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85e26f6546
Original commit message from CVS: * gst/rtpmanager/gstrtpbin.c: (on_sender_timeout), (create_session), (gst_rtp_bin_associate), (gst_rtp_bin_sync_chain), (gst_rtp_bin_class_init), (gst_rtp_bin_request_new_pad): * gst/rtpmanager/gstrtpbin.h: Add signal to notify listeners when a sender becomes a receiver. Tweak lip-sync code, don't store our own copy of the ts-offset of the jitterbuffer, don't adjust sync if the change is less than 4msec. Get the RTP timestamp <-> GStreamer timestamp relation directly from the jitterbuffer instead of our inaccurate version from the source. * gst/rtpmanager/gstrtpjitterbuffer.c: (gst_rtp_jitter_buffer_chain), (gst_rtp_jitter_buffer_loop), (gst_rtp_jitter_buffer_get_sync): * gst/rtpmanager/gstrtpjitterbuffer.h: Add G_LIKELY macros, use global defines for max packet reorder and dropouts. Reset the jitterbuffer clock skew detection when packets seqnums are changed unexpectedly. * gst/rtpmanager/gstrtpsession.c: (on_sender_timeout), (gst_rtp_session_class_init), (gst_rtp_session_init): * gst/rtpmanager/gstrtpsession.h: Add sender timeout signal. * gst/rtpmanager/rtpjitterbuffer.c: (rtp_jitter_buffer_reset_skew), (calculate_skew), (rtp_jitter_buffer_insert), (rtp_jitter_buffer_get_sync): * gst/rtpmanager/rtpjitterbuffer.h: Add some G_LIKELY macros. Keep track of the extended RTP timestamp so that we can report the RTP timestamp <-> GStreamer timestamp relation for lip-sync. Remove server timestamp gap detection code, the server can sometimes make a huge gap in timestamps (talk spurts,...) see #549774. Detect timetamp weirdness instead by observing the sender/receiver timestamp relation and resync if it changes more than 1 second. Add method to report about the current rtp <-> gst timestamp relation which is needed for lip-sync. * gst/rtpmanager/rtpsession.c: (rtp_session_class_init), (on_sender_timeout), (check_collision), (rtp_session_process_sr), (session_cleanup): * gst/rtpmanager/rtpsession.h: Add sender timeout signal. Remove inaccurate rtp <-> gst timestamp relation code, the jitterbuffer can now do an accurate reporting about this. * gst/rtpmanager/rtpsource.c: (rtp_source_init), (rtp_source_update_caps), (calculate_jitter), (rtp_source_process_rtp): * gst/rtpmanager/rtpsource.h: Remove inaccurate rtp <-> gst timestamp relation code. * gst/rtpmanager/rtpstats.h: Define global max-reorder and max-dropout constants for use in various subsystems.
1888 lines
60 KiB
C
1888 lines
60 KiB
C
/* GStreamer
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* Copyright (C) <2007> Wim Taymans <wim.taymans@gmail.com>
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*
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* This library is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Library General Public
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* License as published by the Free Software Foundation; either
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* version 2 of the License, or (at your option) any later version.
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*
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* This library is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Library General Public License for more details.
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*
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* You should have received a copy of the GNU Library General Public
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* License along with this library; if not, write to the
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* Free Software Foundation, Inc., 59 Temple Place - Suite 330,
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* Boston, MA 02111-1307, USA.
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*/
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/**
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* SECTION:element-gstrtpsession
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* @see_also: gstrtpjitterbuffer, gstrtpbin, gstrtpptdemux, gstrtpssrcdemux
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*
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* The RTP session manager models one participant with a unique SSRC in an RTP
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* session. This session can be used to send and receive RTP and RTCP packets.
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* Based on what REQUEST pads are requested from the session manager, specific
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* functionality can be activated.
