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Diffstat (limited to 'gst/rtpmanager/gstrtpsession.c')
| -rw-r--r-- | gst/rtpmanager/gstrtpsession.c | 2186 |
1 files changed, 2186 insertions, 0 deletions
diff --git a/gst/rtpmanager/gstrtpsession.c b/gst/rtpmanager/gstrtpsession.c new file mode 100644 index 0000000..ebeb3fd --- /dev/null +++ b/gst/rtpmanager/gstrtpsession.c @@ -0,0 +1,2186 @@ +/* GStreamer + * Copyright (C) <2007> Wim Taymans <wim.taymans@gmail.com> + * + * This library is free software; you can redistribute it and/or + * modify it under the terms of the GNU Library General Public + * License as published by the Free Software Foundation; either + * version 2 of the License, or (at your option) any later version. + * + * This library is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * Library General Public License for more details. + * + * You should have received a copy of the GNU Library General Public + * License along with this library; if not, write to the + * Free Software Foundation, Inc., 59 Temple Place - Suite 330, + * Boston, MA 02111-1307, USA. + */ + +/** + * SECTION:element-gstrtpsession + * @see_also: gstrtpjitterbuffer, gstrtpbin, gstrtpptdemux, gstrtpssrcdemux + * + * 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. + * + * 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> + * + * 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> + * |[ + * gst-launch udpsrc port=5000 caps="application/x-rtp, ..." ! .recv_rtp_sink gstrtpsession .recv_rtp_src ! rtptheoradepay ! theoradec ! xvimagesink + * ]| 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. + * |[ + * gst-launch udpsrc port=5000 caps="application/x-rtp, ..." ! .recv_rtp_sink gstrtpsession name=session \ + * .recv_rtp_src ! rtptheoradepay ! theoradec ! xvimagesink \ + * udpsrc port=5001 caps="application/x-rtcp" ! session.recv_rtcp_sink + * ]| 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. + * |[ + * gst-launch videotestsrc ! theoraenc ! rtptheorapay ! .send_rtp_sink gstrtpsession .send_rtp_src ! udpsink port=5000 + * ]| Send theora RTP packets through the session manager and out on UDP port + * 5000. + * |[ + * gst-launch videotestsrc ! theoraenc ! rtptheorapay ! .send_rtp_sink gstrtpsession name=session .send_rtp_src \ + * ! udpsink port=5000 session.send_rtcp_src ! udpsink port=5001 + * ]| Send theora RTP packets through the session manager and out on UDP port + * 5000. Send RTCP packets on port 5001. Note 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. + * </refsect2> + * + * Last reviewed on 2007-05-28 (0.10.5) + */ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +#include <gst/rtp/gstrtpbuffer.h> + +#include "gstrtpbin-marshal.h" +#include "gstrtpsession.h" +#include "rtpsession.h" + +GST_DEBUG_CATEGORY_STATIC (gst_rtp_session_debug); +#define GST_CAT_DEFAULT gst_rtp_session_debug + +/* sink pads */ +static GstStaticPadTemplate rtpsession_recv_rtp_sink_template = +GST_STATIC_PAD_TEMPLATE ("recv_rtp_sink", + GST_PAD_SINK, + GST_PAD_REQUEST, + GST_STATIC_CAPS ("application/x-rtp") + ); + +static GstStaticPadTemplate rtpsession_recv_rtcp_sink_template = +GST_STATIC_PAD_TEMPLATE ("recv_rtcp_sink", + GST_PAD_SINK, + GST_PAD_REQUEST, + GST_STATIC_CAPS ("application/x-rtcp") + ); + +static GstStaticPadTemplate rtpsession_send_rtp_sink_template = +GST_STATIC_PAD_TEMPLATE ("send_rtp_sink", + GST_PAD_SINK, + GST_PAD_REQUEST, + GST_STATIC_CAPS ("application/x-rtp") + ); + +/* src pads */ +static GstStaticPadTemplate rtpsession_recv_rtp_src_template = +GST_STATIC_PAD_TEMPLATE ("recv_rtp_src", + GST_PAD_SRC, + GST_PAD_SOMETIMES, + GST_STATIC_CAPS ("application/x-rtp") + ); + +static GstStaticPadTemplate rtpsession_sync_src_template = +GST_STATIC_PAD_TEMPLATE ("sync_src", + GST_PAD_SRC, + GST_PAD_SOMETIMES, + GST_STATIC_CAPS ("application/x-rtcp") + ); + +static GstStaticPadTemplate rtpsession_send_rtp_src_template = +GST_STATIC_PAD_TEMPLATE ("send_rtp_src", + GST_PAD_SRC, + GST_PAD_SOMETIMES, + GST_STATIC_CAPS ("application/x-rtp") + ); + +static GstStaticPadTemplate rtpsession_send_rtcp_src_template = +GST_STATIC_PAD_TEMPLATE ("send_rtcp_src", + GST_PAD_SRC, + GST_PAD_REQUEST, + GST_STATIC_CAPS ("application/x-rtcp") + ); + +/* signals and args */ +enum +{ + SIGNAL_REQUEST_PT_MAP, + SIGNAL_CLEAR_PT_MAP, + + SIGNAL_ON_NEW_SSRC, + SIGNAL_ON_SSRC_COLLISION, + SIGNAL_ON_SSRC_VALIDATED, + SIGNAL_ON_SSRC_ACTIVE, + SIGNAL_ON_SSRC_SDES, + SIGNAL_ON_BYE_SSRC, + SIGNAL_ON_BYE_TIMEOUT, + SIGNAL_ON_TIMEOUT, + SIGNAL_ON_SENDER_TIMEOUT, + LAST_SIGNAL +}; + +#define DEFAULT_NTP_NS_BASE 0 +#define DEFAULT_BANDWIDTH RTP_STATS_BANDWIDTH +#define DEFAULT_RTCP_FRACTION (RTP_STATS_BANDWIDTH * RTP_STATS_RTCP_FRACTION) +#define DEFAULT_RTCP_RR_BANDWIDTH -1 +#define DEFAULT_RTCP_RS_BANDWIDTH -1 +#define DEFAULT_SDES NULL +#define DEFAULT_NUM_SOURCES 0 +#define DEFAULT_NUM_ACTIVE_SOURCES 0 +#define DEFAULT_USE_PIPELINE_CLOCK FALSE +#define DEFAULT_RTCP_MIN_INTERVAL (RTP_STATS_MIN_INTERVAL * GST_SECOND) + +enum +{ + PROP_0, + PROP_NTP_NS_BASE, + PROP_BANDWIDTH, + PROP_RTCP_FRACTION, + PROP_RTCP_RR_BANDWIDTH, + PROP_RTCP_RS_BANDWIDTH, + PROP_SDES, + PROP_NUM_SOURCES, + PROP_NUM_ACTIVE_SOURCES, + PROP_INTERNAL_SESSION, + PROP_USE_PIPELINE_CLOCK, + PROP_RTCP_MIN_INTERVAL, + PROP_LAST +}; + +#define GST_RTP_SESSION_GET_PRIVATE(obj) \ + (G_TYPE_INSTANCE_GET_PRIVATE ((obj), GST_TYPE_RTP_SESSION, GstRtpSessionPrivate)) + +#define GST_RTP_SESSION_LOCK(sess) g_mutex_lock ((sess)->priv->lock) +#define GST_RTP_SESSION_UNLOCK(sess) g_mutex_unlock ((sess)->priv->lock) + +struct _GstRtpSessionPrivate +{ + GMutex *lock; + GstClock *sysclock; + + RTPSession *session; + + /* thread for sending out RTCP */ + GstClockID id; + gboolean stop_thread; + GThread *thread; + gboolean thread_stopped; + + /* caps mapping */ + GHashTable *ptmap; + + /* NTP base time */ + guint64 ntpnsbase; + gboolean use_pipeline_clock; +}; + +/* callbacks to handle actions from the session manager */ +static GstFlowReturn gst_rtp_session_process_rtp (RTPSession * sess, + RTPSource * src, GstBuffer * buffer, gpointer user_data); +static GstFlowReturn gst_rtp_session_send_rtp (RTPSession * sess, + RTPSource * src, gpointer data, gpointer user_data); +static GstFlowReturn gst_rtp_session_send_rtcp (RTPSession * sess, + RTPSource * src, GstBuffer * buffer, gboolean eos, gpointer user_data); +static GstFlowReturn gst_rtp_session_sync_rtcp (RTPSession * sess, + RTPSource * src, GstBuffer * buffer, gpointer user_data); +static gint gst_rtp_session_clock_rate (RTPSession * sess, guint8 payload, + gpointer user_data); +static void gst_rtp_session_reconsider (RTPSession * sess, gpointer user_data); +static void gst_rtp_session_request_key_unit (RTPSession * sess, + gboolean all_headers, gpointer user_data); +static GstClockTime gst_rtp_session_request_time (RTPSession * session, + gpointer user_data); + +static RTPSessionCallbacks callbacks = { + gst_rtp_session_process_rtp, + gst_rtp_session_send_rtp, + gst_rtp_session_sync_rtcp, + gst_rtp_session_send_rtcp, + gst_rtp_session_clock_rate, + gst_rtp_session_reconsider, + gst_rtp_session_request_key_unit, + gst_rtp_session_request_time +}; + +/* GObject vmethods */ +static void gst_rtp_session_finalize (GObject * object); +static void gst_rtp_session_set_property (GObject * object, guint prop_id, + const GValue * value, GParamSpec * pspec); +static void gst_rtp_session_get_property (GObject * object, guint prop_id, + GValue * value, GParamSpec * pspec); + +/* GstElement vmethods */ +static GstStateChangeReturn gst_rtp_session_change_state (GstElement * element, + GstStateChange transition); +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 }; + +static void +on_new_ssrc (RTPSession * session, RTPSource * src, GstRtpSession * sess) +{ + g_signal_emit (sess, gst_rtp_session_signals[SIGNAL_ON_NEW_SSRC], 0, + src->ssrc); +} + +static void +on_ssrc_collision (RTPSession * session, RTPSource * src, GstRtpSession * sess) +{ + g_signal_emit (sess, gst_rtp_session_signals[SIGNAL_ON_SSRC_COLLISION], 0, + src->ssrc); +} + +static void +on_ssrc_validated (RTPSession * session, RTPSource * src, GstRtpSession * sess) +{ + g_signal_emit (sess, gst_rtp_session_signals[SIGNAL_ON_SSRC_VALIDATED], 0, + src->ssrc); +} + +static void +on_ssrc_active (RTPSession * session, RTPSource * src, GstRtpSession * sess) +{ + g_signal_emit (sess, gst_rtp_session_signals[SIGNAL_ON_SSRC_ACTIVE], 0, + src->ssrc); +} + +static void +on_ssrc_sdes (RTPSession * session, RTPSource * src, GstRtpSession * sess) +{ + GstStructure *s; + GstMessage *m; + + /* convert the new SDES info into a message */ + RTP_SESSION_LOCK (session); + g_object_get (src, "sdes", &s, NULL); + RTP_SESSION_UNLOCK (session); + + m = gst_message_new_custom (GST_MESSAGE_ELEMENT, GST_OBJECT (sess), s); + gst_element_post_message (GST_ELEMENT_CAST (sess), m); + + g_signal_emit (sess, gst_rtp_session_signals[SIGNAL_ON_SSRC_SDES], 0, + src->ssrc); +} + +static void +on_bye_ssrc (RTPSession * session, RTPSource * src, GstRtpSession * sess) +{ + g_signal_emit (sess, gst_rtp_session_signals[SIGNAL_ON_BYE_SSRC], 0, + src->ssrc); +} + +static void +on_bye_timeout (RTPSession * session, RTPSource * src, GstRtpSession * sess) +{ + g_signal_emit (sess, gst_rtp_session_signals[SIGNAL_ON_BYE_TIMEOUT], 0, + src->ssrc); +} + +static void +on_timeout (RTPSession * session, RTPSource * src, GstRtpSession * sess) +{ + g_signal_emit (sess, gst_rtp_session_signals[SIGNAL_ON_TIMEOUT], 0, + src->ssrc); +} + +static void +on_sender_timeout (RTPSession * session, RTPSource * src, GstRtpSession * sess) +{ + g_signal_emit (sess, gst_rtp_session_signals[SIGNAL_ON_SENDER_TIMEOUT], 0, + src->ssrc); +} + +GST_BOILERPLATE (GstRtpSession, gst_rtp_session, GstElement, GST_TYPE_ELEMENT); + +static void +gst_rtp_session_base_init (gpointer klass) +{ + GstElementClass *element_class = GST_ELEMENT_CLASS (klass); + + /* sink pads */ + gst_element_class_add_pad_template (element_class, + gst_static_pad_template_get (&rtpsession_recv_rtp_sink_template)); + gst_element_class_add_pad_template (element_class, + gst_static_pad_template_get (&rtpsession_recv_rtcp_sink_template)); + gst_element_class_add_pad_template (element_class, + gst_static_pad_template_get (&rtpsession_send_rtp_sink_template)); + + /* src pads */ + gst_element_class_add_pad_template (element_class, + gst_static_pad_template_get (&rtpsession_recv_rtp_src_template)); + gst_element_class_add_pad_template (element_class, + gst_static_pad_template_get (&rtpsession_sync_src_template)); + gst_element_class_add_pad_template (element_class, + gst_static_pad_template_get (&rtpsession_send_rtp_src_template)); + gst_element_class_add_pad_template (element_class, + gst_static_pad_template_get (&rtpsession_send_rtcp_src_template)); + + gst_element_class_set_details_simple (element_class, "RTP Session", + "Filter/Network/RTP", + "Implement an RTP session", "Wim Taymans <wim.taymans@gmail.com>"); +} + +static void +gst_rtp_session_class_init (GstRtpSessionClass * klass) +{ + GObjectClass *gobject_class; + GstElementClass *gstelement_class; + + gobject_class = (GObjectClass *) klass; + gstelement_class = (GstElementClass *) klass; + + g_type_class_add_private (klass, sizeof (GstRtpSessionPrivate)); + + gobject_class->finalize = gst_rtp_session_finalize; + gobject_class->set_property = gst_rtp_session_set_property; + gobject_class->get_property = gst_rtp_session_get_property; + + /** + * GstRtpSession::request-pt-map: + * @sess: the object which received the signal + * @pt: the pt + * + * Request the payload type as #GstCaps for @pt. + */ + gst_rtp_session_signals[SIGNAL_REQUEST_PT_MAP] = + g_signal_new ("request-pt-map", G_TYPE_FROM_CLASS (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); + + /** + * GstRtpSession::on-new-ssrc: + * @sess: the object which received the signal + * @ssrc: the SSRC + * + * Notify of a new SSRC that entered @session. + */ + gst_rtp_session_signals[SIGNAL_ON_NEW_SSRC] = + g_signal_new ("on-new-ssrc", G_TYPE_FROM_CLASS (klass), + G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRtpSessionClass, on_new_ssrc), + NULL, NULL, g_cclosure_marshal_VOID__UINT, G_TYPE_NONE, 1, G_TYPE_UINT); + /** + * GstRtpSession::on-ssrc_collision: + * @sess: the object which received the signal + * @ssrc: the SSRC + * + * Notify when we have an SSRC collision + */ + gst_rtp_session_signals[SIGNAL_ON_SSRC_COLLISION] = + g_signal_new ("on-ssrc-collision", G_TYPE_FROM_CLASS (klass), + G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRtpSessionClass, + on_ssrc_collision), NULL, NULL, g_cclosure_marshal_VOID__UINT, + G_TYPE_NONE, 1, G_TYPE_UINT); + /** + * GstRtpSession::on-ssrc_validated: + * @sess: the object which received the signal + * @ssrc: the SSRC + * + * Notify of a new SSRC that became validated. + */ + gst_rtp_session_signals[SIGNAL_ON_SSRC_VALIDATED] = + g_signal_new ("on-ssrc-validated", G_TYPE_FROM_CLASS (klass), + G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRtpSessionClass, + on_ssrc_validated), NULL, NULL, g_cclosure_marshal_VOID__UINT, + G_TYPE_NONE, 1, G_TYPE_UINT); + /** + * GstRtpSession::on-ssrc_active: + * @sess: the object which received the signal + * @ssrc: the SSRC + * + * Notify of a SSRC that is active, i.e., sending RTCP. + */ + gst_rtp_session_signals[SIGNAL_ON_SSRC_ACTIVE] = + g_signal_new ("on-ssrc-active", G_TYPE_FROM_CLASS (klass), + G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRtpSessionClass, + on_ssrc_active), NULL, NULL, g_cclosure_marshal_VOID__UINT, + G_TYPE_NONE, 1, G_TYPE_UINT); + /** + * GstRtpSession::on-ssrc-sdes: + * @session: the object which received the signal + * @src: the SSRC + * + * Notify that a new SDES was received for SSRC. + */ + gst_rtp_session_signals[SIGNAL_ON_SSRC_SDES] = + g_signal_new ("on-ssrc-sdes", G_TYPE_FROM_CLASS (klass), + G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRtpSessionClass, on_ssrc_sdes), + NULL, NULL, g_cclosure_marshal_VOID__UINT, G_TYPE_NONE, 1, G_TYPE_UINT); + + /** + * GstRtpSession::on-bye-ssrc: + * @sess: the object which received the signal + * @ssrc: the SSRC + * + * Notify of an SSRC that became inactive because of a BYE packet. + */ + gst_rtp_session_signals[SIGNAL_ON_BYE_SSRC] = + g_signal_new ("on-bye-ssrc", G_TYPE_FROM_CLASS (klass), + G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRtpSessionClass, on_bye_ssrc), + NULL, NULL, g_cclosure_marshal_VOID__UINT, G_TYPE_NONE, 1, G_TYPE_UINT); + /** + * GstRtpSession::on-bye-timeout: + * @sess: the object which received the signal + * @ssrc: the SSRC + * + * Notify of an SSRC that has timed out because of BYE + */ + gst_rtp_session_signals[SIGNAL_ON_BYE_TIMEOUT] = + g_signal_new ("on-bye-timeout", G_TYPE_FROM_CLASS (klass), + G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRtpSessionClass, on_bye_timeout), + NULL, NULL, g_cclosure_marshal_VOID__UINT, G_TYPE_NONE, 1, G_TYPE_UINT); + /** + * GstRtpSession::on-timeout: + * @sess: the object which received the signal + * @ssrc: the SSRC + * + * Notify of an SSRC that has timed out + */ + gst_rtp_session_signals[SIGNAL_ON_TIMEOUT] = + g_signal_new ("on-timeout", G_TYPE_FROM_CLASS (klass), + G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRtpSessionClass, on_timeout), + NULL, NULL, g_cclosure_marshal_VOID__UINT, G_TYPE_NONE, 1, G_TYPE_UINT); + /** + * GstRtpSession::on-sender-timeout: + * @sess: the object which received the signal + * @ssrc: the SSRC + * + * Notify of a sender SSRC that has timed out and became a receiver + */ + gst_rtp_session_signals[SIGNAL_ON_SENDER_TIMEOUT] = + g_signal_new ("on-sender-timeout", G_TYPE_FROM_CLASS (klass), + G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRtpSessionClass, + on_sender_timeout), NULL, NULL, g_cclosure_marshal_VOID__UINT, + G_TYPE_NONE, 1, G_TYPE_UINT); + + g_object_class_install_property (gobject_class, PROP_NTP_NS_BASE, + g_param_spec_uint64 ("ntp-ns-base", "NTP base time", + "The NTP base time corresponding to running_time 0 (deprecated)", 0, + G_MAXUINT64, DEFAULT_NTP_NS_BASE, + G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)); + + g_object_class_install_property (gobject_class, PROP_BANDWIDTH, + g_param_spec_double ("bandwidth", "Bandwidth", + "The bandwidth of the session in bytes per second (0 for auto-discover)", + 0.0, G_MAXDOUBLE, DEFAULT_BANDWIDTH, + G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)); + + g_object_class_install_property (gobject_class, PROP_RTCP_FRACTION, + g_param_spec_double ("rtcp-fraction", "RTCP Fraction", + "The RTCP bandwidth of the session in bytes per second " + "(or as a real fraction of the RTP bandwidth if < 1.0)", + 0.0, G_MAXDOUBLE, DEFAULT_RTCP_FRACTION, + G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)); + + g_object_class_install_property (gobject_class, PROP_RTCP_RR_BANDWIDTH, + g_param_spec_int ("rtcp-rr-bandwidth", "RTCP RR bandwidth", + "The RTCP bandwidth used for receivers in bytes per second (-1 = default)", + -1, G_MAXINT, DEFAULT_RTCP_RR_BANDWIDTH, + G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)); + + g_object_class_install_property (gobject_class, PROP_RTCP_RS_BANDWIDTH, + g_param_spec_int ("rtcp-rs-bandwidth", "RTCP RS bandwidth", + "The RTCP bandwidth used for senders in bytes per second (-1 = default)", + -1, G_MAXINT, DEFAULT_RTCP_RS_BANDWIDTH, + G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)); + + g_object_class_install_property (gobject_class, PROP_SDES, + g_param_spec_boxed ("sdes", "SDES", + "The SDES items of this session", + GST_TYPE_STRUCTURE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)); + + g_object_class_install_property (gobject_class, PROP_NUM_SOURCES, + g_param_spec_uint ("num-sources", "Num Sources", + "The number of sources in the session", 0, G_MAXUINT, + DEFAULT_NUM_SOURCES, G_PARAM_READABLE | G_PARAM_STATIC_STRINGS)); + + g_object_class_install_property (gobject_class, PROP_NUM_ACTIVE_SOURCES, + g_param_spec_uint ("num-active-sources", "Num Active Sources", + "The number of active sources in the session", 0, G_MAXUINT, + DEFAULT_NUM_ACTIVE_SOURCES, + G_PARAM_READABLE | G_PARAM_STATIC_STRINGS)); + + g_object_class_install_property (gobject_class, PROP_INTERNAL_SESSION, + g_param_spec_object ("internal-session", "Internal Session", + "The internal RTPSession object", RTP_TYPE_SESSION, + G_PARAM_READABLE | G_PARAM_STATIC_STRINGS)); + + g_object_class_install_property (gobject_class, PROP_USE_PIPELINE_CLOCK, + g_param_spec_boolean ("use-pipeline-clock", "Use pipeline clock", + "Use the pipeline clock to set the NTP time in the RTCP SR messages", + DEFAULT_USE_PIPELINE_CLOCK, + G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)); + + g_object_class_install_property (gobject_class, PROP_RTCP_MIN_INTERVAL, + g_param_spec_uint64 ("rtcp-min-interval", "Minimum RTCP interval", + "Minimum interval between Regular RTCP packet (in ns)", + 0, G_MAXUINT64, DEFAULT_RTCP_MIN_INTERVAL, + G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)); + + gstelement_class->change_state = + GST_DEBUG_FUNCPTR (gst_rtp_session_change_state); + gstelement_class->request_new_pad = + GST_DEBUG_FUNCPTR (gst_rtp_session_request_new_pad); + 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"); +} + +static void +gst_rtp_session_init (GstRtpSession * rtpsession, GstRtpSessionClass * klass) +{ + rtpsession->priv = GST_RTP_SESSION_GET_PRIVATE (rtpsession); + rtpsession->priv->lock = g_mutex_new (); + rtpsession->priv->sysclock = gst_system_clock_obtain (); + rtpsession->priv->session = rtp_session_new (); + rtpsession->priv->use_pipeline_clock = DEFAULT_USE_PIPELINE_CLOCK; + + /* configure callbacks */ + rtp_session_set_callbacks (rtpsession->priv->session, &callbacks, rtpsession); + /* configure signals */ + g_signal_connect (rtpsession->priv->session, "on-new-ssrc", + (GCallback) on_new_ssrc, rtpsession); + g_signal_connect (rtpsession->priv->session, "on-ssrc-collision", + (GCallback) on_ssrc_collision, rtpsession); + g_signal_connect (rtpsession->priv->session, "on-ssrc-validated", + (GCallback) on_ssrc_validated, rtpsession); + g_signal_connect (rtpsession->priv->session, "on-ssrc-active", + (GCallback) on_ssrc_active, rtpsession); + g_signal_connect (rtpsession->priv->session, "on-ssrc-sdes", + (GCallback) on_ssrc_sdes, rtpsession); + g_signal_connect (rtpsession->priv->session, "on-bye-ssrc", + (GCallback) on_bye_ssrc, rtpsession); + g_signal_connect (rtpsession->priv->session, "on-bye-timeout", + (GCallback) on_bye_timeout, rtpsession); + g_signal_connect (rtpsession->priv->session, "on-timeout", + (GCallback) on_timeout, rtpsession); + g_signal_connect (rtpsession->priv->session, "on-sender-timeout", + (GCallback) on_sender_timeout, rtpsession); + rtpsession->priv->ptmap = g_hash_table_new_full (NULL, NULL, NULL, + (GDestroyNotify) gst_caps_unref); + + 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 +gst_rtp_session_finalize (GObject * object) +{ + GstRtpSession *rtpsession; + + rtpsession = GST_RTP_SESSION (object); + + g_hash_table_destroy (rtpsession->priv->ptmap); + g_mutex_free (rtpsession->priv->lock); + g_object_unref (rtpsession->priv->sysclock); + g_object_unref (rtpsession->priv->session); + + G_OBJECT_CLASS (parent_class)->finalize (object); +} + +static void +gst_rtp_session_set_property (GObject * object, guint prop_id, + const GValue * value, GParamSpec * pspec) +{ + GstRtpSession *rtpsession; + GstRtpSessionPrivate *priv; + + rtpsession = GST_RTP_SESSION (object); + priv = rtpsession->priv; + + switch (prop_id) { + case PROP_NTP_NS_BASE: + GST_OBJECT_LOCK (rtpsession); + priv->ntpnsbase = g_value_get_uint64 (value); + GST_DEBUG_OBJECT (rtpsession, "setting NTP base to %" GST_TIME_FORMAT, + GST_TIME_ARGS (priv->ntpnsbase)); + GST_OBJECT_UNLOCK (rtpsession); + break; + case PROP_BANDWIDTH: + g_object_set_property (G_OBJECT (priv->session), "bandwidth", value); + break; + case PROP_RTCP_FRACTION: + g_object_set_property (G_OBJECT (priv->session), "rtcp-fraction", value); + break; + case PROP_RTCP_RR_BANDWIDTH: + g_object_set_property (G_OBJECT (priv->session), "rtcp-rr-bandwidth", + value); + break; + case PROP_RTCP_RS_BANDWIDTH: + g_object_set_property (G_OBJECT (priv->session), "rtcp-rs-bandwidth", + value); + break; + case PROP_SDES: + rtp_session_set_sdes_struct (priv->session, g_value_get_boxed (value)); + break; + case PROP_USE_PIPELINE_CLOCK: + priv->use_pipeline_clock = g_value_get_boolean (value); + break; + case PROP_RTCP_MIN_INTERVAL: + g_object_set_property (G_OBJECT (priv->session), "rtcp-min-interval", + value); + break; + default: + G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec); + break; + } +} + +static