| /* |
| * 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_AEC_RESAMPLER_H_ |
| #define WEBRTC_MODULES_AUDIO_PROCESSING_AEC_AEC_RESAMPLER_H_ |
| |
| #include "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. |
| // 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_AEC_RESAMPLER_H_ |