Mercurial > libavcodec.hg
annotate aac.c @ 8105:b122cb798801 libavcodec
Fix illegal identifier, __ are reserved for the system.
author | diego |
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date | Mon, 03 Nov 2008 21:59:35 +0000 |
parents | e70975d5ff80 |
children | 7baa25290231 |
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 /** | |
24 * @file aac.c | |
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 | |
44 * N frequency domain prediction | |
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" | |
80 #include "bitstream.h" | |
81 #include "dsputil.h" | |
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82 #include "lpc.h" |
7501 | 83 |
84 #include "aac.h" | |
85 #include "aactab.h" | |
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86 #include "aacdectab.h" |
7501 | 87 #include "mpeg4audio.h" |
88 | |
89 #include <assert.h> | |
90 #include <errno.h> | |
91 #include <math.h> | |
92 #include <string.h> | |
93 | |
94 static VLC vlc_scalefactors; | |
95 static VLC vlc_spectral[11]; | |
96 | |
97 | |
7539 | 98 /** |
7578 | 99 * Configure output channel order based on the current program configuration element. |
100 * | |
101 * @param che_pos current channel position configuration | |
102 * @param new_che_pos New channel position configuration - we only do something if it differs from the current one. | |
103 * | |
104 * @return Returns error status. 0 - OK, !0 - error | |
105 */ | |
106 static int output_configure(AACContext *ac, enum ChannelPosition che_pos[4][MAX_ELEM_ID], | |
107 enum ChannelPosition new_che_pos[4][MAX_ELEM_ID]) { | |
108 AVCodecContext *avctx = ac->avccontext; | |
109 int i, type, channels = 0; | |
110 | |
111 if(!memcmp(che_pos, new_che_pos, 4 * MAX_ELEM_ID * sizeof(new_che_pos[0][0]))) | |
112 return 0; /* no change */ | |
113 | |
114 memcpy(che_pos, new_che_pos, 4 * MAX_ELEM_ID * sizeof(new_che_pos[0][0])); | |
115 | |
116 /* Allocate or free elements depending on if they are in the | |
117 * current program configuration. | |
118 * | |
119 * Set up default 1:1 output mapping. | |
120 * | |
121 * For a 5.1 stream the output order will be: | |
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122 * [ Center ] [ Front Left ] [ Front Right ] [ LFE ] [ Surround Left ] [ Surround Right ] |
7578 | 123 */ |
124 | |
125 for(i = 0; i < MAX_ELEM_ID; i++) { | |
126 for(type = 0; type < 4; type++) { | |
127 if(che_pos[type][i]) { | |
128 if(!ac->che[type][i] && !(ac->che[type][i] = av_mallocz(sizeof(ChannelElement)))) | |
129 return AVERROR(ENOMEM); | |
130 if(type != TYPE_CCE) { | |
131 ac->output_data[channels++] = ac->che[type][i]->ch[0].ret; | |
132 if(type == TYPE_CPE) { | |
133 ac->output_data[channels++] = ac->che[type][i]->ch[1].ret; | |
134 } | |
135 } | |
136 } else | |
137 av_freep(&ac->che[type][i]); | |
138 } | |
139 } | |
140 | |
141 avctx->channels = channels; | |
142 return 0; | |
143 } | |
144 | |
145 /** | |
7539 | 146 * Decode an array of 4 bit element IDs, optionally interleaved with a stereo/mono switching bit. |
147 * | |
148 * @param cpe_map Stereo (Channel Pair Element) map, NULL if stereo bit is not present. | |
149 * @param sce_map mono (Single Channel Element) map | |
150 * @param type speaker type/position for these channels | |
151 */ | |
152 static void decode_channel_map(enum ChannelPosition *cpe_map, | |
153 enum ChannelPosition *sce_map, enum ChannelPosition type, GetBitContext * gb, int n) { | |
154 while(n--) { | |
155 enum ChannelPosition *map = cpe_map && get_bits1(gb) ? cpe_map : sce_map; // stereo or mono map | |
156 map[get_bits(gb, 4)] = type; | |
157 } | |
158 } | |
159 | |
160 /** | |
161 * Decode program configuration element; reference: table 4.2. | |
162 * | |
163 * @param new_che_pos New channel position configuration - we only do something if it differs from the current one. | |
164 * | |
165 * @return Returns error status. 0 - OK, !0 - error | |
166 */ | |
167 static int decode_pce(AACContext * ac, enum ChannelPosition new_che_pos[4][MAX_ELEM_ID], | |
168 GetBitContext * gb) { | |
169 int num_front, num_side, num_back, num_lfe, num_assoc_data, num_cc; | |
170 | |
171 skip_bits(gb, 2); // object_type | |
172 | |
173 ac->m4ac.sampling_index = get_bits(gb, 4); | |
174 if(ac->m4ac.sampling_index > 11) { | |
175 av_log(ac->avccontext, AV_LOG_ERROR, "invalid sampling rate index %d\n", ac->m4ac.sampling_index); | |
176 return -1; | |
177 } | |
178 ac->m4ac.sample_rate = ff_mpeg4audio_sample_rates[ac->m4ac.sampling_index]; | |
7501 | 179 num_front = get_bits(gb, 4); |
180 num_side = get_bits(gb, 4); | |
181 num_back = get_bits(gb, 4); | |
182 num_lfe = get_bits(gb, 2); | |
183 num_assoc_data = get_bits(gb, 3); | |
184 num_cc = get_bits(gb, 4); | |
185 | |
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186 if (get_bits1(gb)) |
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187 skip_bits(gb, 4); // mono_mixdown_tag |
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188 if (get_bits1(gb)) |
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189 skip_bits(gb, 4); // stereo_mixdown_tag |
7501 | 190 |
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191 if (get_bits1(gb)) |
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192 skip_bits(gb, 3); // mixdown_coeff_index and pseudo_surround |
7501 | 193 |
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194 decode_channel_map(new_che_pos[TYPE_CPE], new_che_pos[TYPE_SCE], AAC_CHANNEL_FRONT, gb, num_front); |
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195 decode_channel_map(new_che_pos[TYPE_CPE], new_che_pos[TYPE_SCE], AAC_CHANNEL_SIDE, gb, num_side ); |
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196 decode_channel_map(new_che_pos[TYPE_CPE], new_che_pos[TYPE_SCE], AAC_CHANNEL_BACK, gb, num_back ); |
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197 decode_channel_map(NULL, new_che_pos[TYPE_LFE], AAC_CHANNEL_LFE, gb, num_lfe ); |
7501 | 198 |
199 skip_bits_long(gb, 4 * num_assoc_data); | |
200 | |
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201 decode_channel_map(new_che_pos[TYPE_CCE], new_che_pos[TYPE_CCE], AAC_CHANNEL_CC, gb, num_cc ); |
7501 | 202 |
203 align_get_bits(gb); | |
204 | |
205 /* comment field, first byte is length */ | |
206 skip_bits_long(gb, 8 * get_bits(gb, 8)); | |
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207 return 0; |
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208 } |
7501 | 209 |
7539 | 210 /** |
211 * Set up channel positions based on a default channel configuration | |
212 * as specified in table 1.17. | |
213 * | |
214 * @param new_che_pos New channel position configuration - we only do something if it differs from the current one. | |
215 * | |
216 * @return Returns error status. 0 - OK, !0 - error | |
217 */ | |
218 static int set_default_channel_config(AACContext *ac, enum ChannelPosition new_che_pos[4][MAX_ELEM_ID], | |
219 int channel_config) | |
220 { | |
221 if(channel_config < 1 || channel_config > 7) { | |
222 av_log(ac->avccontext, AV_LOG_ERROR, "invalid default channel configuration (%d)\n", | |
223 channel_config); | |
224 return -1; | |
225 } | |
226 | |
227 /* default channel configurations: | |
228 * | |
229 * 1ch : front center (mono) | |
230 * 2ch : L + R (stereo) | |
231 * 3ch : front center + L + R | |
232 * 4ch : front center + L + R + back center | |
233 * 5ch : front center + L + R + back stereo | |
234 * 6ch : front center + L + R + back stereo + LFE | |
235 * 7ch : front center + L + R + outer front left + outer front right + back stereo + LFE | |
236 */ | |
237 | |
238 if(channel_config != 2) | |
239 new_che_pos[TYPE_SCE][0] = AAC_CHANNEL_FRONT; // front center (or mono) | |
240 if(channel_config > 1) | |
241 new_che_pos[TYPE_CPE][0] = AAC_CHANNEL_FRONT; // L + R (or stereo) | |
242 if(channel_config == 4) | |
243 new_che_pos[TYPE_SCE][1] = AAC_CHANNEL_BACK; // back center | |
244 if(channel_config > 4) | |
245 new_che_pos[TYPE_CPE][(channel_config == 7) + 1] | |
246 = AAC_CHANNEL_BACK; // back stereo | |
247 if(channel_config > 5) | |
248 new_che_pos[TYPE_LFE][0] = AAC_CHANNEL_LFE; // LFE | |
249 if(channel_config == 7) | |
250 new_che_pos[TYPE_CPE][1] = AAC_CHANNEL_FRONT; // outer front left + outer front right | |
251 | |
252 return 0; | |
253 } | |
254 | |
7578 | 255 /** |
256 * Decode GA "General Audio" specific configuration; reference: table 4.1. | |
257 * | |
258 * @return Returns error status. 0 - OK, !0 - error | |
259 */ | |
260 static int decode_ga_specific_config(AACContext * ac, GetBitContext * gb, int channel_config) { | |
261 enum ChannelPosition new_che_pos[4][MAX_ELEM_ID]; | |
262 int extension_flag, ret; | |
263 | |
264 if(get_bits1(gb)) { // frameLengthFlag | |
265 av_log_missing_feature(ac->avccontext, "960/120 MDCT window is", 1); | |
7539 | 266 return -1; |
267 } | |
268 | |
269 if (get_bits1(gb)) // dependsOnCoreCoder | |
