Mercurial > libavcodec.hg
annotate aac.c @ 7578:a05954c505ab libavcodec
More OKed sections of AAC decoder code
author | superdump |
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date | Fri, 15 Aug 2008 00:05:09 +0000 |
parents | 5298e05f931d |
children | c49ab52db74c |
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 #ifndef CONFIG_HARDCODED_TABLES | |
94 static float ff_aac_ivquant_tab[IVQUANT_SIZE]; | |
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95 static float ff_aac_pow2sf_tab[316]; |
7501 | 96 #endif /* CONFIG_HARDCODED_TABLES */ |
97 | |
98 static VLC vlc_scalefactors; | |
99 static VLC vlc_spectral[11]; | |
100 | |
101 | |
7539 | 102 /** |
7578 | 103 * Configure output channel order based on the current program configuration element. |
104 * | |
105 * @param che_pos current channel position configuration | |
106 * @param new_che_pos New channel position configuration - we only do something if it differs from the current one. | |
107 * | |
108 * @return Returns error status. 0 - OK, !0 - error | |
109 */ | |
110 static int output_configure(AACContext *ac, enum ChannelPosition che_pos[4][MAX_ELEM_ID], | |
111 enum ChannelPosition new_che_pos[4][MAX_ELEM_ID]) { | |
112 AVCodecContext *avctx = ac->avccontext; | |
113 int i, type, channels = 0; | |
114 | |
115 if(!memcmp(che_pos, new_che_pos, 4 * MAX_ELEM_ID * sizeof(new_che_pos[0][0]))) | |
116 return 0; /* no change */ | |
117 | |
118 memcpy(che_pos, new_che_pos, 4 * MAX_ELEM_ID * sizeof(new_che_pos[0][0])); | |
119 | |
120 /* Allocate or free elements depending on if they are in the | |
121 * current program configuration. | |
122 * | |
123 * Set up default 1:1 output mapping. | |
124 * | |
125 * For a 5.1 stream the output order will be: | |
126 * [ Front Left ] [ Front Right ] [ Center ] [ LFE ] [ Surround Left ] [ Surround Right ] | |
127 */ | |
128 | |
129 for(i = 0; i < MAX_ELEM_ID; i++) { | |
130 for(type = 0; type < 4; type++) { | |
131 if(che_pos[type][i]) { | |
132 if(!ac->che[type][i] && !(ac->che[type][i] = av_mallocz(sizeof(ChannelElement)))) | |
133 return AVERROR(ENOMEM); | |
134 if(type != TYPE_CCE) { | |
135 ac->output_data[channels++] = ac->che[type][i]->ch[0].ret; | |
136 if(type == TYPE_CPE) { | |
137 ac->output_data[channels++] = ac->che[type][i]->ch[1].ret; | |
138 } | |
139 } | |
140 } else | |
141 av_freep(&ac->che[type][i]); | |
142 } | |
143 } | |
144 | |
145 avctx->channels = channels; | |
146 return 0; | |
147 } | |
148 | |
149 /** | |
7539 | 150 * Decode an array of 4 bit element IDs, optionally interleaved with a stereo/mono switching bit. |
151 * | |
152 * @param cpe_map Stereo (Channel Pair Element) map, NULL if stereo bit is not present. | |
153 * @param sce_map mono (Single Channel Element) map | |
154 * @param type speaker type/position for these channels | |
155 */ | |
156 static void decode_channel_map(enum ChannelPosition *cpe_map, | |
157 enum ChannelPosition *sce_map, enum ChannelPosition type, GetBitContext * gb, int n) { | |
158 while(n--) { | |
159 enum ChannelPosition *map = cpe_map && get_bits1(gb) ? cpe_map : sce_map; // stereo or mono map | |
160 map[get_bits(gb, 4)] = type; | |
161 } | |
162 } | |
163 | |
164 /** | |
165 * Decode program configuration element; reference: table 4.2. | |
166 * | |
167 * @param new_che_pos New channel position configuration - we only do something if it differs from the current one. | |
168 * | |
169 * @return Returns error status. 0 - OK, !0 - error | |
170 */ | |
171 static int decode_pce(AACContext * ac, enum ChannelPosition new_che_pos[4][MAX_ELEM_ID], | |
172 GetBitContext * gb) { | |
173 int num_front, num_side, num_back, num_lfe, num_assoc_data, num_cc; | |
174 | |
175 skip_bits(gb, 2); // object_type | |
176 | |
177 ac->m4ac.sampling_index = get_bits(gb, 4); | |
178 if(ac->m4ac.sampling_index > 11) { | |
179 av_log(ac->avccontext, AV_LOG_ERROR, "invalid sampling rate index %d\n", ac->m4ac.sampling_index); | |
180 return -1; | |
181 } | |
182 ac->m4ac.sample_rate = ff_mpeg4audio_sample_rates[ac->m4ac.sampling_index]; | |
7501 | 183 num_front = get_bits(gb, 4); |
184 num_side = get_bits(gb, 4); | |
185 num_back = get_bits(gb, 4); | |
186 num_lfe = get_bits(gb, 2); | |
187 num_assoc_data = get_bits(gb, 3); | |
188 num_cc = get_bits(gb, 4); | |
189 | |
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190 if (get_bits1(gb)) |
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191 skip_bits(gb, 4); // mono_mixdown_tag |
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192 if (get_bits1(gb)) |
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193 skip_bits(gb, 4); // stereo_mixdown_tag |
7501 | 194 |
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195 if (get_bits1(gb)) |
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196 skip_bits(gb, 3); // mixdown_coeff_index and pseudo_surround |
7501 | 197 |
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198 decode_channel_map(new_che_pos[TYPE_CPE], new_che_pos[TYPE_SCE], AAC_CHANNEL_FRONT, gb, num_front); |
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199 decode_channel_map(new_che_pos[TYPE_CPE], new_che_pos[TYPE_SCE], AAC_CHANNEL_SIDE, gb, num_side ); |
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200 decode_channel_map(new_che_pos[TYPE_CPE], new_che_pos[TYPE_SCE], AAC_CHANNEL_BACK, gb, num_back ); |
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201 decode_channel_map(NULL, new_che_pos[TYPE_LFE], AAC_CHANNEL_LFE, gb, num_lfe ); |
7501 | 202 |
203 skip_bits_long(gb, 4 * num_assoc_data); | |
204 | |
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205 decode_channel_map(new_che_pos[TYPE_CCE], new_che_pos[TYPE_CCE], AAC_CHANNEL_CC, gb, num_cc ); |
7501 | 206 |
207 align_get_bits(gb); | |
208 | |
209 /* comment field, first byte is length */ | |
210 skip_bits_long(gb, 8 * get_bits(gb, 8)); | |
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211 return 0; |
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212 } |
7501 | 213 |
7539 | 214 /** |
215 * Set up channel positions based on a default channel configuration | |
216 * as specified in table 1.17. | |
