1 /* 2 * various filters for ACELP-based codecs 3 * 4 * Copyright (c) 2008 Vladimir Voroshilov 5 * 6 * This file is part of FFmpeg. 7 * 8 * FFmpeg 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 * FFmpeg 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 FFmpeg; 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 typedef struct ACELPFContext { 29 /** 30 * Floating point version of ff_acelp_interpolate() 31 */ 32 void (*acelp_interpolatef)(float *out, const float *in, 33 const float *filter_coeffs, int precision, 34 int frac_pos, int filter_length, int length); 35 36 /** 37 * Apply an order 2 rational transfer function in-place. 38 * 39 * @param out output buffer for filtered speech samples 40 * @param in input buffer containing speech data (may be the same as out) 41 * @param zero_coeffs z^-1 and z^-2 coefficients of the numerator 42 * @param pole_coeffs z^-1 and z^-2 coefficients of the denominator 43 * @param gain scale factor for final output 44 * @param mem intermediate values used by filter (should be 0 initially) 45 * @param n number of samples (should be a multiple of eight) 46 */ 47 void (*acelp_apply_order_2_transfer_function)(float *out, const float *in, 48 const float zero_coeffs[2], 49 const float pole_coeffs[2], 50 float gain, 51 float mem[2], int n); 52 53 }ACELPFContext; 54 55 /** 56 * Initialize ACELPFContext. 57 */ 58 void ff_acelp_filter_init(ACELPFContext *c); 59 void ff_acelp_filter_init_mips(ACELPFContext *c); 60 61 /** 62 * low-pass Finite Impulse Response filter coefficients. 63 * 64 * Hamming windowed sinc filter with cutoff freq 3/40 of the sampling freq, 65 * the coefficients are scaled by 2^15. 66 * This array only contains the right half of the filter. 67 * This filter is likely identical to the one used in G.729, though this 68 * could not be determined from the original comments with certainty. 69 */ 70 extern const int16_t ff_acelp_interp_filter[61]; 71 72 /** 73 * Generic FIR interpolation routine. 74 * @param[out] out buffer for interpolated data 75 * @param in input data 76 * @param filter_coeffs interpolation filter coefficients (0.15) 77 * @param precision sub sample factor, that is the precision of the position 78 * @param frac_pos fractional part of position [0..precision-1] 79 * @param filter_length filter length 80 * @param length length of output 81 * 82 * filter_coeffs contains coefficients of the right half of the symmetric 83 * interpolation filter. filter_coeffs[0] should the central (unpaired) coefficient. 84 * See ff_acelp_interp_filter for an example. 85 */ 86 void ff_acelp_interpolate(int16_t* out, const int16_t* in, 87 const int16_t* filter_coeffs, int precision, 88 int frac_pos, int filter_length, int length); 89 90 /** 91 * Floating point version of ff_acelp_interpolate() 92 */ 93 void ff_acelp_interpolatef(float *out, const float *in, 94 const float *filter_coeffs, int precision, 95 int frac_pos, int filter_length, int length); 96 97 98 /** 99 * high-pass filtering and upscaling (4.2.5 of G.729). 100 * @param[out] out output buffer for filtered speech data 101 * @param[in,out] hpf_f past filtered data from previous (2 items long) 102 * frames (-0x20000000 <= (14.13) < 0x20000000) 103 * @param in speech data to process 104 * @param length input data size 105 * 106 * out[i] = 0.93980581 * in[i] - 1.8795834 * in[i-1] + 0.93980581 * in[i-2] + 107 * 1.9330735 * out[i-1] - 0.93589199 * out[i-2] 108 * 109 * The filter has a cut-off frequency of 1/80 of the sampling freq 110 * 111 * @note Two items before the top of the in buffer must contain two items from the 112 * tail of the previous subframe. 113 * 114 * @remark It is safe to pass the same array in in and out parameters. 115 * 116 * @remark AMR uses mostly the same filter (cut-off frequency 60Hz, same formula, 117 * but constants differs in 5th sign after comma). Fortunately in 118 * fixed-point all coefficients are the same as in G.729. Thus this 119 * routine can be used for the fixed-point AMR decoder, too. 120 */ 121 void ff_acelp_high_pass_filter(int16_t* out, int hpf_f[2], 122 const int16_t* in, int length); 123 124 /** 125 * Apply an order 2 rational transfer function in-place. 126 * 127 * @param out output buffer for filtered speech samples 128 * @param in input buffer containing speech data (may be the same as out) 129 * @param zero_coeffs z^-1 and z^-2 coefficients of the numerator 130 * @param pole_coeffs z^-1 and z^-2 coefficients of the denominator 131 * @param gain scale factor for final output 132 * @param mem intermediate values used by filter (should be 0 initially) 133 * @param n number of samples 134 */ 135 void ff_acelp_apply_order_2_transfer_function(float *out, const float *in, 136 const float zero_coeffs[2], 137 const float pole_coeffs[2], 138 float gain, 139 float mem[2], int n); 140 141 /** 142 * Apply tilt compensation filter, 1 - tilt * z-1. 143 * 144 * @param mem pointer to the filter's state (one single float) 145 * @param tilt tilt factor 146 * @param samples array where the filter is applied 147 * @param size the size of the samples array 148 */ 149 void ff_tilt_compensation(float *mem, float tilt, float *samples, int size); 150 151 152 #endif /* AVCODEC_ACELP_FILTERS_H */ 153