summaryrefslogtreecommitdiffstats
path: root/libavcodec/dca_core.c
blob: 46825edbd4198ca27e73b02dc509e768e6baa333 (plain)
1
2
3
4
5
6
7
8
9
10
11
12
13
14
15
16
17
18
19
20
21
22
23
24
25
26
27
28
29
30
31
32
33
34
35
36
37
38
39
40
41
42
43
44
45
46
47
48
49
50
51
52
53
54
55
56
57
58
59
60
61
62
63
64
65
66
67
68
69
70
71
72
73
74
75
76
77
78
79
80
81
82
83
84
85
86
87
88
89
90
91
92
93
94
95
96
97
98
99
100
101
102
103
104
105
106
107
108
109
110
111
112
113
114
115
116
117
118
119
120
121
122
123
124
125
126
127
128
129
130
131
132
133
134
135
136
137
138
139
140
141
142
143
144
145
146
147
148
149
150
151
152
153
154
155
156
157
158
159
160
161
162
163
164
165
166
167
168
169
170
171
172
173
174
175
176
177
178
179
180
181
182
183
184
185
186
187
188
189
190
191
192
193
194
195
196
197
198
199
200
201
202
203
204
205
206
207
208
209
210
211
212
213
214
215
216
217
218
219
220
221
222
223
224
225
226
227
228
229
230
231
232
233
234
235
236
237
238
239
240
241
242
243
244
245
246
247
248
249
250
251
252
253
254
255
256
257
258
259
260
261
262
263
264
265
266
267
268
269
270
271
272
273
274
275
276
277
278
279
280
281
282
283
284
285
286
287
288
289
290
291
292
293
294
295
296
297
298
299
300
301
302
303
304
305
306
307
308
309
310
311
312
313
314
315
316
317
318
319
320
321
322
323
324
325
326
327
328
329
330
331
332
333
334
335
336
337
338
339
340
341
342
343
344
345
346
347
348
349
350
351
352
353
354
355
356
357
358
359
360
361
362
363
364
365
366
367
368
369
370
371
372
373
374
375
376
377
378
379
380
381
382
383
384
385
386
387
388
389
390
391
392
393
394
395
396
397
398
399
400
401
402
403
404
405
406
407
408
409
410
411
412
413
414
415
416
417
418
419
420
421
422
423
424
425
426
427
428
429
430
431
432
433
434
435
436
437
438
439
440
441
442
443
444
445
446
447
448
449
450
451
452
453
454
455
456
457
458
459
460
461
462
463
464
465
466
467
468
469
470
471
472
473
474
475
476
477
478
479
480
481
482
483
484
485
486
487
488
489
490
491
492
493
494
495
496
497
498
499
500
501
502
503
504
505
506
507
508
509
510
511
512
513
514
515
516
517
518
519
520
521
522
523
524
525
526
527
528
529
530
531
532
533
534
535
536
537
538
539
540
541
542
543
544
545
546
547
548
549
550
551
552
553
554
555
556
557
558
559
560
561
562
563
564
565
566
567
568
569
570
571
572
573
574
575
576
577
578
579
580
581
582
583
584
585
586
587
588
589
590
591
592
593
594
595
596
597
598
599
600
601
602
603
604
605
606
607
608
609
610
611
612
613
614
615
616
617
618
619
620
621
622
623
624
625
626
627
628
629
630
631
632
633
634
635
636
637
638
639
640
641
642
643
644
645
646
647
648
649
650
651
652
653
654
655
656
657
658
659
660
661
662
663
664
665
666
667
668
669
670
671
672
673
674
675
676
677
678
679
680
681
682
683
684
685
686
687
688
689
690
691
692
693
694
695
696
697
698
699
700
701
702
703
704
705
706
707
708
709
710
711
712
713
714
715
716
717
718
719
720
721
722
723
724
725
726
727
728
729
730
731
732
733
734
735
736
737
738
739
740
741
742
743
744
745
746
747
748
749
750
751
752
753
754
755
756
757
758
759
760
761
762
763
764
765
766
767
768
769
770
771
772
773
774
775
776
777
778
779
780
781
782
783
784
785
786
787
788
789
790
791
792
793
794
795
796
797
798
799
800
801
802
803
804
805
806
807
808
809
810
811
812
813
814
815
816
817
818
819
820
821
822
823
824
825
826
827
828
829
830
831
832
833
834
835
836
837
838
839
840
841
842
843
844
845
846
847
848
849
850
851
852
853
854
855
856
857
858
859
860
861
862
863
864
865
866
867
868
869
870
871
872
873
874
875
876
877
878
879
880
881
882
883
884
885
886
887
888
889
890
891
892
893
894
895
896
897
898
899
900
901
902
903
904
905
906
907
908
909
910
911
912
913
914
915
916
917
918
919
920
921
922
923
924
925
926
927
928
929
930
931
932
933
934
935
936
937
938
939
940
941
942
943
944
945
946
947
948
949
950
951
952
953
954
955
956
957
958
959
960
961
962
963
964
965
966
967
968
969
970
971
972
973
974
975
976
977
978
979
980
981
982
983
984
985
986
987
988
989
990
991
992
993
994
995
996
997
998
999
1000
1001
1002
1003
1004
1005
1006
1007
1008
1009
1010
1011
1012
1013
1014
1015
1016
1017
1018
1019
1020
1021
1022
1023
1024
1025
1026
1027
1028
1029
1030
1031
1032
1033
1034
1035
1036
1037
1038
1039
1040
1041
1042
1043
1044
1045
1046
1047
1048
1049
1050
1051
1052
1053
1054
1055
1056
1057
1058
1059
1060
1061
1062
1063
1064
1065
1066
1067
1068
1069
1070
1071
1072
1073
1074
1075
1076
1077
1078
1079
1080
1081
1082
1083
1084
1085
1086
1087
1088
1089
1090
1091
1092
1093
1094
1095
1096
1097
1098
1099
1100
1101
1102
1103
1104
1105
1106
1107
1108
1109
1110
1111
1112
1113
1114
1115
1116
1117
1118
1119
1120
1121
1122
1123
1124
1125
1126
1127
1128
1129
1130
1131
1132
1133
1134
1135
1136
1137
1138
1139
1140
1141
1142
1143
1144
1145
1146
1147
1148
1149
1150
1151
1152
1153
1154
1155
1156
1157
1158
1159
1160
1161
1162
1163
1164
1165
1166
1167
1168
1169
1170
1171
1172
1173
1174
1175
1176
1177
1178
1179
1180
1181
1182
1183
1184
1185
1186
1187
1188
1189
1190
1191
1192
1193
1194
1195
1196
1197
1198
1199
1200
1201
1202
1203
1204
1205
1206
1207
1208
1209
1210
1211
1212
1213
1214
1215
1216
1217
1218
1219
1220
1221
1222
1223
1224
1225
1226
1227
1228
1229
1230
1231
1232
1233
1234
1235
1236
1237
1238
1239
1240
1241
1242
1243
1244
1245
1246
1247
1248
1249
1250
1251
1252
1253
1254
1255
1256
1257
1258
1259
1260
1261
1262
1263
1264
1265
1266
1267
1268
1269
1270
1271
1272
1273
1274
1275
1276
1277
1278
1279
1280
1281
1282
1283
1284
1285
1286
1287
1288
1289
1290
1291
1292
1293
1294
1295
1296
1297
1298
1299
1300
1301
1302
1303
1304
1305
1306
1307
1308
1309
1310
1311
1312
1313
1314
1315
1316
1317
1318
1319
1320
1321
1322
1323
1324
1325
1326
1327
1328
1329
1330
1331
1332
1333
1334
1335
1336
1337
1338
1339
1340
1341
1342
1343
1344
1345
1346
1347
1348
1349
1350
1351
1352
1353
1354
1355
1356
1357
1358
1359
1360
1361
1362
1363
1364
1365
1366
1367
1368
1369
1370
1371
1372
1373
1374
1375
1376
1377
1378
1379
1380
1381
1382
1383
1384
1385
1386
1387
1388
1389
1390
1391
1392
1393
1394
1395
1396
1397
1398
1399
1400
1401
1402
1403
1404
1405
1406
1407
1408
1409
1410
1411
1412
1413
1414
1415
1416
1417
1418
1419
1420
1421
1422
1423
1424
1425
1426
1427
1428
1429
1430
1431
1432
1433
1434
1435
1436
1437
1438
1439
1440
1441
1442
1443
1444
1445
1446
1447
1448
1449
1450
1451
1452
1453
1454
1455
1456
1457
1458
1459
1460
1461
1462
1463
1464
1465
1466
1467
1468
1469
1470
1471
1472
1473
1474
1475
1476
1477
1478
1479
1480
1481
1482
1483
1484
1485
1486
1487
1488
1489
1490
1491
1492
1493
1494
1495
1496
1497
1498
1499
1500
1501
1502
1503
1504
1505
1506
1507
1508
1509
1510
1511
1512
1513
1514
1515
1516
1517
1518
1519
1520
1521
1522
1523
1524
1525
1526
1527
1528
1529
1530
1531
1532
1533
1534
1535
1536
1537
1538
1539
1540
1541
1542
1543
1544
1545
1546
1547
1548
1549
1550
1551
1552
1553
1554
1555
1556
1557
1558
1559
1560
1561
1562
1563
1564
1565
1566
1567
1568
1569
1570
1571
1572
1573
1574
1575
1576
1577
1578
1579
1580
1581
1582
1583
1584
1585
1586
1587
1588
1589
1590
1591
1592
1593
1594
1595
1596
1597
1598
1599
1600
1601
1602
1603
1604
1605
1606
1607
1608
1609
1610
1611
1612
1613
1614
1615
1616
1617
1618
1619
1620
1621
1622
1623
1624
1625
1626
1627
1628
1629
1630
1631
1632
1633
1634
1635
1636
1637
1638
1639
1640
1641
1642
1643
1644
1645
1646
1647
1648
1649
1650
1651
1652
1653
1654
1655
1656
1657
1658
1659
1660
1661
1662
1663
1664
1665
1666
1667
1668
1669
1670
1671
1672
1673
1674
1675
1676
1677
1678
1679
1680
1681
1682
1683
1684
1685
1686
1687
1688
1689
1690
1691
1692
1693
1694
1695
1696
1697
1698
1699
1700
1701
1702
1703
1704
1705
1706
1707
1708
1709
1710
1711
1712
1713
1714
1715
1716
1717
1718
1719
1720
1721
1722
1723
1724
1725
1726
1727
1728
1729
1730
1731
1732
1733
1734
1735
1736
1737
1738
1739
1740
1741
1742
1743
1744
1745
1746
1747
1748
1749
1750
1751
1752
1753
1754
1755
1756
1757
1758
1759
1760
1761
1762
1763
1764
1765
1766
1767
1768
1769
1770
1771
1772
1773
1774
1775
1776
1777
1778
1779
1780
1781
1782
1783
1784
1785
1786
1787
1788
1789
1790
1791
1792
1793
1794
1795
1796
1797
1798
1799
1800
1801
1802
1803
1804
1805
1806
1807
1808
1809
1810
1811
1812
1813
1814
1815
1816
1817
1818
1819
1820
1821
1822
1823
1824
1825
1826
1827
1828
1829
1830
1831
1832
1833
1834
1835
1836
1837
1838
1839
1840
1841
1842
1843
1844
1845
1846
1847
1848
1849
1850
1851
1852
1853
1854
1855
1856
1857
1858
1859
1860
1861
1862
1863
1864
1865
1866
1867
1868
1869
1870
1871
1872
1873
1874
1875
1876
1877
1878
1879
1880
1881
1882
1883
1884
1885
1886
1887
1888
1889
1890
1891
1892
1893
1894
1895
1896
1897
1898
1899
1900
1901
1902
1903
1904
1905
1906
1907
1908
1909
1910
1911
1912
1913
1914
1915
1916
1917
1918
1919
1920
1921
1922
1923
1924
1925
1926
1927
1928
1929
1930
1931
1932
1933
1934
1935
1936
1937
1938
1939
1940
1941
1942
1943
1944
1945
1946
1947
1948
1949
1950
1951
1952
1953
1954
1955
1956
1957
1958
1959
1960
1961
1962
1963
1964
1965
1966
1967
1968
1969
1970
1971
1972
1973
1974
1975
1976
1977
1978
1979
1980
1981
1982
1983
1984
1985
1986
1987
1988
1989
1990
1991
1992
1993
1994
1995
1996
1997
1998
1999
2000
2001
2002
2003
2004
2005
2006
2007
2008
2009
2010
2011
2012
2013
2014
2015
2016
2017
2018
2019
2020
2021
2022
2023
2024
2025
2026
2027
2028
2029
2030
2031
2032
2033
2034
2035
2036
2037
2038
2039
2040
2041
2042
2043
2044
2045
2046
2047
2048
2049
2050
2051
2052
2053
2054
2055
2056
2057
2058
2059
2060
2061
2062
2063
2064
2065
2066
2067
2068
2069
2070
2071
2072
2073
2074
2075
2076
2077
2078
2079
2080
2081
2082
2083
2084
2085
2086
2087
2088
2089
2090
2091
2092
2093
2094
2095
2096
2097
2098
2099
2100
2101
2102
2103
2104
2105
2106
2107
2108
2109
2110
2111
2112
2113
2114
2115
2116
2117
2118
2119
2120
2121
2122
2123
2124
2125
2126
2127
2128
2129
2130
2131
2132
2133
2134
2135
2136
2137
2138
2139
2140
2141
2142
2143
2144
2145
2146
2147
2148
2149
2150
2151
2152
2153
2154
2155
2156
2157
2158
2159
2160
2161
2162
2163
2164
2165
2166
2167
2168
2169
2170
2171
2172
2173
2174
2175
2176
2177
2178
2179
2180
2181
2182
2183
2184
2185
2186
2187
2188
2189
2190
2191
2192
2193
2194
2195
2196
2197
2198
2199
2200
2201
2202
2203
2204
2205
2206
2207
2208
2209
2210
2211
2212
2213
2214
2215
2216
2217
2218
2219
2220
2221
2222
2223
2224
2225
2226
2227
2228
2229
2230
2231
2232
2233
2234
2235
2236
2237
2238
2239
2240
2241
2242
2243
2244
2245
2246
2247
2248
2249
2250
2251
2252
2253
2254
2255
2256
2257
2258
2259
2260
2261
2262
2263
2264
2265
2266
2267
2268
2269
2270
2271
2272
2273
2274
2275
2276
2277
2278
2279
2280
2281
2282
2283
2284
2285
2286
2287
2288
2289
2290
2291
2292
2293
2294
2295
2296
2297
2298
2299
2300
2301
2302
2303
2304
2305
2306
2307
2308
2309
2310
2311
2312
2313
2314
2315
2316
2317
2318
2319
2320
2321
2322
2323
2324
2325
2326
2327
2328
2329
2330
2331
2332
2333
2334
2335
2336
2337
2338
2339
2340
2341
2342
2343
2344
2345
2346
2347
2348
2349
2350
2351
2352
2353
2354
2355
2356
2357
2358
2359
2360
2361
2362
2363
2364
2365
2366
2367
2368
2369
2370
2371
2372
2373
2374
2375
2376
2377
2378
2379
2380
2381
2382
2383
2384
2385
2386
2387
2388
2389
2390
2391
2392
2393
2394
2395
2396
2397
2398
2399
2400
2401
2402
2403
2404
2405
2406
2407
2408
2409
2410
2411
2412
2413
2414
2415
2416
2417
2418
2419
2420
2421
2422
2423
2424
2425
2426
2427
2428
2429
2430
2431
2432
2433
2434
2435
2436
2437
2438
2439
2440
2441
2442
2443
2444
2445
2446
2447
2448
2449
2450
2451
2452
2453
2454
2455
2456
2457
2458
2459
2460
2461
2462
2463
2464
2465
2466
2467
2468
2469
2470
2471
2472
2473
2474
2475
2476
2477
2478
2479
2480
2481
2482
2483
2484
2485
2486
2487
2488
2489
2490
2491
2492
2493
2494
2495
2496
2497
2498
2499
2500
2501
2502
2503
2504
2505
2506
2507
2508
2509
2510
2511
2512
2513
2514
2515
2516
2517
2518
2519
2520
2521
2522
2523
2524
2525
2526
2527
2528
2529
2530
2531
2532
2533
2534
2535
2536
2537
2538
2539
2540
2541
2542
2543
2544
2545
2546
2547
2548
2549
2550
/*
 * Copyright (C) 2016 foo86
 *
 * This file is part of FFmpeg.
 *
 * FFmpeg is free software; you can redistribute it and/or
 * modify it under the terms of the GNU Lesser General Public
 * License as published by the Free Software Foundation; either
 * version 2.1 of the License, or (at your option) any later version.
 *
 * FFmpeg is distributed in the hope that it will be useful,
 * but WITHOUT ANY WARRANTY; without even the implied warranty of
 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
 * Lesser General Public License for more details.
 *
 * You should have received a copy of the GNU Lesser General Public
 * License along with FFmpeg; if not, write to the Free Software
 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
 */

