mirrors/gstreamer
mirrors/gstreamer
1
1
Fork 0
mirror of https://gitlab.freedesktop.org/gstreamer/gstreamer.git synced 2024-11-08 18:39:54 +00:00
Code Issues 1 Projects Releases Wiki Activity
gstreamer/gst/rtpmanager/gstrtpsession.c
Peter Kjellerstedt 11ebf3a6aa gst/rtpmanager/gstrtpsession.c: Make sure to unref the rtpsession returned by gst_pad_get_parent() to prevent a memor... 
Original commit message from CVS:
* gst/rtpmanager/gstrtpsession.c: (gst_rtp_session_query_send_rtcp_src):
Make sure to unref the rtpsession returned by gst_pad_get_parent() to
prevent a memory leak.
2008-05-12 14:28:09 +00:00

1871 lines
59 KiB
C
Raw Blame History

/* GStreamer
* Copyright (C) <2007> Wim Taymans <wim.taymans@gmail.com>
*
* This library is free software; you can redistribute it and/or
* modify it under the terms of the GNU Library General Public
* License as published by the Free Software Foundation; either
* version 2 of the License, or (at your option) any later version.
*
* This library is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Library General Public License for more details.
*
* You should have received a copy of the GNU Library General Public
* License along with this library; if not, write to the
* Free Software Foundation, Inc., 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 <gst/rtp/gstrtpbuffer.h>
#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.taymans@gmail.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_SSRC_ACTIVE,
SIGNAL_ON_SSRC_SDES,
SIGNAL_ON_BYE_SSRC,
SIGNAL_ON_BYE_TIMEOUT,
SIGNAL_ON_TIMEOUT,
LAST_SIGNAL
};
#define DEFAULT_NTP_NS_BASE 0
#define DEFAULT_BANDWIDTH RTP_STATS_BANDWIDTH
#define DEFAULT_RTCP_FRACTION RTP_STATS_RTCP_BANDWIDTH
#define DEFAULT_SDES_CNAME NULL
#define DEFAULT_SDES_NAME NULL
#define DEFAULT_SDES_EMAIL NULL
#define DEFAULT_SDES_PHONE NULL
#define DEFAULT_SDES_LOCATION NULL
#define DEFAULT_SDES_TOOL NULL
#define DEFAULT_SDES_NOTE NULL
#define DEFAULT_NUM_SOURCES 0
#define DEFAULT_NUM_ACTIVE_SOURCES 0
enum
{
PROP_0,
PROP_NTP_NS_BASE,
PROP_BANDWIDTH,
PROP_RTCP_FRACTION,
PROP_SDES_CNAME,
PROP_SDES_NAME,
PROP_SDES_EMAIL,
PROP_SDES_PHONE,
PROP_SDES_LOCATION,
PROP_SDES_TOOL,
PROP_SDES_NOTE,
PROP_NUM_SOURCES,
PROP_NUM_ACTIVE_SOURCES,
PROP_LAST
};
#define GST_RTP_SESSION_GET_PRIVATE(obj) \
(G_TYPE_INSTANCE_GET_PRIVATE ((obj), GST_TYPE_RTP_SESSION, GstRtpSessionPrivate))
#define GST_RTP_SESSION_LOCK(sess) g_mutex_lock ((sess)->priv->lock)
#define GST_RTP_SESSION_UNLOCK(sess) g_mutex_unlock ((sess)->priv->lock)
struct _GstRtpSessionPrivate
{
GMutex *lock;
RTPSession *session;
/* thread for sending out RTCP */
GstClockID id;
gboolean stop_thread;
GThread *thread;
gboolean thread_stopped;
/* caps mapping */
GHashTable *ptmap;
/* NTP base time */
guint64 ntpnsbase;
};
/* 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 GstFlowReturn gst_rtp_session_sync_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 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_sync_rtcp,
gst_rtp_session_send_rtcp,
gst_rtp_session_clock_rate,
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_ssrc_active (RTPSession * session, RTPSource * src, GstRtpSession * sess)
{
g_signal_emit (sess, gst_rtp_session_signals[SIGNAL_ON_SSRC_ACTIVE], 0,
src->ssrc);
}
static GstStructure *
source_get_sdes_structure (RTPSource * src)
{
GstStructure *result;
GValue val = { 0 };
gchar *str;
result = gst_structure_empty_new ("GstRTPSessionSDES");
gst_structure_set (result, "ssrc", G_TYPE_UINT, src->ssrc, NULL);
g_value_init (&val, G_TYPE_STRING);
str = rtp_source_get_sdes_string (src, GST_RTCP_SDES_CNAME);
if (str) {
g_value_take_string (&val, str);
gst_structure_set_value (result, "cname", &val);
}
str = rtp_source_get_sdes_string (src, GST_RTCP_SDES_NAME);
if (str) {
g_value_take_string (&val, str);
gst_structure_set_value (result, "name", &val);
}
str = rtp_source_get_sdes_string (src, GST_RTCP_SDES_EMAIL);
if (str) {
g_value_take_string (&val, str);
gst_structure_set_value (result, "email", &val);
}
str = rtp_source_get_sdes_string (src, GST_RTCP_SDES_PHONE);
if (str) {
g_value_take_string (&val, str);
gst_structure_set_value (result, "phone", &val);
}
str = rtp_source_get_sdes_string (src, GST_RTCP_SDES_LOC);
if (str) {
g_value_take_string (&val, str);
gst_structure_set_value (result, "location", &val);
}
str = rtp_source_get_sdes_string (src, GST_RTCP_SDES_TOOL);
if (str) {
g_value_take_string (&val, str);
gst_structure_set_value (result, "tool", &val);
}
str = rtp_source_get_sdes_string (src, GST_RTCP_SDES_NOTE);
if (str) {
g_value_take_string (&val, str);
gst_structure_set_value (result, "note", &val);
}
str = rtp_source_get_sdes_string (src, GST_RTCP_SDES_PRIV);
if (str) {
g_value_take_string (&val, str);
gst_structure_set_value (result, "priv", &val);
}
g_value_unset (&val);
return result;
}
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);
s = source_get_sdes_structure (src);
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);
}
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-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);
g_object_class_install_property (gobject_class, PROP_NTP_NS_BASE,
g_param_spec_uint64 ("ntp-ns-base", "NTP base time",
"The NTP base time corresponding to running_time 0", 0,
G_MAXUINT64, DEFAULT_NTP_NS_BASE, G_PARAM_READWRITE));
g_object_class_install_property (gobject_class, PROP_BANDWIDTH,
g_param_spec_double ("bandwidth", "Bandwidth",
"The bandwidth of the session",
0.0, G_MAXDOUBLE, DEFAULT_BANDWIDTH, G_PARAM_READWRITE));
g_object_class_install_property (gobject_class, PROP_RTCP_FRACTION,
g_param_spec_double ("rtcp-fraction", "RTCP Fraction",
"The fraction of the bandwidth used for RTCP",
0.0, G_MAXDOUBLE, DEFAULT_RTCP_FRACTION, G_PARAM_READWRITE));
g_object_class_install_property (gobject_class, PROP_SDES_CNAME,
g_param_spec_string ("sdes-cname", "SDES CNAME",
"The CNAME to put in SDES messages of this session",
DEFAULT_SDES_CNAME, G_PARAM_READWRITE));
g_object_class_install_property (gobject_class, PROP_SDES_NAME,
g_param_spec_string ("sdes-name", "SDES NAME",
"The NAME to put in SDES messages of this session",
DEFAULT_SDES_NAME, G_PARAM_READWRITE));
g_object_class_install_property (gobject_class, PROP_SDES_EMAIL,
g_param_spec_string ("sdes-email", "SDES EMAIL",
"The EMAIL to put in SDES messages of this session",
DEFAULT_SDES_EMAIL, G_PARAM_READWRITE));
g_object_class_install_property (gobject_class, PROP_SDES_PHONE,
g_param_spec_string ("sdes-phone", "SDES PHONE",
"The PHONE to put in SDES messages of this session",
DEFAULT_SDES_PHONE, G_PARAM_READWRITE));
g_object_class_install_property (gobject_class, PROP_SDES_LOCATION,
g_param_spec_string ("sdes-location", "SDES LOCATION",
"The LOCATION to put in SDES messages of this session",
DEFAULT_SDES_LOCATION, G_PARAM_READWRITE));
g_object_class_install_property (gobject_class, PROP_SDES_TOOL,
g_param_spec_string ("sdes-tool", "SDES TOOL",
"The TOOL to put in SDES messages of this session",
DEFAULT_SDES_TOOL, G_PARAM_READWRITE));
g_object_class_install_property (gobject_class, PROP_SDES_NOTE,
g_param_spec_string ("sdes-note", "SDES NOTE",
"The NOTE to put in SDES messages of this session",
DEFAULT_SDES_NOTE, G_PARAM_READWRITE));
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_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));
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-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);
rtpsession->priv->ptmap = g_hash_table_new_full (NULL, NULL, NULL,
(GDestroyNotify) gst_caps_unref);
gst_segment_init (&rtpsession->recv_rtp_seg, GST_FORMAT_UNDEFINED);
gst_segment_init (&rtpsession->send_rtp_seg, GST_FORMAT_UNDEFINED);
rtpsession->priv->thread_stopped = TRUE;
}
static void
gst_rtp_session_finalize (GObject * object)
{
GstRtpSession *rtpsession;
rtpsession = GST_RTP_SESSION (object);
if (rtpsession->recv_rtp_sink != NULL)
gst_object_unref (rtpsession->recv_rtp_sink);
if (rtpsession->recv_rtcp_sink != NULL)
gst_object_unref (rtpsession->recv_rtcp_sink);
if (rtpsession->send_rtp_sink != NULL)
gst_object_unref (rtpsession->send_rtp_sink);
if (rtpsession->send_rtcp_src != NULL)
gst_object_unref (rtpsession->send_rtcp_src);
g_hash_table_destroy (rtpsession->priv->ptmap);
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;
GstRtpSessionPrivate *priv;
rtpsession = GST_RTP_SESSION (object);
priv = rtpsession->priv;
switch (prop_id) {
case PROP_NTP_NS_BASE:
GST_OBJECT_LOCK (rtpsession);
priv->ntpnsbase = g_value_get_uint64 (value);
GST_DEBUG_OBJECT (rtpsession, "setting NTP base to %" GST_TIME_FORMAT,
GST_TIME_ARGS (priv->ntpnsbase));
GST_OBJECT_UNLOCK (rtpsession);
break;
case PROP_BANDWIDTH:
rtp_session_set_bandwidth (priv->session, g_value_get_double (value));
break;
case PROP_RTCP_FRACTION:
rtp_session_set_rtcp_fraction (priv->session, g_value_get_double (value));
break;
case PROP_SDES_CNAME:
rtp_session_set_sdes_string (priv->session, GST_RTCP_SDES_CNAME,
g_value_get_string (value));
break;
case PROP_SDES_NAME:
rtp_session_set_sdes_string (priv->session, GST_RTCP_SDES_NAME,
g_value_get_string (value));
break;
case PROP_SDES_EMAIL:
rtp_session_set_sdes_string (priv->session, GST_RTCP_SDES_EMAIL,
g_value_get_string (value));
break;
case PROP_SDES_PHONE:
rtp_session_set_sdes_string (priv->session, GST_RTCP_SDES_PHONE,
g_value_get_string (value));
break;
case PROP_SDES_LOCATION:
rtp_session_set_sdes_string (priv->session, GST_RTCP_SDES_LOC,
g_value_get_string (value));
break;
case PROP_SDES_TOOL:
rtp_session_set_sdes_string (priv->session, GST_RTCP_SDES_TOOL,
g_value_get_string (value));
break;
case PROP_SDES_NOTE:
rtp_session_set_sdes_string (priv->session, GST_RTCP_SDES_NOTE,
g_value_get_string (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_NTP_NS_BASE:
GST_OBJECT_LOCK (rtpsession);
g_value_set_uint64 (value, priv->ntpnsbase);
GST_OBJECT_UNLOCK (rtpsession);
break;
case PROP_BANDWIDTH:
g_value_set_double (value, rtp_session_get_bandwidth (priv->session));
break;
case PROP_RTCP_FRACTION:
g_value_set_double (value, rtp_session_get_rtcp_fraction (priv->session));
break;
case PROP_SDES_CNAME:
g_value_take_string (value, rtp_session_get_sdes_string (priv->session,
GST_RTCP_SDES_CNAME));
break;
case PROP_SDES_NAME:
g_value_take_string (value, rtp_session_get_sdes_string (priv->session,
GST_RTCP_SDES_NAME));
break;
case PROP_SDES_EMAIL:
g_value_take_string (value, rtp_session_get_sdes_string (priv->session,
GST_RTCP_SDES_EMAIL));
break;
case PROP_SDES_PHONE:
g_value_take_string (value, rtp_session_get_sdes_string (priv->session,
GST_RTCP_SDES_PHONE));
break;
case PROP_SDES_LOCATION:
