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gstreamer/subprojects/gst-plugins-good/gst/rtpmanager/rtpsession.c
Ognyan Tonchev 03b6226772 rtpmanager: skip RTPSources which are not ready in the RTCP generation 
If a stream has an 'irregular' frame rate (e.g. metadata) RTCP SR
may be generated way too early, before the RTPSource has received
the first packet after Latency was configured in the pipeline.
We skip such RTPSources in the RTCP generation.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/7740>
2024-10-29 02:10:47 +00:00

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/* 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 <stdlib.h>
#include <gst/rtp/gstrtpbuffer.h>
#include <gst/rtp/gstrtcpbuffer.h>
#include <gst/glib-compat-private.h>
#include "rtpsession.h"
#include "gstrtputils.h"
GST_DEBUG_CATEGORY (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,
SIGNAL_ON_SENDING_NACKS,
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
#define DEFAULT_RTCP_DISABLE_SR_TIMESTAMP FALSE
#define DEFAULT_FAVOR_NEW FALSE
#define DEFAULT_TWCC_FEEDBACK_INTERVAL GST_CLOCK_TIME_NONE
#define DEFAULT_UPDATE_NTP64_HEADER_EXT TRUE
#define DEFAULT_TIMEOUT_INACTIVE_SOURCES TRUE
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,
PROP_RTCP_DISABLE_SR_TIMESTAMP,
PROP_TWCC_FEEDBACK_INTERVAL,
PROP_UPDATE_NTP64_HEADER_EXT,
PROP_TIMEOUT_INACTIVE_SOURCES,
PROP_LAST,
};
static GParamSpec *properties[PROP_LAST];
/* update average packet size */
#define INIT_AVG(avg, val) \
(avg) = (val);
#define UPDATE_AVG(avg, val) \
if ((avg) == 0) \
(avg) = (val); \
else \
(avg) = ((val) + (15 * (avg))) >> 4;
/* 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 gboolean rtp_session_send_rtcp_with_deadline (RTPSession * sess,
GstClockTime deadline);
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, NULL,
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, NULL, 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, NULL, 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, NULL, 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, NULL, 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, NULL, 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, NULL, 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, NULL, 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, NULL, 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, NULL, 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, NULL, 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, NULL, 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, NULL, 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,
NULL, 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,
NULL, 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, NULL, 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, NULL, 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, NULL,
G_TYPE_NONE, 1, RTP_TYPE_SOURCE);
/**
* RTPSession::on-sending-nack
* @session: the object which received the signal
* @sender_ssrc: the sender ssrc
* @media_ssrc: the media ssrc
* @nacks: (element-type guint16): the list of seqnum to be nacked
* @buffer: the #GstBuffer containing the RTCP packet about to be sent
*
* This signal is emitted before NACK packets are added into the RTCP
* packet. This signal can be used to override the conversion of the NACK
* seqnum array into packets. This can be used if your protocol uses
* different type of NACK (e.g. based on RTCP APP).
*
* The handler should transform the seqnum from @nacks array into packets.
* @nacks seqnum must be consumed from the start. The remaining will be
* rescheduled for later base on bandwidth. Only one handler will be
* signalled.
*
* A handler may return 0 to signal that generic NACKs should be created
* for this set. This can be useful if the signal is used for other purpose
* or if the other type of NACK would use more space.
*
* Returns: the number of NACK seqnum that was consumed from @nacks.
*
* Since: 1.16
*/
rtp_session_signals[SIGNAL_ON_SENDING_NACKS] =
g_signal_new ("on-sending-nacks", G_TYPE_FROM_CLASS (klass),
G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (RTPSessionClass, on_sending_nacks),
g_signal_accumulator_first_wins, NULL, NULL,
G_TYPE_UINT, 4, G_TYPE_UINT, G_TYPE_UINT, G_TYPE_ARRAY,
GST_TYPE_BUFFER | G_SIGNAL_TYPE_STATIC_SCOPE);
properties[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 | GST_PARAM_DOC_SHOW_DEFAULT);
properties[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);
properties[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);
properties[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);
properties[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);
properties[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);
properties[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);
properties[PROP_SDES] =
g_param_spec_boxed ("sdes", "SDES",
"The SDES items of this session",
GST_TYPE_STRUCTURE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS
| GST_PARAM_DOC_SHOW_DEFAULT);
properties[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);
properties[PROP_NUM_ACTIVE_SOURCES] =
g_param_spec_uint ("num-active-sources", "Num Active Sources",
"The number of active sources in the session", 0, G_MAXUINT,
DEFAULT_NUM_ACTIVE_SOURCES, G_PARAM_READABLE | G_PARAM_STATIC_STRINGS);
/**
* RTPSource:sources
*
* Get a GValue Array of all sources in the session.
*
* ## Getting the #RTPSources of a session
*
* ``` C
* {
* 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);
* }
* ```
*/
properties[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);
properties[PROP_FAVOR_NEW] =
g_param_spec_boolean ("favor-new", "Favor new sources",
"Resolve SSRC conflict in favor of new sources", DEFAULT_FAVOR_NEW,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS);
properties[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);
properties[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);
properties[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);
properties[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);
properties[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);
properties[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
*/
properties[PROP_STATS] =
g_param_spec_boxed ("stats", "Statistics",
"Various statistics", GST_TYPE_STRUCTURE,
G_PARAM_READABLE | G_PARAM_STATIC_STRINGS);
properties[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);
properties[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);
/**
* RTPSession:disable-sr-timestamp:
*
* Whether sender reports should be timestamped.
