mirrors/gstreamer
mirrors/gstreamer
1
1
Fork 0
mirror of https://gitlab.freedesktop.org/gstreamer/gstreamer.git synced 2024-12-01 06:01:04 +00:00
Code Issues 1 Projects Releases Wiki Activity
gstreamer/gst/rtpmanager/rtpsession.c
Wim Taymans 25fa19fba6 gst/rtpmanager/gstrtpjitterbuffer.c: Add some debug info. 
Original commit message from CVS:
* gst/rtpmanager/gstrtpjitterbuffer.c:
(gst_rtp_jitter_buffer_query):
Add some debug info.
* gst/rtpmanager/rtpsession.c: (rtp_session_init),
(rtp_session_send_rtp):
Store real user name in the session.
2007-05-04 12:32:27 +00:00

1897 lines
50 KiB
C
Raw Blame History

/* GStreamer
* Copyright (C) <2007> Wim Taymans <wim@fluendo.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 "rtpsession.h"
GST_DEBUG_CATEGORY_STATIC (rtp_session_debug);
#define GST_CAT_DEFAULT rtp_session_debug
/* signals and args */
enum
{
SIGNAL_ON_NEW_SSRC,
SIGNAL_ON_SSRC_COLLISION,
SIGNAL_ON_SSRC_VALIDATED,
SIGNAL_ON_BYE_SSRC,
SIGNAL_ON_BYE_TIMEOUT,
SIGNAL_ON_TIMEOUT,
LAST_SIGNAL
};
#define RTP_DEFAULT_BANDWIDTH 64000.0
#define RTP_DEFAULT_RTCP_BANDWIDTH 1000
enum
{
PROP_0
};
/* update average packet size, we keep this scaled by 16 to keep enough
* precision. */
#define UPDATE_AVG(avg, val) \
if ((avg) == 0) \
(avg) = (val) << 4; \
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 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 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::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,
G_TYPE_OBJECT);
/**
* 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,
G_TYPE_OBJECT);
/**
* 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,
G_TYPE_OBJECT);
/**
* 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,
G_TYPE_OBJECT);
/**
* 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,
G_TYPE_OBJECT);
/**
* 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,
G_TYPE_OBJECT);
GST_DEBUG_CATEGORY_INIT (rtp_session_debug, "rtpsession", 0, "RTP Session");
}
static void
rtp_session_init (RTPSession * sess)
{
gint i;
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);
/* create an active SSRC for this session manager */
sess->source = rtp_session_create_source (sess);
sess->source->validated = TRUE;
sess->stats.active_sources++;
/* default UDP header length */
sess->header_len = 28;
sess->mtu = 1400;
/* some default SDES entries */
sess->cname =
g_strdup_printf ("%s@%s", g_get_user_name (), g_get_host_name ());
sess->name = g_strdup (g_get_real_name ());
sess->tool = g_strdup ("GStreamer");
sess->first_rtcp = TRUE;
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_hash_table_destroy (sess->cnames);
g_object_unref (sess->source);
g_free (sess->cname);
g_free (sess->tool);
g_free (sess->bye_reason);
G_OBJECT_CLASS (rtp_session_parent_class)->finalize (object);
}
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) {
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) {
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
static void
on_new_ssrc (RTPSession * sess, RTPSource * source)
{
g_signal_emit (sess, rtp_session_signals[SIGNAL_ON_NEW_SSRC], 0, source);
}
static void
on_ssrc_collision (RTPSession * sess, RTPSource * source)
{
g_signal_emit (sess, rtp_session_signals[SIGNAL_ON_SSRC_COLLISION], 0,
source);
}
static void
on_ssrc_validated (RTPSession * sess, RTPSource * source)
{
g_signal_emit (sess, rtp_session_signals[SIGNAL_ON_SSRC_VALIDATED], 0,
source);
}
static void
on_bye_ssrc (RTPSession * sess, RTPSource * source)
{
/* notify app that reconsideration should be performed */
g_signal_emit (sess, rtp_session_signals[SIGNAL_ON_BYE_SSRC], 0, source);
}
static void
on_bye_timeout (RTPSession * sess, RTPSource * source)
{
g_signal_emit (sess, rtp_session_signals[SIGNAL_ON_BYE_TIMEOUT], 0, source);
}
static void
on_timeout (RTPSession * sess, RTPSource * source)
{
g_signal_emit (sess, rtp_session_signals[SIGNAL_ON_TIMEOUT], 0, 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));
sess->callbacks.process_rtp = callbacks->process_rtp;
sess->callbacks.send_rtp = callbacks->send_rtp;
sess->callbacks.send_rtcp = callbacks->send_rtcp;
sess->callbacks.clock_rate = callbacks->clock_rate;
sess->callbacks.get_time = callbacks->get_time;
sess->callbacks.reconsider = callbacks->reconsider;
sess->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));
sess->stats.bandwidth = bandwidth;
}
/**
* rtp_session_get_bandwidth:
* @sess: an #RTPSession
*
* Get the session bandwidth.
