mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer.git
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308af145ed
This is difficult to encounter in ordinary networks, but is encountered when using tc-netem to add random delays to packets, and also when your UDP stream is bonded over multiple links with varying characteristics. Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/1979>
2906 lines
92 KiB
C
2906 lines
92 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., 51 Franklin St, Fifth Floor,
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* Boston, MA 02110-1301, USA.
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*/
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/**
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* SECTION:element-rtpsession
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* @title: rtpsession
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* @see_also: rtpjitterbuffer, rtpbin, rtpptdemux, rtpssrcdemux
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*
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* The RTP session manager models participants with 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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*
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* * RTP packet validation based on consecutive sequence numbers.
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*
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* * Maintenance of the SSRC participant database.
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*
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* * Keeping per participant statistics based on received RTCP packets.
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*
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* * Scheduling of RR/SR RTCP packets.
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*
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* * Support for multiple sender SSRC.
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*
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* The rtpsession 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
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* forward the packets on the 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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* ## Example pipelines
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* |[
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* gst-launch-1.0 udpsrc port=5000 caps="application/x-rtp, ..." ! .recv_rtp_sink rtpsession .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-1.0 udpsrc port=5000 caps="application/x-rtp, ..." ! .recv_rtp_sink rtpsession 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-1.0 videotestsrc ! theoraenc ! rtptheorapay ! .send_rtp_sink rtpsession .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-1.0 videotestsrc ! theoraenc ! rtptheorapay ! .send_rtp_sink rtpsession 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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*
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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 <gst/glib-compat-private.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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#define GST_TYPE_RTP_NTP_TIME_SOURCE (gst_rtp_ntp_time_source_get_type ())
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GType
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gst_rtp_ntp_time_source_get_type (void)
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{
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static GType type = 0;
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static const GEnumValue values[] = {
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{GST_RTP_NTP_TIME_SOURCE_NTP, "NTP time based on realtime clock", "ntp"},
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{GST_RTP_NTP_TIME_SOURCE_UNIX, "UNIX time based on realtime clock", "unix"},
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{GST_RTP_NTP_TIME_SOURCE_RUNNING_TIME,
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"Running time based on pipeline clock",
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"running-time"},
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{GST_RTP_NTP_TIME_SOURCE_CLOCK_TIME, "Pipeline clock time", "clock-time"},
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{0, NULL, NULL},
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};
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if (!type) {
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type = g_enum_register_static ("GstRtpNtpTimeSource", values);
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}
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return type;
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}
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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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SIGNAL_ON_NEW_SENDER_SSRC,
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SIGNAL_ON_SENDER_SSRC_ACTIVE,
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LAST_SIGNAL
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};
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#define DEFAULT_BANDWIDTH 0
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#define DEFAULT_RTCP_FRACTION RTP_STATS_RTCP_FRACTION
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#define DEFAULT_RTCP_RR_BANDWIDTH -1
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#define DEFAULT_RTCP_RS_BANDWIDTH -1
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#define DEFAULT_SDES NULL
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#define DEFAULT_NUM_SOURCES 0
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#define DEFAULT_NUM_ACTIVE_SOURCES 0
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#define DEFAULT_USE_PIPELINE_CLOCK FALSE
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#define DEFAULT_RTCP_MIN_INTERVAL (RTP_STATS_MIN_INTERVAL * GST_SECOND)
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#define DEFAULT_PROBATION RTP_DEFAULT_PROBATION
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#define DEFAULT_MAX_DROPOUT_TIME 60000
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#define DEFAULT_MAX_MISORDER_TIME 2000
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#define DEFAULT_RTP_PROFILE GST_RTP_PROFILE_AVP
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#define DEFAULT_NTP_TIME_SOURCE GST_RTP_NTP_TIME_SOURCE_NTP
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#define DEFAULT_RTCP_SYNC_SEND_TIME TRUE
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enum
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{
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PROP_0,
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PROP_BANDWIDTH,
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PROP_RTCP_FRACTION,
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PROP_RTCP_RR_BANDWIDTH,
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PROP_RTCP_RS_BANDWIDTH,
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PROP_SDES,
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PROP_NUM_SOURCES,
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PROP_NUM_ACTIVE_SOURCES,
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PROP_INTERNAL_SESSION,
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PROP_USE_PIPELINE_CLOCK,
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PROP_RTCP_MIN_INTERVAL,
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PROP_PROBATION,
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PROP_MAX_DROPOUT_TIME,
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PROP_MAX_MISORDER_TIME,
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PROP_STATS,
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PROP_TWCC_STATS,
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PROP_RTP_PROFILE,
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PROP_NTP_TIME_SOURCE,
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PROP_RTCP_SYNC_SEND_TIME
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};
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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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#define GST_RTP_SESSION_WAIT(sess) g_cond_wait (&(sess)->priv->cond, &(sess)->priv->lock)
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#define GST_RTP_SESSION_SIGNAL(sess) g_cond_signal (&(sess)->priv->cond)
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struct _GstRtpSessionPrivate
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{
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GMutex lock;
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GCond cond;
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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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gboolean wait_send;
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/* caps mapping */
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GHashTable *ptmap;
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GstClockTime send_latency;
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gboolean use_pipeline_clock;
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GstRtpNtpTimeSource ntp_time_source;
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gboolean rtcp_sync_send_time;
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guint recv_rtx_req_count;
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guint sent_rtx_req_count;
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GstStructure *last_twcc_stats;
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/*
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* This is the list of processed packets in the receive path when upstream
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* pushed a buffer list.
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*/
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GstBufferList *processed_list;
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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, gpointer data, 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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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 void gst_rtp_session_request_key_unit (RTPSession * sess, guint32 ssrc,
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gboolean all_headers, gpointer user_data);
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static GstClockTime gst_rtp_session_request_time (RTPSession * session,
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gpointer user_data);
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static void gst_rtp_session_notify_nack (RTPSession * sess,
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guint16 seqnum, guint16 blp, guint32 ssrc, gpointer user_data);
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static void gst_rtp_session_notify_twcc (RTPSession * sess,
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GstStructure * twcc_packets, GstStructure * twcc_stats, gpointer user_data);
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static void gst_rtp_session_reconfigure (RTPSession * sess, gpointer user_data);
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static void gst_rtp_session_notify_early_rtcp (RTPSession * sess,
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gpointer user_data);
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static GstFlowReturn gst_rtp_session_chain_recv_rtp (GstPad * pad,
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GstObject * parent, GstBuffer * buffer);
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static GstFlowReturn gst_rtp_session_chain_recv_rtp_list (GstPad * pad,
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GstObject * parent, GstBufferList * list);
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static GstFlowReturn gst_rtp_session_chain_recv_rtcp (GstPad * pad,
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GstObject * parent, GstBuffer * buffer);
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static GstFlowReturn gst_rtp_session_chain_send_rtp (GstPad * pad,
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GstObject * parent, GstBuffer * buffer);
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static GstFlowReturn gst_rtp_session_chain_send_rtp_list (GstPad * pad,
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GstObject * parent, GstBufferList * list);
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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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gst_rtp_session_request_key_unit,
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gst_rtp_session_request_time,
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gst_rtp_session_notify_nack,
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gst_rtp_session_notify_twcc,
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gst_rtp_session_reconfigure,
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gst_rtp_session_notify_early_rtcp
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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, const GstCaps * caps);
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static void gst_rtp_session_release_pad (GstElement * element, GstPad * pad);
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static gboolean gst_rtp_session_sink_setcaps (GstPad * pad,
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GstRtpSession * rtpsession, GstCaps * caps);
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static gboolean gst_rtp_session_setcaps_send_rtp (GstPad * pad,
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GstRtpSession * rtpsession, GstCaps * caps);
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static void gst_rtp_session_clear_pt_map (GstRtpSession * rtpsession);
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static GstStructure *gst_rtp_session_create_stats (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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GstPad *send_rtp_sink;
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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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GST_RTP_SESSION_LOCK (sess);
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if ((send_rtp_sink = sess->send_rtp_sink))
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gst_object_ref (send_rtp_sink);
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GST_RTP_SESSION_UNLOCK (sess);
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if (send_rtp_sink) {
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GstStructure *structure;
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GstEvent *event;
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RTPSource *internal_src;
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guint32 suggested_ssrc;
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structure = gst_structure_new ("GstRTPCollision", "ssrc", G_TYPE_UINT,
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(guint) src->ssrc, NULL);
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/* if there is no source using the suggested ssrc, most probably because
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* this ssrc has just collided, suggest upstream to use it */
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suggested_ssrc = rtp_session_suggest_ssrc (session, NULL);
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internal_src = rtp_session_get_source_by_ssrc (session, suggested_ssrc);
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if (!internal_src)
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gst_structure_set (structure, "suggested-ssrc", G_TYPE_UINT,
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(guint) suggested_ssrc, NULL);
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else
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g_object_unref (internal_src);
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event = gst_event_new_custom (GST_EVENT_CUSTOM_UPSTREAM, structure);
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gst_pad_push_event (send_rtp_sink, event);
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gst_object_unref (send_rtp_sink);
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}
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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
|
|
on_ssrc_active (RTPSession * session, RTPSource * src, GstRtpSession * sess)
|
|
{
|
|
g_signal_emit (sess, gst_rtp_session_signals[SIGNAL_ON_SSRC_ACTIVE], 0,
|
|
src->ssrc);
|
|
}
|
|
|
|
static void
|
|
on_ssrc_sdes (RTPSession * session, RTPSource * src, GstRtpSession * sess)
|
|
{
|
|
GstStructure *s;
|
|
GstMessage *m;
|
|
|
|
/* convert the new SDES info into a message */
|
|
RTP_SESSION_LOCK (session);
|
|
g_object_get (src, "sdes", &s, NULL);
|
|
RTP_SESSION_UNLOCK (session);
|
|
|
|
m = gst_message_new_custom (GST_MESSAGE_ELEMENT, GST_OBJECT (sess), s);
|
|
gst_element_post_message (GST_ELEMENT_CAST (sess), m);
|
|
|
|
g_signal_emit (sess, gst_rtp_session_signals[SIGNAL_ON_SSRC_SDES], 0,
|
|
src->ssrc);
|
|
}
|
|
|
|
static void
|
|
on_bye_ssrc (RTPSession * session, RTPSource * src, GstRtpSession * sess)
|
|
{
|
|
g_signal_emit (sess, gst_rtp_session_signals[SIGNAL_ON_BYE_SSRC], 0,
|
|
src->ssrc);
|
|
}
|
|
|
|
static void
|
|
on_bye_timeout (RTPSession * session, RTPSource * src, GstRtpSession * sess)
|
|
{
|
|
g_signal_emit (sess, gst_rtp_session_signals[SIGNAL_ON_BYE_TIMEOUT], 0,
|
|
src->ssrc);
|
|
}
|
|
|
|
static void
|
|
on_timeout (RTPSession * session, RTPSource * src, GstRtpSession * sess)
|
|
{
|
|
g_signal_emit (sess, gst_rtp_session_signals[SIGNAL_ON_TIMEOUT], 0,
|
|
src->ssrc);
|
|
}
|
|
|
|
static void
|
|
on_sender_timeout (RTPSession * session, RTPSource * src, GstRtpSession * sess)
|
|
{
|
|
g_signal_emit (sess, gst_rtp_session_signals[SIGNAL_ON_SENDER_TIMEOUT], 0,
|
|
src->ssrc);
|
|
}
|
|
|
|
static void
|
|
on_new_sender_ssrc (RTPSession * session, RTPSource * src, GstRtpSession * sess)
|
|
{
|
|
g_signal_emit (sess, gst_rtp_session_signals[SIGNAL_ON_NEW_SENDER_SSRC], 0,
|
|
src->ssrc);
|
|
}
|
|
|
|
static void
|
|
on_sender_ssrc_active (RTPSession * session, RTPSource * src,
|
|
GstRtpSession * sess)
|
|
{
|
|
g_signal_emit (sess, gst_rtp_session_signals[SIGNAL_ON_SENDER_SSRC_ACTIVE], 0,
|
|
src->ssrc);
|
|
}
|
|
|
|
static void
|
|
on_notify_stats (RTPSession * session, GParamSpec * spec,
|
|
GstRtpSession * rtpsession)
|
|
{
|
|
g_object_notify (G_OBJECT (rtpsession), "stats");
|
|
}
|
|
|
|
#define gst_rtp_session_parent_class parent_class
|
|
G_DEFINE_TYPE_WITH_PRIVATE (GstRtpSession, gst_rtp_session, GST_TYPE_ELEMENT);
|
|
GST_ELEMENT_REGISTER_DEFINE (rtpsession, "rtpsession", GST_RANK_NONE,
|
|
GST_TYPE_RTP_SESSION);
|
|
|
|
static void
|
|
gst_rtp_session_class_init (GstRtpSessionClass * klass)
|
|
{
|
|
GObjectClass *gobject_class;
|
|
GstElementClass *gstelement_class;
|
|
|
|
gobject_class = (GObjectClass *) klass;
|
|
gstelement_class = (GstElementClass *) klass;
|
|
|
|
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, NULL, 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_RUN_LAST | G_SIGNAL_ACTION,
|
|
G_STRUCT_OFFSET (GstRtpSessionClass, clear_pt_map),
|
|
NULL, NULL, NULL, 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, NULL, 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, NULL, 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, NULL, 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, NULL, 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, NULL, 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, NULL, 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, NULL, 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, NULL, 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, NULL, G_TYPE_NONE, 1, G_TYPE_UINT);
|
|
|
|
/**
|
|
* GstRtpSession::on-new-sender-ssrc:
|
|
* @sess: the object which received the signal
|
|
* @ssrc: the sender SSRC
|
|
*
|
|
* Notify of a new sender SSRC that entered @session.
|
|
*
|
|
* Since: 1.8
|
|
*/
|
|
gst_rtp_session_signals[SIGNAL_ON_NEW_SENDER_SSRC] =
|
|
g_signal_new ("on-new-sender-ssrc", G_TYPE_FROM_CLASS (klass),
|
|
G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRtpSessionClass, on_new_ssrc),
|
|
NULL, NULL, NULL, G_TYPE_NONE, 1, G_TYPE_UINT);
|
|
|
|
/**
|
|
* GstRtpSession::on-sender-ssrc-active:
|
|
* @sess: the object which received the signal
|
|
* @ssrc: the sender SSRC
|
|
*
|
|
* Notify of a sender SSRC that is active, i.e., sending RTCP.
