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    /*
    
     * Copyright (C) 2011-2013 Michael Niedermayer (michaelni@gmx.at)
    
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     *
     * 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
    
     * Audio resampling, sample format conversion and mixing library.
    
     *
     * 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.
     *
    
     * The first thing you will need to do in order to use lswr is to allocate
     * SwrContext. This can be done with swr_alloc() or swr_alloc_set_opts(). If you
     * are using the former, you must set options through the @ref avoptions API.
     * The latter function provides the same feature, but it allows you to set some
     * common options in the same statement.
     *
    
     * 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). This is using the swr_alloc() function.
    
     * @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);
    
     * The same job can be done using swr_alloc_set_opts() as well:
     * @code
     * SwrContext *swr = swr_alloc_set_opts(NULL,  // we're allocating a new context
     *                       AV_CH_LAYOUT_STEREO,  // out_ch_layout
     *                       AV_SAMPLE_FMT_S16,    // out_sample_fmt
     *                       44100,                // out_sample_rate
     *                       AV_CH_LAYOUT_5POINT1, // in_ch_layout
     *                       AV_SAMPLE_FMT_FLTP,   // in_sample_fmt
     *                       48000,                // in_sample_rate
     *                       0,                    // log_offset
     *                       NULL);                // log_ctx
     * @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
    
     * using @ref AVOptions, as described above in the first example; or by using
     * swr_alloc_set_opts(), but with the first argument the allocated context.
     * You must 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 samples used in the conversion process can be managed with the libavutil
     * @ref lavu_sampmanip "samples manipulation" API, including av_samples_alloc()
     * function used in the following example.
     *
    
     * 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);
    
     *
     * When the conversion is finished, the conversion
     * context and everything associated with it must be freed with swr_free().
    
     * A swr_close() function is also available, but it exists mainly for
     * compatibility with libavresample, and is not required to be called.
     *
    
     * There will be no memory leak if the data is not completely flushed before
     * swr_free().
     */
    
    #include <stdint.h>
    
    #include "libavutil/channel_layout.h"
    
    #include "libavutil/frame.h"
    
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    #include "libavutil/samplefmt.h"
    
    
    #include "libswresample/version.h"
    
    #if LIBSWRESAMPLE_VERSION_MAJOR < 1
    
    #define SWR_CH_MAX 32   ///< Maximum number of channels
    
    /**
     * @name Option constants
     * These constants are used for the @ref avoptions interface for lswr.
     * @{
     *
     */
    
    
    #define SWR_FLAG_RESAMPLE 1 ///< Force resampling even if equal sample rate
    
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    //TODO use int resample ?
    //long term TODO can we enable this dynamically?
    
    
    /** Dithering algorithms */
    
    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 */
    
    /**
     * The libswresample context. Unlike libavcodec and libavformat, this structure
     * is opaque. This means that if you would like to set options, you must use
     * the @ref avoptions API and cannot directly set values to members of the
     * structure.
     */
    
    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().
    
     * @return the AVClass of SwrContext
    
     */
    const AVClass *swr_get_class(void);
    
    
    /**
     * @name SwrContext constructor functions
     * @{
     */
    
    
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    /**
     * 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
    
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     */
    struct SwrContext *swr_alloc(void);
    
    /**
     * Initialize context after user parameters have been set.
    
     * @note The context must be configured using the AVOption API.
     *
     * @see av_opt_set_int()
     * @see av_opt_set_dict()
    
     * @param[in,out]   s Swr context to initialize
    
     * @return AVERROR error code in case of failure.
    
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     */
    int swr_init(struct SwrContext *s);
    
    
    /**
     * Check whether an swr context has been initialized or not.
     *
    
     * @param[in]       s Swr context to check
     * @see swr_init()
    
     * @return positive if it has been initialized, 0 if not initialized
     */
    int swr_is_initialized(struct SwrContext *s);
    
    
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    /**
    
     * 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               existing Swr context if available, or NULL if not
    
     * @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
    
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     */
    
    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,
    
    /**
     * @}
     *
     * @name SwrContext destructor functions
     * @{
     */
    
    
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    /**
    
     * Free the given SwrContext and set the pointer to NULL.
    
