Mercurial > libavcodec.hg
annotate alac.c @ 9291:6220870901f5 libavcodec
Move declaration of prn before any assignment.
author | benoit |
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date | Tue, 31 Mar 2009 07:37:55 +0000 |
parents | 42a7dfa5b3e4 |
children | 54bc8a2727b0 |
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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 libavcodec/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" |
9097 | 59 #include "mathops.h" |
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60 |
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61 #define ALAC_EXTRADATA_SIZE 36 |
5272 | 62 #define MAX_CHANNELS 2 |
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63 |
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64 typedef struct { |
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65 |
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66 AVCodecContext *avctx; |
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67 GetBitContext gb; |
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68 /* init to 0; first frame decode should initialize from extradata and |
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69 * set this to 1 */ |
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70 int context_initialized; |
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71 |
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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_sample_size; /* 0x10 */ |
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83 uint8_t setinfo_rice_historymult; /* 0x28 */ |
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84 uint8_t setinfo_rice_initialhistory; /* 0x0a */ |
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85 uint8_t setinfo_rice_kmodifier; /* 0x0e */ |
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86 /* end setinfo stuff */ |
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87 |
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88 } ALACContext; |
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89 |
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90 static void allocate_buffers(ALACContext *alac) |
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91 { |
5272 | 92 int chan; |
93 for (chan = 0; chan < MAX_CHANNELS; chan++) { | |
94 alac->predicterror_buffer[chan] = | |
95 av_malloc(alac->setinfo_max_samples_per_frame * 4); | |
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96 |
5272 | 97 alac->outputsamples_buffer[chan] = |
98 av_malloc(alac->setinfo_max_samples_per_frame * 4); | |
99 } | |
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100 } |
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101 |
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102 static int alac_set_info(ALACContext *alac) |
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103 { |
6221 | 104 const unsigned char *ptr = alac->avctx->extradata; |
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105 |
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106 ptr += 4; /* size */ |
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107 ptr += 4; /* alac */ |
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108 ptr += 4; /* 0 ? */ |
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109 |
4364 | 110 if(AV_RB32(ptr) >= UINT_MAX/4){ |
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111 av_log(alac->avctx, AV_LOG_ERROR, "setinfo_max_samples_per_frame too large\n"); |
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112 return -1; |
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113 } |
5276 | 114 |
115 /* buffer size / 2 ? */ | |
116 alac->setinfo_max_samples_per_frame = bytestream_get_be32(&ptr); | |
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117 ptr++; /* ??? */ |
5360 | 118 alac->setinfo_sample_size = *ptr++; |
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119 if (alac->setinfo_sample_size > 32) { |
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120 av_log(alac->avctx, AV_LOG_ERROR, "setinfo_sample_size too large\n"); |
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121 return -1; |
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122 } |
5360 | 123 alac->setinfo_rice_historymult = *ptr++; |
124 alac->setinfo_rice_initialhistory = *ptr++; | |
125 alac->setinfo_rice_kmodifier = *ptr++; | |
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126 ptr++; /* channels? */ |
6633 | 127 bytestream_get_be16(&ptr); /* ??? */ |
128 bytestream_get_be32(&ptr); /* max coded frame size */ | |
129 bytestream_get_be32(&ptr); /* bitrate ? */ | |
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130 bytestream_get_be32(&ptr); /* samplerate */ |
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131 |
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132 allocate_buffers(alac); |
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133 |
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134 return 0; |
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135 } |
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136 |
6626 | 137 static inline int decode_scalar(GetBitContext *gb, int k, int limit, int readsamplesize){ |
138 /* read x - number of 1s before 0 represent the rice */ | |
139 int x = get_unary_0_9(gb); | |
140 | |
141 if (x > 8) { /* RICE THRESHOLD */ | |
142 /* use alternative encoding */ | |
143 x = get_bits(gb, readsamplesize); | |
144 } else { | |
6627 | 145 if (k >= limit) |
146 k = limit; | |
6625 | 147 |
6627 | 148 if (k != 1) { |
149 int extrabits = show_bits(gb, k); | |
6625 | 150 |
6627 | 151 /* multiply x by 2^k - 1, as part of their strange algorithm */ |
152 x = (x << k) - x; | |
6625 | 153 |
6627 | 154 if (extrabits > 1) { |
155 x += extrabits - 1; | |
156 skip_bits(gb, k); | |
157 } else | |
158 skip_bits(gb, k - 1); | |
159 } | |
6626 | 160 } |
6625 | 161 return x; |
162 } | |
163 | |
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164 static void bastardized_rice_decompress(ALACContext *alac, |
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165 int32_t *output_buffer, |
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166 int output_size, |
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167 int readsamplesize, /* arg_10 */ |
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168 int rice_initialhistory, /* arg424->b */ |
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169 int rice_kmodifier, /* arg424->d */ |
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170 int rice_historymult, /* arg424->c */ |
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171 int rice_kmodifier_mask /* arg424->e */ |
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172 ) |
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173 { |
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174 int output_count; |
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175 unsigned int history = rice_initialhistory; |
