mirror of
https://github.com/oxen-io/session-android.git
synced 2024-11-25 02:55:23 +00:00
40 lines
1.4 KiB
C
40 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 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_
|