Move talk/app/webrtc to webrtc/api

talk/app/webrtc/peerconnectioninterface.h

Index: talk/app/webrtc/peerconnectioninterface.h
diff --git a/talk/app/webrtc/peerconnectioninterface.h b/talk/app/webrtc/peerconnectioninterface.h
deleted file mode 100644
index 940f0fb9e6f8940ada3cdaf1d6e51f2bae054287..0000000000000000000000000000000000000000
--- a/talk/app/webrtc/peerconnectioninterface.h
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-/*
- * 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.
- */
-
-// This file contains the PeerConnection interface as defined in
-// http://dev.w3.org/2011/webrtc/editor/webrtc.html#peer-to-peer-connections.
-// Applications must use this interface to implement peerconnection.
-// PeerConnectionFactory class provides factory methods to create
-// peerconnection, mediastream and media tracks objects.
-//
-// The Following steps are needed to setup a typical call using Jsep.
-// 1. Create a PeerConnectionFactoryInterface. Check constructors for more
-// information about input parameters.
-// 2. Create a PeerConnection object. Provide a configuration string which
-// points either to stun or turn server to generate ICE candidates and provide
-// an object that implements the PeerConnectionObserver interface.
-// 3. Create local MediaStream and MediaTracks using the PeerConnectionFactory
-// and add it to PeerConnection by calling AddStream.
-// 4. Create an offer and serialize it and send it to the remote peer.
-// 5. Once an ice candidate have been found PeerConnection will call the
-// observer function OnIceCandidate. The candidates must also be serialized and
-// sent to the remote peer.
-// 6. Once an answer is received from the remote peer, call
-// SetLocalSessionDescription with the offer and SetRemoteSessionDescription
-// with the remote answer.
-// 7. Once a remote candidate is received from the remote peer, provide it to
-// the peerconnection by calling AddIceCandidate.
-
-
-// The Receiver of a call can decide to accept or reject the call.
-// This decision will be taken by the application not peerconnection.
-// If application decides to accept the call
-// 1. Create PeerConnectionFactoryInterface if it doesn't exist.
-// 2. Create a new PeerConnection.
-// 3. Provide the remote offer to the new PeerConnection object by calling
-// SetRemoteSessionDescription.
-// 4. Generate an answer to the remote offer by calling CreateAnswer and send it
-// back to the remote peer.
-// 5. Provide the local answer to the new PeerConnection by calling
-// SetLocalSessionDescription with the answer.
-// 6. Provide the remote ice candidates by calling AddIceCandidate.
-// 7. Once a candidate have been found PeerConnection will call the observer
-// function OnIceCandidate. Send these candidates to the remote peer.
-
-#ifndef TALK_APP_WEBRTC_PEERCONNECTIONINTERFACE_H_
-#define TALK_APP_WEBRTC_PEERCONNECTIONINTERFACE_H_
-
-#include <string>
-#include <utility>
-#include <vector>
-
-#include "talk/app/webrtc/datachannelinterface.h"
-#include "talk/app/webrtc/dtlsidentitystore.h"
-#include "talk/app/webrtc/dtlsidentitystore.h"
-#include "talk/app/webrtc/dtmfsenderinterface.h"
-#include "talk/app/webrtc/jsep.h"
-#include "talk/app/webrtc/mediastreaminterface.h"
-#include "talk/app/webrtc/rtpreceiverinterface.h"
-#include "talk/app/webrtc/rtpsenderinterface.h"
-#include "talk/app/webrtc/statstypes.h"
-#include "talk/app/webrtc/umametrics.h"
-#include "webrtc/base/fileutils.h"
-#include "webrtc/base/network.h"
-#include "webrtc/base/rtccertificate.h"
-#include "webrtc/base/socketaddress.h"
-#include "webrtc/base/sslstreamadapter.h"
-#include "webrtc/p2p/base/portallocator.h"
-
-namespace rtc {
-class SSLIdentity;
-class Thread;
-}
-
-namespace cricket {
-class WebRtcVideoDecoderFactory;
-class WebRtcVideoEncoderFactory;
-}
-
-namespace webrtc {
-class AudioDeviceModule;
-class MediaConstraintsInterface;
-
-// MediaStream container interface.
-class StreamCollectionInterface : public rtc::RefCountInterface {
- public:
-  // TODO(ronghuawu): Update the function names to c++ style, e.g. find -> Find.
