/*************************************************************************** * __________ __ ___. * Open \______ \ ____ ____ | | _\_ |__ _______ ___ * Source | _// _ \_/ ___\| |/ /| __ \ / _ \ \/ / * Jukebox | | ( <_> ) \___| < | \_\ ( <_> > < < * Firmware |____|_ /\____/ \___ >__|_ \|___ /\____/__/\_ \ * \/ \/ \/ \/ \/ * * Copyright (C) 2005 Stepan Moskovchenko * * All files in this archive are subject to the GNU General Public License. * See the file COPYING in the source tree root for full license agreement. * * This software is distributed on an "AS IS" basis, WITHOUT WARRANTY OF ANY * KIND, either express or implied. * ****************************************************************************/ #define BYTE unsigned char //Data chunk ID types, returned by readID() #define ID_UNKNOWN -1 #define ID_MTHD 1 #define ID_MTRK 2 #define ID_EOF 3 //MIDI Commands #define MIDI_NOTE_OFF 128 #define MIDI_NOTE_ON 144 #define MIDI_AFTERTOUCH 160 #define MIDI_CONTROL 176 #define MIDI_PRGM 192 #define MIDI_PITCHW 224 //MIDI Controllers #define CTRL_VOLUME 7 #define CTRL_BALANCE 8 #define CTRL_PANNING 10 #define CHANNEL 1 //Most of these are deprecated.. rampdown is used, maybe one other one too #define STATE_ATTACK 1 #define STATE_DECAY 2 #define STATE_SUSTAIN 3 #define STATE_RELEASE 4 #define STATE_RAMPDOWN 5 //Loop states #define STATE_LOOPING 7 #define STATE_NONLOOPING 8 //Various bits in the GUS mode byte #define LOOP_ENABLED 4 #define LOOP_PINGPONG 8 #define LOOP_REVERSE 16 #define LOOPDIR_FORWARD 0 #define LOOPDIR_REVERSE 1 extern struct plugin_api * rb; int chVol[16] IDATA_ATTR; /* Channel volume */ int chPanLeft[16] IDATA_ATTR; /* Channel panning */ int chPanRight[16] IDATA_ATTR; int chPat[16]; /* Channel patch */ int chPW[16]; /* Channel pitch wheel, MSB only */ struct GPatch * gusload(char *); struct GPatch * patchSet[128]; struct GPatch * drumSet[128]; struct Event { unsigned int delta; unsigned char status, d1, d2; unsigned int len; unsigned char * evData; }; struct Track { unsigned int size; unsigned int numEvents; unsigned int delta; /* For sequencing */ unsigned int pos; /* For sequencing */ void * dataBlock; }; struct MIDIfile { int Length; unsigned short numTracks; unsigned short div; /* Time division, X ticks per millisecond */ struct Track * tracks[48]; unsigned char patches[128]; int numPatches; }; /* struct SynthObject { struct GWaveform * wf; unsigned int delta; unsigned int decay; unsigned int cp; unsigned char state, loopState, loopDir; unsigned char note, vol, ch, isUsed; int curRate, curOffset, targetOffset; unsigned int curPoint; }; */ struct SynthObject { struct GWaveform * wf; int delta; int decay; unsigned int cp; int state, loopState, loopDir; int note, vol, ch, isUsed; int curRate, curOffset, targetOffset; int curPoint; }; struct SynthObject voices[MAX_VOICES] IDATA_ATTR; void sendEvent(struct Event * ev); int tick(struct MIDIfile * mf); inline void setPoint(struct SynthObject * so, int pt); struct Event * getEvent(struct Track * tr, int evNum); int readTwoBytes(int file); int readFourBytes(int file); int readVarData(int file); int midimain(void * filename); void *alloc(int size) { static char *offset = NULL; static int totalSize = 0; char *ret; int remainder = size % 4; size = size + 4-remainder; if (offset == NULL) { offset = rb->plugin_get_audio_buffer(&totalSize); } if (size + 4 > totalSize) { return NULL; } ret = offset + 4; *((unsigned int *)offset) = size; offset += size + 4; totalSize -= size + 4; return ret; } /* Rick's code */ /* void *alloc(int size) { static char *offset = NULL; static int totalSize = 0; char *ret; if (offset == NULL) { offset = rb->plugin_get_audio_buffer(&totalSize); } if (size + 4 > totalSize) { return NULL; } ret = offset + 4; *((unsigned int *)offset) = size; offset += size + 4; totalSize -= size + 4; return ret; }*/ void * allocate(int size) { return alloc(size); } unsigned char readChar(int file) { char buf[2]; rb->read(file, &buf, 1); return buf[0]; } unsigned char * readData(int file, int len) { unsigned char * dat = allocate(len); rb->read(file, dat, len); return dat; } int eof(int fd) { int curPos = rb->lseek(fd, 0, SEEK_CUR); int size = rb->lseek(fd, 0, SEEK_END); rb->lseek(fd, curPos, SEEK_SET); return size+1 == rb->lseek(fd, 0, SEEK_CUR); } void printf(char *fmt, ...) {fmt=fmt; } void exit(int code) { code = code; /* Stub function, kill warning for now */ } '#n75'>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 
/***************************************************************************
 *             __________               __   ___.
 *   Open      \______   \ ____   ____ |  | _\_ |__   _______  ___
 *   Source     |       _//  _ \_/ ___\|  |/ /| __ \ /  _ \  \/  /
 *   Jukebox    |    |   (  <_> )  \___|    < | \_\ (  <_> > <  <
 *   Firmware   |____|_  /\____/ \___  >__|_ \|___  /\____/__/\_ \
 *                     \/            \/     \/    \/            \/
 * $Id$
 *
 * Copyright (C) 2005 Miika Pekkarinen
 *
 * This program is free software; you can redistribute it and/or
 * modify it under the terms of the GNU General Public License
 * as published by the Free Software Foundation; either version 2
 * of the License, or (at your option) any later version.
 *
 * This software is distributed on an "AS IS" basis, WITHOUT WARRANTY OF ANY
 * KIND, either express or implied.
 *
 ****************************************************************************/
#include "config.h"
#include <stdbool.h>
#include <inttypes.h>
#include <string.h>
#include <sound.h>
#include "dsp.h"
#include "eq.h"
#include "kernel.h"
#include "system.h"
#include "settings.h"
#include "replaygain.h"
#include "tdspeed.h"
#include "buffer.h"
#include "fixedpoint.h"
#include "fracmul.h"

