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

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This commit is contained in:
Sebastian Dröge 2011-09-26 15:45:40 +02:00
parent 61ffd7cb42
commit 16c3d6b3d5
2 changed files with 44 additions and 12 deletions
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49
gst-libs/gst/audio/gstaudioencoder.c
View file

@ -368,6 +368,8 @@ gst_audio_encoder_init (GstAudioEncoder * enc, GstAudioEncoderClass * bclass)
enc->priv->adapter = gst_adapter_new ();
g_static_rec_mutex_init (&enc->stream_lock);
/* property default */
enc->priv->granule = DEFAULT_GRANULE;
enc->priv->perfect_ts = DEFAULT_PERFECT_TS;
@ -382,7 +384,7 @@ gst_audio_encoder_init (GstAudioEncoder * enc, GstAudioEncoderClass * bclass)
static void
gst_audio_encoder_reset (GstAudioEncoder * enc, gboolean full)
{
GST_OBJECT_LOCK (enc);
GST_AUDIO_ENCODER_STREAM_LOCK (enc);
GST_LOG_OBJECT (enc, "reset full %d", full);
@ -413,7 +415,7 @@ gst_audio_encoder_reset (GstAudioEncoder * enc, gboolean full)
enc->priv->samples = 0;
enc->priv->discont = FALSE;
GST_OBJECT_UNLOCK (enc);
GST_AUDIO_ENCODER_STREAM_UNLOCK (enc);
}
static void
@ -423,6 +425,8 @@ gst_audio_encoder_finalize (GObject * object)
g_object_unref (enc->priv->adapter);
g_static_rec_mutex_free (&enc->stream_lock);
G_OBJECT_CLASS (parent_class)->finalize (object);
}
@ -470,6 +474,8 @@ gst_audio_encoder_finish_frame (GstAudioEncoder * enc, GstBuffer * buf,
g_return_val_if_fail (buf == NULL || GST_BUFFER_SIZE (buf) > 0,
GST_FLOW_ERROR);
GST_AUDIO_ENCODER_STREAM_LOCK (enc);
if (G_UNLIKELY (enc->priv->tags)) {
GstTagList *tags;
@ -493,10 +499,9 @@ gst_audio_encoder_finish_frame (GstAudioEncoder * enc, GstBuffer * buf,
if (priv->pending_events) {
GList *pending_events, *l;
GST_OBJECT_LOCK (enc);
pending_events = priv->pending_events;
priv->pending_events = NULL;
GST_OBJECT_UNLOCK (enc);
GST_DEBUG_OBJECT (enc, "Pushing pending events");
for (l = priv->pending_events; l; l = l->next)
@ -650,6 +655,8 @@ gst_audio_encoder_finish_frame (GstAudioEncoder * enc, GstBuffer * buf,
}
exit:
GST_AUDIO_ENCODER_STREAM_UNLOCK (enc);
return ret;
/* ERRORS */
@ -660,7 +667,8 @@ overflow:
samples, priv->offset / ctx->info.bpf), (NULL));
if (buf)
gst_buffer_unref (buf);
return GST_FLOW_ERROR;
ret = GST_FLOW_ERROR;
goto exit;
}
}
@ -800,6 +808,8 @@ gst_audio_encoder_chain (GstPad * pad, GstBuffer * buffer)
priv = enc->priv;
ctx = &enc->priv->ctx;
GST_AUDIO_ENCODER_STREAM_LOCK (enc);
/* should know what is coming by now */
if (!ctx->info.bpf)
goto not_negotiated;
@ -931,6 +941,9 @@ gst_audio_encoder_chain (GstPad * pad, GstBuffer * buffer)
done:
GST_LOG_OBJECT (enc, "chain leaving");
GST_AUDIO_ENCODER_STREAM_UNLOCK (enc);
return ret;
/* ERRORS */
@ -939,7 +952,8 @@ not_negotiated:
GST_ELEMENT_ERROR (enc, CORE, NEGOTIATION, (NULL),
("encoder not initialized"));
gst_buffer_unref (buffer);
return GST_FLOW_NOT_NEGOTIATED;
ret = GST_FLOW_NOT_NEGOTIATED;
goto done;
}
wrong_buffer:
{
@ -947,7 +961,8 @@ wrong_buffer:
