Implemented the GetSources() in native code.

webrtc/audio/audio_receive_stream.cc

 /*
  *  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 201 matching lines @@
 void AudioReceiveStream::SetSink(std::unique_ptr<AudioSinkInterface> sink) {
   RTC_DCHECK_RUN_ON(&worker_thread_checker_);
   channel_proxy_->SetSink(std::move(sink));
 }
 
 void AudioReceiveStream::SetGain(float gain) {
   RTC_DCHECK_RUN_ON(&worker_thread_checker_);
   channel_proxy_->SetChannelOutputVolumeScaling(gain);
 }
 
+std::vector<RtpSource> AudioReceiveStream::GetSources() const {
+  RTC_DCHECK_RUN_ON(&worker_thread_checker_);
+  return channel_proxy_->GetSources();
+}
+
 AudioMixer::Source::AudioFrameInfo AudioReceiveStream::GetAudioFrameWithInfo(
     int sample_rate_hz,
     AudioFrame* audio_frame) {
   return channel_proxy_->GetAudioFrameWithInfo(sample_rate_hz, audio_frame);
 }
 
 int AudioReceiveStream::Ssrc() const {
   return config_.rtp.remote_ssrc;
 }
 
@@ skipping 96 matching lines @@
 int AudioReceiveStream::SetVoiceEnginePlayout(bool playout) {
   ScopedVoEInterface<VoEBase> base(voice_engine());
   if (playout) {
     return base->StartPlayout(config_.voe_channel_id);
   } else {
     return base->StopPlayout(config_.voe_channel_id);
   }
 }
 }  // namespace internal
 }  // namespace webrtc