3 * Copyright (c) 2005-2006 Oded Shimon ( ods15 ods15 dyndns org )
4 * Copyright (c) 2006-2007 Maxim Gavrilov ( maxim.gavrilov gmail com )
6 * This file is part of FFmpeg.
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.
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.
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
24 * @file libavcodec/aac.c
26 * @author Oded Shimon ( ods15 ods15 dyndns org )
27 * @author Maxim Gavrilov ( maxim.gavrilov gmail com )
34 * N (code in SoC repo) gain control
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
44 * Y frequency domain prediction
45 * Y Perceptual Noise Substitution
47 * N Scalable Inverse AAC Quantization
48 * N Frequency Selective Switch
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
57 * N Silence Compression
60 * N Structured Audio tools
61 * N Structured Audio Sample Bank Format
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
73 * Note: - HE AAC v1 comprises LC AAC with Spectral Band Replication.
74 * - HE AAC v2 comprises LC AAC with Spectral Band Replication and
87 #include "aacdectab.h"
88 #include "mpeg4audio.h"
89 #include "aac_parser.h"
105 static VLC vlc_scalefactors;
106 static VLC vlc_spectral[11];
108 static uint32_t cbrt_tab[1<<13];
110 static const char overread_err[] = "Input buffer exhausted before END element found\n";
112 static ChannelElement *get_che(AACContext *ac, int type, int elem_id)
114 if (ac->tag_che_map[type][elem_id]) {
115 return ac->tag_che_map[type][elem_id];
117 if (ac->tags_mapped >= tags_per_config[ac->m4ac.chan_config]) {
120 switch (ac->m4ac.chan_config) {
122 if (ac->tags_mapped == 3 && type == TYPE_CPE) {
124 return ac->tag_che_map[TYPE_CPE][elem_id] = ac->che[TYPE_CPE][2];
127 /* Some streams incorrectly code 5.1 audio as SCE[0] CPE[0] CPE[1] SCE[1]
128 instead of SCE[0] CPE[0] CPE[0] LFE[0]. If we seem to have
129 encountered such a stream, transfer the LFE[0] element to SCE[1] */
130 if (ac->tags_mapped == tags_per_config[ac->m4ac.chan_config] - 1 && (type == TYPE_LFE || type == TYPE_SCE)) {
132 return ac->tag_che_map[type][elem_id] = ac->che[TYPE_LFE][0];
135 if (ac->tags_mapped == 2 && type == TYPE_CPE) {
137 return ac->tag_che_map[TYPE_CPE][elem_id] = ac->che[TYPE_CPE][1];
140 if (ac->tags_mapped == 2 && ac->m4ac.chan_config == 4 && type == TYPE_SCE) {
142 return ac->tag_che_map[TYPE_SCE][elem_id] = ac->che[TYPE_SCE][1];
146 if (ac->tags_mapped == (ac->m4ac.chan_config != 2) && type == TYPE_CPE) {
148 return ac->tag_che_map[TYPE_CPE][elem_id] = ac->che[TYPE_CPE][0];
149 } else if (ac->m4ac.chan_config == 2) {
153 if (!ac->tags_mapped && type == TYPE_SCE) {
155 return ac->tag_che_map[TYPE_SCE][elem_id] = ac->che[TYPE_SCE][0];
163 * Check for the channel element in the current channel position configuration.
164 * If it exists, make sure the appropriate element is allocated and map the
165 * channel order to match the internal FFmpeg channel layout.
167 * @param che_pos current channel position configuration
168 * @param type channel element type
169 * @param id channel element id
170 * @param channels count of the number of channels in the configuration
172 * @return Returns error status. 0 - OK, !0 - error
174 static av_cold int che_configure(AACContext *ac,
175 enum ChannelPosition che_pos[4][MAX_ELEM_ID],
179 if (che_pos[type][id]) {
180 if (!ac->che[type][id] && !(ac->che[type][id] = av_mallocz(sizeof(ChannelElement))))
181 return AVERROR(ENOMEM);
182 if (type != TYPE_CCE) {
183 ac->output_data[(*channels)++] = ac->che[type][id]->ch[0].ret;
184 if (type == TYPE_CPE) {
185 ac->output_data[(*channels)++] = ac->che[type][id]->ch[1].ret;
189 av_freep(&ac->che[type][id]);
194 * Configure output channel order based on the current program configuration element.
196 * @param che_pos current channel position configuration
197 * @param new_che_pos New channel position configuration - we only do something if it differs from the current one.
199 * @return Returns error status. 0 - OK, !0 - error
201 static av_cold int output_configure(AACContext *ac,
202 enum ChannelPosition che_pos[4][MAX_ELEM_ID],
203 enum ChannelPosition new_che_pos[4][MAX_ELEM_ID],
204 int channel_config, enum OCStatus oc_type)
206 AVCodecContext *avctx = ac->avccontext;
207 int i, type, channels = 0, ret;
209 memcpy(che_pos, new_che_pos, 4 * MAX_ELEM_ID * sizeof(new_che_pos[0][0]));
211 if (channel_config) {
212 for (i = 0; i < tags_per_config[channel_config]; i++) {
213 if ((ret = che_configure(ac, che_pos,
214 aac_channel_layout_map[channel_config - 1][i][0],
215 aac_channel_layout_map[channel_config - 1][i][1],
220 memset(ac->tag_che_map, 0, 4 * MAX_ELEM_ID * sizeof(ac->che[0][0]));
223 avctx->channel_layout = aac_channel_layout[channel_config - 1];
225 /* Allocate or free elements depending on if they are in the
226 * current program configuration.
228 * Set up default 1:1 output mapping.
230 * For a 5.1 stream the output order will be:
231 * [ Center ] [ Front Left ] [ Front Right ] [ LFE ] [ Surround Left ] [ Surround Right ]
234 for (i = 0; i < MAX_ELEM_ID; i++) {
235 for (type = 0; type < 4; type++) {
236 if ((ret = che_configure(ac, che_pos, type, i, &channels)))
241 memcpy(ac->tag_che_map, ac->che, 4 * MAX_ELEM_ID * sizeof(ac->che[0][0]));
242 ac->tags_mapped = 4 * MAX_ELEM_ID;
244 avctx->channel_layout = 0;
247 avctx->channels = channels;
249 ac->output_configured = oc_type;
255 * Decode an array of 4 bit element IDs, optionally interleaved with a stereo/mono switching bit.
257 * @param cpe_map Stereo (Channel Pair Element) map, NULL if stereo bit is not present.
258 * @param sce_map mono (Single Channel Element) map
259 * @param type speaker type/position for these channels
261 static void decode_channel_map(enum ChannelPosition *cpe_map,
262 enum ChannelPosition *sce_map,
263 enum ChannelPosition type,
264 GetBitContext *gb, int n)
267 enum ChannelPosition *map = cpe_map && get_bits1(gb) ? cpe_map : sce_map; // stereo or mono map
268 map[get_bits(gb, 4)] = type;
273 * Decode program configuration element; reference: table 4.2.
275 * @param new_che_pos New channel position configuration - we only do something if it differs from the current one.
277 * @return Returns error status. 0 - OK, !0 - error
279 static int decode_pce(AACContext *ac, enum ChannelPosition new_che_pos[4][MAX_ELEM_ID],
282 int num_front, num_side, num_back, num_lfe, num_assoc_data, num_cc, sampling_index;
285 skip_bits(gb, 2); // object_type
287 sampling_index = get_bits(gb, 4);
288 if (ac->m4ac.sampling_index != sampling_index)
289 av_log(ac->avccontext, AV_LOG_WARNING, "Sample rate index in program config element does not match the sample rate index configured by the container.\n");
291 num_front = get_bits(gb, 4);
292 num_side = get_bits(gb, 4);
293 num_back = get_bits(gb, 4);
294 num_lfe = get_bits(gb, 2);
295 num_assoc_data = get_bits(gb, 3);
296 num_cc = get_bits(gb, 4);
299 skip_bits(gb, 4); // mono_mixdown_tag
301 skip_bits(gb, 4); // stereo_mixdown_tag
304 skip_bits(gb, 3); // mixdown_coeff_index and pseudo_surround
306 decode_channel_map(new_che_pos[TYPE_CPE], new_che_pos[TYPE_SCE], AAC_CHANNEL_FRONT, gb, num_front);
307 decode_channel_map(new_che_pos[TYPE_CPE], new_che_pos[TYPE_SCE], AAC_CHANNEL_SIDE, gb, num_side );
308 decode_channel_map(new_che_pos[TYPE_CPE], new_che_pos[TYPE_SCE], AAC_CHANNEL_BACK, gb, num_back );
309 decode_channel_map(NULL, new_che_pos[TYPE_LFE], AAC_CHANNEL_LFE, gb, num_lfe );
311 skip_bits_long(gb, 4 * num_assoc_data);
313 decode_channel_map(new_che_pos[TYPE_CCE], new_che_pos[TYPE_CCE], AAC_CHANNEL_CC, gb, num_cc );
317 /* comment field, first byte is length */
318 comment_len = get_bits(gb, 8) * 8;
319 if (get_bits_left(gb) < comment_len) {
320 av_log(ac->avccontext, AV_LOG_ERROR, overread_err);
323 skip_bits_long(gb, comment_len);
328 * Set up channel positions based on a default channel configuration
329 * as specified in table 1.17.
