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

[S Boucher]

I think what is happening is that after the flush/new segment, the audio sink is already moving along, so that the audio data supposed to match the new segment, when it arrives, is already too late and gets dropped.


I hit the following in gst_ring_buffer_commit_full():

      /* segment too far ahead, writer too slow, we need to drop, hopefully UNLIKELY */
      if (G_UNLIKELY (diff < 0)) {
        /* we need to drop one segment at a time, pretend we wrote a
         * segment. */
        skip = TRUE;
        break;
      }


So, how do I go about getting my pipeline to preroll some after a flush?

--- On Fri, 2/12/10, S Boucher <stbya at yahoo.com> wrote:

> From: S Boucher <stbya at yahoo.com>
> Subject: Re: [gst-devel] gst_event_new_flush_(start|stop) messing up the pipeline...
> To: "Discussion of the development of GStreamer" <gstreamer-devel at lists.sourceforge.net>
> Received: Friday, February 12, 2010, 12:32 PM
> 
> 
> --- On Fri, 2/12/10, Tim-Philipp Müller <t.i.m at zen.co.uk>
> 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.
> 
> 
> 
>      
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