1
2
3
4
5
6
7
8
9
10
11
12
13
14
15
16
17
18
19
20
21
22
23
24
25
26
27
28
29
30
31
32
33
34
35
36
37
38
39
40
41
42
43
44
45
46
47
48
49
50
51
52
53
54
55
56
57
58
59
60
61
62
63
64
65
66
67
68
69
70
71
72
73
74
75
76
77
78
79
80
81
82
83
84
85
86
87
88
89
90
91
92
93
94
95
96
97
98
99
100
101
102
103
104
105
106
107
108
109
110
111
112
113
114
115
116
117
118
119
120
121
122
123
124
125
126
127
128
129
130
131
132
133
134
135
136
137
138
139
140
141
142
143
144
145
146
147
148
149
150
151
152
153
|
/*
* various filters for ACELP-based codecs
*
* Copyright (c) 2008 Vladimir Voroshilov
*
* This file is part of FFmpeg.
*
* FFmpeg is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
* FFmpeg is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
#ifndef AVCODEC_ACELP_FILTERS_H
#define AVCODEC_ACELP_FILTERS_H
#include <stdint.h>
typedef struct ACELPFContext {
/**
* Floating point version of ff_acelp_interpolate()
*/
void (*acelp_interpolatef)(float *out, const float *in,
const float *filter_coeffs, int precision,
int frac_pos, int filter_length, int length);
/**
* Apply an order 2 rational transfer function in-place.
*
* @param out output buffer for filtered speech samples
* @param in input buffer containing speech data (may be the same as out)
* @param zero_coeffs z^-1 and z^-2 coefficients of the numerator
* @param pole_coeffs z^-1 and z^-2 coefficients of the denominator
* @param gain scale factor for final output
* @param mem intermediate values used by filter (should be 0 initially)
* @param n number of samples (should be a multiple of eight)
*/
void (*acelp_apply_order_2_transfer_function)(float *out, const float *in,
const float zero_coeffs[2],
const float pole_coeffs[2],
float gain,
float mem[2], int n);
}ACELPFContext;
/**
* Initialize ACELPFContext.
*/
void ff_acelp_filter_init(ACELPFContext *c);
void ff_acelp_filter_init_mips(ACELPFContext *c);
/**
* low-pass Finite Impulse Response filter coefficients.
*
* Hamming windowed sinc filter with cutoff freq 3/40 of the sampling freq,
* the coefficients are scaled by 2^15.
* This array only contains the right half of the filter.
* This filter is likely identical to the one used in G.729, though this
* could not be determined from the original comments with certainty.
*/
extern const int16_t ff_acelp_interp_filter[61];
/**
* Generic FIR interpolation routine.
* @param[out] out buffer for interpolated data
* @param in input data
* @param filter_coeffs interpolation filter coefficients (0.15)
* @param precision sub sample factor, that is the precision of the position
* @param frac_pos fractional part of position [0..precision-1]
* @param filter_length filter length
* @param length length of output
*
* filter_coeffs contains coefficients of the right half of the symmetric
* interpolation filter. filter_coeffs[0] should the central (unpaired) coefficient.
* See ff_acelp_interp_filter for an example.
*
*/
void ff_acelp_interpolate(int16_t* out, const int16_t* in,
const int16_t* filter_coeffs, int precision,
int frac_pos, int filter_length, int length);
/**
* Floating point version of ff_acelp_interpolate()
*/
void ff_acelp_interpolatef(float *out, const float *in,
const float *filter_coeffs, int precision,
int frac_pos, int filter_length, int length);
/**
* high-pass filtering and upscaling (4.2.5 of G.729).
* @param[out] out output buffer for filtered speech data
* @param[in,out] hpf_f past filtered data from previous (2 items long)
* frames (-0x20000000 <= (14.13) < 0x20000000)
* @param in speech data to process
* @param length input data size
*
* out[i] = 0.93980581 * in[i] - 1.8795834 * in[i-1] + 0.93980581 * in[i-2] +
* 1.9330735 * out[i-1] - 0.93589199 * out[i-2]
*
* The filter has a cut-off frequency of 1/80 of the sampling freq
*
* @note Two items before the top of the in buffer must contain two items from the
* tail of the previous subframe.
*
* @remark It is safe to pass the same array in in and out parameters.
*
* @remark AMR uses mostly the same filter (cut-off frequency 60Hz, same formula,
* but constants differs in 5th sign after comma). Fortunately in
* fixed-point all coefficients are the same as in G.729. Thus this
* routine can be used for the fixed-point AMR decoder, too.
*/
void ff_acelp_high_pass_filter(int16_t* out, int hpf_f[2],
const int16_t* in, int length);
/**
* Apply an order 2 rational transfer function in-place.
*
* @param out output buffer for filtered speech samples
* @param in input buffer containing speech data (may be the same as out)
* @param zero_coeffs z^-1 and z^-2 coefficients of the numerator
* @param pole_coeffs z^-1 and z^-2 coefficients of the denominator
* @param gain scale factor for final output
* @param mem intermediate values used by filter (should be 0 initially)
* @param n number of samples
*/
void ff_acelp_apply_order_2_transfer_function(float *out, const float *in,
const float zero_coeffs[2],
const float pole_coeffs[2],
float gain,
float mem[2], int n);
/**
* Apply tilt compensation filter, 1 - tilt * z-1.
*
* @param mem pointer to the filter's state (one single float)
* @param tilt tilt factor
* @param samples array where the filter is applied
* @param size the size of the samples array
*/
void ff_tilt_compensation(float *mem, float tilt, float *samples, int size);
#endif /* AVCODEC_ACELP_FILTERS_H */
|