331 * @param new_che_pos New channel position configuration - we only do something if it differs from the current one.
333 * @return Returns error status. 0 - OK, !0 - error
335 static av_cold int set_default_channel_config(AACContext *ac,
336 enum ChannelPosition new_che_pos[4][MAX_ELEM_ID],
339 if (channel_config < 1 || channel_config > 7) {
340 av_log(ac->avccontext, AV_LOG_ERROR, "invalid default channel configuration (%d)\n",
345 /* default channel configurations:
347 * 1ch : front center (mono)
348 * 2ch : L + R (stereo)
349 * 3ch : front center + L + R
350 * 4ch : front center + L + R + back center
351 * 5ch : front center + L + R + back stereo
352 * 6ch : front center + L + R + back stereo + LFE
353 * 7ch : front center + L + R + outer front left + outer front right + back stereo + LFE
356 if (channel_config != 2)
357 new_che_pos[TYPE_SCE][0] = AAC_CHANNEL_FRONT; // front center (or mono)
358 if (channel_config > 1)
359 new_che_pos[TYPE_CPE][0] = AAC_CHANNEL_FRONT; // L + R (or stereo)
360 if (channel_config == 4)
361 new_che_pos[TYPE_SCE][1] = AAC_CHANNEL_BACK; // back center
362 if (channel_config > 4)
363 new_che_pos[TYPE_CPE][(channel_config == 7) + 1]
364 = AAC_CHANNEL_BACK; // back stereo
365 if (channel_config > 5)
366 new_che_pos[TYPE_LFE][0] = AAC_CHANNEL_LFE; // LFE
367 if (channel_config == 7)
368 new_che_pos[TYPE_CPE][1] = AAC_CHANNEL_FRONT; // outer front left + outer front right
374 * Decode GA "General Audio" specific configuration; reference: table 4.1.
376 * @return Returns error status. 0 - OK, !0 - error
378 static int decode_ga_specific_config(AACContext *ac, GetBitContext *gb,
381 enum ChannelPosition new_che_pos[4][MAX_ELEM_ID];
382 int extension_flag, ret;
384 if (get_bits1(gb)) { // frameLengthFlag
385 av_log_missing_feature(ac->avccontext, "960/120 MDCT window is", 1);
389 if (get_bits1(gb)) // dependsOnCoreCoder
390 skip_bits(gb, 14); // coreCoderDelay
391 extension_flag = get_bits1(gb);
393 if (ac->m4ac.object_type == AOT_AAC_SCALABLE ||
394 ac->m4ac.object_type == AOT_ER_AAC_SCALABLE)
395 skip_bits(gb, 3); // layerNr
397 memset(new_che_pos, 0, 4 * MAX_ELEM_ID * sizeof(new_che_pos[0][0]));
398 if (channel_config == 0) {
399 skip_bits(gb, 4); // element_instance_tag
400 if ((ret = decode_pce(ac, new_che_pos, gb)))
403 if ((ret = set_default_channel_config(ac, new_che_pos, channel_config)))
406 if ((ret = output_configure(ac, ac->che_pos, new_che_pos, channel_config, OC_GLOBAL_HDR)))
409 if (extension_flag) {
410 switch (ac->m4ac.object_type) {
412 skip_bits(gb, 5); // numOfSubFrame
413 skip_bits(gb, 11); // layer_length
417 case AOT_ER_AAC_SCALABLE:
419 skip_bits(gb, 3); /* aacSectionDataResilienceFlag
420 * aacScalefactorDataResilienceFlag
421 * aacSpectralDataResilienceFlag
425 skip_bits1(gb); // extensionFlag3 (TBD in version 3)
431 * Decode audio specific configuration; reference: table 1.13.
433 * @param data pointer to AVCodecContext extradata
434 * @param data_size size of AVCCodecContext extradata
436 * @return Returns error status. 0 - OK, !0 - error
438 static int decode_audio_specific_config(AACContext *ac, void *data,
444 init_get_bits(&gb, data, data_size * 8);
446 if ((i = ff_mpeg4audio_get_config(&ac->m4ac, data, data_size)) < 0)
448 if (ac->m4ac.sampling_index > 12) {
449 av_log(ac->avccontext, AV_LOG_ERROR, "invalid sampling rate index %d\n", ac->m4ac.sampling_index);
453 skip_bits_long(&gb, i);
455 switch (ac->m4ac.object_type) {
458 if (decode_ga_specific_config(ac, &gb, ac->m4ac.chan_config))
462 av_log(ac->avccontext, AV_LOG_ERROR, "Audio object type %s%d is not supported.\n",
463 ac->m4ac.sbr == 1? "SBR+" : "", ac->m4ac.object_type);
470 * linear congruential pseudorandom number generator
472 * @param previous_val pointer to the current state of the generator
474 * @return Returns a 32-bit pseudorandom integer
476 static av_always_inline int lcg_random(int previous_val)
478 return previous_val * 1664525 + 1013904223;
481 static void reset_predict_state(PredictorState *ps)
491 static void reset_all_predictors(PredictorState *ps)
494 for (i = 0; i < MAX_PREDICTORS; i++)
495 reset_predict_state(&ps[i]);
498 static void reset_predictor_group(PredictorState *ps, int group_num)
501 for (i = group_num - 1; i < MAX_PREDICTORS; i += 30)
502 reset_predict_state(&ps[i]);
505 static av_cold int aac_decode_init(AVCodecContext *avccontext)
507 AACContext *ac = avccontext->priv_data;
510 ac->avccontext = avccontext;
512 if (avccontext->extradata_size > 0) {
513 if (decode_audio_specific_config(ac, avccontext->extradata, avccontext->extradata_size))
515 avccontext->sample_rate = ac->m4ac.sample_rate;
516 } else if (avccontext->channels > 0) {
517 ac->m4ac.sample_rate = avccontext->sample_rate;
520 avccontext->sample_fmt = SAMPLE_FMT_S16;
521 avccontext->frame_size = 1024;
523 AAC_INIT_VLC_STATIC( 0, 304);
524 AAC_INIT_VLC_STATIC( 1, 270);
525 AAC_INIT_VLC_STATIC( 2, 550);
526 AAC_INIT_VLC_STATIC( 3, 300);
527 AAC_INIT_VLC_STATIC( 4, 328);
528 AAC_INIT_VLC_STATIC( 5, 294);
529 AAC_INIT_VLC_STATIC( 6, 306);
530 AAC_INIT_VLC_STATIC( 7, 268);
531 AAC_INIT_VLC_STATIC( 8, 510);
532 AAC_INIT_VLC_STATIC( 9, 366);
533 AAC_INIT_VLC_STATIC(10, 462);
535 dsputil_init(&ac->dsp, avccontext);
537 ac->random_state = 0x1f2e3d4c;
539 // -1024 - Compensate wrong IMDCT method.
540 // 32768 - Required to scale values to the correct range for the bias method
541 // for float to int16 conversion.
543 if (ac->dsp.float_to_int16_interleave == ff_float_to_int16_interleave_c) {
544 ac->add_bias = 385.0f;
545 ac->sf_scale = 1. / (-1024. * 32768.);
549 ac->sf_scale = 1. / -1024.;
553 #if !CONFIG_HARDCODED_TABLES
554 for (i = 0; i < 428; i++)
555 ff_aac_pow2sf_tab[i] = pow(2, (i - 200) / 4.);
556 #endif /* CONFIG_HARDCODED_TABLES */
558 INIT_VLC_STATIC(&vlc_scalefactors,7,FF_ARRAY_ELEMS(ff_aac_scalefactor_code),
559 ff_aac_scalefactor_bits, sizeof(ff_aac_scalefactor_bits[0]), sizeof(ff_aac_scalefactor_bits[0]),
560 ff_aac_scalefactor_code, sizeof(ff_aac_scalefactor_code[0]), sizeof(ff_aac_scalefactor_code[0]),
563 ff_mdct_init(&ac->mdct, 11, 1, 1.0);
564 ff_mdct_init(&ac->mdct_small, 8, 1, 1.0);
565 // window initialization
566 ff_kbd_window_init(ff_aac_kbd_long_1024, 4.0, 1024);
567 ff_kbd_window_init(ff_aac_kbd_short_128, 6.0, 128);
568 ff_init_ff_sine_windows(10);
569 ff_init_ff_sine_windows( 7);
571 if (!cbrt_tab[(1<<13) - 1]) {
572 for (i = 0; i < 1<<13; i++) {
583 * Skip data_stream_element; reference: table 4.10.
