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gstreamer/gst/rtpmanager/gstrtpsession.c
Alejandro G. Castro eeea2a7fe8 rtpbin: pipeline gets an EOS when any rtpsources byes 
Instead of sending EOS when a source byes we have to wait for
all the sources to be gone, which means they already sent BYE and
were removed from the session. We now handle the EOS in the rtcp
loop checking the amount of sources in the session.

https://bugzilla.gnome.org/show_bug.cgi?id=773218
2016-11-01 20:16:18 +02:00

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/* GStreamer
* Copyright (C) <2007> Wim Taymans <wim.taymans@gmail.com>
*
* This library is free software; you can redistribute it and/or
* modify it under the terms of the GNU Library General Public
* License as published by the Free Software Foundation; either
* version 2 of the License, or (at your option) any later version.
*
* This library is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Library General Public License for more details.
*
* You should have received a copy of the GNU Library General Public
* License along with this library; if not, write to the
* Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
* Boston, MA 02110-1301, USA.
*/
/**
* SECTION:element-rtpsession
* @see_also: rtpjitterbuffer, rtpbin, rtpptdemux, rtpssrcdemux
*
* The RTP session manager models participants with 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>
* <listitem>
* <para>Support for multiple sender SSRC.</para>
* </listitem>
* </itemizedlist>
*
* The rtpsession 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
* 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 rtpsession .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 rtpsession 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 rtpsession .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 rtpsession 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>
*/
#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
GType
gst_rtp_ntp_time_source_get_type (void)
{
static GType type = 0;
static const GEnumValue values[] = {
{GST_RTP_NTP_TIME_SOURCE_NTP, "NTP time based on realtime clock", "ntp"},
{GST_RTP_NTP_TIME_SOURCE_UNIX, "UNIX time based on realtime clock", "unix"},
{GST_RTP_NTP_TIME_SOURCE_RUNNING_TIME,
"Running time based on pipeline clock",
"running-time"},
{GST_RTP_NTP_TIME_SOURCE_CLOCK_TIME, "Pipeline clock time", "clock-time"},
{0, NULL, NULL},
};
if (!type) {
type = g_enum_register_static ("GstRtpNtpTimeSource", values);
}
return type;
}
/* 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,
SIGNAL_ON_NEW_SENDER_SSRC,
SIGNAL_ON_SENDER_SSRC_ACTIVE,
LAST_SIGNAL
};
#define DEFAULT_BANDWIDTH 0
#define DEFAULT_RTCP_FRACTION 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
#define DEFAULT_MAX_DROPOUT_TIME 60000
#define DEFAULT_MAX_MISORDER_TIME 2000
#define DEFAULT_RTP_PROFILE GST_RTP_PROFILE_AVP
#define DEFAULT_NTP_TIME_SOURCE GST_RTP_NTP_TIME_SOURCE_NTP
#define DEFAULT_RTCP_SYNC_SEND_TIME TRUE
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_MAX_DROPOUT_TIME,
PROP_MAX_MISORDER_TIME,
PROP_STATS,
PROP_RTP_PROFILE,
PROP_NTP_TIME_SOURCE,
PROP_RTCP_SYNC_SEND_TIME
};
#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;
GstRtpNtpTimeSource ntp_time_source;
gboolean rtcp_sync_send_time;
guint rtx_count;
};
/* 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, 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, guint32 ssrc, gpointer user_data);
static void gst_rtp_session_reconfigure (RTPSession * sess, 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,
gst_rtp_session_reconfigure
};
/* 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 GstStructure *gst_rtp_session_create_stats (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 *send_rtp_sink;
g_signal_emit (sess, gst_rtp_session_signals[SIGNAL_ON_SSRC_COLLISION], 0,
src->ssrc);
GST_RTP_SESSION_LOCK (sess);
if ((send_rtp_sink = sess->send_rtp_sink))
gst_object_ref (send_rtp_sink);
GST_RTP_SESSION_UNLOCK (sess);
if (send_rtp_sink) {
GstStructure *structure;
GstEvent *event;
RTPSource *internal_src;
guint32 suggested_ssrc;
structure = gst_structure_new ("GstRTPCollision", "ssrc", G_TYPE_UINT,
(guint) src->ssrc, NULL);
/* if there is no source using the suggested ssrc, most probably because
* this ssrc has just collided, suggest upstream to use it */
suggested_ssrc = rtp_session_suggest_ssrc (session, NULL);
internal_src = rtp_session_get_source_by_ssrc (session, suggested_ssrc);
if (!internal_src)
gst_structure_set (structure, "suggested-ssrc", G_TYPE_UINT,
(guint) suggested_ssrc, NULL);
else
g_object_unref (internal_src);
event = gst_event_new_custom (GST_EVENT_CUSTOM_UPSTREAM, structure);
gst_pad_push_event (send_rtp_sink, event);
gst_object_unref (send_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);
}
static void
on_new_sender_ssrc (RTPSession * session, RTPSource * src, GstRtpSession * sess)
{
g_signal_emit (sess, gst_rtp_session_signals[SIGNAL_ON_NEW_SENDER_SSRC], 0,
src->ssrc);
}
static void
on_sender_ssrc_active (RTPSession * session, RTPSource * src,
GstRtpSession * sess)
{
g_signal_emit (sess, gst_rtp_session_signals[SIGNAL_ON_SENDER_SSRC_ACTIVE], 0,
src->ssrc);
}
static void
on_notify_stats (RTPSession * session, GParamSpec * spec,
GstRtpSession * rtpsession)
{
g_object_notify (G_OBJECT (rtpsession), "stats");
}
#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);
/**
* GstRtpSession::on-new-sender-ssrc:
* @sess: the object which received the signal
* @ssrc: the sender SSRC
*
* Since: 1.8
*
* Notify of a new sender SSRC that entered @session.
*/
gst_rtp_session_signals[SIGNAL_ON_NEW_SENDER_SSRC] =
g_signal_new ("on-new-sender-ssrc", G_TYPE_FROM_CLASS (klass),
G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRtpSessionClass, on_new_ssrc),
NULL, NULL, g_cclosure_marshal_VOID__UINT, G_TYPE_NONE, 1, G_TYPE_UINT);
/**
* GstRtpSession::on-sender-ssrc-active:
* @sess: the object which received the signal
* @ssrc: the sender SSRC
*
* Since: 1.8
*
* Notify of a sender SSRC that is active, i.e., sending RTCP.
*/
gst_rtp_session_signals[SIGNAL_ON_SENDER_SSRC_ACTIVE] =
g_signal_new ("on-sender-ssrc-active", G_TYPE_FROM_CLASS (klass),
G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRtpSessionClass,
on_ssrc_active), 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 "
"(DEPRECATED: Use ntp-time-source property)",
DEFAULT_USE_PIPELINE_CLOCK,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS | G_PARAM_DEPRECATED));
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));
g_object_class_install_property (gobject_class, PROP_MAX_DROPOUT_TIME,
g_param_spec_uint ("max-dropout-time", "Max dropout time",
"The maximum time (milliseconds) of missing packets tolerated.",
0, G_MAXUINT, DEFAULT_MAX_DROPOUT_TIME,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (gobject_class, PROP_MAX_MISORDER_TIME,
g_param_spec_uint ("max-misorder-time", "Max misorder time",
"The maximum time (milliseconds) of misordered packets tolerated.",
0, G_MAXUINT, DEFAULT_MAX_MISORDER_TIME,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
/**
* GstRtpSession::stats:
*
* Various session statistics. This property returns a GstStructure
* with name application/x-rtp-session-stats with the following fields:
*
* "rtx-count" G_TYPE_UINT The number of retransmission events
* received from downstream (in receiver mode)
* "rtx-drop-count" G_TYPE_UINT The number of retransmission events
* dropped (due to bandwidth constraints)
* "sent-nack-count" G_TYPE_UINT Number of NACKs sent
* "recv-nack-count" G_TYPE_UINT Number of NACKs received
* "source-stats" G_TYPE_BOXED GValueArray of #RTPSource::stats for all
* RTP sources (Since 1.8)
*
* Since: 1.4
*/
g_object_class_install_property (gobject_class, PROP_STATS,
g_param_spec_boxed ("stats", "Statistics",
"Various statistics", GST_TYPE_STRUCTURE,
G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (gobject_class, PROP_RTP_PROFILE,
g_param_spec_enum ("rtp-profile", "RTP Profile",
"RTP profile to use", GST_TYPE_RTP_PROFILE, DEFAULT_RTP_PROFILE,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (gobject_class, PROP_NTP_TIME_SOURCE,
g_param_spec_enum ("ntp-time-source", "NTP Time Source",
"NTP time source for RTCP packets",
gst_rtp_ntp_time_source_get_type (), DEFAULT_NTP_TIME_SOURCE,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (gobject_class, PROP_RTCP_SYNC_SEND_TIME,
g_param_spec_boolean ("rtcp-sync-send-time", "RTCP Sync Send Time",
"Use send time or capture time for RTCP sync "
"(TRUE = send time, FALSE = capture time)",
DEFAULT_RTCP_SYNC_SEND_TIME,
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_static_pad_template (gstelement_class,
&rtpsession_recv_rtp_sink_template);
gst_element_class_add_static_pad_template (gstelement_class,
&rtpsession_recv_rtcp_sink_template);
gst_element_class_add_static_pad_template (gstelement_class,
&rtpsession_send_rtp_sink_template);
/* src pads */
gst_element_class_add_static_pad_template (gstelement_class,
&rtpsession_recv_rtp_src_template);
gst_element_class_add_static_pad_template (gstelement_class,
&rtpsession_sync_src_template);
gst_element_class_add_static_pad_template (gstelement_class,
&rtpsession_send_rtp_src_template);
gst_element_class_add_static_pad_template (gstelement_class,
&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;
rtpsession->priv->rtcp_sync_send_time = DEFAULT_RTCP_SYNC_SEND_TIME;
/* 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);
g_signal_connect (rtpsession->priv->session, "on-new-sender-ssrc",
(GCallback) on_new_sender_ssrc, rtpsession);
g_signal_connect (rtpsession->priv->session, "on-sender-ssrc-active",
(GCallback) on_sender_ssrc_active, rtpsession);
g_signal_connect (rtpsession->priv->session, "notify::stats",
