Mercurial > libavcodec.hg
annotate alac.c @ 12266:48d6738904a9 libavcodec
Fix SPLATB_REG mess. Used to be a if/elseif/elseif/elseif spaghetti, so this
splits it into small optimization-specific macros which are selected for each
DSP function. The advantage of this approach is that the sse4 functions now
use the ssse3 codepath also without needing an explicit sse4 codepath.
author | rbultje |
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date | Sat, 24 Jul 2010 19:33:05 +0000 |
parents | 1d6362654d4c |
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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 |
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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" |
9428 | 56 #include "get_bits.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 |
11077 | 80 int32_t *wasted_bits_buffer[MAX_CHANNELS]; |
81 | |
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82 /* stuff from setinfo */ |
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83 uint32_t setinfo_max_samples_per_frame; /* 0x1000 = 4096 */ /* max samples per frame? */ |
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84 uint8_t setinfo_sample_size; /* 0x10 */ |
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85 uint8_t setinfo_rice_historymult; /* 0x28 */ |
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86 uint8_t setinfo_rice_initialhistory; /* 0x0a */ |
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87 uint8_t setinfo_rice_kmodifier; /* 0x0e */ |
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88 /* end setinfo stuff */ |
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89 |
11077 | 90 int wasted_bits; |
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91 } ALACContext; |
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92 |
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93 static void allocate_buffers(ALACContext *alac) |
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94 { |
5272 | 95 int chan; |
96 for (chan = 0; chan < MAX_CHANNELS; chan++) { | |
97 alac->predicterror_buffer[chan] = | |
98 av_malloc(alac->setinfo_max_samples_per_frame * 4); | |
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99 |
5272 | 100 alac->outputsamples_buffer[chan] = |
101 av_malloc(alac->setinfo_max_samples_per_frame * 4); | |
11077 | 102 |
103 alac->wasted_bits_buffer[chan] = av_malloc(alac->setinfo_max_samples_per_frame * 4); | |
5272 | 104 } |
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105 } |
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106 |
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107 static int alac_set_info(ALACContext *alac) |
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108 { |
6221 | 109 const unsigned char *ptr = alac->avctx->extradata; |
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110 |
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111 ptr += 4; /* size */ |
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112 ptr += 4; /* alac */ |
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113 ptr += 4; /* 0 ? */ |
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114 |
4364 | 115 if(AV_RB32(ptr) >= UINT_MAX/4){ |
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116 av_log(alac->avctx, AV_LOG_ERROR, "setinfo_max_samples_per_frame too large\n"); |
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117 return -1; |
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118 } |
5276 | 119 |
120 /* buffer size / 2 ? */ | |
121 alac->setinfo_max_samples_per_frame = bytestream_get_be32(&ptr); | |
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122 ptr++; /* ??? */ |
5360 | 123 alac->setinfo_sample_size = *ptr++; |
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124 if (alac->setinfo_sample_size > 32) { |
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125 av_log(alac->avctx, AV_LOG_ERROR, "setinfo_sample_size too large\n"); |
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126 return -1; |
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127 } |
5360 | 128 alac->setinfo_rice_historymult = *ptr++; |
129 alac->setinfo_rice_initialhistory = *ptr++; | |
130 alac->setinfo_rice_kmodifier = *ptr++; | |
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131 ptr++; /* channels? */ |
6633 | 132 bytestream_get_be16(&ptr); /* ??? */ |
133 bytestream_get_be32(&ptr); /* max coded frame size */ | |
134 bytestream_get_be32(&ptr); /* bitrate ? */ | |
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135 bytestream_get_be32(&ptr); /* samplerate */ |
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136 |
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137 allocate_buffers(alac); |
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138 |
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139 return 0; |
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140 } |
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141 |
6626 | 142 static inline int decode_scalar(GetBitContext *gb, int k, int limit, int readsamplesize){ |
143 /* read x - number of 1s before 0 represent the rice */ | |
144 int x = get_unary_0_9(gb); | |
145 | |
146 if (x > 8) { /* RICE THRESHOLD */ | |
147 /* use alternative encoding */ | |
148 x = get_bits(gb, readsamplesize); | |
149 } else { | |
6627 | 150 if (k >= limit) |
151 k = limit; | |
6625 | 152 |
6627 | 153 if (k != 1) { |
154 int extrabits = show_bits(gb, k); | |
6625 | 155 |
6627 | 156 /* multiply x by 2^k - 1, as part of their strange algorithm */ |
157 x = (x << k) - x; | |
6625 | 158 |
6627 | 159 if (extrabits > 1) { |
160 x += extrabits - 1; | |
161 skip_bits(gb, k); | |
