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 MODULES_AUDIO_PROCESSING_AEC_AEC_RESAMPLER_H_
12 #define MODULES_AUDIO_PROCESSING_AEC_AEC_RESAMPLER_H_
13 
14 #include "modules/audio_processing/aec/aec_core.h"
15 
16 namespace webrtc {
17 
18 enum { kResamplingDelay = 1 };
19 enum { kResamplerBufferSize = FRAME_LEN * 4 };
20 
21 // Unless otherwise specified, functions return 0 on success and -1 on error.
22 void* WebRtcAec_CreateResampler();  // Returns NULL on error.
23 int WebRtcAec_InitResampler(void* resampInst, int deviceSampleRateHz);
24 void WebRtcAec_FreeResampler(void* resampInst);
25 
26 // Estimates skew from raw measurement.
27 int WebRtcAec_GetSkew(void* resampInst, int rawSkew, float* skewEst);
28 
29 // Resamples input using linear interpolation.
30 void WebRtcAec_ResampleLinear(void* resampInst,
31                               const float* inspeech,
32                               size_t size,
33                               float skew,
34                               float* outspeech,
35                               size_t* size_out);
36 
37 }  // namespace webrtc
38 
39 #endif  // MODULES_AUDIO_PROCESSING_AEC_AEC_RESAMPLER_H_
40