-  virtual size_t count() = 0;
-  virtual MediaStreamInterface* at(size_t index) = 0;
-  virtual MediaStreamInterface* find(const std::string& label) = 0;
-  virtual MediaStreamTrackInterface* FindAudioTrack(
-      const std::string& id) = 0;
-  virtual MediaStreamTrackInterface* FindVideoTrack(
-      const std::string& id) = 0;
-
- protected:
-  // Dtor protected as objects shouldn't be deleted via this interface.
-  ~StreamCollectionInterface() {}
-};
-
-class StatsObserver : public rtc::RefCountInterface {
- public:
-  virtual void OnComplete(const StatsReports& reports) = 0;
-
- protected:
-  virtual ~StatsObserver() {}
-};
-
-class MetricsObserverInterface : public rtc::RefCountInterface {
- public:
-
-  // |type| is the type of the enum counter to be incremented. |counter|
-  // is the particular counter in that type. |counter_max| is the next sequence
-  // number after the highest counter.
-  virtual void IncrementEnumCounter(PeerConnectionEnumCounterType type,
-                                    int counter,
-                                    int counter_max) {}
-
-  // This is used to handle sparse counters like SSL cipher suites.
-  // TODO(guoweis): Remove the implementation once the dependency's interface
-  // definition is updated.
-  virtual void IncrementSparseEnumCounter(PeerConnectionEnumCounterType type,
-                                          int counter) {
-    IncrementEnumCounter(type, counter, 0 /* Ignored */);
-  }
-
-  virtual void AddHistogramSample(PeerConnectionMetricsName type,
-                                  int value) = 0;
-
- protected:
-  virtual ~MetricsObserverInterface() {}
-};
-
-typedef MetricsObserverInterface UMAObserver;
-
-class PeerConnectionInterface : public rtc::RefCountInterface {
- public:
-  // See http://dev.w3.org/2011/webrtc/editor/webrtc.html#state-definitions .
-  enum SignalingState {
-    kStable,
-    kHaveLocalOffer,
-    kHaveLocalPrAnswer,
-    kHaveRemoteOffer,
-    kHaveRemotePrAnswer,
-    kClosed,
-  };
-
-  // TODO(bemasc): Remove IceState when callers are changed to
-  // IceConnection/GatheringState.
-  enum IceState {
-    kIceNew,
-    kIceGathering,
-    kIceWaiting,
-    kIceChecking,
-    kIceConnected,
-    kIceCompleted,
-    kIceFailed,
-    kIceClosed,
-  };
-
-  enum IceGatheringState {
-    kIceGatheringNew,
-    kIceGatheringGathering,
-    kIceGatheringComplete
-  };
-
-  enum IceConnectionState {
-    kIceConnectionNew,
-    kIceConnectionChecking,
-    kIceConnectionConnected,
-    kIceConnectionCompleted,
-    kIceConnectionFailed,
-    kIceConnectionDisconnected,
-    kIceConnectionClosed,
-    kIceConnectionMax,
-  };
-
-  struct IceServer {
-    // TODO(jbauch): Remove uri when all code using it has switched to urls.
-    std::string uri;
-    std::vector<std::string> urls;
-    std::string username;
-    std::string password;
-  };
-  typedef std::vector<IceServer> IceServers;
-
-  enum IceTransportsType {
-    // TODO(pthatcher): Rename these kTransporTypeXXX, but update
-    // Chromium at the same time.
-    kNone,
-    kRelay,
-    kNoHost,
-    kAll
-  };
-
-  // https://tools.ietf.org/html/draft-ietf-rtcweb-jsep-08#section-4.1.1
-  enum BundlePolicy {
-    kBundlePolicyBalanced,
-    kBundlePolicyMaxBundle,
-    kBundlePolicyMaxCompat
-  };
-
-  // https://tools.ietf.org/html/draft-ietf-rtcweb-jsep-09#section-4.1.1
-  enum RtcpMuxPolicy {
-    kRtcpMuxPolicyNegotiate,
-    kRtcpMuxPolicyRequire,
-  };
-
-  enum TcpCandidatePolicy {
-    kTcpCandidatePolicyEnabled,
-    kTcpCandidatePolicyDisabled
-  };
-
-  enum ContinualGatheringPolicy {
-    GATHER_ONCE,
-    GATHER_CONTINUALLY
-  };
-
-  // TODO(hbos): Change into class with private data and public getters.
