/*
  * Copyright (c) 2012 Justin Ruggles <justin.ruggles@gmail.com>
  *

  * 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 AVRESAMPLE_AVRESAMPLE_H
 #define AVRESAMPLE_AVRESAMPLE_H
 
 /**
  * @file

  * @ingroup lavr

  * external API header
  */
 

 /**

  * @defgroup lavr libavresample

  * @{
  *
  * Libavresample (lavr) is a library that handles audio resampling, sample
  * format conversion and mixing.
  *
  * Interaction with lavr is done through AVAudioResampleContext, which is
  * allocated with avresample_alloc_context(). 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
  * AVAudioResampleContext *avr = avresample_alloc_context();
  * av_opt_set_int(avr, "in_channel_layout",  AV_CH_LAYOUT_5POINT1, 0);
  * av_opt_set_int(avr, "out_channel_layout", AV_CH_LAYOUT_STEREO,  0);
  * av_opt_set_int(avr, "in_sample_rate",     48000,                0);
  * av_opt_set_int(avr, "out_sample_rate",    44100,                0);
  * av_opt_set_int(avr, "in_sample_fmt",      AV_SAMPLE_FMT_FLTP,   0);

  * av_opt_set_int(avr, "out_sample_fmt",     AV_SAMPLE_FMT_S16,    0);

  * @endcode
  *
  * Once the context is initialized, it must be opened with avresample_open(). If
  * you need to change the conversion parameters, you must close the context with
  * avresample_close(), change the parameters as described above, then reopen it
  * again.
  *
  * The conversion itself is done by repeatedly calling avresample_convert().
  * Note that the samples may get buffered in two places in lavr. The first one
  * is the output FIFO, where the samples end up if the output buffer is not
  * large enough. The data stored in there may be retrieved at any time with
  * avresample_read(). The second place is the resampling delay buffer,
  * applicable only when resampling is done. The samples in it require more input
  * before they can be processed. Their current amount is returned by
  * avresample_get_delay(). At the end of conversion the resampling buffer can be
  * flushed by calling avresample_convert() with NULL input.
  *
  * 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_linesize, in_samples;
  *
  * while (get_input(&input, &in_linesize, &in_samples)) {
  *     uint8_t *output
  *     int out_linesize;

  *     int out_samples = avresample_get_out_samples(avr, in_samples);
  *

  *     av_samples_alloc(&output, &out_linesize, 2, out_samples,
  *                      AV_SAMPLE_FMT_S16, 0);
  *     out_samples = avresample_convert(avr, &output, out_linesize, out_samples,
  *                                      input, in_linesize, in_samples);
  *     handle_output(output, out_linesize, out_samples);
  *     av_freep(&output);
  *  }
  *  @endcode
  *
  *  When the conversion is finished and the FIFOs are flushed if required, the
  *  conversion context and everything associated with it must be freed with
  *  avresample_free().
  */
 

 #include "libavutil/avutil.h"

 #include "libavutil/channel_layout.h"

 #include "libavutil/dict.h"

 #include "libavutil/frame.h"

 #include "libavutil/log.h"

 #include "libavutil/mathematics.h"

 
 #include "libavresample/version.h"
 
 #define AVRESAMPLE_MAX_CHANNELS 32
 
 typedef struct AVAudioResampleContext AVAudioResampleContext;
 
 /** Mixing Coefficient Types */
 enum AVMixCoeffType {

     AV_MIX_COEFF_TYPE_Q8,   /** 16-bit 8.8 fixed-point                      */

     AV_MIX_COEFF_TYPE_Q15,  /** 32-bit 17.15 fixed-point                    */
     AV_MIX_COEFF_TYPE_FLT,  /** floating-point                              */
     AV_MIX_COEFF_TYPE_NB,   /** Number of coeff types. Not part of ABI      */
 };
 

 /** Resampling Filter Types */
 enum AVResampleFilterType {
     AV_RESAMPLE_FILTER_TYPE_CUBIC,              /**< Cubic */
     AV_RESAMPLE_FILTER_TYPE_BLACKMAN_NUTTALL,   /**< Blackman Nuttall Windowed Sinc */
     AV_RESAMPLE_FILTER_TYPE_KAISER,             /**< Kaiser Windowed Sinc */
 };
 

 enum AVResampleDitherMethod {
     AV_RESAMPLE_DITHER_NONE,            /**< Do not use dithering */
     AV_RESAMPLE_DITHER_RECTANGULAR,     /**< Rectangular Dither */
     AV_RESAMPLE_DITHER_TRIANGULAR,      /**< Triangular Dither*/
     AV_RESAMPLE_DITHER_TRIANGULAR_HP,   /**< Triangular Dither with High Pass */
     AV_RESAMPLE_DITHER_TRIANGULAR_NS,   /**< Triangular Dither with Noise Shaping */
     AV_RESAMPLE_DITHER_NB,              /**< Number of dither types. Not part of ABI. */
 };
 

 /**
  * Return the LIBAVRESAMPLE_VERSION_INT constant.
  */
 unsigned avresample_version(void);
 
 /**
  * Return the libavresample build-time configuration.
  * @return  configure string
  */
 const char *avresample_configuration(void);
 
 /**
  * Return the libavresample license.
  */
 const char *avresample_license(void);
 
 /**
  * Get the AVClass for AVAudioResampleContext.
  *
  * Can be used in combination with AV_OPT_SEARCH_FAKE_OBJ for examining options
  * without allocating a context.
  *
  * @see av_opt_find().
  *
  * @return AVClass for AVAudioResampleContext
  */
 const AVClass *avresample_get_class(void);
 
 /**
  * Allocate AVAudioResampleContext and set options.
  *
  * @return  allocated audio resample context, or NULL on failure
  */
 AVAudioResampleContext *avresample_alloc_context(void);
 
 /**
  * Initialize AVAudioResampleContext.

  * @note The context must be configured using the AVOption API.

  * @note The fields "in_channel_layout", "out_channel_layout",
  *       "in_sample_rate", "out_sample_rate", "in_sample_fmt",
  *       "out_sample_fmt" must be set.

  *
  * @see av_opt_set_int()
  * @see av_opt_set_dict()

  * @see av_get_default_channel_layout()

  *
  * @param avr  audio resample context
  * @return     0 on success, negative AVERROR code on failure
  */
 int avresample_open(AVAudioResampleContext *avr);
 
 /**

  * Check whether an AVAudioResampleContext is open or closed.
  *
  * @param avr AVAudioResampleContext to check
  * @return 1 if avr is open, 0 if avr is closed.
  */
 int avresample_is_open(AVAudioResampleContext *avr);
 
 /**

  * Close AVAudioResampleContext.
  *
  * This closes the context, but it does not change the parameters. The context
  * can be reopened with avresample_open(). It does, however, clear the output
  * FIFO and any remaining leftover samples in the resampling delay buffer. If
  * there was a custom matrix being used, that is also cleared.
  *
  * @see avresample_convert()
  * @see avresample_set_matrix()
  *
  * @param avr  audio resample context
  */
 void avresample_close(AVAudioResampleContext *avr);
 
 /**
  * Free AVAudioResampleContext and associated AVOption values.
  *
  * This also calls avresample_close() before freeing.
  *
  * @param avr  audio resample context
  */
 void avresample_free(AVAudioResampleContext **avr);
 
 /**
  * Generate a channel mixing matrix.
  *
  * This function is the one used internally by libavresample for building the
  * default mixing matrix. It is made public just as a utility function for
  * building custom matrices.
  *
  * @param in_layout           input channel layout
  * @param out_layout          output channel layout
  * @param center_mix_level    mix level for the center channel
  * @param surround_mix_level  mix level for the surround channel(s)
  * @param lfe_mix_level       mix level for the low-frequency effects channel
  * @param normalize           if 1, coefficients will be normalized to prevent
  *                            overflow. if 0, coefficients will not be
  *                            normalized.
  * @param[out] matrix         mixing coefficients; matrix[i + stride * o] is
  *                            the weight of input channel i in output channel o.
  * @param stride              distance between adjacent input channels in the
  *                            matrix array

  * @param matrix_encoding     matrixed stereo downmix mode (e.g. dplii)

  * @return                    0 on success, negative AVERROR code on failure
  */
 int avresample_build_matrix(uint64_t in_layout, uint64_t out_layout,
                             double center_mix_level, double surround_mix_level,
                             double lfe_mix_level, int normalize, double *matrix,

                             int stride, enum AVMatrixEncoding matrix_encoding);

 
 /**
  * Get the current channel mixing matrix.
  *

  * If no custom matrix has been previously set or the AVAudioResampleContext is
  * not open, an error is returned.
  *

  * @param avr     audio resample context
  * @param matrix  mixing coefficients; matrix[i + stride * o] is the weight of
  *                input channel i in output channel o.
  * @param stride  distance between adjacent input channels in the matrix array
  * @return        0 on success, negative AVERROR code on failure
  */
 int avresample_get_matrix(AVAudioResampleContext *avr, double *matrix,
                           int stride);
 
 /**
  * Set channel mixing matrix.
  *
  * Allows for setting a custom mixing matrix, overriding the default matrix
  * generated internally during avresample_open(). This function can be called
  * anytime on an allocated context, either before or after calling

  * avresample_open(), as long as the channel layouts have been set.
  * avresample_convert() always uses the current matrix.

