Mercurial > libavcodec.hg
annotate aac.c @ 7580:c49ab52db74c libavcodec
Synchronise AAC decoder code with that from SoC
author | superdump |
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date | Fri, 15 Aug 2008 00:19:14 +0000 |
parents | a05954c505ab |
children | 6fdffa4836a7 |
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 | |
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94 static float ff_aac_pow2sf_tab[316]; |
7501 | 95 #endif /* CONFIG_HARDCODED_TABLES */ |
96 | |
97 static VLC vlc_scalefactors; | |
98 static VLC vlc_spectral[11]; | |
99 | |
100 | |
7539 | 101 /** |
7578 | 102 * Configure output channel order based on the current program configuration element. |
103 * | |
104 * @param che_pos current channel position configuration | |
105 * @param new_che_pos New channel position configuration - we only do something if it differs from the current one. | |
106 * | |
107 * @return Returns error status. 0 - OK, !0 - error | |
108 */ | |
109 static int output_configure(AACContext *ac, enum ChannelPosition che_pos[4][MAX_ELEM_ID], | |
110 enum ChannelPosition new_che_pos[4][MAX_ELEM_ID]) { | |
111 AVCodecContext *avctx = ac->avccontext; | |
112 int i, type, channels = 0; | |
113 | |
114 if(!memcmp(che_pos, new_che_pos, 4 * MAX_ELEM_ID * sizeof(new_che_pos[0][0]))) | |
115 return 0; /* no change */ | |
116 | |
117 memcpy(che_pos, new_che_pos, 4 * MAX_ELEM_ID * sizeof(new_che_pos[0][0])); | |
118 | |
119 /* Allocate or free elements depending on if they are in the | |
120 * current program configuration. | |
121 * | |
122 * Set up default 1:1 output mapping. | |
123 * | |
124 * For a 5.1 stream the output order will be: | |
125 * [ Front Left ] [ Front Right ] [ Center ] [ LFE ] [ Surround Left ] [ Surround Right ] | |
126 */ | |
127 | |
128 for(i = 0; i < MAX_ELEM_ID; i++) { | |
129 for(type = 0; type < 4; type++) { | |
130 if(che_pos[type][i]) { | |
131 if(!ac->che[type][i] && !(ac->che[type][i] = av_mallocz(sizeof(ChannelElement)))) | |
132 return AVERROR(ENOMEM); | |
133 if(type != TYPE_CCE) { | |
134 ac->output_data[channels++] = ac->che[type][i]->ch[0].ret; | |
135 if(type == TYPE_CPE) { | |
136 ac->output_data[channels++] = ac->che[type][i]->ch[1].ret; | |
137 } | |
138 } | |
139 } else | |
140 av_freep(&ac->che[type][i]); | |
141 } | |
142 } | |
143 | |
144 avctx->channels = channels; | |
145 return 0; | |
146 } | |
147 | |
148 /** | |
7539 | 149 * Decode an array of 4 bit element IDs, optionally interleaved with a stereo/mono switching bit. |
150 * | |
151 * @param cpe_map Stereo (Channel Pair Element) map, NULL if stereo bit is not present. | |
152 * @param sce_map mono (Single Channel Element) map | |
153 * @param type speaker type/position for these channels | |
154 */ | |
155 static void decode_channel_map(enum ChannelPosition *cpe_map, | |
156 enum ChannelPosition *sce_map, enum ChannelPosition type, GetBitContext * gb, int n) { | |
157 while(n--) { | |
158 enum ChannelPosition *map = cpe_map && get_bits1(gb) ? cpe_map : sce_map; // stereo or mono map | |
159 map[get_bits(gb, 4)] = type; | |
160 } | |
161 } | |
162 | |
163 /** | |
164 * Decode program configuration element; reference: table 4.2. | |
165 * | |
166 * @param new_che_pos New channel position configuration - we only do something if it differs from the current one. | |
167 * | |
168 * @return Returns error status. 0 - OK, !0 - error | |
169 */ | |
170 static int decode_pce(AACContext * ac, enum ChannelPosition new_che_pos[4][MAX_ELEM_ID], | |
171 GetBitContext * gb) { | |
172 int num_front, num_side, num_back, num_lfe, num_assoc_data, num_cc; | |
173 | |
174 skip_bits(gb, 2); // object_type | |
175 | |
176 ac->m4ac.sampling_index = get_bits(gb, 4); | |
177 if(ac->m4ac.sampling_index > 11) { | |
178 av_log(ac->avccontext, AV_LOG_ERROR, "invalid sampling rate index %d\n", ac->m4ac.sampling_index); | |
179 return -1; | |
180 } | |
181 ac->m4ac.sample_rate = ff_mpeg4audio_sample_rates[ac->m4ac.sampling_index]; | |
7501 | 182 num_front = get_bits(gb, 4); |
183 num_side = get_bits(gb, 4); | |
184 num_back = get_bits(gb, 4); | |
185 num_lfe = get_bits(gb, 2); | |
186 num_assoc_data = get_bits(gb, 3); | |
187 num_cc = get_bits(gb, 4); | |
188 | |
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189 if (get_bits1(gb)) |
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190 skip_bits(gb, 4); // mono_mixdown_tag |
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191 if (get_bits1(gb)) |
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192 skip_bits(gb, 4); // stereo_mixdown_tag |
7501 | 193 |
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194 if (get_bits1(gb)) |
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195 skip_bits(gb, 3); // mixdown_coeff_index and pseudo_surround |
7501 | 196 |
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197 decode_channel_map(new_che_pos[TYPE_CPE], new_che_pos[TYPE_SCE], AAC_CHANNEL_FRONT, gb, num_front); |
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198 decode_channel_map(new_che_pos[TYPE_CPE], new_che_pos[TYPE_SCE], AAC_CHANNEL_SIDE, gb, num_side ); |
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199 decode_channel_map(new_che_pos[TYPE_CPE], new_che_pos[TYPE_SCE], AAC_CHANNEL_BACK, gb, num_back ); |
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200 decode_channel_map(NULL, new_che_pos[TYPE_LFE], AAC_CHANNEL_LFE, gb, num_lfe ); |
7501 | 201 |
202 skip_bits_long(gb, 4 * num_assoc_data); | |
203 | |
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204 decode_channel_map(new_che_pos[TYPE_CCE], new_che_pos[TYPE_CCE], AAC_CHANNEL_CC, gb, num_cc ); |
7501 | 205 |
206 align_get_bits(gb); | |
207 | |
208 /* comment field, first byte is length */ | |
209 skip_bits_long(gb, 8 * get_bits(gb, 8)); | |
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210 return 0; |
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211 } |
7501 | 212 |
7539 | 213 /** |
214 * Set up channel positions based on a default channel configuration | |
