This reverts commit ba97ba7af917d4152f5f3363aba1c1561c6673dc. Reason for revert: The new tests have caused several test failures on the test bots; the method FakeAudioMediaStreamTrack:GetSignalLevel, which is not supposed to be called is sometimes called anyway. Original change's description: > Added PeerConnectionObserver::OnRemoveTrack. > > This corresponds to processing the removal of a remote track step of > the spec, with processing the addition of a remote track already > covered by OnAddTrack. > https://w3c.github.io/webrtc-pc/#processing-remote-mediastreamtracks > > Bug: webrtc:8260, webrtc:8315 > Change-Id: Ica8be92369733eb3cf1397fb60385d45a9b58700 > Reviewed-on: https://webrtc-review.googlesource.com/4722 > Commit-Queue: Henrik Boström <hbos@webrtc.org> > Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org> > Reviewed-by: Steve Anton <steveanton@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#20214} TBR=steveanton@webrtc.org,deadbeef@webrtc.org,hbos@webrtc.org Change-Id: Id2d9533e27227254769b4280a8ff10a47313e714 No-Presubmit: true No-Tree-Checks: true No-Try: true Bug: webrtc:8260, webrtc:8315 Reviewed-on: https://webrtc-review.googlesource.com/7940 Reviewed-by: Alex Loiko <aleloi@webrtc.org> Commit-Queue: Alex Loiko <aleloi@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20218}
515 lines
22 KiB
C++
515 lines
22 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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#ifndef PC_PEERCONNECTION_H_
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#define PC_PEERCONNECTION_H_
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#include <string>
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#include <map>
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#include <memory>
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#include <vector>
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#include "api/peerconnectioninterface.h"
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#include "pc/iceserverparsing.h"
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#include "pc/peerconnectionfactory.h"
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#include "pc/rtcstatscollector.h"
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#include "pc/rtpreceiver.h"
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#include "pc/rtpsender.h"
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#include "pc/statscollector.h"
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#include "pc/streamcollection.h"
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#include "pc/webrtcsession.h"
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namespace webrtc {
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class MediaStreamObserver;
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class VideoRtpReceiver;
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class RtcEventLog;
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// TODO(zhihuang): Remove this declaration when the WebRtcSession tests don't
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// need it.
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void ExtractSharedMediaSessionOptions(
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const PeerConnectionInterface::RTCOfferAnswerOptions& rtc_options,
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cricket::MediaSessionOptions* session_options);
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// PeerConnection implements the PeerConnectionInterface interface.
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// It uses WebRtcSession to implement the PeerConnection functionality.
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class PeerConnection : public PeerConnectionInterface,
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public IceObserver,
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public rtc::MessageHandler,
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public sigslot::has_slots<> {
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public:
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explicit PeerConnection(PeerConnectionFactory* factory,
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std::unique_ptr<RtcEventLog> event_log,
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std::unique_ptr<Call> call);
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bool Initialize(
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const PeerConnectionInterface::RTCConfiguration& configuration,
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std::unique_ptr<cricket::PortAllocator> allocator,
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std::unique_ptr<rtc::RTCCertificateGeneratorInterface> cert_generator,
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PeerConnectionObserver* observer);
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rtc::scoped_refptr<StreamCollectionInterface> local_streams() override;
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rtc::scoped_refptr<StreamCollectionInterface> remote_streams() override;
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bool AddStream(MediaStreamInterface* local_stream) override;
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void RemoveStream(MediaStreamInterface* local_stream) override;
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rtc::scoped_refptr<RtpSenderInterface> AddTrack(
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MediaStreamTrackInterface* track,
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std::vector<MediaStreamInterface*> streams) override;
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bool RemoveTrack(RtpSenderInterface* sender) override;
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// TODO(steveanton): Remove this once all clients have switched to using the
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// PeerConnection shims for WebRtcSession methods instead of the methods
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// directly via this getter.
