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The goal of this work is to make it easier to experiment with the bandwidth estimation implementation. For this reason network control functionality is moved from SendSideCongestionController(SSCC), PacedSender and BitrateController to the newly created GoogCcNetworkController which implements the newly created NetworkControllerInterface. This allows the implementation to be replaced at runtime in the future. This is the first part of a split of a larger CL, see: https://webrtc-review.googlesource.com/c/src/+/39788/8 For further explanations. Bug: webrtc:8415 Change-Id: I770189c04cc31b313bd4e57821acff55fbcb1ad3 Reviewed-on: https://webrtc-review.googlesource.com/43840 Commit-Queue: Sebastian Jansson <srte@webrtc.org> Reviewed-by: Björn Terelius <terelius@webrtc.org> Reviewed-by: Stefan Holmer <stefan@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21868}
175 lines
6.1 KiB
C++
175 lines
6.1 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/alr_detector.h"
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#include "rtc_base/experiments/alr_experiment.h"
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#include "test/field_trial.h"
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#include "test/gtest.h"
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namespace {
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constexpr int kEstimatedBitrateBps = 300000;
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} // namespace
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namespace webrtc {
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namespace {
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class SimulateOutgoingTrafficIn {
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public:
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explicit SimulateOutgoingTrafficIn(AlrDetector* alr_detector,
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int64_t* timestamp_ms)
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: alr_detector_(alr_detector), timestamp_ms_(timestamp_ms) {
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RTC_CHECK(alr_detector_);
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}
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SimulateOutgoingTrafficIn& ForTimeMs(int time_ms) {
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interval_ms_ = time_ms;
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ProduceTraffic();
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return *this;
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}
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SimulateOutgoingTrafficIn& AtPercentOfEstimatedBitrate(int usage_percentage) {
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usage_percentage_.emplace(usage_percentage);
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ProduceTraffic();
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return *this;
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}
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private:
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void ProduceTraffic() {
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if (!interval_ms_ || !usage_percentage_)
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return;
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const int kTimeStepMs = 10;
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for (int t = 0; t < *interval_ms_; t += kTimeStepMs) {
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*timestamp_ms_ += kTimeStepMs;
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alr_detector_->OnBytesSent(kEstimatedBitrateBps * *usage_percentage_ *
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kTimeStepMs / (8 * 100 * 1000),
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*timestamp_ms_);
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}
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int remainder_ms = *interval_ms_ % kTimeStepMs;
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if (remainder_ms > 0) {
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*timestamp_ms_ += kTimeStepMs;
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alr_detector_->OnBytesSent(kEstimatedBitrateBps * *usage_percentage_ *
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remainder_ms / (8 * 100 * 1000),
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*timestamp_ms_);
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}
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}
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AlrDetector* const alr_detector_;
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int64_t* timestamp_ms_;
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rtc::Optional<int> interval_ms_;
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rtc::Optional<int> usage_percentage_;
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};
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} // namespace
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class AlrDetectorTest : public testing::Test {
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public:
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void SetUp() override {
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alr_detector_.SetEstimatedBitrate(kEstimatedBitrateBps);
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}
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protected:
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AlrDetector alr_detector_;
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int64_t timestamp_ms_ = 1000;
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};
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TEST_F(AlrDetectorTest, AlrDetection) {
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// Start in non-ALR state.
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EXPECT_FALSE(alr_detector_.GetApplicationLimitedRegionStartTime());
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// Stay in non-ALR state when usage is close to 100%.
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SimulateOutgoingTrafficIn(&alr_detector_, ×tamp_ms_)
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.ForTimeMs(1000)
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.AtPercentOfEstimatedBitrate(90);
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EXPECT_FALSE(alr_detector_.GetApplicationLimitedRegionStartTime());
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// Verify that we ALR starts when bitrate drops below 20%.
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SimulateOutgoingTrafficIn(&alr_detector_, ×tamp_ms_)
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.ForTimeMs(1500)
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.AtPercentOfEstimatedBitrate(20);
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EXPECT_TRUE(alr_detector_.GetApplicationLimitedRegionStartTime());
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// Verify that ALR ends when usage is above 65%.
