Mercurial > libavcodec.hg
annotate alac.c @ 6710:a4104482ceef libavcodec
Add long names to many AVCodec declarations.
patch by Stefano Sabatini, stefano.sabatini-lala poste it
| author | diego |
|---|---|
| date | Sun, 27 Apr 2008 10:52:44 +0000 |
| parents | 25a963680a88 |
| children | 81ec037b6151 |
| 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 numchannels; |
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72 int bytespersample; |
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73 |
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74 /* buffers */ |
| 5272 | 75 int32_t *predicterror_buffer[MAX_CHANNELS]; |
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76 |
| 5272 | 77 int32_t *outputsamples_buffer[MAX_CHANNELS]; |
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78 |
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79 /* stuff from setinfo */ |
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80 uint32_t setinfo_max_samples_per_frame; /* 0x1000 = 4096 */ /* max samples per frame? */ |
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81 uint8_t setinfo_sample_size; /* 0x10 */ |
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82 uint8_t setinfo_rice_historymult; /* 0x28 */ |
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83 uint8_t setinfo_rice_initialhistory; /* 0x0a */ |
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84 uint8_t setinfo_rice_kmodifier; /* 0x0e */ |
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85 /* end setinfo stuff */ |
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86 |
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87 } ALACContext; |
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88 |
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89 static void allocate_buffers(ALACContext *alac) |
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90 { |
| 5272 | 91 int chan; |
| 92 for (chan = 0; chan < MAX_CHANNELS; chan++) { | |
| 93 alac->predicterror_buffer[chan] = | |
| 94 av_malloc(alac->setinfo_max_samples_per_frame * 4); | |
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95 |
| 5272 | 96 alac->outputsamples_buffer[chan] = |
| 97 av_malloc(alac->setinfo_max_samples_per_frame * 4); | |
| 98 } | |
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99 } |
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100 |
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101 static int alac_set_info(ALACContext *alac) |
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102 { |
| 6221 | 103 const unsigned char *ptr = alac->avctx->extradata; |
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104 |
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105 ptr += 4; /* size */ |
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106 ptr += 4; /* alac */ |
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107 ptr += 4; /* 0 ? */ |
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108 |
| 4364 | 109 if(AV_RB32(ptr) >= UINT_MAX/4){ |
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110 av_log(alac->avctx, AV_LOG_ERROR, "setinfo_max_samples_per_frame too large\n"); |
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111 return -1; |
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112 } |
| 5276 | 113 |
| 114 /* buffer size / 2 ? */ | |
| 115 alac->setinfo_max_samples_per_frame = bytestream_get_be32(&ptr); | |
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116 ptr++; /* ??? */ |
| 5360 | 117 alac->setinfo_sample_size = *ptr++; |
| 118 alac->setinfo_rice_historymult = *ptr++; | |
| 119 alac->setinfo_rice_initialhistory = *ptr++; | |
| 120 alac->setinfo_rice_kmodifier = *ptr++; | |
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121 ptr++; /* channels? */ |
| 6633 | 122 bytestream_get_be16(&ptr); /* ??? */ |
| 123 bytestream_get_be32(&ptr); /* max coded frame size */ | |
| 124 bytestream_get_be32(&ptr); /* bitrate ? */ | |
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125 bytestream_get_be32(&ptr); /* samplerate */ |
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126 |
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127 allocate_buffers(alac); |
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128 |
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129 return 0; |
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130 } |
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131 |
| 6626 | 132 static inline int decode_scalar(GetBitContext *gb, int k, int limit, int readsamplesize){ |
| 133 /* read x - number of 1s before 0 represent the rice */ | |
| 134 int x = get_unary_0_9(gb); | |
| 135 | |
| 136 if (x > 8) { /* RICE THRESHOLD */ | |
| 137 /* use alternative encoding */ | |
| 138 x = get_bits(gb, readsamplesize); | |
| 139 } else { | |
| 6627 | 140 if (k >= limit) |
| 141 k = limit; | |
| 6625 | 142 |
| 6627 | 143 if (k != 1) { |
| 144 int extrabits = show_bits(gb, k); | |
| 6625 | 145 |
| 6627 | 146 /* multiply x by 2^k - 1, as part of their strange algorithm */ |
| 147 x = (x << k) - x; | |
| 6625 | 148 |
| 6627 | 149 if (extrabits > 1) { |
| 150 x += extrabits - 1; | |
| 151 skip_bits(gb, k); | |
| 152 } else | |
| 153 skip_bits(gb, k - 1); | |
| 154 } | |
| 6626 | 155 } |
| 6625 | 156 return x; |
| 157 } | |
| 158 | |
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159 static void bastardized_rice_decompress(ALACContext *alac, |
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160 int32_t *output_buffer, |
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161 int output_size, |
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162 int readsamplesize, /* arg_10 */ |
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163 int rice_initialhistory, /* arg424->b */ |
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164 int rice_kmodifier, /* arg424->d */ |
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165 int rice_historymult, /* arg424->c */ |
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166 int rice_kmodifier_mask /* arg424->e */ |
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167 ) |
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168 { |
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169 int output_count; |
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170 unsigned int history = rice_initialhistory; |
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171 int sign_modifier = 0; |
