diff options
| -rw-r--r-- | ChangeLog | 11 | ||||
| -rw-r--r-- | gst/rtpmanager/gstrtpsession.c | 40 |
2 files changed, 47 insertions, 4 deletions
@@ -1,5 +1,16 @@ 2008-03-11 Wim Taymans <wim.taymans@collabora.co.uk> + Based on patch by: Ole André Vadla Ravnås <ole.andre.ravnas@tandberg.com> + + * gst/rtpmanager/gstrtpsession.c: (gst_rtp_session_init), + (rtcp_thread), (start_rtcp_thread), (stop_rtcp_thread), + (join_rtcp_thread), (gst_rtp_session_change_state): + Avoid a deadlock when joining the RTCP thread in PAUSED because it might + be blocked downstream. Also avoid spawning multiple rtcp threads. + Fixes #520894. + +2008-03-11 Wim Taymans <wim.taymans@collabora.co.uk> + Patch by: Stefan Kost <ensonic@users.sf.net> * gst/rtpmanager/rtpjitterbuffer.c: (calculate_skew): diff --git a/gst/rtpmanager/gstrtpsession.c b/gst/rtpmanager/gstrtpsession.c index 462dc6fb..6de1d3f3 100644 --- a/gst/rtpmanager/gstrtpsession.c +++ b/gst/rtpmanager/gstrtpsession.c @@ -262,6 +262,7 @@ struct _GstRtpSessionPrivate GstClockID id; gboolean stop_thread; GThread *thread; + gboolean thread_stopped; /* caps mapping */ GHashTable *ptmap; @@ -693,6 +694,8 @@ gst_rtp_session_init (GstRtpSession * rtpsession, GstRtpSessionClass * klass) gst_segment_init (&rtpsession->recv_rtp_seg, GST_FORMAT_UNDEFINED); gst_segment_init (&rtpsession->send_rtp_seg, GST_FORMAT_UNDEFINED); + + rtpsession->priv->thread_stopped = TRUE; } static void @@ -923,6 +926,8 @@ rtcp_thread (GstRtpSession * rtpsession) rtp_session_on_timeout (rtpsession->priv->session, current_time, ntpnstime); GST_RTP_SESSION_LOCK (rtpsession); } + /* mark the thread as stopped now */ + rtpsession->priv->thread_stopped = TRUE; GST_RTP_SESSION_UNLOCK (rtpsession); gst_object_unref (sysclock); @@ -949,8 +954,13 @@ start_rtcp_thread (GstRtpSession * rtpsession) GST_RTP_SESSION_LOCK (rtpsession); rtpsession->priv->stop_thread = FALSE; - rtpsession->priv->thread = - g_thread_create ((GThreadFunc) rtcp_thread, rtpsession, TRUE, &error); + if (rtpsession->priv->thread_stopped) { + /* only create a new thread if the old one was stopped. Otherwise we can + * just reuse the currently running one. */ + rtpsession->priv->thread = + g_thread_create ((GThreadFunc) rtcp_thread, rtpsession, TRUE, &error); + rtpsession->priv->thread_stopped = FALSE; + } GST_RTP_SESSION_UNLOCK (rtpsession); if (error != NULL) { @@ -973,9 +983,25 @@ stop_rtcp_thread (GstRtpSession * rtpsession) if (rtpsession->priv->id) gst_clock_id_unschedule (rtpsession->priv->id); GST_RTP_SESSION_UNLOCK (rtpsession); +} - /* FIXME, can deadlock because the thread might be blocked in a push */ - g_thread_join (rtpsession->priv->thread); +static void +join_rtcp_thread (GstRtpSession * rtpsession) +{ + GST_RTP_SESSION_LOCK (rtpsession); + /* don't try to join when we have no thread */ + if (rtpsession->priv->thread != NULL) { + GST_DEBUG_OBJECT (rtpsession, "joining RTCP thread"); + GST_RTP_SESSION_UNLOCK (rtpsession); + + g_thread_join (rtpsession->priv->thread); + + GST_RTP_SESSION_LOCK (rtpsession); + /* after the join, take the lock and clear the thread structure. The caller + * is supposed to not concurrently call start and join. */ + rtpsession->priv->thread = NULL; + } + GST_RTP_SESSION_UNLOCK (rtpsession); } static GstStateChangeReturn @@ -996,6 +1022,10 @@ gst_rtp_session_change_state (GstElement * element, GstStateChange transition) case GST_STATE_CHANGE_PAUSED_TO_PLAYING: break; case GST_STATE_CHANGE_PLAYING_TO_PAUSED: + case GST_STATE_CHANGE_PAUSED_TO_READY: + /* no need to join yet, we might want to continue later. Also, the + * dataflow could block downstream so that a join could just block + * forever. */ stop_rtcp_thread (rtpsession); break; default: @@ -1012,6 +1042,8 @@ gst_rtp_session_change_state (GstElement * element, GstStateChange transition) case GST_STATE_CHANGE_PLAYING_TO_PAUSED: break; case GST_STATE_CHANGE_PAUSED_TO_READY: + /* downstream is now releasing the dataflow and we can join. */ + join_rtcp_thread (rtpsession); break; case GST_STATE_CHANGE_READY_TO_NULL: break; |