Relanding: Implement RTCConfiguration.iceCandidatePoolSize.

webrtc/api/peerconnection.h

 /*
  *  Copyright 2012 The WebRTC project authors. All Rights Reserved.
  *
  *  Use of this source code is governed by a BSD-style license
  *  that can be found in the LICENSE file in the root of the source
  *  tree. An additional intellectual property rights grant can be found
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
@@ skipping 113 matching lines @@
   // Deprecated, use version without constraints.
   void CreateAnswer(CreateSessionDescriptionObserver* observer,
                     const MediaConstraintsInterface* constraints) override;
   void CreateAnswer(CreateSessionDescriptionObserver* observer,
                     const RTCOfferAnswerOptions& options) override;
   void SetLocalDescription(SetSessionDescriptionObserver* observer,
                            SessionDescriptionInterface* desc) override;
   void SetRemoteDescription(SetSessionDescriptionObserver* observer,
                             SessionDescriptionInterface* desc) override;
   bool SetConfiguration(
-      const PeerConnectionInterface::RTCConfiguration& config) override;
+      const PeerConnectionInterface::RTCConfiguration& configuration) override;
   bool AddIceCandidate(const IceCandidateInterface* candidate) override;
   bool RemoveIceCandidates(
       const std::vector<cricket::Candidate>& candidates) override;
 
   void RegisterUMAObserver(UMAObserver* observer) override;
 
   void Close() override;
 
   // Virtual for unit tests.
   virtual const std::vector<rtc::scoped_refptr<DataChannel>>&
@@ skipping 58 matching lines @@
                            MediaStreamInterface* stream);
   void OnVideoTrackAdded(VideoTrackInterface* track,
                          MediaStreamInterface* stream);
   void OnVideoTrackRemoved(VideoTrackInterface* track,
                            MediaStreamInterface* stream);
 
   rtc::Thread* signaling_thread() const {
     return factory_->signaling_thread();
   }
 
+  rtc::Thread* worker_thread() const { return factory_->worker_thread(); }
+
   void PostSetSessionDescriptionFailure(SetSessionDescriptionObserver* observer,
                                         const std::string& error);
   void PostCreateSessionDescriptionFailure(
       CreateSessionDescriptionObserver* observer,
       const std::string& error);
 
   bool IsClosed() const {
     return signaling_state_ == PeerConnectionInterface::kClosed;
   }
 
@@ skipping 163 matching lines @@
   // label -> DataChannel
   std::map<std::string, rtc::scoped_refptr<DataChannel>> rtp_data_channels_;
   std::vector<rtc::scoped_refptr<DataChannel>> sctp_data_channels_;
   std::vector<rtc::scoped_refptr<DataChannel>> sctp_data_channels_to_free_;
 
   bool remote_peer_supports_msid_ = false;
 
   std::vector<rtc::scoped_refptr<RtpSenderInterface>> senders_;
   std::vector<rtc::scoped_refptr<RtpReceiverInterface>> receivers_;
 
-  // The session_ unique_ptr is declared at the bottom of PeerConnection
-  // because its destruction fires signals (such as VoiceChannelDestroyed)
-  // which will trigger some final actions in PeerConnection...
   std::unique_ptr<WebRtcSession> session_;
-  // ... But stats_ depends on session_ so it should be destroyed even earlier.
   std::unique_ptr<StatsCollector> stats_;
 };
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_API_PEERCONNECTION_H_