alsasrc: different behaviour and clocking between 1.2 and 1.1.4

[Robert Krakora]

Also what does "top -H" look like while your pipeline is running?


On Fri, Oct 11, 2013 at 10:05 AM, Robert Krakora <
rob.krakora at messagenetsystems.com> wrote:

> 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
>
> MessageNetSystems.com<http://www.messagenetcommunicationsystems.com/?utm_source=email+signature&utm_medium=email&utm_campaign=email+signature+to+homepage>
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>
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-- 

Rob Krakora,
Senior Software Engineer

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

MessageNetSystems.com<http://www.messagenetcommunicationsystems.com/?utm_source=email+signature&utm_medium=email&utm_campaign=email+signature+to+homepage>
Rob.Krakora at MessageNetSystems.com <rob.krakora at messagenetsystems.com>
P: 317.566.1677, 212
F: 317.663.0808

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