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baseaudiodecoder: Fix thread safety issues if both pads have different streaming threads

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This commit is contained in:
Sebastian Dröge 2011-08-18 09:34:38 +02:00
parent fc511eb45d
commit affd6e2eb9
2 changed files with 53 additions and 14 deletions
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59
omx/gstbaseaudiodecoder.c
View file

@ -370,6 +370,8 @@ gst_base_audio_decoder_init (GstBaseAudioDecoder * dec,
g_queue_init (&dec->priv->frames);
dec->ctx = &dec->priv->ctx;
g_static_rec_mutex_init (&dec->stream_lock);
/* property default */
dec->latency = DEFAULT_LATENCY;
dec->tolerance = DEFAULT_TOLERANCE;
@ -384,7 +386,7 @@ gst_base_audio_decoder_reset (GstBaseAudioDecoder * dec, gboolean full)
{
GST_DEBUG_OBJECT (dec, "gst_base_audio_decoder_reset");
GST_OBJECT_LOCK (dec);
GST_BASE_AUDIO_DECODER_STREAM_LOCK (dec);
if (full) {
dec->priv->active = FALSE;
@ -422,7 +424,7 @@ gst_base_audio_decoder_reset (GstBaseAudioDecoder * dec, gboolean full)
dec->priv->discont = TRUE;
dec->priv->sync_flush = FALSE;
GST_OBJECT_UNLOCK (dec);
GST_BASE_AUDIO_DECODER_STREAM_UNLOCK (dec);
}
static void
@ -440,6 +442,8 @@ gst_base_audio_decoder_finalize (GObject * object)
g_object_unref (dec->priv->adapter_out);
}
g_static_rec_mutex_free (&dec->stream_lock);
G_OBJECT_CLASS (parent_class)->finalize (object);
}
@ -457,6 +461,8 @@ gst_base_audio_decoder_src_setcaps (GstPad * pad, GstCaps * caps)
GST_DEBUG_OBJECT (dec, "setting src caps %" GST_PTR_FORMAT, caps);
GST_BASE_AUDIO_DECODER_STREAM_LOCK (dec);
/* parse caps here to check subclass;
* also makes us aware of output format */
if (!gst_caps_is_fixed (caps))
@ -472,6 +478,9 @@ gst_base_audio_decoder_src_setcaps (GstPad * pad, GstCaps * caps)
if (!gst_base_audio_parse_caps (caps, state, &changed))
goto refuse_caps;
done:
GST_BASE_AUDIO_DECODER_STREAM_UNLOCK (dec);
gst_object_unref (dec);
return res;
@ -479,8 +488,8 @@ gst_base_audio_decoder_src_setcaps (GstPad * pad, GstCaps * caps)
refuse_caps:
{
GST_WARNING_OBJECT (dec, "rejected caps %" GST_PTR_FORMAT, caps);
gst_object_unref (dec);
return res;
res = FALSE;
goto done;
}
}
@ -496,6 +505,7 @@ gst_base_audio_decoder_sink_setcaps (GstPad * pad, GstCaps * caps)
GST_DEBUG_OBJECT (dec, "caps: %" GST_PTR_FORMAT, caps);
GST_BASE_AUDIO_DECODER_STREAM_LOCK (dec);
/* NOTE pbutils only needed here */
/* TODO maybe (only) upstream demuxer/parser etc should handle this ? */
if (dec->priv->taglist)
@ -507,6 +517,8 @@ gst_base_audio_decoder_sink_setcaps (GstPad * pad, GstCaps * caps)
if (klass->set_format)
res = klass->set_format (dec, caps);
GST_BASE_AUDIO_DECODER_STREAM_UNLOCK (dec);
g_object_unref (dec);
return res;
}
@ -680,6 +692,7 @@ gst_base_audio_decoder_finish_frame (GstBaseAudioDecoder * dec, GstBuffer * buf,
GstBaseAudioDecoderContext *ctx;
gint samples = 0;
GstClockTime ts, next_ts;
GstFlowReturn ret = GST_FLOW_OK;
/* subclass should know what it is producing by now */
g_return_val_if_fail (buf == NULL || GST_PAD_CAPS (dec->srcpad) != NULL,
@ -697,13 +710,13 @@ gst_base_audio_decoder_finish_frame (GstBaseAudioDecoder * dec, GstBuffer * buf,
buf ? GST_BUFFER_SIZE (buf) : -1,
buf ? GST_BUFFER_SIZE (buf) / ctx->state.bpf : -1, frames);
GST_BASE_AUDIO_DECODER_STREAM_LOCK (dec);
if (priv->pending_events) {
GList *pending_events, *l;
GST_OBJECT_LOCK (dec);
pending_events = priv->pending_events;
priv->pending_events = NULL;
GST_OBJECT_UNLOCK (dec);
GST_DEBUG_OBJECT (dec, "Pushing pending events");
for (l = priv->pending_events; l; l = l->next)
@ -816,7 +829,11 @@ gst_base_audio_decoder_finish_frame (GstBaseAudioDecoder * dec, GstBuffer * buf,
dec->priv->error_count--;
exit:
return gst_base_audio_decoder_output (dec, buf);
ret = gst_base_audio_decoder_output (dec, buf);
GST_BASE_AUDIO_DECODER_STREAM_UNLOCK (dec);
return ret;
/* ERRORS */
wrong_buffer:
@ -825,7 +842,8 @@ wrong_buffer:
("buffer size %d not a multiple of %d", GST_BUFFER_SIZE (buf),
