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

Les Cargill  <lcargill99@comcast.com> wrote:

>Steve Pope wrote:

>> 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?

>What is the expected effect of this? 

Some combination of edge-case source material and golden-ear listener
will surface an artifact.  

> I can see how delay spread might matter for multipath in
> wireless comms because the frequencies are so much higher,
> but I fail to see how this would make any difference in audio.

I think in both cases a dimensional argument provides intuition.

In comms, when the delay spread approaches the symbol time, performance
starts to degrade, unless you can do something about it algorithmically.

In audio, the analogy to the "symbol time" is the interval over which
the signal is short-term stationary, and the analogy to the "delay spread"
is some low multiple of the inverse of the width of any filter transition 
band.  

For a music signal with fidelity requirements, this can be a pretty short 
interval.  So, if a steep filter rolls off over 1 or 2 KHz, take the
inverse of that and multiply by something (I'd say 3 to 5) and then start 
hoping that your source signal is stationary over that time interval.

If you instead have a filter that rolls off over 10 to 20 KHz, things
look much much nicer -- the odds are much better that the time constants
in that filter are comfortably shorter than those for any source 
signal statistics.  

That's how I see it, anyway -- hardly an airtight proof, but I think
a recipe for staying out of trouble.

Steve

On Sat, 29 Oct 2016 23:46:38 -0500, Les Cargill <lcargill99@comcast.com>
wrote:

>> OK, so if ringing is an issue, sample a little higher. 64 kHz? Why do we
>> need to go to 96 or 192?!?!
>
>Because 48KHz is commonly used for film work. Having the available
>rates be 48/96/192 would unify the music recording business with the 
>film audio business.

That is a point well worth mentioning. As I recall, it was one of the
reasons that original DAT recorders sampled at 48K ... another reason
being that the industry didn't want for consumers to be able to make
bit-perfect digital DAT copies of their 44.1K music CDs.

On Sat, 29 Oct 2016 23:51:20 -0500, Les Cargill <lcargill99@comcast.com>
wrote:

>Sorry; it will always be audiphoolery to me. There is nothing of actual 
>value above 14k in almost all program material. Besides; what will you
>play it back on? Show me a speaker that can represent 25k accurately.

That is the crux of the objective-vs-subjective battle that raged for so
long, but has definitely eased now that higher sampling rates are
readily available. To say, "There is nothing of actual value above 14k
in almost all program material" is to make a subjective judgment with
which others might not agree. To impose that judgment upon others is no
more appropriate than for them to impose theirs upon you. 

If you are satisfied with 44.1K, then it's a wonderful thing because
there is a lot of source material available. If others prefer higher
sampling rates, even if their preference baseless, well, then, let their
money be heard.

Greg

On Wednesday, October 26, 2016 at 9:20:45 PM UTC-5, 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.
> -- 
> Randy Yates, DSP/Embedded Firmware Developer
> Digital Signal Labs
> http://www.digitalsignallabs.com

I would like to see someone address difference in the waveform shape of the highest violin note sampled originally at 44khz vs 192 kHz (think of how a signal progressively becomes a square wave as more odd harmonics are added). Then address the perceived sound difference (i.e. playback quality) of the two stored waveforms when played back at 44khz.

Maurice Givens

On 10/30/2016 12:54 AM, Les Cargill wrote:
> rickman wrote:
>> 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.
>>
>
> But this is making the razor cheaper and the blades more expensive.
> Markets don't usually work that way.

They do if it gives them even one tiny iota of improved performance they 
can then crow about in advertising.  Heck, years ago Verizon spent huge 
money to install a specific form of voice compression that had only the 
slightest edge in accuracy of reproduction.  Most people couldn't hear 
the difference.  But it gave them the right to claim superior performance.

-- 

Rick C

rickman <gnuarm@gmail.com> writes:

> 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") 

"...the best possible audio..."? Define "best possible."

What I am advocating is uncoupling techniques which are useful in A/D
conversion from that which is necessary to communicate the recorded
audio information to the playback system.

> and minimizing the requirements on the much lower cost playback
> mechanisms. A higher sample rate on the disk would do that.

