e99c68dd21
The eventual goal is to replace sigslot entirely, but we need to start small, tread carefully, and evaluate how it works out. Also add a few more RoboCaller unit tests to cover the types we now use with RoboCaller. Change-Id: I9a5814d1668a37546ea484ca88ec9c2be1913d25 Bug: webrtc:11943 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/184660 Commit-Queue: Lahiru Ginnaliya Gamathige <glahiru@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#32266}
5130 lines
197 KiB
C++
5130 lines
197 KiB
C++
/*
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* Copyright 2012 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#include "pc/peer_connection.h"
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#include <algorithm>
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#include <limits>
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#include <memory>
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#include <queue>
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#include <set>
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#include <utility>
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#include <vector>
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#include "absl/algorithm/container.h"
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#include "absl/strings/match.h"
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#include "api/jsep_ice_candidate.h"
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#include "api/jsep_session_description.h"
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#include "api/media_stream_proxy.h"
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#include "api/media_stream_track_proxy.h"
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#include "api/rtc_error.h"
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#include "api/rtc_event_log/rtc_event_log.h"
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#include "api/rtc_event_log_output_file.h"
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#include "api/rtp_parameters.h"
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#include "api/uma_metrics.h"
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#include "api/video/builtin_video_bitrate_allocator_factory.h"
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#include "call/call.h"
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#include "logging/rtc_event_log/ice_logger.h"
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#include "media/base/rid_description.h"
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#include "media/sctp/sctp_transport.h"
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#include "pc/audio_rtp_receiver.h"
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#include "pc/audio_track.h"
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#include "pc/channel.h"
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#include "pc/channel_manager.h"
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#include "pc/dtmf_sender.h"
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#include "pc/media_stream.h"
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#include "pc/media_stream_observer.h"
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#include "pc/remote_audio_source.h"
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#include "pc/rtp_media_utils.h"
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#include "pc/rtp_receiver.h"
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#include "pc/rtp_sender.h"
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#include "pc/sctp_transport.h"
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#include "pc/sctp_utils.h"
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#include "pc/sdp_offer_answer.h"
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#include "pc/sdp_utils.h"
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#include "pc/stream_collection.h"
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#include "pc/video_rtp_receiver.h"
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#include "pc/video_track.h"
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#include "rtc_base/bind.h"
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#include "rtc_base/checks.h"
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#include "rtc_base/logging.h"
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#include "rtc_base/string_encode.h"
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#include "rtc_base/strings/string_builder.h"
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#include "rtc_base/task_utils/to_queued_task.h"
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#include "rtc_base/trace_event.h"
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#include "system_wrappers/include/clock.h"
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#include "system_wrappers/include/metrics.h"
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using cricket::ContentInfo;
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using cricket::ContentInfos;
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using cricket::MediaContentDescription;
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using cricket::MediaProtocolType;
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using cricket::RidDescription;
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using cricket::RidDirection;
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using cricket::SessionDescription;
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using cricket::SimulcastDescription;
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using cricket::SimulcastLayer;
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using cricket::SimulcastLayerList;
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using cricket::StreamParams;
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using cricket::TransportInfo;
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using cricket::LOCAL_PORT_TYPE;
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using cricket::PRFLX_PORT_TYPE;
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using cricket::RELAY_PORT_TYPE;
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using cricket::STUN_PORT_TYPE;
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namespace webrtc {
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// Error messages
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const char kSessionError[] = "Session error code: ";
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const char kSessionErrorDesc[] = "Session error description: ";
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namespace {
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// UMA metric names.
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const char kSimulcastNumberOfEncodings[] =
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"WebRTC.PeerConnection.Simulcast.NumberOfSendEncodings";
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static const char kDefaultStreamId[] = "default";
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static const char kDefaultAudioSenderId[] = "defaulta0";
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static const char kDefaultVideoSenderId[] = "defaultv0";
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// The length of RTCP CNAMEs.
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static const int kRtcpCnameLength = 16;
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enum {
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MSG_SET_SESSIONDESCRIPTION_SUCCESS = 0,
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MSG_SET_SESSIONDESCRIPTION_FAILED,
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MSG_CREATE_SESSIONDESCRIPTION_FAILED,
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MSG_GETSTATS,
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MSG_REPORT_USAGE_PATTERN,
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};
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static const int REPORT_USAGE_PATTERN_DELAY_MS = 60000;
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struct SetSessionDescriptionMsg : public rtc::MessageData {
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explicit SetSessionDescriptionMsg(
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webrtc::SetSessionDescriptionObserver* observer)
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: observer(observer) {}
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rtc::scoped_refptr<webrtc::SetSessionDescriptionObserver> observer;
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RTCError error;
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};
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struct CreateSessionDescriptionMsg : public rtc::MessageData {
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explicit CreateSessionDescriptionMsg(
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webrtc::CreateSessionDescriptionObserver* observer)
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: observer(observer) {}
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rtc::scoped_refptr<webrtc::CreateSessionDescriptionObserver> observer;
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RTCError error;
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};
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struct GetStatsMsg : public rtc::MessageData {
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GetStatsMsg(webrtc::StatsObserver* observer,
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webrtc::MediaStreamTrackInterface* track)
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: observer(observer), track(track) {}
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rtc::scoped_refptr<webrtc::StatsObserver> observer;
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rtc::scoped_refptr<webrtc::MediaStreamTrackInterface> track;
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};
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// Check if we can send |new_stream| on a PeerConnection.
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bool CanAddLocalMediaStream(webrtc::StreamCollectionInterface* current_streams,
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webrtc::MediaStreamInterface* new_stream) {
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if (!new_stream || !current_streams) {
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return false;
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}
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if (current_streams->find(new_stream->id()) != nullptr) {
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RTC_LOG(LS_ERROR) << "MediaStream with ID " << new_stream->id()
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<< " is already added.";
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return false;
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}
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return true;
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}
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// Add options to |[audio/video]_media_description_options| from |senders|.
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void AddPlanBRtpSenderOptions(
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const std::vector<rtc::scoped_refptr<
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RtpSenderProxyWithInternal<RtpSenderInternal>>>& senders,
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cricket::MediaDescriptionOptions* audio_media_description_options,
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cricket::MediaDescriptionOptions* video_media_description_options,
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int num_sim_layers) {
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for (const auto& sender : senders) {
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if (sender->media_type() == cricket::MEDIA_TYPE_AUDIO) {
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if (audio_media_description_options) {
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audio_media_description_options->AddAudioSender(
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sender->id(), sender->internal()->stream_ids());
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}
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} else {
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RTC_DCHECK(sender->media_type() == cricket::MEDIA_TYPE_VIDEO);
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if (video_media_description_options) {
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video_media_description_options->AddVideoSender(
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sender->id(), sender->internal()->stream_ids(), {},
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SimulcastLayerList(), num_sim_layers);
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}
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}
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}
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}
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// Add options to |session_options| from |rtp_data_channels|.
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void AddRtpDataChannelOptions(
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const std::map<std::string, rtc::scoped_refptr<RtpDataChannel>>&
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rtp_data_channels,
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cricket::MediaDescriptionOptions* data_media_description_options) {
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if (!data_media_description_options) {
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return;
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}
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// Check for data channels.
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for (const auto& kv : rtp_data_channels) {
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const RtpDataChannel* channel = kv.second;
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if (channel->state() == RtpDataChannel::kConnecting ||
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channel->state() == RtpDataChannel::kOpen) {
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// Legacy RTP data channels are signaled with the track/stream ID set to
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// the data channel's label.
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data_media_description_options->AddRtpDataChannel(channel->label(),
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channel->label());
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}
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}
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}
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uint32_t ConvertIceTransportTypeToCandidateFilter(
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PeerConnectionInterface::IceTransportsType type) {
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switch (type) {
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case PeerConnectionInterface::kNone:
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return cricket::CF_NONE;
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case PeerConnectionInterface::kRelay:
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return cricket::CF_RELAY;
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case PeerConnectionInterface::kNoHost:
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return (cricket::CF_ALL & ~cricket::CF_HOST);
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case PeerConnectionInterface::kAll:
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return cricket::CF_ALL;
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default:
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RTC_NOTREACHED();
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}
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return cricket::CF_NONE;
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}
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IceCandidatePairType GetIceCandidatePairCounter(
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const cricket::Candidate& local,
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const cricket::Candidate& remote) {
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const auto& l = local.type();
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const auto& r = remote.type();
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const auto& host = LOCAL_PORT_TYPE;
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const auto& srflx = STUN_PORT_TYPE;
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const auto& relay = RELAY_PORT_TYPE;
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const auto& prflx = PRFLX_PORT_TYPE;
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if (l == host && r == host) {
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bool local_hostname =
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!local.address().hostname().empty() && local.address().IsUnresolvedIP();
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bool remote_hostname = !remote.address().hostname().empty() &&
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remote.address().IsUnresolvedIP();
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bool local_private = IPIsPrivate(local.address().ipaddr());
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bool remote_private = IPIsPrivate(remote.address().ipaddr());
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if (local_hostname) {
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if (remote_hostname) {
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return kIceCandidatePairHostNameHostName;
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} else if (remote_private) {
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return kIceCandidatePairHostNameHostPrivate;
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} else {
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return kIceCandidatePairHostNameHostPublic;
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}
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} else if (local_private) {
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if (remote_hostname) {
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return kIceCandidatePairHostPrivateHostName;
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} else if (remote_private) {
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return kIceCandidatePairHostPrivateHostPrivate;
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} else {
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return kIceCandidatePairHostPrivateHostPublic;
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}
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} else {
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if (remote_hostname) {
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return kIceCandidatePairHostPublicHostName;
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} else if (remote_private) {
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return kIceCandidatePairHostPublicHostPrivate;
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} else {
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return kIceCandidatePairHostPublicHostPublic;
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}
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}
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}
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if (l == host && r == srflx)
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return kIceCandidatePairHostSrflx;
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if (l == host && r == relay)
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return kIceCandidatePairHostRelay;
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if (l == host && r == prflx)
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return kIceCandidatePairHostPrflx;
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if (l == srflx && r == host)
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return kIceCandidatePairSrflxHost;
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if (l == srflx && r == srflx)
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return kIceCandidatePairSrflxSrflx;
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if (l == srflx && r == relay)
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return kIceCandidatePairSrflxRelay;
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if (l == srflx && r == prflx)
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return kIceCandidatePairSrflxPrflx;
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if (l == relay && r == host)
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return kIceCandidatePairRelayHost;
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if (l == relay && r == srflx)
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return kIceCandidatePairRelaySrflx;
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if (l == relay && r == relay)
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return kIceCandidatePairRelayRelay;
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if (l == relay && r == prflx)
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return kIceCandidatePairRelayPrflx;
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if (l == prflx && r == host)
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return kIceCandidatePairPrflxHost;
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if (l == prflx && r == srflx)
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return kIceCandidatePairPrflxSrflx;
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if (l == prflx && r == relay)
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return kIceCandidatePairPrflxRelay;
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return kIceCandidatePairMax;
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}
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absl::optional<int> RTCConfigurationToIceConfigOptionalInt(
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int rtc_configuration_parameter) {
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if (rtc_configuration_parameter ==
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webrtc::PeerConnectionInterface::RTCConfiguration::kUndefined) {
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return absl::nullopt;
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}
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return rtc_configuration_parameter;
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}
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// Check if the changes of IceTransportsType motives an ice restart.
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bool NeedIceRestart(bool surface_ice_candidates_on_ice_transport_type_changed,
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PeerConnectionInterface::IceTransportsType current,
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PeerConnectionInterface::IceTransportsType modified) {
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if (current == modified) {
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return false;
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}
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if (!surface_ice_candidates_on_ice_transport_type_changed) {
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return true;
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}
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auto current_filter = ConvertIceTransportTypeToCandidateFilter(current);
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auto modified_filter = ConvertIceTransportTypeToCandidateFilter(modified);
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// If surface_ice_candidates_on_ice_transport_type_changed is true and we
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// extend the filter, then no ice restart is needed.
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return (current_filter & modified_filter) != current_filter;
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}
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} // namespace
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bool PeerConnectionInterface::RTCConfiguration::operator==(
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const PeerConnectionInterface::RTCConfiguration& o) const {
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// This static_assert prevents us from accidentally breaking operator==.
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// Note: Order matters! Fields must be ordered the same as RTCConfiguration.
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struct stuff_being_tested_for_equality {
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IceServers servers;
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IceTransportsType type;
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BundlePolicy bundle_policy;
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RtcpMuxPolicy rtcp_mux_policy;
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std::vector<rtc::scoped_refptr<rtc::RTCCertificate>> certificates;
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int ice_candidate_pool_size;
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bool disable_ipv6;
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bool disable_ipv6_on_wifi;
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int max_ipv6_networks;
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bool disable_link_local_networks;
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bool enable_rtp_data_channel;
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absl::optional<int> screencast_min_bitrate;
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absl::optional<bool> combined_audio_video_bwe;
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absl::optional<bool> enable_dtls_srtp;
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TcpCandidatePolicy tcp_candidate_policy;
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CandidateNetworkPolicy candidate_network_policy;
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int audio_jitter_buffer_max_packets;
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bool audio_jitter_buffer_fast_accelerate;
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int audio_jitter_buffer_min_delay_ms;
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bool audio_jitter_buffer_enable_rtx_handling;
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int ice_connection_receiving_timeout;
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int ice_backup_candidate_pair_ping_interval;
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ContinualGatheringPolicy continual_gathering_policy;
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bool prioritize_most_likely_ice_candidate_pairs;
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struct cricket::MediaConfig media_config;
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bool prune_turn_ports;
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PortPrunePolicy turn_port_prune_policy;
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bool presume_writable_when_fully_relayed;
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bool enable_ice_renomination;
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bool redetermine_role_on_ice_restart;
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bool surface_ice_candidates_on_ice_transport_type_changed;
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absl::optional<int> ice_check_interval_strong_connectivity;
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absl::optional<int> ice_check_interval_weak_connectivity;
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absl::optional<int> ice_check_min_interval;
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absl::optional<int> ice_unwritable_timeout;
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absl::optional<int> ice_unwritable_min_checks;
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absl::optional<int> ice_inactive_timeout;
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absl::optional<int> stun_candidate_keepalive_interval;
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webrtc::TurnCustomizer* turn_customizer;
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SdpSemantics sdp_semantics;
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absl::optional<rtc::AdapterType> network_preference;
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bool active_reset_srtp_params;
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absl::optional<CryptoOptions> crypto_options;
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bool offer_extmap_allow_mixed;
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std::string turn_logging_id;
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bool enable_implicit_rollback;
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absl::optional<bool> allow_codec_switching;
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};
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static_assert(sizeof(stuff_being_tested_for_equality) == sizeof(*this),
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"Did you add something to RTCConfiguration and forget to "
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"update operator==?");
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return type == o.type && servers == o.servers &&
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bundle_policy == o.bundle_policy &&
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rtcp_mux_policy == o.rtcp_mux_policy &&
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tcp_candidate_policy == o.tcp_candidate_policy &&
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candidate_network_policy == o.candidate_network_policy &&
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audio_jitter_buffer_max_packets == o.audio_jitter_buffer_max_packets &&
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audio_jitter_buffer_fast_accelerate ==
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o.audio_jitter_buffer_fast_accelerate &&
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audio_jitter_buffer_min_delay_ms ==
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o.audio_jitter_buffer_min_delay_ms &&
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audio_jitter_buffer_enable_rtx_handling ==
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o.audio_jitter_buffer_enable_rtx_handling &&
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ice_connection_receiving_timeout ==
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o.ice_connection_receiving_timeout &&
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ice_backup_candidate_pair_ping_interval ==
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o.ice_backup_candidate_pair_ping_interval &&
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continual_gathering_policy == o.continual_gathering_policy &&
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certificates == o.certificates &&
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prioritize_most_likely_ice_candidate_pairs ==
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o.prioritize_most_likely_ice_candidate_pairs &&
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media_config == o.media_config && disable_ipv6 == o.disable_ipv6 &&
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disable_ipv6_on_wifi == o.disable_ipv6_on_wifi &&
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max_ipv6_networks == o.max_ipv6_networks &&
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disable_link_local_networks == o.disable_link_local_networks &&
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enable_rtp_data_channel == o.enable_rtp_data_channel &&
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screencast_min_bitrate == o.screencast_min_bitrate &&
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combined_audio_video_bwe == o.combined_audio_video_bwe &&
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enable_dtls_srtp == o.enable_dtls_srtp &&
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ice_candidate_pool_size == o.ice_candidate_pool_size &&
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prune_turn_ports == o.prune_turn_ports &&
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turn_port_prune_policy == o.turn_port_prune_policy &&
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presume_writable_when_fully_relayed ==
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o.presume_writable_when_fully_relayed &&
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enable_ice_renomination == o.enable_ice_renomination &&
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redetermine_role_on_ice_restart == o.redetermine_role_on_ice_restart &&
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surface_ice_candidates_on_ice_transport_type_changed ==
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o.surface_ice_candidates_on_ice_transport_type_changed &&
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ice_check_interval_strong_connectivity ==
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o.ice_check_interval_strong_connectivity &&
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ice_check_interval_weak_connectivity ==
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o.ice_check_interval_weak_connectivity &&
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ice_check_min_interval == o.ice_check_min_interval &&
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ice_unwritable_timeout == o.ice_unwritable_timeout &&
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ice_unwritable_min_checks == o.ice_unwritable_min_checks &&
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ice_inactive_timeout == o.ice_inactive_timeout &&
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stun_candidate_keepalive_interval ==
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o.stun_candidate_keepalive_interval &&
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turn_customizer == o.turn_customizer &&
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sdp_semantics == o.sdp_semantics &&
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network_preference == o.network_preference &&
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active_reset_srtp_params == o.active_reset_srtp_params &&
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crypto_options == o.crypto_options &&
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offer_extmap_allow_mixed == o.offer_extmap_allow_mixed &&
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turn_logging_id == o.turn_logging_id &&
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enable_implicit_rollback == o.enable_implicit_rollback &&
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allow_codec_switching == o.allow_codec_switching;
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}
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bool PeerConnectionInterface::RTCConfiguration::operator!=(
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const PeerConnectionInterface::RTCConfiguration& o) const {
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return !(*this == o);
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}
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|
|
// Generate a RTCP CNAME when a PeerConnection is created.
|
|
std::string GenerateRtcpCname() {
|
|
std::string cname;
|
|
if (!rtc::CreateRandomString(kRtcpCnameLength, &cname)) {
|
|
RTC_LOG(LS_ERROR) << "Failed to generate CNAME.";
|
|
RTC_NOTREACHED();
|
|
}
|
|
return cname;
|
|
}
|
|
|
|
// From |rtc_options|, fill parts of |session_options| shared by all generated
|
|
// m= sectionss (in other words, nothing that involves a map/array).
|
|
void ExtractSharedMediaSessionOptions(
|
|
const PeerConnectionInterface::RTCOfferAnswerOptions& rtc_options,
|
|
cricket::MediaSessionOptions* session_options) {
|
|
session_options->vad_enabled = rtc_options.voice_activity_detection;
|
|
session_options->bundle_enabled = rtc_options.use_rtp_mux;
|
|
session_options->raw_packetization_for_video =
|
|
rtc_options.raw_packetization_for_video;
|
|
}
|
|
|
|
PeerConnection::PeerConnection(PeerConnectionFactory* factory,
|
|
std::unique_ptr<RtcEventLog> event_log,
|
|
std::unique_ptr<Call> call)
|
|
: factory_(factory),
|
|
event_log_(std::move(event_log)),
|
|
event_log_ptr_(event_log_.get()),
|
|
rtcp_cname_(GenerateRtcpCname()),
|
|
local_streams_(StreamCollection::Create()),
|
|
remote_streams_(StreamCollection::Create()),
|
|
call_(std::move(call)),
|
|
call_ptr_(call_.get()),
|
|
sdp_handler_(this),
|
|
data_channel_controller_(this) {}
|
|
|
|
PeerConnection::~PeerConnection() {
|
|
TRACE_EVENT0("webrtc", "PeerConnection::~PeerConnection");
|
|
RTC_DCHECK_RUN_ON(signaling_thread());
|
|
|
|
sdp_handler_.PrepareForShutdown();
|
|
|
|
// Need to stop transceivers before destroying the stats collector because
|
|
// AudioRtpSender has a reference to the StatsCollector it will update when
|
|
// stopping.
|
|
for (const auto& transceiver : transceivers_.List()) {
|
|
transceiver->StopInternal();
|
|
}
|
|
|
|
stats_.reset(nullptr);
|
|
if (stats_collector_) {
|
|
stats_collector_->WaitForPendingRequest();
|
|
stats_collector_ = nullptr;
|
|
}
|
|
|
|
// Don't destroy BaseChannels until after stats has been cleaned up so that
|
|
// the last stats request can still read from the channels.
|
|
DestroyAllChannels();
|
|
|
|
RTC_LOG(LS_INFO) << "Session: " << session_id() << " is destroyed.";
|
|
|
|
sdp_handler_.ResetSessionDescFactory();
|
|
transport_controller_.reset();
|
|
|
|
// port_allocator_ lives on the network thread and should be destroyed there.
|
|
network_thread()->Invoke<void>(RTC_FROM_HERE, [this] {
|
|
RTC_DCHECK_RUN_ON(network_thread());
|
|
port_allocator_.reset();
|
|
});
|
|
// call_ and event_log_ must be destroyed on the worker thread.
|
|
worker_thread()->Invoke<void>(RTC_FROM_HERE, [this] {
|
|
RTC_DCHECK_RUN_ON(worker_thread());
|
|
call_safety_.reset();
|
|
call_.reset();
|
|
// The event log must outlive call (and any other object that uses it).
|
|
event_log_.reset();
|
|
});
|
|
|
|
// Process all pending notifications in the message queue. If we don't do
|
|
// this, requests will linger and not know they succeeded or failed.
|
|
rtc::MessageList list;
|
|
signaling_thread()->Clear(this, rtc::MQID_ANY, &list);
|
|
for (auto& msg : list) {
|
|
if (msg.message_id == MSG_CREATE_SESSIONDESCRIPTION_FAILED) {
|
|
// Processing CreateOffer() and CreateAnswer() messages ensures their
|
|
// observers are invoked even if the PeerConnection is destroyed early.
|
|
OnMessage(&msg);
|
|
} else {
|
|
// TODO(hbos): Consider processing all pending messages. This would mean
|
|
// that SetLocalDescription() and SetRemoteDescription() observers are
|
|
// informed of successes and failures; this is currently NOT the case.
|
|
delete msg.pdata;
|
|
}
|
|
}
|
|
}
|
|
|
|
void PeerConnection::DestroyAllChannels() {
|
|
// Destroy video channels first since they may have a pointer to a voice
|
|
// channel.
|
|
for (const auto& transceiver : transceivers_.List()) {
|
|
if (transceiver->media_type() == cricket::MEDIA_TYPE_VIDEO) {
|
|
DestroyTransceiverChannel(transceiver);
|
|
}
|
|
}
|
|
for (const auto& transceiver : transceivers_.List()) {
|
|
if (transceiver->media_type() == cricket::MEDIA_TYPE_AUDIO) {
|
|
DestroyTransceiverChannel(transceiver);
|
|
}
|
|
}
|
|
DestroyDataChannelTransport();
|
|
}
|
|
|
|
bool PeerConnection::Initialize(
|
|
const PeerConnectionInterface::RTCConfiguration& configuration,
|
|
PeerConnectionDependencies dependencies) {
|
|
RTC_DCHECK_RUN_ON(signaling_thread());
|
|
TRACE_EVENT0("webrtc", "PeerConnection::Initialize");
|
|
|
|
RTCError config_error = ValidateConfiguration(configuration);
|
|
if (!config_error.ok()) {
|
|
RTC_LOG(LS_ERROR) << "Invalid configuration: " << config_error.message();
|
|
return false;
|
|
}
|
|
|
|
if (!dependencies.allocator) {
|
|
RTC_LOG(LS_ERROR)
|
|
<< "PeerConnection initialized without a PortAllocator? "
|
|
"This shouldn't happen if using PeerConnectionFactory.";
|
|
return false;
|
|
}
|
|
|
|
if (!dependencies.observer) {
|
|
// TODO(deadbeef): Why do we do this?
|
|
RTC_LOG(LS_ERROR) << "PeerConnection initialized without a "
|
|
"PeerConnectionObserver";
|
|
return false;
|
|
}
|
|
|
|
observer_ = dependencies.observer;
|
|
async_resolver_factory_ = std::move(dependencies.async_resolver_factory);
|
|
port_allocator_ = std::move(dependencies.allocator);
|
|
packet_socket_factory_ = std::move(dependencies.packet_socket_factory);
|
|
ice_transport_factory_ = std::move(dependencies.ice_transport_factory);
|
|
tls_cert_verifier_ = std::move(dependencies.tls_cert_verifier);
|
|
|
|
cricket::ServerAddresses stun_servers;
|
|
std::vector<cricket::RelayServerConfig> turn_servers;
|
|
|
|
RTCErrorType parse_error =
|
|
ParseIceServers(configuration.servers, &stun_servers, &turn_servers);
|
|
if (parse_error != RTCErrorType::NONE) {
|
|
return false;
|
|
}
|
|
|
|
// Add the turn logging id to all turn servers
|
|
for (cricket::RelayServerConfig& turn_server : turn_servers) {
|
|
turn_server.turn_logging_id = configuration.turn_logging_id;
|
|
}
|
|
|
|
// The port allocator lives on the network thread and should be initialized
|
|
// there.
|
|
const auto pa_result =
|
|
network_thread()->Invoke<InitializePortAllocatorResult>(
|
|
RTC_FROM_HERE,
|
|
rtc::Bind(&PeerConnection::InitializePortAllocator_n, this,
|
|
stun_servers, turn_servers, configuration));
|
|
|
|
// If initialization was successful, note if STUN or TURN servers
|
|
// were supplied.
|
|
if (!stun_servers.empty()) {
|
|
NoteUsageEvent(UsageEvent::STUN_SERVER_ADDED);
|
|
}
|
|
if (!turn_servers.empty()) {
|
|
NoteUsageEvent(UsageEvent::TURN_SERVER_ADDED);
|
|
}
|
|
|
|
// Send information about IPv4/IPv6 status.
|
|
PeerConnectionAddressFamilyCounter address_family;
|
|
if (pa_result.enable_ipv6) {
|
|
address_family = kPeerConnection_IPv6;
|
|
} else {
|
|
address_family = kPeerConnection_IPv4;
|
|
}
|
|
RTC_HISTOGRAM_ENUMERATION("WebRTC.PeerConnection.IPMetrics", address_family,
|
|
kPeerConnectionAddressFamilyCounter_Max);
|
|
|
|
const PeerConnectionFactoryInterface::Options& options = factory_->options();
|
|
|
|
// RFC 3264: The numeric value of the session id and version in the
|
|
// o line MUST be representable with a "64 bit signed integer".
|
|
// Due to this constraint session id |session_id_| is max limited to
|
|
// LLONG_MAX.
|
|
session_id_ = rtc::ToString(rtc::CreateRandomId64() & LLONG_MAX);
|
|
JsepTransportController::Config config;
|
|
config.redetermine_role_on_ice_restart =
|
|
configuration.redetermine_role_on_ice_restart;
|
|
config.ssl_max_version = factory_->options().ssl_max_version;
|
|
config.disable_encryption = options.disable_encryption;
|
|
config.bundle_policy = configuration.bundle_policy;
|
|
config.rtcp_mux_policy = configuration.rtcp_mux_policy;
|
|
// TODO(bugs.webrtc.org/9891) - Remove options.crypto_options then remove this
|
|
// stub.
|
|
config.crypto_options = configuration.crypto_options.has_value()
|
|
? *configuration.crypto_options
|
|
: options.crypto_options;
|
|
config.transport_observer = this;
|
|
config.rtcp_handler = InitializeRtcpCallback();
|
|
config.event_log = event_log_ptr_;
|
|
#if defined(ENABLE_EXTERNAL_AUTH)
|
|
config.enable_external_auth = true;
|
|
#endif
|
|
config.active_reset_srtp_params = configuration.active_reset_srtp_params;
|
|
|
|
// Obtain a certificate from RTCConfiguration if any were provided (optional).
|
|
rtc::scoped_refptr<rtc::RTCCertificate> certificate;
|
|
if (!configuration.certificates.empty()) {
|
|
// TODO(hbos,torbjorng): Decide on certificate-selection strategy instead of
|
|
// just picking the first one. The decision should be made based on the DTLS
|
|
// handshake. The DTLS negotiations need to know about all certificates.
|
|
certificate = configuration.certificates[0];
|
|
}
|
|
|
|
if (options.disable_encryption) {
|
|
dtls_enabled_ = false;
|
|
} else {
|
|
// Enable DTLS by default if we have an identity store or a certificate.
|
|
dtls_enabled_ = (dependencies.cert_generator || certificate);
|
|
// |configuration| can override the default |dtls_enabled_| value.
|
|
if (configuration.enable_dtls_srtp) {
|
|
dtls_enabled_ = *(configuration.enable_dtls_srtp);
|
|
}
|
|
}
|
|
|
|
if (configuration.enable_rtp_data_channel) {
|
|
// Enable creation of RTP data channels if the kEnableRtpDataChannels is
|
|
// set. It takes precendence over the disable_sctp_data_channels
|
|
// PeerConnectionFactoryInterface::Options.
|
|
data_channel_controller_.set_data_channel_type(cricket::DCT_RTP);
|
|
} else {
|
|
// DTLS has to be enabled to use SCTP.
