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
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 * N 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
80 #include "bitstream.h"
85 #include "aacdectab.h"
86 #include "mpeg4audio.h"
93 #ifndef CONFIG_HARDCODED_TABLES
94 static float ff_aac_ivquant_tab[IVQUANT_SIZE];
95 static float ff_aac_pow2sf_tab[316];
96 #endif /* CONFIG_HARDCODED_TABLES */
98 static VLC vlc_scalefactors;
99 static VLC vlc_spectral[11];
103 * Decode an array of 4 bit element IDs, optionally interleaved with a stereo/mono switching bit.
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
109 static void decode_channel_map(enum ChannelPosition *cpe_map,
110 enum ChannelPosition *sce_map, enum ChannelPosition type, GetBitContext * gb, int 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;
118 * Decode program configuration element; reference: table 4.2.
120 * @param new_che_pos New channel position configuration - we only do something if it differs from the current one.
122 * @return Returns error status. 0 - OK, !0 - error
124 static 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;
128 skip_bits(gb, 2); // object_type
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);
135 ac->m4ac.sample_rate = ff_mpeg4audio_sample_rates[ac->m4ac.sampling_index];
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);
144 skip_bits(gb, 4); // mono_mixdown_tag
146 skip_bits(gb, 4); // stereo_mixdown_tag
149 skip_bits(gb, 3); // mixdown_coeff_index and pseudo_surround
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 );
156 skip_bits_long(gb, 4 * num_assoc_data);
158 decode_channel_map(new_che_pos[TYPE_CCE], new_che_pos[TYPE_CCE], AAC_CHANNEL_CC, gb, num_cc );
162 /* comment field, first byte is length */
163 skip_bits_long(gb, 8 * get_bits(gb, 8));
168 * Set up channel positions based on a default channel configuration
169 * as specified in table 1.17.
171 * @param new_che_pos New channel position configuration - we only do something if it differs from the current one.
173 * @return Returns error status. 0 - OK, !0 - error
175 static int set_default_channel_config(AACContext *ac, enum ChannelPosition new_che_pos[4][MAX_ELEM_ID],
178 if(channel_config < 1 || channel_config > 7) {
179 av_log(ac->avccontext, AV_LOG_ERROR, "invalid default channel configuration (%d)\n",
184 /* default channel configurations:
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
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
215 if (get_bits1(gb)) // dependsOnCoreCoder
216 skip_bits(gb, 14); // coreCoderDelay
217 extension_flag = get_bits1(gb);
219 if(ac->m4ac.object_type == AOT_AAC_SCALABLE ||
220 ac->m4ac.object_type == AOT_ER_AAC_SCALABLE)
221 skip_bits(gb, 3); // layerNr
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)))
229 if((ret = set_default_channel_config(ac, new_che_pos, channel_config)))
232 if((ret = output_configure(ac, ac->che_pos, new_che_pos)))
235 if (extension_flag) {
236 switch (ac->m4ac.object_type) {
238 skip_bits(gb, 5); // numOfSubFrame
239 skip_bits(gb, 11); // layer_length
243 case AOT_ER_AAC_SCALABLE:
245 skip_bits(gb, 3); /* aacSectionDataResilienceFlag
246 * aacScalefactorDataResilienceFlag
247 * aacSpectralDataResilienceFlag
251 skip_bits1(gb); // extensionFlag3 (TBD in version 3)
257 * Decode audio specific configuration; reference: table 1.13.
259 * @param data pointer to AVCodecContext extradata
260 * @param data_size size of AVCCodecContext extradata
262 * @return Returns error status. 0 - OK, !0 - error
264 static int decode_audio_specific_config(AACContext * ac, void *data, int data_size) {
268 init_get_bits(&gb, data, data_size * 8);
270 if((i = ff_mpeg4audio_get_config(&ac->m4ac, data, data_size)) < 0)
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);
277 skip_bits_long(&gb, i);
279 switch (ac->m4ac.object_type) {
281 if (decode_ga_specific_config(ac, &gb, ac->m4ac.chan_config))
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);
292 static av_cold int aac_decode_init(AVCodecContext * avccontext) {
293 AACContext * ac = avccontext->priv_data;
296 ac->avccontext = avccontext;
298 if (avccontext->extradata_size <= 0 ||
299 decode_audio_specific_config(ac, avccontext->extradata, avccontext->extradata_size))
302 avccontext->sample_fmt = SAMPLE_FMT_S16;
303 avccontext->sample_rate = ac->m4ac.sample_rate;
304 avccontext->frame_size = 1024;
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);
318 dsputil_init(&ac->dsp, avccontext);
320 ac->random_state = 0x1f2e3d4c;
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.
