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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 SWR_INTERNAL_H
    #define SWR_INTERNAL_H
    
    #include "swresample.h"
    
    #include "libavutil/channel_layout.h"
    
    #include "config.h"
    
    
    #define SQRT3_2      1.22474487139158904909  /* sqrt(3/2) */
    
    #define NS_TAPS 20
    
    
    #if ARCH_X86_64
    typedef int64_t integer;
    #else
    typedef int integer;
    #endif
    
    typedef void (mix_1_1_func_type)(void *out, const void *in, void *coeffp, integer index, integer len);
    typedef void (mix_2_1_func_type)(void *out, const void *in1, const void *in2, void *coeffp, integer index1, integer index2, integer len);
    
    typedef void (mix_any_func_type)(uint8_t **out, const uint8_t **in1, void *coeffp, integer len);
    
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    typedef struct AudioData{
    
        uint8_t *ch[SWR_CH_MAX];    ///< samples buffer per channel
        uint8_t *data;              ///< samples buffer
        int ch_count;               ///< number of channels
        int bps;                    ///< bytes per sample
        int count;                  ///< number of samples
        int planar;                 ///< 1 if planar audio, 0 otherwise
    
        enum AVSampleFormat fmt;    ///< sample format
    
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    } AudioData;
    
    
    struct DitherContext {
        enum SwrDitherType method;
        int dither_pos;
        float scale;
    
        int ns_taps;                                    ///< Noise shaping dither taps
        float ns_scale;                                 ///< Noise shaping dither scale
        float ns_scale_1;                               ///< Noise shaping dither scale^-1
        int ns_pos;                                     ///< Noise shaping dither position
        float ns_coeffs[NS_TAPS];                       ///< Noise shaping filter coefficients
        float ns_errors[SWR_CH_MAX][2*NS_TAPS];
        AudioData noise;                                ///< noise used for dithering
    };
    
    
        const AVClass *av_class;                        ///< AVClass used for AVOption and av_log()
        int log_level_offset;                           ///< logging level offset
        void *log_ctx;                                  ///< parent logging context
        enum AVSampleFormat  in_sample_fmt;             ///< input sample format
    
        enum AVSampleFormat int_sample_fmt;             ///< internal sample format (AV_SAMPLE_FMT_FLTP or AV_SAMPLE_FMT_S16P)
    
        enum AVSampleFormat out_sample_fmt;             ///< output sample format
        int64_t  in_ch_layout;                          ///< input channel layout
        int64_t out_ch_layout;                          ///< output channel layout
        int      in_sample_rate;                        ///< input sample rate
        int     out_sample_rate;                        ///< output sample rate
        int flags;                                      ///< miscellaneous flags such as SWR_FLAG_RESAMPLE
    
        float slev;                                     ///< surround mixing level
    
        float clev;                                     ///< center mixing level
    
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        float lfe_mix_level;                            ///< LFE mixing level
    
        float rematrix_volume;                          ///< rematrixing volume coefficient
    
        enum AVMatrixEncoding matrix_encoding;          /**< matrixed stereo encoding */
    
        const int *channel_map;                         ///< channel index (or -1 if muted channel) map
        int used_ch_count;                              ///< number of used input channels (mapped channel count if channel_map, otherwise in.ch_count)
    
        enum SwrEngine engine;
    
    
        struct DitherContext dither;
    
        int filter_size;                                /**< length of each FIR filter in the resampling filterbank relative to the cutoff frequency */
        int phase_shift;                                /**< log2 of the number of entries in the resampling polyphase filterbank */
        int linear_interp;                              /**< if 1 then the resampling FIR filter will be linearly interpolated */
    
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        double cutoff;                                  /**< resampling cutoff frequency (swr: 6dB point; soxr: 0dB point). 1.0 corresponds to half the output sample rate */
        enum SwrFilterType filter_type;                 /**< swr resampling filter type */
        int kaiser_beta;                                /**< swr beta value for Kaiser window (only applicable if filter_type == AV_FILTER_TYPE_KAISER) */
        double precision;                               /**< soxr resampling precision (in bits) */
        int cheby;                                      /**< soxr: if 1 then passband rolloff will be none (Chebyshev) & irrational ratio approximation precision will be higher */
    
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        float min_compensation;                         ///< swr minimum below which no compensation will happen
        float min_hard_compensation;                    ///< swr minimum below which no silence inject / sample drop will happen
        float soft_compensation_duration;               ///< swr duration over which soft compensation is applied
        float max_soft_compensation;                    ///< swr maximum soft compensation in seconds over soft_compensation_duration
        float async;                                    ///< swr simple 1 parameter async, similar to ffmpegs -async
    
        int resample_first;                             ///< 1 if resampling must come first, 0 if rematrixing
        int rematrix;                                   ///< flag to indicate if rematrixing is needed (basically if input and output layouts mismatch)
    
        int rematrix_custom;                            ///< flag to indicate that a custom matrix has been defined
    