162 } else | |
163 skip_bits(gb, k - 1); | |
164 } | |
6626 | 165 } |
6625 | 166 return x; |
167 } | |
168 | |
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169 static void bastardized_rice_decompress(ALACContext *alac, |
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170 int32_t *output_buffer, |
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171 int output_size, |
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172 int readsamplesize, /* arg_10 */ |
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173 int rice_initialhistory, /* arg424->b */ |
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174 int rice_kmodifier, /* arg424->d */ |
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175 int rice_historymult, /* arg424->c */ |
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176 int rice_kmodifier_mask /* arg424->e */ |
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177 ) |
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178 { |
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179 int output_count; |
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180 unsigned int history = rice_initialhistory; |
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181 int sign_modifier = 0; |
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182 |
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183 for (output_count = 0; output_count < output_size; output_count++) { |
5628 | 184 int32_t x; |
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185 int32_t x_modified; |
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186 int32_t final_val; |
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187 |
6627 | 188 /* standard rice encoding */ |
189 int k; /* size of extra bits */ | |
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190 |
6627 | 191 /* read k, that is bits as is */ |
6632 | 192 k = av_log2((history >> 9) + 3); |
6627 | 193 x= decode_scalar(&alac->gb, k, rice_kmodifier, readsamplesize); |
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194 |
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195 x_modified = sign_modifier + x; |
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196 final_val = (x_modified + 1) / 2; |
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197 if (x_modified & 1) final_val *= -1; |
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198 |
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199 output_buffer[output_count] = final_val; |
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200 |
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201 sign_modifier = 0; |
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202 |
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203 /* now update the history */ |
5365 | 204 history += x_modified * rice_historymult |
205 - ((history * rice_historymult) >> 9); | |
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206 |
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207 if (x_modified > 0xffff) |
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208 history = 0xffff; |
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209 |
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210 /* special case: there may be compressed blocks of 0 */ |
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211 if ((history < 128) && (output_count+1 < output_size)) { |
6742 | 212 int k; |
213 unsigned int block_size; | |
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214 |
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215 sign_modifier = 1; |
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216 |
6632 | 217 k = 7 - av_log2(history) + ((history + 16) >> 6 /* / 64 */); |
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218 |
6627 | 219 block_size= decode_scalar(&alac->gb, k, rice_kmodifier, 16); |
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220 |
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221 if (block_size > 0) { |
6742 | 222 if(block_size >= output_size - output_count){ |
223 av_log(alac->avctx, AV_LOG_ERROR, "invalid zero block size of %d %d %d\n", block_size, output_size, output_count); | |
224 block_size= output_size - output_count - 1; | |
225 } | |
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226 memset(&output_buffer[output_count+1], 0, block_size * 4); |
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227 output_count += block_size; |
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228 } |
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229 |
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230 if (block_size > 0xffff) |
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231 sign_modifier = 0; |
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232 |
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233 history = 0; |
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234 } |
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235 } |
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236 } |
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237 |
5616 | 238 static inline int sign_only(int v) |
239 { | |
240 return v ? FFSIGN(v) : 0; | |
241 } | |
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242 |
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243 static void predictor_decompress_fir_adapt(int32_t *error_buffer, |
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244 int32_t *buffer_out, |
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245 int output_size, |
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246 int readsamplesize, |
