Mercurial > libavcodec.hg
annotate aac.c @ 7700:53fb5eb360d5 libavcodec
simplify mpeg12 avctx->pix_fmt set
author | bcoudurier |
---|---|
date | Tue, 26 Aug 2008 23:25:29 +0000 |
parents | 66fb1f3937fa |
children | 5d93ceaa7a2c |
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" | |
82 | |
83 #include "aac.h" | |
84 #include "aactab.h" | |
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85 #include "aacdectab.h" |
7501 | 86 #include "mpeg4audio.h" |
87 | |
88 #include <assert.h> | |
89 #include <errno.h> | |
90 #include <math.h> | |
91 #include <string.h> | |
92 | |
93 static VLC vlc_scalefactors; | |
94 static VLC vlc_spectral[11]; | |
95 | |
96 | |
7539 | 97 /** |
7578 | 98 * Configure output channel order based on the current program configuration element. |
99 * | |
100 * @param che_pos current channel position configuration | |
101 * @param new_che_pos New channel position configuration - we only do something if it differs from the current one. | |
102 * | |
103 * @return Returns error status. 0 - OK, !0 - error | |
104 */ | |
105 static int output_configure(AACContext *ac, enum ChannelPosition che_pos[4][MAX_ELEM_ID], | |
106 enum ChannelPosition new_che_pos[4][MAX_ELEM_ID]) { | |
107 AVCodecContext *avctx = ac->avccontext; | |
108 int i, type, channels = 0; | |
109 | |
110 if(!memcmp(che_pos, new_che_pos, 4 * MAX_ELEM_ID * sizeof(new_che_pos[0][0]))) | |
111 return 0; /* no change */ | |
112 | |
113 memcpy(che_pos, new_che_pos, 4 * MAX_ELEM_ID * sizeof(new_che_pos[0][0])); | |
114 | |
115 /* Allocate or free elements depending on if they are in the | |
116 * current program configuration. | |
117 * | |
118 * Set up default 1:1 output mapping. | |
119 * | |
120 * For a 5.1 stream the output order will be: | |
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121 * [ Center ] [ Front Left ] [ Front Right ] [ LFE ] [ Surround Left ] [ Surround Right ] |
7578 | 122 */ |
123 | |
124 for(i = 0; i < MAX_ELEM_ID; i++) { | |
125 for(type = 0; type < 4; type++) { | |
126 if(che_pos[type][i]) { | |
127 if(!ac->che[type][i] && !(ac->che[type][i] = av_mallocz(sizeof(ChannelElement)))) | |
128 return AVERROR(ENOMEM); | |
129 if(type != TYPE_CCE) { | |
130 ac->output_data[channels++] = ac->che[type][i]->ch[0].ret; | |
131 if(type == TYPE_CPE) { | |
132 ac->output_data[channels++] = ac->che[type][i]->ch[1].ret; | |
133 } | |
134 } | |
135 } else | |
136 av_freep(&ac->che[type][i]); | |
137 } | |
138 } | |
139 | |
140 avctx->channels = channels; | |
141 return 0; | |
142 } | |
143 | |
144 /** | |
7539 | 145 * Decode an array of 4 bit element IDs, optionally interleaved with a stereo/mono switching bit. |
146 * | |
147 * @param cpe_map Stereo (Channel Pair Element) map, NULL if stereo bit is not present. | |
148 * @param sce_map mono (Single Channel Element) map | |
149 * @param type speaker type/position for these channels | |
150 */ | |
151 static void decode_channel_map(enum ChannelPosition *cpe_map, | |
152 enum ChannelPosition *sce_map, enum ChannelPosition type, GetBitContext * gb, int n) { | |
153 while(n--) { | |
154 enum ChannelPosition *map = cpe_map && get_bits1(gb) ? cpe_map : sce_map; // stereo or mono map | |
155 map[get_bits(gb, 4)] = type; | |
156 } | |
157 } | |
158 | |
159 /** | |
160 * Decode program configuration element; reference: table 4.2. | |
161 * | |
162 * @param new_che_pos New channel position configuration - we only do something if it differs from the current one. | |
163 * | |
164 * @return Returns error status. 0 - OK, !0 - error | |
165 */ | |
166 static int decode_pce(AACContext * ac, enum ChannelPosition new_che_pos[4][MAX_ELEM_ID], | |
167 GetBitContext * gb) { | |
168 int num_front, num_side, num_back, num_lfe, num_assoc_data, num_cc; | |
169 | |
170 skip_bits(gb, 2); // object_type | |
171 | |
172 ac->m4ac.sampling_index = get_bits(gb, 4); | |
173 if(ac->m4ac.sampling_index > 11) { | |
174 av_log(ac->avccontext, AV_LOG_ERROR, "invalid sampling rate index %d\n", ac->m4ac.sampling_index); | |
175 return -1; | |
176 } | |
177 ac->m4ac.sample_rate = ff_mpeg4audio_sample_rates[ac->m4ac.sampling_index]; | |
7501 | 178 num_front = get_bits(gb, 4); |
179 num_side = get_bits(gb, 4); | |
180 num_back = get_bits(gb, 4); | |
181 num_lfe = get_bits(gb, 2); | |
182 num_assoc_data = get_bits(gb, 3); | |
183 num_cc = get_bits(gb, 4); | |
184 | |
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185 if (get_bits1(gb)) |
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186 skip_bits(gb, 4); // mono_mixdown_tag |
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187 if (get_bits1(gb)) |
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188 skip_bits(gb, 4); // stereo_mixdown_tag |
7501 | 189 |
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190 if (get_bits1(gb)) |
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191 skip_bits(gb, 3); // mixdown_coeff_index and pseudo_surround |
7501 | 192 |
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193 decode_channel_map(new_che_pos[TYPE_CPE], new_che_pos[TYPE_SCE], AAC_CHANNEL_FRONT, gb, num_front); |
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194 decode_channel_map(new_che_pos[TYPE_CPE], new_che_pos[TYPE_SCE], AAC_CHANNEL_SIDE, gb, num_side ); |
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195 decode_channel_map(new_che_pos[TYPE_CPE], new_che_pos[TYPE_SCE], AAC_CHANNEL_BACK, gb, num_back ); |
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196 decode_channel_map(NULL, new_che_pos[TYPE_LFE], AAC_CHANNEL_LFE, gb, num_lfe ); |
7501 | 197 |
198 skip_bits_long(gb, 4 * num_assoc_data); | |
199 | |
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200 decode_channel_map(new_che_pos[TYPE_CCE], new_che_pos[TYPE_CCE], AAC_CHANNEL_CC, gb, num_cc ); |
7501 | 201 |
202 align_get_bits(gb); | |
203 | |
204 /* comment field, first byte is length */ | |
205 skip_bits_long(gb, 8 * get_bits(gb, 8)); | |
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206 return 0; |
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207 } |
7501 | 208 |
7539 | 209 /** |
210 * Set up channel positions based on a default channel configuration | |
211 * as specified in table 1.17. | |
212 * | |
213 * @param new_che_pos New channel position configuration - we only do something if it differs from the current one. | |
214 * | |
215 * @return Returns error status. 0 - OK, !0 - error | |
216 */ | |
217 static int set_default_channel_config(AACContext *ac, enum ChannelPosition new_che_pos[4][MAX_ELEM_ID], | |
218 int channel_config) | |
219 { | |
220 if(channel_config < 1 || channel_config > 7) { | |
221 av_log(ac->avccontext, AV_LOG_ERROR, "invalid default channel configuration (%d)\n", | |
222 channel_config); | |
223 return -1; | |
224 } | |
225 | |
226 /* default channel configurations: | |
227 * | |
228 * 1ch : front center (mono) | |
229 * 2ch : L + R (stereo) | |
230 * 3ch : front center + L + R | |
231 * 4ch : front center + L + R + back center | |
232 * 5ch : front center + L + R + back stereo | |
233 * 6ch : front center + L + R + back stereo + LFE | |
234 * 7ch : front center + L + R + outer front left + outer front right + back stereo + LFE | |
235 */ | |
236 | |
237 if(channel_config != 2) | |
238 new_che_pos[TYPE_SCE][0] = AAC_CHANNEL_FRONT; // front center (or mono) | |
239 if(channel_config > 1) | |
240 new_che_pos[TYPE_CPE][0] = AAC_CHANNEL_FRONT; // L + R (or stereo) | |
241 if(channel_config == 4) | |
242 new_che_pos[TYPE_SCE][1] = AAC_CHANNEL_BACK; // back center | |
243 if(channel_config > 4) | |
244 new_che_pos[TYPE_CPE][(channel_config == 7) + 1] | |
245 = AAC_CHANNEL_BACK; // back stereo | |
246 if(channel_config > 5) | |
247 new_che_pos[TYPE_LFE][0] = AAC_CHANNEL_LFE; // LFE | |
248 if(channel_config == 7) | |
249 new_che_pos[TYPE_CPE][1] = AAC_CHANNEL_FRONT; // outer front left + outer front right | |
250 | |
251 return 0; | |
252 } | |
253 | |
7578 | 254 /** |
255 * Decode GA "General Audio" specific configuration; reference: table 4.1. | |
256 * | |
257 * @return Returns error status. 0 - OK, !0 - error | |
258 */ | |
259 static int decode_ga_specific_config(AACContext * ac, GetBitContext * gb, int channel_config) { | |
260 enum ChannelPosition new_che_pos[4][MAX_ELEM_ID]; | |
261 int extension_flag, ret; | |
262 | |
