blob: 60b4db13cb28af938e7065a9e5669fe6bd9a5c75 [file] [log] [blame]
/*
* 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/audio_coding/audio_network_adaptor/audio_network_adaptor_impl.h"
#include <utility>
#include <vector>
#include "api/rtc_event_log/rtc_event.h"
#include "logging/rtc_event_log/events/rtc_event_audio_network_adaptation.h"
#include "logging/rtc_event_log/mock/mock_rtc_event_log.h"
#include "modules/audio_coding/audio_network_adaptor/mock/mock_controller.h"
#include "modules/audio_coding/audio_network_adaptor/mock/mock_controller_manager.h"
#include "modules/audio_coding/audio_network_adaptor/mock/mock_debug_dump_writer.h"
#include "rtc_base/fake_clock.h"
#include "test/field_trial.h"
#include "test/gtest.h"
namespace webrtc {
using ::testing::_;
using ::testing::NiceMock;
using ::testing::Return;
using ::testing::SetArgPointee;
namespace {
constexpr size_t kNumControllers = 2;
constexpr int64_t kClockInitialTimeMs = 12345678;
MATCHER_P(NetworkMetricsIs, metric, "") {
return arg.uplink_bandwidth_bps == metric.uplink_bandwidth_bps &&
arg.target_audio_bitrate_bps == metric.target_audio_bitrate_bps &&
arg.rtt_ms == metric.rtt_ms &&
arg.overhead_bytes_per_packet == metric.overhead_bytes_per_packet &&
arg.uplink_packet_loss_fraction == metric.uplink_packet_loss_fraction;
}
MATCHER_P(IsRtcEventAnaConfigEqualTo, config, "") {
if (arg->GetType() != RtcEvent::Type::AudioNetworkAdaptation) {
return false;
}
auto ana_event = static_cast<RtcEventAudioNetworkAdaptation*>(arg);
return ana_event->config() == config;
}
MATCHER_P(EncoderRuntimeConfigIs, config, "") {
return arg.bitrate_bps == config.bitrate_bps &&
arg.frame_length_ms == config.frame_length_ms &&
arg.uplink_packet_loss_fraction ==
config.uplink_packet_loss_fraction &&
arg.enable_fec == config.enable_fec &&
arg.enable_dtx == config.enable_dtx &&
arg.num_channels == config.num_channels;
}
struct AudioNetworkAdaptorStates {
std::unique_ptr<AudioNetworkAdaptorImpl> audio_network_adaptor;
std::vector<std::unique_ptr<MockController>> mock_controllers;
std::unique_ptr<MockRtcEventLog> event_log;
MockDebugDumpWriter* mock_debug_dump_writer;
};
AudioNetworkAdaptorStates CreateAudioNetworkAdaptor() {
AudioNetworkAdaptorStates states;
std::vector<Controller*> controllers;
for (size_t i = 0; i < kNumControllers; ++i) {
auto controller =
std::unique_ptr<MockController>(new NiceMock<MockController>());
EXPECT_CALL(*controller, Die());
controllers.push_back(controller.get());
states.mock_controllers.push_back(std::move(controller));
}
auto controller_manager = std::unique_ptr<MockControllerManager>(
new NiceMock<MockControllerManager>());
EXPECT_CALL(*controller_manager, Die());
EXPECT_CALL(*controller_manager, GetControllers())
.WillRepeatedly(Return(controllers));
EXPECT_CALL(*controller_manager, GetSortedControllers(_))
.WillRepeatedly(Return(controllers));
states.event_log.reset(new NiceMock<MockRtcEventLog>());
auto debug_dump_writer =
std::unique_ptr<MockDebugDumpWriter>(new NiceMock<MockDebugDumpWriter>());
EXPECT_CALL(*debug_dump_writer, Die());
states.mock_debug_dump_writer = debug_dump_writer.get();
AudioNetworkAdaptorImpl::Config config;
config.event_log = states.event_log.get();
// AudioNetworkAdaptorImpl governs the lifetime of controller manager.