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247 int16_t *predictor_coef_table, |
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248 int predictor_coef_num, |
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249 int predictor_quantitization) |
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250 { |
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251 int i; |
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252 |
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253 /* first sample always copies */ |
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254 *buffer_out = *error_buffer; |
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255 |
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256 if (!predictor_coef_num) { |
5365 | 257 if (output_size <= 1) |
258 return; | |
259 | |
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260 memcpy(buffer_out+1, error_buffer+1, (output_size-1) * 4); |
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261 return; |
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262 } |
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263 |
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264 if (predictor_coef_num == 0x1f) { /* 11111 - max value of predictor_coef_num */ |
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265 /* second-best case scenario for fir decompression, |
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266 * error describes a small difference from the previous sample only |
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267 */ |
5365 | 268 if (output_size <= 1) |
269 return; | |
2542
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270 for (i = 0; i < output_size - 1; i++) { |
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271 int32_t prev_value; |
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272 int32_t error_value; |
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273 |
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274 prev_value = buffer_out[i]; |
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275 error_value = error_buffer[i+1]; |
5365 | 276 buffer_out[i+1] = |
9097 | 277 sign_extend((prev_value + error_value), readsamplesize); |
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278 } |
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279 return; |
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280 } |
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281 |
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282 /* read warm-up samples */ |
5365 | 283 if (predictor_coef_num > 0) |
2542
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284 for (i = 0; i < predictor_coef_num; i++) { |
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285 int32_t val; |
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286 |
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287 val = buffer_out[i] + error_buffer[i+1]; |
9097 | 288 val = sign_extend(val, readsamplesize); |
2542
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289 buffer_out[i+1] = val; |
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290 } |
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291 |
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292 #if 0 |
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293 /* 4 and 8 are very common cases (the only ones i've seen). these |
5963 | 294 * should be unrolled and optimized |
2542
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295 */ |
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296 if (predictor_coef_num == 4) { |
5963 | 297 /* FIXME: optimized general case */ |
2542
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298 return; |
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299 } |
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300 |
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301 if (predictor_coef_table == 8) { |
5963 | 302 /* FIXME: optimized general case */ |
2542
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303 return; |
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304 } |
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305 #endif |
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306 |
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307 /* general case */ |
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308 if (predictor_coef_num > 0) { |
5365 | 309 for (i = predictor_coef_num + 1; i < output_size; i++) { |
2542
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310 int j; |
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311 int sum = 0; |
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312 int outval; |
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313 int error_val = error_buffer[i]; |
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314 |
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315 for (j = 0; j < predictor_coef_num; j++) { |
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316 sum += (buffer_out[predictor_coef_num-j] - buffer_out[0]) * |
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317 predictor_coef_table[j]; |
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318 } |
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319 |
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320 outval = (1 << (predictor_quantitization-1)) + sum; |
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321 outval = outval >> predictor_quantitization; |
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322 outval = outval + buffer_out[0] + error_val; |
9097 | 323 outval = sign_extend(outval, readsamplesize); |
2542