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*
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* The session manager currently implements RFC 3550 including:
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* <itemizedlist>
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* <listitem>
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* <para>RTP packet validation based on consecutive sequence numbers.</para>
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* </listitem>
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* <listitem>
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* <para>Maintainance of the SSRC participant database.</para>
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* </listitem>
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* <listitem>
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* <para>Keeping per participant statistics based on received RTCP packets.</para>
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* </listitem>
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* <listitem>
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* <para>Scheduling of RR/SR RTCP packets.</para>
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* </listitem>
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* </itemizedlist>
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*
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* The gstrtpsession will not demux packets based on SSRC or payload type, nor will
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* it correct for packet reordering and jitter. Use #GstRtpsSrcDemux,
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* #GstRtpPtDemux and GstRtpJitterBuffer in addition to #GstRtpSession to
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* perform these tasks. It is usually a good idea to use #GstRtpBin, which
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* combines all these features in one element.
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*
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* To use #GstRtpSession as an RTP receiver, request a recv_rtp_sink pad, which will
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* automatically create recv_rtp_src pad. Data received on the recv_rtp_sink pad
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* will be processed in the session and after being validated forwarded on the
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* recv_rtp_src pad.
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*
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* To also use #GstRtpSession as an RTCP receiver, request a recv_rtcp_sink pad,
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* which will automatically create a sync_src pad. Packets received on the RTCP
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* pad will be used by the session manager to update the stats and database of
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* the other participants. SR packets will be forwarded on the sync_src pad
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* so that they can be used to perform inter-stream synchronisation when needed.
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*
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* If you want the session manager to generate and send RTCP packets, request
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* the send_rtcp_src pad. Packet pushed on this pad contain SR/RR RTCP reports
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* that should be sent to all participants in the session.
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*
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* To use #GstRtpSession as a sender, request a send_rtp_sink pad, which will
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* automatically create a send_rtp_src pad. The session manager will modify the
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* SSRC in the RTP packets to its own SSRC and wil forward the packets on the
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* send_rtp_src pad after updating its internal state.
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*
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* The session manager needs the clock-rate of the payload types it is handling
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* and will signal the #GstRtpSession::request-pt-map signal when it needs such a
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* mapping. One can clear the cached values with the #GstRtpSession::clear-pt-map
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* signal.
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*
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* <refsect2>
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* <title>Example pipelines</title>
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* |[
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* gst-launch udpsrc port=5000 caps="application/x-rtp, ..." ! .recv_rtp_sink gstrtpsession .recv_rtp_src ! rtptheoradepay ! theoradec ! xvimagesink
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* ]| Receive theora RTP packets from port 5000 and send them to the depayloader,
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* decoder and display. Note that the application/x-rtp caps on udpsrc should be
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* configured based on some negotiation process such as RTSP for this pipeline
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* to work correctly.
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* |[
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* gst-launch udpsrc port=5000 caps="application/x-rtp, ..." ! .recv_rtp_sink gstrtpsession name=session \
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* .recv_rtp_src ! rtptheoradepay ! theoradec ! xvimagesink \
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* udpsrc port=5001 caps="application/x-rtcp" ! session.recv_rtcp_sink
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* ]| Receive theora RTP packets from port 5000 and send them to the depayloader,
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* decoder and display. Receive RTCP packets from port 5001 and process them in
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* the session manager.
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* Note that the application/x-rtp caps on udpsrc should be
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* configured based on some negotiation process such as RTSP for this pipeline
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* to work correctly.
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* |[
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* gst-launch videotestsrc ! theoraenc ! rtptheorapay ! .send_rtp_sink gstrtpsession .send_rtp_src ! udpsink port=5000
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* ]| Send theora RTP packets through the session manager and out on UDP port
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* 5000.
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* |[
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* gst-launch videotestsrc ! theoraenc ! rtptheorapay ! .send_rtp_sink gstrtpsession name=session .send_rtp_src \
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* ! udpsink port=5000 session.send_rtcp_src ! udpsink port=5001
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* ]| Send theora RTP packets through the session manager and out on UDP port
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* 5000. Send RTCP packets on port 5001. Note that this pipeline will not preroll
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* correctly because the second udpsink will not preroll correctly (no RTCP
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* packets are sent in the PAUSED state). Applications should manually set and
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* keep (see gst_element_set_locked_state()) the RTCP udpsink to the PLAYING state.