/* Define LOGF_ENABLE to enable logf output in this file */
/*#define LOGF_ENABLE*/
#include "logf.h"

/* 16-bit samples are scaled based on these constants. The shift should be
 * no more than 15.
 */
#define WORD_SHIFT              12
#define WORD_FRACBITS           27

#define NATIVE_DEPTH            16
/* If the small buffer size changes, check the assembly code! */
#define SMALL_SAMPLE_BUF_COUNT  256
#define DEFAULT_GAIN            0x01000000

/* enums to index conversion properly with stereo mode and other settings */
enum
{
    SAMPLE_INPUT_LE_NATIVE_I_STEREO  = STEREO_INTERLEAVED,
    SAMPLE_INPUT_LE_NATIVE_NI_STEREO = STEREO_NONINTERLEAVED,
    SAMPLE_INPUT_LE_NATIVE_MONO      = STEREO_MONO,
    SAMPLE_INPUT_GT_NATIVE_I_STEREO  = STEREO_INTERLEAVED + STEREO_NUM_MODES,
    SAMPLE_INPUT_GT_NATIVE_NI_STEREO = STEREO_NONINTERLEAVED + STEREO_NUM_MODES,
    SAMPLE_INPUT_GT_NATIVE_MONO      = STEREO_MONO + STEREO_NUM_MODES,
    SAMPLE_INPUT_GT_NATIVE_1ST_INDEX = STEREO_NUM_MODES
};

enum
{
    SAMPLE_OUTPUT_MONO = 0,
    SAMPLE_OUTPUT_STEREO,
    SAMPLE_OUTPUT_DITHERED_MONO,
    SAMPLE_OUTPUT_DITHERED_STEREO
};

/****************************************************************************
 * NOTE: Any assembly routines that use these structures must be updated
 * if current data members are moved or changed.
 */
struct resample_data
{
    uint32_t delta;                     /* 00h */
    uint32_t phase;                     /* 04h */
    int32_t last_sample[2];             /* 08h */
                                        /* 10h */
};

/* This is for passing needed data to assembly dsp routines. If another
 * dsp parameter needs to be passed, add to the end of the structure
 * and remove from dsp_config.
 * If another function type becomes assembly optimized and requires dsp
 * config info, add a pointer paramter of type "struct dsp_data *".
 * If removing something from other than the end, reserve the spot or
 * else update every implementation for every target.
 * Be sure to add the offset of the new member for easy viewing as well. :)
 * It is the first member of dsp_config and all members can be accessesed
 * through the main aggregate but this is intended to make a safe haven
 * for these items whereas the c part can be rearranged at will. dsp_data
 * could even moved within dsp_config without disurbing the order.
 */
struct dsp_data
{
    int output_scale;                   /* 00h */
    int num_channels;                   /* 04h */
    struct resample_data resample_data; /* 08h */
    int32_t clip_min;                   /* 18h */
    int32_t clip_max;                   /* 1ch */
    int32_t gain;                       /* 20h - Note that this is in S8.23 format. */
                                        /* 24h */
};

/* No asm...yet */
struct dither_data
{
    long error[3];  /* 00h */
    long random;    /* 0ch */
                    /* 10h */
};

struct crossfeed_data
{
    int32_t gain;           /* 00h - Direct path gain */
    int32_t coefs[3];       /* 04h - Coefficients for the shelving filter */
    int32_t history[4];     /* 10h - Format is x[n - 1], y[n - 1] for both channels */
    int32_t delay[13][2];   /* 20h */
    int32_t *index;         /* 88h - Current pointer into the delay line */
                            /* 8ch */
};

/* Current setup is one lowshelf filters three peaking filters and one
 *  highshelf filter. Varying the number of shelving filters make no sense,
 *  but adding peaking filters is possible.
 */
struct eq_state
{
    char enabled[5];            /* 00h - Flags for active filters */
    struct eqfilter filters[5]; /* 08h - packing is 4? */
                                /* 10ch */
};

struct compressor_menu
{
    int  threshold;     /* dB - from menu */
    bool auto_gain;     /* 0 = off, 1 = auto */
    int  ratio;         /* from menu */
    bool soft_knee;     /* 0 = hard knee, 1 = soft knee */     
    int  release;       /* samples - from menu */
};

/* Include header with defines which functions are implemented in assembly
   code for the target */
#include <dsp_asm.h>

/* Typedefs keep things much neater in this case */
typedef void (*sample_input_fn_type)(int count, const char *src[],
                                     int32_t *dst[]);
typedef int (*resample_fn_type)(int count, struct dsp_data *data,
                                const int32_t *src[], int32_t *dst[]);
typedef void (*sample_output_fn_type)(int count, struct dsp_data *data,
                                      const int32_t *src[], int16_t *dst);

/* Single-DSP channel processing in place */
typedef void (*channels_process_fn_type)(int count, int32_t *buf[]);
/* DSP local channel processing in place */
typedef void (*channels_process_dsp_fn_type)(int count, struct dsp_data *data,
                                             int32_t *buf[]);

/*
 ***************************************************************************/

struct dsp_config
{
    struct dsp_data data; /* Config members for use in asm routines */
    long codec_frequency; /* Sample rate of data coming from the codec */
    long frequency;       /* Effective sample rate after pitch shift (if any) */
    int  sample_depth;
    int  sample_bytes;
    int  stereo_mode;
    int32_t  tdspeed_percent; /* Speed% * PITCH_SPEED_PRECISION */
    bool tdspeed_active;  /* Timestretch is in use */
    int  frac_bits;
#ifdef HAVE_SW_TONE_CONTROLS
    /* Filter struct for software bass/treble controls */
    struct eqfilter tone_filter;
#endif
    /* Functions that change depending upon settings - NULL if stage is
       disabled */
    sample_input_fn_type         input_samples;
    resample_fn_type             resample;
    sample_output_fn_type        output_samples;
    /* These will be NULL for the voice codec and is more economical that
       way */
    channels_process_dsp_fn_type apply_gain;
    channels_process_fn_type     apply_crossfeed;
    channels_process_fn_type     eq_process;
    channels_process_fn_type     channels_process;
    channels_process_fn_type     compressor_process;
};

