alsasrc: different behaviour and clocking between 1.2 and 1.1.4

Robert Krakora rob.krakora at messagenetsystems.com
Fri Oct 11 16:05:02 CEST 2013


Hi Adrian,

What is the result if you run audio only with 1.2.0?  What is the result if
you run video only with 1.2.0?

Best Regards,

Rob Krakora



On Fri, Oct 11, 2013 at 9:52 AM, Adrian Pardini <listas at tangopardo.com.ar>wrote:

> Hello all,
>
> I'm working on a python (with gi) application where most of the time
> my pipeline is something like:
>
> audio bin: [ alsasrc ! queue2 ! caps ! volume ! tee ] ! queue2 !
> (adder/liveadder , volume and sink)
> video bin: [ v4l2src ! queue2 ! caps ! tee ] ! queue2 ! (videomixer and
> sink)
>
> At some point I create a bin to hold a muxer (avimux or matroskamux)
> linked to a filesink, then I link the mux to the tees with a queue2
> between and after that call sync_state_with_parent() on the bin.
>
> With 1.1.4 it works flawlesly, however with 1.2.0 the pipeline freezes
> for a while and sometimes resumes playing but with a lot of video
> stuttering and no audio.
>
> Thinking about https://bugzilla.gnome.org/show_bug.cgi?id=692953 I
> replaced the alsasink I had by a fakesink with sync=true and I got on
> the console:
>
> audiobasesrc gstaudiobasesrc.c:852:gst_audio_base_src_create:<alsasrc0>
> create DISCONT of 6400 samples at sample 676800
> audiobasesrc gstaudiobasesrc.c:857:gst_audio_base_src_create:<alsasrc0>
> warning: Can't record audio fast enough
> audiobasesrc gstaudiobasesrc.c:857:gst_audio_base_src_create:<alsasrc0>
> warning: Dropped 6400 samples. This is most likely because downstream
> can't keep up and is consuming samples too slowly.
>
>
> With the alsasink back also appears:
>
> audiobasesink
> gstaudiobasesink.c:1274:gst_audio_base_sink_skew_slaving:<audio
> sink> correct clock skew 3170231239 > 20000000
> audiobasesink
> gstaudiobasesink.c:1580:gst_audio_base_sink_get_alignment:<audio
> sink> Unexpected discontinuity in audio timestamps of
> +0:00:00.040000000, resyncing
> audiobasesink
> gstaudiobasesink.c:1274:gst_audio_base_sink_skew_slaving:<audio
> sink> correct clock skew 3503665481 > 20000000
>
> but still fails. Using slave-method=0 (resample) on the alsasrc kind
> of improves things but not by much.
> Setting the whole pipeline to READY or NULL before linking the muxing
> bin works but it is not acceptable for my use case.
>
> I tried using gst-plugins-base from 1.1.4 with the rest of the system
> at 1.2.0, same results. I compiled a few versions of gst* in between
> but as far as my testing goes 1.1.4 is the first that works.
>
> I'm kind of lost as to where to look now, any thoughts?
> _______________________________________________
> gstreamer-devel mailing list
> gstreamer-devel at lists.freedesktop.org
> http://lists.freedesktop.org/mailman/listinfo/gstreamer-devel
>



-- 

Rob Krakora,
Senior Software Engineer

MessageNet Systems
101 E Carmel Dr, Suite 105
Carmel, IN 46032

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