| /* |
| * Copyright 2012 The WebRTC project authors. All Rights Reserved. |
| * |
| * Use of this source code is governed by a BSD-style license |
| * that can be found in the LICENSE file in the root of the source |
| * tree. An additional intellectual property rights grant can be found |
| * in the file PATENTS. All contributing project authors may |
| * be found in the AUTHORS file in the root of the source tree. |
| */ |
| |
| #ifndef PC_PEER_CONNECTION_H_ |
| #define PC_PEER_CONNECTION_H_ |
| |
| #include <map> |
| #include <memory> |
| #include <set> |
| #include <string> |
| #include <vector> |
| |
| #include "api/media_transport_interface.h" |
| #include "api/peer_connection_interface.h" |
| #include "api/turn_customizer.h" |
| #include "pc/ice_server_parsing.h" |
| #include "pc/jsep_transport_controller.h" |
| #include "pc/peer_connection_factory.h" |
| #include "pc/peer_connection_internal.h" |
| #include "pc/rtc_stats_collector.h" |
| #include "pc/rtp_transceiver.h" |
| #include "pc/sctp_transport.h" |
| #include "pc/stats_collector.h" |
| #include "pc/stream_collection.h" |
| #include "pc/webrtc_session_description_factory.h" |
| #include "rtc_base/race_checker.h" |
| #include "rtc_base/unique_id_generator.h" |
| |
| namespace webrtc { |
| |
| class MediaStreamObserver; |
| class VideoRtpReceiver; |
| class RtcEventLog; |
| |
| // PeerConnection is the implementation of the PeerConnection object as defined |
| // by the PeerConnectionInterface API surface. |
| // The class currently is solely responsible for the following: |
| // - Managing the session state machine (signaling state). |
| // - Creating and initializing lower-level objects, like PortAllocator and |
| // BaseChannels. |
| // - Owning and managing the life cycle of the RtpSender/RtpReceiver and track |
| // objects. |
| // - Tracking the current and pending local/remote session descriptions. |
| // The class currently is jointly responsible for the following: |
| // - Parsing and interpreting SDP. |
| // - Generating offers and answers based on the current state. |
| // - The ICE state machine. |
| // - Generating stats. |
| class PeerConnection : public PeerConnectionInternal, |
| public DataChannelProviderInterface, |
| public DataChannelSink, |
| public JsepTransportController::Observer, |
| public rtc::MessageHandler, |
| public sigslot::has_slots<> { |
| public: |
| enum class UsageEvent : int { |
| TURN_SERVER_ADDED = 0x01, |
| STUN_SERVER_ADDED = 0x02, |
| DATA_ADDED = 0x04, |
| AUDIO_ADDED = 0x08, |
| VIDEO_ADDED = 0x10, |
| SET_LOCAL_DESCRIPTION_CALLED = 0x20, |
| SET_REMOTE_DESCRIPTION_CALLED = 0x40, |
| CANDIDATE_COLLECTED = 0x80, |
| REMOTE_CANDIDATE_ADDED = 0x100, |
| ICE_STATE_CONNECTED = 0x200, |
| CLOSE_CALLED = 0x400, |
| PRIVATE_CANDIDATE_COLLECTED = 0x800, |
| MAX_VALUE = 0x1000, |
| }; |
| |
| explicit PeerConnection(PeerConnectionFactory* factory, |
| std::unique_ptr<RtcEventLog> event_log, |
| std::unique_ptr<Call> call); |
| |
| bool Initialize( |
| const PeerConnectionInterface::RTCConfiguration& configuration, |
| PeerConnectionDependencies dependencies); |
| |
| rtc::scoped_refptr<StreamCollectionInterface> local_streams() override; |
| rtc::scoped_refptr<StreamCollectionInterface> remote_streams() override; |
| bool AddStream(MediaStreamInterface* local_stream) override; |
| void RemoveStream(MediaStreamInterface* local_stream) override; |
| |
| RTCErrorOr<rtc::scoped_refptr<RtpSenderInterface>> AddTrack( |
| rtc::scoped_refptr<MediaStreamTrackInterface> track, |
| const std::vector<std::string>& stream_ids) override; |
| bool RemoveTrack(RtpSenderInterface* sender) override; |
| RTCError RemoveTrackNew( |
| rtc::scoped_refptr<RtpSenderInterface> sender) override; |
| |
| RTCErrorOr<rtc::scoped_refptr<RtpTransceiverInterface>> AddTransceiver( |
| rtc::scoped_refptr<MediaStreamTrackInterface> track) override; |
| RTCErrorOr<rtc::scoped_refptr<RtpTransceiverInterface>> AddTransceiver( |
| rtc::scoped_refptr<MediaStreamTrackInterface> track, |
| const RtpTransceiverInit& init) override; |
| RTCErrorOr<rtc::scoped_refptr<RtpTransceiverInterface>> AddTransceiver( |
| cricket::MediaType media_type) override; |
| RTCErrorOr<rtc::scoped_refptr<RtpTransceiverInterface>> AddTransceiver( |
| cricket::MediaType media_type, |
| const RtpTransceiverInit& init) override; |
| |
| // Gets the DTLS SSL certificate associated with the audio transport on the |
| // remote side. This will become populated once the DTLS connection with the |
| // peer has been completed, as indicated by the ICE connection state |
| // transitioning to kIceConnectionCompleted. |
| // Note that this will be removed once we implement RTCDtlsTransport which |
| // has standardized method for getting this information. |
| // See https://www.w3.org/TR/webrtc/#rtcdtlstransport-interface |
| std::unique_ptr<rtc::SSLCertificate> GetRemoteAudioSSLCertificate(); |
| |
| // Version of the above method that returns the full certificate chain. |
| std::unique_ptr<rtc::SSLCertChain> GetRemoteAudioSSLCertChain(); |
| |
| rtc::scoped_refptr<RtpSenderInterface> CreateSender( |
| const std::string& kind, |
| const std::string& stream_id) override; |
| |
| std::vector<rtc::scoped_refptr<RtpSenderInterface>> GetSenders() |
| const override; |
| std::vector<rtc::scoped_refptr<RtpReceiverInterface>> GetReceivers() |
| const override; |
| std::vector<rtc::scoped_refptr<RtpTransceiverInterface>> GetTransceivers() |
| const override; |
| |
| rtc::scoped_refptr<DataChannelInterface> CreateDataChannel( |
| const std::string& label, |
| const DataChannelInit* config) override; |
| // WARNING: LEGACY. See peerconnectioninterface.h |
| bool GetStats(StatsObserver* observer, |
| webrtc::MediaStreamTrackInterface* track, |
| StatsOutputLevel level) override; |
| // Spec-complaint GetStats(). See peerconnectioninterface.h |
| void GetStats(RTCStatsCollectorCallback* callback) override; |
| void GetStats( |
| rtc::scoped_refptr<RtpSenderInterface> selector, |
| rtc::scoped_refptr<RTCStatsCollectorCallback> callback) override; |
| void GetStats( |
| rtc::scoped_refptr<RtpReceiverInterface> selector, |
| rtc::scoped_refptr<RTCStatsCollectorCallback> callback) override; |
| void ClearStatsCache() override; |
| |
| SignalingState signaling_state() override; |
| |
| IceConnectionState ice_connection_state() override; |
| IceConnectionState standardized_ice_connection_state() override; |
| PeerConnectionState peer_connection_state() override; |
| IceGatheringState ice_gathering_state() override; |
| |
| const SessionDescriptionInterface* local_description() const override; |
| const SessionDescriptionInterface* remote_description() const override; |
| const SessionDescriptionInterface* current_local_description() const override; |
| const SessionDescriptionInterface* current_remote_description() |
| const override; |
| const SessionDescriptionInterface* pending_local_description() const override; |
| const SessionDescriptionInterface* pending_remote_description() |
| const override; |
| |
| // JSEP01 |
| void CreateOffer(CreateSessionDescriptionObserver* observer, |
| const RTCOfferAnswerOptions& options) override; |
| void CreateAnswer(CreateSessionDescriptionObserver* observer, |
| const RTCOfferAnswerOptions& options) override; |
| void SetLocalDescription(SetSessionDescriptionObserver* observer, |
| SessionDescriptionInterface* desc) override; |
| void SetRemoteDescription(SetSessionDescriptionObserver* observer, |
| SessionDescriptionInterface* desc) override; |
