wtay gst-plugins-base: gst-plugins-base/ gst-plugins-base/gst-libs/gst/audio/

CVS Root:       /cvs/gstreamer
Module:         gst-plugins-base
Changes by:     wtay
Date:           Tue Nov 25 2008  10:33:04 UTC

Log message:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_drain), (gst_base_audio_sink_event),
(gst_base_audio_sink_sync_latency), (gst_base_audio_sink_render),
(gst_base_audio_sink_change_state):
Really fix audiosink drain handling by keeping track of the running_time
of the last sample.

Modified files:
    .               : ChangeLog
    gst-libs/gst/audio: gstbaseaudiosink.c

Links:
http://freedesktop.org/cgi-bin/viewcvs.cgi/gstreamer/gst-plugins-base/ChangeLog.diff?r1=1.4213&r2=1.4214
http://freedesktop.org/cgi-bin/viewcvs.cgi/gstreamer/gst-plugins-base/gst-libs/gst/audio/gstbaseaudiosink.c.diff?r1=1.110&r2=1.111

====Begin Diffs====
Index: ChangeLog
===================================================================
RCS file: /cvs/gstreamer/gst-plugins-base/ChangeLog,v
retrieving revision 1.4213
retrieving revision 1.4214
diff -u -d -r1.4213 -r1.4214
--- ChangeLog	24 Nov 2008 20:25:23 -0000	1.4213
+++ ChangeLog	25 Nov 2008 10:32:47 -0000	1.4214
@@ -1,3 +1,12 @@
+2008-11-25  Wim Taymans  <[email protected]>
+
+	* gst-libs/gst/audio/gstbaseaudiosink.c:
+	(gst_base_audio_sink_drain), (gst_base_audio_sink_event),
+	(gst_base_audio_sink_sync_latency), (gst_base_audio_sink_render),
+	(gst_base_audio_sink_change_state):
+	Really fix audiosink drain handling by keeping track of the running_time
+	of the last sample.
 2008-11-24  Michael Smith <[email protected]>
 
 	* gst/playback/gstplaybin2.c:
Index: gstbaseaudiosink.c
RCS file: /cvs/gstreamer/gst-plugins-base/gst-libs/gst/audio/gstbaseaudiosink.c,v
retrieving revision 1.110
retrieving revision 1.111
diff -u -d -r1.110 -r1.111
--- gstbaseaudiosink.c	24 Nov 2008 20:11:52 -0000	1.110
+++ gstbaseaudiosink.c	25 Nov 2008 10:32:49 -0000	1.111
@@ -54,6 +54,8 @@
   gint64 last_align;
   gboolean sync_latency;
+  GstClockTime eos_time;
 };
 /* BaseAudioSink signals and args */
@@ -720,23 +722,15 @@
   if (gst_ring_buffer_is_acquired (sink->ringbuffer))
     gst_ring_buffer_start (sink->ringbuffer);
-  if (sink->next_sample != -1) {
-    GstClockTime time;
-
-    /* convert next expected sample to time */
-    time =
-        gst_util_uint64_scale_int (sink->next_sample, GST_SECOND,
-        sink->ringbuffer->spec.rate);
+  if (sink->priv->eos_time != -1) {
     GST_DEBUG_OBJECT (sink,
-        "last sample %" G_GUINT64_FORMAT ", time %" GST_TIME_FORMAT,
-        sink->next_sample, GST_TIME_ARGS (time));
+        "last sample time %" GST_TIME_FORMAT,
+        GST_TIME_ARGS (sink->priv->eos_time));
     /* wait for the EOS time to be reached, this is the time when the last
      * sample is played. */
-    gst_base_sink_wait_eos (GST_BASE_SINK (sink), time, NULL);
+    gst_base_sink_wait_eos (GST_BASE_SINK (sink), sink->priv->eos_time, NULL);
-    sink->next_sample = -1;
     GST_DEBUG_OBJECT (sink, "drained audio");
   }
   return TRUE;
@@ -756,6 +750,7 @@
       /* always resync on sample after a flush */
       sink->priv->avg_skew = -1;
       sink->next_sample = -1;
+      sink->priv->eos_time = -1;
       if (sink->ringbuffer)
         gst_ring_buffer_set_flushing (sink->ringbuffer, FALSE);
       break;
@@ -1169,6 +1164,7 @@
   sink->priv->avg_skew = -1;
   sink->next_sample = -1;
+  sink->priv->eos_time = -1;
   return GST_FLOW_OK;
@@ -1354,12 +1350,13 @@
       "running: start %" GST_TIME_FORMAT " - stop %" GST_TIME_FORMAT,
       GST_TIME_ARGS (render_start), GST_TIME_ARGS (render_stop));
-  GST_DEBUG_OBJECT (sink, "adding base_time %" GST_TIME_FORMAT,
-      GST_TIME_ARGS (base_time));
-  /* add base time to sync against the clock */
-  render_start += base_time;
-  render_stop += base_time;
+  /* store the time of the last sample, we'll use this to perform sync on the
+   * last sample when draining the buffer */
+  if (bsink->segment.rate >= 0.0) {
+    sink->priv->eos_time = render_stop;
+  } else {
+    sink->priv->eos_time = render_start;
+  }
   /* compensate for ts-offset and delay we know this will not underflow because we
    * clipped above. */
@@ -1369,6 +1366,13 @@
   render_start += sync_offset;
   render_stop += sync_offset;
+  GST_DEBUG_OBJECT (sink, "adding base_time %" GST_TIME_FORMAT,
+      GST_TIME_ARGS (base_time));
+  /* add base time to sync against the clock */
+  render_start += base_time;
+  render_stop += base_time;
   GST_DEBUG_OBJECT (sink,
       "after compensation: start %" GST_TIME_FORMAT " - stop %" GST_TIME_FORMAT,
@@ -1737,6 +1741,7 @@
     case GST_STATE_CHANGE_READY_TO_PAUSED:
       sink->priv->last_align = -1;
       gst_ring_buffer_set_flushing (sink->ringbuffer, FALSE);
       gst_ring_buffer_may_start (sink->ringbuffer, FALSE);

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