Implemented the GetSources() in native code.

webrtc/pc/channel.h

Index: webrtc/pc/channel.h
diff --git a/webrtc/pc/channel.h b/webrtc/pc/channel.h
index 6ff0556c9798525ffbe3d29b739dc55085c8681c..4ed563306fce33ba2bce3599f71c6f86dc12779f 100644
--- a/webrtc/pc/channel.h
+++ b/webrtc/pc/channel.h
@@ -19,6 +19,7 @@
 #include <vector>
 
 #include "webrtc/api/call/audio_sink.h"
+#include "webrtc/api/rtpreceiverinterface.h"
 #include "webrtc/base/asyncinvoker.h"
 #include "webrtc/base/asyncudpsocket.h"
 #include "webrtc/base/criticalsection.h"
@@ -491,6 +492,8 @@ class VoiceChannel : public BaseChannel {
   // Get statistics about the current media session.
   bool GetStats(VoiceMediaInfo* stats);
 
+  const std::vector<webrtc::RtpSource> GetSources(uint32_t ssrc);

[hbos, 2017/04/05 11:15:42]

Make the return value non-const since it's returned by value.

[Zhi Huang, 2017/04/06 03:09:50]

Done.

+
   // Monitoring functions
   sigslot::signal2<VoiceChannel*, const std::vector<ConnectionInfo>&>
       SignalConnectionMonitor;
@@ -532,7 +535,6 @@ class VoiceChannel : public BaseChannel {
   void HandleEarlyMediaTimeout();
   bool InsertDtmf_w(uint32_t ssrc, int event, int duration);
   bool SetOutputVolume_w(uint32_t ssrc, double volume);
-  bool GetStats_w(VoiceMediaInfo* stats);
 
   void OnMessage(rtc::Message* pmsg) override;
   void GetSrtpCryptoSuites_n(std::vector<int>* crypto_suites) const override;