-  struct RTCConfiguration {
-    static const int kUndefined = -1;
-    // Default maximum number of packets in the audio jitter buffer.
-    static const int kAudioJitterBufferMaxPackets = 50;
-    // TODO(pthatcher): Rename this ice_transport_type, but update
-    // Chromium at the same time.
-    IceTransportsType type;
-    // TODO(pthatcher): Rename this ice_servers, but update Chromium
-    // at the same time.
-    IceServers servers;
-    BundlePolicy bundle_policy;
-    RtcpMuxPolicy rtcp_mux_policy;
-    TcpCandidatePolicy tcp_candidate_policy;
-    int audio_jitter_buffer_max_packets;
-    bool audio_jitter_buffer_fast_accelerate;
-    int ice_connection_receiving_timeout;         // ms
-    int ice_backup_candidate_pair_ping_interval;  // ms
-    ContinualGatheringPolicy continual_gathering_policy;
-    std::vector<rtc::scoped_refptr<rtc::RTCCertificate>> certificates;
-    bool disable_prerenderer_smoothing;
-    RTCConfiguration()
-        : type(kAll),
-          bundle_policy(kBundlePolicyBalanced),
-          rtcp_mux_policy(kRtcpMuxPolicyNegotiate),
-          tcp_candidate_policy(kTcpCandidatePolicyEnabled),
-          audio_jitter_buffer_max_packets(kAudioJitterBufferMaxPackets),
-          audio_jitter_buffer_fast_accelerate(false),
-          ice_connection_receiving_timeout(kUndefined),
-          ice_backup_candidate_pair_ping_interval(kUndefined),
-          continual_gathering_policy(GATHER_ONCE),
-          disable_prerenderer_smoothing(false) {}
-  };
-
-  struct RTCOfferAnswerOptions {
-    static const int kUndefined = -1;
-    static const int kMaxOfferToReceiveMedia = 1;
-
-    // The default value for constraint offerToReceiveX:true.
-    static const int kOfferToReceiveMediaTrue = 1;
-
-    int offer_to_receive_video;
-    int offer_to_receive_audio;
-    bool voice_activity_detection;
-    bool ice_restart;
-    bool use_rtp_mux;
-
-    RTCOfferAnswerOptions()
-        : offer_to_receive_video(kUndefined),
-          offer_to_receive_audio(kUndefined),
-          voice_activity_detection(true),
-          ice_restart(false),
-          use_rtp_mux(true) {}
-
-    RTCOfferAnswerOptions(int offer_to_receive_video,
-                          int offer_to_receive_audio,
-                          bool voice_activity_detection,
-                          bool ice_restart,
-                          bool use_rtp_mux)
-        : offer_to_receive_video(offer_to_receive_video),
-          offer_to_receive_audio(offer_to_receive_audio),
-          voice_activity_detection(voice_activity_detection),
-          ice_restart(ice_restart),
-          use_rtp_mux(use_rtp_mux) {}
-  };
-
-  // Used by GetStats to decide which stats to include in the stats reports.
-  // |kStatsOutputLevelStandard| includes the standard stats for Javascript API;
-  // |kStatsOutputLevelDebug| includes both the standard stats and additional
-  // stats for debugging purposes.
-  enum StatsOutputLevel {
-    kStatsOutputLevelStandard,
-    kStatsOutputLevelDebug,
-  };
-
-  // Accessor methods to active local streams.
-  virtual rtc::scoped_refptr<StreamCollectionInterface>
-      local_streams() = 0;
-
-  // Accessor methods to remote streams.
-  virtual rtc::scoped_refptr<StreamCollectionInterface>
-      remote_streams() = 0;
-
-  // Add a new MediaStream to be sent on this PeerConnection.
-  // Note that a SessionDescription negotiation is needed before the
-  // remote peer can receive the stream.
-  virtual bool AddStream(MediaStreamInterface* stream) = 0;
-
-  // Remove a MediaStream from this PeerConnection.
-  // Note that a SessionDescription negotiation is need before the
-  // remote peer is notified.
-  virtual void RemoveStream(MediaStreamInterface* stream) = 0;
-
-  // TODO(deadbeef): Make the following two methods pure virtual once
-  // implemented by all subclasses of PeerConnectionInterface.
-  // Add a new MediaStreamTrack to be sent on this PeerConnection.
