(*
  * This file is part of ffm.
  *
  * 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
*)
unit ffm.samplefmt;

{$i ffmpeg.inc}

interface

(*
  * 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.
*)
Type
  pAVSampleFormat = ^TAVSampleFormat;
  TAVSampleFormat = ( //
    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);
function av_get_sample_fmt_name(sample_fmt: TAVSampleFormat): pAnsiChar; cdecl;

(*
  * 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);

{$IFDEF FF_API_GET_BITS_PER_SAMPLE_FMT}
(*
  * @deprecated Use av_get_bytes_per_sample() instead.
*)
// attribute_deprecated
// int av_get_bits_per_sample_fmt(enum AVSampleFormat sample_fmt);
{$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);
function av_get_bytes_per_sample(sample_fmt: TAVSampleFormat): integer; cdecl;

(*
  * 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);
function av_samples_get_buffer_size(linesize: pInteger; nb_channels: integer; nb_samples: integer;
  sample_fmt: TAVSampleFormat; align: integer): integer; cdecl;

(*
  * 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);

implementation

uses ffm.lib;

function av_get_bytes_per_sample; external samplefmt_dll;
function av_get_sample_fmt_name; external samplefmt_dll;
function av_samples_get_buffer_size; external samplefmt_dll;

end.