More OKed hunks of the AAC decoder from SoC
[libav.git] / libavcodec / aac.c
CommitLineData
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1/*
2 * AAC decoder
3 * Copyright (c) 2005-2006 Oded Shimon ( ods15 ods15 dyndns org )
4 * Copyright (c) 2006-2007 Maxim Gavrilov ( maxim.gavrilov gmail com )
5 *
6 * This file is part of FFmpeg.
7 *
8 * FFmpeg is free software; you can redistribute it and/or
9 * modify it under the terms of the GNU Lesser General Public
10 * License as published by the Free Software Foundation; either
11 * version 2.1 of the License, or (at your option) any later version.
12 *
13 * FFmpeg is distributed in the hope that it will be useful,
14 * but WITHOUT ANY WARRANTY; without even the implied warranty of
15 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
16 * Lesser General Public License for more details.
17 *
18 * You should have received a copy of the GNU Lesser General Public
19 * License along with FFmpeg; if not, write to the Free Software
20 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
21 */
22
23/**
24 * @file aac.c
25 * AAC decoder
26 * @author Oded Shimon ( ods15 ods15 dyndns org )
27 * @author Maxim Gavrilov ( maxim.gavrilov gmail com )
28 */
29
30/*
31 * supported tools
32 *
33 * Support? Name
34 * N (code in SoC repo) gain control
35 * Y block switching
36 * Y window shapes - standard
37 * N window shapes - Low Delay
38 * Y filterbank - standard
39 * N (code in SoC repo) filterbank - Scalable Sample Rate
40 * Y Temporal Noise Shaping
41 * N (code in SoC repo) Long Term Prediction
42 * Y intensity stereo
43 * Y channel coupling
44 * N frequency domain prediction
45 * Y Perceptual Noise Substitution
46 * Y Mid/Side stereo
47 * N Scalable Inverse AAC Quantization
48 * N Frequency Selective Switch
49 * N upsampling filter
50 * Y quantization & coding - AAC
51 * N quantization & coding - TwinVQ
52 * N quantization & coding - BSAC
53 * N AAC Error Resilience tools
54 * N Error Resilience payload syntax
55 * N Error Protection tool
56 * N CELP
57 * N Silence Compression
58 * N HVXC
59 * N HVXC 4kbits/s VR
60 * N Structured Audio tools
61 * N Structured Audio Sample Bank Format
62 * N MIDI
63 * N Harmonic and Individual Lines plus Noise
64 * N Text-To-Speech Interface
65 * N (in progress) Spectral Band Replication
66 * Y (not in this code) Layer-1
67 * Y (not in this code) Layer-2
68 * Y (not in this code) Layer-3
69 * N SinuSoidal Coding (Transient, Sinusoid, Noise)
70 * N (planned) Parametric Stereo
71 * N Direct Stream Transfer
72 *
73 * Note: - HE AAC v1 comprises LC AAC with Spectral Band Replication.
74 * - HE AAC v2 comprises LC AAC with Spectral Band Replication and
75 Parametric Stereo.
76 */
77
78
79#include "avcodec.h"
80#include "bitstream.h"
81#include "dsputil.h"
82
83#include "aac.h"
84#include "aactab.h"
cc0591da 85#include "aacdectab.h"
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86#include "mpeg4audio.h"
87
88#include <assert.h>
89#include <errno.h>
90#include <math.h>
91#include <string.h>
92
93#ifndef CONFIG_HARDCODED_TABLES
94 static float ff_aac_ivquant_tab[IVQUANT_SIZE];
cc0591da 95 static float ff_aac_pow2sf_tab[316];
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96#endif /* CONFIG_HARDCODED_TABLES */
97
98static VLC vlc_scalefactors;
99static VLC vlc_spectral[11];
100
101
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102/**
103 * Decode an array of 4 bit element IDs, optionally interleaved with a stereo/mono switching bit.
104 *
105 * @param cpe_map Stereo (Channel Pair Element) map, NULL if stereo bit is not present.
106 * @param sce_map mono (Single Channel Element) map
107 * @param type speaker type/position for these channels
108 */
109static void decode_channel_map(enum ChannelPosition *cpe_map,
110 enum ChannelPosition *sce_map, enum ChannelPosition type, GetBitContext * gb, int n) {
111 while(n--) {
112 enum ChannelPosition *map = cpe_map && get_bits1(gb) ? cpe_map : sce_map; // stereo or mono map
113 map[get_bits(gb, 4)] = type;
114 }
115}
116
117/**
118 * Decode program configuration element; reference: table 4.2.
119 *
120 * @param new_che_pos New channel position configuration - we only do something if it differs from the current one.
121 *
122 * @return Returns error status. 0 - OK, !0 - error
123 */
124static int decode_pce(AACContext * ac, enum ChannelPosition new_che_pos[4][MAX_ELEM_ID],
125 GetBitContext * gb) {
126 int num_front, num_side, num_back, num_lfe, num_assoc_data, num_cc;
127
128 skip_bits(gb, 2); // object_type
129
130 ac->m4ac.sampling_index = get_bits(gb, 4);
131 if(ac->m4ac.sampling_index > 11) {
132 av_log(ac->avccontext, AV_LOG_ERROR, "invalid sampling rate index %d\n", ac->m4ac.sampling_index);
133 return -1;
134 }
135 ac->m4ac.sample_rate = ff_mpeg4audio_sample_rates[ac->m4ac.sampling_index];
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136 num_front = get_bits(gb, 4);
137 num_side = get_bits(gb, 4);
138 num_back = get_bits(gb, 4);
139 num_lfe = get_bits(gb, 2);
140 num_assoc_data = get_bits(gb, 3);
141 num_cc = get_bits(gb, 4);
142
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143 if (get_bits1(gb))
144 skip_bits(gb, 4); // mono_mixdown_tag
145 if (get_bits1(gb))
146 skip_bits(gb, 4); // stereo_mixdown_tag
71e9a1b8 147
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148 if (get_bits1(gb))
149 skip_bits(gb, 3); // mixdown_coeff_index and pseudo_surround
71e9a1b8 150
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151 decode_channel_map(new_che_pos[TYPE_CPE], new_che_pos[TYPE_SCE], AAC_CHANNEL_FRONT, gb, num_front);
152 decode_channel_map(new_che_pos[TYPE_CPE], new_che_pos[TYPE_SCE], AAC_CHANNEL_SIDE, gb, num_side );
153 decode_channel_map(new_che_pos[TYPE_CPE], new_che_pos[TYPE_SCE], AAC_CHANNEL_BACK, gb, num_back );
154 decode_channel_map(NULL, new_che_pos[TYPE_LFE], AAC_CHANNEL_LFE, gb, num_lfe );
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155
156 skip_bits_long(gb, 4 * num_assoc_data);
157
cc0591da 158 decode_channel_map(new_che_pos[TYPE_CCE], new_che_pos[TYPE_CCE], AAC_CHANNEL_CC, gb, num_cc );
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159
160 align_get_bits(gb);
161
162 /* comment field, first byte is length */
163 skip_bits_long(gb, 8 * get_bits(gb, 8));
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164 return 0;
165}
71e9a1b8 166
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167/**
168 * Set up channel positions based on a default channel configuration
169 * as specified in table 1.17.
170 *
171 * @param new_che_pos New channel position configuration - we only do something if it differs from the current one.
172 *
173 * @return Returns error status. 0 - OK, !0 - error
174 */
175static int set_default_channel_config(AACContext *ac, enum ChannelPosition new_che_pos[4][MAX_ELEM_ID],
176 int channel_config)
177{
178 if(channel_config < 1 || channel_config > 7) {
179 av_log(ac->avccontext, AV_LOG_ERROR, "invalid default channel configuration (%d)\n",
180 channel_config);
181 return -1;
182 }
183
184 /* default channel configurations:
185 *
186 * 1ch : front center (mono)
187 * 2ch : L + R (stereo)
188 * 3ch : front center + L + R
189 * 4ch : front center + L + R + back center
190 * 5ch : front center + L + R + back stereo
191 * 6ch : front center + L + R + back stereo + LFE
192 * 7ch : front center + L + R + outer front left + outer front right + back stereo + LFE
193 */
194
195 if(channel_config != 2)
196 new_che_pos[TYPE_SCE][0] = AAC_CHANNEL_FRONT; // front center (or mono)
197 if(channel_config > 1)
198 new_che_pos[TYPE_CPE][0] = AAC_CHANNEL_FRONT; // L + R (or stereo)
199 if(channel_config == 4)
200 new_che_pos[TYPE_SCE][1] = AAC_CHANNEL_BACK; // back center
201 if(channel_config > 4)
202 new_che_pos[TYPE_CPE][(channel_config == 7) + 1]
203 = AAC_CHANNEL_BACK; // back stereo