263 if(get_bits1(gb)) { // frameLengthFlag | |
264 av_log_missing_feature(ac->avccontext, "960/120 MDCT window is", 1); | |
7539 | 265 return -1; |
266 } | |
267 | |
268 if (get_bits1(gb)) // dependsOnCoreCoder | |
269 skip_bits(gb, 14); // coreCoderDelay | |
270 extension_flag = get_bits1(gb); | |
271 | |
272 if(ac->m4ac.object_type == AOT_AAC_SCALABLE || | |
273 ac->m4ac.object_type == AOT_ER_AAC_SCALABLE) | |
274 skip_bits(gb, 3); // layerNr | |
275 | |
276 memset(new_che_pos, 0, 4 * MAX_ELEM_ID * sizeof(new_che_pos[0][0])); | |
277 if (channel_config == 0) { | |
278 skip_bits(gb, 4); // element_instance_tag | |
279 if((ret = decode_pce(ac, new_che_pos, gb))) | |
280 return ret; | |
281 } else { | |
282 if((ret = set_default_channel_config(ac, new_che_pos, channel_config))) | |
283 return ret; | |
284 } | |
285 if((ret = output_configure(ac, ac->che_pos, new_che_pos))) | |
286 return ret; | |
287 | |
288 if (extension_flag) { | |
289 switch (ac->m4ac.object_type) { | |
290 case AOT_ER_BSAC: | |
291 skip_bits(gb, 5); // numOfSubFrame | |
292 skip_bits(gb, 11); // layer_length | |
293 break; | |
294 case AOT_ER_AAC_LC: | |
295 case AOT_ER_AAC_LTP: | |
296 case AOT_ER_AAC_SCALABLE: | |
297 case AOT_ER_AAC_LD: | |
298 skip_bits(gb, 3); /* aacSectionDataResilienceFlag | |
299 * aacScalefactorDataResilienceFlag | |
300 * aacSpectralDataResilienceFlag | |
301 */ | |
302 break; | |
303 } | |
304 skip_bits1(gb); // extensionFlag3 (TBD in version 3) | |
305 } | |
306 return 0; | |
307 } | |
308 | |
309 /** | |
310 * Decode audio specific configuration; reference: table 1.13. | |
311 * | |
312 * @param data pointer to AVCodecContext extradata | |
313 * @param data_size size of AVCCodecContext extradata | |
314 * | |
315 * @return Returns error status. 0 - OK, !0 - error | |
316 */ | |
317 static int decode_audio_specific_config(AACContext * ac, void *data, int data_size) { | |
318 GetBitContext gb; | |
319 int i; | |
320 | |
321 init_get_bits(&gb, data, data_size * 8); | |
322 | |
323 if((i = ff_mpeg4audio_get_config(&ac->m4ac, data, data_size)) < 0) | |
324 return -1; | |
325 if(ac->m4ac.sampling_index > 11) { | |
326 av_log(ac->avccontext, AV_LOG_ERROR, "invalid sampling rate index %d\n", ac->m4ac.sampling_index); | |
327 return -1; | |
328 } | |
329 | |
330 skip_bits_long(&gb, i); | |
331 | |
332 switch (ac->m4ac.object_type) { | |
333 case AOT_AAC_LC: | |
334 if (decode_ga_specific_config(ac, &gb, ac->m4ac.chan_config)) | |
335 return -1; | |
336 break; | |
337 default: | |
338 av_log(ac->avccontext, AV_LOG_ERROR, "Audio object type %s%d is not supported.\n", | |
339 ac->m4ac.sbr == 1? "SBR+" : "", ac->m4ac.object_type); | |
340 return -1; | |
341 } | |
342 return 0; | |
343 } | |
344 | |
7578 | 345 /** |
346 * linear congruential pseudorandom number generator | |
347 * | |
348 * @param previous_val pointer to the current state of the generator | |
349 * | |
350 * @return Returns a 32-bit pseudorandom integer | |
351 */ | |
352 static av_always_inline int lcg_random(int previous_val) { | |
353 return previous_val * 1664525 + 1013904223; | |
354 } | |
355 | |
7501 | 356 static av_cold int aac_decode_init(AVCodecContext * avccontext) { |
357 AACContext * ac = avccontext->priv_data; | |
358 int i; | |
359 | |
360 ac->avccontext = avccontext; | |
361 | |
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362 if (avccontext->extradata_size <= 0 || |
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363 decode_audio_specific_config(ac, avccontext->extradata, avccontext->extradata_size)) |
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364 return -1; |
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365 |
7539 | 366 avccontext->sample_fmt = SAMPLE_FMT_S16; |
7501 | 367 avccontext->sample_rate = ac->m4ac.sample_rate; |
368 avccontext->frame_size = 1024; | |
369 | |
370 AAC_INIT_VLC_STATIC( 0, 144); | |
371 AAC_INIT_VLC_STATIC( 1, 114); | |
372 AAC_INIT_VLC_STATIC( 2, 188); | |
373 AAC_INIT_VLC_STATIC( 3, 180); | |
374 AAC_INIT_VLC_STATIC( 4, 172); | |
375 AAC_INIT_VLC_STATIC( 5, 140); | |
376 AAC_INIT_VLC_STATIC( 6, 168); | |
377 AAC_INIT_VLC_STATIC( 7, 114); | |
378 AAC_INIT_VLC_STATIC( 8, 262); | |
379 AAC_INIT_VLC_STATIC( 9, 248); | |
380 AAC_INIT_VLC_STATIC(10, 384); | |
381 | |
382 dsputil_init(&ac->dsp, avccontext); | |
383 | |
7539 | 384 ac->random_state = 0x1f2e3d4c; |
385 | |
7501 | 386 // -1024 - Compensate wrong IMDCT method. |
387 // 32768 - Required to scale values to the correct range for the bias method | |
388 // for float to int16 conversion. | |
389 | |
390 if(ac->dsp.float_to_int16 == ff_float_to_int16_c) { | |
391 ac->add_bias = 385.0f; | |
392 ac->sf_scale = 1. / (-1024. * 32768.); | |
393 ac->sf_offset = 0; | |
394 } else { | |
395 ac->add_bias = 0.0f; | |
396 ac->sf_scale = 1. / -1024.; | |
397 ac->sf_offset = 60; | |
398 } | |
399 | |
400 #ifndef CONFIG_HARDCODED_TABLES | |
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401 for (i = 0; i < 316; i++) |
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402 ff_aac_pow2sf_tab[i] = pow(2, (i - 200)/4.); |
7501 | 403 #endif /* CONFIG_HARDCODED_TABLES */ |
404 | |
405 INIT_VLC_STATIC(&vlc_scalefactors, 7, sizeof(ff_aac_scalefactor_code)/sizeof(ff_aac_scalefactor_code[0]), | |
406 ff_aac_scalefactor_bits, sizeof(ff_aac_scalefactor_bits[0]), sizeof(ff_aac_scalefactor_bits[0]), | |
407 ff_aac_scalefactor_code, sizeof(ff_aac_scalefactor_code[0]), sizeof(ff_aac_scalefactor_code[0]), | |
408 352); | |
409 | |
410 ff_mdct_init(&ac->mdct, 11, 1); | |
411 ff_mdct_init(&ac->mdct_small, 8, 1); | |
7581 | 412 // window initialization |
413 ff_kbd_window_init(ff_aac_kbd_long_1024, 4.0, 1024); | |
414 ff_kbd_window_init(ff_aac_kbd_short_128, 6.0, 128); | |
415 ff_sine_window_init(ff_sine_1024, 1024); | |
416 ff_sine_window_init(ff_sine_128, 128); | |
417 | |
7501 | 418 return 0; |
419 } | |
420 | |
7539 | 421 /** |
422 * Skip data_stream_element; reference: table 4.10. | |
423 */ | |
424 static void skip_data_stream_element(GetBitContext * gb) { | |
7501 | 425 int byte_align = get_bits1(gb); |
426 int count = get_bits(gb, 8); | |
427 if (count == 255) | |
428 count += get_bits(gb, 8); | |
429 if (byte_align) | |
430 align_get_bits(gb); | |
431 skip_bits_long(gb, 8 * count); | |
432 } | |
433 | |
434 /** | |
7539 | 435 * Decode Individual Channel Stream info; reference: table 4.6. |
436 * | |
437 * @param common_window Channels have independent [0], or shared [1], Individual Channel Stream information. | |
438 */ | |
439 static int decode_ics_info(AACContext * ac, IndividualChannelStream * ics, GetBitContext * gb, int common_window) { | |
440 if (get_bits1(gb)) { | |
441 av_log(ac->avccontext, AV_LOG_ERROR, "Reserved bit set.\n"); | |
442 memset(ics, 0, sizeof(IndividualChannelStream)); | |
443 return -1; | |
444 } | |
445 ics->window_sequence[1] = ics->window_sequence[0]; | |
446 ics->window_sequence[0] = get_bits(gb, 2); | |
447 ics->use_kb_window[1] = ics->use_kb_window[0]; | |
448 ics->use_kb_window[0] = get_bits1(gb); | |
449 ics->num_window_groups = 1; | |
450 ics->group_len[0] = 1; | |
7581 | 451 if (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) { |
452 int i; | |
453 ics->max_sfb = get_bits(gb, 4); | |
454 for (i = 0; i < 7; i++) { | |
455 if (get_bits1(gb)) { | |
456 ics->group_len[ics->num_window_groups-1]++; | |
457 } else { | |
458 ics->num_window_groups++; | |
459 ics->group_len[ics->num_window_groups-1] = 1; | |
460 } | |
461 } | |
462 ics->num_windows = 8; | |
463 ics->swb_offset = swb_offset_128[ac->m4ac.sampling_index]; | |
464 ics->num_swb = ff_aac_num_swb_128[ac->m4ac.sampling_index]; | |
465 ics->tns_max_bands = tns_max_bands_128[ac->m4ac.sampling_index]; | |
466 } else { | |
467 ics->max_sfb = get_bits(gb, 6); | |
468 ics->num_windows = 1; | |
469 ics->swb_offset = swb_offset_1024[ac->m4ac.sampling_index]; | |
470 ics->num_swb = ff_aac_num_swb_1024[ac->m4ac.sampling_index]; | |
471 ics->tns_max_bands = tns_max_bands_1024[ac->m4ac.sampling_index]; | |
7578 | 472 if (get_bits1(gb)) { |
473 av_log_missing_feature(ac->avccontext, "Predictor bit set but LTP is", 1); | |
474 memset(ics, 0, sizeof(IndividualChannelStream)); | |
475 return -1; | |
476 } | |
477 } | |
478 | |
479 if(ics->max_sfb > ics->num_swb) { | |
