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rtpsession: use proper locking for pads and caps

Browse source 
Use the sesion lock and shotdown variable to protect and ref the pads we are
going to push on.

fixes #561825
This commit is contained in:
Wim Taymans 2009-08-31 16:34:14 +02:00
parent a522a2d4d2
commit a74c385b7b
1 changed files with 85 additions and 34 deletions
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119
gst/rtpmanager/gstrtpsession.c
View file

@ -25,7 +25,7 @@
* session. This session can be used to send and receive RTP and RTCP packets.
* Based on what REQUEST pads are requested from the session manager, specific
* functionality can be activated.
*
*
* The session manager currently implements RFC 3550 including:
* <itemizedlist>
* <listitem>
@ -41,38 +41,38 @@
* <para>Scheduling of RR/SR RTCP packets.</para>
* </listitem>
* </itemizedlist>
*
*
* The gstrtpsession will not demux packets based on SSRC or payload type, nor will
* it correct for packet reordering and jitter. Use #GstRtpsSrcDemux,
* #GstRtpPtDemux and GstRtpJitterBuffer in addition to #GstRtpSession to
* perform these tasks. It is usually a good idea to use #GstRtpBin, which
* combines all these features in one element.
*
*
* To use #GstRtpSession as an RTP receiver, request a recv_rtp_sink pad, which will
* automatically create recv_rtp_src pad. Data received on the recv_rtp_sink pad
* will be processed in the session and after being validated forwarded on the
* recv_rtp_src pad.
*
*
* To also use #GstRtpSession as an RTCP receiver, request a recv_rtcp_sink pad,
* which will automatically create a sync_src pad. Packets received on the RTCP
* pad will be used by the session manager to update the stats and database of
* the other participants. SR packets will be forwarded on the sync_src pad
* so that they can be used to perform inter-stream synchronisation when needed.
*
*
* If you want the session manager to generate and send RTCP packets, request
* the send_rtcp_src pad. Packet pushed on this pad contain SR/RR RTCP reports
* that should be sent to all participants in the session.
*
*
* To use #GstRtpSession as a sender, request a send_rtp_sink pad, which will
* automatically create a send_rtp_src pad. The session manager will modify the
* SSRC in the RTP packets to its own SSRC and wil forward the packets on the
* send_rtp_src pad after updating its internal state.
*
*
* The session manager needs the clock-rate of the payload types it is handling
* and will signal the #GstRtpSession::request-pt-map signal when it needs such a
* mapping. One can clear the cached values with the #GstRtpSession::clear-pt-map
* signal.
*
*
* <refsect2>
* <title>Example pipelines</title>
* |[
@ -426,7 +426,7 @@ gst_rtp_session_class_init (GstRtpSessionClass * klass)
/**
* GstRtpSession::on-new-ssrc:
* @sess: the object which received the signal
* @ssrc: the SSRC
* @ssrc: the SSRC
*
* Notify of a new SSRC that entered @session.
*/
@ -437,7 +437,7 @@ gst_rtp_session_class_init (GstRtpSessionClass * klass)
/**
* GstRtpSession::on-ssrc_collision:
* @sess: the object which received the signal
* @ssrc: the SSRC
* @ssrc: the SSRC
*
* Notify when we have an SSRC collision
*/
@ -449,7 +449,7 @@ gst_rtp_session_class_init (GstRtpSessionClass * klass)
/**
* GstRtpSession::on-ssrc_validated:
* @sess: the object which received the signal
* @ssrc: the SSRC
* @ssrc: the SSRC
*
* Notify of a new SSRC that became validated.
*/
@ -485,7 +485,7 @@ gst_rtp_session_class_init (GstRtpSessionClass * klass)
/**
* GstRtpSession::on-bye-ssrc:
* @sess: the object which received the signal
* @ssrc: the SSRC
* @ssrc: the SSRC
*
* Notify of an SSRC that became inactive because of a BYE packet.
