mirrors/gstreamer
mirrors/gstreamer
1
1
Fork 0
mirror of https://gitlab.freedesktop.org/gstreamer/gstreamer.git synced 2024-12-21 07:46:38 +00:00
Code Issues 1 Projects Releases Wiki Activity
gstreamer/gst/rtpmanager/gstrtpsession.c
Olivier Crête b9ceafe5af rtpsession: Demux RTCP buffers from the RTP stream 
If there are RTCP buffers in the RTP stream, process them as
RTCP. This way, we want receive streams following RFC 5761

https://bugzilla.gnome.org/show_bug.cgi?id=687657
2013-09-13 16:25:49 +02:00

2383 lines
73 KiB
C
Raw Blame History

/* GStreamer
* Copyright (C) <2007> Wim Taymans <wim.taymans@gmail.com>
*
* This library is free software; you can redistribute it and/or
* modify it under the terms of the GNU Library General Public
* License as published by the Free Software Foundation; either
* version 2 of the License, or (at your option) any later version.
*
* This library is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Library General Public License for more details.
*
* You should have received a copy of the GNU Library General Public
* License along with this library; if not, write to the
* Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
* Boston, MA 02110-1301, USA.
*/
/**
* SECTION:element-gstrtpsession
* @see_also: gstrtpjitterbuffer, gstrtpbin, gstrtpptdemux, gstrtpssrcdemux
*
* The RTP session manager models one participant with a unique SSRC in an RTP
* session. This session can be used to send and receive RTP and RTCP packets.
* Based on what REQUEST pads are requested from the session manager, specific
* functionality can be activated.
*
* The session manager currently implements RFC 3550 including:
* <itemizedlist>
* <listitem>
* <para>RTP packet validation based on consecutive sequence numbers.</para>
* </listitem>
* <listitem>
* <para>Maintainance of the SSRC participant database.</para>
* </listitem>
* <listitem>
* <para>Keeping per participant statistics based on received RTCP packets.</para>
* </listitem>
* <listitem>
* <para>Scheduling of RR/SR RTCP packets.</para>
* </listitem>
* </itemizedlist>
*
* The gstrtpsession will not demux packets based on SSRC or payload type, nor will
* it correct for packet reordering and jitter. Use #GstRtpsSrcDemux,
* #GstRtpPtDemux and GstRtpJitterBuffer in addition to #GstRtpSession to
* perform these tasks. It is usually a good idea to use #GstRtpBin, which
* combines all these features in one element.
*
* To use #GstRtpSession as an RTP receiver, request a recv_rtp_sink pad, which will
* automatically create recv_rtp_src pad. Data received on the recv_rtp_sink pad
* will be processed in the session and after being validated forwarded on the
* recv_rtp_src pad.
*
* To also use #GstRtpSession as an RTCP receiver, request a recv_rtcp_sink pad,
* which will automatically create a sync_src pad. Packets received on the RTCP
* pad will be used by the session manager to update the stats and database of
* the other participants. SR packets will be forwarded on the sync_src pad
* so that they can be used to perform inter-stream synchronisation when needed.
*
* If you want the session manager to generate and send RTCP packets, request
* the send_rtcp_src pad. Packet pushed on this pad contain SR/RR RTCP reports
* that should be sent to all participants in the session.
*
* To use #GstRtpSession as a sender, request a send_rtp_sink pad, which will
* automatically create a send_rtp_src pad. The session manager will modify the
* SSRC in the RTP packets to its own SSRC and wil forward the packets on the
* send_rtp_src pad after updating its internal state.
*
* The session manager needs the clock-rate of the payload types it is handling
* and will signal the #GstRtpSession::request-pt-map signal when it needs such a
* mapping. One can clear the cached values with the #GstRtpSession::clear-pt-map
* signal.
*
* <refsect2>
* <title>Example pipelines</title>
* |[
* gst-launch-1.0 udpsrc port=5000 caps="application/x-rtp, ..." ! .recv_rtp_sink gstrtpsession .recv_rtp_src ! rtptheoradepay ! theoradec ! xvimagesink
* ]| Receive theora RTP packets from port 5000 and send them to the depayloader,
* decoder and display. Note that the application/x-rtp caps on udpsrc should be
* configured based on some negotiation process such as RTSP for this pipeline
* to work correctly.
* |[
* gst-launch-1.0 udpsrc port=5000 caps="application/x-rtp, ..." ! .recv_rtp_sink gstrtpsession name=session \
* .recv_rtp_src ! rtptheoradepay ! theoradec ! xvimagesink \
* udpsrc port=5001 caps="application/x-rtcp" ! session.recv_rtcp_sink
* ]| Receive theora RTP packets from port 5000 and send them to the depayloader,
* decoder and display. Receive RTCP packets from port 5001 and process them in
* the session manager.
* Note that the application/x-rtp caps on udpsrc should be
* configured based on some negotiation process such as RTSP for this pipeline
* to work correctly.
* |[
* gst-launch-1.0 videotestsrc ! theoraenc ! rtptheorapay ! .send_rtp_sink gstrtpsession .send_rtp_src ! udpsink port=5000
* ]| Send theora RTP packets through the session manager and out on UDP port
* 5000.
* |[
* gst-launch-1.0 videotestsrc ! theoraenc ! rtptheorapay ! .send_rtp_sink gstrtpsession name=session .send_rtp_src \
* ! udpsink port=5000 session.send_rtcp_src ! udpsink port=5001
* ]| Send theora RTP packets through the session manager and out on UDP port
* 5000. Send RTCP packets on port 5001. Note that this pipeline will not preroll
* correctly because the second udpsink will not preroll correctly (no RTCP
* packets are sent in the PAUSED state). Applications should manually set and
* keep (see gst_element_set_locked_state()) the RTCP udpsink to the PLAYING state.
* </refsect2>
*
* Last reviewed on 2007-05-28 (0.10.5)
*/
#ifdef HAVE_CONFIG_H
#include "config.h"
#endif
#include <gst/rtp/gstrtpbuffer.h>
#include <gst/glib-compat-private.h>
#include "gstrtpsession.h"
#include "rtpsession.h"
GST_DEBUG_CATEGORY_STATIC (gst_rtp_session_debug);
#define GST_CAT_DEFAULT gst_rtp_session_debug
/* sink pads */
static GstStaticPadTemplate rtpsession_recv_rtp_sink_template =
GST_STATIC_PAD_TEMPLATE ("recv_rtp_sink",
GST_PAD_SINK,
GST_PAD_REQUEST,
GST_STATIC_CAPS ("application/x-rtp")
);
static GstStaticPadTemplate rtpsession_recv_rtcp_sink_template =
GST_STATIC_PAD_TEMPLATE ("recv_rtcp_sink",
GST_PAD_SINK,
GST_PAD_REQUEST,
GST_STATIC_CAPS ("application/x-rtcp")
);
static GstStaticPadTemplate rtpsession_send_rtp_sink_template =
GST_STATIC_PAD_TEMPLATE ("send_rtp_sink",
GST_PAD_SINK,
GST_PAD_REQUEST,
GST_STATIC_CAPS ("application/x-rtp")
);
/* src pads */
static GstStaticPadTemplate rtpsession_recv_rtp_src_template =
GST_STATIC_PAD_TEMPLATE ("recv_rtp_src",
GST_PAD_SRC,
GST_PAD_SOMETIMES,
GST_STATIC_CAPS ("application/x-rtp")
);
static GstStaticPadTemplate rtpsession_sync_src_template =
GST_STATIC_PAD_TEMPLATE ("sync_src",
GST_PAD_SRC,
GST_PAD_SOMETIMES,
GST_STATIC_CAPS ("application/x-rtcp")
);
static GstStaticPadTemplate rtpsession_send_rtp_src_template =
GST_STATIC_PAD_TEMPLATE ("send_rtp_src",
GST_PAD_SRC,
GST_PAD_SOMETIMES,
GST_STATIC_CAPS ("application/x-rtp")
);
static GstStaticPadTemplate rtpsession_send_rtcp_src_template =
GST_STATIC_PAD_TEMPLATE ("send_rtcp_src",
GST_PAD_SRC,
GST_PAD_REQUEST,
GST_STATIC_CAPS ("application/x-rtcp")
);
/* signals and args */
enum
{
SIGNAL_REQUEST_PT_MAP,
SIGNAL_CLEAR_PT_MAP,
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,
LAST_SIGNAL
};
#define DEFAULT_BANDWIDTH RTP_STATS_BANDWIDTH
#define DEFAULT_RTCP_FRACTION (RTP_STATS_BANDWIDTH * RTP_STATS_RTCP_FRACTION)
#define DEFAULT_RTCP_RR_BANDWIDTH -1
#define DEFAULT_RTCP_RS_BANDWIDTH -1
#define DEFAULT_SDES NULL
#define DEFAULT_NUM_SOURCES 0
#define DEFAULT_NUM_ACTIVE_SOURCES 0
#define DEFAULT_USE_PIPELINE_CLOCK FALSE
#define DEFAULT_RTCP_MIN_INTERVAL (RTP_STATS_MIN_INTERVAL * GST_SECOND)
#define DEFAULT_PROBATION RTP_DEFAULT_PROBATION
enum
{
PROP_0,
PROP_BANDWIDTH,
PROP_RTCP_FRACTION,
PROP_RTCP_RR_BANDWIDTH,
PROP_RTCP_RS_BANDWIDTH,
PROP_SDES,
PROP_NUM_SOURCES,
PROP_NUM_ACTIVE_SOURCES,
PROP_INTERNAL_SESSION,
PROP_USE_PIPELINE_CLOCK,
PROP_RTCP_MIN_INTERVAL,
PROP_PROBATION,
PROP_LAST
};
#define GST_RTP_SESSION_GET_PRIVATE(obj) \
(G_TYPE_INSTANCE_GET_PRIVATE ((obj), GST_TYPE_RTP_SESSION, GstRtpSessionPrivate))
#define GST_RTP_SESSION_LOCK(sess) g_mutex_lock (&(sess)->priv->lock)
#define GST_RTP_SESSION_UNLOCK(sess) g_mutex_unlock (&(sess)->priv->lock)
#define GST_RTP_SESSION_WAIT(sess) g_cond_wait (&(sess)->priv->cond, &(sess)->priv->lock)
#define GST_RTP_SESSION_SIGNAL(sess) g_cond_signal (&(sess)->priv->cond)
struct _GstRtpSessionPrivate
{
GMutex lock;
GCond cond;
GstClock *sysclock;
RTPSession *session;
/* thread for sending out RTCP */
GstClockID id;
gboolean stop_thread;
GThread *thread;
gboolean thread_stopped;
gboolean wait_send;
/* caps mapping */
GHashTable *ptmap;
GstClockTime send_latency;
gboolean use_pipeline_clock;
};
/* callbacks to handle actions from the session manager */
static GstFlowReturn gst_rtp_session_process_rtp (RTPSession * sess,
RTPSource * src, GstBuffer * buffer, gpointer user_data);
static GstFlowReturn gst_rtp_session_send_rtp (RTPSession * sess,
RTPSource * src, gpointer data, gpointer user_data);
static GstFlowReturn gst_rtp_session_send_rtcp (RTPSession * sess,
RTPSource * src, GstBuffer * buffer, gboolean eos, gpointer user_data);
static GstFlowReturn gst_rtp_session_sync_rtcp (RTPSession * sess,
GstBuffer * buffer, gpointer user_data);
static gint gst_rtp_session_clock_rate (RTPSession * sess, guint8 payload,
gpointer user_data);
static void gst_rtp_session_reconsider (RTPSession * sess, gpointer user_data);
static void gst_rtp_session_request_key_unit (RTPSession * sess,
gboolean all_headers, gpointer user_data);
static GstClockTime gst_rtp_session_request_time (RTPSession * session,
gpointer user_data);
static void gst_rtp_session_notify_nack (RTPSession * sess,
guint16 seqnum, guint16 blp, gpointer user_data);
static RTPSessionCallbacks callbacks = {
gst_rtp_session_process_rtp,
gst_rtp_session_send_rtp,
gst_rtp_session_sync_rtcp,
gst_rtp_session_send_rtcp,
gst_rtp_session_clock_rate,
gst_rtp_session_reconsider,
gst_rtp_session_request_key_unit,
gst_rtp_session_request_time,
gst_rtp_session_notify_nack
};
/* GObject vmethods */
static void gst_rtp_session_finalize (GObject * object);
static void gst_rtp_session_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec);
static void gst_rtp_session_get_property (GObject * object, guint prop_id,
GValue * value, GParamSpec * pspec);
/* GstElement vmethods */
static GstStateChangeReturn gst_rtp_session_change_state (GstElement * element,
GstStateChange transition);
static GstPad *gst_rtp_session_request_new_pad (GstElement * element,
GstPadTemplate * templ, const gchar * name, const GstCaps * caps);
static void gst_rtp_session_release_pad (GstElement * element, GstPad * pad);
static gboolean gst_rtp_session_sink_setcaps (GstPad * pad,
GstRtpSession * rtpsession, GstCaps * caps);
static gboolean gst_rtp_session_setcaps_send_rtp (GstPad * pad,
GstRtpSession * rtpsession, GstCaps * caps);
static void gst_rtp_session_clear_pt_map (GstRtpSession * rtpsession);
static guint gst_rtp_session_signals[LAST_SIGNAL] = { 0 };
static void
on_new_ssrc (RTPSession * session, RTPSource * src, GstRtpSession * sess)
{
g_signal_emit (sess, gst_rtp_session_signals[SIGNAL_ON_NEW_SSRC], 0,
src->ssrc);
}
static void
on_ssrc_collision (RTPSession * session, RTPSource * src, GstRtpSession * sess)
{
GstPad *recv_rtp_sink;
g_signal_emit (sess, gst_rtp_session_signals[SIGNAL_ON_SSRC_COLLISION], 0,
src->ssrc);
GST_RTP_SESSION_LOCK (sess);
if ((recv_rtp_sink = sess->recv_rtp_sink))
gst_object_ref (recv_rtp_sink);
GST_RTP_SESSION_UNLOCK (sess);
if (recv_rtp_sink) {
gst_pad_push_event (recv_rtp_sink, gst_event_new_reconfigure ());
gst_object_unref (recv_rtp_sink);
}
}
static void
on_ssrc_validated (RTPSession * session, RTPSource * src, GstRtpSession * sess)
{
g_signal_emit (sess, gst_rtp_session_signals[SIGNAL_ON_SSRC_VALIDATED], 0,
src->ssrc);
}
static void
on_ssrc_active (RTPSession * session, RTPSource * src, GstRtpSession * sess)
{
g_signal_emit (sess, gst_rtp_session_signals[SIGNAL_ON_SSRC_ACTIVE], 0,
src->ssrc);
}
static void
on_ssrc_sdes (RTPSession * session, RTPSource * src, GstRtpSession * sess)
{
GstStructure *s;
GstMessage *m;
/* convert the new SDES info into a message */
RTP_SESSION_LOCK (session);
g_object_get (src, "sdes", &s, NULL);
RTP_SESSION_UNLOCK (session);
m = gst_message_new_custom (GST_MESSAGE_ELEMENT, GST_OBJECT (sess), s);
gst_element_post_message (GST_ELEMENT_CAST (sess), m);
g_signal_emit (sess, gst_rtp_session_signals[SIGNAL_ON_SSRC_SDES], 0,
src->ssrc);
}
static void
on_bye_ssrc (RTPSession * session, RTPSource * src, GstRtpSession * sess)
{
g_signal_emit (sess, gst_rtp_session_signals[SIGNAL_ON_BYE_SSRC], 0,
src->ssrc);
}
static void
on_bye_timeout (RTPSession * session, RTPSource * src, GstRtpSession * sess)
{
g_signal_emit (sess, gst_rtp_session_signals[SIGNAL_ON_BYE_TIMEOUT], 0,
src->ssrc);
}
static void
on_timeout (RTPSession * session, RTPSource * src, GstRtpSession * sess)
{
g_signal_emit (sess, gst_rtp_session_signals[SIGNAL_ON_TIMEOUT], 0,
src->ssrc);
}
static void
on_sender_timeout (RTPSession * session, RTPSource * src, GstRtpSession * sess)
{
g_signal_emit (sess, gst_rtp_session_signals[SIGNAL_ON_SENDER_TIMEOUT], 0,
src->ssrc);
}
#define gst_rtp_session_parent_class parent_class
G_DEFINE_TYPE (GstRtpSession, gst_rtp_session, GST_TYPE_ELEMENT);
static void
gst_rtp_session_class_init (GstRtpSessionClass * klass)
{
GObjectClass *gobject_class;
GstElementClass *gstelement_class;
gobject_class = (GObjectClass *) klass;
gstelement_class = (GstElementClass *) klass;
g_type_class_add_private (klass, sizeof (GstRtpSessionPrivate));
gobject_class->finalize = gst_rtp_session_finalize;
gobject_class->set_property = gst_rtp_session_set_property;
gobject_class->get_property = gst_rtp_session_get_property;
/**
* GstRtpSession::request-pt-map:
* @sess: the object which received the signal
* @pt: the pt
*
* Request the payload type as #GstCaps for @pt.
*/
gst_rtp_session_signals[SIGNAL_REQUEST_PT_MAP] =
g_signal_new ("request-pt-map", G_TYPE_FROM_CLASS (klass),
G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRtpSessionClass, request_pt_map),
NULL, NULL, g_cclosure_marshal_generic, GST_TYPE_CAPS, 1, G_TYPE_UINT);
/**
* GstRtpSession::clear-pt-map:
* @sess: the object which received the signal
*
* Clear the cached pt-maps requested with #GstRtpSession::request-pt-map.
*/
gst_rtp_session_signals[SIGNAL_CLEAR_PT_MAP] =
g_signal_new ("clear-pt-map", G_TYPE_FROM_CLASS (klass),
G_SIGNAL_ACTION, G_STRUCT_OFFSET (GstRtpSessionClass, clear_pt_map),
NULL, NULL, g_cclosure_marshal_VOID__VOID, G_TYPE_NONE, 0, G_TYPE_NONE);
/**
* GstRtpSession::on-new-ssrc:
* @sess: the object which received the signal
* @ssrc: the SSRC
*
* Notify of a new SSRC that entered @session.
*/
gst_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 (GstRtpSessionClass, on_new_ssrc),
NULL, NULL, g_cclosure_marshal_VOID__UINT, G_TYPE_NONE, 1, G_TYPE_UINT);
/**
* GstRtpSession::on-ssrc_collision:
* @sess: the object which received the signal
* @ssrc: the SSRC
*
* Notify when we have an SSRC collision
*/
gst_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 (GstRtpSessionClass,
on_ssrc_collision), NULL, NULL, g_cclosure_marshal_VOID__UINT,
G_TYPE_NONE, 1, G_TYPE_UINT);
/**
* GstRtpSession::on-ssrc_validated:
* @sess: the object which received the signal
* @ssrc: the SSRC
*
* Notify of a new SSRC that became validated.
*/
gst_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 (GstRtpSessionClass,
on_ssrc_validated), NULL, NULL, g_cclosure_marshal_VOID__UINT,
G_TYPE_NONE, 1, G_TYPE_UINT);
/**
* GstRtpSession::on-ssrc_active:
* @sess: the object which received the signal
* @ssrc: the SSRC
*
* Notify of a SSRC that is active, i.e., sending RTCP.
*/
gst_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 (GstRtpSessionClass,
on_ssrc_active), NULL, NULL, g_cclosure_marshal_VOID__UINT,
G_TYPE_NONE, 1, G_TYPE_UINT);
/**
* GstRtpSession::on-ssrc-sdes:
* @session: the object which received the signal
* @src: the SSRC
*
* Notify that a new SDES was received for SSRC.
*/
gst_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 (GstRtpSessionClass, on_ssrc_sdes),
NULL, NULL, g_cclosure_marshal_VOID__UINT, G_TYPE_NONE, 1, G_TYPE_UINT);
/**
* GstRtpSession::on-bye-ssrc:
* @sess: the object which received the signal
* @ssrc: the SSRC
*
* Notify of an SSRC that became inactive because of a BYE packet.
*/
gst_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 (GstRtpSessionClass, on_bye_ssrc),
NULL, NULL, g_cclosure_marshal_VOID__UINT, G_TYPE_NONE, 1, G_TYPE_UINT);
/**
* GstRtpSession::on-bye-timeout:
* @sess: the object which received the signal
* @ssrc: the SSRC
*
* Notify of an SSRC that has timed out because of BYE
*/
gst_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 (GstRtpSessionClass, on_bye_timeout),
NULL, NULL, g_cclosure_marshal_VOID__UINT, G_TYPE_NONE, 1, G_TYPE_UINT);
/**
* GstRtpSession::on-timeout:
* @sess: the object which received the signal
* @ssrc: the SSRC
*
* Notify of an SSRC that has timed out
*/
gst_rtp_session_signals[SIGNAL_ON_TIMEOUT] =
g_signal_new ("on-timeout", G_TYPE_FROM_CLASS (klass),
G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRtpSessionClass, on_timeout),
NULL, NULL, g_cclosure_marshal_VOID__UINT, G_TYPE_NONE, 1, G_TYPE_UINT);
/**
* GstRtpSession::on-sender-timeout:
* @sess: the object which received the signal
* @ssrc: the SSRC
*
* Notify of a sender SSRC that has timed out and became a receiver
*/
gst_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 (GstRtpSessionClass,
on_sender_timeout), NULL, NULL, g_cclosure_marshal_VOID__UINT,
G_TYPE_NONE, 1, G_TYPE_UINT);
g_object_class_install_property (gobject_class, PROP_BANDWIDTH,
g_param_spec_double ("bandwidth", "Bandwidth",
"The bandwidth of the session in bytes per second (0 for auto-discover)",
0.0, G_MAXDOUBLE, DEFAULT_BANDWIDTH,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (gobject_class, PROP_RTCP_FRACTION,
g_param_spec_double ("rtcp-fraction", "RTCP Fraction",
"The RTCP bandwidth of the session in bytes per second "
"(or as a real fraction of the RTP bandwidth if < 1.0)",
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_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));
g_object_class_install_property (gobject_class, PROP_INTERNAL_SESSION,
g_param_spec_object ("internal-session", "Internal Session",
"The internal RTPSession object", RTP_TYPE_SESSION,
G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (gobject_class, PROP_USE_PIPELINE_CLOCK,
g_param_spec_boolean ("use-pipeline-clock", "Use pipeline clock",
"Use the pipeline running-time to set the NTP time in the RTCP SR messages",
DEFAULT_USE_PIPELINE_CLOCK,
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_PROBATION,
g_param_spec_uint ("probation", "Number of probations",
"Consecutive packet sequence numbers to accept the source",
0, G_MAXUINT, DEFAULT_PROBATION,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
gstelement_class->change_state =
GST_DEBUG_FUNCPTR (gst_rtp_session_change_state);
gstelement_class->request_new_pad =
GST_DEBUG_FUNCPTR (gst_rtp_session_request_new_pad);
gstelement_class->release_pad =
GST_DEBUG_FUNCPTR (gst_rtp_session_release_pad);
klass->clear_pt_map = GST_DEBUG_FUNCPTR (gst_rtp_session_clear_pt_map);
/* sink pads */
gst_element_class_add_pad_template (gstelement_class,
gst_static_pad_template_get (&rtpsession_recv_rtp_sink_template));
gst_element_class_add_pad_template (gstelement_class,
gst_static_pad_template_get (&rtpsession_recv_rtcp_sink_template));
gst_element_class_add_pad_template (gstelement_class,
gst_static_pad_template_get (&rtpsession_send_rtp_sink_template));
/* src pads */
gst_element_class_add_pad_template (gstelement_class,
gst_static_pad_template_get (&rtpsession_recv_rtp_src_template));
gst_element_class_add_pad_template (gstelement_class,
gst_static_pad_template_get (&rtpsession_sync_src_template));
gst_element_class_add_pad_template (gstelement_class,
gst_static_pad_template_get (&rtpsession_send_rtp_src_template));
gst_element_class_add_pad_template (gstelement_class,
gst_static_pad_template_get (&rtpsession_send_rtcp_src_template));
gst_element_class_set_static_metadata (gstelement_class, "RTP Session",
"Filter/Network/RTP",
"Implement an RTP session", "Wim Taymans <wim.taymans@gmail.com>");
GST_DEBUG_CATEGORY_INIT (gst_rtp_session_debug,
"rtpsession", 0, "RTP Session");
}
static void
gst_rtp_session_init (GstRtpSession * rtpsession)
{
rtpsession->priv = GST_RTP_SESSION_GET_PRIVATE (rtpsession);
g_mutex_init (&rtpsession->priv->lock);
g_cond_init (&rtpsession->priv->cond);
rtpsession->priv->sysclock = gst_system_clock_obtain ();
rtpsession->priv->session = rtp_session_new ();
rtpsession->priv->use_pipeline_clock = DEFAULT_USE_PIPELINE_CLOCK;
/* configure callbacks */
rtp_session_set_callbacks (rtpsession->priv->session, &callbacks, rtpsession);
/* configure signals */
g_signal_connect (rtpsession->priv->session, "on-new-ssrc",
(GCallback) on_new_ssrc, rtpsession);
g_signal_connect (rtpsession->priv->session, "on-ssrc-collision",
(GCallback) on_ssrc_collision, rtpsession);
g_signal_connect (rtpsession->priv->session, "on-ssrc-validated",
(GCallback) on_ssrc_validated, rtpsession);
g_signal_connect (rtpsession->priv->session, "on-ssrc-active",
(GCallback) on_ssrc_active, rtpsession);
g_signal_connect (rtpsession->priv->session, "on-ssrc-sdes",
(GCallback) on_ssrc_sdes, rtpsession);
g_signal_connect (rtpsession->priv->session, "on-bye-ssrc",
(GCallback) on_bye_ssrc, rtpsession);
g_signal_connect (rtpsession->priv->session, "on-bye-timeout",
(GCallback) on_bye_timeout, rtpsession);
g_signal_connect (rtpsession->priv->session, "on-timeout",
(GCallback) on_timeout, rtpsession);
g_signal_connect (rtpsession->priv->session, "on-sender-timeout",
(GCallback) on_sender_timeout, rtpsession);
rtpsession->priv->ptmap = g_hash_table_new_full (NULL, NULL, NULL,
(GDestroyNotify) gst_caps_unref);
gst_segment_init (&rtpsession->recv_rtp_seg, GST_FORMAT_UNDEFINED);
gst_segment_init (&rtpsession->send_rtp_seg, GST_FORMAT_UNDEFINED);
rtpsession->priv->thread_stopped = TRUE;
}
static void
gst_rtp_session_finalize (GObject * object)
{
GstRtpSession *rtpsession;
rtpsession = GST_RTP_SESSION (object);
g_hash_table_destroy (rtpsession->priv->ptmap);
g_mutex_clear (&rtpsession->priv->lock);
g_cond_clear (&rtpsession->priv->cond);
g_object_unref (rtpsession->priv->sysclock);
g_object_unref (rtpsession->priv->session);
G_OBJECT_CLASS (parent_class)->finalize (object);
}
static void
gst_rtp_session_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec)
{
GstRtpSession *rtpsession;
GstRtpSessionPrivate *priv;
rtpsession = GST_RTP_SESSION (object);
priv = rtpsession->priv;
switch (prop_id) {
case PROP_BANDWIDTH:
g_object_set_property (G_OBJECT (priv->session), "bandwidth", value);
break;
case PROP_RTCP_FRACTION:
g_object_set_property (G_OBJECT (priv->session), "rtcp-fraction", value);
break;
case PROP_RTCP_RR_BANDWIDTH:
g_object_set_property (G_OBJECT (priv->session), "rtcp-rr-bandwidth",
value);
break;
case PROP_RTCP_RS_BANDWIDTH:
g_object_set_property (G_OBJECT (priv->session), "rtcp-rs-bandwidth",
value);
break;
case PROP_SDES:
rtp_session_set_sdes_struct (priv->session, g_value_get_boxed (value));
break;
case PROP_USE_PIPELINE_CLOCK:
priv->use_pipeline_clock = g_value_get_boolean (value);
break;
case PROP_RTCP_MIN_INTERVAL:
g_object_set_property (G_OBJECT (priv->session), "rtcp-min-interval",
value);
break;
case PROP_PROBATION:
g_object_set_property (G_OBJECT (priv->session), "probation", value);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
static void
gst_rtp_session_get_property (GObject * object, guint prop_id,
GValue * value, GParamSpec * pspec)
{
GstRtpSession *rtpsession;
GstRtpSessionPrivate *priv;
rtpsession = GST_RTP_SESSION (object);
priv = rtpsession->priv;
switch (prop_id) {
case PROP_BANDWIDTH:
g_object_get_property (G_OBJECT (priv->session), "bandwidth", value);
break;
case PROP_RTCP_FRACTION:
g_object_get_property (G_OBJECT (priv->session), "rtcp-fraction", value);
break;
case PROP_RTCP_RR_BANDWIDTH:
g_object_get_property (G_OBJECT (priv->session), "rtcp-rr-bandwidth",
value);
break;
case PROP_RTCP_RS_BANDWIDTH:
g_object_get_property (G_OBJECT (priv->session), "rtcp-rs-bandwidth",
value);
break;
case PROP_SDES:
g_value_take_boxed (value, rtp_session_get_sdes_struct (priv->session));
break;
case PROP_NUM_SOURCES:
g_value_set_uint (value, rtp_session_get_num_sources (priv->session));
break;
case PROP_NUM_ACTIVE_SOURCES:
g_value_set_uint (value,
rtp_session_get_num_active_sources (priv->session));
break;
case PROP_INTERNAL_SESSION:
g_value_set_object (value, priv->session);
break;
case PROP_USE_PIPELINE_CLOCK:
g_value_set_boolean (value, priv->use_pipeline_clock);
break;
case PROP_RTCP_MIN_INTERVAL:
