Hello everyone,
A lot of audiophiles approves the SACD formats, but some irreducible retorts that the DVD Audio is superior.
And you, of what side are you?
Garf's sig say's it all...
The answer to the SACD vs. DVD-Audio question is: CD -- DigitalMan
Please search and read the FAQ - this has been discussed many, many times here and I don't think another discussion would achieve anything.
Sorry, I didn't know it.
If it helps, I believe there are now hardware players that support both formats.
Thank you indybrett, but I know this domain very well.
Ayre, Linn, Méridian already develops for a long time this principle.
I wanted to know that rightly that uses each among us.
In my opinion, the only way to answer this is to perform blind comparisons. And to be sure that both tracks are coming from the same mastering. It seems that nobody did it.
I approve your reasoning on the principle of this comparison.
But what to say formats audios that constitutes the world of music.
Must one to trust the decoders audios, that identifies and read the formats ( on PC) or to electronics (sources, readers DVD or SACD of lounge) that treat it also.
My sig says another thing about that discussion, actually, from one of the foremost experts here
Personally I strongly favor DVD-A over SACD/DSD on technical grounds. This has been elaborately discussed here before.
Read for example http://sjeng.org/ftp/SACD.pdf (http://sjeng.org/ftp/SACD.pdf)
Favorite quote:
Finally, consider 8-bit, four-times-oversampled PCM with
noise shaping. This is also a data rate one-half that of DSD and
double that of CD, with a sampling rate of 4 × 44,100 =
176,400 Hz. It can achieve a noise floor 120 dB below full
scale up to 20 kHz, using 96 dB of noise shaping, and a total
noise power of –19 dBFS. Its frequency response would be
flat to 80 kHz. This example is perhaps the most instructive of
the lot. For a data rate one-half that of DSD, it achieves a
comparable signal bandwidth, with a similar noise power
density up to 20 kHz, but much lower power above this
frequency, and 28 dB lower total noise power. It is fully
TPDF-dithered, and so is completely artefact free. At one-half
the data rate it outperforms DSD on every count! DSD is a
profligate wastrel of capacity.
Good evening Garf,
Would the quality of a product be bound only to a technical proof?
I don't think it.
It is only my opinion: my ears first then the view.
What I know, an amplifier transistor will never be an lamp amplifier and even less a hybrid amplifier.
The comparison will always be uncertain and function of the individual's entity.
I Would Recommend DVD-A
Would the quality of a product be bound only to a technical proof?
What part of
At one-half the data rate it outperforms DSD on every count!
didn't you understand?
The goal of the carrier is to reproduce the original signal as faithfully as possible. DSD is very suboptimal in that regard. That's a mathematical fact.
Distortion (like the one from your beloved tube amps) can (and should!) always be added later on, since it's entirely subjective what sounds "better" there.
I admire the clarity of your speech because you give proof on the one hand of truth.
Long life to the développemnt of your codec audio.
To soon
In my opinion, the only way to answer this is to perform blind comparisons. And to be sure that both tracks are coming from the same mastering. It seems that nobody did it.
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[span style='font-size:8pt;line-height:100%']emphasis is mine[/span]
I don't know if this is even possible since the processing of DSD and PCM data is so different.
Right now SACD are mostly created from 96/24 masters (at best) and surely the converted DSD signal can not be better.
DVD-A !!
SONY has to be punished for their constant attempts to establish their own 'standards', with us, the users, being the ones paying the bills ...
What's all this talk about optical carriers about? Don't you know that is sooo 20th century...
What I know, an amplifier transistor will never be an lamp amplifier and even less a hybrid amplifier.
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.........
You would have to know, that many electrical and electronic engineers are not in agreement with this.
In many cases this is a market opinion, and it does not reflect the exact reality...
A lot of audiophiles approves the SACD formats, but some irreducible retorts that the DVD Audio is superior.
[a href="index.php?act=findpost&pid=276142"][{POST_SNAPBACK}][/a]
I doubt much that, due to the limitations of the human ear, you notice a difference between SACD, Dvd-a, and Audio CD (for normal stereo listening).
of course, if you notice differences this probably is as consequence of differents masterings...
Hello Acid Orange Juice,
"You would have to know, that many electrical and electronic engineers are not in agreement with this.
In many cases this is a market opinion, and it does not reflect the exact reality... "
Indeed, if you take the example of the very upscale: Naim Nac552/Nap500, Meridian, Dcs elgar, Ayre or Linn, it is true the analysis of the specter, the passing strip and the dynamics is not really the same.
But the charm of the tube (artisanal manufacture) will remain and will give a heat always little present among the big constructors of electronic high-quality.
" of course, if you notice differences this probably is as consequence of differents masterings... "
The mastering is often questioned in this case of face.
Must i think that an excelllent mastering of the SACD can surpassed a DVD Audio of middle quality.
Finally, this analysis makes myself think about the eternal return (psychoanalysis): the mobile picture of one immobile eternity.
It is strange, but that reflects there strongly: Mastering/Quality, Quality/Mastering.
Heh. An excellent mastering on CD has probably nothing to envy to any SACD or DVD-A.
And, again, technically, DVD-A is far superior to SACD. The point is if that makes DVD-A sound any better than SACD. But then, even when both are technically better than redbook CD, it's not clear that this can make them sound any better than good old CD. It's all about mastering.
The key advantage to SACD is that it can easily be distributed as a hybrid CD. I haven't seen any listening-test being conducted that clearly favored one over the other, so unless the music is targeted at bats, I think the world would be better off with SACD becoming the standard.
Just my 2¢.
DVD-A !!
SONY has to be punished for their constant attempts to establish their own 'standards', with us, the users, being the ones paying the bills ...
[a href="index.php?act=findpost&pid=277193"][{POST_SNAPBACK}][/a]
What bills?
Indeed, if you take the example of the very upscale: Naim Nac552/Nap500, Meridian, Dcs elgar, Ayre or Linn, it is true the analysis of the specter, the passing strip and the dynamics is not really the same.
But the charm of the tube (artisanal manufacture) will remain and will give a heat always little present among the big constructors of electronic high-quality.
Hello guada2 ;
My opinions are based in my own experience as electronic engineer and as scientist; not by publicity or by opinions of third parts, that, others repeat like parrots without having idea of the subject which they are treating.
Their arguments are not more than their own and subjective opinions, and they are not based on facts, evidence or science; like either in engineering.
I have 4 years designing amplifiers of high fidelity. If you think that you will obtain better quality with an expensive amplifier of tubes, when you can obtain the same sound with an amplifier of transistors well designed (by a much smaller cost), then, you are free to waste your money...
Must i think that an excelllent mastering of the SACD can surpassed a DVD Audio of middle quality
... again ...
Your opinions are not based on facts or scientific evidence; they are based only in magical suppositions..
you did not pay attention to this:
I doubt much that, due to the limitations of the human ear, you notice a difference between SACD, Dvd-a, and Audio CD (for normal stereo listening).
It's a fact, not a supposition..
It's well known that the audio CD is completely overkill for the human ear. The normal audio CD have more resolution that the human ear is able to perceive. SACD and DVD-A have more resolution that the normal audio CD.. but.. for what?. It's not necessary for your ears.. maybe if you are a dog probably you hear the difference...
Of course, this is for the case of normal stereo listening.. For multichannel is another story..
To see if there are indeed audible differences between DVD-A and CD, would a valid test be to make a recording @ 96khz/24 bit WAV (a dvd-a spec), resample to a 44.1khz 16 bit CD spec WAV, and ABX the two, or would the 96khz recording be invalid because it wasnt first compressed in the DVD-A MLP format? Would differences become apparent through dithering noise?
Hello Acid Orange Juice,
"Their arguments are not more than their own subjective and expressed an opinion, and they are not based one facts, evidence however science; like either in engineering.
I haggard 4 years designing amplifiers of high fidelity. Yew you think that you will obtain better quality with year expensive to amplify of tubes, when you can obtain the same sound with year to amplify of transistors well designed (by has much smaller cost), then, you are free to waste your money... "
I don't put in doubt your experience, But to think that one DCS ELgar Plus/Verdi/Purcell doesn't make the weight, I am not sure. In short, it was only a parenthesis.
Indeed, I join you on a point: these are not the best electronic that pull the best resonant restitution. Because you know it all as much than me, that the quality of a source raises several criterias of tests.
After some years of comparatives and purchases of materials, I resolved to this stereo system:
Source: Linn Ikemi
Integrated amp: Icos Elsberg 270
New loudspeakers: Thiel CS2.4
Case of alimentaion sector: MPC
Cables HP: Synergistic Research
Cables of modulation: Esprit
It is not the dearest, but I am pleased of its capacity to answer best to the system that I searched for.
For what concerns video I reserve the right of all disclosure: scaler + spotlight + player DVD + préamp + amplifier multicannal + loudspeakers 5.1.
The key advantage to SACD is that it can easily be distributed as a hybrid CD. I haven't seen any listening-test being conducted that clearly favored one over the other, so unless the music is targeted at bats, I think the world would be better off with SACD becoming the standard.
Just my 2¢.
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I just noticed, that DVD Forum has (or will) specified DVDPlus or DualDisc format, which is a double sided disc, one side could be DVD Audio, other side is CD. Check this:
[a href="http://www.disctronics.co.uk/technology/dvdaudio/dvdaud_spec.htm]http://www.disctronics.co.uk/technology/dv...dvdaud_spec.htm[/url]
I think this will help somehow spread of DVD Audio discs.
I prefer DVD Audio format!
Hello everyone,
A lot of audiophiles approves the SACD formats, but some irreducible retorts that the DVD Audio is superior.
And you, of what side are you?
[a href="index.php?act=findpost&pid=276142"][{POST_SNAPBACK}][/a]
Mine: CDDA.
Industry: The side they think will be the hardest the be put on P2P networks. And like Garf said, DSD being completely inefficient AND different it is a godsend. Sound quality? Oh yeah... that..................
Guys... you don't see the BIG picture...
Unfortunately, it looks like the SACD is "winning" the war as the next audio format. In records stores, I've seen the DVD-Audio section shrink and wither, while more and more albums are released on (hybrid) SACDs.
As witnessed many times in the past, the technically superior product doesn't necessarily win...
A few years back I used to think DVD-Audio would win, because all PCs would be able to read them with the right software (and a 24/96 sound card, granted), while consumers would need a new player to read SACDs. It may have been precisely the reason why music publishers turned away from DVD-Audio, with the fear that the protection on DVD-Audio would some day be broken as easily as CSS was on DVD-Video.
A few years back I used to think DVD-Audio would win, because all PCs would be able to read them with the right software (and a 24/96 sound card, granted), while consumers would need a new player to read SACDs. It may have been precisely the reason why music publishers turned away from DVD-Audio, with the fear that the protection on DVD-Audio would some day be broken as easily as CSS was on DVD-Video.
[a href="index.php?act=findpost&pid=283633"][{POST_SNAPBACK}][/a]
It's only a matter of time before someone breaks SACD's protection schemes, though. If you can create it, you can also destroy it.
It's only a matter of time before someone breaks SACD's protection schemes, though. If you can create it, you can also destroy it.
[a href="index.php?act=findpost&pid=283781"][{POST_SNAPBACK}][/a]
Actually the hybrid CD/SACD disks have an interesting "protection", since a computer drive doesn't "see" the SACD layer, only the CD layer.
Actually the hybrid CD/SACD disks have an interesting "protection", since a computer drive doesn't "see" the SACD layer, only the CD layer.
[a href="index.php?act=findpost&pid=284780"][{POST_SNAPBACK}][/a]
That's just one of the intellectual property protection schemes built into SACD. Another is physical (visible) watermarking of the disc, as well as data encryption. Since there is no such thing as an SACD "burner," there's less likelihood that an SACD disc will be trivially pirated, as is the case with the DVD family.
As to which encoding method is "better," that’s a tough and lengthy discussion…
First off, any reference to Lipshitz and Vanderkooy’s paper on the inability to optimally dither DSD data is inappropriate and outdated if you pay attention to current technology. As one of the few people who has taken the time to conduct extensive tests with both 192 kHz LPCM and single speed DSD with live sources (my company <www.sonicstudio.com> pioneered and manufactures both PCM and DSD professional production systems), I can say that they are subjectively different but both equally valid.
As was pointed out in earlier postings, it's the details of the formats that really highlights the advantages of one over the other. The SACD format was designed, from the start, as backwards compatible with the Red Book spec. The Dual Disc "format" is a bolt-on afterthought that lacks compatibility with CD players due to the physical characteristics (thickness) of the discs.
DVD-Audio also cannot deliver the highest fidelity LPCM in multichannel and is limited to 192 k stereo. All other factors being equal, 96 kHz LPCM does not sound as close to the source as DSD does, which means you are saddled with lesser quality 5.1 when listening to a DVD-A release.
Production-wise, both 192 k LPCM and DSD make great origination formats; material can be transcoded into any distribution format from either source file, and is typically done in "the real world" to accomodate the vagaries of the marketplace. 48 or 96 k sources just don't contain enough information to produce a subjectively lossless upsample.
Since the DVD Forum has seen fit to not mandate various player features that would make the consumer experience more enjoyable (persistant memory of user settings, group selection controls, complete front panel control of the player), DVD-A users have to have a TV just to listen to a title. As the “multimedia” capabilities of DVD-A, I’ve never found the slideshow feature to be compelling, especially considering the severe limitations placed on the author by the format’s limitations.
Frankly, for the average consumer, there is little in the DVD-A format that isn’t already delivered more easily and more compatibly by the DVD-Video format. Badly produced audio, whether DSD or 192 k LPCM, will always sound worse than expertly and carefully produced 48k/24. Simply looking at numbers on a spec sheet tell you nothing about the intricacies of a complete media delivery system. Read a lot of information from different sources, weight your budetary options (“universal” players are now cheap but titles in both formats are overpriced) and, more importantly, listen critically before making a decision.
DVD-Audio also cannot deliver the highest fidelity LPCM in multichannel and is limited to 192 k stereo. All other factors being equal, 96 kHz LPCM does not sound as close to the source as DSD does, which means you are saddled with lesser quality 5.1 when listening to a DVD-A release.
[a href="index.php?act=findpost&pid=286097"][{POST_SNAPBACK}][/a]
I thought that sacd goes lossy if you put hard-to-encode hi-res multichannel content on it.
DVD-Audio also cannot deliver the highest fidelity LPCM in multichannel and is limited to 192 k stereo. All other factors being equal, 96 kHz LPCM does not sound as close to the source as DSD does, which means you are saddled with lesser quality 5.1 when listening to a DVD-A release.
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(btw: That part of yours smells like a terms-of-service rule #8 violation)
Are you implying that 96/24 isn't sufficient enough to transport sound transparently for us humans ? (I seriously doubt that)
Please elaborate on how you came to this conlcusion (and on the conlcusion itselt).
I thought that sacd goes lossy if you put hard-to-encode hi-res multichannel content on it.
AFAIK the DSD stream isn't stored lossy on the SACD -- no matter what.
SebastianG
DVD-Audio also cannot deliver the highest fidelity LPCM in multichannel and is limited to 192 k stereo. All other factors being equal, 96 kHz LPCM does not sound as close to the source as DSD does, which means you are saddled with lesser quality 5.1 when listening to a DVD-A release.
[a href="index.php?act=findpost&pid=286097"][{POST_SNAPBACK}][/a]
(btw: That part of yours smells like a terms-of-service rule #8 violation)
Are you implying that 96/24 isn't sufficient enough to transport sound transparently for us humans ? (I seriously doubt that)
The 192kHz claim from the poster you have responded seems absurd to me. Forget 192kHz(96Khz bandwidth) -- at this point no confirmed perceptual tests have concluded that anything exceeding a
[1]16kHz bandwidth is needed to be transparent for musical program playback for humans. The accepted JAES standard on this matter still stands as of this day.
-Chris
[1] Which Bandwidth Is Necessary for Optimal Sound Transmission?
G. PLENGE, H. JAKUBOWSKI, AND P. SCHONE
JAES, Volume 28 Number 3 pp. 114-119; March 1980
As one of the few people who has taken the time to conduct extensive tests with both 192 kHz LPCM and single speed DSD with live sources (my company <www.sonicstudio.com> pioneered and manufactures both PCM and DSD professional production systems), I can say that they are subjectively different but both equally valid.
Do you have detailed perceptual studies avaliable for our analysis?
DVD-Audio also cannot deliver the highest fidelity LPCM in multichannel and is limited to 192 k stereo. All other factors being equal, 96 kHz LPCM does not sound as close to the source as DSD does, which means you are saddled with lesser quality 5.1 when listening to a DVD-A release.
As the other poster noted, you have violated TOS of this forum. Audibility claims, when made in this fashion, must be accompanied by at least preliminary DBT results or solid correlation to a valid perceptual study.
-Chris
I thought that sacd goes lossy if you put hard-to-encode hi-res multichannel content on it.
AFAIK the DSD stream isn't stored lossy on the SACD -- no matter what.
[a href="index.php?act=findpost&pid=286138"][{POST_SNAPBACK}][/a]
Allright. Then it must be the MLP (dvd-a) which does lossy when lossless isn't small enough.
DVD-Audio also cannot deliver the highest fidelity LPCM in multichannel and is limited to 192 k stereo. All other factors being equal, 96 kHz LPCM does not sound as close to the source as DSD does, which means you are saddled with lesser quality 5.1 when listening to a DVD-A release.
[a href="index.php?act=findpost&pid=286097"][{POST_SNAPBACK}][/a]
I thought that sacd goes lossy if you put hard-to-encode hi-res multichannel content on it.
[a href="index.php?act=findpost&pid=286105"][{POST_SNAPBACK}][/a]
That was my understanding too. Mind you, it'd be easier to know the answer if it wasn't such a closed protected system. Way to go, Sony.
I thought that sacd goes lossy if you put hard-to-encode hi-res multichannel content on it.
AFAIK the DSD stream isn't stored lossy on the SACD -- no matter what.
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Allright. Then it must be the MLP (dvd-a) which does lossy when lossless isn't small enough.
[a href="index.php?act=findpost&pid=286251"][{POST_SNAPBACK}][/a]
[a href="http://www.hydrogenaudio.org/forums/index.php?showtopic=10520&view=findpost&p=106542]http://www.hydrogenaudio.org/forums/index....ndpost&p=106542[/url]
Unfortunately, it looks like the SACD is "winning" the war as the next audio format. In records stores, I've seen the DVD-Audio section shrink and wither, while more and more albums are released on (hybrid) SACDs.
[a href="index.php?act=findpost&pid=283633"][{POST_SNAPBACK}][/a]
DVD-A and SACD sales are declining (it's early for the "media of the future" )
HA has proved that even PCM is overkill for the human ear.
Talking about multi-channel, ac3 and DTS is widespread. For me DTS is fare superior and difficult to differentiate from the source. I didn’t make a real scientific test though. Just a friend changing the source.
DVD-A and SACD sales are declining (it's early for the "media of the future" )
HA has proved that even PCM is overkill for the human ear.
[a href="index.php?act=findpost&pid=286313"][{POST_SNAPBACK}][/a]
I think you mean that 96kHz 24bit linear PCM is overkill for the human ear. PCM merely refers to the way the signal is sampled and quantized - with no specification of sample rate or bit depth. For example, 22kHz 8bit linear PCM is certainly not overkill for music signals (it is for speech, though).
Talking about multi-channel, ac3 and DTS is widespread. For me DTS is fare superior and difficult to differentiate from the source. I didn’t make a real scientific test though. Just a friend changing the source.
[a href="index.php?act=findpost&pid=286313"][{POST_SNAPBACK}][/a]
If it's not a double blind test, then making any sort of conclusion from it puts you on fairly shaky ground. However, I agree with you that multichannel is the way things are going (people have mixed opinions about this) but I don't see why, with rapidly increasing storage space, lossy multichannel formats should have any advantage over 6 channel LPCM.
DVD-A and SACD sales are declining (it's early for the "media of the future" )
HA has proved that even PCM is overkill for the human ear.
[{POST_SNAPBACK}][/a] (http://index.php?act=findpost&pid=286313")
I think you mean that 96kHz 24bit linear PCM is overkill for the human ear. PCM merely refers to the way the signal is sampled and quantized - with no specification of sample rate or bit depth. For example, 22kHz 8bit linear PCM is certainly not overkill for music signals (it is for speech, though).
Sorry I didn't mention it, I was talking about 16/44 PCM
Talking about multi-channel, ac3 and DTS is widespread. For me DTS is fare superior and difficult to differentiate from the source. I didn’t make a real scientific test though. Just a friend changing the source.
[a href="index.php?act=findpost&pid=286313"][{POST_SNAPBACK}][/a]
If it's not a double blind test, then making any sort of conclusion from it puts you on fairly shaky ground.
[a href="index.php?act=findpost&pid=286328"][{POST_SNAPBACK}][/a]
Yep, that's why I have started the following topic:[a href="http://www.hydrogenaudio.org/forums/index.php?showtopic=32796&hl=]DTS/ac3 tests, Is it worth it?[/url]
However, I agree with you that multichannel is the way things are going (people have mixed opinions about this) but I don't see why, with rapidly increasing storage space, lossy multichannel formats should have any advantage over 6 channel LPCM.
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Outside the size benefits (which still an advantage even with big hdd, ...), with ac3/DTS you have the compatibility with all the DVD player...
DVD-A and SACD sales are declining (it's early for the "media of the future" )
HA has proved that even PCM is overkill for the human ear.
[{POST_SNAPBACK}][/a] (http://index.php?act=findpost&pid=286313")
I think you mean that 96kHz 24bit linear PCM is overkill for the human ear. PCM merely refers to the way the signal is sampled and quantized - with no specification of sample rate or bit depth. For example, 22kHz 8bit linear PCM is certainly not overkill for music signals (it is for speech, though).
Sorry I didn't mention it, I was talking about 16/44 PCM
[a href="index.php?act=findpost&pid=286334"][{POST_SNAPBACK}][/a]
I doubt you will get much support for the opinion that 16/44 PCM is overkill for the human ear. Perhaps "adequate" would be better - because that's exactly what it is - enough, but not too much.
Yep, that's why I have started the following topic:[a href="http://www.hydrogenaudio.org/forums/index.php?showtopic=32796&hl=]DTS/ac3 tests, Is it worth it?[/url] Outside the size benefits (which still an advantage even with big hdd, ...), with ac3/DTS you have the compatibility with all the DVD player...
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Agreed, compatability is a big plus. It will be interesting to see the outcome of your surround listening tests. Unfortunately I don't have a decent 5.1 system, so I can't participate.
I doubt you will get much support for the opinion that 16/44 PCM is overkill for the human ear. Perhaps "adequate" would be better - because that's exactly what it is - enough, but not too much.
Well, it's the principle of lossy encoder. You can’t say the difference between a lossy at around 200 kb/s and a PCM 16/44 at 1411 kb/s. It means pcm 16/44 is overkill for the human ear. I doubt you will get much support to claim the contrary here at HA.
Outside the compatibility, the benefits of pcm being lossless, meaning you have the choice of the lossy encoder without transcoding artefact. You still have lossless compression at around 800 kb/s.
The 192kHz claim from the poster you have responded seems absurd to me. Forget 192kHz(96Khz bandwidth) -- at this point no confirmed perceptual tests have concluded that anything exceeding a [1]16kHz bandwidth is needed to be transparent for musical program playback for humans. The accepted JAES standard on this matter still stands as of this day.
-Chris
[1] Which Bandwidth Is Necessary for Optimal Sound Transmission?
G. PLENGE, H. JAKUBOWSKI, AND P. SCHONE
JAES, Volume 28 Number 3 pp. 114-119; March 1980
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I haven't read that paper, but a clean 16kHz low pass is clearly not transparent to many younger listeners for much source material. This was demonstrated during the tuning of the --alt-presets, and even before during work on the old --r3mix preset - filtering at 19.5kHz was audible to one person who took part in those tests!
Cheers,
David.
