Moved ring-buffer related files from common_audio to audio_processing

webrtc/modules/audio_processing/utility/lapped_transform.h

 /*
  *  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_COMMON_AUDIO_LAPPED_TRANSFORM_H_
-#define WEBRTC_COMMON_AUDIO_LAPPED_TRANSFORM_H_
+#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_UTILITY_LAPPED_TRANSFORM_H_
+#define WEBRTC_MODULES_AUDIO_PROCESSING_UTILITY_LAPPED_TRANSFORM_H_
 
 #include <complex>
 #include <memory>
 
-#include "webrtc/common_audio/blocker.h"
 #include "webrtc/common_audio/real_fourier.h"
+#include "webrtc/modules/audio_processing/utility/blocker.h"
 #include "webrtc/system_wrappers/include/aligned_array.h"
 
 namespace webrtc {
 
 // Helper class for audio processing modules which operate on frequency domain
 // input derived from the windowed time domain audio stream.
 //
 // The input audio chunk is sliced into possibly overlapping blocks, multiplied
 // by a window and transformed with an FFT implementation. The transformed data
 // is supplied to the given callback for processing. The processed output is
 // then inverse transformed into the time domain and spliced back into a chunk
 // which constitutes the final output of this processing module.
 class LappedTransform {
  public:
   class Callback {
    public:
     virtual ~Callback() {}
 
     virtual void ProcessAudioBlock(const std::complex<float>* const* in_block,
-                                   size_t num_in_channels, size_t frames,
+                                   size_t num_in_channels,
+                                   size_t frames,
                                    size_t num_out_channels,
                                    std::complex<float>* const* out_block) = 0;
   };
 
   // Construct a transform instance. |chunk_length| is the number of samples in
   // each channel. |window| defines the window, owned by the caller (a copy is
   // made internally); |window| should have length equal to |block_length|.
   // |block_length| defines the length of a block, in samples.
   // |shift_amount| is in samples. |callback| is the caller-owned audio
   // processing function called for each block of the input chunk.
@@ skipping 65 matching lines @@
 
   std::unique_ptr<RealFourier> fft_;
   const size_t cplx_length_;
   AlignedArray<float> real_buf_;
   AlignedArray<std::complex<float> > cplx_pre_;
   AlignedArray<std::complex<float> > cplx_post_;
 };
 
 }  // namespace webrtc
 
-#endif  // WEBRTC_COMMON_AUDIO_LAPPED_TRANSFORM_H_
-
+#endif  // WEBRTC_MODULES_AUDIO_PROCESSING_UTILITY_LAPPED_TRANSFORM_H_