Mercurial > libavcodec.hg
annotate alac.c @ 5713:f45228f65e06 libavcodec
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author | andoma |
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date | Tue, 25 Sep 2007 08:43:49 +0000 |
parents | cf88751d8ab7 |
children | 80103098c797 |
rev | line source |
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1 /* |
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2 * ALAC (Apple Lossless Audio Codec) decoder |
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3 * Copyright (c) 2005 David Hammerton |
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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 /** |
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23 * @file alac.c |
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24 * ALAC (Apple Lossless Audio Codec) decoder |
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25 * @author 2005 David Hammerton |
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26 * |
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27 * For more information on the ALAC format, visit: |
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28 * http://crazney.net/programs/itunes/alac.html |
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29 * |
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30 * Note: This decoder expects a 36- (0x24-)byte QuickTime atom to be |
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31 * passed through the extradata[_size] fields. This atom is tacked onto |
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32 * the end of an 'alac' stsd atom and has the following format: |
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33 * bytes 0-3 atom size (0x24), big-endian |
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34 * bytes 4-7 atom type ('alac', not the 'alac' tag from start of stsd) |
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35 * bytes 8-35 data bytes needed by decoder |
3135 | 36 * |
37 * Extradata: | |
38 * 32bit size | |
39 * 32bit tag (=alac) | |
40 * 32bit zero? | |
41 * 32bit max sample per frame | |
42 * 8bit ?? (zero?) | |
43 * 8bit sample size | |
44 * 8bit history mult | |
45 * 8bit initial history | |
46 * 8bit kmodifier | |
47 * 8bit channels? | |
48 * 16bit ?? | |
49 * 32bit max coded frame size | |
50 * 32bit bitrate? | |
51 * 32bit samplerate | |
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52 */ |
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53 |
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54 |
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55 #include "avcodec.h" |
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56 #include "bitstream.h" |
5276 | 57 #include "bytestream.h" |
5628 | 58 #include "unary.h" |
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59 |
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60 #define ALAC_EXTRADATA_SIZE 36 |
5272 | 61 #define MAX_CHANNELS 2 |
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62 |
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63 typedef struct { |
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64 |
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65 AVCodecContext *avctx; |
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66 GetBitContext gb; |
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67 /* init to 0; first frame decode should initialize from extradata and |
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68 * set this to 1 */ |
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69 int context_initialized; |
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70 |
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71 int samplesize; |
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72 int numchannels; |
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73 int bytespersample; |
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74 |
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75 /* buffers */ |
5272 | 76 int32_t *predicterror_buffer[MAX_CHANNELS]; |
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77 |
5272 | 78 int32_t *outputsamples_buffer[MAX_CHANNELS]; |
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79 |
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80 /* stuff from setinfo */ |
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81 uint32_t setinfo_max_samples_per_frame; /* 0x1000 = 4096 */ /* max samples per frame? */ |
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82 uint8_t setinfo_7a; /* 0x00 */ |
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83 uint8_t setinfo_sample_size; /* 0x10 */ |
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84 uint8_t setinfo_rice_historymult; /* 0x28 */ |
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85 uint8_t setinfo_rice_initialhistory; /* 0x0a */ |
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86 uint8_t setinfo_rice_kmodifier; /* 0x0e */ |
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87 uint8_t setinfo_7f; /* 0x02 */ |
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88 uint16_t setinfo_80; /* 0x00ff */ |
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89 uint32_t setinfo_82; /* 0x000020e7 */ /* max sample size?? */ |
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90 uint32_t setinfo_86; /* 0x00069fe4 */ /* bit rate (average)?? */ |
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91 uint32_t setinfo_8a_rate; /* 0x0000ac44 */ |
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92 /* end setinfo stuff */ |
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93 |
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94 } ALACContext; |
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95 |
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96 static void allocate_buffers(ALACContext *alac) |
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97 { |
5272 | 98 int chan; |
99 for (chan = 0; chan < MAX_CHANNELS; chan++) { | |
100 alac->predicterror_buffer[chan] = | |
101 av_malloc(alac->setinfo_max_samples_per_frame * 4); | |
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102 |
5272 | 103 alac->outputsamples_buffer[chan] = |
104 av_malloc(alac->setinfo_max_samples_per_frame * 4); | |
105 } | |
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106 } |
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107 |
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108 static int alac_set_info(ALACContext *alac) |
