What's the use of a 192 kHz sample rate?

On Sat, 3 May 2008 09:25:16 -0700 (PDT), rickman <gnuarm@gmail.com>
wrote:

>On May 3, 12:14 pm, nos...@nospam.com (Don Pearce) wrote:
>
>I thought we were having a conversation.  But I don't appreciate being
>called names.  Would you speak to me this way if I were standing in
>front of you?  Either way you come across as being rude.
>
>Rick

I'll have a reasonable conversation with anyone who can make
reasonable points. Your demand that I prove a negative fell well
outside that area. Talk to me like an idiot, and I will treat you that
way. Your choice.

d
-- 
Pearce Consulting
http://www.pearce.uk.com

"Green Xenon [Radium]" <glucegen1@excite.com> wrote in
news:481becfe$0$5141$4c368faf@roadrunner.com: 

> Hi:
> 
> Why does DVD-Audio use 192 kHz sample rate? What's the advantage over 
> 44.1 kHz? Humans can't hear the full range of a 192 kHz sample rate?
> 
> On average, what is the minimum sample rate for a guy in his early to 
> mid 20s who likes treble?
> 
> I agree there are a small percentage of humans who can hear above 20 
> kHz. However, DVD-audio uses a sample-rate of 192 kHz which allows a 
> maximum frequency of 96 kHz. There is no known case of any human being
> able to hear sounds nearly as high as 96 kHz. I can agree with 48 kHz 
> sample rate and even 96 kHz sample-rate [maybe], but 192 kHz is just
> stupid. 
> 
> So whats the justification fur using 192 kHz? If you ask me, its just
> a total waste of bandwidth and energy. Any proof to the contrary?
> 
> Please correct me if I'm wrong but AFAIK, its a waste of time, money, 
> energy to move to 192 kHz.
> 
> 
> Thanks,
> 
> Radium
> 

Is this a sampling rate or an oversampling rate?


-- 
Scott
Reverse name to reply

On May 3, 8:11 am, Vladimir Vassilevsky <antispam_bo...@hotmail.com>
wrote:
> rickman wrote:
> >>Utter nonsense - unless of course you can cite some proper tests.
>
> > And what do you base this statement on?
>
> The ultimate reason for the audio systems is making the people happy. If
> someone is happy because of 192kHz sample rate, and willing to pay for
> that, then why do you need to proove anything? Heck, if someone orders a
> 192MHz audio system, it would be my pleasure to do this project.
>
> Vladimir Vassilevsky
> DSP and Mixed Signal Design Consultanthttp://www.abvolt.com

I guess,  you are in business of separating fool from his money.

vlad

On May 3, 5:26 pm, dpl...@radagast.org (Dave Platt) wrote:
>
> There may be _some_ justification for it, as it eliminates the need to
> place the knee of the anti-aliasing filter anywhere near the range of
> frequencies that one _can_ hear.  One of the criticisms made against
> CD is that the sharp filtering which must be done at around 20 kHz can
> cause artifacts which may be audible to some listeners, either due to
> "pre-ringing" (with a symmetric FIR low-lass filter) or a frequency-
> dependent delay and "smearing" of transients (with an IIR filter).
>
> These effects can be moved up to higher frequencies, and prevented
> from having effects in the human hearing passband, by increasing the
> sampling rate.

Indeed.  But this problem is largely solved by oversampling at the
DAC, without requiring an increased sample rate at the storage-medium
level.


--
Oli

Steve Underwood wrote:
> Vladimir Vassilevsky wrote:
>> rickman wrote:
>>
>>
>>>> Utter nonsense - unless of course you can cite some proper tests.
>>> And what do you base this statement on?
>>
>> The ultimate reason for the audio systems is making the people happy. 
>> If someone is happy because of 192kHz sample rate, and willing to pay 
>> for that, then why do you need to proove anything? Heck, if someone 
>> orders a 192MHz audio system, it would be my pleasure to do this project.
> 
> 
> Quite right, too. If engineers only built what people actually need, you 
> and I would probably be looking for another occupation.

The only real needs are the air, the food and the water. Everything else 
is a luxury, including sex. The ultimate goal for building anything is 
making somebody happier. If this goal is accomplished by one way or 
another, then the project makes sense.


Vladimir Vassilevsky
DSP and Mixed Signal Design Consultant
http://www.abvolt.com

On May 3, 10:34&#4294967295;am, rickman <gnu...@gmail.com> wrote:
>  &#4294967295;I am simply saying that you shouldn't judge what others can
> perceive by what you can or even what the general public can according
> to "proper tests".

what *should* we base it on?  what the Monster Cable people tell us
that they can hear?

by "proper tests", i am convinced what they mean are truly blind tests
where a sufficient number of subjects, with "good" hearing, are called
on to say if they can hear a difference between two possibly different
sounds or not.  rather than "ABX testing", i think it should be "AB
testing" where the subject hears groups of two sounds, and for each
group the only question the subject must answer is if he/she thinks
the sounds are different or the same (*not* which one sounds "better"
or "less distorted or noisy", etc.).  the test would include an equal
number of identical pairs so that we can unbias it from the subject's
pre-conceived biases.  for each subject you would subtract the number
of false positives from the number of true positives and also subtract
the number of false negatives from the number of true negatives.

