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* Copyright (C) <2007> Wim Taymans <wim.taymans@gmail.com>
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* This library is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Library General Public
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* License as published by the Free Software Foundation; either
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* version 2 of the License, or (at your option) any later version.
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* This library is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Library General Public License for more details.
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* You should have received a copy of the GNU Library General Public
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* License along with this library; if not, write to the
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* Free Software Foundation, Inc., 59 Temple Place - Suite 330,
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* Boston, MA 02111-1307, USA.
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#include <gst/rtp/gstrtpbuffer.h>
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#include <gst/rtp/gstrtcpbuffer.h>
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#include <gst/netbuffer/gstnetbuffer.h>
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#include "gstrtpbin-marshal.h"
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#include "rtpsession.h"
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GST_DEBUG_CATEGORY_STATIC (rtp_session_debug);
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#define GST_CAT_DEFAULT rtp_session_debug
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/* signals and args */
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SIGNAL_GET_SOURCE_BY_SSRC,
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SIGNAL_ON_SSRC_COLLISION,
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SIGNAL_ON_SSRC_VALIDATED,
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SIGNAL_ON_SSRC_ACTIVE,
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SIGNAL_ON_BYE_TIMEOUT,
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SIGNAL_ON_SENDER_TIMEOUT,
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#define DEFAULT_INTERNAL_SOURCE NULL
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#define DEFAULT_BANDWIDTH RTP_STATS_BANDWIDTH
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#define DEFAULT_RTCP_FRACTION RTP_STATS_RTCP_BANDWIDTH
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#define DEFAULT_RTCP_MTU 1400
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#define DEFAULT_SDES_CNAME NULL
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#define DEFAULT_SDES_NAME NULL
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#define DEFAULT_SDES_EMAIL NULL
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#define DEFAULT_SDES_PHONE NULL
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#define DEFAULT_SDES_LOCATION NULL
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#define DEFAULT_SDES_TOOL NULL
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#define DEFAULT_SDES_NOTE NULL
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#define DEFAULT_NUM_SOURCES 0
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#define DEFAULT_NUM_ACTIVE_SOURCES 0
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#define DEFAULT_SOURCES NULL
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PROP_NUM_ACTIVE_SOURCES,
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/* update average packet size, we keep this scaled by 16 to keep enough
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#define UPDATE_AVG(avg, val) \
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(avg) = ((val) + (15 * (avg))) >> 4;
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/* The number RTCP intervals after which to timeout entries in the
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#define RTCP_INTERVAL_COLLISION_TIMEOUT 10
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/* GObject vmethods */
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static void rtp_session_finalize (GObject * object);
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static void rtp_session_set_property (GObject * object, guint prop_id,
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const GValue * value, GParamSpec * pspec);
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static void rtp_session_get_property (GObject * object, guint prop_id,
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GValue * value, GParamSpec * pspec);
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static guint rtp_session_signals[LAST_SIGNAL] = { 0 };
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G_DEFINE_TYPE (RTPSession, rtp_session, G_TYPE_OBJECT);
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static RTPSource *obtain_source (RTPSession * sess, guint32 ssrc,
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gboolean * created, RTPArrivalStats * arrival, gboolean rtp);
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static GstFlowReturn rtp_session_schedule_bye_locked (RTPSession * sess,
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const gchar * reason, GstClockTime current_time);
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static GstClockTime calculate_rtcp_interval (RTPSession * sess,
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gboolean deterministic, gboolean first);
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rtp_session_class_init (RTPSessionClass * klass)
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GObjectClass *gobject_class;
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gobject_class = (GObjectClass *) klass;
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gobject_class->finalize = rtp_session_finalize;
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gobject_class->set_property = rtp_session_set_property;
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gobject_class->get_property = rtp_session_get_property;
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* RTPSession::get-source-by-ssrc:
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* @session: the object which received the signal
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* @ssrc: the SSRC of the RTPSource
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* Request the #RTPSource object with SSRC @ssrc in @session.
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rtp_session_signals[SIGNAL_GET_SOURCE_BY_SSRC] =
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g_signal_new ("get-source-by-ssrc", G_TYPE_FROM_CLASS (klass),
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G_SIGNAL_RUN_LAST | G_SIGNAL_ACTION, G_STRUCT_OFFSET (RTPSessionClass,
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get_source_by_ssrc), NULL, NULL, gst_rtp_bin_marshal_OBJECT__UINT,
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RTP_TYPE_SOURCE, 1, G_TYPE_UINT);
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* RTPSession::on-new-ssrc:
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* @session: the object which received the signal
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* @src: the new RTPSource
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* Notify of a new SSRC that entered @session.
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rtp_session_signals[SIGNAL_ON_NEW_SSRC] =
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g_signal_new ("on-new-ssrc", G_TYPE_FROM_CLASS (klass),
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G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (RTPSessionClass, on_new_ssrc),
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NULL, NULL, g_cclosure_marshal_VOID__OBJECT, G_TYPE_NONE, 1,
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* RTPSession::on-ssrc-collision:
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* @session: the object which received the signal
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* @src: the #RTPSource that caused a collision
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* Notify when we have an SSRC collision
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rtp_session_signals[SIGNAL_ON_SSRC_COLLISION] =
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g_signal_new ("on-ssrc-collision", G_TYPE_FROM_CLASS (klass),
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G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (RTPSessionClass, on_ssrc_collision),
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NULL, NULL, g_cclosure_marshal_VOID__OBJECT, G_TYPE_NONE, 1,
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* RTPSession::on-ssrc-validated:
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* @session: the object which received the signal
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* @src: the new validated RTPSource
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* Notify of a new SSRC that became validated.
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rtp_session_signals[SIGNAL_ON_SSRC_VALIDATED] =
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g_signal_new ("on-ssrc-validated", G_TYPE_FROM_CLASS (klass),
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G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (RTPSessionClass, on_ssrc_validated),
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NULL, NULL, g_cclosure_marshal_VOID__OBJECT, G_TYPE_NONE, 1,
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* RTPSession::on-ssrc-active:
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* @session: the object which received the signal
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* @src: the active RTPSource
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* Notify of a SSRC that is active, i.e., sending RTCP.
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rtp_session_signals[SIGNAL_ON_SSRC_ACTIVE] =
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g_signal_new ("on-ssrc-active", G_TYPE_FROM_CLASS (klass),
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G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (RTPSessionClass, on_ssrc_active),
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NULL, NULL, g_cclosure_marshal_VOID__OBJECT, G_TYPE_NONE, 1,
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* RTPSession::on-ssrc-sdes:
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* @session: the object which received the signal
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* @src: the RTPSource
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* Notify that a new SDES was received for SSRC.
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rtp_session_signals[SIGNAL_ON_SSRC_SDES] =
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g_signal_new ("on-ssrc-sdes", G_TYPE_FROM_CLASS (klass),
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G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (RTPSessionClass, on_ssrc_sdes),
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NULL, NULL, g_cclosure_marshal_VOID__OBJECT, G_TYPE_NONE, 1,
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* RTPSession::on-bye-ssrc:
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* @session: the object which received the signal
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* @src: the RTPSource that went away
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* Notify of an SSRC that became inactive because of a BYE packet.
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rtp_session_signals[SIGNAL_ON_BYE_SSRC] =
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g_signal_new ("on-bye-ssrc", G_TYPE_FROM_CLASS (klass),
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G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (RTPSessionClass, on_bye_ssrc),
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NULL, NULL, g_cclosure_marshal_VOID__OBJECT, G_TYPE_NONE, 1,
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* RTPSession::on-bye-timeout:
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* @session: the object which received the signal
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* @src: the RTPSource that timed out
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* Notify of an SSRC that has timed out because of BYE
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rtp_session_signals[SIGNAL_ON_BYE_TIMEOUT] =
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g_signal_new ("on-bye-timeout", G_TYPE_FROM_CLASS (klass),
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G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (RTPSessionClass, on_bye_timeout),
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NULL, NULL, g_cclosure_marshal_VOID__OBJECT, G_TYPE_NONE, 1,
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* RTPSession::on-timeout:
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* @session: the object which received the signal
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* @src: the RTPSource that timed out
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* Notify of an SSRC that has timed out
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rtp_session_signals[SIGNAL_ON_TIMEOUT] =
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g_signal_new ("on-timeout", G_TYPE_FROM_CLASS (klass),
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G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (RTPSessionClass, on_timeout),
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NULL, NULL, g_cclosure_marshal_VOID__OBJECT, G_TYPE_NONE, 1,
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* RTPSession::on-sender-timeout:
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* @session: the object which received the signal
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* @src: the RTPSource that timed out
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* Notify of an SSRC that was a sender but timed out and became a receiver.
