aboutsummaryrefslogtreecommitdiffstats
path: root/Alc/effects/reverb.c
blob: 7f3dfb16964221bdb52020b3a6ae93089c9cddcc (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
/**
 * Ambisonic reverb engine for the OpenAL cross platform audio library
 * Copyright (C) 2008-2017 by Chris Robinson and Christopher Fitzgerald.
 * This library is free software; you can redistribute it and/or
 *  modify it under the terms of the GNU Library General Public
 *  License as published by the Free Software Foundation; either
 *  version 2 of the License, or (at your option) any later version.
 *
 * This library 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
 *  Library General Public License for more details.
 *
 * You should have received a copy of the GNU Library General Public
 *  License along with this library; if not, write to the
 *  Free Software Foundation, Inc.,
 *  51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
 * Or go to http://www.gnu.org/copyleft/lgpl.html
 */

#include "config.h"

#include <stdio.h>
#include <stdlib.h>
#include <math.h>

#include "alMain.h"
#include "alu.h"
#include "alAuxEffectSlot.h"
#include "alEffect.h"
#include "alFilter.h"
#include "alListener.h"
#include "alError.h"
#include "mixer_defs.h"

/* This is a user config option for modifying the overall output of the reverb
 * effect.
 */
ALfloat ReverbBoost = 1.0f;

/* Specifies whether to use a standard reverb effect in place of EAX reverb (no
 * high-pass, modulation, or echo).
 */
ALboolean EmulateEAXReverb = AL_FALSE;

/* This is the maximum number of samples processed for each inner loop
 * iteration. */
#define MAX_UPDATE_SAMPLES  256

/* The number of samples used for cross-faded delay lines.  This can be used
 * to balance the compensation for abrupt line changes and attenuation due to
 * minimally lengthed recursive lines.  Try to keep this below the device
 * update size.
 */
#define FADE_SAMPLES  128

#ifdef __GNUC__
#define UNEXPECTED(x) __builtin_expect((bool)(x), 0)
#else
#define UNEXPECTED(x) (x)
#endif

static RowMixerFunc MixRowSamples = MixRow_C;

static alonce_flag mixfunc_inited = AL_ONCE_FLAG_INIT;
static void init_mixfunc(void)
{
    MixRowSamples = SelectRowMixer();
}

/* The B-Format to A-Format conversion matrix. The arrangement of rows is
 * deliberately chosen to align the resulting lines to their spatial opposites
 * (0:above front left <-> 3:above back right, 1:below front right <-> 2:below
 * back left). It's not quite opposite, since the A-Format results in a
 * tetrahedron, but it's close enough. Should the model be extended to 8-lines
 * in the future, true opposites can be used.
 */
static const aluMatrixf B2A = {{
    { 0.288675134595f,  0.288675134595f,  0.288675134595f,  0.288675134595f },
    { 0.288675134595f, -0.288675134595f, -0.288675134595f,  0.288675134595f },
    { 0.288675134595f,  0.288675134595f, -0.288675134595f, -0.288675134595f },
    { 0.288675134595f, -0.288675134595f,  0.288675134595f, -0.288675134595f }
}};

/* Converts A-Format to B-Format. */
static const aluMatrixf A2B = {{
    { 0.866025403785f,  0.866025403785f,  0.866025403785f,  0.866025403785f },
    { 0.866025403785f, -0.866025403785f,  0.866025403785f, -0.866025403785f },
    { 0.866025403785f, -0.866025403785f, -0.866025403785f,  0.866025403785f },
    { 0.866025403785f,  0.866025403785f, -0.866025403785f, -0.866025403785f }
}};

static const ALfloat FadeStep = 1.0f / FADE_SAMPLES;

/* The all-pass and delay lines have a variable length dependent on the
 * effect's density parameter.  The resulting density multiplier is:
 *
 *     multiplier = 1 + (density * LINE_MULTIPLIER)
 *
 * Thus the line multiplier below will result in a maximum density multiplier
 * of 10.
 */
static const ALfloat LINE_MULTIPLIER = 9.0f;

/* All delay line lengths are specified in seconds.
 *
 * To approximate early reflections, we break them up into primary (those
 * arriving from the same direction as the source) and secondary (those
 * arriving from the opposite direction).
 *
 * The early taps decorrelate the 4-channel signal to approximate an average
 * room response for the primary reflections after the initial early delay.
 *
 * Given an average room dimension (d_a) and the speed of sound (c) we can
 * calculate the average reflection delay (r_a) regardless of listener and
 * source positions as:
 *
 *     r_a = d_a / c
 *     c   = 343.3
 *
 * This can extended to finding the average difference (r_d) between the
 * maximum (r_1) and minimum (r_0) reflection delays:
 *
 *     r_0 = 2 / 3 r_a
 *         = r_a - r_d / 2
 *         = r_d
 *     r_1 = 4 / 3 r_a
 *         = r_a + r_d / 2
 *         = 2 r_d
 *     r_d = 2 / 3 r_a
 *         = r_1 - r_0
 *
 * As can be determined by integrating the 1D model with a source (s) and
 * listener (l) positioned across the dimension of length (d_a):
 *
 *     r_d = int_(l=0)^d_a (int_(s=0)^d_a |2 d_a - 2 (l + s)| ds) dl / c
 *
 * The initial taps (T_(i=0)^N) are then specified by taking a power series
 * that ranges between r_0 and half of r_1 less r_0:
 *
 *     R_i = 2^(i / (2 N - 1)) r_d
 *         = r_0 + (2^(i / (2 N - 1)) - 1) r_d
 *         = r_0 + T_i
 *     T_i = R_i - r_0
 *         = (2^(i / (2 N - 1)) - 1) r_d
 *
 * Assuming an average of 5m (up to 50m with the density multiplier), we get
 * the following taps:
 */
static const ALfloat EARLY_TAP_LENGTHS[4] =
{
    0.000000e+0f, 1.010676e-3f, 2.126553e-3f, 3.358580e-3f
};

/* The early all-pass filter lengths are based on the early tap lengths:
 *
 *     A_i = R_i / a
 *
 * Where a is the approximate maximum all-pass cycle limit (20).
 */
static const ALfloat EARLY_ALLPASS_LENGTHS[4] =
{
    4.854840e-4f, 5.360178e-4f, 5.918117e-4f, 6.534130e-4f
};

/* The early delay lines are used to transform the primary reflections into
 * the secondary reflections.  The A-format is arranged in such a way that
 * the channels/lines are spatially opposite:
 *
 *     C_i is opposite C_(N-i-1)
 *
 * The delays of the two opposing reflections (R_i and O_i) from a source
 * anywhere along a particular dimension always sum to twice its full delay:
 *
 *     2 r_a = R_i + O_i
 *
 * With that in mind we can determine the delay between the two reflections
 * and thus specify our early line lengths (L_(i=0)^N) using:
 *
 *     O_i = 2 r_a - R_(N-i-1)
 *     L_i = O_i - R_(N-i-1)
 *         = 2 (r_a - R_(N-i-1))
 *         = 2 (r_a - T_(N-i-1) - r_0)
 *         = 2 r_a (1 - (2 / 3) 2^((N - i - 1) / (2 N - 1)))
 *
 * Using an average dimension of 5m, we get:
 */
static const ALfloat EARLY_LINE_LENGTHS[4] =
{
    2.992520e-3f, 5.456575e-3f, 7.688329e-3f, 9.709681e-3f
};

/* The late all-pass filter lengths are based on the late line lengths:
 *
 *     A_i = (5 / 3) L_i / r_1
 */
static const ALfloat LATE_ALLPASS_LENGTHS[4] =
{
    8.091400e-4f, 1.019453e-3f, 1.407968e-3f, 1.618280e-3f
};

/* The late lines are used to approximate the decaying cycle of recursive
 * late reflections.
 *
 * Splitting the lines in half, we start with the shortest reflection paths
 * (L_(i=0)^(N/2)):
 *
 *     L_i = 2^(i / (N - 1)) r_d
 *
 * Then for the opposite (longest) reflection paths (L_(i=N/2)^N):
 *
 *     L_i = 2 r_a - L_(i-N/2)
 *         = 2 r_a - 2^((i - N / 2) / (N - 1)) r_d
 *
 * For our 5m average room, we get:
 */
static const ALfloat LATE_LINE_LENGTHS[4] =
{
    9.709681e-3f, 1.223343e-2f, 1.689561e-2f, 1.941936e-2f
};

/* This coefficient is used to define the delay scale from the sinus, according
 * to the modulation depth property. This value must be below the shortest late
 * line length (0.0097), otherwise with certain parameters (high mod time, low
 * density) the downswing can sample before the input.
 */
static const ALfloat MODULATION_DEPTH_COEFF = 0.0032f;


/* Prior to VS2013, MSVC lacks the round() family of functions. */
#if defined(_MSC_VER) && _MSC_VER < 1800
static inline long lroundf(float val)
{
    if(val < 0.0)
        return fastf2i(ceilf(val-0.5f));
    return fastf2i(floorf(val+0.5f));
}
#endif


typedef struct DelayLineI {
    /* The delay lines use interleaved samples, with the lengths being powers
     * of 2 to allow the use of bit-masking instead of a modulus for wrapping.
     */
    ALsizei  Mask;
    ALfloat (*Line)[4];
} DelayLineI;

typedef struct VecAllpass {
    DelayLineI Delay;
    ALsizei Offset[4][2];
} VecAllpass;

typedef struct ALreverbState {
    DERIVE_FROM_TYPE(ALeffectState);

    ALboolean IsEax;

    /* All delay lines are allocated as a single buffer to reduce memory
     * fragmentation and management code.
     */
    ALfloat *SampleBuffer;
    ALuint   TotalSamples;

    /* Master effect filters */
    struct {
        ALfilterState Lp;
        ALfilterState Hp; /* EAX only */
    } Filter[4];

    /* Core delay line (early reflections and late reverb tap from this). */
    DelayLineI Delay;

    /* Tap points for early reflection delay. */
    ALsizei EarlyDelayTap[4][2];
    ALfloat EarlyDelayCoeff[4];

    /* Tap points for late reverb feed and delay. */
    ALsizei LateFeedTap;
    ALsizei LateDelayTap[4][2];

    /* The feed-back and feed-forward all-pass coefficient. */
    ALfloat ApFeedCoeff;

    /* Coefficients for the all-pass and line scattering matrices. */
    ALfloat MixX;
    ALfloat MixY;

    struct {
        /* A Gerzon vector all-pass filter is used to simulate initial
         * diffusion.  The spread from this filter also helps smooth out the
         * reverb tail.
         */
        VecAllpass VecAp;

        /* An echo line is used to complete the second half of the early
         * reflections.
         */
        DelayLineI Delay;
        ALsizei   Offset[4][2];
        ALfloat   Coeff[4];

        /* The gain for each output channel based on 3D panning. */
        ALfloat CurrentGain[4][MAX_OUTPUT_CHANNELS];
        ALfloat PanGain[4][MAX_OUTPUT_CHANNELS];
    } Early;

    struct {
        /* The vibrato time is tracked with an index over a modulus-wrapped
         * range (in samples).
         */
        ALsizei Index;
        ALsizei Range;
        ALfloat Scale;

        /* The depth of frequency change (also in samples) and its filter. */
        ALfloat Depth[2];
    } Mod; /* EAX only */

    struct {
        /* Attenuation to compensate for the modal density and decay rate of
         * the late lines.
         */
        ALfloat DensityGain;

        /* A recursive delay line is used fill in the reverb tail. */
        DelayLineI Delay;
        ALsizei   Offset[4][2];

        /* T60 decay filters are used to simulate absorption. */
        struct {
            ALfloat LFCoeffs[3];
            ALfloat HFCoeffs[3];
            ALfloat MidCoeff;
            /* The LF and HF filters keep a state of the last input and last
             * output sample.
             */
            ALfloat States[2][2];
        } Filters[4];

        /* A Gerzon vector all-pass filter is used to simulate diffusion. */
        VecAllpass VecAp;

        /* The gain for each output channel based on 3D panning. */
        ALfloat CurrentGain[4][MAX_OUTPUT_CHANNELS];
        ALfloat PanGain[4][MAX_OUTPUT_CHANNELS];
    } Late;

    /* Indicates the cross-fade point for delay line reads [0,FADE_SAMPLES]. */
    ALsizei FadeCount;

    /* The current write offset for all delay lines. */
    ALsizei Offset;

