[gst-devel] Dropped frames at gstbaseaudiosrc

[Sebastian Vöcking]

Hello!

I have a small problem. I am trying to record from my tv card with
v4l2src. It works well unless I don't add sound to the pipeline. As soon
as I record sound as well I get warnings about dropped frames and my
videos end up shorter than they should be. If I construct my pipeline
with gst-launch I can reproduce the problem:

gst-launch-0.10 v4l2src ! theoraenc ! oggmux name="ogg" alsasrc !
audio/x-raw-int,rate=32000 ! audioconvert ! vorbisenc ! ogg. ogg. !
filesink location=test.ogg

The output is:

Setting pipeline to PAUSED ...
Pipeline is live and does not need PREROLL ...
Setting pipeline to PLAYING ...
New clock: GstAudioSrcClock
WARNING: from element /pipeline0/alsasrc0: Internal GStreamer error:
clock problem.  Please file a bug at
http://bugzilla.gnome.org/enter_bug.cgi?product=GStreamer.
Additional debug info:
gstbaseaudiosrc.c(583): gst_base_audio_src_create
(): /pipeline0/alsasrc0:
dropped 14322 samples

[...] Many more dropped samples

Caught interrupt -- handling interrupt.
Interrupt: Setting pipeline to PAUSED ...
Execution ended after 3903277000 ns.
Setting pipeline to PAUSED ...
Setting pipeline to READY ...
Setting pipeline to NULL ...
FREEING pipeline ...

Using osssrc or using a different audio device leads to the same
problem. But I exchange v4l2src with videotestsrc or alsasrc with
audiotestsrc everything is fine.

Where is my mistake?

Thanks,
Sebastian