Move RtcEventLog object from inside VoiceEngine to Call.

webrtc/audio/audio_receive_stream.cc

Index: webrtc/audio/audio_receive_stream.cc
diff --git a/webrtc/audio/audio_receive_stream.cc b/webrtc/audio/audio_receive_stream.cc
index 9c253894719278a48b354763b4aef0a3235d0817..c0a29b24b588f846ae3a608933ac59463b4b025a 100644
--- a/webrtc/audio/audio_receive_stream.cc
+++ b/webrtc/audio/audio_receive_stream.cc
@@ -82,7 +82,8 @@ 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()) {
@@ -94,6 +95,7 @@ AudioReceiveStream::AudioReceiveStream(
 
   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);
   for (const auto& extension : config.rtp.extensions) {
     if (extension.name == RtpExtension::kAudioLevel) {
@@ -128,6 +130,7 @@ AudioReceiveStream::~AudioReceiveStream() {
   RTC_DCHECK(thread_checker_.CalledOnValidThread());
   LOG(LS_INFO) << "~AudioReceiveStream: " << config_.ToString();
   channel_proxy_->ResetCongestionControlObjects();
+  channel_proxy_->SetRtcEventLog(nullptr);
   if (remote_bitrate_estimator_) {
     remote_bitrate_estimator_->RemoveStream(config_.rtp.remote_ssrc);
   }