states.audio_network_adaptor.reset(new AudioNetworkAdaptorImpl(
config, std::move(controller_manager), std::move(debug_dump_writer)));
return states;
}
void SetExpectCallToUpdateNetworkMetrics(
const std::vector<std::unique_ptr<MockController>>& controllers,
const Controller::NetworkMetrics& check) {
for (auto& mock_controller : controllers) {
EXPECT_CALL(*mock_controller,
UpdateNetworkMetrics(NetworkMetricsIs(check)));
}
}
} // namespace
TEST(AudioNetworkAdaptorImplTest,
UpdateNetworkMetricsIsCalledOnSetUplinkBandwidth) {
auto states = CreateAudioNetworkAdaptor();
constexpr int kBandwidth = 16000;
Controller::NetworkMetrics check;
check.uplink_bandwidth_bps = kBandwidth;
SetExpectCallToUpdateNetworkMetrics(states.mock_controllers, check);
states.audio_network_adaptor->SetUplinkBandwidth(kBandwidth);
}
TEST(AudioNetworkAdaptorImplTest,
UpdateNetworkMetricsIsCalledOnSetUplinkPacketLossFraction) {
auto states = CreateAudioNetworkAdaptor();
constexpr float kPacketLoss = 0.7f;
Controller::NetworkMetrics check;
check.uplink_packet_loss_fraction = kPacketLoss;
SetExpectCallToUpdateNetworkMetrics(states.mock_controllers, check);
states.audio_network_adaptor->SetUplinkPacketLossFraction(kPacketLoss);
}
TEST(AudioNetworkAdaptorImplTest, UpdateNetworkMetricsIsCalledOnSetRtt) {
auto states = CreateAudioNetworkAdaptor();
constexpr int kRtt = 100;
Controller::NetworkMetrics check;
check.rtt_ms = kRtt;
SetExpectCallToUpdateNetworkMetrics(states.mock_controllers, check);
states.audio_network_adaptor->SetRtt(kRtt);
}
TEST(AudioNetworkAdaptorImplTest,
UpdateNetworkMetricsIsCalledOnSetTargetAudioBitrate) {
auto states = CreateAudioNetworkAdaptor();
constexpr int kTargetAudioBitrate = 15000;
Controller::NetworkMetrics check;
check.target_audio_bitrate_bps = kTargetAudioBitrate;
SetExpectCallToUpdateNetworkMetrics(states.mock_controllers, check);
states.audio_network_adaptor->SetTargetAudioBitrate(kTargetAudioBitrate);
}
TEST(AudioNetworkAdaptorImplTest, UpdateNetworkMetricsIsCalledOnSetOverhead) {
auto states = CreateAudioNetworkAdaptor();
constexpr size_t kOverhead = 64;
Controller::NetworkMetrics check;
check.overhead_bytes_per_packet = kOverhead;
SetExpectCallToUpdateNetworkMetrics(states.mock_controllers, check);
states.audio_network_adaptor->SetOverhead(kOverhead);
}
TEST(AudioNetworkAdaptorImplTest,
MakeDecisionIsCalledOnGetEncoderRuntimeConfig) {
auto states = CreateAudioNetworkAdaptor();
for (auto& mock_controller : states.mock_controllers)
EXPECT_CALL(*mock_controller, MakeDecision(_));
states.audio_network_adaptor->GetEncoderRuntimeConfig();
}
TEST(AudioNetworkAdaptorImplTest,
DumpEncoderRuntimeConfigIsCalledOnGetEncoderRuntimeConfig) {
test::ScopedFieldTrials override_field_trials(
"WebRTC-Audio-FecAdaptation/Enabled/");
rtc::ScopedFakeClock fake_clock;
fake_clock.AdvanceTime(TimeDelta::Millis(kClockInitialTimeMs));
auto states = CreateAudioNetworkAdaptor();
AudioEncoderRuntimeConfig config;
config.bitrate_bps = 32000;
config.enable_fec = true;
EXPECT_CALL(*states.mock_controllers[0], MakeDecision(_))
.WillOnce(SetArgPointee<0>(config));
EXPECT_CALL(*states.mock_debug_dump_writer,
DumpEncoderRuntimeConfig(EncoderRuntimeConfigIs(config),
kClockInitialTimeMs));
states.audio_network_adaptor->GetEncoderRuntimeConfig();
}
TEST(AudioNetworkAdaptorImplTest,
DumpNetworkMetricsIsCalledOnSetNetworkMetrics) {
rtc::ScopedFakeClock fake_clock;
fake_clock.AdvanceTime(TimeDelta::Millis(kClockInitialTimeMs));
auto states = CreateAudioNetworkAdaptor();
constexpr int kBandwidth = 16000;
constexpr float kPacketLoss = 0.7f;
constexpr int kRtt = 100;
