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https://gitlab.freedesktop.org/gstreamer/gstreamer.git
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3577 lines
102 KiB
C
3577 lines
102 KiB
C
/* GStreamer
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* Copyright (C) <2007> Wim Taymans <wim.taymans@gmail.com>
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*
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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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*
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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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*
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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., 51 Franklin St, Fifth Floor,
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* Boston, MA 02110-1301, USA.
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*/
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/* FIXME 0.11: suppress warnings for deprecated API such as GValueArray
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* with newer GLib versions (>= 2.31.0) */
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#define GLIB_DISABLE_DEPRECATION_WARNINGS
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#include <string.h>
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#include <gst/rtp/gstrtpbuffer.h>
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#include <gst/rtp/gstrtcpbuffer.h>
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#include <gst/glib-compat-private.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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enum
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{
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SIGNAL_GET_SOURCE_BY_SSRC,
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SIGNAL_ON_NEW_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_SSRC_SDES,
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SIGNAL_ON_BYE_SSRC,
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SIGNAL_ON_BYE_TIMEOUT,
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SIGNAL_ON_TIMEOUT,
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SIGNAL_ON_SENDER_TIMEOUT,
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SIGNAL_ON_SENDING_RTCP,
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SIGNAL_ON_FEEDBACK_RTCP,
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SIGNAL_SEND_RTCP,
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LAST_SIGNAL
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};
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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_FRACTION * RTP_STATS_BANDWIDTH)
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#define DEFAULT_RTCP_RR_BANDWIDTH -1
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#define DEFAULT_RTCP_RS_BANDWIDTH -1
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#define DEFAULT_RTCP_MTU 1400
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#define DEFAULT_SDES 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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#define DEFAULT_RTCP_MIN_INTERVAL (RTP_STATS_MIN_INTERVAL * GST_SECOND)
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#define DEFAULT_RTCP_FEEDBACK_RETENTION_WINDOW (2 * GST_SECOND)
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#define DEFAULT_RTCP_IMMEDIATE_FEEDBACK_THRESHOLD (3)
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#define DEFAULT_PROBATION RTP_DEFAULT_PROBATION
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enum
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{
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PROP_0,
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PROP_INTERNAL_SSRC,
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PROP_INTERNAL_SOURCE,
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PROP_BANDWIDTH,
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PROP_RTCP_FRACTION,
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PROP_RTCP_RR_BANDWIDTH,
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PROP_RTCP_RS_BANDWIDTH,
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PROP_RTCP_MTU,
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PROP_SDES,
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PROP_NUM_SOURCES,
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PROP_NUM_ACTIVE_SOURCES,
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PROP_SOURCES,
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PROP_FAVOR_NEW,
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PROP_RTCP_MIN_INTERVAL,
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PROP_RTCP_FEEDBACK_RETENTION_WINDOW,
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PROP_RTCP_IMMEDIATE_FEEDBACK_THRESHOLD,
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PROP_PROBATION,
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PROP_LAST
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};
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/* update average packet size */
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#define INIT_AVG(avg, val) \
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(avg) = (val);
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#define UPDATE_AVG(avg, val) \
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if ((avg) == 0) \
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(avg) = (val); \
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else \
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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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* collision table
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*/
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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 void rtp_session_send_rtcp (RTPSession * sess, GstClockTime max_delay);
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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 guint32 rtp_session_create_new_ssrc (RTPSession * sess);
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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 RTPSource *obtain_internal_source (RTPSession * sess,
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guint32 ssrc, gboolean * created);
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static GstFlowReturn rtp_session_schedule_bye_locked (RTPSession * sess,
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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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static gboolean
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accumulate_trues (GSignalInvocationHint * ihint, GValue * return_accu,
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const GValue * handler_return, gpointer data)
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{
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if (g_value_get_boolean (handler_return))
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g_value_set_boolean (return_accu, TRUE);
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return TRUE;
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}
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static void
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rtp_session_class_init (RTPSessionClass * klass)
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{
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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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/**
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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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*
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* Request the #RTPSource object with SSRC @ssrc in @session.
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*/
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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, g_cclosure_marshal_generic,
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RTP_TYPE_SOURCE, 1, G_TYPE_UINT);
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/**
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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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*
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* Notify of a new SSRC that entered @session.
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*/
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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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RTP_TYPE_SOURCE);
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/**
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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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*
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* Notify when we have an SSRC collision
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*/
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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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RTP_TYPE_SOURCE);
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/**
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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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*
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* Notify of a new SSRC that became validated.
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*/
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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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RTP_TYPE_SOURCE);
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/**
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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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*
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* Notify of a SSRC that is active, i.e., sending RTCP.
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*/
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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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RTP_TYPE_SOURCE);
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/**
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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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*
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* Notify that a new SDES was received for SSRC.
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*/
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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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RTP_TYPE_SOURCE);
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/**
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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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*
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* Notify of an SSRC that became inactive because of a BYE packet.
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*/
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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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RTP_TYPE_SOURCE);
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/**
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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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*
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* Notify of an SSRC that has timed out because of BYE
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*/
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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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RTP_TYPE_SOURCE);
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/**
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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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*
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* Notify of an SSRC that has timed out
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*/
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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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RTP_TYPE_SOURCE);
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/**
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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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*
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* Notify of an SSRC that was a sender but timed out and became a receiver.
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*/
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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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RTP_TYPE_SOURCE);
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/**
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* RTPSession::on-sending-rtcp
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* @session: the object which received the signal
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* @buffer: the #GstBuffer containing the RTCP packet about to be sent
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* @early: %TRUE if the packet is early, %FALSE if it is regular
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*
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* This signal is emitted before sending an RTCP packet, it can be used
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* to add extra RTCP Packets.
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*
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* Returns: %TRUE if the RTCP buffer should NOT be suppressed, %FALSE
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* if suppressing it is acceptable
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*/
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rtp_session_signals[SIGNAL_ON_SENDING_RTCP] =
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g_signal_new ("on-sending-rtcp", G_TYPE_FROM_CLASS (klass),
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G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (RTPSessionClass, on_sending_rtcp),
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accumulate_trues, NULL, g_cclosure_marshal_generic, G_TYPE_BOOLEAN, 2,
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GST_TYPE_BUFFER | G_SIGNAL_TYPE_STATIC_SCOPE, G_TYPE_BOOLEAN);
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/**
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* RTPSession::on-feedback-rtcp:
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* @session: the object which received the signal
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* @type: Type of RTCP packet, will be %GST_RTCP_TYPE_RTPFB or
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* %GST_RTCP_TYPE_RTPFB
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* @fbtype: The type of RTCP FB packet, probably part of #GstRTCPFBType
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* @sender_ssrc: The SSRC of the sender
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* @media_ssrc: The SSRC of the media this refers to
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* @fci: a #GstBuffer with the FCI data from the FB packet or %NULL if
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* there was no FCI
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*
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* Notify that a RTCP feedback packet has been received
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*/
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rtp_session_signals[SIGNAL_ON_FEEDBACK_RTCP] =
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g_signal_new ("on-feedback-rtcp", G_TYPE_FROM_CLASS (klass),
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G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (RTPSessionClass, on_feedback_rtcp),
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NULL, NULL, g_cclosure_marshal_generic, G_TYPE_NONE, 5, G_TYPE_UINT,
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G_TYPE_UINT, G_TYPE_UINT, G_TYPE_UINT, GST_TYPE_BUFFER);
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/**
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* RTPSession::send-rtcp:
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* @session: the object which received the signal
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* @max_delay: The maximum delay after which the feedback will not be useful
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* anymore
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*
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* Requests that the #RTPSession initiate a new RTCP packet as soon as
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* possible within the requested delay.
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*/
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rtp_session_signals[SIGNAL_SEND_RTCP] =
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g_signal_new ("send-rtcp", G_TYPE_FROM_CLASS (klass),
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G_SIGNAL_RUN_LAST | G_SIGNAL_ACTION,
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G_STRUCT_OFFSET (RTPSessionClass, send_rtcp), NULL, NULL,
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g_cclosure_marshal_generic, G_TYPE_NONE, 1, G_TYPE_UINT64);
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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 (deprecated)",
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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 (deprecated)",
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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 (0 for auto-discover)",
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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 (or as a real fraction of the RTP bandwidth if < 1)",
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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_RR_BANDWIDTH,
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g_param_spec_int ("rtcp-rr-bandwidth", "RTCP RR bandwidth",
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"The RTCP bandwidth used for receivers in bytes per second (-1 = default)",
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-1, G_MAXINT, DEFAULT_RTCP_RR_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_RS_BANDWIDTH,
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g_param_spec_int ("rtcp-rs-bandwidth", "RTCP RS bandwidth",
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"The RTCP bandwidth used for senders in bytes per second (-1 = default)",
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-1, G_MAXINT, DEFAULT_RTCP_RS_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_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,
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g_param_spec_boxed ("sdes", "SDES",
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"The SDES items of this session",
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GST_TYPE_STRUCTURE, 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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/**
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* RTPSource::sources
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*
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* Get a GValue Array of all sources in the session.
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*
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* <example>
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* <title>Getting the #RTPSources of a session
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* <programlisting>
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* {
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* GValueArray *arr;
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* GValue *val;
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* guint i;
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*
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* g_object_get (sess, "sources", &arr, NULL);
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*
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* for (i = 0; i < arr->n_values; i++) {
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* RTPSource *source;
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*
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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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* }
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* g_value_array_free (arr);