ctx->state.bpf));
gst_buffer_unref (buf);
return GST_FLOW_ERROR;
ret = GST_FLOW_ERROR;
goto exit;
}
overflow:
{
@ -834,7 +852,8 @@ overflow:
priv->frames.length), (NULL));
if (buf)
gst_buffer_unref (buf);
return GST_FLOW_ERROR;
ret = GST_FLOW_ERROR;
goto exit;
}
}
@ -1240,6 +1259,8 @@ gst_base_audio_decoder_chain (GstPad * pad, GstBuffer * buffer)
GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (buffer)),
GST_TIME_ARGS (GST_BUFFER_DURATION (buffer)));
GST_BASE_AUDIO_DECODER_STREAM_LOCK (dec);
if (GST_BUFFER_FLAG_IS_SET (buffer, GST_BUFFER_FLAG_DISCONT)) {
gint64 samples, ts;
@ -1266,6 +1287,8 @@ gst_base_audio_decoder_chain (GstPad * pad, GstBuffer * buffer)
else
ret = gst_base_audio_decoder_chain_reverse (dec, buffer);
GST_BASE_AUDIO_DECODER_STREAM_UNLOCK (dec);
return ret;
}
@ -1292,6 +1315,7 @@ gst_base_audio_decoder_sink_eventfunc (GstBaseAudioDecoder * dec,
gint64 start, stop, time;
gboolean update;
GST_BASE_AUDIO_DECODER_STREAM_LOCK (dec);
gst_event_parse_new_segment_full (event, &update, &rate, &arate, &format,
&start, &stop, &time);
@ -1327,6 +1351,7 @@ gst_base_audio_decoder_sink_eventfunc (GstBaseAudioDecoder * dec,
GST_FORMAT_TIME, start, stop, time);
} else {
GST_DEBUG_OBJECT (dec, "unsupported format; ignoring");
GST_BASE_AUDIO_DECODER_STREAM_UNLOCK (dec);
break;
}
}
@ -1368,8 +1393,10 @@ gst_base_audio_decoder_sink_eventfunc (GstBaseAudioDecoder * dec,
gst_segment_set_newsegment_full (&dec->segment, update, rate, arate,
format, start, stop, time);
gst_pad_push_event (dec->srcpad, event);
dec->priv->pending_events =
g_list_append (dec->priv->pending_events, event);
handled = TRUE;
GST_BASE_AUDIO_DECODER_STREAM_UNLOCK (dec);
break;
}
@ -1377,18 +1404,20 @@ gst_base_audio_decoder_sink_eventfunc (GstBaseAudioDecoder * dec,
break;
case GST_EVENT_FLUSH_STOP:
GST_BASE_AUDIO_DECODER_STREAM_LOCK (dec);
/* prepare for fresh start */
gst_base_audio_decoder_flush (dec, TRUE);
GST_OBJECT_LOCK (dec);
g_list_foreach (dec->priv->pending_events, (GFunc) gst_event_unref, NULL);
g_list_free (dec->priv->pending_events);
dec->priv->pending_events = NULL;
GST_OBJECT_UNLOCK (dec);
GST_BASE_AUDIO_DECODER_STREAM_UNLOCK (dec);
break;
case GST_EVENT_EOS:
GST_BASE_AUDIO_DECODER_STREAM_LOCK (dec);
gst_base_audio_decoder_drain (dec);
GST_BASE_AUDIO_DECODER_STREAM_UNLOCK (dec);
break;
default:
@ -1432,10 +1461,10 @@ gst_base_audio_decoder_sink_event (GstPad * pad, GstEvent * event)
|| GST_EVENT_TYPE (event) == GST_EVENT_FLUSH_STOP) {
ret = gst_pad_event_default (pad, event);
} else {
GST_OBJECT_LOCK (dec);
GST_BASE_AUDIO_DECODER_STREAM_LOCK (dec);
dec->priv->pending_events =
g_list_append (dec->priv->pending_events, event);
GST_OBJECT_UNLOCK (dec);
GST_BASE_AUDIO_DECODER_STREAM_UNLOCK (dec);
ret = TRUE;
}
}
@ -1482,7 +1511,9 @@ gst_base_audio_decoder_do_seek (GstBaseAudioDecoder * dec, GstEvent * event)
return FALSE;
}
GST_BASE_AUDIO_DECODER_STREAM_LOCK (dec);
memcpy (&seek_segment, &dec->segment, sizeof (seek_segment));
GST_BASE_AUDIO_DECODER_STREAM_UNLOCK (dec);
gst_segment_set_seek (&seek_segment, rate, format, flags, start_type,
start_time, end_type, end_time, NULL);
start_time = seek_segment.last_stop;

8
omx/gstbaseaudiodecoder.h
View file

@ -77,6 +77,9 @@ G_BEGIN_DECLS
*/
#define GST_BASE_AUDIO_DECODER_SINK_PAD(obj) (((GstBaseAudioDecoder *) (obj))->sinkpad)
#define GST_BASE_AUDIO_DECODER_STREAM_LOCK(dec) g_static_rec_mutex_lock (&GST_BASE_AUDIO_DECODER (dec)->stream_lock)
#define GST_BASE_AUDIO_DECODER_STREAM_UNLOCK(dec) g_static_rec_mutex_unlock (&GST_BASE_AUDIO_DECODER (dec)->stream_lock)
typedef struct _GstBaseAudioDecoder GstBaseAudioDecoder;
typedef struct _GstBaseAudioDecoderClass GstBaseAudioDecoderClass;
@ -169,6 +172,11 @@ struct _GstBaseAudioDecoder
GstPad *sinkpad;
GstPad *srcpad;
/* protects all data processing, i.e. is locked
* in the chain function, finish_frame and when
* processing serialized events */
GStaticRecMutex stream_lock;
/* MT-protected (with STREAM_LOCK) */
GstSegment segment;
GstBaseAudioDecoderContext *ctx;