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176 int sign_modifier = 0; |
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177 |
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178 for (output_count = 0; output_count < output_size; output_count++) { |
5628 | 179 int32_t x; |
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180 int32_t x_modified; |
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181 int32_t final_val; |
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182 |
6627 | 183 /* standard rice encoding */ |
184 int k; /* size of extra bits */ | |
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185 |
6627 | 186 /* read k, that is bits as is */ |
6632 | 187 k = av_log2((history >> 9) + 3); |
6627 | 188 x= decode_scalar(&alac->gb, k, rice_kmodifier, readsamplesize); |
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189 |
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190 x_modified = sign_modifier + x; |
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191 final_val = (x_modified + 1) / 2; |
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192 if (x_modified & 1) final_val *= -1; |
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193 |
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194 output_buffer[output_count] = final_val; |
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195 |
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196 sign_modifier = 0; |
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197 |
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198 /* now update the history */ |
5365 | 199 history += x_modified * rice_historymult |
200 - ((history * rice_historymult) >> 9); | |
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201 |
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202 if (x_modified > 0xffff) |
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203 history = 0xffff; |
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204 |
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205 /* special case: there may be compressed blocks of 0 */ |
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206 if ((history < 128) && (output_count+1 < output_size)) { |
6742 | 207 int k; |
208 unsigned int block_size; | |
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209 |
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210 sign_modifier = 1; |
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211 |
6632 | 212 k = 7 - av_log2(history) + ((history + 16) >> 6 /* / 64 */); |
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213 |
6627 | 214 block_size= decode_scalar(&alac->gb, k, rice_kmodifier, 16); |
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215 |
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216 if (block_size > 0) { |
6742 | 217 if(block_size >= output_size - output_count){ |
218 av_log(alac->avctx, AV_LOG_ERROR, "invalid zero block size of %d %d %d\n", block_size, output_size, output_count); | |
219 block_size= output_size - output_count - 1; | |
220 } | |
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221 memset(&output_buffer[output_count+1], 0, block_size * 4); |
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222 output_count += block_size; |
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223 } |
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224 |
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225 if (block_size > 0xffff) |
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226 sign_modifier = 0; |
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227 |
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228 history = 0; |
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229 } |
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230 } |
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231 } |
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232 |
5616 | 233 static inline int sign_only(int v) |
234 { | |
235 return v ? FFSIGN(v) : 0; | |
236 } | |
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237 |
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238 static void predictor_decompress_fir_adapt(int32_t *error_buffer, |
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239 int32_t *buffer_out, |
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240 int output_size, |
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241 int readsamplesize, |
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242 int16_t *predictor_coef_table, |
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243 int predictor_coef_num, |
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244 int predictor_quantitization) |
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245 { |
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246 int i; |
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247 |
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248 /* first sample always copies */ |
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249 *buffer_out = *error_buffer; |
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250 |
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251 if (!predictor_coef_num) { |
5365 | 252 if (output_size <= 1) |
253 return; | |
254 | |
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255 memcpy(buffer_out+1, error_buffer+1, (output_size-1) * 4); |
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256 return; |
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257 } |
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258 |
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259 if (predictor_coef_num == 0x1f) { /* 11111 - max value of predictor_coef_num */ |
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260 /* second-best case scenario for fir decompression, |
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261 * error describes a small difference from the previous sample only |
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262 */ |
5365 | 263 if (output_size <= 1) |
264 return; | |
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265 for (i = 0; i < output_size - 1; i++) { |
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266 int32_t prev_value; |
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267 int32_t error_value; |
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268 |
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269 prev_value = buffer_out[i]; |
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270 error_value = error_buffer[i+1]; |
5365 | 271 buffer_out[i+1] = |
9097 | 272 sign_extend((prev_value + error_value), readsamplesize); |
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273 } |
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274 return; |
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275 } |
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276 |
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277 /* read warm-up samples */ |
5365 | 278 if (predictor_coef_num > 0) |
