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 firs
> -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
EMAIL 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: Lama
> -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 sai
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 t
> 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 memo
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
> -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
> 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 la
On Friday 14 June 2002 11:35 am, Richard C. Burnett wrote:
> a 20khz audio signal. At 40Khz sampling that would give you 2 points on
> the wave to reconstruct it.
Which is enough to reconstruct the sine wave if the output is through a proper
post-DAC filter.
> What if you're sampling interval
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 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
> 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
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
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 thi
> 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 orig
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
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 exclud
No, but this is the reason why most organ consoles have a mirror allowing
the organist to watch the conductor and why in many cathedrals there is a
separately located set of pipes for use with choral music (generally
controlled by an upper manual, labelled "choir"). As long as the organist
follows
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 tr
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"
> *
>
> 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, 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/
> 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 pla
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 beyond
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. l
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
> 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 laten
Not really hype, they are slightly different issues.
(I work with Barry and we discussed this issue earlier)
Achieving system latencies down to 1 - 2 ms is good as it's an indication of
the overall latency of the system. Typically a system can spike at a lot
higher than the values you mention
> There's a psychoacoustic phenomenon known as the Haas effect which states
> that a direct sound and it reflections (echos) are percieved as a single
> sound by the brain, where the time difference between the two is less than
> about 30ms. So if the brain can't distinguish between sounds at this
> > >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 1
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
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
> 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 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 g
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 10m
> 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" r
> 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.p
CTED]] On Behalf 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 thou
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 iss
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] htt
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 s
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 t
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/
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
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