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/agc2/rnn_vad/sequence_buffer.h

@@ -0,0 +1,79 @@
+/*
+ *  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_AGC2_RNN_VAD_SEQUENCE_BUFFER_H_
+#define MODULES_AUDIO_PROCESSING_AGC2_RNN_VAD_SEQUENCE_BUFFER_H_
+
+#include <algorithm>
+#include <cstring>
+#include <type_traits>
+#include <vector>
+
+#include "api/array_view.h"
+#include "rtc_base/checks.h"
+
+namespace webrtc {
+namespace rnn_vad {
+
+// Linear buffer implementation to (i) push fixed size chunks of sequential data
+// and (ii) view contiguous parts of the buffer. The buffer and the pushed
+// chunks have size S and N respectively. For instance, when S = 2N the first
+// half of the sequence buffer is replaced with its second half, and the new N
+// values are written at the end of the buffer.
+// The class also provides a view on the most recent M values, where 0 < M <= S
+// and by default M = N.
+template <typename T, size_t S, size_t N, size_t M = N>
+class SequenceBuffer {
+  static_assert(N <= S,
+                "The new chunk size cannot be larger than the sequence buffer "
+                "size.");
+  static_assert(std::is_arithmetic<T>::value,
+                "Integral or floating point required.");
+
+ public:
+  SequenceBuffer() : buffer_(S) {
+    RTC_DCHECK_EQ(S, buffer_.size());
+    Reset();
+  }
+  SequenceBuffer(const SequenceBuffer&) = delete;
+  SequenceBuffer& operator=(const SequenceBuffer&) = delete;
+  ~SequenceBuffer() = default;
+  size_t size() const { return S; }
+  size_t chunks_size() const { return N; }
+  // Sets the sequence buffer values to zero.
+  void Reset() { std::fill(buffer_.begin(), buffer_.end(), 0); }
+  // Returns a view on the whole buffer.
+  rtc::ArrayView<const T, S> GetBufferView() const {
+    return {buffer_.data(), S};
+  }
+  // Returns a view on the M most recent values of the buffer.
+  rtc::ArrayView<const T, M> GetMostRecentValuesView() const {
+    static_assert(M <= S,
+                  "The number of most recent values cannot be larger than the "
+                  "sequence buffer size.");
+    return {buffer_.data() + S - M, M};
+  }
+  // Shifts left the buffer by N items and add new N items at the end.
+  void Push(rtc::ArrayView<const T, N> new_values) {
+    // Make space for the new values.
+    if (S > N)
+      std::memmove(buffer_.data(), buffer_.data() + N, (S - N) * sizeof(T));
+    // Copy the new values at the end of the buffer.
+    std::memcpy(buffer_.data() + S - N, new_values.data(), N * sizeof(T));
+  }
+
+ private:
+  std::vector<T> buffer_;
+};
+
+}  // namespace rnn_vad
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_PROCESSING_AGC2_RNN_VAD_SEQUENCE_BUFFER_H_