Replace Clock with timeutils in AudioEncoder.
BUG=webrtc:7398
Review-Url: https://codereview.webrtc.org/2782563003
Cr-Commit-Position: refs/heads/master@{#17732}
diff --git a/webrtc/modules/audio_coding/audio_network_adaptor/controller_manager.cc b/webrtc/modules/audio_coding/audio_network_adaptor/controller_manager.cc
index 1e6aff1..ce4e2f9 100644
--- a/webrtc/modules/audio_coding/audio_network_adaptor/controller_manager.cc
+++ b/webrtc/modules/audio_coding/audio_network_adaptor/controller_manager.cc
@@ -14,6 +14,7 @@
#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"
@@ -21,7 +22,6 @@
#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()
@@ -41,8 +41,7 @@
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());
@@ -70,7 +69,7 @@
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(
@@ -186,11 +185,9 @@
} // 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;
@@ -203,8 +200,7 @@
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);
@@ -218,7 +214,7 @@
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(
@@ -262,7 +258,7 @@
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();
@@ -299,7 +295,7 @@
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_;