585 static int skip_data_stream_element(AACContext *ac, GetBitContext *gb)
587 int byte_align = get_bits1(gb);
588 int count = get_bits(gb, 8);
590 count += get_bits(gb, 8);
594 if (get_bits_left(gb) < 8 * count) {
595 av_log(ac->avccontext, AV_LOG_ERROR, overread_err);
598 skip_bits_long(gb, 8 * count);
602 static int decode_prediction(AACContext *ac, IndividualChannelStream *ics,
607 ics->predictor_reset_group = get_bits(gb, 5);
608 if (ics->predictor_reset_group == 0 || ics->predictor_reset_group > 30) {
609 av_log(ac->avccontext, AV_LOG_ERROR, "Invalid Predictor Reset Group.\n");
613 for (sfb = 0; sfb < FFMIN(ics->max_sfb, ff_aac_pred_sfb_max[ac->m4ac.sampling_index]); sfb++) {
614 ics->prediction_used[sfb] = get_bits1(gb);
620 * Decode Individual Channel Stream info; reference: table 4.6.
622 * @param common_window Channels have independent [0], or shared [1], Individual Channel Stream information.
624 static int decode_ics_info(AACContext *ac, IndividualChannelStream *ics,
625 GetBitContext *gb, int common_window)
628 av_log(ac->avccontext, AV_LOG_ERROR, "Reserved bit set.\n");
629 memset(ics, 0, sizeof(IndividualChannelStream));
632 ics->window_sequence[1] = ics->window_sequence[0];
633 ics->window_sequence[0] = get_bits(gb, 2);
634 ics->use_kb_window[1] = ics->use_kb_window[0];
635 ics->use_kb_window[0] = get_bits1(gb);
636 ics->num_window_groups = 1;
637 ics->group_len[0] = 1;
638 if (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) {
640 ics->max_sfb = get_bits(gb, 4);
641 for (i = 0; i < 7; i++) {
643 ics->group_len[ics->num_window_groups - 1]++;
645 ics->num_window_groups++;
646 ics->group_len[ics->num_window_groups - 1] = 1;
649 ics->num_windows = 8;
650 ics->swb_offset = ff_swb_offset_128[ac->m4ac.sampling_index];
651 ics->num_swb = ff_aac_num_swb_128[ac->m4ac.sampling_index];
652 ics->tns_max_bands = ff_tns_max_bands_128[ac->m4ac.sampling_index];
653 ics->predictor_present = 0;
655 ics->max_sfb = get_bits(gb, 6);
656 ics->num_windows = 1;
657 ics->swb_offset = ff_swb_offset_1024[ac->m4ac.sampling_index];
658 ics->num_swb = ff_aac_num_swb_1024[ac->m4ac.sampling_index];
659 ics->tns_max_bands = ff_tns_max_bands_1024[ac->m4ac.sampling_index];
660 ics->predictor_present = get_bits1(gb);
661 ics->predictor_reset_group = 0;
662 if (ics->predictor_present) {
663 if (ac->m4ac.object_type == AOT_AAC_MAIN) {
664 if (decode_prediction(ac, ics, gb)) {
665 memset(ics, 0, sizeof(IndividualChannelStream));
668 } else if (ac->m4ac.object_type == AOT_AAC_LC) {
669 av_log(ac->avccontext, AV_LOG_ERROR, "Prediction is not allowed in AAC-LC.\n");
670 memset(ics, 0, sizeof(IndividualChannelStream));
673 av_log_missing_feature(ac->avccontext, "Predictor bit set but LTP is", 1);
674 memset(ics, 0, sizeof(IndividualChannelStream));
680 if (ics->max_sfb > ics->num_swb) {
681 av_log(ac->avccontext, AV_LOG_ERROR,
682 "Number of scalefactor bands in group (%d) exceeds limit (%d).\n",
683 ics->max_sfb, ics->num_swb);
684 memset(ics, 0, sizeof(IndividualChannelStream));
692 * Decode band types (section_data payload); reference: table 4.46.
694 * @param band_type array of the used band type
695 * @param band_type_run_end array of the last scalefactor band of a band type run
697 * @return Returns error status. 0 - OK, !0 - error
699 static int decode_band_types(AACContext *ac, enum BandType band_type[120],
700 int band_type_run_end[120], GetBitContext *gb,
701 IndividualChannelStream *ics)
704 const int bits = (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) ? 3 : 5;
705 for (g = 0; g < ics->num_window_groups; g++) {
707 while (k < ics->max_sfb) {
708 uint8_t sect_end = k;
710 int sect_band_type = get_bits(gb, 4);
711 if (sect_band_type == 12) {
712 av_log(ac->avccontext, AV_LOG_ERROR, "invalid band type\n");
715 while ((sect_len_incr = get_bits(gb, bits)) == (1 << bits) - 1)
716 sect_end += sect_len_incr;
717 sect_end += sect_len_incr;
718 if (sect_end > ics->max_sfb) {
719 av_log(ac->avccontext, AV_LOG_ERROR,
720 "Number of bands (%d) exceeds limit (%d).\n",
721 sect_end, ics->max_sfb);
724 for (; k < sect_end; k++) {
725 band_type [idx] = sect_band_type;
726 band_type_run_end[idx++] = sect_end;
734 * Decode scalefactors; reference: table 4.47.
736 * @param global_gain first scalefactor value as scalefactors are differentially coded
737 * @param band_type array of the used band type
738 * @param band_type_run_end array of the last scalefactor band of a band type run
739 * @param sf array of scalefactors or intensity stereo positions
741 * @return Returns error status. 0 - OK, !0 - error
743 static int decode_scalefactors(AACContext *ac, float sf[120], GetBitContext *gb,
744 unsigned int global_gain,
745 IndividualChannelStream *ics,
746 enum BandType band_type[120],
747 int band_type_run_end[120])
749 const int sf_offset = ac->sf_offset + (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE ? 12 : 0);
751 int offset[3] = { global_gain, global_gain - 90, 100 };
753 static const char *sf_str[3] = { "Global gain", "Noise gain", "Intensity stereo position" };
754 for (g = 0; g < ics->num_window_groups; g++) {
755 for (i = 0; i < ics->max_sfb;) {
756 int run_end = band_type_run_end[idx];
757 if (band_type[idx] == ZERO_BT) {
758 for (; i < run_end; i++, idx++)
760 } else if ((band_type[idx] == INTENSITY_BT) || (band_type[idx] == INTENSITY_BT2)) {
761 for (; i < run_end; i++, idx++) {
762 offset[2] += get_vlc2(gb, vlc_scalefactors.table, 7, 3) - 60;
763 if (offset[2] > 255U) {
764 av_log(ac->avccontext, AV_LOG_ERROR,
765 "%s (%d) out of range.\n", sf_str[2], offset[2]);
768 sf[idx] = ff_aac_pow2sf_tab[-offset[2] + 300];
770 } else if (band_type[idx] == NOISE_BT) {
771 for (; i < run_end; i++, idx++) {
772 if (noise_flag-- > 0)
773 offset[1] += get_bits(gb, 9) - 256;
775 offset[1] += get_vlc2(gb, vlc_scalefactors.table, 7, 3) - 60;
776 if (offset[1] > 255U) {
777 av_log(ac->avccontext, AV_LOG_ERROR,
778 "%s (%d) out of range.\n", sf_str[1], offset[1]);
781 sf[idx] = -ff_aac_pow2sf_tab[offset[1] + sf_offset + 100];
784 for (; i < run_end; i++, idx++) {
785 offset[0] += get_vlc2(gb, vlc_scalefactors.table, 7, 3) - 60;
786 if (offset[0] > 255U) {
787 av_log(ac->avccontext, AV_LOG_ERROR,
788 "%s (%d) out of range.\n", sf_str[0], offset[0]);
791 sf[idx] = -ff_aac_pow2sf_tab[ offset[0] + sf_offset];
800 * Decode pulse data; reference: table 4.7.
