mirrors/gstreamer
mirrors/gstreamer
1
1
Fork 0
mirror of https://gitlab.freedesktop.org/gstreamer/gstreamer.git synced 2024-09-30 07:42:32 +00:00
Code Issues 1 Projects Releases Wiki Activity

audiobasesink: Don't render a GAP silence buffer

Browse source 
Don't render out silence samples to a buffer, just
start the clock running, since any buffer with the
GAP flag will be discarded in render() now anyway.
This commit is contained in:
Jan Schmidt 2015-01-28 05:26:06 +11:00
parent 1df69786c3
commit efe54e50e9
1 changed files with 5 additions and 35 deletions
Show stats Download patch file Download diff file Expand all files Collapse all files

40
gst-libs/gst/audio/gstaudiobasesink.c
View file

@ -1033,6 +1033,8 @@ gst_audio_base_sink_wait_event (GstBaseSink * bsink, GstEvent * event)
GstAudioBaseSink *sink = GST_AUDIO_BASE_SINK (bsink);
GstFlowReturn ret;
/* For both gap and EOS events, make sure the ringbuffer is running
* before trying to wait on the event! */
switch (GST_EVENT_TYPE (event)) {
case GST_EVENT_EOS:
case GST_EVENT_GAP:
@ -1055,41 +1057,9 @@ gst_audio_base_sink_wait_event (GstBaseSink * bsink, GstEvent * event)
switch (GST_EVENT_TYPE (event)) {
case GST_EVENT_GAP:{
GstClockTime timestamp, duration;
GstAudioRingBufferSpec *spec;
GstBuffer *buffer;
gint n_samples = 0;
GstMapInfo minfo;
spec = &sink->ringbuffer->spec;
gst_event_parse_gap (event, &timestamp, &duration);
/* If the GAP event has a duration, handle it like a
* silence buffer of that duration. Otherwise at least
* start the ringbuffer to make sure the clock is running.
*/
if (duration != GST_CLOCK_TIME_NONE) {
n_samples =
gst_util_uint64_scale_ceil (duration, spec->info.rate, GST_SECOND);
buffer = gst_buffer_new_and_alloc (n_samples * spec->info.bpf);
if (n_samples != 0) {
if (!gst_buffer_map (buffer, &minfo, GST_MAP_WRITE))
return FALSE;
gst_audio_format_fill_silence (spec->info.finfo, minfo.data,
minfo.size);
gst_buffer_unmap (buffer, &minfo);
}
GST_BUFFER_PTS (buffer) = timestamp;
GST_BUFFER_DURATION (buffer) = duration;
GST_BUFFER_FLAG_SET (buffer, GST_BUFFER_FLAG_GAP);
ret = gst_audio_base_sink_render (bsink, buffer);
gst_buffer_unref (buffer);
} else {
gst_audio_base_sink_drain (sink);
}
GstClockTime ts, dur;
gst_event_parse_gap (event, &ts, &dur);
gst_audio_base_sink_render_samples (bsink, NULL, ts, dur);
break;
}
case GST_EVENT_EOS: