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/*

 * Copyright (c) 2012 Justin Ruggles 

 *

 * 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_available(avr) +

 *                       av_rescale_rnd(avresample_get_delay(avr) +

 *                                      in_samples, 44100, 48000, AV_ROUND_UP);

 *     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/log.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.

 *

 * @param avr  audio resample context

 * @return     0 on success, negative AVERROR code on failure

 */

int avresample_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);

/**

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

 *

 * The upper bound on the number of output samples is given by

 * avresample_available() + (avresample_get_delay() + number of input samples) *

 * output sample rate / input sample rate.

 *

 * 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_available()

 * @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 **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);

/**

 * @}

 */

#endif /* AVRESAMPLE_AVRESAMPLE_H */