mirrors/gstreamer
mirrors/gstreamer
1
1
Fork 0
mirror of https://gitlab.freedesktop.org/gstreamer/gstreamer.git synced 2024-12-30 04:00:37 +00:00
Code Issues 1 Projects Releases Wiki Activity
gstreamer/gst/rtpmanager/rtpsession.c
Miguel París Díaz 3aa69ca0bb rtpsession: relate received FIRs and PLIs to source 
This is needed in order to:
 - Avoid ignoring requests for different media sources.
 - Add SSRC field in the GstForceKeyUnit event.

https://bugzilla.gnome.org/show_bug.cgi?id=778013
2017-02-02 12:13:59 -05:00

4392 lines
130 KiB
C
Raw Blame History

/* GStreamer
* Copyright (C) <2007> Wim Taymans <wim.taymans@gmail.com>
*
* This library is free software; you can redistribute it and/or
* modify it under the terms of the GNU Library General Public
* License as published by the Free Software Foundation; either
* version 2 of the License, or (at your option) any later version.
*
* This library is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Library General Public License for more details.
*
* You should have received a copy of the GNU Library General Public
* License along with this library; if not, write to the
* Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
* Boston, MA 02110-1301, USA.
*/
/* FIXME 0.11: suppress warnings for deprecated API such as GValueArray
* with newer GLib versions (>= 2.31.0) */
#define GLIB_DISABLE_DEPRECATION_WARNINGS
#include <string.h>
#include <gst/rtp/gstrtpbuffer.h>
#include <gst/rtp/gstrtcpbuffer.h>
#include <gst/glib-compat-private.h>
#include "rtpsession.h"
GST_DEBUG_CATEGORY_STATIC (rtp_session_debug);
#define GST_CAT_DEFAULT rtp_session_debug
/* signals and args */
enum
{
SIGNAL_GET_SOURCE_BY_SSRC,
SIGNAL_ON_NEW_SSRC,
SIGNAL_ON_SSRC_COLLISION,
SIGNAL_ON_SSRC_VALIDATED,
SIGNAL_ON_SSRC_ACTIVE,
SIGNAL_ON_SSRC_SDES,
SIGNAL_ON_BYE_SSRC,
SIGNAL_ON_BYE_TIMEOUT,
SIGNAL_ON_TIMEOUT,
SIGNAL_ON_SENDER_TIMEOUT,
SIGNAL_ON_SENDING_RTCP,
SIGNAL_ON_APP_RTCP,
SIGNAL_ON_FEEDBACK_RTCP,
SIGNAL_SEND_RTCP,
SIGNAL_SEND_RTCP_FULL,
SIGNAL_ON_RECEIVING_RTCP,
SIGNAL_ON_NEW_SENDER_SSRC,
SIGNAL_ON_SENDER_SSRC_ACTIVE,
LAST_SIGNAL
};
#define DEFAULT_INTERNAL_SOURCE NULL
#define DEFAULT_BANDWIDTH 0.0
#define DEFAULT_RTCP_FRACTION RTP_STATS_RTCP_FRACTION
#define DEFAULT_RTCP_RR_BANDWIDTH -1
#define DEFAULT_RTCP_RS_BANDWIDTH -1
#define DEFAULT_RTCP_MTU 1400
#define DEFAULT_SDES NULL
#define DEFAULT_NUM_SOURCES 0
#define DEFAULT_NUM_ACTIVE_SOURCES 0
#define DEFAULT_SOURCES NULL
#define DEFAULT_RTCP_MIN_INTERVAL (RTP_STATS_MIN_INTERVAL * GST_SECOND)
#define DEFAULT_RTCP_FEEDBACK_RETENTION_WINDOW (2 * GST_SECOND)
#define DEFAULT_RTCP_IMMEDIATE_FEEDBACK_THRESHOLD (3)
#define DEFAULT_PROBATION RTP_DEFAULT_PROBATION
#define DEFAULT_MAX_DROPOUT_TIME 60000
#define DEFAULT_MAX_MISORDER_TIME 2000
#define DEFAULT_RTP_PROFILE GST_RTP_PROFILE_AVP
#define DEFAULT_RTCP_REDUCED_SIZE FALSE
enum
{
PROP_0,
PROP_INTERNAL_SSRC,
PROP_INTERNAL_SOURCE,
PROP_BANDWIDTH,
PROP_RTCP_FRACTION,
PROP_RTCP_RR_BANDWIDTH,
PROP_RTCP_RS_BANDWIDTH,
PROP_RTCP_MTU,
PROP_SDES,
PROP_NUM_SOURCES,
PROP_NUM_ACTIVE_SOURCES,
PROP_SOURCES,
PROP_FAVOR_NEW,
PROP_RTCP_MIN_INTERVAL,
PROP_RTCP_FEEDBACK_RETENTION_WINDOW,
PROP_RTCP_IMMEDIATE_FEEDBACK_THRESHOLD,
PROP_PROBATION,
PROP_MAX_DROPOUT_TIME,
PROP_MAX_MISORDER_TIME,
PROP_STATS,
PROP_RTP_PROFILE,
PROP_RTCP_REDUCED_SIZE
};
/* update average packet size */
#define INIT_AVG(avg, val) \
(avg) = (val);
#define UPDATE_AVG(avg, val) \
if ((avg) == 0) \
(avg) = (val); \
else \
(avg) = ((val) + (15 * (avg))) >> 4;
/* GObject vmethods */
static void rtp_session_finalize (GObject * object);
static void rtp_session_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec);
static void rtp_session_get_property (GObject * object, guint prop_id,
GValue * value, GParamSpec * pspec);
static gboolean rtp_session_send_rtcp (RTPSession * sess,
GstClockTime max_delay);
static guint rtp_session_signals[LAST_SIGNAL] = { 0 };
G_DEFINE_TYPE (RTPSession, rtp_session, G_TYPE_OBJECT);
static guint32 rtp_session_create_new_ssrc (RTPSession * sess);
static RTPSource *obtain_source (RTPSession * sess, guint32 ssrc,
gboolean * created, RTPPacketInfo * pinfo, gboolean rtp);
static RTPSource *obtain_internal_source (RTPSession * sess,
guint32 ssrc, gboolean * created, GstClockTime current_time);
static GstFlowReturn rtp_session_schedule_bye_locked (RTPSession * sess,
GstClockTime current_time);
static GstClockTime calculate_rtcp_interval (RTPSession * sess,
gboolean deterministic, gboolean first);
static gboolean
accumulate_trues (GSignalInvocationHint * ihint, GValue * return_accu,
const GValue * handler_return, gpointer data)
{
if (g_value_get_boolean (handler_return))
g_value_set_boolean (return_accu, TRUE);
return TRUE;
}
static void
rtp_session_class_init (RTPSessionClass * klass)
{
GObjectClass *gobject_class;
gobject_class = (GObjectClass *) klass;
gobject_class->finalize = rtp_session_finalize;
gobject_class->set_property = rtp_session_set_property;
gobject_class->get_property = rtp_session_get_property;
/**
* RTPSession::get-source-by-ssrc:
* @session: the object which received the signal
* @ssrc: the SSRC of the RTPSource
*
* Request the #RTPSource object with SSRC @ssrc in @session.
*/
rtp_session_signals[SIGNAL_GET_SOURCE_BY_SSRC] =
g_signal_new ("get-source-by-ssrc", G_TYPE_FROM_CLASS (klass),
G_SIGNAL_RUN_LAST | G_SIGNAL_ACTION, G_STRUCT_OFFSET (RTPSessionClass,
get_source_by_ssrc), NULL, NULL, g_cclosure_marshal_generic,
RTP_TYPE_SOURCE, 1, G_TYPE_UINT);
/**
* RTPSession::on-new-ssrc:
* @session: the object which received the signal
* @src: the new RTPSource
*
* Notify of a new SSRC that entered @session.
*/
rtp_session_signals[SIGNAL_ON_NEW_SSRC] =
g_signal_new ("on-new-ssrc", G_TYPE_FROM_CLASS (klass),
G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (RTPSessionClass, on_new_ssrc),
NULL, NULL, g_cclosure_marshal_VOID__OBJECT, G_TYPE_NONE, 1,
RTP_TYPE_SOURCE);
/**
* RTPSession::on-ssrc-collision:
* @session: the object which received the signal
* @src: the #RTPSource that caused a collision
*
* Notify when we have an SSRC collision
*/
rtp_session_signals[SIGNAL_ON_SSRC_COLLISION] =
g_signal_new ("on-ssrc-collision", G_TYPE_FROM_CLASS (klass),
G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (RTPSessionClass, on_ssrc_collision),
NULL, NULL, g_cclosure_marshal_VOID__OBJECT, G_TYPE_NONE, 1,
RTP_TYPE_SOURCE);
/**
* RTPSession::on-ssrc-validated:
* @session: the object which received the signal
* @src: the new validated RTPSource
*
* Notify of a new SSRC that became validated.
*/
rtp_session_signals[SIGNAL_ON_SSRC_VALIDATED] =
g_signal_new ("on-ssrc-validated", G_TYPE_FROM_CLASS (klass),
G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (RTPSessionClass, on_ssrc_validated),
NULL, NULL, g_cclosure_marshal_VOID__OBJECT, G_TYPE_NONE, 1,
RTP_TYPE_SOURCE);
/**
* RTPSession::on-ssrc-active:
* @session: the object which received the signal
* @src: the active RTPSource
*
* Notify of a SSRC that is active, i.e., sending RTCP.