270 skip_bits(gb, 14); // coreCoderDelay | |
271 extension_flag = get_bits1(gb); | |
272 | |
273 if(ac->m4ac.object_type == AOT_AAC_SCALABLE || | |
274 ac->m4ac.object_type == AOT_ER_AAC_SCALABLE) | |
275 skip_bits(gb, 3); // layerNr | |
276 | |
277 memset(new_che_pos, 0, 4 * MAX_ELEM_ID * sizeof(new_che_pos[0][0])); | |
278 if (channel_config == 0) { | |
279 skip_bits(gb, 4); // element_instance_tag | |
280 if((ret = decode_pce(ac, new_che_pos, gb))) | |
281 return ret; | |
282 } else { | |
283 if((ret = set_default_channel_config(ac, new_che_pos, channel_config))) | |
284 return ret; | |
285 } | |
286 if((ret = output_configure(ac, ac->che_pos, new_che_pos))) | |
287 return ret; | |
288 | |
289 if (extension_flag) { | |
290 switch (ac->m4ac.object_type) { | |
291 case AOT_ER_BSAC: | |
292 skip_bits(gb, 5); // numOfSubFrame | |
293 skip_bits(gb, 11); // layer_length | |
294 break; | |
295 case AOT_ER_AAC_LC: | |
296 case AOT_ER_AAC_LTP: | |
297 case AOT_ER_AAC_SCALABLE: | |
298 case AOT_ER_AAC_LD: | |
299 skip_bits(gb, 3); /* aacSectionDataResilienceFlag | |
300 * aacScalefactorDataResilienceFlag | |
301 * aacSpectralDataResilienceFlag | |
302 */ | |
303 break; | |
304 } | |
305 skip_bits1(gb); // extensionFlag3 (TBD in version 3) | |
306 } | |
307 return 0; | |
308 } | |
309 | |
310 /** | |
311 * Decode audio specific configuration; reference: table 1.13. | |
312 * | |
313 * @param data pointer to AVCodecContext extradata | |
314 * @param data_size size of AVCCodecContext extradata | |
315 * | |
316 * @return Returns error status. 0 - OK, !0 - error | |
317 */ | |
318 static int decode_audio_specific_config(AACContext * ac, void *data, int data_size) { | |
319 GetBitContext gb; | |
320 int i; | |
321 | |
322 init_get_bits(&gb, data, data_size * 8); | |
323 | |
324 if((i = ff_mpeg4audio_get_config(&ac->m4ac, data, data_size)) < 0) | |
325 return -1; | |
326 if(ac->m4ac.sampling_index > 11) { | |
327 av_log(ac->avccontext, AV_LOG_ERROR, "invalid sampling rate index %d\n", ac->m4ac.sampling_index); | |
328 return -1; | |
329 } | |
330 | |
331 skip_bits_long(&gb, i); | |
332 | |
333 switch (ac->m4ac.object_type) { | |
334 case AOT_AAC_LC: | |
335 if (decode_ga_specific_config(ac, &gb, ac->m4ac.chan_config)) | |
336 return -1; | |
337 break; | |
338 default: | |
339 av_log(ac->avccontext, AV_LOG_ERROR, "Audio object type %s%d is not supported.\n", | |
340 ac->m4ac.sbr == 1? "SBR+" : "", ac->m4ac.object_type); | |
341 return -1; | |
342 } | |
343 return 0; | |
344 } | |
345 | |
7578 | 346 /** |
347 * linear congruential pseudorandom number generator | |
348 * | |
349 * @param previous_val pointer to the current state of the generator | |
350 * | |
351 * @return Returns a 32-bit pseudorandom integer | |
352 */ | |
353 static av_always_inline int lcg_random(int previous_val) { | |
354 return previous_val * 1664525 + 1013904223; | |
355 } | |
356 | |
7501 | 357 static av_cold int aac_decode_init(AVCodecContext * avccontext) { |
358 AACContext * ac = avccontext->priv_data; | |
359 int i; | |
360 | |
361 ac->avccontext = avccontext; | |
362 | |
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363 if (avccontext->extradata_size <= 0 || |
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364 decode_audio_specific_config(ac, avccontext->extradata, avccontext->extradata_size)) |
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365 return -1; |
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366 |
7539 | 367 avccontext->sample_fmt = SAMPLE_FMT_S16; |
7501 | 368 avccontext->sample_rate = ac->m4ac.sample_rate; |
369 avccontext->frame_size = 1024; | |
370 | |
371 AAC_INIT_VLC_STATIC( 0, 144); | |
372 AAC_INIT_VLC_STATIC( 1, 114); | |
373 AAC_INIT_VLC_STATIC( 2, 188); | |
374 AAC_INIT_VLC_STATIC( 3, 180); | |
375 AAC_INIT_VLC_STATIC( 4, 172); | |
376 AAC_INIT_VLC_STATIC( 5, 140); | |
377 AAC_INIT_VLC_STATIC( 6, 168); | |
378 AAC_INIT_VLC_STATIC( 7, 114); | |
379 AAC_INIT_VLC_STATIC( 8, 262); | |
380 AAC_INIT_VLC_STATIC( 9, 248); | |
381 AAC_INIT_VLC_STATIC(10, 384); | |
382 | |
383 dsputil_init(&ac->dsp, avccontext); | |
384 | |
7539 | 385 ac->random_state = 0x1f2e3d4c; |
386 | |
7501 | 387 // -1024 - Compensate wrong IMDCT method. |
388 // 32768 - Required to scale values to the correct range for the bias method | |
389 // for float to int16 conversion. | |
390 | |
391 if(ac->dsp.float_to_int16 == ff_float_to_int16_c) { | |
392 ac->add_bias = 385.0f; | |
393 ac->sf_scale = 1. / (-1024. * 32768.); | |
394 ac->sf_offset = 0; | |
395 } else { | |
396 ac->add_bias = 0.0f; | |
397 ac->sf_scale = 1. / -1024.; | |
398 ac->sf_offset = 60; | |
399 } | |
400 | |
401 #ifndef CONFIG_HARDCODED_TABLES | |
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402 for (i = 0; i < 316; i++) |
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403 ff_aac_pow2sf_tab[i] = pow(2, (i - 200)/4.); |
7501 | 404 #endif /* CONFIG_HARDCODED_TABLES */ |
405 | |
8042 | 406 INIT_VLC_STATIC(&vlc_scalefactors,7,FF_ARRAY_ELEMS(ff_aac_scalefactor_code), |
7501 | 407 ff_aac_scalefactor_bits, sizeof(ff_aac_scalefactor_bits[0]), sizeof(ff_aac_scalefactor_bits[0]), |
408 ff_aac_scalefactor_code, sizeof(ff_aac_scalefactor_code[0]), sizeof(ff_aac_scalefactor_code[0]), | |
409 352); | |
410 | |
411 ff_mdct_init(&ac->mdct, 11, 1); | |
412 ff_mdct_init(&ac->mdct_small, 8, 1); | |
7581 | 413 // window initialization |
414 ff_kbd_window_init(ff_aac_kbd_long_1024, 4.0, 1024); | |
415 ff_kbd_window_init(ff_aac_kbd_short_128, 6.0, 128); | |
416 ff_sine_window_init(ff_sine_1024, 1024); | |
417 ff_sine_window_init(ff_sine_128, 128); | |
418 | |
7501 | 419 return 0; |
420 } | |
421 | |
7539 | 422 /** |
423 * Skip data_stream_element; reference: table 4.10. | |
424 */ | |
425 static void skip_data_stream_element(GetBitContext * gb) { | |
7501 | 426 int byte_align = get_bits1(gb); |
427 int count = get_bits(gb, 8); | |
428 if (count == 255) | |
429 count += get_bits(gb, 8); | |
430 if (byte_align) | |
431 align_get_bits(gb); | |
432 skip_bits_long(gb, 8 * count); | |
433 } | |
434 | |
435 /** | |
7539 | 436 * Decode Individual Channel Stream info; reference: table 4.6. |
437 * | |
438 * @param common_window Channels have independent [0], or shared [1], Individual Channel Stream information. | |
439 */ | |
440 static int decode_ics_info(AACContext * ac, IndividualChannelStream * ics, GetBitContext * gb, int common_window) { | |
441 if (get_bits1(gb)) { | |
442 av_log(ac->avccontext, AV_LOG_ERROR, "Reserved bit set.\n"); | |
443 memset(ics, 0, sizeof(IndividualChannelStream)); | |
444 return -1; | |
445 } | |
446 ics->window_sequence[1] = ics->window_sequence[0]; | |
447 ics->window_sequence[0] = get_bits(gb, 2); | |
448 ics->use_kb_window[1] = ics->use_kb_window[0]; | |
449 ics->use_kb_window[0] = get_bits1(gb); | |
450 ics->num_window_groups = 1; | |
451 ics->group_len[0] = 1; | |
7581 | 452 if (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) { |
453 int i; | |
454 ics->max_sfb = get_bits(gb, 4); | |
455 for (i = 0; i < 7; i++) { | |
456 if (get_bits1(gb)) { | |
457 ics->group_len[ics->num_window_groups-1]++; | |
458 } else { | |
459 ics->num_window_groups++; | |
460 ics->group_len[ics->num_window_groups-1] = 1; | |
461 } | |
462 } | |
463 ics->num_windows = 8; | |
464 ics->swb_offset = swb_offset_128[ac->m4ac.sampling_index]; | |
465 ics->num_swb = ff_aac_num_swb_128[ac->m4ac.sampling_index]; | |
466 ics->tns_max_bands = tns_max_bands_128[ac->m4ac.sampling_index]; | |
467 } else { | |
468 ics->max_sfb = get_bits(gb, 6); | |
469 ics->num_windows = 1; | |
470 ics->swb_offset = swb_offset_1024[ac->m4ac.sampling_index]; | |
471 ics->num_swb = ff_aac_num_swb_1024[ac->m4ac.sampling_index]; | |
472 ics->tns_max_bands = tns_max_bands_1024[ac->m4ac.sampling_index]; | |
7578 | 473 if (get_bits1(gb)) { |
474 av_log_missing_feature(ac->avccontext, "Predictor bit set but LTP is", 1); | |
475 memset(ics, 0, sizeof(IndividualChannelStream)); | |
476 return -1; | |
477 } | |
478 } | |
479 | |
480 if(ics->max_sfb > ics->num_swb) { | |
481 av_log(ac->avccontext, AV_LOG_ERROR, | |
482 "Number of scalefactor bands in group (%d) exceeds limit (%d).\n", | |
483 ics->max_sfb, ics->num_swb); | |
484 memset(ics, 0, sizeof(IndividualChannelStream)); | |
485 return -1; | |
486 } | |
487 | |
7539 | 488 return 0; |
489 } | |
490 | |
491 /** | |
492 * Decode band types (section_data payload); reference: table 4.46. | |
493 * | |
494 * @param band_type array of the used band type | |
495 * @param band_type_run_end array of the last scalefactor band of a band type run | |
496 * | |
497 * @return Returns error status. 0 - OK, !0 - error | |
498 */ | |
499 static int decode_band_types(AACContext * ac, enum BandType band_type[120], | |
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500 int band_type_run_end[120], GetBitContext * gb, IndividualChannelStream * ics) { |