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* </refsect2>
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*
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* Last reviewed on 2007-05-28 (0.10.5)
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*/
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#ifdef HAVE_CONFIG_H
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#include "config.h"
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#endif
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#include <gst/rtp/gstrtpbuffer.h>
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#include "gstrtpbin-marshal.h"
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#include "gstrtpsession.h"
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#include "rtpsession.h"
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GST_DEBUG_CATEGORY_STATIC (gst_rtp_session_debug);
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#define GST_CAT_DEFAULT gst_rtp_session_debug
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/* elementfactory information */
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static const GstElementDetails rtpsession_details =
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GST_ELEMENT_DETAILS ("RTP Session",
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"Filter/Network/RTP",
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"Implement an RTP session",
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"Wim Taymans <wim.taymans@gmail.com>");
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/* sink pads */
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static GstStaticPadTemplate rtpsession_recv_rtp_sink_template =
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GST_STATIC_PAD_TEMPLATE ("recv_rtp_sink",
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GST_PAD_SINK,
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GST_PAD_REQUEST,
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GST_STATIC_CAPS ("application/x-rtp")
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);
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static GstStaticPadTemplate rtpsession_recv_rtcp_sink_template =
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GST_STATIC_PAD_TEMPLATE ("recv_rtcp_sink",
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GST_PAD_SINK,
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GST_PAD_REQUEST,
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GST_STATIC_CAPS ("application/x-rtcp")
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);
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static GstStaticPadTemplate rtpsession_send_rtp_sink_template =
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GST_STATIC_PAD_TEMPLATE ("send_rtp_sink",
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GST_PAD_SINK,
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GST_PAD_REQUEST,
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GST_STATIC_CAPS ("application/x-rtp")
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);
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/* src pads */
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static GstStaticPadTemplate rtpsession_recv_rtp_src_template =
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GST_STATIC_PAD_TEMPLATE ("recv_rtp_src",
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GST_PAD_SRC,
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GST_PAD_SOMETIMES,
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GST_STATIC_CAPS ("application/x-rtp")
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);
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static GstStaticPadTemplate rtpsession_sync_src_template =
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GST_STATIC_PAD_TEMPLATE ("sync_src",
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GST_PAD_SRC,
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GST_PAD_SOMETIMES,
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GST_STATIC_CAPS ("application/x-rtcp")
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);
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static GstStaticPadTemplate rtpsession_send_rtp_src_template =
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GST_STATIC_PAD_TEMPLATE ("send_rtp_src",
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GST_PAD_SRC,
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GST_PAD_SOMETIMES,
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GST_STATIC_CAPS ("application/x-rtp")
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);
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static GstStaticPadTemplate rtpsession_send_rtcp_src_template =
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GST_STATIC_PAD_TEMPLATE ("send_rtcp_src",
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GST_PAD_SRC,
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GST_PAD_REQUEST,
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GST_STATIC_CAPS ("application/x-rtcp")
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);
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/* signals and args */
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enum
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{
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SIGNAL_REQUEST_PT_MAP,
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SIGNAL_CLEAR_PT_MAP,
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SIGNAL_ON_NEW_SSRC,
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SIGNAL_ON_SSRC_COLLISION,
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SIGNAL_ON_SSRC_VALIDATED,
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SIGNAL_ON_SSRC_ACTIVE,
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SIGNAL_ON_SSRC_SDES,
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SIGNAL_ON_BYE_SSRC,
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SIGNAL_ON_BYE_TIMEOUT,
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SIGNAL_ON_TIMEOUT,
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SIGNAL_ON_SENDER_TIMEOUT,
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LAST_SIGNAL
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};
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#define DEFAULT_NTP_NS_BASE 0
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#define DEFAULT_BANDWIDTH RTP_STATS_BANDWIDTH
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#define DEFAULT_RTCP_FRACTION RTP_STATS_RTCP_BANDWIDTH
