1/*
2 * This file is part of FFmpeg.
3 *
4 * FFmpeg is free software; you can redistribute it and/or
5 * modify it under the terms of the GNU Lesser General Public
6 * License as published by the Free Software Foundation; either
7 * version 2.1 of the License, or (at your option) any later version.
8 *
9 * FFmpeg is distributed in the hope that it will be useful,
10 * but WITHOUT ANY WARRANTY; without even the implied warranty of
11 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
12 * Lesser General Public License for more details.
13 *
14 * You should have received a copy of the GNU Lesser General Public
15 * License along with FFmpeg; if not, write to the Free Software
16 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
17 */
18
19#ifndef AVUTIL_SAMPLEFMT_H
20#define AVUTIL_SAMPLEFMT_H
21
22#include <stdint.h>
23
24/**
25 * @addtogroup lavu_audio
26 * @{
27 *
28 * @defgroup lavu_sampfmts Audio sample formats
29 *
30 * Audio sample format enumeration and related convenience functions.
31 * @{
32 */
33
34/**
35 * Audio sample formats
36 *
37 * - The data described by the sample format is always in native-endian order.
38 * Sample values can be expressed by native C types, hence the lack of a signed
39 * 24-bit sample format even though it is a common raw audio data format.
40 *
41 * - The floating-point formats are based on full volume being in the range
42 * [-1.0, 1.0]. Any values outside this range are beyond full volume level.
43 *
44 * - The data layout as used in av_samples_fill_arrays() and elsewhere in FFmpeg
45 * (such as AVFrame in libavcodec) is as follows:
46 *
47 * @par
48 * For planar sample formats, each audio channel is in a separate data plane,
49 * and linesize is the buffer size, in bytes, for a single plane. All data
50 * planes must be the same size. For packed sample formats, only the first data
51 * plane is used, and samples for each channel are interleaved. In this case,
52 * linesize is the buffer size, in bytes, for the 1 plane.
53 *
54 */
55enum AVSampleFormat {
56 AV_SAMPLE_FMT_NONE = -1,
57 AV_SAMPLE_FMT_U8, ///< unsigned 8 bits
58 AV_SAMPLE_FMT_S16, ///< signed 16 bits
59 AV_SAMPLE_FMT_S32, ///< signed 32 bits
60 AV_SAMPLE_FMT_FLT, ///< float
61 AV_SAMPLE_FMT_DBL, ///< double
62
63 AV_SAMPLE_FMT_U8P, ///< unsigned 8 bits, planar
64 AV_SAMPLE_FMT_S16P, ///< signed 16 bits, planar
65 AV_SAMPLE_FMT_S32P, ///< signed 32 bits, planar
66 AV_SAMPLE_FMT_FLTP, ///< float, planar
67 AV_SAMPLE_FMT_DBLP, ///< double, planar
68 AV_SAMPLE_FMT_S64, ///< signed 64 bits
69 AV_SAMPLE_FMT_S64P, ///< signed 64 bits, planar
70
71 AV_SAMPLE_FMT_NB ///< Number of sample formats. DO NOT USE if linking dynamically
72};
73
74/**
75 * Return the name of sample_fmt, or NULL if sample_fmt is not
76 * recognized.
77 */
78const char *av_get_sample_fmt_name(enum AVSampleFormat sample_fmt);
79
80/**
81 * Return a sample format corresponding to name, or AV_SAMPLE_FMT_NONE
82 * on error.
83 */
84enum AVSampleFormat av_get_sample_fmt(const char *name);
85
86/**
87 * Return the planar<->packed alternative form of the given sample format, or
88 * AV_SAMPLE_FMT_NONE on error. If the passed sample_fmt is already in the
89 * requested planar/packed format, the format returned is the same as the
90 * input.
91 */
92enum AVSampleFormat av_get_alt_sample_fmt(enum AVSampleFormat sample_fmt, int planar);
93
94/**
95 * Get the packed alternative form of the given sample format.
96 *
97 * If the passed sample_fmt is already in packed format, the format returned is
98 * the same as the input.
99 *
100 * @return the packed alternative form of the given sample format or
101 AV_SAMPLE_FMT_NONE on error.
102 */
103enum AVSampleFormat av_get_packed_sample_fmt(enum AVSampleFormat sample_fmt);
104
105/**
106 * Get the planar alternative form of the given sample format.
107 *
108 * If the passed sample_fmt is already in planar format, the format returned is
109 * the same as the input.
110 *
111 * @return the planar alternative form of the given sample format or
112 AV_SAMPLE_FMT_NONE on error.
113 */
114enum AVSampleFormat av_get_planar_sample_fmt(enum AVSampleFormat sample_fmt);
115
116/**
117 * Generate a string corresponding to the sample format with
118 * sample_fmt, or a header if sample_fmt is negative.
119 *
120 * @param buf the buffer where to write the string
121 * @param buf_size the size of buf
122 * @param sample_fmt the number of the sample format to print the
123 * corresponding info string, or a negative value to print the
124 * corresponding header.
125 * @return the pointer to the filled buffer or NULL if sample_fmt is
126 * unknown or in case of other errors
127 */
128char *av_get_sample_fmt_string(char *buf, int buf_size, enum AVSampleFormat sample_fmt);
129
130/**
131 * Return number of bytes per sample.
