path: root/libavcodec/aacdec_template.c (plain)
blob: 98a32405979707b012c6b74bea2e5da58ff1d777

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	* Copyright (c) 2008-2013 Alex Converse <alex.converse@gmail.com>
6	*
7	* AAC LATM decoder
8	* Copyright (c) 2008-2010 Paul Kendall <paul@kcbbs.gen.nz>
9	* Copyright (c) 2010 Janne Grunau <janne-libav@jannau.net>
10	*
11	* AAC decoder fixed-point implementation
12	* Copyright (c) 2013
13	* MIPS Technologies, Inc., California.
14	*
15	* This file is part of FFmpeg.
16	*
17	* FFmpeg is free software; you can redistribute it and/or
18	* modify it under the terms of the GNU Lesser General Public
19	* License as published by the Free Software Foundation; either
20	* version 2.1 of the License, or (at your option) any later version.
21	*
22	* FFmpeg is distributed in the hope that it will be useful,
23	* but WITHOUT ANY WARRANTY; without even the implied warranty of
24	* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
25	* Lesser General Public License for more details.
26	*
27	* You should have received a copy of the GNU Lesser General Public
28	* License along with FFmpeg; if not, write to the Free Software
29	* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
30	*/
31
32	/**
33	* @file
34	* AAC decoder
35	* @author Oded Shimon ( ods15 ods15 dyndns org )
36	* @author Maxim Gavrilov ( maxim.gavrilov gmail com )
37	*
38	* AAC decoder fixed-point implementation
39	* @author Stanislav Ocovaj ( stanislav.ocovaj imgtec com )
40	* @author Nedeljko Babic ( nedeljko.babic imgtec com )
41	*/
42
43	/*
44	* supported tools
45	*
46	* Support? Name
47	* N (code in SoC repo) gain control
48	* Y block switching
49	* Y window shapes - standard
50	* N window shapes - Low Delay
51	* Y filterbank - standard
52	* N (code in SoC repo) filterbank - Scalable Sample Rate
53	* Y Temporal Noise Shaping
54	* Y Long Term Prediction
55	* Y intensity stereo
56	* Y channel coupling
57	* Y frequency domain prediction
58	* Y Perceptual Noise Substitution
59	* Y Mid/Side stereo
60	* N Scalable Inverse AAC Quantization
61	* N Frequency Selective Switch
62	* N upsampling filter
63	* Y quantization & coding - AAC
64	* N quantization & coding - TwinVQ
65	* N quantization & coding - BSAC
66	* N AAC Error Resilience tools
67	* N Error Resilience payload syntax
68	* N Error Protection tool
69	* N CELP
70	* N Silence Compression
71	* N HVXC
72	* N HVXC 4kbits/s VR
73	* N Structured Audio tools
74	* N Structured Audio Sample Bank Format
75	* N MIDI
76	* N Harmonic and Individual Lines plus Noise
77	* N Text-To-Speech Interface
78	* Y Spectral Band Replication
79	* Y (not in this code) Layer-1
80	* Y (not in this code) Layer-2
81	* Y (not in this code) Layer-3
82	* N SinuSoidal Coding (Transient, Sinusoid, Noise)
83	* Y Parametric Stereo
84	* N Direct Stream Transfer
85	* Y (not in fixed point code) Enhanced AAC Low Delay (ER AAC ELD)
86	*
87	* Note: - HE AAC v1 comprises LC AAC with Spectral Band Replication.
88	* - HE AAC v2 comprises LC AAC with Spectral Band Replication and
89	Parametric Stereo.
90	*/
91
92	#include "libavutil/thread.h"
93
94	static VLC vlc_scalefactors;
95	static VLC vlc_spectral[11];
96
97	static int output_configure(AACContext *ac,
98	uint8_t layout_map[MAX_ELEM_ID*4][3], int tags,
99	enum OCStatus oc_type, int get_new_frame);
100
101	#define overread_err "Input buffer exhausted before END element found\n"
102
103	static int count_channels(uint8_t (*layout)[3], int tags)
104	{
105	int i, sum = 0;
106	for (i = 0; i < tags; i++) {
107	int syn_ele = layout[i][0];
108	int pos = layout[i][2];
109	sum += (1 + (syn_ele == TYPE_CPE)) *
110	(pos != AAC_CHANNEL_OFF && pos != AAC_CHANNEL_CC);
111	}
112	return sum;
113	}
114
115	/**
116	* Check for the channel element in the current channel position configuration.
117	* If it exists, make sure the appropriate element is allocated and map the
118	* channel order to match the internal FFmpeg channel layout.
119	*
120	* @param che_pos current channel position configuration
121	* @param type channel element type
122	* @param id channel element id
123	* @param channels count of the number of channels in the configuration
124	*
125	* @return Returns error status. 0 - OK, !0 - error
126	*/
127	static av_cold int che_configure(AACContext *ac,
128	enum ChannelPosition che_pos,
129	int type, int id, int *channels)
130	{
131	if (*channels >= MAX_CHANNELS)
132	return AVERROR_INVALIDDATA;
133	if (che_pos) {
134	if (!ac->che[type][id]) {
135	if (!(ac->che[type][id] = av_mallocz(sizeof(ChannelElement))))
136	return AVERROR(ENOMEM);
137	AAC_RENAME(ff_aac_sbr_ctx_init)(ac, &ac->che[type][id]->sbr, type);
138	}
139	if (type != TYPE_CCE) {
140	if (*channels >= MAX_CHANNELS - (type == TYPE_CPE || (type == TYPE_SCE && ac->oc[1].m4ac.ps == 1))) {
141	av_log(ac->avctx, AV_LOG_ERROR, "Too many channels\n");
142	return AVERROR_INVALIDDATA;
143	}
144	ac->output_element[(*channels)++] = &ac->che[type][id]->ch[0];
145	if (type == TYPE_CPE ||
146	(type == TYPE_SCE && ac->oc[1].m4ac.ps == 1)) {
147	ac->output_element[(*channels)++] = &ac->che[type][id]->ch[1];
148	}
149	}
150	} else {
151	if (ac->che[type][id])
152	AAC_RENAME(ff_aac_sbr_ctx_close)(&ac->che[type][id]->sbr);
153	av_freep(&ac->che[type][id]);
154	}
155	return 0;
156	}
157
158	static int frame_configure_elements(AVCodecContext *avctx)
159	{
160	AACContext *ac = avctx->priv_data;
161	int type, id, ch, ret;
162
163	/* set channel pointers to internal buffers by default */
164	for (type = 0; type < 4; type++) {
165	for (id = 0; id < MAX_ELEM_ID; id++) {
166	ChannelElement *che = ac->che[type][id];
167	if (che) {
168	che->ch[0].ret = che->ch[0].ret_buf;
169	che->ch[1].ret = che->ch[1].ret_buf;
170	}
171	}
172	}
173
174	/* get output buffer */
175	av_frame_unref(ac->frame);
176	if (!avctx->channels)
177	return 1;
178
179	ac->frame->nb_samples = 2048;
180	if ((ret = ff_get_buffer(avctx, ac->frame, 0)) < 0)
181	return ret;
182
183	/* map output channel pointers to AVFrame data */
184	for (ch = 0; ch < avctx->channels; ch++) {
185	if (ac->output_element[ch])
186	ac->output_element[ch]->ret = (INTFLOAT *)ac->frame->extended_data[ch];
187	}
188
189	return 0;
190	}
191
192	struct elem_to_channel {
193	uint64_t av_position;
194	uint8_t syn_ele;
195	uint8_t elem_id;
196	uint8_t aac_position;
197	};
198
199	static int assign_pair(struct elem_to_channel e2c_vec[MAX_ELEM_ID],
200	uint8_t (*layout_map)[3], int offset, uint64_t left,
201	uint64_t right, int pos)
202	{
203	if (layout_map[offset][0] == TYPE_CPE) {
204	e2c_vec[offset] = (struct elem_to_channel) {
205	.av_position = left | right,
206	.syn_ele = TYPE_CPE,
207	.elem_id = layout_map[offset][1],
208	.aac_position = pos
209	};
210	return 1;
211	} else {
212	e2c_vec[offset] = (struct elem_to_channel) {
213	.av_position = left,
214	.syn_ele = TYPE_SCE,
215	.elem_id = layout_map[offset][1],
216	.aac_position = pos
217	};
218	e2c_vec[offset + 1] = (struct elem_to_channel) {
219	.av_position = right,
220	.syn_ele = TYPE_SCE,
221	.elem_id = layout_map[offset + 1][1],
222	.aac_position = pos
223	};
224	return 2;
225	}
226	}
227
228	static int count_paired_channels(uint8_t (*layout_map)[3], int tags, int pos,
229	int *current)
230	{
231	int num_pos_channels = 0;
232	int first_cpe = 0;
233	int sce_parity = 0;
234	int i;
235	for (i = *current; i < tags; i++) {
236	if (layout_map[i][2] != pos)
237	break;
238	if (layout_map[i][0] == TYPE_CPE) {
239	if (sce_parity) {
240	if (pos == AAC_CHANNEL_FRONT && !first_cpe) {
241	sce_parity = 0;
242	} else {
243	return -1;
244	}
245	}
246	num_pos_channels += 2;
247	first_cpe = 1;
248	} else {
249	num_pos_channels++;
250	sce_parity ^= 1;
251	}
252	}
253	if (sce_parity &&
254	((pos == AAC_CHANNEL_FRONT && first_cpe) || pos == AAC_CHANNEL_SIDE))
255	return -1;
256	*current = i;
257	return num_pos_channels;
258	}
259
260	static uint64_t sniff_channel_order(uint8_t (*layout_map)[3], int tags)
261	{
262	int i, n, total_non_cc_elements;
263	struct elem_to_channel e2c_vec[4 * MAX_ELEM_ID] = { { 0 } };
264	int num_front_channels, num_side_channels, num_back_channels;
265	uint64_t layout;
266
267	if (FF_ARRAY_ELEMS(e2c_vec) < tags)
268	return 0;
269
270	i = 0;
271	num_front_channels =
272	count_paired_channels(layout_map, tags, AAC_CHANNEL_FRONT, &i);
273	if (num_front_channels < 0)
274	return 0;
275	num_side_channels =
276	count_paired_channels(layout_map, tags, AAC_CHANNEL_SIDE, &i);
277	if (num_side_channels < 0)
278	return 0;
279	num_back_channels =
280	count_paired_channels(layout_map, tags, AAC_CHANNEL_BACK, &i);
281	if (num_back_channels < 0)
282	return 0;
283
284	if (num_side_channels == 0 && num_back_channels >= 4) {
285	num_side_channels = 2;
286	num_back_channels -= 2;
287	}
288
289	i = 0;
290	if (num_front_channels & 1) {
291	e2c_vec[i] = (struct elem_to_channel) {
292	.av_position = AV_CH_FRONT_CENTER,
293	.syn_ele = TYPE_SCE,
294	.elem_id = layout_map[i][1],
295	.aac_position = AAC_CHANNEL_FRONT
296	};
297	i++;
298	num_front_channels--;
299	}
300	if (num_front_channels >= 4) {
301	i += assign_pair(e2c_vec, layout_map, i,
302	AV_CH_FRONT_LEFT_OF_CENTER,
303	AV_CH_FRONT_RIGHT_OF_CENTER,
304	AAC_CHANNEL_FRONT);
305	num_front_channels -= 2;
306	}
307	if (num_front_channels >= 2) {
308	i += assign_pair(e2c_vec, layout_map, i,
309	AV_CH_FRONT_LEFT,
310	AV_CH_FRONT_RIGHT,
311	AAC_CHANNEL_FRONT);
312	num_front_channels -= 2;
313	}
314	while (num_front_channels >= 2) {
315	i += assign_pair(e2c_vec, layout_map, i,
316	UINT64_MAX,
317	UINT64_MAX,
318	AAC_CHANNEL_FRONT);
319	num_front_channels -= 2;
320	}
321
322	if (num_side_channels >= 2) {
323	i += assign_pair(e2c_vec, layout_map, i,
324	AV_CH_SIDE_LEFT,
325	AV_CH_SIDE_RIGHT,
326	AAC_CHANNEL_FRONT);
327	num_side_channels -= 2;
328	}
329	while (num_side_channels >= 2) {
330	i += assign_pair(e2c_vec, layout_map, i,
331	UINT64_MAX,
332	UINT64_MAX,
333	AAC_CHANNEL_SIDE);
334	num_side_channels -= 2;
335	}
336
337	while (num_back_channels >= 4) {
338	i += assign_pair(e2c_vec, layout_map, i,
339	UINT64_MAX,
340	UINT64_MAX,
341	AAC_CHANNEL_BACK);
342	num_back_channels -= 2;
343	}
344	if (num_back_channels >= 2) {
345	i += assign_pair(e2c_vec, layout_map, i,
346	AV_CH_BACK_LEFT,
347	AV_CH_BACK_RIGHT,
348	AAC_CHANNEL_BACK);
349	num_back_channels -= 2;
350	}
351	if (num_back_channels) {
352	e2c_vec[i] = (struct elem_to_channel) {
353	.av_position = AV_CH_BACK_CENTER,
354	.syn_ele = TYPE_SCE,
355	.elem_id = layout_map[i][1],
356	.aac_position = AAC_CHANNEL_BACK
357	};
358	i++;
359	num_back_channels--;
360	}
361
362	if (i < tags && layout_map[i][2] == AAC_CHANNEL_LFE) {
363	e2c_vec[i] = (struct elem_to_channel) {
364	.av_position = AV_CH_LOW_FREQUENCY,
365	.syn_ele = TYPE_LFE,
366	.elem_id = layout_map[i][1],
367	.aac_position = AAC_CHANNEL_LFE
368	};
369	i++;
370	}
371	while (i < tags && layout_map[i][2] == AAC_CHANNEL_LFE) {
372	e2c_vec[i] = (struct elem_to_channel) {
373	.av_position = UINT64_MAX,
374	.syn_ele = TYPE_LFE,
375	.elem_id = layout_map[i][1],
376	.aac_position = AAC_CHANNEL_LFE
377	};
378	i++;
379	}
380
381	// Must choose a stable sort
382	total_non_cc_elements = n = i;
383	do {
384	int next_n = 0;
385	for (i = 1; i < n; i++)
386	if (e2c_vec[i - 1].av_position > e2c_vec[i].av_position) {
387	FFSWAP(struct elem_to_channel, e2c_vec[i - 1], e2c_vec[i]);
388	next_n = i;
389	}
390	n = next_n;
391	} while (n > 0);
392
393	layout = 0;
394	for (i = 0; i < total_non_cc_elements; i++) {
395	layout_map[i][0] = e2c_vec[i].syn_ele;
396	layout_map[i][1] = e2c_vec[i].elem_id;
397	layout_map[i][2] = e2c_vec[i].aac_position;
398	if (e2c_vec[i].av_position != UINT64_MAX) {
399	layout |= e2c_vec[i].av_position;
400	}
401	}
402
403	return layout;
404	}
405
406	/**
407	* Save current output configuration if and only if it has been locked.
408	*/
409	static void push_output_configuration(AACContext *ac) {
410	if (ac->oc[1].status == OC_LOCKED || ac->oc[0].status == OC_NONE) {
411	ac->oc[0] = ac->oc[1];
412	}
413	ac->oc[1].status = OC_NONE;
414	}
415
416	/**
417	* Restore the previous output configuration if and only if the current
418	* configuration is unlocked.
419	*/
420	static void pop_output_configuration(AACContext *ac) {
421	if (ac->oc[1].status != OC_LOCKED && ac->oc[0].status != OC_NONE) {
422	ac->oc[1] = ac->oc[0];
423	ac->avctx->channels = ac->oc[1].channels;
424	ac->avctx->channel_layout = ac->oc[1].channel_layout;
425	output_configure(ac, ac->oc[1].layout_map, ac->oc[1].layout_map_tags,
426	ac->oc[1].status, 0);
427	}
428	}
429
430	/**
431	* Configure output channel order based on the current program
432	* configuration element.
433	*
434	* @return Returns error status. 0 - OK, !0 - error
435	*/
436	static int output_configure(AACContext *ac,
437	uint8_t layout_map[MAX_ELEM_ID * 4][3], int tags,
438	enum OCStatus oc_type, int get_new_frame)
439	{
440	AVCodecContext *avctx = ac->avctx;
441	int i, channels = 0, ret;
442	uint64_t layout = 0;
443	uint8_t id_map[TYPE_END][MAX_ELEM_ID] = {{ 0 }};
444	uint8_t type_counts[TYPE_END] = { 0 };
445
446	if (ac->oc[1].layout_map != layout_map) {
447	memcpy(ac->oc[1].layout_map, layout_map, tags * sizeof(layout_map[0]));
448	ac->oc[1].layout_map_tags = tags;
449	}
450	for (i = 0; i < tags; i++) {
451	int type = layout_map[i][0];
452	int id = layout_map[i][1];
453	id_map[type][id] = type_counts[type]++;
454	if (id_map[type][id] >= MAX_ELEM_ID) {
455	avpriv_request_sample(ac->avctx, "Too large remapped id");
456	return AVERROR_PATCHWELCOME;
457	}
458	}
459	// Try to sniff a reasonable channel order, otherwise output the
460	// channels in the order the PCE declared them.
461	if (avctx->request_channel_layout != AV_CH_LAYOUT_NATIVE)
462	layout = sniff_channel_order(layout_map, tags);
463	for (i = 0; i < tags; i++) {
464	int type = layout_map[i][0];
465	int id = layout_map[i][1];
466	int iid = id_map[type][id];
467	int position = layout_map[i][2];
468	// Allocate or free elements depending on if they are in the
469	// current program configuration.
