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Diffstat (limited to 'gst/rtpmanager/gstrtpsession.c')
| -rw-r--r-- | gst/rtpmanager/gstrtpsession.c | 1940 |
1 files changed, 0 insertions, 1940 deletions
diff --git a/gst/rtpmanager/gstrtpsession.c b/gst/rtpmanager/gstrtpsession.c deleted file mode 100644 index dcddb689..00000000 --- a/gst/rtpmanager/gstrtpsession.c +++ /dev/null @@ -1,1940 +0,0 @@ -/* 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 - -/* elementfactory information */ -static const GstElementDetails rtpsession_details = -GST_ELEMENT_DETAILS ("RTP Session", - "Filter/Network/RTP", - "Implement an RTP session", - "Wim Taymans <wim.taymans@gmail.com>"); - -/* 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_RTCP_BANDWIDTH -#define DEFAULT_SDES NULL -#define DEFAULT_NUM_SOURCES 0 -#define DEFAULT_NUM_ACTIVE_SOURCES 0 - -enum -{ - PROP_0, - PROP_NTP_NS_BASE, - PROP_BANDWIDTH, - PROP_RTCP_FRACTION, - PROP_SDES, - PROP_NUM_SOURCES, - PROP_NUM_ACTIVE_SOURCES, - PROP_INTERNAL_SESSION, - 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; -}; - -/* 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 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 -}; - -/* 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 (element_class, &rtpsession_details); -} - -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", 0, - G_MAXUINT64, DEFAULT_NTP_NS_BASE, G_PARAM_READWRITE)); - - g_object_class_install_property (gobject_class, PROP_BANDWIDTH, - g_param_spec_double ("bandwidth", "Bandwidth", - "The bandwidth of the session", - 0.0, G_MAXDOUBLE, DEFAULT_BANDWIDTH, G_PARAM_READWRITE)); - - g_object_class_install_property (gobject_class, PROP_RTCP_FRACTION, - g_param_spec_double ("rtcp-fraction", "RTCP Fraction", - "The fraction of the bandwidth used for RTCP", - 0.0, G_MAXDOUBLE, DEFAULT_RTCP_FRACTION, G_PARAM_READWRITE)); - - 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_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_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_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)); - - 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 (); - - /* 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); - - if (rtpsession->recv_rtp_sink != NULL) - gst_object_unref (rtpsession->recv_rtp_sink); - if (rtpsession->recv_rtcp_sink != NULL) - gst_object_unref (rtpsession->recv_rtcp_sink); - if (rtpsession->send_rtp_sink != NULL) - gst_object_unref (rtpsession->send_rtp_sink); - if (rtpsession->send_rtcp_src != NULL) - gst_object_unref (rtpsession->send_rtcp_src); - - 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: - rtp_session_set_bandwidth (priv->session, g_value_get_double (value)); - break; - case PROP_RTCP_FRACTION: - rtp_session_set_rtcp_fraction (priv->session, g_value_get_double (value)); - break; - case PROP_SDES: - rtp_session_set_sdes_struct (priv->session, g_value_get_boxed (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_value_set_double (value, rtp_session_get_bandwidth (priv->session)); - break; - case PROP_RTCP_FRACTION: - g_value_set_double (value, rtp_session_get_rtcp_fraction (priv->session)); - 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; - 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, ntpnsbase, rt; - - GST_OBJECT_LOCK (rtpsession); - if ((clock = GST_ELEMENT_CLOCK (rtpsession))) { - base_time = GST_ELEMENT_CAST (rtpsession)->base_time; - ntpnsbase = rtpsession->priv->ntpnsbase; - gst_object_ref (clock); - GST_OBJECT_UNLOCK (rtpsession); - - /* get current clock time and convert to running time */ - rt = gst_clock_get_time (clock) - base_time; - /* add NTP base offset to get NTP ns time */ - ntpns = rt + ntpnsbase; - - 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; - - GST_DEBUG_OBJECT (rtpsession, "entering RTCP thread"); - - GST_RTP_SESSION_LOCK (rtpsession); - - current_time = gst_clock_get_time (rtpsession->priv->sysclock); - - while (!rtpsession->priv->stop_thread) { - GstClockReturn res; - - /* get initial estimate */ - 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)); - - /* 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 (rtpsession->priv->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 (rtpsession->priv->sysclock); - - /* get current NTP time */ - get_current_times (rtpsession, NULL, &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 (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_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; - GstRtpSessionPrivate *priv; - - rtpsession = GST_RTP_SESSION (element); - priv = rtpsession->priv; - - 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; - GstRtpSessionPrivate *priv; - - rtpsession = GST_RTP_SESSION (user_data); - priv = rtpsession->priv; - - if (rtpsession->recv_rtp_src) { - GST_LOG_OBJECT (rtpsession, "pushing received RTP packet"); - result = gst_pad_push (rtpsession->recv_rtp_src, buffer); - } 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; - GstRtpSessionPrivate *priv; - - rtpsession = GST_RTP_SESSION (user_data); - priv = rtpsession->priv; - - if (rtpsession->send_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)); - } else { - GST_LOG_OBJECT (rtpsession, "sending RTP list"); - result = gst_pad_push_list (rtpsession->send_rtp_src, - GST_BUFFER_LIST_CAST (data)); - } - } 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; - GstRtpSessionPrivate *priv; - - rtpsession = GST_RTP_SESSION (user_data); - priv = rtpsession->priv; - - if (rtpsession->send_rtcp_src) { - GstCaps *caps; - - /* set rtcp caps on output pad */ - if (!(caps = GST_PAD_CAPS (rtpsession->send_rtcp_src))) { - caps = gst_caps_new_simple ("application/x-rtcp", NULL); - gst_pad_set_caps (rtpsession->send_rtcp_src, caps); - gst_caps_unref (caps); - } - gst_buffer_set_caps (buffer, caps); - GST_LOG_OBJECT (rtpsession, "sending RTCP"); - result = gst_pad_push (rtpsession->send_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 ()); - } - } else { - GST_DEBUG_OBJECT (rtpsession, "not sending RTCP, no output pad"); - gst_buffer_unref (buffer); - result = GST_FLOW_OK; - } - return result; -} - -/* 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; - GstRtpSessionPrivate *priv; - - rtpsession = GST_RTP_SESSION (user_data); - priv = rtpsession->priv; - - if (rtpsession->sync_src) { - GstCaps *caps; - - /* set rtcp caps on output pad */ - if (!(caps = GST_PAD_CAPS (rtpsession->sync_src))) { - caps = gst_caps_new_simple ("application/x-rtcp", NULL); - gst_pad_set_caps (rtpsession->sync_src, caps); - gst_caps_unref (caps); - } - gst_buffer_set_caps (buffer, caps); - GST_LOG_OBJECT (rtpsession, "sending Sync RTCP"); - result = gst_pad_push (rtpsession->sync_src, buffer); - } else { - GST_DEBUG_OBJECT (rtpsession, "not sending Sync RTCP, no output pad"); - gst_buffer_unref (buffer); - result = GST_FLOW_OK; - } - return result; -} - -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)); -} - -/* called when the session manager needs the clock rate */ -static gint -gst_rtp_session_clock_rate (RTPSession * sess, guint8 payload, - gpointer user_data) -{ - gint ipayload, result = -1; - GstRtpSession *rtpsession; - GstRtpSessionPrivate *priv; - GValue ret = { 0 }; - GValue args[2] = { {0}, {0} }; - GstCaps *caps; - const GstStructure *s; - - rtpsession = GST_RTP_SESSION_CAST (user_data); - priv = rtpsession->priv; - - GST_RTP_SESSION_LOCK (rtpsession); - ipayload = payload; /* make compiler happy */ - caps = g_hash_table_lookup (priv->ptmap, GINT_TO_POINTER (ipayload)); - if (caps) { - gst_caps_ref (caps); - goto found; - } - - /* 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); - - g_signal_emitv (args, gst_rtp_session_signals[SIGNAL_REQUEST_PT_MAP], 0, - &ret); - - 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); - -found: - 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: - GST_RTP_SESSION_UNLOCK (rtpsession); - - return result; - - /* ERRORS */ -no_caps: - { - GST_DEBUG_OBJECT (rtpsession, "could not get caps"); - goto done; - } -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; - GstRtpSessionPrivate *priv; - gboolean ret = FALSE; - - rtpsession = GST_RTP_SESSION (gst_pad_get_parent (pad)); - priv = rtpsession->priv; - - 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 GList * -gst_rtp_session_internal_links (GstPad * pad) -{ - GstRtpSession *rtpsession; - GstRtpSessionPrivate *priv; - GList *res = NULL; - - rtpsession = GST_RTP_SESSION (gst_pad_get_parent (pad)); - priv = rtpsession->priv; - - if (pad == rtpsession->recv_rtp_src) { - res = g_list_prepend (res, rtpsession->recv_rtp_sink); - } else if (pad == rtpsession->recv_rtp_sink) { - res = g_list_prepend (res, rtpsession->recv_rtp_src); - } else if (pad == rtpsession->send_rtp_src) { - res = g_list_prepend (res, rtpsession->send_rtp_sink); - } else if (pad == rtpsession->send_rtp_sink) { - res = g_list_prepend (res, rtpsession->send_rtp_src); - } - - gst_object_unref (rtpsession); - - return res; -} - -static gboolean -gst_rtp_session_sink_setcaps (GstPad * pad, GstCaps * caps) -{ - GstRtpSession *rtpsession; - GstRtpSessionPrivate *priv; - - rtpsession = GST_RTP_SESSION (gst_pad_get_parent (pad)); - priv = rtpsession->priv; - - 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; - guint64 ntpnstime; - 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); - /* add constant to convert running time to NTP time */ - ntpnstime = running_time + priv->ntpnsbase; - } else { - get_current_times (rtpsession, &running_time, &ntpnstime); - } - current_time = gst_clock_get_time (priv->sysclock); - - ret = rtp_session_process_rtp (priv->session, buffer, current_time, - running_time, ntpnstime); - 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; - GstRtpSessionPrivate *priv; - gboolean ret = FALSE; - - rtpsession = GST_RTP_SESSION (gst_pad_get_parent (pad)); - priv = rtpsession->priv; - - 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; - GstFlowReturn ret; - - 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); - ret = rtp_session_process_rtcp (priv->session, buffer, current_time); - - gst_object_unref (rtpsession); - - return GST_FLOW_OK; -} - -static gboolean -gst_rtp_session_query_send_rtcp_src (GstPad * pad, GstQuery * query) -{ - GstRtpSession *rtpsession; - GstRtpSessionPrivate *priv; - gboolean ret = FALSE; - - rtpsession = GST_RTP_SESSION (gst_pad_get_parent (pad)); - priv = rtpsession->priv; - - 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; - GstRtpSessionPrivate *priv; - gboolean ret; - - rtpsession = GST_RTP_SESSION (gst_pad_get_parent (pad)); - priv = rtpsession->priv; - - 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; - GstRtpSessionPrivate *priv; - gboolean ret = FALSE; - - rtpsession = GST_RTP_SESSION (gst_pad_get_parent (pad)); - priv = rtpsession->priv; - - 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: - ret = gst_pad_push_event (rtpsession->send_rtp_src, 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; - GstClockTime current_time; - guint64 ntpnstime; - - 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. */ - ntpnstime = - gst_segment_to_running_time (&rtpsession->send_rtp_seg, GST_FORMAT_TIME, - timestamp); - /* convert to NTP time by adding the NTP base */ - ntpnstime += priv->ntpnsbase; - } else { - /* no timestamp, we could take the current running_time and convert it to - * NTP time. */ - ntpnstime = -1; - } - - current_time = gst_clock_get_time (priv->sysclock); - ret = - rtp_session_send_rtp (priv->session, data, is_list, current_time, - ntpnstime); - 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_internal_link_function (rtpsession->recv_rtp_sink, - gst_rtp_session_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_internal_link_function (rtpsession->recv_rtp_src, - gst_rtp_session_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_internal_link_function (rtpsession->recv_rtcp_sink, - gst_rtp_session_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_internal_link_function (rtpsession->sync_src, - gst_rtp_session_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_internal_link_function (rtpsession->send_rtp_sink, - gst_rtp_session_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_internal_link_function (rtpsession->send_rtp_src, - gst_rtp_session_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_internal_link_function (rtpsession->send_rtcp_src, - gst_rtp_session_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; - } -} |