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gstreamer/gst/rtpmanager/gstrtpsession.c
Wim Taymans 6c781b9ca3 gst/rtpmanager/gstrtpjitterbuffer.c: Fix EOS handling. 
Original commit message from CVS:
* gst/rtpmanager/gstrtpjitterbuffer.c:
(gst_rtp_jitter_buffer_sink_event), (gst_rtp_jitter_buffer_chain),
(gst_rtp_jitter_buffer_loop):
Fix EOS handling.
Convert some DEBUG into WARNINGs.
Pause task when flushing.
* gst/rtpmanager/gstrtpsession.c: (gst_rtp_session_class_init),
(rtcp_thread), (gst_rtp_session_event_recv_rtcp_sink):
Use system clock for RTCP session management timeouts.
* gst/rtpmanager/rtpsession.c: (on_new_ssrc), (on_ssrc_collision),
(on_ssrc_validated), (on_bye_ssrc), (on_bye_timeout), (on_timeout):
Release the session lock when emiting signals.
2007-08-16 11:40:16 +00:00

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/* GStreamer
* Copyright (C) <2007> Wim Taymans <wim@fluendo.com>
*
* This library is free software; you can redistribute it and/or
* modify it under the terms of the GNU Library General Public
* License as published by the Free Software Foundation; either
* version 2 of the License, or (at your option) any later version.
*
* This library is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Library General Public License for more details.
*
* You should have received a copy of the GNU Library General Public
* License along with this library; if not, write to the
* Free Software Foundation, Inc., 59 Temple Place - Suite 330,
* Boston, MA 02111-1307, USA.
*/
/**
* SECTION:element-gstrtpsession
* @short_description: an RTP session manager
* @see_also: gstrtpjitterbuffer, gstrtpbin, gstrtpptdemux, gstrtpssrcdemux
*
* <refsect2>
* <para>
* 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.
* </para>
* <para>
* 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>
* </para>
* <para>
* 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.
* </para>
* <para>
* 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.
* </para>
* <para>
* 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.
* </para>
* <para>
* 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.
* </para>
* <para>
* 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.
* </para>
* <para>
* 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.
* </para>
* <title>Example pipelines</title>
* <para>
* <programlisting>
* gst-launch udpsrc port=5000 caps="application/x-rtp, ..." ! .recv_rtp_sink gstrtpsession .recv_rtp_src ! rtptheoradepay ! theoradec ! xvimagesink
* </programlisting>
* 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.
* </para>
* <para>
* <programlisting>
* gst-launch 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
* </programlisting>
* 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.
* </para>
* <para>
* <programlisting>
* gst-launch videotestsrc ! theoraenc ! rtptheorapay ! .send_rtp_sink gstrtpsession .send_rtp_src ! udpsink port=5000
* </programlisting>
* Send theora RTP packets through the session manager and out on UDP port 5000.
* </para>
* <para>
* <programlisting>
* gst-launch videotestsrc ! theoraenc ! rtptheorapay ! .send_rtp_sink gstrtpsession name=session .send_rtp_src \
* ! udpsink port=5000 session.send_rtcp_src ! udpsink port=5001
* </programlisting>
* 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.