*
* Returns: the session bandwidth.
*/
gdouble
rtp_session_get_bandwidth (RTPSession * sess)
{
g_return_val_if_fail (RTP_IS_SESSION (sess), 0);
return sess->stats.bandwidth;
}
/**
* rtp_session_set_rtcp_bandwidth:
* @sess: an #RTPSession
* @bandwidth: the RTCP bandwidth
*
* Set the bandwidth that should be used for RTCP
* messages.
*/
void
rtp_session_set_rtcp_bandwidth (RTPSession * sess, gdouble bandwidth)
{
g_return_if_fail (RTP_IS_SESSION (sess));
sess->stats.rtcp_bandwidth = bandwidth;
}
/**
* rtp_session_get_rtcp_bandwidth:
* @sess: an #RTPSession
*
* Get the session bandwidth used for RTCP.
*
* Returns: The bandwidth used for RTCP messages.
*/
gdouble
rtp_session_get_rtcp_bandwidth (RTPSession * sess)
{
g_return_val_if_fail (RTP_IS_SESSION (sess), 0.0);
return sess->stats.rtcp_bandwidth;
}
/**
* rtp_session_set_cname:
* @sess: an #RTPSession
* @cname: a CNAME for the session
*
* Set the CNAME for the session.
*/
void
rtp_session_set_cname (RTPSession * sess, const gchar * cname)
{
g_return_if_fail (RTP_IS_SESSION (sess));
g_free (sess->cname);
sess->cname = g_strdup (cname);
}
/**
* rtp_session_get_cname:
* @sess: an #RTPSession
*
* Get the currently configured CNAME for the session.
*
* Returns: The CNAME. g_free after usage.
*/
gchar *
rtp_session_get_cname (RTPSession * sess)
{
g_return_val_if_fail (RTP_IS_SESSION (sess), NULL);
return g_strdup (sess->cname);
}
/**
* rtp_session_set_name:
* @sess: an #RTPSession
* @name: a NAME for the session
*
* Set the NAME for the session.
*/
void
rtp_session_set_name (RTPSession * sess, const gchar * name)
{
g_return_if_fail (RTP_IS_SESSION (sess));
g_free (sess->name);
sess->name = g_strdup (name);
}
/**
* rtp_session_get_name:
* @sess: an #RTPSession
*
* Get the currently configured NAME for the session.
*
* Returns: The NAME. g_free after usage.
*/
gchar *
rtp_session_get_name (RTPSession * sess)
{
g_return_val_if_fail (RTP_IS_SESSION (sess), NULL);
return g_strdup (sess->name);
}
/**
* rtp_session_set_email:
* @sess: an #RTPSession
* @email: an EMAIL for the session
*
* Set the EMAIL the session.
*/
void
rtp_session_set_email (RTPSession * sess, const gchar * email)
{
g_return_if_fail (RTP_IS_SESSION (sess));
g_free (sess->email);
sess->email = g_strdup (email);
}
/**
* rtp_session_get_email:
* @sess: an #RTPSession
*
* Get the currently configured EMAIL of the session.
