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      1 /*
      2  * libjingle
      3  * Copyright 2012, Google Inc.
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      5  * Redistribution and use in source and binary forms, with or without
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     27 
     28 // This file contains the PeerConnection interface as defined in
     29 // http://dev.w3.org/2011/webrtc/editor/webrtc.html#peer-to-peer-connections.
     30 // Applications must use this interface to implement peerconnection.
     31 // PeerConnectionFactory class provides factory methods to create
     32 // peerconnection, mediastream and media tracks objects.
     33 //
     34 // The Following steps are needed to setup a typical call using Jsep.
     35 // 1. Create a PeerConnectionFactoryInterface. Check constructors for more
     36 // information about input parameters.
     37 // 2. Create a PeerConnection object. Provide a configuration string which
     38 // points either to stun or turn server to generate ICE candidates and provide
     39 // an object that implements the PeerConnectionObserver interface.
     40 // 3. Create local MediaStream and MediaTracks using the PeerConnectionFactory
     41 // and add it to PeerConnection by calling AddStream.
     42 // 4. Create an offer and serialize it and send it to the remote peer.
     43 // 5. Once an ice candidate have been found PeerConnection will call the
     44 // observer function OnIceCandidate. The candidates must also be serialized and
     45 // sent to the remote peer.
     46 // 6. Once an answer is received from the remote peer, call
     47 // SetLocalSessionDescription with the offer and SetRemoteSessionDescription
     48 // with the remote answer.
     49 // 7. Once a remote candidate is received from the remote peer, provide it to
     50 // the peerconnection by calling AddIceCandidate.
     51 
     52 
     53 // The Receiver of a call can decide to accept or reject the call.
     54 // This decision will be taken by the application not peerconnection.
     55 // If application decides to accept the call
     56 // 1. Create PeerConnectionFactoryInterface if it doesn't exist.
     57 // 2. Create a new PeerConnection.
     58 // 3. Provide the remote offer to the new PeerConnection object by calling
     59 // SetRemoteSessionDescription.
     60 // 4. Generate an answer to the remote offer by calling CreateAnswer and send it
     61 // back to the remote peer.
     62 // 5. Provide the local answer to the new PeerConnection by calling
     63 // SetLocalSessionDescription with the answer.
     64 // 6. Provide the remote ice candidates by calling AddIceCandidate.
     65 // 7. Once a candidate have been found PeerConnection will call the observer
     66 // function OnIceCandidate. Send these candidates to the remote peer.
     67 
     68 #ifndef TALK_APP_WEBRTC_PEERCONNECTIONINTERFACE_H_
     69 #define TALK_APP_WEBRTC_PEERCONNECTIONINTERFACE_H_
     70 
     71 #include <string>
     72 #include <vector>
     73 
     74 #include "talk/app/webrtc/datachannelinterface.h"
     75 #include "talk/app/webrtc/dtmfsenderinterface.h"
     76 #include "talk/app/webrtc/jsep.h"
     77 #include "talk/app/webrtc/mediastreaminterface.h"
     78 #include "talk/app/webrtc/statstypes.h"
     79 #include "talk/app/webrtc/umametrics.h"
     80 #include "webrtc/base/fileutils.h"
     81 #include "webrtc/base/socketaddress.h"
     82 
     83 namespace rtc {
     84 class Thread;
     85 }
     86 
     87 namespace cricket {
     88 class PortAllocator;
     89 class WebRtcVideoDecoderFactory;
     90 class WebRtcVideoEncoderFactory;
     91 }
     92 
     93 namespace webrtc {
     94 class AudioDeviceModule;
     95 class MediaConstraintsInterface;
     96 
     97 // MediaStream container interface.
     98 class StreamCollectionInterface : public rtc::RefCountInterface {
     99  public:
    100   // TODO(ronghuawu): Update the function names to c++ style, e.g. find -> Find.