802 static int decode_pulses(Pulse *pulse, GetBitContext *gb,
803 const uint16_t *swb_offset, int num_swb)
806 pulse->num_pulse = get_bits(gb, 2) + 1;
807 pulse_swb = get_bits(gb, 6);
808 if (pulse_swb >= num_swb)
810 pulse->pos[0] = swb_offset[pulse_swb];
811 pulse->pos[0] += get_bits(gb, 5);
812 if (pulse->pos[0] > 1023)
814 pulse->amp[0] = get_bits(gb, 4);
815 for (i = 1; i < pulse->num_pulse; i++) {
816 pulse->pos[i] = get_bits(gb, 5) + pulse->pos[i - 1];
817 if (pulse->pos[i] > 1023)
819 pulse->amp[i] = get_bits(gb, 4);
825 * Decode Temporal Noise Shaping data; reference: table 4.48.
827 * @return Returns error status. 0 - OK, !0 - error
829 static int decode_tns(AACContext *ac, TemporalNoiseShaping *tns,
830 GetBitContext *gb, const IndividualChannelStream *ics)
832 int w, filt, i, coef_len, coef_res, coef_compress;
833 const int is8 = ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE;
834 const int tns_max_order = is8 ? 7 : ac->m4ac.object_type == AOT_AAC_MAIN ? 20 : 12;
835 for (w = 0; w < ics->num_windows; w++) {
836 if ((tns->n_filt[w] = get_bits(gb, 2 - is8))) {
837 coef_res = get_bits1(gb);
839 for (filt = 0; filt < tns->n_filt[w]; filt++) {
841 tns->length[w][filt] = get_bits(gb, 6 - 2 * is8);
843 if ((tns->order[w][filt] = get_bits(gb, 5 - 2 * is8)) > tns_max_order) {
844 av_log(ac->avccontext, AV_LOG_ERROR, "TNS filter order %d is greater than maximum %d.",
845 tns->order[w][filt], tns_max_order);
846 tns->order[w][filt] = 0;
849 if (tns->order[w][filt]) {
850 tns->direction[w][filt] = get_bits1(gb);
851 coef_compress = get_bits1(gb);
852 coef_len = coef_res + 3 - coef_compress;
853 tmp2_idx = 2 * coef_compress + coef_res;
855 for (i = 0; i < tns->order[w][filt]; i++)
856 tns->coef[w][filt][i] = tns_tmp2_map[tmp2_idx][get_bits(gb, coef_len)];
865 * Decode Mid/Side data; reference: table 4.54.
867 * @param ms_present Indicates mid/side stereo presence. [0] mask is all 0s;
868 * [1] mask is decoded from bitstream; [2] mask is all 1s;
869 * [3] reserved for scalable AAC
871 static void decode_mid_side_stereo(ChannelElement *cpe, GetBitContext *gb,
875 if (ms_present == 1) {
876 for (idx = 0; idx < cpe->ch[0].ics.num_window_groups * cpe->ch[0].ics.max_sfb; idx++)
877 cpe->ms_mask[idx] = get_bits1(gb);
878 } else if (ms_present == 2) {
879 memset(cpe->ms_mask, 1, cpe->ch[0].ics.num_window_groups * cpe->ch[0].ics.max_sfb * sizeof(cpe->ms_mask[0]));
884 static inline float *VMUL2(float *dst, const float *v, unsigned idx,
888 *dst++ = v[idx & 15] * s;
889 *dst++ = v[idx>>4 & 15] * s;
895 static inline float *VMUL4(float *dst, const float *v, unsigned idx,
899 *dst++ = v[idx & 3] * s;
900 *dst++ = v[idx>>2 & 3] * s;
901 *dst++ = v[idx>>4 & 3] * s;
902 *dst++ = v[idx>>6 & 3] * s;
908 static inline float *VMUL2S(float *dst, const float *v, unsigned idx,
909 unsigned sign, const float *scale)
911 union float754 s0, s1;
913 s0.f = s1.f = *scale;
914 s0.i ^= sign >> 1 << 31;
917 *dst++ = v[idx & 15] * s0.f;
918 *dst++ = v[idx>>4 & 15] * s1.f;
925 static inline float *VMUL4S(float *dst, const float *v, unsigned idx,
926 unsigned sign, const float *scale)
928 unsigned nz = idx >> 12;
929 union float754 s = { .f = *scale };
932 t.i = s.i ^ (sign & 1<<31);
933 *dst++ = v[idx & 3] * t.f;
935 sign <<= nz & 1; nz >>= 1;
936 t.i = s.i ^ (sign & 1<<31);
937 *dst++ = v[idx>>2 & 3] * t.f;
939 sign <<= nz & 1; nz >>= 1;
940 t.i = s.i ^ (sign & 1<<31);
941 *dst++ = v[idx>>4 & 3] * t.f;
943 sign <<= nz & 1; nz >>= 1;
944 t.i = s.i ^ (sign & 1<<31);
945 *dst++ = v[idx>>6 & 3] * t.f;
952 * Decode spectral data; reference: table 4.50.
953 * Dequantize and scale spectral data; reference: 4.6.3.3.
955 * @param coef array of dequantized, scaled spectral data
956 * @param sf array of scalefactors or intensity stereo positions
957 * @param pulse_present set if pulses are present
958 * @param pulse pointer to pulse data struct
959 * @param band_type array of the used band type
961 * @return Returns error status. 0 - OK, !0 - error
963 static int decode_spectrum_and_dequant(AACContext *ac, float coef[1024],
964 GetBitContext *gb, const float sf[120],
965 int pulse_present, const Pulse *pulse,
966 const IndividualChannelStream *ics,
967 enum BandType band_type[120])
969 int i, k, g, idx = 0;
970 const int c = 1024 / ics->num_windows;
971 const uint16_t *offsets = ics->swb_offset;
972 float *coef_base = coef;
975 for (g = 0; g < ics->num_windows; g++)
976 memset(coef + g * 128 + offsets[ics->max_sfb], 0, sizeof(float) * (c - offsets[ics->max_sfb]));
978 for (g = 0; g < ics->num_window_groups; g++) {
979 unsigned g_len = ics->group_len[g];
981 for (i = 0; i < ics->max_sfb; i++, idx++) {
982 const unsigned cbt_m1 = band_type[idx] - 1;
983 float *cfo = coef + offsets[i];
984 int off_len = offsets[i + 1] - offsets[i];
987 if (cbt_m1 >= INTENSITY_BT2 - 1) {
988 for (group = 0; group < g_len; group++, cfo+=128) {
989 memset(cfo, 0, off_len * sizeof(float));
991 } else if (cbt_m1 == NOISE_BT - 1) {
992 for (group = 0; group < g_len; group++, cfo+=128) {
996 for (k = 0; k < off_len; k++) {
997 ac->random_state = lcg_random(ac->random_state);
998 cfo[k] = ac->random_state;
1001 band_energy = ac->dsp.scalarproduct_float(cfo, cfo, off_len);
1002 scale = sf[idx] / sqrtf(band_energy);
1003 ac->dsp.vector_fmul_scalar(cfo, cfo, scale, off_len);
1006 const float *vq = ff_aac_codebook_vector_vals[cbt_m1];
1007 const uint16_t *cb_vector_idx = ff_aac_codebook_vector_idx[cbt_m1];
1008 VLC_TYPE (*vlc_tab)[2] = vlc_spectral[cbt_m1].table;
1009 const int cb_size = ff_aac_spectral_sizes[cbt_m1];
1010 OPEN_READER(re, gb);
1012 switch (cbt_m1 >> 1) {
1014 for (group = 0; group < g_len; group++, cfo+=128) {
1022 UPDATE_CACHE(re, gb);
1023 GET_VLC(code, re, gb, vlc_tab, 8, 2);
1025 if (code >= cb_size) {
1027 goto err_cb_overflow;
1030 cb_idx = cb_vector_idx[code];
1031 cf = VMUL4(cf, vq, cb_idx, sf + idx);
1037 for (group = 0; group < g_len; group++, cfo+=128) {
1047 UPDATE_CACHE(re, gb);
1048 GET_VLC(code, re, gb, vlc_tab, 8, 2);
1050 if (code >= cb_size) {
1052 goto err_cb_overflow;
1055 #if MIN_CACHE_BITS < 20
1056 UPDATE_CACHE(re, gb);
1058 cb_idx = cb_vector_idx[code];
1059 nnz = cb_idx >> 8 & 15;
1060 bits = SHOW_UBITS(re, gb, nnz) << (32-nnz);