        AudioData in;                                   ///< input audio data
        AudioData postin;                               ///< post-input audio data: used for rematrix/resample
        AudioData midbuf;                               ///< intermediate audio data (postin/preout)
        AudioData preout;                               ///< pre-output audio data: used for rematrix/resample
        AudioData out;                                  ///< converted output audio data
        AudioData in_buffer;                            ///< cached audio data (convert and resample purpose)
        int in_buffer_index;                            ///< cached buffer position
        int in_buffer_count;                            ///< cached buffer length
        int resample_in_constraint;                     ///< 1 if the input end was reach before the output end, 0 otherwise
    
        int flushed;                                    ///< 1 if data is to be flushed and no further input is expected
    
        int64_t outpts;                                 ///< output PTS
    
        int drop_output;                                ///< number of output samples to drop
    
        struct AudioConvert *in_convert;                ///< input conversion context
        struct AudioConvert *out_convert;               ///< output conversion context
        struct AudioConvert *full_convert;              ///< full conversion context (single conversion for input and output)
        struct ResampleContext *resample;               ///< resampling context
    
        struct Resampler const *resampler;              ///< resampler virtual function table
    
        float matrix[SWR_CH_MAX][SWR_CH_MAX];           ///< floating point rematrixing coefficients
    
        uint8_t *native_matrix;
        uint8_t *native_one;
    
        uint8_t *native_simd_matrix;
    
        int32_t matrix32[SWR_CH_MAX][SWR_CH_MAX];       ///< 17.15 fixed point rematrixing coefficients
        uint8_t matrix_ch[SWR_CH_MAX][SWR_CH_MAX+1];    ///< Lists of input channels per output channel that have non zero rematrixing coefficients
    
        mix_1_1_func_type *mix_1_1_f;
    
        mix_1_1_func_type *mix_1_1_simd;
    
    
        mix_2_1_func_type *mix_2_1_f;
    
        mix_2_1_func_type *mix_2_1_simd;
    
        mix_any_func_type *mix_any_f;
    
    
        /* TODO: callbacks for ASM optimizations */
    
    typedef struct ResampleContext * (* resample_init_func)(struct ResampleContext *c, int out_rate, int in_rate, int filter_size, int phase_shift, int linear,
    
                                        double cutoff, enum AVSampleFormat format, enum SwrFilterType filter_type, int kaiser_beta, double precision, int cheby);
    
    typedef void    (* resample_free_func)(struct ResampleContext **c);
    typedef int     (* multiple_resample_func)(struct ResampleContext *c, AudioData *dst, int dst_size, AudioData *src, int src_size, int *consumed);
    
    typedef int     (* resample_flush_func)(struct SwrContext *c);
    
    typedef int     (* set_compensation_func)(struct ResampleContext *c, int sample_delta, int compensation_distance);
    typedef int64_t (* get_delay_func)(struct SwrContext *s, int64_t base);
    
    struct Resampler {
      resample_init_func            init;
      resample_free_func            free;
      multiple_resample_func        multiple_resample;
    
      resample_flush_func           flush;
    
      set_compensation_func         set_compensation;
      get_delay_func                get_delay;
    };
    
    extern struct Resampler const swri_resampler;
    
    
    int swri_realloc_audio(AudioData *a, int count);
    
    int swri_resample_int16(struct ResampleContext *c, int16_t *dst, const int16_t *src, int *consumed, int src_size, int dst_size, int update_ctx);
    int swri_resample_int32(struct ResampleContext *c, int32_t *dst, const int32_t *src, int *consumed, int src_size, int dst_size, int update_ctx);
    int swri_resample_float(struct ResampleContext *c, float   *dst, const float   *src, int *consumed, int src_size, int dst_size, int update_ctx);
    
    int swri_resample_double(struct ResampleContext *c,double  *dst, const double  *src, int *consumed, int src_size, int dst_size, int update_ctx);
    
    void swri_noise_shaping_int16 (SwrContext *s, AudioData *srcs, AudioData *noises, int count);
    void swri_noise_shaping_int32 (SwrContext *s, AudioData *srcs, AudioData *noises, int count);
    void swri_noise_shaping_float (SwrContext *s, AudioData *srcs, AudioData *noises, int count);
    void swri_noise_shaping_double(SwrContext *s, AudioData *srcs, AudioData *noises, int count);
    
    
    int swri_rematrix_init(SwrContext *s);
    
    void swri_rematrix_free(SwrContext *s);
    
    int swri_rematrix(SwrContext *s, AudioData *out, AudioData *in, int len, int mustcopy);
    
    void swri_rematrix_init_x86(struct SwrContext *s);
    
    void swri_get_dither(SwrContext *s, void *dst, int len, unsigned seed, enum AVSampleFormat noise_fmt);
    
    int swri_dither_init(SwrContext *s, enum AVSampleFormat out_fmt, enum AVSampleFormat in_fmt);
    
    void swri_audio_convert_init_arm(struct AudioConvert *ac,
                                     enum AVSampleFormat out_fmt,
                                     enum AVSampleFormat in_fmt,
                                     int channels);
    
    void swri_audio_convert_init_x86(struct AudioConvert *ac,
                                     enum AVSampleFormat out_fmt,
                                     enum AVSampleFormat in_fmt,
                                     int channels);
    
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    #endif