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.);
332 ac->sf_scale = 1. / -1024.;
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;
339 for (i = 0; i < 316; i++)
340 ff_aac_pow2sf_tab[i] = pow(2, (i - 200)/4.);
341 #endif /* CONFIG_HARDCODED_TABLES */
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]),
348 ff_mdct_init(&ac->mdct, 11, 1);
349 ff_mdct_init(&ac->mdct_small, 8, 1);
354 * Skip data_stream_element; reference: table 4.10.
356 static void skip_data_stream_element(GetBitContext * gb) {
357 int byte_align = get_bits1(gb);
358 int count = get_bits(gb, 8);
360 count += get_bits(gb, 8);
363 skip_bits_long(gb, 8 * count);
367 * Decode Individual Channel Stream info; reference: table 4.6.
369 * @param common_window Channels have independent [0], or shared [1], Individual Channel Stream information.
371 static int decode_ics_info(AACContext * ac, IndividualChannelStream * ics, GetBitContext * gb, int common_window) {
373 av_log(ac->avccontext, AV_LOG_ERROR, "Reserved bit set.\n");
374 memset(ics, 0, sizeof(IndividualChannelStream));
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;
388 * inverse quantization
390 * @param a quantized value to be dequantized
391 * @return Returns dequantized value.
393 static 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];
397 return cbrtf(fabsf(a)) * a;
401 * Decode band types (section_data payload); reference: table 4.46.
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
406 * @return Returns error status. 0 - OK, !0 - error
408 static int decode_band_types(AACContext * ac, enum BandType band_type[120],
409 int band_type_run_end[120], GetBitContext * gb, IndividualChannelStream * ics) {
411 const int bits = (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) ? 3 : 5;
412 for (g = 0; g < ics->num_window_groups; g++) {
414 while (k < ics->max_sfb) {
415 uint8_t sect_len = k;
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");
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);
437 * Decode scalefactors; reference: table 4.47.
439 * @param global_gain first scalefactor value as scalefactors are differentially coded
440 * @param band_type array of the used band type
441 * @param band_type_run_end array of the last scalefactor band of a band type run
442 * @param sf array of scalefactors or intensity stereo positions
444 * @return Returns error status. 0 - OK, !0 - error
446 static int decode_scalefactors(AACContext * ac, float sf[120], GetBitContext * gb,
447 unsigned int global_gain, IndividualChannelStream * ics,
448 enum BandType band_type[120], int band_type_run_end[120]) {
449 const int sf_offset = ac->sf_offset + (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE ? 12 : 0);
451 int offset[3] = { global_gain, global_gain - 90, 100 };
453 static const char *sf_str[3] = { "Global gain", "Noise gain", "Intensity stereo position" };
454 ics->intensity_present = 0;
455 for (g = 0; g < ics->num_window_groups; g++) {
456 for (i = 0; i < ics->max_sfb;) {
457 int run_end = band_type_run_end[idx];
458 if (band_type[idx] == ZERO_BT) {
459 for(; i < run_end; i++, idx++)
461 }else if((band_type[idx] == INTENSITY_BT) || (band_type[idx] == INTENSITY_BT2)) {
462 ics->intensity_present = 1;
463 for(; i < run_end; i++, idx++) {
464 offset[2] += get_vlc2(gb, vlc_scalefactors.table, 7, 3) - 60;
465 if(offset[2] > 255U) {
466 av_log(ac->avccontext, AV_LOG_ERROR,
467 "%s (%d) out of range.\n", sf_str[2], offset[2]);
470 sf[idx] = ff_aac_pow2sf_tab[-offset[2] + 300];
472 }else if(band_type[idx] == NOISE_BT) {
473 for(; i < run_end; i++, idx++) {
475 offset[1] += get_bits(gb, 9) - 256;
477 offset[1] += get_vlc2(gb, vlc_scalefactors.table, 7, 3) - 60;
478 if(offset[1] > 255U) {
479 av_log(ac->avccontext, AV_LOG_ERROR,
480 "%s (%d) out of range.\n", sf_str[1], offset[1]);
483 sf[idx] = -ff_aac_pow2sf_tab[ offset[1] + sf_offset];
486 for(; i < run_end; i++, idx++) {
487 offset[0] += get_vlc2(gb, vlc_scalefactors.table, 7, 3) - 60;
488 if(offset[0] > 255U) {
489 av_log(ac->avccontext, AV_LOG_ERROR,
490 "%s (%d) out of range.\n", sf_str[0], offset[0]);
493 sf[idx] = -ff_aac_pow2sf_tab[ offset[0] + sf_offset];
502 * Decode pulse data; reference: table 4.7.
