[gst-cvs] gst-plugins-base: baseaudiosink: Allocate and free the clock in NULL-> READY and reverse

[Sebastian Dröge]

Module: gst-plugins-base
Branch: master
Commit: cea2644ed86097aadedc9e8731e78a22ffc6246b
URL:    http://cgit.freedesktop.org/gstreamer/gst-plugins-base/commit/?id=cea2644ed86097aadedc9e8731e78a22ffc6246b

Author: Sebastian Dröge <sebastian.droege at collabora.co.uk>
Date:   Wed Jun  2 12:19:00 2010 +0200

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

---

 gst-libs/gst/audio/gstbaseaudiosink.c |   28 ++++++++++++++++++----------
 1 files changed, 18 insertions(+), 10 deletions(-)

diff --git a/gst-libs/gst/audio/gstbaseaudiosink.c b/gst-libs/gst/audio/gstbaseaudiosink.c
index 4e685ae..5c82d71 100644
--- a/gst-libs/gst/audio/gstbaseaudiosink.c
+++ b/gst-libs/gst/audio/gstbaseaudiosink.c
@@ -272,9 +272,6 @@ 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;
@@ -310,10 +307,6 @@ 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;
@@ -1820,10 +1813,8 @@ gst_base_audio_sink_change_state (GstElement * element,
 
   switch (transition) {
     case GST_STATE_CHANGE_NULL_TO_READY:
-      if (sink->ringbuffer == NULL) {
-        gst_audio_clock_reset (GST_AUDIO_CLOCK (sink->provided_clock), 0);
+      if (sink->ringbuffer == NULL)
         sink->ringbuffer = gst_base_audio_sink_create_ringbuffer (sink);
-      }
       if (!gst_ring_buffer_open_device (sink->ringbuffer))
         goto open_failed;
       break;
@@ -1870,6 +1861,15 @@ 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);
@@ -1886,9 +1886,17 @@ 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: