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  * LP synthesis filter.
  46  * @param out [out] pointer to output buffer
  47  * @param filter_coeffs filter coefficients (-0x8000 <= (3.12) < 0x8000)
  48  * @param in input signal
  49  * @param buffer_length amount of data to process
  50  * @param filter_length filter length (10 for 10th order LP filter)
  51  * @param stop_on_overflow   1 - return immediately if overflow occurs
  52  *                           0 - ignore overflows
  53  * @param rounder the amount to add for rounding (usually 0x800 or 0xfff)
  54  *
  55  * @return 1 if overflow occurred, 0 - otherwise
  56  *
  57  * @note Output buffer must contain filter_length samples of past
  58  *       speech data before pointer.
  59  *
  60  * Routine applies 1/A(z) filter to given speech data.
  61  */
  62 int ff_celp_lp_synthesis_filter(int16_t *out,
  63                                 const int16_t* filter_coeffs,
  64                                 const int16_t* in,
  65                                 int buffer_length,
  66                                 int filter_length,
  67                                 int stop_on_overflow,
  68                                 int rounder);
  69
  70 /**
  71  * LP synthesis filter.
  72  * @param out [out] pointer to output buffer
  73  *        - the array out[-filter_length, -1] must
  74  *        contain the previous result of this filter
  75  * @param filter_coeffs filter coefficients.
  76  * @param in input signal
  77  * @param buffer_length amount of data to process
  78  * @param filter_length filter length (10 for 10th order LP filter)
  79  *
  80  * @note Output buffer must contain filter_length samples of past
  81  *       speech data before pointer.
  82  *
  83  * Routine applies 1/A(z) filter to given speech data.
  84  */
  85 void ff_celp_lp_synthesis_filterf(float *out,
  86                                   const float* filter_coeffs,
  87                                   const float* in,
  88                                   int buffer_length,
  89                                   int filter_length);
  90
  91 /**
  92  * LP zero synthesis filter.
  93  * @param out [out] pointer to output buffer
  94  * @param filter_coeffs filter coefficients.
  95  * @param in input signal
  96  *        - the array in[-filter_length, -1] must
  97  *        contain the previous input of this filter
  98  * @param buffer_length amount of data to process
  99  * @param filter_length filter length (10 for 10th order LP filter)
 100  *
 101  * @note Output buffer must contain filter_length samples of past
 102  *       speech data before pointer.
 103  *
 104  * Routine applies A(z) filter to given speech data.
 105  */
 106 void ff_celp_lp_zero_synthesis_filterf(float *out,
 107                                        const float* filter_coeffs,
 108                                        const float* in,
 109                                        int buffer_length,
 110                                        int filter_length);
 111
 112 #endif /* AVCODEC_CELP_FILTERS_H */