Index: webrtc/api/webrtcsession.h |
diff --git a/webrtc/api/webrtcsession.h b/webrtc/api/webrtcsession.h |
deleted file mode 100644 |
index 0030342ee6332afb61dfa8d4aaa4006a337bcce0..0000000000000000000000000000000000000000 |
--- a/webrtc/api/webrtcsession.h |
+++ /dev/null |
@@ -1,654 +0,0 @@ |
-/* |
- * 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_WEBRTCSESSION_H_ |
-#define WEBRTC_API_WEBRTCSESSION_H_ |
- |
-#include <memory> |
-#include <set> |
-#include <string> |
-#include <vector> |
- |
-#include "webrtc/api/datachannel.h" |
-#include "webrtc/api/dtmfsender.h" |
-#include "webrtc/api/mediacontroller.h" |
-#include "webrtc/api/peerconnectioninterface.h" |
-#include "webrtc/api/statstypes.h" |
-#include "webrtc/base/constructormagic.h" |
-#include "webrtc/base/optional.h" |
-#include "webrtc/base/sigslot.h" |
-#include "webrtc/base/sslidentity.h" |
-#include "webrtc/base/thread.h" |
-#include "webrtc/media/base/mediachannel.h" |
-#include "webrtc/p2p/base/candidate.h" |
-#include "webrtc/p2p/base/transportcontroller.h" |
-#include "webrtc/pc/mediasession.h" |
- |
-#ifdef HAVE_QUIC |
-#include "webrtc/api/quicdatatransport.h" |
-#endif // HAVE_QUIC |
- |
-namespace cricket { |
- |
-class ChannelManager; |
-class RtpDataChannel; |
-class SctpTransportInternal; |
-class SctpTransportInternalFactory; |
-class StatsReport; |
-class VideoChannel; |
-class VoiceChannel; |
- |
-#ifdef HAVE_QUIC |
-class QuicTransportChannel; |
-#endif // HAVE_QUIC |
- |
-} // namespace cricket |
- |
-namespace webrtc { |
- |
-class IceRestartAnswerLatch; |
-class JsepIceCandidate; |
-class MediaStreamSignaling; |
-class WebRtcSessionDescriptionFactory; |
- |
-extern const char kBundleWithoutRtcpMux[]; |
-extern const char kCreateChannelFailed[]; |
-extern const char kInvalidCandidates[]; |
-extern const char kInvalidSdp[]; |
-extern const char kMlineMismatch[]; |
-extern const char kPushDownTDFailed[]; |
-extern const char kSdpWithoutDtlsFingerprint[]; |
-extern const char kSdpWithoutSdesCrypto[]; |
-extern const char kSdpWithoutIceUfragPwd[]; |
-extern const char kSdpWithoutSdesAndDtlsDisabled[]; |
-extern const char kSessionError[]; |
-extern const char kSessionErrorDesc[]; |
-extern const char kDtlsSrtpSetupFailureRtp[]; |
-extern const char kDtlsSrtpSetupFailureRtcp[]; |
-extern const char kEnableBundleFailed[]; |
- |
-// Maximum number of received video streams that will be processed by webrtc |
-// even if they are not signalled beforehand. |
-extern const int kMaxUnsignalledRecvStreams; |
- |
-// ICE state callback interface. |
-class IceObserver { |
- public: |
- IceObserver() {} |
- // Called any time the IceConnectionState changes |
- // TODO(honghaiz): Change the name to OnIceConnectionStateChange so as to |
- // conform to the w3c standard. |
- virtual void OnIceConnectionChange( |
- PeerConnectionInterface::IceConnectionState new_state) {} |
- // Called any time the IceGatheringState changes |
- virtual void OnIceGatheringChange( |
- PeerConnectionInterface::IceGatheringState new_state) {} |
- // New Ice candidate have been found. |
- virtual void OnIceCandidate(const IceCandidateInterface* candidate) = 0; |
- |
- // Some local ICE candidates have been removed. |
- virtual void OnIceCandidatesRemoved( |
- const std::vector<cricket::Candidate>& candidates) = 0; |
- |
- // Called whenever the state changes between receiving and not receiving. |
- virtual void OnIceConnectionReceivingChange(bool receiving) {} |
- |
- protected: |
- ~IceObserver() {} |
- |
- private: |
- RTC_DISALLOW_COPY_AND_ASSIGN(IceObserver); |
-}; |
- |
-// Statistics for all the transports of the session. |
-typedef std::map<std::string, cricket::TransportStats> TransportStatsMap; |
-typedef std::map<std::string, std::string> ProxyTransportMap; |
- |
-// TODO(pthatcher): Think of a better name for this. We already have |
