33 lines
1.3 KiB
C
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_
|