|
|
if (!options.disable_sctp_data_channels && dtls_enabled_) {
|
|
data_channel_controller_.set_data_channel_type(cricket::DCT_SCTP);
|
|
config.sctp_factory = factory_->sctp_transport_factory();
|
|
}
|
|
}
|
|
|
|
config.ice_transport_factory = ice_transport_factory_.get();
|
|
|
|
transport_controller_.reset(new JsepTransportController(
|
|
signaling_thread(), network_thread(), port_allocator_.get(),
|
|
async_resolver_factory_.get(), config));
|
|
transport_controller_->SignalStandardizedIceConnectionState.connect(
|
|
this, &PeerConnection::SetStandardizedIceConnectionState);
|
|
transport_controller_->SignalConnectionState.connect(
|
|
this, &PeerConnection::SetConnectionState);
|
|
transport_controller_->SignalIceGatheringState.connect(
|
|
this, &PeerConnection::OnTransportControllerGatheringState);
|
|
transport_controller_->SignalIceCandidatesGathered.connect(
|
|
this, &PeerConnection::OnTransportControllerCandidatesGathered);
|
|
transport_controller_->SignalIceCandidateError.connect(
|
|
this, &PeerConnection::OnTransportControllerCandidateError);
|
|
transport_controller_->SignalIceCandidatesRemoved.connect(
|
|
this, &PeerConnection::OnTransportControllerCandidatesRemoved);
|
|
transport_controller_->SignalDtlsHandshakeError.connect(
|
|
this, &PeerConnection::OnTransportControllerDtlsHandshakeError);
|
|
transport_controller_->SignalIceCandidatePairChanged.connect(
|
|
this, &PeerConnection::OnTransportControllerCandidateChanged);
|
|
|
|
transport_controller_->SignalIceConnectionState.AddReceiver(
|
|
[this](cricket::IceConnectionState s) {
|
|
RTC_DCHECK_RUN_ON(signaling_thread());
|
|
OnTransportControllerConnectionState(s);
|
|
});
|
|
|
|
stats_.reset(new StatsCollector(this));
|
|
stats_collector_ = RTCStatsCollector::Create(this);
|
|
|
|
configuration_ = configuration;
|
|
|
|
transport_controller_->SetIceConfig(ParseIceConfig(configuration));
|
|
|
|
video_options_.screencast_min_bitrate_kbps =
|
|
configuration.screencast_min_bitrate;
|
|
audio_options_.combined_audio_video_bwe =
|
|
configuration.combined_audio_video_bwe;
|
|
|
|
audio_options_.audio_jitter_buffer_max_packets =
|
|
configuration.audio_jitter_buffer_max_packets;
|
|
|
|
audio_options_.audio_jitter_buffer_fast_accelerate =
|
|
configuration.audio_jitter_buffer_fast_accelerate;
|
|
|
|
audio_options_.audio_jitter_buffer_min_delay_ms =
|
|
configuration.audio_jitter_buffer_min_delay_ms;
|
|
|
|
audio_options_.audio_jitter_buffer_enable_rtx_handling =
|
|
configuration.audio_jitter_buffer_enable_rtx_handling;
|
|
|
|
// Whether the certificate generator/certificate is null or not determines
|
|
// what PeerConnectionDescriptionFactory will do, so make sure that we give it
|
|
// the right instructions by clearing the variables if needed.
|
|
if (!dtls_enabled_) {
|
|
dependencies.cert_generator.reset();
|
|
certificate = nullptr;
|
|
} else if (certificate) {
|
|
// Favor generated certificate over the certificate generator.
|
|
dependencies.cert_generator.reset();
|
|
}
|
|
|
|
auto webrtc_session_desc_factory =
|
|
std::make_unique<WebRtcSessionDescriptionFactory>(
|
|
signaling_thread(), channel_manager(), this, session_id(),
|
|
std::move(dependencies.cert_generator), certificate,
|
|
&ssrc_generator_);
|
|
webrtc_session_desc_factory->SignalCertificateReady.connect(
|
|
this, &PeerConnection::OnCertificateReady);
|
|
|
|
if (options.disable_encryption) {
|
|
webrtc_session_desc_factory->SetSdesPolicy(cricket::SEC_DISABLED);
|
|
}
|
|
|
|
webrtc_session_desc_factory->set_enable_encrypted_rtp_header_extensions(
|
|
GetCryptoOptions().srtp.enable_encrypted_rtp_header_extensions);
|
|
webrtc_session_desc_factory->set_is_unified_plan(IsUnifiedPlan());
|
|
sdp_handler_.SetSessionDescFactory(std::move(webrtc_session_desc_factory));
|
|
|
|
// Add default audio/video transceivers for Plan B SDP.
|
|
if (!IsUnifiedPlan()) {
|
|
transceivers_.Add(RtpTransceiverProxyWithInternal<RtpTransceiver>::Create(
|
|
signaling_thread(), new RtpTransceiver(cricket::MEDIA_TYPE_AUDIO)));
|
|
transceivers_.Add(RtpTransceiverProxyWithInternal<RtpTransceiver>::Create(
|
|
signaling_thread(), new RtpTransceiver(cricket::MEDIA_TYPE_VIDEO)));
|
|
}
|
|
int delay_ms =
|
|
return_histogram_very_quickly_ ? 0 : REPORT_USAGE_PATTERN_DELAY_MS;
|
|
signaling_thread()->PostDelayed(RTC_FROM_HERE, delay_ms, this,
|
|
MSG_REPORT_USAGE_PATTERN, nullptr);
|
|
|
|
if (dependencies.video_bitrate_allocator_factory) {
|
|
video_bitrate_allocator_factory_ =
|
|
std::move(dependencies.video_bitrate_allocator_factory);
|
|
} else {
|
|
video_bitrate_allocator_factory_ =
|
|
CreateBuiltinVideoBitrateAllocatorFactory();
|
|
}
|
|
return true;
|
|
}
|
|
|
|
RTCError PeerConnection::ValidateConfiguration(
|
|
const RTCConfiguration& config) const {
|
|
return cricket::P2PTransportChannel::ValidateIceConfig(
|
|
ParseIceConfig(config));
|
|
}
|
|
|
|
rtc::scoped_refptr<StreamCollectionInterface> PeerConnection::local_streams() {
|
|
RTC_DCHECK_RUN_ON(signaling_thread());
|
|
RTC_CHECK(!IsUnifiedPlan()) << "local_streams is not available with Unified "
|
|
"Plan SdpSemantics. Please use GetSenders "
|
|
"instead.";
|
|
return local_streams_;
|
|
}
|
|
|
|
rtc::scoped_refptr<StreamCollectionInterface> PeerConnection::remote_streams() {
|
|
RTC_DCHECK_RUN_ON(signaling_thread());
|
|
RTC_CHECK(!IsUnifiedPlan()) << "remote_streams is not available with Unified "
|
|
"Plan SdpSemantics. Please use GetReceivers "
|
|
"instead.";
|
|
return remote_streams_;
|
|
}
|
|
|
|
bool PeerConnection::AddStream(MediaStreamInterface* local_stream) {
|
|
RTC_DCHECK_RUN_ON(signaling_thread());
|
|
RTC_CHECK(!IsUnifiedPlan()) << "AddStream is not available with Unified Plan "
|
|
"SdpSemantics. Please use AddTrack instead.";
|
|
TRACE_EVENT0("webrtc", "PeerConnection::AddStream");
|
|
if (IsClosed()) {
|
|
return false;
|
|
}
|
|
if (!CanAddLocalMediaStream(local_streams_, local_stream)) {
|
|
return false;
|
|
}
|
|
|
|
local_streams_->AddStream(local_stream);
|
|
MediaStreamObserver* observer = new MediaStreamObserver(local_stream);
|
|
observer->SignalAudioTrackAdded.connect(this,
|
|
&PeerConnection::OnAudioTrackAdded);
|
|
observer->SignalAudioTrackRemoved.connect(
|
|
this, &PeerConnection::OnAudioTrackRemoved);
|
|
observer->SignalVideoTrackAdded.connect(this,
|
|
&PeerConnection::OnVideoTrackAdded);
|
|
observer->SignalVideoTrackRemoved.connect(
|
|
this, &PeerConnection::OnVideoTrackRemoved);
|
|
stream_observers_.push_back(std::unique_ptr<MediaStreamObserver>(observer));
|
|
|
|
for (const auto& track : local_stream->GetAudioTracks()) {
|
|
AddAudioTrack(track.get(), local_stream);
|
|
}
|
|
for (const auto& track : local_stream->GetVideoTracks()) {
|
|
AddVideoTrack(track.get(), local_stream);
|
|
}
|
|
|
|
stats_->AddStream(local_stream);
|
|
sdp_handler_.UpdateNegotiationNeeded();
|
|
return true;
|
|
}
|
|
|
|
void PeerConnection::RemoveStream(MediaStreamInterface* local_stream) {
|
|
RTC_DCHECK_RUN_ON(signaling_thread());
|
|
RTC_CHECK(!IsUnifiedPlan()) << "RemoveStream is not available with Unified "
|
|
"Plan SdpSemantics. Please use RemoveTrack "
|
|
"instead.";
|
|
TRACE_EVENT0("webrtc", "PeerConnection::RemoveStream");
|
|
if (!IsClosed()) {
|
|
for (const auto& track : local_stream->GetAudioTracks()) {
|
|
RemoveAudioTrack(track.get(), local_stream);
|
|
}
|
|
for (const auto& track : local_stream->GetVideoTracks()) {
|
|
RemoveVideoTrack(track.get(), local_stream);
|
|
}
|
|
}
|
|
local_streams_->RemoveStream(local_stream);
|
|
stream_observers_.erase(
|
|
std::remove_if(
|
|
stream_observers_.begin(), stream_observers_.end(),
|
|
[local_stream](const std::unique_ptr<MediaStreamObserver>& observer) {
|
|
return observer->stream()->id().compare(local_stream->id()) == 0;
|
|
}),
|
|
stream_observers_.end());
|
|
|
|
if (IsClosed()) {
|
|
return;
|
|
}
|
|
sdp_handler_.UpdateNegotiationNeeded();
|
|
}
|
|
|
|
RTCErrorOr<rtc::scoped_refptr<RtpSenderInterface>> PeerConnection::AddTrack(
|
|
rtc::scoped_refptr<MediaStreamTrackInterface> track,
|
|
const std::vector<std::string>& stream_ids) {
|
|
RTC_DCHECK_RUN_ON(signaling_thread());
|
|
TRACE_EVENT0("webrtc", "PeerConnection::AddTrack");
|
|
if (!track) {
|
|
LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_PARAMETER, "Track is null.");
|
|
}
|
|
if (!(track->kind() == MediaStreamTrackInterface::kAudioKind ||
|
|
track->kind() == MediaStreamTrackInterface::kVideoKind)) {
|
|
LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_PARAMETER,
|
|
"Track has invalid kind: " + track->kind());
|
|
}
|
|
if (IsClosed()) {
|
|
LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_STATE,
|
|
"PeerConnection is closed.");
|
|
}
|
|
if (FindSenderForTrack(track)) {
|
|
LOG_AND_RETURN_ERROR(
|
|
RTCErrorType::INVALID_PARAMETER,
|
|
"Sender already exists for track " + track->id() + ".");
|
|
}
|
|
auto sender_or_error =
|
|
(IsUnifiedPlan() ? AddTrackUnifiedPlan(track, stream_ids)
|
|
: AddTrackPlanB(track, stream_ids));
|
|
if (sender_or_error.ok()) {
|
|
sdp_handler_.UpdateNegotiationNeeded();
|
|
stats_->AddTrack(track);
|
|
}
|
|
return sender_or_error;
|
|
}
|
|
|
|
RTCErrorOr<rtc::scoped_refptr<RtpSenderInterface>>
|
|
PeerConnection::AddTrackPlanB(
|
|
rtc::scoped_refptr<MediaStreamTrackInterface> track,
|
|
const std::vector<std::string>& stream_ids) {
|
|
if (stream_ids.size() > 1u) {
|
|
LOG_AND_RETURN_ERROR(RTCErrorType::UNSUPPORTED_OPERATION,
|
|
"AddTrack with more than one stream is not "
|
|
"supported with Plan B semantics.");
|
|
}
|
|
std::vector<std::string> adjusted_stream_ids = stream_ids;
|
|
if (adjusted_stream_ids.empty()) {
|
|
adjusted_stream_ids.push_back(rtc::CreateRandomUuid());
|
|
}
|
|
cricket::MediaType media_type =
|
|
(track->kind() == MediaStreamTrackInterface::kAudioKind
|
|
? cricket::MEDIA_TYPE_AUDIO
|
|
: cricket::MEDIA_TYPE_VIDEO);
|
|
auto new_sender =
|
|
CreateSender(media_type, track->id(), track, adjusted_stream_ids, {});
|
|
if (track->kind() == MediaStreamTrackInterface::kAudioKind) {
|
|
new_sender->internal()->SetMediaChannel(voice_media_channel());
|
|
GetAudioTransceiver()->internal()->AddSender(new_sender);
|
|
const RtpSenderInfo* sender_info =
|
|
FindSenderInfo(local_audio_sender_infos_,
|
|
new_sender->internal()->stream_ids()[0], track->id());
|
|
if (sender_info) {
|
|
new_sender->internal()->SetSsrc(sender_info->first_ssrc);
|
|
}
|
|
} else {
|
|
RTC_DCHECK_EQ(MediaStreamTrackInterface::kVideoKind, track->kind());
|
|
new_sender->internal()->SetMediaChannel(video_media_channel());
|
|
GetVideoTransceiver()->internal()->AddSender(new_sender);
|
|
const RtpSenderInfo* sender_info =
|
|
FindSenderInfo(local_video_sender_infos_,
|
|
new_sender->internal()->stream_ids()[0], track->id());
|
|
if (sender_info) {
|
|
new_sender->internal()->SetSsrc(sender_info->first_ssrc);
|
|
}
|
|
}
|
|
return rtc::scoped_refptr<RtpSenderInterface>(new_sender);
|
|
}
|
|
|
|
RTCErrorOr<rtc::scoped_refptr<RtpSenderInterface>>
|
|
PeerConnection::AddTrackUnifiedPlan(
|
|
rtc::scoped_refptr<MediaStreamTrackInterface> track,
|
|
const std::vector<std::string>& stream_ids) {
|
|
auto transceiver = FindFirstTransceiverForAddedTrack(track);
|
|
if (transceiver) {
|
|
RTC_LOG(LS_INFO) << "Reusing an existing "
|
|
<< cricket::MediaTypeToString(transceiver->media_type())
|
|
<< " transceiver for AddTrack.";
|
|
if (transceiver->stopping()) {
|
|
LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_PARAMETER,
|
|
"The existing transceiver is stopping.");
|
|
}
|
|
|
|
if (transceiver->direction() == RtpTransceiverDirection::kRecvOnly) {
|
|
transceiver->internal()->set_direction(
|
|
RtpTransceiverDirection::kSendRecv);
|
|
} else if (transceiver->direction() == RtpTransceiverDirection::kInactive) {
|
|
transceiver->internal()->set_direction(
|
|
RtpTransceiverDirection::kSendOnly);
|
|
}
|
|
transceiver->sender()->SetTrack(track);
|
|
transceiver->internal()->sender_internal()->set_stream_ids(stream_ids);
|
|
transceiver->internal()->set_reused_for_addtrack(true);
|
|
} else {
|
|
cricket::MediaType media_type =
|
|
(track->kind() == MediaStreamTrackInterface::kAudioKind
|
|
? cricket::MEDIA_TYPE_AUDIO
|
|
: cricket::MEDIA_TYPE_VIDEO);
|
|
RTC_LOG(LS_INFO) << "Adding " << cricket::MediaTypeToString(media_type)
|
|
<< " transceiver in response to a call to AddTrack.";
|
|
std::string sender_id = track->id();
|
|
// Avoid creating a sender with an existing ID by generating a random ID.
|
|
// This can happen if this is the second time AddTrack has created a sender
|
|
// for this track.
|
|
if (FindSenderById(sender_id)) {
|
|
sender_id = rtc::CreateRandomUuid();
|
|
}
|
|
auto sender = CreateSender(media_type, sender_id, track, stream_ids, {});
|
|
auto receiver = CreateReceiver(media_type, rtc::CreateRandomUuid());
|
|
transceiver = CreateAndAddTransceiver(sender, receiver);
|
|
transceiver->internal()->set_created_by_addtrack(true);
|
|
transceiver->internal()->set_direction(RtpTransceiverDirection::kSendRecv);
|
|
}
|
|
return transceiver->sender();
|
|
}
|
|
|
|
rtc::scoped_refptr<RtpTransceiverProxyWithInternal<RtpTransceiver>>
|
|
PeerConnection::FindFirstTransceiverForAddedTrack(
|
|
rtc::scoped_refptr<MediaStreamTrackInterface> track) {
|
|
RTC_DCHECK(track);
|
|
for (auto transceiver : transceivers_.List()) {
|
|
if (!transceiver->sender()->track() &&
|
|
cricket::MediaTypeToString(transceiver->media_type()) ==
|
|
track->kind() &&
|
|
!transceiver->internal()->has_ever_been_used_to_send() &&
|
|
!transceiver->stopped()) {
|
|
return transceiver;
|
|
}
|
|
}
|
|
return nullptr;
|
|
}
|
|
|
|
bool PeerConnection::RemoveTrack(RtpSenderInterface* sender) {
|
|
TRACE_EVENT0("webrtc", "PeerConnection::RemoveTrack");
|
|
return RemoveTrackNew(sender).ok();
|
|
}
|
|
|
|
RTCError PeerConnection::RemoveTrackNew(
|
|
rtc::scoped_refptr<RtpSenderInterface> sender) {
|
|
RTC_DCHECK_RUN_ON(signaling_thread());
|
|
if (!sender) {
|
|
LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_PARAMETER, "Sender is null.");
|
|
}
|
|
if (IsClosed()) {
|
|
LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_STATE,
|
|
"PeerConnection is closed.");
|
|
}
|
|
if (IsUnifiedPlan()) {
|
|
auto transceiver = FindTransceiverBySender(sender);
|
|
if (!transceiver || !sender->track()) {
|
|
return RTCError::OK();
|
|
}
|
|
sender->SetTrack(nullptr);
|
|
if (transceiver->direction() == RtpTransceiverDirection::kSendRecv) {
|
|
transceiver->internal()->set_direction(
|
|
RtpTransceiverDirection::kRecvOnly);
|
|
} else if (transceiver->direction() == RtpTransceiverDirection::kSendOnly) {
|
|
transceiver->internal()->set_direction(
|
|
RtpTransceiverDirection::kInactive);
|
|
}
|
|
} else {
|
|
bool removed;
|
|
if (sender->media_type() == cricket::MEDIA_TYPE_AUDIO) {
|
|
removed = GetAudioTransceiver()->internal()->RemoveSender(sender);
|
|
} else {
|
|
RTC_DCHECK_EQ(cricket::MEDIA_TYPE_VIDEO, sender->media_type());
|
|
removed = GetVideoTransceiver()->internal()->RemoveSender(sender);
|
|
}
|
|
if (!removed) {
|
|
LOG_AND_RETURN_ERROR(
|
|
RTCErrorType::INVALID_PARAMETER,
|
|
"Couldn't find sender " + sender->id() + " to remove.");
|
|
}
|
|
}
|
|
sdp_handler_.UpdateNegotiationNeeded();
|
|
return RTCError::OK();
|
|
}
|
|
|
|
rtc::scoped_refptr<RtpTransceiverProxyWithInternal<RtpTransceiver>>
|
|
PeerConnection::FindTransceiverBySender(
|
|
rtc::scoped_refptr<RtpSenderInterface> sender) {
|
|
return transceivers_.FindBySender(sender);
|
|
}
|
|
|
|
RTCErrorOr<rtc::scoped_refptr<RtpTransceiverInterface>>
|
|
PeerConnection::AddTransceiver(
|
|
rtc::scoped_refptr<MediaStreamTrackInterface> track) {
|
|
return AddTransceiver(track, RtpTransceiverInit());
|
|
}
|
|
|
|
RTCErrorOr<rtc::scoped_refptr<RtpTransceiverInterface>>
|
|
PeerConnection::AddTransceiver(
|
|
rtc::scoped_refptr<MediaStreamTrackInterface> track,
|
|
const RtpTransceiverInit& init) {
|
|
RTC_DCHECK_RUN_ON(signaling_thread());
|
|
RTC_CHECK(IsUnifiedPlan())
|
|
<< "AddTransceiver is only available with Unified Plan SdpSemantics";
|
|
if (!track) {
|
|
LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_PARAMETER, "track is null");
|
|
}
|
|
cricket::MediaType media_type;
|
|
if (track->kind() == MediaStreamTrackInterface::kAudioKind) {
|
|
media_type = cricket::MEDIA_TYPE_AUDIO;
|
|
} else if (track->kind() == MediaStreamTrackInterface::kVideoKind) {
|
|
media_type = cricket::MEDIA_TYPE_VIDEO;
|
|
} else {
|
|
LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_PARAMETER,
|
|
"Track kind is not audio or video");
|
|
}
|
|
return AddTransceiver(media_type, track, init);
|
|
}
|
|
|
|
RTCErrorOr<rtc::scoped_refptr<RtpTransceiverInterface>>
|
|
PeerConnection::AddTransceiver(cricket::MediaType media_type) {
|
|
return AddTransceiver(media_type, RtpTransceiverInit());
|
|
}
|
|
|
|
RTCErrorOr<rtc::scoped_refptr<RtpTransceiverInterface>>
|
|
PeerConnection::AddTransceiver(cricket::MediaType media_type,
|
|
const RtpTransceiverInit& init) {
|
|
RTC_DCHECK_RUN_ON(signaling_thread());
|
|
RTC_CHECK(IsUnifiedPlan())
|
|
<< "AddTransceiver is only available with Unified Plan SdpSemantics";
|
|
if (!(media_type == cricket::MEDIA_TYPE_AUDIO ||
|
|
media_type == cricket::MEDIA_TYPE_VIDEO)) {
|
|
LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_PARAMETER,
|
|
"media type is not audio or video");
|
|
}
|
|
return AddTransceiver(media_type, nullptr, init);
|
|
}
|
|
|
|
RTCErrorOr<rtc::scoped_refptr<RtpTransceiverInterface>>
|
|
PeerConnection::AddTransceiver(
|
|
cricket::MediaType media_type,
|
|
rtc::scoped_refptr<MediaStreamTrackInterface> track,
|
|
const RtpTransceiverInit& init,
|
|
bool update_negotiation_needed) {
|
|
RTC_DCHECK((media_type == cricket::MEDIA_TYPE_AUDIO ||
|
|
media_type == cricket::MEDIA_TYPE_VIDEO));
|
|
if (track) {
|
|
RTC_DCHECK_EQ(media_type,
|
|
(track->kind() == MediaStreamTrackInterface::kAudioKind
|
|
? cricket::MEDIA_TYPE_AUDIO
|
|
: cricket::MEDIA_TYPE_VIDEO));
|
|
}
|
|
|
|
RTC_HISTOGRAM_COUNTS_LINEAR(kSimulcastNumberOfEncodings,
|
|
init.send_encodings.size(), 0, 7, 8);
|
|
|
|
size_t num_rids = absl::c_count_if(init.send_encodings,
|
|
[](const RtpEncodingParameters& encoding) {
|
|
return !encoding.rid.empty();
|
|
});
|
|
if (num_rids > 0 && num_rids != init.send_encodings.size()) {
|
|
LOG_AND_RETURN_ERROR(
|
|
RTCErrorType::INVALID_PARAMETER,
|
|
"RIDs must be provided for either all or none of the send encodings.");
|
|
}
|
|
|
|
if (num_rids > 0 && absl::c_any_of(init.send_encodings,
|
|
[](const RtpEncodingParameters& encoding) {
|
|
return !IsLegalRsidName(encoding.rid);
|
|
})) {
|
|
LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_PARAMETER,
|
|
"Invalid RID value provided.");
|
|
}
|
|
|
|
if (absl::c_any_of(init.send_encodings,
|
|
[](const RtpEncodingParameters& encoding) {
|
|
return encoding.ssrc.has_value();
|
|
})) {
|
|
LOG_AND_RETURN_ERROR(
|
|
RTCErrorType::UNSUPPORTED_PARAMETER,
|
|
"Attempted to set an unimplemented parameter of RtpParameters.");
|
|
}
|
|
|
|
RtpParameters parameters;
|
|
parameters.encodings = init.send_encodings;
|
|
|
|
// Encodings are dropped from the tail if too many are provided.
|
|
if (parameters.encodings.size() > kMaxSimulcastStreams) {
|
|
parameters.encodings.erase(
|
|
parameters.encodings.begin() + kMaxSimulcastStreams,
|
|
parameters.encodings.end());
|
|
}
|
|
|
|
// Single RID should be removed.
|
|
if (parameters.encodings.size() == 1 &&
|
|
!parameters.encodings[0].rid.empty()) {
|
|
RTC_LOG(LS_INFO) << "Removing RID: " << parameters.encodings[0].rid << ".";
|
|
parameters.encodings[0].rid.clear();
|
|
}
|
|
|
|
// If RIDs were not provided, they are generated for simulcast scenario.
|
|
if (parameters.encodings.size() > 1 && num_rids == 0) {
|
|
rtc::UniqueStringGenerator rid_generator;
|
|
for (RtpEncodingParameters& encoding : parameters.encodings) {
|
|
encoding.rid = rid_generator();
|
|
}
|
|
}
|
|
|
|
if (UnimplementedRtpParameterHasValue(parameters)) {
|
|
LOG_AND_RETURN_ERROR(
|
|
RTCErrorType::UNSUPPORTED_PARAMETER,
|
|
"Attempted to set an unimplemented parameter of RtpParameters.");
|
|
}
|
|
|
|
auto result = cricket::CheckRtpParametersValues(parameters);
|
|
if (!result.ok()) {
|
|
LOG_AND_RETURN_ERROR(result.type(), result.message());
|
|
}
|
|
|
|
RTC_LOG(LS_INFO) << "Adding " << cricket::MediaTypeToString(media_type)
|
|
<< " transceiver in response to a call to AddTransceiver.";
|
|
// Set the sender ID equal to the track ID if the track is specified unless
|
|
// that sender ID is already in use.
|
|
std::string sender_id =
|
|
(track && !FindSenderById(track->id()) ? track->id()
|
|
: rtc::CreateRandomUuid());
|
|
auto sender = CreateSender(media_type, sender_id, track, init.stream_ids,
|
|
parameters.encodings);
|
|
auto receiver = CreateReceiver(media_type, rtc::CreateRandomUuid());
|
|
auto transceiver = CreateAndAddTransceiver(sender, receiver);
|
|
transceiver->internal()->set_direction(init.direction);
|
|
|
|
if (update_negotiation_needed) {
|
|
sdp_handler_.UpdateNegotiationNeeded();
|
|
}
|
|
|
|
return rtc::scoped_refptr<RtpTransceiverInterface>(transceiver);
|
|
}
|
|
|
|
rtc::scoped_refptr<RtpSenderProxyWithInternal<RtpSenderInternal>>
|
|
PeerConnection::CreateSender(
|
|
cricket::MediaType media_type,
|
|
const std::string& id,
|
|
rtc::scoped_refptr<MediaStreamTrackInterface> track,
|
|
const std::vector<std::string>& stream_ids,
|
|
const std::vector<RtpEncodingParameters>& send_encodings) {
|
|
RTC_DCHECK_RUN_ON(signaling_thread());
|
|
rtc::scoped_refptr<RtpSenderProxyWithInternal<RtpSenderInternal>> sender;
|
|
if (media_type == cricket::MEDIA_TYPE_AUDIO) {
|
|
RTC_DCHECK(!track ||
|
|
(track->kind() == MediaStreamTrackInterface::kAudioKind));
|
|
sender = RtpSenderProxyWithInternal<RtpSenderInternal>::Create(
|
|
signaling_thread(),
|
|
AudioRtpSender::Create(worker_thread(), id, stats_.get(), this));
|
|
NoteUsageEvent(UsageEvent::AUDIO_ADDED);
|
|
} else {
|
|
RTC_DCHECK_EQ(media_type, cricket::MEDIA_TYPE_VIDEO);
|
|
RTC_DCHECK(!track ||
|
|
(track->kind() == MediaStreamTrackInterface::kVideoKind));
|
|
sender = RtpSenderProxyWithInternal<RtpSenderInternal>::Create(
|
|
signaling_thread(), VideoRtpSender::Create(worker_thread(), id, this));
|
|
NoteUsageEvent(UsageEvent::VIDEO_ADDED);
|
|
}
|
|
bool set_track_succeeded = sender->SetTrack(track);
|
|
RTC_DCHECK(set_track_succeeded);
|
|
sender->internal()->set_stream_ids(stream_ids);
|
|
sender->internal()->set_init_send_encodings(send_encodings);
|
|
return sender;
|
|
}
|
|
|
|
rtc::scoped_refptr<RtpReceiverProxyWithInternal<RtpReceiverInternal>>
|
|
PeerConnection::CreateReceiver(cricket::MediaType media_type,
|
|
const std::string& receiver_id) {
|
|
rtc::scoped_refptr<RtpReceiverProxyWithInternal<RtpReceiverInternal>>
|
|
receiver;
|
|
if (media_type == cricket::MEDIA_TYPE_AUDIO) {
|
|
receiver = RtpReceiverProxyWithInternal<RtpReceiverInternal>::Create(
|
|
signaling_thread(), new AudioRtpReceiver(worker_thread(), receiver_id,
|
|
std::vector<std::string>({})));
|
|
NoteUsageEvent(UsageEvent::AUDIO_ADDED);
|
|
} else {
|
|
RTC_DCHECK_EQ(media_type, cricket::MEDIA_TYPE_VIDEO);
|
|
receiver = RtpReceiverProxyWithInternal<RtpReceiverInternal>::Create(
|
|
signaling_thread(), new VideoRtpReceiver(worker_thread(), receiver_id,
|
|
std::vector<std::string>({})));
|
|
NoteUsageEvent(UsageEvent::VIDEO_ADDED);
|
|
}
|
|
return receiver;
|
|
}
|
|
|
|
rtc::scoped_refptr<RtpTransceiverProxyWithInternal<RtpTransceiver>>
|
|
PeerConnection::CreateAndAddTransceiver(
|
|
rtc::scoped_refptr<RtpSenderProxyWithInternal<RtpSenderInternal>> sender,
|
|
rtc::scoped_refptr<RtpReceiverProxyWithInternal<RtpReceiverInternal>>
|
|
receiver) {
|
|
RTC_DCHECK_RUN_ON(signaling_thread());
|
|
// Ensure that the new sender does not have an ID that is already in use by
|
|
// another sender.
|
|
// Allow receiver IDs to conflict since those come from remote SDP (which
|
|
// could be invalid, but should not cause a crash).
