25 Jan
2004
25 Jan
'04
6:39 p.m.
Anyone here using 192kHz for sampling?
I know that lower sampling frequencies (44.1, 48, 88.2, 96) are more
popular, but i wonder if someone tried to use 176.4 or 192kHz with Linux
audio software?
If you did, what was the hardware (sound cards) and software (JACK?
Ardour? etc.) that you used?
Any problems? How many stereo channels were you able to use
simultaneously (and what was your CPU and disk speeds)?
Did you try to apply LADSPA filters on 192kHz channels? How long before
the system became unresponsive and started to drop frames?
--
Florin Andrei
http://florin.myip.org/
26 Jan
26 Jan
5:23 a.m.
Florin Andrei wrote:
Anyone here using 192kHz for sampling?
are you trying to record for dogs or bats ? ;)
seriously, the only measurably effect of 192k is that it fills up your
disks faster and eats more cpu when processing.
imnsho, all this sampling rate hype is a clever industry ploy to keep
people from thinking that their hardware is finally good enough and they
can stop buying now.
people can't hear over 20k. period. 48k sampling rate gives you 24k
minus what's cropped off by the aliasing filter. granted, higher
sampling allows you to use a simpler, less steep aliasing filter, and
some people claim to perceive an improvement from that. but even then,
96k should be enough.
--
"I never use EQ, never, never, never. I previously used to use mic
positioning but I've even given up on that too."
- Jezar on http://www.audiomelody.com
Jörn Nettingsmeier
Kurfürstenstr 49, 45138 Essen, Germany
http://spunk.dnsalias.org (my server)
http://www.linuxaudiodev.org (Linux Audio Developers)
5:43 a.m.
On Mon, Jan 26, 2004 at 10:19:59 +0100, Joern Nettingsmeier wrote:
imnsho, all this sampling rate hype is a clever
industry ploy to keep
people from thinking that their hardware is finally good enough and they
can stop buying now.
Probably true but...
people can't hear over 20k. period. 48k sampling
rate gives you 24k
minus what's cropped off by the aliasing filter. granted, higher
sampling allows you to use a simpler, less steep aliasing filter, and
some people claim to perceive an improvement from that. but even then,
96k should be enough.
This is not quite correct. While humans can not hear continuous tones over
20-odd kHz that is not the whole story, the frequency reproduction also
limits the minimum transient rise time, which is detected by a different
part of the ear (IIRC, IANABiologist and my psychoacoustics textbook is at
work).
I dont know wether I, personally can tell the difference between a 48 and
96 recording, but I'm not going to rule it out.
- Steve
7:01 a.m.
Steve Harris wrote:
On Mon, Jan 26, 2004 at 10:19:59 +0100, Joern
Nettingsmeier wrote:
interesting point. but isn't membrane inertia the limiting factor in
transient reproduction anyway?
my naive understanding is that min rise time = nyquist freq. if we
cannot perceive frequencies above, say, 20k, all we win by faster
sampling is more accurate timing information.
but there is a worst-case "timing error" of 1/24000 sec, which does not
seem much to me...
i'd like to read more, but i haven't been able to google anything up
about transient perception and reproduction. any pointers?
for the record, here's a 1998 aes paper that elaborates on the
consequences of higher sampling on filter artifaces in the audible
range: http://www.nanophon.com/audio/antialia.pdf
--
"I never use EQ, never, never, never. I previously used to use mic
positioning but I've even given up on that too."
- Jezar on http://www.audiomelody.com
Jörn Nettingsmeier
Kurfürstenstr 49, 45138 Essen, Germany
http://spunk.dnsalias.org (my server)
http://www.linuxaudiodev.org (Linux Audio Developers)
imnsho, all this sampling rate hype is a clever
industry ploy to keep
people from thinking that their hardware is finally good enough and they
can stop buying now.
Probably true but...
people can't hear over 20k. period. 48k
sampling rate gives you 24k
minus what's cropped off by the aliasing filter. granted, higher
sampling allows you to use a simpler, less steep aliasing filter, and
some people claim to perceive an improvement from that. but even then,
96k should be enough.
