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Topic: Why 24bit/48kHz/96kHz/ (Read 413644 times)
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Why 24bit/48kHz/96kHz/

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On a sidenote; recorders may want higher frequencies if they want the frequencies generated above 22khz (analog) in their production. For instance, I have had interesting results playing back 96khz recordings of cymbals at 44khz. I believe quite some artists use this technique, especially for electronic music.
[a href="index.php?act=findpost&pid=353442"][{POST_SNAPBACK}][/a]

What do you mean? If you playback anything on 44.1 kHz, you will have no energy content above 22.05 kHz, in fact, due to anti-aliasing filtering the roll-off will even start at a lower frequency.
[a href="index.php?act=findpost&pid=353546"][{POST_SNAPBACK}][/a]

True, but that's not the point.

If I record some instrument at 192khz, I may want the output played back at 44khz more for the piece I am making than for the usual sound of the instrument. The point is that if you play an instrument, it produces interesting sounds above the frequencies we hear that we cannot hear if we don't play back the music at lower rates.

Why 24bit/48kHz/96kHz/

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The point is that if you play an instrument, it produces interesting sounds above the frequencies we hear that we cannot hear if we don't play back the music at lower rates.
[a href="index.php?act=findpost&pid=353588"][{POST_SNAPBACK}][/a]

That statement does not make sense.

-Chris

Why 24bit/48kHz/96kHz/

I think what he's trying to say is this:

If you record up to 192kHz, and then shift the upper frequency range to a range that is within our human hearing range, you can make nice effects.

Why 24bit/48kHz/96kHz/

I think he means playing 192khz sample at 44khz WITHOUT converting it to 44khz beforehand...... so that in effect it will be played at much slower speed and pitched down.
I am arrogant and I can afford it because I deliver.

Why 24bit/48kHz/96kHz/

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As for the case of 24 bits, I supposed 16 bits audio isn't enough to represent the entire dynamic range of the human hearing. At its most sensitive region, around 2 - 4 kHz, this dynamic range is actually greater than 100 dB.
[a href="index.php?act=findpost&pid=353575"][{POST_SNAPBACK}][/a]

Aside from the safety concerns that Garf has pointed out, it should be noted that when you consider the noise floor of even a very quiet(but not typical) environment(35dB range), plus the SPL capabilities of the transducers, it does not make technical sense to have greater than CD's dynamic range abilities. For starters, one would need large horn speakers or massive line arrays as a prerequisite. 2nd, live(unamplified) music simply does not use the dynamic range of CD, even if uncompressed, from an audience perspective. Now, obviously if you are sitting in the band(or in a small practice room with a band playing within it), you will be experiencing very high SPLs(and you had better be wearing hearing protection) that will strain, and possible exceed in some cases, the RBCD dynamic range capability. You will also be exceeding the safe limits of human hearing.

-Chris

Why 24bit/48kHz/96kHz/

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Practise disagrees with your suppositions. Is there any proof that we can hear beyond 96dB SNR?

Practical experiments have shown most listeners already get into problems hearing to the 13th bit (78db SNR). If you think you (or another person) can do better, please let them take the MAD challenge with whatever equipment and ears you want, come back aftwards, and see if you would still tell me there is a practical reason to have more.

For the record, claiming you hear over 100dB SNR equals claiming you can hear the person next to you breathing while listening to a live rock concert. Still sounds reasonable?
[a href="index.php?act=findpost&pid=353304"][{POST_SNAPBACK}][/a]

Actually, I can see only one reason why 24 bits *may* be worth it. While I really doubt you can hear noise with a 96 dB SNR, the extra bit can give you extra dynamic range. You would normally want to set the least significant bit to "match" the absolute threshold of hearing. That means that with 16-bit resolution, the loudest sound you can reproduce is 96 dB louder. That means you couldn't reproduce (in the same track) the sound of someone breathing 5 meters away *followed* by the sound of a jet taking off. Why anyone would want to do that is beyond me, but I'm wondering if 16 bits is actually enough for movie theatres.