g_value_take_string (value, rtp_session_get_sdes_string (priv->session,
GST_RTCP_SDES_LOC));
break;
case PROP_SDES_TOOL:
g_value_take_string (value, rtp_session_get_sdes_string (priv->session,
GST_RTCP_SDES_TOOL));
break;
case PROP_SDES_NOTE:
g_value_take_string (value, rtp_session_get_sdes_string (priv->session,
GST_RTCP_SDES_NOTE));
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;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
static guint64
get_current_ntp_ns_time (GstRtpSession * rtpsession)
{
guint64 ntpnstime;
GstClock *clock;
GstClockTime base_time, ntpnsbase;
GST_OBJECT_LOCK (rtpsession);
if ((clock = GST_ELEMENT_CLOCK (rtpsession))) {
base_time = GST_ELEMENT_CAST (rtpsession)->base_time;
ntpnsbase = rtpsession->priv->ntpnsbase;
gst_object_ref (clock);
GST_OBJECT_UNLOCK (rtpsession);
/* get current NTP time */
ntpnstime = gst_clock_get_time (clock);
/* convert to running time */
ntpnstime -= base_time;
/* add NTP base offset */
ntpnstime += ntpnsbase;
gst_object_unref (clock);
} else {
GST_OBJECT_UNLOCK (rtpsession);
ntpnstime = -1;
}
return ntpnstime;
}
static void
rtcp_thread (GstRtpSession * rtpsession)
{
GstClock *sysclock;
GstClockID id;
GstClockTime current_time;
GstClockTime next_timeout;
guint64 ntpnstime;
/* for RTCP timeouts we use the system clock */
sysclock = gst_system_clock_obtain ();
if (sysclock == NULL)
goto no_sysclock;
current_time = gst_clock_get_time (sysclock);
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 (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 */
ntpnstime = get_current_ntp_ns_time (rtpsession);
/* 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 (rtpsession->priv->session, current_time, ntpnstime);
GST_RTP_SESSION_LOCK (rtpsession);
}
/* mark the thread as stopped now */
rtpsession->priv->thread_stopped = TRUE;
GST_RTP_SESSION_UNLOCK (rtpsession);
gst_object_unref (sysclock);
GST_DEBUG_OBJECT (rtpsession, "leaving RTCP thread");
return;
/* ERRORS */
no_sysclock:
{
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;
if (rtpsession->priv->thread_stopped) {
/* 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_create ((GThreadFunc) rtcp_thread, rtpsession, TRUE, &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;
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;
GstRtpSessionPrivate *priv;
rtpsession = GST_RTP_SESSION (element);
priv = rtpsession->priv;
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:
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 = 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 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;
if (rtpsession->recv_rtp_src) {
GST_DEBUG_OBJECT (rtpsession, "pushing received RTP packet");
result = gst_pad_push (rtpsession->recv_rtp_src, buffer);
} 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,
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;
if (rtpsession->send_rtcp_src) {
GstCaps *caps;
/* set rtcp caps on output pad */
if (!(caps = GST_PAD_CAPS (rtpsession->send_rtcp_src))) {
caps = gst_caps_new_simple ("application/x-rtcp", NULL);
gst_pad_set_caps (rtpsession->send_rtcp_src, caps);
gst_caps_unref (caps);
}
gst_buffer_set_caps (buffer, caps);
GST_DEBUG_OBJECT (rtpsession, "sending RTCP");
result = gst_pad_push (rtpsession->send_rtcp_src, buffer);
} else {
GST_DEBUG_OBJECT (rtpsession, "not sending RTCP, no output pad");
gst_buffer_unref (buffer);
result = GST_FLOW_OK;
}
return result;
}
/* 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,
RTPSource * src, GstBuffer * buffer, gpointer user_data)
{
GstFlowReturn result;
GstRtpSession *rtpsession;
GstRtpSessionPrivate *priv;
rtpsession = GST_RTP_SESSION (user_data);
priv = rtpsession->priv;
if (rtpsession->sync_src) {
GstCaps *caps;
/* set rtcp caps on output pad */
if (!(caps = GST_PAD_CAPS (rtpsession->sync_src))) {
caps = gst_caps_new_simple ("application/x-rtcp", NULL);
gst_pad_set_caps (rtpsession->sync_src, caps);
gst_caps_unref (caps);
}
gst_buffer_set_caps (buffer, caps);
GST_DEBUG_OBJECT (rtpsession, "sending Sync RTCP");
result = gst_pad_push (rtpsession->sync_src, buffer);
} else {
GST_DEBUG_OBJECT (rtpsession, "not sending Sync RTCP, no output pad");
gst_buffer_unref (buffer);
result = GST_FLOW_OK;
}
return result;
}
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));
}
/* called when the session manager needs the clock rate */
static gint
gst_rtp_session_clock_rate (RTPSession * sess, guint8 payload,
gpointer user_data)
{
gint ipayload, result = -1;
GstRtpSession *rtpsession;
GstRtpSessionPrivate *priv;
GValue ret = { 0 };
GValue args[2] = { {0}, {0} };
GstCaps *caps;
const GstStructure *s;
rtpsession = GST_RTP_SESSION_CAST (user_data);
priv = rtpsession->priv;
GST_RTP_SESSION_LOCK (rtpsession);
ipayload = payload; /* make compiler happy */
caps = g_hash_table_lookup (priv->ptmap, GINT_TO_POINTER (ipayload));
if (caps) {
gst_caps_ref (caps);
goto found;
}
/* 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);
g_signal_emitv (args, gst_rtp_session_signals[SIGNAL_REQUEST_PT_MAP], 0,
&ret);
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);
found:
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:
GST_RTP_SESSION_UNLOCK (rtpsession);
return result;
/* ERRORS */
no_caps:
{
GST_DEBUG_OBJECT (rtpsession, "could not get caps");
goto done;
}
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, 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)) {
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_NEWSEGMENT:
{
gboolean update;
gdouble rate, arate;
GstFormat format;
gint64 start, stop, time;
GstSegment *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_parse_new_segment_full (event, &update, &rate, &arate, &format,
&start, &stop, &time);
GST_DEBUG_OBJECT (rtpsession,
"configured NEWSEGMENT update %d, rate %lf, applied rate %lf, "
"format GST_FORMAT_TIME, "
"%" GST_TIME_FORMAT " -- %" GST_TIME_FORMAT
", time %" GST_TIME_FORMAT ", accum %" GST_TIME_FORMAT,
update, rate, arate, GST_TIME_ARGS (segment->start),
GST_TIME_ARGS (segment->stop), GST_TIME_ARGS (segment->time),
GST_TIME_ARGS (segment->accum));
gst_segment_set_newsegment_full (segment, update, rate,
arate, format, start, stop, time);
/* push event forward */
ret = gst_pad_push_event (rtpsession->recv_rtp_src, event);
break;
}
default:
ret = gst_pad_push_event (rtpsession->recv_rtp_src, event);
break;
}
gst_object_unref (rtpsession);
return ret;
}
static GList *
gst_rtp_session_internal_links (GstPad * pad)
{
GstRtpSession *rtpsession;
GstRtpSessionPrivate *priv;
GList *res = NULL;
rtpsession = GST_RTP_SESSION (gst_pad_get_parent (pad));
priv = rtpsession->priv;
if (pad == rtpsession->recv_rtp_src) {
res = g_list_prepend (res, rtpsession->recv_rtp_sink);
} else if (pad == rtpsession->recv_rtp_sink) {
res = g_list_prepend (res, rtpsession->recv_rtp_src);
} else if (pad == rtpsession->send_rtp_src) {
res = g_list_prepend (res, rtpsession->send_rtp_sink);
} else if (pad == rtpsession->send_rtp_sink) {
res = g_list_prepend (res, rtpsession->send_rtp_src);
}
gst_object_unref (rtpsession);
