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SDP's are generated and consumed according to the W3C PeerConnection API available from https://www.w3.org/TR/webrtc/ The SDP is either created initially from the connected sink pads/attached transceivers as in the case of generating an offer or intersected with the connected sink pads/attached transceivers as in the case for creating an answer. In both cases, the rtp payloaded streams sent by the peer are exposed as separate src pads. The implementation supports trickle ICE, RTCP muxing, reduced size RTCP. With contributions from: Nirbheek Chauhan <nirbheek@centricular.com> Mathieu Duponchelle <mathieu@centricular.com> Edward Hervey <edward@centricular.com> https://bugzilla.gnome.org/show_bug.cgi?id=792523
376 lines
13 KiB
C
376 lines
13 KiB
C
/* GStreamer
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* Copyright (C) 2017 Matthew Waters <matthew@centricular.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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#ifdef HAVE_CONFIG_H
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# include "config.h"
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#endif
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#include "transportreceivebin.h"
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#include "utils.h"
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/*
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* ,----------------------------transport_receive_%u-----------------------------,
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* ; (rtp) ;
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* ; ,---nicesrc----, ,-capsfilter-, ,----dtlssrtpdec----, ,--funnel--, ;
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* ; ; src o--o sink src o--o sink rtp_src o------o sink_0 ; ;
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* ; '--------------' '------------' ; ; ; src o--o rtp_src
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* ; ; rtcp_src o-, ,--o sink_1 ; ;
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* ; '-------------------' ; ; '----------' ;
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* ; ; ; ,--funnel--, ;
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* ; '-+--o sink_0 ; ;
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* ; ,-' ; src o--o rtcp_src
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* ; (rtcp) ; ,-o sink_1 ; ;
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* ; ,---nicesrc----, ,-capsfilter-, ,----dtlssrtpdec----, ; ; '----------' ;
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* ; ; src o--o sink src o--o sink rtp_src o-' ; ;
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* ; '--------------' '------------' ; ; ; ;
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* ; ; rtcp_src o----' ;
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* ; '-------------------' ;
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* '-----------------------------------------------------------------------------'
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*
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* Do we really wnat to be *that* permissive in what we accept?
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*
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* FIXME: When and how do we want to clear the possibly stored buffers?
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*/
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#define GST_CAT_DEFAULT gst_webrtc_transport_receive_bin_debug
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GST_DEBUG_CATEGORY_STATIC (GST_CAT_DEFAULT);
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#define transport_receive_bin_parent_class parent_class
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G_DEFINE_TYPE_WITH_CODE (TransportReceiveBin, transport_receive_bin,
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GST_TYPE_BIN,
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GST_DEBUG_CATEGORY_INIT (gst_webrtc_transport_receive_bin_debug,
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"webrtctransportreceivebin", 0, "webrtctransportreceivebin");
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);
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static GstStaticPadTemplate rtp_sink_template =
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GST_STATIC_PAD_TEMPLATE ("rtp_src",
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GST_PAD_SINK,
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GST_PAD_ALWAYS,
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GST_STATIC_CAPS ("application/x-rtp"));
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static GstStaticPadTemplate rtcp_sink_template =
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GST_STATIC_PAD_TEMPLATE ("rtcp_src",
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GST_PAD_SINK,
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GST_PAD_ALWAYS,
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GST_STATIC_CAPS ("application/x-rtp"));
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enum
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{
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PROP_0,
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PROP_STREAM,
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};
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static const gchar *
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_receive_state_to_string (ReceiveState state)
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{
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switch (state) {
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case RECEIVE_STATE_BLOCK:
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return "block";
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case RECEIVE_STATE_DROP:
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return "drop";
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case RECEIVE_STATE_PASS:
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return "pass";
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default:
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return "Unknown";
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}
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}
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static GstPadProbeReturn
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pad_block (GstPad * pad, GstPadProbeInfo * info, TransportReceiveBin * receive)
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{
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GstPadProbeReturn ret;
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g_mutex_lock (&receive->pad_block_lock);
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while (receive->receive_state == RECEIVE_STATE_BLOCK) {
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g_cond_wait (&receive->pad_block_cond, &receive->pad_block_lock);
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GST_DEBUG_OBJECT (pad, "probe waited. new state %s",
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_receive_state_to_string (receive->receive_state));
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}
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ret = GST_PAD_PROBE_PASS;
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if (receive->receive_state == RECEIVE_STATE_DROP) {
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ret = GST_PAD_PROBE_DROP;
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} else if (receive->receive_state == RECEIVE_STATE_PASS) {
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ret = GST_PAD_PROBE_OK;
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}
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g_mutex_unlock (&receive->pad_block_lock);
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return ret;
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}
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void
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transport_receive_bin_set_receive_state (TransportReceiveBin * receive,
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ReceiveState state)
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{
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g_mutex_lock (&receive->pad_block_lock);
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receive->receive_state = state;
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GST_DEBUG_OBJECT (receive, "changing receive state to %s",
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_receive_state_to_string (state));
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g_cond_signal (&receive->pad_block_cond);
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g_mutex_unlock (&receive->pad_block_lock);
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}
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static void
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transport_receive_bin_set_property (GObject * object, guint prop_id,
