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celp_filters.h
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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 */
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