aacps: Adjust some const qualifiers to suppress warnings
[libav.git] / libavcodec / acelp_filters.h
1 /*
2 * various filters for ACELP-based codecs
3 *
4 * Copyright (c) 2008 Vladimir Voroshilov
5 *
6 * This file is part of Libav.
7 *
8 * Libav 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 * Libav 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 Libav; 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 certainty.
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 */