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324 |
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325 buffer_out[predictor_coef_num+1] = outval; |
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326 |
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327 if (error_val > 0) { |
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328 int predictor_num = predictor_coef_num - 1; |
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329 |
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330 while (predictor_num >= 0 && error_val > 0) { |
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331 int val = buffer_out[0] - buffer_out[predictor_coef_num - predictor_num]; |
5616 | 332 int sign = sign_only(val); |
2542
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333 |
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334 predictor_coef_table[predictor_num] -= sign; |
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335 |
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336 val *= sign; /* absolute value */ |
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337 |
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338 error_val -= ((val >> predictor_quantitization) * |
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339 (predictor_coef_num - predictor_num)); |
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340 |
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341 predictor_num--; |
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342 } |
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343 } else if (error_val < 0) { |
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344 int predictor_num = predictor_coef_num - 1; |
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345 |
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346 while (predictor_num >= 0 && error_val < 0) { |
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347 int val = buffer_out[0] - buffer_out[predictor_coef_num - predictor_num]; |
5616 | 348 int sign = - sign_only(val); |
2542
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349 |
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350 predictor_coef_table[predictor_num] -= sign; |
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351 |
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352 val *= sign; /* neg value */ |
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353 |
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354 error_val -= ((val >> predictor_quantitization) * |
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355 (predictor_coef_num - predictor_num)); |
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356 |
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357 predictor_num--; |
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358 } |
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359 } |
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360 |
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361 buffer_out++; |
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362 } |
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363 } |
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364 } |
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365 |
5504 | 366 static void reconstruct_stereo_16(int32_t *buffer[MAX_CHANNELS], |
5505 | 367 int16_t *buffer_out, |
368 int numchannels, int numsamples, | |
369 uint8_t interlacing_shift, | |
370 uint8_t interlacing_leftweight) | |
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371 { |
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372 int i; |
5365 | 373 if (numsamples <= 0) |
374 return; | |
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375 |
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376 /* weighted interlacing */ |
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377 if (interlacing_leftweight) { |
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378 for (i = 0; i < numsamples; i++) { |
5372 | 379 int32_t a, b; |
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380 |
5372 | 381 a = buffer[0][i]; |
382 b = buffer[1][i]; | |
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383 |
5372 | 384 a -= (b * interlacing_leftweight) >> interlacing_shift; |
385 b += a; | |
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386 |
5372 | 387 buffer_out[i*numchannels] = b; |
388 buffer_out[i*numchannels + 1] = a; | |
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389 } |
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390 |
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391 return; |
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392 } |
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393 |
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394 /* otherwise basic interlacing took place */ |
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395 for (i = 0; i < numsamples; i++) { |
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396 int16_t left, right; |
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397 |
5368
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398 left = buffer[0][i]; |
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399 right = buffer[1][i]; |
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400 |
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401 buffer_out[i*numchannels] = left; |
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402 buffer_out[i*numchannels + 1] = right; |
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403 } |
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404 } |
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405 |
