gst-libs/gst/audio/: Make sure the audio clock always returns an increasing value.

Original commit message from CVS:
* gst-libs/gst/audio/TODO:
* gst-libs/gst/audio/gstaudioclock.c: (gst_audio_clock_init),
(gst_audio_clock_get_internal_time):
* gst-libs/gst/audio/gstaudioclock.h:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_init), (gst_base_audio_sink_dispose),
(gst_base_audio_sink_get_time), (gst_base_audio_sink_event),
(gst_base_audio_sink_render),
(gst_base_audio_sink_create_ringbuffer),
(gst_base_audio_sink_change_state):
Make sure the audio clock always returns an increasing value.

diff --git a/ChangeLog b/ChangeLog
index 006ae2a..c45da6a 100644 (file)
--- a/ChangeLog
+++ b/ChangeLog
@@ -1,3 +1,17 @@
+2005-07-20  Wim Taymans  <wim@fluendo.com>
+
+       * gst-libs/gst/audio/TODO:
+       * gst-libs/gst/audio/gstaudioclock.c: (gst_audio_clock_init),
+       (gst_audio_clock_get_internal_time):
+       * gst-libs/gst/audio/gstaudioclock.h:
+       * gst-libs/gst/audio/gstbaseaudiosink.c:
+       (gst_base_audio_sink_init), (gst_base_audio_sink_dispose),
+       (gst_base_audio_sink_get_time), (gst_base_audio_sink_event),
+       (gst_base_audio_sink_render),
+       (gst_base_audio_sink_create_ringbuffer),
+       (gst_base_audio_sink_change_state):
+       Make sure the audio clock always returns an increasing value.
+
 2005-07-19  Andy Wingo  <wingo@pobox.com>
 
        * gst/videotestsrc/: Cleanups.

diff --git a/gst-libs/gst/audio/TODO b/gst-libs/gst/audio/TODO
index 73bdea2..08e0e34 100644 (file)
--- a/gst-libs/gst/audio/TODO
+++ b/gst-libs/gst/audio/TODO
@@ -7,4 +7,4 @@ TODO
         is parsed correctly.
       - implement seek/query/convert
       - implement getrange scheduling
-      
+   - on EOS, only post EOS when the complete ringbuffer has been played.      

diff --git a/gst-libs/gst/audio/gstaudioclock.c b/gst-libs/gst/audio/gstaudioclock.c
index b7faafd..c878f39 100644 (file)
--- a/gst-libs/gst/audio/gstaudioclock.c
+++ b/gst-libs/gst/audio/gstaudioclock.c
@@ -81,6 +81,8 @@ static void
 gst_audio_clock_init (GstAudioClock * clock)
 {
   gst_object_set_name (GST_OBJECT (clock), "GstAudioClock");
+
+  clock->last_time = 0;
 }
 
 GstClock *
@@ -100,6 +102,13 @@ static GstClockTime
 gst_audio_clock_get_internal_time (GstClock * clock)
 {
   GstAudioClock *aclock = GST_AUDIO_CLOCK (clock);
+  GstClockTime result;
+
+  result = aclock->func (clock, aclock->user_data);
+  if (result == GST_CLOCK_TIME_NONE)
+    result = aclock->last_time;
+  else
+    aclock->last_time = result;
 
-  return aclock->func (clock, aclock->user_data);
+  return result;
 }

diff --git a/gst-libs/gst/audio/gstaudioclock.h b/gst-libs/gst/audio/gstaudioclock.h
index 45b6211..d111117 100644 (file)
--- a/gst-libs/gst/audio/gstaudioclock.h
+++ b/gst-libs/gst/audio/gstaudioclock.h
@@ -51,6 +51,8 @@ struct _GstAudioClock {
   GstAudioClockGetTimeFunc func;
   gpointer user_data;
 
+  GstClockTime last_time;
+
   gpointer _gst_reserved[GST_PADDING];
 };
 

diff --git a/gst-libs/gst/audio/gstbaseaudiosink.c b/gst-libs/gst/audio/gstbaseaudiosink.c
index 984a91d..f440f19 100644 (file)
--- a/gst-libs/gst/audio/gstbaseaudiosink.c
+++ b/gst-libs/gst/audio/gstbaseaudiosink.c
@@ -62,6 +62,8 @@ static GstElementStateReturn gst_base_audio_sink_change_state (GstElement *
 static GstClock *gst_base_audio_sink_get_clock (GstElement * elem);
 static GstClockTime gst_base_audio_sink_get_time (GstClock * clock,
     GstBaseAudioSink * sink);
+static void gst_base_audio_sink_callback (GstRingBuffer * rbuf, guint8 * data,
+    guint len, gpointer user_data);
 
 static GstFlowReturn gst_base_audio_sink_preroll (GstBaseSink * bsink,
     GstBuffer * buffer);
@@ -126,8 +128,10 @@ gst_base_audio_sink_init (GstBaseAudioSink * baseaudiosink)
   baseaudiosink->buffer_time = DEFAULT_BUFFER_TIME;
   baseaudiosink->latency_time = DEFAULT_LATENCY_TIME;
 