I haven't read that paper, but a clean 16kHz low pass is clearly not transparent to many younger listeners for much source material. This was demonstrated during the tuning of the --alt-presets, and even before during work on the old --r3mix preset - filtering at 19.5kHz was audible to one person who took part in those tests!
The testing in the JAES article, which was eventually peer reviewed and accepted as a standard, used test signals that were designed to test respective fixed phase and spectrum content, simulating analysed music(with boosted high frequency spectrum in order to increase sensativity ). The conditions of playback were conistant for all test subjects(half of which were audio professionals - the results of both groups were segregrated). It is true, that perhaps very young people could score significantly better in the musical program HF testing, but since very young people were not used, it remains to be a theory.
I don't know specifically which factors/variables were used during this testing of which you refer. I suspect it may have been the standard *remote* testing as is standard during the codec developments on hydrogenaudio.org. In the case of remote listening, a simple matter of a person testing at much higher SPL levels for the test than is realsiticly feasible during actual listening may greatly influence the test results. They might use a device that has a very inaccurate frequency response anamoly that greatly enhances the ability to hear a specific band.These are just two examples, and of course there are others that limit the practical usefulness in such a remote testing method. I recognize this remote method is certainly useful for guidance of codec designs , especially considering the alternative of no group perceptual testing -- but the lack of careful consistant test conditions and fraud suspectibility(I.E.; depending on the test subject's honesty) reduces the merit and reliability of such testing. It could be that the test results of the testing to which you refer were accurate -- but please understand my skeptiscm based on the factors present in such testing.
-Chris
one need only try to ABX musical content with --aps to realise the tests are statistically significant... i can't ABX preset standard except on problem samples (and these are mp3 flaws, not bandwidth related problems), but i can pretty consistently ABX --preset standard -Y, which is basically a 16khz lowpass (except where short-blocks are triggered).
as far as DVD-A vs SACD, that's a no-brainer. the processing is all done in LPCM, so why not leave it there?
i'm not yet a multichannel convert. for most, it's simply a fad. look at the average 5.1 setup in people's homes - the speakers are in all the wrong places, often obscured completely by couches or ornaments or whatever, pointing in bizarre angles, etc etc. to say nothing of the much-too-high subwoofer crossovers, and ludicrously exaggerated bass. most systems are set up to fit the decor, not for good sound.
i'm sure i'll change my opinion if i get a really good 5.1 system, but right now i'll settle for an above-average 2.0 system (movies still sound stunning on anything capable of decent dynamics and flattish response).
The testing in the JAES article, which was eventually peer reviewed and accepted as a standard, used test signals that were designed to test respective fixed phase and spectrum content, simulating analysed music(with boosted high frequency spectrum in order to increase sensativity ). The conditions of playback were conistant for all test subjects(half of which were audio professionals - the results of both groups were segregrated). It is true, that perhaps very young people could score significantly better in the musical program HF testing, but since very young people were not used, it remains to be a theory.
I don't know specifically which factors/variables were used during this testing of which you refer. I suspect it may have been the standard *remote* testing as is standard during the codec developments on hydrogenaudio.org. In the case of remote listening, a simple matter of a person testing at much higher SPL levels for the test than is realsiticly feasible during actual listening may greatly influence the test results. They might use a device that has a very inaccurate frequency response anamoly that greatly enhances the ability to hear a specific band.These are just two examples, and of course there are others that limit the practical usefulness in such a remote testing method. I recognize this remote method is certainly useful for guidance of codec designs , especially considering the alternative of no group perceptual testing -- but the lack of careful consistant test conditions and fraud suspectibility(I.E.; depending on the test subject's honesty) reduces the merit and reliability of such testing. It could be that the test results of the testing to which you refer were accurate -- but please understand my skeptiscm based on the factors present in such testing.
-Chris
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All those problems are certainly possible (or probable!) in such tests - but these "problems" also apply to normal people listening to normal music under what they consider to be "normal" circumstances.
In other words, if playing back the music louder than "normal", or using poorer transducers than "ideal" causes the 16kHz low pass to be audible, then it's audible - and will be audible to some listeners some of the time in what
they consider to be "normal" listening.
If we forget deafening loudness levels for a moment, all the other factors you mention are only partly relevant anyway - does it really matter if someone detects a difference by listening with a pair of headphones that boost HF by 6dB? This simply means that, on a different track with 6dB more HF, they'd
still identify the problem on a pair of headphones with a
flat frequency response.
What's more, taking a "normal" recording, and boosting the HF by 6dB, may well give you
much less HF than is found on some recent electronic or experimental music.
I agree that it would be useful to test people like Garf, Dibrom etc under ideal conditions, but I'd suggest that the results we already have show that 16kHz is too low - unless you believe all audible differences
apparently due to content above 16kHz were due to ITD in the equipment producing actual changes below 16kHz in the output.
Cheers,
David.
but I'd suggest that the results we already have show that 16kHz is too low - unless you believe all audible differences apparently due to content above 16kHz were due to ITD in the equipment producing actual changes below 16kHz in the output.
At the moment, the conclusions that have been drawn in reference to encoder testing and HF that you refer to are valid only for the set of circumstances -- but isolated perceptual data, in a controlled setting, must be aquisitioned if you want to be certain of that variable's specific contribution. Several conditions could also be tested to account for the speculations we have both made RE: HF content, etc.. However, when I am forced to pick between some tests not isolating factors or establishing a set of controls vs. a peer reviewed JAES article/research, it's no contest as to which one I will consider as more credible/useful. It is possible, of course, that some very young people with very HF ability could detect the difference with music that has unusual power distribution into the HF spectra. The article also concluded that(with the even a slightly lower point filtering that was slightly audible) that a lack of the highest frequencies was not detected as a sound quality decrease in blind tests.
Could you link to the test files used in this ABX test to which you refer? I would, as a matter of seperate interest, be interested in taking a look and listen.
Thanks.
-Chris
i'm sure i'll change my opinion if i get a really good 5.1 system, but right now i'll settle for an above-average 2.0 system (movies still sound stunning on anything capable of decent dynamics and flattish response).
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Too bad that the benefit of surround is dependant on competant mixing of the surround channels by the same illustrious folks who bring you the *excellent* average 2 channel recordings of today. Cough. Cough.
-Chris
It's a fact, not a supposition..
It's well known that the audio CD is completely overkill for the human ear. The normal audio CD have more resolution that the human ear is able to perceive.
I think each the SACD and DVD-A camps make claims that its format is insistinguishable in tests from live feed while the other isn't. At least I have heard both claims. I am not sure who has tested this with CD, or how else your result came to be "well known". Take a reference ADC to convert to 16/44 and reference DAC and compare to live analog or else take 24/192 recordings and convert to 16/44 to compare.
It's a fact, not a supposition..
It's well known that the audio CD is completely overkill for the human ear. The normal audio CD have more resolution that the human ear is able to perceive.
I think each the SACD and DVD-A camps make claims that its format is insistinguishable in tests from live feed while the other isn't. At least I have heard both claims. I am not sure who has tested this with CD, or how else your result came to be "well known". Take a reference ADC to convert to 16/44 and reference DAC and compare to live analog or else take 24/192 recordings and convert to 16/44 to compare.
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I believe that you misread important posts of this thread.
You would have that to read the post #43, where, your doubt was answered clearly.
You can’t say the difference between a lossy at around 200 kb/s and a PCM 16/44 at 1411 kb/s. It means pcm 16/44 is overkill for the human ear. I doubt you will get much support to claim the contrary here at HA.
Outside the compatibility, the benefits of pcm being lossless, meaning you have the choice of the lossy encoder without transcoding artefact. You still have lossless compression at around 800 kb/s.
I don't see how that answers the question.
Could you link to the test files used in this ABX test to which you refer? I would, as a matter of seperate interest, be interested in taking a look and listen.
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According to my experience, real-world music that reveals a difference when lowpassed at 16 KHz is not that frequent, thanks to spectral masking. But there is certainly some. One example is the first sample here, the one lowpassed at 16 KHz:
[a href="http://www.kikeg.arrakis.es/lowpass/]http://www.kikeg.arrakis.es/lowpass/[/url]
It's a sample extracted from a pop music cd, where there is a cymbal ringing just over 16 KHz. I can hear the lowpass at ordinary listening levels without any effort, using both monitor speakers and various kind of headphones (Sony MDR-7506, Sennheiser HD-580).
I've come accross at least another example of this type, but I have no samples available online. If you are interested, I'll post them when I have some time.
Edit: now it's two different samples that reveal a difference when lowpassed at 16 KHz.
From my experience, an instrument like harpsichord reveals without excessive difficulties a 16 KHz lowpass.
I thought it was established scientific fact, that humans can hear pure sinusodial signals up to ca. 20kHz. In that regard, I find it a hard claim that in musical content anything above 16kHz is allways masked.
According to my experience, real-world music that reveals a difference when lowpassed at 16 KHz is not that frequent, thanks to spectral masking. But there is certainly some. One example is the first sample here, the one lowpassed at 16 KHz:
http://www.kikeg.arrakis.es/lowpass/ (http://www.kikeg.arrakis.es/lowpass/)
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That's a pretty amazing sample, I had no trouble ABXing 10/10 of the lowpassed version versus the original. On the other hand, I failed to ABX the 16kHz lowpass of ff123's Mustang Sally samples. Interestingly, my girlfriend (same age as me, 21) failed completely on this sample - she is convinced their is no difference. I blame that on the fact that she has been playing in an orchestra for six years.
I thought it was established scientific fact, that humans can hear pure sinusodial signals up to ca. 20kHz. In that regard, I find it a hard claim that in musical content anything above 16kHz is allways masked.
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Most of the time, it appears that data >16Khz is not important. However, in this sample provided by the poster, it has extraordinary HF content, which basicly resembles a sine wave for a signficant duration that is >16Khz frequency.
-Chris
According to my experience, real-world music that reveals a difference when lowpassed at 16 KHz is not that frequent, thanks to spectral masking. But there is certainly some. One example is the first sample here, the one lowpassed at 16 KHz:
http://www.kikeg.arrakis.es/lowpass/ (http://www.kikeg.arrakis.es/lowpass/)
Thank you. This sample seems to have an extraordinary type of HF content; essentially a relative long duration signal(synthetic cymbal?) of high amplitude that resembles a sine wave that is >16kHz in frequency. The perceptual test to which I referred to determine JNDs did not use this sort of signal. But your sample does show a superb example of the few times that the limit needs to be extended >16Khz. It is noted that the developers of redbook still decided that a bandwidth of about 22kHz was needed for a *safeguard* in the final standard. Maybe they feard this very type of sample would show up occasionally?
-Chris
According to my experience, real-world music that reveals a difference when lowpassed at 16 KHz is not that frequent, thanks to spectral masking. But there is certainly some. One example is the first sample here, the one lowpassed at 16 KHz:
http://www.kikeg.arrakis.es/lowpass/ (http://www.kikeg.arrakis.es/lowpass/)
Thank you. This sample seems to have an extraordinary type of HF content; essentially a relative long duration signal(synthetic cymbal?) of high amplitude that resembles a sine wave that is >16kHz in frequency. The perceptual test to which I referred to determine JNDs did not use this sort of signal. But your sample does show a superb example of the few times that the limit needs to be extended >16Khz. It is noted that the developers of redbook still decided that a bandwidth of about 22kHz was needed for a *safeguard* in the final standard. Maybe they feard this very type of sample would show up occasionally?
-Chris
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Even at 39 it's easily abxable
Most of the time, it appears that data >16Khz is not important. However, in this sample provided by the poster, it has extraordinary HF content, which basicly resembles a sine wave for a signficant duration that is >16Khz frequency.
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I just encoded this sample (which is also very easily ABXable for me when lowpassed at 16kHz) with Lame 3.90.3 --alt-preset standard -Y, and according to the spectral view, Lame preserves most of the HF energy in the critical places; I can't ABX this encode, which shows that the -Y switch is very intelligent in its decision how to spend the few bits it can for HF content.