*/
@ -496,7 +496,7 @@ gst_rtp_session_class_init (GstRtpSessionClass * klass)
/**
* GstRtpSession::on-bye-timeout:
* @sess: the object which received the signal
* @ssrc: the SSRC
* @ssrc: the SSRC
*
* Notify of an SSRC that has timed out because of BYE
*/
@ -507,7 +507,7 @@ gst_rtp_session_class_init (GstRtpSessionClass * klass)
/**
* GstRtpSession::on-timeout:
* @sess: the object which received the signal
* @ssrc: the SSRC
* @ssrc: the SSRC
*
* Notify of an SSRC that has timed out
*/
@ -518,7 +518,7 @@ gst_rtp_session_class_init (GstRtpSessionClass * klass)
/**
* GstRtpSession::on-sender-timeout:
* @sess: the object which received the signal
* @ssrc: the SSRC
* @ssrc: the SSRC
*
* Notify of a sender SSRC that has timed out and became a receiver
*/
@ -949,13 +949,20 @@ gst_rtp_session_process_rtp (RTPSession * sess, RTPSource * src,
GstFlowReturn result;
GstRtpSession *rtpsession;
GstRtpSessionPrivate *priv;
GstPad *rtp_src;
rtpsession = GST_RTP_SESSION (user_data);
priv = rtpsession->priv;
if (rtpsession->recv_rtp_src) {
GST_RTP_SESSION_LOCK (rtpsession);
if ((rtp_src = rtpsession->recv_rtp_src))
gst_object_ref (rtp_src);
GST_RTP_SESSION_UNLOCK (rtpsession);
if (rtp_src) {
GST_LOG_OBJECT (rtpsession, "pushing received RTP packet");
result = gst_pad_push (rtpsession->recv_rtp_src, buffer);
result = gst_pad_push (rtp_src, buffer);
gst_object_unref (rtp_src);
} else {
GST_DEBUG_OBJECT (rtpsession, "dropping received RTP packet");
gst_buffer_unref (buffer);
@ -973,19 +980,25 @@ gst_rtp_session_send_rtp (RTPSession * sess, RTPSource * src,
GstFlowReturn result;
GstRtpSession *rtpsession;
GstRtpSessionPrivate *priv;
GstPad *rtp_src;
rtpsession = GST_RTP_SESSION (user_data);
priv = rtpsession->priv;
if (rtpsession->send_rtp_src) {
GST_RTP_SESSION_LOCK (rtpsession);
if ((rtp_src = rtpsession->send_rtp_src))
gst_object_ref (rtp_src);
GST_RTP_SESSION_UNLOCK (rtpsession);
if (rtp_src) {
if (GST_IS_BUFFER (data)) {
GST_LOG_OBJECT (rtpsession, "sending RTP packet");
result = gst_pad_push (rtpsession->send_rtp_src, GST_BUFFER_CAST (data));
result = gst_pad_push (rtp_src, GST_BUFFER_CAST (data));
} else {
GST_LOG_OBJECT (rtpsession, "sending RTP list");
result = gst_pad_push_list (rtpsession->send_rtp_src,
GST_BUFFER_LIST_CAST (data));
result = gst_pad_push_list (rtp_src, GST_BUFFER_LIST_CAST (data));
}
gst_object_unref (rtp_src);
} else {
gst_mini_object_unref (GST_MINI_OBJECT_CAST (data));
result = GST_FLOW_OK;
@ -1003,37 +1016,55 @@ gst_rtp_session_send_rtcp (RTPSession * sess, RTPSource * src,
GstFlowReturn result;
GstRtpSession *rtpsession;
GstRtpSessionPrivate *priv;
GstPad *rtcp_src;
rtpsession = GST_RTP_SESSION (user_data);
priv = rtpsession->priv;
if (rtpsession->send_rtcp_src) {