-// a TransportStats in transport.h. Perhaps TransportsStats? |
-struct SessionStats { |
- ProxyTransportMap proxy_to_transport; |
- TransportStatsMap transport_stats; |
-}; |
- |
-struct ChannelNamePair { |
- ChannelNamePair( |
- const std::string& content_name, const std::string& transport_name) |
- : content_name(content_name), transport_name(transport_name) {} |
- std::string content_name; |
- std::string transport_name; |
-}; |
- |
-struct ChannelNamePairs { |
- rtc::Optional<ChannelNamePair> voice; |
- rtc::Optional<ChannelNamePair> video; |
- rtc::Optional<ChannelNamePair> data; |
-}; |
- |
-// A WebRtcSession manages general session state. This includes negotiation |
-// of both the application-level and network-level protocols: the former |
-// defines what will be sent and the latter defines how it will be sent. Each |
-// network-level protocol is represented by a Transport object. Each Transport |
-// participates in the network-level negotiation. The individual streams of |
-// packets are represented by TransportChannels. The application-level protocol |
-// is represented by SessionDecription objects. |
-class WebRtcSession : |
- |
- public DtmfProviderInterface, |
- public DataChannelProviderInterface, |
- public sigslot::has_slots<> { |
- public: |
- enum State { |
- STATE_INIT = 0, |
- STATE_SENTOFFER, // Sent offer, waiting for answer. |
- STATE_RECEIVEDOFFER, // Received an offer. Need to send answer. |
- STATE_SENTPRANSWER, // Sent provisional answer. Need to send answer. |
- STATE_RECEIVEDPRANSWER, // Received provisional answer, waiting for answer. |
- STATE_INPROGRESS, // Offer/answer exchange completed. |
- STATE_CLOSED, // Close() was called. |
- }; |
- |
- enum Error { |
- ERROR_NONE = 0, // no error |
- ERROR_CONTENT = 1, // channel errors in SetLocalContent/SetRemoteContent |
- ERROR_TRANSPORT = 2, // transport error of some kind |
- }; |
- |
- // |sctp_factory| may be null, in which case SCTP is treated as unsupported. |
- WebRtcSession( |
- webrtc::MediaControllerInterface* media_controller, |
- rtc::Thread* network_thread, |
- rtc::Thread* worker_thread, |
- rtc::Thread* signaling_thread, |
- cricket::PortAllocator* port_allocator, |
- std::unique_ptr<cricket::TransportController> transport_controller, |
- std::unique_ptr<cricket::SctpTransportInternalFactory> sctp_factory); |
- virtual ~WebRtcSession(); |
- |
- // These are const to allow them to be called from const methods. |
- rtc::Thread* network_thread() const { return network_thread_; } |
- rtc::Thread* worker_thread() const { return worker_thread_; } |
- rtc::Thread* signaling_thread() const { return signaling_thread_; } |
- |
- // The ID of this session. |
- const std::string& id() const { return sid_; } |
- |
- bool Initialize( |
- const PeerConnectionFactoryInterface::Options& options, |
- std::unique_ptr<rtc::RTCCertificateGeneratorInterface> cert_generator, |
- const PeerConnectionInterface::RTCConfiguration& rtc_configuration); |
- // Deletes the voice, video and data channel and changes the session state |
- // to STATE_CLOSED. |
- void Close(); |
- |
- // Returns true if we were the initial offerer. |
- bool initial_offerer() const { return initial_offerer_; } |
- |
- // Returns the current state of the session. See the enum above for details. |
- // Each time the state changes, we will fire this signal. |
- State state() const { return state_; } |
- sigslot::signal2<WebRtcSession*, State> SignalState; |
- |
- // Returns the last error in the session. See the enum above for details. |
- Error error() const { return error_; } |
- const std::string& error_desc() const { return error_desc_; } |
- |
- void RegisterIceObserver(IceObserver* observer) { |
- ice_observer_ = observer; |
- } |
- |
- // Exposed for stats collecting. |
- virtual cricket::VoiceChannel* voice_channel() { |
- return voice_channel_.get(); |
- } |
