Mercurial > libavcodec.hg
annotate truespeech.c @ 11961:d7808ddcbcee libavcodec
lsp: convert variable-length arrays to fixed size
Max LP order is defined to be 16, fixed-size buffers are OK.
author | mru |
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date | Sat, 26 Jun 2010 14:34:12 +0000 |
parents | 7dd2a45249a9 |
children | 41f078d6869d |
rev | line source |
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1 /* |
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2 * DSP Group TrueSpeech compatible decoder |
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3 * Copyright (c) 2005 Konstantin Shishkov |
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4 * |
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5 * This file is part of FFmpeg. |
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6 * |
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7 * FFmpeg is free software; you can redistribute it and/or |
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8 * modify it under the terms of the GNU Lesser General Public |
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9 * License as published by the Free Software Foundation; either |
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10 * version 2.1 of the License, or (at your option) any later version. |
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11 * |
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12 * FFmpeg is distributed in the hope that it will be useful, |
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13 * but WITHOUT ANY WARRANTY; without even the implied warranty of |
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14 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU |
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15 * Lesser General Public License for more details. |
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16 * |
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17 * You should have received a copy of the GNU Lesser General Public |
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18 * License along with FFmpeg; if not, write to the Free Software |
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19 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA |
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20 */ |
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21 |
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22 #include "libavutil/intreadwrite.h" |
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23 #include "avcodec.h" |
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24 |
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25 #include "truespeech_data.h" |
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26 /** |
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27 * @file |
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28 * TrueSpeech decoder. |
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29 */ |
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30 |
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31 /** |
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32 * TrueSpeech decoder context |
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33 */ |
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34 typedef struct { |
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35 /* input data */ |
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36 int16_t vector[8]; //< input vector: 5/5/4/4/4/3/3/3 |
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37 int offset1[2]; //< 8-bit value, used in one copying offset |
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38 int offset2[4]; //< 7-bit value, encodes offsets for copying and for two-point filter |
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39 int pulseoff[4]; //< 4-bit offset of pulse values block |
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40 int pulsepos[4]; //< 27-bit variable, encodes 7 pulse positions |
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41 int pulseval[4]; //< 7x2-bit pulse values |
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42 int flag; //< 1-bit flag, shows how to choose filters |
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43 /* temporary data */ |
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44 int filtbuf[146]; // some big vector used for storing filters |
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45 int prevfilt[8]; // filter from previous frame |
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46 int16_t tmp1[8]; // coefficients for adding to out |
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47 int16_t tmp2[8]; // coefficients for adding to out |
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48 int16_t tmp3[8]; // coefficients for adding to out |
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49 int16_t cvector[8]; // correlated input vector |
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50 int filtval; // gain value for one function |
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51 int16_t newvec[60]; // tmp vector |
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52 int16_t filters[32]; // filters for every subframe |
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53 } TSContext; |
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54 |
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55 static av_cold int truespeech_decode_init(AVCodecContext * avctx) |
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56 { |
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57 // TSContext *c = avctx->priv_data; |
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58 |
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59 avctx->sample_fmt = SAMPLE_FMT_S16; |
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60 return 0; |
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61 } |
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62 |
6285 | 63 static void truespeech_read_frame(TSContext *dec, const uint8_t *input) |
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64 { |
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65 uint32_t t; |
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66 |
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67 /* first dword */ |
4364 | 68 t = AV_RL32(input); |
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69 input += 4; |
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70 |
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71 dec->flag = t & 1; |
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72 |
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73 dec->vector[0] = ts_codebook[0][(t >> 1) & 0x1F]; |
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74 dec->vector[1] = ts_codebook[1][(t >> 6) & 0x1F]; |
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75 dec->vector[2] = ts_codebook[2][(t >> 11) & 0xF]; |
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76 dec->vector[3] = ts_codebook[3][(t >> 15) & 0xF]; |
