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gstaudiobasesink: Always reset avg_skew on _reset

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Only case in which it wasn't (READY to PAUSED) should
have had this value reseted too.
This commit is contained in:
Reynaldo H. Verdejo Pinochet 2013-12-20 17:58:43 -03:00
parent adf800087c
commit c1de7cdefb
1 changed files with 7 additions and 10 deletions
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17
gst-libs/gst/audio/gstaudiobasesink.c
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@ -160,8 +160,7 @@ 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 inline void gst_audio_base_sink_reset_sync (GstAudioBaseSink * sink);
static GstClockTime gst_audio_base_sink_get_time (GstClock * clock,
GstAudioBaseSink * sink);
static void gst_audio_base_sink_callback (GstAudioRingBuffer * rbuf,
@ -885,7 +884,7 @@ gst_audio_base_sink_setcaps (GstBaseSink * bsink, GstCaps * caps)
goto acquire_error;
/* We need to resync since the ringbuffer restarted */
gst_audio_base_sink_reset_sync (sink, TRUE);
gst_audio_base_sink_reset_sync (sink);
if (bsink->pad_mode == GST_PAD_MODE_PUSH) {
GST_DEBUG_OBJECT (sink, "activate ringbuffer");
@ -958,14 +957,12 @@ gst_audio_base_sink_fixate (GstBaseSink * bsink, GstCaps * caps)
}
static inline void
gst_audio_base_sink_reset_sync (GstAudioBaseSink * sink, gboolean sync_skew)
gst_audio_base_sink_reset_sync (GstAudioBaseSink * sink)
{
sink->next_sample = -1;
sink->priv->eos_time = -1;
sink->priv->discont_time = -1;
if (sync_skew)
sink->priv->avg_skew = -1;
sink->priv->avg_skew = -1;
}
static void
@ -1081,7 +1078,7 @@ gst_audio_base_sink_event (GstBaseSink * bsink, GstEvent * event)
break;
case GST_EVENT_FLUSH_STOP:
/* always resync on sample after a flush */
gst_audio_base_sink_reset_sync (sink, TRUE);
gst_audio_base_sink_reset_sync (sink);
if (sink->ringbuffer)
gst_audio_ring_buffer_set_flushing (sink->ringbuffer, FALSE);
break;
@ -1497,7 +1494,7 @@ gst_audio_base_sink_sync_latency (GstBaseSink * bsink, GstMiniObject * obj)
break;
}
gst_audio_base_sink_reset_sync (sink, TRUE);
gst_audio_base_sink_reset_sync (sink);
return GST_FLOW_OK;
@ -2166,7 +2163,7 @@ gst_audio_base_sink_change_state (GstElement * element,
goto open_failed;
break;
case GST_STATE_CHANGE_READY_TO_PAUSED:
gst_audio_base_sink_reset_sync (sink, FALSE);
gst_audio_base_sink_reset_sync (sink);
sink->priv->last_align = -1;
gst_audio_ring_buffer_set_flushing (sink->ringbuffer, FALSE);
gst_audio_ring_buffer_may_start (sink->ringbuffer, FALSE);