| void SetRemoteDescription( |
| std::unique_ptr<SessionDescriptionInterface> desc, |
| rtc::scoped_refptr<SetRemoteDescriptionObserverInterface> observer) |
| override; |
| PeerConnectionInterface::RTCConfiguration GetConfiguration() override; |
| bool SetConfiguration( |
| const PeerConnectionInterface::RTCConfiguration& configuration, |
| RTCError* error) override; |
| bool SetConfiguration( |
| const PeerConnectionInterface::RTCConfiguration& configuration) override { |
| return SetConfiguration(configuration, nullptr); |
| } |
| bool AddIceCandidate(const IceCandidateInterface* candidate) override; |
| bool RemoveIceCandidates( |
| const std::vector<cricket::Candidate>& candidates) override; |
| |
| RTCError SetBitrate(const BitrateSettings& bitrate) override; |
| |
| void SetBitrateAllocationStrategy( |
| std::unique_ptr<rtc::BitrateAllocationStrategy> |
| bitrate_allocation_strategy) override; |
| |
| void SetAudioPlayout(bool playout) override; |
| void SetAudioRecording(bool recording) override; |
| |
| rtc::scoped_refptr<DtlsTransportInterface> LookupDtlsTransportByMid( |
| const std::string& mid) override; |
| rtc::scoped_refptr<DtlsTransport> LookupDtlsTransportByMidInternal( |
| const std::string& mid); |
| |
| rtc::scoped_refptr<SctpTransportInterface> GetSctpTransport() const override; |
| |
| RTC_DEPRECATED bool StartRtcEventLog(rtc::PlatformFile file, |
| int64_t max_size_bytes) override; |
| bool StartRtcEventLog(std::unique_ptr<RtcEventLogOutput> output, |
| int64_t output_period_ms) override; |
| void StopRtcEventLog() override; |
| |
| void Close() override; |
| |
| // PeerConnectionInternal implementation. |
| rtc::Thread* network_thread() const final { |
| return factory_->network_thread(); |
| } |
| rtc::Thread* worker_thread() const final { return factory_->worker_thread(); } |
| rtc::Thread* signaling_thread() const final { |
| return factory_->signaling_thread(); |
| } |
| |
| std::string session_id() const override { |
| RTC_DCHECK_RUN_ON(signaling_thread()); |
| return session_id_; |
| } |
| |
| bool initial_offerer() const override { |
| RTC_DCHECK_RUN_ON(signaling_thread()); |
| return transport_controller_ && transport_controller_->initial_offerer(); |
| } |
| |
| std::vector< |
| rtc::scoped_refptr<RtpTransceiverProxyWithInternal<RtpTransceiver>>> |
| GetTransceiversInternal() const override { |
| RTC_DCHECK_RUN_ON(signaling_thread()); |
| return transceivers_; |
| } |
| |
| sigslot::signal1<DataChannel*>& SignalDataChannelCreated() override { |
| return SignalDataChannelCreated_; |
| } |
| |
| cricket::RtpDataChannel* rtp_data_channel() const override { |
| return rtp_data_channel_; |
| } |
| |
| std::vector<rtc::scoped_refptr<DataChannel>> sctp_data_channels() |
| const override { |
| RTC_DCHECK_RUN_ON(signaling_thread()); |
| return sctp_data_channels_; |
| } |
| |
| absl::optional<std::string> sctp_content_name() const override { |
| RTC_DCHECK_RUN_ON(signaling_thread()); |
| return sctp_mid_; |
| } |
| |
| absl::optional<std::string> sctp_transport_name() const override; |
| |
| cricket::CandidateStatsList GetPooledCandidateStats() const override; |
| std::map<std::string, std::string> GetTransportNamesByMid() const override; |
| std::map<std::string, cricket::TransportStats> GetTransportStatsByNames( |
| const std::set<std::string>& transport_names) override; |
| Call::Stats GetCallStats() override; |
| |
| bool GetLocalCertificate( |
| const std::string& transport_name, |
| rtc::scoped_refptr<rtc::RTCCertificate>* certificate) override; |
| std::unique_ptr<rtc::SSLCertChain> GetRemoteSSLCertChain( |
| const std::string& transport_name) override; |
| bool IceRestartPending(const std::string& content_name) const override; |
| bool NeedsIceRestart(const std::string& content_name) const override; |
| bool GetSslRole(const std::string& content_name, rtc::SSLRole* role) override; |
| |
| void ReturnHistogramVeryQuicklyForTesting() { |
| return_histogram_very_quickly_ = true; |
| } |
| void RequestUsagePatternReportForTesting(); |
| |
| protected: |
| ~PeerConnection() override; |
| |
| private: |
| class SetRemoteDescriptionObserverAdapter; |
| friend class SetRemoteDescriptionObserverAdapter; |
| |
| struct RtpSenderInfo { |
| RtpSenderInfo() : first_ssrc(0) {} |
| RtpSenderInfo(const std::string& stream_id, |
| const std::string sender_id, |
| uint32_t ssrc) |
| : stream_id(stream_id), sender_id(sender_id), first_ssrc(ssrc) {} |
| bool operator==(const RtpSenderInfo& other) { |
| return this->stream_id == other.stream_id && |
| this->sender_id == other.sender_id && |
| this->first_ssrc == other.first_ssrc; |
| } |
| std::string stream_id; |
| std::string sender_id; |
| // An RtpSender can have many SSRCs. The first one is used as a sort of ID |
| // for communicating with the lower layers. |
| uint32_t first_ssrc; |
| }; |
| |
| // Implements MessageHandler. |
| void OnMessage(rtc::Message* msg) override; |
| |
| // Plan B helpers for getting the voice/video media channels for the single |
| // audio/video transceiver, if it exists. |
| cricket::VoiceMediaChannel* voice_media_channel() const |
| RTC_RUN_ON(signaling_thread()); |
| cricket::VideoMediaChannel* video_media_channel() const |
| RTC_RUN_ON(signaling_thread()); |
| |
| std::vector<rtc::scoped_refptr<RtpSenderProxyWithInternal<RtpSenderInternal>>> |
| GetSendersInternal() const RTC_RUN_ON(signaling_thread()); |
| std::vector< |
| rtc::scoped_refptr<RtpReceiverProxyWithInternal<RtpReceiverInternal>>> |
| GetReceiversInternal() const RTC_RUN_ON(signaling_thread()); |
| |
| rtc::scoped_refptr<RtpTransceiverProxyWithInternal<RtpTransceiver>> |
| GetAudioTransceiver() const RTC_RUN_ON(signaling_thread()); |
| rtc::scoped_refptr<RtpTransceiverProxyWithInternal<RtpTransceiver>> |
| GetVideoTransceiver() const RTC_RUN_ON(signaling_thread()); |
| |
| rtc::scoped_refptr<RtpTransceiverProxyWithInternal<RtpTransceiver>> |
| GetFirstAudioTransceiver() const RTC_RUN_ON(signaling_thread()); |
| |
| void CreateAudioReceiver(MediaStreamInterface* stream, |
| const RtpSenderInfo& remote_sender_info) |
| RTC_RUN_ON(signaling_thread()); |
| |
| void CreateVideoReceiver(MediaStreamInterface* stream, |
| const RtpSenderInfo& remote_sender_info) |
| RTC_RUN_ON(signaling_thread()); |
| rtc::scoped_refptr<RtpReceiverInterface> RemoveAndStopReceiver( |
| const RtpSenderInfo& remote_sender_info) RTC_RUN_ON(signaling_thread()); |
| |
| // May be called either by AddStream/RemoveStream, or when a track is |
| // added/removed from a stream previously added via AddStream. |
| void AddAudioTrack(AudioTrackInterface* track, MediaStreamInterface* stream) |
| RTC_RUN_ON(signaling_thread()); |
| void RemoveAudioTrack(AudioTrackInterface* track, |
| MediaStreamInterface* stream) |
| RTC_RUN_ON(signaling_thread()); |
| void AddVideoTrack(VideoTrackInterface* track, MediaStreamInterface* stream) |
| RTC_RUN_ON(signaling_thread()); |
| void RemoveVideoTrack(VideoTrackInterface* track, |
| MediaStreamInterface* stream) |
| RTC_RUN_ON(signaling_thread()); |
| |
| // AddTrack implementation when Unified Plan is specified. |
| RTCErrorOr<rtc::scoped_refptr<RtpSenderInterface>> AddTrackUnifiedPlan( |
| rtc::scoped_refptr<MediaStreamTrackInterface> track, |
| const std::vector<std::string>& stream_ids) |
| RTC_RUN_ON(signaling_thread()); |
| // AddTrack implementation when Plan B is specified. |
| RTCErrorOr<rtc::scoped_refptr<RtpSenderInterface>> AddTrackPlanB( |
| rtc::scoped_refptr<MediaStreamTrackInterface> track, |
| const std::vector<std::string>& stream_ids) |
