Update talk to 58113193 together with https://webrtc-codereview.appspot.com/5309005/.
R=mallinath@webrtc.org, niklas.enbom@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/5719004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5274 4adac7df-926f-26a2-2b94-8c16560cd09d
diff --git a/talk/media/webrtc/webrtcvoiceengine.h b/talk/media/webrtc/webrtcvoiceengine.h
index 29807ef..adf4853 100644
--- a/talk/media/webrtc/webrtcvoiceengine.h
+++ b/talk/media/webrtc/webrtcvoiceengine.h
@@ -174,6 +174,9 @@
bool SetAudioDeviceModule(webrtc::AudioDeviceModule* adm,
webrtc::AudioDeviceModule* adm_sc);
+ // Starts AEC dump using existing file.
+ bool StartAecDump(FILE* file);
+
// Check whether the supplied trace should be ignored.
bool ShouldIgnoreTrace(const std::string& trace);
@@ -356,8 +359,10 @@
virtual bool CanInsertDtmf();
virtual bool InsertDtmf(uint32 ssrc, int event, int duration, int flags);
- virtual void OnPacketReceived(talk_base::Buffer* packet);
- virtual void OnRtcpReceived(talk_base::Buffer* packet);
+ virtual void OnPacketReceived(talk_base::Buffer* packet,
+ const talk_base::PacketTime& packet_time);
+ virtual void OnRtcpReceived(talk_base::Buffer* packet,
+ const talk_base::PacketTime& packet_time);
virtual void OnReadyToSend(bool ready) {}
virtual bool MuteStream(uint32 ssrc, bool on);
virtual bool SetSendBandwidth(bool autobw, int bps);