Interface RTCRtpReceiver and RTCPeerConnection.getReceivers() added.

content/renderer/media/mock_peer_connection_impl.h

 // Copyright (c) 2012 The Chromium Authors. All rights reserved.
 // Use of this source code is governed by a BSD-style license that can be
 // found in the LICENSE file.
 
 #ifndef CONTENT_RENDERER_MEDIA_MOCK_PEER_CONNECTION_IMPL_H_
 #define CONTENT_RENDERER_MEDIA_MOCK_PEER_CONNECTION_IMPL_H_
 
 #include <memory>
 #include <string>
 
@@ skipping 28 matching lines @@
       std::vector<webrtc::MediaStreamInterface*> streams) override {
     NOTIMPLEMENTED();
     return nullptr;
   }
   bool RemoveTrack(webrtc::RtpSenderInterface* sender) override {
     NOTIMPLEMENTED();
     return false;
   }
   rtc::scoped_refptr<webrtc::DtmfSenderInterface>
       CreateDtmfSender(webrtc::AudioTrackInterface* track) override;
+  std::vector<rtc::scoped_refptr<webrtc::RtpReceiverInterface>> GetReceivers()
+      const override;
   rtc::scoped_refptr<webrtc::DataChannelInterface>
       CreateDataChannel(const std::string& label,
                         const webrtc::DataChannelInit* config) override;
   bool GetStats(webrtc::StatsObserver* observer,
                 webrtc::MediaStreamTrackInterface* track,
                 StatsOutputLevel level) override;
   void GetStats(webrtc::RTCStatsCollectorCallback* callback) override;
 
   // Call this function to make sure next call to legacy GetStats fail.
   void SetGetStatsResult(bool result) { getstats_result_ = result; }
@@ skipping 94 matching lines @@
   webrtc::RTCErrorType setconfiguration_error_type_ =
       webrtc::RTCErrorType::NONE;
   rtc::scoped_refptr<webrtc::RTCStatsReport> stats_report_;
 
   DISALLOW_COPY_AND_ASSIGN(MockPeerConnectionImpl);
 };
 
 }  // namespace content
 
 #endif  // CONTENT_RENDERER_MEDIA_MOCK_PEER_CONNECTION_IMPL_H_