215 * as specified in table 1.17. | |
216 * | |
217 * @param new_che_pos New channel position configuration - we only do something if it differs from the current one. | |
218 * | |
219 * @return Returns error status. 0 - OK, !0 - error | |
220 */ | |
221 static int set_default_channel_config(AACContext *ac, enum ChannelPosition new_che_pos[4][MAX_ELEM_ID], | |
222 int channel_config) | |
223 { | |
224 if(channel_config < 1 || channel_config > 7) { | |
225 av_log(ac->avccontext, AV_LOG_ERROR, "invalid default channel configuration (%d)\n", | |
226 channel_config); | |
227 return -1; | |
228 } | |
229 | |
230 /* default channel configurations: | |
231 * | |
232 * 1ch : front center (mono) | |
233 * 2ch : L + R (stereo) | |
234 * 3ch : front center + L + R | |
235 * 4ch : front center + L + R + back center | |
236 * 5ch : front center + L + R + back stereo | |
237 * 6ch : front center + L + R + back stereo + LFE | |
238 * 7ch : front center + L + R + outer front left + outer front right + back stereo + LFE | |
239 */ | |
240 | |
241 if(channel_config != 2) | |
242 new_che_pos[TYPE_SCE][0] = AAC_CHANNEL_FRONT; // front center (or mono) | |
243 if(channel_config > 1) | |
244 new_che_pos[TYPE_CPE][0] = AAC_CHANNEL_FRONT; // L + R (or stereo) | |
245 if(channel_config == 4) | |
246 new_che_pos[TYPE_SCE][1] = AAC_CHANNEL_BACK; // back center | |
247 if(channel_config > 4) | |
248 new_che_pos[TYPE_CPE][(channel_config == 7) + 1] | |
249 = AAC_CHANNEL_BACK; // back stereo | |
250 if(channel_config > 5) | |
251 new_che_pos[TYPE_LFE][0] = AAC_CHANNEL_LFE; // LFE | |
252 if(channel_config == 7) | |
253 new_che_pos[TYPE_CPE][1] = AAC_CHANNEL_FRONT; // outer front left + outer front right | |
254 | |
255 return 0; | |
256 } | |
257 | |
7578 | 258 /** |
259 * Decode GA "General Audio" specific configuration; reference: table 4.1. | |
260 * | |
261 * @return Returns error status. 0 - OK, !0 - error | |
262 */ | |
263 static int decode_ga_specific_config(AACContext * ac, GetBitContext * gb, int channel_config) { | |
264 enum ChannelPosition new_che_pos[4][MAX_ELEM_ID]; | |
265 int extension_flag, ret; | |
266 | |
267 if(get_bits1(gb)) { // frameLengthFlag | |
268 av_log_missing_feature(ac->avccontext, "960/120 MDCT window is", 1); | |
7539 | 269 return -1; |
270 } | |
271 | |
272 if (get_bits1(gb)) // dependsOnCoreCoder | |
273 skip_bits(gb, 14); // coreCoderDelay | |
274 extension_flag = get_bits1(gb); | |
275 | |
276 if(ac->m4ac.object_type == AOT_AAC_SCALABLE || | |
277 ac->m4ac.object_type == AOT_ER_AAC_SCALABLE) | |
278 skip_bits(gb, 3); // layerNr | |
279 | |
280 memset(new_che_pos, 0, 4 * MAX_ELEM_ID * sizeof(new_che_pos[0][0])); | |
281 if (channel_config == 0) { | |
282 skip_bits(gb, 4); // element_instance_tag | |
283 if((ret = decode_pce(ac, new_che_pos, gb))) | |
284 return ret; | |
285 } else { | |
286 if((ret = set_default_channel_config(ac, new_che_pos, channel_config))) | |
287 return ret; | |
288 } | |
289 if((ret = output_configure(ac, ac->che_pos, new_che_pos))) | |
290 return ret; | |
291 | |
292 if (extension_flag) { | |
293 switch (ac->m4ac.object_type) { | |
294 case AOT_ER_BSAC: | |
295 skip_bits(gb, 5); // numOfSubFrame | |
296 skip_bits(gb, 11); // layer_length | |
297 break; | |
298 case AOT_ER_AAC_LC: | |
299 case AOT_ER_AAC_LTP: | |
300 case AOT_ER_AAC_SCALABLE: | |
301 case AOT_ER_AAC_LD: | |
302 skip_bits(gb, 3); /* aacSectionDataResilienceFlag | |
303 * aacScalefactorDataResilienceFlag | |
304 * aacSpectralDataResilienceFlag | |
305 */ | |
306 break; | |
307 } | |
308 skip_bits1(gb); // extensionFlag3 (TBD in version 3) | |
309 } | |
310 return 0; | |
311 } | |
312 | |
313 /** | |
314 * Decode audio specific configuration; reference: table 1.13. | |
315 * | |
316 * @param data pointer to AVCodecContext extradata | |
317 * @param data_size size of AVCCodecContext extradata | |
318 * | |
319 * @return Returns error status. 0 - OK, !0 - error | |
320 */ | |
321 static int decode_audio_specific_config(AACContext * ac, void *data, int data_size) { | |
322 GetBitContext gb; | |
323 int i; | |
324 | |
325 init_get_bits(&gb, data, data_size * 8); | |
326 | |
327 if((i = ff_mpeg4audio_get_config(&ac->m4ac, data, data_size)) < 0) | |
328 return -1; | |
329 if(ac->m4ac.sampling_index > 11) { | |
330 av_log(ac->avccontext, AV_LOG_ERROR, "invalid sampling rate index %d\n", ac->m4ac.sampling_index); | |
331 return -1; | |
332 } | |
333 | |
334 skip_bits_long(&gb, i); | |
335 | |
336 switch (ac->m4ac.object_type) { | |
337 case AOT_AAC_LC: | |
338 if (decode_ga_specific_config(ac, &gb, ac->m4ac.chan_config)) | |
339 return -1; | |
340 break; | |
341 default: | |
342 av_log(ac->avccontext, AV_LOG_ERROR, "Audio object type %s%d is not supported.\n", | |
343 ac->m4ac.sbr == 1? "SBR+" : "", ac->m4ac.object_type); | |
344 return -1; | |
345 } | |
346 return 0; | |
347 } | |
348 | |
7578 | 349 /** |
350 * linear congruential pseudorandom number generator | |
351 * | |
352 * @param previous_val pointer to the current state of the generator | |
353 * | |
354 * @return Returns a 32-bit pseudorandom integer | |
355 */ | |
356 static av_always_inline int lcg_random(int previous_val) { | |
357 return previous_val * 1664525 + 1013904223; | |
358 } | |
359 | |
7501 | 360 static av_cold int aac_decode_init(AVCodecContext * avccontext) { |
361 AACContext * ac = avccontext->priv_data; | |
362 int i; | |
363 | |
364 ac->avccontext = avccontext; | |
365 | |
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366 if (avccontext->extradata_size <= 0 || |
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367 decode_audio_specific_config(ac, avccontext->extradata, avccontext->extradata_size)) |
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368 return -1; |
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369 |
7539 | 370 avccontext->sample_fmt = SAMPLE_FMT_S16; |
7501 | 371 avccontext->sample_rate = ac->m4ac.sample_rate; |
372 avccontext->frame_size = 1024; | |
