mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer.git
synced 2024-11-26 03:31:05 +00:00
17db91c7c1
During RTP-Info synchronization, clock_base was temporarily switched from the actual clock-base to the base RTP time and then back some lines later. Instead directly work with the base RTP time. The comment about using a signed variable for convenience doesn't make any sense because all calculations done with the value are unsigned. Similarly, rtp_clock_base was overridden with the rtp_delta when calculating it, which was fine because it is not used anymore afterwards. Instead, introduce a new variable `rtp_delta` to make this calculation clearer. Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/6575>
5715 lines
179 KiB
C
5715 lines
179 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-rtpbin
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* @title: rtpbin
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* @see_also: rtpjitterbuffer, rtpsession, rtpptdemux, rtpssrcdemux
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*
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* RTP bin combines the functions of #GstRtpSession, #GstRtpSsrcDemux,
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* #GstRtpJitterBuffer and #GstRtpPtDemux in one element. It allows for multiple
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* RTP sessions that will be synchronized together using RTCP SR packets.
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*
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* #GstRtpBin is configured with a number of request pads that define the
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* functionality that is activated, similar to the #GstRtpSession element.
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*
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* To use #GstRtpBin as an RTP receiver, request a recv_rtp_sink_\%u pad. The session
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* number must be specified in the pad name.
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* Data received on the recv_rtp_sink_\%u pad will be processed in the #GstRtpSession
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* manager and after being validated forwarded on #GstRtpSsrcDemux element. Each
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* RTP stream is demuxed based on the SSRC and send to a #GstRtpJitterBuffer. After
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* the packets are released from the jitterbuffer, they will be forwarded to a
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* #GstRtpPtDemux element. The #GstRtpPtDemux element will demux the packets based
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* on the payload type and will create a unique pad recv_rtp_src_\%u_\%u_\%u on
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* rtpbin with the session number, SSRC and payload type respectively as the pad
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* name.
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*
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* To also use #GstRtpBin as an RTCP receiver, request a recv_rtcp_sink_\%u pad. The
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* session number must be specified in the pad name.
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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_\%u pad with the session number in the pad name. Packet pushed
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* on this pad contain SR/RR RTCP reports that should be sent to all participants
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* in the session.
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*
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* To use #GstRtpBin as a sender, request a send_rtp_sink_\%u pad, which will
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* automatically create a send_rtp_src_\%u pad. If the session number is not provided,
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* the pad from the lowest available session will be returned. The session manager will modify the
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* SSRC in the RTP packets to its own SSRC and will forward the packets on the
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* send_rtp_src_\%u 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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* Access to the internal statistics of rtpbin is provided with the
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* get-internal-session property. This action signal gives access to the
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* RTPSession object which further provides action signals to retrieve the
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* internal source and other sources.
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*
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* #GstRtpBin also has signals (#GstRtpBin::request-rtp-encoder,
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* #GstRtpBin::request-rtp-decoder, #GstRtpBin::request-rtcp-encoder and
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* #GstRtpBin::request-rtp-decoder) to dynamically request for RTP and RTCP encoders
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* and decoders in order to support SRTP. The encoders must provide the pads
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* rtp_sink_\%u and rtp_src_\%u for RTP and rtcp_sink_\%u and rtcp_src_\%u for
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* RTCP. The session number will be used in the pad name. The decoders must provide
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* rtp_sink and rtp_src for RTP and rtcp_sink and rtcp_src for RTCP. The decoders will
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* be placed before the #GstRtpSession element, thus they must support SSRC demuxing
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* internally.
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*
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* #GstRtpBin has signals (#GstRtpBin::request-aux-sender and
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* #GstRtpBin::request-aux-receiver to dynamically request an element that can be
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* used to create or merge additional RTP streams. AUX elements are needed to
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* implement FEC or retransmission (such as RFC 4588). An AUX sender must have one
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* sink_\%u pad that matches the sessionid in the signal and it should have 1 or
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* more src_\%u pads. For each src_%\u pad, a session will be made (if needed)
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* and the pad will be linked to the session send_rtp_sink pad. Each session will
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* then expose its source pad as send_rtp_src_\%u on #GstRtpBin.
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* An AUX receiver has 1 src_\%u pad that much match the sessionid in the signal
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* and 1 or more sink_\%u pads. A session will be made for each sink_\%u pad
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* when the corresponding recv_rtp_sink_\%u pad is requested on #GstRtpBin.
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* The #GstRtpBin::request-jitterbuffer signal can be used to provide a custom
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* element to perform arrival time smoothing, reordering and optionally packet
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* loss detection and retransmission requests.
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*
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* ## Example pipelines
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*
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* |[
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* gst-launch-1.0 udpsrc port=5000 caps="application/x-rtp, ..." ! .recv_rtp_sink_0 \
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* rtpbin ! rtptheoradepay ! theoradec ! xvimagesink
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* ]| Receive RTP data from port 5000 and send to the session 0 in rtpbin.
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* |[
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* gst-launch-1.0 rtpbin name=rtpbin \
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* v4l2src ! videoconvert ! ffenc_h263 ! rtph263ppay ! rtpbin.send_rtp_sink_0 \
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* rtpbin.send_rtp_src_0 ! udpsink port=5000 \
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* rtpbin.send_rtcp_src_0 ! udpsink port=5001 sync=false async=false \
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* udpsrc port=5005 ! rtpbin.recv_rtcp_sink_0 \
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* audiotestsrc ! amrnbenc ! rtpamrpay ! rtpbin.send_rtp_sink_1 \
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* rtpbin.send_rtp_src_1 ! udpsink port=5002 \
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* rtpbin.send_rtcp_src_1 ! udpsink port=5003 sync=false async=false \
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* udpsrc port=5007 ! rtpbin.recv_rtcp_sink_1
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* ]| Encode and payload H263 video captured from a v4l2src. Encode and payload AMR
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* audio generated from audiotestsrc. The video is sent to session 0 in rtpbin
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* and the audio is sent to session 1. Video packets are sent on UDP port 5000
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* and audio packets on port 5002. The video RTCP packets for session 0 are sent
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* on port 5001 and the audio RTCP packets for session 0 are sent on port 5003.
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* RTCP packets for session 0 are received on port 5005 and RTCP for session 1
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* is received on port 5007. Since RTCP packets from the sender should be sent
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* as soon as possible and do not participate in preroll, sync=false and
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* async=false is configured on udpsink
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* |[
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* gst-launch-1.0 -v rtpbin name=rtpbin \
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* udpsrc caps="application/x-rtp,media=(string)video,clock-rate=(int)90000,encoding-name=(string)H263-1998" \
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* port=5000 ! rtpbin.recv_rtp_sink_0 \
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* rtpbin. ! rtph263pdepay ! ffdec_h263 ! xvimagesink \
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* udpsrc port=5001 ! rtpbin.recv_rtcp_sink_0 \
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* rtpbin.send_rtcp_src_0 ! udpsink port=5005 sync=false async=false \
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* udpsrc caps="application/x-rtp,media=(string)audio,clock-rate=(int)8000,encoding-name=(string)AMR,encoding-params=(string)1,octet-align=(string)1" \
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* port=5002 ! rtpbin.recv_rtp_sink_1 \
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* rtpbin. ! rtpamrdepay ! amrnbdec ! alsasink \
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* udpsrc port=5003 ! rtpbin.recv_rtcp_sink_1 \
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* rtpbin.send_rtcp_src_1 ! udpsink port=5007 sync=false async=false
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* ]| Receive H263 on port 5000, send it through rtpbin in session 0, depayload,
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* decode and display the video.
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* Receive AMR on port 5002, send it through rtpbin in session 1, depayload,
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* decode and play the audio.
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* Receive server RTCP packets for session 0 on port 5001 and RTCP packets for
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* session 1 on port 5003. These packets will be used for session management and
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* synchronisation.
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* Send RTCP reports for session 0 on port 5005 and RTCP reports for session 1
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* on port 5007.
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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 <stdio.h>
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#include <string.h>
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#include <gst/rtp/gstrtpbuffer.h>
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#include <gst/rtp/gstrtcpbuffer.h>
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#include "gstrtpbin.h"
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#include "rtpsession.h"
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#include "gstrtpsession.h"
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#include "gstrtpjitterbuffer.h"
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#include "gstrtputils.h"
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#include <gst/glib-compat-private.h>
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GST_DEBUG_CATEGORY_STATIC (gst_rtp_bin_debug);
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#define GST_CAT_DEFAULT gst_rtp_bin_debug
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/* sink pads */
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static GstStaticPadTemplate rtpbin_recv_rtp_sink_template =
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GST_STATIC_PAD_TEMPLATE ("recv_rtp_sink_%u",
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GST_PAD_SINK,
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GST_PAD_REQUEST,
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GST_STATIC_CAPS ("application/x-rtp;application/x-srtp")
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);
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/**
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* GstRtpBin!recv_fec_sink_%u_%u:
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*
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* Sink template for receiving Forward Error Correction packets,
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* in the form recv_fec_sink_<session_idx>_<fec_stream_idx>
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*
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* See #GstRTPST_2022_1_FecDec for example usage
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*
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* Since: 1.20
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*/
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static GstStaticPadTemplate rtpbin_recv_fec_sink_template =
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GST_STATIC_PAD_TEMPLATE ("recv_fec_sink_%u_%u",
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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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/**
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* GstRtpBin!send_fec_src_%u_%u:
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*
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* Src template for sending Forward Error Correction packets,
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* in the form send_fec_src_<session_idx>_<fec_stream_idx>
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*
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* See #GstRTPST_2022_1_FecEnc for example usage
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*
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* Since: 1.20
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*/
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static GstStaticPadTemplate rtpbin_send_fec_src_template =
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GST_STATIC_PAD_TEMPLATE ("send_fec_src_%u_%u",
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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 rtpbin_recv_rtcp_sink_template =
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GST_STATIC_PAD_TEMPLATE ("recv_rtcp_sink_%u",
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GST_PAD_SINK,
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GST_PAD_REQUEST,
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GST_STATIC_CAPS ("application/x-rtcp;application/x-srtcp")
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);
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static GstStaticPadTemplate rtpbin_send_rtp_sink_template =
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GST_STATIC_PAD_TEMPLATE ("send_rtp_sink_%u",
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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 rtpbin_recv_rtp_src_template =
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GST_STATIC_PAD_TEMPLATE ("recv_rtp_src_%u_%u_%u",
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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 rtpbin_send_rtcp_src_template =
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GST_STATIC_PAD_TEMPLATE ("send_rtcp_src_%u",
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GST_PAD_SRC,
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GST_PAD_REQUEST,
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GST_STATIC_CAPS ("application/x-rtcp;application/x-srtcp")
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);
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static GstStaticPadTemplate rtpbin_send_rtp_src_template =
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GST_STATIC_PAD_TEMPLATE ("send_rtp_src_%u",
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GST_PAD_SRC,
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GST_PAD_SOMETIMES,
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GST_STATIC_CAPS ("application/x-rtp;application/x-srtp")
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);
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#define GST_RTP_BIN_LOCK(bin) g_mutex_lock (&(bin)->priv->bin_lock)
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#define GST_RTP_BIN_UNLOCK(bin) g_mutex_unlock (&(bin)->priv->bin_lock)
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/* lock to protect dynamic callbacks, like pad-added and new ssrc. */
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#define GST_RTP_BIN_DYN_LOCK(bin) g_mutex_lock (&(bin)->priv->dyn_lock)
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#define GST_RTP_BIN_DYN_UNLOCK(bin) g_mutex_unlock (&(bin)->priv->dyn_lock)
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/* lock for shutdown */
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#define GST_RTP_BIN_SHUTDOWN_LOCK(bin,label) \
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G_STMT_START { \
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if (g_atomic_int_get (&bin->priv->shutdown)) \
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goto label; \
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GST_RTP_BIN_DYN_LOCK (bin); \
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if (g_atomic_int_get (&bin->priv->shutdown)) { \
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GST_RTP_BIN_DYN_UNLOCK (bin); \
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goto label; \
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} \
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} G_STMT_END
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/* unlock for shutdown */
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#define GST_RTP_BIN_SHUTDOWN_UNLOCK(bin) \
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GST_RTP_BIN_DYN_UNLOCK (bin); \
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/* Minimum time offset to apply. This compensates for rounding errors in NTP to
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* RTP timestamp conversions */
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#define MIN_TS_OFFSET_ROUND_OFF_COMP (4 * GST_MSECOND)
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struct _GstRtpBinPrivate
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{
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GMutex bin_lock;
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/* lock protecting dynamic adding/removing */
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GMutex dyn_lock;
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/* if we are shutting down or not */
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gint shutdown;
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gboolean autoremove;
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/* NTP time in ns of last SR sync used */
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guint64 last_ntpnstime;
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/* list of extra elements */
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GList *elements;
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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_PAYLOAD_TYPE_CHANGE,
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SIGNAL_CLEAR_PT_MAP,
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SIGNAL_RESET_SYNC,
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SIGNAL_GET_SESSION,
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SIGNAL_GET_INTERNAL_SESSION,
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SIGNAL_GET_STORAGE,
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SIGNAL_GET_INTERNAL_STORAGE,
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SIGNAL_CLEAR_SSRC,
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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_NPT_STOP,
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SIGNAL_REQUEST_RTP_ENCODER,
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SIGNAL_REQUEST_RTP_DECODER,
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SIGNAL_REQUEST_RTCP_ENCODER,
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SIGNAL_REQUEST_RTCP_DECODER,
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SIGNAL_REQUEST_FEC_DECODER,
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SIGNAL_REQUEST_FEC_DECODER_FULL,
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SIGNAL_REQUEST_FEC_ENCODER,
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SIGNAL_REQUEST_JITTERBUFFER,
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SIGNAL_NEW_JITTERBUFFER,
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SIGNAL_NEW_STORAGE,
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SIGNAL_REQUEST_AUX_SENDER,
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SIGNAL_REQUEST_AUX_RECEIVER,
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SIGNAL_ON_NEW_SENDER_SSRC,
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SIGNAL_ON_SENDER_SSRC_ACTIVE,
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SIGNAL_ON_BUNDLED_SSRC,
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LAST_SIGNAL
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};
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#define DEFAULT_LATENCY_MS 200
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#define DEFAULT_DROP_ON_LATENCY FALSE
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#define DEFAULT_SDES NULL
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#define DEFAULT_DO_LOST FALSE
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#define DEFAULT_IGNORE_PT FALSE
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#define DEFAULT_NTP_SYNC FALSE
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#define DEFAULT_AUTOREMOVE FALSE
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#define DEFAULT_BUFFER_MODE RTP_JITTER_BUFFER_MODE_SLAVE
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#define DEFAULT_USE_PIPELINE_CLOCK FALSE
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#define DEFAULT_RTCP_SYNC GST_RTP_BIN_RTCP_SYNC_ALWAYS
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#define DEFAULT_RTCP_SYNC_INTERVAL 0
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#define DEFAULT_DO_SYNC_EVENT FALSE
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#define DEFAULT_DO_RETRANSMISSION FALSE
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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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#define DEFAULT_MAX_RTCP_RTP_TIME_DIFF 1000
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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_RFC7273_SYNC FALSE
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#define DEFAULT_ADD_REFERENCE_TIMESTAMP_META FALSE
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#define DEFAULT_MAX_STREAMS G_MAXUINT
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#define DEFAULT_MAX_TS_OFFSET_ADJUSTMENT G_GUINT64_CONSTANT(0)
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#define DEFAULT_MAX_TS_OFFSET G_GINT64_CONSTANT(3000000000)
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#define DEFAULT_MIN_TS_OFFSET MIN_TS_OFFSET_ROUND_OFF_COMP
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#define DEFAULT_TS_OFFSET_SMOOTHING_FACTOR 0
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#define DEFAULT_UPDATE_NTP64_HEADER_EXT TRUE
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#define DEFAULT_TIMEOUT_INACTIVE_SOURCES TRUE
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enum
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{
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PROP_0,
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PROP_LATENCY,
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PROP_DROP_ON_LATENCY,
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PROP_SDES,
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PROP_DO_LOST,
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PROP_IGNORE_PT,
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PROP_NTP_SYNC,
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PROP_RTCP_SYNC,
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PROP_RTCP_SYNC_INTERVAL,
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PROP_AUTOREMOVE,
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PROP_BUFFER_MODE,
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PROP_USE_PIPELINE_CLOCK,
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PROP_DO_SYNC_EVENT,
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PROP_DO_RETRANSMISSION,
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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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PROP_MAX_RTCP_RTP_TIME_DIFF,
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PROP_MAX_DROPOUT_TIME,
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PROP_MAX_MISORDER_TIME,
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PROP_RFC7273_SYNC,
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PROP_ADD_REFERENCE_TIMESTAMP_META,
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PROP_MAX_STREAMS,
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PROP_MAX_TS_OFFSET_ADJUSTMENT,
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PROP_MAX_TS_OFFSET,
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PROP_MIN_TS_OFFSET,
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PROP_TS_OFFSET_SMOOTHING_FACTOR,
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PROP_FEC_DECODERS,
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PROP_FEC_ENCODERS,
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PROP_UPDATE_NTP64_HEADER_EXT,
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PROP_TIMEOUT_INACTIVE_SOURCES,
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};
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#define GST_RTP_BIN_RTCP_SYNC_TYPE (gst_rtp_bin_rtcp_sync_get_type())
|
|
static GType
|
|
gst_rtp_bin_rtcp_sync_get_type (void)
|
|
{
|
|
static GType rtcp_sync_type = 0;
|
|
static const GEnumValue rtcp_sync_types[] = {
|
|
{GST_RTP_BIN_RTCP_SYNC_ALWAYS, "always", "always"},
|
|
{GST_RTP_BIN_RTCP_SYNC_INITIAL, "initial", "initial"},
|
|
{GST_RTP_BIN_RTCP_SYNC_RTP, "rtp-info", "rtp-info"},
|
|
{0, NULL, NULL},
|
|
};
|
|
|
|
if (!rtcp_sync_type) {
|
|
rtcp_sync_type = g_enum_register_static ("GstRTCPSync", rtcp_sync_types);
|
|
}
|
|
return rtcp_sync_type;
|
|
}
|
|
|
|
/* helper objects */
|
|
typedef struct _GstRtpBinSession GstRtpBinSession;
|
|
typedef struct _GstRtpBinStream GstRtpBinStream;
|
|
typedef struct _GstRtpBinClient GstRtpBinClient;
|
|
|
|
static guint gst_rtp_bin_signals[LAST_SIGNAL] = { 0 };
|
|
|
|
static GstCaps *pt_map_requested (GstElement * element, guint pt,
|
|
GstRtpBinSession * session);
|
|
static void payload_type_change (GstElement * element, guint pt,
|
|
GstRtpBinSession * session);
|
|
static void remove_recv_rtp (GstRtpBin * rtpbin, GstRtpBinSession * session);
|
|
static void remove_recv_rtcp (GstRtpBin * rtpbin, GstRtpBinSession * session);
|
|
static void remove_recv_fec (GstRtpBin * rtpbin, GstRtpBinSession * session);
|
|
static void remove_send_rtp (GstRtpBin * rtpbin, GstRtpBinSession * session);
|
|
static void remove_send_fec (GstRtpBin * rtpbin, GstRtpBinSession * session);
|
|
static void remove_rtcp (GstRtpBin * rtpbin, GstRtpBinSession * session);
|
|
static void free_client (GstRtpBinClient * client, GstRtpBin * bin);
|
|
static void free_stream (GstRtpBinStream * stream, GstRtpBin * bin);
|
|
static GstRtpBinSession *create_session (GstRtpBin * rtpbin, gint id);
|
|
static GstPad *complete_session_sink (GstRtpBin * rtpbin,
|
|
GstRtpBinSession * session);
|
|
static void
|
|
complete_session_receiver (GstRtpBin * rtpbin, GstRtpBinSession * session,
|
|
guint sessid);
|
|
static GstPad *complete_session_rtcp (GstRtpBin * rtpbin,
|
|
GstRtpBinSession * session, guint sessid);
|
|
static GstElement *session_request_element (GstRtpBinSession * session,
|
|
guint signal);
|
|
|
|
/* Manages the RTP stream for one SSRC.
|
|
*
|
|
* We pipe the stream (coming from the SSRC demuxer) into a jitterbuffer.
|
|
* If we see an SDES RTCP packet that links multiple SSRCs together based on a
|
|
* common CNAME, we create a GstRtpBinClient structure to group the SSRCs
|
|
* together (see below).
|
|
*/
|
|
struct _GstRtpBinStream
|
|
{
|
|
/* the SSRC of this stream */
|
|
guint32 ssrc;
|
|
|
|
/* parent bin */
|
|
GstRtpBin *bin;
|
|
|
|
/* the session this SSRC belongs to */
|
|
GstRtpBinSession *session;
|
|
|
|
/* the jitterbuffer of the SSRC */
|
|
GstElement *buffer;
|
|
gulong buffer_handlesync_sig;
|
|
gulong buffer_ptreq_sig;
|
|
gulong buffer_ntpstop_sig;
|
|
gint percent;
|
|
|
|
/* the PT demuxer of the SSRC */
|
|
GstElement *demux;
|
|
gulong demux_newpad_sig;
|
|
gulong demux_padremoved_sig;
|
|
gulong demux_ptreq_sig;
|
|
gulong demux_ptchange_sig;
|
|
|
|
/* if we have calculated a valid rt_delta for this stream */
|
|
gboolean have_sync;
|
|
/* mapping to local RTP and NTP time */
|
|
gint64 rt_delta;
|
|
gint64 rtp_delta;
|
|
gint64 avg_ts_offset;
|
|
gboolean is_initialized;
|
|
/* base rtptime in gst time */
|
|
gint64 clock_base;
|
|
};
|
|
|
|
#define GST_RTP_SESSION_LOCK(sess) g_mutex_lock (&(sess)->lock)
|
|
#define GST_RTP_SESSION_UNLOCK(sess) g_mutex_unlock (&(sess)->lock)
|
|
|
|
/* Manages the receiving end of the packets.
|
|
*
|
|
* There is one such structure for each RTP session (audio/video/...).
|
|
* We get the RTP/RTCP packets and stuff them into the session manager. From
|
|
* there they are pushed into an SSRC demuxer that splits the stream based on
|
|
* SSRC. Each of the SSRC streams go into their own jitterbuffer (managed with
|
|
* the GstRtpBinStream above).
|
|
*
|
|
* Before the SSRC demuxer, a storage element may be inserted for the purpose
|
|
* of Forward Error Correction.
|
|
*/
|
|
struct _GstRtpBinSession
|
|
{
|
|
/* session id */
|
|
gint id;
|
|
/* the parent bin */
|
|
GstRtpBin *bin;
|
|
/* the session element */
|
|
GstElement *session;
|
|
/* the SSRC demuxer */
|
|
GstElement *demux;
|
|
gulong demux_newpad_sig;
|
|
gulong demux_padremoved_sig;
|
|
|
|
/* Fec support */
|
|
GstElement *storage;
|
|
|
|
GMutex lock;
|
|
|
|
/* list of GstRtpBinStream */
|
|
GSList *streams;
|
|
|
|
/* list of elements */
|
|
GSList *elements;
|
|
|
|
/* mapping of payload type to caps */
|
|
GHashTable *ptmap;
|
|
|
|
/* the pads of the session */
|
|
GstPad *recv_rtp_sink;
|
|
GstPad *recv_rtp_sink_ghost;
|
|
GstPad *recv_rtp_src;
|
|
GstPad *recv_rtcp_sink;
|
|
GstPad *recv_rtcp_sink_ghost;
|
|
GstPad *sync_src;
|
|
GstPad *send_rtp_sink;
|
|
GstPad *send_rtp_sink_ghost;
|
|
GstPad *send_rtp_src_ghost;
|
|
GstPad *send_rtcp_src;
|
|
GstPad *send_rtcp_src_ghost;
|
|
|
|
GSList *recv_fec_sinks;
|
|
GSList *recv_fec_sink_ghosts;
|
|
/* fec decoder placed before the rtpjitterbuffer but after the rtpssrcdemux.
|
|
* XXX: This does not yet support multiple ssrc's in the same rtp session
|
|
*/
|
|
GstElement *early_fec_decoder;
|
|
|
|
GSList *send_fec_src_ghosts;
|
|
};
|
|
|
|
/* Manages the RTP streams that come from one client and should therefore be
|
|
* synchronized.