+
   baseaudiosink->clock = gst_audio_clock_new ("clock",
       (GstAudioClockGetTimeFunc) gst_base_audio_sink_get_time, baseaudiosink);
+
 }
 
 static void
@@ -141,6 +145,10 @@ gst_base_audio_sink_dispose (GObject * object)
     gst_object_unref (sink->clock);
   sink->clock = NULL;
 
+  if (sink->ringbuffer)
+    gst_object_unref (sink->ringbuffer);
+  sink->ringbuffer = NULL;
+
   G_OBJECT_CLASS (parent_class)->dispose (object);
 }
 
@@ -161,7 +169,7 @@ gst_base_audio_sink_get_time (GstClock * clock, GstBaseAudioSink * sink)
   GstClockTime result;
 
   if (sink->ringbuffer == NULL || sink->ringbuffer->spec.rate == 0)
-    return 0;
+    return GST_CLOCK_TIME_NONE;
 
   /* our processed samples are always increasing */
   samples = gst_ring_buffer_samples_done (sink->ringbuffer);
@@ -288,6 +296,8 @@ gst_base_audio_sink_event (GstBaseSink * bsink, GstEvent * event)
         gst_ring_buffer_pause (sink->ringbuffer);
       }
       break;
+    case GST_EVENT_EOS:
+      break;
     default:
       break;
   }
@@ -335,7 +345,7 @@ gst_base_audio_sink_render (GstBaseSink * bsink, GstBuffer * buf)
       GST_TIME_ARGS (time), in_offset, GST_TIME_ARGS (bsink->discont_start));
 
   /* samples should be rendered based on their timestamp. All samples
-   * arriving before the discont_start are to be trown away */
+   * arriving before the discont_start are to be thrown away */
   /* FIXME, for now we drop the sample completely, we should
    * in fact clip the sample. Same for the segment_stop, actually. */
   if (time < bsink->discont_start)
@@ -369,10 +379,10 @@ gst_base_audio_sink_render (GstBaseSink * bsink, GstBuffer * buf)
 
 wrong_state:
   {
-    GST_DEBUG ("ringbuffer in wrong state");
+    GST_DEBUG ("ringbuffer not negotiated");
     GST_ELEMENT_ERROR (sink, RESOURCE, NOT_FOUND,
-        ("sink not negotiated."), (NULL));
-    return GST_FLOW_ERROR;
+        ("sink not negotiated."), ("sink not negotiated."));
+    return GST_FLOW_NOT_NEGOTIATED;
   }
 }
 
@@ -386,14 +396,13 @@ gst_base_audio_sink_create_ringbuffer (GstBaseAudioSink * sink)
   if (bclass->create_ringbuffer)
     buffer = bclass->create_ringbuffer (sink);
 
-  if (buffer) {
+  if (buffer)
     gst_object_set_parent (GST_OBJECT (buffer), GST_OBJECT (sink));
-  }
 
   return buffer;
 }
 
-void
+static void
 gst_base_audio_sink_callback (GstRingBuffer * rbuf, guint8 * data, guint len,
     gpointer user_data)
 {
@@ -411,9 +420,11 @@ gst_base_audio_sink_change_state (GstElement * element)
     case GST_STATE_NULL_TO_READY:
       break;
     case GST_STATE_READY_TO_PAUSED:
-      sink->ringbuffer = gst_base_audio_sink_create_ringbuffer (sink);
-      gst_ring_buffer_set_callback (sink->ringbuffer,
-          gst_base_audio_sink_callback, sink);
+      if (sink->ringbuffer == NULL) {
+        sink->ringbuffer = gst_base_audio_sink_create_ringbuffer (sink);
+        gst_ring_buffer_set_callback (sink->ringbuffer,
+            gst_base_audio_sink_callback, sink);
+      }
       break;
     case GST_STATE_PAUSED_TO_PLAYING:
       break;
@@ -430,8 +441,6 @@ gst_base_audio_sink_change_state (GstElement * element)
     case GST_STATE_PAUSED_TO_READY:
       gst_ring_buffer_stop (sink->ringbuffer);
       gst_ring_buffer_release (sink->ringbuffer);
-      gst_object_unref (sink->ringbuffer);
-      sink->ringbuffer = NULL;
       gst_pad_set_caps (GST_BASE_SINK_PAD (sink), NULL);
       break;
     case GST_STATE_READY_TO_NULL: