[0.11] gst-plugins-base: audiodecoder: Fix thread safety issues if both pads have different streaming threads
Wim Taymans
wtay at kemper.freedesktop.org
Mon Sep 26 10:23:02 PDT 2011
Module: gst-plugins-base
Branch: 0.11
Commit: b767be2f689970f3d0ff4875f4254c43e4991fa1
URL: http://cgit.freedesktop.org/gstreamer/gst-plugins-base/commit/?id=b767be2f689970f3d0ff4875f4254c43e4991fa1
Author: Sebastian Dröge <sebastian.droege at collabora.co.uk>
Date: Mon Sep 26 16:22:00 2011 +0200
audiodecoder: Fix thread safety issues if both pads have different streaming threads
---
gst-libs/gst/audio/gstaudiodecoder.c | 57 +++++++++++++++++++++++++--------
gst-libs/gst/audio/gstaudiodecoder.h | 9 +++++-
2 files changed, 51 insertions(+), 15 deletions(-)
diff --git a/gst-libs/gst/audio/gstaudiodecoder.c b/gst-libs/gst/audio/gstaudiodecoder.c
index 84729cb..3a910a8 100644
--- a/gst-libs/gst/audio/gstaudiodecoder.c
+++ b/gst-libs/gst/audio/gstaudiodecoder.c
@@ -385,6 +385,8 @@ gst_audio_decoder_init (GstAudioDecoder * dec, GstAudioDecoderClass * klass)
dec->priv->adapter_out = gst_adapter_new ();
g_queue_init (&dec->priv->frames);
+ g_static_rec_mutex_init (&dec->stream_lock);
+
/* property default */
dec->priv->latency = DEFAULT_LATENCY;
dec->priv->tolerance = DEFAULT_TOLERANCE;
@@ -400,7 +402,7 @@ gst_audio_decoder_reset (GstAudioDecoder * dec, gboolean full)
{
GST_DEBUG_OBJECT (dec, "gst_audio_decoder_reset");
- GST_OBJECT_LOCK (dec);
+ GST_AUDIO_DECODER_STREAM_LOCK (dec);
if (full) {
dec->priv->active = FALSE;
@@ -438,7 +440,7 @@ gst_audio_decoder_reset (GstAudioDecoder * dec, gboolean full)
dec->priv->discont = TRUE;
dec->priv->sync_flush = FALSE;
- GST_OBJECT_UNLOCK (dec);
+ GST_AUDIO_DECODER_STREAM_UNLOCK (dec);
}
static void
@@ -456,6 +458,8 @@ gst_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);
}
@@ -472,6 +476,8 @@ gst_audio_decoder_src_setcaps (GstPad * pad, GstCaps * caps)
GST_DEBUG_OBJECT (dec, "setting src caps %" GST_PTR_FORMAT, caps);
+ GST_AUDIO_DECODER_STREAM_LOCK (dec);
+
/* parse caps here to check subclass;
* also makes us aware of output format */
if (!gst_caps_is_fixed (caps))
@@ -488,6 +494,9 @@ gst_audio_decoder_src_setcaps (GstPad * pad, GstCaps * caps)
if (!gst_audio_info_from_caps (&dec->priv->ctx.info, caps))
goto refuse_caps;
+done:
+ GST_AUDIO_DECODER_STREAM_UNLOCK (dec);
+
gst_object_unref (dec);
return res;
@@ -495,8 +504,8 @@ gst_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;
}
}
@@ -512,6 +521,7 @@ gst_audio_decoder_sink_setcaps (GstPad * pad, GstCaps * caps)
GST_DEBUG_OBJECT (dec, "caps: %" GST_PTR_FORMAT, caps);
+ GST_AUDIO_DECODER_STREAM_LOCK (dec);
/* NOTE pbutils only needed here */
/* TODO maybe (only) upstream demuxer/parser etc should handle this ? */
if (dec->priv->taglist)
@@ -523,6 +533,8 @@ gst_audio_decoder_sink_setcaps (GstPad * pad, GstCaps * caps)
if (klass->set_format)
res = klass->set_format (dec, caps);
+ GST_AUDIO_DECODER_STREAM_UNLOCK (dec);
+
g_object_unref (dec);
return res;
}
@@ -696,6 +708,7 @@ gst_audio_decoder_finish_frame (GstAudioDecoder * dec, GstBuffer * buf,
GstAudioDecoderContext *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,
@@ -713,13 +726,13 @@ gst_audio_decoder_finish_frame (GstAudioDecoder * dec, GstBuffer * buf,
buf ? GST_BUFFER_SIZE (buf) : -1,
buf ? GST_BUFFER_SIZE (buf) / ctx->info.bpf : -1, frames);
+ GST_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)
@@ -833,7 +846,11 @@ gst_audio_decoder_finish_frame (GstAudioDecoder * dec, GstBuffer * buf,
dec->priv->error_count--;
