Chromium Code Reviews| Index: talk/media/webrtc/webrtcvoiceengine.cc |
| diff --git a/talk/media/webrtc/webrtcvoiceengine.cc b/talk/media/webrtc/webrtcvoiceengine.cc |
| index 54fac221d8fb7d6d829f606e8ab5f9897dc2469a..c6937c17678198340179741439d895ff78af6932 100644 |
| --- a/talk/media/webrtc/webrtcvoiceengine.cc |
| +++ b/talk/media/webrtc/webrtcvoiceengine.cc |
| @@ -50,6 +50,7 @@ |
| #include "webrtc/base/logging.h" |
| #include "webrtc/base/stringencode.h" |
| #include "webrtc/base/stringutils.h" |
| +#include "webrtc/call/rtc_event_log.h" |
| #include "webrtc/common.h" |
| #include "webrtc/modules/audio_processing/include/audio_processing.h" |
| @@ -1276,6 +1277,18 @@ void WebRtcVoiceEngine::StopAecDump() { |
| } |
| } |
| +bool WebRtcVoiceEngine::StartRtcEventLog(rtc::PlatformFile file) { |
| + if (!voe_wrapper_->codec()->GetEventLog()->StartLogging(file)) { |
|
the sun
2015/10/14 13:36:15
I think you should just
return voe_wrapper_->codec
ivoc
2015/10/15 15:36:14
Okay, sounds good.
|
| + LOG_RTCERR0(StartLogging); |
| + return false; |
| + } |
| + return true; |
| +} |
| + |
| +void WebRtcVoiceEngine::StopRtcEventLog() { |
| + voe_wrapper_->codec()->GetEventLog()->StopLogging(); |
| +} |
| + |
| int WebRtcVoiceEngine::CreateVoiceChannel(VoEWrapper* voice_engine_wrapper) { |
| return voice_engine_wrapper->base()->CreateChannel(voe_config_); |
| } |