217 * | |
218 * @param new_che_pos New channel position configuration - we only do something if it differs from the current one. | |
219 * | |
220 * @return Returns error status. 0 - OK, !0 - error | |
221 */ | |
222 static int set_default_channel_config(AACContext *ac, enum ChannelPosition new_che_pos[4][MAX_ELEM_ID], | |
223 int channel_config) | |
224 { | |
225 if(channel_config < 1 || channel_config > 7) { | |
226 av_log(ac->avccontext, AV_LOG_ERROR, "invalid default channel configuration (%d)\n", | |
227 channel_config); | |
228 return -1; | |
229 } | |
230 | |
231 /* default channel configurations: | |
232 * | |
233 * 1ch : front center (mono) | |
234 * 2ch : L + R (stereo) | |
235 * 3ch : front center + L + R | |
236 * 4ch : front center + L + R + back center | |
237 * 5ch : front center + L + R + back stereo | |
238 * 6ch : front center + L + R + back stereo + LFE | |
239 * 7ch : front center + L + R + outer front left + outer front right + back stereo + LFE | |
240 */ | |
241 | |
242 if(channel_config != 2) | |
243 new_che_pos[TYPE_SCE][0] = AAC_CHANNEL_FRONT; // front center (or mono) | |
244 if(channel_config > 1) | |
245 new_che_pos[TYPE_CPE][0] = AAC_CHANNEL_FRONT; // L + R (or stereo) | |
246 if(channel_config == 4) | |
247 new_che_pos[TYPE_SCE][1] = AAC_CHANNEL_BACK; // back center | |
248 if(channel_config > 4) | |
249 new_che_pos[TYPE_CPE][(channel_config == 7) + 1] | |
250 = AAC_CHANNEL_BACK; // back stereo | |
251 if(channel_config > 5) | |
252 new_che_pos[TYPE_LFE][0] = AAC_CHANNEL_LFE; // LFE | |
253 if(channel_config == 7) | |
254 new_che_pos[TYPE_CPE][1] = AAC_CHANNEL_FRONT; // outer front left + outer front right | |
255 | |
256 return 0; | |
257 } | |
258 | |
7578 | 259 /** |
260 * Decode GA "General Audio" specific configuration; reference: table 4.1. | |
261 * | |
262 * @return Returns error status. 0 - OK, !0 - error | |
263 */ | |
264 static int decode_ga_specific_config(AACContext * ac, GetBitContext * gb, int channel_config) { | |
265 enum ChannelPosition new_che_pos[4][MAX_ELEM_ID]; | |
266 int extension_flag, ret; | |
267 | |
268 if(get_bits1(gb)) { // frameLengthFlag | |
269 av_log_missing_feature(ac->avccontext, "960/120 MDCT window is", 1); | |
7539 | 270 return -1; |
271 } | |
272 | |
273 if (get_bits1(gb)) // dependsOnCoreCoder | |
274 skip_bits(gb, 14); // coreCoderDelay | |
275 extension_flag = get_bits1(gb); | |
276 | |
277 if(ac->m4ac.object_type == AOT_AAC_SCALABLE || | |
278 ac->m4ac.object_type == AOT_ER_AAC_SCALABLE) | |
279 skip_bits(gb, 3); // layerNr | |
280 | |
281 memset(new_che_pos, 0, 4 * MAX_ELEM_ID * sizeof(new_che_pos[0][0])); | |
282 if (channel_config == 0) { | |
283 skip_bits(gb, 4); // element_instance_tag | |
284 if((ret = decode_pce(ac, new_che_pos, gb))) | |
285 return ret; | |
286 } else { | |
287 if((ret = set_default_channel_config(ac, new_che_pos, channel_config))) | |
288 return ret; | |
289 } | |
290 if((ret = output_configure(ac, ac->che_pos, new_che_pos))) | |
291 return ret; | |
292 | |
293 if (extension_flag) { | |
294 switch (ac->m4ac.object_type) { | |
295 case AOT_ER_BSAC: | |
296 skip_bits(gb, 5); // numOfSubFrame | |
297 skip_bits(gb, 11); // layer_length | |
298 break; | |
299 case AOT_ER_AAC_LC: | |
300 case AOT_ER_AAC_LTP: | |
301 case AOT_ER_AAC_SCALABLE: | |
302 case AOT_ER_AAC_LD: | |
303 skip_bits(gb, 3); /* aacSectionDataResilienceFlag | |
304 * aacScalefactorDataResilienceFlag | |
305 * aacSpectralDataResilienceFlag | |
306 */ | |
307 break; | |
308 } | |
309 skip_bits1(gb); // extensionFlag3 (TBD in version 3) | |
310 } | |
311 return 0; | |
312 } | |
313 | |
314 /** | |
315 * Decode audio specific configuration; reference: table 1.13. | |
316 * | |
317 * @param data pointer to AVCodecContext extradata | |
318 * @param data_size size of AVCCodecContext extradata | |
319 * | |
320 * @return Returns error status. 0 - OK, !0 - error | |
321 */ | |
322 static int decode_audio_specific_config(AACContext * ac, void *data, int data_size) { | |
323 GetBitContext gb; | |
324 int i; | |
325 | |
326 init_get_bits(&gb, data, data_size * 8); | |
327 | |
328 if((i = ff_mpeg4audio_get_config(&ac->m4ac, data, data_size)) < 0) | |
329 return -1; | |
330 if(ac->m4ac.sampling_index > 11) { | |
331 av_log(ac->avccontext, AV_LOG_ERROR, "invalid sampling rate index %d\n", ac->m4ac.sampling_index); | |
332 return -1; | |
333 } | |
334 | |
335 skip_bits_long(&gb, i); | |
336 | |
337 switch (ac->m4ac.object_type) { | |
338 case AOT_AAC_LC: | |
339 if (decode_ga_specific_config(ac, &gb, ac->m4ac.chan_config)) | |
340 return -1; | |
341 break; | |
342 default: | |
343 av_log(ac->avccontext, AV_LOG_ERROR, "Audio object type %s%d is not supported.\n", | |
344 ac->m4ac.sbr == 1? "SBR+" : "", ac->m4ac.object_type); | |
345 return -1; | |
346 } | |
347 return 0; | |
348 } | |
349 | |
7578 | 350 /** |
351 * linear congruential pseudorandom number generator | |
352 * | |
353 * @param previous_val pointer to the current state of the generator | |
354 * | |
355 * @return Returns a 32-bit pseudorandom integer | |
356 */ | |
357 static av_always_inline int lcg_random(int previous_val) { | |
358 return previous_val * 1664525 + 1013904223; | |
359 } | |
360 | |
7501 | 361 static av_cold int aac_decode_init(AVCodecContext * avccontext) { |
362 AACContext * ac = avccontext->priv_data; | |
363 int i; | |
364 | |
365 ac->avccontext = avccontext; | |
366 | |
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367 if (avccontext->extradata_size <= 0 || |
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368 decode_audio_specific_config(ac, avccontext->extradata, avccontext->extradata_size)) |
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369 return -1; |
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370 |
7539 | 371 avccontext->sample_fmt = SAMPLE_FMT_S16; |
7501 | 372 avccontext->sample_rate = ac->m4ac.sample_rate; |
373 avccontext->frame_size = 1024; | |
374 | |
375 AAC_INIT_VLC_STATIC( 0, 144); | |
376 AAC_INIT_VLC_STATIC( 1, 114); | |
377 AAC_INIT_VLC_STATIC( 2, 188); | |
378 AAC_INIT_VLC_STATIC( 3, 180); | |
379 AAC_INIT_VLC_STATIC( 4, 172); | |
