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gstreamer/gst/rtpmanager/rtpsession.c
Wim Taymans 8598aaf81b rtpbin: Get and use the NTP time when receiving RTCP 
When we receive an RTCP packet, get the current NTP time in nanseconds so that
we can correctly calculate the round-trip time.
2011-02-02 18:30:46 +01:00

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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., 59 Temple Place - Suite 330,
* Boston, MA 02111-1307, USA.
*/
#include <string.h>
#include <gst/rtp/gstrtpbuffer.h>
#include <gst/rtp/gstrtcpbuffer.h>
#include <gst/netbuffer/gstnetbuffer.h>
#include "gstrtpbin-marshal.h"
#include "rtpsession.h"
GST_DEBUG_CATEGORY_STATIC (rtp_session_debug);
#define GST_CAT_DEFAULT rtp_session_debug
/* signals and args */
enum
{
SIGNAL_GET_SOURCE_BY_SSRC,
SIGNAL_ON_NEW_SSRC,
SIGNAL_ON_SSRC_COLLISION,
SIGNAL_ON_SSRC_VALIDATED,
SIGNAL_ON_SSRC_ACTIVE,
SIGNAL_ON_SSRC_SDES,
SIGNAL_ON_BYE_SSRC,
SIGNAL_ON_BYE_TIMEOUT,
SIGNAL_ON_TIMEOUT,
SIGNAL_ON_SENDER_TIMEOUT,
SIGNAL_ON_SENDING_RTCP,
SIGNAL_ON_FEEDBACK_RTCP,
SIGNAL_SEND_RTCP,
LAST_SIGNAL
};
#define DEFAULT_INTERNAL_SOURCE NULL
#define DEFAULT_BANDWIDTH RTP_STATS_BANDWIDTH
#define DEFAULT_RTCP_FRACTION (RTP_STATS_RTCP_FRACTION * RTP_STATS_BANDWIDTH)
#define DEFAULT_RTCP_RR_BANDWIDTH -1
#define DEFAULT_RTCP_RS_BANDWIDTH -1
#define DEFAULT_RTCP_MTU 1400
#define DEFAULT_SDES NULL
#define DEFAULT_NUM_SOURCES 0
#define DEFAULT_NUM_ACTIVE_SOURCES 0
#define DEFAULT_SOURCES NULL
#define DEFAULT_RTCP_MIN_INTERVAL (RTP_STATS_MIN_INTERVAL * GST_SECOND)
#define DEFAULT_RTCP_FEEDBACK_RETENTION_WINDOW (2 * GST_SECOND)
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_LAST
};
/* update average packet size */
#define INIT_AVG(avg, val) \
(avg) = (val);
#define UPDATE_AVG(avg, val) \
if ((avg) == 0) \
(avg) = (val); \
else \
(avg) = ((val) + (15 * (avg))) >> 4;
/* The number RTCP intervals after which to timeout entries in the
* collision table
*/
#define RTCP_INTERVAL_COLLISION_TIMEOUT 10
/* GObject vmethods */
static void rtp_session_finalize (GObject * object);
static void rtp_session_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec);
static void rtp_session_get_property (GObject * object, guint prop_id,
GValue * value, GParamSpec * pspec);
static gboolean rtp_session_on_sending_rtcp (RTPSession * sess,
GstBuffer * buffer, gboolean early);
static void rtp_session_send_rtcp (RTPSession * sess,
GstClockTimeDiff max_delay);
static guint rtp_session_signals[LAST_SIGNAL] = { 0 };
G_DEFINE_TYPE (RTPSession, rtp_session, G_TYPE_OBJECT);
static RTPSource *obtain_source (RTPSession * sess, guint32 ssrc,
gboolean * created, RTPArrivalStats * arrival, gboolean rtp);
static GstFlowReturn rtp_session_schedule_bye_locked (RTPSession * sess,
const gchar * reason, GstClockTime current_time);
static GstClockTime calculate_rtcp_interval (RTPSession * sess,
gboolean deterministic, gboolean first);
static gboolean
accumulate_trues (GSignalInvocationHint * ihint, GValue * return_accu,
const GValue * handler_return, gpointer data)
{
if (g_value_get_boolean (handler_return))
g_value_set_boolean (return_accu, TRUE);
return TRUE;
}
static void
rtp_session_class_init (RTPSessionClass * klass)
{
GObjectClass *gobject_class;
gobject_class = (GObjectClass *) klass;
gobject_class->finalize = rtp_session_finalize;
gobject_class->set_property = rtp_session_set_property;
gobject_class->get_property = rtp_session_get_property;
/**
* RTPSession::get-source-by-ssrc:
* @session: the object which received the signal
* @ssrc: the SSRC of the RTPSource
*
* Request the #RTPSource object with SSRC @ssrc in @session.
*/
rtp_session_signals[SIGNAL_GET_SOURCE_BY_SSRC] =
g_signal_new ("get-source-by-ssrc", G_TYPE_FROM_CLASS (klass),
G_SIGNAL_RUN_LAST | G_SIGNAL_ACTION, G_STRUCT_OFFSET (RTPSessionClass,
get_source_by_ssrc), NULL, NULL, gst_rtp_bin_marshal_OBJECT__UINT,
RTP_TYPE_SOURCE, 1, G_TYPE_UINT);
/**
* RTPSession::on-new-ssrc:
* @session: the object which received the signal
* @src: the new RTPSource
*
* Notify of a new SSRC that entered @session.
*/
rtp_session_signals[SIGNAL_ON_NEW_SSRC] =
g_signal_new ("on-new-ssrc", G_TYPE_FROM_CLASS (klass),
G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (RTPSessionClass, on_new_ssrc),
NULL, NULL, g_cclosure_marshal_VOID__OBJECT, G_TYPE_NONE, 1,
RTP_TYPE_SOURCE);
/**
* RTPSession::on-ssrc-collision:
* @session: the object which received the signal
* @src: the #RTPSource that caused a collision
*
* Notify when we have an SSRC collision
*/
rtp_session_signals[SIGNAL_ON_SSRC_COLLISION] =
g_signal_new ("on-ssrc-collision", G_TYPE_FROM_CLASS (klass),
G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (RTPSessionClass, on_ssrc_collision),
NULL, NULL, g_cclosure_marshal_VOID__OBJECT, G_TYPE_NONE, 1,
RTP_TYPE_SOURCE);
/**
* RTPSession::on-ssrc-validated:
* @session: the object which received the signal
* @src: the new validated RTPSource
*
* Notify of a new SSRC that became validated.
*/
rtp_session_signals[SIGNAL_ON_SSRC_VALIDATED] =
g_signal_new ("on-ssrc-validated", G_TYPE_FROM_CLASS (klass),
G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (RTPSessionClass, on_ssrc_validated),
NULL, NULL, g_cclosure_marshal_VOID__OBJECT, G_TYPE_NONE, 1,
RTP_TYPE_SOURCE);
/**
* RTPSession::on-ssrc-active:
* @session: the object which received the signal
* @src: the active RTPSource
*
* Notify of a SSRC that is active, i.e., sending RTCP.
*/
rtp_session_signals[SIGNAL_ON_SSRC_ACTIVE] =
g_signal_new ("on-ssrc-active", G_TYPE_FROM_CLASS (klass),
G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (RTPSessionClass, on_ssrc_active),
NULL, NULL, g_cclosure_marshal_VOID__OBJECT, G_TYPE_NONE, 1,
RTP_TYPE_SOURCE);
/**
* RTPSession::on-ssrc-sdes:
* @session: the object which received the signal
* @src: the RTPSource
*
* Notify that a new SDES was received for SSRC.
*/
rtp_session_signals[SIGNAL_ON_SSRC_SDES] =
g_signal_new ("on-ssrc-sdes", G_TYPE_FROM_CLASS (klass),
G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (RTPSessionClass, on_ssrc_sdes),
NULL, NULL, g_cclosure_marshal_VOID__OBJECT, G_TYPE_NONE, 1,
RTP_TYPE_SOURCE);
/**
* RTPSession::on-bye-ssrc:
* @session: the object which received the signal
* @src: the RTPSource that went away
*
* Notify of an SSRC that became inactive because of a BYE packet.
*/
rtp_session_signals[SIGNAL_ON_BYE_SSRC] =
g_signal_new ("on-bye-ssrc", G_TYPE_FROM_CLASS (klass),
G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (RTPSessionClass, on_bye_ssrc),
NULL, NULL, g_cclosure_marshal_VOID__OBJECT, G_TYPE_NONE, 1,
RTP_TYPE_SOURCE);
/**
* RTPSession::on-bye-timeout:
* @session: the object which received the signal
* @src: the RTPSource that timed out
*
* Notify of an SSRC that has timed out because of BYE
*/
rtp_session_signals[SIGNAL_ON_BYE_TIMEOUT] =
g_signal_new ("on-bye-timeout", G_TYPE_FROM_CLASS (klass),
G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (RTPSessionClass, on_bye_timeout),
NULL, NULL, g_cclosure_marshal_VOID__OBJECT, G_TYPE_NONE, 1,
RTP_TYPE_SOURCE);
/**
* RTPSession::on-timeout:
* @session: the object which received the signal
* @src: the RTPSource that timed out
*
* Notify of an SSRC that has timed out
*/
rtp_session_signals[SIGNAL_ON_TIMEOUT] =
g_signal_new ("on-timeout", G_TYPE_FROM_CLASS (klass),
G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (RTPSessionClass, on_timeout),
NULL, NULL, g_cclosure_marshal_VOID__OBJECT, G_TYPE_NONE, 1,
RTP_TYPE_SOURCE);
/**
* RTPSession::on-sender-timeout:
* @session: the object which received the signal
* @src: the RTPSource that timed out
*
* Notify of an SSRC that was a sender but timed out and became a receiver.
*/
rtp_session_signals[SIGNAL_ON_SENDER_TIMEOUT] =
g_signal_new ("on-sender-timeout", G_TYPE_FROM_CLASS (klass),
G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (RTPSessionClass, on_sender_timeout),
NULL, NULL, g_cclosure_marshal_VOID__OBJECT, G_TYPE_NONE, 1,
RTP_TYPE_SOURCE);
/**
* RTPSession::on-sending-rtcp
* @session: the object which received the signal
* @buffer: the #GstBuffer containing the RTCP packet about to be sent
* @early: %TRUE if the packet is early, %FALSE if it is regular
*
* This signal is emitted before sending an RTCP packet, it can be used
* to add extra RTCP Packets.
*
* Returns: %TRUE if the RTCP buffer should NOT be suppressed, %FALSE
* if suppressing it is acceptable
*/
rtp_session_signals[SIGNAL_ON_SENDING_RTCP] =
g_signal_new ("on-sending-rtcp", G_TYPE_FROM_CLASS (klass),
G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (RTPSessionClass, on_sending_rtcp),
accumulate_trues, NULL, gst_rtp_bin_marshal_BOOLEAN__POINTER_BOOLEAN,
G_TYPE_BOOLEAN, 2, G_TYPE_POINTER, G_TYPE_BOOLEAN);
/**
* RTPSession::on-feedback-rtcp:
* @session: the object which received the signal
* @type: Type of RTCP packet, will be %GST_RTCP_TYPE_RTPFB or
* %GST_RTCP_TYPE_RTPFB
* @fbtype: The type of RTCP FB packet, probably part of #GstRTCPFBType
* @sender_ssrc: The SSRC of the sender
* @media_ssrc: The SSRC of the media this refers to
* @fci: a #GstBuffer with the FCI data from the FB packet or %NULL if
* there was no FCI
*
* Notify that a RTCP feedback packet has been received
*/
rtp_session_signals[SIGNAL_ON_FEEDBACK_RTCP] =
g_signal_new ("on-feedback-rtcp", G_TYPE_FROM_CLASS (klass),
G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (RTPSessionClass, on_feedback_rtcp),
NULL, NULL, gst_rtp_bin_marshal_VOID__UINT_UINT_UINT_UINT_POINTER,
G_TYPE_NONE, 4, G_TYPE_UINT, G_TYPE_UINT, G_TYPE_UINT, G_TYPE_UINT,
G_TYPE_POINTER);
/**
* RTPSession::send-rtcp:
* @session: the object which received the signal
* @max_delay: The maximum delay after which the feedback will not be useful
* anymore
*
* Requests that the #RTPSession initiate a new RTCP packet as soon as
* possible within the requested delay.