504 static void decode_pulses(Pulse * pulse, GetBitContext * gb) {
506 pulse->num_pulse = get_bits(gb, 2) + 1;
507 pulse->start = get_bits(gb, 6);
508 for (i = 0; i < pulse->num_pulse; i++) {
509 pulse->offset[i] = get_bits(gb, 5);
510 pulse->amp [i] = get_bits(gb, 4);
515 * Decode Mid/Side data; reference: table 4.54.
517 * @param ms_present Indicates mid/side stereo presence. [0] mask is all 0s;
518 * [1] mask is decoded from bitstream; [2] mask is all 1s;
519 * [3] reserved for scalable AAC
521 static void decode_mid_side_stereo(ChannelElement * cpe, GetBitContext * gb,
525 * Add pulses with particular amplitudes to the quantized spectral data; reference: 4.6.3.3.
527 * @param pulse pointer to pulse data struct
528 * @param icoef array of quantized spectral data
530 static void add_pulses(int icoef[1024], const Pulse * pulse, const IndividualChannelStream * ics) {
531 int i, off = ics->swb_offset[pulse->start];
532 for (i = 0; i < pulse->num_pulse; i++) {
534 off += pulse->offset[i];
535 ic = (icoef[off] - 1)>>31;
536 icoef[off] += (pulse->amp[i]^ic) - ic;
541 * Decode an individual_channel_stream payload; reference: table 4.44.
543 * @param common_window Channels have independent [0], or shared [1], Individual Channel Stream information.
544 * @param scale_flag scalable [1] or non-scalable [0] AAC (Unused until scalable AAC is implemented.)
546 * @return Returns error status. 0 - OK, !0 - error
548 static int decode_ics(AACContext * ac, SingleChannelElement * sce, GetBitContext * gb, int common_window, int scale_flag) {
551 TemporalNoiseShaping * tns = &sce->tns;
552 IndividualChannelStream * ics = &sce->ics;
553 float * out = sce->coeffs;
554 int global_gain, pulse_present = 0;
556 /* These two assignments are to silence some GCC warnings about the
557 * variables being used uninitialised when in fact they always are.
562 global_gain = get_bits(gb, 8);
564 if (!common_window && !scale_flag) {
565 if (decode_ics_info(ac, ics, gb, 0) < 0)
569 if (decode_band_types(ac, sce->band_type, sce->band_type_run_end, gb, ics) < 0)
571 if (decode_scalefactors(ac, sce->sf, gb, global_gain, ics, sce->band_type, sce->band_type_run_end) < 0)
576 if ((pulse_present = get_bits1(gb))) {
577 if (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) {
578 av_log(ac->avccontext, AV_LOG_ERROR, "Pulse tool not allowed in eight short sequence.\n");
581 decode_pulses(&pulse, gb);
583 if ((tns->present = get_bits1(gb)) && decode_tns(ac, tns, gb, ics))
586 av_log_missing_feature(ac->avccontext, "SSR", 1);
591 if (decode_spectrum(ac, icoeffs, gb, ics, sce->band_type) < 0)
594 add_pulses(icoeffs, &pulse, ics);
595 dequant(ac, out, icoeffs, sce->sf, ics, sce->band_type);
600 * Decode a channel_pair_element; reference: table 4.4.
