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/decimator.h

@@ -0,0 +1,41 @@
+/*
+ *  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_DECIMATOR_H_
+#define MODULES_AUDIO_PROCESSING_AEC3_DECIMATOR_H_
+
+#include <array>
+#include <vector>
+
+#include "api/array_view.h"
+#include "modules/audio_processing/aec3/aec3_common.h"
+#include "modules/audio_processing/utility/cascaded_biquad_filter.h"
+#include "rtc_base/constructor_magic.h"
+
+namespace webrtc {
+
+// Provides functionality for decimating a signal.
+class Decimator {
+ public:
+  explicit Decimator(size_t down_sampling_factor);
+
+  // Downsamples the signal.
+  void Decimate(rtc::ArrayView<const float> in, rtc::ArrayView<float> out);
+
+ private:
+  const size_t down_sampling_factor_;
+  CascadedBiQuadFilter anti_aliasing_filter_;
+  CascadedBiQuadFilter noise_reduction_filter_;
+
+  RTC_DISALLOW_COPY_AND_ASSIGN(Decimator);
+};
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_PROCESSING_AEC3_DECIMATOR_H_