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audiobuffersplit: Use input running time for comparison instead of the currently tracked running time

Browse source 
Otherwise gapless mode would do completely wrong calculations on
discontinuities and cause input/output to drift slowly.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2780>
This commit is contained in:
Sebastian Dröge 2022-07-21 16:10:18 +03:00 committed by GStreamer Marge Bot
parent 0485c354d2
commit fad0a72eca
1 changed files with 3 additions and 3 deletions
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6
subprojects/gst-plugins-bad/gst/audiobuffersplit/gstaudiobuffersplit.c
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@ -549,14 +549,14 @@ gst_audio_buffer_split_handle_discont (GstAudioBufferSplit * self,
GST_TIME_ARGS (current_rt_end), GST_TIME_ARGS (input_rt)); GST_TIME_ARGS (current_rt_end), GST_TIME_ARGS (input_rt));
new_offset = new_offset =
gst_util_uint64_scale (current_rt - self->resync_rt, gst_util_uint64_scale (input_rt - self->resync_rt,
rate * ABS (self->in_segment.rate), GST_SECOND); rate * ABS (self->in_segment.rate), GST_SECOND);
if (current_rt < self->resync_rt) { if (input_rt < self->resync_rt) {
guint64 drop_samples; guint64 drop_samples;
new_offset = new_offset =
gst_util_uint64_scale (self->resync_rt - gst_util_uint64_scale (self->resync_rt -
current_rt, rate * ABS (self->in_segment.rate), GST_SECOND); input_rt, rate * ABS (self->in_segment.rate), GST_SECOND);
drop_samples = self->current_offset + avail_samples + new_offset; drop_samples = self->current_offset + avail_samples + new_offset;
GST_DEBUG_OBJECT (self, GST_DEBUG_OBJECT (self,