g_object_get_property (G_OBJECT (priv->session), "rtcp-min-interval",
value);
break;
case PROP_PROBATION:
g_object_get_property (G_OBJECT (priv->session), "probation", value);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
static void
get_current_times (GstRtpSession * rtpsession, GstClockTime * running_time,
guint64 * ntpnstime)
{
guint64 ntpns;
GstClock *clock;
GstClockTime base_time, rt, clock_time;
GST_OBJECT_LOCK (rtpsession);
if ((clock = GST_ELEMENT_CLOCK (rtpsession))) {
base_time = GST_ELEMENT_CAST (rtpsession)->base_time;
gst_object_ref (clock);
GST_OBJECT_UNLOCK (rtpsession);
/* get current clock time and convert to running time */
clock_time = gst_clock_get_time (clock);
rt = clock_time - base_time;
if (rtpsession->priv->use_pipeline_clock) {
ntpns = rt;
} else {
GTimeVal current;
/* get current NTP time */
g_get_current_time (&current);
ntpns = GST_TIMEVAL_TO_TIME (current);
}
/* add constant to convert from 1970 based time to 1900 based time */
ntpns += (2208988800LL * GST_SECOND);
gst_object_unref (clock);
} else {
GST_OBJECT_UNLOCK (rtpsession);
rt = -1;
ntpns = -1;
}
if (running_time)
*running_time = rt;
if (ntpnstime)
*ntpnstime = ntpns;
}
static void
rtcp_thread (GstRtpSession * rtpsession)
{
GstClockID id;
GstClockTime current_time;
GstClockTime next_timeout;
guint64 ntpnstime;
GstClockTime running_time;
RTPSession *session;
GstClock *sysclock;
GST_DEBUG_OBJECT (rtpsession, "entering RTCP thread");
GST_RTP_SESSION_LOCK (rtpsession);
while (rtpsession->priv->wait_send) {
GST_LOG_OBJECT (rtpsession, "waiting for RTP thread");
GST_RTP_SESSION_WAIT (rtpsession);
GST_LOG_OBJECT (rtpsession, "signaled...");
}
sysclock = rtpsession->priv->sysclock;
current_time = gst_clock_get_time (sysclock);
session = rtpsession->priv->session;
GST_DEBUG_OBJECT (rtpsession, "starting at %" GST_TIME_FORMAT,
GST_TIME_ARGS (current_time));
session->start_time = current_time;
while (!rtpsession->priv->stop_thread) {
GstClockReturn res;
/* get initial estimate */
next_timeout = rtp_session_next_timeout (session, current_time);
GST_DEBUG_OBJECT (rtpsession, "next check time %" GST_TIME_FORMAT,
GST_TIME_ARGS (next_timeout));
/* leave if no more timeouts, the session ended */
if (next_timeout == GST_CLOCK_TIME_NONE)
break;
id = rtpsession->priv->id =
gst_clock_new_single_shot_id (sysclock, next_timeout);
GST_RTP_SESSION_UNLOCK (rtpsession);
res = gst_clock_id_wait (id, NULL);
GST_RTP_SESSION_LOCK (rtpsession);
gst_clock_id_unref (id);
rtpsession->priv->id = NULL;
if (rtpsession->priv->stop_thread)
break;
/* update current time */
current_time = gst_clock_get_time (sysclock);
/* get current NTP time */
get_current_times (rtpsession, &running_time, &ntpnstime);
/* we get unlocked because we need to perform reconsideration, don't perform
* the timeout but get a new reporting estimate. */
GST_DEBUG_OBJECT (rtpsession, "unlocked %d, current %" GST_TIME_FORMAT,
res, GST_TIME_ARGS (current_time));
/* perform actions, we ignore result. Release lock because it might push. */
GST_RTP_SESSION_UNLOCK (rtpsession);
rtp_session_on_timeout (session, current_time, ntpnstime, running_time);
GST_RTP_SESSION_LOCK (rtpsession);
}
/* mark the thread as stopped now */
rtpsession->priv->thread_stopped = TRUE;
GST_RTP_SESSION_UNLOCK (rtpsession);
GST_DEBUG_OBJECT (rtpsession, "leaving RTCP thread");
}
static gboolean
start_rtcp_thread (GstRtpSession * rtpsession)
{
GError *error = NULL;
gboolean res;
GST_DEBUG_OBJECT (rtpsession, "starting RTCP thread");
GST_RTP_SESSION_LOCK (rtpsession);
rtpsession->priv->stop_thread = FALSE;
if (rtpsession->priv->thread_stopped) {
/* if the thread stopped, and we still have a handle to the thread, join it
* now. We can safely join with the lock held, the thread will not take it
* anymore. */
if (rtpsession->priv->thread)
g_thread_join (rtpsession->priv->thread);
/* only create a new thread if the old one was stopped. Otherwise we can
* just reuse the currently running one. */
rtpsession->priv->thread = g_thread_try_new ("rtpsession-rtcp-thread",
(GThreadFunc) rtcp_thread, rtpsession, &error);
rtpsession->priv->thread_stopped = FALSE;
}
GST_RTP_SESSION_UNLOCK (rtpsession);
if (error != NULL) {
res = FALSE;
GST_DEBUG_OBJECT (rtpsession, "failed to start thread, %s", error->message);
g_error_free (error);
} else {
res = TRUE;
}
return res;
}
static void
stop_rtcp_thread (GstRtpSession * rtpsession)
{
GST_DEBUG_OBJECT (rtpsession, "stopping RTCP thread");
GST_RTP_SESSION_LOCK (rtpsession);
rtpsession->priv->stop_thread = TRUE;
rtpsession->priv->wait_send = FALSE;
GST_RTP_SESSION_SIGNAL (rtpsession);
if (rtpsession->priv->id)
gst_clock_id_unschedule (rtpsession->priv->id);
GST_RTP_SESSION_UNLOCK (rtpsession);
}
static void
join_rtcp_thread (GstRtpSession * rtpsession)
{
GST_RTP_SESSION_LOCK (rtpsession);
/* don't try to join when we have no thread */
if (rtpsession->priv->thread != NULL) {
GST_DEBUG_OBJECT (rtpsession, "joining RTCP thread");
GST_RTP_SESSION_UNLOCK (rtpsession);
g_thread_join (rtpsession->priv->thread);
GST_RTP_SESSION_LOCK (rtpsession);
/* after the join, take the lock and clear the thread structure. The caller
* is supposed to not concurrently call start and join. */
rtpsession->priv->thread = NULL;
}
GST_RTP_SESSION_UNLOCK (rtpsession);
}
static GstStateChangeReturn
gst_rtp_session_change_state (GstElement * element, GstStateChange transition)
{
GstStateChangeReturn res;
GstRtpSession *rtpsession;
rtpsession = GST_RTP_SESSION (element);
switch (transition) {
case GST_STATE_CHANGE_NULL_TO_READY:
break;
case GST_STATE_CHANGE_READY_TO_PAUSED:
GST_RTP_SESSION_LOCK (rtpsession);
if (rtpsession->send_rtp_src)
rtpsession->priv->wait_send = TRUE;
GST_RTP_SESSION_UNLOCK (rtpsession);
break;
case GST_STATE_CHANGE_PAUSED_TO_PLAYING:
break;
case GST_STATE_CHANGE_PLAYING_TO_PAUSED:
case GST_STATE_CHANGE_PAUSED_TO_READY:
/* no need to join yet, we might want to continue later. Also, the
* dataflow could block downstream so that a join could just block
* forever. */
stop_rtcp_thread (rtpsession);
break;
default:
break;
}
res = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
switch (transition) {
case GST_STATE_CHANGE_PAUSED_TO_PLAYING:
if (!start_rtcp_thread (rtpsession))
goto failed_thread;
break;
case GST_STATE_CHANGE_PLAYING_TO_PAUSED:
break;
case GST_STATE_CHANGE_PAUSED_TO_READY:
/* downstream is now releasing the dataflow and we can join. */
join_rtcp_thread (rtpsession);
break;
case GST_STATE_CHANGE_READY_TO_NULL:
break;
default:
break;
}
return res;
/* ERRORS */
failed_thread:
{
return GST_STATE_CHANGE_FAILURE;
}
}
static gboolean
return_true (gpointer key, gpointer value, gpointer user_data)
{
return TRUE;
}
static void
gst_rtp_session_clear_pt_map (GstRtpSession * rtpsession)
{
g_hash_table_foreach_remove (rtpsession->priv->ptmap, return_true, NULL);
}
/* called when the session manager has an RTP packet or a list of packets
* ready for further processing */
static GstFlowReturn
gst_rtp_session_process_rtp (RTPSession * sess, RTPSource * src,
GstBuffer * buffer, gpointer user_data)
{
GstFlowReturn result;
GstRtpSession *rtpsession;
GstPad *rtp_src;
rtpsession = GST_RTP_SESSION (user_data);
GST_RTP_SESSION_LOCK (rtpsession);
if ((rtp_src = rtpsession->recv_rtp_src))
gst_object_ref (rtp_src);
GST_RTP_SESSION_UNLOCK (rtpsession);
if (rtp_src) {
GST_LOG_OBJECT (rtpsession, "pushing received RTP packet");
result = gst_pad_push (rtp_src, buffer);
gst_object_unref (rtp_src);
} else {
GST_DEBUG_OBJECT (rtpsession, "dropping received RTP packet");
gst_buffer_unref (buffer);
result = GST_FLOW_OK;
}
return result;
}
/* called when the session manager has an RTP packet ready for further
* sending */
static GstFlowReturn
gst_rtp_session_send_rtp (RTPSession * sess, RTPSource * src,
gpointer data, gpointer user_data)
{
GstFlowReturn result;
GstRtpSession *rtpsession;
GstPad *rtp_src;
rtpsession = GST_RTP_SESSION (user_data);
GST_RTP_SESSION_LOCK (rtpsession);
if ((rtp_src = rtpsession->send_rtp_src))
gst_object_ref (rtp_src);
if (rtpsession->priv->wait_send) {
GST_LOG_OBJECT (rtpsession, "signal RTCP thread");
rtpsession->priv->wait_send = FALSE;
GST_RTP_SESSION_SIGNAL (rtpsession);
}
GST_RTP_SESSION_UNLOCK (rtpsession);
if (rtp_src) {
if (GST_IS_BUFFER (data)) {
GST_LOG_OBJECT (rtpsession, "sending RTP packet");
result = gst_pad_push (rtp_src, GST_BUFFER_CAST (data));
} else {
GST_LOG_OBJECT (rtpsession, "sending RTP list");
result = gst_pad_push_list (rtp_src, GST_BUFFER_LIST_CAST (data));
