| /* |
| * 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 <map> |
| #include <memory> |
| #include <set> |
| #include <string> |
| #include <vector> |
| |
| #include "api/peer_connection_interface.h" |
| #include "call/call.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<std::string> 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 SdpFormatReceived for the given remote description. |
| virtual void ReportSdpFormatReceived( |
| const SessionDescriptionInterface& remote_description) = 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 bool GetSctpSslRole(rtc::SSLRole* role) = 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<std::string, const cricket::ContentGroup*>& |
| bundle_groups_by_mid) = 0; |
| |
| virtual absl::optional<std::string> GetDataMid() const = 0; |
| // Internal implementation for AddTransceiver family of methods. If |
| // `fire_callback` is set, fires OnRenegotiationNeeded callback if successful. |
| virtual RTCErrorOr<rtc::scoped_refptr<RtpTransceiverInterface>> |
| AddTransceiver(cricket::MediaType media_type, |
| rtc::scoped_refptr<MediaStreamTrackInterface> 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; |
| virtual bool SetupDataChannelTransport_n(const std::string& mid) = 0; |
| virtual void TeardownDataChannelTransport_n() = 0; |
| virtual void SetSctpDataMid(const std::string& mid) = 0; |
| virtual void ResetSctpDataMid() = 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 sigslot::has_slots<> { |
| 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<RtpTransceiverProxyWithInternal<RtpTransceiver>>> |
| GetTransceiversInternal() const = 0; |
| |
| virtual sigslot::signal1<SctpDataChannel*>& |
| SignalSctpDataChannelCreated() = 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<DataChannelStats> GetDataChannelStats() const { |
| return {}; |
| } |
| |
| virtual absl::optional<std::string> 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<std::string, cricket::TransportStats> |
| GetTransportStatsByNames(const std::set<std::string>& transport_names) = 0; |
| |
| virtual Call::Stats GetCallStats() = 0; |
| |
| virtual bool GetLocalCertificate( |
| const std::string& transport_name, |
| rtc::scoped_refptr<rtc::RTCCertificate>* certificate) = 0; |
| virtual std::unique_ptr<rtc::SSLCertChain> 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 the "channel closed" signal |
| virtual void OnSctpDataChannelClosed(DataChannelInterface* channel) {} |
| }; |
| |
| } // namespace webrtc |
| |
| #endif // PC_PEER_CONNECTION_INTERNAL_H_ |