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77 dec->vector[4] = ts_codebook[4][(t >> 19) & 0xF]; |
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78 dec->vector[5] = ts_codebook[5][(t >> 23) & 0x7]; |
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79 dec->vector[6] = ts_codebook[6][(t >> 26) & 0x7]; |
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80 dec->vector[7] = ts_codebook[7][(t >> 29) & 0x7]; |
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81 |
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82 /* second dword */ |
4364 | 83 t = AV_RL32(input); |
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84 input += 4; |
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85 |
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86 dec->offset2[0] = (t >> 0) & 0x7F; |
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87 dec->offset2[1] = (t >> 7) & 0x7F; |
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88 dec->offset2[2] = (t >> 14) & 0x7F; |
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89 dec->offset2[3] = (t >> 21) & 0x7F; |
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90 |
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91 dec->offset1[0] = ((t >> 28) & 0xF) << 4; |
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92 |
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93 /* third dword */ |
4364 | 94 t = AV_RL32(input); |
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95 input += 4; |
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96 |
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97 dec->pulseval[0] = (t >> 0) & 0x3FFF; |
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98 dec->pulseval[1] = (t >> 14) & 0x3FFF; |
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99 |
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100 dec->offset1[1] = (t >> 28) & 0x0F; |
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101 |
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102 /* fourth dword */ |
4364 | 103 t = AV_RL32(input); |
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104 input += 4; |
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105 |
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106 dec->pulseval[2] = (t >> 0) & 0x3FFF; |
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107 dec->pulseval[3] = (t >> 14) & 0x3FFF; |
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108 |
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109 dec->offset1[1] |= ((t >> 28) & 0x0F) << 4; |
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110 |
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111 /* fifth dword */ |
4364 | 112 t = AV_RL32(input); |
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113 input += 4; |
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114 |
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115 dec->pulsepos[0] = (t >> 4) & 0x7FFFFFF; |
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116 |
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117 dec->pulseoff[0] = (t >> 0) & 0xF; |
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118 |
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119 dec->offset1[0] |= (t >> 31) & 1; |
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120 |
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121 /* sixth dword */ |
4364 | 122 t = AV_RL32(input); |
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123 input += 4; |
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124 |
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125 dec->pulsepos[1] = (t >> 4) & 0x7FFFFFF; |
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126 |
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127 dec->pulseoff[1] = (t >> 0) & 0xF; |
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128 |
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129 dec->offset1[0] |= ((t >> 31) & 1) << 1; |
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130 |
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131 /* seventh dword */ |
4364 | 132 t = AV_RL32(input); |
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133 input += 4; |
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134 |
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135 dec->pulsepos[2] = (t >> 4) & 0x7FFFFFF; |
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136 |
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137 dec->pulseoff[2] = (t >> 0) & 0xF; |
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138 |
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139 dec->offset1[0] |= ((t >> 31) & 1) << 2; |
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140 |
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141 /* eighth dword */ |
4364 | 142 t = AV_RL32(input); |
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143 input += 4; |
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144 |
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145 dec->pulsepos[3] = (t >> 4) & 0x7FFFFFF; |
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146 |
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147 dec->pulseoff[3] = (t >> 0) & 0xF; |
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148 |
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149 dec->offset1[0] |= ((t >> 31) & 1) << 3; |
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150 |
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151 } |
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152 |
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153 static void truespeech_correlate_filter(TSContext *dec) |
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154 { |
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155 int16_t tmp[8]; |
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156 int i, j; |
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157 |
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158 for(i = 0; i < 8; i++){ |
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159 if(i > 0){ |
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160 memcpy(tmp, dec->cvector, i * 2); |
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161 for(j = 0; j < i; j++) |
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162 dec->cvector[j] = ((tmp[i - j - 1] * dec->vector[i]) + |
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163 (dec->cvector[j] << 15) + 0x4000) >> 15; |
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164 } |
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165 dec->cvector[i] = (8 - dec->vector[i]) >> 3; |
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166 } |
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167 for(i = 0; i < 8; i++) |
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168 dec->cvector[i] = (dec->cvector[i] * ts_230[i]) >> 15; |
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169 |
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170 dec->filtval = dec->vector[0]; |
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171 } |
