| Index: webrtc/pc/channel.h | 
| diff --git a/webrtc/pc/channel.h b/webrtc/pc/channel.h | 
| index 6ff0556c9798525ffbe3d29b739dc55085c8681c..97ae3ba6f6647d2a86beeb9519531a9e651545d1 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); | 
|  | 
| +  std::vector<webrtc::RtpSource> GetSources(uint32_t ssrc) const; | 
| + | 
| // 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; | 
|  |