void +gst_rtp_session_get_property (GObject * object, guint prop_id, + GValue * value, GParamSpec * pspec) +{ + GstRtpSession *rtpsession; + GstRtpSessionPrivate *priv; + + rtpsession = GST_RTP_SESSION (object); + priv = rtpsession->priv; + + switch (prop_id) { + case PROP_NTP_NS_BASE: + GST_OBJECT_LOCK (rtpsession); + g_value_set_uint64 (value, priv->ntpnsbase); + GST_OBJECT_UNLOCK (rtpsession); + break; + case PROP_BANDWIDTH: + g_object_get_property (G_OBJECT (priv->session), "bandwidth", value); + break; + case PROP_RTCP_FRACTION: + g_object_get_property (G_OBJECT (priv->session), "rtcp-fraction", value); + break; + case PROP_RTCP_RR_BANDWIDTH: + g_object_get_property (G_OBJECT (priv->session), "rtcp-rr-bandwidth", + value); + break; + case PROP_RTCP_RS_BANDWIDTH: + g_object_get_property (G_OBJECT (priv->session), "rtcp-rs-bandwidth", + value); + break; + case PROP_SDES: + g_value_take_boxed (value, rtp_session_get_sdes_struct (priv->session)); + break; + case PROP_NUM_SOURCES: + g_value_set_uint (value, rtp_session_get_num_sources (priv->session)); + break; + case PROP_NUM_ACTIVE_SOURCES: + g_value_set_uint (value, + rtp_session_get_num_active_sources (priv->session)); + break; + case PROP_INTERNAL_SESSION: + g_value_set_object (value, priv->session); + break; + case PROP_USE_PIPELINE_CLOCK: + g_value_set_boolean (value, priv->use_pipeline_clock); + break; + case PROP_RTCP_MIN_INTERVAL: + g_object_get_property (G_OBJECT (priv->session), "rtcp-min-interval", + value); + break; + default: + G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec); + break; + } +} + +static void +get_current_times (GstRtpSession * rtpsession, GstClockTime * running_time, + guint64 * ntpnstime) +{ + guint64 ntpns; + GstClock *clock; + GstClockTime base_time, rt, clock_time; + + GST_OBJECT_LOCK (rtpsession); + if ((clock = GST_ELEMENT_CLOCK (rtpsession))) { + base_time = GST_ELEMENT_CAST (rtpsession)->base_time; + gst_object_ref (clock); + GST_OBJECT_UNLOCK (rtpsession); + + clock_time = gst_clock_get_time (clock); + + if (rtpsession->priv->use_pipeline_clock) { + ntpns = clock_time; + } else { + GTimeVal current; + + /* get current NTP time */ + g_get_current_time (¤t); + ntpns = GST_TIMEVAL_TO_TIME (current); + } + + /* add constant to convert from 1970 based time to 1900 based time */ + ntpns += (2208988800LL * GST_SECOND); + + /* get current clock time and convert to running time */ + rt = clock_time - base_time; + + gst_object_unref (clock); + } else { + GST_OBJECT_UNLOCK (rtpsession); + rt = -1; + ntpns = -1; + } + if (running_time) + *running_time = rt; + if (ntpnstime) + *ntpnstime = ntpns; +} + +static void +rtcp_thread (GstRtpSession * rtpsession) +{ + GstClockID id; + GstClockTime current_time; + GstClockTime next_timeout; + guint64 ntpnstime; + GstClockTime running_time; + RTPSession *session; + GstClock *sysclock; + + GST_DEBUG_OBJECT (rtpsession, "entering RTCP thread"); + + GST_RTP_SESSION_LOCK (rtpsession); + + sysclock = rtpsession->priv->sysclock; + current_time = gst_clock_get_time (sysclock); + + session = rtpsession->priv->session; + + while (!rtpsession->priv->stop_thread) { + GstClockReturn res; + + /* get initial estimate */ + next_timeout = rtp_session_next_timeout (session, current_time); + + GST_DEBUG_OBJECT (rtpsession, "next check time %" GST_TIME_FORMAT, + GST_TIME_ARGS (next_timeout)); + + /* leave if no more timeouts, the session ended */ + if (next_timeout == GST_CLOCK_TIME_NONE) + break; + + id = rtpsession->priv->id = + gst_clock_new_single_shot_id (sysclock, next_timeout); + GST_RTP_SESSION_UNLOCK (rtpsession); + + res = gst_clock_id_wait (id, NULL); + + GST_RTP_SESSION_LOCK (rtpsession); + gst_clock_id_unref (id); + rtpsession->priv->id = NULL; + + if (rtpsession->priv->stop_thread) + break; + + /* update current time */ + current_time = gst_clock_get_time (sysclock); + + /* get current NTP time */ + get_current_times (rtpsession, &running_time, &ntpnstime); + + /* we get unlocked because we need to perform reconsideration, don't perform + * the timeout but get a new reporting estimate. */ + GST_DEBUG_OBJECT (rtpsession, "unlocked %d, current %" GST_TIME_FORMAT, + res, GST_TIME_ARGS (current_time)); + + /* perform actions, we ignore result. Release lock because it might push. */ + GST_RTP_SESSION_UNLOCK (rtpsession); + rtp_session_on_timeout (session, current_time, ntpnstime, running_time); + GST_RTP_SESSION_LOCK (rtpsession); + } + /* mark the thread as stopped now */ + rtpsession->priv->thread_stopped = TRUE; + GST_RTP_SESSION_UNLOCK (rtpsession); + + GST_DEBUG_OBJECT (rtpsession, "leaving RTCP thread"); +} + +static gboolean +start_rtcp_thread (GstRtpSession * rtpsession) +{ + GError *error = NULL; + gboolean res; + + GST_DEBUG_OBJECT (rtpsession, "starting RTCP thread"); + + GST_RTP_SESSION_LOCK (rtpsession); + rtpsession->priv->stop_thread = FALSE; + if (rtpsession->priv->thread_stopped) { + /* if the thread stopped, and we still have a handle to the thread, join it + * now. We can safely join with the lock held, the thread will not take it + * anymore. */ + if (rtpsession->priv->thread) + g_thread_join (rtpsession->priv->thread); + /* 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) { + res = FALSE; + GST_DEBUG_OBJECT (rtpsession, "failed to start thread, %s", error->message); + g_error_free (error); + } else { + res = TRUE; + } + return res; +} + +static void +stop_rtcp_thread (GstRtpSession * rtpsession) +{ + GST_DEBUG_OBJECT (rtpsession, "stopping RTCP thread"); + + GST_RTP_SESSION_LOCK (rtpsession); + rtpsession->priv->stop_thread = TRUE; + if (rtpsession->priv->id) + gst_clock_id_unschedule (rtpsession->priv->id); + GST_RTP_SESSION_UNLOCK (rtpsession); +} + +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 +gst_rtp_session_change_state (GstElement * element, GstStateChange transition) +{ + GstStateChangeReturn res; + GstRtpSession *rtpsession; + + rtpsession = GST_RTP_SESSION (element); + + switch (transition) { + case GST_STATE_CHANGE_NULL_TO_READY: + break; + case GST_STATE_CHANGE_READY_TO_PAUSED: + break; + 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: + break; + } + + res = parent_class->change_state (element, transition); + + switch (transition) { + case GST_STATE_CHANGE_PAUSED_TO_PLAYING: + if (!start_rtcp_thread (rtpsession)) + goto failed_thread; + break; + 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; + default: + break; + } + return res; + + /* ERRORS */ +failed_thread: + { + return GST_STATE_CHANGE_FAILURE; + } +} + +static gboolean +return_true (gpointer key, gpointer value, gpointer user_data) +{ + return TRUE; +} + +static void +gst_rtp_session_clear_pt_map (GstRtpSession * rtpsession) +{ + g_hash_table_foreach_remove (rtpsession->priv->ptmap, return_true, NULL); +} + +/* called when the session manager has an RTP packet or a list of packets + * ready for further processing */ +static GstFlowReturn +gst_rtp_session_process_rtp (RTPSession * sess, RTPSource * src, + GstBuffer * buffer, gpointer user_data) +{ + GstFlowReturn result; + GstRtpSession *rtpsession; + GstPad *rtp_src; + + rtpsession = GST_RTP_SESSION (user_data); + + 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 (rtp_src, buffer); + gst_object_unref (rtp_src); + } else { + GST_DEBUG_OBJECT (rtpsession, "dropping received RTP packet"); + gst_buffer_unref (buffer); + result = GST_FLOW_OK; + } + return result; +} + +/* called when the session manager has an RTP packet ready for further + * sending */ +static GstFlowReturn +gst_rtp_session_send_rtp (RTPSession * sess, RTPSource * src, + gpointer data, gpointer user_data) +{ + GstFlowReturn result; + GstRtpSession *rtpsession; + GstPad *rtp_src; + + rtpsession = GST_RTP_SESSION (user_data); + + 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 (rtp_src, GST_BUFFER_CAST (data)); + } else { + GST_LOG_OBJECT (rtpsession, "sending RTP list"); + 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; + } + return result; +} + +/* called when the session manager has an RTCP packet ready for further + * sending. The eos flag is set when an EOS event should be sent downstream as + * well. */ +static GstFlowReturn +gst_rtp_session_send_rtcp (RTPSession * sess, RTPSource * src, + GstBuffer * buffer, gboolean eos, gpointer user_data) +{ + GstFlowReturn result; + GstRtpSession *rtpsession; + GstPad *rtcp_src; + + rtpsession = GST_RTP_SESSION (user_data); + + 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 */ + if (!