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
|
/*
* madplay - MPEG audio decoder and player
* Copyright (C) 2000-2004 Robert Leslie
*
* This program is free software; you can redistribute it and/or modify
* it under the terms of the GNU General Public License as published by
* the Free Software Foundation; either version 2 of the License, or
* (at your option) any later version.
*
* This program 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 General Public License for more details.
*
* You should have received a copy of the GNU General Public License
* along with this program; if not, write to the Free Software
* Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
*
* $Id: resample.c,v 1.11 2004/01/23 09:41:32 rob Exp $
*/
# ifdef HAVE_CONFIG_H
# include "config.h"
# endif
# include "global.h"
# include <string.h>
# include <mad.h>
# include "resample.h"
# include "audio.h"
/*
* NAME: resample_init()
* DESCRIPTION: initialize resampling state
*/
int resample_init(struct resample_state *state,
unsigned int oldrate, unsigned int newrate)
{
mad_fixed_t ratio;
if (newrate == 0)
return -1;
ratio = mad_f_div(oldrate, newrate);
if (ratio <= 0 || ratio > MAX_RESAMPLEFACTOR * MAD_F_ONE)
return -1;
state->ratio = ratio;
state->step = 0;
state->last = 0;
return 0;
}
/*
* NAME: resample_block()
* DESCRIPTION: algorithmically change the sampling rate of a PCM sample block
*/
unsigned int resample_block(struct resample_state *state,
unsigned int nsamples, mad_fixed_t const *old,
mad_fixed_t *new)
{
mad_fixed_t const *end, *begin;
/*
* This resampling algorithm is based on a linear interpolation, which is
* not at all the best sounding but is relatively fast and efficient.
*
* A better algorithm would be one that implements a bandlimited
* interpolation.
*/
if (state->ratio == MAD_F_ONE) {
memcpy(new, old, nsamples * sizeof(mad_fixed_t));
return nsamples;
}
end = old + nsamples;
begin = new;
if (state->step < 0) {
state->step = mad_f_fracpart(-state->step);
while (state->step < MAD_F_ONE) {
*new++ = state->step ?
state->last + mad_f_mul(*old - state->last, state->step) : state->last;
state->step += state->ratio;
if (((state->step + 0x00000080L) & 0x0fffff00L) == 0)
state->step = (state->step + 0x00000080L) & ~0x0fffffffL;
}
state->step -= MAD_F_ONE;
}
while (end - old > 1 + mad_f_intpart(state->step)) {
old += mad_f_intpart(state->step);
state->step = mad_f_fracpart(state->step);
*new++ = state->step ?
*old + mad_f_mul(old[1] - old[0], state->step) : *old;
state->step += state->ratio;
if (((state->step + 0x00000080L) & 0x0fffff00L) == 0)
state->step = (state->step + 0x00000080L) & ~0x0fffffffL;
}
if (end - old == 1 + mad_f_intpart(state->step)) {
state->last = end[-1];
state->step = -state->step;
}
else
state->step -= mad_f_fromint(end - old);
return new - begin;
}
|