/* General DSP config */
static struct dsp_config dsp_conf[2] IBSS_ATTR;     /* 0=A, 1=V */
/* Dithering */
static struct dither_data dither_data[2] IBSS_ATTR; /* 0=left, 1=right */
static long   dither_mask IBSS_ATTR;
static long   dither_bias IBSS_ATTR;
/* Crossfeed */
struct crossfeed_data crossfeed_data IDATA_ATTR =    /* A */
{
    .index = (int32_t *)crossfeed_data.delay
};

/* Equalizer */
static struct eq_state eq_data;                     /* A */

/* Software tone controls */
#ifdef HAVE_SW_TONE_CONTROLS
static int prescale;                                /* A/V */
static int bass;                                    /* A/V */
static int treble;                                  /* A/V */
#endif

/* Settings applicable to audio codec only */
static int32_t  pitch_ratio = PITCH_SPEED_100;
static int  channels_mode;
       long dsp_sw_gain;
       long dsp_sw_cross;
static bool dither_enabled;
static long eq_precut;
static long track_gain;
static bool new_gain;
static long album_gain;
static long track_peak;
static long album_peak;
static long replaygain;
static bool crossfeed_enabled;

#define AUDIO_DSP (dsp_conf[CODEC_IDX_AUDIO])
#define VOICE_DSP (dsp_conf[CODEC_IDX_VOICE])

/* The internal format is 32-bit samples, non-interleaved, stereo. This
 * format is similar to the raw output from several codecs, so the amount
 * of copying needed is minimized for that case.
 */

#define RESAMPLE_RATIO          4 /* Enough for 11,025 Hz -> 44,100 Hz */

static int32_t small_sample_buf[SMALL_SAMPLE_BUF_COUNT] IBSS_ATTR;
static int32_t small_resample_buf[SMALL_SAMPLE_BUF_COUNT * RESAMPLE_RATIO] IBSS_ATTR;

static int32_t *big_sample_buf = NULL;
static int32_t *big_resample_buf = NULL;
static int big_sample_buf_count = -1;  /* -1=unknown, 0=not available */

static int sample_buf_count;
static int32_t *sample_buf;
static int32_t *resample_buf;

#define SAMPLE_BUF_LEFT_CHANNEL 0
#define SAMPLE_BUF_RIGHT_CHANNEL (sample_buf_count/2)
#define RESAMPLE_BUF_LEFT_CHANNEL 0
#define RESAMPLE_BUF_RIGHT_CHANNEL (sample_buf_count/2 * RESAMPLE_RATIO)

/* compressor */
static struct   compressor_menu c_menu;
static int32_t comp_rel_slope IBSS_ATTR;   /* S7.24 format */
static int32_t comp_makeup_gain IBSS_ATTR; /* S7.24 format */
static int32_t comp_curve[66] IBSS_ATTR;   /* S7.24 format */
static int32_t release_gain IBSS_ATTR;     /* S7.24 format */
#define UNITY (1L << 24)                   /* unity gain in S7.24 format */
static void     compressor_process(int count, int32_t *buf[]);


/* Clip sample to signed 16 bit range */
static inline int32_t clip_sample_16(int32_t sample)
{
    if ((int16_t)sample != sample)
        sample = 0x7fff ^ (sample >> 31);
    return sample;
}

int32_t sound_get_pitch(void)
{
    return pitch_ratio;
}

void sound_set_pitch(int32_t percent)
{
    pitch_ratio = percent;
    dsp_configure(&AUDIO_DSP, DSP_SWITCH_FREQUENCY,
                  AUDIO_DSP.codec_frequency);
}

static void tdspeed_setup(struct dsp_config *dspc)
{
    /* Assume timestretch will not be used */
    dspc->tdspeed_active = false;
    sample_buf = small_sample_buf;
    resample_buf = small_resample_buf;
    sample_buf_count = SMALL_SAMPLE_BUF_COUNT;

    if(!dsp_timestretch_available())
        return; /* Timestretch not enabled or buffer not allocated */
    if (dspc->tdspeed_percent == 0)
        dspc->tdspeed_percent = PITCH_SPEED_100;
    if (!tdspeed_config(
        dspc->codec_frequency == 0 ? NATIVE_FREQUENCY : dspc->codec_frequency,
        dspc->stereo_mode != STEREO_MONO,
        dspc->tdspeed_percent))
        return; /* Timestretch not possible or needed with these parameters */

    /* Timestretch is to be used */
    dspc->tdspeed_active = true;
    sample_buf = big_sample_buf;
    sample_buf_count = big_sample_buf_count;
    resample_buf = big_resample_buf;
}

void dsp_timestretch_enable(bool enabled)
{
    /* Hook to set up timestretch buffer on first call to settings_apply() */
    if (big_sample_buf_count < 0) /* Only do something on first call */
    {
        if (enabled)
        {
            /* Set up timestretch buffers */
            big_sample_buf_count = SMALL_SAMPLE_BUF_COUNT * RESAMPLE_RATIO;
            big_sample_buf = small_resample_buf;
            big_resample_buf = (int32_t *) buffer_alloc(big_sample_buf_count * RESAMPLE_RATIO * sizeof(int32_t));
        }
        else
        {
            /* Not enabled at startup, "big" buffers will never be available */
            big_sample_buf_count = 0;
        }
        tdspeed_setup(&AUDIO_DSP);
    }
}

void dsp_set_timestretch(int32_t percent)
{
    AUDIO_DSP.tdspeed_percent = percent;
    tdspeed_setup(&AUDIO_DSP);
}

int32_t dsp_get_timestretch()
{
    return AUDIO_DSP.tdspeed_percent;
}

bool dsp_timestretch_available()
{
    return (global_settings.timestretch_enabled && big_sample_buf_count > 0);
}

/* Convert count samples to the internal format, if needed.  Updates src
 * to point past the samples "consumed" and dst is set to point to the
 * samples to consume. Note that for mono, dst[0] equals dst[1], as there
 * is no point in processing the same data twice.
 */

/* convert count 16-bit mono to 32-bit mono */
static void sample_input_lte_native_mono(
    int count, const char *src[], int32_t *dst[])
{
    const int16_t *s = (int16_t *) src[0];
    const int16_t * const send = s + count;
    int32_t *d = dst[0] = dst[1] = &sample_buf[SAMPLE_BUF_LEFT_CHANNEL];
    int scale = WORD_SHIFT;

    while (s < send)
    {
        *d++ = *s++ << scale;
    }

    src[0] = (char *)s;
}

/* convert count 16-bit interleaved stereo to 32-bit noninterleaved */
static void sample_input_lte_native_i_stereo(
    int count, const char *src[], int32_t *dst[])
{
    const int32_t *s = (int32_t *) src[0];
    const int32_t * const send = s + count;
    int32_t *dl = dst[0] = &sample_buf[SAMPLE_BUF_LEFT_CHANNEL];
    int32_t *dr = dst[1] = &sample_buf[SAMPLE_BUF_RIGHT_CHANNEL];
    int scale = WORD_SHIFT;

    while (s < send)
    {
        int32_t slr = *s++;
#ifdef ROCKBOX_LITTLE_ENDIAN
        *dl++ = (slr >> 16) << scale;
        *dr++ = (int32_t)(int16_t)slr << scale;
#else  /* ROCKBOX_BIG_ENDIAN */
        *dl++ = (int32_t)(int16_t)slr << scale;
        *dr++ = (slr >> 16) << scale;
#endif
    }

    src[0] = (char *)s;
}

/* convert count 16-bit noninterleaved stereo to 32-bit noninterleaved */
static void sample_input_lte_native_ni_stereo(
    int count, const char *src[], int32_t *dst[])
{
    const int16_t *sl = (int16_t *) src[0];
    const int16_t *sr = (int16_t *) src[1];
    const int16_t * const slend = sl + count;
    int32_t *dl = dst[0] = &sample_buf[SAMPLE_BUF_LEFT_CHANNEL];
    int32_t *dr = dst[1] = &sample_buf[SAMPLE_BUF_RIGHT_CHANNEL];
    int scale = WORD_SHIFT;

    while (sl < slend)
    {
        *dl++ = *sl++ << scale;
        *dr++ = *sr++ << scale;
    }

    src[0] = (char *)sl;
    src[1] = (char *)sr;
}

/* convert count 32-bit mono to 32-bit mono */
static void sample_input_gt_native_mono(
    int count, const char *src[], int32_t *dst[])
{
    dst[0] = dst[1] = (int32_t *)src[0];
    src[0] = (char *)(dst[0] + count);
}

/* convert count 32-bit interleaved stereo to 32-bit noninterleaved stereo */
static void sample_input_gt_native_i_stereo(
    int count, const char *src[], int32_t *dst[])
{
    const int32_t *s = (int32_t *)src[0];
    const int32_t * const send = s + 2*count;
    int32_t *dl = dst[0] = &sample_buf[SAMPLE_BUF_LEFT_CHANNEL];
    int32_t *dr = dst[1] = &sample_buf[SAMPLE_BUF_RIGHT_CHANNEL];

    while (s < send)
    {
        *dl++ = *s++;
        *dr++ = *s++;
    }

    src[0] = (char *)send;
}

/* convert 32 bit-noninterleaved stereo to 32-bit noninterleaved stereo */
static void sample_input_gt_native_ni_stereo(
    int count, const char *src[], int32_t *dst[])
{
    dst[0] = (int32_t *)src[0];
    dst[1] = (int32_t *)src[1];
    src[0] = (char *)(dst[0] + count);
    src[1] = (char *)(dst[1] + count);
}

/**
 * sample_input_new_format()
 *
 * set the to-native sample conversion function based on dsp sample parameters
 *
 * !DSPPARAMSYNC
 * needs syncing with changes to the following dsp parameters:
 *  * dsp->stereo_mode (A/V)
 *  * dsp->sample_depth (A/V)
 */
static void sample_input_new_format(struct dsp_config *dsp)
{
    static const sample_input_fn_type sample_input_functions[] =
    {
        [SAMPLE_INPUT_LE_NATIVE_I_STEREO]  = sample_input_lte_native_i_stereo,
        [SAMPLE_INPUT_LE_NATIVE_NI_STEREO] = sample_input_lte_native_ni_stereo,
        [SAMPLE_INPUT_LE_NATIVE_MONO]      = sample_input_lte_native_mono,
        [SAMPLE_INPUT_GT_NATIVE_I_STEREO]  = sample_input_gt_native_i_stereo,
        [SAMPLE_INPUT_GT_NATIVE_NI_STEREO] = sample_input_gt_native_ni_stereo,
        [SAMPLE_INPUT_GT_NATIVE_MONO]      = sample_input_gt_native_mono,
    };

    int convert = dsp->stereo_mode;

    if (dsp->sample_depth > NATIVE_DEPTH)
        convert += SAMPLE_INPUT_GT_NATIVE_1ST_INDEX;

    dsp->input_samples = sample_input_functions[convert];
}


#ifndef DSP_HAVE_ASM_SAMPLE_OUTPUT_MONO
/* write mono internal format to output format */
static void sample_output_mono(int count, struct dsp_data *data,
                               const int32_t *src[], int16_t *dst)
{
    const int32_t *s0 = src[0];
    const int scale = data->output_scale;
    const int dc_bias = 1 << (scale - 1);

    while (count-- > 0)
    {
        int32_t lr = clip_sample_16((*s0++ + dc_bias) >> scale);
        *dst++ = lr;
        *dst++ = lr;
    }
}
#endif /* DSP_HAVE_ASM_SAMPLE_OUTPUT_MONO */

/* write stereo internal format to output format */
#ifndef DSP_HAVE_ASM_SAMPLE_OUTPUT_STEREO
static void sample_output_stereo(int count, struct dsp_data *data,
                                 const int32_t *src[], int16_t *dst)
{
    const int32_t *s0 = src[0];
    const int32_t *s1 = src[1];
    const int scale = data->output_scale;
    const int dc_bias = 1 << (scale - 1);

    while (count-- > 0)
    {
        *dst++ = clip_sample_16((*s0++ + dc_bias) >> scale);
        *dst++ = clip_sample_16((*s1++ + dc_bias) >> scale);
    }
}
#endif /* DSP_HAVE_ASM_SAMPLE_OUTPUT_STEREO */

