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(int16_t* out, const int16_t* in,
  55                           const int16_t* filter_coeffs, int precision,
  56                           int frac_pos, int filter_length, int length);
  57
  58 /**
  59  * Floating point version of ff_acelp_interpolate()
  60  */
  61 void ff_acelp_interpolatef(float *out, const float *in,
  62                            const float *filter_coeffs, int precision,
  63                            int frac_pos, int filter_length, int length);
  64
  65
  66 /**
  67  * high-pass filtering and upscaling (4.2.5 of G.729).
  68  * @param out [out] output buffer for filtered speech data
  69  * @param hpf_f [in/out] past filtered data from previous (2 items long)
  70  *                       frames (-0x20000000 <= (14.13) < 0x20000000)
  71  * @param in speech data to process
  72  * @param length input data size
  73  *
  74  * out[i] = 0.93980581 * in[i] - 1.8795834 * in[i-1] + 0.93980581 * in[i-2] +
  75  *          1.9330735 * out[i-1] - 0.93589199 * out[i-2]
  76  *
  77  * The filter has a cut-off frequency of 1/80 of the sampling freq
  78  *
  79  * @note Two items before the top of the out buffer must contain two items from the
  80  *       tail of the previous subframe.
  81  *
  82  * @remark It is safe to pass the same array in in and out parameters.
  83  *
  84  * @remark AMR uses mostly the same filter (cut-off frequency 60Hz, same formula,
  85  *         but constants differs in 5th sign after comma). Fortunately in
  86  *         fixed-point all coefficients are the same as in G.729. Thus this
  87  *         routine can be used for the fixed-point AMR decoder, too.
  88  */
  89 void ff_acelp_high_pass_filter(int16_t* out, int hpf_f[2],
  90                                const int16_t* in, int length);
  91
  92 /**
  93  * Apply an order 2 rational transfer function in-place.
  94  *
  95  * @param samples [in/out]
  96  * @param zero_coeffs z^-1 and z^-2 coefficients of the numerator
  97  * @param pole_coeffs z^-1 and z^-2 coefficients of the denominator
  98  * @param gain scale factor for final output
  99  * @param mem intermediate values used by filter (should be 0 initially)
 100  * @param n number of samples
 101  */
 102 void ff_acelp_apply_order_2_transfer_function(float *samples,
 103                                               const float zero_coeffs[2],
 104                                               const float pole_coeffs[2],
 105                                               float gain,
 106                                               float mem[2], int n);
 107
 108 /**
 109  * Apply tilt compensation filter, 1 - tilt * z-1.
 110  *
 111  * @param mem pointer to the filter's state (one single float)
 112  * @param tilt tilt factor
 113  * @param samples array where the filter is applied
 114  * @param size the size of the samples array
 115  */
 116 void ff_tilt_compensation(float *mem, float tilt, float *samples, int size);
 117
 118
 119 #endif /* AVCODEC_ACELP_FILTERS_H */