Skip to main content

Notice

Please note that most of the software linked on this forum is likely to be safe to use. If you are unsure, feel free to ask in the relevant topics, or send a private message to an administrator or moderator. To help curb the problems of false positives, or in the event that you do find actual malware, you can contribute through the article linked here.
Topic: Why 24bit/48kHz/96kHz/ (Read 392085 times) previous topic - next topic
0 Members and 1 Guest are viewing this topic.

Why 24bit/48kHz/96kHz/

Reply #250
classic stuff from the Linn Unidisc universal player measurements in Stereophile

Quote
The occasional low-level clicks [produced on CD playback only] may well be specific to our review sample, but the Unidisk's somewhat disappointing measured performance on CD playback compared with SACD and DVD caused my eyebrows to rise a little


If the manufacturers can't be trusted to give all formats a fair shake in their 'uni-players' , it's not reasonable to assume audible differences are due to *the formats*

And this was a pricey ($11,000) 'high-end' player!


I remember Stereophile proving the Redbook layer of the 30th Anniversary Edition of Pink Floyd Dark Side Of The Moon was deliberately flawed compared to the SACD layer.  The Redbook layer was very often clipped (the wave would hit 0dB and just be flat for a while... hugh !!).  The SACD layer wasn't touching 0dB once.

The weirdest part was the fact that STEREOPHILE made that public.

Anyway.. the moral of the story is that Sony has a monetary interest in making sure Redbook CDs don't sound as good as SACD.

I am just glad Vinyl sold more than SACD and DVD-A combined.

p.s.: sorry, I can't back-up my claims with references.  Too long and my coffee is getting cold.

Why 24bit/48kHz/96kHz/

Reply #251
[...]
I remember Stereophile proving the Redbook layer of the 30th Anniversary Edition of Pink Floyd Dark Side Of The Moon was deliberately flawed compared to the SACD layer.  The Redbook layer was very often clipped (the wave would hit 0dB and just be flat for a while... hugh !!).  The SACD layer wasn't touching 0dB once.
[...]

Old story and it is very known that mastering is the issue here. Not that it is clipped, although it is a sign on the wall. I believe that the 30th anniversary DSOTM PCM layer was not a downsample from the DSD master. It was the same as a previously released version, probably from a cost cutting point of view.

Regards,
Jacco
Logical reasoning brings you from a to b, imagination brings you everywhere.

Why 24bit/48kHz/96kHz/

Reply #252

[...]
I remember Stereophile proving the Redbook layer of the 30th Anniversary Edition of Pink Floyd Dark Side Of The Moon was deliberately flawed compared to the SACD layer.  The Redbook layer was very often clipped (the wave would hit 0dB and just be flat for a while... hugh !!).  The SACD layer wasn't touching 0dB once.
[...]

Old story and it is very known that mastering is the issue here. Not that it is clipped, although it is a sign on the wall. I believe that the 30th anniversary DSOTM PCM layer was not a downsample from the DSD master. It was the same as a previously released version, probably from a cost cutting point of view.

Regards,
Jacco


Funny thing is, I have both the 1992 Shine On Box Set version and the MFSL Ultradisc II version and neither are clipped.

... they went out of their way to make it clipped.  They had an economic reason to do so.  SACD cannot sound better (technically speaking, it does not make sense).  So what are you gonna do if you are an exec at Sony and you want to push the SACD format?  Hmmm... why not re-release CDs but badly mastered!!

The only thing that holds them back, IMHO, is that there are already correct versions out there.  It would be too obvious if they re-released everything clipped or otherwise incorrect.  People could easily see what is going on.

Nahhh... IMHO, again, the thing SACD has is multichannel.  Now, remix stuff properly by really taking advantage of that technology (not like what they did with Kind of Blue for example) AND remove the pesky copy-protection and I MIGHT look at it.

But hey... if you really want to throw your money at SACD, be my guest.  I don't care.

Why 24bit/48kHz/96kHz/

Reply #253

[...]
I remember Stereophile proving the Redbook layer of the 30th Anniversary Edition of Pink Floyd Dark Side Of The Moon was deliberately flawed compared to the SACD layer.  The Redbook layer was very often clipped (the wave would hit 0dB and just be flat for a while... hugh !!).  The SACD layer wasn't touching 0dB once.
[...]

Old story and it is very known that mastering is the issue here. Not that it is clipped, although it is a sign on the wall. I believe that the 30th anniversary DSOTM PCM layer was not a downsample from the DSD master. It was the same as a previously released version, probably from a cost cutting point of view.