#include "dcadec.h"
#include "dcadata.h"
#include "dcahuff.h"
#include "dcamath.h"
#include "dca_syncwords.h"

#if ARCH_ARM
#include "arm/dca.h"
#endif

enum HeaderType {
    HEADER_CORE,
    HEADER_XCH,
    HEADER_XXCH
};

enum AudioMode {
    AMODE_MONO,             // Mode 0: A (mono)
    AMODE_MONO_DUAL,        // Mode 1: A + B (dual mono)
    AMODE_STEREO,           // Mode 2: L + R (stereo)
    AMODE_STEREO_SUMDIFF,   // Mode 3: (L+R) + (L-R) (sum-diff)
    AMODE_STEREO_TOTAL,     // Mode 4: LT + RT (left and right total)
    AMODE_3F,               // Mode 5: C + L + R
    AMODE_2F1R,             // Mode 6: L + R + S
    AMODE_3F1R,             // Mode 7: C + L + R + S
    AMODE_2F2R,             // Mode 8: L + R + SL + SR
    AMODE_3F2R,             // Mode 9: C + L + R + SL + SR

    AMODE_COUNT
};

enum ExtAudioType {
    EXT_AUDIO_XCH   = 0,
    EXT_AUDIO_X96   = 2,
    EXT_AUDIO_XXCH  = 6
};

enum LFEFlag {
    LFE_FLAG_NONE,
    LFE_FLAG_128,
    LFE_FLAG_64,
    LFE_FLAG_INVALID
};

static const int8_t prm_ch_to_spkr_map[AMODE_COUNT][5] = {
    { DCA_SPEAKER_C,            -1,             -1,             -1,             -1 },
    { DCA_SPEAKER_L, DCA_SPEAKER_R,             -1,             -1,             -1 },
    { DCA_SPEAKER_L, DCA_SPEAKER_R,             -1,             -1,             -1 },
    { DCA_SPEAKER_L, DCA_SPEAKER_R,             -1,             -1,             -1 },
    { DCA_SPEAKER_L, DCA_SPEAKER_R,             -1,             -1,             -1 },
    { DCA_SPEAKER_C, DCA_SPEAKER_L, DCA_SPEAKER_R ,             -1,             -1 },
    { DCA_SPEAKER_L, DCA_SPEAKER_R, DCA_SPEAKER_Cs,             -1,             -1 },
    { DCA_SPEAKER_C, DCA_SPEAKER_L, DCA_SPEAKER_R , DCA_SPEAKER_Cs,             -1 },
    { DCA_SPEAKER_L, DCA_SPEAKER_R, DCA_SPEAKER_Ls, DCA_SPEAKER_Rs,             -1 },
    { DCA_SPEAKER_C, DCA_SPEAKER_L, DCA_SPEAKER_R,  DCA_SPEAKER_Ls, DCA_SPEAKER_Rs }
};

static const uint8_t audio_mode_ch_mask[AMODE_COUNT] = {
    DCA_SPEAKER_LAYOUT_MONO,
    DCA_SPEAKER_LAYOUT_STEREO,
    DCA_SPEAKER_LAYOUT_STEREO,
    DCA_SPEAKER_LAYOUT_STEREO,
    DCA_SPEAKER_LAYOUT_STEREO,
    DCA_SPEAKER_LAYOUT_3_0,
    DCA_SPEAKER_LAYOUT_2_1,
    DCA_SPEAKER_LAYOUT_3_1,
    DCA_SPEAKER_LAYOUT_2_2,
    DCA_SPEAKER_LAYOUT_5POINT0
};

static const uint8_t block_code_nbits[7] = {
    7, 10, 12, 13, 15, 17, 19
};

static const uint8_t quant_index_sel_nbits[DCA_CODE_BOOKS] = {
    1, 2, 2, 2, 2, 3, 3, 3, 3, 3
};

static const uint8_t quant_index_group_size[DCA_CODE_BOOKS] = {
    1, 3, 3, 3, 3, 7, 7, 7, 7, 7
};

static int dca_get_vlc(GetBitContext *s, DCAVLC *v, int i)
{
    return get_vlc2(s, v->vlc[i].table, v->vlc[i].bits, v->max_depth) + v->offset;
}

static void get_array(GetBitContext *s, int32_t *array, int size, int n)
{
    int i;

    for (i = 0; i < size; i++)
        array[i] = get_sbits(s, n);
}

// 5.3.1 - Bit stream header
static int parse_frame_header(DCACoreDecoder *s)
{
    int normal_frame, pcmr_index;

    // Frame type
    normal_frame = get_bits1(&s->gb);

    // Deficit sample count
    if (get_bits(&s->gb, 5) != DCA_PCMBLOCK_SAMPLES - 1) {
        av_log(s->avctx, AV_LOG_ERROR, "Deficit samples are not supported\n");
        return normal_frame ? AVERROR_INVALIDDATA : AVERROR_PATCHWELCOME;
    }

    // CRC present flag
    s->crc_present = get_bits1(&s->gb);

    // Number of PCM sample blocks
    s->npcmblocks = get_bits(&s->gb, 7) + 1;
    if (s->npcmblocks & (DCA_SUBBAND_SAMPLES - 1)) {
        av_log(s->avctx, AV_LOG_ERROR, "Unsupported number of PCM sample blocks (%d)\n", s->npcmblocks);
        return (s->npcmblocks < 6 || normal_frame) ? AVERROR_INVALIDDATA : AVERROR_PATCHWELCOME;
    }

    // Primary frame byte size
    s->frame_size = get_bits(&s->gb, 14) + 1;
    if (s->frame_size < 96) {
        av_log(s->avctx, AV_LOG_ERROR, "Invalid core frame size (%d bytes)\n", s->frame_size);
        return AVERROR_INVALIDDATA;
    }

    // Audio channel arrangement
    s->audio_mode = get_bits(&s->gb, 6);
    if (s->audio_mode >= AMODE_COUNT) {
        av_log(s->avctx, AV_LOG_ERROR, "Unsupported audio channel arrangement (%d)\n", s->audio_mode);
        return AVERROR_PATCHWELCOME;
    }

    // Core audio sampling frequency
    s->sample_rate = avpriv_dca_sample_rates[get_bits(&s->gb, 4)];
    if (!s->sample_rate) {
        av_log(s->avctx, AV_LOG_ERROR, "Invalid core audio sampling frequency\n");
        return AVERROR_INVALIDDATA;
    }

    // Transmission bit rate
    s->bit_rate = ff_dca_bit_rates[get_bits(&s->gb, 5)];

    // Reserved field
    skip_bits1(&s->gb);

    // Embedded dynamic range flag
    s->drc_present = get_bits1(&s->gb);

    // Embedded time stamp flag
    s->ts_present = get_bits1(&s->gb);

    // Auxiliary data flag
    s->aux_present = get_bits1(&s->gb);

    // HDCD mastering flag
    skip_bits1(&s->gb);

    // Extension audio descriptor flag
    s->ext_audio_type = get_bits(&s->gb, 3);

    // Extended coding flag
    s->ext_audio_present = get_bits1(&s->gb);

    // Audio sync word insertion flag
    s->sync_ssf = get_bits1(&s->gb);

    // Low frequency effects flag
    s->lfe_present = get_bits(&s->gb, 2);
    if (s->lfe_present == LFE_FLAG_INVALID) {
        av_log(s->avctx, AV_LOG_ERROR, "Invalid low frequency effects flag\n");
        return AVERROR_INVALIDDATA;
    }

    // Predictor history flag switch
    s->predictor_history = get_bits1(&s->gb);

    // Header CRC check bytes
    if (s->crc_present)
        skip_bits(&s->gb, 16);

    // Multirate interpolator switch
    s->filter_perfect = get_bits1(&s->gb);

    // Encoder software revision
    skip_bits(&s->gb, 4);

    // Copy history
    skip_bits(&s->gb, 2);

    // Source PCM resolution
    s->source_pcm_res = ff_dca_bits_per_sample[pcmr_index = get_bits(&s->gb, 3)];
    if (!s->source_pcm_res) {
        av_log(s->avctx, AV_LOG_ERROR, "Invalid source PCM resolution\n");
        return AVERROR_INVALIDDATA;
    }
    s->es_format = pcmr_index & 1;

    // Front sum/difference flag
    s->sumdiff_front = get_bits1(&s->gb);

    // Surround sum/difference flag
    s->sumdiff_surround = get_bits1(&s->gb);

    // Dialog normalization / unspecified
    skip_bits(&s->gb, 4);

    return 0;
}

// 5.3.2 - Primary audio coding header
static int parse_coding_header(DCACoreDecoder *s, enum HeaderType header, int xch_base)
{
    int n, ch, nchannels, header_size = 0, header_pos = get_bits_count(&s->gb);
    unsigned int mask, index;

    if (get_bits_left(&s->gb) < 0)
        return AVERROR_INVALIDDATA;

    switch (header) {
    case HEADER_CORE:
        // Number of subframes
        s->nsubframes = get_bits(&s->gb, 4) + 1;

        // Number of primary audio channels
        s->nchannels = get_bits(&s->gb, 3) + 1;
        if (s->nchannels != ff_dca_channels[s->audio_mode]) {
            av_log(s->avctx, AV_LOG_ERROR, "Invalid number of primary audio channels (%d) for audio channel arrangement (%d)\n", s->nchannels, s->audio_mode);
            return AVERROR_INVALIDDATA;
        }
        av_assert1(s->nchannels <= DCA_CHANNELS - 2);

        s->ch_mask = audio_mode_ch_mask[s->audio_mode];

        // Add LFE channel if present
        if (s->lfe_present)
            s->ch_mask |= DCA_SPEAKER_MASK_LFE1;
        break;

    case HEADER_XCH:
        s->nchannels = ff_dca_channels[s->audio_mode] + 1;
        av_assert1(s->nchannels <= DCA_CHANNELS - 1);
        s->ch_mask |= DCA_SPEAKER_MASK_Cs;
        break;

    case HEADER_XXCH:
        // Channel set header length
        header_size = get_bits(&s->gb, 7) + 1;

        // Check CRC
        if (s->xxch_crc_present
            && ff_dca_check_crc(s->avctx, &s->gb, header_pos, header_pos + header_size * 8)) {
            av_log(s->avctx, AV_LOG_ERROR, "Invalid XXCH channel set header checksum\n");
            return AVERROR_INVALIDDATA;
        }

        // Number of channels in a channel set
        nchannels = get_bits(&s->gb, 3) + 1;
        if (nchannels > DCA_XXCH_CHANNELS_MAX) {
            avpriv_request_sample(s->avctx, "%d XXCH channels", nchannels);
            return AVERROR_PATCHWELCOME;
        }
        s->nchannels = ff_dca_channels[s->audio_mode] + nchannels;
        av_assert1(s->nchannels <= DCA_CHANNELS);

        // Loudspeaker layout mask
        mask = get_bits_long(&s->gb, s->xxch_mask_nbits - DCA_SPEAKER_Cs);
        s->xxch_spkr_mask = mask << DCA_SPEAKER_Cs;

        if (av_popcount(s->xxch_spkr_mask) != nchannels) {
            av_log(s->avctx, AV_LOG_ERROR, "Invalid XXCH speaker layout mask (%#x)\n", s->xxch_spkr_mask);
            return AVERROR_INVALIDDATA;
        }

        if (s->xxch_core_mask & s->xxch_spkr_mask) {
            av_log(s->avctx, AV_LOG_ERROR, "XXCH speaker layout mask (%#x) overlaps with core (%#x)\n", s->xxch_spkr_mask, s->xxch_core_mask);
            return AVERROR_INVALIDDATA;
        }

        // Combine core and XXCH masks together
        s->ch_mask = s->xxch_core_mask | s->xxch_spkr_mask;

        // Downmix coefficients present in stream
        if (get_bits1(&s->gb)) {
            int *coeff_ptr = s->xxch_dmix_coeff;

            // Downmix already performed by encoder
            s->xxch_dmix_embedded = get_bits1(&s->gb);