now, long ago, when DSD was just coming out, i participated in
something like this that an audio guy (now an author) named Bob Katz
did.  this was before Pro Tools HD and they were using some expensive
system from a company called Sonic Solutions that could do 192 kHz.
they recorded at 192 kHz some test sounds (including some high
frequency percussive sounds like castenets, cabasas, cymbals) along
with synthesized bandlimited (to 96 kHz) tones of all sorts of
waveshapes at a variety of frequencies from below 10 kHz up to 90
kHz.  i don't remember all of the test sounds, but they made sure that
most, if not all, had content well above 30 kHz.  the subject listened
to those sounds in two different forms, but both with a 192 kHz
playback rate.  one form was the raw recorded sound, the other was
processed through a phase-linear FIR filter with a lot of taps (i
thought it was around 300 something taps, it was not a real-time FIR
filter but proceessed one sound file into another) that was flat to
within 0.01 dB up to 20 kHz, had a smooth transition band from 20 to
22 kHz, and then was down by more than 130 dB for 22 kHz to 96 kHz.
we padded the beginning an end of the original sound file with zeros,
of half the length of the FIR on both ends and the non-real-time FIR
was not "causal" and had a delay of 0 (so it started responding before
the first non-zero samples) and the filtered sound file was lined up
in time with the original.

now this isn't what Bob Katz did, but it is what i wished he did:
called the original sound "A", and the LPFed sound "B".  so sometimes
the subject hears AA, sometimes AB, sometimes BA, and sometimes BB
(all four permutations exist in equal quantity) and for each pair, the
subject simply has to say if they think the sounds are the same or
different.  every time a subject says that AA or BB sound different,
we count that as a false positive and subtract that count from the
number of times the subject says that AB or BA sound different.
likewise for false negatives.

now Bob didn't do that, but he did something like it (i think it was
ABX) and there was no statistically measured difference.  people could
simply not reliably tell if the stuff above 20 kHz was removed or
not.  they could not tell at all.  no one could.

now, if they cannot tell if the content existed above 22 kHz or not,
is there a need to have it there in storage or in transmission?   if
there is no need to have it there, and it is removed, what does the
sampling theorem tell us regarding sufficient sampling rate?

> I'm not trying to "prove" anything. &#4294967295;I am presenting information which
> you can consider and believe or can ignore. &#4294967295;But you can't say my
> statements are false unless you have some information to "prove" they
> are. &#4294967295;Human hearing is not a microphone connected to an amplifier. &#4294967295;It
> is a very complex process which even includes the brain and we
> certainly don't understand it completely.

but we can't hear anything above 20 kHz.  even with percussive sounds
with sharp attacks.  and if we cannot hear anything above 20 kHz, then
40.0001 kHz sampling rate can store all of the information we need.
for practical reconstruction purposes, 44.1 and 48 kHz are sufficient.

now, in *processing* sounds with some nasty non-linearities in the
process, it very well may be necessary to upsample to 192 kHz or
higher to do that non-linear processing, and when it is done, LPF to
20 kHz and downsample back to 48 kHz.  but, except for experimental
purposes, 192 kHz storage or transimssion is not necessary.

r b-j

On Sat, 3 May 2008 13:36:10 -0700 (PDT), robert bristow-johnson
<rbj@audioimagination.com> wrote:

> but we can't hear anything above 20 kHz.  even with percussive sounds
> with sharp attacks.  and if we cannot hear anything above 20 kHz, then
> 40.0001 kHz sampling rate can store all of the information we need.
> for practical reconstruction purposes, 44.1 and 48 kHz are sufficient.
 
No. When I was 35 years I could hear up to 24 kHz. 
Now I'm 54 years and can easy hear 19 kHz (haven't tested higher). 
 

Steve Underwood schrieb:
> 
> For people who say supersonic sound can't play a part in a listening 
> experience, trying being in a room with a high intensity of supersonic 
> energy. Under some conditions (I'm not clear which) you can sense it, 
> even though you can't hear it. It actually feels like something loud 
> that you can't hear is going on. Its a very odd feeling. That said, I've 
> never found any evidence that this plays a part in any musical 
> experience. I see no reason to try to capture that energy in a 
> recording, unless you feel your dog should enjoy a greater musical 
> experience.

I remember that there was some kind of ultrasound sound-gun to play with 
on the expo2000 in hannover/germany. It modulated AF-sound onto an 
ultrasound signal and transmitted it in a focused sound-beam.

You could point the beam at people from a far distance, say something 
into a microphone and the sound demodulated right inside their bones. It 
was very direct and you had to be somewhere in a very narrow funnel to 
notice it.

I can tell you: It's a very scary experience because your brain 
localizes these sounds somewhere in your own body.

On Sat, 3 May 2008 13:36:10 -0700 (PDT), robert bristow-johnson
<rbj@audioimagination.com> wrote:

>now, long ago, when DSD was just coming out, i participated in
>something like this that an audio guy (now an author) named Bob Katz
>did.  ...  people could
>simply not reliably tell if the stuff above 20 kHz was removed or
>not.  they could not tell at all.  no one could.

Way back in 1999 I attended a demonstration of SACD by Tom Jung.  He
brought along some sample SACD titles that contained the same material
on both the SACD layer and the Redbook CD layer
(http://www.dmprecords.com/technology.htm).  The difference between
them was quite audible.  (Testing was not blind.)  But the question
is, was it because of the differences in information above 20 kHz, or
was it something related to the filtering applied during Super Bit
Mapping?  I suspect the latter.

Greg

On May 3, 4:58 pm, Ken <ke...@telia.com> wrote:
> On Sat, 3 May 2008 13:36:10 -0700 (PDT), robert bristow-johnson
>
> <r...@audioimagination.com> wrote:
> > but we can't hear anything above 20 kHz.  even with percussive sounds
> > with sharp attacks.  and if we cannot hear anything above 20 kHz, then
> > 40.0001 kHz sampling rate can store all of the information we need.
> > for practical reconstruction purposes, 44.1 and 48 kHz are sufficient.
>
> No. When I was 35 years I could hear up to 24 kHz.
> Now I'm 54 years and can easy hear 19 kHz (haven't tested higher).


well, good for you.  still would like to see how you would do in such
a blind test.

r b-j