    /* Temporary storage used when processing. */
    alignas(16) ALsizei ModulationDelays[4][MAX_UPDATE_SAMPLES][2];
    alignas(16) ALfloat AFormatSamples[4][MAX_UPDATE_SAMPLES];
    alignas(16) ALfloat ReverbSamples[4][MAX_UPDATE_SAMPLES];
    alignas(16) ALfloat EarlySamples[4][MAX_UPDATE_SAMPLES];
} ALreverbState;

static ALvoid ALreverbState_Destruct(ALreverbState *State);
static ALboolean ALreverbState_deviceUpdate(ALreverbState *State, ALCdevice *Device);
static ALvoid ALreverbState_update(ALreverbState *State, const ALCcontext *Context, const ALeffectslot *Slot, const ALeffectProps *props);
static ALvoid ALreverbState_process(ALreverbState *State, ALsizei SamplesToDo, const ALfloat (*restrict SamplesIn)[BUFFERSIZE], ALfloat (*restrict SamplesOut)[BUFFERSIZE], ALsizei NumChannels);
DECLARE_DEFAULT_ALLOCATORS(ALreverbState)

DEFINE_ALEFFECTSTATE_VTABLE(ALreverbState);

static void ALreverbState_Construct(ALreverbState *state)
{
    ALsizei i, j;

    ALeffectState_Construct(STATIC_CAST(ALeffectState, state));
    SET_VTABLE2(ALreverbState, ALeffectState, state);

    state->IsEax = AL_FALSE;

    state->TotalSamples = 0;
    state->SampleBuffer = NULL;

    for(i = 0;i < 4;i++)
    {
        ALfilterState_clear(&state->Filter[i].Lp);
        ALfilterState_clear(&state->Filter[i].Hp);
    }

    state->Delay.Mask = 0;
    state->Delay.Line = NULL;

    for(i = 0;i < 4;i++)
    {
        state->EarlyDelayTap[i][0] = 0;
        state->EarlyDelayTap[i][1] = 0;
        state->EarlyDelayCoeff[i] = 0.0f;
    }

    state->LateFeedTap = 0;

    for(i = 0;i < 4;i++)
    {
        state->LateDelayTap[i][0] = 0;
        state->LateDelayTap[i][1] = 0;
    }

    state->ApFeedCoeff = 0.0f;
    state->MixX = 0.0f;
    state->MixY = 0.0f;

    state->Early.VecAp.Delay.Mask = 0;
    state->Early.VecAp.Delay.Line = NULL;
    state->Early.Delay.Mask = 0;
    state->Early.Delay.Line = NULL;
    for(i = 0;i < 4;i++)
    {
        state->Early.VecAp.Offset[i][0] = 0;
        state->Early.VecAp.Offset[i][1] = 0;
        state->Early.Offset[i][0] = 0;
        state->Early.Offset[i][1] = 0;
        state->Early.Coeff[i] = 0.0f;
    }

    state->Mod.Index = 0;
    state->Mod.Range = 1;
    state->Mod.Scale = 0.0f;
    state->Mod.Depth[0] = 0.0f;
    state->Mod.Depth[1] = 0.0f;

    state->Late.DensityGain = 0.0f;

    state->Late.Delay.Mask = 0;
    state->Late.Delay.Line = NULL;
    state->Late.VecAp.Delay.Mask = 0;
    state->Late.VecAp.Delay.Line = NULL;
    for(i = 0;i < 4;i++)
    {
        state->Late.Offset[i][0] = 0;
        state->Late.Offset[i][1] = 0;

        state->Late.VecAp.Offset[i][0] = 0;
        state->Late.VecAp.Offset[i][1] = 0;

        for(j = 0;j < 3;j++)
        {
            state->Late.Filters[i].LFCoeffs[j] = 0.0f;
            state->Late.Filters[i].HFCoeffs[j] = 0.0f;
        }
        state->Late.Filters[i].MidCoeff = 0.0f;

        state->Late.Filters[i].States[0][0] = 0.0f;
        state->Late.Filters[i].States[0][1] = 0.0f;
        state->Late.Filters[i].States[1][0] = 0.0f;
        state->Late.Filters[i].States[1][1] = 0.0f;
    }

    for(i = 0;i < 4;i++)
    {
        for(j = 0;j < MAX_OUTPUT_CHANNELS;j++)
        {
            state->Early.CurrentGain[i][j] = 0.0f;
            state->Early.PanGain[i][j] = 0.0f;
            state->Late.CurrentGain[i][j] = 0.0f;
            state->Late.PanGain[i][j] = 0.0f;
        }
    }

    state->FadeCount = 0;
    state->Offset = 0;
}

static ALvoid ALreverbState_Destruct(ALreverbState *State)
{
    al_free(State->SampleBuffer);
    State->SampleBuffer = NULL;

    ALeffectState_Destruct(STATIC_CAST(ALeffectState,State));
}

/**************************************
 *  Device Update                     *
 **************************************/

/* Given the allocated sample buffer, this function updates each delay line
 * offset.
 */
static inline ALvoid RealizeLineOffset(ALfloat *sampleBuffer, DelayLineI *Delay)
{
    union {
        ALfloat *f;
        ALfloat (*f4)[4];
    } u;
    u.f = &sampleBuffer[(ptrdiff_t)Delay->Line * 4];
    Delay->Line = u.f4;
}

/* Calculate the length of a delay line and store its mask and offset. */
static ALuint CalcLineLength(const ALfloat length, const ptrdiff_t offset, const ALuint frequency,
                             const ALuint extra, DelayLineI *Delay)
{
    ALuint samples;

    /* All line lengths are powers of 2, calculated from their lengths in
     * seconds, rounded up.
     */
    samples = fastf2i(ceilf(length*frequency));
    samples = NextPowerOf2(samples + extra);

    /* All lines share a single sample buffer. */
    Delay->Mask = samples - 1;
    Delay->Line = (ALfloat(*)[4])offset;

    /* Return the sample count for accumulation. */
    return samples;
}

/* Calculates the delay line metrics and allocates the shared sample buffer
 * for all lines given the sample rate (frequency).  If an allocation failure
 * occurs, it returns AL_FALSE.
 */
static ALboolean AllocLines(const ALuint frequency, ALreverbState *State)
{
    ALuint totalSamples, i;
    ALfloat multiplier, length;

    /* All delay line lengths are calculated to accomodate the full range of
     * lengths given their respective paramters.
     */
    totalSamples = 0;

    /* Multiplier for the maximum density value, i.e. density=1, which is
     * actually the least density...
     */
    multiplier = 1.0f + AL_EAXREVERB_MAX_DENSITY*LINE_MULTIPLIER;

    /* The main delay length includes the maximum early reflection delay, the
     * largest early tap width, the maximum late reverb delay, and the
     * largest late tap width.  Finally, it must also be extended by the
     * update size (MAX_UPDATE_SAMPLES) for block processing.
     */
    length = AL_EAXREVERB_MAX_REFLECTIONS_DELAY + EARLY_TAP_LENGTHS[3]*multiplier +
             AL_EAXREVERB_MAX_LATE_REVERB_DELAY +
             (LATE_LINE_LENGTHS[3] - LATE_LINE_LENGTHS[0])*0.25f*multiplier;
    totalSamples += CalcLineLength(length, totalSamples, frequency, MAX_UPDATE_SAMPLES,
                                   &State->Delay);

    /* The early vector all-pass line. */
    length = EARLY_ALLPASS_LENGTHS[3] * multiplier;
    totalSamples += CalcLineLength(length, totalSamples, frequency, 0,
                                   &State->Early.VecAp.Delay);

    /* The early reflection line. */
    length = EARLY_LINE_LENGTHS[3] * multiplier;
    totalSamples += CalcLineLength(length, totalSamples, frequency, 0,
                                   &State->Early.Delay);

    /* The late vector all-pass line. */
    length = LATE_ALLPASS_LENGTHS[3] * multiplier;
    totalSamples += CalcLineLength(length, totalSamples, frequency, 0,
                                   &State->Late.VecAp.Delay);

    /* The late delay lines are calculated from the larger of the maximum
     * density line length or the maximum echo time, and includes the maximum
     * modulation-related delay. The modulator's delay is calculated from the
     * maximum modulation time and depth coefficient, and halved for the low-
     * to-high frequency swing.
     */
    length = maxf(AL_EAXREVERB_MAX_ECHO_TIME, LATE_LINE_LENGTHS[3]*multiplier) +
             AL_EAXREVERB_MAX_MODULATION_TIME*MODULATION_DEPTH_COEFF/2.0f;
    totalSamples += CalcLineLength(length, totalSamples, frequency, 0,
                                   &State->Late.Delay);

    if(totalSamples != State->TotalSamples)
    {
        ALfloat *newBuffer;

        TRACE("New reverb buffer length: %ux4 samples\n", totalSamples);
        newBuffer = al_calloc(16, sizeof(ALfloat[4]) * totalSamples);
        if(!newBuffer) return AL_FALSE;

        al_free(State->SampleBuffer);
        State->SampleBuffer = newBuffer;
        State->TotalSamples = totalSamples;
    }

    /* Update all delays to reflect the new sample buffer. */
    RealizeLineOffset(State->SampleBuffer, &State->Delay);
    RealizeLineOffset(State->SampleBuffer, &State->Early.VecAp.Delay);
    RealizeLineOffset(State->SampleBuffer, &State->Early.Delay);
    RealizeLineOffset(State->SampleBuffer, &State->Late.VecAp.Delay);
    RealizeLineOffset(State->SampleBuffer, &State->Late.Delay);

    /* Clear the sample buffer. */
    for(i = 0;i < State->TotalSamples;i++)
        State->SampleBuffer[i] = 0.0f;

    return AL_TRUE;
}

static ALboolean ALreverbState_deviceUpdate(ALreverbState *State, ALCdevice *Device)
{
    ALuint frequency = Device->Frequency;
    ALfloat multiplier;

    /* Allocate the delay lines. */
    if(!AllocLines(frequency, State))
        return AL_FALSE;

    multiplier = 1.0f + AL_EAXREVERB_MAX_DENSITY*LINE_MULTIPLIER;

    /* The late feed taps are set a fixed position past the latest delay tap. */
    State->LateFeedTap = fastf2i((AL_EAXREVERB_MAX_REFLECTIONS_DELAY +
                                  EARLY_TAP_LENGTHS[3]*multiplier) *
                                 frequency);

    return AL_TRUE;
}

/**************************************
 *  Effect Update                     *
 **************************************/

/* Calculate a decay coefficient given the length of each cycle and the time
 * until the decay reaches -60 dB.
 */
static inline ALfloat CalcDecayCoeff(const ALfloat length, const ALfloat decayTime)
{
    return powf(REVERB_DECAY_GAIN, length/decayTime);
}

/* Calculate a decay length from a coefficient and the time until the decay
 * reaches -60 dB.
 */
static inline ALfloat CalcDecayLength(const ALfloat coeff, const ALfloat decayTime)
{
    return log10f(coeff) * decayTime / log10f(REVERB_DECAY_GAIN);
}

/* Calculate an attenuation to be applied to the input of any echo models to
 * compensate for modal density and decay time.
 */
static inline ALfloat CalcDensityGain(const ALfloat a)
{
    /* The energy of a signal can be obtained by finding the area under the
     * squared signal.  This takes the form of Sum(x_n^2), where x is the
     * amplitude for the sample n.
     *
     * Decaying feedback matches exponential decay of the form Sum(a^n),
     * where a is the attenuation coefficient, and n is the sample.  The area
     * under this decay curve can be calculated as:  1 / (1 - a).
     *
     * Modifying the above equation to find the area under the squared curve
     * (for energy) yields:  1 / (1 - a^2).  Input attenuation can then be
     * calculated by inverting the square root of this approximation,
     * yielding:  1 / sqrt(1 / (1 - a^2)), simplified to: sqrt(1 - a^2).
     */
    return sqrtf(1.0f - a*a);
}

/* Calculate the scattering matrix coefficients given a diffusion factor. */
static inline ALvoid CalcMatrixCoeffs(const ALfloat diffusion, ALfloat *x, ALfloat *y)
{
    ALfloat n, t;

    /* The matrix is of order 4, so n is sqrt(4 - 1). */
    n = sqrtf(3.0f);
    t = diffusion * atanf(n);

    /* Calculate the first mixing matrix coefficient. */
    *x = cosf(t);
    /* Calculate the second mixing matrix coefficient. */
    *y = sinf(t) / n;
}

