| /* |
| * 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 <functional> |
| #include <map> |
| #include <memory> |
| #include <set> |
| #include <string> |
| #include <utility> |
| #include <vector> |
| |
| #include "api/peer_connection_interface.h" |
| #include "api/transport/data_channel_transport_interface.h" |
| #include "api/turn_customizer.h" |
| #include "pc/connection_context.h" |
| #include "pc/data_channel_controller.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/peer_connection_message_handler.h" |
| #include "pc/rtc_stats_collector.h" |
| #include "pc/rtp_sender.h" |
| #include "pc/rtp_transceiver.h" |
| #include "pc/sctp_transport.h" |
| #include "pc/sdp_offer_answer.h" |
| #include "pc/stats_collector.h" |
| #include "pc/stream_collection.h" |
| #include "pc/transceiver_list.h" |
| #include "pc/usage_pattern.h" |
| #include "pc/webrtc_session_description_factory.h" |
| #include "rtc_base/experiments/field_trial_parser.h" |
| #include "rtc_base/operations_chain.h" |
| #include "rtc_base/race_checker.h" |
| #include "rtc_base/task_utils/pending_task_safety_flag.h" |
| #include "rtc_base/unique_id_generator.h" |
| #include "rtc_base/weak_ptr.h" |
| |
| namespace webrtc { |
| |
| class MediaStreamObserver; |
| class VideoRtpReceiver; |
| class RtcEventLog; |
| class SdpOfferAnswerHandler; |
| |
| // 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 JsepTransportController::Observer, |
| public RtpSenderBase::SetStreamsObserver, |
| public sigslot::has_slots<> { |
| public: |
| explicit PeerConnection(rtc::scoped_refptr<ConnectionContext> context, |
| 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; |
| absl::optional<bool> can_trickle_ice_candidates() 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; |
| |
| void RestartIce() override; |
| |
| // JSEP01 |
| void CreateOffer(CreateSessionDescriptionObserver* observer, |
| const RTCOfferAnswerOptions& options) override; |
| void CreateAnswer(CreateSessionDescriptionObserver* observer, |
| const RTCOfferAnswerOptions& options) override; |
| |
| void SetLocalDescription( |
| std::unique_ptr<SessionDescriptionInterface> desc, |
| rtc::scoped_refptr<SetLocalDescriptionObserverInterface> observer) |
| override; |
| void SetLocalDescription( |
| rtc::scoped_refptr<SetLocalDescriptionObserverInterface> observer) |
| override; |
| // TODO(https://crbug.com/webrtc/11798): Delete these methods in favor of the |
| // ones taking SetLocalDescriptionObserverInterface as argument. |
| void SetLocalDescription(SetSessionDescriptionObserver* observer, |
| SessionDescriptionInterface* desc) override; |
| void SetLocalDescription(SetSessionDescriptionObserver* observer) override; |
| |
| void SetRemoteDescription( |
| std::unique_ptr<SessionDescriptionInterface> desc, |
| rtc::scoped_refptr<SetRemoteDescriptionObserverInterface> observer) |
| override; |
| // TODO(https://crbug.com/webrtc/11798): Delete this methods in favor of the |
| // ones taking SetRemoteDescriptionObserverInterface as argument. |
| void SetRemoteDescription(SetSessionDescriptionObserver* observer, |
| SessionDescriptionInterface* desc) override; |
| |
| PeerConnectionInterface::RTCConfiguration GetConfiguration() override; |
| RTCError SetConfiguration( |
| const PeerConnectionInterface::RTCConfiguration& configuration) override; |
| bool AddIceCandidate(const IceCandidateInterface* candidate) override; |
| void AddIceCandidate(std::unique_ptr<IceCandidateInterface> candidate, |
| std::function<void(RTCError)> callback) override; |
| bool RemoveIceCandidates( |
| const std::vector<cricket::Candidate>& candidates) override; |
| |
| RTCError SetBitrate(const BitrateSettings& bitrate) 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; |
| |
| void AddAdaptationResource(rtc::scoped_refptr<Resource> resource) override; |
| |
| bool StartRtcEventLog(std::unique_ptr<RtcEventLogOutput> output, |