Here are the bitrates:
APS: 208 (can't ABX)
APS -Y: 199 (can't ABX)
APS --lowpass 16: 196 (very easy to ABX)
What I'm missing in this interesting discussion so far is the distinction between "production format" and "delivery format".
A production format is used to store audio (usually the output of one or more microphones) for future use in the production process. It is very likely that there will be subsequent processing and therefore overkill in audio quality is required (and legitimate imo).
A delivery format is used to provide the final (mastered) content to the consumer. Subsequent processing of the audio is not intended although it can't be completely excluded (radio broadcasting, consumer tools like EQ, dsp etc.). Therefore the quality requirements for a delivery format can be lower (no overkill).
Another item is the difficulty in separating the quality of a format and its implementation.
For DSD DAC's e.g. choice is rather limited at the moment and no matter how you compare models, there will be too many variables to draw objective conclusions.
If you want to listen to a DSD ADC, you'll need a DAC as well, thereby increasing the variables again. You're testing the implementation, not only the format.
Therefore I think that objective testing, which is relatively easy when you stay in the digital domain, becomes almost impossible when it comes to transducers. Good tests will be very time and budget consuming, probably exceeding the financial limits of this industry.
For me personally SACD is the best delivery format for surround audio at this moment. The hybrid format is great and several cd-plants can deliver the format (at least here in Europe).
Just my € 0,02
I am very interested in this forum, and you will have to excuse me for my lack of technical savvy on this topic. I am not quite up to speed on all of the lingo which is being used.
I look at this a little differently, and yet share some of the same views as some of you.
When I buy music, or movies. I objectively listen for all parts of it. I have been a musician for 2/3 of my life and grew up with an old school audiophile father. I can appreciate the different formats and for this reason I question the guys on here who soley resort to their math skills.
First off.
Who decides which sounds are, or aren't, important? It is my opinion that it is the complete sound which recreates the music, and omission of any of that data is therefore compromising the sound quality.
My father is 54, has over $150,000 invested in his setup and he still to this day goes for vinyl. I still remeber the first time I heard Dark Side of The Moon on vinyl.
The sound was far different from any other format I had heard it on, mainly because the records don't have the limitations of many of the other formats.
For this reason, I say that anyone who says you cannot "hear" the difference between the same recording on CD & SACD can kiss it. I believe hybrid discs are a great testament to this. I have a very modest setup in comparison to that of my father, but I am using sony es and nht 2.5i for fronts. Listen to the remastered Dark Side of The Moon on SACD/CD Hybrid, it was engineered by the ORIGINAL engineer and both recordings thus should be from the same source material. This to me provides an obvious win for SACD, as its' clarity and accuracy in representing the original audio is far greater than that of CD.
Off of that rant, I will say, I have not been able to obtain a recording in CD,SACD,and DVD-A. But to date SACD has best been able to make me feel as if it were live in my living room. Also, I find that a DTS DVD CD sounds just as good if not better that DVD-A (this of course depends on the source and engineer).
So, can anyone answer this. Why create the DVD-A format, when the DVD-V format does just as good.
For me, the choice is still SACD.
Good post, but can you ABX it? ABX is not about the maths, only the truth.
It has already been covered that the SACD/CD sides of DSOTM have different content.
It is my opinion that it is the complete sound which recreates the music, and omission of any of that data is therefore compromising the sound quality.
This is exactly why people use ABX tests
The problem with this whole SACD/DVD-A/CD issue is, that we CAN't do any meaningful objective ABX test at our home "conditions". There are too many variables that could alter the outcome of such a test.
Let's say you have got two discs, one DVD-A the other SACD, with the same content, with the 'SAME' mastering, both originally recorded in analog and then transfered to the respective audio format with SOTA converters. Now you want to do an ABX test, but with which equipment? The processing of the digital signal and it's conversion into analog is so different in both cases that if the outcome of the test would be positive, then we can conclude nothing about the format, just about the used electronics.
A consumer can never approach the ideal state. With their use of some universal player and DVDA and SACD discs of the same material he or she can get some results, but it won't tell anything about the formats.
I won't discuss here which one of these formats is theoretically superior. Many heated words have already been spoken. As everybody knows, practice is very different from theory. A theoretically better method can yield poorer results in real-world use.
However one practical test could be conduced. Let's gather as many as possible DVD-A and SACD players of different price ranges and let's conduce an ABX test with identical source discs. This comparison could tell something about real possibilities of both formats in everyday employment.
The above test falsely entitled "practical" is quite difficult for normal consumer to achieve. So we are quite left in the dark...
Now to the even more sensitive issue - is there any ABXable difference between:
1) good-old 16/44.1 PCM and let's say 24/96 or even 24/192?
2) 16/44.1 PCM layer and DSD layer of the hybrid SACD disc?
I won't cover the first point, since I don't have much experience with such comparison and some tests have been already made (with negative results outweighting ).
The second point is practically feasible as a single or double-blind test. The necessary preconditions are:
- SACD player with DSD and PCM stages as much as similar (utilising the same DACs) with no conversion of the DSD bitstream to the PCM (bass management, etc.)
- Hybrid SACD disc derived from an original analog or DSD recording (analog prefered, to keep off any bias), with no usage of aditional effects in the PCM domain, overdubing, etc. with exactly the same material used for the PCM layer (this is absolute necessity - DSOTM and other discs with altered PCM layer is a joke for this test) and with the same loudness (very important). Classical recordings captured with few mikes and with few alterations on the engineer side are great candidates.
- Quick switching of the DSD/PCM layer in the SACD player. With the old players the layer switching could take for ages, but with the recent ones it's usually below 5 sec.
I've managed many single-blind listening tests consistent with the above scenario in my headphone setup (Modified Sony SCD-XA3000ES player, Dynahi headphone amplifier, HD650 headphones) with a help of a family member who switched randomly the layers.
Because of the TOS of this forum I cannot post any results of these tests, since there is no way how can I prove my claims in this virtual space (silly isn't it? - alternatively you can come by and see (listen) for yourself ). I'll do a little infringement and only say that the tests were in the case of original analog or DSD recordings mostly positive, in the case of PCM sourced discs I often failed.
Forget about the PCM layer on a SACD.
Listen to the remastered Dark Side of The Moon on SACD/CD Hybrid, it was engineered by the ORIGINAL engineer and both recordings thus should be from the same source material. This to me provides an obvious win for SACD, as its' clarity and accuracy in representing the original audio is far greater than that of CD.
This is an interesting example. If you record the analog signal that comes from the DSD layer to a cd, with eg a stand alone cd recorder, then you must conclude that this PCM version is far better (maybe indistinguisable from the SACD) than the PCM layer on the SACD!! If you ask me the PCM layer of the SACD is the 20th anniversary edition and not the 30th anniversary.
Regards,
Jacco
If there's any audible difference (equipment aside) between DVD-A and SACD, then surely DVD-A has to be the better one, simply because it has more data. It might not sound better, but I can't see how it could sound worse. By good I mean more accurate, of course.
If there's any audible difference (equipment aside) between DVD-A and SACD, then surely DVD-A has to be the better one, simply because it has more data. It might not sound better, but I can't see how it could sound worse. By good I mean more accurate, of course.
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No offense intended, but I think you might want to reconsider your logic.
The accoustical accuracy of the media cannot be determined merely by the size of the data disc. Liken this to a computer program.
If two programmers write two completely different programs to perform the exact same function, the programs will not necessarily be the same. Much data can be erroneous. Your program will only be as good as the person who wrote it, and will only be efficient as the person who wrote it.
Who really cares about the capacity of the media in terms of data? Quite frankly, I'm not considering this when I listen to my music, nor do I care about the size of the files that comprise it. Give me media the size of laser-disc, I dont care. That will only effect production/distributions costs. I WANT GREAT SOUND, not an effecient compression.
Earlier in this forum, it is referenced from technical docs that DVD-A is the more effecient of the two, but lacks in certain aspects. See post #9.
SACD may sound better because of different mastering and 5.1 channel distribution, not because it has greater resolution. In fact, DSD is not better than 44.1 KHz PCM, just read this: http://sjeng.org/ftp/SACD.pdf (http://sjeng.org/ftp/SACD.pdf)
And that is the answer to everything. 8)
Listen to the remastered Dark Side of The Moon on SACD/CD Hybrid, it was engineered by the ORIGINAL engineer and both recordings thus should be from the same source material. This to me provides an obvious win for SACD, as its' clarity and accuracy in representing the original audio is far greater than that of CD.
Careful. Just because it was mastered by the same engineer does not in any way mean it was mastered the same. It is virtually impossible to do so. First, the mastering tools available for the DSD environment are different from those available for PCM. Second, traditional PCM mastering involves normalizing and limiting in ways that are actually illegal in the SACD scarlet book - the disk would be rejected. The result is that mastering engineers on SACD are actually forced to use less compression and allow more dynamics. It is not to say that the PCM mastering engineer couldn't do the same, but they don't HAVE to, so they don't.
The result is that SACD disks often use more dynamic range, have less compression, and sound that way. But they don't have to...
Nika
Listen to the remastered Dark Side of The Moon on SACD/CD Hybrid, it was engineered by the ORIGINAL engineer and both recordings thus should be from the same source material. This to me provides an obvious win for SACD, as its' clarity and accuracy in representing the original audio is far greater than that of CD.
Careful. Just because it was mastered by the same engineer does not in any way mean it was mastered the same. It is virtually impossible to do so. First, the mastering tools available for the DSD environment are different from those available for PCM. Second, traditional PCM mastering involves normalizing and limiting in ways that are actually illegal in the SACD scarlet book - the disk would be rejected. The result is that mastering engineers on SACD are actually forced to use less compression and allow more dynamics. It is not to say that the PCM mastering engineer couldn't do the same, but they don't HAVE to, so they don't.
The result is that SACD disks often use more dynamic range, have less compression, and sound that way. But they don't have to...
Nika
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I understand this, that is why I referenced the Dark Side of The Moon Hybrid. Both layers are 30th anniversary remasters done by the original engineer.
Multi-Channel aside, compare the two 2 channel recordings.
The reason I referenced the CD was to address earlier posts by people who try to use the "scientific limitations of the human ear" to excuse the fact that they are unable to distinguish between CD and SACD.
In fact, DSD is not better than 44.1 KHz PCM
As far as I understood the dynamic range is about 30 dB larger for SACD, but it is not distortion-free. Due to the inability of proper dithering.
DVD-A has even more dynamic range and can be made distortion-free.
Regards,
Jacco
Jacco...
you seem very knowledgeable on DVD, so I'll ask you. What is the technical difference in DVD-V and DVD-A in terms or the capabilities?
From page #4:
Finally, consider 8-bit, four-times-oversampled PCM with
noise shaping. This is also a data rate one-half that of DSD and
double that of CD, with a sampling rate of 4 × 44,100 =
176,400 Hz. It can achieve a noise floor 120 dB below full
scale up to 20 kHz, using 96 dB of noise shaping, and a total
noise power of –19 dBFS. Its frequency response would be
flat to 80 kHz. This example is perhaps the most instructive of
the lot. For a data rate one-half that of DSD, it achieves a
comparable signal bandwidth, with a similar noise power
density up to 20 kHz, but much lower power above this
frequency, and 28 dB lower total noise power. It is fully
TPDF-dithered, and so is completely artefact free. At one-half
the data rate it outperforms DSD on every count! DSD is a
profligate wastrel of capacity.
Where it boils down to is the data rate, resulting in a limited amount of high resolution multi channel playback for DVD-V. Not all DVD-V players can handle even the stereo high resolution track. With DVD-A you can go up to 6 channels 24 bit 96 kHz lossless audio.
I truly believe that 16 bit 44k1 is good enough and it should be possible to put a 6 channel 16 bits 44k1 stream on a DVD-V, taking away the necessity of SACD and DVD-A.
Regards,
Jacco
DSD is a
profligate wastrel of capacity.
True. But this statement does not relate cd with sacd in the sense that 44k1 PCM is better than DSD. It is more efficient.
Regards,
Jacco
I understand this, that is why I referenced the Dark Side of The Moon Hybrid. Both layers are 30th anniversary remasters done by the original engineer.