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501 int g, idx = 0; |
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502 const int bits = (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) ? 3 : 5; |
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503 for (g = 0; g < ics->num_window_groups; g++) { |
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504 int k = 0; |
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505 while (k < ics->max_sfb) { |
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506 uint8_t sect_len = k; |
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507 int sect_len_incr; |
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508 int sect_band_type = get_bits(gb, 4); |
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509 if (sect_band_type == 12) { |
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510 av_log(ac->avccontext, AV_LOG_ERROR, "invalid band type\n"); |
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511 return -1; |
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512 } |
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513 while ((sect_len_incr = get_bits(gb, bits)) == (1 << bits)-1) |
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514 sect_len += sect_len_incr; |
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515 sect_len += sect_len_incr; |
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516 if (sect_len > ics->max_sfb) { |
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517 av_log(ac->avccontext, AV_LOG_ERROR, |
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518 "Number of bands (%d) exceeds limit (%d).\n", |
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519 sect_len, ics->max_sfb); |
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520 return -1; |
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521 } |
7581 | 522 for (; k < sect_len; k++) { |
523 band_type [idx] = sect_band_type; | |
524 band_type_run_end[idx++] = sect_len; | |
525 } | |
7539 | 526 } |
527 } | |
528 return 0; | |
529 } | |
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530 |
7539 | 531 /** |
532 * Decode scalefactors; reference: table 4.47. | |
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533 * |
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534 * @param global_gain first scalefactor value as scalefactors are differentially coded |
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535 * @param band_type array of the used band type |
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536 * @param band_type_run_end array of the last scalefactor band of a band type run |
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537 * @param sf array of scalefactors or intensity stereo positions |
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538 * |
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539 * @return Returns error status. 0 - OK, !0 - error |
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540 */ |
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541 static int decode_scalefactors(AACContext * ac, float sf[120], GetBitContext * gb, |
7540 | 542 unsigned int global_gain, IndividualChannelStream * ics, |
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543 enum BandType band_type[120], int band_type_run_end[120]) { |
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544 const int sf_offset = ac->sf_offset + (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE ? 12 : 0); |
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545 int g, i, idx = 0; |
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546 int offset[3] = { global_gain, global_gain - 90, 100 }; |
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547 int noise_flag = 1; |
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548 static const char *sf_str[3] = { "Global gain", "Noise gain", "Intensity stereo position" }; |
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549 for (g = 0; g < ics->num_window_groups; g++) { |
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550 for (i = 0; i < ics->max_sfb;) { |
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551 int run_end = band_type_run_end[idx]; |
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552 if (band_type[idx] == ZERO_BT) { |
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553 for(; i < run_end; i++, idx++) |
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554 sf[idx] = 0.; |
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555 }else if((band_type[idx] == INTENSITY_BT) || (band_type[idx] == INTENSITY_BT2)) { |
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556 for(; i < run_end; i++, idx++) { |
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557 offset[2] += get_vlc2(gb, vlc_scalefactors.table, 7, 3) - 60; |
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558 if(offset[2] > 255U) { |
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559 av_log(ac->avccontext, AV_LOG_ERROR, |
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560 "%s (%d) out of range.\n", sf_str[2], offset[2]); |
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561 return -1; |
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562 } |
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563 sf[idx] = ff_aac_pow2sf_tab[-offset[2] + 300]; |
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564 } |
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565 }else if(band_type[idx] == NOISE_BT) { |
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566 for(; i < run_end; i++, idx++) { |
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567 if(noise_flag-- > 0) |
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568 offset[1] += get_bits(gb, 9) - 256; |
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569 else |
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570 offset[1] += get_vlc2(gb, vlc_scalefactors.table, 7, 3) - 60; |
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571 if(offset[1] > 255U) { |
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572 av_log(ac->avccontext, AV_LOG_ERROR, |
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573 "%s (%d) out of range.\n", sf_str[1], offset[1]); |
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574 return -1; |
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575 } |
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576 sf[idx] = -ff_aac_pow2sf_tab[ offset[1] + sf_offset]; |
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577 } |
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578 }else { |
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579 for(; i < run_end; i++, idx++) { |
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580 offset[0] += get_vlc2(gb, vlc_scalefactors.table, 7, 3) - 60; |
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581 if(offset[0] > 255U) { |
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582 av_log(ac->avccontext, AV_LOG_ERROR, |
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583 "%s (%d) out of range.\n", sf_str[0], offset[0]); |
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584 return -1; |
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585 } |
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586 sf[idx] = -ff_aac_pow2sf_tab[ offset[0] + sf_offset]; |
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587 } |
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588 } |
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589 } |
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590 } |
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591 return 0; |
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592 } |
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593 |
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594 /** |
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595 * Decode pulse data; reference: table 4.7. |
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596 */ |
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597 static int decode_pulses(Pulse * pulse, GetBitContext * gb, const uint16_t * swb_offset, int num_swb) { |
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598 int i, pulse_swb; |
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599 pulse->num_pulse = get_bits(gb, 2) + 1; |
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600 pulse_swb = get_bits(gb, 6); |
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601 if (pulse_swb >= num_swb) |
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602 return -1; |
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603 pulse->pos[0] = swb_offset[pulse_swb]; |
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604 pulse->pos[0] += get_bits(gb, 5); |
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605 if (pulse->pos[0] > 1023) |