1061 LAST_SKIP_BITS(re, gb, nnz);
1062 cf = VMUL4S(cf, vq, cb_idx, bits, sf + idx);
1068 for (group = 0; group < g_len; group++, cfo+=128) {
1076 UPDATE_CACHE(re, gb);
1077 GET_VLC(code, re, gb, vlc_tab, 8, 2);
1079 if (code >= cb_size) {
1081 goto err_cb_overflow;
1084 cb_idx = cb_vector_idx[code];
1085 cf = VMUL2(cf, vq, cb_idx, sf + idx);
1092 for (group = 0; group < g_len; group++, cfo+=128) {
1102 UPDATE_CACHE(re, gb);
1103 GET_VLC(code, re, gb, vlc_tab, 8, 2);
1105 if (code >= cb_size) {
1107 goto err_cb_overflow;
1110 cb_idx = cb_vector_idx[code];
1111 nnz = cb_idx >> 8 & 15;
1112 sign = SHOW_UBITS(re, gb, nnz) << (cb_idx >> 12);
1113 LAST_SKIP_BITS(re, gb, nnz);
1114 cf = VMUL2S(cf, vq, cb_idx, sign, sf + idx);
1120 for (group = 0; group < g_len; group++, cfo+=128) {
1122 uint32_t *icf = (uint32_t *) cf;
1132 UPDATE_CACHE(re, gb);
1133 GET_VLC(code, re, gb, vlc_tab, 8, 2);
1141 if (code >= cb_size) {
1143 goto err_cb_overflow;
1146 cb_idx = cb_vector_idx[code];
1149 bits = SHOW_UBITS(re, gb, nnz) << (32-nnz);
1150 LAST_SKIP_BITS(re, gb, nnz);
1152 for (j = 0; j < 2; j++) {
1156 /* The total length of escape_sequence must be < 22 bits according
1157 to the specification (i.e. max is 111111110xxxxxxxxxxxx). */
1158 UPDATE_CACHE(re, gb);
1159 b = GET_CACHE(re, gb);
1160 b = 31 - av_log2(~b);
1163 av_log(ac->avccontext, AV_LOG_ERROR, "error in spectral data, ESC overflow\n");
1167 #if MIN_CACHE_BITS < 21
1168 LAST_SKIP_BITS(re, gb, b + 1);
1169 UPDATE_CACHE(re, gb);
1171 SKIP_BITS(re, gb, b + 1);
1174 n = (1 << b) + SHOW_UBITS(re, gb, b);
1175 LAST_SKIP_BITS(re, gb, b);
1176 *icf++ = cbrt_tab[n] | (bits & 1<<31);
1179 unsigned v = ((const uint32_t*)vq)[cb_idx & 15];
1180 *icf++ = (bits & 1<<31) | v;
1187 ac->dsp.vector_fmul_scalar(cfo, cfo, sf[idx], off_len);
1191 CLOSE_READER(re, gb);
1197 if (pulse_present) {
1199 for (i = 0; i < pulse->num_pulse; i++) {
1200 float co = coef_base[ pulse->pos[i] ];
1201 while (offsets[idx + 1] <= pulse->pos[i])
1203 if (band_type[idx] != NOISE_BT && sf[idx]) {
1204 float ico = -pulse->amp[i];
1207 ico = co / sqrtf(sqrtf(fabsf(co))) + (co > 0 ? -ico : ico);
1209 coef_base[ pulse->pos[i] ] = cbrtf(fabsf(ico)) * ico * sf[idx];
1216 av_log(ac->avccontext, AV_LOG_ERROR,
1217 "Read beyond end of ff_aac_codebook_vectors[%d][]. index %d >= %d\n",
1218 band_type[idx], err_idx, ff_aac_spectral_sizes[band_type[idx]]);
1222 static av_always_inline float flt16_round(float pf)
1226 tmp.i = (tmp.i + 0x00008000U) & 0xFFFF0000U;
1230 static av_always_inline float flt16_even(float pf)
1234 tmp.i = (tmp.i + 0x00007FFFU + (tmp.i & 0x00010000U >> 16)) & 0xFFFF0000U;
1238 static av_always_inline float flt16_trunc(float pf)
1242 pun.i &= 0xFFFF0000U;
1246 static void predict(AACContext *ac, PredictorState *ps, float *coef,
1249 const float a = 0.953125; // 61.0 / 64
1250 const float alpha = 0.90625; // 29.0 / 32
1255 k1 = ps->var0 > 1 ? ps->cor0 * flt16_even(a / ps->var0) : 0;
1256 k2 = ps->var1 > 1 ? ps->cor1 * flt16_even(a / ps->var1) : 0;
1258 pv = flt16_round(k1 * ps->r0 + k2 * ps->r1);
1260 *coef += pv * ac->sf_scale;
1262 e0 = *coef / ac->sf_scale;
1263 e1 = e0 - k1 * ps->r0;
1265 ps->cor1 = flt16_trunc(alpha * ps->cor1 + ps->r1 * e1);
1266 ps->var1 = flt16_trunc(alpha * ps->var1 + 0.5 * (ps->r1 * ps->r1 + e1 * e1));
1267 ps->cor0 = flt16_trunc(alpha * ps->cor0 + ps->r0 * e0);
1268 ps->var0 = flt16_trunc(alpha * ps->var0 + 0.5 * (ps->r0 * ps->r0 + e0 * e0));
1270 ps->r1 = flt16_trunc(a * (ps->r0 - k1 * e0));
1271 ps->r0 = flt16_trunc(a * e0);
1275 * Apply AAC-Main style frequency domain prediction.
1277 static void apply_prediction(AACContext *ac, SingleChannelElement *sce)
1281 if (!sce->ics.predictor_initialized) {
1282 reset_all_predictors(sce->predictor_state);
1283 sce->ics.predictor_initialized = 1;
1286 if (sce->ics.window_sequence[0] != EIGHT_SHORT_SEQUENCE) {
1287 for (sfb = 0; sfb < ff_aac_pred_sfb_max[ac->m4ac.sampling_index]; sfb++) {
1288 for (k = sce->ics.swb_offset[sfb]; k < sce->ics.swb_offset[sfb + 1]; k++) {
1289 predict(ac, &sce->predictor_state[k], &sce->coeffs[k],
1290 sce->ics.predictor_present && sce->ics.prediction_used[sfb]);
1293 if (sce->ics.predictor_reset_group)
1294 reset_predictor_group(sce->predictor_state, sce->ics.predictor_reset_group);
1296 reset_all_predictors(sce->predictor_state);
1300 * Decode an individual_channel_stream payload; reference: table 4.44.
1302 * @param common_window Channels have independent [0], or shared [1], Individual Channel Stream information.
1303 * @param scale_flag scalable [1] or non-scalable [0] AAC (Unused until scalable AAC is implemented.)
1305 * @return Returns error status. 0 - OK, !0 - error
1307 static int decode_ics(AACContext *ac, SingleChannelElement *sce,
1308 GetBitContext *gb, int common_window, int scale_flag)
1311 TemporalNoiseShaping *tns = &sce->tns;
1312 IndividualChannelStream *ics = &sce->ics;
1313 float *out = sce->coeffs;
1314 int global_gain, pulse_present = 0;
1316 /* This assignment is to silence a GCC warning about the variable being used
1317 * uninitialized when in fact it always is.
1319 pulse.num_pulse = 0;
1321 global_gain = get_bits(gb, 8);
1323 if (!common_window && !scale_flag) {
1324 if (decode_ics_info(ac, ics, gb, 0) < 0)
1328 if (decode_band_types(ac, sce->band_type, sce->band_type_run_end, gb, ics) < 0)
1330 if (decode_scalefactors(ac, sce->sf, gb, global_gain, ics, sce->band_type, sce->band_type_run_end) < 0)
1335 if ((pulse_present = get_bits1(gb))) {
1336 if (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) {
1337 av_log(ac->avccontext, AV_LOG_ERROR, "Pulse tool not allowed in eight short sequence.\n");
1340 if (decode_pulses(&pulse, gb, ics->swb_offset, ics->num_swb)) {
1341 av_log(ac->avccontext, AV_LOG_ERROR, "Pulse data corrupt or invalid.\n");
1345 if ((tns->present = get_bits1(gb)) && decode_tns(ac, tns, gb, ics))
1347 if (get_bits1(gb)) {
1348 av_log_missing_feature(ac->avccontext, "SSR", 1);
1353 if (decode_spectrum_and_dequant(ac, out, gb, sce->sf, pulse_present, &pulse, ics, sce->band_type) < 0)
1356 if (ac->m4ac.object_type == AOT_AAC_MAIN && !common_window)
1357 apply_prediction(ac, sce);
1363 * Mid/Side stereo decoding; reference: 4.6.8.1.3.
