/*
  * 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 "avutil.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 Libav
  * (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 channel data pointers and linesize for samples with sample

  * format sample_fmt.
  *
  * The pointers array is filled with the pointers to the samples data:

  * 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 linesize array is filled with the aligned size of each channel's data
  * buffer for planar layout, or the aligned size of the buffer for all channels
  * for packed layout.

  *

  * @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

  */

 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])

  *

  * @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

  * @see av_samples_fill_arrays()
  */

 int av_samples_alloc(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 */