return res;
}
static gboolean
gst_rtp_session_sink_setcaps (GstPad * pad, GstCaps * caps)
{
GstRtpSession *rtpsession;
GstRtpSessionPrivate *priv;
rtpsession = GST_RTP_SESSION (gst_pad_get_parent (pad));
priv = rtpsession->priv;
GST_RTP_SESSION_LOCK (rtpsession);
gst_rtp_session_cache_caps (rtpsession, caps);
GST_RTP_SESSION_UNLOCK (rtpsession);
gst_object_unref (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, GstBuffer * buffer)
{
GstRtpSession *rtpsession;
GstRtpSessionPrivate *priv;
GstFlowReturn ret;
guint64 ntpnstime;
GstClockTime timestamp;
rtpsession = GST_RTP_SESSION (gst_pad_get_parent (pad));
priv = rtpsession->priv;
GST_DEBUG_OBJECT (rtpsession, "received RTP packet");
/* get NTP time when this packet was captured, this depends on the timestamp. */
timestamp = GST_BUFFER_TIMESTAMP (buffer);
if (GST_CLOCK_TIME_IS_VALID (timestamp)) {
/* convert to running time using the segment values */
ntpnstime =
gst_segment_to_running_time (&rtpsession->recv_rtp_seg, GST_FORMAT_TIME,
timestamp);
/* add constant to convert running time to NTP time */
ntpnstime += priv->ntpnsbase;
} else {
ntpnstime = get_current_ntp_ns_time (rtpsession);
}
ret = rtp_session_process_rtp (priv->session, buffer, ntpnstime);
if (ret != GST_FLOW_OK)
goto push_error;
done:
gst_object_unref (rtpsession);
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, 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 gboolean
gst_rtp_session_query_send_rtcp_src (GstPad * pad, GstQuery * query)
{
GstRtpSession *rtpsession;
GstRtpSessionPrivate *priv;
gboolean ret = FALSE;
rtpsession = GST_RTP_SESSION (gst_pad_get_parent (pad));
priv = rtpsession->priv;
GST_DEBUG_OBJECT (rtpsession, "received 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;
}
gst_object_unref (rtpsession);
return ret;
}
static gboolean
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)) {
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_NEWSEGMENT:
{
gboolean update;
gdouble rate, arate;
GstFormat format;
gint64 start, stop, time;
GstSegment *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_parse_new_segment_full (event, &update, &rate, &arate, &format,
&start, &stop, &time);
GST_DEBUG_OBJECT (rtpsession,
"configured NEWSEGMENT update %d, rate %lf, applied rate %lf, "
"format GST_FORMAT_TIME, "
"%" GST_TIME_FORMAT " -- %" GST_TIME_FORMAT
", time %" GST_TIME_FORMAT ", accum %" GST_TIME_FORMAT,
update, rate, arate, GST_TIME_ARGS (segment->start),
GST_TIME_ARGS (segment->stop), GST_TIME_ARGS (segment->time),
GST_TIME_ARGS (segment->accum));
gst_segment_set_newsegment_full (segment, update, rate,
arate, format, start, stop, time);
/* push event forward */
ret = gst_pad_push_event (rtpsession->send_rtp_src, event);
break;
}
case GST_EVENT_EOS:
ret = gst_pad_push_event (rtpsession->send_rtp_src, event);
rtp_session_send_bye (rtpsession->priv->session, "End of stream");
break;
default:
ret = gst_pad_push_event (rtpsession->send_rtp_src, event);
break;
}
gst_object_unref (rtpsession);
return ret;
}
static GstCaps *
gst_rtp_session_getcaps_send_rtp (GstPad * pad)
{
GstRtpSession *rtpsession;
GstRtpSessionPrivate *priv;
GstCaps *result;
GstStructure *s1, *s2;
rtpsession = GST_RTP_SESSION (gst_pad_get_parent (pad));
priv = rtpsession->priv;
/* we can basically accept anything but we prefer to receive packets with our
* internal SSRC so that we don't have to patch it. Create a structure with
* the SSRC and another one without. */
s1 = gst_structure_new ("application/x-rtp",
"ssrc", G_TYPE_UINT, priv->session->source->ssrc, NULL);