204 if(channel_config > 5)
205 new_che_pos[TYPE_LFE][0] = AAC_CHANNEL_LFE; // LFE
206 if(channel_config == 7)
207 new_che_pos[TYPE_CPE][1] = AAC_CHANNEL_FRONT; // outer front left + outer front right
208
209 return 0;
210}
211
212 return -1;
213 }
214
215 if (get_bits1(gb)) // dependsOnCoreCoder
216 skip_bits(gb, 14); // coreCoderDelay
217 extension_flag = get_bits1(gb);
218
219 if(ac->m4ac.object_type == AOT_AAC_SCALABLE ||
220 ac->m4ac.object_type == AOT_ER_AAC_SCALABLE)
221 skip_bits(gb, 3); // layerNr
222
223 memset(new_che_pos, 0, 4 * MAX_ELEM_ID * sizeof(new_che_pos[0][0]));
224 if (channel_config == 0) {
225 skip_bits(gb, 4); // element_instance_tag
226 if((ret = decode_pce(ac, new_che_pos, gb)))
227 return ret;
228 } else {
229 if((ret = set_default_channel_config(ac, new_che_pos, channel_config)))
230 return ret;
231 }
232 if((ret = output_configure(ac, ac->che_pos, new_che_pos)))
233 return ret;
234
235 if (extension_flag) {
236 switch (ac->m4ac.object_type) {
237 case AOT_ER_BSAC:
238 skip_bits(gb, 5); // numOfSubFrame
239 skip_bits(gb, 11); // layer_length
240 break;
241 case AOT_ER_AAC_LC:
242 case AOT_ER_AAC_LTP:
243 case AOT_ER_AAC_SCALABLE:
244 case AOT_ER_AAC_LD:
245 skip_bits(gb, 3); /* aacSectionDataResilienceFlag
246 * aacScalefactorDataResilienceFlag
247 * aacSpectralDataResilienceFlag
248 */
249 break;
250 }
251 skip_bits1(gb); // extensionFlag3 (TBD in version 3)
252 }
253 return 0;
254}
255
256/**
257 * Decode audio specific configuration; reference: table 1.13.
258 *
259 * @param data pointer to AVCodecContext extradata
260 * @param data_size size of AVCCodecContext extradata
261 *
262 * @return Returns error status. 0 - OK, !0 - error
263 */
264static int decode_audio_specific_config(AACContext * ac, void *data, int data_size) {
265 GetBitContext gb;
266 int i;
267
268 init_get_bits(&gb, data, data_size * 8);
269
270 if((i = ff_mpeg4audio_get_config(&ac->m4ac, data, data_size)) < 0)
271 return -1;
272 if(ac->m4ac.sampling_index > 11) {
273 av_log(ac->avccontext, AV_LOG_ERROR, "invalid sampling rate index %d\n", ac->m4ac.sampling_index);
274 return -1;
275 }
276
277 skip_bits_long(&gb, i);
278
279 switch (ac->m4ac.object_type) {
280 case AOT_AAC_LC:
281 if (decode_ga_specific_config(ac, &gb, ac->m4ac.chan_config))
282 return -1;
283 break;
284 default:
285 av_log(ac->avccontext, AV_LOG_ERROR, "Audio object type %s%d is not supported.\n",
286 ac->m4ac.sbr == 1? "SBR+" : "", ac->m4ac.object_type);
287 return -1;
288 }
289 return 0;
290}
291
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292static av_cold int aac_decode_init(AVCodecContext * avccontext) {
293 AACContext * ac = avccontext->priv_data;
294 int i;
295
296 ac->avccontext = avccontext;
297
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298 if (avccontext->extradata_size <= 0 ||
299 decode_audio_specific_config(ac, avccontext->extradata, avccontext->extradata_size))
300 return -1;
301
9cc04edf 302 avccontext->sample_fmt = SAMPLE_FMT_S16;
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303 avccontext->sample_rate = ac->m4ac.sample_rate;
304 avccontext->frame_size = 1024;
305
306 AAC_INIT_VLC_STATIC( 0, 144);
307 AAC_INIT_VLC_STATIC( 1, 114);
308 AAC_INIT_VLC_STATIC( 2, 188);
309 AAC_INIT_VLC_STATIC( 3, 180);
310 AAC_INIT_VLC_STATIC( 4, 172);
311 AAC_INIT_VLC_STATIC( 5, 140);
312 AAC_INIT_VLC_STATIC( 6, 168);
313 AAC_INIT_VLC_STATIC( 7, 114);
314 AAC_INIT_VLC_STATIC( 8, 262);
315 AAC_INIT_VLC_STATIC( 9, 248);
316 AAC_INIT_VLC_STATIC(10, 384);
317
318 dsputil_init(&ac->dsp, avccontext);
319
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320 ac->random_state = 0x1f2e3d4c;
321
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322 // -1024 - Compensate wrong IMDCT method.
