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gstaudiobasesink: Factor out reset sync routine

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This commit is contained in:
Reynaldo H. Verdejo Pinochet 2013-12-20 15:13:54 -03:00
parent b324d67586
commit d35db35258
1 changed files with 18 additions and 15 deletions
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33
gst-libs/gst/audio/gstaudiobasesink.c
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@ -160,6 +160,8 @@ static gboolean gst_audio_base_sink_query (GstElement * element, GstQuery *
query);
static GstClock *gst_audio_base_sink_provide_clock (GstElement * elem);
static inline void gst_audio_base_sink_reset_sync (GstAudioBaseSink * sink,
gboolean sync_skew);
static GstClockTime gst_audio_base_sink_get_time (GstClock * clock,
GstAudioBaseSink * sink);
static void gst_audio_base_sink_callback (GstAudioRingBuffer * rbuf,
@ -465,6 +467,7 @@ gst_audio_base_sink_query (GstElement * element, GstQuery * query)
* amount of time. */
max_latency = (max_l == -1) ? -1 : (base_latency + max_l);
GST_DEBUG_OBJECT (basesink,
"peer min %" GST_TIME_FORMAT ", our min latency: %"
GST_TIME_FORMAT, GST_TIME_ARGS (min_l),
@ -882,10 +885,7 @@ gst_audio_base_sink_setcaps (GstBaseSink * bsink, GstCaps * caps)
goto acquire_error;
/* We need to resync since the ringbuffer restarted */
sink->priv->avg_skew = -1;
sink->next_sample = -1;
sink->priv->eos_time = -1;
sink->priv->discont_time = -1;
gst_audio_base_sink_reset_sync (sink, TRUE);
if (bsink->pad_mode == GST_PAD_MODE_PUSH) {
GST_DEBUG_OBJECT (sink, "activate ringbuffer");
@ -957,6 +957,17 @@ gst_audio_base_sink_fixate (GstBaseSink * bsink, GstCaps * caps)
return caps;
}
static inline void
gst_audio_base_sink_reset_sync (GstAudioBaseSink * sink, gboolean sync_skew)
{
sink->next_sample = -1;
sink->priv->eos_time = -1;
sink->priv->discont_time = -1;
if (sync_skew)
sink->priv->avg_skew = -1;
}
static void
gst_audio_base_sink_get_times (GstBaseSink * bsink, GstBuffer * buffer,
GstClockTime * start, GstClockTime * end)
@ -1070,10 +1081,7 @@ gst_audio_base_sink_event (GstBaseSink * bsink, GstEvent * event)
break;
case GST_EVENT_FLUSH_STOP:
/* always resync on sample after a flush */
sink->priv->avg_skew = -1;
sink->next_sample = -1;
sink->priv->eos_time = -1;
sink->priv->discont_time = -1;
gst_audio_base_sink_reset_sync (sink, TRUE);
if (sink->ringbuffer)
gst_audio_ring_buffer_set_flushing (sink->ringbuffer, FALSE);
break;
@ -1489,10 +1497,7 @@ gst_audio_base_sink_sync_latency (GstBaseSink * bsink, GstMiniObject * obj)
break;
}
sink->priv->avg_skew = -1;
sink->next_sample = -1;
sink->priv->eos_time = -1;
sink->priv->discont_time = -1;
gst_audio_base_sink_reset_sync (sink, TRUE);
return GST_FLOW_OK;
@ -2161,10 +2166,8 @@ gst_audio_base_sink_change_state (GstElement * element,
goto open_failed;
break;
case GST_STATE_CHANGE_READY_TO_PAUSED:
sink->next_sample = -1;
gst_audio_base_sink_reset_sync (sink, FALSE);
sink->priv->last_align = -1;
sink->priv->eos_time = -1;
sink->priv->discont_time = -1;
gst_audio_ring_buffer_set_flushing (sink->ringbuffer, FALSE);
gst_audio_ring_buffer_may_start (sink->ringbuffer, FALSE);