s2 = gst_structure_new ("application/x-rtp", NULL);
result = gst_caps_new_full (s1, s2, NULL);
GST_DEBUG_OBJECT (rtpsession, "getting caps %" GST_PTR_FORMAT, result);
gst_object_unref (rtpsession);
return result;
}
/* 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;
GstClockTime timestamp;
guint64 ntpnstime;
rtpsession = GST_RTP_SESSION (gst_pad_get_parent (pad));
priv = rtpsession->priv;
GST_DEBUG_OBJECT (rtpsession, "received RTP packet");
/* get NTP time when this packet was captured, this depends on the timestamp. */
timestamp = GST_BUFFER_TIMESTAMP (buffer);
if (GST_CLOCK_TIME_IS_VALID (timestamp)) {
/* convert to running time using the segment start value. */
ntpnstime =
gst_segment_to_running_time (&rtpsession->send_rtp_seg, GST_FORMAT_TIME,
timestamp);
/* convert to NTP time by adding the NTP base */
ntpnstime += priv->ntpnsbase;
} else {
/* no timestamp, we could take the current running_time and convert it to
* NTP time. */
ntpnstime = -1;
}
ret = rtp_session_send_rtp (priv->session, buffer, ntpnstime);
if (ret != GST_FLOW_OK)
goto push_error;
done:
gst_object_unref (rtpsession);
return ret;
/* ERRORS */
push_error:
{
GST_DEBUG_OBJECT (rtpsession, "process returned %s",
gst_flow_get_name (ret));
goto done;
}
}
/* 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,
(GstPadEventFunction) gst_rtp_session_event_recv_rtp_sink);
gst_pad_set_setcaps_function (rtpsession->recv_rtp_sink,
gst_rtp_session_sink_setcaps);
gst_pad_set_internal_link_function (rtpsession->recv_rtp_sink,
gst_rtp_session_internal_links);
gst_pad_set_active (rtpsession->recv_rtp_sink, TRUE);
gst_element_add_pad (GST_ELEMENT_CAST (rtpsession),
rtpsession->recv_rtp_sink);
GST_DEBUG_OBJECT (rtpsession, "creating RTP src pad");
rtpsession->recv_rtp_src =
gst_pad_new_from_static_template (&rtpsession_recv_rtp_src_template,
"recv_rtp_src");
gst_pad_set_internal_link_function (rtpsession->recv_rtp_src,
gst_rtp_session_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;
}
/* 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,
(GstPadEventFunction) gst_rtp_session_event_recv_rtcp_sink);
gst_pad_set_internal_link_function (rtpsession->recv_rtcp_sink,
gst_rtp_session_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_internal_link_function (rtpsession->sync_src,
gst_rtp_session_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;
}
/* 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_getcaps_function (rtpsession->send_rtp_sink,
gst_rtp_session_getcaps_send_rtp);
gst_pad_set_event_function (rtpsession->send_rtp_sink,
(GstPadEventFunction) gst_rtp_session_event_send_rtp_sink);
gst_pad_set_internal_link_function (rtpsession->send_rtp_sink,
gst_rtp_session_internal_links);
gst_pad_set_active (rtpsession->send_rtp_sink, TRUE);
gst_element_add_pad (GST_ELEMENT_CAST (rtpsession),
rtpsession->send_rtp_sink);
rtpsession->send_rtp_src =
gst_pad_new_from_static_template (&rtpsession_send_rtp_src_template,
"send_rtp_src");
gst_pad_set_internal_link_function (rtpsession->send_rtp_src,
gst_rtp_session_internal_links);
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_use_fixed_caps (rtpsession->send_rtcp_src);
gst_pad_set_active (rtpsession->send_rtcp_src, TRUE);
gst_pad_set_internal_link_function (rtpsession->send_rtcp_src,
gst_rtp_session_internal_links);
gst_pad_set_query_function (rtpsession->send_rtcp_src,
gst_rtp_session_query_send_rtcp_src);
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)
{
}
Reference in a new issue View git blame Copy permalink