Integrate Intelligibility with APM

webrtc/modules/audio_processing/audio_buffer.h

 /*
  *  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.
  */
 
@@ skipping 91 matching lines @@
 
   const float* keyboard_data() const;
 
   void set_activity(AudioFrame::VADActivity activity);
   AudioFrame::VADActivity activity() const;
 
   // Use for int16 interleaved data.
   void DeinterleaveFrom(AudioFrame* audioFrame);
   // If |data_changed| is false, only the non-audio data members will be copied
   // to |frame|.
-  void InterleaveTo(AudioFrame* frame, bool data_changed) const;
+  void InterleaveTo(AudioFrame* frame, bool data_changed);
 
   // Use for float deinterleaved data.
   void CopyFrom(const float* const* data,
                 int num_frames,
                 AudioProcessing::ChannelLayout layout);
   void CopyTo(int num_frames,
               AudioProcessing::ChannelLayout layout,
               float* const* data);
   void CopyLowPassToReference();
 
@@ skipping 26 matching lines @@
   bool reference_copied_;
   AudioFrame::VADActivity activity_;
 
   const float* keyboard_data_;
   rtc::scoped_ptr<IFChannelBuffer> data_;
   rtc::scoped_ptr<IFChannelBuffer> split_data_;
   rtc::scoped_ptr<SplittingFilter> splitting_filter_;
   rtc::scoped_ptr<ChannelBuffer<int16_t> > mixed_low_pass_channels_;
   rtc::scoped_ptr<ChannelBuffer<int16_t> > low_pass_reference_channels_;
   rtc::scoped_ptr<IFChannelBuffer> input_buffer_;
+  rtc::scoped_ptr<IFChannelBuffer> output_buffer_;
   rtc::scoped_ptr<ChannelBuffer<float> > process_buffer_;
   ScopedVector<PushSincResampler> input_resamplers_;
   ScopedVector<PushSincResampler> output_resamplers_;
 };
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_AUDIO_PROCESSING_AUDIO_BUFFER_H_