/* GStreamer * Copyright (C) <2007> Wim Taymans * * 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. */ /* FIXME 0.11: suppress warnings for deprecated API such as GValueArray * with newer GLib versions (>= 2.31.0) */ #define GLIB_DISABLE_DEPRECATION_WARNINGS #include #include #include #include #include "gstrtpbin-marshal.h" #include "rtpsession.h" GST_DEBUG_CATEGORY_STATIC (rtp_session_debug); #define GST_CAT_DEFAULT rtp_session_debug /* signals and args */ enum { SIGNAL_GET_SOURCE_BY_SSRC, 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, SIGNAL_ON_SENDING_RTCP, SIGNAL_ON_FEEDBACK_RTCP, SIGNAL_SEND_RTCP, LAST_SIGNAL }; #define DEFAULT_INTERNAL_SOURCE NULL #define DEFAULT_BANDWIDTH RTP_STATS_BANDWIDTH #define DEFAULT_RTCP_FRACTION (RTP_STATS_RTCP_FRACTION * RTP_STATS_BANDWIDTH) #define DEFAULT_RTCP_RR_BANDWIDTH -1 #define DEFAULT_RTCP_RS_BANDWIDTH -1 #define DEFAULT_RTCP_MTU 1400 #define DEFAULT_SDES NULL #define DEFAULT_NUM_SOURCES 0 #define DEFAULT_NUM_ACTIVE_SOURCES 0 #define DEFAULT_SOURCES NULL #define DEFAULT_RTCP_MIN_INTERVAL (RTP_STATS_MIN_INTERVAL * GST_SECOND) #define DEFAULT_RTCP_FEEDBACK_RETENTION_WINDOW (2 * GST_SECOND) #define DEFAULT_RTCP_IMMEDIATE_FEEDBACK_THRESHOLD (3) #define DEFAULT_PROBATION RTP_DEFAULT_PROBATION enum { PROP_0, PROP_INTERNAL_SSRC, PROP_INTERNAL_SOURCE, PROP_BANDWIDTH, PROP_RTCP_FRACTION, PROP_RTCP_RR_BANDWIDTH, PROP_RTCP_RS_BANDWIDTH, PROP_RTCP_MTU, PROP_SDES, PROP_NUM_SOURCES, PROP_NUM_ACTIVE_SOURCES, PROP_SOURCES, PROP_FAVOR_NEW, PROP_RTCP_MIN_INTERVAL, PROP_RTCP_FEEDBACK_RETENTION_WINDOW, PROP_RTCP_IMMEDIATE_FEEDBACK_THRESHOLD, PROP_PROBATION, PROP_LAST }; /* update average packet size */ #define INIT_AVG(avg, val) \ (avg) = (val); #define UPDATE_AVG(avg, val) \ if ((avg) == 0) \ (avg) = (val); \ else \ (avg) = ((val) + (15 * (avg))) >> 4; /* The number RTCP intervals after which to timeout entries in the * collision table */ #define RTCP_INTERVAL_COLLISION_TIMEOUT 10 /* GObject vmethods */ static void rtp_session_finalize (GObject * object); static void rtp_session_set_property (GObject * object, guint prop_id, const GValue * value, GParamSpec * pspec); static void rtp_session_get_property (GObject * object, guint prop_id, GValue * value, GParamSpec * pspec); static gboolean rtp_session_on_sending_rtcp (RTPSession * sess, GstBuffer * buffer, gboolean early); static void rtp_session_send_rtcp (RTPSession * sess, GstClockTimeDiff max_delay); static guint rtp_session_signals[LAST_SIGNAL] = { 0 }; G_DEFINE_TYPE (RTPSession, rtp_session, G_TYPE_OBJECT); static RTPSource *obtain_source (RTPSession * sess, guint32 ssrc, gboolean * created, RTPArrivalStats * arrival, gboolean rtp); static GstFlowReturn rtp_session_schedule_bye_locked (RTPSession * sess, const gchar * reason, GstClockTime current_time); static GstClockTime calculate_rtcp_interval (RTPSession * sess, gboolean deterministic, gboolean first); static gboolean accumulate_trues (GSignalInvocationHint * ihint, GValue * return_accu, const GValue * handler_return, gpointer data) { if (g_value_get_boolean (handler_return)) g_value_set_boolean (return_accu, TRUE); return TRUE; } static void rtp_session_class_init (RTPSessionClass * klass) { GObjectClass *gobject_class; gobject_class = (GObjectClass *) klass; gobject_class->finalize = rtp_session_finalize; gobject_class->set_property = rtp_session_set_property; gobject_class->get_property = rtp_session_get_property; /** * RTPSession::get-source-by-ssrc: * @session: the object which received the signal * @ssrc: the SSRC of the RTPSource * * Request the #RTPSource object with SSRC @ssrc in @session. */ rtp_session_signals[SIGNAL_GET_SOURCE_BY_SSRC] = g_signal_new ("get-source-by-ssrc", G_TYPE_FROM_CLASS (klass), G_SIGNAL_RUN_LAST | G_SIGNAL_ACTION, G_STRUCT_OFFSET (RTPSessionClass, get_source_by_ssrc), NULL, NULL, gst_rtp_bin_marshal_OBJECT__UINT, RTP_TYPE_SOURCE, 1, G_TYPE_UINT); /** * RTPSession::on-new-ssrc: * @session: the object which received the signal * @src: the new RTPSource * * Notify of a new SSRC that entered @session. */ 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 (RTPSessionClass, on_new_ssrc), NULL, NULL, g_cclosure_marshal_VOID__OBJECT, G_TYPE_NONE, 1, RTP_TYPE_SOURCE); /** * RTPSession::on-ssrc-collision: * @session: the object which received the signal * @src: the #RTPSource that caused a collision * * Notify when we have an SSRC collision */ 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 (RTPSessionClass, on_ssrc_collision), NULL, NULL, g_cclosure_marshal_VOID__OBJECT, G_TYPE_NONE, 1, RTP_TYPE_SOURCE); /** * RTPSession::on-ssrc-validated: * @session: the object which received the signal * @src: the new validated RTPSource * * Notify of a new SSRC that became validated. */ 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 (RTPSessionClass, on_ssrc_validated), NULL, NULL, g_cclosure_marshal_VOID__OBJECT, G_TYPE_NONE, 1, RTP_TYPE_SOURCE); /** * RTPSession::on-ssrc-active: * @session: the object which received the signal * @src: the active RTPSource * * Notify of a SSRC that is active, i.e., sending RTCP. */ 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 (RTPSessionClass, on_ssrc_active), NULL, NULL, g_cclosure_marshal_VOID__OBJECT, G_TYPE_NONE, 1, RTP_TYPE_SOURCE); /** * RTPSession::on-ssrc-sdes: * @session: the object which received the signal * @src: the RTPSource * * Notify that a new SDES was received for SSRC. */ 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 (RTPSessionClass, on_ssrc_sdes), NULL, NULL, g_cclosure_marshal_VOID__OBJECT, G_TYPE_NONE, 1, RTP_TYPE_SOURCE); /** * RTPSession::on-bye-ssrc: * @session: the object which received the signal * @src: the RTPSource that went away * * Notify of an SSRC that became inactive because of a BYE packet. */ 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 (RTPSessionClass, on_bye_ssrc), NULL, NULL, g_cclosure_marshal_VOID__OBJECT, G_TYPE_NONE, 1, RTP_TYPE_SOURCE); /** * RTPSession::on-bye-timeout: * @session: the object which received the signal * @src: the RTPSource that timed out * * Notify of an SSRC that has timed out because of BYE */ 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 (RTPSessionClass, on_bye_timeout), NULL, NULL, g_cclosure_marshal_VOID__OBJECT, G_TYPE_NONE, 1, RTP_TYPE_SOURCE); /** * RTPSession::on-timeout: * @session: the object which received the signal * @src: the RTPSource that timed out * * Notify of an SSRC that has timed out */ rtp_session_signals[SIGNAL_ON_TIMEOUT] = g_signal_new ("on-timeout", G_TYPE_FROM_CLASS (klass), G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (RTPSessionClass, on_timeout), NULL, NULL, g_cclosure_marshal_VOID__OBJECT, G_TYPE_NONE, 1, RTP_TYPE_SOURCE); /** * RTPSession::on-sender-timeout: * @session: the object which received the signal * @src: the RTPSource that timed out * * Notify of an SSRC that was a sender but timed out and became a receiver. */ 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 (RTPSessionClass, on_sender_timeout), NULL, NULL, g_cclosure_marshal_VOID__OBJECT, G_TYPE_NONE, 1, RTP_TYPE_SOURCE); /** * RTPSession::on-sending-rtcp * @session: the object which received the signal * @buffer: the #GstBuffer containing the RTCP packet about to be sent * @early: %TRUE if the packet is early, %FALSE if it is regular * * This signal is emitted before sending an RTCP packet, it can be used * to add extra RTCP Packets. * * Returns: %TRUE if the RTCP buffer should NOT be suppressed, %FALSE * if suppressing it is acceptable */ rtp_session_signals[SIGNAL_ON_SENDING_RTCP] = g_signal_new ("on-sending-rtcp", G_TYPE_FROM_CLASS (klass), G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (RTPSessionClass, on_sending_rtcp), accumulate_trues, NULL, gst_rtp_bin_marshal_BOOLEAN__BOXED_BOOLEAN, G_TYPE_BOOLEAN, 2, GST_TYPE_BUFFER | G_SIGNAL_TYPE_STATIC_SCOPE, G_TYPE_BOOLEAN); /** * RTPSession::on-feedback-rtcp: * @session: the object which received the signal * @type: Type of RTCP packet, will be %GST_RTCP_TYPE_RTPFB or * %GST_RTCP_TYPE_RTPFB * @fbtype: The type of RTCP FB packet, probably part of #GstRTCPFBType * @sender_ssrc: The SSRC of the sender * @media_ssrc: The SSRC of the media this refers to * @fci: a #GstBuffer with the FCI data from the FB packet or %NULL if * there was no FCI * * Notify that a RTCP feedback packet has been received */ rtp_session_signals[SIGNAL_ON_FEEDBACK_RTCP] = g_signal_new ("on-feedback-rtcp", G_TYPE_FROM_CLASS (klass), G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (RTPSessionClass, on_feedback_rtcp), NULL, NULL, gst_rtp_bin_marshal_VOID__UINT_UINT_UINT_UINT_BOXED, G_TYPE_NONE, 5, G_TYPE_UINT, G_TYPE_UINT, G_TYPE_UINT, G_TYPE_UINT, GST_TYPE_BUFFER); /** * RTPSession::send-rtcp: * @session: the object which received the signal * @max_delay: The maximum delay after which the feedback will not be useful * anymore * * Requests that the #RTPSession initiate a new RTCP packet as soon as * possible within the requested delay. */ rtp_session_signals[SIGNAL_SEND_RTCP] = g_signal_new ("send-rtcp", G_TYPE_FROM_CLASS (klass), G_SIGNAL_RUN_LAST | G_SIGNAL_ACTION, G_STRUCT_OFFSET (RTPSessionClass, send_rtcp), NULL, NULL, gst_rtp_bin_marshal_VOID__UINT64, G_TYPE_NONE, 1, G_TYPE_UINT64); g_object_class_install_property (gobject_class, PROP_INTERNAL_SSRC, g_param_spec_uint ("internal-ssrc", "Internal SSRC", "The internal SSRC used for the session", 0, G_MAXUINT, 0, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)); g_object_class_install_property (gobject_class, PROP_INTERNAL_SOURCE, g_param_spec_object ("internal-source", "Internal Source", "The internal source element of the session", RTP_TYPE_SOURCE, G_PARAM_READABLE | G_PARAM_STATIC_STRINGS)); g_object_class_install_property (gobject_class, PROP_BANDWIDTH, g_param_spec_double ("bandwidth", "Bandwidth", "The bandwidth of the session (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 fraction of the bandwidth used for RTCP (or as a real fraction of the RTP bandwidth if < 1)", 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_RTCP_MTU, g_param_spec_uint ("rtcp-mtu", "RTCP MTU", "The maximum size of the RTCP packets", 16, G_MAXINT16, DEFAULT_RTCP_MTU, 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)); /** * RTPSource::sources * * Get a GValue Array of all sources in the session. * * * Getting the #RTPSources of a session * <programlisting> * { * GValueArray *arr; * GValue *val; * guint i; * * g_object_get (sess, "sources", &arr, NULL); * * for (i = 0; i < arr->n_values; i++) { * RTPSource *source; * * val = g_value_array_get_nth (arr, i); * source = g_value_get_object (val); * } * g_value_array_free (arr); * } * </programlisting> * </example> */ g_object_class_install_property (gobject_class, PROP_SOURCES, g_param_spec_boxed ("sources", "Sources", "An array of all known sources in the session", G_TYPE_VALUE_ARRAY, G_PARAM_READABLE | G_PARAM_STATIC_STRINGS)); g_object_class_install_property (gobject_class, PROP_FAVOR_NEW, g_param_spec_boolean ("favor-new", "Favor new sources", "Resolve SSRC conflict in favor of new sources", FALSE, 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)); g_object_class_install_property (gobject_class, PROP_RTCP_FEEDBACK_RETENTION_WINDOW, g_param_spec_uint64 ("rtcp-feedback-retention-window", "RTCP Feedback retention window", "Duration during which RTCP Feedback packets are retained (in ns)", 0, G_MAXUINT64, DEFAULT_RTCP_FEEDBACK_RETENTION_WINDOW, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)); g_object_class_install_property (gobject_class, PROP_RTCP_IMMEDIATE_FEEDBACK_THRESHOLD, g_param_spec_uint ("rtcp-immediate-feedback-threshold", "RTCP Immediate Feedback threshold", "The maximum number of members of a RTP session for which immediate" " feedback is used", 0, G_MAXUINT, DEFAULT_RTCP_IMMEDIATE_FEEDBACK_THRESHOLD, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)); g_object_class_install_property (gobject_class, PROP_PROBATION, g_param_spec_uint ("probation", "Number of probations", "Consecutive packet sequence numbers to accept the source", 0, G_MAXUINT, DEFAULT_PROBATION, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)); klass->get_source_by_ssrc = GST_DEBUG_FUNCPTR (rtp_session_get_source_by_ssrc); klass->on_sending_rtcp = GST_DEBUG_FUNCPTR (rtp_session_on_sending_rtcp); klass->send_rtcp = GST_DEBUG_FUNCPTR (rtp_session_send_rtcp); GST_DEBUG_CATEGORY_INIT (rtp_session_debug, "rtpsession", 0, "RTP Session"); } static void rtp_session_init (RTPSession * sess) { gint i; gchar *str; g_mutex_init (&sess->lock); sess->key = g_random_int (); sess->mask_idx = 0; sess->mask = 0; for (i = 0; i < 32; i++) { sess->ssrcs[i] = g_hash_table_new_full (NULL, NULL, NULL, (GDestroyNotify) g_object_unref); } sess->cnames = g_hash_table_new_full (NULL, NULL, g_free, NULL); rtp_stats_init_defaults (&sess->stats); sess->recalc_bandwidth = TRUE; sess->bandwidth = DEFAULT_BANDWIDTH; sess->rtcp_bandwidth = DEFAULT_RTCP_FRACTION; sess->rtcp_rr_bandwidth = DEFAULT_RTCP_RR_BANDWIDTH; sess->rtcp_rs_bandwidth = DEFAULT_RTCP_RS_BANDWIDTH; /* create an active SSRC for this session manager */ sess->source = rtp_session_create_source (sess); sess->source->validated = TRUE; sess->source->internal = TRUE; sess->stats.active_sources++; INIT_AVG (sess->stats.avg_rtcp_packet_size, 100); sess->source->stats.prev_rtcptime = 0; sess->source->stats.last_rtcptime = 1; rtp_stats_set_min_interval (&sess->stats, (gdouble) DEFAULT_RTCP_MIN_INTERVAL / GST_SECOND); /* default UDP header length */ sess->header_len = 28; sess->mtu = DEFAULT_RTCP_MTU; sess->probation = DEFAULT_PROBATION; /* some default SDES entries */ /* we do not want to leak details like the username or hostname here */ str = g_strdup_printf ("user%u@host-%x", g_random_int (), g_random_int ()); rtp_source_set_sdes_string (sess->source, GST_RTCP_SDES_CNAME, str); g_free (str); #if 0 /* we do not want to leak the user's real name here */ str = g_strdup_printf ("Anon%u", g_random_int ()); rtp_source_set_sdes_string (sess->source, GST_RTCP_SDES_NAME, str); g_free (str); #endif rtp_source_set_sdes_string (sess->source, GST_RTCP_SDES_TOOL, "GStreamer"); sess->first_rtcp = TRUE; sess->allow_early = TRUE; sess->rtcp_feedback_retention_window = DEFAULT_RTCP_FEEDBACK_RETENTION_WINDOW; sess->rtcp_immediate_feedback_threshold = DEFAULT_RTCP_IMMEDIATE_FEEDBACK_THRESHOLD; sess->last_keyframe_request = GST_CLOCK_TIME_NONE; GST_DEBUG ("%p: session using SSRC: %08x", sess, sess->source->ssrc); } static void rtp_session_finalize (GObject * object) { RTPSession *sess; gint i; sess = RTP_SESSION_CAST (object); g_mutex_clear (&sess->lock); for (i = 0; i < 32; i++) g_hash_table_destroy (sess->ssrcs[i]); g_free (sess->bye_reason); g_hash_table_destroy (sess->cnames); g_object_unref (sess->source); G_OBJECT_CLASS (rtp_session_parent_class)->finalize (object); } static void copy_source (gpointer key, RTPSource * source, GValueArray * arr) { GValue value = { 0 }; g_value_init (&value, RTP_TYPE_SOURCE); g_value_take_object (&value, source); /* copies the value */ g_value_array_append (arr, &value); } static GValueArray * rtp_session_create_sources (RTPSession * sess) { GValueArray *res; guint size; RTP_SESSION_LOCK (sess); /* get number of elements in the table */ size = g_hash_table_size (sess->ssrcs[sess->mask_idx]); /* create the result value array */ res = g_value_array_new (size); /* and copy all values into the array */ g_hash_table_foreach (sess->ssrcs[sess->mask_idx], (GHFunc) copy_source, res); RTP_SESSION_UNLOCK (sess); return res; } static void rtp_session_set_property (GObject * object, guint prop_id, const GValue * value, GParamSpec * pspec) { RTPSession *sess; sess = RTP_SESSION (object); switch (prop_id) { case PROP_INTERNAL_SSRC: rtp_session_set_internal_ssrc (sess, g_value_get_uint (value)); break; case PROP_BANDWIDTH: sess->bandwidth = g_value_get_double (value); sess->recalc_bandwidth = TRUE; break; case PROP_RTCP_FRACTION: sess->rtcp_bandwidth = g_value_get_double (value); sess->recalc_bandwidth = TRUE; break; case PROP_RTCP_RR_BANDWIDTH: sess->rtcp_rr_bandwidth = g_value_get_int (value); sess->recalc_bandwidth = TRUE; break; case PROP_RTCP_RS_BANDWIDTH: sess->rtcp_rs_bandwidth = g_value_get_int (value); sess->recalc_bandwidth = TRUE; break; case PROP_RTCP_MTU: sess->mtu = g_value_get_uint (value); break; case PROP_SDES: rtp_session_set_sdes_struct (sess, g_value_get_boxed (value)); break; case PROP_FAVOR_NEW: sess->favor_new = g_value_get_boolean (value); break; case PROP_RTCP_MIN_INTERVAL: rtp_stats_set_min_interval (&sess->stats, (gdouble) g_value_get_uint64 (value) / GST_SECOND); /* trigger reconsideration */ RTP_SESSION_LOCK (sess); sess->next_rtcp_check_time = 0; RTP_SESSION_UNLOCK (sess); if (sess->callbacks.reconsider) sess->callbacks.reconsider (sess, sess->reconsider_user_data); break; case PROP_RTCP_IMMEDIATE_FEEDBACK_THRESHOLD: sess->rtcp_immediate_feedback_threshold = g_value_get_uint (value); break; case PROP_PROBATION: sess->probation = g_value_get_uint (value); g_object_set_property (G_OBJECT (sess->source), "probation", value); break; default: G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec); break; } } static void rtp_session_get_property (GObject * object, guint prop_id, GValue * value, GParamSpec * pspec) { RTPSession *sess; sess = RTP_SESSION (object); switch (prop_id) { case PROP_INTERNAL_SSRC: g_value_set_uint (value, rtp_session_get_internal_ssrc (sess)); break; case PROP_INTERNAL_SOURCE: g_value_take_object (value, rtp_session_get_internal_source (sess)); break; case PROP_BANDWIDTH: g_value_set_double (value, sess->bandwidth); break; case PROP_RTCP_FRACTION: g_value_set_double (value, sess->rtcp_bandwidth); break; case PROP_RTCP_RR_BANDWIDTH: g_value_set_int (value, sess->rtcp_rr_bandwidth); break; case PROP_RTCP_RS_BANDWIDTH: g_value_set_int (value, sess->rtcp_rs_bandwidth); break; case PROP_RTCP_MTU: g_value_set_uint (value, sess->mtu); break; case PROP_SDES: g_value_take_boxed (value, rtp_session_get_sdes_struct (sess)); break; case PROP_NUM_SOURCES: g_value_set_uint (value, rtp_session_get_num_sources (sess)); break; case PROP_NUM_ACTIVE_SOURCES: g_value_set_uint (value, rtp_session_get_num_active_sources (sess)); break; case PROP_SOURCES: g_value_take_boxed (value, rtp_session_create_sources (sess)); break; case PROP_FAVOR_NEW: g_value_set_boolean (value, sess->favor_new); break; case PROP_RTCP_MIN_INTERVAL: g_value_set_uint64 (value, sess->stats.min_interval * GST_SECOND); break; case PROP_RTCP_IMMEDIATE_FEEDBACK_THRESHOLD: g_value_set_uint (value, sess->rtcp_immediate_feedback_threshold); break; case PROP_PROBATION: g_value_set_uint (value, sess->probation); g_object_get_property (G_OBJECT (sess->source), "probation", value); break; default: G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec); break; } } static void on_new_ssrc (RTPSession * sess, RTPSource * source) { g_object_ref (source); RTP_SESSION_UNLOCK (sess); g_signal_emit (sess, rtp_session_signals[SIGNAL_ON_NEW_SSRC], 0, source); RTP_SESSION_LOCK (sess); g_object_unref (source); } static void on_ssrc_collision (RTPSession * sess, RTPSource * source) { g_object_ref (source); RTP_SESSION_UNLOCK (sess); g_signal_emit (sess, rtp_session_signals[SIGNAL_ON_SSRC_COLLISION], 0, source); RTP_SESSION_LOCK (sess); g_object_unref (source); } static void on_ssrc_validated (RTPSession * sess, RTPSource * source) { g_object_ref (source); RTP_SESSION_UNLOCK (sess); g_signal_emit (sess, rtp_session_signals[SIGNAL_ON_SSRC_VALIDATED], 0, source); RTP_SESSION_LOCK (sess); g_object_unref (source); } static void on_ssrc_active (RTPSession * sess, RTPSource * source) { g_object_ref (source); RTP_SESSION_UNLOCK (sess); g_signal_emit (sess, rtp_session_signals[SIGNAL_ON_SSRC_ACTIVE], 0, source); RTP_SESSION_LOCK (sess); g_object_unref (source); } static void on_ssrc_sdes (RTPSession * sess, RTPSource * source) { g_object_ref (source); GST_DEBUG ("SDES changed for SSRC %08x", source->ssrc); RTP_SESSION_UNLOCK (sess); g_signal_emit (sess, rtp_session_signals[SIGNAL_ON_SSRC_SDES], 0, source); RTP_SESSION_LOCK (sess); g_object_unref (source); } static void on_bye_ssrc (RTPSession * sess, RTPSource * source) { g_object_ref (source); RTP_SESSION_UNLOCK (sess); g_signal_emit (sess, rtp_session_signals[SIGNAL_ON_BYE_SSRC], 0, source); RTP_SESSION_LOCK (sess); g_object_unref (source); } static void on_bye_timeout (RTPSession * sess, RTPSource * source) { g_object_ref (source); RTP_SESSION_UNLOCK (sess); g_signal_emit (sess, rtp_session_signals[SIGNAL_ON_BYE_TIMEOUT], 0, source); RTP_SESSION_LOCK (sess); g_object_unref (source); } static void on_timeout (RTPSession * sess, RTPSource * source) { g_object_ref (source); RTP_SESSION_UNLOCK (sess); g_signal_emit (sess, rtp_session_signals[SIGNAL_ON_TIMEOUT], 0, source); RTP_SESSION_LOCK (sess); g_object_unref (source); } static void on_sender_timeout (RTPSession * sess, RTPSource * source) { g_object_ref (source); RTP_SESSION_UNLOCK (sess); g_signal_emit (sess, rtp_session_signals[SIGNAL_ON_SENDER_TIMEOUT], 0, source); RTP_SESSION_LOCK (sess); g_object_unref (source); } /** * rtp_session_new: * * Create a new session object. * * Returns: a new #RTPSession. g_object_unref() after usage. */ RTPSession * rtp_session_new (void) { RTPSession *sess; sess = g_object_new (RTP_TYPE_SESSION, NULL); return sess; } /** * rtp_session_set_callbacks: * @sess: an #RTPSession * @callbacks: callbacks to configure * @user_data: user data passed in the callbacks * * Configure a set of callbacks to be notified of actions. */ void rtp_session_set_callbacks (RTPSession * sess, RTPSessionCallbacks * callbacks, gpointer user_data) { g_return_if_fail (RTP_IS_SESSION (sess)); if (callbacks->process_rtp) { sess->callbacks.process_rtp = callbacks->process_rtp; sess->process_rtp_user_data = user_data; } if (callbacks->send_rtp) { sess->callbacks.send_rtp = callbacks->send_rtp; sess->send_rtp_user_data = user_data; } if (callbacks->send_rtcp) { sess->callbacks.send_rtcp = callbacks->send_rtcp; sess->send_rtcp_user_data = user_data; } if (callbacks->sync_rtcp) { sess->callbacks.sync_rtcp = callbacks->sync_rtcp; sess->sync_rtcp_user_data = user_data; } if (callbacks->clock_rate) { sess->callbacks.clock_rate = callbacks->clock_rate; sess->clock_rate_user_data = user_data; } if (callbacks->reconsider) { sess->callbacks.reconsider = callbacks->reconsider; sess->reconsider_user_data = user_data; } if (callbacks->request_key_unit) { sess->callbacks.request_key_unit = callbacks->request_key_unit; sess->request_key_unit_user_data = user_data; } if (callbacks->request_time) { sess->callbacks.request_time = callbacks->request_time; sess->request_time_user_data = user_data; } } /** * rtp_session_set_process_rtp_callback: * @sess: an #RTPSession * @callback: callback to set * @user_data: user data passed in the callback * * Configure only the process_rtp callback to be notified of the process_rtp action. */ void rtp_session_set_process_rtp_callback (RTPSession * sess, RTPSessionProcessRTP callback, gpointer user_data) { g_return_if_fail (RTP_IS_SESSION (sess)); sess->callbacks.process_rtp = callback; sess->process_rtp_user_data = user_data; } /** * rtp_session_set_send_rtp_callback: * @sess: an #RTPSession * @callback: callback to set * @user_data: user data passed in the callback * * Configure only the send_rtp callback to be notified of the send_rtp action. */ void rtp_session_set_send_rtp_callback (RTPSession * sess, RTPSessionSendRTP callback, gpointer user_data) { g_return_if_fail (RTP_IS_SESSION (sess)); sess->callbacks.send_rtp = callback; sess->send_rtp_user_data = user_data; } /** * rtp_session_set_send_rtcp_callback: * @sess: an #RTPSession * @callback: callback to set * @user_data: user data passed in the callback * * Configure only the send_rtcp callback to be notified of the send_rtcp action. */ void rtp_session_set_send_rtcp_callback (RTPSession * sess, RTPSessionSendRTCP callback, gpointer user_data) { g_return_if_fail (RTP_IS_SESSION (sess)); sess->callbacks.send_rtcp = callback; sess->send_rtcp_user_data = user_data; } /** * rtp_session_set_sync_rtcp_callback: * @sess: an #RTPSession * @callback: callback to set * @user_data: user data passed in the callback * * Configure only the sync_rtcp callback to be notified of the sync_rtcp action. */ void rtp_session_set_sync_rtcp_callback (RTPSession * sess, RTPSessionSyncRTCP callback, gpointer user_data) { g_return_if_fail (RTP_IS_SESSION (sess)); sess->callbacks.sync_rtcp = callback; sess->sync_rtcp_user_data = user_data; } /** * rtp_session_set_clock_rate_callback: * @sess: an #RTPSession * @callback: callback to set * @user_data: user data passed in the callback * * Configure only the clock_rate callback to be notified of the clock_rate action. */ void rtp_session_set_clock_rate_callback (RTPSession * sess, RTPSessionClockRate callback, gpointer user_data) { g_return_if_fail (RTP_IS_SESSION (sess)); sess->callbacks.clock_rate = callback; sess->clock_rate_user_data = user_data; } /** * rtp_session_set_reconsider_callback: * @sess: an #RTPSession * @callback: callback to set * @user_data: user data passed in the callback * * Configure only the reconsider callback to be notified of the reconsider action. */ void rtp_session_set_reconsider_callback (RTPSession * sess, RTPSessionReconsider callback, gpointer user_data) { g_return_if_fail (RTP_IS_SESSION (sess)); sess->callbacks.reconsider = callback; sess->reconsider_user_data = user_data; } /** * rtp_session_set_request_time_callback: * @sess: an #RTPSession * @callback: callback to set * @user_data: user data passed in the callback * * Configure only the request_time callback */ void rtp_session_set_request_time_callback (RTPSession * sess, RTPSessionRequestTime callback, gpointer user_data) { g_return_if_fail (RTP_IS_SESSION (sess)); sess->callbacks.request_time = callback; sess->request_time_user_data = user_data; } /** * rtp_session_set_bandwidth: * @sess: an #RTPSession * @bandwidth: the bandwidth allocated * * Set the session bandwidth in bytes per second. */ void rtp_session_set_bandwidth (RTPSession * sess, gdouble bandwidth) { g_return_if_fail (RTP_IS_SESSION (sess)); RTP_SESSION_LOCK (sess); sess->stats.bandwidth = bandwidth; RTP_SESSION_UNLOCK (sess); } /** * rtp_session_get_bandwidth: * @sess: an #RTPSession * * Get the session bandwidth. * * Returns: the session bandwidth. */ gdouble rtp_session_get_bandwidth (RTPSession * sess) { gdouble result; g_return_val_if_fail (RTP_IS_SESSION (sess), 0); RTP_SESSION_LOCK (sess); result = sess->stats.bandwidth; RTP_SESSION_UNLOCK (sess); return result; } /** * rtp_session_set_rtcp_fraction: * @sess: an #RTPSession * @bandwidth: the RTCP bandwidth * * Set the bandwidth in bytes per second that should be used for RTCP * messages. */ void rtp_session_set_rtcp_fraction (RTPSession * sess, gdouble bandwidth) { g_return_if_fail (RTP_IS_SESSION (sess)); RTP_SESSION_LOCK (sess); sess->stats.rtcp_bandwidth = bandwidth; RTP_SESSION_UNLOCK (sess); } /** * rtp_session_get_rtcp_fraction: * @sess: an #RTPSession * * Get the session bandwidth used for RTCP. * * Returns: The bandwidth used for RTCP messages. */ gdouble rtp_session_get_rtcp_fraction (RTPSession * sess) { gdouble result; g_return_val_if_fail (RTP_IS_SESSION (sess), 0.0); RTP_SESSION_LOCK (sess); result = sess->stats.rtcp_bandwidth; RTP_SESSION_UNLOCK (sess); return result; } /** * rtp_session_set_sdes_string: * @sess: an #RTPSession * @type: the type of the SDES item * @item: a null-terminated string to set. * * Store an SDES item of @type in @sess. * * Returns: %FALSE if the data was unchanged @type is invalid. */ gboolean rtp_session_set_sdes_string (RTPSession * sess, GstRTCPSDESType type, const gchar * item) { gboolean result; g_return_val_if_fail (RTP_IS_SESSION (sess), FALSE); RTP_SESSION_LOCK (sess); result = rtp_source_set_sdes_string (sess->source, type, item); RTP_SESSION_UNLOCK (sess); return result; } /** * rtp_session_get_sdes_string: * @sess: an #RTPSession * @type: the type of the SDES item * * Get the SDES item of @type from @sess. * * Returns: a null-terminated copy of the SDES item or NULL when @type was not * valid. g_free() after usage. */ gchar * rtp_session_get_sdes_string (RTPSession * sess, GstRTCPSDESType type) { gchar *result; g_return_val_if_fail (RTP_IS_SESSION (sess), NULL); RTP_SESSION_LOCK (sess); result = rtp_source_get_sdes_string (sess->source, type); RTP_SESSION_UNLOCK (sess); return result; } /** * rtp_session_get_sdes_struct: * @sess: an #RTSPSession * * Get the SDES data as a #GstStructure * * Returns: a GstStructure with SDES items for @sess. This function returns a * copy of the SDES structure, use gst_structure_free() after usage. */ GstStructure * rtp_session_get_sdes_struct (RTPSession * sess) { const GstStructure *sdes; GstStructure *result = NULL; g_return_val_if_fail (RTP_IS_SESSION (sess), NULL); RTP_SESSION_LOCK (sess); sdes = rtp_source_get_sdes_struct (sess->source); if (sdes) result = gst_structure_copy (sdes); RTP_SESSION_UNLOCK (sess); return result; } /** * rtp_session_set_sdes_struct: * @sess: an #RTSPSession * @sdes: a #GstStructure * * Set the SDES data as a #GstStructure. This function makes a copy of @sdes. */ void rtp_session_set_sdes_struct (RTPSession * sess, const GstStructure * sdes) { g_return_if_fail (sdes); g_return_if_fail (RTP_IS_SESSION (sess)); RTP_SESSION_LOCK (sess); rtp_source_set_sdes_struct (sess->source, gst_structure_copy (sdes)); RTP_SESSION_UNLOCK (sess); } static GstFlowReturn source_push_rtp (RTPSource * source, gpointer data, RTPSession * session) { GstFlowReturn result = GST_FLOW_OK; if (source == session->source) { GST_LOG ("source %08x pushed sender RTP packet", source->ssrc); RTP_SESSION_UNLOCK (session); if (session->callbacks.send_rtp) result = session->callbacks.send_rtp (session, source, data, session->send_rtp_user_data); else { gst_mini_object_unref (GST_MINI_OBJECT_CAST (data)); } } else { GST_LOG ("source %08x pushed receiver RTP packet", source->ssrc); RTP_SESSION_UNLOCK (session); if (session->callbacks.process_rtp) result = session->callbacks.process_rtp (session, source, GST_BUFFER_CAST (data), session->process_rtp_user_data); else gst_buffer_unref (GST_BUFFER_CAST (data)); } RTP_SESSION_LOCK (session); return result; } static gint source_clock_rate (RTPSource * source, guint8 pt, RTPSession * session) { gint result; RTP_SESSION_UNLOCK (session); if (session->callbacks.clock_rate) result = session->callbacks.clock_rate (session, pt, session->clock_rate_user_data); else result = -1; RTP_SESSION_LOCK (session); GST_DEBUG ("got clock-rate %d for pt %d", result, pt); return result; } static RTPSourceCallbacks callbacks = { (RTPSourcePushRTP) source_push_rtp, (RTPSourceClockRate) source_clock_rate, }; static gboolean check_collision (RTPSession * sess, RTPSource * source, RTPArrivalStats * arrival, gboolean rtp) { /* If we have no arrival address, we can't do collision checking */ if (!arrival->address) return FALSE; if (sess->source != source) { GSocketAddress *from; /* This is not our local source, but lets check if two remote * source collide */ if (rtp) { from = source->rtp_from; } else { from = source->rtcp_from; } if (from) { if (__g_socket_address_equal (from, arrival->address)) { /* Address is the same */ return FALSE; } else { GST_LOG ("we have a third-party collision or loop ssrc:%x", rtp_source_get_ssrc (source)); if (sess->favor_new) { if (rtp_source_find_conflicting_address (source, arrival->address, arrival->current_time)) { gchar *buf1; buf1 = __g_socket_address_to_string (arrival->address); GST_LOG ("Known conflict on %x for %s, dropping packet", rtp_source_get_ssrc (source), buf1); g_free (buf1); return TRUE; } else { gchar *buf1, *buf2; /* Current address is not a known conflict, lets assume this is * a new source. Save old address in possible conflict list */ rtp_source_add_conflicting_address (source, from, arrival->current_time); buf1 = __g_socket_address_to_string (from); buf2 = __g_socket_address_to_string (arrival->address); GST_DEBUG ("New conflict for ssrc %x, replacing %s with %s," " saving old as known conflict", rtp_source_get_ssrc (source), buf1, buf2); if (rtp) rtp_source_set_rtp_from (source, arrival->address); else rtp_source_set_rtcp_from (source, arrival->address); g_free (buf1); g_free (buf2); return FALSE; } } else { /* Don't need to save old addresses, we ignore new sources */ return TRUE; } } } else { /* We don't already have a from address for RTP, just set it */ if (rtp) rtp_source_set_rtp_from (source, arrival->address); else rtp_source_set_rtcp_from (source, arrival->address); return FALSE; } /* FIXME: Log 3rd party collision somehow * Maybe should be done in upper layer, only the SDES can tell us * if its a collision or a loop */ /* If the source has been inactive for some time, we assume that it has * simply changed its transport source address. Hence, there is no true * third-party collision - only a simulated one. */ if (arrival->current_time > source->last_activity) { GstClockTime inactivity_period = arrival->current_time - source->last_activity; if (inactivity_period > 1 * GST_SECOND) { /* Use new network address */ if (rtp) { g_assert (source->rtp_from); rtp_source_set_rtp_from (source, arrival->address); } else { g_assert (source->rtcp_from); rtp_source_set_rtcp_from (source, arrival->address); } return FALSE; } } } else { /* This is sending with our ssrc, is it an address we already know */ if (rtp_source_find_conflicting_address (source, arrival->address, arrival->current_time)) { /* Its a known conflict, its