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109 { |
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110 unsigned char *ptr = alac->avctx->extradata; |
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111 |
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112 ptr += 4; /* size */ |
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113 ptr += 4; /* alac */ |
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114 ptr += 4; /* 0 ? */ |
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115 |
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117 av_log(alac->avctx, AV_LOG_ERROR, "setinfo_max_samples_per_frame too large\n"); |
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118 return -1; |
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119 } |
5276 | 120 |
121 /* buffer size / 2 ? */ | |
122 alac->setinfo_max_samples_per_frame = bytestream_get_be32(&ptr); | |
5360 | 123 alac->setinfo_7a = *ptr++; |
124 alac->setinfo_sample_size = *ptr++; | |
125 alac->setinfo_rice_historymult = *ptr++; | |
126 alac->setinfo_rice_initialhistory = *ptr++; | |
127 alac->setinfo_rice_kmodifier = *ptr++; | |
128 /* channels? */ | |
129 alac->setinfo_7f = *ptr++; | |
5276 | 130 alac->setinfo_80 = bytestream_get_be16(&ptr); |
131 /* max coded frame size */ | |
132 alac->setinfo_82 = bytestream_get_be32(&ptr); | |
133 /* bitrate ? */ | |
134 alac->setinfo_86 = bytestream_get_be32(&ptr); | |
135 /* samplerate */ | |
136 alac->setinfo_8a_rate = bytestream_get_be32(&ptr); | |
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137 |
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138 allocate_buffers(alac); |
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139 |
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140 return 0; |
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141 } |
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142 |
5364 | 143 static inline int count_leading_zeros(int32_t input) |
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144 { |
5364 | 145 return 31-av_log2(input); |
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146 } |
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147 |
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148 static void bastardized_rice_decompress(ALACContext *alac, |
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149 int32_t *output_buffer, |
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150 int output_size, |
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151 int readsamplesize, /* arg_10 */ |
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152 int rice_initialhistory, /* arg424->b */ |
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153 int rice_kmodifier, /* arg424->d */ |
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154 int rice_historymult, /* arg424->c */ |
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155 int rice_kmodifier_mask /* arg424->e */ |
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156 ) |
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157 { |
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158 int output_count; |
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159 unsigned int history = rice_initialhistory; |
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160 int sign_modifier = 0; |
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161 |
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162 for (output_count = 0; output_count < output_size; output_count++) { |
5628 | 163 int32_t x; |
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164 int32_t x_modified; |
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165 int32_t final_val; |
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166 |
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167 /* read x - number of 1s before 0 represent the rice */ |
5628 | 168 x = get_unary_0_9(&alac->gb); |
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169 |
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170 if (x > 8) { /* RICE THRESHOLD */ |
5365 | 171 /* use alternative encoding */ |
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172 int32_t value; |
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173 |
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174 value = get_bits(&alac->gb, readsamplesize); |
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175 |
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176 /* mask value to readsamplesize size */ |
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177 if (readsamplesize != 32) |
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178 value &= (0xffffffff >> (32 - readsamplesize)); |
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179 |
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180 x = value; |
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181 } else { |
5365 | 182 /* standard rice encoding */ |
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183 int extrabits; |
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184 int k; /* size of extra bits */ |
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185 |
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186 /* read k, that is bits as is */ |
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187 k = 31 - rice_kmodifier - count_leading_zeros((history >> 9) + 3); |
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188 |
2967 | 189 if (k < 0) |
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190 k += rice_kmodifier; |
2967 | 191 else |
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192 k = rice_kmodifier; |
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193 |
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194 if (k != 1) { |
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195 extrabits = show_bits(&alac->gb, k); |
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196 |
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197 /* multiply x by 2^k - 1, as part of their strange algorithm */ |
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198 x = (x << k) - x; |
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199 |
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200 if (extrabits > 1) { |
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201 x += extrabits - 1; |
5518 | 202 skip_bits(&alac->gb, k); |
5365 | 203 } else |
5518 | 204 skip_bits(&alac->gb, k - 1); |
