Commit b296c961 authored by Sebastian Dröge's avatar Sebastian Dröge
Browse files

baseaudiosink/baseaudiosrc: Post CLOCK-LOST/CLOCK-PROVIDE when going to/from READY

Otherwise the clocks are redistributed every time the pipeline
goes to PAUSED, which is quite expensive.
parent f7ee8163
......@@ -1835,6 +1835,16 @@ gst_base_audio_sink_change_state (GstElement * element,
sink->priv->eos_rendering = 0;
gst_ring_buffer_set_flushing (sink->ringbuffer, FALSE);
gst_ring_buffer_may_start (sink->ringbuffer, FALSE);
/* Only post clock-provide messages if this is the clock that
* we've created. If the subclass has overriden it the subclass
* should post this messages whenever necessary */
if (sink->provided_clock && GST_IS_AUDIO_CLOCK (sink->provided_clock) &&
GST_AUDIO_CLOCK_CAST (sink->provided_clock)->func ==
(GstAudioClockGetTimeFunc) gst_base_audio_sink_get_time)
gst_element_post_message (element,
gst_message_new_clock_provide (GST_OBJECT_CAST (element),
sink->provided_clock, TRUE));
break;
case GST_STATE_CHANGE_PAUSED_TO_PLAYING:
GST_OBJECT_LOCK (sink);
......@@ -1849,18 +1859,17 @@ gst_base_audio_sink_change_state (GstElement * element,
/* sync rendering on eos needs running clock */
gst_ring_buffer_start (sink->ringbuffer);
}
/* Only post clock-provide messages if this is the clock that
* we've created. If the subclass has overriden it the subclass
* should post this messages whenever necessary */
if (sink->provided_clock && GST_IS_AUDIO_CLOCK (sink->provided_clock) &&
GST_AUDIO_CLOCK_CAST (sink->provided_clock)->func ==
(GstAudioClockGetTimeFunc) gst_base_audio_sink_get_time)
gst_element_post_message (element,
gst_message_new_clock_provide (GST_OBJECT_CAST (element),
sink->provided_clock, TRUE));
break;
case GST_STATE_CHANGE_PLAYING_TO_PAUSED:
/* ringbuffer cannot start anymore */
gst_ring_buffer_may_start (sink->ringbuffer, FALSE);
gst_ring_buffer_pause (sink->ringbuffer);
GST_OBJECT_LOCK (sink);
sink->priv->sync_latency = FALSE;
GST_OBJECT_UNLOCK (sink);
break;
case GST_STATE_CHANGE_PAUSED_TO_READY:
/* Only post clock-lost messages if this is the clock that
* we've created. If the subclass has overriden it the subclass
* should post this messages whenever necessary */
......@@ -1871,15 +1880,6 @@ gst_base_audio_sink_change_state (GstElement * element,
gst_message_new_clock_lost (GST_OBJECT_CAST (element),
sink->provided_clock));
/* ringbuffer cannot start anymore */
gst_ring_buffer_may_start (sink->ringbuffer, FALSE);
gst_ring_buffer_pause (sink->ringbuffer);
GST_OBJECT_LOCK (sink);
sink->priv->sync_latency = FALSE;
GST_OBJECT_UNLOCK (sink);
break;
case GST_STATE_CHANGE_PAUSED_TO_READY:
/* make sure we unblock before calling the parent state change
* so it can grab the STREAM_LOCK */
gst_ring_buffer_set_flushing (sink->ringbuffer, TRUE);
......
......@@ -1076,11 +1076,6 @@ gst_base_audio_src_change_state (GstElement * element,
src->next_sample = -1;
gst_ring_buffer_set_flushing (src->ringbuffer, FALSE);
gst_ring_buffer_may_start (src->ringbuffer, FALSE);
break;
case GST_STATE_CHANGE_PAUSED_TO_PLAYING:
GST_DEBUG_OBJECT (src, "PAUSED->PLAYING");
gst_ring_buffer_may_start (src->ringbuffer, TRUE);
/* Only post clock-provide messages if this is the clock that
* we've created. If the subclass has overriden it the subclass
* should post this messages whenever necessary */
......@@ -1091,8 +1086,17 @@ gst_base_audio_src_change_state (GstElement * element,
gst_message_new_clock_provide (GST_OBJECT_CAST (element),
src->clock, TRUE));
break;
case GST_STATE_CHANGE_PAUSED_TO_PLAYING:
GST_DEBUG_OBJECT (src, "PAUSED->PLAYING");
gst_ring_buffer_may_start (src->ringbuffer, TRUE);
break;
case GST_STATE_CHANGE_PLAYING_TO_PAUSED:
GST_DEBUG_OBJECT (src, "PLAYING->PAUSED");
gst_ring_buffer_may_start (src->ringbuffer, FALSE);
gst_ring_buffer_pause (src->ringbuffer);
break;
case GST_STATE_CHANGE_PAUSED_TO_READY:
GST_DEBUG_OBJECT (src, "PAUSED->READY");
/* Only post clock-lost messages if this is the clock that
* we've created. If the subclass has overriden it the subclass
* should post this messages whenever necessary */
......@@ -1101,12 +1105,6 @@ gst_base_audio_src_change_state (GstElement * element,
(GstAudioClockGetTimeFunc) gst_base_audio_src_get_time)
gst_element_post_message (element,
gst_message_new_clock_lost (GST_OBJECT_CAST (element), src->clock));
gst_ring_buffer_may_start (src->ringbuffer, FALSE);
gst_ring_buffer_pause (src->ringbuffer);
break;
case GST_STATE_CHANGE_PAUSED_TO_READY:
GST_DEBUG_OBJECT (src, "PAUSED->READY");
gst_ring_buffer_set_flushing (src->ringbuffer, TRUE);
break;
default:
......
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