probably a loop, not a collision * lets just drop the incoming packet */ GST_DEBUG ("Our packets are being looped back to us, dropping"); } else { /* Its a new collision, lets change our SSRC */ rtp_source_add_conflicting_address (source, arrival->address, arrival->current_time); GST_DEBUG ("Collision for SSRC %x", rtp_source_get_ssrc (source)); on_ssrc_collision (sess, source); rtp_session_schedule_bye_locked (sess, "SSRC Collision", arrival->current_time); sess->change_ssrc = TRUE; } } return TRUE; } /* must be called with the session lock, the returned source needs to be * unreffed after usage. */ static RTPSource * obtain_source (RTPSession * sess, guint32 ssrc, gboolean * created, RTPArrivalStats * arrival, gboolean rtp) { RTPSource *source; source = g_hash_table_lookup (sess->ssrcs[sess->mask_idx], GINT_TO_POINTER (ssrc)); if (source == NULL) { /* make new Source in probation and insert */ source = rtp_source_new (ssrc); /* for RTP packets we need to set the source in probation. Receiving RTCP * packets of an SSRC, on the other hand, is a strong indication that we * are dealing with a valid source. */ if (rtp) g_object_set (source, "probation", sess->probation, NULL); else g_object_set (source, "probation", 0, NULL); /* store from address, if any */ if (arrival->address) { if (rtp) rtp_source_set_rtp_from (source, arrival->address); else rtp_source_set_rtcp_from (source, arrival->address); } /* configure a callback on the source */ rtp_source_set_callbacks (source, &callbacks, sess); g_hash_table_insert (sess->ssrcs[sess->mask_idx], GINT_TO_POINTER (ssrc), source); /* we have one more source now */ sess->total_sources++; *created = TRUE; } else { *created = FALSE; /* check for collision, this updates the address when not previously set */ if (check_collision (sess, source, arrival, rtp)) { return NULL; } /* Receiving RTCP packets of an SSRC is a strong indication that we * are dealing with a valid source. */ if (!rtp) g_object_set (source, "probation", 0, NULL); } /* update last activity */ source->last_activity = arrival->current_time; if (rtp) source->last_rtp_activity = arrival->current_time; g_object_ref (source); return source; } /** * rtp_session_get_internal_source: * @sess: a #RTPSession * * Get the internal #RTPSource of @sess. * * Returns: The internal #RTPSource. g_object_unref() after usage. */ RTPSource * rtp_session_get_internal_source (RTPSession * sess) { RTPSource *result; g_return_val_if_fail (RTP_IS_SESSION (sess), NULL); result = g_object_ref (sess->source); return result; } /** * rtp_session_set_internal_ssrc: * @sess: a #RTPSession * @ssrc: an SSRC * * Set the SSRC of @sess to @ssrc. */ void rtp_session_set_internal_ssrc (RTPSession * sess, guint32 ssrc) { RTP_SESSION_LOCK (sess); if (ssrc != sess->source->ssrc) { g_hash_table_steal (sess->ssrcs[sess->mask_idx], GINT_TO_POINTER (sess->source->ssrc)); GST_DEBUG ("setting internal SSRC to %08x", ssrc); /* After this call, any receiver of the old SSRC either in RTP or RTCP * packets will timeout on the old SSRC, we could potentially schedule a * BYE RTCP for the old SSRC... */ sess->source->ssrc = ssrc; rtp_source_reset (sess->source); /* rehash with the new SSRC */ g_hash_table_insert (sess->ssrcs[sess->mask_idx], GINT_TO_POINTER (sess->source->ssrc), sess->source); } RTP_SESSION_UNLOCK (sess); g_object_notify (G_OBJECT (sess), "internal-ssrc"); } /** * rtp_session_get_internal_ssrc: * @sess: a #RTPSession * * Get the internal SSRC of @sess. * * Returns: The SSRC of the session. */ guint32 rtp_session_get_internal_ssrc (RTPSession * sess) { guint32 ssrc; RTP_SESSION_LOCK (sess); ssrc = sess->source->ssrc; RTP_SESSION_UNLOCK (sess); return ssrc; } /** * rtp_session_add_source: * @sess: a #RTPSession * @src: #RTPSource to add * * Add @src to @session. * * Returns: %TRUE on success, %FALSE if a source with the same SSRC already * existed in the session. */ gboolean rtp_session_add_source (RTPSession * sess, RTPSource * src) { gboolean result = FALSE; RTPSource *find; g_return_val_if_fail (RTP_IS_SESSION (sess), FALSE); g_return_val_if_fail (src != NULL, FALSE); RTP_SESSION_LOCK (sess); find = g_hash_table_lookup (sess->ssrcs[sess->mask_idx], GINT_TO_POINTER (src->ssrc)); if (find == NULL) { g_hash_table_insert (sess->ssrcs[sess->mask_idx], GINT_TO_POINTER (src->ssrc), src); /* we have one more source now */ sess->total_sources++; result = TRUE; } RTP_SESSION_UNLOCK (sess); return result; } /** * rtp_session_get_num_sources: * @sess: an #RTPSession * * Get the number of sources in @sess. * * Returns: The number of sources in @sess. */ guint rtp_session_get_num_sources (RTPSession * sess) { guint result; g_return_val_if_fail (RTP_IS_SESSION (sess), FALSE); RTP_SESSION_LOCK (sess); result = sess->total_sources; RTP_SESSION_UNLOCK (sess); return result; } /** * rtp_session_get_num_active_sources: * @sess: an #RTPSession * * Get the number of active sources in @sess. A source is considered active when * it has been validated and has not yet received a BYE RTCP message. * * Returns: The number of active sources in @sess. */ guint rtp_session_get_num_active_sources (RTPSession * sess) { guint result; g_return_val_if_fail (RTP_IS_SESSION (sess), 0); RTP_SESSION_LOCK (sess); result = sess->stats.active_sources; RTP_SESSION_UNLOCK (sess); return result; } /** * rtp_session_get_source_by_ssrc: * @sess: an #RTPSession * @ssrc: an SSRC * * Find the source with @ssrc in @sess. * * Returns: a #RTPSource with SSRC @ssrc or NULL if the source was not found. * g_object_unref() after usage. */ RTPSource * rtp_session_get_source_by_ssrc (RTPSession * sess, guint32 ssrc) { RTPSource *result; g_return_val_if_fail (RTP_IS_SESSION (sess), NULL); RTP_SESSION_LOCK (sess); result = g_hash_table_lookup (sess->ssrcs[sess->mask_idx], GINT_TO_POINTER (ssrc)); if (result) g_object_ref (result); RTP_SESSION_UNLOCK (sess); return result; } /** * rtp_session_get_source_by_cname: * @sess: a #RTPSession * @cname: an CNAME * * Find the source with @cname in @sess. * * Returns: a #RTPSource with CNAME @cname or NULL if the source was not found. * g_object_unref() after usage. */ RTPSource * rtp_session_get_source_by_cname (RTPSession * sess, const gchar * cname) { RTPSource *result; g_return_val_if_fail (RTP_IS_SESSION (sess), NULL); g_return_val_if_fail (cname != NULL, NULL); RTP_SESSION_LOCK (sess); result = g_hash_table_lookup (sess->cnames, cname); if (result) g_object_ref (result); RTP_SESSION_UNLOCK (sess); return result; } /* should be called with the SESSION lock */ static guint32 rtp_session_create_new_ssrc (RTPSession * sess) { guint32 ssrc; while (TRUE) { ssrc = g_random_int (); /* see if it exists in the session, we're done if it doesn't */ if (g_hash_table_lookup (sess->ssrcs[sess->mask_idx], GINT_TO_POINTER (ssrc)) == NULL) break; } return ssrc; } /** * rtp_session_create_source: * @sess: an #RTPSession * * Create an #RTPSource for use in @sess. This function will create a source * with an ssrc that is currently not used by any participants in the session. * * Returns: an #RTPSource. */ RTPSource * rtp_session_create_source (RTPSession * sess) { guint32 ssrc; RTPSource *source; RTP_SESSION_LOCK (sess); ssrc = rtp_session_create_new_ssrc (sess); source = rtp_source_new (ssrc); rtp_source_set_callbacks (source, &callbacks, sess); /* we need an additional ref for the source in the hashtable */ g_object_ref (source); g_hash_table_insert (sess->ssrcs[sess->mask_idx], GINT_TO_POINTER (ssrc), source); /* we have one more source now */ sess->total_sources++; RTP_SESSION_UNLOCK (sess); return source; } /* update the RTPArrivalStats structure with the current time and other bits * about the current buffer we are handling. * This function is typically called when a validated packet is received. * This function should be called with the SESSION_LOCK */ static void update_arrival_stats (RTPSession * sess, RTPArrivalStats * arrival, gboolean rtp, GstBuffer * buffer, GstClockTime current_time, GstClockTime running_time, guint64 ntpnstime) { GstNetAddressMeta *meta; GstRTPBuffer rtpb = { NULL }; /* get time of arrival */ arrival->current_time = current_time; arrival->running_time = running_time; arrival->ntpnstime = ntpnstime; /* get packet size including header overhead */ arrival->bytes = gst_buffer_get_size (buffer) + sess->header_len; if (rtp) { gst_rtp_buffer_map (buffer, GST_MAP_READ, &rtpb); arrival->payload_len = gst_rtp_buffer_get_payload_len (&rtpb); gst_rtp_buffer_unmap (&rtpb); } else { arrival->payload_len = 0; } /* for netbuffer we can store the IP address to check for collisions */ meta = gst_buffer_get_net_address_meta (buffer); if (arrival->address) g_object_unref (arrival->address); if (meta) { arrival->address = G_SOCKET_ADDRESS (g_object_ref (meta->addr)); } else { arrival->address = NULL; } } static void clean_arrival_stats (RTPArrivalStats * arrival) { if (arrival->address) g_object_unref (arrival->address); } /** * rtp_session_process_rtp: * @sess: and #RTPSession * @buffer: an RTP buffer * @current_time: the current system time * @running_time: the running_time of @buffer * * Process an RTP buffer in the session manager. This function takes ownership * of @buffer. * * Returns: a #GstFlowReturn. */ GstFlowReturn rtp_session_process_rtp (RTPSession * sess, GstBuffer * buffer, GstClockTime current_time, GstClockTime running_time) { GstFlowReturn result; guint32 ssrc; RTPSource *source; gboolean created; gboolean prevsender, prevactive; RTPArrivalStats arrival = { NULL, }; guint32 csrcs[16]; guint8 i, count; guint64 oldrate; GstRTPBuffer rtp = { NULL }; g_return_val_if_fail (RTP_IS_SESSION (sess), GST_FLOW_ERROR); g_return_val_if_fail (GST_IS_BUFFER (buffer), GST_FLOW_ERROR); if (!gst_rtp_buffer_map (buffer, GST_MAP_READ, &rtp)) goto invalid_packet; RTP_SESSION_LOCK (sess); /* ignore more RTP packets when we left the session */ if (sess->source->received_bye) goto ignore; /* update arrival stats */ update_arrival_stats (sess, &arrival, TRUE, buffer, current_time, running_time, -1); /* get SSRC and look up in session database */ ssrc = gst_rtp_buffer_get_ssrc (&rtp); source = obtain_source (sess, ssrc, &created, &arrival, TRUE); if (!source) goto collision; /* copy available csrc for later */ count = gst_rtp_buffer_get_csrc_count (&rtp); /* make sure to not overflow our array. An RTP buffer can maximally contain * 16 CSRCs */ count = MIN (count, 16); for (i = 0; i < count; i++) csrcs[i] = gst_rtp_buffer_get_csrc (&rtp, i); gst_rtp_buffer_unmap (&rtp); prevsender = RTP_SOURCE_IS_SENDER (source); prevactive = RTP_SOURCE_IS_ACTIVE (source); oldrate = source->bitrate; /* let source process the packet */ result = rtp_source_process_rtp (source, buffer, &arrival); /* source became active */ if (prevactive != RTP_SOURCE_IS_ACTIVE (source)) { sess->stats.active_sources++; GST_DEBUG ("source: %08x became active, %d active sources", ssrc, sess->stats.active_sources); on_ssrc_validated (sess, source); } if (prevsender != RTP_SOURCE_IS_SENDER (source)) { sess->stats.sender_sources++; GST_DEBUG ("source: %08x became sender, %d sender sources", ssrc, sess->stats.sender_sources); } if (oldrate != source->bitrate) sess->recalc_bandwidth = TRUE; if (created) on_new_ssrc (sess, source); if (source->validated) { gboolean created; /* for validated sources, we add the CSRCs as well */ for (i = 0; i < count; i++) { guint32 csrc; RTPSource *csrc_src; csrc = csrcs[i]; /* get source */ csrc_src = obtain_source (sess, csrc, &created, &arrival, TRUE); if (!csrc_src) continue; if (created) { GST_DEBUG ("created new CSRC: %08x", csrc); rtp_source_set_as_csrc (csrc_src); if (RTP_SOURCE_IS_ACTIVE (csrc_src)) sess->stats.active_sources++; on_new_ssrc (sess, csrc_src); } g_object_unref (csrc_src); } } g_object_unref (source); RTP_SESSION_UNLOCK (sess); clean_arrival_stats (&arrival); return result; /* ERRORS */ invalid_packet: { gst_buffer_unref (buffer); GST_DEBUG ("invalid RTP packet received"); return GST_FLOW_OK; } ignore: { RTP_SESSION_UNLOCK (sess); gst_rtp_buffer_unmap (&rtp); gst_buffer_unref (buffer); GST_DEBUG ("ignoring RTP packet because we are leaving"); return GST_FLOW_OK; } collision: { RTP_SESSION_UNLOCK (sess); gst_rtp_buffer_unmap (&rtp); gst_buffer_unref (buffer); clean_arrival_stats (&arrival); GST_DEBUG ("ignoring packet because its collisioning"); return GST_FLOW_OK; } } static void rtp_session_process_rb (RTPSession * sess, RTPSource * source, GstRTCPPacket * packet, RTPArrivalStats * arrival) { guint count, i; count = gst_rtcp_packet_get_rb_count (packet); for (i = 0; i < count; i++) { guint32 ssrc, exthighestseq, jitter, lsr, dlsr; guint8 fractionlost; gint32 packetslost; gst_rtcp_packet_get_rb (packet, i, &ssrc, &fractionlost, &packetslost, &exthighestseq, &jitter, &lsr, &dlsr); GST_DEBUG ("RB %d: SSRC %08x, jitter %" G_GUINT32_FORMAT, i, ssrc, jitter); if (ssrc == sess->source->ssrc) { /* only deal with report blocks for our session, we update the stats of * the sender of the RTCP message. We could also compare our stats against * the other sender to see if we are better or worse. */ rtp_source_process_rb (source, arrival->ntpnstime, fractionlost, packetslost, exthighestseq, jitter, lsr, dlsr); } } on_ssrc_active (sess, source); } /* A Sender report contains statistics about how the sender is doing. This * includes timing informataion such as the relation between RTP and NTP * timestamps and the number of packets/bytes it sent to us. * * In this report is also included a set of report blocks related to how this * sender is receiving data (in case we (or somebody else) is also sending stuff * to it). This info includes the packet loss, jitter and seqnum. It also * contains information to calculate the round trip time (LSR/DLSR). */ static void rtp_session_process_sr (RTPSession * sess, GstRTCPPacket * packet, RTPArrivalStats * arrival, gboolean * do_sync) { guint32 senderssrc, rtptime, packet_count, octet_count; guint64 ntptime; RTPSource *source; gboolean created, prevsender; gst_rtcp_packet_sr_get_sender_info (packet, &senderssrc, &ntptime, &rtptime, &packet_count, &octet_count); GST_DEBUG ("got SR packet: SSRC %08x, time %" GST_TIME_FORMAT, senderssrc, GST_TIME_ARGS (arrival->current_time)); source = obtain_source (sess, senderssrc, &created, arrival, FALSE); if (!source) return; /* don't try to do lip-sync for sources that sent a BYE */ if (rtp_source_received_bye (source)) *do_sync = FALSE; else *do_sync = TRUE; prevsender = RTP_SOURCE_IS_SENDER (source); /* first update the source */ rtp_source_process_sr (source, arrival->current_time, ntptime, rtptime, packet_count, octet_count); if (prevsender != RTP_SOURCE_IS_SENDER (source)) { sess->stats.sender_sources++; GST_DEBUG ("source: %08x became sender, %d sender sources", senderssrc, sess->stats.sender_sources); } if (created) on_new_ssrc (sess, source); rtp_session_process_rb (sess, source, packet, arrival); g_object_unref (source); } /* A receiver report contains statistics about how a receiver is doing. It * includes stuff like packet loss, jitter and the seqnum it received last. It * also contains info to calculate the round trip time. * * We are only interested in how the sender of this report is doing wrt to us. */ static void rtp_session_process_rr (RTPSession * sess, GstRTCPPacket * packet, RTPArrivalStats * arrival) { guint32 senderssrc; RTPSource *source; gboolean created; senderssrc = gst_rtcp_packet_rr_get_ssrc (packet); GST_DEBUG ("got RR packet: SSRC %08x", senderssrc); source = obtain_source (sess, senderssrc, &created, arrival, FALSE); if (!source) return; if (created) on_new_ssrc (sess, source); rtp_session_process_rb (sess, source, packet, arrival); g_object_unref (source); } /* Get SDES items and store them in the SSRC */ static void rtp_session_process_sdes (RTPSession * sess, GstRTCPPacket * packet, RTPArrivalStats * arrival) { guint items, i, j; gboolean more_items, more_entries; items = gst_rtcp_packet_sdes_get_item_count (packet); GST_DEBUG ("got SDES packet with %d items", items); more_items = gst_rtcp_packet_sdes_first_item (packet); i = 0; while (more_items) { guint32 ssrc; gboolean changed, created, validated; RTPSource *source; GstStructure *sdes; ssrc = gst_rtcp_packet_sdes_get_ssrc (packet); GST_DEBUG ("item %d, SSRC %08x", i, ssrc); changed = FALSE; /* find src, no probation when dealing with RTCP */ source = obtain_source (sess, ssrc, &created, arrival, FALSE); if (!source) return; sdes = gst_structure_new_empty ("application/x-rtp-source-sdes"); more_entries = gst_rtcp_packet_sdes_first_entry (packet); j = 0; while (more_entries) { GstRTCPSDESType type; guint8 len; guint8 *data; gchar *name; gchar *value; gst_rtcp_packet_sdes_get_entry (packet, &type, &len, &data); GST_DEBUG ("entry %d, type %d, len %d, data %.*s", j, type, len, len, data); if (type == GST_RTCP_SDES_PRIV) { name = g_strndup ((const gchar *) &data[1], data[0]); len -= data[0] + 1; data += data[0] + 1; } else { name = g_strdup (gst_rtcp_sdes_type_to_name (type)); } value = g_strndup ((const gchar *) data, len); gst_structure_set (sdes, name, G_TYPE_STRING, value, NULL); g_free (name); g_free (value); more_entries = gst_rtcp_packet_sdes_next_entry (packet); j++; } /* takes ownership of sdes */ changed = rtp_source_set_sdes_struct (source, sdes); validated = !RTP_SOURCE_IS_ACTIVE (source); source->validated = TRUE; if (created) on_new_ssrc (sess, source); /* source became active */ if (validated) { sess->stats.active_sources++; GST_DEBUG ("source: %08x became active, %d active sources", ssrc, sess->stats.active_sources); on_ssrc_validated (sess, source); } if (changed) on_ssrc_sdes (sess, source); g_object_unref (source); more_items = gst_rtcp_packet_sdes_next_item (packet); i++; } } /* BYE is sent when a client leaves the session */ static void rtp_session_process_bye (RTPSession * sess, GstRTCPPacket * packet, RTPArrivalStats * arrival) { guint count, i; gchar *reason; gboolean reconsider = FALSE; reason = gst_rtcp_packet_bye_get_reason (packet); GST_DEBUG ("got BYE packet (reason: %s)", GST_STR_NULL (reason)); count = gst_rtcp_packet_bye_get_ssrc_count (packet); for (i = 0; i < count; i++) { guint32 ssrc; RTPSource *source; gboolean created, prevactive, prevsender; guint pmembers, members; ssrc = gst_rtcp_packet_bye_get_nth_ssrc (packet, i); GST_DEBUG ("SSRC: %08x", ssrc); if (ssrc == sess->source->ssrc) return; /* find src and mark bye, no probation when dealing with RTCP */ source = obtain_source (sess, ssrc, &created, arrival, FALSE); if (!source) return; /* store time for when we need to time out this source */ source->bye_time = arrival->current_time; prevactive = RTP_SOURCE_IS_ACTIVE (source); prevsender = RTP_SOURCE_IS_SENDER (source); /* let the source handle the rest */ rtp_source_process_bye (source, reason); pmembers = sess->stats.active_sources; if (prevactive && !RTP_SOURCE_IS_ACTIVE (source)) { sess->stats.active_sources--; GST_DEBUG ("source: %08x became inactive, %d active sources", ssrc, sess->stats.active_sources); } if (prevsender && !RTP_SOURCE_IS_SENDER (source)) { sess->stats.sender_sources--; GST_DEBUG ("source: %08x became non sender, %d sender sources", ssrc, sess->stats.sender_sources); } members = sess->stats.active_sources; if (!sess->source->received_bye && members < pmembers) { /* some members went away since the previous timeout estimate. * Perform reverse reconsideration but only when we are not scheduling a * BYE ourselves. */ if (arrival->current_time < sess->next_rtcp_check_time) { GstClockTime time_remaining; time_remaining = sess->next_rtcp_check_time - arrival->current_time; sess->next_rtcp_check_time = gst_util_uint64_scale (time_remaining, members, pmembers); GST_DEBUG ("reverse reconsideration %" GST_TIME_FORMAT, GST_TIME_ARGS (sess->next_rtcp_check_time)); sess->next_rtcp_check_time += arrival->current_time; /* mark pending reconsider. We only want to signal the reconsideration * once after we handled all the source in the bye packet */ reconsider = TRUE; } } if (created) on_new_ssrc (sess, source); on_bye_ssrc (sess, source); g_object_unref (source); } if (reconsider) { RTP_SESSION_UNLOCK (sess); /* notify app of reconsideration */ if (sess->callbacks.reconsider) sess->callbacks.reconsider (sess, sess->reconsider_user_data); RTP_SESSION_LOCK (sess); } g_free (reason); } static void rtp_session_process_app (RTPSession * sess, GstRTCPPacket * packet, RTPArrivalStats * arrival) { GST_DEBUG ("received APP"); } static gboolean rtp_session_request_local_key_unit (RTPSession * sess, RTPSource * src, gboolean fir, GstClockTime current_time) { guint32 round_trip = 0; rtp_source_get_last_rb (src, NULL, NULL, NULL, NULL, NULL, NULL, &round_trip); if (sess->last_keyframe_request != GST_CLOCK_TIME_NONE && round_trip) { GstClockTime round_trip_in_ns = gst_util_uint64_scale (round_trip, GST_SECOND, 65536); if (sess->last_keyframe_request != GST_CLOCK_TIME_NONE && current_time - sess->last_keyframe_request < 2 * round_trip_in_ns) { GST_DEBUG ("Ignoring %s request because one was send without one " "RTT (%" GST_TIME_FORMAT " < %" GST_TIME_FORMAT ")", fir ? "FIR" : "PLI", GST_TIME_ARGS (current_time - sess->last_keyframe_request), GST_TIME_ARGS (round_trip_in_ns));; return FALSE; } } sess->last_keyframe_request = current_time; GST_LOG ("received %s request from %X %p(%p)", fir ? "FIR" : "PLI", rtp_source_get_ssrc (src), sess->callbacks.process_rtp, sess->callbacks.request_key_unit); RTP_SESSION_UNLOCK (sess); sess->callbacks.request_key_unit (sess, fir, sess->request_key_unit_user_data); RTP_SESSION_LOCK (sess); return TRUE; } static void rtp_session_process_pli (RTPSession * sess, guint32 sender_ssrc, guint32 media_ssrc, GstClockTime current_time) { RTPSource *src; if (!sess->callbacks.request_key_unit) return; src = g_hash_table_lookup (sess->ssrcs[sess->mask_idx], GINT_TO_POINTER (sender_ssrc)); if (!src) return; rtp_session_request_local_key_unit (sess, src, FALSE, current_time); } static void rtp_session_process_fir (RTPSession * sess, guint32 sender_ssrc, guint8 * fci_data, guint fci_length, GstClockTime current_time) { RTPSource *src; guint32 ssrc; guint position = 0; gboolean our_request = FALSE; if (!sess->callbacks.request_key_unit) return; if (fci_length < 8) return; src = g_hash_table_lookup (sess->ssrcs[sess->mask_idx], GINT_TO_POINTER (sender_ssrc)); /* Hack because Google fails to set the sender_ssrc correctly */ if (!src && sender_ssrc == 1) { GHashTableIter iter; if (sess->stats.sender_sources > RTP_SOURCE_IS_SENDER (sess->source) ? 