/* Calculate the limited HF ratio for use with the late reverb low-pass
 * filters.
 */
static ALfloat CalcLimitedHfRatio(const ALfloat hfRatio, const ALfloat airAbsorptionGainHF,
                                  const ALfloat decayTime, const ALfloat SpeedOfSound)
{
    ALfloat limitRatio;

    /* Find the attenuation due to air absorption in dB (converting delay
     * time to meters using the speed of sound).  Then reversing the decay
     * equation, solve for HF ratio.  The delay length is cancelled out of
     * the equation, so it can be calculated once for all lines.
     */
    limitRatio = 1.0f / (CalcDecayLength(airAbsorptionGainHF, decayTime) * SpeedOfSound);

    /* Using the limit calculated above, apply the upper bound to the HF
     * ratio. Also need to limit the result to a minimum of 0.1, just like
     * the HF ratio parameter.
     */
    return clampf(limitRatio, 0.1f, hfRatio);
}

/* Calculates the first-order high-pass coefficients following the I3DL2
 * reference model.  This is the transfer function:
 *
 *                1 - z^-1
 *     H(z) = p ------------
 *               1 - p z^-1
 *
 * And this is the I3DL2 coefficient calculation given gain (g) and reference
 * angular frequency (w):
 *
 *                                    g
 *      p = ------------------------------------------------------
 *          g cos(w) + sqrt((cos(w) - 1) (g^2 cos(w) + g^2 - 2))
 *
 * The coefficient is applied to the partial differential filter equation as:
 *
 *     c_0 = p
 *     c_1 = -p
 *     c_2 = p
 *     y_i = c_0 x_i + c_1 x_(i-1) + c_2 y_(i-1)
 *
 */
static inline void CalcHighpassCoeffs(const ALfloat gain, const ALfloat w, ALfloat coeffs[3])
{
    ALfloat g, g2, cw, p;

    if(gain >= 1.0f)
    {
        coeffs[0] = 1.0f;
        coeffs[1] = 0.0f;
        coeffs[2] = 0.0f;

        return;
    }

    g = maxf(0.001f, gain);
    g2 = g * g;
    cw = cosf(w);
    p = g / (g*cw + sqrtf((cw - 1.0f) * (g2*cw + g2 - 2.0f)));

    coeffs[0] = p;
    coeffs[1] = -p;
    coeffs[2] = p;
}

/* Calculates the first-order low-pass coefficients following the I3DL2
 * reference model.  This is the transfer function:
 *
 *              (1 - a) z^0
 *     H(z) = ----------------
 *             1 z^0 - a z^-1
 *
 * And this is the I3DL2 coefficient calculation given gain (g) and reference
 * angular frequency (w):
 *
 *          1 - g^2 cos(w) - sqrt(2 g^2 (1 - cos(w)) - g^4 (1 - cos(w)^2))
 *     a = ----------------------------------------------------------------
 *                                    1 - g^2
 *
 * The coefficient is applied to the partial differential filter equation as:
 *
 *     c_0 = 1 - a
 *     c_1 = 0
 *     c_2 = a
 *     y_i = c_0 x_i + c_1 x_(i-1) + c_2 y_(i-1)
 *
 */
static inline void CalcLowpassCoeffs(const ALfloat gain, const ALfloat w, ALfloat coeffs[3])
{
    ALfloat g, g2, cw, a;

    if(gain >= 1.0f)
    {
        coeffs[0] = 1.0f;
        coeffs[1] = 0.0f;
        coeffs[2] = 0.0f;

        return;
    }

    /* Be careful with gains < 0.001, as that causes the coefficient
     * to head towards 1, which will flatten the signal. */
    g = maxf(0.001f, gain);
    g2 = g * g;
    cw = cosf(w);
    a = (1.0f - g2*cw - sqrtf((2.0f*g2*(1.0f - cw)) - g2*g2*(1.0f - cw*cw))) /
        (1.0f - g2);

    coeffs[0] = 1.0f - a;
    coeffs[1] = 0.0f;
    coeffs[2] = a;
}

/* Calculates the first-order low-shelf coefficients.  The shelf filters are
 * used in place of low/high-pass filters to preserve the mid-band.  This is
 * the transfer function:
 *
 *             a_0 + a_1 z^-1
 *     H(z) = ----------------
 *              1 + b_1 z^-1
 *
 * And these are the coefficient calculations given cut gain (g) and a center
 * angular frequency (w):
 *
 *          sin(0.5 (pi - w) - 0.25 pi)
 *     p = -----------------------------
 *          sin(0.5 (pi - w) + 0.25 pi)
 *
 *          g + 1           g + 1
 *     a = ------- + sqrt((-------)^2 - 1)
 *          g - 1           g - 1
 *
 *            1 + g + (1 - g) a
 *     b_0 = -------------------
 *                    2
 *
 *            1 - g + (1 + g) a
 *     b_1 = -------------------
 *                    2
 *
 * The coefficients are applied to the partial differential filter equation
 * as:
 *
 *            b_0 + p b_1
 *     c_0 = -------------
 *              1 + p a
 *
 *            -(b_1 + p b_0)
 *     c_1 = ----------------
 *               1 + p a
 *
 *             p + a
 *     c_2 = ---------
 *            1 + p a
 *
 *     y_i = c_0 x_i + c_1 x_(i-1) + c_2 y_(i-1)
 *
 */
static inline void CalcLowShelfCoeffs(const ALfloat gain, const ALfloat w, ALfloat coeffs[3])
{
    ALfloat g, rw, p, n;
    ALfloat alpha, beta0, beta1;

    if(gain >= 1.0f)
    {
        coeffs[0] = 1.0f;
        coeffs[1] = 0.0f;
        coeffs[2] = 0.0f;

        return;
    }

    g = maxf(0.001f, gain);
    rw = F_PI - w;
    p = sinf(0.5f*rw - 0.25f*F_PI) / sinf(0.5f*rw + 0.25f*F_PI);
    n = (g + 1.0f) / (g - 1.0f);
    alpha = n + sqrtf(n*n - 1.0f);
    beta0 = (1.0f + g + (1.0f - g)*alpha) / 2.0f;
    beta1 = (1.0f - g + (1.0f + g)*alpha) / 2.0f;

    coeffs[0] = (beta0 + p*beta1) / (1.0f + p*alpha);
    coeffs[1] = -(beta1 + p*beta0) / (1.0f + p*alpha);
    coeffs[2] = (p + alpha) / (1.0f + p*alpha);
}

/* Calculates the first-order high-shelf coefficients.  The shelf filters are
 * used in place of low/high-pass filters to preserve the mid-band.  This is
 * the transfer function:
 *
 *             a_0 + a_1 z^-1
 *     H(z) = ----------------
 *              1 + b_1 z^-1
 *
 * And these are the coefficient calculations given cut gain (g) and a center
 * angular frequency (w):
 *
 *          sin(0.5 w - 0.25 pi)
 *     p = ----------------------
 *          sin(0.5 w + 0.25 pi)
 *
 *          g + 1           g + 1
 *     a = ------- + sqrt((-------)^2 - 1)
 *          g - 1           g - 1
 *
 *            1 + g + (1 - g) a
 *     b_0 = -------------------
 *                    2
 *
 *            1 - g + (1 + g) a
 *     b_1 = -------------------
 *                    2
 *
 * The coefficients are applied to the partial differential filter equation
 * as:
 *
 *            b_0 + p b_1
 *     c_0 = -------------
 *              1 + p a
 *
 *            b_1 + p b_0
 *     c_1 = -------------
 *              1 + p a
 *
 *            -(p + a)
 *     c_2 = ----------
 *            1 + p a
 *
 *     y_i = c_0 x_i + c_1 x_(i-1) + c_2 y_(i-1)
 *
 */
static inline void CalcHighShelfCoeffs(const ALfloat gain, const ALfloat w, ALfloat coeffs[3])
{
    ALfloat g, p, n;
    ALfloat alpha, beta0, beta1;

    if(gain >= 1.0f)
    {
        coeffs[0] = 1.0f;
        coeffs[1] = 0.0f;
        coeffs[2] = 0.0f;

        return;
    }

    g = maxf(0.001f, gain);
    p = sinf(0.5f*w - 0.25f*F_PI) / sinf(0.5f*w + 0.25f*F_PI);
    n = (g + 1.0f) / (g - 1.0f);
    alpha = n + sqrtf(n*n - 1.0f);
    beta0 = (1.0f + g + (1.0f - g)*alpha) / 2.0f;
    beta1 = (1.0f - g + (1.0f + g)*alpha) / 2.0f;

    coeffs[0] = (beta0 + p*beta1) / (1.0f + p*alpha);
    coeffs[1] = (beta1 + p*beta0) / (1.0f + p*alpha);
    coeffs[2] = -(p + alpha) / (1.0f + p*alpha);
}

/* Calculates the 3-band T60 damping coefficients for a particular delay line
 * of specified length using a combination of two low/high-pass/shelf or
 * pass-through filter sections (producing 3 coefficients each) and a general
 * gain (7th coefficient) given decay times for each band split at two (LF/
 * HF) reference frequencies (w).
 */
static void CalcT60DampingCoeffs(const ALfloat length, const ALfloat lfDecayTime,
                                 const ALfloat mfDecayTime, const ALfloat hfDecayTime,
                                 const ALfloat lfW, const ALfloat hfW, ALfloat lfcoeffs[3],
                                 ALfloat hfcoeffs[3], ALfloat *midcoeff)
{
    ALfloat lfGain = CalcDecayCoeff(length, lfDecayTime);
    ALfloat mfGain = CalcDecayCoeff(length, mfDecayTime);
    ALfloat hfGain = CalcDecayCoeff(length, hfDecayTime);

    if(lfGain < mfGain)
    {
        if(mfGain < hfGain)
        {
            CalcLowShelfCoeffs(mfGain / hfGain, hfW, lfcoeffs);
            CalcHighpassCoeffs(lfGain / mfGain, lfW, hfcoeffs);
            *midcoeff = hfGain;
        }
        else if(mfGain > hfGain)
        {
            CalcHighpassCoeffs(lfGain / mfGain, lfW, lfcoeffs);
            CalcLowpassCoeffs(hfGain / mfGain, hfW, hfcoeffs);
            *midcoeff = mfGain;
        }
        else
        {
            lfcoeffs[0] = 1.0f;
            lfcoeffs[1] = 0.0f;
            lfcoeffs[2] = 0.0f;
            CalcHighpassCoeffs(lfGain / mfGain, lfW, hfcoeffs);
            *midcoeff = mfGain;
        }
    }
    else if(lfGain > mfGain)
    {
        if(mfGain < hfGain)
        {
            ALfloat hg = mfGain / lfGain;
            ALfloat lg = mfGain / hfGain;

            CalcHighShelfCoeffs(hg, lfW, lfcoeffs);
            CalcLowShelfCoeffs(lg, hfW, hfcoeffs);
            *midcoeff = maxf(lfGain, hfGain) / maxf(hg, lg);
        }
        else if(mfGain > hfGain)
        {
            CalcHighShelfCoeffs(mfGain / lfGain, lfW, lfcoeffs);
            CalcLowpassCoeffs(hfGain / mfGain, hfW, hfcoeffs);
            *midcoeff = lfGain;
        }
        else
        {
            lfcoeffs[0] = 1.0f;
            lfcoeffs[1] = 0.0f;
            lfcoeffs[2] = 0.0f;
            CalcHighShelfCoeffs(mfGain / lfGain, lfW, hfcoeffs);
            *midcoeff = lfGain;
        }
    }
    else
    {
        lfcoeffs[0] = 1.0f;
        lfcoeffs[1] = 0.0f;
        lfcoeffs[2] = 0.0f;

        if(mfGain < hfGain)
        {
            CalcLowShelfCoeffs(mfGain / hfGain, hfW, hfcoeffs);
            *midcoeff = hfGain;
        }
        else if(mfGain > hfGain)
        {
            CalcLowpassCoeffs(hfGain / mfGain, hfW, hfcoeffs);
            *midcoeff = mfGain;
        }
        else
        {
            hfcoeffs[3] = 1.0f;
            hfcoeffs[4] = 0.0f;
            hfcoeffs[5] = 0.0f;
            *midcoeff = mfGain;
        }
    }
}

/* Update the EAX modulation index, range, and depth.  Keep in mind that this
 * kind of vibrato is additive and not multiplicative as one may expect.  The
 * downswing will sound stronger than the upswing.
 */
static ALvoid UpdateModulator(const ALfloat modTime, const ALfloat modDepth,
                              const ALuint frequency, ALreverbState *State)
{
    ALsizei range;