Multi-Channel aside, compare the two 2 channel recordings.
The reason I referenced the CD was to address earlier posts by people who try to use the "scientific limitations of the human ear" to excuse the fact that they are unable to distinguish between CD and SACD.
This is still, however, to imply that the two releases are identical other than the fact that one is encoded to DSD and the other to more traditional PCM. This is an assertion that I do not believe it safe to make.
And even if it were, a gentleman above pointed out various other reasons that an AB test between the two cannot be conclusive with respect to the formats themselves - only to the very specific (and heavily variable laden) equipment and situation that was tested.
Nika
Jacco,
do you know of any DVD-V recorded in high res? I would like to locate these and play some!
In fact, DSD is not better than 44.1 KHz PCM
As far as I understood the dynamic range is about 30 dB larger for SACD, but it is not distortion-free. Due to the inability of proper dithering.
DVD-A has even more dynamic range and can be made distortion-free.
Regards,
Jacco
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We have to specify a frequency range before we can discuss dynamic range. SACD has only 6dB of dynamic range if we talk about the broadband capabilities of the system (from 0Hz to 1.411MHz) but in narrow frequency bands the dyanmic range is much greater. For example, in the audible region of 0Hz to 20kHz the dynamic range is often somewheres between 108dB and 120dB. If we look at, say, up to 100kHz of frequency range (though I don't know why we would) the dynamic range drops precipitously.
DVD-A with good converters has approximately the same dynamic range in the audible band but above the audible band (again, why do we care?) it is greater.
Indeed, DSD has inherent distortion problems, especially noticeable with low level signals due to the lack of dither in the feedback loop of the delta sigma modulator.
Nika
I found this quote from "Why 1-Bit Sigma-Delta Conversion is Unsuitable for High-Quality Applications, Stanley P. Lipshitz and John Vanderkooy" very interesting indeed (originally pointed to by Mo0zOoH).
The high levels of ultrasonic noise and spuriae produced by an inadequately-filtered 1-bit sigma-delta converter pose a problem for audio amplifiers and loudspeakers, which can generate nonlinear distortion products in the baseband when subjected to this type of indignity. One wonders how many of the perceived “differences” noted in Super Audio CD listening comparisons might be due to such nonlinear effects.
It suggests that if successful ABX tests are performed to show that DSD sounds different from PCM, then it would be worthwhile repeating the test with a low pass filter before the amplifier. It is well understood that inaudible high frequency signals can cause audible distortion in tweeters and power amplifiers.
It suggests that if successful ABX tests are performed to show that DSD sounds different from PCM, then it would be worthwhile repeating the test with a low pass filter before the amplifier.
Yes, of course. And if you use an analog one (as recommended by Sony) then you end up with phase distortion which should be audible.
And if you use a digital, linear phase, low-pass filter to do the same then you end up with....
If there's any audible difference (equipment aside) between DVD-A and SACD, then surely DVD-A has to be the better one, simply because it has more data. It might not sound better, but I can't see how it could sound worse. By good I mean more accurate, of course.
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No offense intended, but I think you might want to reconsider your logic.
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No offense intended, but I think you might want to read up on the Nyquist theorem.
Earlier in this forum, it is referenced from technical docs that DVD-A is the more effecient of the two, but lacks in certain aspects. See post #9.
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Excuse me? Post #9 (from Garf) does not mention anything about DVD-A lacking in certain aspects. Quite the contrary, Garf clearly stated his preference for DVD-A.
If there's any audible difference (equipment aside) between DVD-A and SACD, then surely DVD-A has to be the better one, simply because it has more data. It might not sound better, but I can't see how it could sound worse. By good I mean more accurate, of course.
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No offense intended, but I think you might want to reconsider your logic.
[a href="index.php?act=findpost&pid=304824"][{POST_SNAPBACK}][/a]
No offense intended, but I think you might want to read up on the Nyquist theorem.
Just to try to put this one to bed, indeed whether or not the result is correct, the logic expressed by tgoose is suspect. We cannot say that just because one format uses more data that it will inherently sound better. Reductio ad absurdum: a 192bit 10kS/s sample rate will sound better than 24 bit 48kS/s.
DVD-A uses more data but this unto itself cannot be used as an indicator of whether or not it is superior to competing formats. Coastalalbum was correct to call this logic into question.
Nika
When I read it back, what I said isn't quite what I meant. Reading the technical info a bit, what I meant isn't right either, but it's more right than what I said .
I was saying it based on an idea that it'd be possible to losslessy convert DSD to some sort of linear PCM - if it were, it would have both a lower bitrate and sample rate than high end DVD-A, and so, forgetting any resampling problems incurred (or put another way, assuming the sample rate is a factor of DVD-A's sample rate), it wouldn't be possible for it to sound better.
Hmph, that's even less sensical than before.
coastalbumm,
do you know of any DVD-V recorded in high res?
No, but I suppose you search for lossless DTS disks. However, it can be the case that these disks are not compatible with all DVD-V players.
it would be worthwhile repeating the test with a low pass filter before the amplifier
In practise this is the case at 80 kHz or so. Other players such as the DVD963SA uses an even dramatic filtering: they use different filters for the Front and Surround/Center/LFE channels. One with a cutoff of 50 kHz and the other with 40 kHz. Unbelievable.
We have to specify a frequency range before we can discuss dynamic range.
Right. Up to 20 kHz I would say, because I cannot hear signals above 20 kHz at normal listening levels. Furthermore, the 120 dB in the audible range should be seen as a system specification. I have seen with my own eyes that such a converter is designed as such, without stating that every DSD converter is made like that! Also the distortion is kept out of the audible range, but it is not removed especially at frequencies closer to Nyquist.
Regards,
Jacco
Also the distortion is kept out of the audible range[a href="index.php?act=findpost&pid=305006"][{POST_SNAPBACK}][/a]
Stanley P. Lipshitz and John Vanderkooy disagree, as you should well know!
Cheers,
David.
Stanley P. Lipshitz and John Vanderkooy disagree, as you should well know!
Oh yes, pardon me. I was not talking about a standard 2 level quantised system such as DSD as it is posed in the beginning. After the paper of Lipshitz and others it became clear that we have been "mislead", so to speak. After that improvements in DA converter design are proposed (I am not fully aware of the technical inside details) to overcome the distortion problem in the audible range. These distortion components are not removed (as well as the noise modulation) but can be made as small as wanted.
This is a far cry from the originally claimed super-audio performance. Also I made big mistakes in the beginning, you can read them elsewhere on this forum. DVD-A systems are better, in theory. In fact, every 3 or 4 bits quantizer based PCM system is in theory better in the sense of distortion and noise modulation provided that they are properly dithered.
Regards,
Jacco
I'm not sure if you guys have finished arguing the range of human hearing. here is a link.
http://www.norsonic.com/web_pages/human_hearing.html (http://www.norsonic.com/web_pages/human_hearing.html)
Also, babies and small children are the only peple who can hear 20hz-20K
the rest of us have lost some hearing.
The sound was far different from any other format I had heard it on, mainly because the records don't have the limitations of many of the other formats.
On the contrary, records have the severe limitations. But, you can not compare two different formats fairly using standard commercial releases/versions of the same albums. They will probably be different. Nothing guarantees they will be the same.
For this reason, I say that anyone who says you cannot "hear" the difference between the same recording on CD & SACD can kiss it.
Nice.
Let's examine the difference(huge compression differences) between Diana Krall's "The Look of Love" CD track samples vs. SACD track sample.
CD:
(http://www.audioholics.com/techtips/specsformats/images/clip_image001.gif)
SACD:
(http://www.audioholics.com/techtips/specsformats/images/clip_image003.gif)
The CD version is purposely designed to be bad/different in this case, as compared to the SACD version. The dynamics were squashed so that they could increase loudness as far as possible. But, many CDs seem to be designed to be purposely bad these days. Not a problem with the format. A problem with the morons mastering to the format.
-Chris
The CD version is purposely designed to be bad/different in this case, as compared to the SACD version. The dynamics were squashed so that they could increase loudness as far as possible. But, many CDs seem to be designed to be purposely bad these days. Not a problem with the format. A problem with the morons mastering to the format.
-Chris
[a href="index.php?act=findpost&pid=306708"][{POST_SNAPBACK}][/a]
I don't believe we can say for certain that the CD version is purposely designed to be bad/different. The CD version is not limited by modulator overload. It is possible to clip a compact disk even when all of the samples are legal - the signal, when reconstructed, can exceed full scale even though the samples don't. Since D/A converters can't reconstruct above full scale, it is therefore possible to create a compact disk with "illegal" content in that the signal "clips" all over the place when reconstructed despite adhering to redbook standards and having no "digital overs" by definition. This phenomenon happens far more often when significant compression/limiting (and of course normalizing) is used. With all of the compression used in CD mastering and with the loudness wars still happening there can be no surprise that CDs suffer from a potentially "brittle," distorted sound. Not only is the compression and heat itself fatiguing to the ears, but the distortion on top of it can be troublesome. This is again not inherently a limitation of the format (the CD itself) but is really more a limitation of the way in which it is utilized. Of course I don't think any mastering engineer would complain if the redbook standard were suddenly to change to prevent inter-sample digital overs, essentially forcing the industry to adopt either a quieter level on the disk or more dynamic range.
[pause and digest that]
[/pause and digest that]
Funny, that's exactly what the SACD format does. It has a mathematical formula in it (if you will) that prevents digital overs between samples so that the modulator never has to try to reconstruct a signal that could cause it to overload. In the SACD "scarlet book" this is determined by any 28 consecutive identical sample values (28 1s or 28 0s in a row). Such a signal, when reconstructed, is equivalent in the example above to exceeding full scale and clipping the converters.
Therefore, the SACD does not allow that hyper-compression and limiting that CDs can (illegally) allow, forcing the mastering engineer to take one of two approaches:
1. Either hyper compress it and limit it but then turn it down a respective amount (essentially defeating the entire purpose of limiting it in the first place, no?), or
2. Let it breathe and allow it to have the dynamics.
Which choice do you think mastering engineers are taking? We can look at the Diana Krall track above as an indicator. Can there be any surprise that people generally prefer the SACD version of tracks to their hyper-compressed, distorted, (inter-sample) clipping CD tracks? Is this an inherent benefit to SACD or just a misuse of CD?
Nika
I don't believe we can say for certain that the CD version is purposely designed to be bad/different. The CD version is not limited by modulator overload. It is possible to clip a compact disk even when all of the samples are legal - the signal, when reconstructed, can exceed full scale even though the samples don't. Since D/A converters can't reconstruct above full scale, it is therefore possible to create a compact disk with "illegal" content in that the signal "clips" all over the place when reconstruct
The CD was definately compressed more than the SACD, as can be seen in the waveforms. This, in my book = bad, as an automatic qualification. And it had to be purposely designed to sound different. The compression difference is subtantial(enough to be easily seen in waveform windows). It's not as if the compression happened all by itself on the CD: a mastering engineer dialed it in to be that way
on purpose. You refer to the errors of some DACs, and how they differentiate when approaching 0dBfs. It is easy to prevent the anomolies that can occur from different DACs when they approach 0dBfs: don't allow signals to come close to 0dBfs. Keep peak signals 0.3dBfs or lower. If the recording was modified(compressed/limited) in the first place to try and be as loud as possible on the CD, but was left alone for the SACD; then it was purposely designed to sound different between the two examples.
-Chris
The CD was definately compressed more than the SACD, as can be seen in the waveforms. This, in my book = bad, as an automatic qualification. And it had to be purposely designed to sound different.
Chris,
I think the difference in our opinions has to do with intent. It is most likely that the two pieces were mastered independently. The CD version may have been mastered first, but even if it wasn't, it was mastered the typical way CDs are mastered, with no intent of TRYING to get it to sound like the SACD. So it was overcompressed just like the rest of CD mastered material on the market today.
The SACD version was mastered probably with the intent of getting it as hot as possible, but the restrictions in this force it to maintain more dynamic range when it was.
I really think it is unlikely (am I an optimist?) that some record exec said, "Hey, master this one well and make this other one sound better." I think there was probably no intent of getting them to sound different or to skew toward the SACD version. It is more likely, I believe, that that's just the way it came out.
