blob: be9f5627914dbf41bb8dfd7e3f6286a7bd9c2453
1 | /* |
2 | * Copyright (c) 2004 Michael Niedermayer <michaelni@gmx.at> |
3 | * |
4 | * This file is part of FFmpeg. |
5 | * |
6 | * FFmpeg is free software; you can redistribute it and/or |
7 | * modify it under the terms of the GNU Lesser General Public |
8 | * License as published by the Free Software Foundation; either |
9 | * version 2.1 of the License, or (at your option) any later version. |
10 | * |
11 | * FFmpeg is distributed in the hope that it will be useful, |
12 | * but WITHOUT ANY WARRANTY; without even the implied warranty of |
13 | * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU |
14 | * Lesser General Public License for more details. |
15 | * |
16 | * You should have received a copy of the GNU Lesser General Public |
17 | * License along with FFmpeg; if not, write to the Free Software |
18 | * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA |
19 | */ |
20 | |
21 | #ifndef AVRESAMPLE_RESAMPLE_H |
22 | #define AVRESAMPLE_RESAMPLE_H |
23 | |
24 | #include "avresample.h" |
25 | #include "internal.h" |
26 | #include "audio_data.h" |
27 | |
28 | struct ResampleContext { |
29 | AVAudioResampleContext *avr; |
30 | AudioData *buffer; |
31 | uint8_t *filter_bank; |
32 | int filter_length; |
33 | int ideal_dst_incr; |
34 | int dst_incr; |
35 | unsigned int index; |
36 | int frac; |
37 | int src_incr; |
38 | int compensation_distance; |
39 | int phase_shift; |
40 | int phase_mask; |
41 | int linear; |
42 | enum AVResampleFilterType filter_type; |
43 | int kaiser_beta; |
44 | void (*set_filter)(void *filter, double *tab, int phase, int tap_count); |
45 | void (*resample_one)(struct ResampleContext *c, void *dst0, |
46 | int dst_index, const void *src0, |
47 | unsigned int index, int frac); |
48 | void (*resample_nearest)(void *dst0, int dst_index, |
49 | const void *src0, unsigned int index); |
50 | int padding_size; |
51 | int initial_padding_filled; |
52 | int initial_padding_samples; |
53 | int final_padding_filled; |
54 | int final_padding_samples; |
55 | }; |
56 | |
57 | /** |
58 | * Allocate and initialize a ResampleContext. |
59 | * |
60 | * The parameters in the AVAudioResampleContext are used to initialize the |
61 | * ResampleContext. |
62 | * |
63 | * @param avr AVAudioResampleContext |
64 | * @return newly-allocated ResampleContext |
65 | */ |
66 | ResampleContext *ff_audio_resample_init(AVAudioResampleContext *avr); |
67 | |
68 | /** |
69 | * Free a ResampleContext. |
70 | * |
71 | * @param c ResampleContext |
72 | */ |
73 | void ff_audio_resample_free(ResampleContext **c); |
74 | |
75 | /** |
76 | * Resample audio data. |
77 | * |
78 | * Changes the sample rate. |
79 | * |
80 | * @par |
81 | * All samples in the source data may not be consumed depending on the |
82 | * resampling parameters and the size of the output buffer. The unconsumed |
83 | * samples are automatically added to the start of the source in the next call. |
84 | * If the destination data can be reallocated, that may be done in this function |
85 | * in order to fit all available output. If it cannot be reallocated, fewer |
86 | * input samples will be consumed in order to have the output fit in the |
87 | * destination data buffers. |
88 | * |
89 | * @param c ResampleContext |
90 | * @param dst destination audio data |
91 | * @param src source audio data |
92 | * @return 0 on success, negative AVERROR code on failure |
93 | */ |
94 | int ff_audio_resample(ResampleContext *c, AudioData *dst, AudioData *src); |
95 | |
96 | #endif /* AVRESAMPLE_RESAMPLE_H */ |
97 |