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205 } |
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206 } |
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207 |
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208 x_modified = sign_modifier + x; |
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209 final_val = (x_modified + 1) / 2; |
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210 if (x_modified & 1) final_val *= -1; |
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211 |
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212 output_buffer[output_count] = final_val; |
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213 |
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214 sign_modifier = 0; |
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215 |
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216 /* now update the history */ |
5365 | 217 history += x_modified * rice_historymult |
218 - ((history * rice_historymult) >> 9); | |
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219 |
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220 if (x_modified > 0xffff) |
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221 history = 0xffff; |
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222 |
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223 /* special case: there may be compressed blocks of 0 */ |
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224 if ((history < 128) && (output_count+1 < output_size)) { |
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225 int block_size; |
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226 |
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227 sign_modifier = 1; |
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228 |
5628 | 229 x = get_unary_0_9(&alac->gb); |
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230 |
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231 if (x > 8) { |
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232 block_size = get_bits(&alac->gb, 16); |
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233 block_size &= 0xffff; |
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234 } else { |
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235 int k; |
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236 int extrabits; |
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237 |
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238 k = count_leading_zeros(history) + ((history + 16) >> 6 /* / 64 */) - 24; |
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239 |
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240 extrabits = show_bits(&alac->gb, k); |
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241 |
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242 block_size = (((1 << k) - 1) & rice_kmodifier_mask) * x |
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243 + extrabits - 1; |
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244 |
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245 if (extrabits < 2) { |
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246 x = 1 - extrabits; |
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247 block_size += x; |
5518 | 248 skip_bits(&alac->gb, k - 1); |
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249 } else { |
5518 | 250 skip_bits(&alac->gb, k); |
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251 } |
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252 } |
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253 |
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254 if (block_size > 0) { |
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255 memset(&output_buffer[output_count+1], 0, block_size * 4); |
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256 output_count += block_size; |
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257 } |
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258 |
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259 if (block_size > 0xffff) |
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260 sign_modifier = 0; |
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261 |
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262 history = 0; |
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263 } |
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264 } |
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265 } |
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266 |
5617 | 267 static inline int32_t extend_sign32(int32_t val, int bits) |
5616 | 268 { |
269 return (val << (32 - bits)) >> (32 - bits); | |
270 } | |
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271 |
5616 | 272 static inline int sign_only(int v) |
273 { | |
274 return v ? FFSIGN(v) : 0; | |
275 } | |
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276 |
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277 static void predictor_decompress_fir_adapt(int32_t *error_buffer, |
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278 int32_t *buffer_out, |
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279 int output_size, |
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280 int readsamplesize, |
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281 int16_t *predictor_coef_table, |
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282 int predictor_coef_num, |
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283 int predictor_quantitization) |
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284 { |
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285 int i; |
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286 |
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287 /* first sample always copies */ |
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288 *buffer_out = *error_buffer; |
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289 |
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290 if (!predictor_coef_num) { |
5365 | 291 if (output_size <= 1) |
292 return; | |
293 | |
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294 memcpy(buffer_out+1, error_buffer+1, (output_size-1) * 4); |
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295 return; |
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296 } |
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297 |
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298 if (predictor_coef_num == 0x1f) { /* 11111 - max value of predictor_coef_num */ |
