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

talk/media/webrtc/webrtcvoiceengine.cc

 /*
  * libjingle
  * Copyright 2004 Google Inc.
  *
  * Redistribution and use in source and binary forms, with or without
  * modification, are permitted provided that the following conditions are met:
  *
  *  1. Redistributions of source code must retain the above copyright notice,
  *     this list of conditions and the following disclaimer.
  *  2. Redistributions in binary form must reproduce the above copyright notice,
@@ skipping 32 matching lines @@
 #include "talk/media/base/constants.h"
 #include "talk/media/base/streamparams.h"
 #include "talk/media/webrtc/webrtcvoe.h"
 #include "webrtc/base/base64.h"
 #include "webrtc/base/byteorder.h"
 #include "webrtc/base/common.h"
 #include "webrtc/base/helpers.h"
 #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"
 
 namespace cricket {
 
 static const int kMaxNumPacketSize = 6;
 struct CodecPref {
   const char* name;
   int clockrate;
   int channels;
@@ skipping 1266 matching lines @@
   if (is_dumping_aec_) {
     // Stop dumping AEC when we are dumping.
     if (voe_wrapper_->processing()->StopDebugRecording() !=
         webrtc::AudioProcessing::kNoError) {
       LOG_RTCERR0(StopDebugRecording);
     }
     is_dumping_aec_ = false;
   }
 }
 
+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.";

[Henrik Grunell WebRTC, 2015/10/02 10:23:42]

Update text.

[ivoc, 2015/10/07 14:27:27]

Done.

+    if (!rtc::ClosePlatformFile(file))

[pthatcher1, 2015/10/06 20:49:09]

{}s please

[ivoc, 2015/10/07 14:27:27]

Done.

+      LOG(LS_WARNING) << "Could not close file.";
+    return false;
+  }
+  if (voe_wrapper_->codec()->GetEventLog()->StartLogging(event_log_file) != 0) {
+    LOG_RTCERR0(StartLogging);

[pthatcher1, 2015/10/06 20:49:09]

Why does this go through the voice engine?  Why doesn't it go straight from the
ChannelManager/MediaController to webrtc::Call?  It's not audio-specific, is it?

[ivoc, 2015/10/07 14:27:27]

See other reply. For now VoE owns the logging object, but that will change soon.

[pthatcher1, 2015/10/07 15:59:23]

Then can you put a comment in PeerConnectionFactory explaining this? 
- How the method really shouldn't belong there, but it has to because.... 
- TODO(...): Move this into being PeerConnection => MediaController =>
webrtc::Call.  

Or something like that.

[ivoc, 2015/10/08 16:20:17]

Good idea, I added some comments there like you suggested.

+    fclose(event_log_file);
+    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_);
 }
 
 int WebRtcVoiceEngine::CreateMediaVoiceChannel() {
   return CreateVoiceChannel(voe_wrapper_.get());
 }
 
 class WebRtcVoiceMediaChannel::WebRtcVoiceChannelRenderer
     : public AudioRenderer::Sink {
@@ skipping 1936 matching lines @@
       LOG(LS_WARNING) << "Unknown codec " << ToString(codec);
       return false;
     }
   }
   return true;
 }
 
 }  // namespace cricket
 
 #endif  // HAVE_WEBRTC_VOICE