libtgvoip/webrtc_dsp/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 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_