  * Calling avresample_close() on the context will clear the current matrix.
  *
  * @see avresample_close()
  *
  * @param avr     audio resample context
  * @param matrix  mixing coefficients; matrix[i + stride * o] is the weight of
  *                input channel i in output channel o.
  * @param stride  distance between adjacent input channels in the matrix array
  * @return        0 on success, negative AVERROR code on failure
  */
 int avresample_set_matrix(AVAudioResampleContext *avr, const double *matrix,
                           int stride);
 
 /**

  * Set a customized input channel mapping.
  *
  * This function can only be called when the allocated context is not open.
  * Also, the input channel layout must have already been set.
  *
  * Calling avresample_close() on the context will clear the channel mapping.
  *
  * The map for each input channel specifies the channel index in the source to
  * use for that particular channel, or -1 to mute the channel. Source channels
  * can be duplicated by using the same index for multiple input channels.
  *
  * Examples:
  *

  * Reordering 5.1 AAC order (C,L,R,Ls,Rs,LFE) to FFmpeg order (L,R,C,LFE,Ls,Rs):

  * { 1, 2, 0, 5, 3, 4 }
  *
  * Muting the 3rd channel in 4-channel input:
  * { 0, 1, -1, 3 }
  *
  * Duplicating the left channel of stereo input:
  * { 0, 0 }
  *
  * @param avr         audio resample context
  * @param channel_map customized input channel mapping
  * @return            0 on success, negative AVERROR code on failure
  */
 int avresample_set_channel_mapping(AVAudioResampleContext *avr,
                                    const int *channel_map);
 
 /**

  * Set compensation for resampling.
  *

  * This can be called anytime after avresample_open(). If resampling is not
  * automatically enabled because of a sample rate conversion, the
  * "force_resampling" option must have been set to 1 when opening the context
  * in order to use resampling compensation.

  *
  * @param avr                    audio resample context
  * @param sample_delta           compensation delta, in samples
  * @param compensation_distance  compensation distance, in samples
  * @return                       0 on success, negative AVERROR code on failure
  */
 int avresample_set_compensation(AVAudioResampleContext *avr, int sample_delta,
                                 int compensation_distance);
 
 /**

  * Provide the upper bound on the number of samples the configured
  * conversion would output.
  *
  * @param avr           audio resample context
  * @param in_nb_samples number of input samples
  *
  * @return              number of samples or AVERROR(EINVAL) if the value
  *                      would exceed INT_MAX
  */
 
 int avresample_get_out_samples(AVAudioResampleContext *avr, int in_nb_samples);
 
 /**

  * Convert input samples and write them to the output FIFO.
  *

  * The upper bound on the number of output samples can be obtained through
  * avresample_get_out_samples().

  *

  * The output data can be NULL or have fewer allocated samples than required.
  * In this case, any remaining samples not written to the output will be added
  * to an internal FIFO buffer, to be returned at the next call to this function
  * or to avresample_read().
  *
  * If converting sample rate, there may be data remaining in the internal
  * resampling delay buffer. avresample_get_delay() tells the number of remaining
  * samples. To get this data as output, call avresample_convert() with NULL
  * input.
  *
  * At the end of the conversion process, there may be data remaining in the
  * internal FIFO buffer. avresample_available() tells the number of remaining
  * samples. To get this data as output, either call avresample_convert() with
  * NULL input or call avresample_read().
  *

  * @see avresample_get_out_samples()

  * @see avresample_read()
  * @see avresample_get_delay()
  *
  * @param avr             audio resample context
  * @param output          output data pointers
  * @param out_plane_size  output plane size, in bytes.
  *                        This can be 0 if unknown, but that will lead to
  *                        optimized functions not being used directly on the
  *                        output, which could slow down some conversions.
  * @param out_samples     maximum number of samples that the output buffer can hold
  * @param input           input data pointers
  * @param in_plane_size   input plane size, in bytes
  *                        This can be 0 if unknown, but that will lead to
  *                        optimized functions not being used directly on the
  *                        input, which could slow down some conversions.
  * @param in_samples      number of input samples to convert
  * @return                number of samples written to the output buffer,
  *                        not including converted samples added to the internal
  *                        output FIFO
  */

 int avresample_convert(AVAudioResampleContext *avr, uint8_t **output,

                        int out_plane_size, int out_samples,
                        uint8_t * const *input, int in_plane_size,
                        int in_samples);

