Mercurial > libavcodec.hg
annotate truespeech.c @ 8563:b59279494d84 libavcodec
Cosmetics: Rename ff_vdpau_h264_add_data_chunk as ff_vdpau_add_data_chunk.
author | cehoyos |
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date | Sat, 10 Jan 2009 23:00:48 +0000 |
parents | 85ab7655ad4d |
children | 2acf0ae7b041 |
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 #include "avcodec.h" |
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22 |
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23 #include "truespeech_data.h" |
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24 /** |
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25 * @file truespeech.c |
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26 * TrueSpeech decoder. |
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27 */ |
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28 |
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29 /** |
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30 * TrueSpeech decoder context |
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31 */ |
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32 typedef struct { |
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33 /* input data */ |
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34 int16_t vector[8]; //< input vector: 5/5/4/4/4/3/3/3 |
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35 int offset1[2]; //< 8-bit value, used in one copying offset |
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36 int offset2[4]; //< 7-bit value, encodes offsets for copying and for two-point filter |
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37 int pulseoff[4]; //< 4-bit offset of pulse values block |
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38 int pulsepos[4]; //< 27-bit variable, encodes 7 pulse positions |
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39 int pulseval[4]; //< 7x2-bit pulse values |
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40 int flag; //< 1-bit flag, shows how to choose filters |
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41 /* temporary data */ |
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42 int filtbuf[146]; // some big vector used for storing filters |
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43 int prevfilt[8]; // filter from previous frame |
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44 int16_t tmp1[8]; // coefficients for adding to out |
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45 int16_t tmp2[8]; // coefficients for adding to out |
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46 int16_t tmp3[8]; // coefficients for adding to out |
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47 int16_t cvector[8]; // correlated input vector |
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48 int filtval; // gain value for one function |
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49 int16_t newvec[60]; // tmp vector |
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50 int16_t filters[32]; // filters for every subframe |
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51 } TSContext; |
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52 |
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53 static av_cold int truespeech_decode_init(AVCodecContext * avctx) |
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54 { |
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55 // TSContext *c = avctx->priv_data; |
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56 |
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57 avctx->sample_fmt = SAMPLE_FMT_S16; |
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58 return 0; |
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59 } |
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60 |
6285 | 61 static void truespeech_read_frame(TSContext *dec, const uint8_t *input) |
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62 { |
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63 uint32_t t; |
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64 |
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65 /* first dword */ |
4364 | 66 t = AV_RL32(input); |
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67 input += 4; |
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68 |
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69 dec->flag = t & 1; |
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70 |
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71 dec->vector[0] = ts_codebook[0][(t >> 1) & 0x1F]; |
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72 dec->vector[1] = ts_codebook[1][(t >> 6) & 0x1F]; |
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73 dec->vector[2] = ts_codebook[2][(t >> 11) & 0xF]; |
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74 dec->vector[3] = ts_codebook[3][(t >> 15) & 0xF]; |
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75 dec->vector[4] = ts_codebook[4][(t >> 19) & 0xF]; |
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76 dec->vector[5] = ts_codebook[5][(t >> 23) & 0x7]; |
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77 dec->vector[6] = ts_codebook[6][(t >> 26) & 0x7]; |
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78 dec->vector[7] = ts_codebook[7][(t >> 29) & 0x7]; |
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79 |
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80 /* second dword */ |
4364 | 81 t = AV_RL32(input); |
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82 input += 4; |
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83 |
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84 dec->offset2[0] = (t >> 0) & 0x7F; |
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85 dec->offset2[1] = (t >> 7) & 0x7F; |
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86 dec->offset2[2] = (t >> 14) & 0x7F; |
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87 dec->offset2[3] = (t >> 21) & 0x7F; |
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88 |
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89 dec->offset1[0] = ((t >> 28) & 0xF) << 4; |
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90 |
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91 /* third dword */ |
4364 | 92 t = AV_RL32(input); |
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93 input += 4; |
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94 |
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95 dec->pulseval[0] = (t >> 0) & 0x3FFF; |
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96 dec->pulseval[1] = (t >> 14) & 0x3FFF; |
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97 |
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98 dec->offset1[1] = (t >> 28) & 0x0F; |
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99 |
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100 /* fourth dword */ |
4364 | 101 t = AV_RL32(input); |
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102 input += 4; |
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103 |
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104 dec->pulseval[2] = (t >> 0) & 0x3FFF; |
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105 dec->pulseval[3] = (t >> 14) & 0x3FFF; |
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106 |
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107 dec->offset1[1] |= ((t >> 28) & 0x0F) << 4; |
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108 |
