[gst-devel] Dropped frames at gstbaseaudiosrc

[Sebastian Voecking]

Hello!

Thanks for your quick reply.

> What to do about it (possibly);
> allow for "looser/more" threading and/or buffering, e.g.
> - use queue in one or both of the streams going into the muxer
> (e.g. before the encoder element)

> - simlarly, alsasrc (and osssrc) properties buffer-time and latency-time
> determine the total buffer size and individual buffer size

I tried that and now I am here:

gst-launch-0.10 v4l2src device="/dev/video0" !
video/x-raw-yuv,width=640,height=480 ! queue ! theoraenc ! oggmux
name="ogg" alsasrc device="hw:1,0" buffer-time=40000000
latency-time=400000 ! audio/x-raw-int,rate=32000 ! queue !
audioconvert ! vorbisenc ! ogg. ogg. ! filesink location=test.ogg

No more frame get dropped now. But the resulting video and audio are
both very choppy.

What else should I try? BTW I am using Ubuntu Feisty Fawn here with
gstreamer 0.10.12.

Regards,
Sebastian