26 lines
845 B
C++
26 lines
845 B
C++
|
/*
|
||
|
* 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
|