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109 /* fifth dword */ |
4364 | 110 t = AV_RL32(input); |
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111 input += 4; |
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112 |
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113 dec->pulsepos[0] = (t >> 4) & 0x7FFFFFF; |
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114 |
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115 dec->pulseoff[0] = (t >> 0) & 0xF; |
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116 |
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117 dec->offset1[0] |= (t >> 31) & 1; |
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118 |
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119 /* sixth dword */ |
4364 | 120 t = AV_RL32(input); |
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121 input += 4; |
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122 |
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123 dec->pulsepos[1] = (t >> 4) & 0x7FFFFFF; |
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124 |
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125 dec->pulseoff[1] = (t >> 0) & 0xF; |
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126 |
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127 dec->offset1[0] |= ((t >> 31) & 1) << 1; |
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128 |
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129 /* seventh dword */ |
4364 | 130 t = AV_RL32(input); |
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131 input += 4; |
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132 |
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133 dec->pulsepos[2] = (t >> 4) & 0x7FFFFFF; |
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134 |
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135 dec->pulseoff[2] = (t >> 0) & 0xF; |
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136 |
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137 dec->offset1[0] |= ((t >> 31) & 1) << 2; |
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138 |
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139 /* eighth dword */ |
4364 | 140 t = AV_RL32(input); |
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141 input += 4; |
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142 |
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143 dec->pulsepos[3] = (t >> 4) & 0x7FFFFFF; |
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144 |
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145 dec->pulseoff[3] = (t >> 0) & 0xF; |
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146 |
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147 dec->offset1[0] |= ((t >> 31) & 1) << 3; |
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148 |
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149 } |
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150 |
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151 static void truespeech_correlate_filter(TSContext *dec) |
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152 { |
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153 int16_t tmp[8]; |
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154 int i, j; |
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155 |
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156 for(i = 0; i < 8; i++){ |
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157 if(i > 0){ |
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158 memcpy(tmp, dec->cvector, i * 2); |
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159 for(j = 0; j < i; j++) |
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160 dec->cvector[j] = ((tmp[i - j - 1] * dec->vector[i]) + |
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161 (dec->cvector[j] << 15) + 0x4000) >> 15; |
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162 } |
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163 dec->cvector[i] = (8 - dec->vector[i]) >> 3; |
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164 } |
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165 for(i = 0; i < 8; i++) |
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166 dec->cvector[i] = (dec->cvector[i] * ts_230[i]) >> 15; |
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167 |
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168 dec->filtval = dec->vector[0]; |
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169 } |
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170 |
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171 static void truespeech_filters_merge(TSContext *dec) |
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172 { |
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173 int i; |
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174 |
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175 if(!dec->flag){ |
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176 for(i = 0; i < 8; i++){ |
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177 dec->filters[i + 0] = dec->prevfilt[i]; |
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178 dec->filters[i + 8] = dec->prevfilt[i]; |
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179 } |
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180 }else{ |
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181 for(i = 0; i < 8; i++){ |
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182 dec->filters[i + 0]=(dec->cvector[i] * 21846 + dec->prevfilt[i] * 10923 + 16384) >> 15; |
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183 dec->filters[i + 8]=(dec->cvector[i] * 10923 + dec->prevfilt[i] * 21846 + 16384) >> 15; |
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184 } |
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185 } |
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186 for(i = 0; i < 8; i++){ |
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187 dec->filters[i + 16] = dec->cvector[i]; |
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188 dec->filters[i + 24] = dec->cvector[i]; |
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189 } |
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190 } |
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191 |
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192 static void truespeech_apply_twopoint_filter(TSContext *dec, int quart) |
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193 { |
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194 int16_t tmp[146 + 60], *ptr0, *ptr1; |
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195 const int16_t *filter; |
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196 int i, t, off; |
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197 |
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198 t = dec->offset2[quart]; |
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199 if(t == 127){ |
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200 memset(dec->newvec, 0, 60 * 2); |
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201 return; |
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202 } |
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TrueSpeech compatible audio decoder by Konstantin Shishkov
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203 for(i = 0; i < 146; i++) |
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204 tmp[i] = dec->filtbuf[i]; |
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205 off = (t / 25) + dec->offset1[quart >> 1] + 18; |
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206 ptr0 = tmp + 145 - off; |
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207 ptr1 = tmp + 146; |
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208 filter = (const int16_t*)ts_240 + (t % 25) * 2; |
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209 for(i = 0; i < 60; i++){ |
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210 t = (ptr0[0] * filter[0] + ptr0[1] * filter[1] + 0x2000) >> 14; |
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211 ptr0++; |
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212 dec->newvec[i] = t; |
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213 ptr1[i] = t; |
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214 } |
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215 } |
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216 |
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217 static void truespeech_place_pulses(TSContext *dec, int16_t *out, int quart) |
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218 { |
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219 int16_t tmp[7]; |
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220 int i, j, t; |
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221 const int16_t *ptr1; |
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222 int16_t *ptr2; |
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223 int coef; |
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224 |
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225 memset(out, 0, 60 * 2); |
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226 for(i = 0; i < 7; i++) { |
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227 t = dec->pulseval[quart] & 3; |
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228 dec->pulseval[quart] >>= 2; |
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229 tmp[6 - i] = ts_562[dec->pulseoff[quart] * 4 + t]; |
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230 } |
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231 |
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232 coef = dec->pulsepos[quart] >> 15; |
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233 ptr1 = (const int16_t*)ts_140 + 30; |
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234 ptr2 = tmp; |
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235 for(i = 0, j = 3; (i < 30) && (j > 0); i++){ |
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236 t = *ptr1++; |
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237 if(coef >= t) |
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238 coef -= t; |
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TrueSpeech compatible audio decoder by Konstantin Shishkov
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239 else{ |
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240 out[i] = *ptr2++; |
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TrueSpeech compatible audio decoder by Konstantin Shishkov
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241 ptr1 += 30; |
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242 j--; |
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243 } |
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244 } |
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245 coef = dec->pulsepos[quart] & 0x7FFF; |
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246 ptr1 = (const int16_t*)ts_140; |
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247 for(i = 30, j = 4; (i < 60) && (j > 0); i++){ |
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248 t = *ptr1++; |
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249 if(coef >= t) |
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250 coef -= t; |
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TrueSpeech compatible audio decoder by Konstantin Shishkov
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251 else{ |
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TrueSpeech compatible audio decoder by Konstantin Shishkov
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252 out[i] = *ptr2++; |
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TrueSpeech compatible audio decoder by Konstantin Shishkov
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253 ptr1 += 30; |
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254 j--; |
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255 } |
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256 } |
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257 |
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258 } |
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259 |
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260 static void truespeech_update_filters(TSContext *dec, int16_t *out, int quart) |
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261 { |
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262 int i; |
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263 |
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264 for(i = 0; i < 86; i++) |
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265 dec->filtbuf[i] = dec->filtbuf[i + 60]; |
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266 for(i = 0; i < 60; i++){ |
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267 dec->filtbuf[i + 86] = out[i] + dec->newvec[i] - (dec->newvec[i] >> 3); |
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268 out[i] += dec->newvec[i]; |
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269 } |
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270 } |
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271 |
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272 static void truespeech_synth(TSContext *dec, int16_t *out, int quart) |
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273 { |
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274 int i,k; |
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275 int t[8]; |
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276 int16_t *ptr0, *ptr1; |
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277 |
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278 ptr0 = dec->tmp1; |
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279 ptr1 = dec->filters + quart * 8; |
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280 for(i = 0; i < 60; i++){ |
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281 int sum = 0; |
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282 for(k = 0; k < 8; k++) |
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283 sum += ptr0[k] * ptr1[k]; |
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284 sum = (sum + (out[i] << 12) + 0x800) >> 12; |
4594 | 285 out[i] = av_clip(sum, -0x7FFE, 0x7FFE); |
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286 for(k = 7; k > 0; k--) |
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287 ptr0[k] = ptr0[k - 1]; |
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288 ptr0[0] = out[i]; |
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289 } |
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290 |
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291 for(i = 0; i < 8; i++) |
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292 t[i] = (ts_5E2[i] * ptr1[i]) >> 15; |
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293 |
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294 ptr0 = dec->tmp2; |
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295 for(i = 0; i < 60; i++){ |
