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/include/audio_frame_view.h

@@ -0,0 +1,67 @@
+/*
+ *  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_INCLUDE_AUDIO_FRAME_VIEW_H_
+#define MODULES_AUDIO_PROCESSING_INCLUDE_AUDIO_FRAME_VIEW_H_
+
+#include "api/array_view.h"
+
+namespace webrtc {
+
+// Class to pass audio data in T** format, where T is a numeric type.
+template <class T>
+class AudioFrameView {
+ public:
+  // |num_channels| and |channel_size| describe the T**
+  // |audio_samples|. |audio_samples| is assumed to point to a
+  // two-dimensional |num_channels * channel_size| array of floats.
+  AudioFrameView(T* const* audio_samples,
+                 size_t num_channels,
+                 size_t channel_size)
+      : audio_samples_(audio_samples),
+        num_channels_(num_channels),
+        channel_size_(channel_size) {}
+
+  // Implicit cast to allow converting Frame<float> to
+  // Frame<const float>.
+  template <class U>
+  AudioFrameView(AudioFrameView<U> other)
+      : audio_samples_(other.data()),
+        num_channels_(other.num_channels()),
+        channel_size_(other.samples_per_channel()) {}
+
+  AudioFrameView() = delete;
+
+  size_t num_channels() const { return num_channels_; }
+
+  size_t samples_per_channel() const { return channel_size_; }
+
+  rtc::ArrayView<T> channel(size_t idx) {
+    RTC_DCHECK_LE(0, idx);
+    RTC_DCHECK_LE(idx, num_channels_);
+    return rtc::ArrayView<T>(audio_samples_[idx], channel_size_);
+  }
+
+  rtc::ArrayView<const T> channel(size_t idx) const {
+    RTC_DCHECK_LE(0, idx);
+    RTC_DCHECK_LE(idx, num_channels_);
+    return rtc::ArrayView<const T>(audio_samples_[idx], channel_size_);
+  }
+
+  T* const* data() { return audio_samples_; }
+
+ private:
+  T* const* audio_samples_;
+  size_t num_channels_;
+  size_t channel_size_;
+};
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_PROCESSING_INCLUDE_AUDIO_FRAME_VIEW_H_