1365 static void apply_mid_side_stereo(AACContext *ac, ChannelElement *cpe)
1367 const IndividualChannelStream *ics = &cpe->ch[0].ics;
1368 float *ch0 = cpe->ch[0].coeffs;
1369 float *ch1 = cpe->ch[1].coeffs;
1370 int g, i, group, idx = 0;
1371 const uint16_t *offsets = ics->swb_offset;
1372 for (g = 0; g < ics->num_window_groups; g++) {
1373 for (i = 0; i < ics->max_sfb; i++, idx++) {
1374 if (cpe->ms_mask[idx] &&
1375 cpe->ch[0].band_type[idx] < NOISE_BT && cpe->ch[1].band_type[idx] < NOISE_BT) {
1376 for (group = 0; group < ics->group_len[g]; group++) {
1377 ac->dsp.butterflies_float(ch0 + group * 128 + offsets[i],
1378 ch1 + group * 128 + offsets[i],
1379 offsets[i+1] - offsets[i]);
1383 ch0 += ics->group_len[g] * 128;
1384 ch1 += ics->group_len[g] * 128;
1389 * intensity stereo decoding; reference: 4.6.8.2.3
1391 * @param ms_present Indicates mid/side stereo presence. [0] mask is all 0s;
1392 * [1] mask is decoded from bitstream; [2] mask is all 1s;
1393 * [3] reserved for scalable AAC
1395 static void apply_intensity_stereo(ChannelElement *cpe, int ms_present)
1397 const IndividualChannelStream *ics = &cpe->ch[1].ics;
1398 SingleChannelElement *sce1 = &cpe->ch[1];
1399 float *coef0 = cpe->ch[0].coeffs, *coef1 = cpe->ch[1].coeffs;
1400 const uint16_t *offsets = ics->swb_offset;
1401 int g, group, i, k, idx = 0;
1404 for (g = 0; g < ics->num_window_groups; g++) {
1405 for (i = 0; i < ics->max_sfb;) {
1406 if (sce1->band_type[idx] == INTENSITY_BT || sce1->band_type[idx] == INTENSITY_BT2) {
1407 const int bt_run_end = sce1->band_type_run_end[idx];
1408 for (; i < bt_run_end; i++, idx++) {
1409 c = -1 + 2 * (sce1->band_type[idx] - 14);
1411 c *= 1 - 2 * cpe->ms_mask[idx];
1412 scale = c * sce1->sf[idx];
1413 for (group = 0; group < ics->group_len[g]; group++)
1414 for (k = offsets[i]; k < offsets[i + 1]; k++)
1415 coef1[group * 128 + k] = scale * coef0[group * 128 + k];
1418 int bt_run_end = sce1->band_type_run_end[idx];
1419 idx += bt_run_end - i;
1423 coef0 += ics->group_len[g] * 128;
1424 coef1 += ics->group_len[g] * 128;
1429 * Decode a channel_pair_element; reference: table 4.4.
1431 * @param elem_id Identifies the instance of a syntax element.
1433 * @return Returns error status. 0 - OK, !0 - error
1435 static int decode_cpe(AACContext *ac, GetBitContext *gb, ChannelElement *cpe)
1437 int i, ret, common_window, ms_present = 0;
1439 common_window = get_bits1(gb);
1440 if (common_window) {
1441 if (decode_ics_info(ac, &cpe->ch[0].ics, gb, 1))
1443 i = cpe->ch[1].ics.use_kb_window[0];
1444 cpe->ch[1].ics = cpe->ch[0].ics;
1445 cpe->ch[1].ics.use_kb_window[1] = i;
1446 ms_present = get_bits(gb, 2);
1447 if (ms_present == 3) {
1448 av_log(ac->avccontext, AV_LOG_ERROR, "ms_present = 3 is reserved.\n");
1450 } else if (ms_present)
1451 decode_mid_side_stereo(cpe, gb, ms_present);
1453 if ((ret = decode_ics(ac, &cpe->ch[0], gb, common_window, 0)))
1455 if ((ret = decode_ics(ac, &cpe->ch[1], gb, common_window, 0)))
1458 if (common_window) {
1460 apply_mid_side_stereo(ac, cpe);
1461 if (ac->m4ac.object_type == AOT_AAC_MAIN) {
1462 apply_prediction(ac, &cpe->ch[0]);
1463 apply_prediction(ac, &cpe->ch[1]);
1467 apply_intensity_stereo(cpe, ms_present);
1472 * Decode coupling_channel_element; reference: table 4.8.
1474 * @param elem_id Identifies the instance of a syntax element.
1476 * @return Returns error status. 0 - OK, !0 - error
1478 static int decode_cce(AACContext *ac, GetBitContext *gb, ChannelElement *che)
1484 SingleChannelElement *sce = &che->ch[0];
1485 ChannelCoupling *coup = &che->coup;
1487 coup->coupling_point = 2 * get_bits1(gb);
1488 coup->num_coupled = get_bits(gb, 3);
1489 for (c = 0; c <= coup->num_coupled; c++) {
1491 coup->type[c] = get_bits1(gb) ? TYPE_CPE : TYPE_SCE;
1492 coup->id_select[c] = get_bits(gb, 4);
1493 if (coup->type[c] == TYPE_CPE) {
1494 coup->ch_select[c] = get_bits(gb, 2);
1495 if (coup->ch_select[c] == 3)
1498 coup->ch_select[c] = 2;
1500 coup->coupling_point += get_bits1(gb) || (coup->coupling_point >> 1);
1502 sign = get_bits(gb, 1);
1503 scale = pow(2., pow(2., (int)get_bits(gb, 2) - 3));
1505 if ((ret = decode_ics(ac, sce, gb, 0, 0)))
1508 for (c = 0; c < num_gain; c++) {
1512 float gain_cache = 1.;
1514 cge = coup->coupling_point == AFTER_IMDCT ? 1 : get_bits1(gb);
1515 gain = cge ? get_vlc2(gb, vlc_scalefactors.table, 7, 3) - 60: 0;
1516 gain_cache = pow(scale, -gain);
1518 if (coup->coupling_point == AFTER_IMDCT) {
1519 coup->gain[c][0] = gain_cache;
1521 for (g = 0; g < sce->ics.num_window_groups; g++) {
1522 for (sfb = 0; sfb < sce->ics.max_sfb; sfb++, idx++) {
1523 if (sce->band_type[idx] != ZERO_BT) {
1525 int t = get_vlc2(gb, vlc_scalefactors.table, 7, 3) - 60;
1533 gain_cache = pow(scale, -t) * s;
1536 coup->gain[c][idx] = gain_cache;
1546 * Decode Spectral Band Replication extension data; reference: table 4.55.
1548 * @param crc flag indicating the presence of CRC checksum
1549 * @param cnt length of TYPE_FIL syntactic element in bytes
1551 * @return Returns number of bytes consumed from the TYPE_FIL element.
1553 static int decode_sbr_extension(AACContext *ac, GetBitContext *gb,
1556 // TODO : sbr_extension implementation
1557 av_log_missing_feature(ac->avccontext, "SBR", 0);
1558 skip_bits_long(gb, 8 * cnt - 4); // -4 due to reading extension type
1563 * Parse whether channels are to be excluded from Dynamic Range Compression; reference: table 4.53.
1565 * @return Returns number of bytes consumed.
1567 static int decode_drc_channel_exclusions(DynamicRangeControl *che_drc,
1571 int num_excl_chan = 0;
1574 for (i = 0; i < 7; i++)
1575 che_drc->exclude_mask[num_excl_chan++] = get_bits1(gb);
1576 } while (num_excl_chan < MAX_CHANNELS - 7 && get_bits1(gb));
1578 return num_excl_chan / 7;
1582 * Decode dynamic range information; reference: table 4.52.
1584 * @param cnt length of TYPE_FIL syntactic element in bytes
1586 * @return Returns number of bytes consumed.
1588 static int decode_dynamic_range(DynamicRangeControl *che_drc,
1589 GetBitContext *gb, int cnt)
1592 int drc_num_bands = 1;
1595 /* pce_tag_present? */
1596 if (get_bits1(gb)) {
1597 che_drc->pce_instance_tag = get_bits(gb, 4);
1598 skip_bits(gb, 4); // tag_reserved_bits
1602 /* excluded_chns_present? */
1603 if (get_bits1(gb)) {
1604 n += decode_drc_channel_exclusions(che_drc, gb);
1607 /* drc_bands_present? */
1608 if (get_bits1(gb)) {
1609 che_drc->band_incr = get_bits(gb, 4);
1610 che_drc->interpolation_scheme = get_bits(gb, 4);
1612 drc_num_bands += che_drc->band_incr;
1613 for (i = 0; i < drc_num_bands; i++) {
1614 che_drc->band_top[i] = get_bits(gb, 8);
1619 /* prog_ref_level_present? */
1620 if (get_bits1(gb)) {
1621 che_drc->prog_ref_level = get_bits(gb, 7);
1622 skip_bits1(gb); // prog_ref_level_reserved_bits
1626 for (i = 0; i < drc_num_bands; i++) {
1627 che_drc->dyn_rng_sgn[i] = get_bits1(gb);
1628 che_drc->dyn_rng_ctl[i] = get_bits(gb, 7);
1636 * Decode extension data (incomplete); reference: table 4.51.