This is not quite correct. While humans can not hear continuous tones over
20-odd kHz that is not the whole story, the frequency reproduction also
limits the minimum transient rise time, which is detected by a different
part of the ear (IIRC, IANABiologist and my psychoacoustics textbook is at
work).
7:31 a.m.
On Mon, Jan 26, 2004 at 11:54:26 +0100, Joern Nettingsmeier wrote:
Which membrane? The basilar membrane is responsible for frequency alaysis
(AFAIR), but its limited by its geometry. Doesn't it detect resonances in
the narrowing tube? Transients could be detected before the pressure wave
reaches the basilar membrane, though I'm just guessing here.
This is not
quite correct. While humans can not hear continuous tones over
20-odd kHz that is not the whole story, the frequency reproduction also
limits the minimum transient rise time, which is detected by a different
part of the ear (IIRC, IANABiologist and my psychoacoustics textbook is at
work).
interesting point. but isn't membrane inertia the limiting factor in
transient reproduction anyway? my naive understanding is that min rise time = nyquist
freq. if we
cannot perceive frequencies above, say, 20k, all we win by faster
sampling is more accurate timing information.
but there is a worst-case "timing error" of 1/24000 sec, which does not
seem much to me...
Its not the timing error - it wont be quatised to the nearest sample
AFAIK, but you will get substantial ripple from the reconstruction filters
fi you try to reproduce transients near nyquist- dunno how that is perceived.
i'd like to read more, but i haven't been able
to google anything up
about transient perception and reproduction. any pointers?
There seems to be some relevent stuff here:
http://arjournals.annualreviews.org/doi/full/10.1146/annurev.neuro.23.1.285
but its a bit over my head.
for the record, here's a 1998 aes paper that
elaborates on the
consequences of higher sampling on filter artifaces in the audible
range: http://www.nanophon.com/audio/antialia.pdf
Thanks, thats interesting.
- Steve
9:45 a.m.
On Mon, 26 Jan 2004, Steve Harris wrote:
On Mon, Jan 26, 2004 at 11:54:26 +0100, Joern
Nettingsmeier wrote:
Which membrane?
sorry for not being clear, i was thinking of the speaker membrane of the
playback system. just guessing, though.
This is
not quite correct. While humans can not hear continuous tones over
20-odd kHz that is not the whole story, the frequency reproduction also
limits the minimum transient rise time, which is detected by a different
part of the ear (IIRC, IANABiologist and my psychoacoustics textbook is at
work).
interesting point. but isn't membrane inertia the limiting factor in
transient reproduction anyway? The basilar membrane is responsible for frequency
alaysis
(AFAIR), but its limited by its geometry. Doesn't it detect resonances in
the narrowing tube? Transients could be detected before the pressure wave
reaches the basilar membrane, though I'm just guessing here.
drat. it's "pay per view". :(
but i guess i wouldn't understand anything anyway.
my naive understanding is that min rise time =
nyquist freq. if we
cannot perceive frequencies above, say, 20k, all we win by faster
sampling is more accurate timing information.
but there is a worst-case "timing error" of 1/24000 sec, which does not
seem much to me...
Its not the timing error - it wont be quatised to the nearest sample
AFAIK, but you will get substantial ripple from the reconstruction filters
fi you try to reproduce transients near nyquist- dunno how that is perceived.
i'd like to read more, but i haven't been
able to google anything up
about transient perception and reproduction. any pointers?
There seems to be some relevent stuff here:
http://arjournals.annualreviews.org/doi/full/10.1146/annurev.neuro.23.1.285
but its a bit over my head.
11:46 a.m.
On Mon, Jan 26, 2004 at 02:41:34 +0100, Joern Nettingsmeier wrote:
sorry for not being clear, i was thinking of the speaker membrane of the
playback system. just guessing, though.
Ahh, I see, well not if it can reproduce 20kHz sinewaves.
drat. it's "pay per view". :(
but i guess i wouldn't understand anything anyway.
Ah, right, we have a sitelicence by the look of it. It was pretty
inpenetrable anyway.
- Steve
interesting point. but isn't membrane inertia the limiting factor in
transient reproduction anyway?