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rtp_session_signals[SIGNAL_ON_SENDER_TIMEOUT] =
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g_signal_new ("on-sender-timeout", G_TYPE_FROM_CLASS (klass),
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G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (RTPSessionClass, on_sender_timeout),
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NULL, NULL, g_cclosure_marshal_VOID__OBJECT, G_TYPE_NONE, 1,
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g_object_class_install_property (gobject_class, PROP_INTERNAL_SSRC,
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g_param_spec_uint ("internal-ssrc", "Internal SSRC",
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"The internal SSRC used for the session",
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0, G_MAXUINT, 0, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
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g_object_class_install_property (gobject_class, PROP_INTERNAL_SOURCE,
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g_param_spec_object ("internal-source", "Internal Source",
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"The internal source element of the session",
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RTP_TYPE_SOURCE, G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
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g_object_class_install_property (gobject_class, PROP_BANDWIDTH,
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g_param_spec_double ("bandwidth", "Bandwidth",
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"The bandwidth of the session",
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0.0, G_MAXDOUBLE, DEFAULT_BANDWIDTH,
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G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
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g_object_class_install_property (gobject_class, PROP_RTCP_FRACTION,
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g_param_spec_double ("rtcp-fraction", "RTCP Fraction",
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"The fraction of the bandwidth used for RTCP",
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0.0, G_MAXDOUBLE, DEFAULT_RTCP_FRACTION,
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G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
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g_object_class_install_property (gobject_class, PROP_RTCP_MTU,
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g_param_spec_uint ("rtcp-mtu", "RTCP MTU",
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"The maximum size of the RTCP packets",
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16, G_MAXINT16, DEFAULT_RTCP_MTU,
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G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
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g_object_class_install_property (gobject_class, PROP_SDES_CNAME,
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g_param_spec_string ("sdes-cname", "SDES CNAME",
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"The CNAME to put in SDES messages of this session",
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DEFAULT_SDES_CNAME, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
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g_object_class_install_property (gobject_class, PROP_SDES_NAME,
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g_param_spec_string ("sdes-name", "SDES NAME",
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"The NAME to put in SDES messages of this session",
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DEFAULT_SDES_NAME, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
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g_object_class_install_property (gobject_class, PROP_SDES_EMAIL,
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g_param_spec_string ("sdes-email", "SDES EMAIL",
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"The EMAIL to put in SDES messages of this session",
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DEFAULT_SDES_EMAIL, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
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g_object_class_install_property (gobject_class, PROP_SDES_PHONE,
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g_param_spec_string ("sdes-phone", "SDES PHONE",
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"The PHONE to put in SDES messages of this session",
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DEFAULT_SDES_PHONE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
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g_object_class_install_property (gobject_class, PROP_SDES_LOCATION,
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g_param_spec_string ("sdes-location", "SDES LOCATION",
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"The LOCATION to put in SDES messages of this session",
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DEFAULT_SDES_LOCATION, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
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g_object_class_install_property (gobject_class, PROP_SDES_TOOL,
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g_param_spec_string ("sdes-tool", "SDES TOOL",
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"The TOOL to put in SDES messages of this session",
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DEFAULT_SDES_TOOL, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
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g_object_class_install_property (gobject_class, PROP_SDES_NOTE,
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g_param_spec_string ("sdes-note", "SDES NOTE",
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"The NOTE to put in SDES messages of this session",
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DEFAULT_SDES_NOTE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
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g_object_class_install_property (gobject_class, PROP_NUM_SOURCES,
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g_param_spec_uint ("num-sources", "Num Sources",
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"The number of sources in the session", 0, G_MAXUINT,
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DEFAULT_NUM_SOURCES, G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
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g_object_class_install_property (gobject_class, PROP_NUM_ACTIVE_SOURCES,
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g_param_spec_uint ("num-active-sources", "Num Active Sources",
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"The number of active sources in the session", 0, G_MAXUINT,
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DEFAULT_NUM_ACTIVE_SOURCES,
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G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
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* Get a GValue Array of all sources in the session.
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* <title>Getting the #RTPSources of a session
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* g_object_get (sess, "sources", &arr, NULL);
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* for (i = 0; i < arr->n_values; i++) {
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* val = g_value_array_get_nth (arr, i);
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* source = g_value_get_object (val);
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* g_value_array_free (arr);
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g_object_class_install_property (gobject_class, PROP_SOURCES,
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g_param_spec_boxed ("sources", "Sources",
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"An array of all known sources in the session",
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G_TYPE_VALUE_ARRAY, G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
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klass->get_source_by_ssrc =
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GST_DEBUG_FUNCPTR (rtp_session_get_source_by_ssrc);
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GST_DEBUG_CATEGORY_INIT (rtp_session_debug, "rtpsession", 0, "RTP Session");
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rtp_session_init (RTPSession * sess)
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sess->lock = g_mutex_new ();
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sess->key = g_random_int ();
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for (i = 0; i < 32; i++) {
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g_hash_table_new_full (NULL, NULL, NULL,
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(GDestroyNotify) g_object_unref);
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sess->cnames = g_hash_table_new_full (NULL, NULL, g_free, NULL);
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rtp_stats_init_defaults (&sess->stats);
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/* create an active SSRC for this session manager */
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sess->source = rtp_session_create_source (sess);
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sess->source->validated = TRUE;
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sess->source->internal = TRUE;
382
sess->stats.active_sources++;
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/* default UDP header length */
385
sess->header_len = 28;
386
sess->mtu = DEFAULT_RTCP_MTU;
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/* some default SDES entries */
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str = g_strdup_printf ("%s@%s", g_get_user_name (), g_get_host_name ());
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rtp_source_set_sdes_string (sess->source, GST_RTCP_SDES_CNAME, str);
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rtp_source_set_sdes_string (sess->source, GST_RTCP_SDES_NAME,
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rtp_source_set_sdes_string (sess->source, GST_RTCP_SDES_TOOL, "GStreamer");
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sess->first_rtcp = TRUE;
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GST_DEBUG ("%p: session using SSRC: %08x", sess, sess->source->ssrc);
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rtp_session_finalize (GObject * object)
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sess = RTP_SESSION_CAST (object);
410
g_mutex_free (sess->lock);
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for (i = 0; i < 32; i++)
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g_hash_table_destroy (sess->ssrcs[i]);
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g_free (sess->bye_reason);
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g_hash_table_destroy (sess->cnames);
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g_object_unref (sess->source);
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G_OBJECT_CLASS (rtp_session_parent_class)->finalize (object);
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copy_source (gpointer key, RTPSource * source, GValueArray * arr)
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GValue value = { 0 };
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g_value_init (&value, RTP_TYPE_SOURCE);
428
g_value_take_object (&value, source);
429
g_value_array_append (arr, &value);
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rtp_session_create_sources (RTPSession * sess)
438
RTP_SESSION_LOCK (sess);
439
/* get number of elements in the table */
440
size = g_hash_table_size (sess->ssrcs[sess->mask_idx]);
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/* create the result value array */
442
res = g_value_array_new (size);
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/* and copy all values into the array */
445
g_hash_table_foreach (sess->ssrcs[sess->mask_idx], (GHFunc) copy_source, res);
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RTP_SESSION_UNLOCK (sess);
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rtp_session_set_property (GObject * object, guint prop_id,
453
const GValue * value, GParamSpec * pspec)
457
sess = RTP_SESSION (object);
460
case PROP_INTERNAL_SSRC:
461
rtp_session_set_internal_ssrc (sess, g_value_get_uint (value));
464
rtp_session_set_bandwidth (sess, g_value_get_double (value));
466
case PROP_RTCP_FRACTION:
467
rtp_session_set_rtcp_fraction (sess, g_value_get_double (value));
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sess->mtu = g_value_get_uint (value);
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case PROP_SDES_CNAME:
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rtp_session_set_sdes_string (sess, GST_RTCP_SDES_CNAME,
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g_value_get_string (value));
477
rtp_session_set_sdes_string (sess, GST_RTCP_SDES_NAME,
478
g_value_get_string (value));
480
case PROP_SDES_EMAIL:
481
rtp_session_set_sdes_string (sess, GST_RTCP_SDES_EMAIL,
482
g_value_get_string (value));
484
case PROP_SDES_PHONE:
485
rtp_session_set_sdes_string (sess, GST_RTCP_SDES_PHONE,
486
g_value_get_string (value));
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case PROP_SDES_LOCATION:
489
rtp_session_set_sdes_string (sess, GST_RTCP_SDES_LOC,
490
g_value_get_string (value));
493
rtp_session_set_sdes_string (sess, GST_RTCP_SDES_TOOL,
494
g_value_get_string (value));
497
rtp_session_set_sdes_string (sess, GST_RTCP_SDES_NOTE,
498
g_value_get_string (value));
501
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
507
rtp_session_get_property (GObject * object, guint prop_id,
508
GValue * value, GParamSpec * pspec)
512
sess = RTP_SESSION (object);
515
case PROP_INTERNAL_SSRC:
516
g_value_set_uint (value, rtp_session_get_internal_ssrc (sess));
518
case PROP_INTERNAL_SOURCE:
519
g_value_take_object (value, rtp_session_get_internal_source (sess));
522
g_value_set_double (value, rtp_session_get_bandwidth (sess));
524
case PROP_RTCP_FRACTION:
525
g_value_set_double (value, rtp_session_get_rtcp_fraction (sess));
528
g_value_set_uint (value, sess->mtu);
530
case PROP_SDES_CNAME:
531
g_value_take_string (value, rtp_session_get_sdes_string (sess,
532
GST_RTCP_SDES_CNAME));
535
g_value_take_string (value, rtp_session_get_sdes_string (sess,
536
GST_RTCP_SDES_NAME));
538
case PROP_SDES_EMAIL:
539
g_value_take_string (value, rtp_session_get_sdes_string (sess,
540
GST_RTCP_SDES_EMAIL));
542
case PROP_SDES_PHONE:
543
g_value_take_string (value, rtp_session_get_sdes_string (sess,
544
GST_RTCP_SDES_PHONE));
546
case PROP_SDES_LOCATION:
547
g_value_take_string (value, rtp_session_get_sdes_string (sess,
551
g_value_take_string (value, rtp_session_get_sdes_string (sess,
552
GST_RTCP_SDES_TOOL));
555
g_value_take_string (value, rtp_session_get_sdes_string (sess,
556
GST_RTCP_SDES_NOTE));
558
case PROP_NUM_SOURCES:
559
g_value_set_uint (value, rtp_session_get_num_sources (sess));
561
case PROP_NUM_ACTIVE_SOURCES:
562
g_value_set_uint (value, rtp_session_get_num_active_sources (sess));
565
g_value_take_boxed (value, rtp_session_create_sources (sess));
568
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
574
on_new_ssrc (RTPSession * sess, RTPSource * source)
576
g_object_ref (source);
577
RTP_SESSION_UNLOCK (sess);
578
g_signal_emit (sess, rtp_session_signals[SIGNAL_ON_NEW_SSRC], 0, source);
579
RTP_SESSION_LOCK (sess);
580
g_object_unref (source);
584
on_ssrc_collision (RTPSession * sess, RTPSource * source)
586
g_object_ref (source);
587
RTP_SESSION_UNLOCK (sess);
588
g_signal_emit (sess, rtp_session_signals[SIGNAL_ON_SSRC_COLLISION], 0,
590
RTP_SESSION_LOCK (sess);
591
g_object_unref (source);
595
on_ssrc_validated (RTPSession * sess, RTPSource * source)
597
g_object_ref (source);
598
RTP_SESSION_UNLOCK (sess);
599
g_signal_emit (sess, rtp_session_signals[SIGNAL_ON_SSRC_VALIDATED], 0,
601
RTP_SESSION_LOCK (sess);
602
g_object_unref (source);
606
on_ssrc_active (RTPSession * sess, RTPSource * source)
608
g_object_ref (source);
609
RTP_SESSION_UNLOCK (sess);
610
g_signal_emit (sess, rtp_session_signals[SIGNAL_ON_SSRC_ACTIVE], 0, source);
611
RTP_SESSION_LOCK (sess);
612
g_object_unref (source);
616
on_ssrc_sdes (RTPSession * sess, RTPSource * source)
618
g_object_ref (source);
619
GST_DEBUG ("SDES changed for SSRC %08x", source->ssrc);
620
RTP_SESSION_UNLOCK (sess);
621
g_signal_emit (sess, rtp_session_signals[SIGNAL_ON_SSRC_SDES], 0, source);
622
RTP_SESSION_LOCK (sess);
623
g_object_unref (source);
627
on_bye_ssrc (RTPSession * sess, RTPSource * source)
629
g_object_ref (source);
630
RTP_SESSION_UNLOCK (sess);
631
g_signal_emit (sess, rtp_session_signals[SIGNAL_ON_BYE_SSRC], 0, source);
632
RTP_SESSION_LOCK (sess);
633
g_object_unref (source);
637
on_bye_timeout (RTPSession * sess, RTPSource * source)
639
g_object_ref (source);
640
RTP_SESSION_UNLOCK (sess);
641
g_signal_emit (sess, rtp_session_signals[SIGNAL_ON_BYE_TIMEOUT], 0, source);
642
RTP_SESSION_LOCK (sess);
643
g_object_unref (source);
647
on_timeout (RTPSession * sess, RTPSource * source)
649
g_object_ref (source);
650
RTP_SESSION_UNLOCK (sess);
651
g_signal_emit (sess, rtp_session_signals[SIGNAL_ON_TIMEOUT], 0, source);
652
RTP_SESSION_LOCK (sess);
653
g_object_unref (source);
657
on_sender_timeout (RTPSession * sess, RTPSource * source)
659
g_object_ref (source);
660
RTP_SESSION_UNLOCK (sess);
661
g_signal_emit (sess, rtp_session_signals[SIGNAL_ON_SENDER_TIMEOUT], 0,
663
RTP_SESSION_LOCK (sess);
664
g_object_unref (source);
670
* Create a new session object.