     *
     * @param[in] s a pointer to a pointer to Swr context
    
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     */
    void swr_free(struct SwrContext **s);
    
    
    /**
     * Closes the context so that swr_is_initialized() returns 0.
     *
    
     * The context can be brought back to life by running swr_init(),
    
     * swr_init() can also be used without swr_close().
     * This function is mainly provided for simplifying the usecase
    
     * where one tries to support libavresample and libswresample.
    
     *
     * @param[in,out] s Swr context to be closed
    
     */
    void swr_close(struct SwrContext *s);
    
    
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    /**
    
     * @}
     *
     * @name Core conversion functions
     * @{
     */
    
    /** 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 using swr_get_out_samples() to retrieve an
     * upper bound on the required number of output samples for the given number of
     * input samples. Conversion 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
    
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     */
    
    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.
    
     *       @li When automatic timestamp compensation is not used, (min_compensation >= FLT_MAX)
    
     *              in this case timestamps will be passed through with delays compensated
    
     *       @li When automatic timestamp compensation is used, (min_compensation < FLT_MAX)
     *              in this case the output timestamps will match output sample numbers.
     *              See ffmpeg-resampler(1) for the two modes of compensation.
     *
     * @param s[in]     initialized Swr context
     * @param pts[in]   timestamp for the next input sample, INT64_MIN if unknown
     * @see swr_set_compensation(), swr_drop_output(), and swr_inject_silence() are
     *      function used internally for timestamp compensation.
    
     * @return the output timestamp for the next output sample
    
     */
    int64_t swr_next_pts(struct SwrContext *s, int64_t pts);
    
    
    /**
     * @}
     *
     * @name Low-level option setting functions
     * These functons provide a means to set low-level options that is not possible
     * with the AVOption API.
     * @{
     */
    
    
     * Activate resampling compensation ("soft" compensation). This function is
     * internally called when needed in swr_next_pts().
     *
     * @param[in,out] s             allocated Swr context. If it is not initialized,
     *                              or SWR_FLAG_RESAMPLE is not set, swr_init() is
     *                              called with the flag set.
     * @param[in]     sample_delta  delta in PTS per sample
     * @param[in]     compensation_distance number of samples to compensate for
     * @return    >= 0 on success, AVERROR error codes if:
     *            @li @c s is NULL,
     *            @li @c compensation_distance is less than 0,
     *            @li @c compensation_distance is 0 but sample_delta is not,
     *            @li compensation unsupported by resampler, or
     *            @li swr_init() fails when called.
    
    int swr_set_compensation(struct SwrContext *s, int sample_delta, int compensation_distance);
    
    /**
     * Set a customized input channel mapping.
     *
    
     * @param[in,out] s           allocated Swr context, not yet initialized
     * @param[in]     channel_map customized input channel mapping (array of channel
     *                            indexes, -1 for a muted channel)
     * @return >= 0 on success, or AVERROR error code in case of failure.
    
     */
    int swr_set_channel_mapping(struct SwrContext *s, const int *channel_map);
    
    
    /**
     * Generate a channel mixing matrix.
     *
     * This function is the one used internally by libswresample 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 rematrix_maxval     if 1.0, coefficients will be normalized to prevent
     *                            overflow. if INT_MAX, 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)
     * @param log_ctx             parent logging context, can be NULL
     * @return                    0 on success, negative AVERROR code on failure
     */
    int swr_build_matrix(uint64_t in_layout, uint64_t out_layout,
                         double center_mix_level, double surround_mix_level,
                         double lfe_mix_level, double rematrix_maxval,
                         double rematrix_volume, double *matrix,
                         int stride, enum AVMatrixEncoding matrix_encoding,
                         void *log_ctx);
    
    
    /**
     * 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  >= 0 on success, or AVERROR error code in case of failure.
    
     */
    int swr_set_matrix(struct SwrContext *s, const double *matrix, int stride);
    
    
    /**
     * @}
     *
     * @name Sample handling functions
     * @{
     */
    
    
    /**
     * Drops the specified number of output samples.
    
