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988cc0870b
This CL is the result of running include-what-you-use tool on part of the code base (audio target and dependencies) plus manual fixes. bug: webrtc:8311 Change-Id: I277d281ce943c3ecc1bd45fd8d83055931743604 Reviewed-on: https://webrtc-review.googlesource.com/c/106280 Commit-Queue: Yves Gerey <yvesg@google.com> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25311}
25 lines
845 B
C++
25 lines
845 B
C++
/*
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* Copyright (c) 2017 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/aec3/downsampled_render_buffer.h"
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#include <algorithm>
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namespace webrtc {
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DownsampledRenderBuffer::DownsampledRenderBuffer(size_t downsampled_buffer_size)
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: size(static_cast<int>(downsampled_buffer_size)),
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buffer(downsampled_buffer_size, 0.f) {
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std::fill(buffer.begin(), buffer.end(), 0.f);
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}
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DownsampledRenderBuffer::~DownsampledRenderBuffer() = default;
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} // namespace webrtc
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