tor-browser

The Tor Browser
git clone https://git.dasho.dev/tor-browser.git
Log | Files | Refs | README | LICENSE

samplefmt.h (10211B)


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