Implemented the GetSources() in native code.

webrtc/voice_engine/channel.h

Index: webrtc/voice_engine/channel.h
diff --git a/webrtc/voice_engine/channel.h b/webrtc/voice_engine/channel.h
index 13d8e49877a3d74f924dcded2105013babfb10ae..fb9d61ddc84b744c7c1cd3d787e20b30519c86d8 100644
--- a/webrtc/voice_engine/channel.h
+++ b/webrtc/voice_engine/channel.h
@@ -28,6 +28,7 @@
 #include "webrtc/modules/audio_processing/rms_level.h"
 #include "webrtc/modules/rtp_rtcp/include/remote_ntp_time_estimator.h"
 #include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.h"
+#include "webrtc/modules/rtp_rtcp/include/rtp_receiver.h"
 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h"
 #include "webrtc/voice_engine/audio_level.h"
 #include "webrtc/voice_engine/file_player.h"
@@ -52,7 +53,6 @@ class ReceiveStatistics;
 class RemoteNtpTimeEstimator;
 class RtcEventLog;
 class RTPPayloadRegistry;
-class RtpReceiver;
 class RTPReceiverAudio;
 class RtpPacketReceived;
 class RtpRtcp;
@@ -399,6 +399,10 @@ class Channel
 
   void OnRecoverableUplinkPacketLossRate(float recoverable_packet_loss_rate);
 
+  std::vector<RtpSource> GetSources() const {
+    return rtp_receiver_->GetSources();
+  }
+
  private:
   class ProcessAndEncodeAudioTask;