380 AAC_INIT_VLC_STATIC( 5, 140); | |
381 AAC_INIT_VLC_STATIC( 6, 168); | |
382 AAC_INIT_VLC_STATIC( 7, 114); | |
383 AAC_INIT_VLC_STATIC( 8, 262); | |
384 AAC_INIT_VLC_STATIC( 9, 248); | |
385 AAC_INIT_VLC_STATIC(10, 384); | |
386 | |
387 dsputil_init(&ac->dsp, avccontext); | |
388 | |
7539 | 389 ac->random_state = 0x1f2e3d4c; |
390 | |
7501 | 391 // -1024 - Compensate wrong IMDCT method. |
392 // 32768 - Required to scale values to the correct range for the bias method | |
393 // for float to int16 conversion. | |
394 | |
395 if(ac->dsp.float_to_int16 == ff_float_to_int16_c) { | |
396 ac->add_bias = 385.0f; | |
397 ac->sf_scale = 1. / (-1024. * 32768.); | |
398 ac->sf_offset = 0; | |
399 } else { | |
400 ac->add_bias = 0.0f; | |
401 ac->sf_scale = 1. / -1024.; | |
402 ac->sf_offset = 60; | |
403 } | |
404 | |
405 #ifndef CONFIG_HARDCODED_TABLES | |
406 for (i = 1 - IVQUANT_SIZE/2; i < IVQUANT_SIZE/2; i++) | |
407 ff_aac_ivquant_tab[i + IVQUANT_SIZE/2 - 1] = cbrt(fabs(i)) * i; | |
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408 for (i = 0; i < 316; i++) |
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409 ff_aac_pow2sf_tab[i] = pow(2, (i - 200)/4.); |
7501 | 410 #endif /* CONFIG_HARDCODED_TABLES */ |
411 | |
412 INIT_VLC_STATIC(&vlc_scalefactors, 7, sizeof(ff_aac_scalefactor_code)/sizeof(ff_aac_scalefactor_code[0]), | |
413 ff_aac_scalefactor_bits, sizeof(ff_aac_scalefactor_bits[0]), sizeof(ff_aac_scalefactor_bits[0]), | |
414 ff_aac_scalefactor_code, sizeof(ff_aac_scalefactor_code[0]), sizeof(ff_aac_scalefactor_code[0]), | |
415 352); | |
416 | |
417 ff_mdct_init(&ac->mdct, 11, 1); | |
418 ff_mdct_init(&ac->mdct_small, 8, 1); | |
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; | |
452 | |
7578 | 453 if (get_bits1(gb)) { |
454 av_log_missing_feature(ac->avccontext, "Predictor bit set but LTP is", 1); | |
455 memset(ics, 0, sizeof(IndividualChannelStream)); | |
456 return -1; | |
457 } | |
458 } | |
459 | |
460 if(ics->max_sfb > ics->num_swb) { | |
461 av_log(ac->avccontext, AV_LOG_ERROR, | |
462 "Number of scalefactor bands in group (%d) exceeds limit (%d).\n", | |
463 ics->max_sfb, ics->num_swb); | |
464 memset(ics, 0, sizeof(IndividualChannelStream)); | |
465 return -1; | |
466 } | |
467 | |
7539 | 468 return 0; |
469 } | |
470 | |
471 /** | |
7501 | 472 * inverse quantization |
473 * | |
474 * @param a quantized value to be dequantized | |
475 * @return Returns dequantized value. | |
476 */ | |
477 static inline float ivquant(int a) { | |
478 if (a + (unsigned int)IVQUANT_SIZE/2 - 1 < (unsigned int)IVQUANT_SIZE - 1) | |
479 return ff_aac_ivquant_tab[a + IVQUANT_SIZE/2 - 1]; | |
480 else | |
481 return cbrtf(fabsf(a)) * a; | |
482 } | |
483 | |
7539 | 484 /** |
485 * Decode band types (section_data payload); reference: table 4.46. | |
486 * | |
487 * @param band_type array of the used band type | |
488 * @param band_type_run_end array of the last scalefactor band of a band type run | |
489 * | |
490 * @return Returns error status. 0 - OK, !0 - error | |
491 */ | |
492 static int decode_band_types(AACContext * ac, enum BandType band_type[120], | |
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493 int band_type_run_end[120], GetBitContext * gb, IndividualChannelStream * ics) { |
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494 int g, idx = 0; |
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495 const int bits = (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) ? 3 : 5; |
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496 for (g = 0; g < ics->num_window_groups; g++) { |
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497 int k = 0; |
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498 while (k < ics->max_sfb) { |
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499 uint8_t sect_len = k; |
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500 int sect_len_incr; |
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501 int sect_band_type = get_bits(gb, 4); |
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502 if (sect_band_type == 12) { |
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503 av_log(ac->avccontext, AV_LOG_ERROR, "invalid band type\n"); |
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504 return -1; |
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505 } |
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506 while ((sect_len_incr = get_bits(gb, bits)) == (1 << bits)-1) |
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507 sect_len += sect_len_incr; |
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508 sect_len += sect_len_incr; |
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509 if (sect_len > ics->max_sfb) { |
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510 av_log(ac->avccontext, AV_LOG_ERROR, |
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511 "Number of bands (%d) exceeds limit (%d).\n", |
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512 sect_len, ics->max_sfb); |
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513 return -1; |
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514 } |
7539 | 515 } |
516 } | |
517 return 0; | |
518 } | |
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519 |
7539 | 520 /** |
521 * Decode scalefactors; reference: table 4.47. | |
7523
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522 * |
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523 * @param global_gain first scalefactor value as scalefactors are differentially coded |
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524 * @param band_type array of the used band type |
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525 * @param band_type_run_end array of the last scalefactor band of a band type run |
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526 * @param sf array of scalefactors or intensity stereo positions |
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527 * |
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528 * @return Returns error status. 0 - OK, !0 - error |
