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