-  // |streams| indicates which stream labels the track should be associated
-  // with.
-  virtual rtc::scoped_refptr<RtpSenderInterface> AddTrack(
-      MediaStreamTrackInterface* track,
-      std::vector<MediaStreamInterface*> streams) {
-    return nullptr;
-  }
-
-  // Remove an RtpSender from this PeerConnection.
-  // Returns true on success.
-  virtual bool RemoveTrack(RtpSenderInterface* sender) {
-    return false;
-  }
-
-  // Returns pointer to the created DtmfSender on success.
-  // Otherwise returns NULL.
-  virtual rtc::scoped_refptr<DtmfSenderInterface> CreateDtmfSender(
-      AudioTrackInterface* track) = 0;
-
-  // TODO(deadbeef): Make these pure virtual once all subclasses implement them.
-  // |kind| must be "audio" or "video".
-  // |stream_id| is used to populate the msid attribute; if empty, one will
-  // be generated automatically.
-  virtual rtc::scoped_refptr<RtpSenderInterface> CreateSender(
-      const std::string& kind,
-      const std::string& stream_id) {
-    return rtc::scoped_refptr<RtpSenderInterface>();
-  }
-
-  virtual std::vector<rtc::scoped_refptr<RtpSenderInterface>> GetSenders()
-      const {
-    return std::vector<rtc::scoped_refptr<RtpSenderInterface>>();
-  }
-
-  virtual std::vector<rtc::scoped_refptr<RtpReceiverInterface>> GetReceivers()
-      const {
-    return std::vector<rtc::scoped_refptr<RtpReceiverInterface>>();
-  }
-
-  virtual bool GetStats(StatsObserver* observer,
-                        MediaStreamTrackInterface* track,
-                        StatsOutputLevel level) = 0;
-
-  virtual rtc::scoped_refptr<DataChannelInterface> CreateDataChannel(
-      const std::string& label,
-      const DataChannelInit* config) = 0;
-
-  virtual const SessionDescriptionInterface* local_description() const = 0;
-  virtual const SessionDescriptionInterface* remote_description() const = 0;
-
-  // Create a new offer.
-  // The CreateSessionDescriptionObserver callback will be called when done.
-  virtual void CreateOffer(CreateSessionDescriptionObserver* observer,
-                           const MediaConstraintsInterface* constraints) {}
-
-  // TODO(jiayl): remove the default impl and the old interface when chromium
-  // code is updated.
-  virtual void CreateOffer(CreateSessionDescriptionObserver* observer,
-                           const RTCOfferAnswerOptions& options) {}
-
-  // Create an answer to an offer.
-  // The CreateSessionDescriptionObserver callback will be called when done.
-  virtual void CreateAnswer(CreateSessionDescriptionObserver* observer,
-                            const MediaConstraintsInterface* constraints) = 0;
-  // Sets the local session description.
-  // JsepInterface takes the ownership of |desc| even if it fails.
-  // The |observer| callback will be called when done.
-  virtual void SetLocalDescription(SetSessionDescriptionObserver* observer,
-                                   SessionDescriptionInterface* desc) = 0;
-  // Sets the remote session description.
-  // JsepInterface takes the ownership of |desc| even if it fails.
-  // The |observer| callback will be called when done.
-  virtual void SetRemoteDescription(SetSessionDescriptionObserver* observer,
-                                    SessionDescriptionInterface* desc) = 0;
-  // Restarts or updates the ICE Agent process of gathering local candidates
-  // and pinging remote candidates.
-  // TODO(deadbeef): Remove once Chrome is moved over to SetConfiguration.
-  virtual bool UpdateIce(const IceServers& configuration,
-                         const MediaConstraintsInterface* constraints) {
-    return false;
-  }
-  // Sets the PeerConnection's global configuration to |config|.
-  // Any changes to STUN/TURN servers or ICE candidate policy will affect the
-  // next gathering phase, and cause the next call to createOffer to generate
-  // new ICE credentials. Note that the BUNDLE and RTCP-multiplexing policies
-  // cannot be changed with this method.
-  // TODO(deadbeef): Make this pure virtual once all Chrome subclasses of
-  // PeerConnectionInterface implement it.
-  virtual bool SetConfiguration(
-      const PeerConnectionInterface::RTCConfiguration& config) {
-    return false;
-  }
-  // Provides a remote candidate to the ICE Agent.
