| /* |
| * 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 <stdint.h> |
| |
| #include <functional> |
| #include <map> |
| #include <memory> |
| #include <set> |
| #include <string> |
| #include <utility> |
| #include <vector> |
| |
| #include "absl/types/optional.h" |
| #include "api/adaptation/resource.h" |
| #include "api/async_dns_resolver.h" |
| #include "api/async_resolver_factory.h" |
| #include "api/audio_options.h" |
| #include "api/candidate.h" |
| #include "api/crypto/crypto_options.h" |
| #include "api/data_channel_interface.h" |
| #include "api/dtls_transport_interface.h" |
| #include "api/ice_transport_interface.h" |
| #include "api/jsep.h" |
| #include "api/media_stream_interface.h" |
| #include "api/media_types.h" |
| #include "api/packet_socket_factory.h" |
| #include "api/peer_connection_interface.h" |
| #include "api/rtc_error.h" |
| #include "api/rtc_event_log/rtc_event_log.h" |
| #include "api/rtc_event_log_output.h" |
| #include "api/rtp_parameters.h" |
| #include "api/rtp_receiver_interface.h" |
| #include "api/rtp_sender_interface.h" |
| #include "api/rtp_transceiver_interface.h" |
| #include "api/scoped_refptr.h" |
| #include "api/sctp_transport_interface.h" |
| #include "api/sequence_checker.h" |
| #include "api/set_local_description_observer_interface.h" |
| #include "api/set_remote_description_observer_interface.h" |
| #include "api/stats/rtc_stats_collector_callback.h" |
| #include "api/transport/bitrate_settings.h" |
| #include "api/transport/data_channel_transport_interface.h" |
| #include "api/transport/enums.h" |
| #include "api/turn_customizer.h" |
| #include "api/video/video_bitrate_allocator_factory.h" |
| #include "call/call.h" |
| #include "media/base/media_channel.h" |
| #include "media/base/media_engine.h" |
| #include "p2p/base/ice_transport_internal.h" |
| #include "p2p/base/port.h" |
| #include "p2p/base/port_allocator.h" |
| #include "p2p/base/transport_description.h" |
| #include "pc/channel.h" |
| #include "pc/channel_interface.h" |
| #include "pc/channel_manager.h" |
| #include "pc/connection_context.h" |
| #include "pc/data_channel_controller.h" |
| #include "pc/data_channel_utils.h" |
| #include "pc/dtls_transport.h" |
| #include "pc/jsep_transport_controller.h" |
| #include "pc/peer_connection_internal.h" |
| #include "pc/peer_connection_message_handler.h" |
| #include "pc/rtc_stats_collector.h" |
| #include "pc/rtp_receiver.h" |
| #include "pc/rtp_sender.h" |
| #include "pc/rtp_transceiver.h" |
| #include "pc/rtp_transmission_manager.h" |
| #include "pc/rtp_transport_internal.h" |
| #include "pc/sctp_data_channel.h" |
| #include "pc/sctp_transport.h" |
| #include "pc/sdp_offer_answer.h" |
| #include "pc/session_description.h" |
| #include "pc/stats_collector.h" |
| #include "pc/stream_collection.h" |
| #include "pc/transceiver_list.h" |
| #include "pc/transport_stats.h" |
| #include "pc/usage_pattern.h" |
| #include "rtc_base/checks.h" |
| #include "rtc_base/copy_on_write_buffer.h" |
| #include "rtc_base/network/sent_packet.h" |
| #include "rtc_base/rtc_certificate.h" |
| #include "rtc_base/ssl_certificate.h" |
| #include "rtc_base/ssl_stream_adapter.h" |
| #include "rtc_base/task_utils/pending_task_safety_flag.h" |
| #include "rtc_base/third_party/sigslot/sigslot.h" |
| #include "rtc_base/thread.h" |
| #include "rtc_base/thread_annotations.h" |
| #include "rtc_base/unique_id_generator.h" |
| #include "rtc_base/weak_ptr.h" |
| |
| namespace webrtc { |
| |
| // 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 sigslot::has_slots<> { |
| public: |
| // Creates a PeerConnection and initializes it with the given values. |
| // If the initialization fails, the function releases the PeerConnection |
| // and returns nullptr. |
| // |
| // Note that the function takes ownership of dependencies, and will |
| // either use them or release them, whether it succeeds or fails. |
| static RTCErrorOr<rtc::scoped_refptr<PeerConnection>> Create( |
| rtc::scoped_refptr<ConnectionContext> context, |
| const PeerConnectionFactoryInterface::Options& options, |
| std::unique_ptr<RtcEventLog> event_log, |
| std::unique_ptr<Call> call, |
| 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; |
| RTCError RemoveTrackOrError( |
| 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; |
| |
| 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; |