*
* Returns: The EMAIL. g_free after usage.
*/
gchar *
rtp_session_get_email (RTPSession * sess)
{
g_return_val_if_fail (RTP_IS_SESSION (sess), NULL);
return g_strdup (sess->email);
}
/**
* rtp_session_set_phone:
* @sess: an #RTPSession
* @phone: a PHONE for the session
*
* Set the PHONE the session.
*/
void
rtp_session_set_phone (RTPSession * sess, const gchar * phone)
{
g_return_if_fail (RTP_IS_SESSION (sess));
g_free (sess->phone);
sess->phone = g_strdup (phone);
}
/**
* rtp_session_get_location:
* @sess: an #RTPSession
*
* Get the currently configured PHONE of the session.
*
* Returns: The PHONE. g_free after usage.
*/
gchar *
rtp_session_get_phone (RTPSession * sess)
{
g_return_val_if_fail (RTP_IS_SESSION (sess), NULL);
return g_strdup (sess->phone);
}
/**
* rtp_session_set_location:
* @sess: an #RTPSession
* @location: a LOCATION for the session
*
* Set the LOCATION the session.
*/
void
rtp_session_set_location (RTPSession * sess, const gchar * location)
{
g_return_if_fail (RTP_IS_SESSION (sess));
g_free (sess->location);
sess->location = g_strdup (location);
}
/**
* rtp_session_get_location:
* @sess: an #RTPSession
*
* Get the currently configured LOCATION of the session.
*
* Returns: The LOCATION. g_free after usage.
*/
gchar *
rtp_session_get_location (RTPSession * sess)
{
g_return_val_if_fail (RTP_IS_SESSION (sess), NULL);
return g_strdup (sess->location);
}
/**
* rtp_session_set_tool:
* @sess: an #RTPSession
* @tool: a TOOL for the session
*
* Set the TOOL the session.
*/
void
rtp_session_set_tool (RTPSession * sess, const gchar * tool)
{
g_return_if_fail (RTP_IS_SESSION (sess));
g_free (sess->tool);
sess->tool = g_strdup (tool);
}
/**
* rtp_session_get_tool:
* @sess: an #RTPSession
*
* Get the currently configured TOOL of the session.
*
* Returns: The TOOL. g_free after usage.
*/
gchar *
rtp_session_get_tool (RTPSession * sess)
{
g_return_val_if_fail (RTP_IS_SESSION (sess), NULL);
return g_strdup (sess->tool);
}
/**
* rtp_session_set_note:
* @sess: an #RTPSession
* @note: a NOTE for the session
*
* Set the NOTE the session.
*/
void
rtp_session_set_note (RTPSession * sess, const gchar * note)
{
g_return_if_fail (RTP_IS_SESSION (sess));
g_free (sess->note);
sess->note = g_strdup (note);
}
/**
* rtp_session_get_note:
* @sess: an #RTPSession
*
* Get the currently configured NOTE of the session.
*
* Returns: The NOTE. g_free after usage.
