| Index: webrtc/modules/audio_coding/audio_network_adaptor/audio_network_adaptor_impl_unittest.cc
|
| diff --git a/webrtc/modules/audio_coding/audio_network_adaptor/audio_network_adaptor_impl_unittest.cc b/webrtc/modules/audio_coding/audio_network_adaptor/audio_network_adaptor_impl_unittest.cc
|
| index aee41da1b0b5c0523bf853ba483964a5ae488281..65626743adb3a59149b9e856766f7941d0bd9bcc 100644
|
| --- a/webrtc/modules/audio_coding/audio_network_adaptor/audio_network_adaptor_impl_unittest.cc
|
| +++ b/webrtc/modules/audio_coding/audio_network_adaptor/audio_network_adaptor_impl_unittest.cc
|
| @@ -15,25 +15,20 @@
|
| #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;
|
| }
|
|
|
| @@ -44,21 +39,9 @@
|
| 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() {
|
| @@ -81,19 +64,9 @@
|
| 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)));
|
| + AudioNetworkAdaptorImpl::Config(), std::move(controller_manager)));
|
|
|
| return states;
|
| }
|
| @@ -135,52 +108,4 @@
|
| 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
|
|
|