Why 24bit/48kHz/96kHz/

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Actually, I can see only one reason why 24 bits *may* be worth it. While I really doubt you can hear noise with a 96 dB SNR, the extra bit can give you extra dynamic range. You would normally want to set the least significant bit to "match" the absolute threshold of hearing. That means that with 16-bit resolution, the loudest sound you can reproduce is 96 dB louder. That means you couldn't reproduce (in the same track) the sound of someone breathing 5 meters away *followed* by the sound of a jet taking off. Why anyone would want to do that is beyond me, but I'm wondering if 16 bits is actually enough for movie theatres.
[a href="index.php?act=findpost&pid=353600"][{POST_SNAPBACK}][/a]

1) Exactly the same was said a few posts before.

2) Movie theaters don't use CD's.

Why 24bit/48kHz/96kHz/

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but I'm wondering if 16 bits is actually enough for movie theatres.
[a href="index.php?act=findpost&pid=353600"][{POST_SNAPBACK}][/a]

Just to point out the obvious: What's the noisefloor in a theatre full of people when they are being as quiet as possible? 55-60dB at best?

-Chris

Why 24bit/48kHz/96kHz/

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The question is why one would need a SNR of >96dB or a bandwidth of over 22kHz for end user playback.
[a href="index.php?act=findpost&pid=353050"][{POST_SNAPBACK}][/a]

Not needed for playback, of course, but for editing both images and sound, the higher the resolution, the more precise the result.

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I have to step out and disagree that there is no audible difference between 16-bit and 24-bit recordings. The decision to use 16/44.1 was arrived at, not because it was audio nirvana, but because it was the best comprimise possible with the digital to analogue conversion technology of the 80's.
[a href="index.php?act=findpost&pid=353234"][{POST_SNAPBACK}][/a]

Also to make the lenght of most musical works fit into a CD.

Why 24bit/48kHz/96kHz/

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Just to point out the obvious: What's the noisefloor in a theatre full of people when they are being as quiet as possible? 55-60dB at best?

For theatre as in not movies, I've seen 40-45 dB;  of course, I didn't have a sonometer, but I could hear the actess whispering, and I was over 20 meters away.

Why 24bit/48kHz/96kHz/

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,Dec 31 2005, 02:25 PM]
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Just to point out the obvious: What's the noisefloor in a theatre full of people when they are being as quiet as possible? 55-60dB at best?

For theatre as in not movies, I've seen 40-45 dB;  of course, I didn't have a sonometer, but I could hear the actess whispering, and I was over 20 meters away.
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I don't see how you can know it was 40-45dB without a SPL meter, but perhaps you are correct. I have never experienced a floor that I would rate that low in any audience. 40dB would be good for an average home environment, with no one making a sound. But let's say it's 40dB. What is the masking depth of the format floor(which is basicly broadband white noise) under this environment noise floor(which usually resembles braodband noise)? Assuming -10 or -15 dB is sufficient, you have 120 dB peak ability with that lowly 16 bit format. 120dB is extremely loud, and beyond the capability of most home playback systems(exepting large horn speakers and speaker arrays). In the only musical playback perceptual [1]test of SNR of which I'm aware, only -74dB was required for transparent playback on speakers, with a little bit more neded for headphones. But you(or someone here) brought up special effects like a plane flying directly overhead. I don't think that anyone would actually want to reproduce this, even for movie use.

-Chris

[1]Signal-to-Noise Ratio Requirement for Digital Transmission Systems
Spikofski, Gerhard
AES Preprint: 2196

Why 24bit/48kHz/96kHz/

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The main reason why sampling rates > 44.1Khz is used is that at the hardware / player, the required transition bandwidth of the analogue low pass anti-aliasing filter is much - much wider and thus a simple low order analogue filter is sufficient.  Building a low-order linear phase low pass filter is very simple compare to a high order filter!

I know this argument, but does it have any bearing on practise? I mean, is this still a practical problem anywhere? It certainly doesn't seem to be! Even if we can't get the theorethical resolution because of limitations of real world filters, we certainly seem to be able to do well enough very cheaply that the results are perfect in a psychoacoustical sense.