323 // 32768 - Required to scale values to the correct range for the bias method
324 // for float to int16 conversion.
325
326 if(ac->dsp.float_to_int16 == ff_float_to_int16_c) {
327 ac->add_bias = 385.0f;
328 ac->sf_scale = 1. / (-1024. * 32768.);
329 ac->sf_offset = 0;
330 } else {
331 ac->add_bias = 0.0f;
332 ac->sf_scale = 1. / -1024.;
333 ac->sf_offset = 60;
334 }
335
336#ifndef CONFIG_HARDCODED_TABLES
337 for (i = 1 - IVQUANT_SIZE/2; i < IVQUANT_SIZE/2; i++)
338 ff_aac_ivquant_tab[i + IVQUANT_SIZE/2 - 1] = cbrt(fabs(i)) * i;
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339 for (i = 0; i < 316; i++)
340 ff_aac_pow2sf_tab[i] = pow(2, (i - 200)/4.);
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341#endif /* CONFIG_HARDCODED_TABLES */
342
343 INIT_VLC_STATIC(&vlc_scalefactors, 7, sizeof(ff_aac_scalefactor_code)/sizeof(ff_aac_scalefactor_code[0]),
344 ff_aac_scalefactor_bits, sizeof(ff_aac_scalefactor_bits[0]), sizeof(ff_aac_scalefactor_bits[0]),
345 ff_aac_scalefactor_code, sizeof(ff_aac_scalefactor_code[0]), sizeof(ff_aac_scalefactor_code[0]),
346 352);
347
348 ff_mdct_init(&ac->mdct, 11, 1);
349 ff_mdct_init(&ac->mdct_small, 8, 1);
350 return 0;
351}
352
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353/**
354 * Skip data_stream_element; reference: table 4.10.
355 */
356static void skip_data_stream_element(GetBitContext * gb) {
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357 int byte_align = get_bits1(gb);
358 int count = get_bits(gb, 8);
359 if (count == 255)
360 count += get_bits(gb, 8);
361 if (byte_align)
362 align_get_bits(gb);
363 skip_bits_long(gb, 8 * count);
364}
365
366/**
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367 * Decode Individual Channel Stream info; reference: table 4.6.
368 *
369 * @param common_window Channels have independent [0], or shared [1], Individual Channel Stream information.
370 */
371static int decode_ics_info(AACContext * ac, IndividualChannelStream * ics, GetBitContext * gb, int common_window) {
372 if (get_bits1(gb)) {
373 av_log(ac->avccontext, AV_LOG_ERROR, "Reserved bit set.\n");
374 memset(ics, 0, sizeof(IndividualChannelStream));
375 return -1;
376 }
377 ics->window_sequence[1] = ics->window_sequence[0];
378 ics->window_sequence[0] = get_bits(gb, 2);
379 ics->use_kb_window[1] = ics->use_kb_window[0];
380 ics->use_kb_window[0] = get_bits1(gb);
381 ics->num_window_groups = 1;
382 ics->group_len[0] = 1;
383
384 return 0;
385}
386
387/**
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388 * inverse quantization
389 *
390 * @param a quantized value to be dequantized
391 * @return Returns dequantized value.
392 */
393static inline float ivquant(int a) {
394 if (a + (unsigned int)IVQUANT_SIZE/2 - 1 < (unsigned int)IVQUANT_SIZE - 1)
395 return ff_aac_ivquant_tab[a + IVQUANT_SIZE/2 - 1];
396 else
397 return cbrtf(fabsf(a)) * a;
398}
399
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400/**
401 * Decode band types (section_data payload); reference: table 4.46.
402 *
403 * @param band_type array of the used band type
404 * @param band_type_run_end array of the last scalefactor band of a band type run
405 *
406 * @return Returns error status. 0 - OK, !0 - error
407 */
408static int decode_band_types(AACContext * ac, enum BandType band_type[120],
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409 int band_type_run_end[120], GetBitContext * gb, IndividualChannelStream * ics) {
410 int g, idx = 0;
411 const int bits = (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) ? 3 : 5;
412 for (g = 0; g < ics->num_window_groups; g++) {
413 int k = 0;
414 while (k < ics->max_sfb) {
415 uint8_t sect_len = k;
416 int sect_len_incr;
417 int sect_band_type = get_bits(gb, 4);
418 if (sect_band_type == 12) {
419 av_log(ac->avccontext, AV_LOG_ERROR, "invalid band type\n");
420 return -1;
421 }
422 while ((sect_len_incr = get_bits(gb, bits)) == (1 << bits)-1)
423 sect_len += sect_len_incr;
424 sect_len += sect_len_incr;
425 if (sect_len > ics->max_sfb) {
426 av_log(ac->avccontext, AV_LOG_ERROR,
427 "Number of bands (%d) exceeds limit (%d).\n",
428 sect_len, ics->max_sfb);
429 return -1;
430 }
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431 }
432 }
433 return 0;
434}
cc0591da 435
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436/**
437 * Decode scalefactors; reference: table 4.47.
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438 *
439 * @param mix_gain channel gain (Not used by AAC bitstream.)
440 * @param global_gain first scalefactor value as scalefactors are differentially coded
441 * @param band_type array of the used band type
442 * @param band_type_run_end array of the last scalefactor band of a band type run
443 * @param sf array of scalefactors or intensity stereo positions
444 *
445 * @return Returns error status. 0 - OK, !0 - error
446 */
447static int decode_scalefactors(AACContext * ac, float sf[120], GetBitContext * gb,
448 float mix_gain, unsigned int global_gain, IndividualChannelStream * ics,
449 enum BandType band_type[120], int band_type_run_end[120]) {
450 const int sf_offset = ac->sf_offset + (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE ? 12 : 0);
451 int g, i, idx = 0;
452 int offset[3] = { global_gain, global_gain - 90, 100 };
453 int noise_flag = 1;
454 static const char *sf_str[3] = { "Global gain", "Noise gain", "Intensity stereo position" };
455 ics->intensity_present = 0;
456 for (g = 0; g < ics->num_window_groups; g++) {
457 for (i = 0; i < ics->max_sfb;) {
458 int run_end = band_type_run_end[idx];
459 if (band_type[idx] == ZERO_BT) {
460 for(; i < run_end; i++, idx++)
461 sf[idx] = 0.;
462 }else if((band_type[idx] == INTENSITY_BT) || (band_type[idx] == INTENSITY_BT2)) {
463 ics->intensity_present = 1;
464 for(; i < run_end; i++, idx++) {
465 offset[2] += get_vlc2(gb, vlc_scalefactors.table, 7, 3) - 60;
466 if(offset[2] > 255U) {
467 av_log(ac->avccontext, AV_LOG_ERROR,
468 "%s (%d) out of range.\n", sf_str[2], offset[2]);
469 return -1;
470 }
471 sf[idx] = ff_aac_pow2sf_tab[-offset[2] + 300];
472 sf[idx] *= mix_gain;
473 }
474 }else if(band_type[idx] == NOISE_BT) {
475 for(; i < run_end; i++, idx++) {
476 if(noise_flag-- > 0)
477 offset[1] += get_bits(gb, 9) - 256;
478 else
479 offset[1] += get_vlc2(gb, vlc_scalefactors.table, 7, 3) - 60;
480 if(offset[1] > 255U) {
481 av_log(ac->avccontext, AV_LOG_ERROR,
482 "%s (%d) out of range.\n", sf_str[1], offset[1]);
483 return -1;
484 }
485 sf[idx] = -ff_aac_pow2sf_tab[ offset[1] + sf_offset];
486 sf[idx] *= mix_gain;
487 }
488 }else {
489 for(; i < run_end; i++, idx++) {
490 offset[0] += get_vlc2(gb, vlc_scalefactors.table, 7, 3) - 60;
491 if(offset[0] > 255U) {
492 av_log(ac->avccontext, AV_LOG_ERROR,
493 "%s (%d) out of range.\n", sf_str[0], offset[0]);
494 return -1;
495 }
496 sf[idx] = -ff_aac_pow2sf_tab[ offset[0] + sf_offset];
497 sf[idx] *= mix_gain;
498 }
499 }
500 }
501 }
502 return 0;
503}
504
505/**
506 * Decode pulse data; reference: table 4.7.
507 */
508static void decode_pulses(Pulse * pulse, GetBitContext * gb) {
509 int i;
510 pulse->num_pulse = get_bits(gb, 2) + 1;
511 pulse->start = get_bits(gb, 6);
512 for (i = 0; i < pulse->num_pulse; i++) {
513 pulse->offset[i] = get_bits(gb, 5);
514 pulse->amp [i] = get_bits(gb, 4);
515 }
516}
517
518/**
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519 * Decode Mid/Side data; reference: table 4.54.