480 av_log(ac->avccontext, AV_LOG_ERROR, | |
481 "Number of scalefactor bands in group (%d) exceeds limit (%d).\n", | |
482 ics->max_sfb, ics->num_swb); | |
483 memset(ics, 0, sizeof(IndividualChannelStream)); | |
484 return -1; | |
485 } | |
486 | |
7539 | 487 return 0; |
488 } | |
489 | |
490 /** | |
491 * Decode band types (section_data payload); reference: table 4.46. | |
492 * | |
493 * @param band_type array of the used band type | |
494 * @param band_type_run_end array of the last scalefactor band of a band type run | |
495 * | |
496 * @return Returns error status. 0 - OK, !0 - error | |
497 */ | |
498 static int decode_band_types(AACContext * ac, enum BandType band_type[120], | |
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499 int band_type_run_end[120], GetBitContext * gb, IndividualChannelStream * ics) { |
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500 int g, idx = 0; |
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501 const int bits = (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) ? 3 : 5; |
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502 for (g = 0; g < ics->num_window_groups; g++) { |
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503 int k = 0; |
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504 while (k < ics->max_sfb) { |
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505 uint8_t sect_len = k; |
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506 int sect_len_incr; |
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507 int sect_band_type = get_bits(gb, 4); |
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508 if (sect_band_type == 12) { |
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509 av_log(ac->avccontext, AV_LOG_ERROR, "invalid band type\n"); |
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510 return -1; |
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511 } |
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512 while ((sect_len_incr = get_bits(gb, bits)) == (1 << bits)-1) |
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513 sect_len += sect_len_incr; |
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514 sect_len += sect_len_incr; |
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515 if (sect_len > ics->max_sfb) { |
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516 av_log(ac->avccontext, AV_LOG_ERROR, |
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517 "Number of bands (%d) exceeds limit (%d).\n", |
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518 sect_len, ics->max_sfb); |
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519 return -1; |
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520 } |
7581 | 521 for (; k < sect_len; k++) { |
522 band_type [idx] = sect_band_type; | |
523 band_type_run_end[idx++] = sect_len; | |
524 } | |
7539 | 525 } |
526 } | |
527 return 0; | |
528 } | |
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529 |
7539 | 530 /** |
531 * Decode scalefactors; reference: table 4.47. | |
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532 * |
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533 * @param global_gain first scalefactor value as scalefactors are differentially coded |
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534 * @param band_type array of the used band type |
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535 * @param band_type_run_end array of the last scalefactor band of a band type run |
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536 * @param sf array of scalefactors or intensity stereo positions |
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537 * |
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538 * @return Returns error status. 0 - OK, !0 - error |
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539 */ |
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540 static int decode_scalefactors(AACContext * ac, float sf[120], GetBitContext * gb, |
7540 | 541 unsigned int global_gain, IndividualChannelStream * ics, |
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542 enum BandType band_type[120], int band_type_run_end[120]) { |
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543 const int sf_offset = ac->sf_offset + (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE ? 12 : 0); |
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544 int g, i, idx = 0; |
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545 int offset[3] = { global_gain, global_gain - 90, 100 }; |
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546 int noise_flag = 1; |
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547 static const char *sf_str[3] = { "Global gain", "Noise gain", "Intensity stereo position" }; |
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548 for (g = 0; g < ics->num_window_groups; g++) { |
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549 for (i = 0; i < ics->max_sfb;) { |
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550 int run_end = band_type_run_end[idx]; |
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551 if (band_type[idx] == ZERO_BT) { |
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552 for(; i < run_end; i++, idx++) |
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553 sf[idx] = 0.; |
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554 }else if((band_type[idx] == INTENSITY_BT) || (band_type[idx] == INTENSITY_BT2)) { |
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555 for(; i < run_end; i++, idx++) { |
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556 offset[2] += get_vlc2(gb, vlc_scalefactors.table, 7, 3) - 60; |
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557 if(offset[2] > 255U) { |
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558 av_log(ac->avccontext, AV_LOG_ERROR, |
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559 "%s (%d) out of range.\n", sf_str[2], offset[2]); |
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560 return -1; |
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561 } |
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562 sf[idx] = ff_aac_pow2sf_tab[-offset[2] + 300]; |
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563 } |
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564 }else if(band_type[idx] == NOISE_BT) { |
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565 for(; i < run_end; i++, idx++) { |
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566 if(noise_flag-- > 0) |
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567 offset[1] += get_bits(gb, 9) - 256; |
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568 else |
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569 offset[1] += get_vlc2(gb, vlc_scalefactors.table, 7, 3) - 60; |
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570 if(offset[1] > 255U) { |
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571 av_log(ac->avccontext, AV_LOG_ERROR, |
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572 "%s (%d) out of range.\n", sf_str[1], offset[1]); |
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573 return -1; |
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574 } |
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575 sf[idx] = -ff_aac_pow2sf_tab[ offset[1] + sf_offset]; |
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576 } |
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577 }else { |
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578 for(; i < run_end; i++, idx++) { |
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579 offset[0] += get_vlc2(gb, vlc_scalefactors.table, 7, 3) - 60; |
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580 if(offset[0] > 255U) { |
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581 av_log(ac->avccontext, AV_LOG_ERROR, |
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582 "%s (%d) out of range.\n", sf_str[0], offset[0]); |
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583 return -1; |
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584 } |
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585 sf[idx] = -ff_aac_pow2sf_tab[ offset[0] + sf_offset]; |
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586 } |
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587 } |
