repositories / deb_ffmpeg.git / blame
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1 | /* |
| 2 | * Copyright (c) 2012 Justin Ruggles <justin.ruggles@gmail.com> | |
| 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_AUDIO_DATA_H | |
| 22 | #define AVRESAMPLE_AUDIO_DATA_H | |
| 23 | ||
| 24 | #include <stdint.h> | |
| 25 | ||
| 26 | #include "libavutil/audio_fifo.h" | |
| 27 | #include "libavutil/log.h" | |
| 28 | #include "libavutil/samplefmt.h" | |
| 29 | #include "avresample.h" | |
| 30 | #include "internal.h" | |
| 31 | ||
| 32 | int ff_sample_fmt_is_planar(enum AVSampleFormat sample_fmt, int channels); | |
| 33 | ||
| 34 | /** | |
| 35 | * Audio buffer used for intermediate storage between conversion phases. | |
| 36 | */ | |
| 37 | struct AudioData { | |
| 38 | const AVClass *class; /**< AVClass for logging */ | |
| 39 | uint8_t *data[AVRESAMPLE_MAX_CHANNELS]; /**< data plane pointers */ | |
| 40 | uint8_t *buffer; /**< data buffer */ | |
| 41 | unsigned int buffer_size; /**< allocated buffer size */ | |
| 42 | int allocated_samples; /**< number of samples the buffer can hold */ | |
| 43 | int nb_samples; /**< current number of samples */ | |
| 44 | enum AVSampleFormat sample_fmt; /**< sample format */ | |
| 45 | int channels; /**< channel count */ | |
| 46 | int allocated_channels; /**< allocated channel count */ | |
| 47 | int is_planar; /**< sample format is planar */ | |
| 48 | int planes; /**< number of data planes */ | |
| 49 | int sample_size; /**< bytes per sample */ | |
| 50 | int stride; /**< sample byte offset within a plane */ | |
| 51 | int read_only; /**< data is read-only */ | |
| 52 | int allow_realloc; /**< realloc is allowed */ | |
| 53 | int ptr_align; /**< minimum data pointer alignment */ | |
| 54 | int samples_align; /**< allocated samples alignment */ | |
| 55 | const char *name; /**< name for debug logging */ | |
| 56 | }; | |
| 57 | ||
| 58 | int ff_audio_data_set_channels(AudioData *a, int channels); | |
| 59 | ||
| 60 | /** | |
| 61 | * Initialize AudioData using a given source. | |
| 62 | * | |
| 63 | * This does not allocate an internal buffer. It only sets the data pointers | |
| 64 | * and audio parameters. | |
| 65 | * | |
| 66 | * @param a AudioData struct | |
| 67 | * @param src source data pointers | |
| 68 | * @param plane_size plane size, in bytes. | |
| 69 | * This can be 0 if unknown, but that will lead to | |
| 70 | * optimized functions not being used in many cases, | |
| 71 | * which could slow down some conversions. | |
| 72 | * @param channels channel count | |
| 73 | * @param nb_samples number of samples in the source data | |
| 74 | * @param sample_fmt sample format | |
| 75 | * @param read_only indicates if buffer is read only or read/write | |
| 76 | * @param name name for debug logging (can be NULL) | |
| 77 | * @return 0 on success, negative AVERROR value on error | |
| 78 | */ | |
| 79 | int ff_audio_data_init(AudioData *a, uint8_t **src, int plane_size, int channels, | |
| 80 | int nb_samples, enum AVSampleFormat sample_fmt, | |
| 81 | int read_only, const char *name); | |
| 82 | ||
| 83 | /** | |
| 84 | * Allocate AudioData. | |
| 85 | * | |
| 86 | * This allocates an internal buffer and sets audio parameters. | |
| 87 | * | |
| 88 | * @param channels channel count | |
| 89 | * @param nb_samples number of samples to allocate space for | |
| 90 | * @param sample_fmt sample format | |
| 91 | * @param name name for debug logging (can be NULL) | |
| 92 | * @return newly allocated AudioData struct, or NULL on error | |