| RTC_RUN_ON(signaling_thread()); |
| |
| // Returns the first RtpTransceiver suitable for a newly added track, if such |
| // transceiver is available. |
| rtc::scoped_refptr<RtpTransceiverProxyWithInternal<RtpTransceiver>> |
| FindFirstTransceiverForAddedTrack( |
| rtc::scoped_refptr<MediaStreamTrackInterface> track) |
| RTC_RUN_ON(signaling_thread()); |
| |
| rtc::scoped_refptr<RtpTransceiverProxyWithInternal<RtpTransceiver>> |
| FindTransceiverBySender(rtc::scoped_refptr<RtpSenderInterface> sender) |
| RTC_RUN_ON(signaling_thread()); |
| |
| // Internal implementation for AddTransceiver family of methods. If |
| // |fire_callback| is set, fires OnRenegotiationNeeded callback if successful. |
| RTCErrorOr<rtc::scoped_refptr<RtpTransceiverInterface>> AddTransceiver( |
| cricket::MediaType media_type, |
| rtc::scoped_refptr<MediaStreamTrackInterface> track, |
| const RtpTransceiverInit& init, |
| bool fire_callback = true) RTC_RUN_ON(signaling_thread()); |
| |
| rtc::scoped_refptr<RtpSenderProxyWithInternal<RtpSenderInternal>> |
| 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::scoped_refptr<RtpReceiverProxyWithInternal<RtpReceiverInternal>> |
| CreateReceiver(cricket::MediaType media_type, const std::string& receiver_id); |
| |
| // Create a new RtpTransceiver of the given type and add it to the list of |
| // transceivers. |
| rtc::scoped_refptr<RtpTransceiverProxyWithInternal<RtpTransceiver>> |
| CreateAndAddTransceiver( |
| rtc::scoped_refptr<RtpSenderProxyWithInternal<RtpSenderInternal>> sender, |
| rtc::scoped_refptr<RtpReceiverProxyWithInternal<RtpReceiverInternal>> |
| receiver) RTC_RUN_ON(signaling_thread()); |
| |
| void SetIceConnectionState(IceConnectionState new_state) |
| RTC_RUN_ON(signaling_thread()); |
| void SetStandardizedIceConnectionState( |
| PeerConnectionInterface::IceConnectionState new_state) |
| RTC_RUN_ON(signaling_thread()); |
| void SetConnectionState( |
| PeerConnectionInterface::PeerConnectionState new_state) |
| RTC_RUN_ON(signaling_thread()); |
| |
| // Called any time the IceGatheringState changes |
| void OnIceGatheringChange(IceGatheringState new_state) |
| RTC_RUN_ON(signaling_thread()); |
| // New ICE candidate has been gathered. |
| void OnIceCandidate(std::unique_ptr<IceCandidateInterface> candidate) |
| RTC_RUN_ON(signaling_thread()); |
| // Some local ICE candidates have been removed. |
| void OnIceCandidatesRemoved(const std::vector<cricket::Candidate>& candidates) |
| RTC_RUN_ON(signaling_thread()); |
| |
| // Update the state, signaling if necessary. |
| void ChangeSignalingState(SignalingState signaling_state) |
| RTC_RUN_ON(signaling_thread()); |
| |
| // Signals from MediaStreamObserver. |
| void OnAudioTrackAdded(AudioTrackInterface* track, |
| MediaStreamInterface* stream) |
| RTC_RUN_ON(signaling_thread()); |
| void OnAudioTrackRemoved(AudioTrackInterface* track, |
| MediaStreamInterface* stream) |
| RTC_RUN_ON(signaling_thread()); |
| void OnVideoTrackAdded(VideoTrackInterface* track, |
| MediaStreamInterface* stream) |
| RTC_RUN_ON(signaling_thread()); |
| void OnVideoTrackRemoved(VideoTrackInterface* track, |
| MediaStreamInterface* stream) |
| RTC_RUN_ON(signaling_thread()); |
| |
| void PostSetSessionDescriptionSuccess( |
| SetSessionDescriptionObserver* observer); |
| void PostSetSessionDescriptionFailure(SetSessionDescriptionObserver* observer, |
| RTCError&& error); |
| void PostCreateSessionDescriptionFailure( |
| CreateSessionDescriptionObserver* observer, |
| RTCError error); |
| |
| // Synchronous implementations of SetLocalDescription/SetRemoteDescription |
| // that return an RTCError instead of invoking a callback. |
| RTCError ApplyLocalDescription( |
| std::unique_ptr<SessionDescriptionInterface> desc); |
| RTCError ApplyRemoteDescription( |
| std::unique_ptr<SessionDescriptionInterface> desc); |
| |
| // Updates the local RtpTransceivers according to the JSEP rules. Called as |
| // part of setting the local/remote description. |
| RTCError UpdateTransceiversAndDataChannels( |
| cricket::ContentSource source, |
| const SessionDescriptionInterface& new_session, |
| const SessionDescriptionInterface* old_local_description, |
| const SessionDescriptionInterface* old_remote_description) |
| RTC_RUN_ON(signaling_thread()); |
| |
| // Either creates or destroys the transceiver's BaseChannel according to the |
| // given media section. |
| RTCError UpdateTransceiverChannel( |
| rtc::scoped_refptr<RtpTransceiverProxyWithInternal<RtpTransceiver>> |
| transceiver, |
| const cricket::ContentInfo& content, |
| const cricket::ContentGroup* bundle_group) RTC_RUN_ON(signaling_thread()); |
| |
| // Either creates or destroys the local data channel according to the given |
| // media section. |
| RTCError UpdateDataChannel(cricket::ContentSource source, |
| const cricket::ContentInfo& content, |
| const cricket::ContentGroup* bundle_group) |
| RTC_RUN_ON(signaling_thread()); |
| |
| // Associate the given transceiver according to the JSEP rules. |
| RTCErrorOr< |
| rtc::scoped_refptr<RtpTransceiverProxyWithInternal<RtpTransceiver>>> |
| AssociateTransceiver(cricket::ContentSource source, |
| SdpType type, |
| size_t mline_index, |
| const cricket::ContentInfo& content, |
| const cricket::ContentInfo* old_local_content, |
| const cricket::ContentInfo* old_remote_content) |
| RTC_RUN_ON(signaling_thread()); |
| |
| // Returns the RtpTransceiver, if found, that is associated to the given MID. |
| rtc::scoped_refptr<RtpTransceiverProxyWithInternal<RtpTransceiver>> |
| GetAssociatedTransceiver(const std::string& mid) const |
| RTC_RUN_ON(signaling_thread()); |
| |
| // Returns the RtpTransceiver, if found, that was assigned to the given mline |
| // index in CreateOffer. |
| rtc::scoped_refptr<RtpTransceiverProxyWithInternal<RtpTransceiver>> |
| GetTransceiverByMLineIndex(size_t mline_index) const |
| RTC_RUN_ON(signaling_thread()); |
| |
| // Returns an RtpTransciever, if available, that can be used to receive the |
| // given media type according to JSEP rules. |
| rtc::scoped_refptr<RtpTransceiverProxyWithInternal<RtpTransceiver>> |
| FindAvailableTransceiverToReceive(cricket::MediaType media_type) const |
| RTC_RUN_ON(signaling_thread()); |
| |
| // Returns the media section in the given session description that is |
| // associated with the RtpTransceiver. Returns null if none found or this |
| // RtpTransceiver is not associated. Logic varies depending on the |
| // SdpSemantics specified in the configuration. |
| const cricket::ContentInfo* FindMediaSectionForTransceiver( |
| rtc::scoped_refptr<RtpTransceiverProxyWithInternal<RtpTransceiver>> |
| transceiver, |
| const SessionDescriptionInterface* sdesc) const |
| RTC_RUN_ON(signaling_thread()); |
| |
| // Runs the algorithm **set the associated remote streams** specified in |
| // https://w3c.github.io/webrtc-pc/#set-associated-remote-streams. |
| void SetAssociatedRemoteStreams( |
| rtc::scoped_refptr<RtpReceiverInternal> receiver, |
| const std::vector<std::string>& stream_ids, |
| std::vector<rtc::scoped_refptr<MediaStreamInterface>>* added_streams, |
| std::vector<rtc::scoped_refptr<MediaStreamInterface>>* removed_streams) |
| RTC_RUN_ON(signaling_thread()); |
| |
| // Runs the algorithm **process the removal of a remote track** specified in |
| // the WebRTC specification. |
| // This method will update the following lists: |
| // |remove_list| is the list of transceivers for which the receiving track is |
| // being removed. |
