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a4d873786f
This CL was generated by running git ls-files | grep -P "(\.h|\.cc)$" | grep -v 'sdk/' | grep -v 'rtc_base/ssl_' | \ grep -v 'fake_rtc_certificate_generator.h' | grep -v 'modules/audio_device/win/' | \ grep -v 'system_wrappers/source/clock.cc' | grep -v 'rtc_base/trace_event.h' | \ grep -v 'modules/audio_coding/codecs/ilbc/' | grep -v 'screen_capturer_mac.h' | \ grep -v 'spl_inl_mips.h' | grep -v 'data_size_unittest.cc' | grep -v 'timestamp_unittest.cc' \ | xargs clang-format -i ; git cl format Most of these changes are clang-format grouping and reordering includes differently. Bug: webrtc:9340 Change-Id: Ic83ddbc169bfacd21883e381b5181c3dd4fe8a63 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/144051 Commit-Queue: Jonas Olsson <jonasolsson@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28505}
119 lines
4.2 KiB
C++
119 lines
4.2 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/include/receive_side_congestion_controller.h"
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#include "modules/pacing/packet_router.h"
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#include "system_wrappers/include/clock.h"
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#include "test/gmock.h"
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#include "test/gtest.h"
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#include "test/scenario/scenario.h"
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using ::testing::_;
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using ::testing::AtLeast;
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using ::testing::NiceMock;
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using ::testing::Return;
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using ::testing::SaveArg;
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using ::testing::StrictMock;
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namespace webrtc {
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namespace {
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// Helper to convert some time format to resolution used in absolute send time
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// header extension, rounded upwards. |t| is the time to convert, in some
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// resolution. |denom| is the value to divide |t| by to get whole seconds,
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// e.g. |denom| = 1000 if |t| is in milliseconds.
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uint32_t AbsSendTime(int64_t t, int64_t denom) {
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return (((t << 18) + (denom >> 1)) / denom) & 0x00fffffful;
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}
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class MockPacketRouter : public PacketRouter {
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public:
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MOCK_METHOD2(OnReceiveBitrateChanged,
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void(const std::vector<uint32_t>& ssrcs, uint32_t bitrate));
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};
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const uint32_t kInitialBitrateBps = 60000;
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} // namespace
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namespace test {
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TEST(ReceiveSideCongestionControllerTest, OnReceivedPacketWithAbsSendTime) {
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StrictMock<MockPacketRouter> packet_router;
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SimulatedClock clock_(123456);
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ReceiveSideCongestionController controller(&clock_, &packet_router);
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size_t payload_size = 1000;
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RTPHeader header;
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header.ssrc = 0x11eb21c;
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header.extension.hasAbsoluteSendTime = true;
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std::vector<unsigned int> ssrcs;
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EXPECT_CALL(packet_router, OnReceiveBitrateChanged(_, _))
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.WillRepeatedly(SaveArg<0>(&ssrcs));
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for (int i = 0; i < 10; ++i) {
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clock_.AdvanceTimeMilliseconds((1000 * payload_size) / kInitialBitrateBps);
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int64_t now_ms = clock_.TimeInMilliseconds();
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header.extension.absoluteSendTime = AbsSendTime(now_ms, 1000);
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controller.OnReceivedPacket(now_ms, payload_size, header);
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}
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ASSERT_EQ(1u, ssrcs.size());
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EXPECT_EQ(header.ssrc, ssrcs[0]);
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}
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TEST(ReceiveSideCongestionControllerTest, ConvergesToCapacity) {
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Scenario s("recieve_cc_unit/converge");
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NetworkSimulationConfig net_conf;
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net_conf.bandwidth = DataRate::kbps(1000);
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net_conf.delay = TimeDelta::ms(50);
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auto* client = s.CreateClient("send", [&](CallClientConfig* c) {
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c->transport.rates.start_rate = DataRate::kbps(300);
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});
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auto* route = s.CreateRoutes(client, {s.CreateSimulationNode(net_conf)},
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s.CreateClient("return", CallClientConfig()),
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{s.CreateSimulationNode(net_conf)});
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VideoStreamConfig video;
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video.stream.packet_feedback = false;
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s.CreateVideoStream(route->forward(), video);
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s.RunFor(TimeDelta::seconds(30));
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EXPECT_NEAR(client->send_bandwidth().kbps(), 900, 150);
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}
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TEST(ReceiveSideCongestionControllerTest, IsFairToTCP) {
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Scenario s("recieve_cc_unit/tcp_fairness");
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NetworkSimulationConfig net_conf;
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net_conf.bandwidth = DataRate::kbps(1000);
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net_conf.delay = TimeDelta::ms(50);
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auto* client = s.CreateClient("send", [&](CallClientConfig* c) {
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c->transport.rates.start_rate = DataRate::kbps(1000);
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});
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auto send_net = {s.CreateSimulationNode(net_conf)};
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auto ret_net = {s.CreateSimulationNode(net_conf)};
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auto* route = s.CreateRoutes(
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client, send_net, s.CreateClient("return", CallClientConfig()), ret_net);
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VideoStreamConfig video;
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video.stream.packet_feedback = false;
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s.CreateVideoStream(route->forward(), video);
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s.net()->StartFakeTcpCrossTraffic(s.net()->CreateRoute(send_net),
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s.net()->CreateRoute(ret_net),
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FakeTcpConfig());
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s.RunFor(TimeDelta::seconds(30));
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// For some reason we get outcompeted by TCP here, this should probably be
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// fixed and a lower bound should be added to the test.
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EXPECT_LT(client->send_bandwidth().kbps(), 750);
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}
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} // namespace test
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} // namespace webrtc
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