*/
rtp_session_signals[SIGNAL_ON_SSRC_ACTIVE] =
g_signal_new ("on-ssrc-active", G_TYPE_FROM_CLASS (klass),
G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (RTPSessionClass, on_ssrc_active),
NULL, NULL, g_cclosure_marshal_VOID__OBJECT, G_TYPE_NONE, 1,
RTP_TYPE_SOURCE);
/**
* RTPSession::on-ssrc-sdes:
* @session: the object which received the signal
* @src: the RTPSource
*
* Notify that a new SDES was received for SSRC.
*/
rtp_session_signals[SIGNAL_ON_SSRC_SDES] =
g_signal_new ("on-ssrc-sdes", G_TYPE_FROM_CLASS (klass),
G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (RTPSessionClass, on_ssrc_sdes),
NULL, NULL, g_cclosure_marshal_VOID__OBJECT, G_TYPE_NONE, 1,
RTP_TYPE_SOURCE);
/**
* RTPSession::on-bye-ssrc:
* @session: the object which received the signal
* @src: the RTPSource that went away
*
* Notify of an SSRC that became inactive because of a BYE packet.
*/
rtp_session_signals[SIGNAL_ON_BYE_SSRC] =
g_signal_new ("on-bye-ssrc", G_TYPE_FROM_CLASS (klass),
G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (RTPSessionClass, on_bye_ssrc),
NULL, NULL, g_cclosure_marshal_VOID__OBJECT, G_TYPE_NONE, 1,
RTP_TYPE_SOURCE);
/**
* RTPSession::on-bye-timeout:
* @session: the object which received the signal
* @src: the RTPSource that timed out
*
* Notify of an SSRC that has timed out because of BYE
*/
rtp_session_signals[SIGNAL_ON_BYE_TIMEOUT] =
g_signal_new ("on-bye-timeout", G_TYPE_FROM_CLASS (klass),
G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (RTPSessionClass, on_bye_timeout),
NULL, NULL, g_cclosure_marshal_VOID__OBJECT, G_TYPE_NONE, 1,
RTP_TYPE_SOURCE);
/**
* RTPSession::on-timeout:
* @session: the object which received the signal
* @src: the RTPSource that timed out
*
* Notify of an SSRC that has timed out
*/
rtp_session_signals[SIGNAL_ON_TIMEOUT] =
g_signal_new ("on-timeout", G_TYPE_FROM_CLASS (klass),
G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (RTPSessionClass, on_timeout),
NULL, NULL, g_cclosure_marshal_VOID__OBJECT, G_TYPE_NONE, 1,
RTP_TYPE_SOURCE);
/**
* RTPSession::on-sender-timeout:
* @session: the object which received the signal
* @src: the RTPSource that timed out
*
* Notify of an SSRC that was a sender but timed out and became a receiver.
*/
rtp_session_signals[SIGNAL_ON_SENDER_TIMEOUT] =
g_signal_new ("on-sender-timeout", G_TYPE_FROM_CLASS (klass),
G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (RTPSessionClass, on_sender_timeout),
NULL, NULL, g_cclosure_marshal_VOID__OBJECT, G_TYPE_NONE, 1,
RTP_TYPE_SOURCE);
/**
* RTPSession::on-sending-rtcp
* @session: the object which received the signal
* @buffer: the #GstBuffer containing the RTCP packet about to be sent
* @early: %TRUE if the packet is early, %FALSE if it is regular
*
* This signal is emitted before sending an RTCP packet, it can be used
* to add extra RTCP Packets.
*
* Returns: %TRUE if the RTCP buffer should NOT be suppressed, %FALSE
* if suppressing it is acceptable
*/
rtp_session_signals[SIGNAL_ON_SENDING_RTCP] =
g_signal_new ("on-sending-rtcp", G_TYPE_FROM_CLASS (klass),
G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (RTPSessionClass, on_sending_rtcp),
accumulate_trues, NULL, g_cclosure_marshal_generic, G_TYPE_BOOLEAN, 2,
GST_TYPE_BUFFER | G_SIGNAL_TYPE_STATIC_SCOPE, G_TYPE_BOOLEAN);
/**
* RTPSession::on-app-rtcp:
* @session: the object which received the signal
* @subtype: The subtype of the packet
* @ssrc: The SSRC/CSRC of the packet
* @name: The name of the packet
* @data: a #GstBuffer with the application-dependant data or %NULL if
* there was no data
*
* Notify that a RTCP APP packet has been received
*/
rtp_session_signals[SIGNAL_ON_APP_RTCP] =
g_signal_new ("on-app-rtcp", G_TYPE_FROM_CLASS (klass),
G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (RTPSessionClass, on_app_rtcp),
NULL, NULL, g_cclosure_marshal_generic, G_TYPE_NONE, 4,
G_TYPE_UINT, G_TYPE_UINT, G_TYPE_STRING, GST_TYPE_BUFFER);
/**
* RTPSession::on-feedback-rtcp:
* @session: the object which received the signal
* @type: Type of RTCP packet, will be %GST_RTCP_TYPE_RTPFB or
* %GST_RTCP_TYPE_RTPFB
* @fbtype: The type of RTCP FB packet, probably part of #GstRTCPFBType
* @sender_ssrc: The SSRC of the sender
* @media_ssrc: The SSRC of the media this refers to
* @fci: a #GstBuffer with the FCI data from the FB packet or %NULL if
* there was no FCI
*
* Notify that a RTCP feedback packet has been received
*/
rtp_session_signals[SIGNAL_ON_FEEDBACK_RTCP] =
g_signal_new ("on-feedback-rtcp", G_TYPE_FROM_CLASS (klass),
G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (RTPSessionClass, on_feedback_rtcp),
NULL, NULL, g_cclosure_marshal_generic, G_TYPE_NONE, 5, G_TYPE_UINT,
G_TYPE_UINT, G_TYPE_UINT, G_TYPE_UINT, GST_TYPE_BUFFER);
/**
* RTPSession::send-rtcp:
* @session: the object which received the signal
* @max_delay: The maximum delay after which the feedback will not be useful
* anymore
*
* Requests that the #RTPSession initiate a new RTCP packet as soon as
* possible within the requested delay.
*
* This sets feedback to %TRUE if not already done before.
*/
rtp_session_signals[SIGNAL_SEND_RTCP] =
g_signal_new ("send-rtcp", G_TYPE_FROM_CLASS (klass),
G_SIGNAL_RUN_LAST | G_SIGNAL_ACTION,
G_STRUCT_OFFSET (RTPSessionClass, send_rtcp), NULL, NULL,
g_cclosure_marshal_generic, G_TYPE_NONE, 1, G_TYPE_UINT64);
/**
* RTPSession::send-rtcp-full:
* @session: the object which received the signal
* @max_delay: The maximum delay after which the feedback will not be useful
* anymore
*
* Requests that the #RTPSession initiate a new RTCP packet as soon as
* possible within the requested delay.
*
* This sets feedback to %TRUE if not already done before.
*
* Returns: TRUE if the new RTCP packet could be scheduled within the
* requested delay, FALSE otherwise.
*
* Since: 1.6
*/
rtp_session_signals[SIGNAL_SEND_RTCP_FULL] =
g_signal_new ("send-rtcp-full", G_TYPE_FROM_CLASS (klass),
G_SIGNAL_RUN_LAST | G_SIGNAL_ACTION,
G_STRUCT_OFFSET (RTPSessionClass, send_rtcp), NULL, NULL,
g_cclosure_marshal_generic, G_TYPE_BOOLEAN, 1, G_TYPE_UINT64);
/**
* RTPSession::on-receiving-rtcp
* @session: the object which received the signal
* @buffer: the #GstBuffer containing the RTCP packet that was received
*
* This signal is emitted when receiving an RTCP packet before it is handled
* by the session. It can be used to extract custom information from RTCP packets.
*
* Since: 1.6
*/
rtp_session_signals[SIGNAL_ON_RECEIVING_RTCP] =
g_signal_new ("on-receiving-rtcp", G_TYPE_FROM_CLASS (klass),
G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (RTPSessionClass, on_receiving_rtcp),
NULL, NULL, g_cclosure_marshal_generic, G_TYPE_NONE, 1,
GST_TYPE_BUFFER | G_SIGNAL_TYPE_STATIC_SCOPE);
/**
* RTPSession::on-new-sender-ssrc:
* @session: the object which received the signal
* @src: the new sender RTPSource
*
* Notify of a new sender SSRC that entered @session.
*
* Since: 1.8
*/
rtp_session_signals[SIGNAL_ON_NEW_SENDER_SSRC] =
g_signal_new ("on-new-sender-ssrc", G_TYPE_FROM_CLASS (klass),
G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (RTPSessionClass, on_new_sender_ssrc),
NULL, NULL, g_cclosure_marshal_VOID__OBJECT, G_TYPE_NONE, 1,
RTP_TYPE_SOURCE);
/**
* RTPSession::on-sender-ssrc-active:
* @session: the object which received the signal
* @src: the active sender RTPSource
*
* Notify of a sender SSRC that is active, i.e., sending RTCP.