            // Downmix scale factor
            index = get_bits(&s->gb, 6) * 4 - FF_DCA_DMIXTABLE_OFFSET - 3;
            if (index >= FF_DCA_INV_DMIXTABLE_SIZE) {
                av_log(s->avctx, AV_LOG_ERROR, "Invalid XXCH downmix scale index (%d)\n", index);
                return AVERROR_INVALIDDATA;
            }
            s->xxch_dmix_scale_inv = ff_dca_inv_dmixtable[index];

            // Downmix channel mapping mask
            for (ch = 0; ch < nchannels; ch++) {
                mask = get_bits_long(&s->gb, s->xxch_mask_nbits);
                if ((mask & s->xxch_core_mask) != mask) {
                    av_log(s->avctx, AV_LOG_ERROR, "Invalid XXCH downmix channel mapping mask (%#x)\n", mask);
                    return AVERROR_INVALIDDATA;
                }
                s->xxch_dmix_mask[ch] = mask;
            }

            // Downmix coefficients
            for (ch = 0; ch < nchannels; ch++) {
                for (n = 0; n < s->xxch_mask_nbits; n++) {
                    if (s->xxch_dmix_mask[ch] & (1U << n)) {
                        int code = get_bits(&s->gb, 7);
                        int sign = (code >> 6) - 1;
                        if (code &= 63) {
                            index = code * 4 - 3;
                            if (index >= FF_DCA_DMIXTABLE_SIZE) {
                                av_log(s->avctx, AV_LOG_ERROR, "Invalid XXCH downmix coefficient index (%d)\n", index);
                                return AVERROR_INVALIDDATA;
                            }
                            *coeff_ptr++ = (ff_dca_dmixtable[index] ^ sign) - sign;
                        } else {
                            *coeff_ptr++ = 0;
                        }
                    }
                }
            }
        } else {
            s->xxch_dmix_embedded = 0;
        }

        break;
    }

    // Subband activity count
    for (ch = xch_base; ch < s->nchannels; ch++) {
        s->nsubbands[ch] = get_bits(&s->gb, 5) + 2;
        if (s->nsubbands[ch] > DCA_SUBBANDS) {
            av_log(s->avctx, AV_LOG_ERROR, "Invalid subband activity count\n");
            return AVERROR_INVALIDDATA;
        }
    }

    // High frequency VQ start subband
    for (ch = xch_base; ch < s->nchannels; ch++)
        s->subband_vq_start[ch] = get_bits(&s->gb, 5) + 1;

    // Joint intensity coding index
    for (ch = xch_base; ch < s->nchannels; ch++) {
        if ((n = get_bits(&s->gb, 3)) && header == HEADER_XXCH)
            n += xch_base - 1;
        if (n > s->nchannels) {
            av_log(s->avctx, AV_LOG_ERROR, "Invalid joint intensity coding index\n");
            return AVERROR_INVALIDDATA;
        }
        s->joint_intensity_index[ch] = n;
    }

    // Transient mode code book
    for (ch = xch_base; ch < s->nchannels; ch++)
        s->transition_mode_sel[ch] = get_bits(&s->gb, 2);

    // Scale factor code book
    for (ch = xch_base; ch < s->nchannels; ch++) {
        s->scale_factor_sel[ch] = get_bits(&s->gb, 3);
        if (s->scale_factor_sel[ch] == 7) {
            av_log(s->avctx, AV_LOG_ERROR, "Invalid scale factor code book\n");
            return AVERROR_INVALIDDATA;
        }
    }

    // Bit allocation quantizer select
    for (ch = xch_base; ch < s->nchannels; ch++) {
        s->bit_allocation_sel[ch] = get_bits(&s->gb, 3);
        if (s->bit_allocation_sel[ch] == 7) {
            av_log(s->avctx, AV_LOG_ERROR, "Invalid bit allocation quantizer select\n");
            return AVERROR_INVALIDDATA;
        }
    }

    // Quantization index codebook select
    for (n = 0; n < DCA_CODE_BOOKS; n++)
        for (ch = xch_base; ch < s->nchannels; ch++)
            s->quant_index_sel[ch][n] = get_bits(&s->gb, quant_index_sel_nbits[n]);

    // Scale factor adjustment index
    for (n = 0; n < DCA_CODE_BOOKS; n++)
        for (ch = xch_base; ch < s->nchannels; ch++)
            if (s->quant_index_sel[ch][n] < quant_index_group_size[n])
                s->scale_factor_adj[ch][n] = ff_dca_scale_factor_adj[get_bits(&s->gb, 2)];

    if (header == HEADER_XXCH) {
        // Reserved
        // Byte align
        // CRC16 of channel set header
        if (ff_dca_seek_bits(&s->gb, header_pos + header_size * 8)) {
            av_log(s->avctx, AV_LOG_ERROR, "Read past end of XXCH channel set header\n");
            return AVERROR_INVALIDDATA;
        }
    } else {
        // Audio header CRC check word
        if (s->crc_present)
            skip_bits(&s->gb, 16);
    }

    return 0;
}

static inline int parse_scale(DCACoreDecoder *s, int *scale_index, int sel)
{
    const uint32_t *scale_table;
    unsigned int scale_size;

    // Select the root square table
    if (sel > 5) {
        scale_table = ff_dca_scale_factor_quant7;
        scale_size = FF_ARRAY_ELEMS(ff_dca_scale_factor_quant7);
    } else {
        scale_table = ff_dca_scale_factor_quant6;
        scale_size = FF_ARRAY_ELEMS(ff_dca_scale_factor_quant6);
    }

    // If Huffman code was used, the difference of scales was encoded
    if (sel < 5)
        *scale_index += dca_get_vlc(&s->gb, &ff_dca_vlc_scale_factor, sel);
    else
        *scale_index = get_bits(&s->gb, sel + 1);

    // Look up scale factor from the root square table
    if ((unsigned int)*scale_index >= scale_size) {
        av_log(s->avctx, AV_LOG_ERROR, "Invalid scale factor index\n");
        return AVERROR_INVALIDDATA;
    }

    return scale_table[*scale_index];
}

static inline int parse_joint_scale(DCACoreDecoder *s, int sel)
{
    int scale_index;

    // Absolute value was encoded even when Huffman code was used
    if (sel < 5)
        scale_index = dca_get_vlc(&s->gb, &ff_dca_vlc_scale_factor, sel);
    else
        scale_index = get_bits(&s->gb, sel + 1);

    // Bias by 64
    scale_index += 64;

    // Look up joint scale factor
    if ((unsigned int)scale_index >= FF_ARRAY_ELEMS(ff_dca_joint_scale_factors)) {
        av_log(s->avctx, AV_LOG_ERROR, "Invalid joint scale factor index\n");
        return AVERROR_INVALIDDATA;
    }

    return ff_dca_joint_scale_factors[scale_index];
}

// 5.4.1 - Primary audio coding side information
static int parse_subframe_header(DCACoreDecoder *s, int sf,
                                 enum HeaderType header, int xch_base)
{
    int ch, band, ret;

    if (get_bits_left(&s->gb) < 0)
        return AVERROR_INVALIDDATA;

    if (header == HEADER_CORE) {
        // Subsubframe count
        s->nsubsubframes[sf] = get_bits(&s->gb, 2) + 1;

        // Partial subsubframe sample count
        skip_bits(&s->gb, 3);
    }

    // Prediction mode
    for (ch = xch_base; ch < s->nchannels; ch++)
        for (band = 0; band < s->nsubbands[ch]; band++)
            s->prediction_mode[ch][band] = get_bits1(&s->gb);

    // Prediction coefficients VQ address
    for (ch = xch_base; ch < s->nchannels; ch++)
        for (band = 0; band < s->nsubbands[ch]; band++)
            if (s->prediction_mode[ch][band])
                s->prediction_vq_index[ch][band] = get_bits(&s->gb, 12);

    // Bit allocation index
    for (ch = xch_base; ch < s->nchannels; ch++) {
        int sel = s->bit_allocation_sel[ch];

        for (band = 0; band < s->subband_vq_start[ch]; band++) {
            int abits;

            if (sel < 5)
                abits = dca_get_vlc(&s->gb, &ff_dca_vlc_bit_allocation, sel);
            else
                abits = get_bits(&s->gb, sel - 1);

            if (abits > DCA_ABITS_MAX) {
                av_log(s->avctx, AV_LOG_ERROR, "Invalid bit allocation index\n");
                return AVERROR_INVALIDDATA;
            }

            s->bit_allocation[ch][band] = abits;
        }
    }

    // Transition mode
    for (ch = xch_base; ch < s->nchannels; ch++) {
        // Clear transition mode for all subbands
        memset(s->transition_mode[sf][ch], 0, sizeof(s->transition_mode[0][0]));

        // Transient possible only if more than one subsubframe
        if (s->nsubsubframes[sf] > 1) {
            int sel = s->transition_mode_sel[ch];
            for (band = 0; band < s->subband_vq_start[ch]; band++)
                if (s->bit_allocation[ch][band])
                    s->transition_mode[sf][ch][band] = dca_get_vlc(&s->gb, &ff_dca_vlc_transition_mode, sel);
        }
    }

    // Scale factors
    for (ch = xch_base; ch < s->nchannels; ch++) {
        int sel = s->scale_factor_sel[ch];
        int scale_index = 0;

        // Extract scales for subbands up to VQ
        for (band = 0; band < s->subband_vq_start[ch]; band++) {
            if (s->bit_allocation[ch][band]) {
                if ((ret = parse_scale(s, &scale_index, sel)) < 0)
                    return ret;
                s->scale_factors[ch][band][0] = ret;
                if (s->transition_mode[sf][ch][band]) {
                    if ((ret = parse_scale(s, &scale_index, sel)) < 0)
                        return ret;
                    s->scale_factors[ch][band][1] = ret;
                }
            } else {
                s->scale_factors[ch][band][0] = 0;
            }
        }

        // High frequency VQ subbands
        for (band = s->subband_vq_start[ch]; band < s->nsubbands[ch]; band++) {
            if ((ret = parse_scale(s, &scale_index, sel)) < 0)
                return ret;
            s->scale_factors[ch][band][0] = ret;
        }
    }

    // Joint subband codebook select
    for (ch = xch_base; ch < s->nchannels; ch++) {
        if (s->joint_intensity_index[ch]) {
            s->joint_scale_sel[ch] = get_bits(&s->gb, 3);
            if (s->joint_scale_sel[ch] == 7) {
                av_log(s->avctx, AV_LOG_ERROR, "Invalid joint scale factor code book\n");
                return AVERROR_INVALIDDATA;
            }
        }
    }

    // Scale factors for joint subband coding
    for (ch = xch_base; ch < s->nchannels; ch++) {
        int src_ch = s->joint_intensity_index[ch] - 1;
        if (src_ch >= 0) {
            int sel = s->joint_scale_sel[ch];
            for (band = s->nsubbands[ch]; band < s->nsubbands[src_ch]; band++) {
                if ((ret = parse_joint_scale(s, sel)) < 0)
                    return ret;
                s->joint_scale_factors[ch][band] = ret;
            }
        }
    }

    // Dynamic range coefficient
    if (s->drc_present && header == HEADER_CORE)
        skip_bits(&s->gb, 8);

    // Side information CRC check word
    if (s->crc_present)
        skip_bits(&s->gb, 16);

    return 0;
}

#ifndef decode_blockcodes
static inline int decode_blockcodes(int code1, int code2, int levels, int32_t *audio)
{
    int offset = (levels - 1) / 2;
    int n, div;

    for (n = 0; n < DCA_SUBBAND_SAMPLES / 2; n++) {
        div = FASTDIV(code1, levels);
        audio[n] = code1 - div * levels - offset;
        code1 = div;
    }
    for (; n < DCA_SUBBAND_SAMPLES; n++) {
        div = FASTDIV(code2, levels);
        audio[n] = code2 - div * levels - offset;
        code2 = div;
    }

    return code1 | code2;
}
#endif

static inline int parse_block_codes(DCACoreDecoder *s, int32_t *audio, int abits)
{
    // Extract block code indices from the bit stream
    int code1 = get_bits(&s->gb, block_code_nbits[abits - 1]);
    int code2 = get_bits(&s->gb, block_code_nbits[abits - 1]);
    int levels = ff_dca_quant_levels[abits];

    // Look up samples from the block code book
    if (decode_blockcodes(code1, code2, levels, audio)) {
        av_log(s->avctx, AV_LOG_ERROR, "Failed to decode block code(s)\n");
        return AVERROR_INVALIDDATA;
    }

    return 0;
}

static inline int parse_huffman_codes(DCACoreDecoder *s, int32_t *audio, int abits, int sel)
{
    int i;

    // Extract Huffman codes from the bit stream
    for (i = 0; i < DCA_SUBBAND_SAMPLES; i++)
        audio[i] = dca_get_vlc(&s->gb, &ff_dca_vlc_quant_index[abits - 1], sel);

    return 1;
}

static inline int extract_audio(DCACoreDecoder *s, int32_t *audio, int abits, int ch)
{
    av_assert1(abits >= 0 && abits <= DCA_ABITS_MAX);

    if (abits == 0) {
        // No bits allocated
        memset(audio, 0, DCA_SUBBAND_SAMPLES * sizeof(*audio));
        return 0;
    }

    if (abits <= DCA_CODE_BOOKS) {
        int sel = s->quant_index_sel[ch][abits - 1];
        if (sel < quant_index_group_size[abits - 1]) {
            // Huffman codes
            return parse_huffman_codes(s, audio, abits, sel);
        }
        if (abits <= 7) {
            // Block codes
            return parse_block_codes(s, audio, abits);
        }
    }

    // No further encoding
    get_array(&s->gb, audio, DCA_SUBBAND_SAMPLES, abits - 3);
    return 0;
}

static inline void dequantize(int32_t *output, const int32_t *input,
                              int32_t step_size, int32_t scale, int residual)
{
    // Account for quantizer step size
    int64_t step_scale = (int64_t)step_size * scale;
    int n, shift = 0;

    // Limit scale factor resolution to 22 bits
    if (step_scale > (1 << 23)) {
        shift = av_log2(step_scale >> 23) + 1;
        step_scale >>= shift;
    }