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588 } |
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589 } |
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590 return 0; |
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591 } |
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592 |
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593 /** |
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594 * Decode pulse data; reference: table 4.7. |
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595 */ |
7580 | 596 static void decode_pulses(Pulse * pulse, GetBitContext * gb, const uint16_t * swb_offset) { |
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597 int i; |
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598 pulse->num_pulse = get_bits(gb, 2) + 1; |
7580 | 599 pulse->pos[0] = get_bits(gb, 5) + swb_offset[get_bits(gb, 6)]; |
600 pulse->amp[0] = get_bits(gb, 4); | |
601 for (i = 1; i < pulse->num_pulse; i++) { | |
602 pulse->pos[i] = get_bits(gb, 5) + pulse->pos[i-1]; | |
603 pulse->amp[i] = get_bits(gb, 4); | |
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604 } |
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605 } |
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606 |
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607 /** |
7609 | 608 * Decode Temporal Noise Shaping data; reference: table 4.48. |
609 * | |
610 * @return Returns error status. 0 - OK, !0 - error | |
611 */ | |
612 static int decode_tns(AACContext * ac, TemporalNoiseShaping * tns, | |
613 GetBitContext * gb, const IndividualChannelStream * ics) { | |
614 int w, filt, i, coef_len, coef_res, coef_compress; | |
615 const int is8 = ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE; | |
616 const int tns_max_order = is8 ? 7 : ac->m4ac.object_type == AOT_AAC_MAIN ? 20 : 12; | |
617 for (w = 0; w < ics->num_windows; w++) { | |
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618 if ((tns->n_filt[w] = get_bits(gb, 2 - is8))) { |
7609 | 619 coef_res = get_bits1(gb); |
620 | |
7638 | 621 for (filt = 0; filt < tns->n_filt[w]; filt++) { |
622 int tmp2_idx; | |
623 tns->length[w][filt] = get_bits(gb, 6 - 2*is8); | |
7609 | 624 |
7638 | 625 if ((tns->order[w][filt] = get_bits(gb, 5 - 2*is8)) > tns_max_order) { |
626 av_log(ac->avccontext, AV_LOG_ERROR, "TNS filter order %d is greater than maximum %d.", | |
627 tns->order[w][filt], tns_max_order); | |
628 tns->order[w][filt] = 0; | |
629 return -1; | |
630 } | |
631 tns->direction[w][filt] = get_bits1(gb); | |
632 coef_compress = get_bits1(gb); | |
633 coef_len = coef_res + 3 - coef_compress; | |
634 tmp2_idx = 2*coef_compress + coef_res; | |
635 | |
636 for (i = 0; i < tns->order[w][filt]; i++) | |
637 tns->coef[w][filt][i] = tns_tmp2_map[tmp2_idx][get_bits(gb, coef_len)]; | |
7609 | 638 } |
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639 } |
7609 | 640 } |
641 return 0; | |
642 } | |
643 | |
644 /** | |
7539 | 645 * Decode Mid/Side data; reference: table 4.54. |
646 * | |
647 * @param ms_present Indicates mid/side stereo presence. [0] mask is all 0s; | |
648 * [1] mask is decoded from bitstream; [2] mask is all 1s; | |
649 * [3] reserved for scalable AAC | |
650 */ | |
651 static void decode_mid_side_stereo(ChannelElement * cpe, GetBitContext * gb, | |
652 int ms_present) { | |
7578 | 653 int idx; |
654 if (ms_present == 1) { | |
655 for (idx = 0; idx < cpe->ch[0].ics.num_window_groups * cpe->ch[0].ics.max_sfb; idx++) | |
656 cpe->ms_mask[idx] = get_bits1(gb); | |
657 } else if (ms_present == 2) { | |
658 memset(cpe->ms_mask, 1, cpe->ch[0].ics.num_window_groups * cpe->ch[0].ics.max_sfb * sizeof(cpe->ms_mask[0])); | |
659 } | |
660 } | |
7539 | 661 |
662 /** | |
7581 | 663 * Decode spectral data; reference: table 4.50. |
664 * Dequantize and scale spectral data; reference: 4.6.3.3. | |
665 * | |
666 * @param coef array of dequantized, scaled spectral data | |
667 * @param sf array of scalefactors or intensity stereo positions | |
668 * @param pulse_present set if pulses are present | |
669 * @param pulse pointer to pulse data struct | |
670 * @param band_type array of the used band type | |
671 * | |
672 * @return Returns error status. 0 - OK, !0 - error | |
673 */ | |
674 static int decode_spectrum_and_dequant(AACContext * ac, float coef[1024], GetBitContext * gb, float sf[120], | |
675 int pulse_present, const Pulse * pulse, const IndividualChannelStream * ics, enum BandType band_type[120]) { | |
676 int i, k, g, idx = 0; | |
677 const int c = 1024/ics->num_windows; | |
678 const uint16_t * offsets = ics->swb_offset; | |
679 float *coef_base = coef; | |
680 | |
681 for (g = 0; g < ics->num_windows; g++) | |
682 memset(coef + g * 128 + offsets[ics->max_sfb], 0, sizeof(float)*(c - offsets[ics->max_sfb])); | |
683 | |
684 for (g = 0; g < ics->num_window_groups; g++) { | |
685 for (i = 0; i < ics->max_sfb; i++, idx++) { | |
686 const int cur_band_type = band_type[idx]; | |
687 const int dim = cur_band_type >= FIRST_PAIR_BT ? 2 : 4; | |
688 const int is_cb_unsigned = IS_CODEBOOK_UNSIGNED(cur_band_type); | |
689 int group; | |
690 if (cur_band_type == ZERO_BT) { | |
691 for (group = 0; group < ics->group_len[g]; group++) { | |
692 memset(coef + group * 128 + offsets[i], 0, (offsets[i+1] - offsets[i])*sizeof(float)); | |
693 } | |
694 }else if (cur_band_type == NOISE_BT) { | |
695 const float scale = sf[idx] / ((offsets[i+1] - offsets[i]) * PNS_MEAN_ENERGY); | |
696 for (group = 0; group < ics->group_len[g]; group++) { | |
697 for (k = offsets[i]; k < offsets[i+1]; k++) { | |
698 ac->random_state = lcg_random(ac->random_state); | |
699 coef[group*128+k] = ac->random_state * scale; | |
700 } | |
701 } | |
702 }else if (cur_band_type != INTENSITY_BT2 && cur_band_type != INTENSITY_BT) { | |
703 for (group = 0; group < ics->group_len[g]; group++) { | |
704 for (k = offsets[i]; k < offsets[i+1]; k += dim) { | |
705 const int index = get_vlc2(gb, vlc_spectral[cur_band_type - 1].table, 6, 3); | |
706 const int coef_tmp_idx = (group << 7) + k; | |
707 const float *vq_ptr; | |
708 int j; | |
709 if(index >= ff_aac_spectral_sizes[cur_band_type - 1]) { | |
710 av_log(ac->avccontext, AV_LOG_ERROR, | |
711 "Read beyond end of ff_aac_codebook_vectors[%d][]. index %d >= %d\n", | |
712 cur_band_type - 1, index, ff_aac_spectral_sizes[cur_band_type - 1]); | |
713 return -1; | |
714 } | |
715 vq_ptr = &ff_aac_codebook_vectors[cur_band_type - 1][index * dim]; | |
716 if (is_cb_unsigned) { | |
717 for (j = 0; j < dim; j++) | |
718 if (vq_ptr[j]) | |
719 coef[coef_tmp_idx + j] = 1 - 2*(int)get_bits1(gb); | |
720 }else { | |
721 for (j = 0; j < dim; j++) | |
722 coef[coef_tmp_idx + j] = 1.0f; | |
723 } | |
724 if (cur_band_type == ESC_BT) { | |
725 for (j = 0; j < 2; j++) { | |
726 if (vq_ptr[j] == 64.0f) { | |
727 int n = 4; | |
728 /* The total length of escape_sequence must be < 22 bits according | |
729 to the specification (i.e. max is 11111111110xxxxxxxxxx). */ | |
730 while (get_bits1(gb) && n < 15) n++; | |
731 if(n == 15) { | |
732 av_log(ac->avccontext, AV_LOG_ERROR, "error in spectral data, ESC overflow\n"); | |
733 return -1; | |
734 } | |
735 n = (1<<n) + get_bits(gb, n); | |
736 coef[coef_tmp_idx + j] *= cbrtf(fabsf(n)) * n; | |
737 }else | |
738 coef[coef_tmp_idx + j] *= vq_ptr[j]; | |
739 } | |
740 }else | |
741 for (j = 0; j < dim; j++) | |
742 coef[coef_tmp_idx + j] *= vq_ptr[j]; | |
743 for (j = 0; j < dim; j++) | |
744 coef[coef_tmp_idx + j] *= sf[idx]; | |
745 } | |
746 } | |
747 } | |
748 } | |
749 coef += ics->group_len[g]<<7; | |
750 } | |
751 | |
752 if (pulse_present) { | |
753 for(i = 0; i < pulse->num_pulse; i++){ | |
754 float co = coef_base[ pulse->pos[i] ]; | |
755 float ico = co / sqrtf(sqrtf(fabsf(co))) + pulse->amp[i]; | |
756 coef_base[ pulse->pos[i] ] = cbrtf(fabsf(ico)) * ico; | |
757 } | |
758 } | |
759 return 0; | |
760 } | |
761 | |
762 /** | |
7539 | 763 * Decode an individual_channel_stream payload; reference: table 4.44. |
764 * | |
765 * @param common_window Channels have independent [0], or shared [1], Individual Channel Stream information. | |
766 * @param scale_flag scalable [1] or non-scalable [0] AAC (Unused until scalable AAC is implemented.) | |
767 * | |
768 * @return Returns error status. 0 - OK, !0 - error | |