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#define DEFAULT_SDES_CNAME NULL
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#define DEFAULT_SDES_NAME NULL
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#define DEFAULT_SDES_EMAIL NULL
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#define DEFAULT_SDES_PHONE NULL
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#define DEFAULT_SDES_LOCATION NULL
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#define DEFAULT_SDES_TOOL NULL
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#define DEFAULT_SDES_NOTE NULL
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#define DEFAULT_NUM_SOURCES 0
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#define DEFAULT_NUM_ACTIVE_SOURCES 0
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enum
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{
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PROP_0,
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PROP_NTP_NS_BASE,
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PROP_BANDWIDTH,
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PROP_RTCP_FRACTION,
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PROP_SDES_CNAME,
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PROP_SDES_NAME,
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PROP_SDES_EMAIL,
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PROP_SDES_PHONE,
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PROP_SDES_LOCATION,
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PROP_SDES_TOOL,
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PROP_SDES_NOTE,
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PROP_NUM_SOURCES,
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PROP_NUM_ACTIVE_SOURCES,
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PROP_LAST
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};
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#define GST_RTP_SESSION_GET_PRIVATE(obj) \
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(G_TYPE_INSTANCE_GET_PRIVATE ((obj), GST_TYPE_RTP_SESSION, GstRtpSessionPrivate))
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#define GST_RTP_SESSION_LOCK(sess) g_mutex_lock ((sess)->priv->lock)
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#define GST_RTP_SESSION_UNLOCK(sess) g_mutex_unlock ((sess)->priv->lock)
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struct _GstRtpSessionPrivate
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{
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GMutex *lock;
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GstClock *sysclock;
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RTPSession *session;
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/* thread for sending out RTCP */
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GstClockID id;
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gboolean stop_thread;
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GThread *thread;
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gboolean thread_stopped;
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/* caps mapping */
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GHashTable *ptmap;
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/* NTP base time */
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guint64 ntpnsbase;
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};
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/* callbacks to handle actions from the session manager */
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static GstFlowReturn gst_rtp_session_process_rtp (RTPSession * sess,
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RTPSource * src, GstBuffer * buffer, gpointer user_data);
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static GstFlowReturn gst_rtp_session_send_rtp (RTPSession * sess,
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RTPSource * src, GstBuffer * buffer, gpointer user_data);
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static GstFlowReturn gst_rtp_session_send_rtcp (RTPSession * sess,
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RTPSource * src, GstBuffer * buffer, gboolean eos, gpointer user_data);
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static GstFlowReturn gst_rtp_session_sync_rtcp (RTPSession * sess,
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RTPSource * src, GstBuffer * buffer, gpointer user_data);
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static gint gst_rtp_session_clock_rate (RTPSession * sess, guint8 payload,
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gpointer user_data);
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static void gst_rtp_session_reconsider (RTPSession * sess, gpointer user_data);
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static RTPSessionCallbacks callbacks = {
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gst_rtp_session_process_rtp,
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gst_rtp_session_send_rtp,
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gst_rtp_session_sync_rtcp,
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gst_rtp_session_send_rtcp,
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gst_rtp_session_clock_rate,
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gst_rtp_session_reconsider
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};
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/* GObject vmethods */
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static void gst_rtp_session_finalize (GObject * object);
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static void gst_rtp_session_set_property (GObject * object, guint prop_id,
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const GValue * value, GParamSpec * pspec);
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static void gst_rtp_session_get_property (GObject * object, guint prop_id,
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GValue * value, GParamSpec * pspec);
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/* GstElement vmethods */
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static GstStateChangeReturn gst_rtp_session_change_state (GstElement * element,
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GstStateChange transition);