*/
gchar *
rtp_session_get_note (RTPSession * sess)
{
g_return_val_if_fail (RTP_IS_SESSION (sess), NULL);
return g_strdup (sess->note);
}
static GstFlowReturn
source_push_rtp (RTPSource * source, GstBuffer * buffer, RTPSession * session)
{
GstFlowReturn result = GST_FLOW_OK;
if (source == session->source) {
GST_DEBUG ("source %08x pushed sender RTP packet", source->ssrc);
RTP_SESSION_UNLOCK (session);
if (session->callbacks.send_rtp)
result =
session->callbacks.send_rtp (session, source, buffer,
session->user_data);
else
gst_buffer_unref (buffer);
} else {
GST_DEBUG ("source %08x pushed receiver RTP packet", source->ssrc);
RTP_SESSION_UNLOCK (session);
if (session->callbacks.process_rtp)
result =
session->callbacks.process_rtp (session, source, buffer,
session->user_data);
else
gst_buffer_unref (buffer);
}
RTP_SESSION_LOCK (session);
return result;
}
static gint
source_clock_rate (RTPSource * source, guint8 pt, RTPSession * session)
{
gint result;
if (session->callbacks.clock_rate)
result = session->callbacks.clock_rate (session, pt, session->user_data);
else
result = -1;
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)
{
/* FIXME, do collision check */
return FALSE;
}
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);
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))
on_ssrc_collision (sess, source);
}
/* update last activity */
source->last_activity = arrival->time;
if (rtp)
source->last_rtp_activity = arrival->time;
return source;
}
/**
* 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;
}
/**
* 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);
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;
}
source = rtp_source_new (ssrc);
g_object_ref (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++;
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)
{
/* get time or arrival */
if (sess->callbacks.get_time)
arrival->time = sess->callbacks.get_time (sess, sess->user_data);
else
arrival->time = GST_CLOCK_TIME_NONE;
/* 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
*
* 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)
{
GstFlowReturn result;
guint32 ssrc;
RTPSource *source;
gboolean created;
gboolean prevsender, prevactive;
RTPArrivalStats arrival;
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);
/* 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);
prevsender = RTP_SOURCE_IS_SENDER (source);
prevactive = RTP_SOURCE_IS_ACTIVE (source);
/* we need to ref so that we can process the CSRCs later */
gst_buffer_ref (buffer);
/* 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 (created)
on_new_ssrc (sess, source);
if (source->validated) {
guint8 i, count;
gboolean created;
/* for validated sources, we add the CSRCs as well */
count = gst_rtp_buffer_get_csrc_count (buffer);
for (i = 0; i < count; i++) {
guint32 csrc;
RTPSource *csrc_src;
csrc = gst_rtp_buffer_get_csrc (buffer, i);
/* get source */
csrc_src = obtain_source (sess, csrc, &created, &arrival, TRUE);
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, source);
}
}
}
gst_buffer_unref (buffer);
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;
}
}
/* A Sender report contains statistics about how the sender is doing. This
* includes timing informataion about the relation between RTP and NTP
* timestamps is it using 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)
{
guint32 senderssrc, rtptime, packet_count, octet_count;
guint64 ntptime;
guint count, i;
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->time));
source = obtain_source (sess, senderssrc, &created, arrival, FALSE);
prevsender = RTP_SOURCE_IS_SENDER (source);
/* first update the source */
rtp_source_process_sr (source, ntptime, rtptime, packet_count, octet_count,
arrival->time);
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);
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: %08x, %u", 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, fractionlost, packetslost,
exthighestseq, jitter, lsr, dlsr);
}
}
}
/* 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;
guint count, i;
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 (created)
on_new_ssrc (sess, source);
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);
if (ssrc == sess->source->ssrc) {
rtp_source_process_rb (source, fractionlost, packetslost,
exthighestseq, jitter, lsr, dlsr);
}
}
}
/* FIXME, we're just printing this for now... */
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;
ssrc = gst_rtcp_packet_sdes_get_ssrc (packet);
GST_DEBUG ("item %d, SSRC %08x", i, ssrc);
more_entries = gst_rtcp_packet_sdes_first_entry (packet);
j = 0;
while (more_entries) {
GstRTCPSDESType type;
guint8 len;
guint8 *data;
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);
more_entries = gst_rtcp_packet_sdes_next_entry (packet);
j++;
}
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;
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);
/* store time for when we need to time out this source */
source->bye_time = arrival->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->time < sess->next_rtcp_check_time) {
GstClockTime time_remaining;
time_remaining = sess->next_rtcp_check_time - arrival->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->time;
/* notify app of reconsideration */
if (sess->callbacks.reconsider)
sess->callbacks.reconsider (sess, sess->user_data);
}
}
if (created)
on_new_ssrc (sess, source);
on_bye_ssrc (sess, source);
}
g_free (reason);
}
static void
rtp_session_process_app (RTPSession * sess, GstRTCPPacket * packet,
RTPArrivalStats * arrival)
{
GST_DEBUG ("received APP");
}
/**
* rtp_session_process_rtcp:
* @sess: and #RTPSession
* @buffer: an RTCP buffer
*
* Process an RTCP buffer in the session manager.
