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92ea95e34a
In https://webrtc-review.googlesource.com/c/src/+/1560 we moved WebRTC from src/webrtc to src/ (in order to preserve an healthy git history). This CL takes care of fixing header guards, #include paths, etc... NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true TBR=tommi@webrtc.org Bug: chromium:611808 Change-Id: Iea91618212bee0af16aa3f05071eab8f93706578 Reviewed-on: https://webrtc-review.googlesource.com/1561 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Henrik Kjellander <kjellander@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#19846}
85 lines
3.5 KiB
C++
85 lines
3.5 KiB
C++
/*
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* Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#include "modules/audio_processing/test/simulator_buffers.h"
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#include "modules/audio_processing/test/audio_buffer_tools.h"
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#include "rtc_base/checks.h"
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namespace webrtc {
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namespace test {
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SimulatorBuffers::SimulatorBuffers(int render_input_sample_rate_hz,
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int capture_input_sample_rate_hz,
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int render_output_sample_rate_hz,
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int capture_output_sample_rate_hz,
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size_t num_render_input_channels,
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size_t num_capture_input_channels,
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size_t num_render_output_channels,
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size_t num_capture_output_channels) {
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Random rand_gen(42);
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CreateConfigAndBuffer(render_input_sample_rate_hz, num_render_input_channels,
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&rand_gen, &render_input_buffer, &render_input_config,
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&render_input, &render_input_samples);
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CreateConfigAndBuffer(render_output_sample_rate_hz,
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num_render_output_channels, &rand_gen,
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&render_output_buffer, &render_output_config,
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&render_output, &render_output_samples);
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CreateConfigAndBuffer(capture_input_sample_rate_hz,
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num_capture_input_channels, &rand_gen,
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&capture_input_buffer, &capture_input_config,
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&capture_input, &capture_input_samples);
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CreateConfigAndBuffer(capture_output_sample_rate_hz,
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num_capture_output_channels, &rand_gen,
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&capture_output_buffer, &capture_output_config,
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&capture_output, &capture_output_samples);
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UpdateInputBuffers();
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}
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SimulatorBuffers::~SimulatorBuffers() = default;
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void SimulatorBuffers::CreateConfigAndBuffer(
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int sample_rate_hz,
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size_t num_channels,
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Random* rand_gen,
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std::unique_ptr<AudioBuffer>* buffer,
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StreamConfig* config,
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std::vector<float*>* buffer_data,
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std::vector<float>* buffer_data_samples) {
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int samples_per_channel = rtc::CheckedDivExact(sample_rate_hz, 100);
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*config = StreamConfig(sample_rate_hz, num_channels, false);
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buffer->reset(new AudioBuffer(config->num_frames(), config->num_channels(),
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config->num_frames(), config->num_channels(),
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config->num_frames()));
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buffer_data_samples->resize(samples_per_channel * num_channels);
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for (auto& v : *buffer_data_samples) {
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v = rand_gen->Rand<float>();
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}
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buffer_data->resize(num_channels);
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for (size_t ch = 0; ch < num_channels; ++ch) {
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(*buffer_data)[ch] = &(*buffer_data_samples)[ch * samples_per_channel];
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}
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}
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void SimulatorBuffers::UpdateInputBuffers() {
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test::CopyVectorToAudioBuffer(capture_input_config, capture_input_samples,
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capture_input_buffer.get());
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test::CopyVectorToAudioBuffer(render_input_config, render_input_samples,
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render_input_buffer.get());
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}
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} // namespace test
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} // namespace webrtc
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