| 93 | */ | |
| 94 | AudioData *ff_audio_data_alloc(int channels, int nb_samples, | |
| 95 | enum AVSampleFormat sample_fmt, | |
| 96 | const char *name); | |
| 97 | ||
| 98 | /** | |
| 99 | * Reallocate AudioData. | |
| 100 | * | |
| 101 | * The AudioData must have been previously allocated with ff_audio_data_alloc(). | |
| 102 | * | |
| 103 | * @param a AudioData struct | |
| 104 | * @param nb_samples number of samples to allocate space for | |
| 105 | * @return 0 on success, negative AVERROR value on error | |
| 106 | */ | |
| 107 | int ff_audio_data_realloc(AudioData *a, int nb_samples); | |
| 108 | ||
| 109 | /** | |
| 110 | * Free AudioData. | |
| 111 | * | |
| 112 | * The AudioData must have been previously allocated with ff_audio_data_alloc(). | |
| 113 | * | |
| 114 | * @param a AudioData struct | |
| 115 | */ | |
| 116 | void ff_audio_data_free(AudioData **a); | |
| 117 | ||
| 118 | /** | |
| 119 | * Copy data from one AudioData to another. | |
| 120 | * | |
| 121 | * @param out output AudioData | |
| 122 | * @param in input AudioData | |
| 123 | * @param map channel map, NULL if not remapping | |
| 124 | * @return 0 on success, negative AVERROR value on error | |
| 125 | */ | |
| 126 | int ff_audio_data_copy(AudioData *out, AudioData *in, ChannelMapInfo *map); | |
| 127 | ||
| 128 | /** | |
| 129 | * Append data from one AudioData to the end of another. | |
| 130 | * | |
| 131 | * @param dst destination AudioData | |
| 132 | * @param dst_offset offset, in samples, to start writing, relative to the | |
| 133 | * start of dst | |
| 134 | * @param src source AudioData | |
| 135 | * @param src_offset offset, in samples, to start copying, relative to the | |
| 136 | * start of the src | |
| 137 | * @param nb_samples number of samples to copy | |
| 138 | * @return 0 on success, negative AVERROR value on error | |
| 139 | */ | |
| 140 | int ff_audio_data_combine(AudioData *dst, int dst_offset, AudioData *src, | |
| 141 | int src_offset, int nb_samples); | |
| 142 | ||
| 143 | /** | |
| 144 | * Drain samples from the start of the AudioData. | |
| 145 | * | |
| 146 | * Remaining samples are shifted to the start of the AudioData. | |
| 147 | * | |
| 148 | * @param a AudioData struct | |
| 149 | * @param nb_samples number of samples to drain | |
| 150 | */ | |
| 151 | void ff_audio_data_drain(AudioData *a, int nb_samples); | |
| 152 | ||
| 153 | /** | |
| 154 | * Add samples in AudioData to an AVAudioFifo. | |
| 155 | * | |
| 156 | * @param af Audio FIFO Buffer | |
| 157 | * @param a AudioData struct | |
| 158 | * @param offset number of samples to skip from the start of the data | |
| 159 | * @param nb_samples number of samples to add to the FIFO | |
| 160 | * @return number of samples actually added to the FIFO, or | |
| 161 | * negative AVERROR code on error | |
| 162 | */ | |
| 163 | int ff_audio_data_add_to_fifo(AVAudioFifo *af, AudioData *a, int offset, | |
| 164 | int nb_samples); | |
| 165 | ||
| 166 | /** | |
| 167 | * Read samples from an AVAudioFifo to AudioData. | |
| 168 | * | |
| 169 | * @param af Audio FIFO Buffer | |
| 170 | * @param a AudioData struct | |
| 171 | * @param nb_samples number of samples to read from the FIFO | |
| 172 | * @return number of samples actually read from the FIFO, or | |
| 173 | * negative AVERROR code on error | |
| 174 | */ | |
| 175 | int ff_audio_data_read_from_fifo(AVAudioFifo *af, AudioData *a, int nb_samples); | |
| 176 | ||
| 177 | #endif /* AVRESAMPLE_AUDIO_DATA_H */ |