third_party/WebKit/Source/modules/peerconnection/RTCPeerConnection.h

/*
 * Copyright (C) 2012 Google Inc. All rights reserved.
 *
 * 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. Neither the name of Google Inc. nor the names of its contributors
 *    may be used to endorse or promote products derived from this
 *    software without specific prior written permission.
 *
 * THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
 * "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 COPYRIGHT
 * OWNER OR CONTRIBUTORS 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,
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 */

#ifndef RTCPeerConnection_h
#define RTCPeerConnection_h

#include <memory>
#include "bindings/core/v8/ScriptPromise.h"
#include "core/dom/SuspendableObject.h"
#include "modules/EventTargetModules.h"
#include "modules/crypto/NormalizeAlgorithm.h"
#include "modules/mediastream/MediaStream.h"
#include "modules/peerconnection/RTCIceCandidate.h"
#include "platform/AsyncMethodRunner.h"
#include "platform/WebFrameScheduler.h"
#include "platform/bindings/ActiveScriptWrappable.h"
#include "platform/heap/HeapAllocator.h"
#include "public/platform/WebMediaConstraints.h"
#include "public/platform/WebRTCPeerConnectionHandler.h"
#include "public/platform/WebRTCPeerConnectionHandlerClient.h"

namespace blink {
class ExceptionState;
class MediaStreamTrack;
class RTCAnswerOptions;
class RTCConfiguration;
class RTCDTMFSender;
class RTCDataChannel;
class RTCDataChannelInit;
class RTCIceCandidateInitOrRTCIceCandidate;
class RTCOfferOptions;
class RTCPeerConnectionErrorCallback;
class RTCRtpReceiver;
class RTCRtpSender;
class RTCSessionDescription;
class RTCSessionDescriptionCallback;
class RTCSessionDescriptionInit;
class RTCStatsCallback;
class ScriptState;
class VoidCallback;
struct WebRTCConfiguration;

class RTCPeerConnection final : public EventTargetWithInlineData,
                                public WebRTCPeerConnectionHandlerClient,
                                public ActiveScriptWrappable<RTCPeerConnection>,
                                public SuspendableObject {
  DEFINE_WRAPPERTYPEINFO();
  USING_GARBAGE_COLLECTED_MIXIN(RTCPeerConnection);
  USING_PRE_FINALIZER(RTCPeerConnection, Dispose);

 public:
  static RTCPeerConnection* Create(ExecutionContext*,
                                   const RTCConfiguration&,
                                   const Dictionary&,
                                   ExceptionState&);
  ~RTCPeerConnection() override;

  ScriptPromise createOffer(ScriptState*, const RTCOfferOptions&);
  ScriptPromise createOffer(ScriptState*,
                            RTCSessionDescriptionCallback*,
                            RTCPeerConnectionErrorCallback*,
                            const Dictionary&,
                            ExceptionState&);

  ScriptPromise createAnswer(ScriptState*, const RTCAnswerOptions&);
  ScriptPromise createAnswer(ScriptState*,
                             RTCSessionDescriptionCallback*,
                             RTCPeerConnectionErrorCallback*,
                             const Dictionary&);

  ScriptPromise setLocalDescription(ScriptState*,
                                    const RTCSessionDescriptionInit&);
  ScriptPromise setLocalDescription(ScriptState*,
                                    const RTCSessionDescriptionInit&,
                                    VoidCallback*,
                                    RTCPeerConnectionErrorCallback*);
  RTCSessionDescription* localDescription();

  ScriptPromise setRemoteDescription(ScriptState*,
                                     const RTCSessionDescriptionInit&);
  ScriptPromise setRemoteDescription(ScriptState*,
                                     const RTCSessionDescriptionInit&,
                                     VoidCallback*,
                                     RTCPeerConnectionErrorCallback*);
  RTCSessionDescription* remoteDescription();

  String signalingState() const;

  void setConfiguration(ScriptState*, const RTCConfiguration&, ExceptionState&);

  // Certificate management
  // http://w3c.github.io/webrtc-pc/#sec.cert-mgmt
  static ScriptPromise generateCertificate(
      ScriptState*,
      const AlgorithmIdentifier& keygen_algorithm,
      ExceptionState&);

  ScriptPromise addIceCandidate(ScriptState*,
                                const RTCIceCandidateInitOrRTCIceCandidate&);
  ScriptPromise addIceCandidate(ScriptState*,
                                const RTCIceCandidateInitOrRTCIceCandidate&,
                                VoidCallback*,
                                RTCPeerConnectionErrorCallback*);

  String iceGatheringState() const;

  String iceConnectionState() const;

  MediaStreamVector getLocalStreams() const;

  MediaStreamVector getRemoteStreams() const;

  MediaStream* getStreamById(const String& stream_id);

  void addStream(ScriptState*,
                 MediaStream*,
                 const Dictionary& media_constraints,
                 ExceptionState&);

  void removeStream(MediaStream*, ExceptionState&);

  ScriptPromise getStats(ScriptState*,
                         RTCStatsCallback* success_callback,
                         MediaStreamTrack* selector = nullptr);
  ScriptPromise getStats(ScriptState*);

  HeapVector<Member<RTCRtpSender>> getSenders();
  HeapVector<Member<RTCRtpReceiver>> getReceivers();

  RTCDataChannel* createDataChannel(ScriptState*,
                                    String label,
                                    const RTCDataChannelInit&,
                                    ExceptionState&);

