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

Steve Pope wrote:
> Les Cargill  <lcargill99@comcast.com> wrote:
>
>> Greg Berchin wrote:
>
>>> 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.
>
>> That's very far from even remotely settled. What's true is that some
>> classes of plugins work better at higher SR, and that realtime latency
>> can be lower at higher SR. But nobody who knows what they are talking
>> about is proposing a higher end distribution SR.
>
> That's quite an assessment.  It may be true for market reasons
> (nobody ever lost money selling compressors, whereas they surely
> lost money selling SACD), but this conversation is about the technical
> necessities.
>

Okay - for *end playback*, 16/44.1k is extremely entrenched and 
perfectly adequate. Those are big things to move to have any sort of 
change.

It also doesn't help that you get high-profile folks like Neil Young 
spouting nonsense about Pono.


And I think the big problem is more that of demography - the people who
are gonna buy plastic discs are too old to buy too many more, and
the kids might buy *vinyl*, so long as that's a thing, but I don't think
anybody's gonna resell the Stones back catalog again.

Maybe there's a market for high-bandwidth download, but I don't see
one. And who's gonna release essentially the master tapes without
some sort of DRM? DRM would probably kill it.

The real advance of CD was that the playback gear either worked or it 
didn't and in a short amount of time, playback that was very good became
cheap veryu fast. With vinyl. you had to pony up to get stuff that had 
all the resonance taken out, and there was wear.

I had a 1988 model CD player, and it was like $400. Five years later
they were $90 and better - a Discman sounded audibly better.

> Are you saying the observations presented here about pre-ringing, pre-echo
> or delay spread are not technically informed?  (As opposed to, difficult
> to sell, a point to which I will cotton to.)
>

No, I am saying that the perceptual/quality effects of them are 
questionable at best and very likely moot.

Speakers and rooms are much more of a problem.

If you're using a 24 dB/octave filter, no fair complaining about a 
little pre-ringing :)

> That too-steep filters introduce artifacts is a basic fact for anyone
> working in audio.  You only use them when you need to achieve some
> purpose, you don't -- except under extreme pressure -- bake them into
> your entire signal path.
>
Sure. But that's not *really* part of the issue with consumer end 
playback gear in the days of high rate sigma-delta.

> That handful of those veterans who worked on the CD audio format and
> who were kind enough to apprise me of what transpired paint a picture of
> everyone believing it was a horrible underdesign, pushed on the
> standards engineers by fearful managers who didn't really understand
> what was technically possible at market cost.
>

That's interesting. I was totally unaware of that. Now *that* is
a story.

> That it took a decade or so for the industry to tame the artifacts
> with workarounds lends credence to this history.
>

But tamed they are. I gotta say - I'm pretty surprised it had many
people who it scared. But a lot of "DDD" records from that
era were pretty distorted - there was a lot of learning curve.

There *was* a lot of crappy digital gear. One legendary one was the
Panasonic SV-7000.

> My own experience, way back when, in professional audio was that we
> used more resolution (18 bits or more), and did not come nearly as
> close to Nyquist (e.g. typical might be 16 KHz bandwidth at 50 Ks/sec.)

The only old, pro, digital gear I'd used was a Mitsu DASH and I can't 
say enough nice things about it. You needed a good console with one, but 
the shop had a D&R Merlin and it was a great setup.

Really, just get the mic placement right and get out of the way.

> Of course that did not involve mass market with cost pressure yada yada.

Sure.

> But it did involve engineers who would personally lose their jobs
> if the thing didn't actually sound good enough.
>

One of the truly shocking things about CD was when I found out
a lot of the hash from popular recordings ( Hotel California was
the poster child ) was not due to the vinyl process but was on the 
original tracks.

It's kind a lousy sounding record, really. So I will always have to
wonder how good stuff really sounded  off the raw tracks.

But face it; those guys were pretty gacked up and that doesn't
help.

> Steve
>

-- 
Les Cargill

Greg Berchin wrote:
> 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.

Well, it wears the clothing of objectivity anyway :) - LPF some
stuff about 14k and see what's left.

> To impose that judgment upon others is no
> more appropriate than for them to impose theirs upon you.
>

It's more of an observation about population dynamics - there aren't
too many musical instruments that go there.

> 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.
>

And it's easy to do so these days. More power to them.

And to be clear - if I was making an "important" recording, I'd
probably use 24/96k just in case I'm wrong. This being said,
you need the rest of the recording chain to not put stuff up there
that's bad.

