On Thu, Jun 13 2002 at 11:55:11am +0200, Men Muheim wrote:
http://magic.gibson.com/
Looks interesting. Thanks for the info.
[...]
The discussion here should not be about the network but about an audio
API which is independent of the underlying network stack.
[...]
I think the first thing
Maybe the word latency did confuse some people here. Latency is
sometimes used for the OS scheduling time, or for a network packet
delay, or for the audio buffer-size.
In my case I wanted to name the delay from the analogue input signal to
the analogue output signal.
Nonsense! What about
BTW, as I keep moaning, I think network audio is an important next
step
in
LAD development and ideally could be combined with the kind of step
required
to get JACK firmly off the ground. I'm still plugging LADMEA ideas
(www.ladspa.org/ladmea/).
Thank you for getting back to the original
John Lazzaro wrote:
You certainly can't play an instrument with 10ms latency.
See:
http://ccrma-www.stanford.edu/groups/soundwire/delay_p.html
http://www-ccrma.stanford.edu/groups/soundwire/delay.html
http://www-ccrma.stanford.edu/groups/soundwire/WAN_aug17.html
if i read this
Men Muheim wrote:
Thats true, but doesn't match my experience of trying to play with 512
sample buffers... although that probably doesn't equate to 512 samples
of
latency, its probably more.
It is more indeed!
An ALSA audio interface normally uses two double-buffers. One for the
if i read this correctly, it's about latency wrt _another_player_. all
trained ensemble musicians are easily able to compensate for the rather
long delays that occur on normal stages. not *hearing_oneself_in_time*
is a completely different thing. if i try to groove on a softsynth, 10
ms
Joern Nettingsmeier wrote:
it's not hard to hear. at more than 5-8 msec, you feel that beats become
fuzzy, and below that, you can judge by sound coloration (comb filter).
granted, it's a slightly different situation, since you have both a
delayed and an undelayed signal combined, but it
On Fri, Jun 14, 2002 at 08:30:45 -0400, Charles Baker wrote:
what this and the earlier gedankenexperiment about the jazz drummer on an
extremely upbeat tune implies ( and I tend to think it is true) is that
humans can judge delays between direct and indirect sound clearly into the
realm of
Joern Nettingsmeier writes
not *hearing_oneself_in_time* is a completely different thing.
Yes, I agree its a completely different thing, but ...
if i try to groove on a softsynth, 10
ms response time feels ugly on the verge of unusable (provided my
calculations and measurements on latency
-Original Message-
From: Charles Baker [mailto:[EMAIL PROTECTED]]
...
1) Humans can and do play instruments with much worse response than
the 10ms we want as minimum in out linux apps.
...
most (all) naturally introduced latencies are fairly constant in time, one
can get 'feel; for
Charles Baker wrote:
There is a whole WORLD of instrument control issues that are just
beginning to be addressed, and to focus on latency is only to ignore what
I believe to be much more important issues of control.
I just find it damn difficult to play a piano two or three notes ahead of
Going back to the issue of latency, it should be pointed out that while
it might not be a big deal if your softsynth takes 25 ms to trigger,
It is unless you only use it with a sequencer.
latency on the PCI bus is a big problem. If you can't get data from
your HD (or RAM)
From memory I
On Friday 14 June 2002 01:58 pm, Richard C. Burnett wrote:
Which is enough to reconstruct the sine wave if the output is through a
proper post-DAC filter.
amplitude modulation from the different sample points. Have you ever
plotted a sine wave where you you don't pick enough points to
-Original Message-
From: Lamar Owen [mailto:[EMAIL PROTECTED]]
...
The idea is that you get the integrated value of the
amplitude of the sine
wave, since a sine wave always has the same shape. But the
amplitude, at the
Nyquist frequency, cannot change. Yes, I really said
PROTECTED]
[mailto:[EMAIL PROTECTED]]On Behalf Of STEFFL,
ERIK *Internet* (SBCSI)
Sent: Friday, June 14, 2002 12:01 PM
To: '[EMAIL PROTECTED]'
Subject: RE: [linux-audio-dev] RFC: API for audio across network -
inter-host audio routing
-Original Message-
From: Lamar Owen [mailto:[EMAIL
-Original Message-
From: Bob Colwell [mailto:[EMAIL PROTECTED]]
Sent: Friday, June 14, 2002 4:39 PM
To: [EMAIL PROTECTED]
Subject: RE: [linux-audio-dev] RFC: API for audio across network -
inter-host audio routing
Nyquist says that if you sample a repeating waveform, ANY
Peter Hanappe wrote:
I wondered if it would be possible to write a JACK driver (i.e.
replacement for current ALSA driver) that would stream the audio over
a network. The driver is a shared object, so it's technically possible.
I was thinking of the timing issues.
Concerning the timing issues,
Of Steve Harris
Sent: Mittwoch, 12. Juni 2002 20:03
To: [EMAIL PROTECTED]
Subject: Re: [linux-audio-dev] RFC: API for audio across network -
inter-
host audio routing
On Wed, Jun 12, 2002 at 06:25:14 +0200, Men Muheim wrote:
Has anyone ever thought of implementing a library for transfer
Concerning the timing issues, one of the problem raised by audio
transmission is the audio cards clock skew of the different stations
involved in the transmission.
I've done some work on this topic. It's available as a technical
report at
ftp://ftp.grame.fr/pub/Documents/AudioClockSkew.pdf
Indeed I have and it is, in fact, what I plan to be spending most of
this summer working on. I don't know if you've seen gison's magic,
but
it sounds very similar what you're doing:
http://magic.gibson.com/
Looks interesting. Thanks for the info.