769 */ | |
770 static int decode_ics(AACContext * ac, SingleChannelElement * sce, GetBitContext * gb, int common_window, int scale_flag) { | |
771 Pulse pulse; | |
772 TemporalNoiseShaping * tns = &sce->tns; | |
773 IndividualChannelStream * ics = &sce->ics; | |
774 float * out = sce->coeffs; | |
775 int global_gain, pulse_present = 0; | |
776 | |
7580 | 777 /* This assignment is to silence a GCC warning about the variable being used |
778 * uninitialized when in fact it always is. | |
7539 | 779 */ |
780 pulse.num_pulse = 0; | |
781 | |
782 global_gain = get_bits(gb, 8); | |
783 | |
784 if (!common_window && !scale_flag) { | |
785 if (decode_ics_info(ac, ics, gb, 0) < 0) | |
786 return -1; | |
787 } | |
788 | |
789 if (decode_band_types(ac, sce->band_type, sce->band_type_run_end, gb, ics) < 0) | |
790 return -1; | |
791 if (decode_scalefactors(ac, sce->sf, gb, global_gain, ics, sce->band_type, sce->band_type_run_end) < 0) | |
792 return -1; | |
793 | |
794 pulse_present = 0; | |
795 if (!scale_flag) { | |
796 if ((pulse_present = get_bits1(gb))) { | |
797 if (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) { | |
798 av_log(ac->avccontext, AV_LOG_ERROR, "Pulse tool not allowed in eight short sequence.\n"); | |
799 return -1; | |
800 } | |
7580 | 801 decode_pulses(&pulse, gb, ics->swb_offset); |
7539 | 802 } |
803 if ((tns->present = get_bits1(gb)) && decode_tns(ac, tns, gb, ics)) | |
804 return -1; | |
805 if (get_bits1(gb)) { | |
806 av_log_missing_feature(ac->avccontext, "SSR", 1); | |
807 return -1; | |
808 } | |
809 } | |
810 | |
7580 | 811 if (decode_spectrum_and_dequant(ac, out, gb, sce->sf, pulse_present, &pulse, ics, sce->band_type) < 0) |
7539 | 812 return -1; |
813 return 0; | |
814 } | |
815 | |
816 /** | |
7581 | 817 * Mid/Side stereo decoding; reference: 4.6.8.1.3. |
818 */ | |
819 static void apply_mid_side_stereo(ChannelElement * cpe) { | |
820 const IndividualChannelStream * ics = &cpe->ch[0].ics; | |
821 float *ch0 = cpe->ch[0].coeffs; | |
822 float *ch1 = cpe->ch[1].coeffs; | |
823 int g, i, k, group, idx = 0; | |
824 const uint16_t * offsets = ics->swb_offset; | |
825 for (g = 0; g < ics->num_window_groups; g++) { | |
826 for (i = 0; i < ics->max_sfb; i++, idx++) { | |
827 if (cpe->ms_mask[idx] && | |
828 cpe->ch[0].band_type[idx] < NOISE_BT && cpe->ch[1].band_type[idx] < NOISE_BT) { | |
829 for (group = 0; group < ics->group_len[g]; group++) { | |
830 for (k = offsets[i]; k < offsets[i+1]; k++) { | |
831 float tmp = ch0[group*128 + k] - ch1[group*128 + k]; | |
832 ch0[group*128 + k] += ch1[group*128 + k]; | |
833 ch1[group*128 + k] = tmp; | |
834 } | |
835 } | |
836 } | |
837 } | |
838 ch0 += ics->group_len[g]*128; | |
839 ch1 += ics->group_len[g]*128; | |
840 } | |
841 } | |
842 | |
843 /** | |
844 * intensity stereo decoding; reference: 4.6.8.2.3 | |
845 * | |
846 * @param ms_present Indicates mid/side stereo presence. [0] mask is all 0s; | |
847 * [1] mask is decoded from bitstream; [2] mask is all 1s; | |
848 * [3] reserved for scalable AAC | |
849 */ | |
850 static void apply_intensity_stereo(ChannelElement * cpe, int ms_present) { | |
851 const IndividualChannelStream * ics = &cpe->ch[1].ics; | |
852 SingleChannelElement * sce1 = &cpe->ch[1]; | |
853 float *coef0 = cpe->ch[0].coeffs, *coef1 = cpe->ch[1].coeffs; | |
854 const uint16_t * offsets = ics->swb_offset; | |
855 int g, group, i, k, idx = 0; | |
856 int c; | |
857 float scale; | |
858 for (g = 0; g < ics->num_window_groups; g++) { | |
859 for (i = 0; i < ics->max_sfb;) { | |
860 if (sce1->band_type[idx] == INTENSITY_BT || sce1->band_type[idx] == INTENSITY_BT2) { | |
861 const int bt_run_end = sce1->band_type_run_end[idx]; | |
862 for (; i < bt_run_end; i++, idx++) { | |
863 c = -1 + 2 * (sce1->band_type[idx] - 14); | |
864 if (ms_present) | |
865 c *= 1 - 2 * cpe->ms_mask[idx]; | |
866 scale = c * sce1->sf[idx]; | |
867 for (group = 0; group < ics->group_len[g]; group++) | |
868 for (k = offsets[i]; k < offsets[i+1]; k++) | |
869 coef1[group*128 + k] = scale * coef0[group*128 + k]; | |
870 } | |
871 } else { | |
872 int bt_run_end = sce1->band_type_run_end[idx]; | |
873 idx += bt_run_end - i; | |
874 i = bt_run_end; | |
875 } | |
876 } | |
877 coef0 += ics->group_len[g]*128; | |
878 coef1 += ics->group_len[g]*128; | |
879 } | |
880 } | |
881 | |
882 /** | |
7539 | 883 * Decode a channel_pair_element; reference: table 4.4. |
884 * | |
885 * @param elem_id Identifies the instance of a syntax element. | |
886 * | |
887 * @return Returns error status. 0 - OK, !0 - error | |
888 */ | |
889 static int decode_cpe(AACContext * ac, GetBitContext * gb, int elem_id) { | |
890 int i, ret, common_window, ms_present = 0; | |
891 ChannelElement * cpe; | |
892 | |
893 cpe = ac->che[TYPE_CPE][elem_id]; | |
894 common_window = get_bits1(gb); | |
895 if (common_window) { | |
896 if (decode_ics_info(ac, &cpe->ch[0].ics, gb, 1)) | |
897 return -1; | |
898 i = cpe->ch[1].ics.use_kb_window[0]; | |
899 cpe->ch[1].ics = cpe->ch[0].ics; | |
900 cpe->ch[1].ics.use_kb_window[1] = i; | |
901 ms_present = get_bits(gb, 2); | |
902 if(ms_present == 3) { | |
903 av_log(ac->avccontext, AV_LOG_ERROR, "ms_present = 3 is reserved.\n"); | |
904 return -1; | |
905 } else if(ms_present) | |
906 decode_mid_side_stereo(cpe, gb, ms_present); | |
907 } | |
908 if ((ret = decode_ics(ac, &cpe->ch[0], gb, common_window, 0))) | |
909 return ret; | |
910 if ((ret = decode_ics(ac, &cpe->ch[1], gb, common_window, 0))) | |
911 return ret; | |
912 | |
913 if (common_window && ms_present) | |
914 apply_mid_side_stereo(cpe); | |
915 | |
7580 | 916 apply_intensity_stereo(cpe, ms_present); |
7539 | 917 return 0; |
918 } | |
919 | |
7581 | 920 /** |
921 * Decode coupling_channel_element; reference: table 4.8. | |
922 * | |
923 * @param elem_id Identifies the instance of a syntax element. | |
924 * | |
925 * @return Returns error status. 0 - OK, !0 - error | |
926 */ | |
927 static int decode_cce(AACContext * ac, GetBitContext * gb, ChannelElement * che) { | |
928 int num_gain = 0; | |
929 int c, g, sfb, ret, idx = 0; | |
930 int sign; | |
931 float scale; | |
932 SingleChannelElement * sce = &che->ch[0]; | |
933 ChannelCoupling * coup = &che->coup; | |
934 | |
7578 | 935 coup->coupling_point = 2*get_bits1(gb); |
936 coup->num_coupled = get_bits(gb, 3); | |
937 for (c = 0; c <= coup->num_coupled; c++) { | |
938 num_gain++; | |
939 coup->type[c] = get_bits1(gb) ? TYPE_CPE : TYPE_SCE; | |
940 coup->id_select[c] = get_bits(gb, 4); | |
941 if (coup->type[c] == TYPE_CPE) { | |
942 coup->ch_select[c] = get_bits(gb, 2); | |
943 if (coup->ch_select[c] == 3) | |
944 num_gain++; | |
945 } else | |
946 coup->ch_select[c] = 1; | |
947 } | |
948 coup->coupling_point += get_bits1(gb); | |
949 | |
950 if (coup->coupling_point == 2) { | |
951 av_log(ac->avccontext, AV_LOG_ERROR, | |
952 "Independently switched CCE with 'invalid' domain signalled.\n"); | |
953 memset(coup, 0, sizeof(ChannelCoupling)); | |
954 return -1; | |
955 } | |
956 | |
957 sign = get_bits(gb, 1); | |
958 scale = pow(2., pow(2., get_bits(gb, 2) - 3)); | |
959 | |
960 if ((ret = decode_ics(ac, sce, gb, 0, 0))) | |
961 return ret; | |
962 | |
963 for (c = 0; c < num_gain; c++) { | |
964 int cge = 1; | |
965 int gain = 0; | |
966 float gain_cache = 1.; | |
967 if (c) { | |
968 cge = coup->coupling_point == AFTER_IMDCT ? 1 : get_bits1(gb); | |
969 gain = cge ? get_vlc2(gb, vlc_scalefactors.table, 7, 3) - 60: 0; | |
970 gain_cache = pow(scale, gain); | |
971 } | |
972 for (g = 0; g < sce->ics.num_window_groups; g++) | |
973 for (sfb = 0; sfb < sce->ics.max_sfb; sfb++, idx++) | |
974 if (sce->band_type[idx] != ZERO_BT) { | |
975 if (!cge) { | |
976 int t = get_vlc2(gb, vlc_scalefactors.table, 7, 3) - 60; | |
977 if (t) { | |
978 int s = 1; | |
979 if (sign) { | |
980 s -= 2 * (t & 0x1); | |
981 t >>= 1; | |
982 } | |
983 gain += t; | |
984 gain_cache = pow(scale, gain) * s; | |
985 } | |
986 } | |
987 coup->gain[c][idx] = gain_cache; | |
988 } | |
989 } | |
990 return 0; | |
991 } | |
992 | |
7539 | 993 /** |
994 * Decode Spectral Band Replication extension data; reference: table 4.55. | |
7523
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995 * |
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996 * @param crc flag indicating the presence of CRC checksum |