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const GValue * value, GParamSpec * pspec)
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{
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TransportReceiveBin *receive = TRANSPORT_RECEIVE_BIN (object);
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GST_OBJECT_LOCK (receive);
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switch (prop_id) {
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case PROP_STREAM:
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/* XXX: weak-ref this? */
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receive->stream = TRANSPORT_STREAM (g_value_get_object (value));
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break;
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default:
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G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
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break;
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}
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GST_OBJECT_UNLOCK (receive);
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}
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static void
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transport_receive_bin_get_property (GObject * object, guint prop_id,
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GValue * value, GParamSpec * pspec)
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{
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TransportReceiveBin *receive = TRANSPORT_RECEIVE_BIN (object);
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GST_OBJECT_LOCK (receive);
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switch (prop_id) {
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case PROP_STREAM:
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g_value_set_object (value, receive->stream);
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break;
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default:
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G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
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break;
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}
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GST_OBJECT_UNLOCK (receive);
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}
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static void
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transport_receive_bin_finalize (GObject * object)
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{
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TransportReceiveBin *receive = TRANSPORT_RECEIVE_BIN (object);
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g_mutex_clear (&receive->pad_block_lock);
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g_cond_clear (&receive->pad_block_cond);
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G_OBJECT_CLASS (parent_class)->finalize (object);
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}
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static GstStateChangeReturn
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transport_receive_bin_change_state (GstElement * element,
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GstStateChange transition)
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{
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TransportReceiveBin *receive = TRANSPORT_RECEIVE_BIN (element);
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GstStateChangeReturn ret = GST_STATE_CHANGE_SUCCESS;
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GST_DEBUG ("changing state: %s => %s",
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gst_element_state_get_name (GST_STATE_TRANSITION_CURRENT (transition)),
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gst_element_state_get_name (GST_STATE_TRANSITION_NEXT (transition)));
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switch (transition) {
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case GST_STATE_CHANGE_NULL_TO_READY:{
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GstElement *elem;
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receive->rtp_block =
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_create_pad_block (GST_ELEMENT (receive), receive->rtp_src, 0, NULL,
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NULL);
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receive->rtp_block->block_id =
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gst_pad_add_probe (receive->rtp_src, GST_PAD_PROBE_TYPE_ALL_BOTH,
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(GstPadProbeCallback) pad_block, receive, NULL);
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/* XXX: because nice needs the nicesrc internal main loop running in order
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* correctly STUN... */
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/* FIXME: this races with the pad exposure later and may get not-linked */
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elem = receive->stream->transport->transport->src;
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gst_element_set_locked_state (elem, TRUE);
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gst_element_set_state (elem, GST_STATE_PLAYING);
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elem = receive->stream->rtcp_transport->transport->src;
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gst_element_set_locked_state (elem, TRUE);
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gst_element_set_state (elem, GST_STATE_PLAYING);
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break;
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}
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default:
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break;
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}
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ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
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if (ret == GST_STATE_CHANGE_FAILURE)
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return ret;
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switch (transition) {
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case GST_STATE_CHANGE_READY_TO_NULL:{
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GstElement *elem;
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elem = receive->stream->transport->transport->src;
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gst_element_set_locked_state (elem, FALSE);
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gst_element_set_state (elem, GST_STATE_NULL);
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elem = receive->stream->rtcp_transport->transport->src;
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gst_element_set_locked_state (elem, FALSE);
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gst_element_set_state (elem, GST_STATE_NULL);
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if (receive->rtp_block)
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_free_pad_block (receive->rtp_block);
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receive->rtp_block = NULL;
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break;
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}
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default:
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break;
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}
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return ret;
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}
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static void
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rtp_queue_overrun (GstElement * queue, TransportReceiveBin * receive)
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{
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GST_WARNING_OBJECT (receive, "Internal receive queue overrun. Dropping data");
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}
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static void
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transport_receive_bin_constructed (GObject * object)
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{
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TransportReceiveBin *receive = TRANSPORT_RECEIVE_BIN (object);
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GstWebRTCDTLSTransport *transport;
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GstPad *ghost, *pad;
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GstElement *capsfilter, *funnel, *queue;
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GstCaps *caps;
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g_return_if_fail (receive->stream);
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/* link ice src, dtlsrtp together for rtp */
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transport = receive->stream->transport;
