Index: webrtc/media/engine/webrtcvoiceengine.cc |
diff --git a/webrtc/media/engine/webrtcvoiceengine.cc b/webrtc/media/engine/webrtcvoiceengine.cc |
index d329270ce5af736f709a4167abab364c2871a490..2ddf67dd3a6d76d70236b8562cf2d5d8208bbd70 100644 |
--- a/webrtc/media/engine/webrtcvoiceengine.cc |
+++ b/webrtc/media/engine/webrtcvoiceengine.cc |
@@ -28,6 +28,7 @@ |
#include "webrtc/base/stringencode.h" |
#include "webrtc/base/stringutils.h" |
#include "webrtc/base/trace_event.h" |
+#include "webrtc/call/rtc_event_log.h" |
#include "webrtc/common.h" |
#include "webrtc/media/base/audiosource.h" |
#include "webrtc/media/base/mediaconstants.h" |
@@ -1037,6 +1038,27 @@ |
} |
is_dumping_aec_ = false; |
} |
+} |
+ |
+bool WebRtcVoiceEngine::StartRtcEventLog(rtc::PlatformFile file, |
+ int64_t max_size_bytes) { |
+ RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); |
+ webrtc::RtcEventLog* event_log = voe_wrapper_->codec()->GetEventLog(); |
+ if (event_log) { |
+ return event_log->StartLogging(file, max_size_bytes); |
+ } |
+ LOG_RTCERR0(StartRtcEventLog); |
+ return false; |
+} |
+ |
+void WebRtcVoiceEngine::StopRtcEventLog() { |
+ RTC_DCHECK(worker_thread_checker_.CalledOnValidThread()); |
+ webrtc::RtcEventLog* event_log = voe_wrapper_->codec()->GetEventLog(); |
+ if (event_log) { |
+ event_log->StopLogging(); |
+ return; |
+ } |
+ LOG_RTCERR0(StopRtcEventLog); |
} |
int WebRtcVoiceEngine::CreateVoEChannel() { |