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529 */ |
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530 static int decode_scalefactors(AACContext * ac, float sf[120], GetBitContext * gb, |
7540 | 531 unsigned int global_gain, IndividualChannelStream * ics, |
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532 enum BandType band_type[120], int band_type_run_end[120]) { |
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533 const int sf_offset = ac->sf_offset + (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE ? 12 : 0); |
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534 int g, i, idx = 0; |
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535 int offset[3] = { global_gain, global_gain - 90, 100 }; |
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536 int noise_flag = 1; |
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537 static const char *sf_str[3] = { "Global gain", "Noise gain", "Intensity stereo position" }; |
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538 ics->intensity_present = 0; |
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539 for (g = 0; g < ics->num_window_groups; g++) { |
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540 for (i = 0; i < ics->max_sfb;) { |
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541 int run_end = band_type_run_end[idx]; |
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542 if (band_type[idx] == ZERO_BT) { |
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543 for(; i < run_end; i++, idx++) |
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544 sf[idx] = 0.; |
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545 }else if((band_type[idx] == INTENSITY_BT) || (band_type[idx] == INTENSITY_BT2)) { |
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546 ics->intensity_present = 1; |
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547 for(; i < run_end; i++, idx++) { |
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548 offset[2] += get_vlc2(gb, vlc_scalefactors.table, 7, 3) - 60; |
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549 if(offset[2] > 255U) { |
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550 av_log(ac->avccontext, AV_LOG_ERROR, |
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551 "%s (%d) out of range.\n", sf_str[2], offset[2]); |
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552 return -1; |
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553 } |
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554 sf[idx] = ff_aac_pow2sf_tab[-offset[2] + 300]; |
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555 } |
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556 }else if(band_type[idx] == NOISE_BT) { |
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557 for(; i < run_end; i++, idx++) { |
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558 if(noise_flag-- > 0) |
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559 offset[1] += get_bits(gb, 9) - 256; |
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560 else |
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561 offset[1] += get_vlc2(gb, vlc_scalefactors.table, 7, 3) - 60; |
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562 if(offset[1] > 255U) { |
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563 av_log(ac->avccontext, AV_LOG_ERROR, |
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564 "%s (%d) out of range.\n", sf_str[1], offset[1]); |
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565 return -1; |
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566 } |
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567 sf[idx] = -ff_aac_pow2sf_tab[ offset[1] + sf_offset]; |
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568 } |
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569 }else { |
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570 for(; i < run_end; i++, idx++) { |
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571 offset[0] += get_vlc2(gb, vlc_scalefactors.table, 7, 3) - 60; |
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572 if(offset[0] > 255U) { |
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573 av_log(ac->avccontext, AV_LOG_ERROR, |
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574 "%s (%d) out of range.\n", sf_str[0], offset[0]); |
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575 return -1; |
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576 } |
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577 sf[idx] = -ff_aac_pow2sf_tab[ offset[0] + sf_offset]; |
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578 } |
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579 } |
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580 } |
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581 } |
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582 return 0; |
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583 } |
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584 |
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585 /** |
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586 * Decode pulse data; reference: table 4.7. |
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587 */ |
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588 static void decode_pulses(Pulse * pulse, GetBitContext * gb) { |
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589 int i; |
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590 pulse->num_pulse = get_bits(gb, 2) + 1; |
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591 pulse->start = get_bits(gb, 6); |
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592 for (i = 0; i < pulse->num_pulse; i++) { |
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593 pulse->offset[i] = get_bits(gb, 5); |
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594 pulse->amp [i] = get_bits(gb, 4); |
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595 } |
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596 } |
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597 |
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598 /** |