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virtual WebRtcSession* session() { return session_; }
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// Gets the DTLS SSL certificate associated with the audio transport on the
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// remote side. This will become populated once the DTLS connection with the
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// peer has been completed, as indicated by the ICE connection state
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// transitioning to kIceConnectionCompleted.
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// Note that this will be removed once we implement RTCDtlsTransport which
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// has standardized method for getting this information.
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// See https://www.w3.org/TR/webrtc/#rtcdtlstransport-interface
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std::unique_ptr<rtc::SSLCertificate> GetRemoteAudioSSLCertificate();
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rtc::scoped_refptr<DtmfSenderInterface> CreateDtmfSender(
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AudioTrackInterface* track) override;
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rtc::scoped_refptr<RtpSenderInterface> CreateSender(
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const std::string& kind,
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const std::string& stream_id) override;
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std::vector<rtc::scoped_refptr<RtpSenderInterface>> GetSenders()
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const override;
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std::vector<rtc::scoped_refptr<RtpReceiverInterface>> GetReceivers()
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const override;
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rtc::scoped_refptr<DataChannelInterface> CreateDataChannel(
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const std::string& label,
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const DataChannelInit* config) override;
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bool GetStats(StatsObserver* observer,
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webrtc::MediaStreamTrackInterface* track,
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StatsOutputLevel level) override;
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void GetStats(RTCStatsCollectorCallback* callback) override;
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SignalingState signaling_state() override;
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IceConnectionState ice_connection_state() override;
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IceGatheringState ice_gathering_state() override;
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const SessionDescriptionInterface* local_description() const override;
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const SessionDescriptionInterface* remote_description() const override;
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const SessionDescriptionInterface* current_local_description() const override;
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const SessionDescriptionInterface* current_remote_description()
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const override;
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const SessionDescriptionInterface* pending_local_description() const override;
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const SessionDescriptionInterface* pending_remote_description()
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const override;
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// JSEP01
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// Deprecated, use version without constraints.
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void CreateOffer(CreateSessionDescriptionObserver* observer,
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const MediaConstraintsInterface* constraints) override;
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void CreateOffer(CreateSessionDescriptionObserver* observer,
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const RTCOfferAnswerOptions& options) override;
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// Deprecated, use version without constraints.
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void CreateAnswer(CreateSessionDescriptionObserver* observer,
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const MediaConstraintsInterface* constraints) override;
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void CreateAnswer(CreateSessionDescriptionObserver* observer,
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const RTCOfferAnswerOptions& options) override;
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void SetLocalDescription(SetSessionDescriptionObserver* observer,
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SessionDescriptionInterface* desc) override;
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void SetRemoteDescription(SetSessionDescriptionObserver* observer,
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SessionDescriptionInterface* desc) override;
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PeerConnectionInterface::RTCConfiguration GetConfiguration() override;
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bool SetConfiguration(
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const PeerConnectionInterface::RTCConfiguration& configuration,
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RTCError* error) override;
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bool SetConfiguration(
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const PeerConnectionInterface::RTCConfiguration& configuration) override {
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return SetConfiguration(configuration, nullptr);
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}
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bool AddIceCandidate(const IceCandidateInterface* candidate) override;
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bool RemoveIceCandidates(
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const std::vector<cricket::Candidate>& candidates) override;
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void RegisterUMAObserver(UMAObserver* observer) override;
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RTCError SetBitrate(const BitrateParameters& bitrate) override;
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bool StartRtcEventLog(rtc::PlatformFile file,
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int64_t max_size_bytes) override;
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void StopRtcEventLog() override;
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void Close() override;
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sigslot::signal1<DataChannel*> SignalDataChannelCreated;
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// Virtual for unit tests.
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virtual const std::vector<rtc::scoped_refptr<DataChannel>>&
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sctp_data_channels() const {
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return sctp_data_channels_;
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}
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// TODO(steveanton): These methods are temporarily added to facilitate work
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// towards merging WebRtcSession into PeerConnection. To make this easier, we
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// want only PeerConnection to interact with WebRtcSession so they can be
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// merged easily. A few outside classes still access WebRtcSession methods
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// directly, so these have been added to PeerConnection to remove the
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// dependency from WebRtcSession.
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rtc::Thread* network_thread() const { return factory_->network_thread(); }
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rtc::Thread* worker_thread() const { return factory_->worker_thread(); }
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rtc::Thread* signaling_thread() const { return factory_->signaling_thread(); }
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virtual const std::string& session_id() const { return session_->id(); }
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virtual bool session_created() const { return session_ != nullptr; }
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virtual bool initial_offerer() const { return session_->initial_offerer(); }
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virtual std::unique_ptr<SessionStats> GetSessionStats_s() {
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return session_->GetStats_s();
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}
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virtual std::unique_ptr<SessionStats> GetSessionStats(
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const ChannelNamePairs& channel_name_pairs) {
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return session_->GetStats(channel_name_pairs);
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}
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virtual bool GetLocalCertificate(
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const std::string& transport_name,
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rtc::scoped_refptr<rtc::RTCCertificate>* certificate) {
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return session_->GetLocalCertificate(transport_name, certificate);
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}
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virtual std::unique_ptr<rtc::SSLCertificate> GetRemoteSSLCertificate(
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const std::string& transport_name) {
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return session_->GetRemoteSSLCertificate(transport_name);
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}
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virtual Call::Stats GetCallStats() { return session_->GetCallStats(); }
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virtual cricket::VoiceChannel* voice_channel() {
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return session_->voice_channel();
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}
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virtual cricket::VideoChannel* video_channel() {
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return session_->video_channel();
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}
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virtual cricket::RtpDataChannel* rtp_data_channel() {
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return session_->rtp_data_channel();
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}
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virtual rtc::Optional<std::string> sctp_content_name() const {
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return session_->sctp_content_name();
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}
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virtual rtc::Optional<std::string> sctp_transport_name() const {
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return session_->sctp_transport_name();
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}
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virtual bool GetLocalTrackIdBySsrc(uint32_t ssrc, std::string* track_id) {
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return session_->GetLocalTrackIdBySsrc(ssrc, track_id);
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}
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virtual bool GetRemoteTrackIdBySsrc(uint32_t ssrc, std::string* track_id) {
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return session_->GetRemoteTrackIdBySsrc(ssrc, track_id);
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}
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// This is needed for stats tests to inject a MockWebRtcSession. Once
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// WebRtcSession has been merged in, this will no longer be needed.
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void set_session_for_testing(WebRtcSession* session) {
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session_ = session;
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}
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protected:
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~PeerConnection() override;
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private:
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struct TrackInfo {
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TrackInfo() : ssrc(0) {}
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TrackInfo(const std::string& stream_label,
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const std::string track_id,
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uint32_t ssrc)
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: stream_label(stream_label), track_id(track_id), ssrc(ssrc) {}
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bool operator==(const TrackInfo& other) {
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return this->stream_label == other.stream_label &&
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this->track_id == other.track_id && this->ssrc == other.ssrc;
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}
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std::string stream_label;
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std::string track_id;
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uint32_t ssrc;
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};
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typedef std::vector<TrackInfo> TrackInfos;
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// Implements MessageHandler.
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void OnMessage(rtc::Message* msg) override;
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void CreateAudioReceiver(MediaStreamInterface* stream,
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const std::string& track_id,
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uint32_t ssrc);
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void CreateVideoReceiver(MediaStreamInterface* stream,
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const std::string& track_id,
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uint32_t ssrc);
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void DestroyReceiver(const std::string& track_id);
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// May be called either by AddStream/RemoveStream, or when a track is
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// added/removed from a stream previously added via AddStream.
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void AddAudioTrack(AudioTrackInterface* track, MediaStreamInterface* stream);
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void RemoveAudioTrack(AudioTrackInterface* track,
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MediaStreamInterface* stream);
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void AddVideoTrack(VideoTrackInterface* track, MediaStreamInterface* stream);
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void RemoveVideoTrack(VideoTrackInterface* track,
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MediaStreamInterface* stream);
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// Implements IceObserver
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void OnIceConnectionStateChange(IceConnectionState new_state) override;
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void OnIceGatheringChange(IceGatheringState new_state) override;
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void OnIceCandidate(
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std::unique_ptr<IceCandidateInterface> candidate) override;
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void OnIceCandidatesRemoved(
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const std::vector<cricket::Candidate>& candidates) override;
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void OnIceConnectionReceivingChange(bool receiving) override;
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// Signals from WebRtcSession.
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void OnSessionStateChange(WebRtcSession* session, WebRtcSession::State state);
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void ChangeSignalingState(SignalingState signaling_state);
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// Signals from MediaStreamObserver.
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void OnAudioTrackAdded(AudioTrackInterface* track,
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MediaStreamInterface* stream);
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void OnAudioTrackRemoved(AudioTrackInterface* track,
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MediaStreamInterface* stream);
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void OnVideoTrackAdded(VideoTrackInterface* track,
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MediaStreamInterface* stream);
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void OnVideoTrackRemoved(VideoTrackInterface* track,
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MediaStreamInterface* stream);
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void PostSetSessionDescriptionFailure(SetSessionDescriptionObserver* observer,
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const std::string& error);
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void PostCreateSessionDescriptionFailure(
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CreateSessionDescriptionObserver* observer,
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const std::string& error);
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bool IsClosed() const {
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return signaling_state_ == PeerConnectionInterface::kClosed;
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}
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// Returns a MediaSessionOptions struct with options decided by |options|,
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// the local MediaStreams and DataChannels.
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void GetOptionsForOffer(
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const PeerConnectionInterface::RTCOfferAnswerOptions& rtc_options,
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cricket::MediaSessionOptions* session_options);
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// Returns a MediaSessionOptions struct with options decided by
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// |constraints|, the local MediaStreams and DataChannels.
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void GetOptionsForAnswer(const RTCOfferAnswerOptions& options,
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cricket::MediaSessionOptions* session_options);
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// Generates MediaDescriptionOptions for the |session_opts| based on existing
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// local description or remote description.
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void GenerateMediaDescriptionOptions(
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const SessionDescriptionInterface* session_desc,
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cricket::RtpTransceiverDirection audio_direction,
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cricket::RtpTransceiverDirection video_direction,
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rtc::Optional<size_t>* audio_index,
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rtc::Optional<size_t>* video_index,
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rtc::Optional<size_t>* data_index,
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cricket::MediaSessionOptions* session_options);
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// Remove all local and remote tracks of type |media_type|.
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// Called when a media type is rejected (m-line set to port 0).
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void RemoveTracks(cricket::MediaType media_type);
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// Makes sure a MediaStreamTrack is created for each StreamParam in |streams|,
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// and existing MediaStreamTracks are removed if there is no corresponding
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// StreamParam. If |default_track_needed| is true, a default MediaStreamTrack
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// is created if it doesn't exist; if false, it's removed if it exists.
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// |media_type| is the type of the |streams| and can be either audio or video.
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// If a new MediaStream is created it is added to |new_streams|.
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void UpdateRemoteStreamsList(
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const std::vector<cricket::StreamParams>& streams,
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bool default_track_needed,
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cricket::MediaType media_type,
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StreamCollection* new_streams);
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// Triggered when a remote track has been seen for the first time in a remote
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// session description. It creates a remote MediaStreamTrackInterface
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// implementation and triggers CreateAudioReceiver or CreateVideoReceiver.
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void OnRemoteTrackSeen(const std::string& stream_label,
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const std::string& track_id,
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uint32_t ssrc,
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cricket::MediaType media_type);
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// Triggered when a remote track has been removed from a remote session
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// description. It removes the remote track with id |track_id| from a remote
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// MediaStream and triggers DestroyAudioReceiver or DestroyVideoReceiver.
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void OnRemoteTrackRemoved(const std::string& stream_label,
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const std::string& track_id,
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cricket::MediaType media_type);
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// Finds remote MediaStreams without any tracks and removes them from
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// |remote_streams_| and notifies the observer that the MediaStreams no longer
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// exist.
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void UpdateEndedRemoteMediaStreams();
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// Loops through the vector of |streams| and finds added and removed
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// StreamParams since last time this method was called.
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// For each new or removed StreamParam, OnLocalTrackSeen or
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// OnLocalTrackRemoved is invoked.
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void UpdateLocalTracks(const std::vector<cricket::StreamParams>& streams,
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cricket::MediaType media_type);
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// Triggered when a local track has been seen for the first time in a local
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// session description.
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// This method triggers CreateAudioSender or CreateVideoSender if the rtp
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// streams in the local SessionDescription can be mapped to a MediaStreamTrack
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// in a MediaStream in |local_streams_|
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void OnLocalTrackSeen(const std::string& stream_label,
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const std::string& track_id,
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uint32_t ssrc,
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cricket::MediaType media_type);
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// Triggered when a local track has been removed from a local session
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// description.
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// This method triggers DestroyAudioSender or DestroyVideoSender if a stream
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// has been removed from the local SessionDescription and the stream can be
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// mapped to a MediaStreamTrack in a MediaStream in |local_streams_|.
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void OnLocalTrackRemoved(const std::string& stream_label,
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const std::string& track_id,
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uint32_t ssrc,
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cricket::MediaType media_type);
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void UpdateLocalRtpDataChannels(const cricket::StreamParamsVec& streams);
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void UpdateRemoteRtpDataChannels(const cricket::StreamParamsVec& streams);
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void UpdateClosingRtpDataChannels(
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const std::vector<std::string>& active_channels,
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bool is_local_update);
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void CreateRemoteRtpDataChannel(const std::string& label,
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uint32_t remote_ssrc);
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// Creates channel and adds it to the collection of DataChannels that will
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// be offered in a SessionDescription.
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rtc::scoped_refptr<DataChannel> InternalCreateDataChannel(
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const std::string& label,
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const InternalDataChannelInit* config);
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// Checks if any data channel has been added.
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bool HasDataChannels() const;
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void AllocateSctpSids(rtc::SSLRole role);
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void OnSctpDataChannelClosed(DataChannel* channel);
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// Notifications from WebRtcSession relating to BaseChannels.
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void OnVoiceChannelCreated();
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void OnVoiceChannelDestroyed();
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void OnVideoChannelCreated();
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void OnVideoChannelDestroyed();
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void OnDataChannelCreated();
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void OnDataChannelDestroyed();
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// Called when the cricket::DataChannel receives a message indicating that a
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// webrtc::DataChannel should be opened.
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void OnDataChannelOpenMessage(const std::string& label,
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const InternalDataChannelInit& config);
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bool HasRtpSender(cricket::MediaType type) const;
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RtpSenderInternal* FindSenderById(const std::string& id);
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std::vector<rtc::scoped_refptr<
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RtpSenderProxyWithInternal<RtpSenderInternal>>>::iterator
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FindSenderForTrack(MediaStreamTrackInterface* track);
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std::vector<rtc::scoped_refptr<
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RtpReceiverProxyWithInternal<RtpReceiverInternal>>>::iterator
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FindReceiverForTrack(const std::string& track_id);
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TrackInfos* GetRemoteTracks(cricket::MediaType media_type);
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TrackInfos* GetLocalTracks(cricket::MediaType media_type);
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const TrackInfo* FindTrackInfo(const TrackInfos& infos,
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const std::string& stream_label,
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const std::string track_id) const;
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// Returns the specified SCTP DataChannel in sctp_data_channels_,
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// or nullptr if not found.