  RTCDTMFSender* createDTMFSender(MediaStreamTrack*, ExceptionState&);

  void close(ExceptionState&);

  // We allow getStats after close, but not other calls or callbacks.
  bool ShouldFireDefaultCallbacks() { return !closed_ && !stopped_; }
  bool ShouldFireGetStatsCallback() { return !stopped_; }

  DEFINE_ATTRIBUTE_EVENT_LISTENER(negotiationneeded);
  DEFINE_ATTRIBUTE_EVENT_LISTENER(icecandidate);
  DEFINE_ATTRIBUTE_EVENT_LISTENER(signalingstatechange);
  DEFINE_ATTRIBUTE_EVENT_LISTENER(addstream);
  DEFINE_ATTRIBUTE_EVENT_LISTENER(removestream);
  DEFINE_ATTRIBUTE_EVENT_LISTENER(iceconnectionstatechange);
  DEFINE_ATTRIBUTE_EVENT_LISTENER(icegatheringstatechange);
  DEFINE_ATTRIBUTE_EVENT_LISTENER(datachannel);

  // WebRTCPeerConnectionHandlerClient
  void NegotiationNeeded() override;
  void DidGenerateICECandidate(const WebRTCICECandidate&) override;
  void DidChangeSignalingState(SignalingState) override;
  void DidChangeICEGatheringState(ICEGatheringState) override;
  void DidChangeICEConnectionState(ICEConnectionState) override;
  void DidAddRemoteStream(const WebMediaStream&) override;
  void DidRemoveRemoteStream(const WebMediaStream&) override;
  void DidAddRemoteDataChannel(WebRTCDataChannelHandler*) override;
  void ReleasePeerConnectionHandler() override;
  void ClosePeerConnection() override;

  // EventTarget
  const AtomicString& InterfaceName() const override;
  ExecutionContext* GetExecutionContext() const override;

  // SuspendableObject
  void Suspend() override;
  void Resume() override;
  void ContextDestroyed(ExecutionContext*) override;

  // ScriptWrappable
  // We keep the this object alive until either stopped or closed.
  bool HasPendingActivity() const final { return !closed_ && !stopped_; }

  DECLARE_VIRTUAL_TRACE();

 private:
  typedef Function<bool()> BoolFunction;
  class EventWrapper : public GarbageCollectedFinalized<EventWrapper> {
   public:
    EventWrapper(Event*, std::unique_ptr<BoolFunction>);
    // Returns true if |m_setupFunction| returns true or it is null.
    // |m_event| will only be fired if setup() returns true;
    bool Setup();

    DECLARE_TRACE();

    Member<Event> event_;

   private:
    std::unique_ptr<BoolFunction> setup_function_;
  };

  RTCPeerConnection(ExecutionContext*,
                    const WebRTCConfiguration&,
                    WebMediaConstraints,
                    ExceptionState&);
  void Dispose();

  void ScheduleDispatchEvent(Event*);
  void ScheduleDispatchEvent(Event*, std::unique_ptr<BoolFunction>);
  void DispatchScheduledEvent();
  MediaStreamTrack* GetLocalTrackById(const String& track_id) const;
  MediaStreamTrack* GetRemoteTrackById(const String& track_id) const;
  // Senders and receivers returned by the handler are in use by the peer
  // connection, a sender or receiver that is no longer in use is permanently
  // inactive and does not need to be referenced anymore. These methods removes
  // such senders/receivers from |rtp_senders_|/|rtp_receivers_|.
  void RemoveInactiveSenders();
  void RemoveInactiveReceivers();

  void ChangeSignalingState(WebRTCPeerConnectionHandlerClient::SignalingState);
  void ChangeIceGatheringState(
      WebRTCPeerConnectionHandlerClient::ICEGatheringState);
  // Changes the state immediately; does not fire an event.
  // Returns true if the state was changed.
  bool SetIceConnectionState(
      WebRTCPeerConnectionHandlerClient::ICEConnectionState);
  // Changes the state asynchronously and fires an event immediately after
  // changing the state.
  void ChangeIceConnectionState(
      WebRTCPeerConnectionHandlerClient::ICEConnectionState);

  void CloseInternal();

  void RecordRapporMetrics();

  SignalingState signaling_state_;
  ICEGatheringState ice_gathering_state_;
  ICEConnectionState ice_connection_state_;

  // TODO(hbos): Move away from "addStream" and "removeStream" in favor of
  // "addTrack" and "removeTrack". Update tracks, senders and receivers on
  // relevant events. https://crbug.com/705901
  MediaStreamVector local_streams_;
  MediaStreamVector remote_streams_;
  HeapHashMap<uintptr_t, Member<RTCRtpSender>> rtp_senders_;
  HeapHashMap<uintptr_t, Member<RTCRtpReceiver>> rtp_receivers_;

  std::unique_ptr<WebRTCPeerConnectionHandler> peer_handler_;

  Member<AsyncMethodRunner<RTCPeerConnection>> dispatch_scheduled_event_runner_;
  HeapVector<Member<EventWrapper>> scheduled_events_;

  // This handle notifies scheduler about an active connection associated
  // with a frame. Handle should be destroyed when connection is closed.
  std::unique_ptr<WebFrameScheduler::ActiveConnectionHandle>
      connection_handle_for_scheduler_;

  bool stopped_;
  bool closed_;

  bool has_data_channels_;  // For RAPPOR metrics
};

}  // namespace blink

#endif  // RTCPeerConnection_h