chromium/third_party/webrtc/modules/audio_processing/test/simulator_buffers.cc

/*
 *  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) { … }

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) { … }

void SimulatorBuffers::UpdateInputBuffers() { … }

}  // namespace test
}  // namespace webrtc