| |
| RTCErrorOr<rtc::scoped_refptr<DataChannelInterface>> CreateDataChannelOrError( |
| 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 { |
| 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 rtp_manager()->transceivers()->List(); |
| } |
| |
| sigslot::signal1<SctpDataChannel*>& SignalSctpDataChannelCreated() override { |
| RTC_DCHECK_RUN_ON(signaling_thread()); |
| return data_channel_controller_.SignalSctpDataChannelCreated(); |
| } |
| |
| std::vector<DataChannelStats> GetDataChannelStats() const override; |
| |
| absl::optional<std::string> sctp_transport_name() const override; |
| absl::optional<std::string> sctp_mid() const override; |
| |
| cricket::CandidateStatsList GetPooledCandidateStats() 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_ || |
| 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_; |
| } |
| PeerConnectionMessageHandler* message_handler() { |
| RTC_DCHECK_RUN_ON(signaling_thread()); |
| return &message_handler_; |
| } |
| |
| RtpTransmissionManager* rtp_manager() { return rtp_manager_.get(); } |
| const RtpTransmissionManager* rtp_manager() const { |
| return rtp_manager_.get(); |
| } |
| cricket::ChannelManager* channel_manager() const; |
| |
| JsepTransportController* transport_controller() { |
| return transport_controller_.get(); |
| } |
| cricket::PortAllocator* port_allocator() { return port_allocator_.get(); } |
| Call* call_ptr() { return call_ptr_; } |
| |
| ConnectionContext* context() { return context_.get(); } |
| const PeerConnectionFactoryInterface::Options* options() const { |
| return &options_; |
| } |
| void SetIceConnectionState(IceConnectionState new_state); |
| void NoteUsageEvent(UsageEvent event); |
| |
| // Asynchronously adds a remote candidate on the network thread. |
| void AddRemoteCandidate(const std::string& mid, |
| const cricket::Candidate& candidate); |
| |
| // Report the UMA metric SdpFormatReceived for the given remote description. |
| void ReportSdpFormatReceived( |
| const SessionDescriptionInterface& remote_description); |
| |
| // Report the UMA metric BundleUsage for the given remote description. |
| void ReportSdpBundleUsage( |
| const SessionDescriptionInterface& remote_description); |
| |
| // 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 is_unified_plan_; |
| } |
| bool ValidateBundleSettings( |
| const cricket::SessionDescription* desc, |
| const std::map<std::string, const cricket::ContentGroup*>& |
| bundle_groups_by_mid); |
| |
| // Returns the MID for the data section associated with the |
| // SCTP data channel, if it has been set. If no data |
| // channels are configured this will return nullopt. |
| absl::optional<std::string> GetDataMid() const; |
| |
| void SetSctpDataMid(const std::string& mid); |
| |
| void ResetSctpDataMid(); |
| |
| // Asynchronously calls SctpTransport::Start() on the network thread for |
| // `sctp_mid()` if set. Called as part of setting the local description. |
| void StartSctpTransport(int local_port, |
| int remote_port, |
| int max_message_size); |
| |
| // 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(); |
| |
| // 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); |
| |
| // Returns rtp transport, result can not be nullptr. |
| RtpTransportInternal* GetRtpTransport(const std::string& mid); |
| |
| // Returns true if SRTP (either using DTLS-SRTP or SDES) is required by |
| // this session. |
| bool SrtpRequired() const; |
| |
| bool SetupDataChannelTransport_n(const std::string& mid) |
| RTC_RUN_ON(network_thread()); |
| void TeardownDataChannelTransport_n() RTC_RUN_ON(network_thread()); |
| cricket::ChannelInterface* GetChannel(const std::string& content_name) |
| RTC_RUN_ON(network_thread()); |
| |
| // Functions made public for testing. |
| void ReturnHistogramVeryQuicklyForTesting() { |
| RTC_DCHECK_RUN_ON(signaling_thread()); |
| return_histogram_very_quickly_ = true; |
| } |
| void RequestUsagePatternReportForTesting(); |
| |
| protected: |
| // Available for rtc::scoped_refptr creation |
| PeerConnection(rtc::scoped_refptr<ConnectionContext> context, |
| const PeerConnectionFactoryInterface::Options& options, |
| bool is_unified_plan, |
| std::unique_ptr<RtcEventLog> event_log, |
| std::unique_ptr<Call> call, |
| PeerConnectionDependencies& dependencies, |
| bool dtls_enabled); |
| |
| ~PeerConnection() override; |
| |
| private: |
| RTCError Initialize( |
| const PeerConnectionInterface::RTCConfiguration& configuration, |
| PeerConnectionDependencies dependencies); |
| void InitializeTransportController_n( |
| const RTCConfiguration& configuration, |
| const PeerConnectionDependencies& dependencies) |
| RTC_RUN_ON(network_thread()); |
| |
| rtc::scoped_refptr<RtpTransceiverProxyWithInternal<RtpTransceiver>> |
| FindTransceiverBySender(rtc::scoped_refptr<RtpSenderInterface> sender) |
| 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()); |
| // 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()); |
| |
| void OnNegotiationNeeded(); |
| |
| // 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(); |
| |
| // 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()); |
| |
| // 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); |
| |
| // Invoked when TransportController connection completion is signaled. |
| // Reports stats for all transports in use. |
| void ReportTransportStats() RTC_RUN_ON(network_thread()); |
| |
| // Gather the usage of IPv4/IPv6 as best connection. |
| static void ReportBestConnectionState(const cricket::TransportStats& stats); |
| |
| static void ReportNegotiatedCiphers( |
| bool dtls_enabled, |
| const cricket::TransportStats& stats, |
| const std::set<cricket::MediaType>& media_types); |
| void ReportIceCandidateCollected(const cricket::Candidate& candidate) |
| RTC_RUN_ON(signaling_thread()); |
| |
| void ReportUsagePattern() const RTC_RUN_ON(signaling_thread()); |
| |
| void ReportRemoteIceCandidateAdded(const cricket::Candidate& candidate); |
| |
| // 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; |
| |
| std::function<void(const rtc::CopyOnWriteBuffer& packet, |
| int64_t packet_time_us)> |
| InitializeRtcpCallback(); |
| |
| const rtc::scoped_refptr<ConnectionContext> context_; |
| const PeerConnectionFactoryInterface::Options options_; |
| PeerConnectionObserver* observer_ RTC_GUARDED_BY(signaling_thread()) = |
| nullptr; |
| |
| const bool is_unified_plan_; |
| |
| // 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()); |
| |
| const std::unique_ptr<AsyncDnsResolverFactoryInterface> |
| async_dns_resolver_factory_; |
| std::unique_ptr<cricket::PortAllocator> |
| port_allocator_; // TODO(bugs.webrtc.org/9987): Accessed on both |
| // signaling and network thread. |
| const 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. |
| const std::unique_ptr<rtc::SSLCertificateVerifier> tls_cert_verifier_ |
| RTC_GUARDED_BY(network_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()); |
| ScopedTaskSafety signaling_thread_safety_; |
| rtc::scoped_refptr<PendingTaskSafetyFlag> network_thread_safety_; |
| rtc::scoped_refptr<PendingTaskSafetyFlag> worker_thread_safety_; |
| |
| // 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()); |
| |
| const std::string session_id_; |
| |
| 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()); |
| std::string sctp_transport_name_s_ RTC_GUARDED_BY(signaling_thread()); |
| |
| // The machinery for handling offers and answers. Const after initialization. |
| std::unique_ptr<SdpOfferAnswerHandler> sdp_handler_ |
| RTC_GUARDED_BY(signaling_thread()) RTC_PT_GUARDED_BY(signaling_thread()); |
| |
| const bool dtls_enabled_; |
| |
| UsagePattern usage_pattern_ RTC_GUARDED_BY(signaling_thread()); |
| bool return_histogram_very_quickly_ RTC_GUARDED_BY(signaling_thread()) = |
| false; |
| |
| // The DataChannelController is accessed from both the signaling thread |
| // and networking thread. It is a thread-aware object. |
| DataChannelController data_channel_controller_; |
| |
| // Machinery for handling messages posted to oneself |
| PeerConnectionMessageHandler message_handler_ |
| RTC_GUARDED_BY(signaling_thread()); |
| |
| // Administration of senders, receivers and transceivers |
| // Accessed on both signaling and network thread. Const after Initialize(). |
| std::unique_ptr<RtpTransmissionManager> rtp_manager_; |
| |
| // Did the connectionState ever change to `connected`? |
| // Used to gather metrics only the first such state change. |
| bool was_ever_connected_ RTC_GUARDED_BY(signaling_thread()) = false; |
| |
| // This variable needs to be the last one in the class. |
| rtc::WeakPtrFactory<PeerConnection> weak_factory_; |
| }; |
| |
| } // namespace webrtc |
| |
| #endif // PC_PEER_CONNECTION_H_ |