373 | |
374 AAC_INIT_VLC_STATIC( 0, 144); | |
375 AAC_INIT_VLC_STATIC( 1, 114); | |
376 AAC_INIT_VLC_STATIC( 2, 188); | |
377 AAC_INIT_VLC_STATIC( 3, 180); | |
378 AAC_INIT_VLC_STATIC( 4, 172); | |
379 AAC_INIT_VLC_STATIC( 5, 140); | |
380 AAC_INIT_VLC_STATIC( 6, 168); | |
381 AAC_INIT_VLC_STATIC( 7, 114); | |
382 AAC_INIT_VLC_STATIC( 8, 262); | |
383 AAC_INIT_VLC_STATIC( 9, 248); | |
384 AAC_INIT_VLC_STATIC(10, 384); | |
385 | |
386 dsputil_init(&ac->dsp, avccontext); | |
387 | |
7539 | 388 ac->random_state = 0x1f2e3d4c; |
389 | |
7501 | 390 // -1024 - Compensate wrong IMDCT method. |
391 // 32768 - Required to scale values to the correct range for the bias method | |
392 // for float to int16 conversion. | |
393 | |
394 if(ac->dsp.float_to_int16 == ff_float_to_int16_c) { | |
395 ac->add_bias = 385.0f; | |
396 ac->sf_scale = 1. / (-1024. * 32768.); | |
397 ac->sf_offset = 0; | |
398 } else { | |
399 ac->add_bias = 0.0f; | |
400 ac->sf_scale = 1. / -1024.; | |
401 ac->sf_offset = 60; | |
402 } | |
403 | |
404 #ifndef CONFIG_HARDCODED_TABLES | |
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405 for (i = 0; i < 316; i++) |
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406 ff_aac_pow2sf_tab[i] = pow(2, (i - 200)/4.); |
7501 | 407 #endif /* CONFIG_HARDCODED_TABLES */ |
408 | |
409 INIT_VLC_STATIC(&vlc_scalefactors, 7, sizeof(ff_aac_scalefactor_code)/sizeof(ff_aac_scalefactor_code[0]), | |
410 ff_aac_scalefactor_bits, sizeof(ff_aac_scalefactor_bits[0]), sizeof(ff_aac_scalefactor_bits[0]), | |
411 ff_aac_scalefactor_code, sizeof(ff_aac_scalefactor_code[0]), sizeof(ff_aac_scalefactor_code[0]), | |
412 352); | |
413 | |
414 ff_mdct_init(&ac->mdct, 11, 1); | |
415 ff_mdct_init(&ac->mdct_small, 8, 1); | |
416 return 0; | |
417 } | |
418 | |
7539 | 419 /** |
420 * Skip data_stream_element; reference: table 4.10. | |
421 */ | |
422 static void skip_data_stream_element(GetBitContext * gb) { | |
7501 | 423 int byte_align = get_bits1(gb); |
424 int count = get_bits(gb, 8); | |
425 if (count == 255) | |
426 count += get_bits(gb, 8); | |
427 if (byte_align) | |
428 align_get_bits(gb); | |
429 skip_bits_long(gb, 8 * count); | |
430 } | |
431 | |
432 /** | |
7539 | 433 * Decode Individual Channel Stream info; reference: table 4.6. |
434 * | |
435 * @param common_window Channels have independent [0], or shared [1], Individual Channel Stream information. | |
436 */ | |
437 static int decode_ics_info(AACContext * ac, IndividualChannelStream * ics, GetBitContext * gb, int common_window) { | |
438 if (get_bits1(gb)) { | |
439 av_log(ac->avccontext, AV_LOG_ERROR, "Reserved bit set.\n"); | |
440 memset(ics, 0, sizeof(IndividualChannelStream)); | |
441 return -1; | |
442 } | |
443 ics->window_sequence[1] = ics->window_sequence[0]; | |
444 ics->window_sequence[0] = get_bits(gb, 2); | |
445 ics->use_kb_window[1] = ics->use_kb_window[0]; | |
446 ics->use_kb_window[0] = get_bits1(gb); | |
447 ics->num_window_groups = 1; | |
448 ics->group_len[0] = 1; | |
449 | |
7578 | 450 if (get_bits1(gb)) { |
451 av_log_missing_feature(ac->avccontext, "Predictor bit set but LTP is", 1); | |
452 memset(ics, 0, sizeof(IndividualChannelStream)); | |
453 return -1; | |
454 } | |
455 } | |
456 | |
457 if(ics->max_sfb > ics->num_swb) { | |
458 av_log(ac->avccontext, AV_LOG_ERROR, | |
459 "Number of scalefactor bands in group (%d) exceeds limit (%d).\n", | |
460 ics->max_sfb, ics->num_swb); | |
461 memset(ics, 0, sizeof(IndividualChannelStream)); | |
462 return -1; | |
463 } | |
464 | |
7539 | 465 return 0; |
466 } | |
467 | |
468 /** | |
469 * Decode band types (section_data payload); reference: table 4.46. | |
470 * | |
471 * @param band_type array of the used band type | |
472 * @param band_type_run_end array of the last scalefactor band of a band type run | |
473 * | |
474 * @return Returns error status. 0 - OK, !0 - error | |
475 */ | |
476 static int decode_band_types(AACContext * ac, enum BandType band_type[120], | |
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477 int band_type_run_end[120], GetBitContext * gb, IndividualChannelStream * ics) { |
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478 int g, idx = 0; |
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479 const int bits = (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) ? 3 : 5; |
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480 for (g = 0; g < ics->num_window_groups; g++) { |
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481 int k = 0; |
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482 while (k < ics->max_sfb) { |
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483 uint8_t sect_len = k; |
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484 int sect_len_incr; |
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485 int sect_band_type = get_bits(gb, 4); |
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486 if (sect_band_type == 12) { |
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487 av_log(ac->avccontext, AV_LOG_ERROR, "invalid band type\n"); |
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488 return -1; |
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489 } |
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490 while ((sect_len_incr = get_bits(gb, bits)) == (1 << bits)-1) |
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491 sect_len += sect_len_incr; |
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492 sect_len += sect_len_incr; |
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493 if (sect_len > ics->max_sfb) { |
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494 av_log(ac->avccontext, AV_LOG_ERROR, |
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495 "Number of bands (%d) exceeds limit (%d).\n", |
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496 sect_len, ics->max_sfb); |
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497 return -1; |
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498 } |
7539 | 499 } |