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299 /* second-best case scenario for fir decompression, |
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300 * error describes a small difference from the previous sample only |
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301 */ |
5365 | 302 if (output_size <= 1) |
303 return; | |
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304 for (i = 0; i < output_size - 1; i++) { |
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305 int32_t prev_value; |
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306 int32_t error_value; |
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307 |
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308 prev_value = buffer_out[i]; |
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309 error_value = error_buffer[i+1]; |
5365 | 310 buffer_out[i+1] = |
5617 | 311 extend_sign32((prev_value + error_value), readsamplesize); |
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312 } |
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313 return; |
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314 } |
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315 |
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316 /* read warm-up samples */ |
5365 | 317 if (predictor_coef_num > 0) |
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318 for (i = 0; i < predictor_coef_num; i++) { |
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319 int32_t val; |
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320 |
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321 val = buffer_out[i] + error_buffer[i+1]; |
5617 | 322 val = extend_sign32(val, readsamplesize); |
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323 buffer_out[i+1] = val; |
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324 } |
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325 |
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326 #if 0 |
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327 /* 4 and 8 are very common cases (the only ones i've seen). these |
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328 * should be unrolled and optimised |
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329 */ |
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330 if (predictor_coef_num == 4) { |
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331 /* FIXME: optimised general case */ |
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332 return; |
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333 } |
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334 |
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335 if (predictor_coef_table == 8) { |
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336 /* FIXME: optimised general case */ |
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337 return; |
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338 } |
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339 #endif |
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340 |
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341 /* general case */ |
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342 if (predictor_coef_num > 0) { |
5365 | 343 for (i = predictor_coef_num + 1; i < output_size; i++) { |
2542
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344 int j; |
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345 int sum = 0; |
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346 int outval; |
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347 int error_val = error_buffer[i]; |
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348 |
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349 for (j = 0; j < predictor_coef_num; j++) { |
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350 sum += (buffer_out[predictor_coef_num-j] - buffer_out[0]) * |
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351 predictor_coef_table[j]; |
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352 } |
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353 |
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354 outval = (1 << (predictor_quantitization-1)) + sum; |
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355 outval = outval >> predictor_quantitization; |
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356 outval = outval + buffer_out[0] + error_val; |
5617 | 357 outval = extend_sign32(outval, readsamplesize); |
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358 |
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359 buffer_out[predictor_coef_num+1] = outval; |
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360 |
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361 if (error_val > 0) { |
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362 int predictor_num = predictor_coef_num - 1; |
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363 |
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364 while (predictor_num >= 0 && error_val > 0) { |
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365 int val = buffer_out[0] - buffer_out[predictor_coef_num - predictor_num]; |
5616 | 366 int sign = sign_only(val); |
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367 |
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368 predictor_coef_table[predictor_num] -= sign; |
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369 |
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370 val *= sign; /* absolute value */ |
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371 |
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372 error_val -= ((val >> predictor_quantitization) * |
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373 (predictor_coef_num - predictor_num)); |
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374 |
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375 predictor_num--; |
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376 } |
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377 } else if (error_val < 0) { |
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378 int predictor_num = predictor_coef_num - 1; |
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379 |
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380 while (predictor_num >= 0 && error_val < 0) { |
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381 int val = buffer_out[0] - buffer_out[predictor_coef_num - predictor_num]; |