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static GstPad *gst_rtp_session_request_new_pad (GstElement * element,
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GstPadTemplate * templ, const gchar * name);
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static void gst_rtp_session_release_pad (GstElement * element, GstPad * pad);
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static void gst_rtp_session_clear_pt_map (GstRtpSession * rtpsession);
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static guint gst_rtp_session_signals[LAST_SIGNAL] = { 0 };
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static void
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on_new_ssrc (RTPSession * session, RTPSource * src, GstRtpSession * sess)
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{
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g_signal_emit (sess, gst_rtp_session_signals[SIGNAL_ON_NEW_SSRC], 0,
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src->ssrc);
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}
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static void
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on_ssrc_collision (RTPSession * session, RTPSource * src, GstRtpSession * sess)
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{
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g_signal_emit (sess, gst_rtp_session_signals[SIGNAL_ON_SSRC_COLLISION], 0,
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src->ssrc);
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}
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static void
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on_ssrc_validated (RTPSession * session, RTPSource * src, GstRtpSession * sess)
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{
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g_signal_emit (sess, gst_rtp_session_signals[SIGNAL_ON_SSRC_VALIDATED], 0,
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src->ssrc);
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}
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static void
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on_ssrc_active (RTPSession * session, RTPSource * src, GstRtpSession * sess)
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{
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g_signal_emit (sess, gst_rtp_session_signals[SIGNAL_ON_SSRC_ACTIVE], 0,
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src->ssrc);
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}
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static GstStructure *
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source_get_sdes_structure (RTPSource * src)
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{
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GstStructure *result;
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GValue val = { 0 };
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gchar *str;
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result = gst_structure_empty_new ("GstRTPSessionSDES");
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gst_structure_set (result, "ssrc", G_TYPE_UINT, src->ssrc, NULL);
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g_value_init (&val, G_TYPE_STRING);
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str = rtp_source_get_sdes_string (src, GST_RTCP_SDES_CNAME);
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if (str) {
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g_value_take_string (&val, str);
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gst_structure_set_value (result, "cname", &val);
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}
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str = rtp_source_get_sdes_string (src, GST_RTCP_SDES_NAME);
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if (str) {
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g_value_take_string (&val, str);
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gst_structure_set_value (result, "name", &val);
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}
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str = rtp_source_get_sdes_string (src, GST_RTCP_SDES_EMAIL);
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if (str) {
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g_value_take_string (&val, str);
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gst_structure_set_value (result, "email", &val);
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}
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str = rtp_source_get_sdes_string (src, GST_RTCP_SDES_PHONE);
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if (str) {
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g_value_take_string (&val, str);
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gst_structure_set_value (result, "phone", &val);
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}
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str = rtp_source_get_sdes_string (src, GST_RTCP_SDES_LOC);
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if (str) {
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g_value_take_string (&val, str);
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gst_structure_set_value (result, "location", &val);
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}
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str = rtp_source_get_sdes_string (src, GST_RTCP_SDES_TOOL);
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if (str) {
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g_value_take_string (&val, str);
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gst_structure_set_value (result, "tool", &val);
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}
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str = rtp_source_get_sdes_string (src, GST_RTCP_SDES_NOTE);
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if (str) {
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g_value_take_string (&val, str);
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gst_structure_set_value (result, "note", &val);
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}
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str = rtp_source_get_sdes_string (src, GST_RTCP_SDES_PRIV);
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if (str) {
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g_value_take_string (&val, str);