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172 |
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173 static void truespeech_filters_merge(TSContext *dec) |
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174 { |
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175 int i; |
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176 |
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177 if(!dec->flag){ |
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178 for(i = 0; i < 8; i++){ |
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179 dec->filters[i + 0] = dec->prevfilt[i]; |
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180 dec->filters[i + 8] = dec->prevfilt[i]; |
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181 } |
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182 }else{ |
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183 for(i = 0; i < 8; i++){ |
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184 dec->filters[i + 0]=(dec->cvector[i] * 21846 + dec->prevfilt[i] * 10923 + 16384) >> 15; |
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185 dec->filters[i + 8]=(dec->cvector[i] * 10923 + dec->prevfilt[i] * 21846 + 16384) >> 15; |
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186 } |
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187 } |
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188 for(i = 0; i < 8; i++){ |
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189 dec->filters[i + 16] = dec->cvector[i]; |
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190 dec->filters[i + 24] = dec->cvector[i]; |
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191 } |
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192 } |
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193 |
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194 static void truespeech_apply_twopoint_filter(TSContext *dec, int quart) |
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195 { |
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196 int16_t tmp[146 + 60], *ptr0, *ptr1; |
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197 const int16_t *filter; |
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198 int i, t, off; |
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199 |
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200 t = dec->offset2[quart]; |
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201 if(t == 127){ |
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202 memset(dec->newvec, 0, 60 * 2); |
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203 return; |
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204 } |
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TrueSpeech compatible audio decoder by Konstantin Shishkov
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205 for(i = 0; i < 146; i++) |
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206 tmp[i] = dec->filtbuf[i]; |
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207 off = (t / 25) + dec->offset1[quart >> 1] + 18; |
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208 ptr0 = tmp + 145 - off; |
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209 ptr1 = tmp + 146; |
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210 filter = (const int16_t*)ts_240 + (t % 25) * 2; |
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211 for(i = 0; i < 60; i++){ |
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212 t = (ptr0[0] * filter[0] + ptr0[1] * filter[1] + 0x2000) >> 14; |
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213 ptr0++; |
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214 dec->newvec[i] = t; |
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215 ptr1[i] = t; |
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216 } |
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217 } |
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218 |
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219 static void truespeech_place_pulses(TSContext *dec, int16_t *out, int quart) |
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220 { |
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221 int16_t tmp[7]; |
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222 int i, j, t; |
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223 const int16_t *ptr1; |
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224 int16_t *ptr2; |
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225 int coef; |
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226 |
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227 memset(out, 0, 60 * 2); |
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228 for(i = 0; i < 7; i++) { |
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229 t = dec->pulseval[quart] & 3; |
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230 dec->pulseval[quart] >>= 2; |
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231 tmp[6 - i] = ts_562[dec->pulseoff[quart] * 4 + t]; |
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232 } |
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233 |
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234 coef = dec->pulsepos[quart] >> 15; |
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235 ptr1 = (const int16_t*)ts_140 + 30; |
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236 ptr2 = tmp; |
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237 for(i = 0, j = 3; (i < 30) && (j > 0); i++){ |
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238 t = *ptr1++; |
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239 if(coef >= t) |
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240 coef -= t; |
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241 else{ |
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TrueSpeech compatible audio decoder by Konstantin Shishkov
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242 out[i] = *ptr2++; |
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TrueSpeech compatible audio decoder by Konstantin Shishkov
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243 ptr1 += 30; |
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244 j--; |
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TrueSpeech compatible audio decoder by Konstantin Shishkov
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245 } |
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TrueSpeech compatible audio decoder by Konstantin Shishkov
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246 } |
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247 coef = dec->pulsepos[quart] & 0x7FFF; |
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248 ptr1 = (const int16_t*)ts_140; |
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249 for(i = 30, j = 4; (i < 60) && (j > 0); i++){ |
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250 t = *ptr1++; |
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TrueSpeech compatible audio decoder by Konstantin Shishkov
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251 if(coef >= t) |