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606 return -1; |
7580 | 607 pulse->amp[0] = get_bits(gb, 4); |
608 for (i = 1; i < pulse->num_pulse; i++) { | |
609 pulse->pos[i] = get_bits(gb, 5) + pulse->pos[i-1]; | |
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610 if (pulse->pos[i] > 1023) |
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611 return -1; |
7580 | 612 pulse->amp[i] = get_bits(gb, 4); |
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613 } |
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614 return 0; |
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615 } |
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616 |
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617 /** |
7609 | 618 * Decode Temporal Noise Shaping data; reference: table 4.48. |
619 * | |
620 * @return Returns error status. 0 - OK, !0 - error | |
621 */ | |
622 static int decode_tns(AACContext * ac, TemporalNoiseShaping * tns, | |
623 GetBitContext * gb, const IndividualChannelStream * ics) { | |
624 int w, filt, i, coef_len, coef_res, coef_compress; | |
625 const int is8 = ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE; | |
626 const int tns_max_order = is8 ? 7 : ac->m4ac.object_type == AOT_AAC_MAIN ? 20 : 12; | |
627 for (w = 0; w < ics->num_windows; w++) { | |
7637
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628 if ((tns->n_filt[w] = get_bits(gb, 2 - is8))) { |
7609 | 629 coef_res = get_bits1(gb); |
630 | |
7638 | 631 for (filt = 0; filt < tns->n_filt[w]; filt++) { |
632 int tmp2_idx; | |
633 tns->length[w][filt] = get_bits(gb, 6 - 2*is8); | |
7609 | 634 |
7638 | 635 if ((tns->order[w][filt] = get_bits(gb, 5 - 2*is8)) > tns_max_order) { |
636 av_log(ac->avccontext, AV_LOG_ERROR, "TNS filter order %d is greater than maximum %d.", | |
637 tns->order[w][filt], tns_max_order); | |
638 tns->order[w][filt] = 0; | |
639 return -1; | |
640 } | |
7850
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Only read some of TNS bitstream data in the case that the TNS filter order is
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diff
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|
641 if (tns->order[w][filt]) { |
7851 | 642 tns->direction[w][filt] = get_bits1(gb); |
643 coef_compress = get_bits1(gb); | |
644 coef_len = coef_res + 3 - coef_compress; | |
645 tmp2_idx = 2*coef_compress + coef_res; | |
7638 | 646 |
7851 | 647 for (i = 0; i < tns->order[w][filt]; i++) |
648 tns->coef[w][filt][i] = tns_tmp2_map[tmp2_idx][get_bits(gb, coef_len)]; | |
7850
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|
649 } |
7609 | 650 } |
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|
651 } |
7609 | 652 } |
653 return 0; | |
654 } | |
655 | |
656 /** | |
7539 | 657 * Decode Mid/Side data; reference: table 4.54. |
658 * | |
659 * @param ms_present Indicates mid/side stereo presence. [0] mask is all 0s; | |
660 * [1] mask is decoded from bitstream; [2] mask is all 1s; | |
661 * [3] reserved for scalable AAC | |
662 */ | |
663 static void decode_mid_side_stereo(ChannelElement * cpe, GetBitContext * gb, | |
664 int ms_present) { | |
7578 | 665 int idx; |
666 if (ms_present == 1) { | |
667 for (idx = 0; idx < cpe->ch[0].ics.num_window_groups * cpe->ch[0].ics.max_sfb; idx++) | |
668 cpe->ms_mask[idx] = get_bits1(gb); | |
669 } else if (ms_present == 2) { | |
670 memset(cpe->ms_mask, 1, cpe->ch[0].ics.num_window_groups * cpe->ch[0].ics.max_sfb * sizeof(cpe->ms_mask[0])); | |
671 } | |
672 } | |
7539 | 673 |
674 /** | |
7581 | 675 * Decode spectral data; reference: table 4.50. |
676 * Dequantize and scale spectral data; reference: 4.6.3.3. | |
677 * | |
678 * @param coef array of dequantized, scaled spectral data | |
679 * @param sf array of scalefactors or intensity stereo positions | |
680 * @param pulse_present set if pulses are present | |
681 * @param pulse pointer to pulse data struct | |
682 * @param band_type array of the used band type | |
683 * | |
684 * @return Returns error status. 0 - OK, !0 - error | |
685 */ | |
686 static int decode_spectrum_and_dequant(AACContext * ac, float coef[1024], GetBitContext * gb, float sf[120], | |
687 int pulse_present, const Pulse * pulse, const IndividualChannelStream * ics, enum BandType band_type[120]) { | |
688 int i, k, g, idx = 0; | |
689 const int c = 1024/ics->num_windows; | |
690 const uint16_t * offsets = ics->swb_offset; | |
691 float *coef_base = coef; | |
692 | |
693 for (g = 0; g < ics->num_windows; g++) | |
694 memset(coef + g * 128 + offsets[ics->max_sfb], 0, sizeof(float)*(c - offsets[ics->max_sfb])); | |
695 | |
696 for (g = 0; g < ics->num_window_groups; g++) { | |
697 for (i = 0; i < ics->max_sfb; i++, idx++) { | |
698 const int cur_band_type = band_type[idx]; | |
699 const int dim = cur_band_type >= FIRST_PAIR_BT ? 2 : 4; | |
700 const int is_cb_unsigned = IS_CODEBOOK_UNSIGNED(cur_band_type); | |
701 int group; | |
702 if (cur_band_type == ZERO_BT) { | |
703 for (group = 0; group < ics->group_len[g]; group++) { | |
704 memset(coef + group * 128 + offsets[i], 0, (offsets[i+1] - offsets[i])*sizeof(float)); | |
705 } | |
706 }else if (cur_band_type == NOISE_BT) { | |
707 const float scale = sf[idx] / ((offsets[i+1] - offsets[i]) * PNS_MEAN_ENERGY); | |
708 for (group = 0; group < ics->group_len[g]; group++) { | |
709 for (k = offsets[i]; k < offsets[i+1]; k++) { | |
710 ac->random_state = lcg_random(ac->random_state); | |
711 coef[group*128+k] = ac->random_state * scale; | |
712 } | |
713 } | |
714 }else if (cur_band_type != INTENSITY_BT2 && cur_band_type != INTENSITY_BT) { | |
715 for (group = 0; group < ics->group_len[g]; group++) { | |
716 for (k = offsets[i]; k < offsets[i+1]; k += dim) { | |
717 const int index = get_vlc2(gb, vlc_spectral[cur_band_type - 1].table, 6, 3); | |
718 const int coef_tmp_idx = (group << 7) + k; | |
719 const float *vq_ptr; | |
720 int j; | |
721 if(index >= ff_aac_spectral_sizes[cur_band_type - 1]) { | |
722 av_log(ac->avccontext, AV_LOG_ERROR, | |
723 "Read beyond end of ff_aac_codebook_vectors[%d][]. index %d >= %d\n", | |
724 cur_band_type - 1, index, ff_aac_spectral_sizes[cur_band_type - 1]); | |
725 return -1; | |
726 } | |
727 vq_ptr = &ff_aac_codebook_vectors[cur_band_type - 1][index * dim]; | |
728 if (is_cb_unsigned) { | |
729 for (j = 0; j < dim; j++) | |
730 if (vq_ptr[j]) | |
731 coef[coef_tmp_idx + j] = 1 - 2*(int)get_bits1(gb); | |
732 }else { | |
733 for (j = 0; j < dim; j++) | |
734 coef[coef_tmp_idx + j] = 1.0f; | |
735 } | |
736 if (cur_band_type == ESC_BT) { | |
737 for (j = 0; j < 2; j++) { | |
738 if (vq_ptr[j] == 64.0f) { | |
739 int n = 4; | |
740 /* The total length of escape_sequence must be < 22 bits according | |
741 to the specification (i.e. max is 11111111110xxxxxxxxxx). */ | |
742 while (get_bits1(gb) && n < 15) n++; | |
743 if(n == 15) { | |
744 av_log(ac->avccontext, AV_LOG_ERROR, "error in spectral data, ESC overflow\n"); | |
745 return -1; | |
746 } | |
747 n = (1<<n) + get_bits(gb, n); | |
748 coef[coef_tmp_idx + j] *= cbrtf(fabsf(n)) * n; | |
749 }else | |
750 coef[coef_tmp_idx + j] *= vq_ptr[j]; | |
751 } | |
752 }else | |
753 for (j = 0; j < dim; j++) | |
754 coef[coef_tmp_idx + j] *= vq_ptr[j]; | |
755 for (j = 0; j < dim; j++) | |
756 coef[coef_tmp_idx + j] *= sf[idx]; | |
757 } | |
758 } | |
759 } | |
760 } | |
761 coef += ics->group_len[g]<<7; | |
762 } | |
763 | |
764 if (pulse_present) { | |
7857
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765 idx = 0; |
7581 | 766 for(i = 0; i < pulse->num_pulse; i++){ |
767 float co = coef_base[ pulse->pos[i] ]; | |
7857
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768 while(offsets[idx + 1] <= pulse->pos[i]) |
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|
769 idx++; |
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|
770 if (band_type[idx] != NOISE_BT && sf[idx]) { |
7858 | 771 float ico = -pulse->amp[i]; |
772 if (co) { | |
773 co /= sf[idx]; | |
774 ico = co / sqrtf(sqrtf(fabsf(co))) + (co > 0 ? -ico : ico); | |
775 } | |
776 coef_base[ pulse->pos[i] ] = cbrtf(fabsf(ico)) * ico * sf[idx]; | |
7857
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|
777 } |
7581 | 778 } |
779 } | |
780 return 0; | |
781 } | |
782 | |
783 /** | |
7539 | 784 * Decode an individual_channel_stream payload; reference: table 4.44. |
785 * | |
786 * @param common_window Channels have independent [0], or shared [1], Individual Channel Stream information. | |
787 * @param scale_flag scalable [1] or non-scalable [0] AAC (Unused until scalable AAC is implemented.) | |
788 * | |
789 * @return Returns error status. 0 - OK, !0 - error | |
790 */ | |
791 static int decode_ics(AACContext * ac, SingleChannelElement * sce, GetBitContext * gb, int common_window, int scale_flag) { | |
792 Pulse pulse; | |
793 TemporalNoiseShaping * tns = &sce->tns; | |
794 IndividualChannelStream * ics = &sce->ics; | |
795 float * out = sce->coeffs; | |