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gst_structure_set_value (result, "priv", &val);
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}
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g_value_unset (&val);
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return result;
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}
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static void
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on_ssrc_sdes (RTPSession * session, RTPSource * src, GstRtpSession * sess)
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{
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GstStructure *s;
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GstMessage *m;
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/* convert the new SDES info into a message */
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RTP_SESSION_LOCK (session);
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s = source_get_sdes_structure (src);
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RTP_SESSION_UNLOCK (session);
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m = gst_message_new_custom (GST_MESSAGE_ELEMENT, GST_OBJECT (sess), s);
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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);
|
|
}
|
|
|
|
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);
|
|
/**
|
|
* GstRtpSession::on-sender-timeout:
|
|
* @sess: the object which received the signal
|
|
* @ssrc: the SSRC
|
|
*
|
|
* Notify of a sender SSRC that has timed out and became a receiver
|
|
*/
|
|
gst_rtp_session_signals[SIGNAL_ON_SENDER_TIMEOUT] =
|
|
g_signal_new ("on-sender-timeout", G_TYPE_FROM_CLASS (klass),
|
|
G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRtpSessionClass,
|
|
on_sender_timeout), NULL, NULL, g_cclosure_marshal_VOID__UINT,
|
|
G_TYPE_NONE, 1, G_TYPE_UINT);
|
|
|
|
g_object_class_install_property (gobject_class, PROP_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->sysclock = gst_system_clock_obtain ();
|
|
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);
|
|
g_signal_connect (rtpsession->priv->session, "on-sender-timeout",
|
|
(GCallback) on_sender_timeout, rtpsession);
|
|
rtpsession->priv->ptmap = g_hash_table_new_full (NULL, NULL, NULL,
|
|
(GDestroyNotify) gst_caps_unref);
|
|
|
|
gst_segment_init (&rtpsession->recv_rtp_seg, GST_FORMAT_UNDEFINED);
|
|
gst_segment_init (&rtpsession->send_rtp_seg, GST_FORMAT_UNDEFINED);
|
|
|
|
rtpsession->priv->thread_stopped = TRUE;
|
|
}
|
|
|
|
static void
|
|
gst_rtp_session_finalize (GObject * object)
|
|
{
|
|
GstRtpSession *rtpsession;
|
|
|
|
rtpsession = GST_RTP_SESSION (object);
|
|
|
|
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->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_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)
|
|
{
|
|
GstClockID id;
|
|
GstClockTime current_time;
|
|
GstClockTime next_timeout;
|
|
guint64 ntpnstime;
|
|
|
|
GST_DEBUG_OBJECT (rtpsession, "entering RTCP thread");
|
|
|
|
GST_RTP_SESSION_LOCK (rtpsession);
|
|
|
|
current_time = gst_clock_get_time (rtpsession->priv->sysclock);
|
|
|
|
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 (rtpsession->priv->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 (rtpsession->priv->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_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) {
|
|
/* 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_LOG_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_LOG_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. The eos flag is set when an EOS event should be sent downstream as
|
|
* well. */
|
|
static GstFlowReturn
|
|
gst_rtp_session_send_rtcp (RTPSession * sess, RTPSource * src,
|
|
GstBuffer * buffer, gboolean eos, gpointer user_data)
|
|
{
|
|
GstFlowReturn result;
|
|
GstRtpSession *rtpsession;
|
|
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_LOG_OBJECT (rtpsession, "sending RTCP");
|
|
result = gst_pad_push (rtpsession->send_rtcp_src, buffer);
|
|
|
|
/* we have to send EOS after this packet */
|
|
if (eos) {
|
|
GST_LOG_OBJECT (rtpsession, "sending EOS");
|
|
gst_pad_push_event (rtpsession->send_rtcp_src, gst_event_new_eos ());
|
|
}
|
|
} 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_LOG_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;
|
|
GstClockTime current_time;
|
|
guint64 ntpnstime;
|
|
GstClockTime timestamp;
|
|
|
|
rtpsession = GST_RTP_SESSION (gst_pad_get_parent (pad));
|
|
priv = rtpsession->priv;
|
|
|
|
GST_LOG_OBJECT (rtpsession, "received RTP packet");
|
|
|
|
/* get NTP time when this packet was captured, this depends on the timestamp. */
|
|
timestamp = GST_BUFFER_TIMESTAMP (buffer);
|
|
if (GST_CLOCK_TIME_IS_VALID (timestamp)) {
|
|
/* convert to running time using the segment values */
|
|
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);
|
|
}
|
|
|
|
current_time = gst_clock_get_time (priv->sysclock);
|
|
ret = rtp_session_process_rtp (priv->session, buffer, current_time,
|
|
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;
|
|
GstClockTime current_time;
|
|
GstFlowReturn ret;
|
|
|
|
rtpsession = GST_RTP_SESSION (gst_pad_get_parent (pad));
|
|
priv = rtpsession->priv;
|
|
|
|
GST_LOG_OBJECT (rtpsession, "received RTCP packet");
|
|
|
|
current_time = gst_clock_get_time (priv->sysclock);
|
|
ret = rtp_session_process_rtcp (priv->session, buffer, current_time);
|
|
|
|
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:{
|
|
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_send_bye (rtpsession->priv->session, "End of stream",
|
|
current_time);
|
|
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;
|
|
GstClockTime current_time;
|
|
guint64 ntpnstime;
|
|
|
|
rtpsession = GST_RTP_SESSION (gst_pad_get_parent (pad));
|
|
priv = rtpsession->priv;
|
|
|
|
GST_LOG_OBJECT (rtpsession, "received RTP packet");
|
|
|
|
/* get NTP time when this packet was captured, this depends on the timestamp. */
|
|
timestamp = GST_BUFFER_TIMESTAMP (buffer);
|
|
if (GST_CLOCK_TIME_IS_VALID (timestamp)) {
|
|
/* convert to running time using the segment 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;
|
|
}
|
|
|
|
current_time = gst_clock_get_time (priv->sysclock);
|
|
ret = rtp_session_send_rtp (priv->session, buffer, current_time, 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)
|
|
{
|
|
}
|
|
|
|
void
|
|
gst_rtp_session_set_ssrc (GstRtpSession * sess, guint32 ssrc)
|
|
{
|
|
rtp_session_set_internal_ssrc (sess->priv->session, ssrc);
|
|
}
|