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95429f1d4b
rtpbin can now send a custom in-band downstream event which informs downstream that the bin has received an RTCP SR packet. This is useful for applications which want to drop the initial unsynchronized received RTP packets. Fixes https://bugzilla.gnome.org/show_bug.cgi?id=703560 Signed-off-by: Carlos Rafael Giani <dv@pseudoterminal.org>
3190 lines
100 KiB
C
3190 lines
100 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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* @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 wil 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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* <refsect2>
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* <title>Example pipelines</title>
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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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* </refsect2>
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*
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* Last reviewed on 2007-08-30 (0.10.6)
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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-marshal.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 <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")
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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")
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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")
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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")
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);
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#define GST_RTP_BIN_GET_PRIVATE(obj) \
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(G_TYPE_INSTANCE_GET_PRIVATE ((obj), GST_TYPE_RTP_BIN, GstRtpBinPrivate))
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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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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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/* UNIX (ntp) time of last SR sync used */
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guint64 last_unix;
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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_INTERNAL_SESSION,
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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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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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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_LAST
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};
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enum
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{
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GST_RTP_BIN_RTCP_SYNC_ALWAYS,
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GST_RTP_BIN_RTCP_SYNC_INITIAL,
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GST_RTP_BIN_RTCP_SYNC_RTP
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};
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#define GST_RTP_BIN_RTCP_SYNC_TYPE (gst_rtp_bin_rtcp_sync_get_type())
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static GType
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gst_rtp_bin_rtcp_sync_get_type (void)
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{
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static GType rtcp_sync_type = 0;
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static const GEnumValue rtcp_sync_types[] = {
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{GST_RTP_BIN_RTCP_SYNC_ALWAYS, "always", "always"},
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{GST_RTP_BIN_RTCP_SYNC_INITIAL, "initial", "initial"},
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{GST_RTP_BIN_RTCP_SYNC_RTP, "rtp-info", "rtp-info"},
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{0, NULL, NULL},
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};
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if (!rtcp_sync_type) {
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rtcp_sync_type = g_enum_register_static ("GstRTCPSync", rtcp_sync_types);
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}
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return rtcp_sync_type;
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}
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/* helper objects */
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typedef struct _GstRtpBinSession GstRtpBinSession;
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typedef struct _GstRtpBinStream GstRtpBinStream;
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typedef struct _GstRtpBinClient GstRtpBinClient;
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static guint gst_rtp_bin_signals[LAST_SIGNAL] = { 0 };
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static GstCaps *pt_map_requested (GstElement * element, guint pt,
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GstRtpBinSession * session);
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static void payload_type_change (GstElement * element, guint pt,
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GstRtpBinSession * session);
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static void remove_recv_rtp (GstRtpBin * rtpbin, GstRtpBinSession * session);
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static void remove_recv_rtcp (GstRtpBin * rtpbin, GstRtpBinSession * session);
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static void remove_send_rtp (GstRtpBin * rtpbin, GstRtpBinSession * session);
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static void remove_rtcp (GstRtpBin * rtpbin, GstRtpBinSession * session);
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static void free_client (GstRtpBinClient * client, GstRtpBin * bin);
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static void free_stream (GstRtpBinStream * stream, GstRtpBin * bin);
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/* Manages the RTP stream for one SSRC.
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*
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* We pipe the stream (comming from the SSRC demuxer) into a jitterbuffer.
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* If we see an SDES RTCP packet that links multiple SSRCs together based on a
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* common CNAME, we create a GstRtpBinClient structure to group the SSRCs
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* together (see below).
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*/
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struct _GstRtpBinStream
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{
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/* the SSRC of this stream */
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guint32 ssrc;
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/* parent bin */
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GstRtpBin *bin;
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/* the session this SSRC belongs to */
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GstRtpBinSession *session;
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/* the jitterbuffer of the SSRC */
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GstElement *buffer;
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gulong buffer_handlesync_sig;
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gulong buffer_ptreq_sig;
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gulong buffer_ntpstop_sig;
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gint percent;
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/* the PT demuxer of the SSRC */
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GstElement *demux;
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gulong demux_newpad_sig;
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gulong demux_padremoved_sig;
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gulong demux_ptreq_sig;
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gulong demux_ptchange_sig;
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/* if we have calculated a valid rt_delta for this stream */
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gboolean have_sync;
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/* mapping to local RTP and NTP time */
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gint64 rt_delta;
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gint64 rtp_delta;
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/* base rtptime in gst time */
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gint64 clock_base;
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};
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#define GST_RTP_SESSION_LOCK(sess) g_mutex_lock (&(sess)->lock)
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#define GST_RTP_SESSION_UNLOCK(sess) g_mutex_unlock (&(sess)->lock)
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/* Manages the receiving end of the packets.