*/
rtp_session_signals[SIGNAL_SEND_RTCP] =
g_signal_new ("send-rtcp", G_TYPE_FROM_CLASS (klass),
G_SIGNAL_RUN_LAST | G_SIGNAL_ACTION,
G_STRUCT_OFFSET (RTPSessionClass, send_rtcp), NULL, NULL,
gst_rtp_bin_marshal_VOID__UINT64, G_TYPE_NONE, 1, G_TYPE_UINT64);
g_object_class_install_property (gobject_class, PROP_INTERNAL_SSRC,
g_param_spec_uint ("internal-ssrc", "Internal SSRC",
"The internal SSRC used for the session",
0, G_MAXUINT, 0, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (gobject_class, PROP_INTERNAL_SOURCE,
g_param_spec_object ("internal-source", "Internal Source",
"The internal source element of the session",
RTP_TYPE_SOURCE, G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (gobject_class, PROP_BANDWIDTH,
g_param_spec_double ("bandwidth", "Bandwidth",
"The bandwidth of the session (0 for auto-discover)",
0.0, G_MAXDOUBLE, DEFAULT_BANDWIDTH,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (gobject_class, PROP_RTCP_FRACTION,
g_param_spec_double ("rtcp-fraction", "RTCP Fraction",
"The fraction of the bandwidth used for RTCP (or as a real fraction of the RTP bandwidth if < 1)",
0.0, G_MAXDOUBLE, DEFAULT_RTCP_FRACTION,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (gobject_class, PROP_RTCP_RR_BANDWIDTH,
g_param_spec_int ("rtcp-rr-bandwidth", "RTCP RR bandwidth",
"The RTCP bandwidth used for receivers in bytes per second (-1 = default)",
-1, G_MAXINT, DEFAULT_RTCP_RR_BANDWIDTH,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (gobject_class, PROP_RTCP_RS_BANDWIDTH,
g_param_spec_int ("rtcp-rs-bandwidth", "RTCP RS bandwidth",
"The RTCP bandwidth used for senders in bytes per second (-1 = default)",
-1, G_MAXINT, DEFAULT_RTCP_RS_BANDWIDTH,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (gobject_class, PROP_RTCP_MTU,
g_param_spec_uint ("rtcp-mtu", "RTCP MTU",
"The maximum size of the RTCP packets",
16, G_MAXINT16, DEFAULT_RTCP_MTU,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (gobject_class, PROP_SDES,
g_param_spec_boxed ("sdes", "SDES",
"The SDES items of this session",
GST_TYPE_STRUCTURE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (gobject_class, PROP_NUM_SOURCES,
g_param_spec_uint ("num-sources", "Num Sources",
"The number of sources in the session", 0, G_MAXUINT,
DEFAULT_NUM_SOURCES, G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (gobject_class, PROP_NUM_ACTIVE_SOURCES,
g_param_spec_uint ("num-active-sources", "Num Active Sources",
"The number of active sources in the session", 0, G_MAXUINT,
DEFAULT_NUM_ACTIVE_SOURCES,
G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
/**
* RTPSource::sources
*
* Get a GValue Array of all sources in the session.
*
* <example>
* <title>Getting the #RTPSources of a session
* <programlisting>
* {
* GValueArray *arr;
* GValue *val;
* guint i;
*
* g_object_get (sess, "sources", &arr, NULL);
*
* for (i = 0; i < arr->n_values; i++) {
* RTPSource *source;
*
* val = g_value_array_get_nth (arr, i);
* source = g_value_get_object (val);
* }
* g_value_array_free (arr);
* }
* </programlisting>
* </example>
*/
g_object_class_install_property (gobject_class, PROP_SOURCES,
g_param_spec_boxed ("sources", "Sources",
"An array of all known sources in the session",
G_TYPE_VALUE_ARRAY, G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (gobject_class, PROP_FAVOR_NEW,
g_param_spec_boolean ("favor-new", "Favor new sources",
"Resolve SSRC conflict in favor of new sources", FALSE,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (gobject_class, PROP_RTCP_MIN_INTERVAL,
g_param_spec_uint64 ("rtcp-min-interval", "Minimum RTCP interval",
"Minimum interval between Regular RTCP packet (in ns)",
0, G_MAXUINT64, DEFAULT_RTCP_MIN_INTERVAL,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (gobject_class,
PROP_RTCP_FEEDBACK_RETENTION_WINDOW,
g_param_spec_uint64 ("rtcp-feedback-retention-window",
"RTCP Feedback retention window",
"Duration during which RTCP Feedback packets are retained (in ns)",
0, G_MAXUINT64, DEFAULT_RTCP_FEEDBACK_RETENTION_WINDOW,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
klass->get_source_by_ssrc =
GST_DEBUG_FUNCPTR (rtp_session_get_source_by_ssrc);
klass->on_sending_rtcp = GST_DEBUG_FUNCPTR (rtp_session_on_sending_rtcp);
klass->send_rtcp = GST_DEBUG_FUNCPTR (rtp_session_send_rtcp);
GST_DEBUG_CATEGORY_INIT (rtp_session_debug, "rtpsession", 0, "RTP Session");
}
static void
rtp_session_init (RTPSession * sess)
{
gint i;
gchar *str;
sess->lock = g_mutex_new ();
sess->key = g_random_int ();
sess->mask_idx = 0;
sess->mask = 0;
for (i = 0; i < 32; i++) {
sess->ssrcs[i] =
g_hash_table_new_full (NULL, NULL, NULL,
(GDestroyNotify) g_object_unref);
}
sess->cnames = g_hash_table_new_full (NULL, NULL, g_free, NULL);
rtp_stats_init_defaults (&sess->stats);
sess->recalc_bandwidth = TRUE;
sess->bandwidth = DEFAULT_BANDWIDTH;
sess->rtcp_bandwidth = DEFAULT_RTCP_FRACTION;
sess->rtcp_rr_bandwidth = DEFAULT_RTCP_RR_BANDWIDTH;
sess->rtcp_rs_bandwidth = DEFAULT_RTCP_RS_BANDWIDTH;
/* create an active SSRC for this session manager */
sess->source = rtp_session_create_source (sess);
sess->source->validated = TRUE;
sess->source->internal = TRUE;
sess->stats.active_sources++;
INIT_AVG (sess->stats.avg_rtcp_packet_size, 100);
/* default UDP header length */
sess->header_len = 28;
sess->mtu = DEFAULT_RTCP_MTU;
/* some default SDES entries */
str = g_strdup_printf ("%s@%s", g_get_user_name (), g_get_host_name ());
rtp_source_set_sdes_string (sess->source, GST_RTCP_SDES_CNAME, str);
g_free (str);
rtp_source_set_sdes_string (sess->source, GST_RTCP_SDES_NAME,
g_get_real_name ());
rtp_source_set_sdes_string (sess->source, GST_RTCP_SDES_TOOL, "GStreamer");
sess->first_rtcp = TRUE;
sess->allow_early = TRUE;
sess->rtcp_feedback_retention_window = DEFAULT_RTCP_FEEDBACK_RETENTION_WINDOW;
sess->rtcp_pli_requests = g_array_new (FALSE, FALSE, sizeof (guint32));
GST_DEBUG ("%p: session using SSRC: %08x", sess, sess->source->ssrc);
}
static void
rtp_session_finalize (GObject * object)
{
RTPSession *sess;
gint i;
sess = RTP_SESSION_CAST (object);
g_mutex_free (sess->lock);
for (i = 0; i < 32; i++)
g_hash_table_destroy (sess->ssrcs[i]);
g_free (sess->bye_reason);
g_hash_table_destroy (sess->cnames);
g_object_unref (sess->source);
g_array_free (sess->rtcp_pli_requests, TRUE);
G_OBJECT_CLASS (rtp_session_parent_class)->finalize (object);
}
static void
copy_source (gpointer key, RTPSource * source, GValueArray * arr)
{
GValue value = { 0 };
g_value_init (&value, RTP_TYPE_SOURCE);
g_value_take_object (&value, source);
/* copies the value */
g_value_array_append (arr, &value);
}
static GValueArray *
rtp_session_create_sources (RTPSession * sess)
{
GValueArray *res;
guint size;
RTP_SESSION_LOCK (sess);
/* get number of elements in the table */
size = g_hash_table_size (sess->ssrcs[sess->mask_idx]);
/* create the result value array */
res = g_value_array_new (size);
/* and copy all values into the array */
g_hash_table_foreach (sess->ssrcs[sess->mask_idx], (GHFunc) copy_source, res);
RTP_SESSION_UNLOCK (sess);
return res;
}
static void
rtp_session_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec)
{
RTPSession *sess;
sess = RTP_SESSION (object);
switch (prop_id) {
case PROP_INTERNAL_SSRC:
rtp_session_set_internal_ssrc (sess, g_value_get_uint (value));
break;
case PROP_BANDWIDTH:
sess->bandwidth = g_value_get_double (value);
sess->recalc_bandwidth = TRUE;
break;
case PROP_RTCP_FRACTION:
sess->rtcp_bandwidth = g_value_get_double (value);
sess->recalc_bandwidth = TRUE;
break;
case PROP_RTCP_RR_BANDWIDTH:
sess->rtcp_rr_bandwidth = g_value_get_int (value);
sess->recalc_bandwidth = TRUE;
break;
case PROP_RTCP_RS_BANDWIDTH:
sess->rtcp_rs_bandwidth = g_value_get_int (value);
sess->recalc_bandwidth = TRUE;
break;
case PROP_RTCP_MTU:
sess->mtu = g_value_get_uint (value);
break;
case PROP_SDES:
rtp_session_set_sdes_struct (sess, g_value_get_boxed (value));
break;
case PROP_FAVOR_NEW:
sess->favor_new = g_value_get_boolean (value);
break;
case PROP_RTCP_MIN_INTERVAL:
rtp_stats_set_min_interval (&sess->stats,
(gdouble) g_value_get_uint64 (value) / GST_SECOND);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
static void
rtp_session_get_property (GObject * object, guint prop_id,
GValue * value, GParamSpec * pspec)
{
RTPSession *sess;
sess = RTP_SESSION (object);
switch (prop_id) {
case PROP_INTERNAL_SSRC:
g_value_set_uint (value, rtp_session_get_internal_ssrc (sess));
break;
case PROP_INTERNAL_SOURCE:
g_value_take_object (value, rtp_session_get_internal_source (sess));
break;
case PROP_BANDWIDTH:
g_value_set_double (value, sess->bandwidth);
break;
case PROP_RTCP_FRACTION:
g_value_set_double (value, sess->rtcp_bandwidth);
break;
case PROP_RTCP_RR_BANDWIDTH:
g_value_set_int (value, sess->rtcp_rr_bandwidth);
break;
case PROP_RTCP_RS_BANDWIDTH:
g_value_set_int (value, sess->rtcp_rs_bandwidth);
break;
case PROP_RTCP_MTU:
g_value_set_uint (value, sess->mtu);
break;
case PROP_SDES:
g_value_take_boxed (value, rtp_session_get_sdes_struct (sess));
break;
case PROP_NUM_SOURCES:
g_value_set_uint (value, rtp_session_get_num_sources (sess));
break;
case PROP_NUM_ACTIVE_SOURCES:
g_value_set_uint (value, rtp_session_get_num_active_sources (sess));
break;
case PROP_SOURCES:
g_value_take_boxed (value, rtp_session_create_sources (sess));
break;
case PROP_FAVOR_NEW:
g_value_set_boolean (value, sess->favor_new);
break;
case PROP_RTCP_MIN_INTERVAL:
g_value_set_uint64 (value, sess->stats.min_interval * GST_SECOND);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
static void
on_new_ssrc (RTPSession * sess, RTPSource * source)
{
g_object_ref (source);
RTP_SESSION_UNLOCK (sess);
g_signal_emit (sess, rtp_session_signals[SIGNAL_ON_NEW_SSRC], 0, source);
RTP_SESSION_LOCK (sess);
g_object_unref (source);
}
static void
on_ssrc_collision (RTPSession * sess, RTPSource * source)
{
g_object_ref (source);
RTP_SESSION_UNLOCK (sess);
g_signal_emit (sess, rtp_session_signals[SIGNAL_ON_SSRC_COLLISION], 0,
source);
RTP_SESSION_LOCK (sess);
g_object_unref (source);
}
static void
on_ssrc_validated (RTPSession * sess, RTPSource * source)
{
g_object_ref (source);
RTP_SESSION_UNLOCK (sess);
g_signal_emit (sess, rtp_session_signals[SIGNAL_ON_SSRC_VALIDATED], 0,
source);
RTP_SESSION_LOCK (sess);
g_object_unref (source);
}
static void
on_ssrc_active (RTPSession * sess, RTPSource * source)
{
g_object_ref (source);
RTP_SESSION_UNLOCK (sess);
g_signal_emit (sess, rtp_session_signals[SIGNAL_ON_SSRC_ACTIVE], 0, source);
RTP_SESSION_LOCK (sess);
g_object_unref (source);
}
static void
on_ssrc_sdes (RTPSession * sess, RTPSource * source)
{
g_object_ref (source);
GST_DEBUG ("SDES changed for SSRC %08x", source->ssrc);
RTP_SESSION_UNLOCK (sess);
g_signal_emit (sess, rtp_session_signals[SIGNAL_ON_SSRC_SDES], 0, source);
RTP_SESSION_LOCK (sess);
g_object_unref (source);
}
static void
on_bye_ssrc (RTPSession * sess, RTPSource * source)
{
g_object_ref (source);
RTP_SESSION_UNLOCK (sess);
g_signal_emit (sess, rtp_session_signals[SIGNAL_ON_BYE_SSRC], 0, source);
RTP_SESSION_LOCK (sess);
g_object_unref (source);
}
static void
on_bye_timeout (RTPSession * sess, RTPSource * source)
{
g_object_ref (source);
RTP_SESSION_UNLOCK (sess);
g_signal_emit (sess, rtp_session_signals[SIGNAL_ON_BYE_TIMEOUT], 0, source);
RTP_SESSION_LOCK (sess);
g_object_unref (source);
}
static void
on_timeout (RTPSession * sess, RTPSource * source)
{
g_object_ref (source);
RTP_SESSION_UNLOCK (sess);
g_signal_emit (sess, rtp_session_signals[SIGNAL_ON_TIMEOUT], 0, source);
RTP_SESSION_LOCK (sess);
g_object_unref (source);
}
static void
on_sender_timeout (RTPSession * sess, RTPSource * source)
{
g_object_ref (source);
RTP_SESSION_UNLOCK (sess);
g_signal_emit (sess, rtp_session_signals[SIGNAL_ON_SENDER_TIMEOUT], 0,
source);
RTP_SESSION_LOCK (sess);
g_object_unref (source);
}
/**
* rtp_session_new:
*
* Create a new session object.
