behind either DSD or DVD-A, having relied on quoting pro-DSD propaganda in every step of the debate. I'm sorry, but I won't bother debating with you anymore unless you can demonstrate that you at least understand some of what you are quoting.

 

[Joe Bloggs]

attachment

 

[theaudiohobby]

Quote:

And you have not once demonstrated that you have any knowledge of the science behind either DSD or DVD-A, having relied on quoting pro-DSD propaganda in every step of the debate. I'm sorry, but I won't bother debating with you anymore unless you can demonstrate that you at least understand some of what you are quoting

Has the deluge of information showing the superiority in the time domain too much for you

, as I stated at the beginning of this thread, DVDA does indeed have more constant bit resolution over it bandwidth, however does that in itself constitute superiority, the information so obviously points otherwise. In this thread, I have taken a different approach to you and that is show citations that validate my original statements and most of contentions you cite have been comprehensively answered by these citations. Dont forget that 96/24 and 192/24 were around before DVDA came on the scene. The whole thrust of your responses is that the inherent noise in DSD precludes it from delivering anything near its textbook performance and this position has been disbunked long ago. I started this thread out by quoting a dissenting paper that stated some theoretical performance advantages that DSD which paper felt could not be translated into practical performance and then I proceeded to show citations that show otherwise. For the record after philips response I did not read any further contentions on the issues addressed by that paper. In conclusion, sometimes all you need is another line of thinking.

Quote:

We recognise the advantage of bitstream in a basic system and the natural extension of ultrasonic bandwidth

This quote comes from the dissenting paper and their objections were
Quote:

however this is easily lost in post-processing where there is a danger of intermodulation with high-frequency audio and out-of-band shaped noise.

and this is also your line however all these issues have already been answered long ago.

 

[Joe Bloggs]

Quote:

their objections were quote: -------------------------------------------------------------------------------- however this is easily lost in post-processing where there is a danger of intermodulation with high-frequency audio and out-of-band shaped noise. -------------------------------------------------------------------------------- and this is also your line however all these issues have already been answered long ago

I never talked about such things

 

[theaudiohobby]

Joe,

What DACs are you citing?

 

[Joe Bloggs]

SACD: http://www.stereophile.com/showarchives.cgi?814:6
DVD-A: http://www.stereophile.com/fullarchives.cgi?515
Total jitter for the SACD player: 147.8ps
for the DVD-A player: 142ps
Which are pretty much in the same ballpark, so should be a good basis to compare the amplitude of signal-related frequencies caused by jitter, which in the graphs are circled and highlighted in purple.

You're welcome to make your own citations if you're not satisfied with this pair. But you might want to address more important points first. Like, how if we could hear above 20kHz at all we'd all hear the superiority of DVD-A

One more time: my stance is that SACD and DVD-A are equally good from the human ear's perspective. If you'd stop treading on my toes, I'd stop treading on yours

 

[theaudiohobby]

I will continue to thread on your toes because I am sure I have you in a corner

, I am in the process of delivering what is commonly known as the knock out punch.

Quote:

In order to get acceptable performance in the audible band, DSD has to push more noise into these high frequencies. With such high amounts of noise are you still willing to have such a gentle filter that it leaves all the noise in? As I said before, if your amp can go only go up to 130kHz (even that is probably marketing crap) you'd have to cut at 130kHz, and the impulse response for that is not going to look much different from cutting at 96kHz.

Really, why does sampling at 192KHz offer better time resolution than sampling @96Khz with a corresponding improvement in performance

Quote:

Now here's the kicker: if you can hear above 20kHz but can't hear anywhere close to 96kHz, you can never hear the ringing at 96kHz in a 24/192 recording. However, you CAN hear the extra noise from the DSD system.

Really, look at the graphs you so graciously provided and noise is measured at -130db @ 20Khz. are you seriously suggesting audibility @-130db and those values are consistent with the values quoted on the philips paper you are disparaging. The DAC that you quote the dcs Elgar/Elgar plus gives best CD performance with CD upsampled to DSD, how is that for irony.