*
* Since: 1.8
*/
rtp_session_signals[SIGNAL_ON_SENDER_SSRC_ACTIVE] =
g_signal_new ("on-sender-ssrc-active", G_TYPE_FROM_CLASS (klass),
G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (RTPSessionClass,
on_sender_ssrc_active), NULL, NULL, g_cclosure_marshal_VOID__OBJECT,
G_TYPE_NONE, 1, RTP_TYPE_SOURCE);
g_object_class_install_property (gobject_class, PROP_INTERNAL_SSRC,
g_param_spec_uint ("internal-ssrc", "Internal SSRC",
"The internal SSRC used for the session (deprecated)",
0, G_MAXUINT, 0, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (gobject_class, PROP_INTERNAL_SOURCE,
g_param_spec_object ("internal-source", "Internal Source",
"The internal source element of the session (deprecated)",
RTP_TYPE_SOURCE, G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (gobject_class, PROP_BANDWIDTH,
g_param_spec_double ("bandwidth", "Bandwidth",
"The bandwidth of the session in bits per second (0 for auto-discover)",
0.0, G_MAXDOUBLE, DEFAULT_BANDWIDTH,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (gobject_class, PROP_RTCP_FRACTION,
g_param_spec_double ("rtcp-fraction", "RTCP Fraction",
"The fraction of the bandwidth used for RTCP in bits per second (or as a real fraction of the RTP bandwidth if < 1)",
0.0, G_MAXDOUBLE, DEFAULT_RTCP_FRACTION,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (gobject_class, PROP_RTCP_RR_BANDWIDTH,
g_param_spec_int ("rtcp-rr-bandwidth", "RTCP RR bandwidth",
"The RTCP bandwidth used for receivers in bits per second (-1 = default)",
-1, G_MAXINT, DEFAULT_RTCP_RR_BANDWIDTH,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (gobject_class, PROP_RTCP_RS_BANDWIDTH,
g_param_spec_int ("rtcp-rs-bandwidth", "RTCP RS bandwidth",
"The RTCP bandwidth used for senders in bits per second (-1 = default)",
-1, G_MAXINT, DEFAULT_RTCP_RS_BANDWIDTH,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (gobject_class, PROP_RTCP_MTU,
g_param_spec_uint ("rtcp-mtu", "RTCP MTU",
"The maximum size of the RTCP packets",
16, G_MAXINT16, DEFAULT_RTCP_MTU,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (gobject_class, PROP_SDES,
g_param_spec_boxed ("sdes", "SDES",
"The SDES items of this session",
GST_TYPE_STRUCTURE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (gobject_class, PROP_NUM_SOURCES,
g_param_spec_uint ("num-sources", "Num Sources",
"The number of sources in the session", 0, G_MAXUINT,
DEFAULT_NUM_SOURCES, G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (gobject_class, PROP_NUM_ACTIVE_SOURCES,
g_param_spec_uint ("num-active-sources", "Num Active Sources",
"The number of active sources in the session", 0, G_MAXUINT,
DEFAULT_NUM_ACTIVE_SOURCES,
G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
/**
* RTPSource::sources
*
* Get a GValue Array of all sources in the session.
*
* <example>
* <title>Getting the #RTPSources of a session
* <programlisting>
* {
* GValueArray *arr;
* GValue *val;
* guint i;
*
* g_object_get (sess, "sources", &arr, NULL);
*
* for (i = 0; i < arr->n_values; i++) {
* RTPSource *source;
*
* val = g_value_array_get_nth (arr, i);
* source = g_value_get_object (val);
* }
* g_value_array_free (arr);
* }
* </programlisting>
* </example>
*/
g_object_class_install_property (gobject_class, PROP_SOURCES,
g_param_spec_boxed ("sources", "Sources",
"An array of all known sources in the session",
G_TYPE_VALUE_ARRAY, G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (gobject_class, PROP_FAVOR_NEW,
g_param_spec_boolean ("favor-new", "Favor new sources",
"Resolve SSRC conflict in favor of new sources", FALSE,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (gobject_class, PROP_RTCP_MIN_INTERVAL,
g_param_spec_uint64 ("rtcp-min-interval", "Minimum RTCP interval",
"Minimum interval between Regular RTCP packet (in ns)",
0, G_MAXUINT64, DEFAULT_RTCP_MIN_INTERVAL,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (gobject_class,
PROP_RTCP_FEEDBACK_RETENTION_WINDOW,
g_param_spec_uint64 ("rtcp-feedback-retention-window",
"RTCP Feedback retention window",
"Duration during which RTCP Feedback packets are retained (in ns)",
0, G_MAXUINT64, DEFAULT_RTCP_FEEDBACK_RETENTION_WINDOW,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (gobject_class,
PROP_RTCP_IMMEDIATE_FEEDBACK_THRESHOLD,
g_param_spec_uint ("rtcp-immediate-feedback-threshold",
"RTCP Immediate Feedback threshold",
"The maximum number of members of a RTP session for which immediate"
" feedback is used (DEPRECATED: has no effect and is not needed)",
0, G_MAXUINT, DEFAULT_RTCP_IMMEDIATE_FEEDBACK_THRESHOLD,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS | G_PARAM_DEPRECATED));
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));
g_object_class_install_property (gobject_class, PROP_MAX_DROPOUT_TIME,
g_param_spec_uint ("max-dropout-time", "Max dropout time",
"The maximum time (milliseconds) of missing packets tolerated.",
0, G_MAXUINT, DEFAULT_MAX_DROPOUT_TIME,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (gobject_class, PROP_MAX_MISORDER_TIME,
g_param_spec_uint ("max-misorder-time", "Max misorder time",
"The maximum time (milliseconds) of misordered packets tolerated.",
0, G_MAXUINT, DEFAULT_MAX_MISORDER_TIME,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
/**
* RTPSession::stats:
*
* Various session statistics. This property returns a GstStructure
* with name application/x-rtp-session-stats with the following fields:
*
* "rtx-drop-count" G_TYPE_UINT The number of retransmission events
* dropped (due to bandwidth constraints)
* "sent-nack-count" G_TYPE_UINT Number of NACKs sent
* "recv-nack-count" G_TYPE_UINT Number of NACKs received
* "source-stats" G_TYPE_BOXED GValueArray of #RTPSource::stats for all
* RTP sources (Since 1.8)
*
* Since: 1.4
*/
g_object_class_install_property (gobject_class, PROP_STATS,
g_param_spec_boxed ("stats", "Statistics",
"Various statistics", GST_TYPE_STRUCTURE,
G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (gobject_class, PROP_RTP_PROFILE,
g_param_spec_enum ("rtp-profile", "RTP Profile",
"RTP profile to use for this session", GST_TYPE_RTP_PROFILE,
DEFAULT_RTP_PROFILE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (gobject_class, PROP_RTCP_REDUCED_SIZE,
g_param_spec_boolean ("rtcp-reduced-size", "RTCP Reduced Size",
"Use Reduced Size RTCP for feedback packets",
DEFAULT_RTCP_REDUCED_SIZE,
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;
/* TODO: We currently only use the first hash table but this is the
* beginning of an implementation for RFC2762
for (i = 0; i < 32; i++) {
*/
for (i = 0; i < 1; 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;
sess->max_dropout_time = DEFAULT_MAX_DROPOUT_TIME;
sess->max_misorder_time = DEFAULT_MAX_MISORDER_TIME;
/* 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->internal_ssrc_set = FALSE;
sess->first_rtcp = TRUE;
sess->next_rtcp_check_time = GST_CLOCK_TIME_NONE;
sess->last_rtcp_check_time = GST_CLOCK_TIME_NONE;
sess->last_rtcp_send_time = GST_CLOCK_TIME_NONE;
sess->last_rtcp_interval = GST_CLOCK_TIME_NONE;
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->rtp_profile = DEFAULT_RTP_PROFILE;
sess->reduced_size_rtcp = DEFAULT_RTCP_REDUCED_SIZE;
sess->is_doing_ptp = TRUE;
}
static void
rtp_session_finalize (GObject * object)
{
RTPSession *sess;
gint i;
sess = RTP_SESSION_CAST (object);
gst_structure_free (sess->sdes);
g_list_free_full (sess->conflicting_addresses,
(GDestroyNotify) rtp_conflicting_address_free);
/* TODO: Change this again when implementing RFC 2762
* for (i = 0; i < 32; i++)
*/
for (i = 0; i < 1; 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
create_source_stats (gpointer key, RTPSource * source, GValueArray * arr)
{
GValue value = G_VALUE_INIT;
GstStructure *s;
g_object_get (source, "stats", &s, NULL);
g_value_init (&value, GST_TYPE_STRUCTURE);
gst_value_set_structure (&value, s);
g_value_array_append (arr, &value);
gst_structure_free (s);
g_value_unset (&value);
}
static GstStructure *
rtp_session_create_stats (RTPSession * sess)
{
GstStructure *s;
GValueArray *source_stats;
GValue source_stats_v = G_VALUE_INIT;
guint size;
RTP_SESSION_LOCK (sess);
s = gst_structure_new ("application/x-rtp-session-stats",
"rtx-drop-count", G_TYPE_UINT, sess->stats.nacks_dropped,
"sent-nack-count", G_TYPE_UINT, sess->stats.nacks_sent,
"recv-nack-count", G_TYPE_UINT, sess->stats.nacks_received, NULL);
size = g_hash_table_size (sess->ssrcs[sess->mask_idx]);
source_stats = g_value_array_new (size);
g_hash_table_foreach (sess->ssrcs[sess->mask_idx],
(GHFunc) create_source_stats, source_stats);
RTP_SESSION_UNLOCK (sess);
g_value_init (&source_stats_v, G_TYPE_VALUE_ARRAY);
g_value_take_boxed (&source_stats_v, source_stats);
gst_structure_take_value (s, "source-stats", &source_stats_v);
return s;
}
static void
rtp_session_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec)
