ap->input->AddRef ();
timestamp = p->capture_time;
- duration =
- gst_util_uint64_scale_int (sample_count, GST_SECOND, self->info.rate);
// Jitter and discontinuity handling, based on audiobasesrc
start_time = timestamp;
- end_time = p->capture_time + duration;
// 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_time = gst_util_uint64_scale_int (end_offset, GST_SECOND,
+ self->info.rate);
+
+ duration = end_time - start_time;
if (self->next_offset == (guint64) - 1) {
discont = TRUE;