Mercurial > libavcodec.hg
annotate aac.c @ 8349:920ded20b469 libavcodec
Horizontal and vertical prediction functions needed for new lossless mode
of H.264.
author | michael |
---|---|
date | Tue, 16 Dec 2008 19:39:01 +0000 |
parents | e06b2b848b74 |
children | 2d48043ab521 |
rev | line source |
---|---|
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 | |
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44 * Y frequency domain prediction |
7501 | 45 * Y Perceptual Noise Substitution |
46 * Y Mid/Side stereo | |
47 * N Scalable Inverse AAC Quantization | |
48 * N Frequency Selective Switch | |
49 * N upsampling filter | |
50 * Y quantization & coding - AAC | |
51 * N quantization & coding - TwinVQ | |
52 * N quantization & coding - BSAC | |
53 * N AAC Error Resilience tools | |
54 * N Error Resilience payload syntax | |
55 * N Error Protection tool | |
56 * N CELP | |
57 * N Silence Compression | |
58 * N HVXC | |
59 * N HVXC 4kbits/s VR | |
60 * N Structured Audio tools | |
61 * N Structured Audio Sample Bank Format | |
62 * N MIDI | |
63 * N Harmonic and Individual Lines plus Noise | |
64 * N Text-To-Speech Interface | |
65 * N (in progress) Spectral Band Replication | |
66 * Y (not in this code) Layer-1 | |
67 * Y (not in this code) Layer-2 | |
68 * Y (not in this code) Layer-3 | |
69 * N SinuSoidal Coding (Transient, Sinusoid, Noise) | |
70 * N (planned) Parametric Stereo | |
71 * N Direct Stream Transfer | |
72 * | |
73 * Note: - HE AAC v1 comprises LC AAC with Spectral Band Replication. | |
74 * - HE AAC v2 comprises LC AAC with Spectral Band Replication and | |
75 Parametric Stereo. | |
76 */ | |
77 | |
78 | |
79 #include "avcodec.h" | |
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80 #include "internal.h" |
7501 | 81 #include "bitstream.h" |
82 #include "dsputil.h" | |
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83 #include "lpc.h" |
7501 | 84 |
85 #include "aac.h" | |
86 #include "aactab.h" | |
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87 #include "aacdectab.h" |
7501 | 88 #include "mpeg4audio.h" |
89 | |
90 #include <assert.h> | |
91 #include <errno.h> | |
92 #include <math.h> | |
93 #include <string.h> | |
94 | |
95 static VLC vlc_scalefactors; | |
96 static VLC vlc_spectral[11]; | |
97 | |
98 | |
7539 | 99 /** |
7578 | 100 * Configure output channel order based on the current program configuration element. |
101 * | |
102 * @param che_pos current channel position configuration | |
103 * @param new_che_pos New channel position configuration - we only do something if it differs from the current one. | |
104 * | |
105 * @return Returns error status. 0 - OK, !0 - error | |
106 */ | |
107 static int output_configure(AACContext *ac, enum ChannelPosition che_pos[4][MAX_ELEM_ID], | |
108 enum ChannelPosition new_che_pos[4][MAX_ELEM_ID]) { | |
109 AVCodecContext *avctx = ac->avccontext; | |
110 int i, type, channels = 0; | |
111 | |
112 if(!memcmp(che_pos, new_che_pos, 4 * MAX_ELEM_ID * sizeof(new_che_pos[0][0]))) | |
113 return 0; /* no change */ | |
114 | |
115 memcpy(che_pos, new_che_pos, 4 * MAX_ELEM_ID * sizeof(new_che_pos[0][0])); | |
116 | |
117 /* Allocate or free elements depending on if they are in the | |
118 * current program configuration. | |
119 * | |
120 * Set up default 1:1 output mapping. | |
121 * | |
122 * For a 5.1 stream the output order will be: | |
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123 * [ Center ] [ Front Left ] [ Front Right ] [ LFE ] [ Surround Left ] [ Surround Right ] |
7578 | 124 */ |
125 | |
126 for(i = 0; i < MAX_ELEM_ID; i++) { | |
127 for(type = 0; type < 4; type++) { | |
128 if(che_pos[type][i]) { | |
129 if(!ac->che[type][i] && !(ac->che[type][i] = av_mallocz(sizeof(ChannelElement)))) | |
130 return AVERROR(ENOMEM); | |
131 if(type != TYPE_CCE) { | |
132 ac->output_data[channels++] = ac->che[type][i]->ch[0].ret; | |
133 if(type == TYPE_CPE) { | |
134 ac->output_data[channels++] = ac->che[type][i]->ch[1].ret; | |
135 } | |
136 } | |
137 } else | |
138 av_freep(&ac->che[type][i]); | |
139 } | |
140 } | |
141 | |
142 avctx->channels = channels; | |
143 return 0; | |
144 } | |
145 | |
146 /** | |
7539 | 147 * Decode an array of 4 bit element IDs, optionally interleaved with a stereo/mono switching bit. |
148 * | |
149 * @param cpe_map Stereo (Channel Pair Element) map, NULL if stereo bit is not present. | |
150 * @param sce_map mono (Single Channel Element) map | |
151 * @param type speaker type/position for these channels | |
152 */ | |
153 static void decode_channel_map(enum ChannelPosition *cpe_map, | |
154 enum ChannelPosition *sce_map, enum ChannelPosition type, GetBitContext * gb, int n) { | |
155 while(n--) { | |
156 enum ChannelPosition *map = cpe_map && get_bits1(gb) ? cpe_map : sce_map; // stereo or mono map | |
157 map[get_bits(gb, 4)] = type; | |
158 } | |
159 } | |
160 | |
161 /** | |
162 * Decode program configuration element; reference: table 4.2. | |
163 * | |
164 * @param new_che_pos New channel position configuration - we only do something if it differs from the current one. | |
165 * | |
166 * @return Returns error status. 0 - OK, !0 - error | |
167 */ | |
168 static int decode_pce(AACContext * ac, enum ChannelPosition new_che_pos[4][MAX_ELEM_ID], | |
169 GetBitContext * gb) { | |
170 int num_front, num_side, num_back, num_lfe, num_assoc_data, num_cc; | |
171 | |
172 skip_bits(gb, 2); // object_type | |
173 | |
174 ac->m4ac.sampling_index = get_bits(gb, 4); | |
175 if(ac->m4ac.sampling_index > 11) { | |
176 av_log(ac->avccontext, AV_LOG_ERROR, "invalid sampling rate index %d\n", ac->m4ac.sampling_index); | |
177 return -1; | |
178 } | |
179 ac->m4ac.sample_rate = ff_mpeg4audio_sample_rates[ac->m4ac.sampling_index]; | |
7501 | 180 num_front = get_bits(gb, 4); |
181 num_side = get_bits(gb, 4); | |
182 num_back = get_bits(gb, 4); | |
183 num_lfe = get_bits(gb, 2); | |
184 num_assoc_data = get_bits(gb, 3); | |
185 num_cc = get_bits(gb, 4); | |
186 | |
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187 if (get_bits1(gb)) |
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188 skip_bits(gb, 4); // mono_mixdown_tag |
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189 if (get_bits1(gb)) |
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190 skip_bits(gb, 4); // stereo_mixdown_tag |
7501 | 191 |
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192 if (get_bits1(gb)) |
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193 skip_bits(gb, 3); // mixdown_coeff_index and pseudo_surround |
7501 | 194 |
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195 decode_channel_map(new_che_pos[TYPE_CPE], new_che_pos[TYPE_SCE], AAC_CHANNEL_FRONT, gb, num_front); |
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196 decode_channel_map(new_che_pos[TYPE_CPE], new_che_pos[TYPE_SCE], AAC_CHANNEL_SIDE, gb, num_side ); |
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197 decode_channel_map(new_che_pos[TYPE_CPE], new_che_pos[TYPE_SCE], AAC_CHANNEL_BACK, gb, num_back ); |
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198 decode_channel_map(NULL, new_che_pos[TYPE_LFE], AAC_CHANNEL_LFE, gb, num_lfe ); |
7501 | 199 |
200 skip_bits_long(gb, 4 * num_assoc_data); | |
201 | |
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202 decode_channel_map(new_che_pos[TYPE_CCE], new_che_pos[TYPE_CCE], AAC_CHANNEL_CC, gb, num_cc ); |
7501 | 203 |
204 align_get_bits(gb); | |
205 | |
206 /* comment field, first byte is length */ | |
207 skip_bits_long(gb, 8 * get_bits(gb, 8)); | |
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208 return 0; |
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209 } |
7501 | 210 |
7539 | 211 /** |
212 * Set up channel positions based on a default channel configuration | |
213 * as specified in table 1.17. | |
214 * | |
215 * @param new_che_pos New channel position configuration - we only do something if it differs from the current one. | |
216 * | |
217 * @return Returns error status. 0 - OK, !0 - error | |
218 */ | |
219 static int set_default_channel_config(AACContext *ac, enum ChannelPosition new_che_pos[4][MAX_ELEM_ID], | |
220 int channel_config) | |
221 { | |
222 if(channel_config < 1 || channel_config > 7) { | |
223 av_log(ac->avccontext, AV_LOG_ERROR, "invalid default channel configuration (%d)\n", | |
224 channel_config); | |
225 return -1; | |
226 } | |
227 | |
228 /* default channel configurations: | |
229 * | |
230 * 1ch : front center (mono) | |
231 * 2ch : L + R (stereo) | |
232 * 3ch : front center + L + R | |
233 * 4ch : front center + L + R + back center | |
234 * 5ch : front center + L + R + back stereo | |
235 * 6ch : front center + L + R + back stereo + LFE | |
236 * 7ch : front center + L + R + outer front left + outer front right + back stereo + LFE | |
237 */ | |
238 | |
239 if(channel_config != 2) | |
240 new_che_pos[TYPE_SCE][0] = AAC_CHANNEL_FRONT; // front center (or mono) | |
241 if(channel_config > 1) | |
242 new_che_pos[TYPE_CPE][0] = AAC_CHANNEL_FRONT; // L + R (or stereo) | |
243 if(channel_config == 4) | |
244 new_che_pos[TYPE_SCE][1] = AAC_CHANNEL_BACK; // back center | |
245 if(channel_config > 4) | |
246 new_che_pos[TYPE_CPE][(channel_config == 7) + 1] | |
247 = AAC_CHANNEL_BACK; // back stereo | |
248 if(channel_config > 5) | |
249 new_che_pos[TYPE_LFE][0] = AAC_CHANNEL_LFE; // LFE | |
250 if(channel_config == 7) | |
251 new_che_pos[TYPE_CPE][1] = AAC_CHANNEL_FRONT; // outer front left + outer front right | |
252 | |
253 return 0; | |
254 } | |
255 | |
7578 | 256 /** |
257 * Decode GA "General Audio" specific configuration; reference: table 4.1. | |
258 * | |
259 * @return Returns error status. 0 - OK, !0 - error | |
260 */ | |
261 static int decode_ga_specific_config(AACContext * ac, GetBitContext * gb, int channel_config) { | |
262 enum ChannelPosition new_che_pos[4][MAX_ELEM_ID]; | |
263 int extension_flag, ret; | |
264 | |
265 if(get_bits1(gb)) { // frameLengthFlag | |
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266 ff_log_missing_feature(ac->avccontext, "960/120 MDCT window is", 1); |
7539 | 267 return -1; |
268 } | |
269 | |
270 if (get_bits1(gb)) // dependsOnCoreCoder | |
271 skip_bits(gb, 14); // coreCoderDelay | |
272 extension_flag = get_bits1(gb); | |
273 | |
274 if(ac->m4ac.object_type == AOT_AAC_SCALABLE || | |
275 ac->m4ac.object_type == AOT_ER_AAC_SCALABLE) | |
276 skip_bits(gb, 3); // layerNr | |
277 | |
278 memset(new_che_pos, 0, 4 * MAX_ELEM_ID * sizeof(new_che_pos[0][0])); | |
279 if (channel_config == 0) { | |
280 skip_bits(gb, 4); // element_instance_tag | |
281 if((ret = decode_pce(ac, new_che_pos, gb))) | |
282 return ret; | |
283 } else { | |
284 if((ret = set_default_channel_config(ac, new_che_pos, channel_config))) | |
285 return ret; | |
286 } | |
287 if((ret = output_configure(ac, ac->che_pos, new_che_pos))) | |
288 return ret; | |
289 | |
290 if (extension_flag) { | |
291 switch (ac->m4ac.object_type) { | |
292 case AOT_ER_BSAC: | |
293 skip_bits(gb, 5); // numOfSubFrame | |
294 skip_bits(gb, 11); // layer_length | |
295 break; | |
296 case AOT_ER_AAC_LC: | |
297 case AOT_ER_AAC_LTP: | |
298 case AOT_ER_AAC_SCALABLE: | |
299 case AOT_ER_AAC_LD: | |
300 skip_bits(gb, 3); /* aacSectionDataResilienceFlag | |
301 * aacScalefactorDataResilienceFlag | |
302 * aacSpectralDataResilienceFlag | |
303 */ | |
304 break; | |
305 } | |
306 skip_bits1(gb); // extensionFlag3 (TBD in version 3) | |
307 } | |
308 return 0; | |
309 } | |
310 | |
311 /** | |
312 * Decode audio specific configuration; reference: table 1.13. | |
313 * | |
314 * @param data pointer to AVCodecContext extradata | |
315 * @param data_size size of AVCCodecContext extradata | |
316 * | |
317 * @return Returns error status. 0 - OK, !0 - error | |
318 */ | |
319 static int decode_audio_specific_config(AACContext * ac, void *data, int data_size) { | |
320 GetBitContext gb; | |
321 int i; | |
322 | |
323 init_get_bits(&gb, data, data_size * 8); | |
324 | |
325 if((i = ff_mpeg4audio_get_config(&ac->m4ac, data, data_size)) < 0) | |
326 return -1; | |
327 if(ac->m4ac.sampling_index > 11) { | |
328 av_log(ac->avccontext, AV_LOG_ERROR, "invalid sampling rate index %d\n", ac->m4ac.sampling_index); | |
329 return -1; | |
330 } | |
331 | |
332 skip_bits_long(&gb, i); | |
333 | |
334 switch (ac->m4ac.object_type) { | |
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335 case AOT_AAC_MAIN: |