672
* Returns: a new #RTPSession. g_object_unref() after usage.
675
rtp_session_new (void)
679
sess = g_object_new (RTP_TYPE_SESSION, NULL);
685
* rtp_session_set_callbacks:
686
* @sess: an #RTPSession
687
* @callbacks: callbacks to configure
688
* @user_data: user data passed in the callbacks
690
* Configure a set of callbacks to be notified of actions.
693
rtp_session_set_callbacks (RTPSession * sess, RTPSessionCallbacks * callbacks,
696
g_return_if_fail (RTP_IS_SESSION (sess));
698
if (callbacks->process_rtp) {
699
sess->callbacks.process_rtp = callbacks->process_rtp;
700
sess->process_rtp_user_data = user_data;
702
if (callbacks->send_rtp) {
703
sess->callbacks.send_rtp = callbacks->send_rtp;
704
sess->send_rtp_user_data = user_data;
706
if (callbacks->send_rtcp) {
707
sess->callbacks.send_rtcp = callbacks->send_rtcp;
708
sess->send_rtcp_user_data = user_data;
710
if (callbacks->sync_rtcp) {
711
sess->callbacks.sync_rtcp = callbacks->sync_rtcp;
712
sess->sync_rtcp_user_data = user_data;
714
if (callbacks->clock_rate) {
715
sess->callbacks.clock_rate = callbacks->clock_rate;
716
sess->clock_rate_user_data = user_data;
718
if (callbacks->reconsider) {
719
sess->callbacks.reconsider = callbacks->reconsider;
720
sess->reconsider_user_data = user_data;
725
* rtp_session_set_process_rtp_callback:
726
* @sess: an #RTPSession
727
* @callback: callback to set
728
* @user_data: user data passed in the callback
730
* Configure only the process_rtp callback to be notified of the process_rtp action.
733
rtp_session_set_process_rtp_callback (RTPSession * sess,
734
RTPSessionProcessRTP callback, gpointer user_data)
736
g_return_if_fail (RTP_IS_SESSION (sess));
738
sess->callbacks.process_rtp = callback;
739
sess->process_rtp_user_data = user_data;
743
* rtp_session_set_send_rtp_callback:
744
* @sess: an #RTPSession
745
* @callback: callback to set
746
* @user_data: user data passed in the callback
748
* Configure only the send_rtp callback to be notified of the send_rtp action.
751
rtp_session_set_send_rtp_callback (RTPSession * sess,
752
RTPSessionSendRTP callback, gpointer user_data)
754
g_return_if_fail (RTP_IS_SESSION (sess));
756
sess->callbacks.send_rtp = callback;
757
sess->send_rtp_user_data = user_data;
761
* rtp_session_set_send_rtcp_callback:
762
* @sess: an #RTPSession
763
* @callback: callback to set
764
* @user_data: user data passed in the callback
766
* Configure only the send_rtcp callback to be notified of the send_rtcp action.
769
rtp_session_set_send_rtcp_callback (RTPSession * sess,
770
RTPSessionSendRTCP callback, gpointer user_data)
772
g_return_if_fail (RTP_IS_SESSION (sess));
774
sess->callbacks.send_rtcp = callback;
775
sess->send_rtcp_user_data = user_data;
779
* rtp_session_set_sync_rtcp_callback:
780
* @sess: an #RTPSession
781
* @callback: callback to set
782
* @user_data: user data passed in the callback
784
* Configure only the sync_rtcp callback to be notified of the sync_rtcp action.
787
rtp_session_set_sync_rtcp_callback (RTPSession * sess,
788
RTPSessionSyncRTCP callback, gpointer user_data)
790
g_return_if_fail (RTP_IS_SESSION (sess));
792
sess->callbacks.sync_rtcp = callback;
793
sess->sync_rtcp_user_data = user_data;
797
* rtp_session_set_clock_rate_callback:
798
* @sess: an #RTPSession
799
* @callback: callback to set
800
* @user_data: user data passed in the callback
802
* Configure only the clock_rate callback to be notified of the clock_rate action.
805
rtp_session_set_clock_rate_callback (RTPSession * sess,
806
RTPSessionClockRate callback, gpointer user_data)
808
g_return_if_fail (RTP_IS_SESSION (sess));
810
sess->callbacks.clock_rate = callback;
811
sess->clock_rate_user_data = user_data;
815
* rtp_session_set_reconsider_callback:
816
* @sess: an #RTPSession
817
* @callback: callback to set
818
* @user_data: user data passed in the callback
820
* Configure only the reconsider callback to be notified of the reconsider action.
823
rtp_session_set_reconsider_callback (RTPSession * sess,
824
RTPSessionReconsider callback, gpointer user_data)
826
g_return_if_fail (RTP_IS_SESSION (sess));
828
sess->callbacks.reconsider = callback;
829
sess->reconsider_user_data = user_data;
833
* rtp_session_set_bandwidth:
834
* @sess: an #RTPSession
835
* @bandwidth: the bandwidth allocated
837
* Set the session bandwidth in bytes per second.
840
rtp_session_set_bandwidth (RTPSession * sess, gdouble bandwidth)
842
g_return_if_fail (RTP_IS_SESSION (sess));
844
RTP_SESSION_LOCK (sess);
845
sess->stats.bandwidth = bandwidth;
846
RTP_SESSION_UNLOCK (sess);
850
* rtp_session_get_bandwidth:
851
* @sess: an #RTPSession
853
* Get the session bandwidth.
855
* Returns: the session bandwidth.
858
rtp_session_get_bandwidth (RTPSession * sess)
862
g_return_val_if_fail (RTP_IS_SESSION (sess), 0);
864
RTP_SESSION_LOCK (sess);
865
result = sess->stats.bandwidth;
866
RTP_SESSION_UNLOCK (sess);
872
* rtp_session_set_rtcp_fraction:
873
* @sess: an #RTPSession
874
* @bandwidth: the RTCP bandwidth
876
* Set the bandwidth that should be used for RTCP
880
rtp_session_set_rtcp_fraction (RTPSession * sess, gdouble bandwidth)
882
g_return_if_fail (RTP_IS_SESSION (sess));
884
RTP_SESSION_LOCK (sess);
885
sess->stats.rtcp_bandwidth = bandwidth;
886
RTP_SESSION_UNLOCK (sess);
890
* rtp_session_get_rtcp_fraction:
891
* @sess: an #RTPSession
893
* Get the session bandwidth used for RTCP.
895
* Returns: The bandwidth used for RTCP messages.
898
rtp_session_get_rtcp_fraction (RTPSession * sess)
902
g_return_val_if_fail (RTP_IS_SESSION (sess), 0.0);
904
RTP_SESSION_LOCK (sess);
905
result = sess->stats.rtcp_bandwidth;
906
RTP_SESSION_UNLOCK (sess);
912
* rtp_session_set_sdes_string:
913
* @sess: an #RTPSession
914
* @type: the type of the SDES item
915
* @item: a null-terminated string to set.
917
* Store an SDES item of @type in @sess.
919
* Returns: %FALSE if the data was unchanged @type is invalid.