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172 |
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173 for (output_count = 0; output_count < output_size; output_count++) { |
| 5628 | 174 int32_t x; |
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175 int32_t x_modified; |
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176 int32_t final_val; |
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177 |
| 6627 | 178 /* standard rice encoding */ |
| 179 int k; /* size of extra bits */ | |
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180 |
| 6627 | 181 /* read k, that is bits as is */ |
| 6632 | 182 k = av_log2((history >> 9) + 3); |
| 6627 | 183 x= decode_scalar(&alac->gb, k, rice_kmodifier, readsamplesize); |
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184 |
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185 x_modified = sign_modifier + x; |
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186 final_val = (x_modified + 1) / 2; |
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187 if (x_modified & 1) final_val *= -1; |
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188 |
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189 output_buffer[output_count] = final_val; |
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190 |
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191 sign_modifier = 0; |
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192 |
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193 /* now update the history */ |
| 5365 | 194 history += x_modified * rice_historymult |
| 195 - ((history * rice_historymult) >> 9); | |
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196 |
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197 if (x_modified > 0xffff) |
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198 history = 0xffff; |
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199 |
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200 /* special case: there may be compressed blocks of 0 */ |
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201 if ((history < 128) && (output_count+1 < output_size)) { |
| 6626 | 202 int block_size, k; |
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203 |
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204 sign_modifier = 1; |
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205 |
| 6632 | 206 k = 7 - av_log2(history) + ((history + 16) >> 6 /* / 64 */); |
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207 |
| 6627 | 208 block_size= decode_scalar(&alac->gb, k, rice_kmodifier, 16); |
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209 |
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210 if (block_size > 0) { |
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211 memset(&output_buffer[output_count+1], 0, block_size * 4); |
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212 output_count += block_size; |
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213 } |
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214 |
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215 if (block_size > 0xffff) |
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216 sign_modifier = 0; |
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217 |
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218 history = 0; |
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219 } |
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220 } |
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221 } |
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222 |
| 5617 | 223 static inline int32_t extend_sign32(int32_t val, int bits) |
| 5616 | 224 { |
| 225 return (val << (32 - bits)) >> (32 - bits); | |
| 226 } | |
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227 |
| 5616 | 228 static inline int sign_only(int v) |
| 229 { | |
| 230 return v ? FFSIGN(v) : 0; | |
| 231 } | |
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232 |
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233 static void predictor_decompress_fir_adapt(int32_t *error_buffer, |
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234 int32_t *buffer_out, |
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235 int output_size, |
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236 int readsamplesize, |
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237 int16_t *predictor_coef_table, |
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238 int predictor_coef_num, |
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239 int predictor_quantitization) |
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240 { |
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241 int i; |
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242 |
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243 /* first sample always copies */ |
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244 *buffer_out = *error_buffer; |
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245 |
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246 if (!predictor_coef_num) { |
| 5365 | 247 if (output_size <= 1) |
| 248 return; | |
| 249 | |
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250 memcpy(buffer_out+1, error_buffer+1, (output_size-1) * 4); |
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251 return; |
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252 } |
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253 |
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254 if (predictor_coef_num == 0x1f) { /* 11111 - max value of predictor_coef_num */ |
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255 /* second-best case scenario for fir decompression, |
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256 * error describes a small difference from the previous sample only |
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257 */ |
| 5365 | 258 if (output_size <= 1) |
| 259 return; | |
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260 for (i = 0; i < output_size - 1; i++) { |
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261 int32_t prev_value; |
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262 int32_t error_value; |
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263 |
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264 prev_value = buffer_out[i]; |
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265 error_value = error_buffer[i+1]; |