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SimulateOutgoingTrafficIn(&alr_detector_, ×tamp_ms_)
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.ForTimeMs(4000)
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.AtPercentOfEstimatedBitrate(100);
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EXPECT_FALSE(alr_detector_.GetApplicationLimitedRegionStartTime());
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}
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TEST_F(AlrDetectorTest, ShortSpike) {
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// Start in non-ALR state.
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EXPECT_FALSE(alr_detector_.GetApplicationLimitedRegionStartTime());
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// Verify that we ALR starts when bitrate drops below 20%.
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SimulateOutgoingTrafficIn(&alr_detector_, ×tamp_ms_)
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.ForTimeMs(1000)
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.AtPercentOfEstimatedBitrate(20);
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EXPECT_TRUE(alr_detector_.GetApplicationLimitedRegionStartTime());
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// Verify that we stay in ALR region even after a short bitrate spike.
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SimulateOutgoingTrafficIn(&alr_detector_, ×tamp_ms_)
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.ForTimeMs(100)
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.AtPercentOfEstimatedBitrate(150);
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EXPECT_TRUE(alr_detector_.GetApplicationLimitedRegionStartTime());
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// ALR ends when usage is above 65%.
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SimulateOutgoingTrafficIn(&alr_detector_, ×tamp_ms_)
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.ForTimeMs(3000)
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.AtPercentOfEstimatedBitrate(100);
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EXPECT_FALSE(alr_detector_.GetApplicationLimitedRegionStartTime());
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}
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TEST_F(AlrDetectorTest, BandwidthEstimateChanges) {
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// Start in non-ALR state.
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EXPECT_FALSE(alr_detector_.GetApplicationLimitedRegionStartTime());
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// ALR starts when bitrate drops below 20%.
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SimulateOutgoingTrafficIn(&alr_detector_, ×tamp_ms_)
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.ForTimeMs(1000)
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.AtPercentOfEstimatedBitrate(20);
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EXPECT_TRUE(alr_detector_.GetApplicationLimitedRegionStartTime());
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// When bandwidth estimate drops the detector should stay in ALR mode and quit
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// it shortly afterwards as the sender continues sending the same amount of
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// traffic. This is necessary to ensure that ProbeController can still react
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// to the BWE drop by initiating a new probe.
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alr_detector_.SetEstimatedBitrate(kEstimatedBitrateBps / 5);
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EXPECT_TRUE(alr_detector_.GetApplicationLimitedRegionStartTime());
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SimulateOutgoingTrafficIn(&alr_detector_, ×tamp_ms_)
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.ForTimeMs(1000)
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.AtPercentOfEstimatedBitrate(50);
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EXPECT_FALSE(alr_detector_.GetApplicationLimitedRegionStartTime());
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}
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TEST_F(AlrDetectorTest, ParseControlFieldTrial) {
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webrtc::test::ScopedFieldTrials field_trial(
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"WebRTC-ProbingScreenshareBwe/Control/");
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rtc::Optional<AlrExperimentSettings> parsed_params =
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AlrExperimentSettings::CreateFromFieldTrial(
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"WebRTC-ProbingScreenshareBwe");
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EXPECT_FALSE(static_cast<bool>(parsed_params));
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}
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TEST_F(AlrDetectorTest, ParseActiveFieldTrial) {
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webrtc::test::ScopedFieldTrials field_trial(
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"WebRTC-ProbingScreenshareBwe/1.1,2875,85,20,-20,1/");
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rtc::Optional<AlrExperimentSettings> parsed_params =
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AlrExperimentSettings::CreateFromFieldTrial(
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"WebRTC-ProbingScreenshareBwe");
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ASSERT_TRUE(static_cast<bool>(parsed_params));
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EXPECT_EQ(1.1f, parsed_params->pacing_factor);
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EXPECT_EQ(2875, parsed_params->max_paced_queue_time);
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EXPECT_EQ(85, parsed_params->alr_bandwidth_usage_percent);
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EXPECT_EQ(20, parsed_params->alr_start_budget_level_percent);
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EXPECT_EQ(-20, parsed_params->alr_stop_budget_level_percent);
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EXPECT_EQ(1, parsed_params->group_id);
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}
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} // namespace webrtc
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