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As for the case of 24 bits, I supposed 16 bits audio isn't enough to represent the entire dynamic range of the human hearing. At its most sensitive region, around 2 - 4 kHz, this dynamic range is actually greater than 100 dB.
[a href="index.php?act=findpost&pid=353575"][{POST_SNAPBACK}][/a]

Yes, and the amount of records that actually need such a dynamic range is? One in ten million?

As I already explained, I do not think that we need a dynamic range so far that the lower end of audibility corresponds with a higher end where the hearing of the listener is damaged by anything but very short term exposure. That's not only useless, it's actually dangerous.
[a href="index.php?act=findpost&pid=353578"][{POST_SNAPBACK}][/a]

Well I have to admit that I am not really an expert in analogue filter design but from what I have read "somewhere", it is very difficult to design a high order filter with linear phase characteristics besides requiring very expensive high precision resistors and capacitors and inductors !

I supposed this must have been the main drive to higher sampling rate systems as the audio hardware industry is a "cut-throat" business.

Why 24bit/48kHz/96kHz/

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The main reason why sampling rates > 44.1Khz is used is that at the hardware / player, the required transition bandwidth of the analogue low pass anti-aliasing filter is much - much wider and thus a simple low order analogue filter is sufficient.  Building a low-order linear phase low pass filter is very simple compare to a high order filter!

I know this argument, but does it have any bearing on practise? I mean, is this still a practical problem anywhere? It certainly doesn't seem to be! Even if we can't get the theorethical resolution because of limitations of real world filters, we certainly seem to be able to do well enough very cheaply that the results are perfect in a psychoacoustical sense.

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As for the case of 24 bits, I supposed 16 bits audio isn't enough to represent the entire dynamic range of the human hearing. At its most sensitive region, around 2 - 4 kHz, this dynamic range is actually greater than 100 dB.
[a href="index.php?act=findpost&pid=353575"][{POST_SNAPBACK}][/a]

Yes, and the amount of records that actually need such a dynamic range is? One in ten million?

As I already explained, I do not think that we need a dynamic range so far that the lower end of audibility corresponds with a higher end where the hearing of the listener is damaged by anything but very short term exposure. That's not only useless, it's actually dangerous.
[a href="index.php?act=findpost&pid=353578"][{POST_SNAPBACK}][/a]

Well I have to admit that I am not really an expert in analogue filter design but from what I have read "somewhere", it is very difficult to design a high order filter with linear phase characteristics besides requiring very expensive high precision resistors and capacitors and inductors !

I supposed this must have been the main drive to higher sampling rate systems as the audio hardware industry is a "cut-throat" business.
[a href="index.php?act=findpost&pid=353693"][{POST_SNAPBACK}][/a]

Sorry, perhaps higher sampling rate allows the usage of better DA converters instead of the usual sigma-delta oversampling DA converters?

Why 24bit/48kHz/96kHz/

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Do you have many subsequent A/D converters in your recording chain

In most cases, yes.

When editing "purely" digital audio, I agree there is not much sense in upsampling and downsampling (this is not true for changing the bit depth), but when recording from analogue source, my ears told me many times that it is better to have higher sampling rates when editing.

Cheerz =]
Only the best is good enough.

Why 24bit/48kHz/96kHz/

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Another thing to consider is that with proper dithering and noiseshaping, the effective resolution at 44.1kHz can be boosted by as much as 15dB. This means that a properly produced CD (unfortunately very few exist, but HDCD are an example) will have an effective SNR of 111dB.

How many ADC or DAC's have you seen that can attain that resolution?
[a href="index.php?act=findpost&pid=353582"][{POST_SNAPBACK}][/a]

Hi Garf,

How does the calculation exactly go? Is there less noise involved, or is the effective resolution associated with a hearing curve?

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

Why 24bit/48kHz/96kHz/

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1) The 44.1kHz has its historical reasons, but how about 48kHz, 96kHz, or even 192kHz?
How are these numbers chosen? Any technical and practical advantage over 44.1kHz? And why creative cards resamples to these sampling rates?

