| /* |
| * Copyright (c) 2017 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. |
| */ |
| #ifndef WEBRTC_AUDIO_TEST_LOW_BANDWIDTH_AUDIO_TEST_H_ |
| #define WEBRTC_AUDIO_TEST_LOW_BANDWIDTH_AUDIO_TEST_H_ |
| |
| #include <memory> |
| #include <string> |
| #include <vector> |
| |
| #include "webrtc/test/call_test.h" |
| #include "webrtc/test/fake_audio_device.h" |
| |
| namespace webrtc { |
| namespace test { |
| |
| class AudioQualityTest : public test::EndToEndTest { |
| public: |
| AudioQualityTest(); |
| |
| protected: |
| virtual std::string AudioInputFile(); |
| virtual std::string AudioOutputFile(); |
| |
| virtual FakeNetworkPipe::Config GetNetworkPipeConfig(); |
| |
| size_t GetNumVideoStreams() const override; |
| size_t GetNumAudioStreams() const override; |
| size_t GetNumFlexfecStreams() const override; |
| |
| std::unique_ptr<test::FakeAudioDevice::Capturer> CreateCapturer() override; |
| std::unique_ptr<test::FakeAudioDevice::Renderer> CreateRenderer() override; |
| |
| void OnFakeAudioDevicesCreated( |
| test::FakeAudioDevice* send_audio_device, |
| test::FakeAudioDevice* recv_audio_device) override; |
| |
| test::PacketTransport* CreateSendTransport( |
| SingleThreadedTaskQueueForTesting* task_queue, |
| Call* sender_call) override; |
| test::PacketTransport* CreateReceiveTransport( |
| SingleThreadedTaskQueueForTesting* task_queue) override; |
| |
| void ModifyAudioConfigs( |
| AudioSendStream::Config* send_config, |
| std::vector<AudioReceiveStream::Config>* receive_configs) override; |
| |
| void PerformTest() override; |
| void OnTestFinished() override; |
| |
| private: |
| test::FakeAudioDevice* send_audio_device_; |
| }; |
| |
| } // namespace test |
| } // namespace webrtc |
| |
| #endif // WEBRTC_AUDIO_TEST_LOW_BANDWIDTH_AUDIO_TEST_H_ |