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| author | Edward Hervey <edward.hervey@collabora.co.uk> | 2011-12-30 11:41:17 +0100 |
|---|---|---|
| committer | Edward Hervey <edward.hervey@collabora.co.uk> | 2011-12-30 11:41:17 +0100 |
| commit | f70a623418a52eada0278002f8d266d49957b3d5 (patch) | |
| tree | fc54fd9fab400e886b11a9a2b313478b66dd7969 /ext/dts/gstdtsdec.c | |
| parent | d5aaefa59f9ef4153644a2aa254c39a3a9d108e3 (diff) | |
| parent | d465188879cd4dd0735e3fbcaeb83d98f217ed88 (diff) | |
| download | gstreamer-plugins-bad-f70a623418a52eada0278002f8d266d49957b3d5.tar.gz | |
Merge remote-tracking branch 'origin/master' into 0.11-premerge
Conflicts:
docs/libs/Makefile.am
ext/kate/gstkatetiger.c
ext/opus/gstopusdec.c
ext/xvid/gstxvidenc.c
gst-libs/gst/basecamerabinsrc/Makefile.am
gst-libs/gst/basecamerabinsrc/gstbasecamerasrc.c
gst-libs/gst/basecamerabinsrc/gstbasecamerasrc.h
gst-libs/gst/video/gstbasevideocodec.c
gst-libs/gst/video/gstbasevideocodec.h
gst-libs/gst/video/gstbasevideodecoder.c
gst-libs/gst/video/gstbasevideoencoder.c
gst/asfmux/gstasfmux.c
gst/audiovisualizers/gstwavescope.c
gst/camerabin2/gstcamerabin2.c
gst/debugutils/gstcompare.c
gst/frei0r/gstfrei0rmixer.c
gst/mpegpsmux/mpegpsmux.c
gst/mpegtsmux/mpegtsmux.c
gst/mxf/mxfmux.c
gst/videomeasure/gstvideomeasure_ssim.c
gst/videoparsers/gsth264parse.c
gst/videoparsers/gstmpeg4videoparse.c
Diffstat (limited to 'ext/dts/gstdtsdec.c')
| -rw-r--r-- | ext/dts/gstdtsdec.c | 9 |
1 files changed, 7 insertions, 2 deletions
diff --git a/ext/dts/gstdtsdec.c b/ext/dts/gstdtsdec.c index 2a762e903..f71219478 100644 --- a/ext/dts/gstdtsdec.c +++ b/ext/dts/gstdtsdec.c @@ -494,11 +494,15 @@ gst_dtsdec_handle_frame (GstAudioDecoder * bdec, GstBuffer * buffer) guint8 *data; gint size, chans; gint length = 0, flags, sample_rate, bit_rate, frame_length; - GstFlowReturn result = GST_FLOW_UNEXPECTED; + GstFlowReturn result = GST_FLOW_OK; GstBuffer *outbuf; dts = GST_DTSDEC (bdec); + /* no fancy draining */ + if (G_UNLIKELY (!buffer)) + return GST_FLOW_OK; + /* parsed stuff already, so this should work out fine */ data = GST_BUFFER_DATA (buffer); size = GST_BUFFER_SIZE (buffer); @@ -599,7 +603,8 @@ gst_dtsdec_handle_frame (GstAudioDecoder * bdec, GstBuffer * buffer) /* negotiate if required */ if (need_renegotiation) { - GST_DEBUG ("dtsdec: sample_rate:%d stream_chans:0x%x using_chans:0x%x", + GST_DEBUG_OBJECT (dts, + "dtsdec: sample_rate:%d stream_chans:0x%x using_chans:0x%x", dts->sample_rate, dts->stream_channels, dts->using_channels); if (!gst_dtsdec_renegotiate (dts)) goto failed_negotiation; |