Mercurial > libavcodec.hg
annotate aac.c @ 9158:0d5eed0f1b45 libavcodec
Fix unaligned access in ff_copy_bits()
author | mru |
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date | Mon, 09 Mar 2009 23:27:38 +0000 |
parents | 1c478e02676a |
children | 4172c9ef331a |
rev | line source |
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7501 | 1 /* |
2 * AAC decoder | |
3 * Copyright (c) 2005-2006 Oded Shimon ( ods15 ods15 dyndns org ) | |
4 * Copyright (c) 2006-2007 Maxim Gavrilov ( maxim.gavrilov gmail com ) | |
5 * | |
6 * This file is part of FFmpeg. | |
7 * | |
8 * FFmpeg is free software; you can redistribute it and/or | |
9 * modify it under the terms of the GNU Lesser General Public | |
10 * License as published by the Free Software Foundation; either | |
11 * version 2.1 of the License, or (at your option) any later version. | |
12 * | |
13 * FFmpeg is distributed in the hope that it will be useful, | |
14 * but WITHOUT ANY WARRANTY; without even the implied warranty of | |
15 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU | |
16 * Lesser General Public License for more details. | |
17 * | |
18 * You should have received a copy of the GNU Lesser General Public | |
19 * License along with FFmpeg; if not, write to the Free Software | |
20 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA | |
21 */ | |
22 | |
23 /** | |
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24 * @file libavcodec/aac.c |
7501 | 25 * AAC decoder |
26 * @author Oded Shimon ( ods15 ods15 dyndns org ) | |
27 * @author Maxim Gavrilov ( maxim.gavrilov gmail com ) | |
28 */ | |
29 | |
30 /* | |
31 * supported tools | |
32 * | |
33 * Support? Name | |
34 * N (code in SoC repo) gain control | |
35 * Y block switching | |
36 * Y window shapes - standard | |
37 * N window shapes - Low Delay | |
38 * Y filterbank - standard | |
39 * N (code in SoC repo) filterbank - Scalable Sample Rate | |
40 * Y Temporal Noise Shaping | |
41 * N (code in SoC repo) Long Term Prediction | |
42 * Y intensity stereo | |
43 * Y channel coupling | |
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44 * Y frequency domain prediction |
7501 | 45 * Y Perceptual Noise Substitution |
46 * Y Mid/Side stereo | |
47 * N Scalable Inverse AAC Quantization | |
48 * N Frequency Selective Switch | |
49 * N upsampling filter | |
50 * Y quantization & coding - AAC | |
51 * N quantization & coding - TwinVQ | |
52 * N quantization & coding - BSAC | |
53 * N AAC Error Resilience tools | |
54 * N Error Resilience payload syntax | |
55 * N Error Protection tool | |
56 * N CELP | |
57 * N Silence Compression | |
58 * N HVXC | |
59 * N HVXC 4kbits/s VR | |
60 * N Structured Audio tools | |
61 * N Structured Audio Sample Bank Format | |
62 * N MIDI | |
63 * N Harmonic and Individual Lines plus Noise | |
64 * N Text-To-Speech Interface | |
65 * N (in progress) Spectral Band Replication | |
66 * Y (not in this code) Layer-1 | |
67 * Y (not in this code) Layer-2 | |
68 * Y (not in this code) Layer-3 | |
69 * N SinuSoidal Coding (Transient, Sinusoid, Noise) | |
70 * N (planned) Parametric Stereo | |
71 * N Direct Stream Transfer | |
72 * | |
73 * Note: - HE AAC v1 comprises LC AAC with Spectral Band Replication. | |
74 * - HE AAC v2 comprises LC AAC with Spectral Band Replication and | |
75 Parametric Stereo. | |
76 */ | |
77 | |
78 | |
79 #include "avcodec.h" | |
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80 #include "internal.h" |
7501 | 81 #include "bitstream.h" |
82 #include "dsputil.h" | |
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83 #include "lpc.h" |
7501 | 84 |
85 #include "aac.h" | |
86 #include "aactab.h" | |
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87 #include "aacdectab.h" |
7501 | 88 #include "mpeg4audio.h" |
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89 #include "aac_parser.h" |
7501 | 90 |
91 #include <assert.h> | |
92 #include <errno.h> | |
93 #include <math.h> | |
94 #include <string.h> | |
95 | |
96 static VLC vlc_scalefactors; | |
97 static VLC vlc_spectral[11]; | |
98 | |
99 | |
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100 static ChannelElement* get_che(AACContext *ac, int type, int elem_id) { |
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101 static const int8_t tags_per_config[16] = { 0, 1, 1, 2, 3, 3, 4, 5, 0, 0, 0, 0, 0, 0, 0, 0 }; |
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102 if (ac->tag_che_map[type][elem_id]) { |
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103 return ac->tag_che_map[type][elem_id]; |
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104 } |
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105 if (ac->tags_mapped >= tags_per_config[ac->m4ac.chan_config]) { |
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106 return NULL; |
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107 } |
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108 switch (ac->m4ac.chan_config) { |
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109 case 7: |
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110 if (ac->tags_mapped == 3 && type == TYPE_CPE) { |
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111 ac->tags_mapped++; |
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112 return ac->tag_che_map[TYPE_CPE][elem_id] = ac->che[TYPE_CPE][2]; |
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113 } |
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114 case 6: |
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115 /* Some streams incorrectly code 5.1 audio as SCE[0] CPE[0] CPE[1] SCE[1] |
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116 instead of SCE[0] CPE[0] CPE[0] LFE[0]. If we seem to have |
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117 encountered such a stream, transfer the LFE[0] element to SCE[1] */ |
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118 if (ac->tags_mapped == tags_per_config[ac->m4ac.chan_config] - 1 && (type == TYPE_LFE || type == TYPE_SCE)) { |
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119 ac->tags_mapped++; |
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120 return ac->tag_che_map[type][elem_id] = ac->che[TYPE_LFE][0]; |
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121 } |
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122 case 5: |
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123 if (ac->tags_mapped == 2 && type == TYPE_CPE) { |
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124 ac->tags_mapped++; |
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125 return ac->tag_che_map[TYPE_CPE][elem_id] = ac->che[TYPE_CPE][1]; |
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126 } |
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127 case 4: |
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128 if (ac->tags_mapped == 2 && ac->m4ac.chan_config == 4 && type == TYPE_SCE) { |
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129 ac->tags_mapped++; |
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130 return ac->tag_che_map[TYPE_SCE][elem_id] = ac->che[TYPE_SCE][1]; |
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131 } |
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132 case 3: |
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133 case 2: |
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134 if (ac->tags_mapped == (ac->m4ac.chan_config != 2) && type == TYPE_CPE) { |
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135 ac->tags_mapped++; |
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136 return ac->tag_che_map[TYPE_CPE][elem_id] = ac->che[TYPE_CPE][0]; |
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137 } else if (ac->m4ac.chan_config == 2) { |
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138 return NULL; |
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139 } |
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140 case 1: |
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141 if (!ac->tags_mapped && type == TYPE_SCE) { |
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142 ac->tags_mapped++; |
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143 return ac->tag_che_map[TYPE_SCE][elem_id] = ac->che[TYPE_SCE][0]; |
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144 } |
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145 default: |
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146 return NULL; |
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147 } |
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148 } |
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149 |
7539 | 150 /** |
7578 | 151 * Configure output channel order based on the current program configuration element. |
152 * | |
153 * @param che_pos current channel position configuration | |
154 * @param new_che_pos New channel position configuration - we only do something if it differs from the current one. | |
155 * | |
156 * @return Returns error status. 0 - OK, !0 - error | |
157 */ | |
158 static int output_configure(AACContext *ac, enum ChannelPosition che_pos[4][MAX_ELEM_ID], | |
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159 enum ChannelPosition new_che_pos[4][MAX_ELEM_ID], int channel_config) { |
7578 | 160 AVCodecContext *avctx = ac->avccontext; |
161 int i, type, channels = 0; | |
162 | |
163 if(!memcmp(che_pos, new_che_pos, 4 * MAX_ELEM_ID * sizeof(new_che_pos[0][0]))) | |
164 return 0; /* no change */ | |
165 | |
166 memcpy(che_pos, new_che_pos, 4 * MAX_ELEM_ID * sizeof(new_che_pos[0][0])); | |
167 | |
168 /* Allocate or free elements depending on if they are in the | |
169 * current program configuration. | |
170 * | |
171 * Set up default 1:1 output mapping. | |
172 * | |
173 * For a 5.1 stream the output order will be: | |
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174 * [ Center ] [ Front Left ] [ Front Right ] [ LFE ] [ Surround Left ] [ Surround Right ] |
7578 | 175 */ |
176 | |
177 for(i = 0; i < MAX_ELEM_ID; i++) { | |
178 for(type = 0; type < 4; type++) { | |
179 if(che_pos[type][i]) { | |
180 if(!ac->che[type][i] && !(ac->che[type][i] = av_mallocz(sizeof(ChannelElement)))) | |
181 return AVERROR(ENOMEM); | |
182 if(type != TYPE_CCE) { | |
183 ac->output_data[channels++] = ac->che[type][i]->ch[0].ret; | |
184 if(type == TYPE_CPE) { | |
185 ac->output_data[channels++] = ac->che[type][i]->ch[1].ret; | |
186 } | |
187 } | |
188 } else | |
189 av_freep(&ac->che[type][i]); | |
190 } | |
191 } | |
192 | |
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193 if (channel_config) { |
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194 memset(ac->tag_che_map, 0, 4 * MAX_ELEM_ID * sizeof(ac->che[0][0])); |
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195 ac->tags_mapped = 0; |
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196 } else { |
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197 memcpy(ac->tag_che_map, ac->che, 4 * MAX_ELEM_ID * sizeof(ac->che[0][0])); |
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198 ac->tags_mapped = 4*MAX_ELEM_ID; |
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199 } |
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200 |
7578 | 201 avctx->channels = channels; |
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202 |
7578 | 203 return 0; |
204 } | |
205 | |
206 /** | |
7539 | 207 * Decode an array of 4 bit element IDs, optionally interleaved with a stereo/mono switching bit. |
208 * | |
209 * @param cpe_map Stereo (Channel Pair Element) map, NULL if stereo bit is not present. | |
210 * @param sce_map mono (Single Channel Element) map | |
211 * @param type speaker type/position for these channels | |
212 */ | |
213 static void decode_channel_map(enum ChannelPosition *cpe_map, | |
214 enum ChannelPosition *sce_map, enum ChannelPosition type, GetBitContext * gb, int n) { | |
215 while(n--) { | |
216 enum ChannelPosition *map = cpe_map && get_bits1(gb) ? cpe_map : sce_map; // stereo or mono map | |
217 map[get_bits(gb, 4)] = type; | |
218 } | |
219 } | |
220 | |
221 /** | |
222 * Decode program configuration element; reference: table 4.2. | |
223 * | |
224 * @param new_che_pos New channel position configuration - we only do something if it differs from the current one. | |
225 * | |
226 * @return Returns error status. 0 - OK, !0 - error | |
227 */ | |
228 static int decode_pce(AACContext * ac, enum ChannelPosition new_che_pos[4][MAX_ELEM_ID], | |
229 GetBitContext * gb) { | |
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230 int num_front, num_side, num_back, num_lfe, num_assoc_data, num_cc, sampling_index; |
7539 | 231 |
232 skip_bits(gb, 2); // object_type | |
233 | |
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234 sampling_index = get_bits(gb, 4); |
8812 | 235 if(sampling_index > 12) { |
7539 | 236 av_log(ac->avccontext, AV_LOG_ERROR, "invalid sampling rate index %d\n", ac->m4ac.sampling_index); |
237 return -1; | |
238 } | |
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239 ac->m4ac.sampling_index = sampling_index; |
7539 | 240 ac->m4ac.sample_rate = ff_mpeg4audio_sample_rates[ac->m4ac.sampling_index]; |
7501 | 241 num_front = get_bits(gb, 4); |
242 num_side = get_bits(gb, 4); | |
243 num_back = get_bits(gb, 4); | |
244 num_lfe = get_bits(gb, 2); | |
245 num_assoc_data = get_bits(gb, 3); | |
246 num_cc = get_bits(gb, 4); | |
247 | |
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248 if (get_bits1(gb)) |
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249 skip_bits(gb, 4); // mono_mixdown_tag |
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250 if (get_bits1(gb)) |
