| 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 | */ |
| 87 | void ff_acelp_interpolate(int16_t* out, const int16_t* in, |
| 88 | const int16_t* filter_coeffs, int precision, |
| 89 | int frac_pos, int filter_length, int length); |
| 90 | |
| 91 | /** |
| 92 | * Floating point version of ff_acelp_interpolate() |
| 93 | */ |
| 94 | void ff_acelp_interpolatef(float *out, const float *in, |
| 95 | const float *filter_coeffs, int precision, |
| 96 | int frac_pos, int filter_length, int length); |
| 97 | |
| 98 | |
| 99 | /** |
| 100 | * high-pass filtering and upscaling (4.2.5 of G.729). |
| 101 | * @param[out] out output buffer for filtered speech data |
| 102 | * @param[in,out] hpf_f past filtered data from previous (2 items long) |
| 103 | * frames (-0x20000000 <= (14.13) < 0x20000000) |
| 104 | * @param in speech data to process |
| 105 | * @param length input data size |
| 106 | * |
| 107 | * out[i] = 0.93980581 * in[i] - 1.8795834 * in[i-1] + 0.93980581 * in[i-2] + |
| 108 | * 1.9330735 * out[i-1] - 0.93589199 * out[i-2] |
| 109 | * |
| 110 | * The filter has a cut-off frequency of 1/80 of the sampling freq |
| 111 | * |
| 112 | * @note Two items before the top of the in buffer must contain two items from the |
| 113 | * tail of the previous subframe. |
| 114 | * |
| 115 | * @remark It is safe to pass the same array in in and out parameters. |
| 116 | * |
| 117 | * @remark AMR uses mostly the same filter (cut-off frequency 60Hz, same formula, |
| 118 | * but constants differs in 5th sign after comma). Fortunately in |
| 119 | * fixed-point all coefficients are the same as in G.729. Thus this |
| 120 | * routine can be used for the fixed-point AMR decoder, too. |
| 121 | */ |
| 122 | void ff_acelp_high_pass_filter(int16_t* out, int hpf_f[2], |
| 123 | const int16_t* in, int length); |
| 124 | |
| 125 | /** |
| 126 | * Apply an order 2 rational transfer function in-place. |
| 127 | * |
| 128 | * @param out output buffer for filtered speech samples |
| 129 | * @param in input buffer containing speech data (may be the same as out) |
| 130 | * @param zero_coeffs z^-1 and z^-2 coefficients of the numerator |
| 131 | * @param pole_coeffs z^-1 and z^-2 coefficients of the denominator |
| 132 | * @param gain scale factor for final output |
| 133 | * @param mem intermediate values used by filter (should be 0 initially) |
| 134 | * @param n number of samples |
| 135 | */ |
| 136 | void ff_acelp_apply_order_2_transfer_function(float *out, const float *in, |
| 137 | const float zero_coeffs[2], |
| 138 | const float pole_coeffs[2], |
| 139 | float gain, |
| 140 | float mem[2], int n); |
| 141 | |
| 142 | /** |
| 143 | * Apply tilt compensation filter, 1 - tilt * z-1. |
| 144 | * |
| 145 | * @param mem pointer to the filter's state (one single float) |
| 146 | * @param tilt tilt factor |
| 147 | * @param samples array where the filter is applied |
| 148 | * @param size the size of the samples array |
| 149 | */ |
| 150 | void ff_tilt_compensation(float *mem, float tilt, float *samples, int size); |
| 151 | |
| 152 | |
| 153 | #endif /* AVCODEC_ACELP_FILTERS_H */ |