*
* Returns: a #GstFlowReturn.
*/
GstFlowReturn
rtp_session_process_rtcp (RTPSession * sess, GstBuffer * buffer)
{
GstRTCPPacket packet;
gboolean more, is_bye = FALSE;
RTPArrivalStats arrival;
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);
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);
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;
rtp_session_process_bye (sess, &packet, &arrival);
break;
case GST_RTCP_TYPE_APP:
rtp_session_process_app (sess, &packet, &arrival);
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);
}
RTP_SESSION_UNLOCK (sess);
gst_buffer_unref (buffer);
return GST_FLOW_OK;
/* ERRORS */
invalid_packet:
{
GST_DEBUG ("invalid RTCP packet received");
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
* @buffer: an RTP buffer
*
* Send the RTP buffer in the session manager. This function takes ownership of
* @buffer.
*
* Returns: a #GstFlowReturn.
*/
GstFlowReturn
rtp_session_send_rtp (RTPSession * sess, GstBuffer * buffer)
{
GstFlowReturn result;
RTPSource *source;
gboolean prevsender;
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;
GST_DEBUG ("received RTP packet for sending");
RTP_SESSION_LOCK (sess);
source = sess->source;
/* update last activity */
if (sess->callbacks.get_time)
source->last_rtp_activity =
sess->callbacks.get_time (sess, sess->user_data);
prevsender = RTP_SOURCE_IS_SENDER (source);
/* we use our own source to send */
result = rtp_source_send_rtp (sess->source, buffer);
if (RTP_SOURCE_IS_SENDER (source) && !prevsender)
sess->stats.sender_sources++;
RTP_SESSION_UNLOCK (sess);
return result;
/* ERRORS */
invalid_packet:
{
gst_buffer_unref (buffer);
GST_DEBUG ("invalid RTP packet received");
return GST_FLOW_OK;
}
}
static GstClockTime
calculate_rtcp_interval (RTPSession * sess, gboolean deterministic,
gboolean first)
{
GstClockTime result;
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 = rtp_stats_add_rtcp_jitter (&sess->stats, result);
GST_DEBUG ("next interval: %" GST_TIME_FORMAT, GST_TIME_ARGS (result));
return result;
}
/**
* rtp_session_send_bye:
* @sess: an #RTPSession
* @reason: a reason or NULL
*
* Stop the current @sess and schedule a BYE message for the other members.
*
* Returns: a #GstFlowReturn.
*/
GstFlowReturn
rtp_session_send_bye (RTPSession * sess, const gchar * reason)
{
GstFlowReturn result = GST_FLOW_OK;
RTPSource *source;
GstClockTime current, interval;
g_return_val_if_fail (RTP_IS_SESSION (sess), GST_FLOW_ERROR);
RTP_SESSION_LOCK (sess);
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 */
sess->bye_reason = g_strdup (reason);
sess->stats.avg_rtcp_packet_size = 100;
sess->stats.bye_members = 1;
sess->first_rtcp = TRUE;
sess->sent_bye = FALSE;
/* get current time */
if (sess->callbacks.get_time)
current = sess->callbacks.get_time (sess, sess->user_data);
else
current = 0;
/* reschedule transmission */
sess->last_rtcp_send_time = current;
interval = calculate_rtcp_interval (sess, FALSE, TRUE);
sess->next_rtcp_check_time = current + interval;
GST_DEBUG ("Schedule BYE for %" GST_TIME_FORMAT ", %" GST_TIME_FORMAT,
GST_TIME_ARGS (interval), GST_TIME_ARGS (sess->next_rtcp_check_time));
/* notify app of reconsideration */
if (sess->callbacks.reconsider)
sess->callbacks.reconsider (sess, sess->user_data);
done:
RTP_SESSION_UNLOCK (sess);
return result;
}
/**
* rtp_session_next_timeout:
* @sess: an #RTPSession
* @time: the current 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 time.