Cheap delta-sigma D/As abound, don't they? That is, there is little need
to STORE stuff at super-high sample rates on the digital audio delivery
medium.
-- 
Randy Yates, DSP/Embedded Firmware Developer
Digital Signal Labs
http://www.digitalsignallabs.com

Randy Yates  <yates@digitalsignallabs.com> wrote:

>What I am advocating is uncoupling techniques which are useful in A/D
>conversion from that which is necessary to communicate the recorded
>audio information to the playback system.

To a large extent, this is already how things work.  Singal
paths in recording studio equipment are much better spec'ed
than the CD Audio standard, particularly in dynamic range.
As part of mixdown and final production, the product is
assessed for how it will sound in a variety of (inferiorly
spec'ed) final targets.  

Steve

spope33@speedymail.org (Steve Pope) writes:

> Randy Yates  <yates@digitalsignallabs.com> wrote:
>
>>What I am advocating is uncoupling techniques which are useful in A/D
>>conversion from that which is necessary to communicate the recorded
>>audio information to the playback system.
>
> To a large extent, this is already how things work.  Singal
> paths in recording studio equipment are much better spec'ed
> than the CD Audio standard, particularly in dynamic range.
> As part of mixdown and final production, the product is
> assessed for how it will sound in a variety of (inferiorly
> spec'ed) final targets.  

Agreed. The problem is that some people may reasonably argue that the
current CD standard is too limited in resolution and/or sample rate. As
Tim can attest, some people can hear up to 24 kHz. Also I think in some
special borderline cases >16 bits resolution could be useful. So an
updated audio standard may be reasonable (for some value of "reasonable"
ha ha).

The motivation I had for starting this thread was to question the
argument which I've heard some people make that such an updated audio
standard would need astronomical sample rates "to ease the ADC
conversion."
-- 
Randy Yates, DSP/Embedded Firmware Developer
Digital Signal Labs
http://www.digitalsignallabs.com

Randy Yates  <yates@digitalsignallabs.com> wrote:

>spope33@speedymail.org (Steve Pope) writes:

>> Randy Yates  <yates@digitalsignallabs.com> wrote:

>>>What I am advocating is uncoupling techniques which are useful in A/D
>>>conversion from that which is necessary to communicate the recorded
>>>audio information to the playback system.
>>
>> To a large extent, this is already how things work.  Singal
>> paths in recording studio equipment are much better spec'ed
>> than the CD Audio standard, particularly in dynamic range.
>> As part of mixdown and final production, the product is
>> assessed for how it will sound in a variety of (inferiorly
>> spec'ed) final targets.  

>Agreed. The problem is that some people may reasonably argue that the
>current CD standard is too limited in resolution and/or sample rate. As
>Tim can attest, some people can hear up to 24 kHz. Also I think in some
>special borderline cases >16 bits resolution could be useful. So an
>updated audio standard may be reasonable (for some value of "reasonable"
>ha ha).

>The motivation I had for starting this thread was to question the
>argument which I've heard some people make that such an updated audio
>standard would need astronomical sample rates "to ease the ADC
>conversion."

I haven't encountered that argument.

I suppose a case could be made that a single sample rate (say, 96 Ks/sec)
should apply to the entire signal path: all ADC's, all studio equipment, 
through production and distribution all the way to the listener who
finally converts this 96 Ks/sec stream into analog.  That way you 
avoid once and for all the sorts of artifacts that can arise from 
too-sharp filtering as one juggles multiple sample rates.  Thing is, 
there is nobody who can make that kind of decision by fiat, and
it may in fact make no economic sense.

Steve

> 
> I suppose a case could be made that a single sample rate (say, 96 Ks/sec)
> should apply to the entire signal path: all ADC's, all studio equipment, 
> through production and distribution all the way to the listener who
> finally converts this 96 Ks/sec stream into analog.  That way you 
> avoid once and for all the sorts of artifacts that can arise from 
> too-sharp filtering as one juggles multiple sample rates.  Thing is, 
> there is nobody who can make that kind of decision by fiat, and
> it may in fact make no economic sense.
> 
> Steve

right now, by far, the biggest impediments to quality sound delivery to the consumer have nothing to do with the technology but rather are the intentional decisions made by producers ( or their bosses for various reasons) to  squash the dynamic range of the material to make it "louder then the competition".    aka the loudness wars.  

Even if we had 24 bits, the material would all be jammed up into the top 6 dB of the range.





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