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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.
This commit is contained in:
parent
f7ee816355
commit
b296c96169
2 changed files with 28 additions and 30 deletions
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@ -1835,6 +1835,16 @@ gst_base_audio_sink_change_state (GstElement * element,
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sink->priv->eos_rendering = 0;
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gst_ring_buffer_set_flushing (sink->ringbuffer, FALSE);
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gst_ring_buffer_may_start (sink->ringbuffer, FALSE);
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/* Only post clock-provide messages if this is the clock that
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* we've created. If the subclass has overriden it the subclass
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* should post this messages whenever necessary */
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if (sink->provided_clock && GST_IS_AUDIO_CLOCK (sink->provided_clock) &&
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GST_AUDIO_CLOCK_CAST (sink->provided_clock)->func ==
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(GstAudioClockGetTimeFunc) gst_base_audio_sink_get_time)
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gst_element_post_message (element,
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gst_message_new_clock_provide (GST_OBJECT_CAST (element),
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sink->provided_clock, TRUE));
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break;
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case GST_STATE_CHANGE_PAUSED_TO_PLAYING:
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GST_OBJECT_LOCK (sink);
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@ -1849,18 +1859,17 @@ gst_base_audio_sink_change_state (GstElement * element,
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/* sync rendering on eos needs running clock */
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gst_ring_buffer_start (sink->ringbuffer);
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}
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/* Only post clock-provide messages if this is the clock that
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* we've created. If the subclass has overriden it the subclass
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* should post this messages whenever necessary */
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if (sink->provided_clock && GST_IS_AUDIO_CLOCK (sink->provided_clock) &&
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GST_AUDIO_CLOCK_CAST (sink->provided_clock)->func ==
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(GstAudioClockGetTimeFunc) gst_base_audio_sink_get_time)
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gst_element_post_message (element,
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gst_message_new_clock_provide (GST_OBJECT_CAST (element),
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sink->provided_clock, TRUE));
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break;
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case GST_STATE_CHANGE_PLAYING_TO_PAUSED:
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/* ringbuffer cannot start anymore */
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gst_ring_buffer_may_start (sink->ringbuffer, FALSE);
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gst_ring_buffer_pause (sink->ringbuffer);
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GST_OBJECT_LOCK (sink);
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sink->priv->sync_latency = FALSE;
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GST_OBJECT_UNLOCK (sink);
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break;
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case GST_STATE_CHANGE_PAUSED_TO_READY:
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/* Only post clock-lost messages if this is the clock that
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* we've created. If the subclass has overriden it the subclass
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* should post this messages whenever necessary */
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@ -1871,15 +1880,6 @@ gst_base_audio_sink_change_state (GstElement * element,
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gst_message_new_clock_lost (GST_OBJECT_CAST (element),
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sink->provided_clock));
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/* ringbuffer cannot start anymore */
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gst_ring_buffer_may_start (sink->ringbuffer, FALSE);
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gst_ring_buffer_pause (sink->ringbuffer);
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GST_OBJECT_LOCK (sink);
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sink->priv->sync_latency = FALSE;
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GST_OBJECT_UNLOCK (sink);
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break;
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case GST_STATE_CHANGE_PAUSED_TO_READY:
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/* make sure we unblock before calling the parent state change
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* so it can grab the STREAM_LOCK */
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gst_ring_buffer_set_flushing (sink->ringbuffer, TRUE);
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@ -1076,11 +1076,6 @@ gst_base_audio_src_change_state (GstElement * element,
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src->next_sample = -1;
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gst_ring_buffer_set_flushing (src->ringbuffer, FALSE);
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gst_ring_buffer_may_start (src->ringbuffer, FALSE);
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break;
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case GST_STATE_CHANGE_PAUSED_TO_PLAYING:
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GST_DEBUG_OBJECT (src, "PAUSED->PLAYING");
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gst_ring_buffer_may_start (src->ringbuffer, TRUE);
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/* Only post clock-provide messages if this is the clock that
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* we've created. If the subclass has overriden it the subclass
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* should post this messages whenever necessary */
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@ -1091,8 +1086,17 @@ gst_base_audio_src_change_state (GstElement * element,
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gst_message_new_clock_provide (GST_OBJECT_CAST (element),
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src->clock, TRUE));
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break;
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case GST_STATE_CHANGE_PAUSED_TO_PLAYING:
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GST_DEBUG_OBJECT (src, "PAUSED->PLAYING");
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gst_ring_buffer_may_start (src->ringbuffer, TRUE);
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break;
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case GST_STATE_CHANGE_PLAYING_TO_PAUSED:
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GST_DEBUG_OBJECT (src, "PLAYING->PAUSED");
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gst_ring_buffer_may_start (src->ringbuffer, FALSE);
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gst_ring_buffer_pause (src->ringbuffer);
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break;
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case GST_STATE_CHANGE_PAUSED_TO_READY:
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GST_DEBUG_OBJECT (src, "PAUSED->READY");
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/* Only post clock-lost messages if this is the clock that
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* we've created. If the subclass has overriden it the subclass
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* should post this messages whenever necessary */
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@ -1101,12 +1105,6 @@ gst_base_audio_src_change_state (GstElement * element,
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(GstAudioClockGetTimeFunc) gst_base_audio_src_get_time)
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gst_element_post_message (element,
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gst_message_new_clock_lost (GST_OBJECT_CAST (element), src->clock));
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gst_ring_buffer_may_start (src->ringbuffer, FALSE);
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gst_ring_buffer_pause (src->ringbuffer);
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break;
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case GST_STATE_CHANGE_PAUSED_TO_READY:
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GST_DEBUG_OBJECT (src, "PAUSED->READY");
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gst_ring_buffer_set_flushing (src->ringbuffer, TRUE);
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break;
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default:
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