Regards,
Jacco



I know that the mastering of the DSD and CD layers are significantly different on this disc -- I've verified that with my own rips and AD captures --  but I have never heard that the CD layers was the same as a previous version -- do you remember which one?

What's funny to me is that the CD layer is louder than the DSD layer , due to the compression...and thus in light of psychoacoustic 'loudness bias', it could well be judged  *better sounding* than the DSD layer to lots of people in lots of situations.    (Though of course, difference in EQ might mitigate the 'loudness' effect...)

Why 24bit/48kHz/96kHz/

Reply #254


[...]
I remember Stereophile proving the Redbook layer of the 30th Anniversary Edition of Pink Floyd Dark Side Of The Moon was deliberately flawed compared to the SACD layer.  The Redbook layer was very often clipped (the wave would hit 0dB and just be flat for a while... hugh !!).  The SACD layer wasn't touching 0dB once.
[...]

Old story and it is very known that mastering is the issue here. Not that it is clipped, although it is a sign on the wall. I believe that the 30th anniversary DSOTM PCM layer was not a downsample from the DSD master. It was the same as a previously released version, probably from a cost cutting point of view.

Regards,
Jacco


Funny thing is, I have both the 1992 Shine On Box Set version and the MFSL Ultradisc II version and neither are clipped.

... they went out of their way to make it clipped.  They had an economic reason to do so.  SACD cannot sound better (technically speaking, it does not make sense).  So what are you gonna do if you are an exec at Sony and you want to push the SACD format?  Hmmm... why not re-release CDs but badly mastered!!



That is onle possibility -- teh otehr is that they were simply following common 'modern' CD remastering practice, which seems to assume that people will do most of their CD listening in cars or other noisy environments, where louder and less dynamic  *is* often subjectively better.

What *is* underhanded, though, is implying that such discs are demonstrations of how SACD (the format) sounds better than CD.  I've seen some ignorant reviewers do just that...and I've never seen Sony put out any disclaimers regarding its own hype of SACDs.

Quote
The only thing that holds them back, IMHO, is that there are already correct versions out there.  It would be too obvious if they re-released everything clipped or otherwise incorrect.  People could easily see what is going on.

Nahhh... IMHO, again, the thing SACD has is multichannel.  Now, remix stuff properly by really taking advantage of that technology (not like what they did with Kind of Blue for example) AND remove the pesky copy-protection and I MIGHT look at it.



I like the Kind of Blue multichannel mix -- it's definittely a laid-back use of multichannel (essentially the 'surround' is for ambience), but I don't really think this music needs something more aggressive.  'In A Silent Way' has more actual 'surround' content, and that fits it.


Quote
But hey... if you really want to throw your money at SACD, be my guest.  I don't care.


SACDs *are* often mastered more tastefully than their CD remaster counterparts (whether hybrid or independent) -- perhaps because the Scarlet Book spec forbids 'clipping' in the DSD domain (though IAUI you can take it through a PCM step, clip the hell out of it, lower the level, then transcode to DSD --  if you wanted to)

Why 24bit/48kHz/96kHz/

Reply #255
SACDs *are* often mastered more tastefully than their CD remaster counterparts (whether hybrid or independent) -- perhaps because the Scarlet Book spec forbids 'clipping' in the DSD domain (though IAUI you can take it through a PCM step, clip the hell out of it, lower the level, then transcode to DSD --  if you wanted to)


I know the Analog Productions Bill Evans Hybrid SACD have been equally well mastered on their SACD layer and on their Red Book layer.  They CAN do a great job on Red Book. Oh well...

Redbooks are supposedly (I don't have the expertise to make a blanket statement) so badly mastered these days that people are often looking foir the non-re-mastered version of their favourite records (look at SteveHoffman.tv Music forums for an idea of what I am talking about).

IMHO, the current model of the music industry should be completely rehauled.  Image if remastering engineers could select whatever tapes they wished to work on and people could choose the version they wanted.  That would generate competition between mastering engineers and they would have to make a great job.

Why 24bit/48kHz/96kHz/

Reply #256
Well, i haven't really read all of the topic's posts thoroughly. I did scan them reasonably, though, so i'm going to post a question regarding samplerates :

I read somewhere (don't remember exactly where right now, and it's 3 a.m., so...  ) that psychoacoustically, our hearing response to spatial perception was based on differences regarding the sound that reaches each ear, and there were a mainly 3 key factors - amplitude (volume) , spectral differences (high frequency rolloff) and phase differences. The cool remark was that we (human beings ?!  ) can't distiguish the directionality of sounds intervalled below about 5ms.

