mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer.git
synced 2025-03-27 11:32:51 +00:00
audiodecoder: Fix thread safety issues if both pads have different streaming threads
This commit is contained in:
Sebastian Dröge
parent
d0bf465248
commit
b767be2f68
2 changed files with 51 additions and 15 deletions
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@ -385,6 +385,8 @@ gst_audio_decoder_init (GstAudioDecoder * dec, GstAudioDecoderClass * klass)
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dec->priv->adapter_out = gst_adapter_new ();
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g_queue_init (&dec->priv->frames);
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g_static_rec_mutex_init (&dec->stream_lock);
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/* property default */
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dec->priv->latency = DEFAULT_LATENCY;
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dec->priv->tolerance = DEFAULT_TOLERANCE;
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@ -400,7 +402,7 @@ gst_audio_decoder_reset (GstAudioDecoder * dec, gboolean full)
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{
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GST_DEBUG_OBJECT (dec, "gst_audio_decoder_reset");
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GST_OBJECT_LOCK (dec);
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GST_AUDIO_DECODER_STREAM_LOCK (dec);
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if (full) {
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dec->priv->active = FALSE;
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@ -438,7 +440,7 @@ gst_audio_decoder_reset (GstAudioDecoder * dec, gboolean full)
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dec->priv->discont = TRUE;
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dec->priv->sync_flush = FALSE;
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GST_OBJECT_UNLOCK (dec);
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GST_AUDIO_DECODER_STREAM_UNLOCK (dec);
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}
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static void
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@ -456,6 +458,8 @@ gst_audio_decoder_finalize (GObject * object)
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g_object_unref (dec->priv->adapter_out);
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}
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g_static_rec_mutex_free (&dec->stream_lock);
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G_OBJECT_CLASS (parent_class)->finalize (object);
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}
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@ -472,6 +476,8 @@ gst_audio_decoder_src_setcaps (GstPad * pad, GstCaps * caps)
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GST_DEBUG_OBJECT (dec, "setting src caps %" GST_PTR_FORMAT, caps);
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GST_AUDIO_DECODER_STREAM_LOCK (dec);
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/* parse caps here to check subclass;
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* also makes us aware of output format */
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if (!gst_caps_is_fixed (caps))
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@ -488,6 +494,9 @@ gst_audio_decoder_src_setcaps (GstPad * pad, GstCaps * caps)
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if (!gst_audio_info_from_caps (&dec->priv->ctx.info, caps))
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goto refuse_caps;
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done:
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GST_AUDIO_DECODER_STREAM_UNLOCK (dec);
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gst_object_unref (dec);
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return res;
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@ -495,8 +504,8 @@ gst_audio_decoder_src_setcaps (GstPad * pad, GstCaps * caps)
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refuse_caps:
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{
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GST_WARNING_OBJECT (dec, "rejected caps %" GST_PTR_FORMAT, caps);
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gst_object_unref (dec);
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return res;
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res = FALSE;
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goto done;
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}
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}
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@ -512,6 +521,7 @@ gst_audio_decoder_sink_setcaps (GstPad * pad, GstCaps * caps)
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GST_DEBUG_OBJECT (dec, "caps: %" GST_PTR_FORMAT, caps);
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GST_AUDIO_DECODER_STREAM_LOCK (dec);
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/* NOTE pbutils only needed here */
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/* TODO maybe (only) upstream demuxer/parser etc should handle this ? */
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if (dec->priv->taglist)
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@ -523,6 +533,8 @@ gst_audio_decoder_sink_setcaps (GstPad * pad, GstCaps * caps)
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if (klass->set_format)
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res = klass->set_format (dec, caps);
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GST_AUDIO_DECODER_STREAM_UNLOCK (dec);
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g_object_unref (dec);
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return res;
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}
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@ -696,6 +708,7 @@ gst_audio_decoder_finish_frame (GstAudioDecoder * dec, GstBuffer * buf,
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GstAudioDecoderContext *ctx;
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gint samples = 0;
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GstClockTime ts, next_ts;
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GstFlowReturn ret = GST_FLOW_OK;
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/* subclass should know what it is producing by now */
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g_return_val_if_fail (buf == NULL || GST_PAD_CAPS (dec->srcpad) != NULL,
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@ -713,13 +726,13 @@ gst_audio_decoder_finish_frame (GstAudioDecoder * dec, GstBuffer * buf,
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buf ? GST_BUFFER_SIZE (buf) : -1,
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buf ? GST_BUFFER_SIZE (buf) / ctx->info.bpf : -1, frames);
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GST_AUDIO_DECODER_STREAM_LOCK (dec);
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if (priv->pending_events) {
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GList *pending_events, *l;
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GST_OBJECT_LOCK (dec);
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pending_events = priv->pending_events;
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priv->pending_events = NULL;
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GST_OBJECT_UNLOCK (dec);
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GST_DEBUG_OBJECT (dec, "Pushing pending events");
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for (l = priv->pending_events; l; l = l->next)
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@ -833,7 +846,11 @@ gst_audio_decoder_finish_frame (GstAudioDecoder * dec, GstBuffer * buf,
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dec->priv->error_count--;
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exit:
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return gst_audio_decoder_output (dec, buf);
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ret = gst_audio_decoder_output (dec, buf);
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GST_AUDIO_DECODER_STREAM_UNLOCK (dec);
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return ret;
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/* ERRORS */
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wrong_buffer:
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@ -842,7 +859,8 @@ wrong_buffer:
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("buffer size %d not a multiple of %d", GST_BUFFER_SIZE (buf),
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ctx->info.bpf));
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gst_buffer_unref (buf);
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return GST_FLOW_ERROR;
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ret = GST_FLOW_ERROR;
