Hola Jochen and all,
yes, of course it is a trade-off between xruns and delay, but i do
that adaptively as well - start with a quite low framing, measure the
drop-out rate and reopen the soundcard in case of too much drop-outs.
this only impacts the quality of the start-up phase and
stan wrote:
Florian Faber wrote:
You want hardware monitoring - there are sound cards that support
hardware mixing. With good converters you have latencies down to 5
samples at 192kHz, that would be 0.026ms for each way, 0.052ms over
all.
I'm not the original poster, but I'm curious
Alexander Carôt wrote:
3.) Rather than using a double buffer for the playout wouldn't it be
possible to choose only one physical playout buffer and parse the
captured data in right at the interrupt.
It's unlikely that any code could be fast enough to write the entire
buffer before the hardware
Did you try the settings in /etc/security/limits.conf suggested on the
Frinika
front page? (http://frinika.sourceforge.net). I noticed quite some
difference in delay for
Terratec Aureon 5.1 Fun cards using JavaSound.
Helge F.
On Wed, Jun 11, 2008 at 10:11 AM, Clemens Ladisch [EMAIL PROTECTED]
]
An: Clemens Ladisch [EMAIL PROTECTED]
CC: alsa-user@lists.sourceforge.net, Alexander Carôt [EMAIL PROTECTED]
Betreff: Re: [Alsa-user] Output latency
Did you try the settings in /etc/security/limits.conf suggested on the
Frinika
front page? (http://frinika.sourceforge.net). I noticed quite some
Hi Jochen,
2. use a lower frame size, than my codec/systems framing. (e.g. 128
instead of 256, but still transmit 256 in one pass)
Yes - a good idea, however, sometimes depending on the actual machine and OS
(or even low-latency patches) problems might occur when running below 256
If I understand your question correctly, it is because they use two
different buffers. If you aren't trying to play the capture buffer, it
would wreak havoc to try to use it for playback while capture is going
on. So there is a buffer for capture and a buffer for playback. And
each has
Alex,
The idea is the following :
1.) Of course there has to be an input double buffer which generates
the desired block of samples.
You want hardware monitoring - there are sound cards that support
hardware mixing. With good converters you have latencies down to 5
samples at 192kHz, that
Hej,
thanks for getting back to me.
What do you want to do? Realtime monitoring/mixing?
As subject of my PhD research I am working on realtime network music
performances. In other words : I am bassplayer and I play livemusic via the
Internet with people in different locations.
Florian Faber wrote:
Alex,
The idea is the following :
1.) Of course there has to be an input double buffer which generates
the desired block of samples.
You want hardware monitoring - there are sound cards that support
hardware mixing. With good converters you have latencies
I think you would have to customize the driver to do this. In the
special case that you are playing back the recorded input, you write the
input buffer directly to the output buffer every time the input buffer
interrupts because it is time to empty it. Adds an extra branch in the
driver
Alexander Carôt wrote:
If I understand your question correctly, it is because they use two
different buffers. If you aren't trying to play the capture buffer, it
would wreak havoc to try to use it for playback while capture is going
on. So there is a buffer for capture and a buffer for
Hi all,
can anyone give me an explanation why the blocking delay of a soundcard appears
twice using the ALSA driver ? E.g. with 48 kHz at 128 samles / frame I
understand that the capturing process requires 2,6 ms to actually fill one
block of audio samples (= blocking delay of 2,6 ms).
On Sun, Jun 8, 2008 at 5:12 AM, Alexander Carôt
[EMAIL PROTECTED] wrote:
Hi all,
can anyone give me an explanation why the blocking delay of a soundcard
appears twice using the ALSA driver ? E.g. with 48 kHz at 128 samles / frame
I understand that the capturing process requires 2,6 ms to
Alexander Carôt wrote:
Hi all,
can anyone give me an explanation why the blocking delay of a soundcard
appears twice using the ALSA driver ? E.g. with 48 kHz at 128 samles / frame
I understand that the capturing process requires 2,6 ms to actually fill one
block of audio samples (=
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