    // Scale the samples
    if (residual) {
        for (n = 0; n < DCA_SUBBAND_SAMPLES; n++)
            output[n] += clip23(norm__(input[n] * step_scale, 22 - shift));
    } else {
        for (n = 0; n < DCA_SUBBAND_SAMPLES; n++)
            output[n]  = clip23(norm__(input[n] * step_scale, 22 - shift));
    }
}

static inline void inverse_adpcm(int32_t **subband_samples,
                                 const int16_t *vq_index,
                                 const int8_t *prediction_mode,
                                 int sb_start, int sb_end,
                                 int ofs, int len)
{
    int i, j, k;

    for (i = sb_start; i < sb_end; i++) {
        if (prediction_mode[i]) {
            const int16_t *coeff = ff_dca_adpcm_vb[vq_index[i]];
            int32_t *ptr = subband_samples[i] + ofs;
            for (j = 0; j < len; j++) {
                int64_t err = 0;
                for (k = 0; k < DCA_ADPCM_COEFFS; k++)
                    err += (int64_t)ptr[j - k - 1] * coeff[k];
                ptr[j] = clip23(ptr[j] + clip23(norm13(err)));
            }
        }
    }
}

// 5.5 - Primary audio data arrays
static int parse_subframe_audio(DCACoreDecoder *s, int sf, enum HeaderType header,
                                int xch_base, int *sub_pos, int *lfe_pos)
{
    int32_t audio[16], scale;
    int n, ssf, ofs, ch, band;

    // Check number of subband samples in this subframe
    int nsamples = s->nsubsubframes[sf] * DCA_SUBBAND_SAMPLES;
    if (*sub_pos + nsamples > s->npcmblocks) {
        av_log(s->avctx, AV_LOG_ERROR, "Subband sample buffer overflow\n");
        return AVERROR_INVALIDDATA;
    }

    if (get_bits_left(&s->gb) < 0)
        return AVERROR_INVALIDDATA;

    // VQ encoded subbands
    for (ch = xch_base; ch < s->nchannels; ch++) {
        int32_t vq_index[DCA_SUBBANDS];

        for (band = s->subband_vq_start[ch]; band < s->nsubbands[ch]; band++)
            // Extract the VQ address from the bit stream
            vq_index[band] = get_bits(&s->gb, 10);

        if (s->subband_vq_start[ch] < s->nsubbands[ch]) {
            s->dcadsp->decode_hf(s->subband_samples[ch], vq_index,
                                 ff_dca_high_freq_vq, s->scale_factors[ch],
                                 s->subband_vq_start[ch], s->nsubbands[ch],
                                 *sub_pos, nsamples);
        }
    }

    // Low frequency effect data
    if (s->lfe_present && header == HEADER_CORE) {
        unsigned int index;

        // Determine number of LFE samples in this subframe
        int nlfesamples = 2 * s->lfe_present * s->nsubsubframes[sf];
        av_assert1((unsigned int)nlfesamples <= FF_ARRAY_ELEMS(audio));

        // Extract LFE samples from the bit stream
        get_array(&s->gb, audio, nlfesamples, 8);

        // Extract scale factor index from the bit stream
        index = get_bits(&s->gb, 8);
        if (index >= FF_ARRAY_ELEMS(ff_dca_scale_factor_quant7)) {
            av_log(s->avctx, AV_LOG_ERROR, "Invalid LFE scale factor index\n");
            return AVERROR_INVALIDDATA;
        }

        // Look up the 7-bit root square quantization table
        scale = ff_dca_scale_factor_quant7[index];

        // Account for quantizer step size which is 0.035
        scale = mul23(4697620 /* 0.035 * (1 << 27) */, scale);

        // Scale and take the LFE samples
        for (n = 0, ofs = *lfe_pos; n < nlfesamples; n++, ofs++)
            s->lfe_samples[ofs] = clip23(audio[n] * scale >> 4);

        // Advance LFE sample pointer for the next subframe
        *lfe_pos = ofs;
    }

    // Audio data
    for (ssf = 0, ofs = *sub_pos; ssf < s->nsubsubframes[sf]; ssf++) {
        for (ch = xch_base; ch < s->nchannels; ch++) {
            if (get_bits_left(&s->gb) < 0)
                return AVERROR_INVALIDDATA;

            // Not high frequency VQ subbands
            for (band = 0; band < s->subband_vq_start[ch]; band++) {
                int ret, trans_ssf, abits = s->bit_allocation[ch][band];
                int32_t step_size;

                // Extract bits from the bit stream
                if ((ret = extract_audio(s, audio, abits, ch)) < 0)
                    return ret;

                // Select quantization step size table and look up
                // quantization step size
                if (s->bit_rate == 3)
                    step_size = ff_dca_lossless_quant[abits];
                else
                    step_size = ff_dca_lossy_quant[abits];

                // Identify transient location
                trans_ssf = s->transition_mode[sf][ch][band];

                // Determine proper scale factor
                if (trans_ssf == 0 || ssf < trans_ssf)
                    scale = s->scale_factors[ch][band][0];
                else
                    scale = s->scale_factors[ch][band][1];

                // Adjust scale factor when SEL indicates Huffman code
                if (ret > 0) {
                    int64_t adj = s->scale_factor_adj[ch][abits - 1];
                    scale = clip23(adj * scale >> 22);
                }

                dequantize(s->subband_samples[ch][band] + ofs,
                           audio, step_size, scale, 0);
            }
        }

        // DSYNC
        if ((ssf == s->nsubsubframes[sf] - 1 || s->sync_ssf) && get_bits(&s->gb, 16) != 0xffff) {
            av_log(s->avctx, AV_LOG_ERROR, "DSYNC check failed\n");
            return AVERROR_INVALIDDATA;
        }

        ofs += DCA_SUBBAND_SAMPLES;
    }

    // Inverse ADPCM
    for (ch = xch_base; ch < s->nchannels; ch++) {
        inverse_adpcm(s->subband_samples[ch], s->prediction_vq_index[ch],
                      s->prediction_mode[ch], 0, s->nsubbands[ch],
                      *sub_pos, nsamples);
    }

    // Joint subband coding
    for (ch = xch_base; ch < s->nchannels; ch++) {
        int src_ch = s->joint_intensity_index[ch] - 1;
        if (src_ch >= 0) {
            s->dcadsp->decode_joint(s->subband_samples[ch], s->subband_samples[src_ch],
                                    s->joint_scale_factors[ch], s->nsubbands[ch],
                                    s->nsubbands[src_ch], *sub_pos, nsamples);
        }
    }

    // Advance subband sample pointer for the next subframe
    *sub_pos = ofs;
    return 0;
}

static void erase_adpcm_history(DCACoreDecoder *s)
{
    int ch, band;

    // Erase ADPCM history from previous frame if
    // predictor history switch was disabled
    for (ch = 0; ch < DCA_CHANNELS; ch++)
        for (band = 0; band < DCA_SUBBANDS; band++)
            AV_ZERO128(s->subband_samples[ch][band] - DCA_ADPCM_COEFFS);

    emms_c();
}

static int alloc_sample_buffer(DCACoreDecoder *s)
{
    int nchsamples = DCA_ADPCM_COEFFS + s->npcmblocks;
    int nframesamples = nchsamples * DCA_CHANNELS * DCA_SUBBANDS;
    int nlfesamples = DCA_LFE_HISTORY + s->npcmblocks / 2;
    unsigned int size = s->subband_size;
    int ch, band;

    // Reallocate subband sample buffer
    av_fast_mallocz(&s->subband_buffer, &s->subband_size,
                    (nframesamples + nlfesamples) * sizeof(int32_t));
    if (!s->subband_buffer)
        return AVERROR(ENOMEM);

    if (size != s->subband_size) {
        for (ch = 0; ch < DCA_CHANNELS; ch++)
            for (band = 0; band < DCA_SUBBANDS; band++)
                s->subband_samples[ch][band] = s->subband_buffer +
                    (ch * DCA_SUBBANDS + band) * nchsamples + DCA_ADPCM_COEFFS;
        s->lfe_samples = s->subband_buffer + nframesamples;
    }

    if (!s->predictor_history)
        erase_adpcm_history(s);

    return 0;
}

static int parse_frame_data(DCACoreDecoder *s, enum HeaderType header, int xch_base)
{
    int sf, ch, ret, band, sub_pos, lfe_pos;

    if ((ret = parse_coding_header(s, header, xch_base)) < 0)
        return ret;

    for (sf = 0, sub_pos = 0, lfe_pos = DCA_LFE_HISTORY; sf < s->nsubframes; sf++) {
        if ((ret = parse_subframe_header(s, sf, header, xch_base)) < 0)
            return ret;
        if ((ret = parse_subframe_audio(s, sf, header, xch_base, &sub_pos, &lfe_pos)) < 0)
            return ret;
    }

    for (ch = xch_base; ch < s->nchannels; ch++) {
        // Determine number of active subbands for this channel
        int nsubbands = s->nsubbands[ch];
        if (s->joint_intensity_index[ch])
            nsubbands = FFMAX(nsubbands, s->nsubbands[s->joint_intensity_index[ch] - 1]);

        // Update history for ADPCM
        for (band = 0; band < nsubbands; band++) {
            int32_t *samples = s->subband_samples[ch][band] - DCA_ADPCM_COEFFS;
            AV_COPY128(samples, samples + s->npcmblocks);
        }

        // Clear inactive subbands
        for (; band < DCA_SUBBANDS; band++) {
            int32_t *samples = s->subband_samples[ch][band] - DCA_ADPCM_COEFFS;
            memset(samples, 0, (DCA_ADPCM_COEFFS + s->npcmblocks) * sizeof(int32_t));
        }
    }

    emms_c();

    return 0;
}

static int parse_xch_frame(DCACoreDecoder *s)
{
    int ret;

    if (s->ch_mask & DCA_SPEAKER_MASK_Cs) {
        av_log(s->avctx, AV_LOG_ERROR, "XCH with Cs speaker already present\n");
        return AVERROR_INVALIDDATA;
    }

    if ((ret = parse_frame_data(s, HEADER_XCH, s->nchannels)) < 0)
        return ret;

    // Seek to the end of core frame, don't trust XCH frame size
    if (ff_dca_seek_bits(&s->gb, s->frame_size * 8)) {
        av_log(s->avctx, AV_LOG_ERROR, "Read past end of XCH frame\n");
        return AVERROR_INVALIDDATA;
    }

    return 0;
}

static int parse_xxch_frame(DCACoreDecoder *s)
{
    int xxch_nchsets, xxch_frame_size;
    int ret, mask, header_size, header_pos = get_bits_count(&s->gb);

    // XXCH sync word
    if (get_bits_long(&s->gb, 32) != DCA_SYNCWORD_XXCH) {
        av_log(s->avctx, AV_LOG_ERROR, "Invalid XXCH sync word\n");
        return AVERROR_INVALIDDATA;
    }

    // XXCH frame header length
    header_size = get_bits(&s->gb, 6) + 1;

    // Check XXCH frame header CRC
    if (ff_dca_check_crc(s->avctx, &s->gb, header_pos + 32, header_pos + header_size * 8)) {
        av_log(s->avctx, AV_LOG_ERROR, "Invalid XXCH frame header checksum\n");
        return AVERROR_INVALIDDATA;
    }

    // CRC presence flag for channel set header
    s->xxch_crc_present = get_bits1(&s->gb);

    // Number of bits for loudspeaker mask
    s->xxch_mask_nbits = get_bits(&s->gb, 5) + 1;
    if (s->xxch_mask_nbits <= DCA_SPEAKER_Cs) {
        av_log(s->avctx, AV_LOG_ERROR, "Invalid number of bits for XXCH speaker mask (%d)\n", s->xxch_mask_nbits);
        return AVERROR_INVALIDDATA;
    }

    // Number of channel sets
    xxch_nchsets = get_bits(&s->gb, 2) + 1;
    if (xxch_nchsets > 1) {
        avpriv_request_sample(s->avctx, "%d XXCH channel sets", xxch_nchsets);
        return AVERROR_PATCHWELCOME;
    }

    // Channel set 0 data byte size
    xxch_frame_size = get_bits(&s->gb, 14) + 1;

    // Core loudspeaker activity mask
    s->xxch_core_mask = get_bits_long(&s->gb, s->xxch_mask_nbits);

    // Validate the core mask
    mask = s->ch_mask;

    if ((mask & DCA_SPEAKER_MASK_Ls) && (s->xxch_core_mask & DCA_SPEAKER_MASK_Lss))
        mask = (mask & ~DCA_SPEAKER_MASK_Ls) | DCA_SPEAKER_MASK_Lss;

    if ((mask & DCA_SPEAKER_MASK_Rs) && (s->xxch_core_mask & DCA_SPEAKER_MASK_Rss))
        mask = (mask & ~DCA_SPEAKER_MASK_Rs) | DCA_SPEAKER_MASK_Rss;

    if (mask != s->xxch_core_mask) {
        av_log(s->avctx, AV_LOG_ERROR, "XXCH core speaker activity mask (%#x) disagrees with core (%#x)\n", s->xxch_core_mask, mask);
        return AVERROR_INVALIDDATA;
    }

    // Reserved
    // Byte align
    // CRC16 of XXCH frame header
    if (ff_dca_seek_bits(&s->gb, header_pos + header_size * 8)) {
        av_log(s->avctx, AV_LOG_ERROR, "Read past end of XXCH frame header\n");
        return AVERROR_INVALIDDATA;
    }

    // Parse XXCH channel set 0
    if ((ret = parse_frame_data(s, HEADER_XXCH, s->nchannels)) < 0)
        return ret;

    if (ff_dca_seek_bits(&s->gb, header_pos + header_size * 8 + xxch_frame_size * 8)) {
        av_log(s->avctx, AV_LOG_ERROR, "Read past end of XXCH channel set\n");
        return AVERROR_INVALIDDATA;
    }

    return 0;
}

static int parse_xbr_subframe(DCACoreDecoder *s, int xbr_base_ch, int xbr_nchannels,
                              int *xbr_nsubbands, int xbr_transition_mode, int sf, int *sub_pos)
{
    int     xbr_nabits[DCA_CHANNELS];
    int     xbr_bit_allocation[DCA_CHANNELS][DCA_SUBBANDS];
    int     xbr_scale_nbits[DCA_CHANNELS];
    int32_t xbr_scale_factors[DCA_CHANNELS][DCA_SUBBANDS][2];
    int     ssf, ch, band, ofs;

    // Check number of subband samples in this subframe
    if (*sub_pos + s->nsubsubframes[sf] * DCA_SUBBAND_SAMPLES > s->npcmblocks) {
        av_log(s->avctx, AV_LOG_ERROR, "Subband sample buffer overflow\n");
        return AVERROR_INVALIDDATA;
    }

    if (get_bits_left(&s->gb) < 0)
        return AVERROR_INVALIDDATA;

    // Number of bits for XBR bit allocation index
    for (ch = xbr_base_ch; ch < xbr_nchannels; ch++)
        xbr_nabits[ch] = get_bits(&s->gb, 2) + 2;

