1
2
3
4
5
6
7
8
9
10
11
12
13
14
15
16
17
18
19
20
21
22
23
24
25
26
27
28
29
30
31
32
33
34
35
36
37
38
39
40
41
42
43
44
45
46
47
48
49
50
51
52
53
54
55
56
57
58
59
60
61
62
63
64
65
66
67
68
69
70
71
72
73
74
75
76
77
78
79
80
81
82
83
84
85
86
87
88
89
90
91
92
93
94
95
96
97
98
99
100
101
102
103
104
105
106
107
108
109
110
111
112
113
114
115
116
117
118
119
120
121
122
123
124
125
126
127
128
129
130
131
132
133
134
135
136
137
138
139
140
141
142
143
144
145
146
147
148
149
150
151
152
153
154
155
156
157
158
159
160
161
162
163
164
165
166
167
168
169
170
171
172
173
174
175
176
177
178
179
180
181
182
183
184
185
186
187
188
189
190
191
192
193
194
195
196
197
198
199
200
201
202
203
204
205
206
207
208
209
210
211
212
213
214
215
216
217
218
219
220
221
222
223
224
225
226
227
|
/*
* This file is part of Libav.
*
* Libav is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
* Libav is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with Libav; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
#ifndef AVUTIL_SAMPLEFMT_H
#define AVUTIL_SAMPLEFMT_H
#include <stdint.h>
#include "avutil.h"
#include "attributes.h"
/**
* Audio Sample Formats
*
* @par
* The data described by the sample format is always in native-endian order.
* Sample values can be expressed by native C types, hence the lack of a signed
* 24-bit sample format even though it is a common raw audio data format.
*
* @par
* The floating-point formats are based on full volume being in the range
* [-1.0, 1.0]. Any values outside this range are beyond full volume level.
*
* @par
* The data layout as used in av_samples_fill_arrays() and elsewhere in Libav
* (such as AVFrame in libavcodec) is as follows:
*
* For planar sample formats, each audio channel is in a separate data plane,
* and linesize is the buffer size, in bytes, for a single plane. All data
* planes must be the same size. For packed sample formats, only the first data
* plane is used, and samples for each channel are interleaved. In this case,
* linesize is the buffer size, in bytes, for the 1 plane.
*/
enum AVSampleFormat {
AV_SAMPLE_FMT_NONE = -1,
AV_SAMPLE_FMT_U8, ///< unsigned 8 bits
AV_SAMPLE_FMT_S16, ///< signed 16 bits
AV_SAMPLE_FMT_S32, ///< signed 32 bits
AV_SAMPLE_FMT_FLT, ///< float
AV_SAMPLE_FMT_DBL, ///< double
AV_SAMPLE_FMT_U8P, ///< unsigned 8 bits, planar
AV_SAMPLE_FMT_S16P, ///< signed 16 bits, planar
AV_SAMPLE_FMT_S32P, ///< signed 32 bits, planar
AV_SAMPLE_FMT_FLTP, ///< float, planar
AV_SAMPLE_FMT_DBLP, ///< double, planar
AV_SAMPLE_FMT_NB ///< Number of sample formats. DO NOT USE if linking dynamically
};
/**
* Return the name of sample_fmt, or NULL if sample_fmt is not
* recognized.
*/
const char *av_get_sample_fmt_name(enum AVSampleFormat sample_fmt);
/**
* Return a sample format corresponding to name, or AV_SAMPLE_FMT_NONE
* on error.
*/
enum AVSampleFormat av_get_sample_fmt(const char *name);
/**
* Get the packed alternative form of the given sample format.
*
* If the passed sample_fmt is already in packed format, the format returned is
* the same as the input.
*
* @return the packed alternative form of the given sample format or
AV_SAMPLE_FMT_NONE on error.
*/
enum AVSampleFormat av_get_packed_sample_fmt(enum AVSampleFormat sample_fmt);
/**
* Get the planar alternative form of the given sample format.
*
* If the passed sample_fmt is already in planar format, the format returned is
* the same as the input.
*
* @return the planar alternative form of the given sample format or
AV_SAMPLE_FMT_NONE on error.
*/
enum AVSampleFormat av_get_planar_sample_fmt(enum AVSampleFormat sample_fmt);
/**
* Generate a string corresponding to the sample format with
* sample_fmt, or a header if sample_fmt is negative.
*
* @param buf the buffer where to write the string
* @param buf_size the size of buf
* @param sample_fmt the number of the sample format to print the
* corresponding info string, or a negative value to print the
* corresponding header.