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* }
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* </programlisting>
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* </example>
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*/
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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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g_object_class_install_property (gobject_class, PROP_FAVOR_NEW,
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g_param_spec_boolean ("favor-new", "Favor new sources",
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"Resolve SSRC conflict in favor of new sources", FALSE,
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G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
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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->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);
|
|
}
|
|
|
|
rtp_stats_init_defaults (&sess->stats);
|
|
INIT_AVG (sess->stats.avg_rtcp_packet_size, 100);
|
|
rtp_stats_set_min_interval (&sess->stats,
|
|
(gdouble) DEFAULT_RTCP_MIN_INTERVAL / GST_SECOND);
|
|
|
|
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;
|
|
|
|
/* default UDP header length */
|
|
sess->header_len = 28;
|
|
sess->mtu = DEFAULT_RTCP_MTU;
|
|
|
|
sess->probation = DEFAULT_PROBATION;
|
|
|
|
/* some default SDES entries */
|
|
sess->sdes = gst_structure_new_empty ("application/x-rtp-source-sdes");
|
|
|
|
/* 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 ());
|
|
gst_structure_set (sess->sdes, "cname", G_TYPE_STRING, str, NULL);
|
|
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 ());
|
|
gst_structure_set (sdes, "name", G_TYPE_STRING, str, NULL);
|
|
g_free (str);
|
|
#endif
|
|
|
|
gst_structure_set (sess->sdes, "tool", G_TYPE_STRING, "GStreamer", NULL);
|
|
|
|
/* this is the SSRC we suggest */
|
|
sess->suggested_ssrc = rtp_session_create_new_ssrc (sess);
|
|
|
|
sess->first_rtcp = TRUE;
|
|
sess->next_rtcp_check_time = GST_CLOCK_TIME_NONE;
|
|
|
|
sess->allow_early = TRUE;
|
|
sess->next_early_rtcp_time = GST_CLOCK_TIME_NONE;
|
|
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;
|
|
}
|
|
|
|
static void
|
|
rtp_session_finalize (GObject * object)
|
|
{
|
|
RTPSession *sess;
|
|
gint i;
|
|
|
|
sess = RTP_SESSION_CAST (object);
|
|
|
|
gst_structure_free (sess->sdes);
|
|
|
|
for (i = 0; i < 32; i++)
|
|
g_hash_table_destroy (sess->ssrcs[i]);
|
|
|
|
g_mutex_clear (&sess->lock);
|
|
|
|
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:
|
|
break;
|
|
case PROP_BANDWIDTH:
|
|
RTP_SESSION_LOCK (sess);
|
|
sess->bandwidth = g_value_get_double (value);
|
|
sess->recalc_bandwidth = TRUE;
|
|
RTP_SESSION_UNLOCK (sess);
|
|
break;
|
|
case PROP_RTCP_FRACTION:
|
|
RTP_SESSION_LOCK (sess);
|
|
sess->rtcp_bandwidth = g_value_get_double (value);
|
|
sess->recalc_bandwidth = TRUE;
|
|
RTP_SESSION_UNLOCK (sess);
|
|
break;
|
|
case PROP_RTCP_RR_BANDWIDTH:
|
|
RTP_SESSION_LOCK (sess);
|
|
sess->rtcp_rr_bandwidth = g_value_get_int (value);
|
|
sess->recalc_bandwidth = TRUE;
|
|
RTP_SESSION_UNLOCK (sess);
|
|
break;
|
|
case PROP_RTCP_RS_BANDWIDTH:
|
|
RTP_SESSION_LOCK (sess);
|
|
sess->rtcp_rs_bandwidth = g_value_get_int (value);
|
|
sess->recalc_bandwidth = TRUE;
|
|
RTP_SESSION_UNLOCK (sess);
|
|
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);
|
|
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_suggest_ssrc (sess));
|
|
break;
|
|
case PROP_INTERNAL_SOURCE:
|
|
/* FIXME, return a random source */
|
|
g_value_set_object (value, NULL);
|
|
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);
|
|
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_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)
|
|
{
|
|
GstStructure *result = NULL;
|
|
|
|
g_return_val_if_fail (RTP_IS_SESSION (sess), NULL);
|
|
|
|
RTP_SESSION_LOCK (sess);
|
|
if (sess->sdes)
|
|
result = gst_structure_copy (sess->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);
|
|
if (sess->sdes)
|
|
gst_structure_free (sess->sdes);
|
|
sess->sdes = 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->internal) {
|
|
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)
|
|
{
|
|
guint32 ssrc;
|
|
|
|
/* If we have no arrival address, we can't do collision checking */
|
|
if (!arrival->address)
|
|
return FALSE;
|
|
|
|
ssrc = rtp_source_get_ssrc (source);
|
|
|
|
if (!source->internal) {
|
|
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", ssrc);
|
|
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", ssrc,
|
|
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", ssrc, 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
|
|
*/
|
|
} 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", ssrc);
|
|
/* mark the source BYE */
|
|
rtp_source_mark_bye (source, "SSRC Collision");
|
|
/* if we were suggesting this SSRC, change to something else */
|
|
if (sess->suggested_ssrc == ssrc)
|
|
sess->suggested_ssrc = rtp_session_create_new_ssrc (sess);
|
|
|
|
on_ssrc_collision (sess, source);
|
|
|
|
rtp_session_schedule_bye_locked (sess, arrival->current_time);
|
|
}
|
|
}
|
|
|
|
return TRUE;
|
|
}
|
|
|
|
static RTPSource *
|
|
find_source (RTPSession * sess, guint32 ssrc)
|
|
{
|
|
return g_hash_table_lookup (sess->ssrcs[sess->mask_idx],
|
|
GINT_TO_POINTER (ssrc));
|
|
}
|
|
|
|
static void
|
|
add_source (RTPSession * sess, RTPSource * src)
|
|
{
|
|
g_hash_table_insert (sess->ssrcs[sess->mask_idx],
|
|
GINT_TO_POINTER (src->ssrc), src);
|
|
/* report the new source ASAP */
|
|
src->generation = sess->generation;
|
|
/* we have one more source now */
|
|
sess->total_sources++;
|
|
if (RTP_SOURCE_IS_ACTIVE (src))
|
|
sess->stats.active_sources++;
|
|
if (src->internal) {
|
|
sess->stats.internal_sources++;
|
|
if (sess->suggested_ssrc != src->ssrc)
|
|
sess->suggested_ssrc = src->ssrc;
|
|
}
|
|
}
|
|
|
|
/* 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 = find_source (sess, ssrc);
|
|
if (source == NULL) {
|
|
/* make new Source in probation and insert */
|
|
source = rtp_source_new (ssrc);
|
|
|
|
GST_DEBUG ("creating new source %08x %p", ssrc, source);
|
|
|
|
/* 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);
|
|
|
|
add_source (sess, source);
|
|
*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;
|
|
}
|
|
|
|
/* must be called with the session lock, the returned source needs to be
|
|
* unreffed after usage. */
|
|
static RTPSource *
|
|
obtain_internal_source (RTPSession * sess, guint32 ssrc, gboolean * created)
|
|
{
|
|
RTPSource *source;
|
|
|
|
source = find_source (sess, ssrc);
|
|
if (source == NULL) {
|
|
/* make new internal Source and insert */
|
|
source = rtp_source_new (ssrc);
|
|
|
|
GST_DEBUG ("creating new internal source %08x %p", ssrc, source);
|
|
|
|
source->validated = TRUE;
|
|
source->internal = TRUE;
|
|
rtp_source_set_sdes_struct (source, gst_structure_copy (sess->sdes));
|
|
rtp_source_set_callbacks (source, &callbacks, sess);
|
|
|
|
add_source (sess, source);
|
|
*created = TRUE;
|
|
} else {
|
|
*created = FALSE;
|
|
}
|
|
g_object_ref (source);
|
|
|
|
return source;
|
|
}
|
|
|
|
/**
|
|
* rtp_session_suggest_ssrc:
|
|
* @sess: a #RTPSession
|
|
*
|
|
* Suggest an unused SSRC in @sess.
