/*
  * various filters for ACELP-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_ACELP_FILTERS_H
 #define AVCODEC_ACELP_FILTERS_H

 #include <stdint.h>
 

 typedef struct ACELPFContext {
     /**
     * Floating point version of ff_acelp_interpolate()
     */
     void (*acelp_interpolatef)(float *out, const float *in,
                             const float *filter_coeffs, int precision,
                             int frac_pos, int filter_length, int length);
 
     /**
      * Apply an order 2 rational transfer function in-place.
      *
      * @param out output buffer for filtered speech samples
      * @param in input buffer containing speech data (may be the same as out)
      * @param zero_coeffs z^-1 and z^-2 coefficients of the numerator
      * @param pole_coeffs z^-1 and z^-2 coefficients of the denominator
      * @param gain scale factor for final output
      * @param mem intermediate values used by filter (should be 0 initially)
      * @param n number of samples (should be a multiple of eight)
      */
     void (*acelp_apply_order_2_transfer_function)(float *out, const float *in,
                                                   const float zero_coeffs[2],
                                                   const float pole_coeffs[2],
                                                   float gain,
                                                   float mem[2], int n);
 
 }ACELPFContext;
 
 /**
  * Initialize ACELPFContext.
  */
 void ff_acelp_filter_init(ACELPFContext *c);
 void ff_acelp_filter_init_mips(ACELPFContext *c);
 

 /**

  * low-pass Finite Impulse Response filter coefficients.

  *

  * Hamming windowed sinc filter with cutoff freq 3/40 of the sampling freq,
  * the coefficients are scaled by 2^15.

  * This array only contains the right half of the filter.
  * This filter is likely identical to the one used in G.729, though this

  * could not be determined from the original comments with certainty.

  */
 extern const int16_t ff_acelp_interp_filter[61];
 
 /**

  * Generic FIR interpolation routine.

  * @param[out] out buffer for interpolated data

  * @param in input data
  * @param filter_coeffs interpolation filter coefficients (0.15)

  * @param precision sub sample factor, that is the precision of the position
  * @param frac_pos fractional part of position [0..precision-1]

  * @param filter_length filter length

  * @param length length of output

  *

  * filter_coeffs contains coefficients of the right half of the symmetric

  * interpolation filter. filter_coeffs[0] should the central (unpaired) coefficient.

  * See ff_acelp_interp_filter for an example.

  */

 void ff_acelp_interpolate(int16_t* out, const int16_t* in,
                           const int16_t* filter_coeffs, int precision,
                           int frac_pos, int filter_length, int length);

 
 /**

  * Floating point version of ff_acelp_interpolate()
  */
 void ff_acelp_interpolatef(float *out, const float *in,
                            const float *filter_coeffs, int precision,
                            int frac_pos, int filter_length, int length);
 
 
 /**

  * high-pass filtering and upscaling (4.2.5 of G.729).

  * @param[out]     out   output buffer for filtered speech data
  * @param[in,out]  hpf_f past filtered data from previous (2 items long)

  *                       frames (-0x20000000 <= (14.13) < 0x20000000)

  * @param in speech data to process
  * @param length input data size

  *
  * out[i] = 0.93980581 * in[i] - 1.8795834 * in[i-1] + 0.93980581 * in[i-2] +
  *          1.9330735 * out[i-1] - 0.93589199 * out[i-2]
  *

  * The filter has a cut-off frequency of 1/80 of the sampling freq

  *

  * @note Two items before the top of the in buffer must contain two items from the

  *       tail of the previous subframe.
  *

  * @remark It is safe to pass the same array in in and out parameters.

  *

  * @remark AMR uses mostly the same filter (cut-off frequency 60Hz, same formula,

  *         but constants differs in 5th sign after comma). Fortunately in
  *         fixed-point all coefficients are the same as in G.729. Thus this
  *         routine can be used for the fixed-point AMR decoder, too.
  */

 void ff_acelp_high_pass_filter(int16_t* out, int hpf_f[2],
                                const int16_t* in, int length);

 /**
  * Apply an order 2 rational transfer function in-place.
  *

  * @param out output buffer for filtered speech samples
  * @param in input buffer containing speech data (may be the same as out)

  * @param zero_coeffs z^-1 and z^-2 coefficients of the numerator
  * @param pole_coeffs z^-1 and z^-2 coefficients of the denominator
  * @param gain scale factor for final output
  * @param mem intermediate values used by filter (should be 0 initially)
  * @param n number of samples
  */

 void ff_acelp_apply_order_2_transfer_function(float *out, const float *in,

                                               const float zero_coeffs[2],
                                               const float pole_coeffs[2],
                                               float gain,
                                               float mem[2], int n);
 

 /**
  * Apply tilt compensation filter, 1 - tilt * z-1.
  *
  * @param mem pointer to the filter's state (one single float)
  * @param tilt tilt factor
  * @param samples array where the filter is applied
  * @param size the size of the samples array
  */
 void ff_tilt_compensation(float *mem, float tilt, float *samples, int size);
 
 

 #endif /* AVCODEC_ACELP_FILTERS_H */