(*
 * Copyright (C) 2011-2013 Michael Niedermayer (michaelni@gmx.at)
 *
 * This file is part of libswresample
 *
 * libswresample 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.
 *
 * libswresample 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 libswresample; if not, write to the Free Software
 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
 *)
unit swresample;

{$include ffmpeg.inc}

interface
(*
 * @defgroup lswr Libswresample
 * @{
 *
 * Libswresample (lswr) is a library that handles audio resampling, sample
 * format conversion and mixing.
 *
 * Interaction with lswr is done through SwrContext, which is
 * allocated with swr_alloc() or swr_alloc_set_opts(). It is opaque, so all parameters
 * must be set with the @ref avoptions API.
 *
 * For example the following code will setup conversion from planar float sample
 * format to interleaved signed 16-bit integer, downsampling from 48kHz to
 * 44.1kHz and downmixing from 5.1 channels to stereo (using the default mixing
 * matrix):
 * @code
 * SwrContext *swr = swr_alloc();
 * av_opt_set_channel_layout(swr, "in_channel_layout",  AV_CH_LAYOUT_5POINT1, 0);
 * av_opt_set_channel_layout(swr, "out_channel_layout", AV_CH_LAYOUT_STEREO,  0);
 * av_opt_set_int(swr, "in_sample_rate",     48000,                0);
 * av_opt_set_int(swr, "out_sample_rate",    44100,                0);
 * av_opt_set_sample_fmt(swr, "in_sample_fmt",  AV_SAMPLE_FMT_FLTP, 0);
 * av_opt_set_sample_fmt(swr, "out_sample_fmt", AV_SAMPLE_FMT_S16,  0);
 * @endcode
 *
 * Once all values have been set, it must be initialized with swr_init(). If
 * you need to change the conversion parameters, you can change the parameters
 * as described above, or by using swr_alloc_set_opts(), then call swr_init()
 * again.
 *
 * The conversion itself is done by repeatedly calling swr_convert().
 * Note that the samples may get buffered in swr if you provide insufficient
 * output space or if sample rate conversion is done, which requires "future"
 * samples. Samples that do not require future input can be retrieved at any
 * time by using swr_convert() (in_count can be set to 0).
 * At the end of conversion the resampling buffer can be flushed by calling
 * swr_convert() with NULL in and 0 in_count.
 *
 * The delay between input and output, can at any time be found by using
 * swr_get_delay().
 *
 * The following code demonstrates the conversion loop assuming the parameters
 * from above and caller-defined functions get_input() and handle_output():
 * @code
 * uint8_t **input;
 * int in_samples;
 *
 * while (get_input(&input, &in_samples)) {
 *     uint8_t *output;
 *     int out_samples = av_rescale_rnd(swr_get_delay(swr, 48000) +
 *                                      in_samples, 44100, 48000, AV_ROUND_UP);
 *     av_samples_alloc(&output, NULL, 2, out_samples,
 *                      AV_SAMPLE_FMT_S16, 0);
 *     out_samples = swr_convert(swr, &output, out_samples,
 *                                      input, in_samples);
 *     handle_output(output, out_samples);
 *     av_freep(&output);
 * }
 * @endcode
 *
 * When the conversion is finished, the conversion
 * context and everything associated with it must be freed with swr_free().
 * There will be no memory leak if the data is not completely flushed before
 * swr_free().
 *)

//#if LIBSWRESAMPLE_VERSION_MAJOR < 1
//#define SWR_CH_MAX 32   ///< Maximum number of channels
//#endif

//#define SWR_FLAG_RESAMPLE 1 ///< Force resampling even if equal sample rate
//TODO use int resample ?
//long term TODO can we enable this dynamically?

