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/block_delay_buffer.h

@@ -0,0 +1,43 @@
+/*
+ *  Copyright (c) 2018 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_DELAY_BUFFER_H_
+#define MODULES_AUDIO_PROCESSING_AEC3_BLOCK_DELAY_BUFFER_H_
+
+#include <stddef.h>
+
+#include <vector>
+
+#include "modules/audio_processing/audio_buffer.h"
+
+namespace webrtc {
+
+// Class for applying a fixed delay to the samples in a signal partitioned using
+// the audiobuffer band-splitting scheme.
+class BlockDelayBuffer {
+ public:
+  BlockDelayBuffer(size_t num_channels,
+                   size_t num_bands,
+                   size_t frame_length,
+                   size_t delay_samples);
+  ~BlockDelayBuffer();
+
+  // Delays the samples by the specified delay.
+  void DelaySignal(AudioBuffer* frame);
+
+ private:
+  const size_t frame_length_;
+  const size_t delay_;
+  std::vector<std::vector<std::vector<float>>> buf_;
+  size_t last_insert_ = 0;
+};
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_PROCESSING_AEC3_BLOCK_DELAY_BUFFER_H_