* </para>
* </refsect2>
*
* Last reviewed on 2007-05-28 (0.10.5)
*/
#ifdef HAVE_CONFIG_H
#include "config.h"
#endif
#include "gstrtpbin-marshal.h"
#include "gstrtpsession.h"
#include "rtpsession.h"
GST_DEBUG_CATEGORY_STATIC (gst_rtp_session_debug);
#define GST_CAT_DEFAULT gst_rtp_session_debug
/* elementfactory information */
static const GstElementDetails rtpsession_details =
GST_ELEMENT_DETAILS ("RTP Session",
"Filter/Network/RTP",
"Implement an RTP session",
"Wim Taymans <wim@fluendo.com>");
/* 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_BYE_SSRC,
SIGNAL_ON_BYE_TIMEOUT,
SIGNAL_ON_TIMEOUT,
LAST_SIGNAL
};
enum
{
PROP_0
};
#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)
struct _GstRTPSessionPrivate
{
GMutex *lock;
RTPSession *session;
/* thread for sending out RTCP */
GstClockID id;
gboolean stop_thread;
GThread *thread;
};
/* 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, GstBuffer * buffer, gpointer user_data);
static GstFlowReturn gst_rtp_session_send_rtcp (RTPSession * sess,
RTPSource * src, GstBuffer * buffer, gpointer user_data);
static gint gst_rtp_session_clock_rate (RTPSession * sess, guint8 payload,
gpointer user_data);
static GstClockTime gst_rtp_session_get_time (RTPSession * sess,
gpointer user_data);
static void gst_rtp_session_reconsider (RTPSession * sess, gpointer user_data);
static RTPSessionCallbacks callbacks = {
gst_rtp_session_process_rtp,
gst_rtp_session_send_rtp,
gst_rtp_session_send_rtcp,
gst_rtp_session_clock_rate,
gst_rtp_session_get_time,
gst_rtp_session_reconsider
};
/* 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);
static void gst_rtp_session_release_pad (GstElement * element, GstPad * pad);
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)
{
g_signal_emit (sess, gst_rtp_session_signals[SIGNAL_ON_SSRC_COLLISION], 0,
src->ssrc);
}
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_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);
}
GST_BOILERPLATE (GstRTPSession, gst_rtp_session, GstElement, GST_TYPE_ELEMENT);
static void
gst_rtp_session_base_init (gpointer klass)
{
GstElementClass *element_class = GST_ELEMENT_CLASS (klass);
/* sink pads */
gst_element_class_add_pad_template (element_class,
gst_static_pad_template_get (&rtpsession_recv_rtp_sink_template));
gst_element_class_add_pad_template (element_class,
gst_static_pad_template_get (&rtpsession_recv_rtcp_sink_template));
gst_element_class_add_pad_template (element_class,
gst_static_pad_template_get (&rtpsession_send_rtp_sink_template));
/* src pads */
gst_element_class_add_pad_template (element_class,
gst_static_pad_template_get (&rtpsession_recv_rtp_src_template));
gst_element_class_add_pad_template (element_class,
gst_static_pad_template_get (&rtpsession_sync_src_template));
gst_element_class_add_pad_template (element_class,
gst_static_pad_template_get (&rtpsession_send_rtp_src_template));
gst_element_class_add_pad_template (element_class,
gst_static_pad_template_get (&rtpsession_send_rtcp_src_template));
gst_element_class_set_details (element_class, &rtpsession_details);
}
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, gst_rtp_bin_marshal_BOXED__UINT, 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-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);
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);
GST_DEBUG_CATEGORY_INIT (gst_rtp_session_debug,
"rtpsession", 0, "RTP Session");
}
static void
gst_rtp_session_init (GstRTPSession * rtpsession, GstRTPSessionClass * klass)
{
rtpsession->priv = GST_RTP_SESSION_GET_PRIVATE (rtpsession);
rtpsession->priv->lock = g_mutex_new ();
rtpsession->priv->session = rtp_session_new ();
/* 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-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);
}
static void
gst_rtp_session_finalize (GObject * object)
{
GstRTPSession *rtpsession;
rtpsession = GST_RTP_SESSION (object);
g_mutex_free (rtpsession->priv->lock);
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;
rtpsession = GST_RTP_SESSION (object);
switch (prop_id) {
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;
rtpsession = GST_RTP_SESSION (object);
switch (prop_id) {
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
static void
rtcp_thread (GstRTPSession * rtpsession)
{
GstClock *clock;