Magic seems to be a network like
On Thu, Jun 13, 2002 at 11:45:13 +0200, Men Muheim wrote:
Thanx for the info. Unfortunately it does not really get clear to me
what the project does. I think the difference to my approach is that I
am talking about LAN and low latency (10ms)
MAS works over LANs, and should be capable of 10ms
On Thursday 13 June 2002 09:29 am, Charlieb wrote:
On Thu, 13 Jun 2002, Steve Harris wrote:
MAS works over LANs, and should be capable of 10ms latency, which isn't
very low by BTW. You certainly can't play an instrument with 10ms
latency.
Nonsense! What about tubas ?
I admit a guitar
You certainly can't play an instrument with 10ms
latency.
in 10ms sound travels somewhat more than 3 meters.
that why i use nearfield monitors :)
--martijn
On Thu, Jun 13, 2002 at 01:29:11 +, Charlieb wrote:
MAS works over LANs, and should be capable of 10ms latency, which isn't
very low by BTW. You certainly can't play an instrument with 10ms
latency.
Nonsense! What about tubas ?
I admit a guitar would feel pretty awful w/ 10 ms
On Thu, Jun 13, 2002 at 09:43:44 -0400, Lamar Owen wrote:
[latency and instruments]
there is a definite, barely detectable delay between initiation of the note
and the perception of the note (which is both by ear and by hand in the case
of the horn). The long bore is one reason the horn's
You certainly can't play an instrument with 10ms
latency.
Really? You might want to check my math on this, but if the speed of sound
in air at 75 degrees F is about 1135 feet/second, then it takes about
0.00088 seconds, or 0.88 ms, for the sound to travel 1 foot. So in 10 ms
the
You certainly can't play an instrument with 10ms latency.
See:
http://ccrma-www.stanford.edu/groups/soundwire/delay_p.html
http://www-ccrma.stanford.edu/groups/soundwire/delay.html
http://www-ccrma.stanford.edu/groups/soundwire/WAN_aug17.html
These experiments show the limits of musical
On Thursday 13 June 2002 11:59 am, dgm4 wrote:
Lamar Owen wrote:
French Horn. Its bore is as long as a tuba's. I have played horn before,
and there is a definite, barely detectable delay between initiation of
the note and the perception of the note (which is both by ear and by hand
in
On Thursday 13 June 2002 13:15, xk wrote:
I'm not a professional musician, but a 25 ms latency makes me more than
happy.
It really depends upon the specific application. For some problem-spaces
(radio automation, for example), latency is just not a very important issue.
For others (e.g.
In my experience, audible separation of acoustic events normally happens
around 20ms (ignoring phase effects). Most instruments (including guitar)
are entirely playable with this sort of delay.
The pipe organ example is a good one - there is a huge variety of delay on
pipe organs, probably
How about the 1.0-1.5 ms latencies that everbody tries to obtain (or
already
has) in both Linux/Win world? That always made me wonder if this isn't
just
hype like the 192 kHz issue.
I'm not a professional musician, but a 25 ms latency makes me more than
happy.
I would say that for playing
On Thu, 2002-06-13 at 01:39, Dan Hollis wrote:
We hold a patent on MaGIC
Curious.
--
Bob Ham: [EMAIL PROTECTED] http://pkl.net/~node/
My music: http://mp3.com/obelisk_uk
GNU Hurd: http://hurd.gnu.org/
The pipe organ example is a good one - there is a huge variety of delay on
pipe organs, probably beyond the half second (I don't have the figures, but
there's often a significant delay between keypress and note as well as the
acoustic delay). I'm fine with small delays, but fast passages
On Thu, Jun 13, 2002 at 01:29:11PM +, Charlieb wrote:
*
Charlie Baker [EMAIL PROTECTED]
when everything isn't roses, you don't get
any headroom - Thomas Dolby New Toy
*
On
On Thu, Jun 13, 2002 at 12:41:52 -0400, Paul Winkler wrote:
Nonsense! What about tubas ?
I admit a guitar would feel pretty awful w/ 10 ms latency,
And yet electric guitarists do it all the time. Ever stood 10 feet away
from your amp? It's no big deal, you get used to it.
Thats true,
This prompted me to look at my course notes, and here's a quote:
A patent may be granted for an invention only if
following conditions are satisfied:
The invention is new;
It involves an inventive step;
It is capable of industrial application;
The grant for a patent for it is not excluded
Has anyone ever thought of implementing a library for transfer of audio
across networks? The API could be similar to JACK but would allow
inter-host communication. This library would simplify the routing of
audio across networks by solving synchronization issues etc. Therefore
this RFC is closely
On Wed, Jun 12, 2002 at 06:25:14 +0200, Men Muheim wrote:
Has anyone ever thought of implementing a library for transfer of audio
across networks? The API could be similar to JACK but would allow
Have a look at MAS, they had an impressive demo at LinuxTag:
http://mediaapplicationserver.net/
On Wed, Jun 12, 2002 at 07:45:51 +0200, Peter Hanappe wrote:
I wondered if it would be possible to write a JACK driver (i.e.
replacement for current ALSA driver) that would stream the audio over
a network. The driver is a shared object, so it's technically possible.
I was thinking of the
On Wed, 2002-06-12 at 17:25, Men Muheim wrote:
Has anyone ever thought of implementing a library for transfer of audio
across networks?
Indeed I have and it is, in fact, what I plan to be spending most of
this summer working on. I don't know if you've seen gison's magic, but
it sounds very
On Wed, 2002-06-12 at 21:18, Dan Hollis wrote:
I talked to gibson directly about magic. They stated it's patented and
they won't give permission for open source implementation.
I wonder what they have patents on, and whether they're applicable to
the UK.
--
Bob Ham: [EMAIL PROTECTED]
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