Resolve cyclic dependency between audio network adaptor and event log api

webrtc/modules/audio_coding/audio_network_adaptor/include/audio_network_adaptor.h

 /*
  *  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.
  */
 
 #ifndef WEBRTC_MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_INCLUDE_AUDIO_NETWORK_ADAPTOR_H_
 #define WEBRTC_MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_INCLUDE_AUDIO_NETWORK_ADAPTOR_H_
 
 #include "webrtc/base/optional.h"
 
 namespace webrtc {
 
+struct AudioEncoderRuntimeConfig {
+  AudioEncoderRuntimeConfig();
+  AudioEncoderRuntimeConfig(const AudioEncoderRuntimeConfig& other);
+  ~AudioEncoderRuntimeConfig();
+  rtc::Optional<int> bitrate_bps;
+  rtc::Optional<int> frame_length_ms;
+  // Note: This is what we tell the encoder. It doesn't have to reflect
+  // the actual NetworkMetrics; it's subject to our decision.
+  rtc::Optional<float> uplink_packet_loss_fraction;
+  rtc::Optional<bool> enable_fec;
+  rtc::Optional<bool> enable_dtx;
+
+  // Some encoders can encode fewer channels than the actual input to make
+  // better use of the bandwidth. |num_channels| sets the number of channels
+  // to encode.
+  rtc::Optional<size_t> num_channels;
+};
+
 // An AudioNetworkAdaptor optimizes the audio experience by suggesting a
 // suitable runtime configuration (bit rate, frame length, FEC, etc.) to the
 // encoder based on network metrics.
 class AudioNetworkAdaptor {
  public:
-  struct EncoderRuntimeConfig {
-    EncoderRuntimeConfig();
-    EncoderRuntimeConfig(const EncoderRuntimeConfig& other);
-    ~EncoderRuntimeConfig();
-    rtc::Optional<int> bitrate_bps;
-    rtc::Optional<int> frame_length_ms;
-    // Note: This is what we tell the encoder. It doesn't have to reflect
-    // the actual NetworkMetrics; it's subject to our decision.
-    rtc::Optional<float> uplink_packet_loss_fraction;
-    rtc::Optional<bool> enable_fec;
-    rtc::Optional<bool> enable_dtx;
-
-    // Some encoders can encode fewer channels than the actual input to make
-    // better use of the bandwidth. |num_channels| sets the number of channels
-    // to encode.
-    rtc::Optional<size_t> num_channels;
-  };
 
   virtual ~AudioNetworkAdaptor() = default;
 
   virtual void SetUplinkBandwidth(int uplink_bandwidth_bps) = 0;
 
   virtual void SetUplinkPacketLossFraction(
       float uplink_packet_loss_fraction) = 0;
 
   virtual void SetUplinkRecoverablePacketLossFraction(
       float uplink_recoverable_packet_loss_fraction) = 0;
 
   virtual void SetRtt(int rtt_ms) = 0;
 
   virtual void SetTargetAudioBitrate(int target_audio_bitrate_bps) = 0;
 
   virtual void SetOverhead(size_t overhead_bytes_per_packet) = 0;
 
-  virtual EncoderRuntimeConfig GetEncoderRuntimeConfig() = 0;
+  virtual AudioEncoderRuntimeConfig GetEncoderRuntimeConfig() = 0;
 
   virtual void StartDebugDump(FILE* file_handle) = 0;
 
   virtual void StopDebugDump() = 0;
 };
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_AUDIO_CODING_AUDIO_NETWORK_ADAPTOR_INCLUDE_AUDIO_NETWORK_ADAPTOR_H_