GstClockID id;
GstClockTime current_time;
GstClockTime next_timeout;
/* RTCP timeouts we use the system clock */
clock = gst_system_clock_obtain ();
if (clock == NULL)
goto no_clock;
current_time = gst_clock_get_time (clock);
GST_DEBUG_OBJECT (rtpsession, "entering RTCP thread");
GST_RTP_SESSION_LOCK (rtpsession);
while (!rtpsession->priv->stop_thread) {
GstClockReturn res;
/* get initial estimate */
next_timeout =
rtp_session_next_timeout (rtpsession->priv->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 (clock, 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 (clock);
/* 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. */
rtp_session_on_timeout (rtpsession->priv->session, current_time);
}
GST_RTP_SESSION_UNLOCK (rtpsession);
gst_object_unref (clock);
GST_DEBUG_OBJECT (rtpsession, "leaving RTCP thread");
return;
/* ERRORS */
no_clock:
{
GST_ELEMENT_ERROR (rtpsession, CORE, CLOCK, (NULL),
("Could not get system clock"));
return;
}
}
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;
rtpsession->priv->thread =
g_thread_create ((GThreadFunc) rtcp_thread, rtpsession, TRUE, &error);
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;
if (rtpsession->priv->id)
gst_clock_id_unschedule (rtpsession->priv->id);
GST_RTP_SESSION_UNLOCK (rtpsession);
g_thread_join (rtpsession->priv->thread);
}
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:
break;
case GST_STATE_CHANGE_PAUSED_TO_PLAYING:
break;
case GST_STATE_CHANGE_PLAYING_TO_PAUSED:
stop_rtcp_thread (rtpsession);
default:
break;
}
res = 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:
break;
case GST_STATE_CHANGE_READY_TO_NULL:
break;
default:
break;
}
return res;
/* ERRORS */
failed_thread:
{
return GST_STATE_CHANGE_FAILURE;
}
}
static void
gst_rtp_session_clear_pt_map (GstRTPSession * rtpsession)
{
/* FIXME, do something */
}
/* called when the session manager has an RTP packet ready for further
* processing */
static GstFlowReturn
gst_rtp_session_process_rtp (RTPSession * sess, RTPSource * src,
GstBuffer * buffer, gpointer user_data)
{
GstFlowReturn result;
GstRTPSession *rtpsession;
GstRTPSessionPrivate *priv;
rtpsession = GST_RTP_SESSION (user_data);
priv = rtpsession->priv;
GST_DEBUG_OBJECT (rtpsession, "reading receiving RTP packet");
if (rtpsession->recv_rtp_src) {
result = gst_pad_push (rtpsession->recv_rtp_src, buffer);
} else {
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,
GstBuffer * buffer, gpointer user_data)
{
GstFlowReturn result;
GstRTPSession *rtpsession;
GstRTPSessionPrivate *priv;
rtpsession = GST_RTP_SESSION (user_data);
priv = rtpsession->priv;
GST_DEBUG_OBJECT (rtpsession, "sending RTP packet");
if (rtpsession->send_rtp_src) {
result = gst_pad_push (rtpsession->send_rtp_src, buffer);
} else {
gst_buffer_unref (buffer);
result = GST_FLOW_OK;
}
return result;
}
/* 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;
GstRTPSessionPrivate *priv;
rtpsession = GST_RTP_SESSION (user_data);
priv = rtpsession->priv;
GST_DEBUG_OBJECT (rtpsession, "sending RTCP");
if (rtpsession->send_rtcp_src) {
result = gst_pad_push (rtpsession->send_rtcp_src, buffer);
} else {
gst_buffer_unref (buffer);
result = GST_FLOW_OK;
}
return result;
}
/* 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;
GValue ret = { 0 };
GValue args[2] = { {0}, {0} };
GstCaps *caps;
const GstStructure *caps_struct;
rtpsession = GST_RTP_SESSION_CAST (user_data);
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);
g_signal_emitv (args, gst_rtp_session_signals[SIGNAL_REQUEST_PT_MAP], 0,
&ret);
caps = (GstCaps *) g_value_get_boxed (&ret);
if (!caps)
goto no_caps;
caps_struct = gst_caps_get_structure (caps, 0);
if (!gst_structure_get_int (caps_struct, "clock-rate", &result))
goto no_clock_rate;
GST_DEBUG_OBJECT (rtpsession, "parsed clock-rate %d", result);
return result;
/* ERRORS */
no_caps:
{
GST_DEBUG_OBJECT (rtpsession, "could not get caps");
return -1;
}
no_clock_rate:
{
GST_DEBUG_OBJECT (rtpsession, "could not clock-rate from caps");
return -1;
}
}
/* called when the session manager needs the time of clock */
static GstClockTime
gst_rtp_session_get_time (RTPSession * sess, gpointer user_data)
{
GstClockTime result;
GstRTPSession *rtpsession;
GstClock *clock;
rtpsession = GST_RTP_SESSION_CAST (user_data);
