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 module ffmpeg.libavutil.samplefmt; 20 21 import ffmpeg.libavutil.avutil_version; 22 import std.stdint; 23 24 25 @nogc nothrow extern(C): 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 72 AV_SAMPLE_FMT_NB ///< Number of sample formats. DO NOT USE if linking dynamically 73 } 74 75 /** 76 * Return the name of sample_fmt, or NULL if sample_fmt is not 77 * recognized. 78 */ 79 char* av_get_sample_fmt_name(AVSampleFormat sample_fmt); 80 81 /** 82 * Return a sample format corresponding to name, or AV_SAMPLE_FMT_NONE 83 * on error. 84 */ 85 AVSampleFormat av_get_sample_fmt(const char* name); 86 87 /** 88 * Return the planar<->packed alternative form of the given sample format, or 89 * AV_SAMPLE_FMT_NONE on error. If the passed sample_fmt is already in the 90 * requested planar/packed format, the format returned is the same as the 91 * input. 92 */ 93 AVSampleFormat av_get_alt_sample_fmt(AVSampleFormat sample_fmt, int planar); 94 95 /** 96 * Get the packed alternative form of the given sample format. 97 * 98 * If the passed sample_fmt is already in packed format, the format returned is 99 * the same as the input. 100 * 101 * @return the packed alternative form of the given sample format or 102 AV_SAMPLE_FMT_NONE on error. 103 */ 104 AVSampleFormat av_get_packed_sample_fmt(AVSampleFormat sample_fmt); 105 106 /** 107 * Get the planar alternative form of the given sample format. 108 * 109 * If the passed sample_fmt is already in planar format, the format returned is 110 * the same as the input. 111 * 112 * @return the planar alternative form of the given sample format or 113 AV_SAMPLE_FMT_NONE on error. 114 */ 115 AVSampleFormat av_get_planar_sample_fmt(AVSampleFormat sample_fmt); 116 117 /** 118 * Generate a string corresponding to the sample format with 119 * sample_fmt, or a header if sample_fmt is negative. 120 * 121 * @param buf the buffer where to write the string 122 * @param buf_size the size of buf 123 * @param sample_fmt the number of the sample format to print the 124 * corresponding info string, or a negative value to print the 125 * corresponding header. 126 * @return the pointer to the filled buffer or NULL if sample_fmt is 127 * unknown or in case of other errors 128 */ 129 char *av_get_sample_fmt_string(char *buf, int buf_size, AVSampleFormat sample_fmt); 130 131 /** 132 * Return number of bytes per sample. 133 * 134 * @param sample_fmt the sample format 135 * @return number of bytes per sample or zero if unknown for the given 136 * sample format 137 */ 138 int av_get_bytes_per_sample(AVSampleFormat sample_fmt); 139 140 /** 141 * Check if the sample format is planar. 142 * 143 * @param sample_fmt the sample format to inspect 144 * @return 1 if the sample format is planar, 0 if it is interleaved 145 */ 146 int av_sample_fmt_is_planar(AVSampleFormat sample_fmt); 147 148 /** 149 * Get the required buffer size for the given audio parameters. 150 * 151 * @param[out] linesize calculated linesize, may be NULL 152 * @param nb_channels the number of channels 153 * @param nb_samples the number of samples in a single channel 154 * @param sample_fmt the sample format 155 * @param align buffer size alignment (0 = default, 1 = no alignment) 156 * @return required buffer size, or negative error code on failure 157 */ 158 int av_samples_get_buffer_size(int *linesize, int nb_channels, int nb_samples, 159 AVSampleFormat sample_fmt, int alignment); 160 161 /** 162 * @} 163 * 164 * @defgroup lavu_sampmanip Samples manipulation 165 * 166 * Functions that manipulate audio samples 167 * @{ 168 */ 169 170 /** 171 * Fill plane data pointers and linesize for samples with sample 172 * format sample_fmt. 173 * 174 * The audio_data array is filled with the pointers to the samples data planes: 175 * for planar, set the start point of each channel's data within the buffer, 176 * for packed, set the start point of the entire buffer only. 177 * 178 * The value pointed to by linesize is set to the aligned size of each 179 * channel's data buffer for planar layout, or to the aligned size of the 180 * buffer for all channels for packed layout. 