7539 | 336 case AOT_AAC_LC: |
337 if (decode_ga_specific_config(ac, &gb, ac->m4ac.chan_config)) | |
338 return -1; | |
339 break; | |
340 default: | |
341 av_log(ac->avccontext, AV_LOG_ERROR, "Audio object type %s%d is not supported.\n", | |
342 ac->m4ac.sbr == 1? "SBR+" : "", ac->m4ac.object_type); | |
343 return -1; | |
344 } | |
345 return 0; | |
346 } | |
347 | |
7578 | 348 /** |
349 * linear congruential pseudorandom number generator | |
350 * | |
351 * @param previous_val pointer to the current state of the generator | |
352 * | |
353 * @return Returns a 32-bit pseudorandom integer | |
354 */ | |
355 static av_always_inline int lcg_random(int previous_val) { | |
356 return previous_val * 1664525 + 1013904223; | |
357 } | |
358 | |
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359 static void reset_predict_state(PredictorState * ps) { |
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360 ps->r0 = 0.0f; |
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361 ps->r1 = 0.0f; |
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362 ps->cor0 = 0.0f; |
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363 ps->cor1 = 0.0f; |
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364 ps->var0 = 1.0f; |
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365 ps->var1 = 1.0f; |
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366 } |
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367 |
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368 static void reset_all_predictors(PredictorState * ps) { |
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369 int i; |
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370 for (i = 0; i < MAX_PREDICTORS; i++) |
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371 reset_predict_state(&ps[i]); |
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372 } |
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373 |
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374 static void reset_predictor_group(PredictorState * ps, int group_num) { |
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375 int i; |
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376 for (i = group_num-1; i < MAX_PREDICTORS; i+=30) |
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377 reset_predict_state(&ps[i]); |
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378 } |
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379 |
7501 | 380 static av_cold int aac_decode_init(AVCodecContext * avccontext) { |
381 AACContext * ac = avccontext->priv_data; | |
382 int i; | |
383 | |
384 ac->avccontext = avccontext; | |
385 | |
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386 if (avccontext->extradata_size <= 0 || |
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387 decode_audio_specific_config(ac, avccontext->extradata, avccontext->extradata_size)) |
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388 return -1; |
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389 |
7539 | 390 avccontext->sample_fmt = SAMPLE_FMT_S16; |
7501 | 391 avccontext->sample_rate = ac->m4ac.sample_rate; |
392 avccontext->frame_size = 1024; | |
393 | |
394 AAC_INIT_VLC_STATIC( 0, 144); | |
395 AAC_INIT_VLC_STATIC( 1, 114); | |
396 AAC_INIT_VLC_STATIC( 2, 188); | |
397 AAC_INIT_VLC_STATIC( 3, 180); | |
398 AAC_INIT_VLC_STATIC( 4, 172); | |
399 AAC_INIT_VLC_STATIC( 5, 140); | |
400 AAC_INIT_VLC_STATIC( 6, 168); | |
401 AAC_INIT_VLC_STATIC( 7, 114); | |
402 AAC_INIT_VLC_STATIC( 8, 262); | |
403 AAC_INIT_VLC_STATIC( 9, 248); | |
404 AAC_INIT_VLC_STATIC(10, 384); | |
405 | |
406 dsputil_init(&ac->dsp, avccontext); | |
407 | |
7539 | 408 ac->random_state = 0x1f2e3d4c; |
409 | |
7501 | 410 // -1024 - Compensate wrong IMDCT method. |
411 // 32768 - Required to scale values to the correct range for the bias method | |
412 // for float to int16 conversion. | |
413 | |
414 if(ac->dsp.float_to_int16 == ff_float_to_int16_c) { | |
415 ac->add_bias = 385.0f; | |
416 ac->sf_scale = 1. / (-1024. * 32768.); | |
417 ac->sf_offset = 0; | |
418 } else { | |
419 ac->add_bias = 0.0f; | |
420 ac->sf_scale = 1. / -1024.; | |
421 ac->sf_offset = 60; | |
422 } | |
423 | |
424 #ifndef CONFIG_HARDCODED_TABLES | |
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425 for (i = 0; i < 428; i++) |
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426 ff_aac_pow2sf_tab[i] = pow(2, (i - 200)/4.); |
7501 | 427 #endif /* CONFIG_HARDCODED_TABLES */ |
428 | |
8042 | 429 INIT_VLC_STATIC(&vlc_scalefactors,7,FF_ARRAY_ELEMS(ff_aac_scalefactor_code), |
7501 | 430 ff_aac_scalefactor_bits, sizeof(ff_aac_scalefactor_bits[0]), sizeof(ff_aac_scalefactor_bits[0]), |
431 ff_aac_scalefactor_code, sizeof(ff_aac_scalefactor_code[0]), sizeof(ff_aac_scalefactor_code[0]), | |
432 352); | |
433 | |
434 ff_mdct_init(&ac->mdct, 11, 1); | |
435 ff_mdct_init(&ac->mdct_small, 8, 1); | |
7581 | 436 // window initialization |
437 ff_kbd_window_init(ff_aac_kbd_long_1024, 4.0, 1024); | |
438 ff_kbd_window_init(ff_aac_kbd_short_128, 6.0, 128); | |
439 ff_sine_window_init(ff_sine_1024, 1024); | |
440 ff_sine_window_init(ff_sine_128, 128); | |
441 | |
7501 | 442 return 0; |
443 } | |
444 | |
7539 | 445 /** |
446 * Skip data_stream_element; reference: table 4.10. | |
447 */ | |
448 static void skip_data_stream_element(GetBitContext * gb) { | |
7501 | 449 int byte_align = get_bits1(gb); |
450 int count = get_bits(gb, 8); | |
451 if (count == 255) | |
452 count += get_bits(gb, 8); | |
453 if (byte_align) | |
454 align_get_bits(gb); | |
455 skip_bits_long(gb, 8 * count); | |
456 } | |
457 | |
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458 static int decode_prediction(AACContext * ac, IndividualChannelStream * ics, GetBitContext * gb) { |
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459 int sfb; |
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460 if (get_bits1(gb)) { |
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461 ics->predictor_reset_group = get_bits(gb, 5); |
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462 if (ics->predictor_reset_group == 0 || ics->predictor_reset_group > 30) { |
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463 av_log(ac->avccontext, AV_LOG_ERROR, "Invalid Predictor Reset Group.\n"); |
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464 return -1; |
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465 } |
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466 } |
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467 for (sfb = 0; sfb < FFMIN(ics->max_sfb, ff_aac_pred_sfb_max[ac->m4ac.sampling_index]); sfb++) { |
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468 ics->prediction_used[sfb] = get_bits1(gb); |
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469 } |
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470 return 0; |
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471 } |
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472 |
7501 | 473 /** |
7539 | 474 * Decode Individual Channel Stream info; reference: table 4.6. |
475 * | |
476 * @param common_window Channels have independent [0], or shared [1], Individual Channel Stream information. | |
477 */ | |
478 static int decode_ics_info(AACContext * ac, IndividualChannelStream * ics, GetBitContext * gb, int common_window) { | |
479 if (get_bits1(gb)) { | |
480 av_log(ac->avccontext, AV_LOG_ERROR, "Reserved bit set.\n"); | |
481 memset(ics, 0, sizeof(IndividualChannelStream)); | |
482 return -1; | |
483 } | |
484 ics->window_sequence[1] = ics->window_sequence[0]; | |
485 ics->window_sequence[0] = get_bits(gb, 2); | |
486 ics->use_kb_window[1] = ics->use_kb_window[0]; | |
487 ics->use_kb_window[0] = get_bits1(gb); | |
488 ics->num_window_groups = 1; | |
489 ics->group_len[0] = 1; | |
7581 | 490 if (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) { |
491 int i; | |
492 ics->max_sfb = get_bits(gb, 4); | |
493 for (i = 0; i < 7; i++) { | |
494 if (get_bits1(gb)) { | |
495 ics->group_len[ics->num_window_groups-1]++; | |
496 } else { | |
497 ics->num_window_groups++; | |
498 ics->group_len[ics->num_window_groups-1] = 1; | |
499 } | |
500 } | |
501 ics->num_windows = 8; | |
502 ics->swb_offset = swb_offset_128[ac->m4ac.sampling_index]; | |
503 ics->num_swb = ff_aac_num_swb_128[ac->m4ac.sampling_index]; | |
504 ics->tns_max_bands = tns_max_bands_128[ac->m4ac.sampling_index]; | |
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505 ics->predictor_present = 0; |
7581 | 506 } else { |
507 ics->max_sfb = get_bits(gb, 6); | |
508 ics->num_windows = 1; | |
509 ics->swb_offset = swb_offset_1024[ac->m4ac.sampling_index]; | |
510 ics->num_swb = ff_aac_num_swb_1024[ac->m4ac.sampling_index]; | |
511 ics->tns_max_bands = tns_max_bands_1024[ac->m4ac.sampling_index]; | |
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512 ics->predictor_present = get_bits1(gb); |
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513 ics->predictor_reset_group = 0; |
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514 if (ics->predictor_present) { |
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515 if (ac->m4ac.object_type == AOT_AAC_MAIN) { |
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516 if (decode_prediction(ac, ics, gb)) { |
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517 memset(ics, 0, sizeof(IndividualChannelStream)); |
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518 return -1; |
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519 } |
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520 } else if (ac->m4ac.object_type == AOT_AAC_LC) { |
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521 av_log(ac->avccontext, AV_LOG_ERROR, "Prediction is not allowed in AAC-LC.\n"); |
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522 memset(ics, 0, sizeof(IndividualChannelStream)); |
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523 return -1; |
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524 } else { |
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525 ff_log_missing_feature(ac->avccontext, "Predictor bit set but LTP is", 1); |
8200 | 526 memset(ics, 0, sizeof(IndividualChannelStream)); |
527 return -1; | |
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528 } |
7578 | 529 } |
530 } | |
531 | |
532 if(ics->max_sfb > ics->num_swb) { | |
533 av_log(ac->avccontext, AV_LOG_ERROR, | |
534 "Number of scalefactor bands in group (%d) exceeds limit (%d).\n", | |
535 ics->max_sfb, ics->num_swb); | |
536 memset(ics, 0, sizeof(IndividualChannelStream)); | |
537 return -1; | |
538 } | |
539 | |
7539 | 540 return 0; |
541 } | |
542 | |
543 /** | |
544 * Decode band types (section_data payload); reference: table 4.46. | |
545 * | |
546 * @param band_type array of the used band type | |
547 * @param band_type_run_end array of the last scalefactor band of a band type run | |
548 * | |
549 * @return Returns error status. 0 - OK, !0 - error | |
550 */ | |
551 static int decode_band_types(AACContext * ac, enum BandType band_type[120], | |
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552 int band_type_run_end[120], GetBitContext * gb, IndividualChannelStream * ics) { |
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553 int g, idx = 0; |
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554 const int bits = (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) ? 3 : 5; |
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555 for (g = 0; g < ics->num_window_groups; g++) { |
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556 int k = 0; |
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557 while (k < ics->max_sfb) { |
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558 uint8_t sect_len = k; |
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559 int sect_len_incr; |
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560 int sect_band_type = get_bits(gb, 4); |
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561 if (sect_band_type == 12) { |
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562 av_log(ac->avccontext, AV_LOG_ERROR, "invalid band type\n"); |
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563 return -1; |
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564 } |
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565 while ((sect_len_incr = get_bits(gb, bits)) == (1 << bits)-1) |
