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audiobuffersplit: Keep incoming and outgoing segments separate

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We might have to drain already queued input based on the old segment
before forwarding the new segment event. The new segment is only
forwarded after a discont as otherwise we might cause unnecessary
timestamp jumps as we output buffers timestamped based on sample counts.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1254>
This commit is contained in:
Sebastian Dröge 2020-05-08 21:36:44 +03:00 committed by GStreamer Merge Bot
parent 9937101e51
commit 20756e3387
2 changed files with 35 additions and 11 deletions
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42
gst/audiobuffersplit/gstaudiobuffersplit.c
View file

@ -321,7 +321,9 @@ gst_audio_buffer_split_change_state (GstElement * element,
switch (transition) {
case GST_STATE_CHANGE_READY_TO_PAUSED:
gst_audio_info_init (&self->info);
gst_segment_init (&self->segment, GST_FORMAT_TIME);
gst_segment_init (&self->in_segment, GST_FORMAT_TIME);
gst_segment_init (&self->out_segment, GST_FORMAT_UNDEFINED);
self->segment_pending = FALSE;
GST_OBJECT_LOCK (self);
gst_audio_stream_align_mark_discont (self->stream_align);
GST_OBJECT_UNLOCK (self);
@ -384,7 +386,7 @@ gst_audio_buffer_split_output (GstAudioBufferSplit * self, gboolean force,
resync_time_diff =
gst_util_uint64_scale (self->current_offset, GST_SECOND, rate);
if (self->segment.rate < 0.0) {
if (self->out_segment.rate < 0.0) {
if (resync_time > resync_time_diff)
GST_BUFFER_TIMESTAMP (buffer) = resync_time - resync_time_diff;
else
@ -446,7 +448,7 @@ gst_audio_buffer_split_handle_discont (GstAudioBufferSplit * self,
GST_OBJECT_LOCK (self);
discont =
gst_audio_stream_align_process (self->stream_align,
self->segment.rate < 0 ? FALSE : GST_BUFFER_IS_DISCONT (buffer)
self->in_segment.rate < 0 ? FALSE : GST_BUFFER_IS_DISCONT (buffer)
|| GST_BUFFER_FLAG_IS_SET (buffer, GST_BUFFER_FLAG_RESYNC),
GST_BUFFER_PTS (buffer), gst_buffer_get_size (buffer) / bpf, NULL, NULL,
NULL);
@ -458,7 +460,7 @@ gst_audio_buffer_split_handle_discont (GstAudioBufferSplit * self,
/* Reset */
self->drop_samples = 0;
if (self->segment.rate < 0.0) {
if (self->in_segment.rate < 0.0) {
current_timestamp =
self->resync_time - gst_util_uint64_scale (self->current_offset +
avail_samples, GST_SECOND, rate);
@ -584,6 +586,18 @@ gst_audio_buffer_split_handle_discont (GstAudioBufferSplit * self,
self->current_offset = 0;
self->accumulated_error = 0;
self->resync_time = GST_BUFFER_PTS (buffer);
if (self->segment_pending) {
GstEvent *event;
self->out_segment = self->in_segment;
GST_DEBUG_OBJECT (self, "Updating output segment %" GST_SEGMENT_FORMAT,
&self->out_segment);
event = gst_event_new_segment (&self->out_segment);
gst_event_set_seqnum (event, self->segment_seqnum);
gst_pad_push_event (self->srcpad, event);
self->segment_pending = FALSE;
}
}
return ret;
@ -616,7 +630,7 @@ gst_audio_buffer_split_clip_buffer_start_for_gapless (GstAudioBufferSplit *
return NULL;
}
if (self->segment.rate < 0.0) {
if (self->out_segment.rate < 0.0) {
buffer =
gst_audio_buffer_truncate (buffer, bpf, 0,
nsamples - self->drop_samples);
@ -655,7 +669,7 @@ gst_audio_buffer_split_sink_chain (GstPad * pad, GstObject * parent,
}
buffer =
gst_audio_buffer_split_clip_buffer (self, buffer, &self->segment, rate,
gst_audio_buffer_split_clip_buffer (self, buffer, &self->in_segment, rate,
bpf);
if (!buffer)
return GST_FLOW_OK;
@ -736,7 +750,9 @@ gst_audio_buffer_split_sink_event (GstPad * pad, GstObject * parent,
break;
}
case GST_EVENT_FLUSH_STOP:
gst_segment_init (&self->segment, GST_FORMAT_TIME);
gst_segment_init (&self->in_segment, GST_FORMAT_TIME);
gst_segment_init (&self->out_segment, GST_FORMAT_UNDEFINED);
self->segment_pending = FALSE;
GST_OBJECT_LOCK (self);
gst_audio_stream_align_mark_discont (self->stream_align);
GST_OBJECT_UNLOCK (self);
@ -746,12 +762,18 @@ gst_audio_buffer_split_sink_event (GstPad * pad, GstObject * parent,
ret = gst_pad_event_default (pad, parent, event);
break;
case GST_EVENT_SEGMENT:
gst_event_copy_segment (event, &self->segment);
if (self->segment.format != GST_FORMAT_TIME) {
gst_event_copy_segment (event, &self->in_segment);
if (self->in_segment.format != GST_FORMAT_TIME) {
gst_event_unref (event);
ret = FALSE;
} else {
ret = gst_pad_event_default (pad, parent, event);
GST_DEBUG_OBJECT (self,
"Received new input segment %" GST_SEGMENT_FORMAT,
&self->in_segment);
self->segment_pending = TRUE;
self->segment_seqnum = gst_event_get_seqnum (event);
gst_event_unref (event);
ret = TRUE;
}
break;
case GST_EVENT_EOS:

4
gst/audiobuffersplit/gstaudiobuffersplit.h
View file

@ -47,7 +47,9 @@ struct _GstAudioBufferSplit {
gint output_buffer_duration_d;
/* State */
GstSegment segment;
GstSegment in_segment, out_segment;
guint32 segment_seqnum;
gboolean segment_pending;
GstAudioInfo info;
GstAdapter *adapter;