The time resolution is proportional to sampling frequency, surely a jump from 192Khz to 1.4Mhz means something and that is what the dissenting paper referred to when it referred to natural extension of ultrasonic bandwidth. Surely if moving up in sampling frequency is better for PCM wrt time resolution. Why all of a sudden does that cease to apply because we now have DSD? Whether you accept it or not the transient response and time resolution of a DSD system is an order of magnitude better than 192/24, its transient response will be about 4/5 less and certainly much closer to the theoretical analog limit than PCM at 192KHz could ever hope to achieve and that is why dcs said the results are very favourable in their newsletter.

The phrase 'non-correlated noise' was first cited in the dissenting paper why are you suddenly unwilling to use that paper as a basis of any further discussion seeing that you used in the past when it suited your purpose. Non-correlated noise means exactly that, i.e. it has no correlaton to the signal and can be easily filtered out, when you talking about DSD @2.8MHz or 1.4MHz the nyquist frequency, we are only interested in <10% of the available frequency bandwidth as opposed to 192/24 where you run against your nyquist frequency limits much earlier @96K and that limitation is not benign if it were then the logic for moving from 96Khz to 192Khz will cease to exist since the nyquist limit of 24/96khz is also outside the audible band.

Why does the jitter theory so upset you, In a DSD system running @2.8Mhz, we also interested in <.05 % of the available bandwidth. The theory has great validity cos the ratio of jitter with respect to sampling frequency in DSD systems is much lower and you have so kindly provided in the last few minutes, measurements that show that SOTA machines show only a difference of 5 picoseconds. How is that for irony?

If you accept that DSD offers better time resolution even though DVDA has better bit resolution, then I will call it quits, otherwise I will continue the torture.

 

[Dusty Chalk]

Quote:

Originally posted by theaudiohobby I will continue to thread on your toes because I am sure I have you in a corner , I am in the process of delivering what is commonly known as the knock out punch.

No, I think Joe is winning this. At least this from a not-so-innocent bystander. Quote:

Really, look at the graphs you so graciously provided and noise is measured at -130db @ 20Khz. are you seriously suggesting audibility @-130db and those values are consistent with the values quoted on the philips paper you are disparaging.

Actually, I would have expected this from the "they both sound really good" camp (I.E. Joe).

And funny how you mention that the presence of noise can have a beneficial "side-effect" -- it's like you already bought the noise, hook, line and sinker. I think Joe's point (and it's a good one, so it's mine, too) is that the noise's existence in the first place is a problem.

More later, when I have time (I.E. never).

 

[theaudiohobby]

Dusty Chalk,

Joe made a big deal about the noise and then committed the ultimate booboo and undoing of his argument by picking the DCS Elgar Plus as the proof for part of his argument, check the graph, it's SACD linearity is near perfect. This DAC goes further than any other in nailing down the superiority of DSD.

Why should jitter getting swallowed up in noise surprise you, they already applying noiseshaping to the signal anyway to get the required resolution, would the noiseshaping magically exclude jitter? DCS made that statement and produced a DAC that had near perfect linearity with SACDwhen independently measured,

read the comments it says Quote:

Fig.3 plots the Elgar's linearity error for SACD playback, using the 1kHz spot tones on the Sony Test SACD. The measured level error is effectively zero down almost to -120dBFS, which is superb. It rises below that level due to the increasing contribution of noise

those comments are a vindacation of the DCS original comment in their news letter.

Now look at the Weiss that is one of the best DACs out there

What do you think is causing those jagged readings,huh?

There really is no further need to continue this discussion anymore, you guys have forced me to do some hardwork to validate my statements, however resolution finally came from an unexpected source Joe B himself by committing the ultimate booboo that is by introducing DCS Elgar Plus measurements into the discussion. I am done here, DSD/SACD has superior time resolution and handles jitter more elegantly and that superiority has been demonstrated in a commerically available component

 

[Joe Bloggs]

Quote:

Originally posted by theaudiohobby I will continue to thread on your toes because I am sure I have you in a corner , I am in the process of delivering what is commonly known as the knock out punch.

From my experience with online debates it is the losing side that likes to declare victory prematurely. Just ask bbs.stardestroyer.net

Quote:

Really, why does sampling at 192KHz offer better time resolution than sampling @96Khz with a corresponding improvement in performance

This only has any meaning if you can hear above 48kHz.

