Mercurial > libavcodec.hg
annotate acelp_filters.h @ 7642:1fbf9b2060ce libavcodec
Make doxygen comments consistent with the rest of FFmpeg.
author | michael |
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date | Thu, 21 Aug 2008 21:37:53 +0000 |
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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 | |
23 #ifndef FFMPEG_ACELP_FILTERS_H | |
24 #define FFMPEG_ACELP_FILTERS_H | |
25 | |
6796 | 26 #include <stdint.h> |
27 | |
6772 | 28 /** |
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29 * low-pass FIR (Finite Impulse Response) filter coefficients |
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30 * |
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31 * A similar filter is named b30 in G.729. |
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32 * |
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33 * G.729 specification says: |
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34 * b30 is based on Hamming windowed sinc functions, truncated at +/-29 and |
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35 * padded with zeros at +/-30 b30[30]=0. |
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36 * The filter has a cut-off frequency (-3 dB) at 3600 Hz in the oversampled |
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37 * domain. |
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38 * |
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39 * After some analysis, I found this approximation: |
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40 * |
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41 * PI * x |
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42 * Hamm(x,N) = 0.53836-0.46164*cos(--------) |
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43 * N-1 |
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44 * --- |
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45 * 2 |
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46 * |
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47 * PI * x |
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48 * Hamm'(x,k) = Hamm(x - k, 2*k+1) = 0.53836 + 0.46164*cos(--------) |
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49 * k |
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50 * |
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51 * sin(PI * x) |
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52 * Sinc(x) = ----------- (normalized sinc function) |
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53 * PI * x |
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54 * |
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55 * h(t,B) = 2 * B * Sinc(2 * B * t) (impulse response of sinc low-pass filter) |
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56 * |
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57 * b(k,B, n) = Hamm'(n, k) * h(n, B) |
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58 * |
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59 * |
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60 * 3600 |
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61 * B = ---- |
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62 * 8000 |
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63 * |
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64 * 3600 - cut-off frequency |
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65 * 8000 - sampling rate |
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66 * k - filter order |
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67 * |
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68 * ff_acelp_interp_filter[6*i+j] = b(10, 3600/8000, i+j/6) |
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69 * |
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70 * The filter assumes the following order of fractions (X - integer delay): |
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71 * |
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72 * 1/3 precision: X 1/3 2/3 X 1/3 2/3 X |
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73 * 1/6 precision: X 1/6 2/6 3/6 4/6 5/6 X 1/6 2/6 3/6 4/6 5/6 X |
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74 * |
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75 * The filter can be used for 1/3 precision, too, by |
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76 * passing 2*pitch_delay_frac as third parameter to the interpolation routine. |
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77 * |
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78 */ |
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79 extern const int16_t ff_acelp_interp_filter[61]; |
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80 |
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81 /** |
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82 * Generic interpolation routine. |
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83 * @param out [out] buffer for interpolated data |
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84 * @param in input data |
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85 * @param filter_coeffs interpolation filter coefficients (0.15) |
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86 * @param precision filter is able to interpolate with 1/precision precision of pitch delay |
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87 * @param pitch_delay_frac pitch delay, fractional part [0..precision-1] |
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88 * @param filter_length filter length |
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89 * @param length length of speech data to process |
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90 * |
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91 * filter_coeffs contains coefficients of the positive half of the symmetric |
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92 * interpolation filter. filter_coeffs[0] should the central (unpaired) coefficient. |