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566 sect_len += sect_len_incr; |
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567 sect_len += sect_len_incr; |
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568 if (sect_len > ics->max_sfb) { |
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569 av_log(ac->avccontext, AV_LOG_ERROR, |
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570 "Number of bands (%d) exceeds limit (%d).\n", |
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571 sect_len, ics->max_sfb); |
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572 return -1; |
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573 } |
7581 | 574 for (; k < sect_len; k++) { |
575 band_type [idx] = sect_band_type; | |
576 band_type_run_end[idx++] = sect_len; | |
577 } | |
7539 | 578 } |
579 } | |
580 return 0; | |
581 } | |
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582 |
7539 | 583 /** |
584 * Decode scalefactors; reference: table 4.47. | |
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585 * |
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586 * @param global_gain first scalefactor value as scalefactors are differentially coded |
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587 * @param band_type array of the used band type |
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588 * @param band_type_run_end array of the last scalefactor band of a band type run |
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589 * @param sf array of scalefactors or intensity stereo positions |
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590 * |
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591 * @return Returns error status. 0 - OK, !0 - error |
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592 */ |
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593 static int decode_scalefactors(AACContext * ac, float sf[120], GetBitContext * gb, |
7540 | 594 unsigned int global_gain, IndividualChannelStream * ics, |
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595 enum BandType band_type[120], int band_type_run_end[120]) { |
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596 const int sf_offset = ac->sf_offset + (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE ? 12 : 0); |
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597 int g, i, idx = 0; |
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598 int offset[3] = { global_gain, global_gain - 90, 100 }; |
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599 int noise_flag = 1; |
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600 static const char *sf_str[3] = { "Global gain", "Noise gain", "Intensity stereo position" }; |
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601 for (g = 0; g < ics->num_window_groups; g++) { |
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602 for (i = 0; i < ics->max_sfb;) { |
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603 int run_end = band_type_run_end[idx]; |
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604 if (band_type[idx] == ZERO_BT) { |
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605 for(; i < run_end; i++, idx++) |
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606 sf[idx] = 0.; |
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607 }else if((band_type[idx] == INTENSITY_BT) || (band_type[idx] == INTENSITY_BT2)) { |
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608 for(; i < run_end; i++, idx++) { |
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609 offset[2] += get_vlc2(gb, vlc_scalefactors.table, 7, 3) - 60; |
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610 if(offset[2] > 255U) { |
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611 av_log(ac->avccontext, AV_LOG_ERROR, |
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612 "%s (%d) out of range.\n", sf_str[2], offset[2]); |
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613 return -1; |
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614 } |
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615 sf[idx] = ff_aac_pow2sf_tab[-offset[2] + 300]; |
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616 } |
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617 }else if(band_type[idx] == NOISE_BT) { |
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618 for(; i < run_end; i++, idx++) { |
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619 if(noise_flag-- > 0) |
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620 offset[1] += get_bits(gb, 9) - 256; |
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621 else |
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622 offset[1] += get_vlc2(gb, vlc_scalefactors.table, 7, 3) - 60; |
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623 if(offset[1] > 255U) { |
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624 av_log(ac->avccontext, AV_LOG_ERROR, |
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625 "%s (%d) out of range.\n", sf_str[1], offset[1]); |
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626 return -1; |
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627 } |
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628 sf[idx] = -ff_aac_pow2sf_tab[ offset[1] + sf_offset + 100]; |
7523
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629 } |
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630 }else { |
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631 for(; i < run_end; i++, idx++) { |
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632 offset[0] += get_vlc2(gb, vlc_scalefactors.table, 7, 3) - 60; |
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633 if(offset[0] > 255U) { |
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634 av_log(ac->avccontext, AV_LOG_ERROR, |
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635 "%s (%d) out of range.\n", sf_str[0], offset[0]); |
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636 return -1; |
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637 } |
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638 sf[idx] = -ff_aac_pow2sf_tab[ offset[0] + sf_offset]; |
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639 } |
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640 } |
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641 } |
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642 } |
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643 return 0; |
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644 } |
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645 |
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646 /** |
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647 * Decode pulse data; reference: table 4.7. |
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648 */ |
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649 static int decode_pulses(Pulse * pulse, GetBitContext * gb, const uint16_t * swb_offset, int num_swb) { |
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650 int i, pulse_swb; |
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651 pulse->num_pulse = get_bits(gb, 2) + 1; |
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652 pulse_swb = get_bits(gb, 6); |
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653 if (pulse_swb >= num_swb) |
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654 return -1; |
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655 pulse->pos[0] = swb_offset[pulse_swb]; |
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656 pulse->pos[0] += get_bits(gb, 5); |
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657 if (pulse->pos[0] > 1023) |
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658 return -1; |
7580 | 659 pulse->amp[0] = get_bits(gb, 4); |
660 for (i = 1; i < pulse->num_pulse; i++) { | |
661 pulse->pos[i] = get_bits(gb, 5) + pulse->pos[i-1]; | |
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662 if (pulse->pos[i] > 1023) |
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663 return -1; |
7580 | 664 pulse->amp[i] = get_bits(gb, 4); |
7523
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665 } |
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666 return 0; |
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667 } |
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668 |
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669 /** |
7609 | 670 * Decode Temporal Noise Shaping data; reference: table 4.48. |
671 * | |
672 * @return Returns error status. 0 - OK, !0 - error | |
673 */ | |
674 static int decode_tns(AACContext * ac, TemporalNoiseShaping * tns, | |
675 GetBitContext * gb, const IndividualChannelStream * ics) { | |
676 int w, filt, i, coef_len, coef_res, coef_compress; | |
677 const int is8 = ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE; | |
678 const int tns_max_order = is8 ? 7 : ac->m4ac.object_type == AOT_AAC_MAIN ? 20 : 12; | |
679 for (w = 0; w < ics->num_windows; w++) { | |
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680 if ((tns->n_filt[w] = get_bits(gb, 2 - is8))) { |
7609 | 681 coef_res = get_bits1(gb); |
682 | |
7638 | 683 for (filt = 0; filt < tns->n_filt[w]; filt++) { |
684 int tmp2_idx; | |
685 tns->length[w][filt] = get_bits(gb, 6 - 2*is8); | |
7609 | 686 |
7638 | 687 if ((tns->order[w][filt] = get_bits(gb, 5 - 2*is8)) > tns_max_order) { |
688 av_log(ac->avccontext, AV_LOG_ERROR, "TNS filter order %d is greater than maximum %d.", | |
689 tns->order[w][filt], tns_max_order); | |
690 tns->order[w][filt] = 0; | |
691 return -1; | |
692 } | |
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693 if (tns->order[w][filt]) { |
7851 | 694 tns->direction[w][filt] = get_bits1(gb); |
695 coef_compress = get_bits1(gb); | |
696 coef_len = coef_res + 3 - coef_compress; | |
697 tmp2_idx = 2*coef_compress + coef_res; | |
7638 | 698 |
7851 | 699 for (i = 0; i < tns->order[w][filt]; i++) |
700 tns->coef[w][filt][i] = tns_tmp2_map[tmp2_idx][get_bits(gb, coef_len)]; | |
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701 } |
7609 | 702 } |
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703 } |
7609 | 704 } |
705 return 0; | |
706 } | |
707 | |
708 /** | |
7539 | 709 * Decode Mid/Side data; reference: table 4.54. |
710 * | |
711 * @param ms_present Indicates mid/side stereo presence. [0] mask is all 0s; | |
712 * [1] mask is decoded from bitstream; [2] mask is all 1s; | |
713 * [3] reserved for scalable AAC | |
714 */ | |
715 static void decode_mid_side_stereo(ChannelElement * cpe, GetBitContext * gb, | |
716 int ms_present) { | |
7578 | 717 int idx; |
718 if (ms_present == 1) { | |
719 for (idx = 0; idx < cpe->ch[0].ics.num_window_groups * cpe->ch[0].ics.max_sfb; idx++) | |
720 cpe->ms_mask[idx] = get_bits1(gb); | |
721 } else if (ms_present == 2) { | |
722 memset(cpe->ms_mask, 1, cpe->ch[0].ics.num_window_groups * cpe->ch[0].ics.max_sfb * sizeof(cpe->ms_mask[0])); | |
723 } | |
724 } | |
7539 | 725 |
726 /** | |
7581 | 727 * Decode spectral data; reference: table 4.50. |
728 * Dequantize and scale spectral data; reference: 4.6.3.3. | |
729 * | |
730 * @param coef array of dequantized, scaled spectral data | |
731 * @param sf array of scalefactors or intensity stereo positions | |
732 * @param pulse_present set if pulses are present | |
733 * @param pulse pointer to pulse data struct | |
734 * @param band_type array of the used band type | |
735 * | |
736 * @return Returns error status. 0 - OK, !0 - error | |
737 */ | |
738 static int decode_spectrum_and_dequant(AACContext * ac, float coef[1024], GetBitContext * gb, float sf[120], | |
739 int pulse_present, const Pulse * pulse, const IndividualChannelStream * ics, enum BandType band_type[120]) { | |
740 int i, k, g, idx = 0; | |
741 const int c = 1024/ics->num_windows; | |
742 const uint16_t * offsets = ics->swb_offset; | |
743 float *coef_base = coef; | |
8284 | 744 static const float sign_lookup[] = { 1.0f, -1.0f }; |
7581 | 745 |
746 for (g = 0; g < ics->num_windows; g++) | |
747 memset(coef + g * 128 + offsets[ics->max_sfb], 0, sizeof(float)*(c - offsets[ics->max_sfb])); | |
748 | |
749 for (g = 0; g < ics->num_window_groups; g++) { | |
750 for (i = 0; i < ics->max_sfb; i++, idx++) { | |
751 const int cur_band_type = band_type[idx]; | |
752 const int dim = cur_band_type >= FIRST_PAIR_BT ? 2 : 4; | |
