Benno == Benno Senoner [EMAIL PROTECTED] writes:
Benno Not sure if it's enough for you, but since most pro-audio
Benno equipement has some intrinsic latency in it, then I think it's
Benno not worth the trouble to try to get single sample latency. In
Benno theory it's possible
Benno Senoner wrote:
On Wednesday 11 July 2001 19:11, Paul Davis wrote:
So what do you suggest Paul ?
Is my problem normal ?
(I can't even run alsactl store, while apps in OSS emulation mode work
perfectly)
if you have one soundcard, then ALSA should work out-of-the-box, so to
On Thursday 12 July 2001 08:04, Abramo Bagnara wrote:
still no luck.
You have not installed alsa.conf. There was a bug I fixed some time ago
pointed to me by Paul. I don't remember if 0.9b5 contains the fix or
not.
Please use current CVS.
Ok I will do so, but will I be able to run
Paul Davis wrote:
As someone responsible for distributing a sample .asoundrc file with
ardour, I must remind people that the boilerplate text quoted below
should NOT show up in a regular .asoundrc. The sample that came with
ardour is from a couple-of-ALSA-generations ago.
--p
aah. sorry
Benno Senoner wrote:
On Thursday 12 July 2001 08:04, Abramo Bagnara wrote:
still no luck.
You have not installed alsa.conf. There was a bug I fixed some time ago
pointed to me by Paul. I don't remember if 0.9b5 contains the fix or
not.
Please use current CVS.
Ok I will do
I've read that the fastest x86 intel architecture can do interrupts is w/
max latencies of 40us using rtlinux. You might be able to do at least the
interrupt part with a PPC-based system (1us interrupts).
not for 96kHz. you've only got 10usec per sample. sorry.
--p
But N is not fixed! The host is free to call the plugin's run() or
runAdding() function with any non-zero argument. It might call it
like:
run(16);
run(21467);
run(1);
run(480);
run(16384);
May i just ask, out of curiousity, why the host would want to do
Paul Davis [EMAIL PROTECTED] writes:
You're right. But don't you see that this means that the audio sample rate is
different from the control sample rate? For every process cycle, there will
be one new control sample and N new audio samples (N is the fragment length),
implying that the
On Thu, Jul 12, 2001 at 01:08:14AM +0300, n++k wrote:
[Greg Berchin [EMAIL PROTECTED]]
| Steve Harris wrote:
|
| I don't understand the hardware issues, but as the filter coefficent
| appraoches 0.0f the number of cycles taken to multiply it goes up
|
| This confuses me. In reading
Dave Phillips wrote:
Jörn Nettingsmeier wrote:
while talking about things-to-do, the linux audio user list is
active but hasn't been announced yet. will do that after the show.
dave p., if you read this, any ideas ?
Hi, Jörn: I'm back from LSM, swamped with work, but I really want
On 2001-07-12 00:39 -0500, Christopher Lee wrote:
I've read that the fastest x86 intel architecture can do interrupts is w/
max latencies of 40us using rtlinux. You might be able to do at least the
interrupt part with a PPC-based system (1us interrupts).
Did you write 40 microseconds ?
In reading the documentation for the FPU on the Athlon,
they state that it can perform two pipelined double precision floating
point multiplies per cycle.
Are we talking about SIMD instructions here? I've never used them, but
with generic i686 as produced by gcc -O6:
My mistake. It's one
OK,
I'm looking at how to write LAGGA clients, the command line one is pretty
straightforward, but what if I want to use GTK (say)?
Is the correct thing to multi thread inside main, and one thread does
LAAGA stuff, and the other calls gtk_main()?
- Steve
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