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diff --git a/src/modules/audio_processing/main/source/audio_buffer.h b/src/modules/audio_processing/main/source/audio_buffer.h
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+++ b/src/modules/audio_processing/main/source/audio_buffer.h
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+/*
+ *  Copyright (c) 2011 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_MAIN_SOURCE_AUDIO_BUFFER_H_
+#define WEBRTC_MODULES_AUDIO_PROCESSING_MAIN_SOURCE_AUDIO_BUFFER_H_
+
+#include "typedefs.h"
+
+
+namespace webrtc {
+
+struct AudioChannel;
+struct SplitAudioChannel;
+class AudioFrame;
+
+class AudioBuffer {
+ public:
+  AudioBuffer(WebRtc_Word32 max_num_channels, WebRtc_Word32 samples_per_channel);
+  virtual ~AudioBuffer();
+
+  WebRtc_Word32 num_channels() const;
+  WebRtc_Word32 samples_per_channel() const;
+  WebRtc_Word32 samples_per_split_channel() const;
+
+  WebRtc_Word16* data(WebRtc_Word32 channel) const;
+  WebRtc_Word16* low_pass_split_data(WebRtc_Word32 channel) const;
+  WebRtc_Word16* high_pass_split_data(WebRtc_Word32 channel) const;
+  WebRtc_Word16* mixed_low_pass_data(WebRtc_Word32 channel) const;
+  WebRtc_Word16* low_pass_reference(WebRtc_Word32 channel) const;
+
+  WebRtc_Word32* analysis_filter_state1(WebRtc_Word32 channel) const;
+  WebRtc_Word32* analysis_filter_state2(WebRtc_Word32 channel) const;
+  WebRtc_Word32* synthesis_filter_state1(WebRtc_Word32 channel) const;
+  WebRtc_Word32* synthesis_filter_state2(WebRtc_Word32 channel) const;
+
+  void DeinterleaveFrom(AudioFrame* audioFrame);
+  void InterleaveTo(AudioFrame* audioFrame) const;
+  void Mix(WebRtc_Word32 num_mixed_channels);
+  void CopyAndMixLowPass(WebRtc_Word32 num_mixed_channels);
+  void CopyLowPassToReference();
+
+ private:
+  const WebRtc_Word32 max_num_channels_;
+  WebRtc_Word32 num_channels_;
+  WebRtc_Word32 num_mixed_channels_;
+  WebRtc_Word32 num_mixed_low_pass_channels_;
+  const WebRtc_Word32 samples_per_channel_;
+  WebRtc_Word32 samples_per_split_channel_;
+  bool reference_copied_;
+
+  WebRtc_Word16* data_;
+  // TODO(ajm): Prefer to make these vectors if permitted...
+  AudioChannel* channels_;
+  SplitAudioChannel* split_channels_;
+  // TODO(ajm): improve this, we don't need the full 32 kHz space here.
+  AudioChannel* mixed_low_pass_channels_;
+  AudioChannel* low_pass_reference_channels_;
+};
+}  // namespace webrtc
+
+#endif  // WEBRTC_MODULES_AUDIO_PROCESSING_MAIN_SOURCE_AUDIO_BUFFER_H_