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997 * @param cnt length of TYPE_FIL syntactic element in bytes |
7539 | 998 * |
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999 * @return Returns number of bytes consumed from the TYPE_FIL element. |
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1000 */ |
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1001 static int decode_sbr_extension(AACContext * ac, GetBitContext * gb, int crc, int cnt) { |
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1002 // TODO : sbr_extension implementation |
7540 | 1003 av_log_missing_feature(ac->avccontext, "SBR", 0); |
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1004 skip_bits_long(gb, 8*cnt - 4); // -4 due to reading extension type |
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1005 return cnt; |
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1006 } |
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1007 |
7539 | 1008 /** |
7578 | 1009 * Parse whether channels are to be excluded from Dynamic Range Compression; reference: table 4.53. |
1010 * | |
1011 * @return Returns number of bytes consumed. | |
1012 */ | |
1013 static int decode_drc_channel_exclusions(DynamicRangeControl *che_drc, GetBitContext * gb) { | |
1014 int i; | |
1015 int num_excl_chan = 0; | |
1016 | |
1017 do { | |
1018 for (i = 0; i < 7; i++) | |
1019 che_drc->exclude_mask[num_excl_chan++] = get_bits1(gb); | |
1020 } while (num_excl_chan < MAX_CHANNELS - 7 && get_bits1(gb)); | |
1021 | |
1022 return num_excl_chan / 7; | |
1023 } | |
1024 | |
1025 /** | |
7539 | 1026 * Decode dynamic range information; reference: table 4.52. |
1027 * | |
1028 * @param cnt length of TYPE_FIL syntactic element in bytes | |
1029 * | |
1030 * @return Returns number of bytes consumed. | |
1031 */ | |
1032 static int decode_dynamic_range(DynamicRangeControl *che_drc, GetBitContext * gb, int cnt) { | |
1033 int n = 1; | |
1034 int drc_num_bands = 1; | |
1035 int i; | |
1036 | |
1037 /* pce_tag_present? */ | |
1038 if(get_bits1(gb)) { | |
1039 che_drc->pce_instance_tag = get_bits(gb, 4); | |
1040 skip_bits(gb, 4); // tag_reserved_bits | |
1041 n++; | |
1042 } | |
1043 | |
1044 /* excluded_chns_present? */ | |
1045 if(get_bits1(gb)) { | |
1046 n += decode_drc_channel_exclusions(che_drc, gb); | |
1047 } | |
1048 | |
1049 /* drc_bands_present? */ | |
1050 if (get_bits1(gb)) { | |
1051 che_drc->band_incr = get_bits(gb, 4); | |
1052 che_drc->interpolation_scheme = get_bits(gb, 4); | |
1053 n++; | |
1054 drc_num_bands += che_drc->band_incr; | |
1055 for (i = 0; i < drc_num_bands; i++) { | |
1056 che_drc->band_top[i] = get_bits(gb, 8); | |
1057 n++; | |
1058 } | |
1059 } | |
1060 | |
1061 /* prog_ref_level_present? */ | |
1062 if (get_bits1(gb)) { | |
1063 che_drc->prog_ref_level = get_bits(gb, 7); | |
1064 skip_bits1(gb); // prog_ref_level_reserved_bits | |
1065 n++; | |
1066 } | |
1067 | |
1068 for (i = 0; i < drc_num_bands; i++) { | |
1069 che_drc->dyn_rng_sgn[i] = get_bits1(gb); | |
1070 che_drc->dyn_rng_ctl[i] = get_bits(gb, 7); | |
1071 n++; | |
1072 } | |
1073 | |
1074 return n; | |
1075 } | |
1076 | |
1077 /** | |
1078 * Decode extension data (incomplete); reference: table 4.51. | |
1079 * | |
1080 * @param cnt length of TYPE_FIL syntactic element in bytes | |
1081 * | |
1082 * @return Returns number of bytes consumed | |
1083 */ | |
1084 static int decode_extension_payload(AACContext * ac, GetBitContext * gb, int cnt) { | |
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1085 int crc_flag = 0; |
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1086 int res = cnt; |
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1087 switch (get_bits(gb, 4)) { // extension type |
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1088 case EXT_SBR_DATA_CRC: |
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1089 crc_flag++; |
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1090 case EXT_SBR_DATA: |
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1091 res = decode_sbr_extension(ac, gb, crc_flag, cnt); |
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1092 break; |
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1093 case EXT_DYNAMIC_RANGE: |
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1094 res = decode_dynamic_range(&ac->che_drc, gb, cnt); |
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1095 break; |
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1096 case EXT_FILL: |
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1097 case EXT_FILL_DATA: |
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1098 case EXT_DATA_ELEMENT: |
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1099 default: |
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1100 skip_bits_long(gb, 8*cnt - 4); |
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1101 break; |
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1102 }; |
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1103 return res; |
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1104 } |
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1105 |
7632
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1106 /** |
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1107 * Decode Temporal Noise Shaping filter coefficients and apply all-pole filters; reference: 4.6.9.3. |
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1108 * |
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1109 * @param decode 1 if tool is used normally, 0 if tool is used in LTP. |
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1110 * @param coef spectral coefficients |
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1111 */ |
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1112 static void apply_tns(float coef[1024], TemporalNoiseShaping * tns, IndividualChannelStream * ics, int decode) { |
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1113 const int mmm = FFMIN(ics->tns_max_bands, ics->max_sfb); |
7636 | 1114 int w, filt, m, i; |
7632
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1115 int bottom, top, order, start, end, size, inc; |
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1116 float lpc[TNS_MAX_ORDER]; |
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1117 |
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1118 for (w = 0; w < ics->num_windows; w++) { |
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1119 bottom = ics->num_swb; |
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1120 for (filt = 0; filt < tns->n_filt[w]; filt++) { |
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1121 top = bottom; |
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1122 bottom = FFMAX(0, top - tns->length[w][filt]); |
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1123 order = tns->order[w][filt]; |
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1124 if (order == 0) |
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1125 continue; |
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1126 |
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1127 /* tns_decode_coef |
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1128 * FIXME: This duplicates the functionality of some double code in lpc.c. |
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1129 */ |
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1130 for (m = 0; m < order; m++) { |
7634 | 1131 float tmp; |
1132 lpc[m] = tns->coef[w][filt][m]; | |
1133 for (i = 0; i < m/2; i++) { | |
1134 tmp = lpc[i]; | |
1135 lpc[i] += lpc[m] * lpc[m-1-i]; | |
1136 lpc[m-1-i] += lpc[m] * tmp; | |
1137 } | |
1138 if(m & 1) | |
1139 lpc[i] += lpc[m] * lpc[i]; | |
7632
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1140 } |
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1141 |
7609 | 1142 start = ics->swb_offset[FFMIN(bottom, mmm)]; |
1143 end = ics->swb_offset[FFMIN( top, mmm)]; | |
1144 if ((size = end - start) <= 0) | |
1145 continue; | |
1146 if (tns->direction[w][filt]) { | |
1147 inc = -1; start = end - 1; | |
1148 } else { | |
1149 inc = 1; | |
1150 } | |
1151 start += w * 128; | |
1152 | |
1153 // ar filter | |
1154 for (m = 0; m < size; m++, start += inc) | |