{
RTPSession *sess;
sess = RTP_SESSION (object);
switch (prop_id) {
case PROP_INTERNAL_SSRC:
RTP_SESSION_LOCK (sess);
sess->suggested_ssrc = g_value_get_uint (value);
sess->internal_ssrc_set = TRUE;
sess->internal_ssrc_from_caps_or_property = TRUE;
RTP_SESSION_UNLOCK (sess);
if (sess->callbacks.reconfigure)
sess->callbacks.reconfigure (sess, sess->reconfigure_user_data);
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;
case PROP_MAX_DROPOUT_TIME:
sess->max_dropout_time = g_value_get_uint (value);
break;
case PROP_MAX_MISORDER_TIME:
sess->max_misorder_time = g_value_get_uint (value);
break;
case PROP_RTP_PROFILE:
sess->rtp_profile = g_value_get_enum (value);
/* 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_REDUCED_SIZE:
sess->reduced_size_rtcp = g_value_get_boolean (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, NULL));
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;
case PROP_MAX_DROPOUT_TIME:
g_value_set_uint (value, sess->max_dropout_time);
break;
case PROP_MAX_MISORDER_TIME:
g_value_set_uint (value, sess->max_misorder_time);
break;
case PROP_STATS:
g_value_take_boxed (value, rtp_session_create_stats (sess));
break;
case PROP_RTP_PROFILE:
g_value_set_enum (value, sess->rtp_profile);
break;
case PROP_RTCP_REDUCED_SIZE:
g_value_set_boolean (value, sess->reduced_size_rtcp);
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);
}
static void
on_new_sender_ssrc (RTPSession * sess, RTPSource * source)
{
g_object_ref (source);
RTP_SESSION_UNLOCK (sess);
g_signal_emit (sess, rtp_session_signals[SIGNAL_ON_NEW_SENDER_SSRC], 0,
source);
RTP_SESSION_LOCK (sess);
g_object_unref (source);
}
static void
on_sender_ssrc_active (RTPSession * sess, RTPSource * source)
{
g_object_ref (source);
RTP_SESSION_UNLOCK (sess);
g_signal_emit (sess, rtp_session_signals[SIGNAL_ON_SENDER_SSRC_ACTIVE], 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;
}
if (callbacks->notify_nack) {
sess->callbacks.notify_nack = callbacks->notify_nack;
sess->notify_nack_user_data = user_data;
}
if (callbacks->reconfigure) {
sess->callbacks.reconfigure = callbacks->reconfigure;
sess->reconfigure_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 bits 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 bits 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,
};
/**
* rtp_session_find_conflicting_address:
* @session: The session the packet came in
* @address: address to check for
* @time: The time when the packet that is possibly in conflict arrived
*
* Checks if an address which has a conflict is already known. If it is
* a known conflict, remember the time
*
* Returns: TRUE if it was a known conflict, FALSE otherwise
*/
static gboolean
rtp_session_find_conflicting_address (RTPSession * session,
GSocketAddress * address, GstClockTime time)
{
return find_conflicting_address (session->conflicting_addresses, address,
time);
}
/**
* rtp_session_add_conflicting_address:
* @session: The session the packet came in
* @address: address to remember
* @time: The time when the packet that is in conflict arrived
*
* Adds a new conflict address
*/
static void
rtp_session_add_conflicting_address (RTPSession * sess,
GSocketAddress * address, GstClockTime time)
{
sess->conflicting_addresses =
add_conflicting_address (sess->conflicting_addresses, address, time);
}
static gboolean
check_collision (RTPSession * sess, RTPSource * source,
RTPPacketInfo * pinfo, gboolean rtp)
{
guint32 ssrc;
/* If we have no pinfo address, we can't do collision checking */
if (!pinfo->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, pinfo->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,
pinfo->address, pinfo->current_time)) {
gchar *buf1;
buf1 = __g_socket_address_to_string (pinfo->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,
pinfo->current_time);
buf1 = __g_socket_address_to_string (from);
buf2 = __g_socket_address_to_string (pinfo->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, pinfo->address);
else
rtp_source_set_rtcp_from (source, pinfo->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, pinfo->address);
else
rtp_source_set_rtcp_from (source, pinfo->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_session_find_conflicting_address (sess, pinfo->address,
pinfo->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_session_add_conflicting_address (sess, pinfo->address,
pinfo->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);
sess->internal_ssrc_set = TRUE;
}
on_ssrc_collision (sess, source);
rtp_session_schedule_bye_locked (sess, pinfo->current_time);
}
}
return TRUE;
}
typedef struct
{
gboolean is_doing_ptp;
GSocketAddress *new_addr;
} CompareAddrData;
/* check if the two given ip addr are the same (do not care about the port) */
static gboolean
ip_addr_equal (GSocketAddress * a, GSocketAddress * b)
{
return
g_inet_address_equal (g_inet_socket_address_get_address
(G_INET_SOCKET_ADDRESS (a)),
g_inet_socket_address_get_address (G_INET_SOCKET_ADDRESS (b)));
}
static void
compare_rtp_source_addr (const gchar * key, RTPSource * source,
CompareAddrData * data)
{
/* only compare ip addr of remote sources which are also not closing */
if (!source->internal && !source->closing && source->rtp_from) {
/* look for the first rtp source */
if (!data->new_addr)
data->new_addr = source->rtp_from;
/* compare current ip addr with the first one */
else
data->is_doing_ptp &= ip_addr_equal (data->new_addr, source->rtp_from);
}
}
static void
compare_rtcp_source_addr (const gchar * key, RTPSource * source,
CompareAddrData * data)
{
/* only compare ip addr of remote sources which are also not closing */
if (!source->internal && !source->closing && source->rtcp_from) {
/* look for the first rtcp source */
if (!data->new_addr)
data->new_addr = source->rtcp_from;
else
/* compare current ip addr with the first one */
data->is_doing_ptp &= ip_addr_equal (data->new_addr, source->rtcp_from);
}
}
/* loop over our non-internal source to know if the session
* is doing point-to-point */
static void
session_update_ptp (RTPSession * sess)
{
/* to know if the session is doing point to point, the ip addr
* of each non-internal (=remotes) source have to be compared
* to each other.
*/
gboolean is_doing_rtp_ptp;
gboolean is_doing_rtcp_ptp;
CompareAddrData data;
/* compare the first remote source's ip addr that receive rtp packets
* with other remote rtp source.
* it's enough because the session just needs to know if they are all
* equals or not
*/
data.is_doing_ptp = TRUE;
data.new_addr = NULL;
g_hash_table_foreach (sess->ssrcs[sess->mask_idx],
(GHFunc) compare_rtp_source_addr, (gpointer) & data);
is_doing_rtp_ptp = data.is_doing_ptp;
/* same but about rtcp */
data.is_doing_ptp = TRUE;
data.new_addr = NULL;
g_hash_table_foreach (sess->ssrcs[sess->mask_idx],
(GHFunc) compare_rtcp_source_addr, (gpointer) & data);
is_doing_rtcp_ptp = data.is_doing_ptp;
/* the session is doing point-to-point if all rtp remote have the same
* ip addr and if all rtcp remote sources have the same ip addr */
sess->is_doing_ptp = is_doing_rtp_ptp && is_doing_rtcp_ptp;
GST_DEBUG ("doing point-to-point: %d", sess->is_doing_ptp);
}
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->internal_ssrc_from_caps_or_property
&& sess->suggested_ssrc != src->ssrc) {
sess->suggested_ssrc = src->ssrc;
sess->internal_ssrc_set = TRUE;
}
}
/* update point-to-point status */
if (!src->internal)
session_update_ptp (sess);
}
static RTPSource *
find_source (RTPSession * sess, guint32 ssrc)
{
return g_hash_table_lookup (sess->ssrcs[sess->mask_idx],
GINT_TO_POINTER (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,
RTPPacketInfo * pinfo, 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. */
g_object_set (source, "probation", rtp ? sess->probation : 0,
"max-dropout-time", sess->max_dropout_time, "max-misorder-time",
sess->max_misorder_time, NULL);
/* store from address, if any */
if (pinfo->address) {
if (rtp)
rtp_source_set_rtp_from (source, pinfo->address);
else
rtp_source_set_rtcp_from (source, pinfo->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, pinfo, 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 = pinfo->current_time;
if (rtp)
source->last_rtp_activity = pinfo->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,
GstClockTime current_time)
{
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;
source->probation = FALSE;
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;
}
/* update last activity */
if (current_time != GST_CLOCK_TIME_NONE) {
source->last_activity = current_time;
source->last_rtp_activity = current_time;
}
g_object_ref (source);
return source;
}
/**
* rtp_session_suggest_ssrc:
* @sess: a #RTPSession
* @is_random: if the suggested ssrc is random
*
* Suggest an unused SSRC in @sess.