132 *
133 * @param sample_fmt the sample format
134 * @return number of bytes per sample or zero if unknown for the given
135 * sample format
136 */
137int av_get_bytes_per_sample(enum AVSampleFormat sample_fmt);
138
139/**
140 * Check if the sample format is planar.
141 *
142 * @param sample_fmt the sample format to inspect
143 * @return 1 if the sample format is planar, 0 if it is interleaved
144 */
145int av_sample_fmt_is_planar(enum AVSampleFormat sample_fmt);
146
147/**
148 * Get the required buffer size for the given audio parameters.
149 *
150 * @param[out] linesize calculated linesize, may be NULL
151 * @param nb_channels the number of channels
152 * @param nb_samples the number of samples in a single channel
153 * @param sample_fmt the sample format
154 * @param align buffer size alignment (0 = default, 1 = no alignment)
155 * @return required buffer size, or negative error code on failure
156 */
157int av_samples_get_buffer_size(int *linesize, int nb_channels, int nb_samples,
158 enum AVSampleFormat sample_fmt, int align);
159
160/**
161 * @}
162 *
163 * @defgroup lavu_sampmanip Samples manipulation
164 *
165 * Functions that manipulate audio samples
166 * @{
167 */
168
169/**
170 * Fill plane data pointers and linesize for samples with sample
171 * format sample_fmt.
172 *
173 * The audio_data array is filled with the pointers to the samples data planes:
174 * for planar, set the start point of each channel's data within the buffer,
175 * for packed, set the start point of the entire buffer only.
176 *
177 * The value pointed to by linesize is set to the aligned size of each
178 * channel's data buffer for planar layout, or to the aligned size of the
179 * buffer for all channels for packed layout.
180 *
181 * The buffer in buf must be big enough to contain all the samples
182 * (use av_samples_get_buffer_size() to compute its minimum size),
183 * otherwise the audio_data pointers will point to invalid data.
184 *
185 * @see enum AVSampleFormat
186 * The documentation for AVSampleFormat describes the data layout.
187 *
188 * @param[out] audio_data array to be filled with the pointer for each channel
189 * @param[out] linesize calculated linesize, may be NULL
190 * @param buf the pointer to a buffer containing the samples
191 * @param nb_channels the number of channels
192 * @param nb_samples the number of samples in a single channel
193 * @param sample_fmt the sample format
194 * @param align buffer size alignment (0 = default, 1 = no alignment)
195 * @return minimum size in bytes required for the buffer on success,
196 * or a negative error code on failure
197 */
198int av_samples_fill_arrays(uint8_t **audio_data, int *linesize,
199 const uint8_t *buf,
200 int nb_channels, int nb_samples,
201 enum AVSampleFormat sample_fmt, int align);
202
203/**
204 * Allocate a samples buffer for nb_samples samples, and fill data pointers and
205 * linesize accordingly.
206 * The allocated samples buffer can be freed by using av_freep(&audio_data[0])
207 * Allocated data will be initialized to silence.
208 *
209 * @see enum AVSampleFormat
210 * The documentation for AVSampleFormat describes the data layout.
211 *
212 * @param[out] audio_data array to be filled with the pointer for each channel
213 * @param[out] linesize aligned size for audio buffer(s), may be NULL
214 * @param nb_channels number of audio channels
215 * @param nb_samples number of samples per channel
216 * @param sample_fmt the sample format
217 * @param align buffer size alignment (0 = default, 1 = no alignment)
218 * @return >=0 on success or a negative error code on failure
219 * @todo return the size of the allocated buffer in case of success at the next bump
220 * @see av_samples_fill_arrays()
221 * @see av_samples_alloc_array_and_samples()
222 */
223int av_samples_alloc(uint8_t **audio_data, int *linesize, int nb_channels,
224 int nb_samples, enum AVSampleFormat sample_fmt, int align);
225
226/**
227 * Allocate a data pointers array, samples buffer for nb_samples
228 * samples, and fill data pointers and linesize accordingly.
229 *
230 * This is the same as av_samples_alloc(), but also allocates the data
231 * pointers array.
232 *
233 * @see av_samples_alloc()
234 */
235int av_samples_alloc_array_and_samples(uint8_t ***audio_data, int *linesize, int nb_channels,
236 int nb_samples, enum AVSampleFormat sample_fmt, int align);
237
238/**
239 * Copy samples from src to dst.
240 *
241 * @param dst destination array of pointers to data planes
242 * @param src source array of pointers to data planes
243 * @param dst_offset offset in samples at which the data will be written to dst
244 * @param src_offset offset in samples at which the data will be read from src
245 * @param nb_samples number of samples to be copied
246 * @param nb_channels number of audio channels
247 * @param sample_fmt audio sample format
248 */
249int av_samples_copy(uint8_t * const *dst, uint8_t * const *src, int dst_offset,
250 int src_offset, int nb_samples, int nb_channels,
251 enum AVSampleFormat sample_fmt);
252
253/**
254 * Fill an audio buffer with silence.
255 *
256 * @param audio_data array of pointers to data planes
257 * @param offset offset in samples at which to start filling
258 * @param nb_samples number of samples to fill
259 * @param nb_channels number of audio channels
260 * @param sample_fmt audio sample format
261 */
262int av_samples_set_silence(uint8_t * const *audio_data, int offset, int nb_samples,
263 int nb_channels, enum AVSampleFormat sample_fmt);
264
265/**
266 * @}
267 * @}
268 */
269#endif /* AVUTIL_SAMPLEFMT_H */
270