- virtual cricket::VideoChannel* video_channel() { |
- return video_channel_.get(); |
- } |
- // Only valid when using deprecated RTP data channels. |
- virtual cricket::RtpDataChannel* rtp_data_channel() { |
- return rtp_data_channel_.get(); |
- } |
- virtual rtc::Optional<std::string> sctp_content_name() const { |
- return sctp_content_name_; |
- } |
- virtual rtc::Optional<std::string> sctp_transport_name() const { |
- return sctp_transport_name_; |
- } |
- |
- cricket::BaseChannel* GetChannel(const std::string& content_name); |
- |
- cricket::SecurePolicy SdesPolicy() const; |
- |
- // Get current SSL role used by SCTP's underlying transport. |
- bool GetSctpSslRole(rtc::SSLRole* role); |
- // Get SSL role for an arbitrary m= section (handles bundling correctly). |
- // TODO(deadbeef): This is only used internally by the session description |
- // factory, it shouldn't really be public). |
- bool GetSslRole(const std::string& content_name, rtc::SSLRole* role); |
- |
- void CreateOffer( |
- CreateSessionDescriptionObserver* observer, |
- const PeerConnectionInterface::RTCOfferAnswerOptions& options, |
- const cricket::MediaSessionOptions& session_options); |
- void CreateAnswer(CreateSessionDescriptionObserver* observer, |
- const cricket::MediaSessionOptions& session_options); |
- // The ownership of |desc| will be transferred after this call. |
- bool SetLocalDescription(SessionDescriptionInterface* desc, |
- std::string* err_desc); |
- // The ownership of |desc| will be transferred after this call. |
- bool SetRemoteDescription(SessionDescriptionInterface* desc, |
- std::string* err_desc); |
- |
- bool ProcessIceMessage(const IceCandidateInterface* ice_candidate); |
- |
- bool RemoveRemoteIceCandidates( |
- const std::vector<cricket::Candidate>& candidates); |
- |
- cricket::IceConfig ParseIceConfig( |
- const PeerConnectionInterface::RTCConfiguration& config) const; |
- |
- void SetIceConfig(const cricket::IceConfig& ice_config); |
- |
- // Start gathering candidates for any new transports, or transports doing an |
- // ICE restart. |
- void MaybeStartGathering(); |
- |
- const SessionDescriptionInterface* local_description() const { |
- return pending_local_description_ ? pending_local_description_.get() |
- : current_local_description_.get(); |
- } |
- const SessionDescriptionInterface* remote_description() const { |
- return pending_remote_description_ ? pending_remote_description_.get() |
- : current_remote_description_.get(); |
- } |
- const SessionDescriptionInterface* current_local_description() const { |
- return current_local_description_.get(); |
- } |
- const SessionDescriptionInterface* current_remote_description() const { |
- return current_remote_description_.get(); |
- } |
- const SessionDescriptionInterface* pending_local_description() const { |
- return pending_local_description_.get(); |
- } |
- const SessionDescriptionInterface* pending_remote_description() const { |
- return pending_remote_description_.get(); |
- } |
- |
- // Get the id used as a media stream track's "id" field from ssrc. |
- virtual bool GetLocalTrackIdBySsrc(uint32_t ssrc, std::string* track_id); |
- virtual bool GetRemoteTrackIdBySsrc(uint32_t ssrc, std::string* track_id); |
- |
- // Implements DtmfProviderInterface. |
- bool CanInsertDtmf(const std::string& track_id) override; |
- bool InsertDtmf(const std::string& track_id, |
- int code, int duration) override; |
- sigslot::signal0<>* GetOnDestroyedSignal() override; |
- |
- // Implements DataChannelProviderInterface. |
- bool SendData(const cricket::SendDataParams& params, |
- const rtc::CopyOnWriteBuffer& payload, |
- cricket::SendDataResult* result) override; |
- bool ConnectDataChannel(DataChannel* webrtc_data_channel) override; |
- void DisconnectDataChannel(DataChannel* webrtc_data_channel) override; |
- void AddSctpDataStream(int sid) override; |
