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gst-libs/gst/audio/gstbaseaudiosink.c: Really fix audiosink drain handling by keeping track of the running_time of th...

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Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_drain), (gst_base_audio_sink_event),
(gst_base_audio_sink_sync_latency), (gst_base_audio_sink_render),
(gst_base_audio_sink_change_state):
Really fix audiosink drain handling by keeping track of the running_time
of the last sample.
This commit is contained in:
Wim Taymans 2008-11-25 10:32:49 +00:00
parent aec03a4535
commit 6983c1c85b
2 changed files with 32 additions and 18 deletions
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9
ChangeLog
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@ -1,3 +1,12 @@
2008-11-25 Wim Taymans <wim.taymans@collabora.co.uk>
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_drain), (gst_base_audio_sink_event),
(gst_base_audio_sink_sync_latency), (gst_base_audio_sink_render),
(gst_base_audio_sink_change_state):
Really fix audiosink drain handling by keeping track of the running_time
of the last sample.
2008-11-24 Michael Smith <msmith@songbirdnest.com> 2008-11-24 Michael Smith <msmith@songbirdnest.com>
* gst/playback/gstplaybin2.c: * gst/playback/gstplaybin2.c:

41
gst-libs/gst/audio/gstbaseaudiosink.c
View file

@ -54,6 +54,8 @@ struct _GstBaseAudioSinkPrivate
gint64 last_align; gint64 last_align;
gboolean sync_latency; gboolean sync_latency;
GstClockTime eos_time;
}; };
/* BaseAudioSink signals and args */ /* BaseAudioSink signals and args */
@ -720,23 +722,15 @@ gst_base_audio_sink_drain (GstBaseAudioSink * sink)
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->priv->eos_time != -1) {
GstClockTime time;
/* convert next expected sample to time */
time =
gst_util_uint64_scale_int (sink->next_sample, GST_SECOND,
sink->ringbuffer->spec.rate);
GST_DEBUG_OBJECT (sink, GST_DEBUG_OBJECT (sink,
"last sample %" G_GUINT64_FORMAT ", time %" GST_TIME_FORMAT, "last sample time %" GST_TIME_FORMAT,
sink->next_sample, GST_TIME_ARGS (time)); GST_TIME_ARGS (sink->priv->eos_time));
/* wait for the EOS time to be reached, this is the time when the last /* wait for the EOS time to be reached, this is the time when the last
* sample is played. */ * sample is played. */
gst_base_sink_wait_eos (GST_BASE_SINK (sink), time, NULL); gst_base_sink_wait_eos (GST_BASE_SINK (sink), sink->priv->eos_time, NULL);
sink->next_sample = -1;
GST_DEBUG_OBJECT (sink, "drained audio"); GST_DEBUG_OBJECT (sink, "drained audio");
} }
return TRUE; return TRUE;
@ -756,6 +750,7 @@ gst_base_audio_sink_event (GstBaseSink * bsink, GstEvent * event)
/* always resync on sample after a flush */ /* always resync on sample after a flush */
sink->priv->avg_skew = -1; sink->priv->avg_skew = -1;
sink->next_sample = -1; sink->next_sample = -1;
sink->priv->eos_time = -1;
if (sink->ringbuffer) if (sink->ringbuffer)
gst_ring_buffer_set_flushing (sink->ringbuffer, FALSE); gst_ring_buffer_set_flushing (sink->ringbuffer, FALSE);
break; break;
@ -1169,6 +1164,7 @@ gst_base_audio_sink_sync_latency (GstBaseSink * bsink, GstMiniObject * obj)
sink->priv->avg_skew = -1; sink->priv->avg_skew = -1;
sink->next_sample = -1; sink->next_sample = -1;
sink->priv->eos_time = -1;
return GST_FLOW_OK; return GST_FLOW_OK;
@ -1354,12 +1350,13 @@ gst_base_audio_sink_render (GstBaseSink * bsink, GstBuffer * buf)
"running: start %" GST_TIME_FORMAT " - stop %" GST_TIME_FORMAT, "running: start %" GST_TIME_FORMAT " - stop %" GST_TIME_FORMAT,
GST_TIME_ARGS (render_start), GST_TIME_ARGS (render_stop)); GST_TIME_ARGS (render_start), GST_TIME_ARGS (render_stop));
GST_DEBUG_OBJECT (sink, "adding base_time %" GST_TIME_FORMAT, /* store the time of the last sample, we'll use this to perform sync on the
GST_TIME_ARGS (base_time)); * last sample when draining the buffer */
if (bsink->segment.rate >= 0.0) {
/* add base time to sync against the clock */ sink->priv->eos_time = render_stop;
render_start += base_time; } else {
render_stop += base_time; sink->priv->eos_time = render_start;
}
/* compensate for ts-offset and delay we know this will not underflow because we /* compensate for ts-offset and delay we know this will not underflow because we
* clipped above. */ * clipped above. */
@ -1369,6 +1366,13 @@ gst_base_audio_sink_render (GstBaseSink * bsink, GstBuffer * buf)
render_start += sync_offset; render_start += sync_offset;
render_stop += sync_offset; render_stop += sync_offset;
GST_DEBUG_OBJECT (sink, "adding base_time %" GST_TIME_FORMAT,
GST_TIME_ARGS (base_time));
/* add base time to sync against the clock */
render_start += base_time;
render_stop += base_time;
GST_DEBUG_OBJECT (sink, GST_DEBUG_OBJECT (sink,
"after compensation: start %" GST_TIME_FORMAT " - stop %" GST_TIME_FORMAT, "after compensation: start %" GST_TIME_FORMAT " - stop %" GST_TIME_FORMAT,
GST_TIME_ARGS (render_start), GST_TIME_ARGS (render_stop)); GST_TIME_ARGS (render_start), GST_TIME_ARGS (render_stop));
@ -1737,6 +1741,7 @@ gst_base_audio_sink_change_state (GstElement * element,
case GST_STATE_CHANGE_READY_TO_PAUSED: case GST_STATE_CHANGE_READY_TO_PAUSED:
sink->next_sample = -1; sink->next_sample = -1;
sink->priv->last_align = -1; sink->priv->last_align = -1;
sink->priv->eos_time = -1;
gst_ring_buffer_set_flushing (sink->ringbuffer, FALSE); gst_ring_buffer_set_flushing (sink->ringbuffer, FALSE);
gst_ring_buffer_may_start (sink->ringbuffer, FALSE); gst_ring_buffer_may_start (sink->ringbuffer, FALSE);
break; break;