|
|
*
|
|
* Since: 1.8
|
|
*/
|
|
gst_rtp_session_signals[SIGNAL_ON_SENDER_SSRC_ACTIVE] =
|
|
g_signal_new ("on-sender-ssrc-active", G_TYPE_FROM_CLASS (klass),
|
|
G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRtpSessionClass,
|
|
on_ssrc_active), NULL, NULL, NULL, G_TYPE_NONE, 1, G_TYPE_UINT);
|
|
|
|
g_object_class_install_property (gobject_class, PROP_BANDWIDTH,
|
|
g_param_spec_double ("bandwidth", "Bandwidth",
|
|
"The bandwidth of the session in bytes per second (0 for auto-discover)",
|
|
0.0, G_MAXDOUBLE, DEFAULT_BANDWIDTH,
|
|
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
|
|
|
|
g_object_class_install_property (gobject_class, PROP_RTCP_FRACTION,
|
|
g_param_spec_double ("rtcp-fraction", "RTCP Fraction",
|
|
"The RTCP bandwidth of the session in bytes per second "
|
|
"(or as a real fraction of the RTP bandwidth if < 1.0)",
|
|
0.0, G_MAXDOUBLE, DEFAULT_RTCP_FRACTION,
|
|
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
|
|
|
|
g_object_class_install_property (gobject_class, PROP_RTCP_RR_BANDWIDTH,
|
|
g_param_spec_int ("rtcp-rr-bandwidth", "RTCP RR bandwidth",
|
|
"The RTCP bandwidth used for receivers in bytes per second (-1 = default)",
|
|
-1, G_MAXINT, DEFAULT_RTCP_RR_BANDWIDTH,
|
|
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
|
|
|
|
g_object_class_install_property (gobject_class, PROP_RTCP_RS_BANDWIDTH,
|
|
g_param_spec_int ("rtcp-rs-bandwidth", "RTCP RS bandwidth",
|
|
"The RTCP bandwidth used for senders in bytes per second (-1 = default)",
|
|
-1, G_MAXINT, DEFAULT_RTCP_RS_BANDWIDTH,
|
|
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
|
|
|
|
g_object_class_install_property (gobject_class, PROP_SDES,
|
|
g_param_spec_boxed ("sdes", "SDES",
|
|
"The SDES items of this session",
|
|
GST_TYPE_STRUCTURE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS
|
|
| GST_PARAM_DOC_SHOW_DEFAULT));
|
|
|
|
g_object_class_install_property (gobject_class, PROP_NUM_SOURCES,
|
|
g_param_spec_uint ("num-sources", "Num Sources",
|
|
"The number of sources in the session", 0, G_MAXUINT,
|
|
DEFAULT_NUM_SOURCES, G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
|
|
|
|
g_object_class_install_property (gobject_class, PROP_NUM_ACTIVE_SOURCES,
|
|
g_param_spec_uint ("num-active-sources", "Num Active Sources",
|
|
"The number of active sources in the session", 0, G_MAXUINT,
|
|
DEFAULT_NUM_ACTIVE_SOURCES,
|
|
G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
|
|
|
|
g_object_class_install_property (gobject_class, PROP_INTERNAL_SESSION,
|
|
g_param_spec_object ("internal-session", "Internal Session",
|
|
"The internal RTPSession object", RTP_TYPE_SESSION,
|
|
G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
|
|
|
|
g_object_class_install_property (gobject_class, PROP_USE_PIPELINE_CLOCK,
|
|
g_param_spec_boolean ("use-pipeline-clock", "Use pipeline clock",
|
|
"Use the pipeline running-time to set the NTP time in the RTCP SR messages "
|
|
"(DEPRECATED: Use ntp-time-source property)",
|
|
DEFAULT_USE_PIPELINE_CLOCK,
|
|
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS | G_PARAM_DEPRECATED));
|
|
|
|
g_object_class_install_property (gobject_class, PROP_RTCP_MIN_INTERVAL,
|
|
g_param_spec_uint64 ("rtcp-min-interval", "Minimum RTCP interval",
|
|
"Minimum interval between Regular RTCP packet (in ns)",
|
|
0, G_MAXUINT64, DEFAULT_RTCP_MIN_INTERVAL,
|
|
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
|
|
|
|
g_object_class_install_property (gobject_class, PROP_PROBATION,
|
|
g_param_spec_uint ("probation", "Number of probations",
|
|
"Consecutive packet sequence numbers to accept the source",
|
|
0, G_MAXUINT, DEFAULT_PROBATION,
|
|
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
|
|
|
|
g_object_class_install_property (gobject_class, PROP_MAX_DROPOUT_TIME,
|
|
g_param_spec_uint ("max-dropout-time", "Max dropout time",
|
|
"The maximum time (milliseconds) of missing packets tolerated.",
|
|
0, G_MAXUINT, DEFAULT_MAX_DROPOUT_TIME,
|
|
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
|
|
|
|
g_object_class_install_property (gobject_class, PROP_MAX_MISORDER_TIME,
|
|
g_param_spec_uint ("max-misorder-time", "Max misorder time",
|
|
"The maximum time (milliseconds) of misordered packets tolerated.",
|
|
0, G_MAXUINT, DEFAULT_MAX_MISORDER_TIME,
|
|
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
|
|
|
|
/**
|
|
* GstRtpSession:stats:
|
|
*
|
|
* Various session statistics. This property returns a #GstStructure
|
|
* with name `application/x-rtp-session-stats` with the following fields:
|
|
*
|
|
* * "recv-rtx-req-count" G_TYPE_UINT The number of retransmission events
|
|
* received from downstream (in receiver mode) (Since 1.16)
|
|
* * "sent-rtx-req-count" G_TYPE_UINT The number of retransmission events
|
|
* sent downstream (in sender mode) (Since 1.16)
|
|
* * "rtx-count" G_TYPE_UINT DEPRECATED Since 1.16, same as
|
|
* "recv-rtx-req-count".
|
|
* * "rtx-drop-count" G_TYPE_UINT The number of retransmission events
|
|
* dropped (due to bandwidth constraints)
|
|
* * "sent-nack-count" G_TYPE_UINT Number of NACKs sent
|
|
* * "recv-nack-count" G_TYPE_UINT Number of NACKs received
|
|
* * "source-stats" G_TYPE_BOXED GValueArray of #RTPSource:stats for all
|
|
* RTP sources (Since 1.8)
|
|
*
|
|
* Since: 1.4
|
|
*/
|
|
g_object_class_install_property (gobject_class, PROP_STATS,
|
|
g_param_spec_boxed ("stats", "Statistics",
|
|
"Various statistics", GST_TYPE_STRUCTURE,
|
|
G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
|
|
|
|
/**
|
|
* GstRtpSession:twcc-stats:
|
|
*
|
|
* Various statistics derived from TWCC. This property returns a GstStructure
|
|
* with name RTPTWCCStats with the following fields:
|
|
*
|
|
* "bitrate-sent" G_TYPE_UINT The actual sent bitrate of TWCC packets
|
|
* "bitrate-recv" G_TYPE_UINT The estimated bitrate for the receiver.
|
|
* "packets-sent" G_TYPE_UINT Number of packets sent
|
|
* "packets-recv" G_TYPE_UINT Number of packets reported recevied
|
|
* "packet-loss-pct" G_TYPE_DOUBLE Packetloss percentage, based on
|
|
* packets reported as lost from the receiver. Note: depending on the
|
|
* implementation of the receiver and due to the nature of the TWCC
|
|
* RRs being sent with high frequency, out of order packets may not
|
|
* be fully accounted for and this number could be higher than other
|
|
* measurement sources of packet loss.
|
|
* "avg-delta-of-delta", G_TYPE_INT64 In nanoseconds, a moving window
|
|
* average of the difference in inter-packet spacing between
|
|
* sender and receiver. A sudden increase in this number can indicate
|
|
* network congestion.
|
|
*
|
|
* Since: 1.18
|
|
*/
|
|
g_object_class_install_property (gobject_class, PROP_TWCC_STATS,
|
|
g_param_spec_boxed ("twcc-stats", "TWCC Statistics",
|
|
"Various statistics from TWCC", GST_TYPE_STRUCTURE,
|
|
G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
|
|
|
|
g_object_class_install_property (gobject_class, PROP_RTP_PROFILE,
|
|
g_param_spec_enum ("rtp-profile", "RTP Profile",
|
|
"RTP profile to use", GST_TYPE_RTP_PROFILE, DEFAULT_RTP_PROFILE,
|
|
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
|
|
|
|
g_object_class_install_property (gobject_class, PROP_NTP_TIME_SOURCE,
|
|
g_param_spec_enum ("ntp-time-source", "NTP Time Source",
|
|
"NTP time source for RTCP packets",
|
|
GST_TYPE_RTP_NTP_TIME_SOURCE, DEFAULT_NTP_TIME_SOURCE,
|
|
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
|
|
|
|
g_object_class_install_property (gobject_class, PROP_RTCP_SYNC_SEND_TIME,
|
|
g_param_spec_boolean ("rtcp-sync-send-time", "RTCP Sync Send Time",
|
|
"Use send time or capture time for RTCP sync "
|
|
"(TRUE = send time, FALSE = capture time)",
|
|
DEFAULT_RTCP_SYNC_SEND_TIME,
|
|
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
|
|
|
|
gstelement_class->change_state =
|
|
GST_DEBUG_FUNCPTR (gst_rtp_session_change_state);
|
|
gstelement_class->request_new_pad =
|
|
GST_DEBUG_FUNCPTR (gst_rtp_session_request_new_pad);
|
|
gstelement_class->release_pad =
|
|
GST_DEBUG_FUNCPTR (gst_rtp_session_release_pad);
|
|
|
|
klass->clear_pt_map = GST_DEBUG_FUNCPTR (gst_rtp_session_clear_pt_map);
|
|
|
|
/* sink pads */
|
|
gst_element_class_add_static_pad_template (gstelement_class,
|
|
&rtpsession_recv_rtp_sink_template);
|
|
gst_element_class_add_static_pad_template (gstelement_class,
|
|
&rtpsession_recv_rtcp_sink_template);
|
|
gst_element_class_add_static_pad_template (gstelement_class,
|
|
&rtpsession_send_rtp_sink_template);
|
|
|
|
/* src pads */
|
|
gst_element_class_add_static_pad_template (gstelement_class,
|
|
&rtpsession_recv_rtp_src_template);
|
|
gst_element_class_add_static_pad_template (gstelement_class,
|
|
&rtpsession_sync_src_template);
|
|
gst_element_class_add_static_pad_template (gstelement_class,
|
|
&rtpsession_send_rtp_src_template);
|
|
gst_element_class_add_static_pad_template (gstelement_class,
|
|
&rtpsession_send_rtcp_src_template);
|
|
|
|
gst_element_class_set_static_metadata (gstelement_class, "RTP Session",
|
|
"Filter/Network/RTP",
|
|
"Implement an RTP session", "Wim Taymans <wim.taymans@gmail.com>");
|
|
|
|
GST_DEBUG_CATEGORY_INIT (gst_rtp_session_debug,
|
|
"rtpsession", 0, "RTP Session");
|
|
|
|
GST_DEBUG_REGISTER_FUNCPTR (gst_rtp_session_chain_recv_rtp);
|
|
GST_DEBUG_REGISTER_FUNCPTR (gst_rtp_session_chain_recv_rtp_list);
|
|
GST_DEBUG_REGISTER_FUNCPTR (gst_rtp_session_chain_recv_rtcp);
|
|
GST_DEBUG_REGISTER_FUNCPTR (gst_rtp_session_chain_send_rtp);
|
|
GST_DEBUG_REGISTER_FUNCPTR (gst_rtp_session_chain_send_rtp_list);
|
|
|
|
gst_type_mark_as_plugin_api (GST_TYPE_RTP_NTP_TIME_SOURCE, 0);
|
|
gst_type_mark_as_plugin_api (RTP_TYPE_SESSION, 0);
|
|
gst_type_mark_as_plugin_api (RTP_TYPE_SOURCE, 0);
|
|
}
|
|
|
|
static void
|
|
gst_rtp_session_init (GstRtpSession * rtpsession)
|
|
{
|
|
rtpsession->priv = gst_rtp_session_get_instance_private (rtpsession);
|
|
g_mutex_init (&rtpsession->priv->lock);
|
|
g_cond_init (&rtpsession->priv->cond);
|
|
rtpsession->priv->sysclock = gst_system_clock_obtain ();
|
|
rtpsession->priv->session = rtp_session_new ();
|
|
rtpsession->priv->use_pipeline_clock = DEFAULT_USE_PIPELINE_CLOCK;
|
|
rtpsession->priv->rtcp_sync_send_time = DEFAULT_RTCP_SYNC_SEND_TIME;
|
|
|
|
/* configure callbacks */
|
|
rtp_session_set_callbacks (rtpsession->priv->session, &callbacks, rtpsession);
|
|
/* configure signals */
|
|
g_signal_connect (rtpsession->priv->session, "on-new-ssrc",
|
|
(GCallback) on_new_ssrc, rtpsession);
|
|
g_signal_connect (rtpsession->priv->session, "on-ssrc-collision",
|
|
(GCallback) on_ssrc_collision, rtpsession);
|
|
g_signal_connect (rtpsession->priv->session, "on-ssrc-validated",
|
|
(GCallback) on_ssrc_validated, rtpsession);
|
|
g_signal_connect (rtpsession->priv->session, "on-ssrc-active",