922
rtp_session_set_sdes_string (RTPSession * sess, GstRTCPSDESType type,
927
g_return_val_if_fail (RTP_IS_SESSION (sess), FALSE);
929
RTP_SESSION_LOCK (sess);
930
result = rtp_source_set_sdes_string (sess->source, type, item);
931
RTP_SESSION_UNLOCK (sess);
937
* rtp_session_get_sdes_string:
938
* @sess: an #RTPSession
939
* @type: the type of the SDES item
941
* Get the SDES item of @type from @sess.
943
* Returns: a null-terminated copy of the SDES item or NULL when @type was not
944
* valid. g_free() after usage.
947
rtp_session_get_sdes_string (RTPSession * sess, GstRTCPSDESType type)
951
g_return_val_if_fail (RTP_IS_SESSION (sess), NULL);
953
RTP_SESSION_LOCK (sess);
954
result = rtp_source_get_sdes_string (sess->source, type);
955
RTP_SESSION_UNLOCK (sess);
961
source_push_rtp (RTPSource * source, GstBuffer * buffer, RTPSession * session)
963
GstFlowReturn result = GST_FLOW_OK;
965
if (source == session->source) {
966
GST_LOG ("source %08x pushed sender RTP packet", source->ssrc);
968
RTP_SESSION_UNLOCK (session);
970
if (session->callbacks.send_rtp)
972
session->callbacks.send_rtp (session, source, buffer,
973
session->send_rtp_user_data);
975
gst_buffer_unref (buffer);
978
GST_LOG ("source %08x pushed receiver RTP packet", source->ssrc);
979
RTP_SESSION_UNLOCK (session);
981
if (session->callbacks.process_rtp)
983
session->callbacks.process_rtp (session, source, buffer,
984
session->process_rtp_user_data);
986
gst_buffer_unref (buffer);
988
RTP_SESSION_LOCK (session);
994
source_clock_rate (RTPSource * source, guint8 pt, RTPSession * session)
998
RTP_SESSION_UNLOCK (session);
1000
if (session->callbacks.clock_rate)
1002
session->callbacks.clock_rate (session, pt,
1003
session->clock_rate_user_data);
1007
RTP_SESSION_LOCK (session);
1009
GST_DEBUG ("got clock-rate %d for pt %d", result, pt);
1014
static RTPSourceCallbacks callbacks = {
1015
(RTPSourcePushRTP) source_push_rtp,
1016
(RTPSourceClockRate) source_clock_rate,
1020
* find_add_conflicting_addresses:
1021
* @sess: The session to check in
1022
* @arrival: The arrival stats for the buffer
1024
* Checks if an address which has a conflict is already known,
1025
* otherwise remembers it to prevent loops.
1027
* Returns: TRUE if it was a known conflict, FALSE otherwise
1031
find_add_conflicting_addresses (RTPSession * sess, RTPArrivalStats * arrival)
1034
RTPConflictingAddress *new_conflict;
1036
for (item = g_list_first (sess->conflicting_addresses);
1037
item; item = g_list_next (item)) {
1038
RTPConflictingAddress *known_conflict = item->data;
1040
if (gst_netaddress_equal (&arrival->address, &known_conflict->address)) {
1041
known_conflict->time = arrival->time;
1046
new_conflict = g_new0 (RTPConflictingAddress, 1);
1048
memcpy (&new_conflict->address, &arrival->address, sizeof (GstNetAddress));
1049
new_conflict->time = arrival->time;
1051
sess->conflicting_addresses = g_list_prepend (sess->conflicting_addresses,
1058
check_collision (RTPSession * sess, RTPSource * source,
1059
RTPArrivalStats * arrival, gboolean rtp)
1061
/* If we have no arrival address, we can't do collision checking */
1062
if (!arrival->have_address)
1065
if (sess->source != source) {
1066
/* This is not our local source, but lets check if two remote
1070
if (source->have_rtp_from) {
1071
if (gst_netaddress_equal (&source->rtp_from, &arrival->address))
1072
/* Address is the same */
1075
/* We don't already have a from address for RTP, just set it */
1076
rtp_source_set_rtp_from (source, &arrival->address);
1080
if (source->have_rtcp_from) {
1081
if (gst_netaddress_equal (&source->rtcp_from, &arrival->address))
1082
/* Address is the same */
1085
/* We don't already have a from address for RTCP, just set it */
1086
rtp_source_set_rtcp_from (source, &arrival->address);
1090
/* We received RTP or RTCP from this source before but the network address
1091
* changed. In this case, we have third-party collision or loop */
1092
GST_DEBUG ("we have a third-party collision or loop");
1094
/* FIXME: Log 3rd party collision somehow
1095
* Maybe should be done in upper layer, only the SDES can tell us
1096
* if its a collision or a loop
1099
/* This is sending with our ssrc, is it an address we already know */
1101
if (find_add_conflicting_addresses (sess, arrival)) {
1102
/* Its a known conflict, its probably a loop, not a collision
1103
* lets just drop the incoming packet
1105
GST_DEBUG ("Our packets are being looped back to us, dropping");
1107
/* Its a new collision, lets change our SSRC */
1109
GST_DEBUG ("Collision for SSRC %x", rtp_source_get_ssrc (source));
1110
on_ssrc_collision (sess, source);
1112
rtp_session_schedule_bye_locked (sess, "SSRC Collision", arrival->time);
1114
sess->change_ssrc = TRUE;
1122
/* must be called with the session lock, the returned source needs to be
1123
* unreffed after usage. */
1125
obtain_source (RTPSession * sess, guint32 ssrc, gboolean * created,
1126
RTPArrivalStats * arrival, gboolean rtp)
1131
g_hash_table_lookup (sess->ssrcs[sess->mask_idx], GINT_TO_POINTER (ssrc));
1132
if (source == NULL) {
1133
/* make new Source in probation and insert */
1134
source = rtp_source_new (ssrc);
1136
/* for RTP packets we need to set the source in probation. Receiving RTCP
1137
* packets of an SSRC, on the other hand, is a strong indication that we
1138
* are dealing with a valid source. */
1140
source->probation = RTP_DEFAULT_PROBATION;
1142
source->probation = 0;
1144
/* store from address, if any */
1145
if (arrival->have_address) {
1147
rtp_source_set_rtp_from (source, &arrival->address);
1149
rtp_source_set_rtcp_from (source, &arrival->address);
1152
/* configure a callback on the source */
1153
rtp_source_set_callbacks (source, &callbacks, sess);
1155
g_hash_table_insert (sess->ssrcs[sess->mask_idx], GINT_TO_POINTER (ssrc),
1158
/* we have one more source now */
1159
sess->total_sources++;
1163
/* check for collision, this updates the address when not previously set */
1164
if (check_collision (sess, source, arrival, rtp)) {
1168
/* update last activity */
1169
source->last_activity = arrival->time;
1171
source->last_rtp_activity = arrival->time;
1172
g_object_ref (source);
1178
* rtp_session_get_internal_source:
1179
* @sess: a #RTPSession
1181
* Get the internal #RTPSource of @sess.
1183
* Returns: The internal #RTPSource. g_object_unref() after usage.
1186
rtp_session_get_internal_source (RTPSession * sess)
1190
g_return_val_if_fail (RTP_IS_SESSION (sess), NULL);
1192
result = g_object_ref (sess->source);
1198
* rtp_session_set_internal_ssrc:
1199
* @sess: a #RTPSession
1202
* Set the SSRC of @sess to @ssrc.
1205
rtp_session_set_internal_ssrc (RTPSession * sess, guint32 ssrc)
1207
RTP_SESSION_LOCK (sess);
1208
if (ssrc != sess->source->ssrc) {
1209
g_hash_table_steal (sess->ssrcs[sess->mask_idx],
1210
GINT_TO_POINTER (sess->source->ssrc));
1212
GST_DEBUG ("setting internal SSRC to %08x", ssrc);
1213
/* After this call, any receiver of the old SSRC either in RTP or RTCP
1214
* packets will timeout on the old SSRC, we could potentially schedule a
1215
* BYE RTCP for the old SSRC... */
1216
sess->source->ssrc = ssrc;
1217
rtp_source_reset (sess->source);
1219
/* rehash with the new SSRC */
1220
g_hash_table_insert (sess->ssrcs[sess->mask_idx],
1221
GINT_TO_POINTER (sess->source->ssrc), sess->source);
1223
RTP_SESSION_UNLOCK (sess);
1225
g_object_notify (G_OBJECT (sess), "internal-ssrc");
1229
* rtp_session_get_internal_ssrc:
1230
* @sess: a #RTPSession
1232
* Get the internal SSRC of @sess.
1234
* Returns: The SSRC of the session.
1237
rtp_session_get_internal_ssrc (RTPSession * sess)
1241
RTP_SESSION_LOCK (sess);
1242
ssrc = sess->source->ssrc;
1243
RTP_SESSION_UNLOCK (sess);
1249
* rtp_session_add_source:
1250
* @sess: a #RTPSession
1251
* @src: #RTPSource to add
1253
* Add @src to @session.
1255
* Returns: %TRUE on success, %FALSE if a source with the same SSRC already
1256
* existed in the session.
1259
rtp_session_add_source (RTPSession * sess, RTPSource * src)
1261
gboolean result = FALSE;
1264
g_return_val_if_fail (RTP_IS_SESSION (sess), FALSE);
1265
g_return_val_if_fail (src != NULL, FALSE);
1267
RTP_SESSION_LOCK (sess);
1269
g_hash_table_lookup (sess->ssrcs[sess->mask_idx],
1270
GINT_TO_POINTER (src->ssrc));
1272
g_hash_table_insert (sess->ssrcs[sess->mask_idx],
1273
GINT_TO_POINTER (src->ssrc), src);
1274
/* we have one more source now */
1275
sess->total_sources++;
1278
RTP_SESSION_UNLOCK (sess);
1284
* rtp_session_get_num_sources:
1285
* @sess: an #RTPSession
1287
* Get the number of sources in @sess.
1289
* Returns: The number of sources in @sess.