*
* Returns: a new #RTPSession. g_object_unref() after usage.
*/
RTPSession *
rtp_session_new (void)
{
RTPSession *sess;
sess = g_object_new (RTP_TYPE_SESSION, NULL);
return sess;
}
/**
* rtp_session_set_callbacks:
* @sess: an #RTPSession
* @callbacks: callbacks to configure
* @user_data: user data passed in the callbacks
*
* Configure a set of callbacks to be notified of actions.
*/
void
rtp_session_set_callbacks (RTPSession * sess, RTPSessionCallbacks * callbacks,
gpointer user_data)
{
g_return_if_fail (RTP_IS_SESSION (sess));
if (callbacks->process_rtp) {
sess->callbacks.process_rtp = callbacks->process_rtp;
sess->process_rtp_user_data = user_data;
}
if (callbacks->send_rtp) {
sess->callbacks.send_rtp = callbacks->send_rtp;
sess->send_rtp_user_data = user_data;
}
if (callbacks->send_rtcp) {
sess->callbacks.send_rtcp = callbacks->send_rtcp;
sess->send_rtcp_user_data = user_data;
}
if (callbacks->sync_rtcp) {
sess->callbacks.sync_rtcp = callbacks->sync_rtcp;
sess->sync_rtcp_user_data = user_data;
}
if (callbacks->clock_rate) {
sess->callbacks.clock_rate = callbacks->clock_rate;
sess->clock_rate_user_data = user_data;
}
if (callbacks->reconsider) {
sess->callbacks.reconsider = callbacks->reconsider;
sess->reconsider_user_data = user_data;
}
if (callbacks->request_key_unit) {
sess->callbacks.request_key_unit = callbacks->request_key_unit;
sess->request_key_unit_user_data = user_data;
}
if (callbacks->request_time) {
sess->callbacks.request_time = callbacks->request_time;
sess->request_time_user_data = user_data;
}
}
/**
* rtp_session_set_process_rtp_callback:
* @sess: an #RTPSession
* @callback: callback to set
* @user_data: user data passed in the callback
*
* Configure only the process_rtp callback to be notified of the process_rtp action.
*/
void
rtp_session_set_process_rtp_callback (RTPSession * sess,
RTPSessionProcessRTP callback, gpointer user_data)
{
g_return_if_fail (RTP_IS_SESSION (sess));
sess->callbacks.process_rtp = callback;
sess->process_rtp_user_data = user_data;
}
/**
* rtp_session_set_send_rtp_callback:
* @sess: an #RTPSession
* @callback: callback to set
* @user_data: user data passed in the callback
*
* Configure only the send_rtp callback to be notified of the send_rtp action.
*/
void
rtp_session_set_send_rtp_callback (RTPSession * sess,
RTPSessionSendRTP callback, gpointer user_data)
{
g_return_if_fail (RTP_IS_SESSION (sess));
sess->callbacks.send_rtp = callback;
sess->send_rtp_user_data = user_data;
}
/**
* rtp_session_set_send_rtcp_callback:
* @sess: an #RTPSession
* @callback: callback to set
* @user_data: user data passed in the callback
*
* Configure only the send_rtcp callback to be notified of the send_rtcp action.
*/
void
rtp_session_set_send_rtcp_callback (RTPSession * sess,
RTPSessionSendRTCP callback, gpointer user_data)
{
g_return_if_fail (RTP_IS_SESSION (sess));
sess->callbacks.send_rtcp = callback;
sess->send_rtcp_user_data = user_data;
}
/**
* rtp_session_set_sync_rtcp_callback:
* @sess: an #RTPSession
* @callback: callback to set
* @user_data: user data passed in the callback
*
* Configure only the sync_rtcp callback to be notified of the sync_rtcp action.
*/
void
rtp_session_set_sync_rtcp_callback (RTPSession * sess,
RTPSessionSyncRTCP callback, gpointer user_data)
{
g_return_if_fail (RTP_IS_SESSION (sess));
sess->callbacks.sync_rtcp = callback;
sess->sync_rtcp_user_data = user_data;
}
/**
* rtp_session_set_clock_rate_callback:
* @sess: an #RTPSession
* @callback: callback to set
* @user_data: user data passed in the callback
*
* Configure only the clock_rate callback to be notified of the clock_rate action.
*/
void
rtp_session_set_clock_rate_callback (RTPSession * sess,
RTPSessionClockRate callback, gpointer user_data)
{
g_return_if_fail (RTP_IS_SESSION (sess));
sess->callbacks.clock_rate = callback;
sess->clock_rate_user_data = user_data;
}
/**
* rtp_session_set_reconsider_callback:
* @sess: an #RTPSession
* @callback: callback to set
* @user_data: user data passed in the callback
*
* Configure only the reconsider callback to be notified of the reconsider action.
*/
void
rtp_session_set_reconsider_callback (RTPSession * sess,
RTPSessionReconsider callback, gpointer user_data)
{
g_return_if_fail (RTP_IS_SESSION (sess));
sess->callbacks.reconsider = callback;
sess->reconsider_user_data = user_data;
}
/**
* rtp_session_set_request_time_callback:
* @sess: an #RTPSession
* @callback: callback to set
* @user_data: user data passed in the callback
*
* Configure only the request_time callback
*/
void
rtp_session_set_request_time_callback (RTPSession * sess,
RTPSessionRequestTime callback, gpointer user_data)
{
g_return_if_fail (RTP_IS_SESSION (sess));
sess->callbacks.request_time = callback;
sess->request_time_user_data = user_data;
}
/**
* rtp_session_set_bandwidth:
* @sess: an #RTPSession
* @bandwidth: the bandwidth allocated
*
* Set the session bandwidth in bytes per second.
*/
void
rtp_session_set_bandwidth (RTPSession * sess, gdouble bandwidth)
{
g_return_if_fail (RTP_IS_SESSION (sess));
RTP_SESSION_LOCK (sess);
sess->stats.bandwidth = bandwidth;
RTP_SESSION_UNLOCK (sess);
}
/**
* rtp_session_get_bandwidth:
* @sess: an #RTPSession
*
* Get the session bandwidth.
*
* Returns: the session bandwidth.
*/
gdouble
rtp_session_get_bandwidth (RTPSession * sess)
{
gdouble result;
g_return_val_if_fail (RTP_IS_SESSION (sess), 0);
RTP_SESSION_LOCK (sess);
result = sess->stats.bandwidth;
RTP_SESSION_UNLOCK (sess);
return result;
}
/**
* rtp_session_set_rtcp_fraction:
* @sess: an #RTPSession
* @bandwidth: the RTCP bandwidth
*
* Set the bandwidth in bytes per second that should be used for RTCP
* messages.
*/
void
rtp_session_set_rtcp_fraction (RTPSession * sess, gdouble bandwidth)
{
g_return_if_fail (RTP_IS_SESSION (sess));
RTP_SESSION_LOCK (sess);
sess->stats.rtcp_bandwidth = bandwidth;
RTP_SESSION_UNLOCK (sess);
}
/**
* rtp_session_get_rtcp_fraction:
* @sess: an #RTPSession
*
* Get the session bandwidth used for RTCP.
*
* Returns: The bandwidth used for RTCP messages.
*/
gdouble
rtp_session_get_rtcp_fraction (RTPSession * sess)
{
gdouble result;
g_return_val_if_fail (RTP_IS_SESSION (sess), 0.0);
RTP_SESSION_LOCK (sess);
result = sess->stats.rtcp_bandwidth;
RTP_SESSION_UNLOCK (sess);
return result;
}
/**
* rtp_session_set_sdes_string:
* @sess: an #RTPSession
* @type: the type of the SDES item
* @item: a null-terminated string to set.
*
* Store an SDES item of @type in @sess.
*
* Returns: %FALSE if the data was unchanged @type is invalid.
*/
gboolean
rtp_session_set_sdes_string (RTPSession * sess, GstRTCPSDESType type,
const gchar * item)
{
gboolean result;
g_return_val_if_fail (RTP_IS_SESSION (sess), FALSE);
RTP_SESSION_LOCK (sess);
result = rtp_source_set_sdes_string (sess->source, type, item);
RTP_SESSION_UNLOCK (sess);
return result;
}
/**
* rtp_session_get_sdes_string:
* @sess: an #RTPSession
* @type: the type of the SDES item
*
* Get the SDES item of @type from @sess.
*
* Returns: a null-terminated copy of the SDES item or NULL when @type was not
* valid. g_free() after usage.
*/
gchar *
rtp_session_get_sdes_string (RTPSession * sess, GstRTCPSDESType type)
{
gchar *result;
g_return_val_if_fail (RTP_IS_SESSION (sess), NULL);
RTP_SESSION_LOCK (sess);
result = rtp_source_get_sdes_string (sess->source, type);
RTP_SESSION_UNLOCK (sess);
return result;
}
/**
* rtp_session_get_sdes_struct:
* @sess: an #RTSPSession
*
* Get the SDES data as a #GstStructure
*
* Returns: a GstStructure with SDES items for @sess. This function returns a
* copy of the SDES structure, use gst_structure_free() after usage.
*/
GstStructure *
rtp_session_get_sdes_struct (RTPSession * sess)
{
const GstStructure *sdes;
GstStructure *result = NULL;
g_return_val_if_fail (RTP_IS_SESSION (sess), NULL);
RTP_SESSION_LOCK (sess);
sdes = rtp_source_get_sdes_struct (sess->source);
if (sdes)
result = gst_structure_copy (sdes);
RTP_SESSION_UNLOCK (sess);
return result;
}
/**
* rtp_session_set_sdes_struct:
* @sess: an #RTSPSession
* @sdes: a #GstStructure
*
* Set the SDES data as a #GstStructure. This function makes a copy of @sdes.