*
* Returns: a free unused SSRC
*/
guint32
rtp_session_suggest_ssrc (RTPSession * sess, gboolean * is_random)
{
guint32 result;
g_return_val_if_fail (RTP_IS_SESSION (sess), 0);
RTP_SESSION_LOCK (sess);
result = sess->suggested_ssrc;
if (is_random)
*is_random = !sess->internal_ssrc_set;
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 != NULL)
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;
}
static gboolean
update_packet (GstBuffer ** buffer, guint idx, RTPPacketInfo * pinfo)
{
GstNetAddressMeta *meta;
/* get packet size including header overhead */
pinfo->bytes += gst_buffer_get_size (*buffer) + pinfo->header_len;
pinfo->packets++;
if (pinfo->rtp) {
GstRTPBuffer rtp = { NULL };
if (!gst_rtp_buffer_map (*buffer, GST_MAP_READ, &rtp))
goto invalid_packet;
pinfo->payload_len += gst_rtp_buffer_get_payload_len (&rtp);
if (idx == 0) {
gint i;
/* only keep info for first buffer */
pinfo->ssrc = gst_rtp_buffer_get_ssrc (&rtp);
pinfo->seqnum = gst_rtp_buffer_get_seq (&rtp);
pinfo->pt = gst_rtp_buffer_get_payload_type (&rtp);
pinfo->rtptime = gst_rtp_buffer_get_timestamp (&rtp);
/* copy available csrc */
pinfo->csrc_count = gst_rtp_buffer_get_csrc_count (&rtp);
for (i = 0; i < pinfo->csrc_count; i++)
pinfo->csrcs[i] = gst_rtp_buffer_get_csrc (&rtp, i);
}
gst_rtp_buffer_unmap (&rtp);
}
if (idx == 0) {
/* for netbuffer we can store the IP address to check for collisions */
meta = gst_buffer_get_net_address_meta (*buffer);
if (pinfo->address)
g_object_unref (pinfo->address);
if (meta) {
pinfo->address = G_SOCKET_ADDRESS (g_object_ref (meta->addr));
} else {
pinfo->address = NULL;
}
}
return TRUE;
/* ERRORS */
invalid_packet:
{
GST_DEBUG ("invalid RTP packet received");
return FALSE;
}
}
/* update the RTPPacketInfo 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 gboolean
update_packet_info (RTPSession * sess, RTPPacketInfo * pinfo,
gboolean send, gboolean rtp, gboolean is_list, gpointer data,
GstClockTime current_time, GstClockTime running_time, guint64 ntpnstime)
{
gboolean res;
pinfo->send = send;
pinfo->rtp = rtp;
pinfo->is_list = is_list;
pinfo->data = data;
pinfo->current_time = current_time;
pinfo->running_time = running_time;
pinfo->ntpnstime = ntpnstime;
pinfo->header_len = sess->header_len;
pinfo->bytes = 0;
pinfo->payload_len = 0;
pinfo->packets = 0;
if (is_list) {
GstBufferList *list = GST_BUFFER_LIST_CAST (data);
res =
gst_buffer_list_foreach (list, (GstBufferListFunc) update_packet,
pinfo);
} else {
GstBuffer *buffer = GST_BUFFER_CAST (data);
res = update_packet (&buffer, 0, pinfo);
}
return res;
}
static void
clean_packet_info (RTPPacketInfo * pinfo)
{
if (pinfo->address)
g_object_unref (pinfo->address);
if (pinfo->data) {
gst_mini_object_unref (pinfo->data);
pinfo->data = NULL;
}
}
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, guint64 ntpnstime)
{
GstFlowReturn result;
guint32 ssrc;
RTPSource *source;
gboolean created;
gboolean prevsender, prevactive;
RTPPacketInfo pinfo = { 0, };
guint64 oldrate;
g_return_val_if_fail (RTP_IS_SESSION (sess), GST_FLOW_ERROR);
g_return_val_if_fail (GST_IS_BUFFER (buffer), GST_FLOW_ERROR);
RTP_SESSION_LOCK (sess);
/* update pinfo stats */
if (!update_packet_info (sess, &pinfo, FALSE, TRUE, FALSE, buffer,
current_time, running_time, ntpnstime)) {
GST_DEBUG ("invalid RTP packet received");
RTP_SESSION_UNLOCK (sess);
return rtp_session_process_rtcp (sess, buffer, current_time, ntpnstime);
}
ssrc = pinfo.ssrc;
source = obtain_source (sess, ssrc, &created, &pinfo, 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, &pinfo);
/* 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;
gint i;
/* for validated sources, we add the CSRCs as well */
for (i = 0; i < pinfo.csrc_count; i++) {
guint32 csrc;
RTPSource *csrc_src;
csrc = pinfo.csrcs[i];
/* get source */
csrc_src = obtain_source (sess, csrc, &created, &pinfo, 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_packet_info (&pinfo);
return result;
/* ERRORS */
collision:
{
RTP_SESSION_UNLOCK (sess);
clean_packet_info (&pinfo);
GST_DEBUG ("ignoring packet because its collisioning");
return GST_FLOW_OK;
}
}
static void
rtp_session_process_rb (RTPSession * sess, RTPSource * source,
GstRTCPPacket * packet, RTPPacketInfo * pinfo)
{
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, pinfo->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,
RTPPacketInfo * pinfo, 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 (pinfo->current_time));
source = obtain_source (sess, senderssrc, &created, pinfo, FALSE);
if (!source)
return;
/* skip non-bye packets for sources that are marked BYE */
if (sess->scheduled_bye && RTP_SOURCE_IS_MARKED_BYE (source))
goto out;
/* 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, pinfo->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, pinfo);
out:
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,
RTPPacketInfo * pinfo)
{
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, pinfo, FALSE);
if (!source)
return;
/* skip non-bye packets for sources that are marked BYE */
if (sess->scheduled_bye && RTP_SOURCE_IS_MARKED_BYE (source))
goto out;
if (created)
on_new_ssrc (sess, source);
rtp_session_process_rb (sess, source, packet, pinfo);
out:
g_object_unref (source);
}
/* Get SDES items and store them in the SSRC */
static void
rtp_session_process_sdes (RTPSession * sess, GstRTCPPacket * packet,
RTPPacketInfo * pinfo)
{
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, pinfo, FALSE);
if (!source)
return;
/* skip non-bye packets for sources that are marked BYE */
if (sess->scheduled_bye && RTP_SOURCE_IS_MARKED_BYE (source))
goto next;
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);
if (g_utf8_validate (value, -1, NULL)) {
gst_structure_set (sdes, name, G_TYPE_STRING, value, NULL);
} else {
GST_WARNING ("ignore SDES field %s with non-utf8 data %s", name, value);
}
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);
next:
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,
RTPPacketInfo * pinfo)
{
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 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 = find_source (sess, ssrc);
if (!source || source->internal) {
GST_DEBUG ("Ignoring suspicious BYE packet (reason: %s)",
!source ? "can't find source" : "has internal source SSRC");
break;
}
/* store time for when we need to time out this source */
source->bye_time = pinfo->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 &&
pinfo->current_time < sess->next_rtcp_check_time) {
GstClockTime time_remaining;
/* Scale our next RTCP check time according to the change of numbers
* of members. But only if a) this is the first RTCP, or b) this is not
* a feedback session, or c) this is a feedback session but we schedule
* for every RTCP interval (aka no t-rr-interval set).
*
* FIXME: a) and b) are not great as we will possibly go below Tmin
* for non-feedback profiles and in case of a) below
* Tmin/t-rr-interval in any case.