520 *
521 * @param ms_present Indicates mid/side stereo presence. [0] mask is all 0s;
522 * [1] mask is decoded from bitstream; [2] mask is all 1s;
523 * [3] reserved for scalable AAC
524 */
525static void decode_mid_side_stereo(ChannelElement * cpe, GetBitContext * gb,
526 int ms_present) {
527
528/**
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529 * Add pulses with particular amplitudes to the quantized spectral data; reference: 4.6.3.3.
530 *
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531 * @param pulse pointer to pulse data struct
532 * @param icoef array of quantized spectral data
533 */
534static void add_pulses(int icoef[1024], const Pulse * pulse, const IndividualChannelStream * ics) {
535 int i, off = ics->swb_offset[pulse->start];
536 for (i = 0; i < pulse->num_pulse; i++) {
537 int ic;
538 off += pulse->offset[i];
539 ic = (icoef[off] - 1)>>31;
540 icoef[off] += (pulse->amp[i]^ic) - ic;
541 }
542}
543
cc0591da 544/**
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545 * Decode an individual_channel_stream payload; reference: table 4.44.
546 *
547 * @param common_window Channels have independent [0], or shared [1], Individual Channel Stream information.
548 * @param scale_flag scalable [1] or non-scalable [0] AAC (Unused until scalable AAC is implemented.)
549 *
550 * @return Returns error status. 0 - OK, !0 - error
551 */
552static int decode_ics(AACContext * ac, SingleChannelElement * sce, GetBitContext * gb, int common_window, int scale_flag) {
553 int icoeffs[1024];
554 Pulse pulse;
555 TemporalNoiseShaping * tns = &sce->tns;
556 IndividualChannelStream * ics = &sce->ics;
557 float * out = sce->coeffs;
558 int global_gain, pulse_present = 0;
559
560 /* These two assignments are to silence some GCC warnings about the
561 * variables being used uninitialised when in fact they always are.
562 */
563 pulse.num_pulse = 0;
564 pulse.start = 0;
565
566 global_gain = get_bits(gb, 8);
567
568 if (!common_window && !scale_flag) {
569 if (decode_ics_info(ac, ics, gb, 0) < 0)
570 return -1;
571 }
572
573 if (decode_band_types(ac, sce->band_type, sce->band_type_run_end, gb, ics) < 0)
574 return -1;
575 if (decode_scalefactors(ac, sce->sf, gb, global_gain, ics, sce->band_type, sce->band_type_run_end) < 0)
576 return -1;
577
578 pulse_present = 0;
579 if (!scale_flag) {
580 if ((pulse_present = get_bits1(gb))) {
581 if (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) {
582 av_log(ac->avccontext, AV_LOG_ERROR, "Pulse tool not allowed in eight short sequence.\n");
583 return -1;
584 }
585 decode_pulses(&pulse, gb);
586 }
587 if ((tns->present = get_bits1(gb)) && decode_tns(ac, tns, gb, ics))
588 return -1;
589 if (get_bits1(gb)) {
590 av_log_missing_feature(ac->avccontext, "SSR", 1);
591 return -1;
592 }
593 }
594
595 if (decode_spectrum(ac, icoeffs, gb, ics, sce->band_type) < 0)
596 return -1;
597 if (pulse_present)
598 add_pulses(icoeffs, &pulse, ics);
599 dequant(ac, out, icoeffs, sce->sf, ics, sce->band_type);
600 return 0;
601}
602
603/**
604 * Decode a channel_pair_element; reference: table 4.4.
605 *
606 * @param elem_id Identifies the instance of a syntax element.
