Mercurial > libavcodec.hg
annotate celp_filters.h @ 8629:04423b2f6e0b libavcodec
cosmetics: Remove pointless period after copyright statement non-sentences.
author | diego |
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date | Mon, 19 Jan 2009 15:46:40 +0000 |
parents | b9678ea8e5fb |
children | 2838045383c5 |
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 */ |
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39 void ff_celp_convolve_circ( |
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40 int16_t* fc_out, |
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41 const int16_t* fc_in, |
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42 const int16_t* filter, |
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43 int len); |
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44 |
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45 /** |
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46 * LP synthesis filter. |
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47 * @param out [out] pointer to output buffer |
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48 * @param filter_coeffs filter coefficients (-0x8000 <= (3.12) < 0x8000) |
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49 * @param in input signal |
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50 * @param buffer_length amount of data to process |
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51 * @param filter_length filter length (10 for 10th order LP filter) |
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52 * @param stop_on_overflow 1 - return immediately if overflow occurs |
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53 * 0 - ignore overflows |
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54 * @param rounder the amount to add for rounding (usually 0x800 or 0xfff) |
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55 * |
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56 * @return 1 if overflow occurred, 0 - otherwise |
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57 * |
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58 * @note Output buffer must contain filter_length samples of past |
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59 * speech data before pointer. |
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60 * |
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61 * Routine applies 1/A(z) filter to given speech data. |
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62 */ |
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63 int ff_celp_lp_synthesis_filter( |
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64 int16_t *out, |
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65 const int16_t* filter_coeffs, |
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66 const int16_t* in, |
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67 int buffer_length, |
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68 int filter_length, |
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69 int stop_on_overflow, |
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70 int rounder); |
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71 |
8091 | 72 /** |
73 * LP synthesis filter. | |
74 * @param out [out] pointer to output buffer | |
75 * - the array out[-filter_length, -1] must | |
76 * contain the previous result of this filter | |
77 * @param filter_coeffs filter coefficients. | |
78 * @param in input signal | |
79 * @param buffer_length amount of data to process | |
80 * @param filter_length filter length (10 for 10th order LP filter) | |
81 * | |
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82 * @note Output buffer must contain filter_length samples of past |
8091 | 83 * speech data before pointer. |
84 * | |
85 * Routine applies 1/A(z) filter to given speech data. | |
86 */ | |
87 void ff_celp_lp_synthesis_filterf( | |
88 float *out, | |
89 const float* filter_coeffs, | |
90 const float* in, | |
91 int buffer_length, | |
92 int filter_length); | |
93 | |
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94 #endif /* AVCODEC_CELP_FILTERS_H */ |