|
|
*/
|
|
struct _GstRtpBinClient
|
|
{
|
|
/* the common CNAME for the streams */
|
|
gchar *cname;
|
|
guint cname_len;
|
|
|
|
/* the streams */
|
|
guint nstreams;
|
|
GSList *streams;
|
|
};
|
|
|
|
/* find a session with the given id. Must be called with RTP_BIN_LOCK */
|
|
static GstRtpBinSession *
|
|
find_session_by_id (GstRtpBin * rtpbin, gint id)
|
|
{
|
|
GSList *walk;
|
|
|
|
for (walk = rtpbin->sessions; walk; walk = g_slist_next (walk)) {
|
|
GstRtpBinSession *sess = (GstRtpBinSession *) walk->data;
|
|
|
|
if (sess->id == id)
|
|
return sess;
|
|
}
|
|
return NULL;
|
|
}
|
|
|
|
static gboolean
|
|
pad_is_recv_fec (GstRtpBinSession * session, GstPad * pad)
|
|
{
|
|
return g_slist_find (session->recv_fec_sink_ghosts, pad) != NULL;
|
|
}
|
|
|
|
/* find a session with the given request pad. Must be called with RTP_BIN_LOCK */
|
|
static GstRtpBinSession *
|
|
find_session_by_pad (GstRtpBin * rtpbin, GstPad * pad)
|
|
{
|
|
GSList *walk;
|
|
|
|
for (walk = rtpbin->sessions; walk; walk = g_slist_next (walk)) {
|
|
GstRtpBinSession *sess = (GstRtpBinSession *) walk->data;
|
|
|
|
if ((sess->recv_rtp_sink_ghost == pad) ||
|
|
(sess->recv_rtcp_sink_ghost == pad) ||
|
|
(sess->send_rtp_sink_ghost == pad) ||
|
|
(sess->send_rtcp_src_ghost == pad) || pad_is_recv_fec (sess, pad))
|
|
return sess;
|
|
}
|
|
return NULL;
|
|
}
|
|
|
|
static void
|
|
on_new_ssrc (GstElement * session, guint32 ssrc, GstRtpBinSession * sess)
|
|
{
|
|
g_signal_emit (sess->bin, gst_rtp_bin_signals[SIGNAL_ON_NEW_SSRC], 0,
|
|
sess->id, ssrc);
|
|
}
|
|
|
|
static void
|
|
on_ssrc_collision (GstElement * session, guint32 ssrc, GstRtpBinSession * sess)
|
|
{
|
|
g_signal_emit (sess->bin, gst_rtp_bin_signals[SIGNAL_ON_SSRC_COLLISION], 0,
|
|
sess->id, ssrc);
|
|
}
|
|
|
|
static void
|
|
on_ssrc_validated (GstElement * session, guint32 ssrc, GstRtpBinSession * sess)
|
|
{
|
|
g_signal_emit (sess->bin, gst_rtp_bin_signals[SIGNAL_ON_SSRC_VALIDATED], 0,
|
|
sess->id, ssrc);
|
|
}
|
|
|
|
static void
|
|
on_ssrc_active (GstElement * session, guint32 ssrc, GstRtpBinSession * sess)
|
|
{
|
|
g_signal_emit (sess->bin, gst_rtp_bin_signals[SIGNAL_ON_SSRC_ACTIVE], 0,
|
|
sess->id, ssrc);
|
|
}
|
|
|
|
static void
|
|
on_ssrc_sdes (GstElement * session, guint32 ssrc, GstRtpBinSession * sess)
|
|
{
|
|
g_signal_emit (sess->bin, gst_rtp_bin_signals[SIGNAL_ON_SSRC_SDES], 0,
|
|
sess->id, ssrc);
|
|
}
|
|
|
|
static void
|
|
on_bye_ssrc (GstElement * session, guint32 ssrc, GstRtpBinSession * sess)
|
|
{
|
|
g_signal_emit (sess->bin, gst_rtp_bin_signals[SIGNAL_ON_BYE_SSRC], 0,
|
|
sess->id, ssrc);
|
|
}
|
|
|
|
static void
|
|
on_bye_timeout (GstElement * session, guint32 ssrc, GstRtpBinSession * sess)
|
|
{
|
|
g_signal_emit (sess->bin, gst_rtp_bin_signals[SIGNAL_ON_BYE_TIMEOUT], 0,
|
|
sess->id, ssrc);
|
|
|
|
if (sess->bin->priv->autoremove)
|
|
g_signal_emit_by_name (sess->demux, "clear-ssrc", ssrc, NULL);
|
|
}
|
|
|
|
static void
|
|
on_timeout (GstElement * session, guint32 ssrc, GstRtpBinSession * sess)
|
|
{
|
|
g_signal_emit (sess->bin, gst_rtp_bin_signals[SIGNAL_ON_TIMEOUT], 0,
|
|
sess->id, ssrc);
|
|
|
|
if (sess->bin->priv->autoremove)
|
|
g_signal_emit_by_name (sess->demux, "clear-ssrc", ssrc, NULL);
|
|
}
|
|
|
|
static void
|
|
on_sender_timeout (GstElement * session, guint32 ssrc, GstRtpBinSession * sess)
|
|
{
|
|
g_signal_emit (sess->bin, gst_rtp_bin_signals[SIGNAL_ON_SENDER_TIMEOUT], 0,
|
|
sess->id, ssrc);
|
|
}
|
|
|
|
static void
|
|
on_npt_stop (GstElement * jbuf, GstRtpBinStream * stream)
|
|
{
|
|
g_signal_emit (stream->bin, gst_rtp_bin_signals[SIGNAL_ON_NPT_STOP], 0,
|
|
stream->session->id, stream->ssrc);
|
|
}
|
|
|
|
static void
|
|
on_new_sender_ssrc (GstElement * session, guint32 ssrc, GstRtpBinSession * sess)
|
|
{
|
|
g_signal_emit (sess->bin, gst_rtp_bin_signals[SIGNAL_ON_NEW_SENDER_SSRC], 0,
|
|
sess->id, ssrc);
|
|
}
|
|
|
|
static void
|
|
on_sender_ssrc_active (GstElement * session, guint32 ssrc,
|
|
GstRtpBinSession * sess)
|
|
{
|
|
g_signal_emit (sess->bin, gst_rtp_bin_signals[SIGNAL_ON_SENDER_SSRC_ACTIVE],
|
|
0, sess->id, ssrc);
|
|
}
|
|
|
|
/* must be called with the SESSION lock */
|
|
static GstRtpBinStream *
|
|
find_stream_by_ssrc (GstRtpBinSession * session, guint32 ssrc)
|
|
{
|
|
GSList *walk;
|
|
|
|
for (walk = session->streams; walk; walk = g_slist_next (walk)) {
|
|
GstRtpBinStream *stream = (GstRtpBinStream *) walk->data;
|
|
|
|
if (stream->ssrc == ssrc)
|
|
return stream;
|
|
}
|
|
return NULL;
|
|
}
|
|
|
|
static void
|
|
ssrc_demux_pad_removed (GstElement * element, guint ssrc, GstPad * pad,
|
|
GstRtpBinSession * session)
|
|
{
|
|
GstRtpBinStream *stream = NULL;
|
|
GstRtpBin *rtpbin;
|
|
|
|
rtpbin = session->bin;
|
|
|
|
GST_RTP_BIN_LOCK (rtpbin);
|
|
|
|
GST_RTP_SESSION_LOCK (session);
|
|
if ((stream = find_stream_by_ssrc (session, ssrc)))
|
|
session->streams = g_slist_remove (session->streams, stream);
|
|
GST_RTP_SESSION_UNLOCK (session);
|
|
|
|
if (stream)
|
|
free_stream (stream, rtpbin);
|
|
|
|
GST_RTP_BIN_UNLOCK (rtpbin);
|
|
}
|
|
|
|
/* create a session with the given id. Must be called with RTP_BIN_LOCK */
|
|
static GstRtpBinSession *
|
|
create_session (GstRtpBin * rtpbin, gint id)
|
|
{
|
|
GstRtpBinSession *sess;
|
|
GstElement *session, *demux;
|
|
GstElement *storage = NULL;
|
|
GstState target;
|
|
|
|
if (!(session = gst_element_factory_make ("rtpsession", NULL)))
|
|
goto no_session;
|
|
|
|
if (!(demux = gst_element_factory_make ("rtpssrcdemux", NULL)))
|
|
goto no_demux;
|
|
|
|
if (!(storage = gst_element_factory_make ("rtpstorage", NULL)))
|
|
goto no_storage;
|
|
|
|
/* need to sink the storage or otherwise signal handlers from bindings will
|
|
* take ownership of it and we don't own it anymore */
|
|
gst_object_ref_sink (storage);
|
|
g_signal_emit (rtpbin, gst_rtp_bin_signals[SIGNAL_NEW_STORAGE], 0, storage,
|
|
id);
|
|
|
|
sess = g_new0 (GstRtpBinSession, 1);
|
|
g_mutex_init (&sess->lock);
|
|
sess->id = id;
|
|
sess->bin = rtpbin;
|
|
sess->session = session;
|
|
sess->demux = demux;
|
|
sess->storage = storage;
|
|
|
|
sess->ptmap = g_hash_table_new_full (NULL, NULL, NULL,
|
|
(GDestroyNotify) gst_caps_unref);
|
|
rtpbin->sessions = g_slist_prepend (rtpbin->sessions, sess);
|
|
|
|
/* configure SDES items */
|
|
GST_OBJECT_LOCK (rtpbin);
|
|
g_object_set (demux, "max-streams", rtpbin->max_streams, NULL);
|
|
g_object_set (session, "sdes", rtpbin->sdes, "rtp-profile",
|
|
rtpbin->rtp_profile, "rtcp-sync-send-time", rtpbin->rtcp_sync_send_time,
|
|
NULL);
|
|
if (rtpbin->use_pipeline_clock)
|
|
g_object_set (session, "use-pipeline-clock", rtpbin->use_pipeline_clock,
|
|
NULL);
|
|
else
|
|
g_object_set (session, "ntp-time-source", rtpbin->ntp_time_source, NULL);
|
|
|
|
g_object_set (session, "max-dropout-time", rtpbin->max_dropout_time,
|
|
"max-misorder-time", rtpbin->max_misorder_time, NULL);
|
|
|
|
g_object_set (session, "update-ntp64-header-ext",
|
|
rtpbin->update_ntp64_header_ext, NULL);
|
|
|
|
g_object_set (session, "timeout-inactive-sources",
|
|
rtpbin->timeout_inactive_sources, NULL);
|
|
|
|
GST_OBJECT_UNLOCK (rtpbin);
|
|
|
|
/* provide clock_rate to the session manager when needed */
|
|
g_signal_connect (session, "request-pt-map",
|
|
(GCallback) pt_map_requested, sess);
|
|
|
|
g_signal_connect (sess->session, "on-new-ssrc",
|
|
(GCallback) on_new_ssrc, sess);
|
|
g_signal_connect (sess->session, "on-ssrc-collision",
|
|
(GCallback) on_ssrc_collision, sess);
|
|
g_signal_connect (sess->session, "on-ssrc-validated",
|
|
(GCallback) on_ssrc_validated, sess);
|
|
g_signal_connect (sess->session, "on-ssrc-active",
|
|
(GCallback) on_ssrc_active, sess);
|
|
g_signal_connect (sess->session, "on-ssrc-sdes",
|
|
(GCallback) on_ssrc_sdes, sess);
|
|
g_signal_connect (sess->session, "on-bye-ssrc",
|
|
(GCallback) on_bye_ssrc, sess);
|
|
g_signal_connect (sess->session, "on-bye-timeout",
|
|
(GCallback) on_bye_timeout, sess);
|
|
g_signal_connect (sess->session, "on-timeout", (GCallback) on_timeout, sess);
|
|
g_signal_connect (sess->session, "on-sender-timeout",
|
|
(GCallback) on_sender_timeout, sess);
|
|
g_signal_connect (sess->session, "on-new-sender-ssrc",
|
|
(GCallback) on_new_sender_ssrc, sess);
|
|
g_signal_connect (sess->session, "on-sender-ssrc-active",
|
|
(GCallback) on_sender_ssrc_active, sess);
|
|
|
|
gst_bin_add (GST_BIN_CAST (rtpbin), session);
|
|
gst_bin_add (GST_BIN_CAST (rtpbin), demux);
|
|
gst_bin_add (GST_BIN_CAST (rtpbin), storage);
|
|
|
|
/* unref the storage again, the bin has a reference now and
|
|
* we don't need it anymore */
|
|
gst_object_unref (storage);
|
|
|
|
GST_OBJECT_LOCK (rtpbin);
|
|
target = GST_STATE_TARGET (rtpbin);
|
|
GST_OBJECT_UNLOCK (rtpbin);
|
|
|
|
/* change state only to what's needed */
|
|
gst_element_set_state (demux, target);
|
|
gst_element_set_state (session, target);
|
|
gst_element_set_state (storage, target);
|
|
|
|
return sess;
|
|
|
|
/* ERRORS */
|
|
no_session:
|
|
{
|
|
g_warning ("rtpbin: could not create rtpsession element");
|
|
return NULL;
|
|
}
|
|
no_demux:
|
|
{
|
|
gst_object_unref (session);
|
|
g_warning ("rtpbin: could not create rtpssrcdemux element");
|
|
return NULL;
|
|
}
|
|
no_storage:
|
|
{
|
|
gst_object_unref (session);
|
|
gst_object_unref (demux);
|
|
g_warning ("rtpbin: could not create rtpstorage element");
|
|
return NULL;
|
|
}
|
|
}
|
|
|
|
static gboolean
|
|
bin_manage_element (GstRtpBin * bin, GstElement * element)
|
|
{
|
|
GstRtpBinPrivate *priv = bin->priv;
|
|
|
|
if (g_list_find (priv->elements, element)) {
|
|
GST_DEBUG_OBJECT (bin, "requested element %p already in bin", element);
|
|
} else {
|
|
GST_DEBUG_OBJECT (bin, "adding requested element %p", element);
|
|
|
|
if (g_object_is_floating (element))
|
|
element = gst_object_ref_sink (element);
|
|
|
|
if (!gst_bin_add (GST_BIN_CAST (bin), element))
|
|
goto add_failed;
|
|
if (!gst_element_sync_state_with_parent (element))
|
|
GST_WARNING_OBJECT (bin, "unable to sync element state with rtpbin");
|
|
}
|
|
/* we add the element multiple times, each we need an equal number of
|
|
* removes to really remove the element from the bin */
|
|
priv->elements = g_list_prepend (priv->elements, element);
|
|
|
|
return TRUE;
|
|
|
|
/* ERRORS */
|
|
add_failed:
|
|
{
|
|
GST_WARNING_OBJECT (bin, "unable to add element");
|
|
gst_object_unref (element);
|
|
return FALSE;
|
|
}
|
|
}
|
|
|
|
static void
|
|
remove_bin_element (GstElement * element, GstRtpBin * bin)
|
|
{
|
|
GstRtpBinPrivate *priv = bin->priv;
|
|
GList *find;
|
|
|
|
find = g_list_find (priv->elements, element);
|
|
if (find) {
|
|
priv->elements = g_list_delete_link (priv->elements, find);
|
|
|
|
if (!g_list_find (priv->elements, element)) {
|
|
gst_element_set_locked_state (element, TRUE);
|
|
gst_bin_remove (GST_BIN_CAST (bin), element);
|
|
gst_element_set_state (element, GST_STATE_NULL);
|
|
}
|
|
|
|
gst_object_unref (element);
|
|
}
|
|
}
|
|
|
|
/* called with RTP_BIN_LOCK */
|
|
static void
|
|
free_session (GstRtpBinSession * sess, GstRtpBin * bin)
|
|
{
|
|
GST_DEBUG_OBJECT (bin, "freeing session %p", sess);
|
|
|
|
gst_element_set_locked_state (sess->demux, TRUE);
|
|
gst_element_set_locked_state (sess->session, TRUE);
|
|
gst_element_set_locked_state (sess->storage, TRUE);
|
|
|
|
gst_element_set_state (sess->demux, GST_STATE_NULL);
|
|
gst_element_set_state (sess->session, GST_STATE_NULL);
|
|
gst_element_set_state (sess->storage, GST_STATE_NULL);
|
|
|
|
remove_recv_rtp (bin, sess);
|
|
remove_recv_rtcp (bin, sess);
|
|
remove_recv_fec (bin, sess);
|
|
remove_send_rtp (bin, sess);
|
|
remove_send_fec (bin, sess);
|
|
remove_rtcp (bin, sess);
|
|
|
|
gst_bin_remove (GST_BIN_CAST (bin), sess->session);
|
|
gst_bin_remove (GST_BIN_CAST (bin), sess->demux);
|
|
gst_bin_remove (GST_BIN_CAST (bin), sess->storage);
|
|
|
|
g_slist_foreach (sess->elements, (GFunc) remove_bin_element, bin);
|
|
g_slist_free (sess->elements);
|
|
sess->elements = NULL;
|
|
|
|
g_slist_foreach (sess->streams, (GFunc) free_stream, bin);
|
|
g_slist_free (sess->streams);
|
|
|
|
g_mutex_clear (&sess->lock);
|
|
g_hash_table_destroy (sess->ptmap);
|
|
|
|
g_free (sess);
|
|
}
|
|
|
|
/* get the payload type caps for the specific payload @pt in @session */
|
|
static GstCaps *
|
|
get_pt_map (GstRtpBinSession * session, guint pt)
|
|
{
|
|
GstCaps *caps = NULL;
|
|
GstRtpBin *bin;
|
|
GValue ret = { 0 };
|
|
GValue args[3] = { {0}, {0}, {0} };
|
|
|
|
GST_DEBUG ("searching pt %u in cache", pt);
|
|
|
|
GST_RTP_SESSION_LOCK (session);
|
|
|
|
/* first look in the cache */
|
|
caps = g_hash_table_lookup (session->ptmap, GINT_TO_POINTER (pt));
|
|
if (caps) {
|
|
gst_caps_ref (caps);
|
|
goto done;
|
|
}
|
|
|
|
bin = session->bin;
|
|
|
|
GST_DEBUG ("emitting signal for pt %u in session %u", pt, session->id);
|
|
|
|
/* not in cache, send signal to request caps */
|
|
g_value_init (&args[0], GST_TYPE_ELEMENT);
|
|
g_value_set_object (&args[0], bin);
|
|
g_value_init (&args[1], G_TYPE_UINT);
|
|
g_value_set_uint (&args[1], session->id);
|
|
g_value_init (&args[2], G_TYPE_UINT);
|
|
g_value_set_uint (&args[2], pt);
|
|
|
|
g_value_init (&ret, GST_TYPE_CAPS);
|
|
g_value_set_boxed (&ret, NULL);
|
|
|
|
GST_RTP_SESSION_UNLOCK (session);
|
|
|
|
g_signal_emitv (args, gst_rtp_bin_signals[SIGNAL_REQUEST_PT_MAP], 0, &ret);
|
|
|
|
GST_RTP_SESSION_LOCK (session);
|
|
|
|
g_value_unset (&args[0]);
|
|
g_value_unset (&args[1]);
|
|
g_value_unset (&args[2]);
|
|
|
|
/* look in the cache again because we let the lock go */
|
|
caps = g_hash_table_lookup (session->ptmap, GINT_TO_POINTER (pt));
|
|
if (caps) {
|
|
gst_caps_ref (caps);
|
|
g_value_unset (&ret);
|
|
goto done;
|
|
}
|
|
|
|
caps = (GstCaps *) g_value_dup_boxed (&ret);
|
|
g_value_unset (&ret);
|
|
if (!caps)
|
|
goto no_caps;
|
|
|
|
GST_DEBUG ("caching pt %u as %" GST_PTR_FORMAT, pt, caps);
|
|
|
|
/* store in cache, take additional ref */
|
|
g_hash_table_insert (session->ptmap, GINT_TO_POINTER (pt),
|
|
gst_caps_ref (caps));
|
|
|
|
done:
|
|
GST_RTP_SESSION_UNLOCK (session);
|
|
|
|
return caps;
|
|
|
|
/* ERRORS */
|
|
no_caps:
|
|
{
|
|
GST_RTP_SESSION_UNLOCK (session);
|
|
GST_DEBUG ("no pt map could be obtained");
|
|
return NULL;
|
|
}
|
|
}
|
|
|
|
static gboolean
|
|
return_true (gpointer key, gpointer value, gpointer user_data)
|
|
{
|
|
return TRUE;
|
|
}
|
|
|
|
static void
|
|
gst_rtp_bin_reset_sync (GstRtpBin * rtpbin)
|
|
{
|
|
GSList *clients, *streams;
|
|
|
|
GST_DEBUG_OBJECT (rtpbin, "Reset sync on all clients");
|
|
|
|
GST_RTP_BIN_LOCK (rtpbin);
|
|
for (clients = rtpbin->clients; clients; clients = g_slist_next (clients)) {
|
|
GstRtpBinClient *client = (GstRtpBinClient *) clients->data;
|
|
|
|
/* reset sync on all streams for this client */
|
|
for (streams = client->streams; streams; streams = g_slist_next (streams)) {
|
|
GstRtpBinStream *stream = (GstRtpBinStream *) streams->data;
|
|
|
|
/* make use require a new SR packet for this stream before we attempt new
|
|
* lip-sync */
|
|
stream->have_sync = FALSE;
|
|
stream->rt_delta = 0;
|
|
stream->avg_ts_offset = 0;
|
|
stream->is_initialized = FALSE;
|
|
stream->rtp_delta = 0;
|
|
stream->clock_base = -100 * GST_SECOND;
|
|
}
|
|
}
|
|
GST_RTP_BIN_UNLOCK (rtpbin);
|
|
}
|
|
|
|
static void
|
|
gst_rtp_bin_clear_pt_map (GstRtpBin * bin)
|
|
{
|
|
GSList *sessions, *streams;
|
|
|
|
GST_RTP_BIN_LOCK (bin);
|
|
GST_DEBUG_OBJECT (bin, "clearing pt map");
|
|
for (sessions = bin->sessions; sessions; sessions = g_slist_next (sessions)) {
|
|
GstRtpBinSession *session = (GstRtpBinSession *) sessions->data;
|
|
|
|
GST_DEBUG_OBJECT (bin, "clearing session %p", session);
|
|
g_signal_emit_by_name (session->session, "clear-pt-map", NULL);
|
|
|
|
GST_RTP_SESSION_LOCK (session);
|
|
g_hash_table_foreach_remove (session->ptmap, return_true, NULL);
|
|
|
|
for (streams = session->streams; streams; streams = g_slist_next (streams)) {
|
|
GstRtpBinStream *stream = (GstRtpBinStream *) streams->data;
|
|
|
|
GST_DEBUG_OBJECT (bin, "clearing stream %p", stream);
|
|
if (g_signal_lookup ("clear-pt-map", G_OBJECT_TYPE (stream->buffer)) != 0)
|
|
g_signal_emit_by_name (stream->buffer, "clear-pt-map", NULL);
|
|
if (stream->demux)
|
|
g_signal_emit_by_name (stream->demux, "clear-pt-map", NULL);
|
|
}
|
|
GST_RTP_SESSION_UNLOCK (session);
|
|
}
|
|
GST_RTP_BIN_UNLOCK (bin);
|
|
|
|
/* reset sync too */
|
|
gst_rtp_bin_reset_sync (bin);
|
|
}
|
|
|
|
static GstElement *
|
|
gst_rtp_bin_get_session (GstRtpBin * bin, guint session_id)
|
|
{
|
|
GstRtpBinSession *session;
|
|
GstElement *ret = NULL;
|
|
|
|
GST_RTP_BIN_LOCK (bin);
|
|
GST_DEBUG_OBJECT (bin, "retrieving GstRtpSession, index: %u", session_id);
|
|
session = find_session_by_id (bin, (gint) session_id);
|
|
if (session) {
|
|
ret = gst_object_ref (session->session);
|
|
}
|
|
GST_RTP_BIN_UNLOCK (bin);
|
|
|
|
return ret;
|
|
}
|
|
|
|
static RTPSession *
|
|
gst_rtp_bin_get_internal_session (GstRtpBin * bin, guint session_id)
|
|
{
|
|
RTPSession *internal_session = NULL;
|
|
GstRtpBinSession *session;
|
|
|
|
GST_RTP_BIN_LOCK (bin);
|
|
GST_DEBUG_OBJECT (bin, "retrieving internal RTPSession object, index: %u",
|
|
session_id);
|
|
session = find_session_by_id (bin, (gint) session_id);
|
|
if (session) {
|
|
g_object_get (session->session, "internal-session", &internal_session,
|
|
NULL);
|
|
}
|
|
GST_RTP_BIN_UNLOCK (bin);
|
|
|
|
return internal_session;
|
|
}
|
|
|
|
static GstElement *
|
|
gst_rtp_bin_get_storage (GstRtpBin * bin, guint session_id)
|
|
{
|
|
GstRtpBinSession *session;
|
|
GstElement *res = NULL;
|
|
|
|
GST_RTP_BIN_LOCK (bin);
|
|
GST_DEBUG_OBJECT (bin, "retrieving internal storage object, index: %u",
|
|
session_id);
|
|
session = find_session_by_id (bin, (gint) session_id);
|
|
if (session && session->storage) {
|
|
res = gst_object_ref (session->storage);
|
|
}
|
|
GST_RTP_BIN_UNLOCK (bin);
|
|
|
|
return res;
|
|
}
|
|
|
|
static GObject *
|
|
gst_rtp_bin_get_internal_storage (GstRtpBin * bin, guint session_id)
|
|
{
|
|
GObject *internal_storage = NULL;
|
|
GstRtpBinSession *session;
|
|
|
|
GST_RTP_BIN_LOCK (bin);
|
|
GST_DEBUG_OBJECT (bin, "retrieving internal storage object, index: %u",
|
|
session_id);
|
|
session = find_session_by_id (bin, (gint) session_id);
|
|
if (session && session->storage) {
|
|
g_object_get (session->storage, "internal-storage", &internal_storage,
|
|
NULL);
|
|
}
|
|
GST_RTP_BIN_UNLOCK (bin);
|
|
|
|
return internal_storage;
|
|
}
|
|
|
|
static void
|
|
gst_rtp_bin_clear_ssrc (GstRtpBin * bin, guint session_id, guint32 ssrc)
|
|
{
|
|
GstRtpBinSession *session;
|
|
GstElement *demux = NULL;
|
|
|
|
GST_RTP_BIN_LOCK (bin);
|
|
GST_DEBUG_OBJECT (bin, "clearing ssrc %u for session %u", ssrc, session_id);
|
|
session = find_session_by_id (bin, (gint) session_id);
|
|
if (session)
|
|
demux = gst_object_ref (session->demux);
|
|
GST_RTP_BIN_UNLOCK (bin);
|
|
|
|
if (demux) {
|
|
g_signal_emit_by_name (demux, "clear-ssrc", ssrc, NULL);
|
|
gst_object_unref (demux);
|
|
}
|
|
}
|
|
|
|
static GstElement *
|
|
gst_rtp_bin_request_encoder (GstRtpBin * bin, guint session_id)
|
|
{
|
|
GST_DEBUG_OBJECT (bin, "return NULL encoder");
|
|
return NULL;
|
|
}
|
|
|
|
static GstElement *
|
|
gst_rtp_bin_request_decoder (GstRtpBin * bin, guint session_id)
|
|
{
|
|
GST_DEBUG_OBJECT (bin, "return NULL decoder");
|
|
return NULL;
|
|
}
|
|
|
|
static GstElement *
|
|
gst_rtp_bin_request_jitterbuffer (GstRtpBin * bin, guint session_id)
|
|
{
|
|
return gst_element_factory_make ("rtpjitterbuffer", NULL);
|
|
}
|
|
|
|
static void
|
|
gst_rtp_bin_propagate_property_to_jitterbuffer (GstRtpBin * bin,
|
|
const gchar * name, const GValue * value)
|
|
{
|
|
GSList *sessions, *streams;
|
|
|
|
GST_RTP_BIN_LOCK (bin);
|
|
for (sessions = bin->sessions; sessions; sessions = g_slist_next (sessions)) {
|
|
GstRtpBinSession *session = (GstRtpBinSession *) sessions->data;
|
|
|
|
GST_RTP_SESSION_LOCK (session);
|
|
for (streams = session->streams; streams; streams = g_slist_next (streams)) {
|
|
GstRtpBinStream *stream = (GstRtpBinStream *) streams->data;
|
|
GObjectClass *jb_class;
|
|
|
|
jb_class = G_OBJECT_GET_CLASS (G_OBJECT (stream->buffer));
|
|
if (g_object_class_find_property (jb_class, name))
|
|
g_object_set_property (G_OBJECT (stream->buffer), name, value);
|
|
else
|
|
GST_WARNING_OBJECT (bin,
|
|
"Stream jitterbuffer does not expose property %s", name);
|
|
}
|
|
GST_RTP_SESSION_UNLOCK (session);
|
|
}
|
|
GST_RTP_BIN_UNLOCK (bin);
|
|
}
|
|
|
|
static void
|
|
gst_rtp_bin_propagate_property_to_session (GstRtpBin * bin,
|
|
const gchar * name, const GValue * value)
|
|
{
|
|
GSList *sessions;
|
|
|
|
GST_RTP_BIN_LOCK (bin);
|
|
for (sessions = bin->sessions; sessions; sessions = g_slist_next (sessions)) {
|
|
GstRtpBinSession *sess = (GstRtpBinSession *) sessions->data;
|
|
|
|
g_object_set_property (G_OBJECT (sess->session), name, value);
|
|
}
|
|
GST_RTP_BIN_UNLOCK (bin);
|
|
}
|
|
|
|
/* get a client with the given SDES name. Must be called with RTP_BIN_LOCK */
|
|
static GstRtpBinClient *
|
|
get_client (GstRtpBin * bin, guint8 len, const guint8 * data,
|
|
gboolean * created)
|
|
{
|
|
GstRtpBinClient *result = NULL;
|
|
GSList *walk;
|
|
|
|
for (walk = bin->clients; walk; walk = g_slist_next (walk)) {
|
|
GstRtpBinClient *client = (GstRtpBinClient *) walk->data;
|
|
|
|
if (len != client->cname_len)
|
|
continue;
|
|
|
|
if (!strncmp ((gchar *) data, client->cname, client->cname_len)) {
|
|
GST_DEBUG_OBJECT (bin, "found existing client %p with CNAME %s", client,
|
|
client->cname);
|
|
result = client;
|
|
break;
|
|
}
|
|
}
|
|
|
|
/* nothing found, create one */
|
|
if (result == NULL) {
|
|
result = g_new0 (GstRtpBinClient, 1);
|
|
result->cname = g_strndup ((gchar *) data, len);
|
|
result->cname_len = len;
|
|
bin->clients = g_slist_prepend (bin->clients, result);
|
|
GST_DEBUG_OBJECT (bin, "created new client %p with CNAME %s", result,
|
|
result->cname);
|
|
}
|
|
return result;
|
|
}
|
|
|
|
static void
|
|
free_client (GstRtpBinClient * client, GstRtpBin * bin)
|
|
{
|
|
GST_DEBUG_OBJECT (bin, "freeing client %p", client);
|
|
g_slist_free (client->streams);
|
|
g_free (client->cname);
|
|
g_free (client);
|
|
}
|
|
|
|
static void
|
|
get_current_times (GstRtpBin * bin, GstClockTime * running_time,
|
|
guint64 * ntpnstime)
|
|
{
|
|
guint64 ntpns = -1;
|
|
GstClock *clock;
|
|
GstClockTime base_time, rt, clock_time;
|
|
|
|
GST_OBJECT_LOCK (bin);
|
|
if ((clock = GST_ELEMENT_CLOCK (bin))) {
|
|
base_time = GST_ELEMENT_CAST (bin)->base_time;
|
|
gst_object_ref (clock);
|
|
GST_OBJECT_UNLOCK (bin);
|
|
|
|
/* get current clock time and convert to running time */
|
|
clock_time = gst_clock_get_time (clock);
|
|
rt = clock_time - base_time;
|
|
|
|
if (bin->use_pipeline_clock) {
|
|
ntpns = rt;
|
|
/* add constant to convert from 1970 based time to 1900 based time */
|
|
ntpns += (GST_RTP_NTP_UNIX_OFFSET * GST_SECOND);
|
|
} else {
|
|
switch (bin->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 (bin->ntp_time_source == GST_RTP_NTP_TIME_SOURCE_NTP)
|
|
ntpns += (GST_RTP_NTP_UNIX_OFFSET * 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; /* Fix uninited compiler warning */
|
|
g_assert_not_reached ();
|
|
break;
|
|
}
|
|
}
|
|
|
|
gst_object_unref (clock);
|
|
} else {
|
|
GST_OBJECT_UNLOCK (bin);
|
|
rt = -1;
|
|
ntpns = -1;
|
|
}
|
|
if (running_time)
|
|
*running_time = rt;
|
|
if (ntpnstime)
|
|
*ntpnstime = ntpns;
|
|
}
|
|
|
|
static void
|
|
stream_set_ts_offset (GstRtpBin * bin, GstRtpBinStream * stream,
|
|
gint64 ts_offset, gint64 max_ts_offset, guint64 min_ts_offset,
|
|
gboolean allow_positive_ts_offset)
|
|
{
|
|
gint64 prev_ts_offset;
|
|
GObjectClass *jb_class;
|
|
|
|
jb_class = G_OBJECT_GET_CLASS (G_OBJECT (stream->buffer));
|
|
|
|
if (!g_object_class_find_property (jb_class, "ts-offset")) {
|
|
GST_LOG_OBJECT (bin,
|
|
"stream's jitterbuffer does not expose ts-offset property");
|
|
return;
|
|
}
|
|
|
|
if (bin->ts_offset_smoothing_factor > 0) {
|
|
if (!stream->is_initialized) {
|
|
stream->avg_ts_offset = ts_offset;
|
|
stream->is_initialized = TRUE;
|
|
} else {
|
|
/* RMA algorithm using smoothing factor is following, but split into
|
|
* parts to check for overflows:
|
|
* stream->avg_ts_offset =
|
|
* ((bin->ts_offset_smoothing_factor - 1) * stream->avg_ts_offset
|
|
* + ts_offset) / bin->ts_offset_smoothing_factor
|
|
*/
|
|
guint64 max_possible_smoothing_factor = G_MAXUINT64;
|
|
gint64 cur_avg_product =
|
|
(bin->ts_offset_smoothing_factor - 1) * stream->avg_ts_offset;
|
|
if (stream->avg_ts_offset != 0)
|
|
max_possible_smoothing_factor =
|
|
G_MAXINT64 / ABS (stream->avg_ts_offset);
|
|
|
|
if ((max_possible_smoothing_factor < bin->ts_offset_smoothing_factor) ||
|
|
(cur_avg_product > 0 && G_MAXINT64 - cur_avg_product < ts_offset) ||
|
|
(cur_avg_product < 0 && G_MININT64 - cur_avg_product > ts_offset)) {
|
|
GST_WARNING_OBJECT (bin,
|
|
"ts-offset-smoothing-factor calculation overflow, fallback to using ts-offset directly");
|
|
stream->avg_ts_offset = ts_offset;
|
|
} else {
|
|
stream->avg_ts_offset =
|
|
(cur_avg_product + ts_offset) / bin->ts_offset_smoothing_factor;
|
|
}
|
|
}
|
|
} else {
|
|
stream->avg_ts_offset = ts_offset;
|
|
}
|
|
|
|
g_object_get (stream->buffer, "ts-offset", &prev_ts_offset, NULL);
|
|
|
|
/* delta changed, see how much */
|
|
if (prev_ts_offset != stream->avg_ts_offset) {
|
|
gint64 diff;
|
|
|
|
diff = prev_ts_offset - stream->avg_ts_offset;
|
|
|
|
GST_DEBUG_OBJECT (bin,
|
|
"ts-offset %" G_GINT64_FORMAT ", prev %" G_GINT64_FORMAT
|
|
", diff: %" G_GINT64_FORMAT, stream->avg_ts_offset, prev_ts_offset,
|
|
diff);
|
|
|
|
/* ignore minor offsets */
|
|
if (ABS (diff) < min_ts_offset) {
|
|
GST_DEBUG_OBJECT (bin, "offset too small, ignoring");
|
|
return;
|
|
}
|
|
|
|
/* sanity check offset */
|
|
if (max_ts_offset > 0) {
|
|
if (stream->avg_ts_offset > 0 && !allow_positive_ts_offset) {
|
|
GST_DEBUG_OBJECT (bin,
|
|
"offset is positive (clocks are out of sync), ignoring");
|
|
return;
|
|
}
|
|
if (ABS (stream->avg_ts_offset) > max_ts_offset) {
|
|
GST_DEBUG_OBJECT (bin, "offset too large, ignoring");
|
|
return;
|
|
}
|
|
}
|
|
|
|
g_object_set (stream->buffer, "ts-offset", stream->avg_ts_offset, NULL);
|
|
}
|
|
GST_DEBUG_OBJECT (bin, "stream SSRC %08x, delta %" G_GINT64_FORMAT,
|
|
stream->ssrc, stream->avg_ts_offset);
|
|
}
|
|
|
|
static void
|
|
gst_rtp_bin_send_sync_event (GstRtpBinStream * stream)
|
|
{
|
|
if (stream->bin->send_sync_event) {
|
|
GstEvent *event;
|
|
GstPad *srcpad;
|
|
|
|
GST_DEBUG_OBJECT (stream->bin,
|
|
"sending GstRTCPSRReceived event downstream");
|
|
|
|
event = gst_event_new_custom (GST_EVENT_CUSTOM_DOWNSTREAM,
|
|
gst_structure_new_empty ("GstRTCPSRReceived"));
|
|
|
|
srcpad = gst_element_get_static_pad (stream->buffer, "src");
|
|
gst_pad_push_event (srcpad, event);
|
|
gst_object_unref (srcpad);
|
|
}
|
|
}
|
|
|
|
/* associate a stream to the given CNAME. This will make sure all streams for
|
|
* that CNAME are synchronized together.