607 *
608 * @return Returns error status. 0 - OK, !0 - error
609 */
610static int decode_cpe(AACContext * ac, GetBitContext * gb, int elem_id) {
611 int i, ret, common_window, ms_present = 0;
612 ChannelElement * cpe;
613
614 cpe = ac->che[TYPE_CPE][elem_id];
615 common_window = get_bits1(gb);
616 if (common_window) {
617 if (decode_ics_info(ac, &cpe->ch[0].ics, gb, 1))
618 return -1;
619 i = cpe->ch[1].ics.use_kb_window[0];
620 cpe->ch[1].ics = cpe->ch[0].ics;
621 cpe->ch[1].ics.use_kb_window[1] = i;
622 ms_present = get_bits(gb, 2);
623 if(ms_present == 3) {
624 av_log(ac->avccontext, AV_LOG_ERROR, "ms_present = 3 is reserved.\n");
625 return -1;
626 } else if(ms_present)
627 decode_mid_side_stereo(cpe, gb, ms_present);
628 }
629 if ((ret = decode_ics(ac, &cpe->ch[0], gb, common_window, 0)))
630 return ret;
631 if ((ret = decode_ics(ac, &cpe->ch[1], gb, common_window, 0)))
632 return ret;
633
634 if (common_window && ms_present)
635 apply_mid_side_stereo(cpe);
636
637 if (cpe->ch[1].ics.intensity_present)
638 apply_intensity_stereo(cpe, ms_present);
639 return 0;
640}
641
642/**
643 * Decode Spectral Band Replication extension data; reference: table 4.55.
cc0591da
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644 *
645 * @param crc flag indicating the presence of CRC checksum
646 * @param cnt length of TYPE_FIL syntactic element in bytes
9cc04edf 647 *
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648 * @return Returns number of bytes consumed from the TYPE_FIL element.
649 */
650static int decode_sbr_extension(AACContext * ac, GetBitContext * gb, int crc, int cnt) {
651 // TODO : sbr_extension implementation
652 av_log(ac->avccontext, AV_LOG_DEBUG, "aac: SBR not yet supported.\n");
653 skip_bits_long(gb, 8*cnt - 4); // -4 due to reading extension type
654 return cnt;
655}
656
9cc04edf
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657/**
658 * Decode dynamic range information; reference: table 4.52.
659 *
660 * @param cnt length of TYPE_FIL syntactic element in bytes
661 *
662 * @return Returns number of bytes consumed.
663 */
664static int decode_dynamic_range(DynamicRangeControl *che_drc, GetBitContext * gb, int cnt) {
665 int n = 1;
666 int drc_num_bands = 1;
667 int i;
668
669 /* pce_tag_present? */
670 if(get_bits1(gb)) {
671 che_drc->pce_instance_tag = get_bits(gb, 4);
672 skip_bits(gb, 4); // tag_reserved_bits
673 n++;
674 }
675
676 /* excluded_chns_present? */
677 if(get_bits1(gb)) {
678 n += decode_drc_channel_exclusions(che_drc, gb);
679 }
680
681 /* drc_bands_present? */
682 if (get_bits1(gb)) {
683 che_drc->band_incr = get_bits(gb, 4);
684 che_drc->interpolation_scheme = get_bits(gb, 4);
685 n++;
686 drc_num_bands += che_drc->band_incr;
687 for (i = 0; i < drc_num_bands; i++) {
688 che_drc->band_top[i] = get_bits(gb, 8);
689 n++;
690 }
691 }
692
693 /* prog_ref_level_present? */
694 if (get_bits1(gb)) {
695 che_drc->prog_ref_level = get_bits(gb, 7);
696 skip_bits1(gb); // prog_ref_level_reserved_bits
697 n++;
698 }
699
700 for (i = 0; i < drc_num_bands; i++) {
701 che_drc->dyn_rng_sgn[i] = get_bits1(gb);
702 che_drc->dyn_rng_ctl[i] = get_bits(gb, 7);
703 n++;
704 }
705
706 return n;
707}
708
709/**
710 * Decode extension data (incomplete); reference: table 4.51.
711 *
712 * @param cnt length of TYPE_FIL syntactic element in bytes
713 *
714 * @return Returns number of bytes consumed
715 */
716static int decode_extension_payload(AACContext * ac, GetBitContext * gb, int cnt) {
cc0591da
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717 int crc_flag = 0;
718 int res = cnt;
719 switch (get_bits(gb, 4)) { // extension type
720 case EXT_SBR_DATA_CRC:
721 crc_flag++;
722 case EXT_SBR_DATA:
723 res = decode_sbr_extension(ac, gb, crc_flag, cnt);
724 break;
725 case EXT_DYNAMIC_RANGE:
726 res = decode_dynamic_range(&ac->che_drc, gb, cnt);
727 break;
728 case EXT_FILL:
729 case EXT_FILL_DATA:
730 case EXT_DATA_ELEMENT:
731 default:
732 skip_bits_long(gb, 8*cnt - 4);
733 break;
734 };
735 return res;
736}
737
738/**
9cc04edf
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739 * Conduct IMDCT and windowing.