*/
void
rtp_session_set_sdes_struct (RTPSession * sess, const GstStructure * sdes)
{
g_return_if_fail (sdes);
g_return_if_fail (RTP_IS_SESSION (sess));
RTP_SESSION_LOCK (sess);
rtp_source_set_sdes_struct (sess->source, gst_structure_copy (sdes));
RTP_SESSION_UNLOCK (sess);
}
static GstFlowReturn
source_push_rtp (RTPSource * source, gpointer data, RTPSession * session)
{
GstFlowReturn result = GST_FLOW_OK;
if (source == session->source) {
GST_LOG ("source %08x pushed sender RTP packet", source->ssrc);
RTP_SESSION_UNLOCK (session);
if (session->callbacks.send_rtp)
result =
session->callbacks.send_rtp (session, source, data,
session->send_rtp_user_data);
else {
gst_mini_object_unref (GST_MINI_OBJECT_CAST (data));
}
} else {
GST_LOG ("source %08x pushed receiver RTP packet", source->ssrc);
RTP_SESSION_UNLOCK (session);
if (session->callbacks.process_rtp)
result =
session->callbacks.process_rtp (session, source,
GST_BUFFER_CAST (data), session->process_rtp_user_data);
else
gst_buffer_unref (GST_BUFFER_CAST (data));
}
RTP_SESSION_LOCK (session);
return result;
}
static gint
source_clock_rate (RTPSource * source, guint8 pt, RTPSession * session)
{
gint result;
RTP_SESSION_UNLOCK (session);
if (session->callbacks.clock_rate)
result =
session->callbacks.clock_rate (session, pt,
session->clock_rate_user_data);
else
result = -1;
RTP_SESSION_LOCK (session);
GST_DEBUG ("got clock-rate %d for pt %d", result, pt);
return result;
}
static RTPSourceCallbacks callbacks = {
(RTPSourcePushRTP) source_push_rtp,
(RTPSourceClockRate) source_clock_rate,
};
static gboolean
check_collision (RTPSession * sess, RTPSource * source,
RTPArrivalStats * arrival, gboolean rtp)
{
/* If we have no arrival address, we can't do collision checking */
if (!arrival->have_address)
return FALSE;
if (sess->source != source) {
GstNetAddress *from;
gboolean have_from;
/* This is not our local source, but lets check if two remote
* source collide
*/
if (rtp) {
from = &source->rtp_from;
have_from = source->have_rtp_from;
} else {
from = &source->rtcp_from;
have_from = source->have_rtcp_from;
}
if (have_from) {
if (gst_netaddress_equal (from, &arrival->address)) {
/* Address is the same */
return FALSE;
} else {
GST_LOG ("we have a third-party collision or loop ssrc:%x",
rtp_source_get_ssrc (source));
if (sess->favor_new) {
if (rtp_source_find_conflicting_address (source,
&arrival->address, arrival->current_time)) {
gchar buf1[40];
gst_netaddress_to_string (&arrival->address, buf1, 40);
GST_LOG ("Known conflict on %x for %s, dropping packet",
rtp_source_get_ssrc (source), buf1);
return TRUE;
} else {
gchar buf1[40], buf2[40];
/* Current address is not a known conflict, lets assume this is
* a new source. Save old address in possible conflict list
*/
rtp_source_add_conflicting_address (source, from,
arrival->current_time);
gst_netaddress_to_string (from, buf1, 40);
gst_netaddress_to_string (&arrival->address, buf2, 40);
GST_DEBUG ("New conflict for ssrc %x, replacing %s with %s,"
" saving old as known conflict",
rtp_source_get_ssrc (source), buf1, buf2);
if (rtp)
rtp_source_set_rtp_from (source, &arrival->address);
else
rtp_source_set_rtcp_from (source, &arrival->address);
return FALSE;
}
} else {
/* Don't need to save old addresses, we ignore new sources */
return TRUE;
}
}
} else {
/* We don't already have a from address for RTP, just set it */
if (rtp)
rtp_source_set_rtp_from (source, &arrival->address);
else
rtp_source_set_rtcp_from (source, &arrival->address);
return FALSE;
}
/* FIXME: Log 3rd party collision somehow
* Maybe should be done in upper layer, only the SDES can tell us
* if its a collision or a loop
*/
/* If the source has been inactive for some time, we assume that it has
* simply changed its transport source address. Hence, there is no true
* third-party collision - only a simulated one. */
if (arrival->current_time > source->last_activity) {
GstClockTime inactivity_period =
arrival->current_time - source->last_activity;
if (inactivity_period > 1 * GST_SECOND) {
/* Use new network address */
if (rtp) {
g_assert (source->have_rtp_from);
rtp_source_set_rtp_from (source, &arrival->address);
} else {
g_assert (source->have_rtcp_from);
rtp_source_set_rtcp_from (source, &arrival->address);
}
return FALSE;
}
}
} else {
/* This is sending with our ssrc, is it an address we already know */
if (rtp_source_find_conflicting_address (source, &arrival->address,
arrival->current_time)) {
/* Its a known conflict, its probably a loop, not a collision
* lets just drop the incoming packet
*/
GST_DEBUG ("Our packets are being looped back to us, dropping");
} else {
/* Its a new collision, lets change our SSRC */
rtp_source_add_conflicting_address (source, &arrival->address,
arrival->current_time);
GST_DEBUG ("Collision for SSRC %x", rtp_source_get_ssrc (source));
on_ssrc_collision (sess, source);
rtp_session_schedule_bye_locked (sess, "SSRC Collision",
arrival->current_time);
sess->change_ssrc = TRUE;
}
}
return TRUE;
}
/* must be called with the session lock, the returned source needs to be
* unreffed after usage. */
static RTPSource *
obtain_source (RTPSession * sess, guint32 ssrc, gboolean * created,
RTPArrivalStats * arrival, gboolean rtp)
{
RTPSource *source;
source =
g_hash_table_lookup (sess->ssrcs[sess->mask_idx], GINT_TO_POINTER (ssrc));
if (source == NULL) {
/* make new Source in probation and insert */
source = rtp_source_new (ssrc);
/* for RTP packets we need to set the source in probation. Receiving RTCP
* packets of an SSRC, on the other hand, is a strong indication that we
* are dealing with a valid source. */
if (rtp)
source->probation = RTP_DEFAULT_PROBATION;
else
source->probation = 0;
/* store from address, if any */
if (arrival->have_address) {
if (rtp)
rtp_source_set_rtp_from (source, &arrival->address);
else
rtp_source_set_rtcp_from (source, &arrival->address);
}
/* configure a callback on the source */
rtp_source_set_callbacks (source, &callbacks, sess);
g_hash_table_insert (sess->ssrcs[sess->mask_idx], GINT_TO_POINTER (ssrc),
source);
/* we have one more source now */
sess->total_sources++;
*created = TRUE;
} else {
*created = FALSE;
/* check for collision, this updates the address when not previously set */
if (check_collision (sess, source, arrival, rtp)) {
return NULL;
}
}
/* update last activity */
source->last_activity = arrival->current_time;
if (rtp)
source->last_rtp_activity = arrival->current_time;
g_object_ref (source);
return source;
}
/**
* rtp_session_get_internal_source:
* @sess: a #RTPSession
*
* Get the internal #RTPSource of @sess.
*
* Returns: The internal #RTPSource. g_object_unref() after usage.
*/
RTPSource *
rtp_session_get_internal_source (RTPSession * sess)
{
RTPSource *result;
g_return_val_if_fail (RTP_IS_SESSION (sess), NULL);
result = g_object_ref (sess->source);
return result;
}
/**
* rtp_session_set_internal_ssrc:
* @sess: a #RTPSession
* @ssrc: an SSRC
*
* Set the SSRC of @sess to @ssrc.
*/
void
rtp_session_set_internal_ssrc (RTPSession * sess, guint32 ssrc)
{
RTP_SESSION_LOCK (sess);
if (ssrc != sess->source->ssrc) {
g_hash_table_steal (sess->ssrcs[sess->mask_idx],
GINT_TO_POINTER (sess->source->ssrc));
GST_DEBUG ("setting internal SSRC to %08x", ssrc);
/* After this call, any receiver of the old SSRC either in RTP or RTCP
* packets will timeout on the old SSRC, we could potentially schedule a
* BYE RTCP for the old SSRC... */
sess->source->ssrc = ssrc;
rtp_source_reset (sess->source);
/* rehash with the new SSRC */
g_hash_table_insert (sess->ssrcs[sess->mask_idx],
GINT_TO_POINTER (sess->source->ssrc), sess->source);
}
RTP_SESSION_UNLOCK (sess);
g_object_notify (G_OBJECT (sess), "internal-ssrc");
}
/**
* rtp_session_get_internal_ssrc:
* @sess: a #RTPSession
*
* Get the internal SSRC of @sess.
*
* Returns: The SSRC of the session.
*/
guint32
rtp_session_get_internal_ssrc (RTPSession * sess)
{
guint32 ssrc;
RTP_SESSION_LOCK (sess);
ssrc = sess->source->ssrc;
RTP_SESSION_UNLOCK (sess);
return ssrc;
}
/**
* rtp_session_add_source:
* @sess: a #RTPSession
* @src: #RTPSource to add
*
* Add @src to @session.
*
* Returns: %TRUE on success, %FALSE if a source with the same SSRC already
* existed in the session.
*/
gboolean
rtp_session_add_source (RTPSession * sess, RTPSource * src)
{
gboolean result = FALSE;
RTPSource *find;
g_return_val_if_fail (RTP_IS_SESSION (sess), FALSE);
g_return_val_if_fail (src != NULL, FALSE);
RTP_SESSION_LOCK (sess);
find =
g_hash_table_lookup (sess->ssrcs[sess->mask_idx],
GINT_TO_POINTER (src->ssrc));
if (find == NULL) {
g_hash_table_insert (sess->ssrcs[sess->mask_idx],
GINT_TO_POINTER (src->ssrc), src);
/* we have one more source now */
sess->total_sources++;
result = TRUE;
}
RTP_SESSION_UNLOCK (sess);
return result;
}
/**
* rtp_session_get_num_sources:
* @sess: an #RTPSession
*
* Get the number of sources in @sess.
*
* Returns: The number of sources in @sess.
*/
guint
rtp_session_get_num_sources (RTPSession * sess)
{
guint result;
g_return_val_if_fail (RTP_IS_SESSION (sess), FALSE);
RTP_SESSION_LOCK (sess);
result = sess->total_sources;
RTP_SESSION_UNLOCK (sess);
return result;
}
/**
* rtp_session_get_num_active_sources:
* @sess: an #RTPSession
*
* Get the number of active sources in @sess. A source is considered active when
* it has been validated and has not yet received a BYE RTCP message.
*
* Returns: The number of active sources in @sess.
*/
guint
rtp_session_get_num_active_sources (RTPSession * sess)
{
guint result;
g_return_val_if_fail (RTP_IS_SESSION (sess), 0);
RTP_SESSION_LOCK (sess);
result = sess->stats.active_sources;
RTP_SESSION_UNLOCK (sess);
return result;
}
/**
* rtp_session_get_source_by_ssrc:
* @sess: an #RTPSession
* @ssrc: an SSRC
*
* Find the source with @ssrc in @sess.
*
* Returns: a #RTPSource with SSRC @ssrc or NULL if the source was not found.
* g_object_unref() after usage.
*/
RTPSource *
rtp_session_get_source_by_ssrc (RTPSession * sess, guint32 ssrc)
{
RTPSource *result;
g_return_val_if_fail (RTP_IS_SESSION (sess), NULL);
RTP_SESSION_LOCK (sess);
result =
g_hash_table_lookup (sess->ssrcs[sess->mask_idx], GINT_TO_POINTER (ssrc));
if (result)
g_object_ref (result);
RTP_SESSION_UNLOCK (sess);
return result;
}
/**
* rtp_session_get_source_by_cname:
* @sess: a #RTPSession
* @cname: an CNAME
*
* Find the source with @cname in @sess.
*
* Returns: a #RTPSource with CNAME @cname or NULL if the source was not found.
* g_object_unref() after usage.
*/
RTPSource *
rtp_session_get_source_by_cname (RTPSession * sess, const gchar * cname)
{
RTPSource *result;
g_return_val_if_fail (RTP_IS_SESSION (sess), NULL);
g_return_val_if_fail (cname != NULL, NULL);
RTP_SESSION_LOCK (sess);
result = g_hash_table_lookup (sess->cnames, cname);
if (result)
g_object_ref (result);
RTP_SESSION_UNLOCK (sess);
return result;
}
/* should be called with the SESSION lock */
static guint32
rtp_session_create_new_ssrc (RTPSession * sess)
{
guint32 ssrc;
while (TRUE) {
ssrc = g_random_int ();
/* see if it exists in the session, we're done if it doesn't */
if (g_hash_table_lookup (sess->ssrcs[sess->mask_idx],
GINT_TO_POINTER (ssrc)) == NULL)
break;
}
return ssrc;
}
/**
* rtp_session_create_source:
* @sess: an #RTPSession
*
* Create an #RTPSource for use in @sess. This function will create a source
* with an ssrc that is currently not used by any participants in the session.
*
* Returns: an #RTPSource.
*/
RTPSource *
rtp_session_create_source (RTPSession * sess)
{
guint32 ssrc;
RTPSource *source;
RTP_SESSION_LOCK (sess);
ssrc = rtp_session_create_new_ssrc (sess);
source = rtp_source_new (ssrc);
rtp_source_set_callbacks (source, &callbacks, sess);
/* we need an additional ref for the source in the hashtable */
g_object_ref (source);
g_hash_table_insert (sess->ssrcs[sess->mask_idx], GINT_TO_POINTER (ssrc),
source);
/* we have one more source now */
sess->total_sources++;
RTP_SESSION_UNLOCK (sess);
return source;
}
/* update the RTPArrivalStats structure with the current time and other bits
* about the current buffer we are handling.
* This function is typically called when a validated packet is received.
