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audio_data.h
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1 /*
2  * Copyright (c) 2012 Justin Ruggles <justin.ruggles@gmail.com>
3  *
4  * This file is part of Libav.
5  *
6  * Libav 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  * Libav 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 Libav; 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 
31 /**
32  * Audio buffer used for intermediate storage between conversion phases.
33  */
34 typedef struct AudioData {
35  const AVClass *class; /**< AVClass for logging */
36  uint8_t *data[AVRESAMPLE_MAX_CHANNELS]; /**< data plane pointers */
37  uint8_t *buffer; /**< data buffer */
38  unsigned int buffer_size; /**< allocated buffer size */
39  int allocated_samples; /**< number of samples the buffer can hold */
40  int nb_samples; /**< current number of samples */
41  enum AVSampleFormat sample_fmt; /**< sample format */
42  int channels; /**< channel count */
43  int allocated_channels; /**< allocated channel count */
44  int is_planar; /**< sample format is planar */
45  int planes; /**< number of data planes */
46  int sample_size; /**< bytes per sample */
47  int stride; /**< sample byte offset within a plane */
48  int read_only; /**< data is read-only */
49  int allow_realloc; /**< realloc is allowed */
50  int ptr_align; /**< minimum data pointer alignment */
51  int samples_align; /**< allocated samples alignment */
52  const char *name; /**< name for debug logging */
53 } AudioData;
54 
55 int ff_audio_data_set_channels(AudioData *a, int channels);
56 
57 /**
58  * Initialize AudioData using a given source.
59  *
60  * This does not allocate an internal buffer. It only sets the data pointers
61  * and audio parameters.
62  *
63  * @param a AudioData struct
64  * @param src source data pointers
65  * @param plane_size plane size, in bytes.
66  * This can be 0 if unknown, but that will lead to
67  * optimized functions not being used in many cases,
68  * which could slow down some conversions.
69  * @param channels channel count
70  * @param nb_samples number of samples in the source data
71  * @param sample_fmt sample format
72  * @param read_only indicates if buffer is read only or read/write
73  * @param name name for debug logging (can be NULL)
74  * @return 0 on success, negative AVERROR value on error
75  */
76 int ff_audio_data_init(AudioData *a, uint8_t **src, int plane_size, int channels,
77  int nb_samples, enum AVSampleFormat sample_fmt,
78  int read_only, const char *name);
79 
80 /**
81  * Allocate AudioData.
82  *
83  * This allocates an internal buffer and sets audio parameters.
84  *
85  * @param channels channel count
86  * @param nb_samples number of samples to allocate space for
87  * @param sample_fmt sample format
88  * @param name name for debug logging (can be NULL)
89  * @return newly allocated AudioData struct, or NULL on error
90  */
91 AudioData *ff_audio_data_alloc(int channels, int nb_samples,
92  enum AVSampleFormat sample_fmt,
93  const char *name);
94 
95 /**
96  * Reallocate AudioData.
97  *
98  * The AudioData must have been previously allocated with ff_audio_data_alloc().
99  *
100  * @param a AudioData struct
101  * @param nb_samples number of samples to allocate space for
102  * @return 0 on success, negative AVERROR value on error
103  */
105 
106 /**
107  * Free AudioData.
108  *
109  * The AudioData must have been previously allocated with ff_audio_data_alloc().
110  *
111  * @param a AudioData struct
112  */
114 
115 /**
116  * Copy data from one AudioData to another.
117  *
118  * @param out output AudioData
119  * @param in input AudioData
120  * @return 0 on success, negative AVERROR value on error
121  */
122 int ff_audio_data_copy(AudioData *out, AudioData *in);
123 
124 /**
125  * Append data from one AudioData to the end of another.
126  *
127  * @param dst destination AudioData
128  * @param dst_offset offset, in samples, to start writing, relative to the
129  * start of dst
130  * @param src source AudioData
131  * @param src_offset offset, in samples, to start copying, relative to the
132  * start of the src
133  * @param nb_samples number of samples to copy
134  * @return 0 on success, negative AVERROR value on error
135  */
136 int ff_audio_data_combine(AudioData *dst, int dst_offset, AudioData *src,
137  int src_offset, int nb_samples);
138 
139 /**
140  * Drain samples from the start of the AudioData.
141  *
142  * Remaining samples are shifted to the start of the AudioData.
143  *
144  * @param a AudioData struct
145  * @param nb_samples number of samples to drain
146  */
148 
149 /**
150  * Add samples in AudioData to an AVAudioFifo.
151  *
152  * @param af Audio FIFO Buffer
153  * @param a AudioData struct
154  * @param offset number of samples to skip from the start of the data
155  * @param nb_samples number of samples to add to the FIFO
156  * @return number of samples actually added to the FIFO, or
157  * negative AVERROR code on error
158  */
160  int nb_samples);
161 
162 /**
163  * Read samples from an AVAudioFifo to AudioData.
164  *
165  * @param af Audio FIFO Buffer
166  * @param a AudioData struct
167  * @param nb_samples number of samples to read from the FIFO
168  * @return number of samples actually read from the FIFO, or
169  * negative AVERROR code on error
170  */
172 
173 #endif /* AVRESAMPLE_AUDIO_DATA_H */