 
 /**
  * Return the number of samples currently in the resampling delay buffer.
  *
  * When resampling, there may be a delay between the input and output. Any
  * unconverted samples in each call are stored internally in a delay buffer.
  * This function allows the user to determine the current number of samples in
  * the delay buffer, which can be useful for synchronization.
  *
  * @see avresample_convert()
  *
  * @param avr  audio resample context
  * @return     number of samples currently in the resampling delay buffer
  */
 int avresample_get_delay(AVAudioResampleContext *avr);
 
 /**
  * Return the number of available samples in the output FIFO.
  *
  * During conversion, if the user does not specify an output buffer or
  * specifies an output buffer that is smaller than what is needed, remaining
  * samples that are not written to the output are stored to an internal FIFO
  * buffer. The samples in the FIFO can be read with avresample_read() or
  * avresample_convert().
  *
  * @see avresample_read()
  * @see avresample_convert()
  *
  * @param avr  audio resample context
  * @return     number of samples available for reading
  */
 int avresample_available(AVAudioResampleContext *avr);
 
 /**
  * Read samples from the output FIFO.
  *
  * During conversion, if the user does not specify an output buffer or
  * specifies an output buffer that is smaller than what is needed, remaining
  * samples that are not written to the output are stored to an internal FIFO
  * buffer. This function can be used to read samples from that internal FIFO.
  *
  * @see avresample_available()
  * @see avresample_convert()
  *
  * @param avr         audio resample context

  * @param output      output data pointers. May be NULL, in which case
  *                    nb_samples of data is discarded from output FIFO.

  * @param nb_samples  number of samples to read from the FIFO
  * @return            the number of samples written to output
  */

 int avresample_read(AVAudioResampleContext *avr, uint8_t **output, int nb_samples);

 /**

  * Convert the samples in the input AVFrame and write them to the output AVFrame.
  *
  * Input and output AVFrames must have channel_layout, sample_rate and format set.
  *
  * The upper bound on the number of output samples is obtained through
  * avresample_get_out_samples().
  *
  * If the output AVFrame does not have the data pointers allocated the nb_samples
  * field will be set using avresample_get_out_samples() and av_frame_get_buffer()
  * is called to allocate the frame.
  *
  * The output AVFrame can be NULL or have fewer allocated samples than required.
  * In this case, any remaining samples not written to the output will be added
  * to an internal FIFO buffer, to be returned at the next call to this function
  * or to avresample_convert() or to avresample_read().
  *
  * If converting sample rate, there may be data remaining in the internal
  * resampling delay buffer. avresample_get_delay() tells the number of
  * remaining samples. To get this data as output, call this function or
  * avresample_convert() with NULL input.
  *
  * At the end of the conversion process, there may be data remaining in the
  * internal FIFO buffer. avresample_available() tells the number of remaining
  * samples. To get this data as output, either call this function or
  * avresample_convert() with NULL input or call avresample_read().
  *
  * If the AVAudioResampleContext configuration does not match the output and
  * input AVFrame settings the conversion does not take place and depending on
  * which AVFrame is not matching AVERROR_OUTPUT_CHANGED, AVERROR_INPUT_CHANGED
  * or AVERROR_OUTPUT_CHANGED|AVERROR_INPUT_CHANGED is returned.
  *
  * @see avresample_get_out_samples()
  * @see avresample_available()
  * @see avresample_convert()
  * @see avresample_read()
  * @see avresample_get_delay()
  *
  * @param avr             audio resample context
  * @param output          output AVFrame
  * @param input           input AVFrame
  * @return                0 on success, AVERROR on failure or nonmatching
  *                        configuration.
  */
 int avresample_convert_frame(AVAudioResampleContext *avr,
                              AVFrame *output, AVFrame *input);
 
 /**
  * Configure or reconfigure the AVAudioResampleContext using the information
  * provided by the AVFrames.
  *
  * The original resampling context is reset even on failure.
  * The function calls avresample_close() internally if the context is open.
  *
  * @see avresample_open();
  * @see avresample_close();
  *
  * @param avr             audio resample context

  * @param out             output AVFrame
  * @param in              input AVFrame

  * @return                0 on success, AVERROR on failure.
  */
 int avresample_config(AVAudioResampleContext *avr, AVFrame *out, AVFrame *in);
 
 /**

  * @}
  */
 

 #endif /* AVRESAMPLE_AVRESAMPLE_H */