470	ret = che_configure(ac, position, type, iid, &channels);
471	if (ret < 0)
472	return ret;
473	ac->tag_che_map[type][id] = ac->che[type][iid];
474	}
475	if (ac->oc[1].m4ac.ps == 1 && channels == 2) {
476	if (layout == AV_CH_FRONT_CENTER) {
477	layout = AV_CH_FRONT_LEFT|AV_CH_FRONT_RIGHT;
478	} else {
479	layout = 0;
480	}
481	}
482
483	if (layout) avctx->channel_layout = layout;
484	ac->oc[1].channel_layout = layout;
485	avctx->channels = ac->oc[1].channels = channels;
486	ac->oc[1].status = oc_type;
487
488	if (get_new_frame) {
489	if ((ret = frame_configure_elements(ac->avctx)) < 0)
490	return ret;
491	}
492
493	return 0;
494	}
495
496	static void flush(AVCodecContext *avctx)
497	{
498	AACContext *ac= avctx->priv_data;
499	int type, i, j;
500
501	for (type = 3; type >= 0; type--) {
502	for (i = 0; i < MAX_ELEM_ID; i++) {
503	ChannelElement *che = ac->che[type][i];
504	if (che) {
505	for (j = 0; j <= 1; j++) {
506	memset(che->ch[j].saved, 0, sizeof(che->ch[j].saved));
507	}
508	}
509	}
510	}
511	}
512
513	/**
514	* Set up channel positions based on a default channel configuration
515	* as specified in table 1.17.
516	*
517	* @return Returns error status. 0 - OK, !0 - error
518	*/
519	static int set_default_channel_config(AVCodecContext *avctx,
520	uint8_t (*layout_map)[3],
521	int *tags,
522	int channel_config)
523	{
524	if (channel_config < 1 || (channel_config > 7 && channel_config < 11) ||
525	channel_config > 12) {
526	av_log(avctx, AV_LOG_ERROR,
527	"invalid default channel configuration (%d)\n",
528	channel_config);
529	return AVERROR_INVALIDDATA;
530	}
531	*tags = tags_per_config[channel_config];
532	memcpy(layout_map, aac_channel_layout_map[channel_config - 1],
533	*tags * sizeof(*layout_map));
534
535	/*
536	* AAC specification has 7.1(wide) as a default layout for 8-channel streams.
537	* However, at least Nero AAC encoder encodes 7.1 streams using the default
538	* channel config 7, mapping the side channels of the original audio stream
539	* to the second AAC_CHANNEL_FRONT pair in the AAC stream. Similarly, e.g. FAAD
540	* decodes the second AAC_CHANNEL_FRONT pair as side channels, therefore decoding
541	* the incorrect streams as if they were correct (and as the encoder intended).
542	*
543	* As actual intended 7.1(wide) streams are very rare, default to assuming a
544	* 7.1 layout was intended.
545	*/
546	if (channel_config == 7 && avctx->strict_std_compliance < FF_COMPLIANCE_STRICT) {
547	av_log(avctx, AV_LOG_INFO, "Assuming an incorrectly encoded 7.1 channel layout"
548	" instead of a spec-compliant 7.1(wide) layout, use -strict %d to decode"
549	" according to the specification instead.\n", FF_COMPLIANCE_STRICT);
550	layout_map[2][2] = AAC_CHANNEL_SIDE;
551	}
552
553	return 0;
554	}
555
556	static ChannelElement *get_che(AACContext *ac, int type, int elem_id)
557	{
558	/* For PCE based channel configurations map the channels solely based
559	* on tags. */
560	if (!ac->oc[1].m4ac.chan_config) {
561	return ac->tag_che_map[type][elem_id];
562	}
563	// Allow single CPE stereo files to be signalled with mono configuration.
564	if (!ac->tags_mapped && type == TYPE_CPE &&
565	ac->oc[1].m4ac.chan_config == 1) {
566	uint8_t layout_map[MAX_ELEM_ID*4][3];
567	int layout_map_tags;
568	push_output_configuration(ac);
569
570	av_log(ac->avctx, AV_LOG_DEBUG, "mono with CPE\n");
571
572	if (set_default_channel_config(ac->avctx, layout_map,
573	&layout_map_tags, 2) < 0)
574	return NULL;
575	if (output_configure(ac, layout_map, layout_map_tags,
576	OC_TRIAL_FRAME, 1) < 0)
577	return NULL;
578
579	ac->oc[1].m4ac.chan_config = 2;
580	ac->oc[1].m4ac.ps = 0;
581	}
582	// And vice-versa
583	if (!ac->tags_mapped && type == TYPE_SCE &&
584	ac->oc[1].m4ac.chan_config == 2) {
585	uint8_t layout_map[MAX_ELEM_ID * 4][3];
586	int layout_map_tags;
587	push_output_configuration(ac);
588
589	av_log(ac->avctx, AV_LOG_DEBUG, "stereo with SCE\n");
590
591	if (set_default_channel_config(ac->avctx, layout_map,
592	&layout_map_tags, 1) < 0)
593	return NULL;
594	if (output_configure(ac, layout_map, layout_map_tags,
595	OC_TRIAL_FRAME, 1) < 0)
596	return NULL;
597
598	ac->oc[1].m4ac.chan_config = 1;
599	if (ac->oc[1].m4ac.sbr)
600	ac->oc[1].m4ac.ps = -1;
601	}
602	/* For indexed channel configurations map the channels solely based
603	* on position. */
604	switch (ac->oc[1].m4ac.chan_config) {
605	case 12:
606	case 7:
607	if (ac->tags_mapped == 3 && type == TYPE_CPE) {
608	ac->tags_mapped++;
609	return ac->tag_che_map[TYPE_CPE][elem_id] = ac->che[TYPE_CPE][2];
610	}
611	case 11:
612	if (ac->tags_mapped == 2 &&
613	ac->oc[1].m4ac.chan_config == 11 &&
614	type == TYPE_SCE) {
615	ac->tags_mapped++;
616	return ac->tag_che_map[TYPE_SCE][elem_id] = ac->che[TYPE_SCE][1];
617	}
618	case 6:
619	/* Some streams incorrectly code 5.1 audio as
620	* SCE[0] CPE[0] CPE[1] SCE[1]
621	* instead of
622	* SCE[0] CPE[0] CPE[1] LFE[0].
623	* If we seem to have encountered such a stream, transfer
624	* the LFE[0] element to the SCE[1]'s mapping */
625	if (ac->tags_mapped == tags_per_config[ac->oc[1].m4ac.chan_config] - 1 && (type == TYPE_LFE || type == TYPE_SCE)) {
626	if (!ac->warned_remapping_once && (type != TYPE_LFE || elem_id != 0)) {
627	av_log(ac->avctx, AV_LOG_WARNING,
628	"This stream seems to incorrectly report its last channel as %s[%d], mapping to LFE[0]\n",
629	type == TYPE_SCE ? "SCE" : "LFE", elem_id);
630	ac->warned_remapping_once++;
631	}
632	ac->tags_mapped++;
633	return ac->tag_che_map[type][elem_id] = ac->che[TYPE_LFE][0];
634	}
635	case 5:
636	if (ac->tags_mapped == 2 && type == TYPE_CPE) {
637	ac->tags_mapped++;
638	return ac->tag_che_map[TYPE_CPE][elem_id] = ac->che[TYPE_CPE][1];
639	}
640	case 4:
641	/* Some streams incorrectly code 4.0 audio as
642	* SCE[0] CPE[0] LFE[0]
643	* instead of
644	* SCE[0] CPE[0] SCE[1].
645	* If we seem to have encountered such a stream, transfer
646	* the SCE[1] element to the LFE[0]'s mapping */
647	if (ac->tags_mapped == tags_per_config[ac->oc[1].m4ac.chan_config] - 1 && (type == TYPE_LFE || type == TYPE_SCE)) {
648	if (!ac->warned_remapping_once && (type != TYPE_SCE || elem_id != 1)) {
649	av_log(ac->avctx, AV_LOG_WARNING,
650	"This stream seems to incorrectly report its last channel as %s[%d], mapping to SCE[1]\n",
651	type == TYPE_SCE ? "SCE" : "LFE", elem_id);
652	ac->warned_remapping_once++;
653	}
654	ac->tags_mapped++;
655	return ac->tag_che_map[type][elem_id] = ac->che[TYPE_SCE][1];
656	}
657	if (ac->tags_mapped == 2 &&
658	ac->oc[1].m4ac.chan_config == 4 &&
659	type == TYPE_SCE) {
660	ac->tags_mapped++;
661	return ac->tag_che_map[TYPE_SCE][elem_id] = ac->che[TYPE_SCE][1];
662	}
663	case 3:
664	case 2:
665	if (ac->tags_mapped == (ac->oc[1].m4ac.chan_config != 2) &&
666	type == TYPE_CPE) {
667	ac->tags_mapped++;
668	return ac->tag_che_map[TYPE_CPE][elem_id] = ac->che[TYPE_CPE][0];
669	} else if (ac->oc[1].m4ac.chan_config == 2) {
670	return NULL;
671	}
672	case 1:
673	if (!ac->tags_mapped && type == TYPE_SCE) {
674	ac->tags_mapped++;
675	return ac->tag_che_map[TYPE_SCE][elem_id] = ac->che[TYPE_SCE][0];
676	}
677	default:
678	return NULL;
679	}
680	}
681
682	/**
683	* Decode an array of 4 bit element IDs, optionally interleaved with a
684	* stereo/mono switching bit.
685	*
686	* @param type speaker type/position for these channels
687	*/
688	static void decode_channel_map(uint8_t layout_map[][3],
689	enum ChannelPosition type,
690	GetBitContext *gb, int n)
691	{
692	while (n--) {
693	enum RawDataBlockType syn_ele;
694	switch (type) {
695	case AAC_CHANNEL_FRONT:
696	case AAC_CHANNEL_BACK:
697	case AAC_CHANNEL_SIDE:
698	syn_ele = get_bits1(gb);
699	break;
700	case AAC_CHANNEL_CC:
701	skip_bits1(gb);
702	syn_ele = TYPE_CCE;
703	break;
704	case AAC_CHANNEL_LFE:
705	syn_ele = TYPE_LFE;
706	break;
707	default:
708	// AAC_CHANNEL_OFF has no channel map
709	av_assert0(0);
710	}
711	layout_map[0][0] = syn_ele;
712	layout_map[0][1] = get_bits(gb, 4);
713	layout_map[0][2] = type;
714	layout_map++;
715	}
716	}
717
718	static inline void relative_align_get_bits(GetBitContext *gb,
719	int reference_position) {
720	int n = (reference_position - get_bits_count(gb) & 7);
721	if (n)
722	skip_bits(gb, n);
723	}
724
725	/**
726	* Decode program configuration element; reference: table 4.2.
727	*
728	* @return Returns error status. 0 - OK, !0 - error
729	*/
730	static int decode_pce(AVCodecContext *avctx, MPEG4AudioConfig *m4ac,
731	uint8_t (*layout_map)[3],
732	GetBitContext *gb, int byte_align_ref)
733	{
734	int num_front, num_side, num_back, num_lfe, num_assoc_data, num_cc;
735	int sampling_index;
736	int comment_len;
737	int tags;
738
739	skip_bits(gb, 2); // object_type
740
741	sampling_index = get_bits(gb, 4);
742	if (m4ac->sampling_index != sampling_index)
743	av_log(avctx, AV_LOG_WARNING,
744	"Sample rate index in program config element does not "
745	"match the sample rate index configured by the container.\n");
746
747	num_front = get_bits(gb, 4);
748	num_side = get_bits(gb, 4);
749	num_back = get_bits(gb, 4);
750	num_lfe = get_bits(gb, 2);
751	num_assoc_data = get_bits(gb, 3);
752	num_cc = get_bits(gb, 4);
753
754	if (get_bits1(gb))
755	skip_bits(gb, 4); // mono_mixdown_tag
756	if (get_bits1(gb))
757	skip_bits(gb, 4); // stereo_mixdown_tag
758
759	if (get_bits1(gb))
760	skip_bits(gb, 3); // mixdown_coeff_index and pseudo_surround
761
762	if (get_bits_left(gb) < 4 * (num_front + num_side + num_back + num_lfe + num_assoc_data + num_cc)) {
763	av_log(avctx, AV_LOG_ERROR, "decode_pce: " overread_err);
764	return -1;
765	}
766	decode_channel_map(layout_map , AAC_CHANNEL_FRONT, gb, num_front);
767	tags = num_front;
768	decode_channel_map(layout_map + tags, AAC_CHANNEL_SIDE, gb, num_side);
769	tags += num_side;
770	decode_channel_map(layout_map + tags, AAC_CHANNEL_BACK, gb, num_back);
771	tags += num_back;
772	decode_channel_map(layout_map + tags, AAC_CHANNEL_LFE, gb, num_lfe);
773	tags += num_lfe;
774
775	skip_bits_long(gb, 4 * num_assoc_data);
776
777	decode_channel_map(layout_map + tags, AAC_CHANNEL_CC, gb, num_cc);
778	tags += num_cc;
779
780	relative_align_get_bits(gb, byte_align_ref);
781
782	/* comment field, first byte is length */
783	comment_len = get_bits(gb, 8) * 8;
784	if (get_bits_left(gb) < comment_len) {
785	av_log(avctx, AV_LOG_ERROR, "decode_pce: " overread_err);
786	return AVERROR_INVALIDDATA;
787	}
788	skip_bits_long(gb, comment_len);
789	return tags;
790	}
791
792	/**
793	* Decode GA "General Audio" specific configuration; reference: table 4.1.
794	*
795	* @param ac pointer to AACContext, may be null
796	* @param avctx pointer to AVCCodecContext, used for logging
797	*
798	* @return Returns error status. 0 - OK, !0 - error
799	*/
800	static int decode_ga_specific_config(AACContext *ac, AVCodecContext *avctx,
801	GetBitContext *gb,
802	int get_bit_alignment,
803	MPEG4AudioConfig *m4ac,
804	int channel_config)
805	{
806	int extension_flag, ret, ep_config, res_flags;
807	uint8_t layout_map[MAX_ELEM_ID*4][3];
808	int tags = 0;
809
810	if (get_bits1(gb)) { // frameLengthFlag
811	avpriv_request_sample(avctx, "960/120 MDCT window");
812	return AVERROR_PATCHWELCOME;
813	}
814	m4ac->frame_length_short = 0;
815
816	if (get_bits1(gb)) // dependsOnCoreCoder
817	skip_bits(gb, 14); // coreCoderDelay
818	extension_flag = get_bits1(gb);
819
820	if (m4ac->object_type == AOT_AAC_SCALABLE ||
821	m4ac->object_type == AOT_ER_AAC_SCALABLE)
822	skip_bits(gb, 3); // layerNr
823
824	if (channel_config == 0) {
825	skip_bits(gb, 4); // element_instance_tag
826	tags = decode_pce(avctx, m4ac, layout_map, gb, get_bit_alignment);
827	if (tags < 0)
828	return tags;
829	} else {
830	if ((ret = set_default_channel_config(avctx, layout_map,
831	&tags, channel_config)))
832	return ret;
833	}
834
835	if (count_channels(layout_map, tags) > 1) {
836	m4ac->ps = 0;
837	} else if (m4ac->sbr == 1 && m4ac->ps == -1)
838	m4ac->ps = 1;
839
840	if (ac && (ret = output_configure(ac, layout_map, tags, OC_GLOBAL_HDR, 0)))
841	return ret;
842
843	if (extension_flag) {
844	switch (m4ac->object_type) {
845	case AOT_ER_BSAC:
846	skip_bits(gb, 5); // numOfSubFrame
847	skip_bits(gb, 11); // layer_length
848	break;
849	case AOT_ER_AAC_LC:
850	case AOT_ER_AAC_LTP:
851	case AOT_ER_AAC_SCALABLE:
852	case AOT_ER_AAC_LD:
853	res_flags = get_bits(gb, 3);
854	if (res_flags) {
855	avpriv_report_missing_feature(avctx,
856	"AAC data resilience (flags %x)",
857	res_flags);
858	return AVERROR_PATCHWELCOME;
859	}
860	break;
861	}
862	skip_bits1(gb); // extensionFlag3 (TBD in version 3)
863	}
864	switch (m4ac->object_type) {
865	case AOT_ER_AAC_LC:
866	case AOT_ER_AAC_LTP:
867	case AOT_ER_AAC_SCALABLE:
868	case AOT_ER_AAC_LD:
869	ep_config = get_bits(gb, 2);
870	if (ep_config) {
871	avpriv_report_missing_feature(avctx,
872	"epConfig %d", ep_config);
873	return AVERROR_PATCHWELCOME;
874	}
875	}
876	return 0;
877	}
878
879	static int decode_eld_specific_config(AACContext *ac, AVCodecContext *avctx,
880	GetBitContext *gb,
881	MPEG4AudioConfig *m4ac,
882	int channel_config)
883	{
884	int ret, ep_config, res_flags;
885	uint8_t layout_map[MAX_ELEM_ID*4][3];
886	int tags = 0;
887	const int ELDEXT_TERM = 0;
888
889	m4ac->ps = 0;
890	m4ac->sbr = 0;
891	#if USE_FIXED
892	if (get_bits1(gb)) { // frameLengthFlag
893	avpriv_request_sample(avctx, "960/120 MDCT window");
894	return AVERROR_PATCHWELCOME;
895	}
896	#else
897	m4ac->frame_length_short = get_bits1(gb);
898	#endif
899	res_flags = get_bits(gb, 3);
900	if (res_flags) {
901	avpriv_report_missing_feature(avctx,
902	"AAC data resilience (flags %x)",
903	res_flags);
904	return AVERROR_PATCHWELCOME;
905	}
906
907	if (get_bits1(gb)) { // ldSbrPresentFlag
908	avpriv_report_missing_feature(avctx,
909	"Low Delay SBR");
910	return AVERROR_PATCHWELCOME;
911	}
912
913	while (get_bits(gb, 4) != ELDEXT_TERM) {
914	int len = get_bits(gb, 4);
915	if (len == 15)
916	len += get_bits(gb, 8);
917	if (len == 15 + 255)
918	len += get_bits(gb, 16);
919	if (get_bits_left(gb) < len * 8 + 4) {
920	av_log(avctx, AV_LOG_ERROR, overread_err);
921	return AVERROR_INVALIDDATA;
922	}
923	skip_bits_long(gb, 8 * len);
924	}
925
926	if ((ret = set_default_channel_config(avctx, layout_map,
927	&tags, channel_config)))
928	return ret;
929
930	if (ac && (ret = output_configure(ac, layout_map, tags, OC_GLOBAL_HDR, 0)))
931	return ret;
932
933	ep_config = get_bits(gb, 2);
934	if (ep_config) {
935	avpriv_report_missing_feature(avctx,
936	"epConfig %d", ep_config);
937	return AVERROR_PATCHWELCOME;
938	}
939	return 0;
940	}
941
942	/**
943	* Decode audio specific configuration; reference: table 1.13.