    // XBR bit allocation index
    for (ch = xbr_base_ch; ch < xbr_nchannels; ch++) {
        for (band = 0; band < xbr_nsubbands[ch]; band++) {
            xbr_bit_allocation[ch][band] = get_bits(&s->gb, xbr_nabits[ch]);
            if (xbr_bit_allocation[ch][band] > DCA_ABITS_MAX) {
                av_log(s->avctx, AV_LOG_ERROR, "Invalid XBR bit allocation index\n");
                return AVERROR_INVALIDDATA;
            }
        }
    }

    // Number of bits for scale indices
    for (ch = xbr_base_ch; ch < xbr_nchannels; ch++) {
        xbr_scale_nbits[ch] = get_bits(&s->gb, 3);
        if (!xbr_scale_nbits[ch]) {
            av_log(s->avctx, AV_LOG_ERROR, "Invalid number of bits for XBR scale factor index\n");
            return AVERROR_INVALIDDATA;
        }
    }

    // XBR scale factors
    for (ch = xbr_base_ch; ch < xbr_nchannels; ch++) {
        const uint32_t *scale_table;
        int scale_size;

        // Select the root square table
        if (s->scale_factor_sel[ch] > 5) {
            scale_table = ff_dca_scale_factor_quant7;
            scale_size = FF_ARRAY_ELEMS(ff_dca_scale_factor_quant7);
        } else {
            scale_table = ff_dca_scale_factor_quant6;
            scale_size = FF_ARRAY_ELEMS(ff_dca_scale_factor_quant6);
        }

        // Parse scale factor indices and look up scale factors from the root
        // square table
        for (band = 0; band < xbr_nsubbands[ch]; band++) {
            if (xbr_bit_allocation[ch][band]) {
                int scale_index = get_bits(&s->gb, xbr_scale_nbits[ch]);
                if (scale_index >= scale_size) {
                    av_log(s->avctx, AV_LOG_ERROR, "Invalid XBR scale factor index\n");
                    return AVERROR_INVALIDDATA;
                }
                xbr_scale_factors[ch][band][0] = scale_table[scale_index];
                if (xbr_transition_mode && s->transition_mode[sf][ch][band]) {
                    scale_index = get_bits(&s->gb, xbr_scale_nbits[ch]);
                    if (scale_index >= scale_size) {
                        av_log(s->avctx, AV_LOG_ERROR, "Invalid XBR scale factor index\n");
                        return AVERROR_INVALIDDATA;
                    }
                    xbr_scale_factors[ch][band][1] = scale_table[scale_index];
                }
            }
        }
    }

    // Audio data
    for (ssf = 0, ofs = *sub_pos; ssf < s->nsubsubframes[sf]; ssf++) {
        for (ch = xbr_base_ch; ch < xbr_nchannels; ch++) {
            if (get_bits_left(&s->gb) < 0)
                return AVERROR_INVALIDDATA;

            for (band = 0; band < xbr_nsubbands[ch]; band++) {
                int ret, trans_ssf, abits = xbr_bit_allocation[ch][band];
                int32_t audio[DCA_SUBBAND_SAMPLES], step_size, scale;

                // Extract bits from the bit stream
                if (abits > 7) {
                    // No further encoding
                    get_array(&s->gb, audio, DCA_SUBBAND_SAMPLES, abits - 3);
                } else if (abits > 0) {
                    // Block codes
                    if ((ret = parse_block_codes(s, audio, abits)) < 0)
                        return ret;
                } else {
                    // No bits allocated
                    continue;
                }

                // Look up quantization step size
                step_size = ff_dca_lossless_quant[abits];

                // Identify transient location
                if (xbr_transition_mode)
                    trans_ssf = s->transition_mode[sf][ch][band];
                else
                    trans_ssf = 0;

                // Determine proper scale factor
                if (trans_ssf == 0 || ssf < trans_ssf)
                    scale = xbr_scale_factors[ch][band][0];
                else
                    scale = xbr_scale_factors[ch][band][1];

                dequantize(s->subband_samples[ch][band] + ofs,
                           audio, step_size, scale, 1);
            }
        }

        // DSYNC
        if ((ssf == s->nsubsubframes[sf] - 1 || s->sync_ssf) && get_bits(&s->gb, 16) != 0xffff) {
            av_log(s->avctx, AV_LOG_ERROR, "XBR-DSYNC check failed\n");
            return AVERROR_INVALIDDATA;
        }

        ofs += DCA_SUBBAND_SAMPLES;
    }

    // Advance subband sample pointer for the next subframe
    *sub_pos = ofs;
    return 0;
}

static int parse_xbr_frame(DCACoreDecoder *s)
{
    int     xbr_frame_size[DCA_EXSS_CHSETS_MAX];
    int     xbr_nchannels[DCA_EXSS_CHSETS_MAX];
    int     xbr_nsubbands[DCA_EXSS_CHSETS_MAX * DCA_EXSS_CHANNELS_MAX];
    int     xbr_nchsets, xbr_transition_mode, xbr_band_nbits, xbr_base_ch;
    int     i, ch1, ch2, ret, header_size, header_pos = get_bits_count(&s->gb);

    // XBR sync word
    if (get_bits_long(&s->gb, 32) != DCA_SYNCWORD_XBR) {
        av_log(s->avctx, AV_LOG_ERROR, "Invalid XBR sync word\n");
        return AVERROR_INVALIDDATA;
    }

    // XBR frame header length
    header_size = get_bits(&s->gb, 6) + 1;

    // Check XBR frame header CRC
    if (ff_dca_check_crc(s->avctx, &s->gb, header_pos + 32, header_pos + header_size * 8)) {
        av_log(s->avctx, AV_LOG_ERROR, "Invalid XBR frame header checksum\n");
        return AVERROR_INVALIDDATA;
    }

    // Number of channel sets
    xbr_nchsets = get_bits(&s->gb, 2) + 1;

    // Channel set data byte size
    for (i = 0; i < xbr_nchsets; i++)
        xbr_frame_size[i] = get_bits(&s->gb, 14) + 1;

    // Transition mode flag
    xbr_transition_mode = get_bits1(&s->gb);

    // Channel set headers
    for (i = 0, ch2 = 0; i < xbr_nchsets; i++) {
        xbr_nchannels[i] = get_bits(&s->gb, 3) + 1;
        xbr_band_nbits = get_bits(&s->gb, 2) + 5;
        for (ch1 = 0; ch1 < xbr_nchannels[i]; ch1++, ch2++) {
            xbr_nsubbands[ch2] = get_bits(&s->gb, xbr_band_nbits) + 1;
            if (xbr_nsubbands[ch2] > DCA_SUBBANDS) {
                av_log(s->avctx, AV_LOG_ERROR, "Invalid number of active XBR subbands (%d)\n", xbr_nsubbands[ch2]);
                return AVERROR_INVALIDDATA;
            }
        }
    }

    // Reserved
    // Byte align
    // CRC16 of XBR frame header
    if (ff_dca_seek_bits(&s->gb, header_pos + header_size * 8)) {
        av_log(s->avctx, AV_LOG_ERROR, "Read past end of XBR frame header\n");
        return AVERROR_INVALIDDATA;
    }

    // Channel set data
    for (i = 0, xbr_base_ch = 0; i < xbr_nchsets; i++) {
        header_pos = get_bits_count(&s->gb);

        if (xbr_base_ch + xbr_nchannels[i] <= s->nchannels) {
            int sf, sub_pos;

            for (sf = 0, sub_pos = 0; sf < s->nsubframes; sf++) {
                if ((ret = parse_xbr_subframe(s, xbr_base_ch,
                                              xbr_base_ch + xbr_nchannels[i],
                                              xbr_nsubbands, xbr_transition_mode,
                                              sf, &sub_pos)) < 0)
                    return ret;
            }
        }

        xbr_base_ch += xbr_nchannels[i];

        if (ff_dca_seek_bits(&s->gb, header_pos + xbr_frame_size[i] * 8)) {
            av_log(s->avctx, AV_LOG_ERROR, "Read past end of XBR channel set\n");
            return AVERROR_INVALIDDATA;
        }
    }

    return 0;
}

// Modified ISO/IEC 9899 linear congruential generator
// Returns pseudorandom integer in range [-2^30, 2^30 - 1]
static int rand_x96(DCACoreDecoder *s)
{
    s->x96_rand = 1103515245U * s->x96_rand + 12345U;
    return (s->x96_rand & 0x7fffffff) - 0x40000000;
}

static int parse_x96_subframe_audio(DCACoreDecoder *s, int sf, int xch_base, int *sub_pos)
{
    int n, ssf, ch, band, ofs;

    // Check number of subband samples in this subframe
    int nsamples = s->nsubsubframes[sf] * DCA_SUBBAND_SAMPLES;
    if (*sub_pos + nsamples > s->npcmblocks) {
        av_log(s->avctx, AV_LOG_ERROR, "Subband sample buffer overflow\n");
        return AVERROR_INVALIDDATA;
    }

    if (get_bits_left(&s->gb) < 0)
        return AVERROR_INVALIDDATA;

    // VQ encoded or unallocated subbands
    for (ch = xch_base; ch < s->x96_nchannels; ch++) {
        for (band = s->x96_subband_start; band < s->nsubbands[ch]; band++) {
            // Get the sample pointer and scale factor
            int32_t *samples = s->x96_subband_samples[ch][band] + *sub_pos;
            int32_t scale    = s->scale_factors[ch][band >> 1][band & 1];

            switch (s->bit_allocation[ch][band]) {
            case 0: // No bits allocated for subband
                if (scale <= 1)
                    memset(samples, 0, nsamples * sizeof(int32_t));
                else for (n = 0; n < nsamples; n++)
                    // Generate scaled random samples
                    samples[n] = mul31(rand_x96(s), scale);
                break;

            case 1: // VQ encoded subband
                for (ssf = 0; ssf < (s->nsubsubframes[sf] + 1) / 2; ssf++) {
                    // Extract the VQ address from the bit stream and look up
                    // the VQ code book for up to 16 subband samples
                    const int8_t *vq_samples = ff_dca_high_freq_vq[get_bits(&s->gb, 10)];
                    // Scale and take the samples
                    for (n = 0; n < FFMIN(nsamples - ssf * 16, 16); n++)
                        *samples++ = clip23(vq_samples[n] * scale + (1 << 3) >> 4);
                }
                break;
            }
        }
    }

    // Audio data
    for (ssf = 0, ofs = *sub_pos; ssf < s->nsubsubframes[sf]; ssf++) {
        for (ch = xch_base; ch < s->x96_nchannels; ch++) {
            if (get_bits_left(&s->gb) < 0)
                return AVERROR_INVALIDDATA;

            for (band = s->x96_subband_start; band < s->nsubbands[ch]; band++) {
                int ret, abits = s->bit_allocation[ch][band] - 1;
                int32_t audio[DCA_SUBBAND_SAMPLES], step_size, scale;

                // Not VQ encoded or unallocated subbands
                if (abits < 1)
                    continue;

                // Extract bits from the bit stream
                if ((ret = extract_audio(s, audio, abits, ch)) < 0)
                    return ret;

                // Select quantization step size table and look up quantization
                // step size
                if (s->bit_rate == 3)
                    step_size = ff_dca_lossless_quant[abits];
                else
                    step_size = ff_dca_lossy_quant[abits];

                // Get the scale factor
                scale = s->scale_factors[ch][band >> 1][band & 1];

                dequantize(s->x96_subband_samples[ch][band] + ofs,
                           audio, step_size, scale, 0);
            }
        }

        // DSYNC
        if ((ssf == s->nsubsubframes[sf] - 1 || s->sync_ssf) && get_bits(&s->gb, 16) != 0xffff) {
            av_log(s->avctx, AV_LOG_ERROR, "X96-DSYNC check failed\n");
            return AVERROR_INVALIDDATA;
        }

        ofs += DCA_SUBBAND_SAMPLES;
    }

    // Inverse ADPCM
    for (ch = xch_base; ch < s->x96_nchannels; ch++) {
        inverse_adpcm(s->x96_subband_samples[ch], s->prediction_vq_index[ch],
                      s->prediction_mode[ch], s->x96_subband_start, s->nsubbands[ch],
                      *sub_pos, nsamples);
    }

    // Joint subband coding
    for (ch = xch_base; ch < s->x96_nchannels; ch++) {
        int src_ch = s->joint_intensity_index[ch] - 1;
        if (src_ch >= 0) {
            s->dcadsp->decode_joint(s->x96_subband_samples[ch], s->x96_subband_samples[src_ch],
                                    s->joint_scale_factors[ch], s->nsubbands[ch],
                                    s->nsubbands[src_ch], *sub_pos, nsamples);
        }
    }

    // Advance subband sample pointer for the next subframe
    *sub_pos = ofs;
    return 0;
}

static void erase_x96_adpcm_history(DCACoreDecoder *s)
{
    int ch, band;

    // Erase ADPCM history from previous frame if
    // predictor history switch was disabled
    for (ch = 0; ch < DCA_CHANNELS; ch++)
        for (band = 0; band < DCA_SUBBANDS_X96; band++)
            AV_ZERO128(s->x96_subband_samples[ch][band] - DCA_ADPCM_COEFFS);

    emms_c();
}

static int alloc_x96_sample_buffer(DCACoreDecoder *s)
{
    int nchsamples = DCA_ADPCM_COEFFS + s->npcmblocks;
    int nframesamples = nchsamples * DCA_CHANNELS * DCA_SUBBANDS_X96;
    unsigned int size = s->x96_subband_size;
    int ch, band;

    // Reallocate subband sample buffer
    av_fast_mallocz(&s->x96_subband_buffer, &s->x96_subband_size,
                    nframesamples * sizeof(int32_t));
    if (!s->x96_subband_buffer)
        return AVERROR(ENOMEM);

    if (size != s->x96_subband_size) {
        for (ch = 0; ch < DCA_CHANNELS; ch++)
            for (band = 0; band < DCA_SUBBANDS_X96; band++)
                s->x96_subband_samples[ch][band] = s->x96_subband_buffer +
                    (ch * DCA_SUBBANDS_X96 + band) * nchsamples + DCA_ADPCM_COEFFS;
    }

    if (!s->predictor_history)
        erase_x96_adpcm_history(s);

    return 0;
}

static int parse_x96_subframe_header(DCACoreDecoder *s, int xch_base)
{
    int ch, band, ret;

    if (get_bits_left(&s->gb) < 0)
        return AVERROR_INVALIDDATA;

    // Prediction mode
    for (ch = xch_base; ch < s->x96_nchannels; ch++)
        for (band = s->x96_subband_start; band < s->nsubbands[ch]; band++)
            s->prediction_mode[ch][band] = get_bits1(&s->gb);