| 5365 | 266 buffer_out[i+1] = |
| 5617 | 267 extend_sign32((prev_value + error_value), readsamplesize); |
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268 } |
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269 return; |
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270 } |
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271 |
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272 /* read warm-up samples */ |
| 5365 | 273 if (predictor_coef_num > 0) |
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274 for (i = 0; i < predictor_coef_num; i++) { |
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275 int32_t val; |
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276 |
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277 val = buffer_out[i] + error_buffer[i+1]; |
| 5617 | 278 val = extend_sign32(val, readsamplesize); |
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279 buffer_out[i+1] = val; |
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280 } |
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281 |
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282 #if 0 |
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283 /* 4 and 8 are very common cases (the only ones i've seen). these |
| 5963 | 284 * should be unrolled and optimized |
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285 */ |
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286 if (predictor_coef_num == 4) { |
| 5963 | 287 /* FIXME: optimized general case */ |
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288 return; |
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289 } |
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290 |
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291 if (predictor_coef_table == 8) { |
| 5963 | 292 /* FIXME: optimized general case */ |
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293 return; |
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294 } |
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295 #endif |
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296 |
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297 /* general case */ |
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298 if (predictor_coef_num > 0) { |
| 5365 | 299 for (i = predictor_coef_num + 1; i < output_size; i++) { |
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300 int j; |
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301 int sum = 0; |
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302 int outval; |
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303 int error_val = error_buffer[i]; |
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304 |
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305 for (j = 0; j < predictor_coef_num; j++) { |
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306 sum += (buffer_out[predictor_coef_num-j] - buffer_out[0]) * |
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307 predictor_coef_table[j]; |
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308 } |
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309 |
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310 outval = (1 << (predictor_quantitization-1)) + sum; |
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311 outval = outval >> predictor_quantitization; |
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312 outval = outval + buffer_out[0] + error_val; |
| 5617 | 313 outval = extend_sign32(outval, readsamplesize); |
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314 |
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315 buffer_out[predictor_coef_num+1] = outval; |
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316 |
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317 if (error_val > 0) { |
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318 int predictor_num = predictor_coef_num - 1; |
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319 |
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320 while (predictor_num >= 0 && error_val > 0) { |
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321 int val = buffer_out[0] - buffer_out[predictor_coef_num - predictor_num]; |
| 5616 | 322 int sign = sign_only(val); |
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323 |
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324 predictor_coef_table[predictor_num] -= sign; |
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325 |
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326 val *= sign; /* absolute value */ |
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327 |
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328 error_val -= ((val >> predictor_quantitization) * |
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329 (predictor_coef_num - predictor_num)); |
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330 |
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331 predictor_num--; |
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332 } |
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333 } else if (error_val < 0) { |
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334 int predictor_num = predictor_coef_num - 1; |
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335 |
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336 while (predictor_num >= 0 && error_val < 0) { |
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337 int val = buffer_out[0] - buffer_out[predictor_coef_num - predictor_num]; |
| 5616 | 338 int sign = - sign_only(val); |
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339 |
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340 predictor_coef_table[predictor_num] -= sign; |
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341 |
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342 val *= sign; /* neg value */ |
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343 |
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344 error_val -= ((val >> predictor_quantitization) * |
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345 (predictor_coef_num - predictor_num)); |
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346 |
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347 predictor_num--; |
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348 } |
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349 } |
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350 |
