/* * Copyright 2018 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_INTERNAL_H_ #define PC_PEER_CONNECTION_INTERNAL_H_ #include #include #include #include #include #include "absl/types/optional.h" #include "api/peer_connection_interface.h" #include "call/call.h" #include "modules/audio_device/include/audio_device.h" #include "pc/jsep_transport_controller.h" #include "pc/peer_connection_message_handler.h" #include "pc/rtp_transceiver.h" #include "pc/rtp_transmission_manager.h" #include "pc/sctp_data_channel.h" namespace webrtc { class DataChannelController; class LegacyStatsCollector; // This interface defines the functions that are needed for // SdpOfferAnswerHandler to access PeerConnection internal state. class PeerConnectionSdpMethods { public: virtual ~PeerConnectionSdpMethods() = default; // The SDP session ID as defined by RFC 3264. virtual std::string session_id() const = 0; // Returns true if the ICE restart flag above was set, and no ICE restart has // occurred yet for this transport (by applying a local description with // changed ufrag/password). If the transport has been deleted as a result of // bundling, returns false. virtual bool NeedsIceRestart(const std::string& content_name) const = 0; virtual absl::optional sctp_mid() const = 0; // Functions below this comment are known to only be accessed // from SdpOfferAnswerHandler. // Return a pointer to the active configuration. virtual const PeerConnectionInterface::RTCConfiguration* configuration() const = 0; // Report the UMA metric BundleUsage for the given remote description. virtual void ReportSdpBundleUsage( const SessionDescriptionInterface& remote_description) = 0; virtual PeerConnectionMessageHandler* message_handler() = 0; virtual RtpTransmissionManager* rtp_manager() = 0; virtual const RtpTransmissionManager* rtp_manager() const = 0; virtual bool dtls_enabled() const = 0; virtual const PeerConnectionFactoryInterface::Options* options() const = 0; // 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. virtual CryptoOptions GetCryptoOptions() = 0; virtual JsepTransportController* transport_controller_s() = 0; virtual JsepTransportController* transport_controller_n() = 0; virtual DataChannelController* data_channel_controller() = 0; virtual cricket::PortAllocator* port_allocator() = 0; virtual LegacyStatsCollector* legacy_stats() = 0; // Returns the observer. Will crash on CHECK if the observer is removed. virtual PeerConnectionObserver* Observer() const = 0; virtual absl::optional GetSctpSslRole_n() = 0; virtual PeerConnectionInterface::IceConnectionState ice_connection_state_internal() = 0; virtual void SetIceConnectionState( PeerConnectionInterface::IceConnectionState new_state) = 0; virtual void NoteUsageEvent(UsageEvent event) = 0; virtual bool IsClosed() const = 0; // 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. virtual bool IsUnifiedPlan() const = 0; virtual bool ValidateBundleSettings( const cricket::SessionDescription* desc, const std::map& bundle_groups_by_mid) = 0; // Internal implementation for AddTransceiver family of methods. If // `fire_callback` is set, fires OnRenegotiationNeeded callback if successful. virtual RTCErrorOr> AddTransceiver(cricket::MediaType media_type, rtc::scoped_refptr track, const RtpTransceiverInit& init, bool fire_callback = true) = 0; // Asynchronously calls SctpTransport::Start() on the network thread for // `sctp_mid()` if set. Called as part of setting the local description. virtual void StartSctpTransport(int local_port, int remote_port, int max_message_size) = 0; // Asynchronously adds a remote candidate on the network thread. virtual void AddRemoteCandidate(const std::string& mid, const cricket::Candidate& candidate) = 0; virtual Call* call_ptr() = 0; // Returns true if SRTP (either using DTLS-SRTP or SDES) is required by // this session. virtual bool SrtpRequired() const = 0; // Initializes the data channel transport for the peerconnection instance. // This will have the effect that `sctp_mid()` and `sctp_transport_name()` // will return a set value (even though it might be an empty string) and the // dc transport will be initialized on the network thread. virtual bool CreateDataChannelTransport(absl::string_view mid) = 0; // Tears down the data channel transport state and clears the `sctp_mid()` and // `sctp_transport_name()` properties. virtual void DestroyDataChannelTransport(RTCError error) = 0; virtual const FieldTrialsView& trials() const = 0; virtual void ClearStatsCache() = 0; }; // Functions defined in this class are called by other objects, // but not by SdpOfferAnswerHandler. class PeerConnectionInternal : public PeerConnectionInterface, public PeerConnectionSdpMethods { public: virtual rtc::Thread* network_thread() const = 0; virtual rtc::Thread* worker_thread() const = 0; // Returns true if we were the initial offerer. virtual bool initial_offerer() const = 0; virtual std::vector< rtc::scoped_refptr>> GetTransceiversInternal() const = 0; // Call on the network thread to fetch stats for all the data channels. // TODO(tommi): Make pure virtual after downstream updates. virtual std::vector GetDataChannelStats() const { return {}; } virtual absl::optional sctp_transport_name() const = 0; virtual cricket::CandidateStatsList GetPooledCandidateStats() const = 0; // Returns a map from transport name to transport stats for all given // transport names. // Must be called on the network thread. virtual std::map GetTransportStatsByNames(const std::set& transport_names) = 0; virtual Call::Stats GetCallStats() = 0; virtual absl::optional GetAudioDeviceStats() = 0; virtual bool GetLocalCertificate( const std::string& transport_name, rtc::scoped_refptr* certificate) = 0; virtual std::unique_ptr GetRemoteSSLCertChain( const std::string& transport_name) = 0; // Returns true if there was an ICE restart initiated by the remote offer. virtual bool IceRestartPending(const std::string& content_name) const = 0; // Get SSL role for an arbitrary m= section (handles bundling correctly). virtual bool GetSslRole(const std::string& content_name, rtc::SSLRole* role) = 0; // Functions needed by DataChannelController virtual void NoteDataAddedEvent() {} // Handler for sctp data channel state changes. // The `channel_id` is the same unique identifier as used in // `DataChannelStats::internal_id and // `RTCDataChannelStats::data_channel_identifier`. virtual void OnSctpDataChannelStateChanged( int channel_id, DataChannelInterface::DataState state) {} }; } // namespace webrtc #endif // PC_PEER_CONNECTION_INTERNAL_H_