GST_RTP_SESSION_LOCK (rtpsession);
if (rtpsession->priv->stop_thread)
goto stopping;
if ((rtcp_src = rtpsession->send_rtcp_src)) {
GstCaps *caps;
/* set rtcp caps on output pad */
caps = GST_PAD_CAPS (rtpsession->send_rtcp_src);
if (!caps) {
if ((caps = GST_PAD_CAPS (rtcp_src))) {
caps = gst_caps_new_simple ("application/x-rtcp", NULL);
gst_pad_set_caps (rtpsession->send_rtcp_src, caps);
} else {
gst_caps_ref (caps);
gst_pad_set_caps (rtcp_src, caps);
gst_caps_unref (caps);
}
gst_buffer_set_caps (buffer, caps);
gst_caps_unref (caps);
GST_LOG_OBJECT (rtpsession, "sending RTCP");
result = gst_pad_push (rtpsession->send_rtcp_src, buffer);
gst_object_ref (rtcp_src);
GST_RTP_SESSION_UNLOCK (rtpsession);
result = gst_pad_push (rtcp_src, buffer);
/* we have to send EOS after this packet */
if (eos) {
GST_LOG_OBJECT (rtpsession, "sending EOS");
gst_pad_push_event (rtpsession->send_rtcp_src, gst_event_new_eos ());
gst_pad_push_event (rtcp_src, gst_event_new_eos ());
}
gst_object_unref (rtcp_src);
} else {
GST_RTP_SESSION_UNLOCK (rtpsession);
GST_DEBUG_OBJECT (rtpsession, "not sending RTCP, no output pad");
gst_buffer_unref (buffer);
result = GST_FLOW_OK;
}
return result;
/* ERRORS */
stopping:
{
GST_DEBUG_OBJECT (rtpsession, "we are stopping");
gst_buffer_unref (buffer);
GST_RTP_SESSION_UNLOCK (rtpsession);
return GST_FLOW_OK;
}
}
/* called when the session manager has an SR RTCP packet ready for handling
@ -1045,28 +1076,48 @@ gst_rtp_session_sync_rtcp (RTPSession * sess,
GstFlowReturn result;
GstRtpSession *rtpsession;
GstRtpSessionPrivate *priv;
GstPad *sync_src;
rtpsession = GST_RTP_SESSION (user_data);
priv = rtpsession->priv;
if (rtpsession->sync_src) {
GST_RTP_SESSION_LOCK (rtpsession);
if (rtpsession->priv->stop_thread)
goto stopping;
if ((sync_src = rtpsession->sync_src)) {
GstCaps *caps;
/* set rtcp caps on output pad */
if (!(caps = GST_PAD_CAPS (rtpsession->sync_src))) {
if (!(caps = GST_PAD_CAPS (sync_src))) {
caps = gst_caps_new_simple ("application/x-rtcp", NULL);
gst_pad_set_caps (rtpsession->sync_src, caps);
gst_pad_set_caps (sync_src, caps);
gst_caps_unref (caps);
}
gst_buffer_set_caps (buffer, caps);
gst_object_ref (sync_src);
GST_RTP_SESSION_UNLOCK (rtpsession);
GST_LOG_OBJECT (rtpsession, "sending Sync RTCP");
result = gst_pad_push (rtpsession->sync_src, buffer);
result = gst_pad_push (sync_src, buffer);
gst_object_unref (sync_src);
} else {
GST_RTP_SESSION_UNLOCK (rtpsession);
GST_DEBUG_OBJECT (rtpsession, "not sending Sync RTCP, no output pad");
gst_buffer_unref (buffer);
result = GST_FLOW_OK;
}
return result;
/* ERRORS */
stopping:
{
GST_DEBUG_OBJECT (rtpsession, "we are stopping");
gst_buffer_unref (buffer);
GST_RTP_SESSION_UNLOCK (rtpsession);
return GST_FLOW_OK;
}
}
static void