Update to current webrtc library
This is from the upstream library commit id 3326535126e435f1ba647885ce43a8f0f3d317eb, corresponding to Chromium 88.0.4290.1.
This commit is contained in:
39
webrtc/modules/audio_processing/aec3/block_buffer.cc
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39
webrtc/modules/audio_processing/aec3/block_buffer.cc
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/*
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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/block_buffer.h"
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#include <algorithm>
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namespace webrtc {
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BlockBuffer::BlockBuffer(size_t size,
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size_t num_bands,
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size_t num_channels,
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size_t frame_length)
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: size(static_cast<int>(size)),
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buffer(size,
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std::vector<std::vector<std::vector<float>>>(
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num_bands,
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std::vector<std::vector<float>>(
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num_channels,
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std::vector<float>(frame_length, 0.f)))) {
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for (auto& block : buffer) {
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for (auto& band : block) {
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for (auto& channel : band) {
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std::fill(channel.begin(), channel.end(), 0.f);
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}
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}
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}
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}
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BlockBuffer::~BlockBuffer() = default;
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} // namespace webrtc
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