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
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
samples
en" <[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 sugges
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]>
w
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 hardwa
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 c
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
> 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
> dr
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
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.
http://www
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, t
> 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
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 (= b
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
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