* This function should be called with the SESSION_LOCK
*/
static void
update_arrival_stats (RTPSession * sess, RTPArrivalStats * arrival,
gboolean rtp, GstBuffer * buffer, GstClockTime current_time,
GstClockTime running_time, guint64 ntpnstime)
{
/* get time of arrival */
arrival->current_time = current_time;
arrival->running_time = running_time;
arrival->ntpnstime = ntpnstime;
/* get packet size including header overhead */
arrival->bytes = GST_BUFFER_SIZE (buffer) + sess->header_len;
if (rtp) {
arrival->payload_len = gst_rtp_buffer_get_payload_len (buffer);
} else {
arrival->payload_len = 0;
}
/* for netbuffer we can store the IP address to check for collisions */
arrival->have_address = GST_IS_NETBUFFER (buffer);
if (arrival->have_address) {
GstNetBuffer *netbuf = (GstNetBuffer *) buffer;
memcpy (&arrival->address, &netbuf->from, sizeof (GstNetAddress));
}
}
/**
* rtp_session_process_rtp:
* @sess: and #RTPSession
* @buffer: an RTP buffer
* @current_time: the current system time
* @running_time: the running_time of @buffer
*
* Process an RTP buffer in the session manager. This function takes ownership
* of @buffer.
*
* Returns: a #GstFlowReturn.
*/
GstFlowReturn
rtp_session_process_rtp (RTPSession * sess, GstBuffer * buffer,
GstClockTime current_time, GstClockTime running_time)
{
GstFlowReturn result;
guint32 ssrc;
RTPSource *source;
gboolean created;
gboolean prevsender, prevactive;
RTPArrivalStats arrival;
guint32 csrcs[16];
guint8 i, count;
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);
if (!gst_rtp_buffer_validate (buffer))
goto invalid_packet;
RTP_SESSION_LOCK (sess);
/* update arrival stats */
update_arrival_stats (sess, &arrival, TRUE, buffer, current_time,
running_time, -1);
/* ignore more RTP packets when we left the session */
if (sess->source->received_bye)
goto ignore;
/* get SSRC and look up in session database */
ssrc = gst_rtp_buffer_get_ssrc (buffer);
source = obtain_source (sess, ssrc, &created, &arrival, TRUE);
if (!source)
goto collision;
prevsender = RTP_SOURCE_IS_SENDER (source);
prevactive = RTP_SOURCE_IS_ACTIVE (source);
oldrate = source->bitrate;
/* copy available csrc for later */
count = gst_rtp_buffer_get_csrc_count (buffer);
/* make sure to not overflow our array. An RTP buffer can maximally contain
* 16 CSRCs */
count = MIN (count, 16);
for (i = 0; i < count; i++)
csrcs[i] = gst_rtp_buffer_get_csrc (buffer, i);
/* let source process the packet */
result = rtp_source_process_rtp (source, buffer, &arrival);
/* source became active */
if (prevactive != RTP_SOURCE_IS_ACTIVE (source)) {
sess->stats.active_sources++;
GST_DEBUG ("source: %08x became active, %d active sources", ssrc,
sess->stats.active_sources);
on_ssrc_validated (sess, source);
}
if (prevsender != RTP_SOURCE_IS_SENDER (source)) {
sess->stats.sender_sources++;
GST_DEBUG ("source: %08x became sender, %d sender sources", ssrc,
sess->stats.sender_sources);
}
if (oldrate != source->bitrate)
sess->recalc_bandwidth = TRUE;
if (created)
on_new_ssrc (sess, source);
if (source->validated) {
gboolean created;
/* for validated sources, we add the CSRCs as well */
for (i = 0; i < count; i++) {
guint32 csrc;
RTPSource *csrc_src;
csrc = csrcs[i];
/* get source */
csrc_src = obtain_source (sess, csrc, &created, &arrival, TRUE);
if (!csrc_src)
continue;
if (created) {
GST_DEBUG ("created new CSRC: %08x", csrc);
rtp_source_set_as_csrc (csrc_src);
if (RTP_SOURCE_IS_ACTIVE (csrc_src))
sess->stats.active_sources++;
on_new_ssrc (sess, csrc_src);
}
g_object_unref (csrc_src);
}
}
g_object_unref (source);
RTP_SESSION_UNLOCK (sess);
return result;
/* ERRORS */
invalid_packet:
{
gst_buffer_unref (buffer);
GST_DEBUG ("invalid RTP packet received");
return GST_FLOW_OK;
}
ignore:
{
gst_buffer_unref (buffer);
RTP_SESSION_UNLOCK (sess);
GST_DEBUG ("ignoring RTP packet because we are leaving");
return GST_FLOW_OK;
}
collision:
{
gst_buffer_unref (buffer);
RTP_SESSION_UNLOCK (sess);
GST_DEBUG ("ignoring packet because its collisioning");
return GST_FLOW_OK;
}
}
static void
rtp_session_process_rb (RTPSession * sess, RTPSource * source,
GstRTCPPacket * packet, RTPArrivalStats * arrival)
{
guint count, i;
count = gst_rtcp_packet_get_rb_count (packet);
for (i = 0; i < count; i++) {
guint32 ssrc, exthighestseq, jitter, lsr, dlsr;
guint8 fractionlost;
gint32 packetslost;
gst_rtcp_packet_get_rb (packet, i, &ssrc, &fractionlost,
&packetslost, &exthighestseq, &jitter, &lsr, &dlsr);
GST_DEBUG ("RB %d: SSRC %08x, jitter %" G_GUINT32_FORMAT, i, ssrc, jitter);
if (ssrc == sess->source->ssrc) {
/* only deal with report blocks for our session, we update the stats of
* the sender of the RTCP message. We could also compare our stats against
* the other sender to see if we are better or worse. */
rtp_source_process_rb (source, arrival->ntpnstime, fractionlost,
packetslost, exthighestseq, jitter, lsr, dlsr);
}
}
on_ssrc_active (sess, source);
}
/* A Sender report contains statistics about how the sender is doing. This
* includes timing informataion such as the relation between RTP and NTP
* timestamps and the number of packets/bytes it sent to us.
*
* In this report is also included a set of report blocks related to how this
* sender is receiving data (in case we (or somebody else) is also sending stuff
* to it). This info includes the packet loss, jitter and seqnum. It also
* contains information to calculate the round trip time (LSR/DLSR).
*/
static void
rtp_session_process_sr (RTPSession * sess, GstRTCPPacket * packet,
RTPArrivalStats * arrival, gboolean * do_sync)
{
guint32 senderssrc, rtptime, packet_count, octet_count;
guint64 ntptime;
RTPSource *source;
gboolean created, prevsender;
gst_rtcp_packet_sr_get_sender_info (packet, &senderssrc, &ntptime, &rtptime,
&packet_count, &octet_count);
GST_DEBUG ("got SR packet: SSRC %08x, time %" GST_TIME_FORMAT,
senderssrc, GST_TIME_ARGS (arrival->current_time));
source = obtain_source (sess, senderssrc, &created, arrival, FALSE);
if (!source)
return;
/* don't try to do lip-sync for sources that sent a BYE */
if (rtp_source_received_bye (source))
*do_sync = FALSE;
else
*do_sync = TRUE;
prevsender = RTP_SOURCE_IS_SENDER (source);
/* first update the source */
rtp_source_process_sr (source, arrival->current_time, ntptime, rtptime,
packet_count, octet_count);
if (prevsender != RTP_SOURCE_IS_SENDER (source)) {
sess->stats.sender_sources++;
GST_DEBUG ("source: %08x became sender, %d sender sources", senderssrc,
sess->stats.sender_sources);
}
if (created)
on_new_ssrc (sess, source);
rtp_session_process_rb (sess, source, packet, arrival);
g_object_unref (source);
}
/* A receiver report contains statistics about how a receiver is doing. It
* includes stuff like packet loss, jitter and the seqnum it received last. It
* also contains info to calculate the round trip time.
*
* We are only interested in how the sender of this report is doing wrt to us.
*/
static void
rtp_session_process_rr (RTPSession * sess, GstRTCPPacket * packet,
RTPArrivalStats * arrival)
{
guint32 senderssrc;
RTPSource *source;
gboolean created;
senderssrc = gst_rtcp_packet_rr_get_ssrc (packet);
GST_DEBUG ("got RR packet: SSRC %08x", senderssrc);
source = obtain_source (sess, senderssrc, &created, arrival, FALSE);
if (!source)
return;
if (created)
on_new_ssrc (sess, source);
rtp_session_process_rb (sess, source, packet, arrival);
g_object_unref (source);
}
/* Get SDES items and store them in the SSRC */
static void
rtp_session_process_sdes (RTPSession * sess, GstRTCPPacket * packet,
RTPArrivalStats * arrival)
{
guint items, i, j;
gboolean more_items, more_entries;
items = gst_rtcp_packet_sdes_get_item_count (packet);
GST_DEBUG ("got SDES packet with %d items", items);
more_items = gst_rtcp_packet_sdes_first_item (packet);
i = 0;
while (more_items) {
guint32 ssrc;
gboolean changed, created, validated;
RTPSource *source;
GstStructure *sdes;
ssrc = gst_rtcp_packet_sdes_get_ssrc (packet);
GST_DEBUG ("item %d, SSRC %08x", i, ssrc);
changed = FALSE;
/* find src, no probation when dealing with RTCP */
source = obtain_source (sess, ssrc, &created, arrival, FALSE);
if (!source)
return;
sdes = gst_structure_new ("application/x-rtp-source-sdes", NULL);
more_entries = gst_rtcp_packet_sdes_first_entry (packet);
j = 0;
while (more_entries) {
GstRTCPSDESType type;
guint8 len;
guint8 *data;
gchar *name;
gchar *value;
gst_rtcp_packet_sdes_get_entry (packet, &type, &len, &data);
GST_DEBUG ("entry %d, type %d, len %d, data %.*s", j, type, len, len,
data);
if (type == GST_RTCP_SDES_PRIV) {
name = g_strndup ((const gchar *) &data[1], data[0]);
len -= data[0] + 1;
data += data[0] + 1;
} else {
name = g_strdup (gst_rtcp_sdes_type_to_name (type));
}
value = g_strndup ((const gchar *) data, len);
gst_structure_set (sdes, name, G_TYPE_STRING, value, NULL);
g_free (name);
g_free (value);
more_entries = gst_rtcp_packet_sdes_next_entry (packet);
j++;
}
/* takes ownership of sdes */
changed = rtp_source_set_sdes_struct (source, sdes);
validated = !RTP_SOURCE_IS_ACTIVE (source);
source->validated = TRUE;
if (created)
on_new_ssrc (sess, source);
if (validated)
on_ssrc_validated (sess, source);
if (changed)
on_ssrc_sdes (sess, source);
g_object_unref (source);
more_items = gst_rtcp_packet_sdes_next_item (packet);
i++;
}
}
/* BYE is sent when a client leaves the session
*/
static void
rtp_session_process_bye (RTPSession * sess, GstRTCPPacket * packet,
RTPArrivalStats * arrival)
{
guint count, i;
gchar *reason;
gboolean reconsider = FALSE;
reason = gst_rtcp_packet_bye_get_reason (packet);
GST_DEBUG ("got BYE packet (reason: %s)", GST_STR_NULL (reason));
count = gst_rtcp_packet_bye_get_ssrc_count (packet);
for (i = 0; i < count; i++) {
guint32 ssrc;
RTPSource *source;
gboolean created, prevactive, prevsender;
guint pmembers, members;
ssrc = gst_rtcp_packet_bye_get_nth_ssrc (packet, i);
GST_DEBUG ("SSRC: %08x", ssrc);
/* find src and mark bye, no probation when dealing with RTCP */
source = obtain_source (sess, ssrc, &created, arrival, FALSE);
if (!source)
return;
/* store time for when we need to time out this source */
source->bye_time = arrival->current_time;
prevactive = RTP_SOURCE_IS_ACTIVE (source);
prevsender = RTP_SOURCE_IS_SENDER (source);
/* let the source handle the rest */
rtp_source_process_bye (source, reason);
pmembers = sess->stats.active_sources;
if (prevactive && !RTP_SOURCE_IS_ACTIVE (source)) {
sess->stats.active_sources--;
GST_DEBUG ("source: %08x became inactive, %d active sources", ssrc,
sess->stats.active_sources);
}
if (prevsender && !RTP_SOURCE_IS_SENDER (source)) {
sess->stats.sender_sources--;
GST_DEBUG ("source: %08x became non sender, %d sender sources", ssrc,
sess->stats.sender_sources);
}
members = sess->stats.active_sources;
if (!sess->source->received_bye && members < pmembers) {
/* some members went away since the previous timeout estimate.