(caps = GST_PAD_CAPS (rtcp_src))) { + caps = gst_caps_new_simple ("application/x-rtcp", NULL); + gst_pad_set_caps (rtcp_src, caps); + } else + gst_caps_ref (caps); + gst_buffer_set_caps (buffer, caps); + gst_caps_unref (caps); + + gst_object_ref (rtcp_src); + GST_RTP_SESSION_UNLOCK (rtpsession); + + GST_LOG_OBJECT (rtpsession, "sending RTCP"); + 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 (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 + * inter stream synchronisation */ +static GstFlowReturn +gst_rtp_session_sync_rtcp (RTPSession * sess, RTPSource * src, + GstBuffer * buffer, gpointer user_data) +{ + GstFlowReturn result; + GstRtpSession *rtpsession; + GstPad *sync_src; + + rtpsession = GST_RTP_SESSION (user_data); + + 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 (sync_src))) { + caps = gst_caps_new_simple ("application/x-rtcp", NULL); + gst_pad_set_caps (sync_src, caps); + } else + gst_caps_ref (caps); + gst_buffer_set_caps (buffer, caps); + gst_caps_unref (caps); + + gst_object_ref (sync_src); + GST_RTP_SESSION_UNLOCK (rtpsession); + + GST_LOG_OBJECT (rtpsession, "sending Sync RTCP"); + 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 +gst_rtp_session_cache_caps (GstRtpSession * rtpsession, GstCaps * caps) +{ + GstRtpSessionPrivate *priv; + const GstStructure *s; + gint payload; + + priv = rtpsession->priv; + + GST_DEBUG_OBJECT (rtpsession, "parsing caps"); + + s = gst_caps_get_structure (caps, 0); + if (!gst_structure_get_int (s, "payload", &payload)) + return; + + if (g_hash_table_lookup (priv->ptmap, GINT_TO_POINTER (payload))) + return; + + g_hash_table_insert (priv->ptmap, GINT_TO_POINTER (payload), + gst_caps_ref (caps)); +} + +static GstCaps * +gst_rtp_session_get_caps_for_pt (GstRtpSession * rtpsession, guint payload) +{ + GstCaps *caps = NULL; + GValue args[2] = { {0}, {0} }; + GValue ret = { 0 }; + + GST_RTP_SESSION_LOCK (rtpsession); + caps = g_hash_table_lookup (rtpsession->priv->ptmap, + GINT_TO_POINTER (payload)); + if (caps) { + gst_caps_ref (caps); + goto done; + } + + /* not found in the cache, try to get it with a signal */ + g_value_init (&args[0], GST_TYPE_ELEMENT); + g_value_set_object (&args[0], rtpsession); + g_value_init (&args[1], G_TYPE_UINT); + g_value_set_uint (&args[1], payload); + + g_value_init (&ret, GST_TYPE_CAPS); + g_value_set_boxed (&ret, NULL); + + GST_RTP_SESSION_UNLOCK (rtpsession); + + g_signal_emitv (args, gst_rtp_session_signals[SIGNAL_REQUEST_PT_MAP], 0, + &ret); + + GST_RTP_SESSION_LOCK (rtpsession); + + g_value_unset (&args[0]); + g_value_unset (&args[1]); + caps = (GstCaps *) g_value_dup_boxed (&ret); + g_value_unset (&ret); + if (!caps) + goto no_caps; + + gst_rtp_session_cache_caps (rtpsession, caps); + +done: + GST_RTP_SESSION_UNLOCK (rtpsession); + + return caps; + +no_caps: + { + GST_DEBUG_OBJECT (rtpsession, "could not get caps"); + goto done; + } +} + +/* called when the session manager needs the clock rate */ +static gint +gst_rtp_session_clock_rate (RTPSession * sess, guint8 payload, + gpointer user_data) +{ + gint result = -1; + GstRtpSession *rtpsession; + GstCaps *caps; + const GstStructure *s; + + rtpsession = GST_RTP_SESSION_CAST (user_data); + + caps = gst_rtp_session_get_caps_for_pt (rtpsession, payload); + + if (!caps) + goto done; + + s = gst_caps_get_structure (caps, 0); + if (!gst_structure_get_int (s, "clock-rate", &result)) + goto no_clock_rate; + + gst_caps_unref (caps); + + GST_DEBUG_OBJECT (rtpsession, "parsed clock-rate %d", result); + +done: + + return result; + + /* ERRORS */ +no_clock_rate: + { + gst_caps_unref (caps); + GST_DEBUG_OBJECT (rtpsession, "No clock-rate in caps!"); + goto done; + } +} + +/* called when the session manager asks us to reconsider the timeout */ +static void +gst_rtp_session_reconsider (RTPSession * sess, gpointer user_data) +{ + GstRtpSession *rtpsession; + + rtpsession = GST_RTP_SESSION_CAST (user_data); + + GST_RTP_SESSION_LOCK (rtpsession); + GST_DEBUG_OBJECT (rtpsession, "unlock timer for reconsideration"); + if (rtpsession->priv->id) + gst_clock_id_unschedule (rtpsession->priv->id); + GST_RTP_SESSION_UNLOCK (rtpsession); +} + +static gboolean +gst_rtp_session_event_recv_rtp_sink (GstPad * pad, GstEvent * event) +{ + GstRtpSession *rtpsession; + gboolean ret = FALSE; + + rtpsession = GST_RTP_SESSION (gst_pad_get_parent (pad)); + if (G_UNLIKELY (rtpsession == NULL)) { + gst_event_unref (event); + return FALSE; + } + + GST_DEBUG_OBJECT (rtpsession, "received event %s", + GST_EVENT_TYPE_NAME (event)); + + switch (GST_EVENT_TYPE (event)) { + case GST_EVENT_FLUSH_STOP: + gst_segment_init (&rtpsession->recv_rtp_seg, GST_FORMAT_UNDEFINED); + ret = gst_pad_push_event (rtpsession->recv_rtp_src, event); + break; + case GST_EVENT_NEWSEGMENT: + { + gboolean update; + gdouble rate, arate; + GstFormat format; + gint64 start, stop, time; + GstSegment *segment; + + segment = &rtpsession->recv_rtp_seg; + + /* the newsegment event is needed to convert the RTP timestamp to + * running_time, which is needed to generate a mapping from RTP to NTP + * timestamps in SR reports */ + gst_event_parse_new_segment_full (event, &update, &rate, &arate, &format, + &start, &stop, &time); + + GST_DEBUG_OBJECT (rtpsession, + "configured NEWSEGMENT update %d, rate %lf, applied rate %lf, " + "format GST_FORMAT_TIME, " + "%" GST_TIME_FORMAT " -- %" GST_TIME_FORMAT + ", time %" GST_TIME_FORMAT ", accum %" GST_TIME_FORMAT, + update, rate, arate, GST_TIME_ARGS (segment->start), + GST_TIME_ARGS (segment->stop), GST_TIME_ARGS (segment->time), + GST_TIME_ARGS (segment->accum)); + + gst_segment_set_newsegment_full (segment, update, rate, + arate, format, start, stop, time); + + /* push event forward */ + ret = gst_pad_push_event (rtpsession->recv_rtp_src, event); + break; + } + default: + ret = gst_pad_push_event (rtpsession->recv_rtp_src, event); + break; + } + gst_object_unref (rtpsession); + + return ret; + +} + +static gboolean +gst_rtp_session_request_remote_key_unit (GstRtpSession * rtpsession, + guint32 ssrc, guint payload, gboolean all_headers) +{ + GstCaps *caps; + gboolean requested = FALSE; + + caps = gst_rtp_session_get_caps_for_pt (rtpsession, payload); + + if (caps) { + const GstStructure *s = gst_caps_get_structure (caps, 0); + gboolean pli; + + pli = gst_structure_has_field (s, "rtcp-fb-nack-pli"); + + gst_caps_unref (caps); + + if (pli) { + rtp_session_request_key_unit (rtpsession->priv->session, ssrc); + rtp_session_request_early_rtcp (rtpsession->priv->session, + gst_clock_get_time (rtpsession->priv->sysclock), 200 * GST_MSECOND); + requested = TRUE; + } + } + + return