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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 <cstdint>
#include <memory>
#include <vector>
#include "api/environment/environment_factory.h"
#include "api/media_types.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/rtp_rtcp/include/rtp_header_extension_map.h"
#include "modules/rtp_rtcp/source/rtcp_packet.h"
#include "modules/rtp_rtcp/source/rtcp_packet/common_header.h"
#include "modules/rtp_rtcp/source/rtcp_packet/congestion_control_feedback.h"
#include "modules/rtp_rtcp/source/rtp_header_extensions.h"
#include "modules/rtp_rtcp/source/rtp_packet_received.h"
#include "rtc_base/buffer.h"
#include "system_wrappers/include/clock.h"
#include "test/explicit_key_value_config.h"
#include "test/gmock.h"
#include "test/gtest.h"
#include "test/scenario/scenario.h"
#include "test/scenario/scenario_config.h"
namespace webrtc {
namespace test {
namespace {
using ::testing::_;
using ::testing::AtLeast;
using ::testing::ElementsAre;
using ::testing::MockFunction;
using ::testing::SizeIs;
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(CreateEnvironment(&clock),
feedback_sender.AsStdFunction(),
remb_sender.AsStdFunction());
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(CreateEnvironment(&clock),
feedback_sender.AsStdFunction(),
remb_sender.AsStdFunction());
EXPECT_CALL(remb_sender, Call(123, _));
controller.SetMaxDesiredReceiveBitrate(DataRate::BitsPerSec(123));
}
void CheckRfc8888Feedback(
const std::vector<std::unique_ptr<rtcp::RtcpPacket>>& rtcp_packets) {
ASSERT_THAT(rtcp_packets, SizeIs(1));
Buffer buffer = rtcp_packets[0]->Build();
rtcp::CommonHeader header;
EXPECT_TRUE(header.Parse(buffer.data(), buffer.size()));
// Check for RFC 8888 format message type 11(CCFB)
EXPECT_EQ(header.fmt(),
rtcp::CongestionControlFeedback::kFeedbackMessageType);
}
TEST(ReceiveSideCongestionControllerTest, SendsRfc8888FeedbackIfForced) {
test::ExplicitKeyValueConfig field_trials(
"WebRTC-RFC8888CongestionControlFeedback/force_send:true/");
MockFunction<void(std::vector<std::unique_ptr<rtcp::RtcpPacket>>)>
rtcp_sender;
MockFunction<void(uint64_t, std::vector<uint32_t>)> remb_sender;
SimulatedClock clock(123456);
ReceiveSideCongestionController controller(
CreateEnvironment(&clock, &field_trials), rtcp_sender.AsStdFunction(),
remb_sender.AsStdFunction());
// Expect that RTCP feedback is sent.
EXPECT_CALL(rtcp_sender, Call)
.WillOnce(
[&](std::vector<std::unique_ptr<rtcp::RtcpPacket>> rtcp_packets) {
CheckRfc8888Feedback(rtcp_packets);
});
// Expect that REMB is not sent.
EXPECT_CALL(remb_sender, Call).Times(0);
RtpPacketReceived packet;
packet.set_arrival_time(clock.CurrentTime());
controller.OnReceivedPacket(packet, MediaType::VIDEO);
TimeDelta next_process = controller.MaybeProcess();
clock.AdvanceTime(next_process);
next_process = controller.MaybeProcess();
}
TEST(ReceiveSideCongestionControllerTest, SendsRfc8888FeedbackIfEnabled) {
MockFunction<void(std::vector<std::unique_ptr<rtcp::RtcpPacket>>)>
rtcp_sender;
MockFunction<void(uint64_t, std::vector<uint32_t>)> remb_sender;
SimulatedClock clock(123456);
ReceiveSideCongestionController controller(CreateEnvironment(&clock),
rtcp_sender.AsStdFunction(),
remb_sender.AsStdFunction());
controller.EnableSendCongestionControlFeedbackAccordingToRfc8888();
// Expect that RTCP feedback is sent.
EXPECT_CALL(rtcp_sender, Call)
.WillOnce(
[&](std::vector<std::unique_ptr<rtcp::RtcpPacket>> rtcp_packets) {
CheckRfc8888Feedback(rtcp_packets);
});
// Expect that REMB is not sent.
EXPECT_CALL(remb_sender, Call).Times(0);
RtpPacketReceived packet;
packet.set_arrival_time(clock.CurrentTime());
controller.OnReceivedPacket(packet, MediaType::VIDEO);
TimeDelta next_process = controller.MaybeProcess();
clock.AdvanceTime(next_process);
next_process = controller.MaybeProcess();
}
TEST(ReceiveSideCongestionControllerTest,
SendsNoFeedbackIfNotRfcRfc8888EnabledAndNoTransportFeedback) {
MockFunction<void(std::vector<std::unique_ptr<rtcp::RtcpPacket>>)>
rtcp_sender;
MockFunction<void(uint64_t, std::vector<uint32_t>)> remb_sender;
SimulatedClock clock(123456);
ReceiveSideCongestionController controller(CreateEnvironment(&clock),
rtcp_sender.AsStdFunction(),
remb_sender.AsStdFunction());
// No Transport feedback is sent because received packet does not have
// transport sequence number rtp header extension.
EXPECT_CALL(rtcp_sender, Call).Times(0);
RtpPacketReceived packet;
packet.set_arrival_time(clock.CurrentTime());
controller.OnReceivedPacket(packet, MediaType::VIDEO);
TimeDelta next_process = controller.MaybeProcess();
clock.AdvanceTime(next_process);
next_process = controller.MaybeProcess();
}
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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