796 int global_gain, pulse_present = 0; | |
797 | |
7580 | 798 /* This assignment is to silence a GCC warning about the variable being used |
799 * uninitialized when in fact it always is. | |
7539 | 800 */ |
801 pulse.num_pulse = 0; | |
802 | |
803 global_gain = get_bits(gb, 8); | |
804 | |
805 if (!common_window && !scale_flag) { | |
806 if (decode_ics_info(ac, ics, gb, 0) < 0) | |
807 return -1; | |
808 } | |
809 | |
810 if (decode_band_types(ac, sce->band_type, sce->band_type_run_end, gb, ics) < 0) | |
811 return -1; | |
812 if (decode_scalefactors(ac, sce->sf, gb, global_gain, ics, sce->band_type, sce->band_type_run_end) < 0) | |
813 return -1; | |
814 | |
815 pulse_present = 0; | |
816 if (!scale_flag) { | |
817 if ((pulse_present = get_bits1(gb))) { | |
818 if (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) { | |
819 av_log(ac->avccontext, AV_LOG_ERROR, "Pulse tool not allowed in eight short sequence.\n"); | |
820 return -1; | |
821 } | |
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822 if (decode_pulses(&pulse, gb, ics->swb_offset, ics->num_swb)) { |
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823 av_log(ac->avccontext, AV_LOG_ERROR, "Pulse data corrupt or invalid.\n"); |
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Validate pulse position and error out if an invalid position is encountered.
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diff
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|
824 return -1; |
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Validate pulse position and error out if an invalid position is encountered.
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|
825 } |
7539 | 826 } |
827 if ((tns->present = get_bits1(gb)) && decode_tns(ac, tns, gb, ics)) | |
828 return -1; | |
829 if (get_bits1(gb)) { | |
830 av_log_missing_feature(ac->avccontext, "SSR", 1); | |
831 return -1; | |
832 } | |
833 } | |
834 | |
7580 | 835 if (decode_spectrum_and_dequant(ac, out, gb, sce->sf, pulse_present, &pulse, ics, sce->band_type) < 0) |
7539 | 836 return -1; |
837 return 0; | |
838 } | |
839 | |
840 /** | |
7581 | 841 * Mid/Side stereo decoding; reference: 4.6.8.1.3. |
842 */ | |
843 static void apply_mid_side_stereo(ChannelElement * cpe) { | |
844 const IndividualChannelStream * ics = &cpe->ch[0].ics; | |
845 float *ch0 = cpe->ch[0].coeffs; | |
846 float *ch1 = cpe->ch[1].coeffs; | |
847 int g, i, k, group, idx = 0; | |
848 const uint16_t * offsets = ics->swb_offset; | |
849 for (g = 0; g < ics->num_window_groups; g++) { | |
850 for (i = 0; i < ics->max_sfb; i++, idx++) { | |
851 if (cpe->ms_mask[idx] && | |
852 cpe->ch[0].band_type[idx] < NOISE_BT && cpe->ch[1].band_type[idx] < NOISE_BT) { | |
853 for (group = 0; group < ics->group_len[g]; group++) { | |
854 for (k = offsets[i]; k < offsets[i+1]; k++) { | |
855 float tmp = ch0[group*128 + k] - ch1[group*128 + k]; | |
856 ch0[group*128 + k] += ch1[group*128 + k]; | |
857 ch1[group*128 + k] = tmp; | |
858 } | |
859 } | |
860 } | |
861 } | |
862 ch0 += ics->group_len[g]*128; | |
863 ch1 += ics->group_len[g]*128; | |
864 } | |
865 } | |
866 | |
867 /** | |
868 * intensity stereo decoding; reference: 4.6.8.2.3 | |
869 * | |
870 * @param ms_present Indicates mid/side stereo presence. [0] mask is all 0s; | |
871 * [1] mask is decoded from bitstream; [2] mask is all 1s; | |
872 * [3] reserved for scalable AAC | |
873 */ | |
874 static void apply_intensity_stereo(ChannelElement * cpe, int ms_present) { | |
875 const IndividualChannelStream * ics = &cpe->ch[1].ics; | |
876 SingleChannelElement * sce1 = &cpe->ch[1]; | |
877 float *coef0 = cpe->ch[0].coeffs, *coef1 = cpe->ch[1].coeffs; | |
878 const uint16_t * offsets = ics->swb_offset; | |
879 int g, group, i, k, idx = 0; | |
880 int c; | |
881 float scale; | |
882 for (g = 0; g < ics->num_window_groups; g++) { | |
883 for (i = 0; i < ics->max_sfb;) { | |
884 if (sce1->band_type[idx] == INTENSITY_BT || sce1->band_type[idx] == INTENSITY_BT2) { | |
885 const int bt_run_end = sce1->band_type_run_end[idx]; | |
886 for (; i < bt_run_end; i++, idx++) { | |
887 c = -1 + 2 * (sce1->band_type[idx] - 14); | |
888 if (ms_present) | |
889 c *= 1 - 2 * cpe->ms_mask[idx]; | |
890 scale = c * sce1->sf[idx]; | |
891 for (group = 0; group < ics->group_len[g]; group++) | |
892 for (k = offsets[i]; k < offsets[i+1]; k++) | |
893 coef1[group*128 + k] = scale * coef0[group*128 + k]; | |
894 } | |
895 } else { | |
896 int bt_run_end = sce1->band_type_run_end[idx]; | |
897 idx += bt_run_end - i; | |
898 i = bt_run_end; | |
899 } | |
900 } | |
901 coef0 += ics->group_len[g]*128; | |
902 coef1 += ics->group_len[g]*128; | |
903 } | |
904 } | |
905 | |
906 /** | |
7539 | 907 * Decode a channel_pair_element; reference: table 4.4. |
908 * | |
909 * @param elem_id Identifies the instance of a syntax element. | |
910 * | |
911 * @return Returns error status. 0 - OK, !0 - error | |
912 */ | |
913 static int decode_cpe(AACContext * ac, GetBitContext * gb, int elem_id) { | |
914 int i, ret, common_window, ms_present = 0; | |
915 ChannelElement * cpe; | |
916 | |
917 cpe = ac->che[TYPE_CPE][elem_id]; | |
918 common_window = get_bits1(gb); | |
919 if (common_window) { | |
920 if (decode_ics_info(ac, &cpe->ch[0].ics, gb, 1)) | |
921 return -1; | |
922 i = cpe->ch[1].ics.use_kb_window[0]; | |
923 cpe->ch[1].ics = cpe->ch[0].ics; | |
924 cpe->ch[1].ics.use_kb_window[1] = i; | |
925 ms_present = get_bits(gb, 2); | |
926 if(ms_present == 3) { | |
927 av_log(ac->avccontext, AV_LOG_ERROR, "ms_present = 3 is reserved.\n"); | |
928 return -1; | |
929 } else if(ms_present) | |
930 decode_mid_side_stereo(cpe, gb, ms_present); | |
931 } | |
932 if ((ret = decode_ics(ac, &cpe->ch[0], gb, common_window, 0))) | |
933 return ret; | |
934 if ((ret = decode_ics(ac, &cpe->ch[1], gb, common_window, 0))) | |
935 return ret; | |
936 | |
937 if (common_window && ms_present) | |
938 apply_mid_side_stereo(cpe); | |
939 | |
7580 | 940 apply_intensity_stereo(cpe, ms_present); |
7539 | 941 return 0; |
942 } | |
943 | |
7581 | 944 /** |
945 * Decode coupling_channel_element; reference: table 4.8. | |
946 * | |
947 * @param elem_id Identifies the instance of a syntax element. | |
948 * | |
949 * @return Returns error status. 0 - OK, !0 - error | |
950 */ | |
951 static int decode_cce(AACContext * ac, GetBitContext * gb, ChannelElement * che) { | |
952 int num_gain = 0; | |
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953 int c, g, sfb, ret; |
7581 | 954 int sign; |
955 float scale; | |
956 SingleChannelElement * sce = &che->ch[0]; | |
957 ChannelCoupling * coup = &che->coup; | |
958 | |
7578 | 959 coup->coupling_point = 2*get_bits1(gb); |
960 coup->num_coupled = get_bits(gb, 3); | |
961 for (c = 0; c <= coup->num_coupled; c++) { | |
962 num_gain++; | |
963 coup->type[c] = get_bits1(gb) ? TYPE_CPE : TYPE_SCE; | |
964 coup->id_select[c] = get_bits(gb, 4); | |
965 if (coup->type[c] == TYPE_CPE) { | |
966 coup->ch_select[c] = get_bits(gb, 2); | |
967 if (coup->ch_select[c] == 3) | |
968 num_gain++; | |
969 } else | |
8004
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970 coup->ch_select[c] = 2; |
7578 | 971 } |
972 coup->coupling_point += get_bits1(gb); | |
973 | |
974 if (coup->coupling_point == 2) { | |
975 av_log(ac->avccontext, AV_LOG_ERROR, | |
976 "Independently switched CCE with 'invalid' domain signalled.\n"); | |
977 memset(coup, 0, sizeof(ChannelCoupling)); | |
978 return -1; | |
979 } | |
980 | |
981 sign = get_bits(gb, 1); | |
7889
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982 scale = pow(2., pow(2., (int)get_bits(gb, 2) - 3)); |
7578 | 983 |
984 if ((ret = decode_ics(ac, sce, gb, 0, 0))) | |
985 return ret; | |
986 | |
987 for (c = 0; c < num_gain; c++) { | |
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988 int idx = 0; |
7578 | 989 int cge = 1; |
990 int gain = 0; | |
991 float gain_cache = 1.; | |
992 if (c) { | |
993 cge = coup->coupling_point == AFTER_IMDCT ? 1 : get_bits1(gb); | |
994 gain = cge ? get_vlc2(gb, vlc_scalefactors.table, 7, 3) - 60: 0; | |
8004
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995 gain_cache = pow(scale, -gain); |
7578 | 996 } |
7890 | 997 for (g = 0; g < sce->ics.num_window_groups; g++) { |
998 for (sfb = 0; sfb < sce->ics.max_sfb; sfb++, idx++) { | |
7578 | 999 if (sce->band_type[idx] != ZERO_BT) { |
1000 if (!cge) { | |
1001 int t = get_vlc2(gb, vlc_scalefactors.table, 7, 3) - 60; | |
1002 if (t) { | |
1003 int s = 1; | |
8004
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1004 t = gain += t; |
7578 | 1005 if (sign) { |
1006 s -= 2 * (t & 0x1); | |
1007 t >>= 1; | |
1008 } | |
8004
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1009 gain_cache = pow(scale, -t) * s; |
7578 | 1010 } |
1011 } | |
1012 coup->gain[c][idx] = gain_cache; | |
1013 } | |
7890 | 1014 } |
1015 } | |
7578 | 1016 } |
1017 return 0; | |
1018 } | |