    /* Modulation is calculated in two parts.
     *
     * The modulation time effects the speed of the sinus. An index out of the
     * current range (both in samples) is incremented each sample, so a longer
     * time implies a larger range. The range is bound to a reasonable minimum
     * (1 sample) and when the timing changes, the index is rescaled to the new
     * range to keep the sinus consistent.
     */
    range = fastf2i(modTime*frequency + 0.5f);
    State->Mod.Index = (ALsizei)(State->Mod.Index * (ALint64)range /
                                 State->Mod.Range) % range;
    State->Mod.Range = range;
    State->Mod.Scale = F_TAU / range;

    /* The modulation depth effects the scale of the sinus, which varies the
     * delay for the tapped output. This delay changing over time changes the
     * pitch, creating the modulation effect. The scale needs to be multiplied
     * by the modulation time (itself scaled by the max modulation time) so
     * that a given depth produces a consistent shift in frequency over all
     * ranges of time.
     */
    State->Mod.Depth[1] = modDepth * MODULATION_DEPTH_COEFF *
                          (modTime / AL_EAXREVERB_MAX_MODULATION_TIME) *
                          frequency * FRACTIONONE;
}

/* Update the offsets for the main effect delay line. */
static ALvoid UpdateDelayLine(const ALfloat earlyDelay, const ALfloat lateDelay, const ALfloat density, const ALfloat decayTime, const ALuint frequency, ALreverbState *State)
{
    ALfloat multiplier, length;
    ALuint i;

    multiplier = 1.0f + density*LINE_MULTIPLIER;

    /* Early reflection taps are decorrelated by means of an average room
     * reflection approximation described above the definition of the taps.
     * This approximation is linear and so the above density multiplier can
     * be applied to adjust the width of the taps.  A single-band decay
     * coefficient is applied to simulate initial attenuation and absorption.
     *
     * Late reverb taps are based on the late line lengths to allow a zero-
     * delay path and offsets that would continue the propagation naturally
     * into the late lines.
     */
    for(i = 0;i < 4;i++)
    {
        length = earlyDelay + EARLY_TAP_LENGTHS[i]*multiplier;
        State->EarlyDelayTap[i][1] = fastf2i(length * frequency);

        length = EARLY_TAP_LENGTHS[i]*multiplier;
        State->EarlyDelayCoeff[i] = CalcDecayCoeff(length, decayTime);

        length = lateDelay + (LATE_LINE_LENGTHS[i] - LATE_LINE_LENGTHS[0])*0.25f*multiplier;
        State->LateDelayTap[i][1] = State->LateFeedTap + fastf2i(length * frequency);
    }
}

/* Update the early reflection line lengths and gain coefficients. */
static ALvoid UpdateEarlyLines(const ALfloat density, const ALfloat decayTime, const ALuint frequency, ALreverbState *State)
{
    ALfloat multiplier, length;
    ALsizei i;

    multiplier = 1.0f + density*LINE_MULTIPLIER;

    for(i = 0;i < 4;i++)
    {
        /* Calculate the length (in seconds) of each all-pass line. */
        length = EARLY_ALLPASS_LENGTHS[i] * multiplier;

        /* Calculate the delay offset for each all-pass line. */
        State->Early.VecAp.Offset[i][1] = fastf2i(length * frequency);

        /* Calculate the length (in seconds) of each delay line. */
        length = EARLY_LINE_LENGTHS[i] * multiplier;

        /* Calculate the delay offset for each delay line. */
        State->Early.Offset[i][1] = fastf2i(length * frequency);

        /* Calculate the gain (coefficient) for each line. */
        State->Early.Coeff[i] = CalcDecayCoeff(length, decayTime);
    }
}

/* Update the late reverb line lengths and T60 coefficients. */
static ALvoid UpdateLateLines(const ALfloat density, const ALfloat diffusion, const ALfloat lfDecayTime, const ALfloat mfDecayTime, const ALfloat hfDecayTime, const ALfloat lfW, const ALfloat hfW, const ALfloat echoTime, const ALfloat echoDepth, const ALuint frequency, ALreverbState *State)
{
    ALfloat multiplier, length, bandWeights[3];
    ALsizei i;

    /* To compensate for changes in modal density and decay time of the late
     * reverb signal, the input is attenuated based on the maximal energy of
     * the outgoing signal.  This approximation is used to keep the apparent
     * energy of the signal equal for all ranges of density and decay time.
     *
     * The average length of the delay lines is used to calculate the
     * attenuation coefficient.
     */
    multiplier = 1.0f + density*LINE_MULTIPLIER;
    length = (LATE_LINE_LENGTHS[0] + LATE_LINE_LENGTHS[1] +
              LATE_LINE_LENGTHS[2] + LATE_LINE_LENGTHS[3]) / 4.0f * multiplier;
    /* Include the echo transformation (see below). */
    length = lerp(length, echoTime, echoDepth);
    length += (LATE_ALLPASS_LENGTHS[0] + LATE_ALLPASS_LENGTHS[1] +
               LATE_ALLPASS_LENGTHS[2] + LATE_ALLPASS_LENGTHS[3]) / 4.0f * multiplier;
    /* The density gain calculation uses an average decay time weighted by
     * approximate bandwidth.  This attempts to compensate for losses of
     * energy that reduce decay time due to scattering into highly attenuated
     * bands.
     */
    bandWeights[0] = lfW;
    bandWeights[1] = hfW - lfW;
    bandWeights[2] = F_TAU - hfW;
    State->Late.DensityGain = CalcDensityGain(
        CalcDecayCoeff(length, (bandWeights[0]*lfDecayTime + bandWeights[1]*mfDecayTime +
                                bandWeights[2]*hfDecayTime) / F_TAU)
    );

    for(i = 0;i < 4;i++)
    {
        /* Calculate the length (in seconds) of each all-pass line. */
        length = LATE_ALLPASS_LENGTHS[i] * multiplier;

        /* Calculate the delay offset for each all-pass line. */
        State->Late.VecAp.Offset[i][1] = fastf2i(length * frequency);

        /* Calculate the length (in seconds) of each delay line.  This also
         * applies the echo transformation.  As the EAX echo depth approaches
         * 1, the line lengths approach a length equal to the echoTime.  This
         * helps to produce distinct echoes along the tail.
         */
        length = lerp(LATE_LINE_LENGTHS[i] * multiplier, echoTime, echoDepth);

        /* Calculate the delay offset for each delay line. */
        State->Late.Offset[i][1] = fastf2i(length*frequency*FRACTIONONE + 0.5f);

        /* Approximate the absorption that the vector all-pass would exhibit
         * given the current diffusion so we don't have to process a full T60
         * filter for each of its four lines.
         */
        length += lerp(LATE_ALLPASS_LENGTHS[i],
                       (LATE_ALLPASS_LENGTHS[0] + LATE_ALLPASS_LENGTHS[1] +
                        LATE_ALLPASS_LENGTHS[2] + LATE_ALLPASS_LENGTHS[3]) / 4.0f,
                       diffusion) * multiplier;

        /* Calculate the T60 damping coefficients for each line. */
        CalcT60DampingCoeffs(length, lfDecayTime, mfDecayTime, hfDecayTime,
                             lfW, hfW, State->Late.Filters[i].LFCoeffs,
                             State->Late.Filters[i].HFCoeffs,
                             &State->Late.Filters[i].MidCoeff);
    }
}

/* Creates a transform matrix given a reverb vector. This works by creating a
 * Z-focus transform, then a rotate transform around X, then Y, to place the
 * focal point in the direction of the vector, using the vector length as a
 * focus strength.
 *
 * This isn't technically correct since the vector is supposed to define the
 * aperture and not rotate the perceived soundfield, but in practice it's
 * probably good enough.
 */
static aluMatrixf GetTransformFromVector(const ALfloat *vec)
{
    aluMatrixf zfocus, xrot, yrot;
    aluMatrixf tmp1, tmp2;
    ALfloat length;
    ALfloat sa, a;

    length = sqrtf(vec[0]*vec[0] + vec[1]*vec[1] + vec[2]*vec[2]);

    /* Define a Z-focus (X in Ambisonics) transform, given the panning vector
     * length.
     */
    sa = sinf(minf(length, 1.0f) * (F_PI/4.0f));
    aluMatrixfSet(&zfocus,
                     1.0f/(1.0f+sa),                       0.0f,                       0.0f, (sa/(1.0f+sa))/1.732050808f,
                               0.0f, sqrtf((1.0f-sa)/(1.0f+sa)),                       0.0f,                        0.0f,
                               0.0f,                       0.0f, sqrtf((1.0f-sa)/(1.0f+sa)),                        0.0f,
        (sa/(1.0f+sa))*1.732050808f,                       0.0f,                       0.0f,              1.0f/(1.0f+sa)
    );

    /* Define rotation around X (Y in Ambisonics) */
    a = atan2f(vec[1], sqrtf(vec[0]*vec[0] + vec[2]*vec[2]));
    aluMatrixfSet(&xrot,
        1.0f, 0.0f,     0.0f,    0.0f,
        0.0f, 1.0f,     0.0f,    0.0f,
        0.0f, 0.0f,  cosf(a), sinf(a),
        0.0f, 0.0f, -sinf(a), cosf(a)
    );

    /* Define rotation around Y (Z in Ambisonics). NOTE: EFX's reverb vectors
     * use a right-handled coordinate system, compared to the rest of OpenAL
     * which uses left-handed. This is fixed by negating Z, however it would
     * need to also be negated to get a proper Ambisonics angle, thus
     * cancelling it out.
     */
    a = atan2f(-vec[0], vec[2]);
    aluMatrixfSet(&yrot,
        1.0f,     0.0f, 0.0f,    0.0f,
        0.0f,  cosf(a), 0.0f, sinf(a),
        0.0f,     0.0f, 1.0f,    0.0f,
        0.0f, -sinf(a), 0.0f, cosf(a)
    );

#define MATRIX_MULT(_res, _m1, _m2) do {                                      \
    int row, col;                                                             \
    for(col = 0;col < 4;col++)                                                \
    {                                                                         \
        for(row = 0;row < 4;row++)                                            \
            _res.m[row][col] = _m1.m[row][0]*_m2.m[0][col] + _m1.m[row][1]*_m2.m[1][col] + \
                               _m1.m[row][2]*_m2.m[2][col] + _m1.m[row][3]*_m2.m[3][col];  \
    }                                                                         \
} while(0)
    /* Define a matrix that first focuses on Z, then rotates around X then Y to
     * focus the output in the direction of the vector.
     */
    MATRIX_MULT(tmp1, xrot, zfocus);
    MATRIX_MULT(tmp2, yrot, tmp1);
#undef MATRIX_MULT

    return tmp2;
}

/* Update the early and late 3D panning gains. */
static ALvoid Update3DPanning(const ALCdevice *Device, const ALfloat *ReflectionsPan, const ALfloat *LateReverbPan, const ALfloat gain, const ALfloat earlyGain, const ALfloat lateGain, ALreverbState *State)
{
    aluMatrixf transform, rot;
    ALsizei i;

    STATIC_CAST(ALeffectState,State)->OutBuffer = Device->FOAOut.Buffer;
    STATIC_CAST(ALeffectState,State)->OutChannels = Device->FOAOut.NumChannels;

    /* Note: _res is transposed. */
#define MATRIX_MULT(_res, _m1, _m2) do {                                                   \
    int row, col;                                                                          \
    for(col = 0;col < 4;col++)                                                             \
    {                                                                                      \
        for(row = 0;row < 4;row++)                                                         \
            _res.m[col][row] = _m1.m[row][0]*_m2.m[0][col] + _m1.m[row][1]*_m2.m[1][col] + \
                               _m1.m[row][2]*_m2.m[2][col] + _m1.m[row][3]*_m2.m[3][col];  \
    }                                                                                      \
} while(0)
    /* Create a matrix that first converts A-Format to B-Format, then rotates
     * the B-Format soundfield according to the panning vector.
     */
    rot = GetTransformFromVector(ReflectionsPan);
    MATRIX_MULT(transform, rot, A2B);
    memset(&State->Early.PanGain, 0, sizeof(State->Early.PanGain));
    for(i = 0;i < MAX_EFFECT_CHANNELS;i++)
        ComputeFirstOrderGains(Device->FOAOut, transform.m[i], gain*earlyGain, State->Early.PanGain[i]);

    rot = GetTransformFromVector(LateReverbPan);
    MATRIX_MULT(transform, rot, A2B);
    memset(&State->Late.PanGain, 0, sizeof(State->Late.PanGain));
    for(i = 0;i < MAX_EFFECT_CHANNELS;i++)
        ComputeFirstOrderGains(Device->FOAOut, transform.m[i], gain*lateGain, State->Late.PanGain[i]);
#undef MATRIX_MULT
}

static ALvoid ALreverbState_update(ALreverbState *State, const ALCcontext *Context, const ALeffectslot *Slot, const ALeffectProps *props)
{
    const ALCdevice *Device = Context->Device;
    const ALlistener *Listener = Context->Listener;
    ALuint frequency = Device->Frequency;
    ALfloat lfScale, hfScale, hfRatio;
    ALfloat lfDecayTime, hfDecayTime;
    ALfloat gain, gainlf, gainhf;
    ALsizei i;

    if(Slot->Params.EffectType == AL_EFFECT_EAXREVERB && !EmulateEAXReverb)
        State->IsEax = AL_TRUE;
    else if(Slot->Params.EffectType == AL_EFFECT_REVERB || EmulateEAXReverb)
        State->IsEax = AL_FALSE;