exit:
- return gst_audio_decoder_output (dec, buf);
+ ret = gst_audio_decoder_output (dec, buf);
+
+ GST_AUDIO_DECODER_STREAM_UNLOCK (dec);
+
+ return ret;
/* ERRORS */
wrong_buffer:
@@ -842,7 +859,8 @@ wrong_buffer:
("buffer size %d not a multiple of %d", GST_BUFFER_SIZE (buf),
ctx->info.bpf));
gst_buffer_unref (buf);
- return GST_FLOW_ERROR;
+ ret = GST_FLOW_ERROR;
+ goto exit;
}
overflow:
{
@@ -851,7 +869,8 @@ overflow:
priv->frames.length), (NULL));
if (buf)
gst_buffer_unref (buf);
- return GST_FLOW_ERROR;
+ ret = GST_FLOW_ERROR;
+ goto exit;
}
}
@@ -1255,6 +1274,8 @@ gst_audio_decoder_chain (GstPad * pad, GstBuffer * buffer)
GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (buffer)),
GST_TIME_ARGS (GST_BUFFER_DURATION (buffer)));
+ GST_AUDIO_DECODER_STREAM_LOCK (dec);
+
if (GST_BUFFER_FLAG_IS_SET (buffer, GST_BUFFER_FLAG_DISCONT)) {
gint64 samples, ts;
@@ -1281,6 +1302,8 @@ gst_audio_decoder_chain (GstPad * pad, GstBuffer * buffer)
else
ret = gst_audio_decoder_chain_reverse (dec, buffer);
+ GST_AUDIO_DECODER_STREAM_UNLOCK (dec);
+
return ret;
}
@@ -1306,6 +1329,7 @@ gst_audio_decoder_sink_eventfunc (GstAudioDecoder * dec, GstEvent * event)
gint64 start, stop, time;
gboolean update;
+ GST_AUDIO_DECODER_STREAM_LOCK (dec);
gst_event_parse_new_segment_full (event, &update, &rate, &arate, &format,
&start, &stop, &time);
@@ -1341,6 +1365,7 @@ gst_audio_decoder_sink_eventfunc (GstAudioDecoder * dec, GstEvent * event)
GST_FORMAT_TIME, start, stop, time);
} else {
GST_DEBUG_OBJECT (dec, "unsupported format; ignoring");
+ GST_AUDIO_DECODER_STREAM_UNLOCK (dec);
break;
}
}
@@ -1383,8 +1408,10 @@ gst_audio_decoder_sink_eventfunc (GstAudioDecoder * dec, GstEvent * event)
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_AUDIO_DECODER_STREAM_UNLOCK (dec);
break;
}
@@ -1392,18 +1419,20 @@ gst_audio_decoder_sink_eventfunc (GstAudioDecoder * dec, GstEvent * event)
break;
case GST_EVENT_FLUSH_STOP:
+ GST_AUDIO_DECODER_STREAM_LOCK (dec);
/* prepare for fresh start */
gst_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_AUDIO_DECODER_STREAM_UNLOCK (dec);
break;
case GST_EVENT_EOS:
+ GST_AUDIO_DECODER_STREAM_LOCK (dec);
gst_audio_decoder_drain (dec);
+ GST_AUDIO_DECODER_STREAM_UNLOCK (dec);
break;
default:
@@ -1447,10 +1476,10 @@ gst_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_AUDIO_DECODER_STREAM_LOCK (dec);
dec->priv->pending_events =
g_list_append (dec->priv->pending_events, event);
- GST_OBJECT_UNLOCK (dec);
+ GST_AUDIO_DECODER_STREAM_UNLOCK (dec);
ret = TRUE;
}
}
diff --git a/gst-libs/gst/audio/gstaudiodecoder.h b/gst-libs/gst/audio/gstaudiodecoder.h
index 1c47e1a..783f83e 100644
--- a/gst-libs/gst/audio/gstaudiodecoder.h
+++ b/gst-libs/gst/audio/gstaudiodecoder.h
@@ -20,7 +20,6 @@
* Free Software Foundation, Inc., 59 Temple Place - Suite 330,
* Boston, MA 02111-1307, USA.
*/
-
#ifndef _GST_AUDIO_DECODER_H_
#define _GST_AUDIO_DECODER_H_
@@ -85,6 +84,9 @@ G_BEGIN_DECLS
*/
#define GST_AUDIO_DECODER_SINK_PAD(obj) (((GstAudioDecoder *) (obj))->sinkpad)
+#define GST_AUDIO_DECODER_STREAM_LOCK(dec) g_static_rec_mutex_lock (&GST_AUDIO_DECODER (dec)->stream_lock)
+#define GST_AUDIO_DECODER_STREAM_UNLOCK(dec) g_static_rec_mutex_unlock (&GST_AUDIO_DECODER (dec)->stream_lock)
+
typedef struct _GstAudioDecoder GstAudioDecoder;
typedef struct _GstAudioDecoderClass GstAudioDecoderClass;
@@ -146,6 +148,11 @@ struct _GstAudioDecoder
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;
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