1019 | |
7539 | 1020 /** |
1021 * Decode Spectral Band Replication extension data; reference: table 4.55. | |
7523
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1022 * |
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1023 * @param crc flag indicating the presence of CRC checksum |
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1024 * @param cnt length of TYPE_FIL syntactic element in bytes |
7539 | 1025 * |
7523
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1026 * @return Returns number of bytes consumed from the TYPE_FIL element. |
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1027 */ |
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1028 static int decode_sbr_extension(AACContext * ac, GetBitContext * gb, int crc, int cnt) { |
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1029 // TODO : sbr_extension implementation |
7540 | 1030 av_log_missing_feature(ac->avccontext, "SBR", 0); |
7523
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1031 skip_bits_long(gb, 8*cnt - 4); // -4 due to reading extension type |
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1032 return cnt; |
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1033 } |
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1034 |
7539 | 1035 /** |
7578 | 1036 * Parse whether channels are to be excluded from Dynamic Range Compression; reference: table 4.53. |
1037 * | |
1038 * @return Returns number of bytes consumed. | |
1039 */ | |
1040 static int decode_drc_channel_exclusions(DynamicRangeControl *che_drc, GetBitContext * gb) { | |
1041 int i; | |
1042 int num_excl_chan = 0; | |
1043 | |
1044 do { | |
1045 for (i = 0; i < 7; i++) | |
1046 che_drc->exclude_mask[num_excl_chan++] = get_bits1(gb); | |
1047 } while (num_excl_chan < MAX_CHANNELS - 7 && get_bits1(gb)); | |
1048 | |
1049 return num_excl_chan / 7; | |
1050 } | |
1051 | |
1052 /** | |
7539 | 1053 * Decode dynamic range information; reference: table 4.52. |
1054 * | |
1055 * @param cnt length of TYPE_FIL syntactic element in bytes | |
1056 * | |
1057 * @return Returns number of bytes consumed. | |
1058 */ | |
1059 static int decode_dynamic_range(DynamicRangeControl *che_drc, GetBitContext * gb, int cnt) { | |
1060 int n = 1; | |
1061 int drc_num_bands = 1; | |
1062 int i; | |
1063 | |
1064 /* pce_tag_present? */ | |
1065 if(get_bits1(gb)) { | |
1066 che_drc->pce_instance_tag = get_bits(gb, 4); | |
1067 skip_bits(gb, 4); // tag_reserved_bits | |
1068 n++; | |
1069 } | |
1070 | |
1071 /* excluded_chns_present? */ | |
1072 if(get_bits1(gb)) { | |
1073 n += decode_drc_channel_exclusions(che_drc, gb); | |
1074 } | |
1075 | |
1076 /* drc_bands_present? */ | |
1077 if (get_bits1(gb)) { | |
1078 che_drc->band_incr = get_bits(gb, 4); | |
1079 che_drc->interpolation_scheme = get_bits(gb, 4); | |
1080 n++; | |
1081 drc_num_bands += che_drc->band_incr; | |
1082 for (i = 0; i < drc_num_bands; i++) { | |
1083 che_drc->band_top[i] = get_bits(gb, 8); | |
1084 n++; | |
1085 } | |
1086 } | |
1087 | |
1088 /* prog_ref_level_present? */ | |
1089 if (get_bits1(gb)) { | |
1090 che_drc->prog_ref_level = get_bits(gb, 7); | |
1091 skip_bits1(gb); // prog_ref_level_reserved_bits | |
1092 n++; | |
1093 } | |
1094 | |
1095 for (i = 0; i < drc_num_bands; i++) { | |
1096 che_drc->dyn_rng_sgn[i] = get_bits1(gb); | |
1097 che_drc->dyn_rng_ctl[i] = get_bits(gb, 7); | |
1098 n++; | |
1099 } | |
1100 | |
1101 return n; | |
1102 } | |
1103 | |
1104 /** | |
1105 * Decode extension data (incomplete); reference: table 4.51. | |
1106 * | |
1107 * @param cnt length of TYPE_FIL syntactic element in bytes | |
1108 * | |
1109 * @return Returns number of bytes consumed | |
1110 */ | |
1111 static int decode_extension_payload(AACContext * ac, GetBitContext * gb, int cnt) { | |
7523
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1112 int crc_flag = 0; |
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1113 int res = cnt; |
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1114 switch (get_bits(gb, 4)) { // extension type |
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1115 case EXT_SBR_DATA_CRC: |
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1116 crc_flag++; |
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1117 case EXT_SBR_DATA: |
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1118 res = decode_sbr_extension(ac, gb, crc_flag, cnt); |
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1119 break; |
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1120 case EXT_DYNAMIC_RANGE: |
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1121 res = decode_dynamic_range(&ac->che_drc, gb, cnt); |
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1122 break; |
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1123 case EXT_FILL: |
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1124 case EXT_FILL_DATA: |
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1125 case EXT_DATA_ELEMENT: |
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1126 default: |
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1127 skip_bits_long(gb, 8*cnt - 4); |
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1128 break; |
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1129 }; |
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1130 return res; |
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1131 } |
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1132 |
7632
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1133 /** |
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1134 * Decode Temporal Noise Shaping filter coefficients and apply all-pole filters; reference: 4.6.9.3. |
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1135 * |
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1136 * @param decode 1 if tool is used normally, 0 if tool is used in LTP. |
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1137 * @param coef spectral coefficients |
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1138 */ |
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1139 static void apply_tns(float coef[1024], TemporalNoiseShaping * tns, IndividualChannelStream * ics, int decode) { |
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1140 const int mmm = FFMIN(ics->tns_max_bands, ics->max_sfb); |
7636 | 1141 int w, filt, m, i; |
7632
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1142 int bottom, top, order, start, end, size, inc; |
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1143 float lpc[TNS_MAX_ORDER]; |
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1144 |
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1145 for (w = 0; w < ics->num_windows; w++) { |
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1146 bottom = ics->num_swb; |
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1147 for (filt = 0; filt < tns->n_filt[w]; filt++) { |
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1148 top = bottom; |
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1149 bottom = FFMAX(0, top - tns->length[w][filt]); |
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1150 order = tns->order[w][filt]; |
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1151 if (order == 0) |
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1152 continue; |
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1153 |
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1154 // tns_decode_coef |
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1155 compute_lpc_coefs(tns->coef[w][filt], order, lpc, 0, 0, 0); |
7632
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1156 |
7609 | 1157 start = ics->swb_offset[FFMIN(bottom, mmm)]; |
1158 end = ics->swb_offset[FFMIN( top, mmm)]; | |
1159 if ((size = end - start) <= 0) | |
1160 continue; | |
1161 if (tns->direction[w][filt]) { | |
1162 inc = -1; start = end - 1; | |
1163 } else { | |
1164 inc = 1; | |
1165 } | |
1166 start += w * 128; | |
1167 | |
1168 // ar filter | |
1169 for (m = 0; m < size; m++, start += inc) | |
1170 for (i = 1; i <= FFMIN(m, order); i++) | |
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1171 coef[start] -= coef[start - i*inc] * lpc[i-1]; |
7609 | 1172 } |
1173 } | |
1174 } | |
1175 | |
7523
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1176 /** |
7539 | 1177 * Conduct IMDCT and windowing. |
1178 */ | |
1179 static void imdct_and_windowing(AACContext * ac, SingleChannelElement * sce) { | |
1180 IndividualChannelStream * ics = &sce->ics; | |
1181 float * in = sce->coeffs; | |
1182 float * out = sce->ret; | |
1183 float * saved = sce->saved; | |
7580 | 1184 const float * swindow = ics->use_kb_window[0] ? ff_aac_kbd_short_128 : ff_sine_128; |
1185 const float * lwindow_prev = ics->use_kb_window[1] ? ff_aac_kbd_long_1024 : ff_sine_1024; | |
1186 const float * swindow_prev = ics->use_kb_window[1] ? ff_aac_kbd_short_128 : ff_sine_128; | |
