mirrors/gstreamer
mirrors/gstreamer
1
1
Fork 0
mirror of https://gitlab.freedesktop.org/gstreamer/gstreamer.git synced 2024-11-08 10:31:05 +00:00
Code Issues 1 Projects Releases Wiki Activity

audiomixer: Add locking to fill_buffer and fix mix_buffer

Browse source 
The audiomixer pad struct fields may be changed from other threads
This commit is contained in:
Nirbheek Chauhan 2015-03-12 01:49:58 +05:30 committed by Sebastian Dröge
parent 8227310d22
commit 4e221b7a65
1 changed files with 13 additions and 4 deletions
Show stats Download patch file Download diff file Expand all files Collapse all files

17
gst/audiomixer/gstaudiomixer.c
View file

@ -675,7 +675,6 @@ gst_audiomixer_sink_event (GstAggregator * agg, GstAggregatorPad * aggpad,
}
case GST_EVENT_SEGMENT:
{
GstAudioMixerPad *pad = GST_AUDIO_MIXER_PAD (aggpad);
const GstSegment *segment;
gst_event_parse_segment (event, &segment);
@ -692,9 +691,13 @@ gst_audiomixer_sink_event (GstAggregator * agg, GstAggregatorPad * aggpad,
gst_event_unref (event);
event = NULL;
} else {
GstAudioMixerPad *pad = GST_AUDIO_MIXER_PAD (aggpad);
/* Ideally, this should only be set when the new segment causes running
* times to change, and hence needs discont calculation in fill_buffer */
GST_OBJECT_LOCK (pad);
pad->new_segment = TRUE;
GST_OBJECT_UNLOCK (pad);
}
break;
}
@ -991,7 +994,7 @@ gst_audio_mixer_fill_buffer (GstAudioMixer * audiomixer, GstAudioMixerPad * pad,
rate = GST_AUDIO_INFO_RATE (&audiomixer->info);
bpf = GST_AUDIO_INFO_BPF (&audiomixer->info);
timestamp = GST_BUFFER_TIMESTAMP (inbuf);
timestamp = GST_BUFFER_PTS (inbuf);
stream_time = gst_segment_to_stream_time (&aggpad->segment, GST_FORMAT_TIME,
timestamp);
@ -1000,6 +1003,7 @@ gst_audio_mixer_fill_buffer (GstAudioMixer * audiomixer, GstAudioMixerPad * pad,
if (GST_CLOCK_TIME_IS_VALID (stream_time))
gst_object_sync_values (GST_OBJECT (pad), stream_time);
GST_OBJECT_LOCK (pad);
pad->position = 0;
pad->size = gst_buffer_get_size (inbuf);
@ -1088,6 +1092,7 @@ gst_audio_mixer_fill_buffer (GstAudioMixer * audiomixer, GstAudioMixerPad * pad,
GST_DEBUG_OBJECT (pad,
"Buffer before segment or current position: %" G_GUINT64_FORMAT " < %"
G_GUINT64_FORMAT, end_running_time_offset, audiomixer->offset);
GST_OBJECT_UNLOCK (pad);
return FALSE;
}
@ -1107,6 +1112,7 @@ gst_audio_mixer_fill_buffer (GstAudioMixer * audiomixer, GstAudioMixerPad * pad,
"Buffer before segment or current position: %" G_GUINT64_FORMAT
" < %" G_GUINT64_FORMAT, end_running_time_offset,
audiomixer->offset);
GST_OBJECT_UNLOCK (pad);
return FALSE;
}
}
@ -1122,6 +1128,7 @@ gst_audio_mixer_fill_buffer (GstAudioMixer * audiomixer, GstAudioMixerPad * pad,
"Queued new buffer at offset %" G_GUINT64_FORMAT, pad->output_offset);
pad->buffer = inbuf;
GST_OBJECT_UNLOCK (pad);
return TRUE;
}
@ -1145,6 +1152,7 @@ gst_audio_mixer_mix_buffer (GstAudioMixer * audiomixer, GstAudioMixerPad * pad,
bpf = GST_AUDIO_INFO_BPF (&audiomixer->info);
GST_OBJECT_LOCK (pad);
/* Overlap => mix */
if (audiomixer->offset < pad->output_offset)
out_start = pad->output_offset - audiomixer->offset;
@ -1156,10 +1164,11 @@ gst_audio_mixer_mix_buffer (GstAudioMixer * audiomixer, GstAudioMixerPad * pad,
overlap = blocksize - out_start;
inbuf = gst_aggregator_pad_get_buffer (aggpad);
if (inbuf == NULL)
if (inbuf == NULL) {
GST_OBJECT_UNLOCK (pad);
return;
}
GST_OBJECT_LOCK (pad);
if (pad->mute || pad->volume < G_MINDOUBLE) {
GST_DEBUG_OBJECT (pad, "Skipping muted pad");
gst_buffer_unref (inbuf);