- void RemoveSctpDataStream(int sid) override; |
- bool ReadyToSendData() const override; |
- |
- // Returns stats for all channels of all transports. |
- // This avoids exposing the internal structures used to track them. |
- // The parameterless version creates |ChannelNamePairs| from |voice_channel|, |
- // |video_channel| and |voice_channel| if available - this requires it to be |
- // called on the signaling thread - and invokes the other |GetStats|. The |
- // other |GetStats| can be invoked on any thread; if not invoked on the |
- // network thread a thread hop will happen. |
- std::unique_ptr<SessionStats> GetStats_s(); |
- virtual std::unique_ptr<SessionStats> GetStats( |
- const ChannelNamePairs& channel_name_pairs); |
- |
- // virtual so it can be mocked in unit tests |
- virtual bool GetLocalCertificate( |
- const std::string& transport_name, |
- rtc::scoped_refptr<rtc::RTCCertificate>* certificate); |
- |
- // Caller owns returned certificate |
- virtual std::unique_ptr<rtc::SSLCertificate> GetRemoteSSLCertificate( |
- const std::string& transport_name); |
- |
- cricket::DataChannelType data_channel_type() const; |
- |
- // Returns true if there was an ICE restart initiated by the remote offer. |
- bool IceRestartPending(const std::string& content_name) const; |
- |
- // Set the "needs-ice-restart" flag as described in JSEP. After the flag is |
- // set, offers should generate new ufrags/passwords until an ICE restart |
- // occurs. |
- void SetNeedsIceRestartFlag(); |
- // 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. |
- bool NeedsIceRestart(const std::string& content_name) const; |
- |
- // Called when an RTCCertificate is generated or retrieved by |
- // WebRTCSessionDescriptionFactory. Should happen before setLocalDescription. |
- void OnCertificateReady( |
- const rtc::scoped_refptr<rtc::RTCCertificate>& certificate); |
- void OnDtlsSrtpSetupFailure(cricket::BaseChannel*, bool rtcp); |
- |
- // For unit test. |
- bool waiting_for_certificate_for_testing() const; |
- const rtc::scoped_refptr<rtc::RTCCertificate>& certificate_for_testing(); |
- |
- void set_metrics_observer( |
- webrtc::MetricsObserverInterface* metrics_observer) { |
- metrics_observer_ = metrics_observer; |
- transport_controller_->SetMetricsObserver(metrics_observer); |
- } |
- |
- // Called when voice_channel_, video_channel_ and |
- // rtp_data_channel_/sctp_transport_ are created and destroyed. As a result |
- // of, for example, setting a new description. |
- sigslot::signal0<> SignalVoiceChannelCreated; |
- sigslot::signal0<> SignalVoiceChannelDestroyed; |
- sigslot::signal0<> SignalVideoChannelCreated; |
- sigslot::signal0<> SignalVideoChannelDestroyed; |
- sigslot::signal0<> SignalDataChannelCreated; |
- sigslot::signal0<> SignalDataChannelDestroyed; |
- // Called when the whole session is destroyed. |
- sigslot::signal0<> SignalDestroyed; |
- |
- // Called when a valid data channel OPEN message is received. |
- // std::string represents the data channel label. |
- sigslot::signal2<const std::string&, const InternalDataChannelInit&> |
- SignalDataChannelOpenMessage; |
-#ifdef HAVE_QUIC |
- QuicDataTransport* quic_data_transport() { |
- return quic_data_transport_.get(); |
- } |
-#endif // HAVE_QUIC |
- |
- private: |
- // Indicates the type of SessionDescription in a call to SetLocalDescription |
- // and SetRemoteDescription. |
- enum Action { |
- kOffer, |
- kPrAnswer, |
- kAnswer, |
- }; |
- |
- // Non-const versions of local_description()/remote_description(), for use |
- // internally. |
- SessionDescriptionInterface* mutable_local_description() { |
- return pending_local_description_ ? pending_local_description_.get() |
- : current_local_description_.get(); |
- } |
- SessionDescriptionInterface* mutable_remote_description() { |
- return pending_remote_description_ ? pending_remote_description_.get() |
