blob: f644ec325e0dae110ef618a709a2b1d44d8b5229
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 */ |
170 |