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Diffstat (limited to 'lib/rbcodec/codecs/libwmavoice/acelp_filters.h')
-rw-r--r-- | lib/rbcodec/codecs/libwmavoice/acelp_filters.h | 120 |
1 files changed, 120 insertions, 0 deletions
diff --git a/lib/rbcodec/codecs/libwmavoice/acelp_filters.h b/lib/rbcodec/codecs/libwmavoice/acelp_filters.h new file mode 100644 index 0000000000..0b1ccf4e71 --- /dev/null +++ b/lib/rbcodec/codecs/libwmavoice/acelp_filters.h | |||
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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 | /** | ||
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 | */ | ||
37 | extern 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 | */ | ||
54 | void 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 | */ | ||
61 | void 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 | */ | ||
89 | void 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 | */ | ||
103 | void 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 | */ | ||
117 | void ff_tilt_compensation(float *mem, float tilt, float *samples, int size); | ||
118 | |||
119 | |||
120 | #endif /* AVCODEC_ACELP_FILTERS_H */ | ||