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/*
* 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/include/receive_side_congestion_controller.h"
#include "api/test/network_emulation/create_cross_traffic.h"
#include "api/test/network_emulation/cross_traffic.h"
#include "api/units/data_rate.h"
#include "api/units/data_size.h"
#include "api/units/time_delta.h"
#include "api/units/timestamp.h"
#include "modules/pacing/packet_router.h"
#include "modules/rtp_rtcp/include/rtp_header_extension_map.h"
#include "modules/rtp_rtcp/source/rtp_header_extensions.h"
#include "modules/rtp_rtcp/source/rtp_packet_received.h"
#include "system_wrappers/include/clock.h"
#include "test/gmock.h"
#include "test/gtest.h"
#include "test/scenario/scenario.h"
namespace webrtc {
namespace test {
namespace {
using ::testing::_;
using ::testing::AtLeast;
using ::testing::ElementsAre;
using ::testing::MockFunction;
constexpr DataRate kInitialBitrate = DataRate::BitsPerSec(60'000);
TEST(ReceiveSideCongestionControllerTest, SendsRembWithAbsSendTime) {
static constexpr DataSize kPayloadSize = DataSize::Bytes(1000);
MockFunction<void(std::vector<std::unique_ptr<rtcp::RtcpPacket>>)>
feedback_sender;
MockFunction<void(uint64_t, std::vector<uint32_t>)> remb_sender;
SimulatedClock clock_(123456);
ReceiveSideCongestionController controller(
&clock_, feedback_sender.AsStdFunction(), remb_sender.AsStdFunction(),
nullptr);
RtpHeaderExtensionMap extensions;
extensions.Register<AbsoluteSendTime>(1);
RtpPacketReceived packet(&extensions);
packet.SetSsrc(0x11eb21c);
packet.ReserveExtension<AbsoluteSendTime>();
packet.SetPayloadSize(kPayloadSize.bytes());
EXPECT_CALL(remb_sender, Call(_, ElementsAre(packet.Ssrc())))
.Times(AtLeast(1));
for (int i = 0; i < 10; ++i) {
clock_.AdvanceTime(kPayloadSize / kInitialBitrate);
Timestamp now = clock_.CurrentTime();
packet.SetExtension<AbsoluteSendTime>(AbsoluteSendTime::To24Bits(now));
packet.set_arrival_time(now);
controller.OnReceivedPacket(packet, MediaType::VIDEO);
}
}
TEST(ReceiveSideCongestionControllerTest,
SendsRembAfterSetMaxDesiredReceiveBitrate) {
MockFunction<void(std::vector<std::unique_ptr<rtcp::RtcpPacket>>)>
feedback_sender;
MockFunction<void(uint64_t, std::vector<uint32_t>)> remb_sender;
SimulatedClock clock_(123456);
ReceiveSideCongestionController controller(
&clock_, feedback_sender.AsStdFunction(), remb_sender.AsStdFunction(),
nullptr);
EXPECT_CALL(remb_sender, Call(123, _));
controller.SetMaxDesiredReceiveBitrate(DataRate::BitsPerSec(123));
}
TEST(ReceiveSideCongestionControllerTest, ConvergesToCapacity) {
Scenario s("receive_cc_unit/converge");
NetworkSimulationConfig net_conf;
net_conf.bandwidth = DataRate::KilobitsPerSec(1000);
net_conf.delay = TimeDelta::Millis(50);
auto* client = s.CreateClient("send", [&](CallClientConfig* c) {
c->transport.rates.start_rate = DataRate::KilobitsPerSec(300);
});
auto* route = s.CreateRoutes(client, {s.CreateSimulationNode(net_conf)},
s.CreateClient("return", CallClientConfig()),
{s.CreateSimulationNode(net_conf)});
VideoStreamConfig video;
video.stream.packet_feedback = false;
s.CreateVideoStream(route->forward(), video);
s.RunFor(TimeDelta::Seconds(30));
EXPECT_NEAR(client->send_bandwidth().kbps(), 900, 150);
}
TEST(ReceiveSideCongestionControllerTest, IsFairToTCP) {
Scenario s("receive_cc_unit/tcp_fairness");
NetworkSimulationConfig net_conf;
net_conf.bandwidth = DataRate::KilobitsPerSec(1000);
net_conf.delay = TimeDelta::Millis(50);
auto* client = s.CreateClient("send", [&](CallClientConfig* c) {
c->transport.rates.start_rate = DataRate::KilobitsPerSec(1000);
});
auto send_net = {s.CreateSimulationNode(net_conf)};
auto ret_net = {s.CreateSimulationNode(net_conf)};
auto* route = s.CreateRoutes(
client, send_net, s.CreateClient("return", CallClientConfig()), ret_net);
VideoStreamConfig video;
video.stream.packet_feedback = false;
s.CreateVideoStream(route->forward(), video);
s.net()->StartCrossTraffic(CreateFakeTcpCrossTraffic(
s.net()->CreateRoute(send_net), s.net()->CreateRoute(ret_net),
FakeTcpConfig()));
s.RunFor(TimeDelta::Seconds(30));
// For some reason we get outcompeted by TCP here, this should probably be
// fixed and a lower bound should be added to the test.
EXPECT_LT(client->send_bandwidth().kbps(), 750);
}
} // namespace
} // namespace test
} // namespace webrtc
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