ivoc | 3cfb3ef | 2016-11-24 04:17:28 -0800 | [diff] [blame] | 1 | /* |
| 2 | * Copyright (c) 2016 The WebRTC project authors. All Rights Reserved. |
| 3 | * |
| 4 | * Use of this source code is governed by a BSD-style license |
| 5 | * that can be found in the LICENSE file in the root of the source |
| 6 | * tree. An additional intellectual property rights grant can be found |
| 7 | * in the file PATENTS. All contributing project authors may |
| 8 | * be found in the AUTHORS file in the root of the source tree. |
| 9 | */ |
| 10 | |
| 11 | #ifndef WEBRTC_MODULES_AUDIO_PROCESSING_TEST_SIMULATOR_BUFFERS_H_ |
| 12 | #define WEBRTC_MODULES_AUDIO_PROCESSING_TEST_SIMULATOR_BUFFERS_H_ |
| 13 | |
| 14 | #include <memory> |
| 15 | #include <vector> |
| 16 | |
ivoc | 3cfb3ef | 2016-11-24 04:17:28 -0800 | [diff] [blame] | 17 | #include "webrtc/modules/audio_processing/audio_buffer.h" |
| 18 | #include "webrtc/modules/audio_processing/include/audio_processing.h" |
Edward Lemur | c20978e | 2017-07-06 19:44:34 +0200 | [diff] [blame] | 19 | #include "webrtc/rtc_base/random.h" |
ivoc | 3cfb3ef | 2016-11-24 04:17:28 -0800 | [diff] [blame] | 20 | |
| 21 | namespace webrtc { |
| 22 | namespace test { |
| 23 | |
| 24 | struct SimulatorBuffers { |
| 25 | SimulatorBuffers(int render_input_sample_rate_hz, |
| 26 | int capture_input_sample_rate_hz, |
| 27 | int render_output_sample_rate_hz, |
| 28 | int capture_output_sample_rate_hz, |
| 29 | size_t num_render_input_channels, |
| 30 | size_t num_capture_input_channels, |
| 31 | size_t num_render_output_channels, |
| 32 | size_t num_capture_output_channels); |
| 33 | ~SimulatorBuffers(); |
| 34 | |
| 35 | void CreateConfigAndBuffer(int sample_rate_hz, |
| 36 | size_t num_channels, |
| 37 | Random* rand_gen, |
| 38 | std::unique_ptr<AudioBuffer>* buffer, |
| 39 | StreamConfig* config, |
| 40 | std::vector<float*>* buffer_data, |
| 41 | std::vector<float>* buffer_data_samples); |
| 42 | |
| 43 | void UpdateInputBuffers(); |
| 44 | |
| 45 | std::unique_ptr<AudioBuffer> render_input_buffer; |
| 46 | std::unique_ptr<AudioBuffer> capture_input_buffer; |
| 47 | std::unique_ptr<AudioBuffer> render_output_buffer; |
| 48 | std::unique_ptr<AudioBuffer> capture_output_buffer; |
| 49 | StreamConfig render_input_config; |
| 50 | StreamConfig capture_input_config; |
| 51 | StreamConfig render_output_config; |
| 52 | StreamConfig capture_output_config; |
| 53 | std::vector<float*> render_input; |
| 54 | std::vector<float> render_input_samples; |
| 55 | std::vector<float*> capture_input; |
| 56 | std::vector<float> capture_input_samples; |
| 57 | std::vector<float*> render_output; |
| 58 | std::vector<float> render_output_samples; |
| 59 | std::vector<float*> capture_output; |
| 60 | std::vector<float> capture_output_samples; |
| 61 | }; |
| 62 | |
| 63 | } // namespace test |
| 64 | } // namespace webrtc |
| 65 | |
| 66 | #endif // WEBRTC_MODULES_AUDIO_PROCESSING_TEST_SIMULATOR_BUFFERS_H_ |