|
|
*
|
|
* Returns: a free unused SSRC
|
|
*/
|
|
guint32
|
|
rtp_session_suggest_ssrc (RTPSession * sess)
|
|
{
|
|
guint32 result;
|
|
|
|
g_return_val_if_fail (RTP_IS_SESSION (sess), 0);
|
|
|
|
RTP_SESSION_LOCK (sess);
|
|
result = sess->suggested_ssrc;
|
|
RTP_SESSION_UNLOCK (sess);
|
|
|
|
return result;
|
|
}
|
|
|
|
/**
|
|
* 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 = find_source (sess, src->ssrc);
|
|
if (find == NULL) {
|
|
add_source (sess, src);
|
|
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 = find_source (sess, ssrc);
|
|
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 (find_source (sess, 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);
|
|
add_source (sess, source);
|
|
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);
|
|
}
|
|
|
|
static gboolean
|
|
source_update_active (RTPSession * sess, RTPSource * source,
|
|
gboolean prevactive)
|
|
{
|
|
gboolean active = RTP_SOURCE_IS_ACTIVE (source);
|
|
guint32 ssrc = source->ssrc;
|
|
|
|
if (prevactive == active)
|
|
return FALSE;
|
|
|
|
if (active) {
|
|
sess->stats.active_sources++;
|
|
GST_DEBUG ("source: %08x became active, %d active sources", ssrc,
|
|
sess->stats.active_sources);
|
|
} else {
|
|
sess->stats.active_sources--;
|
|
GST_DEBUG ("source: %08x became inactive, %d active sources", ssrc,
|
|
sess->stats.active_sources);
|
|
}
|
|
return TRUE;
|
|
}
|
|
|
|
static gboolean
|
|
source_update_sender (RTPSession * sess, RTPSource * source,
|
|
gboolean prevsender)
|
|
{
|
|
gboolean sender = RTP_SOURCE_IS_SENDER (source);
|
|
guint32 ssrc = source->ssrc;
|
|
|
|
if (prevsender == sender)
|
|
return FALSE;
|
|
|
|
if (sender) {
|
|
sess->stats.sender_sources++;
|
|
if (source->internal)
|
|
sess->stats.internal_sender_sources++;
|
|
GST_DEBUG ("source: %08x became sender, %d sender sources", ssrc,
|
|
sess->stats.sender_sources);
|
|
} else {
|
|
sess->stats.sender_sources--;
|
|
if (source->internal)
|
|
sess->stats.internal_sender_sources--;
|
|
GST_DEBUG ("source: %08x became non sender, %d sender sources", ssrc,
|
|
sess->stats.sender_sources);
|
|
}
|
|
return TRUE;
|
|
}
|
|
|
|
/**
|
|
* 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;
|
|
|
|
/* get SSRC to look up in session database */
|
|
ssrc = gst_rtp_buffer_get_ssrc (&rtp);
|
|
/* 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);
|
|
|
|
RTP_SESSION_LOCK (sess);
|
|
|
|
/* update arrival stats */
|
|
update_arrival_stats (sess, &arrival, TRUE, buffer, current_time,
|
|
running_time, -1);
|
|
|
|
source = obtain_source (sess, ssrc, &created, &arrival, TRUE);
|
|
if (!source)
|
|
goto collision;
|
|
|
|
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 (source_update_active (sess, source, prevactive))
|
|
on_ssrc_validated (sess, source);
|
|
|
|
source_update_sender (sess, source, prevsender);
|
|
|
|
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);
|
|
source_update_active (sess, csrc_src, FALSE);
|
|
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;
|
|
}
|
|
collision:
|
|
{
|
|
RTP_SESSION_UNLOCK (sess);
|
|
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;
|
|
RTPSource *src;
|
|
|
|
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);
|
|
|
|
/* find our own source */
|
|
src = find_source (sess, ssrc);
|
|
if (src == NULL)
|
|
continue;
|
|
|
|
if (src->internal && RTP_SOURCE_IS_ACTIVE (src)) {
|
|
/* 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. */
|
|
/* FIXME, need to keep track who the RB block is from */
|
|
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_IS_MARKED_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);
|
|
|
|
source_update_sender (sess, source, prevsender);
|
|
|
|
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, prevactive;
|
|
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);
|
|
|
|
prevactive = RTP_SOURCE_IS_ACTIVE (source);
|
|
source->validated = TRUE;
|
|
|
|
if (created)
|
|
on_new_ssrc (sess, source);
|
|
|
|
/* source became active */
|
|
if (source_update_active (sess, source, prevactive))
|
|
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);
|
|
|
|
/* find src and mark bye, no probation when dealing with RTCP */
|
|
source = obtain_source (sess, ssrc, &created, arrival, FALSE);
|
|
if (!source)
|
|
return;
|
|
|
|
if (source->internal) {
|
|
/* our own source, something weird with this packet */
|
|
g_object_unref (source);
|
|
continue;
|
|
}
|
|
|
|
/* 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);
|
|
|
|
/* mark the source BYE */
|
|
rtp_source_mark_bye (source, reason);
|
|
|
|
pmembers = sess->stats.active_sources;
|
|
|
|
source_update_active (sess, source, prevactive);
|
|
source_update_sender (sess, source, prevsender);
|
|
|
|
members = sess->stats.active_sources;
|
|
|
|
if (!sess->scheduled_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 (sess->next_rtcp_check_time != GST_CLOCK_TIME_NONE &&
|
|
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 (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 = find_source (sess, 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 = find_source (sess, sender_ssrc);
|
|
|
|
/* Hack because Google fails to set the sender_ssrc correctly */
|
|
if (!src && sender_ssrc == 1) {
|
|
GHashTableIter iter;
|
|
|
|
/* we can't find the source if there are multiple */
|
|
if (sess->stats.sender_sources > sess->stats.internal_sender_sources + 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->internal && rtp_source_is_sender (src))