| // |removed_streams| is the list of streams which no longer have a receiving |
| // track so should be removed. |
| // https://w3c.github.io/webrtc-pc/#process-remote-track-removal |
| void ProcessRemovalOfRemoteTrack( |
| rtc::scoped_refptr<RtpTransceiverProxyWithInternal<RtpTransceiver>> |
| transceiver, |
| std::vector<rtc::scoped_refptr<RtpTransceiverInterface>>* remove_list, |
| std::vector<rtc::scoped_refptr<MediaStreamInterface>>* removed_streams) |
| RTC_RUN_ON(signaling_thread()); |
| |
| void RemoveRemoteStreamsIfEmpty( |
| const std::vector<rtc::scoped_refptr<MediaStreamInterface>>& |
| remote_streams, |
| std::vector<rtc::scoped_refptr<MediaStreamInterface>>* removed_streams) |
| RTC_RUN_ON(signaling_thread()); |
| |
| void OnNegotiationNeeded(); |
| |
| bool IsClosed() const { |
| RTC_DCHECK_RUN_ON(signaling_thread()); |
| return signaling_state_ == PeerConnectionInterface::kClosed; |
| } |
| |
| // Returns a MediaSessionOptions struct with options decided by |options|, |
| // the local MediaStreams and DataChannels. |
| void GetOptionsForOffer(const PeerConnectionInterface::RTCOfferAnswerOptions& |
| offer_answer_options, |
| cricket::MediaSessionOptions* session_options) |
| RTC_RUN_ON(signaling_thread()); |
| void GetOptionsForPlanBOffer( |
| const PeerConnectionInterface::RTCOfferAnswerOptions& |
| offer_answer_options, |
| cricket::MediaSessionOptions* session_options) |
| RTC_RUN_ON(signaling_thread()); |
| void GetOptionsForUnifiedPlanOffer( |
| const PeerConnectionInterface::RTCOfferAnswerOptions& |
| offer_answer_options, |
| cricket::MediaSessionOptions* session_options) |
| RTC_RUN_ON(signaling_thread()); |
| |
| RTCError HandleLegacyOfferOptions(const RTCOfferAnswerOptions& options) |
| RTC_RUN_ON(signaling_thread()); |
| void RemoveRecvDirectionFromReceivingTransceiversOfType( |
| cricket::MediaType media_type) RTC_RUN_ON(signaling_thread()); |
| void AddUpToOneReceivingTransceiverOfType(cricket::MediaType media_type); |
| std::vector< |
| rtc::scoped_refptr<RtpTransceiverProxyWithInternal<RtpTransceiver>>> |
| GetReceivingTransceiversOfType(cricket::MediaType media_type) |
| RTC_RUN_ON(signaling_thread()); |
| |
| // Returns a MediaSessionOptions struct with options decided by |
| // |constraints|, the local MediaStreams and DataChannels. |
| void GetOptionsForAnswer(const RTCOfferAnswerOptions& offer_answer_options, |
| cricket::MediaSessionOptions* session_options) |
| RTC_RUN_ON(signaling_thread()); |
| void GetOptionsForPlanBAnswer( |
| const PeerConnectionInterface::RTCOfferAnswerOptions& |
| offer_answer_options, |
| cricket::MediaSessionOptions* session_options) |
| RTC_RUN_ON(signaling_thread()); |
| void GetOptionsForUnifiedPlanAnswer( |
| const PeerConnectionInterface::RTCOfferAnswerOptions& |
| offer_answer_options, |
| cricket::MediaSessionOptions* session_options) |
| RTC_RUN_ON(signaling_thread()); |
| |
| // Generates MediaDescriptionOptions for the |session_opts| based on existing |
| // local description or remote description. |
| void 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) |
| RTC_RUN_ON(signaling_thread()); |
| |
| // Generates the active MediaDescriptionOptions for the local data channel |
| // given the specified MID. |
| cricket::MediaDescriptionOptions GetMediaDescriptionOptionsForActiveData( |
| const std::string& mid) const RTC_RUN_ON(signaling_thread()); |
| |
| // Generates the rejected MediaDescriptionOptions for the local data channel |
| // given the specified MID. |
| cricket::MediaDescriptionOptions GetMediaDescriptionOptionsForRejectedData( |
| const std::string& mid) const RTC_RUN_ON(signaling_thread()); |
| |
| // Returns the MID for the data section associated with either the |
| // RtpDataChannel or SCTP data channel, if it has been set. If no data |
| // channels are configured this will return nullopt. |
| absl::optional<std::string> GetDataMid() const RTC_RUN_ON(signaling_thread()); |
| |
| // Remove all local and remote senders of type |media_type|. |
| // Called when a media type is rejected (m-line set to port 0). |
| void RemoveSenders(cricket::MediaType media_type) |
| RTC_RUN_ON(signaling_thread()); |
| |
| // Makes sure a MediaStreamTrack is created for each StreamParam in |streams|, |
| // and existing MediaStreamTracks are removed if there is no corresponding |
| // StreamParam. If |default_track_needed| is true, a default MediaStreamTrack |
| // is created if it doesn't exist; if false, it's removed if it exists. |
| // |media_type| is the type of the |streams| and can be either audio or video. |
| // If a new MediaStream is created it is added to |new_streams|. |
| void UpdateRemoteSendersList( |
| const std::vector<cricket::StreamParams>& streams, |
| bool default_track_needed, |
| cricket::MediaType media_type, |
| StreamCollection* new_streams) RTC_RUN_ON(signaling_thread()); |
| |
| // Triggered when a remote sender has been seen for the first time in a remote |
| // session description. It creates a remote MediaStreamTrackInterface |
| // implementation and triggers CreateAudioReceiver or CreateVideoReceiver. |
| void OnRemoteSenderAdded(const RtpSenderInfo& sender_info, |
| cricket::MediaType media_type) |
| RTC_RUN_ON(signaling_thread()); |
| |
| // Triggered when a remote sender has been removed from a remote session |
| // description. It removes the remote sender with id |sender_id| from a remote |
| // MediaStream and triggers DestroyAudioReceiver or DestroyVideoReceiver. |
| void OnRemoteSenderRemoved(const RtpSenderInfo& sender_info, |
| cricket::MediaType media_type) |
| RTC_RUN_ON(signaling_thread()); |
| |
| // Finds remote MediaStreams without any tracks and removes them from |
| // |remote_streams_| and notifies the observer that the MediaStreams no longer |
| // exist. |
| void UpdateEndedRemoteMediaStreams() RTC_RUN_ON(signaling_thread()); |
| |
| // Loops through the vector of |streams| and finds added and removed |
| // StreamParams since last time this method was called. |
| // For each new or removed StreamParam, OnLocalSenderSeen or |
| // OnLocalSenderRemoved is invoked. |
| void UpdateLocalSenders(const std::vector<cricket::StreamParams>& streams, |
| cricket::MediaType media_type) |
| RTC_RUN_ON(signaling_thread()); |
| |
| // Triggered when a local sender has been seen for the first time in a local |
| // session description. |
| // This method triggers CreateAudioSender or CreateVideoSender if the rtp |
| // streams in the local SessionDescription can be mapped to a MediaStreamTrack |
| // in a MediaStream in |local_streams_| |
| void OnLocalSenderAdded(const RtpSenderInfo& sender_info, |
| cricket::MediaType media_type) |
| RTC_RUN_ON(signaling_thread()); |
| |
| // Triggered when a local sender has been removed from a local session |
| // description. |
| // This method triggers DestroyAudioSender or DestroyVideoSender if a stream |
| // has been removed from the local SessionDescription and the stream can be |
| // mapped to a MediaStreamTrack in a MediaStream in |local_streams_|. |
| void OnLocalSenderRemoved(const RtpSenderInfo& sender_info, |
| cricket::MediaType media_type) |
| RTC_RUN_ON(signaling_thread()); |
| |
| void UpdateLocalRtpDataChannels(const cricket::StreamParamsVec& streams) |
| RTC_RUN_ON(signaling_thread()); |
| void UpdateRemoteRtpDataChannels(const cricket::StreamParamsVec& streams) |
| RTC_RUN_ON(signaling_thread()); |
| void UpdateClosingRtpDataChannels( |
| const std::vector<std::string>& active_channels, |
| bool is_local_update) RTC_RUN_ON(signaling_thread()); |
| void CreateRemoteRtpDataChannel(const std::string& label, |
| uint32_t remote_ssrc) |
| RTC_RUN_ON(signaling_thread()); |
| |
| // Creates channel and adds it to the collection of DataChannels that will |
| // be offered in a SessionDescription. |
| rtc::scoped_refptr<DataChannel> InternalCreateDataChannel( |
| const std::string& label, |
| const InternalDataChannelInit* config) RTC_RUN_ON(signaling_thread()); |
| |
| // Checks if any data channel has been added. |
| bool HasDataChannels() const RTC_RUN_ON(signaling_thread()); |
| |
| void AllocateSctpSids(rtc::SSLRole role) RTC_RUN_ON(signaling_thread()); |
| void OnSctpDataChannelClosed(DataChannel* channel) |
| RTC_RUN_ON(signaling_thread()); |
| |
| void OnDataChannelDestroyed() RTC_RUN_ON(signaling_thread()); |
| // Called when a valid data channel OPEN message is received. |
| void OnDataChannelOpenMessage(const std::string& label, |
| const InternalDataChannelInit& config) |
| RTC_RUN_ON(signaling_thread()); |
| |
| // Parses and handles open messages. Returns true if the message is an open |
| // message, false otherwise. |
| bool HandleOpenMessage_s(const cricket::ReceiveDataParams& params, |
| const rtc::CopyOnWriteBuffer& buffer) |
| RTC_RUN_ON(signaling_thread()); |
| |
| // Returns true if the PeerConnection is configured to use Unified Plan |
| // semantics for creating offers/answers and setting local/remote |
| // descriptions. If this is true the RtpTransceiver API will also be available |
| // to the user. If this is false, Plan B semantics are assumed. |
| // TODO(bugs.webrtc.org/8530): Flip the default to be Unified Plan once |
| // sufficient time has passed. |
| bool IsUnifiedPlan() const RTC_RUN_ON(signaling_thread()) { |
| return configuration_.sdp_semantics == SdpSemantics::kUnifiedPlan; |
| } |
| |
| // The offer/answer machinery assumes the media section MID is present and |
| // unique. To support legacy end points that do not supply a=mid lines, this |
| // method will modify the session description to add MIDs generated according |
| // to the SDP semantics. |
| void FillInMissingRemoteMids(cricket::SessionDescription* remote_description) |
| RTC_RUN_ON(signaling_thread()); |
| |
| // Is there an RtpSender of the given type? |
| bool HasRtpSender(cricket::MediaType type) const |
| RTC_RUN_ON(signaling_thread()); |
| |
| // Return the RtpSender with the given track attached. |
| rtc::scoped_refptr<RtpSenderProxyWithInternal<RtpSenderInternal>> |
| FindSenderForTrack(MediaStreamTrackInterface* track) const |
| RTC_RUN_ON(signaling_thread()); |
| |
| // Return the RtpSender with the given id, or null if none exists. |
| rtc::scoped_refptr<RtpSenderProxyWithInternal<RtpSenderInternal>> |
| FindSenderById(const std::string& sender_id) const |
| RTC_RUN_ON(signaling_thread()); |
| |
| // Return the RtpReceiver with the given id, or null if none exists. |
| rtc::scoped_refptr<RtpReceiverProxyWithInternal<RtpReceiverInternal>> |
| FindReceiverById(const std::string& receiver_id) const |
| RTC_RUN_ON(signaling_thread()); |
| |
| std::vector<RtpSenderInfo>* GetRemoteSenderInfos( |
| cricket::MediaType media_type); |
| std::vector<RtpSenderInfo>* GetLocalSenderInfos( |
| cricket::MediaType media_type); |
| const RtpSenderInfo* FindSenderInfo(const std::vector<RtpSenderInfo>& infos, |
| const std::string& stream_id, |
| const std::string sender_id) const; |
| |
| // Returns the specified SCTP DataChannel in sctp_data_channels_, |
| // or nullptr if not found. |
| DataChannel* FindDataChannelBySid(int sid) const |
| RTC_RUN_ON(signaling_thread()); |
| |
| // Called when first configuring the port allocator. |
| struct InitializePortAllocatorResult { |
| bool enable_ipv6; |
| }; |
| InitializePortAllocatorResult InitializePortAllocator_n( |
| const cricket::ServerAddresses& stun_servers, |
| const std::vector<cricket::RelayServerConfig>& turn_servers, |
| const RTCConfiguration& configuration); |
| // Called when SetConfiguration is called to apply the supported subset |
| // of the configuration on the network thread. |
| bool ReconfigurePortAllocator_n( |
| const cricket::ServerAddresses& stun_servers, |
| const std::vector<cricket::RelayServerConfig>& turn_servers, |
| IceTransportsType type, |
| int candidate_pool_size, |
| bool prune_turn_ports, |
| webrtc::TurnCustomizer* turn_customizer, |
| absl::optional<int> stun_candidate_keepalive_interval, |
| bool have_local_description); |
| |
| // Starts output of an RTC event log to the given output object. |
| // This function should only be called from the worker thread. |
| bool StartRtcEventLog_w(std::unique_ptr<RtcEventLogOutput> output, |
| int64_t output_period_ms); |
| |
| // Stops recording an RTC event log. |
| // This function should only be called from the worker thread. |
| void StopRtcEventLog_w(); |
| |
| // Ensures the configuration doesn't have any parameters with invalid values, |
| // or values that conflict with other parameters. |
| // |
| // Returns RTCError::OK() if there are no issues. |
| RTCError ValidateConfiguration(const RTCConfiguration& config) const; |
| |
| cricket::ChannelManager* channel_manager() const; |
| |
| enum class SessionError { |
| kNone, // No error. |
| kContent, // Error in BaseChannel SetLocalContent/SetRemoteContent. |
| kTransport, // Error from the underlying transport. |
| }; |
| |
| // Returns the last error in the session. See the enum above for details. |
| SessionError session_error() const RTC_RUN_ON(signaling_thread()) { |
| return session_error_; |
| } |
| const std::string& session_error_desc() const { return session_error_desc_; } |
| |
| cricket::ChannelInterface* GetChannel(const std::string& content_name); |
| |
| // Get current SSL role used by SCTP's underlying transport. |
| bool GetSctpSslRole(rtc::SSLRole* role); |
| |
| cricket::IceConfig ParseIceConfig( |
| const PeerConnectionInterface::RTCConfiguration& config) const; |
| |
| // Implements DataChannelProviderInterface. |
| bool SendData(const cricket::SendDataParams& params, |
| const rtc::CopyOnWriteBuffer& payload, |
| cricket::SendDataResult* result) override; |
| bool ConnectDataChannel(DataChannel* webrtc_data_channel) override; |
| void DisconnectDataChannel(DataChannel* webrtc_data_channel) override; |
| void AddSctpDataStream(int sid) override; |
| void RemoveSctpDataStream(int sid) override; |
| bool ReadyToSendData() const override; |
| |
| cricket::DataChannelType data_channel_type() const; |
| |
| // Implements DataChannelSink. |
| void OnDataReceived(int channel_id, |
| DataMessageType type, |
| const rtc::CopyOnWriteBuffer& buffer) override; |
| void OnChannelClosing(int channel_id) override; |
| void OnChannelClosed(int channel_id) override; |
| |
| // Called when an RTCCertificate is generated or retrieved by |
| // WebRTCSessionDescriptionFactory. Should happen before setLocalDescription. |
| void OnCertificateReady( |
| const rtc::scoped_refptr<rtc::RTCCertificate>& certificate); |
| void OnDtlsSrtpSetupFailure(cricket::BaseChannel*, bool rtcp); |
| |
| // Non-const versions of local_description()/remote_description(), for use |
| // internally. |
| SessionDescriptionInterface* mutable_local_description() |
| RTC_RUN_ON(signaling_thread()) { |
| return pending_local_description_ ? pending_local_description_.get() |
| : current_local_description_.get(); |
| } |
| SessionDescriptionInterface* mutable_remote_description() |
| RTC_RUN_ON(signaling_thread()) { |
| return pending_remote_description_ ? pending_remote_description_.get() |
| : current_remote_description_.get(); |
| } |
| |
| // Updates the error state, signaling if necessary. |
| void SetSessionError(SessionError error, const std::string& error_desc); |
| |
| RTCError UpdateSessionState(SdpType type, |
| cricket::ContentSource source, |
| const cricket::SessionDescription* description); |
| // Push the media parts of the local or remote session description |
| // down to all of the channels. |
| RTCError PushdownMediaDescription(SdpType type, cricket::ContentSource source) |
| RTC_RUN_ON(signaling_thread()); |
| bool PushdownSctpParameters_n(cricket::ContentSource source, |
| int local_sctp_port, |
| int remote_sctp_port); |
| |
| RTCError PushdownTransportDescription(cricket::ContentSource source, |
| SdpType type); |
| |
| // Returns true and the TransportInfo of the given |content_name| |
| // from |description|. Returns false if it's not available. |
| static bool GetTransportDescription( |
| const cricket::SessionDescription* description, |
| const std::string& content_name, |
| cricket::TransportDescription* info); |
| |
| // Enables media channels to allow sending of media. |
| // This enables media to flow on all configured audio/video channels and the |
| // RtpDataChannel. |
| void EnableSending() RTC_RUN_ON(signaling_thread()); |
| |
| // Destroys all BaseChannels and destroys the SCTP data channel, if present. |
| void DestroyAllChannels() RTC_RUN_ON(signaling_thread()); |
| |
| // Returns the media index for a local ice candidate given the content name. |
| // Returns false if the local session description does not have a media |
| // content called |content_name|. |
| bool GetLocalCandidateMediaIndex(const std::string& content_name, |
| int* sdp_mline_index) |
| RTC_RUN_ON(signaling_thread()); |
| // Uses all remote candidates in |remote_desc| in this session. |
| bool UseCandidatesInSessionDescription( |
| const SessionDescriptionInterface* remote_desc) |
| RTC_RUN_ON(signaling_thread()); |
| // Uses |candidate| in this session. |
| bool UseCandidate(const IceCandidateInterface* candidate) |
| RTC_RUN_ON(signaling_thread()); |
| // Deletes the corresponding channel of contents that don't exist in |desc|. |
| // |desc| can be null. This means that all channels are deleted. |
| void RemoveUnusedChannels(const cricket::SessionDescription* desc) |
| RTC_RUN_ON(signaling_thread()); |
| |
| // Allocates media channels based on the |desc|. If |desc| doesn't have |
| // the BUNDLE option, this method will disable BUNDLE in PortAllocator. |
| // This method will also delete any existing media channels before creating. |
| RTCError CreateChannels(const cricket::SessionDescription& desc) |
| RTC_RUN_ON(signaling_thread()); |
| |
| // If the BUNDLE policy is max-bundle, then we know for sure that all |
| // transports will be bundled from the start. This method returns the BUNDLE |
| // group if that's the case, or null if BUNDLE will be negotiated later. An |
| // error is returned if max-bundle is specified but the session description |
| // does not have a BUNDLE group. |
| RTCErrorOr<const cricket::ContentGroup*> GetEarlyBundleGroup( |
| const cricket::SessionDescription& desc) const |
| RTC_RUN_ON(signaling_thread()); |
| |
| // Helper methods to create media channels. |
| cricket::VoiceChannel* CreateVoiceChannel(const std::string& mid) |
| RTC_RUN_ON(signaling_thread()); |
| cricket::VideoChannel* CreateVideoChannel(const std::string& mid) |
| RTC_RUN_ON(signaling_thread()); |
| bool CreateDataChannel(const std::string& mid) RTC_RUN_ON(signaling_thread()); |
| |
| bool CreateSctpTransport_n(const std::string& mid); |
| // For bundling. |
| void DestroySctpTransport_n(); |
| // SctpTransport signal handlers. Needed to marshal signals from the network |
| // to signaling thread. |
| void OnSctpTransportReadyToSendData_n(); |
| // This may be called with "false" if the direction of the m= section causes |
| // us to tear down the SCTP connection. |
| void OnSctpTransportReadyToSendData_s(bool ready); |
| void OnSctpTransportDataReceived_n(const cricket::ReceiveDataParams& params, |
| const rtc::CopyOnWriteBuffer& payload); |
| // Beyond just firing the signal to the signaling thread, listens to SCTP |
| // CONTROL messages on unused SIDs and processes them as OPEN messages. |
| void OnSctpTransportDataReceived_s(const cricket::ReceiveDataParams& params, |
| const rtc::CopyOnWriteBuffer& payload); |
| void OnSctpClosingProcedureStartedRemotely_n(int sid); |
| void OnSctpClosingProcedureComplete_n(int sid); |
| |
| bool SetupMediaTransportForDataChannels_n(const std::string& mid) |
| RTC_RUN_ON(network_thread()); |
| void OnMediaTransportStateChanged_n() RTC_RUN_ON(network_thread()); |
| void TeardownMediaTransportForDataChannels_n() RTC_RUN_ON(network_thread()); |
| |
| bool ValidateBundleSettings(const cricket::SessionDescription* desc); |
| bool HasRtcpMuxEnabled(const cricket::ContentInfo* content); |
| // Below methods are helper methods which verifies SDP. |
| RTCError ValidateSessionDescription(const SessionDescriptionInterface* sdesc, |
| cricket::ContentSource source) |
| RTC_RUN_ON(signaling_thread()); |
| |
| // Check if a call to SetLocalDescription is acceptable with a session |
| // description of the given type. |
| bool ExpectSetLocalDescription(SdpType type); |
| // Check if a call to SetRemoteDescription is acceptable with a session |
| // description of the given type. |
| bool ExpectSetRemoteDescription(SdpType type); |
| // Verifies a=setup attribute as per RFC 5763. |
| bool ValidateDtlsSetupAttribute(const cricket::SessionDescription* desc, |
| SdpType type); |
| |
| // Returns true if we are ready to push down the remote candidate. |
| // |remote_desc| is the new remote description, or NULL if the current remote |
| // description should be used. Output |valid| is true if the candidate media |
| // index is valid. |
| bool ReadyToUseRemoteCandidate(const IceCandidateInterface* candidate, |
| const SessionDescriptionInterface* remote_desc, |
| bool* valid) RTC_RUN_ON(signaling_thread()); |
| |
| // Returns true if SRTP (either using DTLS-SRTP or SDES) is required by |
| // this session. |
| bool SrtpRequired() const RTC_RUN_ON(signaling_thread()); |
| |
| // JsepTransportController signal handlers. |
| void OnTransportControllerConnectionState(cricket::IceConnectionState state) |
| RTC_RUN_ON(signaling_thread()); |
| void OnTransportControllerGatheringState(cricket::IceGatheringState state) |
| RTC_RUN_ON(signaling_thread()); |
| void OnTransportControllerCandidatesGathered( |
| const std::string& transport_name, |
| const std::vector<cricket::Candidate>& candidates) |
| RTC_RUN_ON(signaling_thread()); |
| void OnTransportControllerCandidatesRemoved( |
| const std::vector<cricket::Candidate>& candidates) |
| RTC_RUN_ON(signaling_thread()); |
| void OnTransportControllerDtlsHandshakeError(rtc::SSLHandshakeError error); |
| |
| const char* SessionErrorToString(SessionError error) const; |
| std::string GetSessionErrorMsg() RTC_RUN_ON(signaling_thread()); |
| |
| // Report the UMA metric SdpFormatReceived for the given remote offer. |
| void ReportSdpFormatReceived(const SessionDescriptionInterface& remote_offer); |
| |
| // Report inferred negotiated SDP semantics from a local/remote answer to the |
| // UMA observer. |
| void ReportNegotiatedSdpSemantics(const SessionDescriptionInterface& answer); |
| |
| // Invoked when TransportController connection completion is signaled. |
| // Reports stats for all transports in use. |
| void ReportTransportStats() RTC_RUN_ON(signaling_thread()); |
| |
| // Gather the usage of IPv4/IPv6 as best connection. |
| void ReportBestConnectionState(const cricket::TransportStats& stats); |
| |
| void ReportNegotiatedCiphers(const cricket::TransportStats& stats, |
| const std::set<cricket::MediaType>& media_types) |
| RTC_RUN_ON(signaling_thread()); |
| |
| void NoteUsageEvent(UsageEvent event); |
| void ReportUsagePattern() const RTC_RUN_ON(signaling_thread()); |
| |
| void OnSentPacket_w(const rtc::SentPacket& sent_packet); |
| |
| const std::string GetTransportName(const std::string& content_name) |
| RTC_RUN_ON(signaling_thread()); |
| |
| // Destroys and clears the BaseChannel associated with the given transceiver, |
| // if such channel is set. |
| void DestroyTransceiverChannel( |
| rtc::scoped_refptr<RtpTransceiverProxyWithInternal<RtpTransceiver>> |
| transceiver); |
| |
| // Destroys the RTP data channel and/or the SCTP data channel and clears it. |
| void DestroyDataChannel() RTC_RUN_ON(signaling_thread()); |
| |
| // Destroys the given ChannelInterface. |
| // The channel cannot be accessed after this method is called. |
| void DestroyChannelInterface(cricket::ChannelInterface* channel); |
| |
| // JsepTransportController::Observer override. |
| // |
| // Called by |transport_controller_| when processing transport information |
| // from a session description, and the mapping from m= sections to transports |
| // changed (as a result of BUNDLE negotiation, or m= sections being |
| // rejected). |
| bool OnTransportChanged(const std::string& mid, |
| RtpTransportInternal* rtp_transport, |
| rtc::scoped_refptr<DtlsTransport> dtls_transport, |
| MediaTransportInterface* media_transport) override; |
| |
| // Returns the observer. Will crash on CHECK if the observer is removed. |
| PeerConnectionObserver* Observer() const RTC_RUN_ON(signaling_thread()); |
| |
| // Returns the CryptoOptions for this PeerConnection. This will always |
| // return the RTCConfiguration.crypto_options if set and will only default |
| // back to the PeerConnectionFactory settings if nothing was set. |
| CryptoOptions GetCryptoOptions() RTC_RUN_ON(signaling_thread()); |
| |
| // Returns rtp transport, result can not be nullptr. |
| RtpTransportInternal* GetRtpTransport(const std::string& mid) |
| RTC_RUN_ON(signaling_thread()) { |
| auto rtp_transport = transport_controller_->GetRtpTransport(mid); |
| RTC_DCHECK(rtp_transport); |
| return rtp_transport; |
| } |
| |
| // Returns media transport, if PeerConnection was created with configuration |
| // to use media transport. Otherwise returns nullptr. |
| MediaTransportInterface* GetMediaTransport(const std::string& mid) |
| RTC_RUN_ON(signaling_thread()) { |
| auto media_transport = transport_controller_->GetMediaTransport(mid); |
| RTC_DCHECK((configuration_.use_media_transport || |
| configuration_.use_media_transport_for_data_channels) == |
| (media_transport != nullptr)) |
| << "configuration_.use_media_transport=" |
| << configuration_.use_media_transport |
| << ", configuration_.use_media_transport_for_data_channels=" |
| << configuration_.use_media_transport_for_data_channels |
| << ", (media_transport != nullptr)=" << (media_transport != nullptr); |
| return media_transport; |
| } |
| |
| sigslot::signal1<DataChannel*> SignalDataChannelCreated_ |
| RTC_GUARDED_BY(signaling_thread()); |
| |
| // Storing the factory as a scoped reference pointer ensures that the memory |
| // in the PeerConnectionFactoryImpl remains available as long as the |
| // PeerConnection is running. It is passed to PeerConnection as a raw pointer. |
| // However, since the reference counting is done in the |
| // PeerConnectionFactoryInterface all instances created using the raw pointer |
| // will refer to the same reference count. |
| const rtc::scoped_refptr<PeerConnectionFactory> factory_; |
| PeerConnectionObserver* observer_ RTC_GUARDED_BY(signaling_thread()) = |
| nullptr; |
| |
| // The EventLog needs to outlive |call_| (and any other object that uses it). |
| std::unique_ptr<RtcEventLog> event_log_ RTC_GUARDED_BY(worker_thread()); |
| |
| // Points to the same thing as `event_log_`. Since it's const, we may read the |
| // pointer (but not touch the object) from any thread. |
| RtcEventLog* const event_log_ptr_ RTC_PT_GUARDED_BY(worker_thread()); |
| |
| SignalingState signaling_state_ RTC_GUARDED_BY(signaling_thread()) = kStable; |
| IceConnectionState ice_connection_state_ RTC_GUARDED_BY(signaling_thread()) = |
| kIceConnectionNew; |
| PeerConnectionInterface::IceConnectionState standardized_ice_connection_state_ |
| RTC_GUARDED_BY(signaling_thread()) = kIceConnectionNew; |
| PeerConnectionInterface::PeerConnectionState connection_state_ |
| RTC_GUARDED_BY(signaling_thread()) = PeerConnectionState::kNew; |
| |
| IceGatheringState ice_gathering_state_ RTC_GUARDED_BY(signaling_thread()) = |
| kIceGatheringNew; |
| PeerConnectionInterface::RTCConfiguration configuration_ |
| RTC_GUARDED_BY(signaling_thread()); |
| |
| // Cache configuration_.use_media_transport so that we can access it from |
| // other threads. |
| // TODO(bugs.webrtc.org/9987): Caching just this bool and allowing the data |
| // it's derived from to change is not necessarily sound. Stop doing it. |
| rtc::RaceChecker use_media_transport_race_checker_; |
| bool use_media_transport_ RTC_GUARDED_BY(use_media_transport_race_checker_) = |
| configuration_.use_media_transport; |
| |
| // TODO(zstein): |async_resolver_factory_| can currently be nullptr if it |
| // is not injected. It should be required once chromium supplies it. |
| std::unique_ptr<AsyncResolverFactory> async_resolver_factory_ |
| RTC_GUARDED_BY(signaling_thread()); |
| std::unique_ptr<cricket::PortAllocator> |
| port_allocator_; // TODO(bugs.webrtc.org/9987): Accessed on both |
| // signaling and network thread. |
| std::unique_ptr<rtc::SSLCertificateVerifier> |
| tls_cert_verifier_; // TODO(bugs.webrtc.org/9987): Accessed on both |
| // signaling and network thread. |
| |
| // One PeerConnection has only one RTCP CNAME. |
| // https://tools.ietf.org/html/draft-ietf-rtcweb-rtp-usage-26#section-4.9 |
| const std::string rtcp_cname_; |
| |
| // Streams added via AddStream. |
| const rtc::scoped_refptr<StreamCollection> local_streams_ |
| RTC_GUARDED_BY(signaling_thread()); |
| // Streams created as a result of SetRemoteDescription. |
| const rtc::scoped_refptr<StreamCollection> remote_streams_ |
| RTC_GUARDED_BY(signaling_thread()); |
| |
| std::vector<std::unique_ptr<MediaStreamObserver>> stream_observers_ |
| RTC_GUARDED_BY(signaling_thread()); |
| |
| // These lists store sender info seen in local/remote descriptions. |
| std::vector<RtpSenderInfo> remote_audio_sender_infos_ |
| RTC_GUARDED_BY(signaling_thread()); |
| std::vector<RtpSenderInfo> remote_video_sender_infos_ |
| RTC_GUARDED_BY(signaling_thread()); |
| std::vector<RtpSenderInfo> local_audio_sender_infos_ |
| RTC_GUARDED_BY(signaling_thread()); |
| std::vector<RtpSenderInfo> local_video_sender_infos_ |
| RTC_GUARDED_BY(signaling_thread()); |
| |
| SctpSidAllocator sid_allocator_ RTC_GUARDED_BY(signaling_thread()); |
| // label -> DataChannel |
| std::map<std::string, rtc::scoped_refptr<DataChannel>> rtp_data_channels_ |
| RTC_GUARDED_BY(signaling_thread()); |
| std::vector<rtc::scoped_refptr<DataChannel>> sctp_data_channels_ |
| RTC_GUARDED_BY(signaling_thread()); |
| std::vector<rtc::scoped_refptr<DataChannel>> sctp_data_channels_to_free_ |
| RTC_GUARDED_BY(signaling_thread()); |
| |