1292
rtp_session_get_num_sources (RTPSession * sess)
1296
g_return_val_if_fail (RTP_IS_SESSION (sess), FALSE);
1298
RTP_SESSION_LOCK (sess);
1299
result = sess->total_sources;
1300
RTP_SESSION_UNLOCK (sess);
1306
* rtp_session_get_num_active_sources:
1307
* @sess: an #RTPSession
1309
* Get the number of active sources in @sess. A source is considered active when
1310
* it has been validated and has not yet received a BYE RTCP message.
1312
* Returns: The number of active sources in @sess.
1315
rtp_session_get_num_active_sources (RTPSession * sess)
1319
g_return_val_if_fail (RTP_IS_SESSION (sess), 0);
1321
RTP_SESSION_LOCK (sess);
1322
result = sess->stats.active_sources;
1323
RTP_SESSION_UNLOCK (sess);
1329
* rtp_session_get_source_by_ssrc:
1330
* @sess: an #RTPSession
1333
* Find the source with @ssrc in @sess.
1335
* Returns: a #RTPSource with SSRC @ssrc or NULL if the source was not found.
1336
* g_object_unref() after usage.
1339
rtp_session_get_source_by_ssrc (RTPSession * sess, guint32 ssrc)
1343
g_return_val_if_fail (RTP_IS_SESSION (sess), NULL);
1345
RTP_SESSION_LOCK (sess);
1347
g_hash_table_lookup (sess->ssrcs[sess->mask_idx], GINT_TO_POINTER (ssrc));
1349
g_object_ref (result);
1350
RTP_SESSION_UNLOCK (sess);
1356
* rtp_session_get_source_by_cname:
1357
* @sess: a #RTPSession
1360
* Find the source with @cname in @sess.
1362
* Returns: a #RTPSource with CNAME @cname or NULL if the source was not found.
1363
* g_object_unref() after usage.
1366
rtp_session_get_source_by_cname (RTPSession * sess, const gchar * cname)
1370
g_return_val_if_fail (RTP_IS_SESSION (sess), NULL);
1371
g_return_val_if_fail (cname != NULL, NULL);
1373
RTP_SESSION_LOCK (sess);
1374
result = g_hash_table_lookup (sess->cnames, cname);
1376
g_object_ref (result);
1377
RTP_SESSION_UNLOCK (sess);
1383
rtp_session_create_new_ssrc (RTPSession * sess)
1388
ssrc = g_random_int ();
1390
/* see if it exists in the session, we're done if it doesn't */
1391
if (g_hash_table_lookup (sess->ssrcs[sess->mask_idx],
1392
GINT_TO_POINTER (ssrc)) == NULL)
1401
* rtp_session_create_source:
1402
* @sess: an #RTPSession
1404
* Create an #RTPSource for use in @sess. This function will create a source
1405
* with an ssrc that is currently not used by any participants in the session.
1407
* Returns: an #RTPSource.
1410
rtp_session_create_source (RTPSession * sess)
1415
RTP_SESSION_LOCK (sess);
1416
ssrc = rtp_session_create_new_ssrc (sess);
1417
source = rtp_source_new (ssrc);
1418
rtp_source_set_callbacks (source, &callbacks, sess);
1419
/* we need an additional ref for the source in the hashtable */
1420
g_object_ref (source);
1421
g_hash_table_insert (sess->ssrcs[sess->mask_idx], GINT_TO_POINTER (ssrc),
1423
/* we have one more source now */
1424
sess->total_sources++;
1425
RTP_SESSION_UNLOCK (sess);
1430
/* update the RTPArrivalStats structure with the current time and other bits
1431
* about the current buffer we are handling.
1432
* This function is typically called when a validated packet is received.
1433
* This function should be called with the SESSION_LOCK
1436
update_arrival_stats (RTPSession * sess, RTPArrivalStats * arrival,
1437
gboolean rtp, GstBuffer * buffer, GstClockTime current_time,
1438
GstClockTime running_time, guint64 ntpnstime)
1440
/* get time of arrival */
1441
arrival->time = current_time;
1442
arrival->running_time = running_time;
1443
arrival->ntpnstime = ntpnstime;
1445
/* get packet size including header overhead */
1446
arrival->bytes = GST_BUFFER_SIZE (buffer) + sess->header_len;
1449
arrival->payload_len = gst_rtp_buffer_get_payload_len (buffer);
1451
arrival->payload_len = 0;
1454
/* for netbuffer we can store the IP address to check for collisions */
1455
arrival->have_address = GST_IS_NETBUFFER (buffer);
1456
if (arrival->have_address) {
1457
GstNetBuffer *netbuf = (GstNetBuffer *) buffer;
1459
memcpy (&arrival->address, &netbuf->from, sizeof (GstNetAddress));
1464
* rtp_session_process_rtp:
1465
* @sess: and #RTPSession
1466
* @buffer: an RTP buffer
1467
* @current_time: the current system time
1468
* @ntpnstime: the NTP arrival time in nanoseconds
1470
* Process an RTP buffer in the session manager. This function takes ownership
1473
* Returns: a #GstFlowReturn.
1476
rtp_session_process_rtp (RTPSession * sess, GstBuffer * buffer,
1477
GstClockTime current_time, GstClockTime running_time, guint64 ntpnstime)
1479
GstFlowReturn result;
1483
gboolean prevsender, prevactive;
1484
RTPArrivalStats arrival;
1486
g_return_val_if_fail (RTP_IS_SESSION (sess), GST_FLOW_ERROR);
1487
g_return_val_if_fail (GST_IS_BUFFER (buffer), GST_FLOW_ERROR);
1489
if (!gst_rtp_buffer_validate (buffer))
1490
goto invalid_packet;
1492
RTP_SESSION_LOCK (sess);
1493
/* update arrival stats */
1494
update_arrival_stats (sess, &arrival, TRUE, buffer, current_time,
1495
running_time, ntpnstime);
1497
/* ignore more RTP packets when we left the session */
1498
if (sess->source->received_bye)
1501
/* get SSRC and look up in session database */
1502
ssrc = gst_rtp_buffer_get_ssrc (buffer);
1503
source = obtain_source (sess, ssrc, &created, &arrival, TRUE);
1507
prevsender = RTP_SOURCE_IS_SENDER (source);
1508
prevactive = RTP_SOURCE_IS_ACTIVE (source);
1510
/* we need to ref so that we can process the CSRCs later */
1511
gst_buffer_ref (buffer);
1513
/* let source process the packet */
1514
result = rtp_source_process_rtp (source, buffer, &arrival);
1516
/* source became active */
1517
if (prevactive != RTP_SOURCE_IS_ACTIVE (source)) {
1518
sess->stats.active_sources++;
1519
GST_DEBUG ("source: %08x became active, %d active sources", ssrc,
1520
sess->stats.active_sources);
1521
on_ssrc_validated (sess, source);
1523
if (prevsender != RTP_SOURCE_IS_SENDER (source)) {
1524
sess->stats.sender_sources++;
1525
GST_DEBUG ("source: %08x became sender, %d sender sources", ssrc,
1526
sess->stats.sender_sources);
1530
on_new_ssrc (sess, source);
1532
if (source->validated) {
1536
/* for validated sources, we add the CSRCs as well */
1537
count = gst_rtp_buffer_get_csrc_count (buffer);
1539
for (i = 0; i < count; i++) {
1541
RTPSource *csrc_src;
1543
csrc = gst_rtp_buffer_get_csrc (buffer, i);
1546
csrc_src = obtain_source (sess, csrc, &created, &arrival, TRUE);
1551
GST_DEBUG ("created new CSRC: %08x", csrc);
1552
rtp_source_set_as_csrc (csrc_src);
1553
if (RTP_SOURCE_IS_ACTIVE (csrc_src))
1554
sess->stats.active_sources++;
1555
on_new_ssrc (sess, csrc_src);
1557
g_object_unref (csrc_src);
1560
g_object_unref (source);
1561
gst_buffer_unref (buffer);
1563
RTP_SESSION_UNLOCK (sess);
1570
gst_buffer_unref (buffer);
1571
GST_DEBUG ("invalid RTP packet received");
1576
gst_buffer_unref (buffer);
1577
RTP_SESSION_UNLOCK (sess);
1578
GST_DEBUG ("ignoring RTP packet because we are leaving");
1583
gst_buffer_unref (buffer);
1584
RTP_SESSION_UNLOCK (sess);
1585
GST_DEBUG ("ignoring packet because its collisioning");
1591
rtp_session_process_rb (RTPSession * sess, RTPSource * source,
1592
GstRTCPPacket * packet, RTPArrivalStats * arrival)
1596
count = gst_rtcp_packet_get_rb_count (packet);
1597
for (i = 0; i < count; i++) {
1598
guint32 ssrc, exthighestseq, jitter, lsr, dlsr;
1599
guint8 fractionlost;
1602
gst_rtcp_packet_get_rb (packet, i, &ssrc, &fractionlost,
1603
&packetslost, &exthighestseq, &jitter, &lsr, &dlsr);
1605
GST_DEBUG ("RB %d: SSRC %08x, jitter %" G_GUINT32_FORMAT, i, ssrc, jitter);
1607
if (ssrc == sess->source->ssrc) {
1608
/* only deal with report blocks for our session, we update the stats of
1609
* the sender of the RTCP message. We could also compare our stats against
1610
* the other sender to see if we are better or worse. */
1611
rtp_source_process_rb (source, arrival->time, fractionlost, packetslost,
1612
exthighestseq, jitter, lsr, dlsr);
1614
on_ssrc_active (sess, source);
1619
/* A Sender report contains statistics about how the sender is doing. This
1620
* includes timing informataion such as the relation between RTP and NTP
1621
* timestamps and the number of packets/bytes it sent to us.
1623
* In this report is also included a set of report blocks related to how this
1624
* sender is receiving data (in case we (or somebody else) is also sending stuff
1625
* to it). This info includes the packet loss, jitter and seqnum. It also
1626
* contains information to calculate the round trip time (LSR/DLSR).