|
|
RTC_DCHECK(!FindSenderById(sender->id()));
|
|
auto transceiver = RtpTransceiverProxyWithInternal<RtpTransceiver>::Create(
|
|
signaling_thread(),
|
|
new RtpTransceiver(
|
|
sender, receiver, channel_manager(),
|
|
sender->media_type() == cricket::MEDIA_TYPE_AUDIO
|
|
? channel_manager()->GetSupportedAudioRtpHeaderExtensions()
|
|
: channel_manager()->GetSupportedVideoRtpHeaderExtensions()));
|
|
transceivers_.Add(transceiver);
|
|
transceiver->internal()->SignalNegotiationNeeded.connect(
|
|
this, &PeerConnection::OnNegotiationNeeded);
|
|
return transceiver;
|
|
}
|
|
|
|
void PeerConnection::OnNegotiationNeeded() {
|
|
RTC_DCHECK_RUN_ON(signaling_thread());
|
|
RTC_DCHECK(!IsClosed());
|
|
sdp_handler_.UpdateNegotiationNeeded();
|
|
}
|
|
|
|
rtc::scoped_refptr<RtpSenderInterface> PeerConnection::CreateSender(
|
|
const std::string& kind,
|
|
const std::string& stream_id) {
|
|
RTC_DCHECK_RUN_ON(signaling_thread());
|
|
RTC_CHECK(!IsUnifiedPlan()) << "CreateSender is not available with Unified "
|
|
"Plan SdpSemantics. Please use AddTransceiver "
|
|
"instead.";
|
|
TRACE_EVENT0("webrtc", "PeerConnection::CreateSender");
|
|
if (IsClosed()) {
|
|
return nullptr;
|
|
}
|
|
|
|
// Internally we need to have one stream with Plan B semantics, so we
|
|
// generate a random stream ID if not specified.
|
|
std::vector<std::string> stream_ids;
|
|
if (stream_id.empty()) {
|
|
stream_ids.push_back(rtc::CreateRandomUuid());
|
|
RTC_LOG(LS_INFO)
|
|
<< "No stream_id specified for sender. Generated stream ID: "
|
|
<< stream_ids[0];
|
|
} else {
|
|
stream_ids.push_back(stream_id);
|
|
}
|
|
|
|
// TODO(steveanton): Move construction of the RtpSenders to RtpTransceiver.
|
|
rtc::scoped_refptr<RtpSenderProxyWithInternal<RtpSenderInternal>> new_sender;
|
|
if (kind == MediaStreamTrackInterface::kAudioKind) {
|
|
auto audio_sender = AudioRtpSender::Create(
|
|
worker_thread(), rtc::CreateRandomUuid(), stats_.get(), this);
|
|
audio_sender->SetMediaChannel(voice_media_channel());
|
|
new_sender = RtpSenderProxyWithInternal<RtpSenderInternal>::Create(
|
|
signaling_thread(), audio_sender);
|
|
GetAudioTransceiver()->internal()->AddSender(new_sender);
|
|
} else if (kind == MediaStreamTrackInterface::kVideoKind) {
|
|
auto video_sender =
|
|
VideoRtpSender::Create(worker_thread(), rtc::CreateRandomUuid(), this);
|
|
video_sender->SetMediaChannel(video_media_channel());
|
|
new_sender = RtpSenderProxyWithInternal<RtpSenderInternal>::Create(
|
|
signaling_thread(), video_sender);
|
|
GetVideoTransceiver()->internal()->AddSender(new_sender);
|
|
} else {
|
|
RTC_LOG(LS_ERROR) << "CreateSender called with invalid kind: " << kind;
|
|
return nullptr;
|
|
}
|
|
new_sender->internal()->set_stream_ids(stream_ids);
|
|
|
|
return new_sender;
|
|
}
|
|
|
|
std::vector<rtc::scoped_refptr<RtpSenderInterface>> PeerConnection::GetSenders()
|
|
const {
|
|
RTC_DCHECK_RUN_ON(signaling_thread());
|
|
std::vector<rtc::scoped_refptr<RtpSenderInterface>> ret;
|
|
for (const auto& sender : GetSendersInternal()) {
|
|
ret.push_back(sender);
|
|
}
|
|
return ret;
|
|
}
|
|
|
|
std::vector<rtc::scoped_refptr<RtpSenderProxyWithInternal<RtpSenderInternal>>>
|
|
PeerConnection::GetSendersInternal() const {
|
|
std::vector<rtc::scoped_refptr<RtpSenderProxyWithInternal<RtpSenderInternal>>>
|
|
all_senders;
|
|
for (const auto& transceiver : transceivers_.List()) {
|
|
if (IsUnifiedPlan() && transceiver->internal()->stopped())
|
|
continue;
|
|
|
|
auto senders = transceiver->internal()->senders();
|
|
all_senders.insert(all_senders.end(), senders.begin(), senders.end());
|
|
}
|
|
return all_senders;
|
|
}
|
|
|
|
std::vector<rtc::scoped_refptr<RtpReceiverInterface>>
|
|
PeerConnection::GetReceivers() const {
|
|
RTC_DCHECK_RUN_ON(signaling_thread());
|
|
std::vector<rtc::scoped_refptr<RtpReceiverInterface>> ret;
|
|
for (const auto& receiver : GetReceiversInternal()) {
|
|
ret.push_back(receiver);
|
|
}
|
|
return ret;
|
|
}
|
|
|
|
std::vector<
|
|
rtc::scoped_refptr<RtpReceiverProxyWithInternal<RtpReceiverInternal>>>
|
|
PeerConnection::GetReceiversInternal() const {
|
|
std::vector<
|
|
rtc::scoped_refptr<RtpReceiverProxyWithInternal<RtpReceiverInternal>>>
|
|
all_receivers;
|
|
for (const auto& transceiver : transceivers_.List()) {
|
|
if (IsUnifiedPlan() && transceiver->internal()->stopped())
|
|
continue;
|
|
|
|
auto receivers = transceiver->internal()->receivers();
|
|
all_receivers.insert(all_receivers.end(), receivers.begin(),
|
|
receivers.end());
|
|
}
|
|
return all_receivers;
|
|
}
|
|
|
|
std::vector<rtc::scoped_refptr<RtpTransceiverInterface>>
|
|
PeerConnection::GetTransceivers() const {
|
|
RTC_DCHECK_RUN_ON(signaling_thread());
|
|
RTC_CHECK(IsUnifiedPlan())
|
|
<< "GetTransceivers is only supported with Unified Plan SdpSemantics.";
|
|
std::vector<rtc::scoped_refptr<RtpTransceiverInterface>> all_transceivers;
|
|
for (const auto& transceiver : transceivers_.List()) {
|
|
all_transceivers.push_back(transceiver);
|
|
}
|
|
return all_transceivers;
|
|
}
|
|
|
|
bool PeerConnection::GetStats(StatsObserver* observer,
|
|
MediaStreamTrackInterface* track,
|
|
StatsOutputLevel level) {
|
|
TRACE_EVENT0("webrtc", "PeerConnection::GetStats");
|
|
RTC_DCHECK_RUN_ON(signaling_thread());
|
|
if (!observer) {
|
|
RTC_LOG(LS_ERROR) << "GetStats - observer is NULL.";
|
|
return false;
|
|
}
|
|
|
|
stats_->UpdateStats(level);
|
|
// The StatsCollector is used to tell if a track is valid because it may
|
|
// remember tracks that the PeerConnection previously removed.
|
|
if (track && !stats_->IsValidTrack(track->id())) {
|
|
RTC_LOG(LS_WARNING) << "GetStats is called with an invalid track: "
|
|
<< track->id();
|
|
return false;
|
|
}
|
|
signaling_thread()->Post(RTC_FROM_HERE, this, MSG_GETSTATS,
|
|
new GetStatsMsg(observer, track));
|
|
return true;
|
|
}
|
|
|
|
void PeerConnection::GetStats(RTCStatsCollectorCallback* callback) {
|
|
TRACE_EVENT0("webrtc", "PeerConnection::GetStats");
|
|
RTC_DCHECK_RUN_ON(signaling_thread());
|
|
RTC_DCHECK(stats_collector_);
|
|
RTC_DCHECK(callback);
|
|
stats_collector_->GetStatsReport(callback);
|
|
}
|
|
|
|
void PeerConnection::GetStats(
|
|
rtc::scoped_refptr<RtpSenderInterface> selector,
|
|
rtc::scoped_refptr<RTCStatsCollectorCallback> callback) {
|
|
TRACE_EVENT0("webrtc", "PeerConnection::GetStats");
|
|
RTC_DCHECK_RUN_ON(signaling_thread());
|
|
RTC_DCHECK(callback);
|
|
RTC_DCHECK(stats_collector_);
|
|
rtc::scoped_refptr<RtpSenderInternal> internal_sender;
|
|
if (selector) {
|
|
for (const auto& proxy_transceiver : transceivers_.List()) {
|
|
for (const auto& proxy_sender :
|
|
proxy_transceiver->internal()->senders()) {
|
|
if (proxy_sender == selector) {
|
|
internal_sender = proxy_sender->internal();
|
|
break;
|
|
}
|
|
}
|
|
if (internal_sender)
|
|
break;
|
|
}
|
|
}
|
|
// If there is no |internal_sender| then |selector| is either null or does not
|
|
// belong to the PeerConnection (in Plan B, senders can be removed from the
|
|
// PeerConnection). This means that "all the stats objects representing the
|
|
// selector" is an empty set. Invoking GetStatsReport() with a null selector
|
|
// produces an empty stats report.
|
|
stats_collector_->GetStatsReport(internal_sender, callback);
|
|
}
|
|
|
|
void PeerConnection::GetStats(
|
|
rtc::scoped_refptr<RtpReceiverInterface> selector,
|
|
rtc::scoped_refptr<RTCStatsCollectorCallback> callback) {
|
|
TRACE_EVENT0("webrtc", "PeerConnection::GetStats");
|
|
RTC_DCHECK_RUN_ON(signaling_thread());
|
|
RTC_DCHECK(callback);
|
|
RTC_DCHECK(stats_collector_);
|
|
rtc::scoped_refptr<RtpReceiverInternal> internal_receiver;
|
|
if (selector) {
|
|
for (const auto& proxy_transceiver : transceivers_.List()) {
|
|
for (const auto& proxy_receiver :
|
|
proxy_transceiver->internal()->receivers()) {
|
|
if (proxy_receiver == selector) {
|
|
internal_receiver = proxy_receiver->internal();
|
|
break;
|
|
}
|
|
}
|
|
if (internal_receiver)
|
|
break;
|
|
}
|
|
}
|
|
// If there is no |internal_receiver| then |selector| is either null or does
|
|
// not belong to the PeerConnection (in Plan B, receivers can be removed from
|
|
// the PeerConnection). This means that "all the stats objects representing
|
|
// the selector" is an empty set. Invoking GetStatsReport() with a null
|
|
// selector produces an empty stats report.
|
|
stats_collector_->GetStatsReport(internal_receiver, callback);
|
|
}
|
|
|
|
PeerConnectionInterface::SignalingState PeerConnection::signaling_state() {
|
|
RTC_DCHECK_RUN_ON(signaling_thread());
|
|
return sdp_handler_.signaling_state();
|
|
}
|
|
|
|
PeerConnectionInterface::IceConnectionState
|
|
PeerConnection::ice_connection_state() {
|
|
RTC_DCHECK_RUN_ON(signaling_thread());
|
|
return ice_connection_state_;
|
|
}
|
|
|
|
PeerConnectionInterface::IceConnectionState
|
|
PeerConnection::standardized_ice_connection_state() {
|
|
RTC_DCHECK_RUN_ON(signaling_thread());
|
|
return standardized_ice_connection_state_;
|
|
}
|
|
|
|
PeerConnectionInterface::PeerConnectionState
|
|
PeerConnection::peer_connection_state() {
|
|
RTC_DCHECK_RUN_ON(signaling_thread());
|
|
return connection_state_;
|
|
}
|
|
|
|
PeerConnectionInterface::IceGatheringState
|
|
PeerConnection::ice_gathering_state() {
|
|
RTC_DCHECK_RUN_ON(signaling_thread());
|
|
return ice_gathering_state_;
|
|
}
|
|
|
|
absl::optional<bool> PeerConnection::can_trickle_ice_candidates() {
|
|
RTC_DCHECK_RUN_ON(signaling_thread());
|
|
const SessionDescriptionInterface* description = current_remote_description();
|
|
if (!description) {
|
|
description = pending_remote_description();
|
|
}
|
|
if (!description) {
|
|
return absl::nullopt;
|
|
}
|
|
// TODO(bugs.webrtc.org/7443): Change to retrieve from session-level option.
|
|
if (description->description()->transport_infos().size() < 1) {
|
|
return absl::nullopt;
|
|
}
|
|
return description->description()->transport_infos()[0].description.HasOption(
|
|
"trickle");
|
|
}
|
|
|
|
rtc::scoped_refptr<DataChannelInterface> PeerConnection::CreateDataChannel(
|
|
const std::string& label,
|
|
const DataChannelInit* config) {
|
|
RTC_DCHECK_RUN_ON(signaling_thread());
|
|
TRACE_EVENT0("webrtc", "PeerConnection::CreateDataChannel");
|
|
|
|
bool first_datachannel = !data_channel_controller_.HasDataChannels();
|
|
|
|
std::unique_ptr<InternalDataChannelInit> internal_config;
|
|
if (config) {
|
|
internal_config.reset(new InternalDataChannelInit(*config));
|
|
}
|
|
rtc::scoped_refptr<DataChannelInterface> channel(
|
|
data_channel_controller_.InternalCreateDataChannelWithProxy(
|
|
label, internal_config.get()));
|
|
if (!channel.get()) {
|
|
return nullptr;
|
|
}
|
|
|
|
// Trigger the onRenegotiationNeeded event for every new RTP DataChannel, or
|
|
// the first SCTP DataChannel.
|
|
if (data_channel_type() == cricket::DCT_RTP || first_datachannel) {
|
|
sdp_handler_.UpdateNegotiationNeeded();
|
|
}
|
|
NoteUsageEvent(UsageEvent::DATA_ADDED);
|
|
return channel;
|
|
}
|
|
|
|
void PeerConnection::RestartIce() {
|
|
RTC_DCHECK_RUN_ON(signaling_thread());
|
|
sdp_handler_.RestartIce();
|
|
}
|
|
|
|
void PeerConnection::CreateOffer(CreateSessionDescriptionObserver* observer,
|
|
const RTCOfferAnswerOptions& options) {
|
|
RTC_DCHECK_RUN_ON(signaling_thread());
|
|
sdp_handler_.CreateOffer(observer, options);
|
|
}
|
|
|
|
void PeerConnection::CreateAnswer(CreateSessionDescriptionObserver* observer,
|
|
const RTCOfferAnswerOptions& options) {
|
|
RTC_DCHECK_RUN_ON(signaling_thread());
|
|
sdp_handler_.CreateAnswer(observer, options);
|
|
}
|
|
|
|
RTCError PeerConnection::HandleLegacyOfferOptions(
|
|
const RTCOfferAnswerOptions& options) {
|
|
RTC_DCHECK_RUN_ON(signaling_thread());
|
|
RTC_DCHECK(IsUnifiedPlan());
|
|
|
|
if (options.offer_to_receive_audio == 0) {
|
|
RemoveRecvDirectionFromReceivingTransceiversOfType(
|
|
cricket::MEDIA_TYPE_AUDIO);
|
|
} else if (options.offer_to_receive_audio == 1) {
|
|
AddUpToOneReceivingTransceiverOfType(cricket::MEDIA_TYPE_AUDIO);
|
|
} else if (options.offer_to_receive_audio > 1) {
|
|
LOG_AND_RETURN_ERROR(RTCErrorType::UNSUPPORTED_PARAMETER,
|
|
"offer_to_receive_audio > 1 is not supported.");
|
|
}
|
|
|
|
if (options.offer_to_receive_video == 0) {
|
|
RemoveRecvDirectionFromReceivingTransceiversOfType(
|
|
cricket::MEDIA_TYPE_VIDEO);
|
|
} else if (options.offer_to_receive_video == 1) {
|
|
AddUpToOneReceivingTransceiverOfType(cricket::MEDIA_TYPE_VIDEO);
|
|
} else if (options.offer_to_receive_video > 1) {
|
|
LOG_AND_RETURN_ERROR(RTCErrorType::UNSUPPORTED_PARAMETER,
|
|
"offer_to_receive_video > 1 is not supported.");
|
|
}
|
|
|
|
return RTCError::OK();
|
|
}
|
|
|
|
void PeerConnection::RemoveRecvDirectionFromReceivingTransceiversOfType(
|
|
cricket::MediaType media_type) {
|
|
for (const auto& transceiver : GetReceivingTransceiversOfType(media_type)) {
|
|
RtpTransceiverDirection new_direction =
|
|
RtpTransceiverDirectionWithRecvSet(transceiver->direction(), false);
|
|
if (new_direction != transceiver->direction()) {
|
|
RTC_LOG(LS_INFO) << "Changing " << cricket::MediaTypeToString(media_type)
|
|
<< " transceiver (MID="
|
|
<< transceiver->mid().value_or("<not set>") << ") from "
|
|
<< RtpTransceiverDirectionToString(
|
|
transceiver->direction())
|
|
<< " to "
|
|
<< RtpTransceiverDirectionToString(new_direction)
|
|
<< " since CreateOffer specified offer_to_receive=0";
|
|
transceiver->internal()->set_direction(new_direction);
|
|
}
|
|
}
|
|
}
|
|
|
|
void PeerConnection::AddUpToOneReceivingTransceiverOfType(
|
|
cricket::MediaType media_type) {
|
|
RTC_DCHECK_RUN_ON(signaling_thread());
|
|
if (GetReceivingTransceiversOfType(media_type).empty()) {
|
|
RTC_LOG(LS_INFO)
|
|
<< "Adding one recvonly " << cricket::MediaTypeToString(media_type)
|
|
<< " transceiver since CreateOffer specified offer_to_receive=1";
|
|
RtpTransceiverInit init;
|
|
init.direction = RtpTransceiverDirection::kRecvOnly;
|
|
AddTransceiver(media_type, nullptr, init,
|
|
/*update_negotiation_needed=*/false);
|
|
}
|
|
}
|
|
|
|
std::vector<rtc::scoped_refptr<RtpTransceiverProxyWithInternal<RtpTransceiver>>>
|
|
PeerConnection::GetReceivingTransceiversOfType(cricket::MediaType media_type) {
|
|
std::vector<
|
|
rtc::scoped_refptr<RtpTransceiverProxyWithInternal<RtpTransceiver>>>
|
|
receiving_transceivers;
|
|
for (const auto& transceiver : transceivers_.List()) {
|
|
if (!transceiver->stopped() && transceiver->media_type() == media_type &&
|
|
RtpTransceiverDirectionHasRecv(transceiver->direction())) {
|
|
receiving_transceivers.push_back(transceiver);
|
|
}
|
|
}
|
|
return receiving_transceivers;
|
|
}
|
|
|
|
void PeerConnection::SetLocalDescription(
|
|
SetSessionDescriptionObserver* observer,
|
|
SessionDescriptionInterface* desc_ptr) {
|
|
RTC_DCHECK_RUN_ON(signaling_thread());
|
|
sdp_handler_.SetLocalDescription(observer, desc_ptr);
|
|
}
|
|
|
|
void PeerConnection::SetLocalDescription(
|
|
std::unique_ptr<SessionDescriptionInterface> desc,
|
|
rtc::scoped_refptr<SetLocalDescriptionObserverInterface> observer) {
|
|
RTC_DCHECK_RUN_ON(signaling_thread());
|
|
sdp_handler_.SetLocalDescription(std::move(desc), observer);
|
|
}
|
|
|
|
void PeerConnection::SetLocalDescription(
|
|
SetSessionDescriptionObserver* observer) {
|
|
RTC_DCHECK_RUN_ON(signaling_thread());
|
|
sdp_handler_.SetLocalDescription(observer);
|
|
}
|
|
|
|
void PeerConnection::SetLocalDescription(
|
|
rtc::scoped_refptr<SetLocalDescriptionObserverInterface> observer) {
|
|
RTC_DCHECK_RUN_ON(signaling_thread());
|
|
sdp_handler_.SetLocalDescription(observer);
|
|
}
|
|
|
|
void PeerConnection::RemoveStoppedTransceivers() {
|
|
RTC_DCHECK_RUN_ON(signaling_thread());
|
|
// 3.2.10.1: For each transceiver in the connection's set of transceivers
|
|
// run the following steps:
|
|
if (!IsUnifiedPlan())
|
|
return;
|
|
// Traverse a copy of the transceiver list.
|
|
auto transceiver_list = transceivers_.List();
|
|
for (auto transceiver : transceiver_list) {
|
|
// 3.2.10.1.1: If transceiver is stopped, associated with an m= section
|
|
// and the associated m= section is rejected in
|
|
// connection.[[CurrentLocalDescription]] or
|
|
// connection.[[CurrentRemoteDescription]], remove the
|
|
// transceiver from the connection's set of transceivers.
|
|
if (!transceiver->stopped()) {
|
|
continue;
|
|
}
|
|
const ContentInfo* local_content =
|
|
FindMediaSectionForTransceiver(transceiver, local_description());
|
|
const ContentInfo* remote_content =
|
|
FindMediaSectionForTransceiver(transceiver, remote_description());
|
|
if ((local_content && local_content->rejected) ||
|
|
(remote_content && remote_content->rejected)) {
|
|
RTC_LOG(LS_INFO) << "Dissociating transceiver"
|
|
<< " since the media section is being recycled.";
|
|
transceiver->internal()->set_mid(absl::nullopt);
|
|
transceiver->internal()->set_mline_index(absl::nullopt);
|
|
transceivers_.Remove(transceiver);
|
|
continue;
|
|
}
|
|
if (!local_content && !remote_content) {
|
|
// TODO(bugs.webrtc.org/11973): Consider if this should be removed already
|
|
// See https://github.com/w3c/webrtc-pc/issues/2576
|
|
RTC_LOG(LS_INFO)
|
|
<< "Dropping stopped transceiver that was never associated";
|
|
transceivers_.Remove(transceiver);
|
|
continue;
|
|
}
|
|
}
|
|
}
|
|
|
|
// The SDP parser used to populate these values by default for the 'content
|
|
// name' if an a=mid line was absent.
|
|
static absl::string_view GetDefaultMidForPlanB(cricket::MediaType media_type) {
|
|
switch (media_type) {
|
|
case cricket::MEDIA_TYPE_AUDIO:
|
|
return cricket::CN_AUDIO;
|
|
case cricket::MEDIA_TYPE_VIDEO:
|
|
return cricket::CN_VIDEO;
|
|
case cricket::MEDIA_TYPE_DATA:
|
|
return cricket::CN_DATA;
|
|
}
|
|
RTC_NOTREACHED();
|
|
return "";
|
|
}
|
|
|
|
void PeerConnection::FillInMissingRemoteMids(
|
|
cricket::SessionDescription* new_remote_description) {
|
|
RTC_DCHECK_RUN_ON(signaling_thread());
|
|
RTC_DCHECK(new_remote_description);
|
|
const cricket::ContentInfos no_infos;
|
|
const cricket::ContentInfos& local_contents =
|
|
(local_description() ? local_description()->description()->contents()
|
|
: no_infos);
|
|
const cricket::ContentInfos& remote_contents =
|
|
(remote_description() ? remote_description()->description()->contents()
|
|
: no_infos);
|
|
for (size_t i = 0; i < new_remote_description->contents().size(); ++i) {
|
|
cricket::ContentInfo& content = new_remote_description->contents()[i];
|
|
if (!content.name.empty()) {
|
|
continue;
|
|
}
|
|
std::string new_mid;
|
|
absl::string_view source_explanation;
|
|
if (IsUnifiedPlan()) {
|
|
if (i < local_contents.size()) {
|
|
new_mid = local_contents[i].name;
|
|
source_explanation = "from the matching local media section";
|
|
} else if (i < remote_contents.size()) {
|
|
new_mid = remote_contents[i].name;
|
|
source_explanation = "from the matching previous remote media section";
|
|
} else {
|
|
new_mid = mid_generator_();
|
|
source_explanation = "generated just now";
|
|
}
|
|
} else {
|
|
new_mid = std::string(
|
|
GetDefaultMidForPlanB(content.media_description()->type()));
|
|
source_explanation = "to match pre-existing behavior";
|
|
}
|
|
RTC_DCHECK(!new_mid.empty());
|
|
content.name = new_mid;
|
|
new_remote_description->transport_infos()[i].content_name = new_mid;
|
|
RTC_LOG(LS_INFO) << "SetRemoteDescription: Remote media section at i=" << i
|
|
<< " is missing an a=mid line. Filling in the value '"
|
|
<< new_mid << "' " << source_explanation << ".";
|
|
}
|
|
}
|
|
|
|
void PeerConnection::SetRemoteDescription(
|
|
SetSessionDescriptionObserver* observer,
|
|
SessionDescriptionInterface* desc_ptr) {
|
|
RTC_DCHECK_RUN_ON(signaling_thread());
|
|
sdp_handler_.SetRemoteDescription(observer, desc_ptr);
|
|
}
|
|
|
|
void PeerConnection::SetRemoteDescription(
|
|
std::unique_ptr<SessionDescriptionInterface> desc,
|
|
rtc::scoped_refptr<SetRemoteDescriptionObserverInterface> observer) {
|
|
RTC_DCHECK_RUN_ON(signaling_thread());
|
|
sdp_handler_.SetRemoteDescription(std::move(desc), observer);
|
|
}
|
|
|
|
void PeerConnection::ProcessRemovalOfRemoteTrack(
|
|
rtc::scoped_refptr<RtpTransceiverProxyWithInternal<RtpTransceiver>>
|
|
transceiver,
|
|
std::vector<rtc::scoped_refptr<RtpTransceiverInterface>>* remove_list,
|
|
std::vector<rtc::scoped_refptr<MediaStreamInterface>>* removed_streams) {
|
|
RTC_DCHECK(transceiver->mid());
|
|
RTC_LOG(LS_INFO) << "Processing the removal of a track for MID="
|
|
<< *transceiver->mid();
|
|
std::vector<rtc::scoped_refptr<MediaStreamInterface>> previous_streams =
|
|
transceiver->internal()->receiver_internal()->streams();
|
|
// This will remove the remote track from the streams.
|
|
transceiver->internal()->receiver_internal()->set_stream_ids({});
|
|
remove_list->push_back(transceiver);
|
|
RemoveRemoteStreamsIfEmpty(previous_streams, removed_streams);
|
|
}
|
|
|
|
void PeerConnection::RemoveRemoteStreamsIfEmpty(
|
|
const std::vector<rtc::scoped_refptr<MediaStreamInterface>>& remote_streams,
|
|
std::vector<rtc::scoped_refptr<MediaStreamInterface>>* removed_streams) {
|
|
RTC_DCHECK_RUN_ON(signaling_thread());
|
|
// TODO(https://crbug.com/webrtc/9480): When we use stream IDs instead of
|
|
// streams, see if the stream was removed by checking if this was the last
|
|
// receiver with that stream ID.
|
|
for (const auto& remote_stream : remote_streams) {
|
|
if (remote_stream->GetAudioTracks().empty() &&
|
|
remote_stream->GetVideoTracks().empty()) {
|
|
remote_streams_->RemoveStream(remote_stream);
|
|
removed_streams->push_back(remote_stream);
|
|
}
|
|
}
|
|
}
|
|
|
|
rtc::scoped_refptr<RtpTransceiverProxyWithInternal<RtpTransceiver>>
|
|
PeerConnection::GetAssociatedTransceiver(const std::string& mid) const {
|
|
RTC_DCHECK_RUN_ON(signaling_thread());
|
|
RTC_DCHECK(IsUnifiedPlan());
|
|
return transceivers_.FindByMid(mid);
|
|
}
|
|
|
|
rtc::scoped_refptr<RtpTransceiverProxyWithInternal<RtpTransceiver>>
|
|
PeerConnection::GetTransceiverByMLineIndex(size_t mline_index) const {
|
|
RTC_DCHECK_RUN_ON(signaling_thread());
|
|
RTC_DCHECK(IsUnifiedPlan());
|
|
return transceivers_.FindByMLineIndex(mline_index);
|
|
}
|
|
|
|
rtc::scoped_refptr<RtpTransceiverProxyWithInternal<RtpTransceiver>>
|
|
PeerConnection::FindAvailableTransceiverToReceive(
|
|
cricket::MediaType media_type) const {
|
|
RTC_DCHECK_RUN_ON(signaling_thread());
|
|
RTC_DCHECK(IsUnifiedPlan());
|
|
// From JSEP section 5.10 (Applying a Remote Description):
|
|
// If the m= section is sendrecv or recvonly, and there are RtpTransceivers of
|
|
// the same type that were added to the PeerConnection by addTrack and are not
|
|
// associated with any m= section and are not stopped, find the first such
|
|
// RtpTransceiver.
|
|
for (auto transceiver : transceivers_.List()) {
|
|
if (transceiver->media_type() == media_type &&
|
|
transceiver->internal()->created_by_addtrack() && !transceiver->mid() &&
|
|
!transceiver->stopped()) {
|
|
return transceiver;
|
|
}
|
|
}
|
|
return nullptr;
|
|
}
|
|
|
|
const cricket::ContentInfo* PeerConnection::FindMediaSectionForTransceiver(
|
|
rtc::scoped_refptr<RtpTransceiverProxyWithInternal<RtpTransceiver>>
|
|
transceiver,
|
|
const SessionDescriptionInterface* sdesc) const {
|
|
RTC_DCHECK_RUN_ON(signaling_thread());
|
|
RTC_DCHECK(transceiver);
|
|
RTC_DCHECK(sdesc);
|
|
if (IsUnifiedPlan()) {
|
|
if (!transceiver->internal()->mid()) {
|
|
// This transceiver is not associated with a media section yet.
|
|
return nullptr;
|
|
}
|
|
return sdesc->description()->GetContentByName(
|
|
*transceiver->internal()->mid());
|
|
} else {
|
|
// Plan B only allows at most one audio and one video section, so use the
|
|
// first media section of that type.
|
|
return cricket::GetFirstMediaContent(sdesc->description()->contents(),
|
|
transceiver->media_type());
|
|
}
|
|
}
|
|
|
|
PeerConnectionInterface::RTCConfiguration PeerConnection::GetConfiguration() {
|
|
RTC_DCHECK_RUN_ON(signaling_thread());
|
|
return configuration_;
|
|
}
|
|
|
|
RTCError PeerConnection::SetConfiguration(
|
|
const RTCConfiguration& configuration) {
|
|
RTC_DCHECK_RUN_ON(signaling_thread());
|
|
TRACE_EVENT0("webrtc", "PeerConnection::SetConfiguration");
|
|
if (IsClosed()) {
|
|
LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_STATE,
|
|
"SetConfiguration: PeerConnection is closed.");
|
|
}
|
|
|
|
// According to JSEP, after setLocalDescription, changing the candidate pool
|
|
// size is not allowed, and changing the set of ICE servers will not result
|
|
// in new candidates being gathered.