    // Prediction coefficients VQ address
    for (ch = xch_base; ch < s->x96_nchannels; ch++)
        for (band = s->x96_subband_start; band < s->nsubbands[ch]; band++)
            if (s->prediction_mode[ch][band])
                s->prediction_vq_index[ch][band] = get_bits(&s->gb, 12);

    // Bit allocation index
    for (ch = xch_base; ch < s->x96_nchannels; ch++) {
        int sel = s->bit_allocation_sel[ch];
        int abits = 0;

        for (band = s->x96_subband_start; band < s->nsubbands[ch]; band++) {
            // If Huffman code was used, the difference of abits was encoded
            if (sel < 7)
                abits += dca_get_vlc(&s->gb, &ff_dca_vlc_quant_index[5 + 2 * s->x96_high_res], sel);
            else
                abits = get_bits(&s->gb, 3 + s->x96_high_res);

            if (abits < 0 || abits > 7 + 8 * s->x96_high_res) {
                av_log(s->avctx, AV_LOG_ERROR, "Invalid X96 bit allocation index\n");
                return AVERROR_INVALIDDATA;
            }

            s->bit_allocation[ch][band] = abits;
        }
    }

    // Scale factors
    for (ch = xch_base; ch < s->x96_nchannels; ch++) {
        int sel = s->scale_factor_sel[ch];
        int scale_index = 0;

        // Extract scales for subbands which are transmitted even for
        // unallocated subbands
        for (band = s->x96_subband_start; band < s->nsubbands[ch]; band++) {
            if ((ret = parse_scale(s, &scale_index, sel)) < 0)
                return ret;
            s->scale_factors[ch][band >> 1][band & 1] = ret;
        }
    }

    // Joint subband codebook select
    for (ch = xch_base; ch < s->x96_nchannels; ch++) {
        if (s->joint_intensity_index[ch]) {
            s->joint_scale_sel[ch] = get_bits(&s->gb, 3);
            if (s->joint_scale_sel[ch] == 7) {
                av_log(s->avctx, AV_LOG_ERROR, "Invalid X96 joint scale factor code book\n");
                return AVERROR_INVALIDDATA;
            }
        }
    }

    // Scale factors for joint subband coding
    for (ch = xch_base; ch < s->x96_nchannels; ch++) {
        int src_ch = s->joint_intensity_index[ch] - 1;
        if (src_ch >= 0) {
            int sel = s->joint_scale_sel[ch];
            for (band = s->nsubbands[ch]; band < s->nsubbands[src_ch]; band++) {
                if ((ret = parse_joint_scale(s, sel)) < 0)
                    return ret;
                s->joint_scale_factors[ch][band] = ret;
            }
        }
    }

    // Side information CRC check word
    if (s->crc_present)
        skip_bits(&s->gb, 16);

    return 0;
}

static int parse_x96_coding_header(DCACoreDecoder *s, int exss, int xch_base)
{
    int n, ch, header_size = 0, header_pos = get_bits_count(&s->gb);

    if (get_bits_left(&s->gb) < 0)
        return AVERROR_INVALIDDATA;

    if (exss) {
        // Channel set header length
        header_size = get_bits(&s->gb, 7) + 1;

        // Check CRC
        if (s->x96_crc_present
            && ff_dca_check_crc(s->avctx, &s->gb, header_pos, header_pos + header_size * 8)) {
            av_log(s->avctx, AV_LOG_ERROR, "Invalid X96 channel set header checksum\n");
            return AVERROR_INVALIDDATA;
        }
    }

    // High resolution flag
    s->x96_high_res = get_bits1(&s->gb);

    // First encoded subband
    if (s->x96_rev_no < 8) {
        s->x96_subband_start = get_bits(&s->gb, 5);
        if (s->x96_subband_start > 27) {
            av_log(s->avctx, AV_LOG_ERROR, "Invalid X96 subband start index (%d)\n", s->x96_subband_start);
            return AVERROR_INVALIDDATA;
        }
    } else {
        s->x96_subband_start = DCA_SUBBANDS;
    }

    // Subband activity count
    for (ch = xch_base; ch < s->x96_nchannels; ch++) {
        s->nsubbands[ch] = get_bits(&s->gb, 6) + 1;
        if (s->nsubbands[ch] < DCA_SUBBANDS) {
            av_log(s->avctx, AV_LOG_ERROR, "Invalid X96 subband activity count (%d)\n", s->nsubbands[ch]);
            return AVERROR_INVALIDDATA;
        }
    }

    // Joint intensity coding index
    for (ch = xch_base; ch < s->x96_nchannels; ch++) {
        if ((n = get_bits(&s->gb, 3)) && xch_base)
            n += xch_base - 1;
        if (n > s->x96_nchannels) {
            av_log(s->avctx, AV_LOG_ERROR, "Invalid X96 joint intensity coding index\n");
            return AVERROR_INVALIDDATA;
        }
        s->joint_intensity_index[ch] = n;
    }

    // Scale factor code book
    for (ch = xch_base; ch < s->x96_nchannels; ch++) {
        s->scale_factor_sel[ch] = get_bits(&s->gb, 3);
        if (s->scale_factor_sel[ch] >= 6) {
            av_log(s->avctx, AV_LOG_ERROR, "Invalid X96 scale factor code book\n");
            return AVERROR_INVALIDDATA;
        }
    }

    // Bit allocation quantizer select
    for (ch = xch_base; ch < s->x96_nchannels; ch++)
        s->bit_allocation_sel[ch] = get_bits(&s->gb, 3);

    // Quantization index codebook select
    for (n = 0; n < 6 + 4 * s->x96_high_res; n++)
        for (ch = xch_base; ch < s->x96_nchannels; ch++)
            s->quant_index_sel[ch][n] = get_bits(&s->gb, quant_index_sel_nbits[n]);

    if (exss) {
        // Reserved
        // Byte align
        // CRC16 of channel set header
        if (ff_dca_seek_bits(&s->gb, header_pos + header_size * 8)) {
            av_log(s->avctx, AV_LOG_ERROR, "Read past end of X96 channel set header\n");
            return AVERROR_INVALIDDATA;
        }
    } else {
        if (s->crc_present)
            skip_bits(&s->gb, 16);
    }

    return 0;
}

static int parse_x96_frame_data(DCACoreDecoder *s, int exss, int xch_base)
{
    int sf, ch, ret, band, sub_pos;

    if ((ret = parse_x96_coding_header(s, exss, xch_base)) < 0)
        return ret;

    for (sf = 0, sub_pos = 0; sf < s->nsubframes; sf++) {
        if ((ret = parse_x96_subframe_header(s, xch_base)) < 0)
            return ret;
        if ((ret = parse_x96_subframe_audio(s, sf, xch_base, &sub_pos)) < 0)
            return ret;
    }

    for (ch = xch_base; ch < s->x96_nchannels; ch++) {
        // Determine number of active subbands for this channel
        int nsubbands = s->nsubbands[ch];
        if (s->joint_intensity_index[ch])
            nsubbands = FFMAX(nsubbands, s->nsubbands[s->joint_intensity_index[ch] - 1]);

        // Update history for ADPCM and clear inactive subbands
        for (band = 0; band < DCA_SUBBANDS_X96; band++) {
            int32_t *samples = s->x96_subband_samples[ch][band] - DCA_ADPCM_COEFFS;
            if (band >= s->x96_subband_start && band < nsubbands)
                AV_COPY128(samples, samples + s->npcmblocks);
            else
                memset(samples, 0, (DCA_ADPCM_COEFFS + s->npcmblocks) * sizeof(int32_t));
        }
    }

    emms_c();

    return 0;
}

static int parse_x96_frame(DCACoreDecoder *s)
{
    int ret;

    // Revision number
    s->x96_rev_no = get_bits(&s->gb, 4);
    if (s->x96_rev_no < 1 || s->x96_rev_no > 8) {
        av_log(s->avctx, AV_LOG_ERROR, "Invalid X96 revision (%d)\n", s->x96_rev_no);
        return AVERROR_INVALIDDATA;
    }

    s->x96_crc_present = 0;
    s->x96_nchannels = s->nchannels;

    if ((ret = alloc_x96_sample_buffer(s)) < 0)
        return ret;

    if ((ret = parse_x96_frame_data(s, 0, 0)) < 0)
        return ret;

    // Seek to the end of core frame
    if (ff_dca_seek_bits(&s->gb, s->frame_size * 8)) {
        av_log(s->avctx, AV_LOG_ERROR, "Read past end of X96 frame\n");
        return AVERROR_INVALIDDATA;
    }

    return 0;
}

static int parse_x96_frame_exss(DCACoreDecoder *s)
{
    int     x96_frame_size[DCA_EXSS_CHSETS_MAX];
    int     x96_nchannels[DCA_EXSS_CHSETS_MAX];
    int     x96_nchsets, x96_base_ch;
    int     i, ret, header_size, header_pos = get_bits_count(&s->gb);

    // X96 sync word
    if (get_bits_long(&s->gb, 32) != DCA_SYNCWORD_X96) {
        av_log(s->avctx, AV_LOG_ERROR, "Invalid X96 sync word\n");
        return AVERROR_INVALIDDATA;
    }

    // X96 frame header length
    header_size = get_bits(&s->gb, 6) + 1;

    // Check X96 frame header CRC
    if (ff_dca_check_crc(s->avctx, &s->gb, header_pos + 32, header_pos + header_size * 8)) {
        av_log(s->avctx, AV_LOG_ERROR, "Invalid X96 frame header checksum\n");
        return AVERROR_INVALIDDATA;
    }

    // Revision number
    s->x96_rev_no = get_bits(&s->gb, 4);
    if (s->x96_rev_no < 1 || s->x96_rev_no > 8) {
        av_log(s->avctx, AV_LOG_ERROR, "Invalid X96 revision (%d)\n", s->x96_rev_no);
        return AVERROR_INVALIDDATA;
    }

    // CRC presence flag for channel set header
    s->x96_crc_present = get_bits1(&s->gb);

    // Number of channel sets
    x96_nchsets = get_bits(&s->gb, 2) + 1;

    // Channel set data byte size
    for (i = 0; i < x96_nchsets; i++)
        x96_frame_size[i] = get_bits(&s->gb, 12) + 1;

    // Number of channels in channel set
    for (i = 0; i < x96_nchsets; i++)
        x96_nchannels[i] = get_bits(&s->gb, 3) + 1;

    // Reserved
    // Byte align
    // CRC16 of X96 frame header
    if (ff_dca_seek_bits(&s->gb, header_pos + header_size * 8)) {
        av_log(s->avctx, AV_LOG_ERROR, "Read past end of X96 frame header\n");
        return AVERROR_INVALIDDATA;
    }

    if ((ret = alloc_x96_sample_buffer(s)) < 0)
        return ret;

    // Channel set data
    s->x96_nchannels = 0;
    for (i = 0, x96_base_ch = 0; i < x96_nchsets; i++) {
        header_pos = get_bits_count(&s->gb);

        if (x96_base_ch + x96_nchannels[i] <= s->nchannels) {
            s->x96_nchannels = x96_base_ch + x96_nchannels[i];
            if ((ret = parse_x96_frame_data(s, 1, x96_base_ch)) < 0)
                return ret;
        }

        x96_base_ch += x96_nchannels[i];

        if (ff_dca_seek_bits(&s->gb, header_pos + x96_frame_size[i] * 8)) {
            av_log(s->avctx, AV_LOG_ERROR, "Read past end of X96 channel set\n");
            return AVERROR_INVALIDDATA;
        }
    }

    return 0;
}

static int parse_aux_data(DCACoreDecoder *s)
{
    int aux_pos;

    if (get_bits_left(&s->gb) < 0)
        return AVERROR_INVALIDDATA;

    // Auxiliary data byte count (can't be trusted)
    skip_bits(&s->gb, 6);

    // 4-byte align
    skip_bits_long(&s->gb, -get_bits_count(&s->gb) & 31);

    // Auxiliary data sync word
    if (get_bits_long(&s->gb, 32) != DCA_SYNCWORD_REV1AUX) {
        av_log(s->avctx, AV_LOG_ERROR, "Invalid auxiliary data sync word\n");
        return AVERROR_INVALIDDATA;
    }

    aux_pos = get_bits_count(&s->gb);

    // Auxiliary decode time stamp flag
    if (get_bits1(&s->gb))
        skip_bits_long(&s->gb, 47);

    // Auxiliary dynamic downmix flag
    if (s->prim_dmix_embedded = get_bits1(&s->gb)) {
        int i, m, n;

        // Auxiliary primary channel downmix type
        s->prim_dmix_type = get_bits(&s->gb, 3);
        if (s->prim_dmix_type >= DCA_DMIX_TYPE_COUNT) {
            av_log(s->avctx, AV_LOG_ERROR, "Invalid primary channel set downmix type\n");
            return AVERROR_INVALIDDATA;
        }

        // Size of downmix coefficients matrix
        m = ff_dca_dmix_primary_nch[s->prim_dmix_type];
        n = ff_dca_channels[s->audio_mode] + !!s->lfe_present;

        // Dynamic downmix code coefficients
        for (i = 0; i < m * n; i++) {
            int code = get_bits(&s->gb, 9);
            int sign = (code >> 8) - 1;
            unsigned int index = code & 0xff;
            if (index >= FF_DCA_DMIXTABLE_SIZE) {
                av_log(s->avctx, AV_LOG_ERROR, "Invalid downmix coefficient index\n");
                return AVERROR_INVALIDDATA;
            }
            s->prim_dmix_coeff[i] = (ff_dca_dmixtable[index] ^ sign) - sign;
        }
    }

    // Byte align
    skip_bits(&s->gb, -get_bits_count(&s->gb) & 7);

    // CRC16 of auxiliary data
    skip_bits(&s->gb, 16);

    // Check CRC
    if (ff_dca_check_crc(s->avctx, &s->gb, aux_pos, get_bits_count(&s->gb))) {
        av_log(s->avctx, AV_LOG_ERROR, "Invalid auxiliary data checksum\n");
        return AVERROR_INVALIDDATA;
    }

    return 0;
}

static int parse_optional_info(DCACoreDecoder *s)
{
    DCAContext *dca = s->avctx->priv_data;
    int ret = -1;

    // Time code stamp
    if (s->ts_present)
        skip_bits_long(&s->gb, 32);

    // Auxiliary data
    if (s->aux_present && (ret = parse_aux_data(s)) < 0
        && (s->avctx->err_recognition & AV_EF_EXPLODE))
        return ret;

    if (ret < 0)
        s->prim_dmix_embedded = 0;

