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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 * 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  */
95 AudioData *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  */
108 int 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  */
117 void 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  */
127 int 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  */
141 int 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  */
152 void 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  */
164 int 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  */
176 int ff_audio_data_read_from_fifo(AVAudioFifo *af, AudioData *a, int nb_samples);
177 
178 #endif /* AVRESAMPLE_AUDIO_DATA_H */
179