    /* Calculate the master filters */
    hfScale = props->Reverb.HFReference / frequency;
    /* Restrict the filter gains from going below -60dB to keep the filter from
     * killing most of the signal.
     */
    gainhf = maxf(props->Reverb.GainHF, 0.001f);
    ALfilterState_setParams(&State->Filter[0].Lp, ALfilterType_HighShelf,
                            gainhf, hfScale, calc_rcpQ_from_slope(gainhf, 1.0f));
    lfScale = props->Reverb.LFReference / frequency;
    gainlf = maxf(props->Reverb.GainLF, 0.001f);
    ALfilterState_setParams(&State->Filter[0].Hp, ALfilterType_LowShelf,
                            gainlf, lfScale, calc_rcpQ_from_slope(gainlf, 1.0f));
    for(i = 1;i < 4;i++)
    {
        ALfilterState_copyParams(&State->Filter[i].Lp, &State->Filter[0].Lp);
        ALfilterState_copyParams(&State->Filter[i].Hp, &State->Filter[0].Hp);
    }

    /* Update the main effect delay and associated taps. */
    UpdateDelayLine(props->Reverb.ReflectionsDelay, props->Reverb.LateReverbDelay,
                    props->Reverb.Density, props->Reverb.DecayTime, frequency,
                    State);

    /* Calculate the all-pass feed-back/forward coefficient. */
    State->ApFeedCoeff = sqrtf(0.5f) * powf(props->Reverb.Diffusion, 2.0f);

    /* Update the early lines. */
    UpdateEarlyLines(props->Reverb.Density, props->Reverb.DecayTime,
                     frequency, State);

    /* Get the mixing matrix coefficients. */
    CalcMatrixCoeffs(props->Reverb.Diffusion, &State->MixX, &State->MixY);

    /* If the HF limit parameter is flagged, calculate an appropriate limit
     * based on the air absorption parameter.
     */
    hfRatio = props->Reverb.DecayHFRatio;
    if(props->Reverb.DecayHFLimit && props->Reverb.AirAbsorptionGainHF < 1.0f)
        hfRatio = CalcLimitedHfRatio(hfRatio, props->Reverb.AirAbsorptionGainHF,
            props->Reverb.DecayTime, Listener->Params.ReverbSpeedOfSound
        );

    /* Calculate the LF/HF decay times. */
    lfDecayTime = clampf(props->Reverb.DecayTime * props->Reverb.DecayLFRatio,
                         AL_EAXREVERB_MIN_DECAY_TIME, AL_EAXREVERB_MAX_DECAY_TIME);
    hfDecayTime = clampf(props->Reverb.DecayTime * hfRatio,
                         AL_EAXREVERB_MIN_DECAY_TIME, AL_EAXREVERB_MAX_DECAY_TIME);

    /* Update the modulator line. */
    UpdateModulator(props->Reverb.ModulationTime, props->Reverb.ModulationDepth,
                    frequency, State);

    /* Update the late lines. */
    UpdateLateLines(props->Reverb.Density, props->Reverb.Diffusion,
                    lfDecayTime, props->Reverb.DecayTime, hfDecayTime,
                    F_TAU * lfScale, F_TAU * hfScale,
                    props->Reverb.EchoTime, props->Reverb.EchoDepth,
                    frequency, State);

    /* Update early and late 3D panning. */
    gain = props->Reverb.Gain * Slot->Params.Gain * ReverbBoost;
    Update3DPanning(Device, props->Reverb.ReflectionsPan,
                    props->Reverb.LateReverbPan, gain,
                    props->Reverb.ReflectionsGain,
                    props->Reverb.LateReverbGain, State);

    /* Determine if delay-line cross-fading is required. */
    for(i = 0;i < 4;i++)
    {
        if((State->EarlyDelayTap[i][1] != State->EarlyDelayTap[i][0]) ||
           (State->Early.VecAp.Offset[i][1] != State->Early.VecAp.Offset[i][0]) ||
           (State->Early.Offset[i][1] != State->Early.Offset[i][0]) ||
           (State->LateDelayTap[i][1] != State->LateDelayTap[i][0]) ||
           (State->Late.VecAp.Offset[i][1] != State->Late.VecAp.Offset[i][0]) ||
           (State->Late.Offset[i][1] != State->Late.Offset[i][0]) ||
           (State->Mod.Depth[1] != State->Mod.Depth[0]))
        {
            State->FadeCount = 0;
            break;
        }
    }
}


/**************************************
 *  Effect Processing                 *
 **************************************/

/* Basic delay line input/output routines. */
static inline ALfloat DelayLineOut(const DelayLineI *Delay, const ALsizei offset, const ALsizei c)
{
    return Delay->Line[offset&Delay->Mask][c];
}

/* Cross-faded delay line output routine.  Instead of interpolating the
 * offsets, this interpolates (cross-fades) the outputs at each offset.
 */
static inline ALfloat FadedDelayLineOut(const DelayLineI *Delay, const ALsizei off0,
                                        const ALsizei off1, const ALsizei c, const ALfloat mu)
{
    return lerp(Delay->Line[off0&Delay->Mask][c], Delay->Line[off1&Delay->Mask][c], mu);
}
#define DELAY_OUT_Faded(d, o0, o1, c, mu) FadedDelayLineOut(d, o0, o1, c, mu)
#define DELAY_OUT_Unfaded(d, o0, o1, c, mu) DelayLineOut(d, o0, c)

static inline ALvoid DelayLineIn(DelayLineI *Delay, const ALsizei offset, const ALsizei c, const ALfloat in)
{
    Delay->Line[offset&Delay->Mask][c] = in;
}

static inline ALvoid DelayLineIn4(DelayLineI *Delay, ALsizei offset, const ALfloat in[4])
{
    ALsizei i;
    offset &= Delay->Mask;
    for(i = 0;i < 4;i++)
        Delay->Line[offset][i] = in[i];
}

static inline ALvoid DelayLineIn4Rev(DelayLineI *Delay, ALsizei offset, const ALfloat in[4])
{
    ALsizei i;
    offset &= Delay->Mask;
    for(i = 0;i < 4;i++)
        Delay->Line[offset][i] = in[3-i];
}

static void CalcModulationDelays(ALreverbState *State,
                                 ALsizei (*restrict delays)[MAX_UPDATE_SAMPLES][2],
                                 const ALsizei (*restrict offsets)[2], const ALsizei todo)
{
    ALsizei phase_offset = State->Mod.Range >> 2;
    ALsizei index, c, i;
    ALfloat sinus;

    for(c = 0;c < 4;c++)
    {
        index = State->Mod.Index + phase_offset*c;
        for(i = 0;i < todo;i++)
        {
            /* Calculate the sinus rhythm (dependent on modulation time and the
             * sampling rate).
             */
            sinus = sinf(index * State->Mod.Scale) + 1.0f;
            index = (index+1) % State->Mod.Range;

            /* Calculate the read offset. */
            delays[c][i][0] = fastf2i(sinus*State->Mod.Depth[0]) + offsets[c][0];
            delays[c][i][1] = fastf2i(sinus*State->Mod.Depth[1]) + offsets[c][1];
        }
    }
    State->Mod.Index = (State->Mod.Index+todo) % State->Mod.Range;
}

/* Applies a scattering matrix to the 4-line (vector) input.  This is used
 * for both the below vector all-pass model and to perform modal feed-back
 * delay network (FDN) mixing.
 *
 * The matrix is derived from a skew-symmetric matrix to form a 4D rotation
 * matrix with a single unitary rotational parameter:
 *
 *     [  d,  a,  b,  c ]          1 = a^2 + b^2 + c^2 + d^2
 *     [ -a,  d,  c, -b ]
 *     [ -b, -c,  d,  a ]
 *     [ -c,  b, -a,  d ]
 *
 * The rotation is constructed from the effect's diffusion parameter,
 * yielding:
 *
 *     1 = x^2 + 3 y^2
 *
 * Where a, b, and c are the coefficient y with differing signs, and d is the
 * coefficient x.  The final matrix is thus:
 *
 *     [  x,  y, -y,  y ]          n = sqrt(matrix_order - 1)
 *     [ -y,  x,  y,  y ]          t = diffusion_parameter * atan(n)
 *     [  y, -y,  x,  y ]          x = cos(t)
 *     [ -y, -y, -y,  x ]          y = sin(t) / n
 *
 * Any square orthogonal matrix with an order that is a power of two will
 * work (where ^T is transpose, ^-1 is inverse):
 *
 *     M^T = M^-1
 *
 * Using that knowledge, finding an appropriate matrix can be accomplished
 * naively by searching all combinations of:
 *
 *     M = D + S - S^T
 *
 * Where D is a diagonal matrix (of x), and S is a triangular matrix (of y)
 * whose combination of signs are being iterated.
 */
static inline void VectorPartialScatter(ALfloat *restrict vec, const ALfloat xCoeff, const ALfloat yCoeff)
{
    const ALfloat f[4] = { vec[0], vec[1], vec[2], vec[3] };

    vec[0] = xCoeff*f[0] + yCoeff*(         f[1] + -f[2] +  f[3]);
    vec[1] = xCoeff*f[1] + yCoeff*(-f[0]         +  f[2] +  f[3]);
    vec[2] = xCoeff*f[2] + yCoeff*( f[0] + -f[1]         +  f[3]);
    vec[3] = xCoeff*f[3] + yCoeff*(-f[0] + -f[1] + -f[2]        );
}

/* This applies a Gerzon multiple-in/multiple-out (MIMO) vector all-pass
 * filter to the 4-line input.
 *
 * It works by vectorizing a regular all-pass filter and replacing the delay
 * element with a scattering matrix (like the one above) and a diagonal
 * matrix of delay elements.
 *
 * Two static specializations are used for transitional (cross-faded) delay
 * line processing and non-transitional processing.
 */
#define DECL_TEMPLATE(T)                                                      \
static void VectorAllpass_##T(ALfloat *restrict vec, const ALsizei offset,    \
                              const ALfloat feedCoeff, const ALfloat xCoeff,  \
                              const ALfloat yCoeff, const ALfloat mu,         \
                              VecAllpass *Vap)                                \
{                                                                             \
    ALfloat input;                                                            \
    ALfloat f[4];                                                             \
    ALsizei i;                                                                \
                                                                              \
    (void)mu; /* Ignore for Unfaded. */                                       \
                                                                              \
    for(i = 0;i < 4;i++)                                                      \
    {                                                                         \
        input = vec[i];                                                       \
        vec[i] = DELAY_OUT_##T(&Vap->Delay, offset-Vap->Offset[i][0],         \
                               offset-Vap->Offset[i][1], i, mu) -             \
                 feedCoeff*input;                                             \
        f[i] = input + feedCoeff*vec[i];                                      \
    }                                                                         \
                                                                              \
    VectorPartialScatter(f, xCoeff, yCoeff);                                  \
                                                                              \
    DelayLineIn4(&Vap->Delay, offset, f);                                     \
}
DECL_TEMPLATE(Unfaded)
DECL_TEMPLATE(Faded)
#undef DECL_TEMPLATE

