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 #include "avutil.h"
25 #include "attributes.h"
26 
27 /**
28  * @addtogroup lavu_audio
29  * @{
30  *
31  * @defgroup lavu_sampfmts Audio sample formats
32  *
33  * Audio sample format enumeration and related convenience functions.
34  * @{
35  */
36 
37 /**
38  * Audio sample formats
39  *
40  * - The data described by the sample format is always in native-endian order.
41  *   Sample values can be expressed by native C types, hence the lack of a signed
42  *   24-bit sample format even though it is a common raw audio data format.
43  *
44  * - The floating-point formats are based on full volume being in the range
45  *   [-1.0, 1.0]. Any values outside this range are beyond full volume level.
46  *
47  * - The data layout as used in av_samples_fill_arrays() and elsewhere in FFmpeg
48  *   (such as AVFrame in libavcodec) is as follows:
49  *
50  * @par
51  * For planar sample formats, each audio channel is in a separate data plane,
52  * and linesize is the buffer size, in bytes, for a single plane. All data
53  * planes must be the same size. For packed sample formats, only the first data
54  * plane is used, and samples for each channel are interleaved. In this case,
55  * linesize is the buffer size, in bytes, for the 1 plane.
56  *
57  */
58 enum AVSampleFormat {
59     AV_SAMPLE_FMT_NONE = -1,
60     AV_SAMPLE_FMT_U8,          ///< unsigned 8 bits
61     AV_SAMPLE_FMT_S16,         ///< signed 16 bits
62     AV_SAMPLE_FMT_S32,         ///< signed 32 bits
63     AV_SAMPLE_FMT_FLT,         ///< float
64     AV_SAMPLE_FMT_DBL,         ///< double
65 
66     AV_SAMPLE_FMT_U8P,         ///< unsigned 8 bits, planar
67     AV_SAMPLE_FMT_S16P,        ///< signed 16 bits, planar
68     AV_SAMPLE_FMT_S32P,        ///< signed 32 bits, planar
69     AV_SAMPLE_FMT_FLTP,        ///< float, planar
70     AV_SAMPLE_FMT_DBLP,        ///< double, planar
71     AV_SAMPLE_FMT_S64,         ///< signed 64 bits
72     AV_SAMPLE_FMT_S64P,        ///< signed 64 bits, planar
73 
74     AV_SAMPLE_FMT_NB           ///< Number of sample formats. DO NOT USE if linking dynamically
75 };
76 
77 /**
78  * Return the name of sample_fmt, or NULL if sample_fmt is not
79  * recognized.
80  */
81 const char *av_get_sample_fmt_name(enum AVSampleFormat sample_fmt);
82 
83 /**
84  * Return a sample format corresponding to name, or AV_SAMPLE_FMT_NONE
85  * on error.
86  */
87 enum AVSampleFormat av_get_sample_fmt(const char *name);
88 
89 /**
90  * Return the planar<->packed alternative form of the given sample format, or
91  * AV_SAMPLE_FMT_NONE on error. If the passed sample_fmt is already in the
92  * requested planar/packed format, the format returned is the same as the
93  * input.
94  */
95 enum AVSampleFormat av_get_alt_sample_fmt(enum AVSampleFormat sample_fmt, int planar);
96 
97 /**
98  * Get the packed alternative form of the given sample format.
99  *
100  * If the passed sample_fmt is already in packed format, the format returned is
101  * the same as the input.
102  *
103  * @return  the packed alternative form of the given sample format or
104             AV_SAMPLE_FMT_NONE on error.
105  */
106 enum AVSampleFormat av_get_packed_sample_fmt(enum AVSampleFormat sample_fmt);
107 
108 /**
109  * Get the planar alternative form of the given sample format.
110  *
111  * If the passed sample_fmt is already in planar format, the format returned is
112  * the same as the input.
113  *
114  * @return  the planar alternative form of the given sample format or
115             AV_SAMPLE_FMT_NONE on error.
116  */
117 enum AVSampleFormat av_get_planar_sample_fmt(enum AVSampleFormat sample_fmt);
118 
119 /**
120  * Generate a string corresponding to the sample format with
121  * sample_fmt, or a header if sample_fmt is negative.
122  *
123  * @param buf the buffer where to write the string
124  * @param buf_size the size of buf
125  * @param sample_fmt the number of the sample format to print the
126  * corresponding info string, or a negative value to print the
127  * corresponding header.
128  * @return the pointer to the filled buffer or NULL if sample_fmt is
129  * unknown or in case of other errors
130  */
131 char *av_get_sample_fmt_string(char *buf, int buf_size, enum AVSampleFormat sample_fmt);
132 
133 /**
134  * Return number of bytes per sample.
135  *
136  * @param sample_fmt the sample format
137  * @return number of bytes per sample or zero if unknown for the given
138  * sample format
139  */
140 int av_get_bytes_per_sample(enum AVSampleFormat sample_fmt);
141 
142 /**
143  * Check if the sample format is planar.
144  *
145  * @param sample_fmt the sample format to inspect
146  * @return 1 if the sample format is planar, 0 if it is interleaved
147  */
148 int av_sample_fmt_is_planar(enum AVSampleFormat sample_fmt);
149 
150 /**
151  * Get the required buffer size for the given audio parameters.
