annotate celp_filters.h @ 10557:f2f4d6fe3f6d libavcodec

Make sorting function used in TwinVQ a shared function
author vitor
date Sun, 22 Nov 2009 22:25:58 +0000
parents d35904b4fe3f
children d124d9b688d0
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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,
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40 const int16_t* fc_in,
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41 const int16_t* filter,
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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 */
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77 int ff_celp_lp_synthesis_filter(int16_t *out,
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78 const int16_t* filter_coeffs,
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79 const int16_t* in,
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80 int buffer_length,
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81 int filter_length,
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82 int stop_on_overflow,
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83 int rounder);
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84
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85 /**
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86 * LP synthesis filter.
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87 * @param out [out] pointer to output buffer
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88 * - the array out[-filter_length, -1] must
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89 * contain the previous result of this filter
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90 * @param filter_coeffs filter coefficients.
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91 * @param in input signal
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92 * @param buffer_length amount of data to process
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93 * @param filter_length filter length (10 for 10th order LP filter)
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94 *
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95 * @note Output buffer must contain filter_length samples of past
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96 * speech data before pointer.
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97 *
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98 * Routine applies 1/A(z) filter to given speech data.
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99 */
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100 void ff_celp_lp_synthesis_filterf(float *out,
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101 const float* filter_coeffs,
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102 const float* in,
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103 int buffer_length,
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104 int filter_length);
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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 */
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121 void ff_celp_lp_zero_synthesis_filterf(float *out,
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122 const float* filter_coeffs,
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123 const float* in,
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124 int buffer_length,
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125 int filter_length);
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126
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127 #endif /* AVCODEC_CELP_FILTERS_H */