libavcodec/acelp_filters.h

   1 /*
   2  * various filters for ACELP-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_ACELP_FILTERS_H
  24 #define AVCODEC_ACELP_FILTERS_H
  25
  26 #include <stdint.h>
  27
  28 /**
  29  * low-pass Finite Impulse Response filter coefficients.
  30  *
  31  * Hamming windowed sinc filter with cutoff freq 3/40 of the sampling freq,
  32  * the coefficients are scaled by 2^15.
  33  * This array only contains the right half of the filter.
  34  * This filter is likely identical to the one used in G.729, though this
  35  * could not be determined from the original comments with certainity.
  36  */
  37 extern const int16_t ff_acelp_interp_filter[61];
  38
  39 /**
  40  * Generic FIR interpolation routine.
  41  * @param out [out] buffer for interpolated data
  42  * @param in input data
  43  * @param filter_coeffs interpolation filter coefficients (0.15)
  44  * @param precision sub sample factor, that is the precision of the position
  45  * @param frac_pos fractional part of position [0..precision-1]
  46  * @param filter_length filter length
  47  * @param length length of output
  48  *
  49  * filter_coeffs contains coefficients of the right half of the symmetric
  50  * interpolation filter. filter_coeffs[0] should the central (unpaired) coefficient.
  51  * See ff_acelp_interp_filter for an example.
  52  *
  53  */
  54 void ff_acelp_interpolate(
  55         int16_t* out,
  56         const int16_t* in,
  57         const int16_t* filter_coeffs,
  58         int precision,
  59         int frac_pos,
  60         int filter_length,
  61         int length);
  62
  63 /**
  64  * Circularly convolve fixed vector with a phase dispersion impulse
  65  *        response filter (D.6.2 of G.729 and 6.1.5 of AMR).
  66  * @param fc_out vector with filter applied
  67  * @param fc_in source vector
  68  * @param filter phase filter coefficients
  69  *
  70  *  fc_out[n] = sum(i,0,len-1){ fc_in[i] * filter[(len + n - i)%len] }
  71  *
  72  * \note fc_in and fc_out should not overlap!
  73  */
  74 void ff_acelp_convolve_circ(
  75         int16_t* fc_out,
  76         const int16_t* fc_in,
  77         const int16_t* filter,
  78         int len);
  79
  80 /**
  81  * LP synthesis filter.
  82  * @param out [out] pointer to output buffer
  83  * @param filter_coeffs filter coefficients (-0x8000 <= (3.12) < 0x8000)
  84  * @param in input signal
  85  * @param buffer_length amount of data to process
  86  * @param filter_length filter length (10 for 10th order LP filter)
  87  * @param stop_on_overflow   1 - return immediately if overflow occurs
  88  *                           0 - ignore overflows
  89  * @param rounder the amount to add for rounding (usually 0x800 or 0xfff)
  90  *
  91  * @return 1 if overflow occurred, 0 - otherwise
  92  *
  93  * @note Output buffer must contain 10 samples of past
  94  *       speech data before pointer.
  95  *
  96  * Routine applies 1/A(z) filter to given speech data.
  97  */
  98 int ff_acelp_lp_synthesis_filter(
  99         int16_t *out,
 100         const int16_t* filter_coeffs,
 101         const int16_t* in,
 102         int buffer_length,
 103         int filter_length,
 104         int stop_on_overflow,
 105         int rounder);
 106
 107
 108 /**
 109  * high-pass filtering and upscaling (4.2.5 of G.729).
 110  * @param out [out] output buffer for filtered speech data
 111  * @param hpf_f [in/out] past filtered data from previous (2 items long)
 112  *                       frames (-0x20000000 <= (14.13) < 0x20000000)
 113  * @param in speech data to process
 114  * @param length input data size
 115  *
 116  * out[i] = 0.93980581 * in[i] - 1.8795834 * in[i-1] + 0.93980581 * in[i-2] +
 117  *          1.9330735 * out[i-1] - 0.93589199 * out[i-2]
 118  *
 119  * The filter has a cut-off frequency of 1/80 of the sampling freq
 120  *
 121  * @note Two items before the top of the out buffer must contain two items from the
 122  *       tail of the previous subframe.
 123  *
 124  * @remark It is safe to pass the same array in in and out parameters.
 125  *
 126  * @remark AMR uses mostly the same filter (cut-off frequency 60Hz, same formula,
 127  *         but constants differs in 5th sign after comma). Fortunately in
 128  *         fixed-point all coefficients are the same as in G.729. Thus this
 129  *         routine can be used for the fixed-point AMR decoder, too.
 130  */
 131 void ff_acelp_high_pass_filter(
 132         int16_t* out,
 133         int hpf_f[2],
 134         const int16_t* in,
 135         int length);
 136
 137 #endif /* AVCODEC_ACELP_FILTERS_H */