     *
     * This function, along with swr_inject_silence(), is called by swr_next_pts()
     * if needed for "hard" compensation.
     *
     * @param s     allocated Swr context
     * @param count number of samples to be dropped
     *
     * @return >= 0 on success, or a negative AVERROR code on failure
    
     */
    int swr_drop_output(struct SwrContext *s, int count);
    
    
    /**
     * Injects the specified number of silence samples.
    
     *
     * This function, along with swr_drop_output(), is called by swr_next_pts()
     * if needed for "hard" compensation.
     *
     * @param s     allocated Swr context
     * @param count number of samples to be dropped
     *
     * @return >= 0 on success, or a negative AVERROR code on failure
    
     */
    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 base  timebase in which the returned delay will be:
     *              @li if it's set to 1 the returned delay is in seconds
     *              @li if it's set to 1000 the returned delay is in milliseconds
     *              @li if it's set to the input sample rate then the returned
     *                  delay is in input samples
     *              @li if it's set to the output sample rate then the returned
     *                  delay is in output samples
     *              @li if it's the least common multiple of in_sample_rate and
     *                  out_sample_rate then an exact rounding-free delay will be
     *                  returned
     * @returns     the delay in 1 / @c base units.
    
     */
    int64_t swr_get_delay(struct SwrContext *s, int64_t base);
    
    
    /**
     * Find an upper bound on the number of samples that the next swr_convert
     * call will output, if called with in_samples of input samples. This
     * depends on the internal state, and anything changing the internal state
     * (like further swr_convert() calls) will may change the number of samples
     * swr_get_out_samples() returns for the same number of input samples.
     *
     * @param in_samples    number of input samples.
     * @note any call to swr_inject_silence(), swr_convert(), swr_next_pts()
     *       or swr_set_compensation() invalidates this limit
     * @note it is recommended to pass the correct available buffer size
     *       to all functions like swr_convert() even if swr_get_out_samples()
     *       indicates that less would be used.
     * @returns an upper bound on the number of samples that the next swr_convert
     *          will output or a negative value to indicate an error
     */
    int swr_get_out_samples(struct SwrContext *s, int in_samples);
    
    
    /**
     * @}
     *
     * @name Configuration accessors
     * @{
     */
    
    
     * Return the @ref LIBSWRESAMPLE_VERSION_INT constant.
     *
     * This is useful to check if the build-time libswresample has the same version
     * as the run-time one.
     *
     * @returns     the unsigned int-typed version
    
     */
    unsigned swresample_version(void);
    
    /**
     * Return the swr build-time configuration.
    
     *
     * @returns     the build-time @c ./configure flags
    
     */
    const char *swresample_configuration(void);
    
    /**
     * Return the swr license.
    
     *
     * @returns     the license of libswresample, determined at build-time
    
    /**
     * @}
     *
     * @name AVFrame based API
     * @{
     */
    
    /**
     * 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.
     *
     * If the output AVFrame does not have the data pointers allocated the nb_samples
     * field will be set using 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 swr_convert().
     *
     * If converting sample rate, there may be data remaining in the internal
     * resampling delay buffer. swr_get_delay() tells the number of
     * remaining samples. To get this data as output, call this function or
     * swr_convert() with NULL input.
     *
     * If the SwrContext 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 the result of a bitwise-OR of them is returned.
     *
     * @see swr_delay()
     * @see swr_convert()
     * @see swr_get_delay()
     *
     * @param swr             audio resample context
     * @param output          output AVFrame
     * @param input           input AVFrame
     * @return                0 on success, AVERROR on failure or nonmatching
     *                        configuration.
     */
    int swr_convert_frame(SwrContext *swr,
                          AVFrame *output, const AVFrame *input);
    
    /**
     * Configure or reconfigure the SwrContext using the information
     * provided by the AVFrames.
     *
     * The original resampling context is reset even on failure.
     * The function calls swr_close() internally if the context is open.
     *
     * @see swr_close();
     *
     * @param swr             audio resample context
     * @param output          output AVFrame
     * @param input           input AVFrame
     * @return                0 on success, AVERROR on failure.
     */
    int swr_config_frame(SwrContext *swr, const AVFrame *out, const AVFrame *in);
    
    
    #endif /* SWRESAMPLE_SWRESAMPLE_H */