text: /*
   *  Copyright (c) 2012 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 MODULES_AUDIO_PROCESSING_AEC_AEC_RESAMPLER_H_
  #define MODULES_AUDIO_PROCESSING_AEC_AEC_RESAMPLER_H_
  
  #include <stddef.h>
  
  #include "modules/audio_processing/aec/aec_core.h"
  
  namespace webrtc
  {
  
  enum
  {
      kResamplingDelay = 1
  };
  enum
  {
      kResamplerBufferSize = FRAME_LEN * 4
  };
  
  // Unless otherwise specified, functions return 0 on success and -1 on error.
  void* WebRtcAec_CreateResampler(); // Returns NULL on error.
  int WebRtcAec_InitResampler(void* resampInst, int deviceSampleRateHz);
  void WebRtcAec_FreeResampler(void* resampInst);
  
  // Estimates skew from raw measurement.
  int WebRtcAec_GetSkew(void* resampInst, int rawSkew, float* skewEst);
  
  // Resamples input using linear interpolation.
  void WebRtcAec_ResampleLinear(void* resampInst,
      const float* inspeech,
      size_t size,
      float skew,
      float* outspeech,
      size_t* size_out);
  
  } // namespace webrtc
  
  #endif // MODULES_AUDIO_PROCESSING_AEC_AEC_RESAMPLER_H_
  