5616 | 382 int sign = - sign_only(val); |
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383 |
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384 predictor_coef_table[predictor_num] -= sign; |
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385 |
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386 val *= sign; /* neg value */ |
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387 |
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388 error_val -= ((val >> predictor_quantitization) * |
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389 (predictor_coef_num - predictor_num)); |
a27a580f292e
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390 |
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391 predictor_num--; |
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392 } |
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393 } |
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394 |
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395 buffer_out++; |
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396 } |
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397 } |
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398 } |
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399 |
5504 | 400 static void reconstruct_stereo_16(int32_t *buffer[MAX_CHANNELS], |
5505 | 401 int16_t *buffer_out, |
402 int numchannels, int numsamples, | |
403 uint8_t interlacing_shift, | |
404 uint8_t interlacing_leftweight) | |
2543
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405 { |
2542
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406 int i; |
5365 | 407 if (numsamples <= 0) |
408 return; | |
2542
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409 |
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410 /* weighted interlacing */ |
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411 if (interlacing_leftweight) { |
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412 for (i = 0; i < numsamples; i++) { |
5372 | 413 int32_t a, b; |
2542
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414 |
5372 | 415 a = buffer[0][i]; |
416 b = buffer[1][i]; | |
2542
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417 |
5372 | 418 a -= (b * interlacing_leftweight) >> interlacing_shift; |
419 b += a; | |
2542
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420 |
5372 | 421 buffer_out[i*numchannels] = b; |
422 buffer_out[i*numchannels + 1] = a; | |
2542
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423 } |
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424 |
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425 return; |
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426 } |
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427 |
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428 /* otherwise basic interlacing took place */ |
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429 for (i = 0; i < numsamples; i++) { |
a27a580f292e
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|
430 int16_t left, right; |
a27a580f292e
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|
431 |
5368
a31f6ae7e83d
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vitor
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5367
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432 left = buffer[0][i]; |
a31f6ae7e83d
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5367
diff
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433 right = buffer[1][i]; |
2542
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|
434 |
a27a580f292e
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parents:
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|
435 buffer_out[i*numchannels] = left; |
a27a580f292e
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diff
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|
436 buffer_out[i*numchannels + 1] = right; |
a27a580f292e
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|
437 } |
a27a580f292e
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parents:
diff
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|
438 } |
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439 |
2544
8c426f5a09ae
decoder works fine now, when fed properly-sized chunks by the demuxer;
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2543
diff
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440 static int alac_decode_frame(AVCodecContext *avctx, |
8c426f5a09ae
decoder works fine now, when fed properly-sized chunks by the demuxer;
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441 void *outbuffer, int *outputsize, |
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decoder works fine now, when fed properly-sized chunks by the demuxer;
melanson
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442 uint8_t *inbuffer, int input_buffer_size) |
2543
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443 { |
2559
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melanson
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2546
diff
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444 ALACContext *alac = avctx->priv_data; |
2544
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decoder works fine now, when fed properly-sized chunks by the demuxer;
melanson
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2543
diff
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|
445 |
2542
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|
446 int channels; |
2543
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do not use a variable before proper initialization
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diff
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447 int32_t outputsamples; |
5271 | 448 int hassize; |
449 int readsamplesize; | |
450 int wasted_bytes; | |
451 int isnotcompressed; | |
5358 | 452 uint8_t interlacing_shift; |
453 uint8_t interlacing_leftweight; | |
2542
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|
454 |
2544
8c426f5a09ae
decoder works fine now, when fed properly-sized chunks by the demuxer;
melanson
parents:
2543
diff
changeset
|
455 /* short-circuit null buffers */ |
8c426f5a09ae
decoder works fine now, when fed properly-sized chunks by the demuxer;
melanson
parents:
2543
diff
changeset
|
456 if (!inbuffer || !input_buffer_size) |
8c426f5a09ae
decoder works fine now, when fed properly-sized chunks by the demuxer;
melanson
parents:
2543
