blob: 1280307a955dd1f1b3c453c51ec2b863684f4ec2
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 |