    101   virtual size_t count() = 0;
    102   virtual MediaStreamInterface* at(size_t index) = 0;
    103   virtual MediaStreamInterface* find(const std::string& label) = 0;
    104   virtual MediaStreamTrackInterface* FindAudioTrack(
    105       const std::string& id) = 0;
    106   virtual MediaStreamTrackInterface* FindVideoTrack(
    107       const std::string& id) = 0;
    108 
    109  protected:
    110   // Dtor protected as objects shouldn't be deleted via this interface.
    111   ~StreamCollectionInterface() {}
    112 };
    113 
    114 class StatsObserver : public rtc::RefCountInterface {
    115  public:
    116   // TODO(tommi): Remove.
    117   virtual void OnComplete(const std::vector<StatsReport>& reports) {}
    118 
    119   // TODO(tommi): Make pure virtual and remove implementation.
    120   virtual void OnComplete(const StatsReports& reports) {
    121     std::vector<StatsReportCopyable> report_copies;
    122     for (size_t i = 0; i < reports.size(); ++i)
    123       report_copies.push_back(StatsReportCopyable(*reports[i]));
    124     std::vector<StatsReport>* r =
    125         reinterpret_cast<std::vector<StatsReport>*>(&report_copies);
    126      OnComplete(*r);
    127    }
    128 
    129  protected:
    130   virtual ~StatsObserver() {}
    131 };
    132 
    133 class UMAObserver : public rtc::RefCountInterface {
    134  public:
    135   virtual void IncrementCounter(PeerConnectionUMAMetricsCounter type) = 0;
    136   virtual void AddHistogramSample(PeerConnectionUMAMetricsName type,
    137                                   int value) = 0;
    138 
    139  protected:
    140   virtual ~UMAObserver() {}
    141 };
    142 
    143 class PeerConnectionInterface : public rtc::RefCountInterface {
    144  public:
    145   // See http://dev.w3.org/2011/webrtc/editor/webrtc.html#state-definitions .
    146   enum SignalingState {
    147     kStable,
    148     kHaveLocalOffer,
    149     kHaveLocalPrAnswer,
    150     kHaveRemoteOffer,
    151     kHaveRemotePrAnswer,
    152     kClosed,
    153   };
    154 
    155   // TODO(bemasc): Remove IceState when callers are changed to
    156   // IceConnection/GatheringState.
    157   enum IceState {
    158     kIceNew,
    159     kIceGathering,
    160     kIceWaiting,
    161     kIceChecking,
    162     kIceConnected,
    163     kIceCompleted,
    164     kIceFailed,
    165     kIceClosed,
    166   };
    167 
    168   enum IceGatheringState {
    169     kIceGatheringNew,
    170     kIceGatheringGathering,
    171     kIceGatheringComplete
    172   };
    173 
    174   enum IceConnectionState {
    175     kIceConnectionNew,
    176     kIceConnectionChecking,
    177     kIceConnectionConnected,
    178     kIceConnectionCompleted,
    179     kIceConnectionFailed,
    180     kIceConnectionDisconnected,
    181     kIceConnectionClosed,
    182   };
    183 
    184   struct IceServer {
    185     std::string uri;
    186     std::string username;
    187     std::string password;
    188   };
    189   typedef std::vector<IceServer> IceServers;
    190 
    191   enum IceTransportsType {
    192     kNone,
    193     kRelay,
    194     kNoHost,
    195     kAll
    196   };
    197 
    198   struct RTCConfiguration {
    199     IceTransportsType type;
    200     IceServers servers;
    201 
    202     RTCConfiguration() : type(kAll) {}
    203     explicit RTCConfiguration(IceTransportsType type) : type(type) {}
    204   };
    205 
    206   struct RTCOfferAnswerOptions {
    207     static const int kUndefined = -1;
    208     static const int kMaxOfferToReceiveMedia = 1;
    209 
    210     // The default value for constraint offerToReceiveX:true.
