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56da2f7868
This CL activates unit tests for the task queue based send side congestion controller that will replace the current one in the future. To be able to have the tests run side by side with the tests of the old congestion controller, the old tests have been prefixed with "Legacy". This CL also contains some minor fixes to the new congestion controller code. Bug: webrtc:8415 Change-Id: I5e7474d42f17fcbfef402e26f638846fa3424695 Reviewed-on: https://webrtc-review.googlesource.com/55381 Reviewed-by: Philip Eliasson <philipel@webrtc.org> Reviewed-by: Björn Terelius <terelius@webrtc.org> Commit-Queue: Sebastian Jansson <srte@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22229}
72 lines
2.4 KiB
C++
72 lines
2.4 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/median_slope_estimator.h"
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#include "rtc_base/random.h"
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#include "test/gtest.h"
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namespace webrtc {
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namespace {
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constexpr size_t kWindowSize = 20;
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constexpr double kGain = 1;
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constexpr int64_t kAvgTimeBetweenPackets = 10;
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constexpr size_t kPacketCount = 2 * kWindowSize + 1;
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void TestEstimator(double slope, double jitter_stddev, double tolerance) {
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MedianSlopeEstimator estimator(kWindowSize, kGain);
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Random random(0x1234567);
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int64_t send_times[kPacketCount];
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int64_t recv_times[kPacketCount];
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int64_t send_start_time = random.Rand(1000000);
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int64_t recv_start_time = random.Rand(1000000);
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for (size_t i = 0; i < kPacketCount; ++i) {
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send_times[i] = send_start_time + i * kAvgTimeBetweenPackets;
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double latency = i * kAvgTimeBetweenPackets / (1 - slope);
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double jitter = random.Gaussian(0, jitter_stddev);
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recv_times[i] = recv_start_time + latency + jitter;
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}
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for (size_t i = 1; i < kPacketCount; ++i) {
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double recv_delta = recv_times[i] - recv_times[i - 1];
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double send_delta = send_times[i] - send_times[i - 1];
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estimator.Update(recv_delta, send_delta, recv_times[i]);
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if (i < kWindowSize)
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EXPECT_NEAR(estimator.trendline_slope(), 0, 0.001);
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else
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EXPECT_NEAR(estimator.trendline_slope(), slope, tolerance);
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}
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}
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} // namespace
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TEST(DeprecatedMedianSlopeEstimator, PerfectLineSlopeOneHalf) {
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TestEstimator(0.5, 0, 0.001);
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}
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TEST(DeprecatedMedianSlopeEstimator, PerfectLineSlopeMinusOne) {
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TestEstimator(-1, 0, 0.001);
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}
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TEST(DeprecatedMedianSlopeEstimator, PerfectLineSlopeZero) {
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TestEstimator(0, 0, 0.001);
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}
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TEST(DeprecatedMedianSlopeEstimator, JitteryLineSlopeOneHalf) {
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TestEstimator(0.5, kAvgTimeBetweenPackets / 3.0, 0.01);
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}
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TEST(DeprecatedMedianSlopeEstimator, JitteryLineSlopeMinusOne) {
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TestEstimator(-1, kAvgTimeBetweenPackets / 3.0, 0.05);
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}
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TEST(DeprecatedMedianSlopeEstimator, JitteryLineSlopeZero) {
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TestEstimator(0, kAvgTimeBetweenPackets / 3.0, 0.02);
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}
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} // namespace webrtc
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