1629
rtp_session_process_sr (RTPSession * sess, GstRTCPPacket * packet,
1630
RTPArrivalStats * arrival)
1632
guint32 senderssrc, rtptime, packet_count, octet_count;
1635
gboolean created, prevsender;
1637
gst_rtcp_packet_sr_get_sender_info (packet, &senderssrc, &ntptime, &rtptime,
1638
&packet_count, &octet_count);
1640
GST_DEBUG ("got SR packet: SSRC %08x, time %" GST_TIME_FORMAT,
1641
senderssrc, GST_TIME_ARGS (arrival->time));
1643
source = obtain_source (sess, senderssrc, &created, arrival, FALSE);
1647
prevsender = RTP_SOURCE_IS_SENDER (source);
1649
/* first update the source */
1650
rtp_source_process_sr (source, arrival->time, ntptime, rtptime, packet_count,
1653
if (prevsender != RTP_SOURCE_IS_SENDER (source)) {
1654
sess->stats.sender_sources++;
1655
GST_DEBUG ("source: %08x became sender, %d sender sources", senderssrc,
1656
sess->stats.sender_sources);
1660
on_new_ssrc (sess, source);
1662
rtp_session_process_rb (sess, source, packet, arrival);
1663
g_object_unref (source);
1666
/* A receiver report contains statistics about how a receiver is doing. It
1667
* includes stuff like packet loss, jitter and the seqnum it received last. It
1668
* also contains info to calculate the round trip time.
1670
* We are only interested in how the sender of this report is doing wrt to us.
1673
rtp_session_process_rr (RTPSession * sess, GstRTCPPacket * packet,
1674
RTPArrivalStats * arrival)
1680
senderssrc = gst_rtcp_packet_rr_get_ssrc (packet);
1682
GST_DEBUG ("got RR packet: SSRC %08x", senderssrc);
1684
source = obtain_source (sess, senderssrc, &created, arrival, FALSE);
1689
on_new_ssrc (sess, source);
1691
rtp_session_process_rb (sess, source, packet, arrival);
1692
g_object_unref (source);
1695
/* Get SDES items and store them in the SSRC */
1697
rtp_session_process_sdes (RTPSession * sess, GstRTCPPacket * packet,
1698
RTPArrivalStats * arrival)
1701
gboolean more_items, more_entries;
1703
items = gst_rtcp_packet_sdes_get_item_count (packet);
1704
GST_DEBUG ("got SDES packet with %d items", items);
1706
more_items = gst_rtcp_packet_sdes_first_item (packet);
1708
while (more_items) {
1710
gboolean changed, created;
1713
ssrc = gst_rtcp_packet_sdes_get_ssrc (packet);
1715
GST_DEBUG ("item %d, SSRC %08x", i, ssrc);
1719
/* find src, no probation when dealing with RTCP */
1720
source = obtain_source (sess, ssrc, &created, arrival, FALSE);
1724
more_entries = gst_rtcp_packet_sdes_first_entry (packet);
1726
while (more_entries) {
1727
GstRTCPSDESType type;
1731
gst_rtcp_packet_sdes_get_entry (packet, &type, &len, &data);
1733
GST_DEBUG ("entry %d, type %d, len %d, data %.*s", j, type, len, len,
1736
changed |= rtp_source_set_sdes (source, type, data, len);
1738
more_entries = gst_rtcp_packet_sdes_next_entry (packet);
1742
source->validated = TRUE;
1745
on_new_ssrc (sess, source);
1747
on_ssrc_sdes (sess, source);
1749
g_object_unref (source);
1751
more_items = gst_rtcp_packet_sdes_next_item (packet);
1756
/* BYE is sent when a client leaves the session
1759
rtp_session_process_bye (RTPSession * sess, GstRTCPPacket * packet,
1760
RTPArrivalStats * arrival)
1765
reason = gst_rtcp_packet_bye_get_reason (packet);
1766
GST_DEBUG ("got BYE packet (reason: %s)", GST_STR_NULL (reason));
1768
count = gst_rtcp_packet_bye_get_ssrc_count (packet);
1769
for (i = 0; i < count; i++) {
1772
gboolean created, prevactive, prevsender;
1773
guint pmembers, members;
1775
ssrc = gst_rtcp_packet_bye_get_nth_ssrc (packet, i);
1776
GST_DEBUG ("SSRC: %08x", ssrc);
1778
/* find src and mark bye, no probation when dealing with RTCP */
1779
source = obtain_source (sess, ssrc, &created, arrival, FALSE);
1783
/* store time for when we need to time out this source */
1784
source->bye_time = arrival->time;
1786
prevactive = RTP_SOURCE_IS_ACTIVE (source);
1787
prevsender = RTP_SOURCE_IS_SENDER (source);
1789
/* let the source handle the rest */
1790
rtp_source_process_bye (source, reason);
1792
pmembers = sess->stats.active_sources;
1794
if (prevactive && !RTP_SOURCE_IS_ACTIVE (source)) {
1795
sess->stats.active_sources--;
1796
GST_DEBUG ("source: %08x became inactive, %d active sources", ssrc,
1797
sess->stats.active_sources);
1799
if (prevsender && !RTP_SOURCE_IS_SENDER (source)) {
1800
sess->stats.sender_sources--;
1801
GST_DEBUG ("source: %08x became non sender, %d sender sources", ssrc,
1802
sess->stats.sender_sources);
1804
members = sess->stats.active_sources;
1806
if (!sess->source->received_bye && members < pmembers) {
1807
/* some members went away since the previous timeout estimate.
1808
* Perform reverse reconsideration but only when we are not scheduling a
1810
if (arrival->time < sess->next_rtcp_check_time) {
1811
GstClockTime time_remaining;
1813
time_remaining = sess->next_rtcp_check_time - arrival->time;
1814
sess->next_rtcp_check_time =
1815
gst_util_uint64_scale (time_remaining, members, pmembers);
1817
GST_DEBUG ("reverse reconsideration %" GST_TIME_FORMAT,
1818
GST_TIME_ARGS (sess->next_rtcp_check_time));
1820
sess->next_rtcp_check_time += arrival->time;
1822
RTP_SESSION_UNLOCK (sess);
1823
/* notify app of reconsideration */
1824
if (sess->callbacks.reconsider)
1825
sess->callbacks.reconsider (sess, sess->reconsider_user_data);
1826
RTP_SESSION_LOCK (sess);
1831
on_new_ssrc (sess, source);
1833
on_bye_ssrc (sess, source);
1835
g_object_unref (source);
1841
rtp_session_process_app (RTPSession * sess, GstRTCPPacket * packet,
1842
RTPArrivalStats * arrival)
1844
GST_DEBUG ("received APP");
1848
* rtp_session_process_rtcp:
1849
* @sess: and #RTPSession
1850
* @buffer: an RTCP buffer
1851
* @current_time: the current system time
1853
* Process an RTCP buffer in the session manager. This function takes ownership
1856
* Returns: a #GstFlowReturn.
1859
rtp_session_process_rtcp (RTPSession * sess, GstBuffer * buffer,
1860
GstClockTime current_time)
1862
GstRTCPPacket packet;
1863
gboolean more, is_bye = FALSE, is_sr = FALSE;
1864
RTPArrivalStats arrival;
1865
GstFlowReturn result = GST_FLOW_OK;
1867
g_return_val_if_fail (RTP_IS_SESSION (sess), GST_FLOW_ERROR);
1868
g_return_val_if_fail (GST_IS_BUFFER (buffer), GST_FLOW_ERROR);
1870
if (!gst_rtcp_buffer_validate (buffer))
1871
goto invalid_packet;
1873
GST_DEBUG ("received RTCP packet");
1875
RTP_SESSION_LOCK (sess);
1876
/* update arrival stats */
1877
update_arrival_stats (sess, &arrival, FALSE, buffer, current_time, -1, -1);
1882
/* make writable, we might want to change the buffer */
1883
buffer = gst_buffer_make_metadata_writable (buffer);
1885
/* start processing the compound packet */
1886
more = gst_rtcp_buffer_get_first_packet (buffer, &packet);
1890
type = gst_rtcp_packet_get_type (&packet);
1892
/* when we are leaving the session, we should ignore all non-BYE messages */
1893
if (sess->source->received_bye && type != GST_RTCP_TYPE_BYE) {
1894
GST_DEBUG ("ignoring non-BYE RTCP packet because we are leaving");
1899
case GST_RTCP_TYPE_SR:
1900
rtp_session_process_sr (sess, &packet, &arrival);
1903
case GST_RTCP_TYPE_RR:
1904
rtp_session_process_rr (sess, &packet, &arrival);
1906
case GST_RTCP_TYPE_SDES:
1907
rtp_session_process_sdes (sess, &packet, &arrival);
1909
case GST_RTCP_TYPE_BYE:
1911
rtp_session_process_bye (sess, &packet, &arrival);
1913
case GST_RTCP_TYPE_APP:
1914
rtp_session_process_app (sess, &packet, &arrival);
1917
GST_WARNING ("got unknown RTCP packet");
1921
more = gst_rtcp_packet_move_to_next (&packet);
1924
/* if we are scheduling a BYE, we only want to count bye packets, else we
1925
* count everything */
1926
if (sess->source->received_bye) {
1928
sess->stats.bye_members++;
1929
UPDATE_AVG (sess->stats.avg_rtcp_packet_size, arrival.bytes);
1932
/* keep track of average packet size */
1933
UPDATE_AVG (sess->stats.avg_rtcp_packet_size, arrival.bytes);
1935
RTP_SESSION_UNLOCK (sess);
1937
/* notify caller of sr packets in the callback */
1938
if (is_sr && sess->callbacks.sync_rtcp)
1939
result = sess->callbacks.sync_rtcp (sess, sess->source, buffer,
1940
sess->sync_rtcp_user_data);
1942
gst_buffer_unref (buffer);
1949
GST_DEBUG ("invalid RTCP packet received");
1950
gst_buffer_unref (buffer);
1955
gst_buffer_unref (buffer);
1956
RTP_SESSION_UNLOCK (sess);
1957
GST_DEBUG ("ignoring RTP packet because we left");
1963
* rtp_session_send_rtp:
1964
* @sess: an #RTPSession
1965
* @buffer: an RTP buffer
1966
* @current_time: the current system time
1967
* @ntpnstime: the NTP time in nanoseconds of when this buffer was captured.
