blob: c105911aa8834df2049d281c84577fdfa558ec23 [file] [log] [blame]
* 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/downsampled_render_buffer.h"
#include <algorithm>
namespace webrtc {
DownsampledRenderBuffer::DownsampledRenderBuffer(size_t downsampled_buffer_size)
: size(static_cast<int>(downsampled_buffer_size)),
buffer(downsampled_buffer_size, 0.f) {
std::fill(buffer.begin(), buffer.end(), 0.f);
DownsampledRenderBuffer::~DownsampledRenderBuffer() = default;
} // namespace webrtc