*/
if (sess->last_rtcp_send_time == GST_CLOCK_TIME_NONE ||
!(sess->rtp_profile == GST_RTP_PROFILE_AVPF
|| sess->rtp_profile == GST_RTP_PROFILE_SAVPF) ||
sess->next_rtcp_check_time - sess->last_rtcp_send_time ==
sess->last_rtcp_interval) {
time_remaining = sess->next_rtcp_check_time - pinfo->current_time;
sess->next_rtcp_check_time =
gst_util_uint64_scale (time_remaining, members, pmembers);
sess->next_rtcp_check_time += pinfo->current_time;
}
sess->last_rtcp_interval =
gst_util_uint64_scale (sess->last_rtcp_interval, members, pmembers);
GST_DEBUG ("reverse reconsideration %" GST_TIME_FORMAT,
GST_TIME_ARGS (sess->next_rtcp_check_time));
/* mark pending reconsider. We only want to signal the reconsideration
* once after we handled all the source in the bye packet */
reconsider = TRUE;
}
}
on_bye_ssrc (sess, 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,
RTPPacketInfo * pinfo)
{
GST_DEBUG ("received APP");
if (g_signal_has_handler_pending (sess,
rtp_session_signals[SIGNAL_ON_APP_RTCP], 0, TRUE)) {
GstBuffer *data_buffer = NULL;
guint16 data_length;
gchar name[5];
data_length = gst_rtcp_packet_app_get_data_length (packet) * 4;
if (data_length > 0) {
guint8 *data = gst_rtcp_packet_app_get_data (packet);
data_buffer = gst_buffer_copy_region (packet->rtcp->buffer,
GST_BUFFER_COPY_MEMORY, data - packet->rtcp->map.data, data_length);
GST_BUFFER_PTS (data_buffer) = pinfo->running_time;
}
memcpy (name, gst_rtcp_packet_app_get_name (packet), 4);
name[4] = '\0';
RTP_SESSION_UNLOCK (sess);
g_signal_emit (sess, rtp_session_signals[SIGNAL_ON_APP_RTCP], 0,
gst_rtcp_packet_app_get_subtype (packet),
gst_rtcp_packet_app_get_ssrc (packet), name, data_buffer);
RTP_SESSION_LOCK (sess);
if (data_buffer)
gst_buffer_unref (data_buffer);
}
}
static gboolean
rtp_session_request_local_key_unit (RTPSession * sess, RTPSource * src,
guint32 media_ssrc, gboolean fir, GstClockTime current_time)
{
guint32 round_trip = 0;
rtp_source_get_last_rb (src, NULL, NULL, NULL, NULL, NULL, NULL, &round_trip);
if (src->last_keyframe_request != GST_CLOCK_TIME_NONE && round_trip) {
GstClockTime round_trip_in_ns = gst_util_uint64_scale (round_trip,
GST_SECOND, 65536);
/* Sanity check to avoid always ignoring PLI/FIR if we receive RTCP
* packets with erroneous values resulting in crazy high RTT. */
if (round_trip_in_ns > 5 * GST_SECOND)
round_trip_in_ns = GST_SECOND / 2;
if (current_time - src->last_keyframe_request < 2 * round_trip_in_ns) {
GST_DEBUG ("Ignoring %s request from %X because one was send without one "
"RTT (%" GST_TIME_FORMAT " < %" GST_TIME_FORMAT ")",
fir ? "FIR" : "PLI", rtp_source_get_ssrc (src),
GST_TIME_ARGS (current_time - src->last_keyframe_request),
GST_TIME_ARGS (round_trip_in_ns));
return FALSE;
}
}
src->last_keyframe_request = current_time;
GST_LOG ("received %s request from %X about %X %p(%p)", fir ? "FIR" : "PLI",
rtp_source_get_ssrc (src), media_ssrc, sess->callbacks.process_rtp,
sess->callbacks.request_key_unit);
RTP_SESSION_UNLOCK (sess);
sess->callbacks.request_key_unit (sess, media_ssrc, 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 == NULL)
return;
rtp_session_request_local_key_unit (sess, src, media_ssrc, FALSE,
current_time);
}
static void
rtp_session_process_fir (RTPSession * sess, guint32 sender_ssrc,
guint32 media_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 == NULL)
continue;
if (own->internal) {
our_request = TRUE;
break;
}
}
if (!our_request)
return;
rtp_session_request_local_key_unit (sess, src, media_ssrc, TRUE,
current_time);
}
static void
rtp_session_process_nack (RTPSession * sess, guint32 sender_ssrc,
guint32 media_ssrc, guint8 * fci_data, guint fci_length,
GstClockTime current_time)
{
sess->stats.nacks_received++;
if (!sess->callbacks.notify_nack)
return;
while (fci_length > 0) {
guint16 seqnum, blp;
seqnum = GST_READ_UINT16_BE (fci_data);
blp = GST_READ_UINT16_BE (fci_data + 2);
GST_DEBUG ("NACK #%u, blp %04x, SSRC 0x%08x", seqnum, blp, media_ssrc);
RTP_SESSION_UNLOCK (sess);
sess->callbacks.notify_nack (sess, seqnum, blp, media_ssrc,
sess->notify_nack_user_data);
RTP_SESSION_LOCK (sess);
fci_data += 4;
fci_length -= 4;
}
}
static void
rtp_session_process_feedback (RTPSession * sess, GstRTCPPacket * packet,
RTPPacketInfo * pinfo, 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;
src = find_source (sess, media_ssrc);
/* skip non-bye packets for sources that are marked BYE */
if (sess->scheduled_bye && src && RTP_SOURCE_IS_MARKED_BYE (src))
return;
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_PTS (fci_buffer) = pinfo->running_time;
}
RTP_SESSION_UNLOCK (sess);
g_signal_emit (sess, rtp_session_signals[SIGNAL_ON_FEEDBACK_RTCP], 0,
type, fbtype, sender_ssrc, media_ssrc, fci_buffer);
RTP_SESSION_LOCK (sess);
if (fci_buffer)
gst_buffer_unref (fci_buffer);
}
if (src && sess->rtcp_feedback_retention_window) {
rtp_source_retain_rtcp_packet (src, packet, pinfo->running_time);
}
if ((src && 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:
if (src)
src->stats.recv_pli_count++;
rtp_session_process_pli (sess, sender_ssrc, media_ssrc,
current_time);
break;
case GST_RTCP_PSFB_TYPE_FIR:
if (src)
src->stats.recv_fir_count++;
rtp_session_process_fir (sess, sender_ssrc, media_ssrc, fci_data,
fci_length, current_time);
break;
default:
break;
}
break;
case GST_RTCP_TYPE_RTPFB:
switch (fbtype) {
case GST_RTCP_RTPFB_TYPE_NACK:
if (src)
src->stats.recv_nack_count++;
rtp_session_process_nack (sess, sender_ssrc, media_ssrc,
fci_data, fci_length, current_time);
break;
default:
break;
}
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;
RTPPacketInfo pinfo = { 0, };
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_reduced (buffer))
goto invalid_packet;
GST_DEBUG ("received RTCP packet");
g_signal_emit (sess, rtp_session_signals[SIGNAL_ON_RECEIVING_RTCP], 0,
buffer);
RTP_SESSION_LOCK (sess);
/* update pinfo stats */
update_packet_info (sess, &pinfo, FALSE, FALSE, 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);
switch (type) {
case GST_RTCP_TYPE_SR:
rtp_session_process_sr (sess, &packet, &pinfo, &do_sync);
break;
case GST_RTCP_TYPE_RR:
rtp_session_process_rr (sess, &packet, &pinfo);
break;
case GST_RTCP_TYPE_SDES:
rtp_session_process_sdes (sess, &packet, &pinfo);
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, &pinfo);
break;
case GST_RTCP_TYPE_APP:
rtp_session_process_app (sess, &packet, &pinfo);
break;
case GST_RTCP_TYPE_RTPFB:
case GST_RTCP_TYPE_PSFB:
rtp_session_process_feedback (sess, &packet, &pinfo, current_time);
break;
default:
GST_WARNING ("got unknown RTCP packet");
break;
}
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 && is_bye) {
sess->bye_stats.bye_members++;
UPDATE_AVG (sess->bye_stats.avg_rtcp_packet_size, pinfo.bytes);
}
/* keep track of average packet size */
UPDATE_AVG (sess->stats.avg_rtcp_packet_size, pinfo.bytes);
GST_DEBUG ("%p, received RTCP packet, avg size %u, %u", &sess->stats,
sess->stats.avg_rtcp_packet_size, pinfo.bytes);
RTP_SESSION_UNLOCK (sess);
pinfo.data = NULL;
clean_packet_info (&pinfo);
/* 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, GST_CLOCK_TIME_NONE);
sess->suggested_ssrc = ssrc;
sess->internal_ssrc_set = TRUE;
sess->internal_ssrc_from_caps_or_property = TRUE;
if (source) {
rtp_source_update_caps (source, caps);
if (created)
on_new_sender_ssrc (sess, source);
g_object_unref (source);
}
if (gst_structure_get_uint (s, "rtx-ssrc", &ssrc)) {
source =
obtain_internal_source (sess, ssrc, &created, GST_CLOCK_TIME_NONE);
if (source) {
rtp_source_update_caps (source, caps);
g_object_unref (source);
}
}
RTP_SESSION_UNLOCK (sess);
} else {
sess->internal_ssrc_from_caps_or_property = FALSE;
}
}
/**
* 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;
RTPPacketInfo pinfo = { 0, };
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");
RTP_SESSION_LOCK (sess);
if (!update_packet_info (sess, &pinfo, TRUE, TRUE, is_list, data,
current_time, running_time, -1))
goto invalid_packet;
source = obtain_internal_source (sess, pinfo.ssrc, &created, current_time);
if (created)
on_new_sender_ssrc (sess, source);
prevsender = RTP_SOURCE_IS_SENDER (source);
oldrate = source->bitrate;
/* we use our own source to send */
result = rtp_source_send_rtp (source, &pinfo);
source_update_sender (sess, source, prevsender);
if (oldrate != source->bitrate)
sess->recalc_bandwidth = TRUE;
RTP_SESSION_UNLOCK (sess);
g_object_unref (source);
clean_packet_info (&pinfo);
return result;
invalid_packet:
{
gst_mini_object_unref (GST_MINI_OBJECT_CAST (data));
RTP_SESSION_UNLOCK (sess);
GST_DEBUG ("invalid RTP packet received");
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;
RTPSessionStats *stats;
/* 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);
}
if (bandwidth < RTP_STATS_BANDWIDTH)
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) {
stats = &sess->bye_stats;
result = rtp_stats_calculate_bye_interval (stats);
} else {
session_update_ptp (sess);
stats = &sess->stats;
result = rtp_stats_calculate_rtcp_interval (stats,
stats->internal_sender_sources > 0, sess->rtp_profile,
sess->is_doing_ptp, 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 (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 */
memcpy (&sess->bye_stats, &sess->stats, sizeof (RTPSessionStats));
INIT_AVG (sess->bye_stats.avg_rtcp_packet_size, 100);
sess->bye_stats.bye_members = 1;
sess->first_rtcp = TRUE;
/* reschedule transmission */
sess->last_rtcp_send_time = current_time;
sess->last_rtcp_check_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;
sess->last_rtcp_interval = interval;
GST_DEBUG ("Schedule BYE for %" GST_TIME_FORMAT ", %" GST_TIME_FORMAT,
GST_TIME_ARGS (interval), GST_TIME_ARGS (sess->next_rtcp_check_time));
RTP_SESSION_UNLOCK (sess);
/* notify app of reconsideration */
if (sess->callbacks.reconsider)
sess->callbacks.reconsider (sess, sess->reconsider_user_data);
RTP_SESSION_LOCK (sess);
done:
return result;
}
/**
* rtp_session_schedule_bye:
* @sess: an #RTPSession
* @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;
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)) {
GST_DEBUG ("have 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->bye_stats.active_sources >= 50) {
GST_DEBUG ("reconsider BYE, more than 50 sources");
/* reconsider BYE if members >= 50 */
interval = calculate_rtcp_interval (sess, FALSE, TRUE);
sess->last_rtcp_interval = interval;
}
} else {
if (sess->first_rtcp) {
GST_DEBUG ("first RTCP packet");
/* we are called for the first time */
interval = calculate_rtcp_interval (sess, FALSE, TRUE);
sess->last_rtcp_interval = interval;
} 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);
sess->last_rtcp_interval = interval;
if ((sess->rtp_profile == GST_RTP_PROFILE_AVPF
|| sess->rtp_profile == GST_RTP_PROFILE_SAVPF)
&& interval != GST_CLOCK_TIME_NONE) {
/* Apply the rules from RFC 4585 section 3.5.3 */
if (sess->stats.min_interval != 0) {
GstClockTime T_rr_current_interval = g_random_double_range (0.5,
1.5) * sess->stats.min_interval * GST_SECOND;
if (T_rr_current_interval > interval) {
GST_DEBUG ("Adjusting interval for t-rr-interval: %" GST_TIME_FORMAT
" > %" GST_TIME_FORMAT, GST_TIME_ARGS (T_rr_current_interval),
GST_TIME_ARGS (interval));
interval = T_rr_current_interval;
}
}
}
}
}
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;
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;
guint nacked_seqnums;
} 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 (data->is_early && sess->reduced_size_rtcp)
return;
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;
}
if (g_hash_table_contains (source->reported_in_sr_of,
GUINT_TO_POINTER (data->source->ssrc))) {
GST_DEBUG ("source %08x already reported in this generation", source->ssrc);
return;
}
if (gst_rtcp_packet_get_rb_count (packet) == GST_RTCP_MAX_RB_COUNT) {
GST_DEBUG ("max RB count reached");
return;
}
/* only report about other sender */
if (source == data->source)
goto reported;
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:
g_hash_table_add (source->reported_in_sr_of,
GUINT_TO_POINTER (data->source->ssrc));
}
/* 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;
source->stats.sent_fir_count++;
}
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;
source->stats.sent_pli_count++;
}
/* 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;
data->nacked_seqnums++;
}
rtp_source_clear_nacks (source);
data->may_suppress = FALSE;
source->stats.sent_nack_count += n_nacks;
}
/* 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,
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 && 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;
}
}
if (source->internal && source->sent_bye) {
GST_DEBUG ("removing internal source that has sent BYE %08x", source->ssrc);
remove = 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));
if (source->internal) {
/* this is an internal source that is not using our suggested ssrc.
