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/*
* 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.
*/
#include "modules/audio_processing/test/simulator_buffers.h"
#include "modules/audio_processing/test/audio_buffer_tools.h"
#include "rtc_base/checks.h"
namespace webrtc {
namespace test {
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) {
Random rand_gen(42);
CreateConfigAndBuffer(render_input_sample_rate_hz, num_render_input_channels,
&rand_gen, &render_input_buffer, &render_input_config,
&render_input, &render_input_samples);
CreateConfigAndBuffer(render_output_sample_rate_hz,
num_render_output_channels, &rand_gen,
&render_output_buffer, &render_output_config,
&render_output, &render_output_samples);
CreateConfigAndBuffer(capture_input_sample_rate_hz,
num_capture_input_channels, &rand_gen,
&capture_input_buffer, &capture_input_config,
&capture_input, &capture_input_samples);
CreateConfigAndBuffer(capture_output_sample_rate_hz,
num_capture_output_channels, &rand_gen,
&capture_output_buffer, &capture_output_config,
&capture_output, &capture_output_samples);
UpdateInputBuffers();
}
SimulatorBuffers::~SimulatorBuffers() = default;
void SimulatorBuffers::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) {
int samples_per_channel = rtc::CheckedDivExact(sample_rate_hz, 100);
*config = StreamConfig(sample_rate_hz, num_channels, false);
buffer->reset(new AudioBuffer(config->num_frames(), config->num_channels(),
config->num_frames(), config->num_channels(),
config->num_frames()));
buffer_data_samples->resize(samples_per_channel * num_channels);
for (auto& v : *buffer_data_samples) {
v = rand_gen->Rand<float>();
}
buffer_data->resize(num_channels);
for (size_t ch = 0; ch < num_channels; ++ch) {
(*buffer_data)[ch] = &(*buffer_data_samples)[ch * samples_per_channel];
}
}
void SimulatorBuffers::UpdateInputBuffers() {
test::CopyVectorToAudioBuffer(capture_input_config, capture_input_samples,
capture_input_buffer.get());
test::CopyVectorToAudioBuffer(render_input_config, render_input_samples,
render_input_buffer.get());
}
} // namespace test
} // namespace webrtc
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