/*
  * various filters for CELP-based codecs
  *
  * Copyright (c) 2008 Vladimir Voroshilov
  *
  * 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 AVCODEC_CELP_FILTERS_H
 #define AVCODEC_CELP_FILTERS_H
 
 #include <stdint.h>
 

 typedef struct CELPFContext {
     /**
      * LP synthesis filter.
      * @param[out] out pointer to output buffer
      *        - the array out[-filter_length, -1] must
      *        contain the previous result of this filter
      * @param filter_coeffs filter coefficients.
      * @param in input signal
      * @param buffer_length amount of data to process
      * @param filter_length filter length (10 for 10th order LP filter). Must be
      *                      greater than 4 and even.
      *
      * @note Output buffer must contain filter_length samples of past
      *       speech data before pointer.
      *
      * Routine applies 1/A(z) filter to given speech data.
      */
     void (*celp_lp_synthesis_filterf)(float *out, const float *filter_coeffs,
                                       const float *in, int buffer_length,
                                       int filter_length);
 
     /**
      * LP zero synthesis filter.
      * @param[out] out pointer to output buffer
      * @param filter_coeffs filter coefficients.
      * @param in input signal
      *        - the array in[-filter_length, -1] must
      *        contain the previous input of this filter
      * @param buffer_length amount of data to process (should be a multiple of eight)
      * @param filter_length filter length (10 for 10th order LP filter;
      *                                      should be a multiple of two)
      *
      * @note Output buffer must contain filter_length samples of past
      *       speech data before pointer.
      *
      * Routine applies A(z) filter to given speech data.
      */
     void (*celp_lp_zero_synthesis_filterf)(float *out, const float *filter_coeffs,
                                            const float *in, int buffer_length,
                                            int filter_length);
 
 }CELPFContext;
 
 /**
  * Initialize CELPFContext.
  */
 void ff_celp_filter_init(CELPFContext *c);
 void ff_celp_filter_init_mips(CELPFContext *c);
 

 /**
  * Circularly convolve fixed vector with a phase dispersion impulse
  *        response filter (D.6.2 of G.729 and 6.1.5 of AMR).
  * @param fc_out vector with filter applied
  * @param fc_in source vector
  * @param filter phase filter coefficients
  *
  *  fc_out[n] = sum(i,0,len-1){ fc_in[i] * filter[(len + n - i)%len] }
  *

  * @note fc_in and fc_out should not overlap!

  */

 void ff_celp_convolve_circ(int16_t *fc_out, const int16_t *fc_in,
                            const int16_t *filter, int len);

 
 /**

  * Add an array to a rotated array.
  *
  * out[k] = in[k] + fac * lagged[k-lag] with wrap-around
  *
  * @param out result vector
  * @param in samples to be added unfiltered
  * @param lagged samples to be rotated, multiplied and added
  * @param lag lagged vector delay in the range [0, n]
  * @param fac scalefactor for lagged samples
  * @param n number of samples
  */
 void ff_celp_circ_addf(float *out, const float *in,
                        const float *lagged, int lag, float fac, int n);
 
 /**

  * LP synthesis filter.

  * @param[out] out pointer to output buffer

  * @param filter_coeffs filter coefficients (-0x8000 <= (3.12) < 0x8000)
  * @param in input signal
  * @param buffer_length amount of data to process
  * @param filter_length filter length (10 for 10th order LP filter)
  * @param stop_on_overflow   1 - return immediately if overflow occurs
  *                           0 - ignore overflows

  * @param shift the result is shifted right by this value

  * @param rounder the amount to add for rounding (usually 0x800 or 0xfff)
  *
  * @return 1 if overflow occurred, 0 - otherwise
  *

  * @note Output buffer must contain filter_length samples of past

  *       speech data before pointer.
  *
  * Routine applies 1/A(z) filter to given speech data.
  */

 int ff_celp_lp_synthesis_filter(int16_t *out, const int16_t *filter_coeffs,
                                 const int16_t *in, int buffer_length,
                                 int filter_length, int stop_on_overflow,

                                 int shift, int rounder);

 /**
  * LP synthesis filter.

  * @param[out] out pointer to output buffer

  *        - the array out[-filter_length, -1] must
  *        contain the previous result of this filter
  * @param filter_coeffs filter coefficients.
  * @param in input signal
  * @param buffer_length amount of data to process

  * @param filter_length filter length (10 for 10th order LP filter). Must be
  *                      greater than 4 and even.

  *

  * @note Output buffer must contain filter_length samples of past

  *       speech data before pointer.
  *
  * Routine applies 1/A(z) filter to given speech data.
  */

 void ff_celp_lp_synthesis_filterf(float *out, const float *filter_coeffs,
                                   const float *in, int buffer_length,

                                   int filter_length);

 /**
  * LP zero synthesis filter.

  * @param[out] out pointer to output buffer

  * @param filter_coeffs filter coefficients.
  * @param in input signal
  *        - the array in[-filter_length, -1] must
  *        contain the previous input of this filter
  * @param buffer_length amount of data to process
  * @param filter_length filter length (10 for 10th order LP filter)
  *
  * @note Output buffer must contain filter_length samples of past
  *       speech data before pointer.
  *
  * Routine applies A(z) filter to given speech data.
  */

 void ff_celp_lp_zero_synthesis_filterf(float *out, const float *filter_coeffs,
                                        const float *in, int buffer_length,

                                        int filter_length);

 #endif /* AVCODEC_CELP_FILTERS_H */