annotate acelp_filters.h @ 7650:1e4ae5de68a4 libavcodec

Rename subframe_size to a name that is not specific to some specific use of ff_acelp_convolve_circ()
author michael
date Thu, 21 Aug 2008 22:41:20 +0000
parents 4df1c17c034c
children de64f2e14b13
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1 /*
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2 * various filters for ACELP-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 FFMPEG_ACELP_FILTERS_H
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24 #define FFMPEG_ACELP_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 * low-pass Finite Impulse Response filter coefficients.
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30 *
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31 * Hamming windowed sinc filter with cutoff freq 3/40 of the sampling freq,
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32 * the coefficients are scaled by 2^15.
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33 * This array only contains the right half of the filter.
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34 * This filter is likely identical to the one used in G.729, though this
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35 * could not be determined from the original comments with certainity.
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36 */
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37 extern const int16_t ff_acelp_interp_filter[61];
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38
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39 /**
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40 * Generic FIR interpolation routine.
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41 * @param out [out] buffer for interpolated data
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42 * @param in input data
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43 * @param filter_coeffs interpolation filter coefficients (0.15)
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44 * @param precision sub sample factor, that is the precision of the position
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45 * @param frac_pos fractional part of position [0..precision-1]
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46 * @param filter_length filter length
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47 * @param length length of output
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48 *
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49 * filter_coeffs contains coefficients of the right half of the symmetric
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50 * interpolation filter. filter_coeffs[0] should the central (unpaired) coefficient.
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51 * See ff_acelp_interp_filter for an example.
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52 *
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53 */
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54 void ff_acelp_interpolate(
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55 int16_t* out,
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56 const int16_t* in,
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57 const int16_t* filter_coeffs,
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58 int precision,
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59 int frac_pos,
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60 int filter_length,
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61 int length);
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62
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63 /**
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64 * Circularly convolve fixed vector with a phase dispersion impulse
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65 * response filter (D.6.2 of G.729 and 6.1.5 of AMR).
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66 * @param fc_out vector with filter applied
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67 * @param fc_in source vector
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68 * @param filter phase filter coefficients
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69 *
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70 * fc_out[n] = sum(i,0,len-1){ fc_in[i] * filter[(len + n - i)%len] }
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71 *
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72 * \note fc_in and fc_out should not overlap!
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73 */
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74 void ff_acelp_convolve_circ(
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75 int16_t* fc_out,
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76 const int16_t* fc_in,
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77 const int16_t* filter,
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78 int len);
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79
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80 /**
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81 * LP synthesis filter.
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82 * @param out [out] pointer to output buffer
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83 * @param filter_coeffs filter coefficients (-0x8000 <= (3.12) < 0x8000)
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84 * @param in input signal
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85 * @param buffer_length amount of data to process
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86 * @param filter_length filter length (10 for 10th order LP filter)
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87 * @param stop_on_overflow 1 - return immediately if overflow occurs
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88 * 0 - ignore overflows
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89 * @param rounder the amount to add for rounding (usually 0x800 or 0xfff)
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90 *
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91 * @return 1 if overflow occurred, 0 - otherwise
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92 *
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93 * @note Output buffer must contain 10 samples of past
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94 * speech data before pointer.
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95 *
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96 * Routine applies 1/A(z) filter to given speech data.
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97 */
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98 int ff_acelp_lp_synthesis_filter(
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99 int16_t *out,
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100 const int16_t* filter_coeffs,
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101 const int16_t* in,
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102 int buffer_length,
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103 int filter_length,
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104 int stop_on_overflow,
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105 int rounder);
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106
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107 /**
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108 * Calculates coefficients of weighted A(z/weight) filter.
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109 * @param out [out] weighted A(z/weight) result
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110 * filter (-0x8000 <= (3.12) < 0x8000)
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111 * @param in source filter (-0x8000 <= (3.12) < 0x8000)
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112 * @param weight_pow array containing weight^i (-0x8000 <= (0.15) < 0x8000)
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113 * @param filter_length filter length (11 for 10th order LP filter)
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114 *
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115 * out[i]=weight_pow[i]*in[i] , i=0..9
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116 */
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117 void ff_acelp_weighted_filter(
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118 int16_t *out,
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119 const int16_t* in,
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120 const int16_t *weight_pow,
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121 int filter_length);
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122
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123 /**
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124 * high-pass filtering and upscaling (4.2.5 of G.729).
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125 * @param out [out] output buffer for filtered speech data
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126 * @param hpf_f [in/out] past filtered data from previous (2 items long)
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127 * frames (-0x20000000 <= (14.13) < 0x20000000)
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128 * @param in speech data to process
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129 * @param length input data size
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130 *
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131 * out[i] = 0.93980581 * in[i] - 1.8795834 * in[i-1] + 0.93980581 * in[i-2] +
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132 * 1.9330735 * out[i-1] - 0.93589199 * out[i-2]
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133 *
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134 * The filter has a cut-off frequency of 100Hz
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135 *
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136 * @note Two items before the top of the out buffer must contain two items from the
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137 * tail of the previous subframe.
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138 *
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139 * @remark It is safe to pass the same array in in and out parameters.
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140 *
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141 * @remark AMR uses mostly the same filter (cut-off frequency 60Hz, same formula,
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142 * but constants differs in 5th sign after comma). Fortunately in
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143 * fixed-point all coefficients are the same as in G.729. Thus this
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144 * routine can be used for the fixed-point AMR decoder, too.
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145 */
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146 void ff_acelp_high_pass_filter(
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147 int16_t* out,
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148 int hpf_f[2],
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149 const int16_t* in,
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150 int length);
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151
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152 #endif /* FFMPEG_ACELP_FILTERS_H */