1968
* This is the buffer timestamp converted to NTP time.
1970
* Send the RTP buffer in the session manager. This function takes ownership of
1973
* Returns: a #GstFlowReturn.
1976
rtp_session_send_rtp (RTPSession * sess, GstBuffer * buffer,
1977
GstClockTime current_time, guint64 ntpnstime)
1979
GstFlowReturn result;
1981
gboolean prevsender;
1983
g_return_val_if_fail (RTP_IS_SESSION (sess), GST_FLOW_ERROR);
1984
g_return_val_if_fail (GST_IS_BUFFER (buffer), GST_FLOW_ERROR);
1986
if (!gst_rtp_buffer_validate (buffer))
1987
goto invalid_packet;
1989
GST_LOG ("received RTP packet for sending");
1991
RTP_SESSION_LOCK (sess);
1992
source = sess->source;
1994
/* update last activity */
1995
source->last_rtp_activity = current_time;
1997
prevsender = RTP_SOURCE_IS_SENDER (source);
1999
/* we use our own source to send */
2000
result = rtp_source_send_rtp (source, buffer, ntpnstime);
2002
if (RTP_SOURCE_IS_SENDER (source) && !prevsender)
2003
sess->stats.sender_sources++;
2004
RTP_SESSION_UNLOCK (sess);
2011
gst_buffer_unref (buffer);
2012
GST_DEBUG ("invalid RTP packet received");
2018
calculate_rtcp_interval (RTPSession * sess, gboolean deterministic,
2021
GstClockTime result;
2023
if (sess->source->received_bye) {
2024
result = rtp_stats_calculate_bye_interval (&sess->stats);
2026
result = rtp_stats_calculate_rtcp_interval (&sess->stats,
2027
RTP_SOURCE_IS_SENDER (sess->source), first);
2030
GST_DEBUG ("next deterministic interval: %" GST_TIME_FORMAT ", first %d",
2031
GST_TIME_ARGS (result), first);
2034
result = rtp_stats_add_rtcp_jitter (&sess->stats, result);
2036
GST_DEBUG ("next interval: %" GST_TIME_FORMAT, GST_TIME_ARGS (result));
2041
/* Stop the current @sess and schedule a BYE message for the other members.
2042
* One must have the session lock to call this function
2044
static GstFlowReturn
2045
rtp_session_schedule_bye_locked (RTPSession * sess, const gchar * reason,
2046
GstClockTime current_time)
2048
GstFlowReturn result = GST_FLOW_OK;
2050
GstClockTime interval;
2052
g_return_val_if_fail (RTP_IS_SESSION (sess), GST_FLOW_ERROR);
2054
source = sess->source;
2056
/* ignore more BYEs */
2057
if (source->received_bye)
2060
/* we have BYE now */
2061
source->received_bye = TRUE;
2062
/* at least one member wants to send a BYE */
2063
g_free (sess->bye_reason);
2064
sess->bye_reason = g_strdup (reason);
2065
sess->stats.avg_rtcp_packet_size = 100;
2066
sess->stats.bye_members = 1;
2067
sess->first_rtcp = TRUE;
2068
sess->sent_bye = FALSE;
2070
/* reschedule transmission */
2071
sess->last_rtcp_send_time = current_time;
2072
interval = calculate_rtcp_interval (sess, FALSE, TRUE);
2073
sess->next_rtcp_check_time = current_time + interval;
2075
GST_DEBUG ("Schedule BYE for %" GST_TIME_FORMAT ", %" GST_TIME_FORMAT,
2076
GST_TIME_ARGS (interval), GST_TIME_ARGS (sess->next_rtcp_check_time));
2078
RTP_SESSION_UNLOCK (sess);
2079
/* notify app of reconsideration */
2080
if (sess->callbacks.reconsider)
2081
sess->callbacks.reconsider (sess, sess->reconsider_user_data);
2082
RTP_SESSION_LOCK (sess);
2089
* rtp_session_schedule_bye:
2090
* @sess: an #RTPSession
2091
* @reason: a reason or NULL
2092
* @current_time: the current system time
2094
* Stop the current @sess and schedule a BYE message for the other members.
2096
* Returns: a #GstFlowReturn.
2099
rtp_session_schedule_bye (RTPSession * sess, const gchar * reason,
2100
GstClockTime current_time)
2102
GstFlowReturn result = GST_FLOW_OK;
2104
g_return_val_if_fail (RTP_IS_SESSION (sess), GST_FLOW_ERROR);
2106
RTP_SESSION_LOCK (sess);
2107
result = rtp_session_schedule_bye_locked (sess, reason, current_time);
2108
RTP_SESSION_UNLOCK (sess);
2114
* rtp_session_next_timeout:
2115
* @sess: an #RTPSession
2116
* @current_time: the current system time
2118
* Get the next time we should perform session maintenance tasks.
2120
* Returns: a time when rtp_session_on_timeout() should be called with the
2121
* current system time.
2124
rtp_session_next_timeout (RTPSession * sess, GstClockTime current_time)
2126
GstClockTime result;
2128
g_return_val_if_fail (RTP_IS_SESSION (sess), GST_FLOW_ERROR);
2130
RTP_SESSION_LOCK (sess);
2132
result = sess->next_rtcp_check_time;
2134
GST_DEBUG ("current time: %" GST_TIME_FORMAT ", next :%" GST_TIME_FORMAT,
2135
GST_TIME_ARGS (current_time), GST_TIME_ARGS (result));
2137
if (result < current_time) {
2138
GST_DEBUG ("take current time as base");
2139
/* our previous check time expired, start counting from the current time
2141
result = current_time;
2144
if (sess->source->received_bye) {
2145
if (sess->sent_bye) {
2146
GST_DEBUG ("we sent BYE already");
2147
result = GST_CLOCK_TIME_NONE;
2148
} else if (sess->stats.active_sources >= 50) {
2149
GST_DEBUG ("reconsider BYE, more than 50 sources");
2150
/* reconsider BYE if members >= 50 */
2151
result += calculate_rtcp_interval (sess, FALSE, TRUE);
2154
if (sess->first_rtcp) {
2155
GST_DEBUG ("first RTCP packet");
2156
/* we are called for the first time */
2157
result += calculate_rtcp_interval (sess, FALSE, TRUE);
2158
} else if (sess->next_rtcp_check_time < current_time) {
2159
GST_DEBUG ("old check time expired, getting new timeout");
2160
/* get a new timeout when we need to */
2161
result += calculate_rtcp_interval (sess, FALSE, FALSE);
2164
sess->next_rtcp_check_time = result;
2166
GST_DEBUG ("next timeout: %" GST_TIME_FORMAT, GST_TIME_ARGS (result));
2167
RTP_SESSION_UNLOCK (sess);
2176
GstClockTime current_time;
2178
GstClockTime interval;
2179
GstRTCPPacket packet;
2185
session_start_rtcp (RTPSession * sess, ReportData * data)
2187
GstRTCPPacket *packet = &data->packet;
2188
RTPSource *own = sess->source;
2190
data->rtcp = gst_rtcp_buffer_new (sess->mtu);
2192
if (RTP_SOURCE_IS_SENDER (own)) {
2195
guint32 packet_count, octet_count;
2197
/* we are a sender, create SR */
2198
GST_DEBUG ("create SR for SSRC %08x", own->ssrc);
2199
gst_rtcp_buffer_add_packet (data->rtcp, GST_RTCP_TYPE_SR, packet);
2201
/* get latest stats */
2202
rtp_source_get_new_sr (own, data->ntpnstime, &ntptime, &rtptime,
2203
&packet_count, &octet_count);
2205
rtp_source_process_sr (own, data->current_time, ntptime, rtptime,
2206
packet_count, octet_count);
2208
/* fill in sender report info */
2209
gst_rtcp_packet_sr_set_sender_info (packet, own->ssrc,
2210
ntptime, rtptime, packet_count, octet_count);
2212
/* we are only receiver, create RR */
2213
GST_DEBUG ("create RR for SSRC %08x", own->ssrc);
2214
gst_rtcp_buffer_add_packet (data->rtcp, GST_RTCP_TYPE_RR, packet);
2215
gst_rtcp_packet_rr_set_ssrc (packet, own->ssrc);
2219
/* construct a Sender or Receiver Report */
2221
session_report_blocks (const gchar * key, RTPSource * source, ReportData * data)
2223
RTPSession *sess = data->sess;
2224
GstRTCPPacket *packet = &data->packet;
2226
/* create a new buffer if needed */
2227
if (data->rtcp == NULL) {
2228
session_start_rtcp (sess, data);
2230
if (gst_rtcp_packet_get_rb_count (packet) < GST_RTCP_MAX_RB_COUNT) {
2231
/* only report about other sender sources */
2232
if (source != sess->source && RTP_SOURCE_IS_SENDER (source)) {
2233
guint8 fractionlost;
2235
guint32 exthighestseq, jitter;
2239
rtp_source_get_new_rb (source, data->current_time, &fractionlost,
2240
&packetslost, &exthighestseq, &jitter, &lsr, &dlsr);
2242
/* packet is not yet filled, add report block for this source. */
2243
gst_rtcp_packet_add_rb (packet, source->ssrc, fractionlost, packetslost,
2244
exthighestseq, jitter, lsr, dlsr);
2249
/* perform cleanup of sources that timed out */
2251
session_cleanup (const gchar * key, RTPSource * source, ReportData * data)
2253
gboolean remove = FALSE;
2254
gboolean byetimeout = FALSE;
2255
gboolean sendertimeout = FALSE;
2256
gboolean is_sender, is_active;
2257
RTPSession *sess = data->sess;
2258
GstClockTime interval;
2260
is_sender = RTP_SOURCE_IS_SENDER (source);
2261
is_active = RTP_SOURCE_IS_ACTIVE (source);
2263
/* check for our own source, we don't want to delete our own source. */
2264
if (!(source == sess->source)) {
