gst-libs/gst/audio/gstbaseaudiosrc.c: Add debug info.

Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosrc.c:
(gst_base_audio_src_create), (gst_base_audio_src_change_state):
Add debug info.
When going from PLAYING to PAUSED, pause the ringbuffer before calling
the parent state change function, just like the audiosink, because the
parent waits for the element to finish its processing before completing
the state change. This makes going to PAUSED a lot snappier.
When going from READY to PAUSED, don't allow the ringbuffer to start
yet.

diff --git a/ChangeLog b/ChangeLog
index a717b8c..5893066 100644 (file)
--- a/ChangeLog
+++ b/ChangeLog
@@ -1,3 +1,15 @@
+2007-12-17  Wim Taymans  <wim.taymans@collabora.co.uk>
+
+       * gst-libs/gst/audio/gstbaseaudiosrc.c:
+       (gst_base_audio_src_create), (gst_base_audio_src_change_state):
+       Add debug info.
+       When going from PLAYING to PAUSED, pause the ringbuffer before calling
+       the parent state change function, just like the audiosink, because the
+       parent waits for the element to finish its processing before completing
+       the state change. This makes going to PAUSED a lot snappier.
+       When going from READY to PAUSED, don't allow the ringbuffer to start
+       yet.
+
 2007-12-17  Edward Hervey  <edward.hervey@collabora.co.uk>
 
        * gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):

diff --git a/gst-libs/gst/audio/gstbaseaudiosrc.c b/gst-libs/gst/audio/gstbaseaudiosrc.c
index 16aef20..c21c1a3 100644 (file)
--- a/gst-libs/gst/audio/gstbaseaudiosrc.c
+++ b/gst-libs/gst/audio/gstbaseaudiosrc.c
@@ -642,9 +642,12 @@ gst_base_audio_src_create (GstBaseSrc * bsrc, guint64 offset, guint length,
       break;
 
     /* else something interrupted us and we wait for playing again. */
+    GST_DEBUG_OBJECT (src, "wait playing");
     if (gst_base_src_wait_playing (bsrc) != GST_FLOW_OK)
       goto stopped;
 
+    GST_DEBUG_OBJECT (src, "continue playing");
+
     /* read next samples */
     sample += read;
     samples -= read;
@@ -760,6 +763,7 @@ gst_base_audio_src_change_state (GstElement * element,
 
   switch (transition) {
     case GST_STATE_CHANGE_NULL_TO_READY:
+      GST_DEBUG_OBJECT (src, "NULL->READY");
       if (src->ringbuffer == NULL) {
         src->ringbuffer = gst_base_audio_src_create_ringbuffer (src);
       }
@@ -767,13 +771,22 @@ gst_base_audio_src_change_state (GstElement * element,
         goto open_failed;
       break;
     case GST_STATE_CHANGE_READY_TO_PAUSED:
+      GST_DEBUG_OBJECT (src, "READY->PAUSED");
       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);
       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");
       gst_ring_buffer_set_flushing (src->ringbuffer, TRUE);
       break;
     default:
@@ -783,14 +796,12 @@ gst_base_audio_src_change_state (GstElement * element,
   ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
 
   switch (transition) {
-    case GST_STATE_CHANGE_PLAYING_TO_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");
       gst_ring_buffer_release (src->ringbuffer);
       break;
     case GST_STATE_CHANGE_READY_TO_NULL:
+      GST_DEBUG_OBJECT (src, "READY->NULL");
       gst_ring_buffer_close_device (src->ringbuffer);
       gst_object_unparent (GST_OBJECT_CAST (src->ringbuffer));
       src->ringbuffer = NULL;