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gst_bin_add (GST_BIN (receive), GST_ELEMENT (transport->dtlssrtpdec));
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capsfilter = gst_element_factory_make ("capsfilter", NULL);
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caps = gst_caps_new_empty_simple ("application/x-rtp");
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g_object_set (capsfilter, "caps", caps, NULL);
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gst_caps_unref (caps);
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gst_bin_add (GST_BIN (receive), GST_ELEMENT (capsfilter));
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if (!gst_element_link_pads (capsfilter, "src", transport->dtlssrtpdec,
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"sink"))
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g_warn_if_reached ();
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gst_bin_add (GST_BIN (receive), GST_ELEMENT (transport->transport->src));
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if (!gst_element_link_pads (GST_ELEMENT (transport->transport->src), "src",
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GST_ELEMENT (capsfilter), "sink"))
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g_warn_if_reached ();
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/* link ice src, dtlsrtp together for rtcp */
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transport = receive->stream->rtcp_transport;
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gst_bin_add (GST_BIN (receive), GST_ELEMENT (transport->dtlssrtpdec));
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capsfilter = gst_element_factory_make ("capsfilter", NULL);
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caps = gst_caps_new_empty_simple ("application/x-rtcp");
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g_object_set (capsfilter, "caps", caps, NULL);
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gst_caps_unref (caps);
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gst_bin_add (GST_BIN (receive), GST_ELEMENT (capsfilter));
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if (!gst_element_link_pads (capsfilter, "src", transport->dtlssrtpdec,
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"sink"))
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g_warn_if_reached ();
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gst_bin_add (GST_BIN (receive), GST_ELEMENT (transport->transport->src));
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if (!gst_element_link_pads (GST_ELEMENT (transport->transport->src), "src",
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GST_ELEMENT (capsfilter), "sink"))
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g_warn_if_reached ();
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/* create funnel for rtp_src */
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funnel = gst_element_factory_make ("funnel", NULL);
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gst_bin_add (GST_BIN (receive), funnel);
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if (!gst_element_link_pads (receive->stream->transport->dtlssrtpdec,
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"rtp_src", funnel, "sink_0"))
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g_warn_if_reached ();
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if (!gst_element_link_pads (receive->stream->rtcp_transport->dtlssrtpdec,
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"rtp_src", funnel, "sink_1"))
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g_warn_if_reached ();
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queue = gst_element_factory_make ("queue", NULL);
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/* FIXME: make this configurable? */
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g_object_set (queue, "leaky", 2, "max-size-time", (guint64) 0,
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"max-size-buffers", 0, "max-size-bytes", 5 * 1024 * 1024, NULL);
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g_signal_connect (queue, "overrun", G_CALLBACK (rtp_queue_overrun), receive);
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gst_bin_add (GST_BIN (receive), queue);
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if (!gst_element_link_pads (funnel, "src", queue, "sink"))
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g_warn_if_reached ();
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pad = gst_element_get_static_pad (queue, "src");
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receive->rtp_src = gst_ghost_pad_new ("rtp_src", pad);
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gst_element_add_pad (GST_ELEMENT (receive), receive->rtp_src);
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gst_object_unref (pad);
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/* create funnel for rtcp_src */
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funnel = gst_element_factory_make ("funnel", NULL);
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gst_bin_add (GST_BIN (receive), funnel);
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if (!gst_element_link_pads (receive->stream->transport->dtlssrtpdec,
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"rtcp_src", funnel, "sink_0"))
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g_warn_if_reached ();
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if (!gst_element_link_pads (receive->stream->rtcp_transport->dtlssrtpdec,
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"rtcp_src", funnel, "sink_1"))
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g_warn_if_reached ();
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pad = gst_element_get_static_pad (funnel, "src");
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ghost = gst_ghost_pad_new ("rtcp_src", pad);
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gst_element_add_pad (GST_ELEMENT (receive), ghost);
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gst_object_unref (pad);
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G_OBJECT_CLASS (parent_class)->constructed (object);
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}
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static void
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transport_receive_bin_class_init (TransportReceiveBinClass * klass)
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{
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GObjectClass *gobject_class = (GObjectClass *) klass;
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GstElementClass *element_class = (GstElementClass *) klass;
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element_class->change_state = transport_receive_bin_change_state;
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gst_element_class_add_static_pad_template (element_class, &rtp_sink_template);
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gst_element_class_add_static_pad_template (element_class,
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&rtcp_sink_template);
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gst_element_class_set_metadata (element_class, "WebRTC Transport Receive Bin",
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"Filter/Network/WebRTC", "A bin for webrtc connections",
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"Matthew Waters <matthew@centricular.com>");
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gobject_class->constructed = transport_receive_bin_constructed;
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gobject_class->get_property = transport_receive_bin_get_property;
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gobject_class->set_property = transport_receive_bin_set_property;
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gobject_class->finalize = transport_receive_bin_finalize;
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g_object_class_install_property (gobject_class,
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PROP_STREAM,
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g_param_spec_object ("stream", "Stream",
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"The TransportStream for this receiveing bin",
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transport_stream_get_type (),
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G_PARAM_READWRITE | G_PARAM_CONSTRUCT_ONLY | G_PARAM_STATIC_STRINGS));
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}
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static void
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transport_receive_bin_init (TransportReceiveBin * receive)
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{
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g_mutex_init (&receive->pad_block_lock);
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g_cond_init (&receive->pad_block_cond);
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}
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