1/* 2 * various filters for ACELP-based codecs 3 * 4 * Copyright (c) 2008 Vladimir Voroshilov 5 * 6 * This file is part of Libav. 7 * 8 * Libav is free software; you can redistribute it and/or 9 * modify it under the terms of the GNU Lesser General Public 10 * License as published by the Free Software Foundation; either 11 * version 2.1 of the License, or (at your option) any later version. 12 * 13 * Libav is distributed in the hope that it will be useful, 14 * but WITHOUT ANY WARRANTY; without even the implied warranty of 15 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU 16 * Lesser General Public License for more details. 17 * 18 * You should have received a copy of the GNU Lesser General Public 19 * License along with Libav; if not, write to the Free Software 20 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA 21 */ 22 23#ifndef AVCODEC_ACELP_FILTERS_H 24#define AVCODEC_ACELP_FILTERS_H 25 26#include <stdint.h> 27 28/** 29 * low-pass Finite Impulse Response filter coefficients. 30 * 31 * Hamming windowed sinc filter with cutoff freq 3/40 of the sampling freq, 32 * the coefficients are scaled by 2^15. 33 * This array only contains the right half of the filter. 34 * This filter is likely identical to the one used in G.729, though this 35 * could not be determined from the original comments with certainity. 36 */ 37extern const int16_t ff_acelp_interp_filter[61]; 38 39/** 40 * Generic FIR interpolation routine. 41 * @param[out] out buffer for interpolated data 42 * @param in input data 43 * @param filter_coeffs interpolation filter coefficients (0.15) 44 * @param precision sub sample factor, that is the precision of the position 45 * @param frac_pos fractional part of position [0..precision-1] 46 * @param filter_length filter length 47 * @param length length of output 48 * 49 * filter_coeffs contains coefficients of the right half of the symmetric 50 * interpolation filter. filter_coeffs[0] should the central (unpaired) coefficient. 51 * See ff_acelp_interp_filter for an example. 52 * 53 */ 54void ff_acelp_interpolate(int16_t* out, const int16_t* in, 55 const int16_t* filter_coeffs, int precision, 56 int frac_pos, int filter_length, int length); 57 58/** 59 * Floating point version of ff_acelp_interpolate() 60 */ 61void ff_acelp_interpolatef(float *out, const float *in, 62 const float *filter_coeffs, int precision, 63 int frac_pos, int filter_length, int length); 64 65 66/** 67 * high-pass filtering and upscaling (4.2.5 of G.729). 68 * @param[out] out output buffer for filtered speech data 69 * @param[in,out] hpf_f past filtered data from previous (2 items long) 70 * frames (-0x20000000 <= (14.13) < 0x20000000) 71 * @param in speech data to process 72 * @param length input data size 73 * 74 * out[i] = 0.93980581 * in[i] - 1.8795834 * in[i-1] + 0.93980581 * in[i-2] + 75 * 1.9330735 * out[i-1] - 0.93589199 * out[i-2] 76 * 77 * The filter has a cut-off frequency of 1/80 of the sampling freq 78 * 79 * @note Two items before the top of the out buffer must contain two items from the 80 * tail of the previous subframe. 81 * 82 * @remark It is safe to pass the same array in in and out parameters. 83 * 84 * @remark AMR uses mostly the same filter (cut-off frequency 60Hz, same formula, 85 * but constants differs in 5th sign after comma). Fortunately in 86 * fixed-point all coefficients are the same as in G.729. Thus this 87 * routine can be used for the fixed-point AMR decoder, too. 88 */ 89void ff_acelp_high_pass_filter(int16_t* out, int hpf_f[2], 90 const int16_t* in, int length); 91 92/** 93 * Apply an order 2 rational transfer function in-place. 94 * 95 * @param out output buffer for filtered speech samples 96 * @param in input buffer containing speech data (may be the same as out) 97 * @param zero_coeffs z^-1 and z^-2 coefficients of the numerator 98 * @param pole_coeffs z^-1 and z^-2 coefficients of the denominator 99 * @param gain scale factor for final output 100 * @param mem intermediate values used by filter (should be 0 initially) 101 * @param n number of samples 102 */ 103void ff_acelp_apply_order_2_transfer_function(float *out, const float *in, 104 const float zero_coeffs[2], 105 const float pole_coeffs[2], 106 float gain, 107 float mem[2], int n); 108 109/** 110 * Apply tilt compensation filter, 1 - tilt * z-1. 111 * 112 * @param mem pointer to the filter's state (one single float) 113 * @param tilt tilt factor 114 * @param samples array where the filter is applied 115 * @param size the size of the samples array 116 */ 117void ff_tilt_compensation(float *mem, float tilt, float *samples, int size); 118 119 120#endif /* AVCODEC_ACELP_FILTERS_H */ 121