753 const int is_cb_unsigned = IS_CODEBOOK_UNSIGNED(cur_band_type); | |
754 int group; | |
755 if (cur_band_type == ZERO_BT) { | |
756 for (group = 0; group < ics->group_len[g]; group++) { | |
757 memset(coef + group * 128 + offsets[i], 0, (offsets[i+1] - offsets[i])*sizeof(float)); | |
758 } | |
759 }else if (cur_band_type == NOISE_BT) { | |
760 for (group = 0; group < ics->group_len[g]; group++) { | |
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761 float scale; |
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762 float band_energy = 0; |
7581 | 763 for (k = offsets[i]; k < offsets[i+1]; k++) { |
764 ac->random_state = lcg_random(ac->random_state); | |
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765 coef[group*128+k] = ac->random_state; |
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766 band_energy += coef[group*128+k]*coef[group*128+k]; |
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767 } |
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768 scale = sf[idx] / sqrtf(band_energy); |
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769 for (k = offsets[i]; k < offsets[i+1]; k++) { |
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770 coef[group*128+k] *= scale; |
7581 | 771 } |
772 } | |
773 }else if (cur_band_type != INTENSITY_BT2 && cur_band_type != INTENSITY_BT) { | |
774 for (group = 0; group < ics->group_len[g]; group++) { | |
775 for (k = offsets[i]; k < offsets[i+1]; k += dim) { | |
776 const int index = get_vlc2(gb, vlc_spectral[cur_band_type - 1].table, 6, 3); | |
777 const int coef_tmp_idx = (group << 7) + k; | |
778 const float *vq_ptr; | |
779 int j; | |
780 if(index >= ff_aac_spectral_sizes[cur_band_type - 1]) { | |
781 av_log(ac->avccontext, AV_LOG_ERROR, | |
782 "Read beyond end of ff_aac_codebook_vectors[%d][]. index %d >= %d\n", | |
783 cur_band_type - 1, index, ff_aac_spectral_sizes[cur_band_type - 1]); | |
784 return -1; | |
785 } | |
786 vq_ptr = &ff_aac_codebook_vectors[cur_band_type - 1][index * dim]; | |
787 if (is_cb_unsigned) { | |
8284 | 788 if (vq_ptr[0]) coef[coef_tmp_idx ] = sign_lookup[get_bits1(gb)]; |
789 if (vq_ptr[1]) coef[coef_tmp_idx + 1] = sign_lookup[get_bits1(gb)]; | |
8283
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790 if (dim == 4) { |
8284 | 791 if (vq_ptr[2]) coef[coef_tmp_idx + 2] = sign_lookup[get_bits1(gb)]; |
792 if (vq_ptr[3]) coef[coef_tmp_idx + 3] = sign_lookup[get_bits1(gb)]; | |
8283
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793 } |
7581 | 794 }else { |
8283
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795 coef[coef_tmp_idx ] = 1.0f; |
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796 coef[coef_tmp_idx + 1] = 1.0f; |
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797 if (dim == 4) { |
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798 coef[coef_tmp_idx + 2] = 1.0f; |
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799 coef[coef_tmp_idx + 3] = 1.0f; |
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800 } |
7581 | 801 } |
802 if (cur_band_type == ESC_BT) { | |
803 for (j = 0; j < 2; j++) { | |
804 if (vq_ptr[j] == 64.0f) { | |
805 int n = 4; | |
806 /* The total length of escape_sequence must be < 22 bits according | |
807 to the specification (i.e. max is 11111111110xxxxxxxxxx). */ | |
808 while (get_bits1(gb) && n < 15) n++; | |
809 if(n == 15) { | |
810 av_log(ac->avccontext, AV_LOG_ERROR, "error in spectral data, ESC overflow\n"); | |
811 return -1; | |
812 } | |
813 n = (1<<n) + get_bits(gb, n); | |
8282 | 814 coef[coef_tmp_idx + j] *= cbrtf(n) * n; |
7581 | 815 }else |
816 coef[coef_tmp_idx + j] *= vq_ptr[j]; | |
817 } | |
818 }else | |
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819 { |
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820 coef[coef_tmp_idx ] *= vq_ptr[0]; |
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821 coef[coef_tmp_idx + 1] *= vq_ptr[1]; |
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822 if (dim == 4) { |
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823 coef[coef_tmp_idx + 2] *= vq_ptr[2]; |
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824 coef[coef_tmp_idx + 3] *= vq_ptr[3]; |
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825 } |
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826 } |
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827 coef[coef_tmp_idx ] *= sf[idx]; |
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828 coef[coef_tmp_idx + 1] *= sf[idx]; |
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829 if (dim == 4) { |
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830 coef[coef_tmp_idx + 2] *= sf[idx]; |
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|
831 coef[coef_tmp_idx + 3] *= sf[idx]; |
ea8e4095d980
AAC: Unroll code within decode_spectrum_and_dequant(). Gives 12% net speed gain.
superdump
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diff
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|
832 } |
7581 | 833 } |
834 } | |
835 } | |
836 } | |
837 coef += ics->group_len[g]<<7; | |
838 } | |
839 | |
840 if (pulse_present) { | |
7857
d7801471cd52
Apply pulses to unscaled coefficients rather than scaled.
superdump
parents:
7856
diff
changeset
|
841 idx = 0; |
7581 | 842 for(i = 0; i < pulse->num_pulse; i++){ |
843 float co = coef_base[ pulse->pos[i] ]; | |
7857
d7801471cd52
Apply pulses to unscaled coefficients rather than scaled.
superdump
parents:
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diff
changeset
|
844 while(offsets[idx + 1] <= pulse->pos[i]) |
d7801471cd52
Apply pulses to unscaled coefficients rather than scaled.
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parents:
7856
diff
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|
845 idx++; |
d7801471cd52
Apply pulses to unscaled coefficients rather than scaled.
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parents:
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diff
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|
846 if (band_type[idx] != NOISE_BT && sf[idx]) { |
7858 | 847 float ico = -pulse->amp[i]; |
848 if (co) { | |
849 co /= sf[idx]; | |
850 ico = co / sqrtf(sqrtf(fabsf(co))) + (co > 0 ? -ico : ico); | |
851 } | |
852 coef_base[ pulse->pos[i] ] = cbrtf(fabsf(ico)) * ico * sf[idx]; | |
7857
d7801471cd52
Apply pulses to unscaled coefficients rather than scaled.
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diff
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|
853 } |
7581 | 854 } |
855 } | |
856 return 0; | |
857 } | |
858 | |
8199
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AAC: Frequency domain prediction and hence Main profile support
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|
859 static av_always_inline float flt16_round(float pf) { |
8a8d40fe77d9
AAC: Frequency domain prediction and hence Main profile support
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diff
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|
860 int exp; |
8a8d40fe77d9
AAC: Frequency domain prediction and hence Main profile support
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diff
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|
861 pf = frexpf(pf, &exp); |
8a8d40fe77d9
AAC: Frequency domain prediction and hence Main profile support
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|
862 pf = ldexpf(roundf(ldexpf(pf, 8)), exp-8); |
8a8d40fe77d9
AAC: Frequency domain prediction and hence Main profile support
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|
863 return pf; |
8a8d40fe77d9
AAC: Frequency domain prediction and hence Main profile support
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diff
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|
864 } |
8a8d40fe77d9
AAC: Frequency domain prediction and hence Main profile support
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|
865 |
8a8d40fe77d9
AAC: Frequency domain prediction and hence Main profile support
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|
866 static av_always_inline float flt16_even(float pf) { |
8a8d40fe77d9
AAC: Frequency domain prediction and hence Main profile support
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|
867 int exp; |
8a8d40fe77d9
AAC: Frequency domain prediction and hence Main profile support
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diff
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|
868 pf = frexpf(pf, &exp); |
8a8d40fe77d9
AAC: Frequency domain prediction and hence Main profile support
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diff
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|
869 pf = ldexpf(rintf(ldexpf(pf, 8)), exp-8); |
8a8d40fe77d9
AAC: Frequency domain prediction and hence Main profile support
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|
870 return pf; |
8a8d40fe77d9
AAC: Frequency domain prediction and hence Main profile support
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diff
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|
871 } |
8a8d40fe77d9
AAC: Frequency domain prediction and hence Main profile support
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|
872 |
8a8d40fe77d9
AAC: Frequency domain prediction and hence Main profile support
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|
873 static av_always_inline float flt16_trunc(float pf) { |
8a8d40fe77d9
AAC: Frequency domain prediction and hence Main profile support
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diff
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|
874 int exp; |
8a8d40fe77d9
AAC: Frequency domain prediction and hence Main profile support
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|
875 pf = frexpf(pf, &exp); |
8a8d40fe77d9
AAC: Frequency domain prediction and hence Main profile support
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|
876 pf = ldexpf(truncf(ldexpf(pf, 8)), exp-8); |
8a8d40fe77d9
AAC: Frequency domain prediction and hence Main profile support
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|
877 return pf; |
8a8d40fe77d9
AAC: Frequency domain prediction and hence Main profile support
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diff
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|
878 } |
8a8d40fe77d9
AAC: Frequency domain prediction and hence Main profile support
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|
879 |
8a8d40fe77d9
AAC: Frequency domain prediction and hence Main profile support
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|
880 static void predict(AACContext * ac, PredictorState * ps, float* coef, int output_enable) { |
8a8d40fe77d9
AAC: Frequency domain prediction and hence Main profile support
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|
881 const float a = 0.953125; // 61.0/64 |
8a8d40fe77d9
AAC: Frequency domain prediction and hence Main profile support
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|
882 const float alpha = 0.90625; // 29.0/32 |
8a8d40fe77d9
AAC: Frequency domain prediction and hence Main profile support
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|
883 float e0, e1; |
8a8d40fe77d9
AAC: Frequency domain prediction and hence Main profile support
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|
884 float pv; |
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AAC: Frequency domain prediction and hence Main profile support
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|
885 float k1, k2; |
8a8d40fe77d9
AAC: Frequency domain prediction and hence Main profile support
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|
886 |
8a8d40fe77d9
AAC: Frequency domain prediction and hence Main profile support
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|
887 k1 = ps->var0 > 1 ? ps->cor0 * flt16_even(a / ps->var0) : 0; |
8a8d40fe77d9
AAC: Frequency domain prediction and hence Main profile support
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|
888 k2 = ps->var1 > 1 ? ps->cor1 * flt16_even(a / ps->var1) : 0; |
8a8d40fe77d9
AAC: Frequency domain prediction and hence Main profile support
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|
889 |
8a8d40fe77d9
AAC: Frequency domain prediction and hence Main profile support
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|
890 pv = flt16_round(k1 * ps->r0 + k2 * ps->r1); |
8a8d40fe77d9
AAC: Frequency domain prediction and hence Main profile support