944	*
945	* @param ac pointer to AACContext, may be null
946	* @param avctx pointer to AVCCodecContext, used for logging
947	* @param m4ac pointer to MPEG4AudioConfig, used for parsing
948	* @param gb buffer holding an audio specific config
949	* @param get_bit_alignment relative alignment for byte align operations
950	* @param sync_extension look for an appended sync extension
951	*
952	* @return Returns error status or number of consumed bits. <0 - error
953	*/
954	static int decode_audio_specific_config_gb(AACContext *ac,
955	AVCodecContext *avctx,
956	MPEG4AudioConfig *m4ac,
957	GetBitContext *gb,
958	int get_bit_alignment,
959	int sync_extension)
960	{
961	int i, ret;
962	GetBitContext gbc = *gb;
963
964	if ((i = ff_mpeg4audio_get_config_gb(m4ac, &gbc, sync_extension)) < 0)
965	return AVERROR_INVALIDDATA;
966
967	if (m4ac->sampling_index > 12) {
968	av_log(avctx, AV_LOG_ERROR,
969	"invalid sampling rate index %d\n",
970	m4ac->sampling_index);
971	return AVERROR_INVALIDDATA;
972	}
973	if (m4ac->object_type == AOT_ER_AAC_LD &&
974	(m4ac->sampling_index < 3 || m4ac->sampling_index > 7)) {
975	av_log(avctx, AV_LOG_ERROR,
976	"invalid low delay sampling rate index %d\n",
977	m4ac->sampling_index);
978	return AVERROR_INVALIDDATA;
979	}
980
981	skip_bits_long(gb, i);
982
983	switch (m4ac->object_type) {
984	case AOT_AAC_MAIN:
985	case AOT_AAC_LC:
986	case AOT_AAC_LTP:
987	case AOT_ER_AAC_LC:
988	case AOT_ER_AAC_LD:
989	if ((ret = decode_ga_specific_config(ac, avctx, gb, get_bit_alignment,
990	m4ac, m4ac->chan_config)) < 0)
991	return ret;
992	break;
993	case AOT_ER_AAC_ELD:
994	if ((ret = decode_eld_specific_config(ac, avctx, gb,
995	m4ac, m4ac->chan_config)) < 0)
996	return ret;
997	break;
998	default:
999	avpriv_report_missing_feature(avctx,
1000	"Audio object type %s%d",
1001	m4ac->sbr == 1 ? "SBR+" : "",
1002	m4ac->object_type);
1003	return AVERROR(ENOSYS);
1004	}
1005
1006	ff_dlog(avctx,
1007	"AOT %d chan config %d sampling index %d (%d) SBR %d PS %d\n",
1008	m4ac->object_type, m4ac->chan_config, m4ac->sampling_index,
1009	m4ac->sample_rate, m4ac->sbr,
1010	m4ac->ps);
1011
1012	return get_bits_count(gb);
1013	}
1014
1015	static int decode_audio_specific_config(AACContext *ac,
1016	AVCodecContext *avctx,
1017	MPEG4AudioConfig *m4ac,
1018	const uint8_t *data, int64_t bit_size,
1019	int sync_extension)
1020	{
1021	int i, ret;
1022	GetBitContext gb;
1023
1024	if (bit_size < 0 || bit_size > INT_MAX) {
1025	av_log(avctx, AV_LOG_ERROR, "Audio specific config size is invalid\n");
1026	return AVERROR_INVALIDDATA;
1027	}
1028
1029	ff_dlog(avctx, "audio specific config size %d\n", (int)bit_size >> 3);
1030	for (i = 0; i < bit_size >> 3; i++)
1031	ff_dlog(avctx, "%02x ", data[i]);
1032	ff_dlog(avctx, "\n");
1033
1034	if ((ret = init_get_bits(&gb, data, bit_size)) < 0)
1035	return ret;
1036
1037	return decode_audio_specific_config_gb(ac, avctx, m4ac, &gb, 0,
1038	sync_extension);
1039	}
1040
1041	/**
1042	* linear congruential pseudorandom number generator
1043	*
1044	* @param previous_val pointer to the current state of the generator
1045	*
1046	* @return Returns a 32-bit pseudorandom integer
1047	*/
1048	static av_always_inline int lcg_random(unsigned previous_val)
1049	{
1050	union { unsigned u; int s; } v = { previous_val * 1664525u + 1013904223 };
1051	return v.s;
1052	}
1053
1054	static void reset_all_predictors(PredictorState *ps)
1055	{
1056	int i;
1057	for (i = 0; i < MAX_PREDICTORS; i++)
1058	reset_predict_state(&ps[i]);
1059	}
1060
1061	static int sample_rate_idx (int rate)
1062	{
1063	if (92017 <= rate) return 0;
1064	else if (75132 <= rate) return 1;
1065	else if (55426 <= rate) return 2;
1066	else if (46009 <= rate) return 3;
1067	else if (37566 <= rate) return 4;
1068	else if (27713 <= rate) return 5;
1069	else if (23004 <= rate) return 6;
1070	else if (18783 <= rate) return 7;
1071	else if (13856 <= rate) return 8;
1072	else if (11502 <= rate) return 9;
1073	else if (9391 <= rate) return 10;
1074	else return 11;
1075	}
1076
1077	static void reset_predictor_group(PredictorState *ps, int group_num)
1078	{
1079	int i;
1080	for (i = group_num - 1; i < MAX_PREDICTORS; i += 30)
1081	reset_predict_state(&ps[i]);
1082	}
1083
1084	#define AAC_INIT_VLC_STATIC(num, size) \
1085	INIT_VLC_STATIC(&vlc_spectral[num], 8, ff_aac_spectral_sizes[num], \
1086	ff_aac_spectral_bits[num], sizeof(ff_aac_spectral_bits[num][0]), \
1087	sizeof(ff_aac_spectral_bits[num][0]), \
1088	ff_aac_spectral_codes[num], sizeof(ff_aac_spectral_codes[num][0]), \
1089	sizeof(ff_aac_spectral_codes[num][0]), \
1090	size);
1091
1092	static void aacdec_init(AACContext *ac);
1093
1094	static av_cold void aac_static_table_init(void)
1095	{
1096	AAC_INIT_VLC_STATIC( 0, 304);
1097	AAC_INIT_VLC_STATIC( 1, 270);
1098	AAC_INIT_VLC_STATIC( 2, 550);
1099	AAC_INIT_VLC_STATIC( 3, 300);
1100	AAC_INIT_VLC_STATIC( 4, 328);
1101	AAC_INIT_VLC_STATIC( 5, 294);
1102	AAC_INIT_VLC_STATIC( 6, 306);
1103	AAC_INIT_VLC_STATIC( 7, 268);
1104	AAC_INIT_VLC_STATIC( 8, 510);
1105	AAC_INIT_VLC_STATIC( 9, 366);
1106	AAC_INIT_VLC_STATIC(10, 462);
1107
1108	AAC_RENAME(ff_aac_sbr_init)();
1109
1110	ff_aac_tableinit();
1111
1112	INIT_VLC_STATIC(&vlc_scalefactors, 7,
1113	FF_ARRAY_ELEMS(ff_aac_scalefactor_code),
1114	ff_aac_scalefactor_bits,
1115	sizeof(ff_aac_scalefactor_bits[0]),
1116	sizeof(ff_aac_scalefactor_bits[0]),
1117	ff_aac_scalefactor_code,
1118	sizeof(ff_aac_scalefactor_code[0]),
1119	sizeof(ff_aac_scalefactor_code[0]),
1120	352);
1121
1122	// window initialization
1123	AAC_RENAME(ff_kbd_window_init)(AAC_RENAME(ff_aac_kbd_long_1024), 4.0, 1024);
1124	AAC_RENAME(ff_kbd_window_init)(AAC_RENAME(ff_aac_kbd_short_128), 6.0, 128);
1125	AAC_RENAME(ff_init_ff_sine_windows)(10);
1126	AAC_RENAME(ff_init_ff_sine_windows)( 9);
1127	AAC_RENAME(ff_init_ff_sine_windows)( 7);
1128
1129	AAC_RENAME(ff_cbrt_tableinit)();
1130	}
1131
1132	static AVOnce aac_table_init = AV_ONCE_INIT;
1133
1134	static av_cold int aac_decode_init(AVCodecContext *avctx)
1135	{
1136	AACContext *ac = avctx->priv_data;
1137	int ret;
1138
1139	ret = ff_thread_once(&aac_table_init, &aac_static_table_init);
1140	if (ret != 0)
1141	return AVERROR_UNKNOWN;
1142
1143	ac->avctx = avctx;
1144	ac->oc[1].m4ac.sample_rate = avctx->sample_rate;
1145
1146	aacdec_init(ac);
1147	#if USE_FIXED
1148	avctx->sample_fmt = AV_SAMPLE_FMT_S32P;
1149	#else
1150	avctx->sample_fmt = AV_SAMPLE_FMT_FLTP;
1151	#endif /* USE_FIXED */
1152
1153	if (avctx->extradata_size > 0) {
1154	if ((ret = decode_audio_specific_config(ac, ac->avctx, &ac->oc[1].m4ac,
1155	avctx->extradata,
1156	avctx->extradata_size * 8LL,
1157	1)) < 0)
1158	return ret;
1159	} else {
1160	int sr, i;
1161	uint8_t layout_map[MAX_ELEM_ID*4][3];
1162	int layout_map_tags;
1163
1164	sr = sample_rate_idx(avctx->sample_rate);
1165	ac->oc[1].m4ac.sampling_index = sr;
1166	ac->oc[1].m4ac.channels = avctx->channels;
1167	ac->oc[1].m4ac.sbr = -1;
1168	ac->oc[1].m4ac.ps = -1;
1169
1170	for (i = 0; i < FF_ARRAY_ELEMS(ff_mpeg4audio_channels); i++)
1171	if (ff_mpeg4audio_channels[i] == avctx->channels)
1172	break;
1173	if (i == FF_ARRAY_ELEMS(ff_mpeg4audio_channels)) {
1174	i = 0;
1175	}
1176	ac->oc[1].m4ac.chan_config = i;
1177
1178	if (ac->oc[1].m4ac.chan_config) {
1179	int ret = set_default_channel_config(avctx, layout_map,
1180	&layout_map_tags, ac->oc[1].m4ac.chan_config);
1181	if (!ret)
1182	output_configure(ac, layout_map, layout_map_tags,
1183	OC_GLOBAL_HDR, 0);
1184	else if (avctx->err_recognition & AV_EF_EXPLODE)
1185	return AVERROR_INVALIDDATA;
1186	}
1187	}
1188
1189	if (avctx->channels > MAX_CHANNELS) {
1190	av_log(avctx, AV_LOG_ERROR, "Too many channels\n");
1191	return AVERROR_INVALIDDATA;
1192	}
1193
1194	#if USE_FIXED
1195	ac->fdsp = avpriv_alloc_fixed_dsp(avctx->flags & AV_CODEC_FLAG_BITEXACT);
1196	#else
1197	ac->fdsp = avpriv_float_dsp_alloc(avctx->flags & AV_CODEC_FLAG_BITEXACT);
1198	#endif /* USE_FIXED */
1199	if (!ac->fdsp) {
1200	return AVERROR(ENOMEM);
1201	}
1202
1203	ac->random_state = 0x1f2e3d4c;
1204
1205	AAC_RENAME_32(ff_mdct_init)(&ac->mdct, 11, 1, 1.0 / RANGE15(1024.0));
1206	AAC_RENAME_32(ff_mdct_init)(&ac->mdct_ld, 10, 1, 1.0 / RANGE15(512.0));
1207	AAC_RENAME_32(ff_mdct_init)(&ac->mdct_small, 8, 1, 1.0 / RANGE15(128.0));
1208	AAC_RENAME_32(ff_mdct_init)(&ac->mdct_ltp, 11, 0, RANGE15(-2.0));
1209	#if !USE_FIXED
1210	ret = ff_mdct15_init(&ac->mdct480, 1, 5, -1.0f);
1211	if (ret < 0)
1212	return ret;
1213	#endif
1214
1215	return 0;
1216	}
1217
1218	/**
1219	* Skip data_stream_element; reference: table 4.10.
1220	*/
1221	static int skip_data_stream_element(AACContext *ac, GetBitContext *gb)
1222	{
1223	int byte_align = get_bits1(gb);
1224	int count = get_bits(gb, 8);
1225	if (count == 255)
1226	count += get_bits(gb, 8);
1227	if (byte_align)
1228	align_get_bits(gb);
1229
1230	if (get_bits_left(gb) < 8 * count) {
1231	av_log(ac->avctx, AV_LOG_ERROR, "skip_data_stream_element: "overread_err);
1232	return AVERROR_INVALIDDATA;
1233	}
1234	skip_bits_long(gb, 8 * count);
1235	return 0;
1236	}
1237
1238	static int decode_prediction(AACContext *ac, IndividualChannelStream *ics,
1239	GetBitContext *gb)
1240	{
1241	int sfb;
1242	if (get_bits1(gb)) {
1243	ics->predictor_reset_group = get_bits(gb, 5);
1244	if (ics->predictor_reset_group == 0 ||
1245	ics->predictor_reset_group > 30) {
1246	av_log(ac->avctx, AV_LOG_ERROR,
1247	"Invalid Predictor Reset Group.\n");
1248	return AVERROR_INVALIDDATA;
1249	}
1250	}
1251	for (sfb = 0; sfb < FFMIN(ics->max_sfb, ff_aac_pred_sfb_max[ac->oc[1].m4ac.sampling_index]); sfb++) {
1252	ics->prediction_used[sfb] = get_bits1(gb);
1253	}
1254	return 0;
1255	}
1256
1257	/**
1258	* Decode Long Term Prediction data; reference: table 4.xx.
1259	*/
1260	static void decode_ltp(LongTermPrediction *ltp,
1261	GetBitContext *gb, uint8_t max_sfb)
1262	{
1263	int sfb;
1264
1265	ltp->lag = get_bits(gb, 11);
1266	ltp->coef = ltp_coef[get_bits(gb, 3)];
1267	for (sfb = 0; sfb < FFMIN(max_sfb, MAX_LTP_LONG_SFB); sfb++)
1268	ltp->used[sfb] = get_bits1(gb);
1269	}
1270
1271	/**
1272	* Decode Individual Channel Stream info; reference: table 4.6.
