42 lines
1.4 KiB
C++
42 lines
1.4 KiB
C++
/*
|
|
* 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_
|