Resolve cyclic dependency between audio network adaptor and event log api

webrtc/modules/audio_coding/audio_network_adaptor/event_log_writer_unittest.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 25 matching lines @@
          arg.uplink_packet_loss_fraction ==
              config.uplink_packet_loss_fraction &&
          arg.enable_fec == config.enable_fec &&
          arg.enable_dtx == config.enable_dtx &&
          arg.num_channels == config.num_channels;
 }
 
 struct EventLogWriterStates {
   std::unique_ptr<EventLogWriter> event_log_writer;
   std::unique_ptr<testing::StrictMock<MockRtcEventLog>> event_log;
-  AudioNetworkAdaptor::EncoderRuntimeConfig runtime_config;
+  AudioEncoderRuntimeConfig runtime_config;
 };
 
 EventLogWriterStates CreateEventLogWriter() {
   EventLogWriterStates state;
   state.event_log.reset(new testing::StrictMock<MockRtcEventLog>());
   state.event_log_writer.reset(new EventLogWriter(
       state.event_log.get(), kMinBitrateChangeBps, kMinBitrateChangeFraction,
       kMinPacketLossChangeFraction));
   state.runtime_config.bitrate_bps = rtc::Optional<int>(kHighBitrateBps);
   state.runtime_config.frame_length_ms = rtc::Optional<int>(kFrameLengthMs);
@@ skipping 199 matching lines @@
       *state.event_log,
       LogAudioNetworkAdaptation(EncoderRuntimeConfigIs(state.runtime_config)))
       .Times(1);
   for (int bitrate_bps = kHighBitrateBps;
        bitrate_bps <= kHighBitrateBps + kMinBitrateChangeBps; bitrate_bps++) {
     state.runtime_config.bitrate_bps = rtc::Optional<int>(bitrate_bps);
     state.event_log_writer->MaybeLogEncoderConfig(state.runtime_config);
   }
 }
 }  // namespace webrtc