Move RtcEventLog object from inside VoiceEngine to Call.

webrtc/media/engine/webrtcvoiceengine.cc

 /*
  *  Copyright (c) 2004 The WebRTC project authors. All Rights Reserved.
  *
  *  Use of this source code is governed by a BSD-style license
  *  that can be found in the LICENSE file in the root of the source
  *  tree. An additional intellectual property rights grant can be found
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
@@ skipping 12 matching lines @@
 
 #include "webrtc/audio/audio_sink.h"
 #include "webrtc/base/arraysize.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/media/base/audioframe.h"
 #include "webrtc/media/base/audiorenderer.h"
 #include "webrtc/media/base/constants.h"
 #include "webrtc/media/base/streamparams.h"
 #include "webrtc/media/engine/webrtcmediaengine.h"
 #include "webrtc/media/engine/webrtcvoe.h"
 #include "webrtc/modules/audio_coding/acm2/rent_a_codec.h"
 #include "webrtc/modules/audio_processing/include/audio_processing.h"
 #include "webrtc/system_wrappers/include/field_trial.h"
@@ skipping 1059 matching lines @@
   if (is_dumping_aec_) {
     // Stop dumping AEC when we are dumping.
     if (voe_wrapper_->base()->audio_processing()->StopDebugRecording() !=
         webrtc::AudioProcessing::kNoError) {
       LOG_RTCERR0(StopDebugRecording);
     }
     is_dumping_aec_ = false;
   }
 }
 
-bool WebRtcVoiceEngine::StartRtcEventLog(rtc::PlatformFile file) {
-  RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
-  webrtc::RtcEventLog* event_log = voe_wrapper_->codec()->GetEventLog();
-  if (event_log) {
-    return event_log->StartLogging(file);
-  }
-  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() {
   RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
   return voe_wrapper_->base()->CreateChannel(voe_config_);
 }
 
 class WebRtcVoiceMediaChannel::WebRtcAudioSendStream
     : public AudioRenderer::Sink {
  public:
   WebRtcAudioSendStream(int ch, webrtc::AudioTransport* voe_audio_transport,
                         uint32_t ssrc, const std::string& c_name,
@@ skipping 1377 matching lines @@
     }
   } else {
     LOG(LS_INFO) << "Stopping playout for channel #" << channel;
     engine()->voe()->base()->StopPlayout(channel);
   }
   return true;
 }
 }  // namespace cricket
 
 #endif  // HAVE_WEBRTC_VOICE