constexpr int kTargetAudioBitrate = 15000;
constexpr size_t kOverhead = 64;
Controller::NetworkMetrics check;
check.uplink_bandwidth_bps = kBandwidth;
int64_t timestamp_check = kClockInitialTimeMs;
EXPECT_CALL(*states.mock_debug_dump_writer,
DumpNetworkMetrics(NetworkMetricsIs(check), timestamp_check));
states.audio_network_adaptor->SetUplinkBandwidth(kBandwidth);
fake_clock.AdvanceTime(TimeDelta::Millis(100));
timestamp_check += 100;
check.uplink_packet_loss_fraction = kPacketLoss;
EXPECT_CALL(*states.mock_debug_dump_writer,
DumpNetworkMetrics(NetworkMetricsIs(check), timestamp_check));
states.audio_network_adaptor->SetUplinkPacketLossFraction(kPacketLoss);
fake_clock.AdvanceTime(TimeDelta::Millis(50));
timestamp_check += 50;
fake_clock.AdvanceTime(TimeDelta::Millis(200));
timestamp_check += 200;
check.rtt_ms = kRtt;
EXPECT_CALL(*states.mock_debug_dump_writer,
DumpNetworkMetrics(NetworkMetricsIs(check), timestamp_check));
states.audio_network_adaptor->SetRtt(kRtt);
fake_clock.AdvanceTime(TimeDelta::Millis(150));
timestamp_check += 150;
check.target_audio_bitrate_bps = kTargetAudioBitrate;
EXPECT_CALL(*states.mock_debug_dump_writer,
DumpNetworkMetrics(NetworkMetricsIs(check), timestamp_check));
states.audio_network_adaptor->SetTargetAudioBitrate(kTargetAudioBitrate);
fake_clock.AdvanceTime(TimeDelta::Millis(50));
timestamp_check += 50;
check.overhead_bytes_per_packet = kOverhead;
EXPECT_CALL(*states.mock_debug_dump_writer,
DumpNetworkMetrics(NetworkMetricsIs(check), timestamp_check));
states.audio_network_adaptor->SetOverhead(kOverhead);
}
TEST(AudioNetworkAdaptorImplTest, LogRuntimeConfigOnGetEncoderRuntimeConfig) {
test::ScopedFieldTrials override_field_trials(
"WebRTC-Audio-FecAdaptation/Enabled/");
auto states = CreateAudioNetworkAdaptor();
AudioEncoderRuntimeConfig config;
config.bitrate_bps = 32000;
config.enable_fec = true;
EXPECT_CALL(*states.mock_controllers[0], MakeDecision(_))
.WillOnce(SetArgPointee<0>(config));
EXPECT_CALL(*states.event_log, LogProxy(IsRtcEventAnaConfigEqualTo(config)))
.Times(1);
states.audio_network_adaptor->GetEncoderRuntimeConfig();
}
TEST(AudioNetworkAdaptorImplTest, TestANAStats) {
auto states = CreateAudioNetworkAdaptor();
// Simulate some adaptation, otherwise the stats will not show anything.
AudioEncoderRuntimeConfig config1, config2;
config1.bitrate_bps = 32000;
config1.num_channels = 2;
config1.enable_fec = true;
config1.enable_dtx = true;
config1.frame_length_ms = 120;
config1.uplink_packet_loss_fraction = 0.1f;
config2.bitrate_bps = 16000;
config2.num_channels = 1;
config2.enable_fec = false;
config2.enable_dtx = false;
config2.frame_length_ms = 60;
config1.uplink_packet_loss_fraction = 0.1f;
EXPECT_CALL(*states.mock_controllers[0], MakeDecision(_))
.WillOnce(SetArgPointee<0>(config1));
states.audio_network_adaptor->GetEncoderRuntimeConfig();
EXPECT_CALL(*states.mock_controllers[0], MakeDecision(_))
.WillOnce(SetArgPointee<0>(config2));
states.audio_network_adaptor->GetEncoderRuntimeConfig();
EXPECT_CALL(*states.mock_controllers[0], MakeDecision(_))
.WillOnce(SetArgPointee<0>(config1));
states.audio_network_adaptor->GetEncoderRuntimeConfig();
auto ana_stats = states.audio_network_adaptor->GetStats();
EXPECT_EQ(ana_stats.bitrate_action_counter, 2u);
EXPECT_EQ(ana_stats.channel_action_counter, 2u);
EXPECT_EQ(ana_stats.dtx_action_counter, 2u);
EXPECT_EQ(ana_stats.fec_action_counter, 2u);
EXPECT_EQ(ana_stats.frame_length_increase_counter, 1u);
EXPECT_EQ(ana_stats.frame_length_decrease_counter, 1u);
EXPECT_EQ(ana_stats.uplink_packet_loss_fraction, 0.1f);
}
} // namespace webrtc