| bool remote_peer_supports_msid_ RTC_GUARDED_BY(signaling_thread()) = false; |
| |
| // The unique_ptr belongs to the worker thread, but the Call object manages |
| // its own thread safety. |
| std::unique_ptr<Call> call_ RTC_GUARDED_BY(worker_thread()); |
| |
| // Points to the same thing as `call_`. Since it's const, we may read the |
| // pointer from any thread. |
| Call* const call_ptr_; |
| |
| std::unique_ptr<StatsCollector> stats_ |
| RTC_GUARDED_BY(signaling_thread()); // A pointer is passed to senders_ |
| rtc::scoped_refptr<RTCStatsCollector> stats_collector_ |
| RTC_GUARDED_BY(signaling_thread()); |
| |
| std::vector< |
| rtc::scoped_refptr<RtpTransceiverProxyWithInternal<RtpTransceiver>>> |
| transceivers_; // TODO(bugs.webrtc.org/9987): Accessed on both signaling |
| // and network thread. |
| |
| // In Unified Plan, if we encounter remote SDP that does not contain an a=msid |
| // line we create and use a stream with a random ID for our receivers. This is |
| // to support legacy endpoints that do not support the a=msid attribute (as |
| // opposed to streamless tracks with "a=msid:-"). |
| rtc::scoped_refptr<MediaStreamInterface> missing_msid_default_stream_ |
| RTC_GUARDED_BY(signaling_thread()); |
| // MIDs will be generated using this generator which will keep track of |
| // all the MIDs that have been seen over the life of the PeerConnection. |
| rtc::UniqueStringGenerator mid_generator_ RTC_GUARDED_BY(signaling_thread()); |
| |
| SessionError session_error_ RTC_GUARDED_BY(signaling_thread()) = |
| SessionError::kNone; |
| std::string session_error_desc_ RTC_GUARDED_BY(signaling_thread()); |
| |
| std::string session_id_ RTC_GUARDED_BY(signaling_thread()); |
| |
| std::unique_ptr<JsepTransportController> |
| transport_controller_; // TODO(bugs.webrtc.org/9987): Accessed on both |
| // signaling and network thread. |
| std::unique_ptr<cricket::SctpTransportInternalFactory> |
| sctp_factory_; // TODO(bugs.webrtc.org/9987): Accessed on both |
| // signaling and network thread. |
| // |rtp_data_channel_| is used if in RTP data channel mode, |sctp_transport_| |
| // when using SCTP. |
| cricket::RtpDataChannel* rtp_data_channel_ = |
| nullptr; // TODO(bugs.webrtc.org/9987): Accessed on both |
| // signaling and some other thread. |
| |
| cricket::SctpTransportInternal* cricket_sctp_transport() { |
| return sctp_transport_->internal(); |
| } |
| rtc::scoped_refptr<SctpTransport> |
| sctp_transport_; // TODO(bugs.webrtc.org/9987): Accessed on both |
| // signaling and network thread. |
| |
| // |sctp_mid_| is the content name (MID) in SDP. |
| absl::optional<std::string> |
| sctp_mid_; // TODO(bugs.webrtc.org/9987): Accessed on both signaling |
| // and network thread. |
| |
| // Value cached on signaling thread. Only updated when SctpReadyToSendData |
| // fires on the signaling thread. |
| bool sctp_ready_to_send_data_ RTC_GUARDED_BY(signaling_thread()) = false; |
| |
| // Same as signals provided by SctpTransport, but these are guaranteed to |
| // fire on the signaling thread, whereas SctpTransport fires on the networking |
| // thread. |
| // |sctp_invoker_| is used so that any signals queued on the signaling thread |
| // from the network thread are immediately discarded if the SctpTransport is |
| // destroyed (due to m= section being rejected). |
| // TODO(deadbeef): Use a proxy object to ensure that method calls/signals |
| // are marshalled to the right thread. Could almost use proxy.h for this, |
| // but it doesn't have a mechanism for marshalling sigslot::signals |
| std::unique_ptr<rtc::AsyncInvoker> sctp_invoker_ |
| RTC_GUARDED_BY(network_thread()); |
| sigslot::signal1<bool> SignalSctpReadyToSendData |
| RTC_GUARDED_BY(signaling_thread()); |
| sigslot::signal2<const cricket::ReceiveDataParams&, |
| const rtc::CopyOnWriteBuffer&> |
| SignalSctpDataReceived RTC_GUARDED_BY(signaling_thread()); |
| sigslot::signal1<int> SignalSctpClosingProcedureStartedRemotely |
| RTC_GUARDED_BY(signaling_thread()); |
| sigslot::signal1<int> SignalSctpClosingProcedureComplete |
| RTC_GUARDED_BY(signaling_thread()); |
| |
| // Whether this peer is the caller. Set when the local description is applied. |
| absl::optional<bool> is_caller_ RTC_GUARDED_BY(signaling_thread()); |
| |
| // Content name (MID) for media transport data channels in SDP. |
| absl::optional<std::string> |
| media_transport_data_mid_; // TODO(bugs.webrtc.org/9987): Accessed on |
| // both signaling and network thread. |
| |
| // Media transport used for data channels. Thread-safe. |
| MediaTransportInterface* media_transport_ = |
| nullptr; // TODO(bugs.webrtc.org/9987): Object is thread safe, but |
| // pointer accessed on both signaling and network thread. |
| |
| // Cached value of whether the media transport is ready to send. |
| bool media_transport_ready_to_send_data_ RTC_GUARDED_BY(signaling_thread()) = |
| false; |
| |
| // Used to invoke media transport signals on the signaling thread. |
| std::unique_ptr<rtc::AsyncInvoker> media_transport_invoker_ |
| RTC_GUARDED_BY(network_thread()); |
| |
| // Identical to the signals for SCTP, but from media transport: |
| sigslot::signal1<bool> SignalMediaTransportWritable_s |
| RTC_GUARDED_BY(signaling_thread()); |
| sigslot::signal2<const cricket::ReceiveDataParams&, |
| const rtc::CopyOnWriteBuffer&> |
| SignalMediaTransportReceivedData_s RTC_GUARDED_BY(signaling_thread()); |
| sigslot::signal1<int> SignalMediaTransportChannelClosing_s |
| RTC_GUARDED_BY(signaling_thread()); |
| sigslot::signal1<int> SignalMediaTransportChannelClosed_s |
| RTC_GUARDED_BY(signaling_thread()); |
| |
| std::unique_ptr<SessionDescriptionInterface> current_local_description_ |
| RTC_GUARDED_BY(signaling_thread()); |
| std::unique_ptr<SessionDescriptionInterface> pending_local_description_ |
| RTC_GUARDED_BY(signaling_thread()); |
| std::unique_ptr<SessionDescriptionInterface> current_remote_description_ |
| RTC_GUARDED_BY(signaling_thread()); |
| std::unique_ptr<SessionDescriptionInterface> pending_remote_description_ |
| RTC_GUARDED_BY(signaling_thread()); |
| bool dtls_enabled_ RTC_GUARDED_BY(signaling_thread()) = false; |
| // Specifies which kind of data channel is allowed. This is controlled |
| // by the chrome command-line flag and constraints: |
| // 1. If chrome command-line switch 'enable-sctp-data-channels' is enabled, |
| // constraint kEnableDtlsSrtp is true, and constaint kEnableRtpDataChannels is |
| // not set or false, SCTP is allowed (DCT_SCTP); |
| // 2. If constraint kEnableRtpDataChannels is true, RTP is allowed (DCT_RTP); |
| // 3. If both 1&2 are false, data channel is not allowed (DCT_NONE). |
| cricket::DataChannelType data_channel_type_ = cricket::DCT_NONE; |
| // List of content names for which the remote side triggered an ICE restart. |
| std::set<std::string> pending_ice_restarts_; |
| |
| std::unique_ptr<WebRtcSessionDescriptionFactory> webrtc_session_desc_factory_; |
| |
| // Member variables for caching global options. |
| cricket::AudioOptions audio_options_; |
| cricket::VideoOptions video_options_; |
| |
| int usage_event_accumulator_ = 0; |
| bool return_histogram_very_quickly_ = false; |
| |
| // This object should be used to generate any SSRC that is not explicitly |
| // specified by the user (or by the remote party). |
| // The generator is not used directly, instead it is passed on to the |
| // channel manager and the session description factory. |
| rtc::UniqueRandomIdGenerator ssrc_generator_; |
| }; |
| |
| } // namespace webrtc |
| |
| #endif // PC_PEER_CONNECTION_H_ |
