ffmpeg / libavcodec / acelp_filters.h @ 2912e87a
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/*


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* various filters for ACELPbased codecs

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*

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* Copyright (c) 2008 Vladimir Voroshilov

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*

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* This file is part of Libav.

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*

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* Libav is free software; you can redistribute it and/or

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* modify it under the terms of the GNU Lesser General Public

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* License as published by the Free Software Foundation; either

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* version 2.1 of the License, or (at your option) any later version.

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*

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* Libav is distributed in the hope that it will be useful,

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* but WITHOUT ANY WARRANTY; without even the implied warranty of

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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU

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* Lesser General Public License for more details.

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*

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* You should have received a copy of the GNU Lesser General Public

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* License along with Libav; if not, write to the Free Software

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* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 021101301 USA

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*/

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#ifndef AVCODEC_ACELP_FILTERS_H

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#define AVCODEC_ACELP_FILTERS_H

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#include <stdint.h> 
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/**

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* lowpass Finite Impulse Response filter coefficients.

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*

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* Hamming windowed sinc filter with cutoff freq 3/40 of the sampling freq,

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* the coefficients are scaled by 2^15.

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* This array only contains the right half of the filter.

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* This filter is likely identical to the one used in G.729, though this

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* could not be determined from the original comments with certainity.

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*/

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extern const int16_t ff_acelp_interp_filter[61]; 
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/**

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* Generic FIR interpolation routine.

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* @param[out] out buffer for interpolated data

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* @param in input data

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* @param filter_coeffs interpolation filter coefficients (0.15)

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* @param precision sub sample factor, that is the precision of the position

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* @param frac_pos fractional part of position [0..precision1]

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* @param filter_length filter length

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* @param length length of output

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*

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* filter_coeffs contains coefficients of the right half of the symmetric

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* interpolation filter. filter_coeffs[0] should the central (unpaired) coefficient.

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* See ff_acelp_interp_filter for an example.

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*

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*/

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void ff_acelp_interpolate(int16_t* out, const int16_t* in, 
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const int16_t* filter_coeffs, int precision, 
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int frac_pos, int filter_length, int length); 
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/**

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* Floating point version of ff_acelp_interpolate()

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*/

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void ff_acelp_interpolatef(float *out, const float *in, 
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const float *filter_coeffs, int precision, 
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int frac_pos, int filter_length, int length); 
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/**

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* highpass filtering and upscaling (4.2.5 of G.729).

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* @param[out] out output buffer for filtered speech data

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* @param[in,out] hpf_f past filtered data from previous (2 items long)

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* frames (0x20000000 <= (14.13) < 0x20000000)

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* @param in speech data to process

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* @param length input data size

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*

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* out[i] = 0.93980581 * in[i]  1.8795834 * in[i1] + 0.93980581 * in[i2] +

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* 1.9330735 * out[i1]  0.93589199 * out[i2]

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*

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* The filter has a cutoff frequency of 1/80 of the sampling freq

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*

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* @note Two items before the top of the out buffer must contain two items from the

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* tail of the previous subframe.

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*

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* @remark It is safe to pass the same array in in and out parameters.

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*

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* @remark AMR uses mostly the same filter (cutoff frequency 60Hz, same formula,

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* but constants differs in 5th sign after comma). Fortunately in

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* fixedpoint all coefficients are the same as in G.729. Thus this

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* routine can be used for the fixedpoint AMR decoder, too.

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*/

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void ff_acelp_high_pass_filter(int16_t* out, int hpf_f[2], 
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const int16_t* in, int length); 
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/**

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* Apply an order 2 rational transfer function inplace.

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*

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* @param out output buffer for filtered speech samples

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* @param in input buffer containing speech data (may be the same as out)

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* @param zero_coeffs z^1 and z^2 coefficients of the numerator

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* @param pole_coeffs z^1 and z^2 coefficients of the denominator

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* @param gain scale factor for final output

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* @param mem intermediate values used by filter (should be 0 initially)

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* @param n number of samples

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*/

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void ff_acelp_apply_order_2_transfer_function(float *out, const float *in, 
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const float zero_coeffs[2], 
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const float pole_coeffs[2], 
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float gain,

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float mem[2], int n); 
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/**

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* Apply tilt compensation filter, 1  tilt * z1.

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*

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* @param mem pointer to the filter's state (one single float)

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* @param tilt tilt factor

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* @param samples array where the filter is applied

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* @param size the size of the samples array

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*/

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void ff_tilt_compensation(float *mem, float tilt, float *samples, int size); 
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#endif /* AVCODEC_ACELP_FILTERS_H */ 