/* * This file is part of FFmpeg. * * FFmpeg 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. * * FFmpeg 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 FFmpeg; 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" /** * @addtogroup lavu_audio * @{ * * @defgroup lavu_sampfmts Audio sample formats * * Audio sample format enumeration and related convenience functions. * @{ */ /** * Audio sample formats * * - 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. * * - 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. * * - The data layout as used in av_samples_fill_arrays() and elsewhere in FFmpeg * (such as AVFrame in libavcodec) is as follows: * * @par * 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_S64, ///< signed 64 bits AV_SAMPLE_FMT_S64P, ///< signed 64 bits, 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); /** * Return the planar<->packed alternative form of the given sample format, or * AV_SAMPLE_FMT_NONE on error. If the passed sample_fmt is already in the * requested planar/packed format, the format returned is the same as the * input. */ enum AVSampleFormat av_get_alt_sample_fmt(enum AVSampleFormat sample_fmt, int planar); /** * 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); /** * 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); /** * @} * * @defgroup lavu_sampmanip Samples manipulation * * Functions that manipulate audio samples * @{ */ /** * Fill plane data pointers and linesize for samples with sample * format sample_fmt. * * The audio_data array is filled with the pointers to the samples data planes: * 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 value pointed to by linesize is set to the aligned size of each * channel's data buffer for planar layout, or to the aligned size of the * buffer for all channels for packed layout. * * The buffer in buf must be big enough to contain all the samples * (use av_samples_get_buffer_size() to compute its minimum size), * otherwise the audio_data pointers will point to invalid data. * * @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 minimum size in bytes required for the buffer 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]) * Allocated data will be initialized to silence. * * @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 * @todo return the size of the allocated buffer in case of success at the next bump * @see av_samples_fill_arrays() * @see av_samples_alloc_array_and_samples() */ int av_samples_alloc(uint8_t **audio_data, int *linesize, int nb_channels, int nb_samples, enum AVSampleFormat sample_fmt, int align); /** * Allocate a data pointers array, samples buffer for nb_samples * samples, and fill data pointers and linesize accordingly. * * This is the same as av_samples_alloc(), but also allocates the data * pointers array. * * @see av_samples_alloc() */ int av_samples_alloc_array_and_samples(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 */