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351 buffer_out++; |
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352 } |
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353 } |
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354 } |
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355 |
| 5504 | 356 static void reconstruct_stereo_16(int32_t *buffer[MAX_CHANNELS], |
| 5505 | 357 int16_t *buffer_out, |
| 358 int numchannels, int numsamples, | |
| 359 uint8_t interlacing_shift, | |
| 360 uint8_t interlacing_leftweight) | |
|
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361 { |
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362 int i; |
| 5365 | 363 if (numsamples <= 0) |
| 364 return; | |
|
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365 |
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366 /* weighted interlacing */ |
|
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367 if (interlacing_leftweight) { |
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368 for (i = 0; i < numsamples; i++) { |
| 5372 | 369 int32_t a, b; |
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370 |
| 5372 | 371 a = buffer[0][i]; |
| 372 b = buffer[1][i]; | |
|
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373 |
| 5372 | 374 a -= (b * interlacing_leftweight) >> interlacing_shift; |
| 375 b += a; | |
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376 |
| 5372 | 377 buffer_out[i*numchannels] = b; |
| 378 buffer_out[i*numchannels + 1] = a; | |
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379 } |
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380 |
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381 return; |
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382 } |
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383 |
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384 /* otherwise basic interlacing took place */ |
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385 for (i = 0; i < numsamples; i++) { |
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386 int16_t left, right; |
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387 |
|
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388 left = buffer[0][i]; |
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389 right = buffer[1][i]; |
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390 |
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391 buffer_out[i*numchannels] = left; |
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392 buffer_out[i*numchannels + 1] = right; |
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393 } |
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394 } |
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395 |
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396 static int alac_decode_frame(AVCodecContext *avctx, |
|
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397 void *outbuffer, int *outputsize, |
| 6221 | 398 const uint8_t *inbuffer, int input_buffer_size) |
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399 { |
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400 ALACContext *alac = avctx->priv_data; |
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401 |
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402 int channels; |
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403 int32_t outputsamples; |
| 5271 | 404 int hassize; |
| 405 int readsamplesize; | |
| 406 int wasted_bytes; | |
| 407 int isnotcompressed; | |
| 5358 | 408 uint8_t interlacing_shift; |
| 409 uint8_t interlacing_leftweight; | |
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410 |
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411 /* short-circuit null buffers */ |
|
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412 if (!inbuffer || !input_buffer_size) |
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413 return input_buffer_size; |
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414 |
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415 /* initialize from the extradata */ |
|
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416 if (!alac->context_initialized) { |
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417 if (alac->avctx->extradata_size != ALAC_EXTRADATA_SIZE) { |
| 3135 | 418 av_log(avctx, AV_LOG_ERROR, "alac: expected %d extradata bytes\n", |
|
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419 ALAC_EXTRADATA_SIZE); |
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420 return input_buffer_size; |
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|
421 } |
|
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422 if (alac_set_info(alac)) { |
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423 av_log(avctx, AV_LOG_ERROR, "alac: set_info failed\n"); |
|
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424 return input_buffer_size; |
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425 } |
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|
426 alac->context_initialized = 1; |
|
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|
427 } |
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428 |
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429 init_get_bits(&alac->gb, inbuffer, input_buffer_size * 8); |
|
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|
430 |
|
5298
e00f7048c50c
Make "channels" variable mean the number of channels, not the number of
vitor
parents:
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diff
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|
431 channels = get_bits(&alac->gb, 3) + 1; |
| 5362 | 432 if (channels > MAX_CHANNELS) { |
| 433 av_log(avctx, AV_LOG_ERROR, "channels > %d not supported\n", | |
| 434 MAX_CHANNELS); | |
| 435 return input_buffer_size; | |
| 436 } | |
|
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|
437 |
| 5361 | 438 /* 2^result = something to do with output waiting. |
| 439 * perhaps matters if we read > 1 frame in a pass? | |
| 440 */ | |
| 5518 | 441 skip_bits(&alac->gb, 4); |
| 5361 | 442 |
| 5518 | 443 skip_bits(&alac->gb, 12); /* unknown, skip 12 bits */ |
|
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|
444 |
| 5361 | 445 /* the output sample size is stored soon */ |
| 5513 | 446 hassize = get_bits1(&alac->gb); |
|
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|
447 |
| 5361 | 448 wasted_bytes = get_bits(&alac->gb, 2); /* unknown ? */ |