|
|
break;
|
|
src = NULL;
|
|
}
|
|
}
|
|
if (!src)
|
|
return;
|
|
|
|
for (position = 0; position < fci_length; position += 8) {
|
|
guint8 *data = fci_data + position;
|
|
RTPSource *own;
|
|
|
|
ssrc = GST_READ_UINT32_BE (data);
|
|
|
|
own = find_source (sess, ssrc);
|
|
if (own->internal) {
|
|
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);
|
|
RTPSource *src;
|
|
|
|
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);
|
|
}
|
|
|
|
src = find_source (sess, media_ssrc);
|
|
if (!src)
|
|
return;
|
|
|
|
if (sess->rtcp_feedback_retention_window) {
|
|
rtp_source_retain_rtcp_packet (src, packet, arrival->running_time);
|
|
}
|
|
|
|
if (src->internal ||
|
|
/* 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);
|
|
|
|
/* 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->scheduled_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->scheduled_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) {
|
|
result = sess->callbacks.sync_rtcp (sess, 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;
|
|
}
|
|
}
|
|
|
|
/**
|
|
* 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)
|
|
{
|
|
GstStructure *s;
|
|
guint ssrc;
|
|
|
|
g_return_if_fail (RTP_IS_SESSION (sess));
|
|
g_return_if_fail (GST_IS_CAPS (caps));
|
|
|
|
GST_LOG ("received caps %" GST_PTR_FORMAT, caps);
|
|
|
|
s = gst_caps_get_structure (caps, 0);
|
|
|
|
if (gst_structure_get_uint (s, "ssrc", &ssrc)) {
|
|
RTPSource *source;
|
|
gboolean created;
|
|
|
|
RTP_SESSION_LOCK (sess);
|
|
source = obtain_internal_source (sess, ssrc, &created);
|
|
if (source) {
|
|
rtp_source_update_caps (source, caps);
|
|
g_object_unref (source);
|
|
}
|
|
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;
|
|
GstBuffer *buffer;
|
|
GstRTPBuffer rtp = { NULL };
|
|
guint32 ssrc;
|
|
gboolean created;
|
|
|
|
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");
|
|
|
|
if (is_list) {
|
|
GstBufferList *list = GST_BUFFER_LIST_CAST (data);
|
|
|
|
buffer = gst_buffer_list_get (list, 0);
|
|
if (!buffer)
|
|
goto no_buffer;
|
|
} else {
|
|
buffer = GST_BUFFER_CAST (data);
|
|
}
|
|
|
|
if (!gst_rtp_buffer_map (buffer, GST_MAP_READ, &rtp))
|
|
goto invalid_packet;
|
|
|
|
/* get SSRC and look up in session database */
|
|
ssrc = gst_rtp_buffer_get_ssrc (&rtp);
|
|
|
|
gst_rtp_buffer_unmap (&rtp);
|
|
|
|
RTP_SESSION_LOCK (sess);
|
|
source = obtain_internal_source (sess, ssrc, &created);
|
|
|
|
/* 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);
|
|
|
|
source_update_sender (sess, source, prevsender);
|
|
|
|
if (oldrate != source->bitrate)
|
|
sess->recalc_bandwidth = TRUE;
|
|
RTP_SESSION_UNLOCK (sess);
|
|
|
|
g_object_unref (source);
|
|
|
|
return result;
|
|
|
|
invalid_packet:
|
|
{
|
|
gst_mini_object_unref (GST_MINI_OBJECT_CAST (data));
|
|
GST_DEBUG ("invalid RTP packet received");
|
|
return GST_FLOW_OK;
|
|
}
|
|
no_buffer:
|
|
{
|
|
gst_mini_object_unref (GST_MINI_OBJECT_CAST (data));
|
|
GST_DEBUG ("no buffer in list");
|
|
return GST_FLOW_OK;
|
|
}
|
|
}
|
|
|
|
static void
|
|
add_bitrates (gpointer key, RTPSource * source, gdouble * bandwidth)
|
|
{
|
|
*bandwidth += source->bitrate;
|
|
}
|
|
|
|
/* must be called with session lock */
|
|
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 = 0;
|
|
|
|
g_hash_table_foreach (sess->ssrcs[sess->mask_idx],
|
|
(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->scheduled_bye) {
|
|
result = rtp_stats_calculate_bye_interval (&sess->stats);
|
|
} else {
|
|
result = rtp_stats_calculate_rtcp_interval (&sess->stats,
|
|
sess->stats.internal_sender_sources > 0, 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;
|
|
}
|
|
|
|
static void
|
|
source_mark_bye (const gchar * key, RTPSource * source, const gchar * reason)
|
|
{
|
|
if (source->internal)
|
|
rtp_source_mark_bye (source, reason);
|
|
}
|
|
|
|
/**
|
|
* rtp_session_mark_all_bye:
|
|
* @sess: an #RTPSession
|
|
* @reason: a reason
|
|
*
|
|
* Mark all internal sources of the session as BYE with @reason.
|
|
*/
|
|
void
|
|
rtp_session_mark_all_bye (RTPSession * sess, const gchar * reason)
|
|
{
|
|
g_return_if_fail (RTP_IS_SESSION (sess));
|
|
|
|
RTP_SESSION_LOCK (sess);
|
|
g_hash_table_foreach (sess->ssrcs[sess->mask_idx],
|
|
(GHFunc) source_mark_bye, (gpointer) reason);
|
|
RTP_SESSION_UNLOCK (sess);
|
|
}
|
|
|
|
/* 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, GstClockTime current_time)
|
|
{
|
|
GstFlowReturn result = GST_FLOW_OK;
|
|
GstClockTime interval;
|
|
|
|
/* nothing to do it we already scheduled bye */
|
|
if (sess->scheduled_bye)
|
|
goto done;
|
|
|
|
/* we schedule BYE now */
|
|
sess->scheduled_bye = TRUE;
|
|
/* at least one member wants to send a BYE */
|
|
INIT_AVG (sess->stats.avg_rtcp_packet_size, 100);
|
|
sess->stats.bye_members = 1;
|
|
sess->first_rtcp = TRUE;
|
|
sess->allow_early = TRUE;
|
|
|
|
/* reschedule transmission */
|
|
sess->last_rtcp_send_time = current_time;
|
|
interval = calculate_rtcp_interval (sess, FALSE, TRUE);
|
|
|
|
if (interval != GST_CLOCK_TIME_NONE)
|
|
sess->next_rtcp_check_time = current_time + interval;
|
|
else
|
|
sess->next_rtcp_check_time = GST_CLOCK_TIME_NONE;
|
|
|
|
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
|
|
* @current_time: the current system time
|
|
*
|
|
* Schedule a BYE message for all sources marked as BYE in @sess.
|
|
*
|
|
* Returns: a #GstFlowReturn.
