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2ee83c1784
Environment includes propagated field trials that can be later passed to RemoteBitrateEstimators member, and would allow not to rely on the global field trial string Bug: webrtc:42220378 Change-Id: Icf75a433c20352b2c22829c2148c92f69a2517aa Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/349645 Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/main@{#42242}
128 lines
4.9 KiB
C++
128 lines
4.9 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 "api/environment/environment_factory.h"
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#include "api/test/network_emulation/create_cross_traffic.h"
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#include "api/test/network_emulation/cross_traffic.h"
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#include "api/units/data_rate.h"
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#include "api/units/data_size.h"
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#include "api/units/time_delta.h"
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#include "api/units/timestamp.h"
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#include "modules/pacing/packet_router.h"
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#include "modules/rtp_rtcp/include/rtp_header_extension_map.h"
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#include "modules/rtp_rtcp/source/rtp_header_extensions.h"
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#include "modules/rtp_rtcp/source/rtp_packet_received.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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namespace webrtc {
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namespace test {
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namespace {
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using ::testing::_;
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using ::testing::AtLeast;
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using ::testing::ElementsAre;
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using ::testing::MockFunction;
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constexpr DataRate kInitialBitrate = DataRate::BitsPerSec(60'000);
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TEST(ReceiveSideCongestionControllerTest, SendsRembWithAbsSendTime) {
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static constexpr DataSize kPayloadSize = DataSize::Bytes(1000);
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MockFunction<void(std::vector<std::unique_ptr<rtcp::RtcpPacket>>)>
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feedback_sender;
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MockFunction<void(uint64_t, std::vector<uint32_t>)> remb_sender;
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SimulatedClock clock(123456);
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ReceiveSideCongestionController controller(
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CreateEnvironment(&clock), feedback_sender.AsStdFunction(),
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remb_sender.AsStdFunction(), nullptr);
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RtpHeaderExtensionMap extensions;
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extensions.Register<AbsoluteSendTime>(1);
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RtpPacketReceived packet(&extensions);
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packet.SetSsrc(0x11eb21c);
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packet.ReserveExtension<AbsoluteSendTime>();
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packet.SetPayloadSize(kPayloadSize.bytes());
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EXPECT_CALL(remb_sender, Call(_, ElementsAre(packet.Ssrc())))
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.Times(AtLeast(1));
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for (int i = 0; i < 10; ++i) {
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clock.AdvanceTime(kPayloadSize / kInitialBitrate);
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Timestamp now = clock.CurrentTime();
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packet.SetExtension<AbsoluteSendTime>(AbsoluteSendTime::To24Bits(now));
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packet.set_arrival_time(now);
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controller.OnReceivedPacket(packet, MediaType::VIDEO);
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}
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}
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TEST(ReceiveSideCongestionControllerTest,
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SendsRembAfterSetMaxDesiredReceiveBitrate) {
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MockFunction<void(std::vector<std::unique_ptr<rtcp::RtcpPacket>>)>
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feedback_sender;
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MockFunction<void(uint64_t, std::vector<uint32_t>)> remb_sender;
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SimulatedClock clock(123456);
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ReceiveSideCongestionController controller(
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CreateEnvironment(&clock), feedback_sender.AsStdFunction(),
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remb_sender.AsStdFunction(), nullptr);
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EXPECT_CALL(remb_sender, Call(123, _));
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controller.SetMaxDesiredReceiveBitrate(DataRate::BitsPerSec(123));
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}
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TEST(ReceiveSideCongestionControllerTest, ConvergesToCapacity) {
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Scenario s("receive_cc_unit/converge");
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NetworkSimulationConfig net_conf;
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net_conf.bandwidth = DataRate::KilobitsPerSec(1000);
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net_conf.delay = TimeDelta::Millis(50);
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auto* client = s.CreateClient("send", [&](CallClientConfig* c) {
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c->transport.rates.start_rate = DataRate::KilobitsPerSec(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("receive_cc_unit/tcp_fairness");
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NetworkSimulationConfig net_conf;
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net_conf.bandwidth = DataRate::KilobitsPerSec(1000);
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net_conf.delay = TimeDelta::Millis(50);
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auto* client = s.CreateClient("send", [&](CallClientConfig* c) {
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c->transport.rates.start_rate = DataRate::KilobitsPerSec(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()->StartCrossTraffic(CreateFakeTcpCrossTraffic(
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s.net()->CreateRoute(send_net), 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
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} // namespace test
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} // namespace webrtc
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