*/
GstClockTime
rtp_session_next_timeout (RTPSession * sess, GstClockTime time)
{
GstClockTime result;
g_return_val_if_fail (RTP_IS_SESSION (sess), GST_FLOW_ERROR);
RTP_SESSION_LOCK (sess);
result = sess->next_rtcp_check_time;
if (sess->source->received_bye) {
if (sess->sent_bye)
result = GST_CLOCK_TIME_NONE;
else if (sess->stats.active_sources >= 50)
/* reconsider BYE if members >= 50 */
result = time + calculate_rtcp_interval (sess, FALSE, TRUE);
} else {
if (sess->first_rtcp)
/* we are called for the first time */
result = time + calculate_rtcp_interval (sess, FALSE, TRUE);
else if (sess->next_rtcp_check_time < time)
/* get a new timeout when we need to */
result = time + calculate_rtcp_interval (sess, FALSE, FALSE);
}
sess->next_rtcp_check_time = result;
GST_DEBUG ("next timeout: %" GST_TIME_FORMAT, GST_TIME_ARGS (result));
RTP_SESSION_UNLOCK (sess);
return result;
}
typedef struct
{
RTPSession *sess;
GstBuffer *rtcp;
GstClockTime time;
GstClockTime interval;
GstRTCPPacket packet;
gboolean is_bye;
gboolean has_sdes;
} 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;
/* 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);
/* convert clock time to NTP time */
ntptime = gst_util_uint64_scale (data->time, (1LL << 32), GST_SECOND);
ntptime += (2208988800LL << 32);
rtptime = 0;
/* fill in sender report info, FIXME RTP timestamps missing */
gst_rtcp_packet_sr_set_sender_info (packet, own->ssrc,
ntptime, rtptime, own->stats.packets_sent, own->stats.octets_sent);
} 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);
}
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)) {
RTPSourceStats *stats;
guint64 extended_max, expected;
guint64 expected_interval, received_interval, ntptime;
gint64 lost, lost_interval;
guint32 fraction, LSR, DLSR;
GstClockTime time;
stats = &source->stats;
extended_max = stats->cycles + stats->max_seq;
expected = extended_max - stats->base_seq + 1;
GST_DEBUG ("ext_max %d, expected %d, received %d, base_seq %d",
extended_max, expected, stats->packets_received, stats->base_seq);
lost = expected - stats->packets_received;
lost = CLAMP (lost, -0x800000, 0x7fffff);
expected_interval = expected - stats->prev_expected;
stats->prev_expected = expected;
received_interval = stats->packets_received - stats->prev_received;
stats->prev_received = stats->packets_received;
lost_interval = expected_interval - received_interval;
if (expected_interval == 0 || lost_interval <= 0)
fraction = 0;
else
fraction = (lost_interval << 8) / expected_interval;
GST_DEBUG ("add RR for SSRC %08x", source->ssrc);
/* we scaled the jitter up for additional precision */
GST_DEBUG ("fraction %d, lost %d, extseq %u, jitter %d", fraction, lost,
extended_max, stats->jitter >> 4);
if (rtp_source_get_last_sr (source, &ntptime, NULL, NULL, NULL, &time)) {
GstClockTime diff;
/* LSR is middle bits of the last ntptime */
LSR = (ntptime >> 16) & 0xffffffff;
diff = data->time - time;
GST_DEBUG ("last SR time diff %" GST_TIME_FORMAT, GST_TIME_ARGS (diff));
/* DLSR, delay since last SR is expressed in 1/65536 second units */
DLSR = gst_util_uint64_scale_int (diff, 65536, GST_SECOND);
} else {
/* No valid SR received, LSR/DLSR are set to 0 then */
LSR = 0;
DLSR = 0;
}
GST_DEBUG ("LSR %08x, DLSR %08x", LSR, DLSR);
/* packet is not yet filled, add report block for this source. */
gst_rtcp_packet_add_rb (packet, source->ssrc, fraction, lost,
extended_max, stats->jitter >> 4, LSR, DLSR);
}
}
}
/* perform cleanup of sources that timed out */
static gboolean
session_cleanup (const gchar * key, RTPSource * source, ReportData * data)
{
gboolean remove = FALSE;