7539 | 1187 float * buf = ac->buf_mdct; |
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1188 DECLARE_ALIGNED(16, float, temp[128]); |
7539 | 1189 int i; |
1190 | |
7661
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1191 // imdct |
7578 | 1192 if (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) { |
1193 if (ics->window_sequence[1] == ONLY_LONG_SEQUENCE || ics->window_sequence[1] == LONG_STOP_SEQUENCE) | |
1194 av_log(ac->avccontext, AV_LOG_WARNING, | |
1195 "Transition from an ONLY_LONG or LONG_STOP to an EIGHT_SHORT sequence detected. " | |
1196 "If you heard an audible artifact, please submit the sample to the FFmpeg developers.\n"); | |
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1197 for (i = 0; i < 1024; i += 128) |
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1198 ff_imdct_half(&ac->mdct_small, buf + i, in + i); |
7661
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1199 } else |
7663
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1200 ff_imdct_half(&ac->mdct, buf, in); |
7661
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1201 |
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1202 /* window overlapping |
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1203 * NOTE: To simplify the overlapping code, all 'meaningless' short to long |
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1204 * and long to short transitions are considered to be short to short |
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1205 * transitions. This leaves just two cases (long to long and short to short) |
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1206 * with a little special sauce for EIGHT_SHORT_SEQUENCE. |
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1207 */ |
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1208 if ((ics->window_sequence[1] == ONLY_LONG_SEQUENCE || ics->window_sequence[1] == LONG_STOP_SEQUENCE) && |
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1209 (ics->window_sequence[0] == ONLY_LONG_SEQUENCE || ics->window_sequence[0] == LONG_START_SEQUENCE)) { |
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1210 ac->dsp.vector_fmul_window( out, saved, buf, lwindow_prev, ac->add_bias, 512); |
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1211 } else { |
7662 | 1212 for (i = 0; i < 448; i++) |
1213 out[i] = saved[i] + ac->add_bias; | |
7578 | 1214 |
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1215 if (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) { |
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1216 ac->dsp.vector_fmul_window(out + 448 + 0*128, saved + 448, buf + 0*128, swindow_prev, ac->add_bias, 64); |
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1217 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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1218 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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1219 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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1220 ac->dsp.vector_fmul_window(temp, buf + 3*128 + 64, buf + 4*128, swindow, ac->add_bias, 64); |
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1221 memcpy( out + 448 + 4*128, temp, 64 * sizeof(float)); |
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1222 } else { |
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1223 ac->dsp.vector_fmul_window(out + 448, saved + 448, buf, swindow_prev, ac->add_bias, 64); |
7662 | 1224 for (i = 576; i < 1024; i++) |
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1225 out[i] = buf[i-512] + ac->add_bias; |
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1226 } |
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1227 } |
7578 | 1228 |
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1229 // buffer update |
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1230 if (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) { |
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1231 for (i = 0; i < 64; i++) |
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1232 saved[i] = temp[64 + i] - ac->add_bias; |
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1233 ac->dsp.vector_fmul_window(saved + 64, buf + 4*128 + 64, buf + 5*128, swindow, 0, 64); |
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1234 ac->dsp.vector_fmul_window(saved + 192, buf + 5*128 + 64, buf + 6*128, swindow, 0, 64); |
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1235 ac->dsp.vector_fmul_window(saved + 320, buf + 6*128 + 64, buf + 7*128, swindow, 0, 64); |
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1236 memcpy( saved + 448, buf + 7*128 + 64, 64 * sizeof(float)); |
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1237 } else if (ics->window_sequence[0] == LONG_START_SEQUENCE) { |
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1238 memcpy( saved, buf + 512, 448 * sizeof(float)); |
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1239 memcpy( saved + 448, buf + 7*128 + 64, 64 * sizeof(float)); |
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1240 } else { // LONG_STOP or ONLY_LONG |
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1241 memcpy( saved, buf + 512, 512 * sizeof(float)); |
7578 | 1242 } |
1243 } | |
1244 | |
7539 | 1245 /** |
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1246 * Apply dependent channel coupling (applied before IMDCT). |
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1247 * |
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1248 * @param index index into coupling gain array |
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1249 */ |
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1250 static void apply_dependent_coupling(AACContext * ac, SingleChannelElement * target, ChannelElement * cce, int index) { |
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1251 IndividualChannelStream * ics = &cce->ch[0].ics; |
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1252 const uint16_t * offsets = ics->swb_offset; |
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1253 float * dest = target->coeffs; |
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1254 const float * src = cce->ch[0].coeffs; |
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1255 int g, i, group, k, idx = 0; |
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1256 if(ac->m4ac.object_type == AOT_AAC_LTP) { |
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1257 av_log(ac->avccontext, AV_LOG_ERROR, |
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1258 "Dependent coupling is not supported together with LTP\n"); |
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1259 return; |
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1260 } |
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1261 for (g = 0; g < ics->num_window_groups; g++) { |
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1262 for (i = 0; i < ics->max_sfb; i++, idx++) { |
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1263 if (cce->ch[0].band_type[idx] != ZERO_BT) { |
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1264 for (group = 0; group < ics->group_len[g]; group++) { |
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1265 for (k = offsets[i]; k < offsets[i+1]; k++) { |
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1266 // XXX dsputil-ize |
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1267 dest[group*128+k] += cce->coup.gain[index][idx] * src[group*128+k]; |
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1268 } |
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1269 } |
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1270 } |
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1271 } |
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1272 dest += ics->group_len[g]*128; |
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1273 src += ics->group_len[g]*128; |
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1274 } |
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1275 } |
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1276 |
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1277 /** |
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1278 * Apply independent channel coupling (applied after IMDCT). |
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1279 * |
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1280 * @param index index into coupling gain array |
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1281 */ |
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1282 static void apply_independent_coupling(AACContext * ac, SingleChannelElement * target, ChannelElement * cce, int index) { |
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1283 int i; |
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1284 for (i = 0; i < 1024; i++) |
