Resolve cyclic dependency between audio network adaptor and event log api

webrtc/logging/rtc_event_log/rtc_event_log_unittest.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 <map>
 #include <memory>
 #include <string>
 #include <utility>
 #include <vector>
 
 #include "webrtc/base/buffer.h"
 #include "webrtc/base/checks.h"
 #include "webrtc/base/fakeclock.h"
 #include "webrtc/base/random.h"
 #include "webrtc/call/call.h"
 #include "webrtc/logging/rtc_event_log/rtc_event_log.h"
 #include "webrtc/logging/rtc_event_log/rtc_event_log_parser.h"
 #include "webrtc/logging/rtc_event_log/rtc_event_log_unittest_helper.h"
+#include "webrtc/modules/audio_coding/audio_network_adaptor/include/audio_network_adaptor.h"
 #include "webrtc/modules/rtp_rtcp/source/rtcp_packet.h"
 #include "webrtc/modules/rtp_rtcp/source/rtcp_packet/sender_report.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_header_extension.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_header_extensions.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_packet_to_send.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
@@ skipping 185 matching lines @@
   config->rtp.ssrc = prng->Rand<uint32_t>();
   // Add header extensions.
   for (unsigned i = 0; i < kNumExtensions; i++) {
     if (extensions_bitvector & (1u << i)) {
       config->rtp.extensions.push_back(
           RtpExtension(kExtensionNames[i], prng->Rand<int>()));
     }
   }
 }
 
-void GenerateAudioNetworkAdaptation(
-    uint32_t extensions_bitvector,
-    AudioNetworkAdaptor::EncoderRuntimeConfig* config,
-    Random* prng) {
+void GenerateAudioNetworkAdaptation(uint32_t extensions_bitvector,
+                                    AudioEncoderRuntimeConfig* config,
+                                    Random* prng) {
   config->bitrate_bps = rtc::Optional<int>(prng->Rand(0, 3000000));
   config->enable_fec = rtc::Optional<bool>(prng->Rand<bool>());
   config->enable_dtx = rtc::Optional<bool>(prng->Rand<bool>());
   config->frame_length_ms = rtc::Optional<int>(prng->Rand(10, 120));
   config->num_channels = rtc::Optional<size_t>(prng->Rand(1, 2));
   config->uplink_packet_loss_fraction =
       rtc::Optional<float>(prng->Rand<float>());
 }
 
 // Test for the RtcEventLog class. Dumps some RTP packets and other events
@@ skipping 608 matching lines @@
     GenerateAudioNetworkAdaptation(extensions_bitvector, &config, &prng);
   }
   void LogConfig(RtcEventLog* event_log) override {
     event_log->LogAudioNetworkAdaptation(config);
   }
   void VerifyConfig(const ParsedRtcEventLog& parsed_log,
                     size_t index) override {
     RtcEventLogTestHelper::VerifyAudioNetworkAdaptation(parsed_log, index,
                                                         config);
   }
-  AudioNetworkAdaptor::EncoderRuntimeConfig config;
+  AudioEncoderRuntimeConfig config;
 };
 
 TEST(RtcEventLogTest, LogAudioReceiveConfig) {
   AudioReceiveConfigReadWriteTest test;
   test.DoTest();
 }
 
 TEST(RtcEventLogTest, LogAudioSendConfig) {
   AudioSendConfigReadWriteTest test;
   test.DoTest();
 }
 
 TEST(RtcEventLogTest, LogVideoReceiveConfig) {
   VideoReceiveConfigReadWriteTest test;
   test.DoTest();
 }
 
 TEST(RtcEventLogTest, LogVideoSendConfig) {
   VideoSendConfigReadWriteTest test;
   test.DoTest();
 }
 
 TEST(RtcEventLogTest, LogAudioNetworkAdaptation) {
   AudioNetworkAdaptationReadWriteTest test;
   test.DoTest();
 }
 
 }  // namespace webrtc