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251 skip_bits(gb, 4); // stereo_mixdown_tag |
7501 | 252 |
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253 if (get_bits1(gb)) |
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254 skip_bits(gb, 3); // mixdown_coeff_index and pseudo_surround |
7501 | 255 |
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256 decode_channel_map(new_che_pos[TYPE_CPE], new_che_pos[TYPE_SCE], AAC_CHANNEL_FRONT, gb, num_front); |
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257 decode_channel_map(new_che_pos[TYPE_CPE], new_che_pos[TYPE_SCE], AAC_CHANNEL_SIDE, gb, num_side ); |
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258 decode_channel_map(new_che_pos[TYPE_CPE], new_che_pos[TYPE_SCE], AAC_CHANNEL_BACK, gb, num_back ); |
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259 decode_channel_map(NULL, new_che_pos[TYPE_LFE], AAC_CHANNEL_LFE, gb, num_lfe ); |
7501 | 260 |
261 skip_bits_long(gb, 4 * num_assoc_data); | |
262 | |
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263 decode_channel_map(new_che_pos[TYPE_CCE], new_che_pos[TYPE_CCE], AAC_CHANNEL_CC, gb, num_cc ); |
7501 | 264 |
265 align_get_bits(gb); | |
266 | |
267 /* comment field, first byte is length */ | |
268 skip_bits_long(gb, 8 * get_bits(gb, 8)); | |
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269 return 0; |
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270 } |
7501 | 271 |
7539 | 272 /** |
273 * Set up channel positions based on a default channel configuration | |
274 * as specified in table 1.17. | |
275 * | |
276 * @param new_che_pos New channel position configuration - we only do something if it differs from the current one. | |
277 * | |
278 * @return Returns error status. 0 - OK, !0 - error | |
279 */ | |
280 static int set_default_channel_config(AACContext *ac, enum ChannelPosition new_che_pos[4][MAX_ELEM_ID], | |
281 int channel_config) | |
282 { | |
283 if(channel_config < 1 || channel_config > 7) { | |
284 av_log(ac->avccontext, AV_LOG_ERROR, "invalid default channel configuration (%d)\n", | |
285 channel_config); | |
286 return -1; | |
287 } | |
288 | |
289 /* default channel configurations: | |
290 * | |
291 * 1ch : front center (mono) | |
292 * 2ch : L + R (stereo) | |
293 * 3ch : front center + L + R | |
294 * 4ch : front center + L + R + back center | |
295 * 5ch : front center + L + R + back stereo | |
296 * 6ch : front center + L + R + back stereo + LFE | |
297 * 7ch : front center + L + R + outer front left + outer front right + back stereo + LFE | |
298 */ | |
299 | |
300 if(channel_config != 2) | |
301 new_che_pos[TYPE_SCE][0] = AAC_CHANNEL_FRONT; // front center (or mono) | |
302 if(channel_config > 1) | |
303 new_che_pos[TYPE_CPE][0] = AAC_CHANNEL_FRONT; // L + R (or stereo) | |
304 if(channel_config == 4) | |
305 new_che_pos[TYPE_SCE][1] = AAC_CHANNEL_BACK; // back center | |
306 if(channel_config > 4) | |
307 new_che_pos[TYPE_CPE][(channel_config == 7) + 1] | |
308 = AAC_CHANNEL_BACK; // back stereo | |
309 if(channel_config > 5) | |
310 new_che_pos[TYPE_LFE][0] = AAC_CHANNEL_LFE; // LFE | |
311 if(channel_config == 7) | |
312 new_che_pos[TYPE_CPE][1] = AAC_CHANNEL_FRONT; // outer front left + outer front right | |
313 | |
314 return 0; | |
315 } | |
316 | |
7578 | 317 /** |
318 * Decode GA "General Audio" specific configuration; reference: table 4.1. | |
319 * | |
320 * @return Returns error status. 0 - OK, !0 - error | |
321 */ | |
322 static int decode_ga_specific_config(AACContext * ac, GetBitContext * gb, int channel_config) { | |
323 enum ChannelPosition new_che_pos[4][MAX_ELEM_ID]; | |
324 int extension_flag, ret; | |
325 | |
326 if(get_bits1(gb)) { // frameLengthFlag | |
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327 ff_log_missing_feature(ac->avccontext, "960/120 MDCT window is", 1); |
7539 | 328 return -1; |
329 } | |
330 | |
331 if (get_bits1(gb)) // dependsOnCoreCoder | |
332 skip_bits(gb, 14); // coreCoderDelay | |
333 extension_flag = get_bits1(gb); | |
334 | |
335 if(ac->m4ac.object_type == AOT_AAC_SCALABLE || | |
336 ac->m4ac.object_type == AOT_ER_AAC_SCALABLE) | |
337 skip_bits(gb, 3); // layerNr | |
338 | |
339 memset(new_che_pos, 0, 4 * MAX_ELEM_ID * sizeof(new_che_pos[0][0])); | |
340 if (channel_config == 0) { | |
341 skip_bits(gb, 4); // element_instance_tag | |
342 if((ret = decode_pce(ac, new_che_pos, gb))) | |
343 return ret; | |
344 } else { | |
345 if((ret = set_default_channel_config(ac, new_che_pos, channel_config))) | |
346 return ret; | |
347 } | |
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348 if((ret = output_configure(ac, ac->che_pos, new_che_pos, channel_config))) |
7539 | 349 return ret; |
350 | |
351 if (extension_flag) { | |
352 switch (ac->m4ac.object_type) { | |
353 case AOT_ER_BSAC: | |
354 skip_bits(gb, 5); // numOfSubFrame | |
355 skip_bits(gb, 11); // layer_length | |
356 break; | |
357 case AOT_ER_AAC_LC: | |
358 case AOT_ER_AAC_LTP: | |
359 case AOT_ER_AAC_SCALABLE: | |
360 case AOT_ER_AAC_LD: | |
361 skip_bits(gb, 3); /* aacSectionDataResilienceFlag | |
362 * aacScalefactorDataResilienceFlag | |
363 * aacSpectralDataResilienceFlag | |
364 */ | |
365 break; | |
366 } | |
367 skip_bits1(gb); // extensionFlag3 (TBD in version 3) | |
368 } | |
369 return 0; | |
370 } | |
371 | |
372 /** | |
373 * Decode audio specific configuration; reference: table 1.13. | |
374 * | |
375 * @param data pointer to AVCodecContext extradata | |
376 * @param data_size size of AVCCodecContext extradata | |
377 * | |
378 * @return Returns error status. 0 - OK, !0 - error | |
379 */ | |
380 static int decode_audio_specific_config(AACContext * ac, void *data, int data_size) { | |
381 GetBitContext gb; | |
382 int i; | |
383 | |
384 init_get_bits(&gb, data, data_size * 8); | |
385 | |
386 if((i = ff_mpeg4audio_get_config(&ac->m4ac, data, data_size)) < 0) | |
387 return -1; | |
8812 | 388 if(ac->m4ac.sampling_index > 12) { |
7539 | 389 av_log(ac->avccontext, AV_LOG_ERROR, "invalid sampling rate index %d\n", ac->m4ac.sampling_index); |
390 return -1; | |
391 } | |
392 | |
393 skip_bits_long(&gb, i); | |
394 | |
395 switch (ac->m4ac.object_type) { | |
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396 case AOT_AAC_MAIN: |
7539 | 397 case AOT_AAC_LC: |
398 if (decode_ga_specific_config(ac, &gb, ac->m4ac.chan_config)) | |
399 return -1; | |
400 break; | |
401 default: | |
402 av_log(ac->avccontext, AV_LOG_ERROR, "Audio object type %s%d is not supported.\n", | |
403 ac->m4ac.sbr == 1? "SBR+" : "", ac->m4ac.object_type); | |
404 return -1; | |
405 } | |
406 return 0; | |
407 } | |
408 | |
7578 | 409 /** |
410 * linear congruential pseudorandom number generator | |
411 * | |
412 * @param previous_val pointer to the current state of the generator | |
413 * | |
414 * @return Returns a 32-bit pseudorandom integer | |
415 */ | |
416 static av_always_inline int lcg_random(int previous_val) { | |
417 return previous_val * 1664525 + 1013904223; | |
418 } | |
419 | |
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420 static void reset_predict_state(PredictorState * ps) { |
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421 ps->r0 = 0.0f; |
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422 ps->r1 = 0.0f; |
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423 ps->cor0 = 0.0f; |
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424 ps->cor1 = 0.0f; |
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425 ps->var0 = 1.0f; |
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426 ps->var1 = 1.0f; |
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427 } |
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428 |
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429 static void reset_all_predictors(PredictorState * ps) { |
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430 int i; |
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431 for (i = 0; i < MAX_PREDICTORS; i++) |
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432 reset_predict_state(&ps[i]); |
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433 } |
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434 |
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435 static void reset_predictor_group(PredictorState * ps, int group_num) { |
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436 int i; |
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437 for (i = group_num-1; i < MAX_PREDICTORS; i+=30) |
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438 reset_predict_state(&ps[i]); |
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439 } |
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440 |
7501 | 441 static av_cold int aac_decode_init(AVCodecContext * avccontext) { |
442 AACContext * ac = avccontext->priv_data; | |
443 int i; | |
444 | |
445 ac->avccontext = avccontext; | |
446 | |
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447 if (avccontext->extradata_size > 0) { |
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448 if(decode_audio_specific_config(ac, avccontext->extradata, avccontext->extradata_size)) |
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449 return -1; |
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450 avccontext->sample_rate = ac->m4ac.sample_rate; |
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451 } else if (avccontext->channels > 0) { |
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452 enum ChannelPosition new_che_pos[4][MAX_ELEM_ID]; |
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453 memset(new_che_pos, 0, 4 * MAX_ELEM_ID * sizeof(new_che_pos[0][0])); |
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454 if(set_default_channel_config(ac, new_che_pos, avccontext->channels - (avccontext->channels == 8))) |
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455 return -1; |
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456 if(output_configure(ac, ac->che_pos, new_che_pos, 1)) |
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457 return -1; |
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458 ac->m4ac.sample_rate = avccontext->sample_rate; |
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459 } else { |
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460 ff_log_missing_feature(ac->avccontext, "Implicit channel configuration is", 0); |
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461 return -1; |
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462 } |
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463 |
7539 | 464 avccontext->sample_fmt = SAMPLE_FMT_S16; |
7501 | 465 avccontext->frame_size = 1024; |
466 | |
467 AAC_INIT_VLC_STATIC( 0, 144); | |
468 AAC_INIT_VLC_STATIC( 1, 114); | |
469 AAC_INIT_VLC_STATIC( 2, 188); | |
470 AAC_INIT_VLC_STATIC( 3, 180); | |
471 AAC_INIT_VLC_STATIC( 4, 172); | |
472 AAC_INIT_VLC_STATIC( 5, 140); | |
473 AAC_INIT_VLC_STATIC( 6, 168); | |
474 AAC_INIT_VLC_STATIC( 7, 114); | |
475 AAC_INIT_VLC_STATIC( 8, 262); | |
476 AAC_INIT_VLC_STATIC( 9, 248); | |
477 AAC_INIT_VLC_STATIC(10, 384); | |
478 | |
479 dsputil_init(&ac->dsp, avccontext); | |
480 | |
7539 | 481 ac->random_state = 0x1f2e3d4c; |
482 | |
7501 | 483 // -1024 - Compensate wrong IMDCT method. |
484 // 32768 - Required to scale values to the correct range for the bias method | |
485 // for float to int16 conversion. | |
486 | |
487 if(ac->dsp.float_to_int16 == ff_float_to_int16_c) { | |
488 ac->add_bias = 385.0f; | |
489 ac->sf_scale = 1. / (-1024. * 32768.); | |
490 ac->sf_offset = 0; | |
491 } else { | |
492 ac->add_bias = 0.0f; | |
493 ac->sf_scale = 1. / -1024.; | |
494 ac->sf_offset = 60; | |
495 } | |
496 | |
8590 | 497 #if !CONFIG_HARDCODED_TABLES |
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498 for (i = 0; i < 428; i++) |
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499 ff_aac_pow2sf_tab[i] = pow(2, (i - 200)/4.); |
7501 | 500 #endif /* CONFIG_HARDCODED_TABLES */ |
501 | |
8042 | 502 INIT_VLC_STATIC(&vlc_scalefactors,7,FF_ARRAY_ELEMS(ff_aac_scalefactor_code), |
7501 | 503 ff_aac_scalefactor_bits, sizeof(ff_aac_scalefactor_bits[0]), sizeof(ff_aac_scalefactor_bits[0]), |
504 ff_aac_scalefactor_code, sizeof(ff_aac_scalefactor_code[0]), sizeof(ff_aac_scalefactor_code[0]), | |
505 352); | |
506 | |
507 ff_mdct_init(&ac->mdct, 11, 1); | |
508 ff_mdct_init(&ac->mdct_small, 8, 1); | |
7581 | 509 // window initialization |
510 ff_kbd_window_init(ff_aac_kbd_long_1024, 4.0, 1024); | |
511 ff_kbd_window_init(ff_aac_kbd_short_128, 6.0, 128); | |
512 ff_sine_window_init(ff_sine_1024, 1024); | |
513 ff_sine_window_init(ff_sine_128, 128); | |
514 | |
7501 | 515 return 0; |
516 } | |
517 | |
7539 | 518 /** |
519 * Skip data_stream_element; reference: table 4.10. | |
520 */ | |
521 static void skip_data_stream_element(GetBitContext * gb) { | |
7501 | 522 int byte_align = get_bits1(gb); |
523 int count = get_bits(gb, 8); | |
524 if (count == 255) | |
525 count += get_bits(gb, 8); | |
526 if (byte_align) | |
527 align_get_bits(gb); | |
528 skip_bits_long(gb, 8 * count); | |
529 } | |
530 | |
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531 static int decode_prediction(AACContext * ac, IndividualChannelStream * ics, GetBitContext * gb) { |
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532 int sfb; |
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533 if (get_bits1(gb)) { |
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534 ics->predictor_reset_group = get_bits(gb, 5); |
