| /* |
| * 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 "modules/audio_processing/audio_buffer.h" |
| #include "modules/audio_processing/include/audio_processing.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_ |