* Perform reverse reconsideration but only when we are not scheduling a
* BYE ourselves. */
if (arrival->current_time < sess->next_rtcp_check_time) {
GstClockTime time_remaining;
time_remaining = sess->next_rtcp_check_time - arrival->current_time;
sess->next_rtcp_check_time =
gst_util_uint64_scale (time_remaining, members, pmembers);
GST_DEBUG ("reverse reconsideration %" GST_TIME_FORMAT,
GST_TIME_ARGS (sess->next_rtcp_check_time));
sess->next_rtcp_check_time += arrival->current_time;
/* mark pending reconsider. We only want to signal the reconsideration
* once after we handled all the source in the bye packet */
reconsider = TRUE;
}
}
if (created)
on_new_ssrc (sess, source);
on_bye_ssrc (sess, source);
g_object_unref (source);
}
if (reconsider) {
RTP_SESSION_UNLOCK (sess);
/* notify app of reconsideration */
if (sess->callbacks.reconsider)
sess->callbacks.reconsider (sess, sess->reconsider_user_data);
RTP_SESSION_LOCK (sess);
}
g_free (reason);
}
static void
rtp_session_process_app (RTPSession * sess, GstRTCPPacket * packet,
RTPArrivalStats * arrival)
{
GST_DEBUG ("received APP");
}
static void
rtp_session_process_pli (RTPSession * sess, guint32 sender_ssrc,
guint32 media_ssrc, GstClockTime current_time)
{
RTPSource *src;
guint32 round_trip = 0;
if (!sess->callbacks.request_key_unit)
return;
src = g_hash_table_lookup (sess->ssrcs[sess->mask_idx],
GINT_TO_POINTER (sender_ssrc));
if (!src)
return;
if (sess->last_keyframe_request != GST_CLOCK_TIME_NONE &&
rtp_source_get_last_rb (src, NULL, NULL, NULL, NULL, NULL, NULL,
&round_trip)) {
GstClockTime round_trip_in_ns = gst_util_uint64_scale (round_trip,
GST_SECOND, 65536);
if (sess->last_keyframe_request != GST_CLOCK_TIME_NONE &&
current_time - sess->last_keyframe_request < round_trip_in_ns) {
GST_DEBUG ("Ignoring PLI because one was send without one RTT (%"
GST_TIME_FORMAT " < %" GST_TIME_FORMAT ")",
GST_TIME_ARGS (current_time - sess->last_keyframe_request),
GST_TIME_ARGS (round_trip_in_ns));;
return;
}
}
sess->last_keyframe_request = current_time;
GST_LOG ("received PLI from %X %p(%p)", sender_ssrc,
sess->callbacks.process_rtp, sess->callbacks.request_key_unit);
sess->callbacks.request_key_unit (sess, FALSE,
sess->request_key_unit_user_data);
}
static void
rtp_session_process_feedback (RTPSession * sess, GstRTCPPacket * packet,
RTPArrivalStats * arrival, GstClockTime current_time)
{
GstRTCPType type = gst_rtcp_packet_get_type (packet);
GstRTCPFBType fbtype = gst_rtcp_packet_fb_get_type (packet);
guint32 sender_ssrc = gst_rtcp_packet_fb_get_sender_ssrc (packet);
guint32 media_ssrc = gst_rtcp_packet_fb_get_media_ssrc (packet);
guint length = 4 * (gst_rtcp_packet_get_length (packet) - 2);
GST_DEBUG ("received feedback %d:%d from %08X about %08X"
" with FCI of length %d", type, fbtype, sender_ssrc, media_ssrc, length);
if (g_signal_has_handler_pending (sess,
rtp_session_signals[SIGNAL_ON_FEEDBACK_RTCP], 0, TRUE)) {
GstBuffer *fci = NULL;
if (length) {
fci = gst_buffer_create_sub (packet->buffer, packet->offset + 72, length);
GST_BUFFER_TIMESTAMP (fci) = arrival->running_time;
}
RTP_SESSION_UNLOCK (sess);
g_signal_emit (sess, rtp_session_signals[SIGNAL_ON_FEEDBACK_RTCP], 0,
type, fbtype, sender_ssrc, media_ssrc, fci);
RTP_SESSION_LOCK (sess);
if (fci)
gst_buffer_unref (fci);
}
if (sess->rtcp_feedback_retention_window) {
RTPSource *src = g_hash_table_lookup (sess->ssrcs[sess->mask_idx],
GINT_TO_POINTER (media_ssrc));
if (src)
rtp_source_retain_rtcp_packet (src, packet, arrival->running_time);
}
if (rtp_source_get_ssrc (sess->source) == media_ssrc) {
switch (type) {
case GST_RTCP_TYPE_PSFB:
switch (fbtype) {
case GST_RTCP_PSFB_TYPE_PLI:
rtp_session_process_pli (sess, sender_ssrc, media_ssrc,
current_time);
break;
default:
break;
}
break;
case GST_RTCP_TYPE_RTPFB:
default:
break;
}
}
}
/**
* rtp_session_process_rtcp:
* @sess: and #RTPSession
* @buffer: an RTCP buffer
* @current_time: the current system time
* @ntpnstime: the current NTP time in nanoseconds
*
* Process an RTCP buffer in the session manager. This function takes ownership
* of @buffer.
*
* Returns: a #GstFlowReturn.
*/
GstFlowReturn
rtp_session_process_rtcp (RTPSession * sess, GstBuffer * buffer,
GstClockTime current_time, guint64 ntpnstime)
{
GstRTCPPacket packet;
gboolean more, is_bye = FALSE, do_sync = FALSE;
RTPArrivalStats arrival;
GstFlowReturn result = GST_FLOW_OK;
g_return_val_if_fail (RTP_IS_SESSION (sess), GST_FLOW_ERROR);
g_return_val_if_fail (GST_IS_BUFFER (buffer), GST_FLOW_ERROR);
if (!gst_rtcp_buffer_validate (buffer))
goto invalid_packet;
GST_DEBUG ("received RTCP packet");
RTP_SESSION_LOCK (sess);
/* update arrival stats */
update_arrival_stats (sess, &arrival, FALSE, buffer, current_time, -1,
ntpnstime);
if (sess->sent_bye)
goto ignore;
/* start processing the compound packet */
more = gst_rtcp_buffer_get_first_packet (buffer, &packet);
while (more) {
GstRTCPType type;
type = gst_rtcp_packet_get_type (&packet);
/* when we are leaving the session, we should ignore all non-BYE messages */
if (sess->source->received_bye && type != GST_RTCP_TYPE_BYE) {
GST_DEBUG ("ignoring non-BYE RTCP packet because we are leaving");
goto next;
}
switch (type) {
case GST_RTCP_TYPE_SR:
rtp_session_process_sr (sess, &packet, &arrival, &do_sync);
break;
case GST_RTCP_TYPE_RR:
rtp_session_process_rr (sess, &packet, &arrival);
break;
case GST_RTCP_TYPE_SDES:
rtp_session_process_sdes (sess, &packet, &arrival);
break;
case GST_RTCP_TYPE_BYE:
is_bye = TRUE;
/* don't try to attempt lip-sync anymore for streams with a BYE */
do_sync = FALSE;
rtp_session_process_bye (sess, &packet, &arrival);
break;
case GST_RTCP_TYPE_APP:
rtp_session_process_app (sess, &packet, &arrival);
break;
case GST_RTCP_TYPE_RTPFB:
case GST_RTCP_TYPE_PSFB:
rtp_session_process_feedback (sess, &packet, &arrival, current_time);
break;
default:
GST_WARNING ("got unknown RTCP packet");
break;
}
next:
more = gst_rtcp_packet_move_to_next (&packet);
}
/* if we are scheduling a BYE, we only want to count bye packets, else we
* count everything */
if (sess->source->received_bye) {
if (is_bye) {
sess->stats.bye_members++;
UPDATE_AVG (sess->stats.avg_rtcp_packet_size, arrival.bytes);
}
} else {
/* keep track of average packet size */
UPDATE_AVG (sess->stats.avg_rtcp_packet_size, arrival.bytes);
}
GST_DEBUG ("%p, received RTCP packet, avg size %u, %u", &sess->stats,
sess->stats.avg_rtcp_packet_size, arrival.bytes);
RTP_SESSION_UNLOCK (sess);
/* notify caller of sr packets in the callback */
if (do_sync && sess->callbacks.sync_rtcp) {
/* make writable, we might want to change the buffer */
buffer = gst_buffer_make_metadata_writable (buffer);
result = sess->callbacks.sync_rtcp (sess, sess->source, buffer,
sess->sync_rtcp_user_data);
} else
gst_buffer_unref (buffer);
return result;
/* ERRORS */
invalid_packet:
{
GST_DEBUG ("invalid RTCP packet received");
gst_buffer_unref (buffer);
return GST_FLOW_OK;
}
ignore:
{
gst_buffer_unref (buffer);
RTP_SESSION_UNLOCK (sess);
GST_DEBUG ("ignoring RTP packet because we left");
return GST_FLOW_OK;
}
}
/**
* rtp_session_send_rtp:
* @sess: an #RTPSession
* @data: pointer to either an RTP buffer or a list of RTP buffers
* @is_list: TRUE when @data is a buffer list
* @current_time: the current system time
* @running_time: the running time of @data
*
* Send the RTP buffer in the session manager. This function takes ownership of
* @buffer.
*
* Returns: a #GstFlowReturn.
*/
GstFlowReturn
rtp_session_send_rtp (RTPSession * sess, gpointer data, gboolean is_list,
GstClockTime current_time, GstClockTime running_time)
{
GstFlowReturn result;
RTPSource *source;
gboolean prevsender;
gboolean valid_packet;
guint64 oldrate;
g_return_val_if_fail (RTP_IS_SESSION (sess), GST_FLOW_ERROR);
g_return_val_if_fail (is_list || GST_IS_BUFFER (data), GST_FLOW_ERROR);
if (is_list) {
valid_packet = gst_rtp_buffer_list_validate (GST_BUFFER_LIST_CAST (data));
} else {
valid_packet = gst_rtp_buffer_validate (GST_BUFFER_CAST (data));
}
if (!valid_packet)
goto invalid_packet;
GST_LOG ("received RTP %s for sending", is_list ? "list" : "packet");
RTP_SESSION_LOCK (sess);
source = sess->source;
/* update last activity */
source->last_rtp_activity = current_time;
prevsender = RTP_SOURCE_IS_SENDER (source);
oldrate = source->bitrate;
/* we use our own source to send */
result = rtp_source_send_rtp (source, data, is_list, running_time);
if (RTP_SOURCE_IS_SENDER (source) && !prevsender)
sess->stats.sender_sources++;
if (oldrate != source->bitrate)
sess->recalc_bandwidth = TRUE;
RTP_SESSION_UNLOCK (sess);
return result;
/* ERRORS */
invalid_packet:
{
gst_mini_object_unref (GST_MINI_OBJECT_CAST (data));
GST_DEBUG ("invalid RTP packet received");
return GST_FLOW_OK;
}
}
static void
add_bitrates (gpointer key, RTPSource * source, gdouble * bandwidth)
{
*bandwidth += source->bitrate;
}
static GstClockTime
calculate_rtcp_interval (RTPSession * sess, gboolean deterministic,
gboolean first)
{
GstClockTime result;
/* recalculate bandwidth when it changed */
if (sess->recalc_bandwidth) {
gdouble bandwidth;
if (sess->bandwidth > 0)
bandwidth = sess->bandwidth;
else {
/* If it is <= 0, then try to estimate the actual bandwidth */
bandwidth = sess->source->bitrate;
g_hash_table_foreach (sess->cnames, (GHFunc) add_bitrates, &bandwidth);
bandwidth /= 8.0;
}
if (bandwidth == 0)
bandwidth = RTP_STATS_BANDWIDTH;
rtp_stats_set_bandwidths (&sess->stats, bandwidth,
sess->rtcp_bandwidth, sess->rtcp_rs_bandwidth, sess->rtcp_rr_bandwidth);
sess->recalc_bandwidth = FALSE;
}
if (sess->source->received_bye) {
result = rtp_stats_calculate_bye_interval (&sess->stats);
} else {
result = rtp_stats_calculate_rtcp_interval (&sess->stats,
RTP_SOURCE_IS_SENDER (sess->source), first);
}
GST_DEBUG ("next deterministic interval: %" GST_TIME_FORMAT ", first %d",
GST_TIME_ARGS (result), first);
if (!deterministic && result != GST_CLOCK_TIME_NONE)
result = rtp_stats_add_rtcp_jitter (&sess->stats, result);
GST_DEBUG ("next interval: %" GST_TIME_FORMAT, GST_TIME_ARGS (result));
return result;
}
/* Stop the current @sess and schedule a BYE message for the other members.
