Implemented the GetSources() in native code.

webrtc/media/engine/fakewebrtccall.h

 /*
  *  Copyright (c) 2015 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 79 matching lines @@
 
  private:
   // webrtc::AudioReceiveStream implementation.
   void Start() override { started_ = true; }
   void Stop() override { started_ = false; }
 
   webrtc::AudioReceiveStream::Stats GetStats() const override;
   int GetOutputLevel() const override { return 0; }
   void SetSink(std::unique_ptr<webrtc::AudioSinkInterface> sink) override;
   void SetGain(float gain) override;
+  std::vector<webrtc::RtpSource> GetSources() const override {
+    return std::vector<webrtc::RtpSource>();
+  }
 
   int id_ = -1;
   webrtc::AudioReceiveStream::Config config_;
   webrtc::AudioReceiveStream::Stats stats_;
   int received_packets_ = 0;
   std::unique_ptr<webrtc::AudioSinkInterface> sink_;
   float gain_ = 1.0f;
   rtc::Buffer last_packet_;
   bool started_ = false;
 };
@@ skipping 201 matching lines @@
 
   int num_created_send_streams_;
   int num_created_receive_streams_;
 
   int audio_transport_overhead_;
   int video_transport_overhead_;
 };
 
 }  // namespace cricket
 #endif  // WEBRTC_MEDIA_ENGINE_FAKEWEBRTCCALL_H_