2 : 1) return; g_hash_table_iter_init (&iter, sess->ssrcs[sess->mask_idx]); while (g_hash_table_iter_next (&iter, NULL, (gpointer *) & src)) { if (src != sess->source && rtp_source_is_sender (src)) break; src = NULL; } } if (!src) return; for (position = 0; position < fci_length; position += 8) { guint8 *data = fci_data + position; ssrc = GST_READ_UINT32_BE (data); if (ssrc == rtp_source_get_ssrc (sess->source)) { our_request = TRUE; break; } } if (!our_request) return; rtp_session_request_local_key_unit (sess, src, TRUE, current_time); } static void rtp_session_process_feedback (RTPSession * sess, GstRTCPPacket * packet, RTPArrivalStats * arrival, GstClockTime current_time) { GstRTCPType type = gst_rtcp_packet_get_type (packet); GstRTCPFBType fbtype = gst_rtcp_packet_fb_get_type (packet); guint32 sender_ssrc = gst_rtcp_packet_fb_get_sender_ssrc (packet); guint32 media_ssrc = gst_rtcp_packet_fb_get_media_ssrc (packet); guint8 *fci_data = gst_rtcp_packet_fb_get_fci (packet); guint fci_length = 4 * gst_rtcp_packet_fb_get_fci_length (packet); GST_DEBUG ("received feedback %d:%d from %08X about %08X with FCI of " "length %d", type, fbtype, sender_ssrc, media_ssrc, fci_length); if (g_signal_has_handler_pending (sess, rtp_session_signals[SIGNAL_ON_FEEDBACK_RTCP], 0, TRUE)) { GstBuffer *fci_buffer = NULL; if (fci_length > 0) { fci_buffer = gst_buffer_copy_region (packet->rtcp->buffer, GST_BUFFER_COPY_MEMORY, fci_data - packet->rtcp->map.data, fci_length); GST_BUFFER_TIMESTAMP (fci_buffer) = arrival->running_time; } RTP_SESSION_UNLOCK (sess); g_signal_emit (sess, rtp_session_signals[SIGNAL_ON_FEEDBACK_RTCP], 0, type, fbtype, sender_ssrc, media_ssrc, fci_buffer); RTP_SESSION_LOCK (sess); if (fci_buffer) gst_buffer_unref (fci_buffer); } if (sess->rtcp_feedback_retention_window) { RTPSource *src = g_hash_table_lookup (sess->ssrcs[sess->mask_idx], GINT_TO_POINTER (media_ssrc)); if (src) rtp_source_retain_rtcp_packet (src, packet, arrival->running_time); } if (rtp_source_get_ssrc (sess->source) == media_ssrc || /* PSFB FIR puts the media ssrc inside the FCI */ (type == GST_RTCP_TYPE_PSFB && fbtype == GST_RTCP_PSFB_TYPE_FIR)) { switch (type) { case GST_RTCP_TYPE_PSFB: switch (fbtype) { case GST_RTCP_PSFB_TYPE_PLI: rtp_session_process_pli (sess, sender_ssrc, media_ssrc, current_time); break; case GST_RTCP_PSFB_TYPE_FIR: rtp_session_process_fir (sess, sender_ssrc, fci_data, fci_length, current_time); break; default: break; } break; case GST_RTCP_TYPE_RTPFB: default: break; } } } /** * rtp_session_process_rtcp: * @sess: and #RTPSession * @buffer: an RTCP buffer * @current_time: the current system time * @ntpnstime: the current NTP time in nanoseconds * * Process an RTCP buffer in the session manager. This function takes ownership * of @buffer. * * Returns: a #GstFlowReturn. */ GstFlowReturn rtp_session_process_rtcp (RTPSession * sess, GstBuffer * buffer, GstClockTime current_time, guint64 ntpnstime) { GstRTCPPacket packet; gboolean more, is_bye = FALSE, do_sync = FALSE; RTPArrivalStats arrival = { NULL, }; GstFlowReturn result = GST_FLOW_OK; GstRTCPBuffer rtcp = { NULL, }; g_return_val_if_fail (RTP_IS_SESSION (sess), GST_FLOW_ERROR); g_return_val_if_fail (GST_IS_BUFFER (buffer), GST_FLOW_ERROR); if (!gst_rtcp_buffer_validate (buffer)) goto invalid_packet; GST_DEBUG ("received RTCP packet"); RTP_SESSION_LOCK (sess); /* update arrival stats */ update_arrival_stats (sess, &arrival, FALSE, buffer, current_time, -1, ntpnstime); if (sess->sent_bye) goto ignore; /* start processing the compound packet */ gst_rtcp_buffer_map (buffer, GST_MAP_READ, &rtcp); more = gst_rtcp_buffer_get_first_packet (&rtcp, &packet); while (more) { GstRTCPType type; type = gst_rtcp_packet_get_type (&packet); /* when we are leaving the session, we should ignore all non-BYE messages */ if (sess->source->received_bye && type != GST_RTCP_TYPE_BYE) { GST_DEBUG ("ignoring non-BYE RTCP packet because we are leaving"); goto next; } switch (type) { case GST_RTCP_TYPE_SR: rtp_session_process_sr (sess, &packet, &arrival, &do_sync); break; case GST_RTCP_TYPE_RR: rtp_session_process_rr (sess, &packet, &arrival); break; case GST_RTCP_TYPE_SDES: rtp_session_process_sdes (sess, &packet, &arrival); break; case GST_RTCP_TYPE_BYE: is_bye = TRUE; /* don't try to attempt lip-sync anymore for streams with a BYE */ do_sync = FALSE; rtp_session_process_bye (sess, &packet, &arrival); break; case GST_RTCP_TYPE_APP: rtp_session_process_app (sess, &packet, &arrival); break; case GST_RTCP_TYPE_RTPFB: case GST_RTCP_TYPE_PSFB: rtp_session_process_feedback (sess, &packet, &arrival, current_time); break; default: GST_WARNING ("got unknown RTCP packet"); break; } next: more = gst_rtcp_packet_move_to_next (&packet); } gst_rtcp_buffer_unmap (&rtcp); /* if we are scheduling a BYE, we only want to count bye packets, else we * count everything */ if (sess->source->received_bye) { if (is_bye) { sess->stats.bye_members++; UPDATE_AVG (sess->stats.avg_rtcp_packet_size, arrival.bytes); } } else { /* keep track of average packet size */ UPDATE_AVG (sess->stats.avg_rtcp_packet_size, arrival.bytes); } GST_DEBUG ("%p, received RTCP packet, avg size %u, %u", &sess->stats, sess->stats.avg_rtcp_packet_size, arrival.bytes); RTP_SESSION_UNLOCK (sess); clean_arrival_stats (&arrival); /* notify caller of sr packets in the callback */ if (do_sync && sess->callbacks.sync_rtcp) { /* make writable, we might want to change the buffer */ buffer = gst_buffer_make_writable (buffer); result = sess->callbacks.sync_rtcp (sess, sess->source, buffer, sess->sync_rtcp_user_data); } else gst_buffer_unref (buffer); return result; /* ERRORS */ invalid_packet: { GST_DEBUG ("invalid RTCP packet received"); gst_buffer_unref (buffer); return GST_FLOW_OK; } ignore: { RTP_SESSION_UNLOCK (sess); gst_buffer_unref (buffer); clean_arrival_stats (&arrival); GST_DEBUG ("ignoring RTCP packet because we left"); return GST_FLOW_OK; } } /** * rtp_session_update_send_caps: * @sess: an #RTPSession * @caps: a #GstCaps * * Update the caps of the sender in the rtp session. */ void rtp_session_update_send_caps (RTPSession * sess, GstCaps * caps) { g_return_val_if_fail (RTP_IS_SESSION (sess), GST_FLOW_ERROR); g_return_val_if_fail (GST_IS_CAPS (caps), GST_FLOW_ERROR); GST_LOG ("received caps %" GST_PTR_FORMAT, caps); RTP_SESSION_LOCK (sess); rtp_source_update_caps (sess->source, caps); RTP_SESSION_UNLOCK (sess); } /** * rtp_session_send_rtp: * @sess: an #RTPSession * @data: pointer to either an RTP buffer or a list of RTP buffers * @is_list: TRUE when @data is a buffer list * @current_time: the current system time * @running_time: the running time of @data * * Send the RTP buffer in the session manager. This function takes ownership of * @buffer. * * Returns: a #GstFlowReturn. */ GstFlowReturn rtp_session_send_rtp (RTPSession * sess, gpointer data, gboolean is_list, GstClockTime current_time, GstClockTime running_time) { GstFlowReturn result; RTPSource *source; gboolean prevsender; guint64 oldrate; g_return_val_if_fail (RTP_IS_SESSION (sess), GST_FLOW_ERROR); g_return_val_if_fail (is_list || GST_IS_BUFFER (data), GST_FLOW_ERROR); GST_LOG ("received RTP %s for sending", is_list ? "list" : "packet"); RTP_SESSION_LOCK (sess); source = sess->source; /* update last activity */ source->last_rtp_activity = current_time; prevsender = RTP_SOURCE_IS_SENDER (source); oldrate = source->bitrate; /* we use our own source to send */ result = rtp_source_send_rtp (source, data, is_list, running_time); if (RTP_SOURCE_IS_SENDER (source) && !prevsender) sess->stats.sender_sources++; if (oldrate != source->bitrate) sess->recalc_bandwidth = TRUE; RTP_SESSION_UNLOCK (sess); return result; } static void add_bitrates (gpointer key, RTPSource * source, gdouble * bandwidth) { *bandwidth += source->bitrate; } static GstClockTime calculate_rtcp_interval (RTPSession * sess, gboolean deterministic, gboolean first) { GstClockTime result; /* recalculate bandwidth when it changed */ if (sess->recalc_bandwidth) { gdouble bandwidth; if (sess->bandwidth > 0) bandwidth = sess->bandwidth; else { /* If it is <= 0, then try to estimate the actual bandwidth */ bandwidth = sess->source->bitrate; g_hash_table_foreach (sess->cnames, (GHFunc) add_bitrates, &bandwidth); bandwidth /= 8.0; } if (bandwidth < 8000) bandwidth = RTP_STATS_BANDWIDTH; rtp_stats_set_bandwidths (&sess->stats, bandwidth, sess->rtcp_bandwidth, sess->rtcp_rs_bandwidth, sess->rtcp_rr_bandwidth); sess->recalc_bandwidth = FALSE; } if (sess->source->received_bye) { result = rtp_stats_calculate_bye_interval (&sess->stats); } else { result = rtp_stats_calculate_rtcp_interval (&sess->stats, RTP_SOURCE_IS_SENDER (sess->source), first); } GST_DEBUG ("next deterministic interval: %" GST_TIME_FORMAT ", first %d", GST_TIME_ARGS (result), first); if (!deterministic && result != GST_CLOCK_TIME_NONE) result = rtp_stats_add_rtcp_jitter (&sess->stats, result); GST_DEBUG ("next interval: %" GST_TIME_FORMAT, GST_TIME_ARGS (result)); return result; } /* Stop the current @sess and schedule a BYE message for the other members. * One must have the session lock to call this function */ static GstFlowReturn rtp_session_schedule_bye_locked (RTPSession * sess, const gchar * reason, GstClockTime current_time) { GstFlowReturn result = GST_FLOW_OK; RTPSource *source; GstClockTime interval; g_return_val_if_fail (RTP_IS_SESSION (sess), GST_FLOW_ERROR); source = sess->source; /* ignore more BYEs */ if (source->received_bye) goto done; /* we have BYE now */ source->received_bye = TRUE; /* at least one member wants to send a BYE */ g_free (sess->bye_reason); sess->bye_reason = g_strdup (reason); INIT_AVG (sess->stats.avg_rtcp_packet_size, 100); sess->stats.bye_members = 1; sess->first_rtcp = TRUE; sess->sent_bye = FALSE; sess->allow_early = TRUE; /* reschedule transmission */ sess->last_rtcp_send_time = current_time; interval = calculate_rtcp_interval (sess, FALSE, TRUE); sess->next_rtcp_check_time = current_time + interval; GST_DEBUG ("Schedule BYE for %" GST_TIME_FORMAT ", %" GST_TIME_FORMAT, GST_TIME_ARGS (interval), GST_TIME_ARGS (sess->next_rtcp_check_time)); RTP_SESSION_UNLOCK (sess); /* notify app of reconsideration */ if (sess->callbacks.reconsider) sess->callbacks.reconsider (sess, sess->reconsider_user_data); RTP_SESSION_LOCK (sess); done: return result; } /** * rtp_session_schedule_bye: * @sess: an #RTPSession * @reason: a reason or NULL * @current_time: the current system time * * Stop the current @sess and schedule a BYE message for the other members. * * Returns: a #GstFlowReturn. */ GstFlowReturn rtp_session_schedule_bye (RTPSession * sess, const gchar * reason, GstClockTime current_time) { GstFlowReturn result = GST_FLOW_OK; g_return_val_if_fail (RTP_IS_SESSION (sess), GST_FLOW_ERROR); RTP_SESSION_LOCK (sess); result = rtp_session_schedule_bye_locked (sess, reason, current_time); RTP_SESSION_UNLOCK (sess); return result; } /** * rtp_session_next_timeout: * @sess: an #RTPSession * @current_time: the current system time * * Get the next time we should perform session maintenance