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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 */ |