Mercurial > libavcodec.hg
annotate celp_filters.h @ 10612:d6860312274c libavcodec
Support ASCII pnms.
Implements issue1452.
author | michael |
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date | Tue, 01 Dec 2009 16:56:13 +0000 |
parents | d35904b4fe3f |
children | d124d9b688d0 |
rev | line source |
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1 /* |
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2 * various filters for CELP-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_CELP_FILTERS_H |
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24 #define AVCODEC_CELP_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 * Circularly convolve fixed vector with a phase dispersion impulse |
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30 * response filter (D.6.2 of G.729 and 6.1.5 of AMR). |
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31 * @param fc_out vector with filter applied |
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32 * @param fc_in source vector |
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33 * @param filter phase filter coefficients |
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34 * |
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35 * fc_out[n] = sum(i,0,len-1){ fc_in[i] * filter[(len + n - i)%len] } |
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36 * |
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37 * \note fc_in and fc_out should not overlap! |
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38 */ |
10003 | 39 void ff_celp_convolve_circ(int16_t* fc_out, |
40 const int16_t* fc_in, | |
41 const int16_t* filter, | |
42 int len); | |
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43 |
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44 /** |
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45 * Add an array to a rotated array. |
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46 * |
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47 * out[k] = in[k] + fac * lagged[k-lag] with wrap-around |
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48 * |
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49 * @param out result vector |
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50 * @param in samples to be added unfiltered |
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51 * @param lagged samples to be rotated, multiplied and added |
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52 * @param lag lagged vector delay in the range [0, n] |
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53 * @param fac scalefactor for lagged samples |
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54 * @param n number of samples |
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55 */ |
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56 void ff_celp_circ_addf(float *out, const float *in, |
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57 const float *lagged, int lag, float fac, int n); |
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58 |
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59 /** |
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60 * LP synthesis filter. |
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61 * @param out [out] pointer to output buffer |
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62 * @param filter_coeffs filter coefficients (-0x8000 <= (3.12) < 0x8000) |
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63 * @param in input signal |
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64 * @param buffer_length amount of data to process |
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65 * @param filter_length filter length (10 for 10th order LP filter) |
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66 * @param stop_on_overflow 1 - return immediately if overflow occurs |
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67 * 0 - ignore overflows |
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68 * @param rounder the amount to add for rounding (usually 0x800 or 0xfff) |
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69 * |
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70 * @return 1 if overflow occurred, 0 - otherwise |
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71 * |
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72 * @note Output buffer must contain filter_length samples of past |
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73 * speech data before pointer. |
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74 * |
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75 * Routine applies 1/A(z) filter to given speech data. |
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76 */ |
10003 | 77 int ff_celp_lp_synthesis_filter(int16_t *out, |
78 const int16_t* filter_coeffs, | |
79 const int16_t* in, | |
80 int buffer_length, | |
81 int filter_length, | |
82 int stop_on_overflow, | |
83 int rounder); | |
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84 |
8091 | 85 /** |
86 * LP synthesis filter. | |
87 * @param out [out] pointer to output buffer | |
88 * - the array out[-filter_length, -1] must | |
89 * contain the previous result of this filter | |
90 * @param filter_coeffs filter coefficients. | |
91 * @param in input signal | |
92 * @param buffer_length amount of data to process | |
93 * @param filter_length filter length (10 for 10th order LP filter) | |
94 * | |
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95 * @note Output buffer must contain filter_length samples of past |
8091 | 96 * speech data before pointer. |
97 * | |
98 * Routine applies 1/A(z) filter to given speech data. | |
99 */ | |
10003 | 100 void ff_celp_lp_synthesis_filterf(float *out, |
101 const float* filter_coeffs, | |
102 const float* in, | |
103 int buffer_length, | |
104 int filter_length); | |
8091 | 105 |
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106 /** |
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107 * LP zero synthesis filter. |
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108 * @param out [out] pointer to output buffer |
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109 * @param filter_coeffs filter coefficients. |
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110 * @param in input signal |
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111 * - the array in[-filter_length, -1] must |
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112 * contain the previous input of this filter |
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113 * @param buffer_length amount of data to process |
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114 * @param filter_length filter length (10 for 10th order LP filter) |
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115 * |
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116 * @note Output buffer must contain filter_length samples of past |
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117 * speech data before pointer. |
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118 * |
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119 * Routine applies A(z) filter to given speech data. |
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120 */ |
10003 | 121 void ff_celp_lp_zero_synthesis_filterf(float *out, |
122 const float* filter_coeffs, | |
123 const float* in, | |
124 int buffer_length, | |
125 int filter_length); | |
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126 |
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127 #endif /* AVCODEC_CELP_FILTERS_H */ |