Set the event log in Channel from AudioSendStream

webrtc/audio/audio_send_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 45 matching lines @@
   ss << '}';
   return ss.str();
 }
 
 namespace internal {
 AudioSendStream::AudioSendStream(
     const webrtc::AudioSendStream::Config& config,
     const rtc::scoped_refptr<webrtc::AudioState>& audio_state,
     rtc::TaskQueue* worker_queue,
     CongestionController* congestion_controller,
-    BitrateAllocator* bitrate_allocator)
+    BitrateAllocator* bitrate_allocator,
+    RtcEventLog* event_log)
     : worker_queue_(worker_queue),
       config_(config),
       audio_state_(audio_state),
       bitrate_allocator_(bitrate_allocator) {
   LOG(LS_INFO) << "AudioSendStream: " << config_.ToString();
   RTC_DCHECK_NE(config_.voe_channel_id, -1);
   RTC_DCHECK(audio_state_.get());
   RTC_DCHECK(congestion_controller);
 
   VoiceEngineImpl* voe_impl = static_cast<VoiceEngineImpl*>(voice_engine());
   channel_proxy_ = voe_impl->GetChannelProxy(config_.voe_channel_id);
+  channel_proxy_->SetRtcEventLog(event_log);
   channel_proxy_->RegisterSenderCongestionControlObjects(
       congestion_controller->pacer(),
       congestion_controller->GetTransportFeedbackObserver(),
       congestion_controller->packet_router());
   channel_proxy_->SetRTCPStatus(true);
   channel_proxy_->SetLocalSSRC(config.rtp.ssrc);
   channel_proxy_->SetRTCP_CNAME(config.rtp.c_name);
   // 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,
@@ skipping 12 matching lines @@
       RTC_NOTREACHED() << "Registering unsupported RTP extension.";
     }
   }
 }
 
 AudioSendStream::~AudioSendStream() {
   RTC_DCHECK(thread_checker_.CalledOnValidThread());
   LOG(LS_INFO) << "~AudioSendStream: " << config_.ToString();
   channel_proxy_->DeRegisterExternalTransport();
   channel_proxy_->ResetCongestionControlObjects();
+  channel_proxy_->SetRtcEventLog(nullptr);
 }
 
 void AudioSendStream::Start() {
   RTC_DCHECK(thread_checker_.CalledOnValidThread());
   if (config_.min_bitrate_kbps != -1 && config_.max_bitrate_kbps != -1) {
     RTC_DCHECK_GE(config_.max_bitrate_kbps, config_.min_bitrate_kbps);
     rtc::Event thread_sync_event(false /* manual_reset */, false);
     worker_queue_->PostTask([this, &thread_sync_event] {
       bitrate_allocator_->AddObserver(this, config_.min_bitrate_kbps * 1000,
                                       config_.max_bitrate_kbps * 1000, 0, true);
@@ skipping 157 matching lines @@
 
 VoiceEngine* AudioSendStream::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