|
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|
449 |
| 5361 | 450 /* whether the frame is compressed */ |
| 5513 | 451 isnotcompressed = get_bits1(&alac->gb); |
|
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|
452 |
| 5361 | 453 if (hassize) { |
| 454 /* now read the number of samples as a 32bit integer */ | |
| 455 outputsamples = get_bits(&alac->gb, 32); | |
| 456 } else | |
| 457 outputsamples = alac->setinfo_max_samples_per_frame; | |
|
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|
458 |
| 5361 | 459 *outputsize = outputsamples * alac->bytespersample; |
| 460 readsamplesize = alac->setinfo_sample_size - (wasted_bytes * 8) + channels - 1; | |
| 5271 | 461 |
| 5361 | 462 if (!isnotcompressed) { |
| 463 /* so it is compressed */ | |
| 464 int16_t predictor_coef_table[channels][32]; | |
| 465 int predictor_coef_num[channels]; | |
| 466 int prediction_type[channels]; | |
| 467 int prediction_quantitization[channels]; | |
| 468 int ricemodifier[channels]; | |
| 469 int i, chan; | |
|
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|
470 |
| 5361 | 471 interlacing_shift = get_bits(&alac->gb, 8); |
| 472 interlacing_leftweight = get_bits(&alac->gb, 8); | |
|
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|
473 |
| 5361 | 474 for (chan = 0; chan < channels; chan++) { |
|
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vitor
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|
475 prediction_type[chan] = get_bits(&alac->gb, 4); |
|
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|
476 prediction_quantitization[chan] = get_bits(&alac->gb, 4); |
|
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diff
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|
477 |
|
5357
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|
478 ricemodifier[chan] = get_bits(&alac->gb, 3); |
|
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vitor
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|
479 predictor_coef_num[chan] = get_bits(&alac->gb, 5); |
|
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|
480 |
|
a27a580f292e
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diff
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|
481 /* read the predictor table */ |
| 5365 | 482 for (i = 0; i < predictor_coef_num[chan]; i++) |
|
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vitor
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|
483 predictor_coef_table[chan][i] = (int16_t)get_bits(&alac->gb, 16); |
| 5361 | 484 } |
|
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diff
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|
485 |
| 5400 | 486 if (wasted_bytes) |
| 5361 | 487 av_log(avctx, AV_LOG_ERROR, "FIXME: unimplemented, unhandling of wasted_bytes\n"); |
|
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|
488 |
| 5361 | 489 for (chan = 0; chan < channels; chan++) { |
|
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diff
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|
490 bastardized_rice_decompress(alac, |
|
5357
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Remove code duplication. Based on a patch by Matthieu Castet.
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diff
changeset
|
491 alac->predicterror_buffer[chan], |
|
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|
492 outputsamples, |
|
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parents:
diff
changeset
|
493 readsamplesize, |
|
a27a580f292e
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melanson
parents:
diff
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|
494 alac->setinfo_rice_initialhistory, |
|
a27a580f292e
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diff
changeset
|
495 alac->setinfo_rice_kmodifier, |
|
5357
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Remove code duplication. Based on a patch by Matthieu Castet.
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|
496 ricemodifier[chan] * alac->setinfo_rice_historymult / 4, |
|
2542
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|
497 (1 << alac->setinfo_rice_kmodifier) - 1); |
|
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|
498 |
|
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|
499 if (prediction_type[chan] == 0) { |
| 5361 | 500 /* adaptive fir */ |
|
5357
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|
501 predictor_decompress_fir_adapt(alac->predicterror_buffer[chan], |
|
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|
502 alac->outputsamples_buffer[chan], |
|
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|
503 outputsamples, |
|
a27a580f292e
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parents:
diff
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|
504 readsamplesize, |
|
5357
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diff
changeset
|
505 predictor_coef_table[chan], |
|
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|
506 predictor_coef_num[chan], |
|
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|
507 prediction_quantitization[chan]); |
|
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|
508 } else { |
|
5357
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|
509 av_log(avctx, AV_LOG_ERROR, "FIXME: unhandled prediction type: %i\n", prediction_type[chan]); |
| 5366 | 510 /* I think the only other prediction type (or perhaps this is |
| 511 * just a boolean?) runs adaptive fir twice.. like: | |
|
5359
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diff
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|
512 * predictor_decompress_fir_adapt(predictor_error, tempout, ...) |
|
4743a1e95531
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|
513 * predictor_decompress_fir_adapt(predictor_error, outputsamples ...) |
|
4743a1e95531
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|
514 * little strange.. |
|
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|
515 */ |
|
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|
516 } |
| 5361 | 517 } |
| 518 } else { | |
| 519 /* not compressed, easy case */ | |
| 520 if (alac->setinfo_sample_size <= 16) { | |
| 521 int i, chan; | |
| 5366 | 522 for (chan = 0; chan < channels; chan++) |
|
2542
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changeset
|
523 for (i = 0; i < outputsamples; i++) { |
|
5357
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Remove code duplication. Based on a patch by Matthieu Castet.