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|
891 if (output_enable) |
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AAC: Frequency domain prediction and hence Main profile support
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|
892 *coef += pv * ac->sf_scale; |
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AAC: Frequency domain prediction and hence Main profile support
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diff
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|
893 |
8a8d40fe77d9
AAC: Frequency domain prediction and hence Main profile support
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|
894 e0 = *coef / ac->sf_scale; |
8a8d40fe77d9
AAC: Frequency domain prediction and hence Main profile support
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|
895 e1 = e0 - k1 * ps->r0; |
8a8d40fe77d9
AAC: Frequency domain prediction and hence Main profile support
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diff
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|
896 |
8a8d40fe77d9
AAC: Frequency domain prediction and hence Main profile support
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|
897 ps->cor1 = flt16_trunc(alpha * ps->cor1 + ps->r1 * e1); |
8a8d40fe77d9
AAC: Frequency domain prediction and hence Main profile support
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|
898 ps->var1 = flt16_trunc(alpha * ps->var1 + 0.5 * (ps->r1 * ps->r1 + e1 * e1)); |
8a8d40fe77d9
AAC: Frequency domain prediction and hence Main profile support
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|
899 ps->cor0 = flt16_trunc(alpha * ps->cor0 + ps->r0 * e0); |
8a8d40fe77d9
AAC: Frequency domain prediction and hence Main profile support
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|
900 ps->var0 = flt16_trunc(alpha * ps->var0 + 0.5 * (ps->r0 * ps->r0 + e0 * e0)); |
8a8d40fe77d9
AAC: Frequency domain prediction and hence Main profile support
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|
901 |
8a8d40fe77d9
AAC: Frequency domain prediction and hence Main profile support
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|
902 ps->r1 = flt16_trunc(a * (ps->r0 - k1 * e0)); |
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AAC: Frequency domain prediction and hence Main profile support
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|
903 ps->r0 = flt16_trunc(a * e0); |
8a8d40fe77d9
AAC: Frequency domain prediction and hence Main profile support
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diff
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|
904 } |
8a8d40fe77d9
AAC: Frequency domain prediction and hence Main profile support
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|
905 |
8a8d40fe77d9
AAC: Frequency domain prediction and hence Main profile support
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diff
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|
906 /** |
8a8d40fe77d9
AAC: Frequency domain prediction and hence Main profile support
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diff
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|
907 * Apply AAC-Main style frequency domain prediction. |
8a8d40fe77d9
AAC: Frequency domain prediction and hence Main profile support
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diff
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|
908 */ |
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AAC: Frequency domain prediction and hence Main profile support
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|
909 static void apply_prediction(AACContext * ac, SingleChannelElement * sce) { |
8a8d40fe77d9
AAC: Frequency domain prediction and hence Main profile support
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|
910 int sfb, k; |
8a8d40fe77d9
AAC: Frequency domain prediction and hence Main profile support
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|
911 |
8a8d40fe77d9
AAC: Frequency domain prediction and hence Main profile support
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|
912 if (!sce->ics.predictor_initialized) { |
8a8d40fe77d9
AAC: Frequency domain prediction and hence Main profile support
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|
913 reset_all_predictors(sce->ics.predictor_state); |
8a8d40fe77d9
AAC: Frequency domain prediction and hence Main profile support
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|
914 sce->ics.predictor_initialized = 1; |
8a8d40fe77d9
AAC: Frequency domain prediction and hence Main profile support
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|
915 } |
8a8d40fe77d9
AAC: Frequency domain prediction and hence Main profile support
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|
916 |
8a8d40fe77d9
AAC: Frequency domain prediction and hence Main profile support
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|
917 if (sce->ics.window_sequence[0] != EIGHT_SHORT_SEQUENCE) { |
8a8d40fe77d9
AAC: Frequency domain prediction and hence Main profile support
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|
918 for (sfb = 0; sfb < ff_aac_pred_sfb_max[ac->m4ac.sampling_index]; sfb++) { |
8a8d40fe77d9
AAC: Frequency domain prediction and hence Main profile support
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|
919 for (k = sce->ics.swb_offset[sfb]; k < sce->ics.swb_offset[sfb + 1]; k++) { |
8a8d40fe77d9
AAC: Frequency domain prediction and hence Main profile support
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|
920 predict(ac, &sce->ics.predictor_state[k], &sce->coeffs[k], |
8a8d40fe77d9
AAC: Frequency domain prediction and hence Main profile support
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|
921 sce->ics.predictor_present && sce->ics.prediction_used[sfb]); |
8a8d40fe77d9
AAC: Frequency domain prediction and hence Main profile support
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|
922 } |
8a8d40fe77d9
AAC: Frequency domain prediction and hence Main profile support
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|
923 } |
8a8d40fe77d9
AAC: Frequency domain prediction and hence Main profile support
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|
924 if (sce->ics.predictor_reset_group) |
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AAC: Frequency domain prediction and hence Main profile support
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|
925 reset_predictor_group(sce->ics.predictor_state, sce->ics.predictor_reset_group); |
8a8d40fe77d9
AAC: Frequency domain prediction and hence Main profile support
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|
926 } else |
8a8d40fe77d9
AAC: Frequency domain prediction and hence Main profile support
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|
927 reset_all_predictors(sce->ics.predictor_state); |
8a8d40fe77d9
AAC: Frequency domain prediction and hence Main profile support
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|
928 } |
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AAC: Frequency domain prediction and hence Main profile support
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|
929 |
7581 | 930 /** |
7539 | 931 * Decode an individual_channel_stream payload; reference: table 4.44. |
932 * | |
933 * @param common_window Channels have independent [0], or shared [1], Individual Channel Stream information. | |
934 * @param scale_flag scalable [1] or non-scalable [0] AAC (Unused until scalable AAC is implemented.) | |
935 * | |
936 * @return Returns error status. 0 - OK, !0 - error | |
937 */ | |
938 static int decode_ics(AACContext * ac, SingleChannelElement * sce, GetBitContext * gb, int common_window, int scale_flag) { | |
939 Pulse pulse; | |
940 TemporalNoiseShaping * tns = &sce->tns; | |
941 IndividualChannelStream * ics = &sce->ics; | |
942 float * out = sce->coeffs; | |
943 int global_gain, pulse_present = 0; | |
944 | |
7580 | 945 /* This assignment is to silence a GCC warning about the variable being used |
946 * uninitialized when in fact it always is. | |
7539 | 947 */ |
948 pulse.num_pulse = 0; | |
949 | |
950 global_gain = get_bits(gb, 8); | |
951 | |
952 if (!common_window && !scale_flag) { | |
953 if (decode_ics_info(ac, ics, gb, 0) < 0) | |
954 return -1; | |
955 } | |
956 | |
957 if (decode_band_types(ac, sce->band_type, sce->band_type_run_end, gb, ics) < 0) | |
958 return -1; | |
959 if (decode_scalefactors(ac, sce->sf, gb, global_gain, ics, sce->band_type, sce->band_type_run_end) < 0) | |
960 return -1; | |
961 | |
962 pulse_present = 0; | |
963 if (!scale_flag) { | |
964 if ((pulse_present = get_bits1(gb))) { | |
965 if (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) { | |
966 av_log(ac->avccontext, AV_LOG_ERROR, "Pulse tool not allowed in eight short sequence.\n"); | |
967 return -1; | |
968 } | |
7871
8277c41b7160
Validate pulse position and error out if an invalid position is encountered.
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|
969 if (decode_pulses(&pulse, gb, ics->swb_offset, ics->num_swb)) { |
8277c41b7160
Validate pulse position and error out if an invalid position is encountered.
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|
970 av_log(ac->avccontext, AV_LOG_ERROR, "Pulse data corrupt or invalid.\n"); |
8277c41b7160
Validate pulse position and error out if an invalid position is encountered.
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diff
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|
971 return -1; |
8277c41b7160
Validate pulse position and error out if an invalid position is encountered.
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|
972 } |
7539 | 973 } |
974 if ((tns->present = get_bits1(gb)) && decode_tns(ac, tns, gb, ics)) | |
975 return -1; | |
976 if (get_bits1(gb)) { | |
8281
f93efc084e41
Make av_log_missing_feature an internal function, and change its name
stefano
parents:
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diff
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|
977 ff_log_missing_feature(ac->avccontext, "SSR", 1); |
7539 | 978 return -1; |
979 } | |
980 } | |
981 | |
7580 | 982 if (decode_spectrum_and_dequant(ac, out, gb, sce->sf, pulse_present, &pulse, ics, sce->band_type) < 0) |
7539 | 983 return -1; |
8199
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AAC: Frequency domain prediction and hence Main profile support
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|
984 |
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|
985 if(ac->m4ac.object_type == AOT_AAC_MAIN) |
8a8d40fe77d9
AAC: Frequency domain prediction and hence Main profile support
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|
986 apply_prediction(ac, sce); |
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|
987 |
7539 | 988 return 0; |
989 } | |
990 | |
991 /** | |
7581 | 992 * Mid/Side stereo decoding; reference: 4.6.8.1.3. |
993 */ | |
994 static void apply_mid_side_stereo(ChannelElement * cpe) { | |
995 const IndividualChannelStream * ics = &cpe->ch[0].ics; | |
996 float *ch0 = cpe->ch[0].coeffs; | |
997 float *ch1 = cpe->ch[1].coeffs; | |
998 int g, i, k, group, idx = 0; | |
999 const uint16_t * offsets = ics->swb_offset; | |
1000 for (g = 0; g < ics->num_window_groups; g++) { | |
1001 for (i = 0; i < ics->max_sfb; i++, idx++) { | |
1002 if (cpe->ms_mask[idx] && | |
1003 cpe->ch[0].band_type[idx] < NOISE_BT && cpe->ch[1].band_type[idx] < NOISE_BT) { | |
1004 for (group = 0; group < ics->group_len[g]; group++) { | |
1005 for (k = offsets[i]; k < offsets[i+1]; k++) { | |
1006 float tmp = ch0[group*128 + k] - ch1[group*128 + k]; | |
1007 ch0[group*128 + k] += ch1[group*128 + k]; | |
1008 ch1[group*128 + k] = tmp; | |
1009 } | |
1010 } | |
1011 } | |
1012 } | |
1013 ch0 += ics->group_len[g]*128; | |
1014 ch1 += ics->group_len[g]*128; | |
1015 } | |
1016 } | |
1017 | |
1018 /** | |
1019 * intensity stereo decoding; reference: 4.6.8.2.3 | |
1020 * | |
1021 * @param ms_present Indicates mid/side stereo presence. [0] mask is all 0s; | |
1022 * [1] mask is decoded from bitstream; [2] mask is all 1s; | |
1023 * [3] reserved for scalable AAC | |
1024 */ | |
1025 static void apply_intensity_stereo(ChannelElement * cpe, int ms_present) { | |
1026 const IndividualChannelStream * ics = &cpe->ch[1].ics; | |
1027 SingleChannelElement * sce1 = &cpe->ch[1]; | |
1028 float *coef0 = cpe->ch[0].coeffs, *coef1 = cpe->ch[1].coeffs; | |
1029 const uint16_t * offsets = ics->swb_offset; | |