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TrueSpeech compatible audio decoder by Konstantin Shishkov
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296 int sum = 0; |
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TrueSpeech compatible audio decoder by Konstantin Shishkov
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297 for(k = 0; k < 8; k++) |
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298 sum += ptr0[k] * t[k]; |
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299 for(k = 7; k > 0; k--) |
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300 ptr0[k] = ptr0[k - 1]; |
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301 ptr0[0] = out[i]; |
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302 out[i] = ((out[i] << 12) - sum) >> 12; |
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303 } |
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TrueSpeech compatible audio decoder by Konstantin Shishkov
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304 |
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TrueSpeech compatible audio decoder by Konstantin Shishkov
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305 for(i = 0; i < 8; i++) |
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TrueSpeech compatible audio decoder by Konstantin Shishkov
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306 t[i] = (ts_5F2[i] * ptr1[i]) >> 15; |
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307 |
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308 ptr0 = dec->tmp3; |
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309 for(i = 0; i < 60; i++){ |
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310 int sum = out[i] << 12; |
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311 for(k = 0; k < 8; k++) |
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312 sum += ptr0[k] * t[k]; |
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313 for(k = 7; k > 0; k--) |
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314 ptr0[k] = ptr0[k - 1]; |
4594 | 315 ptr0[0] = av_clip((sum + 0x800) >> 12, -0x7FFE, 0x7FFE); |
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316 |
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317 sum = ((ptr0[1] * (dec->filtval - (dec->filtval >> 2))) >> 4) + sum; |
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318 sum = sum - (sum >> 3); |
4594 | 319 out[i] = av_clip((sum + 0x800) >> 12, -0x7FFE, 0x7FFE); |
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320 } |
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321 } |
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322 |
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323 static void truespeech_save_prevvec(TSContext *c) |
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324 { |
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325 int i; |
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326 |
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327 for(i = 0; i < 8; i++) |
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328 c->prevfilt[i] = c->cvector[i]; |
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329 } |
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330 |
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331 static int truespeech_decode_frame(AVCodecContext *avctx, |
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332 void *data, int *data_size, |
6285 | 333 const uint8_t *buf, int buf_size) |
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334 { |
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335 TSContext *c = avctx->priv_data; |
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336 |
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337 int i, j; |
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338 short *samples = data; |
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339 int consumed = 0; |
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340 int16_t out_buf[240]; |
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341 int iterations; |
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342 |
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343 if (!buf_size) |
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344 return 0; |
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345 |
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346 iterations = FFMIN(buf_size / 32, *data_size / 480); |
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347 for(j = 0; j < iterations; j++) { |
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348 truespeech_read_frame(c, buf + consumed); |
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349 consumed += 32; |
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350 |
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351 truespeech_correlate_filter(c); |
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352 truespeech_filters_merge(c); |
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353 |
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354 memset(out_buf, 0, 240 * 2); |
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355 for(i = 0; i < 4; i++) { |
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356 truespeech_apply_twopoint_filter(c, i); |
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357 truespeech_place_pulses(c, out_buf + i * 60, i); |
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358 truespeech_update_filters(c, out_buf + i * 60, i); |
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359 truespeech_synth(c, out_buf + i * 60, i); |
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360 } |
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361 |
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362 truespeech_save_prevvec(c); |
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363 |
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364 /* finally output decoded frame */ |
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365 for(i = 0; i < 240; i++) |
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366 *samples++ = out_buf[i]; |
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367 |
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368 } |
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369 |
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370 *data_size = consumed * 15; |
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371 |
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372 return consumed; |
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373 } |
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374 |
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375 AVCodec truespeech_decoder = { |
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376 "truespeech", |
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377 CODEC_TYPE_AUDIO, |
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378 CODEC_ID_TRUESPEECH, |
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379 sizeof(TSContext), |
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380 truespeech_decode_init, |
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381 NULL, |
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382 NULL, |
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383 truespeech_decode_frame, |
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384 .long_name = NULL_IF_CONFIG_SMALL("DSP Group TrueSpeech"), |
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385 }; |