/* A helper to reverse vector components. */
static inline void VectorReverse(ALfloat vec[4])
{
    const ALfloat f[4] = { vec[0], vec[1], vec[2], vec[3] };

    vec[0] = f[3];
    vec[1] = f[2];
    vec[2] = f[1];
    vec[3] = f[0];
}

/* This generates early reflections.
 *
 * This is done by obtaining the primary reflections (those arriving from the
 * same direction as the source) from the main delay line.  These are
 * attenuated and all-pass filtered (based on the diffusion parameter).
 *
 * The early lines are then fed in reverse (according to the approximately
 * opposite spatial location of the A-Format lines) to create the secondary
 * reflections (those arriving from the opposite direction as the source).
 *
 * The early response is then completed by combining the primary reflections
 * with the delayed and attenuated output from the early lines.
 *
 * Finally, the early response is reversed, scattered (based on diffusion),
 * and fed into the late reverb section of the main delay line.
 *
 * Two static specializations are used for transitional (cross-faded) delay
 * line processing and non-transitional processing.
 */
#define DECL_TEMPLATE(T)                                                      \
static ALvoid EarlyReflection_##T(ALreverbState *State, const ALsizei todo,   \
                                  ALfloat fade,                               \
                                  ALfloat (*restrict out)[MAX_UPDATE_SAMPLES])\
{                                                                             \
    ALsizei offset = State->Offset;                                           \
    const ALfloat apFeedCoeff = State->ApFeedCoeff;                           \
    const ALfloat mixX = State->MixX;                                         \
    const ALfloat mixY = State->MixY;                                         \
    ALfloat f[4];                                                             \
    ALsizei i, j;                                                             \
                                                                              \
    for(i = 0;i < todo;i++)                                                   \
    {                                                                         \
        for(j = 0;j < 4;j++)                                                  \
            f[j] = DELAY_OUT_##T(&State->Delay,                               \
                offset-State->EarlyDelayTap[j][0],                            \
                offset-State->EarlyDelayTap[j][1], j, fade                    \
            ) * State->EarlyDelayCoeff[j];                                    \
                                                                              \
        VectorAllpass_##T(f, offset, apFeedCoeff, mixX, mixY, fade,           \
                          &State->Early.VecAp);                               \
                                                                              \
        DelayLineIn4Rev(&State->Early.Delay, offset, f);                      \
                                                                              \
        for(j = 0;j < 4;j++)                                                  \
            f[j] += DELAY_OUT_##T(&State->Early.Delay,                        \
                offset-State->Early.Offset[j][0],                             \
                offset-State->Early.Offset[j][1], j, fade                     \
            ) * State->Early.Coeff[j];                                        \
                                                                              \
        for(j = 0;j < 4;j++)                                                  \
            out[j][i] = f[j];                                                 \
                                                                              \
        VectorReverse(f);                                                     \
                                                                              \
        VectorPartialScatter(f, mixX, mixY);                                  \
                                                                              \
        DelayLineIn4(&State->Delay, offset-State->LateFeedTap, f);            \
                                                                              \
        offset++;                                                             \
        fade += FadeStep;                                                     \
    }                                                                         \
}
DECL_TEMPLATE(Unfaded)
DECL_TEMPLATE(Faded)
#undef DECL_TEMPLATE

/* Applies a first order filter section. */
static inline ALfloat FirstOrderFilter(const ALfloat in, const ALfloat coeffs[3], ALfloat state[2])
{
    ALfloat out = coeffs[0]*in + coeffs[1]*state[0] + coeffs[2]*state[1];

    state[0] = in;
    state[1] = out;

    return out;
}

/* Applies the two T60 damping filter sections. */
static inline ALfloat LateT60Filter(const ALsizei index, const ALfloat in, ALreverbState *State)
{
    ALfloat out = FirstOrderFilter(in, State->Late.Filters[index].LFCoeffs,
                                   State->Late.Filters[index].States[0]);

    return State->Late.Filters[index].MidCoeff *
           FirstOrderFilter(out, State->Late.Filters[index].HFCoeffs,
                            State->Late.Filters[index].States[1]);
}

/* This generates the reverb tail using a modified feed-back delay network
 * (FDN).
 *
 * Results from the early reflections are attenuated by the density gain and
 * mixed with the output from the late delay lines.
 *
 * The late response is then completed by T60 and all-pass filtering the mix.
 *
 * Finally, the lines are reversed (so they feed their opposite directions)
 * and scattered with the FDN matrix before re-feeding the delay lines.
 *
 * Two static specializations are used for transitional (cross-faded) delay
 * line processing and non-transitional processing.
 */
static ALvoid LateReverb_Faded(ALreverbState *State, const ALsizei todo,
                             ALfloat fade,
                             ALfloat (*restrict out)[MAX_UPDATE_SAMPLES])
{
    ALsizei (*restrict moddelay)[MAX_UPDATE_SAMPLES][2] = State->ModulationDelays;
    const ALfloat apFeedCoeff = State->ApFeedCoeff;
    const ALfloat mixX = State->MixX;
    const ALfloat mixY = State->MixY;
    ALsizei offset;
    ALsizei i, j;

    CalcModulationDelays(State, moddelay, State->Late.Offset, todo);

    offset = State->Offset;
    for(i = 0;i < todo;i++)
    {
        ALfloat f[4];

        for(j = 0;j < 4;j++)
            f[j] = DELAY_OUT_Faded(&State->Delay,
                offset-State->LateDelayTap[j][0],
                offset-State->LateDelayTap[j][1], j, fade
            ) * State->Late.DensityGain;

        for(j = 0;j < 4;j++)
        {
            ALsizei delay0 = offset - (moddelay[j][i][0]>>FRACTIONBITS);
            ALfloat modmu0 = (moddelay[j][i][0]&FRACTIONBITS) * (1.0f/FRACTIONONE);
            ALsizei delay1 = offset - (moddelay[j][i][1]>>FRACTIONBITS);
            ALfloat modmu1 = (moddelay[j][i][1]&FRACTIONBITS) * (1.0f/FRACTIONONE);
            ALfloat r = DelayLineOut(&State->Late.Delay, delay0  , j)*(1.0f-modmu0)*(1.0f-fade) +
                        DelayLineOut(&State->Late.Delay, delay0-1, j)*(     modmu0)*(1.0f-fade) +
                        DelayLineOut(&State->Late.Delay, delay1  , j)*(1.0f-modmu1)*(     fade) +
                        DelayLineOut(&State->Late.Delay, delay1-1, j)*(     modmu1)*(     fade);
            out[j][i] = f[j] + r;
        }

        for(j = 0;j < 4;j++)
            f[j] += DELAY_OUT_Faded(&State->Late.Delay,
                offset - (State->Late.Offset[j][0]>>FRACTIONBITS),
                offset - (State->Late.Offset[j][1]>>FRACTIONBITS), j, fade
            );

        for(j = 0;j < 4;j++)
            f[j] = LateT60Filter(j, f[j], State);

        VectorAllpass_Faded(f, offset, apFeedCoeff, mixX, mixY, fade,
                            &State->Late.VecAp);

        VectorReverse(f);

        VectorPartialScatter(f, mixX, mixY);

        DelayLineIn4(&State->Late.Delay, offset, f);

        offset++;
        fade += FadeStep;
    }
}
static ALvoid LateReverb_Unfaded(ALreverbState *State, const ALsizei todo,
                                 ALfloat fade,
                                 ALfloat (*restrict out)[MAX_UPDATE_SAMPLES])
{
    ALsizei (*restrict moddelay)[MAX_UPDATE_SAMPLES][2] = State->ModulationDelays;
    const ALfloat apFeedCoeff = State->ApFeedCoeff;
    const ALfloat mixX = State->MixX;
    const ALfloat mixY = State->MixY;
    ALsizei offset;
    ALsizei i, j;

    CalcModulationDelays(State, moddelay, State->Late.Offset, todo);

    offset = State->Offset;
    for(i = 0;i < todo;i++)
    {
        ALfloat f[4];

        for(j = 0;j < 4;j++)
            f[j] = DelayLineOut(&State->Delay, offset-State->LateDelayTap[j][0], j) *
                   State->Late.DensityGain;

        for(j = 0;j < 4;j++)
        {
            ALsizei delay = offset - (moddelay[j][i][0]>>FRACTIONBITS);
            ALfloat modmu = (moddelay[j][i][0]&FRACTIONBITS) * (1.0f/FRACTIONONE);
            ALfloat r = DelayLineOut(&State->Late.Delay,   delay, j)*(1.0-modmu) +
                        DelayLineOut(&State->Late.Delay, delay-1, j)*(    modmu);
            out[j][i] = f[j] + r;
        }

        for(j = 0;j < 4;j++)
            f[j] += DelayLineOut(&State->Late.Delay,
                offset - (State->Late.Offset[j][0]>>FRACTIONBITS), j);

        for(j = 0;j < 4;j++)
            f[j] = LateT60Filter(j, f[j], State);

        VectorAllpass_Unfaded(f, offset, apFeedCoeff, mixX, mixY, fade,
                              &State->Late.VecAp);

        VectorReverse(f);

        VectorPartialScatter(f, mixX, mixY);

        DelayLineIn4(&State->Late.Delay, offset, f);

        offset++;
    }
}

typedef ALfloat (*ProcMethodType)(ALreverbState *State, const ALsizei todo, ALfloat fade,
    const ALfloat (*restrict input)[MAX_UPDATE_SAMPLES],
    ALfloat (*restrict early)[MAX_UPDATE_SAMPLES], ALfloat (*restrict late)[MAX_UPDATE_SAMPLES]);

/* Perform the non-EAX reverb pass on a given input sample, resulting in
 * four-channel output.
 */
static ALfloat VerbPass(ALreverbState *State, const ALsizei todo, ALfloat fade,
                        const ALfloat (*restrict input)[MAX_UPDATE_SAMPLES],
                        ALfloat (*restrict early)[MAX_UPDATE_SAMPLES],
                        ALfloat (*restrict late)[MAX_UPDATE_SAMPLES])
{
    ALsizei i, c;

    for(c = 0;c < 4;c++)
    {
        /* Low-pass filter the incoming samples (use the early buffer as temp
         * storage).
         */
        ALfilterState_process(&State->Filter[c].Lp, &early[0][0], input[c], todo);

        /* Feed the initial delay line. */
        for(i = 0;i < todo;i++)
            DelayLineIn(&State->Delay, State->Offset+i, c, early[0][i]);
    }

    if(fade < 1.0f)
    {
        /* Generate early reflections. */
        EarlyReflection_Faded(State, todo, fade, early);

        /* Generate late reverb. */
        LateReverb_Faded(State, todo, fade, late);
        fade = minf(1.0f, fade + todo*FadeStep);
    }
    else
    {
        /* Generate early reflections. */
        EarlyReflection_Unfaded(State, todo, fade, early);

        /* Generate late reverb. */
        LateReverb_Unfaded(State, todo, fade, late);
    }

    /* Step all delays forward one sample. */
    State->Offset += todo;

    return fade;
}

/* Perform the EAX reverb pass on a given input sample, resulting in four-
 * channel output.
 */
static ALfloat EAXVerbPass(ALreverbState *State, const ALsizei todo, ALfloat fade,
                           const ALfloat (*restrict input)[MAX_UPDATE_SAMPLES],
                           ALfloat (*restrict early)[MAX_UPDATE_SAMPLES],
                           ALfloat (*restrict late)[MAX_UPDATE_SAMPLES])
{
    ALsizei i, c;

    for(c = 0;c < 4;c++)
    {
        /* Band-pass the incoming samples. Use the early output lines for temp
         * storage.
         */
        ALfilterState_process(&State->Filter[c].Lp, early[0], input[c], todo);
        ALfilterState_process(&State->Filter[c].Hp, early[1], early[0], todo);

        /* Feed the initial delay line. */
        for(i = 0;i < todo;i++)
            DelayLineIn(&State->Delay, State->Offset+i, c, early[1][i]);
    }

    if(fade < 1.0f)
    {
        /* Generate early reflections. */
        EarlyReflection_Faded(State, todo, fade, early);

        /* Generate late reverb. */
        LateReverb_Faded(State, todo, fade, late);
        fade = minf(1.0f, fade + todo*FadeStep);
    }
    else
    {
        /* Generate early reflections. */
        EarlyReflection_Unfaded(State, todo, fade, early);

        /* Generate late reverb. */
        LateReverb_Unfaded(State, todo, fade, late);
    }

    /* Step all delays forward. */
    State->Offset += todo;

    return fade;
}

static ALvoid ALreverbState_process(ALreverbState *State, ALsizei SamplesToDo, const ALfloat (*restrict SamplesIn)[BUFFERSIZE], ALfloat (*restrict SamplesOut)[BUFFERSIZE], ALsizei NumChannels)
{
    ProcMethodType ReverbProc = State->IsEax ? EAXVerbPass : VerbPass;
    ALfloat (*restrict afmt)[MAX_UPDATE_SAMPLES] = State->AFormatSamples;
    ALfloat (*restrict early)[MAX_UPDATE_SAMPLES] = State->EarlySamples;
    ALfloat (*restrict late)[MAX_UPDATE_SAMPLES] = State->ReverbSamples;
    ALsizei fadeCount = State->FadeCount;
    ALfloat fade = (ALfloat)fadeCount / FADE_SAMPLES;
    ALsizei base, c;