*
* Since: 1.16
*/
properties[PROP_RTCP_DISABLE_SR_TIMESTAMP] =
g_param_spec_boolean ("disable-sr-timestamp",
"Disable Sender Report Timestamp",
"Whether sender reports should be timestamped",
DEFAULT_RTCP_DISABLE_SR_TIMESTAMP,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS);
/**
* RTPSession:twcc-feedback-interval:
*
* The interval to send TWCC reports on.
* This overrides the default behavior of sending reports
* based on marker-bits.
*
* Since: 1.20
*/
properties[PROP_TWCC_FEEDBACK_INTERVAL] =
g_param_spec_uint64 ("twcc-feedback-interval",
"TWCC Feedback Interval",
"The interval to send TWCC reports on",
0, G_MAXUINT64, DEFAULT_TWCC_FEEDBACK_INTERVAL,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS);
/**
* RTPSession:update-ntp64-header-ext:
*
* Whether RTP NTP header extension should be updated with actual
* NTP time. If not, use the NTP time from buffer timestamp metadata
*
* Since: 1.22
*/
properties[PROP_UPDATE_NTP64_HEADER_EXT] =
g_param_spec_boolean ("update-ntp64-header-ext",
"Update NTP-64 RTP Header Extension",
"Whether RTP NTP header extension should be updated with actual NTP time",
DEFAULT_UPDATE_NTP64_HEADER_EXT,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS);
/**
* RTPSession:timeout-inactive-sources:
*
* Whether inactive sources should be timed out
*
* Since: 1.24
*/
properties[PROP_TIMEOUT_INACTIVE_SOURCES] =
g_param_spec_boolean ("timeout-inactive-sources",
"Time out inactive sources",
"Whether sources that don't receive RTP or RTCP packets for longer "
"than 5x RTCP interval should be removed",
DEFAULT_TIMEOUT_INACTIVE_SOURCES,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS);
g_object_class_install_properties (gobject_class, PROP_LAST, properties);
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 = UDP_IP_HEADER_OVERHEAD;
sess->mtu = DEFAULT_RTCP_MTU;
sess->update_ntp64_header_ext = DEFAULT_UPDATE_NTP64_HEADER_EXT;
sess->timeout_inactive_sources = DEFAULT_TIMEOUT_INACTIVE_SOURCES;
sess->probation = DEFAULT_PROBATION;
sess->max_dropout_time = DEFAULT_MAX_DROPOUT_TIME;
sess->max_misorder_time = DEFAULT_MAX_MISORDER_TIME;
sess->favor_new = DEFAULT_FAVOR_NEW;
/* 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->timestamp_sender_reports = !DEFAULT_RTCP_DISABLE_SR_TIMESTAMP;
sess->is_doing_ptp = TRUE;
sess->twcc = rtp_twcc_manager_new (sess->mtu);
sess->twcc_stats = rtp_twcc_stats_new ();
}
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_object_unref (sess->twcc);
rtp_twcc_stats_free (sess->twcc_stats);
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;
GstStructure *s;
g_object_get (source, "stats", &s, NULL);
g_value_array_append (arr, NULL);
value = g_value_array_get_nth (arr, arr->n_values - 1);
g_value_init (value, GST_TYPE_STRUCTURE);
g_value_take_boxed (value, s);
}
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);
rtp_twcc_manager_set_mtu (sess->twcc, sess->mtu);
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_FEEDBACK_RETENTION_WINDOW:
sess->rtcp_feedback_retention_window = g_value_get_uint64 (value);
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;
case PROP_RTCP_DISABLE_SR_TIMESTAMP:
sess->timestamp_sender_reports = !g_value_get_boolean (value);
break;
case PROP_TWCC_FEEDBACK_INTERVAL:
rtp_twcc_manager_set_feedback_interval (sess->twcc,
g_value_get_uint64 (value));
break;
case PROP_UPDATE_NTP64_HEADER_EXT:
sess->update_ntp64_header_ext = g_value_get_boolean (value);
break;
case PROP_TIMEOUT_INACTIVE_SOURCES:
sess->timeout_inactive_sources = 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_FEEDBACK_RETENTION_WINDOW:
g_value_set_uint64 (value, sess->rtcp_feedback_retention_window);
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;
case PROP_RTCP_DISABLE_SR_TIMESTAMP:
g_value_set_boolean (value, !sess->timestamp_sender_reports);
break;
case PROP_TWCC_FEEDBACK_INTERVAL:
g_value_set_uint64 (value,
rtp_twcc_manager_get_feedback_interval (sess->twcc));
break;
case PROP_UPDATE_NTP64_HEADER_EXT:
g_value_set_boolean (value, sess->update_ntp64_header_ext);
break;
case PROP_TIMEOUT_INACTIVE_SOURCES:
g_value_set_boolean (value, sess->timeout_inactive_sources);
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_reset:
* @sess: an #RTPSession
*
* Reset the sources of @sess.
