Mercurial > libavcodec.hg
annotate acelp_filters.h @ 7787:681a05d9b04f libavcodec
Ignore the spec, and calculate aspect ratio on width/height.
Fixes issue562 and issue621.
author | michael |
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date | Wed, 03 Sep 2008 19:04:46 +0000 |
parents | c4a4495715dd |
children | 611a21e4b01b |
rev | line source |
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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 */ |
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54 void ff_acelp_interpolate( |
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55 int16_t* out, |
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56 const int16_t* in, |
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57 const int16_t* filter_coeffs, |
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58 int precision, |
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59 int frac_pos, |
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60 int filter_length, |
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61 int length); |
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62 |
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63 /** |
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64 * Circularly convolve fixed vector with a phase dispersion impulse |
6772 | 65 * response filter (D.6.2 of G.729 and 6.1.5 of AMR). |
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66 * @param fc_out vector with filter applied |
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67 * @param fc_in source vector |
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68 * @param filter phase filter coefficients |
6772 | 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, | |
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78 int len); |
6772 | 79 |
80 /** | |
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81 * LP synthesis filter. |
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82 * @param out [out] pointer to output buffer |
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83 * @param filter_coeffs filter coefficients (-0x8000 <= (3.12) < 0x8000) |
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84 * @param in input signal |
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85 * @param buffer_length amount of data to process |
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86 * @param filter_length filter length (10 for 10th order LP filter) |
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87 * @param stop_on_overflow 1 - return immediately if overflow occurs |
6772 | 88 * 0 - ignore overflows |
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89 * @param rounder the amount to add for rounding (usually 0x800 or 0xfff) |
6772 | 90 * |
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91 * @return 1 if overflow occurred, 0 - otherwise |
6772 | 92 * |
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93 * @note Output buffer must contain 10 samples of past |
6772 | 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, | |
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104 int stop_on_overflow, |
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105 int rounder); |
6772 | 106 |
107 | |
108 /** | |
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109 * high-pass filtering and upscaling (4.2.5 of G.729). |
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110 * @param out [out] output buffer for filtered speech data |
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111 * @param hpf_f [in/out] past filtered data from previous (2 items long) |
6772 | 112 * frames (-0x20000000 <= (14.13) < 0x20000000) |
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113 * @param in speech data to process |
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114 * @param length input data size |
6772 | 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 * | |
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119 * The filter has a cut-off frequency of 1/80 of the sampling freq |
6772 | 120 * |
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121 * @note Two items before the top of the out buffer must contain two items from the |
6772 | 122 * tail of the previous subframe. |
123 * | |
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124 * @remark It is safe to pass the same array in in and out parameters. |
6772 | 125 * |
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126 * @remark AMR uses mostly the same filter (cut-off frequency 60Hz, same formula, |
6772 | 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 | |
7760 | 137 #endif /* AVCODEC_ACELP_FILTERS_H */ |