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DataChannel* FindDataChannelBySid(int sid) const;
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// Called when first configuring the port allocator.
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bool InitializePortAllocator_n(const RTCConfiguration& configuration);
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// Called when SetConfiguration is called to apply the supported subset
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// of the configuration on the network thread.
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bool ReconfigurePortAllocator_n(
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const cricket::ServerAddresses& stun_servers,
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const std::vector<cricket::RelayServerConfig>& turn_servers,
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IceTransportsType type,
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int candidate_pool_size,
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bool prune_turn_ports);
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// Starts recording an RTC event log using the supplied platform file.
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// This function should only be called from the worker thread.
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bool StartRtcEventLog_w(rtc::PlatformFile file, int64_t max_size_bytes);
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// Stops recording an RTC event log.
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// This function should only be called from the worker thread.
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void StopRtcEventLog_w();
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// Ensures the configuration doesn't have any parameters with invalid values,
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// or values that conflict with other parameters.
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//
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// Returns RTCError::OK() if there are no issues.
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RTCError ValidateConfiguration(const RTCConfiguration& config) const;
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// Storing the factory as a scoped reference pointer ensures that the memory
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// in the PeerConnectionFactoryImpl remains available as long as the
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// PeerConnection is running. It is passed to PeerConnection as a raw pointer.
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// However, since the reference counting is done in the
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// PeerConnectionFactoryInterface all instances created using the raw pointer
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// will refer to the same reference count.
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rtc::scoped_refptr<PeerConnectionFactory> factory_;
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PeerConnectionObserver* observer_;
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UMAObserver* uma_observer_;
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// The EventLog needs to outlive |call_| (and any other object that uses it).
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std::unique_ptr<RtcEventLog> event_log_;
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SignalingState signaling_state_;
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IceConnectionState ice_connection_state_;
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IceGatheringState ice_gathering_state_;
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PeerConnectionInterface::RTCConfiguration configuration_;
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std::unique_ptr<cricket::PortAllocator> port_allocator_;
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// One PeerConnection has only one RTCP CNAME.
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// https://tools.ietf.org/html/draft-ietf-rtcweb-rtp-usage-26#section-4.9
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std::string rtcp_cname_;
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// Streams added via AddStream.
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rtc::scoped_refptr<StreamCollection> local_streams_;
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// Streams created as a result of SetRemoteDescription.
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rtc::scoped_refptr<StreamCollection> remote_streams_;
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std::vector<std::unique_ptr<MediaStreamObserver>> stream_observers_;
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// These lists store track info seen in local/remote descriptions.
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TrackInfos remote_audio_tracks_;
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TrackInfos remote_video_tracks_;
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TrackInfos local_audio_tracks_;
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TrackInfos local_video_tracks_;
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SctpSidAllocator sid_allocator_;
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// label -> DataChannel
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std::map<std::string, rtc::scoped_refptr<DataChannel>> rtp_data_channels_;
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std::vector<rtc::scoped_refptr<DataChannel>> sctp_data_channels_;
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std::vector<rtc::scoped_refptr<DataChannel>> sctp_data_channels_to_free_;
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bool remote_peer_supports_msid_ = false;
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std::unique_ptr<Call> call_;
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WebRtcSession* session_;
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std::unique_ptr<WebRtcSession> owned_session_;
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std::unique_ptr<StatsCollector> stats_; // A pointer is passed to senders_
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rtc::scoped_refptr<RTCStatsCollector> stats_collector_;
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std::vector<rtc::scoped_refptr<RtpSenderProxyWithInternal<RtpSenderInternal>>>
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senders_;
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std::vector<
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rtc::scoped_refptr<RtpReceiverProxyWithInternal<RtpReceiverInternal>>>
|
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receivers_;
|
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};
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} // namespace webrtc
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#endif // PC_PEERCONNECTION_H_
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