1030 int g, group, i, k, idx = 0; | |
1031 int c; | |
1032 float scale; | |
1033 for (g = 0; g < ics->num_window_groups; g++) { | |
1034 for (i = 0; i < ics->max_sfb;) { | |
1035 if (sce1->band_type[idx] == INTENSITY_BT || sce1->band_type[idx] == INTENSITY_BT2) { | |
1036 const int bt_run_end = sce1->band_type_run_end[idx]; | |
1037 for (; i < bt_run_end; i++, idx++) { | |
1038 c = -1 + 2 * (sce1->band_type[idx] - 14); | |
1039 if (ms_present) | |
1040 c *= 1 - 2 * cpe->ms_mask[idx]; | |
1041 scale = c * sce1->sf[idx]; | |
1042 for (group = 0; group < ics->group_len[g]; group++) | |
1043 for (k = offsets[i]; k < offsets[i+1]; k++) | |
1044 coef1[group*128 + k] = scale * coef0[group*128 + k]; | |
1045 } | |
1046 } else { | |
1047 int bt_run_end = sce1->band_type_run_end[idx]; | |
1048 idx += bt_run_end - i; | |
1049 i = bt_run_end; | |
1050 } | |
1051 } | |
1052 coef0 += ics->group_len[g]*128; | |
1053 coef1 += ics->group_len[g]*128; | |
1054 } | |
1055 } | |
1056 | |
1057 /** | |
7539 | 1058 * Decode a channel_pair_element; reference: table 4.4. |
1059 * | |
1060 * @param elem_id Identifies the instance of a syntax element. | |
1061 * | |
1062 * @return Returns error status. 0 - OK, !0 - error | |
1063 */ | |
1064 static int decode_cpe(AACContext * ac, GetBitContext * gb, int elem_id) { | |
1065 int i, ret, common_window, ms_present = 0; | |
1066 ChannelElement * cpe; | |
1067 | |
1068 cpe = ac->che[TYPE_CPE][elem_id]; | |
1069 common_window = get_bits1(gb); | |
1070 if (common_window) { | |
1071 if (decode_ics_info(ac, &cpe->ch[0].ics, gb, 1)) | |
1072 return -1; | |
1073 i = cpe->ch[1].ics.use_kb_window[0]; | |
1074 cpe->ch[1].ics = cpe->ch[0].ics; | |
1075 cpe->ch[1].ics.use_kb_window[1] = i; | |
1076 ms_present = get_bits(gb, 2); | |
1077 if(ms_present == 3) { | |
1078 av_log(ac->avccontext, AV_LOG_ERROR, "ms_present = 3 is reserved.\n"); | |
1079 return -1; | |
1080 } else if(ms_present) | |
1081 decode_mid_side_stereo(cpe, gb, ms_present); | |
1082 } | |
1083 if ((ret = decode_ics(ac, &cpe->ch[0], gb, common_window, 0))) | |
1084 return ret; | |
1085 if ((ret = decode_ics(ac, &cpe->ch[1], gb, common_window, 0))) | |
1086 return ret; | |
1087 | |
1088 if (common_window && ms_present) | |
1089 apply_mid_side_stereo(cpe); | |
1090 | |
7580 | 1091 apply_intensity_stereo(cpe, ms_present); |
7539 | 1092 return 0; |
1093 } | |
1094 | |
7581 | 1095 /** |
1096 * Decode coupling_channel_element; reference: table 4.8. | |
1097 * | |
1098 * @param elem_id Identifies the instance of a syntax element. | |
1099 * | |
1100 * @return Returns error status. 0 - OK, !0 - error | |
1101 */ | |
1102 static int decode_cce(AACContext * ac, GetBitContext * gb, ChannelElement * che) { | |
1103 int num_gain = 0; | |
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1104 int c, g, sfb, ret; |
7581 | 1105 int sign; |
1106 float scale; | |
1107 SingleChannelElement * sce = &che->ch[0]; | |
1108 ChannelCoupling * coup = &che->coup; | |
1109 | |
7578 | 1110 coup->coupling_point = 2*get_bits1(gb); |
1111 coup->num_coupled = get_bits(gb, 3); | |
1112 for (c = 0; c <= coup->num_coupled; c++) { | |
1113 num_gain++; | |
1114 coup->type[c] = get_bits1(gb) ? TYPE_CPE : TYPE_SCE; | |
1115 coup->id_select[c] = get_bits(gb, 4); | |
1116 if (coup->type[c] == TYPE_CPE) { | |
1117 coup->ch_select[c] = get_bits(gb, 2); | |
1118 if (coup->ch_select[c] == 3) | |
1119 num_gain++; | |
1120 } else | |
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1121 coup->ch_select[c] = 2; |
7578 | 1122 } |
1123 coup->coupling_point += get_bits1(gb); | |
1124 | |
1125 if (coup->coupling_point == 2) { | |
1126 av_log(ac->avccontext, AV_LOG_ERROR, | |
1127 "Independently switched CCE with 'invalid' domain signalled.\n"); | |
1128 memset(coup, 0, sizeof(ChannelCoupling)); | |
1129 return -1; | |
1130 } | |
1131 | |
1132 sign = get_bits(gb, 1); | |
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1133 scale = pow(2., pow(2., (int)get_bits(gb, 2) - 3)); |
7578 | 1134 |
1135 if ((ret = decode_ics(ac, sce, gb, 0, 0))) | |
1136 return ret; | |
1137 | |
1138 for (c = 0; c < num_gain; c++) { | |
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1139 int idx = 0; |
7578 | 1140 int cge = 1; |
1141 int gain = 0; | |
1142 float gain_cache = 1.; | |
1143 if (c) { | |
1144 cge = coup->coupling_point == AFTER_IMDCT ? 1 : get_bits1(gb); | |
1145 gain = cge ? get_vlc2(gb, vlc_scalefactors.table, 7, 3) - 60: 0; | |
8004
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1146 gain_cache = pow(scale, -gain); |
7578 | 1147 } |
7890 | 1148 for (g = 0; g < sce->ics.num_window_groups; g++) { |
1149 for (sfb = 0; sfb < sce->ics.max_sfb; sfb++, idx++) { | |
7578 | 1150 if (sce->band_type[idx] != ZERO_BT) { |
1151 if (!cge) { | |
1152 int t = get_vlc2(gb, vlc_scalefactors.table, 7, 3) - 60; | |
1153 if (t) { | |
1154 int s = 1; | |
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1155 t = gain += t; |
7578 | 1156 if (sign) { |
1157 s -= 2 * (t & 0x1); | |
1158 t >>= 1; | |
1159 } | |
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1160 gain_cache = pow(scale, -t) * s; |
7578 | 1161 } |
1162 } | |
1163 coup->gain[c][idx] = gain_cache; | |
1164 } | |
7890 | 1165 } |
1166 } | |
7578 | 1167 } |
1168 return 0; | |
1169 } | |
1170 | |
7539 | 1171 /** |
1172 * Decode Spectral Band Replication extension data; reference: table 4.55. | |
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1173 * |
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1174 * @param crc flag indicating the presence of CRC checksum |
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1175 * @param cnt length of TYPE_FIL syntactic element in bytes |
7539 | 1176 * |
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1177 * @return Returns number of bytes consumed from the TYPE_FIL element. |
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1178 */ |
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1179 static int decode_sbr_extension(AACContext * ac, GetBitContext * gb, int crc, int cnt) { |
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1180 // TODO : sbr_extension implementation |
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1181 ff_log_missing_feature(ac->avccontext, "SBR", 0); |
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1182 skip_bits_long(gb, 8*cnt - 4); // -4 due to reading extension type |
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1183 return cnt; |
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1184 } |
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1185 |
7539 | 1186 /** |
7578 | 1187 * Parse whether channels are to be excluded from Dynamic Range Compression; reference: table 4.53. |
1188 * | |
1189 * @return Returns number of bytes consumed. | |
1190 */ | |
1191 static int decode_drc_channel_exclusions(DynamicRangeControl *che_drc, GetBitContext * gb) { | |
1192 int i; | |
1193 int num_excl_chan = 0; | |
1194 | |
1195 do { | |
1196 for (i = 0; i < 7; i++) | |
1197 che_drc->exclude_mask[num_excl_chan++] = get_bits1(gb); | |
1198 } while (num_excl_chan < MAX_CHANNELS - 7 && get_bits1(gb)); | |
1199 | |
1200 return num_excl_chan / 7; | |
1201 } | |
1202 | |
1203 /** | |
7539 | 1204 * Decode dynamic range information; reference: table 4.52. |
1205 * | |
1206 * @param cnt length of TYPE_FIL syntactic element in bytes | |
1207 * | |
1208 * @return Returns number of bytes consumed. | |
1209 */ | |
1210 static int decode_dynamic_range(DynamicRangeControl *che_drc, GetBitContext * gb, int cnt) { | |
1211 int n = 1; | |
1212 int drc_num_bands = 1; | |
1213 int i; | |
1214 | |
1215 /* pce_tag_present? */ | |
1216 if(get_bits1(gb)) { | |
1217 che_drc->pce_instance_tag = get_bits(gb, 4); | |
1218 skip_bits(gb, 4); // tag_reserved_bits | |
1219 n++; | |
1220 } | |
1221 | |
1222 /* excluded_chns_present? */ | |
1223 if(get_bits1(gb)) { | |
1224 n += decode_drc_channel_exclusions(che_drc, gb); | |
1225 } | |
1226 | |
1227 /* drc_bands_present? */ | |
1228 if (get_bits1(gb)) { | |
1229 che_drc->band_incr = get_bits(gb, 4); | |
1230 che_drc->interpolation_scheme = get_bits(gb, 4); | |
1231 n++; | |
1232 drc_num_bands += che_drc->band_incr; | |
1233 for (i = 0; i < drc_num_bands; i++) { | |
1234 che_drc->band_top[i] = get_bits(gb, 8); | |
1235 n++; | |
1236 } | |
1237 } | |
1238 | |
1239 /* prog_ref_level_present? */ | |
1240 if (get_bits1(gb)) { | |
1241 che_drc->prog_ref_level = get_bits(gb, 7); | |
1242 skip_bits1(gb); // prog_ref_level_reserved_bits | |
1243 n++; | |
1244 } | |
1245 | |
1246 for (i = 0; i < drc_num_bands; i++) { | |
1247 che_drc->dyn_rng_sgn[i] = get_bits1(gb); | |
1248 che_drc->dyn_rng_ctl[i] = get_bits(gb, 7); | |
1249 n++; | |
1250 } | |
1251 | |
1252 return n; | |
1253 } | |
1254 | |
1255 /** | |
1256 * Decode extension data (incomplete); reference: table 4.51. | |
1257 * | |
1258 * @param cnt length of TYPE_FIL syntactic element in bytes | |
1259 * | |
1260 * @return Returns number of bytes consumed | |
1261 */ | |
1262 static int decode_extension_payload(AACContext * ac, GetBitContext * gb, int cnt) { | |
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1263 int crc_flag = 0; |
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1264 int res = cnt; |
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1265 switch (get_bits(gb, 4)) { // extension type |
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1266 case EXT_SBR_DATA_CRC: |
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1267 crc_flag++; |
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1268 case EXT_SBR_DATA: |
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1269 res = decode_sbr_extension(ac, gb, crc_flag, cnt); |
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1270 break; |
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1271 case EXT_DYNAMIC_RANGE: |
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1272 res = decode_dynamic_range(&ac->che_drc, gb, cnt); |
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1273 break; |
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1274 case EXT_FILL: |
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1275 case EXT_FILL_DATA: |
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1276 case EXT_DATA_ELEMENT: |
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1277 default: |
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1278 skip_bits_long(gb, 8*cnt - 4); |
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1279 break; |
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1280 }; |
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1281 return res; |
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1282 } |
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1283 |
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1284 /** |
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1285 * Decode Temporal Noise Shaping filter coefficients and apply all-pole filters; reference: 4.6.9.3. |
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1286 * |
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1287 * @param decode 1 if tool is used normally, 0 if tool is used in LTP. |
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1288 * @param coef spectral coefficients |
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1289 */ |
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1290 static void apply_tns(float coef[1024], TemporalNoiseShaping * tns, IndividualChannelStream * ics, int decode) { |
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1291 const int mmm = FFMIN(ics->tns_max_bands, ics->max_sfb); |
7636 | 1292 int w, filt, m, i; |
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1293 int bottom, top, order, start, end, size, inc; |
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1294 float lpc[TNS_MAX_ORDER]; |
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1295 |
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1296 for (w = 0; w < ics->num_windows; w++) { |
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1297 bottom = ics->num_swb; |
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1298 for (filt = 0; filt < tns->n_filt[w]; filt++) { |
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1299 top = bottom; |
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1300 bottom = FFMAX(0, top - tns->length[w][filt]); |