Nika
I really think it is unlikely (am I an optimist?) that some record exec said, "Hey, master this one well and make this other one sound better." I think there was probably no intent of getting them to sound different or to skew toward the SACD version. It is more likely, I believe, that that's just the way it came out.
[a href="index.php?act=findpost&pid=306995"][{POST_SNAPBACK}][/a]
Call me paranoid, but the fact that record companies would be glad to phase out CDs sooner rather than later due to missing DRM/copy protection capabilities AND the fact that CD audio is already overkill for 99% of people on 99% of all equipment AND the fact that SACDs can be sold with a higher mark-up makes it very likely for me that CDs are screwed up on purpose to fool people into buying SACD/DVD-A
Chris,
I think the difference in our opinions has to do with intent.
I really think it is unlikely (am I an optimist?) that some record exec said, "Hey, master this one well and make this other one sound better." I think there was probably no intent of getting them to sound different or to skew toward the SACD version.
Yes, there appears to be a difference in our interpretation of intent. But what does not help, is that about a year ago, I had some e-mail discussions with a recording engineer(Michael Bishop) regarding a CD release(Tierney Sutton, Album: Dancing In The Dark) that he engineered. I was concerned about the
horrible sound quality. By horrible, I mean that it was very compressed and even had *audible digital clipping(!) on some parts of the recordings(clipping readily available on waveform windows, very audible hash/static sound where these clips occur, and the voltage overshoot[exceeding 0dBfs] on clipping from a CD player DAC is confirmed with a digital scope). This behaviour was confirmed to exist on two different copies of the CD, obtained several months apart, and from different states. Back to the discussion with the engineer: Mr. Bishop disclosed that the CD version was 'compromised' so that it would sound as loud as popular CDs, if playing one after another in a disc changer, for example. He denied that the CD was clipped in any spot -- but a simple exam of the ripped CDDA data demonstrates that portions are clipped--no question. Now, please consider that this record company in question is Telarc. An audiophile label! This has biased me, and I automaticly assume the worst now.
-Chris
* TOS #8 violation is possible with this claim. But an ABX is not possible in this situation, so please allow this exception. Since a DBT was not possible, I did this: I listened to a few suspect tracks and wrote down the time positions where I believed to hear digital clipping. I then checked the waveforms using these written down time positions, to discover that the waveforms were clipped at the positions I wrote down. I will upload a clip of one of the worst occurances on the CD, but I can't at the moment. I can't access my ftp(internet problems) at the moment.
Update: I have now uploaded a sample section from the album referenced above:www.linaeum.com/downloads/misc/ts_ditd_t10_clipsample.wav
DVD-A is nothing but a CD, with the same idea and principles of signal coding, just with increased parameters. SACD has different principles, and it is closer to analog recording. It was designed for recording studios to store there tape archives and for audiophile listeners. Studios never used and will never use CD or DVD-Audio for serious backups.
Call me paranoid, but the fact that record companies would be glad to phase out CDs sooner rather than later due to missing DRM/copy protection capabilities AND the fact that CD audio is already overkill for 99% of people on 99% of all equipment AND the fact that SACDs can be sold with a higher mark-up makes it very likely for me that CDs are screwed up on purpose to fool people into buying SACD/DVD-A
[a href="index.php?act=findpost&pid=307005"][{POST_SNAPBACK}][/a]
That's a bold statement and I disagree. This loudness war has started long before these new formats were known.
Perhaps you're underestimating the importance of having the music sound as loud as possible.
Even in the old, analog times, mastering engineers tried to push the medium to its limits, partly for signal to noise reasons, but also for increased impact when played next to competitors music.
With all other things being equal, a louder version of a track is usually preferred over the softer one. Engineers know this and try to go as far as possible, even if this means increased distortion.
Just do the test yourself. Make a playlist of say 10 of your favorite songs and reduce the level of just one somewhere in the middle by 3 or 6 dB and play the entire(!) list again. The softer song will sound different and, probably, less interesting. Not by itself, but in the context.
Consumers are now used to randomly play thousands of different tracks on their mp3-jukebox and it's the task of a mastering engineer to make sure his/her version sounds as good as possible to 99% of the people. A difficult task.
* TOS #8 violation is possible with this claim. But an ABX is not possible in this situation, so please allow this exception. Since a DBT was not possible, I did this: I listened to a few suspect tracks and wrote down the time positions where I believed to hear digital clipping. I then checked the waveforms using these written down time positions, to discover that the waveforms were clipped at the positions I wrote down. I will upload a clip of one of the worst occurances on the CD, but I can't at the moment. I can't access my ftp(internet problems) at the moment.
[a href="index.php?act=findpost&pid=307066"][{POST_SNAPBACK}][/a]
Digital clipping can be shown on a good digital peak-level meter with calibrated overload indication. The number of consecutive clipped samples (full scale) is an indication for the amount of audible clipping. No clipped samples means the audio can be reconstructed with a good DAC.
One or more FS samples indicate clipping, but it doesn't have to be audible.
I'd gladly check your files for digital clipping and even try to remove the clipping if you like. The music will be softer then, but that's exactly the mastering engineer's dilemma.
Call me paranoid, but the fact that record companies would be glad to phase out CDs sooner rather than later due to missing DRM/copy protection capabilities AND the fact that CD audio is already overkill for 99% of people on 99% of all equipment AND the fact that SACDs can be sold with a higher mark-up makes it very likely for me that CDs are screwed up on purpose to fool people into buying SACD/DVD-A
[a href="index.php?act=findpost&pid=307005"][{POST_SNAPBACK}][/a]
That's a bold statement and I disagree. This loudness war has started long before these new formats were known.
Perhaps you're underestimating the importance of having the music sound as loud as possible.
[a href="index.php?act=findpost&pid=307097"][{POST_SNAPBACK}][/a]
I agree that there are other (and certainly equally important) factors that affect the loudness of CDs, but still the companies have good reasons for pushing SACD/DVD-A. Their main selling point is that they sound much better than CDs, which is plainly wrong. They
may sound better with some kinds of music on high-end systems and with experienced listeners, but it's really ridiculously little advancement which in no way justifies changing systems on the part of consumers as it was the case with Vinyl vs. CD. So if an album is released simultaneously on CD and SACD, the record company has to make sure that the SACD version sounds better in order to justify its existence. Since this is only possible by making the CD worse than it could be, even with musical genres that are usually not much affected by the loudness race, they do it. At least that's what
I would do if I were in their place
ok, so, the debate rages on between formats... ive a question then.
What would be the idea way to try to represent analogue sound in a digital medium? How can we duplicate, with optimal results, real sound in digital format?
ok, so, the debate rages on between formats... ive a question then.
What would be the idea way to try to represent analogue sound in a digital medium? How can we duplicate, with optimal results, real sound in digital format?
[a href="index.php?act=findpost&pid=307126"][{POST_SNAPBACK}][/a]
SACD is the first step to do that.
You can find technical info regarding SACD's coding in the web.
Well the truth is that PCM is actually a very ideal format. The problem is that many knowledgeable fools make fantastic pages like this (http://www.mother-of-tone.com/cd.htm) that make people misunderstand the way PCM works.
Remember maths class when you find that there is only one parabola that can fit through 3 given points? Well PCM is working on this principal, using the sampled points to guide a continuous waveform. It is -not- the jagged and "digital" (but good enough) approximation that is often described.
Remember maths class when you find that there is only one parabola that can fit through 3 given points? Well PCM is working on this principal, using the sampled points to guide a continuous waveform. It is -not- the jagged and "digital" (but good enough) approximation that is often described.
[a href="index.php?act=findpost&pid=307136"][{POST_SNAPBACK}][/a]
Right. The only problem are "sampled points" and there 16-bit distance . All the rest is fine .
If they think that
louder is better, why would they make CD louder if they want SACD to sound better ??
filtering at 19.5kHz was audible to one person who took part in those tests!
[{POST_SNAPBACK}][/a] (http://index.php?act=findpost&pid=286619")
Filtering at 28 kHz was audible for this person : [a href="http://www.hydrogenaudio.org/forums/index.php?showtopic=17118&view=findpost&p=198806]http://www.hydrogenaudio.org/forums/index....ndpost&p=198806[/url]
at this point no confirmed perceptual tests have concluded that anything exceeding a [1]16kHz bandwidth is needed to be transparent for musical program playback for humans. [{POST_SNAPBACK}][/a] (http://index.php?act=findpost&pid=286167")
What about this one ?
Yoshikawa, Shokichiro; Noge, Satoru; Yamamoto, Takeo; Saito, Keishi
Does High Sampling Frequency Improve Perceptual Time-Axis Resolution of Digital Audio Signal?
Preprint 4562 in
[a href="http://www.aes.org/publications/preprints/search.cfm]http://www.aes.org/publications/preprints/search.cfm[/url]
What about this one ?
Oops, sorry, it doesn't deal with
musical program
If they think that louder is better, why would they make CD louder if they want SACD to sound better ??
Excellent question. Now, who has an answer?
Filtering at 28 kHz was audible for this person : http://www.hydrogenaudio.org/forums/index....ndpost&p=198806 (http://www.hydrogenaudio.org/forums/index.php?showtopic=17118&view=findpost&p=198806)
I should have been more specific. 1st, I did ammend the 16kHz part of my statement in this thread already, because an unusual sample(with a strong >16kHz constant component at high amplitude, which I suspect is basicly as audible as a sine wave, in the confined circumstances) was easily ABXed, by basicly anyone, that was provided by a previous poster. Fear of this type of sample was probably one of the reasons that the designers of RBCD decided to go ahead and leave room for error, and have a bandwidth >20kHz. As for the person who ABXed a sample with 28kHz filtering, you realize this is a questionable result, especially since this person stated they could not even hear an 18kHz sine wave. I already participated in the thread linked, and we(you and I) already discussed the requirements for a test to have high credibility/reliability. I don't see a need for us to rehash that in this current thread.
What about this one ?
Yoshikawa, Shokichiro; Noge, Satoru; Yamamoto, Takeo; Saito, Keishi
Does High Sampling Frequency Improve Perceptual Time-Axis Resolution of Digital Audio Signal?
Preprint 4562 in
http://www.aes.org/publications/preprints/search.cfm (http://www.aes.org/publications/preprints/search.cfm)
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1. I have not reviewed the paper, so I can't comment on the test specifically(do you have a copy you could e-mail me? It would be appreciated if you could. ).
2. As you already stated in the last reply, the test did not use musical program material.
-Chris
ok, so, the debate rages on between formats... ive a question then.
What would be the idea way to try to represent analogue sound in a digital medium? How can we duplicate, with optimal results, real sound in digital format?
[a href="index.php?act=findpost&pid=307126"][{POST_SNAPBACK}][/a]
I don't understand the distinction you are making between "real sound" and "digital sound". When an analog audio signal is converted into PCM (CD and DVD) it is sampled (limiting it's bandwidth) and quantized (limiting it's signal to noise ratio). If we do the sampling process at 96kHz (with correct filtering) the sampled signal can be used to perfectly reconstruct the original signal up to 48kHz. The quantization step reduces the signal to noise ratio, effectively adding wide-band quantization noise to the signal. For 24bit, the quantization noise is at -144dB (this can be improved in the audible band by noise shaping).
Converting back from PCM to analog gives us our input signal, bandlimited and with a small amount of added wide band noise. The output of the DAC is "real sound" - it's not been tainted by it's trip to the world of PCM in any way other than the bandlimiting and noise (both of which should be inaudible for 24/96). Real-world DACs and ADCs do mangle the sound, but decent ones hardly mangle it at all. What do you base your distinction between "real sound" and "digital sound" on?
SACD is the first step to do that.
You can find technical info regarding SACD's coding in the web.
[a href="index.php?act=findpost&pid=307129"][{POST_SNAPBACK}][/a]
What makes you believe that DSD is any more "analog" than L-PCM? I suggest you read up on some of the research into the properties of DSD, starting with "Why 1-Bit Sigma-Delta Conversion is Unsuitable for High-Quality Applications" by Lipshitz and Vanderkooy (AES 110th convention). Please go back and read some of the previous discussions of this topic here on HydrogenAudio. The search function will guide you on your way.
