| Index: webrtc/pc/channel.h
|
| diff --git a/webrtc/pc/channel.h b/webrtc/pc/channel.h
|
| index 97ae3ba6f6647d2a86beeb9519531a9e651545d1..6ff0556c9798525ffbe3d29b739dc55085c8681c 100644
|
| --- a/webrtc/pc/channel.h
|
| +++ b/webrtc/pc/channel.h
|
| @@ -19,7 +19,6 @@
|
| #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"
|
| @@ -492,8 +491,6 @@
|
| // 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;
|
| @@ -535,6 +532,7 @@
|
| 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;
|
|
|