* since there must be another source using this ssrc, we can remove
* this one instead of making it a receiver forever */
if (source->ssrc != sess->suggested_ssrc) {
rtp_source_mark_bye (source, "timed out");
/* do not schedule bye here, since we are inside the RTCP timeout
* processing and scheduling bye will interfere with SR/RR sending */
}
} else {
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) {
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;
GstClockTime interval;
RTPSessionStats *stats;
if (sess->scheduled_bye)
stats = &sess->bye_stats;
else
stats = &sess->stats;
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) {
GST_DEBUG ("early feedback %" GST_TIME_FORMAT " < now %"
GST_TIME_FORMAT, GST_TIME_ARGS (sess->next_early_rtcp_time),
GST_TIME_ARGS (current_time));
} else 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;
}
/* take interval and add jitter */
interval = data->interval;
if (interval != GST_CLOCK_TIME_NONE)
interval = rtp_stats_add_rtcp_jitter (stats, interval);
if (sess->last_rtcp_check_time != GST_CLOCK_TIME_NONE) {
/* perform forward reconsideration */
if (interval != GST_CLOCK_TIME_NONE) {
GstClockTime elapsed;
/* get elapsed time since we last reported */
elapsed = current_time - sess->last_rtcp_check_time;
GST_DEBUG ("forward reconsideration %" GST_TIME_FORMAT ", elapsed %"
GST_TIME_FORMAT, GST_TIME_ARGS (interval), GST_TIME_ARGS (elapsed));
new_send_time = interval + sess->last_rtcp_check_time;
} else {
new_send_time = sess->last_rtcp_check_time;
}
} else {
/* If this is the first RTCP packet, we can reconsider anything based
* on the last RTCP send time because there was none.
*/
g_warn_if_fail (!data->is_early);
data->is_early = FALSE;
new_send_time = current_time;
}
if (!data->is_early) {
/* 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;
sess->last_rtcp_interval = interval;
return FALSE;
}
sess->last_rtcp_interval = interval;
if ((sess->rtp_profile == GST_RTP_PROFILE_AVPF
|| sess->rtp_profile == GST_RTP_PROFILE_SAVPF)
&& interval != GST_CLOCK_TIME_NONE) {
/* Apply the rules from RFC 4585 section 3.5.3 */
if (stats->min_interval != 0 && !sess->first_rtcp) {
GstClockTime T_rr_current_interval =
g_random_double_range (0.5, 1.5) * stats->min_interval * GST_SECOND;
if (T_rr_current_interval > interval) {
GST_DEBUG ("Adjusting interval for t-rr-interval: %" GST_TIME_FORMAT
" > %" GST_TIME_FORMAT, GST_TIME_ARGS (T_rr_current_interval),
GST_TIME_ARGS (interval));
interval = T_rr_current_interval;
}
}
}
sess->next_rtcp_check_time = current_time + interval;
}
GST_DEBUG ("can send RTCP now, next %" GST_TIME_FORMAT,
GST_TIME_ARGS (sess->next_rtcp_check_time));
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;
ReportOutput *output;
/* only generate RTCP for active internal sources */
if (!source->internal || source->sent_bye)
return;
/* ignore other sources when we do the timeout after a scheduled BYE */
if (sess->scheduled_bye && !source->marked_bye)
return;
data->source = source;
/* open packet */
session_start_rtcp (sess, data);
if (source->marked_bye) {
/* send BYE */
make_source_bye (sess, source, data);
} 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 && (!data->is_early || !sess->reduced_size_rtcp))
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->buffer = data->rtcp;
/* queue the RTCP packet to push later */
g_queue_push_tail (&data->output, output);
}
static void
update_generation (const gchar * key, RTPSource * source, ReportData * data)
{
RTPSession *sess = data->sess;
if (g_hash_table_size (source->reported_in_sr_of) >=
sess->stats.internal_sources) {
/* source is reported, move to next generation */
source->generation = sess->generation + 1;
g_hash_table_remove_all (source->reported_in_sr_of);
GST_LOG ("reported source %x, new generation: %d", source->ssrc,
source->generation);
/* 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);
}
}
}
/**
* 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;
gboolean all_empty = FALSE;
g_return_val_if_fail (RTP_IS_SESSION (sess), GST_FLOW_ERROR);
GST_DEBUG ("reporting at %" GST_TIME_FORMAT ", NTP time %" GST_TIME_FORMAT
", 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;
data.nacked_seqnums = 0;
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);
GST_DEBUG ("interval %" GST_TIME_FORMAT, GST_TIME_ARGS (data.interval));
/* 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,
current_time);
sess->internal_ssrc_set = TRUE;
if (created)
on_new_sender_ssrc (sess, source);
g_object_unref (source);
}
sess->conflicting_addresses =
timeout_conflicting_addresses (sess->conflicting_addresses, current_time);
/* 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);
/* update point-to-point status */
session_update_ptp (sess);
/* see if we need to generate SR or RR packets */
if (!is_rtcp_time (sess, current_time, &data))
goto done;
/* check if all the buffers are empty afer generation */
all_empty = TRUE;
GST_DEBUG
("doing RTCP generation %u for %u sources, early %d, may suppress %d",
sess->generation, data.num_to_report, data.is_early, data.may_suppress);
/* generate RTCP for all internal sources */
g_hash_table_foreach (sess->ssrcs[sess->mask_idx],
(GHFunc) generate_rtcp, &data);
/* update the generation for all the sources that have been reported */
g_hash_table_foreach (sess->ssrcs[sess->mask_idx],
(GHFunc) update_generation, &data);
/* we keep track of the last report time in order to timeout inactive
* receivers or senders */
if (!data.is_early) {
GST_DEBUG ("Time since last regular RTCP: %" GST_TIME_FORMAT " - %"
GST_TIME_FORMAT " = %" GST_TIME_FORMAT,
GST_TIME_ARGS (data.current_time),
GST_TIME_ARGS (sess->last_rtcp_send_time),
GST_TIME_ARGS (data.current_time - sess->last_rtcp_send_time));
sess->last_rtcp_send_time = data.current_time;
}
GST_DEBUG ("Time since last RTCP: %" GST_TIME_FORMAT " - %" GST_TIME_FORMAT
" = %" GST_TIME_FORMAT, GST_TIME_ARGS (data.current_time),
GST_TIME_ARGS (sess->last_rtcp_send_time),
GST_TIME_ARGS (data.current_time - sess->last_rtcp_check_time));
sess->last_rtcp_check_time = data.current_time;
sess->first_rtcp = FALSE;
sess->next_early_rtcp_time = GST_CLOCK_TIME_NONE;
sess->scheduled_bye = FALSE;
done:
RTP_SESSION_UNLOCK (sess);
/* notify about updated statistics */
g_object_notify (G_OBJECT (sess), "stats");
/* push out the RTCP packets */
while ((output = g_queue_pop_head (&data.output))) {
gboolean do_not_suppress, empty_buffer;
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);
empty_buffer = gst_buffer_get_size (buffer) == 0;
if (!empty_buffer)
all_empty = FALSE;
if (sess->callbacks.send_rtcp &&
!empty_buffer && (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,
sess->send_rtcp_user_data);
RTP_SESSION_LOCK (sess);
sess->stats.nacks_sent += data.nacked_seqnums;
on_sender_ssrc_active (sess, source);
RTP_SESSION_UNLOCK (sess);
} else {
GST_DEBUG ("freeing packet callback: %p"
" empty_buffer: %d, "
" do_not_suppress: %d may_suppress: %d", sess->callbacks.send_rtcp,
empty_buffer, do_not_suppress, data.may_suppress);
if (!empty_buffer) {
RTP_SESSION_LOCK (sess);
sess->stats.nacks_dropped += data.nacked_seqnums;
RTP_SESSION_UNLOCK (sess);
}
gst_buffer_unref (buffer);
}
g_object_unref (source);
g_slice_free (ReportOutput, output);
}
if (all_empty)
GST_ERROR ("generated empty RTCP messages for all the sources");
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
*
* Returns: %TRUE if the related RTCP can be scheduled.