|
|
(GCallback) on_ssrc_active, rtpsession);
|
|
g_signal_connect (rtpsession->priv->session, "on-ssrc-sdes",
|
|
(GCallback) on_ssrc_sdes, rtpsession);
|
|
g_signal_connect (rtpsession->priv->session, "on-bye-ssrc",
|
|
(GCallback) on_bye_ssrc, rtpsession);
|
|
g_signal_connect (rtpsession->priv->session, "on-bye-timeout",
|
|
(GCallback) on_bye_timeout, rtpsession);
|
|
g_signal_connect (rtpsession->priv->session, "on-timeout",
|
|
(GCallback) on_timeout, rtpsession);
|
|
g_signal_connect (rtpsession->priv->session, "on-sender-timeout",
|
|
(GCallback) on_sender_timeout, rtpsession);
|
|
g_signal_connect (rtpsession->priv->session, "on-new-sender-ssrc",
|
|
(GCallback) on_new_sender_ssrc, rtpsession);
|
|
g_signal_connect (rtpsession->priv->session, "on-sender-ssrc-active",
|
|
(GCallback) on_sender_ssrc_active, rtpsession);
|
|
g_signal_connect (rtpsession->priv->session, "notify::stats",
|
|
(GCallback) on_notify_stats, rtpsession);
|
|
rtpsession->priv->ptmap = g_hash_table_new_full (NULL, NULL, NULL,
|
|
(GDestroyNotify) gst_caps_unref);
|
|
|
|
rtpsession->recv_rtcp_segment_seqnum = GST_SEQNUM_INVALID;
|
|
|
|
gst_segment_init (&rtpsession->recv_rtp_seg, GST_FORMAT_UNDEFINED);
|
|
gst_segment_init (&rtpsession->send_rtp_seg, GST_FORMAT_UNDEFINED);
|
|
|
|
rtpsession->priv->thread_stopped = TRUE;
|
|
|
|
rtpsession->priv->recv_rtx_req_count = 0;
|
|
rtpsession->priv->sent_rtx_req_count = 0;
|
|
|
|
rtpsession->priv->ntp_time_source = DEFAULT_NTP_TIME_SOURCE;
|
|
}
|
|
|
|
static void
|
|
gst_rtp_session_finalize (GObject * object)
|
|
{
|
|
GstRtpSession *rtpsession;
|
|
|
|
rtpsession = GST_RTP_SESSION (object);
|
|
|
|
g_hash_table_destroy (rtpsession->priv->ptmap);
|
|
g_mutex_clear (&rtpsession->priv->lock);
|
|
g_cond_clear (&rtpsession->priv->cond);
|
|
g_object_unref (rtpsession->priv->sysclock);
|
|
g_object_unref (rtpsession->priv->session);
|
|
if (rtpsession->priv->last_twcc_stats)
|
|
gst_structure_free (rtpsession->priv->last_twcc_stats);
|
|
|
|
G_OBJECT_CLASS (parent_class)->finalize (object);
|
|
}
|
|
|
|
static void
|
|
gst_rtp_session_set_property (GObject * object, guint prop_id,
|
|
const GValue * value, GParamSpec * pspec)
|
|
{
|
|
GstRtpSession *rtpsession;
|
|
GstRtpSessionPrivate *priv;
|
|
|
|
rtpsession = GST_RTP_SESSION (object);
|
|
priv = rtpsession->priv;
|
|
|
|
switch (prop_id) {
|
|
case PROP_BANDWIDTH:
|
|
g_object_set_property (G_OBJECT (priv->session), "bandwidth", value);
|
|
break;
|
|
case PROP_RTCP_FRACTION:
|
|
g_object_set_property (G_OBJECT (priv->session), "rtcp-fraction", value);
|
|
break;
|
|
case PROP_RTCP_RR_BANDWIDTH:
|
|
g_object_set_property (G_OBJECT (priv->session), "rtcp-rr-bandwidth",
|
|
value);
|
|
break;
|
|
case PROP_RTCP_RS_BANDWIDTH:
|
|
g_object_set_property (G_OBJECT (priv->session), "rtcp-rs-bandwidth",
|
|
value);
|
|
break;
|
|
case PROP_SDES:
|
|
rtp_session_set_sdes_struct (priv->session, g_value_get_boxed (value));
|
|
break;
|
|
case PROP_USE_PIPELINE_CLOCK:
|
|
priv->use_pipeline_clock = g_value_get_boolean (value);
|
|
break;
|
|
case PROP_RTCP_MIN_INTERVAL:
|
|
g_object_set_property (G_OBJECT (priv->session), "rtcp-min-interval",
|
|
value);
|
|
break;
|
|
case PROP_PROBATION:
|
|
g_object_set_property (G_OBJECT (priv->session), "probation", value);
|
|
break;
|
|
case PROP_MAX_DROPOUT_TIME:
|
|
g_object_set_property (G_OBJECT (priv->session), "max-dropout-time",
|
|
value);
|
|
break;
|
|
case PROP_MAX_MISORDER_TIME:
|
|
g_object_set_property (G_OBJECT (priv->session), "max-misorder-time",
|
|
value);
|
|
break;
|
|
case PROP_RTP_PROFILE:
|
|
g_object_set_property (G_OBJECT (priv->session), "rtp-profile", value);
|
|
break;
|
|
case PROP_NTP_TIME_SOURCE:
|
|
priv->ntp_time_source = g_value_get_enum (value);
|
|
break;
|
|
case PROP_RTCP_SYNC_SEND_TIME:
|
|
priv->rtcp_sync_send_time = g_value_get_boolean (value);
|
|
break;
|
|
default:
|
|
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
|
|
break;
|
|
}
|
|
}
|
|
|
|
static void
|
|
gst_rtp_session_get_property (GObject * object, guint prop_id,
|
|
GValue * value, GParamSpec * pspec)
|
|
{
|
|
GstRtpSession *rtpsession;
|
|
GstRtpSessionPrivate *priv;
|
|
|
|
rtpsession = GST_RTP_SESSION (object);
|
|
priv = rtpsession->priv;
|
|
|
|
switch (prop_id) {
|
|
case PROP_BANDWIDTH:
|
|
g_object_get_property (G_OBJECT (priv->session), "bandwidth", value);
|
|
break;
|
|
case PROP_RTCP_FRACTION:
|
|
g_object_get_property (G_OBJECT (priv->session), "rtcp-fraction", value);
|
|
break;
|
|
case PROP_RTCP_RR_BANDWIDTH:
|
|
g_object_get_property (G_OBJECT (priv->session), "rtcp-rr-bandwidth",
|
|
value);
|
|
break;
|
|
case PROP_RTCP_RS_BANDWIDTH:
|
|
g_object_get_property (G_OBJECT (priv->session), "rtcp-rs-bandwidth",
|
|
value);
|
|
break;
|
|
case PROP_SDES:
|
|
g_value_take_boxed (value, rtp_session_get_sdes_struct (priv->session));
|
|
break;
|
|
case PROP_NUM_SOURCES:
|
|
g_value_set_uint (value, rtp_session_get_num_sources (priv->session));
|
|
break;
|
|
case PROP_NUM_ACTIVE_SOURCES:
|
|
g_value_set_uint (value,
|
|
rtp_session_get_num_active_sources (priv->session));
|
|
break;
|
|
case PROP_INTERNAL_SESSION:
|
|
g_value_set_object (value, priv->session);
|
|
break;
|
|
case PROP_USE_PIPELINE_CLOCK:
|
|
g_value_set_boolean (value, priv->use_pipeline_clock);
|
|
break;
|
|
case PROP_RTCP_MIN_INTERVAL:
|
|
g_object_get_property (G_OBJECT (priv->session), "rtcp-min-interval",
|
|
value);
|
|
break;
|
|
case PROP_PROBATION:
|
|
g_object_get_property (G_OBJECT (priv->session), "probation", value);
|
|
break;
|
|
case PROP_MAX_DROPOUT_TIME:
|
|
g_object_get_property (G_OBJECT (priv->session), "max-dropout-time",
|
|
value);
|
|
break;
|
|
case PROP_MAX_MISORDER_TIME:
|
|
g_object_get_property (G_OBJECT (priv->session), "max-misorder-time",
|
|
value);
|
|
break;
|
|
case PROP_STATS:
|
|
g_value_take_boxed (value, gst_rtp_session_create_stats (rtpsession));
|
|
break;
|
|
case PROP_TWCC_STATS:
|
|
GST_RTP_SESSION_LOCK (rtpsession);
|
|
g_value_set_boxed (value, priv->last_twcc_stats);
|
|
GST_RTP_SESSION_UNLOCK (rtpsession);
|
|
break;
|
|
case PROP_RTP_PROFILE:
|
|
g_object_get_property (G_OBJECT (priv->session), "rtp-profile", value);
|
|
break;
|
|
case PROP_NTP_TIME_SOURCE:
|
|
g_value_set_enum (value, priv->ntp_time_source);
|
|
break;
|
|
case PROP_RTCP_SYNC_SEND_TIME:
|
|
g_value_set_boolean (value, priv->rtcp_sync_send_time);
|
|
break;
|
|
default:
|
|
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
|
|
break;
|
|
}
|
|
}
|
|
|
|
static GstStructure *
|
|
gst_rtp_session_create_stats (GstRtpSession * rtpsession)
|
|
{
|
|
GstStructure *s;
|
|
|
|
g_object_get (rtpsession->priv->session, "stats", &s, NULL);
|
|
gst_structure_set (s, "rtx-count", G_TYPE_UINT,
|
|
rtpsession->priv->recv_rtx_req_count, "recv-rtx-req-count", G_TYPE_UINT,
|
|
rtpsession->priv->recv_rtx_req_count, "sent-rtx-req-count", G_TYPE_UINT,
|
|
rtpsession->priv->sent_rtx_req_count, NULL);
|
|
|
|
return s;
|
|
}
|
|
|
|
static void
|
|
get_current_times (GstRtpSession * rtpsession, GstClockTime * running_time,
|
|
guint64 * ntpnstime)
|
|
{
|
|
guint64 ntpns = -1;
|
|
GstClock *clock;
|
|
GstClockTime base_time, rt, clock_time;
|
|
|
|
GST_OBJECT_LOCK (rtpsession);
|
|
if ((clock = GST_ELEMENT_CLOCK (rtpsession))) {
|
|
base_time = GST_ELEMENT_CAST (rtpsession)->base_time;
|
|
gst_object_ref (clock);
|
|
GST_OBJECT_UNLOCK (rtpsession);
|
|
|
|
/* get current clock time and convert to running time */
|
|
clock_time = gst_clock_get_time (clock);
|
|
rt = clock_time - base_time;
|
|
|
|
if (rtpsession->priv->use_pipeline_clock) {
|
|
ntpns = rt;
|
|
/* add constant to convert from 1970 based time to 1900 based time */
|
|
ntpns += (2208988800LL * GST_SECOND);
|
|
} else {
|
|
switch (rtpsession->priv->ntp_time_source) {
|
|
case GST_RTP_NTP_TIME_SOURCE_NTP:
|
|
case GST_RTP_NTP_TIME_SOURCE_UNIX:{
|
|
/* get current NTP time */
|
|
ntpns = g_get_real_time () * GST_USECOND;
|
|
|
|
/* add constant to convert from 1970 based time to 1900 based time */
|
|
if (rtpsession->priv->ntp_time_source == GST_RTP_NTP_TIME_SOURCE_NTP)
|
|
ntpns += (2208988800LL * GST_SECOND);
|
|
break;
|
|
}
|
|
case GST_RTP_NTP_TIME_SOURCE_RUNNING_TIME:
|
|
ntpns = rt;
|
|
break;
|
|
case GST_RTP_NTP_TIME_SOURCE_CLOCK_TIME:
|
|
ntpns = clock_time;
|
|
break;
|
|
default:
|
|
ntpns = -1;
|
|
g_assert_not_reached ();
|
|
break;
|
|
}
|
|
}
|
|
|
|
gst_object_unref (clock);
|
|
} else {
|
|
GST_OBJECT_UNLOCK (rtpsession);
|
|
rt = -1;
|
|
ntpns = -1;
|
|
}
|
|
if (running_time)
|
|
*running_time = rt;
|
|
if (ntpnstime)
|
|
*ntpnstime = ntpns;
|
|
}
|
|
|
|
/* must be called with GST_RTP_SESSION_LOCK */
|
|
static void
|
|
signal_waiting_rtcp_thread_unlocked (GstRtpSession * rtpsession)
|
|
{
|
|
if (rtpsession->priv->wait_send) {
|
|
GST_LOG_OBJECT (rtpsession, "signal RTCP thread");
|
|
rtpsession->priv->wait_send = FALSE;
|
|
GST_RTP_SESSION_SIGNAL (rtpsession);
|
|
}
|
|
}
|
|
|
|
static void
|
|
rtcp_thread (GstRtpSession * rtpsession)
|
|
{
|
|
GstClockID id;
|
|
GstClockTime current_time;
|
|
GstClockTime next_timeout;
|
|
guint64 ntpnstime;
|
|
GstClockTime running_time;
|
|
RTPSession *session;
|
|
GstClock *sysclock;
|
|
|
|
GST_DEBUG_OBJECT (rtpsession, "entering RTCP thread");
|
|
|
|
GST_RTP_SESSION_LOCK (rtpsession);
|
|
|
|
while (rtpsession->priv->wait_send) {
|
|
GST_LOG_OBJECT (rtpsession, "waiting for getting started");
|
|
GST_RTP_SESSION_WAIT (rtpsession);
|
|
GST_LOG_OBJECT (rtpsession, "signaled...");
|
|
}
|
|
|
|
sysclock = rtpsession->priv->sysclock;
|
|
current_time = gst_clock_get_time (sysclock);
|
|
|
|
session = rtpsession->priv->session;
|
|
|
|
GST_DEBUG_OBJECT (rtpsession, "starting at %" GST_TIME_FORMAT,
|
|
GST_TIME_ARGS (current_time));
|
|
session->start_time = current_time;
|
|
|
|
while (!rtpsession->priv->stop_thread) {
|
|
GstClockReturn res;
|
|
|
|
/* get initial estimate */
|
|
next_timeout = rtp_session_next_timeout (session, current_time);
|
|
|
|
GST_DEBUG_OBJECT (rtpsession, "next check time %" GST_TIME_FORMAT,
|
|
GST_TIME_ARGS (next_timeout));
|
|
|
|
/* leave if no more timeouts, the session ended */
|
|
if (next_timeout == GST_CLOCK_TIME_NONE)
|
|
break;
|
|
|
|
id = rtpsession->priv->id =
|
|
gst_clock_new_single_shot_id (sysclock, next_timeout);
|
|
GST_RTP_SESSION_UNLOCK (rtpsession);
|
|
|
|
res = gst_clock_id_wait (id, NULL);
|
|
|
|
GST_RTP_SESSION_LOCK (rtpsession);
|
|
gst_clock_id_unref (id);
|
|
rtpsession->priv->id = NULL;
|
|
|
|
if (rtpsession->priv->stop_thread)
|
|
break;
|
|
|
|
/* update current time */
|
|
current_time = gst_clock_get_time (sysclock);
|
|
|
|
/* get current NTP time */
|
|
get_current_times (rtpsession, &running_time, &ntpnstime);
|
|
|
|
/* we get unlocked because we need to perform reconsideration, don't perform
|
|
* the timeout but get a new reporting estimate. */
|
|
GST_DEBUG_OBJECT (rtpsession, "unlocked %d, current %" GST_TIME_FORMAT,
|
|
res, GST_TIME_ARGS (current_time));
|
|
|
|
/* perform actions, we ignore result. Release lock because it might push. */
|
|
GST_RTP_SESSION_UNLOCK (rtpsession);
|
|
rtp_session_on_timeout (session, current_time, ntpnstime, running_time);
|
|
GST_RTP_SESSION_LOCK (rtpsession);
|
|
}
|
|
/* mark the thread as stopped now */
|
|
rtpsession->priv->thread_stopped = TRUE;
|
|
GST_RTP_SESSION_UNLOCK (rtpsession);
|
|
|
|
GST_DEBUG_OBJECT (rtpsession, "leaving RTCP thread");
|
|
}
|
|
|
|
static gboolean
|
|
start_rtcp_thread (GstRtpSession * rtpsession)
|
|
{
|
|
GError *error = NULL;
|
|
gboolean res;
|
|
|
|
GST_DEBUG_OBJECT (rtpsession, "starting RTCP thread");
|
|
|
|
GST_RTP_SESSION_LOCK (rtpsession);
|
|
rtpsession->priv->stop_thread = FALSE;
|
|
if (rtpsession->priv->thread_stopped) {
|
|
/* if the thread stopped, and we still have a handle to the thread, join it
|
|
* now. We can safely join with the lock held, the thread will not take it
|
|
* anymore. */
|
|
if (rtpsession->priv->thread)
|
|
g_thread_join (rtpsession->priv->thread);
|
|
/* only create a new thread if the old one was stopped. Otherwise we can
|
|
* just reuse the currently running one. */
|
|
rtpsession->priv->thread = g_thread_try_new ("rtpsession-rtcp",
|
|
(GThreadFunc) rtcp_thread, rtpsession, &error);
|
|
rtpsession->priv->thread_stopped = FALSE;
|
|
}
|
|
GST_RTP_SESSION_UNLOCK (rtpsession);
|
|
|
|
if (error != NULL) {
|
|
res = FALSE;
|
|
GST_DEBUG_OBJECT (rtpsession, "failed to start thread, %s", error->message);
|
|
g_error_free (error);
|
|
} else {
|
|
res = TRUE;
|
|
}
|
|
return res;
|
|
}
|
|
|
|
static void
|
|
stop_rtcp_thread (GstRtpSession * rtpsession)
|
|
{
|
|
GST_DEBUG_OBJECT (rtpsession, "stopping RTCP thread");
|
|
|
|
GST_RTP_SESSION_LOCK (rtpsession);
|
|
rtpsession->priv->stop_thread = TRUE;
|
|
signal_waiting_rtcp_thread_unlocked (rtpsession);
|
|
if (rtpsession->priv->id)
|
|
gst_clock_id_unschedule (rtpsession->priv->id);
|
|
GST_RTP_SESSION_UNLOCK (rtpsession);
|
|
}
|
|
|
|
static void
|
|
join_rtcp_thread (GstRtpSession * rtpsession)
|
|
{
|
|
GST_RTP_SESSION_LOCK (rtpsession);
|
|
/* don't try to join when we have no thread */
|
|
if (rtpsession->priv->thread != NULL) {
|
|
GST_DEBUG_OBJECT (rtpsession, "joining RTCP thread");
|
|
GST_RTP_SESSION_UNLOCK (rtpsession);
|
|
|
|
g_thread_join (rtpsession->priv->thread);
|
|
|
|
GST_RTP_SESSION_LOCK (rtpsession);
|
|
/* after the join, take the lock and clear the thread structure. The caller
|
|
* is supposed to not concurrently call start and join. */
|
|
rtpsession->priv->thread = NULL;
|
|
}
|
|
GST_RTP_SESSION_UNLOCK (rtpsession);
|
|
}
|
|
|
|
static GstStateChangeReturn
|
|
gst_rtp_session_change_state (GstElement * element, GstStateChange transition)
|
|
{
|
|
GstStateChangeReturn res;
|
|
GstRtpSession *rtpsession;
|
|
|
|
rtpsession = GST_RTP_SESSION (element);
|
|
|
|
switch (transition) {
|
|
case GST_STATE_CHANGE_NULL_TO_READY:
|
|
break;
|
|
case GST_STATE_CHANGE_READY_TO_PAUSED:
|
|
GST_RTP_SESSION_LOCK (rtpsession);
|
|
rtpsession->priv->wait_send = TRUE;
|
|
GST_RTP_SESSION_UNLOCK (rtpsession);
|
|
break;
|
|
case GST_STATE_CHANGE_PAUSED_TO_PLAYING:
|
|
break;
|
|
case GST_STATE_CHANGE_PLAYING_TO_PAUSED:
|
|
case GST_STATE_CHANGE_PAUSED_TO_READY:
|
|
/* no need to join yet, we might want to continue later. Also, the
|
|
* dataflow could block downstream so that a join could just block
|
|
* forever. */
|
|
stop_rtcp_thread (rtpsession);
|
|
break;
|
|
default:
|
|
break;
|
|
}
|
|
|
|
res = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
|
|
|
|
switch (transition) {
|
|
case GST_STATE_CHANGE_PAUSED_TO_PLAYING:
|
|
if (!start_rtcp_thread (rtpsession))
|
|
goto failed_thread;
|
|
break;
|
|
case GST_STATE_CHANGE_PLAYING_TO_PAUSED:
|
|
break;
|
|
case GST_STATE_CHANGE_PAUSED_TO_READY:
|
|
/* downstream is now releasing the dataflow and we can join. */
|
|
join_rtcp_thread (rtpsession);
|
|
rtp_session_reset (rtpsession->priv->session);
|
|
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)
|
|
{
|
|
GST_RTP_SESSION_LOCK (rtpsession);
|
|
g_hash_table_foreach_remove (rtpsession->priv->ptmap, return_true, NULL);
|
|
GST_RTP_SESSION_UNLOCK (rtpsession);
|
|
}
|
|
|
|
/* called when the session manager has an RTP packet ready to be pushed */
|
|
static GstFlowReturn
|
|
gst_rtp_session_process_rtp (RTPSession * sess, RTPSource * src,
|
|
GstBuffer * buffer, gpointer user_data)
|
|
{
|
|
GstFlowReturn result;
|
|
GstRtpSession *rtpsession;
|
|
GstPad *rtp_src;
|
|
|
|
rtpsession = GST_RTP_SESSION (user_data);
|
|
|
|
GST_RTP_SESSION_LOCK (rtpsession);
|
|
if ((rtp_src = rtpsession->recv_rtp_src))
|
|
gst_object_ref (rtp_src);
|
|
GST_RTP_SESSION_UNLOCK (rtpsession);
|
|
|
|
if (rtp_src) {
|
|
if (rtpsession->priv->processed_list) {
|
|
GST_LOG_OBJECT (rtpsession, "queueing received RTP packet");
|
|
gst_buffer_list_add (rtpsession->priv->processed_list, buffer);
|
|
result = GST_FLOW_OK;
|
|
} else {
|
|
GST_LOG_OBJECT (rtpsession, "pushing received RTP packet");
|
|
result = gst_pad_push (rtp_src, buffer);
|
|
}
|
|
gst_object_unref (rtp_src);
|
|
} else {
|
|
GST_DEBUG_OBJECT (rtpsession, "dropping received RTP packet");
|
|
gst_buffer_unref (buffer);
|
|
result = GST_FLOW_OK;
|
|
}
|
|
return result;
|
|
}
|
|
|
|
/* called when the session manager has an RTP packet ready for further
|
|
* sending */
|
|
static GstFlowReturn
|
|
gst_rtp_session_send_rtp (RTPSession * sess, RTPSource * src,
|
|
gpointer data, gpointer user_data)
|
|
{
|
|
GstFlowReturn result;
|
|
GstRtpSession *rtpsession;
|
|
GstPad *rtp_src;
|
|
|
|
rtpsession = GST_RTP_SESSION (user_data);
|
|
|
|
GST_RTP_SESSION_LOCK (rtpsession);
|
|
if ((rtp_src = rtpsession->send_rtp_src))
|
|
gst_object_ref (rtp_src);
|
|
signal_waiting_rtcp_thread_unlocked (rtpsession);
|
|
GST_RTP_SESSION_UNLOCK (rtpsession);
|
|
|
|
if (rtp_src) {
|
|
if (GST_IS_BUFFER (data)) {
|
|
GST_LOG_OBJECT (rtpsession, "sending RTP packet");
|
|
result = gst_pad_push (rtp_src, GST_BUFFER_CAST (data));
|
|
} else {
|
|
GST_LOG_OBJECT (rtpsession, "sending RTP list");
|
|
result = gst_pad_push_list (rtp_src, GST_BUFFER_LIST_CAST (data));
|
|
}
|
|
gst_object_unref (rtp_src);
|
|
} else {
|
|
gst_mini_object_unref (GST_MINI_OBJECT_CAST (data));
|
|
result = GST_FLOW_OK;
|
|
}
|
|
return result;
|
|
}
|
|
|
|
static void
|
|
do_rtcp_events (GstRtpSession * rtpsession, GstPad * srcpad)
|
|
{
|
|
GstCaps *caps;
|
|
GstSegment seg;
|
|
GstEvent *event;
|
|
gchar *stream_id;
|
|
gboolean have_group_id;
|
|
guint group_id;
|
|
|
|
stream_id =
|
|
g_strdup_printf ("%08x%08x%08x%08x", g_random_int (), g_random_int (),
|
|
g_random_int (), g_random_int ());
|
|
|
|
GST_RTP_SESSION_LOCK (rtpsession);
|
|
if (rtpsession->recv_rtp_sink) {
|
|
event =
|
|
gst_pad_get_sticky_event (rtpsession->recv_rtp_sink,
|
|
GST_EVENT_STREAM_START, 0);
|
|
if (event) {
|
|
if (gst_event_parse_group_id (event, &group_id))
|
|
have_group_id = TRUE;
|
|
else
|
|
have_group_id = FALSE;
|
|
gst_event_unref (event);
|
|
} else {
|
|
have_group_id = TRUE;
|
|
group_id = gst_util_group_id_next ();
|
|
}
|
|
} else {
|
|
have_group_id = TRUE;
|
|
group_id = gst_util_group_id_next ();
|
|
}
|
|
GST_RTP_SESSION_UNLOCK (rtpsession);
|
|
|
|
event = gst_event_new_stream_start (stream_id);
|
|
rtpsession->recv_rtcp_segment_seqnum = gst_event_get_seqnum (event);
|
|
gst_event_set_seqnum (event, rtpsession->recv_rtcp_segment_seqnum);
|
|
if (have_group_id)
|
|
gst_event_set_group_id (event, group_id);
|
|
gst_pad_push_event (srcpad, event);
|
|
g_free (stream_id);
|
|
|
|
caps = gst_caps_new_empty_simple ("application/x-rtcp");
|
|
gst_pad_set_caps (srcpad, caps);
|
|
gst_caps_unref (caps);
|
|
|
|
gst_segment_init (&seg, GST_FORMAT_TIME);
|
|
event = gst_event_new_segment (&seg);
|
|
gst_event_set_seqnum (event, rtpsession->recv_rtcp_segment_seqnum);
|
|
gst_pad_push_event (srcpad, event);
|
|
}
|
|
|
|
/* called when the session manager has an RTCP packet ready for further
|
|
* sending. The eos flag is set when an EOS event should be sent downstream as
|
|
* well. */
|
|
static GstFlowReturn
|
|
gst_rtp_session_send_rtcp (RTPSession * sess, RTPSource * src,
|
|
GstBuffer * buffer, gboolean all_sources_bye, gpointer user_data)
|
|
{
|
|
GstFlowReturn result;
|
|
GstRtpSession *rtpsession;
|
|
GstPad *rtcp_src;
|
|
|
|
rtpsession = GST_RTP_SESSION (user_data);
|
|
|
|
GST_RTP_SESSION_LOCK (rtpsession);
|
|
if (rtpsession->priv->stop_thread)
|
|
goto stopping;
|
|
|
|
if ((rtcp_src = rtpsession->send_rtcp_src)) {
|
|
gst_object_ref (rtcp_src);
|
|
GST_RTP_SESSION_UNLOCK (rtpsession);
|
|
|
|
/* set rtcp caps on output pad */
|
|
if (!gst_pad_has_current_caps (rtcp_src))
|
|
do_rtcp_events (rtpsession, rtcp_src);
|
|
|
|
GST_LOG_OBJECT (rtpsession, "sending RTCP");
|
|
result = gst_pad_push (rtcp_src, buffer);
|
|
|
|
/* Forward send an EOS on the RTCP sink if we received an EOS on the
|
|
* send_rtp_sink. We don't need to check the recv_rtp_sink since in this
|
|
* case the EOS event would already have been sent */
|
|
if (all_sources_bye && rtpsession->send_rtp_sink &&
|
|
GST_PAD_IS_EOS (rtpsession->send_rtp_sink)) {
|
|
GstEvent *event;
|
|
|
|
GST_LOG_OBJECT (rtpsession, "sending EOS");
|
|
|
|
event = gst_event_new_eos ();
|
|
gst_event_set_seqnum (event, rtpsession->recv_rtcp_segment_seqnum);
|
|
gst_pad_push_event (rtcp_src, event);
|
|
}
|
|
gst_object_unref (rtcp_src);
|
|
} else {
|
|
GST_RTP_SESSION_UNLOCK (rtpsession);
|
|
|
|
GST_DEBUG_OBJECT (rtpsession, "not sending RTCP, no output pad");
|
|
gst_buffer_unref (buffer);
|
|
result = GST_FLOW_OK;
|
|
}
|
|
return result;
|
|
|
|
/* ERRORS */
|
|
stopping:
|
|
{
|
|
GST_DEBUG_OBJECT (rtpsession, "we are stopping");
|
|
gst_buffer_unref (buffer);
|
|
GST_RTP_SESSION_UNLOCK (rtpsession);
|
|
return GST_FLOW_OK;
|
|
}
|
|
}
|
|
|
|
/* called when the session manager has an SR RTCP packet ready for handling
|
|
* inter stream synchronisation */
|
|
static GstFlowReturn
|
|
gst_rtp_session_sync_rtcp (RTPSession * sess,
|
|
GstBuffer * buffer, gpointer user_data)
|
|
{
|
|
GstFlowReturn result;
|
|
GstRtpSession *rtpsession;
|
|
GstPad *sync_src;
|
|
|
|
rtpsession = GST_RTP_SESSION (user_data);
|
|
|
|
GST_RTP_SESSION_LOCK (rtpsession);
|
|
if (rtpsession->priv->stop_thread)
|
|
goto stopping;
|
|
|
|
if ((sync_src = rtpsession->sync_src)) {
|
|
gst_object_ref (sync_src);
|
|
GST_RTP_SESSION_UNLOCK (rtpsession);
|
|
|
|
/* set rtcp caps on output pad, this happens
|
|
* when we receive RTCP muxed with RTP according
|
|
* to RFC5761. Otherwise we would have forwarded
|
|
* the events from the recv_rtcp_sink pad already
|
|
*/
|
|
if (!gst_pad_has_current_caps (sync_src))
|
|
do_rtcp_events (rtpsession, sync_src);
|
|
|
|
GST_LOG_OBJECT (rtpsession, "sending Sync RTCP");
|
|
result = gst_pad_push (sync_src, buffer);
|
|
gst_object_unref (sync_src);
|
|
} else {
|
|
GST_RTP_SESSION_UNLOCK (rtpsession);
|
|
|
|
GST_DEBUG_OBJECT (rtpsession, "not sending Sync RTCP, no output pad");
|
|
gst_buffer_unref (buffer);
|
|
result = GST_FLOW_OK;
|
|
}
|
|
return result;
|
|
|
|
/* ERRORS */
|
|
stopping:
|
|
{
|
|
GST_DEBUG_OBJECT (rtpsession, "we are stopping");
|
|
gst_buffer_unref (buffer);
|
|
GST_RTP_SESSION_UNLOCK (rtpsession);
|
|
return GST_FLOW_OK;
|
|
}
|
|
}
|
|
|
|
static void
|
|
gst_rtp_session_cache_caps (GstRtpSession * rtpsession, GstCaps * caps)
|
|
{
|
|
GstRtpSessionPrivate *priv;
|
|
const GstStructure *s;
|
|
gint payload;
|
|
|
|
priv = rtpsession->priv;
|
|
|
|
GST_DEBUG_OBJECT (rtpsession, "parsing caps");
|
|
|
|
s = gst_caps_get_structure (caps, 0);
|
|
|
|
if (!gst_structure_get_int (s, "payload", &payload))
|
|
return;
|
|
|
|
if (g_hash_table_lookup (priv->ptmap, GINT_TO_POINTER (payload)))
|
|
return;
|
|
|
|
rtp_session_update_recv_caps_structure (rtpsession->priv->session, s);
|
|
|
|
g_hash_table_insert (priv->ptmap, GINT_TO_POINTER (payload),
|
|
gst_caps_ref (caps));
|
|
}
|
|
|
|
static GstCaps *
|
|
gst_rtp_session_get_caps_for_pt (GstRtpSession * rtpsession, guint payload)
|
|
{
|
|
GstCaps *caps = NULL;
|
|
GValue args[2] = { {0}, {0} };
|
|
GValue ret = { 0 };
|
|
|
|
GST_RTP_SESSION_LOCK (rtpsession);
|
|
caps = g_hash_table_lookup (rtpsession->priv->ptmap,
|
|
GINT_TO_POINTER (payload));
|
|
if (caps) {
|
|
gst_caps_ref (caps);
|
|
goto done;
|
|
}
|
|
|
|
/* not found in the cache, try to get it with a signal */
|
|
g_value_init (&args[0], GST_TYPE_ELEMENT);
|
|
g_value_set_object (&args[0], rtpsession);
|
|
g_value_init (&args[1], G_TYPE_UINT);
|
|
g_value_set_uint (&args[1], payload);
|
|
|
|
g_value_init (&ret, GST_TYPE_CAPS);
|
|
g_value_set_boxed (&ret, NULL);
|
|
|
|
GST_RTP_SESSION_UNLOCK (rtpsession);
|
|
|
|
g_signal_emitv (args, gst_rtp_session_signals[SIGNAL_REQUEST_PT_MAP], 0,
|
|
&ret);
|
|
|
|
GST_RTP_SESSION_LOCK (rtpsession);
|
|
|
|
g_value_unset (&args[0]);
|
|
g_value_unset (&args[1]);
|
|
caps = (GstCaps *) g_value_dup_boxed (&ret);
|
|
g_value_unset (&ret);
|
|
if (!caps)
|
|
goto no_caps;
|
|
|
|
gst_rtp_session_cache_caps (rtpsession, caps);
|
|
|
|
done:
|
|
GST_RTP_SESSION_UNLOCK (rtpsession);
|
|
|
|
return caps;
|
|
|
|
no_caps:
|
|
{
|
|
GST_DEBUG_OBJECT (rtpsession, "could not get caps");
|
|
goto done;
|
|
}
|
|
}
|
|
|
|
/* called when the session manager needs the clock rate */
|
|
static gint
|
|
gst_rtp_session_clock_rate (RTPSession * sess, guint8 payload,
|
|
gpointer user_data)
|
|
{
|
|
gint result = -1;
|
|
GstRtpSession *rtpsession;
|
|
GstCaps *caps;
|
|
const GstStructure *s;
|
|
|
|
rtpsession = GST_RTP_SESSION_CAST (user_data);
|
|
|
|
caps = gst_rtp_session_get_caps_for_pt (rtpsession, payload);
|
|
|
|
if (!caps)
|
|
goto done;
|
|
|
|
s = gst_caps_get_structure (caps, 0);
|
|
if (!gst_structure_get_int (s, "clock-rate", &result))
|
|
goto no_clock_rate;
|
|
|
|
gst_caps_unref (caps);
|
|
|
|
GST_DEBUG_OBJECT (rtpsession, "parsed clock-rate %d", result);
|
|
|
|
done:
|
|
|
|
return result;
|
|
|
|
/* ERRORS */
|
|
no_clock_rate:
|
|
{
|
|
gst_caps_unref (caps);
|
|
GST_DEBUG_OBJECT (rtpsession, "No clock-rate in caps!");
|
|
goto done;
|
|
}
|
|
}
|
|
|
|
/* called when the session manager asks us to reconsider the timeout */
|
|
static void
|
|
gst_rtp_session_reconsider (RTPSession * sess, gpointer user_data)
|
|
{
|
|
GstRtpSession *rtpsession;
|
|
|
|
rtpsession = GST_RTP_SESSION_CAST (user_data);
|
|
|
|
GST_RTP_SESSION_LOCK (rtpsession);
|
|
GST_DEBUG_OBJECT (rtpsession, "unlock timer for reconsideration");
|
|
if (rtpsession->priv->id)
|
|
gst_clock_id_unschedule (rtpsession->priv->id);
|
|
GST_RTP_SESSION_UNLOCK (rtpsession);
|
|
}
|
|
|
|
static gboolean
|
|
gst_rtp_session_event_recv_rtp_sink (GstPad * pad, GstObject * parent,
|
|
GstEvent * event)
|
|
{
|
|
GstRtpSession *rtpsession;
|
|
gboolean ret = FALSE;
|
|
|
|
rtpsession = GST_RTP_SESSION (parent);
|
|
|
|
GST_DEBUG_OBJECT (rtpsession, "received event %s",
|
|
GST_EVENT_TYPE_NAME (event));
|
|
|
|
switch (GST_EVENT_TYPE (event)) {
|
|
case GST_EVENT_CAPS:
|
|
{
|
|
GstCaps *caps;
|
|
|
|
/* process */
|
|
gst_event_parse_caps (event, &caps);
|
|
gst_rtp_session_sink_setcaps (pad, rtpsession, caps);
|
|
ret = gst_pad_push_event (rtpsession->recv_rtp_src, event);
|
|
break;
|
|
}
|
|
case GST_EVENT_FLUSH_STOP:
|
|
gst_segment_init (&rtpsession->recv_rtp_seg, GST_FORMAT_UNDEFINED);
|
|
rtpsession->recv_rtcp_segment_seqnum = GST_SEQNUM_INVALID;
|
|
ret = gst_pad_push_event (rtpsession->recv_rtp_src, event);
|
|
break;
|
|
case GST_EVENT_SEGMENT:
|
|
{
|
|
GstSegment *segment, in_segment;
|
|
|
|
segment = &rtpsession->recv_rtp_seg;
|
|
|
|
/* the newsegment event is needed to convert the RTP timestamp to
|
|
* running_time, which is needed to generate a mapping from RTP to NTP
|
|
* timestamps in SR reports */
|
|
gst_event_copy_segment (event, &in_segment);
|
|
GST_DEBUG_OBJECT (rtpsession, "received segment %" GST_SEGMENT_FORMAT,
|
|
&in_segment);
|
|
|
|
/* accept upstream */
|
|
gst_segment_copy_into (&in_segment, segment);
|
|
|
|
/* push event forward */
|
|
ret = gst_pad_push_event (rtpsession->recv_rtp_src, event);
|
|
break;
|
|
}
|
|
case GST_EVENT_EOS:
|
|
{
|
|
GstPad *rtcp_src;
|
|
|
|
ret =
|
|
gst_pad_push_event (rtpsession->recv_rtp_src, gst_event_ref (event));
|
|
|
|
GST_RTP_SESSION_LOCK (rtpsession);
|
|
if ((rtcp_src = rtpsession->send_rtcp_src))
|
|
gst_object_ref (rtcp_src);
|
|
GST_RTP_SESSION_UNLOCK (rtpsession);
|
|
|
|
gst_event_unref (event);
|
|
|
|
if (rtcp_src) {
|
|
event = gst_event_new_eos ();
|
|
if (rtpsession->recv_rtcp_segment_seqnum != GST_SEQNUM_INVALID)
|
|
gst_event_set_seqnum (event, rtpsession->recv_rtcp_segment_seqnum);
|
|
ret = gst_pad_push_event (rtcp_src, event);
|
|
gst_object_unref (rtcp_src);
|
|
} else {
|
|
ret = TRUE;
|
|
}
|
|
break;
|
|
}
|
|
default:
|
|
ret = gst_pad_push_event (rtpsession->recv_rtp_src, event);
|
|
break;
|
|
}
|
|
|
|
return ret;
|
|
|
|
}
|
|
|
|
static gboolean
|
|
gst_rtp_session_request_remote_key_unit (GstRtpSession * rtpsession,
|
|
guint32 ssrc, guint payload, gboolean all_headers, gint count)
|
|
{
|
|
GstCaps *caps;
|
|
|
|
caps = gst_rtp_session_get_caps_for_pt (rtpsession, payload);
|
|
|
|
if (caps) {
|
|
const GstStructure *s = gst_caps_get_structure (caps, 0);
|
|
gboolean pli;
|
|
gboolean fir;
|
|
|
|
pli = gst_structure_has_field (s, "rtcp-fb-nack-pli");
|
|
fir = gst_structure_has_field (s, "rtcp-fb-ccm-fir") && all_headers;
|
|
|
|
/* Google Talk uses FIR for repair, so send it even if we just want a
|
|
* regular PLI */
|
|
if (!pli &&
|
|
gst_structure_has_field (s, "rtcp-fb-x-gstreamer-fir-as-repair"))
|
|
fir = TRUE;
|
|
|
|
gst_caps_unref (caps);
|
|
|
|
if (pli || fir)
|
|
return rtp_session_request_key_unit (rtpsession->priv->session, ssrc,
|
|
fir, count);
|
|
}
|
|
|
|
return FALSE;
|
|
}
|
|
|
|
static gboolean
|
|
gst_rtp_session_event_recv_rtp_src (GstPad * pad, GstObject * parent,
|
|
GstEvent * event)
|
|
{
|
|
GstRtpSession *rtpsession;
|
|
gboolean forward = TRUE;
|
|
gboolean ret = TRUE;
|
|
const GstStructure *s;
|
|
guint32 ssrc;
|
|
guint pt;
|
|
|
|
rtpsession = GST_RTP_SESSION (parent);
|
|
|
|
switch (GST_EVENT_TYPE (event)) {
|
|
case GST_EVENT_CUSTOM_UPSTREAM:
|
|
s = gst_event_get_structure (event);
|
|
if (gst_structure_has_name (s, "GstForceKeyUnit") &&
|
|
gst_structure_get_uint (s, "ssrc", &ssrc) &&
|
|
gst_structure_get_uint (s, "payload", &pt)) {
|
|
gboolean all_headers = FALSE;
|
|
gint count = -1;
|
|
|
|
gst_structure_get_boolean (s, "all-headers", &all_headers);
|
|
if (gst_structure_get_int (s, "count", &count) && count < 0)
|
|
count += G_MAXINT; /* Make sure count is positive if present */
|
|
if (gst_rtp_session_request_remote_key_unit (rtpsession, ssrc, pt,
|
|
all_headers, count))
|
|
forward = FALSE;
|
|
} else if (gst_structure_has_name (s, "GstRTPRetransmissionRequest")) {
|
|
guint seqnum, delay, deadline, max_delay, avg_rtt;
|
|
|
|
GST_RTP_SESSION_LOCK (rtpsession);
|
|
rtpsession->priv->recv_rtx_req_count++;
|
|
GST_RTP_SESSION_UNLOCK (rtpsession);
|
|
|
|
if (!gst_structure_get_uint (s, "ssrc", &ssrc))
|
|
ssrc = -1;
|
|
if (!gst_structure_get_uint (s, "seqnum", &seqnum))
|
|
seqnum = -1;
|
|
if (!gst_structure_get_uint (s, "delay", &delay))
|
|
delay = 0;
|
|
if (!gst_structure_get_uint (s, "deadline", &deadline))
|
|
deadline = 100;
|
|
if (!gst_structure_get_uint (s, "avg-rtt", &avg_rtt))
|
|
avg_rtt = 40;
|
|
|
|
/* remaining time to receive the packet */
|
|
max_delay = deadline;
|
|
if (max_delay > delay)
|
|
max_delay -= delay;
|
|
/* estimated RTT */
|
|
if (max_delay > avg_rtt)
|
|
max_delay -= avg_rtt;
|
|
else
|
|
max_delay = 0;
|
|
|
|
if (rtp_session_request_nack (rtpsession->priv->session, ssrc, seqnum,
|
|
max_delay * GST_MSECOND))
|
|
forward = FALSE;
|
|
}
|
|
break;
|
|
default:
|
|
break;
|
|
}
|
|
|
|
if (forward) {
|
|
GstPad *recv_rtp_sink;
|
|
|
|
GST_RTP_SESSION_LOCK (rtpsession);
|
|
if ((recv_rtp_sink = rtpsession->recv_rtp_sink))
|
|
gst_object_ref (recv_rtp_sink);
|
|
GST_RTP_SESSION_UNLOCK (rtpsession);
|
|
|
|
if (recv_rtp_sink) {
|
|
ret = gst_pad_push_event (recv_rtp_sink, event);
|
|
gst_object_unref (recv_rtp_sink);
|
|
} else
|
|
gst_event_unref (event);
|
|
} else {
|
|
gst_event_unref (event);
|
|
}
|
|
|
|
return ret;
|
|
}
|
|
|
|
|
|
static GstIterator *
|
|
gst_rtp_session_iterate_internal_links (GstPad * pad, GstObject * parent)
|
|
{
|
|
GstRtpSession *rtpsession;
|
|
GstPad *otherpad = NULL;
|
|
GstIterator *it = NULL;
|
|
|
|
rtpsession = GST_RTP_SESSION (parent);
|
|
|
|
GST_RTP_SESSION_LOCK (rtpsession);
|
|
if (pad == rtpsession->recv_rtp_src) {
|
|
otherpad = gst_object_ref (rtpsession->recv_rtp_sink);
|
|
} else if (pad == rtpsession->recv_rtp_sink) {
|
|
otherpad = gst_object_ref (rtpsession->recv_rtp_src);
|
|
} else if (pad == rtpsession->send_rtp_src) {
|
|
otherpad = gst_object_ref (rtpsession->send_rtp_sink);
|
|
} else if (pad == rtpsession->send_rtp_sink) {
|
|
otherpad = gst_object_ref (rtpsession->send_rtp_src);
|
|
}
|
|
GST_RTP_SESSION_UNLOCK (rtpsession);
|
|
|
|
if (otherpad) {
|
|
GValue val = { 0, };
|
|
|
|
g_value_init (&val, GST_TYPE_PAD);
|
|
g_value_set_object (&val, otherpad);
|
|
it = gst_iterator_new_single (GST_TYPE_PAD, &val);
|
|
g_value_unset (&val);
|
|
gst_object_unref (otherpad);
|
|
} else {
|
|
it = gst_iterator_new_single (GST_TYPE_PAD, NULL);
|
|
}
|
|
|
|
return it;
|
|
}
|
|
|
|
static gboolean
|
|
gst_rtp_session_sink_setcaps (GstPad * pad, GstRtpSession * rtpsession,
|
|
GstCaps * caps)
|
|
{
|
|
GST_RTP_SESSION_LOCK (rtpsession);
|
|
gst_rtp_session_cache_caps (rtpsession, caps);
|
|
GST_RTP_SESSION_UNLOCK (rtpsession);
|
|
|
|
return TRUE;
|
|
}
|
|
|
|
/* receive a packet from a sender, send it to the RTP session manager and
|
|
* forward the packet on the rtp_src pad
|
|
*/
|
|
static GstFlowReturn
|
|
gst_rtp_session_chain_recv_rtp (GstPad * pad, GstObject * parent,
|
|
GstBuffer * buffer)
|
|
{
|
|
GstRtpSession *rtpsession;
|
|
GstRtpSessionPrivate *priv;
|
|
GstFlowReturn ret;
|
|
GstClockTime current_time, running_time;
|
|
GstClockTime timestamp;
|
|
guint64 ntpnstime;
|
|
|
|
rtpsession = GST_RTP_SESSION (parent);
|
|
priv = rtpsession->priv;
|
|
|
|
GST_LOG_OBJECT (rtpsession, "received RTP packet");
|
|
|
|
GST_RTP_SESSION_LOCK (rtpsession);
|
|
signal_waiting_rtcp_thread_unlocked (rtpsession);
|
|
GST_RTP_SESSION_UNLOCK (rtpsession);
|
|
|
|
/* get NTP time when this packet was captured, this depends on the timestamp. */
|
|
timestamp = GST_BUFFER_PTS (buffer);
|
|
if (GST_CLOCK_TIME_IS_VALID (timestamp)) {
|
|
/* convert to running time using the segment values */
|
|
running_time =
|
|
gst_segment_to_running_time (&rtpsession->recv_rtp_seg, GST_FORMAT_TIME,
|
|
timestamp);
|
|
ntpnstime = GST_CLOCK_TIME_NONE;
|
|
} else {
|
|
get_current_times (rtpsession, &running_time, &ntpnstime);
|
|
}
|
|
current_time = gst_clock_get_time (priv->sysclock);
|
|
|
|
ret = rtp_session_process_rtp (priv->session, buffer, current_time,
|
|
running_time, ntpnstime);
|
|
if (ret != GST_FLOW_OK)
|
|
goto push_error;
|
|
|
|
done:
|
|
|
|
return ret;
|
|
|
|
/* ERRORS */
|
|
push_error:
|
|
{
|
|
GST_DEBUG_OBJECT (rtpsession, "process returned %s",
|
|
gst_flow_get_name (ret));
|
|
goto done;
|
|
}
|
|
}
|
|
|
|
static gboolean
|
|
process_received_buffer_in_list (GstBuffer ** buffer, guint idx, gpointer data)
|
|
{
|
|
gint ret;
|
|
|
|
ret = gst_rtp_session_chain_recv_rtp (NULL, data, *buffer);
|
|
if (ret != GST_FLOW_OK)
|
|
GST_ERROR ("Processing individual buffer in a list failed");
|
|
|
|
/*
|
|
* The buffer has been processed, remove it from the original list, if it was
|
|
* a valid RTP buffer it has been added to the "processed" list in
|
|
* gst_rtp_session_process_rtp().
|
|
*/
|
|
*buffer = NULL;
|
|
return TRUE;
|
|
}
|
|
|
|
static GstFlowReturn
|
|
gst_rtp_session_chain_recv_rtp_list (GstPad * pad, GstObject * parent,
|
|
GstBufferList * list)
|
|
{
|
|
GstRtpSession *rtpsession = GST_RTP_SESSION (parent);
|
|
GstBufferList *processed_list;
|
|
|
|
processed_list = gst_buffer_list_new ();
|
|
|
|
/* Set some private data to detect that a buffer list is being pushed. */
|
|
rtpsession->priv->processed_list = processed_list;
|
|
|
|
/*
|
|
* Individually process the buffers from the incoming buffer list as the
|
|
* incoming RTP packets in the list can be mixed in all sorts of ways:
|
|
* - different frames,
|
|
* - different sources,
|
|
* - different types (RTP or RTCP)
|
|
*/
|
|
gst_buffer_list_foreach (list,
|
|
(GstBufferListFunc) process_received_buffer_in_list, parent);
|
|
|
|
gst_buffer_list_unref (list);
|
|
|
|
/* Clean up private data in case the next push does not use a buffer list. */
|
|
rtpsession->priv->processed_list = NULL;
|
|
|
|
if (gst_buffer_list_length (processed_list) == 0 || !rtpsession->recv_rtp_src) {
|
|
gst_buffer_list_unref (processed_list);
|
|
return GST_FLOW_OK;
|
|
}
|
|
|
|
GST_LOG_OBJECT (rtpsession, "pushing received RTP list");
|
|
return gst_pad_push_list (rtpsession->recv_rtp_src, processed_list);
|
|
}
|
|
|
|
static gboolean
|
|
gst_rtp_session_event_recv_rtcp_sink (GstPad * pad, GstObject * parent,
|
|
GstEvent * event)
|
|
{
|
|
GstRtpSession *rtpsession;
|
|
gboolean ret = FALSE;
|
|
|
|
rtpsession = GST_RTP_SESSION (parent);
|
|
|
|
GST_DEBUG_OBJECT (rtpsession, "received event %s",
|
|
GST_EVENT_TYPE_NAME (event));
|
|
|
|
switch (GST_EVENT_TYPE (event)) {
|
|
case GST_EVENT_SEGMENT:
|
|
/* Make sure that the sync_src pad has caps before the segment event.
|
|
* Otherwise we might get a segment event before caps from the receive
|
|
* RTCP pad, and then later when receiving RTCP packets will set caps.
|
|
* This will results in a sticky event misordering warning
|
|
*/
|
|
if (!gst_pad_has_current_caps (rtpsession->sync_src)) {
|
|
GstCaps *caps = gst_caps_new_empty_simple ("application/x-rtcp");
|
|
gst_pad_set_caps (rtpsession->sync_src, caps);
|
|
gst_caps_unref (caps);
|
|
}
|
|
/* fall through */
|
|
default:
|
|
ret = gst_pad_push_event (rtpsession->sync_src, event);
|
|
break;
|
|
}
|
|
|
|
return ret;
|
|
}
|
|
|
|
/* Receive an RTCP packet from a sender, send it to the RTP session manager and
|
|
* forward the SR packets to the sync_src pad.
|
|
*/
|
|
static GstFlowReturn
|
|
gst_rtp_session_chain_recv_rtcp (GstPad * pad, GstObject * parent,
|
|
GstBuffer * buffer)
|
|
{
|
|
GstRtpSession *rtpsession;
|
|
GstRtpSessionPrivate *priv;
|
|
GstClockTime current_time;
|
|
GstClockTime running_time;
|
|
guint64 ntpnstime;
|
|
|
|
rtpsession = GST_RTP_SESSION (parent);
|
|
priv = rtpsession->priv;
|
|
|
|
GST_LOG_OBJECT (rtpsession, "received RTCP packet");
|
|
|
|
GST_RTP_SESSION_LOCK (rtpsession);
|
|
signal_waiting_rtcp_thread_unlocked (rtpsession);
|
|
GST_RTP_SESSION_UNLOCK (rtpsession);
|
|
|
|
current_time = gst_clock_get_time (priv->sysclock);
|
|
get_current_times (rtpsession, &running_time, &ntpnstime);
|
|
|
|
rtp_session_process_rtcp (priv->session, buffer, current_time, running_time,
|
|
ntpnstime);
|
|
|
|
return GST_FLOW_OK; /* always return OK */
|
|
}
|
|
|
|
static gboolean
|
|
gst_rtp_session_query_send_rtcp_src (GstPad * pad, GstObject * parent,
|
|
GstQuery * query)
|
|
{
|
|
GstRtpSession *rtpsession;
|
|
gboolean ret = FALSE;
|
|
|
|
rtpsession = GST_RTP_SESSION (parent);
|
|
|
|
GST_DEBUG_OBJECT (rtpsession, "received QUERY %s",
|
|
GST_QUERY_TYPE_NAME (query));
|
|
|
|
switch (GST_QUERY_TYPE (query)) {
|
|
case GST_QUERY_LATENCY:
|
|
ret = TRUE;
|
|
/* use the defaults for the latency query. */
|
|
gst_query_set_latency (query, FALSE, 0, -1);
|
|
break;
|
|
default:
|
|
/* other queries simply fail for now */
|
|
break;
|
|
}
|
|
|
|
return ret;
|
|
}
|
|
|
|
static gboolean
|
|
gst_rtp_session_event_send_rtcp_src (GstPad * pad, GstObject * parent,
|
|
GstEvent * event)
|
|
{
|
|
GstRtpSession *rtpsession;
|
|
gboolean ret = TRUE;
|
|
|
|
rtpsession = GST_RTP_SESSION (parent);
|
|
GST_DEBUG_OBJECT (rtpsession, "received EVENT %s",
|
|
GST_EVENT_TYPE_NAME (event));
|
|
|
|
switch (GST_EVENT_TYPE (event)) {
|
|
case GST_EVENT_SEEK:
|
|
case GST_EVENT_LATENCY:
|
|
gst_event_unref (event);
|
|
ret = TRUE;
|
|
break;
|
|
default:
|
|
/* other events simply fail for now */
|
|
gst_event_unref (event);
|
|
ret = FALSE;
|
|
break;
|
|
}
|
|
|
|
return ret;
|
|
}
|
|
|
|
|
|
static gboolean
|
|
gst_rtp_session_event_send_rtp_sink (GstPad * pad, GstObject * parent,
|
|
GstEvent * event)
|
|
{
|
|
GstRtpSession *rtpsession;
|
|
gboolean ret = FALSE;
|
|
|
|
rtpsession = GST_RTP_SESSION (parent);
|
|
|
|
GST_DEBUG_OBJECT (rtpsession, "received EVENT %s",
|
|
GST_EVENT_TYPE_NAME (event));
|
|
|
|
switch (GST_EVENT_TYPE (event)) {
|
|
case GST_EVENT_CAPS:
|
|
{
|
|
GstCaps *caps;
|
|
|
|
/* process */
|
|
gst_event_parse_caps (event, &caps);
|
|
gst_rtp_session_setcaps_send_rtp (pad, rtpsession, caps);
|
|
ret = gst_pad_push_event (rtpsession->send_rtp_src, event);
|
|
break;
|
|
}
|
|
case GST_EVENT_FLUSH_STOP:
|
|
gst_segment_init (&rtpsession->send_rtp_seg, GST_FORMAT_UNDEFINED);
|
|
ret = gst_pad_push_event (rtpsession->send_rtp_src, event);
|
|
break;
|
|
case GST_EVENT_SEGMENT:{
|
|
GstSegment *segment, in_segment;
|
|
|
|
segment = &rtpsession->send_rtp_seg;
|
|
|
|
/* the newsegment event is needed to convert the RTP timestamp to
|
|
* running_time, which is needed to generate a mapping from RTP to NTP
|
|
* timestamps in SR reports */
|
|
gst_event_copy_segment (event, &in_segment);
|
|
GST_DEBUG_OBJECT (rtpsession, "received segment %" GST_SEGMENT_FORMAT,
|
|
&in_segment);
|
|
|
|
/* accept upstream */
|
|
gst_segment_copy_into (&in_segment, segment);
|
|
|
|
/* push event forward */
|
|
ret = gst_pad_push_event (rtpsession->send_rtp_src, event);
|
|
break;
|
|
}
|
|
case GST_EVENT_EOS:{
|
|
GstClockTime current_time;
|
|
|
|
/* push downstream FIXME, we are not supposed to leave the session just
|
|
* because we stop sending. */
|
|
ret = gst_pad_push_event (rtpsession->send_rtp_src, event);
|
|
current_time = gst_clock_get_time (rtpsession->priv->sysclock);
|
|
|
|
GST_DEBUG_OBJECT (rtpsession, "scheduling BYE message");
|
|
rtp_session_mark_all_bye (rtpsession->priv->session, "End Of Stream");
|
|
rtp_session_schedule_bye (rtpsession->priv->session, current_time);
|
|
break;
|
|
}
|
|
default:{
|
|
GstPad *send_rtp_src;
|
|
|
|
GST_RTP_SESSION_LOCK (rtpsession);
|
|
if ((send_rtp_src = rtpsession->send_rtp_src))
|
|
gst_object_ref (send_rtp_src);
|
|
GST_RTP_SESSION_UNLOCK (rtpsession);
|
|
|
|
if (send_rtp_src) {
|
|
ret = gst_pad_push_event (send_rtp_src, event);
|
|
gst_object_unref (send_rtp_src);
|
|
} else
|
|
gst_event_unref (event);
|
|
|
|
break;
|
|
}
|
|
}
|
|
|
|
return ret;
|
|
}
|
|
|
|
static gboolean
|
|
gst_rtp_session_event_send_rtp_src (GstPad * pad, GstObject * parent,
|
|
GstEvent * event)
|
|
{
|
|
GstRtpSession *rtpsession;
|
|
gboolean ret = FALSE;
|
|
|
|
rtpsession = GST_RTP_SESSION (parent);
|
|
|
|
GST_DEBUG_OBJECT (rtpsession, "received EVENT %s",
|
|
GST_EVENT_TYPE_NAME (event));
|
|
|
|
switch (GST_EVENT_TYPE (event)) {
|
|
case GST_EVENT_LATENCY:
|
|
/* save the latency, we need this to know when an RTP packet will be
|
|
* rendered by the sink */
|
|
gst_event_parse_latency (event, &rtpsession->priv->send_latency);
|
|
|
|
ret = gst_pad_event_default (pad, parent, event);
|
|
break;
|
|
default:
|
|
ret = gst_pad_event_default (pad, parent, event);
|
|
break;
|
|
}
|
|
return ret;
|
|
}
|
|
|
|
static GstCaps *
|
|
gst_rtp_session_getcaps_send_rtp (GstPad * pad, GstRtpSession * rtpsession,
|
|
GstCaps * filter)
|
|
{
|
|
GstRtpSessionPrivate *priv;
|
|
GstCaps *result;
|
|
GstStructure *s1, *s2;
|
|
guint ssrc;
|
|
gboolean is_random;
|
|
|
|
priv = rtpsession->priv;
|
|
|
|
ssrc = rtp_session_suggest_ssrc (priv->session, &is_random);
|
|
|
|
/* we can basically accept anything but we prefer to receive packets with our
|
|
* internal SSRC so that we don't have to patch it. Create a structure with
|
|
* the SSRC and another one without.
