Resolve cyclic dependency between audio network adaptor and event log api

webrtc/logging/rtc_event_log/rtc_event_log.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.
  */
 
 #include "webrtc/logging/rtc_event_log/rtc_event_log.h"
 
 #include <limits>
 #include <vector>
 
 #include "webrtc/base/checks.h"
 #include "webrtc/base/constructormagic.h"
 #include "webrtc/base/event.h"
 #include "webrtc/base/swap_queue.h"
 #include "webrtc/base/thread_checker.h"
 #include "webrtc/base/timeutils.h"
 #include "webrtc/call/call.h"
 #include "webrtc/logging/rtc_event_log/rtc_event_log_helper_thread.h"
+#include "webrtc/modules/audio_coding/audio_network_adaptor/include/audio_network_adaptor.h"
 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
 #include "webrtc/modules/rtp_rtcp/source/byte_io.h"
 #include "webrtc/modules/rtp_rtcp/source/rtcp_packet/app.h"
 #include "webrtc/modules/rtp_rtcp/source/rtcp_packet/bye.h"
 #include "webrtc/modules/rtp_rtcp/source/rtcp_packet/common_header.h"
 #include "webrtc/modules/rtp_rtcp/source/rtcp_packet/extended_jitter_report.h"
 #include "webrtc/modules/rtp_rtcp/source/rtcp_packet/extended_reports.h"
+#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/psfb.h"
 #include "webrtc/modules/rtp_rtcp/source/rtcp_packet/receiver_report.h"
-#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/psfb.h"
 #include "webrtc/modules/rtp_rtcp/source/rtcp_packet/rtpfb.h"
 #include "webrtc/modules/rtp_rtcp/source/rtcp_packet/sdes.h"
 #include "webrtc/modules/rtp_rtcp/source/rtcp_packet/sender_report.h"
 #include "webrtc/system_wrappers/include/file_wrapper.h"
 #include "webrtc/system_wrappers/include/logging.h"
 
 #ifdef ENABLE_RTC_EVENT_LOG
 // Files generated at build-time by the protobuf compiler.
 #ifdef WEBRTC_ANDROID_PLATFORM_BUILD
 #include "external/webrtc/webrtc/logging/rtc_event_log/rtc_event_log.pb.h"
@@ skipping 35 matching lines @@
                      MediaType media_type,
                      const uint8_t* packet,
                      size_t length) override;
   void LogAudioPlayout(uint32_t ssrc) override;
   void LogLossBasedBweUpdate(int32_t bitrate_bps,
                              uint8_t fraction_loss,
                              int32_t total_packets) override;
   void LogDelayBasedBweUpdate(int32_t bitrate_bps,
                               BandwidthUsage detector_state) override;
   void LogAudioNetworkAdaptation(
-      const AudioNetworkAdaptor::EncoderRuntimeConfig& config) override;
+      const AudioEncoderRuntimeConfig& config) override;
   void LogProbeClusterCreated(int id,
                               int bitrate_bps,
                               int min_probes,
                               int min_bytes) override;
   void LogProbeResultSuccess(int id, int bitrate_bps) override;
   void LogProbeResultFailure(int id,
                              ProbeFailureReason failure_reason) override;
 
  private:
   void StoreEvent(std::unique_ptr<rtclog::Event>* event);
@@ skipping 398 matching lines @@
   std::unique_ptr<rtclog::Event> event(new rtclog::Event());
   event->set_timestamp_us(rtc::TimeMicros());
   event->set_type(rtclog::Event::DELAY_BASED_BWE_UPDATE);
   auto bwe_event = event->mutable_delay_based_bwe_update();
   bwe_event->set_bitrate_bps(bitrate_bps);
   bwe_event->set_detector_state(ConvertDetectorState(detector_state));
   StoreEvent(&event);
 }
 
 void RtcEventLogImpl::LogAudioNetworkAdaptation(
-    const AudioNetworkAdaptor::EncoderRuntimeConfig& config) {
+    const AudioEncoderRuntimeConfig& config) {
   std::unique_ptr<rtclog::Event> event(new rtclog::Event());
   event->set_timestamp_us(rtc::TimeMicros());
   event->set_type(rtclog::Event::AUDIO_NETWORK_ADAPTATION_EVENT);
   auto audio_network_adaptation = event->mutable_audio_network_adaptation();
   if (config.bitrate_bps)
     audio_network_adaptation->set_bitrate_bps(*config.bitrate_bps);
   if (config.frame_length_ms)
     audio_network_adaptation->set_frame_length_ms(*config.frame_length_ms);
   if (config.uplink_packet_loss_fraction) {
     audio_network_adaptation->set_uplink_packet_loss_fraction(
@@ skipping 91 matching lines @@
 #else
   return std::unique_ptr<RtcEventLog>(new RtcEventLogNullImpl());
 #endif  // ENABLE_RTC_EVENT_LOG
 }
 
 std::unique_ptr<RtcEventLog> RtcEventLog::CreateNull() {
   return std::unique_ptr<RtcEventLog>(new RtcEventLogNullImpl());
 }
 
 }  // namespace webrtc