Given this, at 192khz samplerates, we have 5,2083(3) microseconds between samples, which is a very close call to the beferementioned value. Now i ask you - wouldn't this, per se, give us an extremely more detailed stereo/multichannel image? Does anybody work on a studio/facility that has equipment capable of performing tests on this?

Anyway, don't bother to kill me if what i just said is complete nonsense. I don't have any TonMeister or other acoustics degrees, and i'm way beyond my usual lack of sleep threshold, so... 

Why 24bit/48kHz/96kHz/

Reply #257
Given this, at 192khz samplerates, we have 5,2083(3) microseconds between samples, which is a very close call to the beferementioned value. Now i ask you - wouldn't this, per se, give us an extremely more detailed stereo/multichannel image?


No.  The sample period has absolutely nothing to do with that.  It only determines the maxium possible frequency in the signal.  For frequencies below half the sample rate, the signal produced by your DAC is continuous (and thus the distance between adjacent samples is 0).  Or at least it is for an ideal DAC (what you actually hear will depend on your equipment)!

Does anybody work on a studio/facility that has equipment capable of performing tests on this?


The effect of sample rate was mathamatically proved in the 1930s.  No need to test, but you're welcome to if you feel like proving that mathamatics works

Anyway, don't bother to kill me if what i just said is complete nonsense. I don't have any TonMeister or other acoustics degrees, and i'm way beyond my usual lack of sleep threshold, so... 


EE degree would be best for this sort of thing

Why 24bit/48kHz/96kHz/

Reply #258
Given this, at 192khz samplerates, we have 5,2083(3) microseconds between samples, which is a very close call to the beferementioned value. Now i ask you - wouldn't this, per se, give us an extremely more detailed stereo/multichannel image?


You can easily resolve your 5 or 10 microsecond interaural ITD's with a 44.1 kHz sampled signal.

Try it, using a Gaussian pulse, yourself.
-----
J. D. (jj) Johnston

Why 24bit/48kHz/96kHz/

Reply #259
The cool remark was that we (human beings ?!  ) can't distiguish the directionality of sounds intervalled below about 5ms.

Given this, at 192khz samplerates, we have 5,2083(3) microseconds between samples, which is a very close call to the beferementioned value.

Did you notice that "ms" means millisecond and that one millisecond equals 1000 microseconds? 

Why 24bit/48kHz/96kHz/

Reply #260
ms can refer to microseconds in a few circumstances.(But IIRC, they mostly involve SPICE plots.)

Why 24bit/48kHz/96kHz/

Reply #261

The cool remark was that we (human beings ?!  ) can't distiguish the directionality of sounds intervalled below about 5ms.

Given this, at 192khz samplerates, we have 5,2083(3) microseconds between samples, which is a very close call to the beferementioned value.

Did you notice that "ms" means millisecond and that one millisecond equals 1000 microseconds? 



ITD's have been detected (by humans) in the 5 to 10 MICROsecond range.

In fact, they have been so detected in digital audio signals at 44.1/16, which ought to be a great big hint for those who think they can't be.
-----
J. D. (jj) Johnston

Why 24bit/48kHz/96kHz/

Reply #262
A big thumbs up for the replies. Thanks for helping a rookie on this question. Is there any practical advantage on working on such high samplerates (96k+)?

Why 24bit/48kHz/96kHz/

Reply #263
There are very few cut and dry reasons why you'd really want to. If you're doing an analog capture of a high res digital format (SACD/DVD-A) it would make sense. You'd also need 96khz (and maybe a little more?) to record quadraphonic records. And of course if you're recording something that you know has huge amounts of ultrasound (car keys, castanets, etc), that needs to be recorded for some reason.

Why 24bit/48kHz/96kHz/

Reply #264
A big thumbs up for the replies. Thanks for helping a rookie on this question. Is there any practical advantage on working on such high samplerates (96k+)?

For recording/production/mixing, it might make sense. Converters and plugins (digital effects) have a better result with 96 or 88.2 khz (you better get 88.2 if you want a final release digitally resample at 44.1).
However, you have to work on a music that require very open high end which is not the case with typical pop/rock (Beatles for example rarely went higher than 12 khz), because it takes more CPU and HD space

Why 24bit/48kHz/96kHz/

Reply #265
Sorry for buming old thread, but this has been lying in my head for months: someone said, that you can perfectly "capture" all frequencies up to samplerate/2. So, how do you then capture a 22049Hz sine wave with a sampling frequency of 44100Hz without a 1hz "breathing volume" ?? Sorry if this question is stupid, but I have no answer!