|
|
* Only do this if the session actually decided on an ssrc already,
|
|
* otherwise we give upstream the opportunity to select an ssrc itself */
|
|
if (!is_random) {
|
|
s1 = gst_structure_new ("application/x-rtp", "ssrc", G_TYPE_UINT, ssrc,
|
|
NULL);
|
|
s2 = gst_structure_new_empty ("application/x-rtp");
|
|
|
|
result = gst_caps_new_full (s1, s2, NULL);
|
|
} else {
|
|
result = gst_caps_new_empty_simple ("application/x-rtp");
|
|
}
|
|
|
|
if (filter) {
|
|
GstCaps *caps = result;
|
|
|
|
result = gst_caps_intersect_full (filter, caps, GST_CAPS_INTERSECT_FIRST);
|
|
gst_caps_unref (caps);
|
|
}
|
|
|
|
GST_DEBUG_OBJECT (rtpsession, "getting caps %" GST_PTR_FORMAT, result);
|
|
|
|
return result;
|
|
}
|
|
|
|
static gboolean
|
|
gst_rtp_session_query_send_rtp (GstPad * pad, GstObject * parent,
|
|
GstQuery * query)
|
|
{
|
|
gboolean res = FALSE;
|
|
GstRtpSession *rtpsession;
|
|
|
|
rtpsession = GST_RTP_SESSION (parent);
|
|
|
|
switch (GST_QUERY_TYPE (query)) {
|
|
case GST_QUERY_CAPS:
|
|
{
|
|
GstCaps *filter, *caps;
|
|
|
|
gst_query_parse_caps (query, &filter);
|
|
caps = gst_rtp_session_getcaps_send_rtp (pad, rtpsession, filter);
|
|
gst_query_set_caps_result (query, caps);
|
|
gst_caps_unref (caps);
|
|
res = TRUE;
|
|
break;
|
|
}
|
|
default:
|
|
res = gst_pad_query_default (pad, parent, query);
|
|
break;
|
|
}
|
|
|
|
return res;
|
|
}
|
|
|
|
static gboolean
|
|
gst_rtp_session_setcaps_send_rtp (GstPad * pad, GstRtpSession * rtpsession,
|
|
GstCaps * caps)
|
|
{
|
|
GstRtpSessionPrivate *priv;
|
|
|
|
priv = rtpsession->priv;
|
|
|
|
rtp_session_update_send_caps (priv->session, caps);
|
|
|
|
return TRUE;
|
|
}
|
|
|
|
/* Receive an RTP packet or a list of packets to be sent to the receivers,
|
|
* send to RTP session manager and forward to send_rtp_src.
|
|
*/
|
|
static GstFlowReturn
|
|
gst_rtp_session_chain_send_rtp_common (GstRtpSession * rtpsession,
|
|
gpointer data, gboolean is_list)
|
|
{
|
|
GstRtpSessionPrivate *priv;
|
|
GstFlowReturn ret;
|
|
GstClockTime timestamp, running_time;
|
|
GstClockTime current_time;
|
|
|
|
priv = rtpsession->priv;
|
|
|
|
GST_LOG_OBJECT (rtpsession, "received RTP %s", is_list ? "list" : "packet");
|
|
|
|
/* get NTP time when this packet was captured, this depends on the timestamp. */
|
|
if (is_list) {
|
|
GstBuffer *buffer = NULL;
|
|
|
|
/* All buffers in a list have the same timestamp.
|
|
* So, just take it from the first buffer. */
|
|
buffer = gst_buffer_list_get (GST_BUFFER_LIST_CAST (data), 0);
|
|
if (buffer)
|
|
timestamp = GST_BUFFER_PTS (buffer);
|
|
else
|
|
timestamp = -1;
|
|
} else {
|
|
timestamp = GST_BUFFER_PTS (GST_BUFFER_CAST (data));
|
|
}
|
|
|
|
if (GST_CLOCK_TIME_IS_VALID (timestamp)) {
|
|
/* convert to running time using the segment start value. */
|
|
running_time =
|
|
gst_segment_to_running_time (&rtpsession->send_rtp_seg, GST_FORMAT_TIME,
|
|
timestamp);
|
|
if (priv->rtcp_sync_send_time)
|
|
running_time += priv->send_latency;
|
|
} else {
|
|
/* no timestamp. */
|
|
running_time = -1;
|
|
}
|
|
|
|
current_time = gst_clock_get_time (priv->sysclock);
|
|
ret = rtp_session_send_rtp (priv->session, data, is_list, current_time,
|
|
running_time);
|
|
if (ret != GST_FLOW_OK)
|
|
goto push_error;
|
|
|
|
done:
|
|
|
|
return ret;
|
|
|
|
/* ERRORS */
|
|
push_error:
|
|
{
|
|
GST_DEBUG_OBJECT (rtpsession, "process returned %s",
|
|
gst_flow_get_name (ret));
|
|
goto done;
|
|
}
|
|
}
|
|
|
|
static GstFlowReturn
|
|
gst_rtp_session_chain_send_rtp (GstPad * pad, GstObject * parent,
|
|
GstBuffer * buffer)
|
|
{
|
|
GstRtpSession *rtpsession = GST_RTP_SESSION (parent);
|
|
|
|
return gst_rtp_session_chain_send_rtp_common (rtpsession, buffer, FALSE);
|
|
}
|
|
|
|
static GstFlowReturn
|
|
gst_rtp_session_chain_send_rtp_list (GstPad * pad, GstObject * parent,
|
|
GstBufferList * list)
|
|
{
|
|
GstRtpSession *rtpsession = GST_RTP_SESSION (parent);
|
|
|
|
return gst_rtp_session_chain_send_rtp_common (rtpsession, list, TRUE);
|
|
}
|
|
|
|
/* Create sinkpad to receive RTP packets from senders. This will also create a
|
|
* srcpad for the RTP packets.
|
|
*/
|
|
static GstPad *
|
|
create_recv_rtp_sink (GstRtpSession * rtpsession)
|
|
{
|
|
GST_DEBUG_OBJECT (rtpsession, "creating RTP sink pad");
|
|
|
|
rtpsession->recv_rtp_sink =
|
|
gst_pad_new_from_static_template (&rtpsession_recv_rtp_sink_template,
|
|
"recv_rtp_sink");
|
|
gst_pad_set_chain_function (rtpsession->recv_rtp_sink,
|
|
gst_rtp_session_chain_recv_rtp);
|
|
gst_pad_set_chain_list_function (rtpsession->recv_rtp_sink,
|
|
gst_rtp_session_chain_recv_rtp_list);
|
|
gst_pad_set_event_function (rtpsession->recv_rtp_sink,
|
|
gst_rtp_session_event_recv_rtp_sink);
|
|
gst_pad_set_iterate_internal_links_function (rtpsession->recv_rtp_sink,
|
|
gst_rtp_session_iterate_internal_links);
|
|
GST_PAD_SET_PROXY_ALLOCATION (rtpsession->recv_rtp_sink);
|
|
gst_pad_set_active (rtpsession->recv_rtp_sink, TRUE);
|
|
gst_element_add_pad (GST_ELEMENT_CAST (rtpsession),
|
|
rtpsession->recv_rtp_sink);
|
|
|
|
GST_DEBUG_OBJECT (rtpsession, "creating RTP src pad");
|
|
rtpsession->recv_rtp_src =
|
|
gst_pad_new_from_static_template (&rtpsession_recv_rtp_src_template,
|
|
"recv_rtp_src");
|
|
gst_pad_set_event_function (rtpsession->recv_rtp_src,
|
|
gst_rtp_session_event_recv_rtp_src);
|
|
gst_pad_set_iterate_internal_links_function (rtpsession->recv_rtp_src,
|
|
gst_rtp_session_iterate_internal_links);
|
|
gst_pad_use_fixed_caps (rtpsession->recv_rtp_src);
|
|
gst_pad_set_active (rtpsession->recv_rtp_src, TRUE);
|
|
gst_element_add_pad (GST_ELEMENT_CAST (rtpsession), rtpsession->recv_rtp_src);
|
|
|
|
return rtpsession->recv_rtp_sink;
|
|
}
|
|
|
|
/* Remove sinkpad to receive RTP packets from senders. This will also remove
|
|
* the srcpad for the RTP packets.
|
|
*/
|
|
static void
|
|
remove_recv_rtp_sink (GstRtpSession * rtpsession)
|
|
{
|
|
GST_DEBUG_OBJECT (rtpsession, "removing RTP sink pad");
|
|
|
|
/* deactivate from source to sink */
|
|
gst_pad_set_active (rtpsession->recv_rtp_src, FALSE);
|
|
gst_pad_set_active (rtpsession->recv_rtp_sink, FALSE);
|
|
|
|
/* remove pads */
|
|
gst_element_remove_pad (GST_ELEMENT_CAST (rtpsession),
|
|
rtpsession->recv_rtp_sink);
|
|
rtpsession->recv_rtp_sink = NULL;
|
|
|
|
GST_DEBUG_OBJECT (rtpsession, "removing RTP src pad");
|
|
gst_element_remove_pad (GST_ELEMENT_CAST (rtpsession),
|
|
rtpsession->recv_rtp_src);
|
|
rtpsession->recv_rtp_src = NULL;
|
|
}
|
|
|
|
/* Create a sinkpad to receive RTCP messages from senders, this will also create a
|
|
* sync_src pad for the SR packets.
|
|
*/
|
|
static GstPad *
|
|
create_recv_rtcp_sink (GstRtpSession * rtpsession)
|
|
{
|
|
GST_DEBUG_OBJECT (rtpsession, "creating RTCP sink pad");
|
|
|
|
rtpsession->recv_rtcp_sink =
|
|
gst_pad_new_from_static_template (&rtpsession_recv_rtcp_sink_template,
|
|
"recv_rtcp_sink");
|
|
gst_pad_set_chain_function (rtpsession->recv_rtcp_sink,
|
|
gst_rtp_session_chain_recv_rtcp);
|
|
gst_pad_set_event_function (rtpsession->recv_rtcp_sink,
|
|
gst_rtp_session_event_recv_rtcp_sink);
|
|
gst_pad_set_iterate_internal_links_function (rtpsession->recv_rtcp_sink,
|
|
gst_rtp_session_iterate_internal_links);
|
|
gst_pad_set_active (rtpsession->recv_rtcp_sink, TRUE);
|
|
gst_element_add_pad (GST_ELEMENT_CAST (rtpsession),
|
|
rtpsession->recv_rtcp_sink);
|
|
|
|
GST_DEBUG_OBJECT (rtpsession, "creating sync src pad");
|
|
rtpsession->sync_src =
|
|
gst_pad_new_from_static_template (&rtpsession_sync_src_template,
|
|
"sync_src");
|
|
gst_pad_set_iterate_internal_links_function (rtpsession->sync_src,
|
|
gst_rtp_session_iterate_internal_links);
|
|
gst_pad_use_fixed_caps (rtpsession->sync_src);
|
|
gst_pad_set_active (rtpsession->sync_src, TRUE);
|
|
gst_element_add_pad (GST_ELEMENT_CAST (rtpsession), rtpsession->sync_src);
|
|
|
|
return rtpsession->recv_rtcp_sink;
|
|
}
|
|
|
|
static void
|
|
remove_recv_rtcp_sink (GstRtpSession * rtpsession)
|
|
{
|
|
GST_DEBUG_OBJECT (rtpsession, "removing RTCP sink pad");
|
|
|
|
gst_pad_set_active (rtpsession->sync_src, FALSE);
|
|
gst_pad_set_active (rtpsession->recv_rtcp_sink, FALSE);
|
|
|
|
gst_element_remove_pad (GST_ELEMENT_CAST (rtpsession),
|
|
rtpsession->recv_rtcp_sink);
|
|
rtpsession->recv_rtcp_sink = NULL;
|
|
|
|
GST_DEBUG_OBJECT (rtpsession, "removing sync src pad");
|
|
gst_element_remove_pad (GST_ELEMENT_CAST (rtpsession), rtpsession->sync_src);
|
|
rtpsession->sync_src = NULL;
|
|
}
|
|
|
|
/* Create a sinkpad to receive RTP packets for receivers. This will also create a
|
|
* send_rtp_src pad.