1638 * @param cnt length of TYPE_FIL syntactic element in bytes
1640 * @return Returns number of bytes consumed
1642 static int decode_extension_payload(AACContext *ac, GetBitContext *gb, int cnt)
1646 switch (get_bits(gb, 4)) { // extension type
1647 case EXT_SBR_DATA_CRC:
1650 res = decode_sbr_extension(ac, gb, crc_flag, cnt);
1652 case EXT_DYNAMIC_RANGE:
1653 res = decode_dynamic_range(&ac->che_drc, gb, cnt);
1657 case EXT_DATA_ELEMENT:
1659 skip_bits_long(gb, 8 * cnt - 4);
1666 * Decode Temporal Noise Shaping filter coefficients and apply all-pole filters; reference: 4.6.9.3.
1668 * @param decode 1 if tool is used normally, 0 if tool is used in LTP.
1669 * @param coef spectral coefficients
1671 static void apply_tns(float coef[1024], TemporalNoiseShaping *tns,
1672 IndividualChannelStream *ics, int decode)
1674 const int mmm = FFMIN(ics->tns_max_bands, ics->max_sfb);
1676 int bottom, top, order, start, end, size, inc;
1677 float lpc[TNS_MAX_ORDER];
1679 for (w = 0; w < ics->num_windows; w++) {
1680 bottom = ics->num_swb;
1681 for (filt = 0; filt < tns->n_filt[w]; filt++) {
1683 bottom = FFMAX(0, top - tns->length[w][filt]);
1684 order = tns->order[w][filt];
1689 compute_lpc_coefs(tns->coef[w][filt], order, lpc, 0, 0, 0);
1691 start = ics->swb_offset[FFMIN(bottom, mmm)];
1692 end = ics->swb_offset[FFMIN( top, mmm)];
1693 if ((size = end - start) <= 0)
1695 if (tns->direction[w][filt]) {
1704 for (m = 0; m < size; m++, start += inc)
1705 for (i = 1; i <= FFMIN(m, order); i++)
1706 coef[start] -= coef[start - i * inc] * lpc[i - 1];
1712 * Conduct IMDCT and windowing.
1714 static void imdct_and_windowing(AACContext *ac, SingleChannelElement *sce)
1716 IndividualChannelStream *ics = &sce->ics;
1717 float *in = sce->coeffs;
1718 float *out = sce->ret;
1719 float *saved = sce->saved;
1720 const float *swindow = ics->use_kb_window[0] ? ff_aac_kbd_short_128 : ff_sine_128;
1721 const float *lwindow_prev = ics->use_kb_window[1] ? ff_aac_kbd_long_1024 : ff_sine_1024;
1722 const float *swindow_prev = ics->use_kb_window[1] ? ff_aac_kbd_short_128 : ff_sine_128;
1723 float *buf = ac->buf_mdct;
1724 float *temp = ac->temp;
1728 if (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) {
1729 if (ics->window_sequence[1] == ONLY_LONG_SEQUENCE || ics->window_sequence[1] == LONG_STOP_SEQUENCE)
1730 av_log(ac->avccontext, AV_LOG_WARNING,
1731 "Transition from an ONLY_LONG or LONG_STOP to an EIGHT_SHORT sequence detected. "
1732 "If you heard an audible artifact, please submit the sample to the FFmpeg developers.\n");
1733 for (i = 0; i < 1024; i += 128)
1734 ff_imdct_half(&ac->mdct_small, buf + i, in + i);
1736 ff_imdct_half(&ac->mdct, buf, in);
1738 /* window overlapping
1739 * NOTE: To simplify the overlapping code, all 'meaningless' short to long
1740 * and long to short transitions are considered to be short to short
1741 * transitions. This leaves just two cases (long to long and short to short)
1742 * with a little special sauce for EIGHT_SHORT_SEQUENCE.
1744 if ((ics->window_sequence[1] == ONLY_LONG_SEQUENCE || ics->window_sequence[1] == LONG_STOP_SEQUENCE) &&
1745 (ics->window_sequence[0] == ONLY_LONG_SEQUENCE || ics->window_sequence[0] == LONG_START_SEQUENCE)) {
1746 ac->dsp.vector_fmul_window( out, saved, buf, lwindow_prev, ac->add_bias, 512);
1748 for (i = 0; i < 448; i++)
1749 out[i] = saved[i] + ac->add_bias;
1751 if (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) {
1752 ac->dsp.vector_fmul_window(out + 448 + 0*128, saved + 448, buf + 0*128, swindow_prev, ac->add_bias, 64);
1753 ac->dsp.vector_fmul_window(out + 448 + 1*128, buf + 0*128 + 64, buf + 1*128, swindow, ac->add_bias, 64);
1754 ac->dsp.vector_fmul_window(out + 448 + 2*128, buf + 1*128 + 64, buf + 2*128, swindow, ac->add_bias, 64);
1755 ac->dsp.vector_fmul_window(out + 448 + 3*128, buf + 2*128 + 64, buf + 3*128, swindow, ac->add_bias, 64);
1756 ac->dsp.vector_fmul_window(temp, buf + 3*128 + 64, buf + 4*128, swindow, ac->add_bias, 64);
1757 memcpy( out + 448 + 4*128, temp, 64 * sizeof(float));
1759 ac->dsp.vector_fmul_window(out + 448, saved + 448, buf, swindow_prev, ac->add_bias, 64);
1760 for (i = 576; i < 1024; i++)
1761 out[i] = buf[i-512] + ac->add_bias;
1766 if (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) {
1767 for (i = 0; i < 64; i++)
1768 saved[i] = temp[64 + i] - ac->add_bias;
1769 ac->dsp.vector_fmul_window(saved + 64, buf + 4*128 + 64, buf + 5*128, swindow, 0, 64);
1770 ac->dsp.vector_fmul_window(saved + 192, buf + 5*128 + 64, buf + 6*128, swindow, 0, 64);
1771 ac->dsp.vector_fmul_window(saved + 320, buf + 6*128 + 64, buf + 7*128, swindow, 0, 64);
1772 memcpy( saved + 448, buf + 7*128 + 64, 64 * sizeof(float));
1773 } else if (ics->window_sequence[0] == LONG_START_SEQUENCE) {
1774 memcpy( saved, buf + 512, 448 * sizeof(float));
1775 memcpy( saved + 448, buf + 7*128 + 64, 64 * sizeof(float));
1776 } else { // LONG_STOP or ONLY_LONG
1777 memcpy( saved, buf + 512, 512 * sizeof(float));
1782 * Apply dependent channel coupling (applied before IMDCT).
1784 * @param index index into coupling gain array
1786 static void apply_dependent_coupling(AACContext *ac,
1787 SingleChannelElement *target,
1788 ChannelElement *cce, int index)
1790 IndividualChannelStream *ics = &cce->ch[0].ics;
1791 const uint16_t *offsets = ics->swb_offset;
1792 float *dest = target->coeffs;
1793 const float *src = cce->ch[0].coeffs;
1794 int g, i, group, k, idx = 0;
1795 if (ac->m4ac.object_type == AOT_AAC_LTP) {
1796 av_log(ac->avccontext, AV_LOG_ERROR,
1797 "Dependent coupling is not supported together with LTP\n");
1800 for (g = 0; g < ics->num_window_groups; g++) {
1801 for (i = 0; i < ics->max_sfb; i++, idx++) {
1802 if (cce->ch[0].band_type[idx] != ZERO_BT) {
1803 const float gain = cce->coup.gain[index][idx];
1804 for (group = 0; group < ics->group_len[g]; group++) {
1805 for (k = offsets[i]; k < offsets[i + 1]; k++) {
1807 dest[group * 128 + k] += gain * src[group * 128 + k];
1812 dest += ics->group_len[g] * 128;
1813 src += ics->group_len[g] * 128;
1818 * Apply independent channel coupling (applied after IMDCT).
