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
32int ff_sample_fmt_is_planar(enum AVSampleFormat sample_fmt, int channels);
33
34/**
35 * Audio buffer used for intermediate storage between conversion phases.
36 */
37struct 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
58int 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 */
79int ff_audio_data_init(AudioData *a, uint8_t * const *src, int plane_size,
80                       int channels, int nb_samples,
81                       enum AVSampleFormat sample_fmt, int read_only,
82                       const char *name);
83
84/**
85 * Allocate AudioData.
86 *
87 * This allocates an internal buffer and sets audio parameters.
88 *
89 * @param channels        channel count
90 * @param nb_samples      number of samples to allocate space for
91 * @param sample_fmt      sample format
92 * @param name            name for debug logging (can be NULL)
93 * @return                newly allocated AudioData struct, or NULL on error
94 */
95AudioData *ff_audio_data_alloc(int channels, int nb_samples,
96                               enum AVSampleFormat sample_fmt,
97                               const char *name);
98
99/**
100 * Reallocate AudioData.
101 *
102 * The AudioData must have been previously allocated with ff_audio_data_alloc().
103 *
104 * @param a           AudioData struct
105 * @param nb_samples  number of samples to allocate space for
106 * @return            0 on success, negative AVERROR value on error
107 */
108int ff_audio_data_realloc(AudioData *a, int nb_samples);
109
110/**
111 * Free AudioData.
112 *
113 * The AudioData must have been previously allocated with ff_audio_data_alloc().
114 *
115 * @param a  AudioData struct
116 */
117void ff_audio_data_free(AudioData **a);
118
119/**
120 * Copy data from one AudioData to another.
121 *
122 * @param out  output AudioData
123 * @param in   input AudioData
124 * @param map  channel map, NULL if not remapping
125 * @return     0 on success, negative AVERROR value on error
126 */
127int ff_audio_data_copy(AudioData *out, AudioData *in, ChannelMapInfo *map);
128
129/**
130 * Append data from one AudioData to the end of another.
131 *
132 * @param dst         destination AudioData
133 * @param dst_offset  offset, in samples, to start writing, relative to the
134 *                    start of dst
135 * @param src         source AudioData
136 * @param src_offset  offset, in samples, to start copying, relative to the
137 *                    start of the src
138 * @param nb_samples  number of samples to copy
139 * @return            0 on success, negative AVERROR value on error
140 */
141int ff_audio_data_combine(AudioData *dst, int dst_offset, AudioData *src,
142                          int src_offset, int nb_samples);
143
144/**
145 * Drain samples from the start of the AudioData.
146 *
147 * Remaining samples are shifted to the start of the AudioData.
148 *
149 * @param a           AudioData struct
150 * @param nb_samples  number of samples to drain
151 */
152void ff_audio_data_drain(AudioData *a, int nb_samples);
153
154/**
155 * Add samples in AudioData to an AVAudioFifo.
156 *
157 * @param af          Audio FIFO Buffer
158 * @param a           AudioData struct
159 * @param offset      number of samples to skip from the start of the data
160 * @param nb_samples  number of samples to add to the FIFO
161 * @return            number of samples actually added to the FIFO, or
162 *                    negative AVERROR code on error
163 */
164int ff_audio_data_add_to_fifo(AVAudioFifo *af, AudioData *a, int offset,
165                              int nb_samples);
166
167/**
168 * Read samples from an AVAudioFifo to AudioData.
169 *
170 * @param af          Audio FIFO Buffer
171 * @param a           AudioData struct
172 * @param nb_samples  number of samples to read from the FIFO
173 * @return            number of samples actually read from the FIFO, or
174 *                    negative AVERROR code on error
175 */
176int ff_audio_data_read_from_fifo(AVAudioFifo *af, AudioData *a, int nb_samples);
177
178#endif /* AVRESAMPLE_AUDIO_DATA_H */
179