500 } | |
501 return 0; | |
502 } | |
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503 |
7539 | 504 /** |
505 * Decode scalefactors; reference: table 4.47. | |
7523
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506 * |
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507 * @param global_gain first scalefactor value as scalefactors are differentially coded |
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508 * @param band_type array of the used band type |
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509 * @param band_type_run_end array of the last scalefactor band of a band type run |
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510 * @param sf array of scalefactors or intensity stereo positions |
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511 * |
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512 * @return Returns error status. 0 - OK, !0 - error |
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513 */ |
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514 static int decode_scalefactors(AACContext * ac, float sf[120], GetBitContext * gb, |
7540 | 515 unsigned int global_gain, IndividualChannelStream * ics, |
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516 enum BandType band_type[120], int band_type_run_end[120]) { |
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517 const int sf_offset = ac->sf_offset + (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE ? 12 : 0); |
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518 int g, i, idx = 0; |
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519 int offset[3] = { global_gain, global_gain - 90, 100 }; |
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520 int noise_flag = 1; |
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521 static const char *sf_str[3] = { "Global gain", "Noise gain", "Intensity stereo position" }; |
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522 for (g = 0; g < ics->num_window_groups; g++) { |
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523 for (i = 0; i < ics->max_sfb;) { |
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524 int run_end = band_type_run_end[idx]; |
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525 if (band_type[idx] == ZERO_BT) { |
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526 for(; i < run_end; i++, idx++) |
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527 sf[idx] = 0.; |
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528 }else if((band_type[idx] == INTENSITY_BT) || (band_type[idx] == INTENSITY_BT2)) { |
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529 for(; i < run_end; i++, idx++) { |
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530 offset[2] += get_vlc2(gb, vlc_scalefactors.table, 7, 3) - 60; |
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531 if(offset[2] > 255U) { |
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532 av_log(ac->avccontext, AV_LOG_ERROR, |
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533 "%s (%d) out of range.\n", sf_str[2], offset[2]); |
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534 return -1; |
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535 } |
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536 sf[idx] = ff_aac_pow2sf_tab[-offset[2] + 300]; |
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537 } |
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538 }else if(band_type[idx] == NOISE_BT) { |
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539 for(; i < run_end; i++, idx++) { |
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540 if(noise_flag-- > 0) |
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541 offset[1] += get_bits(gb, 9) - 256; |
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542 else |
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543 offset[1] += get_vlc2(gb, vlc_scalefactors.table, 7, 3) - 60; |
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544 if(offset[1] > 255U) { |
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545 av_log(ac->avccontext, AV_LOG_ERROR, |
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546 "%s (%d) out of range.\n", sf_str[1], offset[1]); |
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547 return -1; |
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548 } |
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549 sf[idx] = -ff_aac_pow2sf_tab[ offset[1] + sf_offset]; |
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550 } |
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551 }else { |
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552 for(; i < run_end; i++, idx++) { |
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553 offset[0] += get_vlc2(gb, vlc_scalefactors.table, 7, 3) - 60; |
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554 if(offset[0] > 255U) { |
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555 av_log(ac->avccontext, AV_LOG_ERROR, |
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556 "%s (%d) out of range.\n", sf_str[0], offset[0]); |
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557 return -1; |
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558 } |
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559 sf[idx] = -ff_aac_pow2sf_tab[ offset[0] + sf_offset]; |
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560 } |
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561 } |
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562 } |
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563 } |
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564 return 0; |
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565 } |
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566 |
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567 /** |
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568 * Decode pulse data; reference: table 4.7. |
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569 */ |