Why 24bit/48kHz/96kHz/

Reply #266
A good enough reconstruction filter will reproduce a 22049 hz sine wave, um, good enough.

Why 24bit/48kHz/96kHz/

Reply #267
Sorry for buming old thread, but this has been lying in my head for months: someone said, that you can perfectly "capture" all frequencies up to samplerate/2. So, how do you then capture a 22049Hz sine wave with a sampling frequency of 44100Hz without a 1hz "breathing volume" ?? Sorry if this question is stupid, but I have no answer!


Maybe there is the answer

http://www.hydrogenaudio.org/forums/index....amp;#entry44506

http://www.hydrogenaudio.org/forums/index....d&pid=19328

Why 24bit/48kHz/96kHz/

Reply #268
Sorry for buming old thread, but this has been lying in my head for months: someone said, that you can perfectly "capture" all frequencies up to samplerate/2. So, how do you then capture a 22049Hz sine wave with a sampling frequency of 44100Hz without a 1hz "breathing volume" ?? Sorry if this question is stupid, but I have no answer!



As previously mentioned, the DAC will have to do a bit of waveform reconstruction in some shape or form, because you just can't sample above the Nyquist, and I guess that's (well, it IS) where the quality of the circuitry and chips used comes into play.


Here's a thought... You have your 24/96, your 192, your 44.1 and 88.2... DSD only has a bitdepth of ONE bit, but its samplerate is 2.8224MHz. How cool is that  (though granted it is sigma delta modulation, not plain pulse code, when we're at these kind of levels of samplerates it really is just 'you say potatoe, I say potatoe') And I know this is common knowledge, it's just a part of my brain still marvels at the fact that you can have a single bit of bitdepth and still have amazing quality recordings as long as your samplerate is sufficient. It's madness, madness I say!
Don't forget International Talk Like A Pirate Day! September the 19th!

Why 24bit/48kHz/96kHz/

Reply #269
Sorry for buming old thread, but this has been lying in my head for months: someone said, that you can perfectly "capture" all frequencies up to samplerate/2. So, how do you then capture a 22049Hz sine wave with a sampling frequency of 44100Hz without a 1hz "breathing volume" ?? Sorry if this question is stupid, but I have no answer!


By using a reconstruction filter whose impulse response is based on a sin wave.  Specifically a sinc, a type of decaying, sin-like function.  If you really do add infinitely many of them together (while multiplying them times the sample value as you go) you will find that everything cancels, except a single sin wave at 22049Hz.

One of the reasons this works is that the sampled version without the DAC contains lots of jumps, where the amplitude changes very quickly between samples.  These jumps are high frequencies.  The sinc function is actually a filter with a flat passband, and perfect frequency rejection in the stop band.  So when you add and multiply those infinate number of sincs together, you're actually using a perfect filter to remove all the noise added by sampling in the first place.  With all the noise removed, the signal is perfectly reconstructed (SNR==infinity), and if you look at the waveform, you'll see that all the space between samples has been filled in by the filter.

Sorry if thats hard to follow.  Its the best I can do without lots of math.

Why 24bit/48kHz/96kHz/

Reply #270
As previously mentioned, the DAC will have to do a bit of waveform reconstruction in some shape or form, because you just can't sample above the Nyquist, and I guess that's (well, it IS) where the quality of the circuitry and chips used comes into play.
The DAC's reconstruction filter has to make sure that no frequencies above 0.5 Fs come through. In practice this means that the filter has to start attenuating at a lower frequency. Some real world DAC chip numbers are (CS4340A): -0.05dB at 0.4535Fs and -3dB at 0.4998Fs. At 44.1 kHz sampling rate 0.4535 Fs corresponds to 20 kHz. Your 22.049 kHz will be reproduced, but at a lower level. (microphones, speakers and headphones are likely to have much more roll-off, be it less steep) Increasing the sampling rate will move this "problem" to a higher frequency.

Why 24bit/48kHz/96kHz/

Reply #271
Sorry for buming old thread, but this has been lying in my head for months: someone said, that you can perfectly "capture" all frequencies up to samplerate/2. So, how do you then capture a 22049Hz sine wave with a sampling frequency of 44100Hz without a 1hz "breathing volume" ?? Sorry if this question is stupid, but I have no answer!