|
|
* Must be called with GST_RTP_BIN_LOCK */
|
|
static void
|
|
gst_rtp_bin_associate (GstRtpBin * bin, GstRtpBinStream * stream, guint8 len,
|
|
const guint8 * data, guint64 ntpnstime, guint64 last_extrtptime,
|
|
guint64 base_rtptime, guint64 base_time, guint clock_rate,
|
|
gint64 rtp_clock_base)
|
|
{
|
|
GstRtpBinClient *client;
|
|
gboolean created;
|
|
GSList *walk;
|
|
GstClockTime running_time, running_time_rtp;
|
|
|
|
/* first find or create the CNAME */
|
|
client = get_client (bin, len, data, &created);
|
|
|
|
/* find stream in the client */
|
|
for (walk = client->streams; walk; walk = g_slist_next (walk)) {
|
|
GstRtpBinStream *ostream = (GstRtpBinStream *) walk->data;
|
|
|
|
if (ostream == stream)
|
|
break;
|
|
}
|
|
/* not found, add it to the list */
|
|
if (walk == NULL) {
|
|
GST_DEBUG_OBJECT (bin,
|
|
"new association of SSRC %08x with client %p with CNAME %s",
|
|
stream->ssrc, client, client->cname);
|
|
client->streams = g_slist_prepend (client->streams, stream);
|
|
client->nstreams++;
|
|
} else {
|
|
GST_DEBUG_OBJECT (bin,
|
|
"found association of SSRC %08x with client %p with CNAME %s",
|
|
stream->ssrc, client, client->cname);
|
|
}
|
|
|
|
if (!GST_CLOCK_TIME_IS_VALID (last_extrtptime)) {
|
|
GST_DEBUG_OBJECT (bin, "invalidated sync data");
|
|
if (bin->rtcp_sync == GST_RTP_BIN_RTCP_SYNC_RTP) {
|
|
/* we don't need that data, so carry on,
|
|
* but make some values look saner */
|
|
last_extrtptime = base_rtptime;
|
|
} else {
|
|
/* nothing we can do with this data in this case */
|
|
GST_DEBUG_OBJECT (bin, "bailing out");
|
|
return;
|
|
}
|
|
}
|
|
|
|
/* Take the extended rtptime we found in the SR packet and map it to the
|
|
* local rtptime. The local rtp time is used to construct timestamps on the
|
|
* buffers so we will calculate what running_time corresponds to the RTP
|
|
* timestamp in the SR packet. */
|
|
running_time_rtp = last_extrtptime - base_rtptime;
|
|
|
|
GST_DEBUG_OBJECT (bin,
|
|
"base %" G_GUINT64_FORMAT ", extrtptime %" G_GUINT64_FORMAT
|
|
", local RTP %" G_GUINT64_FORMAT ", clock-rate %d, "
|
|
"clock-base %" G_GINT64_FORMAT, base_rtptime,
|
|
last_extrtptime, running_time_rtp, clock_rate, rtp_clock_base);
|
|
|
|
/* calculate local RTP time in gstreamer timestamp, we essentially perform the
|
|
* same conversion that a jitterbuffer would use to convert an rtp timestamp
|
|
* into a corresponding gstreamer timestamp. Note that the base_time also
|
|
* contains the drift between sender and receiver. */
|
|
running_time =
|
|
gst_util_uint64_scale_int (running_time_rtp, GST_SECOND, clock_rate);
|
|
running_time += base_time;
|
|
|
|
stream->have_sync = TRUE;
|
|
|
|
GST_DEBUG_OBJECT (bin,
|
|
"SR RTP running time %" G_GUINT64_FORMAT ", SR NTP %" G_GUINT64_FORMAT,
|
|
running_time, ntpnstime);
|
|
|
|
/* recalc inter stream playout offset, but only if there is more than one
|
|
* stream or we're doing NTP sync. */
|
|
if (bin->ntp_sync) {
|
|
gint64 ntpdiff, rtdiff;
|
|
guint64 local_ntpnstime;
|
|
GstClockTime local_running_time;
|
|
|
|
/* For NTP sync we need to first get a snapshot of running_time and NTP
|
|
* time. We know at what running_time we play a certain RTP time, we also
|
|
* calculated when we would play the RTP time in the SR packet. Now we need
|
|
* to know how the running_time and the NTP time relate to each other. */
|
|
get_current_times (bin, &local_running_time, &local_ntpnstime);
|
|
|
|
/* see how far away the NTP time is. This is the difference between the
|
|
* current NTP time and the NTP time in the last SR packet. */
|
|
ntpdiff = local_ntpnstime - ntpnstime;
|
|
/* see how far away the running_time is. This is the difference between the
|
|
* current running_time and the running_time of the RTP timestamp in the
|
|
* last SR packet. */
|
|
rtdiff = local_running_time - running_time;
|
|
|
|
GST_DEBUG_OBJECT (bin,
|
|
"local NTP time %" G_GUINT64_FORMAT ", SR NTP time %" G_GUINT64_FORMAT,
|
|
local_ntpnstime, ntpnstime);
|
|
GST_DEBUG_OBJECT (bin,
|
|
"local running time %" G_GUINT64_FORMAT ", SR RTP running time %"
|
|
G_GUINT64_FORMAT, local_running_time, running_time);
|
|
GST_DEBUG_OBJECT (bin,
|
|
"NTP diff %" G_GINT64_FORMAT ", RT diff %" G_GINT64_FORMAT, ntpdiff,
|
|
rtdiff);
|
|
|
|
/* combine to get the final diff to apply to the running_time */
|
|
stream->rt_delta = rtdiff - ntpdiff;
|
|
|
|
stream_set_ts_offset (bin, stream, stream->rt_delta, bin->max_ts_offset,
|
|
bin->min_ts_offset, FALSE);
|
|
} else {
|
|
gint64 min, rtp_min, clock_base;
|
|
gboolean all_sync, use_rtp;
|
|
gboolean rtcp_sync = g_atomic_int_get (&bin->rtcp_sync);
|
|
|
|
/* calculate delta between server and receiver. ntpnstime is created by
|
|
* converting the ntptime in the last SR packet to a gstreamer timestamp. This
|
|
* delta expresses the difference to our timeline and the server timeline. The
|
|
* difference in itself doesn't mean much but we can combine the delta of
|
|
* multiple streams to create a stream specific offset. */
|
|
stream->rt_delta = ntpnstime - running_time;
|
|
|
|
/* calculate the min of all deltas, ignoring streams that did not yet have a
|
|
* valid rt_delta because we did not yet receive an SR packet for those
|
|
* streams.
|
|
* We calculate the minimum because we would like to only apply positive
|
|
* offsets to streams, delaying their playback instead of trying to speed up
|
|
* other streams (which might be impossible when we have to create negative
|
|
* latencies).
|
|
* The stream that has the smallest diff is selected as the reference stream,
|
|
* all other streams will have a positive offset to this difference. */
|
|
|
|
/* some alternative setting allow ignoring RTCP as much as possible,
|
|
* for servers generating bogus ntp timeline */
|
|
min = rtp_min = G_MAXINT64;
|
|
use_rtp = FALSE;
|
|
if (rtcp_sync == GST_RTP_BIN_RTCP_SYNC_RTP) {
|
|
guint64 ext_base = -1;
|
|
gint64 rtp_delta = 0;
|
|
|
|
use_rtp = TRUE;
|
|
/* convert to extended RTP time */
|
|
rtp_clock_base = gst_rtp_buffer_ext_timestamp (&ext_base, rtp_clock_base);
|
|
/* sanity check; base rtp and provided clock_base should be close */
|
|
if (rtp_clock_base >= base_rtptime) {
|
|
if (rtp_clock_base - base_rtptime < 10 * clock_rate) {
|
|
rtp_delta = base_time +
|
|
gst_util_uint64_scale_int (rtp_clock_base - base_rtptime,
|
|
GST_SECOND, clock_rate);
|
|
} else {
|
|
use_rtp = FALSE;
|
|
}
|
|
} else {
|
|
if (base_rtptime - rtp_clock_base < 10 * clock_rate) {
|
|
rtp_delta = base_time -
|
|
gst_util_uint64_scale_int (base_rtptime - rtp_clock_base,
|
|
GST_SECOND, clock_rate);
|
|
} else {
|
|
use_rtp = FALSE;
|
|
}
|
|
}
|
|
/* warn and bail for clarity out if no sane values */
|
|
if (!use_rtp) {
|
|
GST_WARNING_OBJECT (bin, "unable to sync to provided rtptime");
|
|
return;
|
|
}
|
|
/* store to track changes */
|
|
clock_base = rtp_delta;
|
|
/* generate a fake as before,
|
|
* now equating rtptime obtained from RTP-Info,
|
|
* where the large time represent the otherwise irrelevant npt/ntp time */
|
|
stream->rtp_delta = (GST_SECOND << 28) - rtp_delta;
|
|
} else {
|
|
clock_base = rtp_clock_base;
|
|
}
|
|
|
|
all_sync = TRUE;
|
|
for (walk = client->streams; walk; walk = g_slist_next (walk)) {
|
|
GstRtpBinStream *ostream = (GstRtpBinStream *) walk->data;
|
|
|
|
if (!ostream->have_sync) {
|
|
all_sync = FALSE;
|
|
continue;
|
|
}
|
|
|
|
/* change in current stream's base from previously init'ed value
|
|
* leads to reset of all stream's base */
|
|
if (stream != ostream && stream->clock_base >= 0 &&
|
|
(stream->clock_base != clock_base)) {
|
|
GST_DEBUG_OBJECT (bin, "reset upon clock base change");
|
|
ostream->clock_base = -100 * GST_SECOND;
|
|
ostream->rtp_delta = 0;
|
|
}
|
|
|
|
if (ostream->rt_delta < min)
|
|
min = ostream->rt_delta;
|
|
if (ostream->rtp_delta < rtp_min)
|
|
rtp_min = ostream->rtp_delta;
|
|
}
|
|
|
|
/* arrange to re-sync for each stream upon significant change,
|
|
* e.g. post-seek */
|
|
all_sync = all_sync && (stream->clock_base == clock_base);
|
|
stream->clock_base = clock_base;
|
|
|
|
/* may need init performed above later on, but nothing more to do now */
|
|
if (client->nstreams <= 1)
|
|
return;
|
|
|
|
GST_DEBUG_OBJECT (bin, "client %p min delta %" G_GINT64_FORMAT
|
|
" all sync %d", client, min, all_sync);
|
|
GST_DEBUG_OBJECT (bin, "rtcp sync mode %d, use_rtp %d", rtcp_sync, use_rtp);
|
|
|
|
switch (rtcp_sync) {
|
|
case GST_RTP_BIN_RTCP_SYNC_RTP:
|
|
if (!use_rtp)
|
|
break;
|
|
GST_DEBUG_OBJECT (bin, "using rtp generated reports; "
|
|
"client %p min rtp delta %" G_GINT64_FORMAT, client, rtp_min);
|
|
/* fall-through */
|
|
case GST_RTP_BIN_RTCP_SYNC_INITIAL:
|
|
/* if all have been synced already, do not bother further */
|
|
if (all_sync) {
|
|
GST_DEBUG_OBJECT (bin, "all streams already synced; done");
|
|
return;
|
|
}
|
|
break;
|
|
default:
|
|
break;
|
|
}
|
|
|
|
/* bail out if we adjusted recently enough */
|
|
if (all_sync && (ntpnstime - bin->priv->last_ntpnstime) <
|
|
bin->rtcp_sync_interval * GST_MSECOND) {
|
|
GST_DEBUG_OBJECT (bin, "discarding RTCP sender packet for sync; "
|
|
"previous sender info too recent "
|
|
"(previous NTP %" G_GUINT64_FORMAT ")", bin->priv->last_ntpnstime);
|
|
return;
|
|
}
|
|
bin->priv->last_ntpnstime = ntpnstime;
|
|
|
|
/* calculate offsets for each stream */
|
|
for (walk = client->streams; walk; walk = g_slist_next (walk)) {
|
|
GstRtpBinStream *ostream = (GstRtpBinStream *) walk->data;
|
|
gint64 ts_offset;
|
|
|
|
/* ignore streams for which we didn't receive an SR packet yet, we
|
|
* can't synchronize them yet. We can however sync other streams just
|
|
* fine. */
|
|
if (!ostream->have_sync)
|
|
continue;
|
|
|
|
/* calculate offset to our reference stream, this should always give a
|
|
* positive number. */
|
|
if (use_rtp)
|
|
ts_offset = ostream->rtp_delta - rtp_min;
|
|
else
|
|
ts_offset = ostream->rt_delta - min;
|
|
|
|
stream_set_ts_offset (bin, ostream, ts_offset, bin->max_ts_offset,
|
|
bin->min_ts_offset, TRUE);
|
|
}
|
|
}
|
|
gst_rtp_bin_send_sync_event (stream);
|
|
|
|
return;
|
|
}
|
|
|
|
#define GST_RTCP_BUFFER_FOR_PACKETS(b,buffer,packet) \
|
|
for ((b) = gst_rtcp_buffer_get_first_packet ((buffer), (packet)); (b); \
|
|
(b) = gst_rtcp_packet_move_to_next ((packet)))
|
|
|
|
#define GST_RTCP_SDES_FOR_ITEMS(b,packet) \
|
|
for ((b) = gst_rtcp_packet_sdes_first_item ((packet)); (b); \
|
|
(b) = gst_rtcp_packet_sdes_next_item ((packet)))
|
|
|
|
#define GST_RTCP_SDES_FOR_ENTRIES(b,packet) \
|
|
for ((b) = gst_rtcp_packet_sdes_first_entry ((packet)); (b); \
|
|
(b) = gst_rtcp_packet_sdes_next_entry ((packet)))
|
|
|
|
static void
|
|
gst_rtp_bin_handle_sync (GstElement * jitterbuffer, GstStructure * s,
|
|
GstRtpBinStream * stream)
|
|
{
|
|
GstRtpBin *bin;
|
|
GstRTCPPacket packet;
|
|
guint32 ssrc;
|
|
guint64 ntpnstime, inband_ntpnstime;
|
|
gboolean have_sr;
|
|
gboolean more;
|
|
guint64 base_rtptime;
|
|
guint64 base_time;
|
|
guint clock_rate;
|
|
guint64 clock_base;
|
|
guint64 extrtptime, inband_ext_rtptime;
|
|
GstBuffer *buffer;
|
|
const gchar *cname;
|
|
GstRTCPBuffer rtcp = { NULL, };
|
|
|
|
bin = stream->bin;
|
|
|
|
GST_DEBUG_OBJECT (bin, "sync handler called");
|
|
|
|
/* get the last relation between the rtp timestamps and the gstreamer
|
|
* timestamps. We get this info directly from the jitterbuffer which
|
|
* constructs gstreamer timestamps from rtp timestamps and so it know exactly
|
|
* what the current situation is. */
|
|
if (!gst_structure_get_uint64 (s, "base-rtptime", &base_rtptime) ||
|
|
!gst_structure_get_uint64 (s, "base-time", &base_time) ||
|
|
!gst_structure_get_uint (s, "clock-rate", &clock_rate) ||
|
|
!gst_structure_get_uint64 (s, "clock-base", &clock_base)) {
|
|
/* invalid structure */
|
|
return;
|
|
}
|
|
|
|
cname = gst_structure_get_string (s, "cname");
|
|
|
|
/* if the jitterbuffer directly got the NTP timestamp then don't work
|
|
* through the RTCP SR, otherwise extract it from there */
|
|
if (gst_structure_get_uint64 (s, "inband-ntpnstime", &inband_ntpnstime)
|
|
&& gst_structure_get_uint64 (s, "inband-ext-rtptime", &inband_ext_rtptime)
|
|
&& (cname = gst_structure_get_string (s, "cname"))
|
|
&& gst_structure_get_uint (s, "ssrc", &ssrc)) {
|
|
GST_DEBUG_OBJECT (bin,
|
|
"handle sync from inband NTP-64 information for SSRC %08x", ssrc);
|
|
|
|
if (ssrc != stream->ssrc)
|
|
return;
|
|
|
|
GST_RTP_BIN_LOCK (bin);
|
|
gst_rtp_bin_associate (bin, stream, strlen (cname), (const guint8 *) cname,
|
|
inband_ntpnstime, inband_ext_rtptime, base_rtptime, base_time,
|
|
clock_rate, clock_base);
|
|
GST_RTP_BIN_UNLOCK (bin);
|
|
return;
|
|
}
|
|
|
|
if (!gst_structure_get_uint64 (s, "sr-ext-rtptime", &extrtptime)
|
|
|| !gst_structure_has_field_typed (s, "sr-buffer", GST_TYPE_BUFFER)) {
|
|
/* invalid structure */
|
|
return;
|
|
}
|
|
|
|
GST_DEBUG_OBJECT (bin, "handle sync from RTCP SR information");
|
|
|
|
/* get RTCP SR ntpnstime if available */
|
|
if (gst_structure_get_uint64 (s, "sr-ntpnstime", &ntpnstime) && cname) {
|
|
GST_RTP_BIN_LOCK (bin);
|
|
/* associate the stream to CNAME */
|
|
gst_rtp_bin_associate (bin, stream, strlen (cname),
|
|
(const guint8 *) cname, ntpnstime, extrtptime, base_rtptime,
|
|
base_time, clock_rate, clock_base);
|
|
GST_RTP_BIN_UNLOCK (bin);
|
|
return;
|
|
}
|
|
|
|
/* otherwise parse the RTCP packet */
|
|
buffer = gst_value_get_buffer (gst_structure_get_value (s, "sr-buffer"));
|
|
|
|
have_sr = FALSE;
|
|
|
|
gst_rtcp_buffer_map (buffer, GST_MAP_READ, &rtcp);
|
|
|
|
GST_RTCP_BUFFER_FOR_PACKETS (more, &rtcp, &packet) {
|
|
/* first packet must be SR or RR or else the validate would have failed */
|
|
switch (gst_rtcp_packet_get_type (&packet)) {
|
|
case GST_RTCP_TYPE_SR:
|
|
/* only parse first. There is only supposed to be one SR in the packet
|
|
* but we will deal with malformed packets gracefully by trying the
|
|
* next RTCP packet. */
|
|
if (have_sr)
|
|
continue;
|
|
|
|
/* get NTP time */
|
|
gst_rtcp_packet_sr_get_sender_info (&packet, &ssrc, &ntpnstime, NULL,
|
|
NULL, NULL);
|
|
|
|
/* convert ntptime to nanoseconds */
|
|
ntpnstime = gst_util_uint64_scale (ntpnstime, GST_SECOND,
|
|
(G_GINT64_CONSTANT (1) << 32));
|
|
|
|
GST_DEBUG_OBJECT (bin, "received sync packet from SSRC %08x", ssrc);
|
|
|
|
/* ignore SR that is not ours and check the next RTCP packet */
|
|
if (ssrc != stream->ssrc)
|
|
continue;
|
|
|
|
have_sr = TRUE;
|
|
|
|
/* If we already have the CNAME don't require parsing SDES */
|
|
if (cname) {
|
|
GST_RTP_BIN_LOCK (bin);
|
|
/* associate the stream to CNAME */
|
|
gst_rtp_bin_associate (bin, stream, strlen (cname),
|
|
(const guint8 *) cname, ntpnstime, extrtptime, base_rtptime,
|
|
base_time, clock_rate, clock_base);
|
|
GST_RTP_BIN_UNLOCK (bin);
|
|
|
|
goto out;
|
|
}
|
|
|
|
break;
|
|
case GST_RTCP_TYPE_SDES:
|
|
{
|
|
gboolean more_items;
|
|
|
|
/* Bail out if we have not seen an SR item yet. */
|
|
if (!have_sr)
|
|
goto out;
|
|
|
|
GST_RTCP_SDES_FOR_ITEMS (more_items, &packet) {
|
|
gboolean more_entries;
|
|
|
|
/* skip items that are not about the SSRC of the sender */
|
|
if (gst_rtcp_packet_sdes_get_ssrc (&packet) != ssrc)
|
|
continue;
|
|
|
|
/* find the CNAME entry */
|
|
GST_RTCP_SDES_FOR_ENTRIES (more_entries, &packet) {
|
|
GstRTCPSDESType type;
|
|
guint8 len;
|
|
const guint8 *data;
|
|
|
|
gst_rtcp_packet_sdes_get_entry (&packet, &type, &len,
|
|
(guint8 **) & data);
|
|
|
|
if (type == GST_RTCP_SDES_CNAME) {
|
|
GST_RTP_BIN_LOCK (bin);
|
|
/* associate the stream to CNAME */
|
|
gst_rtp_bin_associate (bin, stream, len, data,
|
|
ntpnstime, extrtptime, base_rtptime, base_time, clock_rate,
|
|
clock_base);
|
|
GST_RTP_BIN_UNLOCK (bin);
|
|
|
|
goto out;
|
|
}
|
|
}
|
|
}
|
|
|
|
/* only deal with first SDES, there is only supposed to be one SDES in
|
|
* the RTCP packet but we deal with bad packets gracefully. */
|
|
goto out;
|
|
}
|
|
default:
|
|
/* we can ignore these packets */
|
|
break;
|
|
}
|
|
}
|
|
out:
|
|
gst_rtcp_buffer_unmap (&rtcp);
|
|
}
|
|
|
|
/* create a new stream with @ssrc in @session. Must be called with
|
|
* RTP_SESSION_LOCK. */
|
|
static GstRtpBinStream *
|
|
create_stream (GstRtpBinSession * session, guint32 ssrc)
|
|
{
|
|
GstElement *buffer, *demux = NULL;
|
|
GstRtpBinStream *stream;
|
|
GstRtpBin *rtpbin;
|
|
GstState target;
|
|
GObjectClass *jb_class;
|
|
|
|
rtpbin = session->bin;
|
|
|
|
if (g_slist_length (session->streams) >= rtpbin->max_streams)
|
|
goto max_streams;
|
|
|
|
if (!(buffer =
|
|
session_request_element (session, SIGNAL_REQUEST_JITTERBUFFER)))
|
|
goto no_jitterbuffer;
|
|
|
|
if (!rtpbin->ignore_pt) {
|
|
if (!(demux = gst_element_factory_make ("rtpptdemux", NULL)))
|
|
goto no_demux;
|
|
}
|
|
|
|
stream = g_new0 (GstRtpBinStream, 1);
|
|
stream->ssrc = ssrc;
|
|
stream->bin = rtpbin;
|
|
stream->session = session;
|
|
stream->buffer = gst_object_ref (buffer);
|
|
stream->demux = demux;
|
|
|
|
stream->have_sync = FALSE;
|
|
stream->rt_delta = 0;
|
|
stream->avg_ts_offset = 0;
|
|
stream->is_initialized = FALSE;
|
|
stream->rtp_delta = 0;
|
|
stream->percent = 100;
|
|
stream->clock_base = -100 * GST_SECOND;
|
|
session->streams = g_slist_prepend (session->streams, stream);
|
|
|
|
jb_class = G_OBJECT_GET_CLASS (G_OBJECT (buffer));
|
|
|
|
if (g_signal_lookup ("request-pt-map", G_OBJECT_TYPE (buffer)) != 0) {
|
|
/* provide clock_rate to the jitterbuffer when needed */
|
|
stream->buffer_ptreq_sig = g_signal_connect (buffer, "request-pt-map",
|
|
(GCallback) pt_map_requested, session);
|
|
}
|
|
if (g_signal_lookup ("on-npt-stop", G_OBJECT_TYPE (buffer)) != 0) {
|
|
stream->buffer_ntpstop_sig = g_signal_connect (buffer, "on-npt-stop",
|
|
(GCallback) on_npt_stop, stream);
|
|
}
|
|
|
|
g_object_set_data (G_OBJECT (buffer), "GstRTPBin.session", session);
|
|
g_object_set_data (G_OBJECT (buffer), "GstRTPBin.stream", stream);
|
|
|
|
/* configure latency and packet lost */
|
|
if (g_object_class_find_property (jb_class, "latency"))
|
|
g_object_set (buffer, "latency", rtpbin->latency_ms, NULL);
|
|
if (g_object_class_find_property (jb_class, "drop-on-latency"))
|
|
g_object_set (buffer, "drop-on-latency", rtpbin->drop_on_latency, NULL);
|
|
if (g_object_class_find_property (jb_class, "do-lost"))
|
|
g_object_set (buffer, "do-lost", rtpbin->do_lost, NULL);
|
|
if (g_object_class_find_property (jb_class, "mode"))
|
|
g_object_set (buffer, "mode", rtpbin->buffer_mode, NULL);
|
|
if (g_object_class_find_property (jb_class, "do-retransmission"))
|
|
g_object_set (buffer, "do-retransmission", rtpbin->do_retransmission, NULL);
|
|
if (g_object_class_find_property (jb_class, "max-rtcp-rtp-time-diff"))
|
|
g_object_set (buffer, "max-rtcp-rtp-time-diff",
|
|
rtpbin->max_rtcp_rtp_time_diff, NULL);
|
|
if (g_object_class_find_property (jb_class, "max-dropout-time"))
|
|
g_object_set (buffer, "max-dropout-time", rtpbin->max_dropout_time, NULL);
|
|
if (g_object_class_find_property (jb_class, "max-misorder-time"))
|
|
g_object_set (buffer, "max-misorder-time", rtpbin->max_misorder_time, NULL);
|
|
if (g_object_class_find_property (jb_class, "rfc7273-sync"))
|
|
g_object_set (buffer, "rfc7273-sync", rtpbin->rfc7273_sync, NULL);
|
|
if (g_object_class_find_property (jb_class, "add-reference-timestamp-meta"))
|
|
g_object_set (buffer, "add-reference-timestamp-meta",
|
|
rtpbin->add_reference_timestamp_meta, NULL);
|
|
if (g_object_class_find_property (jb_class, "max-ts-offset-adjustment"))
|
|
g_object_set (buffer, "max-ts-offset-adjustment",
|
|
rtpbin->max_ts_offset_adjustment, NULL);
|
|
if (g_object_class_find_property (jb_class, "sync-interval"))
|
|
g_object_set (buffer, "sync-interval", rtpbin->rtcp_sync_interval, NULL);
|
|
|
|
g_signal_emit (rtpbin, gst_rtp_bin_signals[SIGNAL_NEW_JITTERBUFFER], 0,
|
|
buffer, session->id, ssrc);
|
|
|
|
if (!rtpbin->ignore_pt)
|
|
gst_bin_add (GST_BIN_CAST (rtpbin), demux);
|
|
|
|
/* link stuff */
|
|
if (demux)
|
|
gst_element_link_pads_full (buffer, "src", demux, "sink",
|
|
GST_PAD_LINK_CHECK_NOTHING);
|
|
|
|
if (rtpbin->buffering) {
|
|
guint64 last_out;
|
|
|
|
if (g_signal_lookup ("set-active", G_OBJECT_TYPE (buffer)) != 0) {
|
|
GST_INFO_OBJECT (rtpbin,
|
|
"bin is buffering, set jitterbuffer as not active");
|
|
g_signal_emit_by_name (buffer, "set-active", FALSE, (gint64) 0,
|
|
&last_out);
|
|
}
|
|
}
|
|
|
|
|
|
GST_OBJECT_LOCK (rtpbin);
|
|
target = GST_STATE_TARGET (rtpbin);
|
|
GST_OBJECT_UNLOCK (rtpbin);
|
|
|
|
/* from sink to source */
|
|
if (demux)
|
|
gst_element_set_state (demux, target);
|
|
|
|
gst_element_set_state (buffer, target);
|
|
|
|
return stream;
|
|
|
|
/* ERRORS */
|
|
max_streams:
|
|
{
|
|
GST_WARNING_OBJECT (rtpbin, "stream exceeds maximum (%d)",
|
|
rtpbin->max_streams);
|
|
return NULL;
|
|
}
|
|
no_jitterbuffer:
|
|
{
|
|
g_warning ("rtpbin: could not create rtpjitterbuffer element");
|
|
return NULL;
|
|
}
|
|
no_demux:
|
|
{
|
|
gst_object_unref (buffer);
|
|
g_warning ("rtpbin: could not create rtpptdemux element");
|
|
return NULL;
|
|
}
|
|
}
|
|
|
|
/* called with RTP_BIN_LOCK */
|
|
static void
|
|
free_stream (GstRtpBinStream * stream, GstRtpBin * bin)
|
|
{
|
|
GstRtpBinSession *sess = stream->session;
|
|
GSList *clients, *next_client;
|
|
|
|
GST_DEBUG_OBJECT (bin, "freeing stream %p", stream);
|
|
|
|
gst_element_set_locked_state (stream->buffer, TRUE);
|
|
if (stream->demux)
|
|
gst_element_set_locked_state (stream->demux, TRUE);
|
|
|
|
gst_element_set_state (stream->buffer, GST_STATE_NULL);
|
|
if (stream->demux)
|
|
gst_element_set_state (stream->demux, GST_STATE_NULL);
|
|
|
|
if (stream->demux) {
|
|
g_signal_handler_disconnect (stream->demux, stream->demux_newpad_sig);
|
|
g_signal_handler_disconnect (stream->demux, stream->demux_ptreq_sig);
|
|
g_signal_handler_disconnect (stream->demux, stream->demux_ptchange_sig);
|
|
g_signal_handler_disconnect (stream->demux, stream->demux_padremoved_sig);
|
|
}
|
|
|
|
if (stream->buffer_handlesync_sig)
|
|
g_signal_handler_disconnect (stream->buffer, stream->buffer_handlesync_sig);
|
|
if (stream->buffer_ptreq_sig)
|
|
g_signal_handler_disconnect (stream->buffer, stream->buffer_ptreq_sig);
|
|
if (stream->buffer_ntpstop_sig)
|
|
g_signal_handler_disconnect (stream->buffer, stream->buffer_ntpstop_sig);
|
|
|
|
sess->elements = g_slist_remove (sess->elements, stream->buffer);
|
|
remove_bin_element (stream->buffer, bin);
|
|
gst_object_unref (stream->buffer);
|
|
|
|
if (stream->demux)
|
|
gst_bin_remove (GST_BIN_CAST (bin), stream->demux);
|
|
|
|
for (clients = bin->clients; clients; clients = next_client) {
|
|
GstRtpBinClient *client = (GstRtpBinClient *) clients->data;
|
|
GSList *streams, *next_stream;
|
|
|
|
next_client = g_slist_next (clients);
|
|
|
|
for (streams = client->streams; streams; streams = next_stream) {
|
|
GstRtpBinStream *ostream = (GstRtpBinStream *) streams->data;
|
|
|
|
next_stream = g_slist_next (streams);
|
|
|
|
if (ostream == stream) {
|
|
client->streams = g_slist_delete_link (client->streams, streams);
|
|
/* If this was the last stream belonging to this client,
|
|
* clean up the client. */
|
|
if (--client->nstreams == 0) {
|
|
bin->clients = g_slist_delete_link (bin->clients, clients);
|
|
free_client (client, bin);
|
|
break;
|
|
}
|
|
}
|
|
}
|
|
}
|
|
g_free (stream);
|
|
}
|
|
|
|
/* GObject vmethods */
|
|
static void gst_rtp_bin_dispose (GObject * object);
|
|
static void gst_rtp_bin_finalize (GObject * object);
|
|
static void gst_rtp_bin_set_property (GObject * object, guint prop_id,
|
|
const GValue * value, GParamSpec * pspec);
|
|
static void gst_rtp_bin_get_property (GObject * object, guint prop_id,
|
|
GValue * value, GParamSpec * pspec);
|
|
|
|
/* GstElement vmethods */
|
|
static GstStateChangeReturn gst_rtp_bin_change_state (GstElement * element,
|
|
GstStateChange transition);
|
|
static GstPad *gst_rtp_bin_request_new_pad (GstElement * element,
|
|
GstPadTemplate * templ, const gchar * name, const GstCaps * caps);
|
|
static void gst_rtp_bin_release_pad (GstElement * element, GstPad * pad);
|
|
static void gst_rtp_bin_handle_message (GstBin * bin, GstMessage * message);
|
|
|
|
#define gst_rtp_bin_parent_class parent_class
|
|
G_DEFINE_TYPE_WITH_PRIVATE (GstRtpBin, gst_rtp_bin, GST_TYPE_BIN);
|
|
GST_ELEMENT_REGISTER_DEFINE (rtpbin, "rtpbin", GST_RANK_NONE, GST_TYPE_RTP_BIN);
|
|
|
|
static gboolean
|
|
_gst_element_accumulator (GSignalInvocationHint * ihint,
|
|
GValue * return_accu, const GValue * handler_return, gpointer dummy)
|
|
{
|
|
GstElement *element;
|
|
|
|
element = g_value_get_object (handler_return);
|
|
GST_DEBUG ("got element %" GST_PTR_FORMAT, element);
|
|
|
|
g_value_set_object (return_accu, element);
|
|
|
|
/* stop emission if we have an element */
|
|
return (element == NULL);
|
|
}
|
|
|
|
static gboolean
|
|
_gst_caps_accumulator (GSignalInvocationHint * ihint,
|
|
GValue * return_accu, const GValue * handler_return, gpointer dummy)
|
|
{
|
|
GstCaps *caps;
|
|
|
|
caps = g_value_get_boxed (handler_return);
|
|
GST_DEBUG ("got caps %" GST_PTR_FORMAT, caps);
|
|
|
|
g_value_set_boxed (return_accu, caps);
|
|
|
|
/* stop emission if we have a caps */
|
|
return (caps == NULL);
|
|
}
|
|
|
|
static void
|
|
gst_rtp_bin_class_init (GstRtpBinClass * klass)
|
|
{
|
|
GObjectClass *gobject_class;
|
|
GstElementClass *gstelement_class;
|
|
GstBinClass *gstbin_class;
|
|
|
|
gobject_class = (GObjectClass *) klass;
|
|
gstelement_class = (GstElementClass *) klass;
|
|
gstbin_class = (GstBinClass *) klass;
|
|
|
|
gobject_class->dispose = gst_rtp_bin_dispose;
|
|
gobject_class->finalize = gst_rtp_bin_finalize;
|
|
gobject_class->set_property = gst_rtp_bin_set_property;
|
|
gobject_class->get_property = gst_rtp_bin_get_property;
|
|
|
|
g_object_class_install_property (gobject_class, PROP_LATENCY,
|
|
g_param_spec_uint ("latency", "Buffer latency in ms",
|
|
"Default amount of ms to buffer in the jitterbuffers", 0,
|
|
G_MAXUINT, DEFAULT_LATENCY_MS,
|
|
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
|
|
|
|
g_object_class_install_property (gobject_class, PROP_DROP_ON_LATENCY,
|
|
g_param_spec_boolean ("drop-on-latency",
|
|
"Drop buffers when maximum latency is reached",
|
|
"Tells the jitterbuffer to never exceed the given latency in size",
|
|
DEFAULT_DROP_ON_LATENCY, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
|
|
|
|
/**
|
|
* GstRtpBin::request-pt-map:
|
|
* @rtpbin: the object which received the signal
|
|
* @session: the session
|
|
* @pt: the pt
|
|
*
|
|
* Request the payload type as #GstCaps for @pt in @session.
|
|
*/
|
|
gst_rtp_bin_signals[SIGNAL_REQUEST_PT_MAP] =
|
|
g_signal_new ("request-pt-map", G_TYPE_FROM_CLASS (klass),
|
|
G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRtpBinClass, request_pt_map),
|
|
_gst_caps_accumulator, NULL, NULL, GST_TYPE_CAPS, 2, G_TYPE_UINT,
|
|
G_TYPE_UINT);
|
|
|
|
/**
|
|
* GstRtpBin::payload-type-change:
|
|
* @rtpbin: the object which received the signal
|
|
* @session: the session
|
|
* @pt: the pt
|
|
*
|
|
* Signal that the current payload type changed to @pt in @session.
|
|
*/
|
|
gst_rtp_bin_signals[SIGNAL_PAYLOAD_TYPE_CHANGE] =
|
|
g_signal_new ("payload-type-change", G_TYPE_FROM_CLASS (klass),
|
|
G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRtpBinClass, payload_type_change),
|
|
NULL, NULL, NULL, G_TYPE_NONE, 2, G_TYPE_UINT, G_TYPE_UINT);
|
|
|
|
/**
|
|
* GstRtpBin::clear-pt-map:
|
|
* @rtpbin: the object which received the signal
|
|
*
|
|
* Clear all previously cached pt-mapping obtained with
|
|
* #GstRtpBin::request-pt-map.
|
|
*/
|
|
gst_rtp_bin_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 (GstRtpBinClass,
|
|
clear_pt_map), NULL, NULL, NULL, G_TYPE_NONE, 0, G_TYPE_NONE);
|
|
|
|
/**
|
|
* GstRtpBin::reset-sync:
|
|
* @rtpbin: the object which received the signal
|
|
*
|
|
* Reset all currently configured lip-sync parameters and require new SR
|
|
* packets for all streams before lip-sync is attempted again.
|
|
*/
|
|
gst_rtp_bin_signals[SIGNAL_RESET_SYNC] =
|
|
g_signal_new ("reset-sync", G_TYPE_FROM_CLASS (klass),
|
|
G_SIGNAL_RUN_LAST | G_SIGNAL_ACTION, G_STRUCT_OFFSET (GstRtpBinClass,
|
|
reset_sync), NULL, NULL, NULL, G_TYPE_NONE, 0, G_TYPE_NONE);
|
|
|
|
/**
|
|
* GstRtpBin::get-session:
|
|
* @rtpbin: the object which received the signal
|
|
* @id: the session id
|
|
*
|
|
* Request the related GstRtpSession as #GstElement related with session @id.
|
|
*
|
|
* Since: 1.8
|
|
*/
|
|
gst_rtp_bin_signals[SIGNAL_GET_SESSION] =
|
|
g_signal_new ("get-session", G_TYPE_FROM_CLASS (klass),
|
|
G_SIGNAL_RUN_LAST | G_SIGNAL_ACTION, G_STRUCT_OFFSET (GstRtpBinClass,
|
|
get_session), NULL, NULL, NULL, GST_TYPE_ELEMENT, 1, G_TYPE_UINT);
|
|
|
|
/**
|
|
* GstRtpBin::get-internal-session:
|
|
* @rtpbin: the object which received the signal
|
|
* @id: the session id
|
|
*
|
|
* Request the internal RTPSession object as #GObject in session @id.
|
|
*/
|
|
gst_rtp_bin_signals[SIGNAL_GET_INTERNAL_SESSION] =
|
|
g_signal_new ("get-internal-session", G_TYPE_FROM_CLASS (klass),
|
|
G_SIGNAL_RUN_LAST | G_SIGNAL_ACTION, G_STRUCT_OFFSET (GstRtpBinClass,
|
|
get_internal_session), NULL, NULL, NULL, RTP_TYPE_SESSION, 1,
|
|
G_TYPE_UINT);
|
|
|
|
/**
|
|
* GstRtpBin::get-internal-storage:
|
|
* @rtpbin: the object which received the signal
|
|
* @id: the session id
|
|
*
|
|
* Request the internal RTPStorage object as #GObject in session @id. This
|
|
* is the internal storage used by the RTPStorage element, which is used to
|
|
* keep a backlog of received RTP packets for the session @id.
|
|
*
|
|
* Since: 1.14
|
|
*/
|
|
gst_rtp_bin_signals[SIGNAL_GET_INTERNAL_STORAGE] =
|
|
g_signal_new ("get-internal-storage", G_TYPE_FROM_CLASS (klass),
|
|
G_SIGNAL_RUN_LAST | G_SIGNAL_ACTION, G_STRUCT_OFFSET (GstRtpBinClass,
|
|
get_internal_storage), NULL, NULL, NULL, G_TYPE_OBJECT, 1,
|
|
G_TYPE_UINT);
|
|
|
|
/**
|
|
* GstRtpBin::get-storage:
|
|
* @rtpbin: the object which received the signal
|
|
* @id: the session id
|
|
*
|
|
* Request the RTPStorage element as #GObject in session @id. This element
|
|
* is used to keep a backlog of received RTP packets for the session @id.
|
|
*
|
|
* Since: 1.16
|
|
*/
|
|
gst_rtp_bin_signals[SIGNAL_GET_STORAGE] =
|
|
g_signal_new ("get-storage", G_TYPE_FROM_CLASS (klass),
|
|
G_SIGNAL_RUN_LAST | G_SIGNAL_ACTION, G_STRUCT_OFFSET (GstRtpBinClass,
|
|
get_storage), NULL, NULL, NULL, GST_TYPE_ELEMENT, 1, G_TYPE_UINT);
|
|
|
|
/**
|
|
* GstRtpBin::clear-ssrc:
|
|
* @rtpbin: the object which received the signal
|
|
* @id: the session id
|
|
* @ssrc: the ssrc
|
|
*
|
|
* Remove all pads from rtpssrcdemux element associated with the specified
|
|
* ssrc. This delegate the action signal to the rtpssrcdemux element
|
|
* associated with the specified session.
|
|
*
|
|
* Since: 1.20
|
|
*/
|
|
gst_rtp_bin_signals[SIGNAL_CLEAR_SSRC] =
|
|
g_signal_new ("clear-ssrc", G_TYPE_FROM_CLASS (klass),
|
|
G_SIGNAL_RUN_LAST | G_SIGNAL_ACTION, G_STRUCT_OFFSET (GstRtpBinClass,
|
|
clear_ssrc), NULL, NULL, NULL, G_TYPE_NONE, 2,
|
|
G_TYPE_UINT, G_TYPE_UINT);
|
|
|
|
/**
|
|
* GstRtpBin::on-new-ssrc:
|
|
* @rtpbin: the object which received the signal
|
|
* @session: the session
|
|
* @ssrc: the SSRC
|
|
*
|
|
* Notify of a new SSRC that entered @session.
|
|
*/
|
|
gst_rtp_bin_signals[SIGNAL_ON_NEW_SSRC] =
|
|
g_signal_new ("on-new-ssrc", G_TYPE_FROM_CLASS (klass),
|
|
G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRtpBinClass, on_new_ssrc),
|
|
NULL, NULL, NULL, G_TYPE_NONE, 2, G_TYPE_UINT, G_TYPE_UINT);
|
|
/**
|
|
* GstRtpBin::on-ssrc-collision:
|
|
* @rtpbin: the object which received the signal
|
|
* @session: the session
|
|
* @ssrc: the SSRC
|
|
*
|
|
* Notify when we have an SSRC collision
|
|
*/
|
|
gst_rtp_bin_signals[SIGNAL_ON_SSRC_COLLISION] =
|
|
g_signal_new ("on-ssrc-collision", G_TYPE_FROM_CLASS (klass),
|
|
G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRtpBinClass, on_ssrc_collision),
|
|
NULL, NULL, NULL, G_TYPE_NONE, 2, G_TYPE_UINT, G_TYPE_UINT);
|
|
/**
|
|
* GstRtpBin::on-ssrc-validated:
|
|
* @rtpbin: the object which received the signal
|
|
* @session: the session
|
|
* @ssrc: the SSRC
|
|
*
|
|
* Notify of a new SSRC that became validated.
|
|
*/
|
|
gst_rtp_bin_signals[SIGNAL_ON_SSRC_VALIDATED] =
|
|
g_signal_new ("on-ssrc-validated", G_TYPE_FROM_CLASS (klass),
|
|
G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRtpBinClass, on_ssrc_validated),
|
|
NULL, NULL, NULL, G_TYPE_NONE, 2, G_TYPE_UINT, G_TYPE_UINT);
|
|
/**
|
|
* GstRtpBin::on-ssrc-active:
|
|
* @rtpbin: the object which received the signal
|
|
* @session: the session
|
|
* @ssrc: the SSRC
|
|
*
|
|
* Notify of a SSRC that is active, i.e., sending RTCP.
|
|
*/
|
|
gst_rtp_bin_signals[SIGNAL_ON_SSRC_ACTIVE] =
|
|
g_signal_new ("on-ssrc-active", G_TYPE_FROM_CLASS (klass),
|
|
G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRtpBinClass, on_ssrc_active),
|
|
NULL, NULL, NULL, G_TYPE_NONE, 2, G_TYPE_UINT, G_TYPE_UINT);
|
|
/**
|
|
* GstRtpBin::on-ssrc-sdes:
|
|
* @rtpbin: the object which received the signal
|
|
* @session: the session
|
|
* @ssrc: the SSRC
|
|
*
|
|
* Notify of a SSRC that is active, i.e., sending RTCP.
|
|
*/
|
|
gst_rtp_bin_signals[SIGNAL_ON_SSRC_SDES] =
|
|
g_signal_new ("on-ssrc-sdes", G_TYPE_FROM_CLASS (klass),
|
|
G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRtpBinClass, on_ssrc_sdes),
|
|
NULL, NULL, NULL, G_TYPE_NONE, 2, G_TYPE_UINT, G_TYPE_UINT);
|
|
|
|
/**
|
|
* GstRtpBin::on-bye-ssrc:
|
|
* @rtpbin: the object which received the signal
|
|
* @session: the session
|
|
* @ssrc: the SSRC
|
|
*
|
|
* Notify of an SSRC that became inactive because of a BYE packet.
|
|
*/
|
|
gst_rtp_bin_signals[SIGNAL_ON_BYE_SSRC] =
|
|
g_signal_new ("on-bye-ssrc", G_TYPE_FROM_CLASS (klass),
|
|
G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRtpBinClass, on_bye_ssrc),
|
|
NULL, NULL, NULL, G_TYPE_NONE, 2, G_TYPE_UINT, G_TYPE_UINT);
|
|
/**
|
|
* GstRtpBin::on-bye-timeout:
|
|
* @rtpbin: the object which received the signal
|
|
* @session: the session
|
|
* @ssrc: the SSRC
|
|
*
|
|
* Notify of an SSRC that has timed out because of BYE
|
|
*/
|
|
gst_rtp_bin_signals[SIGNAL_ON_BYE_TIMEOUT] =
|
|
g_signal_new ("on-bye-timeout", G_TYPE_FROM_CLASS (klass),
|
|
G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRtpBinClass, on_bye_timeout),
|
|
NULL, NULL, NULL, G_TYPE_NONE, 2, G_TYPE_UINT, G_TYPE_UINT);
|
|
/**
|
|
* GstRtpBin::on-timeout:
|
|
* @rtpbin: the object which received the signal
|
|
* @session: the session
|
|
* @ssrc: the SSRC
|
|
*
|
|
* Notify of an SSRC that has timed out
|
|
*/
|
|
gst_rtp_bin_signals[SIGNAL_ON_TIMEOUT] =
|
|
g_signal_new ("on-timeout", G_TYPE_FROM_CLASS (klass),
|
|
G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRtpBinClass, on_timeout),
|
|
NULL, NULL, NULL, G_TYPE_NONE, 2, G_TYPE_UINT, G_TYPE_UINT);
|
|
/**
|
|
* GstRtpBin::on-sender-timeout:
|
|
* @rtpbin: the object which received the signal
|
|
* @session: the session
|
|
* @ssrc: the SSRC
|
|
*
|
|
* Notify of a sender SSRC that has timed out and became a receiver
|
|
*/
|
|
gst_rtp_bin_signals[SIGNAL_ON_SENDER_TIMEOUT] =
|
|
g_signal_new ("on-sender-timeout", G_TYPE_FROM_CLASS (klass),
|
|
G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRtpBinClass, on_sender_timeout),
|
|
NULL, NULL, NULL, G_TYPE_NONE, 2, G_TYPE_UINT, G_TYPE_UINT);
|
|
|
|
/**
|
|
* GstRtpBin::on-npt-stop:
|
|
* @rtpbin: the object which received the signal
|
|
* @session: the session
|
|
* @ssrc: the SSRC
|
|
*
|
|
* Notify that SSRC sender has sent data up to the configured NPT stop time.
|
|
*/
|
|
gst_rtp_bin_signals[SIGNAL_ON_NPT_STOP] =
|
|
g_signal_new ("on-npt-stop", G_TYPE_FROM_CLASS (klass),
|
|
G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRtpBinClass, on_npt_stop),
|
|
NULL, NULL, NULL, G_TYPE_NONE, 2, G_TYPE_UINT, G_TYPE_UINT);
|
|
|
|
/**
|
|
* GstRtpBin::request-rtp-encoder:
|
|
* @rtpbin: the object which received the signal
|
|
* @session: the session
|
|
*
|
|
* Request an RTP encoder element for the given @session. The encoder
|
|
* element will be added to the bin if not previously added.
|
|
*
|
|
* If no handler is connected, no encoder will be used.
|
|
*
|
|
* Since: 1.4
|
|
*/
|
|
gst_rtp_bin_signals[SIGNAL_REQUEST_RTP_ENCODER] =
|
|
g_signal_new ("request-rtp-encoder", G_TYPE_FROM_CLASS (klass),
|
|
G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRtpBinClass,
|
|
request_rtp_encoder), _gst_element_accumulator, NULL, NULL,
|
|
GST_TYPE_ELEMENT, 1, G_TYPE_UINT);
|
|
|
|
/**
|
|
* GstRtpBin::request-rtp-decoder:
|
|
* @rtpbin: the object which received the signal
|
|
* @session: the session
|
|
*
|
|
* Request an RTP decoder element for the given @session. The decoder
|
|
* element will be added to the bin if not previously added.
|
|
*
|
|
* If no handler is connected, no encoder will be used.
|
|
*
|
|
* Since: 1.4
|
|
*/
|
|
gst_rtp_bin_signals[SIGNAL_REQUEST_RTP_DECODER] =
|
|
g_signal_new ("request-rtp-decoder", G_TYPE_FROM_CLASS (klass),
|
|
G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRtpBinClass,
|
|
request_rtp_decoder), _gst_element_accumulator, NULL,
|
|
NULL, GST_TYPE_ELEMENT, 1, G_TYPE_UINT);
|
|
|
|
/**
|
|
* GstRtpBin::request-rtcp-encoder:
|
|
* @rtpbin: the object which received the signal
|
|
* @session: the session
|
|
*
|
|
* Request an RTCP encoder element for the given @session. The encoder
|
|
* element will be added to the bin if not previously added.
|
|
*
|
|
* If no handler is connected, no encoder will be used.
|
|
*
|
|
* Since: 1.4
|
|
*/
|
|
gst_rtp_bin_signals[SIGNAL_REQUEST_RTCP_ENCODER] =
|
|
g_signal_new ("request-rtcp-encoder", G_TYPE_FROM_CLASS (klass),
|
|
G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRtpBinClass,
|
|
request_rtcp_encoder), _gst_element_accumulator, NULL, NULL,
|
|
GST_TYPE_ELEMENT, 1, G_TYPE_UINT);
|
|
|
|
/**
|
|
* GstRtpBin::request-rtcp-decoder:
|
|
* @rtpbin: the object which received the signal
|
|
* @session: the session
|
|
*
|
|
* Request an RTCP decoder element for the given @session. The decoder
|
|
* element will be added to the bin if not previously added.
|
|
*
|
|
* If no handler is connected, no encoder will be used.
|
|
*
|
|
* Since: 1.4
|
|
*/
|
|
gst_rtp_bin_signals[SIGNAL_REQUEST_RTCP_DECODER] =
|
|
g_signal_new ("request-rtcp-decoder", G_TYPE_FROM_CLASS (klass),
|
|
G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRtpBinClass,
|
|
request_rtcp_decoder), _gst_element_accumulator, NULL, NULL,
|
|
GST_TYPE_ELEMENT, 1, G_TYPE_UINT);
|
|
|
|
/**
|
|
* GstRtpBin::request-jitterbuffer:
|
|
* @rtpbin: the object which received the signal
|
|
* @session: the session
|
|
*
|
|
* Request a jitterbuffer element for the given @session.
|
|
*
|
|
* If no handler is connected, the default jitterbuffer will be used.
|
|
*
|
|
* Note: The provided element is expected to conform to the API exposed
|
|
* by the standard #GstRtpJitterBuffer. Runtime checks will be made to
|
|
* determine whether it exposes properties and signals before attempting
|
|
* to set, call or connect to them, and some functionalities of #GstRtpBin
|
|
* may not be available when that is not the case.
|
|
*
|
|
* This should be considered experimental API, as the standard jitterbuffer
|
|
* API is susceptible to change, provided elements will have to update their
|
|
* custom jitterbuffer's API to match the API of #GstRtpJitterBuffer if and
|
|
* when it changes.
|
|
*
|
|
* Since: 1.18
|
|
*/
|
|
gst_rtp_bin_signals[SIGNAL_REQUEST_JITTERBUFFER] =
|
|
g_signal_new ("request-jitterbuffer", G_TYPE_FROM_CLASS (klass),
|
|
G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRtpBinClass,
|
|
request_jitterbuffer), _gst_element_accumulator, NULL,
|
|
g_cclosure_marshal_generic, GST_TYPE_ELEMENT, 1, G_TYPE_UINT);
|
|
|
|
/**
|
|
* GstRtpBin::new-jitterbuffer:
|
|
* @rtpbin: the object which received the signal
|
|
* @jitterbuffer: the new jitterbuffer
|
|
* @session: the session
|
|
* @ssrc: the SSRC
|
|
*
|
|
* Notify that a new @jitterbuffer was created for @session and @ssrc.
|
|
* This signal can, for example, be used to configure @jitterbuffer.
|
|
*
|
|
* Since: 1.4
|
|
*/
|
|
gst_rtp_bin_signals[SIGNAL_NEW_JITTERBUFFER] =
|
|
g_signal_new ("new-jitterbuffer", G_TYPE_FROM_CLASS (klass),
|
|
G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRtpBinClass,
|
|
new_jitterbuffer), NULL, NULL, NULL,
|
|
G_TYPE_NONE, 3, GST_TYPE_ELEMENT, G_TYPE_UINT, G_TYPE_UINT);
|
|
|
|
/**
|
|
* GstRtpBin::new-storage:
|
|
* @rtpbin: the object which received the signal
|
|
* @storage: the new storage
|
|
* @session: the session
|
|
*
|
|
* Notify that a new @storage was created for @session.
|
|
* This signal can, for example, be used to configure @storage.
|
|
*
|
|
* Since: 1.14
|
|
*/
|
|
gst_rtp_bin_signals[SIGNAL_NEW_STORAGE] =
|
|
g_signal_new ("new-storage", G_TYPE_FROM_CLASS (klass),
|
|
G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRtpBinClass,
|
|
new_storage), NULL, NULL, NULL,
|
|
G_TYPE_NONE, 2, GST_TYPE_ELEMENT, G_TYPE_UINT);
|
|
|
|
/**
|
|
* GstRtpBin::request-aux-sender:
|
|
* @rtpbin: the object which received the signal
|
|
* @session: the session
|
|
*
|
|
* Request an AUX sender element for the given @session. The AUX
|
|
* element will be added to the bin.
|
|
*
|
|
* If no handler is connected, no AUX element will be used.
|
|
*
|
|
* Since: 1.4
|
|
*/
|
|
gst_rtp_bin_signals[SIGNAL_REQUEST_AUX_SENDER] =
|
|
g_signal_new ("request-aux-sender", G_TYPE_FROM_CLASS (klass),
|
|
G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRtpBinClass,
|
|
request_aux_sender), _gst_element_accumulator, NULL, NULL,
|
|
GST_TYPE_ELEMENT, 1, G_TYPE_UINT);
|
|
|
|
/**
|
|
* GstRtpBin::request-aux-receiver:
|
|
* @rtpbin: the object which received the signal
|
|
* @session: the session
|
|
*
|
|
* Request an AUX receiver element for the given @session. The AUX
|
|
* element will be added to the bin.
|
|
*
|
|
* If no handler is connected, no AUX element will be used.
|
|
*
|
|
* Since: 1.4
|
|
*/
|
|
gst_rtp_bin_signals[SIGNAL_REQUEST_AUX_RECEIVER] =
|
|
g_signal_new ("request-aux-receiver", G_TYPE_FROM_CLASS (klass),
|
|
G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRtpBinClass,
|
|
request_aux_receiver), _gst_element_accumulator, NULL, NULL,
|
|
GST_TYPE_ELEMENT, 1, G_TYPE_UINT);
|
|
|
|
/**
|
|
* GstRtpBin::request-fec-decoder:
|
|
* @rtpbin: the object which received the signal
|
|
* @session: the session index
|
|
*
|
|
* Request a FEC decoder element for the given @session. The element
|
|
* will be added to the bin after the pt demuxer. If there are multiple
|
|
* ssrc's and pt's in @session, this signal may be called multiple times for
|
|
* the same @session each corresponding to a newly discovered ssrc.
|
|
*
|
|
* If no handler is connected, no FEC decoder will be used.
|
|
*
|
|
* Warning: usage of this signal is not appropriate for the BUNDLE case,
|
|
* connect to #GstRtpBin::request-fec-decoder-full instead.
|
|
*
|
|
* Since: 1.14
|
|
*/
|
|
gst_rtp_bin_signals[SIGNAL_REQUEST_FEC_DECODER] =
|
|
g_signal_new ("request-fec-decoder", G_TYPE_FROM_CLASS (klass),
|
|
G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRtpBinClass,
|
|
request_fec_decoder), _gst_element_accumulator, NULL, NULL,
|
|
GST_TYPE_ELEMENT, 1, G_TYPE_UINT);
|
|
|
|
/**
|
|
* GstRtpBin::request-fec-decoder-full:
|
|
* @rtpbin: the object which received the signal
|
|
* @session: the session index
|
|
* @ssrc: the ssrc of the stream
|
|
* @pt: the payload type
|
|
*
|
|
* Request a FEC decoder element for the given @session. The element
|
|
* will be added to the bin after the pt demuxer. If there are multiple
|
|
* ssrc's and pt's in @session, this signal may be called multiple times for
|
|
* the same @session each corresponding to a newly discovered ssrc and payload
|
|
* type, those are provided as parameters.
|
|
*
|
|
* If no handler is connected, no FEC decoder will be used.
|
|
*
|
|
* Since: 1.20
|
|
*/
|
|
gst_rtp_bin_signals[SIGNAL_REQUEST_FEC_DECODER_FULL] =
|
|
g_signal_new ("request-fec-decoder-full", G_TYPE_FROM_CLASS (klass),
|
|
G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRtpBinClass,
|
|
request_fec_decoder), _gst_element_accumulator, NULL, NULL,
|
|
GST_TYPE_ELEMENT, 3, G_TYPE_UINT, G_TYPE_UINT, G_TYPE_UINT);
|
|
|
|
/**
|
|
* GstRtpBin::request-fec-encoder:
|
|
* @rtpbin: the object which received the signal
|
|
* @session: the session index
|
|
*
|
|
* Request a FEC encoder element for the given @session. The element
|
|
* will be added to the bin after the RTPSession.
|
|
*
|
|
* If no handler is connected, no FEC encoder will be used.
|
|
*
|
|
* Since: 1.14
|
|
*/
|
|
gst_rtp_bin_signals[SIGNAL_REQUEST_FEC_ENCODER] =
|
|
g_signal_new ("request-fec-encoder", G_TYPE_FROM_CLASS (klass),
|
|
G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstRtpBinClass,
|
|
request_fec_encoder), _gst_element_accumulator, NULL, NULL,
|
|
GST_TYPE_ELEMENT, 1, G_TYPE_UINT);
|
|
|
|
/**
|
|
* GstRtpBin::on-new-sender-ssrc:
|
|
* @rtpbin: the object which received the signal
|
|
* @session: the session
|
|
* @ssrc: the sender SSRC
|
|
*
|
|
* Notify of a new sender SSRC that entered @session.
|
|
*
|
|
* Since: 1.8
|
|
*/
|
|
gst_rtp_bin_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 (GstRtpBinClass, on_new_sender_ssrc),
|
|
NULL, NULL, NULL, G_TYPE_NONE, 2, G_TYPE_UINT, G_TYPE_UINT);
|
|
/**
|
|
* GstRtpBin::on-sender-ssrc-active:
|
|
* @rtpbin: the object which received the signal
|
|
* @session: the session
|
|
* @ssrc: the sender SSRC
|
|
*
|
|
* Notify of a sender SSRC that is active, i.e., sending RTCP.
|
|
*
|
|
* Since: 1.8
|
|
*/
|
|
gst_rtp_bin_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 (GstRtpBinClass,
|
|
on_sender_ssrc_active), NULL, NULL, NULL,
|
|
G_TYPE_NONE, 2, G_TYPE_UINT, G_TYPE_UINT);
|
|
|
|
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_DO_LOST,
|
|
g_param_spec_boolean ("do-lost", "Do Lost",
|
|
"Send an event downstream when a packet is lost", DEFAULT_DO_LOST,
|
|
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
|
|
|
|
g_object_class_install_property (gobject_class, PROP_AUTOREMOVE,
|
|
g_param_spec_boolean ("autoremove", "Auto Remove",
|
|
"Automatically remove timed out sources", DEFAULT_AUTOREMOVE,
|
|
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
|
|
|
|
g_object_class_install_property (gobject_class, PROP_IGNORE_PT,
|
|
g_param_spec_boolean ("ignore-pt", "Ignore PT",
|
|
"Do not demultiplex based on PT values", DEFAULT_IGNORE_PT,
|
|
G_PARAM_READWRITE | 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));
|
|
/**
|
|
* GstRtpBin:buffer-mode:
|
|
*
|
|
* Control the buffering and timestamping mode used by the jitterbuffer.
|
|
*/
|
|
g_object_class_install_property (gobject_class, PROP_BUFFER_MODE,
|
|
g_param_spec_enum ("buffer-mode", "Buffer Mode",
|
|
"Control the buffering algorithm in use", RTP_TYPE_JITTER_BUFFER_MODE,
|
|
DEFAULT_BUFFER_MODE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
|
|
/**
|
|
* GstRtpBin:ntp-sync:
|
|
*
|
|
* Set the NTP time from the sender reports as the running-time on the
|
|
* buffers. When both the sender and receiver have sychronized
|
|
* running-time, i.e. when the clock and base-time is shared
|
|
* between the receivers and the and the senders, this option can be
|
|
* used to synchronize receivers on multiple machines.
|
|
*/
|
|
g_object_class_install_property (gobject_class, PROP_NTP_SYNC,
|
|
g_param_spec_boolean ("ntp-sync", "Sync on NTP clock",
|
|
"Synchronize received streams to the NTP clock", DEFAULT_NTP_SYNC,
|
|
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
|
|
|
|
/**
|
|
* GstRtpBin:rtcp-sync:
|
|
*
|
|
* If not synchronizing (directly) to the NTP clock, determines how to sync
|
|
* the various streams.
|
|
*/
|
|
g_object_class_install_property (gobject_class, PROP_RTCP_SYNC,
|
|
g_param_spec_enum ("rtcp-sync", "RTCP Sync",
|
|
"Use of RTCP SR in synchronization", GST_RTP_BIN_RTCP_SYNC_TYPE,
|
|
DEFAULT_RTCP_SYNC, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
|
|
|
|
/**
|
|
* GstRtpBin:rtcp-sync-interval:
|
|
*
|
|
* Determines how often to sync streams using RTCP data or inband NTP-64
|
|
* header extensions.
|
|
*/
|
|
g_object_class_install_property (gobject_class, PROP_RTCP_SYNC_INTERVAL,
|
|
g_param_spec_uint ("rtcp-sync-interval", "RTCP Sync Interval",
|
|
"RTCP SR / NTP-64 interval synchronization (ms) (0 = always)",
|
|
0, G_MAXUINT, DEFAULT_RTCP_SYNC_INTERVAL,
|
|
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
|
|
|
|
g_object_class_install_property (gobject_class, PROP_DO_SYNC_EVENT,
|
|
g_param_spec_boolean ("do-sync-event", "Do Sync Event",
|
|
"Send event downstream when a stream is synchronized to the sender",
|
|
DEFAULT_DO_SYNC_EVENT, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
|
|
|
|
/**
|
|
* GstRtpBin:do-retransmission:
|
|
*
|
|
* Enables RTP retransmission on all streams. To control retransmission on
|
|
* a per-SSRC basis, connect to the #GstRtpBin::new-jitterbuffer signal and
|
|
* set the #GstRtpJitterBuffer:do-retransmission property on the
|
|
* #GstRtpJitterBuffer object instead.
|
|
*/
|
|
g_object_class_install_property (gobject_class, PROP_DO_RETRANSMISSION,
|
|
g_param_spec_boolean ("do-retransmission", "Do retransmission",
|
|
"Enable retransmission on all streams",
|
|
DEFAULT_DO_RETRANSMISSION,
|
|
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
|
|
|
|
/**
|
|
* GstRtpBin:rtp-profile:
|
|
*
|
|
* Sets the default RTP profile of newly created RTP sessions. The
|
|
* profile can be changed afterwards on a per-session basis.
|
|
*/
|
|
g_object_class_install_property (gobject_class, PROP_RTP_PROFILE,
|
|
g_param_spec_enum ("rtp-profile", "RTP Profile",
|
|
"Default RTP profile of newly created sessions",
|
|
GST_TYPE_RTP_PROFILE, DEFAULT_RTP_PROFILE,
|
|
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
|
|
|
|
g_object_class_install_property (gobject_class, PROP_NTP_TIME_SOURCE,
|
|
g_param_spec_enum ("ntp-time-source", "NTP Time Source",
|
|
"NTP time source for RTCP packets",
|
|
gst_rtp_ntp_time_source_get_type (), DEFAULT_NTP_TIME_SOURCE,
|
|
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
|
|
|
|
g_object_class_install_property (gobject_class, PROP_RTCP_SYNC_SEND_TIME,
|
|
g_param_spec_boolean ("rtcp-sync-send-time", "RTCP Sync Send Time",
|
|
"Use send time or capture time for RTCP sync "
|
|
"(TRUE = send time, FALSE = capture time)",
|
|
DEFAULT_RTCP_SYNC_SEND_TIME,
|
|
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
|
|
|
|
g_object_class_install_property (gobject_class, PROP_MAX_RTCP_RTP_TIME_DIFF,
|
|
g_param_spec_int ("max-rtcp-rtp-time-diff", "Max RTCP RTP Time Diff",
|
|
"Maximum amount of time in ms that the RTP time in RTCP SRs "
|
|
"is allowed to be ahead (-1 disabled)", -1, G_MAXINT,
|
|
DEFAULT_MAX_RTCP_RTP_TIME_DIFF,
|
|
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));
|
|
|
|
g_object_class_install_property (gobject_class, PROP_RFC7273_SYNC,
|
|
g_param_spec_boolean ("rfc7273-sync", "Sync on RFC7273 clock",
|
|
"Synchronize received streams to the RFC7273 clock "
|
|
"(requires clock and offset to be provided)", DEFAULT_RFC7273_SYNC,
|
|
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
|
|
|
|
/**
|
|
* GstRtpBin:add-reference-timestamp-meta:
|
|
*
|
|
* When syncing to a RFC7273 clock or after clock synchronization via RTCP or
|
|
* inband NTP-64 header extensions has happened, add #GstReferenceTimestampMeta
|
|
* to buffers with the original reconstructed reference clock timestamp.
|
|
*
|
|
* Since: 1.22
|
|
*/
|
|
g_object_class_install_property (gobject_class,
|
|
PROP_ADD_REFERENCE_TIMESTAMP_META,
|
|
g_param_spec_boolean ("add-reference-timestamp-meta",
|
|
"Add Reference Timestamp Meta",
|
|
"Add Reference Timestamp Meta to buffers with the original clock timestamp "
|
|
"before any adjustments when syncing to an RFC7273 clock or after clock "
|
|
"synchronization via RTCP or inband NTP-64 header extensions has happened.",
|
|
DEFAULT_ADD_REFERENCE_TIMESTAMP_META,
|
|
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
|
|
|
|
g_object_class_install_property (gobject_class, PROP_MAX_STREAMS,
|
|
g_param_spec_uint ("max-streams", "Max Streams",
|
|
"The maximum number of streams to create for one session",
|
|
0, G_MAXUINT, DEFAULT_MAX_STREAMS,
|
|
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
|
|
|
|
/**
|
|
* GstRtpBin:max-ts-offset-adjustment:
|
|
*
|
|
* Syncing time stamps to NTP time adds a time offset. This parameter
|
|
* specifies the maximum number of nanoseconds per frame that this time offset
|
|
* may be adjusted with. This is used to avoid sudden large changes to time
|
|
* stamps.
|
|
*
|
|
* Since: 1.14
|
|
*/
|
|
g_object_class_install_property (gobject_class, PROP_MAX_TS_OFFSET_ADJUSTMENT,
|
|
g_param_spec_uint64 ("max-ts-offset-adjustment",
|
|
"Max Timestamp Offset Adjustment",
|
|
"The maximum number of nanoseconds per frame that time stamp offsets "
|
|
"may be adjusted (0 = no limit).", 0, G_MAXUINT64,
|
|
DEFAULT_MAX_TS_OFFSET_ADJUSTMENT, G_PARAM_READWRITE |
|
|
G_PARAM_STATIC_STRINGS));
|
|
|
|
/**
|
|
* GstRtpBin:max-ts-offset:
|
|
*
|
|
* Used to set an upper limit of how large a time offset may be. This
|
|
* is used to protect against unrealistic values as a result of either
|
|
* client,server or clock issues.
|
|
*
|
|
* Since: 1.14
|
|
*/
|
|
g_object_class_install_property (gobject_class, PROP_MAX_TS_OFFSET,
|
|
g_param_spec_int64 ("max-ts-offset", "Max TS Offset",
|
|
"The maximum absolute value of the time offset in (nanoseconds). "
|
|
"Note, if the ntp-sync parameter is set the default value is "
|
|
"changed to 0 (no limit)", 0, G_MAXINT64, DEFAULT_MAX_TS_OFFSET,
|
|
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
|
|
|
|
/**
|
|
* GstRtpBin:min-ts-offset:
|
|
*
|
|
* Used to set an lower limit for when a time offset is deemed large enough
|
|
* to be useful for sync corrections.
|
|
*
|
|
* When streaming for instance audio, even very small ts_offsets cause
|
|
* audible glitches. This property is used for controlling how sensitive the
|
|
* adjustments should be to small deviations in ts_offset, occurring for
|
|
* instance due to jittery network conditions or system load.
|
|
*
|
|
* Since: 1.22
|
|
*/
|
|
g_object_class_install_property (gobject_class, PROP_MIN_TS_OFFSET,
|
|
g_param_spec_uint64 ("min-ts-offset", "Min TS Offset",
|
|
"The minimum absolute value of the time offset in (nanoseconds). "
|
|
"Used to set an lower limit for when a time offset is deemed large "
|
|
"enough to be useful for sync corrections."
|
|
"Note, if the ntp-sync parameter is set the default value is "
|
|
"changed to 0 (no limit)", 0, G_MAXUINT64, DEFAULT_MIN_TS_OFFSET,
|
|
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
|
|
|
|
/**
|
|
* GstRtpBin:ts-offset-smoothing-factor:
|
|
*
|
|
* Controls the weighting between previous and current timestamp offsets in
|
|
* a running moving average (RMA):
|
|
* ts_offset_average(n) =
|
|
* ((ts-offset-smoothing-factor - 1) * ts_offset_average(n - 1) + ts_offset(n)) /
|
|
* ts-offset-smoothing-factor
|
|
*
|
|
* This can stabilize the timestamp offset and prevent unnecessary skew
|
|
* corrections due to jitter introduced by network or system load.
|
|
*
|
|
* Since: 1.22
|
|
*/
|
|
g_object_class_install_property (gobject_class,
|
|
PROP_TS_OFFSET_SMOOTHING_FACTOR,
|
|
g_param_spec_uint ("ts-offset-smoothing-factor",
|
|
"Timestamp Offset Smoothing Factor",
|
|
"Sets a smoothing factor for the timestamp offset in number of "
|
|
"values for a calculated running moving average. "
|
|
"(0 = no smoothing factor)", 0, G_MAXUINT,
|
|
DEFAULT_TS_OFFSET_SMOOTHING_FACTOR,
|
|
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
|
|
|
|
/**
|
|
* GstRtpBin:fec-decoders:
|
|
*
|
|
* Used to provide a factory used to build the FEC decoder for a
|
|
* given session, as a command line alternative to
|
|
* #GstRtpBin::request-fec-decoder.
|
|
*
|
|
* Expects a GstStructure in the form session_id (gint) -> factory (string)
|
|
*
|
|
* Since: 1.20
|
|
*/
|
|
g_object_class_install_property (gobject_class, PROP_FEC_DECODERS,
|
|
g_param_spec_boxed ("fec-decoders", "Fec Decoders",
|
|
"GstStructure mapping from session index to FEC decoder "
|
|
"factory, eg "
|
|
"fec-decoders='fec,0=\"rtpst2022-1-fecdec\\ size-time\\=1000000000\";'",
|
|
GST_TYPE_STRUCTURE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
|
|
|
|
/**
|
|
* GstRtpBin:fec-encoders:
|
|
*
|
|
* Used to provide a factory used to build the FEC encoder for a
|
|
* given session, as a command line alternative to
|
|
* #GstRtpBin::request-fec-encoder.
|
|
*
|
|
* Expects a GstStructure in the form session_id (gint) -> factory (string)
|
|
*
|
|
* Since: 1.20
|
|
*/
|
|
g_object_class_install_property (gobject_class, PROP_FEC_ENCODERS,
|
|
g_param_spec_boxed ("fec-encoders", "Fec Encoders",
|
|
"GstStructure mapping from session index to FEC encoder "
|
|
"factory, eg "
|
|
"fec-encoders='fec,0=\"rtpst2022-1-fecenc\\ rows\\=5\\ columns\\=5\";'",
|
|
GST_TYPE_STRUCTURE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
|
|
|
|
/**
|
|
* GstRtpBin:update-ntp64-header-ext:
|
|
*
|
|
* Whether RTP NTP header extension should be updated with actual
|
|
* NTP time. If not, use the NTP time from buffer timestamp metadata
|
|
*
|
|
* Since: 1.22
|
|
*/
|
|
g_object_class_install_property (gobject_class,
|
|
PROP_UPDATE_NTP64_HEADER_EXT,
|
|
g_param_spec_boolean ("update-ntp64-header-ext",
|
|
"Update NTP-64 RTP Header Extension",
|
|
"Whether RTP NTP header extension should be updated with actual NTP time",
|
|
DEFAULT_UPDATE_NTP64_HEADER_EXT,
|
|
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
|
|
|
|
/**
|
|
* GstRtpBin:timeout-inactive-sources:
|
|
*
|
|
* Whether inactive sources should be timed out
|
|
*
|
|
* Since: 1.24
|
|
*/
|
|
g_object_class_install_property (gobject_class,
|
|
PROP_TIMEOUT_INACTIVE_SOURCES,
|
|
g_param_spec_boolean ("timeout-inactive-sources",
|
|
"Time out inactive sources",
|
|
"Whether sources that don't receive RTP or RTCP packets for longer "
|
|
"than 5x RTCP interval should be removed",
|
|
DEFAULT_TIMEOUT_INACTIVE_SOURCES,
|
|
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
|
|
|
|
gstelement_class->change_state = GST_DEBUG_FUNCPTR (gst_rtp_bin_change_state);
|
|
gstelement_class->request_new_pad =
|
|
GST_DEBUG_FUNCPTR (gst_rtp_bin_request_new_pad);
|
|
gstelement_class->release_pad = GST_DEBUG_FUNCPTR (gst_rtp_bin_release_pad);
|
|
|
|
/* sink pads */
|
|
gst_element_class_add_static_pad_template (gstelement_class,
|
|
&rtpbin_recv_rtp_sink_template);
|
|
gst_element_class_add_static_pad_template (gstelement_class,
|
|
&rtpbin_recv_fec_sink_template);
|
|
gst_element_class_add_static_pad_template (gstelement_class,
|
|
&rtpbin_recv_rtcp_sink_template);
|
|
gst_element_class_add_static_pad_template (gstelement_class,
|
|
&rtpbin_send_rtp_sink_template);
|
|
|
|
/* src pads */
|
|
gst_element_class_add_static_pad_template (gstelement_class,
|
|
&rtpbin_recv_rtp_src_template);
|
|
gst_element_class_add_static_pad_template (gstelement_class,
|
|
&rtpbin_send_rtcp_src_template);
|
|
gst_element_class_add_static_pad_template (gstelement_class,
|
|
&rtpbin_send_rtp_src_template);
|
|
gst_element_class_add_static_pad_template (gstelement_class,
|
|
&rtpbin_send_fec_src_template);
|
|
|
|
gst_element_class_set_static_metadata (gstelement_class, "RTP Bin",
|
|
"Filter/Network/RTP",
|
|
"Real-Time Transport Protocol bin",
|
|
"Wim Taymans <wim.taymans@gmail.com>");
|
|
|
|
gstbin_class->handle_message = GST_DEBUG_FUNCPTR (gst_rtp_bin_handle_message);
|
|
|
|
klass->clear_pt_map = GST_DEBUG_FUNCPTR (gst_rtp_bin_clear_pt_map);
|
|
klass->reset_sync = GST_DEBUG_FUNCPTR (gst_rtp_bin_reset_sync);
|
|
klass->get_session = GST_DEBUG_FUNCPTR (gst_rtp_bin_get_session);
|
|
klass->get_internal_session =
|
|
GST_DEBUG_FUNCPTR (gst_rtp_bin_get_internal_session);
|
|
klass->get_storage = GST_DEBUG_FUNCPTR (gst_rtp_bin_get_storage);
|
|
klass->get_internal_storage =
|
|
GST_DEBUG_FUNCPTR (gst_rtp_bin_get_internal_storage);
|
|
klass->clear_ssrc = GST_DEBUG_FUNCPTR (gst_rtp_bin_clear_ssrc);
|
|
klass->request_rtp_encoder = GST_DEBUG_FUNCPTR (gst_rtp_bin_request_encoder);
|
|
klass->request_rtp_decoder = GST_DEBUG_FUNCPTR (gst_rtp_bin_request_decoder);
|
|
klass->request_rtcp_encoder = GST_DEBUG_FUNCPTR (gst_rtp_bin_request_encoder);
|
|
klass->request_rtcp_decoder = GST_DEBUG_FUNCPTR (gst_rtp_bin_request_decoder);
|
|
klass->request_jitterbuffer =
|
|
GST_DEBUG_FUNCPTR (gst_rtp_bin_request_jitterbuffer);
|
|
|
|
GST_DEBUG_CATEGORY_INIT (gst_rtp_bin_debug, "rtpbin", 0, "RTP bin");
|
|
|
|
gst_type_mark_as_plugin_api (GST_RTP_BIN_RTCP_SYNC_TYPE, 0);
|
|
}
|
|
|
|
static void
|
|
gst_rtp_bin_init (GstRtpBin * rtpbin)
|
|
{
|
|
gchar *cname;
|
|
|
|
rtpbin->priv = gst_rtp_bin_get_instance_private (rtpbin);
|
|
g_mutex_init (&rtpbin->priv->bin_lock);
|
|
g_mutex_init (&rtpbin->priv->dyn_lock);
|
|
|
|
rtpbin->latency_ms = DEFAULT_LATENCY_MS;
|
|
rtpbin->latency_ns = DEFAULT_LATENCY_MS * GST_MSECOND;
|
|
rtpbin->drop_on_latency = DEFAULT_DROP_ON_LATENCY;
|
|
rtpbin->do_lost = DEFAULT_DO_LOST;
|
|
rtpbin->ignore_pt = DEFAULT_IGNORE_PT;
|
|
rtpbin->ntp_sync = DEFAULT_NTP_SYNC;
|
|
rtpbin->rtcp_sync = DEFAULT_RTCP_SYNC;
|
|
rtpbin->rtcp_sync_interval = DEFAULT_RTCP_SYNC_INTERVAL;
|
|
rtpbin->priv->autoremove = DEFAULT_AUTOREMOVE;
|
|
rtpbin->buffer_mode = DEFAULT_BUFFER_MODE;
|
|
rtpbin->use_pipeline_clock = DEFAULT_USE_PIPELINE_CLOCK;
|
|
rtpbin->send_sync_event = DEFAULT_DO_SYNC_EVENT;
|
|
rtpbin->do_retransmission = DEFAULT_DO_RETRANSMISSION;
|
|
rtpbin->rtp_profile = DEFAULT_RTP_PROFILE;
|
|
rtpbin->ntp_time_source = DEFAULT_NTP_TIME_SOURCE;
|
|
rtpbin->rtcp_sync_send_time = DEFAULT_RTCP_SYNC_SEND_TIME;
|
|
rtpbin->max_rtcp_rtp_time_diff = DEFAULT_MAX_RTCP_RTP_TIME_DIFF;
|
|
rtpbin->max_dropout_time = DEFAULT_MAX_DROPOUT_TIME;
|
|
rtpbin->max_misorder_time = DEFAULT_MAX_MISORDER_TIME;
|
|
rtpbin->rfc7273_sync = DEFAULT_RFC7273_SYNC;
|
|
rtpbin->add_reference_timestamp_meta = DEFAULT_ADD_REFERENCE_TIMESTAMP_META;
|
|
rtpbin->max_streams = DEFAULT_MAX_STREAMS;
|
|
rtpbin->max_ts_offset_adjustment = DEFAULT_MAX_TS_OFFSET_ADJUSTMENT;
|
|
rtpbin->max_ts_offset = DEFAULT_MAX_TS_OFFSET;
|
|
rtpbin->max_ts_offset_is_set = FALSE;
|
|
rtpbin->min_ts_offset = DEFAULT_MIN_TS_OFFSET;
|
|
rtpbin->min_ts_offset_is_set = FALSE;
|
|
rtpbin->ts_offset_smoothing_factor = DEFAULT_TS_OFFSET_SMOOTHING_FACTOR;
|
|
rtpbin->update_ntp64_header_ext = DEFAULT_UPDATE_NTP64_HEADER_EXT;
|
|
rtpbin->timeout_inactive_sources = DEFAULT_TIMEOUT_INACTIVE_SOURCES;
|
|
|
|
/* some default SDES entries */
|
|
cname = g_strdup_printf ("user%u@host-%x", g_random_int (), g_random_int ());
|
|
rtpbin->sdes = gst_structure_new ("application/x-rtp-source-sdes",
|
|
"cname", G_TYPE_STRING, cname, "tool", G_TYPE_STRING, "GStreamer", NULL);
|
|
rtpbin->fec_decoders =
|
|
gst_structure_new_empty ("application/x-rtp-fec-decoders");
|
|
rtpbin->fec_encoders =
|
|
gst_structure_new_empty ("application/x-rtp-fec-encoders");
|
|
g_free (cname);
|
|
}
|
|
|
|
static void
|
|
gst_rtp_bin_dispose (GObject * object)
|
|
{
|
|
GstRtpBin *rtpbin;
|
|
|
|
rtpbin = GST_RTP_BIN (object);
|
|
|
|
GST_RTP_BIN_LOCK (rtpbin);
|
|
GST_DEBUG_OBJECT (object, "freeing sessions");
|
|
g_slist_foreach (rtpbin->sessions, (GFunc) free_session, rtpbin);
|
|
g_slist_free (rtpbin->sessions);
|
|
rtpbin->sessions = NULL;
|
|
GST_RTP_BIN_UNLOCK (rtpbin);
|
|
|
|
G_OBJECT_CLASS (parent_class)->dispose (object);
|
|
}
|
|
|
|
static void
|
|
gst_rtp_bin_finalize (GObject * object)
|
|
{
|
|
GstRtpBin *rtpbin;
|
|
|
|
rtpbin = GST_RTP_BIN (object);
|
|
|
|
if (rtpbin->sdes)
|
|
gst_structure_free (rtpbin->sdes);
|
|
|
|
if (rtpbin->fec_decoders)
|
|
gst_structure_free (rtpbin->fec_decoders);
|
|
|
|
if (rtpbin->fec_encoders)
|
|
gst_structure_free (rtpbin->fec_encoders);
|
|
|
|
g_mutex_clear (&rtpbin->priv->bin_lock);
|
|
g_mutex_clear (&rtpbin->priv->dyn_lock);
|
|
|
|
G_OBJECT_CLASS (parent_class)->finalize (object);
|
|
}
|
|
|
|
|
|
static void
|
|
gst_rtp_bin_set_sdes_struct (GstRtpBin * bin, const GstStructure * sdes)
|
|
{
|
|
GSList *item;
|
|
|
|
if (sdes == NULL)
|
|
return;
|
|
|
|
GST_RTP_BIN_LOCK (bin);
|
|
|
|
GST_OBJECT_LOCK (bin);
|
|
if (bin->sdes)
|
|
gst_structure_free (bin->sdes);
|
|
bin->sdes = gst_structure_copy (sdes);
|
|
GST_OBJECT_UNLOCK (bin);
|
|
|
|
/* store in all sessions */
|
|
for (item = bin->sessions; item; item = g_slist_next (item)) {
|
|
GstRtpBinSession *session = item->data;
|
|
g_object_set (session->session, "sdes", sdes, NULL);
|
|
}
|
|
|
|
GST_RTP_BIN_UNLOCK (bin);
|
|
}
|
|
|
|
static void
|
|
gst_rtp_bin_set_fec_decoders_struct (GstRtpBin * bin,
|
|
const GstStructure * decoders)
|
|
{
|
|
if (decoders == NULL)
|
|
return;
|
|
|
|
GST_RTP_BIN_LOCK (bin);
|
|
|
|
GST_OBJECT_LOCK (bin);
|
|
if (bin->fec_decoders)
|
|
gst_structure_free (bin->fec_decoders);
|
|
bin->fec_decoders = gst_structure_copy (decoders);
|
|
|
|
GST_OBJECT_UNLOCK (bin);
|
|
|
|
GST_RTP_BIN_UNLOCK (bin);
|
|
}
|
|
|
|
static void
|
|
gst_rtp_bin_set_fec_encoders_struct (GstRtpBin * bin,
|
|
const GstStructure * encoders)
|
|
{
|
|
if (encoders == NULL)
|
|
return;
|
|
|
|
GST_RTP_BIN_LOCK (bin);
|
|
|
|
GST_OBJECT_LOCK (bin);
|
|
if (bin->fec_encoders)
|
|
gst_structure_free (bin->fec_encoders);
|
|
bin->fec_encoders = gst_structure_copy (encoders);
|
|
|
|
GST_OBJECT_UNLOCK (bin);
|
|
|
|
GST_RTP_BIN_UNLOCK (bin);
|
|
}
|
|
|
|
static GstStructure *
|
|
gst_rtp_bin_get_sdes_struct (GstRtpBin * bin)
|
|
{
|
|
GstStructure *result;
|
|
|
|
GST_OBJECT_LOCK (bin);
|
|
result = gst_structure_copy (bin->sdes);
|
|
GST_OBJECT_UNLOCK (bin);
|
|
|
|
return result;
|
|
}
|
|
|
|
static GstStructure *
|
|
gst_rtp_bin_get_fec_decoders_struct (GstRtpBin * bin)
|
|
{
|
|
GstStructure *result;
|
|
|
|
GST_OBJECT_LOCK (bin);
|
|
result = gst_structure_copy (bin->fec_decoders);
|
|
GST_OBJECT_UNLOCK (bin);
|
|
|
|
return result;
|
|
}
|
|
|
|
static GstStructure *
|
|
gst_rtp_bin_get_fec_encoders_struct (GstRtpBin * bin)
|
|
{
|
|
GstStructure *result;
|
|
|
|
GST_OBJECT_LOCK (bin);
|
|
result = gst_structure_copy (bin->fec_encoders);
|
|
GST_OBJECT_UNLOCK (bin);
|
|
|
|
return result;
|
|
}
|
|
|
|
static void
|
|
gst_rtp_bin_set_property (GObject * object, guint prop_id,
|
|
const GValue * value, GParamSpec * pspec)
|
|
{
|
|
GstRtpBin *rtpbin;
|
|
|
|
rtpbin = GST_RTP_BIN (object);
|
|
|
|
switch (prop_id) {
|
|
case PROP_LATENCY:
|
|
GST_RTP_BIN_LOCK (rtpbin);
|
|
rtpbin->latency_ms = g_value_get_uint (value);
|
|
rtpbin->latency_ns = rtpbin->latency_ms * GST_MSECOND;
|
|
GST_RTP_BIN_UNLOCK (rtpbin);
|
|
/* propagate the property down to the jitterbuffer */
|
|
gst_rtp_bin_propagate_property_to_jitterbuffer (rtpbin, "latency", value);
|
|
break;
|
|
case PROP_DROP_ON_LATENCY:
|
|
GST_RTP_BIN_LOCK (rtpbin);
|
|
rtpbin->drop_on_latency = g_value_get_boolean (value);
|
|
GST_RTP_BIN_UNLOCK (rtpbin);
|
|
/* propagate the property down to the jitterbuffer */
|
|
gst_rtp_bin_propagate_property_to_jitterbuffer (rtpbin,
|
|
"drop-on-latency", value);
|
|
break;
|
|
case PROP_SDES:
|
|
gst_rtp_bin_set_sdes_struct (rtpbin, g_value_get_boxed (value));
|
|
break;
|
|
case PROP_DO_LOST:
|
|
GST_RTP_BIN_LOCK (rtpbin);
|
|
rtpbin->do_lost = g_value_get_boolean (value);
|
|
GST_RTP_BIN_UNLOCK (rtpbin);
|
|
gst_rtp_bin_propagate_property_to_jitterbuffer (rtpbin, "do-lost", value);
|
|
break;
|
|
case PROP_NTP_SYNC:
|
|
rtpbin->ntp_sync = g_value_get_boolean (value);
|
|
/* The default value of max_ts_offset depends on ntp_sync. If user
|
|
* hasn't set it then change default value */
|
|
if (!rtpbin->max_ts_offset_is_set) {
|
|
if (rtpbin->ntp_sync) {
|
|
rtpbin->max_ts_offset = 0;
|
|
} else {
|
|
rtpbin->max_ts_offset = DEFAULT_MAX_TS_OFFSET;
|
|
}
|
|
}
|
|
if (!rtpbin->min_ts_offset_is_set) {
|
|
if (rtpbin->ntp_sync) {
|
|
rtpbin->min_ts_offset = 0;
|
|
} else {
|
|
rtpbin->min_ts_offset = DEFAULT_MIN_TS_OFFSET;
|
|
}
|
|
}
|
|
break;
|
|
case PROP_RTCP_SYNC:
|
|
g_atomic_int_set (&rtpbin->rtcp_sync, g_value_get_enum (value));
|
|
break;
|
|
case PROP_RTCP_SYNC_INTERVAL:
|
|
rtpbin->rtcp_sync_interval = g_value_get_uint (value);
|
|
break;
|
|
case PROP_IGNORE_PT:
|
|
rtpbin->ignore_pt = g_value_get_boolean (value);
|
|
break;
|
|
case PROP_AUTOREMOVE:
|
|
rtpbin->priv->autoremove = g_value_get_boolean (value);
|
|
break;
|
|
case PROP_USE_PIPELINE_CLOCK:
|
|
{
|
|
GSList *sessions;
|
|
GST_RTP_BIN_LOCK (rtpbin);
|
|
rtpbin->use_pipeline_clock = g_value_get_boolean (value);
|
|
for (sessions = rtpbin->sessions; sessions;
|
|
sessions = g_slist_next (sessions)) {
|
|
GstRtpBinSession *session = (GstRtpBinSession *) sessions->data;
|
|
|
|
g_object_set (G_OBJECT (session->session),
|
|
"use-pipeline-clock", rtpbin->use_pipeline_clock, NULL);
|
|
}
|
|
GST_RTP_BIN_UNLOCK (rtpbin);
|
|
}
|
|
break;
|
|
case PROP_DO_SYNC_EVENT:
|
|
rtpbin->send_sync_event = g_value_get_boolean (value);
|
|
break;
|
|
case PROP_BUFFER_MODE:
|
|
GST_RTP_BIN_LOCK (rtpbin);
|
|
rtpbin->buffer_mode = g_value_get_enum (value);
|
|
GST_RTP_BIN_UNLOCK (rtpbin);
|
|
/* propagate the property down to the jitterbuffer */
|
|
gst_rtp_bin_propagate_property_to_jitterbuffer (rtpbin, "mode", value);
|
|
break;
|
|
case PROP_DO_RETRANSMISSION:
|
|
GST_RTP_BIN_LOCK (rtpbin);
|
|
rtpbin->do_retransmission = g_value_get_boolean (value);
|
|
GST_RTP_BIN_UNLOCK (rtpbin);
|
|
gst_rtp_bin_propagate_property_to_jitterbuffer (rtpbin,
|
|
"do-retransmission", value);
|
|
break;
|
|
case PROP_RTP_PROFILE:
|
|
rtpbin->rtp_profile = g_value_get_enum (value);
|
|
break;
|
|
case PROP_NTP_TIME_SOURCE:{
|
|
GSList *sessions;
|
|
GST_RTP_BIN_LOCK (rtpbin);
|
|
rtpbin->ntp_time_source = g_value_get_enum (value);
|
|
for (sessions = rtpbin->sessions; sessions;
|
|
sessions = g_slist_next (sessions)) {
|
|
GstRtpBinSession *session = (GstRtpBinSession *) sessions->data;
|
|
|
|
g_object_set (G_OBJECT (session->session),
|
|
"ntp-time-source", rtpbin->ntp_time_source, NULL);
|
|
}
|
|
GST_RTP_BIN_UNLOCK (rtpbin);
|
|
break;
|
|
}
|
|
case PROP_RTCP_SYNC_SEND_TIME:{
|
|
GSList *sessions;
|
|
GST_RTP_BIN_LOCK (rtpbin);
|
|
rtpbin->rtcp_sync_send_time = g_value_get_boolean (value);
|
|
for (sessions = rtpbin->sessions; sessions;
|
|
sessions = g_slist_next (sessions)) {
|
|
GstRtpBinSession *session = (GstRtpBinSession *) sessions->data;
|
|
|
|
g_object_set (G_OBJECT (session->session),
|
|
"rtcp-sync-send-time", rtpbin->rtcp_sync_send_time, NULL);
|
|
}
|
|
GST_RTP_BIN_UNLOCK (rtpbin);
|
|
break;
|
|
}
|
|
case PROP_MAX_RTCP_RTP_TIME_DIFF:
|
|
GST_RTP_BIN_LOCK (rtpbin);
|
|
rtpbin->max_rtcp_rtp_time_diff = g_value_get_int (value);
|
|
GST_RTP_BIN_UNLOCK (rtpbin);
|
|
gst_rtp_bin_propagate_property_to_jitterbuffer (rtpbin,
|
|
"max-rtcp-rtp-time-diff", value);
|
|
break;
|
|
case PROP_MAX_DROPOUT_TIME:
|
|
GST_RTP_BIN_LOCK (rtpbin);
|
|
rtpbin->max_dropout_time = g_value_get_uint (value);
|
|
GST_RTP_BIN_UNLOCK (rtpbin);
|
|
gst_rtp_bin_propagate_property_to_jitterbuffer (rtpbin,
|
|
"max-dropout-time", value);
|
|
gst_rtp_bin_propagate_property_to_session (rtpbin, "max-dropout-time",
|
|
value);
|
|
break;
|
|
case PROP_MAX_MISORDER_TIME:
|
|
GST_RTP_BIN_LOCK (rtpbin);
|
|
rtpbin->max_misorder_time = g_value_get_uint (value);
|
|
GST_RTP_BIN_UNLOCK (rtpbin);
|
|
gst_rtp_bin_propagate_property_to_jitterbuffer (rtpbin,
|
|
"max-misorder-time", value);
|
|
gst_rtp_bin_propagate_property_to_session (rtpbin, "max-misorder-time",
|
|
value);
|
|
break;
|
|
case PROP_RFC7273_SYNC:
|
|
rtpbin->rfc7273_sync = g_value_get_boolean (value);
|
|
gst_rtp_bin_propagate_property_to_jitterbuffer (rtpbin,
|
|
"rfc7273-sync", value);
|
|
break;
|
|
case PROP_ADD_REFERENCE_TIMESTAMP_META:
|
|
rtpbin->add_reference_timestamp_meta = g_value_get_boolean (value);
|
|
gst_rtp_bin_propagate_property_to_jitterbuffer (rtpbin,
|
|
"add-reference-timestamp-meta", value);
|
|
break;
|
|
case PROP_MAX_STREAMS:
|
|
rtpbin->max_streams = g_value_get_uint (value);
|
|
break;
|
|
case PROP_MAX_TS_OFFSET_ADJUSTMENT:
|
|
rtpbin->max_ts_offset_adjustment = g_value_get_uint64 (value);
|
|
gst_rtp_bin_propagate_property_to_jitterbuffer (rtpbin,
|
|
"max-ts-offset-adjustment", value);
|
|
break;
|
|
case PROP_MAX_TS_OFFSET:
|
|
rtpbin->max_ts_offset = g_value_get_int64 (value);
|
|
rtpbin->max_ts_offset_is_set = TRUE;
|
|
break;
|
|
case PROP_MIN_TS_OFFSET:
|
|
rtpbin->min_ts_offset = g_value_get_uint64 (value);
|
|
rtpbin->min_ts_offset_is_set = TRUE;
|
|
break;
|
|
case PROP_TS_OFFSET_SMOOTHING_FACTOR:
|
|
rtpbin->ts_offset_smoothing_factor = g_value_get_uint (value);
|
|
break;
|
|
case PROP_FEC_DECODERS:
|
|
gst_rtp_bin_set_fec_decoders_struct (rtpbin, g_value_get_boxed (value));
|
|
break;
|
|
case PROP_FEC_ENCODERS:
|
|
gst_rtp_bin_set_fec_encoders_struct (rtpbin, g_value_get_boxed (value));
|
|
break;
|
|
case PROP_UPDATE_NTP64_HEADER_EXT:
|
|
GST_RTP_BIN_LOCK (rtpbin);
|
|
rtpbin->update_ntp64_header_ext = g_value_get_boolean (value);
|
|
GST_RTP_BIN_UNLOCK (rtpbin);
|
|
gst_rtp_bin_propagate_property_to_session (rtpbin,
|
|
"update-ntp64-header-ext", value);
|
|
break;
|
|
case PROP_TIMEOUT_INACTIVE_SOURCES:
|
|
GST_RTP_BIN_LOCK (rtpbin);
|
|
rtpbin->timeout_inactive_sources = g_value_get_boolean (value);
|
|
GST_RTP_BIN_UNLOCK (rtpbin);
|
|
gst_rtp_bin_propagate_property_to_session (rtpbin,
|
|
"timeout-inactive-sources", value);
|
|
break;
|
|
default:
|
|
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
|
|
break;
|
|
}
|
|
}
|
|
|
|
static void
|
|
gst_rtp_bin_get_property (GObject * object, guint prop_id,
|
|
GValue * value, GParamSpec * pspec)
|
|
{
|
|
GstRtpBin *rtpbin;
|
|
|
|
rtpbin = GST_RTP_BIN (object);
|
|
|
|
switch (prop_id) {
|
|
case PROP_LATENCY:
|
|
GST_RTP_BIN_LOCK (rtpbin);
|
|
g_value_set_uint (value, rtpbin->latency_ms);
|
|
GST_RTP_BIN_UNLOCK (rtpbin);
|
|
break;
|
|
case PROP_DROP_ON_LATENCY:
|
|
GST_RTP_BIN_LOCK (rtpbin);
|
|
g_value_set_boolean (value, rtpbin->drop_on_latency);
|
|
GST_RTP_BIN_UNLOCK (rtpbin);
|
|
break;
|
|
case PROP_SDES:
|
|
g_value_take_boxed (value, gst_rtp_bin_get_sdes_struct (rtpbin));
|
|
break;
|
|
case PROP_DO_LOST:
|
|
GST_RTP_BIN_LOCK (rtpbin);
|
|
g_value_set_boolean (value, rtpbin->do_lost);
|
|
GST_RTP_BIN_UNLOCK (rtpbin);
|
|
break;
|
|
case PROP_IGNORE_PT:
|
|
g_value_set_boolean (value, rtpbin->ignore_pt);
|
|
break;
|
|
case PROP_NTP_SYNC:
|
|
g_value_set_boolean (value, rtpbin->ntp_sync);
|
|
break;
|
|
case PROP_RTCP_SYNC:
|
|
g_value_set_enum (value, g_atomic_int_get (&rtpbin->rtcp_sync));
|
|
break;
|
|
case PROP_RTCP_SYNC_INTERVAL:
|
|
g_value_set_uint (value, rtpbin->rtcp_sync_interval);
|
|
break;
|
|
case PROP_AUTOREMOVE:
|
|
g_value_set_boolean (value, rtpbin->priv->autoremove);
|
|
break;
|
|
case PROP_BUFFER_MODE:
|
|
g_value_set_enum (value, rtpbin->buffer_mode);
|
|
break;
|
|
case PROP_USE_PIPELINE_CLOCK:
|
|
g_value_set_boolean (value, rtpbin->use_pipeline_clock);
|
|
break;
|
|
case PROP_DO_SYNC_EVENT:
|
|
g_value_set_boolean (value, rtpbin->send_sync_event);
|
|
break;
|
|
case PROP_DO_RETRANSMISSION:
|
|
GST_RTP_BIN_LOCK (rtpbin);
|
|
g_value_set_boolean (value, rtpbin->do_retransmission);
|
|
GST_RTP_BIN_UNLOCK (rtpbin);
|
|
break;
|
|
case PROP_RTP_PROFILE:
|
|
g_value_set_enum (value, rtpbin->rtp_profile);
|
|
break;
|
|
case PROP_NTP_TIME_SOURCE:
|
|
g_value_set_enum (value, rtpbin->ntp_time_source);
|
|
break;
|
|
case PROP_RTCP_SYNC_SEND_TIME:
|
|
g_value_set_boolean (value, rtpbin->rtcp_sync_send_time);
|
|
break;
|
|
case PROP_MAX_RTCP_RTP_TIME_DIFF:
|
|
GST_RTP_BIN_LOCK (rtpbin);
|
|
g_value_set_int (value, rtpbin->max_rtcp_rtp_time_diff);
|
|
GST_RTP_BIN_UNLOCK (rtpbin);
|
|
break;
|
|
case PROP_MAX_DROPOUT_TIME:
|
|
g_value_set_uint (value, rtpbin->max_dropout_time);
|
|
break;
|
|
case PROP_MAX_MISORDER_TIME:
|
|
g_value_set_uint (value, rtpbin->max_misorder_time);
|
|
break;
|
|
case PROP_RFC7273_SYNC:
|
|
g_value_set_boolean (value, rtpbin->rfc7273_sync);
|
|
break;
|
|
case PROP_ADD_REFERENCE_TIMESTAMP_META:
|
|
g_value_set_boolean (value, rtpbin->add_reference_timestamp_meta);
|
|
break;
|
|
case PROP_MAX_STREAMS:
|
|
g_value_set_uint (value, rtpbin->max_streams);
|
|
break;
|
|
case PROP_MAX_TS_OFFSET_ADJUSTMENT:
|
|
g_value_set_uint64 (value, rtpbin->max_ts_offset_adjustment);
|
|
break;
|
|
case PROP_MAX_TS_OFFSET:
|
|
g_value_set_int64 (value, rtpbin->max_ts_offset);
|
|
break;
|
|
case PROP_MIN_TS_OFFSET:
|
|
g_value_set_uint64 (value, rtpbin->min_ts_offset);
|
|
break;
|
|
case PROP_TS_OFFSET_SMOOTHING_FACTOR:
|
|
g_value_set_uint (value, rtpbin->ts_offset_smoothing_factor);
|
|
break;
|
|
case PROP_FEC_DECODERS:
|
|
g_value_take_boxed (value, gst_rtp_bin_get_fec_decoders_struct (rtpbin));
|
|
break;
|
|
case PROP_FEC_ENCODERS:
|
|
g_value_take_boxed (value, gst_rtp_bin_get_fec_encoders_struct (rtpbin));
|
|
break;
|
|
case PROP_UPDATE_NTP64_HEADER_EXT:
|
|
g_value_set_boolean (value, rtpbin->update_ntp64_header_ext);
|
|
break;
|
|
case PROP_TIMEOUT_INACTIVE_SOURCES:
|
|
g_value_set_boolean (value, rtpbin->timeout_inactive_sources);
|
|
break;
|
|
default:
|
|
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
|
|
break;
|
|
}
|
|
}
|
|
|
|
static void
|
|
gst_rtp_bin_handle_message (GstBin * bin, GstMessage * message)
|
|
{
|
|
GstRtpBin *rtpbin;
|
|
|
|
rtpbin = GST_RTP_BIN (bin);
|
|
|
|
switch (GST_MESSAGE_TYPE (message)) {
|
|
case GST_MESSAGE_ELEMENT:
|
|
{
|
|
const GstStructure *s = gst_message_get_structure (message);
|
|
|
|
/* we change the structure name and add the session ID to it */
|
|
if (gst_structure_has_name (s, "application/x-rtp-source-sdes")) {
|
|
GstRtpBinSession *sess;
|
|
|
|
/* find the session we set it as object data */
|
|
sess = g_object_get_data (G_OBJECT (GST_MESSAGE_SRC (message)),
|
|
"GstRTPBin.session");
|
|
|
|
if (G_LIKELY (sess)) {
|
|
message = gst_message_make_writable (message);
|
|
s = gst_message_get_structure (message);
|
|
gst_structure_set ((GstStructure *) s, "session", G_TYPE_UINT,
|
|
sess->id, NULL);
|
|
}
|
|
}
|
|
GST_BIN_CLASS (parent_class)->handle_message (bin, message);
|
|
break;
|
|
}
|
|
case GST_MESSAGE_BUFFERING:
|
|
{
|
|
gint percent;
|
|
gint min_percent = 100;
|
|
GSList *sessions, *streams;
|
|
GstRtpBinStream *stream;
|
|
gboolean change = FALSE, active = FALSE;
|
|
GstClockTime min_out_time;
|
|
GstBufferingMode mode;
|
|
gint avg_in, avg_out;
|
|
gint64 buffering_left;
|
|
|
|
gst_message_parse_buffering (message, &percent);
|
|
gst_message_parse_buffering_stats (message, &mode, &avg_in, &avg_out,
|
|
&buffering_left);
|
|
|
|
stream =
|
|
g_object_get_data (G_OBJECT (GST_MESSAGE_SRC (message)),
|
|
"GstRTPBin.stream");
|
|
|
|
GST_DEBUG_OBJECT (bin, "got percent %d from stream %p", percent, stream);
|
|
|
|
/* get the stream */
|
|
if (G_LIKELY (stream)) {
|
|
GST_RTP_BIN_LOCK (rtpbin);
|
|
/* fill in the percent */
|
|
stream->percent = percent;
|
|
|
|
/* calculate the min value for all streams */
|
|
for (sessions = rtpbin->sessions; sessions;
|
|
sessions = g_slist_next (sessions)) {
|
|
GstRtpBinSession *session = (GstRtpBinSession *) sessions->data;
|
|
|
|
GST_RTP_SESSION_LOCK (session);
|
|
if (session->streams) {
|
|
for (streams = session->streams; streams;
|
|
streams = g_slist_next (streams)) {
|
|
GstRtpBinStream *stream = (GstRtpBinStream *) streams->data;
|
|
|
|
GST_DEBUG_OBJECT (bin, "stream %p percent %d", stream,
|
|
stream->percent);
|
|
|
|
/* find min percent */
|
|
if (min_percent > stream->percent)
|
|
min_percent = stream->percent;
|
|
}
|
|
} else {
|
|
GST_INFO_OBJECT (bin,
|
|
"session has no streams, setting min_percent to 0");
|
|
min_percent = 0;
|
|
}
|
|
GST_RTP_SESSION_UNLOCK (session);
|
|
}
|
|
GST_DEBUG_OBJECT (bin, "min percent %d", min_percent);
|
|
|
|
if (rtpbin->buffering) {
|
|
if (min_percent == 100) {
|
|
rtpbin->buffering = FALSE;
|
|
active = TRUE;
|
|
change = TRUE;
|
|
}
|
|
} else {
|
|
if (min_percent < 100) {
|
|
/* pause the streams */
|
|
rtpbin->buffering = TRUE;
|
|
active = FALSE;
|
|
change = TRUE;
|
|
}
|
|
}
|
|
GST_RTP_BIN_UNLOCK (rtpbin);
|
|
|
|
gst_message_unref (message);
|
|
|
|
/* make a new buffering message with the min value */
|
|
message =
|
|
gst_message_new_buffering (GST_OBJECT_CAST (bin), min_percent);
|
|
gst_message_set_buffering_stats (message, mode, avg_in, avg_out,
|
|
buffering_left);
|
|
|
|
if (G_UNLIKELY (change)) {
|
|
GstClock *clock;
|
|
guint64 running_time = 0;
|
|
guint64 offset = 0;
|
|
|
|
/* figure out the running time when we have a clock */
|
|
if (G_LIKELY ((clock =
|
|
gst_element_get_clock (GST_ELEMENT_CAST (bin))))) {
|
|
guint64 now, base_time;
|
|
|
|
now = gst_clock_get_time (clock);
|
|
base_time = gst_element_get_base_time (GST_ELEMENT_CAST (bin));
|
|
running_time = now - base_time;
|
|
gst_object_unref (clock);
|
|
}
|
|
GST_DEBUG_OBJECT (bin,
|
|
"running time now %" GST_TIME_FORMAT,
|
|
GST_TIME_ARGS (running_time));
|
|
|
|
GST_RTP_BIN_LOCK (rtpbin);
|
|
|
|
/* when we reactivate, calculate the offsets so that all streams have
|
|
* an output time that is at least as big as the running_time */
|
|
offset = 0;
|
|
if (active) {
|
|
if (running_time > rtpbin->buffer_start) {
|
|
offset = running_time - rtpbin->buffer_start;
|
|
if (offset >= rtpbin->latency_ns)
|
|
offset -= rtpbin->latency_ns;
|
|
else
|
|
offset = 0;
|
|
}
|
|
}
|
|
|
|
/* pause all streams */
|
|
min_out_time = -1;
|
|
for (sessions = rtpbin->sessions; sessions;
|
|
sessions = g_slist_next (sessions)) {
|
|
GstRtpBinSession *session = (GstRtpBinSession *) sessions->data;
|
|
|
|
GST_RTP_SESSION_LOCK (session);
|
|
for (streams = session->streams; streams;
|
|
streams = g_slist_next (streams)) {
|
|
GstRtpBinStream *stream = (GstRtpBinStream *) streams->data;
|
|
GstElement *element = stream->buffer;
|
|
guint64 last_out = -1;
|
|
|
|
if (g_signal_lookup ("set-active", G_OBJECT_TYPE (element)) != 0) {
|
|
g_signal_emit_by_name (element, "set-active", active, offset,
|
|
&last_out);
|
|
}
|
|
|
|
if (!active) {
|
|
g_object_get (element, "percent", &stream->percent, NULL);
|
|
|
|
if (last_out == -1)
|
|
last_out = 0;
|
|
if (min_out_time == -1 || last_out < min_out_time)
|
|
min_out_time = last_out;
|
|
}
|
|
|
|
GST_DEBUG_OBJECT (bin,
|
|
"setting %p to %d, offset %" GST_TIME_FORMAT ", last %"
|
|
GST_TIME_FORMAT ", percent %d", element, active,
|
|
GST_TIME_ARGS (offset), GST_TIME_ARGS (last_out),
|
|
stream->percent);
|
|
}
|
|
GST_RTP_SESSION_UNLOCK (session);
|
|
}
|
|
GST_DEBUG_OBJECT (bin,
|
|
"min out time %" GST_TIME_FORMAT, GST_TIME_ARGS (min_out_time));
|
|
|
|
/* the buffer_start is the min out time of all paused jitterbuffers */
|
|
if (!active)
|
|
rtpbin->buffer_start = min_out_time;
|
|
|
|
GST_RTP_BIN_UNLOCK (rtpbin);
|
|
}
|
|
}
|
|
GST_BIN_CLASS (parent_class)->handle_message (bin, message);
|
|
break;
|
|
}
|
|
default:
|
|
{
|
|
GST_BIN_CLASS (parent_class)->handle_message (bin, message);
|
|
break;
|
|
}
|
|
}
|
|
}
|
|
|
|
static GstStateChangeReturn
|
|
gst_rtp_bin_change_state (GstElement * element, GstStateChange transition)
|
|
{
|
|
GstStateChangeReturn res;
|
|
GstRtpBin *rtpbin;
|
|
GstRtpBinPrivate *priv;
|
|
|
|
rtpbin = GST_RTP_BIN (element);
|
|
priv = rtpbin->priv;
|
|
|
|
switch (transition) {
|
|
case GST_STATE_CHANGE_NULL_TO_READY:
|
|
break;
|
|
case GST_STATE_CHANGE_READY_TO_PAUSED:
|
|
priv->last_ntpnstime = 0;
|
|
GST_LOG_OBJECT (rtpbin, "clearing shutdown flag");
|
|
g_atomic_int_set (&priv->shutdown, 0);
|
|
break;
|
|
case GST_STATE_CHANGE_PAUSED_TO_READY:
|
|
GST_LOG_OBJECT (rtpbin, "setting shutdown flag");
|
|
g_atomic_int_set (&priv->shutdown, 1);
|
|
/* wait for all callbacks to end by taking the lock. No new callbacks will
|
|
* be able to happen as we set the shutdown flag. */
|
|
GST_RTP_BIN_DYN_LOCK (rtpbin);
|
|
GST_LOG_OBJECT (rtpbin, "dynamic lock taken, we can continue shutdown");
|
|
GST_RTP_BIN_DYN_UNLOCK (rtpbin);
|
|
break;
|
|
default:
|
|
break;
|
|
}
|
|
|
|
res = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
|
|
|
|
switch (transition) {
|
|
case GST_STATE_CHANGE_PLAYING_TO_PAUSED:
|
|
break;
|
|
case GST_STATE_CHANGE_PAUSED_TO_READY:
|
|
break;
|
|
case GST_STATE_CHANGE_READY_TO_NULL:
|
|
break;
|
|
default:
|
|
break;
|
|
}
|
|
return res;
|
|
}
|
|
|
|
static GstElement *
|
|
session_request_element_full (GstRtpBinSession * session, guint signal,
|
|
guint ssrc, guint8 pt)
|
|
{
|
|
GstElement *element = NULL;
|
|
GstRtpBin *bin = session->bin;
|
|
|
|
g_signal_emit (bin, gst_rtp_bin_signals[signal], 0, session->id, ssrc, pt,
|
|
&element);
|
|
|
|
if (element) {
|
|
if (!bin_manage_element (bin, element))
|
|
goto manage_failed;
|
|
session->elements = g_slist_prepend (session->elements, element);
|
|
}
|
|
return element;
|
|
|
|
/* ERRORS */
|
|
manage_failed:
|
|
{
|
|
GST_WARNING_OBJECT (bin, "unable to manage element");
|
|
gst_object_unref (element);
|
|
return NULL;
|
|
}
|
|
}
|
|
|
|
static GstElement *
|
|
session_request_element (GstRtpBinSession * session, guint signal)
|
|
{
|
|
GstElement *element = NULL;
|
|
GstRtpBin *bin = session->bin;
|
|
|
|
g_signal_emit (bin, gst_rtp_bin_signals[signal], 0, session->id, &element);
|
|
|
|
if (element) {
|
|
if (!bin_manage_element (bin, element))
|
|
goto manage_failed;
|
|
session->elements = g_slist_prepend (session->elements, element);
|
|
}
|
|
return element;
|
|
|
|
/* ERRORS */
|
|
manage_failed:
|
|
{
|
|
GST_WARNING_OBJECT (bin, "unable to manage element");
|
|
gst_object_unref (element);
|
|
return NULL;
|
|
}
|
|
}
|
|
|
|
static gboolean
|
|
copy_sticky_events (GstPad * pad, GstEvent ** event, gpointer user_data)
|
|
{
|
|
GstPad *gpad = GST_PAD_CAST (user_data);
|
|
|
|
GST_DEBUG_OBJECT (gpad, "store sticky event %" GST_PTR_FORMAT, *event);
|
|
gst_pad_store_sticky_event (gpad, *event);
|
|
|
|
return TRUE;
|
|
}
|
|
|
|
static gboolean
|
|
ensure_early_fec_decoder (GstRtpBin * rtpbin, GstRtpBinSession * session)
|
|
{
|
|
const gchar *factory;
|
|
gchar *sess_id_str;
|
|
|
|
if (session->early_fec_decoder)
|
|
goto done;
|
|
|
|
sess_id_str = g_strdup_printf ("%u", session->id);
|
|
factory = gst_structure_get_string (rtpbin->fec_decoders, sess_id_str);
|
|
g_free (sess_id_str);
|
|
|
|
/* First try the property */
|
|
if (factory) {
|
|
GError *err = NULL;
|
|
|
|
session->early_fec_decoder =
|
|
gst_parse_bin_from_description_full (factory, TRUE, NULL,
|
|
GST_PARSE_FLAG_NO_SINGLE_ELEMENT_BINS | GST_PARSE_FLAG_FATAL_ERRORS,
|
|
&err);
|
|
if (!session->early_fec_decoder) {
|
|
GST_ERROR_OBJECT (rtpbin, "Failed to build decoder from factory: %s",
|
|
err->message);
|
|
}
|
|
|
|
bin_manage_element (session->bin, session->early_fec_decoder);
|
|
session->elements =
|
|
g_slist_prepend (session->elements, session->early_fec_decoder);
|
|
GST_INFO_OBJECT (rtpbin, "Built FEC decoder: %" GST_PTR_FORMAT
|
|
" for session %u", session->early_fec_decoder, session->id);
|
|
}
|
|
|
|
/* Do not fallback to the signal as the signal expects a fec decoder to
|
|
* be placed at a different place in the pipeline */
|
|
|
|
done:
|
|
return session->early_fec_decoder != NULL;
|
|
}
|
|
|
|
static void
|
|
expose_recv_src_pad (GstRtpBin * rtpbin, GstPad * pad, GstRtpBinStream * stream,
|
|
guint8 pt)
|
|
{
|
|
GstElementClass *klass;
|
|
GstPadTemplate *templ;
|
|
gchar *padname;
|
|
GstPad *gpad;
|
|
|
|
gst_object_ref (pad);
|
|
|
|
if (stream->session->storage) {
|
|
/* First try the legacy signal, with no ssrc and pt as parameters.
|
|
* This will likely cause issues for the BUNDLE case. */
|
|
GstElement *fec_decoder =
|
|
session_request_element (stream->session, SIGNAL_REQUEST_FEC_DECODER);
|
|
|
|
/* Now try the new signal, where the application can provide a FEC
|
|
* decoder according to ssrc and pt. */
|
|
if (!fec_decoder) {
|
|
fec_decoder =
|
|
session_request_element_full (stream->session,
|
|
SIGNAL_REQUEST_FEC_DECODER_FULL, stream->ssrc, pt);
|
|
}
|
|
|
|
if (fec_decoder) {
|
|
GstPad *sinkpad, *srcpad;
|
|
GstPadLinkReturn ret;
|
|
|
|
sinkpad = gst_element_get_static_pad (fec_decoder, "sink");
|
|
|
|
if (!sinkpad)
|
|
goto fec_decoder_sink_failed;
|
|
|
|
ret = gst_pad_link (pad, sinkpad);
|
|
gst_object_unref (sinkpad);
|
|
|
|
if (ret != GST_PAD_LINK_OK)
|
|
goto fec_decoder_link_failed;
|
|
|
|
srcpad = gst_element_get_static_pad (fec_decoder, "src");
|
|
|
|
if (!srcpad)
|
|
goto fec_decoder_src_failed;
|
|
|
|
gst_pad_sticky_events_foreach (pad, copy_sticky_events, srcpad);
|
|
gst_object_unref (pad);
|
|
pad = srcpad;
|
|
}
|
|
}
|
|
|
|
GST_RTP_BIN_SHUTDOWN_LOCK (rtpbin, shutdown);
|
|
|
|
/* ghost the pad to the parent */
|
|
klass = GST_ELEMENT_GET_CLASS (rtpbin);
|
|
templ = gst_element_class_get_pad_template (klass, "recv_rtp_src_%u_%u_%u");
|
|
padname = g_strdup_printf ("recv_rtp_src_%u_%u_%u",
|
|
stream->session->id, stream->ssrc, pt);
|
|
gpad = gst_ghost_pad_new_from_template (padname, pad, templ);
|
|
g_free (padname);
|
|
g_object_set_data (G_OBJECT (pad), "GstRTPBin.ghostpad", gpad);
|
|
|
|
gst_pad_set_active (gpad, TRUE);
|
|
GST_RTP_BIN_SHUTDOWN_UNLOCK (rtpbin);
|
|
|
|
gst_pad_sticky_events_foreach (pad, copy_sticky_events, gpad);
|
|
gst_element_add_pad (GST_ELEMENT_CAST (rtpbin), gpad);
|
|
|
|
done:
|
|
gst_object_unref (pad);
|
|
|
|
return;
|
|
|
|
shutdown:
|
|
{
|
|
GST_DEBUG ("ignoring, we are shutting down");
|
|
goto done;
|
|
}
|
|
fec_decoder_sink_failed:
|
|
{
|
|
g_warning ("rtpbin: failed to get fec encoder sink pad for session %u",
|
|
stream->session->id);
|
|
goto done;
|
|
}
|
|
fec_decoder_src_failed:
|
|
{
|
|
g_warning ("rtpbin: failed to get fec encoder src pad for session %u",
|
|
stream->session->id);
|
|
goto done;
|
|
}
|
|
fec_decoder_link_failed:
|
|
{
|
|
g_warning ("rtpbin: failed to link fec decoder for session %u",
|
|
stream->session->id);
|
|
goto done;
|
|
}
|
|
}
|
|
|
|
/* a new pad (SSRC) was created in @session. This signal is emitted from the
|
|
* payload demuxer. */
|
|
static void
|
|
new_payload_found (GstElement * element, guint pt, GstPad * pad,
|
|
GstRtpBinStream * stream)
|
|
{
|
|
GstRtpBin *rtpbin;
|
|
|
|
rtpbin = stream->bin;
|
|
|
|
GST_DEBUG_OBJECT (rtpbin, "new payload pad %u", pt);
|
|
|
|
expose_recv_src_pad (rtpbin, pad, stream, pt);
|
|
}
|
|
|
|
static void
|
|
payload_pad_removed (GstElement * element, GstPad * pad,
|
|
GstRtpBinStream * stream)
|
|
{
|
|
GstRtpBin *rtpbin;
|
|
GstPad *gpad;
|
|
|
|
rtpbin = stream->bin;
|
|
|
|
GST_DEBUG ("payload pad removed");
|
|
|
|
GST_RTP_BIN_DYN_LOCK (rtpbin);
|
|
if ((gpad = g_object_get_data (G_OBJECT (pad), "GstRTPBin.ghostpad"))) {
|
|
g_object_set_data (G_OBJECT (pad), "GstRTPBin.ghostpad", NULL);
|
|
|
|
gst_pad_set_active (gpad, FALSE);
|
|
gst_element_remove_pad (GST_ELEMENT_CAST (rtpbin), gpad);
|
|
}
|
|
GST_RTP_BIN_DYN_UNLOCK (rtpbin);
|
|
}
|
|
|
|
static GstCaps *
|
|
pt_map_requested (GstElement * element, guint pt, GstRtpBinSession * session)
|
|
{
|
|
GstRtpBin *rtpbin;
|
|
GstCaps *caps;
|
|
|
|
rtpbin = session->bin;
|
|
|
|
GST_DEBUG_OBJECT (rtpbin, "payload map requested for pt %u in session %u", pt,
|
|
session->id);
|
|
|
|
caps = get_pt_map (session, pt);
|
|
if (!caps)
|
|
goto no_caps;
|
|
|
|
return caps;
|
|
|
|
/* ERRORS */
|
|
no_caps:
|
|
{
|
|
GST_DEBUG_OBJECT (rtpbin, "could not get caps");
|
|
return NULL;
|
|
}
|
|
}
|
|
|
|
static GstCaps *
|
|
ptdemux_pt_map_requested (GstElement * element, guint pt,
|
|
GstRtpBinSession * session)
|
|
{
|
|
GstCaps *ret = pt_map_requested (element, pt, session);
|
|
|
|
if (ret && gst_caps_get_size (ret) == 1) {
|
|
const GstStructure *s = gst_caps_get_structure (ret, 0);
|
|
gboolean is_fec;
|
|
|
|
if (gst_structure_get_boolean (s, "is-fec", &is_fec) && is_fec) {
|
|
GValue v = G_VALUE_INIT;
|
|
GValue v2 = G_VALUE_INIT;
|
|
|
|
GST_INFO_OBJECT (session->bin, "Will ignore FEC pt %u in session %u", pt,
|
|
session->id);
|
|
g_value_init (&v, GST_TYPE_ARRAY);
|
|
g_value_init (&v2, G_TYPE_INT);
|
|
g_object_get_property (G_OBJECT (element), "ignored-payload-types", &v);
|
|
g_value_set_int (&v2, pt);
|
|
gst_value_array_append_value (&v, &v2);
|
|
g_value_unset (&v2);
|
|
g_object_set_property (G_OBJECT (element), "ignored-payload-types", &v);
|
|
g_value_unset (&v);
|
|
}
|
|
}
|
|
|
|
return ret;
|
|
}
|
|
|
|
static void
|
|
payload_type_change (GstElement * element, guint pt, GstRtpBinSession * session)
|
|
{
|
|
GST_DEBUG_OBJECT (session->bin,
|
|
"emitting signal for pt type changed to %u in session %u", pt,
|
|
session->id);
|
|
|
|
g_signal_emit (session->bin, gst_rtp_bin_signals[SIGNAL_PAYLOAD_TYPE_CHANGE],
|
|
0, session->id, pt);
|
|
}
|
|
|
|
/* emitted when caps changed for the session */
|
|
static void
|
|
caps_changed (GstPad * pad, GParamSpec * pspec, GstRtpBinSession * session)
|
|
{
|
|
GstRtpBin *bin;
|
|
GstCaps *caps;
|
|
gint payload;
|
|
const GstStructure *s;
|
|
|
|
bin = session->bin;
|
|
|
|
g_object_get (pad, "caps", &caps, NULL);
|
|
|
|
if (caps == NULL)
|
|
return;
|
|
|
|
GST_DEBUG_OBJECT (bin, "got caps %" GST_PTR_FORMAT, caps);
|
|
|
|
s = gst_caps_get_structure (caps, 0);
|
|
|
|
/* get payload, finish when it's not there */
|
|
if (!gst_structure_get_int (s, "payload", &payload)) {
|
|
gst_caps_unref (caps);
|
|
return;
|
|
}
|
|
|
|
GST_RTP_SESSION_LOCK (session);
|
|
GST_DEBUG_OBJECT (bin, "insert caps for payload %d", payload);
|
|
g_hash_table_insert (session->ptmap, GINT_TO_POINTER (payload), caps);
|
|
GST_RTP_SESSION_UNLOCK (session);
|
|
}
|
|
|
|
/* a new pad (SSRC) was created in @session */
|
|
static void
|
|
new_ssrc_pad_found (GstElement * element, guint ssrc, GstPad * pad,
|
|
GstRtpBinSession * session)
|
|
{
|
|
GstRtpBin *rtpbin;
|
|
GstRtpBinStream *stream;
|
|
GstPad *sinkpad, *srcpad;
|
|
gchar *padname;
|
|
|
|
rtpbin = session->bin;
|
|
|
|
GST_DEBUG_OBJECT (rtpbin, "new SSRC pad %08x, %s:%s", ssrc,
|
|
GST_DEBUG_PAD_NAME (pad));
|
|
|
|
GST_RTP_BIN_SHUTDOWN_LOCK (rtpbin, shutdown);
|
|
|
|
GST_RTP_SESSION_LOCK (session);
|
|
|
|
/* create new stream */
|
|
stream = create_stream (session, ssrc);
|
|
if (!stream)
|
|
goto no_stream;
|
|
|
|
/* get pad and link */
|
|
GST_DEBUG_OBJECT (rtpbin, "linking jitterbuffer RTP");
|
|
padname = g_strdup_printf ("src_%u", ssrc);
|
|
srcpad = gst_element_get_static_pad (element, padname);
|
|
g_free (padname);
|
|
|
|
if (session->early_fec_decoder) {
|
|
GST_DEBUG_OBJECT (rtpbin, "linking fec decoder");
|
|
sinkpad = gst_element_get_static_pad (session->early_fec_decoder, "sink");
|
|
gst_pad_link_full (srcpad, sinkpad, GST_PAD_LINK_CHECK_NOTHING);
|
|
gst_object_unref (sinkpad);
|
|
gst_object_unref (srcpad);
|
|
srcpad = gst_element_get_static_pad (session->early_fec_decoder, "src");
|
|
}
|
|
|
|
sinkpad = gst_element_get_static_pad (stream->buffer, "sink");
|
|
gst_pad_link_full (srcpad, sinkpad, GST_PAD_LINK_CHECK_NOTHING);
|
|
gst_object_unref (sinkpad);
|
|
gst_object_unref (srcpad);
|
|
|
|
sinkpad = gst_element_request_pad_simple (stream->buffer, "sink_rtcp");
|
|
if (sinkpad) {
|
|
GST_DEBUG_OBJECT (rtpbin, "linking jitterbuffer RTCP");
|
|
padname = g_strdup_printf ("rtcp_src_%u", ssrc);
|
|
srcpad = gst_element_get_static_pad (element, padname);
|
|
g_free (padname);
|
|
gst_pad_link_full (srcpad, sinkpad, GST_PAD_LINK_CHECK_NOTHING);
|
|
gst_object_unref (sinkpad);
|
|
gst_object_unref (srcpad);
|
|
}
|
|
|
|
if (g_signal_lookup ("handle-sync", G_OBJECT_TYPE (stream->buffer)) != 0) {
|
|
/* connect to the RTCP sync signal from the jitterbuffer */
|
|
GST_DEBUG_OBJECT (rtpbin, "connecting sync signal");
|
|
stream->buffer_handlesync_sig = g_signal_connect (stream->buffer,
|
|
"handle-sync", (GCallback) gst_rtp_bin_handle_sync, stream);
|
|
}
|
|
|
|
if (stream->demux) {
|
|
/* connect to the new-pad signal of the payload demuxer, this will expose the
|
|
* new pad by ghosting it. */
|
|
stream->demux_newpad_sig = g_signal_connect (stream->demux,
|
|
"new-payload-type", (GCallback) new_payload_found, stream);
|
|
stream->demux_padremoved_sig = g_signal_connect (stream->demux,
|
|
"pad-removed", (GCallback) payload_pad_removed, stream);
|
|
|
|
/* connect to the request-pt-map signal. This signal will be emitted by the
|
|
* demuxer so that it can apply a proper caps on the buffers for the
|
|
* depayloaders. */
|
|
stream->demux_ptreq_sig = g_signal_connect (stream->demux,
|
|
"request-pt-map", (GCallback) ptdemux_pt_map_requested, session);
|
|
/* connect to the signal so it can be forwarded. */
|
|
stream->demux_ptchange_sig = g_signal_connect (stream->demux,
|
|
"payload-type-change", (GCallback) payload_type_change, session);
|
|
|
|
GST_RTP_SESSION_UNLOCK (session);
|
|
GST_RTP_BIN_SHUTDOWN_UNLOCK (rtpbin);
|
|
} else {
|
|
/* add rtpjitterbuffer src pad to pads */
|
|
GstPad *pad;
|
|
|
|
pad = gst_element_get_static_pad (stream->buffer, "src");
|
|
|
|
GST_RTP_SESSION_UNLOCK (session);
|
|
GST_RTP_BIN_SHUTDOWN_UNLOCK (rtpbin);
|
|
|
|
expose_recv_src_pad (rtpbin, pad, stream, 255);
|
|
|
|
gst_object_unref (pad);
|
|
}
|
|
|
|
return;
|
|
|
|
/* ERRORS */
|
|
shutdown:
|
|
{
|
|
GST_DEBUG_OBJECT (rtpbin, "we are shutting down");
|
|
return;
|
|
}
|
|
no_stream:
|
|
{
|
|
GST_RTP_SESSION_UNLOCK (session);
|
|
GST_RTP_BIN_SHUTDOWN_UNLOCK (rtpbin);
|
|
GST_DEBUG_OBJECT (rtpbin, "could not create stream");
|
|
return;
|
|
}
|
|
}
|
|
|
|
static GstPad *
|
|
complete_session_sink (GstRtpBin * rtpbin, GstRtpBinSession * session)
|
|
{
|
|
guint sessid = session->id;
|
|
GstPad *recv_rtp_sink;
|
|
GstElement *decoder;
|
|
|
|
g_assert (!session->recv_rtp_sink);
|
|
|
|
/* get recv_rtp pad and store */
|
|
session->recv_rtp_sink =
|
|
gst_element_request_pad_simple (session->session, "recv_rtp_sink");
|
|
if (session->recv_rtp_sink == NULL)
|
|
goto pad_failed;
|
|
|
|
g_signal_connect (session->recv_rtp_sink, "notify::caps",
|
|
(GCallback) caps_changed, session);
|
|
|
|
GST_DEBUG_OBJECT (rtpbin, "requesting RTP decoder");
|
|
decoder = session_request_element (session, SIGNAL_REQUEST_RTP_DECODER);
|
|
if (decoder) {
|
|
GstPad *decsrc, *decsink;
|
|
GstPadLinkReturn ret;
|
|
|
|
GST_DEBUG_OBJECT (rtpbin, "linking RTP decoder");
|
|
decsink = gst_element_get_static_pad (decoder, "rtp_sink");
|
|
if (decsink == NULL)
|
|
goto dec_sink_failed;
|
|
|
|
recv_rtp_sink = decsink;
|
|
|
|
decsrc = gst_element_get_static_pad (decoder, "rtp_src");
|
|
if (decsrc == NULL)
|
|
goto dec_src_failed;
|
|
|
|
ret = gst_pad_link (decsrc, session->recv_rtp_sink);
|
|
|
|
gst_object_unref (decsrc);
|
|
|
|
if (ret != GST_PAD_LINK_OK)
|
|
goto dec_link_failed;
|
|
|
|
} else {
|
|
GST_DEBUG_OBJECT (rtpbin, "no RTP decoder given");
|
|
recv_rtp_sink = gst_object_ref (session->recv_rtp_sink);
|
|
}
|
|
|
|
return recv_rtp_sink;
|
|
|
|
/* ERRORS */
|
|
pad_failed:
|
|
{
|
|
g_warning ("rtpbin: failed to get session recv_rtp_sink pad");
|
|
return NULL;
|
|
}
|
|
dec_sink_failed:
|
|
{
|
|
g_warning ("rtpbin: failed to get decoder sink pad for session %u", sessid);
|
|
return NULL;
|
|
}
|
|
dec_src_failed:
|
|
{
|
|
g_warning ("rtpbin: failed to get decoder src pad for session %u", sessid);
|
|
gst_object_unref (recv_rtp_sink);
|
|
return NULL;
|
|
}
|
|
dec_link_failed:
|
|
{
|
|
g_warning ("rtpbin: failed to link rtp decoder for session %u", sessid);
|
|
gst_object_unref (recv_rtp_sink);
|
|
return NULL;
|
|
}
|
|
}
|
|
|
|
static void
|
|
complete_session_receiver (GstRtpBin * rtpbin, GstRtpBinSession * session,
|
|
guint sessid)
|
|
{
|
|
GstElement *aux;
|
|
GstPad *recv_rtp_src;
|
|
|
|
g_assert (!session->recv_rtp_src);
|
|
|
|
session->recv_rtp_src =
|
|
gst_element_get_static_pad (session->session, "recv_rtp_src");
|
|
if (session->recv_rtp_src == NULL)
|
|
goto pad_failed;
|
|
|
|
/* find out if we need AUX elements */
|
|
aux = session_request_element (session, SIGNAL_REQUEST_AUX_RECEIVER);
|
|
if (aux) {
|
|
gchar *pname;
|
|
GstPad *auxsink;
|
|
GstPadLinkReturn ret;
|
|
|
|
GST_DEBUG_OBJECT (rtpbin, "linking AUX receiver");
|
|
|
|
pname = g_strdup_printf ("sink_%u", sessid);
|
|
auxsink = gst_element_get_static_pad (aux, pname);
|
|
g_free (pname);
|
|
if (auxsink == NULL)
|
|
goto aux_sink_failed;
|
|
|
|
ret = gst_pad_link (session->recv_rtp_src, auxsink);
|
|
gst_object_unref (auxsink);
|
|
if (ret != GST_PAD_LINK_OK)
|
|
goto aux_link_failed;
|
|
|
|
/* this can be NULL when this AUX element is not to be linked any further */
|
|
pname = g_strdup_printf ("src_%u", sessid);
|
|
recv_rtp_src = gst_element_get_static_pad (aux, pname);
|
|
g_free (pname);
|
|
} else {
|
|
recv_rtp_src = gst_object_ref (session->recv_rtp_src);
|
|
}
|
|
|
|
/* Add a storage element if needed */
|
|
if (recv_rtp_src && session->storage) {
|
|
GstPadLinkReturn ret;
|
|
GstPad *sinkpad = gst_element_get_static_pad (session->storage, "sink");
|
|
|
|
ret = gst_pad_link (recv_rtp_src, sinkpad);
|
|
|
|
gst_object_unref (sinkpad);
|
|
gst_object_unref (recv_rtp_src);
|
|
|
|
if (ret != GST_PAD_LINK_OK)
|
|
goto storage_link_failed;
|
|
|
|
recv_rtp_src = gst_element_get_static_pad (session->storage, "src");
|
|
}
|
|
|
|
if (recv_rtp_src) {
|
|
GstPad *sinkdpad;
|
|
|
|
GST_DEBUG_OBJECT (rtpbin, "getting demuxer RTP sink pad");
|
|
sinkdpad = gst_element_get_static_pad (session->demux, "sink");
|
|
GST_DEBUG_OBJECT (rtpbin, "linking demuxer RTP sink pad");
|
|
gst_pad_link_full (recv_rtp_src, sinkdpad, GST_PAD_LINK_CHECK_NOTHING);
|
|
gst_object_unref (sinkdpad);
|
|
gst_object_unref (recv_rtp_src);
|
|
|
|
/* connect to the new-ssrc-pad signal of the SSRC demuxer */
|
|
session->demux_newpad_sig = g_signal_connect (session->demux,
|
|
"new-ssrc-pad", (GCallback) new_ssrc_pad_found, session);
|
|
session->demux_padremoved_sig = g_signal_connect (session->demux,
|
|
"removed-ssrc-pad", (GCallback) ssrc_demux_pad_removed, session);
|
|
}
|
|
|
|
return;
|
|
|
|
pad_failed:
|
|
{
|
|
g_warning ("rtpbin: failed to get session recv_rtp_src pad");
|
|
return;
|
|
}
|
|
aux_sink_failed:
|
|
{
|
|
g_warning ("rtpbin: failed to get AUX sink pad for session %u", sessid);
|
|
return;
|
|
}
|
|
aux_link_failed:
|
|
{
|
|
g_warning ("rtpbin: failed to link AUX pad to session %u", sessid);
|
|
return;
|
|
}
|
|
storage_link_failed:
|
|
{
|
|
g_warning ("rtpbin: failed to link storage");
|
|
return;
|
|
}
|
|
}
|
|
|
|
/* Create a pad for receiving RTP for the session in @name. Must be called with
|
|
* RTP_BIN_LOCK.
|
|
*/
|
|
static GstPad *
|
|
create_recv_rtp (GstRtpBin * rtpbin, GstPadTemplate * templ, const gchar * name)
|
|
{
|
|
guint sessid;
|
|
GstRtpBinSession *session;
|
|
GstPad *recv_rtp_sink;
|
|
|
|
/* first get the session number */
|
|
if (name == NULL || sscanf (name, "recv_rtp_sink_%u", &sessid) != 1)
|
|
goto no_name;
|
|
|
|
GST_DEBUG_OBJECT (rtpbin, "finding session %u", sessid);
|
|
|
|
/* get or create session */
|
|
session = find_session_by_id (rtpbin, sessid);
|
|
if (!session) {
|
|
GST_DEBUG_OBJECT (rtpbin, "creating session %u", sessid);
|
|
/* create session now */
|
|
session = create_session (rtpbin, sessid);
|
|
if (session == NULL)
|
|
goto create_error;
|
|
}
|
|
|
|
/* check if pad was requested */
|
|
if (session->recv_rtp_sink_ghost != NULL)
|
|
return session->recv_rtp_sink_ghost;
|
|
|
|
/* setup the session sink pad */
|
|
recv_rtp_sink = complete_session_sink (rtpbin, session);
|
|
if (!recv_rtp_sink)
|
|
goto session_sink_failed;
|
|
|
|
GST_DEBUG_OBJECT (rtpbin, "ghosting session sink pad");
|
|
session->recv_rtp_sink_ghost =
|
|
gst_ghost_pad_new_from_template (name, recv_rtp_sink, templ);
|
|
gst_object_unref (recv_rtp_sink);
|
|
|
|
complete_session_receiver (rtpbin, session, sessid);
|
|
|
|
return session->recv_rtp_sink_ghost;
|
|
|
|
/* ERRORS */
|
|
no_name:
|
|
{
|
|
g_warning ("rtpbin: cannot find session id for pad: %s",
|
|
GST_STR_NULL (name));
|
|
return NULL;
|
|
}
|
|
create_error:
|
|
{
|
|
/* create_session already warned */
|
|
return NULL;
|
|
}
|
|
session_sink_failed:
|
|
{
|
|
/* warning already done */
|
|
return NULL;
|
|
}
|
|
}
|
|
|
|
static void
|
|
remove_recv_rtp (GstRtpBin * rtpbin, GstRtpBinSession * session)
|
|
{
|
|
if (session->demux_newpad_sig) {
|
|
g_signal_handler_disconnect (session->demux, session->demux_newpad_sig);
|
|
session->demux_newpad_sig = 0;
|
|
}
|
|
if (session->demux_padremoved_sig) {
|
|
g_signal_handler_disconnect (session->demux, session->demux_padremoved_sig);
|
|
session->demux_padremoved_sig = 0;
|
|
}
|
|
if (session->recv_rtp_src) {
|
|
gst_object_unref (session->recv_rtp_src);
|
|
session->recv_rtp_src = NULL;
|
|
}
|
|
if (session->recv_rtp_sink) {
|
|
gst_element_release_request_pad (session->session, session->recv_rtp_sink);
|
|
gst_object_unref (session->recv_rtp_sink);
|
|
session->recv_rtp_sink = NULL;
|
|
}
|
|
if (session->recv_rtp_sink_ghost) {
|
|
gst_pad_set_active (session->recv_rtp_sink_ghost, FALSE);
|
|
gst_element_remove_pad (GST_ELEMENT_CAST (rtpbin),
|
|
session->recv_rtp_sink_ghost);
|
|
session->recv_rtp_sink_ghost = NULL;
|
|
}
|
|
}
|
|
|
|
static gint
|
|
fec_sinkpad_find (const GValue * item, gchar * padname)
|
|
{
|
|
GstPad *pad = g_value_get_object (item);
|
|
return g_strcmp0 (GST_PAD_NAME (pad), padname);
|
|
}
|
|
|
|
static GstPad *
|
|
complete_session_fec (GstRtpBin * rtpbin, GstRtpBinSession * session,
|
|
guint fec_idx)
|
|
{
|
|
gboolean have_static_pad;
|
|
gchar *padname;
|
|
|
|
GstPad *ret;
|
|
GstIterator *it;
|
|
GValue item = { 0, };
|
|
|
|
if (!ensure_early_fec_decoder (rtpbin, session))
|
|
goto no_decoder;
|
|
|
|
padname = g_strdup_printf ("fec_%u", fec_idx);
|
|
|
|
GST_DEBUG_OBJECT (rtpbin, "getting FEC sink pad %s", padname);
|
|
|
|
/* First try to find the decoder static pad that matches the padname */
|
|
it = gst_element_iterate_sink_pads (session->early_fec_decoder);
|
|
have_static_pad =
|
|
gst_iterator_find_custom (it, (GCompareFunc) fec_sinkpad_find, &item,
|
|
padname);
|
|
|
|
if (have_static_pad) {
|
|
ret = g_value_get_object (&item);
|
|
gst_object_ref (ret);
|
|
g_value_unset (&item);
|
|
} else {
|
|
ret = gst_element_request_pad_simple (session->early_fec_decoder, padname);
|
|
}
|
|
|
|
g_free (padname);
|
|
gst_iterator_free (it);
|
|
|
|
if (ret == NULL)
|
|
goto pad_failed;
|
|
|
|
session->recv_fec_sinks = g_slist_prepend (session->recv_fec_sinks, ret);
|
|
|
|
return ret;
|
|
|
|
pad_failed:
|
|
{
|
|
g_warning ("rtpbin: failed to get decoder fec pad");
|
|
return NULL;
|
|
}
|
|
no_decoder:
|
|
{
|
|
g_warning ("rtpbin: failed to build FEC decoder for session %u",
|
|
session->id);
|
|
return NULL;
|
|
}
|
|
}
|
|
|
|
static GstPad *
|
|
complete_session_rtcp (GstRtpBin * rtpbin, GstRtpBinSession * session,
|
|
guint sessid)
|
|
{
|
|
GstElement *decoder;
|
|
GstPad *sinkdpad;
|
|
GstPad *decsink = NULL;
|
|
|
|
/* get recv_rtp pad and store */
|
|
GST_DEBUG_OBJECT (rtpbin, "getting RTCP sink pad");
|
|
session->recv_rtcp_sink =
|
|
gst_element_request_pad_simple (session->session, "recv_rtcp_sink");
|
|
if (session->recv_rtcp_sink == NULL)
|
|
goto pad_failed;
|
|
|
|
GST_DEBUG_OBJECT (rtpbin, "getting RTCP decoder");
|
|
decoder = session_request_element (session, SIGNAL_REQUEST_RTCP_DECODER);
|
|
if (decoder) {
|
|
GstPad *decsrc;
|
|
GstPadLinkReturn ret;
|
|
|
|
GST_DEBUG_OBJECT (rtpbin, "linking RTCP decoder");
|
|
decsink = gst_element_get_static_pad (decoder, "rtcp_sink");
|
|
decsrc = gst_element_get_static_pad (decoder, "rtcp_src");
|
|
|
|
if (decsink == NULL)
|
|
goto dec_sink_failed;
|
|
|
|
if (decsrc == NULL)
|
|
goto dec_src_failed;
|
|
|
|
ret = gst_pad_link (decsrc, session->recv_rtcp_sink);
|
|
|
|
gst_object_unref (decsrc);
|
|
|
|
if (ret != GST_PAD_LINK_OK)
|
|
goto dec_link_failed;
|
|
} else {
|
|
GST_DEBUG_OBJECT (rtpbin, "no RTCP decoder given");
|
|
decsink = gst_object_ref (session->recv_rtcp_sink);
|
|
}
|
|
|
|
/* get srcpad, link to SSRCDemux */
|
|
GST_DEBUG_OBJECT (rtpbin, "getting sync src pad");
|
|
session->sync_src = gst_element_get_static_pad (session->session, "sync_src");
|
|
if (session->sync_src == NULL)
|
|
goto src_pad_failed;
|
|
|
|
GST_DEBUG_OBJECT (rtpbin, "getting demuxer RTCP sink pad");
|
|
sinkdpad = gst_element_get_static_pad (session->demux, "rtcp_sink");
|
|
gst_pad_link_full (session->sync_src, sinkdpad, GST_PAD_LINK_CHECK_NOTHING);
|
|
gst_object_unref (sinkdpad);
|
|
|
|
return decsink;
|
|
|
|
pad_failed:
|
|
{
|
|
g_warning ("rtpbin: failed to get session rtcp_sink pad");
|
|
return NULL;
|
|
}
|
|
dec_sink_failed:
|
|
{
|
|
g_warning ("rtpbin: failed to get decoder sink pad for session %u", sessid);
|
|
return NULL;
|
|
}
|
|
dec_src_failed:
|
|
{
|
|
g_warning ("rtpbin: failed to get decoder src pad for session %u", sessid);
|
|
goto cleanup;
|
|
}
|
|
dec_link_failed:
|
|
{
|
|
g_warning ("rtpbin: failed to link rtcp decoder for session %u", sessid);
|
|
goto cleanup;
|
|
}
|
|
src_pad_failed:
|
|
{
|
|
g_warning ("rtpbin: failed to get session sync_src pad");
|
|
}
|
|
|
|
cleanup:
|
|
gst_object_unref (decsink);
|
|
return NULL;
|
|
}
|
|
|
|
/* Create a pad for receiving RTCP for the session in @name. Must be called with
|
|
* RTP_BIN_LOCK.
|
|
*/
|
|
static GstPad *
|
|
create_recv_rtcp (GstRtpBin * rtpbin, GstPadTemplate * templ,
|
|
const gchar * name)
|
|
{
|
|
guint sessid;
|
|
GstRtpBinSession *session;
|
|
GstPad *decsink = NULL;
|
|
|
|
/* first get the session number */
|
|
if (name == NULL || sscanf (name, "recv_rtcp_sink_%u", &sessid) != 1)
|
|
goto no_name;
|
|
|
|
GST_DEBUG_OBJECT (rtpbin, "finding session %u", sessid);
|
|
|
|
/* get or create the session */
|
|
session = find_session_by_id (rtpbin, sessid);
|
|
if (!session) {
|
|
GST_DEBUG_OBJECT (rtpbin, "creating session %u", sessid);
|
|
/* create session now */
|
|
session = create_session (rtpbin, sessid);
|
|
if (session == NULL)
|
|
goto create_error;
|
|
}
|
|
|
|
/* check if pad was requested */
|
|
if (session->recv_rtcp_sink_ghost != NULL)
|
|
return session->recv_rtcp_sink_ghost;
|
|
|
|
decsink = complete_session_rtcp (rtpbin, session, sessid);
|
|
if (!decsink)
|
|
goto create_error;
|
|
|
|
session->recv_rtcp_sink_ghost =
|
|
gst_ghost_pad_new_from_template (name, decsink, templ);
|
|
gst_object_unref (decsink);
|
|
|
|
return session->recv_rtcp_sink_ghost;
|
|
|
|
/* ERRORS */
|
|
no_name:
|
|
{
|
|
g_warning ("rtpbin: cannot find session id for pad: %s",
|
|
GST_STR_NULL (name));
|
|
return NULL;
|
|
}
|
|
create_error:
|
|
{
|
|
/* create_session already warned */
|
|
return NULL;
|
|
}
|
|
}
|
|
|
|
static GstPad *
|
|
create_recv_fec (GstRtpBin * rtpbin, GstPadTemplate * templ, const gchar * name)
|
|
{
|
|
guint sessid, fec_idx;
|
|
GstRtpBinSession *session;
|
|
GstPad *decsink = NULL;
|
|
GstPad *ghost;
|
|
|
|
/* first get the session number */
|
|
if (name == NULL
|
|
|| sscanf (name, "recv_fec_sink_%u_%u", &sessid, &fec_idx) != 2)
|
|
goto no_name;
|
|
|
|
if (fec_idx > 1)
|
|
goto invalid_idx;
|
|
|
|
GST_DEBUG_OBJECT (rtpbin, "finding session %u", sessid);
|
|
|
|
/* get or create the session */
|
|
session = find_session_by_id (rtpbin, sessid);
|
|
if (!session) {
|
|
GST_DEBUG_OBJECT (rtpbin, "creating session %u", sessid);
|
|
/* create session now */
|
|
session = create_session (rtpbin, sessid);
|
|
if (session == NULL)
|
|
goto create_error;
|
|
}
|
|
|
|
decsink = complete_session_fec (rtpbin, session, fec_idx);
|
|
if (!decsink)
|
|
goto create_error;
|
|
|
|
ghost = gst_ghost_pad_new_from_template (name, decsink, templ);
|
|
session->recv_fec_sink_ghosts =
|
|
g_slist_prepend (session->recv_fec_sink_ghosts, ghost);
|
|
gst_object_unref (decsink);
|
|
|
|
return ghost;
|
|
|
|
/* ERRORS */
|
|
no_name:
|
|
{
|
|
g_warning ("rtpbin: cannot find session id for pad: %s",
|
|
GST_STR_NULL (name));
|
|
return NULL;
|
|
}
|
|
invalid_idx:
|
|
{
|
|
g_warning ("rtpbin: invalid FEC index: %s", GST_STR_NULL (name));
|
|
return NULL;
|
|
}
|
|
create_error:
|
|
{
|
|
/* create_session already warned */
|
|
return NULL;
|
|
}
|
|
}
|
|
|
|
static void
|
|
remove_recv_rtcp (GstRtpBin * rtpbin, GstRtpBinSession * session)
|
|
{
|
|
if (session->recv_rtcp_sink_ghost) {
|
|
gst_pad_set_active (session->recv_rtcp_sink_ghost, FALSE);
|
|
gst_element_remove_pad (GST_ELEMENT_CAST (rtpbin),
|
|
session->recv_rtcp_sink_ghost);
|
|
session->recv_rtcp_sink_ghost = NULL;
|
|
}
|
|
if (session->sync_src) {
|
|
/* releasing the request pad should also unref the sync pad */
|
|
gst_object_unref (session->sync_src);
|
|
session->sync_src = NULL;
|
|
}
|
|
if (session->recv_rtcp_sink) {
|
|
gst_element_release_request_pad (session->session, session->recv_rtcp_sink);
|
|
gst_object_unref (session->recv_rtcp_sink);
|
|
session->recv_rtcp_sink = NULL;
|
|
}
|
|
}
|
|
|
|
static void
|
|
remove_recv_fec_for_pad (GstRtpBin * rtpbin, GstRtpBinSession * session,
|
|
GstPad * ghost)
|
|
{
|
|
GSList *item;
|
|
GstPad *target;
|
|
|
|
target = gst_ghost_pad_get_target (GST_GHOST_PAD (ghost));
|
|
|
|
if (target) {
|
|
item = g_slist_find (session->recv_fec_sinks, target);
|
|
if (item) {
|
|
GstPadTemplate *templ;
|
|
GstPad *pad;
|
|
|
|
pad = item->data;
|
|
templ = gst_pad_get_pad_template (pad);
|
|
|
|
if (GST_PAD_TEMPLATE_PRESENCE (templ) == GST_PAD_REQUEST) {
|
|
GST_DEBUG_OBJECT (rtpbin,
|
|
"Releasing FEC decoder pad %" GST_PTR_FORMAT, pad);
|
|
gst_element_release_request_pad (session->early_fec_decoder, pad);
|
|
} else {
|
|
gst_object_unref (pad);
|
|
}
|
|
|
|
session->recv_fec_sinks =
|
|
g_slist_delete_link (session->recv_fec_sinks, item);
|
|
|
|
gst_object_unref (templ);
|
|
}
|
|
gst_object_unref (target);
|
|
}
|
|
|
|
item = g_slist_find (session->recv_fec_sink_ghosts, ghost);
|
|
if (item)
|
|
session->recv_fec_sink_ghosts =
|
|
g_slist_delete_link (session->recv_fec_sink_ghosts, item);
|
|
|
|
gst_pad_set_active (ghost, FALSE);
|
|
gst_element_remove_pad (GST_ELEMENT_CAST (rtpbin), ghost);
|
|
}
|
|
|
|
static void
|
|
remove_recv_fec (GstRtpBin * rtpbin, GstRtpBinSession * session)
|
|
{
|
|
GSList *copy;
|
|
GSList *tmp;
|
|
|
|
copy = g_slist_copy (session->recv_fec_sink_ghosts);
|
|
|
|
for (tmp = copy; tmp; tmp = tmp->next) {
|
|
remove_recv_fec_for_pad (rtpbin, session, (GstPad *) tmp->data);
|
|
}
|
|
|
|
g_slist_free (copy);
|
|
}
|
|
|
|
static gboolean
|
|
complete_session_src (GstRtpBin * rtpbin, GstRtpBinSession * session)
|
|
{
|
|
gchar *gname;
|
|
guint sessid = session->id;
|
|
GstPad *send_rtp_src;
|
|
GstElement *encoder;
|
|
GstElementClass *klass;
|
|
GstPadTemplate *templ;
|
|
gboolean ret = FALSE;
|
|
|
|
/* get srcpad */
|
|
send_rtp_src = gst_element_get_static_pad (session->session, "send_rtp_src");
|
|
|
|
if (send_rtp_src == NULL)
|
|
goto no_srcpad;
|
|
|
|
GST_DEBUG_OBJECT (rtpbin, "getting RTP encoder");
|
|
encoder = session_request_element (session, SIGNAL_REQUEST_RTP_ENCODER);
|
|
if (encoder) {
|
|
gchar *ename;
|
|
GstPad *encsrc, *encsink;
|
|
GstPadLinkReturn ret;
|
|
|
|
GST_DEBUG_OBJECT (rtpbin, "linking RTP encoder");
|
|
ename = g_strdup_printf ("rtp_src_%u", sessid);
|
|
encsrc = gst_element_get_static_pad (encoder, ename);
|
|
g_free (ename);
|
|
|
|
if (encsrc == NULL)
|
|
goto enc_src_failed;
|
|
|
|
ename = g_strdup_printf ("rtp_sink_%u", sessid);
|
|
encsink = gst_element_get_static_pad (encoder, ename);
|
|
g_free (ename);
|
|
if (encsink == NULL)
|
|
goto enc_sink_failed;
|
|
|
|
ret = gst_pad_link (send_rtp_src, encsink);
|
|
gst_object_unref (encsink);
|
|
gst_object_unref (send_rtp_src);
|
|
|
|
send_rtp_src = encsrc;
|
|
|
|
if (ret != GST_PAD_LINK_OK)
|
|
goto enc_link_failed;
|
|
} else {
|
|
GST_DEBUG_OBJECT (rtpbin, "no RTP encoder given");
|
|
}
|
|
|
|
/* ghost the new source pad */
|
|
klass = GST_ELEMENT_GET_CLASS (rtpbin);
|
|
gname = g_strdup_printf ("send_rtp_src_%u", sessid);
|
|
templ = gst_element_class_get_pad_template (klass, "send_rtp_src_%u");
|
|
session->send_rtp_src_ghost =
|
|
gst_ghost_pad_new_from_template (gname, send_rtp_src, templ);
|
|
gst_pad_set_active (session->send_rtp_src_ghost, TRUE);
|
|
gst_pad_sticky_events_foreach (send_rtp_src, copy_sticky_events,
|
|
session->send_rtp_src_ghost);
|
|
gst_element_add_pad (GST_ELEMENT_CAST (rtpbin), session->send_rtp_src_ghost);
|
|
g_free (gname);
|
|
|
|
ret = TRUE;
|
|
|
|
done:
|
|
if (send_rtp_src)
|
|
gst_object_unref (send_rtp_src);
|
|
|
|
return ret;
|
|
|
|
/* ERRORS */
|
|
no_srcpad:
|
|
{
|
|
g_warning ("rtpbin: failed to get rtp source pad for session %u", sessid);
|
|
goto done;
|
|
}
|
|
enc_src_failed:
|
|
{
|
|
g_warning ("rtpbin: failed to get %" GST_PTR_FORMAT
|
|
" src pad for session %u", encoder, sessid);
|
|
goto done;
|
|
}
|
|
enc_sink_failed:
|
|
{
|
|
g_warning ("rtpbin: failed to get %" GST_PTR_FORMAT
|
|
" sink pad for session %u", encoder, sessid);
|
|
goto done;
|
|
}
|
|
enc_link_failed:
|
|
{
|
|
g_warning ("rtpbin: failed to link %" GST_PTR_FORMAT " for session %u",
|
|
encoder, sessid);
|
|
goto done;
|
|
}
|
|
}
|
|
|
|
static gboolean
|
|
setup_aux_sender_fold (const GValue * item, GValue * result, gpointer user_data)
|
|
{
|
|
GstPad *pad;
|
|
gchar *name;
|
|
guint sessid;
|
|
GstRtpBinSession *session = user_data, *newsess;
|
|
GstRtpBin *rtpbin = session->bin;
|
|
GstPadLinkReturn ret;
|
|
|
|
pad = g_value_get_object (item);
|
|
name = gst_pad_get_name (pad);
|
|
|
|
if (name == NULL || sscanf (name, "src_%u", &sessid) != 1)
|
|
goto no_name;
|
|
|
|
g_free (name);
|
|
|
|
newsess = find_session_by_id (rtpbin, sessid);
|
|
if (newsess == NULL) {
|
|
/* create new session */
|
|
newsess = create_session (rtpbin, sessid);
|
|
if (newsess == NULL)
|
|
goto create_error;
|
|
} else if (newsess->send_rtp_sink != NULL)
|
|
goto existing_session;
|
|
|
|
/* get send_rtp pad and store */
|
|
newsess->send_rtp_sink =
|
|
gst_element_request_pad_simple (newsess->session, "send_rtp_sink");
|
|
if (newsess->send_rtp_sink == NULL)
|
|
goto pad_failed;
|
|
|
|
ret = gst_pad_link (pad, newsess->send_rtp_sink);
|
|
if (ret != GST_PAD_LINK_OK)
|
|
goto aux_link_failed;
|
|
|
|
if (!complete_session_src (rtpbin, newsess))
|
|
goto session_src_failed;
|
|
|
|
return TRUE;
|
|
|
|
/* ERRORS */
|
|
no_name:
|
|
{
|
|
GST_WARNING ("ignoring invalid pad name %s", GST_STR_NULL (name));
|
|
g_free (name);
|
|
return TRUE;
|
|
}
|
|
create_error:
|
|
{
|
|
/* create_session already warned */
|
|
return FALSE;
|
|
}
|
|
existing_session:
|
|
{
|
|
GST_DEBUG_OBJECT (rtpbin,
|
|
"skipping src_%i setup, since it is already configured.", sessid);
|
|
return TRUE;
|
|
}
|
|
pad_failed:
|
|
{
|
|
g_warning ("rtpbin: failed to get session pad for session %u", sessid);
|
|
return FALSE;
|
|
}
|
|
aux_link_failed:
|
|
{
|
|
g_warning ("rtpbin: failed to link AUX for session %u", sessid);
|
|
return FALSE;
|
|
}
|
|
session_src_failed:
|
|
{
|
|
g_warning ("rtpbin: failed to complete AUX for session %u", sessid);
|
|
return FALSE;
|
|
}
|
|
}
|
|
|
|
static gboolean
|
|
setup_aux_sender (GstRtpBin * rtpbin, GstRtpBinSession * session,
|
|
GstElement * aux)
|
|
{
|
|
GstIterator *it;
|
|
GValue result = { 0, };
|
|
GstIteratorResult res;
|
|
|
|
it = gst_element_iterate_src_pads (aux);
|
|
res = gst_iterator_fold (it, setup_aux_sender_fold, &result, session);
|
|
gst_iterator_free (it);
|
|
|
|
return res == GST_ITERATOR_DONE;
|
|
}
|
|
|
|
static void
|
|
fec_encoder_add_pad_unlocked (GstPad * pad, GstRtpBinSession * session)
|
|
{
|
|
GstElementClass *klass;
|
|
gchar *gname;
|
|
GstPadTemplate *templ;
|
|
guint fec_idx;
|
|
GstPad *ghost;
|
|
|
|
if (sscanf (GST_PAD_NAME (pad), "fec_%u", &fec_idx) != 1) {
|
|
GST_WARNING_OBJECT (session->bin,
|
|
"FEC encoder added pad with name not matching fec_%%u (%s)",
|
|
GST_PAD_NAME (pad));
|
|
goto done;
|
|
}
|
|
|
|
GST_INFO_OBJECT (session->bin, "FEC encoder for session %u exposed new pad",
|
|
session->id);
|
|
|
|
klass = GST_ELEMENT_GET_CLASS (session->bin);
|
|
gname = g_strdup_printf ("send_fec_src_%u_%u", session->id, fec_idx);
|
|
templ = gst_element_class_get_pad_template (klass, "send_fec_src_%u_%u");
|
|
ghost = gst_ghost_pad_new_from_template (gname, pad, templ);
|
|
session->send_fec_src_ghosts =
|
|
g_slist_prepend (session->send_fec_src_ghosts, ghost);
|
|
gst_pad_set_active (ghost, TRUE);
|
|
gst_pad_sticky_events_foreach (pad, copy_sticky_events, ghost);
|
|
gst_element_add_pad (GST_ELEMENT (session->bin), ghost);
|
|
g_free (gname);
|
|
|
|
done:
|
|
return;
|
|
}
|
|
|
|
static void
|
|
fec_encoder_add_pad (GstPad * pad, GstRtpBinSession * session)
|
|
{
|
|
GST_RTP_BIN_LOCK (session->bin);
|
|
fec_encoder_add_pad_unlocked (pad, session);
|
|
GST_RTP_BIN_UNLOCK (session->bin);
|
|
}
|
|
|
|
static gint
|
|
fec_srcpad_iterator_filter (const GValue * item, GValue * unused)
|
|
{
|
|
guint fec_idx;
|
|
GstPad *pad = g_value_get_object (item);
|
|
GstPadTemplate *templ = gst_pad_get_pad_template (pad);
|
|
|
|
gint have_static_pad =
|
|
(GST_PAD_TEMPLATE_PRESENCE (templ) == GST_PAD_ALWAYS) &&
|
|
(sscanf (GST_PAD_NAME (pad), "fec_%u", &fec_idx) == 1);
|
|
|
|
gst_object_unref (templ);
|
|
|
|
/* return 0 to retain pad in filtered iterator */
|
|
return !have_static_pad;
|
|
}
|
|
|
|
static void
|
|
fec_srcpad_iterator_foreach (const GValue * item, GstRtpBinSession * session)
|
|
{
|
|
GstPad *pad = g_value_get_object (item);
|
|
fec_encoder_add_pad_unlocked (pad, session);
|
|
}
|
|
|
|
static void
|
|
fec_encoder_pad_added_cb (GstElement * encoder, GstPad * pad,
|
|
GstRtpBinSession * session)
|
|
{
|
|
fec_encoder_add_pad (pad, session);
|
|
}
|
|
|
|
static GstElement *
|
|
request_fec_encoder (GstRtpBin * rtpbin, GstRtpBinSession * session,
|
|
guint sessid)
|
|
{
|
|
GstElement *ret = NULL;
|
|
const gchar *factory;
|
|
gchar *sess_id_str;
|
|
|
|
sess_id_str = g_strdup_printf ("%u", sessid);
|
|
factory = gst_structure_get_string (rtpbin->fec_encoders, sess_id_str);
|
|
g_free (sess_id_str);
|
|
|
|
/* First try the property */
|
|
if (factory) {
|
|
GError *err = NULL;
|
|
|
|
ret =
|
|
gst_parse_bin_from_description_full (factory, TRUE, NULL,
|
|
GST_PARSE_FLAG_NO_SINGLE_ELEMENT_BINS | GST_PARSE_FLAG_FATAL_ERRORS,
|
|
&err);
|
|
if (!ret) {
|
|
GST_ERROR_OBJECT (rtpbin, "Failed to build encoder from factory: %s",
|
|
err->message);
|
|
goto done;
|
|
}
|
|
|
|
bin_manage_element (session->bin, ret);
|
|
session->elements = g_slist_prepend (session->elements, ret);
|
|
GST_INFO_OBJECT (rtpbin, "Built FEC encoder: %" GST_PTR_FORMAT
|
|
" for session %u", ret, sessid);
|
|
}
|
|
|
|
/* Fallback to the signal */
|
|
if (!ret)
|
|
ret = session_request_element (session, SIGNAL_REQUEST_FEC_ENCODER);
|
|
|
|
if (ret) {
|
|
/* First, add encoder pads that match fec_% template and are already present */
|
|
GstIterator *it, *filter;
|
|
GstIteratorResult it_ret = GST_ITERATOR_OK;
|
|
|
|
it = gst_element_iterate_src_pads (ret);
|
|
filter =
|
|
gst_iterator_filter (it, (GCompareFunc) fec_srcpad_iterator_filter,
|
|
NULL);
|
|
|
|
while (it_ret == GST_ITERATOR_OK || it_ret == GST_ITERATOR_RESYNC) {
|
|
it_ret =
|
|
gst_iterator_foreach (filter,
|
|
(GstIteratorForeachFunction) fec_srcpad_iterator_foreach, session);
|
|
|
|
if (it_ret == GST_ITERATOR_RESYNC)
|
|
gst_iterator_resync (filter);
|
|
}
|
|
|
|
gst_iterator_free (filter);
|
|
|
|
/* Finally, connect to pad-added signal if any of the encoder pads are
|
|
* added later */
|
|
g_signal_connect (ret, "pad-added", G_CALLBACK (fec_encoder_pad_added_cb),
|
|
session);
|
|
}
|
|
|
|
done:
|
|
return ret;
|
|
}
|
|
|
|
/* Create a pad for sending RTP for the session in @name. Must be called with
|
|
* RTP_BIN_LOCK.
|
|
*/
|
|
static GstPad *
|
|
create_send_rtp (GstRtpBin * rtpbin, GstPadTemplate * templ, const gchar * name)
|
|
{
|
|
gchar *pname;
|
|
guint sessid;
|
|
GstPad *send_rtp_sink;
|
|
GstElement *aux;
|
|
GstElement *encoder;
|
|
GstElement *prev = NULL;
|
|
GstRtpBinSession *session;
|
|
|
|
/* first get the session number */
|
|
if (name == NULL || sscanf (name, "send_rtp_sink_%u", &sessid) != 1)
|
|
goto no_name;
|
|
|
|
/* get or create session */
|
|
session = find_session_by_id (rtpbin, sessid);
|
|
if (!session) {
|
|
/* create session now */
|
|
session = create_session (rtpbin, sessid);
|
|
if (session == NULL)
|
|
goto create_error;
|
|
}
|
|
|
|
/* check if pad was requested */
|
|
if (session->send_rtp_sink_ghost != NULL)
|
|
return session->send_rtp_sink_ghost;
|
|
|
|
/* check if we are already using this session as a sender */
|
|
if (session->send_rtp_sink != NULL)
|
|
goto existing_session;
|
|
|
|
encoder = request_fec_encoder (rtpbin, session, sessid);
|
|
|
|
if (encoder) {
|
|
GST_DEBUG_OBJECT (rtpbin, "Linking FEC encoder");
|
|
|
|
send_rtp_sink = gst_element_get_static_pad (encoder, "sink");
|
|
|
|
if (!send_rtp_sink)
|
|
goto enc_sink_failed;
|
|
|
|
prev = encoder;
|
|
}
|
|
|
|
GST_DEBUG_OBJECT (rtpbin, "getting RTP AUX sender");
|
|
aux = session_request_element (session, SIGNAL_REQUEST_AUX_SENDER);
|
|
if (aux) {
|
|
GstPad *sinkpad;
|
|
GST_DEBUG_OBJECT (rtpbin, "linking AUX sender");
|
|
if (!setup_aux_sender (rtpbin, session, aux))
|
|
goto aux_session_failed;
|
|
|
|
pname = g_strdup_printf ("sink_%u", sessid);
|
|
sinkpad = gst_element_get_static_pad (aux, pname);
|
|
g_free (pname);
|
|
|
|
if (sinkpad == NULL)
|
|
goto aux_sink_failed;
|
|
|
|
if (!prev) {
|
|
send_rtp_sink = sinkpad;
|
|
} else {
|
|
GstPad *srcpad = gst_element_get_static_pad (prev, "src");
|
|
GstPadLinkReturn ret;
|
|
|
|
ret = gst_pad_link (srcpad, sinkpad);
|
|
gst_object_unref (srcpad);
|
|
if (ret != GST_PAD_LINK_OK) {
|
|
goto aux_link_failed;
|
|
}
|
|
gst_object_unref (sinkpad);
|
|
}
|
|
prev = aux;
|
|
} else {
|
|
/* get send_rtp pad and store */
|
|
session->send_rtp_sink =
|
|
gst_element_request_pad_simple (session->session, "send_rtp_sink");
|
|
if (session->send_rtp_sink == NULL)
|
|
goto pad_failed;
|
|
|
|
if (!complete_session_src (rtpbin, session))
|
|
goto session_src_failed;
|
|
|
|
if (!prev) {
|
|
send_rtp_sink = gst_object_ref (session->send_rtp_sink);
|
|
} else {
|
|
GstPad *srcpad = gst_element_get_static_pad (prev, "src");
|
|
GstPadLinkReturn ret;
|
|
|
|
ret = gst_pad_link (srcpad, session->send_rtp_sink);
|
|
gst_object_unref (srcpad);
|
|
if (ret != GST_PAD_LINK_OK)
|
|
goto session_link_failed;
|
|
}
|
|
}
|
|
|
|
session->send_rtp_sink_ghost =
|
|
gst_ghost_pad_new_from_template (name, send_rtp_sink, templ);
|
|
gst_object_unref (send_rtp_sink);
|
|
|
|
return session->send_rtp_sink_ghost;
|
|
|
|
/* ERRORS */
|
|
no_name:
|
|
{
|
|
g_warning ("rtpbin: cannot find session id for pad: %s",
|
|
GST_STR_NULL (name));
|
|
return NULL;
|
|
}
|
|
create_error:
|
|
{
|
|
/* create_session already warned */
|
|
return NULL;
|
|
}
|
|
existing_session:
|
|
{
|
|
g_warning ("rtpbin: session %u is already in use", sessid);
|
|
return NULL;
|
|
}
|
|
aux_session_failed:
|
|
{
|
|
g_warning ("rtpbin: failed to get AUX sink pad for session %u", sessid);
|
|
return NULL;
|
|
}
|
|
aux_sink_failed:
|
|
{
|
|
g_warning ("rtpbin: failed to get AUX sink pad for session %u", sessid);
|
|
return NULL;
|
|
}
|
|
aux_link_failed:
|
|
{
|
|
g_warning ("rtpbin: failed to link %" GST_PTR_FORMAT " for session %u",
|
|
aux, sessid);
|
|
return NULL;
|
|
}
|
|
pad_failed:
|
|
{
|
|
g_warning ("rtpbin: failed to get session pad for session %u", sessid);
|
|
return NULL;
|
|
}
|
|
session_src_failed:
|
|
{
|
|
g_warning ("rtpbin: failed to setup source pads for session %u", sessid);
|
|
return NULL;
|
|
}
|
|
session_link_failed:
|
|
{
|
|
g_warning ("rtpbin: failed to link %" GST_PTR_FORMAT " for session %u",
|
|
session, sessid);
|
|
return NULL;
|
|
}
|
|
enc_sink_failed:
|
|
{
|
|
g_warning ("rtpbin: failed to get %" GST_PTR_FORMAT
|
|
" sink pad for session %u", encoder, sessid);
|
|
return NULL;
|
|
}
|
|
}
|
|
|
|
static void
|
|
remove_send_rtp (GstRtpBin * rtpbin, GstRtpBinSession * session)
|
|
{
|
|
if (session->send_rtp_src_ghost) {
|
|
gst_pad_set_active (session->send_rtp_src_ghost, FALSE);
|
|
gst_element_remove_pad (GST_ELEMENT_CAST (rtpbin),
|
|
session->send_rtp_src_ghost);
|
|
session->send_rtp_src_ghost = NULL;
|
|
}
|
|
if (session->send_rtp_sink) {
|
|
gst_element_release_request_pad (GST_ELEMENT_CAST (session->session),
|
|
session->send_rtp_sink);
|
|
gst_object_unref (session->send_rtp_sink);
|
|
session->send_rtp_sink = NULL;
|
|
}
|
|
if (session->send_rtp_sink_ghost) {
|
|
gst_pad_set_active (session->send_rtp_sink_ghost, FALSE);
|
|
gst_element_remove_pad (GST_ELEMENT_CAST (rtpbin),
|
|
session->send_rtp_sink_ghost);
|
|
session->send_rtp_sink_ghost = NULL;
|
|
}
|
|
}
|
|
|
|
static void
|
|
remove_send_fec (GstRtpBin * rtpbin, GstRtpBinSession * session)
|
|
{
|
|
GSList *tmp;
|
|
|
|
for (tmp = session->send_fec_src_ghosts; tmp; tmp = tmp->next) {
|
|
GstPad *ghost = GST_PAD (tmp->data);
|
|
gst_pad_set_active (ghost, FALSE);
|
|
gst_element_remove_pad (GST_ELEMENT_CAST (rtpbin), ghost);
|
|
}
|
|
|
|
g_slist_free (session->send_fec_src_ghosts);
|
|
session->send_fec_src_ghosts = NULL;
|
|
}
|
|
|
|
/* Create a pad for sending RTCP for the session in @name. Must be called with
|
|
* RTP_BIN_LOCK.
|
|
*/
|
|
static GstPad *
|
|
create_send_rtcp (GstRtpBin * rtpbin, GstPadTemplate * templ,
|
|
const gchar * name)
|
|
{
|
|
guint sessid;
|
|
GstPad *encsrc;
|
|
GstElement *encoder;
|
|
GstRtpBinSession *session;
|
|
|
|
/* first get the session number */
|
|
if (name == NULL || sscanf (name, "send_rtcp_src_%u", &sessid) != 1)
|
|
goto no_name;
|
|
|
|
/* get or create session */
|
|
session = find_session_by_id (rtpbin, sessid);
|
|
if (!session) {
|
|
GST_DEBUG_OBJECT (rtpbin, "creating session %u", sessid);
|
|
/* create session now */
|
|
session = create_session (rtpbin, sessid);
|
|
if (session == NULL)
|
|
goto create_error;
|
|
}
|
|
|
|
/* check if pad was requested */
|
|
if (session->send_rtcp_src_ghost != NULL)
|
|
return session->send_rtcp_src_ghost;
|
|
|
|
/* get rtcp_src pad and store */
|
|
session->send_rtcp_src =
|
|
gst_element_request_pad_simple (session->session, "send_rtcp_src");
|
|
if (session->send_rtcp_src == NULL)
|
|
goto pad_failed;
|
|
|
|
GST_DEBUG_OBJECT (rtpbin, "getting RTCP encoder");
|
|
encoder = session_request_element (session, SIGNAL_REQUEST_RTCP_ENCODER);
|
|
if (encoder) {
|
|
gchar *ename;
|
|
GstPad *encsink;
|
|
GstPadLinkReturn ret;
|
|
|
|
GST_DEBUG_OBJECT (rtpbin, "linking RTCP encoder");
|
|
|
|
ename = g_strdup_printf ("rtcp_src_%u", sessid);
|
|
encsrc = gst_element_get_static_pad (encoder, ename);
|
|
g_free (ename);
|
|
if (encsrc == NULL)
|
|
goto enc_src_failed;
|
|
|
|
ename = g_strdup_printf ("rtcp_sink_%u", sessid);
|
|
encsink = gst_element_get_static_pad (encoder, ename);
|
|
g_free (ename);
|
|
if (encsink == NULL)
|
|
goto enc_sink_failed;
|
|
|
|
ret = gst_pad_link (session->send_rtcp_src, encsink);
|
|
gst_object_unref (encsink);
|
|
|
|
if (ret != GST_PAD_LINK_OK)
|
|
goto enc_link_failed;
|
|
} else {
|
|
GST_DEBUG_OBJECT (rtpbin, "no RTCP encoder given");
|
|
encsrc = gst_object_ref (session->send_rtcp_src);
|
|
}
|
|
|
|
session->send_rtcp_src_ghost =
|
|
gst_ghost_pad_new_from_template (name, encsrc, templ);
|
|
gst_object_unref (encsrc);
|
|
|
|
return session->send_rtcp_src_ghost;
|
|
|
|
/* ERRORS */
|
|
no_name:
|
|
{
|
|
g_warning ("rtpbin: cannot find session id for pad: %s",
|
|
GST_STR_NULL (name));
|
|
return NULL;
|
|
}
|
|
create_error:
|
|
{
|
|
/* create_session already warned */
|
|
return NULL;
|
|
}
|
|
pad_failed:
|
|
{
|
|
g_warning ("rtpbin: failed to get rtcp pad for session %u", sessid);
|
|
return NULL;
|
|
}
|
|
enc_src_failed:
|
|
{
|
|
g_warning ("rtpbin: failed to get encoder src pad for session %u", sessid);
|
|
return NULL;
|
|
}
|
|
enc_sink_failed:
|
|
{
|
|
g_warning ("rtpbin: failed to get encoder sink pad for session %u", sessid);
|
|
gst_object_unref (encsrc);
|
|
return NULL;
|
|
}
|
|
enc_link_failed:
|
|
{
|
|
g_warning ("rtpbin: failed to link rtcp encoder for session %u", sessid);
|
|
gst_object_unref (encsrc);
|
|
return NULL;
|
|
}
|
|
}
|
|
|
|
static void
|
|
remove_rtcp (GstRtpBin * rtpbin, GstRtpBinSession * session)
|
|
{
|
|
if (session->send_rtcp_src_ghost) {
|
|
gst_pad_set_active (session->send_rtcp_src_ghost, FALSE);
|
|
gst_element_remove_pad (GST_ELEMENT_CAST (rtpbin),
|
|
session->send_rtcp_src_ghost);
|
|
session->send_rtcp_src_ghost = NULL;
|
|
}
|
|
if (session->send_rtcp_src) {
|
|
gst_element_release_request_pad (session->session, session->send_rtcp_src);
|
|
gst_object_unref (session->send_rtcp_src);
|
|
session->send_rtcp_src = NULL;
|
|
}
|
|
}
|
|
|
|
/* If the requested name is NULL we should create a name with
|
|
* the session number assuming we want the lowest possible session
|
|
* with a free pad like the template */
|
|
static gchar *
|
|
gst_rtp_bin_get_free_pad_name (GstElement * element, GstPadTemplate * templ)
|
|
{
|
|
gboolean name_found = FALSE;
|
|
gint session = 0;
|
|
GstIterator *pad_it = NULL;
|
|
gchar *pad_name = NULL;
|
|
GValue data = { 0, };
|
|
|
|
GST_DEBUG_OBJECT (element, "find a free pad name for template");
|
|
while (!name_found) {
|
|
gboolean done = FALSE;
|
|
|
|
g_free (pad_name);
|
|
pad_name = g_strdup_printf (templ->name_template, session++);
|
|
pad_it = gst_element_iterate_pads (GST_ELEMENT (element));
|
|
name_found = TRUE;
|
|
while (!done) {
|
|
switch (gst_iterator_next (pad_it, &data)) {
|
|
case GST_ITERATOR_OK:
|
|
{
|
|
GstPad *pad;
|
|
gchar *name;
|
|
|
|
pad = g_value_get_object (&data);
|
|
name = gst_pad_get_name (pad);
|
|
|
|
if (strcmp (name, pad_name) == 0) {
|
|
done = TRUE;
|
|
name_found = FALSE;
|
|
}
|
|
g_free (name);
|
|
g_value_reset (&data);
|
|
break;
|
|
}
|
|
case GST_ITERATOR_ERROR:
|
|
case GST_ITERATOR_RESYNC:
|
|
/* restart iteration */
|
|
done = TRUE;
|
|
name_found = FALSE;
|
|
session = 0;
|
|
break;
|
|
case GST_ITERATOR_DONE:
|
|
done = TRUE;
|
|
break;
|
|
}
|
|
}
|
|
g_value_unset (&data);
|
|
gst_iterator_free (pad_it);
|
|
}
|
|
|
|
GST_DEBUG_OBJECT (element, "free pad name found: '%s'", pad_name);
|
|
return pad_name;
|
|
}
|
|
|
|
/*
|
|
*/
|
|
static GstPad *
|
|
gst_rtp_bin_request_new_pad (GstElement * element,
|
|
GstPadTemplate * templ, const gchar * name, const GstCaps * caps)
|
|
{
|
|
GstRtpBin *rtpbin;
|
|
GstElementClass *klass;
|
|
GstPad *result;
|
|
|
|
gchar *pad_name = NULL;
|
|
|
|
g_return_val_if_fail (templ != NULL, NULL);
|
|
g_return_val_if_fail (GST_IS_RTP_BIN (element), NULL);
|
|
|
|
rtpbin = GST_RTP_BIN (element);
|
|
klass = GST_ELEMENT_GET_CLASS (element);
|
|
|
|
GST_RTP_BIN_LOCK (rtpbin);
|
|
|
|
if (name == NULL) {
|
|
/* use a free pad name */
|
|
pad_name = gst_rtp_bin_get_free_pad_name (element, templ);
|
|
} else {
|
|
/* use the provided name */
|
|
pad_name = g_strdup (name);
|
|
}
|
|
|
|
GST_DEBUG_OBJECT (rtpbin, "Trying to request a pad with name %s", pad_name);
|
|
|
|
/* figure out the template */
|
|
if (templ == gst_element_class_get_pad_template (klass, "recv_rtp_sink_%u")) {
|
|
result = create_recv_rtp (rtpbin, templ, pad_name);
|
|
} else if (templ == gst_element_class_get_pad_template (klass,
|
|
"recv_rtcp_sink_%u")) {
|
|
result = create_recv_rtcp (rtpbin, templ, pad_name);
|
|
} else if (templ == gst_element_class_get_pad_template (klass,
|
|
"send_rtp_sink_%u")) {
|
|
result = create_send_rtp (rtpbin, templ, pad_name);
|
|
} else if (templ == gst_element_class_get_pad_template (klass,
|
|
"send_rtcp_src_%u")) {
|
|
result = create_send_rtcp (rtpbin, templ, pad_name);
|
|
} else if (templ == gst_element_class_get_pad_template (klass,
|
|
"recv_fec_sink_%u_%u")) {
|
|
result = create_recv_fec (rtpbin, templ, pad_name);
|
|
} else
|
|
goto wrong_template;
|
|
|
|
g_free (pad_name);
|
|
GST_RTP_BIN_UNLOCK (rtpbin);
|
|
|
|
if (result) {
|
|
gst_pad_set_active (result, TRUE);
|
|
gst_element_add_pad (GST_ELEMENT_CAST (rtpbin), result);
|
|
}
|
|
|
|
return result;
|
|
|
|
/* ERRORS */
|
|
wrong_template:
|
|
{
|
|
g_free (pad_name);
|
|
GST_RTP_BIN_UNLOCK (rtpbin);
|
|
g_warning ("rtpbin: this is not our template");
|
|
return NULL;
|
|
}
|
|
}
|
|
|
|
static void
|
|
gst_rtp_bin_release_pad (GstElement * element, GstPad * pad)
|
|
{
|
|
GstRtpBinSession *session;
|
|
GstRtpBin *rtpbin;
|
|
|
|
g_return_if_fail (GST_IS_GHOST_PAD (pad));
|
|
g_return_if_fail (GST_IS_RTP_BIN (element));
|
|
|
|
rtpbin = GST_RTP_BIN (element);
|
|
|
|
GST_RTP_BIN_LOCK (rtpbin);
|
|
GST_DEBUG_OBJECT (rtpbin, "Trying to release pad %s:%s",
|
|
GST_DEBUG_PAD_NAME (pad));
|
|
|
|
if (!(session = find_session_by_pad (rtpbin, pad)))
|
|
goto unknown_pad;
|
|
|
|
if (session->recv_rtp_sink_ghost == pad) {
|
|
remove_recv_rtp (rtpbin, session);
|
|
} else if (session->recv_rtcp_sink_ghost == pad) {
|
|
remove_recv_rtcp (rtpbin, session);
|
|
} else if (session->send_rtp_sink_ghost == pad) {
|
|
remove_send_rtp (rtpbin, session);
|
|
} else if (session->send_rtcp_src_ghost == pad) {
|
|
remove_rtcp (rtpbin, session);
|
|
} else if (pad_is_recv_fec (session, pad)) {
|
|
remove_recv_fec_for_pad (rtpbin, session, pad);
|
|
}
|
|
|
|
/* no more request pads, free the complete session */
|
|
if (session->recv_rtp_sink_ghost == NULL
|
|
&& session->recv_rtcp_sink_ghost == NULL
|
|
&& session->send_rtp_sink_ghost == NULL
|
|
&& session->send_rtcp_src_ghost == NULL
|
|
&& session->recv_fec_sink_ghosts == NULL) {
|
|
GST_DEBUG_OBJECT (rtpbin, "no more pads for session %p", session);
|
|
rtpbin->sessions = g_slist_remove (rtpbin->sessions, session);
|
|
free_session (session, rtpbin);
|
|
}
|
|
GST_RTP_BIN_UNLOCK (rtpbin);
|
|
|
|
return;
|
|
|
|
/* ERROR */
|
|
unknown_pad:
|
|
{
|
|
GST_RTP_BIN_UNLOCK (rtpbin);
|
|
g_warning ("rtpbin: %s:%s is not one of our request pads",
|
|
GST_DEBUG_PAD_NAME (pad));
|
|
return;
|
|
}
|
|
}
|