annotate acelp_filters.h @ 7051:4f1331b0d428 libavcodec

simplify
author michael
date Mon, 16 Jun 2008 23:31:23 +0000
parents 94465a2c3b34
children 2b763a495c07
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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 FIR (Finite Impulse Response) filter coefficients
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30 *
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31 * A similar filter is named b30 in G.729.
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32 *
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33 * G.729 specification says:
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34 * b30 is based on Hamming windowed sinc functions, truncated at +/-29 and
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35 * padded with zeros at +/-30 b30[30]=0.
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36 * The filter has a cut-off frequency (-3 dB) at 3600 Hz in the oversampled
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37 * domain.
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38 *
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39 * After some analysis, I found this approximation:
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40 *
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41 * PI * x
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42 * Hamm(x,N) = 0.53836-0.46164*cos(--------)
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43 * N-1
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44 * ---
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45 * 2
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46 *
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47 * PI * x
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48 * Hamm'(x,k) = Hamm(x - k, 2*k+1) = 0.53836 + 0.46164*cos(--------)
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49 * k
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50 *
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51 * sin(PI * x)
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52 * Sinc(x) = ----------- (normalized sinc function)
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53 * PI * x
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54 *
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55 * h(t,B) = 2 * B * Sinc(2 * B * t) (impulse response of sinc low-pass filter)
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56 *
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57 * b(k,B, n) = Hamm'(n, k) * h(n, B)
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58 *
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59 *
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60 * 3600
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61 * B = ----
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62 * 8000
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63 *
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64 * 3600 - cut-off frequency
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65 * 8000 - sampling rate
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66 * k - filter order
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67 *
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68 * ff_acelp_interp_filter[6*i+j] = b(10, 3600/8000, i+j/6)
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69 *
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70 * The filter assumes the following order of fractions (X - integer delay):
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71 *
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72 * 1/3 precision: X 1/3 2/3 X 1/3 2/3 X
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73 * 1/6 precision: X 1/6 2/6 3/6 4/6 5/6 X 1/6 2/6 3/6 4/6 5/6 X
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74 *
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75 * The filter can be used for 1/3 precision, too, by
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76 * passing 2*pitch_delay_frac as third parameter to the interpolation routine.
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77 *
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78 */
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79 extern const int16_t ff_acelp_interp_filter[61];
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80
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81 /**
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82 * \brief Generic interpolation routine
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83 * \param out [out] buffer for interpolated data
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84 * \param in input data
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85 * \param filter_coeffs interpolation filter coefficients (0.15)
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86 * \param precision filter is able to interpolate with 1/precision precision of pitch delay
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87 * \param pitch_delay_frac pitch delay, fractional part [0..precision-1]
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88 * \param filter_length filter length
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89 * \param length length of speech data to process
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90 *
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91 * filter_coeffs contains coefficients of the positive half of the symmetric
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92 * interpolation filter. filter_coeffs[0] should the central (unpaired) coefficient.
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93 * See ff_acelp_interp_filter fot example.
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94 *
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95 */
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96 void ff_acelp_interpolate(
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97 int16_t* out,
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98 const int16_t* in,
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99 const int16_t* filter_coeffs,
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100 int precision,
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101 int pitch_delay_frac,
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102 int filter_length,
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103 int length);
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104
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105 /**
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106 * \brief Circularly convolve fixed vector with a phase dispersion impulse
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107 * response filter (D.6.2 of G.729 and 6.1.5 of AMR).
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108 * \param fc_out vector with filter applied
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109 * \param fc_in source vector
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110 * \param filter phase filter coefficients
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111 *
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112 * fc_out[n] = sum(i,0,len-1){ fc_in[i] * filter[(len + n - i)%len] }
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113 *
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114 * \note fc_in and fc_out should not overlap!
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115 */
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116 void ff_acelp_convolve_circ(
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117 int16_t* fc_out,
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118 const int16_t* fc_in,
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119 const int16_t* filter,
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120 int subframe_size);
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121
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122 /**
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123 * \brief LP synthesis filter
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124 * \param out [out] pointer to output buffer
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125 * \param filter_coeffs filter coefficients (-0x8000 <= (3.12) < 0x8000)
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126 * \param in input signal
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127 * \param buffer_length amount of data to process
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128 * \param filter_length filter length (11 for 10th order LP filter)
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129 * \param stop_on_overflow 1 - return immediately if overflow occurs
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130 * 0 - ignore overflows
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131 *
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132 * \return 1 if overflow occurred, 0 - otherwise
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133 *
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134 * \note Output buffer must contain 10 samples of past
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135 * speech data before pointer.
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136 *
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137 * Routine applies 1/A(z) filter to given speech data.
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138 */
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139 int ff_acelp_lp_synthesis_filter(
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140 int16_t *out,
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141 const int16_t* filter_coeffs,
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142 const int16_t* in,
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143 int buffer_length,
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144 int filter_length,
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145 int stop_on_overflow);
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146
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147 /**
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148 * \brief Calculates coefficients of weighted A(z/weight) filter.
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149 * \param out [out] weighted A(z/weight) result
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150 * filter (-0x8000 <= (3.12) < 0x8000)
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151 * \param in source filter (-0x8000 <= (3.12) < 0x8000)
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152 * \param weight_pow array containing weight^i (-0x8000 <= (0.15) < 0x8000)
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153 * \param filter_length filter length (11 for 10th order LP filter)
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154 *
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155 * out[i]=weight_pow[i]*in[i] , i=0..9
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156 */
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157 void ff_acelp_weighted_filter(
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158 int16_t *out,
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159 const int16_t* in,
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160 const int16_t *weight_pow,
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161 int filter_length);
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162
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163 /**
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164 * \brief high-pass filtering and upscaling (4.2.5 of G.729)
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165 * \param out [out] output buffer for filtered speech data
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166 * \param hpf_f [in/out] past filtered data from previous (2 items long)
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167 * frames (-0x20000000 <= (14.13) < 0x20000000)
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168 * \param in speech data to process
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169 * \param length input data size
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170 *
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171 * out[i] = 0.93980581 * in[i] - 1.8795834 * in[i-1] + 0.93980581 * in[i-2] +
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172 * 1.9330735 * out[i-1] - 0.93589199 * out[i-2]
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173 *
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174 * The filter has a cut-off frequency of 100Hz
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175 *
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176 * \note Two items before the top of the out buffer must contain two items from the
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177 * tail of the previous subframe.
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178 *
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179 * \remark It is safe to pass the same array in in and out parameters.
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180 *
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181 * \remark AMR uses mostly the same filter (cut-off frequency 60Hz, same formula,
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182 * but constants differs in 5th sign after comma). Fortunately in
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183 * fixed-point all coefficients are the same as in G.729. Thus this
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184 * routine can be used for the fixed-point AMR decoder, too.
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185 */
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186 void ff_acelp_high_pass_filter(
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187 int16_t* out,
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188 int hpf_f[2],
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189 const int16_t* in,
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190 int length);
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191
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192 #endif /* FFMPEG_ACELP_FILTERS_H */