Re: [gst-devel] gst_event_new_flush_(start|stop) messing up the pipeline...

[<thi...@gm...>]

On Fri, Feb 12, 2010 at 2:32 PM, S Boucher <st...@ya...> wrote:

>
>
> --- On Fri, 2/12/10, Tim-Philipp Müller <t....@ze...> wrote:
>
> > If audio is not played any longer, then either buffers are
> > dropped
> > somewhere before the sink, or buffers are modified to
> > silence data, or
> > the sink drops/clips them.
>
> All buffers a pushed successfully the 1st and 2nd time (its the same data
> being pushed both times).
>
> Inserting an identity element prior to the sink and dumping the buffers
> show that the buffers are, as expected, the same the 1st and 2nd time.
>
> the audiosink appears to be consuming the buffers correctly the first and
> 2nd time. i.e. I get a series of (with the "at" increasing normally, and for
> the 1st and 2nd time through):
>
> 0:00:04.207250106 12211  0x84b2a80 DEBUG        baseaudiosink
> gstbaseaudiosink.c:1475:gst_base_audio_sink_render:<audiosink-actual-sink-pulse>
> rendering at 0 368/368
> 0:00:04.207293501 12211  0x84b2a80 DEBUG        baseaudiosink
> gstbaseaudiosink.c:1485:gst_base_audio_sink_render:<audiosink-actual-sink-pulse>
> wrote 368 of 368
> 0:00:04.207318364 12211  0x84b2a80 DEBUG        baseaudiosink
> gstbaseaudiosink.c:1508:gst_base_audio_sink_render:<audiosink-actual-sink-pulse>
> next sample expected at 368
>
> So, something's getting busted in the pulse audio sink due to the
> flush_start/flush_stop sent between the first and 2nd time.
>

You can compare what your element does with some other element of the same
kind (demuxer, decoder, source) that's on -base or -good and check if
something is different.


>
>
>
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-- 
Thiago Sousa Santos