7580 | 570 static void decode_pulses(Pulse * pulse, GetBitContext * gb, const uint16_t * swb_offset) { |
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571 int i; |
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572 pulse->num_pulse = get_bits(gb, 2) + 1; |
7580 | 573 pulse->pos[0] = get_bits(gb, 5) + swb_offset[get_bits(gb, 6)]; |
574 pulse->amp[0] = get_bits(gb, 4); | |
575 for (i = 1; i < pulse->num_pulse; i++) { | |
576 pulse->pos[i] = get_bits(gb, 5) + pulse->pos[i-1]; | |
577 pulse->amp[i] = get_bits(gb, 4); | |
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578 } |
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579 } |
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580 |
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581 /** |
7539 | 582 * Decode Mid/Side data; reference: table 4.54. |
583 * | |
584 * @param ms_present Indicates mid/side stereo presence. [0] mask is all 0s; | |
585 * [1] mask is decoded from bitstream; [2] mask is all 1s; | |
586 * [3] reserved for scalable AAC | |
587 */ | |
588 static void decode_mid_side_stereo(ChannelElement * cpe, GetBitContext * gb, | |
589 int ms_present) { | |
7578 | 590 int idx; |
591 if (ms_present == 1) { | |
592 for (idx = 0; idx < cpe->ch[0].ics.num_window_groups * cpe->ch[0].ics.max_sfb; idx++) | |
593 cpe->ms_mask[idx] = get_bits1(gb); | |
594 } else if (ms_present == 2) { | |
595 memset(cpe->ms_mask, 1, cpe->ch[0].ics.num_window_groups * cpe->ch[0].ics.max_sfb * sizeof(cpe->ms_mask[0])); | |
596 } | |
597 } | |
7539 | 598 |
599 /** | |
600 * Decode an individual_channel_stream payload; reference: table 4.44. | |
601 * | |
602 * @param common_window Channels have independent [0], or shared [1], Individual Channel Stream information. | |
603 * @param scale_flag scalable [1] or non-scalable [0] AAC (Unused until scalable AAC is implemented.) | |
604 * | |
605 * @return Returns error status. 0 - OK, !0 - error | |
606 */ | |
607 static int decode_ics(AACContext * ac, SingleChannelElement * sce, GetBitContext * gb, int common_window, int scale_flag) { | |
608 Pulse pulse; | |
609 TemporalNoiseShaping * tns = &sce->tns; | |
610 IndividualChannelStream * ics = &sce->ics; | |
611 float * out = sce->coeffs; | |
612 int global_gain, pulse_present = 0; | |
613 | |
7580 | 614 /* This assignment is to silence a GCC warning about the variable being used |
615 * uninitialized when in fact it always is. | |
7539 | 616 */ |
617 pulse.num_pulse = 0; | |
618 | |
619 global_gain = get_bits(gb, 8); | |
620 | |
621 if (!common_window && !scale_flag) { | |
622 if (decode_ics_info(ac, ics, gb, 0) < 0) | |
623 return -1; | |
624 } | |
625 | |
626 if (decode_band_types(ac, sce->band_type, sce->band_type_run_end, gb, ics) < 0) | |
627 return -1; | |
628 if (decode_scalefactors(ac, sce->sf, gb, global_gain, ics, sce->band_type, sce->band_type_run_end) < 0) | |
629 return -1; | |
630 | |
631 pulse_present = 0; | |
632 if (!scale_flag) { | |
633 if ((pulse_present = get_bits1(gb))) { | |
634 if (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) { | |
635 av_log(ac->avccontext, AV_LOG_ERROR, "Pulse tool not allowed in eight short sequence.\n"); | |
636 return -1; | |
637 } | |
7580 | 638 decode_pulses(&pulse, gb, ics->swb_offset); |
7539 | 639 } |
640 if ((tns->present = get_bits1(gb)) && decode_tns(ac, tns, gb, ics)) | |
641 return -1; | |
642 if (get_bits1(gb)) { | |
643 av_log_missing_feature(ac->avccontext, "SSR", 1); | |
644 return -1; | |
645 } | |
646 } | |
647 | |
7580 | 648 if (decode_spectrum_and_dequant(ac, out, gb, sce->sf, pulse_present, &pulse, ics, sce->band_type) < 0) |
7539 | 649 return -1; |
650 return 0; | |
651 } | |
652 | |
653 /** | |
654 * Decode a channel_pair_element; reference: table 4.4. | |
655 * | |
656 * @param elem_id Identifies the instance of a syntax element. | |
657 * | |
658 * @return Returns error status. 0 - OK, !0 - error | |
659 */ | |
660 static int decode_cpe(AACContext * ac, GetBitContext * gb, int elem_id) { | |
661 int i, ret, common_window, ms_present = 0; | |
662 ChannelElement * cpe; | |
663 | |
664 cpe = ac->che[TYPE_CPE][elem_id]; | |
665 common_window = get_bits1(gb); | |
666 if (common_window) { | |
667 if (decode_ics_info(ac, &cpe->ch[0].ics, gb, 1)) | |
668 return -1; | |
669 i = cpe->ch[1].ics.use_kb_window[0]; | |
670 cpe->ch[1].ics = cpe->ch[0].ics; | |
671 cpe->ch[1].ics.use_kb_window[1] = i; | |
672 ms_present = get_bits(gb, 2); | |
673 if(ms_present == 3) { | |
674 av_log(ac->avccontext, AV_LOG_ERROR, "ms_present = 3 is reserved.\n"); | |
675 return -1; | |
676 } else if(ms_present) | |
677 decode_mid_side_stereo(cpe, gb, ms_present); | |
678 } | |
679 if ((ret = decode_ics(ac, &cpe->ch[0], gb, common_window, 0))) | |
680 return ret; | |
681 if ((ret = decode_ics(ac, &cpe->ch[1], gb, common_window, 0))) | |
682 return ret; | |
683 | |
684 if (common_window && ms_present) | |
685 apply_mid_side_stereo(cpe); | |
686 | |
7580 | 687 apply_intensity_stereo(cpe, ms_present); |
7539 | 688 return 0; |
689 } | |
690 | |
7578 | 691 coup->coupling_point = 2*get_bits1(gb); |
692 coup->num_coupled = get_bits(gb, 3); | |
693 for (c = 0; c <= coup->num_coupled; c++) { | |
694 num_gain++; | |
695 coup->type[c] = get_bits1(gb) ? TYPE_CPE : TYPE_SCE; | |
696 coup->id_select[c] = get_bits(gb, 4); | |
697 if (coup->type[c] == TYPE_CPE) { | |
698 coup->ch_select[c] = get_bits(gb, 2); | |
699 if (coup->ch_select[c] == 3) | |
700 num_gain++; | |
701 } else | |
702 coup->ch_select[c] = 1; | |
703 } | |
704 coup->coupling_point += get_bits1(gb); | |
705 | |
706 if (coup->coupling_point == 2) { | |
707 av_log(ac->avccontext, AV_LOG_ERROR, | |
708 "Independently switched CCE with 'invalid' domain signalled.\n"); | |
709 memset(coup, 0, sizeof(ChannelCoupling)); | |
710 return -1; | |
711 } | |
712 | |
713 sign = get_bits(gb, 1); | |
714 scale = pow(2., pow(2., get_bits(gb, 2) - 3)); | |
715 | |
716 if ((ret = decode_ics(ac, sce, gb, 0, 0))) | |
