Move talk/app/webrtc to webrtc/api

talk/app/webrtc/webrtcsession.h

Index: talk/app/webrtc/webrtcsession.h
diff --git a/talk/app/webrtc/webrtcsession.h b/talk/app/webrtc/webrtcsession.h
deleted file mode 100644
index 7378736b15fee0bd86b035fd8d4953ab8829b942..0000000000000000000000000000000000000000
--- a/talk/app/webrtc/webrtcsession.h
+++ /dev/null
@@ -1,522 +0,0 @@
-/*
- * libjingle
- * Copyright 2012 Google Inc.
- *
- * Redistribution and use in source and binary forms, with or without
- * modification, are permitted provided that the following conditions are met:
- *
- *  1. Redistributions of source code must retain the above copyright notice,
- *     this list of conditions and the following disclaimer.
- *  2. Redistributions in binary form must reproduce the above copyright notice,
- *     this list of conditions and the following disclaimer in the documentation
- *     and/or other materials provided with the distribution.
- *  3. The name of the author may not be used to endorse or promote products
- *     derived from this software without specific prior written permission.
- *
- * THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED
- * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
- * MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO
- * EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
- * SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
- * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS;
- * OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
- * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR
- * OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF
- * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
- */
-
-#ifndef TALK_APP_WEBRTC_WEBRTCSESSION_H_
-#define TALK_APP_WEBRTC_WEBRTCSESSION_H_
-
-#include <string>
-#include <vector>
-
-#include "talk/app/webrtc/datachannel.h"
-#include "talk/app/webrtc/dtmfsender.h"
-#include "talk/app/webrtc/mediacontroller.h"
-#include "talk/app/webrtc/mediastreamprovider.h"
-#include "talk/app/webrtc/peerconnectioninterface.h"
-#include "talk/app/webrtc/statstypes.h"
-#include "talk/session/media/mediasession.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/transportcontroller.h"
-
-namespace cricket {
-
-class ChannelManager;
-class DataChannel;
-class StatsReport;
-class VideoCapturer;
-class VideoChannel;
-class VoiceChannel;
-
-}  // 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 kDtlsSetupFailureRtp[];
-extern const char kDtlsSetupFailureRtcp[];
-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;
-
-  // 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;
-};
-
-// 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 AudioProviderInterface,
-                      public VideoProviderInterface,
-                      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
-  };
-
-  WebRtcSession(webrtc::MediaControllerInterface* media_controller,
-                rtc::Thread* signaling_thread,
-                rtc::Thread* worker_thread,
-                cricket::PortAllocator* port_allocator);
-  virtual ~WebRtcSession();
-
-  // These are const to allow them to be called from const methods.
-  rtc::Thread* signaling_thread() const { return signaling_thread_; }
-  rtc::Thread* worker_thread() const { return worker_thread_; }
-  cricket::PortAllocator* port_allocator() const { return port_allocator_; }
-
-  // The ID of this session.
-  const std::string& id() const { return sid_; }
-
-  bool Initialize(
-      const PeerConnectionFactoryInterface::Options& options,
-      const MediaConstraintsInterface* constraints,
-      rtc::scoped_ptr<DtlsIdentityStoreInterface> dtls_identity_store,
-      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;
-  }
-
-  virtual cricket::VoiceChannel* voice_channel() {
-    return voice_channel_.get();
-  }
-  virtual cricket::VideoChannel* video_channel() {
-    return video_channel_.get();
-  }
-  virtual cricket::DataChannel* data_channel() {
-    return data_channel_.get();
-  }
-
-  void SetSdesPolicy(cricket::SecurePolicy secure_policy);
-  cricket::SecurePolicy SdesPolicy() const;
-
-  // Get current ssl role from transport.
-  bool GetSslRole(const std::string& transport_name, rtc::SSLRole* role);
-
-  // Get current SSL role for this channel's transport.
-  // If |transport| is null, returns false.
-  bool GetSslRole(const cricket::BaseChannel* channel, rtc::SSLRole* role);
-
-  void CreateOffer(
-      CreateSessionDescriptionObserver* observer,
-      const PeerConnectionInterface::RTCOfferAnswerOptions& options,
-      const cricket::MediaSessionOptions& session_options);
-  void CreateAnswer(CreateSessionDescriptionObserver* observer,
-                    const MediaConstraintsInterface* constraints,
-                    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 SetIceTransports(PeerConnectionInterface::IceTransportsType type);
-
-  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 local_desc_.get();
-  }
-  const SessionDescriptionInterface* remote_description() const {
-    return remote_desc_.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);
-
-  // AudioMediaProviderInterface implementation.