    211     static const int kOfferToReceiveMediaTrue = 1;
    212 
    213     int offer_to_receive_video;
    214     int offer_to_receive_audio;
    215     bool voice_activity_detection;
    216     bool ice_restart;
    217     bool use_rtp_mux;
    218 
    219     RTCOfferAnswerOptions()
    220         : offer_to_receive_video(kUndefined),
    221           offer_to_receive_audio(kUndefined),
    222           voice_activity_detection(true),
    223           ice_restart(false),
    224           use_rtp_mux(true) {}
    225 
    226     RTCOfferAnswerOptions(int offer_to_receive_video,
    227                           int offer_to_receive_audio,
    228                           bool voice_activity_detection,
    229                           bool ice_restart,
    230                           bool use_rtp_mux)
    231         : offer_to_receive_video(offer_to_receive_video),
    232           offer_to_receive_audio(offer_to_receive_audio),
    233           voice_activity_detection(voice_activity_detection),
    234           ice_restart(ice_restart),
    235           use_rtp_mux(use_rtp_mux) {}
    236   };
    237 
    238   // Used by GetStats to decide which stats to include in the stats reports.
    239   // |kStatsOutputLevelStandard| includes the standard stats for Javascript API;
    240   // |kStatsOutputLevelDebug| includes both the standard stats and additional
    241   // stats for debugging purposes.
    242   enum StatsOutputLevel {
    243     kStatsOutputLevelStandard,
    244     kStatsOutputLevelDebug,
    245   };
    246 
    247   // Accessor methods to active local streams.
    248   virtual rtc::scoped_refptr<StreamCollectionInterface>
    249       local_streams() = 0;
    250 
    251   // Accessor methods to remote streams.
    252   virtual rtc::scoped_refptr<StreamCollectionInterface>
    253       remote_streams() = 0;
    254 
    255   // Add a new MediaStream to be sent on this PeerConnection.
    256   // Note that a SessionDescription negotiation is needed before the
    257   // remote peer can receive the stream.
    258   virtual bool AddStream(MediaStreamInterface* stream,
    259                          const MediaConstraintsInterface* constraints) = 0;
    260 
    261   // Remove a MediaStream from this PeerConnection.
    262   // Note that a SessionDescription negotiation is need before the
    263   // remote peer is notified.
    264   virtual void RemoveStream(MediaStreamInterface* stream) = 0;
    265 
    266   // Returns pointer to the created DtmfSender on success.
    267   // Otherwise returns NULL.
    268   virtual rtc::scoped_refptr<DtmfSenderInterface> CreateDtmfSender(
    269       AudioTrackInterface* track) = 0;
    270 
    271   virtual bool GetStats(StatsObserver* observer,
    272                         MediaStreamTrackInterface* track,
    273                         StatsOutputLevel level) = 0;
    274 
    275   virtual rtc::scoped_refptr<DataChannelInterface> CreateDataChannel(
    276       const std::string& label,
    277       const DataChannelInit* config) = 0;
    278 
    279   virtual const SessionDescriptionInterface* local_description() const = 0;
    280   virtual const SessionDescriptionInterface* remote_description() const = 0;
    281 
    282   // Create a new offer.
    283   // The CreateSessionDescriptionObserver callback will be called when done.
    284   virtual void CreateOffer(CreateSessionDescriptionObserver* observer,
    285                            const MediaConstraintsInterface* constraints) {}
    286 
    287   // TODO(jiayl): remove the default impl and the old interface when chromium
    288   // code is updated.
    289   virtual void CreateOffer(CreateSessionDescriptionObserver* observer,
    290                            const RTCOfferAnswerOptions& options) {}
    291 
    292   // Create an answer to an offer.
    293   // The CreateSessionDescriptionObserver callback will be called when done.
    294   virtual void CreateAnswer(CreateSessionDescriptionObserver* observer,
    295                             const MediaConstraintsInterface* constraints) = 0;
    296   // Sets the local session description.
    297   // JsepInterface takes the ownership of |desc| even if it fails.
    298   // The |observer| callback will be called when done.
    299   virtual void SetLocalDescription(SetSessionDescriptionObserver* observer,
    300                                    SessionDescriptionInterface* desc) = 0;
    301   // Sets the remote session description.
    302   // JsepInterface takes the ownership of |desc| even if it fails.
    303   // The |observer| callback will be called when done.