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279 for (i = 0; i < predictor_coef_num; i++) { |
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280 int32_t val; |
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281 |
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282 val = buffer_out[i] + error_buffer[i+1]; |
9097 | 283 val = sign_extend(val, readsamplesize); |
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284 buffer_out[i+1] = val; |
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285 } |
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286 |
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287 #if 0 |
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288 /* 4 and 8 are very common cases (the only ones i've seen). these |
5963 | 289 * should be unrolled and optimized |
2542
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290 */ |
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291 if (predictor_coef_num == 4) { |
5963 | 292 /* FIXME: optimized general case */ |
2542
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293 return; |
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294 } |
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295 |
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296 if (predictor_coef_table == 8) { |
5963 | 297 /* FIXME: optimized general case */ |
2542
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298 return; |
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299 } |
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300 #endif |
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301 |
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302 /* general case */ |
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303 if (predictor_coef_num > 0) { |
5365 | 304 for (i = predictor_coef_num + 1; i < output_size; i++) { |
2542
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305 int j; |
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306 int sum = 0; |
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307 int outval; |
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308 int error_val = error_buffer[i]; |
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309 |
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310 for (j = 0; j < predictor_coef_num; j++) { |
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311 sum += (buffer_out[predictor_coef_num-j] - buffer_out[0]) * |
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312 predictor_coef_table[j]; |
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313 } |
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314 |
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315 outval = (1 << (predictor_quantitization-1)) + sum; |
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316 outval = outval >> predictor_quantitization; |
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317 outval = outval + buffer_out[0] + error_val; |
9097 | 318 outval = sign_extend(outval, readsamplesize); |
2542
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319 |
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320 buffer_out[predictor_coef_num+1] = outval; |
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321 |
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322 if (error_val > 0) { |
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323 int predictor_num = predictor_coef_num - 1; |
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324 |
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325 while (predictor_num >= 0 && error_val > 0) { |
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326 int val = buffer_out[0] - buffer_out[predictor_coef_num - predictor_num]; |
5616 | 327 int sign = sign_only(val); |
2542
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328 |
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329 predictor_coef_table[predictor_num] -= sign; |
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330 |
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331 val *= sign; /* absolute value */ |
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332 |
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333 error_val -= ((val >> predictor_quantitization) * |
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334 (predictor_coef_num - predictor_num)); |
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335 |
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336 predictor_num--; |
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337 } |
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338 } else if (error_val < 0) { |
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339 int predictor_num = predictor_coef_num - 1; |
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340 |
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341 while (predictor_num >= 0 && error_val < 0) { |
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342 int val = buffer_out[0] - buffer_out[predictor_coef_num - predictor_num]; |
5616 | 343 int sign = - sign_only(val); |
2542
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344 |
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345 predictor_coef_table[predictor_num] -= sign; |
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346 |
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347 val *= sign; /* neg value */ |
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348 |
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349 error_val -= ((val >> predictor_quantitization) * |
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350 (predictor_coef_num - predictor_num)); |
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351 |
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352 predictor_num--; |
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353 } |
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354 } |
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355 |
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356 buffer_out++; |
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357 } |
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358 } |
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359 } |
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360 |
5504 | 361 static void reconstruct_stereo_16(int32_t *buffer[MAX_CHANNELS], |
5505 | 362 int16_t *buffer_out, |
363 int numchannels, int numsamples, | |
364 uint8_t interlacing_shift, | |
365 uint8_t interlacing_leftweight) | |
2543
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366 { |
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367 int i; |
5365 | 368 if (numsamples <= 0) |
369 return; | |
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370 |
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371 /* weighted interlacing */ |
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|
372 if (interlacing_leftweight) { |
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373 for (i = 0; i < numsamples; i++) { |