11077 | 406 static void decorrelate_stereo_24(int32_t *buffer[MAX_CHANNELS], |
407 int32_t *buffer_out, | |
408 int32_t *wasted_bits_buffer[MAX_CHANNELS], | |
409 int wasted_bits, | |
410 int numchannels, int numsamples, | |
411 uint8_t interlacing_shift, | |
412 uint8_t interlacing_leftweight) | |
413 { | |
414 int i; | |
415 | |
416 if (numsamples <= 0) | |
417 return; | |
418 | |
419 /* weighted interlacing */ | |
420 if (interlacing_leftweight) { | |
421 for (i = 0; i < numsamples; i++) { | |
422 int32_t a, b; | |
423 | |
424 a = buffer[0][i]; | |
425 b = buffer[1][i]; | |
426 | |
427 a -= (b * interlacing_leftweight) >> interlacing_shift; | |
428 b += a; | |
429 | |
430 if (wasted_bits) { | |
431 b = (b << wasted_bits) | wasted_bits_buffer[0][i]; | |
432 a = (a << wasted_bits) | wasted_bits_buffer[1][i]; | |
433 } | |
434 | |
435 buffer_out[i * numchannels] = b << 8; | |
436 buffer_out[i * numchannels + 1] = a << 8; | |
437 } | |
438 } else { | |
439 for (i = 0; i < numsamples; i++) { | |
440 int32_t left, right; | |
441 | |
442 left = buffer[0][i]; | |
443 right = buffer[1][i]; | |
444 | |
445 if (wasted_bits) { | |
446 left = (left << wasted_bits) | wasted_bits_buffer[0][i]; | |
447 right = (right << wasted_bits) | wasted_bits_buffer[1][i]; | |
448 } | |
449 | |
450 buffer_out[i * numchannels] = left << 8; | |
451 buffer_out[i * numchannels + 1] = right << 8; | |
452 } | |
453 } | |
454 } | |
455 | |
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456 static int alac_decode_frame(AVCodecContext *avctx, |
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457 void *outbuffer, int *outputsize, |
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458 AVPacket *avpkt) |
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459 { |
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460 const uint8_t *inbuffer = avpkt->data; |
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461 int input_buffer_size = avpkt->size; |
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462 ALACContext *alac = avctx->priv_data; |
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463 |
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464 int channels; |
6743 | 465 unsigned int outputsamples; |
5271 | 466 int hassize; |
7660
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467 unsigned int readsamplesize; |
5271 | 468 int isnotcompressed; |
5358 | 469 uint8_t interlacing_shift; |
470 uint8_t interlacing_leftweight; | |
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471 |
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472 /* short-circuit null buffers */ |
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473 if (!inbuffer || !input_buffer_size) |
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474 return input_buffer_size; |
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475 |
2542
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476 /* initialize from the extradata */ |
2559
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477 if (!alac->context_initialized) { |
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478 if (alac->avctx->extradata_size != ALAC_EXTRADATA_SIZE) { |
3135 | 479 av_log(avctx, AV_LOG_ERROR, "alac: expected %d extradata bytes\n", |
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480 ALAC_EXTRADATA_SIZE); |
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481 return input_buffer_size; |
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482 } |
4226
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483 if (alac_set_info(alac)) { |
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484 av_log(avctx, AV_LOG_ERROR, "alac: set_info failed\n"); |
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485 return input_buffer_size; |
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486 } |
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487 alac->context_initialized = 1; |
2542
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488 } |
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489 |
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490 init_get_bits(&alac->gb, inbuffer, input_buffer_size * 8); |
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491 |
5298
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492 channels = get_bits(&alac->gb, 3) + 1; |
5362 | 493 if (channels > MAX_CHANNELS) { |
494 av_log(avctx, AV_LOG_ERROR, "channels > %d not supported\n", | |
495 MAX_CHANNELS); | |
496 return input_buffer_size; | |
497 } | |
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498 |
5361 | 499 /* 2^result = something to do with output waiting. |
500 * perhaps matters if we read > 1 frame in a pass? | |
501 */ | |
5518 | 502 skip_bits(&alac->gb, 4); |
5361 | 503 |
5518 | 504 skip_bits(&alac->gb, 12); /* unknown, skip 12 bits */ |
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505 |
5361 | 506 /* the output sample size is stored soon */ |
5513 | 507 hassize = get_bits1(&alac->gb); |
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508 |
11077 | 509 alac->wasted_bits = get_bits(&alac->gb, 2) << 3; |
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510 |
5361 | 511 /* whether the frame is compressed */ |
5513 | 512 isnotcompressed = get_bits1(&alac->gb); |
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513 |
5361 | 514 if (hassize) { |
515 /* now read the number of samples as a 32bit integer */ | |
6773 | 516 outputsamples = get_bits_long(&alac->gb, 32); |
6743 | 517 if(outputsamples > alac->setinfo_max_samples_per_frame){ |
518 av_log(avctx, AV_LOG_ERROR, "outputsamples %d > %d\n", outputsamples, alac->setinfo_max_samples_per_frame); | |