7249 | 93 * See ff_acelp_interp_filter for an example. |
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94 * |
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95 */ |
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96 void ff_acelp_interpolate( |
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97 int16_t* out, |
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98 const int16_t* in, |
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99 const int16_t* filter_coeffs, |
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100 int precision, |
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101 int pitch_delay_frac, |
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102 int filter_length, |
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103 int length); |
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104 |
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105 /** |
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106 * Circularly convolve fixed vector with a phase dispersion impulse |
6772 | 107 * response filter (D.6.2 of G.729 and 6.1.5 of AMR). |
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108 * @param fc_out vector with filter applied |
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109 * @param fc_in source vector |
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110 * @param filter phase filter coefficients |
6772 | 111 * |
112 * fc_out[n] = sum(i,0,len-1){ fc_in[i] * filter[(len + n - i)%len] } | |
113 * | |
114 * \note fc_in and fc_out should not overlap! | |
115 */ | |
116 void ff_acelp_convolve_circ( | |
117 int16_t* fc_out, | |
118 const int16_t* fc_in, | |
119 const int16_t* filter, | |
120 int subframe_size); | |
121 | |
122 /** | |
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123 * LP synthesis filter. |
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124 * @param out [out] pointer to output buffer |
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125 * @param filter_coeffs filter coefficients (-0x8000 <= (3.12) < 0x8000) |
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126 * @param in input signal |
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127 * @param buffer_length amount of data to process |
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128 * @param filter_length filter length (10 for 10th order LP filter) |
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129 * @param stop_on_overflow 1 - return immediately if overflow occurs |
6772 | 130 * 0 - ignore overflows |
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131 * @param rounder the amount to add for rounding (usually 0x800 or 0xfff) |
6772 | 132 * |
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133 * @return 1 if overflow occurred, 0 - otherwise |
6772 | 134 * |
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135 * @note Output buffer must contain 10 samples of past |
6772 | 136 * speech data before pointer. |
137 * | |
138 * Routine applies 1/A(z) filter to given speech data. | |
139 */ | |
140 int ff_acelp_lp_synthesis_filter( | |
141 int16_t *out, | |
142 const int16_t* filter_coeffs, | |
143 const int16_t* in, | |
144 int buffer_length, | |
145 int filter_length, | |
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146 int stop_on_overflow, |
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147 int rounder); |
6772 | 148 |
149 /** | |
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150 * Calculates coefficients of weighted A(z/weight) filter. |
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151 * @param out [out] weighted A(z/weight) result |
6772 | 152 * filter (-0x8000 <= (3.12) < 0x8000) |
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153 * @param in source filter (-0x8000 <= (3.12) < 0x8000) |
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154 * @param weight_pow array containing weight^i (-0x8000 <= (0.15) < 0x8000) |
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155 * @param filter_length filter length (11 for 10th order LP filter) |
6772 | 156 * |
157 * out[i]=weight_pow[i]*in[i] , i=0..9 | |
158 */ | |
159 void ff_acelp_weighted_filter( | |
160 int16_t *out, | |
161 const int16_t* in, | |
162 const int16_t *weight_pow, | |
163 int filter_length); | |
164 | |
165 /** | |
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166 * high-pass filtering and upscaling (4.2.5 of G.729). |
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167 * @param out [out] output buffer for filtered speech data |
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168 * @param hpf_f [in/out] past filtered data from previous (2 items long) |
6772 | 169 * frames (-0x20000000 <= (14.13) < 0x20000000) |
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170 * @param in speech data to process |
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171 * @param length input data size |
6772 | 172 * |
173 * out[i] = 0.93980581 * in[i] - 1.8795834 * in[i-1] + 0.93980581 * in[i-2] + | |
174 * 1.9330735 * out[i-1] - 0.93589199 * out[i-2] | |
175 * | |
176 * The filter has a cut-off frequency of 100Hz | |
177 * | |
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178 * @note Two items before the top of the out buffer must contain two items from the |
6772 | 179 * tail of the previous subframe. |
180 * | |
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181 * @remark It is safe to pass the same array in in and out parameters. |
6772 | 182 * |
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183 * @remark AMR uses mostly the same filter (cut-off frequency 60Hz, same formula, |
6772 | 184 * but constants differs in 5th sign after comma). Fortunately in |
185 * fixed-point all coefficients are the same as in G.729. Thus this | |
186 * routine can be used for the fixed-point AMR decoder, too. | |
187 */ | |
188 void ff_acelp_high_pass_filter( | |
189 int16_t* out, | |
190 int hpf_f[2], | |
191 const int16_t* in, | |
192 int length); | |
193 | |
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194 #endif /* FFMPEG_ACELP_FILTERS_H */ |