Bump to WebRTC M120 release

Some API deprecation -- ExperimentalAgc and ExperimentalNs are gone.
We're continuing to carry iSAC even though it's gone upstream, but maybe
we'll want to drop that soon.

webrtc/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_