Removing webrtc::PortAllocatorFactoryInterface.

talk/app/webrtc/peerconnection.h

 /*
  * 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,
@@ skipping 26 matching lines @@
 #include "talk/app/webrtc/rtpsenderinterface.h"
 #include "talk/app/webrtc/statscollector.h"
 #include "talk/app/webrtc/streamcollection.h"
 #include "talk/app/webrtc/webrtcsession.h"
 #include "webrtc/base/scoped_ptr.h"
 
 namespace webrtc {
 
 class RemoteMediaStreamFactory;
 
-typedef std::vector<PortAllocatorFactoryInterface::StunConfiguration>
-    StunConfigurations;
-typedef std::vector<PortAllocatorFactoryInterface::TurnConfiguration>
-    TurnConfigurations;
-
 // Populates |session_options| from |rtc_options|, and returns true if options
 // are valid.
 bool ConvertRtcOptionsForOffer(
     const PeerConnectionInterface::RTCOfferAnswerOptions& rtc_options,
     cricket::MediaSessionOptions* session_options);
 
 // Populates |session_options| from |constraints|, and returns true if all
 // mandatory constraints are satisfied.
 bool ParseConstraintsForAnswer(const MediaConstraintsInterface* constraints,
                                cricket::MediaSessionOptions* session_options);
 
-// Parses the URLs for each server in |servers| to build |stun_config| and
-// |turn_config|.
+// Parses the URLs for each server in |servers| to build |stun_servers| and
+// |turn_servers|.
 bool ParseIceServers(const PeerConnectionInterface::IceServers& servers,
-                     StunConfigurations* stun_config,
-                     TurnConfigurations* turn_config);
+                     cricket::ServerAddresses* stun_servers,
+                     std::vector<cricket::RelayServerConfig>* turn_servers);
 
 // PeerConnection implements the PeerConnectionInterface interface.
 // It uses WebRtcSession to implement the PeerConnection functionality.
 class PeerConnection : public PeerConnectionInterface,
                        public IceObserver,
                        public rtc::MessageHandler,
                        public sigslot::has_slots<> {
  public:
   explicit PeerConnection(PeerConnectionFactory* factory);
 
-  // TODO(deadbeef): Remove this overload of Initialize once everyone is moved
-  // to the new version.
-  bool Initialize(
-      const PeerConnectionInterface::RTCConfiguration& configuration,
-      const MediaConstraintsInterface* constraints,
-      PortAllocatorFactoryInterface* allocator_factory,
-      rtc::scoped_ptr<DtlsIdentityStoreInterface> dtls_identity_store,
-      PeerConnectionObserver* observer);
-
   bool Initialize(
       const PeerConnectionInterface::RTCConfiguration& configuration,
       const MediaConstraintsInterface* constraints,
       rtc::scoped_ptr<cricket::PortAllocator> allocator,
       rtc::scoped_ptr<DtlsIdentityStoreInterface> dtls_identity_store,
       PeerConnectionObserver* observer);
 
   rtc::scoped_refptr<StreamCollectionInterface> local_streams() override;
   rtc::scoped_refptr<StreamCollectionInterface> remote_streams() override;
   bool AddStream(MediaStreamInterface* local_stream) override;
@@ skipping 288 matching lines @@
   // because its destruction fires signals (such as VoiceChannelDestroyed)
   // which will trigger some final actions in PeerConnection...
   rtc::scoped_ptr<WebRtcSession> session_;
   // ... But stats_ depends on session_ so it should be destroyed even earlier.
   rtc::scoped_ptr<StatsCollector> stats_;
 };
 
 }  // namespace webrtc
 
 #endif  // TALK_APP_WEBRTC_PEERCONNECTION_H_