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gst-libs/gst/audio/gstbaseaudiosrc.c: Fix duration when no clock was provided. Fixes #520300.

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Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosrc.c:
(gst_base_audio_src_create):
Fix duration when no clock was provided. Fixes #520300.
This commit is contained in:
Wim Taymans 2008-03-10 17:19:56 +00:00
parent cf273d8add
commit 579949e2c5
3 changed files with 18 additions and 11 deletions
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6
ChangeLog
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@ -1,3 +1,9 @@
2008-03-10 Wim Taymans <wim.taymans@collabora.co.uk>
* gst-libs/gst/audio/gstbaseaudiosrc.c:
(gst_base_audio_src_create):
Fix duration when no clock was provided. Fixes #520300.
2008-03-07 Wim Taymans <wim.taymans@collabora.co.uk>
Patch by: Olivier Crete <tester at tester ca>

2
common

@ -1 +1 @@
Subproject commit e02bd43fe6b9e45536eccbf5b7a5f9eae62030fd
Subproject commit 170f8e91adc7157f6e708ffa58ca22d10e4e45da

21
gst-libs/gst/audio/gstbaseaudiosrc.c
View file

@ -598,7 +598,7 @@ gst_base_audio_src_create (GstBaseSrc * bsrc, guint64 offset, guint length,
GstRingBuffer *ringbuffer;
GstRingBufferSpec *spec;
guint read;
GstClockTime timestamp;
GstClockTime timestamp, duration;
GstClock *clock;
ringbuffer = src->ringbuffer;
@ -669,18 +669,21 @@ gst_base_audio_src_create (GstBaseSrc * bsrc, guint64 offset, guint length,
GST_BUFFER_FLAG_SET (buf, GST_BUFFER_FLAG_DISCONT);
}
src->next_sample = sample + samples;
/* get the normal timestamp to get the duration. */
timestamp = gst_util_uint64_scale_int (sample, GST_SECOND, spec->rate);
duration = gst_util_uint64_scale_int (src->next_sample, GST_SECOND,
spec->rate) - timestamp;
GST_OBJECT_LOCK (src);
clock = GST_ELEMENT_CLOCK (src);
if (clock == NULL || clock == src->clock) {
/* timestamp against our own clock. We do this also when no external clock
* was provided to us. */
timestamp = gst_util_uint64_scale_int (sample, GST_SECOND, spec->rate);
} else {
if (clock != NULL && clock != src->clock) {
GstClockTime base_time, latency;
/* We are slaved to another clock, take running time of the clock and just
* timestamp against it. Somebody else in the pipeline should figure out the
* clock drift, for now. */
* clock drift, for now. We keep the duration we calculated above. */
timestamp = gst_clock_get_time (clock);
base_time = GST_ELEMENT_CAST (src)->base_time;
@ -699,9 +702,7 @@ gst_base_audio_src_create (GstBaseSrc * bsrc, guint64 offset, guint length,
GST_OBJECT_UNLOCK (src);
GST_BUFFER_TIMESTAMP (buf) = timestamp;
src->next_sample = sample + samples;
GST_BUFFER_DURATION (buf) = gst_util_uint64_scale_int (src->next_sample,
GST_SECOND, spec->rate) - GST_BUFFER_TIMESTAMP (buf);
GST_BUFFER_DURATION (buf) = duration;
GST_BUFFER_OFFSET (buf) = sample;
GST_BUFFER_OFFSET_END (buf) = sample + samples;