-  // A copy of the |candidate| will be created and added to the remote
-  // description. So the caller of this method still has the ownership of the
-  // |candidate|.
-  // TODO(ronghuawu): Consider to change this so that the AddIceCandidate will
-  // take the ownership of the |candidate|.
-  virtual bool AddIceCandidate(const IceCandidateInterface* candidate) = 0;
-
-  virtual void RegisterUMAObserver(UMAObserver* observer) = 0;
-
-  // Returns the current SignalingState.
-  virtual SignalingState signaling_state() = 0;
-
-  // TODO(bemasc): Remove ice_state when callers are changed to
-  // IceConnection/GatheringState.
-  // Returns the current IceState.
-  virtual IceState ice_state() = 0;
-  virtual IceConnectionState ice_connection_state() = 0;
-  virtual IceGatheringState ice_gathering_state() = 0;
-
-  // Terminates all media and closes the transport.
-  virtual void Close() = 0;
-
- protected:
-  // Dtor protected as objects shouldn't be deleted via this interface.
-  ~PeerConnectionInterface() {}
-};
-
-// PeerConnection callback interface. Application should implement these
-// methods.
-class PeerConnectionObserver {
- public:
-  enum StateType {
-    kSignalingState,
-    kIceState,
-  };
-
-  // Triggered when the SignalingState changed.
-  virtual void OnSignalingChange(
-      PeerConnectionInterface::SignalingState new_state) = 0;
-
-  // Triggered when media is received on a new stream from remote peer.
-  virtual void OnAddStream(MediaStreamInterface* stream) = 0;
-
-  // Triggered when a remote peer close a stream.
-  virtual void OnRemoveStream(MediaStreamInterface* stream) = 0;
-
-  // Triggered when a remote peer open a data channel.
-  virtual void OnDataChannel(DataChannelInterface* data_channel) = 0;
-
-  // Triggered when renegotiation is needed, for example the ICE has restarted.
-  virtual void OnRenegotiationNeeded() = 0;
-
-  // Called any time the IceConnectionState changes
-  virtual void OnIceConnectionChange(
-      PeerConnectionInterface::IceConnectionState new_state) = 0;
-
-  // Called any time the IceGatheringState changes
-  virtual void OnIceGatheringChange(
-      PeerConnectionInterface::IceGatheringState new_state) = 0;
-
-  // New Ice candidate have been found.
-  virtual void OnIceCandidate(const IceCandidateInterface* candidate) = 0;
-
-  // Called when the ICE connection receiving status changes.
-  virtual void OnIceConnectionReceivingChange(bool receiving) {}
-
- protected:
-  // Dtor protected as objects shouldn't be deleted via this interface.
-  ~PeerConnectionObserver() {}
-};
-
-// PeerConnectionFactoryInterface is the factory interface use for creating
-// PeerConnection, MediaStream and media tracks.
-// PeerConnectionFactoryInterface will create required libjingle threads,
-// socket and network manager factory classes for networking.
-// If an application decides to provide its own threads and network
-// implementation of these classes it should use the alternate
-// CreatePeerConnectionFactory method which accepts threads as input and use the
-// CreatePeerConnection version that takes a PortAllocator as an
-// argument.
-class PeerConnectionFactoryInterface : public rtc::RefCountInterface {
- public:
-  class Options {
-   public:
-    Options()
-        : disable_encryption(false),
-          disable_sctp_data_channels(false),
-          disable_network_monitor(false),
-          network_ignore_mask(rtc::kDefaultNetworkIgnoreMask),
-          ssl_max_version(rtc::SSL_PROTOCOL_DTLS_12) {}
-    bool disable_encryption;
-    bool disable_sctp_data_channels;
-    bool disable_network_monitor;
-
-    // Sets the network types to ignore. For instance, calling this with
-    // ADAPTER_TYPE_ETHERNET | ADAPTER_TYPE_LOOPBACK will ignore Ethernet and
-    // loopback interfaces.
-    int network_ignore_mask;
-
-    // Sets the maximum supported protocol version. The highest version
-    // supported by both ends will be used for the connection, i.e. if one
-    // party supports DTLS 1.0 and the other DTLS 1.2, DTLS 1.0 will be used.