5372 | 374 int32_t a, b; |
2542
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375 |
5372 | 376 a = buffer[0][i]; |
377 b = buffer[1][i]; | |
2542
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378 |
5372 | 379 a -= (b * interlacing_leftweight) >> interlacing_shift; |
380 b += a; | |
2542
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381 |
5372 | 382 buffer_out[i*numchannels] = b; |
383 buffer_out[i*numchannels + 1] = a; | |
2542
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384 } |
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385 |
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386 return; |
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387 } |
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388 |
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389 /* otherwise basic interlacing took place */ |
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390 for (i = 0; i < numsamples; i++) { |
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|
391 int16_t left, right; |
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|
392 |
5368
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Make deinterlace_16 receive an array as a parameter and not two separated vars
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393 left = buffer[0][i]; |
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394 right = buffer[1][i]; |
2542
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395 |
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396 buffer_out[i*numchannels] = left; |
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397 buffer_out[i*numchannels + 1] = right; |
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398 } |
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399 } |
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400 |
2544
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401 static int alac_decode_frame(AVCodecContext *avctx, |
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402 void *outbuffer, int *outputsize, |
6221 | 403 const uint8_t *inbuffer, int input_buffer_size) |
2543
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404 { |
2559
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405 ALACContext *alac = avctx->priv_data; |
2544
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406 |
2542
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407 int channels; |
6743 | 408 unsigned int outputsamples; |
5271 | 409 int hassize; |
7660
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410 unsigned int readsamplesize; |
5271 | 411 int wasted_bytes; |
412 int isnotcompressed; | |
5358 | 413 uint8_t interlacing_shift; |
414 uint8_t interlacing_leftweight; | |
2542
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415 |
2544
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416 /* short-circuit null buffers */ |
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417 if (!inbuffer || !input_buffer_size) |
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decoder works fine now, when fed properly-sized chunks by the demuxer;
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418 return input_buffer_size; |
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decoder works fine now, when fed properly-sized chunks by the demuxer;
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419 |
2542
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420 /* initialize from the extradata */ |
2559
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421 if (!alac->context_initialized) { |
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422 if (alac->avctx->extradata_size != ALAC_EXTRADATA_SIZE) { |
3135 | 423 av_log(avctx, AV_LOG_ERROR, "alac: expected %d extradata bytes\n", |
2542
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|
424 ALAC_EXTRADATA_SIZE); |
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|
425 return input_buffer_size; |
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|
426 } |
4226
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bcoudurier
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427 if (alac_set_info(alac)) { |
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fix some potential security issues, patch by Matthias Hopf, mat at mshopf dot de
bcoudurier
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428 av_log(avctx, AV_LOG_ERROR, "alac: set_info failed\n"); |
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bcoudurier
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429 return input_buffer_size; |
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bcoudurier
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430 } |
2559
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431 alac->context_initialized = 1; |
2542
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432 } |
2543
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433 |
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434 init_get_bits(&alac->gb, inbuffer, input_buffer_size * 8); |
2542
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435 |
5298
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Make "channels" variable mean the number of channels, not the number of
vitor
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5292
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436 channels = get_bits(&alac->gb, 3) + 1; |
5362 | 437 if (channels > MAX_CHANNELS) { |
438 av_log(avctx, AV_LOG_ERROR, "channels > %d not supported\n", | |
439 MAX_CHANNELS); | |
440 return input_buffer_size; | |
441 } | |
2542
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442 |
5361 | 443 /* 2^result = something to do with output waiting. |
444 * perhaps matters if we read > 1 frame in a pass? | |
445 */ | |
5518 | 446 skip_bits(&alac->gb, 4); |
5361 | 447 |
5518 | 448 skip_bits(&alac->gb, 12); /* unknown, skip 12 bits */ |
2542
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diff
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|
449 |
5361 | 450 /* the output sample size is stored soon */ |
5513 | 451 hassize = get_bits1(&alac->gb); |
2542
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452 |
5361 | 453 wasted_bytes = get_bits(&alac->gb, 2); /* unknown ? */ |
2542
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454 |
5361 | 455 /* whether the frame is compressed */ |
5513 | 456 isnotcompressed = get_bits1(&alac->gb); |
2542
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457 |
5361 | 458 if (hassize) { |
459 /* now read the number of samples as a 32bit integer */ | |
6773 | 460 outputsamples = get_bits_long(&alac->gb, 32); |
6743 | 461 if(outputsamples > alac->setinfo_max_samples_per_frame){ |
462 av_log(avctx, AV_LOG_ERROR, "outputsamples %d > %d\n", outputsamples, alac->setinfo_max_samples_per_frame); | |
463 return -1; | |
464 } | |
5361 | 465 } else |
466 outputsamples = alac->setinfo_max_samples_per_frame; | |
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467 |
