Resolve cyclic dependency between audio network adaptor and event log api

webrtc/logging/rtc_event_log/rtc_event_log_unittest_helper.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_unittest_helper.h"
 
 #include <string.h>
 
 #include <string>
 
 #include "webrtc/base/checks.h"
+#include "webrtc/modules/audio_coding/audio_network_adaptor/include/audio_network_adaptor.h"
 #include "webrtc/test/gtest.h"
 #include "webrtc/test/testsupport/fileutils.h"
 
 // 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"
 #else
 #include "webrtc/logging/rtc_event_log/rtc_event_log.pb.h"
 #endif
 
@@ skipping 513 matching lines @@
   BandwidthUsage parsed_detector_state;
   parsed_log.GetDelayBasedBweUpdate(index, &parsed_bitrate,
                                     &parsed_detector_state);
   EXPECT_EQ(bitrate, parsed_bitrate);
   EXPECT_EQ(detector_state, parsed_detector_state);
 }
 
 void RtcEventLogTestHelper::VerifyAudioNetworkAdaptation(
     const ParsedRtcEventLog& parsed_log,
     size_t index,
-    const AudioNetworkAdaptor::EncoderRuntimeConfig& config) {
-  AudioNetworkAdaptor::EncoderRuntimeConfig parsed_config;
+    const AudioEncoderRuntimeConfig& config) {
+  AudioEncoderRuntimeConfig parsed_config;
   parsed_log.GetAudioNetworkAdaptation(index, &parsed_config);
   EXPECT_EQ(config.bitrate_bps, parsed_config.bitrate_bps);
   EXPECT_EQ(config.enable_dtx, parsed_config.enable_dtx);
   EXPECT_EQ(config.enable_fec, parsed_config.enable_fec);
   EXPECT_EQ(config.frame_length_ms, parsed_config.frame_length_ms);
   EXPECT_EQ(config.num_channels, parsed_config.num_channels);
   EXPECT_EQ(config.uplink_packet_loss_fraction,
             parsed_config.uplink_packet_loss_fraction);
 }
 
@@ skipping 70 matching lines @@
   ASSERT_TRUE(bwe_event.has_id());
   EXPECT_EQ(id, bwe_event.id());
   ASSERT_TRUE(bwe_event.has_result());
   EXPECT_EQ(GetProbeResultType(failure_reason), bwe_event.result());
   ASSERT_FALSE(bwe_event.has_bitrate_bps());
 
   // TODO(philipel): Verify the parser when parsing has been implemented.
 }
 
 }  // namespace webrtc