diff options
Diffstat (limited to 'gst/rtpmanager/gstrtpsession.c')
| -rw-r--r-- | gst/rtpmanager/gstrtpsession.c | 122 |
1 files changed, 117 insertions, 5 deletions
diff --git a/gst/rtpmanager/gstrtpsession.c b/gst/rtpmanager/gstrtpsession.c index 80f340a5..431098d9 100644 --- a/gst/rtpmanager/gstrtpsession.c +++ b/gst/rtpmanager/gstrtpsession.c @@ -20,20 +20,112 @@ /** * SECTION:element-rtpsession * @short_description: an RTP session manager - * @see_also: rtpjitterbuffer, rtpbin + * @see_also: rtpjitterbuffer, rtpbin, rtpptdemux, rtpssrcdemux * * <refsect2> * <para> + * The RTP session manager models one participant with a unique SSRC in an RTP + * 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. + * </para> + * <para> + * The session manager currently implements RFC 3550 including: + * <itemizedlist> + * <listitem> + * <para>RTP packet validation based on consecutive sequence numbers.</para> + * </listitem> + * <listitem> + * <para>Maintainance of the SSRC participant database.</para> + * </listitem> + * <listitem> + * <para>Keeping per participant statistics based on received RTCP packets.</para> + * </listitem> + * <listitem> + * <para>Scheduling of RR/SR RTCP packets.</para> + * </listitem> + * </itemizedlist> + * </para> + * <para> + * The rtpsession will not demux packets based on SSRC or payload type, nor will + * it correct for packet reordering and jitter. Use rtpssrcdemux, rtpptdemux and + * rtpjitterbuffer in addition to rtpsession to perform these tasks. It is + * usually a good idea to use rtpbin, which combines all these features in one + * element. + * </para> + * <para> + * To use rtpsession 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. + * </para> + * <para> + * To also use rtpsession 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. + * </para> + * <para> + * 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. + * </para> + * <para> + * To use rtpsession 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. + * </para> + * <para> + * 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. * </para> * <title>Example pipelines</title> * <para> * <programlisting> - * gst-launch -v filesrc location=sine.ogg ! oggdemux ! vorbisdec ! audioconvert ! alsasink + * gst-launch udpsrc port=5000 caps="application/x-rtp, ..." ! .recv_rtp_sink rtpsession .recv_rtp_src ! rtptheoradepay ! theoradec ! xvimagesink * </programlisting> + * Receive theora RTP packets from port 5000 and send them to the depayloader, + * decoder and display. Note that the application/x-rtp caps on udpsrc should be + * configured based on some negotiation process such as RTSP for this pipeline + * to work correctly. + * </para> + * <para> + * <programlisting> + * gst-launch udpsrc port=5000 caps="application/x-rtp, ..." ! .recv_rtp_sink rtpsession name=session \ + * .recv_rtp_src ! rtptheoradepay ! theoradec ! xvimagesink \ + * udpsrc port=5001 caps="application/x-rtcp" ! session.recv_rtcp_sink + * </programlisting> + * Receive theora RTP packets from port 5000 and send them to the depayloader, + * decoder and display. Receive RTCP packets from port 5001 and process them in + * the session manager. + * Note that the application/x-rtp caps on udpsrc should be + * configured based on some negotiation process such as RTSP for this pipeline + * to work correctly. + * </para> + * <para> + * <programlisting> + * gst-launch videotestsrc ! theoraenc ! rtptheorapay ! .send_rtp_sink rtpsession .send_rtp_src ! udpsink port=5000 + * </programlisting> + * Send theora RTP packets through the session manager and out on UDP port 5000. + * </para> + * <para> + * <programlisting> + * gst-launch videotestsrc ! theoraenc ! rtptheorapay ! .send_rtp_sink rtpsession name=session .send_rtp_src \ + * ! udpsink port=5000 session.send_rtcp_src ! udpsink port=5001 + * </programlisting> + * Send theora RTP packets through the session manager and out on UDP port 5000. + * Send RTCP packets on port 5001. Not that this pipeline will not preroll + * correctly because the second udpsink will not preroll correctly (no RTCP + * packets are sent in the PAUSED state). Applications should manually set and + * keep (see #gst_element_set_locked_state()) the RTCP udpsink to the PLAYING state. * </para> * </refsect2> * - * Last reviewed on 2007-04-02 (0.10.6) + * Last reviewed on 2007-05-23 (0.10.6) */ #ifdef HAVE_CONFIG_H @@ -50,7 +142,7 @@ GST_DEBUG_CATEGORY_STATIC (gst_rtp_session_debug); /* elementfactory information */ static const GstElementDetails rtpsession_details = GST_ELEMENT_DETAILS ("RTP Session", - "Filter/Editor/Video", + "Filter/Network/RTP", "Implement an RTP session", "Wim Taymans <wim@fluendo.com>"); @@ -109,6 +201,7 @@ GST_STATIC_PAD_TEMPLATE ("send_rtcp_src", enum { SIGNAL_REQUEST_PT_MAP, + SIGNAL_CLEAR_PT_MAP, LAST_SIGNAL }; @@ -169,6 +262,8 @@ static GstPad *gst_rtp_session_request_new_pad (GstElement * element, GstPadTemplate * templ, const gchar * name); static void gst_rtp_session_release_pad (GstElement * element, GstPad * pad); +static void gst_rtp_session_clear_pt_map (GstRTPSession * rtpsession); + static guint gst_rtp_session_signals[LAST_SIGNAL] = { 0 }; GST_BOILERPLATE (GstRTPSession, gst_rtp_session, GstElement, GST_TYPE_ELEMENT); @@ -226,6 +321,16 @@ gst_rtp_session_class_init (GstRTPSessionClass * klass) G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRTPSessionClass, request_pt_map), NULL, NULL, gst_rtp_bin_marshal_BOXED__UINT, GST_TYPE_CAPS, 1, G_TYPE_UINT); + /** + * GstRTPSession::clear-pt-map: + * @sess: the object which received the signal + * + * Clear the cached pt-maps requested with GstRTPSession::request-pt-map. + */ + gst_rtp_session_signals[SIGNAL_CLEAR_PT_MAP] = + g_signal_new ("clear-pt-map", G_TYPE_FROM_CLASS (klass), + G_SIGNAL_ACTION, G_STRUCT_OFFSET (GstRTPSessionClass, clear_pt_map), + NULL, NULL, g_cclosure_marshal_VOID__VOID, G_TYPE_NONE, 0, G_TYPE_NONE); gstelement_class->change_state = GST_DEBUG_FUNCPTR (gst_rtp_session_change_state); @@ -234,6 +339,8 @@ gst_rtp_session_class_init (GstRTPSessionClass * klass) gstelement_class->release_pad = GST_DEBUG_FUNCPTR (gst_rtp_session_release_pad); + klass->clear_pt_map = GST_DEBUG_FUNCPTR (gst_rtp_session_clear_pt_map); + GST_DEBUG_CATEGORY_INIT (gst_rtp_session_debug, "rtpsession", 0, "RTP Session"); } @@ -315,7 +422,6 @@ rtcp_thread (GstRTPSession * rtpsession) next_timeout = rtp_session_next_timeout (rtpsession->priv->session, current_time); - GST_DEBUG_OBJECT (rtpsession, "next check time %" GST_TIME_FORMAT, GST_TIME_ARGS (next_timeout)); @@ -438,6 +544,12 @@ failed_thread: } } +static void +gst_rtp_session_clear_pt_map (GstRTPSession * rtpsession) +{ + /* FIXME, do something */ +} + /* called when the session manager has an RTP packet ready for further * processing */ static GstFlowReturn |