| 1 | /* |
| 2 | * various filters for CELP-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_CELP_FILTERS_H |
| 24 | #define AVCODEC_CELP_FILTERS_H |
| 25 | |
| 26 | #include <stdint.h> |
| 27 | |
| 28 | typedef struct CELPFContext { |
| 29 | /** |
| 30 | * LP synthesis filter. |
| 31 | * @param[out] out pointer to output buffer |
| 32 | * - the array out[-filter_length, -1] must |
| 33 | * contain the previous result of this filter |
| 34 | * @param filter_coeffs filter coefficients. |
| 35 | * @param in input signal |
| 36 | * @param buffer_length amount of data to process |
| 37 | * @param filter_length filter length (10 for 10th order LP filter). Must be |
| 38 | * greater than 4 and even. |
| 39 | * |
| 40 | * @note Output buffer must contain filter_length samples of past |
| 41 | * speech data before pointer. |
| 42 | * |
| 43 | * Routine applies 1/A(z) filter to given speech data. |
| 44 | */ |
| 45 | void (*celp_lp_synthesis_filterf)(float *out, const float *filter_coeffs, |
| 46 | const float *in, int buffer_length, |
| 47 | int filter_length); |
| 48 | |
| 49 | /** |
| 50 | * LP zero synthesis filter. |
| 51 | * @param[out] out pointer to output buffer |
| 52 | * @param filter_coeffs filter coefficients. |
| 53 | * @param in input signal |
| 54 | * - the array in[-filter_length, -1] must |
| 55 | * contain the previous input of this filter |
| 56 | * @param buffer_length amount of data to process (should be a multiple of eight) |
| 57 | * @param filter_length filter length (10 for 10th order LP filter; |
| 58 | * should be a multiple of two) |
| 59 | * |
| 60 | * @note Output buffer must contain filter_length samples of past |
| 61 | * speech data before pointer. |
| 62 | * |
| 63 | * Routine applies A(z) filter to given speech data. |
| 64 | */ |
| 65 | void (*celp_lp_zero_synthesis_filterf)(float *out, const float *filter_coeffs, |
| 66 | const float *in, int buffer_length, |
| 67 | int filter_length); |
| 68 | |
| 69 | }CELPFContext; |
| 70 | |
| 71 | /** |
| 72 | * Initialize CELPFContext. |
| 73 | */ |
| 74 | void ff_celp_filter_init(CELPFContext *c); |
| 75 | void ff_celp_filter_init_mips(CELPFContext *c); |
| 76 | |
| 77 | /** |
| 78 | * Circularly convolve fixed vector with a phase dispersion impulse |
| 79 | * response filter (D.6.2 of G.729 and 6.1.5 of AMR). |
| 80 | * @param fc_out vector with filter applied |
| 81 | * @param fc_in source vector |
| 82 | * @param filter phase filter coefficients |
| 83 | * |
| 84 | * fc_out[n] = sum(i,0,len-1){ fc_in[i] * filter[(len + n - i)%len] } |
| 85 | * |
| 86 | * @note fc_in and fc_out should not overlap! |
| 87 | */ |
| 88 | void ff_celp_convolve_circ(int16_t *fc_out, const int16_t *fc_in, |
| 89 | const int16_t *filter, int len); |
| 90 | |
| 91 | /** |
| 92 | * Add an array to a rotated array. |
| 93 | * |
| 94 | * out[k] = in[k] + fac * lagged[k-lag] with wrap-around |
| 95 | * |
| 96 | * @param out result vector |
| 97 | * @param in samples to be added unfiltered |
| 98 | * @param lagged samples to be rotated, multiplied and added |
| 99 | * @param lag lagged vector delay in the range [0, n] |
| 100 | * @param fac scalefactor for lagged samples |
| 101 | * @param n number of samples |
| 102 | */ |
| 103 | void ff_celp_circ_addf(float *out, const float *in, |
| 104 | const float *lagged, int lag, float fac, int n); |
| 105 | |
| 106 | /** |
| 107 | * LP synthesis filter. |
| 108 | * @param[out] out pointer to output buffer |
| 109 | * @param filter_coeffs filter coefficients (-0x8000 <= (3.12) < 0x8000) |
| 110 | * @param in input signal |
| 111 | * @param buffer_length amount of data to process |
| 112 | * @param filter_length filter length (10 for 10th order LP filter) |
| 113 | * @param stop_on_overflow 1 - return immediately if overflow occurs |
| 114 | * 0 - ignore overflows |
| 115 | * @param shift the result is shifted right by this value |
| 116 | * @param rounder the amount to add for rounding (usually 0x800 or 0xfff) |
| 117 | * |
| 118 | * @return 1 if overflow occurred, 0 - otherwise |
| 119 | * |
| 120 | * @note Output buffer must contain filter_length samples of past |
| 121 | * speech data before pointer. |
| 122 | * |
| 123 | * Routine applies 1/A(z) filter to given speech data. |
| 124 | */ |
| 125 | int ff_celp_lp_synthesis_filter(int16_t *out, const int16_t *filter_coeffs, |
| 126 | const int16_t *in, int buffer_length, |
| 127 | int filter_length, int stop_on_overflow, |
| 128 | int shift, int rounder); |
| 129 | |
| 130 | /** |
| 131 | * LP synthesis filter. |
| 132 | * @param[out] out pointer to output buffer |
| 133 | * - the array out[-filter_length, -1] must |
| 134 | * contain the previous result of this filter |
| 135 | * @param filter_coeffs filter coefficients. |
| 136 | * @param in input signal |
| 137 | * @param buffer_length amount of data to process |
| 138 | * @param filter_length filter length (10 for 10th order LP filter). Must be |
| 139 | * greater than 4 and even. |
| 140 | * |
| 141 | * @note Output buffer must contain filter_length samples of past |
| 142 | * speech data before pointer. |
| 143 | * |
| 144 | * Routine applies 1/A(z) filter to given speech data. |
| 145 | */ |
| 146 | void ff_celp_lp_synthesis_filterf(float *out, const float *filter_coeffs, |
| 147 | const float *in, int buffer_length, |
| 148 | int filter_length); |
| 149 | |
| 150 | /** |
| 151 | * LP zero synthesis filter. |
| 152 | * @param[out] out pointer to output buffer |
| 153 | * @param filter_coeffs filter coefficients. |
| 154 | * @param in input signal |
| 155 | * - the array in[-filter_length, -1] must |
| 156 | * contain the previous input of this filter |
| 157 | * @param buffer_length amount of data to process |
| 158 | * @param filter_length filter length (10 for 10th order LP filter) |
| 159 | * |
| 160 | * @note Output buffer must contain filter_length samples of past |
| 161 | * speech data before pointer. |
| 162 | * |
| 163 | * Routine applies A(z) filter to given speech data. |
| 164 | */ |
| 165 | void ff_celp_lp_zero_synthesis_filterf(float *out, const float *filter_coeffs, |
| 166 | const float *in, int buffer_length, |
| 167 | int filter_length); |
| 168 | |
| 169 | #endif /* AVCODEC_CELP_FILTERS_H */ |