Update to current webrtc library

This is from the upstream library commit id
3326535126e435f1ba647885ce43a8f0f3d317eb, corresponding to Chromium
88.0.4290.1.

webrtc/modules/audio_processing/aec3/fft_buffer.h

@@ -0,0 +1,60 @@
+/*
+ *  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_FFT_BUFFER_H_
+#define MODULES_AUDIO_PROCESSING_AEC3_FFT_BUFFER_H_
+
+#include <stddef.h>
+
+#include <vector>
+
+#include "modules/audio_processing/aec3/fft_data.h"
+#include "rtc_base/checks.h"
+
+namespace webrtc {
+
+// Struct for bundling a circular buffer of FftData objects together with the
+// read and write indices.
+struct FftBuffer {
+  FftBuffer(size_t size, size_t num_channels);
+  ~FftBuffer();
+
+  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<std::vector<FftData>> buffer;
+  int write = 0;
+  int read = 0;
+};
+
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_PROCESSING_AEC3_FFT_BUFFER_H_