6b0679ff70a6225e146178e15730e33d877ba9b0
[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 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 FFMPEG_ACELP_FILTERS_H
24 #define FFMPEG_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(
55 int16_t* out,
56 const int16_t* in,
57 const int16_t* filter_coeffs,
58 int precision,
59 int frac_pos,
60 int filter_length,
61 int length);
62
63 /**
64 * Circularly convolve fixed vector with a phase dispersion impulse
65 * response filter (D.6.2 of G.729 and 6.1.5 of AMR).
66 * @param fc_out vector with filter applied
67 * @param fc_in source vector
68 * @param filter phase filter coefficients
69 *
70 * fc_out[n] = sum(i,0,len-1){ fc_in[i] * filter[(len + n - i)%len] }
71 *
72 * \note fc_in and fc_out should not overlap!
73 */
74 void ff_acelp_convolve_circ(
75 int16_t* fc_out,
76 const int16_t* fc_in,
77 const int16_t* filter,
78 int len);
79
80 /**
81 * LP synthesis filter.
82 * @param out [out] pointer to output buffer
83 * @param filter_coeffs filter coefficients (-0x8000 <= (3.12) < 0x8000)
84 * @param in input signal
85 * @param buffer_length amount of data to process
86 * @param filter_length filter length (10 for 10th order LP filter)
87 * @param stop_on_overflow 1 - return immediately if overflow occurs
88 * 0 - ignore overflows
89 * @param rounder the amount to add for rounding (usually 0x800 or 0xfff)
90 *
91 * @return 1 if overflow occurred, 0 - otherwise
92 *
93 * @note Output buffer must contain 10 samples of past
94 * speech data before pointer.
95 *
96 * Routine applies 1/A(z) filter to given speech data.
97 */
98 int ff_acelp_lp_synthesis_filter(
99 int16_t *out,
100 const int16_t* filter_coeffs,
101 const int16_t* in,
102 int buffer_length,
103 int filter_length,
104 int stop_on_overflow,
105 int rounder);
106
107
108 /**
109 * high-pass filtering and upscaling (4.2.5 of G.729).
110 * @param out [out] output buffer for filtered speech data
111 * @param hpf_f [in/out] past filtered data from previous (2 items long)
112 * frames (-0x20000000 <= (14.13) < 0x20000000)
113 * @param in speech data to process
114 * @param length input data size
115 *
116 * out[i] = 0.93980581 * in[i] - 1.8795834 * in[i-1] + 0.93980581 * in[i-2] +
117 * 1.9330735 * out[i-1] - 0.93589199 * out[i-2]
118 *
119 * The filter has a cut-off frequency of 1/80 of the sampling freq
120 *
121 * @note Two items before the top of the out buffer must contain two items from the
122 * tail of the previous subframe.
123 *
124 * @remark It is safe to pass the same array in in and out parameters.
125 *
126 * @remark AMR uses mostly the same filter (cut-off frequency 60Hz, same formula,
127 * but constants differs in 5th sign after comma). Fortunately in
128 * fixed-point all coefficients are the same as in G.729. Thus this
129 * routine can be used for the fixed-point AMR decoder, too.
130 */
131 void ff_acelp_high_pass_filter(
132 int16_t* out,
133 int hpf_f[2],
134 const int16_t* in,
135 int length);
136
137 #endif /* FFMPEG_ACELP_FILTERS_H */