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535 if (ics->predictor_reset_group == 0 || ics->predictor_reset_group > 30) { |
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536 av_log(ac->avccontext, AV_LOG_ERROR, "Invalid Predictor Reset Group.\n"); |
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537 return -1; |
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538 } |
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539 } |
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540 for (sfb = 0; sfb < FFMIN(ics->max_sfb, ff_aac_pred_sfb_max[ac->m4ac.sampling_index]); sfb++) { |
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541 ics->prediction_used[sfb] = get_bits1(gb); |
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542 } |
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543 return 0; |
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544 } |
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545 |
7501 | 546 /** |
7539 | 547 * Decode Individual Channel Stream info; reference: table 4.6. |
548 * | |
549 * @param common_window Channels have independent [0], or shared [1], Individual Channel Stream information. | |
550 */ | |
551 static int decode_ics_info(AACContext * ac, IndividualChannelStream * ics, GetBitContext * gb, int common_window) { | |
552 if (get_bits1(gb)) { | |
553 av_log(ac->avccontext, AV_LOG_ERROR, "Reserved bit set.\n"); | |
554 memset(ics, 0, sizeof(IndividualChannelStream)); | |
555 return -1; | |
556 } | |
557 ics->window_sequence[1] = ics->window_sequence[0]; | |
558 ics->window_sequence[0] = get_bits(gb, 2); | |
559 ics->use_kb_window[1] = ics->use_kb_window[0]; | |
560 ics->use_kb_window[0] = get_bits1(gb); | |
561 ics->num_window_groups = 1; | |
562 ics->group_len[0] = 1; | |
7581 | 563 if (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) { |
564 int i; | |
565 ics->max_sfb = get_bits(gb, 4); | |
566 for (i = 0; i < 7; i++) { | |
567 if (get_bits1(gb)) { | |
568 ics->group_len[ics->num_window_groups-1]++; | |
569 } else { | |
570 ics->num_window_groups++; | |
571 ics->group_len[ics->num_window_groups-1] = 1; | |
572 } | |
573 } | |
574 ics->num_windows = 8; | |
575 ics->swb_offset = swb_offset_128[ac->m4ac.sampling_index]; | |
576 ics->num_swb = ff_aac_num_swb_128[ac->m4ac.sampling_index]; | |
577 ics->tns_max_bands = tns_max_bands_128[ac->m4ac.sampling_index]; | |
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578 ics->predictor_present = 0; |
7581 | 579 } else { |
580 ics->max_sfb = get_bits(gb, 6); | |
581 ics->num_windows = 1; | |
582 ics->swb_offset = swb_offset_1024[ac->m4ac.sampling_index]; | |
583 ics->num_swb = ff_aac_num_swb_1024[ac->m4ac.sampling_index]; | |
584 ics->tns_max_bands = tns_max_bands_1024[ac->m4ac.sampling_index]; | |
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585 ics->predictor_present = get_bits1(gb); |
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586 ics->predictor_reset_group = 0; |
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587 if (ics->predictor_present) { |
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588 if (ac->m4ac.object_type == AOT_AAC_MAIN) { |
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AAC: Frequency domain prediction and hence Main profile support
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diff
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589 if (decode_prediction(ac, ics, gb)) { |
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590 memset(ics, 0, sizeof(IndividualChannelStream)); |
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591 return -1; |
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|
592 } |
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593 } else if (ac->m4ac.object_type == AOT_AAC_LC) { |
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594 av_log(ac->avccontext, AV_LOG_ERROR, "Prediction is not allowed in AAC-LC.\n"); |
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595 memset(ics, 0, sizeof(IndividualChannelStream)); |
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596 return -1; |
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597 } else { |
8281
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598 ff_log_missing_feature(ac->avccontext, "Predictor bit set but LTP is", 1); |
8200 | 599 memset(ics, 0, sizeof(IndividualChannelStream)); |
600 return -1; | |
8199
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601 } |
7578 | 602 } |
603 } | |
604 | |
605 if(ics->max_sfb > ics->num_swb) { | |
606 av_log(ac->avccontext, AV_LOG_ERROR, | |
607 "Number of scalefactor bands in group (%d) exceeds limit (%d).\n", | |
608 ics->max_sfb, ics->num_swb); | |
609 memset(ics, 0, sizeof(IndividualChannelStream)); | |
610 return -1; | |
611 } | |
612 | |
7539 | 613 return 0; |
614 } | |
615 | |
616 /** | |
617 * Decode band types (section_data payload); reference: table 4.46. | |
618 * | |
619 * @param band_type array of the used band type | |
620 * @param band_type_run_end array of the last scalefactor band of a band type run | |
621 * | |
622 * @return Returns error status. 0 - OK, !0 - error | |
623 */ | |
624 static int decode_band_types(AACContext * ac, enum BandType band_type[120], | |
7523
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625 int band_type_run_end[120], GetBitContext * gb, IndividualChannelStream * ics) { |
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626 int g, idx = 0; |
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627 const int bits = (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) ? 3 : 5; |
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628 for (g = 0; g < ics->num_window_groups; g++) { |
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629 int k = 0; |
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630 while (k < ics->max_sfb) { |
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631 uint8_t sect_len = k; |
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632 int sect_len_incr; |
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633 int sect_band_type = get_bits(gb, 4); |
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634 if (sect_band_type == 12) { |
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635 av_log(ac->avccontext, AV_LOG_ERROR, "invalid band type\n"); |
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636 return -1; |
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637 } |
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638 while ((sect_len_incr = get_bits(gb, bits)) == (1 << bits)-1) |
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639 sect_len += sect_len_incr; |
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640 sect_len += sect_len_incr; |
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641 if (sect_len > ics->max_sfb) { |
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642 av_log(ac->avccontext, AV_LOG_ERROR, |
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643 "Number of bands (%d) exceeds limit (%d).\n", |
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644 sect_len, ics->max_sfb); |
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645 return -1; |
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646 } |
7581 | 647 for (; k < sect_len; k++) { |
648 band_type [idx] = sect_band_type; | |
649 band_type_run_end[idx++] = sect_len; | |
650 } | |
7539 | 651 } |
652 } | |
653 return 0; | |
654 } | |
7523
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655 |
7539 | 656 /** |
657 * Decode scalefactors; reference: table 4.47. | |
7523
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658 * |
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659 * @param global_gain first scalefactor value as scalefactors are differentially coded |
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660 * @param band_type array of the used band type |
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661 * @param band_type_run_end array of the last scalefactor band of a band type run |
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662 * @param sf array of scalefactors or intensity stereo positions |
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663 * |
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664 * @return Returns error status. 0 - OK, !0 - error |
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665 */ |
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666 static int decode_scalefactors(AACContext * ac, float sf[120], GetBitContext * gb, |
7540 | 667 unsigned int global_gain, IndividualChannelStream * ics, |
7523
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668 enum BandType band_type[120], int band_type_run_end[120]) { |
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669 const int sf_offset = ac->sf_offset + (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE ? 12 : 0); |
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670 int g, i, idx = 0; |
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671 int offset[3] = { global_gain, global_gain - 90, 100 }; |
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672 int noise_flag = 1; |
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673 static const char *sf_str[3] = { "Global gain", "Noise gain", "Intensity stereo position" }; |
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674 for (g = 0; g < ics->num_window_groups; g++) { |
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675 for (i = 0; i < ics->max_sfb;) { |
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676 int run_end = band_type_run_end[idx]; |
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677 if (band_type[idx] == ZERO_BT) { |
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678 for(; i < run_end; i++, idx++) |
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679 sf[idx] = 0.; |
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680 }else if((band_type[idx] == INTENSITY_BT) || (band_type[idx] == INTENSITY_BT2)) { |
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681 for(; i < run_end; i++, idx++) { |
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682 offset[2] += get_vlc2(gb, vlc_scalefactors.table, 7, 3) - 60; |
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683 if(offset[2] > 255U) { |
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684 av_log(ac->avccontext, AV_LOG_ERROR, |
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685 "%s (%d) out of range.\n", sf_str[2], offset[2]); |
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686 return -1; |
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687 } |
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688 sf[idx] = ff_aac_pow2sf_tab[-offset[2] + 300]; |
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689 } |
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690 }else if(band_type[idx] == NOISE_BT) { |
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691 for(; i < run_end; i++, idx++) { |
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692 if(noise_flag-- > 0) |
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693 offset[1] += get_bits(gb, 9) - 256; |
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694 else |
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695 offset[1] += get_vlc2(gb, vlc_scalefactors.table, 7, 3) - 60; |
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696 if(offset[1] > 255U) { |
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697 av_log(ac->avccontext, AV_LOG_ERROR, |
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698 "%s (%d) out of range.\n", sf_str[1], offset[1]); |
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699 return -1; |
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700 } |
8109
7baa25290231
Fix perceptual noise shaping scaling including an oversight of the offset for
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changeset
|
701 sf[idx] = -ff_aac_pow2sf_tab[ offset[1] + sf_offset + 100]; |
7523
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702 } |
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703 }else { |
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704 for(; i < run_end; i++, idx++) { |
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705 offset[0] += get_vlc2(gb, vlc_scalefactors.table, 7, 3) - 60; |
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706 if(offset[0] > 255U) { |
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707 av_log(ac->avccontext, AV_LOG_ERROR, |
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708 "%s (%d) out of range.\n", sf_str[0], offset[0]); |
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709 return -1; |
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710 } |
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711 sf[idx] = -ff_aac_pow2sf_tab[ offset[0] + sf_offset]; |
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712 } |
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713 } |
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714 } |
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715 } |
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716 return 0; |
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717 } |
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718 |
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719 /** |
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720 * Decode pulse data; reference: table 4.7. |
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721 */ |
7871
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Validate pulse position and error out if an invalid position is encountered.