- : current_remote_description_.get(); |
- } |
- |
- // Log session state. |
- void LogState(State old_state, State new_state); |
- |
- // Updates the state, signaling if necessary. |
- virtual void SetState(State state); |
- |
- // Updates the error state, signaling if necessary. |
- // TODO(ronghuawu): remove the SetError method that doesn't take |error_desc|. |
- virtual void SetError(Error error, const std::string& error_desc); |
- |
- bool UpdateSessionState(Action action, cricket::ContentSource source, |
- std::string* err_desc); |
- static Action GetAction(const std::string& type); |
- // Push the media parts of the local or remote session description |
- // down to all of the channels. |
- bool PushdownMediaDescription(cricket::ContentAction action, |
- cricket::ContentSource source, |
- std::string* error_desc); |
- bool PushdownSctpParameters_n(cricket::ContentSource source); |
- |
- bool PushdownTransportDescription(cricket::ContentSource source, |
- cricket::ContentAction action, |
- std::string* error_desc); |
- |
- // Helper methods to push local and remote transport descriptions. |
- bool PushdownLocalTransportDescription( |
- const cricket::SessionDescription* sdesc, |
- cricket::ContentAction action, |
- std::string* error_desc); |
- bool PushdownRemoteTransportDescription( |
- const cricket::SessionDescription* sdesc, |
- cricket::ContentAction action, |
- std::string* error_desc); |
- |
- // 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 name of the transport channel when BUNDLE is enabled, or |
- // nullptr if the channel is not part of any bundle. |
- const std::string* GetBundleTransportName( |
- const cricket::ContentInfo* content, |
- const cricket::ContentGroup* bundle); |
- |
- // Cause all the BaseChannels in the bundle group to have the same |
- // transport channel. |
- bool EnableBundle(const cricket::ContentGroup& bundle); |
- |
- // Enables media channels to allow sending of media. |
- void EnableChannels(); |
- // 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); |
- // Uses all remote candidates in |remote_desc| in this session. |
- bool UseCandidatesInSessionDescription( |
- const SessionDescriptionInterface* remote_desc); |
- // Uses |candidate| in this session. |
- bool UseCandidate(const IceCandidateInterface* candidate); |
- // Deletes the corresponding channel of contents that don't exist in |desc|. |
- // |desc| can be null. This means that all channels are deleted. |
- void RemoveUnusedChannels(const cricket::SessionDescription* desc); |
- |
- // Allocates media channels based on the |desc|. If |desc| doesn't have |
- // the BUNDLE option, this method will disable BUNDLE in PortAllocator. |
- // This method will also delete any existing media channels before creating. |
- bool CreateChannels(const cricket::SessionDescription* desc); |
- |
- // Helper methods to create media channels. |
- bool CreateVoiceChannel(const cricket::ContentInfo* content, |
- const std::string* bundle_transport); |
- bool CreateVideoChannel(const cricket::ContentInfo* content, |
- const std::string* bundle_transport); |
- bool CreateDataChannel(const cricket::ContentInfo* content, |
- const std::string* bundle_transport); |
- |
- std::unique_ptr<SessionStats> GetStats_n( |
- const ChannelNamePairs& channel_name_pairs); |
- |
- bool CreateSctpTransport_n(const std::string& content_name, |
- const std::string& transport_name); |
- // For bundling. |
- void ChangeSctpTransport_n(const std::string& transport_name); |
- void DestroySctpTransport_n(); |
- // SctpTransport signal handlers. Needed to marshal signals from the network |
- // to signaling thread. |
- void OnSctpTransportReadyToSendData_n(); |
- // This may be called with "false" if the direction of the m= section causes |
- // us to tear down the SCTP connection. |
- void OnSctpTransportReadyToSendData_s(bool ready); |
