chromium/third_party/webrtc/modules/audio_processing/aec3/render_buffer.cc

/*
 *  Copyright (c) 2017 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/aec3/render_buffer.h"

#include <algorithm>
#include <functional>

#include "modules/audio_processing/aec3/aec3_common.h"
#include "rtc_base/checks.h"

namespace webrtc {

RenderBuffer::RenderBuffer(BlockBuffer* block_buffer,
                           SpectrumBuffer* spectrum_buffer,
                           FftBuffer* fft_buffer)
    : … { … }

RenderBuffer::~RenderBuffer() = default;

void RenderBuffer::SpectralSum(
    size_t num_spectra,
    std::array<float, kFftLengthBy2Plus1>* X2) const { … }

void RenderBuffer::SpectralSums(
    size_t num_spectra_shorter,
    size_t num_spectra_longer,
    std::array<float, kFftLengthBy2Plus1>* X2_shorter,
    std::array<float, kFftLengthBy2Plus1>* X2_longer) const { … }

}  // namespace webrtc