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/**
29 * Circularly convolve fixed vector with a phase dispersion impulse
30 *        response filter (D.6.2 of G.729 and 6.1.5 of AMR).
31 * @param fc_out vector with filter applied
32 * @param fc_in source vector
33 * @param filter phase filter coefficients
34 *
35 *  fc_out[n] = sum(i,0,len-1){ fc_in[i] * filter[(len + n - i)%len] }
36 *
37 * \note fc_in and fc_out should not overlap!
38 */
39void ff_celp_convolve_circ(int16_t *fc_out, const int16_t *fc_in,
40                           const int16_t *filter, int len);
41
42/**
43 * Add an array to a rotated array.
44 *
45 * out[k] = in[k] + fac * lagged[k-lag] with wrap-around
46 *
47 * @param out result vector
48 * @param in samples to be added unfiltered
49 * @param lagged samples to be rotated, multiplied and added
50 * @param lag lagged vector delay in the range [0, n]
51 * @param fac scalefactor for lagged samples
52 * @param n number of samples
53 */
54void ff_celp_circ_addf(float *out, const float *in,
55                       const float *lagged, int lag, float fac, int n);
56
57/**
58 * LP synthesis filter.
59 * @param out [out] pointer to output buffer
60 * @param filter_coeffs filter coefficients (-0x8000 <= (3.12) < 0x8000)
61 * @param in input signal
62 * @param buffer_length amount of data to process
63 * @param filter_length filter length (10 for 10th order LP filter)
64 * @param stop_on_overflow   1 - return immediately if overflow occurs
65 *                           0 - ignore overflows
66 * @param rounder the amount to add for rounding (usually 0x800 or 0xfff)
67 *
68 * @return 1 if overflow occurred, 0 - otherwise
69 *
70 * @note Output buffer must contain filter_length samples of past
71 *       speech data before pointer.
72 *
73 * Routine applies 1/A(z) filter to given speech data.
74 */
75int ff_celp_lp_synthesis_filter(int16_t *out, const int16_t *filter_coeffs,
76                                const int16_t *in, int buffer_length,
77                                int filter_length, int stop_on_overflow,
78                                int rounder);
79
80/**
81 * LP synthesis filter.
82 * @param out [out] pointer to output buffer
83 *        - the array out[-filter_length, -1] must
84 *        contain the previous result of this filter
85 * @param filter_coeffs filter coefficients.
86 * @param in input signal
87 * @param buffer_length amount of data to process
88 * @param filter_length filter length (10 for 10th order LP filter). Must be
89 *                      greater than 4 and even.
90 *
91 * @note Output buffer must contain filter_length samples of past
92 *       speech data before pointer.
93 *
94 * Routine applies 1/A(z) filter to given speech data.
95 */
96void ff_celp_lp_synthesis_filterf(float *out, const float *filter_coeffs,
97                                  const float *in, int buffer_length,
98                                  int filter_length);
99
100/**
101 * LP zero synthesis filter.
102 * @param out [out] pointer to output buffer
103 * @param filter_coeffs filter coefficients.
104 * @param in input signal
105 *        - the array in[-filter_length, -1] must
106 *        contain the previous input of this filter
107 * @param buffer_length amount of data to process
108 * @param filter_length filter length (10 for 10th order LP filter)
109 *
110 * @note Output buffer must contain filter_length samples of past
111 *       speech data before pointer.
112 *
113 * Routine applies A(z) filter to given speech data.
114 */
115void ff_celp_lp_zero_synthesis_filterf(float *out, const float *filter_coeffs,
116                                       const float *in, int buffer_length,
117                                       int filter_length);
118
119#endif /* AVCODEC_CELP_FILTERS_H */
120