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decklinkaudiosrc: Don't subtract the duration from the capture time

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We already have the real capture time, not the time when we received
the end of the packet.
This commit is contained in:
Sebastian Dröge 2015-03-04 16:04:18 +01:00
parent 44c913413c
commit 1f9d37c924
1 changed files with 3 additions and 6 deletions
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9
sys/decklink/gstdecklinkaudiosrc.cpp
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@ -427,6 +427,8 @@ gst_decklink_audio_src_got_packet (GstElement * element,
gst_decklink_video_src_convert_to_external_clock (videosrc, &capture_time, gst_decklink_video_src_convert_to_external_clock (videosrc, &capture_time,
NULL); NULL);
gst_object_unref (videosrc); gst_object_unref (videosrc);
GST_LOG_OBJECT (self, "Actual timestamp %" GST_TIME_FORMAT,
GST_TIME_ARGS (capture_time));
} }
g_mutex_lock (&self->lock); g_mutex_lock (&self->lock);
@ -496,14 +498,9 @@ gst_decklink_audio_src_create (GstPushSrc * bsrc, GstBuffer ** buffer)
ap->input = self->input->input; ap->input = self->input->input;
ap->input->AddRef (); ap->input->AddRef ();
timestamp = p->capture_time;
duration = duration =
gst_util_uint64_scale_int (sample_count, GST_SECOND, self->info.rate); gst_util_uint64_scale_int (sample_count, GST_SECOND, self->info.rate);
// Our capture time is the end timestamp, subtract the
// duration to get the start timestamp
if (p->capture_time >= duration)
timestamp = p->capture_time - duration;
else
timestamp = 0;
// Jitter and discontinuity handling, based on audiobasesrc // Jitter and discontinuity handling, based on audiobasesrc
start_time = timestamp; start_time = timestamp;