Implemented the GetSources() in native code.

webrtc/call/audio_receive_stream.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.
  */
 
 #ifndef WEBRTC_CALL_AUDIO_RECEIVE_STREAM_H_
 #define WEBRTC_CALL_AUDIO_RECEIVE_STREAM_H_
 
 #include <map>
 #include <memory>
 #include <string>
 #include <vector>
 
 #include "webrtc/api/audio_codecs/audio_decoder_factory.h"
 #include "webrtc/api/call/transport.h"
+#include "webrtc/api/rtpreceiverinterface.h"
 #include "webrtc/base/optional.h"
 #include "webrtc/base/scoped_ref_ptr.h"
 #include "webrtc/common_types.h"
 #include "webrtc/config.h"
 #include "webrtc/typedefs.h"
 
 namespace webrtc {
 class AudioSinkInterface;
 
 // WORK IN PROGRESS
@@ skipping 95 matching lines @@
   // NOTE: Audio must still somehow be pulled through AudioTransport for audio
   // to stream through this sink. In practice, this happens if mixed audio
   // is being pulled+rendered and/or if audio is being pulled for the purposes
   // of feeding to the AEC.
   virtual void SetSink(std::unique_ptr<AudioSinkInterface> sink) = 0;
 
   // Sets playback gain of the stream, applied when mixing, and thus after it
   // is potentially forwarded to any attached AudioSinkInterface implementation.
   virtual void SetGain(float gain) = 0;
 
+  virtual std::vector<RtpSource> GetSources() const = 0;
+
  protected:
   virtual ~AudioReceiveStream() {}
 };
 }  // namespace webrtc
 
 #endif  // WEBRTC_CALL_AUDIO_RECEIVE_STREAM_H_