* @return the pointer to the filled buffer or NULL if sample_fmt is
* unknown or in case of other errors
*/
char *av_get_sample_fmt_string(char *buf, int buf_size, enum AVSampleFormat sample_fmt);
#if FF_API_GET_BITS_PER_SAMPLE_FMT
/**
* @deprecated Use av_get_bytes_per_sample() instead.
*/
attribute_deprecated
int av_get_bits_per_sample_fmt(enum AVSampleFormat sample_fmt);
#endif
/**
* Return number of bytes per sample.
*
* @param sample_fmt the sample format
* @return number of bytes per sample or zero if unknown for the given
* sample format
*/
int av_get_bytes_per_sample(enum AVSampleFormat sample_fmt);
/**
* Check if the sample format is planar.
*
* @param sample_fmt the sample format to inspect
* @return 1 if the sample format is planar, 0 if it is interleaved
*/
int av_sample_fmt_is_planar(enum AVSampleFormat sample_fmt);
/**
* Get the required buffer size for the given audio parameters.
*
* @param[out] linesize calculated linesize, may be NULL
* @param nb_channels the number of channels
* @param nb_samples the number of samples in a single channel
* @param sample_fmt the sample format
* @param align buffer size alignment (0 = default, 1 = no alignment)
* @return required buffer size, or negative error code on failure
*/
int av_samples_get_buffer_size(int *linesize, int nb_channels, int nb_samples,
enum AVSampleFormat sample_fmt, int align);
/**
* Fill channel data pointers and linesize for samples with sample
* format sample_fmt.
*
* The pointers array is filled with the pointers to the samples data:
* for planar, set the start point of each channel's data within the buffer,
* for packed, set the start point of the entire buffer only.
*
* The linesize array is filled with the aligned size of each channel's data
* buffer for planar layout, or the aligned size of the buffer for all channels
* for packed layout.
*
* @see enum AVSampleFormat
* The documentation for AVSampleFormat describes the data layout.
*
* @param[out] audio_data array to be filled with the pointer for each channel
* @param[out] linesize calculated linesize, may be NULL
* @param buf the pointer to a buffer containing the samples
* @param nb_channels the number of channels
* @param nb_samples the number of samples in a single channel
* @param sample_fmt the sample format
* @param align buffer size alignment (0 = default, 1 = no alignment)
* @return 0 on success or a negative error code on failure
*/
int av_samples_fill_arrays(uint8_t **audio_data, int *linesize,
const uint8_t *buf,
int nb_channels, int nb_samples,
enum AVSampleFormat sample_fmt, int align);
/**
* Allocate a samples buffer for nb_samples samples, and fill data pointers and
* linesize accordingly.
* The allocated samples buffer can be freed by using av_freep(&audio_data[0])
*
* @see enum AVSampleFormat
* The documentation for AVSampleFormat describes the data layout.
*
* @param[out] audio_data array to be filled with the pointer for each channel
* @param[out] linesize aligned size for audio buffer(s), may be NULL
* @param nb_channels number of audio channels
* @param nb_samples number of samples per channel
* @param align buffer size alignment (0 = default, 1 = no alignment)
* @return 0 on success or a negative error code on failure
* @see av_samples_fill_arrays()
*/
int av_samples_alloc(uint8_t **audio_data, int *linesize, int nb_channels,
int nb_samples, enum AVSampleFormat sample_fmt, int align);
/**
* Copy samples from src to dst.
*
* @param dst destination array of pointers to data planes
* @param src source array of pointers to data planes
* @param dst_offset offset in samples at which the data will be written to dst
* @param src_offset offset in samples at which the data will be read from src
* @param nb_samples number of samples to be copied
* @param nb_channels number of audio channels
* @param sample_fmt audio sample format
*/
int av_samples_copy(uint8_t **dst, uint8_t * const *src, int dst_offset,
int src_offset, int nb_samples, int nb_channels,
enum AVSampleFormat sample_fmt);
/**
* Fill an audio buffer with silence.
*
* @param audio_data array of pointers to data planes
* @param offset offset in samples at which to start filling
* @param nb_samples number of samples to fill
* @param nb_channels number of audio channels
* @param sample_fmt audio sample format
*/
int av_samples_set_silence(uint8_t **audio_data, int offset, int nb_samples,
int nb_channels, enum AVSampleFormat sample_fmt);
#endif /* AVUTIL_SAMPLEFMT_H */
|
