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