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722 static int decode_pulses(Pulse * pulse, GetBitContext * gb, const uint16_t * swb_offset, int num_swb) { |
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723 int i, pulse_swb; |
7523
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724 pulse->num_pulse = get_bits(gb, 2) + 1; |
7871
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725 pulse_swb = get_bits(gb, 6); |
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726 if (pulse_swb >= num_swb) |
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727 return -1; |
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728 pulse->pos[0] = swb_offset[pulse_swb]; |
7856
0882c93a0c11
Operands of '+' are not necessarily evaluated in order so r15311 was pointless.
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729 pulse->pos[0] += get_bits(gb, 5); |
7871
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730 if (pulse->pos[0] > 1023) |
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731 return -1; |
7580 | 732 pulse->amp[0] = get_bits(gb, 4); |
733 for (i = 1; i < pulse->num_pulse; i++) { | |
734 pulse->pos[i] = get_bits(gb, 5) + pulse->pos[i-1]; | |
7871
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735 if (pulse->pos[i] > 1023) |
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736 return -1; |
7580 | 737 pulse->amp[i] = get_bits(gb, 4); |
7523
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738 } |
7871
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739 return 0; |
7523
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740 } |
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741 |
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742 /** |
7609 | 743 * Decode Temporal Noise Shaping data; reference: table 4.48. |
744 * | |
745 * @return Returns error status. 0 - OK, !0 - error | |
746 */ | |
747 static int decode_tns(AACContext * ac, TemporalNoiseShaping * tns, | |
748 GetBitContext * gb, const IndividualChannelStream * ics) { | |
749 int w, filt, i, coef_len, coef_res, coef_compress; | |
750 const int is8 = ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE; | |
751 const int tns_max_order = is8 ? 7 : ac->m4ac.object_type == AOT_AAC_MAIN ? 20 : 12; | |
752 for (w = 0; w < ics->num_windows; w++) { | |
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753 if ((tns->n_filt[w] = get_bits(gb, 2 - is8))) { |
7609 | 754 coef_res = get_bits1(gb); |
755 | |
7638 | 756 for (filt = 0; filt < tns->n_filt[w]; filt++) { |
757 int tmp2_idx; | |
758 tns->length[w][filt] = get_bits(gb, 6 - 2*is8); | |
7609 | 759 |
7638 | 760 if ((tns->order[w][filt] = get_bits(gb, 5 - 2*is8)) > tns_max_order) { |
761 av_log(ac->avccontext, AV_LOG_ERROR, "TNS filter order %d is greater than maximum %d.", | |
762 tns->order[w][filt], tns_max_order); | |
763 tns->order[w][filt] = 0; | |
764 return -1; | |
765 } | |
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766 if (tns->order[w][filt]) { |
7851 | 767 tns->direction[w][filt] = get_bits1(gb); |
768 coef_compress = get_bits1(gb); | |
769 coef_len = coef_res + 3 - coef_compress; | |
770 tmp2_idx = 2*coef_compress + coef_res; | |
7638 | 771 |
7851 | 772 for (i = 0; i < tns->order[w][filt]; i++) |
773 tns->coef[w][filt][i] = tns_tmp2_map[tmp2_idx][get_bits(gb, coef_len)]; | |
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774 } |
7609 | 775 } |
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776 } |
7609 | 777 } |
778 return 0; | |
779 } | |
780 | |
781 /** | |
7539 | 782 * Decode Mid/Side data; reference: table 4.54. |
783 * | |
784 * @param ms_present Indicates mid/side stereo presence. [0] mask is all 0s; | |
785 * [1] mask is decoded from bitstream; [2] mask is all 1s; | |
786 * [3] reserved for scalable AAC | |
787 */ | |
788 static void decode_mid_side_stereo(ChannelElement * cpe, GetBitContext * gb, | |
789 int ms_present) { | |
7578 | 790 int idx; |
791 if (ms_present == 1) { | |
792 for (idx = 0; idx < cpe->ch[0].ics.num_window_groups * cpe->ch[0].ics.max_sfb; idx++) | |
793 cpe->ms_mask[idx] = get_bits1(gb); | |
794 } else if (ms_present == 2) { | |
795 memset(cpe->ms_mask, 1, cpe->ch[0].ics.num_window_groups * cpe->ch[0].ics.max_sfb * sizeof(cpe->ms_mask[0])); | |
796 } | |
797 } | |
7539 | 798 |
799 /** | |
7581 | 800 * Decode spectral data; reference: table 4.50. |
801 * Dequantize and scale spectral data; reference: 4.6.3.3. | |
802 * | |
803 * @param coef array of dequantized, scaled spectral data | |
804 * @param sf array of scalefactors or intensity stereo positions | |
805 * @param pulse_present set if pulses are present | |
806 * @param pulse pointer to pulse data struct | |
807 * @param band_type array of the used band type | |
808 * | |
809 * @return Returns error status. 0 - OK, !0 - error | |
810 */ | |
811 static int decode_spectrum_and_dequant(AACContext * ac, float coef[1024], GetBitContext * gb, float sf[120], | |
812 int pulse_present, const Pulse * pulse, const IndividualChannelStream * ics, enum BandType band_type[120]) { | |
813 int i, k, g, idx = 0; | |
814 const int c = 1024/ics->num_windows; | |
815 const uint16_t * offsets = ics->swb_offset; | |
816 float *coef_base = coef; | |
8284 | 817 static const float sign_lookup[] = { 1.0f, -1.0f }; |
7581 | 818 |
819 for (g = 0; g < ics->num_windows; g++) | |
820 memset(coef + g * 128 + offsets[ics->max_sfb], 0, sizeof(float)*(c - offsets[ics->max_sfb])); | |
821 | |
822 for (g = 0; g < ics->num_window_groups; g++) { | |
823 for (i = 0; i < ics->max_sfb; i++, idx++) { | |
824 const int cur_band_type = band_type[idx]; | |
825 const int dim = cur_band_type >= FIRST_PAIR_BT ? 2 : 4; | |
826 const int is_cb_unsigned = IS_CODEBOOK_UNSIGNED(cur_band_type); | |
827 int group; | |
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828 if (cur_band_type == ZERO_BT || cur_band_type == INTENSITY_BT2 || cur_band_type == INTENSITY_BT) { |
7581 | 829 for (group = 0; group < ics->group_len[g]; group++) { |
830 memset(coef + group * 128 + offsets[i], 0, (offsets[i+1] - offsets[i])*sizeof(float)); | |
831 } | |
832 }else if (cur_band_type == NOISE_BT) { | |
833 for (group = 0; group < ics->group_len[g]; group++) { | |
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834 float scale; |
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835 float band_energy = 0; |
7581 | 836 for (k = offsets[i]; k < offsets[i+1]; k++) { |
837 ac->random_state = lcg_random(ac->random_state); | |
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838 coef[group*128+k] = ac->random_state; |
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839 band_energy += coef[group*128+k]*coef[group*128+k]; |
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840 } |
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841 scale = sf[idx] / sqrtf(band_energy); |
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842 for (k = offsets[i]; k < offsets[i+1]; k++) { |
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843 coef[group*128+k] *= scale; |
7581 | 844 } |
845 } | |
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846 }else { |
7581 | 847 for (group = 0; group < ics->group_len[g]; group++) { |
848 for (k = offsets[i]; k < offsets[i+1]; k += dim) { | |
849 const int index = get_vlc2(gb, vlc_spectral[cur_band_type - 1].table, 6, 3); | |
850 const int coef_tmp_idx = (group << 7) + k; | |
851 const float *vq_ptr; | |
852 int j; | |
853 if(index >= ff_aac_spectral_sizes[cur_band_type - 1]) { | |
854 av_log(ac->avccontext, AV_LOG_ERROR, | |
855 "Read beyond end of ff_aac_codebook_vectors[%d][]. index %d >= %d\n", | |
856 cur_band_type - 1, index, ff_aac_spectral_sizes[cur_band_type - 1]); | |
857 return -1; | |
858 } | |
859 vq_ptr = &ff_aac_codebook_vectors[cur_band_type - 1][index * dim]; | |
860 if (is_cb_unsigned) { | |
8284 | 861 if (vq_ptr[0]) coef[coef_tmp_idx ] = sign_lookup[get_bits1(gb)]; |
862 if (vq_ptr[1]) coef[coef_tmp_idx + 1] = sign_lookup[get_bits1(gb)]; | |
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863 if (dim == 4) { |
8284 | 864 if (vq_ptr[2]) coef[coef_tmp_idx + 2] = sign_lookup[get_bits1(gb)]; |
865 if (vq_ptr[3]) coef[coef_tmp_idx + 3] = sign_lookup[get_bits1(gb)]; | |
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866 } |
9053 | 867 if (cur_band_type == ESC_BT) { |
868 for (j = 0; j < 2; j++) { | |
869 if (vq_ptr[j] == 64.0f) { | |
870 int n = 4; | |
871 /* The total length of escape_sequence must be < 22 bits according | |
872 to the specification (i.e. max is 11111111110xxxxxxxxxx). */ | |
873 while (get_bits1(gb) && n < 15) n++; | |
874 if(n == 15) { | |
875 av_log(ac->avccontext, AV_LOG_ERROR, "error in spectral data, ESC overflow\n"); | |
876 return -1; | |
877 } | |
878 n = (1<<n) + get_bits(gb, n); | |
879 coef[coef_tmp_idx + j] *= cbrtf(n) * n; | |
880 }else | |
881 coef[coef_tmp_idx + j] *= vq_ptr[j]; | |
882 } | |
883 }else | |
884 { | |
885 coef[coef_tmp_idx ] *= vq_ptr[0]; | |
886 coef[coef_tmp_idx + 1] *= vq_ptr[1]; | |
887 if (dim == 4) { | |
888 coef[coef_tmp_idx + 2] *= vq_ptr[2]; | |
889 coef[coef_tmp_idx + 3] *= vq_ptr[3]; | |
890 } | |
7581 | 891 } |
9052
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892 }else { |
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|
893 coef[coef_tmp_idx ] = vq_ptr[0]; |
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894 coef[coef_tmp_idx + 1] = vq_ptr[1]; |
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895 if (dim == 4) { |
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|
896 coef[coef_tmp_idx + 2] = vq_ptr[2]; |
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|
897 coef[coef_tmp_idx + 3] = vq_ptr[3]; |
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|
898 } |
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|
899 } |
8283
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|
900 coef[coef_tmp_idx ] *= sf[idx]; |
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901 coef[coef_tmp_idx + 1] *= sf[idx]; |
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902 if (dim == 4) { |
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903 coef[coef_tmp_idx + 2] *= sf[idx]; |
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904 coef[coef_tmp_idx + 3] *= sf[idx]; |
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905 } |
7581 | 906 } |
907 } | |
908 } | |
909 } | |
910 coef += ics->group_len[g]<<7; | |
911 } | |
912 | |
913 if (pulse_present) { | |
7857
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914 idx = 0; |
7581 | 915 for(i = 0; i < pulse->num_pulse; i++){ |
916 float co = coef_base[ pulse->pos[i] ]; | |
7857
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917 while(offsets[idx + 1] <= pulse->pos[i]) |
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|
918 idx++; |
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919 if (band_type[idx] != NOISE_BT && sf[idx]) { |
7858 | 920 float ico = -pulse->amp[i]; |
921 if (co) { | |
922 co /= sf[idx]; | |
923 ico = co / sqrtf(sqrtf(fabsf(co))) + (co > 0 ? -ico : ico); | |
924 } | |
925 coef_base[ pulse->pos[i] ] = cbrtf(fabsf(ico)) * ico * sf[idx]; | |
7857
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926 } |
7581 | 927 } |
928 } | |
929 return 0; | |
930 } | |
931 | |
8199
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932 static av_always_inline float flt16_round(float pf) { |
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933 int exp; |
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934 pf = frexpf(pf, &exp); |