1155 for (i = 1; i <= FFMIN(m, order); i++) | |
7632
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1156 coef[start] -= coef[start - i*inc] * lpc[i-1]; |
7609 | 1157 } |
1158 } | |
1159 } | |
1160 | |
7523
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1161 /** |
7539 | 1162 * Conduct IMDCT and windowing. |
1163 */ | |
1164 static void imdct_and_windowing(AACContext * ac, SingleChannelElement * sce) { | |
1165 IndividualChannelStream * ics = &sce->ics; | |
1166 float * in = sce->coeffs; | |
1167 float * out = sce->ret; | |
1168 float * saved = sce->saved; | |
7580 | 1169 const float * swindow = ics->use_kb_window[0] ? ff_aac_kbd_short_128 : ff_sine_128; |
1170 const float * lwindow_prev = ics->use_kb_window[1] ? ff_aac_kbd_long_1024 : ff_sine_1024; | |
1171 const float * swindow_prev = ics->use_kb_window[1] ? ff_aac_kbd_short_128 : ff_sine_128; | |
7539 | 1172 float * buf = ac->buf_mdct; |
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1173 DECLARE_ALIGNED(16, float, temp[128]); |
7539 | 1174 int i; |
1175 | |
7661
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1176 // imdct |
7578 | 1177 if (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) { |
1178 if (ics->window_sequence[1] == ONLY_LONG_SEQUENCE || ics->window_sequence[1] == LONG_STOP_SEQUENCE) | |
1179 av_log(ac->avccontext, AV_LOG_WARNING, | |
1180 "Transition from an ONLY_LONG or LONG_STOP to an EIGHT_SHORT sequence detected. " | |
1181 "If you heard an audible artifact, please submit the sample to the FFmpeg developers.\n"); | |
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1182 for (i = 0; i < 1024; i += 128) |
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1183 ff_imdct_half(&ac->mdct_small, buf + i, in + i); |
7661
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1184 } else |
7663
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1185 ff_imdct_half(&ac->mdct, buf, in); |
7661
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1186 |
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1187 /* window overlapping |
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1188 * NOTE: To simplify the overlapping code, all 'meaningless' short to long |
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1189 * and long to short transitions are considered to be short to short |
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1190 * transitions. This leaves just two cases (long to long and short to short) |
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1191 * with a little special sauce for EIGHT_SHORT_SEQUENCE. |
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1192 */ |
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1193 if ((ics->window_sequence[1] == ONLY_LONG_SEQUENCE || ics->window_sequence[1] == LONG_STOP_SEQUENCE) && |
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1194 (ics->window_sequence[0] == ONLY_LONG_SEQUENCE || ics->window_sequence[0] == LONG_START_SEQUENCE)) { |
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1195 ac->dsp.vector_fmul_window( out, saved, buf, lwindow_prev, ac->add_bias, 512); |
7661
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1196 } else { |
7662 | 1197 for (i = 0; i < 448; i++) |
1198 out[i] = saved[i] + ac->add_bias; | |
7578 | 1199 |
7661
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1200 if (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) { |
7663
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1201 ac->dsp.vector_fmul_window(out + 448 + 0*128, saved + 448, buf + 0*128, swindow_prev, ac->add_bias, 64); |
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1202 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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1203 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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1204 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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1205 ac->dsp.vector_fmul_window(temp, buf + 3*128 + 64, buf + 4*128, swindow, ac->add_bias, 64); |
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1206 memcpy( out + 448 + 4*128, temp, 64 * sizeof(float)); |
7661
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1207 } else { |
7663
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1208 ac->dsp.vector_fmul_window(out + 448, saved + 448, buf, swindow_prev, ac->add_bias, 64); |
7662 | 1209 for (i = 576; i < 1024; i++) |
7663
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1210 out[i] = buf[i-512] + ac->add_bias; |
7661
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1211 } |
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1212 } |
7578 | 1213 |
7661
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1214 // buffer update |
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1215 if (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) { |
7663
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1216 for (i = 0; i < 64; i++) |
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1217 saved[i] = temp[64 + i] - ac->add_bias; |
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1218 ac->dsp.vector_fmul_window(saved + 64, buf + 4*128 + 64, buf + 5*128, swindow, 0, 64); |
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1219 ac->dsp.vector_fmul_window(saved + 192, buf + 5*128 + 64, buf + 6*128, swindow, 0, 64); |
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1220 ac->dsp.vector_fmul_window(saved + 320, buf + 6*128 + 64, buf + 7*128, swindow, 0, 64); |
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1221 memcpy( saved + 448, buf + 7*128 + 64, 64 * sizeof(float)); |
7661
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1222 } else if (ics->window_sequence[0] == LONG_START_SEQUENCE) { |
7663
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1223 memcpy( saved, buf + 512, 448 * sizeof(float)); |
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1224 memcpy( saved + 448, buf + 7*128 + 64, 64 * sizeof(float)); |
7661
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1225 } else { // LONG_STOP or ONLY_LONG |
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1226 memcpy( saved, buf + 512, 512 * sizeof(float)); |
7578 | 1227 } |
1228 } | |
1229 | |
7539 | 1230 /** |
7523
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1231 * Apply dependent channel coupling (applied before IMDCT). |
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1232 * |
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1233 * @param index index into coupling gain array |
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1234 */ |
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1235 static void apply_dependent_coupling(AACContext * ac, SingleChannelElement * sce, ChannelElement * cc, int index) { |
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1236 IndividualChannelStream * ics = &cc->ch[0].ics; |
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1237 const uint16_t * offsets = ics->swb_offset; |
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1238 float * dest = sce->coeffs; |
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1239 const float * src = cc->ch[0].coeffs; |
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1240 int g, i, group, k, idx = 0; |
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1241 if(ac->m4ac.object_type == AOT_AAC_LTP) { |
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1242 av_log(ac->avccontext, AV_LOG_ERROR, |
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1243 "Dependent coupling is not supported together with LTP\n"); |
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1244 return; |
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1245 } |
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1246 for (g = 0; g < ics->num_window_groups; g++) { |
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1247 for (i = 0; i < ics->max_sfb; i++, idx++) { |
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1248 if (cc->ch[0].band_type[idx] != ZERO_BT) { |
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1249 for (group = 0; group < ics->group_len[g]; group++) { |
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1250 for (k = offsets[i]; k < offsets[i+1]; k++) { |