*/
gboolean
rtp_session_request_early_rtcp (RTPSession * sess, GstClockTime current_time,
GstClockTime max_delay)
{
GstClockTime T_dither_max, T_rr, offset = 0;
gboolean ret;
gboolean allow_early;
/* Implements the algorithm described in RFC 4585 section 3.5.2 */
RTP_SESSION_LOCK (sess);
/* We assume a feedback profile if something is requesting RTCP
* to be sent */
sess->rtp_profile = GST_RTP_PROFILE_AVPF;
/* Check if already requested */
/* RFC 4585 section 3.5.2 step 2 */
if (GST_CLOCK_TIME_IS_VALID (sess->next_early_rtcp_time)) {
GST_LOG_OBJECT (sess, "already have next early rtcp time");
ret = (current_time + max_delay > sess->next_early_rtcp_time);
goto end;
}
if (!GST_CLOCK_TIME_IS_VALID (sess->next_rtcp_check_time)) {
GST_LOG_OBJECT (sess, "no next RTCP check time");
ret = FALSE;
goto end;
}
/* RFC 4585 section 3.5.3 step 1
* If no regular RTCP packet has been sent before, then a regular
* RTCP packet has to be scheduled first and FB messages might be
* included there
*/
if (!GST_CLOCK_TIME_IS_VALID (sess->last_rtcp_send_time)) {
GST_LOG_OBJECT (sess, "no RTCP sent yet");
if (current_time + max_delay > sess->next_rtcp_check_time) {
GST_LOG_OBJECT (sess,
"next scheduled time is soon %" GST_TIME_FORMAT " + %" GST_TIME_FORMAT
" > %" GST_TIME_FORMAT, GST_TIME_ARGS (current_time),
GST_TIME_ARGS (max_delay),
GST_TIME_ARGS (sess->next_rtcp_check_time));
ret = TRUE;
} else {
GST_LOG_OBJECT (sess,
"can't allow early feedback, next scheduled time is too late %"
GST_TIME_FORMAT " + %" GST_TIME_FORMAT " < %" GST_TIME_FORMAT,
GST_TIME_ARGS (current_time), GST_TIME_ARGS (max_delay),
GST_TIME_ARGS (sess->next_rtcp_check_time));
ret = FALSE;
}
goto end;
}
T_rr = sess->last_rtcp_interval;
/* RFC 4585 section 3.5.2 step 2b */
/* If the total sources is <=2, then there is only us and one peer */
/* When there is one auxiliary stream the session can still do point
* to point.
*/
if (sess->is_doing_ptp) {
T_dither_max = 0;
} else {
/* Divide by 2 because l = 0.5 */
T_dither_max = T_rr;
T_dither_max /= 2;
}
/* RFC 4585 section 3.5.2 step 3 */
if (current_time + T_dither_max > sess->next_rtcp_check_time) {
GST_LOG_OBJECT (sess,
"don't send because of dither, next scheduled time is too soon %"
GST_TIME_FORMAT " + %" GST_TIME_FORMAT " > %" GST_TIME_FORMAT,
GST_TIME_ARGS (current_time), GST_TIME_ARGS (T_dither_max),
GST_TIME_ARGS (sess->next_rtcp_check_time));
ret = T_dither_max <= max_delay;
goto end;
}
/* RFC 4585 section 3.5.2 step 4a and
* RFC 4585 section 3.5.2 step 6 */
allow_early = FALSE;
if (sess->last_rtcp_check_time == sess->last_rtcp_send_time) {
/* Last time we sent a full RTCP packet, we can now immediately
* send an early one as allow_early was reset to TRUE */
allow_early = TRUE;
} else if (sess->last_rtcp_check_time + T_rr <= current_time + max_delay) {
/* Last packet we sent was an early RTCP packet and more than
* T_rr has passed since then, meaning we would have suppressed
* a regular RTCP packet already and reset allow_early to TRUE */
allow_early = TRUE;
/* We have to offset a bit as T_rr has not passed yet, but will before
* max_delay */
if (sess->last_rtcp_check_time + T_rr > current_time)
offset = (sess->last_rtcp_check_time + T_rr) - current_time;
} else {
GST_DEBUG_OBJECT (sess,
"can't allow early RTCP yet: last regular %" GST_TIME_FORMAT ", %"
GST_TIME_FORMAT " + %" GST_TIME_FORMAT " > %" GST_TIME_FORMAT " + %"
GST_TIME_FORMAT, GST_TIME_ARGS (sess->last_rtcp_send_time),
GST_TIME_ARGS (sess->last_rtcp_check_time), GST_TIME_ARGS (T_rr),
GST_TIME_ARGS (current_time), GST_TIME_ARGS (max_delay));
}
if (!allow_early) {
/* Ignore the request a scheduled packet will be in time anyway */
if (current_time + max_delay > sess->next_rtcp_check_time) {
GST_LOG_OBJECT (sess,
"next scheduled time is soon %" GST_TIME_FORMAT " + %" GST_TIME_FORMAT
" > %" GST_TIME_FORMAT, GST_TIME_ARGS (current_time),
GST_TIME_ARGS (max_delay),
GST_TIME_ARGS (sess->next_rtcp_check_time));
ret = TRUE;
} else {
GST_LOG_OBJECT (sess,
"can't allow early feedback and next scheduled time is too late %"
GST_TIME_FORMAT " + %" GST_TIME_FORMAT " < %" GST_TIME_FORMAT,
GST_TIME_ARGS (current_time), GST_TIME_ARGS (max_delay),
GST_TIME_ARGS (sess->next_rtcp_check_time));
ret = FALSE;
}
goto end;
}
/* RFC 4585 section 3.5.2 step 4b */
if (T_dither_max) {
/* Schedule an early transmission later */
sess->next_early_rtcp_time = g_random_double () * T_dither_max +
current_time + offset;
} else {
/* If no dithering, schedule it for NOW */
sess->next_early_rtcp_time = current_time + offset;
}
GST_LOG_OBJECT (sess, "next early RTCP time %" GST_TIME_FORMAT
", next regular RTCP time %" GST_TIME_FORMAT,
GST_TIME_ARGS (sess->next_early_rtcp_time),
GST_TIME_ARGS (sess->next_rtcp_check_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 TRUE;
end:
RTP_SESSION_UNLOCK (sess);
return ret;
}
static gboolean
rtp_session_send_rtcp (RTPSession * sess, GstClockTime max_delay)
{
GstClockTime now;
if (!sess->callbacks.send_rtcp)
return FALSE;
now = sess->callbacks.request_time (sess, sess->request_time_user_data);
return 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;
if (!rtp_session_send_rtcp (sess, 5 * GST_SECOND)) {
GST_DEBUG ("FIR/PLI not sent");
return FALSE;
}
RTP_SESSION_LOCK (sess);
src = find_source (sess, ssrc);
if (src == NULL)
goto no_source;
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_UNLOCK (sess);
return TRUE;
/* ERRORS */
no_source:
{
RTP_SESSION_UNLOCK (sess);
return FALSE;
}
}
/**
* rtp_session_request_nack:
* @sess: a #RTPSession
* @ssrc: the SSRC
* @seqnum: the missing seqnum
* @max_delay: max delay to request NACK
*
* Request scheduling of a NACK feedback packet for @seqnum in @ssrc.
*
* Returns: %TRUE if the NACK feedback could be scheduled
*/
gboolean
rtp_session_request_nack (RTPSession * sess, guint32 ssrc, guint16 seqnum,
GstClockTime max_delay)
{
RTPSource *source;
if (!rtp_session_send_rtcp (sess, max_delay)) {
GST_DEBUG ("NACK not sent");
return FALSE;
}
RTP_SESSION_LOCK (sess);
source = find_source (sess, ssrc);
if (source == NULL)
goto no_source;
GST_DEBUG ("request NACK for %08x, #%u", ssrc, seqnum);
rtp_source_register_nack (source, seqnum);
RTP_SESSION_UNLOCK (sess);
return TRUE;
/* ERRORS */
no_source:
{
RTP_SESSION_UNLOCK (sess);
return FALSE;
}
}
Reference in a new issue View git blame Copy permalink