|
|
if (local_description() && configuration.ice_candidate_pool_size !=
|
|
configuration_.ice_candidate_pool_size) {
|
|
LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_MODIFICATION,
|
|
"Can't change candidate pool size after calling "
|
|
"SetLocalDescription.");
|
|
}
|
|
|
|
if (local_description() &&
|
|
configuration.crypto_options != configuration_.crypto_options) {
|
|
LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_MODIFICATION,
|
|
"Can't change crypto_options after calling "
|
|
"SetLocalDescription.");
|
|
}
|
|
|
|
// The simplest (and most future-compatible) way to tell if the config was
|
|
// modified in an invalid way is to copy each property we do support
|
|
// modifying, then use operator==. There are far more properties we don't
|
|
// support modifying than those we do, and more could be added.
|
|
RTCConfiguration modified_config = configuration_;
|
|
modified_config.servers = configuration.servers;
|
|
modified_config.type = configuration.type;
|
|
modified_config.ice_candidate_pool_size =
|
|
configuration.ice_candidate_pool_size;
|
|
modified_config.prune_turn_ports = configuration.prune_turn_ports;
|
|
modified_config.turn_port_prune_policy = configuration.turn_port_prune_policy;
|
|
modified_config.surface_ice_candidates_on_ice_transport_type_changed =
|
|
configuration.surface_ice_candidates_on_ice_transport_type_changed;
|
|
modified_config.ice_check_min_interval = configuration.ice_check_min_interval;
|
|
modified_config.ice_check_interval_strong_connectivity =
|
|
configuration.ice_check_interval_strong_connectivity;
|
|
modified_config.ice_check_interval_weak_connectivity =
|
|
configuration.ice_check_interval_weak_connectivity;
|
|
modified_config.ice_unwritable_timeout = configuration.ice_unwritable_timeout;
|
|
modified_config.ice_unwritable_min_checks =
|
|
configuration.ice_unwritable_min_checks;
|
|
modified_config.ice_inactive_timeout = configuration.ice_inactive_timeout;
|
|
modified_config.stun_candidate_keepalive_interval =
|
|
configuration.stun_candidate_keepalive_interval;
|
|
modified_config.turn_customizer = configuration.turn_customizer;
|
|
modified_config.network_preference = configuration.network_preference;
|
|
modified_config.active_reset_srtp_params =
|
|
configuration.active_reset_srtp_params;
|
|
modified_config.turn_logging_id = configuration.turn_logging_id;
|
|
modified_config.allow_codec_switching = configuration.allow_codec_switching;
|
|
if (configuration != modified_config) {
|
|
LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_MODIFICATION,
|
|
"Modifying the configuration in an unsupported way.");
|
|
}
|
|
|
|
// Validate the modified configuration.
|
|
RTCError validate_error = ValidateConfiguration(modified_config);
|
|
if (!validate_error.ok()) {
|
|
return validate_error;
|
|
}
|
|
|
|
// Note that this isn't possible through chromium, since it's an unsigned
|
|
// short in WebIDL.
|
|
if (configuration.ice_candidate_pool_size < 0 ||
|
|
configuration.ice_candidate_pool_size > static_cast<int>(UINT16_MAX)) {
|
|
return RTCError(RTCErrorType::INVALID_RANGE);
|
|
}
|
|
|
|
// Parse ICE servers before hopping to network thread.
|
|
cricket::ServerAddresses stun_servers;
|
|
std::vector<cricket::RelayServerConfig> turn_servers;
|
|
RTCErrorType parse_error =
|
|
ParseIceServers(configuration.servers, &stun_servers, &turn_servers);
|
|
if (parse_error != RTCErrorType::NONE) {
|
|
return RTCError(parse_error);
|
|
}
|
|
// Add the turn logging id to all turn servers
|
|
for (cricket::RelayServerConfig& turn_server : turn_servers) {
|
|
turn_server.turn_logging_id = configuration.turn_logging_id;
|
|
}
|
|
|
|
// Note if STUN or TURN servers were supplied.
|
|
if (!stun_servers.empty()) {
|
|
NoteUsageEvent(UsageEvent::STUN_SERVER_ADDED);
|
|
}
|
|
if (!turn_servers.empty()) {
|
|
NoteUsageEvent(UsageEvent::TURN_SERVER_ADDED);
|
|
}
|
|
|
|
// In theory this shouldn't fail.
|
|
if (!network_thread()->Invoke<bool>(
|
|
RTC_FROM_HERE,
|
|
rtc::Bind(&PeerConnection::ReconfigurePortAllocator_n, this,
|
|
stun_servers, turn_servers, modified_config.type,
|
|
modified_config.ice_candidate_pool_size,
|
|
modified_config.GetTurnPortPrunePolicy(),
|
|
modified_config.turn_customizer,
|
|
modified_config.stun_candidate_keepalive_interval,
|
|
static_cast<bool>(local_description())))) {
|
|
LOG_AND_RETURN_ERROR(RTCErrorType::INTERNAL_ERROR,
|
|
"Failed to apply configuration to PortAllocator.");
|
|
}
|
|
|
|
// As described in JSEP, calling setConfiguration with new ICE servers or
|
|
// candidate policy must set a "needs-ice-restart" bit so that the next offer
|
|
// triggers an ICE restart which will pick up the changes.
|
|
if (modified_config.servers != configuration_.servers ||
|
|
NeedIceRestart(
|
|
configuration_.surface_ice_candidates_on_ice_transport_type_changed,
|
|
configuration_.type, modified_config.type) ||
|
|
modified_config.GetTurnPortPrunePolicy() !=
|
|
configuration_.GetTurnPortPrunePolicy()) {
|
|
transport_controller_->SetNeedsIceRestartFlag();
|
|
}
|
|
|
|
transport_controller_->SetIceConfig(ParseIceConfig(modified_config));
|
|
|
|
if (configuration_.active_reset_srtp_params !=
|
|
modified_config.active_reset_srtp_params) {
|
|
transport_controller_->SetActiveResetSrtpParams(
|
|
modified_config.active_reset_srtp_params);
|
|
}
|
|
|
|
if (modified_config.allow_codec_switching.has_value()) {
|
|
std::vector<cricket::VideoMediaChannel*> channels;
|
|
for (const auto& transceiver : transceivers_.List()) {
|
|
if (transceiver->media_type() != cricket::MEDIA_TYPE_VIDEO)
|
|
continue;
|
|
|
|
auto* video_channel = static_cast<cricket::VideoChannel*>(
|
|
transceiver->internal()->channel());
|
|
if (video_channel)
|
|
channels.push_back(video_channel->media_channel());
|
|
}
|
|
|
|
worker_thread()->Invoke<void>(
|
|
RTC_FROM_HERE,
|
|
[channels = std::move(channels),
|
|
allow_codec_switching = *modified_config.allow_codec_switching]() {
|
|
for (auto* ch : channels)
|
|
ch->SetVideoCodecSwitchingEnabled(allow_codec_switching);
|
|
});
|
|
}
|
|
|
|
configuration_ = modified_config;
|
|
return RTCError::OK();
|
|
}
|
|
|
|
bool PeerConnection::AddIceCandidate(
|
|
const IceCandidateInterface* ice_candidate) {
|
|
RTC_DCHECK_RUN_ON(signaling_thread());
|
|
return sdp_handler_.AddIceCandidate(ice_candidate);
|
|
}
|
|
|
|
void PeerConnection::AddIceCandidate(
|
|
std::unique_ptr<IceCandidateInterface> candidate,
|
|
std::function<void(RTCError)> callback) {
|
|
RTC_DCHECK_RUN_ON(signaling_thread());
|
|
sdp_handler_.AddIceCandidate(std::move(candidate), callback);
|
|
}
|
|
|
|
bool PeerConnection::RemoveIceCandidates(
|
|
const std::vector<cricket::Candidate>& candidates) {
|
|
TRACE_EVENT0("webrtc", "PeerConnection::RemoveIceCandidates");
|
|
RTC_DCHECK_RUN_ON(signaling_thread());
|
|
return sdp_handler_.RemoveIceCandidates(candidates);
|
|
}
|
|
|
|
RTCError PeerConnection::SetBitrate(const BitrateSettings& bitrate) {
|
|
if (!worker_thread()->IsCurrent()) {
|
|
return worker_thread()->Invoke<RTCError>(
|
|
RTC_FROM_HERE, [&]() { return SetBitrate(bitrate); });
|
|
}
|
|
RTC_DCHECK_RUN_ON(worker_thread());
|
|
|
|
const bool has_min = bitrate.min_bitrate_bps.has_value();
|
|
const bool has_start = bitrate.start_bitrate_bps.has_value();
|
|
const bool has_max = bitrate.max_bitrate_bps.has_value();
|
|
if (has_min && *bitrate.min_bitrate_bps < 0) {
|
|
LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_PARAMETER,
|
|
"min_bitrate_bps <= 0");
|
|
}
|
|
if (has_start) {
|
|
if (has_min && *bitrate.start_bitrate_bps < *bitrate.min_bitrate_bps) {
|
|
LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_PARAMETER,
|
|
"start_bitrate_bps < min_bitrate_bps");
|
|
} else if (*bitrate.start_bitrate_bps < 0) {
|
|
LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_PARAMETER,
|
|
"curent_bitrate_bps < 0");
|
|
}
|
|
}
|
|
if (has_max) {
|
|
if (has_start && *bitrate.max_bitrate_bps < *bitrate.start_bitrate_bps) {
|
|
LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_PARAMETER,
|
|
"max_bitrate_bps < start_bitrate_bps");
|
|
} else if (has_min && *bitrate.max_bitrate_bps < *bitrate.min_bitrate_bps) {
|
|
LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_PARAMETER,
|
|
"max_bitrate_bps < min_bitrate_bps");
|
|
} else if (*bitrate.max_bitrate_bps < 0) {
|
|
LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_PARAMETER,
|
|
"max_bitrate_bps < 0");
|
|
}
|
|
}
|
|
|
|
RTC_DCHECK(call_.get());
|
|
call_->SetClientBitratePreferences(bitrate);
|
|
|
|
return RTCError::OK();
|
|
}
|
|
|
|
void PeerConnection::SetAudioPlayout(bool playout) {
|
|
if (!worker_thread()->IsCurrent()) {
|
|
worker_thread()->Invoke<void>(
|
|
RTC_FROM_HERE,
|
|
rtc::Bind(&PeerConnection::SetAudioPlayout, this, playout));
|
|
return;
|
|
}
|
|
auto audio_state =
|
|
factory_->channel_manager()->media_engine()->voice().GetAudioState();
|
|
audio_state->SetPlayout(playout);
|
|
}
|
|
|
|
void PeerConnection::SetAudioRecording(bool recording) {
|
|
if (!worker_thread()->IsCurrent()) {
|
|
worker_thread()->Invoke<void>(
|
|
RTC_FROM_HERE,
|
|
rtc::Bind(&PeerConnection::SetAudioRecording, this, recording));
|
|
return;
|
|
}
|
|
auto audio_state =
|
|
factory_->channel_manager()->media_engine()->voice().GetAudioState();
|
|
audio_state->SetRecording(recording);
|
|
}
|
|
|
|
std::unique_ptr<rtc::SSLCertificate>
|
|
PeerConnection::GetRemoteAudioSSLCertificate() {
|
|
std::unique_ptr<rtc::SSLCertChain> chain = GetRemoteAudioSSLCertChain();
|
|
if (!chain || !chain->GetSize()) {
|
|
return nullptr;
|
|
}
|
|
return chain->Get(0).Clone();
|
|
}
|
|
|
|
std::unique_ptr<rtc::SSLCertChain>
|
|
PeerConnection::GetRemoteAudioSSLCertChain() {
|
|
RTC_DCHECK_RUN_ON(signaling_thread());
|
|
auto audio_transceiver = GetFirstAudioTransceiver();
|
|
if (!audio_transceiver || !audio_transceiver->internal()->channel()) {
|
|
return nullptr;
|
|
}
|
|
return transport_controller_->GetRemoteSSLCertChain(
|
|
audio_transceiver->internal()->channel()->transport_name());
|
|
}
|
|
|
|
rtc::scoped_refptr<RtpTransceiverProxyWithInternal<RtpTransceiver>>
|
|
PeerConnection::GetFirstAudioTransceiver() const {
|
|
for (auto transceiver : transceivers_.List()) {
|
|
if (transceiver->media_type() == cricket::MEDIA_TYPE_AUDIO) {
|
|
return transceiver;
|
|
}
|
|
}
|
|
return nullptr;
|
|
}
|
|
|
|
void PeerConnection::AddAdaptationResource(
|
|
rtc::scoped_refptr<Resource> resource) {
|
|
if (!worker_thread()->IsCurrent()) {
|
|
return worker_thread()->Invoke<void>(RTC_FROM_HERE, [this, resource]() {
|
|
return AddAdaptationResource(resource);
|
|
});
|
|
}
|
|
RTC_DCHECK_RUN_ON(worker_thread());
|
|
if (!call_) {
|
|
// The PeerConnection has been closed.
|
|
return;
|
|
}
|
|
call_->AddAdaptationResource(resource);
|
|
}
|
|
|
|
bool PeerConnection::StartRtcEventLog(std::unique_ptr<RtcEventLogOutput> output,
|
|
int64_t output_period_ms) {
|
|
return worker_thread()->Invoke<bool>(
|
|
RTC_FROM_HERE,
|
|
[this, output = std::move(output), output_period_ms]() mutable {
|
|
return StartRtcEventLog_w(std::move(output), output_period_ms);
|
|
});
|
|
}
|
|
|
|
bool PeerConnection::StartRtcEventLog(
|
|
std::unique_ptr<RtcEventLogOutput> output) {
|
|
int64_t output_period_ms = webrtc::RtcEventLog::kImmediateOutput;
|
|
if (absl::StartsWith(factory_->trials().Lookup("WebRTC-RtcEventLogNewFormat"),
|
|
"Enabled")) {
|
|
output_period_ms = 5000;
|
|
}
|
|
return StartRtcEventLog(std::move(output), output_period_ms);
|
|
}
|
|
|
|
void PeerConnection::StopRtcEventLog() {
|
|
worker_thread()->Invoke<void>(
|
|
RTC_FROM_HERE, rtc::Bind(&PeerConnection::StopRtcEventLog_w, this));
|
|
}
|
|
|
|
rtc::scoped_refptr<DtlsTransportInterface>
|
|
PeerConnection::LookupDtlsTransportByMid(const std::string& mid) {
|
|
RTC_DCHECK_RUN_ON(signaling_thread());
|
|
return transport_controller_->LookupDtlsTransportByMid(mid);
|
|
}
|
|
|
|
rtc::scoped_refptr<DtlsTransport>
|
|
PeerConnection::LookupDtlsTransportByMidInternal(const std::string& mid) {
|
|
RTC_DCHECK_RUN_ON(signaling_thread());
|
|
return transport_controller_->LookupDtlsTransportByMid(mid);
|
|
}
|
|
|
|
rtc::scoped_refptr<SctpTransportInterface> PeerConnection::GetSctpTransport()
|
|
const {
|
|
RTC_DCHECK_RUN_ON(signaling_thread());
|
|
if (!sctp_mid_s_) {
|
|
return nullptr;
|
|
}
|
|
return transport_controller_->GetSctpTransport(*sctp_mid_s_);
|
|
}
|
|
|
|
const SessionDescriptionInterface* PeerConnection::local_description() const {
|
|
RTC_DCHECK_RUN_ON(signaling_thread());
|
|
return sdp_handler_.local_description();
|
|
}
|
|
|
|
const SessionDescriptionInterface* PeerConnection::remote_description() const {
|
|
RTC_DCHECK_RUN_ON(signaling_thread());
|
|
return sdp_handler_.remote_description();
|
|
}
|
|
|
|
const SessionDescriptionInterface* PeerConnection::current_local_description()
|
|
const {
|
|
RTC_DCHECK_RUN_ON(signaling_thread());
|
|
return sdp_handler_.current_local_description();
|
|
}
|
|
|
|
const SessionDescriptionInterface* PeerConnection::current_remote_description()
|
|
const {
|
|
RTC_DCHECK_RUN_ON(signaling_thread());
|
|
return sdp_handler_.current_remote_description();
|
|
}
|
|
|
|
const SessionDescriptionInterface* PeerConnection::pending_local_description()
|
|
const {
|
|
RTC_DCHECK_RUN_ON(signaling_thread());
|
|
return sdp_handler_.pending_local_description();
|
|
}
|
|
|
|
const SessionDescriptionInterface* PeerConnection::pending_remote_description()
|
|
const {
|
|
RTC_DCHECK_RUN_ON(signaling_thread());
|
|
return sdp_handler_.pending_remote_description();
|
|
}
|
|
|
|
void PeerConnection::Close() {
|
|
RTC_DCHECK_RUN_ON(signaling_thread());
|
|
TRACE_EVENT0("webrtc", "PeerConnection::Close");
|
|
|
|
if (IsClosed()) {
|
|
return;
|
|
}
|
|
// Update stats here so that we have the most recent stats for tracks and
|
|
// streams before the channels are closed.
|
|
stats_->UpdateStats(kStatsOutputLevelStandard);
|
|
|
|
ice_connection_state_ = PeerConnectionInterface::kIceConnectionClosed;
|
|
Observer()->OnIceConnectionChange(ice_connection_state_);
|
|
standardized_ice_connection_state_ =
|
|
PeerConnectionInterface::IceConnectionState::kIceConnectionClosed;
|
|
connection_state_ = PeerConnectionInterface::PeerConnectionState::kClosed;
|
|
Observer()->OnConnectionChange(connection_state_);
|
|
|
|
sdp_handler_.Close();
|
|
|
|
NoteUsageEvent(UsageEvent::CLOSE_CALLED);
|
|
|
|
for (const auto& transceiver : transceivers_.List()) {
|
|
transceiver->internal()->SetPeerConnectionClosed();
|
|
if (!transceiver->stopped())
|
|
transceiver->StopInternal();
|
|
}
|
|
|
|
// Ensure that all asynchronous stats requests are completed before destroying
|
|
// the transport controller below.
|
|
if (stats_collector_) {
|
|
stats_collector_->WaitForPendingRequest();
|
|
}
|
|
|
|
// Don't destroy BaseChannels until after stats has been cleaned up so that
|
|
// the last stats request can still read from the channels.
|
|
DestroyAllChannels();
|
|
|
|
// The event log is used in the transport controller, which must be outlived
|
|
// by the former. CreateOffer by the peer connection is implemented
|
|
// asynchronously and if the peer connection is closed without resetting the
|
|
// WebRTC session description factory, the session description factory would
|
|
// call the transport controller.
|
|
sdp_handler_.ResetSessionDescFactory();
|
|
transport_controller_.reset();
|
|
|
|
network_thread()->Invoke<void>(
|
|
RTC_FROM_HERE, rtc::Bind(&cricket::PortAllocator::DiscardCandidatePool,
|
|
port_allocator_.get()));
|
|
|
|
worker_thread()->Invoke<void>(RTC_FROM_HERE, [this] {
|
|
RTC_DCHECK_RUN_ON(worker_thread());
|
|
call_safety_.reset();
|
|
call_.reset();
|
|
// The event log must outlive call (and any other object that uses it).
|
|
event_log_.reset();
|
|
});
|
|
ReportUsagePattern();
|
|
// The .h file says that observer can be discarded after close() returns.
|
|
// Make sure this is true.
|
|
observer_ = nullptr;
|
|
}
|
|
|
|
void PeerConnection::OnMessage(rtc::Message* msg) {
|
|
RTC_DCHECK_RUN_ON(signaling_thread());
|
|
switch (msg->message_id) {
|
|
case MSG_SET_SESSIONDESCRIPTION_SUCCESS: {
|
|
SetSessionDescriptionMsg* param =
|
|
static_cast<SetSessionDescriptionMsg*>(msg->pdata);
|
|
param->observer->OnSuccess();
|
|
delete param;
|
|
break;
|
|
}
|
|
case MSG_SET_SESSIONDESCRIPTION_FAILED: {
|
|
SetSessionDescriptionMsg* param =
|
|
static_cast<SetSessionDescriptionMsg*>(msg->pdata);
|
|
param->observer->OnFailure(std::move(param->error));
|
|
delete param;
|
|
break;
|
|
}
|
|
case MSG_CREATE_SESSIONDESCRIPTION_FAILED: {
|
|
CreateSessionDescriptionMsg* param =
|
|
static_cast<CreateSessionDescriptionMsg*>(msg->pdata);
|
|
param->observer->OnFailure(std::move(param->error));
|
|
delete param;
|
|
break;
|
|
}
|
|
case MSG_GETSTATS: {
|
|
GetStatsMsg* param = static_cast<GetStatsMsg*>(msg->pdata);
|
|
StatsReports reports;
|
|
stats_->GetStats(param->track, &reports);
|
|
param->observer->OnComplete(reports);
|
|
delete param;
|
|
break;
|
|
}
|
|
case MSG_REPORT_USAGE_PATTERN: {
|
|
ReportUsagePattern();
|
|
break;
|
|
}
|
|
default:
|
|
RTC_NOTREACHED() << "Not implemented";
|
|
break;
|
|
}
|
|
}
|
|
|
|
cricket::VoiceMediaChannel* PeerConnection::voice_media_channel() const {
|
|
RTC_DCHECK(!IsUnifiedPlan());
|
|
auto* voice_channel = static_cast<cricket::VoiceChannel*>(
|
|
GetAudioTransceiver()->internal()->channel());
|
|
if (voice_channel) {
|
|
return voice_channel->media_channel();
|
|
} else {
|
|
return nullptr;
|
|
}
|
|
}
|
|
|
|
cricket::VideoMediaChannel* PeerConnection::video_media_channel() const {
|
|
RTC_DCHECK(!IsUnifiedPlan());
|
|
auto* video_channel = static_cast<cricket::VideoChannel*>(
|
|
GetVideoTransceiver()->internal()->channel());
|
|
if (video_channel) {
|
|
return video_channel->media_channel();
|
|
} else {
|
|
return nullptr;
|
|
}
|
|
}
|
|
|
|
void PeerConnection::CreateAudioReceiver(
|
|
MediaStreamInterface* stream,
|
|
const RtpSenderInfo& remote_sender_info) {
|
|
std::vector<rtc::scoped_refptr<MediaStreamInterface>> streams;
|
|
streams.push_back(rtc::scoped_refptr<MediaStreamInterface>(stream));
|
|
// TODO(https://crbug.com/webrtc/9480): When we remove remote_streams(), use
|
|
// the constructor taking stream IDs instead.
|
|
auto* audio_receiver = new AudioRtpReceiver(
|
|
worker_thread(), remote_sender_info.sender_id, streams);
|
|
audio_receiver->SetMediaChannel(voice_media_channel());
|
|
if (remote_sender_info.sender_id == kDefaultAudioSenderId) {
|
|
audio_receiver->SetupUnsignaledMediaChannel();
|
|
} else {
|
|
audio_receiver->SetupMediaChannel(remote_sender_info.first_ssrc);
|
|
}
|
|
auto receiver = RtpReceiverProxyWithInternal<RtpReceiverInternal>::Create(
|
|
signaling_thread(), audio_receiver);
|
|
GetAudioTransceiver()->internal()->AddReceiver(receiver);
|
|
Observer()->OnAddTrack(receiver, streams);
|
|
NoteUsageEvent(UsageEvent::AUDIO_ADDED);
|
|
}
|
|
|
|
void PeerConnection::CreateVideoReceiver(
|
|
MediaStreamInterface* stream,
|
|
const RtpSenderInfo& remote_sender_info) {
|
|
std::vector<rtc::scoped_refptr<MediaStreamInterface>> streams;
|
|
streams.push_back(rtc::scoped_refptr<MediaStreamInterface>(stream));
|
|
// TODO(https://crbug.com/webrtc/9480): When we remove remote_streams(), use
|
|
// the constructor taking stream IDs instead.
|
|
auto* video_receiver = new VideoRtpReceiver(
|
|
worker_thread(), remote_sender_info.sender_id, streams);
|
|
video_receiver->SetMediaChannel(video_media_channel());
|
|
if (remote_sender_info.sender_id == kDefaultVideoSenderId) {
|
|
video_receiver->SetupUnsignaledMediaChannel();
|
|
} else {
|
|
video_receiver->SetupMediaChannel(remote_sender_info.first_ssrc);
|
|
}
|
|
auto receiver = RtpReceiverProxyWithInternal<RtpReceiverInternal>::Create(
|
|
signaling_thread(), video_receiver);
|
|
GetVideoTransceiver()->internal()->AddReceiver(receiver);
|
|
Observer()->OnAddTrack(receiver, streams);
|
|
NoteUsageEvent(UsageEvent::VIDEO_ADDED);
|
|
}
|
|
|
|
// TODO(deadbeef): Keep RtpReceivers around even if track goes away in remote
|
|
// description.
|
|
rtc::scoped_refptr<RtpReceiverInterface> PeerConnection::RemoveAndStopReceiver(
|
|
const RtpSenderInfo& remote_sender_info) {
|
|
auto receiver = FindReceiverById(remote_sender_info.sender_id);
|
|
if (!receiver) {
|
|
RTC_LOG(LS_WARNING) << "RtpReceiver for track with id "
|
|
<< remote_sender_info.sender_id << " doesn't exist.";
|
|
return nullptr;
|
|
}
|
|
if (receiver->media_type() == cricket::MEDIA_TYPE_AUDIO) {
|
|
GetAudioTransceiver()->internal()->RemoveReceiver(receiver);
|
|
} else {
|
|
GetVideoTransceiver()->internal()->RemoveReceiver(receiver);
|
|
}
|
|
return receiver;
|
|
}
|
|
|
|
void PeerConnection::AddAudioTrack(AudioTrackInterface* track,
|
|
MediaStreamInterface* stream) {
|
|
RTC_DCHECK(!IsClosed());
|
|
RTC_DCHECK(track);
|
|
RTC_DCHECK(stream);
|
|
auto sender = FindSenderForTrack(track);
|
|
if (sender) {
|
|
// We already have a sender for this track, so just change the stream_id
|
|
// so that it's correct in the next call to CreateOffer.
|
|
sender->internal()->set_stream_ids({stream->id()});
|
|
return;
|
|
}
|
|
|
|
// Normal case; we've never seen this track before.
|
|
auto new_sender = CreateSender(cricket::MEDIA_TYPE_AUDIO, track->id(), track,
|
|
{stream->id()}, {});
|
|
new_sender->internal()->SetMediaChannel(voice_media_channel());
|
|
GetAudioTransceiver()->internal()->AddSender(new_sender);
|
|
// If the sender has already been configured in SDP, we call SetSsrc,
|
|
// which will connect the sender to the underlying transport. This can
|
|
// occur if a local session description that contains the ID of the sender
|
|
// is set before AddStream is called. It can also occur if the local
|
|
// session description is not changed and RemoveStream is called, and
|
|
// later AddStream is called again with the same stream.
|
|
const RtpSenderInfo* sender_info =
|
|
FindSenderInfo(local_audio_sender_infos_, stream->id(), track->id());
|
|
if (sender_info) {
|
|
new_sender->internal()->SetSsrc(sender_info->first_ssrc);
|
|
}
|
|
}
|
|
|
|
// TODO(deadbeef): Don't destroy RtpSenders here; they should be kept around
|
|
// indefinitely, when we have unified plan SDP.
|
|
void PeerConnection::RemoveAudioTrack(AudioTrackInterface* track,
|
|
MediaStreamInterface* stream) {
|
|
RTC_DCHECK(!IsClosed());
|
|
auto sender = FindSenderForTrack(track);
|
|
if (!sender) {
|
|
RTC_LOG(LS_WARNING) << "RtpSender for track with id " << track->id()
|
|
<< " doesn't exist.";
|
|
return;
|
|
}
|
|
GetAudioTransceiver()->internal()->RemoveSender(sender);
|
|
}
|
|
|
|
void PeerConnection::AddVideoTrack(VideoTrackInterface* track,
|
|
MediaStreamInterface* stream) {
|
|
RTC_DCHECK(!IsClosed());
|
|
RTC_DCHECK(track);
|
|
RTC_DCHECK(stream);
|
|
auto sender = FindSenderForTrack(track);
|
|
if (sender) {
|
|
// We already have a sender for this track, so just change the stream_id
|
|
// so that it's correct in the next call to CreateOffer.
|
|
sender->internal()->set_stream_ids({stream->id()});
|
|
return;
|
|
}
|
|
|
|
// Normal case; we've never seen this track before.
|
|
auto new_sender = CreateSender(cricket::MEDIA_TYPE_VIDEO, track->id(), track,
|
|
{stream->id()}, {});
|
|
new_sender->internal()->SetMediaChannel(video_media_channel());
|
|
GetVideoTransceiver()->internal()->AddSender(new_sender);
|
|
const RtpSenderInfo* sender_info =
|
|
FindSenderInfo(local_video_sender_infos_, stream->id(), track->id());
|
|
if (sender_info) {
|
|
new_sender->internal()->SetSsrc(sender_info->first_ssrc);
|
|
}
|
|
}
|
|
|
|
void PeerConnection::RemoveVideoTrack(VideoTrackInterface* track,
|
|
MediaStreamInterface* stream) {
|
|
RTC_DCHECK(!IsClosed());
|
|
auto sender = FindSenderForTrack(track);
|
|
if (!sender) {
|
|
RTC_LOG(LS_WARNING) << "RtpSender for track with id " << track->id()
|
|
<< " doesn't exist.";
|
|
return;
|
|
}
|
|
GetVideoTransceiver()->internal()->RemoveSender(sender);
|
|
}
|
|
|
|
void PeerConnection::SetIceConnectionState(IceConnectionState new_state) {
|
|
RTC_DCHECK_RUN_ON(signaling_thread());
|
|
if (ice_connection_state_ == new_state) {
|
|
return;
|
|
}
|
|
|
|
// After transitioning to "closed", ignore any additional states from
|
|
// TransportController (such as "disconnected").
|
|
if (IsClosed()) {
|
|
return;
|
|
}
|
|
|
|
RTC_LOG(LS_INFO) << "Changing IceConnectionState " << ice_connection_state_
|
|
<< " => " << new_state;
|
|
RTC_DCHECK(ice_connection_state_ !=
|
|
PeerConnectionInterface::kIceConnectionClosed);
|
|
|
|
ice_connection_state_ = new_state;
|
|
Observer()->OnIceConnectionChange(ice_connection_state_);
|
|
}
|
|
|
|
void PeerConnection::SetStandardizedIceConnectionState(
|
|
PeerConnectionInterface::IceConnectionState new_state) {
|
|
if (standardized_ice_connection_state_ == new_state) {
|
|
return;
|
|
}
|
|
|
|
if (IsClosed()) {
|
|
return;
|
|
}
|
|
|
|
RTC_LOG(LS_INFO) << "Changing standardized IceConnectionState "
|
|
<< standardized_ice_connection_state_ << " => " << new_state;
|
|
|
|
standardized_ice_connection_state_ = new_state;
|
|
Observer()->OnStandardizedIceConnectionChange(new_state);
|
|
}
|
|
|
|
void PeerConnection::SetConnectionState(
|
|
PeerConnectionInterface::PeerConnectionState new_state) {
|
|
if (connection_state_ == new_state)
|
|
return;
|
|
if (IsClosed())
|
|
return;
|
|
connection_state_ = new_state;
|
|
Observer()->OnConnectionChange(new_state);
|
|
}
|
|
|
|
void PeerConnection::OnIceGatheringChange(
|
|
PeerConnectionInterface::IceGatheringState new_state) {
|
|
if (IsClosed()) {
|
|
return;
|
|
}
|
|
ice_gathering_state_ = new_state;
|
|
Observer()->OnIceGatheringChange(ice_gathering_state_);
|
|
}
|
|
|
|
void PeerConnection::OnIceCandidate(
|
|
std::unique_ptr<IceCandidateInterface> candidate) {
|
|
if (IsClosed()) {
|
|
return;
|
|
}
|
|
ReportIceCandidateCollected(candidate->candidate());
|
|
Observer()->OnIceCandidate(candidate.get());
|
|
}
|
|
|
|
void PeerConnection::OnIceCandidateError(const std::string& address,
|
|
int port,
|
|
const std::string& url,
|
|
int error_code,
|
|
const std::string& error_text) {
|
|
if (IsClosed()) {
|
|
return;
|
|
}
|
|
Observer()->OnIceCandidateError(address, port, url, error_code, error_text);
|
|
// Leftover not to break wpt test during migration to the new API.
|
|
Observer()->OnIceCandidateError(address + ":", url, error_code, error_text);
|
|
}
|
|
|
|
void PeerConnection::OnIceCandidatesRemoved(
|
|
const std::vector<cricket::Candidate>& candidates) {
|
|
if (IsClosed()) {
|
|
return;
|
|
}
|
|
Observer()->OnIceCandidatesRemoved(candidates);
|
|
}
|
|
|
|
void PeerConnection::OnSelectedCandidatePairChanged(
|
|
const cricket::CandidatePairChangeEvent& event) {
|
|
if (IsClosed()) {
|
|
return;
|
|
}
|
|
|
|
if (event.selected_candidate_pair.local_candidate().type() ==
|
|
LOCAL_PORT_TYPE &&
|
|
event.selected_candidate_pair.remote_candidate().type() ==
|
|
LOCAL_PORT_TYPE) {
|
|
NoteUsageEvent(UsageEvent::DIRECT_CONNECTION_SELECTED);
|
|
}
|
|
|
|
Observer()->OnIceSelectedCandidatePairChanged(event);
|
|
}
|
|
|
|
void PeerConnection::OnAudioTrackAdded(AudioTrackInterface* track,
|
|
MediaStreamInterface* stream) {
|
|
if (IsClosed()) {
|
|
return;
|
|
}
|
|
AddAudioTrack(track, stream);
|
|
sdp_handler_.UpdateNegotiationNeeded();
|
|
}
|
|
|
|
void PeerConnection::OnAudioTrackRemoved(AudioTrackInterface* track,
|
|
MediaStreamInterface* stream) {
|
|
if (IsClosed()) {
|
|
return;
|
|
}
|
|
RemoveAudioTrack(track, stream);
|
|
sdp_handler_.UpdateNegotiationNeeded();
|
|
}
|
|
|
|
void PeerConnection::OnVideoTrackAdded(VideoTrackInterface* track,
|
|
MediaStreamInterface* stream) {
|
|
if (IsClosed()) {
|
|
return;
|
|
}
|
|
AddVideoTrack(track, stream);
|
|
sdp_handler_.UpdateNegotiationNeeded();
|
|
}
|
|
|
|
void PeerConnection::OnVideoTrackRemoved(VideoTrackInterface* track,
|
|
MediaStreamInterface* stream) {
|
|
if (IsClosed()) {
|
|
return;
|
|
}
|
|
RemoveVideoTrack(track, stream);
|
|
sdp_handler_.UpdateNegotiationNeeded();
|
|
}
|
|
|
|
void PeerConnection::PostSetSessionDescriptionSuccess(
|
|
SetSessionDescriptionObserver* observer) {
|
|
SetSessionDescriptionMsg* msg = new SetSessionDescriptionMsg(observer);
|
|
signaling_thread()->Post(RTC_FROM_HERE, this,
|
|
MSG_SET_SESSIONDESCRIPTION_SUCCESS, msg);
|
|
}
|
|
|
|
void PeerConnection::PostSetSessionDescriptionFailure(
|
|
SetSessionDescriptionObserver* observer,
|
|
RTCError&& error) {
|
|
RTC_DCHECK(!error.ok());
|
|
SetSessionDescriptionMsg* msg = new SetSessionDescriptionMsg(observer);
|
|
msg->error = std::move(error);
|
|
signaling_thread()->Post(RTC_FROM_HERE, this,
|
|
MSG_SET_SESSIONDESCRIPTION_FAILED, msg);
|
|
}
|
|
|
|
void PeerConnection::PostCreateSessionDescriptionFailure(
|
|
CreateSessionDescriptionObserver* observer,
|
|
RTCError error) {
|
|
RTC_DCHECK(!error.ok());
|
|
CreateSessionDescriptionMsg* msg = new CreateSessionDescriptionMsg(observer);
|
|
msg->error = std::move(error);
|
|
signaling_thread()->Post(RTC_FROM_HERE, this,
|
|
MSG_CREATE_SESSIONDESCRIPTION_FAILED, msg);
|
|
}
|
|
|
|
void PeerConnection::GetOptionsForOffer(
|
|
const PeerConnectionInterface::RTCOfferAnswerOptions& offer_answer_options,
|
|
cricket::MediaSessionOptions* session_options) {
|
|
RTC_DCHECK_RUN_ON(signaling_thread());
|
|
ExtractSharedMediaSessionOptions(offer_answer_options, session_options);
|
|
|
|
if (IsUnifiedPlan()) {
|
|
GetOptionsForUnifiedPlanOffer(offer_answer_options, session_options);
|
|
} else {
|
|
GetOptionsForPlanBOffer(offer_answer_options, session_options);
|
|
}
|
|
|
|
// Intentionally unset the data channel type for RTP data channel with the
|
|
// second condition. Otherwise the RTP data channels would be successfully
|
|
// negotiated by default and the unit tests in WebRtcDataBrowserTest will fail
|
|
// when building with chromium. We want to leave RTP data channels broken, so
|
|
// people won't try to use them.
|
|
if (data_channel_controller_.HasRtpDataChannels() ||
|
|
data_channel_type() != cricket::DCT_RTP) {
|
|
session_options->data_channel_type = data_channel_type();
|
|
}
|
|
|
|
// Apply ICE restart flag and renomination flag.
|
|
bool ice_restart =
|
|
offer_answer_options.ice_restart || sdp_handler_.HasNewIceCredentials();
|
|
for (auto& options : session_options->media_description_options) {
|
|
options.transport_options.ice_restart = ice_restart;
|
|
options.transport_options.enable_ice_renomination =
|
|
configuration_.enable_ice_renomination;
|
|
}
|
|
|
|
session_options->rtcp_cname = rtcp_cname_;
|
|
session_options->crypto_options = GetCryptoOptions();
|
|
session_options->pooled_ice_credentials =
|
|
network_thread()->Invoke<std::vector<cricket::IceParameters>>(
|
|
RTC_FROM_HERE,
|
|
rtc::Bind(&cricket::PortAllocator::GetPooledIceCredentials,
|
|
port_allocator_.get()));
|
|
session_options->offer_extmap_allow_mixed =
|
|
configuration_.offer_extmap_allow_mixed;
|
|
|
|
// Allow fallback for using obsolete SCTP syntax.
|
|
// Note that the default in |session_options| is true, while
|
|
// the default in |options| is false.
|
|
session_options->use_obsolete_sctp_sdp =
|
|
offer_answer_options.use_obsolete_sctp_sdp;
|
|
}
|
|
|
|
void PeerConnection::GetOptionsForPlanBOffer(
|
|
const PeerConnectionInterface::RTCOfferAnswerOptions& offer_answer_options,
|
|
cricket::MediaSessionOptions* session_options) {
|
|
// Figure out transceiver directional preferences.
|
|
bool send_audio = !GetAudioTransceiver()->internal()->senders().empty();
|
|
bool send_video = !GetVideoTransceiver()->internal()->senders().empty();
|
|
|
|
// By default, generate sendrecv/recvonly m= sections.
|
|
bool recv_audio = true;
|
|
bool recv_video = true;
|
|
|
|
// By default, only offer a new m= section if we have media to send with it.
|
|
bool offer_new_audio_description = send_audio;
|
|
bool offer_new_video_description = send_video;
|
|
bool offer_new_data_description = data_channel_controller_.HasDataChannels();
|
|
|
|
// The "offer_to_receive_X" options allow those defaults to be overridden.
|
|
if (offer_answer_options.offer_to_receive_audio !=
|
|
RTCOfferAnswerOptions::kUndefined) {
|
|
recv_audio = (offer_answer_options.offer_to_receive_audio > 0);
|
|
offer_new_audio_description =
|
|
offer_new_audio_description ||
|
|
(offer_answer_options.offer_to_receive_audio > 0);
|
|
}
|
|
if (offer_answer_options.offer_to_receive_video !=
|
|
RTCOfferAnswerOptions::kUndefined) {
|
|
recv_video = (offer_answer_options.offer_to_receive_video > 0);
|
|
offer_new_video_description =
|
|
offer_new_video_description ||
|
|
(offer_answer_options.offer_to_receive_video > 0);
|
|
}
|
|
|
|
absl::optional<size_t> audio_index;
|
|
absl::optional<size_t> video_index;
|
|
absl::optional<size_t> data_index;
|
|
// If a current description exists, generate m= sections in the same order,
|
|
// using the first audio/video/data section that appears and rejecting
|
|
// extraneous ones.
|
|
if (local_description()) {
|
|
GenerateMediaDescriptionOptions(
|
|
local_description(),
|
|
RtpTransceiverDirectionFromSendRecv(send_audio, recv_audio),
|
|
RtpTransceiverDirectionFromSendRecv(send_video, recv_video),
|
|
&audio_index, &video_index, &data_index, session_options);
|
|
}
|
|
|
|
// Add audio/video/data m= sections to the end if needed.
|
|
if (!audio_index && offer_new_audio_description) {
|
|
cricket::MediaDescriptionOptions options(
|
|
cricket::MEDIA_TYPE_AUDIO, cricket::CN_AUDIO,
|
|
RtpTransceiverDirectionFromSendRecv(send_audio, recv_audio), false);
|
|
options.header_extensions =
|
|
channel_manager()->GetSupportedAudioRtpHeaderExtensions();
|
|
session_options->media_description_options.push_back(options);
|
|
audio_index = session_options->media_description_options.size() - 1;
|
|
}
|
|
if (!video_index && offer_new_video_description) {
|
|
cricket::MediaDescriptionOptions options(
|
|
cricket::MEDIA_TYPE_VIDEO, cricket::CN_VIDEO,
|
|
RtpTransceiverDirectionFromSendRecv(send_video, recv_video), false);
|
|
options.header_extensions =
|
|
channel_manager()->GetSupportedVideoRtpHeaderExtensions();
|
|
session_options->media_description_options.push_back(options);
|
|
video_index = session_options->media_description_options.size() - 1;
|
|
}
|
|
if (!data_index && offer_new_data_description) {
|
|
session_options->media_description_options.push_back(
|
|
GetMediaDescriptionOptionsForActiveData(cricket::CN_DATA));
|
|
data_index = session_options->media_description_options.size() - 1;
|
|
}
|
|
|
|
cricket::MediaDescriptionOptions* audio_media_description_options =
|
|
!audio_index ? nullptr
|
|
: &session_options->media_description_options[*audio_index];
|
|
cricket::MediaDescriptionOptions* video_media_description_options =
|
|
!video_index ? nullptr
|
|
: &session_options->media_description_options[*video_index];
|
|
|
|
AddPlanBRtpSenderOptions(GetSendersInternal(),
|
|
audio_media_description_options,
|
|
video_media_description_options,
|
|
offer_answer_options.num_simulcast_layers);
|
|
}
|
|
|
|
static cricket::MediaDescriptionOptions
|
|
GetMediaDescriptionOptionsForTransceiver(
|
|
rtc::scoped_refptr<RtpTransceiverProxyWithInternal<RtpTransceiver>>
|
|
transceiver,
|
|
const std::string& mid,
|
|
bool is_create_offer) {
|
|
// NOTE: a stopping transceiver should be treated as a stopped one in
|
|
// createOffer as specified in
|
|
// https://w3c.github.io/webrtc-pc/#dom-rtcpeerconnection-createoffer.
|
|
bool stopped =
|
|
is_create_offer ? transceiver->stopping() : transceiver->stopped();
|
|
cricket::MediaDescriptionOptions media_description_options(
|
|
transceiver->media_type(), mid, transceiver->direction(), stopped);
|
|
media_description_options.codec_preferences =
|
|
transceiver->codec_preferences();
|
|
media_description_options.header_extensions =
|
|
transceiver->HeaderExtensionsToOffer();
|
|
// This behavior is specified in JSEP. The gist is that:
|
|
// 1. The MSID is included if the RtpTransceiver's direction is sendonly or
|
|
// sendrecv.
|
|
// 2. If the MSID is included, then it must be included in any subsequent
|
|
// offer/answer exactly the same until the RtpTransceiver is stopped.
|
|
if (stopped || (!RtpTransceiverDirectionHasSend(transceiver->direction()) &&
|
|
!transceiver->internal()->has_ever_been_used_to_send())) {
|
|
return media_description_options;
|
|
}
|
|
|
|
cricket::SenderOptions sender_options;
|
|
sender_options.track_id = transceiver->sender()->id();
|
|
sender_options.stream_ids = transceiver->sender()->stream_ids();
|
|
|
|
// The following sets up RIDs and Simulcast.
|
|
// RIDs are included if Simulcast is requested or if any RID was specified.
|
|
RtpParameters send_parameters =
|
|
transceiver->internal()->sender_internal()->GetParametersInternal();
|
|
bool has_rids = std::any_of(send_parameters.encodings.begin(),
|
|
send_parameters.encodings.end(),
|
|
[](const RtpEncodingParameters& encoding) {
|
|
return !encoding.rid.empty();
|
|
});
|
|
|
|
std::vector<RidDescription> send_rids;
|
|
SimulcastLayerList send_layers;
|
|
for (const RtpEncodingParameters& encoding : send_parameters.encodings) {
|
|
if (encoding.rid.empty()) {
|
|
continue;
|
|
}
|
|
send_rids.push_back(RidDescription(encoding.rid, RidDirection::kSend));
|
|
send_layers.AddLayer(SimulcastLayer(encoding.rid, !encoding.active));
|
|
}
|
|
|
|
if (has_rids) {
|
|
sender_options.rids = send_rids;
|
|
}
|
|
|
|
sender_options.simulcast_layers = send_layers;
|
|
// When RIDs are configured, we must set num_sim_layers to 0 to.
|
|
// Otherwise, num_sim_layers must be 1 because either there is no
|
|
// simulcast, or simulcast is acheived by munging the SDP.
|
|
sender_options.num_sim_layers = has_rids ? 0 : 1;
|
|
media_description_options.sender_options.push_back(sender_options);
|
|
|
|
return media_description_options;
|
|
}
|
|
|
|
// Returns the ContentInfo at mline index |i|, or null if none exists.
|
|
static const ContentInfo* GetContentByIndex(
|
|
const SessionDescriptionInterface* sdesc,
|
|
size_t i) {
|
|
if (!sdesc) {
|
|
return nullptr;
|
|
}
|
|
const ContentInfos& contents = sdesc->description()->contents();
|
|
return (i < contents.size() ? &contents[i] : nullptr);
|
|
}
|
|
|
|
void PeerConnection::GetOptionsForUnifiedPlanOffer(
|
|
const RTCOfferAnswerOptions& offer_answer_options,
|
|
cricket::MediaSessionOptions* session_options) {
|
|
// Rules for generating an offer are dictated by JSEP sections 5.2.1 (Initial
|
|
// Offers) and 5.2.2 (Subsequent Offers).
|
|
RTC_DCHECK_EQ(session_options->media_description_options.size(), 0);
|
|
const ContentInfos no_infos;
|
|
const ContentInfos& local_contents =
|
|
(local_description() ? local_description()->description()->contents()
|
|
: no_infos);
|
|
const ContentInfos& remote_contents =
|
|
(remote_description() ? remote_description()->description()->contents()
|
|
: no_infos);
|
|
// The mline indices that can be recycled. New transceivers should reuse these
|
|
// slots first.
|
|
std::queue<size_t> recycleable_mline_indices;
|
|
// First, go through each media section that exists in either the local or
|
|
// remote description and generate a media section in this offer for the
|
|
// associated transceiver. If a media section can be recycled, generate a
|
|
// default, rejected media section here that can be later overwritten.
|
|
for (size_t i = 0;
|
|
i < std::max(local_contents.size(), remote_contents.size()); ++i) {
|
|
// Either |local_content| or |remote_content| is non-null.
|
|
const ContentInfo* local_content =
|
|
(i < local_contents.size() ? &local_contents[i] : nullptr);
|
|
const ContentInfo* current_local_content =
|
|
GetContentByIndex(current_local_description(), i);
|
|
const ContentInfo* remote_content =
|
|
(i < remote_contents.size() ? &remote_contents[i] : nullptr);
|
|
const ContentInfo* current_remote_content =
|
|
GetContentByIndex(current_remote_description(), i);
|
|
bool had_been_rejected =
|
|
(current_local_content && current_local_content->rejected) ||
|
|
(current_remote_content && current_remote_content->rejected);
|
|
const std::string& mid =
|
|
(local_content ? local_content->name : remote_content->name);
|
|
cricket::MediaType media_type =
|
|
(local_content ? local_content->media_description()->type()
|
|
: remote_content->media_description()->type());
|
|
if (media_type == cricket::MEDIA_TYPE_AUDIO ||
|
|
media_type == cricket::MEDIA_TYPE_VIDEO) {
|
|
// A media section is considered eligible for recycling if it is marked as
|
|
// rejected in either the current local or current remote description.
|
|
auto transceiver = GetAssociatedTransceiver(mid);
|
|
if (!transceiver) {
|
|
// No associated transceiver. The media section has been stopped.
|
|
recycleable_mline_indices.push(i);
|
|
session_options->media_description_options.push_back(
|
|
cricket::MediaDescriptionOptions(media_type, mid,
|
|
RtpTransceiverDirection::kInactive,
|
|
/*stopped=*/true));
|
|
} else {
|
|
// NOTE: a stopping transceiver should be treated as a stopped one in
|
|
// createOffer as specified in
|
|
// https://w3c.github.io/webrtc-pc/#dom-rtcpeerconnection-createoffer.
|
|
if (had_been_rejected && transceiver->stopping()) {
|
|
session_options->media_description_options.push_back(
|
|
cricket::MediaDescriptionOptions(
|
|
transceiver->media_type(), mid,
|
|
RtpTransceiverDirection::kInactive,
|
|
/*stopped=*/true));
|
|
recycleable_mline_indices.push(i);
|
|
} else {
|
|
session_options->media_description_options.push_back(
|
|
GetMediaDescriptionOptionsForTransceiver(
|
|
transceiver, mid,
|
|
/*is_create_offer=*/true));
|
|
// CreateOffer shouldn't really cause any state changes in
|
|
// PeerConnection, but we need a way to match new transceivers to new
|
|
// media sections in SetLocalDescription and JSEP specifies this is
|
|
// done by recording the index of the media section generated for the
|
|
// transceiver in the offer.
|
|
transceiver->internal()->set_mline_index(i);
|
|
}
|
|
}
|
|
} else {
|
|
RTC_CHECK_EQ(cricket::MEDIA_TYPE_DATA, media_type);
|
|
if (had_been_rejected) {
|
|
session_options->media_description_options.push_back(
|
|
GetMediaDescriptionOptionsForRejectedData(mid));
|
|
} else {
|
|
RTC_CHECK(GetDataMid());
|
|
if (mid == *GetDataMid()) {
|
|
session_options->media_description_options.push_back(
|
|
GetMediaDescriptionOptionsForActiveData(mid));
|
|
} else {
|
|
session_options->media_description_options.push_back(
|
|
GetMediaDescriptionOptionsForRejectedData(mid));
|
|
}
|
|
}
|
|
}
|
|
}
|
|
|
|
// Next, look for transceivers that are newly added (that is, are not stopped
|
|
// and not associated). Reuse media sections marked as recyclable first,
|
|
// otherwise append to the end of the offer. New media sections should be
|
|
// added in the order they were added to the PeerConnection.
|
|
for (const auto& transceiver : transceivers_.List()) {
|
|
if (transceiver->mid() || transceiver->stopping()) {
|
|
continue;
|
|
}
|
|
size_t mline_index;
|
|
if (!recycleable_mline_indices.empty()) {
|
|
mline_index = recycleable_mline_indices.front();
|
|
recycleable_mline_indices.pop();
|
|
session_options->media_description_options[mline_index] =
|
|
GetMediaDescriptionOptionsForTransceiver(
|
|
transceiver, mid_generator_(), /*is_create_offer=*/true);
|
|
} else {
|
|
mline_index = session_options->media_description_options.size();
|
|
session_options->media_description_options.push_back(
|
|
GetMediaDescriptionOptionsForTransceiver(
|
|
transceiver, mid_generator_(), /*is_create_offer=*/true));
|
|
}
|
|
// See comment above for why CreateOffer changes the transceiver's state.
|
|
transceiver->internal()->set_mline_index(mline_index);
|
|
}
|
|
// Lastly, add a m-section if we have local data channels and an m section
|
|
// does not already exist.
|
|
if (!GetDataMid() && data_channel_controller_.HasDataChannels()) {
|
|
session_options->media_description_options.push_back(
|
|
GetMediaDescriptionOptionsForActiveData(mid_generator_()));
|
|
}
|
|
}
|
|
|
|
void PeerConnection::GetOptionsForAnswer(
|
|
const RTCOfferAnswerOptions& offer_answer_options,
|
|
cricket::MediaSessionOptions* session_options) {
|
|
RTC_DCHECK_RUN_ON(signaling_thread());
|
|
ExtractSharedMediaSessionOptions(offer_answer_options, session_options);
|
|
|
|
if (IsUnifiedPlan()) {
|
|
GetOptionsForUnifiedPlanAnswer(offer_answer_options, session_options);
|
|
} else {
|
|
GetOptionsForPlanBAnswer(offer_answer_options, session_options);
|
|
}
|
|
|
|
// Intentionally unset the data channel type for RTP data channel. Otherwise
|
|
// the RTP data channels would be successfully negotiated by default and the
|
|
// unit tests in WebRtcDataBrowserTest will fail when building with chromium.
|
|
// We want to leave RTP data channels broken, so people won't try to use them.
|
|
if (data_channel_controller_.HasRtpDataChannels() ||
|
|
data_channel_type() != cricket::DCT_RTP) {
|
|
session_options->data_channel_type = data_channel_type();
|
|
}
|
|
|
|
// Apply ICE renomination flag.
|
|
for (auto& options : session_options->media_description_options) {
|
|
options.transport_options.enable_ice_renomination =
|
|
configuration_.enable_ice_renomination;
|
|
}
|
|
|
|
session_options->rtcp_cname = rtcp_cname_;
|
|
session_options->crypto_options = GetCryptoOptions();
|
|
session_options->pooled_ice_credentials =
|
|
network_thread()->Invoke<std::vector<cricket::IceParameters>>(
|
|
RTC_FROM_HERE,
|
|
rtc::Bind(&cricket::PortAllocator::GetPooledIceCredentials,
|
|
port_allocator_.get()));
|
|
}
|
|
|
|
void PeerConnection::GetOptionsForPlanBAnswer(
|
|
const PeerConnectionInterface::RTCOfferAnswerOptions& offer_answer_options,
|
|
cricket::MediaSessionOptions* session_options) {
|
|
// Figure out transceiver directional preferences.
|
|
bool send_audio = !GetAudioTransceiver()->internal()->senders().empty();
|
|
bool send_video = !GetVideoTransceiver()->internal()->senders().empty();
|
|
|
|
// By default, generate sendrecv/recvonly m= sections. The direction is also
|
|
// restricted by the direction in the offer.
|
|
bool recv_audio = true;
|
|
bool recv_video = true;
|
|
|
|
// The "offer_to_receive_X" options allow those defaults to be overridden.
|
|
if (offer_answer_options.offer_to_receive_audio !=
|
|
RTCOfferAnswerOptions::kUndefined) {
|
|
recv_audio = (offer_answer_options.offer_to_receive_audio > 0);
|
|
}
|
|
if (offer_answer_options.offer_to_receive_video !=
|
|
RTCOfferAnswerOptions::kUndefined) {
|
|
recv_video = (offer_answer_options.offer_to_receive_video > 0);
|
|
}
|
|
|
|
absl::optional<size_t> audio_index;
|
|
absl::optional<size_t> video_index;
|
|
absl::optional<size_t> data_index;
|
|
|
|
// Generate m= sections that match those in the offer.
|
|
// Note that mediasession.cc will handle intersection our preferred
|
|
// direction with the offered direction.
|
|
GenerateMediaDescriptionOptions(
|
|
remote_description(),
|
|
RtpTransceiverDirectionFromSendRecv(send_audio, recv_audio),
|
|
RtpTransceiverDirectionFromSendRecv(send_video, recv_video), &audio_index,
|
|
&video_index, &data_index, session_options);
|
|
|
|
cricket::MediaDescriptionOptions* audio_media_description_options =
|
|
!audio_index ? nullptr
|
|
: &session_options->media_description_options[*audio_index];
|
|
cricket::MediaDescriptionOptions* video_media_description_options =
|
|
!video_index ? nullptr
|
|
: &session_options->media_description_options[*video_index];
|
|
|
|
AddPlanBRtpSenderOptions(GetSendersInternal(),
|
|
audio_media_description_options,
|
|
video_media_description_options,
|
|
offer_answer_options.num_simulcast_layers);
|
|
}
|
|
|
|
void PeerConnection::GetOptionsForUnifiedPlanAnswer(
|
|
const PeerConnectionInterface::RTCOfferAnswerOptions& offer_answer_options,
|
|
cricket::MediaSessionOptions* session_options) {
|
|
// Rules for generating an answer are dictated by JSEP sections 5.3.1 (Initial
|
|
// Answers) and 5.3.2 (Subsequent Answers).
|
|
RTC_DCHECK(remote_description());
|
|
RTC_DCHECK(remote_description()->GetType() == SdpType::kOffer);
|
|
for (const ContentInfo& content :
|
|
remote_description()->description()->contents()) {
|
|
cricket::MediaType media_type = content.media_description()->type();
|
|
if (media_type == cricket::MEDIA_TYPE_AUDIO ||
|
|
media_type == cricket::MEDIA_TYPE_VIDEO) {
|
|
auto transceiver = GetAssociatedTransceiver(content.name);
|
|
if (transceiver) {
|
|
session_options->media_description_options.push_back(
|
|
GetMediaDescriptionOptionsForTransceiver(
|
|
transceiver, content.name,
|
|
/*is_create_offer=*/false));
|
|
} else {
|
|
// This should only happen with rejected transceivers.
|
|
RTC_DCHECK(content.rejected);
|
|
session_options->media_description_options.push_back(
|
|
cricket::MediaDescriptionOptions(media_type, content.name,
|
|
RtpTransceiverDirection::kInactive,
|
|
/*stopped=*/true));
|
|
}
|
|
} else {
|
|
RTC_CHECK_EQ(cricket::MEDIA_TYPE_DATA, media_type);
|
|
// Reject all data sections if data channels are disabled.
|
|
// Reject a data section if it has already been rejected.
|
|
// Reject all data sections except for the first one.
|
|
if (data_channel_type() == cricket::DCT_NONE || content.rejected ||
|
|
content.name != *GetDataMid()) {
|
|
session_options->media_description_options.push_back(
|
|
GetMediaDescriptionOptionsForRejectedData(content.name));
|
|
} else {
|
|
session_options->media_description_options.push_back(
|
|
GetMediaDescriptionOptionsForActiveData(content.name));
|
|
}
|
|
}
|
|
}
|
|
}
|
|
|
|
void PeerConnection::GenerateMediaDescriptionOptions(
|
|
const SessionDescriptionInterface* session_desc,
|
|
RtpTransceiverDirection audio_direction,
|
|
RtpTransceiverDirection video_direction,
|
|
absl::optional<size_t>* audio_index,
|
|
absl::optional<size_t>* video_index,
|
|
absl::optional<size_t>* data_index,
|
|
cricket::MediaSessionOptions* session_options) {
|
|
for (const cricket::ContentInfo& content :
|
|
session_desc->description()->contents()) {
|
|
if (IsAudioContent(&content)) {
|
|
// If we already have an audio m= section, reject this extra one.
|
|
if (*audio_index) {
|
|
session_options->media_description_options.push_back(
|
|
cricket::MediaDescriptionOptions(
|
|
cricket::MEDIA_TYPE_AUDIO, content.name,
|
|
RtpTransceiverDirection::kInactive, /*stopped=*/true));
|
|
} else {
|
|
bool stopped = (audio_direction == RtpTransceiverDirection::kInactive);
|
|
session_options->media_description_options.push_back(
|
|
cricket::MediaDescriptionOptions(cricket::MEDIA_TYPE_AUDIO,
|
|
content.name, audio_direction,
|
|
stopped));
|
|
*audio_index = session_options->media_description_options.size() - 1;
|
|
}
|
|
session_options->media_description_options.back().header_extensions =
|
|
channel_manager()->GetSupportedAudioRtpHeaderExtensions();
|
|
} else if (IsVideoContent(&content)) {
|
|
// If we already have an video m= section, reject this extra one.
|
|
if (*video_index) {
|
|
session_options->media_description_options.push_back(
|
|
cricket::MediaDescriptionOptions(
|
|
cricket::MEDIA_TYPE_VIDEO, content.name,
|
|
RtpTransceiverDirection::kInactive, /*stopped=*/true));
|
|
} else {
|
|
bool stopped = (video_direction == RtpTransceiverDirection::kInactive);
|
|
session_options->media_description_options.push_back(
|
|
cricket::MediaDescriptionOptions(cricket::MEDIA_TYPE_VIDEO,
|
|
content.name, video_direction,
|
|
stopped));
|
|
*video_index = session_options->media_description_options.size() - 1;
|
|
}
|
|
session_options->media_description_options.back().header_extensions =
|
|
channel_manager()->GetSupportedVideoRtpHeaderExtensions();
|
|
} else {
|
|
RTC_DCHECK(IsDataContent(&content));
|
|
// If we already have an data m= section, reject this extra one.
|
|
if (*data_index) {
|
|
session_options->media_description_options.push_back(
|
|
GetMediaDescriptionOptionsForRejectedData(content.name));
|
|
} else {
|
|
session_options->media_description_options.push_back(
|
|
GetMediaDescriptionOptionsForActiveData(content.name));
|
|
*data_index = session_options->media_description_options.size() - 1;
|
|
}
|
|
}
|
|
}
|
|
}
|
|
|
|
cricket::MediaDescriptionOptions
|
|
PeerConnection::GetMediaDescriptionOptionsForActiveData(
|
|
const std::string& mid) const {
|
|
// Direction for data sections is meaningless, but legacy endpoints might
|
|
// expect sendrecv.
|
|
cricket::MediaDescriptionOptions options(cricket::MEDIA_TYPE_DATA, mid,
|
|
RtpTransceiverDirection::kSendRecv,
|
|
/*stopped=*/false);
|
|
AddRtpDataChannelOptions(*data_channel_controller_.rtp_data_channels(),
|
|
&options);
|
|
return options;
|
|
}
|
|
|
|
cricket::MediaDescriptionOptions
|
|
PeerConnection::GetMediaDescriptionOptionsForRejectedData(
|
|
const std::string& mid) const {
|
|
cricket::MediaDescriptionOptions options(cricket::MEDIA_TYPE_DATA, mid,
|
|
RtpTransceiverDirection::kInactive,
|
|
/*stopped=*/true);
|
|
AddRtpDataChannelOptions(*data_channel_controller_.rtp_data_channels(),
|
|
&options);
|
|
return options;
|
|
}
|
|
|
|
absl::optional<std::string> PeerConnection::GetDataMid() const {
|
|
RTC_DCHECK_RUN_ON(signaling_thread());
|
|
switch (data_channel_type()) {
|
|
case cricket::DCT_RTP:
|
|
if (!data_channel_controller_.rtp_data_channel()) {
|
|
return absl::nullopt;
|
|
}
|
|
return data_channel_controller_.rtp_data_channel()->content_name();
|
|
case cricket::DCT_SCTP:
|
|
return sctp_mid_s_;
|
|
default:
|
|
return absl::nullopt;
|
|
}
|
|
}
|
|
|
|
void PeerConnection::RemoveSenders(cricket::MediaType media_type) {
|
|
RTC_DCHECK_RUN_ON(signaling_thread());
|
|
UpdateLocalSenders(std::vector<cricket::StreamParams>(), media_type);
|
|
UpdateRemoteSendersList(std::vector<cricket::StreamParams>(), false,
|
|
media_type, nullptr);
|
|
}
|
|
|
|
void PeerConnection::UpdateRemoteSendersList(
|
|
const cricket::StreamParamsVec& streams,
|
|
bool default_sender_needed,
|
|
cricket::MediaType media_type,
|
|
StreamCollection* new_streams) {
|
|
RTC_DCHECK_RUN_ON(signaling_thread());
|
|
RTC_DCHECK(!IsUnifiedPlan());
|
|
|
|
std::vector<RtpSenderInfo>* current_senders =
|
|
GetRemoteSenderInfos(media_type);
|
|
|
|
// Find removed senders. I.e., senders where the sender id or ssrc don't match
|
|
// the new StreamParam.
|
|
for (auto sender_it = current_senders->begin();
|
|
sender_it != current_senders->end();
|
|
/* incremented manually */) {
|
|
const RtpSenderInfo& info = *sender_it;
|
|
const cricket::StreamParams* params =
|
|
cricket::GetStreamBySsrc(streams, info.first_ssrc);
|
|
std::string params_stream_id;
|
|
if (params) {
|
|
params_stream_id =
|
|
(!params->first_stream_id().empty() ? params->first_stream_id()
|
|
: kDefaultStreamId);
|
|
}
|
|
bool sender_exists = params && params->id == info.sender_id &&
|
|
params_stream_id == info.stream_id;
|
|
// If this is a default track, and we still need it, don't remove it.
|
|
if ((info.stream_id == kDefaultStreamId && default_sender_needed) ||
|
|
sender_exists) {
|
|
++sender_it;
|
|
} else {
|
|
OnRemoteSenderRemoved(info, media_type);
|
|
sender_it = current_senders->erase(sender_it);
|
|
}
|
|
}
|
|
|
|
// Find new and active senders.
|
|
for (const cricket::StreamParams& params : streams) {
|
|
if (!params.has_ssrcs()) {
|
|
// The remote endpoint has streams, but didn't signal ssrcs. For an active
|
|
// sender, this means it is coming from a Unified Plan endpoint,so we just
|
|
// create a default.
|
|
default_sender_needed = true;
|
|
break;
|
|
}
|
|
|
|
// |params.id| is the sender id and the stream id uses the first of
|
|
// |params.stream_ids|. The remote description could come from a Unified
|
|
// Plan endpoint, with multiple or no stream_ids() signaled. Since this is
|
|
// not supported in Plan B, we just take the first here and create the
|
|
// default stream ID if none is specified.
|
|
const std::string& stream_id =
|
|
(!params.first_stream_id().empty() ? params.first_stream_id()
|
|
: kDefaultStreamId);
|
|
const std::string& sender_id = params.id;
|
|
uint32_t ssrc = params.first_ssrc();
|
|
|
|
rtc::scoped_refptr<MediaStreamInterface> stream =
|
|
remote_streams_->find(stream_id);
|
|
if (!stream) {
|
|
// This is a new MediaStream. Create a new remote MediaStream.
|
|
stream = MediaStreamProxy::Create(rtc::Thread::Current(),
|
|
MediaStream::Create(stream_id));
|
|
remote_streams_->AddStream(stream);
|
|
new_streams->AddStream(stream);
|
|
}
|
|
|
|
const RtpSenderInfo* sender_info =
|
|
FindSenderInfo(*current_senders, stream_id, sender_id);
|
|
if (!sender_info) {
|
|
current_senders->push_back(RtpSenderInfo(stream_id, sender_id, ssrc));
|
|
OnRemoteSenderAdded(current_senders->back(), media_type);
|
|
}
|
|
}
|
|
|
|
// Add default sender if necessary.
|
|
if (default_sender_needed) {
|
|
rtc::scoped_refptr<MediaStreamInterface> default_stream =
|
|
remote_streams_->find(kDefaultStreamId);
|
|
if (!default_stream) {
|
|
// Create the new default MediaStream.
|
|
default_stream = MediaStreamProxy::Create(
|
|
rtc::Thread::Current(), MediaStream::Create(kDefaultStreamId));
|
|
remote_streams_->AddStream(default_stream);
|
|
new_streams->AddStream(default_stream);
|
|
}
|
|
std::string default_sender_id = (media_type == cricket::MEDIA_TYPE_AUDIO)
|
|
? kDefaultAudioSenderId
|
|
: kDefaultVideoSenderId;
|
|
const RtpSenderInfo* default_sender_info =
|
|
FindSenderInfo(*current_senders, kDefaultStreamId, default_sender_id);
|
|
if (!default_sender_info) {
|
|
current_senders->push_back(
|
|
RtpSenderInfo(kDefaultStreamId, default_sender_id, /*ssrc=*/0));
|
|
OnRemoteSenderAdded(current_senders->back(), media_type);
|
|
}
|
|
}
|
|
}
|
|
|
|
void PeerConnection::OnRemoteSenderAdded(const RtpSenderInfo& sender_info,
|
|
cricket::MediaType media_type) {
|
|
RTC_LOG(LS_INFO) << "Creating " << cricket::MediaTypeToString(media_type)
|
|
<< " receiver for track_id=" << sender_info.sender_id
|
|
<< " and stream_id=" << sender_info.stream_id;
|
|
|
|
MediaStreamInterface* stream = remote_streams_->find(sender_info.stream_id);
|
|
if (media_type == cricket::MEDIA_TYPE_AUDIO) {
|
|
CreateAudioReceiver(stream, sender_info);
|
|
} else if (media_type == cricket::MEDIA_TYPE_VIDEO) {
|
|
CreateVideoReceiver(stream, sender_info);
|
|
} else {
|
|
RTC_NOTREACHED() << "Invalid media type";
|
|
}
|
|
}
|
|
|
|
void PeerConnection::OnRemoteSenderRemoved(const RtpSenderInfo& sender_info,
|
|
cricket::MediaType media_type) {
|
|
RTC_LOG(LS_INFO) << "Removing " << cricket::MediaTypeToString(media_type)
|
|
<< " receiver for track_id=" << sender_info.sender_id
|
|
<< " and stream_id=" << sender_info.stream_id;
|
|
|
|
MediaStreamInterface* stream = remote_streams_->find(sender_info.stream_id);
|
|
|
|
rtc::scoped_refptr<RtpReceiverInterface> receiver;
|
|
if (media_type == cricket::MEDIA_TYPE_AUDIO) {
|
|
// When the MediaEngine audio channel is destroyed, the RemoteAudioSource
|
|
// will be notified which will end the AudioRtpReceiver::track().
|
|
receiver = RemoveAndStopReceiver(sender_info);
|
|
rtc::scoped_refptr<AudioTrackInterface> audio_track =
|
|
stream->FindAudioTrack(sender_info.sender_id);
|
|
if (audio_track) {
|
|
stream->RemoveTrack(audio_track);
|
|
}
|
|
} else if (media_type == cricket::MEDIA_TYPE_VIDEO) {
|
|
// Stopping or destroying a VideoRtpReceiver will end the
|
|
// VideoRtpReceiver::track().
|
|
receiver = RemoveAndStopReceiver(sender_info);
|
|
rtc::scoped_refptr<VideoTrackInterface> video_track =
|
|
stream->FindVideoTrack(sender_info.sender_id);
|
|
if (video_track) {
|
|
// There's no guarantee the track is still available, e.g. the track may
|
|
// have been removed from the stream by an application.
|
|
stream->RemoveTrack(video_track);
|
|
}
|
|
} else {
|
|
RTC_NOTREACHED() << "Invalid media type";
|
|
}
|
|
if (receiver) {
|
|
Observer()->OnRemoveTrack(receiver);
|
|
}
|
|
}
|
|
|
|
void PeerConnection::UpdateEndedRemoteMediaStreams() {
|
|
RTC_DCHECK_RUN_ON(signaling_thread());
|
|
std::vector<rtc::scoped_refptr<MediaStreamInterface>> streams_to_remove;
|
|
for (size_t i = 0; i < remote_streams_->count(); ++i) {
|
|
MediaStreamInterface* stream = remote_streams_->at(i);
|
|
if (stream->GetAudioTracks().empty() && stream->GetVideoTracks().empty()) {
|
|
streams_to_remove.push_back(stream);
|
|
}
|
|
}
|
|
|
|
for (auto& stream : streams_to_remove) {
|
|
remote_streams_->RemoveStream(stream);
|
|
Observer()->OnRemoveStream(std::move(stream));
|
|
}
|
|
}
|
|
|
|
void PeerConnection::UpdateLocalSenders(
|
|
const std::vector<cricket::StreamParams>& streams,
|
|
cricket::MediaType media_type) {
|
|
RTC_DCHECK_RUN_ON(signaling_thread());
|
|
std::vector<RtpSenderInfo>* current_senders = GetLocalSenderInfos(media_type);
|
|
|
|
// Find removed tracks. I.e., tracks where the track id, stream id or ssrc
|
|
// don't match the new StreamParam.
|
|
for (auto sender_it = current_senders->begin();
|
|
sender_it != current_senders->end();
|
|
/* incremented manually */) {
|
|
const RtpSenderInfo& info = *sender_it;
|
|
const cricket::StreamParams* params =
|
|
cricket::GetStreamBySsrc(streams, info.first_ssrc);
|
|
if (!params || params->id != info.sender_id ||
|
|
params->first_stream_id() != info.stream_id) {
|
|
OnLocalSenderRemoved(info, media_type);
|
|
sender_it = current_senders->erase(sender_it);
|
|
} else {
|
|
++sender_it;
|
|
}
|
|
}
|
|
|
|
// Find new and active senders.
|
|
for (const cricket::StreamParams& params : streams) {
|
|
// The sync_label is the MediaStream label and the |stream.id| is the
|
|
// sender id.
|
|
const std::string& stream_id = params.first_stream_id();
|
|
const std::string& sender_id = params.id;
|
|
uint32_t ssrc = params.first_ssrc();
|
|
const RtpSenderInfo* sender_info =
|
|
FindSenderInfo(*current_senders, stream_id, sender_id);
|
|
if (!sender_info) {
|
|
current_senders->push_back(RtpSenderInfo(stream_id, sender_id, ssrc));
|
|
OnLocalSenderAdded(current_senders->back(), media_type);
|
|
}
|
|
}
|
|
}
|
|
|
|
void PeerConnection::OnLocalSenderAdded(const RtpSenderInfo& sender_info,
|
|
cricket::MediaType media_type) {
|
|
RTC_DCHECK(!IsUnifiedPlan());
|
|
auto sender = FindSenderById(sender_info.sender_id);
|
|
if (!sender) {
|
|
RTC_LOG(LS_WARNING) << "An unknown RtpSender with id "
|
|
<< sender_info.sender_id
|
|
<< " has been configured in the local description.";
|
|
return;
|
|
}
|
|
|
|
if (sender->media_type() != media_type) {
|
|
RTC_LOG(LS_WARNING) << "An RtpSender has been configured in the local"
|
|
" description with an unexpected media type.";
|
|
return;
|
|
}
|
|
|
|
sender->internal()->set_stream_ids({sender_info.stream_id});
|
|
sender->internal()->SetSsrc(sender_info.first_ssrc);
|
|
}
|
|
|
|
void PeerConnection::OnLocalSenderRemoved(const RtpSenderInfo& sender_info,
|
|
cricket::MediaType media_type) {
|
|
auto sender = FindSenderById(sender_info.sender_id);
|
|
if (!sender) {
|
|
// This is the normal case. I.e., RemoveStream has been called and the
|
|
// SessionDescriptions has been renegotiated.
|
|
return;
|
|
}
|
|
|
|
// A sender has been removed from the SessionDescription but it's still
|
|
// associated with the PeerConnection. This only occurs if the SDP doesn't
|
|
// match with the calls to CreateSender, AddStream and RemoveStream.
|
|
if (sender->media_type() != media_type) {
|
|
RTC_LOG(LS_WARNING) << "An RtpSender has been configured in the local"
|
|
" description with an unexpected media type.";
|
|
return;
|
|
}
|
|
|
|
sender->internal()->SetSsrc(0);
|
|
}
|
|
|
|
void PeerConnection::OnSctpDataChannelClosed(DataChannelInterface* channel) {
|
|
// Since data_channel_controller doesn't do signals, this
|
|
// signal is relayed here.
|
|
data_channel_controller_.OnSctpDataChannelClosed(
|
|
static_cast<SctpDataChannel*>(channel));
|
|
}
|
|
|
|
rtc::scoped_refptr<RtpTransceiverProxyWithInternal<RtpTransceiver>>
|
|
PeerConnection::GetAudioTransceiver() const {
|
|
// This method only works with Plan B SDP, where there is a single
|
|
// audio/video transceiver.
|
|
RTC_DCHECK(!IsUnifiedPlan());
|
|
for (auto transceiver : transceivers_.List()) {
|
|
if (transceiver->media_type() == cricket::MEDIA_TYPE_AUDIO) {
|
|
return transceiver;
|
|
}
|
|
}
|
|
RTC_NOTREACHED();
|
|
return nullptr;
|
|
}
|
|
|
|
rtc::scoped_refptr<RtpTransceiverProxyWithInternal<RtpTransceiver>>
|
|
PeerConnection::GetVideoTransceiver() const {
|
|
// This method only works with Plan B SDP, where there is a single
|
|
// audio/video transceiver.
|
|
RTC_DCHECK(!IsUnifiedPlan());
|
|
for (auto transceiver : transceivers_.List()) {
|
|
if (transceiver->media_type() == cricket::MEDIA_TYPE_VIDEO) {
|
|
return transceiver;
|
|
}
|
|
}
|
|
RTC_NOTREACHED();
|
|
return nullptr;
|
|
}
|
|
|
|
rtc::scoped_refptr<RtpSenderProxyWithInternal<RtpSenderInternal>>
|
|
PeerConnection::FindSenderForTrack(MediaStreamTrackInterface* track) const {
|
|
for (const auto& transceiver : transceivers_.List()) {
|
|
for (auto sender : transceiver->internal()->senders()) {
|
|
if (sender->track() == track) {
|
|
return sender;
|
|
}
|
|
}
|
|
}
|
|
return nullptr;
|
|
}
|
|
|
|
rtc::scoped_refptr<RtpSenderProxyWithInternal<RtpSenderInternal>>
|
|
PeerConnection::FindSenderById(const std::string& sender_id) const {
|
|
for (const auto& transceiver : transceivers_.List()) {
|
|
for (auto sender : transceiver->internal()->senders()) {
|
|
if (sender->id() == sender_id) {
|
|
return sender;
|
|
}
|
|
}
|
|
}
|
|
return nullptr;
|
|
}
|
|
|
|
rtc::scoped_refptr<RtpReceiverProxyWithInternal<RtpReceiverInternal>>
|
|
PeerConnection::FindReceiverById(const std::string& receiver_id) const {
|
|
for (const auto& transceiver : transceivers_.List()) {
|
|
for (auto receiver : transceiver->internal()->receivers()) {
|
|
if (receiver->id() == receiver_id) {
|
|
return receiver;
|
|
}
|
|
}
|
|
}
|
|
return nullptr;
|
|
}
|
|
|
|
std::vector<PeerConnection::RtpSenderInfo>*
|
|
PeerConnection::GetRemoteSenderInfos(cricket::MediaType media_type) {
|
|
RTC_DCHECK(media_type == cricket::MEDIA_TYPE_AUDIO ||
|
|
media_type == cricket::MEDIA_TYPE_VIDEO);
|
|
return (media_type == cricket::MEDIA_TYPE_AUDIO)
|
|
? &remote_audio_sender_infos_
|
|
: &remote_video_sender_infos_;
|
|
}
|
|
|
|
std::vector<PeerConnection::RtpSenderInfo>* PeerConnection::GetLocalSenderInfos(
|
|
cricket::MediaType media_type) {
|
|
RTC_DCHECK(media_type == cricket::MEDIA_TYPE_AUDIO ||
|
|
media_type == cricket::MEDIA_TYPE_VIDEO);
|
|
return (media_type == cricket::MEDIA_TYPE_AUDIO) ? &local_audio_sender_infos_
|
|
: &local_video_sender_infos_;
|
|
}
|
|
|
|
const PeerConnection::RtpSenderInfo* PeerConnection::FindSenderInfo(
|
|
const std::vector<PeerConnection::RtpSenderInfo>& infos,
|
|
const std::string& stream_id,
|
|
const std::string sender_id) const {
|
|
for (const RtpSenderInfo& sender_info : infos) {
|
|
if (sender_info.stream_id == stream_id &&
|
|
sender_info.sender_id == sender_id) {
|
|
return &sender_info;
|
|
}
|
|
}
|
|
return nullptr;
|
|
}
|
|
|
|
SctpDataChannel* PeerConnection::FindDataChannelBySid(int sid) const {
|
|
return data_channel_controller_.FindDataChannelBySid(sid);
|
|
}
|
|
|
|
PeerConnection::InitializePortAllocatorResult
|
|
PeerConnection::InitializePortAllocator_n(
|
|
const cricket::ServerAddresses& stun_servers,
|
|
const std::vector<cricket::RelayServerConfig>& turn_servers,
|
|
const RTCConfiguration& configuration) {
|
|
RTC_DCHECK_RUN_ON(network_thread());
|
|
|
|
port_allocator_->Initialize();
|
|
// To handle both internal and externally created port allocator, we will
|
|
// enable BUNDLE here.
|
|
int port_allocator_flags = port_allocator_->flags();
|
|
port_allocator_flags |= cricket::PORTALLOCATOR_ENABLE_SHARED_SOCKET |
|
|
cricket::PORTALLOCATOR_ENABLE_IPV6 |
|
|
cricket::PORTALLOCATOR_ENABLE_IPV6_ON_WIFI;
|
|
// If the disable-IPv6 flag was specified, we'll not override it
|
|
// by experiment.
|
|
if (configuration.disable_ipv6) {
|
|
port_allocator_flags &= ~(cricket::PORTALLOCATOR_ENABLE_IPV6);
|
|
} else if (absl::StartsWith(factory_->trials().Lookup("WebRTC-IPv6Default"),
|
|
"Disabled")) {
|
|
port_allocator_flags &= ~(cricket::PORTALLOCATOR_ENABLE_IPV6);
|
|
}
|
|
if (configuration.disable_ipv6_on_wifi) {
|
|
port_allocator_flags &= ~(cricket::PORTALLOCATOR_ENABLE_IPV6_ON_WIFI);
|
|
RTC_LOG(LS_INFO) << "IPv6 candidates on Wi-Fi are disabled.";
|
|
}
|
|
|
|
if (configuration.tcp_candidate_policy == kTcpCandidatePolicyDisabled) {
|
|
port_allocator_flags |= cricket::PORTALLOCATOR_DISABLE_TCP;
|
|
RTC_LOG(LS_INFO) << "TCP candidates are disabled.";
|
|
}
|
|
|
|
if (configuration.candidate_network_policy ==
|
|
kCandidateNetworkPolicyLowCost) {
|
|
port_allocator_flags |= cricket::PORTALLOCATOR_DISABLE_COSTLY_NETWORKS;
|
|
RTC_LOG(LS_INFO) << "Do not gather candidates on high-cost networks";
|
|
}
|
|
|
|
if (configuration.disable_link_local_networks) {
|
|
port_allocator_flags |= cricket::PORTALLOCATOR_DISABLE_LINK_LOCAL_NETWORKS;
|
|
RTC_LOG(LS_INFO) << "Disable candidates on link-local network interfaces.";
|
|
}
|
|
|
|
port_allocator_->set_flags(port_allocator_flags);
|
|
// No step delay is used while allocating ports.
|
|
port_allocator_->set_step_delay(cricket::kMinimumStepDelay);
|
|
port_allocator_->SetCandidateFilter(
|
|
ConvertIceTransportTypeToCandidateFilter(configuration.type));
|
|
port_allocator_->set_max_ipv6_networks(configuration.max_ipv6_networks);
|
|
|
|
auto turn_servers_copy = turn_servers;
|
|
for (auto& turn_server : turn_servers_copy) {
|
|
turn_server.tls_cert_verifier = tls_cert_verifier_.get();
|
|
}
|
|
// Call this last since it may create pooled allocator sessions using the
|
|
// properties set above.
|
|
port_allocator_->SetConfiguration(
|
|
stun_servers, std::move(turn_servers_copy),
|
|
configuration.ice_candidate_pool_size,
|
|
configuration.GetTurnPortPrunePolicy(), configuration.turn_customizer,
|
|
configuration.stun_candidate_keepalive_interval);
|
|
|
|
InitializePortAllocatorResult res;
|
|
res.enable_ipv6 = port_allocator_flags & cricket::PORTALLOCATOR_ENABLE_IPV6;
|
|
return res;
|
|
}
|
|
|
|
bool PeerConnection::ReconfigurePortAllocator_n(
|
|
const cricket::ServerAddresses& stun_servers,
|
|
const std::vector<cricket::RelayServerConfig>& turn_servers,
|
|
IceTransportsType type,
|
|
int candidate_pool_size,
|
|
PortPrunePolicy turn_port_prune_policy,
|
|
webrtc::TurnCustomizer* turn_customizer,
|
|
absl::optional<int> stun_candidate_keepalive_interval,
|
|
bool have_local_description) {
|
|
port_allocator_->SetCandidateFilter(
|
|
ConvertIceTransportTypeToCandidateFilter(type));
|
|
// According to JSEP, after setLocalDescription, changing the candidate pool
|
|
// size is not allowed, and changing the set of ICE servers will not result
|
|
// in new candidates being gathered.
|
|
if (have_local_description) {
|
|
port_allocator_->FreezeCandidatePool();
|
|
}
|
|
// Add the custom tls turn servers if they exist.
|
|
auto turn_servers_copy = turn_servers;
|
|
for (auto& turn_server : turn_servers_copy) {
|
|
turn_server.tls_cert_verifier = tls_cert_verifier_.get();
|
|
}
|
|
// Call this last since it may create pooled allocator sessions using the
|
|
// candidate filter set above.
|
|
return port_allocator_->SetConfiguration(
|
|
stun_servers, std::move(turn_servers_copy), candidate_pool_size,
|
|
turn_port_prune_policy, turn_customizer,
|
|
stun_candidate_keepalive_interval);
|
|
}
|
|
|
|
cricket::ChannelManager* PeerConnection::channel_manager() const {
|
|
return factory_->channel_manager();
|
|
}
|
|
|
|
bool PeerConnection::StartRtcEventLog_w(
|
|
std::unique_ptr<RtcEventLogOutput> output,
|
|
int64_t output_period_ms) {
|
|
RTC_DCHECK_RUN_ON(worker_thread());
|
|
if (!event_log_) {
|
|
return false;
|
|
}
|
|
return event_log_->StartLogging(std::move(output), output_period_ms);
|
|
}
|
|
|
|
void PeerConnection::StopRtcEventLog_w() {
|
|
RTC_DCHECK_RUN_ON(worker_thread());
|
|
if (event_log_) {
|
|
event_log_->StopLogging();
|
|
}
|
|
}
|
|
|
|
cricket::ChannelInterface* PeerConnection::GetChannel(
|
|
const std::string& content_name) {
|
|
for (const auto& transceiver : transceivers_.List()) {
|
|
cricket::ChannelInterface* channel = transceiver->internal()->channel();
|
|
if (channel && channel->content_name() == content_name) {
|
|
return channel;
|
|
}
|
|
}
|
|
if (rtp_data_channel() &&
|
|
rtp_data_channel()->content_name() == content_name) {
|
|
return rtp_data_channel();
|
|
}
|
|
return nullptr;
|
|
}
|
|
|
|
bool PeerConnection::GetSctpSslRole(rtc::SSLRole* role) {
|
|
RTC_DCHECK_RUN_ON(signaling_thread());
|
|
if (!local_description() || !remote_description()) {
|
|
RTC_LOG(LS_VERBOSE)
|
|
<< "Local and Remote descriptions must be applied to get the "
|
|
"SSL Role of the SCTP transport.";
|
|
return false;
|
|
}
|
|
if (!data_channel_controller_.data_channel_transport()) {
|
|
RTC_LOG(LS_INFO) << "Non-rejected SCTP m= section is needed to get the "
|
|
"SSL Role of the SCTP transport.";
|
|
return false;
|
|
}
|
|
|
|
absl::optional<rtc::SSLRole> dtls_role;
|
|
if (sctp_mid_s_) {
|
|
dtls_role = transport_controller_->GetDtlsRole(*sctp_mid_s_);
|
|
if (!dtls_role && sdp_handler_.is_caller().has_value()) {
|
|
dtls_role = *sdp_handler_.is_caller() ? rtc::SSL_SERVER : rtc::SSL_CLIENT;
|
|
}
|
|
*role = *dtls_role;
|
|
return true;
|
|
}
|
|
return false;
|
|
}
|
|
|
|
bool PeerConnection::GetSslRole(const std::string& content_name,
|
|
rtc::SSLRole* role) {
|
|
RTC_DCHECK_RUN_ON(signaling_thread());
|
|
if (!local_description() || !remote_description()) {
|
|
RTC_LOG(LS_INFO)
|
|
<< "Local and Remote descriptions must be applied to get the "
|
|
"SSL Role of the session.";
|
|
return false;
|
|
}
|
|
|
|
auto dtls_role = transport_controller_->GetDtlsRole(content_name);
|
|
if (dtls_role) {
|
|
*role = *dtls_role;
|
|
return true;
|
|
}
|
|
return false;
|
|
}
|
|
|
|
void PeerConnection::SetSessionError(SessionError error,
|
|
const std::string& error_desc) {
|
|
RTC_DCHECK_RUN_ON(signaling_thread());
|
|
if (error != session_error_) {
|
|
session_error_ = error;
|
|
session_error_desc_ = error_desc;
|
|
}
|
|
}
|
|
|
|
void PeerConnection::UpdatePayloadTypeDemuxingState(
|
|
cricket::ContentSource source) {
|
|
// We may need to delete any created default streams and disable creation of
|
|
// new ones on the basis of payload type. This is needed to avoid SSRC
|
|
// collisions in Call's RtpDemuxer, in the case that a transceiver has
|
|
// created a default stream, and then some other channel gets the SSRC
|
|
// signaled in the corresponding Unified Plan "m=" section. For more context
|
|
// see https://bugs.chromium.org/p/webrtc/issues/detail?id=11477
|
|
const SessionDescriptionInterface* sdesc =
|
|
(source == cricket::CS_LOCAL ? local_description()
|
|
: remote_description());
|
|
size_t num_receiving_video_transceivers = 0;
|
|
size_t num_receiving_audio_transceivers = 0;
|
|
for (auto& content_info : sdesc->description()->contents()) {
|
|
if (content_info.rejected ||
|
|
(source == cricket::ContentSource::CS_LOCAL &&
|
|
!RtpTransceiverDirectionHasRecv(
|
|
content_info.media_description()->direction())) ||
|
|
(source == cricket::ContentSource::CS_REMOTE &&
|
|
!RtpTransceiverDirectionHasSend(
|
|
content_info.media_description()->direction()))) {
|
|
// Ignore transceivers that are not receiving.
|
|
continue;
|
|
}
|
|
switch (content_info.media_description()->type()) {
|
|
case cricket::MediaType::MEDIA_TYPE_AUDIO:
|
|
++num_receiving_audio_transceivers;
|
|
break;
|
|
case cricket::MediaType::MEDIA_TYPE_VIDEO:
|
|
++num_receiving_video_transceivers;
|
|
break;
|
|
default:
|
|
// Ignore data channels.
|
|
continue;
|
|
}
|
|
}
|
|
bool pt_demuxing_enabled_video = num_receiving_video_transceivers <= 1;
|
|
bool pt_demuxing_enabled_audio = num_receiving_audio_transceivers <= 1;
|
|
|
|
// Gather all updates ahead of time so that all channels can be updated in a
|
|
// single Invoke; necessary due to thread guards.
|
|
std::vector<std::pair<RtpTransceiverDirection, cricket::ChannelInterface*>>
|
|
channels_to_update;
|
|
for (const auto& transceiver : transceivers_.List()) {
|
|
cricket::ChannelInterface* channel = transceiver->internal()->channel();
|
|
const ContentInfo* content =
|
|
FindMediaSectionForTransceiver(transceiver, sdesc);
|
|
if (!channel || !content) {
|
|
continue;
|
|
}
|
|
RtpTransceiverDirection local_direction =
|
|
content->media_description()->direction();
|
|
if (source == cricket::CS_REMOTE) {
|
|
local_direction = RtpTransceiverDirectionReversed(local_direction);
|
|
}
|
|
channels_to_update.emplace_back(local_direction,
|
|
transceiver->internal()->channel());
|
|
}
|
|
|
|
if (!channels_to_update.empty()) {
|
|
worker_thread()->Invoke<void>(
|
|
RTC_FROM_HERE, [&channels_to_update, pt_demuxing_enabled_audio,
|
|
pt_demuxing_enabled_video]() {
|
|
for (const auto& it : channels_to_update) {
|
|
RtpTransceiverDirection local_direction = it.first;
|
|
cricket::ChannelInterface* channel = it.second;
|
|
cricket::MediaType media_type = channel->media_type();
|
|
if (media_type == cricket::MediaType::MEDIA_TYPE_AUDIO) {
|
|
channel->SetPayloadTypeDemuxingEnabled(
|
|
pt_demuxing_enabled_audio &&
|
|
RtpTransceiverDirectionHasRecv(local_direction));
|
|
} else if (media_type == cricket::MediaType::MEDIA_TYPE_VIDEO) {
|
|
channel->SetPayloadTypeDemuxingEnabled(
|
|
pt_demuxing_enabled_video &&
|
|
RtpTransceiverDirectionHasRecv(local_direction));
|
|
}
|
|
}
|
|
});
|
|
}
|
|
}
|
|
|
|
RTCError PeerConnection::PushdownMediaDescription(
|
|
SdpType type,
|
|
cricket::ContentSource source) {
|
|
const SessionDescriptionInterface* sdesc =
|
|
(source == cricket::CS_LOCAL ? local_description()
|
|
: remote_description());
|
|
RTC_DCHECK_RUN_ON(signaling_thread());
|
|
RTC_DCHECK(sdesc);
|
|
|
|
UpdatePayloadTypeDemuxingState(source);
|
|
|
|
// Push down the new SDP media section for each audio/video transceiver.
|
|
for (const auto& transceiver : transceivers_.List()) {
|
|
const ContentInfo* content_info =
|
|
FindMediaSectionForTransceiver(transceiver, sdesc);
|
|
cricket::ChannelInterface* channel = transceiver->internal()->channel();
|
|
if (!channel || !content_info || content_info->rejected) {
|
|
continue;
|
|
}
|
|
const MediaContentDescription* content_desc =
|
|
content_info->media_description();
|
|
if (!content_desc) {
|
|
continue;
|
|
}
|
|
std::string error;
|
|
bool success = (source == cricket::CS_LOCAL)
|
|
? channel->SetLocalContent(content_desc, type, &error)
|
|
: channel->SetRemoteContent(content_desc, type, &error);
|
|
if (!success) {
|
|
LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_PARAMETER, error);
|
|
}
|
|
}
|
|
|
|
// If using the RtpDataChannel, push down the new SDP section for it too.
|
|
if (data_channel_controller_.rtp_data_channel()) {
|
|
const ContentInfo* data_content =
|
|
cricket::GetFirstDataContent(sdesc->description());
|
|
if (data_content && !data_content->rejected) {
|
|
const MediaContentDescription* data_desc =
|
|
data_content->media_description();
|
|
if (data_desc) {
|
|
std::string error;
|
|
bool success =
|
|
(source == cricket::CS_LOCAL)
|
|
? data_channel_controller_.rtp_data_channel()->SetLocalContent(
|
|
data_desc, type, &error)
|
|
: data_channel_controller_.rtp_data_channel()->SetRemoteContent(
|
|
data_desc, type, &error);
|
|
if (!success) {
|
|
LOG_AND_RETURN_ERROR(RTCErrorType::INVALID_PARAMETER, error);
|
|
}
|
|
}
|
|
}
|
|
}
|
|
|
|
// Need complete offer/answer with an SCTP m= section before starting SCTP,
|
|
// according to https://tools.ietf.org/html/draft-ietf-mmusic-sctp-sdp-19
|
|
if (sctp_mid_s_ && local_description() && remote_description()) {
|
|
rtc::scoped_refptr<SctpTransport> sctp_transport =
|
|
transport_controller_->GetSctpTransport(*sctp_mid_s_);
|
|
auto local_sctp_description = cricket::GetFirstSctpDataContentDescription(
|
|
local_description()->description());
|
|
auto remote_sctp_description = cricket::GetFirstSctpDataContentDescription(
|
|
remote_description()->description());
|
|
if (sctp_transport && local_sctp_description && remote_sctp_description) {
|
|
int max_message_size;
|
|
// A remote max message size of zero means "any size supported".
|
|
// We configure the connection with our own max message size.
|
|
if (remote_sctp_description->max_message_size() == 0) {
|
|
max_message_size = local_sctp_description->max_message_size();
|
|
} else {
|
|
max_message_size =
|
|
std::min(local_sctp_description->max_message_size(),
|
|
remote_sctp_description->max_message_size());
|
|
}
|
|
sctp_transport->Start(local_sctp_description->port(),
|
|
remote_sctp_description->port(), max_message_size);
|
|
}
|
|
}
|
|
|
|
return RTCError::OK();
|
|
}
|
|
|
|
RTCError PeerConnection::PushdownTransportDescription(
|
|
cricket::ContentSource source,
|
|
SdpType type) {
|
|
RTC_DCHECK_RUN_ON(signaling_thread());
|
|
|
|
if (source == cricket::CS_LOCAL) {
|
|
const SessionDescriptionInterface* sdesc = local_description();
|
|
RTC_DCHECK(sdesc);
|
|
return transport_controller_->SetLocalDescription(type,
|
|
sdesc->description());
|
|
} else {
|
|
const SessionDescriptionInterface* sdesc = remote_description();
|
|
RTC_DCHECK(sdesc);
|
|
return transport_controller_->SetRemoteDescription(type,
|
|
sdesc->description());
|
|
}
|
|
}
|
|
|
|
bool PeerConnection::GetTransportDescription(
|
|
const SessionDescription* description,
|
|
const std::string& content_name,
|
|
cricket::TransportDescription* tdesc) {
|
|
if (!description || !tdesc) {
|
|
return false;
|
|
}
|
|
const TransportInfo* transport_info =
|
|
description->GetTransportInfoByName(content_name);
|
|
if (!transport_info) {
|
|
return false;
|
|
}
|
|
*tdesc = transport_info->description;
|
|
return true;
|
|
}
|
|
|
|
cricket::IceConfig PeerConnection::ParseIceConfig(
|
|
const PeerConnectionInterface::RTCConfiguration& config) const {
|
|
cricket::ContinualGatheringPolicy gathering_policy;
|
|
switch (config.continual_gathering_policy) {
|
|
case PeerConnectionInterface::GATHER_ONCE:
|
|
gathering_policy = cricket::GATHER_ONCE;
|
|
break;
|
|
case PeerConnectionInterface::GATHER_CONTINUALLY:
|
|
gathering_policy = cricket::GATHER_CONTINUALLY;
|
|
break;
|
|
default:
|
|
RTC_NOTREACHED();
|
|
gathering_policy = cricket::GATHER_ONCE;
|
|
}
|
|
|
|
cricket::IceConfig ice_config;
|
|
ice_config.receiving_timeout = RTCConfigurationToIceConfigOptionalInt(
|
|
config.ice_connection_receiving_timeout);
|
|
ice_config.prioritize_most_likely_candidate_pairs =
|
|
config.prioritize_most_likely_ice_candidate_pairs;
|
|
ice_config.backup_connection_ping_interval =
|
|
RTCConfigurationToIceConfigOptionalInt(
|
|
config.ice_backup_candidate_pair_ping_interval);
|
|
ice_config.continual_gathering_policy = gathering_policy;
|
|
ice_config.presume_writable_when_fully_relayed =
|
|
config.presume_writable_when_fully_relayed;
|
|
ice_config.surface_ice_candidates_on_ice_transport_type_changed =
|
|
config.surface_ice_candidates_on_ice_transport_type_changed;
|
|
ice_config.ice_check_interval_strong_connectivity =
|
|
config.ice_check_interval_strong_connectivity;
|
|
ice_config.ice_check_interval_weak_connectivity =
|
|
config.ice_check_interval_weak_connectivity;
|
|
ice_config.ice_check_min_interval = config.ice_check_min_interval;
|
|
ice_config.ice_unwritable_timeout = config.ice_unwritable_timeout;
|
|
ice_config.ice_unwritable_min_checks = config.ice_unwritable_min_checks;
|
|
ice_config.ice_inactive_timeout = config.ice_inactive_timeout;
|
|
ice_config.stun_keepalive_interval = config.stun_candidate_keepalive_interval;
|
|
ice_config.network_preference = config.network_preference;
|
|
return ice_config;
|
|
}
|
|
|
|
std::vector<DataChannelStats> PeerConnection::GetDataChannelStats() const {
|
|
RTC_DCHECK_RUN_ON(signaling_thread());
|
|
return data_channel_controller_.GetDataChannelStats();
|
|
}
|
|
|
|
absl::optional<std::string> PeerConnection::sctp_transport_name() const {
|
|
RTC_DCHECK_RUN_ON(signaling_thread());
|
|
if (sctp_mid_s_ && transport_controller_) {
|
|
auto dtls_transport = transport_controller_->GetDtlsTransport(*sctp_mid_s_);
|
|
if (dtls_transport) {
|
|
return dtls_transport->transport_name();
|
|
}
|
|
return absl::optional<std::string>();
|
|
}
|
|
return absl::optional<std::string>();
|
|
}
|
|
|
|
cricket::CandidateStatsList PeerConnection::GetPooledCandidateStats() const {
|
|
cricket::CandidateStatsList candidate_states_list;
|
|
network_thread()->Invoke<void>(
|
|
RTC_FROM_HERE,
|
|
rtc::Bind(&cricket::PortAllocator::GetCandidateStatsFromPooledSessions,
|
|
port_allocator_.get(), &candidate_states_list));
|
|
return candidate_states_list;
|
|
}
|
|
|
|
std::map<std::string, std::string> PeerConnection::GetTransportNamesByMid()
|
|
const {
|
|
RTC_DCHECK_RUN_ON(signaling_thread());
|
|
std::map<std::string, std::string> transport_names_by_mid;
|
|
for (const auto& transceiver : transceivers_.List()) {
|
|
cricket::ChannelInterface* channel = transceiver->internal()->channel();
|
|
if (channel) {
|
|
transport_names_by_mid[channel->content_name()] =
|
|
channel->transport_name();
|
|
}
|
|
}
|
|
if (data_channel_controller_.rtp_data_channel()) {
|
|
transport_names_by_mid[data_channel_controller_.rtp_data_channel()
|
|
->content_name()] =
|
|
data_channel_controller_.rtp_data_channel()->transport_name();
|
|
}
|
|
if (data_channel_controller_.data_channel_transport()) {
|
|
absl::optional<std::string> transport_name = sctp_transport_name();
|
|
RTC_DCHECK(transport_name);
|
|
transport_names_by_mid[*sctp_mid_s_] = *transport_name;
|
|
}
|
|
return transport_names_by_mid;
|
|
}
|
|
|
|
std::map<std::string, cricket::TransportStats>
|
|
PeerConnection::GetTransportStatsByNames(
|
|
const std::set<std::string>& transport_names) {
|
|
if (!network_thread()->IsCurrent()) {
|
|
return network_thread()
|
|
->Invoke<std::map<std::string, cricket::TransportStats>>(
|
|
RTC_FROM_HERE,
|
|
[&] { return GetTransportStatsByNames(transport_names); });
|
|
}
|
|
RTC_DCHECK_RUN_ON(network_thread());
|
|
std::map<std::string, cricket::TransportStats> transport_stats_by_name;
|
|
for (const std::string& transport_name : transport_names) {
|
|
cricket::TransportStats transport_stats;
|
|
bool success =
|
|
transport_controller_->GetStats(transport_name, &transport_stats);
|
|
if (success) {
|
|
transport_stats_by_name[transport_name] = std::move(transport_stats);
|
|
} else {
|
|
RTC_LOG(LS_ERROR) << "Failed to get transport stats for transport_name="
|
|
<< transport_name;
|
|
}
|
|
}
|
|
return transport_stats_by_name;
|
|
}
|
|
|
|
bool PeerConnection::GetLocalCertificate(
|
|
const std::string& transport_name,
|
|
rtc::scoped_refptr<rtc::RTCCertificate>* certificate) {
|
|
if (!certificate) {
|
|
return false;
|
|
}
|
|
*certificate = transport_controller_->GetLocalCertificate(transport_name);
|
|
return *certificate != nullptr;
|
|
}
|
|
|
|
std::unique_ptr<rtc::SSLCertChain> PeerConnection::GetRemoteSSLCertChain(
|
|
const std::string& transport_name) {
|
|
return transport_controller_->GetRemoteSSLCertChain(transport_name);
|
|
}
|
|
|
|
cricket::DataChannelType PeerConnection::data_channel_type() const {
|
|
return data_channel_controller_.data_channel_type();
|
|
}
|
|
|
|
bool PeerConnection::IceRestartPending(const std::string& content_name) const {
|
|
RTC_DCHECK_RUN_ON(signaling_thread());
|
|
return sdp_handler_.IceRestartPending(content_name);
|
|
}
|
|
|
|
bool PeerConnection::NeedsIceRestart(const std::string& content_name) const {
|
|
return transport_controller_->NeedsIceRestart(content_name);
|
|
}
|
|
|
|
void PeerConnection::OnCertificateReady(
|
|
const rtc::scoped_refptr<rtc::RTCCertificate>& certificate) {
|
|
transport_controller_->SetLocalCertificate(certificate);
|
|
}
|
|
|
|
void PeerConnection::OnTransportControllerConnectionState(
|
|
cricket::IceConnectionState state) {
|
|
switch (state) {
|
|
case cricket::kIceConnectionConnecting:
|
|
// If the current state is Connected or Completed, then there were
|
|
// writable channels but now there are not, so the next state must
|
|
// be Disconnected.
|
|
// kIceConnectionConnecting is currently used as the default,
|
|
// un-connected state by the TransportController, so its only use is
|
|
// detecting disconnections.
|
|
if (ice_connection_state_ ==
|
|
PeerConnectionInterface::kIceConnectionConnected ||
|
|
ice_connection_state_ ==
|
|
PeerConnectionInterface::kIceConnectionCompleted) {
|
|
SetIceConnectionState(
|
|
PeerConnectionInterface::kIceConnectionDisconnected);
|
|
}
|
|
break;
|
|
case cricket::kIceConnectionFailed:
|
|
SetIceConnectionState(PeerConnectionInterface::kIceConnectionFailed);
|
|
break;
|
|
case cricket::kIceConnectionConnected:
|
|
RTC_LOG(LS_INFO) << "Changing to ICE connected state because "
|
|
"all transports are writable.";
|
|
SetIceConnectionState(PeerConnectionInterface::kIceConnectionConnected);
|
|
NoteUsageEvent(UsageEvent::ICE_STATE_CONNECTED);
|
|
break;
|
|
case cricket::kIceConnectionCompleted:
|
|
RTC_LOG(LS_INFO) << "Changing to ICE completed state because "
|
|
"all transports are complete.";
|
|
if (ice_connection_state_ !=
|
|
PeerConnectionInterface::kIceConnectionConnected) {
|
|
// If jumping directly from "checking" to "connected",
|
|
// signal "connected" first.
|
|
SetIceConnectionState(PeerConnectionInterface::kIceConnectionConnected);
|
|
}
|
|
SetIceConnectionState(PeerConnectionInterface::kIceConnectionCompleted);
|
|
NoteUsageEvent(UsageEvent::ICE_STATE_CONNECTED);
|
|
ReportTransportStats();
|
|
break;
|
|
default:
|
|
RTC_NOTREACHED();
|
|
}
|
|
}
|
|
|
|
void PeerConnection::OnTransportControllerCandidatesGathered(
|
|
const std::string& transport_name,
|
|
const cricket::Candidates& candidates) {
|
|
int sdp_mline_index;
|
|
if (!GetLocalCandidateMediaIndex(transport_name, &sdp_mline_index)) {
|
|
RTC_LOG(LS_ERROR)
|
|
<< "OnTransportControllerCandidatesGathered: content name "
|
|
<< transport_name << " not found";
|
|
return;
|
|
}
|
|
|
|
for (cricket::Candidates::const_iterator citer = candidates.begin();
|
|
citer != candidates.end(); ++citer) {
|
|
// Use transport_name as the candidate media id.
|
|
std::unique_ptr<JsepIceCandidate> candidate(
|
|
new JsepIceCandidate(transport_name, sdp_mline_index, *citer));
|
|
sdp_handler_.AddLocalIceCandidate(candidate.get());
|
|
OnIceCandidate(std::move(candidate));
|
|
}
|
|
}
|
|
|
|
void PeerConnection::OnTransportControllerCandidateError(
|
|
const cricket::IceCandidateErrorEvent& event) {
|
|
OnIceCandidateError(event.address, event.port, event.url, event.error_code,
|
|
event.error_text);
|
|
}
|
|
|
|
void PeerConnection::OnTransportControllerCandidatesRemoved(
|
|
const std::vector<cricket::Candidate>& candidates) {
|
|
// Sanity check.
|
|
for (const cricket::Candidate& candidate : candidates) {
|
|
if (candidate.transport_name().empty()) {
|
|
RTC_LOG(LS_ERROR) << "OnTransportControllerCandidatesRemoved: "
|
|
"empty content name in candidate "
|
|
<< candidate.ToString();
|
|
return;
|
|
}
|
|
}
|
|
sdp_handler_.RemoveLocalIceCandidates(candidates);
|
|
OnIceCandidatesRemoved(candidates);
|
|
}
|
|
|
|
void PeerConnection::OnTransportControllerCandidateChanged(
|
|
const cricket::CandidatePairChangeEvent& event) {
|
|
OnSelectedCandidatePairChanged(event);
|
|
}
|
|
|
|
void PeerConnection::OnTransportControllerDtlsHandshakeError(
|
|
rtc::SSLHandshakeError error) {
|
|
RTC_HISTOGRAM_ENUMERATION(
|
|
"WebRTC.PeerConnection.DtlsHandshakeError", static_cast<int>(error),
|
|
static_cast<int>(rtc::SSLHandshakeError::MAX_VALUE));
|
|
}
|
|
|
|
void PeerConnection::EnableSending() {
|
|
RTC_DCHECK_RUN_ON(signaling_thread());
|
|
for (const auto& transceiver : transceivers_.List()) {
|
|
cricket::ChannelInterface* channel = transceiver->internal()->channel();
|
|
if (channel && !channel->enabled()) {
|
|
channel->Enable(true);
|
|
}
|
|
}
|
|
|
|
if (data_channel_controller_.rtp_data_channel() &&
|
|
!data_channel_controller_.rtp_data_channel()->enabled()) {
|
|
data_channel_controller_.rtp_data_channel()->Enable(true);
|
|
}
|
|
}
|
|
|
|
// Returns the media index for a local ice candidate given the content name.
|
|
bool PeerConnection::GetLocalCandidateMediaIndex(
|
|
const std::string& content_name,
|
|
int* sdp_mline_index) {
|
|
if (!local_description() || !sdp_mline_index) {
|
|
return false;
|
|
}
|
|
|
|
bool content_found = false;
|
|
const ContentInfos& contents = local_description()->description()->contents();
|
|
for (size_t index = 0; index < contents.size(); ++index) {
|
|
if (contents[index].name == content_name) {
|
|
*sdp_mline_index = static_cast<int>(index);
|
|
content_found = true;
|
|
break;
|
|
}
|
|
}
|
|
return content_found;
|
|
}
|
|
|
|
bool PeerConnection::UseCandidatesInSessionDescription(
|
|
const SessionDescriptionInterface* remote_desc) {
|
|
RTC_DCHECK_RUN_ON(signaling_thread());
|
|
if (!remote_desc) {
|
|
return true;
|
|
}
|
|
bool ret = true;
|
|
|
|
for (size_t m = 0; m < remote_desc->number_of_mediasections(); ++m) {
|
|
const IceCandidateCollection* candidates = remote_desc->candidates(m);
|
|
for (size_t n = 0; n < candidates->count(); ++n) {
|
|
const IceCandidateInterface* candidate = candidates->at(n);
|
|
bool valid = false;
|
|
if (!ReadyToUseRemoteCandidate(candidate, remote_desc, &valid)) {
|
|
if (valid) {
|
|
RTC_LOG(LS_INFO)
|
|
<< "UseCandidatesInSessionDescription: Not ready to use "
|
|
"candidate.";
|
|
}
|
|
continue;
|
|
}
|
|
ret = UseCandidate(candidate);
|
|
if (!ret) {
|
|
break;
|
|
}
|
|
}
|
|
}
|
|
return ret;
|
|
}
|
|
|
|
bool PeerConnection::UseCandidate(const IceCandidateInterface* candidate) {
|
|
RTC_DCHECK_RUN_ON(signaling_thread());
|
|
RTCErrorOr<const cricket::ContentInfo*> result =
|
|
FindContentInfo(remote_description(), candidate);
|
|
if (!result.ok()) {
|
|
RTC_LOG(LS_ERROR) << "UseCandidate: Invalid candidate. "
|
|
<< result.error().message();
|
|
return false;
|
|
}
|
|
std::vector<cricket::Candidate> candidates;
|
|
candidates.push_back(candidate->candidate());
|
|
// Invoking BaseSession method to handle remote candidates.
|
|
RTCError error = transport_controller_->AddRemoteCandidates(
|
|
result.value()->name, candidates);
|
|
if (error.ok()) {
|
|
ReportRemoteIceCandidateAdded(candidate->candidate());
|
|
// Candidates successfully submitted for checking.
|
|
if (ice_connection_state_ == PeerConnectionInterface::kIceConnectionNew ||
|
|
ice_connection_state_ ==
|
|
PeerConnectionInterface::kIceConnectionDisconnected) {
|
|
// If state is New, then the session has just gotten its first remote ICE
|
|
// candidates, so go to Checking.
|
|
// If state is Disconnected, the session is re-using old candidates or
|
|
// receiving additional ones, so go to Checking.
|
|
// If state is Connected, stay Connected.
|
|
// TODO(bemasc): If state is Connected, and the new candidates are for a
|
|
// newly added transport, then the state actually _should_ move to
|
|
// checking. Add a way to distinguish that case.
|
|
SetIceConnectionState(PeerConnectionInterface::kIceConnectionChecking);
|
|
}
|
|
// TODO(bemasc): If state is Completed, go back to Connected.
|
|
} else {
|
|
RTC_LOG(LS_WARNING) << error.message();
|
|
}
|
|
return true;
|
|
}
|
|
|
|
RTCErrorOr<const cricket::ContentInfo*> PeerConnection::FindContentInfo(
|
|
const SessionDescriptionInterface* description,
|
|
const IceCandidateInterface* candidate) {
|
|
if (candidate->sdp_mline_index() >= 0) {
|
|
size_t mediacontent_index =
|
|
static_cast<size_t>(candidate->sdp_mline_index());
|
|
size_t content_size = description->description()->contents().size();
|
|
if (mediacontent_index < content_size) {
|
|
return &description->description()->contents()[mediacontent_index];
|
|
} else {
|
|
return RTCError(RTCErrorType::INVALID_RANGE,
|
|
"Media line index (" +
|
|
rtc::ToString(candidate->sdp_mline_index()) +
|
|
") out of range (number of mlines: " +
|
|
rtc::ToString(content_size) + ").");
|
|
}
|
|
} else if (!candidate->sdp_mid().empty()) {
|
|
auto& contents = description->description()->contents();
|
|
auto it = absl::c_find_if(
|
|
contents, [candidate](const cricket::ContentInfo& content_info) {
|
|
return content_info.mid() == candidate->sdp_mid();
|
|
});
|
|
if (it == contents.end()) {
|
|
return RTCError(
|
|
RTCErrorType::INVALID_PARAMETER,
|
|
"Mid " + candidate->sdp_mid() +
|
|
" specified but no media section with that mid found.");
|
|
} else {
|
|
return &*it;
|
|
}
|
|
}
|
|
|
|
return RTCError(RTCErrorType::INVALID_PARAMETER,
|
|
"Neither sdp_mline_index nor sdp_mid specified.");
|
|
}
|
|
|
|
void PeerConnection::RemoveUnusedChannels(const SessionDescription* desc) {
|
|
RTC_DCHECK_RUN_ON(signaling_thread());
|
|
// Destroy video channel first since it may have a pointer to the
|
|
// voice channel.
|
|
const cricket::ContentInfo* video_info = cricket::GetFirstVideoContent(desc);
|
|
if (!video_info || video_info->rejected) {
|
|
DestroyTransceiverChannel(GetVideoTransceiver());
|
|
}
|
|
|
|
const cricket::ContentInfo* audio_info = cricket::GetFirstAudioContent(desc);
|
|
if (!audio_info || audio_info->rejected) {
|
|
DestroyTransceiverChannel(GetAudioTransceiver());
|
|
}
|
|
|
|
const cricket::ContentInfo* data_info = cricket::GetFirstDataContent(desc);
|
|
if (!data_info || data_info->rejected) {
|
|
DestroyDataChannelTransport();
|
|
}
|
|
}
|
|
|
|
RTCError PeerConnection::CreateChannels(const SessionDescription& desc) {
|
|
// Creating the media channels. Transports should already have been created
|
|
// at this point.
|
|
RTC_DCHECK_RUN_ON(signaling_thread());
|
|
const cricket::ContentInfo* voice = cricket::GetFirstAudioContent(&desc);
|
|
if (voice && !voice->rejected &&
|
|
!GetAudioTransceiver()->internal()->channel()) {
|
|
cricket::VoiceChannel* voice_channel = CreateVoiceChannel(voice->name);
|
|
if (!voice_channel) {
|
|
LOG_AND_RETURN_ERROR(RTCErrorType::INTERNAL_ERROR,
|
|
"Failed to create voice channel.");
|
|
}
|
|
GetAudioTransceiver()->internal()->SetChannel(voice_channel);
|
|
}
|
|
|
|
const cricket::ContentInfo* video = cricket::GetFirstVideoContent(&desc);
|
|
if (video && !video->rejected &&
|
|
!GetVideoTransceiver()->internal()->channel()) {
|
|
cricket::VideoChannel* video_channel = CreateVideoChannel(video->name);
|
|
if (!video_channel) {
|
|
LOG_AND_RETURN_ERROR(RTCErrorType::INTERNAL_ERROR,
|
|
"Failed to create video channel.");
|
|
}
|
|
GetVideoTransceiver()->internal()->SetChannel(video_channel);
|
|
}
|
|
|
|
const cricket::ContentInfo* data = cricket::GetFirstDataContent(&desc);
|
|
if (data_channel_type() != cricket::DCT_NONE && data && !data->rejected &&
|
|
!data_channel_controller_.rtp_data_channel() &&
|
|
!data_channel_controller_.data_channel_transport()) {
|
|
if (!CreateDataChannel(data->name)) {
|
|
LOG_AND_RETURN_ERROR(RTCErrorType::INTERNAL_ERROR,
|
|
"Failed to create data channel.");
|
|
}
|
|
}
|
|
|
|
return RTCError::OK();
|
|
}
|
|
|
|
// TODO(steveanton): Perhaps this should be managed by the RtpTransceiver.
|
|
cricket::VoiceChannel* PeerConnection::CreateVoiceChannel(
|
|
const std::string& mid) {
|
|
RTC_DCHECK_RUN_ON(signaling_thread());
|
|
RtpTransportInternal* rtp_transport = GetRtpTransport(mid);
|
|
|
|
// TODO(bugs.webrtc.org/11992): CreateVoiceChannel internally switches to the
|
|
// worker thread. We shouldn't be using the |call_ptr_| hack here but simply
|
|
// be on the worker thread and use |call_| (update upstream code).
|
|
cricket::VoiceChannel* voice_channel = channel_manager()->CreateVoiceChannel(
|
|
call_ptr_, configuration_.media_config, rtp_transport, signaling_thread(),
|
|
mid, SrtpRequired(), GetCryptoOptions(), &ssrc_generator_,
|
|
audio_options_);
|
|
if (!voice_channel) {
|
|
return nullptr;
|
|
}
|
|
voice_channel->SignalSentPacket().connect(this,
|
|
&PeerConnection::OnSentPacket_w);
|
|
voice_channel->SetRtpTransport(rtp_transport);
|
|
|
|
return voice_channel;
|
|
}
|
|
|
|
// TODO(steveanton): Perhaps this should be managed by the RtpTransceiver.
|
|
cricket::VideoChannel* PeerConnection::CreateVideoChannel(
|
|
const std::string& mid) {
|
|
RTC_DCHECK_RUN_ON(signaling_thread());
|
|
RtpTransportInternal* rtp_transport = GetRtpTransport(mid);
|
|
|
|
// TODO(bugs.webrtc.org/11992): CreateVideoChannel internally switches to the
|
|
// worker thread. We shouldn't be using the |call_ptr_| hack here but simply
|
|
// be on the worker thread and use |call_| (update upstream code).
|
|
cricket::VideoChannel* video_channel = channel_manager()->CreateVideoChannel(
|
|
call_ptr_, configuration_.media_config, rtp_transport, signaling_thread(),
|
|
mid, SrtpRequired(), GetCryptoOptions(), &ssrc_generator_, video_options_,
|
|
video_bitrate_allocator_factory_.get());
|
|
if (!video_channel) {
|
|
return nullptr;
|
|
}
|
|
video_channel->SignalSentPacket().connect(this,
|
|
&PeerConnection::OnSentPacket_w);
|
|
video_channel->SetRtpTransport(rtp_transport);
|
|
|
|
return video_channel;
|
|
}
|
|
|
|
bool PeerConnection::CreateDataChannel(const std::string& mid) {
|
|
RTC_DCHECK_RUN_ON(signaling_thread());
|
|
switch (data_channel_type()) {
|
|
case cricket::DCT_SCTP:
|
|
if (network_thread()->Invoke<bool>(
|
|
RTC_FROM_HERE,
|
|
rtc::Bind(&PeerConnection::SetupDataChannelTransport_n, this,
|
|
mid))) {
|
|
sctp_mid_s_ = mid;
|
|
} else {
|
|
return false;
|
|
}
|
|
return true;
|
|
case cricket::DCT_RTP:
|
|
default:
|
|
RtpTransportInternal* rtp_transport = GetRtpTransport(mid);
|
|
// TODO(bugs.webrtc.org/9987): set_rtp_data_channel() should be called on
|
|
// the network thread like set_data_channel_transport is.