    // Core extensions
    if (s->ext_audio_present && !dca->core_only) {
        int sync_pos = FFMIN(s->frame_size / 4, s->gb.size_in_bits / 32) - 1;
        int last_pos = get_bits_count(&s->gb) / 32;
        int size, dist;

        // Search for extension sync words aligned on 4-byte boundary. Search
        // must be done backwards from the end of core frame to work around
        // sync word aliasing issues.
        switch (s->ext_audio_type) {
        case EXT_AUDIO_XCH:
            if (dca->request_channel_layout)
                break;

            // The distance between XCH sync word and end of the core frame
            // must be equal to XCH frame size. Off by one error is allowed for
            // compatibility with legacy bitstreams. Minimum XCH frame size is
            // 96 bytes. AMODE and PCHS are further checked to reduce
            // probability of alias sync detection.
            for (; sync_pos >= last_pos; sync_pos--) {
                if (AV_RB32(s->gb.buffer + sync_pos * 4) == DCA_SYNCWORD_XCH) {
                    s->gb.index = (sync_pos + 1) * 32;
                    size = get_bits(&s->gb, 10) + 1;
                    dist = s->frame_size - sync_pos * 4;
                    if (size >= 96
                        && (size == dist || size - 1 == dist)
                        && get_bits(&s->gb, 7) == 0x08) {
                        s->xch_pos = get_bits_count(&s->gb);
                        break;
                    }
                }
            }

            if (!s->xch_pos) {
                av_log(s->avctx, AV_LOG_ERROR, "XCH sync word not found\n");
                if (s->avctx->err_recognition & AV_EF_EXPLODE)
                    return AVERROR_INVALIDDATA;
            }
            break;

        case EXT_AUDIO_X96:
            // The distance between X96 sync word and end of the core frame
            // must be equal to X96 frame size. Minimum X96 frame size is 96
            // bytes.
            for (; sync_pos >= last_pos; sync_pos--) {
                if (AV_RB32(s->gb.buffer + sync_pos * 4) == DCA_SYNCWORD_X96) {
                    s->gb.index = (sync_pos + 1) * 32;
                    size = get_bits(&s->gb, 12) + 1;
                    dist = s->frame_size - sync_pos * 4;
                    if (size >= 96 && size == dist) {
                        s->x96_pos = get_bits_count(&s->gb);
                        break;
                    }
                }
            }

            if (!s->x96_pos) {
                av_log(s->avctx, AV_LOG_ERROR, "X96 sync word not found\n");
                if (s->avctx->err_recognition & AV_EF_EXPLODE)
                    return AVERROR_INVALIDDATA;
            }
            break;

        case EXT_AUDIO_XXCH:
            if (dca->request_channel_layout)
                break;

            // XXCH frame header CRC must be valid. Minimum XXCH frame header
            // size is 11 bytes.
            for (; sync_pos >= last_pos; sync_pos--) {
                if (AV_RB32(s->gb.buffer + sync_pos * 4) == DCA_SYNCWORD_XXCH) {
                    s->gb.index = (sync_pos + 1) * 32;
                    size = get_bits(&s->gb, 6) + 1;
                    dist = s->gb.size_in_bits / 8 - sync_pos * 4;
                    if (size >= 11 && size <= dist &&
                        !av_crc(dca->crctab, 0xffff, s->gb.buffer +
                                (sync_pos + 1) * 4, size - 4)) {
                        s->xxch_pos = sync_pos * 32;
                        break;
                    }
                }
            }

            if (!s->xxch_pos) {
                av_log(s->avctx, AV_LOG_ERROR, "XXCH sync word not found\n");
                if (s->avctx->err_recognition & AV_EF_EXPLODE)
                    return AVERROR_INVALIDDATA;
            }
            break;
        }
    }

    return 0;
}

int ff_dca_core_parse(DCACoreDecoder *s, uint8_t *data, int size)
{
    int ret;

    s->ext_audio_mask = 0;
    s->xch_pos = s->xxch_pos = s->x96_pos = 0;

    if ((ret = init_get_bits8(&s->gb, data, size)) < 0)
        return ret;

    skip_bits_long(&s->gb, 32);
    if ((ret = parse_frame_header(s)) < 0)
        return ret;
    if ((ret = alloc_sample_buffer(s)) < 0)
        return ret;
    if ((ret = parse_frame_data(s, HEADER_CORE, 0)) < 0)
        return ret;
    if ((ret = parse_optional_info(s)) < 0)
        return ret;

    // Workaround for DTS in WAV
    if (s->frame_size > size && s->frame_size < size + 4)
        s->frame_size = size;

    if (ff_dca_seek_bits(&s->gb, s->frame_size * 8)) {
        av_log(s->avctx, AV_LOG_ERROR, "Read past end of core frame\n");
        if (s->avctx->err_recognition & AV_EF_EXPLODE)
            return AVERROR_INVALIDDATA;
    }

    return 0;
}

int ff_dca_core_parse_exss(DCACoreDecoder *s, uint8_t *data, DCAExssAsset *asset)
{
    AVCodecContext *avctx = s->avctx;
    DCAContext *dca = avctx->priv_data;
    GetBitContext gb = s->gb;
    int exss_mask = asset ? asset->extension_mask : 0;
    int ret = 0, ext = 0;

    // Parse (X)XCH unless downmixing
    if (!dca->request_channel_layout) {
        if (exss_mask & DCA_EXSS_XXCH) {
            if ((ret = init_get_bits8(&s->gb, data + asset->xxch_offset, asset->xxch_size)) < 0)
                return ret;
            ret = parse_xxch_frame(s);
            ext = DCA_EXSS_XXCH;
        } else if (s->xxch_pos) {
            s->gb.index = s->xxch_pos;
            ret = parse_xxch_frame(s);
            ext = DCA_CSS_XXCH;
        } else if (s->xch_pos) {
            s->gb.index = s->xch_pos;
            ret = parse_xch_frame(s);
            ext = DCA_CSS_XCH;
        }

        // Revert to primary channel set in case (X)XCH parsing fails
        if (ret < 0) {
            if (avctx->err_recognition & AV_EF_EXPLODE)
                return ret;
            s->nchannels = ff_dca_channels[s->audio_mode];
            s->ch_mask = audio_mode_ch_mask[s->audio_mode];
            if (s->lfe_present)
                s->ch_mask |= DCA_SPEAKER_MASK_LFE1;
        } else {
            s->ext_audio_mask |= ext;
        }
    }

    // Parse XBR
    if (exss_mask & DCA_EXSS_XBR) {
        if ((ret = init_get_bits8(&s->gb, data + asset->xbr_offset, asset->xbr_size)) < 0)
            return ret;
        if ((ret = parse_xbr_frame(s)) < 0) {
            if (avctx->err_recognition & AV_EF_EXPLODE)
                return ret;
        } else {
            s->ext_audio_mask |= DCA_EXSS_XBR;
        }
    }

    // Parse X96 unless decoding XLL
    if (!(dca->packet & DCA_PACKET_XLL)) {
        if (exss_mask & DCA_EXSS_X96) {
            if ((ret = init_get_bits8(&s->gb, data + asset->x96_offset, asset->x96_size)) < 0)
                return ret;
            if ((ret = parse_x96_frame_exss(s)) < 0) {
                if (ret == AVERROR(ENOMEM) || (avctx->err_recognition & AV_EF_EXPLODE))
                    return ret;
            } else {
                s->ext_audio_mask |= DCA_EXSS_X96;
            }
        } else if (s->x96_pos) {
            s->gb = gb;
            s->gb.index = s->x96_pos;
            if ((ret = parse_x96_frame(s)) < 0) {
                if (ret == AVERROR(ENOMEM) || (avctx->err_recognition & AV_EF_EXPLODE))
                    return ret;
            } else {
                s->ext_audio_mask |= DCA_CSS_X96;
            }
        }
    }

    return 0;
}

static int map_prm_ch_to_spkr(DCACoreDecoder *s, int ch)
{
    int pos, spkr;

    // Try to map this channel to core first
    pos = ff_dca_channels[s->audio_mode];
    if (ch < pos) {
        spkr = prm_ch_to_spkr_map[s->audio_mode][ch];
        if (s->ext_audio_mask & (DCA_CSS_XXCH | DCA_EXSS_XXCH)) {
            if (s->xxch_core_mask & (1U << spkr))
                return spkr;
            if (spkr == DCA_SPEAKER_Ls && (s->xxch_core_mask & DCA_SPEAKER_MASK_Lss))
                return DCA_SPEAKER_Lss;
            if (spkr == DCA_SPEAKER_Rs && (s->xxch_core_mask & DCA_SPEAKER_MASK_Rss))
                return DCA_SPEAKER_Rss;
            return -1;
        }
        return spkr;
    }

    // Then XCH
    if ((s->ext_audio_mask & DCA_CSS_XCH) && ch == pos)
        return DCA_SPEAKER_Cs;

    // Then XXCH
    if (s->ext_audio_mask & (DCA_CSS_XXCH | DCA_EXSS_XXCH)) {
        for (spkr = DCA_SPEAKER_Cs; spkr < s->xxch_mask_nbits; spkr++)
            if (s->xxch_spkr_mask & (1U << spkr))
                if (pos++ == ch)
                    return spkr;
    }

    // No mapping
    return -1;
}

static void erase_dsp_history(DCACoreDecoder *s)
{
    memset(s->dcadsp_data, 0, sizeof(s->dcadsp_data));
    s->output_history_lfe_fixed = 0;
    s->output_history_lfe_float = 0;
}

static void set_filter_mode(DCACoreDecoder *s, int mode)
{
    if (s->filter_mode != mode) {
        erase_dsp_history(s);
        s->filter_mode = mode;
    }
}

int ff_dca_core_filter_fixed(DCACoreDecoder *s, int x96_synth)
{
    int n, ch, spkr, nsamples, x96_nchannels = 0;
    const int32_t *filter_coeff;
    int32_t *ptr;

    // Externally set x96_synth flag implies that X96 synthesis should be
    // enabled, yet actual X96 subband data should be discarded. This is a
    // special case for lossless residual decoder that ignores X96 data if
    // present.
    if (!x96_synth && (s->ext_audio_mask & (DCA_CSS_X96 | DCA_EXSS_X96))) {
        x96_nchannels = s->x96_nchannels;
        x96_synth = 1;
    }
    if (x96_synth < 0)
        x96_synth = 0;

    s->output_rate = s->sample_rate << x96_synth;
    s->npcmsamples = nsamples = (s->npcmblocks * DCA_PCMBLOCK_SAMPLES) << x96_synth;

    // Reallocate PCM output buffer
    av_fast_malloc(&s->output_buffer, &s->output_size,
                   nsamples * av_popcount(s->ch_mask) * sizeof(int32_t));
    if (!s->output_buffer)
        return AVERROR(ENOMEM);

    ptr = (int32_t *)s->output_buffer;
    for (spkr = 0; spkr < DCA_SPEAKER_COUNT; spkr++) {
        if (s->ch_mask & (1U << spkr)) {
            s->output_samples[spkr] = ptr;
            ptr += nsamples;
        } else {
            s->output_samples[spkr] = NULL;
        }
    }

    // Handle change of filtering mode
    set_filter_mode(s, x96_synth | DCA_FILTER_MODE_FIXED);

    // Select filter
    if (x96_synth)
        filter_coeff = ff_dca_fir_64bands_fixed;
    else if (s->filter_perfect)
        filter_coeff = ff_dca_fir_32bands_perfect_fixed;
    else
        filter_coeff = ff_dca_fir_32bands_nonperfect_fixed;

    // Filter primary channels
    for (ch = 0; ch < s->nchannels; ch++) {
        // Map this primary channel to speaker
        spkr = map_prm_ch_to_spkr(s, ch);
        if (spkr < 0)
            return AVERROR(EINVAL);

        // Filter bank reconstruction
        s->dcadsp->sub_qmf_fixed[x96_synth](
            &s->synth,
            &s->dcadct,
            s->output_samples[spkr],
            s->subband_samples[ch],
            ch < x96_nchannels ? s->x96_subband_samples[ch] : NULL,
            s->dcadsp_data[ch].u.fix.hist1,
            &s->dcadsp_data[ch].offset,
            s->dcadsp_data[ch].u.fix.hist2,
            filter_coeff,
            s->npcmblocks);
    }

    // Filter LFE channel
    if (s->lfe_present) {
        int32_t *samples = s->output_samples[DCA_SPEAKER_LFE1];
        int nlfesamples = s->npcmblocks >> 1;

        // Check LFF
        if (s->lfe_present == LFE_FLAG_128) {
            av_log(s->avctx, AV_LOG_ERROR, "Fixed point mode doesn't support LFF=1\n");
            return AVERROR(EINVAL);
        }

        // Offset intermediate buffer for X96
        if (x96_synth)
            samples += nsamples / 2;

        // Interpolate LFE channel
        s->dcadsp->lfe_fir_fixed(samples, s->lfe_samples + DCA_LFE_HISTORY,
                                 ff_dca_lfe_fir_64_fixed, s->npcmblocks);

        if (x96_synth) {
            // Filter 96 kHz oversampled LFE PCM to attenuate high frequency
            // (47.6 - 48.0 kHz) components of interpolation image
            s->dcadsp->lfe_x96_fixed(s->output_samples[DCA_SPEAKER_LFE1],
                                     samples, &s->output_history_lfe_fixed,
                                     nsamples / 2);

        }

        // Update LFE history
        for (n = DCA_LFE_HISTORY - 1; n >= 0; n--)
            s->lfe_samples[n] = s->lfe_samples[nlfesamples + n];
    }

    return 0;
}

static int filter_frame_fixed(DCACoreDecoder *s, AVFrame *frame)
{
    AVCodecContext *avctx = s->avctx;
    DCAContext *dca = avctx->priv_data;
    int i, n, ch, ret, spkr, nsamples;

    // Don't filter twice when falling back from XLL
    if (!(dca->packet & DCA_PACKET_XLL) && (ret = ff_dca_core_filter_fixed(s, 0)) < 0)
        return ret;

    avctx->sample_rate = s->output_rate;
    avctx->sample_fmt = AV_SAMPLE_FMT_S32P;
    avctx->bits_per_raw_sample = 24;

    frame->nb_samples = nsamples = s->npcmsamples;
    if ((ret = ff_get_buffer(avctx, frame, 0)) < 0)
        return ret;

    // Undo embedded XCH downmix
    if (s->es_format && (s->ext_audio_mask & DCA_CSS_XCH)
        && s->audio_mode >= AMODE_2F2R) {
        s->dcadsp->dmix_sub_xch(s->output_samples[DCA_SPEAKER_Ls],
                                s->output_samples[DCA_SPEAKER_Rs],
                                s->output_samples[DCA_SPEAKER_Cs],
                                nsamples);