519 return -1; | |
520 } | |
5361 | 521 } else |
522 outputsamples = alac->setinfo_max_samples_per_frame; | |
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523 |
11077 | 524 switch (alac->setinfo_sample_size) { |
525 case 16: avctx->sample_fmt = SAMPLE_FMT_S16; | |
526 alac->bytespersample = channels << 1; | |
527 break; | |
528 case 24: avctx->sample_fmt = SAMPLE_FMT_S32; | |
529 alac->bytespersample = channels << 2; | |
530 break; | |
531 default: av_log(avctx, AV_LOG_ERROR, "Sample depth %d is not supported.\n", | |
532 alac->setinfo_sample_size); | |
533 return -1; | |
534 } | |
535 | |
6744 | 536 if(outputsamples > *outputsize / alac->bytespersample){ |
537 av_log(avctx, AV_LOG_ERROR, "sample buffer too small\n"); | |
538 return -1; | |
539 } | |
540 | |
5361 | 541 *outputsize = outputsamples * alac->bytespersample; |
11077 | 542 readsamplesize = alac->setinfo_sample_size - (alac->wasted_bits) + channels - 1; |
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543 if (readsamplesize > MIN_CACHE_BITS) { |
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544 av_log(avctx, AV_LOG_ERROR, "readsamplesize too big (%d)\n", readsamplesize); |
6fe7ff34a4db
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545 return -1; |
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|
546 } |
5271 | 547 |
5361 | 548 if (!isnotcompressed) { |
549 /* so it is compressed */ | |
11940 | 550 int16_t predictor_coef_table[MAX_CHANNELS][32]; |
551 int predictor_coef_num[MAX_CHANNELS]; | |
552 int prediction_type[MAX_CHANNELS]; | |
553 int prediction_quantitization[MAX_CHANNELS]; | |
554 int ricemodifier[MAX_CHANNELS]; | |
5361 | 555 int i, chan; |
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556 |
5361 | 557 interlacing_shift = get_bits(&alac->gb, 8); |
558 interlacing_leftweight = get_bits(&alac->gb, 8); | |
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559 |
5361 | 560 for (chan = 0; chan < channels; chan++) { |
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561 prediction_type[chan] = get_bits(&alac->gb, 4); |
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562 prediction_quantitization[chan] = get_bits(&alac->gb, 4); |
2542
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563 |
5357
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564 ricemodifier[chan] = get_bits(&alac->gb, 3); |
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565 predictor_coef_num[chan] = get_bits(&alac->gb, 5); |
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|
566 |
a27a580f292e
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567 /* read the predictor table */ |
5365 | 568 for (i = 0; i < predictor_coef_num[chan]; i++) |
5357
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569 predictor_coef_table[chan][i] = (int16_t)get_bits(&alac->gb, 16); |
5361 | 570 } |
2542
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571 |
11077 | 572 if (alac->wasted_bits) { |
573 int i, ch; | |
574 for (i = 0; i < outputsamples; i++) { | |
575 for (ch = 0; ch < channels; ch++) | |
576 alac->wasted_bits_buffer[ch][i] = get_bits(&alac->gb, alac->wasted_bits); | |
577 } | |
578 } | |
5361 | 579 for (chan = 0; chan < channels; chan++) { |
2542
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580 bastardized_rice_decompress(alac, |
5357
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|
581 alac->predicterror_buffer[chan], |
2542
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|
582 outputsamples, |
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|
583 readsamplesize, |
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|
584 alac->setinfo_rice_initialhistory, |
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|
585 alac->setinfo_rice_kmodifier, |
5357
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586 ricemodifier[chan] * alac->setinfo_rice_historymult / 4, |
2542
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587 (1 << alac->setinfo_rice_kmodifier) - 1); |
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588 |
5357
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589 if (prediction_type[chan] == 0) { |
5361 | 590 /* adaptive fir */ |
5357
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591 predictor_decompress_fir_adapt(alac->predicterror_buffer[chan], |
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592 alac->outputsamples_buffer[chan], |
2542
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|
593 outputsamples, |
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|
594 readsamplesize, |
5357
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595 predictor_coef_table[chan], |
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596 predictor_coef_num[chan], |
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597 prediction_quantitization[chan]); |
2542
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598 } else { |
5357
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599 av_log(avctx, AV_LOG_ERROR, "FIXME: unhandled prediction type: %i\n", prediction_type[chan]); |
5366 | 600 /* I think the only other prediction type (or perhaps this is |
601 * just a boolean?) runs adaptive fir twice.. like: | |
5359
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602 * predictor_decompress_fir_adapt(predictor_error, tempout, ...) |
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603 * predictor_decompress_fir_adapt(predictor_error, outputsamples ...) |
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604 * little strange.. |
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605 */ |
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606 } |