1273	*/
1274	static int decode_ics_info(AACContext *ac, IndividualChannelStream *ics,
1275	GetBitContext *gb)
1276	{
1277	const MPEG4AudioConfig *const m4ac = &ac->oc[1].m4ac;
1278	const int aot = m4ac->object_type;
1279	const int sampling_index = m4ac->sampling_index;
1280	if (aot != AOT_ER_AAC_ELD) {
1281	if (get_bits1(gb)) {
1282	av_log(ac->avctx, AV_LOG_ERROR, "Reserved bit set.\n");
1283	if (ac->avctx->err_recognition & AV_EF_BITSTREAM)
1284	return AVERROR_INVALIDDATA;
1285	}
1286	ics->window_sequence[1] = ics->window_sequence[0];
1287	ics->window_sequence[0] = get_bits(gb, 2);
1288	if (aot == AOT_ER_AAC_LD &&
1289	ics->window_sequence[0] != ONLY_LONG_SEQUENCE) {
1290	av_log(ac->avctx, AV_LOG_ERROR,
1291	"AAC LD is only defined for ONLY_LONG_SEQUENCE but "
1292	"window sequence %d found.\n", ics->window_sequence[0]);
1293	ics->window_sequence[0] = ONLY_LONG_SEQUENCE;
1294	return AVERROR_INVALIDDATA;
1295	}
1296	ics->use_kb_window[1] = ics->use_kb_window[0];
1297	ics->use_kb_window[0] = get_bits1(gb);
1298	}
1299	ics->num_window_groups = 1;
1300	ics->group_len[0] = 1;
1301	if (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) {
1302	int i;
1303	ics->max_sfb = get_bits(gb, 4);
1304	for (i = 0; i < 7; i++) {
1305	if (get_bits1(gb)) {
1306	ics->group_len[ics->num_window_groups - 1]++;
1307	} else {
1308	ics->num_window_groups++;
1309	ics->group_len[ics->num_window_groups - 1] = 1;
1310	}
1311	}
1312	ics->num_windows = 8;
1313	ics->swb_offset = ff_swb_offset_128[sampling_index];
1314	ics->num_swb = ff_aac_num_swb_128[sampling_index];
1315	ics->tns_max_bands = ff_tns_max_bands_128[sampling_index];
1316	ics->predictor_present = 0;
1317	} else {
1318	ics->max_sfb = get_bits(gb, 6);
1319	ics->num_windows = 1;
1320	if (aot == AOT_ER_AAC_LD || aot == AOT_ER_AAC_ELD) {
1321	if (m4ac->frame_length_short) {
1322	ics->swb_offset = ff_swb_offset_480[sampling_index];
1323	ics->num_swb = ff_aac_num_swb_480[sampling_index];
1324	ics->tns_max_bands = ff_tns_max_bands_480[sampling_index];
1325	} else {
1326	ics->swb_offset = ff_swb_offset_512[sampling_index];
1327	ics->num_swb = ff_aac_num_swb_512[sampling_index];
1328	ics->tns_max_bands = ff_tns_max_bands_512[sampling_index];
1329	}
1330	if (!ics->num_swb || !ics->swb_offset)
1331	return AVERROR_BUG;
1332	} else {
1333	ics->swb_offset = ff_swb_offset_1024[sampling_index];
1334	ics->num_swb = ff_aac_num_swb_1024[sampling_index];
1335	ics->tns_max_bands = ff_tns_max_bands_1024[sampling_index];
1336	}
1337	if (aot != AOT_ER_AAC_ELD) {
1338	ics->predictor_present = get_bits1(gb);
1339	ics->predictor_reset_group = 0;
1340	}
1341	if (ics->predictor_present) {
1342	if (aot == AOT_AAC_MAIN) {
1343	if (decode_prediction(ac, ics, gb)) {
1344	goto fail;
1345	}
1346	} else if (aot == AOT_AAC_LC ||
1347	aot == AOT_ER_AAC_LC) {
1348	av_log(ac->avctx, AV_LOG_ERROR,
1349	"Prediction is not allowed in AAC-LC.\n");
1350	goto fail;
1351	} else {
1352	if (aot == AOT_ER_AAC_LD) {
1353	av_log(ac->avctx, AV_LOG_ERROR,
1354	"LTP in ER AAC LD not yet implemented.\n");
1355	return AVERROR_PATCHWELCOME;
1356	}
1357	if ((ics->ltp.present = get_bits(gb, 1)))
1358	decode_ltp(&ics->ltp, gb, ics->max_sfb);
1359	}
1360	}
1361	}
1362
1363	if (ics->max_sfb > ics->num_swb) {
1364	av_log(ac->avctx, AV_LOG_ERROR,
1365	"Number of scalefactor bands in group (%d) "
1366	"exceeds limit (%d).\n",
1367	ics->max_sfb, ics->num_swb);
1368	goto fail;
1369	}
1370
1371	return 0;
1372	fail:
1373	ics->max_sfb = 0;
1374	return AVERROR_INVALIDDATA;
1375	}
1376
1377	/**
1378	* Decode band types (section_data payload); reference: table 4.46.
1379	*
1380	* @param band_type array of the used band type
1381	* @param band_type_run_end array of the last scalefactor band of a band type run
1382	*
1383	* @return Returns error status. 0 - OK, !0 - error
1384	*/
1385	static int decode_band_types(AACContext *ac, enum BandType band_type[120],
1386	int band_type_run_end[120], GetBitContext *gb,
1387	IndividualChannelStream *ics)
1388	{
1389	int g, idx = 0;
1390	const int bits = (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) ? 3 : 5;
1391	for (g = 0; g < ics->num_window_groups; g++) {
1392	int k = 0;
1393	while (k < ics->max_sfb) {
1394	uint8_t sect_end = k;
1395	int sect_len_incr;
1396	int sect_band_type = get_bits(gb, 4);
1397	if (sect_band_type == 12) {
1398	av_log(ac->avctx, AV_LOG_ERROR, "invalid band type\n");
1399	return AVERROR_INVALIDDATA;
1400	}
1401	do {
1402	sect_len_incr = get_bits(gb, bits);
1403	sect_end += sect_len_incr;
1404	if (get_bits_left(gb) < 0) {
1405	av_log(ac->avctx, AV_LOG_ERROR, "decode_band_types: "overread_err);
1406	return AVERROR_INVALIDDATA;
1407	}
1408	if (sect_end > ics->max_sfb) {
1409	av_log(ac->avctx, AV_LOG_ERROR,
1410	"Number of bands (%d) exceeds limit (%d).\n",
1411	sect_end, ics->max_sfb);
1412	return AVERROR_INVALIDDATA;
1413	}
1414	} while (sect_len_incr == (1 << bits) - 1);
1415	for (; k < sect_end; k++) {
1416	band_type [idx] = sect_band_type;
1417	band_type_run_end[idx++] = sect_end;
1418	}
1419	}
1420	}
1421	return 0;
1422	}
1423
1424	/**
1425	* Decode scalefactors; reference: table 4.47.
1426	*
1427	* @param global_gain first scalefactor value as scalefactors are differentially coded
1428	* @param band_type array of the used band type
1429	* @param band_type_run_end array of the last scalefactor band of a band type run
1430	* @param sf array of scalefactors or intensity stereo positions
1431	*
1432	* @return Returns error status. 0 - OK, !0 - error
1433	*/
1434	static int decode_scalefactors(AACContext *ac, INTFLOAT sf[120], GetBitContext *gb,
1435	unsigned int global_gain,
1436	IndividualChannelStream *ics,
1437	enum BandType band_type[120],
1438	int band_type_run_end[120])
1439	{
1440	int g, i, idx = 0;
1441	int offset[3] = { global_gain, global_gain - NOISE_OFFSET, 0 };
1442	int clipped_offset;
1443	int noise_flag = 1;
1444	for (g = 0; g < ics->num_window_groups; g++) {
1445	for (i = 0; i < ics->max_sfb;) {
1446	int run_end = band_type_run_end[idx];
1447	if (band_type[idx] == ZERO_BT) {
1448	for (; i < run_end; i++, idx++)
1449	sf[idx] = FIXR(0.);
1450	} else if ((band_type[idx] == INTENSITY_BT) ||
1451	(band_type[idx] == INTENSITY_BT2)) {
1452	for (; i < run_end; i++, idx++) {
1453	offset[2] += get_vlc2(gb, vlc_scalefactors.table, 7, 3) - SCALE_DIFF_ZERO;
1454	clipped_offset = av_clip(offset[2], -155, 100);
1455	if (offset[2] != clipped_offset) {
1456	avpriv_request_sample(ac->avctx,
1457	"If you heard an audible artifact, there may be a bug in the decoder. "
1458	"Clipped intensity stereo position (%d -> %d)",
1459	offset[2], clipped_offset);
1460	}
1461	#if USE_FIXED
1462	sf[idx] = 100 - clipped_offset;
1463	#else
1464	sf[idx] = ff_aac_pow2sf_tab[-clipped_offset + POW_SF2_ZERO];
1465	#endif /* USE_FIXED */
1466	}
1467	} else if (band_type[idx] == NOISE_BT) {
1468	for (; i < run_end; i++, idx++) {
1469	if (noise_flag-- > 0)
1470	offset[1] += get_bits(gb, NOISE_PRE_BITS) - NOISE_PRE;
1471	else
1472	offset[1] += get_vlc2(gb, vlc_scalefactors.table, 7, 3) - SCALE_DIFF_ZERO;
1473	clipped_offset = av_clip(offset[1], -100, 155);
1474	if (offset[1] != clipped_offset) {
1475	avpriv_request_sample(ac->avctx,
1476	"If you heard an audible artifact, there may be a bug in the decoder. "
1477	"Clipped noise gain (%d -> %d)",
1478	offset[1], clipped_offset);
1479	}
1480	#if USE_FIXED
1481	sf[idx] = -(100 + clipped_offset);
1482	#else
1483	sf[idx] = -ff_aac_pow2sf_tab[clipped_offset + POW_SF2_ZERO];
1484	#endif /* USE_FIXED */
1485	}
1486	} else {
1487	for (; i < run_end; i++, idx++) {
1488	offset[0] += get_vlc2(gb, vlc_scalefactors.table, 7, 3) - SCALE_DIFF_ZERO;
1489	if (offset[0] > 255U) {
1490	av_log(ac->avctx, AV_LOG_ERROR,
1491	"Scalefactor (%d) out of range.\n", offset[0]);
1492	return AVERROR_INVALIDDATA;
1493	}
1494	#if USE_FIXED
1495	sf[idx] = -offset[0];
1496	#else
1497	sf[idx] = -ff_aac_pow2sf_tab[offset[0] - 100 + POW_SF2_ZERO];
1498	#endif /* USE_FIXED */
1499	}
1500	}
1501	}
1502	}
1503	return 0;
1504	}
1505
1506	/**
1507	* Decode pulse data; reference: table 4.7.
1508	*/
1509	static int decode_pulses(Pulse *pulse, GetBitContext *gb,
1510	const uint16_t *swb_offset, int num_swb)
1511	{
1512	int i, pulse_swb;
1513	pulse->num_pulse = get_bits(gb, 2) + 1;
1514	pulse_swb = get_bits(gb, 6);
1515	if (pulse_swb >= num_swb)
1516	return -1;
1517	pulse->pos[0] = swb_offset[pulse_swb];
1518	pulse->pos[0] += get_bits(gb, 5);
1519	if (pulse->pos[0] >= swb_offset[num_swb])
1520	return -1;
1521	pulse->amp[0] = get_bits(gb, 4);
1522	for (i = 1; i < pulse->num_pulse; i++) {
1523	pulse->pos[i] = get_bits(gb, 5) + pulse->pos[i - 1];
1524	if (pulse->pos[i] >= swb_offset[num_swb])
1525	return -1;
1526	pulse->amp[i] = get_bits(gb, 4);
1527	}
1528	return 0;
1529	}
1530
1531	/**
1532	* Decode Temporal Noise Shaping data; reference: table 4.48.
1533	*
1534	* @return Returns error status. 0 - OK, !0 - error
1535	*/
1536	static int decode_tns(AACContext *ac, TemporalNoiseShaping *tns,
1537	GetBitContext *gb, const IndividualChannelStream *ics)
1538	{
1539	int w, filt, i, coef_len, coef_res, coef_compress;
1540	const int is8 = ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE;
1541	const int tns_max_order = is8 ? 7 : ac->oc[1].m4ac.object_type == AOT_AAC_MAIN ? 20 : 12;
1542	for (w = 0; w < ics->num_windows; w++) {
1543	if ((tns->n_filt[w] = get_bits(gb, 2 - is8))) {
1544	coef_res = get_bits1(gb);
1545
1546	for (filt = 0; filt < tns->n_filt[w]; filt++) {
1547	int tmp2_idx;
1548	tns->length[w][filt] = get_bits(gb, 6 - 2 * is8);
1549
1550	if ((tns->order[w][filt] = get_bits(gb, 5 - 2 * is8)) > tns_max_order) {
1551	av_log(ac->avctx, AV_LOG_ERROR,
1552	"TNS filter order %d is greater than maximum %d.\n",
1553	tns->order[w][filt], tns_max_order);
1554	tns->order[w][filt] = 0;
1555	return AVERROR_INVALIDDATA;
1556	}
1557	if (tns->order[w][filt]) {
1558	tns->direction[w][filt] = get_bits1(gb);
1559	coef_compress = get_bits1(gb);
1560	coef_len = coef_res + 3 - coef_compress;
1561	tmp2_idx = 2 * coef_compress + coef_res;
1562
1563	for (i = 0; i < tns->order[w][filt]; i++)
1564	tns->coef[w][filt][i] = tns_tmp2_map[tmp2_idx][get_bits(gb, coef_len)];
1565	}
1566	}
1567	}
1568	}
1569	return 0;
1570	}
1571
1572	/**
1573	* Decode Mid/Side data; reference: table 4.54.
1574	*
1575	* @param ms_present Indicates mid/side stereo presence. [0] mask is all 0s;
1576	* [1] mask is decoded from bitstream; [2] mask is all 1s;
1577	* [3] reserved for scalable AAC
1578	*/
1579	static void decode_mid_side_stereo(ChannelElement *cpe, GetBitContext *gb,
1580	int ms_present)
1581	{
1582	int idx;
1583	int max_idx = cpe->ch[0].ics.num_window_groups * cpe->ch[0].ics.max_sfb;
1584	if (ms_present == 1) {
1585	for (idx = 0; idx < max_idx; idx++)
1586	cpe->ms_mask[idx] = get_bits1(gb);
1587	} else if (ms_present == 2) {
1588	memset(cpe->ms_mask, 1, max_idx * sizeof(cpe->ms_mask[0]));
1589	}
1590	}
1591
1592	/**
1593	* Decode spectral data; reference: table 4.50.
1594	* Dequantize and scale spectral data; reference: 4.6.3.3.