    }

    // Undo embedded XXCH downmix
    if ((s->ext_audio_mask & (DCA_CSS_XXCH | DCA_EXSS_XXCH))
        && s->xxch_dmix_embedded) {
        int scale_inv   = s->xxch_dmix_scale_inv;
        int *coeff_ptr  = s->xxch_dmix_coeff;
        int xch_base    = ff_dca_channels[s->audio_mode];
        av_assert1(s->nchannels - xch_base <= DCA_XXCH_CHANNELS_MAX);

        // Undo embedded core downmix pre-scaling
        for (spkr = 0; spkr < s->xxch_mask_nbits; spkr++) {
            if (s->xxch_core_mask & (1U << spkr)) {
                s->dcadsp->dmix_scale_inv(s->output_samples[spkr],
                                          scale_inv, nsamples);
            }
        }

        // Undo downmix
        for (ch = xch_base; ch < s->nchannels; ch++) {
            int src_spkr = map_prm_ch_to_spkr(s, ch);
            if (src_spkr < 0)
                return AVERROR(EINVAL);
            for (spkr = 0; spkr < s->xxch_mask_nbits; spkr++) {
                if (s->xxch_dmix_mask[ch - xch_base] & (1U << spkr)) {
                    int coeff = mul16(*coeff_ptr++, scale_inv);
                    if (coeff) {
                        s->dcadsp->dmix_sub(s->output_samples[spkr    ],
                                            s->output_samples[src_spkr],
                                            coeff, nsamples);
                    }
                }
            }
        }
    }

    if (!(s->ext_audio_mask & (DCA_CSS_XXCH | DCA_CSS_XCH | DCA_EXSS_XXCH))) {
        // Front sum/difference decoding
        if ((s->sumdiff_front && s->audio_mode > AMODE_MONO)
            || s->audio_mode == AMODE_STEREO_SUMDIFF) {
            s->fixed_dsp->butterflies_fixed(s->output_samples[DCA_SPEAKER_L],
                                            s->output_samples[DCA_SPEAKER_R],
                                            nsamples);
        }

        // Surround sum/difference decoding
        if (s->sumdiff_surround && s->audio_mode >= AMODE_2F2R) {
            s->fixed_dsp->butterflies_fixed(s->output_samples[DCA_SPEAKER_Ls],
                                            s->output_samples[DCA_SPEAKER_Rs],
                                            nsamples);
        }
    }

    // Downmix primary channel set to stereo
    if (s->request_mask != s->ch_mask) {
        ff_dca_downmix_to_stereo_fixed(s->dcadsp,
                                       s->output_samples,
                                       s->prim_dmix_coeff,
                                       nsamples, s->ch_mask);
    }

    for (i = 0; i < avctx->channels; i++) {
        int32_t *samples = s->output_samples[s->ch_remap[i]];
        int32_t *plane = (int32_t *)frame->extended_data[i];
        for (n = 0; n < nsamples; n++)
            plane[n] = clip23(samples[n]) * (1 << 8);
    }

    return 0;
}

static int filter_frame_float(DCACoreDecoder *s, AVFrame *frame)
{
    AVCodecContext *avctx = s->avctx;
    int x96_nchannels = 0, x96_synth = 0;
    int i, n, ch, ret, spkr, nsamples, nchannels;
    float *output_samples[DCA_SPEAKER_COUNT] = { NULL }, *ptr;
    const float *filter_coeff;

    if (s->ext_audio_mask & (DCA_CSS_X96 | DCA_EXSS_X96)) {
        x96_nchannels = s->x96_nchannels;
        x96_synth = 1;
    }

    avctx->sample_rate = s->sample_rate << x96_synth;
    avctx->sample_fmt = AV_SAMPLE_FMT_FLTP;
    avctx->bits_per_raw_sample = 0;

    frame->nb_samples = nsamples = (s->npcmblocks * DCA_PCMBLOCK_SAMPLES) << x96_synth;
    if ((ret = ff_get_buffer(avctx, frame, 0)) < 0)
        return ret;

    // Build reverse speaker to channel mapping
    for (i = 0; i < avctx->channels; i++)
        output_samples[s->ch_remap[i]] = (float *)frame->extended_data[i];

    // Allocate space for extra channels
    nchannels = av_popcount(s->ch_mask) - avctx->channels;
    if (nchannels > 0) {
        av_fast_malloc(&s->output_buffer, &s->output_size,
                       nsamples * nchannels * sizeof(float));
        if (!s->output_buffer)
            return AVERROR(ENOMEM);

        ptr = (float *)s->output_buffer;
        for (spkr = 0; spkr < DCA_SPEAKER_COUNT; spkr++) {
            if (!(s->ch_mask & (1U << spkr)))
                continue;
            if (output_samples[spkr])
                continue;
            output_samples[spkr] = ptr;
            ptr += nsamples;
        }
    }

    // Handle change of filtering mode
    set_filter_mode(s, x96_synth);

    // Select filter
    if (x96_synth)
        filter_coeff = ff_dca_fir_64bands;
    else if (s->filter_perfect)
        filter_coeff = ff_dca_fir_32bands_perfect;
    else
        filter_coeff = ff_dca_fir_32bands_nonperfect;

    // Filter primary channels
    for (ch = 0; ch < s->nchannels; ch++) {
        // Map this primary channel to speaker
        spkr = map_prm_ch_to_spkr(s, ch);
        if (spkr < 0)
            return AVERROR(EINVAL);

        // Filter bank reconstruction
        s->dcadsp->sub_qmf_float[x96_synth](
            &s->synth,
            &s->imdct[x96_synth],
            output_samples[spkr],
            s->subband_samples[ch],
            ch < x96_nchannels ? s->x96_subband_samples[ch] : NULL,
            s->dcadsp_data[ch].u.flt.hist1,
            &s->dcadsp_data[ch].offset,
            s->dcadsp_data[ch].u.flt.hist2,
            filter_coeff,
            s->npcmblocks,
            1.0f / (1 << (17 - x96_synth)));
    }

    // Filter LFE channel
    if (s->lfe_present) {
        int dec_select = (s->lfe_present == LFE_FLAG_128);
        float *samples = output_samples[DCA_SPEAKER_LFE1];
        int nlfesamples = s->npcmblocks >> (dec_select + 1);

        // Offset intermediate buffer for X96
        if (x96_synth)
            samples += nsamples / 2;

        // Select filter
        if (dec_select)
            filter_coeff = ff_dca_lfe_fir_128;
        else
            filter_coeff = ff_dca_lfe_fir_64;

        // Interpolate LFE channel
        s->dcadsp->lfe_fir_float[dec_select](
            samples, s->lfe_samples + DCA_LFE_HISTORY,
            filter_coeff, s->npcmblocks);

        if (x96_synth) {
            // Filter 96 kHz oversampled LFE PCM to attenuate high frequency
            // (47.6 - 48.0 kHz) components of interpolation image
            s->dcadsp->lfe_x96_float(output_samples[DCA_SPEAKER_LFE1],
                                     samples, &s->output_history_lfe_float,
                                     nsamples / 2);
        }

        // Update LFE history
        for (n = DCA_LFE_HISTORY - 1; n >= 0; n--)
            s->lfe_samples[n] = s->lfe_samples[nlfesamples + n];
    }

    // Undo embedded XCH downmix
    if (s->es_format && (s->ext_audio_mask & DCA_CSS_XCH)
        && s->audio_mode >= AMODE_2F2R) {
        s->float_dsp->vector_fmac_scalar(output_samples[DCA_SPEAKER_Ls],
                                         output_samples[DCA_SPEAKER_Cs],
                                         -M_SQRT1_2, nsamples);
        s->float_dsp->vector_fmac_scalar(output_samples[DCA_SPEAKER_Rs],
                                         output_samples[DCA_SPEAKER_Cs],
                                         -M_SQRT1_2, nsamples);
    }

    // Undo embedded XXCH downmix
    if ((s->ext_audio_mask & (DCA_CSS_XXCH | DCA_EXSS_XXCH))
        && s->xxch_dmix_embedded) {
        float scale_inv = s->xxch_dmix_scale_inv * (1.0f / (1 << 16));
        int *coeff_ptr  = s->xxch_dmix_coeff;
        int xch_base    = ff_dca_channels[s->audio_mode];
        av_assert1(s->nchannels - xch_base <= DCA_XXCH_CHANNELS_MAX);

        // Undo downmix
        for (ch = xch_base; ch < s->nchannels; ch++) {
            int src_spkr = map_prm_ch_to_spkr(s, ch);
            if (src_spkr < 0)
                return AVERROR(EINVAL);
            for (spkr = 0; spkr < s->xxch_mask_nbits; spkr++) {
                if (s->xxch_dmix_mask[ch - xch_base] & (1U << spkr)) {
                    int coeff = *coeff_ptr++;
                    if (coeff) {
                        s->float_dsp->vector_fmac_scalar(output_samples[    spkr],
                                                         output_samples[src_spkr],
                                                         coeff * (-1.0f / (1 << 15)),
                                                         nsamples);
                    }
                }
            }
        }

        // Undo embedded core downmix pre-scaling
        for (spkr = 0; spkr < s->xxch_mask_nbits; spkr++) {
            if (s->xxch_core_mask & (1U << spkr)) {
                s->float_dsp->vector_fmul_scalar(output_samples[spkr],
                                                 output_samples[spkr],
                                                 scale_inv, nsamples);
            }
        }
    }

    if (!(s->ext_audio_mask & (DCA_CSS_XXCH | DCA_CSS_XCH | DCA_EXSS_XXCH))) {
        // Front sum/difference decoding
        if ((s->sumdiff_front && s->audio_mode > AMODE_MONO)
            || s->audio_mode == AMODE_STEREO_SUMDIFF) {
            s->float_dsp->butterflies_float(output_samples[DCA_SPEAKER_L],
                                            output_samples[DCA_SPEAKER_R],
                                            nsamples);
        }

        // Surround sum/difference decoding
        if (s->sumdiff_surround && s->audio_mode >= AMODE_2F2R) {
            s->float_dsp->butterflies_float(output_samples[DCA_SPEAKER_Ls],
                                            output_samples[DCA_SPEAKER_Rs],
                                            nsamples);
        }
    }

    // Downmix primary channel set to stereo
    if (s->request_mask != s->ch_mask) {
        ff_dca_downmix_to_stereo_float(s->float_dsp, output_samples,
                                       s->prim_dmix_coeff,
                                       nsamples, s->ch_mask);
    }

    return 0;
}

int ff_dca_core_filter_frame(DCACoreDecoder *s, AVFrame *frame)
{
    AVCodecContext *avctx = s->avctx;
    DCAContext *dca = avctx->priv_data;
    DCAExssAsset *asset = &dca->exss.assets[0];
    enum AVMatrixEncoding matrix_encoding;
    int ret;

    // Handle downmixing to stereo request
    if (dca->request_channel_layout == DCA_SPEAKER_LAYOUT_STEREO
        && s->audio_mode > AMODE_MONO && s->prim_dmix_embedded
        && (s->prim_dmix_type == DCA_DMIX_TYPE_LoRo ||
            s->prim_dmix_type == DCA_DMIX_TYPE_LtRt))
        s->request_mask = DCA_SPEAKER_LAYOUT_STEREO;
    else
        s->request_mask = s->ch_mask;
    if (!ff_dca_set_channel_layout(avctx, s->ch_remap, s->request_mask))
        return AVERROR(EINVAL);

    // Force fixed point mode when falling back from XLL
    if ((avctx->flags & AV_CODEC_FLAG_BITEXACT) || ((dca->packet & DCA_PACKET_EXSS)
                                                    && (asset->extension_mask & DCA_EXSS_XLL)))
        ret = filter_frame_fixed(s, frame);
    else
        ret = filter_frame_float(s, frame);
    if (ret < 0)
        return ret;

    // Set profile, bit rate, etc
    if (s->ext_audio_mask & DCA_EXSS_MASK)
        avctx->profile = FF_PROFILE_DTS_HD_HRA;
    else if (s->ext_audio_mask & (DCA_CSS_XXCH | DCA_CSS_XCH))
        avctx->profile = FF_PROFILE_DTS_ES;
    else if (s->ext_audio_mask & DCA_CSS_X96)
        avctx->profile = FF_PROFILE_DTS_96_24;
    else
        avctx->profile = FF_PROFILE_DTS;

    if (s->bit_rate > 3 && !(s->ext_audio_mask & DCA_EXSS_MASK))
        avctx->bit_rate = s->bit_rate;
    else
        avctx->bit_rate = 0;

    if (s->audio_mode == AMODE_STEREO_TOTAL || (s->request_mask != s->ch_mask &&
                                                s->prim_dmix_type == DCA_DMIX_TYPE_LtRt))
        matrix_encoding = AV_MATRIX_ENCODING_DOLBY;
    else
        matrix_encoding = AV_MATRIX_ENCODING_NONE;
    if ((ret = ff_side_data_update_matrix_encoding(frame, matrix_encoding)) < 0)
        return ret;

    return 0;
}

av_cold void ff_dca_core_flush(DCACoreDecoder *s)
{
    if (s->subband_buffer) {
        erase_adpcm_history(s);
        memset(s->lfe_samples, 0, DCA_LFE_HISTORY * sizeof(int32_t));
    }

    if (s->x96_subband_buffer)
        erase_x96_adpcm_history(s);

    erase_dsp_history(s);
}

av_cold int ff_dca_core_init(DCACoreDecoder *s)
{
    if (!(s->float_dsp = avpriv_float_dsp_alloc(0)))
        return -1;
    if (!(s->fixed_dsp = avpriv_alloc_fixed_dsp(0)))
        return -1;

    ff_dcadct_init(&s->dcadct);
    if (ff_mdct_init(&s->imdct[0], 6, 1, 1.0) < 0)
        return -1;
    if (ff_mdct_init(&s->imdct[1], 7, 1, 1.0) < 0)
        return -1;
    ff_synth_filter_init(&s->synth);

    s->x96_rand = 1;
    return 0;
}

av_cold void ff_dca_core_close(DCACoreDecoder *s)
{
    av_freep(&s->float_dsp);
    av_freep(&s->fixed_dsp);

    ff_mdct_end(&s->imdct[0]);
    ff_mdct_end(&s->imdct[1]);

    av_freep(&s->subband_buffer);
    s->subband_size = 0;

    av_freep(&s->x96_subband_buffer);
    s->x96_subband_size = 0;

    av_freep(&s->output_buffer);
    s->output_size = 0;
}
OpenPOWER on IntegriCloud