/**
 * The "dither" code to convert the 24-bit samples produced by libmad was
 * taken from the coolplayer project - coolplayer.sourceforge.net
 *
 * This function handles mono and stereo outputs.
 */
static void sample_output_dithered(int count, struct dsp_data *data,
                                   const int32_t *src[], int16_t *dst)
{
    const int32_t mask = dither_mask;
    const int32_t bias = dither_bias;
    const int scale = data->output_scale;
    const int32_t min = data->clip_min;
    const int32_t max = data->clip_max;
    const int32_t range = max - min;
    int ch;
    int16_t *d;

    for (ch = 0; ch < data->num_channels; ch++)
    {
        struct dither_data * const dither = &dither_data[ch];
        const int32_t *s = src[ch];
        int i;

        for (i = 0, d = &dst[ch]; i < count; i++, s++, d += 2)
        {
            int32_t output, sample;
            int32_t random;

            /* Noise shape and bias (for correct rounding later) */
            sample = *s;
            sample += dither->error[0] - dither->error[1] + dither->error[2];
            dither->error[2] = dither->error[1];
            dither->error[1] = dither->error[0]/2;

            output = sample + bias;

            /* Dither, highpass triangle PDF */
            random = dither->random*0x0019660dL + 0x3c6ef35fL;
            output += (random & mask) - (dither->random & mask);
            dither->random = random;

            /* Round sample to output range */
            output &= ~mask;

            /* Error feedback */
            dither->error[0] = sample - output;

            /* Clip */
            if ((uint32_t)(output - min) > (uint32_t)range)
            {
                int32_t c = min;
                if (output > min)
                    c += range;
                output = c;
            }

            /* Quantize and store */
            *d = output >> scale;
        }
    }

    if (data->num_channels == 2)
        return;

    /* Have to duplicate left samples into the right channel since
       pcm buffer and hardware is interleaved stereo */
    d = &dst[0];

    while (count-- > 0)
    {
        int16_t s = *d++;
        *d++ = s;
    }
}

/**
 * sample_output_new_format()
 *
 * set the from-native to ouput sample conversion routine
 *
 * !DSPPARAMSYNC
 * needs syncing with changes to the following dsp parameters:
 *  * dsp->stereo_mode (A/V)
 *  * dither_enabled (A)
 */
static void sample_output_new_format(struct dsp_config *dsp)
{
    static const sample_output_fn_type sample_output_functions[] =
    {
        sample_output_mono,
        sample_output_stereo,
        sample_output_dithered,
        sample_output_dithered
    };

    int out = dsp->data.num_channels - 1;

    if (dsp == &AUDIO_DSP && dither_enabled)
        out += 2;

    dsp->output_samples = sample_output_functions[out];
}

/**
 * Linear interpolation resampling that introduces a one sample delay because
 * of our inability to look into the future at the end of a frame.
 */
#ifndef DSP_HAVE_ASM_RESAMPLING
static int dsp_downsample(int count, struct dsp_data *data,
                          const int32_t *src[], int32_t *dst[])
{
    int ch = data->num_channels - 1;
    uint32_t delta = data->resample_data.delta;
    uint32_t phase, pos;
    int32_t *d;

    /* Rolled channel loop actually showed slightly faster. */
    do
    {
        /* Just initialize things and not worry too much about the relatively
         * uncommon case of not being able to spit out a sample for the frame.
         */
        const int32_t *s = src[ch];
        int32_t last = data->resample_data.last_sample[ch];

        data->resample_data.last_sample[ch] = s[count - 1];
        d = dst[ch];
        phase = data->resample_data.phase;
        pos = phase >> 16;

        /* Do we need last sample of previous frame for interpolation? */
        if (pos > 0)
            last = s[pos - 1];

        while (pos < (uint32_t)count)
        {
            *d++ = last + FRACMUL((phase & 0xffff) << 15, s[pos] - last);
            phase += delta;
            pos = phase >> 16;
            last = s[pos - 1];
        }
    }
    while (--ch >= 0);

    /* Wrap phase accumulator back to start of next frame. */
    data->resample_data.phase = phase - (count << 16);
    return d - dst[0];
}

static int dsp_upsample(int count, struct dsp_data *data,
                        const int32_t *src[], int32_t *dst[])
{
    int  ch = data->num_channels - 1;
    uint32_t delta = data->resample_data.delta;
    uint32_t phase, pos;
    int32_t *d;

    /* Rolled channel loop actually showed slightly faster. */
    do
    {
        /* Should always be able to output a sample for a ratio up to RESAMPLE_RATIO */
        const int32_t *s = src[ch];
        int32_t last = data->resample_data.last_sample[ch];

        data->resample_data.last_sample[ch] = s[count - 1];
        d = dst[ch];
        phase = data->resample_data.phase;
        pos = phase >> 16;

        while (pos == 0)
        {
            *d++ = last + FRACMUL((phase & 0xffff) << 15, s[0] - last);
            phase += delta;
            pos = phase >> 16;
        }

        while (pos < (uint32_t)count)
        {
            last = s[pos - 1];
            *d++ = last + FRACMUL((phase & 0xffff) << 15, s[pos] - last);
            phase += delta;
            pos = phase >> 16;
        }
    }
    while (--ch >= 0);

    /* Wrap phase accumulator back to start of next frame. */
    data->resample_data.phase = phase & 0xffff;
    return d - dst[0];
}
#endif /* DSP_HAVE_ASM_RESAMPLING */

static void resampler_new_delta(struct dsp_config *dsp)
{
    dsp->data.resample_data.delta = (unsigned long)
        dsp->frequency * 65536LL / NATIVE_FREQUENCY;

    if (dsp->frequency == NATIVE_FREQUENCY)
    {
        /* NOTE: If fully glitch-free transistions from no resampling to
           resampling are desired, last_sample history should be maintained
           even when not resampling. */
        dsp->resample = NULL;
        dsp->data.resample_data.phase = 0;
        dsp->data.resample_data.last_sample[0] = 0;
        dsp->data.resample_data.last_sample[1] = 0;
    }
    else if (dsp->frequency < NATIVE_FREQUENCY)
        dsp->resample = dsp_upsample;
    else
        dsp->resample = dsp_downsample;
}