> Greg
>

-- 
Les Cargill

Steve Pope wrote:
> 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.

Case in point - I have a cheap Focusrite Scarlett 18i20 and if
you loop it back with a guitar cable, fire a 1KHz tone at it,
all the artifacts are below -96dB and just a few are above -110dB.

This is a consumer level USB interface, made to a price point.

> 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.
>

Quality of sound isn't an actual concern for stuff that people
think will make money. Er, rather, the version of the word "quality"
that you would use, and that I would use, is different from what is
used there.

> Steve
>

-- 
Les Cargill

Steve Pope wrote:
> 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.
>

Seems little fruit for a lotta  bandwidth :)

>> 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.
>

That's in interesting way of looking at it. But my experience is that
you can add *significant distortion* in the 10-20KHz octave, and most or 
many people will perceive this as an improvement. Various aural
exciters do this sort of thing.

My guess about this is that "louder is better" applies in a way that
works differently from lower frequency signals.

> Steve
>

-- 
Les Cargill

makolber@yahoo.com writes:

>> 
>> 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.

mark,

I am tempted to agree with you. I wonder how many people would be
willing to pay a premium for high-definition, uncompressed versions.
Perhaps $$$ would speak to these people.
-- 
Randy Yates, DSP/Embedded Firmware Developer
Digital Signal Labs
http://www.digitalsignallabs.com

On Wednesday, November 2, 2016 at 5:30:22 PM UTC-7, Les Cargill wrote:
> Greg Berchin wrote:
> > 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.
> 
> Well, it wears the clothing of objectivity anyway :) - LPF some
> stuff about 14k and see what's left.
> 
> > To impose that judgment upon others is no
> > more appropriate than for them to impose theirs upon you.
> >
> 
> It's more of an observation about population dynamics - there aren't
> too many musical instruments that go there.
> 
...
> Les Cargill

How about:
http://www.cco.caltech.edu/~boyk/spectra/spectra.htm#I

Dale B. Dalrymple

dbd wrote:
> On Wednesday, November 2, 2016 at 5:30:22 PM UTC-7, Les Cargill wrote:
>> Greg Berchin wrote:
>>> 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.
>>
>> Well, it wears the clothing of objectivity anyway :) - LPF some
>> stuff about 14k and see what's left.
>>
>>> To impose that judgment upon others is no
>>> more appropriate than for them to impose theirs upon you.
>>>
>>
>> It's more of an observation about population dynamics - there aren't
>> too many musical instruments that go there.
>>
> ...
>> Les Cargill
>
> How about:
> http://www.cco.caltech.edu/~boyk/spectra/spectra.htm#I
>
> Dale B. Dalrymple
>

Thanks for the link.

That sort of reinforces my bias going in :) I'm not complaining, mind
you, but it's just difficult to assign a value to.

-- 
Les Cargill

That's an interesting link , better researched than most. 

There a phenomenon called blind site, where people who are blind due to damage to their visual cortex or optic nerve can still walk through a room and avoid objects on the floor. This is caused by other neural pathways that bypass the visual cortex, so people with this condition have an "impression" that something is in their path without being able to "see" it. I've always wondered if there is a similar mechanism at work in the auditory system. 

Bob

On Thu, 3 Nov 2016 06:04:59 -0700 (PDT), radams2000@gmail.com wrote:

>That's an interesting link , better researched than most.=20
>
>There a phenomenon called blind site, where people who are blind due to dam=
>age to their visual cortex or optic nerve can still walk through a room and=
> avoid objects on the floor. This is caused by other neural pathways that b=
>ypass the visual cortex, so people with this condition have an "impression"=
> that something is in their path without being able to "see" it. I've alway=
>s wondered if there is a similar mechanism at work in the auditory system.=
>=20
>
>Bob

Some blind people that learn to navigate with taps or clicks and echo
locate say that they "see" the images of objects around them once they
adapt to processing the audio.   People who were sighted and lost
their sight describe this, so it's probably a good analog.

Dolphins "see" things inside other things as they can routinely pick
out opaque containers that have a particular shaped object inside
them.

So there is a fair amount of evidence that the brain is able to hook
all that stuff together.

On Thursday, October 27, 2016 at 3:20:45 PM UTC+13, 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

Quantization noise is lower too. Ok so maybe you can't hear the difference but I am sure those golden ear hippies that listen to old 33 records can