| /* | |
| * 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. | |
| */ | |
| #ifndef MODULES_AUDIO_PROCESSING_TEST_SIMULATOR_BUFFERS_H_ | |
| #define MODULES_AUDIO_PROCESSING_TEST_SIMULATOR_BUFFERS_H_ | |
| #include <memory> | |
| #include <vector> | |
| #include "api/audio/audio_processing.h" | |
| #include "modules/audio_processing/audio_buffer.h" | |
| #include "rtc_base/random.h" | |
| namespace webrtc { | |
| namespace test { | |
| struct SimulatorBuffers { | |
| SimulatorBuffers(int render_input_sample_rate_hz, | |
| int capture_input_sample_rate_hz, | |
| int render_output_sample_rate_hz, | |
| int capture_output_sample_rate_hz, | |
| size_t num_render_input_channels, | |
| size_t num_capture_input_channels, | |
| size_t num_render_output_channels, | |
| size_t num_capture_output_channels); | |
| ~SimulatorBuffers(); | |
| void CreateConfigAndBuffer(int sample_rate_hz, | |
| size_t num_channels, | |
| Random* rand_gen, | |
| std::unique_ptr<AudioBuffer>* buffer, | |
| StreamConfig* config, | |
| std::vector<float*>* buffer_data, | |
| std::vector<float>* buffer_data_samples); | |
| void UpdateInputBuffers(); | |
| std::unique_ptr<AudioBuffer> render_input_buffer; | |
| std::unique_ptr<AudioBuffer> capture_input_buffer; | |
| std::unique_ptr<AudioBuffer> render_output_buffer; | |
| std::unique_ptr<AudioBuffer> capture_output_buffer; | |
| StreamConfig render_input_config; | |
| StreamConfig capture_input_config; | |
| StreamConfig render_output_config; | |
| StreamConfig capture_output_config; | |
| std::vector<float*> render_input; | |
| std::vector<float> render_input_samples; | |
| std::vector<float*> capture_input; | |
| std::vector<float> capture_input_samples; | |
| std::vector<float*> render_output; | |
| std::vector<float> render_output_samples; | |
| std::vector<float*> capture_output; | |
| std::vector<float> capture_output_samples; | |
| }; | |
| } // namespace test | |
| } // namespace webrtc | |
| #endif // MODULES_AUDIO_PROCESSING_TEST_SIMULATOR_BUFFERS_H_ |