1595	*
1596	* @param coef array of dequantized, scaled spectral data
1597	* @param sf array of scalefactors or intensity stereo positions
1598	* @param pulse_present set if pulses are present
1599	* @param pulse pointer to pulse data struct
1600	* @param band_type array of the used band type
1601	*
1602	* @return Returns error status. 0 - OK, !0 - error
1603	*/
1604	static int decode_spectrum_and_dequant(AACContext *ac, INTFLOAT coef[1024],
1605	GetBitContext *gb, const INTFLOAT sf[120],
1606	int pulse_present, const Pulse *pulse,
1607	const IndividualChannelStream *ics,
1608	enum BandType band_type[120])
1609	{
1610	int i, k, g, idx = 0;
1611	const int c = 1024 / ics->num_windows;
1612	const uint16_t *offsets = ics->swb_offset;
1613	INTFLOAT *coef_base = coef;
1614
1615	for (g = 0; g < ics->num_windows; g++)
1616	memset(coef + g * 128 + offsets[ics->max_sfb], 0,
1617	sizeof(INTFLOAT) * (c - offsets[ics->max_sfb]));
1618
1619	for (g = 0; g < ics->num_window_groups; g++) {
1620	unsigned g_len = ics->group_len[g];
1621
1622	for (i = 0; i < ics->max_sfb; i++, idx++) {
1623	const unsigned cbt_m1 = band_type[idx] - 1;
1624	INTFLOAT *cfo = coef + offsets[i];
1625	int off_len = offsets[i + 1] - offsets[i];
1626	int group;
1627
1628	if (cbt_m1 >= INTENSITY_BT2 - 1) {
1629	for (group = 0; group < (AAC_SIGNE)g_len; group++, cfo+=128) {
1630	memset(cfo, 0, off_len * sizeof(*cfo));
1631	}
1632	} else if (cbt_m1 == NOISE_BT - 1) {
1633	for (group = 0; group < (AAC_SIGNE)g_len; group++, cfo+=128) {
1634	#if !USE_FIXED
1635	float scale;
1636	#endif /* !USE_FIXED */
1637	INTFLOAT band_energy;
1638
1639	for (k = 0; k < off_len; k++) {
1640	ac->random_state = lcg_random(ac->random_state);
1641	#if USE_FIXED
1642	cfo[k] = ac->random_state >> 3;
1643	#else
1644	cfo[k] = ac->random_state;
1645	#endif /* USE_FIXED */
1646	}
1647
1648	#if USE_FIXED
1649	band_energy = ac->fdsp->scalarproduct_fixed(cfo, cfo, off_len);
1650	band_energy = fixed_sqrt(band_energy, 31);
1651	noise_scale(cfo, sf[idx], band_energy, off_len);
1652	#else
1653	band_energy = ac->fdsp->scalarproduct_float(cfo, cfo, off_len);
1654	scale = sf[idx] / sqrtf(band_energy);
1655	ac->fdsp->vector_fmul_scalar(cfo, cfo, scale, off_len);
1656	#endif /* USE_FIXED */
1657	}
1658	} else {
1659	#if !USE_FIXED
1660	const float *vq = ff_aac_codebook_vector_vals[cbt_m1];
1661	#endif /* !USE_FIXED */
1662	const uint16_t *cb_vector_idx = ff_aac_codebook_vector_idx[cbt_m1];
1663	VLC_TYPE (*vlc_tab)[2] = vlc_spectral[cbt_m1].table;
1664	OPEN_READER(re, gb);
1665
1666	switch (cbt_m1 >> 1) {
1667	case 0:
1668	for (group = 0; group < (AAC_SIGNE)g_len; group++, cfo+=128) {
1669	INTFLOAT *cf = cfo;
1670	int len = off_len;
1671
1672	do {
1673	int code;
1674	unsigned cb_idx;
1675
1676	UPDATE_CACHE(re, gb);
1677	GET_VLC(code, re, gb, vlc_tab, 8, 2);
1678	cb_idx = cb_vector_idx[code];
1679	#if USE_FIXED
1680	cf = DEC_SQUAD(cf, cb_idx);
1681	#else
1682	cf = VMUL4(cf, vq, cb_idx, sf + idx);
1683	#endif /* USE_FIXED */
1684	} while (len -= 4);
1685	}
1686	break;
1687
1688	case 1:
1689	for (group = 0; group < (AAC_SIGNE)g_len; group++, cfo+=128) {
1690	INTFLOAT *cf = cfo;
1691	int len = off_len;
1692
1693	do {
1694	int code;
1695	unsigned nnz;
1696	unsigned cb_idx;
1697	uint32_t bits;
1698
1699	UPDATE_CACHE(re, gb);
1700	GET_VLC(code, re, gb, vlc_tab, 8, 2);
1701	cb_idx = cb_vector_idx[code];
1702	nnz = cb_idx >> 8 & 15;
1703	bits = nnz ? GET_CACHE(re, gb) : 0;
1704	LAST_SKIP_BITS(re, gb, nnz);
1705	#if USE_FIXED
1706	cf = DEC_UQUAD(cf, cb_idx, bits);
1707	#else
1708	cf = VMUL4S(cf, vq, cb_idx, bits, sf + idx);
1709	#endif /* USE_FIXED */
1710	} while (len -= 4);
1711	}
1712	break;
1713
1714	case 2:
1715	for (group = 0; group < (AAC_SIGNE)g_len; group++, cfo+=128) {
1716	INTFLOAT *cf = cfo;
1717	int len = off_len;
1718
1719	do {
1720	int code;
1721	unsigned cb_idx;
1722
1723	UPDATE_CACHE(re, gb);
1724	GET_VLC(code, re, gb, vlc_tab, 8, 2);
1725	cb_idx = cb_vector_idx[code];
1726	#if USE_FIXED
1727	cf = DEC_SPAIR(cf, cb_idx);
1728	#else
1729	cf = VMUL2(cf, vq, cb_idx, sf + idx);
1730	#endif /* USE_FIXED */
1731	} while (len -= 2);
1732	}
1733	break;
1734
1735	case 3:
1736	case 4:
1737	for (group = 0; group < (AAC_SIGNE)g_len; group++, cfo+=128) {
1738	INTFLOAT *cf = cfo;
1739	int len = off_len;
1740
1741	do {
1742	int code;
1743	unsigned nnz;
1744	unsigned cb_idx;
1745	unsigned sign;
1746
1747	UPDATE_CACHE(re, gb);
1748	GET_VLC(code, re, gb, vlc_tab, 8, 2);
1749	cb_idx = cb_vector_idx[code];
1750	nnz = cb_idx >> 8 & 15;
1751	sign = nnz ? SHOW_UBITS(re, gb, nnz) << (cb_idx >> 12) : 0;
1752	LAST_SKIP_BITS(re, gb, nnz);
1753	#if USE_FIXED
1754	cf = DEC_UPAIR(cf, cb_idx, sign);
1755	#else
1756	cf = VMUL2S(cf, vq, cb_idx, sign, sf + idx);
1757	#endif /* USE_FIXED */
1758	} while (len -= 2);
1759	}
1760	break;
1761
1762	default:
1763	for (group = 0; group < (AAC_SIGNE)g_len; group++, cfo+=128) {
1764	#if USE_FIXED
1765	int *icf = cfo;
1766	int v;
1767	#else
1768	float *cf = cfo;
1769	uint32_t *icf = (uint32_t *) cf;
1770	#endif /* USE_FIXED */
1771	int len = off_len;
1772
1773	do {
1774	int code;
1775	unsigned nzt, nnz;
1776	unsigned cb_idx;
1777	uint32_t bits;
1778	int j;
1779
1780	UPDATE_CACHE(re, gb);
1781	GET_VLC(code, re, gb, vlc_tab, 8, 2);
1782
1783	if (!code) {
1784	*icf++ = 0;
1785	*icf++ = 0;
1786	continue;
1787	}
1788
1789	cb_idx = cb_vector_idx[code];
1790	nnz = cb_idx >> 12;
1791	nzt = cb_idx >> 8;
1792	bits = SHOW_UBITS(re, gb, nnz) << (32-nnz);
1793	LAST_SKIP_BITS(re, gb, nnz);
1794
1795	for (j = 0; j < 2; j++) {
1796	if (nzt & 1<<j) {
1797	uint32_t b;
1798	int n;
1799	/* The total length of escape_sequence must be < 22 bits according
1800	to the specification (i.e. max is 111111110xxxxxxxxxxxx). */
1801	UPDATE_CACHE(re, gb);
1802	b = GET_CACHE(re, gb);
1803	b = 31 - av_log2(~b);
1804
1805	if (b > 8) {
1806	av_log(ac->avctx, AV_LOG_ERROR, "error in spectral data, ESC overflow\n");
1807	return AVERROR_INVALIDDATA;
1808	}
1809
1810	SKIP_BITS(re, gb, b + 1);
1811	b += 4;
1812	n = (1 << b) + SHOW_UBITS(re, gb, b);
1813	LAST_SKIP_BITS(re, gb, b);
1814	#if USE_FIXED
1815	v = n;
1816	if (bits & 1U<<31)
1817	v = -v;
1818	*icf++ = v;
1819	#else
1820	*icf++ = ff_cbrt_tab[n] | (bits & 1U<<31);
1821	#endif /* USE_FIXED */
1822	bits <<= 1;
1823	} else {
1824	#if USE_FIXED
1825	v = cb_idx & 15;
1826	if (bits & 1U<<31)
1827	v = -v;
1828	*icf++ = v;
1829	#else
1830	unsigned v = ((const uint32_t*)vq)[cb_idx & 15];
1831	*icf++ = (bits & 1U<<31) | v;
1832	#endif /* USE_FIXED */
1833	bits <<= !!v;
1834	}
1835	cb_idx >>= 4;
1836	}
1837	} while (len -= 2);
1838	#if !USE_FIXED
1839	ac->fdsp->vector_fmul_scalar(cfo, cfo, sf[idx], off_len);
1840	#endif /* !USE_FIXED */
1841	}
1842	}
1843
1844	CLOSE_READER(re, gb);
1845	}
1846	}
1847	coef += g_len << 7;
1848	}
1849
1850	if (pulse_present) {
1851	idx = 0;
1852	for (i = 0; i < pulse->num_pulse; i++) {
1853	INTFLOAT co = coef_base[ pulse->pos[i] ];
1854	while (offsets[idx + 1] <= pulse->pos[i])
1855	idx++;
1856	if (band_type[idx] != NOISE_BT && sf[idx]) {
1857	INTFLOAT ico = -pulse->amp[i];
1858	#if USE_FIXED
1859	if (co) {
1860	ico = co + (co > 0 ? -ico : ico);
1861	}
1862	coef_base[ pulse->pos[i] ] = ico;
1863	#else
1864	if (co) {
1865	co /= sf[idx];
1866	ico = co / sqrtf(sqrtf(fabsf(co))) + (co > 0 ? -ico : ico);
1867	}
1868	coef_base[ pulse->pos[i] ] = cbrtf(fabsf(ico)) * ico * sf[idx];
1869	#endif /* USE_FIXED */
1870	}
1871	}
1872	}
1873	#if USE_FIXED
1874	coef = coef_base;
1875	idx = 0;
1876	for (g = 0; g < ics->num_window_groups; g++) {
1877	unsigned g_len = ics->group_len[g];
1878
1879	for (i = 0; i < ics->max_sfb; i++, idx++) {
1880	const unsigned cbt_m1 = band_type[idx] - 1;
1881	int *cfo = coef + offsets[i];
1882	int off_len = offsets[i + 1] - offsets[i];
1883	int group;
1884
1885	if (cbt_m1 < NOISE_BT - 1) {
1886	for (group = 0; group < (int)g_len; group++, cfo+=128) {
1887	ac->vector_pow43(cfo, off_len);
1888	ac->subband_scale(cfo, cfo, sf[idx], 34, off_len);
1889	}
1890	}
1891	}
1892	coef += g_len << 7;
1893	}
1894	#endif /* USE_FIXED */
1895	return 0;
1896	}
1897
1898	/**
1899	* Apply AAC-Main style frequency domain prediction.
1900	*/
1901	static void apply_prediction(AACContext *ac, SingleChannelElement *sce)
1902	{
1903	int sfb, k;
1904
1905	if (!sce->ics.predictor_initialized) {
1906	reset_all_predictors(sce->predictor_state);
1907	sce->ics.predictor_initialized = 1;
1908	}
1909
1910	if (sce->ics.window_sequence[0] != EIGHT_SHORT_SEQUENCE) {
1911	for (sfb = 0;
1912	sfb < ff_aac_pred_sfb_max[ac->oc[1].m4ac.sampling_index];
1913	sfb++) {
1914	for (k = sce->ics.swb_offset[sfb];
1915	k < sce->ics.swb_offset[sfb + 1];
1916	k++) {
1917	predict(&sce->predictor_state[k], &sce->coeffs[k],
1918	sce->ics.predictor_present &&
1919	sce->ics.prediction_used[sfb]);
1920	}
1921	}
1922	if (sce->ics.predictor_reset_group)
1923	reset_predictor_group(sce->predictor_state,
1924	sce->ics.predictor_reset_group);
1925	} else
1926	reset_all_predictors(sce->predictor_state);
1927	}
1928
1929	/**
1930	* Decode an individual_channel_stream payload; reference: table 4.44.
1931	*
1932	* @param common_window Channels have independent [0], or shared [1], Individual Channel Stream information.
1933	* @param scale_flag scalable [1] or non-scalable [0] AAC (Unused until scalable AAC is implemented.)
1934	*
1935	* @return Returns error status. 0 - OK, !0 - error
1936	*/
1937	static int decode_ics(AACContext *ac, SingleChannelElement *sce,
1938	GetBitContext *gb, int common_window, int scale_flag)
1939	{
1940	Pulse pulse;
1941	TemporalNoiseShaping *tns = &sce->tns;
1942	IndividualChannelStream *ics = &sce->ics;
1943	INTFLOAT *out = sce->coeffs;
1944	int global_gain, eld_syntax, er_syntax, pulse_present = 0;
1945	int ret;
1946
1947	eld_syntax = ac->oc[1].m4ac.object_type == AOT_ER_AAC_ELD;
1948	er_syntax = ac->oc[1].m4ac.object_type == AOT_ER_AAC_LC ||
1949	ac->oc[1].m4ac.object_type == AOT_ER_AAC_LTP ||
1950	ac->oc[1].m4ac.object_type == AOT_ER_AAC_LD ||
1951	ac->oc[1].m4ac.object_type == AOT_ER_AAC_ELD;
1952
1953	/* This assignment is to silence a GCC warning about the variable being used
1954	* uninitialized when in fact it always is.
1955	*/
1956	pulse.num_pulse = 0;
1957
1958	global_gain = get_bits(gb, 8);
1959
1960	if (!common_window && !scale_flag) {
1961	if (decode_ics_info(ac, ics, gb) < 0)
1962	return AVERROR_INVALIDDATA;
1963	}
1964
1965	if ((ret = decode_band_types(ac, sce->band_type,
1966	sce->band_type_run_end, gb, ics)) < 0)
1967	return ret;
1968	if ((ret = decode_scalefactors(ac, sce->sf, gb, global_gain, ics,
1969	sce->band_type, sce->band_type_run_end)) < 0)
1970	return ret;
1971
1972	pulse_present = 0;
1973	if (!scale_flag) {
1974	if (!eld_syntax && (pulse_present = get_bits1(gb))) {
1975	if (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) {
1976	av_log(ac->avctx, AV_LOG_ERROR,
1977	"Pulse tool not allowed in eight short sequence.\n");
1978	return AVERROR_INVALIDDATA;
1979	}
1980	if (decode_pulses(&pulse, gb, ics->swb_offset, ics->num_swb)) {
1981	av_log(ac->avctx, AV_LOG_ERROR,
1982	"Pulse data corrupt or invalid.\n");
1983	return AVERROR_INVALIDDATA;
1984	}
1985	}
1986	tns->present = get_bits1(gb);
1987	if (tns->present && !er_syntax)
1988	if (decode_tns(ac, tns, gb, ics) < 0)
1989	return AVERROR_INVALIDDATA;
1990	if (!eld_syntax && get_bits1(gb)) {
1991	avpriv_request_sample(ac->avctx, "SSR");
1992	return AVERROR_PATCHWELCOME;
1993	}
1994	// I see no textual basis in the spec for this occurring after SSR gain
1995	// control, but this is what both reference and real implmentations do
1996	if (tns->present && er_syntax)
1997	if (decode_tns(ac, tns, gb, ics) < 0)
1998	return AVERROR_INVALIDDATA;
1999	}
2000
2001	if (decode_spectrum_and_dequant(ac, out, gb, sce->sf, pulse_present,
2002	&pulse, ics, sce->band_type) < 0)
2003	return AVERROR_INVALIDDATA;
2004
2005	if (ac->oc[1].m4ac.object_type == AOT_AAC_MAIN && !common_window)
2006	apply_prediction(ac, sce);
2007
2008	return 0;
2009	}
2010
2011	/**
2012	* Mid/Side stereo decoding; reference: 4.6.8.1.3.
2013	*/
2014	static void apply_mid_side_stereo(AACContext *ac, ChannelElement *cpe)
2015	{
2016	const IndividualChannelStream *ics = &cpe->ch[0].ics;
2017	INTFLOAT *ch0 = cpe->ch[0].coeffs;
2018	INTFLOAT *ch1 = cpe->ch[1].coeffs;
2019	int g, i, group, idx = 0;
2020	const uint16_t *offsets = ics->swb_offset;
2021	for (g = 0; g < ics->num_window_groups; g++) {
2022	for (i = 0; i < ics->max_sfb; i++, idx++) {
2023	if (cpe->ms_mask[idx] &&
2024	cpe->ch[0].band_type[idx] < NOISE_BT &&
2025	cpe->ch[1].band_type[idx] < NOISE_BT) {
2026	#if USE_FIXED
2027	for (group = 0; group < ics->group_len[g]; group++) {
2028	ac->fdsp->butterflies_fixed(ch0 + group * 128 + offsets[i],
2029	ch1 + group * 128 + offsets[i],
2030	offsets[i+1] - offsets[i]);
2031	#else
2032	for (group = 0; group < ics->group_len[g]; group++) {
2033	ac->fdsp->butterflies_float(ch0 + group * 128 + offsets[i],
2034	ch1 + group * 128 + offsets[i],
2035	offsets[i+1] - offsets[i]);
2036	#endif /* USE_FIXED */
2037	}
2038	}
2039	}
2040	ch0 += ics->group_len[g] * 128;
2041	ch1 += ics->group_len[g] * 128;
2042	}
2043	}
2044
2045	/**
2046	* intensity stereo decoding; reference: 4.6.8.2.3
2047	*
2048	* @param ms_present Indicates mid/side stereo presence. [0] mask is all 0s;
2049	* [1] mask is decoded from bitstream; [2] mask is all 1s;
2050	* [3] reserved for scalable AAC
2051	*/
2052	static void apply_intensity_stereo(AACContext *ac,
2053	ChannelElement *cpe, int ms_present)
2054	{
2055	const IndividualChannelStream *ics = &cpe->ch[1].ics;
2056	SingleChannelElement *sce1 = &cpe->ch[1];
2057	INTFLOAT *coef0 = cpe->ch[0].coeffs, *coef1 = cpe->ch[1].coeffs;
2058	const uint16_t *offsets = ics->swb_offset;
2059	int g, group, i, idx = 0;
2060	int c;
2061	INTFLOAT scale;
2062	for (g = 0; g < ics->num_window_groups; g++) {
2063	for (i = 0; i < ics->max_sfb;) {
2064	if (sce1->band_type[idx] == INTENSITY_BT ||
2065	sce1->band_type[idx] == INTENSITY_BT2) {
2066	const int bt_run_end = sce1->band_type_run_end[idx];
2067	for (; i < bt_run_end; i++, idx++) {
2068	c = -1 + 2 * (sce1->band_type[idx] - 14);
2069	if (ms_present)
2070	c *= 1 - 2 * cpe->ms_mask[idx];
2071	scale = c * sce1->sf[idx];
2072	for (group = 0; group < ics->group_len[g]; group++)
2073	#if USE_FIXED
2074	ac->subband_scale(coef1 + group * 128 + offsets[i],
2075	coef0 + group * 128 + offsets[i],
2076	scale,
2077	23,
2078	offsets[i + 1] - offsets[i]);
2079	#else
2080	ac->fdsp->vector_fmul_scalar(coef1 + group * 128 + offsets[i],
2081	coef0 + group * 128 + offsets[i],
2082	scale,
2083	offsets[i + 1] - offsets[i]);
2084	#endif /* USE_FIXED */
2085	}
2086	} else {
2087	int bt_run_end = sce1->band_type_run_end[idx];
2088	idx += bt_run_end - i;
2089	i = bt_run_end;
2090	}
2091	}
2092	coef0 += ics->group_len[g] * 128;
2093	coef1 += ics->group_len[g] * 128;
2094	}
2095	}
2096
2097	/**
2098	* Decode a channel_pair_element; reference: table 4.4.
2099	*
2100	* @return Returns error status. 0 - OK, !0 - error
2101	*/
2102	static int decode_cpe(AACContext *ac, GetBitContext *gb, ChannelElement *cpe)
2103	{
2104	int i, ret, common_window, ms_present = 0;
2105	int eld_syntax = ac->oc[1].m4ac.object_type == AOT_ER_AAC_ELD;
2106
2107	common_window = eld_syntax || get_bits1(gb);
2108	if (common_window) {
2109	if (decode_ics_info(ac, &cpe->ch[0].ics, gb))
2110	return AVERROR_INVALIDDATA;
2111	i = cpe->ch[1].ics.use_kb_window[0];
2112	cpe->ch[1].ics = cpe->ch[0].ics;
2113	cpe->ch[1].ics.use_kb_window[1] = i;
2114	if (cpe->ch[1].ics.predictor_present &&
2115	(ac->oc[1].m4ac.object_type != AOT_AAC_MAIN))
2116	if ((cpe->ch[1].ics.ltp.present = get_bits(gb, 1)))
2117	decode_ltp(&cpe->ch[1].ics.ltp, gb, cpe->ch[1].ics.max_sfb);
2118	ms_present = get_bits(gb, 2);
2119	if (ms_present == 3) {
2120	av_log(ac->avctx, AV_LOG_ERROR, "ms_present = 3 is reserved.\n");
2121	return AVERROR_INVALIDDATA;
2122	} else if (ms_present)
2123	decode_mid_side_stereo(cpe, gb, ms_present);
2124	}
2125	if ((ret = decode_ics(ac, &cpe->ch[0], gb, common_window, 0)))
2126	return ret;
2127	if ((ret = decode_ics(ac, &cpe->ch[1], gb, common_window, 0)))
2128	return ret;
2129
2130	if (common_window) {
2131	if (ms_present)
2132	apply_mid_side_stereo(ac, cpe);
2133	if (ac->oc[1].m4ac.object_type == AOT_AAC_MAIN) {
2134	apply_prediction(ac, &cpe->ch[0]);
2135	apply_prediction(ac, &cpe->ch[1]);
2136	}
2137	}
2138
2139	apply_intensity_stereo(ac, cpe, ms_present);
2140	return 0;
2141	}
2142
2143	static const float cce_scale[] = {
2144	1.09050773266525765921, //2^(1/8)
2145	1.18920711500272106672, //2^(1/4)
2146	M_SQRT2,
2147	2,
2148	};
2149
2150	/**
2151	* Decode coupling_channel_element; reference: table 4.8.