6744 | 468 if(outputsamples > *outputsize / alac->bytespersample){ |
469 av_log(avctx, AV_LOG_ERROR, "sample buffer too small\n"); | |
470 return -1; | |
471 } | |
472 | |
5361 | 473 *outputsize = outputsamples * alac->bytespersample; |
474 readsamplesize = alac->setinfo_sample_size - (wasted_bytes * 8) + channels - 1; | |
7660
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jai_menon
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7451
diff
changeset
|
475 if (readsamplesize > MIN_CACHE_BITS) { |
6fe7ff34a4db
alac : check readsamplesize to ensure get_bits() doesn't
jai_menon
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7451
diff
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|
476 av_log(avctx, AV_LOG_ERROR, "readsamplesize too big (%d)\n", readsamplesize); |
6fe7ff34a4db
alac : check readsamplesize to ensure get_bits() doesn't
jai_menon
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7451
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|
477 return -1; |
6fe7ff34a4db
alac : check readsamplesize to ensure get_bits() doesn't
jai_menon
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7451
diff
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|
478 } |
5271 | 479 |
5361 | 480 if (!isnotcompressed) { |
481 /* so it is compressed */ | |
482 int16_t predictor_coef_table[channels][32]; | |
483 int predictor_coef_num[channels]; | |
484 int prediction_type[channels]; | |
485 int prediction_quantitization[channels]; | |
486 int ricemodifier[channels]; | |
487 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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|
488 |
5361 | 489 interlacing_shift = get_bits(&alac->gb, 8); |
490 interlacing_leftweight = get_bits(&alac->gb, 8); | |
2542
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first pass at ALAC decoder from David Hammerton; while David's original
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diff
changeset
|
491 |
5361 | 492 for (chan = 0; chan < channels; chan++) { |
5357
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Remove code duplication. Based on a patch by Matthieu Castet.
vitor
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493 prediction_type[chan] = get_bits(&alac->gb, 4); |
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vitor
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5298
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494 prediction_quantitization[chan] = get_bits(&alac->gb, 4); |
2542
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first pass at ALAC decoder from David Hammerton; while David's original
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diff
changeset
|
495 |
5357
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Remove code duplication. Based on a patch by Matthieu Castet.
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5298
diff
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|
496 ricemodifier[chan] = get_bits(&alac->gb, 3); |
07c6fa478918
Remove code duplication. Based on a patch by Matthieu Castet.
vitor
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5298
diff
changeset
|
497 predictor_coef_num[chan] = get_bits(&alac->gb, 5); |
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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|
498 |
a27a580f292e
first pass at ALAC decoder from David Hammerton; while David's original
melanson
parents:
diff
changeset
|
499 /* read the predictor table */ |
5365 | 500 for (i = 0; i < predictor_coef_num[chan]; i++) |
5357
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Remove code duplication. Based on a patch by Matthieu Castet.
vitor
parents:
5298
diff
changeset
|
501 predictor_coef_table[chan][i] = (int16_t)get_bits(&alac->gb, 16); |
5361 | 502 } |
2542
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first pass at ALAC decoder from David Hammerton; while David's original
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parents:
diff
changeset
|
503 |
5400 | 504 if (wasted_bytes) |
5361 | 505 av_log(avctx, AV_LOG_ERROR, "FIXME: unimplemented, unhandling of wasted_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
|
506 |
5361 | 507 for (chan = 0; chan < channels; 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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|
508 bastardized_rice_decompress(alac, |
5357
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Remove code duplication. Based on a patch by Matthieu Castet.
vitor
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5298
diff
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|
509 alac->predicterror_buffer[chan], |
2542
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first pass at ALAC decoder from David Hammerton; while David's original
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diff
changeset
|
510 outputsamples, |
a27a580f292e
first pass at ALAC decoder from David Hammerton; while David's original
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parents:
diff
changeset
|
511 readsamplesize, |
a27a580f292e
first pass at ALAC decoder from David Hammerton; while David's original
melanson
parents:
diff
changeset
|
512 alac->setinfo_rice_initialhistory, |
a27a580f292e
first pass at ALAC decoder from David Hammerton; while David's original
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parents:
diff
changeset
|
513 alac->setinfo_rice_kmodifier, |
5357
07c6fa478918
Remove code duplication. Based on a patch by Matthieu Castet.
vitor
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5298
diff
changeset
|
514 ricemodifier[chan] * alac->setinfo_rice_historymult / 4, |
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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|
515 (1 << alac->setinfo_rice_kmodifier) - 1); |
a27a580f292e
first pass at ALAC decoder from David Hammerton; while David's original
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|
516 |
5357
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vitor
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5298
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|
517 if (prediction_type[chan] == 0) { |
5361 | 518 /* adaptive fir */ |
5357
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vitor
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5298
diff
changeset
|
519 predictor_decompress_fir_adapt(alac->predicterror_buffer[chan], |
07c6fa478918
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vitor
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5298
diff
changeset
|
520 alac->outputsamples_buffer[chan], |
2542
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first pass at ALAC decoder from David Hammerton; while David's original
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parents:
diff
changeset
|
521 outputsamples, |
a27a580f292e
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parents:
diff
changeset
|
522 readsamplesize, |
5357
07c6fa478918
Remove code duplication. Based on a patch by Matthieu Castet.