Which membrane? i'd like to read more, but i haven't been able
to google anything up
about transient perception and reproduction. any pointers?
There seems to be some relevent stuff here:
http://arjournals.annualreviews.org/doi/full/10.1146/annurev.neuro.23.1.285
but its a bit over my head. 
12:57 p.m.
On Mon, Jan 26, 2004 at 09:40:03AM +0000, Steve Harris wrote:
This is not quite correct. While humans can not hear
continuous tones over
20-odd kHz that is not the whole story, the frequency reproduction also
limits the minimum transient rise time, which is detected by a different
part of the ear (IIRC, IANABiologist and my psychoacoustics textbook is at
work).
Which psychoacoustics textbook is it that you have at work?
-Eric Rz.
6:47 a.m.
On Mon, 26 Jan 2004 10:19:59 +0100
Joern Nettingsmeier <nettings(a)folkwang-hochschule.de> wrote:
people can't hear over 20k. period. 48k sampling
rate gives you 24k
minus what's cropped off by the aliasing filter.
Thats not the full story. There was a Japanese study where untrained
listeners could tell the difference between 10kHz sine waves and 10kHz
square waves in double blind tests. That means that the listeners
could tell presence or absence of the 30kHz third harmonic of the
square wave.
Theroretically this should not have been possible which indicates that
theories of hearing perception are far from complete.
Erik
--
+-----------------------------------------------------------+
Erik de Castro Lopo nospam(a)mega-nerd.com (Yes it's valid)
+-----------------------------------------------------------+
"MS apparently now has a team dedicated to tracking problems with Linux
and publicizing them. I guess eventually they'll figure out this back
fires..." -- William Burrow
7:39 a.m.
Erik de Castro Lopo wrote:
On Mon, 26 Jan 2004 10:19:59 +0100
Joern Nettingsmeier <nettings(a)folkwang-hochschule.de> wrote:
or there *may* have been aliasing or reproduction artifacts which
provided clues to the listeners within the 20-20k range.
<babble type="not strictly scientific">
and even if they could tell a difference, this does not imply one was
better or more hifi than the other - most people can hear the difference
of a 192k mp3 and a 44k cd track, but i've yet to meet somebody who
could always determine which is which, or even which of the two is better.
on my monitors, i can hear a 19k sine wave at 6db over my normal
listening level. but: i'm 28 yrs old, and i've been doing about 20 gigs
and/or mixing jobs a year at higher levels that i'm comfortable with for
about 10 years. i really doubt my ears are still that good, and my guess
is it's some artifact from the tweeters.
</babble>
ok, now i shut up :)
in any case, if you have papers on studies about this, send them over.
it seems there is still much to learn.
--
"I never use EQ, never, never, never. I previously used to use mic
positioning but I've even given up on that too."
- Jezar on http://www.audiomelody.com
Jörn Nettingsmeier
Kurfürstenstr 49, 45138 Essen, Germany
http://spunk.dnsalias.org (my server)
http://www.linuxaudiodev.org (Linux Audio Developers)
people can't hear over 20k. period. 48k
sampling rate gives you 24k
minus what's cropped off by the aliasing filter.
Thats not the full story. There was a Japanese study where untrained
listeners could tell the difference between 10kHz sine waves and 10kHz
square waves in double blind tests. That means that the listeners
could tell presence or absence of the 30kHz third harmonic of the
square wave.
Theroretically this should not have been possible which indicates that
theories of hearing perception are far from complete.
3:01 p.m.
On Mon, 26 Jan 2004 12:35:22 +0100
Joern Nettingsmeier <nettings(a)folkwang-hochschule.de> wrote:
in any case, if you have papers on studies about this,
send them over.
I think this is the one that I saw referenceed in an article somewhere
else:
Tsutomi Oohashi, Emi Nishina, Norie Kawai, Yoshitaka Fuwamoto, Hiroshi Imai,
High-Frequency Sound Above the Audible Range Affects Brain Electric Activity
and Sound Perception.
Audio Engineering Society preprint No. 3207 (91st convention, New York City).
