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author | Daniel Baumann <daniel.baumann@progress-linux.org> | 2024-04-19 00:47:55 +0000 |
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committer | Daniel Baumann <daniel.baumann@progress-linux.org> | 2024-04-19 00:47:55 +0000 |
commit | 26a029d407be480d791972afb5975cf62c9360a6 (patch) | |
tree | f435a8308119effd964b339f76abb83a57c29483 /third_party/libwebrtc/modules/audio_processing/aec3/downsampled_render_buffer.h | |
parent | Initial commit. (diff) | |
download | firefox-26a029d407be480d791972afb5975cf62c9360a6.tar.xz firefox-26a029d407be480d791972afb5975cf62c9360a6.zip |
Adding upstream version 124.0.1.upstream/124.0.1
Signed-off-by: Daniel Baumann <daniel.baumann@progress-linux.org>
Diffstat (limited to 'third_party/libwebrtc/modules/audio_processing/aec3/downsampled_render_buffer.h')
-rw-r--r-- | third_party/libwebrtc/modules/audio_processing/aec3/downsampled_render_buffer.h | 58 |
1 files changed, 58 insertions, 0 deletions
diff --git a/third_party/libwebrtc/modules/audio_processing/aec3/downsampled_render_buffer.h b/third_party/libwebrtc/modules/audio_processing/aec3/downsampled_render_buffer.h new file mode 100644 index 0000000000..fbdc9b4e93 --- /dev/null +++ b/third_party/libwebrtc/modules/audio_processing/aec3/downsampled_render_buffer.h @@ -0,0 +1,58 @@ +/* + * 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. + */ + +#ifndef MODULES_AUDIO_PROCESSING_AEC3_DOWNSAMPLED_RENDER_BUFFER_H_ +#define MODULES_AUDIO_PROCESSING_AEC3_DOWNSAMPLED_RENDER_BUFFER_H_ + +#include <stddef.h> + +#include <vector> + +#include "rtc_base/checks.h" + +namespace webrtc { + +// Holds the circular buffer of the downsampled render data. +struct DownsampledRenderBuffer { + explicit DownsampledRenderBuffer(size_t downsampled_buffer_size); + ~DownsampledRenderBuffer(); + + int IncIndex(int index) const { + RTC_DCHECK_EQ(buffer.size(), static_cast<size_t>(size)); + return index < size - 1 ? index + 1 : 0; + } + + int DecIndex(int index) const { + RTC_DCHECK_EQ(buffer.size(), static_cast<size_t>(size)); + return index > 0 ? index - 1 : size - 1; + } + + int OffsetIndex(int index, int offset) const { + RTC_DCHECK_GE(buffer.size(), offset); + RTC_DCHECK_EQ(buffer.size(), static_cast<size_t>(size)); + return (size + index + offset) % size; + } + + void UpdateWriteIndex(int offset) { write = OffsetIndex(write, offset); } + void IncWriteIndex() { write = IncIndex(write); } + void DecWriteIndex() { write = DecIndex(write); } + void UpdateReadIndex(int offset) { read = OffsetIndex(read, offset); } + void IncReadIndex() { read = IncIndex(read); } + void DecReadIndex() { read = DecIndex(read); } + + const int size; + std::vector<float> buffer; + int write = 0; + int read = 0; +}; + +} // namespace webrtc + +#endif // MODULES_AUDIO_PROCESSING_AEC3_DOWNSAMPLED_RENDER_BUFFER_H_ |