|
|
*/
|
|
static GstPad *
|
|
create_send_rtp_sink (GstRtpSession * rtpsession)
|
|
{
|
|
GST_DEBUG_OBJECT (rtpsession, "creating pad");
|
|
|
|
rtpsession->send_rtp_sink =
|
|
gst_pad_new_from_static_template (&rtpsession_send_rtp_sink_template,
|
|
"send_rtp_sink");
|
|
gst_pad_set_chain_function (rtpsession->send_rtp_sink,
|
|
gst_rtp_session_chain_send_rtp);
|
|
gst_pad_set_chain_list_function (rtpsession->send_rtp_sink,
|
|
gst_rtp_session_chain_send_rtp_list);
|
|
gst_pad_set_query_function (rtpsession->send_rtp_sink,
|
|
gst_rtp_session_query_send_rtp);
|
|
gst_pad_set_event_function (rtpsession->send_rtp_sink,
|
|
gst_rtp_session_event_send_rtp_sink);
|
|
gst_pad_set_iterate_internal_links_function (rtpsession->send_rtp_sink,
|
|
gst_rtp_session_iterate_internal_links);
|
|
GST_PAD_SET_PROXY_CAPS (rtpsession->send_rtp_sink);
|
|
GST_PAD_SET_PROXY_ALLOCATION (rtpsession->send_rtp_sink);
|
|
gst_pad_set_active (rtpsession->send_rtp_sink, TRUE);
|
|
gst_element_add_pad (GST_ELEMENT_CAST (rtpsession),
|
|
rtpsession->send_rtp_sink);
|
|
|
|
rtpsession->send_rtp_src =
|
|
gst_pad_new_from_static_template (&rtpsession_send_rtp_src_template,
|
|
"send_rtp_src");
|
|
gst_pad_set_iterate_internal_links_function (rtpsession->send_rtp_src,
|
|
gst_rtp_session_iterate_internal_links);
|
|
gst_pad_set_event_function (rtpsession->send_rtp_src,
|
|
gst_rtp_session_event_send_rtp_src);
|
|
GST_PAD_SET_PROXY_CAPS (rtpsession->send_rtp_src);
|
|
gst_pad_set_active (rtpsession->send_rtp_src, TRUE);
|
|
gst_element_add_pad (GST_ELEMENT_CAST (rtpsession), rtpsession->send_rtp_src);
|
|
|
|
return rtpsession->send_rtp_sink;
|
|
}
|
|
|
|
static void
|
|
remove_send_rtp_sink (GstRtpSession * rtpsession)
|
|
{
|
|
GST_DEBUG_OBJECT (rtpsession, "removing pad");
|
|
|
|
gst_pad_set_active (rtpsession->send_rtp_src, FALSE);
|
|
gst_pad_set_active (rtpsession->send_rtp_sink, FALSE);
|
|
|
|
gst_element_remove_pad (GST_ELEMENT_CAST (rtpsession),
|
|
rtpsession->send_rtp_sink);
|
|
rtpsession->send_rtp_sink = NULL;
|
|
|
|
gst_element_remove_pad (GST_ELEMENT_CAST (rtpsession),
|
|
rtpsession->send_rtp_src);
|
|
rtpsession->send_rtp_src = NULL;
|
|
}
|
|
|
|
/* Create a srcpad with the RTCP packets to send out.
|
|
* This pad will be driven by the RTP session manager when it wants to send out
|
|
* RTCP packets.
|
|
*/
|
|
static GstPad *
|
|
create_send_rtcp_src (GstRtpSession * rtpsession)
|
|
{
|
|
GST_DEBUG_OBJECT (rtpsession, "creating pad");
|
|
|
|
rtpsession->send_rtcp_src =
|
|
gst_pad_new_from_static_template (&rtpsession_send_rtcp_src_template,
|
|
"send_rtcp_src");
|
|
gst_pad_use_fixed_caps (rtpsession->send_rtcp_src);
|
|
gst_pad_set_active (rtpsession->send_rtcp_src, TRUE);
|
|
gst_pad_set_iterate_internal_links_function (rtpsession->send_rtcp_src,
|
|
gst_rtp_session_iterate_internal_links);
|
|
gst_pad_set_query_function (rtpsession->send_rtcp_src,
|
|
gst_rtp_session_query_send_rtcp_src);
|
|
gst_pad_set_event_function (rtpsession->send_rtcp_src,
|
|
gst_rtp_session_event_send_rtcp_src);
|
|
gst_element_add_pad (GST_ELEMENT_CAST (rtpsession),
|
|
rtpsession->send_rtcp_src);
|
|
|
|
return rtpsession->send_rtcp_src;
|
|
}
|
|
|
|
static void
|
|
remove_send_rtcp_src (GstRtpSession * rtpsession)
|
|
{
|
|
GST_DEBUG_OBJECT (rtpsession, "removing pad");
|
|
|
|
gst_pad_set_active (rtpsession->send_rtcp_src, FALSE);
|
|
|
|
gst_element_remove_pad (GST_ELEMENT_CAST (rtpsession),
|
|
rtpsession->send_rtcp_src);
|
|
rtpsession->send_rtcp_src = NULL;
|
|
}
|
|
|
|
static GstPad *
|
|
gst_rtp_session_request_new_pad (GstElement * element,
|
|
GstPadTemplate * templ, const gchar * name, const GstCaps * caps)
|
|
{
|
|
GstRtpSession *rtpsession;
|
|
GstElementClass *klass;
|
|
GstPad *result;
|
|
|
|
g_return_val_if_fail (templ != NULL, NULL);
|
|
g_return_val_if_fail (GST_IS_RTP_SESSION (element), NULL);
|
|
|
|
rtpsession = GST_RTP_SESSION (element);
|
|
klass = GST_ELEMENT_GET_CLASS (element);
|
|
|
|
GST_DEBUG_OBJECT (element, "requesting pad %s", GST_STR_NULL (name));
|
|
|
|
GST_RTP_SESSION_LOCK (rtpsession);
|
|
|
|
/* figure out the template */
|
|
if (templ == gst_element_class_get_pad_template (klass, "recv_rtp_sink")) {
|
|
if (rtpsession->recv_rtp_sink != NULL)
|
|
goto exists;
|
|
|
|
result = create_recv_rtp_sink (rtpsession);
|
|
} else if (templ == gst_element_class_get_pad_template (klass,
|
|
"recv_rtcp_sink")) {
|
|
if (rtpsession->recv_rtcp_sink != NULL)
|
|
goto exists;
|
|
|
|
result = create_recv_rtcp_sink (rtpsession);
|
|
} else if (templ == gst_element_class_get_pad_template (klass,
|
|
"send_rtp_sink")) {
|
|
if (rtpsession->send_rtp_sink != NULL)
|
|
goto exists;
|
|
|
|
result = create_send_rtp_sink (rtpsession);
|
|
} else if (templ == gst_element_class_get_pad_template (klass,
|
|
"send_rtcp_src")) {
|
|
if (rtpsession->send_rtcp_src != NULL)
|
|
goto exists;
|
|
|
|
result = create_send_rtcp_src (rtpsession);
|
|
} else
|
|
goto wrong_template;
|
|
|
|
GST_RTP_SESSION_UNLOCK (rtpsession);
|
|
|
|
return result;
|
|
|
|
/* ERRORS */
|
|
wrong_template:
|
|
{
|
|
GST_RTP_SESSION_UNLOCK (rtpsession);
|
|
g_warning ("rtpsession: this is not our template");
|
|
return NULL;
|
|
}
|
|
exists:
|
|
{
|
|
GST_RTP_SESSION_UNLOCK (rtpsession);
|
|
g_warning ("rtpsession: pad already requested");
|
|
return NULL;
|
|
}
|
|
}
|
|
|
|
static void
|
|
gst_rtp_session_release_pad (GstElement * element, GstPad * pad)
|
|
{
|
|
GstRtpSession *rtpsession;
|
|
|
|
g_return_if_fail (GST_IS_RTP_SESSION (element));
|
|
g_return_if_fail (GST_IS_PAD (pad));
|
|
|
|
rtpsession = GST_RTP_SESSION (element);
|
|
|
|
GST_DEBUG_OBJECT (element, "releasing pad %s:%s", GST_DEBUG_PAD_NAME (pad));
|
|
|
|
GST_RTP_SESSION_LOCK (rtpsession);
|
|
|
|
if (rtpsession->recv_rtp_sink == pad) {
|
|
remove_recv_rtp_sink (rtpsession);
|
|
} else if (rtpsession->recv_rtcp_sink == pad) {
|
|
remove_recv_rtcp_sink (rtpsession);
|
|
} else if (rtpsession->send_rtp_sink == pad) {
|
|
remove_send_rtp_sink (rtpsession);
|
|
} else if (rtpsession->send_rtcp_src == pad) {
|
|
remove_send_rtcp_src (rtpsession);
|
|
} else
|
|
goto wrong_pad;
|
|
|
|
GST_RTP_SESSION_UNLOCK (rtpsession);
|
|
|
|
return;
|
|
|
|
/* ERRORS */
|
|
wrong_pad:
|
|
{
|
|
GST_RTP_SESSION_UNLOCK (rtpsession);
|
|
g_warning ("rtpsession: asked to release an unknown pad");
|
|
return;
|
|
}
|
|
}
|
|
|
|
static void
|
|
gst_rtp_session_request_key_unit (RTPSession * sess,
|
|
guint32 ssrc, gboolean all_headers, gpointer user_data)
|
|
{
|
|
GstRtpSession *rtpsession = GST_RTP_SESSION (user_data);
|
|
GstEvent *event;
|
|
GstPad *send_rtp_sink;
|
|
|
|
GST_RTP_SESSION_LOCK (rtpsession);
|
|
if ((send_rtp_sink = rtpsession->send_rtp_sink))
|
|
gst_object_ref (send_rtp_sink);
|
|
GST_RTP_SESSION_UNLOCK (rtpsession);
|
|
|
|
if (send_rtp_sink) {
|
|
event = gst_event_new_custom (GST_EVENT_CUSTOM_UPSTREAM,
|
|
gst_structure_new ("GstForceKeyUnit", "ssrc", G_TYPE_UINT, ssrc,
|
|
"all-headers", G_TYPE_BOOLEAN, all_headers, NULL));
|
|
gst_pad_push_event (send_rtp_sink, event);
|
|
gst_object_unref (send_rtp_sink);
|
|
}
|
|
}
|
|
|
|
static GstClockTime
|
|
gst_rtp_session_request_time (RTPSession * session, gpointer user_data)
|
|
{
|
|
GstRtpSession *rtpsession = GST_RTP_SESSION (user_data);
|
|
|
|
return gst_clock_get_time (rtpsession->priv->sysclock);
|
|
}
|
|
|
|
static void
|
|
gst_rtp_session_notify_nack (RTPSession * sess, guint16 seqnum,
|
|
guint16 blp, guint32 ssrc, gpointer user_data)
|
|
{
|
|
GstRtpSession *rtpsession = GST_RTP_SESSION (user_data);
|
|
GstEvent *event;
|
|
GstPad *send_rtp_sink;
|
|
|
|
GST_RTP_SESSION_LOCK (rtpsession);
|
|
if ((send_rtp_sink = rtpsession->send_rtp_sink))
|
|
gst_object_ref (send_rtp_sink);
|
|
GST_RTP_SESSION_UNLOCK (rtpsession);
|
|
|
|
if (send_rtp_sink) {
|
|
while (TRUE) {
|
|
event = gst_event_new_custom (GST_EVENT_CUSTOM_UPSTREAM,
|
|
gst_structure_new ("GstRTPRetransmissionRequest",
|
|
"seqnum", G_TYPE_UINT, (guint) seqnum,
|
|
"ssrc", G_TYPE_UINT, (guint) ssrc, NULL));
|
|
gst_pad_push_event (send_rtp_sink, event);
|
|
|
|
GST_RTP_SESSION_LOCK (rtpsession);
|
|
rtpsession->priv->sent_rtx_req_count++;
|
|
GST_RTP_SESSION_UNLOCK (rtpsession);
|
|
|
|
if (blp == 0)
|
|
break;
|
|
|
|
seqnum++;
|
|
while ((blp & 1) == 0) {
|
|
seqnum++;
|
|
blp >>= 1;
|
|
}
|
|
blp >>= 1;
|
|
}
|
|
gst_object_unref (send_rtp_sink);
|
|
}
|
|
}
|
|
|
|
static void
|
|
gst_rtp_session_notify_twcc (RTPSession * sess,
|
|
GstStructure * twcc_packets, GstStructure * twcc_stats, gpointer user_data)
|
|
{
|
|
GstRtpSession *rtpsession = GST_RTP_SESSION (user_data);
|
|
GstEvent *event;
|
|
GstPad *send_rtp_sink;
|
|
|
|
GST_RTP_SESSION_LOCK (rtpsession);
|
|
if ((send_rtp_sink = rtpsession->send_rtp_sink))
|
|
gst_object_ref (send_rtp_sink);
|
|
if (rtpsession->priv->last_twcc_stats)
|
|
gst_structure_free (rtpsession->priv->last_twcc_stats);
|
|
rtpsession->priv->last_twcc_stats = twcc_stats;
|
|
GST_RTP_SESSION_UNLOCK (rtpsession);
|
|
|
|
if (send_rtp_sink) {
|
|
event = gst_event_new_custom (GST_EVENT_CUSTOM_UPSTREAM, twcc_packets);
|
|
gst_pad_push_event (send_rtp_sink, event);
|
|
gst_object_unref (send_rtp_sink);
|
|
}
|
|
|
|
g_object_notify (G_OBJECT (rtpsession), "twcc-stats");
|
|
}
|
|
|
|
static void
|
|
gst_rtp_session_reconfigure (RTPSession * sess, gpointer user_data)
|
|
{
|
|
GstRtpSession *rtpsession = GST_RTP_SESSION (user_data);
|
|
GstPad *send_rtp_sink;
|
|
|
|
GST_RTP_SESSION_LOCK (rtpsession);
|
|
if ((send_rtp_sink = rtpsession->send_rtp_sink))
|
|
gst_object_ref (send_rtp_sink);
|
|
GST_RTP_SESSION_UNLOCK (rtpsession);
|
|
|
|
if (send_rtp_sink) {
|
|
gst_pad_push_event (send_rtp_sink, gst_event_new_reconfigure ());
|
|
gst_object_unref (send_rtp_sink);
|
|
}
|
|
}
|
|
|
|
static void
|
|
gst_rtp_session_notify_early_rtcp (RTPSession * sess, gpointer user_data)
|
|
{
|
|
GstRtpSession *rtpsession = GST_RTP_SESSION (user_data);
|
|
|
|
GST_DEBUG_OBJECT (rtpsession, "Notified of early RTCP");
|
|
/* with an early RTCP request, we might have to start the RTCP thread */
|
|
GST_RTP_SESSION_LOCK (rtpsession);
|
|
signal_waiting_rtcp_thread_unlocked (rtpsession);
|
|
GST_RTP_SESSION_UNLOCK (rtpsession);
|
|
}
|