| /* |
| * 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 WEBRTC_API_PEERCONNECTION_H_ |
| #define WEBRTC_API_PEERCONNECTION_H_ |
| |
| #include <string> |
| #include <map> |
| #include <memory> |
| #include <vector> |
| |
| #include "webrtc/api/peerconnectionfactory.h" |
| #include "webrtc/api/peerconnectioninterface.h" |
| #include "webrtc/api/rtcstatscollector.h" |
| #include "webrtc/api/rtpreceiver.h" |
| #include "webrtc/api/rtpsender.h" |
| #include "webrtc/api/statscollector.h" |
| #include "webrtc/api/streamcollection.h" |
| #include "webrtc/api/webrtcsession.h" |
| |
| namespace webrtc { |
| |
| class MediaStreamObserver; |
| class VideoRtpReceiver; |
| |
| // Populates |session_options| from |rtc_options|, and returns true if options |
| // are valid. |
| // |session_options|->transport_options map entries must exist in order for |
| // them to be populated from |rtc_options|. |
| bool ExtractMediaSessionOptions( |
| const PeerConnectionInterface::RTCOfferAnswerOptions& rtc_options, |
| bool is_offer, |
| cricket::MediaSessionOptions* session_options); |
| |
| // Populates |session_options| from |constraints|, and returns true if all |
| // mandatory constraints are satisfied. |
| // Assumes that |session_options|->transport_options map entries exist. |
| // Will also set defaults if corresponding constraints are not present: |
| // recv_audio=true, recv_video=true, bundle_enabled=true. |
| // Other fields will be left with existing values. |
| // |
| // Deprecated. Will be removed once callers that use constraints are gone. |
| // TODO(hta): Remove when callers are gone. |
| // https://bugs.chromium.org/p/webrtc/issues/detail?id=5617 |
| bool ParseConstraintsForAnswer(const MediaConstraintsInterface* constraints, |
| cricket::MediaSessionOptions* session_options); |
| |
| // Parses the URLs for each server in |servers| to build |stun_servers| and |
| // |turn_servers|. |
| bool ParseIceServers(const PeerConnectionInterface::IceServers& servers, |
| cricket::ServerAddresses* stun_servers, |
| std::vector<cricket::RelayServerConfig>* turn_servers); |
| |
| // PeerConnection implements the PeerConnectionInterface interface. |
| // It uses WebRtcSession to implement the PeerConnection functionality. |
| class PeerConnection : public PeerConnectionInterface, |
| public IceObserver, |
| public rtc::MessageHandler, |
| public sigslot::has_slots<> { |
| public: |
| explicit PeerConnection(PeerConnectionFactory* factory); |
| |
| bool Initialize( |
| const PeerConnectionInterface::RTCConfiguration& configuration, |
| std::unique_ptr<cricket::PortAllocator> allocator, |
| std::unique_ptr<rtc::RTCCertificateGeneratorInterface> cert_generator, |
| PeerConnectionObserver* observer); |
| |
| 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; |
| |
| rtc::scoped_refptr<RtpSenderInterface> AddTrack( |
| MediaStreamTrackInterface* track, |
| std::vector<MediaStreamInterface*> streams) override; |
| bool RemoveTrack(RtpSenderInterface* sender) override; |
| |
| virtual WebRtcSession* session() { return session_.get(); } |
| |
| rtc::scoped_refptr<DtmfSenderInterface> CreateDtmfSender( |
| AudioTrackInterface* track) override; |
| |
| 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; |
| |
| rtc::scoped_refptr<DataChannelInterface> CreateDataChannel( |
| const std::string& label, |
| const DataChannelInit* config) override; |
| bool GetStats(StatsObserver* observer, |
| webrtc::MediaStreamTrackInterface* track, |
| StatsOutputLevel level) override; |
| void GetStats(RTCStatsCollectorCallback* callback) override; |
| |
| SignalingState signaling_state() override; |
| |
| // TODO(bemasc): Remove ice_state() when callers are removed. |
| IceState ice_state() override; |
| IceConnectionState ice_connection_state() override; |
| IceGatheringState ice_gathering_state() override; |
| |
| const SessionDescriptionInterface* local_description() const override; |
| const SessionDescriptionInterface* remote_description() const override; |
