Implemented the GetSources() in native code.

webrtc/media/engine/fakewebrtccall.h

Index: webrtc/media/engine/fakewebrtccall.h
diff --git a/webrtc/media/engine/fakewebrtccall.h b/webrtc/media/engine/fakewebrtccall.h
index 1c9212d02a2a29ccad7ca2b344b57122508aecfd..b3fe05adc11f1daf11b7b6aa715fed057ab30a42 100644
--- a/webrtc/media/engine/fakewebrtccall.h
+++ b/webrtc/media/engine/fakewebrtccall.h
@@ -33,6 +33,10 @@
 #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:
@@ -97,6 +101,10 @@ class FakeAudioReceiveStream final : public webrtc::AudioReceiveStream {
   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_;
@@ -106,6 +114,7 @@ class FakeAudioReceiveStream final : public webrtc::AudioReceiveStream {
   float gain_ = 1.0f;
   rtc::Buffer last_packet_;
   bool started_ = false;
+  std::vector<webrtc::RtpContributingSource*> contributing_sources_;
 };
 
 class FakeVideoSendStream final