gboolean byetimeout = 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->time > source->bye_time &&
data->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->time > source->last_activity) {
interval = MAX (data->interval * 5, 5 * GST_SECOND);
if (data->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->time > source->last_rtp_activity) {
interval = MAX (data->interval * 2, 5 * GST_SECOND);
if (data->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--;
}
}
}
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);
}
return remove;
}
static void
session_sdes (RTPSession * sess, ReportData * data)
{
GstRTCPPacket *packet = &data->packet;
/* 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);
gst_rtcp_packet_sdes_add_entry (packet, GST_RTCP_SDES_CNAME,
strlen (sess->cname), (guint8 *) sess->cname);
/* other SDES items must only be added at regular intervals and only when the
* user requests to since it might be a privacy problem */
#if 0
gst_rtcp_packet_sdes_add_entry (&packet, GST_RTCP_SDES_NAME,
strlen (sess->name), (guint8 *) sess->name);
gst_rtcp_packet_sdes_add_entry (&packet, GST_RTCP_SDES_TOOL,
strlen (sess->tool), (guint8 *) sess->tool);
#endif
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 time, ReportData * data)
{
GstClockTime new_send_time;
gboolean result;
/* no need to check yet */
if (sess->next_rtcp_check_time > time) {
GST_DEBUG ("no check time yet");
return FALSE;
}
/* perform forward reconsideration */
new_send_time = rtp_stats_add_rtcp_jitter (&sess->stats, data->interval);
GST_DEBUG ("forward reconsideration %" GST_TIME_FORMAT,
GST_TIME_ARGS (new_send_time));
new_send_time += sess->last_rtcp_send_time;
/* check if reconsideration */
if (time < new_send_time) {
GST_DEBUG ("reconsider RTCP for %" GST_TIME_FORMAT,
GST_TIME_ARGS (new_send_time));
result = FALSE;
/* store new check time */
sess->next_rtcp_check_time = new_send_time;
} else {
result = TRUE;
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 = time + new_send_time;
}
return result;
}
/**
* rtp_session_on_timeout:
* @sess: an #RTPSession
*
* 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 time)
{
GstFlowReturn result = GST_FLOW_OK;
ReportData data;
g_return_val_if_fail (RTP_IS_SESSION (sess), GST_FLOW_ERROR);
data.sess = sess;
data.rtcp = NULL;
data.time = time;
data.is_bye = FALSE;
data.has_sdes = FALSE;
GST_DEBUG ("reporting at %" GST_TIME_FORMAT, GST_TIME_ARGS (time));
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);
/* first perform cleanups */
g_hash_table_foreach_remove (sess->ssrcs[sess->mask_idx],
(GHRFunc) session_cleanup, &data);
/* see if we need to generate SR or RR packets */
if (is_rtcp_time (sess, time, &data)) {
if (sess->source->received_bye) {
/* generate BYE instead */
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) {
guint size;
/* we keep track of the last report time in order to timeout inactive
* receivers or senders */
sess->last_rtcp_send_time = data.time;
sess->first_rtcp = FALSE;
/* add SDES for this source when not already added */
if (!data.has_sdes)
session_sdes (sess, &data);
/* update average RTCP size before sending */
size = GST_BUFFER_SIZE (data.rtcp) + sess->header_len;
UPDATE_AVG (sess->stats.avg_rtcp_packet_size, size);
}
RTP_SESSION_UNLOCK (sess);
/* push out the RTCP packet */
if (data.rtcp) {
/* close the RTCP packet */
gst_rtcp_buffer_end (data.rtcp);
if (sess->callbacks.send_rtcp)
result = sess->callbacks.send_rtcp (sess, sess->source, data.rtcp,
sess->user_data);
else
gst_buffer_unref (data.rtcp);
}
return result;
}
Reference in a new issue View git blame Copy permalink