requested; +} + +static gboolean +gst_rtp_session_event_recv_rtp_src (GstPad * pad, GstEvent * event) +{ + GstRtpSession *rtpsession; + gboolean forward = TRUE; + gboolean ret = TRUE; + const GstStructure *s; + guint32 ssrc; + guint pt; + + rtpsession = GST_RTP_SESSION (gst_pad_get_parent (pad)); + if (G_UNLIKELY (rtpsession == NULL)) { + gst_event_unref (event); + return FALSE; + } + + switch (GST_EVENT_TYPE (event)) { + case GST_EVENT_CUSTOM_UPSTREAM: + s = gst_event_get_structure (event); + if (gst_structure_has_name (s, "GstForceKeyUnit") && + gst_structure_get_uint (s, "ssrc", &ssrc) && + gst_structure_get_uint (s, "payload", &pt)) { + gboolean all_headers = FALSE; + + gst_structure_get_boolean (s, "all-headers", &all_headers); + if (gst_rtp_session_request_remote_key_unit (rtpsession, ssrc, pt, + all_headers)) + forward = FALSE; + } + break; + default: + break; + } + + if (forward) + ret = gst_pad_push_event (rtpsession->recv_rtp_sink, event); + + gst_object_unref (rtpsession); + + return ret; +} + + +static GstIterator * +gst_rtp_session_iterate_internal_links (GstPad * pad) +{ + GstRtpSession *rtpsession; + GstPad *otherpad = NULL; + GstIterator *it = NULL; + + rtpsession = GST_RTP_SESSION (gst_pad_get_parent (pad)); + if (G_UNLIKELY (rtpsession == NULL)) + return NULL; + + GST_RTP_SESSION_LOCK (rtpsession); + if (pad == rtpsession->recv_rtp_src) { + otherpad = gst_object_ref (rtpsession->recv_rtp_sink); + } else if (pad == rtpsession->recv_rtp_sink) { + otherpad = gst_object_ref (rtpsession->recv_rtp_src); + } else if (pad == rtpsession->send_rtp_src) { + otherpad = gst_object_ref (rtpsession->send_rtp_sink); + } else if (pad == rtpsession->send_rtp_sink) { + otherpad = gst_object_ref (rtpsession->send_rtp_src); + } + GST_RTP_SESSION_UNLOCK (rtpsession); + + if (otherpad) { + it = gst_iterator_new_single (GST_TYPE_PAD, otherpad, + (GstCopyFunction) gst_object_ref, (GFreeFunc) gst_object_unref); + gst_object_unref (otherpad); + } + + gst_object_unref (rtpsession); + + return it; +} + +static gboolean +gst_rtp_session_sink_setcaps (GstPad * pad, GstCaps * caps) +{ + GstRtpSession *rtpsession; + + rtpsession = GST_RTP_SESSION (gst_pad_get_parent (pad)); + + GST_RTP_SESSION_LOCK (rtpsession); + gst_rtp_session_cache_caps (rtpsession, caps); + GST_RTP_SESSION_UNLOCK (rtpsession); + + gst_object_unref (rtpsession); + + return TRUE; +} + +/* receive a packet from a sender, send it to the RTP session manager and + * forward the packet on the rtp_src pad + */ +static GstFlowReturn +gst_rtp_session_chain_recv_rtp (GstPad * pad, GstBuffer * buffer) +{ + GstRtpSession *rtpsession; + GstRtpSessionPrivate *priv; + GstFlowReturn ret; + GstClockTime current_time, running_time; + GstClockTime timestamp; + + rtpsession = GST_RTP_SESSION (gst_pad_get_parent (pad)); + priv = rtpsession->priv; + + GST_LOG_OBJECT (rtpsession, "received RTP packet"); + + /* get NTP time when this packet was captured, this depends on the timestamp. */ + timestamp = GST_BUFFER_TIMESTAMP (buffer); + if (GST_CLOCK_TIME_IS_VALID (timestamp)) { + /* convert to running time using the segment values */ + running_time = + gst_segment_to_running_time (&rtpsession->recv_rtp_seg, GST_FORMAT_TIME, + timestamp); + } else { + get_current_times (rtpsession, &running_time, NULL); + } + current_time = gst_clock_get_time (priv->sysclock); + + ret = rtp_session_process_rtp (priv->session, buffer, current_time, + running_time); + if (ret != GST_FLOW_OK) + goto push_error; + +done: + gst_object_unref (rtpsession); + + return ret; + + /* ERRORS */ +push_error: + { + GST_DEBUG_OBJECT (rtpsession, "process returned %s", + gst_flow_get_name (ret)); + goto done; + } +} + +static gboolean +gst_rtp_session_event_recv_rtcp_sink (GstPad * pad, GstEvent * event) +{ + GstRtpSession *rtpsession; + gboolean ret = FALSE; + + rtpsession = GST_RTP_SESSION (gst_pad_get_parent (pad)); + + GST_DEBUG_OBJECT (rtpsession, "received event %s", + GST_EVENT_TYPE_NAME (event)); + + switch (GST_EVENT_TYPE (event)) { + default: + ret = gst_pad_push_event (rtpsession->sync_src, event); + break; + } + gst_object_unref (rtpsession); + + return ret; +} + +/* Receive an RTCP packet from a sender, send it to the RTP session manager and + * forward the SR packets to the sync_src pad. + */ +static GstFlowReturn +gst_rtp_session_chain_recv_rtcp (GstPad * pad, GstBuffer * buffer) +{ + GstRtpSession *rtpsession; + GstRtpSessionPrivate *priv; + GstClockTime current_time; + guint64 ntpnstime; + + rtpsession = GST_RTP_SESSION (gst_pad_get_parent (pad)); + priv = rtpsession->priv; + + GST_LOG_OBJECT (rtpsession, "received RTCP packet"); + + current_time = gst_clock_get_time (priv->sysclock); + get_current_times (rtpsession, NULL, &ntpnstime); + + rtp_session_process_rtcp (priv->session, buffer, current_time, ntpnstime); + + gst_object_unref (rtpsession); + + return GST_FLOW_OK; /* always return OK */ +} + +static gboolean +gst_rtp_session_query_send_rtcp_src (GstPad * pad, GstQuery * query) +{ + GstRtpSession *rtpsession; + gboolean ret = FALSE; + + rtpsession = GST_RTP_SESSION (gst_pad_get_parent (pad)); + + GST_DEBUG_OBJECT (rtpsession, "received QUERY"); + + switch (GST_QUERY_TYPE (query)) { + case GST_QUERY_LATENCY: + ret = TRUE; + /* use the defaults for the latency query. */ + gst_query_set_latency (query, FALSE, 0, -1); + break; + default: + /* other queries simply fail for now */ + break; + } + + gst_object_unref (rtpsession); + + return ret; +} + +static gboolean +gst_rtp_session_event_send_rtcp_src (GstPad * pad, GstEvent * event) +{ + GstRtpSession *rtpsession; + gboolean ret = TRUE; + + rtpsession = GST_RTP_SESSION (gst_pad_get_parent (pad)); + if (G_UNLIKELY (rtpsession == NULL)) { + gst_event_unref (event); + return FALSE; + } + GST_DEBUG_OBJECT (rtpsession, "received EVENT"); + + switch (GST_EVENT_TYPE (event)) { + case GST_EVENT_SEEK: + case GST_EVENT_LATENCY: + gst_event_unref (event); + ret = TRUE; + break; + default: + /* other events simply fail for now */ + gst_event_unref (event); + ret = FALSE; + break; + } + + gst_object_unref (rtpsession); + return ret; +} + + +static gboolean +gst_rtp_session_event_send_rtp_sink (GstPad * pad, GstEvent * event) +{ + GstRtpSession *rtpsession; + gboolean ret = FALSE; + + rtpsession = GST_RTP_SESSION (gst_pad_get_parent (pad)); + + GST_DEBUG_OBJECT (rtpsession, "received event"); + + switch (GST_EVENT_TYPE (event)) { + case GST_EVENT_FLUSH_STOP: + gst_segment_init (&rtpsession->send_rtp_seg, GST_FORMAT_UNDEFINED); + ret = gst_pad_push_event (rtpsession->send_rtp_src, event); + break; + case GST_EVENT_NEWSEGMENT:{ + gboolean update; + gdouble rate, arate; + GstFormat format; + gint64 start, stop, time; + GstSegment *segment; + + segment = &rtpsession->send_rtp_seg; + + /* the newsegment event is needed to convert the RTP timestamp to + * running_time, which is needed to generate a mapping from RTP to NTP + * timestamps in SR reports */ + gst_event_parse_new_segment_full (event, &update, &rate, &arate, &format, + &start, &stop, &time); + + GST_DEBUG_OBJECT (rtpsession, + "configured NEWSEGMENT update %d, rate %lf, applied rate %lf, " + "format GST_FORMAT_TIME, " + "%" GST_TIME_FORMAT " -- %" GST_TIME_FORMAT + ", time %" GST_TIME_FORMAT ", accum %" GST_TIME_FORMAT, + update, rate, arate, GST_TIME_ARGS (segment->start), + GST_TIME_ARGS (segment->stop), GST_TIME_ARGS (segment->time), + GST_TIME_ARGS (segment->accum)); + + gst_segment_set_newsegment_full (segment, update, rate, + arate, format, start, stop, time); + + /* push event forward */ + ret = gst_pad_push_event (rtpsession->send_rtp_src, event); + break; + } + case GST_EVENT_EOS:{ + GstClockTime current_time; + + /* push downstream FIXME, we are not supposed to leave the session just + * because we stop sending. */ + ret = gst_pad_push_event (rtpsession->send_rtp_src, event); + current_time = gst_clock_get_time (rtpsession->priv->sysclock); + GST_DEBUG_OBJECT (rtpsession, "scheduling BYE message"); + rtp_session_schedule_bye (rtpsession->priv->session, "End of stream", + current_time); + break; + } + default:{ + GstPad *send_rtp_src = NULL; + GST_RTP_SESSION_LOCK (rtpsession); + if (rtpsession->send_rtp_src) + send_rtp_src = gst_object_ref (rtpsession->send_rtp_src); + GST_RTP_SESSION_UNLOCK (rtpsession); + + if (send_rtp_src) { + ret = gst_pad_push_event (send_rtp_src, event); + gst_object_unref (send_rtp_src); + } else + gst_event_unref (event); + + break; + } + } + gst_object_unref (rtpsession); + + return ret; +} + +static GstCaps * +gst_rtp_session_getcaps_send_rtp (GstPad * pad) +{ + GstRtpSession *rtpsession; + GstRtpSessionPrivate *priv; + GstCaps *result; + GstStructure *s1, *s2; + guint ssrc; + + rtpsession = GST_RTP_SESSION (gst_pad_get_parent (pad)); + priv = rtpsession->priv; + + ssrc = rtp_session_get_internal_ssrc (priv->session); + + /* we can basically accept anything but we prefer to receive packets with our + * internal SSRC so that we don't have to patch it. Create a structure with + * the SSRC and another one without. */ + s1 = gst_structure_new ("application/x-rtp", "ssrc", G_TYPE_UINT, ssrc, NULL); + s2 = gst_structure_new ("application/x-rtp", NULL); + + result = gst_caps_new_full (s1, s2, NULL); + + GST_DEBUG_OBJECT (rtpsession, "getting caps %" GST_PTR_FORMAT, result); + + gst_object_unref (rtpsession); + + return result; +} + +static gboolean +gst_rtp_session_setcaps_send_rtp (GstPad * pad, GstCaps * caps) +{ + GstRtpSession *rtpsession; + GstRtpSessionPrivate *priv; + GstStructure *s = gst_caps_get_structure (caps, 0); + guint ssrc; + + rtpsession = GST_RTP_SESSION (gst_pad_get_parent (pad)); + priv = rtpsession->priv; + + if (gst_structure_get_uint (s, "ssrc", &ssrc)) { + GST_DEBUG_OBJECT (rtpsession, "setting internal SSRC to %08x", ssrc); + rtp_session_set_internal_ssrc (priv->session, ssrc); + } + + gst_object_unref (rtpsession); + + return TRUE; +} + +/* Recieve an RTP packet or a list of packets to be send to the receivers, + * send to RTP session manager and forward to send_rtp_src. + */ +static GstFlowReturn +gst_rtp_session_chain_send_rtp_common (GstPad * pad, gpointer data, + gboolean is_list) +{ + GstRtpSession *rtpsession; + GstRtpSessionPrivate *priv; + GstFlowReturn ret; + GstClockTime timestamp, running_time; + GstClockTime current_time; + + rtpsession = GST_RTP_SESSION (gst_pad_get_parent (pad)); + priv = rtpsession->priv; + + GST_LOG_OBJECT (rtpsession, "received RTP %s", is_list ? "list" : "packet"); + + /* get NTP time when this packet was captured, this depends on the timestamp. */ + if (is_list) { + GstBuffer *buffer = NULL; + + /* All groups in an list have the same timestamp. + * So, just take it from the first group. */ + buffer = gst_buffer_list_get (GST_BUFFER_LIST_CAST (data), 0, 0); + if (buffer) + timestamp = GST_BUFFER_TIMESTAMP (buffer); + else + timestamp = -1; + } else { + timestamp = GST_BUFFER_TIMESTAMP (GST_BUFFER_CAST (data)); + } + + if (GST_CLOCK_TIME_IS_VALID (timestamp)) { + /* convert to running time using the segment start value. */ + running_time = + gst_segment_to_running_time (&rtpsession->send_rtp_seg, GST_FORMAT_TIME, + timestamp); + } else { + /* no timestamp. */ + running_time = -1; + } + + current_time = gst_clock_get_time (priv->sysclock); + ret = rtp_session_send_rtp (priv->session, data, is_list, current_time, + running_time); + if (ret != GST_FLOW_OK) + goto push_error; + +done: + gst_object_unref (rtpsession); + + return ret; + + /* ERRORS */ +push_error: + { + GST_DEBUG_OBJECT (rtpsession, "process returned %s", + gst_flow_get_name (ret)); + goto done; + } +} + +static GstFlowReturn +gst_rtp_session_chain_send_rtp (GstPad * pad, GstBuffer * buffer) +{ + return gst_rtp_session_chain_send_rtp_common (pad, buffer, FALSE); +} + +static GstFlowReturn +gst_rtp_session_chain_send_rtp_list (GstPad * pad, GstBufferList * list) +{ + return gst_rtp_session_chain_send_rtp_common (pad, list, TRUE); +} + +/* Create sinkpad to receive RTP packets from senders. This will also create a + * srcpad for the RTP packets. + */ +static GstPad * +create_recv_rtp_sink (GstRtpSession * rtpsession) +{ + GST_DEBUG_OBJECT (rtpsession, "creating RTP sink pad"); + + rtpsession->recv_rtp_sink = + gst_pad_new_from_static_template (&rtpsession_recv_rtp_sink_template, + "recv_rtp_sink"); + gst_pad_set_chain_function (rtpsession->recv_rtp_sink, + gst_rtp_session_chain_recv_rtp); + gst_pad_set_event_function (rtpsession->recv_rtp_sink, + (GstPadEventFunction) gst_rtp_session_event_recv_rtp_sink); + gst_pad_set_setcaps_function (rtpsession->recv_rtp_sink, + gst_rtp_session_sink_setcaps); + gst_pad_set_iterate_internal_links_function (rtpsession->recv_rtp_sink, + gst_rtp_session_iterate_internal_links); + gst_pad_set_active (rtpsession->recv_rtp_sink, TRUE); + gst_element_add_pad (GST_ELEMENT_CAST (rtpsession), + rtpsession->recv_rtp_sink); + + GST_DEBUG_OBJECT (rtpsession, "creating RTP src pad"); + rtpsession->recv_rtp_src = + gst_pad_new_from_static_template (&rtpsession_recv_rtp_src_template, + "recv_rtp_src"); + gst_pad_set_event_function (rtpsession->recv_rtp_src, + (GstPadEventFunction) gst_rtp_session_event_recv_rtp_src); + gst_pad_set_iterate_internal_links_function (rtpsession->recv_rtp_src, + gst_rtp_session_iterate_internal_links); + gst_pad_use_fixed_caps (rtpsession->recv_rtp_src); + gst_pad_set_active (rtpsession->recv_rtp_src, TRUE); + gst_element_add_pad (GST_ELEMENT_CAST (rtpsession), rtpsession->recv_rtp_src); + + return rtpsession->recv_rtp_sink; +} + +/* Remove sinkpad to receive RTP packets from senders. This will also remove + * the srcpad for the RTP packets. + */ +static void +remove_recv_rtp_sink (GstRtpSession * rtpsession) +{ + GST_DEBUG_OBJECT (rtpsession, "removing RTP sink pad"); + + /* deactivate from source to sink */ + gst_pad_set_active (rtpsession->recv_rtp_src, FALSE); + gst_pad_set_active (rtpsession->recv_rtp_sink, FALSE); + + /* remove pads */ + gst_element_remove_pad (GST_ELEMENT_CAST (rtpsession), + rtpsession->recv_rtp_sink); + rtpsession->recv_rtp_sink = NULL; + + GST_DEBUG_OBJECT (rtpsession, "removing RTP src pad"); + gst_element_remove_pad (GST_ELEMENT_CAST (rtpsession), + rtpsession->recv_rtp_src); + rtpsession->recv_rtp_src = NULL; +} + +/* Create a sinkpad to receive RTCP messages from senders, this will also create a + * sync_src pad for the SR packets. + */ +static GstPad * +create_recv_rtcp_sink (GstRtpSession * rtpsession) +{ + GST_DEBUG_OBJECT (rtpsession, "creating RTCP sink pad"); + + rtpsession->recv_rtcp_sink = + gst_pad_new_from_static_template (&rtpsession_recv_rtcp_sink_template, + "recv_rtcp_sink"); + gst_pad_set_chain_function (rtpsession->recv_rtcp_sink, + gst_rtp_session_chain_recv_rtcp); + gst_pad_set_event_function (rtpsession->recv_rtcp_sink, + (GstPadEventFunction) gst_rtp_session_event_recv_rtcp_sink); + gst_pad_set_iterate_internal_links_function (rtpsession->recv_rtcp_sink, + gst_rtp_session_iterate_internal_links); + gst_pad_set_active (rtpsession->recv_rtcp_sink, TRUE); + gst_element_add_pad (GST_ELEMENT_CAST (rtpsession), + rtpsession->recv_rtcp_sink); + + GST_DEBUG_OBJECT (rtpsession, "creating sync src pad"); + rtpsession->sync_src = + gst_pad_new_from_static_template (&rtpsession_sync_src_template, + "sync_src"); + gst_pad_set_iterate_internal_links_function (rtpsession->sync_src, + gst_rtp_session_iterate_internal_links); + gst_pad_use_fixed_caps (rtpsession->sync_src); + gst_pad_set_active (rtpsession->sync_src, TRUE); + gst_element_add_pad (GST_ELEMENT_CAST (rtpsession), rtpsession->sync_src); + + return rtpsession->recv_rtcp_sink; +} + +static void +remove_recv_rtcp_sink (GstRtpSession * rtpsession) +{ + GST_DEBUG_OBJECT (rtpsession, "removing RTCP sink pad"); + + gst_pad_set_active (rtpsession->sync_src, FALSE); + gst_pad_set_active (rtpsession->recv_rtcp_sink, FALSE); + + gst_element_remove_pad (GST_ELEMENT_CAST (rtpsession), + rtpsession->recv_rtcp_sink); + rtpsession->recv_rtcp_sink = NULL; + + GST_DEBUG_OBJECT (rtpsession, "removing sync src pad"); + gst_element_remove_pad (GST_ELEMENT_CAST (rtpsession), rtpsession->sync_src); + rtpsession->sync_src = NULL; +} + +/* Create a sinkpad to receive RTP packets for receivers. This will also create a + * send_rtp_src pad. + */ +static GstPad * +create_send_rtp_sink (GstRtpSession * rtpsession) +{ + GST_DEBUG_OBJECT (rtpsession, "creating pad"); + + rtpsession->send_rtp_sink = + gst_pad_new_from_static_template (&rtpsession_send_rtp_sink_template, + "send_rtp_sink"); + gst_pad_set_chain_function (rtpsession->send_rtp_sink, + gst_rtp_session_chain_send_rtp); + gst_pad_set_chain_list_function (rtpsession->send_rtp_sink, + gst_rtp_session_chain_send_rtp_list); + gst_pad_set_getcaps_function (rtpsession->send_rtp_sink, + gst_rtp_session_getcaps_send_rtp); + gst_pad_set_setcaps_function (rtpsession->send_rtp_sink, + gst_rtp_session_setcaps_send_rtp); + gst_pad_set_event_function (rtpsession->send_rtp_sink, + (GstPadEventFunction) gst_rtp_session_event_send_rtp_sink); + gst_pad_set_iterate_internal_links_function (rtpsession->send_rtp_sink, + gst_rtp_session_iterate_internal_links); + gst_pad_set_active (rtpsession->send_rtp_sink, TRUE); + gst_element_add_pad (GST_ELEMENT_CAST (rtpsession), + rtpsession->send_rtp_sink); + + rtpsession->send_rtp_src = + gst_pad_new_from_static_template (&rtpsession_send_rtp_src_template, + "send_rtp_src"); + gst_pad_set_iterate_internal_links_function (rtpsession->send_rtp_src, + gst_rtp_session_iterate_internal_links); + gst_pad_set_active (rtpsession->send_rtp_src, TRUE); + gst_element_add_pad (GST_ELEMENT_CAST (rtpsession), rtpsession->send_rtp_src); + + return rtpsession->send_rtp_sink; +} + +static void +remove_send_rtp_sink (GstRtpSession * rtpsession) +{ + GST_DEBUG_OBJECT (rtpsession, "removing pad"); + + gst_pad_set_active (rtpsession->send_rtp_src, FALSE); + gst_pad_set_active (rtpsession->send_rtp_sink, FALSE); + + gst_element_remove_pad (GST_ELEMENT_CAST (rtpsession), + rtpsession->send_rtp_sink); + rtpsession->send_rtp_sink = NULL; + + gst_element_remove_pad (GST_ELEMENT_CAST (rtpsession), + rtpsession->send_rtp_src); + rtpsession->send_rtp_src = NULL; +} + +/* Create a srcpad with the RTCP packets to send out. + * This pad will be driven by the RTP session manager when it wants to send out + * RTCP packets. + */ +static GstPad * +create_send_rtcp_src (GstRtpSession * rtpsession) +{ + GST_DEBUG_OBJECT (rtpsession, "creating pad"); + + rtpsession->send_rtcp_src = + gst_pad_new_from_static_template (&rtpsession_send_rtcp_src_template, + "send_rtcp_src"); + gst_pad_use_fixed_caps (rtpsession->send_rtcp_src); + gst_pad_set_active (rtpsession->send_rtcp_src, TRUE); + gst_pad_set_iterate_internal_links_function (rtpsession->send_rtcp_src, + gst_rtp_session_iterate_internal_links); + gst_pad_set_query_function (rtpsession->send_rtcp_src, + gst_rtp_session_query_send_rtcp_src); + gst_pad_set_event_function (rtpsession->send_rtcp_src, + gst_rtp_session_event_send_rtcp_src); + gst_element_add_pad (GST_ELEMENT_CAST (rtpsession), + rtpsession->send_rtcp_src); + + return rtpsession->send_rtcp_src; +} + +static void +remove_send_rtcp_src (GstRtpSession * rtpsession) +{ + GST_DEBUG_OBJECT (rtpsession, "removing pad"); + + gst_pad_set_active (rtpsession->send_rtcp_src, FALSE); + + gst_element_remove_pad (GST_ELEMENT_CAST (rtpsession), + rtpsession->send_rtcp_src); + rtpsession->send_rtcp_src = NULL; +} + +static GstPad * +gst_rtp_session_request_new_pad (GstElement * element, + GstPadTemplate * templ, const gchar * name) +{ + GstRtpSession *rtpsession; + GstElementClass *klass; + GstPad *result; + + g_return_val_if_fail (templ != NULL, NULL); + g_return_val_if_fail (GST_IS_RTP_SESSION (element), NULL); + + rtpsession = GST_RTP_SESSION (element); + klass = GST_ELEMENT_GET_CLASS (element); + + GST_DEBUG_OBJECT (element, "requesting pad %s", GST_STR_NULL (name)); + + GST_RTP_SESSION_LOCK (rtpsession); + + /* figure out the template */ + if (templ == gst_element_class_get_pad_template (klass, "recv_rtp_sink")) { + if (rtpsession->recv_rtp_sink != NULL) + goto exists; + + result = create_recv_rtp_sink (rtpsession); + } else if (templ == gst_element_class_get_pad_template (klass, + "recv_rtcp_sink")) { + if (rtpsession->recv_rtcp_sink != NULL) + goto exists; + + result = create_recv_rtcp_sink (rtpsession); + } else if (templ == gst_element_class_get_pad_template (klass, + "send_rtp_sink")) { + if (rtpsession->send_rtp_sink != NULL) + goto exists; + + result = create_send_rtp_sink (rtpsession); + } else if (templ == gst_element_class_get_pad_template (klass, + "send_rtcp_src")) { + if (rtpsession->send_rtcp_src != NULL) + goto exists; + + result = create_send_rtcp_src (rtpsession); + } else + goto wrong_template; + + GST_RTP_SESSION_UNLOCK (rtpsession); + + return result; + + /* ERRORS */ +wrong_template: + { + GST_RTP_SESSION_UNLOCK (rtpsession); + g_warning ("gstrtpsession: this is not our template"); + return NULL; + } +exists: + { + GST_RTP_SESSION_UNLOCK (rtpsession); + g_warning ("gstrtpsession: pad already requested"); + return NULL; + } +} + +static void +gst_rtp_session_release_pad (GstElement * element, GstPad * pad) +{ + GstRtpSession *rtpsession; + + g_return_if_fail (GST_IS_RTP_SESSION (element)); + g_return_if_fail (GST_IS_PAD (pad)); + + rtpsession = GST_RTP_SESSION (element); + + GST_DEBUG_OBJECT (element, "releasing pad %s:%s", GST_DEBUG_PAD_NAME (pad)); + + GST_RTP_SESSION_LOCK (rtpsession); + + if (rtpsession->recv_rtp_sink == pad) { + remove_recv_rtp_sink (rtpsession); + } else if (rtpsession->recv_rtcp_sink == pad) { + remove_recv_rtcp_sink (rtpsession); + } else if (rtpsession->send_rtp_sink == pad) { + remove_send_rtp_sink (rtpsession); + } else if (rtpsession->send_rtcp_src == pad) { + remove_send_rtcp_src (rtpsession); + } else + goto wrong_pad; + + GST_RTP_SESSION_UNLOCK (rtpsession); + + return; + + /* ERRORS */ +wrong_pad: + { + GST_RTP_SESSION_UNLOCK (rtpsession); + g_warning ("gstrtpsession: asked to release an unknown pad"); + return; + } +} + +static void +gst_rtp_session_request_key_unit (RTPSession * sess, + gboolean all_headers, gpointer user_data) +{ + GstRtpSession *rtpsession = GST_RTP_SESSION (user_data); + GstEvent *event; + + event = gst_event_new_custom (GST_EVENT_CUSTOM_UPSTREAM, + gst_structure_new ("GstForceKeyUnit", + "all-headers", G_TYPE_BOOLEAN, all_headers, NULL)); + gst_pad_push_event (rtpsession->send_rtp_sink, event); +} + +static GstClockTime +gst_rtp_session_request_time (RTPSession * session, gpointer user_data) +{ + GstRtpSession *rtpsession = GST_RTP_SESSION (user_data); + + return gst_clock_get_time (rtpsession->priv->sysclock); +} |