* One must have the session lock to call this function
*/
static GstFlowReturn
rtp_session_schedule_bye_locked (RTPSession * sess, const gchar * reason,
GstClockTime current_time)
{
GstFlowReturn result = GST_FLOW_OK;
RTPSource *source;
GstClockTime interval;
g_return_val_if_fail (RTP_IS_SESSION (sess), GST_FLOW_ERROR);
source = sess->source;
/* ignore more BYEs */
if (source->received_bye)
goto done;
/* we have BYE now */
source->received_bye = TRUE;
/* at least one member wants to send a BYE */
g_free (sess->bye_reason);
sess->bye_reason = g_strdup (reason);
INIT_AVG (sess->stats.avg_rtcp_packet_size, 100);
sess->stats.bye_members = 1;
sess->first_rtcp = TRUE;
sess->sent_bye = FALSE;
sess->allow_early = TRUE;
/* reschedule transmission */
sess->last_rtcp_send_time = current_time;
interval = calculate_rtcp_interval (sess, FALSE, TRUE);
sess->next_rtcp_check_time = current_time + interval;
GST_DEBUG ("Schedule BYE for %" GST_TIME_FORMAT ", %" GST_TIME_FORMAT,
GST_TIME_ARGS (interval), GST_TIME_ARGS (sess->next_rtcp_check_time));
RTP_SESSION_UNLOCK (sess);
/* notify app of reconsideration */
if (sess->callbacks.reconsider)
sess->callbacks.reconsider (sess, sess->reconsider_user_data);
RTP_SESSION_LOCK (sess);
done:
return result;
}
/**
* rtp_session_schedule_bye:
* @sess: an #RTPSession
* @reason: a reason or NULL
* @current_time: the current system time
*
* Stop the current @sess and schedule a BYE message for the other members.
*
* Returns: a #GstFlowReturn.
*/
GstFlowReturn
rtp_session_schedule_bye (RTPSession * sess, const gchar * reason,
GstClockTime current_time)
{
GstFlowReturn result = GST_FLOW_OK;
g_return_val_if_fail (RTP_IS_SESSION (sess), GST_FLOW_ERROR);
RTP_SESSION_LOCK (sess);
result = rtp_session_schedule_bye_locked (sess, reason, current_time);
RTP_SESSION_UNLOCK (sess);
return result;
}
/**
* rtp_session_next_timeout:
* @sess: an #RTPSession
* @current_time: the current system time
*
* Get the next time we should perform session maintenance tasks.
*
* Returns: a time when rtp_session_on_timeout() should be called with the
* current system time.
*/
GstClockTime
rtp_session_next_timeout (RTPSession * sess, GstClockTime current_time)
{
GstClockTime result, interval = 0;
g_return_val_if_fail (RTP_IS_SESSION (sess), GST_CLOCK_TIME_NONE);
RTP_SESSION_LOCK (sess);
if (GST_CLOCK_TIME_IS_VALID (sess->next_early_rtcp_time)) {
result = sess->next_early_rtcp_time;
goto early_exit;
}
result = sess->next_rtcp_check_time;
GST_DEBUG ("current time: %" GST_TIME_FORMAT ", next :%" GST_TIME_FORMAT,
GST_TIME_ARGS (current_time), GST_TIME_ARGS (result));
if (result < current_time) {
GST_DEBUG ("take current time as base");
/* our previous check time expired, start counting from the current time
* again. */
result = current_time;
}
if (sess->source->received_bye) {
if (sess->sent_bye) {
GST_DEBUG ("we sent BYE already");
interval = GST_CLOCK_TIME_NONE;
} else if (sess->stats.active_sources >= 50) {
GST_DEBUG ("reconsider BYE, more than 50 sources");
/* reconsider BYE if members >= 50 */
interval = calculate_rtcp_interval (sess, FALSE, TRUE);
}
} else {
if (sess->first_rtcp) {
GST_DEBUG ("first RTCP packet");
/* we are called for the first time */
interval = calculate_rtcp_interval (sess, FALSE, TRUE);
} else if (sess->next_rtcp_check_time < current_time) {
GST_DEBUG ("old check time expired, getting new timeout");
/* get a new timeout when we need to */
interval = calculate_rtcp_interval (sess, FALSE, FALSE);
}
}
if (interval != GST_CLOCK_TIME_NONE)
result += interval;
else
result = GST_CLOCK_TIME_NONE;
sess->next_rtcp_check_time = result;
early_exit:
GST_DEBUG ("current time: %" GST_TIME_FORMAT
", next time: %" GST_TIME_FORMAT,
GST_TIME_ARGS (current_time), GST_TIME_ARGS (result));
RTP_SESSION_UNLOCK (sess);
return result;
}
typedef struct
{
RTPSession *sess;
GstBuffer *rtcp;
GstClockTime current_time;
guint64 ntpnstime;
GstClockTime running_time;
GstClockTime interval;
GstRTCPPacket packet;
gboolean is_bye;
gboolean has_sdes;
gboolean is_early;
gboolean may_suppress;
} ReportData;
static void
session_start_rtcp (RTPSession * sess, ReportData * data)
{
GstRTCPPacket *packet = &data->packet;
RTPSource *own = sess->source;
data->rtcp = gst_rtcp_buffer_new (sess->mtu);
if (RTP_SOURCE_IS_SENDER (own)) {
guint64 ntptime;
guint32 rtptime;
guint32 packet_count, octet_count;
/* we are a sender, create SR */
GST_DEBUG ("create SR for SSRC %08x", own->ssrc);
gst_rtcp_buffer_add_packet (data->rtcp, GST_RTCP_TYPE_SR, packet);
/* get latest stats */
rtp_source_get_new_sr (own, data->ntpnstime, data->running_time,
&ntptime, &rtptime, &packet_count, &octet_count);
/* store stats */
rtp_source_process_sr (own, data->current_time, ntptime, rtptime,
packet_count, octet_count);
/* fill in sender report info */
gst_rtcp_packet_sr_set_sender_info (packet, own->ssrc,
ntptime, rtptime, packet_count, octet_count);
} else {
/* we are only receiver, create RR */
GST_DEBUG ("create RR for SSRC %08x", own->ssrc);
gst_rtcp_buffer_add_packet (data->rtcp, GST_RTCP_TYPE_RR, packet);
gst_rtcp_packet_rr_set_ssrc (packet, own->ssrc);
}
}
/* construct a Sender or Receiver Report */
static void
session_report_blocks (const gchar * key, RTPSource * source, ReportData * data)
{
RTPSession *sess = data->sess;
GstRTCPPacket *packet = &data->packet;
/* create a new buffer if needed */
if (data->rtcp == NULL) {
session_start_rtcp (sess, data);
} else if (data->is_early) {
/* Put a single RR or SR in minimal compound packets */
return;
}
if (gst_rtcp_packet_get_rb_count (packet) < GST_RTCP_MAX_RB_COUNT) {
/* only report about other sender sources */
if (source != sess->source && RTP_SOURCE_IS_SENDER (source)) {
guint8 fractionlost;
gint32 packetslost;
guint32 exthighestseq, jitter;
guint32 lsr, dlsr;
/* get new stats */
rtp_source_get_new_rb (source, data->current_time, &fractionlost,
&packetslost, &exthighestseq, &jitter, &lsr, &dlsr);
/* store last generated RR packet */
source->last_rr.is_valid = TRUE;
source->last_rr.fractionlost = fractionlost;
source->last_rr.packetslost = packetslost;
source->last_rr.exthighestseq = exthighestseq;
source->last_rr.jitter = jitter;
source->last_rr.lsr = lsr;
source->last_rr.dlsr = dlsr;
/* packet is not yet filled, add report block for this source. */
gst_rtcp_packet_add_rb (packet, source->ssrc, fractionlost, packetslost,
exthighestseq, jitter, lsr, dlsr);
}
}
}
/* perform cleanup of sources that timed out */
static void
session_cleanup (const gchar * key, RTPSource * source, ReportData * data)
{
gboolean remove = FALSE;
gboolean byetimeout = FALSE;
gboolean sendertimeout = FALSE;
gboolean is_sender, is_active;
RTPSession *sess = data->sess;
GstClockTime interval;
is_sender = RTP_SOURCE_IS_SENDER (source);
is_active = RTP_SOURCE_IS_ACTIVE (source);
/* check for our own source, we don't want to delete our own source. */
if (!(source == sess->source)) {
if (source->received_bye) {
/* if we received a BYE from the source, remove the source after some
* time. */
if (data->current_time > source->bye_time &&
data->current_time - source->bye_time > sess->stats.bye_timeout) {
GST_DEBUG ("removing BYE source %08x", source->ssrc);
remove = TRUE;
byetimeout = TRUE;
}
}
/* sources that were inactive for more than 5 times the deterministic reporting
* interval get timed out. the min timeout is 5 seconds. */
if (data->current_time > source->last_activity) {
interval = MAX (data->interval * 5, 5 * GST_SECOND);
if (data->current_time - source->last_activity > interval) {
GST_DEBUG ("removing timeout source %08x, last %" GST_TIME_FORMAT,
source->ssrc, GST_TIME_ARGS (source->last_activity));
remove = TRUE;
}
}
}
/* senders that did not send for a long time become a receiver, this also
* holds for our own source. */
if (is_sender) {
if (data->current_time > source->last_rtp_activity) {
interval = MAX (data->interval * 2, 5 * GST_SECOND);
if (data->current_time - source->last_rtp_activity > interval) {
GST_DEBUG ("sender source %08x timed out and became receiver, last %"
GST_TIME_FORMAT, source->ssrc,
GST_TIME_ARGS (source->last_rtp_activity));
source->is_sender = FALSE;
sess->stats.sender_sources--;
sendertimeout = TRUE;
}
}
}
if (remove) {
sess->total_sources--;
if (is_sender)
sess->stats.sender_sources--;
if (is_active)
sess->stats.active_sources--;
if (byetimeout)
on_bye_timeout (sess, source);
else
on_timeout (sess, source);
} else {
if (sendertimeout)
on_sender_timeout (sess, source);
}
source->closing = remove;
}
static void
session_sdes (RTPSession * sess, ReportData * data)
{
GstRTCPPacket *packet = &data->packet;
const GstStructure *sdes;
gint i, n_fields;
/* add SDES packet */
gst_rtcp_buffer_add_packet (data->rtcp, GST_RTCP_TYPE_SDES, packet);
gst_rtcp_packet_sdes_add_item (packet, sess->source->ssrc);
sdes = rtp_source_get_sdes_struct (sess->source);
/* add all fields in the structure, the order is not important. */
n_fields = gst_structure_n_fields (sdes);
for (i = 0; i < n_fields; ++i) {
const gchar *field;
const gchar *value;
GstRTCPSDESType type;
field = gst_structure_nth_field_name (sdes, i);
if (field == NULL)
continue;
value = gst_structure_get_string (sdes, field);
if (value == NULL)
continue;
type = gst_rtcp_sdes_name_to_type (field);
/* Early packets are minimal and only include the CNAME */
if (data->is_early && type != GST_RTCP_SDES_CNAME)
continue;
if (type > GST_RTCP_SDES_END && type < GST_RTCP_SDES_PRIV) {
gst_rtcp_packet_sdes_add_entry (packet, type, strlen (value),
(const guint8 *) value);
} else if (type == GST_RTCP_SDES_PRIV) {
gsize prefix_len;
gsize value_len;
gsize data_len;
guint8 data[256];
/* don't accept entries that are too big */
prefix_len = strlen (field);
if (prefix_len > 255)
continue;
value_len = strlen (value);
if (value_len > 255)
continue;
data_len = 1 + prefix_len + value_len;
if (data_len > 255)
continue;
data[0] = prefix_len;
memcpy (&data[1], field, prefix_len);
memcpy (&data[1 + prefix_len], value, value_len);
gst_rtcp_packet_sdes_add_entry (packet, type, data_len, data);
}
}
data->has_sdes = TRUE;
}
/* schedule a BYE packet */
static void
session_bye (RTPSession * sess, ReportData * data)
{
GstRTCPPacket *packet = &data->packet;
/* open packet */
session_start_rtcp (sess, data);
/* add SDES */
session_sdes (sess, data);
/* add a BYE packet */
gst_rtcp_buffer_add_packet (data->rtcp, GST_RTCP_TYPE_BYE, packet);
gst_rtcp_packet_bye_add_ssrc (packet, sess->source->ssrc);
if (sess->bye_reason)
gst_rtcp_packet_bye_set_reason (packet, sess->bye_reason);
/* we have a BYE packet now */
data->is_bye = TRUE;
}
static gboolean
is_rtcp_time (RTPSession * sess, GstClockTime current_time, ReportData * data)
{
GstClockTime new_send_time, elapsed;
if (data->is_early && sess->next_early_rtcp_time < current_time)
goto early;
/* no need to check yet */
if (sess->next_rtcp_check_time > current_time) {
GST_DEBUG ("no check time yet, next %" GST_TIME_FORMAT " > now %"
GST_TIME_FORMAT, GST_TIME_ARGS (sess->next_rtcp_check_time),
GST_TIME_ARGS (current_time));
return FALSE;
}
/* get elapsed time since we last reported */
elapsed = current_time - sess->last_rtcp_send_time;
/* perform forward reconsideration */
new_send_time = rtp_stats_add_rtcp_jitter (&sess->stats, data->interval);
GST_DEBUG ("forward reconsideration %" GST_TIME_FORMAT ", elapsed %"
GST_TIME_FORMAT, GST_TIME_ARGS (new_send_time), GST_TIME_ARGS (elapsed));
new_send_time += sess->last_rtcp_send_time;
/* check if reconsideration */
if (current_time < new_send_time) {
GST_DEBUG ("reconsider RTCP for %" GST_TIME_FORMAT,
GST_TIME_ARGS (new_send_time));
/* store new check time */
sess->next_rtcp_check_time = new_send_time;
return FALSE;
}
early:
new_send_time = calculate_rtcp_interval (sess, FALSE, FALSE);
GST_DEBUG ("can send RTCP now, next interval %" GST_TIME_FORMAT,
GST_TIME_ARGS (new_send_time));
sess->next_rtcp_check_time = current_time + new_send_time;
/* Apply the rules from RFC 4585 section 3.5.3 */
if (sess->stats.min_interval != 0 && !sess->first_rtcp) {
GstClockTimeDiff T_rr_current_interval = g_random_double_range (0.5, 1.5) *
sess->stats.min_interval;
/* This will caused the RTCP to be suppressed if no FB packets are added */
if (sess->last_rtcp_send_time + T_rr_current_interval >
sess->next_rtcp_check_time) {
GST_DEBUG ("RTCP packet could be suppressed min: %" GST_TIME_FORMAT
" last: %" GST_TIME_FORMAT
" + T_rr_current_interval: %" GST_TIME_FORMAT
" > sess->next_rtcp_check_time: %" GST_TIME_FORMAT,
GST_TIME_ARGS (sess->stats.min_interval),
GST_TIME_ARGS (sess->last_rtcp_send_time),
GST_TIME_ARGS (T_rr_current_interval),
GST_TIME_ARGS (sess->next_rtcp_check_time));
data->may_suppress = TRUE;
}
}
return TRUE;
}
static void
clone_ssrcs_hashtable (gchar * key, RTPSource * source, GHashTable * hash_table)
{
g_hash_table_insert (hash_table, key, g_object_ref (source));
}
static gboolean
remove_closing_sources (const gchar * key, RTPSource * source, gpointer * data)
{
return source->closing;
}
/**
* rtp_session_on_timeout:
* @sess: an #RTPSession
* @current_time: the current system time
* @ntpnstime: the current NTP time in nanoseconds
* @running_time: the current running_time of the pipeline
*
* Perform maintenance actions after the timeout obtained with
* rtp_session_next_timeout() expired.
