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317a1f09ed
WebRTC is now using C++14 so there is no need to use the Abseil version of std::make_unique. This CL has been created with the following steps: git grep -l absl::make_unique | sort | uniq > /tmp/make_unique.txt git grep -l absl::WrapUnique | sort | uniq > /tmp/wrap_unique.txt git grep -l "#include <memory>" | sort | uniq > /tmp/memory.txt diff --new-line-format="" --unchanged-line-format="" \ /tmp/make_unique.txt /tmp/wrap_unique.txt | sort | \ uniq > /tmp/only_make_unique.txt diff --new-line-format="" --unchanged-line-format="" \ /tmp/only_make_unique.txt /tmp/memory.txt | \ xargs grep -l "absl/memory" > /tmp/add-memory.txt git grep -l "\babsl::make_unique\b" | \ xargs sed -i "s/\babsl::make_unique\b/std::make_unique/g" git checkout PRESUBMIT.py abseil-in-webrtc.md cat /tmp/add-memory.txt | \ xargs sed -i \ 's/#include "absl\/memory\/memory.h"/#include <memory>/g' git cl format # Manual fix order of the new inserted #include <memory> cat /tmp/only_make_unique | xargs grep -l "#include <memory>" | \ xargs sed -i '/#include "absl\/memory\/memory.h"/d' git ls-files | grep BUILD.gn | \ xargs sed -i '/\/\/third_party\/abseil-cpp\/absl\/memory/d' python tools_webrtc/gn_check_autofix.py \ -m tryserver.webrtc -b linux_rel # Repead the gn_check_autofix step for other platforms git ls-files | grep BUILD.gn | \ xargs sed -i 's/absl\/memory:memory/absl\/memory/g' git cl format Bug: webrtc:10945 Change-Id: I3fe28ea80f4dd3ba3cf28effd151d5e1f19aff89 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/153221 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Alessio Bazzica <alessiob@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29209}
112 lines
3.9 KiB
C++
112 lines
3.9 KiB
C++
/*
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* Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#include "modules/congestion_controller/goog_cc/alr_detector.h"
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#include <cstdint>
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#include <cstdio>
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#include <memory>
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#include "api/rtc_event_log/rtc_event.h"
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#include "api/rtc_event_log/rtc_event_log.h"
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#include "logging/rtc_event_log/events/rtc_event_alr_state.h"
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#include "rtc_base/checks.h"
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#include "rtc_base/numerics/safe_conversions.h"
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#include "rtc_base/time_utils.h"
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namespace webrtc {
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namespace {
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AlrDetectorConfig GetConfigFromTrials(
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const WebRtcKeyValueConfig* key_value_config) {
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RTC_CHECK(AlrExperimentSettings::MaxOneFieldTrialEnabled(*key_value_config));
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absl::optional<AlrExperimentSettings> experiment_settings =
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AlrExperimentSettings::CreateFromFieldTrial(
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*key_value_config,
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AlrExperimentSettings::kScreenshareProbingBweExperimentName);
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if (!experiment_settings) {
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experiment_settings = AlrExperimentSettings::CreateFromFieldTrial(
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*key_value_config,
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AlrExperimentSettings::kStrictPacingAndProbingExperimentName);
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}
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AlrDetectorConfig conf;
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if (experiment_settings) {
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conf.bandwidth_usage_ratio =
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experiment_settings->alr_bandwidth_usage_percent / 100.0;
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conf.start_budget_level_ratio =
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experiment_settings->alr_start_budget_level_percent / 100.0;
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conf.stop_budget_level_ratio =
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experiment_settings->alr_stop_budget_level_percent / 100.0;
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}
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conf.Parser()->Parse(
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key_value_config->Lookup("WebRTC-AlrDetectorParameters"));
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return conf;
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}
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} // namespace
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std::unique_ptr<StructParametersParser> AlrDetectorConfig::Parser() {
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return StructParametersParser::Create( //
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"bw_usage", &bandwidth_usage_ratio, //
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"start", &start_budget_level_ratio, //
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"stop", &stop_budget_level_ratio);
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}
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AlrDetector::AlrDetector(AlrDetectorConfig config, RtcEventLog* event_log)
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: conf_(config), alr_budget_(0, true), event_log_(event_log) {}
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AlrDetector::AlrDetector(const WebRtcKeyValueConfig* key_value_config)
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: AlrDetector(GetConfigFromTrials(key_value_config), nullptr) {}
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AlrDetector::AlrDetector(const WebRtcKeyValueConfig* key_value_config,
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RtcEventLog* event_log)
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: AlrDetector(GetConfigFromTrials(key_value_config), event_log) {}
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AlrDetector::~AlrDetector() {}
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void AlrDetector::OnBytesSent(size_t bytes_sent, int64_t send_time_ms) {
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if (!last_send_time_ms_.has_value()) {
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last_send_time_ms_ = send_time_ms;
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// Since the duration for sending the bytes is unknwon, return without
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// updating alr state.
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return;
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}
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int64_t delta_time_ms = send_time_ms - *last_send_time_ms_;
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last_send_time_ms_ = send_time_ms;
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alr_budget_.UseBudget(bytes_sent);
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alr_budget_.IncreaseBudget(delta_time_ms);
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bool state_changed = false;
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if (alr_budget_.budget_ratio() > conf_.start_budget_level_ratio &&
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!alr_started_time_ms_) {
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alr_started_time_ms_.emplace(rtc::TimeMillis());
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state_changed = true;
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} else if (alr_budget_.budget_ratio() < conf_.stop_budget_level_ratio &&
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alr_started_time_ms_) {
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state_changed = true;
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alr_started_time_ms_.reset();
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}
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if (event_log_ && state_changed) {
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event_log_->Log(
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std::make_unique<RtcEventAlrState>(alr_started_time_ms_.has_value()));
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}
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}
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void AlrDetector::SetEstimatedBitrate(int bitrate_bps) {
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RTC_DCHECK(bitrate_bps);
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int target_rate_kbps =
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static_cast<double>(bitrate_bps) * conf_.bandwidth_usage_ratio / 1000;
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alr_budget_.set_target_rate_kbps(target_rate_kbps);
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}
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absl::optional<int64_t> AlrDetector::GetApplicationLimitedRegionStartTime()
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const {
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return alr_started_time_ms_;
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}
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} // namespace webrtc
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