gst-libs/gst/audio/gstaudioclock.*: Add method to inform the clock that the time starts from 0 again. We use this inf...

Original commit message from CVS:
* gst-libs/gst/audio/gstaudioclock.c: (gst_audio_clock_init),
(gst_audio_clock_reset), (gst_audio_clock_get_internal_time):
* gst-libs/gst/audio/gstaudioclock.h:
Add method to inform the clock that the time starts from 0 again. We use
this info to calculate a clock offset so that the time we report in
internal_time is monotonically increasing, as required by the clock base
class. Fixes #521761.
API: GstAudioClock::gst_audio_clock_reset()
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_skew_slaving),
(gst_base_audio_sink_change_state):
* gst-libs/gst/audio/gstbaseaudiosrc.c:
(gst_base_audio_src_create), (gst_base_audio_src_change_state):
Reset reported time when we (re)create the ringbuffer.

ChangeLog

@@ -1,3 +1,21 @@
+2008-05-27  Wim Taymans  <wim.taymans@collabora.co.uk>
+
+	* gst-libs/gst/audio/gstaudioclock.c: (gst_audio_clock_init),
+	(gst_audio_clock_reset), (gst_audio_clock_get_internal_time):
+	* gst-libs/gst/audio/gstaudioclock.h:
+	Add method to inform the clock that the time starts from 0 again. We use
+	this info to calculate a clock offset so that the time we report in
+	internal_time is monotonically increasing, as required by the clock base
+	class. Fixes #521761.
+	API: GstAudioClock::gst_audio_clock_reset()
+
+	* gst-libs/gst/audio/gstbaseaudiosink.c:
+	(gst_base_audio_sink_skew_slaving),
+	(gst_base_audio_sink_change_state):
+	* gst-libs/gst/audio/gstbaseaudiosrc.c:
+	(gst_base_audio_src_create), (gst_base_audio_src_change_state):
+	Reset reported time when we (re)create the ringbuffer.
+
 2008-05-27  Tim-Philipp Müller  <tim.muller at collabora co uk>
 
 	* ext/alsa/gstalsamixertrack.c:

gst-libs/gst/audio/gstaudioclock.c

@@ -94,6 +94,7 @@ static void
 gst_audio_clock_init (GstAudioClock * clock)
 {
   clock->last_time = 0;
+  clock->abidata.ABI.time_offset = 0;
   GST_OBJECT_FLAG_SET (clock, GST_CLOCK_FLAG_CAN_SET_MASTER);
 }
 
@@ -122,17 +123,39 @@ gst_audio_clock_new (gchar * name, GstAudioClockGetTimeFunc func,
   return (GstClock *) aclock;
 }
 
+/**
+ * gst_audio_clock_reset:
+ * @clock: a #GstAudioClock
+ * @time: a #GstClockTime
+ *
+ * Inform @clock that future calls to #GstAudioClockGetTimeFunc will return values
+ * starting from @time. The clock will update an internal offset to make sure that
+ * future calls to internal_time will return an increasing result as required by
+ * the #GstClock object.
+ */
+void
+gst_audio_clock_reset (GstAudioClock * clock, GstClockTime time)
+{
+  if (clock->last_time >= time)
+    clock->abidata.ABI.time_offset = clock->last_time - time;
+  else
+    clock->abidata.ABI.time_offset = -(time - clock->last_time);
+}
+
 static GstClockTime
 gst_audio_clock_get_internal_time (GstClock * clock)
 {
-  GstAudioClock *aclock = GST_AUDIO_CLOCK (clock);
+  GstAudioClock *aclock;
   GstClockTime result;
 
+  aclock = GST_AUDIO_CLOCK (clock);
+
   result = aclock->func (clock, aclock->user_data);
   if (result == GST_CLOCK_TIME_NONE)
     result = aclock->last_time;
-  else
+  else {
+    result += aclock->abidata.ABI.time_offset;
     aclock->last_time = result;
-
+  }
   return result;
 }

gst-libs/gst/audio/gstaudioclock.h

@@ -71,7 +71,13 @@ struct _GstAudioClock {
   GstClockTime last_time;
 
   /*< private >*/
-  gpointer _gst_reserved[GST_PADDING];
+  union {
+    struct {
+      GstClockTimeDiff   time_offset;
+    } ABI;
+    /* adding + 0 to mark ABI change to be undone later */
+    gpointer _gst_reserved[GST_PADDING + 0];
+  } abidata;
 };
 
 struct _GstAudioClockClass {
@@ -84,6 +90,7 @@ struct _GstAudioClockClass {
 GType           gst_audio_clock_get_type        (void);
 GstClock*       gst_audio_clock_new             (gchar *name, GstAudioClockGetTimeFunc func,
                                                  gpointer user_data);
+void            gst_audio_clock_reset           (GstAudioClock *clock, GstClockTime time);
 
 G_END_DECLS
 

gst-libs/gst/audio/gstbaseaudiosink.c

@@ -1604,6 +1604,7 @@ 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);
         sink->ringbuffer = gst_base_audio_sink_create_ringbuffer (sink);
       }
       if (!gst_ring_buffer_open_device (sink->ringbuffer))

gst-libs/gst/audio/gstbaseaudiosrc.c

@@ -870,6 +870,7 @@ gst_base_audio_src_change_state (GstElement * element,
     case GST_STATE_CHANGE_NULL_TO_READY:
       GST_DEBUG_OBJECT (src, "NULL->READY");
       if (src->ringbuffer == NULL) {
+        gst_audio_clock_reset (GST_AUDIO_CLOCK (src->clock), 0);
         src->ringbuffer = gst_base_audio_src_create_ringbuffer (src);
       }
       if (!gst_ring_buffer_open_device (src->ringbuffer))