1820 * @param index index into coupling gain array
1822 static void apply_independent_coupling(AACContext *ac,
1823 SingleChannelElement *target,
1824 ChannelElement *cce, int index)
1827 const float gain = cce->coup.gain[index][0];
1828 const float bias = ac->add_bias;
1829 const float *src = cce->ch[0].ret;
1830 float *dest = target->ret;
1832 for (i = 0; i < 1024; i++)
1833 dest[i] += gain * (src[i] - bias);
1837 * channel coupling transformation interface
1839 * @param index index into coupling gain array
1840 * @param apply_coupling_method pointer to (in)dependent coupling function
1842 static void apply_channel_coupling(AACContext *ac, ChannelElement *cc,
1843 enum RawDataBlockType type, int elem_id,
1844 enum CouplingPoint coupling_point,
1845 void (*apply_coupling_method)(AACContext *ac, SingleChannelElement *target, ChannelElement *cce, int index))
1849 for (i = 0; i < MAX_ELEM_ID; i++) {
1850 ChannelElement *cce = ac->che[TYPE_CCE][i];
1853 if (cce && cce->coup.coupling_point == coupling_point) {
1854 ChannelCoupling *coup = &cce->coup;
1856 for (c = 0; c <= coup->num_coupled; c++) {
1857 if (coup->type[c] == type && coup->id_select[c] == elem_id) {
1858 if (coup->ch_select[c] != 1) {
1859 apply_coupling_method(ac, &cc->ch[0], cce, index);
1860 if (coup->ch_select[c] != 0)
1863 if (coup->ch_select[c] != 2)
1864 apply_coupling_method(ac, &cc->ch[1], cce, index++);
1866 index += 1 + (coup->ch_select[c] == 3);
1873 * Convert spectral data to float samples, applying all supported tools as appropriate.
1875 static void spectral_to_sample(AACContext *ac)
1878 for (type = 3; type >= 0; type--) {
1879 for (i = 0; i < MAX_ELEM_ID; i++) {
1880 ChannelElement *che = ac->che[type][i];
1882 if (type <= TYPE_CPE)
1883 apply_channel_coupling(ac, che, type, i, BEFORE_TNS, apply_dependent_coupling);
1884 if (che->ch[0].tns.present)
1885 apply_tns(che->ch[0].coeffs, &che->ch[0].tns, &che->ch[0].ics, 1);
1886 if (che->ch[1].tns.present)
1887 apply_tns(che->ch[1].coeffs, &che->ch[1].tns, &che->ch[1].ics, 1);
1888 if (type <= TYPE_CPE)
1889 apply_channel_coupling(ac, che, type, i, BETWEEN_TNS_AND_IMDCT, apply_dependent_coupling);
1890 if (type != TYPE_CCE || che->coup.coupling_point == AFTER_IMDCT)
1891 imdct_and_windowing(ac, &che->ch[0]);
1892 if (type == TYPE_CPE)
1893 imdct_and_windowing(ac, &che->ch[1]);
1894 if (type <= TYPE_CCE)
1895 apply_channel_coupling(ac, che, type, i, AFTER_IMDCT, apply_independent_coupling);
1901 static int parse_adts_frame_header(AACContext *ac, GetBitContext *gb)
1904 AACADTSHeaderInfo hdr_info;
1906 size = ff_aac_parse_header(gb, &hdr_info);
1908 if (ac->output_configured != OC_LOCKED && hdr_info.chan_config) {
1909 enum ChannelPosition new_che_pos[4][MAX_ELEM_ID];
1910 memset(new_che_pos, 0, 4 * MAX_ELEM_ID * sizeof(new_che_pos[0][0]));
1911 ac->m4ac.chan_config = hdr_info.chan_config;
1912 if (set_default_channel_config(ac, new_che_pos, hdr_info.chan_config))
1914 if (output_configure(ac, ac->che_pos, new_che_pos, hdr_info.chan_config, OC_TRIAL_FRAME))
1916 } else if (ac->output_configured != OC_LOCKED) {
1917 ac->output_configured = OC_NONE;
1919 if (ac->output_configured != OC_LOCKED)
1921 ac->m4ac.sample_rate = hdr_info.sample_rate;
1922 ac->m4ac.sampling_index = hdr_info.sampling_index;
1923 ac->m4ac.object_type = hdr_info.object_type;
1924 if (!ac->avccontext->sample_rate)
1925 ac->avccontext->sample_rate = hdr_info.sample_rate;
1926 if (hdr_info.num_aac_frames == 1) {
1927 if (!hdr_info.crc_absent)
1930 av_log_missing_feature(ac->avccontext, "More than one AAC RDB per ADTS frame is", 0);
1937 static int aac_decode_frame(AVCodecContext *avccontext, void *data,
1938 int *data_size, AVPacket *avpkt)
1940 const uint8_t *buf = avpkt->data;
1941 int buf_size = avpkt->size;
1942 AACContext *ac = avccontext->priv_data;
1943 ChannelElement *che = NULL;
1945 enum RawDataBlockType elem_type;
1946 int err, elem_id, data_size_tmp;
1948 init_get_bits(&gb, buf, buf_size * 8);
1950 if (show_bits(&gb, 12) == 0xfff) {
1951 if (parse_adts_frame_header(ac, &gb) < 0) {
1952 av_log(avccontext, AV_LOG_ERROR, "Error decoding AAC frame header.\n");
1955 if (ac->m4ac.sampling_index > 12) {
1956 av_log(ac->avccontext, AV_LOG_ERROR, "invalid sampling rate index %d\n", ac->m4ac.sampling_index);
1962 while ((elem_type = get_bits(&gb, 3)) != TYPE_END) {
1963 elem_id = get_bits(&gb, 4);
1965 if (elem_type < TYPE_DSE && !(che=get_che(ac, elem_type, elem_id))) {
1966 av_log(ac->avccontext, AV_LOG_ERROR, "channel element %d.%d is not allocated\n", elem_type, elem_id);
1970 switch (elem_type) {
1973 err = decode_ics(ac, &che->ch[0], &gb, 0, 0);
1977 err = decode_cpe(ac, &gb, che);
1981 err = decode_cce(ac, &gb, che);
1985 err = decode_ics(ac, &che->ch[0], &gb, 0, 0);
1989 err = skip_data_stream_element(ac, &gb);
1993 enum ChannelPosition new_che_pos[4][MAX_ELEM_ID];
1994 memset(new_che_pos, 0, 4 * MAX_ELEM_ID * sizeof(new_che_pos[0][0]));
1995 if ((err = decode_pce(ac, new_che_pos, &gb)))
1997 if (ac->output_configured > OC_TRIAL_PCE)
1998 av_log(avccontext, AV_LOG_ERROR,
1999 "Not evaluating a further program_config_element as this construct is dubious at best.\n");
2001 err = output_configure(ac, ac->che_pos, new_che_pos, 0, OC_TRIAL_PCE);
2007 elem_id += get_bits(&gb, 8) - 1;
2008 if (get_bits_left(&gb) < 8 * elem_id) {
2009 av_log(avccontext, AV_LOG_ERROR, overread_err);
2013 elem_id -= decode_extension_payload(ac, &gb, elem_id);
2014 err = 0; /* FIXME */
2018 err = -1; /* should not happen, but keeps compiler happy */
2025 if (get_bits_left(&gb) < 3) {
2026 av_log(avccontext, AV_LOG_ERROR, overread_err);
2031 spectral_to_sample(ac);
2033 data_size_tmp = 1024 * avccontext->channels * sizeof(int16_t);
2034 if (*data_size < data_size_tmp) {
2035 av_log(avccontext, AV_LOG_ERROR,
2036 "Output buffer too small (%d) or trying to output too many samples (%d) for this frame.\n",
2037 *data_size, data_size_tmp);
2040 *data_size = data_size_tmp;
2042 ac->dsp.float_to_int16_interleave(data, (const float **)ac->output_data, 1024, avccontext->channels);
2044 if (ac->output_configured)
2045 ac->output_configured = OC_LOCKED;
2050 static av_cold int aac_decode_close(AVCodecContext *avccontext)
2052 AACContext *ac = avccontext->priv_data;
2055 for (i = 0; i < MAX_ELEM_ID; i++) {
2056 for (type = 0; type < 4; type++)
2057 av_freep(&ac->che[type][i]);
2060 ff_mdct_end(&ac->mdct);
2061 ff_mdct_end(&ac->mdct_small);
2065 AVCodec aac_decoder = {
2074 .long_name = NULL_IF_CONFIG_SMALL("Advanced Audio Coding"),
2075 .sample_fmts = (const enum SampleFormat[]) {
2076 SAMPLE_FMT_S16,SAMPLE_FMT_NONE
2078 .channel_layouts = aac_channel_layout,