//enum SwrDitherType {
//    SWR_DITHER_NONE = 0,
//    SWR_DITHER_RECTANGULAR,
//    SWR_DITHER_TRIANGULAR,
//    SWR_DITHER_TRIANGULAR_HIGHPASS,
//
//    SWR_DITHER_NS = 64,         ///< not part of API/ABI
//    SWR_DITHER_NS_LIPSHITZ,
//    SWR_DITHER_NS_F_WEIGHTED,
//    SWR_DITHER_NS_MODIFIED_E_WEIGHTED,
//    SWR_DITHER_NS_IMPROVED_E_WEIGHTED,
//    SWR_DITHER_NS_SHIBATA,
//    SWR_DITHER_NS_LOW_SHIBATA,
//    SWR_DITHER_NS_HIGH_SHIBATA,
//    SWR_DITHER_NB,              ///< not part of API/ABI
//};
//
//(* Resampling Engines *)
//enum SwrEngine {
//    SWR_ENGINE_SWR,             (*< SW Resampler *)
//    SWR_ENGINE_SOXR,            (*< SoX Resampler *)
//    SWR_ENGINE_NB,              ///< not part of API/ABI
//};
//
//(* Resampling Filter Types *)
//enum SwrFilterType {
//    SWR_FILTER_TYPE_CUBIC,              (*< Cubic *)
//    SWR_FILTER_TYPE_BLACKMAN_NUTTALL,   (*< Blackman Nuttall Windowed Sinc *)
//    SWR_FILTER_TYPE_KAISER,             (*< Kaiser Windowed Sinc *)
//};
Type
//typedef struct SwrContext SwrContext;
pSwrContext = ^TSwrContext;
TSwrContext = packed record

end;

(*
 * Get the AVClass for swrContext. It can be used in combination with
 * AV_OPT_SEARCH_FAKE_OBJ for examining options.
 *
 * @see av_opt_find().
 *)
//const AVClass *swr_get_class(void);

(*
 * Allocate SwrContext.
 *
 * If you use this function you will need to set the parameters (manually or
 * with swr_alloc_set_opts()) before calling swr_init().
 *
 * @see swr_alloc_set_opts(), swr_init(), swr_free()
 * @return NULL on error, allocated context otherwise
 *)
//struct SwrContext *swr_alloc(void);

(*
 * Initialize context after user parameters have been set.
 *
 * @return AVERROR error code in case of failure.
 *)
//int swr_init(struct SwrContext *s);

(*
 * Allocate SwrContext if needed and set/reset common parameters.
 *
 * This function does not require s to be allocated with swr_alloc(). On the
 * other hand, swr_alloc() can use swr_alloc_set_opts() to set the parameters
 * on the allocated context.
 *
 * @param s               Swr context, can be NULL
 * @param out_ch_layout   output channel layout (AV_CH_LAYOUT_* )
 * @param out_sample_fmt  output sample format (AV_SAMPLE_FMT_* ).
 * @param out_sample_rate output sample rate (frequency in Hz)
 * @param in_ch_layout    input channel layout (AV_CH_LAYOUT_* )
 * @param in_sample_fmt   input sample format (AV_SAMPLE_FMT_* ).
 * @param in_sample_rate  input sample rate (frequency in Hz)
 * @param log_offset      logging level offset
 * @param log_ctx         parent logging context, can be NULL
 *
 * @see swr_init(), swr_free()
 * @return NULL on error, allocated context otherwise
 *)
//struct SwrContext *swr_alloc_set_opts(struct SwrContext *s,
//                                      int64_t out_ch_layout, enum AVSampleFormat out_sample_fmt, int out_sample_rate,
//                                      int64_t  in_ch_layout, enum AVSampleFormat  in_sample_fmt, int  in_sample_rate,
//                                      int log_offset, void *log_ctx);

(*
 * Free the given SwrContext and set the pointer to NULL.
 *)
//void swr_free(struct SwrContext **s);