tasks. * * Returns: a time when rtp_session_on_timeout() should be called with the * current system time. */ GstClockTime rtp_session_next_timeout (RTPSession * sess, GstClockTime current_time) { GstClockTime result, interval = 0; g_return_val_if_fail (RTP_IS_SESSION (sess), GST_CLOCK_TIME_NONE); RTP_SESSION_LOCK (sess); if (GST_CLOCK_TIME_IS_VALID (sess->next_early_rtcp_time)) { result = sess->next_early_rtcp_time; goto early_exit; } result = sess->next_rtcp_check_time; GST_DEBUG ("current time: %" GST_TIME_FORMAT ", next :%" GST_TIME_FORMAT, GST_TIME_ARGS (current_time), GST_TIME_ARGS (result)); if (result < current_time) { GST_DEBUG ("take current time as base"); /* our previous check time expired, start counting from the current time * again. */ result = current_time; } if (sess->source->received_bye) { if (sess->sent_bye) { GST_DEBUG ("we sent BYE already"); interval = GST_CLOCK_TIME_NONE; } else if (sess->stats.active_sources >= 50) { GST_DEBUG ("reconsider BYE, more than 50 sources"); /* reconsider BYE if members >= 50 */ interval = calculate_rtcp_interval (sess, FALSE, TRUE); } } else { if (sess->first_rtcp) { GST_DEBUG ("first RTCP packet"); /* we are called for the first time */ interval = calculate_rtcp_interval (sess, FALSE, TRUE); } else if (sess->next_rtcp_check_time < current_time) { GST_DEBUG ("old check time expired, getting new timeout"); /* get a new timeout when we need to */ interval = calculate_rtcp_interval (sess, FALSE, FALSE); } } if (interval != GST_CLOCK_TIME_NONE) result += interval; else result = GST_CLOCK_TIME_NONE; sess->next_rtcp_check_time = result; early_exit: GST_DEBUG ("current time: %" GST_TIME_FORMAT ", next time: %" GST_TIME_FORMAT, GST_TIME_ARGS (current_time), GST_TIME_ARGS (result)); RTP_SESSION_UNLOCK (sess); return result; } typedef struct { GstRTCPBuffer rtcpbuf; RTPSession *sess; GstBuffer *rtcp; GstClockTime current_time; guint64 ntpnstime; GstClockTime running_time; GstClockTime interval; GstRTCPPacket packet; gboolean is_bye; gboolean has_sdes; gboolean is_early; gboolean may_suppress; } ReportData; static void session_start_rtcp (RTPSession * sess, ReportData * data) { GstRTCPPacket *packet = &data->packet; RTPSource *own = sess->source; GstRTCPBuffer *rtcp = &data->rtcpbuf; data->rtcp = gst_rtcp_buffer_new (sess->mtu); gst_rtcp_buffer_map (data->rtcp, GST_MAP_READWRITE, rtcp); if (RTP_SOURCE_IS_SENDER (own)) { guint64 ntptime; guint32 rtptime; guint32 packet_count, octet_count; /* we are a sender, create SR */ GST_DEBUG ("create SR for SSRC %08x", own->ssrc); gst_rtcp_buffer_add_packet (rtcp, GST_RTCP_TYPE_SR, packet); /* get latest stats */ rtp_source_get_new_sr (own, data->ntpnstime, data->running_time, &ntptime, &rtptime, &packet_count, &octet_count); /* store stats */ rtp_source_process_sr (own, data->current_time, ntptime, rtptime, packet_count, octet_count); /* fill in sender report info */ gst_rtcp_packet_sr_set_sender_info (packet, own->ssrc, ntptime, rtptime, packet_count, octet_count); } else { /* we are only receiver, create RR */ GST_DEBUG ("create RR for SSRC %08x", own->ssrc); gst_rtcp_buffer_add_packet (rtcp, GST_RTCP_TYPE_RR, packet); gst_rtcp_packet_rr_set_ssrc (packet, own->ssrc); } } /* construct a Sender or Receiver Report */ static void session_report_blocks (const gchar * key, RTPSource * source, ReportData * data) { RTPSession *sess = data->sess; GstRTCPPacket *packet = &data->packet; /* create a new buffer if needed */ if (data->rtcp == NULL) { session_start_rtcp (sess, data); } else if (data->is_early) { /* Put a single RR or SR in minimal compound packets */ return; } if (gst_rtcp_packet_get_rb_count (packet) < GST_RTCP_MAX_RB_COUNT) { /* only report about other sender sources */ if (source != sess->source && RTP_SOURCE_IS_SENDER (source)) { guint8 fractionlost; gint32 packetslost; guint32 exthighestseq, jitter; guint32 lsr, dlsr; /* get new stats */ rtp_source_get_new_rb (source, data->current_time, &fractionlost, &packetslost, &exthighestseq, &jitter, &lsr, &dlsr); /* store last generated RR packet */ source->last_rr.is_valid = TRUE; source->last_rr.fractionlost = fractionlost; source->last_rr.packetslost = packetslost; source->last_rr.exthighestseq = exthighestseq; source->last_rr.jitter = jitter; source->last_rr.lsr = lsr; source->last_rr.dlsr = dlsr; /* packet is not yet filled, add report block for this source. */ gst_rtcp_packet_add_rb (packet, source->ssrc, fractionlost, packetslost, exthighestseq, jitter, lsr, dlsr); } } } /* perform cleanup of sources that timed out */ static void session_cleanup (const gchar * key, RTPSource * source, ReportData * data) { gboolean remove = FALSE; gboolean byetimeout = FALSE; gboolean sendertimeout = FALSE; gboolean is_sender, is_active; RTPSession *sess = data->sess; GstClockTime interval, binterval; GstClockTime btime; is_sender = RTP_SOURCE_IS_SENDER (source); is_active = RTP_SOURCE_IS_ACTIVE (source); /* our own rtcp interval may have been forced low by secondary configuration, * while sender side may still operate with higher interval, * so do not just take our interval to decide on timing out sender, * but take (if data->interval <= 5 * GST_SECOND): * interval = CLAMP (sender_interval, data->interval, 5 * GST_SECOND) * where sender_interval is difference between last 2 received RTCP reports */ if (data->interval >= 5 * GST_SECOND || (source == sess->source)) { binterval = data->interval; } else { GST_LOG ("prev_rtcp %" GST_TIME_FORMAT ", last_rtcp %" GST_TIME_FORMAT, GST_TIME_ARGS (source->stats.prev_rtcptime), GST_TIME_ARGS (source->stats.last_rtcptime)); /* if not received enough yet, fallback to larger default */ if (source->stats.last_rtcptime > source->stats.prev_rtcptime) binterval = source->stats.last_rtcptime - source->stats.prev_rtcptime; else binterval = 5 * GST_SECOND; binterval = CLAMP (binterval, data->interval, 5 * GST_SECOND); } GST_LOG ("timeout base interval %" GST_TIME_FORMAT, GST_TIME_ARGS (binterval)); /* check for our own source, we don't want to delete our own source. */ if (!(source == sess->source)) { if (source->received_bye) { /* if we received a BYE from the source, remove the source after some * time. */ if (data->current_time > source->bye_time && data->current_time - source->bye_time > sess->stats.bye_timeout) { GST_DEBUG ("removing BYE source %08x", source->ssrc); remove = TRUE; byetimeout = TRUE; } } /* sources that were inactive for more than 5 times the deterministic reporting * interval get timed out. the min timeout is 5 seconds. */ /* mind old time that might pre-date last time going to PLAYING */ btime = MAX (source->last_activity, sess->start_time); if (data->current_time > btime) { interval = MAX (binterval * 5, 5 * GST_SECOND); if (data->current_time - btime > interval) { GST_DEBUG ("removing timeout source %08x, last %" GST_TIME_FORMAT, source->ssrc, GST_TIME_ARGS (btime)); remove = TRUE; } } } /* senders that did not send for a long time become a receiver, this also * holds for our own source. */ if (is_sender) { /* mind old time that might pre-date last time going to PLAYING */ btime = MAX (source->last_rtp_activity, sess->start_time); if (data->current_time > btime) { interval = MAX (binterval * 2, 5 * GST_SECOND); if (data->current_time - btime > interval) { GST_DEBUG ("sender source %08x timed out and became receiver, last %" GST_TIME_FORMAT, source->ssrc, GST_TIME_ARGS (btime)); source->is_sender = FALSE; sess->stats.sender_sources--; sendertimeout = TRUE; } } } if (remove) { sess->total_sources--; if (is_sender) sess->stats.sender_sources--; if (is_active) sess->stats.active_sources--; if (byetimeout) on_bye_timeout (sess, source); else on_timeout (sess, source); } else { if (sendertimeout) on_sender_timeout (sess, source); } source->closing = remove; } static void session_sdes (RTPSession * sess, ReportData * data) { GstRTCPPacket *packet = &data->packet; const GstStructure *sdes; gint i, n_fields; GstRTCPBuffer *rtcp = &data->rtcpbuf; /* add SDES packet */ gst_rtcp_buffer_add_packet (rtcp, GST_RTCP_TYPE_SDES, packet); gst_rtcp_packet_sdes_add_item (packet, sess->source->ssrc); sdes = rtp_source_get_sdes_struct (sess->source); /* add all fields in the structure, the order is not important. */ n_fields = gst_structure_n_fields (sdes); for (i = 0; i < n_fields; ++i) { const gchar *field; const gchar *value; GstRTCPSDESType type; field = gst_structure_nth_field_name (sdes, i); if (field == NULL) continue; value = gst_structure_get_string (sdes, field); if (value == NULL) continue; type = gst_rtcp_sdes_name_to_type (field); /* Early packets are minimal and only include the CNAME */ if (data->is_early && type != GST_RTCP_SDES_CNAME) continue; if (type > GST_RTCP_SDES_END && type < GST_RTCP_SDES_PRIV) { gst_rtcp_packet_sdes_add_entry (packet, type, strlen (value), (const guint8 *) value); } else if (type == GST_RTCP_SDES_PRIV) { gsize prefix_len; gsize value_len; gsize data_len; guint8 data[256]; /* don't accept entries that are too big */ prefix_len = strlen (field); if (prefix_len > 255) continue; value_len = strlen (value); if (value_len > 255) continue; data_len = 1 + prefix_len + value_len; if (data_len > 255) continue; data[0] = prefix_len; memcpy (&data[1], field, prefix_len); memcpy (&data[1 + prefix_len], value, value_len); gst_rtcp_packet_sdes_add_entry (packet, type, data_len, data); } } data->has_sdes = TRUE; } /* schedule a BYE packet */ static void session_bye (RTPSession * sess, ReportData * data) { GstRTCPPacket *packet = &data->packet; GstRTCPBuffer *rtcp = &data->rtcpbuf; /* open packet */ session_start_rtcp (sess, data); /* add SDES */ session_sdes (sess, data); /* add a BYE packet */ gst_rtcp_buffer_add_packet (rtcp, GST_RTCP_TYPE_BYE, packet); gst_rtcp_packet_bye_add_ssrc (packet, sess->source->ssrc); if (sess->bye_reason) gst_rtcp_packet_bye_set_reason (packet, sess->bye_reason); /* we have a BYE packet now */ data->is_bye = TRUE; } static gboolean is_rtcp_time (RTPSession * sess, GstClockTime current_time, ReportData * data) { GstClockTime new_send_time, elapsed; if (data->is_early && sess->next_early_rtcp_time < current_time) goto early; /* no need to check yet */ if (sess->next_rtcp_check_time > current_time) { GST_DEBUG ("no check time yet, next %" GST_TIME_FORMAT " > now %" GST_TIME_FORMAT, GST_TIME_ARGS (sess->next_rtcp_check_time), GST_TIME_ARGS (current_time)); return FALSE; } /* get elapsed time since we last reported */ elapsed = current_time - sess->last_rtcp_send_time; /* perform forward reconsideration */ new_send_time = rtp_stats_add_rtcp_jitter (&sess->stats, data->interval); GST_DEBUG ("forward reconsideration %" GST_TIME_FORMAT ", elapsed %" GST_TIME_FORMAT, GST_TIME_ARGS (new_send_time), GST_TIME_ARGS (elapsed)); new_send_time += sess->last_rtcp_send_time; /* check if reconsideration */ if (current_time < new_send_time) { GST_DEBUG ("reconsider RTCP for %" GST_TIME_FORMAT, GST_TIME_ARGS (new_send_time)); /* store new check time */ sess->next_rtcp_check_time = new_send_time; return FALSE; } early: new_send_time = calculate_rtcp_interval (sess, FALSE, FALSE); GST_DEBUG ("can send RTCP now, next interval %" GST_TIME_FORMAT, GST_TIME_ARGS (new_send_time)); sess->next_rtcp_check_time = current_time + new_send_time; /* Apply the rules from RFC 4585 section 3.5.3 */ if (sess->stats.min_interval != 0 && !sess->first_rtcp) { GstClockTimeDiff T_rr_current_interval = g_random_double_range (0.5, 1.5) * sess->stats.min_interval; /* This will caused the RTCP to be suppressed if no FB packets are added */ if (sess->last_rtcp_send_time + T_rr_current_interval > sess->next_rtcp_check_time) { GST_DEBUG ("RTCP packet could be suppressed min: %" GST_TIME_FORMAT " last: %" GST_TIME_FORMAT " + T_rr_current_interval: %" GST_TIME_FORMAT " > sess->next_rtcp_check_time: %" GST_TIME_FORMAT, GST_TIME_ARGS (sess->stats.min_interval), GST_TIME_ARGS (sess->last_rtcp_send_time), GST_TIME_ARGS (T_rr_current_interval), GST_TIME_ARGS (sess->next_rtcp_check_time)); data->may_suppress = TRUE; } } return TRUE; } static void clone_ssrcs_hashtable (gchar * key, RTPSource * source, GHashTable * hash_table) { g_hash_table_insert (hash_table, key, g_object_ref (source)); } static gboolean remove_closing_sources (const gchar * key, RTPSource * source, gpointer * data) { return source->closing; } /** * rtp_session_on_timeout: * @sess: an #RTPSession * @current_time: the current system time * @ntpnstime: the current NTP time in nanoseconds * @running_time: the current running_time of the pipeline * * Perform maintenance actions after the timeout obtained with * rtp_session_next_timeout() expired. * * This function will perform timeouts of receivers and senders, send a BYE * packet or generate RTCP packets with current session stats. * * This function can call the #RTPSessionSendRTCP callback, possibly multiple * times, for each packet that should be processed. * * Returns: a #GstFlowReturn. */ GstFlowReturn rtp_session_on_timeout (RTPSession * sess, GstClockTime current_time, guint64 ntpnstime, GstClockTime running_time) { GstFlowReturn result = GST_FLOW_OK; ReportData data = { GST_RTCP_BUFFER_INIT }; RTPSource *own; GHashTable *table_copy; gboolean notify = FALSE; g_return_val_if_fail (RTP_IS_SESSION (sess), GST_FLOW_ERROR); GST_DEBUG ("reporting at %" GST_TIME_FORMAT ", NTP time %" GST_TIME_FORMAT, GST_TIME_ARGS (current_time), GST_TIME_ARGS (ntpnstime)); data.sess = sess; data.rtcp = NULL; data.current_time = current_time; data.ntpnstime = ntpnstime; data.is_bye = FALSE; data.has_sdes = FALSE; data.may_suppress = FALSE; data.running_time = running_time; own = sess->source; RTP_SESSION_LOCK (sess); /* get a new interval, we need this for various cleanups etc */ data.interval = calculate_rtcp_interval (sess, TRUE, sess->first_rtcp); /* Make a local copy of the hashtable. We need to do this because the * cleanup stage below releases the session lock. */ table_copy = g_hash_table_new_full (NULL, NULL, NULL, (GDestroyNotify) g_object_unref); g_hash_table_foreach (sess->ssrcs[sess->mask_idx], (GHFunc) clone_ssrcs_hashtable, table_copy); /* Clean up the session, mark the source for removing, this might release the * session lock. */ g_hash_table_foreach (table_copy, (GHFunc) session_cleanup, &data); g_hash_table_destroy (table_copy); /* Now remove the marked sources */ g_hash_table_foreach_remove (sess->ssrcs[sess->mask_idx], (GHRFunc) remove_closing_sources, NULL); if (GST_CLOCK_TIME_IS_VALID (sess->next_early_rtcp_time)) data.is_early = TRUE; else data.is_early = FALSE; /* see if we need to generate SR or RR packets */ if (is_rtcp_time (sess, current_time, &data)) { if (own->received_bye) { /* generate BYE instead */ GST_DEBUG ("generating BYE message"); session_bye (sess, &data); sess->sent_bye = TRUE; } else { /* loop over all known sources and do something */ g_hash_table_foreach (sess->ssrcs[sess->mask_idx], (GHFunc) session_report_blocks, &data); } } if (data.rtcp) { /* we keep track of the last report time in order to timeout inactive * receivers or senders */ if (!data.is_early && !data.may_suppress) sess->last_rtcp_send_time = data.current_time; sess->first_rtcp = FALSE; sess->next_early_rtcp_time = GST_CLOCK_TIME_NONE; /* add SDES for this source when not already added */ if (!data.has_sdes) session_sdes (sess, &data); } /* check for outdated collisions */ GST_DEBUG ("Timing out collisions"); rtp_source_timeout (sess->source, current_time, /* "a relatively long time" -- RFC 3550 section 8.2 */ RTP_STATS_MIN_INTERVAL * GST_SECOND * 10, running_time - sess->rtcp_feedback_retention_window); if (sess->change_ssrc) { GST_DEBUG ("need to change our SSRC (%08x)", own->ssrc); g_hash_table_steal (sess->ssrcs[sess->mask_idx], GINT_TO_POINTER (own->ssrc)); own->ssrc = rtp_session_create_new_ssrc (sess); rtp_source_reset (own); g_hash_table_insert (sess->ssrcs[sess->mask_idx], GINT_TO_POINTER (own->ssrc), own); g_free (sess->bye_reason); sess->bye_reason = NULL; sess->sent_bye = FALSE; sess->change_ssrc = FALSE; notify = TRUE; GST_DEBUG ("changed our SSRC to %08x", own->ssrc); } sess->allow_early = TRUE; RTP_SESSION_UNLOCK (sess); if (notify) g_object_notify (G_OBJECT (sess), "internal-ssrc"); /* push out the RTCP packet */ if (data.rtcp) { gboolean do_not_suppress; gst_rtcp_buffer_unmap (&data.rtcpbuf); /* Give the user a change to add its own packet */ g_signal_emit (sess, rtp_session_signals[SIGNAL_ON_SENDING_RTCP], 0, data.rtcp, data.is_early, &do_not_suppress); if (sess->callbacks.send_rtcp && (do_not_suppress || !data.may_suppress)) { guint packet_size; packet_size = gst_buffer_get_size (data.rtcp) + sess->header_len; UPDATE_AVG (sess->stats.avg_rtcp_packet_size, packet_size); GST_DEBUG ("%p, sending RTCP packet, avg size %u, %u", &sess->stats, sess->stats.avg_rtcp_packet_size, packet_size); result = sess->callbacks.send_rtcp (sess, own, data.rtcp, sess->sent_bye, sess->send_rtcp_user_data); } else { GST_DEBUG ("freeing packet callback: %p" " do_not_suppress: %d may_suppress: %d", sess->callbacks.send_rtcp, do_not_suppress, data.may_suppress); gst_buffer_unref (data.rtcp); } } return result; } void rtp_session_request_early_rtcp (RTPSession * sess, GstClockTime current_time, GstClockTimeDiff max_delay) { GstClockTime T_dither_max; /* Implements the algorithm described in RFC 4585 section 3.5.2 */ RTP_SESSION_LOCK (sess); /* Check if already requested */ /* RFC 4585 section 3.5.2 step 2 */ if (GST_CLOCK_TIME_IS_VALID (sess->next_early_rtcp_time)) goto dont_send; /* Ignore the request a scheduled packet will be in time anyway */ if (current_time + max_delay > sess->next_rtcp_check_time) goto dont_send; /* RFC 4585 section 3.5.2 step 2b */ /* If the total sources is <=2, then there is only us and one peer */ if (sess->total_sources <= 2) { T_dither_max = 0; } else { /* Divide by 2 because l = 0.5 */ T_dither_max = sess->next_rtcp_check_time - sess->last_rtcp_send_time; T_dither_max /= 2; } /* RFC 4585 section 3.5.2 step 3 */ if (current_time + T_dither_max > sess->next_rtcp_check_time) goto dont_send; /* RFC 4585 section 3.5.2 step 4 * Don't send if allow_early is FALSE, but not if we are in * immediate mode, meaning we are part of a group of at most the * application-specific threshold. */ if (sess->total_sources > sess->rtcp_immediate_feedback_threshold && sess->allow_early == FALSE) goto dont_send; if (T_dither_max) { /* Schedule an early transmission later */ sess->next_early_rtcp_time = g_random_double () * T_dither_max + current_time; } else { /* If no dithering, schedule it for NOW */ sess->next_early_rtcp_time = current_time; } RTP_SESSION_UNLOCK (sess); /* notify app of need to send packet early * and therefore of timeout change */ if (sess->callbacks.reconsider) sess->callbacks.reconsider (sess, sess->reconsider_user_data); return; dont_send: RTP_SESSION_UNLOCK (sess); } gboolean rtp_session_request_key_unit (RTPSession * sess, guint32 ssrc, GstClockTime now, gboolean fir, gint count) { RTPSource *src = g_hash_table_lookup (sess->ssrcs[sess->mask_idx], GUINT_TO_POINTER (ssrc)); if (!src) return FALSE; if (fir) { src->send_pli = FALSE; src->send_fir = TRUE; if (count == -1 || count != src->last_fir_count) src->current_send_fir_seqnum++; src->last_fir_count = count; } else if (!src->send_fir) { src->send_pli = TRUE; } rtp_session_request_early_rtcp (sess, now, 200 * GST_MSECOND); return TRUE; } static gboolean has_pli_compare_func (gconstpointer a, gconstpointer ignored) { GstRTCPPacket packet; GstRTCPBuffer rtcp = { NULL, }; gboolean ret = FALSE; gst_rtcp_buffer_map ((GstBuffer *) a, GST_MAP_READ, &rtcp); if (gst_rtcp_buffer_get_first_packet (&rtcp, &packet)) { if (gst_rtcp_packet_get_type (&packet) == GST_RTCP_TYPE_PSFB && gst_rtcp_packet_fb_get_type (&packet) == GST_RTCP_PSFB_TYPE_PLI) ret = TRUE; } gst_rtcp_buffer_unmap (&rtcp); return ret; } static gboolean rtp_session_on_sending_rtcp (RTPSession * sess, GstBuffer * buffer, gboolean early) { gboolean ret = FALSE; GHashTableIter iter; gpointer key, value; gboolean started_fir = FALSE; GstRTCPPacket fir_rtcppacket; GstRTCPBuffer rtcp = { NULL, }; RTP_SESSION_LOCK (sess); gst_rtcp_buffer_map (buffer, GST_MAP_READWRITE, &rtcp); g_hash_table_iter_init (&iter, sess->ssrcs[sess->mask_idx]); while (g_hash_table_iter_next (&iter, &key, &value)) { guint media_ssrc = GPOINTER_TO_UINT (key); RTPSource *media_src = value; guint8 *fci_data; if (media_src->send_fir) { if (!started_fir) { if (!gst_rtcp_buffer_add_packet (&rtcp, GST_RTCP_TYPE_PSFB, &fir_rtcppacket)) break; gst_rtcp_packet_fb_set_type (&fir_rtcppacket, GST_RTCP_PSFB_TYPE_FIR); gst_rtcp_packet_fb_set_sender_ssrc (&fir_rtcppacket, rtp_source_get_ssrc (sess->source)); gst_rtcp_packet_fb_set_media_ssrc (&fir_rtcppacket, 0); if (!gst_rtcp_packet_fb_set_fci_length (&fir_rtcppacket, 2)) { gst_rtcp_packet_remove (&fir_rtcppacket); break; } ret = TRUE; started_fir = TRUE; } else { if (!gst_rtcp_packet_fb_set_fci_length (&fir_rtcppacket, !gst_rtcp_packet_fb_get_fci_length (&fir_rtcppacket) + 2)) break; } fci_data = gst_rtcp_packet_fb_get_fci (&fir_rtcppacket) - ((gst_rtcp_packet_fb_get_fci_length (&fir_rtcppacket) - 2) * 4); GST_WRITE_UINT32_BE (fci_data, media_ssrc); fci_data += 4; fci_data[0] = media_src->current_send_fir_seqnum; fci_data[1] = fci_data[2] = fci_data[3] = 0; media_src->send_fir = FALSE; } } g_hash_table_iter_init (&iter, sess->ssrcs[sess->mask_idx]); while (g_hash_table_iter_next (&iter, &key, &value)) { guint media_ssrc = GPOINTER_TO_UINT (key); RTPSource *media_src = value; GstRTCPPacket pli_rtcppacket; if (media_src->send_pli && !rtp_source_has_retained (media_src, has_pli_compare_func, NULL)) { if (!gst_rtcp_buffer_add_packet (&rtcp, GST_RTCP_TYPE_PSFB, &pli_rtcppacket)) /* Break because the packet is full, will put next request in a * further packet */ break; gst_rtcp_packet_fb_set_type (&pli_rtcppacket, GST_RTCP_PSFB_TYPE_PLI); gst_rtcp_packet_fb_set_sender_ssrc (&pli_rtcppacket, rtp_source_get_ssrc (sess->source)); gst_rtcp_packet_fb_set_media_ssrc (&pli_rtcppacket, media_ssrc); ret = TRUE; } media_src->send_pli = FALSE; } gst_rtcp_buffer_unmap (&rtcp); RTP_SESSION_UNLOCK (sess); return ret; } static void rtp_session_send_rtcp (RTPSession * sess, GstClockTimeDiff max_delay) { GstClockTime now; if (!sess->callbacks.send_rtcp) return; now = sess->callbacks.request_time (sess, sess->request_time_user_data); rtp_session_request_early_rtcp (sess, now, max_delay); }