annotate acelp_filters.h @ 11498:8e889ce0d616 libavcodec

100L, revert r22560, already present
author bcoudurier
date Mon, 15 Mar 2010 23:40:51 +0000
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1 /*
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2 * various filters for ACELP-based codecs
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3 *
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4 * Copyright (c) 2008 Vladimir Voroshilov
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5 *
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6 * This file is part of FFmpeg.
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7 *
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8 * FFmpeg is free software; you can redistribute it and/or
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9 * modify it under the terms of the GNU Lesser General Public
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10 * License as published by the Free Software Foundation; either
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11 * version 2.1 of the License, or (at your option) any later version.
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12 *
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13 * FFmpeg is distributed in the hope that it will be useful,
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14 * but WITHOUT ANY WARRANTY; without even the implied warranty of
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15 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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16 * Lesser General Public License for more details.
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17 *
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18 * You should have received a copy of the GNU Lesser General Public
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19 * License along with FFmpeg; if not, write to the Free Software
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20 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
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21 */
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22
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23 #ifndef AVCODEC_ACELP_FILTERS_H
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24 #define AVCODEC_ACELP_FILTERS_H
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25
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26 #include <stdint.h>
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27
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28 /**
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29 * low-pass Finite Impulse Response filter coefficients.
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30 *
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31 * Hamming windowed sinc filter with cutoff freq 3/40 of the sampling freq,
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32 * the coefficients are scaled by 2^15.
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33 * This array only contains the right half of the filter.
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34 * This filter is likely identical to the one used in G.729, though this
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35 * could not be determined from the original comments with certainity.
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36 */
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37 extern const int16_t ff_acelp_interp_filter[61];
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38
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39 /**
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40 * Generic FIR interpolation routine.
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41 * @param out [out] buffer for interpolated data
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42 * @param in input data
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43 * @param filter_coeffs interpolation filter coefficients (0.15)
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44 * @param precision sub sample factor, that is the precision of the position
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45 * @param frac_pos fractional part of position [0..precision-1]
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46 * @param filter_length filter length
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47 * @param length length of output
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48 *
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49 * filter_coeffs contains coefficients of the right half of the symmetric
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50 * interpolation filter. filter_coeffs[0] should the central (unpaired) coefficient.
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51 * See ff_acelp_interp_filter for an example.
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52 *
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53 */
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54 void ff_acelp_interpolate(int16_t* out, const int16_t* in,
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55 const int16_t* filter_coeffs, int precision,
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56 int frac_pos, int filter_length, int length);
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57
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58 /**
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59 * Floating point version of ff_acelp_interpolate()
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60 */
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61 void ff_acelp_interpolatef(float *out, const float *in,
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62 const float *filter_coeffs, int precision,
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63 int frac_pos, int filter_length, int length);
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64
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65
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66 /**
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67 * high-pass filtering and upscaling (4.2.5 of G.729).
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68 * @param out [out] output buffer for filtered speech data
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69 * @param hpf_f [in/out] past filtered data from previous (2 items long)
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70 * frames (-0x20000000 <= (14.13) < 0x20000000)
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71 * @param in speech data to process
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72 * @param length input data size
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73 *
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74 * out[i] = 0.93980581 * in[i] - 1.8795834 * in[i-1] + 0.93980581 * in[i-2] +
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75 * 1.9330735 * out[i-1] - 0.93589199 * out[i-2]
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76 *
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77 * The filter has a cut-off frequency of 1/80 of the sampling freq
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78 *
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79 * @note Two items before the top of the out buffer must contain two items from the
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80 * tail of the previous subframe.
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81 *
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82 * @remark It is safe to pass the same array in in and out parameters.
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83 *
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84 * @remark AMR uses mostly the same filter (cut-off frequency 60Hz, same formula,
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85 * but constants differs in 5th sign after comma). Fortunately in
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86 * fixed-point all coefficients are the same as in G.729. Thus this
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87 * routine can be used for the fixed-point AMR decoder, too.
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88 */
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89 void ff_acelp_high_pass_filter(int16_t* out, int hpf_f[2],
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90 const int16_t* in, int length);
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91
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92 /**
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93 * Apply an order 2 rational transfer function in-place.
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94 *
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95 * @param samples [in/out]
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96 * @param zero_coeffs z^-1 and z^-2 coefficients of the numerator
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97 * @param pole_coeffs z^-1 and z^-2 coefficients of the denominator
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98 * @param gain scale factor for final output
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99 * @param mem intermediate values used by filter (should be 0 initially)
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100 * @param n number of samples
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101 */
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102 void ff_acelp_apply_order_2_transfer_function(float *samples,
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103 const float zero_coeffs[2],
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104 const float pole_coeffs[2],
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105 float gain,
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106 float mem[2], int n);
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107
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108 /**
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109 * Apply tilt compensation filter, 1 - tilt * z-1.
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110 *
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111 * @param mem pointer to the filter's state (one single float)
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112 * @param tilt tilt factor
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113 * @param samples array where the filter is applied
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114 * @param size the size of the samples array
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115 */
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116 void ff_tilt_compensation(float *mem, float tilt, float *samples, int size);
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117
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118
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119 #endif /* AVCODEC_ACELP_FILTERS_H */