webrtc/modules/audio_processing/aec/main/source/resampler.h

33 lines
1.3 KiB
C

/*
* Copyright (c) 2011 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_MAIN_SOURCE_RESAMPLER_H_
#define WEBRTC_MODULES_AUDIO_PROCESSING_AEC_MAIN_SOURCE_RESAMPLER_H_
enum { kResamplingDelay = 1 };
// 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.
// Returns size of resampled array.
int WebRtcAec_ResampleLinear(void *resampInst,
const short *inspeech,
int size,
float skew,
short *outspeech);
#endif // WEBRTC_MODULES_AUDIO_PROCESSING_AEC_MAIN_SOURCE_RESAMPLER_H_