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author | Daniel Baumann <daniel.baumann@progress-linux.org> | 2024-04-07 19:33:14 +0000 |
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committer | Daniel Baumann <daniel.baumann@progress-linux.org> | 2024-04-07 19:33:14 +0000 |
commit | 36d22d82aa202bb199967e9512281e9a53db42c9 (patch) | |
tree | 105e8c98ddea1c1e4784a60a5a6410fa416be2de /third_party/libwebrtc/modules/audio_processing/aec3/block.h | |
parent | Initial commit. (diff) | |
download | firefox-esr-36d22d82aa202bb199967e9512281e9a53db42c9.tar.xz firefox-esr-36d22d82aa202bb199967e9512281e9a53db42c9.zip |
Adding upstream version 115.7.0esr.upstream/115.7.0esrupstream
Signed-off-by: Daniel Baumann <daniel.baumann@progress-linux.org>
Diffstat (limited to 'third_party/libwebrtc/modules/audio_processing/aec3/block.h')
-rw-r--r-- | third_party/libwebrtc/modules/audio_processing/aec3/block.h | 91 |
1 files changed, 91 insertions, 0 deletions
diff --git a/third_party/libwebrtc/modules/audio_processing/aec3/block.h b/third_party/libwebrtc/modules/audio_processing/aec3/block.h new file mode 100644 index 0000000000..c1fc70722d --- /dev/null +++ b/third_party/libwebrtc/modules/audio_processing/aec3/block.h @@ -0,0 +1,91 @@ +/* + * Copyright (c) 2022 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_BLOCK_H_ +#define MODULES_AUDIO_PROCESSING_AEC3_BLOCK_H_ + +#include <array> +#include <vector> + +#include "api/array_view.h" +#include "modules/audio_processing/aec3/aec3_common.h" + +namespace webrtc { + +// Contains one or more channels of 4 milliseconds of audio data. +// The audio is split in one or more frequency bands, each with a sampling +// rate of 16 kHz. +class Block { + public: + Block(int num_bands, int num_channels, float default_value = 0.0f) + : num_bands_(num_bands), + num_channels_(num_channels), + data_(num_bands * num_channels * kBlockSize, default_value) {} + + // Returns the number of bands. + int NumBands() const { return num_bands_; } + + // Returns the number of channels. + int NumChannels() const { return num_channels_; } + + // Modifies the number of channels and sets all samples to zero. + void SetNumChannels(int num_channels) { + num_channels_ = num_channels; + data_.resize(num_bands_ * num_channels_ * kBlockSize); + std::fill(data_.begin(), data_.end(), 0.0f); + } + + // Iterators for accessing the data. + auto begin(int band, int channel) { + return data_.begin() + GetIndex(band, channel); + } + + auto begin(int band, int channel) const { + return data_.begin() + GetIndex(band, channel); + } + + auto end(int band, int channel) { return begin(band, channel) + kBlockSize; } + + auto end(int band, int channel) const { + return begin(band, channel) + kBlockSize; + } + + // Access data via ArrayView. + rtc::ArrayView<float, kBlockSize> View(int band, int channel) { + return rtc::ArrayView<float, kBlockSize>(&data_[GetIndex(band, channel)], + kBlockSize); + } + + rtc::ArrayView<const float, kBlockSize> View(int band, int channel) const { + return rtc::ArrayView<const float, kBlockSize>( + &data_[GetIndex(band, channel)], kBlockSize); + } + + // Lets two Blocks swap audio data. + void Swap(Block& b) { + std::swap(num_bands_, b.num_bands_); + std::swap(num_channels_, b.num_channels_); + data_.swap(b.data_); + } + + private: + // Returns the index of the first sample of the requested |band| and + // |channel|. + int GetIndex(int band, int channel) const { + return (band * num_channels_ + channel) * kBlockSize; + } + + int num_bands_; + int num_channels_; + std::vector<float> data_; +}; + +} // namespace webrtc +#endif // MODULES_AUDIO_PROCESSING_AEC3_BLOCK_H_ |