|
|
data_channel_controller_.set_rtp_data_channel(
|
|
channel_manager()->CreateRtpDataChannel(
|
|
configuration_.media_config, rtp_transport, signaling_thread(),
|
|
mid, SrtpRequired(), GetCryptoOptions(), &ssrc_generator_));
|
|
if (!data_channel_controller_.rtp_data_channel()) {
|
|
return false;
|
|
}
|
|
data_channel_controller_.rtp_data_channel()->SignalSentPacket().connect(
|
|
this, &PeerConnection::OnSentPacket_w);
|
|
data_channel_controller_.rtp_data_channel()->SetRtpTransport(
|
|
rtp_transport);
|
|
have_pending_rtp_data_channel_ = true;
|
|
return true;
|
|
}
|
|
return false;
|
|
}
|
|
|
|
Call::Stats PeerConnection::GetCallStats() {
|
|
if (!worker_thread()->IsCurrent()) {
|
|
return worker_thread()->Invoke<Call::Stats>(
|
|
RTC_FROM_HERE, rtc::Bind(&PeerConnection::GetCallStats, this));
|
|
}
|
|
RTC_DCHECK_RUN_ON(worker_thread());
|
|
rtc::Thread::ScopedDisallowBlockingCalls no_blocking_calls;
|
|
if (call_) {
|
|
return call_->GetStats();
|
|
} else {
|
|
return Call::Stats();
|
|
}
|
|
}
|
|
|
|
bool PeerConnection::SetupDataChannelTransport_n(const std::string& mid) {
|
|
DataChannelTransportInterface* transport =
|
|
transport_controller_->GetDataChannelTransport(mid);
|
|
if (!transport) {
|
|
RTC_LOG(LS_ERROR)
|
|
<< "Data channel transport is not available for data channels, mid="
|
|
<< mid;
|
|
return false;
|
|
}
|
|
RTC_LOG(LS_INFO) << "Setting up data channel transport for mid=" << mid;
|
|
|
|
data_channel_controller_.set_data_channel_transport(transport);
|
|
data_channel_controller_.SetupDataChannelTransport_n();
|
|
sctp_mid_n_ = mid;
|
|
|
|
// Note: setting the data sink and checking initial state must be done last,
|
|
// after setting up the data channel. Setting the data sink may trigger
|
|
// callbacks to PeerConnection which require the transport to be completely
|
|
// set up (eg. OnReadyToSend()).
|
|
transport->SetDataSink(&data_channel_controller_);
|
|
return true;
|
|
}
|
|
|
|
void PeerConnection::TeardownDataChannelTransport_n() {
|
|
if (!sctp_mid_n_ && !data_channel_controller_.data_channel_transport()) {
|
|
return;
|
|
}
|
|
RTC_LOG(LS_INFO) << "Tearing down data channel transport for mid="
|
|
<< *sctp_mid_n_;
|
|
|
|
// |sctp_mid_| may still be active through an SCTP transport. If not, unset
|
|
// it.
|
|
sctp_mid_n_.reset();
|
|
data_channel_controller_.TeardownDataChannelTransport_n();
|
|
}
|
|
|
|
// Returns false if bundle is enabled and rtcp_mux is disabled.
|
|
bool PeerConnection::ValidateBundleSettings(const SessionDescription* desc) {
|
|
bool bundle_enabled = desc->HasGroup(cricket::GROUP_TYPE_BUNDLE);
|
|
if (!bundle_enabled)
|
|
return true;
|
|
|
|
const cricket::ContentGroup* bundle_group =
|
|
desc->GetGroupByName(cricket::GROUP_TYPE_BUNDLE);
|
|
RTC_DCHECK(bundle_group != NULL);
|
|
|
|
const cricket::ContentInfos& contents = desc->contents();
|
|
for (cricket::ContentInfos::const_iterator citer = contents.begin();
|
|
citer != contents.end(); ++citer) {
|
|
const cricket::ContentInfo* content = (&*citer);
|
|
RTC_DCHECK(content != NULL);
|
|
if (bundle_group->HasContentName(content->name) && !content->rejected &&
|
|
content->type == MediaProtocolType::kRtp) {
|
|
if (!HasRtcpMuxEnabled(content))
|
|
return false;
|
|
}
|
|
}
|
|
// RTCP-MUX is enabled in all the contents.
|
|
return true;
|
|
}
|
|
|
|
bool PeerConnection::HasRtcpMuxEnabled(const cricket::ContentInfo* content) {
|
|
return content->media_description()->rtcp_mux();
|
|
}
|
|
|
|
bool PeerConnection::ExpectSetLocalDescription(SdpType type) {
|
|
PeerConnectionInterface::SignalingState state = signaling_state();
|
|
if (type == SdpType::kOffer) {
|
|
return (state == PeerConnectionInterface::kStable) ||
|
|
(state == PeerConnectionInterface::kHaveLocalOffer);
|
|
} else {
|
|
RTC_DCHECK(type == SdpType::kPrAnswer || type == SdpType::kAnswer);
|
|
return (state == PeerConnectionInterface::kHaveRemoteOffer) ||
|
|
(state == PeerConnectionInterface::kHaveLocalPrAnswer);
|
|
}
|
|
}
|
|
|
|
bool PeerConnection::ExpectSetRemoteDescription(SdpType type) {
|
|
PeerConnectionInterface::SignalingState state = signaling_state();
|
|
if (type == SdpType::kOffer) {
|
|
return (state == PeerConnectionInterface::kStable) ||
|
|
(state == PeerConnectionInterface::kHaveRemoteOffer);
|
|
} else {
|
|
RTC_DCHECK(type == SdpType::kPrAnswer || type == SdpType::kAnswer);
|
|
return (state == PeerConnectionInterface::kHaveLocalOffer) ||
|
|
(state == PeerConnectionInterface::kHaveRemotePrAnswer);
|
|
}
|
|
}
|
|
|
|
const char* PeerConnection::SessionErrorToString(SessionError error) const {
|
|
switch (error) {
|
|
case SessionError::kNone:
|
|
return "ERROR_NONE";
|
|
case SessionError::kContent:
|
|
return "ERROR_CONTENT";
|
|
case SessionError::kTransport:
|
|
return "ERROR_TRANSPORT";
|
|
}
|
|
RTC_NOTREACHED();
|
|
return "";
|
|
}
|
|
|
|
std::string PeerConnection::GetSessionErrorMsg() {
|
|
RTC_DCHECK_RUN_ON(signaling_thread());
|
|
rtc::StringBuilder desc;
|
|
desc << kSessionError << SessionErrorToString(session_error()) << ". ";
|
|
desc << kSessionErrorDesc << session_error_desc() << ".";
|
|
return desc.Release();
|
|
}
|
|
|
|
void PeerConnection::ReportSdpFormatReceived(
|
|
const SessionDescriptionInterface& remote_offer) {
|
|
int num_audio_mlines = 0;
|
|
int num_video_mlines = 0;
|
|
int num_audio_tracks = 0;
|
|
int num_video_tracks = 0;
|
|
for (const ContentInfo& content : remote_offer.description()->contents()) {
|
|
cricket::MediaType media_type = content.media_description()->type();
|
|
int num_tracks = std::max(
|
|
1, static_cast<int>(content.media_description()->streams().size()));
|
|
if (media_type == cricket::MEDIA_TYPE_AUDIO) {
|
|
num_audio_mlines += 1;
|
|
num_audio_tracks += num_tracks;
|
|
} else if (media_type == cricket::MEDIA_TYPE_VIDEO) {
|
|
num_video_mlines += 1;
|
|
num_video_tracks += num_tracks;
|
|
}
|
|
}
|
|
SdpFormatReceived format = kSdpFormatReceivedNoTracks;
|
|
if (num_audio_mlines > 1 || num_video_mlines > 1) {
|
|
format = kSdpFormatReceivedComplexUnifiedPlan;
|
|
} else if (num_audio_tracks > 1 || num_video_tracks > 1) {
|
|
format = kSdpFormatReceivedComplexPlanB;
|
|
} else if (num_audio_tracks > 0 || num_video_tracks > 0) {
|
|
format = kSdpFormatReceivedSimple;
|
|
}
|
|
RTC_HISTOGRAM_ENUMERATION("WebRTC.PeerConnection.SdpFormatReceived", format,
|
|
kSdpFormatReceivedMax);
|
|
}
|
|
|
|
void PeerConnection::ReportIceCandidateCollected(
|
|
const cricket::Candidate& candidate) {
|
|
NoteUsageEvent(UsageEvent::CANDIDATE_COLLECTED);
|
|
if (candidate.address().IsPrivateIP()) {
|
|
NoteUsageEvent(UsageEvent::PRIVATE_CANDIDATE_COLLECTED);
|
|
}
|
|
if (candidate.address().IsUnresolvedIP()) {
|
|
NoteUsageEvent(UsageEvent::MDNS_CANDIDATE_COLLECTED);
|
|
}
|
|
if (candidate.address().family() == AF_INET6) {
|
|
NoteUsageEvent(UsageEvent::IPV6_CANDIDATE_COLLECTED);
|
|
}
|
|
}
|
|
|
|
void PeerConnection::ReportRemoteIceCandidateAdded(
|
|
const cricket::Candidate& candidate) {
|
|
NoteUsageEvent(UsageEvent::REMOTE_CANDIDATE_ADDED);
|
|
if (candidate.address().IsPrivateIP()) {
|
|
NoteUsageEvent(UsageEvent::REMOTE_PRIVATE_CANDIDATE_ADDED);
|
|
}
|
|
if (candidate.address().IsUnresolvedIP()) {
|
|
NoteUsageEvent(UsageEvent::REMOTE_MDNS_CANDIDATE_ADDED);
|
|
}
|
|
if (candidate.address().family() == AF_INET6) {
|
|
NoteUsageEvent(UsageEvent::REMOTE_IPV6_CANDIDATE_ADDED);
|
|
}
|
|
}
|
|
|
|
void PeerConnection::NoteUsageEvent(UsageEvent event) {
|
|
RTC_DCHECK_RUN_ON(signaling_thread());
|
|
usage_event_accumulator_ |= static_cast<int>(event);
|
|
}
|
|
|
|
void PeerConnection::ReportUsagePattern() const {
|
|
RTC_DLOG(LS_INFO) << "Usage signature is " << usage_event_accumulator_;
|
|
RTC_HISTOGRAM_ENUMERATION_SPARSE("WebRTC.PeerConnection.UsagePattern",
|
|
usage_event_accumulator_,
|
|
static_cast<int>(UsageEvent::MAX_VALUE));
|
|
const int bad_bits =
|
|
static_cast<int>(UsageEvent::SET_LOCAL_DESCRIPTION_SUCCEEDED) |
|
|
static_cast<int>(UsageEvent::CANDIDATE_COLLECTED);
|
|
const int good_bits =
|
|
static_cast<int>(UsageEvent::SET_REMOTE_DESCRIPTION_SUCCEEDED) |
|
|
static_cast<int>(UsageEvent::REMOTE_CANDIDATE_ADDED) |
|
|
static_cast<int>(UsageEvent::ICE_STATE_CONNECTED);
|
|
if ((usage_event_accumulator_ & bad_bits) == bad_bits &&
|
|
(usage_event_accumulator_ & good_bits) == 0) {
|
|
// If called after close(), we can't report, because observer may have
|
|
// been deallocated, and therefore pointer is null. Write to log instead.
|
|
if (observer_) {
|
|
Observer()->OnInterestingUsage(usage_event_accumulator_);
|
|
} else {
|
|
RTC_LOG(LS_INFO) << "Interesting usage signature "
|
|
<< usage_event_accumulator_
|
|
<< " observed after observer shutdown";
|
|
}
|
|
}
|
|
}
|
|
|
|
void PeerConnection::ReportNegotiatedSdpSemantics(
|
|
const SessionDescriptionInterface& answer) {
|
|
SdpSemanticNegotiated semantics_negotiated;
|
|
switch (answer.description()->msid_signaling()) {
|
|
case 0:
|
|
semantics_negotiated = kSdpSemanticNegotiatedNone;
|
|
break;
|
|
case cricket::kMsidSignalingMediaSection:
|
|
semantics_negotiated = kSdpSemanticNegotiatedUnifiedPlan;
|
|
break;
|
|
case cricket::kMsidSignalingSsrcAttribute:
|
|
semantics_negotiated = kSdpSemanticNegotiatedPlanB;
|
|
break;
|
|
case cricket::kMsidSignalingMediaSection |
|
|
cricket::kMsidSignalingSsrcAttribute:
|
|
semantics_negotiated = kSdpSemanticNegotiatedMixed;
|
|
break;
|
|
default:
|
|
RTC_NOTREACHED();
|
|
}
|
|
RTC_HISTOGRAM_ENUMERATION("WebRTC.PeerConnection.SdpSemanticNegotiated",
|
|
semantics_negotiated, kSdpSemanticNegotiatedMax);
|
|
}
|
|
|
|
// We need to check the local/remote description for the Transport instead of
|
|
// the session, because a new Transport added during renegotiation may have
|
|
// them unset while the session has them set from the previous negotiation.
|
|
// Not doing so may trigger the auto generation of transport description and
|
|
// mess up DTLS identity information, ICE credential, etc.
|
|
bool PeerConnection::ReadyToUseRemoteCandidate(
|
|
const IceCandidateInterface* candidate,
|
|
const SessionDescriptionInterface* remote_desc,
|
|
bool* valid) {
|
|
RTC_DCHECK_RUN_ON(signaling_thread());
|
|
*valid = true;
|
|
|
|
const SessionDescriptionInterface* current_remote_desc =
|
|
remote_desc ? remote_desc : remote_description();
|
|
|
|
if (!current_remote_desc) {
|
|
return false;
|
|
}
|
|
|
|
RTCErrorOr<const cricket::ContentInfo*> result =
|
|
FindContentInfo(current_remote_desc, candidate);
|
|
if (!result.ok()) {
|
|
RTC_LOG(LS_ERROR) << "ReadyToUseRemoteCandidate: Invalid candidate. "
|
|
<< result.error().message();
|
|
|
|
*valid = false;
|
|
return false;
|
|
}
|
|
|
|
std::string transport_name = GetTransportName(result.value()->name);
|
|
return !transport_name.empty();
|
|
}
|
|
|
|
bool PeerConnection::SrtpRequired() const {
|
|
return (dtls_enabled_ ||
|
|
sdp_handler_.webrtc_session_desc_factory()->SdesPolicy() ==
|
|
cricket::SEC_REQUIRED);
|
|
}
|
|
|
|
void PeerConnection::OnTransportControllerGatheringState(
|
|
cricket::IceGatheringState state) {
|
|
RTC_DCHECK(signaling_thread()->IsCurrent());
|
|
if (state == cricket::kIceGatheringGathering) {
|
|
OnIceGatheringChange(PeerConnectionInterface::kIceGatheringGathering);
|
|
} else if (state == cricket::kIceGatheringComplete) {
|
|
OnIceGatheringChange(PeerConnectionInterface::kIceGatheringComplete);
|
|
} else if (state == cricket::kIceGatheringNew) {
|
|
OnIceGatheringChange(PeerConnectionInterface::kIceGatheringNew);
|
|
} else {
|
|
RTC_LOG(LS_ERROR) << "Unknown state received: " << state;
|
|
RTC_NOTREACHED();
|
|
}
|
|
}
|
|
|
|
void PeerConnection::ReportTransportStats() {
|
|
std::map<std::string, std::set<cricket::MediaType>>
|
|
media_types_by_transport_name;
|
|
for (const auto& transceiver : transceivers_.List()) {
|
|
if (transceiver->internal()->channel()) {
|
|
const std::string& transport_name =
|
|
transceiver->internal()->channel()->transport_name();
|
|
media_types_by_transport_name[transport_name].insert(
|
|
transceiver->media_type());
|
|
}
|
|
}
|
|
if (rtp_data_channel()) {
|
|
media_types_by_transport_name[rtp_data_channel()->transport_name()].insert(
|
|
cricket::MEDIA_TYPE_DATA);
|
|
}
|
|
|
|
absl::optional<std::string> transport_name = sctp_transport_name();
|
|
if (transport_name) {
|
|
media_types_by_transport_name[*transport_name].insert(
|
|
cricket::MEDIA_TYPE_DATA);
|
|
}
|
|
|
|
for (const auto& entry : media_types_by_transport_name) {
|
|
const std::string& transport_name = entry.first;
|
|
const std::set<cricket::MediaType> media_types = entry.second;
|
|
cricket::TransportStats stats;
|
|
if (transport_controller_->GetStats(transport_name, &stats)) {
|
|
ReportBestConnectionState(stats);
|
|
ReportNegotiatedCiphers(stats, media_types);
|
|
}
|
|
}
|
|
}
|
|
// Walk through the ConnectionInfos to gather best connection usage
|
|
// for IPv4 and IPv6.
|
|
void PeerConnection::ReportBestConnectionState(
|
|
const cricket::TransportStats& stats) {
|
|
for (const cricket::TransportChannelStats& channel_stats :
|
|
stats.channel_stats) {
|
|
for (const cricket::ConnectionInfo& connection_info :
|
|
channel_stats.ice_transport_stats.connection_infos) {
|
|
if (!connection_info.best_connection) {
|
|
continue;
|
|
}
|
|
|
|
const cricket::Candidate& local = connection_info.local_candidate;
|
|
const cricket::Candidate& remote = connection_info.remote_candidate;
|
|
|
|
// Increment the counter for IceCandidatePairType.
|
|
if (local.protocol() == cricket::TCP_PROTOCOL_NAME ||
|
|
(local.type() == RELAY_PORT_TYPE &&
|
|
local.relay_protocol() == cricket::TCP_PROTOCOL_NAME)) {
|
|
RTC_HISTOGRAM_ENUMERATION("WebRTC.PeerConnection.CandidatePairType_TCP",
|
|
GetIceCandidatePairCounter(local, remote),
|
|
kIceCandidatePairMax);
|
|
} else if (local.protocol() == cricket::UDP_PROTOCOL_NAME) {
|
|
RTC_HISTOGRAM_ENUMERATION("WebRTC.PeerConnection.CandidatePairType_UDP",
|
|
GetIceCandidatePairCounter(local, remote),
|
|
kIceCandidatePairMax);
|
|
} else {
|
|
RTC_CHECK(0);
|
|
}
|
|
|
|
// Increment the counter for IP type.
|
|
if (local.address().family() == AF_INET) {
|
|
RTC_HISTOGRAM_ENUMERATION("WebRTC.PeerConnection.IPMetrics",
|
|
kBestConnections_IPv4,
|
|
kPeerConnectionAddressFamilyCounter_Max);
|
|
} else if (local.address().family() == AF_INET6) {
|
|
RTC_HISTOGRAM_ENUMERATION("WebRTC.PeerConnection.IPMetrics",
|
|
kBestConnections_IPv6,
|
|
kPeerConnectionAddressFamilyCounter_Max);
|
|
} else {
|
|
RTC_CHECK(!local.address().hostname().empty() &&
|
|
local.address().IsUnresolvedIP());
|
|
}
|
|
|
|
return;
|
|
}
|
|
}
|
|
}
|
|
|
|
void PeerConnection::ReportNegotiatedCiphers(
|
|
const cricket::TransportStats& stats,
|
|
const std::set<cricket::MediaType>& media_types) {
|
|
if (!dtls_enabled_ || stats.channel_stats.empty()) {
|
|
return;
|
|
}
|
|
|
|
int srtp_crypto_suite = stats.channel_stats[0].srtp_crypto_suite;
|
|
int ssl_cipher_suite = stats.channel_stats[0].ssl_cipher_suite;
|
|
if (srtp_crypto_suite == rtc::SRTP_INVALID_CRYPTO_SUITE &&
|
|
ssl_cipher_suite == rtc::TLS_NULL_WITH_NULL_NULL) {
|
|
return;
|
|
}
|
|
|
|
if (srtp_crypto_suite != rtc::SRTP_INVALID_CRYPTO_SUITE) {
|
|
for (cricket::MediaType media_type : media_types) {
|
|
switch (media_type) {
|
|
case cricket::MEDIA_TYPE_AUDIO:
|
|
RTC_HISTOGRAM_ENUMERATION_SPARSE(
|
|
"WebRTC.PeerConnection.SrtpCryptoSuite.Audio", srtp_crypto_suite,
|
|
rtc::SRTP_CRYPTO_SUITE_MAX_VALUE);
|
|
break;
|
|
case cricket::MEDIA_TYPE_VIDEO:
|
|
RTC_HISTOGRAM_ENUMERATION_SPARSE(
|
|
"WebRTC.PeerConnection.SrtpCryptoSuite.Video", srtp_crypto_suite,
|
|
rtc::SRTP_CRYPTO_SUITE_MAX_VALUE);
|
|
break;
|
|
case cricket::MEDIA_TYPE_DATA:
|
|
RTC_HISTOGRAM_ENUMERATION_SPARSE(
|
|
"WebRTC.PeerConnection.SrtpCryptoSuite.Data", srtp_crypto_suite,
|
|
rtc::SRTP_CRYPTO_SUITE_MAX_VALUE);
|
|
break;
|
|
default:
|
|
RTC_NOTREACHED();
|
|
continue;
|
|
}
|
|
}
|
|
}
|
|
|
|
if (ssl_cipher_suite != rtc::TLS_NULL_WITH_NULL_NULL) {
|
|
for (cricket::MediaType media_type : media_types) {
|
|
switch (media_type) {
|
|
case cricket::MEDIA_TYPE_AUDIO:
|
|
RTC_HISTOGRAM_ENUMERATION_SPARSE(
|
|
"WebRTC.PeerConnection.SslCipherSuite.Audio", ssl_cipher_suite,
|
|
rtc::SSL_CIPHER_SUITE_MAX_VALUE);
|
|
break;
|
|
case cricket::MEDIA_TYPE_VIDEO:
|
|
RTC_HISTOGRAM_ENUMERATION_SPARSE(
|
|
"WebRTC.PeerConnection.SslCipherSuite.Video", ssl_cipher_suite,
|
|
rtc::SSL_CIPHER_SUITE_MAX_VALUE);
|
|
break;
|
|
case cricket::MEDIA_TYPE_DATA:
|
|
RTC_HISTOGRAM_ENUMERATION_SPARSE(
|
|
"WebRTC.PeerConnection.SslCipherSuite.Data", ssl_cipher_suite,
|
|
rtc::SSL_CIPHER_SUITE_MAX_VALUE);
|
|
break;
|
|
default:
|
|
RTC_NOTREACHED();
|
|
continue;
|
|
}
|
|
}
|
|
}
|
|
}
|
|
|
|
void PeerConnection::OnSentPacket_w(const rtc::SentPacket& sent_packet) {
|
|
RTC_DCHECK_RUN_ON(worker_thread());
|
|
RTC_DCHECK(call_);
|
|
call_->OnSentPacket(sent_packet);
|
|
}
|
|
|
|
const std::string PeerConnection::GetTransportName(
|
|
const std::string& content_name) {
|
|
cricket::ChannelInterface* channel = GetChannel(content_name);
|
|
if (channel) {
|
|
return channel->transport_name();
|
|
}
|
|
if (data_channel_controller_.data_channel_transport()) {
|
|
RTC_DCHECK(sctp_mid_s_);
|
|
if (content_name == *sctp_mid_s_) {
|
|
return *sctp_transport_name();
|
|
}
|
|
}
|
|
// Return an empty string if failed to retrieve the transport name.
|
|
return "";
|
|
}
|
|
|
|
void PeerConnection::DestroyTransceiverChannel(
|
|
rtc::scoped_refptr<RtpTransceiverProxyWithInternal<RtpTransceiver>>
|
|
transceiver) {
|
|
RTC_DCHECK(transceiver);
|
|
|
|
cricket::ChannelInterface* channel = transceiver->internal()->channel();
|
|
if (channel) {
|
|
transceiver->internal()->SetChannel(nullptr);
|
|
DestroyChannelInterface(channel);
|
|
}
|
|
}
|
|
|
|
void PeerConnection::DestroyDataChannelTransport() {
|
|
RTC_DCHECK_RUN_ON(signaling_thread());
|
|
if (data_channel_controller_.rtp_data_channel()) {
|
|
data_channel_controller_.OnTransportChannelClosed();
|
|
DestroyChannelInterface(data_channel_controller_.rtp_data_channel());
|
|
data_channel_controller_.set_rtp_data_channel(nullptr);
|
|
}
|
|
|
|
// Note: Cannot use rtc::Bind to create a functor to invoke because it will
|
|
// grab a reference to this PeerConnection. If this is called from the
|
|
// PeerConnection destructor, the RefCountedObject vtable will have already
|
|
// been destroyed (since it is a subclass of PeerConnection) and using
|
|
// rtc::Bind will cause "Pure virtual function called" error to appear.
|
|
|
|
if (sctp_mid_s_) {
|
|
data_channel_controller_.OnTransportChannelClosed();
|
|
network_thread()->Invoke<void>(RTC_FROM_HERE, [this] {
|
|
RTC_DCHECK_RUN_ON(network_thread());
|
|
TeardownDataChannelTransport_n();
|
|
});
|
|
sctp_mid_s_.reset();
|
|
}
|
|
}
|
|
|
|
void PeerConnection::DestroyChannelInterface(
|
|
cricket::ChannelInterface* channel) {
|
|
// TODO(bugs.webrtc.org/11992): All the below methods should be called on the
|
|
// worker thread. (they switch internally anyway). Change
|
|
// DestroyChannelInterface to either be called on the worker thread, or do
|
|
// this asynchronously on the worker.
|
|
RTC_DCHECK(channel);
|
|
switch (channel->media_type()) {
|
|
case cricket::MEDIA_TYPE_AUDIO:
|
|
channel_manager()->DestroyVoiceChannel(
|
|
static_cast<cricket::VoiceChannel*>(channel));
|
|
break;
|
|
case cricket::MEDIA_TYPE_VIDEO:
|
|
channel_manager()->DestroyVideoChannel(
|
|
static_cast<cricket::VideoChannel*>(channel));
|
|
break;
|
|
case cricket::MEDIA_TYPE_DATA:
|
|
channel_manager()->DestroyRtpDataChannel(
|
|
static_cast<cricket::RtpDataChannel*>(channel));
|
|
break;
|
|
default:
|
|
RTC_NOTREACHED() << "Unknown media type: " << channel->media_type();
|
|
break;
|
|
}
|
|
}
|
|
|
|
bool PeerConnection::OnTransportChanged(
|
|
const std::string& mid,
|
|
RtpTransportInternal* rtp_transport,
|
|
rtc::scoped_refptr<DtlsTransport> dtls_transport,
|
|
DataChannelTransportInterface* data_channel_transport) {
|
|
RTC_DCHECK_RUN_ON(network_thread());
|
|
bool ret = true;
|
|
auto base_channel = GetChannel(mid);
|
|
if (base_channel) {
|
|
ret = base_channel->SetRtpTransport(rtp_transport);
|
|
}
|
|
if (mid == sctp_mid_n_) {
|
|
data_channel_controller_.OnTransportChanged(data_channel_transport);
|
|
}
|
|
return ret;
|
|
}
|
|
|
|
void PeerConnection::OnSetStreams() {
|
|
RTC_DCHECK_RUN_ON(signaling_thread());
|
|
if (IsUnifiedPlan())
|
|
sdp_handler_.UpdateNegotiationNeeded();
|
|
}
|
|
|
|
PeerConnectionObserver* PeerConnection::Observer() const {
|
|
RTC_DCHECK_RUN_ON(signaling_thread());
|
|
RTC_DCHECK(observer_);
|
|
return observer_;
|
|
}
|
|
|
|
CryptoOptions PeerConnection::GetCryptoOptions() {
|
|
// TODO(bugs.webrtc.org/9891) - Remove PeerConnectionFactory::CryptoOptions
|
|
// after it has been removed.
|
|
return configuration_.crypto_options.has_value()
|
|
? *configuration_.crypto_options
|
|
: factory_->options().crypto_options;
|
|
}
|
|
|
|
void PeerConnection::ClearStatsCache() {
|
|
RTC_DCHECK_RUN_ON(signaling_thread());
|
|
if (stats_collector_) {
|
|
stats_collector_->ClearCachedStatsReport();
|
|
}
|
|
}
|
|
|
|
void PeerConnection::RequestUsagePatternReportForTesting() {
|
|
signaling_thread()->Post(RTC_FROM_HERE, this, MSG_REPORT_USAGE_PATTERN,
|
|
nullptr);
|
|
}
|
|
|
|
bool PeerConnection::ShouldFireNegotiationNeededEvent(uint32_t event_id) {
|
|
RTC_DCHECK_RUN_ON(signaling_thread());
|
|
return sdp_handler_.ShouldFireNegotiationNeededEvent(event_id);
|
|
}
|
|
|
|
std::function<void(const rtc::CopyOnWriteBuffer& packet,
|
|
int64_t packet_time_us)>
|
|
PeerConnection::InitializeRtcpCallback() {
|
|
RTC_DCHECK_RUN_ON(signaling_thread());
|
|
|
|
auto flag =
|
|
worker_thread()->Invoke<rtc::scoped_refptr<PendingTaskSafetyFlag>>(
|
|
RTC_FROM_HERE, [this] {
|
|
RTC_DCHECK_RUN_ON(worker_thread());
|
|
if (!call_)
|
|
return rtc::scoped_refptr<PendingTaskSafetyFlag>();
|
|
if (!call_safety_)
|
|
call_safety_.reset(new ScopedTaskSafety());
|
|
return call_safety_->flag();
|
|
});
|
|
|
|
if (!flag)
|
|
return [](const rtc::CopyOnWriteBuffer&, int64_t) {};
|
|
|
|
return [this, flag = std::move(flag)](const rtc::CopyOnWriteBuffer& packet,
|
|
int64_t packet_time_us) {
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RTC_DCHECK_RUN_ON(network_thread());
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// TODO(bugs.webrtc.org/11993): We should actually be delivering this call
|
|
// directly to the Call class somehow directly on the network thread and not
|
|
// incur this hop here. The DeliverPacket() method will eventually just have
|
|
// to hop back over to the network thread.
|
|
worker_thread()->PostTask(ToQueuedTask(flag, [this, packet,
|
|
packet_time_us] {
|
|
RTC_DCHECK_RUN_ON(worker_thread());
|
|
call_->Receiver()->DeliverPacket(MediaType::ANY, packet, packet_time_us);
|
|
}));
|
|
};
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}
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} // namespace webrtc
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