    304   virtual void SetRemoteDescription(SetSessionDescriptionObserver* observer,
    305                                     SessionDescriptionInterface* desc) = 0;
    306   // Restarts or updates the ICE Agent process of gathering local candidates
    307   // and pinging remote candidates.
    308   virtual bool UpdateIce(const IceServers& configuration,
    309                          const MediaConstraintsInterface* constraints) = 0;
    310   // Provides a remote candidate to the ICE Agent.
    311   // A copy of the |candidate| will be created and added to the remote
    312   // description. So the caller of this method still has the ownership of the
    313   // |candidate|.
    314   // TODO(ronghuawu): Consider to change this so that the AddIceCandidate will
    315   // take the ownership of the |candidate|.
    316   virtual bool AddIceCandidate(const IceCandidateInterface* candidate) = 0;
    317 
    318   virtual void RegisterUMAObserver(UMAObserver* observer) = 0;
    319 
    320   // Returns the current SignalingState.
    321   virtual SignalingState signaling_state() = 0;
    322 
    323   // TODO(bemasc): Remove ice_state when callers are changed to
    324   // IceConnection/GatheringState.
    325   // Returns the current IceState.
    326   virtual IceState ice_state() = 0;
    327   virtual IceConnectionState ice_connection_state() = 0;
    328   virtual IceGatheringState ice_gathering_state() = 0;
    329 
    330   // Terminates all media and closes the transport.
    331   virtual void Close() = 0;
    332 
    333  protected:
    334   // Dtor protected as objects shouldn't be deleted via this interface.
    335   ~PeerConnectionInterface() {}
    336 };
    337 
    338 // PeerConnection callback interface. Application should implement these
    339 // methods.
    340 class PeerConnectionObserver {
    341  public:
    342   enum StateType {
    343     kSignalingState,
    344     kIceState,
    345   };
    346 
    347   virtual void OnError() = 0;
    348 
    349   // Triggered when the SignalingState changed.
    350   virtual void OnSignalingChange(
    351      PeerConnectionInterface::SignalingState new_state) {}
    352 
    353   // Triggered when SignalingState or IceState have changed.
    354   // TODO(bemasc): Remove once callers transition to OnSignalingChange.
    355   virtual void OnStateChange(StateType state_changed) {}
    356 
    357   // Triggered when media is received on a new stream from remote peer.
    358   virtual void OnAddStream(MediaStreamInterface* stream) = 0;
    359 
    360   // Triggered when a remote peer close a stream.
    361   virtual void OnRemoveStream(MediaStreamInterface* stream) = 0;
    362 
    363   // Triggered when a remote peer open a data channel.
    364   // TODO(perkj): Make pure virtual.
    365   virtual void OnDataChannel(DataChannelInterface* data_channel) {}
    366 
    367   // Triggered when renegotiation is needed, for example the ICE has restarted.
    368   virtual void OnRenegotiationNeeded() = 0;
    369 
    370   // Called any time the IceConnectionState changes
    371   virtual void OnIceConnectionChange(
    372       PeerConnectionInterface::IceConnectionState new_state) {}
    373 
    374   // Called any time the IceGatheringState changes
    375   virtual void OnIceGatheringChange(
    376       PeerConnectionInterface::IceGatheringState new_state) {}
    377 
    378   // New Ice candidate have been found.
    379   virtual void OnIceCandidate(const IceCandidateInterface* candidate) = 0;
    380 
    381   // TODO(bemasc): Remove this once callers transition to OnIceGatheringChange.
    382   // All Ice candidates have been found.
    383   virtual void OnIceComplete() {}
    384 
    385  protected:
    386   // Dtor protected as objects shouldn't be deleted via this interface.
    387   ~PeerConnectionObserver() {}
    388 };
    389 
    390 // Factory class used for creating cricket::PortAllocator that is used
    391 // for ICE negotiation.
    392 class PortAllocatorFactoryInterface : public rtc::RefCountInterface {
    393  public:
    394   struct StunConfiguration {
    395     StunConfiguration(const std::string& address, int port)
    396         : server(address, port) {}
    397     // STUN server address and port.