*/
void
rtp_session_reset (RTPSession * sess)
{
g_return_if_fail (RTP_IS_SESSION (sess));
RTP_SESSION_LOCK (sess);
/* remove all sources */
g_hash_table_remove_all (sess->ssrcs[sess->mask_idx]);
sess->total_sources = 0;
sess->stats.sender_sources = 0;
sess->stats.internal_sender_sources = 0;
sess->stats.internal_sources = 0;
sess->stats.active_sources = 0;
sess->generation = 0;
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->scheduled_bye = FALSE;
/* reset session stats */
sess->stats.bye_members = 0;
sess->stats.nacks_dropped = 0;
sess->stats.nacks_sent = 0;
sess->stats.nacks_received = 0;
sess->is_doing_ptp = TRUE;
g_list_free_full (sess->conflicting_addresses,
(GDestroyNotify) rtp_conflicting_address_free);
sess->conflicting_addresses = NULL;
RTP_SESSION_UNLOCK (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->caps) {
sess->callbacks.caps = callbacks->caps;
sess->caps_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->notify_twcc) {
sess->callbacks.notify_twcc = callbacks->notify_twcc;
sess->notify_twcc_user_data = user_data;
}
if (callbacks->reconfigure) {
sess->callbacks.reconfigure = callbacks->reconfigure;
sess->reconfigure_user_data = user_data;
}
if (callbacks->notify_early_rtcp) {
sess->callbacks.notify_early_rtcp = callbacks->notify_early_rtcp;
sess->notify_early_rtcp_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_caps_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_caps_callback (RTPSession * sess,
RTPSessionCaps callback, gpointer user_data)
{
g_return_if_fail (RTP_IS_SESSION (sess));
sess->callbacks.caps = callback;
sess->caps_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;
}
static void
source_set_sdes (const gchar * key, RTPSource * source, GstStructure * sdes)
{
rtp_source_set_sdes_struct (source, gst_structure_copy (sdes));
}
/**
* 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);
g_hash_table_foreach (sess->ssrcs[sess->mask_idx],
(GHFunc) source_set_sdes, sess->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 GstCaps *
source_caps (RTPSource * source, guint8 pt, RTPSession * session)
{
GstCaps *result = NULL;
RTP_SESSION_UNLOCK (session);
if (session->callbacks.caps)
result = session->callbacks.caps (session, pt, session->caps_user_data);
RTP_SESSION_LOCK (session);
GST_DEBUG ("got caps %" GST_PTR_FORMAT " for pt %d", result, pt);
return result;
}
static RTPSourceCallbacks callbacks = {
(RTPSourcePushRTP) source_push_rtp,
(RTPSourceCaps) source_caps,
};
/**
* 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 void
rtp_session_have_conflict (RTPSession * sess, RTPSource * source,
GSocketAddress * address, GstClockTime current_time)
{
guint32 ssrc = rtp_source_get_ssrc (source);
/* Its a new collision, lets change our SSRC */
rtp_session_add_conflicting_address (sess, address, current_time);
/* 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, current_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 {
GST_DEBUG ("Collision for SSRC %x from new incoming packet,"
" change our sender ssrc", ssrc);
rtp_session_have_conflict (sess, source, pinfo->address,
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 = 0;
source->curr_probation = 0;
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;
}
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);
pinfo->marker = gst_rtp_buffer_get_marker (&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);
/* RTP header extensions */
pinfo->header_ext = gst_rtp_buffer_get_extension_bytes (&rtp,
&pinfo->header_ext_bit_pattern);
}
if (pinfo->ntp64_ext_id != 0 && pinfo->send && !pinfo->have_ntp64_ext) {
guint8 *data;
guint size;
/* Remember here that there is a 64-bit NTP header extension on this buffer
* or any of the other buffers in the buffer list.
* Later we update this after making the buffer(list) writable.
*/
if ((gst_rtp_buffer_get_extension_onebyte_header (&rtp,
pinfo->ntp64_ext_id, 0, (gpointer *) & data, &size)
&& size == 8)
|| (gst_rtp_buffer_get_extension_twobytes_header (&rtp, NULL,
pinfo->ntp64_ext_id, 0, (gpointer *) & data, &size)
&& size == 8)) {
pinfo->have_ntp64_ext = TRUE;
}
}
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 RTP_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;
pinfo->marker = FALSE;
pinfo->ntp64_ext_id = send ? sess->send_ntp64_ext_id : 0;
pinfo->have_ntp64_ext = FALSE;
if (is_list) {
GstBufferList *list = GST_BUFFER_LIST_CAST (data);
res =
gst_buffer_list_foreach (list, (GstBufferListFunc) update_packet,
pinfo);
pinfo->arrival_time = GST_CLOCK_TIME_NONE;
} else {
GstBuffer *buffer = GST_BUFFER_CAST (data);
res = update_packet (&buffer, 0, pinfo);
pinfo->arrival_time = GST_BUFFER_DTS (buffer);
}
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;
}
if (pinfo->header_ext)
g_bytes_unref (pinfo->header_ext);
}
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 void
process_twcc_packet (RTPSession * sess, RTPPacketInfo * pinfo)
{
if (rtp_twcc_manager_recv_packet (sess->twcc, pinfo)) {
RTP_SESSION_UNLOCK (sess);
/* TODO: find a better rational for this number, and possibly tune it based
on factors like framerate / bandwidth etc */
if (!rtp_session_send_rtcp (sess, 100 * GST_MSECOND)) {
GST_INFO ("Could not schedule TWCC straight away");
}
RTP_SESSION_LOCK (sess);
}
}
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, running_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;
if (created)
on_new_ssrc (sess, source);
/* let source process the packet */
result = rtp_source_process_rtp (source, &pinfo);
process_twcc_packet (sess, &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 (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;