I don't understand the distinction you are making between "real sound" and "digital sound". When an analog audio signal is converted into PCM (CD and DVD) it is sampled (limiting it's bandwidth) and quantized (limiting it's signal to noise ratio). If we do the sampling process at 96kHz (with correct filtering) the sampled signal can be used to perfectly reconstruct the original signal up to 48kHz. The quantization step reduces the signal to noise ratio, effectively adding wide-band quantization noise to the signal. For 24bit, the quantization noise is at -144dB (this can be improved in the audible band by noise shaping).
Converting back from PCM to analog gives us our input signal, bandlimited and with a small amount of added wide band noise. The output of the DAC is "real sound" - it's not been tainted by it's trip to the world of PCM in any way other than the bandlimiting and noise (both of which should be inaudible for 24/96). Real-world DACs and ADCs do mangle the sound, but decent ones hardly mangle it at all. What do you base your distinction between "real sound" and "digital sound" on?
my thought of real sound is sound pre-recording; sound that has yet to be imprinted on a medium. Digital sound is sound that has been put into a digital form; 1s and 0s that when processed acordingly gives us American Pie.
... If digital sound and the sound prior to being digitized is, as you say, a "[perfect] reconstruct [of] the original signal", then why is there a debate at all, and why are different formats still being tested and released? If there was a way to reconstruct the original signal 100%, why are we even discussing anything else?
As for the person who ABXed a sample with 28kHz filtering, you realize this is a questionable result
Yes. I was rather answering 2bdecided. From my point of view, this result is not a proof in itself, but it justifies taking some time to test the same sample.
(do you have a copy you could e-mail me? It would be appreciated if you could. ).
[a href="index.php?act=findpost&pid=307168"][{POST_SNAPBACK}][/a]
No. I'd like to read it too, but I don't know if I shall spend 20 $ for it.
at this point no confirmed perceptual tests have concluded that anything exceeding a [1]16kHz bandwidth is needed to be transparent for musical program playback for humans.
[a href="index.php?act=findpost&pid=286167"][{POST_SNAPBACK}][/a]
I could try to do it if needed. I know that I'm sensitive to 16KHz lowpassing with some instruments (harpsichord, cymbals...), which appeared as dull, desaturated with ~16Khz lowpass.
I seriously don't think that 16KHz could be fully transparent, even with 'usual music'. At least not with headphones.
at this point no confirmed perceptual tests have concluded that anything exceeding a [1]16kHz bandwidth is needed to be transparent for musical program playback for humans.
[a href="index.php?act=findpost&pid=286167"][{POST_SNAPBACK}][/a]
I could try to do it if needed. I know that I'm sensitive to 16KHz lowpassing with some instruments (harpsichord, cymbals...), which appeared as dull, desaturated with ~16Khz lowpass.
I seriously don't think that 16KHz could be fully transparent, even with 'usual music'. At least not with headphones.
[a href="index.php?act=findpost&pid=307235"][{POST_SNAPBACK}][/a]
The same with me. While being a student, I tested my hearing in a lab and was able to hear up to 19kHz generated noise. Although, do not know what I can now.
Besides, ears is not the only part of our body we percept sounds with. If I can not hear frequencies above 20kHz it does not mean that I do not percept them and that they do not influence me.
... If digital sound and the sound prior to being digitized is, as you say, a "[perfect] reconstruct [of] the original signal", then why is there a debate at all, and why are different formats still being tested and released? If there was a way to reconstruct the original signal 100%, why are we even discussing anything else?
[a href="index.php?act=findpost&pid=307202"][{POST_SNAPBACK}][/a]
There are several reasons new digital formats are being developed. First, with high rate PCM (DVD-A) there are some technical advantages - 24/96 is better than 16/44.1. Whether that difference is audible is still something that is under discussion. Next, it is easier to build DACs and filters which exhibit closer to ideal behaviour in the audio band when given high-rate data. This makes equipment cheaper and better. DSD (as used in SACD for example) is also very easy to build good DACs for. I believe the industry push towards digital path amplifiers (S-Master and friends) has been a major factor in some company's backing of the format.
Lastly, an most important in my mind, is that the new formats offer DRM to music lables. Whether you think this is a good idea or not, it is in the interests of music lables to limit your freedom as much as possible when it comes to your use of their recordings. In my opinion, the next generation of digital formats has been driven by two factors - 5% technical considerations and 95% business considerations.
Lastly, an most important in my mind, is that the new formats offer DRM to music lables. Whether you think this is a good idea or not, it is in the interests of music lables to limit your freedom as much as possible when it comes to your use of their recordings. In my opinion, the next generation of digital formats has been driven by two factors - 5% technical considerations and 95% business considerations.
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oh of course. I disagree with the premise of their MO, but I understand it completely... and thats been the case since tinpan ally, only getting worse.
oh well.
I've added to my web page ( http://www.kikeg.arrakis.es/lowpass/ (http://www.kikeg.arrakis.es/lowpass/) ) that other sample where a 16 KHz lowpass is audible. It's the Androgyny sample. The difference is not as obvious as in the Chenoa sample, but for me it's still pretty easy to abx (16/16 in 45 sec.)
As for the person who ABXed a sample with 28kHz filtering, you realize this is a questionable result
Yes. I was rather answering 2bdecided. From my point of view, this result is not a proof in itself, but it justifies taking some time to test the same sample.
(do you have a copy you could e-mail me? It would be appreciated if you could. ).
[a href="index.php?act=findpost&pid=307168"][{POST_SNAPBACK}][/a]
No. I'd like to read it too, but I don't know if I shall spend 20 $ for it.
[a href="index.php?act=findpost&pid=307223"][{POST_SNAPBACK}][/a]
I have a friend who has an AES membership(he can download PDFs for free). I'll ask him if he'll let me have a copy. If I can get him to do it, I'll forward you a copy if you give me an e-mail address.
-Chris
I have a friend who has an AES membership(he can download PDFs for free). I'll ask him if he'll let me have a copy. If I can get him to do it, I'll forward you a copy if you give me an e-mail address.
-Chris
[a href="index.php?act=findpost&pid=307457"][{POST_SNAPBACK}][/a]
It says on the site it'll still cost him $5.
Single Convention Preprints and Conference Papers are $5.00 for AES Members and $20.00 for non-Members.
I have a copy of the document but it says that you need "direct permission from the Journal of the Audio Engineering Society" to distribute it.
Edit: I can provide a description of the test without infringing copyright.
The sample material used for the test was recorded on a 96 kHz 16 bit DAT tape. The reference sample (sample R) had a FIR lowpass filter at 40 kHz while the test sample (sample X) had a lowpass at 20 kHz. 11 adult males, 22-24 years of age, were asked to identify which sample was X and which was R. The result was statistically significant (pval < 0.05), and the folllowing was concluded.
From the experiment using a pulse train signal, it was suggested that widening the frequency range of the audio system improves perceptual time-axis resolution. Sound recording and reproduction system using a high sampling (i.e. 96 kHz sampling) will be helpful from the view point of improving time-axis resolution of the digital audio signals.
Digital clipping can be shown on a good digital peak-level meter with calibrated overload indication. The number of consecutive clipped samples (full scale) is an indication for the amount of audible clipping. No clipped samples means the audio can be reconstructed with a good DAC.
One or more FS samples indicate clipping, but it doesn't have to be audible.
I'd gladly check your files for digital clipping and even try to remove the clipping if you like. The music will be softer then, but that's exactly the mastering engineer's dilemma.
[{POST_SNAPBACK}][/a] (http://index.php?act=findpost&pid=307101")
I haven't read far enough ahead in the thread to see if this was addressed. The above statement is not correct. For more read the paper on the consequences of traditional peak meters here:
[a href="http://www.cadenzarecording.com/papers]http://www.cadenzarecording.com/papers[/url]
It is very possible (in fact likely) that signals wherein the samples approach FS will clip the output of even a good D/A converter. The digital reconstruction filters on those D/As simply can't reconstruct above FS even if the analog can.
Cheers!
Nika
... If digital sound and the sound prior to being digitized is, as you say, a "[perfect] reconstruct [of] the original signal", then why is there a debate at all, and why are different formats still being tested and released? If there was a way to reconstruct the original signal 100%, why are we even discussing anything else?
[a href="index.php?act=findpost&pid=307202"][{POST_SNAPBACK}][/a]
Erich,
Your question implies that the changes to formats have been made because they inherently provide an improvement. This is an optimistic view of the marketing and multinational corporations today. Sony has proved inside their building that 44.1kS/s is better than DSD and can sound as good for any recording/mixing/delivery situation as higher L-PCM sample rates such as 96kS/s and 192kS/s. There is a reason that you don't hear about this.
96kS/s was originally proposed back in the early 90s or late 80s as a solution to a problem the industry was having with filter design. That problem was fixed long ago, but the push for higher fs had started and it was viewed as potentially profitable as well, so despite the lack of need for it, the industry pushed on.
SACD (DSD) was similarly created as a patch for a problem that the industry had. But while the industry fixed that problem (that being integral and differential non-linearity in converters yielding high distortion) the DSD marketing machine had already gotten started. And therefore, despite the fact that modern converters designed for L-PCM use inherently have a leg up on DSD, Sony can't back down now and DSD continues as a major marketing push.
In each of these situations the industry had a choice: fix what was wrong or develop a new format altogether. Since there is a lot of profitable money tied with new formats we can't be surprised which way the labels and consumer gear manufacturers went, while the R&D guys at the converter manufacturers were busy fixing the problems...
Nika
Edit: I can provide a description of the test without infringing copyright.
...
[a href="index.php?act=findpost&pid=307458"][{POST_SNAPBACK}][/a]
But, did the test designers take any measures to avoid speaker distortion products falling into audible band, when feed with signals very rich in ultrasonic content, such as the pulse train they make mention of?
The same with me. While being a student, I tested my hearing in a lab and was able to hear up to 19kHz generated noise. Although, do not know what I can now.
Besides, ears is not the only part of our body we percept sounds with. If I can not hear frequencies above 20kHz it does not mean that I do not percept them and that they do not influence me.
[a href="index.php?act=findpost&pid=307240"][{POST_SNAPBACK}][/a]
1. Was this test using linear phase filtering? That will throw off everything immediately, as non-linear phase filtering will create audible phase problems in the audible range, despite the presence of HF content.
2. Were the speakers linear above 20kHz? Non-linear speakers above 20kHz can create audible bandwidth distortion when presented with HF content. Again, however, this isn't an indication of audibility above 20kHz but rather audibility below.
3. Despite the persistent notion that we can hear with organs other than our ears, absolutely nothing has been presented on the market that substantiates that we can do this for HF content at musical program listening levels. We can "feel" HF content at several hundred dB above normal listening levels (you would certainly "feel" it if someone started sonar-welding your skin, for example) but even studies that have been done on this subject have indicated no "perceptibility" of HF content. Nobody asked the "listeners" in such tests to only indicate the HF content that they could "hear" and to ignore the content that they "felt in other ways." U
Nika
From the experiment using a pulse train signal, it was suggested that widening the frequency range of the audio system improves perceptual time-axis resolution. Sound recording and reproduction system using a high sampling (i.e. 96 kHz sampling) will be helpful from the view point of improving time-axis resolution of the digital audio signals.
[a href="index.php?act=findpost&pid=307458"][{POST_SNAPBACK}][/a]
This type of stuff is really annoying. How in the world did they draw that conclusion from that test?
Nika
Single Convention Preprints and Conference Papers are $5.00 for AES Members and $20.00 for non-Members.
That is for the old standard membership model. There is an online subscription model that allows for unlimited downloads of papers, which was recently introduced within this year.
I have a copy of the document but it says that you need "direct permission from the Journal of the Audio Engineering Society" to distribute it.
Then, consider me an infringer(of the trade of AES research papers).
As for the general description; that's not exactly what I need. I need to review a paper in full before I come to any conclusion(s). This is not a published paper, for one thing, so the quality of paper could be anything.
