| /* |
| * 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 AUDIO_TEST_AUDIO_END_TO_END_TEST_H_ |
| #define AUDIO_TEST_AUDIO_END_TO_END_TEST_H_ |
| |
| #include <memory> |
| #include <string> |
| #include <vector> |
| |
| #include "api/audio/audio_device.h" |
| #include "api/task_queue/task_queue_base.h" |
| #include "api/test/simulated_network.h" |
| #include "modules/audio_device/include/test_audio_device.h" |
| #include "test/call_test.h" |
| |
| namespace webrtc { |
| namespace test { |
| |
| class AudioEndToEndTest : public test::EndToEndTest { |
| public: |
| AudioEndToEndTest(); |
| |
| protected: |
| AudioDeviceModule* send_audio_device() { return send_audio_device_; } |
| const AudioSendStream* send_stream() const { return send_stream_; } |
| const AudioReceiveStreamInterface* receive_stream() const { |
| return receive_stream_; |
| } |
| |
| size_t GetNumVideoStreams() const override; |
| size_t GetNumAudioStreams() const override; |
| size_t GetNumFlexfecStreams() const override; |
| |
| std::unique_ptr<TestAudioDeviceModule::Capturer> CreateCapturer() override; |
| std::unique_ptr<TestAudioDeviceModule::Renderer> CreateRenderer() override; |
| |
| void OnFakeAudioDevicesCreated(AudioDeviceModule* send_audio_device, |
| AudioDeviceModule* recv_audio_device) override; |
| |
| void ModifyAudioConfigs(AudioSendStream::Config* send_config, |
| std::vector<AudioReceiveStreamInterface::Config>* |
| receive_configs) override; |
| void OnAudioStreamsCreated(AudioSendStream* send_stream, |
| const std::vector<AudioReceiveStreamInterface*>& |
| receive_streams) override; |
| |
| private: |
| AudioDeviceModule* send_audio_device_ = nullptr; |
| AudioSendStream* send_stream_ = nullptr; |
| AudioReceiveStreamInterface* receive_stream_ = nullptr; |
| }; |
| |
| } // namespace test |
| } // namespace webrtc |
| |
| #endif // AUDIO_TEST_AUDIO_END_TO_END_TEST_H_ |