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1285 target->ret[i] += cce->coup.gain[index][0] * (cce->ch[0].ret[i] - ac->add_bias); |
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1286 } |
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1287 |
7581 | 1288 /** |
1289 * channel coupling transformation interface | |
1290 * | |
1291 * @param index index into coupling gain array | |
1292 * @param apply_coupling_method pointer to (in)dependent coupling function | |
1293 */ | |
1294 static void apply_channel_coupling(AACContext * ac, ChannelElement * cc, | |
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1295 enum RawDataBlockType type, int elem_id, enum CouplingPoint coupling_point, |
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1296 void (*apply_coupling_method)(AACContext * ac, SingleChannelElement * target, ChannelElement * cce, int index)) |
7581 | 1297 { |
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1298 int i, c; |
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1299 |
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1300 for (i = 0; i < MAX_ELEM_ID; i++) { |
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1301 ChannelElement *cce = ac->che[TYPE_CCE][i]; |
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1302 int index = 0; |
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1303 |
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1304 if (cce && cce->coup.coupling_point == coupling_point) { |
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1305 ChannelCoupling * coup = &cce->coup; |
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1306 |
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1307 for (c = 0; c <= coup->num_coupled; c++) { |
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1308 if (coup->type[c] == type && coup->id_select[c] == elem_id) { |
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1309 if (coup->ch_select[c] != 1) { |
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1310 apply_coupling_method(ac, &cc->ch[0], cce, index); |
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1311 if (coup->ch_select[c] != 0) |
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1312 index++; |
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1313 } |
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1314 if (coup->ch_select[c] != 2) |
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1315 apply_coupling_method(ac, &cc->ch[1], cce, index++); |
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1316 } else |
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1317 index += 1 + (coup->ch_select[c] == 3); |
7581 | 1318 } |
1319 } | |
1320 } | |
1321 } | |
1322 | |
1323 /** | |
1324 * Convert spectral data to float samples, applying all supported tools as appropriate. | |
1325 */ | |
1326 static void spectral_to_sample(AACContext * ac) { | |
1327 int i, type; | |
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1328 for(type = 3; type >= 0; type--) { |
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1329 for (i = 0; i < MAX_ELEM_ID; i++) { |
7581 | 1330 ChannelElement *che = ac->che[type][i]; |
1331 if(che) { | |
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1332 if(type <= TYPE_CPE) |
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1333 apply_channel_coupling(ac, che, type, i, BEFORE_TNS, apply_dependent_coupling); |
7581 | 1334 if(che->ch[0].tns.present) |
1335 apply_tns(che->ch[0].coeffs, &che->ch[0].tns, &che->ch[0].ics, 1); | |
1336 if(che->ch[1].tns.present) | |
1337 apply_tns(che->ch[1].coeffs, &che->ch[1].tns, &che->ch[1].ics, 1); | |
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1338 if(type <= TYPE_CPE) |
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1339 apply_channel_coupling(ac, che, type, i, BETWEEN_TNS_AND_IMDCT, apply_dependent_coupling); |
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1340 if(type != TYPE_CCE || che->coup.coupling_point == AFTER_IMDCT) |
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1341 imdct_and_windowing(ac, &che->ch[0]); |
7581 | 1342 if(type == TYPE_CPE) |
1343 imdct_and_windowing(ac, &che->ch[1]); | |
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1344 if(type <= TYPE_CCE) |
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1345 apply_channel_coupling(ac, che, type, i, AFTER_IMDCT, apply_independent_coupling); |
7578 | 1346 } |
1347 } | |
1348 } | |
1349 } | |
1350 | |
1351 static int aac_decode_frame(AVCodecContext * avccontext, void * data, int * data_size, const uint8_t * buf, int buf_size) { | |
1352 AACContext * ac = avccontext->priv_data; | |
1353 GetBitContext gb; | |
1354 enum RawDataBlockType elem_type; | |
1355 int err, elem_id, data_size_tmp; | |
1356 | |
1357 init_get_bits(&gb, buf, buf_size*8); | |
1358 | |
1359 // parse | |
1360 while ((elem_type = get_bits(&gb, 3)) != TYPE_END) { | |
1361 elem_id = get_bits(&gb, 4); | |
1362 err = -1; | |
1363 | |
1364 if(elem_type == TYPE_SCE && elem_id == 1 && | |
1365 !ac->che[TYPE_SCE][elem_id] && ac->che[TYPE_LFE][0]) { | |
1366 /* Some streams incorrectly code 5.1 audio as SCE[0] CPE[0] CPE[1] SCE[1] | |
1367 instead of SCE[0] CPE[0] CPE[0] LFE[0]. If we seem to have | |
1368 encountered such a stream, transfer the LFE[0] element to SCE[1] */ | |
1369 ac->che[TYPE_SCE][elem_id] = ac->che[TYPE_LFE][0]; | |
1370 ac->che[TYPE_LFE][0] = NULL; | |
1371 } | |
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1372 if(elem_type < TYPE_DSE) { |
7578 | 1373 if(!ac->che[elem_type][elem_id]) |
1374 return -1; | |
1375 if(elem_type != TYPE_CCE) | |
1376 ac->che[elem_type][elem_id]->coup.coupling_point = 4; | |
1377 } | |
1378 | |
1379 switch (elem_type) { | |
1380 | |
1381 case TYPE_SCE: | |
1382 err = decode_ics(ac, &ac->che[TYPE_SCE][elem_id]->ch[0], &gb, 0, 0); | |
1383 break; | |
1384 | |
1385 case TYPE_CPE: | |
1386 err = decode_cpe(ac, &gb, elem_id); | |
1387 break; | |
1388 | |
1389 case TYPE_CCE: | |
7733 | 1390 err = decode_cce(ac, &gb, ac->che[TYPE_CCE][elem_id]); |
7578 | 1391 break; |
1392 | |
1393 case TYPE_LFE: | |
1394 err = decode_ics(ac, &ac->che[TYPE_LFE][elem_id]->ch[0], &gb, 0, 0); | |
1395 break; | |
1396 | |
1397 case TYPE_DSE: | |
1398 skip_data_stream_element(&gb); | |
1399 err = 0; | |
1400 break; | |
1401 | |
1402 case TYPE_PCE: | |
1403 { | |
1404 enum ChannelPosition new_che_pos[4][MAX_ELEM_ID]; | |
1405 memset(new_che_pos, 0, 4 * MAX_ELEM_ID * sizeof(new_che_pos[0][0])); | |
1406 if((err = decode_pce(ac, new_che_pos, &gb))) | |
1407 break; | |
1408 err = output_configure(ac, ac->che_pos, new_che_pos); | |
1409 break; | |
1410 } | |
1411 | |
1412 case TYPE_FIL: | |
1413 if (elem_id == 15) | |
1414 elem_id += get_bits(&gb, 8) - 1; | |
1415 while (elem_id > 0) | |
1416 elem_id -= decode_extension_payload(ac, &gb, elem_id); | |
1417 err = 0; /* FIXME */ | |
1418 break; | |
1419 | |
1420 default: | |
1421 err = -1; /* should not happen, but keeps compiler happy */ | |
1422 break; | |
1423 } | |
1424 | |
1425 if(err) | |
1426 return err; | |
1427 } | |
1428 | |
1429 spectral_to_sample(ac); | |
1430 | |
7539 | 1431 if (!ac->is_saved) { |
1432 ac->is_saved = 1; | |
1433 *data_size = 0; | |
7580 | 1434 return buf_size; |
7539 | 1435 } |
1436 | |
1437 data_size_tmp = 1024 * avccontext->channels * sizeof(int16_t); | |
1438 if(*data_size < data_size_tmp) { | |
1439 av_log(avccontext, AV_LOG_ERROR, | |
1440 "Output buffer too small (%d) or trying to output too many samples (%d) for this frame.\n", | |
1441 *data_size, data_size_tmp); | |
1442 return -1; | |
1443 } | |
1444 *data_size = data_size_tmp; | |
1445 | |
1446 ac->dsp.float_to_int16_interleave(data, (const float **)ac->output_data, 1024, avccontext->channels); | |
1447 | |
1448 return buf_size; | |
1449 } | |
1450 | |
7501 | 1451 static av_cold int aac_decode_close(AVCodecContext * avccontext) { |
1452 AACContext * ac = avccontext->priv_data; | |
7540 | 1453 int i, type; |
7501 | 1454 |
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1455 for (i = 0; i < MAX_ELEM_ID; i++) { |
7540 | 1456 for(type = 0; type < 4; type++) |
1457 av_freep(&ac->che[type][i]); | |
7501 | 1458 } |
1459 | |
1460 ff_mdct_end(&ac->mdct); | |
1461 ff_mdct_end(&ac->mdct_small); | |
1462 return 0 ; | |
1463 } | |
1464 | |
1465 AVCodec aac_decoder = { | |
1466 "aac", | |
1467 CODEC_TYPE_AUDIO, | |
1468 CODEC_ID_AAC, | |
1469 sizeof(AACContext), | |
1470 aac_decode_init, | |
1471 NULL, | |
1472 aac_decode_close, | |
1473 aac_decode_frame, | |
1474 .long_name = NULL_IF_CONFIG_SMALL("Advanced Audio Coding"), | |
7523
a3f7ffdb676d
Sync already committed code with that in SoC and commit more OKed hunks of code
superdump
parents:
7501
diff
changeset
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1475 .sample_fmts = (enum SampleFormat[]){SAMPLE_FMT_S16,SAMPLE_FMT_NONE}, |
7501 | 1476 }; |