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gst/rtpmanager/gstrtpsession.c: Avoid a deadlock when joining the RTCP thread in PAUSED because it might be blocked d...

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Original commit message from CVS:
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.
This commit is contained in:
Ole André Vadla Ravnås 2008-03-11 11:36:03 +00:00 committed by Wim Taymans
parent 4e64ace712
commit 0b94354a0d
2 changed files with 47 additions and 4 deletions
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11
ChangeLog
View file

@ -1,3 +1,14 @@
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>

40
gst/rtpmanager/gstrtpsession.c
View file

@ -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;