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diff --git a/src/modules/audio_processing/main/source/audio_processing_impl.h b/src/modules/audio_processing/main/source/audio_processing_impl.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_PROCESSING_IMPL_H_
+#define WEBRTC_MODULES_AUDIO_PROCESSING_MAIN_SOURCE_AUDIO_PROCESSING_IMPL_H_
+
+#include <list>
+
+#include "audio_processing.h"
+
+namespace webrtc {
+class CriticalSectionWrapper;
+class FileWrapper;
+
+class AudioBuffer;
+class EchoCancellationImpl;
+class EchoControlMobileImpl;
+class GainControlImpl;
+class HighPassFilterImpl;
+class LevelEstimatorImpl;
+class NoiseSuppressionImpl;
+class ProcessingComponent;
+class VoiceDetectionImpl;
+
+class AudioProcessingImpl : public AudioProcessing {
+ public:
+  enum {
+    kSampleRate8kHz = 8000,
+    kSampleRate16kHz = 16000,
+    kSampleRate32kHz = 32000
+  };
+
+  explicit AudioProcessingImpl(int id);
+  virtual ~AudioProcessingImpl();
+
+  CriticalSectionWrapper* crit() const;
+
+  int split_sample_rate_hz() const;
+  bool was_stream_delay_set() const;
+
+  // AudioProcessing methods.
+  virtual int Initialize();
+  virtual int InitializeLocked();
+  virtual int set_sample_rate_hz(int rate);
+  virtual int sample_rate_hz() const;
+  virtual int set_num_channels(int input_channels, int output_channels);
+  virtual int num_input_channels() const;
+  virtual int num_output_channels() const;
+  virtual int set_num_reverse_channels(int channels);
+  virtual int num_reverse_channels() const;
+  virtual int ProcessStream(AudioFrame* frame);
+  virtual int AnalyzeReverseStream(AudioFrame* frame);
+  virtual int set_stream_delay_ms(int delay);
+  virtual int stream_delay_ms() const;
+  virtual int StartDebugRecording(const char filename[kMaxFilenameSize]);
+  virtual int StopDebugRecording();
+  virtual EchoCancellation* echo_cancellation() const;
+  virtual EchoControlMobile* echo_control_mobile() const;
+  virtual GainControl* gain_control() const;
+  virtual HighPassFilter* high_pass_filter() const;
+  virtual LevelEstimator* level_estimator() const;
+  virtual NoiseSuppression* noise_suppression() const;
+  virtual VoiceDetection* voice_detection() const;
+
+  // Module methods.
+  virtual WebRtc_Word32 Version(WebRtc_Word8* version,
+                              WebRtc_UWord32& remainingBufferInBytes,
+                              WebRtc_UWord32& position) const;
+  virtual WebRtc_Word32 ChangeUniqueId(const WebRtc_Word32 id);
+
+ private:
+  int id_;
+
+  EchoCancellationImpl* echo_cancellation_;
+  EchoControlMobileImpl* echo_control_mobile_;
+  GainControlImpl* gain_control_;
+  HighPassFilterImpl* high_pass_filter_;
+  LevelEstimatorImpl* level_estimator_;
+  NoiseSuppressionImpl* noise_suppression_;
+  VoiceDetectionImpl* voice_detection_;
+
+  std::list<ProcessingComponent*> component_list_;
+
+  FileWrapper* debug_file_;
+  CriticalSectionWrapper* crit_;
+
+  AudioBuffer* render_audio_;
+  AudioBuffer* capture_audio_;
+
+  int sample_rate_hz_;
+  int split_sample_rate_hz_;
+  int samples_per_channel_;
+  int stream_delay_ms_;
+  bool was_stream_delay_set_;
+
+  int num_render_input_channels_;
+  int num_capture_input_channels_;
+  int num_capture_output_channels_;
+};
+}  // namespace webrtc
+
+#endif  // WEBRTC_MODULES_AUDIO_PROCESSING_MAIN_SOURCE_AUDIO_PROCESSING_IMPL_H_