This reverts commit 65792c5a5c542201f7b9feefded505842692e6ed. Reason for revert: <INSERT REASONING HERE> Original change's description: > Revert "Revert "Reland "Moved congestion controller to task queue.""" > > This reverts commit 4e849f6925b2ac44b0957a228d7131fc391fca54. > > Reason for revert: <INSERT REASONING HERE> > > Original change's description: > > Revert "Reland "Moved congestion controller to task queue."" > > > > This reverts commit 57daeb7ac7f3d80992905b53fea500953fcfd793. > > > > Reason for revert: Cause increased congestion and deadlocks in downstream project > > > > Original change's description: > > > Reland "Moved congestion controller to task queue." > > > > > > This is a reland of 0cbcba7ea0dced1a7f353c64d6cf91d46ccb29f9. > > > > > > Original change's description: > > > > Moved congestion controller to task queue. > > > > > > > > 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} > > > > > > Bug: webrtc:8415 > > > Change-Id: I1d1756a30deed5b421b1c91c1918a13b6bb455da > > > Reviewed-on: https://webrtc-review.googlesource.com/48000 > > > Reviewed-by: Stefan Holmer <stefan@webrtc.org> > > > Commit-Queue: Sebastian Jansson <srte@webrtc.org> > > > Cr-Commit-Position: refs/heads/master@{#21899} > > > > TBR=terelius@webrtc.org,stefan@webrtc.org,srte@webrtc.org > > > > # Not skipping CQ checks because original CL landed > 1 day ago. > > > > Bug: webrtc:8415 > > Change-Id: Ida8074dcac2cc28b3629228eb22846d8a8e81b83 > > Reviewed-on: https://webrtc-review.googlesource.com/52980 > > Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> > > Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#22017} > > TBR=danilchap@webrtc.org,terelius@webrtc.org,stefan@webrtc.org,srte@webrtc.org > > Change-Id: I3393b74370c4f4d0955f50728005b2b925be169b > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Bug: webrtc:8415 > Reviewed-on: https://webrtc-review.googlesource.com/53262 > Reviewed-by: Sebastian Jansson <srte@webrtc.org> > Commit-Queue: Sebastian Jansson <srte@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#22023} TBR=danilchap@webrtc.org,terelius@webrtc.org,stefan@webrtc.org,srte@webrtc.org Change-Id: Id68ad986ee51142b7be3381d0793709b4392fe2c No-Presubmit: true No-Tree-Checks: true No-Try: true Bug: webrtc:8415 Reviewed-on: https://webrtc-review.googlesource.com/53360 Reviewed-by: Sebastian Jansson <srte@webrtc.org> Commit-Queue: Sebastian Jansson <srte@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22024}
171 lines
5.7 KiB
C++
171 lines
5.7 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/pacing/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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: alr_detector_(alr_detector) {
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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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alr_detector_->OnBytesSent(kEstimatedBitrateBps * *usage_percentage_ *
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kTimeStepMs / (8 * 100 * 1000),
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kTimeStepMs);
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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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alr_detector_->OnBytesSent(kEstimatedBitrateBps * *usage_percentage_ *
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remainder_ms / (8 * 100 * 1000),
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kTimeStepMs);
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}
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}
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AlrDetector* const alr_detector_;
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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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};
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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_)
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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_)
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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_)
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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_)
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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_)
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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_)
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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_)
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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_)
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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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