simulator_buffers.cc (3736B)
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 #include "modules/audio_processing/test/simulator_buffers.h" 12 13 #include <cstddef> 14 #include <memory> 15 #include <vector> 16 17 #include "api/audio/audio_processing.h" 18 #include "modules/audio_processing/audio_buffer.h" 19 #include "modules/audio_processing/test/audio_buffer_tools.h" 20 #include "rtc_base/checks.h" 21 #include "rtc_base/random.h" 22 23 namespace webrtc { 24 namespace test { 25 26 SimulatorBuffers::SimulatorBuffers(int render_input_sample_rate_hz, 27 int capture_input_sample_rate_hz, 28 int render_output_sample_rate_hz, 29 int capture_output_sample_rate_hz, 30 size_t num_render_input_channels, 31 size_t num_capture_input_channels, 32 size_t num_render_output_channels, 33 size_t num_capture_output_channels) { 34 Random rand_gen(42); 35 CreateConfigAndBuffer(render_input_sample_rate_hz, num_render_input_channels, 36 &rand_gen, &render_input_buffer, &render_input_config, 37 &render_input, &render_input_samples); 38 39 CreateConfigAndBuffer(render_output_sample_rate_hz, 40 num_render_output_channels, &rand_gen, 41 &render_output_buffer, &render_output_config, 42 &render_output, &render_output_samples); 43 44 CreateConfigAndBuffer(capture_input_sample_rate_hz, 45 num_capture_input_channels, &rand_gen, 46 &capture_input_buffer, &capture_input_config, 47 &capture_input, &capture_input_samples); 48 49 CreateConfigAndBuffer(capture_output_sample_rate_hz, 50 num_capture_output_channels, &rand_gen, 51 &capture_output_buffer, &capture_output_config, 52 &capture_output, &capture_output_samples); 53 54 UpdateInputBuffers(); 55 } 56 57 SimulatorBuffers::~SimulatorBuffers() = default; 58 59 void SimulatorBuffers::CreateConfigAndBuffer( 60 int sample_rate_hz, 61 size_t num_channels, 62 Random* rand_gen, 63 std::unique_ptr<AudioBuffer>* buffer, 64 StreamConfig* config, 65 std::vector<float*>* buffer_data, 66 std::vector<float>* buffer_data_samples) { 67 int samples_per_channel = CheckedDivExact(sample_rate_hz, 100); 68 *config = StreamConfig(sample_rate_hz, num_channels); 69 buffer->reset( 70 new AudioBuffer(config->sample_rate_hz(), config->num_channels(), 71 config->sample_rate_hz(), config->num_channels(), 72 config->sample_rate_hz(), config->num_channels())); 73 74 buffer_data_samples->resize(samples_per_channel * num_channels); 75 for (auto& v : *buffer_data_samples) { 76 v = rand_gen->Rand<float>(); 77 } 78 79 buffer_data->resize(num_channels); 80 for (size_t ch = 0; ch < num_channels; ++ch) { 81 (*buffer_data)[ch] = &(*buffer_data_samples)[ch * samples_per_channel]; 82 } 83 } 84 85 void SimulatorBuffers::UpdateInputBuffers() { 86 test::CopyVectorToAudioBuffer(capture_input_config, capture_input_samples, 87 capture_input_buffer.get()); 88 test::CopyVectorToAudioBuffer(render_input_config, render_input_samples, 89 render_input_buffer.get()); 90 } 91 92 } // namespace test 93 } // namespace webrtc