2152	*
2153	* @return Returns error status. 0 - OK, !0 - error
2154	*/
2155	static int decode_cce(AACContext *ac, GetBitContext *gb, ChannelElement *che)
2156	{
2157	int num_gain = 0;
2158	int c, g, sfb, ret;
2159	int sign;
2160	INTFLOAT scale;
2161	SingleChannelElement *sce = &che->ch[0];
2162	ChannelCoupling *coup = &che->coup;
2163
2164	coup->coupling_point = 2 * get_bits1(gb);
2165	coup->num_coupled = get_bits(gb, 3);
2166	for (c = 0; c <= coup->num_coupled; c++) {
2167	num_gain++;
2168	coup->type[c] = get_bits1(gb) ? TYPE_CPE : TYPE_SCE;
2169	coup->id_select[c] = get_bits(gb, 4);
2170	if (coup->type[c] == TYPE_CPE) {
2171	coup->ch_select[c] = get_bits(gb, 2);
2172	if (coup->ch_select[c] == 3)
2173	num_gain++;
2174	} else
2175	coup->ch_select[c] = 2;
2176	}
2177	coup->coupling_point += get_bits1(gb) || (coup->coupling_point >> 1);
2178
2179	sign = get_bits(gb, 1);
2180	scale = AAC_RENAME(cce_scale)[get_bits(gb, 2)];
2181
2182	if ((ret = decode_ics(ac, sce, gb, 0, 0)))
2183	return ret;
2184
2185	for (c = 0; c < num_gain; c++) {
2186	int idx = 0;
2187	int cge = 1;
2188	int gain = 0;
2189	INTFLOAT gain_cache = FIXR10(1.);
2190	if (c) {
2191	cge = coup->coupling_point == AFTER_IMDCT ? 1 : get_bits1(gb);
2192	gain = cge ? get_vlc2(gb, vlc_scalefactors.table, 7, 3) - 60: 0;
2193	gain_cache = GET_GAIN(scale, gain);
2194	}
2195	if (coup->coupling_point == AFTER_IMDCT) {
2196	coup->gain[c][0] = gain_cache;
2197	} else {
2198	for (g = 0; g < sce->ics.num_window_groups; g++) {
2199	for (sfb = 0; sfb < sce->ics.max_sfb; sfb++, idx++) {
2200	if (sce->band_type[idx] != ZERO_BT) {
2201	if (!cge) {
2202	int t = get_vlc2(gb, vlc_scalefactors.table, 7, 3) - 60;
2203	if (t) {
2204	int s = 1;
2205	t = gain += t;
2206	if (sign) {
2207	s -= 2 * (t & 0x1);
2208	t >>= 1;
2209	}
2210	gain_cache = GET_GAIN(scale, t) * s;
2211	}
2212	}
2213	coup->gain[c][idx] = gain_cache;
2214	}
2215	}
2216	}
2217	}
2218	}
2219	return 0;
2220	}
2221
2222	/**
2223	* Parse whether channels are to be excluded from Dynamic Range Compression; reference: table 4.53.
2224	*
2225	* @return Returns number of bytes consumed.
2226	*/
2227	static int decode_drc_channel_exclusions(DynamicRangeControl *che_drc,
2228	GetBitContext *gb)
2229	{
2230	int i;
2231	int num_excl_chan = 0;
2232
2233	do {
2234	for (i = 0; i < 7; i++)
2235	che_drc->exclude_mask[num_excl_chan++] = get_bits1(gb);
2236	} while (num_excl_chan < MAX_CHANNELS - 7 && get_bits1(gb));
2237
2238	return num_excl_chan / 7;
2239	}
2240
2241	/**
2242	* Decode dynamic range information; reference: table 4.52.
2243	*
2244	* @return Returns number of bytes consumed.
2245	*/
2246	static int decode_dynamic_range(DynamicRangeControl *che_drc,
2247	GetBitContext *gb)
2248	{
2249	int n = 1;
2250	int drc_num_bands = 1;
2251	int i;
2252
2253	/* pce_tag_present? */
2254	if (get_bits1(gb)) {
2255	che_drc->pce_instance_tag = get_bits(gb, 4);
2256	skip_bits(gb, 4); // tag_reserved_bits
2257	n++;
2258	}
2259
2260	/* excluded_chns_present? */
2261	if (get_bits1(gb)) {
2262	n += decode_drc_channel_exclusions(che_drc, gb);
2263	}
2264
2265	/* drc_bands_present? */
2266	if (get_bits1(gb)) {
2267	che_drc->band_incr = get_bits(gb, 4);
2268	che_drc->interpolation_scheme = get_bits(gb, 4);
2269	n++;
2270	drc_num_bands += che_drc->band_incr;
2271	for (i = 0; i < drc_num_bands; i++) {
2272	che_drc->band_top[i] = get_bits(gb, 8);
2273	n++;
2274	}
2275	}
2276
2277	/* prog_ref_level_present? */
2278	if (get_bits1(gb)) {
2279	che_drc->prog_ref_level = get_bits(gb, 7);
2280	skip_bits1(gb); // prog_ref_level_reserved_bits
2281	n++;
2282	}
2283
2284	for (i = 0; i < drc_num_bands; i++) {
2285	che_drc->dyn_rng_sgn[i] = get_bits1(gb);
2286	che_drc->dyn_rng_ctl[i] = get_bits(gb, 7);
2287	n++;
2288	}
2289
2290	return n;
2291	}
2292
2293	static int decode_fill(AACContext *ac, GetBitContext *gb, int len) {
2294	uint8_t buf[256];
2295	int i, major, minor;
2296
2297	if (len < 13+7*8)
2298	goto unknown;
2299
2300	get_bits(gb, 13); len -= 13;
2301
2302	for(i=0; i+1<sizeof(buf) && len>=8; i++, len-=8)
2303	buf[i] = get_bits(gb, 8);
2304
2305	buf[i] = 0;
2306	if (ac->avctx->debug & FF_DEBUG_PICT_INFO)
2307	av_log(ac->avctx, AV_LOG_DEBUG, "FILL:%s\n", buf);
2308
2309	if (sscanf(buf, "libfaac %d.%d", &major, &minor) == 2){
2310	ac->avctx->internal->skip_samples = 1024;
2311	}
2312
2313	unknown:
2314	skip_bits_long(gb, len);
2315
2316	return 0;
2317	}
2318
2319	/**
2320	* Decode extension data (incomplete); reference: table 4.51.
2321	*
2322	* @param cnt length of TYPE_FIL syntactic element in bytes
2323	*
2324	* @return Returns number of bytes consumed
2325	*/
2326	static int decode_extension_payload(AACContext *ac, GetBitContext *gb, int cnt,
2327	ChannelElement *che, enum RawDataBlockType elem_type)
2328	{
2329	int crc_flag = 0;
2330	int res = cnt;
2331	int type = get_bits(gb, 4);
2332
2333	if (ac->avctx->debug & FF_DEBUG_STARTCODE)
2334	av_log(ac->avctx, AV_LOG_DEBUG, "extension type: %d len:%d\n", type, cnt);
2335
2336	switch (type) { // extension type
2337	case EXT_SBR_DATA_CRC:
2338	crc_flag++;
2339	case EXT_SBR_DATA:
2340	if (!che) {
2341	av_log(ac->avctx, AV_LOG_ERROR, "SBR was found before the first channel element.\n");
2342	return res;
2343	} else if (!ac->oc[1].m4ac.sbr) {
2344	av_log(ac->avctx, AV_LOG_ERROR, "SBR signaled to be not-present but was found in the bitstream.\n");
2345	skip_bits_long(gb, 8 * cnt - 4);
2346	return res;
2347	} else if (ac->oc[1].m4ac.sbr == -1 && ac->oc[1].status == OC_LOCKED) {
2348	av_log(ac->avctx, AV_LOG_ERROR, "Implicit SBR was found with a first occurrence after the first frame.\n");
2349	skip_bits_long(gb, 8 * cnt - 4);
2350	return res;
2351	} else if (ac->oc[1].m4ac.ps == -1 && ac->oc[1].status < OC_LOCKED && ac->avctx->channels == 1) {
2352	ac->oc[1].m4ac.sbr = 1;
2353	ac->oc[1].m4ac.ps = 1;
2354	ac->avctx->profile = FF_PROFILE_AAC_HE_V2;
2355	output_configure(ac, ac->oc[1].layout_map, ac->oc[1].layout_map_tags,
2356	ac->oc[1].status, 1);
2357	} else {
2358	ac->oc[1].m4ac.sbr = 1;
2359	ac->avctx->profile = FF_PROFILE_AAC_HE;
2360	}
2361	res = AAC_RENAME(ff_decode_sbr_extension)(ac, &che->sbr, gb, crc_flag, cnt, elem_type);
2362	break;
2363	case EXT_DYNAMIC_RANGE:
2364	res = decode_dynamic_range(&ac->che_drc, gb);
2365	break;
2366	case EXT_FILL:
2367	decode_fill(ac, gb, 8 * cnt - 4);
2368	break;
2369	case EXT_FILL_DATA:
2370	case EXT_DATA_ELEMENT:
2371	default:
2372	skip_bits_long(gb, 8 * cnt - 4);
2373	break;
2374	};
2375	return res;
2376	}
2377
2378	/**
2379	* Decode Temporal Noise Shaping filter coefficients and apply all-pole filters; reference: 4.6.9.3.
2380	*
2381	* @param decode 1 if tool is used normally, 0 if tool is used in LTP.
2382	* @param coef spectral coefficients
2383	*/
2384	static void apply_tns(INTFLOAT coef[1024], TemporalNoiseShaping *tns,
2385	IndividualChannelStream *ics, int decode)
2386	{
2387	const int mmm = FFMIN(ics->tns_max_bands, ics->max_sfb);
2388	int w, filt, m, i;
2389	int bottom, top, order, start, end, size, inc;
2390	INTFLOAT lpc[TNS_MAX_ORDER];
2391	INTFLOAT tmp[TNS_MAX_ORDER+1];
2392
2393	for (w = 0; w < ics->num_windows; w++) {
2394	bottom = ics->num_swb;
2395	for (filt = 0; filt < tns->n_filt[w]; filt++) {
2396	top = bottom;
2397	bottom = FFMAX(0, top - tns->length[w][filt]);
2398	order = tns->order[w][filt];
2399	if (order == 0)
2400	continue;
2401
2402	// tns_decode_coef
2403	AAC_RENAME(compute_lpc_coefs)(tns->coef[w][filt], order, lpc, 0, 0, 0);
2404
2405	start = ics->swb_offset[FFMIN(bottom, mmm)];
2406	end = ics->swb_offset[FFMIN( top, mmm)];
2407	if ((size = end - start) <= 0)
2408	continue;
2409	if (tns->direction[w][filt]) {
2410	inc = -1;
2411	start = end - 1;
2412	} else {
2413	inc = 1;
2414	}
2415	start += w * 128;
2416
2417	if (decode) {
2418	// ar filter
2419	for (m = 0; m < size; m++, start += inc)
2420	for (i = 1; i <= FFMIN(m, order); i++)
2421	coef[start] -= AAC_MUL26(coef[start - i * inc], lpc[i - 1]);
2422	} else {
2423	// ma filter
2424	for (m = 0; m < size; m++, start += inc) {
2425	tmp[0] = coef[start];
2426	for (i = 1; i <= FFMIN(m, order); i++)
2427	coef[start] += AAC_MUL26(tmp[i], lpc[i - 1]);
2428	for (i = order; i > 0; i--)
2429	tmp[i] = tmp[i - 1];
2430	}
2431	}
2432	}
2433	}
2434	}
2435
2436	/**
2437	* Apply windowing and MDCT to obtain the spectral
2438	* coefficient from the predicted sample by LTP.
2439	*/
2440	static void windowing_and_mdct_ltp(AACContext *ac, INTFLOAT *out,
2441	INTFLOAT *in, IndividualChannelStream *ics)
2442	{
2443	const INTFLOAT *lwindow = ics->use_kb_window[0] ? AAC_RENAME(ff_aac_kbd_long_1024) : AAC_RENAME(ff_sine_1024);
2444	const INTFLOAT *swindow = ics->use_kb_window[0] ? AAC_RENAME(ff_aac_kbd_short_128) : AAC_RENAME(ff_sine_128);
2445	const INTFLOAT *lwindow_prev = ics->use_kb_window[1] ? AAC_RENAME(ff_aac_kbd_long_1024) : AAC_RENAME(ff_sine_1024);
2446	const INTFLOAT *swindow_prev = ics->use_kb_window[1] ? AAC_RENAME(ff_aac_kbd_short_128) : AAC_RENAME(ff_sine_128);
2447
2448	if (ics->window_sequence[0] != LONG_STOP_SEQUENCE) {
2449	ac->fdsp->vector_fmul(in, in, lwindow_prev, 1024);
2450	} else {
2451	memset(in, 0, 448 * sizeof(*in));
2452	ac->fdsp->vector_fmul(in + 448, in + 448, swindow_prev, 128);
2453	}
2454	if (ics->window_sequence[0] != LONG_START_SEQUENCE) {
2455	ac->fdsp->vector_fmul_reverse(in + 1024, in + 1024, lwindow, 1024);
2456	} else {
2457	ac->fdsp->vector_fmul_reverse(in + 1024 + 448, in + 1024 + 448, swindow, 128);
2458	memset(in + 1024 + 576, 0, 448 * sizeof(*in));
2459	}
2460	ac->mdct_ltp.mdct_calc(&ac->mdct_ltp, out, in);
2461	}
2462
2463	/**
2464	* Apply the long term prediction
2465	*/
2466	static void apply_ltp(AACContext *ac, SingleChannelElement *sce)
2467	{
2468	const LongTermPrediction *ltp = &sce->ics.ltp;
2469	const uint16_t *offsets = sce->ics.swb_offset;
2470	int i, sfb;
2471
2472	if (sce->ics.window_sequence[0] != EIGHT_SHORT_SEQUENCE) {
2473	INTFLOAT *predTime = sce->ret;
2474	INTFLOAT *predFreq = ac->buf_mdct;
2475	int16_t num_samples = 2048;
2476
2477	if (ltp->lag < 1024)
2478	num_samples = ltp->lag + 1024;
2479	for (i = 0; i < num_samples; i++)
2480	predTime[i] = AAC_MUL30(sce->ltp_state[i + 2048 - ltp->lag], ltp->coef);
2481	memset(&predTime[i], 0, (2048 - i) * sizeof(*predTime));
2482
2483	ac->windowing_and_mdct_ltp(ac, predFreq, predTime, &sce->ics);
2484
2485	if (sce->tns.present)
2486	ac->apply_tns(predFreq, &sce->tns, &sce->ics, 0);
2487
2488	for (sfb = 0; sfb < FFMIN(sce->ics.max_sfb, MAX_LTP_LONG_SFB); sfb++)
2489	if (ltp->used[sfb])
2490	for (i = offsets[sfb]; i < offsets[sfb + 1]; i++)
2491	sce->coeffs[i] += predFreq[i];
2492	}
2493	}
2494
2495	/**
2496	* Update the LTP buffer for next frame
2497	*/
2498	static void update_ltp(AACContext *ac, SingleChannelElement *sce)
2499	{
2500	IndividualChannelStream *ics = &sce->ics;
2501	INTFLOAT *saved = sce->saved;
2502	INTFLOAT *saved_ltp = sce->coeffs;
2503	const INTFLOAT *lwindow = ics->use_kb_window[0] ? AAC_RENAME(ff_aac_kbd_long_1024) : AAC_RENAME(ff_sine_1024);
2504	const INTFLOAT *swindow = ics->use_kb_window[0] ? AAC_RENAME(ff_aac_kbd_short_128) : AAC_RENAME(ff_sine_128);
2505	int i;
2506
2507	if (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) {
2508	memcpy(saved_ltp, saved, 512 * sizeof(*saved_ltp));
2509	memset(saved_ltp + 576, 0, 448 * sizeof(*saved_ltp));
2510	ac->fdsp->vector_fmul_reverse(saved_ltp + 448, ac->buf_mdct + 960, &swindow[64], 64);
2511
2512	for (i = 0; i < 64; i++)
2513	saved_ltp[i + 512] = AAC_MUL31(ac->buf_mdct[1023 - i], swindow[63 - i]);
2514	} else if (ics->window_sequence[0] == LONG_START_SEQUENCE) {
2515	memcpy(saved_ltp, ac->buf_mdct + 512, 448 * sizeof(*saved_ltp));
2516	memset(saved_ltp + 576, 0, 448 * sizeof(*saved_ltp));
2517	ac->fdsp->vector_fmul_reverse(saved_ltp + 448, ac->buf_mdct + 960, &swindow[64], 64);
2518
2519	for (i = 0; i < 64; i++)
2520	saved_ltp[i + 512] = AAC_MUL31(ac->buf_mdct[1023 - i], swindow[63 - i]);
2521	} else { // LONG_STOP or ONLY_LONG
2522	ac->fdsp->vector_fmul_reverse(saved_ltp, ac->buf_mdct + 512, &lwindow[512], 512);
2523
2524	for (i = 0; i < 512; i++)
2525	saved_ltp[i + 512] = AAC_MUL31(ac->buf_mdct[1023 - i], lwindow[511 - i]);
2526	}
2527
2528	memcpy(sce->ltp_state, sce->ltp_state+1024, 1024 * sizeof(*sce->ltp_state));
2529	memcpy(sce->ltp_state+1024, sce->ret, 1024 * sizeof(*sce->ltp_state));
2530	memcpy(sce->ltp_state+2048, saved_ltp, 1024 * sizeof(*sce->ltp_state));
2531	}
2532
2533	/**
2534	* Conduct IMDCT and windowing.