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1301 order = tns->order[w][filt]; |
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1302 if (order == 0) |
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1303 continue; |
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1304 |
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1305 // tns_decode_coef |
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1306 compute_lpc_coefs(tns->coef[w][filt], order, lpc, 0, 0, 0); |
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1307 |
7609 | 1308 start = ics->swb_offset[FFMIN(bottom, mmm)]; |
1309 end = ics->swb_offset[FFMIN( top, mmm)]; | |
1310 if ((size = end - start) <= 0) | |
1311 continue; | |
1312 if (tns->direction[w][filt]) { | |
1313 inc = -1; start = end - 1; | |
1314 } else { | |
1315 inc = 1; | |
1316 } | |
1317 start += w * 128; | |
1318 | |
1319 // ar filter | |
1320 for (m = 0; m < size; m++, start += inc) | |
1321 for (i = 1; i <= FFMIN(m, order); i++) | |
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1322 coef[start] -= coef[start - i*inc] * lpc[i-1]; |
7609 | 1323 } |
1324 } | |
1325 } | |
1326 | |
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1327 /** |
7539 | 1328 * Conduct IMDCT and windowing. |
1329 */ | |
1330 static void imdct_and_windowing(AACContext * ac, SingleChannelElement * sce) { | |
1331 IndividualChannelStream * ics = &sce->ics; | |
1332 float * in = sce->coeffs; | |
1333 float * out = sce->ret; | |
1334 float * saved = sce->saved; | |
7580 | 1335 const float * swindow = ics->use_kb_window[0] ? ff_aac_kbd_short_128 : ff_sine_128; |
1336 const float * lwindow_prev = ics->use_kb_window[1] ? ff_aac_kbd_long_1024 : ff_sine_1024; | |
1337 const float * swindow_prev = ics->use_kb_window[1] ? ff_aac_kbd_short_128 : ff_sine_128; | |
7539 | 1338 float * buf = ac->buf_mdct; |
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1339 float * temp = ac->temp; |
7539 | 1340 int i; |
1341 | |
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1342 // imdct |
7578 | 1343 if (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) { |
1344 if (ics->window_sequence[1] == ONLY_LONG_SEQUENCE || ics->window_sequence[1] == LONG_STOP_SEQUENCE) | |
1345 av_log(ac->avccontext, AV_LOG_WARNING, | |
1346 "Transition from an ONLY_LONG or LONG_STOP to an EIGHT_SHORT sequence detected. " | |
1347 "If you heard an audible artifact, please submit the sample to the FFmpeg developers.\n"); | |
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1348 for (i = 0; i < 1024; i += 128) |
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1349 ff_imdct_half(&ac->mdct_small, buf + i, in + i); |
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1350 } else |
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1351 ff_imdct_half(&ac->mdct, buf, in); |
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1352 |
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1353 /* window overlapping |
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Separate code in imdct_and_windowing() into distinct IMDCT, windowing and
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parents:
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diff
changeset
|
1354 * NOTE: To simplify the overlapping code, all 'meaningless' short to long |
258eb30e6f7d
Separate code in imdct_and_windowing() into distinct IMDCT, windowing and
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parents:
7638
diff
changeset
|
1355 * and long to short transitions are considered to be short to short |
258eb30e6f7d
Separate code in imdct_and_windowing() into distinct IMDCT, windowing and
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parents:
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diff
changeset
|
1356 * transitions. This leaves just two cases (long to long and short to short) |
258eb30e6f7d
Separate code in imdct_and_windowing() into distinct IMDCT, windowing and
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parents:
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diff
changeset
|
1357 * with a little special sauce for EIGHT_SHORT_SEQUENCE. |
258eb30e6f7d
Separate code in imdct_and_windowing() into distinct IMDCT, windowing and
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parents:
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diff
changeset
|
1358 */ |
258eb30e6f7d
Separate code in imdct_and_windowing() into distinct IMDCT, windowing and
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parents:
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diff
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|
1359 if ((ics->window_sequence[1] == ONLY_LONG_SEQUENCE || ics->window_sequence[1] == LONG_STOP_SEQUENCE) && |
258eb30e6f7d
Separate code in imdct_and_windowing() into distinct IMDCT, windowing and
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parents:
7638
diff
changeset
|
1360 (ics->window_sequence[0] == ONLY_LONG_SEQUENCE || ics->window_sequence[0] == LONG_START_SEQUENCE)) { |
7663
66fb1f3937fa
Use ff_imdct_half() and vector_fmul_window() for IMDCT and windowing. Reduce
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parents:
7662
diff
changeset
|
1361 ac->dsp.vector_fmul_window( out, saved, buf, lwindow_prev, ac->add_bias, 512); |
7661
258eb30e6f7d
Separate code in imdct_and_windowing() into distinct IMDCT, windowing and
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parents:
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diff
changeset
|
1362 } else { |
7662 | 1363 for (i = 0; i < 448; i++) |
1364 out[i] = saved[i] + ac->add_bias; | |
7578 | 1365 |
7661
258eb30e6f7d
Separate code in imdct_and_windowing() into distinct IMDCT, windowing and
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parents:
7638
diff
changeset
|
1366 if (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) { |
7663
66fb1f3937fa
Use ff_imdct_half() and vector_fmul_window() for IMDCT and windowing. Reduce
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parents:
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|
1367 ac->dsp.vector_fmul_window(out + 448 + 0*128, saved + 448, buf + 0*128, swindow_prev, ac->add_bias, 64); |
66fb1f3937fa
Use ff_imdct_half() and vector_fmul_window() for IMDCT and windowing. Reduce
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parents:
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changeset
|
1368 ac->dsp.vector_fmul_window(out + 448 + 1*128, buf + 0*128 + 64, buf + 1*128, swindow, ac->add_bias, 64); |
66fb1f3937fa
Use ff_imdct_half() and vector_fmul_window() for IMDCT and windowing. Reduce
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parents:
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diff
changeset
|
1369 ac->dsp.vector_fmul_window(out + 448 + 2*128, buf + 1*128 + 64, buf + 2*128, swindow, ac->add_bias, 64); |
66fb1f3937fa
Use ff_imdct_half() and vector_fmul_window() for IMDCT and windowing. Reduce
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parents:
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changeset
|
1370 ac->dsp.vector_fmul_window(out + 448 + 3*128, buf + 2*128 + 64, buf + 3*128, swindow, ac->add_bias, 64); |
66fb1f3937fa
Use ff_imdct_half() and vector_fmul_window() for IMDCT and windowing. Reduce
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parents:
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|
1371 ac->dsp.vector_fmul_window(temp, buf + 3*128 + 64, buf + 4*128, swindow, ac->add_bias, 64); |
66fb1f3937fa
Use ff_imdct_half() and vector_fmul_window() for IMDCT and windowing. Reduce
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parents:
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diff
changeset
|
1372 memcpy( out + 448 + 4*128, temp, 64 * sizeof(float)); |
7661
258eb30e6f7d
Separate code in imdct_and_windowing() into distinct IMDCT, windowing and
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parents:
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changeset
|
1373 } else { |
7663
66fb1f3937fa
Use ff_imdct_half() and vector_fmul_window() for IMDCT and windowing. Reduce
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parents:
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|
1374 ac->dsp.vector_fmul_window(out + 448, saved + 448, buf, swindow_prev, ac->add_bias, 64); |
7662 | 1375 for (i = 576; i < 1024; i++) |
7663
66fb1f3937fa
Use ff_imdct_half() and vector_fmul_window() for IMDCT and windowing. Reduce
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parents:
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changeset
|
1376 out[i] = buf[i-512] + ac->add_bias; |
7661
258eb30e6f7d
Separate code in imdct_and_windowing() into distinct IMDCT, windowing and
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parents:
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|
1377 } |
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Separate code in imdct_and_windowing() into distinct IMDCT, windowing and
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|
1378 } |
7578 | 1379 |
7661
258eb30e6f7d
Separate code in imdct_and_windowing() into distinct IMDCT, windowing and
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parents:
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changeset
|
1380 // buffer update |
258eb30e6f7d
Separate code in imdct_and_windowing() into distinct IMDCT, windowing and
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parents:
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changeset
|
1381 if (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) { |
7663
66fb1f3937fa
Use ff_imdct_half() and vector_fmul_window() for IMDCT and windowing. Reduce
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parents:
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|
1382 for (i = 0; i < 64; i++) |
66fb1f3937fa
Use ff_imdct_half() and vector_fmul_window() for IMDCT and windowing. Reduce
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|
1383 saved[i] = temp[64 + i] - ac->add_bias; |
66fb1f3937fa
Use ff_imdct_half() and vector_fmul_window() for IMDCT and windowing. Reduce
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|
1384 ac->dsp.vector_fmul_window(saved + 64, buf + 4*128 + 64, buf + 5*128, swindow, 0, 64); |
66fb1f3937fa
Use ff_imdct_half() and vector_fmul_window() for IMDCT and windowing. Reduce
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parents:
7662
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changeset
|
1385 ac->dsp.vector_fmul_window(saved + 192, buf + 5*128 + 64, buf + 6*128, swindow, 0, 64); |
66fb1f3937fa
Use ff_imdct_half() and vector_fmul_window() for IMDCT and windowing. Reduce
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parents:
7662
diff
changeset
|
1386 ac->dsp.vector_fmul_window(saved + 320, buf + 6*128 + 64, buf + 7*128, swindow, 0, 64); |
66fb1f3937fa
Use ff_imdct_half() and vector_fmul_window() for IMDCT and windowing. Reduce
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parents:
7662
diff
changeset
|
1387 memcpy( saved + 448, buf + 7*128 + 64, 64 * sizeof(float)); |
7661
258eb30e6f7d
Separate code in imdct_and_windowing() into distinct IMDCT, windowing and
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parents:
7638
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changeset
|
1388 } else if (ics->window_sequence[0] == LONG_START_SEQUENCE) { |
7663
66fb1f3937fa
Use ff_imdct_half() and vector_fmul_window() for IMDCT and windowing. Reduce
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parents:
7662
diff
changeset
|
1389 memcpy( saved, buf + 512, 448 * sizeof(float)); |
66fb1f3937fa
Use ff_imdct_half() and vector_fmul_window() for IMDCT and windowing. Reduce
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parents:
7662
diff
changeset
|
1390 memcpy( saved + 448, buf + 7*128 + 64, 64 * sizeof(float)); |
7661
258eb30e6f7d
Separate code in imdct_and_windowing() into distinct IMDCT, windowing and
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changeset
|
1391 } else { // LONG_STOP or ONLY_LONG |
7663
66fb1f3937fa
Use ff_imdct_half() and vector_fmul_window() for IMDCT and windowing. Reduce
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|
1392 memcpy( saved, buf + 512, 512 * sizeof(float)); |
7578 | 1393 } |
1394 } | |
1395 | |
7539 | 1396 /** |
7523
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|
1397 * Apply dependent channel coupling (applied before IMDCT). |
