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diff --git a/chromium/third_party/webrtc/modules/audio_processing/common.h b/chromium/third_party/webrtc/modules/audio_processing/common.h
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+++ b/chromium/third_party/webrtc/modules/audio_processing/common.h
@@ -0,0 +1,76 @@
+/*
+ *  Copyright (c) 2014 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 WEBRTC_MODULES_AUDIO_PROCESSING_COMMON_H_
+#define WEBRTC_MODULES_AUDIO_PROCESSING_COMMON_H_
+
+#include <assert.h>
+#include <string.h>
+
+#include "webrtc/modules/audio_processing/include/audio_processing.h"
+#include "webrtc/system_wrappers/interface/scoped_ptr.h"
+
+namespace webrtc {
+
+static inline int ChannelsFromLayout(AudioProcessing::ChannelLayout layout) {
+  switch (layout) {
+    case AudioProcessing::kMono:
+    case AudioProcessing::kMonoAndKeyboard:
+      return 1;
+    case AudioProcessing::kStereo:
+    case AudioProcessing::kStereoAndKeyboard:
+      return 2;
+  }
+  assert(false);
+  return -1;
+}
+
+// Helper to encapsulate a contiguous data buffer with access to a pointer
+// array of the deinterleaved channels.
+template <typename T>
+class ChannelBuffer {
+ public:
+  ChannelBuffer(int samples_per_channel, int num_channels)
+      : data_(new T[samples_per_channel * num_channels]),
+        channels_(new T*[num_channels]),
+        samples_per_channel_(samples_per_channel),
+        num_channels_(num_channels) {
+    memset(data_.get(), 0, sizeof(T) * samples_per_channel * num_channels);
+    for (int i = 0; i < num_channels; ++i)
+      channels_[i] = &data_[i * samples_per_channel];
+  }
+  ~ChannelBuffer() {}
+
+  void CopyFrom(const void* channel_ptr, int i) {
+    assert(i < num_channels_);
+    memcpy(channels_[i], channel_ptr, samples_per_channel_ * sizeof(T));
+  }
+
+  T* data() { return data_.get(); }
+  T* channel(int i) {
+    assert(i < num_channels_);
+    return channels_[i];
+  }
+  T** channels() { return channels_.get(); }
+
+  int samples_per_channel() { return samples_per_channel_; }
+  int num_channels() { return num_channels_; }
+  int length() { return samples_per_channel_ * num_channels_; }
+
+ private:
+  scoped_ptr<T[]> data_;
+  scoped_ptr<T*[]> channels_;
+  int samples_per_channel_;
+  int num_channels_;
+};
+
+}  // namespace webrtc
+
+#endif  // WEBRTC_MODULES_AUDIO_PROCESSING_COMMON_H_