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.cc

@@ -0,0 +1,62 @@
+/*
+ *  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.
+ */
+#include "modules/audio_processing/aec3/block_delay_buffer.h"
+
+#include "api/array_view.h"
+#include "rtc_base/checks.h"
+
+namespace webrtc {
+
+BlockDelayBuffer::BlockDelayBuffer(size_t num_channels,
+                                   size_t num_bands,
+                                   size_t frame_length,
+                                   size_t delay_samples)
+    : frame_length_(frame_length),
+      delay_(delay_samples),
+      buf_(num_channels,
+           std::vector<std::vector<float>>(num_bands,
+                                           std::vector<float>(delay_, 0.f))) {}
+
+BlockDelayBuffer::~BlockDelayBuffer() = default;
+
+void BlockDelayBuffer::DelaySignal(AudioBuffer* frame) {
+  RTC_DCHECK_EQ(buf_.size(), frame->num_channels());
+  if (delay_ == 0) {
+    return;
+  }
+
+  const size_t num_bands = buf_[0].size();
+  const size_t num_channels = buf_.size();
+
+  const size_t i_start = last_insert_;
+  size_t i = 0;
+  for (size_t ch = 0; ch < num_channels; ++ch) {
+    RTC_DCHECK_EQ(buf_[ch].size(), frame->num_bands());
+    RTC_DCHECK_EQ(buf_[ch].size(), num_bands);
+    rtc::ArrayView<float* const> frame_ch(frame->split_bands(ch), num_bands);
+
+    for (size_t band = 0; band < num_bands; ++band) {
+      RTC_DCHECK_EQ(delay_, buf_[ch][band].size());
+      i = i_start;
+
+      for (size_t k = 0; k < frame_length_; ++k) {
+        const float tmp = buf_[ch][band][i];
+        buf_[ch][band][i] = frame_ch[band][k];
+        frame_ch[band][k] = tmp;
+
+        i = i < delay_ - 1 ? i + 1 : 0;
+      }
+    }
+  }
+
+  last_insert_ = i;
+}
+
+}  // namespace webrtc