Move RtcEventLog object from inside VoiceEngine to Call.

webrtc/audio/audio_receive_stream.cc

 /*
  *  Copyright (c) 2015 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 63 matching lines @@
     ss << ", sync_group: " << sync_group;
   }
   ss << '}';
   return ss.str();
 }
 
 namespace internal {
 AudioReceiveStream::AudioReceiveStream(
     CongestionController* congestion_controller,
     const webrtc::AudioReceiveStream::Config& config,
-    const rtc::scoped_refptr<webrtc::AudioState>& audio_state)
+    const rtc::scoped_refptr<webrtc::AudioState>& audio_state,
+    webrtc::RtcEventLog* event_log)
     : config_(config),
       audio_state_(audio_state),
       rtp_header_parser_(RtpHeaderParser::Create()) {
   LOG(LS_INFO) << "AudioReceiveStream: " << config_.ToString();
   RTC_DCHECK_NE(config_.voe_channel_id, -1);
   RTC_DCHECK(audio_state_.get());
   RTC_DCHECK(congestion_controller);
   RTC_DCHECK(rtp_header_parser_);
 
   VoiceEngineImpl* voe_impl = static_cast<VoiceEngineImpl*>(voice_engine());
   channel_proxy_ = voe_impl->GetChannelProxy(config_.voe_channel_id);
+  channel_proxy_->SetRtcEventLog(event_log);
   channel_proxy_->SetLocalSSRC(config.rtp.local_ssrc);
   // TODO(solenberg): Config NACK history window (which is a packet count),
   // using the actual packet size for the configured codec.
   channel_proxy_->SetNACKStatus(config_.rtp.nack.rtp_history_ms != 0,
                                 config_.rtp.nack.rtp_history_ms / 20);
 
   // TODO(ossu): This is where we'd like to set the decoder factory to
   // use. However, since it needs to be included when constructing Channel, we
   // cannot do that until we're able to move Channel ownership into the
   // Audio{Send,Receive}Streams.  The best we can do is check that we're not
@@ skipping 31 matching lines @@
     remote_bitrate_estimator_ =
         congestion_controller->GetRemoteBitrateEstimator(true);
   }
 }
 
 AudioReceiveStream::~AudioReceiveStream() {
   RTC_DCHECK(thread_checker_.CalledOnValidThread());
   LOG(LS_INFO) << "~AudioReceiveStream: " << config_.ToString();
   channel_proxy_->DeRegisterExternalTransport();
   channel_proxy_->ResetCongestionControlObjects();
+  channel_proxy_->SetRtcEventLog(nullptr);
   if (remote_bitrate_estimator_) {
     remote_bitrate_estimator_->RemoveStream(config_.rtp.remote_ssrc);
   }
 }
 
 void AudioReceiveStream::Start() {
   RTC_DCHECK(thread_checker_.CalledOnValidThread());
 }
 
 void AudioReceiveStream::Stop() {
@@ skipping 105 matching lines @@
 
 VoiceEngine* AudioReceiveStream::voice_engine() const {
   internal::AudioState* audio_state =
       static_cast<internal::AudioState*>(audio_state_.get());
   VoiceEngine* voice_engine = audio_state->voice_engine();
   RTC_DCHECK(voice_engine);
   return voice_engine;
 }
 }  // namespace internal
 }  // namespace webrtc