Just thought I'd pipe up and answer this one. I think all sound cards that adhere to the AC'97 spec have to have the ability to resample to 48KHz, some vendors do this in hardware while others just have it as an option.

It's a common complaint on HTPC forums when people can't get bit-perfect output from their PCs into external DACs

Why 24bit/48kHz/96kHz/

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Another thing to consider is that with proper dithering and noiseshaping, the effective resolution at 44.1kHz can be boosted by as much as 15dB. This means that a properly produced CD (unfortunately very few exist, but HDCD are an example) will have an effective SNR of 111dB.

How many ADC or DAC's have you seen that can attain that resolution?
[a href="index.php?act=findpost&pid=353582"][{POST_SNAPBACK}][/a]

Hi Garf,
It's quite perturbing to talk about 111dB of SNR with a DAC having 96dB (16bits).
Actually dithering/NS decrease distortion while increasing the noise. This noise being created in order to be barely audible, we can say that we have a psychoacoustic SNR of 111 dB on a device with a dynamic of 96 dB.
Is that correct?

Why 24bit/48kHz/96kHz/

First: What about ensuring extremely controlled listening conditions, I mean using closed headphones, maybe using noise cancellation, in additon headphones requere much less power than speakers, so clipping of extremely loud peals should be prevented unlikely with the hadling capacity of even the most powerful amplifiers. Could these conditions lead to a significant reduction of the external noise floor in a way that extremely low level sounds could be audable and extremely loud peaks could result undistorted, so that 24 bit could make a difference?

Second: I know phase shift in an audio signal is not audable provided that it is constant in the entire audio spectrum. I was wondering whether filtering introduces frequency depending phase shift so it could become therefore audable?

Third: What do you think of this quote from the article titled "SACD vs. DVD-Audio:
High Definition Formats Evaluation "
http://www.digit-life.com/articles2/sacd-dvd-a/
Do you think It is a pure nonsense?

Why do We Need Frequencies over 20 kHz?

We have come to an important stage. One and the same question has been asked on different forums in threads about formats and their quality - if a human being can't hear frequencies over 20 kHz (in rare instances 21 kHz) then the 22  kHz limitation in CD-DA (according to the Nyquist theorem it's a half of sampling frequency, for 44100 it will be 22050Hz) should pose no problems, shouldn't it? Why do we need 50 kHz, 70 kHz, and even 96 kHz, if we don't hear them anyway? And the second point - if our speakers have the stated pass band of 20Hz-20 kHz on the level of -3dB, then why do we need DVD-A or SACD? Our loudspeakers will not be able to reproduce the required spectrum, and moreover, intermodulation products of ultrasonic frequencies may get into the audible region and distort the original signal. Isn't it better don't have these ultrasonic images?

That's what said Nelson Pass, the well-known guru in the audiophile and audio engineerieng society:

"Although human hearing is generally very poor above 20,000 Hertz, ultrasonic frequency roll-offs produce phase and amplitude effects in the audible region; for example, a single pole (6dB/octave) roll-off at 30  kHz produces about 9 phase lag and 0.5 dB loss at 10  kHz. The effects may be subtle, but their audibility is undesirable in a piece of equipment whose performance is judged by its neutrality." (original). Thus, being aware of our hearing sensitivity to phase distortions, we can presume considerable decrease in the level of such distortions in the systems with a wider signal spectrum (including the quality LP playback).

Acoustical Spectrum of Real Instruments

The second point - if we limit the original signal with the analog anti-aliasing filter for the CD signal spectrum in comparison with the same filter for DVD-A 24bit 96 kHz signal when trying to record the spectrum of such instruments as a trumpet, there will be a considerable difference in phase distortions between the original and the record even before it is delivered to the speakers. The speakers (well-designed) have mildly sloping signal falloff characteristic at high frequencies, which mostly depends on the tweeter design, and thus introduces fewer phase distortions than the filter with a steep amplitude-frequency response falloff, which is in fact the CD-DA format for the wideband spectrum signals.