Erik
--
+-----------------------------------------------------------+
Erik de Castro Lopo nospam(a)mega-nerd.com (Yes it's valid)
+-----------------------------------------------------------+
"The X-files is too optimistic. The truth is not out there."
-- Anthony Ord
9:58 a.m.
AFAIK the only reason for going to 96KHz is to allow for digital effects
processing to do a better job. Maybe some people can hear a difference
on *very* expensive equipment but eventually you have to go to 16 bit
44.1KHz ;-)
Jan
On Mon, 2004-01-26 at 03:19, Joern Nettingsmeier wrote:
Florin Andrei wrote:
Anyone here using 192kHz for sampling?
are you trying to record for dogs or bats ? ;)
seriously, the only measurably effect of 192k is that it fills up your
disks faster and eats more cpu when processing.
imnsho, all this sampling rate hype is a clever industry ploy to keep
people from thinking that their hardware is finally good enough and they
can stop buying now.
people can't hear over 20k. period. 48k sampling rate gives you 24k
minus what's cropped off by the aliasing filter. granted, higher
sampling allows you to use a simpler, less steep aliasing filter, and
some people claim to perceive an improvement from that. but even then,
96k should be enough.
--
"I never use EQ, never, never, never. I previously used to use mic
positioning but I've even given up on that too."
- Jezar on http://www.audiomelody.com
Jörn Nettingsmeier
Kurfürstenstr 49, 45138 Essen, Germany
http://spunk.dnsalias.org (my server)
http://www.linuxaudiodev.org (Linux Audio Developers)
10:06 a.m.
"Jan Depner" <eviltwin69(a)cableone.net> wrote:
AFAIK the only reason for going to 96KHz is to allow
for digital effects
processing to do a better job. Maybe some people can hear a difference
on *very* expensive equipment but eventually you have to go to 16 bit
44.1KHz ;-)
Not if you are burning your tracks to DVD-R (or DVD+R) at 24/96.
Len Moskowitz
Core Sound
www.core-sound.com
10:18 a.m.
On Mon, 2004-01-26 at 08:04, Len Moskowitz wrote:
"Jan Depner" <eviltwin69(a)cableone.net>
wrote:
Yeah, those work real well in my car ;-)
Jan
AFAIK the only reason for going to 96KHz is to
allow for digital effects
processing to do a better job. Maybe some people can hear a difference
on *very* expensive equipment but eventually you have to go to 16 bit
44.1KHz ;-)
Not if you are burning your tracks to DVD-R (or DVD+R) at 24/96.
Len Moskowitz
Core Sound
www.core-sound.com
10:24 a.m.
On Mon, Jan 26, 2004 at 07:56:05AM -0600, Jan Depner wrote:
AFAIK the only reason for going to 96KHz is to allow
for digital effects
processing to do a better job. Maybe some people can hear a difference
on *very* expensive equipment but eventually you have to go to 16 bit
44.1KHz ;-)
For conventional CD's yes, but aren't there plans for some sort
of
consumer audio DVD at a higher sampling rate and word length?
(there's always somebody out there who'll want it at any price "because
it's better")
--
Anahata
anahata(a)treewind.co.uk Tel: 01638 720444
http://www.treewind.co.uk Mob: 07976 263827
10:38 a.m.
AFAIK the only reason for going to 96KHz is to allow
for digital effects
processing to do a better job. Maybe some people can hear a difference
on *very* expensive equipment but eventually you have to go to 16 bit
44.1KHz ;-)
Jan
Nahh....who cares about them. We just care about ourselves sitting in our
192KHz studios, right Jan? Nirvana at the console? ;-)
More seriously though, doesn't having four 24-bit samples at 192KHz give you
the opportunity to do something mathematically when going to one 16-bit
sample at 44.1/48HK? I would think that it's never bad to have more
information. It all comes down to what you do with it when you have it on
disk.
- Mark
7553
days inactive
7554
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linux-audio-user@lists.linuxaudio.org
15 comments
9 participants
participants (9)
-
Anahata -
Eric Dantan Rzewnicki -
Erik de Castro Lopo -
Florin Andrei -
Jan Depner -
Joern Nettingsmeier -
Len Moskowitz -
Mark Knecht -
Steve Harris