/* Resample count stereo samples. Updates the src array, if resampling is
 * done, to refer to the resampled data. Returns number of stereo samples
 * for further processing.
 */
static inline int resample(struct dsp_config *dsp, int count, int32_t *src[])
{
    int32_t *dst[2] =
    {
        &resample_buf[RESAMPLE_BUF_LEFT_CHANNEL],
        &resample_buf[RESAMPLE_BUF_RIGHT_CHANNEL],
    };

    count = dsp->resample(count, &dsp->data, (const int32_t **)src, dst);

    src[0] = dst[0];
    src[1] = dst[dsp->data.num_channels - 1];

    return count;
}

static void dither_init(struct dsp_config *dsp)
{
    memset(dither_data, 0, sizeof (dither_data));
    dither_bias = (1L << (dsp->frac_bits - NATIVE_DEPTH));
    dither_mask = (1L << (dsp->frac_bits + 1 - NATIVE_DEPTH)) - 1;
}

void dsp_dither_enable(bool enable)
{
    struct dsp_config *dsp = &AUDIO_DSP;
    dither_enabled = enable;
    sample_output_new_format(dsp);
}

/* Applies crossfeed to the stereo signal in src.
 * Crossfeed is a process where listening over speakers is simulated. This
 * is good for old hard panned stereo records, which might be quite fatiguing
 * to listen to on headphones with no crossfeed.
 */
#ifndef DSP_HAVE_ASM_CROSSFEED
static void apply_crossfeed(int count, int32_t *buf[])
{
    int32_t *hist_l = &crossfeed_data.history[0];
    int32_t *hist_r = &crossfeed_data.history[2];
    int32_t *delay = &crossfeed_data.delay[0][0];
    int32_t *coefs = &crossfeed_data.coefs[0];
    int32_t gain = crossfeed_data.gain;
    int32_t *di = crossfeed_data.index;

    int32_t acc;
    int32_t left, right;
    int i;

    for (i = 0; i < count; i++)
    {
        left = buf[0][i];
        right = buf[1][i];

        /* Filter delayed sample from left speaker */
        acc = FRACMUL(*di, coefs[0]);
        acc += FRACMUL(hist_l[0], coefs[1]);
        acc += FRACMUL(hist_l[1], coefs[2]);
        /* Save filter history for left speaker */
        hist_l[1] = acc;
        hist_l[0] = *di;
        *di++ = left;
        /* Filter delayed sample from right speaker */
        acc = FRACMUL(*di, coefs[0]);
        acc += FRACMUL(hist_r[0], coefs[1]);
        acc += FRACMUL(hist_r[1], coefs[2]);
        /* Save filter history for right speaker */
        hist_r[1] = acc;
        hist_r[0] = *di;
        *di++ = right;
        /* Now add the attenuated direct sound and write to outputs */
        buf[0][i] = FRACMUL(left, gain) + hist_r[1];
        buf[1][i] = FRACMUL(right, gain) + hist_l[1];

        /* Wrap delay line index if bigger than delay line size */
        if (di >= delay + 13*2)
            di = delay;
    }
    /* Write back local copies of data we've modified */
    crossfeed_data.index = di;
}
#endif /* DSP_HAVE_ASM_CROSSFEED */

/**
 * dsp_set_crossfeed(bool enable)
 *
 * !DSPPARAMSYNC
 * needs syncing with changes to the following dsp parameters:
 *  * dsp->stereo_mode (A)
 */
void dsp_set_crossfeed(bool enable)
{
    crossfeed_enabled = enable;
    AUDIO_DSP.apply_crossfeed = (enable && AUDIO_DSP.data.num_channels > 1)
                                    ? apply_crossfeed : NULL;
}

void dsp_set_crossfeed_direct_gain(int gain)
{
    crossfeed_data.gain = get_replaygain_int(gain * 10) << 7;
    /* If gain is negative, the calculation overflowed and we need to clamp */
    if (crossfeed_data.gain < 0)
        crossfeed_data.gain = 0x7fffffff;
}

/* Both gains should be below 0 dB */
void dsp_set_crossfeed_cross_params(long lf_gain, long hf_gain, long cutoff)
{
    int32_t *c = crossfeed_data.coefs;
    long scaler = get_replaygain_int(lf_gain * 10) << 7;

    cutoff = 0xffffffff/NATIVE_FREQUENCY*cutoff;
    hf_gain -= lf_gain;
    /* Divide cutoff by sqrt(10^(hf_gain/20)) to place cutoff at the -3 dB
     * point instead of shelf midpoint. This is for compatibility with the old
     * crossfeed shelf filter and should be removed if crossfeed settings are
     * ever made incompatible for any other good reason.
     */
    cutoff = fp_div(cutoff, get_replaygain_int(hf_gain*5), 24);
    filter_shelf_coefs(cutoff, hf_gain, false, c);
    /* Scale coefs by LF gain and shift them to s0.31 format. We have no gains
     * over 1 and can do this safely
     */
    c[0] = FRACMUL_SHL(c[0], scaler, 4);
    c[1] = FRACMUL_SHL(c[1], scaler, 4);
    c[2] <<= 4;
}

/* Apply a constant gain to the samples (e.g., for ReplayGain).
 * Note that this must be called before the resampler.
 */
#ifndef DSP_HAVE_ASM_APPLY_GAIN
static void dsp_apply_gain(int count, struct dsp_data *data, int32_t *buf[])
{
    const int32_t gain = data->gain;
    int ch;

    for (ch = 0; ch < data->num_channels; ch++)
    {
        int32_t *d = buf[ch];
        int i;

        for (i = 0; i < count; i++)
            d[i] = FRACMUL_SHL(d[i], gain, 8);
    }
}
#endif /* DSP_HAVE_ASM_APPLY_GAIN */

/* Combine all gains to a global gain. */
static void set_gain(struct dsp_config *dsp)
{
    /* gains are in S7.24 format */
    dsp->data.gain = DEFAULT_GAIN;

    /* Replay gain not relevant to voice */
    if (dsp == &AUDIO_DSP && replaygain)
    {
        dsp->data.gain = replaygain;
    }

    if (dsp->eq_process && eq_precut)
    {
        dsp->data.gain = fp_mul(dsp->data.gain, eq_precut, 24);
    }