740 */
741static void imdct_and_windowing(AACContext * ac, SingleChannelElement * sce) {
742 IndividualChannelStream * ics = &sce->ics;
743 float * in = sce->coeffs;
744 float * out = sce->ret;
745 float * saved = sce->saved;
746 const float * lwindow = ics->use_kb_window[0] ? ff_aac_kbd_long_1024 : ff_aac_sine_long_1024;
747 const float * swindow = ics->use_kb_window[0] ? ff_aac_kbd_short_128 : ff_aac_sine_short_128;
748 const float * lwindow_prev = ics->use_kb_window[1] ? ff_aac_kbd_long_1024 : ff_aac_sine_long_1024;
749 const float * swindow_prev = ics->use_kb_window[1] ? ff_aac_kbd_short_128 : ff_aac_sine_short_128;
750 float * buf = ac->buf_mdct;
751 int i;
752
753/**
cc0591da
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754 * Apply dependent channel coupling (applied before IMDCT).
755 *
756 * @param index index into coupling gain array
757 */
758static void apply_dependent_coupling(AACContext * ac, SingleChannelElement * sce, ChannelElement * cc, int index) {
759 IndividualChannelStream * ics = &cc->ch[0].ics;
760 const uint16_t * offsets = ics->swb_offset;
761 float * dest = sce->coeffs;
762 const float * src = cc->ch[0].coeffs;
763 int g, i, group, k, idx = 0;
764 if(ac->m4ac.object_type == AOT_AAC_LTP) {
765 av_log(ac->avccontext, AV_LOG_ERROR,
766 "Dependent coupling is not supported together with LTP\n");
767 return;
768 }
769 for (g = 0; g < ics->num_window_groups; g++) {
770 for (i = 0; i < ics->max_sfb; i++, idx++) {
771 if (cc->ch[0].band_type[idx] != ZERO_BT) {
772 float gain = cc->coup.gain[index][idx] * sce->mixing_gain;
773 for (group = 0; group < ics->group_len[g]; group++) {
774 for (k = offsets[i]; k < offsets[i+1]; k++) {
775 // XXX dsputil-ize
776 dest[group*128+k] += gain * src[group*128+k];
777 }
778 }
779 }
780 }
781 dest += ics->group_len[g]*128;
782 src += ics->group_len[g]*128;
783 }
784}
785
786/**
787 * Apply independent channel coupling (applied after IMDCT).
788 *
789 * @param index index into coupling gain array
790 */
791static void apply_independent_coupling(AACContext * ac, SingleChannelElement * sce, ChannelElement * cc, int index) {
792 int i;
793 float gain = cc->coup.gain[index][0] * sce->mixing_gain;
794 for (i = 0; i < 1024; i++)
795 sce->ret[i] += gain * (cc->ch[0].ret[i] - ac->add_bias);
796}
797
9cc04edf
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798 if (!ac->is_saved) {
799 ac->is_saved = 1;
800 *data_size = 0;
801 return 0;
802 }
803
804 data_size_tmp = 1024 * avccontext->channels * sizeof(int16_t);
805 if(*data_size < data_size_tmp) {
806 av_log(avccontext, AV_LOG_ERROR,
807 "Output buffer too small (%d) or trying to output too many samples (%d) for this frame.\n",
808 *data_size, data_size_tmp);
809 return -1;
810 }
811 *data_size = data_size_tmp;
812
813 ac->dsp.float_to_int16_interleave(data, (const float **)ac->output_data, 1024, avccontext->channels);
814
815 return buf_size;
816}
817
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818static av_cold int aac_decode_close(AVCodecContext * avccontext) {
819 AACContext * ac = avccontext->priv_data;
820 int i, j;
821
cc0591da 822 for (i = 0; i < MAX_ELEM_ID; i++) {
71e9a1b8
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823 for(j = 0; j < 4; j++)
824 av_freep(&ac->che[j][i]);
825 }
826
827 ff_mdct_end(&ac->mdct);
828 ff_mdct_end(&ac->mdct_small);
71e9a1b8
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829 return 0 ;
830}
831
832AVCodec aac_decoder = {
833 "aac",
834 CODEC_TYPE_AUDIO,
835 CODEC_ID_AAC,
836 sizeof(AACContext),
837 aac_decode_init,
838 NULL,
839 aac_decode_close,
840 aac_decode_frame,
841 .long_name = NULL_IF_CONFIG_SMALL("Advanced Audio Coding"),
cc0591da 842 .sample_fmts = (enum SampleFormat[]){SAMPLE_FMT_S16,SAMPLE_FMT_NONE},
71e9a1b8 843};