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 {
 namespace {
 
 const int kMaxNumPacketSize = 6;
 struct CodecPref {
   const char* name;
   int clockrate;
@@ skipping 1206 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 RtcEventLog file stream.";
+    if (!rtc::ClosePlatformFile(file)) {

[the sun, 2015/10/13 11:00:40]

This isn't completely clear to me.

I assume ClosePlatformFile() is called because ownership of the PlatformFile is
effectively transferred to WVoE when this function is called.

But below, when StartLogging() is called, you do not call ClosePlatformFile(),
and not in StopRtcEventLog() why?

[ivoc, 2015/10/13 13:50:50]

Hmm, that is actually a really good point. The straightforward answer is that
this code is based on the StartAecDump and StopAecDump functions (located just
above), which do exactly the same thing.

However, that not really a good reason. I see two options for fixing this
problem:
- We could store the rtc::PlatformFile here in WebRtcVoiceEngine, and call the
ClosePlatformFile() whenever StopRtcEventLog() is called. The downside is that
it is also possible for the RtcEventLog object to stop logging due to a timeout
(there is an internal timer that will stop the log if it goes on for too long),
in which case the ClosePlatformFile() function will not be called.
- We could pass ownership of the rtc::PlatformFile to the RtcEventLog and make
sure the ClosePlatformFile() function is called whenever the log is stopped.
This would be a little awkward because only when a rtc::PlatformFile is used to
start the log should the ClosePlatformFile() function be called but not in cases
it was started using a path or file pointer.

I'm leaning toward the second option, do you agree?

[terelius, 2015/10/13 14:29:52]

Do we need any other way to pass the file, or can we always use
rtc::PlatformFile?

[the sun, 2015/10/13 14:29:54]

Yes, that seems like the best option. Make sure that Chromium expects the
ownership to be passed, even in the case of the call failing.

[ivoc, 2015/10/13 14:55:06]

@terelius: Possibly, but I don't know if using a rtc::PlatformFile would be
convenient for other applications than Chromium. We can remove the other options
later if they turn out to be useless. 

@the sun: I will keep that in mind when I work on the Chromium part :)

+      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;
+  }
+  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 1931 matching lines @@
       LOG(LS_WARNING) << "Unknown codec " << ToString(codec);
       return false;
     }
   }
   return true;
 }
 
 }  // namespace cricket
 
 #endif  // HAVE_WEBRTC_VOICE