Enrico

Why 24bit/48kHz/96kHz/

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And why creative cards resamples to these sampling rates?[{POST_SNAPBACK}][/a]

They need to be able to mix together different sounds. You would not like your music to stop when you receive a mail or when you launch a game.
Analog mixers would be much more expensive than a soundcard, plus the cost of one DAC per channel.
So the mix is done digitally. But what will be the output sample rate if one input track is 22050 Hz, another 32 kHz, and another one 44100 Hz ? Well, 48 kHz was chosen.

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It's quite perturbing to talk about 111dB of SNR with a DAC having 96dB (16bits).
Actually dithering/NS decrease distortion while increasing the noise. This noise being created in order to be barely audible, we can say that we have a psychoacoustic SNR of 111 dB on a device with a dynamic of 96 dB.
Is that correct?
[a href="index.php?act=findpost&pid=354797"][{POST_SNAPBACK}][/a]

This is just a matter of frequency. It seems to me that in a 16 bits system, you can't get more than 96 dB SNR at 22049 Hz. But you may get more at lower frequencies.

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I was wondering whether filtering introduces frequency depending phase shift so it could become therefore audable?

According to Nika Aldrich (  [a href="http://www.cadenzarecording.com/papers.html]The Audio Engineer's Approach to Understanding Digital Filters[/url], p35 ), filters used in ADCs and DACs are almost exclusively FIR filters, that do not introduce any frequency dependant phase shift, while getting -144 dB at 22050 Hz and 0 dB at 20000 Hz.
This dismisses all the arguments of Nelson Pass quoted above.

Why 24bit/48kHz/96kHz/

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Why do we need 24bit/48kHz/96kHz/192kHz if 16bit/44.1kHz is good enough?
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We dont need it. It's just virtual useless number-games to give people the incentive to buy new equipment and then re-buy all our music. There are some *technical* arguments for using 48khz instead of 44khz.... but the actual benefit for normal endusers is zero.
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The most thorough advocacy case I've ever seen for greater-than-redbook schemes was laid down by J. Robert Stuart of Meridian (who brought us Meridian Lossless Packing for DVD-Audio)
But even *it* boils down to a case of 'we don't *know* for sure that we really need it, but why not use it since we can?'  I just don't see the point of it for home playback, myself.

[a href="http://www.meridian-audio.com/ara/coding2.pdf]http://www.meridian-audio.com/ara/coding2.pdf[/url]

Why 24bit/48kHz/96kHz/

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I agree that it makes no difference to end users...  though there are technical reasons why this would have its advantages...

most of it comes from the way hardware plays back music.... it has to do with the enforcements of peaks... each sample is essentially enforced as a peak in the reproduced sound wave... but this should NOT be the case... the sample may have been taken along any part of the wave (and most likely not at a peak)...

The affects of this are mostly negligable and hardware also works to minimize the affects... but it shouldn't have to... and higher sampling rates allow the hardware to start  with a closer reresentation to the original wave...

I'll look for some articles about this "phenom"... I actually looked them up a year or two ago cause an audiophile engineer I know was telling me about it and I didn't believe him...
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This really doesn't make any sense at all.
[a href="index.php?act=findpost&pid=353156"][{POST_SNAPBACK}][/a]

He's talking about 'phantom overs' -- signals that don't register as 'overs' during recording/production/mastering  due to poor digital metering.  Some hardware handles these intelligently, others just clip.

[a href="http://www.tcelectronic.com/media/Level_paper_AES109.pdf]http://www.tcelectronic.com/media/Level_paper_AES109.pdf[/url]

Why 24bit/48kHz/96kHz/

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Sorry for popping in like that, but here's my answer to the main question here...

I do a lot of audio editing (some people call me a "producer"). For my needs, 16/44.1 kHz is far from sufficient - after applying some filters and effects, artifacts are very audible and audio is heavily damaged.

Then I suggest you make sure all your work stays in the 32-bit domain until you're done, then convert it down to 16.  A higher sampling rate than redbook standard is probably unnecessary unless you are employing converters that are introducing audible aliasing artifacts with 44 kHz.