|
|
*/
|
|
GstFlowReturn
|
|
rtp_session_schedule_bye (RTPSession * sess, 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, 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 time: %" GST_TIME_FORMAT,
|
|
GST_TIME_ARGS (current_time), GST_TIME_ARGS (result));
|
|
|
|
if (result == GST_CLOCK_TIME_NONE || 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->scheduled_bye) {
|
|
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
|
|
{
|
|
RTPSource *source;
|
|
gboolean is_bye;
|
|
GstBuffer *buffer;
|
|
} ReportOutput;
|
|
|
|
typedef struct
|
|
{
|
|
GstRTCPBuffer rtcpbuf;
|
|
RTPSession *sess;
|
|
RTPSource *source;
|
|
guint num_to_report;
|
|
gboolean have_fir;
|
|
gboolean have_pli;
|
|
gboolean have_nack;
|
|
GstBuffer *rtcp;
|
|
GstClockTime current_time;
|
|
guint64 ntpnstime;
|
|
GstClockTime running_time;
|
|
GstClockTime interval;
|
|
GstRTCPPacket packet;
|
|
gboolean has_sdes;
|
|
gboolean is_early;
|
|
gboolean may_suppress;
|
|
GQueue output;
|
|
} ReportData;
|
|
|
|
static void
|
|
session_start_rtcp (RTPSession * sess, ReportData * data)
|
|
{
|
|
GstRTCPPacket *packet = &data->packet;
|
|
RTPSource *own = data->source;
|
|
GstRTCPBuffer *rtcp = &data->rtcpbuf;
|
|
|
|
data->rtcp = gst_rtcp_buffer_new (sess->mtu);
|
|
data->has_sdes = FALSE;
|
|
|
|
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;
|
|
guint8 fractionlost;
|
|
gint32 packetslost;
|
|
guint32 exthighestseq, jitter;
|
|
guint32 lsr, dlsr;
|
|
|
|
/* don't report for sources in future generations */
|
|
if (((gint16) (source->generation - sess->generation)) > 0) {
|
|
GST_DEBUG ("source %08x generation %u > %u", source->ssrc,
|
|
source->generation, sess->generation);
|
|
return;
|
|
}
|
|
|
|
/* only report about other sender */
|
|
if (source == data->source)
|
|
goto reported;
|
|
|
|
if (gst_rtcp_packet_get_rb_count (packet) == GST_RTCP_MAX_RB_COUNT) {
|
|
GST_DEBUG ("max RB count reached");
|
|
return;
|
|
}
|
|
|
|
if (!RTP_SOURCE_IS_SENDER (source)) {
|
|
GST_DEBUG ("source %08x not sender", source->ssrc);
|
|
goto reported;
|
|
}
|
|
|
|
GST_DEBUG ("create RB for SSRC %08x", source->ssrc);
|
|
|
|
/* 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);
|
|
|
|
reported:
|
|
/* source is reported, move to next generation */
|
|
source->generation = sess->generation + 1;
|
|
|
|
/* if we reported all sources in this generation, move to next */
|
|
if (--data->num_to_report == 0) {
|
|
sess->generation++;
|
|
GST_DEBUG ("all reported, generation now %u", sess->generation);
|
|
}
|
|
}
|
|
|
|
/* construct FIR */
|
|
static void
|
|
session_add_fir (const gchar * key, RTPSource * source, ReportData * data)
|
|
{
|
|
GstRTCPPacket *packet = &data->packet;
|
|
guint16 len;
|
|
guint8 *fci_data;
|
|
|
|
if (!source->send_fir)
|
|
return;
|
|
|
|
len = gst_rtcp_packet_fb_get_fci_length (packet);
|
|
if (!gst_rtcp_packet_fb_set_fci_length (packet, len + 2))
|
|
/* exit because the packet is full, will put next request in a
|
|
* further packet */
|
|
return;
|
|
|
|
fci_data = gst_rtcp_packet_fb_get_fci (packet) + (len * 4);
|
|
|
|
GST_WRITE_UINT32_BE (fci_data, source->ssrc);
|
|
fci_data += 4;
|
|
fci_data[0] = source->current_send_fir_seqnum;
|
|
fci_data[1] = fci_data[2] = fci_data[3] = 0;
|
|
|
|
source->send_fir = FALSE;
|
|
}
|
|
|
|
static void
|
|
session_fir (RTPSession * sess, ReportData * data)
|
|
{
|
|
GstRTCPBuffer *rtcp = &data->rtcpbuf;
|
|
GstRTCPPacket *packet = &data->packet;
|
|
|
|
if (!gst_rtcp_buffer_add_packet (rtcp, GST_RTCP_TYPE_PSFB, packet))
|
|
return;
|
|
|
|
gst_rtcp_packet_fb_set_type (packet, GST_RTCP_PSFB_TYPE_FIR);
|
|
gst_rtcp_packet_fb_set_sender_ssrc (packet, data->source->ssrc);
|
|
gst_rtcp_packet_fb_set_media_ssrc (packet, 0);
|
|
|
|
g_hash_table_foreach (sess->ssrcs[sess->mask_idx],
|
|
(GHFunc) session_add_fir, data);
|
|
|
|
if (gst_rtcp_packet_fb_get_fci_length (packet) == 0)
|
|
gst_rtcp_packet_remove (packet);
|
|
else
|
|
data->may_suppress = FALSE;
|
|
}
|
|
|
|
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;
|
|
}
|
|
|
|
/* construct PLI */
|
|
static void
|
|
session_pli (const gchar * key, RTPSource * source, ReportData * data)
|
|
{
|
|
GstRTCPBuffer *rtcp = &data->rtcpbuf;
|
|
GstRTCPPacket *packet = &data->packet;
|
|
|
|
if (!source->send_pli)
|
|
return;
|
|
|
|
if (rtp_source_has_retained (source, has_pli_compare_func, NULL))
|
|
return;
|
|
|
|
if (!gst_rtcp_buffer_add_packet (rtcp, GST_RTCP_TYPE_PSFB, packet))
|
|
/* exit because the packet is full, will put next request in a
|
|
* further packet */
|
|
return;
|
|
|
|
gst_rtcp_packet_fb_set_type (packet, GST_RTCP_PSFB_TYPE_PLI);
|
|
gst_rtcp_packet_fb_set_sender_ssrc (packet, data->source->ssrc);
|
|
gst_rtcp_packet_fb_set_media_ssrc (packet, source->ssrc);
|
|
|
|
source->send_pli = FALSE;
|
|
data->may_suppress = FALSE;
|
|
}
|
|
|
|
/* construct NACK */
|
|
static void
|
|
session_nack (const gchar * key, RTPSource * source, ReportData * data)
|
|
{
|
|
GstRTCPBuffer *rtcp = &data->rtcpbuf;
|
|
GstRTCPPacket *packet = &data->packet;