(*
 * Convert audio.
 *
 * in and in_count can be set to 0 to flush the last few samples out at the
 * end.
 *
 * If more input is provided than output space then the input will be buffered.
 * You can avoid this buffering by providing more output space than input.
 * Convertion will run directly without copying whenever possible.
 *
 * @param s         allocated Swr context, with parameters set
 * @param out       output buffers, only the first one need be set in case of packed audio
 * @param out_count amount of space available for output in samples per channel
 * @param in        input buffers, only the first one need to be set in case of packed audio
 * @param in_count  number of input samples available in one channel
 *
 * @return number of samples output per channel, negative value on error
 *)
//int swr_convert(struct SwrContext *s, uint8_t **out, int out_count,
//                                const uint8_t **in , int in_count);
function swr_convert(s:pSwrContext; Var out_:PByte; out_count:Integer;
                                const in_:pByte;in_count:Integer):Integer;cdecl;

(*
 * Convert the next timestamp from input to output
 * timestamps are in 1/(in_sample_rate * out_sample_rate) units.
 *
 * @note There are 2 slightly differently behaving modes.
 *       First is when automatic timestamp compensation is not used, (min_compensation >= FLT_MAX)
 *              in this case timestamps will be passed through with delays compensated
 *       Second is when automatic timestamp compensation is used, (min_compensation < FLT_MAX)
 *              in this case the output timestamps will match output sample numbers
 *
 * @param pts   timestamp for the next input sample, INT64_MIN if unknown
 * @return the output timestamp for the next output sample
 *)
//int64_t swr_next_pts(struct SwrContext *s, int64_t pts);

(*
 * Activate resampling compensation.
 *)
//int swr_set_compensation(struct SwrContext *s, int sample_delta, int compensation_distance);

(*
 * Set a customized input channel mapping.
 *
 * @param s           allocated Swr context, not yet initialized
 * @param channel_map customized input channel mapping (array of channel
 *                    indexes, -1 for a muted channel)
 * @return AVERROR error code in case of failure.
 *)
//int swr_set_channel_mapping(struct SwrContext *s, const int *channel_map);

(*
 * Set a customized remix matrix.
 *
 * @param s       allocated Swr context, not yet initialized
 * @param matrix  remix coefficients; matrix[i + stride * o] is
 *                the weight of input channel i in output channel o
 * @param stride  offset between lines of the matrix
 * @return  AVERROR error code in case of failure.
 *)
//int swr_set_matrix(struct SwrContext *s, const double *matrix, int stride);

(*
 * Drops the specified number of output samples.
 *)
//int swr_drop_output(struct SwrContext *s, int count);

(*
 * Injects the specified number of silence samples.
 *)
//int swr_inject_silence(struct SwrContext *s, int count);

(*
 * Gets the delay the next input sample will experience relative to the next output sample.
 *
 * Swresample can buffer data if more input has been provided than available
 * output space, also converting between sample rates needs a delay.
 * This function returns the sum of all such delays.
 * The exact delay is not necessarily an integer value in either input or
 * output sample rate. Especially when downsampling by a large value, the
 * output sample rate may be a poor choice to represent the delay, similarly
 * for upsampling and the input sample rate.
 *
 * @param s     swr context
 * @param base  timebase in which the returned delay will be
 *              if its set to 1 the returned delay is in seconds
 *              if its set to 1000 the returned delay is in milli seconds
 *              if its set to the input sample rate then the returned delay is in input samples
 *              if its set to the output sample rate then the returned delay is in output samples
 *              an exact rounding free delay can be found by using LCM(in_sample_rate, out_sample_rate)
 * @returns     the delay in 1/base units.
 *)
//int64_t swr_get_delay(struct SwrContext *s, int64_t base);

(*
 * Return the LIBSWRESAMPLE_VERSION_INT constant.
 *)
//unsigned swresample_version(void);

(*
 * Return the swr build-time configuration.
 *)
//const char *swresample_configuration(void);

(*
 * Return the swr license.
 *)
//const char *swresample_license(void);

implementation

uses ffmpeglib;

function swr_convert; external swscale_dll;

end.