    /* Process reverb for these samples. */
    for(base = 0;base < SamplesToDo;)
    {
        ALsizei todo = mini(SamplesToDo-base, MAX_UPDATE_SAMPLES);
        /* If cross-fading, don't do more samples than there are to fade. */
        if(FADE_SAMPLES-fadeCount > 0)
            todo = mini(todo, FADE_SAMPLES-fadeCount);

        /* Convert B-Format to A-Format for processing. */
        memset(afmt, 0, sizeof(*afmt)*4);
        for(c = 0;c < 4;c++)
            MixRowSamples(afmt[c], B2A.m[c],
                SamplesIn, MAX_EFFECT_CHANNELS, base, todo
            );

        /* Process the samples for reverb. */
        fade = ReverbProc(State, todo, fade, afmt, early, late);
        if(UNEXPECTED(fadeCount < FADE_SAMPLES) && (fadeCount += todo) >= FADE_SAMPLES)
        {
            /* Update the cross-fading delay line taps. */
            fadeCount = FADE_SAMPLES;
            fade = 1.0f;
            for(c = 0;c < 4;c++)
            {
                State->EarlyDelayTap[c][0] = State->EarlyDelayTap[c][1];
                State->Early.VecAp.Offset[c][0] = State->Early.VecAp.Offset[c][1];
                State->Early.Offset[c][0] = State->Early.Offset[c][1];
                State->LateDelayTap[c][0] = State->LateDelayTap[c][1];
                State->Late.VecAp.Offset[c][0] = State->Late.VecAp.Offset[c][1];
                State->Late.Offset[c][0] = State->Late.Offset[c][1];
            }
            State->Mod.Depth[0] = State->Mod.Depth[1];
        }

        /* Mix the A-Format results to output, implicitly converting back to
         * B-Format.
         */
        for(c = 0;c < 4;c++)
            MixSamples(early[c], NumChannels, SamplesOut,
                State->Early.CurrentGain[c], State->Early.PanGain[c],
                SamplesToDo-base, base, todo
            );
        for(c = 0;c < 4;c++)
            MixSamples(late[c], NumChannels, SamplesOut,
                State->Late.CurrentGain[c], State->Late.PanGain[c],
                SamplesToDo-base, base, todo
            );

        base += todo;
    }
    State->FadeCount = fadeCount;
}


typedef struct ALreverbStateFactory {
    DERIVE_FROM_TYPE(ALeffectStateFactory);
} ALreverbStateFactory;

static ALeffectState *ALreverbStateFactory_create(ALreverbStateFactory* UNUSED(factory))
{
    ALreverbState *state;

    alcall_once(&mixfunc_inited, init_mixfunc);

    NEW_OBJ0(state, ALreverbState)();
    if(!state) return NULL;

    return STATIC_CAST(ALeffectState, state);
}

DEFINE_ALEFFECTSTATEFACTORY_VTABLE(ALreverbStateFactory);

ALeffectStateFactory *ALreverbStateFactory_getFactory(void)
{
    static ALreverbStateFactory ReverbFactory = { { GET_VTABLE2(ALreverbStateFactory, ALeffectStateFactory) } };

    return STATIC_CAST(ALeffectStateFactory, &ReverbFactory);
}


void ALeaxreverb_setParami(ALeffect *effect, ALCcontext *context, ALenum param, ALint val)
{
    ALeffectProps *props = &effect->Props;
    switch(param)
    {
        case AL_EAXREVERB_DECAY_HFLIMIT:
            if(!(val >= AL_EAXREVERB_MIN_DECAY_HFLIMIT && val <= AL_EAXREVERB_MAX_DECAY_HFLIMIT))
                SET_ERROR_AND_RETURN(context, AL_INVALID_VALUE);
            props->Reverb.DecayHFLimit = val;
            break;

        default:
            SET_ERROR_AND_RETURN(context, AL_INVALID_ENUM);
    }
}
void ALeaxreverb_setParamiv(ALeffect *effect, ALCcontext *context, ALenum param, const ALint *vals)
{
    ALeaxreverb_setParami(effect, context, param, vals[0]);
}
void ALeaxreverb_setParamf(ALeffect *effect, ALCcontext *context, ALenum param, ALfloat val)
{
    ALeffectProps *props = &effect->Props;
    switch(param)
    {
        case AL_EAXREVERB_DENSITY:
            if(!(val >= AL_EAXREVERB_MIN_DENSITY && val <= AL_EAXREVERB_MAX_DENSITY))
                SET_ERROR_AND_RETURN(context, AL_INVALID_VALUE);
            props->Reverb.Density = val;
            break;

        case AL_EAXREVERB_DIFFUSION:
            if(!(val >= AL_EAXREVERB_MIN_DIFFUSION && val <= AL_EAXREVERB_MAX_DIFFUSION))
                SET_ERROR_AND_RETURN(context, AL_INVALID_VALUE);
            props->Reverb.Diffusion = val;
            break;

        case AL_EAXREVERB_GAIN:
            if(!(val >= AL_EAXREVERB_MIN_GAIN && val <= AL_EAXREVERB_MAX_GAIN))
                SET_ERROR_AND_RETURN(context, AL_INVALID_VALUE);
            props->Reverb.Gain = val;
            break;

        case AL_EAXREVERB_GAINHF:
            if(!(val >= AL_EAXREVERB_MIN_GAINHF && val <= AL_EAXREVERB_MAX_GAINHF))
                SET_ERROR_AND_RETURN(context, AL_INVALID_VALUE);
            props->Reverb.GainHF = val;
            break;

        case AL_EAXREVERB_GAINLF:
            if(!(val >= AL_EAXREVERB_MIN_GAINLF && val <= AL_EAXREVERB_MAX_GAINLF))
                SET_ERROR_AND_RETURN(context, AL_INVALID_VALUE);
            props->Reverb.GainLF = val;
            break;

        case AL_EAXREVERB_DECAY_TIME:
            if(!(val >= AL_EAXREVERB_MIN_DECAY_TIME && val <= AL_EAXREVERB_MAX_DECAY_TIME))
                SET_ERROR_AND_RETURN(context, AL_INVALID_VALUE);
            props->Reverb.DecayTime = val;
            break;

        case AL_EAXREVERB_DECAY_HFRATIO:
            if(!(val >= AL_EAXREVERB_MIN_DECAY_HFRATIO && val <= AL_EAXREVERB_MAX_DECAY_HFRATIO))
                SET_ERROR_AND_RETURN(context, AL_INVALID_VALUE);
            props->Reverb.DecayHFRatio = val;
            break;

        case AL_EAXREVERB_DECAY_LFRATIO:
            if(!(val >= AL_EAXREVERB_MIN_DECAY_LFRATIO && val <= AL_EAXREVERB_MAX_DECAY_LFRATIO))
                SET_ERROR_AND_RETURN(context, AL_INVALID_VALUE);
            props->Reverb.DecayLFRatio = val;
            break;

        case AL_EAXREVERB_REFLECTIONS_GAIN:
            if(!(val >= AL_EAXREVERB_MIN_REFLECTIONS_GAIN && val <= AL_EAXREVERB_MAX_REFLECTIONS_GAIN))
                SET_ERROR_AND_RETURN(context, AL_INVALID_VALUE);
            props->Reverb.ReflectionsGain = val;
            break;

        case AL_EAXREVERB_REFLECTIONS_DELAY:
            if(!(val >= AL_EAXREVERB_MIN_REFLECTIONS_DELAY && val <= AL_EAXREVERB_MAX_REFLECTIONS_DELAY))
                SET_ERROR_AND_RETURN(context, AL_INVALID_VALUE);
            props->Reverb.ReflectionsDelay = val;
            break;

        case AL_EAXREVERB_LATE_REVERB_GAIN:
            if(!(val >= AL_EAXREVERB_MIN_LATE_REVERB_GAIN && val <= AL_EAXREVERB_MAX_LATE_REVERB_GAIN))
                SET_ERROR_AND_RETURN(context, AL_INVALID_VALUE);
            props->Reverb.LateReverbGain = val;
            break;

        case AL_EAXREVERB_LATE_REVERB_DELAY:
            if(!(val >= AL_EAXREVERB_MIN_LATE_REVERB_DELAY && val <= AL_EAXREVERB_MAX_LATE_REVERB_DELAY))
                SET_ERROR_AND_RETURN(context, AL_INVALID_VALUE);
            props->Reverb.LateReverbDelay = val;
            break;

        case AL_EAXREVERB_AIR_ABSORPTION_GAINHF:
            if(!(val >= AL_EAXREVERB_MIN_AIR_ABSORPTION_GAINHF && val <= AL_EAXREVERB_MAX_AIR_ABSORPTION_GAINHF))
                SET_ERROR_AND_RETURN(context, AL_INVALID_VALUE);
            props->Reverb.AirAbsorptionGainHF = val;
            break;

        case AL_EAXREVERB_ECHO_TIME:
            if(!(val >= AL_EAXREVERB_MIN_ECHO_TIME && val <= AL_EAXREVERB_MAX_ECHO_TIME))
                SET_ERROR_AND_RETURN(context, AL_INVALID_VALUE);
            props->Reverb.EchoTime = val;
            break;

        case AL_EAXREVERB_ECHO_DEPTH:
            if(!(val >= AL_EAXREVERB_MIN_ECHO_DEPTH && val <= AL_EAXREVERB_MAX_ECHO_DEPTH))
                SET_ERROR_AND_RETURN(context, AL_INVALID_VALUE);
            props->Reverb.EchoDepth = val;
            break;

        case AL_EAXREVERB_MODULATION_TIME:
            if(!(val >= AL_EAXREVERB_MIN_MODULATION_TIME && val <= AL_EAXREVERB_MAX_MODULATION_TIME))
                SET_ERROR_AND_RETURN(context, AL_INVALID_VALUE);
            props->Reverb.ModulationTime = val;
            break;

        case AL_EAXREVERB_MODULATION_DEPTH:
            if(!(val >= AL_EAXREVERB_MIN_MODULATION_DEPTH && val <= AL_EAXREVERB_MAX_MODULATION_DEPTH))
                SET_ERROR_AND_RETURN(context, AL_INVALID_VALUE);
            props->Reverb.ModulationDepth = val;
            break;

        case AL_EAXREVERB_HFREFERENCE:
            if(!(val >= AL_EAXREVERB_MIN_HFREFERENCE && val <= AL_EAXREVERB_MAX_HFREFERENCE))
                SET_ERROR_AND_RETURN(context, AL_INVALID_VALUE);
            props->Reverb.HFReference = val;
            break;

        case AL_EAXREVERB_LFREFERENCE:
            if(!(val >= AL_EAXREVERB_MIN_LFREFERENCE && val <= AL_EAXREVERB_MAX_LFREFERENCE))
                SET_ERROR_AND_RETURN(context, AL_INVALID_VALUE);
            props->Reverb.LFReference = val;
            break;

        case AL_EAXREVERB_ROOM_ROLLOFF_FACTOR:
            if(!(val >= AL_EAXREVERB_MIN_ROOM_ROLLOFF_FACTOR && val <= AL_EAXREVERB_MAX_ROOM_ROLLOFF_FACTOR))
                SET_ERROR_AND_RETURN(context, AL_INVALID_VALUE);
            props->Reverb.RoomRolloffFactor = val;
            break;

        default:
            SET_ERROR_AND_RETURN(context, AL_INVALID_ENUM);
    }
}
void ALeaxreverb_setParamfv(ALeffect *effect, ALCcontext *context, ALenum param, const ALfloat *vals)
{
    ALeffectProps *props = &effect->Props;
    switch(param)
    {
        case AL_EAXREVERB_REFLECTIONS_PAN:
            if(!(isfinite(vals[0]) && isfinite(vals[1]) && isfinite(vals[2])))
                SET_ERROR_AND_RETURN(context, AL_INVALID_VALUE);
            props->Reverb.ReflectionsPan[0] = vals[0];
            props->Reverb.ReflectionsPan[1] = vals[1];
            props->Reverb.ReflectionsPan[2] = vals[2];
            break;
        case AL_EAXREVERB_LATE_REVERB_PAN:
            if(!(isfinite(vals[0]) && isfinite(vals[1]) && isfinite(vals[2])))
                SET_ERROR_AND_RETURN(context, AL_INVALID_VALUE);
            props->Reverb.LateReverbPan[0] = vals[0];
            props->Reverb.LateReverbPan[1] = vals[1];
            props->Reverb.LateReverbPan[2] = vals[2];
            break;