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1251 // XXX dsputil-ize |
7540 | 1252 dest[group*128+k] += cc->coup.gain[index][idx] * src[group*128+k]; |
7523
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1253 } |
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1254 } |
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1255 } |
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1256 } |
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1257 dest += ics->group_len[g]*128; |
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1258 src += ics->group_len[g]*128; |
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1259 } |
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1260 } |
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1261 |
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1262 /** |
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1263 * Apply independent channel coupling (applied after IMDCT). |
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1264 * |
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1265 * @param index index into coupling gain array |
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1266 */ |
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1267 static void apply_independent_coupling(AACContext * ac, SingleChannelElement * sce, ChannelElement * cc, int index) { |
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1268 int i; |
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1269 for (i = 0; i < 1024; i++) |
7540 | 1270 sce->ret[i] += cc->coup.gain[index][0] * (cc->ch[0].ret[i] - ac->add_bias); |
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1271 } |
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1272 |
7581 | 1273 /** |
1274 * channel coupling transformation interface | |
1275 * | |
1276 * @param index index into coupling gain array | |
1277 * @param apply_coupling_method pointer to (in)dependent coupling function | |
1278 */ | |
1279 static void apply_channel_coupling(AACContext * ac, ChannelElement * cc, | |
1280 void (*apply_coupling_method)(AACContext * ac, SingleChannelElement * sce, ChannelElement * cc, int index)) | |
1281 { | |
1282 int c; | |
1283 int index = 0; | |
1284 ChannelCoupling * coup = &cc->coup; | |
1285 for (c = 0; c <= coup->num_coupled; c++) { | |
1286 if (ac->che[coup->type[c]][coup->id_select[c]]) { | |
1287 if (coup->ch_select[c] != 2) { | |
1288 apply_coupling_method(ac, &ac->che[coup->type[c]][coup->id_select[c]]->ch[0], cc, index); | |
1289 if (coup->ch_select[c] != 0) | |
1290 index++; | |
1291 } | |
1292 if (coup->ch_select[c] != 1) | |
1293 apply_coupling_method(ac, &ac->che[coup->type[c]][coup->id_select[c]]->ch[1], cc, index++); | |
1294 } else { | |
1295 av_log(ac->avccontext, AV_LOG_ERROR, | |
1296 "coupling target %sE[%d] not available\n", | |
1297 coup->type[c] == TYPE_CPE ? "CP" : "SC", coup->id_select[c]); | |
1298 break; | |
1299 } | |
1300 } | |
1301 } | |
1302 | |
1303 /** | |
1304 * Convert spectral data to float samples, applying all supported tools as appropriate. | |
1305 */ | |
1306 static void spectral_to_sample(AACContext * ac) { | |
1307 int i, type; | |
1308 for (i = 0; i < MAX_ELEM_ID; i++) { | |
1309 for(type = 0; type < 4; type++) { | |
1310 ChannelElement *che = ac->che[type][i]; | |
1311 if(che) { | |
1312 if(che->coup.coupling_point == BEFORE_TNS) | |
1313 apply_channel_coupling(ac, che, apply_dependent_coupling); | |
1314 if(che->ch[0].tns.present) | |
1315 apply_tns(che->ch[0].coeffs, &che->ch[0].tns, &che->ch[0].ics, 1); | |
1316 if(che->ch[1].tns.present) | |
1317 apply_tns(che->ch[1].coeffs, &che->ch[1].tns, &che->ch[1].ics, 1); | |
1318 if(che->coup.coupling_point == BETWEEN_TNS_AND_IMDCT) | |
1319 apply_channel_coupling(ac, che, apply_dependent_coupling); | |
1320 imdct_and_windowing(ac, &che->ch[0]); | |
1321 if(type == TYPE_CPE) | |
1322 imdct_and_windowing(ac, &che->ch[1]); | |
1323 if(che->coup.coupling_point == AFTER_IMDCT) | |
1324 apply_channel_coupling(ac, che, apply_independent_coupling); | |
7578 | 1325 } |
1326 } | |
1327 } | |
1328 } | |
1329 | |
1330 static int aac_decode_frame(AVCodecContext * avccontext, void * data, int * data_size, const uint8_t * buf, int buf_size) { | |
1331 AACContext * ac = avccontext->priv_data; | |
1332 GetBitContext gb; | |
1333 enum RawDataBlockType elem_type; | |
1334 int err, elem_id, data_size_tmp; | |
1335 | |
1336 init_get_bits(&gb, buf, buf_size*8); | |
1337 | |
1338 // parse | |
1339 while ((elem_type = get_bits(&gb, 3)) != TYPE_END) { | |
1340 elem_id = get_bits(&gb, 4); | |
1341 err = -1; | |
1342 | |
1343 if(elem_type == TYPE_SCE && elem_id == 1 && | |
1344 !ac->che[TYPE_SCE][elem_id] && ac->che[TYPE_LFE][0]) { | |
1345 /* Some streams incorrectly code 5.1 audio as SCE[0] CPE[0] CPE[1] SCE[1] | |
1346 instead of SCE[0] CPE[0] CPE[0] LFE[0]. If we seem to have | |
1347 encountered such a stream, transfer the LFE[0] element to SCE[1] */ | |
1348 ac->che[TYPE_SCE][elem_id] = ac->che[TYPE_LFE][0]; | |
1349 ac->che[TYPE_LFE][0] = NULL; | |
1350 } | |
1351 if(elem_type && elem_type < TYPE_DSE) { | |
1352 if(!ac->che[elem_type][elem_id]) | |
1353 return -1; | |
1354 if(elem_type != TYPE_CCE) | |
1355 ac->che[elem_type][elem_id]->coup.coupling_point = 4; | |
1356 } | |
1357 | |
1358 switch (elem_type) { | |
1359 | |
1360 case TYPE_SCE: | |
1361 err = decode_ics(ac, &ac->che[TYPE_SCE][elem_id]->ch[0], &gb, 0, 0); | |
1362 break; | |
1363 | |
1364 case TYPE_CPE: | |
1365 err = decode_cpe(ac, &gb, elem_id); | |
1366 break; | |
1367 | |
1368 case TYPE_CCE: | |
1369 err = decode_cce(ac, &gb, ac->che[TYPE_SCE][elem_id]); | |
1370 break; | |
1371 | |
1372 case TYPE_LFE: | |
1373 err = decode_ics(ac, &ac->che[TYPE_LFE][elem_id]->ch[0], &gb, 0, 0); | |
1374 break; | |
1375 | |
1376 case TYPE_DSE: | |
1377 skip_data_stream_element(&gb); | |
1378 err = 0; | |
1379 break; | |
1380 | |
1381 case TYPE_PCE: | |
1382 { | |
1383 enum ChannelPosition new_che_pos[4][MAX_ELEM_ID]; | |
1384 memset(new_che_pos, 0, 4 * MAX_ELEM_ID * sizeof(new_che_pos[0][0])); | |
1385 if((err = decode_pce(ac, new_che_pos, &gb))) | |
1386 break; | |
1387 err = output_configure(ac, ac->che_pos, new_che_pos); | |
1388 break; | |
1389 } | |
1390 | |
1391 case TYPE_FIL: | |
1392 if (elem_id == 15) | |
1393 elem_id += get_bits(&gb, 8) - 1; | |
1394 while (elem_id > 0) | |
1395 elem_id -= decode_extension_payload(ac, &gb, elem_id); | |
1396 err = 0; /* FIXME */ | |
1397 break; | |
1398 | |
1399 default: | |
1400 err = -1; /* should not happen, but keeps compiler happy */ | |
1401 break; | |
1402 } | |
1403 | |
1404 if(err) | |
1405 return err; | |
1406 } | |
1407 | |
1408 spectral_to_sample(ac); | |
1409 | |
7539 | 1410 if (!ac->is_saved) { |
1411 ac->is_saved = 1; | |
1412 *data_size = 0; | |
7580 | 1413 return buf_size; |
7539 | 1414 } |
1415 | |
1416 data_size_tmp = 1024 * avccontext->channels * sizeof(int16_t); | |
1417 if(*data_size < data_size_tmp) { | |
1418 av_log(avccontext, AV_LOG_ERROR, | |
1419 "Output buffer too small (%d) or trying to output too many samples (%d) for this frame.\n", | |
1420 *data_size, data_size_tmp); | |
1421 return -1; | |
1422 } | |
1423 *data_size = data_size_tmp; | |
1424 | |
1425 ac->dsp.float_to_int16_interleave(data, (const float **)ac->output_data, 1024, avccontext->channels); | |
1426 | |
1427 return buf_size; | |
1428 } | |
1429 | |
7501 | 1430 static av_cold int aac_decode_close(AVCodecContext * avccontext) { |
1431 AACContext * ac = avccontext->priv_data; | |
7540 | 1432 int i, type; |
7501 | 1433 |
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1434 for (i = 0; i < MAX_ELEM_ID; i++) { |
7540 | 1435 for(type = 0; type < 4; type++) |
1436 av_freep(&ac->che[type][i]); | |
7501 | 1437 } |
1438 | |
1439 ff_mdct_end(&ac->mdct); | |
1440 ff_mdct_end(&ac->mdct_small); | |
1441 return 0 ; | |
1442 } | |
1443 | |
1444 AVCodec aac_decoder = { | |
1445 "aac", | |
1446 CODEC_TYPE_AUDIO, | |
1447 CODEC_ID_AAC, | |
1448 sizeof(AACContext), | |
1449 aac_decode_init, | |
1450 NULL, | |
1451 aac_decode_close, | |
1452 aac_decode_frame, | |
1453 .long_name = NULL_IF_CONFIG_SMALL("Advanced Audio Coding"), | |
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1454 .sample_fmts = (enum SampleFormat[]){SAMPLE_FMT_S16,SAMPLE_FMT_NONE}, |
7501 | 1455 }; |