2535	*/
2536	static void imdct_and_windowing(AACContext *ac, SingleChannelElement *sce)
2537	{
2538	IndividualChannelStream *ics = &sce->ics;
2539	INTFLOAT *in = sce->coeffs;
2540	INTFLOAT *out = sce->ret;
2541	INTFLOAT *saved = sce->saved;
2542	const INTFLOAT *swindow = ics->use_kb_window[0] ? AAC_RENAME(ff_aac_kbd_short_128) : AAC_RENAME(ff_sine_128);
2543	const INTFLOAT *lwindow_prev = ics->use_kb_window[1] ? AAC_RENAME(ff_aac_kbd_long_1024) : AAC_RENAME(ff_sine_1024);
2544	const INTFLOAT *swindow_prev = ics->use_kb_window[1] ? AAC_RENAME(ff_aac_kbd_short_128) : AAC_RENAME(ff_sine_128);
2545	INTFLOAT *buf = ac->buf_mdct;
2546	INTFLOAT *temp = ac->temp;
2547	int i;
2548
2549	// imdct
2550	if (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) {
2551	for (i = 0; i < 1024; i += 128)
2552	ac->mdct_small.imdct_half(&ac->mdct_small, buf + i, in + i);
2553	} else {
2554	ac->mdct.imdct_half(&ac->mdct, buf, in);
2555	#if USE_FIXED
2556	for (i=0; i<1024; i++)
2557	buf[i] = (buf[i] + 4) >> 3;
2558	#endif /* USE_FIXED */
2559	}
2560
2561	/* window overlapping
2562	* NOTE: To simplify the overlapping code, all 'meaningless' short to long
2563	* and long to short transitions are considered to be short to short
2564	* transitions. This leaves just two cases (long to long and short to short)
2565	* with a little special sauce for EIGHT_SHORT_SEQUENCE.
2566	*/
2567	if ((ics->window_sequence[1] == ONLY_LONG_SEQUENCE || ics->window_sequence[1] == LONG_STOP_SEQUENCE) &&
2568	(ics->window_sequence[0] == ONLY_LONG_SEQUENCE || ics->window_sequence[0] == LONG_START_SEQUENCE)) {
2569	ac->fdsp->vector_fmul_window( out, saved, buf, lwindow_prev, 512);
2570	} else {
2571	memcpy( out, saved, 448 * sizeof(*out));
2572
2573	if (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) {
2574	ac->fdsp->vector_fmul_window(out + 448 + 0*128, saved + 448, buf + 0*128, swindow_prev, 64);
2575	ac->fdsp->vector_fmul_window(out + 448 + 1*128, buf + 0*128 + 64, buf + 1*128, swindow, 64);
2576	ac->fdsp->vector_fmul_window(out + 448 + 2*128, buf + 1*128 + 64, buf + 2*128, swindow, 64);
2577	ac->fdsp->vector_fmul_window(out + 448 + 3*128, buf + 2*128 + 64, buf + 3*128, swindow, 64);
2578	ac->fdsp->vector_fmul_window(temp, buf + 3*128 + 64, buf + 4*128, swindow, 64);
2579	memcpy( out + 448 + 4*128, temp, 64 * sizeof(*out));
2580	} else {
2581	ac->fdsp->vector_fmul_window(out + 448, saved + 448, buf, swindow_prev, 64);
2582	memcpy( out + 576, buf + 64, 448 * sizeof(*out));
2583	}
2584	}
2585
2586	// buffer update
2587	if (ics->window_sequence[0] == EIGHT_SHORT_SEQUENCE) {
2588	memcpy( saved, temp + 64, 64 * sizeof(*saved));
2589	ac->fdsp->vector_fmul_window(saved + 64, buf + 4*128 + 64, buf + 5*128, swindow, 64);
2590	ac->fdsp->vector_fmul_window(saved + 192, buf + 5*128 + 64, buf + 6*128, swindow, 64);
2591	ac->fdsp->vector_fmul_window(saved + 320, buf + 6*128 + 64, buf + 7*128, swindow, 64);
2592	memcpy( saved + 448, buf + 7*128 + 64, 64 * sizeof(*saved));
2593	} else if (ics->window_sequence[0] == LONG_START_SEQUENCE) {
2594	memcpy( saved, buf + 512, 448 * sizeof(*saved));
2595	memcpy( saved + 448, buf + 7*128 + 64, 64 * sizeof(*saved));
2596	} else { // LONG_STOP or ONLY_LONG
2597	memcpy( saved, buf + 512, 512 * sizeof(*saved));
2598	}
2599	}
2600
2601	static void imdct_and_windowing_ld(AACContext *ac, SingleChannelElement *sce)
2602	{
2603	IndividualChannelStream *ics = &sce->ics;
2604	INTFLOAT *in = sce->coeffs;
2605	INTFLOAT *out = sce->ret;
2606	INTFLOAT *saved = sce->saved;
2607	INTFLOAT *buf = ac->buf_mdct;
2608	#if USE_FIXED
2609	int i;
2610	#endif /* USE_FIXED */
2611
2612	// imdct
2613	ac->mdct.imdct_half(&ac->mdct_ld, buf, in);
2614
2615	#if USE_FIXED
2616	for (i = 0; i < 1024; i++)
2617	buf[i] = (buf[i] + 2) >> 2;
2618	#endif /* USE_FIXED */
2619
2620	// window overlapping
2621	if (ics->use_kb_window[1]) {
2622	// AAC LD uses a low overlap sine window instead of a KBD window
2623	memcpy(out, saved, 192 * sizeof(*out));
2624	ac->fdsp->vector_fmul_window(out + 192, saved + 192, buf, AAC_RENAME(ff_sine_128), 64);
2625	memcpy( out + 320, buf + 64, 192 * sizeof(*out));
2626	} else {
2627	ac->fdsp->vector_fmul_window(out, saved, buf, AAC_RENAME(ff_sine_512), 256);
2628	}
2629
2630	// buffer update
2631	memcpy(saved, buf + 256, 256 * sizeof(*saved));
2632	}
2633
2634	static void imdct_and_windowing_eld(AACContext *ac, SingleChannelElement *sce)
2635	{
2636	INTFLOAT *in = sce->coeffs;
2637	INTFLOAT *out = sce->ret;
2638	INTFLOAT *saved = sce->saved;
2639	INTFLOAT *buf = ac->buf_mdct;
2640	int i;
2641	const int n = ac->oc[1].m4ac.frame_length_short ? 480 : 512;
2642	const int n2 = n >> 1;
2643	const int n4 = n >> 2;
2644	const INTFLOAT *const window = n == 480 ? AAC_RENAME(ff_aac_eld_window_480) :
2645	AAC_RENAME(ff_aac_eld_window_512);
2646
2647	// Inverse transform, mapped to the conventional IMDCT by
2648	// Chivukula, R.K.; Reznik, Y.A.; Devarajan, V.,
2649	// "Efficient algorithms for MPEG-4 AAC-ELD, AAC-LD and AAC-LC filterbanks,"
2650	// International Conference on Audio, Language and Image Processing, ICALIP 2008.
2651	// URL: http://ieeexplore.ieee.org/stamp/stamp.jsp?tp=&arnumber=4590245&isnumber=4589950
2652	for (i = 0; i < n2; i+=2) {
2653	INTFLOAT temp;
2654	temp = in[i ]; in[i ] = -in[n - 1 - i]; in[n - 1 - i] = temp;
2655	temp = -in[i + 1]; in[i + 1] = in[n - 2 - i]; in[n - 2 - i] = temp;
2656	}
2657	#if !USE_FIXED
2658	if (n == 480)
2659	ac->mdct480->imdct_half(ac->mdct480, buf, in, 1, -1.f/(16*1024*960));
2660	else
2661	#endif
2662	ac->mdct.imdct_half(&ac->mdct_ld, buf, in);
2663
2664	#if USE_FIXED
2665	for (i = 0; i < 1024; i++)
2666	buf[i] = (buf[i] + 1) >> 1;
2667	#endif /* USE_FIXED */
2668
2669	for (i = 0; i < n; i+=2) {
2670	buf[i] = -buf[i];
2671	}
2672	// Like with the regular IMDCT at this point we still have the middle half
2673	// of a transform but with even symmetry on the left and odd symmetry on
2674	// the right
2675
2676	// window overlapping
2677	// The spec says to use samples [0..511] but the reference decoder uses
2678	// samples [128..639].
2679	for (i = n4; i < n2; i ++) {
2680	out[i - n4] = AAC_MUL31( buf[ n2 - 1 - i] , window[i - n4]) +
2681	AAC_MUL31( saved[ i + n2] , window[i + n - n4]) +
2682	AAC_MUL31(-saved[n + n2 - 1 - i] , window[i + 2*n - n4]) +
2683	AAC_MUL31(-saved[ 2*n + n2 + i] , window[i + 3*n - n4]);
2684	}
2685	for (i = 0; i < n2; i ++) {
2686	out[n4 + i] = AAC_MUL31( buf[ i] , window[i + n2 - n4]) +
2687	AAC_MUL31(-saved[ n - 1 - i] , window[i + n2 + n - n4]) +
2688	AAC_MUL31(-saved[ n + i] , window[i + n2 + 2*n - n4]) +
2689	AAC_MUL31( saved[2*n + n - 1 - i] , window[i + n2 + 3*n - n4]);
2690	}
2691	for (i = 0; i < n4; i ++) {
2692	out[n2 + n4 + i] = AAC_MUL31( buf[ i + n2] , window[i + n - n4]) +
2693	AAC_MUL31(-saved[n2 - 1 - i] , window[i + 2*n - n4]) +
2694	AAC_MUL31(-saved[n + n2 + i] , window[i + 3*n - n4]);
2695	}
2696
2697	// buffer update
2698	memmove(saved + n, saved, 2 * n * sizeof(*saved));
2699	memcpy( saved, buf, n * sizeof(*saved));
2700	}
2701
2702	/**
2703	* channel coupling transformation interface
2704	*
2705	* @param apply_coupling_method pointer to (in)dependent coupling function
2706	*/
2707	static void apply_channel_coupling(AACContext *ac, ChannelElement *cc,
2708	enum RawDataBlockType type, int elem_id,
2709	enum CouplingPoint coupling_point,
2710	void (*apply_coupling_method)(AACContext *ac, SingleChannelElement *target, ChannelElement *cce, int index))
2711	{
2712	int i, c;
2713
2714	for (i = 0; i < MAX_ELEM_ID; i++) {
2715	ChannelElement *cce = ac->che[TYPE_CCE][i];
2716	int index = 0;
2717
2718	if (cce && cce->coup.coupling_point == coupling_point) {
2719	ChannelCoupling *coup = &cce->coup;
2720
2721	for (c = 0; c <= coup->num_coupled; c++) {
2722	if (coup->type[c] == type && coup->id_select[c] == elem_id) {
2723	if (coup->ch_select[c] != 1) {
2724	apply_coupling_method(ac, &cc->ch[0], cce, index);
2725	if (coup->ch_select[c] != 0)
2726	index++;
2727	}
2728	if (coup->ch_select[c] != 2)
2729	apply_coupling_method(ac, &cc->ch[1], cce, index++);
2730	} else
2731	index += 1 + (coup->ch_select[c] == 3);
2732	}
2733	}
2734	}
2735	}
2736
2737	/**
2738	* Convert spectral data to samples, applying all supported tools as appropriate.
2739	*/
2740	static void spectral_to_sample(AACContext *ac, int samples)
2741	{
2742	int i, type;
2743	void (*imdct_and_window)(AACContext *ac, SingleChannelElement *sce);
2744	switch (ac->oc[1].m4ac.object_type) {
2745	case AOT_ER_AAC_LD:
2746	imdct_and_window = imdct_and_windowing_ld;
2747	break;
2748	case AOT_ER_AAC_ELD:
2749	imdct_and_window = imdct_and_windowing_eld;
2750	break;
2751	default:
2752	imdct_and_window = ac->imdct_and_windowing;
2753	}
2754	for (type = 3; type >= 0; type--) {
2755	for (i = 0; i < MAX_ELEM_ID; i++) {
2756	ChannelElement *che = ac->che[type][i];
2757	if (che && che->present) {
2758	if (type <= TYPE_CPE)
2759	apply_channel_coupling(ac, che, type, i, BEFORE_TNS, AAC_RENAME(apply_dependent_coupling));
2760	if (ac->oc[1].m4ac.object_type == AOT_AAC_LTP) {
2761	if (che->ch[0].ics.predictor_present) {
2762	if (che->ch[0].ics.ltp.present)
2763	ac->apply_ltp(ac, &che->ch[0]);
2764	if (che->ch[1].ics.ltp.present && type == TYPE_CPE)
2765	ac->apply_ltp(ac, &che->ch[1]);
2766	}
2767	}
2768	if (che->ch[0].tns.present)
2769	ac->apply_tns(che->ch[0].coeffs, &che->ch[0].tns, &che->ch[0].ics, 1);
2770	if (che->ch[1].tns.present)
2771	ac->apply_tns(che->ch[1].coeffs, &che->ch[1].tns, &che->ch[1].ics, 1);
2772	if (type <= TYPE_CPE)
2773	apply_channel_coupling(ac, che, type, i, BETWEEN_TNS_AND_IMDCT, AAC_RENAME(apply_dependent_coupling));
2774	if (type != TYPE_CCE || che->coup.coupling_point == AFTER_IMDCT) {
2775	imdct_and_window(ac, &che->ch[0]);
2776	if (ac->oc[1].m4ac.object_type == AOT_AAC_LTP)
2777	ac->update_ltp(ac, &che->ch[0]);
2778	if (type == TYPE_CPE) {
2779	imdct_and_window(ac, &che->ch[1]);
2780	if (ac->oc[1].m4ac.object_type == AOT_AAC_LTP)
2781	ac->update_ltp(ac, &che->ch[1]);
2782	}
2783	if (ac->oc[1].m4ac.sbr > 0) {
2784	AAC_RENAME(ff_sbr_apply)(ac, &che->sbr, type, che->ch[0].ret, che->ch[1].ret);
2785	}
2786	}
2787	if (type <= TYPE_CCE)
2788	apply_channel_coupling(ac, che, type, i, AFTER_IMDCT, AAC_RENAME(apply_independent_coupling));
2789
2790	#if USE_FIXED
2791	{
2792	int j;
2793	/* preparation for resampler */
2794	for(j = 0; j<samples; j++){
2795	che->ch[0].ret[j] = (int32_t)av_clipl_int32((int64_t)che->ch[0].ret[j]<<7)+0x8000;
2796	if(type == TYPE_CPE)
2797	che->ch[1].ret[j] = (int32_t)av_clipl_int32((int64_t)che->ch[1].ret[j]<<7)+0x8000;
2798	}
2799	}
2800	#endif /* USE_FIXED */
2801	che->present = 0;
2802	} else if (che) {
2803	av_log(ac->avctx, AV_LOG_VERBOSE, "ChannelElement %d.%d missing \n", type, i);
2804	}
2805	}
2806	}
2807	}
2808
2809	static int parse_adts_frame_header(AACContext *ac, GetBitContext *gb)
2810	{
2811	int size;
2812	AACADTSHeaderInfo hdr_info;
2813	uint8_t layout_map[MAX_ELEM_ID*4][3];
2814	int layout_map_tags, ret;
2815
2816	size = avpriv_aac_parse_header(gb, &hdr_info);
2817	if (size > 0) {
2818	if (!ac->warned_num_aac_frames && hdr_info.num_aac_frames != 1) {
2819	// This is 2 for "VLB " audio in NSV files.
2820	// See samples/nsv/vlb_audio.
2821	avpriv_report_missing_feature(ac->avctx,
2822	"More than one AAC RDB per ADTS frame");
2823	ac->warned_num_aac_frames = 1;
2824	}
2825	push_output_configuration(ac);
2826	if (hdr_info.chan_config) {
2827	ac->oc[1].m4ac.chan_config = hdr_info.chan_config;
2828	if ((ret = set_default_channel_config(ac->avctx,
2829	layout_map,
2830	&layout_map_tags,
2831	hdr_info.chan_config)) < 0)
2832	return ret;
2833	if ((ret = output_configure(ac, layout_map, layout_map_tags,
2834	FFMAX(ac->oc[1].status,
2835	OC_TRIAL_FRAME), 0)) < 0)
2836	return ret;
2837	} else {
2838	ac->oc[1].m4ac.chan_config = 0;
2839	/**
2840	* dual mono frames in Japanese DTV can have chan_config 0
2841	* WITHOUT specifying PCE.
2842	* thus, set dual mono as default.
