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a4d873786f
This CL was generated by running git ls-files | grep -P "(\.h|\.cc)$" | grep -v 'sdk/' | grep -v 'rtc_base/ssl_' | \ grep -v 'fake_rtc_certificate_generator.h' | grep -v 'modules/audio_device/win/' | \ grep -v 'system_wrappers/source/clock.cc' | grep -v 'rtc_base/trace_event.h' | \ grep -v 'modules/audio_coding/codecs/ilbc/' | grep -v 'screen_capturer_mac.h' | \ grep -v 'spl_inl_mips.h' | grep -v 'data_size_unittest.cc' | grep -v 'timestamp_unittest.cc' \ | xargs clang-format -i ; git cl format Most of these changes are clang-format grouping and reordering includes differently. Bug: webrtc:9340 Change-Id: Ic83ddbc169bfacd21883e381b5181c3dd4fe8a63 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/144051 Commit-Queue: Jonas Olsson <jonasolsson@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#28505}
58 lines
1.8 KiB
C++
58 lines
1.8 KiB
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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#ifndef MODULES_AUDIO_PROCESSING_AEC3_DOWNSAMPLED_RENDER_BUFFER_H_
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#define MODULES_AUDIO_PROCESSING_AEC3_DOWNSAMPLED_RENDER_BUFFER_H_
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#include <stddef.h>
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#include <vector>
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#include "rtc_base/checks.h"
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namespace webrtc {
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// Holds the circular buffer of the downsampled render data.
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struct DownsampledRenderBuffer {
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explicit DownsampledRenderBuffer(size_t downsampled_buffer_size);
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~DownsampledRenderBuffer();
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int IncIndex(int index) const {
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RTC_DCHECK_EQ(buffer.size(), static_cast<size_t>(size));
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return index < size - 1 ? index + 1 : 0;
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}
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int DecIndex(int index) const {
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RTC_DCHECK_EQ(buffer.size(), static_cast<size_t>(size));
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return index > 0 ? index - 1 : size - 1;
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}
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int OffsetIndex(int index, int offset) const {
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RTC_DCHECK_GE(buffer.size(), offset);
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RTC_DCHECK_EQ(buffer.size(), static_cast<size_t>(size));
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return (size + index + offset) % size;
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}
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void UpdateWriteIndex(int offset) { write = OffsetIndex(write, offset); }
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void IncWriteIndex() { write = IncIndex(write); }
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void DecWriteIndex() { write = DecIndex(write); }
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void UpdateReadIndex(int offset) { read = OffsetIndex(read, offset); }
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void IncReadIndex() { read = IncIndex(read); }
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void DecReadIndex() { read = DecIndex(read); }
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const int size;
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std::vector<float> buffer;
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int write = 0;
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int read = 0;
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};
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} // namespace webrtc
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#endif // MODULES_AUDIO_PROCESSING_AEC3_DOWNSAMPLED_RENDER_BUFFER_H_
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