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authorDaniel Baumann <daniel.baumann@progress-linux.org>2024-04-07 19:33:14 +0000
committerDaniel Baumann <daniel.baumann@progress-linux.org>2024-04-07 19:33:14 +0000
commit36d22d82aa202bb199967e9512281e9a53db42c9 (patch)
tree105e8c98ddea1c1e4784a60a5a6410fa416be2de /third_party/libwebrtc/modules/audio_processing/aec3/block.h
parentInitial commit. (diff)
downloadfirefox-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')
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+/*
+ * 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_