-Chris
There is a good explanation of the behaviour of DACs when in the presence of levels nearing 0dBfs and the relation to peak output distortion(s), in this paper:
OdBFS+ Levels in Digital Mastering
Sgren H. Nielsen and Thomas Lund
AES Preprint 5251
This paper(unlike most) is available for free download from the authors:
http://www.tcelectronic.com/media/nielsen_...00_0dbfs_le.pdf (http://www.tcelectronic.com/media/nielsen_lund_2000_0dbfs_le.pdf)
-Chris
But, did the test designers take any measures to avoid speaker distortion products falling into audible band, when feed with signals very rich in ultrasonic content, such as the pulse train they make mention of?
[a href="index.php?act=findpost&pid=307499"][{POST_SNAPBACK}][/a]
Good question. While this might have little effect for musical content, the difference can be very big with such signals. I remember, back in school, having played with a generator with a very cheap speaker plugged in (the kind of speaker for mono TV set or FM-Alarm-clock, if you see what I mean), and the 10 kHz sine, square and triangle had a completely different sound because of distortion.
But, did the test designers take any measures to avoid speaker distortion products falling into audible band, when feed with signals very rich in ultrasonic content, such as the pulse train they make mention of?
[a href="index.php?act=findpost&pid=307499"][{POST_SNAPBACK}][/a]
If they did it wasn't mentioned in the preprint.
Listen to the remastered Dark Side of The Moon on SACD/CD Hybrid, it was engineered by the ORIGINAL engineer [a href="index.php?act=findpost&pid=304696"][{POST_SNAPBACK}][/a]
Hi,
Just for correction, the engineer for the new Dark Side Of The Moon SACD was James Guthrie, and the ORIGINAL engineer was Alan Parsons, so that was NOT engineered by the original engineer.
...No clipped samples means the audio can be reconstructed with a good DAC...
[{POST_SNAPBACK}][/a] (http://index.php?act=findpost&pid=307101")
I haven't read far enough ahead in the thread to see if this was addressed. The above statement is not correct. For more read the paper on the consequences of traditional peak meters here:
[a href="http://www.cadenzarecording.com/papers]http://www.cadenzarecording.com/papers[/url]
It is very possible (in fact likely) that signals wherein the samples approach FS will clip the output of even a good D/A converter. The digital reconstruction filters on those D/As simply can't reconstruct above FS even if the analog can.
Cheers!
Nika
[a href="index.php?act=findpost&pid=307495"][{POST_SNAPBACK}][/a]
Hi Nika,
since reconstruction filter "headroom" isn't part of any DAC specification I know of, it's hard to know the performance in that respect. After I read your post I called Prism (my DAC brand) to ask about their specs and although they couldn't give numbers ("would have to look that up") they assured me that their reconstruction filter has been designed with "quite some" margin for overloads.
Do you know of any test or (AES?) specification that can help to quantify DAC overload performance ? I'd love to know the limits of my DAC.
Hi Nika,
since reconstruction filter "headroom" isn't part of any DAC specification I know of, it's hard to know the performance in that respect. After I read your post I called Prism (my DAC brand) to ask about their specs and although they couldn't give numbers ("would have to look that up") they assured me that their reconstruction filter has been designed with "quite some" margin for overloads.
Do you know of any test or (AES?) specification that can help to quantify DAC overload performance ? I'd love to know the limits of my DAC.
[a href="index.php?act=findpost&pid=308393"][{POST_SNAPBACK}][/a]
Kees,
I don't know of any specification that is recognized regarding reconstruction filter headroom. As for a test, you would have to be pretty smart about it, but you could certainly create an artificial signal for testing purposes. The easiest one would be a 1/4 sampling rate sine wave wherein you create it such that the samples end up at .707, .707, -.707, -.707 FS repeatedly throughout (sampling the sine wave halfway between zero and peak throughout). If you normalize the samples then the signal will exceed full scale by about 3dB, and if the signal is clipping the filters then this would be easy to identify on test equipment on the back end.
So: create a digital signal of FS, FS, -FS, -FS (repeat ad nauseum) and this represents a 11.025kHz sine wave with the samples normalized at full scale. How does this look coming out of the DAC? Is it an 11.025kHz sine wave or is it a mess? Now start turning that down. What you see on your scope will probably change in other ways than just amplitude. By the time you get down to 3dB of attenuation you should see a pure sine wave.
My guess is that the folks at Prism will tell you that they have "tons of headroom" on their filters, but they are only speaking about the analog filters. The digital filters that are built on to the converter chips they have almost no control over. I doubt they have any more headroom there than the chips have inherently - which you'd have to speak to the chip designer about (good luck).
I hope this helps?
Nika
So: create a digital signal of FS, FS, -FS, -FS (repeat ad nauseum) and this represents a 11.025kHz sine wave with the samples normalized at full scale. How does this look coming out of the DAC? Is it an 11.025kHz sine wave or is it a mess?
[a href="index.php?act=findpost&pid=308407"][{POST_SNAPBACK}][/a]
Aha, I think you've helped pinning down the reason for my confusion.
During my 20 years with digital audio, I've never allowed the digital signal of my (cd-)masters to reach FS completely. Overload indicators are set to 1 sample at 16 bit.
I'm still trying to find out how DAC overload can occur with signals that have never (nowhere in the signal-path, not even inside the dsp) reached FS.
Lowering overall level by say 0.01dB to avoid overload indication is a nono IMO since it doesn't remove clipping.
From what I understood, a DAC can still overload "between samples" with some signals (probably with lots of high freq. content), even when the signal never reaches FS.
In your FS, FS, -FS, -FS example, you're proposing to create a distorted (by definition, at least in my vocabulary) signal to start with. I have no doubt that this might result in audible distortion.
ps: if this is getting off topic too much I wouldn't mind a new thread.
Aha, I think you've helped pinning down the reason for my confusion.
During my 20 years with digital audio, I've never allowed the digital signal of my (cd-)masters to reach FS completely. Overload indicators are set to 1 sample at 16 bit.
Doesn't matter. Set it as I said except .01dB lower. You now have FS(-1 LSB) FS(-1LSB), -FS(+1 LSB), -FS(+1LSB) etc. yet your waveform is still exceeding full scale by 3dB, and this is a "legal, legitimate" waveform - a sine wave at 11.025kHz.
I'm still trying to find out how DAC overload can occur with signals that have never (nowhere in the signal-path, not even inside the dsp) reached FS.
Go to http://www.cadenzarecording.com/papers (http://www.cadenzarecording.com/papers) and read the paper on "traditional peak meters." There are graphics and whatnot that explain.
Cheers!
Nika
Doesn't matter. Set it as I said except .01dB lower. You now have FS(-1 LSB) FS(-1LSB), -FS(+1 LSB), -FS(+1LSB) etc. yet your waveform is still exceeding full scale by 3dB, and this is a "legal, legitimate" waveform - a sine wave at 11.025kHz.
[a href="index.php?act=findpost&pid=308547"][{POST_SNAPBACK}][/a]
I've tried the test and didn't hear strange things. But then, to my not so young ears there is no difference between a 11.025 kHz sine wave and a ditto square wave. So if there's only harmonic distortion, I won't hear it. When I have some time I should take a look at the spectrum of the analog output of the DAC. Perhaps there's something visable but inaudible.
Listen to the remastered Dark Side of The Moon on SACD/CD Hybrid, it was engineered by the ORIGINAL engineer and both recordings thus should be from the same source material. This to me provides an obvious win for SACD, as its' clarity and accuracy in representing the original audio is far greater than that of CD.
Careful. Just because it was mastered by the same engineer does not in any way mean it was mastered the same. It is virtually impossible to do so. First, the mastering tools available for the DSD environment are different from those available for PCM. Second, traditional PCM mastering involves normalizing and limiting in ways that are actually illegal in the SACD scarlet book - the disk would be rejected. The result is that mastering engineers on SACD are actually forced to use less compression and allow more dynamics. It is not to say that the PCM mastering engineer couldn't do the same, but they don't HAVE to, so they don't.
The result is that SACD disks often use more dynamic range, have less compression, and sound that way. But they don't have to...
Nika
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I understand this, that is why I referenced the Dark Side of The Moon Hybrid. Both layers are 30th anniversary remasters done by the original engineer.
Multi-Channel aside, compare the two 2 channel recordings.
The reason I referenced the CD was to address earlier posts by people who try to use the "scientific limitations of the human ear" to excuse the fact that they are unable to distinguish between CD and SACD.
[a href="index.php?act=findpost&pid=304829"][{POST_SNAPBACK}][/a]
Your example is *useless* for addressing that issue. The two layers opf Dark Side have *clearly* been remastered *very* differently. This is obvious from visible inspection of the waveforms, as well as objective measurement of same. I've done this at home, and so has John Atkinson of Stereophile. This means that using two different A/D transfer chains for the analog output -- Atkinson's being *far* more high-end than mine -- we were both able to see that the two layers are mastered differently. My own experience in doing the same comparison on other hybrid SACDs, btw, is that some CD layers *do* look extremely like the SACD layer (e.g., the Rollling Stones hybrids). DSotM's isn't one of them.
[a href="http://www.stereophile.com/news/11649/]http://www.stereophile.com/news/11649/[/url]
With the mastering of the two layers being this different, you can't separate that effect, from any supposed difference between the *formats*.
Scientific evidence for *audible* difference of SACD/DVD-A/Redbook CD distribution formats *qua formats* simply has not been forthcoming, either from the industry or anywhere else.
The CD version is purposely designed to be bad/different in this case, as compared to the SACD version. The dynamics were squashed so that they could increase loudness as far as possible. But, many CDs seem to be designed to be purposely bad these days. Not a problem with the format. A problem with the morons mastering to the format.
-Chris
[a href="index.php?act=findpost&pid=306708"][{POST_SNAPBACK}][/a]
From what I read on prosound forums, it's more a problem of moron producers/musicians demanding
entry into the loudness wars...and mastering engineers (who are hired hands, in the end) doing the job they're paid to do.
That Krall CD waveform doesn't look that bad, really , compared to lots of the stuff out there that's close to solid-brick green.
If they think that louder is better, why would they make CD louder if they want SACD to sound better ??
Great question. I'd say it's because it's not easy to queue up SACD and CD tracks togther quickly. But it's quite easy to queue up multiple CD tracks (changers have been around for decades, and now of course we have lossy format players in abundance). Also, SACD's afaik don't tend to get much radio play. So it's the 'competition' between CDs, not between CD and SACD tracks, that has producers and musicians worrying that their mixes aren't 'punchy;' enough.
There's another reason for the SACD being quiter...
The DSD signal on SACD is always at digital full scale - that's the nature of DSD.
However, you can't put information in the audio band at a level comparable to digital full scale because it would make the whole DSD conversion process go unstable. You might just get a ~1MHz square wave with absolutely no audio information modulated onto it at all, or something equally unpleasant and unrelated to the original audio. So you keep well away from digital full scale.
How "loud" SACDs appear at the analogue outputs relative to a standard CD depends on how the DAC is calibrated, but for DSD encoding, Sony suggest using 75% modulation as the limit, and not one that you should be aiming to hit.
For these reasons, brick wall limiting is pointless and useless for SACD, unless you want that "sound". Plus (I think) Sony have some control over what is release on SACD, and in an attempt to make the format "sound good" in the ears of consumers, they try to ensure only high quality recordings are released on SACD.
Cheers,
David.
Actually, before each release, all the DSD track are verified in Japan. The dynamic of the high frequencies being limited, they want to be sure that it won't be distorted.
.
Hands down, flat-out, the defacto study of DVD-A vs. SACD...
http://www.hfm-detmold.de/eti/projekte/dip..._paper_6086.pdf (http://www.hfm-detmold.de/eti/projekte/diplomarbeiten/dsdvspcm/aes_paper_6086.pdf)
Andrew D.
cdnav.com (http://www.cdnav.com)
.
This study does not seem to reach any conclusion that either is "better", only that under rare conditions they can be perceptively different. It would have been a lot more interesting to me if it had compared either one to 44.1/16.