    398     rtc::SocketAddress server;
    399   };
    400 
    401   struct TurnConfiguration {
    402     TurnConfiguration(const std::string& address,
    403                       int port,
    404                       const std::string& username,
    405                       const std::string& password,
    406                       const std::string& transport_type,
    407                       bool secure)
    408         : server(address, port),
    409           username(username),
    410           password(password),
    411           transport_type(transport_type),
    412           secure(secure) {}
    413     rtc::SocketAddress server;
    414     std::string username;
    415     std::string password;
    416     std::string transport_type;
    417     bool secure;
    418   };
    419 
    420   virtual cricket::PortAllocator* CreatePortAllocator(
    421       const std::vector<StunConfiguration>& stun_servers,
    422       const std::vector<TurnConfiguration>& turn_configurations) = 0;
    423 
    424  protected:
    425   PortAllocatorFactoryInterface() {}
    426   ~PortAllocatorFactoryInterface() {}
    427 };
    428 
    429 // Used to receive callbacks of DTLS identity requests.
    430 class DTLSIdentityRequestObserver : public rtc::RefCountInterface {
    431  public:
    432   virtual void OnFailure(int error) = 0;
    433   virtual void OnSuccess(const std::string& der_cert,
    434                          const std::string& der_private_key) = 0;
    435  protected:
    436   virtual ~DTLSIdentityRequestObserver() {}
    437 };
    438 
    439 class DTLSIdentityServiceInterface {
    440  public:
    441   // Asynchronously request a DTLS identity, including a self-signed certificate
    442   // and the private key used to sign the certificate, from the identity store
    443   // for the given identity name.
    444   // DTLSIdentityRequestObserver::OnSuccess will be called with the identity if
    445   // the request succeeded; DTLSIdentityRequestObserver::OnFailure will be
    446   // called with an error code if the request failed.
    447   //
    448   // Only one request can be made at a time. If a second request is called
    449   // before the first one completes, RequestIdentity will abort and return
    450   // false.
    451   //
    452   // |identity_name| is an internal name selected by the client to identify an
    453   // identity within an origin. E.g. an web site may cache the certificates used
    454   // to communicate with differnent peers under different identity names.
    455   //
    456   // |common_name| is the common name used to generate the certificate. If the
    457   // certificate already exists in the store, |common_name| is ignored.
    458   //
    459   // |observer| is the object to receive success or failure callbacks.
    460   //
    461   // Returns true if either OnFailure or OnSuccess will be called.
    462   virtual bool RequestIdentity(
    463       const std::string& identity_name,
    464       const std::string& common_name,
    465       DTLSIdentityRequestObserver* observer) = 0;
    466 
    467   virtual ~DTLSIdentityServiceInterface() {}
    468 };
    469 
    470 // PeerConnectionFactoryInterface is the factory interface use for creating
    471 // PeerConnection, MediaStream and media tracks.
    472 // PeerConnectionFactoryInterface will create required libjingle threads,
    473 // socket and network manager factory classes for networking.
    474 // If an application decides to provide its own threads and network
    475 // implementation of these classes it should use the alternate
    476 // CreatePeerConnectionFactory method which accepts threads as input and use the
    477 // CreatePeerConnection version that takes a PortAllocatorFactoryInterface as
    478 // argument.
    479 class PeerConnectionFactoryInterface : public rtc::RefCountInterface {
    480  public:
    481   class Options {
    482    public:
    483     Options() :
    484       disable_encryption(false),
    485       disable_sctp_data_channels(false) {
    486     }
    487     bool disable_encryption;
    488     bool disable_sctp_data_channels;
    489   };
    490 
    491   virtual void SetOptions(const Options& options) = 0;
    492 
    493   virtual rtc::scoped_refptr<PeerConnectionInterface>
    494       CreatePeerConnection(
    495           const PeerConnectionInterface::RTCConfiguration& configuration,
    496           const MediaConstraintsInterface* constraints,
    497           PortAllocatorFactoryInterface* allocator_factory,
    498           DTLSIdentityServiceInterface* dtls_identity_service,
    499           PeerConnectionObserver* observer) = 0;
    500 
    501   // TODO(mallinath) : Remove below versions after clients are updated
    502   // to above method.