        default:
            ALeaxreverb_setParamf(effect, context, param, vals[0]);
            break;
    }
}

void ALeaxreverb_getParami(const ALeffect *effect, ALCcontext *context, ALenum param, ALint *val)
{
    const ALeffectProps *props = &effect->Props;
    switch(param)
    {
        case AL_EAXREVERB_DECAY_HFLIMIT:
            *val = props->Reverb.DecayHFLimit;
            break;

        default:
            SET_ERROR_AND_RETURN(context, AL_INVALID_ENUM);
    }
}
void ALeaxreverb_getParamiv(const ALeffect *effect, ALCcontext *context, ALenum param, ALint *vals)
{
    ALeaxreverb_getParami(effect, context, param, vals);
}
void ALeaxreverb_getParamf(const ALeffect *effect, ALCcontext *context, ALenum param, ALfloat *val)
{
    const ALeffectProps *props = &effect->Props;
    switch(param)
    {
        case AL_EAXREVERB_DENSITY:
            *val = props->Reverb.Density;
            break;

        case AL_EAXREVERB_DIFFUSION:
            *val = props->Reverb.Diffusion;
            break;

        case AL_EAXREVERB_GAIN:
            *val = props->Reverb.Gain;
            break;

        case AL_EAXREVERB_GAINHF:
            *val = props->Reverb.GainHF;
            break;

        case AL_EAXREVERB_GAINLF:
            *val = props->Reverb.GainLF;
            break;

        case AL_EAXREVERB_DECAY_TIME:
            *val = props->Reverb.DecayTime;
            break;

        case AL_EAXREVERB_DECAY_HFRATIO:
            *val = props->Reverb.DecayHFRatio;
            break;

        case AL_EAXREVERB_DECAY_LFRATIO:
            *val = props->Reverb.DecayLFRatio;
            break;

        case AL_EAXREVERB_REFLECTIONS_GAIN:
            *val = props->Reverb.ReflectionsGain;
            break;

        case AL_EAXREVERB_REFLECTIONS_DELAY:
            *val = props->Reverb.ReflectionsDelay;
            break;

        case AL_EAXREVERB_LATE_REVERB_GAIN:
            *val = props->Reverb.LateReverbGain;
            break;

        case AL_EAXREVERB_LATE_REVERB_DELAY:
            *val = props->Reverb.LateReverbDelay;
            break;

        case AL_EAXREVERB_AIR_ABSORPTION_GAINHF:
            *val = props->Reverb.AirAbsorptionGainHF;
            break;

        case AL_EAXREVERB_ECHO_TIME:
            *val = props->Reverb.EchoTime;
            break;

        case AL_EAXREVERB_ECHO_DEPTH:
            *val = props->Reverb.EchoDepth;
            break;

        case AL_EAXREVERB_MODULATION_TIME:
            *val = props->Reverb.ModulationTime;
            break;

        case AL_EAXREVERB_MODULATION_DEPTH:
            *val = props->Reverb.ModulationDepth;
            break;

        case AL_EAXREVERB_HFREFERENCE:
            *val = props->Reverb.HFReference;
            break;

        case AL_EAXREVERB_LFREFERENCE:
            *val = props->Reverb.LFReference;
            break;

        case AL_EAXREVERB_ROOM_ROLLOFF_FACTOR:
            *val = props->Reverb.RoomRolloffFactor;
            break;

        default:
            SET_ERROR_AND_RETURN(context, AL_INVALID_ENUM);
    }
}
void ALeaxreverb_getParamfv(const ALeffect *effect, ALCcontext *context, ALenum param, ALfloat *vals)
{
    const ALeffectProps *props = &effect->Props;
    switch(param)
    {
        case AL_EAXREVERB_REFLECTIONS_PAN:
            vals[0] = props->Reverb.ReflectionsPan[0];
            vals[1] = props->Reverb.ReflectionsPan[1];
            vals[2] = props->Reverb.ReflectionsPan[2];
            break;
        case AL_EAXREVERB_LATE_REVERB_PAN:
            vals[0] = props->Reverb.LateReverbPan[0];
            vals[1] = props->Reverb.LateReverbPan[1];
            vals[2] = props->Reverb.LateReverbPan[2];
            break;

        default:
            ALeaxreverb_getParamf(effect, context, param, vals);
            break;
    }
}

DEFINE_ALEFFECT_VTABLE(ALeaxreverb);

void ALreverb_setParami(ALeffect *effect, ALCcontext *context, ALenum param, ALint val)
{
    ALeffectProps *props = &effect->Props;
    switch(param)
    {
        case AL_REVERB_DECAY_HFLIMIT:
            if(!(val >= AL_REVERB_MIN_DECAY_HFLIMIT && val <= AL_REVERB_MAX_DECAY_HFLIMIT))
                SET_ERROR_AND_RETURN(context, AL_INVALID_VALUE);
            props->Reverb.DecayHFLimit = val;
            break;

        default:
            SET_ERROR_AND_RETURN(context, AL_INVALID_ENUM);
    }
}
void ALreverb_setParamiv(ALeffect *effect, ALCcontext *context, ALenum param, const ALint *vals)
{
    ALreverb_setParami(effect, context, param, vals[0]);
}
void ALreverb_setParamf(ALeffect *effect, ALCcontext *context, ALenum param, ALfloat val)
{
    ALeffectProps *props = &effect->Props;
    switch(param)
    {
        case AL_REVERB_DENSITY:
            if(!(val >= AL_REVERB_MIN_DENSITY && val <= AL_REVERB_MAX_DENSITY))
                SET_ERROR_AND_RETURN(context, AL_INVALID_VALUE);
            props->Reverb.Density = val;
            break;

        case AL_REVERB_DIFFUSION:
            if(!(val >= AL_REVERB_MIN_DIFFUSION && val <= AL_REVERB_MAX_DIFFUSION))
                SET_ERROR_AND_RETURN(context, AL_INVALID_VALUE);
            props->Reverb.Diffusion = val;
            break;

        case AL_REVERB_GAIN:
            if(!(val >= AL_REVERB_MIN_GAIN && val <= AL_REVERB_MAX_GAIN))
                SET_ERROR_AND_RETURN(context, AL_INVALID_VALUE);
            props->Reverb.Gain = val;
            break;

        case AL_REVERB_GAINHF:
            if(!(val >= AL_REVERB_MIN_GAINHF && val <= AL_REVERB_MAX_GAINHF))
                SET_ERROR_AND_RETURN(context, AL_INVALID_VALUE);
            props->Reverb.GainHF = val;
            break;

        case AL_REVERB_DECAY_TIME:
            if(!(val >= AL_REVERB_MIN_DECAY_TIME && val <= AL_REVERB_MAX_DECAY_TIME))
                SET_ERROR_AND_RETURN(context, AL_INVALID_VALUE);
            props->Reverb.DecayTime = val;
            break;

        case AL_REVERB_DECAY_HFRATIO:
            if(!(val >= AL_REVERB_MIN_DECAY_HFRATIO && val <= AL_REVERB_MAX_DECAY_HFRATIO))
                SET_ERROR_AND_RETURN(context, AL_INVALID_VALUE);
            props->Reverb.DecayHFRatio = val;
            break;

        case AL_REVERB_REFLECTIONS_GAIN:
            if(!(val >= AL_REVERB_MIN_REFLECTIONS_GAIN && val <= AL_REVERB_MAX_REFLECTIONS_GAIN))
                SET_ERROR_AND_RETURN(context, AL_INVALID_VALUE);
            props->Reverb.ReflectionsGain = val;
            break;

        case AL_REVERB_REFLECTIONS_DELAY:
            if(!(val >= AL_REVERB_MIN_REFLECTIONS_DELAY && val <= AL_REVERB_MAX_REFLECTIONS_DELAY))
                SET_ERROR_AND_RETURN(context, AL_INVALID_VALUE);
            props->Reverb.ReflectionsDelay = val;
            break;

        case AL_REVERB_LATE_REVERB_GAIN:
            if(!(val >= AL_REVERB_MIN_LATE_REVERB_GAIN && val <= AL_REVERB_MAX_LATE_REVERB_GAIN))
                SET_ERROR_AND_RETURN(context, AL_INVALID_VALUE);
            props->Reverb.LateReverbGain = val;
            break;

        case AL_REVERB_LATE_REVERB_DELAY:
            if(!(val >= AL_REVERB_MIN_LATE_REVERB_DELAY && val <= AL_REVERB_MAX_LATE_REVERB_DELAY))
                SET_ERROR_AND_RETURN(context, AL_INVALID_VALUE);
            props->Reverb.LateReverbDelay = val;
            break;

        case AL_REVERB_AIR_ABSORPTION_GAINHF:
            if(!(val >= AL_REVERB_MIN_AIR_ABSORPTION_GAINHF && val <= AL_REVERB_MAX_AIR_ABSORPTION_GAINHF))
                SET_ERROR_AND_RETURN(context, AL_INVALID_VALUE);
            props->Reverb.AirAbsorptionGainHF = val;
            break;

        case AL_REVERB_ROOM_ROLLOFF_FACTOR:
            if(!(val >= AL_REVERB_MIN_ROOM_ROLLOFF_FACTOR && val <= AL_REVERB_MAX_ROOM_ROLLOFF_FACTOR))
                SET_ERROR_AND_RETURN(context, AL_INVALID_VALUE);
            props->Reverb.RoomRolloffFactor = val;
            break;

        default:
            SET_ERROR_AND_RETURN(context, AL_INVALID_ENUM);
    }
}
void ALreverb_setParamfv(ALeffect *effect, ALCcontext *context, ALenum param, const ALfloat *vals)
{
    ALreverb_setParamf(effect, context, param, vals[0]);
}

void ALreverb_getParami(const ALeffect *effect, ALCcontext *context, ALenum param, ALint *val)
{
    const ALeffectProps *props = &effect->Props;
    switch(param)
    {
        case AL_REVERB_DECAY_HFLIMIT:
            *val = props->Reverb.DecayHFLimit;
            break;

        default:
            SET_ERROR_AND_RETURN(context, AL_INVALID_ENUM);
    }
}
void ALreverb_getParamiv(const ALeffect *effect, ALCcontext *context, ALenum param, ALint *vals)
{
    ALreverb_getParami(effect, context, param, vals);
}
void ALreverb_getParamf(const ALeffect *effect, ALCcontext *context, ALenum param, ALfloat *val)
{
    const ALeffectProps *props = &effect->Props;
    switch(param)
    {
        case AL_REVERB_DENSITY:
            *val = props->Reverb.Density;
            break;

        case AL_REVERB_DIFFUSION:
            *val = props->Reverb.Diffusion;
            break;

        case AL_REVERB_GAIN:
            *val = props->Reverb.Gain;
            break;

        case AL_REVERB_GAINHF:
            *val = props->Reverb.GainHF;
            break;

        case AL_REVERB_DECAY_TIME:
            *val = props->Reverb.DecayTime;
            break;

        case AL_REVERB_DECAY_HFRATIO:
            *val = props->Reverb.DecayHFRatio;
            break;

        case AL_REVERB_REFLECTIONS_GAIN:
            *val = props->Reverb.ReflectionsGain;
            break;

        case AL_REVERB_REFLECTIONS_DELAY:
            *val = props->Reverb.ReflectionsDelay;
            break;

        case AL_REVERB_LATE_REVERB_GAIN:
            *val = props->Reverb.LateReverbGain;
            break;

        case AL_REVERB_LATE_REVERB_DELAY:
            *val = props->Reverb.LateReverbDelay;
            break;

        case AL_REVERB_AIR_ABSORPTION_GAINHF:
            *val = props->Reverb.AirAbsorptionGainHF;
            break;

        case AL_REVERB_ROOM_ROLLOFF_FACTOR:
            *val = props->Reverb.RoomRolloffFactor;
            break;

        default:
            SET_ERROR_AND_RETURN(context, AL_INVALID_ENUM);
    }
}
void ALreverb_getParamfv(const ALeffect *effect, ALCcontext *context, ALenum param, ALfloat *vals)
{
    ALreverb_getParamf(effect, context, param, vals);
}

DEFINE_ALEFFECT_VTABLE(ALreverb);