|
|
guint32 *nacks;
|
|
guint n_nacks, i;
|
|
guint8 *fci_data;
|
|
|
|
if (!source->send_nack)
|
|
return;
|
|
|
|
if (!gst_rtcp_buffer_add_packet (rtcp, GST_RTCP_TYPE_RTPFB, packet))
|
|
/* exit because the packet is full, will put next request in a
|
|
* further packet */
|
|
return;
|
|
|
|
gst_rtcp_packet_fb_set_type (packet, GST_RTCP_RTPFB_TYPE_NACK);
|
|
gst_rtcp_packet_fb_set_sender_ssrc (packet, data->source->ssrc);
|
|
gst_rtcp_packet_fb_set_media_ssrc (packet, source->ssrc);
|
|
|
|
nacks = rtp_source_get_nacks (source, &n_nacks);
|
|
GST_DEBUG ("%u NACKs", n_nacks);
|
|
if (!gst_rtcp_packet_fb_set_fci_length (packet, n_nacks))
|
|
return;
|
|
|
|
fci_data = gst_rtcp_packet_fb_get_fci (packet);
|
|
for (i = 0; i < n_nacks; i++) {
|
|
GST_WRITE_UINT32_BE (fci_data, nacks[i]);
|
|
fci_data += 4;
|
|
}
|
|
|
|
rtp_source_clear_nacks (source);
|
|
data->may_suppress = FALSE;
|
|
}
|
|
|
|
/* 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;
|
|
|
|
GST_DEBUG ("look at %08x, generation %u", source->ssrc, source->generation);
|
|
|
|
/* check for outdated collisions */
|
|
if (source->internal) {
|
|
GST_DEBUG ("Timing out collisions for %x", source->ssrc);
|
|
rtp_source_timeout (source, data->current_time,
|
|
/* "a relatively long time" -- RFC 3550 section 8.2 */
|
|
RTP_STATS_MIN_INTERVAL * GST_SECOND * 10,
|
|
data->running_time - sess->rtcp_feedback_retention_window);
|
|
}
|
|
|
|
/* nothing else to do when without RTCP */
|
|
if (data->interval == GST_CLOCK_TIME_NONE)
|
|
return;
|
|
|
|
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->internal) {
|
|
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));
|
|
|
|
if (!source->internal) {
|
|
if (source->marked_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 sources. */
|
|
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) {
|
|
if (source->internal && source->sent_bye) {
|
|
/* an internal source is BYE and stopped sending RTP, remove */
|
|
GST_DEBUG ("internal BYE source %08x timed out, last %"
|
|
GST_TIME_FORMAT, source->ssrc, GST_TIME_ARGS (btime));
|
|
remove = TRUE;
|
|
} else {
|
|
GST_DEBUG ("sender source %08x timed out and became receiver, last %"
|
|
GST_TIME_FORMAT, source->ssrc, GST_TIME_ARGS (btime));
|
|
sendertimeout = TRUE;
|
|
}
|
|
}
|
|
}
|
|
}
|
|
|
|
if (remove) {
|
|
sess->total_sources--;
|
|
if (is_sender) {
|
|
sess->stats.sender_sources--;
|
|
if (source->internal)
|
|
sess->stats.internal_sender_sources--;
|
|
}
|
|
if (is_active)
|
|
sess->stats.active_sources--;
|
|
|
|
if (source->internal)
|
|
sess->stats.internal_sources--;
|
|
|
|
if (byetimeout)
|
|
on_bye_timeout (sess, source);
|
|
else
|
|
on_timeout (sess, source);
|
|
} else {
|
|
if (sendertimeout) {
|
|
source->is_sender = FALSE;
|
|
sess->stats.sender_sources--;
|
|
if (source->internal)
|
|
sess->stats.internal_sender_sources--;
|
|
|
|
on_sender_timeout (sess, source);
|
|
}
|
|
/* count how many source to report in this generation */
|
|
if (((gint16) (source->generation - sess->generation)) <= 0)
|
|
data->num_to_report++;
|
|
}
|
|
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, data->source->ssrc);
|
|
|
|
sdes = rtp_source_get_sdes_struct (data->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
|
|
make_source_bye (RTPSession * sess, RTPSource * source, ReportData * data)
|
|
{
|
|
GstRTCPPacket *packet = &data->packet;
|
|
GstRTCPBuffer *rtcp = &data->rtcpbuf;
|
|
|
|
/* 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, source->ssrc);
|
|
if (source->bye_reason)
|
|
gst_rtcp_packet_bye_set_reason (packet, source->bye_reason);
|
|
|
|
/* we have a BYE packet now */
|
|
source->sent_bye = TRUE;
|
|
}
|
|
|
|
static gboolean
|
|
is_rtcp_time (RTPSession * sess, GstClockTime current_time, ReportData * data)
|
|
{
|
|
GstClockTime new_send_time, elapsed;
|
|
|
|
if (GST_CLOCK_TIME_IS_VALID (sess->next_early_rtcp_time))
|
|
data->is_early = TRUE;
|
|
else
|
|
data->is_early = FALSE;
|
|
|
|
if (data->is_early && sess->next_early_rtcp_time < current_time)
|
|
goto early;
|
|
|
|
/* no need to check yet */
|
|
if (sess->next_rtcp_check_time == GST_CLOCK_TIME_NONE ||
|
|
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;
|
|
|
|
new_send_time = data->interval;
|
|
/* perform forward reconsideration */
|
|
if (new_send_time != GST_CLOCK_TIME_NONE) {
|
|
new_send_time = rtp_stats_add_rtcp_jitter (&sess->stats, new_send_time);
|
|
|
|
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 (new_send_time == GST_CLOCK_TIME_NONE || 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 = new_send_time;
|
|
if (new_send_time != GST_CLOCK_TIME_NONE) {
|
|
sess->next_rtcp_check_time += current_time;
|
|
|
|
/* Apply the rules from RFC 4585 section 3.5.3 */
|
|
if (sess->stats.min_interval != 0 && !sess->first_rtcp) {
|
|
GstClockTime 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,
|
|
ReportData * data)
|
|
{
|
|
if (source->closing)
|
|
return TRUE;
|
|
|
|
if (source->send_fir)
|
|
data->have_fir = TRUE;
|
|
if (source->send_pli)
|
|
data->have_pli = TRUE;