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935 pf = ldexpf(roundf(ldexpf(pf, 8)), exp-8); |
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936 return pf; |
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|
937 } |
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938 |
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939 static av_always_inline float flt16_even(float pf) { |
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940 int exp; |
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941 pf = frexpf(pf, &exp); |
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942 pf = ldexpf(rintf(ldexpf(pf, 8)), exp-8); |
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943 return pf; |
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944 } |
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945 |
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946 static av_always_inline float flt16_trunc(float pf) { |
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947 int exp; |
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948 pf = frexpf(pf, &exp); |
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949 pf = ldexpf(truncf(ldexpf(pf, 8)), exp-8); |
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950 return pf; |
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951 } |
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952 |
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953 static void predict(AACContext * ac, PredictorState * ps, float* coef, int output_enable) { |
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954 const float a = 0.953125; // 61.0/64 |
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955 const float alpha = 0.90625; // 29.0/32 |
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956 float e0, e1; |
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957 float pv; |
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958 float k1, k2; |
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959 |
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960 k1 = ps->var0 > 1 ? ps->cor0 * flt16_even(a / ps->var0) : 0; |
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961 k2 = ps->var1 > 1 ? ps->cor1 * flt16_even(a / ps->var1) : 0; |
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962 |
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963 pv = flt16_round(k1 * ps->r0 + k2 * ps->r1); |
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964 if (output_enable) |
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965 *coef += pv * ac->sf_scale; |
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966 |
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967 e0 = *coef / ac->sf_scale; |
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968 e1 = e0 - k1 * ps->r0; |
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969 |
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970 ps->cor1 = flt16_trunc(alpha * ps->cor1 + ps->r1 * e1); |
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971 ps->var1 = flt16_trunc(alpha * ps->var1 + 0.5 * (ps->r1 * ps->r1 + e1 * e1)); |
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972 ps->cor0 = flt16_trunc(alpha * ps->cor0 + ps->r0 * e0); |
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973 ps->var0 = flt16_trunc(alpha * ps->var0 + 0.5 * (ps->r0 * ps->r0 + e0 * e0)); |
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974 |
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975 ps->r1 = flt16_trunc(a * (ps->r0 - k1 * e0)); |
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976 ps->r0 = flt16_trunc(a * e0); |
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977 } |
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978 |
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979 /** |
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|
980 * Apply AAC-Main style frequency domain prediction. |
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|
981 */ |
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982 static void apply_prediction(AACContext * ac, SingleChannelElement * sce) { |
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983 int sfb, k; |
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984 |
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985 if (!sce->ics.predictor_initialized) { |
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986 reset_all_predictors(sce->predictor_state); |
8199
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987 sce->ics.predictor_initialized = 1; |
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|
988 } |
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989 |
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990 if (sce->ics.window_sequence[0] != EIGHT_SHORT_SEQUENCE) { |
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991 for (sfb = 0; sfb < ff_aac_pred_sfb_max[ac->m4ac.sampling_index]; sfb++) { |
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992 for (k = sce->ics.swb_offset[sfb]; k < sce->ics.swb_offset[sfb + 1]; k++) { |
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993 predict(ac, &sce->predictor_state[k], &sce->coeffs[k], |
8199
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994 sce->ics.predictor_present && sce->ics.prediction_used[sfb]); |
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995 } |
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996 } |
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997 if (sce->ics.predictor_reset_group) |
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998 reset_predictor_group(sce->predictor_state, sce->ics.predictor_reset_group); |
8199
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999 } else |
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1000 reset_all_predictors(sce->predictor_state); |
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1001 } |
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1002 |
7581 | 1003 /** |
7539 | 1004 * Decode an individual_channel_stream payload; reference: table 4.44. |
1005 * | |
1006 * @param common_window Channels have independent [0], or shared [1], Individual Channel Stream information. | |
1007 * @param scale_flag scalable [1] or non-scalable [0] AAC (Unused until scalable AAC is implemented.) | |
1008 * | |
1009 * @return Returns error status. 0 - OK, !0 - error | |
1010 */ | |
1011 static int decode_ics(AACContext * ac, SingleChannelElement * sce, GetBitContext * gb, int common_window, int scale_flag) { | |
1012 Pulse pulse; | |
1013 TemporalNoiseShaping * tns = &sce->tns; | |
1014 IndividualChannelStream * ics = &sce->ics; | |
1015 float * out = sce->coeffs; | |
1016 int global_gain, pulse_present = 0; | |
1017 | |
7580 | 1018 /* This assignment is to silence a GCC warning about the variable being used |
1019 * uninitialized when in fact it always is. | |
7539 | 1020 */ |
1021 pulse.num_pulse = 0; | |
1022 | |
1023 global_gain = get_bits(gb, 8); | |
1024 | |
1025 if (!common_window && !scale_flag) { | |
1026 if (decode_ics_info(ac, ics, gb, 0) < 0) | |
1027 return -1; | |
1028 } | |
1029 | |
1030 if (decode_band_types(ac, sce->band_type, sce->band_type_run_end, gb, ics) < 0) | |
1031 return -1; | |
1032 if (decode_scalefactors(ac, sce->sf, gb, global_gain, ics, sce->band_type, sce->band_type_run_end) < 0) | |
1033 return -1; | |
1034 | |
1035 pulse_present = 0; | |
1036 if (!scale_flag) { | |
1037 if ((pulse_present = get_bits1(gb))) { | |
1038 if (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) { | |
1039 av_log(ac->avccontext, AV_LOG_ERROR, "Pulse tool not allowed in eight short sequence.\n"); | |
1040 return -1; | |
1041 } | |
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1042 if (decode_pulses(&pulse, gb, ics->swb_offset, ics->num_swb)) { |
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1043 av_log(ac->avccontext, AV_LOG_ERROR, "Pulse data corrupt or invalid.\n"); |
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1044 return -1; |
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1045 } |
7539 | 1046 } |
1047 if ((tns->present = get_bits1(gb)) && decode_tns(ac, tns, gb, ics)) | |
1048 return -1; | |
1049 if (get_bits1(gb)) { | |
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1050 ff_log_missing_feature(ac->avccontext, "SSR", 1); |
7539 | 1051 return -1; |
1052 } | |
1053 } | |
1054 | |
7580 | 1055 if (decode_spectrum_and_dequant(ac, out, gb, sce->sf, pulse_present, &pulse, ics, sce->band_type) < 0) |
7539 | 1056 return -1; |
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1057 |
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1058 if(ac->m4ac.object_type == AOT_AAC_MAIN && !common_window) |
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1059 apply_prediction(ac, sce); |
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1060 |
7539 | 1061 return 0; |
1062 } | |
1063 | |
1064 /** | |
7581 | 1065 * Mid/Side stereo decoding; reference: 4.6.8.1.3. |
1066 */ | |
1067 static void apply_mid_side_stereo(ChannelElement * cpe) { | |
1068 const IndividualChannelStream * ics = &cpe->ch[0].ics; | |
1069 float *ch0 = cpe->ch[0].coeffs; | |
1070 float *ch1 = cpe->ch[1].coeffs; | |
1071 int g, i, k, group, idx = 0; | |
1072 const uint16_t * offsets = ics->swb_offset; | |
1073 for (g = 0; g < ics->num_window_groups; g++) { | |
1074 for (i = 0; i < ics->max_sfb; i++, idx++) { | |
1075 if (cpe->ms_mask[idx] && | |
1076 cpe->ch[0].band_type[idx] < NOISE_BT && cpe->ch[1].band_type[idx] < NOISE_BT) { | |
1077 for (group = 0; group < ics->group_len[g]; group++) { | |
1078 for (k = offsets[i]; k < offsets[i+1]; k++) { | |
1079 float tmp = ch0[group*128 + k] - ch1[group*128 + k]; | |
1080 ch0[group*128 + k] += ch1[group*128 + k]; | |
1081 ch1[group*128 + k] = tmp; | |
1082 } | |
1083 } | |
1084 } | |
1085 } | |
1086 ch0 += ics->group_len[g]*128; | |
1087 ch1 += ics->group_len[g]*128; | |
1088 } | |
1089 } | |
1090 | |
1091 /** | |
1092 * intensity stereo decoding; reference: 4.6.8.2.3 | |
1093 * | |
1094 * @param ms_present Indicates mid/side stereo presence. [0] mask is all 0s; | |
1095 * [1] mask is decoded from bitstream; [2] mask is all 1s; | |
1096 * [3] reserved for scalable AAC | |
1097 */ | |
1098 static void apply_intensity_stereo(ChannelElement * cpe, int ms_present) { | |
1099 const IndividualChannelStream * ics = &cpe->ch[1].ics; | |
1100 SingleChannelElement * sce1 = &cpe->ch[1]; | |
1101 float *coef0 = cpe->ch[0].coeffs, *coef1 = cpe->ch[1].coeffs; | |
1102 const uint16_t * offsets = ics->swb_offset; | |
1103 int g, group, i, k, idx = 0; | |
1104 int c; | |
1105 float scale; | |
1106 for (g = 0; g < ics->num_window_groups; g++) { | |
1107 for (i = 0; i < ics->max_sfb;) { | |
1108 if (sce1->band_type[idx] == INTENSITY_BT || sce1->band_type[idx] == INTENSITY_BT2) { | |
1109 const int bt_run_end = sce1->band_type_run_end[idx]; | |
1110 for (; i < bt_run_end; i++, idx++) { | |
1111 c = -1 + 2 * (sce1->band_type[idx] - 14); | |
1112 if (ms_present) | |
1113 c *= 1 - 2 * cpe->ms_mask[idx]; | |
1114 scale = c * sce1->sf[idx]; | |
1115 for (group = 0; group < ics->group_len[g]; group++) | |
1116 for (k = offsets[i]; k < offsets[i+1]; k++) | |
1117 coef1[group*128 + k] = scale * coef0[group*128 + k]; | |
1118 } | |
1119 } else { | |
1120 int bt_run_end = sce1->band_type_run_end[idx]; | |
1121 idx += bt_run_end - i; | |
1122 i = bt_run_end; | |
1123 } | |
1124 } | |
1125 coef0 += ics->group_len[g]*128; | |
1126 coef1 += ics->group_len[g]*128; | |