Why 24bit/48kHz/96kHz/

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First: What about ensuring extremely controlled listening conditions, I mean using closed headphones, maybe using noise cancellation, in additon headphones requere much less power than speakers, so clipping of extremely loud peals should be prevented unlikely with the hadling capacity of even the most powerful amplifiers. Could these conditions lead to a significant reduction of the external noise floor in a way that extremely low level sounds could be audable and extremely loud peaks could result undistorted, so that 24 bit could make a difference?

Second: I know phase shift in an audio signal is not audable provided that it is constant in the entire audio spectrum. I was wondering whether filtering introduces frequency depending phase shift so it could become therefore audable?

Third: What do you think of this quote from the article titled "SACD vs. DVD-Audio:
High Definition Formats Evaluation "
http://www.digit-life.com/articles2/sacd-dvd-a/
Do you think It is a pure nonsense?

Maybe you want to read an entire HA thread about it -- where the author also chimed in?

http://www.hydrogenaudio.org/forums/index....showtopic=26218

Why 24bit/48kHz/96kHz/

Quote from: krabapple,Jan 6 2006, 11:27 AM

Quote from: enry2k,Jan 5 2006, 07:08 PM
First: What about ensuring extremely Maybe you want to read an entire HA thread about it -- where the author also chimed in?

http://www.hydrogenaudio.org/forums/index....showtopic=26218
[a href="index.php?act=findpost&pid=355101"][{POST_SNAPBACK}][/a]

Thank you for quoting the thread, it is really interesting.

On the same cd audio quality subject, a few yaars ago Ivan Dimkovic, gave me this interesting reply in the audiocoding forum:
Quote [ ]Author: Ivan Dimkovic (---.verat.net)
Date:  01-05-02 06:32

For "studio" quality, sampling rate of 96 kHz and 24-bit fidelity is used. This is not used because somebody would be able to hear 40 kHz tone, or to distinguish between 110 and 105 dB signal strength, but because of filter design and many steps in production.

For "everyday" use, 64 kHz with 24 bit fidelity - with 25 kHz lowpass filter (to avoid AD-DA problems) would be "best of best" - I doubt that somebody would be able to distinguish this from 96 kHz without lowpass.]

So what are the A/D-D/A problems he is reffering to?
Couldn't be possible to use oversampling in the A/D  converter "say 4 times 44.1 Knz " to avoid sharp antialiasing filters and related problems, then in a following stage, throught the useage of FIR filters  to reduce the bandwidth to 20 Knz and sample rate converting the signal down to 44.1 knz without significant loss with no use of analogue steepy filters, to produce a perceptially valuable signal? Oversampling is also commonly employed in D/A converters.

Finally has AES or any other important organizations published any results yet regarding random tests involving the so-called golden ears on 44.1/16 vs 96/24 subjective comparison?

Regards

Enrico

Why 24bit/48kHz/96kHz/

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Another thing to consider is that with proper dithering and noiseshaping, the effective resolution at 44.1kHz can be boosted by as much as 15dB. This means that a properly produced CD (unfortunately very few exist, but HDCD are an example) will have an effective SNR of 111dB.

How many ADC or DAC's have you seen that can attain that resolution?
[a href="index.php?act=findpost&pid=353582"][{POST_SNAPBACK}][/a]

Hi Garf,
It's quite perturbing to talk about 111dB of SNR with a DAC having 96dB (16bits).
Actually dithering/NS decrease distortion while increasing the noise. This noise being created in order to be barely audible, we can say that we have a psychoacoustic SNR of 111 dB on a device with a dynamic of 96 dB.
Is that correct?
[a href="index.php?act=findpost&pid=354797"][{POST_SNAPBACK}][/a]

I don't think dithering necessary adds a noise floor to the signal, in a book titled "Principles of Digital Audio" I read a few years ago, a A/D converter is decribed in which a pseudorandom signal is used to generate noise throught a D/A converter which produces a signal level comparable to the least significant bit. The noise signal is mixed up with the audio signal to be encoded. After A/D conversion, the same random signal is subtracted from the LSB. The resulting signal is said to retain the dither positive effect, without additional noise floor.
I guess this is the same idea  HD CDA was based upon. producing a result similar to 20 bit quatization.

Enrico