guint32 sender_ssrc = source->ssrc;
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 from SSRC %08x: SSRC %08x, jitter %" G_GUINT32_FORMAT, i,
sender_ssrc, 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 ssrc the RTCP message is about. We could also compare our stats against
* the other sender to see if we are better or worse. */
rtp_source_process_rb (src, ssrc, sender_ssrc, pinfo->ntpnstime,
fractionlost, packetslost, exthighestseq, jitter, lsr, dlsr);
/* deprecated: it is also stored in the non-internal source */
rtp_source_process_rb (source, ssrc, sender_ssrc, 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 information 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,
const guint32 * ssrcs, guint num_ssrcs, gboolean fir,
GstClockTime current_time)
{
guint32 round_trip = 0;
gint i;
g_return_val_if_fail (ssrcs != NULL && num_ssrcs > 0, FALSE);
rtp_source_get_last_rb (src, NULL, 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;
for (i = 0; i < num_ssrcs; ++i) {
GST_LOG ("received %s request from %X about %X %p(%p)",
fir ? "FIR" : "PLI",
rtp_source_get_ssrc (src), ssrcs[i], sess->callbacks.process_rtp,
sess->callbacks.request_key_unit);
RTP_SESSION_UNLOCK (sess);
sess->callbacks.request_key_unit (sess, ssrcs[i], 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) {
/* try to find a src with media_ssrc instead */
src = find_source (sess, media_ssrc);
if (src == NULL)
return;
}
rtp_session_request_local_key_unit (sess, src, &media_ssrc, 1, FALSE,
current_time);
}
static void
rtp_session_process_fir (RTPSession * sess, guint32 sender_ssrc,
guint8 * fci_data, guint fci_length, GstClockTime current_time)
{
RTPSource *src;
guint32 ssrc;
guint position = 0;
guint32 ssrcs[32];
guint num_ssrcs = 0;
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 && num_ssrcs < 32) {
ssrcs[num_ssrcs++] = ssrc;
}
}
if (num_ssrcs == 0)
return;
rtp_session_request_local_key_unit (sess, src, ssrcs, num_ssrcs, 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_sr_req (RTPSession * sess, guint32 sender_ssrc,
guint32 media_ssrc)
{
RTPSource *src;
/* Request a new SR in feedback profiles ASAP */
if (sess->rtp_profile != GST_RTP_PROFILE_AVPF
&& sess->rtp_profile != GST_RTP_PROFILE_SAVPF)
return;
src = find_source (sess, sender_ssrc);
/* Our own RTCP packet */
if (src && src->internal)
return;
src = find_source (sess, media_ssrc);
/* Not an SSRC we're producing */
if (!src || !src->internal)
return;
GST_DEBUG_OBJECT (sess, "Handling RTCP-SR-REQ");
/* FIXME: 5s max_delay hard-coded here as we have to give some
* high enough value */
sess->sr_req_pending = TRUE;
rtp_session_send_rtcp (sess, 5 * GST_SECOND);
}
static void
rtp_session_process_twcc (RTPSession * sess, guint32 sender_ssrc,
guint32 media_ssrc, guint8 * fci_data, guint fci_length)
{
GArray *twcc_packets;
GstStructure *twcc_packets_s;
GstStructure *twcc_stats_s;
twcc_packets = rtp_twcc_manager_parse_fci (sess->twcc,
fci_data, fci_length * sizeof (guint32));
if (twcc_packets == NULL)
return;
twcc_packets_s = rtp_twcc_stats_get_packets_structure (twcc_packets);
twcc_stats_s =
rtp_twcc_stats_process_packets (sess->twcc_stats, twcc_packets);
GST_DEBUG_OBJECT (sess, "Parsed TWCC: %" GST_PTR_FORMAT, twcc_packets_s);
GST_INFO_OBJECT (sess, "Current TWCC stats %" GST_PTR_FORMAT, twcc_stats_s);
g_array_unref (twcc_packets);
RTP_SESSION_UNLOCK (sess);
if (sess->callbacks.notify_twcc)
sess->callbacks.notify_twcc (sess, twcc_packets_s, twcc_stats_s,
sess->notify_twcc_user_data);
RTP_SESSION_LOCK (sess);
}
static void
rtp_session_process_feedback (RTPSession * sess, GstRTCPPacket * packet,
RTPPacketInfo * pinfo, GstClockTime current_time)
{
GstRTCPType type;
GstRTCPFBType fbtype;
guint32 sender_ssrc, media_ssrc;
guint8 *fci_data;
guint fci_length;
RTPSource *src;
/* The feedback packet must include both sender SSRC and media SSRC */
if (packet->length < 2)
return;
type = gst_rtcp_packet_get_type (packet);
fbtype = gst_rtcp_packet_fb_get_type (packet);
sender_ssrc = gst_rtcp_packet_fb_get_sender_ssrc (packet);
media_ssrc = gst_rtcp_packet_fb_get_media_ssrc (packet);
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;
if (src)
g_object_ref (src);
fci_data = gst_rtcp_packet_fb_get_fci (packet);
fci_length = gst_rtcp_packet_fb_get_fci_length (packet) * sizeof (guint32);
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 != GST_CLOCK_TIME_NONE) {
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) ||
/* TWCC is for all sources, so a single media-ssrc is not enough */
(type == GST_RTCP_TYPE_RTPFB && fbtype == GST_RTCP_RTPFB_TYPE_TWCC)) {
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, 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;
case GST_RTCP_RTPFB_TYPE_RTCP_SR_REQ:
rtp_session_process_sr_req (sess, sender_ssrc, media_ssrc);
break;
case GST_RTCP_RTPFB_TYPE_TWCC:
rtp_session_process_twcc (sess, sender_ssrc, media_ssrc,
fci_data, fci_length);
break;
default:
break;
}
default:
break;
}
}
if (src)
g_object_unref (src);
}
/**
* 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, GstClockTime running_time, guint64 ntpnstime)
{
GstRTCPPacket packet;
gboolean more, is_bye = FALSE, do_sync = FALSE, has_report = 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,
running_time, 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:
has_report = TRUE;
rtp_session_process_sr (sess, &packet, &pinfo, &do_sync);
break;
case GST_RTCP_TYPE_RR:
has_report = TRUE;
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;
case GST_RTCP_TYPE_XR:
/* FIXME: This block is added to downgrade warning level.