1127 } | |
1128 } | |
1129 | |
1130 /** | |
7539 | 1131 * Decode a channel_pair_element; reference: table 4.4. |
1132 * | |
1133 * @param elem_id Identifies the instance of a syntax element. | |
1134 * | |
1135 * @return Returns error status. 0 - OK, !0 - error | |
1136 */ | |
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1137 static int decode_cpe(AACContext * ac, GetBitContext * gb, ChannelElement * cpe) { |
7539 | 1138 int i, ret, common_window, ms_present = 0; |
1139 | |
1140 common_window = get_bits1(gb); | |
1141 if (common_window) { | |
1142 if (decode_ics_info(ac, &cpe->ch[0].ics, gb, 1)) | |
1143 return -1; | |
1144 i = cpe->ch[1].ics.use_kb_window[0]; | |
1145 cpe->ch[1].ics = cpe->ch[0].ics; | |
1146 cpe->ch[1].ics.use_kb_window[1] = i; | |
1147 ms_present = get_bits(gb, 2); | |
1148 if(ms_present == 3) { | |
1149 av_log(ac->avccontext, AV_LOG_ERROR, "ms_present = 3 is reserved.\n"); | |
1150 return -1; | |
1151 } else if(ms_present) | |
1152 decode_mid_side_stereo(cpe, gb, ms_present); | |
1153 } | |
1154 if ((ret = decode_ics(ac, &cpe->ch[0], gb, common_window, 0))) | |
1155 return ret; | |
1156 if ((ret = decode_ics(ac, &cpe->ch[1], gb, common_window, 0))) | |
1157 return ret; | |
1158 | |
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1159 if (common_window) { |
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|
1160 if (ms_present) |
8414 | 1161 apply_mid_side_stereo(cpe); |
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1162 if (ac->m4ac.object_type == AOT_AAC_MAIN) { |
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1163 apply_prediction(ac, &cpe->ch[0]); |
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1164 apply_prediction(ac, &cpe->ch[1]); |
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|
1165 } |
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|
1166 } |
7539 | 1167 |
7580 | 1168 apply_intensity_stereo(cpe, ms_present); |
7539 | 1169 return 0; |
1170 } | |
1171 | |
7581 | 1172 /** |
1173 * Decode coupling_channel_element; reference: table 4.8. | |
1174 * | |
1175 * @param elem_id Identifies the instance of a syntax element. | |
1176 * | |
1177 * @return Returns error status. 0 - OK, !0 - error | |
1178 */ | |
1179 static int decode_cce(AACContext * ac, GetBitContext * gb, ChannelElement * che) { | |
1180 int num_gain = 0; | |
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1181 int c, g, sfb, ret; |
7581 | 1182 int sign; |
1183 float scale; | |
1184 SingleChannelElement * sce = &che->ch[0]; | |
1185 ChannelCoupling * coup = &che->coup; | |
1186 | |
7578 | 1187 coup->coupling_point = 2*get_bits1(gb); |
1188 coup->num_coupled = get_bits(gb, 3); | |
1189 for (c = 0; c <= coup->num_coupled; c++) { | |
1190 num_gain++; | |
1191 coup->type[c] = get_bits1(gb) ? TYPE_CPE : TYPE_SCE; | |
1192 coup->id_select[c] = get_bits(gb, 4); | |
1193 if (coup->type[c] == TYPE_CPE) { | |
1194 coup->ch_select[c] = get_bits(gb, 2); | |
1195 if (coup->ch_select[c] == 3) | |
1196 num_gain++; | |
1197 } else | |
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1198 coup->ch_select[c] = 2; |
7578 | 1199 } |
1200 coup->coupling_point += get_bits1(gb); | |
1201 | |
1202 if (coup->coupling_point == 2) { | |
1203 av_log(ac->avccontext, AV_LOG_ERROR, | |
1204 "Independently switched CCE with 'invalid' domain signalled.\n"); | |
1205 memset(coup, 0, sizeof(ChannelCoupling)); | |
1206 return -1; | |
1207 } | |
1208 | |
1209 sign = get_bits(gb, 1); | |
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1210 scale = pow(2., pow(2., (int)get_bits(gb, 2) - 3)); |
7578 | 1211 |
1212 if ((ret = decode_ics(ac, sce, gb, 0, 0))) | |
1213 return ret; | |
1214 | |
1215 for (c = 0; c < num_gain; c++) { | |
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1216 int idx = 0; |
7578 | 1217 int cge = 1; |
1218 int gain = 0; | |
1219 float gain_cache = 1.; | |
1220 if (c) { | |
1221 cge = coup->coupling_point == AFTER_IMDCT ? 1 : get_bits1(gb); | |
1222 gain = cge ? get_vlc2(gb, vlc_scalefactors.table, 7, 3) - 60: 0; | |
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1223 gain_cache = pow(scale, -gain); |
7578 | 1224 } |
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1225 if (coup->coupling_point == AFTER_IMDCT) { |
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1226 coup->gain[c][0] = gain_cache; |
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1227 } else { |
8823 | 1228 for (g = 0; g < sce->ics.num_window_groups; g++) { |
1229 for (sfb = 0; sfb < sce->ics.max_sfb; sfb++, idx++) { | |
1230 if (sce->band_type[idx] != ZERO_BT) { | |
1231 if (!cge) { | |
1232 int t = get_vlc2(gb, vlc_scalefactors.table, 7, 3) - 60; | |
1233 if (t) { | |
1234 int s = 1; | |
1235 t = gain += t; | |
1236 if (sign) { | |
1237 s -= 2 * (t & 0x1); | |
1238 t >>= 1; | |
1239 } | |
1240 gain_cache = pow(scale, -t) * s; | |
7578 | 1241 } |
1242 } | |
8823 | 1243 coup->gain[c][idx] = gain_cache; |
7578 | 1244 } |
1245 } | |
7890 | 1246 } |
1247 } | |
7578 | 1248 } |
1249 return 0; | |
1250 } | |
1251 | |
7539 | 1252 /** |
1253 * Decode Spectral Band Replication extension data; reference: table 4.55. | |
7523
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1254 * |
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1255 * @param crc flag indicating the presence of CRC checksum |
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1256 * @param cnt length of TYPE_FIL syntactic element in bytes |
7539 | 1257 * |
7523
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1258 * @return Returns number of bytes consumed from the TYPE_FIL element. |
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1259 */ |
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1260 static int decode_sbr_extension(AACContext * ac, GetBitContext * gb, int crc, int cnt) { |
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1261 // TODO : sbr_extension implementation |
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1262 ff_log_missing_feature(ac->avccontext, "SBR", 0); |
7523
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1263 skip_bits_long(gb, 8*cnt - 4); // -4 due to reading extension type |
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1264 return cnt; |
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1265 } |
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1266 |
7539 | 1267 /** |
7578 | 1268 * Parse whether channels are to be excluded from Dynamic Range Compression; reference: table 4.53. |
1269 * | |
1270 * @return Returns number of bytes consumed. | |
1271 */ | |
1272 static int decode_drc_channel_exclusions(DynamicRangeControl *che_drc, GetBitContext * gb) { | |
1273 int i; | |
1274 int num_excl_chan = 0; | |
1275 | |
1276 do { | |
1277 for (i = 0; i < 7; i++) | |
1278 che_drc->exclude_mask[num_excl_chan++] = get_bits1(gb); | |
1279 } while (num_excl_chan < MAX_CHANNELS - 7 && get_bits1(gb)); | |
1280 | |
1281 return num_excl_chan / 7; | |
1282 } | |
1283 | |
1284 /** | |
7539 | 1285 * Decode dynamic range information; reference: table 4.52. |
1286 * | |
1287 * @param cnt length of TYPE_FIL syntactic element in bytes | |
1288 * | |
1289 * @return Returns number of bytes consumed. | |
1290 */ | |
1291 static int decode_dynamic_range(DynamicRangeControl *che_drc, GetBitContext * gb, int cnt) { | |
1292 int n = 1; | |
1293 int drc_num_bands = 1; | |
1294 int i; | |
1295 | |
1296 /* pce_tag_present? */ | |
1297 if(get_bits1(gb)) { | |
1298 che_drc->pce_instance_tag = get_bits(gb, 4); | |
1299 skip_bits(gb, 4); // tag_reserved_bits | |
1300 n++; | |
1301 } | |
1302 | |
1303 /* excluded_chns_present? */ | |
1304 if(get_bits1(gb)) { | |
1305 n += decode_drc_channel_exclusions(che_drc, gb); | |
1306 } | |
1307 | |
1308 /* drc_bands_present? */ | |
1309 if (get_bits1(gb)) { | |
1310 che_drc->band_incr = get_bits(gb, 4); | |
1311 che_drc->interpolation_scheme = get_bits(gb, 4); | |
1312 n++; | |
1313 drc_num_bands += che_drc->band_incr; | |
1314 for (i = 0; i < drc_num_bands; i++) { | |
1315 che_drc->band_top[i] = get_bits(gb, 8); | |
1316 n++; | |
1317 } | |
1318 } | |
1319 | |
1320 /* prog_ref_level_present? */ | |
1321 if (get_bits1(gb)) { | |
1322 che_drc->prog_ref_level = get_bits(gb, 7); | |
1323 skip_bits1(gb); // prog_ref_level_reserved_bits | |
1324 n++; | |
1325 } | |
1326 | |
1327 for (i = 0; i < drc_num_bands; i++) { | |
1328 che_drc->dyn_rng_sgn[i] = get_bits1(gb); | |
1329 che_drc->dyn_rng_ctl[i] = get_bits(gb, 7); | |
1330 n++; | |
1331 } | |
1332 | |
1333 return n; | |
1334 } | |
1335 | |
1336 /** | |
1337 * Decode extension data (incomplete); reference: table 4.51. | |
1338 * | |
1339 * @param cnt length of TYPE_FIL syntactic element in bytes | |
1340 * | |
1341 * @return Returns number of bytes consumed | |
1342 */ | |
1343 static int decode_extension_payload(AACContext * ac, GetBitContext * gb, int cnt) { | |
7523
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1344 int crc_flag = 0; |
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1345 int res = cnt; |
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1346 switch (get_bits(gb, 4)) { // extension type |
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1347 case EXT_SBR_DATA_CRC: |
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1348 crc_flag++; |
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1349 case EXT_SBR_DATA: |
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1350 res = decode_sbr_extension(ac, gb, crc_flag, cnt); |
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1351 break; |
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1352 case EXT_DYNAMIC_RANGE: |
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1353 res = decode_dynamic_range(&ac->che_drc, gb, cnt); |
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1354 break; |
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1355 case EXT_FILL: |
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1356 case EXT_FILL_DATA: |
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1357 case EXT_DATA_ELEMENT: |
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1358 default: |
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1359 skip_bits_long(gb, 8*cnt - 4); |
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1360 break; |
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1361 }; |
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1362 return res; |
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1363 } |
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1364 |
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1365 /** |
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1366 * Decode Temporal Noise Shaping filter coefficients and apply all-pole filters; reference: 4.6.9.3. |
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1367 * |
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1368 * @param decode 1 if tool is used normally, 0 if tool is used in LTP. |
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1369 * @param coef spectral coefficients |
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1370 */ |
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1371 static void apply_tns(float coef[1024], TemporalNoiseShaping * tns, IndividualChannelStream * ics, int decode) { |
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1372 const int mmm = FFMIN(ics->tns_max_bands, ics->max_sfb); |
7636 | 1373 int w, filt, m, i; |
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1374 int bottom, top, order, start, end, size, inc; |