| int64_t output_period_ms) override; |
| bool StartRtcEventLog(std::unique_ptr<RtcEventLogOutput> output) override; |
| void StopRtcEventLog() override; |
| |
| void Close() override; |
| |
| rtc::Thread* signaling_thread() const final { |
| return context_->signaling_thread(); |
| } |
| |
| // PeerConnectionInternal implementation. |
| rtc::Thread* network_thread() const final { |
| return context_->network_thread(); |
| } |
| rtc::Thread* worker_thread() const final { return context_->worker_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_.List(); |
| } |
| |
| sigslot::signal1<RtpDataChannel*>& SignalRtpDataChannelCreated() override { |
| return data_channel_controller_.SignalRtpDataChannelCreated(); |
| } |
| |
| sigslot::signal1<SctpDataChannel*>& SignalSctpDataChannelCreated() override { |
| return data_channel_controller_.SignalSctpDataChannelCreated(); |
| } |
| |
| cricket::RtpDataChannel* rtp_data_channel() const override { |
| return data_channel_controller_.rtp_data_channel(); |
| } |
| |
| std::vector<DataChannelStats> GetDataChannelStats() const override; |
| |
| 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; |
| |
| // Functions needed by DataChannelController |
| void NoteDataAddedEvent() { NoteUsageEvent(UsageEvent::DATA_ADDED); } |
| // Returns the observer. Will crash on CHECK if the observer is removed. |
| PeerConnectionObserver* Observer() const; |
| bool IsClosed() const { |
| RTC_DCHECK_RUN_ON(signaling_thread()); |
| return sdp_handler_.signaling_state() == PeerConnectionInterface::kClosed; |
| } |
| // Get current SSL role used by SCTP's underlying transport. |
| bool GetSctpSslRole(rtc::SSLRole* role); |
| // Handler for the "channel closed" signal |
| void OnSctpDataChannelClosed(DataChannelInterface* channel); |
| |
| bool ShouldFireNegotiationNeededEvent(uint32_t event_id) override; |
| |
| // Functions needed by SdpOfferAnswerHandler |
| StatsCollector* stats() { |
| RTC_DCHECK_RUN_ON(signaling_thread()); |
| return stats_.get(); |
| } |
| DataChannelController* data_channel_controller() { |
| RTC_DCHECK_RUN_ON(signaling_thread()); |
| return &data_channel_controller_; |
| } |
| bool dtls_enabled() const { |
| RTC_DCHECK_RUN_ON(signaling_thread()); |
| return dtls_enabled_; |
| } |
| const PeerConnectionInterface::RTCConfiguration* configuration() const { |
| RTC_DCHECK_RUN_ON(signaling_thread()); |
| return &configuration_; |
| } |
| absl::optional<std::string> sctp_mid() { |
| RTC_DCHECK_RUN_ON(signaling_thread()); |
| return sctp_mid_s_; |
| } |
| PeerConnectionMessageHandler* message_handler() { |
| RTC_DCHECK_RUN_ON(signaling_thread()); |
| return &message_handler_; |
| } |
| |
| // Functions made public for testing. |
| void ReturnHistogramVeryQuicklyForTesting() { |
| RTC_DCHECK_RUN_ON(signaling_thread()); |
| return_histogram_very_quickly_ = true; |
| } |
| void RequestUsagePatternReportForTesting(); |
| |
| protected: |
| ~PeerConnection() override; |
| |
| private: |
| // While refactoring: Allow access from SDP negotiation |
| // TOOD(https://bugs.webrtc.org/11995): Remove friendship. |
| friend class SdpOfferAnswerHandler; |
| |
| 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; |
| }; |
| |
| // 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; |
| std::vector< |
| rtc::scoped_refptr<RtpReceiverProxyWithInternal<RtpReceiverInternal>>> |
| GetReceiversInternal() const RTC_RUN_ON(signaling_thread()); |
| |
| rtc::scoped_refptr<RtpTransceiverProxyWithInternal<RtpTransceiver>> |
| GetAudioTransceiver() const; |
| rtc::scoped_refptr<RtpTransceiverProxyWithInternal<RtpTransceiver>> |
| GetVideoTransceiver() const; |
| |
| 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); |
| void RemoveAudioTrack(AudioTrackInterface* track, |
| MediaStreamInterface* stream); |
| void AddVideoTrack(VideoTrackInterface* track, MediaStreamInterface* stream); |
| void RemoveVideoTrack(VideoTrackInterface* track, |
| MediaStreamInterface* stream); |
| |
| // 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::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); |
| |
| void SetIceConnectionState(IceConnectionState new_state); |
| 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()); |
| // Gathering of an ICE candidate failed. |
| void OnIceCandidateError(const std::string& address, |
| int port, |
| const std::string& url, |
| int error_code, |
| const std::string& error_text) |
| 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()); |
| |
| void OnSelectedCandidatePairChanged( |
| const cricket::CandidatePairChangeEvent& event) |
| 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 OnNegotiationNeeded(); |
| |
| |
| // 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; |
| |
| // 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, |
| MediaStreamInterface* stream, |
| cricket::MediaType media_type); |
| |
| // 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, |
| MediaStreamInterface* stream, |
| cricket::MediaType media_type); |
| |
| // 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); |
| |
| // 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); |
| |
| // 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_DCHECK_RUN_ON(signaling_thread()); |
| return configuration_.sdp_semantics == SdpSemantics::kUnifiedPlan; |
| } |
| |
| // 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. |
| SctpDataChannel* 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, |
| PortPrunePolicy turn_port_prune_policy, |
| 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; |
| |
| cricket::ChannelInterface* GetChannel(const std::string& content_name); |
| |
| cricket::IceConfig ParseIceConfig( |
| const PeerConnectionInterface::RTCConfiguration& config) const; |
| |
| cricket::DataChannelType data_channel_type() const; |
| |
| // Called when an RTCCertificate is generated or retrieved by |
| // WebRTCSessionDescriptionFactory. Should happen before setLocalDescription. |
| void OnCertificateReady( |
| const rtc::scoped_refptr<rtc::RTCCertificate>& certificate); |
| |
| // 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); |
| |
| // 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()); |
| |
| bool SetupDataChannelTransport_n(const std::string& mid) |
| RTC_RUN_ON(network_thread()); |
| void TeardownDataChannelTransport_n() RTC_RUN_ON(network_thread()); |
| |
| bool ValidateBundleSettings(const cricket::SessionDescription* desc); |
| bool HasRtcpMuxEnabled(const cricket::ContentInfo* content); |
| |
| // Verifies a=setup attribute as per RFC 5763. |
| bool ValidateDtlsSetupAttribute(const cricket::SessionDescription* desc, |
| SdpType type); |
| |
| // 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 OnTransportControllerCandidateError( |
| const cricket::IceCandidateErrorEvent& event) |
| RTC_RUN_ON(signaling_thread()); |
| void OnTransportControllerCandidatesRemoved( |
| const std::vector<cricket::Candidate>& candidates) |
| RTC_RUN_ON(signaling_thread()); |
| void OnTransportControllerCandidateChanged( |
| const cricket::CandidatePairChangeEvent& event) |
| RTC_RUN_ON(signaling_thread()); |
| void OnTransportControllerDtlsHandshakeError(rtc::SSLHandshakeError error); |
| |
| // Report the UMA metric SdpFormatReceived for the given remote offer. |
| void ReportSdpFormatReceived(const SessionDescriptionInterface& remote_offer); |
| |
| // 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 ReportIceCandidateCollected(const cricket::Candidate& candidate) |
| 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); |
| |
| // 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, |
| DataChannelTransportInterface* data_channel_transport) override; |
| |
| // RtpSenderBase::SetStreamsObserver override. |
| void OnSetStreams() override; |
| |
| // 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(); |
| |
| // Returns rtp transport, result can not be nullptr. |
| RtpTransportInternal* GetRtpTransport(const std::string& mid) { |
| RTC_DCHECK_RUN_ON(signaling_thread()); |
| auto rtp_transport = transport_controller_->GetRtpTransport(mid); |
| RTC_DCHECK(rtp_transport); |
| return rtp_transport; |
| } |
| |
| std::function<void(const rtc::CopyOnWriteBuffer& packet, |
| int64_t packet_time_us)> |
| InitializeRtcpCallback(); |
| |
| // 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<ConnectionContext> context_; |
| 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()); |
| |
| 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()); |
| |
| // 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<rtc::PacketSocketFactory> packet_socket_factory_; |
| std::unique_ptr<cricket::PortAllocator> |
| port_allocator_; // TODO(bugs.webrtc.org/9987): Accessed on both |
| // signaling and network thread. |
| std::unique_ptr<webrtc::IceTransportFactory> |
| ice_transport_factory_; // TODO(bugs.webrtc.org/9987): Accessed on the |
| // signaling thread but the underlying raw |
| // pointer is given to |
| // |jsep_transport_controller_| and used on the |
| // network thread. |
| std::unique_ptr<rtc::SSLCertificateVerifier> |
| tls_cert_verifier_; // TODO(bugs.webrtc.org/9987): Accessed on both |
| // signaling and network 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()); |
| |
| // 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()); |
| std::unique_ptr<ScopedTaskSafety> call_safety_ |
| RTC_GUARDED_BY(worker_thread()); |
| |
| // Points to the same thing as `call_`. Since it's const, we may read the |
| // pointer from any thread. |
| // TODO(bugs.webrtc.org/11992): Remove this workaround (and potential dangling |
| // pointer). |
| 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()); |
| TransceiverList transceivers_; |
| |
| 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. |
| |
| // |sctp_mid_| is the content name (MID) in SDP. |
| // Note: this is used as the data channel MID by both SCTP and data channel |
| // transports. It is set when either transport is initialized and unset when |
| // both transports are deleted. |
| // There is one copy on the signaling thread and another copy on the |
| // networking thread. Changes are always initiated from the signaling |
| // thread, but applied first on the networking thread via an invoke(). |
| absl::optional<std::string> sctp_mid_s_ RTC_GUARDED_BY(signaling_thread()); |
| absl::optional<std::string> sctp_mid_n_ RTC_GUARDED_BY(network_thread()); |
| |
| // The machinery for handling offers and answers. |
| SdpOfferAnswerHandler sdp_handler_ RTC_GUARDED_BY(signaling_thread()); |
| |
| bool dtls_enabled_ RTC_GUARDED_BY(signaling_thread()) = false; |
| |
| // Member variables for caching global options. |
| cricket::AudioOptions audio_options_ RTC_GUARDED_BY(signaling_thread()); |
| cricket::VideoOptions video_options_ RTC_GUARDED_BY(signaling_thread()); |
| |
| UsagePattern usage_pattern_ RTC_GUARDED_BY(signaling_thread()); |
| bool return_histogram_very_quickly_ RTC_GUARDED_BY(signaling_thread()) = |
| 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_ |
| RTC_GUARDED_BY(signaling_thread()); |
| |
| // A video bitrate allocator factory. |
| // This can injected using the PeerConnectionDependencies, |
| // or else the CreateBuiltinVideoBitrateAllocatorFactory() will be called. |
| // Note that one can still choose to override this in a MediaEngine |
| // if one wants too. |
| std::unique_ptr<webrtc::VideoBitrateAllocatorFactory> |
| video_bitrate_allocator_factory_; |
| |
| DataChannelController data_channel_controller_; |
| |
| // Machinery for handling messages posted to oneself |
| PeerConnectionMessageHandler message_handler_; |
| }; |
| |
| } // namespace webrtc |
| |
| #endif // PC_PEER_CONNECTION_H_ |