Replace Clock with timeutils in AudioEncoder.

webrtc/modules/audio_coding/audio_network_adaptor/controller_manager.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/controller_manager.h"
 
 #include <cmath>
 #include <utility>
 
 #include "webrtc/base/ignore_wundef.h"
+#include "webrtc/base/timeutils.h"
 #include "webrtc/modules/audio_coding/audio_network_adaptor/bitrate_controller.h"
 #include "webrtc/modules/audio_coding/audio_network_adaptor/channel_controller.h"
 #include "webrtc/modules/audio_coding/audio_network_adaptor/dtx_controller.h"
 #include "webrtc/modules/audio_coding/audio_network_adaptor/fec_controller_plr_based.h"
 #include "webrtc/modules/audio_coding/audio_network_adaptor/fec_controller_rplr_based.h"
 #include "webrtc/modules/audio_coding/audio_network_adaptor/frame_length_controller.h"
 #include "webrtc/modules/audio_coding/audio_network_adaptor/util/threshold_curve.h"
-#include "webrtc/system_wrappers/include/clock.h"
 
 #ifdef WEBRTC_AUDIO_NETWORK_ADAPTOR_DEBUG_DUMP
 RTC_PUSH_IGNORING_WUNDEF()
 #ifdef WEBRTC_ANDROID_PLATFORM_BUILD
 #include "external/webrtc/webrtc/modules/audio_coding/audio_network_adaptor/config.pb.h"
 #else
 #include "webrtc/modules/audio_coding/audio_network_adaptor/config.pb.h"
 #endif
 RTC_POP_IGNORING_WUNDEF()
 #endif
 
 namespace webrtc {
 
 namespace {
 
 #ifdef WEBRTC_AUDIO_NETWORK_ADAPTOR_DEBUG_DUMP
 
 std::unique_ptr<FecControllerPlrBased> CreateFecControllerPlrBased(
     const audio_network_adaptor::config::FecController& config,
-    bool initial_fec_enabled,
-    const Clock* clock) {
+    bool initial_fec_enabled) {
   RTC_CHECK(config.has_fec_enabling_threshold());
   RTC_CHECK(config.has_fec_disabling_threshold());
   RTC_CHECK(config.has_time_constant_ms());
 
   auto& fec_enabling_threshold = config.fec_enabling_threshold();
   RTC_CHECK(fec_enabling_threshold.has_low_bandwidth_bps());
   RTC_CHECK(fec_enabling_threshold.has_low_bandwidth_packet_loss());
   RTC_CHECK(fec_enabling_threshold.has_high_bandwidth_bps());
   RTC_CHECK(fec_enabling_threshold.has_high_bandwidth_packet_loss());
 
   auto& fec_disabling_threshold = config.fec_disabling_threshold();
   RTC_CHECK(fec_disabling_threshold.has_low_bandwidth_bps());
   RTC_CHECK(fec_disabling_threshold.has_low_bandwidth_packet_loss());
   RTC_CHECK(fec_disabling_threshold.has_high_bandwidth_bps());
   RTC_CHECK(fec_disabling_threshold.has_high_bandwidth_packet_loss());
 
   return std::unique_ptr<FecControllerPlrBased>(
       new FecControllerPlrBased(FecControllerPlrBased::Config(
           initial_fec_enabled,
           ThresholdCurve(fec_enabling_threshold.low_bandwidth_bps(),
                          fec_enabling_threshold.low_bandwidth_packet_loss(),
                          fec_enabling_threshold.high_bandwidth_bps(),
                          fec_enabling_threshold.high_bandwidth_packet_loss()),
           ThresholdCurve(fec_disabling_threshold.low_bandwidth_bps(),
                          fec_disabling_threshold.low_bandwidth_packet_loss(),
                          fec_disabling_threshold.high_bandwidth_bps(),
                          fec_disabling_threshold.high_bandwidth_packet_loss()),
-          config.time_constant_ms(), clock)));
+          config.time_constant_ms())));
 }
 
 std::unique_ptr<FecControllerRplrBased> CreateFecControllerRplrBased(
     const audio_network_adaptor::config::FecControllerRplrBased& config,
     bool initial_fec_enabled) {
   RTC_CHECK(config.has_fec_enabling_threshold());
   RTC_CHECK(config.has_fec_disabling_threshold());
 
   auto& fec_enabling_threshold = config.fec_enabling_threshold();
   RTC_CHECK(fec_enabling_threshold.has_low_bandwidth_bps());
@@ skipping 95 matching lines @@
     int initial_bitrate_bps,
     int initial_frame_length_ms) {
   return std::unique_ptr<BitrateController>(new BitrateController(
       BitrateController::Config(initial_bitrate_bps, initial_frame_length_ms)));
 }
 #endif  // WEBRTC_AUDIO_NETWORK_ADAPTOR_DEBUG_DUMP
 
