1 /*
2  *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
3  *
4  *  Use of this source code is governed by a BSD-style license
5  *  that can be found in the LICENSE file in the root of the source
6  *  tree. An additional intellectual property rights grant can be found
7  *  in the file PATENTS.  All contributing project authors may
8  *  be found in the AUTHORS file in the root of the source tree.
9  */
10 
11 #ifndef WEBRTC_MODULES_AUDIO_PROCESSING_AEC_AEC_RESAMPLER_H_
12 #define WEBRTC_MODULES_AUDIO_PROCESSING_AEC_AEC_RESAMPLER_H_
13 
14 #include "webrtc/modules/audio_processing/aec/aec_core.h"
15 
16 enum {
17   kResamplingDelay = 1
18 };
19 enum {
20   kResamplerBufferSize = FRAME_LEN * 4
21 };
22 
23 // Unless otherwise specified, functions return 0 on success and -1 on error.
24 void* WebRtcAec_CreateResampler();  // Returns NULL on error.
25 int WebRtcAec_InitResampler(void* resampInst, int deviceSampleRateHz);
26 void WebRtcAec_FreeResampler(void* resampInst);
27 
28 // Estimates skew from raw measurement.
29 int WebRtcAec_GetSkew(void* resampInst, int rawSkew, float* skewEst);
30 
31 // Resamples input using linear interpolation.
32 void WebRtcAec_ResampleLinear(void* resampInst,
33                               const float* inspeech,
34                               size_t size,
35                               float skew,
36                               float* outspeech,
37                               size_t* size_out);
38 
39 #endif  // WEBRTC_MODULES_AUDIO_PROCESSING_AEC_AEC_RESAMPLER_H_
40