* Once the parser is implemented, it should be replaced with
* a proper process function. */
GST_DEBUG ("got RTCP XR packet, but ignored");
break;
default:
GST_WARNING ("got unknown RTCP packet type: %d", type);
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);
if (has_report) {
g_object_notify_by_pspec (G_OBJECT (sess), properties[PROP_STATS]);
}
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_send_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_send_caps (source, caps);
if (created)
on_new_sender_ssrc (sess, source);
g_object_unref (source);
}
}
RTP_SESSION_UNLOCK (sess);
} else {
sess->internal_ssrc_from_caps_or_property = FALSE;
}
sess->send_ntp64_ext_id =
gst_rtp_get_extmap_id_for_attribute (s,
GST_RTP_HDREXT_BASE GST_RTP_HDREXT_NTP_64);
rtp_twcc_manager_parse_send_ext_id (sess->twcc, s);
}
static void
update_ntp64_header_ext_data (RTPPacketInfo * pinfo, GstBuffer * buffer)
{
GstRTPBuffer rtp = GST_RTP_BUFFER_INIT;
if (gst_rtp_buffer_map (buffer, GST_MAP_READWRITE, &rtp)) {
guint16 bits;
guint8 *data;
guint wordlen;
if (gst_rtp_buffer_get_extension_data (&rtp, &bits, (gpointer *) & data,
&wordlen)) {
gsize len = wordlen * 4;
/* One-byte header */
if (bits == 0xBEDE) {
/* One-byte header extension */
while (TRUE) {
guint8 ext_id, ext_len;
if (len < 1)
break;
ext_id = GST_READ_UINT8 (data) >> 4;
ext_len = (GST_READ_UINT8 (data) & 0xF) + 1;
data += 1;
len -= 1;
if (ext_id == 0) {
/* Skip padding */
continue;
} else if (ext_id == 15) {
/* Stop parsing */
break;
}
/* extension doesn't fit into the header */
if (ext_len > len)
break;
if (ext_id == pinfo->ntp64_ext_id && ext_len == 8) {
if (pinfo->ntpnstime != GST_CLOCK_TIME_NONE) {
guint64 ntptime = gst_util_uint64_scale (pinfo->ntpnstime,
G_GUINT64_CONSTANT (1) << 32,
GST_SECOND);
GST_WRITE_UINT64_BE (data, ntptime);
} else {
/* Replace extension with padding */
memset (data - 1, 0, 1 + ext_len);
}
}
/* skip to the next extension */
data += ext_len;
len -= ext_len;
}
} else if ((bits >> 4) == 0x100) {
/* Two-byte header extension */
while (TRUE) {
guint8 ext_id, ext_len;
if (len < 1)
break;
ext_id = GST_READ_UINT8 (data);
data += 1;
len -= 1;
if (ext_id == 0) {
/* Skip padding */
continue;
}
ext_len = GST_READ_UINT8 (data);
data += 1;
len -= 1;
/* extension doesn't fit into the header */
if (ext_len > len)
break;
if (ext_id == pinfo->ntp64_ext_id && ext_len == 8) {
if (pinfo->ntpnstime != GST_CLOCK_TIME_NONE) {
guint64 ntptime = gst_util_uint64_scale (pinfo->ntpnstime,
G_GUINT64_CONSTANT (1) << 32,
GST_SECOND);
GST_WRITE_UINT64_BE (data, ntptime);
} else {
/* Replace extension with padding */
memset (data - 2, 0, 2 + ext_len);
}
}
/* skip to the next extension */
data += ext_len;
len -= ext_len;
}
}
}
gst_rtp_buffer_unmap (&rtp);
}
}
static void
update_ntp64_header_ext (RTPPacketInfo * pinfo)
{
/* Early return if we don't know the header extension id or the packets
* don't contain the header extension */
if (pinfo->ntp64_ext_id == 0 || !pinfo->have_ntp64_ext)
return;
/* If no NTP time is known then the header extension will be replaced with
* padding, otherwise it will be updated */
GST_TRACE
("Updating NTP-64 header extension for SSRC %08x packet with RTP time %u and running time %"
GST_TIME_FORMAT " to %" GST_TIME_FORMAT, pinfo->ssrc, pinfo->rtptime,
GST_TIME_ARGS (pinfo->running_time), GST_TIME_ARGS (pinfo->ntpnstime));
if (GST_IS_BUFFER_LIST (pinfo->data)) {
GstBufferList *list;
guint i = 0;
pinfo->data = gst_buffer_list_make_writable (pinfo->data);
list = GST_BUFFER_LIST (pinfo->data);
for (i = 0; i < gst_buffer_list_length (list); i++) {
GstBuffer *buffer = gst_buffer_list_get_writable (list, i);
update_ntp64_header_ext_data (pinfo, buffer);
}
} else {
pinfo->data = gst_buffer_make_writable (pinfo->data);
update_ntp64_header_ext_data (pinfo, pinfo->data);
}
}
/**
* 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 data (a buffer or buffer list) in the session manager. This
* function takes ownership of @data.