(GCallback) on_notify_stats, 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;
rtpsession->priv->rtx_count = 0;
rtpsession->priv->ntp_time_source = DEFAULT_NTP_TIME_SOURCE;
}
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;
case PROP_MAX_DROPOUT_TIME:
g_object_set_property (G_OBJECT (priv->session), "max-dropout-time",
value);
break;
case PROP_MAX_MISORDER_TIME:
g_object_set_property (G_OBJECT (priv->session), "max-misorder-time",
value);
break;
case PROP_RTP_PROFILE:
g_object_set_property (G_OBJECT (priv->session), "rtp-profile", value);
break;
case PROP_NTP_TIME_SOURCE:
priv->ntp_time_source = g_value_get_enum (value);
break;
case PROP_RTCP_SYNC_SEND_TIME:
priv->rtcp_sync_send_time = g_value_get_boolean (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;
case PROP_MAX_DROPOUT_TIME:
g_object_get_property (G_OBJECT (priv->session), "max-dropout-time",
value);
break;
case PROP_MAX_MISORDER_TIME:
g_object_get_property (G_OBJECT (priv->session), "max-misorder-time",
value);
break;
case PROP_STATS:
g_value_take_boxed (value, gst_rtp_session_create_stats (rtpsession));
break;
case PROP_RTP_PROFILE:
g_object_get_property (G_OBJECT (priv->session), "rtp-profile", value);
break;
case PROP_NTP_TIME_SOURCE:
g_value_set_enum (value, priv->ntp_time_source);
break;
case PROP_RTCP_SYNC_SEND_TIME:
g_value_set_boolean (value, priv->rtcp_sync_send_time);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
static GstStructure *
gst_rtp_session_create_stats (GstRtpSession * rtpsession)
{
GstStructure *s;
g_object_get (rtpsession->priv->session, "stats", &s, NULL);
gst_structure_set (s, "rtx-count", G_TYPE_UINT, rtpsession->priv->rtx_count,
NULL);
return s;
}
static void
get_current_times (GstRtpSession * rtpsession, GstClockTime * running_time,
guint64 * ntpnstime)
{
guint64 ntpns = -1;
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;
/* add constant to convert from 1970 based time to 1900 based time */
ntpns += (2208988800LL * GST_SECOND);
} else {
switch (rtpsession->priv->ntp_time_source) {
case GST_RTP_NTP_TIME_SOURCE_NTP:
case GST_RTP_NTP_TIME_SOURCE_UNIX:{
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 */
if (rtpsession->priv->ntp_time_source == GST_RTP_NTP_TIME_SOURCE_NTP)
ntpns += (2208988800LL * GST_SECOND);
break;
}
case GST_RTP_NTP_TIME_SOURCE_RUNNING_TIME:
ntpns = rt;
break;
case GST_RTP_NTP_TIME_SOURCE_CLOCK_TIME:
ntpns = clock_time;
break;
default:
ntpns = -1;
g_assert_not_reached ();
break;
}
}
gst_object_unref (clock);
} else {
GST_OBJECT_UNLOCK (rtpsession);
rt = -1;
ntpns = -1;
}
if (running_time)
*running_time = rt;
if (ntpnstime)
*ntpnstime = ntpns;
}
/* must be called with GST_RTP_SESSION_LOCK */
static void
signal_waiting_rtcp_thread_unlocked (GstRtpSession * rtpsession)
{
if (rtpsession->priv->wait_send) {
GST_LOG_OBJECT (rtpsession, "signal RTCP thread");
rtpsession->priv->wait_send = FALSE;
GST_RTP_SESSION_SIGNAL (rtpsession);
}
}
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 getting started");
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);
if (!rtp_session_get_num_sources (session)) {
/* when no sources left in the session, all of the them have went
* BYE at some point and removed, we can send EOS to the
* pipeline. */
GstPad *rtcp_src = rtpsession->send_rtcp_src;
if (rtcp_src) {
gst_object_ref (rtcp_src);
GST_LOG_OBJECT (rtpsession, "sending EOS");
GST_RTP_SESSION_UNLOCK (rtpsession);
gst_pad_push_event (rtpsession->send_rtcp_src, gst_event_new_eos ());
GST_RTP_SESSION_LOCK (rtpsession);
gst_object_unref (rtcp_src);
}
}
}
/* 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;
signal_waiting_rtcp_thread_unlocked (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);
signal_waiting_rtcp_thread_unlocked (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. */
static GstFlowReturn
gst_rtp_session_send_rtcp (RTPSession * sess, RTPSource * src,
GstBuffer * buffer, 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);
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);
/* set rtcp caps on output pad, this happens
* when we receive RTCP muxed with RTP according