|
|
if (source->send_nack)
|
|
data->have_nack = TRUE;
|
|
|
|
return FALSE;
|
|
}
|
|
|
|
static void
|
|
generate_rtcp (const gchar * key, RTPSource * source, ReportData * data)
|
|
{
|
|
RTPSession *sess = data->sess;
|
|
gboolean is_bye = FALSE;
|
|
ReportOutput *output;
|
|
|
|
/* only generate RTCP for active internal sources */
|
|
if (!source->internal || source->sent_bye)
|
|
return;
|
|
|
|
data->source = source;
|
|
|
|
/* open packet */
|
|
session_start_rtcp (sess, data);
|
|
|
|
if (source->marked_bye) {
|
|
/* send BYE */
|
|
make_source_bye (sess, source, data);
|
|
is_bye = TRUE;
|
|
} else if (!data->is_early) {
|
|
/* loop over all known sources and add report blocks. If we are early, we
|
|
* just make a minimal RTCP packet and skip this step */
|
|
g_hash_table_foreach (sess->ssrcs[sess->mask_idx],
|
|
(GHFunc) session_report_blocks, data);
|
|
}
|
|
if (!data->has_sdes)
|
|
session_sdes (sess, data);
|
|
|
|
if (data->have_fir)
|
|
session_fir (sess, data);
|
|
|
|
if (data->have_pli)
|
|
g_hash_table_foreach (sess->ssrcs[sess->mask_idx],
|
|
(GHFunc) session_pli, data);
|
|
|
|
if (data->have_nack)
|
|
g_hash_table_foreach (sess->ssrcs[sess->mask_idx],
|
|
(GHFunc) session_nack, data);
|
|
|
|
gst_rtcp_buffer_unmap (&data->rtcpbuf);
|
|
|
|
output = g_slice_new (ReportOutput);
|
|
output->source = g_object_ref (source);
|
|
output->is_bye = is_bye;
|
|
output->buffer = data->rtcp;
|
|
/* queue the RTCP packet to push later */
|
|
g_queue_push_tail (&data->output, output);
|
|
}
|
|
|
|
/**
|
|
* 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 };
|
|
GHashTable *table_copy;
|
|
ReportOutput *output;
|
|
|
|
g_return_val_if_fail (RTP_IS_SESSION (sess), GST_FLOW_ERROR);
|
|
|
|
GST_DEBUG ("reporting at %" GST_TIME_FORMAT ", NTP time %" GST_TIME_FORMAT
|
|
", running-time %" GST_TIME_FORMAT, GST_TIME_ARGS (current_time),
|
|
GST_TIME_ARGS (ntpnstime), GST_TIME_ARGS (running_time));
|
|
|
|
data.sess = sess;
|
|
data.current_time = current_time;
|
|
data.ntpnstime = ntpnstime;
|
|
data.running_time = running_time;
|
|
data.num_to_report = 0;
|
|
data.may_suppress = FALSE;
|
|
g_queue_init (&data.output);
|
|
|
|
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);
|
|
|
|
/* we need an internal source now */
|
|
if (sess->stats.internal_sources == 0) {
|
|
RTPSource *source;
|
|
gboolean created;
|
|
|
|
source = obtain_internal_source (sess, sess->suggested_ssrc, &created);
|
|
g_object_unref (source);
|
|
}
|
|
|
|
/* 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, &data);
|
|
|
|
/* see if we need to generate SR or RR packets */
|
|
if (!is_rtcp_time (sess, current_time, &data))
|
|
goto done;
|
|
|
|
GST_DEBUG ("doing RTCP generation %u for %u sources", sess->generation,
|
|
data.num_to_report);
|
|
|
|
/* generate RTCP for all internal sources */
|
|
g_hash_table_foreach (sess->ssrcs[sess->mask_idx],
|
|
(GHFunc) generate_rtcp, &data);
|
|
|
|
/* 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;
|
|
|
|
done:
|
|
RTP_SESSION_UNLOCK (sess);
|
|
|
|
/* push out the RTCP packets */
|
|
while ((output = g_queue_pop_head (&data.output))) {
|
|
gboolean do_not_suppress;
|
|
GstBuffer *buffer = output->buffer;
|
|
RTPSource *source = output->source;
|
|
|
|
/* Give the user a change to add its own packet */
|
|
g_signal_emit (sess, rtp_session_signals[SIGNAL_ON_SENDING_RTCP], 0,
|
|
buffer, 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 (buffer) + 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, source, buffer, output->is_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 (buffer);
|
|
}
|
|
g_object_unref (source);
|
|
g_slice_free (ReportOutput, output);
|
|
}
|
|
return result;
|
|
}
|
|
|
|
/**
|
|
* rtp_session_request_early_rtcp:
|
|
* @sess: an #RTPSession
|
|
* @current_time: the current system time
|
|
* @max_delay: maximum delay
|
|
*
|
|
* Request transmission of early RTCP
|
|
*/
|
|
void
|
|
rtp_session_request_early_rtcp (RTPSession * sess, GstClockTime current_time,
|
|
GstClockTime 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;
|
|
|
|
if (!GST_CLOCK_TIME_IS_VALID (sess->next_rtcp_check_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);
|
|
}
|
|
|
|
static void
|
|
rtp_session_send_rtcp (RTPSession * sess, GstClockTime 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);
|
|
}
|
|
|
|
gboolean
|
|
rtp_session_request_key_unit (RTPSession * sess, guint32 ssrc,
|
|
gboolean fir, gint count)
|
|
{
|
|
RTPSource *src = find_source (sess, 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_send_rtcp (sess, 200 * GST_MSECOND);
|
|
|
|
return TRUE;
|
|
}
|
|
|
|
gboolean
|
|
rtp_session_request_nack (RTPSession * sess, guint32 ssrc, guint16 seqnum,
|
|
GstClockTime max_delay)
|
|
{
|
|
RTPSource *source = find_source (sess, ssrc);
|
|
|
|
if (source == NULL)
|
|
return FALSE;
|
|
|
|
GST_DEBUG ("request NACK for %08x, #%u", ssrc, seqnum);
|
|
rtp_source_register_nack (source, seqnum);
|
|
|
|
rtp_session_send_rtcp (sess, max_delay);
|
|
|
|
return TRUE;
|
|
}
|