blob: 14baae7e8164b8d7bd1726c2a7f152589635e2dd [file] [log] [blame]
/* Copyright (c) 2014 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 <stdint.h>
#include <memory>
#include <vector>
#include "absl/types/optional.h"
#include "api/array_view.h"
#include "modules/video_coding/jitter_estimator.h"
#include "rtc_base/experiments/jitter_upper_bound_experiment.h"
#include "rtc_base/numerics/histogram_percentile_counter.h"
#include "rtc_base/strings/string_builder.h"
#include "rtc_base/time_utils.h"
#include "system_wrappers/include/clock.h"
#include "test/field_trial.h"
#include "test/gtest.h"
namespace webrtc {
class TestVCMJitterEstimator : public ::testing::Test {
protected:
TestVCMJitterEstimator() : fake_clock_(0) {}
virtual void SetUp() {
estimator_ = std::make_unique<VCMJitterEstimator>(&fake_clock_);
}
void AdvanceClock(int64_t microseconds) {
fake_clock_.AdvanceTimeMicroseconds(microseconds);
}
SimulatedClock fake_clock_;
std::unique_ptr<VCMJitterEstimator> estimator_;
};
// Generates some simple test data in the form of a sawtooth wave.
class ValueGenerator {
public:
explicit ValueGenerator(int32_t amplitude)
: amplitude_(amplitude), counter_(0) {}
virtual ~ValueGenerator() {}
int64_t Delay() const { return ((counter_ % 11) - 5) * amplitude_; }
uint32_t FrameSize() const { return 1000 + Delay(); }
void Advance() { ++counter_; }
private:
const int32_t amplitude_;
int64_t counter_;
};
// 5 fps, disable jitter delay altogether.
TEST_F(TestVCMJitterEstimator, TestLowRate) {
ValueGenerator gen(10);
uint64_t time_delta_us = rtc::kNumMicrosecsPerSec / 5;
for (int i = 0; i < 60; ++i) {
estimator_->UpdateEstimate(gen.Delay(), gen.FrameSize());
AdvanceClock(time_delta_us);
if (i > 2)
EXPECT_EQ(estimator_->GetJitterEstimate(0, absl::nullopt), 0);
gen.Advance();
}
}
TEST_F(TestVCMJitterEstimator, TestLowRateDisabled) {
test::ScopedFieldTrials field_trials(
"WebRTC-ReducedJitterDelayKillSwitch/Enabled/");
SetUp();
ValueGenerator gen(10);
uint64_t time_delta_us = rtc::kNumMicrosecsPerSec / 5;
for (int i = 0; i < 60; ++i) {
estimator_->UpdateEstimate(gen.Delay(), gen.FrameSize());
AdvanceClock(time_delta_us);
if (i > 2)
EXPECT_GT(estimator_->GetJitterEstimate(0, absl::nullopt), 0);
gen.Advance();
}
}
TEST_F(TestVCMJitterEstimator, TestUpperBound) {
struct TestContext {
TestContext()
: upper_bound(0.0),
rtt_mult(0),
rtt_mult_add_cap_ms(absl::nullopt),
percentiles(1000) {}
double upper_bound;
double rtt_mult;
absl::optional<double> rtt_mult_add_cap_ms;
rtc::HistogramPercentileCounter percentiles;
};
std::vector<TestContext> test_cases(4);
// Large upper bound, rtt_mult = 0, and nullopt for rtt_mult addition cap.
test_cases[0].upper_bound = 100.0;
test_cases[0].rtt_mult = 0;
test_cases[0].rtt_mult_add_cap_ms = absl::nullopt;
// Small upper bound, rtt_mult = 0, and nullopt for rtt_mult addition cap.
test_cases[1].upper_bound = 3.5;
test_cases[1].rtt_mult = 0;
test_cases[1].rtt_mult_add_cap_ms = absl::nullopt;
// Large upper bound, rtt_mult = 1, and large rtt_mult addition cap value.
test_cases[2].upper_bound = 1000.0;
test_cases[2].rtt_mult = 1.0;
test_cases[2].rtt_mult_add_cap_ms = 200.0;
// Large upper bound, rtt_mult = 1, and small rtt_mult addition cap value.
test_cases[3].upper_bound = 1000.0;
test_cases[3].rtt_mult = 1.0;
test_cases[3].rtt_mult_add_cap_ms = 10.0;
// Test jitter buffer upper_bound and rtt_mult addition cap sizes.
for (TestContext& context : test_cases) {
// Set up field trial and reset jitter estimator.
char string_buf[64];
rtc::SimpleStringBuilder ssb(string_buf);
ssb << JitterUpperBoundExperiment::kJitterUpperBoundExperimentName
<< "/Enabled-" << context.upper_bound << "/";
test::ScopedFieldTrials field_trials(ssb.str());
SetUp();
ValueGenerator gen(50);
uint64_t time_delta_us = rtc::kNumMicrosecsPerSec / 30;
constexpr int64_t kRttMs = 250;
for (int i = 0; i < 100; ++i) {
estimator_->UpdateEstimate(gen.Delay(), gen.FrameSize());
AdvanceClock(time_delta_us);
estimator_->FrameNacked(); // To test rtt_mult.
estimator_->UpdateRtt(kRttMs); // To test rtt_mult.
context.percentiles.Add(
static_cast<uint32_t>(estimator_->GetJitterEstimate(
context.rtt_mult, context.rtt_mult_add_cap_ms)));
gen.Advance();
}
}
// Median should be similar after three seconds. Allow 5% error margin.
uint32_t median_unbound = *test_cases[0].percentiles.GetPercentile(0.5);
uint32_t median_bounded = *test_cases[1].percentiles.GetPercentile(0.5);
EXPECT_NEAR(median_unbound, median_bounded, (median_unbound * 5) / 100);
// Max should be lower for the bounded case.
uint32_t max_unbound = *test_cases[0].percentiles.GetPercentile(1.0);
uint32_t max_bounded = *test_cases[1].percentiles.GetPercentile(1.0);
EXPECT_GT(max_unbound, static_cast<uint32_t>(max_bounded * 1.25));
// With rtt_mult = 1, max should be lower with small rtt_mult add cap value.
max_unbound = *test_cases[2].percentiles.GetPercentile(1.0);
max_bounded = *test_cases[3].percentiles.GetPercentile(1.0);
EXPECT_GT(max_unbound, static_cast<uint32_t>(max_bounded * 1.25));
}
} // namespace webrtc