2265
if (source->received_bye) {
2266
/* if we received a BYE from the source, remove the source after some
2268
if (data->current_time > source->bye_time &&
2269
data->current_time - source->bye_time > sess->stats.bye_timeout) {
2270
GST_DEBUG ("removing BYE source %08x", source->ssrc);
2275
/* sources that were inactive for more than 5 times the deterministic reporting
2276
* interval get timed out. the min timeout is 5 seconds. */
2277
if (data->current_time > source->last_activity) {
2278
interval = MAX (data->interval * 5, 5 * GST_SECOND);
2279
if (data->current_time - source->last_activity > interval) {
2280
GST_DEBUG ("removing timeout source %08x, last %" GST_TIME_FORMAT,
2281
source->ssrc, GST_TIME_ARGS (source->last_activity));
2287
/* senders that did not send for a long time become a receiver, this also
2288
* holds for our own source. */
2290
if (data->current_time > source->last_rtp_activity) {
2291
interval = MAX (data->interval * 2, 5 * GST_SECOND);
2292
if (data->current_time - source->last_rtp_activity > interval) {
2293
GST_DEBUG ("sender source %08x timed out and became receiver, last %"
2294
GST_TIME_FORMAT, source->ssrc,
2295
GST_TIME_ARGS (source->last_rtp_activity));
2296
source->is_sender = FALSE;
2297
sess->stats.sender_sources--;
2298
sendertimeout = TRUE;
2304
sess->total_sources--;
2306
sess->stats.sender_sources--;
2308
sess->stats.active_sources--;
2311
on_bye_timeout (sess, source);
2313
on_timeout (sess, source);
2316
on_sender_timeout (sess, source);
2322
session_sdes (RTPSession * sess, ReportData * data)
2324
GstRTCPPacket *packet = &data->packet;
2328
/* add SDES packet */
2329
gst_rtcp_buffer_add_packet (data->rtcp, GST_RTCP_TYPE_SDES, packet);
2331
gst_rtcp_packet_sdes_add_item (packet, sess->source->ssrc);
2333
rtp_source_get_sdes (sess->source, GST_RTCP_SDES_CNAME, &sdes_data,
2335
gst_rtcp_packet_sdes_add_entry (packet, GST_RTCP_SDES_CNAME, sdes_len,
2338
/* other SDES items must only be added at regular intervals and only when the
2339
* user requests to since it might be a privacy problem */
2341
gst_rtcp_packet_sdes_add_entry (&packet, GST_RTCP_SDES_NAME,
2342
strlen (sess->name), (guint8 *) sess->name);
2343
gst_rtcp_packet_sdes_add_entry (&packet, GST_RTCP_SDES_TOOL,
2344
strlen (sess->tool), (guint8 *) sess->tool);
2347
data->has_sdes = TRUE;
2350
/* schedule a BYE packet */
2352
session_bye (RTPSession * sess, ReportData * data)
2354
GstRTCPPacket *packet = &data->packet;
2357
session_start_rtcp (sess, data);
2360
session_sdes (sess, data);
2362
/* add a BYE packet */
2363
gst_rtcp_buffer_add_packet (data->rtcp, GST_RTCP_TYPE_BYE, packet);
2364
gst_rtcp_packet_bye_add_ssrc (packet, sess->source->ssrc);
2365
if (sess->bye_reason)
2366
gst_rtcp_packet_bye_set_reason (packet, sess->bye_reason);
2368
/* we have a BYE packet now */
2369
data->is_bye = TRUE;
2373
is_rtcp_time (RTPSession * sess, GstClockTime current_time, ReportData * data)
2375
GstClockTime new_send_time, elapsed;
2378
/* no need to check yet */
2379
if (sess->next_rtcp_check_time > current_time) {
2380
GST_DEBUG ("no check time yet, next %" GST_TIME_FORMAT " > now %"
2381
GST_TIME_FORMAT, GST_TIME_ARGS (sess->next_rtcp_check_time),
2382
GST_TIME_ARGS (current_time));
2386
/* get elapsed time since we last reported */
2387
elapsed = current_time - sess->last_rtcp_send_time;
2389
/* perform forward reconsideration */
2390
new_send_time = rtp_stats_add_rtcp_jitter (&sess->stats, data->interval);
2392
GST_DEBUG ("forward reconsideration %" GST_TIME_FORMAT ", elapsed %"
2393
GST_TIME_FORMAT, GST_TIME_ARGS (new_send_time), GST_TIME_ARGS (elapsed));
2395
new_send_time += sess->last_rtcp_send_time;
2397
/* check if reconsideration */
2398
if (current_time < new_send_time) {
2399
GST_DEBUG ("reconsider RTCP for %" GST_TIME_FORMAT,
2400
GST_TIME_ARGS (new_send_time));
2402
/* store new check time */
2403
sess->next_rtcp_check_time = new_send_time;
2406
new_send_time = calculate_rtcp_interval (sess, FALSE, FALSE);
2408
GST_DEBUG ("can send RTCP now, next interval %" GST_TIME_FORMAT,
2409
GST_TIME_ARGS (new_send_time));
2410
sess->next_rtcp_check_time = current_time + new_send_time;
2416
* rtp_session_on_timeout:
2417
* @sess: an #RTPSession
2418
* @current_time: the current system time
2419
* @ntpnstime: the current NTP time in nanoseconds
2421
* Perform maintenance actions after the timeout obtained with
2422
* rtp_session_next_timeout() expired.
2424
* This function will perform timeouts of receivers and senders, send a BYE
2425
* packet or generate RTCP packets with current session stats.
2427
* This function can call the #RTPSessionSendRTCP callback, possibly multiple
2428
* times, for each packet that should be processed.
2430
* Returns: a #GstFlowReturn.
2433
rtp_session_on_timeout (RTPSession * sess, GstClockTime current_time,
2436
GstFlowReturn result = GST_FLOW_OK;
2440
gboolean notify = FALSE;
2442
g_return_val_if_fail (RTP_IS_SESSION (sess), GST_FLOW_ERROR);
2444
GST_DEBUG ("reporting at %" GST_TIME_FORMAT ", NTP time %" GST_TIME_FORMAT,
2445
GST_TIME_ARGS (current_time), GST_TIME_ARGS (ntpnstime));
2449
data.current_time = current_time;
2450
data.ntpnstime = ntpnstime;
2451
data.is_bye = FALSE;
2452
data.has_sdes = FALSE;
2456
RTP_SESSION_LOCK (sess);
2457
/* get a new interval, we need this for various cleanups etc */
2458
data.interval = calculate_rtcp_interval (sess, TRUE, sess->first_rtcp);
2460
/* first perform cleanups */
2461
g_hash_table_foreach_remove (sess->ssrcs[sess->mask_idx],
2462
(GHRFunc) session_cleanup, &data);
2464
/* see if we need to generate SR or RR packets */
2465
if (is_rtcp_time (sess, current_time, &data)) {
2466
if (own->received_bye) {
2467
/* generate BYE instead */
2468
GST_DEBUG ("generating BYE message");
2469
session_bye (sess, &data);
2470
sess->sent_bye = TRUE;
2472
/* loop over all known sources and do something */
2473
g_hash_table_foreach (sess->ssrcs[sess->mask_idx],
2474
(GHFunc) session_report_blocks, &data);
2481
/* we keep track of the last report time in order to timeout inactive
2482
* receivers or senders */
2483
sess->last_rtcp_send_time = data.current_time;
2484
sess->first_rtcp = FALSE;
2486
/* add SDES for this source when not already added */
2488
session_sdes (sess, &data);
2490
/* update average RTCP size before sending */
2491
size = GST_BUFFER_SIZE (data.rtcp) + sess->header_len;
2492
UPDATE_AVG (sess->stats.avg_rtcp_packet_size, size);
2495
/* check for outdated collisions */
2496
GST_DEBUG ("checking collision list");
2497
item = g_list_first (sess->conflicting_addresses);
2499
RTPConflictingAddress *known_conflict = item->data;
2500
GList *next_item = g_list_next (item);
2502
if (known_conflict->time < current_time - (data.interval *
2503
RTCP_INTERVAL_COLLISION_TIMEOUT)) {
2504
sess->conflicting_addresses =
2505
g_list_delete_link (sess->conflicting_addresses, item);
2506
GST_DEBUG ("collision %p timed out", known_conflict);
2507
g_free (known_conflict);
2512
if (sess->change_ssrc) {
2513
GST_DEBUG ("need to change our SSRC (%08x)", own->ssrc);
2514
g_hash_table_steal (sess->ssrcs[sess->mask_idx],
2515
GINT_TO_POINTER (own->ssrc));
2517
own->ssrc = rtp_session_create_new_ssrc (sess);
2518
rtp_source_reset (own);
2520
g_hash_table_insert (sess->ssrcs[sess->mask_idx],
2521
GINT_TO_POINTER (own->ssrc), own);
2523
g_free (sess->bye_reason);
2524
sess->bye_reason = NULL;
2525
sess->sent_bye = FALSE;
2526
sess->change_ssrc = FALSE;
2528
GST_DEBUG ("changed our SSRC to %08x", own->ssrc);
2530
RTP_SESSION_UNLOCK (sess);
2533
g_object_notify (G_OBJECT (sess), "internal-ssrc");
2535
/* push out the RTCP packet */
2537
/* close the RTCP packet */
2538
gst_rtcp_buffer_end (data.rtcp);
2540
GST_DEBUG ("sending packet");
2541
if (sess->callbacks.send_rtcp)
2542
result = sess->callbacks.send_rtcp (sess, own, data.rtcp,
2543
sess->sent_bye, sess->send_rtcp_user_data);
2545
GST_DEBUG ("freeing packet");
2546
gst_buffer_unref (data.rtcp);
added
removed
gst/rtpmanager/rtpsession.c