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TrueSpeech compatible audio decoder by Konstantin Shishkov
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252 coef -= t; |
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TrueSpeech compatible audio decoder by Konstantin Shishkov
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253 else{ |
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TrueSpeech compatible audio decoder by Konstantin Shishkov
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254 out[i] = *ptr2++; |
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TrueSpeech compatible audio decoder by Konstantin Shishkov
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255 ptr1 += 30; |
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TrueSpeech compatible audio decoder by Konstantin Shishkov
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256 j--; |
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TrueSpeech compatible audio decoder by Konstantin Shishkov
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257 } |
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TrueSpeech compatible audio decoder by Konstantin Shishkov
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258 } |
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TrueSpeech compatible audio decoder by Konstantin Shishkov
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259 |
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TrueSpeech compatible audio decoder by Konstantin Shishkov
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260 } |
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TrueSpeech compatible audio decoder by Konstantin Shishkov
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261 |
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262 static void truespeech_update_filters(TSContext *dec, int16_t *out, int quart) |
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263 { |
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TrueSpeech compatible audio decoder by Konstantin Shishkov
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264 int i; |
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265 |
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TrueSpeech compatible audio decoder by Konstantin Shishkov
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266 for(i = 0; i < 86; i++) |
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267 dec->filtbuf[i] = dec->filtbuf[i + 60]; |
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268 for(i = 0; i < 60; i++){ |
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269 dec->filtbuf[i + 86] = out[i] + dec->newvec[i] - (dec->newvec[i] >> 3); |
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270 out[i] += dec->newvec[i]; |
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271 } |
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TrueSpeech compatible audio decoder by Konstantin Shishkov
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272 } |
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273 |
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274 static void truespeech_synth(TSContext *dec, int16_t *out, int quart) |
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275 { |
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TrueSpeech compatible audio decoder by Konstantin Shishkov
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276 int i,k; |
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TrueSpeech compatible audio decoder by Konstantin Shishkov
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277 int t[8]; |
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TrueSpeech compatible audio decoder by Konstantin Shishkov
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278 int16_t *ptr0, *ptr1; |
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279 |
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TrueSpeech compatible audio decoder by Konstantin Shishkov
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280 ptr0 = dec->tmp1; |
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TrueSpeech compatible audio decoder by Konstantin Shishkov
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281 ptr1 = dec->filters + quart * 8; |
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TrueSpeech compatible audio decoder by Konstantin Shishkov
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282 for(i = 0; i < 60; i++){ |
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TrueSpeech compatible audio decoder by Konstantin Shishkov
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283 int sum = 0; |
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TrueSpeech compatible audio decoder by Konstantin Shishkov
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284 for(k = 0; k < 8; k++) |
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TrueSpeech compatible audio decoder by Konstantin Shishkov
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285 sum += ptr0[k] * ptr1[k]; |
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286 sum = (sum + (out[i] << 12) + 0x800) >> 12; |
4594 | 287 out[i] = av_clip(sum, -0x7FFE, 0x7FFE); |
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288 for(k = 7; k > 0; k--) |
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289 ptr0[k] = ptr0[k - 1]; |
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290 ptr0[0] = out[i]; |
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291 } |
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292 |
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293 for(i = 0; i < 8; i++) |
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294 t[i] = (ts_5E2[i] * ptr1[i]) >> 15; |
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295 |
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296 ptr0 = dec->tmp2; |
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TrueSpeech compatible audio decoder by Konstantin Shishkov
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297 for(i = 0; i < 60; i++){ |
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TrueSpeech compatible audio decoder by Konstantin Shishkov
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298 int sum = 0; |
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TrueSpeech compatible audio decoder by Konstantin Shishkov
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299 for(k = 0; k < 8; k++) |
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TrueSpeech compatible audio decoder by Konstantin Shishkov
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300 sum += ptr0[k] * t[k]; |
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TrueSpeech compatible audio decoder by Konstantin Shishkov
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301 for(k = 7; k > 0; k--) |
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TrueSpeech compatible audio decoder by Konstantin Shishkov
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302 ptr0[k] = ptr0[k - 1]; |
