/*

  * 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
  */
 

 #ifndef SWRESAMPLE_SWRESAMPLE_H
 #define SWRESAMPLE_SWRESAMPLE_H
 

 /**
  * @file

  * @ingroup lswr

  * libswresample public header
  */
 

 /**
  * @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().
  */

 #include <stdint.h>

 #include "libavutil/samplefmt.h"
 

 #include "libswresample/version.h"

 #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 */
 };
 

 typedef struct SwrContext SwrContext;

 
 /**

  * 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);
 
 /**

  * Check whether an swr context has been initialized or not.
  *
  * @return positive if it has been initialized, 0 if not initialized
  */
 int swr_is_initialized(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);

 /**

  * 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);
 

 /**
  * @}
  */
 

 #endif /* SWRESAMPLE_SWRESAMPLE_H */