*
* Returns: a #GstFlowReturn.
*/
GstFlowReturn
rtp_session_send_rtp (RTPSession * sess, gpointer data, gboolean is_list,
GstClockTime current_time, GstClockTime running_time, guint64 ntpnstime)
{
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, ntpnstime))
goto invalid_packet;
/* Update any 64-bit NTP header extensions with the actual NTP time here */
if (sess->update_ntp64_header_ext)
update_ntp64_header_ext (&pinfo);
rtp_twcc_manager_send_packet (sess->twcc, &pinfo);
source = obtain_internal_source (sess, pinfo.ssrc, &created, current_time);
if (created)
on_new_sender_ssrc (sess, source);
if (!source->internal) {
GSocketAddress *from;
if (source->rtp_from)
from = source->rtp_from;
else
from = source->rtcp_from;
if (from) {
if (rtp_session_find_conflicting_address (sess, from, current_time)) {
/* Its a known conflict, its probably a loop, not a collision
* lets just drop the incoming packet
*/
GST_LOG ("Our packets are being looped back to us, ignoring collision");
} else {
GST_DEBUG ("Collision for SSRC %x, change our sender ssrc", pinfo.ssrc);
rtp_session_have_conflict (sess, source, from, current_time);
}
} else {
GST_LOG ("Ignoring collision on sent SSRC %x because remote source"
" doesn't have an address", pinfo.ssrc);
}
/* the the sending source is not internal, we have to drop the packet,
or else we will end up receving it ourselves! */
goto collision;
}
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;
}
collision:
{
g_object_unref (source);
clean_packet_info (&pinfo);
RTP_SESSION_UNLOCK (sess);
GST_WARNING ("non-internal source with same ssrc %08x, drop packet",
pinfo.ssrc);
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;
gboolean is_bye;
GstBuffer *buffer;
} ReportOutput;
typedef struct
{
GstRTCPBuffer rtcpbuf;
RTPSession *sess;
RTPSource *source;
guint num_to_report;
gboolean have_fir;
gboolean have_pli;
gboolean have_nack;
GstBuffer *rtcp;
GstClockTime current_time;
guint64 ntpnstime;
GstClockTime running_time;
GstClockTime interval;
GstRTCPPacket packet;
gboolean has_sdes;
gboolean is_early;
GQueue output;
guint nacked_seqnums;
gboolean timeout_inactive_sources;
} ReportData;
static gboolean
session_start_rtcp (RTPSession * sess, ReportData * data)
{
GstRTCPPacket *packet = &data->packet;
RTPSource *own = data->source;
GstRTCPBuffer *rtcp = &data->rtcpbuf;
data->rtcp = gst_rtcp_buffer_new (sess->mtu);
data->has_sdes = FALSE;
gst_rtcp_buffer_map (data->rtcp, GST_MAP_READWRITE, rtcp);
if (RTP_SOURCE_IS_SENDER (own) && (!data->is_early || !sess->reduced_size_rtcp
|| sess->sr_req_pending)) {
guint64 ntptime;
guint32 rtptime;
guint32 packet_count, octet_count;
sess->sr_req_pending = FALSE;
/* 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 */
if (!rtp_source_get_new_sr (own, data->ntpnstime, data->running_time,
&ntptime, &rtptime, &packet_count, &octet_count)) {
gst_rtcp_buffer_unmap (&data->rtcpbuf);
return FALSE;
}
/* 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,
sess->timestamp_sender_reports ? ntptime : 0,
sess->timestamp_sender_reports ? rtptime : 0,
packet_count, octet_count);
} else if (!data->is_early || !sess->reduced_size_rtcp) {
/* 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);
}
return TRUE;
}
/* 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 remote sources */
if (source->internal)
goto reported;
if (!RTP_SOURCE_IS_SENDER (source)) {
GST_DEBUG ("source %08x not sender", source->ssrc);
goto reported;
}
if (source->disable_rtcp) {
GST_DEBUG ("source %08x has RTCP disabled", 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.ssrc = data->source->ssrc;
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);
}
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;
source->stats.sent_pli_count++;
}
/* construct NACK */
static void
session_nack (const gchar * key, RTPSource * source, ReportData * data)
{
RTPSession *sess = data->sess;
GstRTCPBuffer *rtcp = &data->rtcpbuf;
GstRTCPPacket *packet = &data->packet;
guint16 *nacks;
GstClockTime *nack_deadlines;
guint n_nacks, i = 0;
guint nacked_seqnums = 0;
guint16 n_fb_nacks = 0;
guint8 *fci_data;
if (!source->send_nack)
return;
nacks = rtp_source_get_nacks (source, &n_nacks);
nack_deadlines = rtp_source_get_nack_deadlines (source, NULL);
GST_DEBUG ("%u NACKs current time %" GST_TIME_FORMAT, n_nacks,
GST_TIME_ARGS (data->current_time));
/* cleanup expired nacks */
for (i = 0; i < n_nacks; i++) {
GST_DEBUG ("#%u deadline %" GST_TIME_FORMAT, nacks[i],
GST_TIME_ARGS (nack_deadlines[i]));
if (nack_deadlines[i] >= data->current_time)
break;
}
if (data->is_early) {