2843	*/
2844	if (ac->dmono_mode && ac->oc[0].status == OC_NONE) {
2845	layout_map_tags = 2;
2846	layout_map[0][0] = layout_map[1][0] = TYPE_SCE;
2847	layout_map[0][2] = layout_map[1][2] = AAC_CHANNEL_FRONT;
2848	layout_map[0][1] = 0;
2849	layout_map[1][1] = 1;
2850	if (output_configure(ac, layout_map, layout_map_tags,
2851	OC_TRIAL_FRAME, 0))
2852	return -7;
2853	}
2854	}
2855	ac->oc[1].m4ac.sample_rate = hdr_info.sample_rate;
2856	ac->oc[1].m4ac.sampling_index = hdr_info.sampling_index;
2857	ac->oc[1].m4ac.object_type = hdr_info.object_type;
2858	ac->oc[1].m4ac.frame_length_short = 0;
2859	if (ac->oc[0].status != OC_LOCKED ||
2860	ac->oc[0].m4ac.chan_config != hdr_info.chan_config ||
2861	ac->oc[0].m4ac.sample_rate != hdr_info.sample_rate) {
2862	ac->oc[1].m4ac.sbr = -1;
2863	ac->oc[1].m4ac.ps = -1;
2864	}
2865	if (!hdr_info.crc_absent)
2866	skip_bits(gb, 16);
2867	}
2868	return size;
2869	}
2870
2871	static int aac_decode_er_frame(AVCodecContext *avctx, void *data,
2872	int *got_frame_ptr, GetBitContext *gb)
2873	{
2874	AACContext *ac = avctx->priv_data;
2875	const MPEG4AudioConfig *const m4ac = &ac->oc[1].m4ac;
2876	ChannelElement *che;
2877	int err, i;
2878	int samples = m4ac->frame_length_short ? 960 : 1024;
2879	int chan_config = m4ac->chan_config;
2880	int aot = m4ac->object_type;
2881
2882	if (aot == AOT_ER_AAC_LD || aot == AOT_ER_AAC_ELD)
2883	samples >>= 1;
2884
2885	ac->frame = data;
2886
2887	if ((err = frame_configure_elements(avctx)) < 0)
2888	return err;
2889
2890	// The FF_PROFILE_AAC_* defines are all object_type - 1
2891	// This may lead to an undefined profile being signaled
2892	ac->avctx->profile = aot - 1;
2893
2894	ac->tags_mapped = 0;
2895
2896	if (chan_config < 0 || (chan_config >= 8 && chan_config < 11) || chan_config >= 13) {
2897	avpriv_request_sample(avctx, "Unknown ER channel configuration %d",
2898	chan_config);
2899	return AVERROR_INVALIDDATA;
2900	}
2901	for (i = 0; i < tags_per_config[chan_config]; i++) {
2902	const int elem_type = aac_channel_layout_map[chan_config-1][i][0];
2903	const int elem_id = aac_channel_layout_map[chan_config-1][i][1];
2904	if (!(che=get_che(ac, elem_type, elem_id))) {
2905	av_log(ac->avctx, AV_LOG_ERROR,
2906	"channel element %d.%d is not allocated\n",
2907	elem_type, elem_id);
2908	return AVERROR_INVALIDDATA;
2909	}
2910	che->present = 1;
2911	if (aot != AOT_ER_AAC_ELD)
2912	skip_bits(gb, 4);
2913	switch (elem_type) {
2914	case TYPE_SCE:
2915	err = decode_ics(ac, &che->ch[0], gb, 0, 0);
2916	break;
2917	case TYPE_CPE:
2918	err = decode_cpe(ac, gb, che);
2919	break;
2920	case TYPE_LFE:
2921	err = decode_ics(ac, &che->ch[0], gb, 0, 0);
2922	break;
2923	}
2924	if (err < 0)
2925	return err;
2926	}
2927
2928	spectral_to_sample(ac, samples);
2929
2930	if (!ac->frame->data[0] && samples) {
2931	av_log(avctx, AV_LOG_ERROR, "no frame data found\n");
2932	return AVERROR_INVALIDDATA;
2933	}
2934
2935	ac->frame->nb_samples = samples;
2936	ac->frame->sample_rate = avctx->sample_rate;
2937	*got_frame_ptr = 1;
2938
2939	skip_bits_long(gb, get_bits_left(gb));
2940	return 0;
2941	}
2942
2943	static int aac_decode_frame_int(AVCodecContext *avctx, void *data,
2944	int *got_frame_ptr, GetBitContext *gb, AVPacket *avpkt)
2945	{
2946	AACContext *ac = avctx->priv_data;
2947	ChannelElement *che = NULL, *che_prev = NULL;
2948	enum RawDataBlockType elem_type, che_prev_type = TYPE_END;
2949	int err, elem_id;
2950	int samples = 0, multiplier, audio_found = 0, pce_found = 0;
2951	int is_dmono, sce_count = 0;
2952	int payload_alignment;
2953
2954	ac->frame = data;
2955
2956	if (show_bits(gb, 12) == 0xfff) {
2957	if ((err = parse_adts_frame_header(ac, gb)) < 0) {
2958	av_log(avctx, AV_LOG_ERROR, "Error decoding AAC frame header.\n");
2959	goto fail;
2960	}
2961	if (ac->oc[1].m4ac.sampling_index > 12) {
2962	av_log(ac->avctx, AV_LOG_ERROR, "invalid sampling rate index %d\n", ac->oc[1].m4ac.sampling_index);
2963	err = AVERROR_INVALIDDATA;
2964	goto fail;
2965	}
2966	}
2967
2968	if ((err = frame_configure_elements(avctx)) < 0)
2969	goto fail;
2970
2971	// The FF_PROFILE_AAC_* defines are all object_type - 1
2972	// This may lead to an undefined profile being signaled
2973	ac->avctx->profile = ac->oc[1].m4ac.object_type - 1;
2974
2975	payload_alignment = get_bits_count(gb);
2976	ac->tags_mapped = 0;
2977	// parse
2978	while ((elem_type = get_bits(gb, 3)) != TYPE_END) {
2979	elem_id = get_bits(gb, 4);
2980
2981	if (avctx->debug & FF_DEBUG_STARTCODE)
2982	av_log(avctx, AV_LOG_DEBUG, "Elem type:%x id:%x\n", elem_type, elem_id);
2983
2984	if (!avctx->channels && elem_type != TYPE_PCE) {
2985	err = AVERROR_INVALIDDATA;
2986	goto fail;
2987	}
2988
2989	if (elem_type < TYPE_DSE) {
2990	if (!(che=get_che(ac, elem_type, elem_id))) {
2991	av_log(ac->avctx, AV_LOG_ERROR, "channel element %d.%d is not allocated\n",
2992	elem_type, elem_id);
2993	err = AVERROR_INVALIDDATA;
2994	goto fail;
2995	}
2996	samples = 1024;
2997	che->present = 1;
2998	}
2999
3000	switch (elem_type) {
3001
3002	case TYPE_SCE:
3003	err = decode_ics(ac, &che->ch[0], gb, 0, 0);
3004	audio_found = 1;
3005	sce_count++;
3006	break;
3007
3008	case TYPE_CPE:
3009	err = decode_cpe(ac, gb, che);
3010	audio_found = 1;
3011	break;
3012
3013	case TYPE_CCE:
3014	err = decode_cce(ac, gb, che);
3015	break;
3016
3017	case TYPE_LFE:
3018	err = decode_ics(ac, &che->ch[0], gb, 0, 0);
3019	audio_found = 1;
3020	break;
3021
3022	case TYPE_DSE:
3023	err = skip_data_stream_element(ac, gb);
3024	break;
3025
3026	case TYPE_PCE: {
3027	uint8_t layout_map[MAX_ELEM_ID*4][3];
3028	int tags;
3029	push_output_configuration(ac);
3030	tags = decode_pce(avctx, &ac->oc[1].m4ac, layout_map, gb,
3031	payload_alignment);
3032	if (tags < 0) {
3033	err = tags;
3034	break;
3035	}
3036	if (pce_found) {
3037	av_log(avctx, AV_LOG_ERROR,
3038	"Not evaluating a further program_config_element as this construct is dubious at best.\n");
3039	pop_output_configuration(ac);
3040	} else {
3041	err = output_configure(ac, layout_map, tags, OC_TRIAL_PCE, 1);
3042	if (!err)
3043	ac->oc[1].m4ac.chan_config = 0;
3044	pce_found = 1;
3045	}
3046	break;
3047	}
3048
3049	case TYPE_FIL:
3050	if (elem_id == 15)
3051	elem_id += get_bits(gb, 8) - 1;
3052	if (get_bits_left(gb) < 8 * elem_id) {
3053	av_log(avctx, AV_LOG_ERROR, "TYPE_FIL: "overread_err);
3054	err = AVERROR_INVALIDDATA;
3055	goto fail;
3056	}
3057	while (elem_id > 0)
3058	elem_id -= decode_extension_payload(ac, gb, elem_id, che_prev, che_prev_type);
3059	err = 0; /* FIXME */
3060	break;
3061
3062	default:
3063	err = AVERROR_BUG; /* should not happen, but keeps compiler happy */
3064	break;
3065	}
3066
3067	if (elem_type < TYPE_DSE) {
3068	che_prev = che;
3069	che_prev_type = elem_type;
3070	}
3071
3072	if (err)
3073	goto fail;
3074
3075	if (get_bits_left(gb) < 3) {
3076	av_log(avctx, AV_LOG_ERROR, overread_err);
3077	err = AVERROR_INVALIDDATA;
3078	goto fail;
3079	}
3080	}
3081
3082	if (!avctx->channels) {
3083	*got_frame_ptr = 0;
3084	return 0;
3085	}
3086
3087	multiplier = (ac->oc[1].m4ac.sbr == 1) ? ac->oc[1].m4ac.ext_sample_rate > ac->oc[1].m4ac.sample_rate : 0;
3088	samples <<= multiplier;
3089
3090	spectral_to_sample(ac, samples);
3091
3092	if (ac->oc[1].status && audio_found) {
3093	avctx->sample_rate = ac->oc[1].m4ac.sample_rate << multiplier;
3094	avctx->frame_size = samples;
3095	ac->oc[1].status = OC_LOCKED;
3096	}
3097
3098	if (multiplier)
3099	avctx->internal->skip_samples_multiplier = 2;
3100
3101	if (!ac->frame->data[0] && samples) {
3102	av_log(avctx, AV_LOG_ERROR, "no frame data found\n");
3103	err = AVERROR_INVALIDDATA;
3104	goto fail;
3105	}
3106
3107	if (samples) {
3108	ac->frame->nb_samples = samples;
3109	ac->frame->sample_rate = avctx->sample_rate;
3110	} else
3111	av_frame_unref(ac->frame);
3112	*got_frame_ptr = !!samples;
3113
3114	/* for dual-mono audio (SCE + SCE) */
3115	is_dmono = ac->dmono_mode && sce_count == 2 &&
3116	ac->oc[1].channel_layout == (AV_CH_FRONT_LEFT | AV_CH_FRONT_RIGHT);
3117	if (is_dmono) {
3118	if (ac->dmono_mode == 1)
3119	((AVFrame *)data)->data[1] =((AVFrame *)data)->data[0];
3120	else if (ac->dmono_mode == 2)
3121	((AVFrame *)data)->data[0] =((AVFrame *)data)->data[1];
3122	}
3123
3124	return 0;
3125	fail:
3126	pop_output_configuration(ac);
3127	return err;
3128	}
3129
3130	static int aac_decode_frame(AVCodecContext *avctx, void *data,
3131	int *got_frame_ptr, AVPacket *avpkt)
3132	{
3133	AACContext *ac = avctx->priv_data;
3134	const uint8_t *buf = avpkt->data;
3135	int buf_size = avpkt->size;
3136	GetBitContext gb;
3137	int buf_consumed;
3138	int buf_offset;
3139	int err;
3140	int new_extradata_size;
3141	const uint8_t *new_extradata = av_packet_get_side_data(avpkt,
3142	AV_PKT_DATA_NEW_EXTRADATA,
3143	&new_extradata_size);
3144	int jp_dualmono_size;
3145	const uint8_t *jp_dualmono = av_packet_get_side_data(avpkt,
3146	AV_PKT_DATA_JP_DUALMONO,
3147	&jp_dualmono_size);
3148
3149	if (new_extradata && 0) {
3150	av_free(avctx->extradata);
3151	avctx->extradata = av_mallocz(new_extradata_size +
3152	AV_INPUT_BUFFER_PADDING_SIZE);
3153	if (!avctx->extradata)
3154	return AVERROR(ENOMEM);
3155	avctx->extradata_size = new_extradata_size;
3156	memcpy(avctx->extradata, new_extradata, new_extradata_size);
3157	push_output_configuration(ac);
3158	if (decode_audio_specific_config(ac, ac->avctx, &ac->oc[1].m4ac,
3159	avctx->extradata,
3160	avctx->extradata_size*8LL, 1) < 0) {
3161	pop_output_configuration(ac);
3162	return AVERROR_INVALIDDATA;
3163	}
3164	}
3165
3166	ac->dmono_mode = 0;
3167	if (jp_dualmono && jp_dualmono_size > 0)
3168	ac->dmono_mode = 1 + *jp_dualmono;
3169	if (ac->force_dmono_mode >= 0)
3170	ac->dmono_mode = ac->force_dmono_mode;
3171
3172	if (INT_MAX / 8 <= buf_size)
3173	return AVERROR_INVALIDDATA;
3174
3175	if ((err = init_get_bits8(&gb, buf, buf_size)) < 0)
3176	return err;
3177
3178	switch (ac->oc[1].m4ac.object_type) {
3179	case AOT_ER_AAC_LC:
3180	case AOT_ER_AAC_LTP:
3181	case AOT_ER_AAC_LD:
3182	case AOT_ER_AAC_ELD:
3183	err = aac_decode_er_frame(avctx, data, got_frame_ptr, &gb);
3184	break;
3185	default:
3186	err = aac_decode_frame_int(avctx, data, got_frame_ptr, &gb, avpkt);
3187	}
3188	if (err < 0)
3189	return err;
3190
3191	buf_consumed = (get_bits_count(&gb) + 7) >> 3;
3192	for (buf_offset = buf_consumed; buf_offset < buf_size; buf_offset++)
3193	if (buf[buf_offset])
3194	break;
3195
3196	return buf_size > buf_offset ? buf_consumed : buf_size;
3197	}
3198
3199	static av_cold int aac_decode_close(AVCodecContext *avctx)
3200	{
3201	AACContext *ac = avctx->priv_data;
3202	int i, type;
3203
3204	for (i = 0; i < MAX_ELEM_ID; i++) {
3205	for (type = 0; type < 4; type++) {
3206	if (ac->che[type][i])
3207	AAC_RENAME(ff_aac_sbr_ctx_close)(&ac->che[type][i]->sbr);
3208	av_freep(&ac->che[type][i]);
3209	}
3210	}
3211
3212	ff_mdct_end(&ac->mdct);
3213	ff_mdct_end(&ac->mdct_small);
3214	ff_mdct_end(&ac->mdct_ld);
3215	ff_mdct_end(&ac->mdct_ltp);
3216	#if !USE_FIXED
3217	ff_mdct15_uninit(&ac->mdct480);
3218	#endif
3219	av_freep(&ac->fdsp);
3220	return 0;
3221	}
3222
3223	static void aacdec_init(AACContext *c)
3224	{
3225	c->imdct_and_windowing = imdct_and_windowing;
3226	c->apply_ltp = apply_ltp;
3227	c->apply_tns = apply_tns;
3228	c->windowing_and_mdct_ltp = windowing_and_mdct_ltp;
3229	c->update_ltp = update_ltp;
3230	#if USE_FIXED
3231	c->vector_pow43 = vector_pow43;
3232	c->subband_scale = subband_scale;
3233	#endif
3234
3235	#if !USE_FIXED
3236	if(ARCH_MIPS)
3237	ff_aacdec_init_mips(c);
3238	#endif /* !USE_FIXED */
3239	}
3240	/**
3241	* AVOptions for Japanese DTV specific extensions (ADTS only)
3242	*/
3243	#define AACDEC_FLAGS AV_OPT_FLAG_DECODING_PARAM | AV_OPT_FLAG_AUDIO_PARAM
3244	static const AVOption options[] = {
3245	{"dual_mono_mode", "Select the channel to decode for dual mono",
3246	offsetof(AACContext, force_dmono_mode), AV_OPT_TYPE_INT, {.i64=-1}, -1, 2,
3247	AACDEC_FLAGS, "dual_mono_mode"},
3248
3249	{"auto", "autoselection", 0, AV_OPT_TYPE_CONST, {.i64=-1}, INT_MIN, INT_MAX, AACDEC_FLAGS, "dual_mono_mode"},
3250	{"main", "Select Main/Left channel", 0, AV_OPT_TYPE_CONST, {.i64= 1}, INT_MIN, INT_MAX, AACDEC_FLAGS, "dual_mono_mode"},
3251	{"sub" , "Select Sub/Right channel", 0, AV_OPT_TYPE_CONST, {.i64= 2}, INT_MIN, INT_MAX, AACDEC_FLAGS, "dual_mono_mode"},
3252	{"both", "Select both channels", 0, AV_OPT_TYPE_CONST, {.i64= 0}, INT_MIN, INT_MAX, AACDEC_FLAGS, "dual_mono_mode"},
3253
3254	{NULL},
3255	};
3256
3257	static const AVClass aac_decoder_class = {
3258	.class_name = "AAC decoder",
3259	.item_name = av_default_item_name,
3260	.option = options,
3261	.version = LIBAVUTIL_VERSION_INT,
3262	};
3263