Added functions on libjingle API to start and stop the recording of an RtcEventLog.

talk/media/webrtc/webrtcvoiceengine.cc

Index: talk/media/webrtc/webrtcvoiceengine.cc
diff --git a/talk/media/webrtc/webrtcvoiceengine.cc b/talk/media/webrtc/webrtcvoiceengine.cc
index 5130232bca4ce880ba5fea277fb0b58d7e39fefa..db8600a10feb059208c14e9fd74ee8b3f3c8e7d9 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"
 
@@ -1336,6 +1337,27 @@ void WebRtcVoiceEngine::StopAecDump() {
   }
 }
 
+bool WebRtcVoiceEngine::StartRtcEventLog(rtc::PlatformFile file) {
+  FILE* event_log_file = rtc::FdopenPlatformFileForWriting(file);
+  if (!event_log_file) {
+    LOG(LS_ERROR) << "Could not open AEC dump file stream.";
+    if (!rtc::ClosePlatformFile(file))
+      LOG(LS_WARNING) << "Could not close file.";
+    return false;
+  }
+  if (voe_wrapper_->codec()->GetEventLog()->StartLogging(event_log_file) != 0) {
+    LOG_RTCERR0(StartLogging);
+    fclose(event_log_file);
+    return false;
+  }
+  is_dumping_aec_ = true;
+  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_);
 }