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 15 matching lines @@
 
 #include "webrtc/api/video/video_frame.h"
 #include "webrtc/base/buffer.h"
 #include "webrtc/call/audio_receive_stream.h"
 #include "webrtc/call/audio_send_stream.h"
 #include "webrtc/call/call.h"
 #include "webrtc/call/flexfec_receive_stream.h"
 #include "webrtc/video_receive_stream.h"
 #include "webrtc/video_send_stream.h"
 
+namespace webrtc {
+class RtpContributingSources;
+}  // namespace webrtc
+
 namespace cricket {
 class FakeAudioSendStream final : public webrtc::AudioSendStream {
  public:
   struct TelephoneEvent {
     int payload_type = -1;
     int payload_frequency = -1;
     int event_code = 0;
     int duration_ms = 0;
   };
 
@@ skipping 44 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;
+  const std::vector<webrtc::RtpContributingSource*>& GetContributingSources()
+      override {
+    return contributing_sources_;
+  }
 
   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;
+  std::vector<webrtc::RtpContributingSource*> contributing_sources_;
 };
 
 class FakeVideoSendStream final
     : public webrtc::VideoSendStream,
       public rtc::VideoSinkInterface<webrtc::VideoFrame> {
  public:
   FakeVideoSendStream(webrtc::VideoSendStream::Config config,
                       webrtc::VideoEncoderConfig encoder_config);
   ~FakeVideoSendStream() override;
   const webrtc::VideoSendStream::Config& GetConfig() const;
@@ skipping 190 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_