diff
changeset
|
457 return input_buffer_size; |
8c426f5a09ae
decoder works fine now, when fed properly-sized chunks by the demuxer;
melanson
parents:
2543
diff
changeset
|
458 |
2542
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first pass at ALAC decoder from David Hammerton; while David's original
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parents:
diff
changeset
|
459 /* initialize from the extradata */ |
2559
7d8ba28e18d9
replace custom bit-reading functions with FFmpeg's internal function;
melanson
parents:
2546
diff
changeset
|
460 if (!alac->context_initialized) { |
7d8ba28e18d9
replace custom bit-reading functions with FFmpeg's internal function;
melanson
parents:
2546
diff
changeset
|
461 if (alac->avctx->extradata_size != ALAC_EXTRADATA_SIZE) { |
3135 | 462 av_log(avctx, AV_LOG_ERROR, "alac: expected %d extradata bytes\n", |
2542
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first pass at ALAC decoder from David Hammerton; while David's original
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parents:
diff
changeset
|
463 ALAC_EXTRADATA_SIZE); |
a27a580f292e
first pass at ALAC decoder from David Hammerton; while David's original
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parents:
diff
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|
464 return input_buffer_size; |
a27a580f292e
first pass at ALAC decoder from David Hammerton; while David's original
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parents:
diff
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|
465 } |
4226
43ebe9279fa0
fix some potential security issues, patch by Matthias Hopf, mat at mshopf dot de
bcoudurier
parents:
3947
diff
changeset
|
466 if (alac_set_info(alac)) { |
43ebe9279fa0
fix some potential security issues, patch by Matthias Hopf, mat at mshopf dot de
bcoudurier
parents:
3947
diff
changeset
|
467 av_log(avctx, AV_LOG_ERROR, "alac: set_info failed\n"); |
43ebe9279fa0
fix some potential security issues, patch by Matthias Hopf, mat at mshopf dot de
bcoudurier
parents:
3947
diff
changeset
|
468 return input_buffer_size; |
43ebe9279fa0
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bcoudurier
parents:
3947
diff
changeset
|
469 } |
2559
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replace custom bit-reading functions with FFmpeg's internal function;
melanson
parents:
2546
diff
changeset
|
470 alac->context_initialized = 1; |
2542
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diff
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|
471 } |
2543
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do not use a variable before proper initialization
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parents:
2542
diff
changeset
|
472 |
2559
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melanson
parents:
2546
diff
changeset
|
473 init_get_bits(&alac->gb, inbuffer, input_buffer_size * 8); |
2542
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|
474 |
5298
e00f7048c50c
Make "channels" variable mean the number of channels, not the number of
vitor
parents:
5292
diff
changeset
|
475 channels = get_bits(&alac->gb, 3) + 1; |
5362 | 476 if (channels > MAX_CHANNELS) { |
477 av_log(avctx, AV_LOG_ERROR, "channels > %d not supported\n", | |
478 MAX_CHANNELS); | |
479 return input_buffer_size; | |
480 } | |
2542
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first pass at ALAC decoder from David Hammerton; while David's original
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diff
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|
481 |
5361 | 482 /* 2^result = something to do with output waiting. |
483 * perhaps matters if we read > 1 frame in a pass? | |
484 */ | |
5518 | 485 skip_bits(&alac->gb, 4); |
5361 | 486 |
5518 | 487 skip_bits(&alac->gb, 12); /* unknown, skip 12 bits */ |
2542
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first pass at ALAC decoder from David Hammerton; while David's original
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diff
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|
488 |
5361 | 489 /* the output sample size is stored soon */ |
5513 | 490 hassize = get_bits1(&alac->gb); |
2542
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first pass at ALAC decoder from David Hammerton; while David's original
melanson
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diff
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|
491 |
5361 | 492 wasted_bytes = get_bits(&alac->gb, 2); /* unknown ? */ |
2542
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first pass at ALAC decoder from David Hammerton; while David's original
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diff
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|
493 |
5361 | 494 /* whether the frame is compressed */ |
5513 | 495 isnotcompressed = get_bits1(&alac->gb); |
2542
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diff
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|
496 |
5361 | 497 if (hassize) { |
498 /* now read the number of samples as a 32bit integer */ | |
499 outputsamples = get_bits(&alac->gb, 32); | |
500 } else | |
501 outputsamples = alac->setinfo_max_samples_per_frame; | |
2542
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first pass at ALAC decoder from David Hammerton; while David's original
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diff
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|
502 |
5361 | 503 *outputsize = outputsamples * alac->bytespersample; |
504 readsamplesize = alac->setinfo_sample_size - (wasted_bytes * 8) + channels - 1; | |
5271 | 505 |
5361 | 506 if (!isnotcompressed) { |
507 /* so it is compressed */ | |
508 int16_t predictor_coef_table[channels][32]; | |
509 int predictor_coef_num[channels]; | |
510 int prediction_type[channels]; | |
511 int prediction_quantitization[channels]; | |
512 int ricemodifier[channels]; | |
513 int i, chan; | |
2542
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first pass at ALAC decoder from David Hammerton; while David's original
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diff
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|
514 |
5361 | 515 interlacing_shift = get_bits(&alac->gb, 8); |
516 interlacing_leftweight = get_bits(&alac->gb, 8); | |
2542
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517 |
5361 | 518 for (chan = 0; chan < channels; chan++) { |
5357
07c6fa478918
Remove code duplication. Based on a patch by Matthieu Castet.
