libavcodec/celp_filters.h

   1 /*
   2  * various filters for CELP-based codecs
   3  *
   4  * Copyright (c) 2008 Vladimir Voroshilov
   5  *
   6  * This file is part of FFmpeg.
   7  *
   8  * FFmpeg is free software; you can redistribute it and/or
   9  * modify it under the terms of the GNU Lesser General Public
  10  * License as published by the Free Software Foundation; either
  11  * version 2.1 of the License, or (at your option) any later version.
  12  *
  13  * FFmpeg is distributed in the hope that it will be useful,
  14  * but WITHOUT ANY WARRANTY; without even the implied warranty of
  15  * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
  16  * Lesser General Public License for more details.
  17  *
  18  * You should have received a copy of the GNU Lesser General Public
  19  * License along with FFmpeg; if not, write to the Free Software
  20  * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
  21  */
  22
  23 #ifndef AVCODEC_CELP_FILTERS_H
  24 #define AVCODEC_CELP_FILTERS_H
  25
  26 #include <stdint.h>
  27
  28 /**
  29  * Circularly convolve fixed vector with a phase dispersion impulse
  30  *        response filter (D.6.2 of G.729 and 6.1.5 of AMR).
  31  * @param fc_out vector with filter applied
  32  * @param fc_in source vector
  33  * @param filter phase filter coefficients
  34  *
  35  *  fc_out[n] = sum(i,0,len-1){ fc_in[i] * filter[(len + n - i)%len] }
  36  *
  37  * \note fc_in and fc_out should not overlap!
  38  */
  39 void ff_celp_convolve_circ(int16_t* fc_out,
  40                            const int16_t* fc_in,
  41                            const int16_t* filter,
  42                            int len);
  43
  44 /**
  45  * Add an array to a rotated array.
  46  *
  47  * out[k] = in[k] + fac * lagged[k-lag] with wrap-around
  48  *
  49  * @param out result vector
  50  * @param in samples to be added unfiltered
  51  * @param lagged samples to be rotated, multiplied and added
  52  * @param lag lagged vector delay in the range [0, n]
  53  * @param fac scalefactor for lagged samples
  54  * @param n number of samples
  55  */
  56 void ff_celp_circ_addf(float *out, const float *in,
  57                        const float *lagged, int lag, float fac, int n);
  58
  59 /**
  60  * LP synthesis filter.
  61  * @param out [out] pointer to output buffer
  62  * @param filter_coeffs filter coefficients (-0x8000 <= (3.12) < 0x8000)
  63  * @param in input signal
  64  * @param buffer_length amount of data to process
  65  * @param filter_length filter length (10 for 10th order LP filter)
  66  * @param stop_on_overflow   1 - return immediately if overflow occurs
  67  *                           0 - ignore overflows
  68  * @param rounder the amount to add for rounding (usually 0x800 or 0xfff)
  69  *
  70  * @return 1 if overflow occurred, 0 - otherwise
  71  *
  72  * @note Output buffer must contain filter_length samples of past
  73  *       speech data before pointer.
  74  *
  75  * Routine applies 1/A(z) filter to given speech data.
  76  */
  77 int ff_celp_lp_synthesis_filter(int16_t *out,
  78                                 const int16_t* filter_coeffs,
  79                                 const int16_t* in,
  80                                 int buffer_length,
  81                                 int filter_length,
  82                                 int stop_on_overflow,
  83                                 int rounder);
  84
  85 /**
  86  * LP synthesis filter.
  87  * @param out [out] pointer to output buffer
  88  *        - the array out[-filter_length, -1] must
  89  *        contain the previous result of this filter
  90  * @param filter_coeffs filter coefficients.
  91  * @param in input signal
  92  * @param buffer_length amount of data to process
  93  * @param filter_length filter length (10 for 10th order LP filter)
  94  *
  95  * @note Output buffer must contain filter_length samples of past
  96  *       speech data before pointer.
  97  *
  98  * Routine applies 1/A(z) filter to given speech data.
  99  */
 100 void ff_celp_lp_synthesis_filterf(float *out,
 101                                   const float* filter_coeffs,
 102                                   const float* in,
 103                                   int buffer_length,
 104                                   int filter_length);
 105
 106 /**
 107  * LP zero synthesis filter.
 108  * @param out [out] pointer to output buffer
 109  * @param filter_coeffs filter coefficients.
 110  * @param in input signal
 111  *        - the array in[-filter_length, -1] must
 112  *        contain the previous input of this filter
 113  * @param buffer_length amount of data to process
 114  * @param filter_length filter length (10 for 10th order LP filter)
 115  *
 116  * @note Output buffer must contain filter_length samples of past
 117  *       speech data before pointer.
 118  *
 119  * Routine applies A(z) filter to given speech data.
 120  */
 121 void ff_celp_lp_zero_synthesis_filterf(float *out,
 122                                        const float* filter_coeffs,
 123                                        const float* in,
 124                                        int buffer_length,
 125                                        int filter_length);
 126
 127 #endif /* AVCODEC_CELP_FILTERS_H */