7539 | 599 * Decode Mid/Side data; reference: table 4.54. |
600 * | |
601 * @param ms_present Indicates mid/side stereo presence. [0] mask is all 0s; | |
602 * [1] mask is decoded from bitstream; [2] mask is all 1s; | |
603 * [3] reserved for scalable AAC | |
604 */ | |
605 static void decode_mid_side_stereo(ChannelElement * cpe, GetBitContext * gb, | |
606 int ms_present) { | |
7578 | 607 int idx; |
608 if (ms_present == 1) { | |
609 for (idx = 0; idx < cpe->ch[0].ics.num_window_groups * cpe->ch[0].ics.max_sfb; idx++) | |
610 cpe->ms_mask[idx] = get_bits1(gb); | |
611 } else if (ms_present == 2) { | |
612 memset(cpe->ms_mask, 1, cpe->ch[0].ics.num_window_groups * cpe->ch[0].ics.max_sfb * sizeof(cpe->ms_mask[0])); | |
613 } | |
614 } | |
7539 | 615 |
616 /** | |
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617 * Add pulses with particular amplitudes to the quantized spectral data; reference: 4.6.3.3. |
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618 * |
7501 | 619 * @param pulse pointer to pulse data struct |
620 * @param icoef array of quantized spectral data | |
621 */ | |
622 static void add_pulses(int icoef[1024], const Pulse * pulse, const IndividualChannelStream * ics) { | |
623 int i, off = ics->swb_offset[pulse->start]; | |
624 for (i = 0; i < pulse->num_pulse; i++) { | |
625 int ic; | |
626 off += pulse->offset[i]; | |
627 ic = (icoef[off] - 1)>>31; | |
628 icoef[off] += (pulse->amp[i]^ic) - ic; | |
629 } | |
630 } | |
631 | |
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632 /** |
7539 | 633 * Decode an individual_channel_stream payload; reference: table 4.44. |
634 * | |
635 * @param common_window Channels have independent [0], or shared [1], Individual Channel Stream information. | |
636 * @param scale_flag scalable [1] or non-scalable [0] AAC (Unused until scalable AAC is implemented.) | |
637 * | |
638 * @return Returns error status. 0 - OK, !0 - error | |
639 */ | |
640 static int decode_ics(AACContext * ac, SingleChannelElement * sce, GetBitContext * gb, int common_window, int scale_flag) { | |
641 int icoeffs[1024]; | |
642 Pulse pulse; | |
643 TemporalNoiseShaping * tns = &sce->tns; | |
644 IndividualChannelStream * ics = &sce->ics; | |
645 float * out = sce->coeffs; | |
646 int global_gain, pulse_present = 0; | |
647 | |
648 /* These two assignments are to silence some GCC warnings about the | |
649 * variables being used uninitialised when in fact they always are. | |
650 */ | |
651 pulse.num_pulse = 0; | |
652 pulse.start = 0; | |
653 | |
654 global_gain = get_bits(gb, 8); | |
655 | |
656 if (!common_window && !scale_flag) { | |
657 if (decode_ics_info(ac, ics, gb, 0) < 0) | |
658 return -1; | |
659 } | |
660 | |
661 if (decode_band_types(ac, sce->band_type, sce->band_type_run_end, gb, ics) < 0) | |
662 return -1; | |
663 if (decode_scalefactors(ac, sce->sf, gb, global_gain, ics, sce->band_type, sce->band_type_run_end) < 0) | |
664 return -1; | |
665 | |
666 pulse_present = 0; | |
667 if (!scale_flag) { | |
668 if ((pulse_present = get_bits1(gb))) { | |
669 if (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) { | |
670 av_log(ac->avccontext, AV_LOG_ERROR, "Pulse tool not allowed in eight short sequence.\n"); | |
671 return -1; | |
672 } | |
673 decode_pulses(&pulse, gb); | |
674 } | |
675 if ((tns->present = get_bits1(gb)) && decode_tns(ac, tns, gb, ics)) | |
676 return -1; | |
677 if (get_bits1(gb)) { | |
678 av_log_missing_feature(ac->avccontext, "SSR", 1); | |
679 return -1; | |
680 } | |
681 } | |
682 | |
683 if (decode_spectrum(ac, icoeffs, gb, ics, sce->band_type) < 0) | |
684 return -1; | |
685 if (pulse_present) | |
686 add_pulses(icoeffs, &pulse, ics); | |
687 dequant(ac, out, icoeffs, sce->sf, ics, sce->band_type); | |
688 return 0; | |
689 } | |
690 | |
691 /** | |
692 * Decode a channel_pair_element; reference: table 4.4. | |
693 * | |
694 * @param elem_id Identifies the instance of a syntax element. | |
695 * | |
696 * @return Returns error status. 0 - OK, !0 - error | |
697 */ | |
698 static int decode_cpe(AACContext * ac, GetBitContext * gb, int elem_id) { | |
699 int i, ret, common_window, ms_present = 0; | |
700 ChannelElement * cpe; | |
701 | |
702 cpe = ac->che[TYPE_CPE][elem_id]; | |
703 common_window = get_bits1(gb); | |
704 if (common_window) { | |
705 if (decode_ics_info(ac, &cpe->ch[0].ics, gb, 1)) | |
706 return -1; | |
707 i = cpe->ch[1].ics.use_kb_window[0]; | |
708 cpe->ch[1].ics = cpe->ch[0].ics; | |
709 cpe->ch[1].ics.use_kb_window[1] = i; | |
710 ms_present = get_bits(gb, 2); | |
711 if(ms_present == 3) { | |
712 av_log(ac->avccontext, AV_LOG_ERROR, "ms_present = 3 is reserved.\n"); | |
713 return -1; | |
714 } else if(ms_present) | |
715 decode_mid_side_stereo(cpe, gb, ms_present); | |
716 } | |
717 if ((ret = decode_ics(ac, &cpe->ch[0], gb, common_window, 0))) | |
718 return ret; | |
719 if ((ret = decode_ics(ac, &cpe->ch[1], gb, common_window, 0))) | |
720 return ret; | |
721 | |
722 if (common_window && ms_present) | |
723 apply_mid_side_stereo(cpe); | |
724 | |
725 if (cpe->ch[1].ics.intensity_present) | |
726 apply_intensity_stereo(cpe, ms_present); | |
727 return 0; | |
728 } | |
729 | |
7578 | 730 coup->coupling_point = 2*get_bits1(gb); |
731 coup->num_coupled = get_bits(gb, 3); | |
732 for (c = 0; c <= coup->num_coupled; c++) { | |
733 num_gain++; | |
734 coup->type[c] = get_bits1(gb) ? TYPE_CPE : TYPE_SCE; | |
735 coup->id_select[c] = get_bits(gb, 4); | |
736 if (coup->type[c] == TYPE_CPE) { | |
737 coup->ch_select[c] = get_bits(gb, 2); | |
738 if (coup->ch_select[c] == 3) | |
739 num_gain++; | |
740 } else | |
741 coup->ch_select[c] = 1; | |
742 } | |
743 coup->coupling_point += get_bits1(gb); | |
744 | |
745 if (coup->coupling_point == 2) { | |
746 av_log(ac->avccontext, AV_LOG_ERROR, | |
747 "Independently switched CCE with 'invalid' domain signalled.\n"); | |
748 memset(coup, 0, sizeof(ChannelCoupling)); | |
749 return -1; | |
750 } | |
751 | |
752 sign = get_bits(gb, 1); | |
753 scale = pow(2., pow(2., get_bits(gb, 2) - 3)); | |
754 | |
755 if ((ret = decode_ics(ac, sce, gb, 0, 0))) | |