/* don't remove them all if this is an early RTCP packet. It may happen
* that the NACKs are late due to high RTT, not sending NACKs at all would
* keep the RTX RTT stats high and maintain a dropping state. */
i = MIN (n_nacks - 1, i);
}
if (i) {
GST_WARNING ("Removing %u expired NACKS", i);
rtp_source_clear_nacks (source, i);
n_nacks -= i;
if (n_nacks == 0)
return;
}
/* allow overriding NACK to packet conversion */
if (g_signal_has_handler_pending (sess,
rtp_session_signals[SIGNAL_ON_SENDING_NACKS], 0, TRUE)) {
/* this is needed as it will actually resize the buffer */
gst_rtcp_buffer_unmap (rtcp);
g_signal_emit (sess, rtp_session_signals[SIGNAL_ON_SENDING_NACKS], 0,
data->source->ssrc, source->ssrc, source->nacks, data->rtcp,
&nacked_seqnums);
/* and now remap for the remaining work */
gst_rtcp_buffer_map (data->rtcp, GST_MAP_READWRITE, rtcp);
if (nacked_seqnums > 0)
goto done;
}
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);
if (!gst_rtcp_packet_fb_set_fci_length (packet, 1)) {
gst_rtcp_packet_remove (packet);
GST_WARNING ("no nacks fit in the packet");
return;
}
fci_data = gst_rtcp_packet_fb_get_fci (packet);
for (i = 0; i < n_nacks; i = nacked_seqnums) {
guint16 seqnum = nacks[i];
guint16 blp = 0;
guint j;
if (!gst_rtcp_packet_fb_set_fci_length (packet, n_fb_nacks + 1))
break;
n_fb_nacks++;
nacked_seqnums++;
for (j = i + 1; j < n_nacks; j++) {
gint diff;
diff = gst_rtp_buffer_compare_seqnum (seqnum, nacks[j]);
GST_TRACE ("[%u][%u] %u %u diff %i", i, j, seqnum, nacks[j], diff);
if (diff > 16)
break;
blp |= 1 << (diff - 1);
nacked_seqnums++;
}
GST_WRITE_UINT32_BE (fci_data, seqnum << 16 | blp);
fci_data += 4;
}
GST_DEBUG ("Sent %u seqnums into %u FB NACKs", nacked_seqnums, n_fb_nacks);
source->stats.sent_nack_count += n_fb_nacks;
done:
data->nacked_seqnums += nacked_seqnums;
rtp_source_clear_nacks (source, nacked_seqnums);
}
/* 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;
}
if (data->timeout_inactive_sources) {
/* 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
&& source->media_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_twcc (const gchar * key, RTPSource * source, ReportData * data)
{
RTPSession *sess = data->sess;
GstBuffer *buf;
/* 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;
/* skip if RTCP is disabled */
if (source->disable_rtcp) {
GST_DEBUG ("source %08x has RTCP disabled", source->ssrc);
return;
}
GST_DEBUG ("generating TWCC feedback for source %08x", source->ssrc);
while ((buf = rtp_twcc_manager_get_feedback (sess->twcc, source->ssrc))) {
ReportOutput *output = g_new (ReportOutput, 1);
output->source = g_object_ref (source);
output->is_bye = FALSE;
output->buffer = buf;
/* queue the RTCP packet to push later */
g_queue_push_tail (&data->output, output);
}
}
static void
generate_rtcp (const gchar * key, RTPSource * source, ReportData * data)
{
RTPSession *sess = data->sess;
gboolean is_bye = FALSE;
ReportOutput *output;
gboolean sr_req_pending = sess->sr_req_pending;
/* 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;
/* skip if RTCP is disabled */
if (source->disable_rtcp) {
GST_DEBUG ("source %08x has RTCP disabled", source->ssrc);
return;
}
data->source = source;
/* open packet */
if (!session_start_rtcp (sess, data)) {
GST_WARNING ("source %08x can not generate RTCP", source->ssrc);
return;
}
if (source->marked_bye) {
/* send BYE */
make_source_bye (sess, source, data);
is_bye = TRUE;
} else if (!data->is_early) {
/* loop over all known sources and add report blocks. If we are early, we
* just make a minimal RTCP packet and skip this step */
g_hash_table_foreach (sess->ssrcs[sess->mask_idx],
(GHFunc) session_report_blocks, data);
}
if (!data->has_sdes && (!data->is_early || !sess->reduced_size_rtcp
|| sr_req_pending))
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_new (ReportOutput, 1);
output->source = g_object_ref (source);
output->is_bye = is_bye;
output->buffer = data->rtcp;
/* queue the RTCP packet to push later */
g_queue_push_tail (&data->output, output);
}
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);
}
}
}
static void
schedule_remaining_nacks (const gchar * key, RTPSource * source,
ReportData * data)
{
RTPSession *sess = data->sess;
GstClockTime *nack_deadlines;
GstClockTime deadline;
guint n_nacks;
if (!source->send_nack)
return;
/* the scheduling is entirely based on available bandwidth, just take the
* biggest seqnum, which will have the largest deadline to request early
* RTCP. */
nack_deadlines = rtp_source_get_nack_deadlines (source, &n_nacks);
deadline = nack_deadlines[n_nacks - 1];
RTP_SESSION_UNLOCK (sess);
rtp_session_send_rtcp_with_deadline (sess, deadline);