}
gst_object_unref (rtp_src);
} else {
gst_mini_object_unref (GST_MINI_OBJECT_CAST (data));
result = GST_FLOW_OK;
}
return result;
}
static void
do_rtcp_events (GstRtpSession * rtpsession, GstPad * srcpad)
{
GstCaps *caps;
GstSegment seg;
GstEvent *event;
gchar *stream_id;
gboolean have_group_id;
guint group_id;
stream_id =
g_strdup_printf ("%08x%08x%08x%08x", g_random_int (), g_random_int (),
g_random_int (), g_random_int ());
GST_RTP_SESSION_LOCK (rtpsession);
if (rtpsession->recv_rtp_sink) {
event =
gst_pad_get_sticky_event (rtpsession->recv_rtp_sink,
GST_EVENT_STREAM_START, 0);
if (event) {
if (gst_event_parse_group_id (event, &group_id))
have_group_id = TRUE;
else
have_group_id = FALSE;
gst_event_unref (event);
} else {
have_group_id = TRUE;
group_id = gst_util_group_id_next ();
}
} else {
have_group_id = TRUE;
group_id = gst_util_group_id_next ();
}
GST_RTP_SESSION_UNLOCK (rtpsession);
event = gst_event_new_stream_start (stream_id);
if (have_group_id)
gst_event_set_group_id (event, group_id);
gst_pad_push_event (srcpad, event);
g_free (stream_id);
caps = gst_caps_new_empty_simple ("application/x-rtcp");
gst_pad_set_caps (srcpad, caps);
gst_caps_unref (caps);
gst_segment_init (&seg, GST_FORMAT_TIME);
event = gst_event_new_segment (&seg);
gst_pad_push_event (srcpad, event);
}
/* called when the session manager has an RTCP packet ready for further
* sending. The eos flag is set when an EOS event should be sent downstream as
* well. */
static GstFlowReturn
gst_rtp_session_send_rtcp (RTPSession * sess, RTPSource * src,
GstBuffer * buffer, gboolean eos, gpointer user_data)
{
GstFlowReturn result;
GstRtpSession *rtpsession;
GstPad *rtcp_src;
rtpsession = GST_RTP_SESSION (user_data);
GST_RTP_SESSION_LOCK (rtpsession);
if (rtpsession->priv->stop_thread)
goto stopping;
if ((rtcp_src = rtpsession->send_rtcp_src)) {
gst_object_ref (rtcp_src);
GST_RTP_SESSION_UNLOCK (rtpsession);
/* set rtcp caps on output pad */
if (!gst_pad_has_current_caps (rtcp_src))
do_rtcp_events (rtpsession, rtcp_src);
GST_LOG_OBJECT (rtpsession, "sending RTCP");
result = gst_pad_push (rtcp_src, buffer);
/* we have to send EOS after this packet */
if (eos) {
GST_LOG_OBJECT (rtpsession, "sending EOS");
gst_pad_push_event (rtcp_src, gst_event_new_eos ());
}
gst_object_unref (rtcp_src);
} else {
GST_RTP_SESSION_UNLOCK (rtpsession);
GST_DEBUG_OBJECT (rtpsession, "not sending RTCP, no output pad");
gst_buffer_unref (buffer);
result = GST_FLOW_OK;
}
return result;
/* ERRORS */
stopping:
{
GST_DEBUG_OBJECT (rtpsession, "we are stopping");
gst_buffer_unref (buffer);
GST_RTP_SESSION_UNLOCK (rtpsession);
return GST_FLOW_OK;
}
}
/* called when the session manager has an SR RTCP packet ready for handling
* inter stream synchronisation */
static GstFlowReturn
gst_rtp_session_sync_rtcp (RTPSession * sess,
GstBuffer * buffer, gpointer user_data)
{
GstFlowReturn result;
GstRtpSession *rtpsession;
GstPad *sync_src;
rtpsession = GST_RTP_SESSION (user_data);
GST_RTP_SESSION_LOCK (rtpsession);
if (rtpsession->priv->stop_thread)
goto stopping;
if ((sync_src = rtpsession->sync_src)) {
gst_object_ref (sync_src);
GST_RTP_SESSION_UNLOCK (rtpsession);
GST_LOG_OBJECT (rtpsession, "sending Sync RTCP");
result = gst_pad_push (sync_src, buffer);
gst_object_unref (sync_src);
} else {
GST_RTP_SESSION_UNLOCK (rtpsession);
GST_DEBUG_OBJECT (rtpsession, "not sending Sync RTCP, no output pad");
gst_buffer_unref (buffer);
result = GST_FLOW_OK;
}
return result;
/* ERRORS */
stopping:
{
GST_DEBUG_OBJECT (rtpsession, "we are stopping");
gst_buffer_unref (buffer);
GST_RTP_SESSION_UNLOCK (rtpsession);
return GST_FLOW_OK;
}
}
static void
gst_rtp_session_cache_caps (GstRtpSession * rtpsession, GstCaps * caps)
{
GstRtpSessionPrivate *priv;
const GstStructure *s;
gint payload;
priv = rtpsession->priv;
GST_DEBUG_OBJECT (rtpsession, "parsing caps");
s = gst_caps_get_structure (caps, 0);
if (!gst_structure_get_int (s, "payload", &payload))
return;
if (g_hash_table_lookup (priv->ptmap, GINT_TO_POINTER (payload)))
return;
g_hash_table_insert (priv->ptmap, GINT_TO_POINTER (payload),
gst_caps_ref (caps));
}
static GstCaps *
gst_rtp_session_get_caps_for_pt (GstRtpSession * rtpsession, guint payload)
{
GstCaps *caps = NULL;
GValue args[2] = { {0}, {0} };
GValue ret = { 0 };
GST_RTP_SESSION_LOCK (rtpsession);
caps = g_hash_table_lookup (rtpsession->priv->ptmap,
GINT_TO_POINTER (payload));
if (caps) {
gst_caps_ref (caps);
goto done;
}
/* not found in the cache, try to get it with a signal */
g_value_init (&args[0], GST_TYPE_ELEMENT);
g_value_set_object (&args[0], rtpsession);
g_value_init (&args[1], G_TYPE_UINT);
g_value_set_uint (&args[1], payload);
g_value_init (&ret, GST_TYPE_CAPS);
g_value_set_boxed (&ret, NULL);
GST_RTP_SESSION_UNLOCK (rtpsession);
g_signal_emitv (args, gst_rtp_session_signals[SIGNAL_REQUEST_PT_MAP], 0,
&ret);
GST_RTP_SESSION_LOCK (rtpsession);
g_value_unset (&args[0]);
g_value_unset (&args[1]);
caps = (GstCaps *) g_value_dup_boxed (&ret);
g_value_unset (&ret);
if (!caps)
goto no_caps;
gst_rtp_session_cache_caps (rtpsession, caps);
done:
GST_RTP_SESSION_UNLOCK (rtpsession);
return caps;
no_caps:
{
GST_DEBUG_OBJECT (rtpsession, "could not get caps");
goto done;
}
}
/* called when the session manager needs the clock rate */
static gint
gst_rtp_session_clock_rate (RTPSession * sess, guint8 payload,
gpointer user_data)
{
gint result = -1;
GstRtpSession *rtpsession;
GstCaps *caps;
const GstStructure *s;
rtpsession = GST_RTP_SESSION_CAST (user_data);
caps = gst_rtp_session_get_caps_for_pt (rtpsession, payload);
if (!caps)
goto done;
s = gst_caps_get_structure (caps, 0);
if (!gst_structure_get_int (s, "clock-rate", &result))
goto no_clock_rate;
gst_caps_unref (caps);
GST_DEBUG_OBJECT (rtpsession, "parsed clock-rate %d", result);
done:
return result;
/* ERRORS */
no_clock_rate:
{
gst_caps_unref (caps);
GST_DEBUG_OBJECT (rtpsession, "No clock-rate in caps!");
goto done;
}
}
/* called when the session manager asks us to reconsider the timeout */
static void
gst_rtp_session_reconsider (RTPSession * sess, gpointer user_data)
{
GstRtpSession *rtpsession;
rtpsession = GST_RTP_SESSION_CAST (user_data);
GST_RTP_SESSION_LOCK (rtpsession);
GST_DEBUG_OBJECT (rtpsession, "unlock timer for reconsideration");
if (rtpsession->priv->id)
gst_clock_id_unschedule (rtpsession->priv->id);
GST_RTP_SESSION_UNLOCK (rtpsession);
}
static gboolean
gst_rtp_session_event_recv_rtp_sink (GstPad * pad, GstObject * parent,
GstEvent * event)
{
GstRtpSession *rtpsession;
gboolean ret = FALSE;
rtpsession = GST_RTP_SESSION (parent);
GST_DEBUG_OBJECT (rtpsession, "received event %s",
GST_EVENT_TYPE_NAME (event));
switch (GST_EVENT_TYPE (event)) {
case GST_EVENT_CAPS:
{
GstCaps *caps;
/* process */
gst_event_parse_caps (event, &caps);
gst_rtp_session_sink_setcaps (pad, rtpsession, caps);
ret = gst_pad_push_event (rtpsession->recv_rtp_src, event);
break;
}
case GST_EVENT_FLUSH_STOP:
gst_segment_init (&rtpsession->recv_rtp_seg, GST_FORMAT_UNDEFINED);
ret = gst_pad_push_event (rtpsession->recv_rtp_src, event);
break;
case GST_EVENT_SEGMENT:
{
GstSegment *segment, in_segment;
segment = &rtpsession->recv_rtp_seg;
/* the newsegment event is needed to convert the RTP timestamp to
* running_time, which is needed to generate a mapping from RTP to NTP
* timestamps in SR reports */
gst_event_copy_segment (event, &in_segment);
GST_DEBUG_OBJECT (rtpsession, "received segment %" GST_SEGMENT_FORMAT,
&in_segment);
/* accept upstream */
gst_segment_copy_into (&in_segment, segment);
/* push event forward */
ret = gst_pad_push_event (rtpsession->recv_rtp_src, event);
break;
}
default:
ret = gst_pad_push_event (rtpsession->recv_rtp_src, event);
break;
}
return ret;
}
static gboolean
gst_rtp_session_request_remote_key_unit (GstRtpSession * rtpsession,
guint32 ssrc, guint payload, gboolean all_headers, gint count)
{
GstCaps *caps;
caps = gst_rtp_session_get_caps_for_pt (rtpsession, payload);
if (caps) {
const GstStructure *s = gst_caps_get_structure (caps, 0);
gboolean pli;
gboolean fir;
pli = gst_structure_has_field (s, "rtcp-fb-nack-pli");
fir = gst_structure_has_field (s, "rtcp-fb-ccm-fir") && all_headers;
/* Google Talk uses FIR for repair, so send it even if we just want a
* regular PLI */
if (!pli &&
gst_structure_has_field (s, "rtcp-fb-x-gstreamer-fir-as-repair"))
fir = TRUE;
gst_caps_unref (caps);
if (pli || fir)
return rtp_session_request_key_unit (rtpsession->priv->session, ssrc,