clock = gst_element_get_clock (GST_ELEMENT_CAST (rtpsession));
if (clock) {
result = gst_clock_get_time (clock);
gst_object_unref (clock);
} else
result = GST_CLOCK_TIME_NONE;
return result;
}
/* 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 GstFlowReturn
gst_rtp_session_event_recv_rtp_sink (GstPad * pad, GstEvent * event)
{
GstRTPSession *rtpsession;
GstRTPSessionPrivate *priv;
gboolean ret = FALSE;
rtpsession = GST_RTP_SESSION (gst_pad_get_parent (pad));
priv = rtpsession->priv;
GST_DEBUG_OBJECT (rtpsession, "received event %s",
GST_EVENT_TYPE_NAME (event));
switch (GST_EVENT_TYPE (event)) {
default:
ret = gst_pad_push_event (rtpsession->recv_rtp_src, event);
break;
}
gst_object_unref (rtpsession);
return ret;
}
/* 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, GstBuffer * buffer)
{
GstRTPSession *rtpsession;
GstRTPSessionPrivate *priv;
GstFlowReturn ret;
rtpsession = GST_RTP_SESSION (gst_pad_get_parent (pad));
priv = rtpsession->priv;
GST_DEBUG_OBJECT (rtpsession, "received RTP packet");
ret = rtp_session_process_rtp (priv->session, buffer);
gst_object_unref (rtpsession);
return ret;
}
static GstFlowReturn
gst_rtp_session_event_recv_rtcp_sink (GstPad * pad, GstEvent * event)
{
GstRTPSession *rtpsession;
GstRTPSessionPrivate *priv;
gboolean ret = FALSE;
rtpsession = GST_RTP_SESSION (gst_pad_get_parent (pad));
priv = rtpsession->priv;
GST_DEBUG_OBJECT (rtpsession, "received event %s",
GST_EVENT_TYPE_NAME (event));
switch (GST_EVENT_TYPE (event)) {
default:
if (rtpsession->send_rtcp_src) {
gst_event_ref (event);
ret = gst_pad_push_event (rtpsession->send_rtcp_src, event);
}
ret = gst_pad_push_event (rtpsession->sync_src, event);
break;
}
gst_object_unref (rtpsession);
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, GstBuffer * buffer)
{
GstRTPSession *rtpsession;
GstRTPSessionPrivate *priv;
GstFlowReturn ret;
rtpsession = GST_RTP_SESSION (gst_pad_get_parent (pad));
priv = rtpsession->priv;
GST_DEBUG_OBJECT (rtpsession, "received RTCP packet");
ret = rtp_session_process_rtcp (priv->session, buffer);
gst_object_unref (rtpsession);
return GST_FLOW_OK;
}
static GstFlowReturn
gst_rtp_session_event_send_rtp_sink (GstPad * pad, GstEvent * event)
{
GstRTPSession *rtpsession;
GstRTPSessionPrivate *priv;
gboolean ret = FALSE;
rtpsession = GST_RTP_SESSION (gst_pad_get_parent (pad));
priv = rtpsession->priv;
GST_DEBUG_OBJECT (rtpsession, "received event");
switch (GST_EVENT_TYPE (event)) {
default:
ret = gst_pad_push_event (rtpsession->send_rtp_src, event);
break;
}
gst_object_unref (rtpsession);
return ret;
}
/* Recieve an RTP packet 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 (GstPad * pad, GstBuffer * buffer)
{
GstRTPSession *rtpsession;
GstRTPSessionPrivate *priv;
GstFlowReturn ret;
rtpsession = GST_RTP_SESSION (gst_pad_get_parent (pad));
priv = rtpsession->priv;
GST_DEBUG_OBJECT (rtpsession, "received RTP packet");
ret = rtp_session_send_rtp (priv->session, buffer);
gst_object_unref (rtpsession);
return ret;
}
/* 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_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_active (rtpsession->recv_rtp_src, TRUE);
gst_element_add_pad (GST_ELEMENT_CAST (rtpsession), rtpsession->recv_rtp_src);
return rtpsession->recv_rtp_sink;
}
/* 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_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_active (rtpsession->sync_src, TRUE);
gst_element_add_pad (GST_ELEMENT_CAST (rtpsession), rtpsession->sync_src);
return rtpsession->recv_rtcp_sink;
}
/* 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_event_function (rtpsession->send_rtp_sink,
gst_rtp_session_event_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_active (rtpsession->send_rtp_src, TRUE);
gst_element_add_pad (GST_ELEMENT_CAST (rtpsession), rtpsession->send_rtp_src);
return rtpsession->send_rtp_sink;
}
/* 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_set_active (rtpsession->send_rtcp_src, TRUE);
gst_element_add_pad (GST_ELEMENT_CAST (rtpsession),
rtpsession->send_rtcp_src);
return rtpsession->send_rtcp_src;
}
static GstPad *
gst_rtp_session_request_new_pad (GstElement * element,
GstPadTemplate * templ, const gchar * name)
{
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)
{
}
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