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*
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* There is one such structure for each RTP session (audio/video/...).
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* We get the RTP/RTCP packets and stuff them into the session manager. From
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* there they are pushed into an SSRC demuxer that splits the stream based on
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* SSRC. Each of the SSRC streams go into their own jitterbuffer (managed with
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* the GstRtpBinStream above).
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*/
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struct _GstRtpBinSession
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{
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/* session id */
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gint id;
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/* the parent bin */
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GstRtpBin *bin;
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/* the session element */
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GstElement *session;
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/* the SSRC demuxer */
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GstElement *demux;
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gulong demux_newpad_sig;
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gulong demux_padremoved_sig;
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GMutex lock;
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/* list of GstRtpBinStream */
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GSList *streams;
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/* mapping of payload type to caps */
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GHashTable *ptmap;
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/* the pads of the session */
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GstPad *recv_rtp_sink;
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GstPad *recv_rtp_sink_ghost;
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GstPad *recv_rtp_src;
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GstPad *recv_rtcp_sink;
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GstPad *recv_rtcp_sink_ghost;
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GstPad *sync_src;
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GstPad *send_rtp_sink;
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GstPad *send_rtp_sink_ghost;
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GstPad *send_rtp_src;
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GstPad *send_rtp_src_ghost;
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GstPad *send_rtcp_src;
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GstPad *send_rtcp_src_ghost;
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};
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|
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/* Manages the RTP streams that come from one client and should therefore be
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* synchronized.
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*/
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struct _GstRtpBinClient
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{
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/* the common CNAME for the streams */
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gchar *cname;
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guint cname_len;
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/* the streams */
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guint nstreams;
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GSList *streams;
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};
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/* find a session with the given id. Must be called with RTP_BIN_LOCK */
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static GstRtpBinSession *
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find_session_by_id (GstRtpBin * rtpbin, gint id)
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{
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GSList *walk;
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|
|
|
for (walk = rtpbin->sessions; walk; walk = g_slist_next (walk)) {
|
|
GstRtpBinSession *sess = (GstRtpBinSession *) walk->data;
|
|
|
|
if (sess->id == id)
|
|
return sess;
|
|
}
|
|
return 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))
|
|
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);
|
|
}
|
|
|
|
/* 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;
|
|
GstState target;
|
|
|
|
if (!(session = gst_element_factory_make ("rtpsession", NULL)))
|
|
goto no_session;
|
|
|
|
if (!(demux = gst_element_factory_make ("rtpssrcdemux", NULL)))
|
|
goto no_demux;
|
|
|
|
sess = g_new0 (GstRtpBinSession, 1);
|
|
g_mutex_init (&sess->lock);
|
|
sess->id = id;
|
|
sess->bin = rtpbin;
|
|
sess->session = session;
|
|
sess->demux = demux;
|
|
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 (session, "sdes", rtpbin->sdes, "use-pipeline-clock",
|
|
rtpbin->use_pipeline_clock, 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);
|
|
|
|
gst_bin_add (GST_BIN_CAST (rtpbin), session);
|
|
gst_bin_add (GST_BIN_CAST (rtpbin), demux);
|
|
|
|
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);
|
|
|
|
return sess;
|
|
|
|
/* ERRORS */
|
|
no_session:
|
|
{
|
|
g_warning ("rtpbin: could not create gstrtpsession element");
|
|
return NULL;
|
|
}
|
|
no_demux:
|
|
{
|
|
gst_object_unref (session);
|
|
g_warning ("rtpbin: could not create gstrtpssrcdemux element");
|
|
return NULL;
|
|
}
|
|
}
|
|
|
|
/* 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_state (sess->demux, GST_STATE_NULL);
|
|
gst_element_set_state (sess->session, GST_STATE_NULL);
|
|
|
|
remove_recv_rtp (bin, sess);
|
|
remove_recv_rtcp (bin, sess);
|
|
remove_send_rtp (bin, sess);
|
|
remove_rtcp (bin, sess);
|
|
|
|
gst_bin_remove (GST_BIN_CAST (bin), sess->session);
|
|
gst_bin_remove (GST_BIN_CAST (bin), sess->demux);
|
|
|
|
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 %d 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 ("emiting signal for pt %d in session %d", 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 %d 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->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);
|
|
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 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: %d",
|
|
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 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;
|
|
|
|
g_object_set_property (G_OBJECT (stream->buffer), name, value);
|
|
}
|
|
GST_RTP_SESSION_UNLOCK (session);
|
|
}
|
|
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, 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;
|
|
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);
|
|
|
|
clock_time = gst_clock_get_time (clock);
|
|
|
|
if (bin->use_pipeline_clock) {
|
|
ntpns = clock_time - base_time;
|
|
} else {
|
|
GTimeVal current;
|
|
|
|
/* get current NTP time */
|
|
g_get_current_time (¤t);
|
|
ntpns = GST_TIMEVAL_TO_TIME (current);
|
|
}
|
|
|
|
/* add constant to convert from 1970 based time to 1900 based time */
|
|
ntpns += (2208988800LL * GST_SECOND);
|
|
|
|
/* get current clock time and convert to running time */
|
|
rt = clock_time - base_time;
|
|
|
|
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, gboolean check)
|
|
{
|
|
gint64 prev_ts_offset;
|
|
|
|
g_object_get (stream->buffer, "ts-offset", &prev_ts_offset, NULL);
|
|
|
|
/* delta changed, see how much */
|
|
if (prev_ts_offset != ts_offset) {
|
|
gint64 diff;
|
|
|
|
diff = prev_ts_offset - ts_offset;
|
|
|
|
GST_DEBUG_OBJECT (bin,
|
|
"ts-offset %" G_GINT64_FORMAT ", prev %" G_GINT64_FORMAT
|
|
", diff: %" G_GINT64_FORMAT, ts_offset, prev_ts_offset, diff);
|
|
|
|
if (check) {
|
|
/* only change diff when it changed more than 4 milliseconds. This
|
|
* compensates for rounding errors in NTP to RTP timestamp
|
|
* conversions */
|
|
if (ABS (diff) < 4 * GST_MSECOND) {
|
|
GST_DEBUG_OBJECT (bin, "offset too small, ignoring");
|
|
return;
|
|
}
|
|
if (ABS (diff) > (3 * GST_SECOND)) {
|
|
GST_WARNING_OBJECT (bin, "offset unusually large, ignoring");
|
|
return;
|
|
}
|
|
}
|
|
g_object_set (stream->buffer, "ts-offset", ts_offset, NULL);
|
|
}
|
|
GST_DEBUG_OBJECT (bin, "stream SSRC %08x, delta %" G_GINT64_FORMAT,
|
|
stream->ssrc, 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 ("GstRTCPSRReceived", NULL));
|
|
|
|
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,
|
|
guint8 * data, guint64 ntptime, guint64 last_extrtptime,
|
|
guint64 base_rtptime, guint64 base_time, guint clock_rate,
|
|
gint64 rtp_clock_base)
|
|
{
|
|
GstRtpBinClient *client;
|
|
gboolean created;
|
|
GSList *walk;
|
|
guint64 local_rt;
|
|
guint64 local_rtp;
|
|
GstClockTime running_time;
|
|
guint64 ntpnstime;
|
|
gint64 ntpdiff, rtdiff;
|
|
guint64 last_unix;
|
|
|
|
/* 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. */
|
|
local_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, local_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. */
|
|
local_rt = gst_util_uint64_scale_int (local_rtp, GST_SECOND, clock_rate);
|
|
local_rt += base_time;
|
|
|
|
/* convert ntptime to unix time since 1900 */
|
|
last_unix = gst_util_uint64_scale (ntptime, GST_SECOND,
|
|
(G_GINT64_CONSTANT (1) << 32));
|
|
|
|
stream->have_sync = TRUE;
|
|
|
|
GST_DEBUG_OBJECT (bin,
|
|
"local UNIX %" G_GUINT64_FORMAT ", remote UNIX %" G_GUINT64_FORMAT,
|
|
local_rt, last_unix);
|
|
|
|
/* recalc inter stream playout offset, but only if there is more than one
|
|
* stream or we're doing NTP sync. */
|
|
if (bin->ntp_sync) {
|
|
/* 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 eachother. */
|
|
get_current_times (bin, &running_time, &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 = ntpnstime - last_unix;
|
|
/* 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 = running_time - local_rt;
|
|
|
|
GST_DEBUG_OBJECT (bin,
|
|
"NTP time %" G_GUINT64_FORMAT ", last unix %" G_GUINT64_FORMAT,
|
|
ntpnstime, last_unix);
|
|
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, FALSE);
|
|
} else {
|
|
gint64 min, rtp_min, clock_base = stream->clock_base;
|
|
gboolean all_sync, use_rtp;
|
|
gboolean rtcp_sync = g_atomic_int_get (&bin->rtcp_sync);
|
|
|
|
/* calculate delta between server and receiver. last_unix 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 = last_unix - local_rt;
|
|
|
|
/* 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 mininum 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 imposible 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;
|
|
|
|
use_rtp = TRUE;
|
|
/* signed version for convienience */
|
|
clock_base = base_rtptime;
|
|
/* deal with possible wrap-around */
|
|
ext_base = base_rtptime;
|
|
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 >= clock_base) {
|
|
if (rtp_clock_base - clock_base < 10 * clock_rate) {
|
|
rtp_clock_base = base_time +
|
|
gst_util_uint64_scale_int (rtp_clock_base - clock_base,
|
|
GST_SECOND, clock_rate);
|
|
} else {
|
|
use_rtp = FALSE;
|
|
}
|
|
} else {
|
|
if (clock_base - rtp_clock_base < 10 * clock_rate) {
|
|
rtp_clock_base = base_time -
|
|
gst_util_uint64_scale_int (clock_base - 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_clock_base;
|
|
/* 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_clock_base;
|
|
} 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 && (last_unix - bin->priv->last_unix) <
|
|
bin->rtcp_sync_interval * GST_MSECOND) {
|
|
GST_DEBUG_OBJECT (bin, "discarding RTCP sender packet for sync; "
|
|
"previous sender info too recent "
|
|
"(previous UNIX %" G_GUINT64_FORMAT ")", bin->priv->last_unix);
|
|
return;
|
|
}
|
|
bin->priv->last_unix = last_unix;
|
|
|
|
/* 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, 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 ntptime;
|
|
gboolean have_sr, have_sdes;
|
|
gboolean more;
|
|
guint64 base_rtptime;
|
|
guint64 base_time;
|
|
guint clock_rate;
|
|
guint64 clock_base;
|
|
guint64 extrtptime;
|
|
GstBuffer *buffer;
|
|
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. */
|
|
base_rtptime =
|
|
g_value_get_uint64 (gst_structure_get_value (s, "base-rtptime"));
|
|
base_time = g_value_get_uint64 (gst_structure_get_value (s, "base-time"));
|
|
clock_rate = g_value_get_uint (gst_structure_get_value (s, "clock-rate"));
|
|
clock_base = g_value_get_uint64 (gst_structure_get_value (s, "clock-base"));
|
|
extrtptime =
|
|
g_value_get_uint64 (gst_structure_get_value (s, "sr-ext-rtptime"));
|
|
buffer = gst_value_get_buffer (gst_structure_get_value (s, "sr-buffer"));
|
|
|
|
have_sr = FALSE;
|
|
have_sdes = 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 */
|
|
if (have_sr)
|
|
break;
|
|
/* get NTP and RTP times */
|
|
gst_rtcp_packet_sr_get_sender_info (&packet, &ssrc, &ntptime, NULL,
|
|
NULL, NULL);
|
|
|
|
GST_DEBUG_OBJECT (bin, "received sync packet from SSRC %08x", ssrc);
|
|
/* ignore SR that is not ours */
|
|
if (ssrc != stream->ssrc)
|
|
continue;
|
|
|
|
have_sr = TRUE;
|
|
break;
|
|
case GST_RTCP_TYPE_SDES:
|
|
{
|
|
gboolean more_items, more_entries;
|
|
|
|
/* only deal with first SDES, there is only supposed to be one SDES in
|
|
* the RTCP packet but we deal with bad packets gracefully. Also bail
|
|
* out if we have not seen an SR item yet. */
|
|
if (have_sdes || !have_sr)
|
|
break;
|
|
|
|
GST_RTCP_SDES_FOR_ITEMS (more_items, &packet) {
|
|
/* 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;
|
|
guint8 *data;
|
|
|
|
gst_rtcp_packet_sdes_get_entry (&packet, &type, &len, &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,
|
|
ntptime, extrtptime, base_rtptime, base_time, clock_rate,
|
|
clock_base);
|
|
GST_RTP_BIN_UNLOCK (bin);
|
|
}
|
|
}
|
|
}
|
|
have_sdes = TRUE;
|
|
break;
|
|
}
|
|
default:
|
|
/* we can ignore these packets */
|
|
break;
|
|
}
|
|
}
|
|
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;
|
|
|
|
rtpbin = session->bin;
|
|
|
|
if (!(buffer = gst_element_factory_make ("rtpjitterbuffer", NULL)))
|
|
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 = buffer;
|
|
stream->demux = demux;
|
|
|
|
stream->have_sync = FALSE;
|
|
stream->rt_delta = 0;
|
|
stream->rtp_delta = 0;
|
|
stream->percent = 100;
|
|
stream->clock_base = -100 * GST_SECOND;
|
|
session->streams = g_slist_prepend (session->streams, stream);
|
|
|
|
/* provide clock_rate to the jitterbuffer when needed */
|
|
stream->buffer_ptreq_sig = g_signal_connect (buffer, "request-pt-map",
|
|
(GCallback) pt_map_requested, session);
|
|
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 */
|
|
g_object_set (buffer, "latency", rtpbin->latency_ms, NULL);
|
|
g_object_set (buffer, "drop-on-latency", rtpbin->drop_on_latency, NULL);
|
|
g_object_set (buffer, "do-lost", rtpbin->do_lost, NULL);
|
|
g_object_set (buffer, "mode", rtpbin->buffer_mode, NULL);
|
|
|
|
if (!rtpbin->ignore_pt)
|
|
gst_bin_add (GST_BIN_CAST (rtpbin), demux);
|
|
gst_bin_add (GST_BIN_CAST (rtpbin), buffer);
|
|
|
|
/* link stuff */
|
|
if (demux)
|
|
gst_element_link (buffer, demux);
|
|
|
|
if (rtpbin->buffering) {
|
|
guint64 last_out;
|
|
|
|
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 */
|
|
no_jitterbuffer:
|
|
{
|
|
g_warning ("rtpbin: could not create gstrtpjitterbuffer element");
|
|
return NULL;
|
|
}
|
|
no_demux:
|
|
{
|
|
gst_object_unref (buffer);
|
|
g_warning ("rtpbin: could not create gstrtpptdemux element");
|
|
return NULL;
|
|
}
|
|
}
|
|
|
|
/* called with RTP_BIN_LOCK */
|
|
static void
|
|
free_stream (GstRtpBinStream * stream, GstRtpBin * bin)
|
|
{
|
|
GSList *clients, *next_client;
|
|
|
|
GST_DEBUG_OBJECT (bin, "freeing stream %p", stream);
|
|
|
|
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->buffer, stream->buffer_handlesync_sig);
|
|
g_signal_handler_disconnect (stream->buffer, stream->buffer_ptreq_sig);
|
|
g_signal_handler_disconnect (stream->buffer, stream->buffer_ntpstop_sig);
|
|
|
|
if (stream->demux)
|
|
gst_element_set_locked_state (stream->demux, TRUE);
|
|
gst_element_set_locked_state (stream->buffer, TRUE);
|
|
|
|
if (stream->demux)
|
|
gst_element_set_state (stream->demux, GST_STATE_NULL);
|
|
gst_element_set_state (stream->buffer, GST_STATE_NULL);
|
|
|
|
/* now remove this signal, we need this while going to NULL because it to
|
|
* do some cleanups */
|
|
if (stream->demux)
|
|
g_signal_handler_disconnect (stream->demux, stream->demux_padremoved_sig);
|
|
|
|
gst_bin_remove (GST_BIN_CAST (bin), 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 (GstRtpBin, gst_rtp_bin, GST_TYPE_BIN);
|
|
|
|
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;
|
|
|
|
g_type_class_add_private (klass, sizeof (GstRtpBinPrivate));
|
|
|
|
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),
|
|
NULL, NULL, gst_rtp_bin_marshal_BOXED__UINT_UINT, 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.
|
|
*
|
|
* Since: 0.10.17
|
|
*/
|
|
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, gst_rtp_bin_marshal_VOID__UINT_UINT, 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, g_cclosure_marshal_VOID__VOID, 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, g_cclosure_marshal_VOID__VOID, G_TYPE_NONE,
|
|
0, G_TYPE_NONE);
|
|
|
|
/**
|
|
* 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, gst_rtp_bin_marshal_OBJECT__UINT,
|
|
RTP_TYPE_SESSION, 1, 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, gst_rtp_bin_marshal_VOID__UINT_UINT, 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, gst_rtp_bin_marshal_VOID__UINT_UINT, 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, gst_rtp_bin_marshal_VOID__UINT_UINT, 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, gst_rtp_bin_marshal_VOID__UINT_UINT, 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, gst_rtp_bin_marshal_VOID__UINT_UINT, 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, gst_rtp_bin_marshal_VOID__UINT_UINT, 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, gst_rtp_bin_marshal_VOID__UINT_UINT, 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, gst_rtp_bin_marshal_VOID__UINT_UINT, 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, gst_rtp_bin_marshal_VOID__UINT_UINT, 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, gst_rtp_bin_marshal_VOID__UINT_UINT, 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));
|
|
|
|
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",
|
|
DEFAULT_USE_PIPELINE_CLOCK,
|
|
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
|
|
/**
|
|
* GstRtpBin::buffer-mode:
|
|
*
|
|
* Control the buffering and timestamping mode used by the jitterbuffer.
|
|
*
|
|
* Since: 0.10.17
|
|
*/
|
|
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.
|
|
*
|
|
* Since: 0.10.21
|
|
*/
|
|
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.
|
|
*
|
|
* Since: 0.10.31
|
|
*/
|
|
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.
|
|
*
|
|
* Since: 0.10.31
|
|
*/
|
|
g_object_class_install_property (gobject_class, PROP_RTCP_SYNC_INTERVAL,
|
|
g_param_spec_uint ("rtcp-sync-interval", "RTCP Sync Interval",
|
|
"RTCP SR 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));
|
|
|
|
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_pad_template (gstelement_class,
|
|
gst_static_pad_template_get (&rtpbin_recv_rtp_sink_template));
|
|
gst_element_class_add_pad_template (gstelement_class,
|
|
gst_static_pad_template_get (&rtpbin_recv_rtcp_sink_template));
|
|
gst_element_class_add_pad_template (gstelement_class,
|
|
gst_static_pad_template_get (&rtpbin_send_rtp_sink_template));
|
|
|
|
/* src pads */
|
|
gst_element_class_add_pad_template (gstelement_class,
|
|
gst_static_pad_template_get (&rtpbin_recv_rtp_src_template));
|
|
gst_element_class_add_pad_template (gstelement_class,
|
|
gst_static_pad_template_get (&rtpbin_send_rtcp_src_template));
|
|
gst_element_class_add_pad_template (gstelement_class,
|
|
gst_static_pad_template_get (&rtpbin_send_rtp_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_internal_session =
|
|
GST_DEBUG_FUNCPTR (gst_rtp_bin_get_internal_session);
|
|
|
|
GST_DEBUG_CATEGORY_INIT (gst_rtp_bin_debug, "rtpbin", 0, "RTP bin");
|
|
}
|
|
|
|
static void
|
|
gst_rtp_bin_init (GstRtpBin * rtpbin)
|
|
{
|
|
gchar *cname;
|
|
|
|
rtpbin->priv = GST_RTP_BIN_GET_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;
|
|
|
|
/* 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);
|
|
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);
|
|
|
|
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 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 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);
|
|
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;
|
|
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;
|
|
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;
|
|
|
|
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_unix = 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;
|
|
}
|
|
|
|
/* a new pad (SSRC) was created in @session. This signal is emited from the
|
|
* payload demuxer. */
|
|
static void
|
|
new_payload_found (GstElement * element, guint pt, GstPad * pad,
|
|
GstRtpBinStream * stream)
|
|
{
|
|
GstRtpBin *rtpbin;
|
|
GstElementClass *klass;
|
|
GstPadTemplate *templ;
|
|
gchar *padname;
|
|
GstPad *gpad;
|
|
|
|
rtpbin = stream->bin;
|
|
|
|
GST_DEBUG ("new payload pad %d", pt);
|
|
|
|
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_element_add_pad (GST_ELEMENT_CAST (rtpbin), gpad);
|
|
|
|
return;
|
|
|
|
shutdown:
|
|
{
|
|
GST_DEBUG ("ignoring, we are shutting down");
|
|
return;
|
|
}
|
|
}
|
|
|
|
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 %d in session %d", 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 void
|
|
payload_type_change (GstElement * element, guint pt, GstRtpBinSession * session)
|
|
{
|
|
GST_DEBUG_OBJECT (session->bin,
|
|
"emiting signal for pt type changed to %d in session %d", pt,
|
|
session->id);
|
|
|
|
g_signal_emit (session->bin, gst_rtp_bin_signals[SIGNAL_PAYLOAD_TYPE_CHANGE],
|
|
0, session->id, pt);
|
|
}
|
|
|
|
/* emited 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))
|
|
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);
|
|
sinkpad = gst_element_get_static_pad (stream->buffer, "sink");
|
|
gst_pad_link (srcpad, sinkpad);
|
|
gst_object_unref (sinkpad);
|
|
gst_object_unref (srcpad);
|
|
|
|
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);
|
|
sinkpad = gst_element_get_request_pad (stream->buffer, "sink_rtcp");
|
|
gst_pad_link (srcpad, sinkpad);
|
|
gst_object_unref (sinkpad);
|
|
gst_object_unref (srcpad);
|
|
|
|
/* 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 emited 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) 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);
|
|
} else {
|
|
/* add gstrtpjitterbuffer src pad to pads */
|
|
GstElementClass *klass;
|
|
GstPadTemplate *templ;
|
|
gchar *padname;
|
|
GstPad *gpad, *pad;
|
|
|
|
pad = gst_element_get_static_pad (stream->buffer, "src");
|
|
|
|
/* 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, 255);
|
|
gpad = gst_ghost_pad_new_from_template (padname, pad, templ);
|
|
g_free (padname);
|
|
|
|
gst_pad_set_active (gpad, TRUE);
|
|
gst_element_add_pad (GST_ELEMENT_CAST (rtpbin), gpad);
|
|
|
|
gst_object_unref (pad);
|
|
}
|
|
|
|
GST_RTP_SESSION_UNLOCK (session);
|
|
GST_RTP_BIN_SHUTDOWN_UNLOCK (rtpbin);
|
|
|
|
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;
|
|
}
|
|
}
|
|
|
|
/* 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)
|
|
{
|
|
GstPad *sinkdpad;
|
|
guint sessid;
|
|
GstRtpBinSession *session;
|
|
GstPadLinkReturn lres;
|
|
|
|
/* 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 %d", sessid);
|
|
|
|
/* get or create session */
|
|
session = find_session_by_id (rtpbin, sessid);
|
|
if (!session) {
|
|
GST_DEBUG_OBJECT (rtpbin, "creating session %d", 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;
|
|
|
|
GST_DEBUG_OBJECT (rtpbin, "getting RTP sink pad");
|
|
/* get recv_rtp pad and store */
|
|
session->recv_rtp_sink =
|
|
gst_element_get_request_pad (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, "getting RTP src pad");
|
|
/* get srcpad, link to SSRCDemux */
|
|
session->recv_rtp_src =
|
|
gst_element_get_static_pad (session->session, "recv_rtp_src");
|
|
if (session->recv_rtp_src == NULL)
|
|
goto pad_failed;
|
|
|
|
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");
|
|
lres = gst_pad_link (session->recv_rtp_src, sinkdpad);
|
|
gst_object_unref (sinkdpad);
|
|
if (lres != GST_PAD_LINK_OK)
|
|
goto link_failed;
|
|
|
|
/* 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);
|
|
|
|
GST_DEBUG_OBJECT (rtpbin, "ghosting session sink pad");
|
|
session->recv_rtp_sink_ghost =
|
|
gst_ghost_pad_new_from_template (name, session->recv_rtp_sink, templ);
|
|
gst_pad_set_active (session->recv_rtp_sink_ghost, TRUE);
|
|
gst_element_add_pad (GST_ELEMENT_CAST (rtpbin), session->recv_rtp_sink_ghost);
|
|
|
|
return session->recv_rtp_sink_ghost;
|
|
|
|
/* ERRORS */
|
|
no_name:
|
|
{
|
|
g_warning ("rtpbin: invalid name given");
|
|
return NULL;
|
|
}
|
|
create_error:
|
|
{
|
|
/* create_session already warned */
|
|
return NULL;
|
|
}
|
|
pad_failed:
|
|
{
|
|
g_warning ("rtpbin: failed to get session pad");
|
|
return NULL;
|
|
}
|
|
link_failed:
|
|
{
|
|
g_warning ("rtpbin: failed to link pads");
|
|
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;
|
|
}
|
|
}
|
|
|
|
/* 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 *sinkdpad;
|
|
GstPadLinkReturn lres;
|
|
|
|
/* 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 %d", sessid);
|
|
|
|
/* get or create the session */
|
|
session = find_session_by_id (rtpbin, sessid);
|
|
if (!session) {
|
|
GST_DEBUG_OBJECT (rtpbin, "creating session %d", 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;
|
|
|
|
/* get recv_rtp pad and store */
|
|
GST_DEBUG_OBJECT (rtpbin, "getting RTCP sink pad");
|
|
session->recv_rtcp_sink =
|
|
gst_element_get_request_pad (session->session, "recv_rtcp_sink");
|
|
if (session->recv_rtcp_sink == NULL)
|
|
goto pad_failed;
|
|
|
|
/* 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 pad_failed;
|
|
|
|
GST_DEBUG_OBJECT (rtpbin, "getting demuxer RTCP sink pad");
|
|
sinkdpad = gst_element_get_static_pad (session->demux, "rtcp_sink");
|
|
lres = gst_pad_link (session->sync_src, sinkdpad);
|
|
gst_object_unref (sinkdpad);
|
|
if (lres != GST_PAD_LINK_OK)
|
|
goto link_failed;
|
|
|
|
session->recv_rtcp_sink_ghost =
|
|
gst_ghost_pad_new_from_template (name, session->recv_rtcp_sink, templ);
