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decklinkaudiosrc: Calculate the duration more accurately from the capture time and numbers of samples

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This should prevent any accumulating rounding errors with the duration.
This commit is contained in:
Sebastian Dröge 2015-03-04 16:13:30 +01:00
parent 912e58c64c
commit 254337365a
1 changed files with 7 additions and 4 deletions
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11
sys/decklink/gstdecklinkaudiosrc.cpp
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@ -499,16 +499,19 @@ gst_decklink_audio_src_create (GstPushSrc * bsrc, GstBuffer ** buffer)
ap->input->AddRef (); ap->input->AddRef ();
timestamp = p->capture_time; timestamp = p->capture_time;
duration =
gst_util_uint64_scale_int (sample_count, GST_SECOND, self->info.rate);
// Jitter and discontinuity handling, based on audiobasesrc // Jitter and discontinuity handling, based on audiobasesrc
start_time = timestamp; start_time = timestamp;
end_time = p->capture_time + duration;
// Convert to the sample numbers // Convert to the sample numbers
start_offset = gst_util_uint64_scale (start_time, self->info.rate, GST_SECOND); start_offset =
gst_util_uint64_scale (start_time, self->info.rate, GST_SECOND);
end_offset = start_offset + sample_count; end_offset = start_offset + sample_count;
end_time = gst_util_uint64_scale_int (end_offset, GST_SECOND,
self->info.rate);
duration = end_time - start_time;
if (self->next_offset == (guint64) - 1) { if (self->next_offset == (guint64) - 1) {
discont = TRUE; discont = TRUE;