| 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 */ |