fir, count);
}
return FALSE;
}
static gboolean
gst_rtp_session_event_recv_rtp_src (GstPad * pad, GstObject * parent,
GstEvent * event)
{
GstRtpSession *rtpsession;
gboolean forward = TRUE;
gboolean ret = TRUE;
const GstStructure *s;
guint32 ssrc;
guint pt;
rtpsession = GST_RTP_SESSION (parent);
switch (GST_EVENT_TYPE (event)) {
case GST_EVENT_CUSTOM_UPSTREAM:
s = gst_event_get_structure (event);
if (gst_structure_has_name (s, "GstForceKeyUnit") &&
gst_structure_get_uint (s, "ssrc", &ssrc) &&
gst_structure_get_uint (s, "payload", &pt)) {
gboolean all_headers = FALSE;
gint count = -1;
gst_structure_get_boolean (s, "all-headers", &all_headers);
if (gst_structure_get_int (s, "count", &count) && count < 0)
count += G_MAXINT; /* Make sure count is positive if present */
if (gst_rtp_session_request_remote_key_unit (rtpsession, ssrc, pt,
all_headers, count))
forward = FALSE;
} else if (gst_structure_has_name (s, "GstRTPRetransmissionRequest")) {
GstClockTime running_time;
guint seqnum, delay, deadline, max_delay;
if (!gst_structure_get_clock_time (s, "running-time", &running_time))
running_time = -1;
if (!gst_structure_get_uint (s, "ssrc", &ssrc))
ssrc = -1;
if (!gst_structure_get_uint (s, "seqnum", &seqnum))
seqnum = -1;
if (!gst_structure_get_uint (s, "delay", &delay))
delay = 0;
if (!gst_structure_get_uint (s, "deadline", &deadline))
deadline = 100;
/* remaining time to receive the packet */
max_delay = deadline;
if (max_delay > delay)
max_delay -= delay;
/* estimated RTT */
if (max_delay > 40)
max_delay -= 40;
else
max_delay = 0;
if (rtp_session_request_nack (rtpsession->priv->session, ssrc, seqnum,
max_delay * GST_MSECOND))
forward = FALSE;
}
break;
default:
break;
}
if (forward) {
GstPad *recv_rtp_sink;
GST_RTP_SESSION_LOCK (rtpsession);
if ((recv_rtp_sink = rtpsession->recv_rtp_sink))
gst_object_ref (recv_rtp_sink);
GST_RTP_SESSION_UNLOCK (rtpsession);
if (recv_rtp_sink) {
ret = gst_pad_push_event (recv_rtp_sink, event);
gst_object_unref (recv_rtp_sink);
} else
gst_event_unref (event);
} else {
gst_event_unref (event);
}
return ret;
}
static GstIterator *
gst_rtp_session_iterate_internal_links (GstPad * pad, GstObject * parent)
{
GstRtpSession *rtpsession;
GstPad *otherpad = NULL;
GstIterator *it = NULL;
rtpsession = GST_RTP_SESSION (parent);
GST_RTP_SESSION_LOCK (rtpsession);
if (pad == rtpsession->recv_rtp_src) {
otherpad = gst_object_ref (rtpsession->recv_rtp_sink);
} else if (pad == rtpsession->recv_rtp_sink) {
otherpad = gst_object_ref (rtpsession->recv_rtp_src);
} else if (pad == rtpsession->send_rtp_src) {
otherpad = gst_object_ref (rtpsession->send_rtp_sink);
} else if (pad == rtpsession->send_rtp_sink) {
otherpad = gst_object_ref (rtpsession->send_rtp_src);
}
GST_RTP_SESSION_UNLOCK (rtpsession);
if (otherpad) {
GValue val = { 0, };
g_value_init (&val, GST_TYPE_PAD);
g_value_set_object (&val, otherpad);
it = gst_iterator_new_single (GST_TYPE_PAD, &val);
g_value_unset (&val);
gst_object_unref (otherpad);
}
return it;
}
static gboolean
gst_rtp_session_sink_setcaps (GstPad * pad, GstRtpSession * rtpsession,
GstCaps * caps)
{
GST_RTP_SESSION_LOCK (rtpsession);
gst_rtp_session_cache_caps (rtpsession, caps);
GST_RTP_SESSION_UNLOCK (rtpsession);
return TRUE;
}
/* receive a packet from a sender, send it to the RTP session manager and
* forward the packet on the rtp_src pad
*/
static GstFlowReturn
gst_rtp_session_chain_recv_rtp (GstPad * pad, GstObject * parent,
GstBuffer * buffer)
{
GstRtpSession *rtpsession;
GstRtpSessionPrivate *priv;
GstFlowReturn ret;
GstClockTime current_time, running_time;
GstClockTime timestamp;
guint64 ntpnstime;
rtpsession = GST_RTP_SESSION (parent);
priv = rtpsession->priv;
GST_LOG_OBJECT (rtpsession, "received RTP packet");
/* get NTP time when this packet was captured, this depends on the timestamp. */
timestamp = GST_BUFFER_TIMESTAMP (buffer);
if (GST_CLOCK_TIME_IS_VALID (timestamp)) {
/* convert to running time using the segment values */
running_time =
gst_segment_to_running_time (&rtpsession->recv_rtp_seg, GST_FORMAT_TIME,
timestamp);
ntpnstime = GST_CLOCK_TIME_NONE;
} else {
get_current_times (rtpsession, &running_time, &ntpnstime);
}
current_time = gst_clock_get_time (priv->sysclock);
ret = rtp_session_process_rtp (priv->session, buffer, current_time,
running_time, ntpnstime);
if (ret != GST_FLOW_OK)
goto push_error;
done:
return ret;
/* ERRORS */
push_error:
{
GST_DEBUG_OBJECT (rtpsession, "process returned %s",
gst_flow_get_name (ret));
goto done;
}
}
static gboolean
gst_rtp_session_event_recv_rtcp_sink (GstPad * pad, GstObject * parent,
GstEvent * event)
{
GstRtpSession *rtpsession;
gboolean ret = FALSE;
rtpsession = GST_RTP_SESSION (parent);
GST_DEBUG_OBJECT (rtpsession, "received event %s",
GST_EVENT_TYPE_NAME (event));
switch (GST_EVENT_TYPE (event)) {
default:
ret = gst_pad_push_event (rtpsession->sync_src, event);
break;
}
return ret;
}
/* Receive an RTCP packet from a sender, send it to the RTP session manager and
* forward the SR packets to the sync_src pad.
*/
static GstFlowReturn
gst_rtp_session_chain_recv_rtcp (GstPad * pad, GstObject * parent,
GstBuffer * buffer)
{
GstRtpSession *rtpsession;
GstRtpSessionPrivate *priv;
GstClockTime current_time;
guint64 ntpnstime;
rtpsession = GST_RTP_SESSION (parent);
priv = rtpsession->priv;
GST_LOG_OBJECT (rtpsession, "received RTCP packet");
current_time = gst_clock_get_time (priv->sysclock);
get_current_times (rtpsession, NULL, &ntpnstime);
rtp_session_process_rtcp (priv->session, buffer, current_time, ntpnstime);
return GST_FLOW_OK; /* always return OK */
}
static gboolean
gst_rtp_session_query_send_rtcp_src (GstPad * pad, GstObject * parent,
GstQuery * query)
{
GstRtpSession *rtpsession;
gboolean ret = FALSE;
rtpsession = GST_RTP_SESSION (parent);
GST_DEBUG_OBJECT (rtpsession, "received QUERY %s",
GST_QUERY_TYPE_NAME (query));
switch (GST_QUERY_TYPE (query)) {
case GST_QUERY_LATENCY:
ret = TRUE;
/* use the defaults for the latency query. */
gst_query_set_latency (query, FALSE, 0, -1);
break;
default:
/* other queries simply fail for now */
break;
}
return ret;
}
static gboolean
gst_rtp_session_event_send_rtcp_src (GstPad * pad, GstObject * parent,
GstEvent * event)
{
GstRtpSession *rtpsession;
gboolean ret = TRUE;
rtpsession = GST_RTP_SESSION (parent);
GST_DEBUG_OBJECT (rtpsession, "received EVENT %s",
GST_EVENT_TYPE_NAME (event));
switch (GST_EVENT_TYPE (event)) {
case GST_EVENT_SEEK:
case GST_EVENT_LATENCY:
gst_event_unref (event);
ret = TRUE;
break;
default:
/* other events simply fail for now */
gst_event_unref (event);
ret = FALSE;
break;
}
return ret;
}
static gboolean
gst_rtp_session_event_send_rtp_sink (GstPad * pad, GstObject * parent,
GstEvent * event)
{
GstRtpSession *rtpsession;
gboolean ret = FALSE;
rtpsession = GST_RTP_SESSION (parent);
GST_DEBUG_OBJECT (rtpsession, "received EVENT %s",
GST_EVENT_TYPE_NAME (event));
switch (GST_EVENT_TYPE (event)) {
case GST_EVENT_CAPS:
{
GstCaps *caps;
/* process */
gst_event_parse_caps (event, &caps);
gst_rtp_session_setcaps_send_rtp (pad, rtpsession, caps);
ret = gst_pad_push_event (rtpsession->send_rtp_src, event);
break;
}
case GST_EVENT_FLUSH_STOP:
gst_segment_init (&rtpsession->send_rtp_seg, GST_FORMAT_UNDEFINED);
ret = gst_pad_push_event (rtpsession->send_rtp_src, event);
break;
case GST_EVENT_SEGMENT:{
GstSegment *segment, in_segment;
segment = &rtpsession->send_rtp_seg;
/* the newsegment event is needed to convert the RTP timestamp to
* running_time, which is needed to generate a mapping from RTP to NTP
* timestamps in SR reports */
gst_event_copy_segment (event, &in_segment);
GST_DEBUG_OBJECT (rtpsession, "received segment %" GST_SEGMENT_FORMAT,
&in_segment);
/* accept upstream */
gst_segment_copy_into (&in_segment, segment);
/* push event forward */
ret = gst_pad_push_event (rtpsession->send_rtp_src, event);
break;
}
case GST_EVENT_EOS:{
GstClockTime current_time;
/* push downstream FIXME, we are not supposed to leave the session just
* because we stop sending. */
ret = gst_pad_push_event (rtpsession->send_rtp_src, event);
current_time = gst_clock_get_time (rtpsession->priv->sysclock);
GST_DEBUG_OBJECT (rtpsession, "scheduling BYE message");
rtp_session_mark_all_bye (rtpsession->priv->session, "End Of Stream");
rtp_session_schedule_bye (rtpsession->priv->session, current_time);
break;
}
default:{
GstPad *send_rtp_src;
GST_RTP_SESSION_LOCK (rtpsession);
if ((send_rtp_src = rtpsession->send_rtp_src))