- void OnSctpTransportDataReceived_n(const cricket::ReceiveDataParams& params, |
- const rtc::CopyOnWriteBuffer& payload); |
- // Beyond just firing the signal to the signaling thread, listens to SCTP |
- // CONTROL messages on unused SIDs and processes them as OPEN messages. |
- void OnSctpTransportDataReceived_s(const cricket::ReceiveDataParams& params, |
- const rtc::CopyOnWriteBuffer& payload); |
- void OnSctpStreamClosedRemotely_n(int sid); |
- |
- std::string BadStateErrMsg(State state); |
- void SetIceConnectionState(PeerConnectionInterface::IceConnectionState state); |
- void SetIceConnectionReceiving(bool receiving); |
- |
- bool ValidateBundleSettings(const cricket::SessionDescription* desc); |
- bool HasRtcpMuxEnabled(const cricket::ContentInfo* content); |
- // Below methods are helper methods which verifies SDP. |
- bool ValidateSessionDescription(const SessionDescriptionInterface* sdesc, |
- cricket::ContentSource source, |
- std::string* err_desc); |
- |
- // Check if a call to SetLocalDescription is acceptable with |action|. |
- bool ExpectSetLocalDescription(Action action); |
- // Check if a call to SetRemoteDescription is acceptable with |action|. |
- bool ExpectSetRemoteDescription(Action action); |
- // Verifies a=setup attribute as per RFC 5763. |
- bool ValidateDtlsSetupAttribute(const cricket::SessionDescription* desc, |
- Action action); |
- |
- // Returns true if we are ready to push down the remote candidate. |
- // |remote_desc| is the new remote description, or NULL if the current remote |
- // description should be used. Output |valid| is true if the candidate media |
- // index is valid. |
- bool ReadyToUseRemoteCandidate(const IceCandidateInterface* candidate, |
- const SessionDescriptionInterface* remote_desc, |
- bool* valid); |
- |
- // Returns true if SRTP (either using DTLS-SRTP or SDES) is required by |
- // this session. |
- bool SrtpRequired() const; |
- |
- // TransportController signal handlers. |
- void OnTransportControllerConnectionState(cricket::IceConnectionState state); |
- void OnTransportControllerReceiving(bool receiving); |
- void OnTransportControllerGatheringState(cricket::IceGatheringState state); |
- void OnTransportControllerCandidatesGathered( |
- const std::string& transport_name, |
- const std::vector<cricket::Candidate>& candidates); |
- void OnTransportControllerCandidatesRemoved( |
- const std::vector<cricket::Candidate>& candidates); |
- void OnTransportControllerDtlsHandshakeError(rtc::SSLHandshakeError error); |
- |
- std::string GetSessionErrorMsg(); |
- |
- // Invoked when TransportController connection completion is signaled. |
- // Reports stats for all transports in use. |
- void ReportTransportStats(); |
- |
- // Gather the usage of IPv4/IPv6 as best connection. |
- void ReportBestConnectionState(const cricket::TransportStats& stats); |
- |
- void ReportNegotiatedCiphers(const cricket::TransportStats& stats); |
- |
- void OnSentPacket_w(const rtc::SentPacket& sent_packet); |
- |
- const std::string GetTransportName(const std::string& content_name); |
- |
- void DestroyRtcpTransport_n(const std::string& transport_name); |
- void DestroyVideoChannel(); |
- void DestroyVoiceChannel(); |
- void DestroyDataChannel(); |
- |
- rtc::Thread* const network_thread_; |
- rtc::Thread* const worker_thread_; |
- rtc::Thread* const signaling_thread_; |
- |
- State state_ = STATE_INIT; |
- Error error_ = ERROR_NONE; |
- std::string error_desc_; |
- |
- const std::string sid_; |
- bool initial_offerer_ = false; |
- |
- const std::unique_ptr<cricket::TransportController> transport_controller_; |
- const std::unique_ptr<cricket::SctpTransportInternalFactory> sctp_factory_; |
- MediaControllerInterface* media_controller_; |
- std::unique_ptr<cricket::VoiceChannel> voice_channel_; |
- std::unique_ptr<cricket::VideoChannel> video_channel_; |
