webrtc/modules/congestion_controller/median_slope_estimator_unittest.cc
Mirko Bonadei 92ea95e34a Fixing WebRTC after moving from src/webrtc to src/
In https://webrtc-review.googlesource.com/c/src/+/1560 we moved WebRTC
from src/webrtc to src/ (in order to preserve an healthy git history).
This CL takes care of fixing header guards, #include paths, etc...

NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
TBR=tommi@webrtc.org


Bug: chromium:611808
Change-Id: Iea91618212bee0af16aa3f05071eab8f93706578
Reviewed-on: https://webrtc-review.googlesource.com/1561
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Henrik Kjellander <kjellander@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19846}
2017-09-15 05:02:56 +00:00

72 lines
2.3 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/congestion_controller/median_slope_estimator.h"
#include "rtc_base/random.h"
#include "test/gtest.h"
namespace webrtc {
namespace {
constexpr size_t kWindowSize = 20;
constexpr double kGain = 1;
constexpr int64_t kAvgTimeBetweenPackets = 10;
constexpr size_t kPacketCount = 2 * kWindowSize + 1;
void TestEstimator(double slope, double jitter_stddev, double tolerance) {
MedianSlopeEstimator estimator(kWindowSize, kGain);
Random random(0x1234567);
int64_t send_times[kPacketCount];
int64_t recv_times[kPacketCount];
int64_t send_start_time = random.Rand(1000000);
int64_t recv_start_time = random.Rand(1000000);
for (size_t i = 0; i < kPacketCount; ++i) {
send_times[i] = send_start_time + i * kAvgTimeBetweenPackets;
double latency = i * kAvgTimeBetweenPackets / (1 - slope);
double jitter = random.Gaussian(0, jitter_stddev);
recv_times[i] = recv_start_time + latency + jitter;
}
for (size_t i = 1; i < kPacketCount; ++i) {
double recv_delta = recv_times[i] - recv_times[i - 1];
double send_delta = send_times[i] - send_times[i - 1];
estimator.Update(recv_delta, send_delta, recv_times[i]);
if (i < kWindowSize)
EXPECT_NEAR(estimator.trendline_slope(), 0, 0.001);
else
EXPECT_NEAR(estimator.trendline_slope(), slope, tolerance);
}
}
} // namespace
TEST(MedianSlopeEstimator, PerfectLineSlopeOneHalf) {
TestEstimator(0.5, 0, 0.001);
}
TEST(MedianSlopeEstimator, PerfectLineSlopeMinusOne) {
TestEstimator(-1, 0, 0.001);
}
TEST(MedianSlopeEstimator, PerfectLineSlopeZero) {
TestEstimator(0, 0, 0.001);
}
TEST(MedianSlopeEstimator, JitteryLineSlopeOneHalf) {
TestEstimator(0.5, kAvgTimeBetweenPackets / 3.0, 0.01);
}
TEST(MedianSlopeEstimator, JitteryLineSlopeMinusOne) {
TestEstimator(-1, kAvgTimeBetweenPackets / 3.0, 0.05);
}
TEST(MedianSlopeEstimator, JitteryLineSlopeZero) {
TestEstimator(0, kAvgTimeBetweenPackets / 3.0, 0.02);
}
} // namespace webrtc