Hi Dominique,
thanks for your answer!
Ye, have already thought that as well. Just thought its maybe not appropriate to ask there since Im not involved in the ALSA project. But will give it a try.
The JACK API was already in my mind, too. Just dont know if its rather for small (synchron but differnet after a new start of the application) delays between streams or it can handle that its always the same delay between streams, even after a new start, as well. Maybe Ill give it a try, too.
Thanks again and cheers!
Gesendet:Donnerstag, 26. Juni 2014 um 12:48 Uhr
Von:Dominique Michel dominique.mic...@vtxnet.ch
An:alsa-user@lists.sourceforge.net
Betreff:Re: [Alsa-user] Constant delay between starting a playback and a capture stream
Le Mon, 23 Jun 2014 18:20:43 +0200,
Max Schmidt schmidti...@web.de a crit :
Hi,
I am not an audio developer but I think the LAD list is a better place
for such highly technical issues. I also think most developers that
want a constant audio latency with their application will use the
JACK API instead of the ALSA API.
http://lists.linuxaudio.org/listinfo/linux-audio-dev
Hey all,
first a big hello to everyone since Im new to this mailing-list.
Ive had a look regarding this issue here and in generell in the
internet but didnt find anything relating. So, sorry if I overlooked
something. Ive got a BeagleBone Black (ARM Cortex A 8 P) with an
Audio-Cape (using McAsp and ALSA-Davinci-Drivers) and wrote an mmap
based playback-capture application (both devices hw:0,0). The
important thing is that the application needs a constant delay (not
necessarily small) between the start of the playback and the capture
stream. So when looping back the signal to the microfone the first
played sample e.g. always (even after a restart of the BeagleBone or
the app) has a delay in the capture stream of exactly 80 samples
(when e.g. played with 48 kHz), and once measured can be seen as
constant. To realize that I use the ALSA API function
snd_pcm_link(c_handle, p_handle). When starting the playback stream
(and therefore linked capture stream as welll) manually its buffer
already is filled. There is no buffer underrun/overrun recovery. It
already works quite fine but I still have some questions and it would
be really nice if someone can help: 1) Looking at a plot of multiple
measurements of looped back and captured square waves (or sines)
there still is a jitter of about two to four samples (at 48 kHz;
which is about 40 to 80 ms), no matter if I run it as RT-app (energy
save modes etc. disabled, just important Processes have a higher
Priority e.g. EDMA) or normal app. Please correct me if Im wrong,
but as far as I understand when the streams are linked, at start the
processor is going through a linked list triggering all linked
streams. And the trigger start is an atomic process so shouldnt get
interrupted. Shouldnt it always take the same time between starting
the playback and capture stream then? And if yes, where could the
variable start delay come from? 2) Looking at the time stamp of both
streams it tells that the difference between the start trigger
normally is between 2 to 7 ms. What, I think, does not really fit to
the observation written above since then there normally should be no
big sample jitter at 48 kHz. Are the time stamp values not precise
enough (actually how precise are the time stamps, just that they can
show microseconds does not imply that the resolution really is
microseconds, or am I wrong?) or are they correct and the difference
in latency seen in measured plots comes from somewhere else? Delay
due to Hardware should be constant so the issue must have something
to do with ALSA or Linux. Would be nice if someone can help or has an
idea! Tipps for an improvement are wlecome as well! Many thanks and
cheers!
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