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Revert "baseaudiosink: Allocate and free the clock in NULL->READY and reverse"

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This reverts commit cea2644ed8.

Many audio sink assume that they can create a clock in
the instance init function and it will be there forever
and not be cleared by the state change functions.
This commit is contained in:
Sebastian Dröge 2010-06-03 13:44:40 +02:00
parent cea2644ed8
commit a5c35621c3
1 changed files with 10 additions and 18 deletions
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28
gst-libs/gst/audio/gstbaseaudiosink.c
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@ -272,6 +272,9 @@ gst_base_audio_sink_init (GstBaseAudioSink * baseaudiosink,
baseaudiosink->provide_clock = DEFAULT_PROVIDE_CLOCK;
baseaudiosink->priv->slave_method = DEFAULT_SLAVE_METHOD;
baseaudiosink->provided_clock = gst_audio_clock_new ("GstAudioSinkClock",
(GstAudioClockGetTimeFunc) gst_base_audio_sink_get_time, baseaudiosink);
GST_BASE_SINK (baseaudiosink)->can_activate_push = TRUE;
GST_BASE_SINK (baseaudiosink)->can_activate_pull = DEFAULT_CAN_ACTIVATE_PULL;
baseaudiosink->priv->drift_tolerance = DEFAULT_DRIFT_TOLERANCE;
@ -307,6 +310,10 @@ gst_base_audio_sink_dispose (GObject * object)
sink = GST_BASE_AUDIO_SINK (object);
if (sink->provided_clock)
gst_object_unref (sink->provided_clock);
sink->provided_clock = NULL;
if (sink->ringbuffer) {
gst_object_unparent (GST_OBJECT_CAST (sink->ringbuffer));
sink->ringbuffer = NULL;
@ -1813,8 +1820,10 @@ gst_base_audio_sink_change_state (GstElement * element,
switch (transition) {
case GST_STATE_CHANGE_NULL_TO_READY:
if (sink->ringbuffer == NULL)
if (sink->ringbuffer == NULL) {
gst_audio_clock_reset (GST_AUDIO_CLOCK (sink->provided_clock), 0);
sink->ringbuffer = gst_base_audio_sink_create_ringbuffer (sink);
}
if (!gst_ring_buffer_open_device (sink->ringbuffer))
goto open_failed;
break;
@ -1861,15 +1870,6 @@ gst_base_audio_sink_change_state (GstElement * element,
ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
switch (transition) {
case GST_STATE_CHANGE_NULL_TO_READY:
/* If the subclass doesn't provide a clock... */
if (!sink->provided_clock)
sink->provided_clock = gst_audio_clock_new ("GstAudioSinkClock",
(GstAudioClockGetTimeFunc) gst_base_audio_sink_get_time, sink);
gst_audio_clock_reset (GST_AUDIO_CLOCK (sink->provided_clock), 0);
gst_element_post_message (element,
gst_message_new_clock_provide (GST_OBJECT_CAST (element),
sink->provided_clock, TRUE));
case GST_STATE_CHANGE_PLAYING_TO_PAUSED:
/* stop slaving ourselves to the master, if any */
gst_clock_set_master (sink->provided_clock, NULL);
@ -1886,17 +1886,9 @@ gst_base_audio_sink_change_state (GstElement * element,
gst_ring_buffer_activate (sink->ringbuffer, FALSE);
gst_ring_buffer_release (sink->ringbuffer);
gst_ring_buffer_close_device (sink->ringbuffer);
gst_element_post_message (element,
gst_message_new_clock_provide (GST_OBJECT_CAST (element),
NULL, FALSE));
GST_OBJECT_LOCK (sink);
gst_object_unparent (GST_OBJECT_CAST (sink->ringbuffer));
sink->ringbuffer = NULL;
if (sink->provided_clock)
gst_object_unref (sink->provided_clock);
sink->provided_clock = NULL;
GST_OBJECT_UNLOCK (sink);
break;
default: