Implemented the GetSources() in native code.

webrtc/pc/rtpreceiver.cc

 /*
  *  Copyright 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.
  */
 
 #include "webrtc/pc/rtpreceiver.h"
 
 #include "webrtc/api/mediastreamtrackproxy.h"
 #include "webrtc/api/videosourceproxy.h"
+#include "webrtc/base/ptr_util.h"
 #include "webrtc/base/trace_event.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_receiver.h"
 #include "webrtc/pc/audiotrack.h"
 #include "webrtc/pc/videotrack.h"
 
 namespace webrtc {
 
 AudioRtpReceiver::AudioRtpReceiver(const std::string& track_id,
                                    uint32_t ssrc,
                                    cricket::VoiceChannel* channel)
     : id_(track_id),
       ssrc_(ssrc),
@@ skipping 64 matching lines @@
     return;
   }
   if (channel_) {
     // Allow that SetOutputVolume fail. This is the normal case when the
     // underlying media channel has already been deleted.
     channel_->SetOutputVolume(ssrc_, 0);
   }
   stopped_ = true;
 }
 
+std::vector<RtpContributingSource> AudioRtpReceiver::GetContributingSources() {
+  return channel_->GetContributingSources(ssrc_);
+}
+
 void AudioRtpReceiver::Reconfigure() {
   RTC_DCHECK(!stopped_);
   if (!channel_) {
     LOG(LS_ERROR) << "AudioRtpReceiver::Reconfigure: No audio channel exists.";
     return;
   }
   if (!channel_->SetOutputVolume(ssrc_,
                                  track_->enabled() ? cached_volume_ : 0)) {
     RTC_NOTREACHED();
   }
@@ skipping 119 matching lines @@
 }
 
 void VideoRtpReceiver::OnFirstPacketReceived(cricket::BaseChannel* channel) {
   if (observer_) {
     observer_->OnFirstPacketReceived(media_type());
   }
   received_first_packet_ = true;
 }
 
 }  // namespace webrtc