gst_object_ref (send_rtp_src);
GST_RTP_SESSION_UNLOCK (rtpsession);
if (send_rtp_src) {
ret = gst_pad_push_event (send_rtp_src, event);
gst_object_unref (send_rtp_src);
} else
gst_event_unref (event);
break;
}
}
return ret;
}
static gboolean
gst_rtp_session_event_send_rtp_src (GstPad * pad, GstObject * parent,
GstEvent * event)
{
GstRtpSession *rtpsession;
gboolean ret = FALSE;
rtpsession = GST_RTP_SESSION (parent);
GST_DEBUG_OBJECT (rtpsession, "received EVENT %s",
GST_EVENT_TYPE_NAME (event));
switch (GST_EVENT_TYPE (event)) {
case GST_EVENT_LATENCY:
/* save the latency, we need this to know when an RTP packet will be
* rendered by the sink */
gst_event_parse_latency (event, &rtpsession->priv->send_latency);
ret = gst_pad_event_default (pad, parent, event);
break;
default:
ret = gst_pad_event_default (pad, parent, event);
break;
}
return ret;
}
static GstCaps *
gst_rtp_session_getcaps_send_rtp (GstPad * pad, GstRtpSession * rtpsession,
GstCaps * filter)
{
GstRtpSessionPrivate *priv;
GstCaps *result;
GstStructure *s1, *s2;
guint ssrc;
priv = rtpsession->priv;
ssrc = rtp_session_suggest_ssrc (priv->session);
/* we can basically accept anything but we prefer to receive packets with our
* internal SSRC so that we don't have to patch it. Create a structure with
* the SSRC and another one without. */
s1 = gst_structure_new ("application/x-rtp", "ssrc", G_TYPE_UINT, ssrc, NULL);
s2 = gst_structure_new_empty ("application/x-rtp");
result = gst_caps_new_full (s1, s2, NULL);
if (filter) {
GstCaps *caps = result;
result = gst_caps_intersect_full (filter, caps, GST_CAPS_INTERSECT_FIRST);
gst_caps_unref (caps);
}
GST_DEBUG_OBJECT (rtpsession, "getting caps %" GST_PTR_FORMAT, result);
return result;
}
static gboolean
gst_rtp_session_query_send_rtp (GstPad * pad, GstObject * parent,
GstQuery * query)
{
gboolean res = FALSE;
GstRtpSession *rtpsession;
rtpsession = GST_RTP_SESSION (parent);
switch (GST_QUERY_TYPE (query)) {
case GST_QUERY_CAPS:
{
GstCaps *filter, *caps;
gst_query_parse_caps (query, &filter);
caps = gst_rtp_session_getcaps_send_rtp (pad, rtpsession, filter);
gst_query_set_caps_result (query, caps);
gst_caps_unref (caps);
res = TRUE;
break;
}
default:
res = gst_pad_query_default (pad, parent, query);
break;
}
return res;
}
static gboolean
gst_rtp_session_setcaps_send_rtp (GstPad * pad, GstRtpSession * rtpsession,
GstCaps * caps)
{
GstRtpSessionPrivate *priv;
priv = rtpsession->priv;
rtp_session_update_send_caps (priv->session, caps);
return TRUE;
}
/* Recieve an RTP packet or a list of packets to be send to the receivers,
* send to RTP session manager and forward to send_rtp_src.
*/
static GstFlowReturn
gst_rtp_session_chain_send_rtp_common (GstRtpSession * rtpsession,
gpointer data, gboolean is_list)
{
GstRtpSessionPrivate *priv;
GstFlowReturn ret;
GstClockTime timestamp, running_time;
GstClockTime current_time;
priv = rtpsession->priv;
GST_LOG_OBJECT (rtpsession, "received RTP %s", is_list ? "list" : "packet");
/* get NTP time when this packet was captured, this depends on the timestamp. */
if (is_list) {
GstBuffer *buffer = NULL;
/* All groups in an list have the same timestamp.
* So, just take it from the first group. */
buffer = gst_buffer_list_get (GST_BUFFER_LIST_CAST (data), 0);
if (buffer)
timestamp = GST_BUFFER_TIMESTAMP (buffer);
else
timestamp = -1;
} else {
timestamp = GST_BUFFER_TIMESTAMP (GST_BUFFER_CAST (data));
}
if (GST_CLOCK_TIME_IS_VALID (timestamp)) {
/* convert to running time using the segment start value. */
running_time =
gst_segment_to_running_time (&rtpsession->send_rtp_seg, GST_FORMAT_TIME,
timestamp);
running_time += priv->send_latency;
} else {
/* no timestamp. */
running_time = -1;
}
current_time = gst_clock_get_time (priv->sysclock);
ret = rtp_session_send_rtp (priv->session, data, is_list, current_time,
running_time);
if (ret != GST_FLOW_OK)
goto push_error;
done:
return ret;
/* ERRORS */
push_error:
{
GST_DEBUG_OBJECT (rtpsession, "process returned %s",
gst_flow_get_name (ret));
goto done;
}
}
static GstFlowReturn
gst_rtp_session_chain_send_rtp (GstPad * pad, GstObject * parent,
GstBuffer * buffer)
{
GstRtpSession *rtpsession = GST_RTP_SESSION (parent);
return gst_rtp_session_chain_send_rtp_common (rtpsession, buffer, FALSE);
}
static GstFlowReturn
gst_rtp_session_chain_send_rtp_list (GstPad * pad, GstObject * parent,
GstBufferList * list)
{
GstRtpSession *rtpsession = GST_RTP_SESSION (parent);
return gst_rtp_session_chain_send_rtp_common (rtpsession, list, TRUE);
}
/* Create sinkpad to receive RTP packets from senders. This will also create a
* srcpad for the RTP packets.
*/
static GstPad *
create_recv_rtp_sink (GstRtpSession * rtpsession)
{
GST_DEBUG_OBJECT (rtpsession, "creating RTP sink pad");
rtpsession->recv_rtp_sink =
gst_pad_new_from_static_template (&rtpsession_recv_rtp_sink_template,
"recv_rtp_sink");
gst_pad_set_chain_function (rtpsession->recv_rtp_sink,
gst_rtp_session_chain_recv_rtp);
gst_pad_set_event_function (rtpsession->recv_rtp_sink,
gst_rtp_session_event_recv_rtp_sink);
gst_pad_set_iterate_internal_links_function (rtpsession->recv_rtp_sink,
gst_rtp_session_iterate_internal_links);
gst_pad_set_active (rtpsession->recv_rtp_sink, TRUE);
gst_element_add_pad (GST_ELEMENT_CAST (rtpsession),
rtpsession->recv_rtp_sink);
GST_DEBUG_OBJECT (rtpsession, "creating RTP src pad");
rtpsession->recv_rtp_src =
gst_pad_new_from_static_template (&rtpsession_recv_rtp_src_template,
"recv_rtp_src");
gst_pad_set_event_function (rtpsession->recv_rtp_src,
gst_rtp_session_event_recv_rtp_src);
gst_pad_set_iterate_internal_links_function (rtpsession->recv_rtp_src,
gst_rtp_session_iterate_internal_links);
gst_pad_use_fixed_caps (rtpsession->recv_rtp_src);
gst_pad_set_active (rtpsession->recv_rtp_src, TRUE);
gst_element_add_pad (GST_ELEMENT_CAST (rtpsession), rtpsession->recv_rtp_src);
return rtpsession->recv_rtp_sink;
}
/* Remove sinkpad to receive RTP packets from senders. This will also remove
* the srcpad for the RTP packets.
*/
static void
remove_recv_rtp_sink (GstRtpSession * rtpsession)
{
GST_DEBUG_OBJECT (rtpsession, "removing RTP sink pad");
/* deactivate from source to sink */
gst_pad_set_active (rtpsession->recv_rtp_src, FALSE);
gst_pad_set_active (rtpsession->recv_rtp_sink, FALSE);
/* remove pads */
gst_element_remove_pad (GST_ELEMENT_CAST (rtpsession),
rtpsession->recv_rtp_sink);
rtpsession->recv_rtp_sink = NULL;
GST_DEBUG_OBJECT (rtpsession, "removing RTP src pad");
gst_element_remove_pad (GST_ELEMENT_CAST (rtpsession),
rtpsession->recv_rtp_src);
rtpsession->recv_rtp_src = NULL;
}
/* Create a sinkpad to receive RTCP messages from senders, this will also create a
* sync_src pad for the SR packets.
*/
static GstPad *
create_recv_rtcp_sink (GstRtpSession * rtpsession)
{
GST_DEBUG_OBJECT (rtpsession, "creating RTCP sink pad");
rtpsession->recv_rtcp_sink =
gst_pad_new_from_static_template (&rtpsession_recv_rtcp_sink_template,
"recv_rtcp_sink");
gst_pad_set_chain_function (rtpsession->recv_rtcp_sink,
gst_rtp_session_chain_recv_rtcp);
gst_pad_set_event_function (rtpsession->recv_rtcp_sink,
gst_rtp_session_event_recv_rtcp_sink);
gst_pad_set_iterate_internal_links_function (rtpsession->recv_rtcp_sink,
gst_rtp_session_iterate_internal_links);
gst_pad_set_active (rtpsession->recv_rtcp_sink, TRUE);
gst_element_add_pad (GST_ELEMENT_CAST (rtpsession),
rtpsession->recv_rtcp_sink);
GST_DEBUG_OBJECT (rtpsession, "creating sync src pad");
rtpsession->sync_src =
gst_pad_new_from_static_template (&rtpsession_sync_src_template,
"sync_src");
gst_pad_set_iterate_internal_links_function (rtpsession->sync_src,
gst_rtp_session_iterate_internal_links);
gst_pad_use_fixed_caps (rtpsession->sync_src);
gst_pad_set_active (rtpsession->sync_src, TRUE);
gst_element_add_pad (GST_ELEMENT_CAST (rtpsession), rtpsession->sync_src);
return rtpsession->recv_rtcp_sink;
}
static void
remove_recv_rtcp_sink (GstRtpSession * rtpsession)
{
GST_DEBUG_OBJECT (rtpsession, "removing RTCP sink pad");
gst_pad_set_active (rtpsession->sync_src, FALSE);
gst_pad_set_active (rtpsession->recv_rtcp_sink, FALSE);
gst_element_remove_pad (GST_ELEMENT_CAST (rtpsession),
rtpsession->recv_rtcp_sink);
rtpsession->recv_rtcp_sink = NULL;
GST_DEBUG_OBJECT (rtpsession, "removing sync src pad");
gst_element_remove_pad (GST_ELEMENT_CAST (rtpsession), rtpsession->sync_src);
rtpsession->sync_src = NULL;
}
/* Create a sinkpad to receive RTP packets for receivers. This will also create a
* send_rtp_src pad.