RTP_SESSION_LOCK (sess);
}
static gboolean
rtp_session_are_all_sources_bye (RTPSession * sess)
{
GHashTableIter iter;
RTPSource *src;
RTP_SESSION_LOCK (sess);
g_hash_table_iter_init (&iter, sess->ssrcs[sess->mask_idx]);
while (g_hash_table_iter_next (&iter, NULL, (gpointer *) & src)) {
if (src->internal && !src->sent_bye) {
RTP_SESSION_UNLOCK (sess);
return FALSE;
}
}
RTP_SESSION_UNLOCK (sess);
return TRUE;
}
/**
* 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.nacked_seqnums = 0;
data.timeout_inactive_sources = sess->timeout_inactive_sources;
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 after generation */
all_empty = TRUE;
/* Make a local copy of the hashtable. We need to do this because the
* generate_rtcp 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);
GST_DEBUG
("doing RTCP generation %u for %u sources, early %d",
sess->generation, data.num_to_report, data.is_early);
/* generate RTCP for all internal sources, this might release the
* session lock. */
g_hash_table_foreach (table_copy, (GHFunc) generate_rtcp, &data);
g_hash_table_foreach (table_copy, (GHFunc) generate_twcc, &data);
/* update the generation for all the sources that have been reported */
g_hash_table_foreach (table_copy, (GHFunc) update_generation, &data);
g_hash_table_destroy (table_copy);
/* 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_check_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_by_pspec (G_OBJECT (sess), properties[PROP_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);
/* do_not_suppress is ignored because we never ever suppress RTCP packets
* here. Suppression of regular RTCP happens by simply not scheduling this
* function unless needed */
empty_buffer = gst_buffer_get_size (buffer) == 0;
if (!empty_buffer)
all_empty = FALSE;
if (sess->callbacks.send_rtcp && !empty_buffer) {
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,
rtp_session_are_all_sources_bye (sess), 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", sess->callbacks.send_rtcp, empty_buffer);
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_free (output);
}
if (all_empty)
GST_ERROR ("generated empty RTCP messages for all the sources");
/* schedule remaining nacks */
RTP_SESSION_LOCK (sess);
g_hash_table_foreach (sess->ssrcs[sess->mask_idx],
(GHFunc) schedule_remaining_nacks, &data);
RTP_SESSION_UNLOCK (sess);
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_internal (RTPSession * sess, GstClockTime now,
GstClockTime max_delay)
{
/* notify the application that we intend to send early RTCP */
if (sess->callbacks.notify_early_rtcp)
sess->callbacks.notify_early_rtcp (sess, sess->notify_early_rtcp_user_data);
return rtp_session_request_early_rtcp (sess, now, max_delay);
}
static gboolean
rtp_session_send_rtcp_with_deadline (RTPSession * sess, GstClockTime deadline)
{
GstClockTime now, max_delay;
if (!sess->callbacks.send_rtcp)
return FALSE;
now = sess->callbacks.request_time (sess, sess->request_time_user_data);
if (deadline < now)
return FALSE;
max_delay = deadline - now;
return rtp_session_send_rtcp_internal (sess, now, max_delay);
}
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_send_rtcp_internal (sess, now, max_delay);
}
gboolean
rtp_session_request_key_unit (RTPSession * sess, guint32 ssrc,
gboolean fir, gint count)
{
RTPSource *src;
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);
if (!rtp_session_send_rtcp (sess, 5 * GST_SECOND)) {
GST_DEBUG ("FIR/PLI not sent early, sending with next regular RTCP");
}
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;
GstClockTime now;
if (!sess->callbacks.send_rtcp)
return FALSE;
now = sess->callbacks.request_time (sess, sess->request_time_user_data);
RTP_SESSION_LOCK (sess);
source = find_source (sess, ssrc);
if (source == NULL)
goto no_source;
GST_DEBUG ("request NACK for SSRC %08x, #%u, deadline %" GST_TIME_FORMAT,
ssrc, seqnum, GST_TIME_ARGS (now + max_delay));
rtp_source_register_nack (source, seqnum, now + max_delay);
RTP_SESSION_UNLOCK (sess);
if (!rtp_session_send_rtcp_internal (sess, now, max_delay)) {
GST_DEBUG ("NACK not sent early, sending with next regular RTCP");
}
return TRUE;
/* ERRORS */
no_source:
{
RTP_SESSION_UNLOCK (sess);
return FALSE;
}
}
/**
* rtp_session_update_recv_caps_structure:
* @sess: an #RTPSession
* @s: a #GstStructure from a #GstCaps
*
* Update the caps of the receiver in the rtp session.
*/
void
rtp_session_update_recv_caps_structure (RTPSession * sess,
const GstStructure * s)
{
rtp_twcc_manager_parse_recv_ext_id (sess->twcc, s);
}
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