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1375 float lpc[TNS_MAX_ORDER]; |
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1376 |
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1377 for (w = 0; w < ics->num_windows; w++) { |
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1378 bottom = ics->num_swb; |
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1379 for (filt = 0; filt < tns->n_filt[w]; filt++) { |
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1380 top = bottom; |
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1381 bottom = FFMAX(0, top - tns->length[w][filt]); |
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1382 order = tns->order[w][filt]; |
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1383 if (order == 0) |
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1384 continue; |
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1385 |
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1386 // tns_decode_coef |
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1387 compute_lpc_coefs(tns->coef[w][filt], order, lpc, 0, 0, 0); |
7632
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1388 |
7609 | 1389 start = ics->swb_offset[FFMIN(bottom, mmm)]; |
1390 end = ics->swb_offset[FFMIN( top, mmm)]; | |
1391 if ((size = end - start) <= 0) | |
1392 continue; | |
1393 if (tns->direction[w][filt]) { | |
1394 inc = -1; start = end - 1; | |
1395 } else { | |
1396 inc = 1; | |
1397 } | |
1398 start += w * 128; | |
1399 | |
1400 // ar filter | |
1401 for (m = 0; m < size; m++, start += inc) | |
1402 for (i = 1; i <= FFMIN(m, order); i++) | |
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1403 coef[start] -= coef[start - i*inc] * lpc[i-1]; |
7609 | 1404 } |
1405 } | |
1406 } | |
1407 | |
7523
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1408 /** |
7539 | 1409 * Conduct IMDCT and windowing. |
1410 */ | |
1411 static void imdct_and_windowing(AACContext * ac, SingleChannelElement * sce) { | |
1412 IndividualChannelStream * ics = &sce->ics; | |
1413 float * in = sce->coeffs; | |
1414 float * out = sce->ret; | |
1415 float * saved = sce->saved; | |
7580 | 1416 const float * swindow = ics->use_kb_window[0] ? ff_aac_kbd_short_128 : ff_sine_128; |
1417 const float * lwindow_prev = ics->use_kb_window[1] ? ff_aac_kbd_long_1024 : ff_sine_1024; | |
1418 const float * swindow_prev = ics->use_kb_window[1] ? ff_aac_kbd_short_128 : ff_sine_128; | |
7539 | 1419 float * buf = ac->buf_mdct; |
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1420 float * temp = ac->temp; |
7539 | 1421 int i; |
1422 | |
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1423 // imdct |
7578 | 1424 if (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) { |
1425 if (ics->window_sequence[1] == ONLY_LONG_SEQUENCE || ics->window_sequence[1] == LONG_STOP_SEQUENCE) | |
1426 av_log(ac->avccontext, AV_LOG_WARNING, | |
1427 "Transition from an ONLY_LONG or LONG_STOP to an EIGHT_SHORT sequence detected. " | |
1428 "If you heard an audible artifact, please submit the sample to the FFmpeg developers.\n"); | |
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1429 for (i = 0; i < 1024; i += 128) |
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1430 ff_imdct_half(&ac->mdct_small, buf + i, in + i); |
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1431 } else |
7663
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1432 ff_imdct_half(&ac->mdct, buf, in); |
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1433 |
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1434 /* window overlapping |
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1435 * NOTE: To simplify the overlapping code, all 'meaningless' short to long |
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1436 * and long to short transitions are considered to be short to short |
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1437 * transitions. This leaves just two cases (long to long and short to short) |
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1438 * with a little special sauce for EIGHT_SHORT_SEQUENCE. |
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1439 */ |
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1440 if ((ics->window_sequence[1] == ONLY_LONG_SEQUENCE || ics->window_sequence[1] == LONG_STOP_SEQUENCE) && |
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1441 (ics->window_sequence[0] == ONLY_LONG_SEQUENCE || ics->window_sequence[0] == LONG_START_SEQUENCE)) { |
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1442 ac->dsp.vector_fmul_window( out, saved, buf, lwindow_prev, ac->add_bias, 512); |
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1443 } else { |
7662 | 1444 for (i = 0; i < 448; i++) |
1445 out[i] = saved[i] + ac->add_bias; | |
7578 | 1446 |
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1447 if (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) { |
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1448 ac->dsp.vector_fmul_window(out + 448 + 0*128, saved + 448, buf + 0*128, swindow_prev, ac->add_bias, 64); |
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1449 ac->dsp.vector_fmul_window(out + 448 + 1*128, buf + 0*128 + 64, buf + 1*128, swindow, ac->add_bias, 64); |
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1450 ac->dsp.vector_fmul_window(out + 448 + 2*128, buf + 1*128 + 64, buf + 2*128, swindow, ac->add_bias, 64); |
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1451 ac->dsp.vector_fmul_window(out + 448 + 3*128, buf + 2*128 + 64, buf + 3*128, swindow, ac->add_bias, 64); |
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1452 ac->dsp.vector_fmul_window(temp, buf + 3*128 + 64, buf + 4*128, swindow, ac->add_bias, 64); |
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1453 memcpy( out + 448 + 4*128, temp, 64 * sizeof(float)); |
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1454 } else { |
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1455 ac->dsp.vector_fmul_window(out + 448, saved + 448, buf, swindow_prev, ac->add_bias, 64); |
7662 | 1456 for (i = 576; i < 1024; i++) |
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1457 out[i] = buf[i-512] + ac->add_bias; |
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1458 } |
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1459 } |
7578 | 1460 |
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1461 // buffer update |
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1462 if (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) { |
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1463 for (i = 0; i < 64; i++) |
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1464 saved[i] = temp[64 + i] - ac->add_bias; |
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1465 ac->dsp.vector_fmul_window(saved + 64, buf + 4*128 + 64, buf + 5*128, swindow, 0, 64); |
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1466 ac->dsp.vector_fmul_window(saved + 192, buf + 5*128 + 64, buf + 6*128, swindow, 0, 64); |
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1467 ac->dsp.vector_fmul_window(saved + 320, buf + 6*128 + 64, buf + 7*128, swindow, 0, 64); |
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1468 memcpy( saved + 448, buf + 7*128 + 64, 64 * sizeof(float)); |
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1469 } else if (ics->window_sequence[0] == LONG_START_SEQUENCE) { |
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|
1470 memcpy( saved, buf + 512, 448 * sizeof(float)); |
66fb1f3937fa
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1471 memcpy( saved + 448, buf + 7*128 + 64, 64 * sizeof(float)); |
7661
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1472 } else { // LONG_STOP or ONLY_LONG |
7663
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|
1473 memcpy( saved, buf + 512, 512 * sizeof(float)); |
7578 | 1474 } |
1475 } | |
1476 | |
7539 | 1477 /** |
7523
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1478 * Apply dependent channel coupling (applied before IMDCT). |
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1479 * |
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1480 * @param index index into coupling gain array |
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1481 */ |
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1482 static void apply_dependent_coupling(AACContext * ac, SingleChannelElement * target, ChannelElement * cce, int index) { |
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1483 IndividualChannelStream * ics = &cce->ch[0].ics; |
7523
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1484 const uint16_t * offsets = ics->swb_offset; |
8006
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1485 float * dest = target->coeffs; |
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1486 const float * src = cce->ch[0].coeffs; |
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1487 int g, i, group, k, idx = 0; |
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1488 if(ac->m4ac.object_type == AOT_AAC_LTP) { |
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1489 av_log(ac->avccontext, AV_LOG_ERROR, |
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1490 "Dependent coupling is not supported together with LTP\n"); |
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1491 return; |
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1492 } |
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1493 for (g = 0; g < ics->num_window_groups; g++) { |
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1494 for (i = 0; i < ics->max_sfb; i++, idx++) { |
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1495 if (cce->ch[0].band_type[idx] != ZERO_BT) { |
9143 | 1496 const float gain = cce->coup.gain[index][idx]; |
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1497 for (group = 0; group < ics->group_len[g]; group++) { |
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1498 for (k = offsets[i]; k < offsets[i+1]; k++) { |
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1499 // XXX dsputil-ize |
9143 | 1500 dest[group*128+k] += gain * src[group*128+k]; |
7523
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1501 } |
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1502 } |
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1503 } |
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1504 } |
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1505 dest += ics->group_len[g]*128; |
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1506 src += ics->group_len[g]*128; |
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1507 } |
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1508 } |
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1509 |
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1510 /** |
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1511 * Apply independent channel coupling (applied after IMDCT). |
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1512 * |
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1513 * @param index index into coupling gain array |
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1514 */ |
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1515 static void apply_independent_coupling(AACContext * ac, SingleChannelElement * target, ChannelElement * cce, int index) { |
7523
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1516 int i; |
9051
