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| 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 |
| 17 #include "webrtc/base/random.h" |
| 18 #include "webrtc/modules/audio_processing/audio_buffer.h" |
| 19 #include "webrtc/modules/audio_processing/include/audio_processing.h" |
| 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_ |
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