*
* This function will perform timeouts of receivers and senders, send a BYE
* packet or generate RTCP packets with current session stats.
*
* This function can call the #RTPSessionSendRTCP callback, possibly multiple
* times, for each packet that should be processed.
*
* Returns: a #GstFlowReturn.
*/
GstFlowReturn
rtp_session_on_timeout (RTPSession * sess, GstClockTime current_time,
guint64 ntpnstime, GstClockTime running_time)
{
GstFlowReturn result = GST_FLOW_OK;
ReportData data;
RTPSource *own;
GHashTable *table_copy;
gboolean notify = FALSE;
g_return_val_if_fail (RTP_IS_SESSION (sess), GST_FLOW_ERROR);
GST_DEBUG ("reporting at %" GST_TIME_FORMAT ", NTP time %" GST_TIME_FORMAT,
GST_TIME_ARGS (current_time), GST_TIME_ARGS (ntpnstime));
data.sess = sess;
data.rtcp = NULL;
data.current_time = current_time;
data.ntpnstime = ntpnstime;
data.is_bye = FALSE;
data.has_sdes = FALSE;
data.may_suppress = FALSE;
data.running_time = running_time;
own = sess->source;
RTP_SESSION_LOCK (sess);
/* get a new interval, we need this for various cleanups etc */
data.interval = calculate_rtcp_interval (sess, TRUE, sess->first_rtcp);
/* Make a local copy of the hashtable. We need to do this because the
* cleanup stage below releases the session lock. */
table_copy = g_hash_table_new_full (NULL, NULL, NULL,
(GDestroyNotify) g_object_unref);
g_hash_table_foreach (sess->ssrcs[sess->mask_idx],
(GHFunc) clone_ssrcs_hashtable, table_copy);
/* Clean up the session, mark the source for removing, this might release the
* session lock. */
g_hash_table_foreach (table_copy, (GHFunc) session_cleanup, &data);
g_hash_table_destroy (table_copy);
/* Now remove the marked sources */
g_hash_table_foreach_remove (sess->ssrcs[sess->mask_idx],
(GHRFunc) remove_closing_sources, NULL);
if (GST_CLOCK_TIME_IS_VALID (sess->next_early_rtcp_time))
data.is_early = TRUE;
else
data.is_early = FALSE;
/* see if we need to generate SR or RR packets */
if (is_rtcp_time (sess, current_time, &data)) {
if (own->received_bye) {
/* generate BYE instead */
GST_DEBUG ("generating BYE message");
session_bye (sess, &data);
sess->sent_bye = TRUE;
} else {
/* loop over all known sources and do something */
g_hash_table_foreach (sess->ssrcs[sess->mask_idx],
(GHFunc) session_report_blocks, &data);
}
}
if (data.rtcp) {
/* we keep track of the last report time in order to timeout inactive
* receivers or senders */
if (!data.is_early && !data.may_suppress)
sess->last_rtcp_send_time = data.current_time;
sess->first_rtcp = FALSE;
sess->next_early_rtcp_time = GST_CLOCK_TIME_NONE;
/* add SDES for this source when not already added */
if (!data.has_sdes)
session_sdes (sess, &data);
}
/* check for outdated collisions */
GST_DEBUG ("Timing out collisions");
rtp_source_timeout (sess->source, current_time,
data.interval * RTCP_INTERVAL_COLLISION_TIMEOUT,
running_time - sess->rtcp_feedback_retention_window);
if (sess->change_ssrc) {
GST_DEBUG ("need to change our SSRC (%08x)", own->ssrc);
g_hash_table_steal (sess->ssrcs[sess->mask_idx],
GINT_TO_POINTER (own->ssrc));
own->ssrc = rtp_session_create_new_ssrc (sess);
rtp_source_reset (own);
g_hash_table_insert (sess->ssrcs[sess->mask_idx],
GINT_TO_POINTER (own->ssrc), own);
g_free (sess->bye_reason);
sess->bye_reason = NULL;
sess->sent_bye = FALSE;
sess->change_ssrc = FALSE;
notify = TRUE;
GST_DEBUG ("changed our SSRC to %08x", own->ssrc);
}
sess->allow_early = TRUE;
RTP_SESSION_UNLOCK (sess);
if (notify)
g_object_notify (G_OBJECT (sess), "internal-ssrc");
/* push out the RTCP packet */
if (data.rtcp) {
gboolean do_not_suppress;
/* Give the user a change to add its own packet */
g_signal_emit (sess, rtp_session_signals[SIGNAL_ON_SENDING_RTCP], 0,
data.rtcp, data.is_early, &do_not_suppress);
if (sess->callbacks.send_rtcp && (do_not_suppress || !data.may_suppress)) {
guint packet_size;
/* close the RTCP packet */
gst_rtcp_buffer_end (data.rtcp);
packet_size = GST_BUFFER_SIZE (data.rtcp) + sess->header_len;
UPDATE_AVG (sess->stats.avg_rtcp_packet_size, packet_size);
GST_DEBUG ("%p, sending RTCP packet, avg size %u, %u", &sess->stats,
sess->stats.avg_rtcp_packet_size, packet_size);
result =
sess->callbacks.send_rtcp (sess, own, data.rtcp, sess->sent_bye,
sess->send_rtcp_user_data);
} else {
GST_DEBUG ("freeing packet callback: %p"
" do_not_suppress: %d may_suppress: %d",
sess->callbacks.send_rtcp, do_not_suppress, data.may_suppress);
gst_buffer_unref (data.rtcp);
}
}
return result;
}
void
rtp_session_request_early_rtcp (RTPSession * sess, GstClockTime current_time,
GstClockTimeDiff max_delay)
{
GstClockTime T_dither_max;
/* Implements the algorithm described in RFC 4585 section 3.5.2 */
RTP_SESSION_LOCK (sess);
/* Check if already requested */
/* RFC 4585 section 3.5.2 step 2 */
if (GST_CLOCK_TIME_IS_VALID (sess->next_early_rtcp_time))
goto dont_send;
/* Ignore the request a scheduled packet will be in time anyway */
if (current_time + max_delay > sess->next_rtcp_check_time)
goto dont_send;
/* RFC 4585 section 3.5.2 step 2b */
/* If the total sources is <=2, then there is only us and one peer */
if (sess->total_sources <= 2) {
T_dither_max = 0;
} else {
/* Divide by 2 because l = 0.5 */
T_dither_max = sess->next_rtcp_check_time - sess->last_rtcp_send_time;
T_dither_max /= 2;
}
/* RFC 4585 section 3.5.2 step 3 */
if (current_time + T_dither_max > sess->next_rtcp_check_time)
goto dont_send;
/* RFC 4585 section 3.5.2 step 4 */
if (sess->allow_early == FALSE)
goto dont_send;
if (T_dither_max) {
/* Schedule an early transmission later */
sess->next_early_rtcp_time = g_random_double () * T_dither_max +
current_time;
} else {
/* If no dithering, schedule it for NOW */
sess->next_early_rtcp_time = current_time;
}
RTP_SESSION_UNLOCK (sess);
/* notify app of need to send packet early
* and therefore of timeout change */
if (sess->callbacks.reconsider)
sess->callbacks.reconsider (sess, sess->reconsider_user_data);
return;
dont_send:
RTP_SESSION_UNLOCK (sess);
}
void
rtp_session_request_key_unit (RTPSession * sess, guint32 ssrc, gboolean fir)
{
guint i;
if (fir)
return;
for (i = 0; i < sess->rtcp_pli_requests->len; i++)
if (ssrc == g_array_index (sess->rtcp_pli_requests, guint32, i))
return;
g_array_append_val (sess->rtcp_pli_requests, ssrc);
}
static gboolean
has_pli_compare_func (gconstpointer a, gconstpointer ignored)
{
GstRTCPPacket packet;
packet.buffer = (GstBuffer *) a;
packet.offset = 0;
if (gst_rtcp_packet_get_type (&packet) == GST_RTCP_TYPE_PSFB &&
gst_rtcp_packet_fb_get_type (&packet) == GST_RTCP_PSFB_TYPE_PLI)
return TRUE;
else
return FALSE;
}
static gboolean
rtp_session_on_sending_rtcp (RTPSession * sess, GstBuffer * buffer,
gboolean early)
{
gboolean ret = FALSE;
RTP_SESSION_LOCK (sess);
while (sess->rtcp_pli_requests->len) {
GstRTCPPacket rtcppacket;
guint media_ssrc = g_array_index (sess->rtcp_pli_requests, guint32, 0);
RTPSource *media_src = g_hash_table_lookup (sess->ssrcs[sess->mask_idx],
GUINT_TO_POINTER (media_ssrc));
if (media_src && !rtp_source_has_retained (media_src,
has_pli_compare_func, NULL)) {
if (gst_rtcp_buffer_add_packet (buffer, GST_RTCP_TYPE_PSFB, &rtcppacket)) {
gst_rtcp_packet_fb_set_type (&rtcppacket, GST_RTCP_PSFB_TYPE_PLI);
gst_rtcp_packet_fb_set_sender_ssrc (&rtcppacket,
rtp_source_get_ssrc (sess->source));
gst_rtcp_packet_fb_set_media_ssrc (&rtcppacket, media_ssrc);
ret = TRUE;
} else {
/* Break because the packet is full, will put next request in a
* further packet
*/
break;
}
}
g_array_remove_index (sess->rtcp_pli_requests, 0);
}
RTP_SESSION_UNLOCK (sess);
return ret;
}
static void
rtp_session_send_rtcp (RTPSession * sess, GstClockTimeDiff max_delay)
{
GstClockTime now;
if (!sess->callbacks.send_rtcp)
return;
now = sess->callbacks.request_time (sess, sess->request_time_user_data);
rtp_session_request_early_rtcp (sess, now, max_delay);
}
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