Replace Clock with timeutils in AudioEncoder.

webrtc/modules/audio_coding/audio_network_adaptor/audio_network_adaptor_impl.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/modules/audio_coding/audio_network_adaptor/audio_network_adaptor_impl.h"
 
 #include <utility>
 
 #include "webrtc/base/logging.h"
+#include "webrtc/base/timeutils.h"
 
 namespace webrtc {
 
 namespace {
 constexpr int kEventLogMinBitrateChangeBps = 5000;
 constexpr float kEventLogMinBitrateChangeFraction = 0.25;
 constexpr float kEventLogMinPacketLossChangeFraction = 0.5;
 }  // namespace
 
-AudioNetworkAdaptorImpl::Config::Config()
-    : event_log(nullptr), clock(nullptr){};
+AudioNetworkAdaptorImpl::Config::Config() : event_log(nullptr){};
 
 AudioNetworkAdaptorImpl::Config::~Config() = default;
 
 AudioNetworkAdaptorImpl::AudioNetworkAdaptorImpl(
     const Config& config,
     std::unique_ptr<ControllerManager> controller_manager,
     std::unique_ptr<DebugDumpWriter> debug_dump_writer)
     : config_(config),
       controller_manager_(std::move(controller_manager)),
       debug_dump_writer_(std::move(debug_dump_writer)),
@@ skipping 75 matching lines @@
   UpdateNetworkMetrics(network_metrics);
 }
 
 AudioEncoderRuntimeConfig AudioNetworkAdaptorImpl::GetEncoderRuntimeConfig() {
   AudioEncoderRuntimeConfig config;
   for (auto& controller :
        controller_manager_->GetSortedControllers(last_metrics_))
     controller->MakeDecision(&config);
 
   if (debug_dump_writer_)
-    debug_dump_writer_->DumpEncoderRuntimeConfig(
-        config, config_.clock->TimeInMilliseconds());
+    debug_dump_writer_->DumpEncoderRuntimeConfig(config, rtc::TimeMillis());
 
   if (event_log_writer_)
     event_log_writer_->MaybeLogEncoderConfig(config);
 
   return config;
 }
 
 void AudioNetworkAdaptorImpl::StartDebugDump(FILE* file_handle) {
   debug_dump_writer_ = DebugDumpWriter::Create(file_handle);
 }
 
 void AudioNetworkAdaptorImpl::StopDebugDump() {
   debug_dump_writer_.reset(nullptr);
 }
 
 void AudioNetworkAdaptorImpl::DumpNetworkMetrics() {
   if (debug_dump_writer_)
-    debug_dump_writer_->DumpNetworkMetrics(last_metrics_,
-                                           config_.clock->TimeInMilliseconds());
+    debug_dump_writer_->DumpNetworkMetrics(last_metrics_, rtc::TimeMillis());
 }
 
 void AudioNetworkAdaptorImpl::UpdateNetworkMetrics(
     const Controller::NetworkMetrics& network_metrics) {
   for (auto& controller : controller_manager_->GetControllers())
     controller->UpdateNetworkMetrics(network_metrics);
 }
 
 }  // namespace webrtc