-    rtc::SSLProtocolVersion ssl_max_version;
-  };
-
-  virtual void SetOptions(const Options& options) = 0;
-
-  virtual rtc::scoped_refptr<PeerConnectionInterface> CreatePeerConnection(
-      const PeerConnectionInterface::RTCConfiguration& configuration,
-      const MediaConstraintsInterface* constraints,
-      rtc::scoped_ptr<cricket::PortAllocator> allocator,
-      rtc::scoped_ptr<DtlsIdentityStoreInterface> dtls_identity_store,
-      PeerConnectionObserver* observer) = 0;
-
-  virtual rtc::scoped_refptr<MediaStreamInterface>
-      CreateLocalMediaStream(const std::string& label) = 0;
-
-  // Creates a AudioSourceInterface.
-  // |constraints| decides audio processing settings but can be NULL.
-  virtual rtc::scoped_refptr<AudioSourceInterface> CreateAudioSource(
-      const MediaConstraintsInterface* constraints) = 0;
-
-  // Creates a VideoSourceInterface. The new source take ownership of
-  // |capturer|. |constraints| decides video resolution and frame rate but can
-  // be NULL.
-  virtual rtc::scoped_refptr<VideoSourceInterface> CreateVideoSource(
-      cricket::VideoCapturer* capturer,
-      const MediaConstraintsInterface* constraints) = 0;
-
-  // Creates a new local VideoTrack. The same |source| can be used in several
-  // tracks.
-  virtual rtc::scoped_refptr<VideoTrackInterface>
-      CreateVideoTrack(const std::string& label,
-                       VideoSourceInterface* source) = 0;
-
-  // Creates an new AudioTrack. At the moment |source| can be NULL.
-  virtual rtc::scoped_refptr<AudioTrackInterface>
-      CreateAudioTrack(const std::string& label,
-                       AudioSourceInterface* source) = 0;
-
-  // Starts AEC dump using existing file. Takes ownership of |file| and passes
-  // it on to VoiceEngine (via other objects) immediately, which will take
-  // the ownerhip. If the operation fails, the file will be closed.
-  // A maximum file size in bytes can be specified. When the file size limit is
-  // reached, logging is stopped automatically. If max_size_bytes is set to a
-  // value <= 0, no limit will be used, and logging will continue until the
-  // StopAecDump function is called.
-  virtual bool StartAecDump(rtc::PlatformFile file, int64_t max_size_bytes) = 0;
-
-  // Stops logging the AEC dump.
-  virtual void StopAecDump() = 0;
-
-  // Starts RtcEventLog using existing file. Takes ownership of |file| and
-  // passes it on to VoiceEngine, which will take the ownership. If the
-  // operation fails the file will be closed. The logging will stop
-  // automatically after 10 minutes have passed, or when the StopRtcEventLog
-  // function is called.
-  // This function as well as the StopRtcEventLog don't really belong on this
-  // interface, this is a temporary solution until we move the logging object
-  // from inside voice engine to webrtc::Call, which will happen when the VoE
-  // restructuring effort is further along.
-  // TODO(ivoc): Move this into being:
-  //             PeerConnection => MediaController => webrtc::Call.
-  virtual bool StartRtcEventLog(rtc::PlatformFile file) = 0;
-
-  // Stops logging the RtcEventLog.
-  virtual void StopRtcEventLog() = 0;
-
- protected:
-  // Dtor and ctor protected as objects shouldn't be created or deleted via
-  // this interface.
-  PeerConnectionFactoryInterface() {}
-  ~PeerConnectionFactoryInterface() {} // NOLINT
-};
-
-// Create a new instance of PeerConnectionFactoryInterface.
-rtc::scoped_refptr<PeerConnectionFactoryInterface>
-CreatePeerConnectionFactory();
-
-// Create a new instance of PeerConnectionFactoryInterface.
-// Ownership of |factory|, |default_adm|, and optionally |encoder_factory| and
-// |decoder_factory| transferred to the returned factory.
-rtc::scoped_refptr<PeerConnectionFactoryInterface>
-CreatePeerConnectionFactory(
-    rtc::Thread* worker_thread,
-    rtc::Thread* signaling_thread,
-    AudioDeviceModule* default_adm,
-    cricket::WebRtcVideoEncoderFactory* encoder_factory,
-    cricket::WebRtcVideoDecoderFactory* decoder_factory);
-
-}  // namespace webrtc
-
-#endif  // TALK_APP_WEBRTC_PEERCONNECTIONINTERFACE_H_