756 return ret; | |
757 | |
758 for (c = 0; c < num_gain; c++) { | |
759 int cge = 1; | |
760 int gain = 0; | |
761 float gain_cache = 1.; | |
762 if (c) { | |
763 cge = coup->coupling_point == AFTER_IMDCT ? 1 : get_bits1(gb); | |
764 gain = cge ? get_vlc2(gb, vlc_scalefactors.table, 7, 3) - 60: 0; | |
765 gain_cache = pow(scale, gain); | |
766 } | |
767 for (g = 0; g < sce->ics.num_window_groups; g++) | |
768 for (sfb = 0; sfb < sce->ics.max_sfb; sfb++, idx++) | |
769 if (sce->band_type[idx] != ZERO_BT) { | |
770 if (!cge) { | |
771 int t = get_vlc2(gb, vlc_scalefactors.table, 7, 3) - 60; | |
772 if (t) { | |
773 int s = 1; | |
774 if (sign) { | |
775 s -= 2 * (t & 0x1); | |
776 t >>= 1; | |
777 } | |
778 gain += t; | |
779 gain_cache = pow(scale, gain) * s; | |
780 } | |
781 } | |
782 coup->gain[c][idx] = gain_cache; | |
783 } | |
784 } | |
785 return 0; | |
786 } | |
787 | |
7539 | 788 /** |
789 * Decode Spectral Band Replication extension data; reference: table 4.55. | |
7523
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790 * |
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791 * @param crc flag indicating the presence of CRC checksum |
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792 * @param cnt length of TYPE_FIL syntactic element in bytes |
7539 | 793 * |
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794 * @return Returns number of bytes consumed from the TYPE_FIL element. |
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795 */ |
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796 static int decode_sbr_extension(AACContext * ac, GetBitContext * gb, int crc, int cnt) { |
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797 // TODO : sbr_extension implementation |
7540 | 798 av_log_missing_feature(ac->avccontext, "SBR", 0); |
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799 skip_bits_long(gb, 8*cnt - 4); // -4 due to reading extension type |
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800 return cnt; |
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801 } |
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802 |
7539 | 803 /** |
7578 | 804 * Parse whether channels are to be excluded from Dynamic Range Compression; reference: table 4.53. |
805 * | |
806 * @return Returns number of bytes consumed. | |
807 */ | |
808 static int decode_drc_channel_exclusions(DynamicRangeControl *che_drc, GetBitContext * gb) { | |
809 int i; | |
810 int num_excl_chan = 0; | |
811 | |
812 do { | |
813 for (i = 0; i < 7; i++) | |
814 che_drc->exclude_mask[num_excl_chan++] = get_bits1(gb); | |
815 } while (num_excl_chan < MAX_CHANNELS - 7 && get_bits1(gb)); | |
816 | |
817 return num_excl_chan / 7; | |
818 } | |
819 | |
820 /** | |
7539 | 821 * Decode dynamic range information; reference: table 4.52. |
822 * | |
823 * @param cnt length of TYPE_FIL syntactic element in bytes | |
824 * | |
825 * @return Returns number of bytes consumed. | |
826 */ | |
827 static int decode_dynamic_range(DynamicRangeControl *che_drc, GetBitContext * gb, int cnt) { | |
828 int n = 1; | |
829 int drc_num_bands = 1; | |
830 int i; | |
831 | |
832 /* pce_tag_present? */ | |
833 if(get_bits1(gb)) { | |
834 che_drc->pce_instance_tag = get_bits(gb, 4); | |
835 skip_bits(gb, 4); // tag_reserved_bits | |
836 n++; | |
837 } | |
838 | |
839 /* excluded_chns_present? */ | |
840 if(get_bits1(gb)) { | |
841 n += decode_drc_channel_exclusions(che_drc, gb); | |
842 } | |
843 | |
844 /* drc_bands_present? */ | |
845 if (get_bits1(gb)) { | |
846 che_drc->band_incr = get_bits(gb, 4); | |
847 che_drc->interpolation_scheme = get_bits(gb, 4); | |
848 n++; | |
849 drc_num_bands += che_drc->band_incr; | |
850 for (i = 0; i < drc_num_bands; i++) { | |
851 che_drc->band_top[i] = get_bits(gb, 8); | |
852 n++; | |
853 } | |
854 } | |
855 | |
856 /* prog_ref_level_present? */ | |
857 if (get_bits1(gb)) { | |
858 che_drc->prog_ref_level = get_bits(gb, 7); | |
859 skip_bits1(gb); // prog_ref_level_reserved_bits | |
860 n++; | |
861 } | |
862 | |
863 for (i = 0; i < drc_num_bands; i++) { | |
864 che_drc->dyn_rng_sgn[i] = get_bits1(gb); | |
865 che_drc->dyn_rng_ctl[i] = get_bits(gb, 7); | |
866 n++; | |
867 } | |
868 | |
869 return n; | |
870 } | |
871 | |
872 /** | |
873 * Decode extension data (incomplete); reference: table 4.51. | |
874 * | |
875 * @param cnt length of TYPE_FIL syntactic element in bytes | |
876 * | |
877 * @return Returns number of bytes consumed | |
878 */ | |
879 static int decode_extension_payload(AACContext * ac, GetBitContext * gb, int cnt) { | |
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880 int crc_flag = 0; |
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881 int res = cnt; |
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882 switch (get_bits(gb, 4)) { // extension type |
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883 case EXT_SBR_DATA_CRC: |
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884 crc_flag++; |
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885 case EXT_SBR_DATA: |
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886 res = decode_sbr_extension(ac, gb, crc_flag, cnt); |
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887 break; |
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888 case EXT_DYNAMIC_RANGE: |
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889 res = decode_dynamic_range(&ac->che_drc, gb, cnt); |
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890 break; |
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891 case EXT_FILL: |
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892 case EXT_FILL_DATA: |
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893 case EXT_DATA_ELEMENT: |
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894 default: |
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895 skip_bits_long(gb, 8*cnt - 4); |
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896 break; |
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897 }; |
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898 return res; |
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899 } |
