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