5361 | 607 } |
608 } else { | |
609 /* not compressed, easy case */ | |
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610 int i, chan; |
11077 | 611 if (alac->setinfo_sample_size <= 16) { |
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612 for (i = 0; i < outputsamples; i++) |
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613 for (chan = 0; chan < channels; chan++) { |
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614 int32_t audiobits; |
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615 |
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616 audiobits = get_sbits_long(&alac->gb, alac->setinfo_sample_size); |
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617 |
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618 alac->outputsamples_buffer[chan][i] = audiobits; |
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619 } |
11077 | 620 } else { |
621 for (i = 0; i < outputsamples; i++) { | |
622 for (chan = 0; chan < channels; chan++) { | |
623 alac->outputsamples_buffer[chan][i] = get_bits(&alac->gb, | |
624 alac->setinfo_sample_size); | |
625 alac->outputsamples_buffer[chan][i] = sign_extend(alac->outputsamples_buffer[chan][i], | |
626 alac->setinfo_sample_size); | |
627 } | |
628 } | |
629 } | |
630 alac->wasted_bits = 0; | |
5361 | 631 interlacing_shift = 0; |
632 interlacing_leftweight = 0; | |
633 } | |
6641 | 634 if (get_bits(&alac->gb, 3) != 7) |
635 av_log(avctx, AV_LOG_ERROR, "Error : Wrong End Of Frame\n"); | |
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636 |
5361 | 637 switch(alac->setinfo_sample_size) { |
5400 | 638 case 16: |
5361 | 639 if (channels == 2) { |
5504 | 640 reconstruct_stereo_16(alac->outputsamples_buffer, |
5506 | 641 (int16_t*)outbuffer, |
642 alac->numchannels, | |
643 outputsamples, | |
644 interlacing_shift, | |
645 interlacing_leftweight); | |
5361 | 646 } else { |
647 int i; | |
648 for (i = 0; i < outputsamples; i++) { | |
11081 | 649 ((int16_t*)outbuffer)[i] = alac->outputsamples_buffer[0][i]; |
5361 | 650 } |
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651 } |
5361 | 652 break; |
653 case 24: | |
11077 | 654 if (channels == 2) { |
655 decorrelate_stereo_24(alac->outputsamples_buffer, | |
656 outbuffer, | |
657 alac->wasted_bits_buffer, | |
658 alac->wasted_bits, | |
659 alac->numchannels, | |
660 outputsamples, | |
661 interlacing_shift, | |
662 interlacing_leftweight); | |
663 } else { | |
664 int i; | |
665 for (i = 0; i < outputsamples; i++) | |
666 ((int32_t *)outbuffer)[i] = alac->outputsamples_buffer[0][i] << 8; | |
667 } | |
5361 | 668 break; |
669 } | |
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670 |
6641 | 671 if (input_buffer_size * 8 - get_bits_count(&alac->gb) > 8) |
672 av_log(avctx, AV_LOG_ERROR, "Error : %d bits left\n", input_buffer_size * 8 - get_bits_count(&alac->gb)); | |
673 | |
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674 return input_buffer_size; |
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675 } |
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676 |
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677 static av_cold int alac_decode_init(AVCodecContext * avctx) |
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678 { |
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679 ALACContext *alac = avctx->priv_data; |
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680 alac->avctx = avctx; |
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681 alac->context_initialized = 0; |
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682 |
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683 alac->numchannels = alac->avctx->channels; |
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684 |
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685 return 0; |
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686 } |
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687 |
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688 static av_cold int alac_decode_close(AVCodecContext *avctx) |
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689 { |
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690 ALACContext *alac = avctx->priv_data; |
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691 |
5272 | 692 int chan; |
693 for (chan = 0; chan < MAX_CHANNELS; chan++) { | |
11079 | 694 av_freep(&alac->predicterror_buffer[chan]); |
695 av_freep(&alac->outputsamples_buffer[chan]); | |
11077 | 696 av_freep(&alac->wasted_bits_buffer[chan]); |
5272 | 697 } |
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698 |
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699 return 0; |
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700 } |
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701 |
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702 AVCodec alac_decoder = { |
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703 "alac", |
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704 AVMEDIA_TYPE_AUDIO, |
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705 CODEC_ID_ALAC, |
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706 sizeof(ALACContext), |
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707 alac_decode_init, |
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708 NULL, |
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709 alac_decode_close, |
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710 alac_decode_frame, |
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711 .long_name = NULL_IF_CONFIG_SMALL("ALAC (Apple Lossless Audio Codec)"), |
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712 }; |