717 return ret; | |
718 | |
719 for (c = 0; c < num_gain; c++) { | |
720 int cge = 1; | |
721 int gain = 0; | |
722 float gain_cache = 1.; | |
723 if (c) { | |
724 cge = coup->coupling_point == AFTER_IMDCT ? 1 : get_bits1(gb); | |
725 gain = cge ? get_vlc2(gb, vlc_scalefactors.table, 7, 3) - 60: 0; | |
726 gain_cache = pow(scale, gain); | |
727 } | |
728 for (g = 0; g < sce->ics.num_window_groups; g++) | |
729 for (sfb = 0; sfb < sce->ics.max_sfb; sfb++, idx++) | |
730 if (sce->band_type[idx] != ZERO_BT) { | |
731 if (!cge) { | |
732 int t = get_vlc2(gb, vlc_scalefactors.table, 7, 3) - 60; | |
733 if (t) { | |
734 int s = 1; | |
735 if (sign) { | |
736 s -= 2 * (t & 0x1); | |
737 t >>= 1; | |
738 } | |
739 gain += t; | |
740 gain_cache = pow(scale, gain) * s; | |
741 } | |
742 } | |
743 coup->gain[c][idx] = gain_cache; | |
744 } | |
745 } | |
746 return 0; | |
747 } | |
748 | |
7539 | 749 /** |
750 * Decode Spectral Band Replication extension data; reference: table 4.55. | |
7523
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751 * |
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752 * @param crc flag indicating the presence of CRC checksum |
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753 * @param cnt length of TYPE_FIL syntactic element in bytes |
7539 | 754 * |
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755 * @return Returns number of bytes consumed from the TYPE_FIL element. |
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756 */ |
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757 static int decode_sbr_extension(AACContext * ac, GetBitContext * gb, int crc, int cnt) { |
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758 // TODO : sbr_extension implementation |
7540 | 759 av_log_missing_feature(ac->avccontext, "SBR", 0); |
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760 skip_bits_long(gb, 8*cnt - 4); // -4 due to reading extension type |
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761 return cnt; |
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762 } |
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763 |
7539 | 764 /** |
7578 | 765 * Parse whether channels are to be excluded from Dynamic Range Compression; reference: table 4.53. |
766 * | |
767 * @return Returns number of bytes consumed. | |
768 */ | |
769 static int decode_drc_channel_exclusions(DynamicRangeControl *che_drc, GetBitContext * gb) { | |
770 int i; | |
771 int num_excl_chan = 0; | |
772 | |
773 do { | |
774 for (i = 0; i < 7; i++) | |
775 che_drc->exclude_mask[num_excl_chan++] = get_bits1(gb); | |
776 } while (num_excl_chan < MAX_CHANNELS - 7 && get_bits1(gb)); | |
777 | |
778 return num_excl_chan / 7; | |
779 } | |
780 | |
781 /** | |
7539 | 782 * Decode dynamic range information; reference: table 4.52. |
783 * | |
784 * @param cnt length of TYPE_FIL syntactic element in bytes | |
785 * | |
786 * @return Returns number of bytes consumed. | |
787 */ | |
788 static int decode_dynamic_range(DynamicRangeControl *che_drc, GetBitContext * gb, int cnt) { | |
789 int n = 1; | |
790 int drc_num_bands = 1; | |
791 int i; | |
792 | |
793 /* pce_tag_present? */ | |
794 if(get_bits1(gb)) { | |
795 che_drc->pce_instance_tag = get_bits(gb, 4); | |
796 skip_bits(gb, 4); // tag_reserved_bits | |
797 n++; | |
798 } | |
799 | |
800 /* excluded_chns_present? */ | |
801 if(get_bits1(gb)) { | |
802 n += decode_drc_channel_exclusions(che_drc, gb); | |
803 } | |
804 | |
805 /* drc_bands_present? */ | |
806 if (get_bits1(gb)) { | |
807 che_drc->band_incr = get_bits(gb, 4); | |
808 che_drc->interpolation_scheme = get_bits(gb, 4); | |
809 n++; | |
810 drc_num_bands += che_drc->band_incr; | |
811 for (i = 0; i < drc_num_bands; i++) { | |
812 che_drc->band_top[i] = get_bits(gb, 8); | |
813 n++; | |
814 } | |
815 } | |
816 | |
817 /* prog_ref_level_present? */ | |
818 if (get_bits1(gb)) { | |
819 che_drc->prog_ref_level = get_bits(gb, 7); | |
820 skip_bits1(gb); // prog_ref_level_reserved_bits | |
821 n++; | |
822 } | |
823 | |
824 for (i = 0; i < drc_num_bands; i++) { | |
825 che_drc->dyn_rng_sgn[i] = get_bits1(gb); | |
826 che_drc->dyn_rng_ctl[i] = get_bits(gb, 7); | |
827 n++; | |
828 } | |
829 | |
830 return n; | |
831 } | |
832 | |
833 /** | |
834 * Decode extension data (incomplete); reference: table 4.51. | |
835 * | |
836 * @param cnt length of TYPE_FIL syntactic element in bytes | |
837 * | |
838 * @return Returns number of bytes consumed | |
839 */ | |
840 static int decode_extension_payload(AACContext * ac, GetBitContext * gb, int cnt) { | |
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841 int crc_flag = 0; |
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842 int res = cnt; |
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843 switch (get_bits(gb, 4)) { // extension type |
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844 case EXT_SBR_DATA_CRC: |
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845 crc_flag++; |
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846 case EXT_SBR_DATA: |
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847 res = decode_sbr_extension(ac, gb, crc_flag, cnt); |
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848 break; |
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849 case EXT_DYNAMIC_RANGE: |
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850 res = decode_dynamic_range(&ac->che_drc, gb, cnt); |
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851 break; |
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852 case EXT_FILL: |
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853 case EXT_FILL_DATA: |
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854 case EXT_DATA_ELEMENT: |
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855 default: |
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856 skip_bits_long(gb, 8*cnt - 4); |
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857 break; |
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858 }; |
