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gst-libs/gst/audio/gstbaseaudiosink.c: Use new basesink method to make our EOS drain interruptable.

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Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_drain):
Use new basesink method to make our EOS drain interruptable.
This commit is contained in:
Wim Taymans 2007-10-10 15:36:56 +00:00
parent c96877779a
commit 02f280a9a0
2 changed files with 16 additions and 17 deletions
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6
ChangeLog
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@ -1,3 +1,9 @@
2007-10-10 Wim Taymans <wim.taymans@gmail.com>
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_drain):
Use new basesink method to make our EOS drain interruptable.
2007-10-10 Jan Schmidt <Jan.Schmidt@sun.com> 2007-10-10 Jan Schmidt <Jan.Schmidt@sun.com>
* gst-libs/gst/rtp/gstrtppayloads.c: * gst-libs/gst/rtp/gstrtppayloads.c:

27
gst-libs/gst/audio/gstbaseaudiosink.c
View file

@ -547,9 +547,7 @@ gst_base_audio_sink_get_times (GstBaseSink * bsink, GstBuffer * buffer,
*end = GST_CLOCK_TIME_NONE; *end = GST_CLOCK_TIME_NONE;
} }
/* FIXME, this waits for the drain to happen but it cannot be /* This waits for the drain to happen and can be canceled */
* canceled.
*/
static gboolean static gboolean
gst_base_audio_sink_drain (GstBaseAudioSink * sink) gst_base_audio_sink_drain (GstBaseAudioSink * sink)
{ {
@ -559,33 +557,28 @@ gst_base_audio_sink_drain (GstBaseAudioSink * sink)
return TRUE; return TRUE;
/* need to start playback before we can drain, but only when /* need to start playback before we can drain, but only when
* we have successfully negotiated a format and thus aqcuired the * we have successfully negotiated a format and thus acquired the
* ringbuffer. */ * ringbuffer. */
if (gst_ring_buffer_is_acquired (sink->ringbuffer)) if (gst_ring_buffer_is_acquired (sink->ringbuffer))
gst_ring_buffer_start (sink->ringbuffer); gst_ring_buffer_start (sink->ringbuffer);
if (sink->next_sample != -1) { if (sink->next_sample != -1) {
GstClockTime time; GstClockTime time;
GstClock *clock;
/* convert next expected sample to time */
time = time =
gst_util_uint64_scale_int (sink->next_sample, GST_SECOND, gst_util_uint64_scale_int (sink->next_sample, GST_SECOND,
sink->ringbuffer->spec.rate); sink->ringbuffer->spec.rate);
GST_OBJECT_LOCK (sink); GST_DEBUG_OBJECT (sink,
if ((clock = GST_ELEMENT_CLOCK (sink)) != NULL) { "last sample %" G_GUINT64_FORMAT ", time %" GST_TIME_FORMAT,
GstClockID id = gst_clock_new_single_shot_id (clock, time); sink->next_sample, GST_TIME_ARGS (time));
GST_OBJECT_UNLOCK (sink); /* wait for the EOS time to be reached, this is the time when the last
* sample is played. */
gst_base_sink_wait_eos (GST_BASE_SINK (sink), time, NULL);
GST_DEBUG_OBJECT (sink, "waiting for last sample to play"); sink->next_sample = -1;
gst_clock_id_wait (id, NULL);
gst_clock_id_unref (id);
sink->next_sample = -1;
} else {
GST_OBJECT_UNLOCK (sink);
}
} }
return TRUE; return TRUE;
} }