 }  // namespace
 
 ControllerManagerImpl::Config::Config(int min_reordering_time_ms,
-                                      float min_reordering_squared_distance,
-                                      const Clock* clock)
+                                      float min_reordering_squared_distance)
     : min_reordering_time_ms(min_reordering_time_ms),
-      min_reordering_squared_distance(min_reordering_squared_distance),
-      clock(clock) {}
+      min_reordering_squared_distance(min_reordering_squared_distance) {}
 
 ControllerManagerImpl::Config::~Config() = default;
 
 std::unique_ptr<ControllerManager> ControllerManagerImpl::Create(
     const ProtoString& config_string,
     size_t num_encoder_channels,
     rtc::ArrayView<const int> encoder_frame_lengths_ms,
     int min_encoder_bitrate_bps,
     size_t intial_channels_to_encode,
     int initial_frame_length_ms,
     int initial_bitrate_bps,
     bool initial_fec_enabled,
-    bool initial_dtx_enabled,
-    const Clock* clock) {
+    bool initial_dtx_enabled) {
 #ifdef WEBRTC_AUDIO_NETWORK_ADAPTOR_DEBUG_DUMP
   audio_network_adaptor::config::ControllerManager controller_manager_config;
   controller_manager_config.ParseFromString(config_string);
 
   std::vector<std::unique_ptr<Controller>> controllers;
   std::map<const Controller*, std::pair<int, float>> chracteristic_points;
 
   for (int i = 0; i < controller_manager_config.controllers_size(); ++i) {
     auto& controller_config = controller_manager_config.controllers(i);
     std::unique_ptr<Controller> controller;
     switch (controller_config.controller_case()) {
       case audio_network_adaptor::config::Controller::kFecController:
         controller = CreateFecControllerPlrBased(
-            controller_config.fec_controller(), initial_fec_enabled, clock);
+            controller_config.fec_controller(), initial_fec_enabled);
         break;
       case audio_network_adaptor::config::Controller::kFecControllerRplrBased:
         controller = CreateFecControllerRplrBased(
             controller_config.fec_controller_rplr_based(), initial_fec_enabled);
         break;
       case audio_network_adaptor::config::Controller::kFrameLengthController:
         controller = CreateFrameLengthController(
             controller_config.frame_length_controller(),
             encoder_frame_lengths_ms, initial_frame_length_ms,
             min_encoder_bitrate_bps);
@@ skipping 23 matching lines @@
           characteristic_point.uplink_packet_loss_fraction());
     }
     controllers.push_back(std::move(controller));
   }
 
   RTC_CHECK(controller_manager_config.has_min_reordering_time_ms());
   RTC_CHECK(controller_manager_config.has_min_reordering_squared_distance());
   return std::unique_ptr<ControllerManagerImpl>(new ControllerManagerImpl(
       ControllerManagerImpl::Config(
           controller_manager_config.min_reordering_time_ms(),
-          controller_manager_config.min_reordering_squared_distance(), clock),
+          controller_manager_config.min_reordering_squared_distance()),
       std::move(controllers), chracteristic_points));
 #else
   RTC_NOTREACHED();
   return nullptr;
 #endif  // WEBRTC_AUDIO_NETWORK_ADAPTOR_DEBUG_DUMP
 }
 
 ControllerManagerImpl::ControllerManagerImpl(const Config& config)
     : ControllerManagerImpl(
           config,
@@ skipping 16 matching lines @@
     controller_scoring_points_.insert(std::make_pair(
         controller_point.first, ScoringPoint(controller_point.second.first,
                                              controller_point.second.second)));
   }
 }
 
 ControllerManagerImpl::~ControllerManagerImpl() = default;
 
 std::vector<Controller*> ControllerManagerImpl::GetSortedControllers(
     const Controller::NetworkMetrics& metrics) {
-  int64_t now_ms = config_.clock->TimeInMilliseconds();
+  int64_t now_ms = rtc::TimeMillis();
 
   if (!metrics.uplink_bandwidth_bps || !metrics.uplink_packet_loss_fraction)
     return sorted_controllers_;
 
   if (last_reordering_time_ms_ &&
       now_ms - *last_reordering_time_ms_ < config_.min_reordering_time_ms)
     return sorted_controllers_;
 
   ScoringPoint scoring_point(*metrics.uplink_bandwidth_bps,
                              *metrics.uplink_packet_loss_fraction);
@@ skipping 73 matching lines @@
       NormalizeUplinkBandwidth(scoring_point.uplink_bandwidth_bps) -
       NormalizeUplinkBandwidth(uplink_bandwidth_bps);
   float diff_normalized_packet_loss =
       NormalizePacketLossFraction(scoring_point.uplink_packet_loss_fraction) -
       NormalizePacketLossFraction(uplink_packet_loss_fraction);
   return std::pow(diff_normalized_bitrate_bps, 2) +
          std::pow(diff_normalized_packet_loss, 2);
 }
 
 }  // namespace webrtc