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859 return res; |
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860 } |
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861 |
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862 /** |
7539 | 863 * Conduct IMDCT and windowing. |
864 */ | |
865 static void imdct_and_windowing(AACContext * ac, SingleChannelElement * sce) { | |
866 IndividualChannelStream * ics = &sce->ics; | |
867 float * in = sce->coeffs; | |
868 float * out = sce->ret; | |
869 float * saved = sce->saved; | |
7580 | 870 const float * lwindow = ics->use_kb_window[0] ? ff_aac_kbd_long_1024 : ff_sine_1024; |
871 const float * swindow = ics->use_kb_window[0] ? ff_aac_kbd_short_128 : ff_sine_128; | |
872 const float * lwindow_prev = ics->use_kb_window[1] ? ff_aac_kbd_long_1024 : ff_sine_1024; | |
873 const float * swindow_prev = ics->use_kb_window[1] ? ff_aac_kbd_short_128 : ff_sine_128; | |
7539 | 874 float * buf = ac->buf_mdct; |
875 int i; | |
876 | |
7578 | 877 if (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) { |
878 if (ics->window_sequence[1] == ONLY_LONG_SEQUENCE || ics->window_sequence[1] == LONG_STOP_SEQUENCE) | |
879 av_log(ac->avccontext, AV_LOG_WARNING, | |
880 "Transition from an ONLY_LONG or LONG_STOP to an EIGHT_SHORT sequence detected. " | |
881 "If you heard an audible artifact, please submit the sample to the FFmpeg developers.\n"); | |
882 for (i = 0; i < 2048; i += 256) { | |
883 ff_imdct_calc(&ac->mdct_small, buf + i, in + i/2); | |
884 ac->dsp.vector_fmul_reverse(ac->revers + i/2, buf + i + 128, swindow, 128); | |
885 } | |
886 for (i = 0; i < 448; i++) out[i] = saved[i] + ac->add_bias; | |
887 | |
888 ac->dsp.vector_fmul_add_add(out + 448 + 0*128, buf + 0*128, swindow_prev, saved + 448 , ac->add_bias, 128, 1); | |
889 ac->dsp.vector_fmul_add_add(out + 448 + 1*128, buf + 2*128, swindow, ac->revers + 0*128, ac->add_bias, 128, 1); | |
890 ac->dsp.vector_fmul_add_add(out + 448 + 2*128, buf + 4*128, swindow, ac->revers + 1*128, ac->add_bias, 128, 1); | |
891 ac->dsp.vector_fmul_add_add(out + 448 + 3*128, buf + 6*128, swindow, ac->revers + 2*128, ac->add_bias, 128, 1); | |
892 ac->dsp.vector_fmul_add_add(out + 448 + 4*128, buf + 8*128, swindow, ac->revers + 3*128, ac->add_bias, 64, 1); | |
893 | |
894 #if 0 | |
895 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); | |
896 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); | |
897 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); | |
898 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); | |
899 #endif | |
900 | |
901 ac->dsp.vector_fmul_add_add(saved, buf + 1024 + 64, swindow + 64, ac->revers + 3*128+64, 0, 64, 1); | |
902 ac->dsp.vector_fmul_add_add(saved + 64, buf + 1024 + 2*128, swindow, ac->revers + 4*128, 0, 128, 1); | |
903 ac->dsp.vector_fmul_add_add(saved + 192, buf + 1024 + 4*128, swindow, ac->revers + 5*128, 0, 128, 1); | |
904 ac->dsp.vector_fmul_add_add(saved + 320, buf + 1024 + 6*128, swindow, ac->revers + 6*128, 0, 128, 1); | |
905 memcpy( saved + 448, ac->revers + 7*128, 128 * sizeof(float)); | |
906 memset( saved + 576, 0, 448 * sizeof(float)); | |
907 } else { | |
908 ff_imdct_calc(&ac->mdct, buf, in); | |
909 if (ics->window_sequence[0] == LONG_STOP_SEQUENCE) { | |
910 for (i = 0; i < 448; i++) out[i] = saved[i] + ac->add_bias; | |
911 ac->dsp.vector_fmul_add_add(out + 448, buf + 448, swindow_prev, saved + 448, ac->add_bias, 128, 1); | |
912 for (i = 576; i < 1024; i++) out[i] = buf[i] + saved[i] + ac->add_bias; | |
913 } else { | |
914 ac->dsp.vector_fmul_add_add(out, buf, lwindow_prev, saved, ac->add_bias, 1024, 1); | |
915 } | |
916 if (ics->window_sequence[0] == LONG_START_SEQUENCE) { | |
917 memcpy(saved, buf + 1024, 448 * sizeof(float)); | |
918 ac->dsp.vector_fmul_reverse(saved + 448, buf + 1024 + 448, swindow, 128); | |
919 memset(saved + 576, 0, 448 * sizeof(float)); | |
920 } else { | |
921 ac->dsp.vector_fmul_reverse(saved, buf + 1024, lwindow, 1024); | |
922 } | |
923 } | |
924 } | |
925 | |
7539 | 926 /** |
7523
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927 * Apply dependent channel coupling (applied before IMDCT). |
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928 * |
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929 * @param index index into coupling gain array |
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930 */ |
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931 static void apply_dependent_coupling(AACContext * ac, SingleChannelElement * sce, ChannelElement * cc, int index) { |
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932 IndividualChannelStream * ics = &cc->ch[0].ics; |
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933 const uint16_t * offsets = ics->swb_offset; |
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934 float * dest = sce->coeffs; |
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935 const float * src = cc->ch[0].coeffs; |
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936 int g, i, group, k, idx = 0; |
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937 if(ac->m4ac.object_type == AOT_AAC_LTP) { |
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938 av_log(ac->avccontext, AV_LOG_ERROR, |
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939 "Dependent coupling is not supported together with LTP\n"); |
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940 return; |
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941 } |
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942 for (g = 0; g < ics->num_window_groups; g++) { |
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943 for (i = 0; i < ics->max_sfb; i++, idx++) { |
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944 if (cc->ch[0].band_type[idx] != ZERO_BT) { |
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945 for (group = 0; group < ics->group_len[g]; group++) { |
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946 for (k = offsets[i]; k < offsets[i+1]; k++) { |