#ifdef HAVE_SW_VOLUME_CONTROL
    if (global_settings.volume < SW_VOLUME_MAX ||
        global_settings.volume > SW_VOLUME_MIN)
    {
        int vol_gain = get_replaygain_int(global_settings.volume * 100);
        dsp->data.gain = (long) (((int64_t) dsp->data.gain * vol_gain) >> 24);
    }
#endif

    if (dsp->data.gain == DEFAULT_GAIN)
    {
        dsp->data.gain = 0;
    }
    else
    {
        dsp->data.gain >>= 1;   /* convert gain to S8.23 format */
    }

    dsp->apply_gain = dsp->data.gain != 0 ? dsp_apply_gain : NULL;
}

/**
 * Update the amount to cut the audio before applying the equalizer.
 *
 * @param precut to apply in decibels (multiplied by 10)
 */
void dsp_set_eq_precut(int precut)
{
    eq_precut = get_replaygain_int(precut * -10);
    set_gain(&AUDIO_DSP);
}

/**
 * Synchronize the equalizer filter coefficients with the global settings.
 *
 * @param band the equalizer band to synchronize
 */
void dsp_set_eq_coefs(int band)
{
    const int *setting;
    long gain;
    unsigned long cutoff, q;

    /* Adjust setting pointer to the band we actually want to change */
    setting = &global_settings.eq_band0_cutoff + (band * 3);

    /* Convert user settings to format required by coef generator functions */
    cutoff = 0xffffffff / NATIVE_FREQUENCY * (*setting++);
    q = *setting++;
    gain = *setting++;

    if (q == 0)
        q = 1;

    /* NOTE: The coef functions assume the EMAC unit is in fractional mode,
       which it should be, since we're executed from the main thread. */

    /* Assume a band is disabled if the gain is zero */
    if (gain == 0)
    {
        eq_data.enabled[band] = 0;
    }
    else
    {
        if (band == 0)
            eq_ls_coefs(cutoff, q, gain, eq_data.filters[band].coefs);
        else if (band == 4)
            eq_hs_coefs(cutoff, q, gain, eq_data.filters[band].coefs);
        else
            eq_pk_coefs(cutoff, q, gain, eq_data.filters[band].coefs);

        eq_data.enabled[band] = 1;
    }
}

/* Apply EQ filters to those bands that have got it switched on. */
static void eq_process(int count, int32_t *buf[])
{
    static const int shifts[] =
    {
        EQ_SHELF_SHIFT,  /* low shelf  */
        EQ_PEAK_SHIFT,   /* peaking    */
        EQ_PEAK_SHIFT,   /* peaking    */
        EQ_PEAK_SHIFT,   /* peaking    */
        EQ_SHELF_SHIFT,  /* high shelf */
    };
    unsigned int channels = AUDIO_DSP.data.num_channels;
    int i;

    /* filter configuration currently is 1 low shelf filter, 3 band peaking
       filters and 1 high shelf filter, in that order. we need to know this
       so we can choose the correct shift factor.
     */
    for (i = 0; i < 5; i++)
    {
        if (!eq_data.enabled[i])
            continue;
        eq_filter(buf, &eq_data.filters[i], count, channels, shifts[i]);
    }
}

/**
 * Use to enable the equalizer.
 *
 * @param enable true to enable the equalizer
 */
void dsp_set_eq(bool enable)
{
    AUDIO_DSP.eq_process = enable ? eq_process : NULL;
    set_gain(&AUDIO_DSP);
}

static void dsp_set_stereo_width(int value)
{
    long width, straight, cross;

    width = value * 0x7fffff / 100;

    if (value <= 100)
    {
        straight = (0x7fffff + width) / 2;
        cross = straight - width;
    }
    else
    {
        /* straight = (1 + width) / (2 * width) */
        straight = ((int64_t)(0x7fffff + width) << 22) / width;
        cross = straight - 0x7fffff;
    }

    dsp_sw_gain  = straight << 8;
    dsp_sw_cross = cross << 8;
}

/**
 * Implements the different channel configurations and stereo width.
 */

/* SOUND_CHAN_STEREO mode is a noop so has no function - just outline one for
 * completeness. */
#if 0
static void channels_process_sound_chan_stereo(int count, int32_t *buf[])
{
    /* The channels are each just themselves */
    (void)count; (void)buf;
}
#endif

#ifndef DSP_HAVE_ASM_SOUND_CHAN_MONO
static void channels_process_sound_chan_mono(int count, int32_t *buf[])
{
    int32_t *sl = buf[0], *sr = buf[1];

    while (count-- > 0)
    {
        int32_t lr = *sl/2 + *sr/2;
        *sl++ = lr;
        *sr++ = lr;
    }
}
#endif /* DSP_HAVE_ASM_SOUND_CHAN_MONO */

#ifndef DSP_HAVE_ASM_SOUND_CHAN_CUSTOM
static void channels_process_sound_chan_custom(int count, int32_t *buf[])
{
    const int32_t gain  = dsp_sw_gain;
    const int32_t cross = dsp_sw_cross;
    int32_t *sl = buf[0], *sr = buf[1];

    while (count-- > 0)
    {
        int32_t l = *sl;
        int32_t r = *sr;
        *sl++ = FRACMUL(l, gain) + FRACMUL(r, cross);
        *sr++ = FRACMUL(r, gain) + FRACMUL(l, cross);
    }
}
#endif /* DSP_HAVE_ASM_SOUND_CHAN_CUSTOM */

static void channels_process_sound_chan_mono_left(int count, int32_t *buf[])
{
    /* Just copy over the other channel */
    memcpy(buf[1], buf[0], count * sizeof (*buf));
}

static void channels_process_sound_chan_mono_right(int count, int32_t *buf[])
{
    /* Just copy over the other channel */
    memcpy(buf[0], buf[1], count * sizeof (*buf));
}