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TrueSpeech compatible audio decoder by Konstantin Shishkov
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303 ptr0[0] = out[i]; |
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TrueSpeech compatible audio decoder by Konstantin Shishkov
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304 out[i] = ((out[i] << 12) - sum) >> 12; |
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TrueSpeech compatible audio decoder by Konstantin Shishkov
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305 } |
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306 |
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307 for(i = 0; i < 8; i++) |
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308 t[i] = (ts_5F2[i] * ptr1[i]) >> 15; |
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309 |
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310 ptr0 = dec->tmp3; |
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311 for(i = 0; i < 60; i++){ |
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312 int sum = out[i] << 12; |
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313 for(k = 0; k < 8; k++) |
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314 sum += ptr0[k] * t[k]; |
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315 for(k = 7; k > 0; k--) |
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316 ptr0[k] = ptr0[k - 1]; |
4594 | 317 ptr0[0] = av_clip((sum + 0x800) >> 12, -0x7FFE, 0x7FFE); |
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318 |
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319 sum = ((ptr0[1] * (dec->filtval - (dec->filtval >> 2))) >> 4) + sum; |
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320 sum = sum - (sum >> 3); |
4594 | 321 out[i] = av_clip((sum + 0x800) >> 12, -0x7FFE, 0x7FFE); |
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322 } |
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323 } |
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324 |
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325 static void truespeech_save_prevvec(TSContext *c) |
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326 { |
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327 int i; |
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328 |
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329 for(i = 0; i < 8; i++) |
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330 c->prevfilt[i] = c->cvector[i]; |
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331 } |
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332 |
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333 static int truespeech_decode_frame(AVCodecContext *avctx, |
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334 void *data, int *data_size, |
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335 AVPacket *avpkt) |
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336 { |
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337 const uint8_t *buf = avpkt->data; |
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338 int buf_size = avpkt->size; |
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339 TSContext *c = avctx->priv_data; |
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340 |
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341 int i, j; |
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342 short *samples = data; |
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343 int consumed = 0; |
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344 int16_t out_buf[240]; |
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345 int iterations; |
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346 |
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347 if (!buf_size) |
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348 return 0; |
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349 |
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350 iterations = FFMIN(buf_size / 32, *data_size / 480); |
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351 for(j = 0; j < iterations; j++) { |
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352 truespeech_read_frame(c, buf + consumed); |
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353 consumed += 32; |
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354 |
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355 truespeech_correlate_filter(c); |
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356 truespeech_filters_merge(c); |
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357 |
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358 memset(out_buf, 0, 240 * 2); |
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359 for(i = 0; i < 4; i++) { |
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360 truespeech_apply_twopoint_filter(c, i); |
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361 truespeech_place_pulses(c, out_buf + i * 60, i); |
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362 truespeech_update_filters(c, out_buf + i * 60, i); |
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363 truespeech_synth(c, out_buf + i * 60, i); |
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364 } |
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365 |
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366 truespeech_save_prevvec(c); |
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367 |
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368 /* finally output decoded frame */ |
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369 for(i = 0; i < 240; i++) |
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370 *samples++ = out_buf[i]; |
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371 |
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372 } |
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373 |
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374 *data_size = consumed * 15; |
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375 |
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376 return consumed; |
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377 } |
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378 |
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379 AVCodec truespeech_decoder = { |
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380 "truespeech", |
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381 AVMEDIA_TYPE_AUDIO, |
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382 CODEC_ID_TRUESPEECH, |
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383 sizeof(TSContext), |
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384 truespeech_decode_init, |
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385 NULL, |
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386 NULL, |
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387 truespeech_decode_frame, |
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388 .long_name = NULL_IF_CONFIG_SMALL("DSP Group TrueSpeech"), |
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389 }; |