    503   // In latest W3C WebRTC draft, PC constructor will take RTCConfiguration,
    504   // and not IceServers. RTCConfiguration is made up of ice servers and
    505   // ice transport type.
    506   // http://dev.w3.org/2011/webrtc/editor/webrtc.html
    507   inline rtc::scoped_refptr<PeerConnectionInterface>
    508       CreatePeerConnection(
    509           const PeerConnectionInterface::IceServers& configuration,
    510           const MediaConstraintsInterface* constraints,
    511           PortAllocatorFactoryInterface* allocator_factory,
    512           DTLSIdentityServiceInterface* dtls_identity_service,
    513           PeerConnectionObserver* observer) {
    514       PeerConnectionInterface::RTCConfiguration rtc_config;
    515       rtc_config.servers = configuration;
    516       return CreatePeerConnection(rtc_config, constraints, allocator_factory,
    517                                   dtls_identity_service, observer);
    518   }
    519 
    520   virtual rtc::scoped_refptr<MediaStreamInterface>
    521       CreateLocalMediaStream(const std::string& label) = 0;
    522 
    523   // Creates a AudioSourceInterface.
    524   // |constraints| decides audio processing settings but can be NULL.
    525   virtual rtc::scoped_refptr<AudioSourceInterface> CreateAudioSource(
    526       const MediaConstraintsInterface* constraints) = 0;
    527 
    528   // Creates a VideoSourceInterface. The new source take ownership of
    529   // |capturer|. |constraints| decides video resolution and frame rate but can
    530   // be NULL.
    531   virtual rtc::scoped_refptr<VideoSourceInterface> CreateVideoSource(
    532       cricket::VideoCapturer* capturer,
    533       const MediaConstraintsInterface* constraints) = 0;
    534 
    535   // Creates a new local VideoTrack. The same |source| can be used in several
    536   // tracks.
    537   virtual rtc::scoped_refptr<VideoTrackInterface>
    538       CreateVideoTrack(const std::string& label,
    539                        VideoSourceInterface* source) = 0;
    540 
    541   // Creates an new AudioTrack. At the moment |source| can be NULL.
    542   virtual rtc::scoped_refptr<AudioTrackInterface>
    543       CreateAudioTrack(const std::string& label,
    544                        AudioSourceInterface* source) = 0;
    545 
    546   // Starts AEC dump using existing file. Takes ownership of |file| and passes
    547   // it on to VoiceEngine (via other objects) immediately, which will take
    548   // the ownerhip. If the operation fails, the file will be closed.
    549   // TODO(grunell): Remove when Chromium has started to use AEC in each source.
    550   // http://crbug.com/264611.
    551   virtual bool StartAecDump(rtc::PlatformFile file) = 0;
    552 
    553  protected:
    554   // Dtor and ctor protected as objects shouldn't be created or deleted via
    555   // this interface.
    556   PeerConnectionFactoryInterface() {}
    557   ~PeerConnectionFactoryInterface() {} // NOLINT
    558 };
    559 
    560 // Create a new instance of PeerConnectionFactoryInterface.
    561 rtc::scoped_refptr<PeerConnectionFactoryInterface>
    562 CreatePeerConnectionFactory();
    563 
    564 // Create a new instance of PeerConnectionFactoryInterface.
    565 // Ownership of |factory|, |default_adm|, and optionally |encoder_factory| and
    566 // |decoder_factory| transferred to the returned factory.
    567 rtc::scoped_refptr<PeerConnectionFactoryInterface>
    568 CreatePeerConnectionFactory(
    569     rtc::Thread* worker_thread,
    570     rtc::Thread* signaling_thread,
    571     AudioDeviceModule* default_adm,
    572     cricket::WebRtcVideoEncoderFactory* encoder_factory,
    573     cricket::WebRtcVideoDecoderFactory* decoder_factory);
    574 
    575 }  // namespace webrtc
    576 
    577 #endif  // TALK_APP_WEBRTC_PEERCONNECTIONINTERFACE_H_
    578