181 * 182 * The buffer in buf must be big enough to contain all the samples 183 * (use av_samples_get_buffer_size() to compute its minimum size), 184 * otherwise the audio_data pointers will point to invalid data. 185 * 186 * @see enum AVSampleFormat 187 * The documentation for AVSampleFormat describes the data layout. 188 * 189 * @param[out] audio_data array to be filled with the pointer for each channel 190 * @param[out] linesize calculated linesize, may be NULL 191 * @param buf the pointer to a buffer containing the samples 192 * @param nb_channels the number of channels 193 * @param nb_samples the number of samples in a single channel 194 * @param sample_fmt the sample format 195 * @param align buffer size alignment (0 = default, 1 = no alignment) 196 * @return >=0 on success or a negative error code on failure 197 * @todo return minimum size in bytes required for the buffer in case 198 * of success at the next bump 199 */ 200 int av_samples_fill_arrays(uint **audio_data, int *linesize, 201 const uint* buf, 202 int nb_channels, int nb_samples, 203 AVSampleFormat sample_fmt, int alignment); 204 205 /** 206 * Allocate a samples buffer for nb_samples samples, and fill data pointers and 207 * linesize accordingly. 208 * The allocated samples buffer can be freed by using av_freep(&audio_data[0]) 209 * Allocated data will be initialized to silence. 210 * 211 * @see enum AVSampleFormat 212 * The documentation for AVSampleFormat describes the data layout. 213 * 214 * @param[out] audio_data array to be filled with the pointer for each channel 215 * @param[out] linesize aligned size for audio buffer(s), may be NULL 216 * @param nb_channels number of audio channels 217 * @param nb_samples number of samples per channel 218 * @param align buffer size alignment (0 = default, 1 = no alignment) 219 * @return >=0 on success or a negative error code on failure 220 * @todo return the size of the allocated buffer in case of success at the next bump 221 * @see av_samples_fill_arrays() 222 * @see av_samples_alloc_array_and_samples() 223 */ 224 int av_samples_alloc(ubyte **audio_data, int *linesize, int nb_channels, 225 int nb_samples, AVSampleFormat sample_fmt, int alignment); 226 227 /** 228 * Allocate a data pointers array, samples buffer for nb_samples 229 * samples, and fill data pointers and linesize accordingly. 230 * 231 * This is the same as av_samples_alloc(), but also allocates the data 232 * pointers array. 233 * 234 * @see av_samples_alloc() 235 */ 236 int av_samples_alloc_array_and_samples(uint8_t ***audio_data, int *linesize, int nb_channels, 237 int nb_samples, AVSampleFormat sample_fmt, int alignment); 238 239 /** 240 * Copy samples from src to dst. 241 * 242 * @param dst destination array of pointers to data planes 243 * @param src source array of pointers to data planes 244 * @param dst_offset offset in samples at which the data will be written to dst 245 * @param src_offset offset in samples at which the data will be read from src 246 * @param nb_samples number of samples to be copied 247 * @param nb_channels number of audio channels 248 * @param sample_fmt audio sample format 249 */ 250 int av_samples_copy(ubyte **dst, const uint **src, int dst_offset, 251 int src_offset, int nb_samples, int nb_channels, 252 AVSampleFormat sample_fmt); 253 254 /** 255 * Fill an audio buffer with silence. 256 * 257 * @param audio_data array of pointers to data planes 258 * @param offset offset in samples at which to start filling 259 * @param nb_samples number of samples to fill 260 * @param nb_channels number of audio channels 261 * @param sample_fmt audio sample format 262 */ 263 int av_samples_set_silence(ubyte **audio_data, int offset, int nb_samples, 264 int nb_channels, AVSampleFormat sample_fmt); 265 266 /** 267 * @} 268 * @} 269 */ 270 //#endif /* AVUTIL_SAMPLEFMT_H */