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900 |
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901 /** |
7539 | 902 * Conduct IMDCT and windowing. |
903 */ | |
904 static void imdct_and_windowing(AACContext * ac, SingleChannelElement * sce) { | |
905 IndividualChannelStream * ics = &sce->ics; | |
906 float * in = sce->coeffs; | |
907 float * out = sce->ret; | |
908 float * saved = sce->saved; | |
909 const float * lwindow = ics->use_kb_window[0] ? ff_aac_kbd_long_1024 : ff_aac_sine_long_1024; | |
910 const float * swindow = ics->use_kb_window[0] ? ff_aac_kbd_short_128 : ff_aac_sine_short_128; | |
911 const float * lwindow_prev = ics->use_kb_window[1] ? ff_aac_kbd_long_1024 : ff_aac_sine_long_1024; | |
912 const float * swindow_prev = ics->use_kb_window[1] ? ff_aac_kbd_short_128 : ff_aac_sine_short_128; | |
913 float * buf = ac->buf_mdct; | |
914 int i; | |
915 | |
7578 | 916 if (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) { |
917 if (ics->window_sequence[1] == ONLY_LONG_SEQUENCE || ics->window_sequence[1] == LONG_STOP_SEQUENCE) | |
918 av_log(ac->avccontext, AV_LOG_WARNING, | |
919 "Transition from an ONLY_LONG or LONG_STOP to an EIGHT_SHORT sequence detected. " | |
920 "If you heard an audible artifact, please submit the sample to the FFmpeg developers.\n"); | |
921 for (i = 0; i < 2048; i += 256) { | |
922 ff_imdct_calc(&ac->mdct_small, buf + i, in + i/2); | |
923 ac->dsp.vector_fmul_reverse(ac->revers + i/2, buf + i + 128, swindow, 128); | |
924 } | |
925 for (i = 0; i < 448; i++) out[i] = saved[i] + ac->add_bias; | |
926 | |
927 ac->dsp.vector_fmul_add_add(out + 448 + 0*128, buf + 0*128, swindow_prev, saved + 448 , ac->add_bias, 128, 1); | |
928 ac->dsp.vector_fmul_add_add(out + 448 + 1*128, buf + 2*128, swindow, ac->revers + 0*128, ac->add_bias, 128, 1); | |
929 ac->dsp.vector_fmul_add_add(out + 448 + 2*128, buf + 4*128, swindow, ac->revers + 1*128, ac->add_bias, 128, 1); | |
930 ac->dsp.vector_fmul_add_add(out + 448 + 3*128, buf + 6*128, swindow, ac->revers + 2*128, ac->add_bias, 128, 1); | |
931 ac->dsp.vector_fmul_add_add(out + 448 + 4*128, buf + 8*128, swindow, ac->revers + 3*128, ac->add_bias, 64, 1); | |
932 | |
933 #if 0 | |
934 vector_fmul_add_add_add(&ac->dsp, out + 448 + 1*128, buf + 2*128, swindow, saved + 448 + 1*128, ac->revers + 0*128, ac->add_bias, 128); | |
935 vector_fmul_add_add_add(&ac->dsp, out + 448 + 2*128, buf + 4*128, swindow, saved + 448 + 2*128, ac->revers + 1*128, ac->add_bias, 128); | |
936 vector_fmul_add_add_add(&ac->dsp, out + 448 + 3*128, buf + 6*128, swindow, saved + 448 + 3*128, ac->revers + 2*128, ac->add_bias, 128); | |
937 vector_fmul_add_add_add(&ac->dsp, out + 448 + 4*128, buf + 8*128, swindow, saved + 448 + 4*128, ac->revers + 3*128, ac->add_bias, 64); | |
938 #endif | |
939 | |
940 ac->dsp.vector_fmul_add_add(saved, buf + 1024 + 64, swindow + 64, ac->revers + 3*128+64, 0, 64, 1); | |
941 ac->dsp.vector_fmul_add_add(saved + 64, buf + 1024 + 2*128, swindow, ac->revers + 4*128, 0, 128, 1); | |
942 ac->dsp.vector_fmul_add_add(saved + 192, buf + 1024 + 4*128, swindow, ac->revers + 5*128, 0, 128, 1); | |
943 ac->dsp.vector_fmul_add_add(saved + 320, buf + 1024 + 6*128, swindow, ac->revers + 6*128, 0, 128, 1); | |
944 memcpy( saved + 448, ac->revers + 7*128, 128 * sizeof(float)); | |
945 memset( saved + 576, 0, 448 * sizeof(float)); | |
946 } else { | |
947 ff_imdct_calc(&ac->mdct, buf, in); | |
948 if (ics->window_sequence[0] == LONG_STOP_SEQUENCE) { | |
949 for (i = 0; i < 448; i++) out[i] = saved[i] + ac->add_bias; | |
950 ac->dsp.vector_fmul_add_add(out + 448, buf + 448, swindow_prev, saved + 448, ac->add_bias, 128, 1); | |
951 for (i = 576; i < 1024; i++) out[i] = buf[i] + saved[i] + ac->add_bias; | |
952 } else { | |
953 ac->dsp.vector_fmul_add_add(out, buf, lwindow_prev, saved, ac->add_bias, 1024, 1); | |
954 } | |
955 if (ics->window_sequence[0] == LONG_START_SEQUENCE) { | |
956 memcpy(saved, buf + 1024, 448 * sizeof(float)); | |
957 ac->dsp.vector_fmul_reverse(saved + 448, buf + 1024 + 448, swindow, 128); | |
958 memset(saved + 576, 0, 448 * sizeof(float)); | |
959 } else { | |
960 ac->dsp.vector_fmul_reverse(saved, buf + 1024, lwindow, 1024); | |
961 } | |
962 } | |
963 } | |
964 | |
7539 | 965 /** |
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966 * Apply dependent channel coupling (applied before IMDCT). |
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967 * |
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968 * @param index index into coupling gain array |
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969 */ |
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970 static void apply_dependent_coupling(AACContext * ac, SingleChannelElement * sce, ChannelElement * cc, int index) { |
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971 IndividualChannelStream * ics = &cc->ch[0].ics; |
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972 const uint16_t * offsets = ics->swb_offset; |
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973 float * dest = sce->coeffs; |
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974 const float * src = cc->ch[0].coeffs; |
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975 int g, i, group, k, idx = 0; |
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976 if(ac->m4ac.object_type == AOT_AAC_LTP) { |
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977 av_log(ac->avccontext, AV_LOG_ERROR, |
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978 "Dependent coupling is not supported together with LTP\n"); |
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979 return; |
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980 } |
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981 for (g = 0; g < ics->num_window_groups; g++) { |
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982 for (i = 0; i < ics->max_sfb; i++, idx++) { |
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983 if (cc->ch[0].band_type[idx] != ZERO_BT) { |
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984 for (group = 0; group < ics->group_len[g]; group++) { |
