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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 

#include "avutil.h"

#include "attributes.h"

/**

 * Audio Sample Formats

 *

 * @par

 * 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.

 *

 * @par

 * 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.

 *

 * @par

 * The data layout as used in av_samples_fill_arrays() and elsewhere in FFmpeg

 * (such as AVFrame in libavcodec) is as follows:

 *

 * 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_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);

#if FF_API_GET_BITS_PER_SAMPLE_FMT

/**

 * @deprecated Use av_get_bytes_per_sample() instead.

 */

attribute_deprecated

int av_get_bits_per_sample_fmt(enum AVSampleFormat sample_fmt);

#endif

/**

 * 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);

/**

 * 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 */