602 * @param elem_id Identifies the instance of a syntax element.
604 * @return Returns error status. 0 - OK, !0 - error
606 static int decode_cpe(AACContext * ac, GetBitContext * gb, int elem_id) {
607 int i, ret, common_window, ms_present = 0;
608 ChannelElement * cpe;
610 cpe = ac->che[TYPE_CPE][elem_id];
611 common_window = get_bits1(gb);
613 if (decode_ics_info(ac, &cpe->ch[0].ics, gb, 1))
615 i = cpe->ch[1].ics.use_kb_window[0];
616 cpe->ch[1].ics = cpe->ch[0].ics;
617 cpe->ch[1].ics.use_kb_window[1] = i;
618 ms_present = get_bits(gb, 2);
619 if(ms_present == 3) {
620 av_log(ac->avccontext, AV_LOG_ERROR, "ms_present = 3 is reserved.\n");
622 } else if(ms_present)
623 decode_mid_side_stereo(cpe, gb, ms_present);
625 if ((ret = decode_ics(ac, &cpe->ch[0], gb, common_window, 0)))
627 if ((ret = decode_ics(ac, &cpe->ch[1], gb, common_window, 0)))
630 if (common_window && ms_present)
631 apply_mid_side_stereo(cpe);
633 if (cpe->ch[1].ics.intensity_present)
634 apply_intensity_stereo(cpe, ms_present);
639 * Decode Spectral Band Replication extension data; reference: table 4.55.
641 * @param crc flag indicating the presence of CRC checksum
642 * @param cnt length of TYPE_FIL syntactic element in bytes
644 * @return Returns number of bytes consumed from the TYPE_FIL element.
646 static int decode_sbr_extension(AACContext * ac, GetBitContext * gb, int crc, int cnt) {
647 // TODO : sbr_extension implementation
648 av_log_missing_feature(ac->avccontext, "SBR", 0);
649 skip_bits_long(gb, 8*cnt - 4); // -4 due to reading extension type
654 * Decode dynamic range information; reference: table 4.52.
656 * @param cnt length of TYPE_FIL syntactic element in bytes
658 * @return Returns number of bytes consumed.
660 static int decode_dynamic_range(DynamicRangeControl *che_drc, GetBitContext * gb, int cnt) {
662 int drc_num_bands = 1;
665 /* pce_tag_present? */
667 che_drc->pce_instance_tag = get_bits(gb, 4);
668 skip_bits(gb, 4); // tag_reserved_bits
672 /* excluded_chns_present? */
674 n += decode_drc_channel_exclusions(che_drc, gb);
677 /* drc_bands_present? */
679 che_drc->band_incr = get_bits(gb, 4);
680 che_drc->interpolation_scheme = get_bits(gb, 4);
682 drc_num_bands += che_drc->band_incr;
683 for (i = 0; i < drc_num_bands; i++) {
684 che_drc->band_top[i] = get_bits(gb, 8);
689 /* prog_ref_level_present? */
691 che_drc->prog_ref_level = get_bits(gb, 7);
692 skip_bits1(gb); // prog_ref_level_reserved_bits
696 for (i = 0; i < drc_num_bands; i++) {
697 che_drc->dyn_rng_sgn[i] = get_bits1(gb);
698 che_drc->dyn_rng_ctl[i] = get_bits(gb, 7);
706 * Decode extension data (incomplete); reference: table 4.51.
