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

rickman  <gnuarm@gmail.com> wrote:

>You don't mention anything about the anti-alias filter on the playback 
>end.  

(I think reconstruction filter is the better term, but that's fine-
tuning.)

>Are you assuming there is another excellent resampling filter and 
>a high sample rate DAC?  

That's certainly how I would try to design it.  (Conveniently
algorithmic DAC's have this property built in, and are cheaper,
but may suffer in other ways.)

Steve

On 10/27/2016 12:32 PM, Randy Yates wrote:
> 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?
>
I was still being annoyed by the 22kHz whine from EGA monitors into my 
early 40s. I don't think hearing a wide bandwidth until middle age is at 
all rare.

Regards,
Steve

On 27.10.16 07.11, rickman wrote:
> 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.

/Every/ nowadays Audio DAC uses oversampling or delta sigma. Every CD 
player, every sound card, including the $20 class. This is quite simple 
and implemented in hardware.

Increasing the data bandwidth definitely does not make things cheaper. 
Think also of cordless transfers, where BT is already insufficient to 
deliver the bandwidth for 16/48 and compression tricks need to be used 
to get reasonable quality.


Marcel

Steve Underwood <steveu@dis.org> writes:

> On 10/27/2016 12:32 PM, Randy Yates wrote:
>> 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?
>>
> I was still being annoyed by the 22kHz whine from EGA monitors into my
> early 40s. I don't think hearing a wide bandwidth until middle age is
> at all rare.

That is a surprising statement, but not really my point, which is,
WHATEVER the 0.0000001 percent probability is, set the sample rate to
cover that (and a little more) and get on with it.

The reason we haven't been able to do that is that there haven't been
any extensive, proper, authoritative experiments done to identify what
that curve is (i.e., highest frequency audible versus percent of
population, down to some TBD low percentage).

BTW, my hearing was never above about 16 kHz since my 20s. But I played in
rock bands when I was young...
-- 
Randy Yates, DSP/Embedded Firmware Developer
Digital Signal Labs
http://www.digitalsignallabs.com

On Wednesday, October 26, 2016 at 11:25:35 PM UTC-5, Randy Yates wrote:

> OK, so if ringing is an issue, sample a little higher. 64 kHz? Why do we
> need to go to 96 or 192?!?!

This argument persisted in audiophile circles for decades. Now that 96 kHz, 192 kHz, and even 384 kHz are relatively common, one doesn't see much argument about it any more. So evidently 44.1/48 kHz really wasn't enough, but somewhere around double that is.

Those high sampling rates are not as wasteful of storage space or transmission bandwidth as one might expect. There is so much redundancy in the signals that lossless compression can be quite effective.

Long ago I read an article that argued 64 kHz / 24 bit would be ideal for audio. I tend to agree with that idea myself, though I cannot hear 21 kHz like I used to. There will always be others for whom too much is never enough.

Greg

robert bristow-johnson <rbj@audioimagination.com> writes:
> [...]
> 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. 

Be careful here: this testing is an "experiment" and there are
requirements (I think - I never studied the "design of experiments") on
the sample size (I think that's the right term..) in order to ensure the
experiment accurately represents the population.

In other words, just because 100 people couldn't hear a difference in
double-blind AB testing doesn't mean 99.9999 percent of the world
population doesn't (e.g.).
-- 
Randy Yates, DSP/Embedded Firmware Developer
Digital Signal Labs
http://www.digitalsignallabs.com

On Thu, 27 Oct 2016 00:25:28 -0400, Randy Yates wrote:

> 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...

Done to audio cables to markedly increase the price:

http://www.jenalabs.com/interconnects/speakercables-13.html  Note the 
prices.

http://www.vhaudio.com/cryo.html

http://snakeriveraudio.com/index.php?main_page=page&id=3

And, of course, tubes: http://www.tubeman.com/cat8_1.htm

And why: http://www.lhup.edu/~DSIMANEK/cargocul.htm

-- 

Tim Wescott
Wescott Design Services
http://www.wescottdesign.com

I'm looking for work -- see my website!

Tim Wescott  <tim@seemywebsite.really> wrote:

> And second, because really steep filters ring.

Bingo.

> 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.

This part I'm not following -- the filters presumably are linear
(so the filter itself adds no new frequency components) and
neither are they underdamped (existing frequency components
are not amplified by the filter).

The ringing artifact shows up as delay spread, as I noted above.
Sounds components that were once separated along the time axis
now overlap.  

Steve

There's an interesting paper written by Cirrus logic on that subject. They call it "pre-echo"

https://www.cirrus.com/en/pubs/whitePaper/DS668WP1.pdf

benjamin.couillard@gmail.com writes:

> There's an interesting paper written by Cirrus logic on that subject. They call it "pre-echo"
>
> https://www.cirrus.com/en/pubs/whitePaper/DS668WP1.pdf

Ben,

Nice reference - thanks!
-- 
Randy Yates, DSP/Embedded Firmware Developer
Digital Signal Labs
http://www.digitalsignallabs.com