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947 // XXX dsputil-ize |
7540 | 948 dest[group*128+k] += cc->coup.gain[index][idx] * src[group*128+k]; |
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949 } |
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950 } |
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951 } |
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952 } |
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953 dest += ics->group_len[g]*128; |
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954 src += ics->group_len[g]*128; |
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955 } |
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956 } |
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957 |
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958 /** |
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959 * Apply independent channel coupling (applied after IMDCT). |
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960 * |
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961 * @param index index into coupling gain array |
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962 */ |
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963 static void apply_independent_coupling(AACContext * ac, SingleChannelElement * sce, ChannelElement * cc, int index) { |
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964 int i; |
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965 for (i = 0; i < 1024; i++) |
7540 | 966 sce->ret[i] += cc->coup.gain[index][0] * (cc->ch[0].ret[i] - ac->add_bias); |
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967 } |
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968 |
7578 | 969 } |
970 } | |
971 } | |
972 } | |
973 | |
974 static int aac_decode_frame(AVCodecContext * avccontext, void * data, int * data_size, const uint8_t * buf, int buf_size) { | |
975 AACContext * ac = avccontext->priv_data; | |
976 GetBitContext gb; | |
977 enum RawDataBlockType elem_type; | |
978 int err, elem_id, data_size_tmp; | |
979 | |
980 init_get_bits(&gb, buf, buf_size*8); | |
981 | |
982 // parse | |
983 while ((elem_type = get_bits(&gb, 3)) != TYPE_END) { | |
984 elem_id = get_bits(&gb, 4); | |
985 err = -1; | |
986 | |
987 if(elem_type == TYPE_SCE && elem_id == 1 && | |
988 !ac->che[TYPE_SCE][elem_id] && ac->che[TYPE_LFE][0]) { | |
989 /* Some streams incorrectly code 5.1 audio as SCE[0] CPE[0] CPE[1] SCE[1] | |
990 instead of SCE[0] CPE[0] CPE[0] LFE[0]. If we seem to have | |
991 encountered such a stream, transfer the LFE[0] element to SCE[1] */ | |
992 ac->che[TYPE_SCE][elem_id] = ac->che[TYPE_LFE][0]; | |
993 ac->che[TYPE_LFE][0] = NULL; | |
994 } | |
995 if(elem_type && elem_type < TYPE_DSE) { | |
996 if(!ac->che[elem_type][elem_id]) | |
997 return -1; | |
998 if(elem_type != TYPE_CCE) | |
999 ac->che[elem_type][elem_id]->coup.coupling_point = 4; | |
1000 } | |
1001 | |
1002 switch (elem_type) { | |
1003 | |
1004 case TYPE_SCE: | |
1005 err = decode_ics(ac, &ac->che[TYPE_SCE][elem_id]->ch[0], &gb, 0, 0); | |
1006 break; | |
1007 | |
1008 case TYPE_CPE: | |
1009 err = decode_cpe(ac, &gb, elem_id); | |
1010 break; | |
1011 | |
1012 case TYPE_CCE: | |
1013 err = decode_cce(ac, &gb, ac->che[TYPE_SCE][elem_id]); | |
1014 break; | |
1015 | |
1016 case TYPE_LFE: | |
1017 err = decode_ics(ac, &ac->che[TYPE_LFE][elem_id]->ch[0], &gb, 0, 0); | |
1018 break; | |
1019 | |
1020 case TYPE_DSE: | |
1021 skip_data_stream_element(&gb); | |
1022 err = 0; | |
1023 break; | |
1024 | |
1025 case TYPE_PCE: | |
1026 { | |
1027 enum ChannelPosition new_che_pos[4][MAX_ELEM_ID]; | |
1028 memset(new_che_pos, 0, 4 * MAX_ELEM_ID * sizeof(new_che_pos[0][0])); | |
1029 if((err = decode_pce(ac, new_che_pos, &gb))) | |
1030 break; | |
1031 err = output_configure(ac, ac->che_pos, new_che_pos); | |
1032 break; | |
1033 } | |
1034 | |
1035 case TYPE_FIL: | |
1036 if (elem_id == 15) | |
1037 elem_id += get_bits(&gb, 8) - 1; | |
1038 while (elem_id > 0) | |
1039 elem_id -= decode_extension_payload(ac, &gb, elem_id); | |
1040 err = 0; /* FIXME */ | |
1041 break; | |
1042 | |
1043 default: | |
1044 err = -1; /* should not happen, but keeps compiler happy */ | |
1045 break; | |
1046 } | |
1047 | |
1048 if(err) | |
1049 return err; | |
1050 } | |
1051 | |
1052 spectral_to_sample(ac); | |
1053 | |
7539 | 1054 if (!ac->is_saved) { |
1055 ac->is_saved = 1; | |
1056 *data_size = 0; | |
7580 | 1057 return buf_size; |
7539 | 1058 } |
1059 | |
1060 data_size_tmp = 1024 * avccontext->channels * sizeof(int16_t); | |
1061 if(*data_size < data_size_tmp) { | |
1062 av_log(avccontext, AV_LOG_ERROR, | |
1063 "Output buffer too small (%d) or trying to output too many samples (%d) for this frame.\n", | |
1064 *data_size, data_size_tmp); | |
1065 return -1; | |
1066 } | |
1067 *data_size = data_size_tmp; | |
1068 | |
1069 ac->dsp.float_to_int16_interleave(data, (const float **)ac->output_data, 1024, avccontext->channels); | |
1070 | |
1071 return buf_size; | |
1072 } | |
1073 | |
7501 | 1074 static av_cold int aac_decode_close(AVCodecContext * avccontext) { |
1075 AACContext * ac = avccontext->priv_data; | |
7540 | 1076 int i, type; |
7501 | 1077 |
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1078 for (i = 0; i < MAX_ELEM_ID; i++) { |
7540 | 1079 for(type = 0; type < 4; type++) |
1080 av_freep(&ac->che[type][i]); | |
7501 | 1081 } |
1082 | |
1083 ff_mdct_end(&ac->mdct); | |
1084 ff_mdct_end(&ac->mdct_small); | |
1085 return 0 ; | |
1086 } | |
1087 | |
1088 AVCodec aac_decoder = { | |
1089 "aac", | |
1090 CODEC_TYPE_AUDIO, | |
1091 CODEC_ID_AAC, | |
1092 sizeof(AACContext), | |
1093 aac_decode_init, | |
1094 NULL, | |
1095 aac_decode_close, | |
1096 aac_decode_frame, | |
1097 .long_name = NULL_IF_CONFIG_SMALL("Advanced Audio Coding"), | |
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1098 .sample_fmts = (enum SampleFormat[]){SAMPLE_FMT_S16,SAMPLE_FMT_NONE}, |
7501 | 1099 }; |