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|
1398 * |
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|
1399 * @param index index into coupling gain array |
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|
1400 */ |
8006
c7c1e85d14bc
Rename variables to clarify the channel coupling element and corresponding
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|
1401 static void apply_dependent_coupling(AACContext * ac, SingleChannelElement * target, ChannelElement * cce, int index) { |
c7c1e85d14bc
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|
1402 IndividualChannelStream * ics = &cce->ch[0].ics; |
7523
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|
1403 const uint16_t * offsets = ics->swb_offset; |
8006
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|
1404 float * dest = target->coeffs; |
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|
1405 const float * src = cce->ch[0].coeffs; |
7523
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1406 int g, i, group, k, idx = 0; |
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1407 if(ac->m4ac.object_type == AOT_AAC_LTP) { |
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|
1408 av_log(ac->avccontext, AV_LOG_ERROR, |
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1409 "Dependent coupling is not supported together with LTP\n"); |
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|
1410 return; |
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1411 } |
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1412 for (g = 0; g < ics->num_window_groups; g++) { |
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|
1413 for (i = 0; i < ics->max_sfb; i++, idx++) { |
8006
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|
1414 if (cce->ch[0].band_type[idx] != ZERO_BT) { |
7523
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1415 for (group = 0; group < ics->group_len[g]; group++) { |
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|
1416 for (k = offsets[i]; k < offsets[i+1]; k++) { |
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1417 // XXX dsputil-ize |
8006
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1418 dest[group*128+k] += cce->coup.gain[index][idx] * src[group*128+k]; |
7523
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1419 } |
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1420 } |
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1421 } |
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1422 } |
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1423 dest += ics->group_len[g]*128; |
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1424 src += ics->group_len[g]*128; |
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1425 } |
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1426 } |
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1427 |
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1428 /** |
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1429 * Apply independent channel coupling (applied after IMDCT). |
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1430 * |
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1431 * @param index index into coupling gain array |
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1432 */ |
8006
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1433 static void apply_independent_coupling(AACContext * ac, SingleChannelElement * target, ChannelElement * cce, int index) { |
7523
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1434 int i; |
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1435 for (i = 0; i < 1024; i++) |
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1436 target->ret[i] += cce->coup.gain[index][0] * (cce->ch[0].ret[i] - ac->add_bias); |
7523
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1437 } |
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1438 |
7581 | 1439 /** |
1440 * channel coupling transformation interface | |
1441 * | |
1442 * @param index index into coupling gain array | |
1443 * @param apply_coupling_method pointer to (in)dependent coupling function | |
1444 */ | |
1445 static void apply_channel_coupling(AACContext * ac, ChannelElement * cc, | |
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1446 enum RawDataBlockType type, int elem_id, enum CouplingPoint coupling_point, |
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1447 void (*apply_coupling_method)(AACContext * ac, SingleChannelElement * target, ChannelElement * cce, int index)) |
7581 | 1448 { |
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1449 int i, c; |
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1450 |
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1451 for (i = 0; i < MAX_ELEM_ID; i++) { |
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1452 ChannelElement *cce = ac->che[TYPE_CCE][i]; |
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1453 int index = 0; |
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1454 |
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1455 if (cce && cce->coup.coupling_point == coupling_point) { |
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1456 ChannelCoupling * coup = &cce->coup; |
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1457 |
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1458 for (c = 0; c <= coup->num_coupled; c++) { |
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1459 if (coup->type[c] == type && coup->id_select[c] == elem_id) { |
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1460 if (coup->ch_select[c] != 1) { |
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1461 apply_coupling_method(ac, &cc->ch[0], cce, index); |
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1462 if (coup->ch_select[c] != 0) |
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1463 index++; |
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1464 } |
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1465 if (coup->ch_select[c] != 2) |
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1466 apply_coupling_method(ac, &cc->ch[1], cce, index++); |
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1467 } else |
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1468 index += 1 + (coup->ch_select[c] == 3); |
7581 | 1469 } |
1470 } | |
1471 } | |
1472 } | |
1473 | |
1474 /** | |
1475 * Convert spectral data to float samples, applying all supported tools as appropriate. | |
1476 */ | |
1477 static void spectral_to_sample(AACContext * ac) { | |
1478 int i, type; | |
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1479 for(type = 3; type >= 0; type--) { |
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1480 for (i = 0; i < MAX_ELEM_ID; i++) { |
7581 | 1481 ChannelElement *che = ac->che[type][i]; |
1482 if(che) { | |
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1483 if(type <= TYPE_CPE) |
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1484 apply_channel_coupling(ac, che, type, i, BEFORE_TNS, apply_dependent_coupling); |
7581 | 1485 if(che->ch[0].tns.present) |
1486 apply_tns(che->ch[0].coeffs, &che->ch[0].tns, &che->ch[0].ics, 1); | |
1487 if(che->ch[1].tns.present) | |
1488 apply_tns(che->ch[1].coeffs, &che->ch[1].tns, &che->ch[1].ics, 1); | |
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1489 if(type <= TYPE_CPE) |
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1490 apply_channel_coupling(ac, che, type, i, BETWEEN_TNS_AND_IMDCT, apply_dependent_coupling); |
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|
1491 if(type != TYPE_CCE || che->coup.coupling_point == AFTER_IMDCT) |
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1492 imdct_and_windowing(ac, &che->ch[0]); |
7581 | 1493 if(type == TYPE_CPE) |
1494 imdct_and_windowing(ac, &che->ch[1]); | |
8004
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1495 if(type <= TYPE_CCE) |
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1496 apply_channel_coupling(ac, che, type, i, AFTER_IMDCT, apply_independent_coupling); |
7578 | 1497 } |
1498 } | |
1499 } | |
1500 } | |
1501 | |
1502 static int aac_decode_frame(AVCodecContext * avccontext, void * data, int * data_size, const uint8_t * buf, int buf_size) { | |
1503 AACContext * ac = avccontext->priv_data; | |
1504 GetBitContext gb; | |
1505 enum RawDataBlockType elem_type; | |
1506 int err, elem_id, data_size_tmp; | |
1507 | |
1508 init_get_bits(&gb, buf, buf_size*8); | |
1509 | |
1510 // parse | |
1511 while ((elem_type = get_bits(&gb, 3)) != TYPE_END) { | |
1512 elem_id = get_bits(&gb, 4); | |
1513 err = -1; | |
1514 | |
1515 if(elem_type == TYPE_SCE && elem_id == 1 && | |
1516 !ac->che[TYPE_SCE][elem_id] && ac->che[TYPE_LFE][0]) { | |
1517 /* Some streams incorrectly code 5.1 audio as SCE[0] CPE[0] CPE[1] SCE[1] | |
1518 instead of SCE[0] CPE[0] CPE[0] LFE[0]. If we seem to have | |
1519 encountered such a stream, transfer the LFE[0] element to SCE[1] */ | |
1520 ac->che[TYPE_SCE][elem_id] = ac->che[TYPE_LFE][0]; | |
1521 ac->che[TYPE_LFE][0] = NULL; | |
1522 } | |
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1523 if(elem_type < TYPE_DSE) { |
7578 | 1524 if(!ac->che[elem_type][elem_id]) |
1525 return -1; | |
1526 if(elem_type != TYPE_CCE) | |
1527 ac->che[elem_type][elem_id]->coup.coupling_point = 4; | |
1528 } | |
1529 | |
1530 switch (elem_type) { | |
1531 | |
1532 case TYPE_SCE: | |
1533 err = decode_ics(ac, &ac->che[TYPE_SCE][elem_id]->ch[0], &gb, 0, 0); | |
1534 break; | |
1535 | |
1536 case TYPE_CPE: | |
1537 err = decode_cpe(ac, &gb, elem_id); | |
1538 break; | |
1539 | |
1540 case TYPE_CCE: | |
7733 | 1541 err = decode_cce(ac, &gb, ac->che[TYPE_CCE][elem_id]); |
7578 | 1542 break; |
1543 | |
1544 case TYPE_LFE: | |
1545 err = decode_ics(ac, &ac->che[TYPE_LFE][elem_id]->ch[0], &gb, 0, 0); | |
1546 break; | |
1547 | |
1548 case TYPE_DSE: | |
1549 skip_data_stream_element(&gb); | |
1550 err = 0; | |
1551 break; | |
1552 | |
1553 case TYPE_PCE: | |
1554 { | |
1555 enum ChannelPosition new_che_pos[4][MAX_ELEM_ID]; | |
1556 memset(new_che_pos, 0, 4 * MAX_ELEM_ID * sizeof(new_che_pos[0][0])); | |
1557 if((err = decode_pce(ac, new_che_pos, &gb))) | |
1558 break; | |
1559 err = output_configure(ac, ac->che_pos, new_che_pos); | |
1560 break; | |
1561 } | |
1562 | |
1563 case TYPE_FIL: | |
1564 if (elem_id == 15) | |
1565 elem_id += get_bits(&gb, 8) - 1; | |
1566 while (elem_id > 0) | |
1567 elem_id -= decode_extension_payload(ac, &gb, elem_id); | |
1568 err = 0; /* FIXME */ | |
1569 break; | |
1570 | |
1571 default: | |
1572 err = -1; /* should not happen, but keeps compiler happy */ | |
1573 break; | |
1574 } | |
1575 | |
1576 if(err) | |
1577 return err; | |
1578 } | |
1579 | |
1580 spectral_to_sample(ac); | |
1581 | |
7539 | 1582 if (!ac->is_saved) { |
1583 ac->is_saved = 1; | |
1584 *data_size = 0; | |
7580 | 1585 return buf_size; |
7539 | 1586 } |
1587 | |
1588 data_size_tmp = 1024 * avccontext->channels * sizeof(int16_t); | |
1589 if(*data_size < data_size_tmp) { | |
1590 av_log(avccontext, AV_LOG_ERROR, | |
1591 "Output buffer too small (%d) or trying to output too many samples (%d) for this frame.\n", | |
1592 *data_size, data_size_tmp); | |
1593 return -1; | |
1594 } | |
1595 *data_size = data_size_tmp; | |
1596 | |
1597 ac->dsp.float_to_int16_interleave(data, (const float **)ac->output_data, 1024, avccontext->channels); | |
1598 | |
1599 return buf_size; | |
1600 } | |
1601 | |
7501 | 1602 static av_cold int aac_decode_close(AVCodecContext * avccontext) { |
1603 AACContext * ac = avccontext->priv_data; | |
7540 | 1604 int i, type; |
7501 | 1605 |
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1606 for (i = 0; i < MAX_ELEM_ID; i++) { |
7540 | 1607 for(type = 0; type < 4; type++) |
1608 av_freep(&ac->che[type][i]); | |
7501 | 1609 } |
1610 | |
1611 ff_mdct_end(&ac->mdct); | |
1612 ff_mdct_end(&ac->mdct_small); | |
1613 return 0 ; | |
1614 } | |
1615 | |
1616 AVCodec aac_decoder = { | |
1617 "aac", | |
1618 CODEC_TYPE_AUDIO, | |
1619 CODEC_ID_AAC, | |
1620 sizeof(AACContext), | |
1621 aac_decode_init, | |
1622 NULL, | |
1623 aac_decode_close, | |
1624 aac_decode_frame, | |
1625 .long_name = NULL_IF_CONFIG_SMALL("Advanced Audio Coding"), | |
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1626 .sample_fmts = (enum SampleFormat[]){SAMPLE_FMT_S16,SAMPLE_FMT_NONE}, |
7501 | 1627 }; |