*/
static GstPad *
create_send_rtp_sink (GstRtpSession * rtpsession)
{
GST_DEBUG_OBJECT (rtpsession, "creating pad");
rtpsession->send_rtp_sink =
gst_pad_new_from_static_template (&rtpsession_send_rtp_sink_template,
"send_rtp_sink");
gst_pad_set_chain_function (rtpsession->send_rtp_sink,
gst_rtp_session_chain_send_rtp);
gst_pad_set_chain_list_function (rtpsession->send_rtp_sink,
gst_rtp_session_chain_send_rtp_list);
gst_pad_set_query_function (rtpsession->send_rtp_sink,
gst_rtp_session_query_send_rtp);
gst_pad_set_event_function (rtpsession->send_rtp_sink,
gst_rtp_session_event_send_rtp_sink);
gst_pad_set_iterate_internal_links_function (rtpsession->send_rtp_sink,
gst_rtp_session_iterate_internal_links);
gst_pad_set_active (rtpsession->send_rtp_sink, TRUE);
gst_element_add_pad (GST_ELEMENT_CAST (rtpsession),
rtpsession->send_rtp_sink);
rtpsession->send_rtp_src =
gst_pad_new_from_static_template (&rtpsession_send_rtp_src_template,
"send_rtp_src");
gst_pad_set_iterate_internal_links_function (rtpsession->send_rtp_src,
gst_rtp_session_iterate_internal_links);
gst_pad_set_event_function (rtpsession->send_rtp_src,
gst_rtp_session_event_send_rtp_src);
gst_pad_set_active (rtpsession->send_rtp_src, TRUE);
gst_element_add_pad (GST_ELEMENT_CAST (rtpsession), rtpsession->send_rtp_src);
return rtpsession->send_rtp_sink;
}
static void
remove_send_rtp_sink (GstRtpSession * rtpsession)
{
GST_DEBUG_OBJECT (rtpsession, "removing pad");
gst_pad_set_active (rtpsession->send_rtp_src, FALSE);
gst_pad_set_active (rtpsession->send_rtp_sink, FALSE);
gst_element_remove_pad (GST_ELEMENT_CAST (rtpsession),
rtpsession->send_rtp_sink);
rtpsession->send_rtp_sink = NULL;
gst_element_remove_pad (GST_ELEMENT_CAST (rtpsession),
rtpsession->send_rtp_src);
rtpsession->send_rtp_src = NULL;
}
/* Create a srcpad with the RTCP packets to send out.
* This pad will be driven by the RTP session manager when it wants to send out
* RTCP packets.
*/
static GstPad *
create_send_rtcp_src (GstRtpSession * rtpsession)
{
GST_DEBUG_OBJECT (rtpsession, "creating pad");
rtpsession->send_rtcp_src =
gst_pad_new_from_static_template (&rtpsession_send_rtcp_src_template,
"send_rtcp_src");
gst_pad_use_fixed_caps (rtpsession->send_rtcp_src);
gst_pad_set_active (rtpsession->send_rtcp_src, TRUE);
gst_pad_set_iterate_internal_links_function (rtpsession->send_rtcp_src,
gst_rtp_session_iterate_internal_links);
gst_pad_set_query_function (rtpsession->send_rtcp_src,
gst_rtp_session_query_send_rtcp_src);
gst_pad_set_event_function (rtpsession->send_rtcp_src,
gst_rtp_session_event_send_rtcp_src);
gst_element_add_pad (GST_ELEMENT_CAST (rtpsession),
rtpsession->send_rtcp_src);
return rtpsession->send_rtcp_src;
}
static void
remove_send_rtcp_src (GstRtpSession * rtpsession)
{
GST_DEBUG_OBJECT (rtpsession, "removing pad");
gst_pad_set_active (rtpsession->send_rtcp_src, FALSE);
gst_element_remove_pad (GST_ELEMENT_CAST (rtpsession),
rtpsession->send_rtcp_src);
rtpsession->send_rtcp_src = NULL;
}
static GstPad *
gst_rtp_session_request_new_pad (GstElement * element,
GstPadTemplate * templ, const gchar * name, const GstCaps * caps)
{
GstRtpSession *rtpsession;
GstElementClass *klass;
GstPad *result;
g_return_val_if_fail (templ != NULL, NULL);
g_return_val_if_fail (GST_IS_RTP_SESSION (element), NULL);
rtpsession = GST_RTP_SESSION (element);
klass = GST_ELEMENT_GET_CLASS (element);
GST_DEBUG_OBJECT (element, "requesting pad %s", GST_STR_NULL (name));
GST_RTP_SESSION_LOCK (rtpsession);
/* figure out the template */
if (templ == gst_element_class_get_pad_template (klass, "recv_rtp_sink")) {
if (rtpsession->recv_rtp_sink != NULL)
goto exists;
result = create_recv_rtp_sink (rtpsession);
} else if (templ == gst_element_class_get_pad_template (klass,
"recv_rtcp_sink")) {
if (rtpsession->recv_rtcp_sink != NULL)
goto exists;
result = create_recv_rtcp_sink (rtpsession);
} else if (templ == gst_element_class_get_pad_template (klass,
"send_rtp_sink")) {
if (rtpsession->send_rtp_sink != NULL)
goto exists;
result = create_send_rtp_sink (rtpsession);
} else if (templ == gst_element_class_get_pad_template (klass,
"send_rtcp_src")) {
if (rtpsession->send_rtcp_src != NULL)
goto exists;
result = create_send_rtcp_src (rtpsession);
} else
goto wrong_template;
GST_RTP_SESSION_UNLOCK (rtpsession);
return result;
/* ERRORS */
wrong_template:
{
GST_RTP_SESSION_UNLOCK (rtpsession);
g_warning ("gstrtpsession: this is not our template");
return NULL;
}
exists:
{
GST_RTP_SESSION_UNLOCK (rtpsession);
g_warning ("gstrtpsession: pad already requested");
return NULL;
}
}
static void
gst_rtp_session_release_pad (GstElement * element, GstPad * pad)
{
GstRtpSession *rtpsession;
g_return_if_fail (GST_IS_RTP_SESSION (element));
g_return_if_fail (GST_IS_PAD (pad));
rtpsession = GST_RTP_SESSION (element);
GST_DEBUG_OBJECT (element, "releasing pad %s:%s", GST_DEBUG_PAD_NAME (pad));
GST_RTP_SESSION_LOCK (rtpsession);
if (rtpsession->recv_rtp_sink == pad) {
remove_recv_rtp_sink (rtpsession);
} else if (rtpsession->recv_rtcp_sink == pad) {
remove_recv_rtcp_sink (rtpsession);
} else if (rtpsession->send_rtp_sink == pad) {
remove_send_rtp_sink (rtpsession);
} else if (rtpsession->send_rtcp_src == pad) {
remove_send_rtcp_src (rtpsession);
} else
goto wrong_pad;
GST_RTP_SESSION_UNLOCK (rtpsession);
return;
/* ERRORS */
wrong_pad:
{
GST_RTP_SESSION_UNLOCK (rtpsession);
g_warning ("gstrtpsession: asked to release an unknown pad");
return;
}
}
static void
gst_rtp_session_request_key_unit (RTPSession * sess,
gboolean all_headers, gpointer user_data)
{
GstRtpSession *rtpsession = GST_RTP_SESSION (user_data);
GstEvent *event;
GstPad *send_rtp_sink;
GST_RTP_SESSION_LOCK (rtpsession);
if ((send_rtp_sink = rtpsession->send_rtp_sink))
gst_object_ref (send_rtp_sink);
GST_RTP_SESSION_UNLOCK (rtpsession);
if (send_rtp_sink) {
event = gst_event_new_custom (GST_EVENT_CUSTOM_UPSTREAM,
gst_structure_new ("GstForceKeyUnit",
"all-headers", G_TYPE_BOOLEAN, all_headers, NULL));
gst_pad_push_event (send_rtp_sink, event);
gst_object_unref (send_rtp_sink);
}
}
static GstClockTime
gst_rtp_session_request_time (RTPSession * session, gpointer user_data)
{
GstRtpSession *rtpsession = GST_RTP_SESSION (user_data);
return gst_clock_get_time (rtpsession->priv->sysclock);
}
static void
gst_rtp_session_notify_nack (RTPSession * sess, guint16 seqnum,
guint16 blp, gpointer user_data)
{
GstRtpSession *rtpsession = GST_RTP_SESSION (user_data);
GstEvent *event;
GstPad *send_rtp_sink;
GST_RTP_SESSION_LOCK (rtpsession);
if ((send_rtp_sink = rtpsession->send_rtp_sink))
gst_object_ref (send_rtp_sink);
GST_RTP_SESSION_UNLOCK (rtpsession);
if (send_rtp_sink) {
while (TRUE) {
event = gst_event_new_custom (GST_EVENT_CUSTOM_UPSTREAM,
gst_structure_new ("GstRTPRetransmissionRequest",
"seqnum", G_TYPE_UINT, (guint) seqnum, NULL));
gst_pad_push_event (send_rtp_sink, event);
if (blp == 0)
break;
seqnum++;
while ((blp & 1) == 0) {
seqnum++;
blp >>= 1;
}
blp >>= 1;
}
gst_object_unref (send_rtp_sink);
}
}
Reference in a new issue View git blame Copy permalink