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 Libav.
   7  *
   8  * Libav 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  * Libav 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 Libav; 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, const int16_t *fc_in,
  40                            const int16_t *filter, int len);
  41
  42 /**
  43  * Add an array to a rotated array.
  44  *
  45  * out[k] = in[k] + fac * lagged[k-lag] with wrap-around
  46  *
  47  * @param out result vector
  48  * @param in samples to be added unfiltered
  49  * @param lagged samples to be rotated, multiplied and added
  50  * @param lag lagged vector delay in the range [0, n]
  51  * @param fac scalefactor for lagged samples
  52  * @param n number of samples
  53  */
  54 void ff_celp_circ_addf(float *out, const float *in,
  55                        const float *lagged, int lag, float fac, int n);
  56
  57 /**
  58  * LP synthesis filter.
  59  * @param[out] out pointer to output buffer
  60  * @param filter_coeffs filter coefficients (-0x8000 <= (3.12) < 0x8000)
  61  * @param in input signal
  62  * @param buffer_length amount of data to process
  63  * @param filter_length filter length (10 for 10th order LP filter)
  64  * @param stop_on_overflow   1 - return immediately if overflow occurs
  65  *                           0 - ignore overflows
  66  * @param shift the result is shifted right by this value
  67  * @param rounder the amount to add for rounding (usually 0x800 or 0xfff)
  68  *
  69  * @return 1 if overflow occurred, 0 - otherwise
  70  *
  71  * @note Output buffer must contain filter_length samples of past
  72  *       speech data before pointer.
  73  *
  74  * Routine applies 1/A(z) filter to given speech data.
  75  */
  76 int ff_celp_lp_synthesis_filter(int16_t *out, const int16_t *filter_coeffs,
  77                                 const int16_t *in, int buffer_length,
  78                                 int filter_length, int stop_on_overflow,
  79                                 int shift, int rounder);
  80
  81 /**
  82  * LP synthesis filter.
  83  * @param[out] out pointer to output buffer
  84  *        - the array out[-filter_length, -1] must
  85  *        contain the previous result of this filter
  86  * @param filter_coeffs filter coefficients.
  87  * @param in input signal
  88  * @param buffer_length amount of data to process
  89  * @param filter_length filter length (10 for 10th order LP filter). Must be
  90  *                      greater than 4 and even.
  91  *
  92  * @note Output buffer must contain filter_length samples of past
  93  *       speech data before pointer.
  94  *
  95  * Routine applies 1/A(z) filter to given speech data.
  96  */
  97 void ff_celp_lp_synthesis_filterf(float *out, const float *filter_coeffs,
  98                                   const float *in, int buffer_length,
  99                                   int filter_length);
 100
 101 /**
 102  * LP zero synthesis filter.
 103  * @param[out] out pointer to output buffer
 104  * @param filter_coeffs filter coefficients.
 105  * @param in input signal
 106  *        - the array in[-filter_length, -1] must
 107  *        contain the previous input of this filter
 108  * @param buffer_length amount of data to process
 109  * @param filter_length filter length (10 for 10th order LP filter)
 110  *
 111  * @note Output buffer must contain filter_length samples of past
 112  *       speech data before pointer.
 113  *
 114  * Routine applies A(z) filter to given speech data.
 115  */
 116 void ff_celp_lp_zero_synthesis_filterf(float *out, const float *filter_coeffs,
 117                                        const float *in, int buffer_length,
 118                                        int filter_length);
 119
 120 #endif /* AVCODEC_CELP_FILTERS_H */