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diff --git a/third_party/libwebrtc/modules/audio_processing/include/audio_frame_view.h b/third_party/libwebrtc/modules/audio_processing/include/audio_frame_view.h
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+++ b/third_party/libwebrtc/modules/audio_processing/include/audio_frame_view.h
@@ -0,0 +1,68 @@
+/*
+ *  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, int num_channels, int channel_size)
+      : audio_samples_(audio_samples),
+        num_channels_(num_channels),
+        channel_size_(channel_size) {
+    RTC_DCHECK_GE(num_channels_, 0);
+    RTC_DCHECK_GE(channel_size_, 0);
+  }
+
+  // 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;
+
+  int num_channels() const { return num_channels_; }
+
+  int samples_per_channel() const { return channel_size_; }
+
+  rtc::ArrayView<T> channel(int 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(int 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_;
+  int num_channels_;
+  int channel_size_;
+};
+}  // namespace webrtc
+
+#endif  // MODULES_AUDIO_PROCESSING_INCLUDE_AUDIO_FRAME_VIEW_H_