Quote:

Really, look at the graphs you so graciously provided and noise is measured at -130db @ 20Khz. are you seriously suggesting audibility @-130db and those values are consistent with the values quoted on the philips paper you are disparaging.

I am saying that you have no chance of hearing anything at 96kHz, and if you can hear anything between 20kHz to 96kHz at all, it is DVD-A that offers higher fidelity at those frequencies. Say you can hear up to 40kHz (rather unlikely!), the noise floor for DSD at 40kHz is -60dB and -130dB for DVD-A

Of course, if you say -130dB@20kHz or -60dB@40kHz, the conclusion is that DVD-A and SACD are equal in quality for human ears. Certainly not that SACD is better because it has more noise that we can't hear

Quote:

The DAC that you quote the dcs Elgar/Elgar plus gives best CD performance with CD upsampled to DSD, how is that for irony.

And the DAC in the Pioneer gives better performance than the Elgar without having to do that.

Quote:

The time resolution is proportional to sampling frequency, surely a jump from 192Khz to 1.4Mhz means something and that is what the dissenting paper referred to when it referred to natural extension of ultrasonic bandwidth. Surely if moving up in sampling frequency is better for PCM wrt time resolution. Why all of a sudden does that cease to apply because we now have DSD?

This only has meaning if you can hear above 96kHz. If your hearing limit is ANYTHING below 96kHz, DVD-A wins. Time resolution that is too high for humans to discern is time resolution wasted.

Whether you accept it or not the transient response and time resolution of a DSD system is an order of magnitude better than 192/24, its transient response will be about 4/5 less and certainly much closer to the theoretical analog limit than PCM at 192KHz could ever hope to achieve and that is why dcs said the results are very favourable in their newsletter.

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The phrase 'non-correlated noise' was first cited in the dissenting paper why are you suddenly unwilling to use that paper as a basis of any further discussion seeing that you used in the past when it suited your purpose. Non-correlated noise means exactly that, i.e. it has no correlaton to the signal and can be easily filtered out,

LMAO.

Tell that to someone trying to remove analog tape hiss (hey guys check out this ignorant ass) If you were correct there would be no such thing as a noise floor to worry about in SACD. What's more jitter is still correlated with signal in the SACD DAC. Every data point marked in purple is signal correlated jitter. Or is that too hard to understand for your pea-sized brain?

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that limitation is not benign if it were then the logic for moving from 96Khz to 192Khz will cease to exist since the nyquist limit of 24/96khz is also outside the audible band.

The logic is mostly not there, except that you might want more resolution for processing stages. Before you point out the listening tests from the dCS paper, there was no mention of double-blind tests and people could have been just shooting their mouths off imagining differences. Not to mention that as someone in the other forum pointed out, the state of the art in recording ADCs are now multibit converters, not the sigma-delta converters DSD was designed for. (at a time when sigma-delta converters were state of the art) If any of the problems stated in the dCS paper were real, they were obviously due to problems in the equipment and not the format itself.

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Why does the jitter theory so upset you, In a DSD system running @2.8Mhz, we also interested in <.05 % of the available bandwidth. The theory has great validity cos the ratio of jitter with respect to sampling frequency in DSD systems is much lower and you have so kindly provided in the last few minutes, measurements that show that SOTA machines show only a difference of 5 picoseconds. How is that for irony?

1. The difference is in favor of the universal, ONE-BOX player that was playing a CD
2. But that wasn't even the point--the point was to check for signal-correlated jitter, which is a subset of the total jitter, not the sum total; i.e. of the total jitter that the SOTA machines could not avoid, how much of it was translated into 'signal-correlated jitter', i.e. jitter that manifests itself as tones at a predictable frequency above or below the signal frequency? And the answer is, it's comparable for the two machines.

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If you accept that DSD offers better time resolution even though DVDA has better bit resolution, then I will call it quits, otherwise I will continue the torture. [/B]

I will accept that DSD offers more time resolution that nobody can hear, while having more and more noise up to 96kHz that you are much more likely to hear than the ringing at 96kHz--but the probabilities for both events are infinitesimal, which is why the two formats are equivalent in quality.