webrtc_m130/modules/pacing/alr_detector_unittest.cc
Sebastian Jansson ea86bb74fc Revert "Revert "Revert "Reland "Moved congestion controller to task queue.""""
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}
2018-02-14 16:53:49 +00:00

171 lines
5.7 KiB
C++

/*
* 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 "modules/pacing/alr_detector.h"
#include "rtc_base/experiments/alr_experiment.h"
#include "test/field_trial.h"
#include "test/gtest.h"
namespace {
constexpr int kEstimatedBitrateBps = 300000;
} // namespace
namespace webrtc {
namespace {
class SimulateOutgoingTrafficIn {
public:
explicit SimulateOutgoingTrafficIn(AlrDetector* alr_detector)
: alr_detector_(alr_detector) {
RTC_CHECK(alr_detector_);
}
SimulateOutgoingTrafficIn& ForTimeMs(int time_ms) {
interval_ms_ = time_ms;
ProduceTraffic();
return *this;
}
SimulateOutgoingTrafficIn& AtPercentOfEstimatedBitrate(int usage_percentage) {
usage_percentage_.emplace(usage_percentage);
ProduceTraffic();
return *this;
}
private:
void ProduceTraffic() {
if (!interval_ms_ || !usage_percentage_)
return;
const int kTimeStepMs = 10;
for (int t = 0; t < *interval_ms_; t += kTimeStepMs) {
alr_detector_->OnBytesSent(kEstimatedBitrateBps * *usage_percentage_ *
kTimeStepMs / (8 * 100 * 1000),
kTimeStepMs);
}
int remainder_ms = *interval_ms_ % kTimeStepMs;
if (remainder_ms > 0) {
alr_detector_->OnBytesSent(kEstimatedBitrateBps * *usage_percentage_ *
remainder_ms / (8 * 100 * 1000),
kTimeStepMs);
}
}
AlrDetector* const alr_detector_;
rtc::Optional<int> interval_ms_;
rtc::Optional<int> usage_percentage_;
};
} // namespace
class AlrDetectorTest : public testing::Test {
public:
void SetUp() override {
alr_detector_.SetEstimatedBitrate(kEstimatedBitrateBps);
}
protected:
AlrDetector alr_detector_;
};
TEST_F(AlrDetectorTest, AlrDetection) {
// Start in non-ALR state.
EXPECT_FALSE(alr_detector_.GetApplicationLimitedRegionStartTime());
// Stay in non-ALR state when usage is close to 100%.
SimulateOutgoingTrafficIn(&alr_detector_)
.ForTimeMs(1000)
.AtPercentOfEstimatedBitrate(90);
EXPECT_FALSE(alr_detector_.GetApplicationLimitedRegionStartTime());
// Verify that we ALR starts when bitrate drops below 20%.
SimulateOutgoingTrafficIn(&alr_detector_)
.ForTimeMs(1500)
.AtPercentOfEstimatedBitrate(20);
EXPECT_TRUE(alr_detector_.GetApplicationLimitedRegionStartTime());
// Verify that ALR ends when usage is above 65%.
SimulateOutgoingTrafficIn(&alr_detector_)
.ForTimeMs(4000)
.AtPercentOfEstimatedBitrate(100);
EXPECT_FALSE(alr_detector_.GetApplicationLimitedRegionStartTime());
}
TEST_F(AlrDetectorTest, ShortSpike) {
// Start in non-ALR state.
EXPECT_FALSE(alr_detector_.GetApplicationLimitedRegionStartTime());
// Verify that we ALR starts when bitrate drops below 20%.
SimulateOutgoingTrafficIn(&alr_detector_)
.ForTimeMs(1000)
.AtPercentOfEstimatedBitrate(20);
EXPECT_TRUE(alr_detector_.GetApplicationLimitedRegionStartTime());
// Verify that we stay in ALR region even after a short bitrate spike.
SimulateOutgoingTrafficIn(&alr_detector_)
.ForTimeMs(100)
.AtPercentOfEstimatedBitrate(150);
EXPECT_TRUE(alr_detector_.GetApplicationLimitedRegionStartTime());
// ALR ends when usage is above 65%.
SimulateOutgoingTrafficIn(&alr_detector_)
.ForTimeMs(3000)
.AtPercentOfEstimatedBitrate(100);
EXPECT_FALSE(alr_detector_.GetApplicationLimitedRegionStartTime());
}
TEST_F(AlrDetectorTest, BandwidthEstimateChanges) {
// Start in non-ALR state.
EXPECT_FALSE(alr_detector_.GetApplicationLimitedRegionStartTime());
// ALR starts when bitrate drops below 20%.
SimulateOutgoingTrafficIn(&alr_detector_)
.ForTimeMs(1000)
.AtPercentOfEstimatedBitrate(20);
EXPECT_TRUE(alr_detector_.GetApplicationLimitedRegionStartTime());
// When bandwidth estimate drops the detector should stay in ALR mode and quit
// it shortly afterwards as the sender continues sending the same amount of
// traffic. This is necessary to ensure that ProbeController can still react
// to the BWE drop by initiating a new probe.
alr_detector_.SetEstimatedBitrate(kEstimatedBitrateBps / 5);
EXPECT_TRUE(alr_detector_.GetApplicationLimitedRegionStartTime());
SimulateOutgoingTrafficIn(&alr_detector_)
.ForTimeMs(1000)
.AtPercentOfEstimatedBitrate(50);
EXPECT_FALSE(alr_detector_.GetApplicationLimitedRegionStartTime());
}
TEST_F(AlrDetectorTest, ParseControlFieldTrial) {
webrtc::test::ScopedFieldTrials field_trial(
"WebRTC-ProbingScreenshareBwe/Control/");
rtc::Optional<AlrExperimentSettings> parsed_params =
AlrExperimentSettings::CreateFromFieldTrial(
"WebRTC-ProbingScreenshareBwe");
EXPECT_FALSE(static_cast<bool>(parsed_params));
}
TEST_F(AlrDetectorTest, ParseActiveFieldTrial) {
webrtc::test::ScopedFieldTrials field_trial(
"WebRTC-ProbingScreenshareBwe/1.1,2875,85,20,-20,1/");
rtc::Optional<AlrExperimentSettings> parsed_params =
AlrExperimentSettings::CreateFromFieldTrial(
"WebRTC-ProbingScreenshareBwe");
ASSERT_TRUE(static_cast<bool>(parsed_params));
EXPECT_EQ(1.1f, parsed_params->pacing_factor);
EXPECT_EQ(2875, parsed_params->max_paced_queue_time);
EXPECT_EQ(85, parsed_params->alr_bandwidth_usage_percent);
EXPECT_EQ(20, parsed_params->alr_start_budget_level_percent);
EXPECT_EQ(-20, parsed_params->alr_stop_budget_level_percent);
EXPECT_EQ(1, parsed_params->group_id);
}
} // namespace webrtc