vitor
parents:
5298
diff
changeset
|
523 predictor_coef_table[chan], |
07c6fa478918
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5298
diff
changeset
|
524 predictor_coef_num[chan], |
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5298
diff
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|
525 prediction_quantitization[chan]); |
2542
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first pass at ALAC decoder from David Hammerton; while David's original
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diff
changeset
|
526 } else { |
5357
07c6fa478918
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vitor
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5298
diff
changeset
|
527 av_log(avctx, AV_LOG_ERROR, "FIXME: unhandled prediction type: %i\n", prediction_type[chan]); |
5366 | 528 /* I think the only other prediction type (or perhaps this is |
529 * just a boolean?) runs adaptive fir twice.. like: | |
5359
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vitor
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5358
diff
changeset
|
530 * predictor_decompress_fir_adapt(predictor_error, tempout, ...) |
4743a1e95531
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vitor
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5358
diff
changeset
|
531 * predictor_decompress_fir_adapt(predictor_error, outputsamples ...) |
4743a1e95531
Remove more code duplication. Based on a patch by Matthieu Castet.
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5358
diff
changeset
|
532 * little strange.. |
4743a1e95531
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5358
diff
changeset
|
533 */ |
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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|
534 } |
5361 | 535 } |
536 } else { | |
537 /* not compressed, easy case */ | |
6779
a25842a4df43
cosmetics: indent after last commit. patch by matthieu castet <castet matthieu free fr>.
jbr
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6778
diff
changeset
|
538 int i, chan; |
7043
e1ed057dc7c3
fix verbatim mode decoding. patch by Jai Menon (realityman gmx net).
jbr
parents:
7040
diff
changeset
|
539 for (i = 0; i < outputsamples; i++) |
e1ed057dc7c3
fix verbatim mode decoding. patch by Jai Menon (realityman gmx net).
jbr
parents:
7040
diff
changeset
|
540 for (chan = 0; chan < channels; chan++) { |
6779
a25842a4df43
cosmetics: indent after last commit. patch by matthieu castet <castet matthieu free fr>.
jbr
parents:
6778
diff
changeset
|
541 int32_t audiobits; |
2542
a27a580f292e
first pass at ALAC decoder from David Hammerton; while David's original
melanson
parents:
diff
changeset
|
542 |
9102
42a7dfa5b3e4
alacdec: Simplify reading of uncompressed samples by using
jbr
parents:
9097
diff
changeset
|
543 audiobits = get_sbits_long(&alac->gb, alac->setinfo_sample_size); |
5359
4743a1e95531
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vitor
parents:
5358
diff
changeset
|
544 |
6779
a25842a4df43
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jbr
parents:
6778
diff
changeset
|
545 alac->outputsamples_buffer[chan][i] = audiobits; |
a25842a4df43
cosmetics: indent after last commit. patch by matthieu castet <castet matthieu free fr>.
jbr
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6778
diff
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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"); | |
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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|
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; |
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595 alac->bytespersample = 2 * alac->numchannels; |
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596 avctx->sample_fmt = SAMPLE_FMT_S16; |
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597 |
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598 return 0; |
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599 } |
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600 |
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601 static av_cold int alac_decode_close(AVCodecContext *avctx) |
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602 { |
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603 ALACContext *alac = avctx->priv_data; |
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604 |
5272 | 605 int chan; |
606 for (chan = 0; chan < MAX_CHANNELS; chan++) { | |
607 av_free(alac->predicterror_buffer[chan]); | |
608 av_free(alac->outputsamples_buffer[chan]); | |
609 } | |
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610 |
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611 return 0; |
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612 } |
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613 |
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614 AVCodec alac_decoder = { |
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615 "alac", |
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616 CODEC_TYPE_AUDIO, |
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617 CODEC_ID_ALAC, |
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618 sizeof(ALACContext), |
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619 alac_decode_init, |
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620 NULL, |
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621 alac_decode_close, |
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622 alac_decode_frame, |
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623 .long_name = NULL_IF_CONFIG_SMALL("ALAC (Apple Lossless Audio Codec)"), |
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624 }; |