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goto exit;
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}
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overflow:
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{
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@ -851,7 +869,8 @@ overflow:
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priv->frames.length), (NULL));
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if (buf)
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gst_buffer_unref (buf);
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return GST_FLOW_ERROR;
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ret = GST_FLOW_ERROR;
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goto exit;
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}
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}
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@ -1255,6 +1274,8 @@ gst_audio_decoder_chain (GstPad * pad, GstBuffer * buffer)
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GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (buffer)),
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GST_TIME_ARGS (GST_BUFFER_DURATION (buffer)));
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GST_AUDIO_DECODER_STREAM_LOCK (dec);
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if (GST_BUFFER_FLAG_IS_SET (buffer, GST_BUFFER_FLAG_DISCONT)) {
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gint64 samples, ts;
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@ -1281,6 +1302,8 @@ gst_audio_decoder_chain (GstPad * pad, GstBuffer * buffer)
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else
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ret = gst_audio_decoder_chain_reverse (dec, buffer);
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GST_AUDIO_DECODER_STREAM_UNLOCK (dec);
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return ret;
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}
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@ -1306,6 +1329,7 @@ gst_audio_decoder_sink_eventfunc (GstAudioDecoder * dec, GstEvent * event)
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gint64 start, stop, time;
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gboolean update;
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GST_AUDIO_DECODER_STREAM_LOCK (dec);
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gst_event_parse_new_segment_full (event, &update, &rate, &arate, &format,
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&start, &stop, &time);
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@ -1341,6 +1365,7 @@ gst_audio_decoder_sink_eventfunc (GstAudioDecoder * dec, GstEvent * event)
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GST_FORMAT_TIME, start, stop, time);
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} else {
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GST_DEBUG_OBJECT (dec, "unsupported format; ignoring");
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GST_AUDIO_DECODER_STREAM_UNLOCK (dec);
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break;
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}
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}
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@ -1383,8 +1408,10 @@ gst_audio_decoder_sink_eventfunc (GstAudioDecoder * dec, GstEvent * event)
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gst_segment_set_newsegment_full (&dec->segment, update, rate, arate,
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format, start, stop, time);
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gst_pad_push_event (dec->srcpad, event);
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dec->priv->pending_events =
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g_list_append (dec->priv->pending_events, event);
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handled = TRUE;
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GST_AUDIO_DECODER_STREAM_UNLOCK (dec);
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break;
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}
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@ -1392,18 +1419,20 @@ gst_audio_decoder_sink_eventfunc (GstAudioDecoder * dec, GstEvent * event)
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break;
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case GST_EVENT_FLUSH_STOP:
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GST_AUDIO_DECODER_STREAM_LOCK (dec);
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/* prepare for fresh start */
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gst_audio_decoder_flush (dec, TRUE);
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GST_OBJECT_LOCK (dec);
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g_list_foreach (dec->priv->pending_events, (GFunc) gst_event_unref, NULL);
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g_list_free (dec->priv->pending_events);
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dec->priv->pending_events = NULL;
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GST_OBJECT_UNLOCK (dec);
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GST_AUDIO_DECODER_STREAM_UNLOCK (dec);
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break;
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case GST_EVENT_EOS:
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GST_AUDIO_DECODER_STREAM_LOCK (dec);
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gst_audio_decoder_drain (dec);
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GST_AUDIO_DECODER_STREAM_UNLOCK (dec);
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break;
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default:
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@ -1447,10 +1476,10 @@ gst_audio_decoder_sink_event (GstPad * pad, GstEvent * event)
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|| GST_EVENT_TYPE (event) == GST_EVENT_FLUSH_STOP) {
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ret = gst_pad_event_default (pad, event);
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} else {
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GST_OBJECT_LOCK (dec);
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GST_AUDIO_DECODER_STREAM_LOCK (dec);
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dec->priv->pending_events =
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g_list_append (dec->priv->pending_events, event);
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GST_OBJECT_UNLOCK (dec);
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GST_AUDIO_DECODER_STREAM_UNLOCK (dec);
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ret = TRUE;
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}
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}
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@ -20,7 +20,6 @@
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* Free Software Foundation, Inc., 59 Temple Place - Suite 330,
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* Boston, MA 02111-1307, USA.
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*/
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#ifndef _GST_AUDIO_DECODER_H_
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#define _GST_AUDIO_DECODER_H_
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@ -85,6 +84,9 @@ G_BEGIN_DECLS
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*/
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#define GST_AUDIO_DECODER_SINK_PAD(obj) (((GstAudioDecoder *) (obj))->sinkpad)
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#define GST_AUDIO_DECODER_STREAM_LOCK(dec) g_static_rec_mutex_lock (&GST_AUDIO_DECODER (dec)->stream_lock)
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#define GST_AUDIO_DECODER_STREAM_UNLOCK(dec) g_static_rec_mutex_unlock (&GST_AUDIO_DECODER (dec)->stream_lock)
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typedef struct _GstAudioDecoder GstAudioDecoder;
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typedef struct _GstAudioDecoderClass GstAudioDecoderClass;
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@ -146,6 +148,11 @@ struct _GstAudioDecoder
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GstPad *sinkpad;
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GstPad *srcpad;
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/* protects all data processing, i.e. is locked
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* in the chain function, finish_frame and when
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* processing serialized events */
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GStaticRecMutex stream_lock;
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/* MT-protected (with STREAM_LOCK) */
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GstSegment segment;
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