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985 for (k = offsets[i]; k < offsets[i+1]; k++) { |
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986 // XXX dsputil-ize |
7540 | 987 dest[group*128+k] += cc->coup.gain[index][idx] * src[group*128+k]; |
7523
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988 } |
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989 } |
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990 } |
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991 } |
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992 dest += ics->group_len[g]*128; |
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993 src += ics->group_len[g]*128; |
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994 } |
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995 } |
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996 |
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997 /** |
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998 * Apply independent channel coupling (applied after IMDCT). |
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999 * |
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1000 * @param index index into coupling gain array |
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1001 */ |
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1002 static void apply_independent_coupling(AACContext * ac, SingleChannelElement * sce, ChannelElement * cc, int index) { |
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1003 int i; |
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1004 for (i = 0; i < 1024; i++) |
7540 | 1005 sce->ret[i] += cc->coup.gain[index][0] * (cc->ch[0].ret[i] - ac->add_bias); |
7523
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1006 } |
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1007 |
7578 | 1008 } |
1009 } | |
1010 } | |
1011 } | |
1012 | |
1013 static int aac_decode_frame(AVCodecContext * avccontext, void * data, int * data_size, const uint8_t * buf, int buf_size) { | |
1014 AACContext * ac = avccontext->priv_data; | |
1015 GetBitContext gb; | |
1016 enum RawDataBlockType elem_type; | |
1017 int err, elem_id, data_size_tmp; | |
1018 | |
1019 init_get_bits(&gb, buf, buf_size*8); | |
1020 | |
1021 // parse | |
1022 while ((elem_type = get_bits(&gb, 3)) != TYPE_END) { | |
1023 elem_id = get_bits(&gb, 4); | |
1024 err = -1; | |
1025 | |
1026 if(elem_type == TYPE_SCE && elem_id == 1 && | |
1027 !ac->che[TYPE_SCE][elem_id] && ac->che[TYPE_LFE][0]) { | |
1028 /* Some streams incorrectly code 5.1 audio as SCE[0] CPE[0] CPE[1] SCE[1] | |
1029 instead of SCE[0] CPE[0] CPE[0] LFE[0]. If we seem to have | |
1030 encountered such a stream, transfer the LFE[0] element to SCE[1] */ | |
1031 ac->che[TYPE_SCE][elem_id] = ac->che[TYPE_LFE][0]; | |
1032 ac->che[TYPE_LFE][0] = NULL; | |
1033 } | |
1034 if(elem_type && elem_type < TYPE_DSE) { | |
1035 if(!ac->che[elem_type][elem_id]) | |
1036 return -1; | |
1037 if(elem_type != TYPE_CCE) | |
1038 ac->che[elem_type][elem_id]->coup.coupling_point = 4; | |
1039 } | |
1040 | |
1041 switch (elem_type) { | |
1042 | |
1043 case TYPE_SCE: | |
1044 err = decode_ics(ac, &ac->che[TYPE_SCE][elem_id]->ch[0], &gb, 0, 0); | |
1045 break; | |
1046 | |
1047 case TYPE_CPE: | |
1048 err = decode_cpe(ac, &gb, elem_id); | |
1049 break; | |
1050 | |
1051 case TYPE_CCE: | |
1052 err = decode_cce(ac, &gb, ac->che[TYPE_SCE][elem_id]); | |
1053 break; | |
1054 | |
1055 case TYPE_LFE: | |
1056 err = decode_ics(ac, &ac->che[TYPE_LFE][elem_id]->ch[0], &gb, 0, 0); | |
1057 break; | |
1058 | |
1059 case TYPE_DSE: | |
1060 skip_data_stream_element(&gb); | |
1061 err = 0; | |
1062 break; | |
1063 | |
1064 case TYPE_PCE: | |
1065 { | |
1066 enum ChannelPosition new_che_pos[4][MAX_ELEM_ID]; | |
1067 memset(new_che_pos, 0, 4 * MAX_ELEM_ID * sizeof(new_che_pos[0][0])); | |
1068 if((err = decode_pce(ac, new_che_pos, &gb))) | |
1069 break; | |
1070 err = output_configure(ac, ac->che_pos, new_che_pos); | |
1071 break; | |
1072 } | |
1073 | |
1074 case TYPE_FIL: | |
1075 if (elem_id == 15) | |
1076 elem_id += get_bits(&gb, 8) - 1; | |
1077 while (elem_id > 0) | |
1078 elem_id -= decode_extension_payload(ac, &gb, elem_id); | |
1079 err = 0; /* FIXME */ | |
1080 break; | |
1081 | |
1082 default: | |
1083 err = -1; /* should not happen, but keeps compiler happy */ | |
1084 break; | |
1085 } | |
1086 | |
1087 if(err) | |
1088 return err; | |
1089 } | |
1090 | |
1091 spectral_to_sample(ac); | |
1092 | |
7539 | 1093 if (!ac->is_saved) { |
1094 ac->is_saved = 1; | |
1095 *data_size = 0; | |
1096 return 0; | |
1097 } | |
1098 | |
1099 data_size_tmp = 1024 * avccontext->channels * sizeof(int16_t); | |
1100 if(*data_size < data_size_tmp) { | |
1101 av_log(avccontext, AV_LOG_ERROR, | |
1102 "Output buffer too small (%d) or trying to output too many samples (%d) for this frame.\n", | |
1103 *data_size, data_size_tmp); | |
1104 return -1; | |
1105 } | |
1106 *data_size = data_size_tmp; | |
1107 | |
1108 ac->dsp.float_to_int16_interleave(data, (const float **)ac->output_data, 1024, avccontext->channels); | |
1109 | |
1110 return buf_size; | |
1111 } | |
1112 | |
7501 | 1113 static av_cold int aac_decode_close(AVCodecContext * avccontext) { |
1114 AACContext * ac = avccontext->priv_data; | |
7540 | 1115 int i, type; |
7501 | 1116 |
7523
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1117 for (i = 0; i < MAX_ELEM_ID; i++) { |
7540 | 1118 for(type = 0; type < 4; type++) |
1119 av_freep(&ac->che[type][i]); | |
7501 | 1120 } |
1121 | |
1122 ff_mdct_end(&ac->mdct); | |
1123 ff_mdct_end(&ac->mdct_small); | |
1124 return 0 ; | |
1125 } | |
1126 | |
1127 AVCodec aac_decoder = { | |
1128 "aac", | |
1129 CODEC_TYPE_AUDIO, | |
1130 CODEC_ID_AAC, | |
1131 sizeof(AACContext), | |
1132 aac_decode_init, | |
1133 NULL, | |
1134 aac_decode_close, | |
1135 aac_decode_frame, | |
1136 .long_name = NULL_IF_CONFIG_SMALL("Advanced Audio Coding"), | |
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1137 .sample_fmts = (enum SampleFormat[]){SAMPLE_FMT_S16,SAMPLE_FMT_NONE}, |
7501 | 1138 }; |