|
|
gst_pad_set_active (session->recv_rtcp_sink_ghost, TRUE);
|
|
gst_element_add_pad (GST_ELEMENT_CAST (rtpbin),
|
|
session->recv_rtcp_sink_ghost);
|
|
|
|
return session->recv_rtcp_sink_ghost;
|
|
|
|
/* ERRORS */
|
|
no_name:
|
|
{
|
|
g_warning ("rtpbin: invalid name given");
|
|
return NULL;
|
|
}
|
|
create_error:
|
|
{
|
|
/* create_session already warned */
|
|
return NULL;
|
|
}
|
|
pad_failed:
|
|
{
|
|
g_warning ("rtpbin: failed to get session pad");
|
|
return NULL;
|
|
}
|
|
link_failed:
|
|
{
|
|
g_warning ("rtpbin: failed to link pads");
|
|
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;
|
|
}
|
|
}
|
|
|
|
/* 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 *gname;
|
|
guint sessid;
|
|
GstRtpBinSession *session;
|
|
GstElementClass *klass;
|
|
|
|
/* 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;
|
|
|
|
/* get send_rtp pad and store */
|
|
session->send_rtp_sink =
|
|
gst_element_get_request_pad (session->session, "send_rtp_sink");
|
|
if (session->send_rtp_sink == NULL)
|
|
goto pad_failed;
|
|
|
|
session->send_rtp_sink_ghost =
|
|
gst_ghost_pad_new_from_template (name, session->send_rtp_sink, templ);
|
|
gst_pad_set_active (session->send_rtp_sink_ghost, TRUE);
|
|
gst_element_add_pad (GST_ELEMENT_CAST (rtpbin), session->send_rtp_sink_ghost);
|
|
|
|
/* get srcpad */
|
|
session->send_rtp_src =
|
|
gst_element_get_static_pad (session->session, "send_rtp_src");
|
|
if (session->send_rtp_src == NULL)
|
|
goto no_srcpad;
|
|
|
|
/* 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, session->send_rtp_src, templ);
|
|
gst_pad_set_active (session->send_rtp_src_ghost, TRUE);
|
|
gst_element_add_pad (GST_ELEMENT_CAST (rtpbin), session->send_rtp_src_ghost);
|
|
g_free (gname);
|
|
|
|
return session->send_rtp_sink_ghost;
|
|
|
|
/* ERRORS */
|
|
no_name:
|
|
{
|
|
g_warning ("rtpbin: invalid name given");
|
|
return NULL;
|
|
}
|
|
create_error:
|
|
{
|
|
/* create_session already warned */
|
|
return NULL;
|
|
}
|
|
pad_failed:
|
|
{
|
|
g_warning ("rtpbin: failed to get session pad for session %d", sessid);
|
|
return NULL;
|
|
}
|
|
no_srcpad:
|
|
{
|
|
g_warning ("rtpbin: failed to get rtp source pad for session %d", 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_src) {
|
|
gst_object_unref (session->send_rtp_src);
|
|
session->send_rtp_src = 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;
|
|
}
|
|
}
|
|
|
|
/* Create a pad for sending RTCP for the session in @name. Must be called with
|
|
* RTP_BIN_LOCK.
|
|
*/
|
|
static GstPad *
|
|
create_rtcp (GstRtpBin * rtpbin, GstPadTemplate * templ, const gchar * name)
|
|
{
|
|
guint sessid;
|
|
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)
|
|
goto no_session;
|
|
|
|
/* 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_get_request_pad (session->session, "send_rtcp_src");
|
|
if (session->send_rtcp_src == NULL)
|
|
goto pad_failed;
|
|
|
|
session->send_rtcp_src_ghost =
|
|
gst_ghost_pad_new_from_template (name, session->send_rtcp_src, templ);
|
|
gst_pad_set_active (session->send_rtcp_src_ghost, TRUE);
|
|
gst_element_add_pad (GST_ELEMENT_CAST (rtpbin), session->send_rtcp_src_ghost);
|
|
|
|
return session->send_rtcp_src_ghost;
|
|
|
|
/* ERRORS */
|
|
no_name:
|
|
{
|
|
g_warning ("rtpbin: invalid name given");
|
|
return NULL;
|
|
}
|
|
no_session:
|
|
{
|
|
g_warning ("rtpbin: session with id %d does not exist", sessid);
|
|
return NULL;
|
|
}
|
|
pad_failed:
|
|
{
|
|
g_warning ("rtpbin: failed to get rtcp pad for session %d", sessid);
|
|
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 posible 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_rtcp (rtpbin, templ, pad_name);
|
|
} else
|
|
goto wrong_template;
|
|
|
|
g_free (pad_name);
|
|
GST_RTP_BIN_UNLOCK (rtpbin);
|
|
|
|
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);
|
|
}
|
|
|
|
/* 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) {
|
|
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;
|
|
}
|
|
}
|