("buffer size %d not a multiple of %d", GST_BUFFER_SIZE (buffer),
ctx->info.bpf));
gst_buffer_unref (buffer);
return GST_FLOW_ERROR;
ret = GST_FLOW_ERROR;
goto done;
}
}
@ -989,6 +1004,8 @@ gst_audio_encoder_sink_setcaps (GstPad * pad, GstCaps * caps)
ctx = &enc->priv->ctx;
state = &ctx->info;
GST_AUDIO_ENCODER_STREAM_LOCK (enc);
GST_DEBUG_OBJECT (enc, "caps: %" GST_PTR_FORMAT, caps);
if (!gst_caps_is_fixed (caps))
@ -1045,13 +1062,17 @@ gst_audio_encoder_sink_setcaps (GstPad * pad, GstCaps * caps)
GST_DEBUG_OBJECT (enc, "new audio format identical to configured format");
}
exit:
GST_AUDIO_ENCODER_STREAM_UNLOCK (enc);
return res;
/* ERRORS */
refuse_caps:
{
GST_WARNING_OBJECT (enc, "rejected caps %" GST_PTR_FORMAT, caps);
return res;
goto exit;
}
}
@ -1191,6 +1212,7 @@ gst_audio_encoder_sink_eventfunc (GstAudioEncoder * enc, GstEvent * event)
break;
}
GST_AUDIO_ENCODER_STREAM_LOCK (enc);
/* finish current segment */
gst_audio_encoder_drain (enc);
/* reset partially for new segment */
@ -1198,6 +1220,7 @@ gst_audio_encoder_sink_eventfunc (GstAudioEncoder * enc, GstEvent * event)
/* and follow along with segment */
gst_segment_set_newsegment_full (&enc->segment, update, rate, arate,
format, start, stop, time);
GST_AUDIO_ENCODER_STREAM_UNLOCK (enc);
break;
}
@ -1205,6 +1228,7 @@ gst_audio_encoder_sink_eventfunc (GstAudioEncoder * enc, GstEvent * event)
break;
case GST_EVENT_FLUSH_STOP:
GST_AUDIO_ENCODER_STREAM_LOCK (enc);
/* discard any pending stuff */
/* TODO route through drain ?? */
if (!enc->priv->drained && klass->flush)
@ -1212,16 +1236,17 @@ gst_audio_encoder_sink_eventfunc (GstAudioEncoder * enc, GstEvent * event)
/* and get (re)set for the sequel */
gst_audio_encoder_reset (enc, FALSE);
GST_OBJECT_LOCK (enc);
g_list_foreach (enc->priv->pending_events, (GFunc) gst_event_unref, NULL);
g_list_free (enc->priv->pending_events);
enc->priv->pending_events = NULL;
GST_OBJECT_UNLOCK (enc);
GST_AUDIO_ENCODER_STREAM_UNLOCK (enc);
break;
case GST_EVENT_EOS:
GST_AUDIO_ENCODER_STREAM_LOCK (enc);
gst_audio_encoder_drain (enc);
GST_AUDIO_ENCODER_STREAM_UNLOCK (enc);
break;
case GST_EVENT_TAG:
@ -1284,10 +1309,10 @@ gst_audio_encoder_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 (enc);
GST_AUDIO_ENCODER_STREAM_LOCK (enc);
enc->priv->pending_events =
g_list_append (enc->priv->pending_events, event);
GST_OBJECT_UNLOCK (enc);
GST_AUDIO_ENCODER_STREAM_UNLOCK (enc);
ret = TRUE;
}
}

7
gst-libs/gst/audio/gstaudioencoder.h
View file

@ -87,6 +87,8 @@ G_BEGIN_DECLS
*/
#define GST_AUDIO_ENCODER_SEGMENT(obj) (GST_AUDIO_ENCODER_CAST (obj)->segment)
#define GST_AUDIO_ENCODER_STREAM_LOCK(enc) g_static_rec_mutex_lock (&GST_AUDIO_ENCODER (enc)->stream_lock)
#define GST_AUDIO_ENCODER_STREAM_UNLOCK(enc) g_static_rec_mutex_unlock (&GST_AUDIO_ENCODER (enc)->stream_lock)
typedef struct _GstAudioEncoder GstAudioEncoder;
typedef struct _GstAudioEncoderClass GstAudioEncoderClass;
@ -108,6 +110,11 @@ struct _GstAudioEncoder {
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;