* to RFC5761. Otherwise we would have forwarded
* the events from the recv_rtcp_sink pad already
*/
if (!gst_pad_has_current_caps (sync_src))
do_rtcp_events (rtpsession, sync_src);
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;
}
case GST_EVENT_EOS:
{
GstPad *rtcp_src;
ret =
gst_pad_push_event (rtpsession->recv_rtp_src, gst_event_ref (event));
GST_RTP_SESSION_LOCK (rtpsession);
if ((rtcp_src = rtpsession->send_rtcp_src))
gst_object_ref (rtcp_src);
GST_RTP_SESSION_UNLOCK (rtpsession);
if (rtcp_src) {
ret = gst_pad_push_event (rtcp_src, event);
gst_object_unref (rtcp_src);
} else {
gst_event_unref (event);
ret = TRUE;
}
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, avg_rtt;
GST_RTP_SESSION_LOCK (rtpsession);
rtpsession->priv->rtx_count++;
GST_RTP_SESSION_UNLOCK (rtpsession);
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;
if (!gst_structure_get_uint (s, "avg-rtt", &avg_rtt))
avg_rtt = 40;
/* remaining time to receive the packet */
max_delay = deadline;
if (max_delay > delay)
max_delay -= delay;
/* estimated RTT */
if (max_delay > avg_rtt)
max_delay -= avg_rtt;
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);
} else {
it = gst_iterator_new_single (GST_TYPE_PAD, NULL);
}
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");
GST_RTP_SESSION_LOCK (rtpsession);
signal_waiting_rtcp_thread_unlocked (rtpsession);
GST_RTP_SESSION_UNLOCK (rtpsession);
/* get NTP time when this packet was captured, this depends on the timestamp. */
timestamp = GST_BUFFER_PTS (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)) {
case GST_EVENT_SEGMENT:
/* Make sure that the sync_src pad has caps before the segment event.
* Otherwise we might get a segment event before caps from the receive
* RTCP pad, and then later when receiving RTCP packets will set caps.
* This will results in a sticky event misordering warning
*/
if (!gst_pad_has_current_caps (rtpsession->sync_src)) {
GstCaps *caps = gst_caps_new_empty_simple ("application/x-rtcp");
gst_pad_set_caps (rtpsession->sync_src, caps);
gst_caps_unref (caps);
}
/* fall through */
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");
GST_RTP_SESSION_LOCK (rtpsession);
signal_waiting_rtcp_thread_unlocked (rtpsession);
GST_RTP_SESSION_UNLOCK (rtpsession);
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;
gboolean is_random;
priv = rtpsession->priv;
ssrc = rtp_session_suggest_ssrc (priv->session, &is_random);
/* 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.
* Only do this if the session actually decided on an ssrc already,
* otherwise we give upstream the opportunity to select an ssrc itself */
if (!is_random) {
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);
} else {
result = gst_caps_new_empty_simple ("application/x-rtp");
}
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_PTS (buffer);
else
timestamp = -1;
} else {
timestamp = GST_BUFFER_PTS (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);
if (priv->rtcp_sync_send_time)
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_PROXY_ALLOCATION (rtpsession->recv_rtp_sink);
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_PROXY_CAPS (rtpsession->send_rtp_sink);
GST_PAD_SET_PROXY_ALLOCATION (rtpsession->send_rtp_sink);
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_PROXY_CAPS (rtpsession->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 ("rtpsession: this is not our template");
return NULL;
}
exists:
{
GST_RTP_SESSION_UNLOCK (rtpsession);
g_warning ("rtpsession: 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 ("rtpsession: 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, guint32 ssrc, 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,
"ssrc", G_TYPE_UINT, (guint) ssrc, 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);
}
}
static void
gst_rtp_session_reconfigure (RTPSession * sess, gpointer user_data)
{
GstRtpSession *rtpsession = GST_RTP_SESSION (user_data);
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) {
gst_pad_push_event (send_rtp_sink, gst_event_new_reconfigure ());
gst_object_unref (send_rtp_sink);
}
}
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