From b767be2f689970f3d0ff4875f4254c43e4991fa1 Mon Sep 17 00:00:00 2001 From: =?utf8?q?Sebastian=20Dr=C3=B6ge?= Date: Mon, 26 Sep 2011 16:22:00 +0200 Subject: [PATCH] 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; -- 2.7.4