Chromium Code Reviews| 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 |
| new file mode 100644 |
| index 0000000000000000000000000000000000000000..46f56595b07871cfd0890ae748f20a3ba2d7a57d |
| --- /dev/null |
| +++ b/webrtc/modules/audio_coding/audio_network_adaptor/audio_network_adaptor_impl_unittest.cc |
| @@ -0,0 +1,110 @@ |
| +/* |
| + * 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" |
| + |
| +namespace webrtc { |
| + |
| +using ::testing::_; |
| +using ::testing::NiceMock; |
| +using ::testing::Return; |
| + |
| +namespace { |
| + |
| +constexpr size_t kNumControllers = 2; |
| + |
| +MATCHER_P(NetworkMetricsIs, metric, "") { |
| + return arg.uplink_bandwidth_bps == metric.uplink_bandwidth_bps && |
| + 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; |
| +} |
| + |
| +struct AudioNetworkAdaptorStates { |
| + std::unique_ptr<AudioNetworkAdaptorImpl> audio_network_adaptor; |
| + std::vector<std::unique_ptr<MockController>> mock_controllers; |
| +}; |
| + |
| +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)); |
| + |
| + // AudioNetworkAdaptorImpl governs the lifetime of controller manager. |
| + states.audio_network_adaptor.reset(new AudioNetworkAdaptorImpl( |
| + AudioNetworkAdaptorImpl::Config(), std::move(controller_manager))); |
| + |
| + return states; |
| +} |
| + |
| +} // namespace |
| + |
| +TEST(AudioNetworkAdaptorImplTest, |
| + MakeDecisionIsCalledOnGetEncoderRuntimeConfig) { |
| + AudioNetworkAdaptorStates states = CreateAudioNetworkAdaptor(); |
|
kwiberg-webrtc
2016/09/13 11:11:42
Excellent! Optionally, shorten this to
auto sta
minyue-webrtc
2016/09/13 11:48:50
yes, shorter is preferred. Thanks!
|
| + |
| + 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) { |
| + AudioNetworkAdaptorStates 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); |
| +} |
| + |
| +} // namespace webrtc |