gst/matroska/matroska-demux.c: Don't calculate the default duration of a frame from the audio sampling rate. This onl...

Original commit message from CVS:
* gst/matroska/matroska-demux.c: (gst_matroska_demux_add_stream):
Don't calculate the default duration of a frame from the audio sampling
rate. This only works for raw audio if every frame contains a single
sample and results in broken buffer durations for other formats
if no specified default duration is given or the blocks have no
duration. Fixes bug #548831.

3 files changed, 9 insertions, 4 deletions

diff --git a/ChangeLog b/ChangeLog
index 4c3d32f1..325926b1 100644
--- a/ChangeLog
+++ b/ChangeLog
@@ -1,5 +1,14 @@
 2008-08-21  Sebastian Dröge  <sebastian.droege@collabora.co.uk>
 
+	* gst/matroska/matroska-demux.c: (gst_matroska_demux_add_stream):
+	Don't calculate the default duration of a frame from the audio sampling
+	rate. This only works for raw audio if every frame contains a single
+	sample and results in broken buffer durations for other formats
+	if no specified default duration is given or the blocks have no
+	duration. Fixes bug #548831.
+
+2008-08-21  Sebastian Dröge  <sebastian.droege@collabora.co.uk>
+
 	* gst/matroska/matroska-demux.c:
 	(gst_matroska_demux_parse_blockgroup_or_simpleblock):
 	Allow zero sized blocks instead of returning GST_FLOW_OK. Such blocks
diff --git a/common b/common
-Subproject d70ca17ae6fbe6020996e4567275d5e14972ed4
+Subproject 8d494854a6018336a80ece82ceb3df0033e2da9
diff --git a/gst/matroska/matroska-demux.c b/gst/matroska/matroska-demux.c
index 0179b4ed..8e0193d9 100644
--- a/gst/matroska/matroska-demux.c
+++ b/gst/matroska/matroska-demux.c
@@ -1391,10 +1391,6 @@ gst_matroska_demux_add_stream (GstMatroskaDemux * demux)
                 break;
               }
 
-              if (context->default_duration == 0)
-                context->default_duration =
-                    gst_gdouble_to_guint64 ((gdouble) GST_SECOND * (1.0 / num));
-
               GST_DEBUG_OBJECT (demux, "TrackAudioSamplingFrequency: %lf", num);
               audiocontext->samplerate = num;
               break;