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Sample Rate of a Digital Audio Delivery Medium (i.e., CD)

Started by Randy Yates October 26, 2016
Perhaps I'm forgetting something, but why would the sample rate of the
digital audio delivery medium such as CD have _anything_ to do with the
ease of sampling the original signal?

It is certainly true that it is better to oversample the original analog
input so that the antialiasing filter requirements can be greatly
relaxed. However, once we're in the digital domain we can run some very
tight digital filter to resample to 44.1 kHz (or whatever). Right?

Am I missing something? I bring this up because I see people arguing for
a high sample rate in a new digital audio delivery medium (HRA) for this
reason and I just don't see it.
-- 
Randy Yates, DSP/Embedded Firmware Developer
Digital Signal Labs
http://www.digitalsignallabs.com
Randy Yates <yates@digitalsignallabs.com> writes:

> Perhaps I'm forgetting something, but why would the sample rate of the > digital audio delivery medium such as CD have _anything_ to do with the > ease of sampling the original signal? > > It is certainly true that it is better to oversample the original analog > input so that the antialiasing filter requirements can be greatly > relaxed. However, once we're in the digital domain we can run some very > tight digital filter to resample to 44.1 kHz (or whatever). Right? > > Am I missing something? I bring this up because I see people arguing for > a high sample rate in a new digital audio delivery medium (HRA) for this > reason and I just don't see it.
Note that the other reason I see people arguing for it (e.g., 96 kHz), higher bandwidth, is of course a fact, even if I don't think it's necessary. -- Randy Yates, DSP/Embedded Firmware Developer Digital Signal Labs http://www.digitalsignallabs.com
On Wed, 26 Oct 2016 22:20:38 -0400, Randy Yates wrote:

> Perhaps I'm forgetting something, but why would the sample rate of the > digital audio delivery medium such as CD have _anything_ to do with the > ease of sampling the original signal? > > It is certainly true that it is better to oversample the original analog > input so that the antialiasing filter requirements can be greatly > relaxed. However, once we're in the digital domain we can run some very > tight digital filter to resample to 44.1 kHz (or whatever). Right? > > Am I missing something? I bring this up because I see people arguing for > a high sample rate in a new digital audio delivery medium (HRA) for this > reason and I just don't see it.
Two reasons. First, because you can't dip flash chips in LN2, so you have to do _something_ to convince yourself that you're cooler than the other guy. And second, because really steep filters ring. My sisters and my boys can hear sounds significantly above 20kHz, which is probably around where the ringing would happen if you're aiming to sample at 44kHz. -- www.wescottdesign.com
On Wednesday, October 26, 2016 at 11:30:45 PM UTC-4, Tim Wescott wrote:
> On Wed, 26 Oct 2016 22:20:38 -0400, Randy Yates wrote: > > > Perhaps I'm forgetting something, but why would the sample rate of the > > digital audio delivery medium such as CD have _anything_ to do with the > > ease of sampling the original signal? > > > > It is certainly true that it is better to oversample the original analog > > input so that the antialiasing filter requirements can be greatly > > relaxed. However, once we're in the digital domain we can run some very > > tight digital filter to resample to 44.1 kHz (or whatever). Right? > > > > Am I missing something? I bring this up because I see people arguing for > > a high sample rate in a new digital audio delivery medium (HRA) for this > > reason and I just don't see it. > > Two reasons. First, because you can't dip flash chips in LN2, so you > have to do _something_ to convince yourself that you're cooler than the > other guy. And second, because really steep filters ring. My sisters > and my boys can hear sounds significantly above 20kHz, which is probably > around where the ringing would happen if you're aiming to sample at 44kHz. >
then your kids and your sis's can hear above 22 kHz and the Nyquist criterion is not satisfied. if it makes no difference whether we remove components above 22 kHz or not, then Fs=44 kHz is sufficient. i pretty much share the same feeling as Randy, but now if speed and memory are getting cheaper and cheaper, i figger "why not Fs = 96 kHz?" so that we just don't the hell worry about it. (of course, internal nonlinearities can be pre-upsampled and post-downsampled at whatever rate.) but, if using double-blind AB testing (not quite the same as ABX testing), it is shown using Fs = 192 kHz that human beings cannot hear the difference between raw full BW audio and the same signal with the content above 22 kHz removed, if people guessing at "same or different" do no better than random guessing, then the information at Fs=44 kHz is sufficient for archiving or anything. but it doesn't cost much more to do it at 96 kHz. r b-j
Tim Wescott <tim@seemywebsite.really> writes:

> On Wed, 26 Oct 2016 22:20:38 -0400, Randy Yates wrote: > >> Perhaps I'm forgetting something, but why would the sample rate of the >> digital audio delivery medium such as CD have _anything_ to do with the >> ease of sampling the original signal? >> >> It is certainly true that it is better to oversample the original analog >> input so that the antialiasing filter requirements can be greatly >> relaxed. However, once we're in the digital domain we can run some very >> tight digital filter to resample to 44.1 kHz (or whatever). Right? >> >> Am I missing something? I bring this up because I see people arguing for >> a high sample rate in a new digital audio delivery medium (HRA) for this >> reason and I just don't see it. > > Two reasons. First, because you can't dip flash chips in LN2,
You totally lost me with that statement! But if you're basically saying marketing, well then...
> so you have to do _something_ to convince yourself that you're cooler > than the other guy. And second, because really steep filters ring. My > sisters and my boys can hear sounds significantly above 20kHz, which > is probably around where the ringing would happen if you're aiming to > sample at 44kHz.
OK, so if ringing is an issue, sample a little higher. 64 kHz? Why do we need to go to 96 or 192?!?! -- Randy Yates, DSP/Embedded Firmware Developer Digital Signal Labs http://www.digitalsignallabs.com
On 27.10.16 05.30, Tim Wescott wrote:
> On Wed, 26 Oct 2016 22:20:38 -0400, Randy Yates wrote: >> Am I missing something? I bring this up because I see people arguing for >> a high sample rate in a new digital audio delivery medium (HRA) for this >> reason and I just don't see it. > > Two reasons. First, because you can't dip flash chips in LN2, so you > have to do _something_ to convince yourself that you're cooler than the > other guy.
That's the only reason.
> And second, because really steep filters ring. My sisters > and my boys can hear sounds significantly above 20kHz, which is probably > around where the ringing would happen if you're aiming to sample at 44kHz.
There is almost no modern Audio ADC/DAC around that does not use oversampling internally. Just because it is one of the cheapest ways to implement the filters. But there is really no need to propagate this oversampling rate to data storage. Yes, in the mid 90s there were sound chips common that had neither an anti aliasing filter nor a reconstruction filter. - And even that worked! Simply because no reasonable Audio signal contains significant energy above 20 kHz. If so then this is noise from bad recordings or whatever. OK, this sometimes happened and it is a good news that this devices died off. About ringing: A filter with 2 kHz transition is not steep. I have never measured a sound card where the filter had more than 0,5 dB overshoot. Sorry, but no one of the human species can ever hear that. Especially not children. Even if they may be physically aware of higher frequencies they do not know what ringing is, hence they will not notice. (This is always the way our brain works. If we can't identify something we ignore it until it hurts. - Probably we are already Borg.) Marcel
Tim Wescott <tim@seemywebsite.really> writes:
> [...] > My sisters and my boys can hear sounds significantly above 20kHz,
How much higher? 22 kHz? 30 kHz? 1 x 10^9 kHz?!? Somebody needs to shoot this "I can hear up to x kHz" beast. Where is the 0.0000001 percent probably on the normal curve for this? -- Randy Yates, DSP/Embedded Firmware Developer Digital Signal Labs http://www.digitalsignallabs.com
On 27.10.16 04.20, Randy Yates wrote:
> It is certainly true that it is better to oversample the original analog > input so that the antialiasing filter requirements can be greatly > relaxed. However, once we're in the digital domain we can run some very > tight digital filter to resample to 44.1 kHz (or whatever). Right?
Ack.
> Am I missing something? I bring this up because I see people arguing for > a high sample rate in a new digital audio delivery medium (HRA) for this > reason and I just don't see it.
Well, there are many thing on earth that are just done because they /can/ be done. Or do you really think that anybody /needs/ a car with 500 PS? The more the better, you know. This also applies to the sampling rate. You can buy a 24/192 sound device for the Raspberry Pi - WTF? OK, it might be interesting for measurement purposes - a $60 network analyzer is a deal even if it only works up to 80 kHz. But for audio processing it is total overkill. Do you remember SACD? They explicitly sold Hybrid SACD where the CD layer was botched on purpose so that the customers realized that they need SACD. Otherwise no one would have ever been able to identify an audible difference to CD. OK - SACD flopped, but not because it is useless just because of license restrictions (e.g. software dercryption was not allowed). Marcel
On 10/26/2016 10:20 PM, Randy Yates wrote:
> Perhaps I'm forgetting something, but why would the sample rate of the > digital audio delivery medium such as CD have _anything_ to do with the > ease of sampling the original signal? > > It is certainly true that it is better to oversample the original analog > input so that the antialiasing filter requirements can be greatly > relaxed. However, once we're in the digital domain we can run some very > tight digital filter to resample to 44.1 kHz (or whatever). Right? > > Am I missing something? I bring this up because I see people arguing for > a high sample rate in a new digital audio delivery medium (HRA) for this > reason and I just don't see it.
You don't mention anything about the anti-alias filter on the playback end. Are you assuming there is another excellent resampling filter and a high sample rate DAC? I believe the idea is to use the fancy stuff in the already expensive recording process to get the best possible audio on the disk (what you seem to be calling the "digital audio delivery medium") and minimizing the requirements on the much lower cost playback mechanisms. A higher sample rate on the disk would do that. -- Rick C
Randy Yates  <yates@digitalsignallabs.com> wrote:

>Perhaps I'm forgetting something, but why would the sample rate of the >digital audio delivery medium such as CD have _anything_ to do with the >ease of sampling the original signal? > >It is certainly true that it is better to oversample the original analog >input so that the antialiasing filter requirements can be greatly >relaxed. However, once we're in the digital domain we can run some very >tight digital filter to resample to 44.1 kHz (or whatever). Right?
>Am I missing something? I bring this up because I see people arguing for >a high sample rate in a new digital audio delivery medium (HRA) for this >reason and I just don't see it.
A major issue is that those "very tight" digital filters have a very non-zero normalized delay spread. The popular fiction is that if they are linear phase, they have no coloration. To realize this is fiction, take it to the extreme -- could you have a delay spread of one minute and it is still inaudible? One second? (I've pointed this out in the past, but it has fallen on deaf ears.) Steve