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audiomixer: Add locking to fill_buffer and fix mix_buffer

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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
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17
gst/audiomixer/gstaudiomixer.c
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@ -675,7 +675,6 @@ gst_audiomixer_sink_event (GstAggregator * agg, GstAggregatorPad * aggpad,
} }
case GST_EVENT_SEGMENT: case GST_EVENT_SEGMENT:
{ {
GstAudioMixerPad *pad = GST_AUDIO_MIXER_PAD (aggpad);
const GstSegment *segment; const GstSegment *segment;
gst_event_parse_segment (event, &segment); gst_event_parse_segment (event, &segment);
@ -692,9 +691,13 @@ gst_audiomixer_sink_event (GstAggregator * agg, GstAggregatorPad * aggpad,
gst_event_unref (event); gst_event_unref (event);
event = NULL; event = NULL;
} else { } else {
GstAudioMixerPad *pad = GST_AUDIO_MIXER_PAD (aggpad);
/* Ideally, this should only be set when the new segment causes running /* Ideally, this should only be set when the new segment causes running
* times to change, and hence needs discont calculation in fill_buffer */ * times to change, and hence needs discont calculation in fill_buffer */
GST_OBJECT_LOCK (pad);
pad->new_segment = TRUE; pad->new_segment = TRUE;
GST_OBJECT_UNLOCK (pad);
} }
break; break;
} }
@ -991,7 +994,7 @@ gst_audio_mixer_fill_buffer (GstAudioMixer * audiomixer, GstAudioMixerPad * pad,
rate = GST_AUDIO_INFO_RATE (&audiomixer->info); rate = GST_AUDIO_INFO_RATE (&audiomixer->info);
bpf = GST_AUDIO_INFO_BPF (&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, stream_time = gst_segment_to_stream_time (&aggpad->segment, GST_FORMAT_TIME,
timestamp); timestamp);
@ -1000,6 +1003,7 @@ gst_audio_mixer_fill_buffer (GstAudioMixer * audiomixer, GstAudioMixerPad * pad,
if (GST_CLOCK_TIME_IS_VALID (stream_time)) if (GST_CLOCK_TIME_IS_VALID (stream_time))
gst_object_sync_values (GST_OBJECT (pad), stream_time); gst_object_sync_values (GST_OBJECT (pad), stream_time);
GST_OBJECT_LOCK (pad);
pad->position = 0; pad->position = 0;
pad->size = gst_buffer_get_size (inbuf); pad->size = gst_buffer_get_size (inbuf);
@ -1088,6 +1092,7 @@ gst_audio_mixer_fill_buffer (GstAudioMixer * audiomixer, GstAudioMixerPad * pad,
GST_DEBUG_OBJECT (pad, GST_DEBUG_OBJECT (pad,
"Buffer before segment or current position: %" G_GUINT64_FORMAT " < %" "Buffer before segment or current position: %" G_GUINT64_FORMAT " < %"
G_GUINT64_FORMAT, end_running_time_offset, audiomixer->offset); G_GUINT64_FORMAT, end_running_time_offset, audiomixer->offset);
GST_OBJECT_UNLOCK (pad);
return FALSE; return FALSE;
} }
@ -1107,6 +1112,7 @@ gst_audio_mixer_fill_buffer (GstAudioMixer * audiomixer, GstAudioMixerPad * pad,
"Buffer before segment or current position: %" G_GUINT64_FORMAT "Buffer before segment or current position: %" G_GUINT64_FORMAT
" < %" G_GUINT64_FORMAT, end_running_time_offset, " < %" G_GUINT64_FORMAT, end_running_time_offset,
audiomixer->offset); audiomixer->offset);
GST_OBJECT_UNLOCK (pad);
return FALSE; 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); "Queued new buffer at offset %" G_GUINT64_FORMAT, pad->output_offset);
pad->buffer = inbuf; pad->buffer = inbuf;
GST_OBJECT_UNLOCK (pad);
return TRUE; return TRUE;
} }
@ -1145,6 +1152,7 @@ gst_audio_mixer_mix_buffer (GstAudioMixer * audiomixer, GstAudioMixerPad * pad,
bpf = GST_AUDIO_INFO_BPF (&audiomixer->info); bpf = GST_AUDIO_INFO_BPF (&audiomixer->info);
GST_OBJECT_LOCK (pad);
/* Overlap => mix */ /* Overlap => mix */
if (audiomixer->offset < pad->output_offset) if (audiomixer->offset < pad->output_offset)
out_start = pad->output_offset - audiomixer->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; overlap = blocksize - out_start;
inbuf = gst_aggregator_pad_get_buffer (aggpad); inbuf = gst_aggregator_pad_get_buffer (aggpad);
if (inbuf == NULL) if (inbuf == NULL) {
GST_OBJECT_UNLOCK (pad);
return; return;
}
GST_OBJECT_LOCK (pad);
if (pad->mute || pad->volume < G_MINDOUBLE) { if (pad->mute || pad->volume < G_MINDOUBLE) {
GST_DEBUG_OBJECT (pad, "Skipping muted pad"); GST_DEBUG_OBJECT (pad, "Skipping muted pad");
gst_buffer_unref (inbuf); gst_buffer_unref (inbuf);