I had asked just that question, here are the answers: http://www.hydrogenaudio.org/forums/index....showtopic=47764
Veni Vidi Vorbis.

Why 24bit/48kHz/96kHz/

Reply #272
Sorry for buming old thread, but this has been lying in my head for months: someone said, that you can perfectly "capture" all frequencies up to samplerate/2. So, how do you then capture a 22049Hz sine wave with a sampling frequency of 44100Hz without a 1hz "breathing volume" ?? Sorry if this question is stupid, but I have no answer!


being a mathematician and professionally working with DSP audio, i can fully confirm your suspicions that perfect A2D-D2A reconstruction does not exist in this world.

First, Nyquist theorem refers to signals that do not exist in this world because any signal with limited spectrum must be unlimited in duration. Second, filters that have transient (0 dB to -98 dB) bandwidth of 1/22000 exist only on paper. Third .... well, take it easy. 

Back to the initial question: you need "breathing space", both in terms of hertz, bits depth and channels. the more - the easier it will be to record and playback. being very tight on headroom requires perfection and leaves no space for mistakes whatsoever. Honestly, I think that recording classical music with 25...35 dB variations in dynamics, huge orchestra, soloists and choirs (as Mahler's 8th, etc) in 2 channel 16/44.1 without serious alterations to material is not possible. that may be reason why classical/jazz jumped to SACD - at least on recording stage.

On another hand... IMHO... currently, proportion of recordings that would benefit from SACD/16b+/48k+ is tiny.

Why 24bit/48kHz/96kHz/

Reply #273
Welcome to HA!

being a mathematician and professionally working with DSP audio, i can fully confirm your suspicions that perfect A2D-D2A reconstruction does not exist in this world.

Does "perfect" anything exist in this world? (Other than mathematics, maybe?)


Quote
First, Nyquist theorem refers to signals that do not exist in this world because any signal with limited spectrum must be unlimited in duration.

True. Let us say it's "good enough"


Quote
Second, filters that have transient (0 dB to -98 dB) bandwidth of 1/22000 exist only on paper.

Well, if they can exist on paper, they can exist in the real world too. Do you have a special PC or DSP that can't do maths?

In any case, we don't need to transition in 1/22050 of the sample rate. Let us keep everything up to 20kHz intact, and make sure we have nothing above 22050kHz. Now the transition is 1/10th of the sample rate - very easy, with a perfect result up to 20kHz, some attenuation above this, and with no aliasing or distortion.

Do your ears work above 20kHz for normal amplitudes?


Quote
Third .... well, take it easy.

No, bring it on - that's the point of this thread - there are many proposed reasons why 44.1kHz isn't enough - most fall down immediately with rigorous scrutiny. Even the best ones are completely unproven.


Quote
Back to the initial question: you need "breathing space", both in terms of hertz, bits depth and channels.

You may need more of everything for ease of acquisition, clean processing etc, but that has never been doubted. I know the benefit of a greater bit depth for processing. I know the benefit of a higher sample rate (at least internally/synthetically) for some kinds of processing. The question is about the delivery method.


Quote
the more - the easier it will be to record and playback. being very tight on headroom requires perfection and leaves no space for mistakes whatsoever. Honestly, I think that recording classical music with 25...35 dB variations in dynamics, huge orchestra, soloists and choirs (as Mahler's 8th, etc) in 2 channel 16/44.1 without serious alterations to material is not possible.

Mahler requires more than 20kHz bandwidth, and more than 96dB dynamic range? I'd better buy some new ears!

To be fair, you have a point about the theoretical dynamic range requirements for a tiny proportion of the worlds recordings if peaks are to be kept, and the quietest parts are to remain noise-free (though noise shaped dither can do the job at 16-bits in practice).

Quote
On another hand... IMHO... currently, proportion of recordings that would benefit from SACD/16b+/48k+ is tiny.

Yes.

Many would benefit from good multichannel, but almost no one is proposing this!

Cheers,
David.

Why 24bit/48kHz/96kHz/

Reply #274
I think that recording classical music with 25...35 dB variations in dynamics, huge orchestra, soloists and choirs (as Mahler's 8th, etc) in 2 channel 16/44.1 without serious alterations to material is not possible. that may be reason why classical/jazz jumped to SACD - at least on recording stage


Do you have ABX results to backup this opinion, or is it mere conjecture ?