- // |rtp_data_channel_| is used if in RTP data channel mode, |sctp_transport_| |
- // when using SCTP. |
- std::unique_ptr<cricket::RtpDataChannel> rtp_data_channel_; |
- |
- std::unique_ptr<cricket::SctpTransportInternal> sctp_transport_; |
- // |sctp_transport_name_| keeps track of what DTLS transport the SCTP |
- // transport is using (which can change due to bundling). |
- rtc::Optional<std::string> sctp_transport_name_; |
- // |sctp_content_name_| is the content name (MID) in SDP. |
- rtc::Optional<std::string> sctp_content_name_; |
- // Value cached on signaling thread. Only updated when SctpReadyToSendData |
- // fires on the signaling thread. |
- bool sctp_ready_to_send_data_ = false; |
- // Same as signals provided by SctpTransport, but these are guaranteed to |
- // fire on the signaling thread, whereas SctpTransport fires on the networking |
- // thread. |
- // |sctp_invoker_| is used so that any signals queued on the signaling thread |
- // from the network thread are immediately discarded if the SctpTransport is |
- // destroyed (due to m= section being rejected). |
- // TODO(deadbeef): Use a proxy object to ensure that method calls/signals |
- // are marshalled to the right thread. Could almost use proxy.h for this, |
- // but it doesn't have a mechanism for marshalling sigslot::signals |
- std::unique_ptr<rtc::AsyncInvoker> sctp_invoker_; |
- sigslot::signal1<bool> SignalSctpReadyToSendData; |
- sigslot::signal2<const cricket::ReceiveDataParams&, |
- const rtc::CopyOnWriteBuffer&> |
- SignalSctpDataReceived; |
- sigslot::signal1<int> SignalSctpStreamClosedRemotely; |
- |
- cricket::ChannelManager* channel_manager_; |
- IceObserver* ice_observer_; |
- PeerConnectionInterface::IceConnectionState ice_connection_state_; |
- bool ice_connection_receiving_; |
- std::unique_ptr<SessionDescriptionInterface> current_local_description_; |
- std::unique_ptr<SessionDescriptionInterface> pending_local_description_; |
- std::unique_ptr<SessionDescriptionInterface> current_remote_description_; |
- std::unique_ptr<SessionDescriptionInterface> pending_remote_description_; |
- // If the remote peer is using a older version of implementation. |
- bool older_version_remote_peer_; |
- bool dtls_enabled_; |
- // Specifies which kind of data channel is allowed. This is controlled |
- // by the chrome command-line flag and constraints: |
- // 1. If chrome command-line switch 'enable-sctp-data-channels' is enabled, |
- // constraint kEnableDtlsSrtp is true, and constaint kEnableRtpDataChannels is |
- // not set or false, SCTP is allowed (DCT_SCTP); |
- // 2. If constraint kEnableRtpDataChannels is true, RTP is allowed (DCT_RTP); |
- // 3. If both 1&2 are false, data channel is not allowed (DCT_NONE). |
- // The data channel type could be DCT_QUIC if the QUIC data channel is |
- // enabled. |
- cricket::DataChannelType data_channel_type_; |
- // List of content names for which the remote side triggered an ICE restart. |
- std::set<std::string> pending_ice_restarts_; |
- |
- std::unique_ptr<WebRtcSessionDescriptionFactory> webrtc_session_desc_factory_; |
- |
- // Member variables for caching global options. |
- cricket::AudioOptions audio_options_; |
- cricket::VideoOptions video_options_; |
- MetricsObserverInterface* metrics_observer_; |
- |
- // Declares the bundle policy for the WebRTCSession. |
- PeerConnectionInterface::BundlePolicy bundle_policy_; |
- |
- // Declares the RTCP mux policy for the WebRTCSession. |
- PeerConnectionInterface::RtcpMuxPolicy rtcp_mux_policy_; |
- |
- bool received_first_video_packet_ = false; |
- bool received_first_audio_packet_ = false; |
- |
-#ifdef HAVE_QUIC |
- std::unique_ptr<QuicDataTransport> quic_data_transport_; |
-#endif // HAVE_QUIC |
- |
- RTC_DISALLOW_COPY_AND_ASSIGN(WebRtcSession); |
-}; |
-} // namespace webrtc |
- |
-#endif // WEBRTC_API_WEBRTCSESSION_H_ |