-  void SetAudioPlayout(uint32_t ssrc, bool enable) override;
-  void SetAudioSend(uint32_t ssrc,
-                    bool enable,
-                    const cricket::AudioOptions& options,
-                    cricket::AudioRenderer* renderer) override;
-  void SetAudioPlayoutVolume(uint32_t ssrc, double volume) override;
-  void SetRawAudioSink(uint32_t ssrc,
-                       rtc::scoped_ptr<AudioSinkInterface> sink) override;
-
-  // Implements VideoMediaProviderInterface.
-  bool SetCaptureDevice(uint32_t ssrc, cricket::VideoCapturer* camera) override;
-  void SetVideoPlayout(
-      uint32_t ssrc,
-      bool enable,
-      rtc::VideoSinkInterface<cricket::VideoFrame>* sink) override;
-  void SetVideoSend(uint32_t ssrc,
-                    bool enable,
-                    const cricket::VideoOptions* options) override;
-
-  // Implements DtmfProviderInterface.
-  virtual bool CanInsertDtmf(const std::string& track_id);
-  virtual bool InsertDtmf(const std::string& track_id,
-                          int code, int duration);
-  virtual sigslot::signal0<>* GetOnDestroyedSignal();
-
-  // Implements DataChannelProviderInterface.
-  bool SendData(const cricket::SendDataParams& params,
-                const rtc::Buffer& 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.
-  virtual bool GetTransportStats(SessionStats* stats);
-
-  // Get stats for a specific channel
-  bool GetChannelTransportStats(cricket::BaseChannel* ch, SessionStats* stats);
-
-  // 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 bool GetRemoteSSLCertificate(const std::string& transport_name,
-                                       rtc::SSLCertificate** cert);
-
-  cricket::DataChannelType data_channel_type() const;
-
-  bool IceRestartPending() const;
-
-  void ResetIceRestartLatch();
-
-  // Called when an RTCCertificate is generated or retrieved by
-  // WebRTCSessionDescriptionFactory. Should happen before setLocalDescription.
-  void OnCertificateReady(
-      const rtc::scoped_refptr<rtc::RTCCertificate>& certificate);
-  void OnDtlsSetupFailure(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;
-  }
-
-  // Called when voice_channel_, video_channel_ and data_channel_ 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;
-
- private:
-  // Indicates the type of SessionDescription in a call to SetLocalDescription
-  // and SetRemoteDescription.
-  enum Action {
-    kOffer,
-    kPrAnswer,
-    kAnswer,
-  };
-
-  // 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 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);
-
-  cricket::BaseChannel* GetChannel(const std::string& content_name);
-  // 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);
-  bool CreateVideoChannel(const cricket::ContentInfo* content);
-  bool CreateDataChannel(const cricket::ContentInfo* content);
-
-  // Listens to SCTP CONTROL messages on unused SIDs and process them as OPEN
-  // messages.
-  void OnDataChannelMessageReceived(cricket::DataChannel* channel,
-                                    const cricket::ReceiveDataParams& params,
-                                    const rtc::Buffer& payload);
-
-  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);
-
-  void OnTransportControllerConnectionState(cricket::IceConnectionState state);
-  void OnTransportControllerReceiving(bool receiving);
-  void OnTransportControllerGatheringState(cricket::IceGatheringState state);
-  void OnTransportControllerCandidatesGathered(
-      const std::string& transport_name,
-      const cricket::Candidates& candidates);
-
-  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(cricket::TransportChannel* channel,
-                      const rtc::SentPacket& sent_packet);
-
-  rtc::Thread* const signaling_thread_;
-  rtc::Thread* const worker_thread_;
-  cricket::PortAllocator* const port_allocator_;
-
-  State state_ = STATE_INIT;
-  Error error_ = ERROR_NONE;
-  std::string error_desc_;
-
-  const std::string sid_;
-  bool initial_offerer_ = false;
-
-  rtc::scoped_ptr<cricket::TransportController> transport_controller_;
-  MediaControllerInterface* media_controller_;
-  rtc::scoped_ptr<cricket::VoiceChannel> voice_channel_;
-  rtc::scoped_ptr<cricket::VideoChannel> video_channel_;
-  rtc::scoped_ptr<cricket::DataChannel> data_channel_;
-  cricket::ChannelManager* channel_manager_;
-  IceObserver* ice_observer_;
-  PeerConnectionInterface::IceConnectionState ice_connection_state_;
-  bool ice_connection_receiving_;
-  rtc::scoped_ptr<SessionDescriptionInterface> local_desc_;
-  rtc::scoped_ptr<SessionDescriptionInterface> remote_desc_;
-  // 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).
-  cricket::DataChannelType data_channel_type_;
-  rtc::scoped_ptr<IceRestartAnswerLatch> ice_restart_latch_;
-
-  rtc::scoped_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_;
-
-  RTC_DISALLOW_COPY_AND_ASSIGN(WebRtcSession);
-};
-}  // namespace webrtc
-
-#endif  // TALK_APP_WEBRTC_WEBRTCSESSION_H_