Mercurial > libavcodec.hg
annotate acelp_filters.h @ 11498:8e889ce0d616 libavcodec
100L, revert r22560, already present
author | bcoudurier |
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date | Mon, 15 Mar 2010 23:40:51 +0000 |
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6772 | 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 | |
7760 | 23 #ifndef AVCODEC_ACELP_FILTERS_H |
24 #define AVCODEC_ACELP_FILTERS_H | |
6772 | 25 |
6796 | 26 #include <stdint.h> |
27 | |
6772 | 28 /** |
7643 | 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. |
7643 | 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. | |
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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 /** |
7647 | 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. |
7249 | 51 * See ff_acelp_interp_filter for an example. |
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52 * |
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53 */ |
9792 | 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); | |
6772 | 57 |
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) |
6772 | 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 |
6772 | 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 * | |
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77 * The filter has a cut-off frequency of 1/80 of the sampling freq |
6772 | 78 * |
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79 * @note Two items before the top of the out buffer must contain two items from the |
6772 | 80 * tail of the previous subframe. |
81 * | |
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82 * @remark It is safe to pass the same array in in and out parameters. |
6772 | 83 * |
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84 * @remark AMR uses mostly the same filter (cut-off frequency 60Hz, same formula, |
6772 | 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 */ | |
9792 | 89 void ff_acelp_high_pass_filter(int16_t* out, int hpf_f[2], |
90 const int16_t* in, int length); | |
6772 | 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 |
7760 | 119 #endif /* AVCODEC_ACELP_FILTERS_H */ |