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diff
changeset
|
524 int32_t audiobits; |
|
2542
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parents:
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|
525 |
|
5357
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diff
changeset
|
526 audiobits = get_bits(&alac->gb, alac->setinfo_sample_size); |
| 5617 | 527 audiobits = extend_sign32(audiobits, readsamplesize); |
|
5359
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|
528 |
|
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diff
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|
529 alac->outputsamples_buffer[chan][i] = audiobits; |
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530 } |
| 5361 | 531 } else { |
| 532 int i, chan; | |
| 5366 | 533 for (chan = 0; chan < channels; chan++) |
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534 for (i = 0; i < outputsamples; i++) { |
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535 int32_t audiobits; |
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536 |
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537 audiobits = get_bits(&alac->gb, 16); |
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538 /* special case of sign extension.. |
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539 * as we'll be ORing the low 16bits into this */ |
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540 audiobits = audiobits << 16; |
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541 audiobits = audiobits >> (32 - alac->setinfo_sample_size); |
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542 audiobits |= get_bits(&alac->gb, alac->setinfo_sample_size - 16); |
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543 |
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544 alac->outputsamples_buffer[chan][i] = audiobits; |
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545 } |
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546 } |
| 5361 | 547 /* wasted_bytes = 0; */ |
| 548 interlacing_shift = 0; | |
| 549 interlacing_leftweight = 0; | |
| 550 } | |
| 6641 | 551 if (get_bits(&alac->gb, 3) != 7) |
| 552 av_log(avctx, AV_LOG_ERROR, "Error : Wrong End Of Frame\n"); | |
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553 |
| 5361 | 554 switch(alac->setinfo_sample_size) { |
| 5400 | 555 case 16: |
| 5361 | 556 if (channels == 2) { |
| 5504 | 557 reconstruct_stereo_16(alac->outputsamples_buffer, |
| 5506 | 558 (int16_t*)outbuffer, |
| 559 alac->numchannels, | |
| 560 outputsamples, | |
| 561 interlacing_shift, | |
| 562 interlacing_leftweight); | |
| 5361 | 563 } else { |
| 564 int i; | |
| 565 for (i = 0; i < outputsamples; i++) { | |
| 566 int16_t sample = alac->outputsamples_buffer[0][i]; | |
| 567 ((int16_t*)outbuffer)[i * alac->numchannels] = sample; | |
| 568 } | |
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569 } |
| 5361 | 570 break; |
| 571 case 20: | |
| 572 case 24: | |
| 5624 | 573 // It is not clear if there exist any encoder that creates 24 bit ALAC |
| 574 // files. iTunes convert 24 bit raw files to 16 bit before encoding. | |
| 5361 | 575 case 32: |
| 576 av_log(avctx, AV_LOG_ERROR, "FIXME: unimplemented sample size %i\n", alac->setinfo_sample_size); | |
| 577 break; | |
| 578 default: | |
| 579 break; | |
| 580 } | |
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581 |
| 6641 | 582 if (input_buffer_size * 8 - get_bits_count(&alac->gb) > 8) |
| 583 av_log(avctx, AV_LOG_ERROR, "Error : %d bits left\n", input_buffer_size * 8 - get_bits_count(&alac->gb)); | |
| 584 | |
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585 return input_buffer_size; |
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586 } |
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587 |
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588 static av_cold int alac_decode_init(AVCodecContext * avctx) |
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589 { |
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590 ALACContext *alac = avctx->priv_data; |
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591 alac->avctx = avctx; |
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592 alac->context_initialized = 0; |
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593 |
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594 alac->numchannels = alac->avctx->channels; |
| 6635 | 595 alac->bytespersample = (avctx->bits_per_sample / 8) * alac->numchannels; |
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596 |
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597 return 0; |
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598 } |
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599 |
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600 static av_cold int alac_decode_close(AVCodecContext *avctx) |
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601 { |
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602 ALACContext *alac = avctx->priv_data; |
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603 |
| 5272 | 604 int chan; |
| 605 for (chan = 0; chan < MAX_CHANNELS; chan++) { | |
| 606 av_free(alac->predicterror_buffer[chan]); | |
| 607 av_free(alac->outputsamples_buffer[chan]); | |
| 608 } | |
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609 |
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610 return 0; |
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611 } |
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612 |
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613 AVCodec alac_decoder = { |
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614 "alac", |
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615 CODEC_TYPE_AUDIO, |
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616 CODEC_ID_ALAC, |
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617 sizeof(ALACContext), |
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618 alac_decode_init, |
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619 NULL, |
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620 alac_decode_close, |
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621 alac_decode_frame, |
| 6710 | 622 .long_name = "ALAC (Apple Lossless Audio Codec)", |
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623 }; |