vitor
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5298
diff
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519 prediction_type[chan] = get_bits(&alac->gb, 4); |
07c6fa478918
Remove code duplication. Based on a patch by Matthieu Castet.
vitor
parents:
5298
diff
changeset
|
520 prediction_quantitization[chan] = get_bits(&alac->gb, 4); |
2542
a27a580f292e
first pass at ALAC decoder from David Hammerton; while David's original
melanson
parents:
diff
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|
521 |
5357
07c6fa478918
Remove code duplication. Based on a patch by Matthieu Castet.
vitor
parents:
5298
diff
changeset
|
522 ricemodifier[chan] = get_bits(&alac->gb, 3); |
07c6fa478918
Remove code duplication. Based on a patch by Matthieu Castet.
vitor
parents:
5298
diff
changeset
|
523 predictor_coef_num[chan] = get_bits(&alac->gb, 5); |
2542
a27a580f292e
first pass at ALAC decoder from David Hammerton; while David's original
melanson
parents:
diff
changeset
|
524 |
a27a580f292e
first pass at ALAC decoder from David Hammerton; while David's original
melanson
parents:
diff
changeset
|
525 /* read the predictor table */ |
5365 | 526 for (i = 0; i < predictor_coef_num[chan]; i++) |
5357
07c6fa478918
Remove code duplication. Based on a patch by Matthieu Castet.
vitor
parents:
5298
diff
changeset
|
527 predictor_coef_table[chan][i] = (int16_t)get_bits(&alac->gb, 16); |
5361 | 528 } |
2542
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first pass at ALAC decoder from David Hammerton; while David's original
melanson
parents:
diff
changeset
|
529 |
5400 | 530 if (wasted_bytes) |
5361 | 531 av_log(avctx, AV_LOG_ERROR, "FIXME: unimplemented, unhandling of wasted_bytes\n"); |
2542
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532 |
5361 | 533 for (chan = 0; chan < channels; chan++) { |
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534 bastardized_rice_decompress(alac, |
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535 alac->predicterror_buffer[chan], |
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536 outputsamples, |
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537 readsamplesize, |
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538 alac->setinfo_rice_initialhistory, |
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539 alac->setinfo_rice_kmodifier, |
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540 ricemodifier[chan] * alac->setinfo_rice_historymult / 4, |
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541 (1 << alac->setinfo_rice_kmodifier) - 1); |
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542 |
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543 if (prediction_type[chan] == 0) { |
5361 | 544 /* adaptive fir */ |
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545 predictor_decompress_fir_adapt(alac->predicterror_buffer[chan], |
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546 alac->outputsamples_buffer[chan], |
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547 outputsamples, |
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548 readsamplesize, |
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549 predictor_coef_table[chan], |
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550 predictor_coef_num[chan], |
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551 prediction_quantitization[chan]); |
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552 } else { |
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553 av_log(avctx, AV_LOG_ERROR, "FIXME: unhandled prediction type: %i\n", prediction_type[chan]); |
5366 | 554 /* I think the only other prediction type (or perhaps this is |
555 * just a boolean?) runs adaptive fir twice.. like: | |
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556 * predictor_decompress_fir_adapt(predictor_error, tempout, ...) |
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557 * predictor_decompress_fir_adapt(predictor_error, outputsamples ...) |
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558 * little strange.. |
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559 */ |
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560 } |
5361 | 561 } |
562 } else { | |
563 /* not compressed, easy case */ | |
564 if (alac->setinfo_sample_size <= 16) { | |
565 int i, chan; | |
5366 | 566 for (chan = 0; chan < channels; chan++) |
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567 for (i = 0; i < outputsamples; i++) { |
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568 int32_t audiobits; |
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569 |
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570 audiobits = get_bits(&alac->gb, alac->setinfo_sample_size); |
5617 | 571 audiobits = extend_sign32(audiobits, readsamplesize); |
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572 |
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573 alac->outputsamples_buffer[chan][i] = audiobits; |
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574 } |
5361 | 575 } else { |
576 int i, chan; | |
5366 | 577 for (chan = 0; chan < channels; chan++) |
