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; |