modules/audio_processing/aec/aec_resampler.h

/*
 *  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 WEBRTC_MODULES_AUDIO_PROCESSING_AEC_AEC_RESAMPLER_H_
#define WEBRTC_MODULES_AUDIO_PROCESSING_AEC_AEC_RESAMPLER_H_

#include "webrtc/modules/audio_processing/aec/aec_core.h"

enum {
  kResamplingDelay = 1
};
enum {
  kResamplerBufferSize = FRAME_LEN * 4
};

// Unless otherwise specified, functions return 0 on success and -1 on error
int WebRtcAec_CreateResampler(void** resampInst);
int WebRtcAec_InitResampler(void* resampInst, int deviceSampleRateHz);
int 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,
                              int size,
                              float skew,
                              float* outspeech,
                              int* size_out);

#endif  // WEBRTC_MODULES_AUDIO_PROCESSING_AEC_AEC_RESAMPLER_H_