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1517 const float gain = cce->coup.gain[index][0]; |
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1518 const float bias = ac->add_bias; |
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1519 const float* src = cce->ch[0].ret; |
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1520 float* dest = target->ret; |
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1521 |
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1522 for (i = 0; i < 1024; i++) |
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1523 dest[i] += gain * (src[i] - bias); |
7523
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1524 } |
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1525 |
7581 | 1526 /** |
1527 * channel coupling transformation interface | |
1528 * | |
1529 * @param index index into coupling gain array | |
1530 * @param apply_coupling_method pointer to (in)dependent coupling function | |
1531 */ | |
1532 static void apply_channel_coupling(AACContext * ac, ChannelElement * cc, | |
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1533 enum RawDataBlockType type, int elem_id, enum CouplingPoint coupling_point, |
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1534 void (*apply_coupling_method)(AACContext * ac, SingleChannelElement * target, ChannelElement * cce, int index)) |
7581 | 1535 { |
8004
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1536 int i, c; |
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1537 |
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1538 for (i = 0; i < MAX_ELEM_ID; i++) { |
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1539 ChannelElement *cce = ac->che[TYPE_CCE][i]; |
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1540 int index = 0; |
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1541 |
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1542 if (cce && cce->coup.coupling_point == coupling_point) { |
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1543 ChannelCoupling * coup = &cce->coup; |
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1544 |
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1545 for (c = 0; c <= coup->num_coupled; c++) { |
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1546 if (coup->type[c] == type && coup->id_select[c] == elem_id) { |
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1547 if (coup->ch_select[c] != 1) { |
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1548 apply_coupling_method(ac, &cc->ch[0], cce, index); |
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1549 if (coup->ch_select[c] != 0) |
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1550 index++; |
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1551 } |
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1552 if (coup->ch_select[c] != 2) |
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1553 apply_coupling_method(ac, &cc->ch[1], cce, index++); |
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1554 } else |
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1555 index += 1 + (coup->ch_select[c] == 3); |
7581 | 1556 } |
1557 } | |
1558 } | |
1559 } | |
1560 | |
1561 /** | |
1562 * Convert spectral data to float samples, applying all supported tools as appropriate. | |
1563 */ | |
1564 static void spectral_to_sample(AACContext * ac) { | |
8502 | 1565 int i, type; |
8004
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1566 for(type = 3; type >= 0; type--) { |
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1567 for (i = 0; i < MAX_ELEM_ID; i++) { |
7581 | 1568 ChannelElement *che = ac->che[type][i]; |
1569 if(che) { | |
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1570 if(type <= TYPE_CPE) |
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1571 apply_channel_coupling(ac, che, type, i, BEFORE_TNS, apply_dependent_coupling); |
7581 | 1572 if(che->ch[0].tns.present) |
1573 apply_tns(che->ch[0].coeffs, &che->ch[0].tns, &che->ch[0].ics, 1); | |
1574 if(che->ch[1].tns.present) | |
1575 apply_tns(che->ch[1].coeffs, &che->ch[1].tns, &che->ch[1].ics, 1); | |
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1576 if(type <= TYPE_CPE) |
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1577 apply_channel_coupling(ac, che, type, i, BETWEEN_TNS_AND_IMDCT, apply_dependent_coupling); |
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1578 if(type != TYPE_CCE || che->coup.coupling_point == AFTER_IMDCT) |
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1579 imdct_and_windowing(ac, &che->ch[0]); |
7581 | 1580 if(type == TYPE_CPE) |
1581 imdct_and_windowing(ac, &che->ch[1]); | |
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1582 if(type <= TYPE_CCE) |
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1583 apply_channel_coupling(ac, che, type, i, AFTER_IMDCT, apply_independent_coupling); |
7578 | 1584 } |
1585 } | |
1586 } | |
1587 } | |
1588 | |
8549
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1589 static int parse_adts_frame_header(AACContext * ac, GetBitContext * gb) { |
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1590 |
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1591 int size; |
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1592 AACADTSHeaderInfo hdr_info; |
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1593 |
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1594 size = ff_aac_parse_header(gb, &hdr_info); |
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1595 if (size > 0) { |
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1596 if (hdr_info.chan_config) |
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1597 ac->m4ac.chan_config = hdr_info.chan_config; |
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1598 ac->m4ac.sample_rate = hdr_info.sample_rate; |
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1599 ac->m4ac.sampling_index = hdr_info.sampling_index; |
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1600 ac->m4ac.object_type = hdr_info.object_type; |
9151 | 1601 if (hdr_info.num_aac_frames == 1) { |
1602 if (!hdr_info.crc_absent) | |
1603 skip_bits(gb, 16); | |
1604 } else { | |
1605 ff_log_missing_feature(ac->avccontext, "More than one AAC RDB per ADTS frame is", 0); | |
1606 return -1; | |
1607 } | |
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1608 } |
8549
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1609 return size; |
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1610 } |
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1611 |
7578 | 1612 static int aac_decode_frame(AVCodecContext * avccontext, void * data, int * data_size, const uint8_t * buf, int buf_size) { |
1613 AACContext * ac = avccontext->priv_data; | |
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1614 ChannelElement * che = NULL; |
7578 | 1615 GetBitContext gb; |
1616 enum RawDataBlockType elem_type; | |
1617 int err, elem_id, data_size_tmp; | |
1618 | |
1619 init_get_bits(&gb, buf, buf_size*8); | |
1620 | |
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1621 if (show_bits(&gb, 12) == 0xfff) { |
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1622 if ((err = parse_adts_frame_header(ac, &gb)) < 0) { |
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1623 av_log(avccontext, AV_LOG_ERROR, "Error decoding AAC frame header.\n"); |
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1624 return -1; |
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1625 } |
8812 | 1626 if (ac->m4ac.sampling_index > 12) { |
8789 | 1627 av_log(ac->avccontext, AV_LOG_ERROR, "invalid sampling rate index %d\n", ac->m4ac.sampling_index); |
1628 return -1; | |
1629 } | |
8549
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1630 } |
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1631 |
7578 | 1632 // parse |
1633 while ((elem_type = get_bits(&gb, 3)) != TYPE_END) { | |
1634 elem_id = get_bits(&gb, 4); | |
1635 err = -1; | |
1636 | |
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1637 if(elem_type < TYPE_DSE && !(che=get_che(ac, elem_type, elem_id))) { |
8814 | 1638 av_log(ac->avccontext, AV_LOG_ERROR, "channel element %d.%d is not allocated\n", elem_type, elem_id); |
1639 return -1; | |
7578 | 1640 } |
1641 | |
1642 switch (elem_type) { | |
1643 | |
1644 case TYPE_SCE: | |
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1645 err = decode_ics(ac, &che->ch[0], &gb, 0, 0); |
7578 | 1646 break; |
1647 | |
1648 case TYPE_CPE: | |
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1649 err = decode_cpe(ac, &gb, che); |
7578 | 1650 break; |
1651 | |
1652 case TYPE_CCE: | |
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1653 err = decode_cce(ac, &gb, che); |
7578 | 1654 break; |
1655 | |
1656 case TYPE_LFE: | |
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1657 err = decode_ics(ac, &che->ch[0], &gb, 0, 0); |
7578 | 1658 break; |
1659 | |
1660 case TYPE_DSE: | |
1661 skip_data_stream_element(&gb); | |
1662 err = 0; | |
1663 break; | |
1664 | |
1665 case TYPE_PCE: | |
1666 { | |
1667 enum ChannelPosition new_che_pos[4][MAX_ELEM_ID]; | |
1668 memset(new_che_pos, 0, 4 * MAX_ELEM_ID * sizeof(new_che_pos[0][0])); | |
1669 if((err = decode_pce(ac, new_che_pos, &gb))) | |
1670 break; | |
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1671 err = output_configure(ac, ac->che_pos, new_che_pos, 0); |
7578 | 1672 break; |
1673 } | |
1674 | |
1675 case TYPE_FIL: | |
1676 if (elem_id == 15) | |
1677 elem_id += get_bits(&gb, 8) - 1; | |
1678 while (elem_id > 0) | |
1679 elem_id -= decode_extension_payload(ac, &gb, elem_id); | |
1680 err = 0; /* FIXME */ | |
1681 break; | |
1682 | |
1683 default: | |
1684 err = -1; /* should not happen, but keeps compiler happy */ | |
1685 break; | |
1686 } | |
1687 | |
1688 if(err) | |
1689 return err; | |
1690 } | |
1691 | |
1692 spectral_to_sample(ac); | |
1693 | |
7539 | 1694 if (!ac->is_saved) { |
1695 ac->is_saved = 1; | |
1696 *data_size = 0; | |
7580 | 1697 return buf_size; |
7539 | 1698 } |
1699 | |
1700 data_size_tmp = 1024 * avccontext->channels * sizeof(int16_t); | |
1701 if(*data_size < data_size_tmp) { | |
1702 av_log(avccontext, AV_LOG_ERROR, | |
1703 "Output buffer too small (%d) or trying to output too many samples (%d) for this frame.\n", | |
1704 *data_size, data_size_tmp); | |
1705 return -1; | |
1706 } | |
1707 *data_size = data_size_tmp; | |
1708 | |
1709 ac->dsp.float_to_int16_interleave(data, (const float **)ac->output_data, 1024, avccontext->channels); | |
1710 | |
1711 return buf_size; | |
1712 } | |
1713 | |
7501 | 1714 static av_cold int aac_decode_close(AVCodecContext * avccontext) { |
1715 AACContext * ac = avccontext->priv_data; | |
7540 | 1716 int i, type; |
7501 | 1717 |
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1718 for (i = 0; i < MAX_ELEM_ID; i++) { |
7540 | 1719 for(type = 0; type < 4; type++) |
1720 av_freep(&ac->che[type][i]); | |
7501 | 1721 } |
1722 | |
1723 ff_mdct_end(&ac->mdct); | |
1724 ff_mdct_end(&ac->mdct_small); | |
1725 return 0 ; | |
1726 } | |
1727 | |
1728 AVCodec aac_decoder = { | |
1729 "aac", | |
1730 CODEC_TYPE_AUDIO, | |
1731 CODEC_ID_AAC, | |
1732 sizeof(AACContext), | |
1733 aac_decode_init, | |
1734 NULL, | |
1735 aac_decode_close, | |
1736 aac_decode_frame, | |
1737 .long_name = NULL_IF_CONFIG_SMALL("Advanced Audio Coding"), | |
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1738 .sample_fmts = (enum SampleFormat[]){SAMPLE_FMT_S16,SAMPLE_FMT_NONE}, |
7501 | 1739 }; |