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/*
* 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 >=0 on success or a negative error code on failure
* @todo return minimum size in bytes required for the buffer in case
* of success at the next bump
*/
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 */
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