Resolve cyclic dependency between audio network adaptor and event log api

webrtc/logging/rtc_event_log/rtc_event_log_parser.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.
  */
 
 #include "webrtc/logging/rtc_event_log/rtc_event_log_parser.h"
 
 #include <stdint.h>
 #include <string.h>
 
 #include <algorithm>
 #include <fstream>
 #include <istream>
 #include <utility>
 
 #include "webrtc/base/checks.h"
 #include "webrtc/base/logging.h"
 #include "webrtc/call/call.h"
 #include "webrtc/logging/rtc_event_log/rtc_event_log.h"
+#include "webrtc/modules/audio_coding/audio_network_adaptor/include/audio_network_adaptor.h"
 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
 
 namespace webrtc {
 
 namespace {
 MediaType GetRuntimeMediaType(rtclog::MediaType media_type) {
   switch (media_type) {
     case rtclog::MediaType::ANY:
       return MediaType::ANY;
     case rtclog::MediaType::AUDIO:
@@ skipping 469 matching lines @@
     *bitrate_bps = delay_event.bitrate_bps();
   }
   RTC_CHECK(delay_event.has_detector_state());
   if (detector_state != nullptr) {
     *detector_state = GetRuntimeDetectorState(delay_event.detector_state());
   }
 }
 
 void ParsedRtcEventLog::GetAudioNetworkAdaptation(
     size_t index,
-    AudioNetworkAdaptor::EncoderRuntimeConfig* config) const {
+    AudioEncoderRuntimeConfig* config) const {
   RTC_CHECK_LT(index, GetNumberOfEvents());
   const rtclog::Event& event = events_[index];
   RTC_CHECK(event.has_type());
   RTC_CHECK_EQ(event.type(), rtclog::Event::AUDIO_NETWORK_ADAPTATION_EVENT);
   RTC_CHECK(event.has_audio_network_adaptation());
   const rtclog::AudioNetworkAdaptation& ana_event =
       event.audio_network_adaptation();
   if (ana_event.has_bitrate_bps())
     config->bitrate_bps = rtc::Optional<int>(ana_event.bitrate_bps());
   if (ana_event.has_enable_fec())
     config->enable_fec = rtc::Optional<bool>(ana_event.enable_fec());
   if (ana_event.has_enable_dtx())
     config->enable_dtx = rtc::Optional<bool>(ana_event.enable_dtx());
   if (ana_event.has_frame_length_ms())
     config->frame_length_ms = rtc::Optional<int>(ana_event.frame_length_ms());
   if (ana_event.has_num_channels())
     config->num_channels = rtc::Optional<size_t>(ana_event.num_channels());
   if (ana_event.has_uplink_packet_loss_fraction())
     config->uplink_packet_loss_fraction =
         rtc::Optional<float>(ana_event.uplink_packet_loss_fraction());
 }
 
 }  // namespace webrtc