| |
| // JSEP01 |
| // Deprecated, use version without constraints. |
| void CreateOffer(CreateSessionDescriptionObserver* observer, |
| const MediaConstraintsInterface* constraints) override; |
| void CreateOffer(CreateSessionDescriptionObserver* observer, |
| const RTCOfferAnswerOptions& options) override; |
| // Deprecated, use version without constraints. |
| void CreateAnswer(CreateSessionDescriptionObserver* observer, |
| const MediaConstraintsInterface* constraints) override; |
| void CreateAnswer(CreateSessionDescriptionObserver* observer, |
| const RTCOfferAnswerOptions& options) override; |
| void SetLocalDescription(SetSessionDescriptionObserver* observer, |
| SessionDescriptionInterface* desc) override; |
| void SetRemoteDescription(SetSessionDescriptionObserver* observer, |
| SessionDescriptionInterface* desc) override; |
| bool SetConfiguration( |
| const PeerConnectionInterface::RTCConfiguration& configuration) override; |
| bool AddIceCandidate(const IceCandidateInterface* candidate) override; |
| bool RemoveIceCandidates( |
| const std::vector<cricket::Candidate>& candidates) override; |
| |
| void RegisterUMAObserver(UMAObserver* observer) override; |
| |
| bool StartRtcEventLog(rtc::PlatformFile file, |
| int64_t max_size_bytes) override; |
| void StopRtcEventLog() override; |
| |
| void Close() override; |
| |
| // Virtual for unit tests. |
| virtual const std::vector<rtc::scoped_refptr<DataChannel>>& |
| sctp_data_channels() const { |
| return sctp_data_channels_; |
| } |
| |
| protected: |
| ~PeerConnection() override; |
| |
| private: |
| struct TrackInfo { |
| TrackInfo() : ssrc(0) {} |
| TrackInfo(const std::string& stream_label, |
| const std::string track_id, |
| uint32_t ssrc) |
| : stream_label(stream_label), track_id(track_id), ssrc(ssrc) {} |
| bool operator==(const TrackInfo& other) { |
| return this->stream_label == other.stream_label && |
| this->track_id == other.track_id && this->ssrc == other.ssrc; |
| } |
| std::string stream_label; |
| std::string track_id; |
| uint32_t ssrc; |
| }; |
| typedef std::vector<TrackInfo> TrackInfos; |
| |
| // Implements MessageHandler. |
| void OnMessage(rtc::Message* msg) override; |
| |
| void CreateAudioReceiver(MediaStreamInterface* stream, |
| const std::string& track_id, |
| uint32_t ssrc); |
| |
| void CreateVideoReceiver(MediaStreamInterface* stream, |
| const std::string& track_id, |
| uint32_t ssrc); |
| void DestroyReceiver(const std::string& track_id); |
| void DestroyAudioSender(MediaStreamInterface* stream, |
| AudioTrackInterface* audio_track, |
| uint32_t ssrc); |
| void DestroyVideoSender(MediaStreamInterface* stream, |
| VideoTrackInterface* video_track); |
| |
| // Implements IceObserver |
| void OnIceConnectionChange(IceConnectionState new_state) override; |
| void OnIceGatheringChange(IceGatheringState new_state) override; |
| void OnIceCandidate(const IceCandidateInterface* candidate) override; |
| void OnIceCandidatesRemoved( |
| const std::vector<cricket::Candidate>& candidates) override; |
| void OnIceConnectionReceivingChange(bool receiving) override; |
| |
| // Signals from WebRtcSession. |
| void OnSessionStateChange(WebRtcSession* session, WebRtcSession::State state); |
| void ChangeSignalingState(SignalingState signaling_state); |
| |
| // Signals from MediaStreamObserver. |
| void OnAudioTrackAdded(AudioTrackInterface* track, |
| MediaStreamInterface* stream); |
| void OnAudioTrackRemoved(AudioTrackInterface* track, |
| MediaStreamInterface* stream); |
| void OnVideoTrackAdded(VideoTrackInterface* track, |
| MediaStreamInterface* stream); |
| void OnVideoTrackRemoved(VideoTrackInterface* track, |
| MediaStreamInterface* stream); |
| |
| rtc::Thread* signaling_thread() const { |
| return factory_->signaling_thread(); |
| } |
| |
| rtc::Thread* network_thread() const { return factory_->network_thread(); } |
| |
| void PostSetSessionDescriptionFailure(SetSessionDescriptionObserver* observer, |
| const std::string& error); |
| void PostCreateSessionDescriptionFailure( |
| CreateSessionDescriptionObserver* observer, |
| const std::string& error); |
| |
| bool IsClosed() const { |
| return signaling_state_ == PeerConnectionInterface::kClosed; |
| } |
| |
| // Returns a MediaSessionOptions struct with options decided by |options|, |
| // the local MediaStreams and DataChannels. |
| virtual bool GetOptionsForOffer( |
| const PeerConnectionInterface::RTCOfferAnswerOptions& rtc_options, |
| cricket::MediaSessionOptions* session_options); |
| |
| // Returns a MediaSessionOptions struct with options decided by |
| // |constraints|, the local MediaStreams and DataChannels. |
| // Deprecated, use version without constraints. |
| virtual bool GetOptionsForAnswer( |
| const MediaConstraintsInterface* constraints, |
| cricket::MediaSessionOptions* session_options); |
| virtual bool GetOptionsForAnswer( |
| const RTCOfferAnswerOptions& options, |
| cricket::MediaSessionOptions* session_options); |
| |
| void InitializeOptionsForAnswer( |
| cricket::MediaSessionOptions* session_options); |
| |
| // Helper function for options processing. |
| // Deprecated. |
| virtual void FinishOptionsForAnswer( |
| cricket::MediaSessionOptions* session_options); |
| |
| // Remove all local and remote tracks of type |media_type|. |
| // Called when a media type is rejected (m-line set to port 0). |
| void RemoveTracks(cricket::MediaType media_type); |
| |
| // 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 UpdateRemoteStreamsList( |
| const std::vector<cricket::StreamParams>& streams, |
| bool default_track_needed, |
| cricket::MediaType media_type, |
| StreamCollection* new_streams); |
| |
| // Triggered when a remote track has been seen for the first time in a remote |
| // session description. It creates a remote MediaStreamTrackInterface |
| // implementation and triggers CreateAudioReceiver or CreateVideoReceiver. |
| void OnRemoteTrackSeen(const std::string& stream_label, |
| const std::string& track_id, |
| uint32_t ssrc, |
| cricket::MediaType media_type); |
| |
| // Triggered when a remote track has been removed from a remote session |
| // description. It removes the remote track with id |track_id| from a remote |
| // MediaStream and triggers DestroyAudioReceiver or DestroyVideoReceiver. |
| void OnRemoteTrackRemoved(const std::string& stream_label, |
| const std::string& track_id, |
| cricket::MediaType media_type); |
| |
| // Finds remote MediaStreams without any tracks and removes them from |
| // |remote_streams_| and notifies the observer that the MediaStreams no longer |
| // exist. |
| void UpdateEndedRemoteMediaStreams(); |
| |
| // 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, OnLocalTrackSeen or |
| // OnLocalTrackRemoved is invoked. |
| void UpdateLocalTracks(const std::vector<cricket::StreamParams>& streams, |
| cricket::MediaType media_type); |
| |
| // Triggered when a local track 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 OnLocalTrackSeen(const std::string& stream_label, |
| const std::string& track_id, |
| uint32_t ssrc, |
| cricket::MediaType media_type); |
| |
| // Triggered when a local track 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 OnLocalTrackRemoved(const std::string& stream_label, |
| const std::string& track_id, |
| uint32_t ssrc, |
| cricket::MediaType media_type); |
| |
| void UpdateLocalRtpDataChannels(const cricket::StreamParamsVec& streams); |
| void UpdateRemoteRtpDataChannels(const cricket::StreamParamsVec& streams); |
| void UpdateClosingRtpDataChannels( |
| const std::vector<std::string>& active_channels, |
| bool is_local_update); |
| void CreateRemoteRtpDataChannel(const std::string& label, |
| uint32_t remote_ssrc); |
| |
| // 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); |
| |
| // Checks if any data channel has been added. |
| bool HasDataChannels() const; |
| |
| void AllocateSctpSids(rtc::SSLRole role); |
| void OnSctpDataChannelClosed(DataChannel* channel); |
| |
| // Notifications from WebRtcSession relating to BaseChannels. |
| void OnVoiceChannelCreated(); |
| void OnVoiceChannelDestroyed(); |
| void OnVideoChannelCreated(); |
| void OnVideoChannelDestroyed(); |
| void OnDataChannelCreated(); |
| void OnDataChannelDestroyed(); |
| // Called when the cricket::DataChannel receives a message indicating that a |
| // webrtc::DataChannel should be opened. |
| void OnDataChannelOpenMessage(const std::string& label, |
| const InternalDataChannelInit& config); |
| |
| RtpSenderInternal* FindSenderById(const std::string& id); |
| |
| std::vector<rtc::scoped_refptr< |
| RtpSenderProxyWithInternal<RtpSenderInternal>>>::iterator |
| FindSenderForTrack(MediaStreamTrackInterface* track); |
| std::vector<rtc::scoped_refptr< |
| RtpReceiverProxyWithInternal<RtpReceiverInternal>>>::iterator |
| FindReceiverForTrack(const std::string& track_id); |
| |
| TrackInfos* GetRemoteTracks(cricket::MediaType media_type); |
| TrackInfos* GetLocalTracks(cricket::MediaType media_type); |
| const TrackInfo* FindTrackInfo(const TrackInfos& infos, |
| const std::string& stream_label, |
| const std::string track_id) const; |
| |
| // Returns the specified SCTP DataChannel in sctp_data_channels_, |
| // or nullptr if not found. |
| DataChannel* FindDataChannelBySid(int sid) const; |
| |
| // Called when first configuring the port allocator. |
| bool InitializePortAllocator_n(const RTCConfiguration& configuration); |
| // Called when SetConfiguration is called. Only a subset of the configuration |
| // is applied. |
| bool ReconfigurePortAllocator_n(const RTCConfiguration& configuration); |
| |
| // Starts recording an Rtc EventLog using the supplied platform file. |
| // This function should only be called from the worker thread. |
| bool StartRtcEventLog_w(rtc::PlatformFile file, int64_t max_size_bytes); |
| // Starts recording an Rtc EventLog using the supplied platform file. |
| // This function should only be called from the worker thread. |
| void StopRtcEventLog_w(); |
| |
| // 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. |
| rtc::scoped_refptr<PeerConnectionFactory> factory_; |
| PeerConnectionObserver* observer_; |
| UMAObserver* uma_observer_; |
| SignalingState signaling_state_; |
| // TODO(bemasc): Remove ice_state_. |
| IceState ice_state_; |
| IceConnectionState ice_connection_state_; |
| IceGatheringState ice_gathering_state_; |
| |
| std::unique_ptr<cricket::PortAllocator> port_allocator_; |
| std::unique_ptr<MediaControllerInterface> media_controller_; |
| |
| // One PeerConnection has only one RTCP CNAME. |
| // https://tools.ietf.org/html/draft-ietf-rtcweb-rtp-usage-26#section-4.9 |
| std::string rtcp_cname_; |
| |
| // Streams added via AddStream. |
| rtc::scoped_refptr<StreamCollection> local_streams_; |
| // Streams created as a result of SetRemoteDescription. |
| rtc::scoped_refptr<StreamCollection> remote_streams_; |
| |
| std::vector<std::unique_ptr<MediaStreamObserver>> stream_observers_; |
| |
| // These lists store track info seen in local/remote descriptions. |
| TrackInfos remote_audio_tracks_; |
| TrackInfos remote_video_tracks_; |
| TrackInfos local_audio_tracks_; |
| TrackInfos local_video_tracks_; |
| |
| SctpSidAllocator sid_allocator_; |
| // label -> DataChannel |
| std::map<std::string, rtc::scoped_refptr<DataChannel>> rtp_data_channels_; |
| std::vector<rtc::scoped_refptr<DataChannel>> sctp_data_channels_; |
| std::vector<rtc::scoped_refptr<DataChannel>> sctp_data_channels_to_free_; |
| |
| bool remote_peer_supports_msid_ = false; |
| |
| bool enable_ice_renomination_ = false; |
| |
| std::vector<rtc::scoped_refptr<RtpSenderProxyWithInternal<RtpSenderInternal>>> |
| senders_; |
| std::vector< |
| rtc::scoped_refptr<RtpReceiverProxyWithInternal<RtpReceiverInternal>>> |
| receivers_; |
| |
| std::unique_ptr<WebRtcSession> session_; |
| std::unique_ptr<StatsCollector> stats_; |
| rtc::scoped_refptr<RTCStatsCollector> stats_collector_; |
| }; |
| |
| } // namespace webrtc |
| |
| #endif // WEBRTC_API_PEERCONNECTION_H_ |