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578 for (i = 0; i < outputsamples; i++) { |
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579 int32_t audiobits; |
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580 |
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581 audiobits = get_bits(&alac->gb, 16); |
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582 /* special case of sign extension.. |
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583 * as we'll be ORing the low 16bits into this */ |
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584 audiobits = audiobits << 16; |
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585 audiobits = audiobits >> (32 - alac->setinfo_sample_size); |
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586 audiobits |= get_bits(&alac->gb, alac->setinfo_sample_size - 16); |
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587 |
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588 alac->outputsamples_buffer[chan][i] = audiobits; |
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589 } |
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590 } |
5361 | 591 /* wasted_bytes = 0; */ |
592 interlacing_shift = 0; | |
593 interlacing_leftweight = 0; | |
594 } | |
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595 |
5361 | 596 switch(alac->setinfo_sample_size) { |
5400 | 597 case 16: |
5361 | 598 if (channels == 2) { |
5504 | 599 reconstruct_stereo_16(alac->outputsamples_buffer, |
5506 | 600 (int16_t*)outbuffer, |
601 alac->numchannels, | |
602 outputsamples, | |
603 interlacing_shift, | |
604 interlacing_leftweight); | |
5361 | 605 } else { |
606 int i; | |
607 for (i = 0; i < outputsamples; i++) { | |
608 int16_t sample = alac->outputsamples_buffer[0][i]; | |
609 ((int16_t*)outbuffer)[i * alac->numchannels] = sample; | |
610 } | |
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611 } |
5361 | 612 break; |
613 case 20: | |
614 case 24: | |
5624 | 615 // It is not clear if there exist any encoder that creates 24 bit ALAC |
616 // files. iTunes convert 24 bit raw files to 16 bit before encoding. | |
5361 | 617 case 32: |
618 av_log(avctx, AV_LOG_ERROR, "FIXME: unimplemented sample size %i\n", alac->setinfo_sample_size); | |
619 break; | |
620 default: | |
621 break; | |
622 } | |
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623 |
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624 return input_buffer_size; |
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625 } |
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626 |
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627 static int alac_decode_init(AVCodecContext * avctx) |
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628 { |
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629 ALACContext *alac = avctx->priv_data; |
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630 alac->avctx = avctx; |
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631 alac->context_initialized = 0; |
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632 |
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633 alac->samplesize = alac->avctx->bits_per_sample; |
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634 alac->numchannels = alac->avctx->channels; |
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635 alac->bytespersample = (alac->samplesize / 8) * alac->numchannels; |
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636 |
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637 return 0; |
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638 } |
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639 |
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640 static int alac_decode_close(AVCodecContext *avctx) |
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641 { |
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642 ALACContext *alac = avctx->priv_data; |
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643 |
5272 | 644 int chan; |
645 for (chan = 0; chan < MAX_CHANNELS; chan++) { | |
646 av_free(alac->predicterror_buffer[chan]); | |
647 av_free(alac->outputsamples_buffer[chan]); | |
648 } | |
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649 |
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650 return 0; |
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651 } |
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652 |
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653 AVCodec alac_decoder = { |
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654 "alac", |
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655 CODEC_TYPE_AUDIO, |
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656 CODEC_ID_ALAC, |
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657 sizeof(ALACContext), |
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658 alac_decode_init, |
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659 NULL, |
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660 alac_decode_close, |
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661 alac_decode_frame, |
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662 }; |