708 * @param cnt length of TYPE_FIL syntactic element in bytes
710 * @return Returns number of bytes consumed
712 static int decode_extension_payload(AACContext * ac, GetBitContext * gb, int cnt) {
715 switch (get_bits(gb, 4)) { // extension type
716 case EXT_SBR_DATA_CRC:
719 res = decode_sbr_extension(ac, gb, crc_flag, cnt);
721 case EXT_DYNAMIC_RANGE:
722 res = decode_dynamic_range(&ac->che_drc, gb, cnt);
726 case EXT_DATA_ELEMENT:
728 skip_bits_long(gb, 8*cnt - 4);
735 * Conduct IMDCT and windowing.
737 static void imdct_and_windowing(AACContext * ac, SingleChannelElement * sce) {
738 IndividualChannelStream * ics = &sce->ics;
739 float * in = sce->coeffs;
740 float * out = sce->ret;
741 float * saved = sce->saved;
742 const float * lwindow = ics->use_kb_window[0] ? ff_aac_kbd_long_1024 : ff_aac_sine_long_1024;
743 const float * swindow = ics->use_kb_window[0] ? ff_aac_kbd_short_128 : ff_aac_sine_short_128;
744 const float * lwindow_prev = ics->use_kb_window[1] ? ff_aac_kbd_long_1024 : ff_aac_sine_long_1024;
745 const float * swindow_prev = ics->use_kb_window[1] ? ff_aac_kbd_short_128 : ff_aac_sine_short_128;
746 float * buf = ac->buf_mdct;
750 * Apply dependent channel coupling (applied before IMDCT).
752 * @param index index into coupling gain array
754 static void apply_dependent_coupling(AACContext * ac, SingleChannelElement * sce, ChannelElement * cc, int index) {
755 IndividualChannelStream * ics = &cc->ch[0].ics;
756 const uint16_t * offsets = ics->swb_offset;
757 float * dest = sce->coeffs;
758 const float * src = cc->ch[0].coeffs;
759 int g, i, group, k, idx = 0;
760 if(ac->m4ac.object_type == AOT_AAC_LTP) {
761 av_log(ac->avccontext, AV_LOG_ERROR,
762 "Dependent coupling is not supported together with LTP\n");
765 for (g = 0; g < ics->num_window_groups; g++) {
766 for (i = 0; i < ics->max_sfb; i++, idx++) {
767 if (cc->ch[0].band_type[idx] != ZERO_BT) {
768 for (group = 0; group < ics->group_len[g]; group++) {
769 for (k = offsets[i]; k < offsets[i+1]; k++) {
771 dest[group*128+k] += cc->coup.gain[index][idx] * src[group*128+k];
776 dest += ics->group_len[g]*128;
777 src += ics->group_len[g]*128;
782 * Apply independent channel coupling (applied after IMDCT).
784 * @param index index into coupling gain array
786 static void apply_independent_coupling(AACContext * ac, SingleChannelElement * sce, ChannelElement * cc, int index) {
788 for (i = 0; i < 1024; i++)
789 sce->ret[i] += cc->coup.gain[index][0] * (cc->ch[0].ret[i] - ac->add_bias);
798 data_size_tmp = 1024 * avccontext->channels * sizeof(int16_t);
799 if(*data_size < data_size_tmp) {
800 av_log(avccontext, AV_LOG_ERROR,
801 "Output buffer too small (%d) or trying to output too many samples (%d) for this frame.\n",
802 *data_size, data_size_tmp);
805 *data_size = data_size_tmp;
807 ac->dsp.float_to_int16_interleave(data, (const float **)ac->output_data, 1024, avccontext->channels);
812 static av_cold int aac_decode_close(AVCodecContext * avccontext) {
813 AACContext * ac = avccontext->priv_data;
816 for (i = 0; i < MAX_ELEM_ID; i++) {
817 for(type = 0; type < 4; type++)
818 av_freep(&ac->che[type][i]);
821 ff_mdct_end(&ac->mdct);
822 ff_mdct_end(&ac->mdct_small);
826 AVCodec aac_decoder = {
835 .long_name = NULL_IF_CONFIG_SMALL("Advanced Audio Coding"),
836 .sample_fmts = (enum SampleFormat[]){SAMPLE_FMT_S16,SAMPLE_FMT_NONE},