Resolve cyclic dependency between audio network adaptor and event log api

webrtc/modules/audio_coding/audio_network_adaptor/channel_controller.cc

 /*
  *  Copyright (c) 2016 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 23 matching lines @@
 }
 
 ChannelController::~ChannelController() = default;
 
 void ChannelController::UpdateNetworkMetrics(
     const NetworkMetrics& network_metrics) {
   if (network_metrics.uplink_bandwidth_bps)
     uplink_bandwidth_bps_ = network_metrics.uplink_bandwidth_bps;
 }
 
-void ChannelController::MakeDecision(
-    AudioNetworkAdaptor::EncoderRuntimeConfig* config) {
+void ChannelController::MakeDecision(AudioEncoderRuntimeConfig* config) {
   // Decision on |num_channels| should not have been made.
   RTC_DCHECK(!config->num_channels);
 
   if (uplink_bandwidth_bps_) {
     if (channels_to_encode_ == 2 &&
         *uplink_bandwidth_bps_ <= config_.channel_2_to_1_bandwidth_bps) {
       channels_to_encode_ = 1;
     } else if (channels_to_encode_ == 1 &&
                *uplink_bandwidth_bps_ >= config_.channel_1_to_2_bandwidth_bps) {
       channels_to_encode_ =
           std::min(static_cast<size_t>(2), config_.num_encoder_channels);
     }
   }
   config->num_channels = rtc::Optional<size_t>(channels_to_encode_);
 }
 
 }  // namespace webrtc