14b43beb7c
TBR=niklas.enbom@webrtc.org Review URL: https://webrtc-codereview.appspot.com/915006 git-svn-id: http://webrtc.googlecode.com/svn/trunk@2963 4adac7df-926f-26a2-2b94-8c16560cd09d
36 lines
1.4 KiB
C
36 lines
1.4 KiB
C
/*
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* Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_AEC_AEC_RESAMPLER_H_
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#define WEBRTC_MODULES_AUDIO_PROCESSING_AEC_AEC_RESAMPLER_H_
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#include "aec_core.h"
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enum { kResamplingDelay = 1 };
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enum { kResamplerBufferSize = FRAME_LEN * 4 };
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// Unless otherwise specified, functions return 0 on success and -1 on error
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int WebRtcAec_CreateResampler(void **resampInst);
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int WebRtcAec_InitResampler(void *resampInst, int deviceSampleRateHz);
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int WebRtcAec_FreeResampler(void *resampInst);
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// Estimates skew from raw measurement.
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int WebRtcAec_GetSkew(void *resampInst, int rawSkew, float *skewEst);
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// Resamples input using linear interpolation.
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void WebRtcAec_ResampleLinear(void *resampInst,
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const short *inspeech,
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int size,
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float skew,
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short *outspeech,
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int *size_out);
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#endif // WEBRTC_MODULES_AUDIO_PROCESSING_AEC_AEC_RESAMPLER_H_
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