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

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This commit is contained in:
Sebastian Dröge 2011-08-17 11:34:04 +02:00
parent ebc740ea06
commit 60a1e0e967
2 changed files with 46 additions and 10 deletions
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49
omx/gstbaseaudioencoder.c
View file

@ -354,6 +354,8 @@ gst_base_audio_encoder_init (GstBaseAudioEncoder * enc,
enc->priv->adapter = gst_adapter_new ();
enc->ctx = &enc->priv->ctx;
g_static_rec_mutex_init (&enc->stream_lock);
/* property default */
enc->perfect_ts = DEFAULT_PERFECT_TS;
enc->hard_resync = DEFAULT_HARD_RESYNC;
@ -367,6 +369,8 @@ gst_base_audio_encoder_init (GstBaseAudioEncoder * enc,
static void
gst_base_audio_encoder_reset (GstBaseAudioEncoder * enc, gboolean full)
{
GST_BASE_AUDIO_ENCODER_STREAM_LOCK (enc);
if (full) {
enc->priv->active = FALSE;
enc->priv->samples_in = 0;
@ -389,6 +393,8 @@ gst_base_audio_encoder_reset (GstBaseAudioEncoder * enc, gboolean full)
enc->priv->base_gp = -1;
enc->priv->samples = 0;
enc->priv->discont = FALSE;
GST_BASE_AUDIO_ENCODER_STREAM_UNLOCK (enc);
}
static void
@ -398,6 +404,8 @@ gst_base_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);
}
@ -443,6 +451,8 @@ gst_base_audio_encoder_finish_frame (GstBaseAudioEncoder * enc, GstBuffer * buf,
g_return_val_if_fail (buf == NULL || GST_BUFFER_SIZE (buf) > 0,
GST_FLOW_ERROR);
GST_BASE_AUDIO_ENCODER_STREAM_LOCK (enc);
GST_LOG_OBJECT (enc, "accepting %d bytes encoded data as %d samples",
buf ? GST_BUFFER_SIZE (buf) : -1, samples);
@ -451,10 +461,9 @@ gst_base_audio_encoder_finish_frame (GstBaseAudioEncoder * 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)
@ -602,6 +611,8 @@ gst_base_audio_encoder_finish_frame (GstBaseAudioEncoder * enc, GstBuffer * buf,
}
exit:
GST_BASE_AUDIO_ENCODER_STREAM_UNLOCK (enc);
return ret;
/* ERRORS */
@ -612,7 +623,8 @@ overflow:
samples, priv->offset / ctx->state.bpf), (NULL));
if (buf)
gst_buffer_unref (buf);
return GST_FLOW_ERROR;
ret = GST_FLOW_ERROR;
goto exit;
}
}
@ -752,6 +764,8 @@ gst_base_audio_encoder_chain (GstPad * pad, GstBuffer * buffer)
priv = enc->priv;
ctx = enc->ctx;
GST_BASE_AUDIO_ENCODER_STREAM_LOCK (enc);
/* should know what is coming by now */
if (!ctx->state.bpf)
goto not_negotiated;
@ -882,6 +896,9 @@ gst_base_audio_encoder_chain (GstPad * pad, GstBuffer * buffer)
done:
GST_LOG_OBJECT (enc, "chain leaving");
GST_BASE_AUDIO_ENCODER_STREAM_UNLOCK (enc);
return ret;
/* ERRORS */
@ -890,7 +907,8 @@ 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:
{
@ -898,7 +916,8 @@ wrong_buffer:
("buffer size %d not a multiple of %d", GST_BUFFER_SIZE (buffer),
ctx->state.bpf));
gst_buffer_unref (buffer);
return GST_FLOW_ERROR;
ret = GST_FLOW_ERROR;
goto done;
}
}
@ -920,6 +939,8 @@ gst_base_audio_encoder_sink_setcaps (GstPad * pad, GstCaps * caps)
ctx = enc->ctx;
state = &ctx->state;
GST_BASE_AUDIO_ENCODER_STREAM_LOCK (enc);
GST_DEBUG_OBJECT (enc, "caps: %" GST_PTR_FORMAT, caps);
if (!gst_caps_is_fixed (caps))
@ -971,13 +992,17 @@ gst_base_audio_encoder_sink_setcaps (GstPad * pad, GstCaps * caps)
GST_DEBUG_OBJECT (enc, "new audio format identical to configured format");
}
exit:
GST_BASE_AUDIO_ENCODER_STREAM_UNLOCK (enc);
return res;
/* ERRORS */
refuse_caps:
{
GST_WARNING_OBJECT (enc, "rejected caps %" GST_PTR_FORMAT, caps);
return res;
goto exit;
}
}
@ -1104,6 +1129,7 @@ gst_base_audio_encoder_sink_eventfunc (GstBaseAudioEncoder * enc,
break;
}
GST_BASE_AUDIO_ENCODER_STREAM_LOCK (enc);
/* finish current segment */
gst_base_audio_encoder_drain (enc);
/* reset partially for new segment */
@ -1111,6 +1137,7 @@ gst_base_audio_encoder_sink_eventfunc (GstBaseAudioEncoder * enc,
/* and follow along with segment */
gst_segment_set_newsegment_full (&enc->segment, update, rate, arate,
format, start, stop, time);
GST_BASE_AUDIO_ENCODER_STREAM_UNLOCK (enc);
break;
}
@ -1118,6 +1145,7 @@ gst_base_audio_encoder_sink_eventfunc (GstBaseAudioEncoder * enc,
break;
case GST_EVENT_FLUSH_STOP:
GST_BASE_AUDIO_ENCODER_STREAM_LOCK (enc);
/* discard any pending stuff */
/* TODO route through drain ?? */
if (!enc->priv->drained && klass->flush)
@ -1125,16 +1153,17 @@ gst_base_audio_encoder_sink_eventfunc (GstBaseAudioEncoder * enc,
/* and get (re)set for the sequel */
gst_base_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_BASE_AUDIO_ENCODER_STREAM_UNLOCK (enc);
break;
case GST_EVENT_EOS:
GST_BASE_AUDIO_ENCODER_STREAM_LOCK (enc);
gst_base_audio_encoder_drain (enc);
GST_BASE_AUDIO_ENCODER_STREAM_UNLOCK (enc);
break;
default:
@ -1178,10 +1207,10 @@ gst_base_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_BASE_AUDIO_ENCODER_STREAM_LOCK (enc);
enc->priv->pending_events =
g_list_append (enc->priv->pending_events, event);
GST_OBJECT_UNLOCK (enc);
GST_BASE_AUDIO_ENCODER_STREAM_UNLOCK (enc);
ret = TRUE;
}
}

7
omx/gstbaseaudioencoder.h
View file

@ -83,6 +83,8 @@ G_BEGIN_DECLS
*/
#define GST_BASE_AUDIO_ENCODER_SEGMENT(obj) (GST_BASE_AUDIO_ENCODER_CAST (obj)->segment)
#define GST_BASE_AUDIO_ENCODER_STREAM_LOCK(enc) g_static_rec_mutex_lock (&GST_BASE_AUDIO_ENCODER (enc)->stream_lock)
#define GST_BASE_AUDIO_ENCODER_STREAM_UNLOCK(enc) g_static_rec_mutex_unlock (&GST_BASE_AUDIO_ENCODER (enc)->stream_lock)
typedef struct _GstBaseAudioEncoder GstBaseAudioEncoder;
typedef struct _GstBaseAudioEncoderClass GstBaseAudioEncoderClass;
@ -130,6 +132,11 @@ struct _GstBaseAudioEncoder {
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;
GstBaseAudioEncoderContext *ctx;