blob: aee41da1b0b5c0523bf853ba483964a5ae488281 [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 <utility>
#include <vector>
#include "testing/gtest/include/gtest/gtest.h"
#include "webrtc/modules/audio_coding/audio_network_adaptor/audio_network_adaptor_impl.h"
#include "webrtc/modules/audio_coding/audio_network_adaptor/mock/mock_controller.h"
#include "webrtc/modules/audio_coding/audio_network_adaptor/mock/mock_controller_manager.h"
#include "webrtc/modules/audio_coding/audio_network_adaptor/mock/mock_debug_dump_writer.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.uplink_packet_loss_fraction == metric.uplink_packet_loss_fraction;
}
MATCHER_P(ConstraintsReceiverFrameLengthRangeIs, frame_length_range, "") {
return arg.receiver_frame_length_range->min_frame_length_ms ==
frame_length_range.min_frame_length_ms &&
arg.receiver_frame_length_range->max_frame_length_ms ==
frame_length_range.max_frame_length_ms;
}
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<SimulatedClock> simulated_clock;
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.simulated_clock.reset(new SimulatedClock(kClockInitialTimeMs * 1000));
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.clock = states.simulated_clock.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;
}
} // namespace
TEST(AudioNetworkAdaptorImplTest,
MakeDecisionIsCalledOnGetEncoderRuntimeConfig) {
auto states = CreateAudioNetworkAdaptor();
constexpr int kBandwidth = 16000;
constexpr float kPacketLoss = 0.7f;
Controller::NetworkMetrics check;
check.uplink_bandwidth_bps = rtc::Optional<int>(kBandwidth);
for (auto& mock_controller : states.mock_controllers) {
EXPECT_CALL(*mock_controller, MakeDecision(NetworkMetricsIs(check), _));
}
states.audio_network_adaptor->SetUplinkBandwidth(kBandwidth);
states.audio_network_adaptor->GetEncoderRuntimeConfig();
check.uplink_packet_loss_fraction = rtc::Optional<float>(kPacketLoss);
for (auto& mock_controller : states.mock_controllers) {
EXPECT_CALL(*mock_controller, MakeDecision(NetworkMetricsIs(check), _));
}
states.audio_network_adaptor->SetUplinkPacketLossFraction(kPacketLoss);
states.audio_network_adaptor->GetEncoderRuntimeConfig();
}
TEST(AudioNetworkAdaptorImplTest, SetConstraintsIsCalledOnSetFrameLengthRange) {
auto states = CreateAudioNetworkAdaptor();
for (auto& mock_controller : states.mock_controllers) {
EXPECT_CALL(*mock_controller,
SetConstraints(ConstraintsReceiverFrameLengthRangeIs(
Controller::Constraints::FrameLengthRange(20, 120))));
}
states.audio_network_adaptor->SetReceiverFrameLengthRange(20, 120);
}
TEST(AudioNetworkAdaptorImplTest,
DumpEncoderRuntimeConfigIsCalledOnGetEncoderRuntimeConfig) {
auto states = CreateAudioNetworkAdaptor();
AudioNetworkAdaptor::EncoderRuntimeConfig config;
config.bitrate_bps = rtc::Optional<int>(32000);
config.enable_fec = rtc::Optional<bool>(true);
EXPECT_CALL(*states.mock_controllers[0], MakeDecision(_, _))
.WillOnce(SetArgPointee<1>(config));
EXPECT_CALL(*states.mock_debug_dump_writer,
DumpEncoderRuntimeConfig(EncoderRuntimeConfigIs(config),
kClockInitialTimeMs));
states.audio_network_adaptor->GetEncoderRuntimeConfig();
}
TEST(AudioNetworkAdaptorImplTest,
DumpNetworkMetricsIsCalledOnSetNetworkMetrics) {
auto states = CreateAudioNetworkAdaptor();
constexpr int kBandwidth = 16000;
constexpr float kPacketLoss = 0.7f;
constexpr int kRtt = 100;
Controller::NetworkMetrics check;
check.uplink_bandwidth_bps = rtc::Optional<int>(kBandwidth);
int64_t timestamp_check = kClockInitialTimeMs;
EXPECT_CALL(*states.mock_debug_dump_writer,
DumpNetworkMetrics(NetworkMetricsIs(check), timestamp_check));
states.audio_network_adaptor->SetUplinkBandwidth(kBandwidth);
states.simulated_clock->AdvanceTimeMilliseconds(100);
timestamp_check += 100;
check.uplink_packet_loss_fraction = rtc::Optional<float>(kPacketLoss);
EXPECT_CALL(*states.mock_debug_dump_writer,
DumpNetworkMetrics(NetworkMetricsIs(check), timestamp_check));
states.audio_network_adaptor->SetUplinkPacketLossFraction(kPacketLoss);
states.simulated_clock->AdvanceTimeMilliseconds(200);
timestamp_check += 200;
check.rtt_ms = rtc::Optional<int>(kRtt);
EXPECT_CALL(*states.mock_debug_dump_writer,
DumpNetworkMetrics(NetworkMetricsIs(check), timestamp_check));
states.audio_network_adaptor->SetRtt(kRtt);
}
} // namespace webrtc