152  *
153  * @param[out] linesize calculated linesize, may be NULL
154  * @param nb_channels   the number of channels
155  * @param nb_samples    the number of samples in a single channel
156  * @param sample_fmt    the sample format
157  * @param align         buffer size alignment (0 = default, 1 = no alignment)
158  * @return              required buffer size, or negative error code on failure
159  */
160 int av_samples_get_buffer_size(int *linesize, int nb_channels, int nb_samples,
161                                enum AVSampleFormat sample_fmt, int align);
162 
163 /**
164  * @}
165  *
166  * @defgroup lavu_sampmanip Samples manipulation
167  *
168  * Functions that manipulate audio samples
169  * @{
170  */
171 
172 /**
173  * Fill plane data pointers and linesize for samples with sample
174  * format sample_fmt.
175  *
176  * The audio_data array is filled with the pointers to the samples data planes:
177  * for planar, set the start point of each channel's data within the buffer,
178  * for packed, set the start point of the entire buffer only.
179  *
180  * The value pointed to by linesize is set to the aligned size of each
181  * channel's data buffer for planar layout, or to the aligned size of the
182  * buffer for all channels for packed layout.
183  *
184  * The buffer in buf must be big enough to contain all the samples
185  * (use av_samples_get_buffer_size() to compute its minimum size),
186  * otherwise the audio_data pointers will point to invalid data.
187  *
188  * @see enum AVSampleFormat
189  * The documentation for AVSampleFormat describes the data layout.
190  *
191  * @param[out] audio_data  array to be filled with the pointer for each channel
192  * @param[out] linesize    calculated linesize, may be NULL
193  * @param buf              the pointer to a buffer containing the samples
194  * @param nb_channels      the number of channels
195  * @param nb_samples       the number of samples in a single channel
196  * @param sample_fmt       the sample format
197  * @param align            buffer size alignment (0 = default, 1 = no alignment)
198  * @return                 >=0 on success or a negative error code on failure
199  * @todo return minimum size in bytes required for the buffer in case
200  * of success at the next bump
201  */
202 int av_samples_fill_arrays(uint8_t **audio_data, int *linesize,
203                            const uint8_t *buf,
204                            int nb_channels, int nb_samples,
205                            enum AVSampleFormat sample_fmt, int align);
206 
207 /**
208  * Allocate a samples buffer for nb_samples samples, and fill data pointers and
209  * linesize accordingly.
210  * The allocated samples buffer can be freed by using av_freep(&audio_data[0])
211  * Allocated data will be initialized to silence.
212  *
213  * @see enum AVSampleFormat
214  * The documentation for AVSampleFormat describes the data layout.
215  *
216  * @param[out] audio_data  array to be filled with the pointer for each channel
217  * @param[out] linesize    aligned size for audio buffer(s), may be NULL
218  * @param nb_channels      number of audio channels
219  * @param nb_samples       number of samples per channel
220  * @param align            buffer size alignment (0 = default, 1 = no alignment)
221  * @return                 >=0 on success or a negative error code on failure
222  * @todo return the size of the allocated buffer in case of success at the next bump
223  * @see av_samples_fill_arrays()
224  * @see av_samples_alloc_array_and_samples()
225  */
226 int av_samples_alloc(uint8_t **audio_data, int *linesize, int nb_channels,
227                      int nb_samples, enum AVSampleFormat sample_fmt, int align);
228 
229 /**
230  * Allocate a data pointers array, samples buffer for nb_samples
231  * samples, and fill data pointers and linesize accordingly.
232  *
233  * This is the same as av_samples_alloc(), but also allocates the data
234  * pointers array.
235  *
236  * @see av_samples_alloc()
237  */
238 int av_samples_alloc_array_and_samples(uint8_t ***audio_data, int *linesize, int nb_channels,
239                                        int nb_samples, enum AVSampleFormat sample_fmt, int align);
240 
241 /**
242  * Copy samples from src to dst.
243  *
244  * @param dst destination array of pointers to data planes
245  * @param src source array of pointers to data planes
246  * @param dst_offset offset in samples at which the data will be written to dst
247  * @param src_offset offset in samples at which the data will be read from src
248  * @param nb_samples number of samples to be copied
249  * @param nb_channels number of audio channels
250  * @param sample_fmt audio sample format
251  */
252 int av_samples_copy(uint8_t **dst, uint8_t * const *src, int dst_offset,
253                     int src_offset, int nb_samples, int nb_channels,
254                     enum AVSampleFormat sample_fmt);
255 
256 /**
257  * Fill an audio buffer with silence.
258  *
259  * @param audio_data  array of pointers to data planes
260  * @param offset      offset in samples at which to start filling
261  * @param nb_samples  number of samples to fill
262  * @param nb_channels number of audio channels
263  * @param sample_fmt  audio sample format
264  */
265 int av_samples_set_silence(uint8_t **audio_data, int offset, int nb_samples,
266                            int nb_channels, enum AVSampleFormat sample_fmt);
267 
268 /**
269  * @}
270  * @}
271  */
272 #endif /* AVUTIL_SAMPLEFMT_H */
273