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
|
/*
* audio resampling
* Copyright (c) 2004-2012 Michael Niedermayer <michaelni@gmx.at>
*
* 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 SWRESAMPLE_RESAMPLE_H
#define SWRESAMPLE_RESAMPLE_H
#include "libavutil/log.h"
#include "libavutil/samplefmt.h"
#include "swresample_internal.h"
typedef struct ResampleContext {
const AVClass *av_class;
uint8_t *filter_bank;
int filter_length;
int filter_alloc;
int ideal_dst_incr;
int dst_incr;
int dst_incr_div;
int dst_incr_mod;
int index;
int frac;
int src_incr;
int compensation_distance;
int phase_shift;
int phase_mask;
int linear;
enum SwrFilterType filter_type;
double kaiser_beta;
double factor;
enum AVSampleFormat format;
int felem_size;
int filter_shift;
struct {
void (*resample_one)(void *dst, const void *src,
int n, int64_t index, int64_t incr);
int (*resample)(struct ResampleContext *c, void *dst,
const void *src, int n, int update_ctx);
} dsp;
} ResampleContext;
void swri_resample_dsp_init(ResampleContext *c);
void swri_resample_dsp_x86_init(ResampleContext *c);
void swri_resample_dsp_arm_init(ResampleContext *c);
#endif /* SWRESAMPLE_RESAMPLE_H */
|
