Mercurial > libavcodec.hg
annotate celp_filters.h @ 11300:1f512abdf680 libavcodec
Simplify code to set cbp_*
this seems 1 cpu cycle slower even though we practically just remove code.
Speed loss seems caused by the merge of if(left_type), iam commiting this
anyway as i cant imagine this to be anything but compiler messup.
author | michael |
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date | Fri, 26 Feb 2010 09:13:40 +0000 |
parents | 63451af5f8f9 |
children | 0885e7a93ed4 |
rev | line source |
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Split off celp_filters.[ch] from acelp_filters.[ch] for the QCELP decoder.
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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(int16_t *fc_out, const int16_t *fc_in, |
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40 const int16_t *filter, int len); |
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41 |
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42 /** |
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43 * Add an array to a rotated array. |
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44 * |
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45 * out[k] = in[k] + fac * lagged[k-lag] with wrap-around |
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46 * |
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47 * @param out result vector |
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48 * @param in samples to be added unfiltered |
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49 * @param lagged samples to be rotated, multiplied and added |
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50 * @param lag lagged vector delay in the range [0, n] |
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51 * @param fac scalefactor for lagged samples |
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52 * @param n number of samples |
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53 */ |
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54 void ff_celp_circ_addf(float *out, const float *in, |
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55 const float *lagged, int lag, float fac, int n); |
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56 |
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57 /** |
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58 * LP synthesis filter. |
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59 * @param out [out] pointer to output buffer |
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60 * @param filter_coeffs filter coefficients (-0x8000 <= (3.12) < 0x8000) |
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61 * @param in input signal |
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62 * @param buffer_length amount of data to process |
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63 * @param filter_length filter length (10 for 10th order LP filter) |
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64 * @param stop_on_overflow 1 - return immediately if overflow occurs |
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65 * 0 - ignore overflows |
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66 * @param rounder the amount to add for rounding (usually 0x800 or 0xfff) |
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67 * |
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68 * @return 1 if overflow occurred, 0 - otherwise |
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69 * |
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70 * @note Output buffer must contain filter_length samples of past |
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71 * speech data before pointer. |
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72 * |
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73 * Routine applies 1/A(z) filter to given speech data. |
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74 */ |
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75 int ff_celp_lp_synthesis_filter(int16_t *out, const int16_t *filter_coeffs, |
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76 const int16_t *in, int buffer_length, |
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77 int filter_length, int stop_on_overflow, |
10003 | 78 int rounder); |
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79 |
8091 | 80 /** |
81 * LP synthesis filter. | |
82 * @param out [out] pointer to output buffer | |
83 * - the array out[-filter_length, -1] must | |
84 * contain the previous result of this filter | |
85 * @param filter_coeffs filter coefficients. | |
86 * @param in input signal | |
87 * @param buffer_length amount of data to process | |
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88 * @param filter_length filter length (10 for 10th order LP filter). Must be |
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89 * greater than 4 and even. |
8091 | 90 * |
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91 * @note Output buffer must contain filter_length samples of past |
8091 | 92 * speech data before pointer. |
93 * | |
94 * Routine applies 1/A(z) filter to given speech data. | |
95 */ | |
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96 void ff_celp_lp_synthesis_filterf(float *out, const float *filter_coeffs, |
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97 const float *in, int buffer_length, |
10003 | 98 int filter_length); |
8091 | 99 |
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100 /** |
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101 * LP zero synthesis filter. |
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102 * @param out [out] pointer to output buffer |
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103 * @param filter_coeffs filter coefficients. |
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104 * @param in input signal |
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105 * - the array in[-filter_length, -1] must |
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106 * contain the previous input of this filter |
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107 * @param buffer_length amount of data to process |
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108 * @param filter_length filter length (10 for 10th order LP filter) |
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109 * |
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110 * @note Output buffer must contain filter_length samples of past |
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111 * speech data before pointer. |
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112 * |
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113 * Routine applies A(z) filter to given speech data. |
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114 */ |
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115 void ff_celp_lp_zero_synthesis_filterf(float *out, const float *filter_coeffs, |
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116 const float *in, int buffer_length, |
10003 | 117 int filter_length); |
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118 |
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119 #endif /* AVCODEC_CELP_FILTERS_H */ |