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/*
* 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_
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