#ifndef DSP_HAVE_ASM_SOUND_CHAN_KARAOKE
static void channels_process_sound_chan_karaoke(int count, int32_t *buf[])
{
    int32_t *sl = buf[0], *sr = buf[1];

    while (count-- > 0)
    {
        int32_t ch = *sl/2 - *sr/2;
        *sl++ = ch;
        *sr++ = -ch;
    }
}
#endif /* DSP_HAVE_ASM_SOUND_CHAN_KARAOKE */

static void dsp_set_channel_config(int value)
{
    static const channels_process_fn_type channels_process_functions[] =
    {
        /* SOUND_CHAN_STEREO = All-purpose index for no channel processing */
        [SOUND_CHAN_STEREO]     = NULL,
        [SOUND_CHAN_MONO]       = channels_process_sound_chan_mono,
        [SOUND_CHAN_CUSTOM]     = channels_process_sound_chan_custom,
        [SOUND_CHAN_MONO_LEFT]  = channels_process_sound_chan_mono_left,
        [SOUND_CHAN_MONO_RIGHT] = channels_process_sound_chan_mono_right,
        [SOUND_CHAN_KARAOKE]    = channels_process_sound_chan_karaoke,
    };

    if ((unsigned)value >= ARRAYLEN(channels_process_functions) ||
        AUDIO_DSP.stereo_mode == STEREO_MONO)
    {
        value = SOUND_CHAN_STEREO;
    }

    /* This doesn't apply to voice */
    channels_mode = value;
    AUDIO_DSP.channels_process = channels_process_functions[value];
}

#if CONFIG_CODEC == SWCODEC

#ifdef HAVE_SW_TONE_CONTROLS
static void set_tone_controls(void)
{
    filter_bishelf_coefs(0xffffffff/NATIVE_FREQUENCY*200,
                         0xffffffff/NATIVE_FREQUENCY*3500,
                         bass, treble, -prescale,
                         AUDIO_DSP.tone_filter.coefs);
    /* Sync the voice dsp coefficients */
    memcpy(&VOICE_DSP.tone_filter.coefs, AUDIO_DSP.tone_filter.coefs,
           sizeof (VOICE_DSP.tone_filter.coefs));
}
#endif

/* Hook back from firmware/ part of audio, which can't/shouldn't call apps/
 * code directly.
 */
int dsp_callback(int msg, intptr_t param)
{
    switch (msg)
    {
#ifdef HAVE_SW_TONE_CONTROLS
    case DSP_CALLBACK_SET_PRESCALE:
        prescale = param;
        set_tone_controls();
        break;
    /* prescaler is always set after calling any of these, so we wait with
     * calculating coefs until the above case is hit.
     */
    case DSP_CALLBACK_SET_BASS:
        bass = param;
        break;
    case DSP_CALLBACK_SET_TREBLE:
        treble = param;
        break;
#ifdef HAVE_SW_VOLUME_CONTROL
    case DSP_CALLBACK_SET_SW_VOLUME:
        set_gain(&AUDIO_DSP);
        break;
#endif
#endif
    case DSP_CALLBACK_SET_CHANNEL_CONFIG:
        dsp_set_channel_config(param);
        break;
    case DSP_CALLBACK_SET_STEREO_WIDTH:
        dsp_set_stereo_width(param);
        break;
    default:
        break;
    }
    return 0;
}
#endif

/* Process and convert src audio to dst based on the DSP configuration,
 * reading count number of audio samples. dst is assumed to be large
 * enough; use dsp_output_count() to get the required number. src is an
 * array of pointers; for mono and interleaved stereo, it contains one
 * pointer to the start of the audio data and the other is ignored; for
 * non-interleaved stereo, it contains two pointers, one for each audio
 * channel. Returns number of bytes written to dst.
 */
int dsp_process(struct dsp_config *dsp, char *dst, const char *src[], int count)
{
    int32_t *tmp[2];
    static long last_yield;
    long tick;
    int written = 0;

#if defined(CPU_COLDFIRE)
    /* set emac unit for dsp processing, and save old macsr, we're running in
       codec thread context at this point, so can't clobber it */
    unsigned long old_macsr = coldfire_get_macsr();
    coldfire_set_macsr(EMAC_FRACTIONAL | EMAC_SATURATE);
#endif

    if (new_gain)
        dsp_set_replaygain(); /* Gain has changed */

    /* Perform at least one yield before starting */
    last_yield = current_tick;
    yield();

    /* Testing function pointers for NULL is preferred since the pointer
       will be preloaded to be used for the call if not. */
    while (count > 0)
    {
        int samples = MIN(sample_buf_count/2, count);
        count -= samples;

        dsp->input_samples(samples, src, tmp);

        if (dsp->tdspeed_active)
            samples = tdspeed_doit(tmp, samples);
        
        int chunk_offset = 0;
        while (samples > 0)
        {
            int32_t *t2[2];
            t2[0] = tmp[0]+chunk_offset;
            t2[1] = tmp[1]+chunk_offset;

            int chunk = MIN(sample_buf_count/2, samples);
            chunk_offset += chunk;
            samples -= chunk;

            if (dsp->apply_gain)
                dsp->apply_gain(chunk, &dsp->data, t2);

            if (dsp->resample && (chunk = resample(dsp, chunk, t2)) <= 0)
                break; /* I'm pretty sure we're downsampling here */

            if (dsp->apply_crossfeed)
                dsp->apply_crossfeed(chunk, t2);

            if (dsp->eq_process)
                dsp->eq_process(chunk, t2);

#ifdef HAVE_SW_TONE_CONTROLS
            if ((bass | treble) != 0)
                eq_filter(t2, &dsp->tone_filter, chunk,
                      dsp->data.num_channels, FILTER_BISHELF_SHIFT);
#endif

            if (dsp->channels_process)
                dsp->channels_process(chunk, t2);
            
            if (dsp->compressor_process)
                dsp->compressor_process(chunk, t2);

            dsp->output_samples(chunk, &dsp->data, (const int32_t **)t2, (int16_t *)dst);

            written += chunk;
            dst += chunk * sizeof (int16_t) * 2;

            /* yield at least once each tick */
            tick = current_tick;
            if (TIME_AFTER(tick, last_yield))
            {
                last_yield = tick;
                yield();
            }
        }
    }

#if defined(CPU_COLDFIRE)
    /* set old macsr again */
    coldfire_set_macsr(old_macsr);
#endif
    return written;
}

/* Given count number of input samples, calculate the maximum number of
 * samples of output data that would be generated (the calculation is not
 * entirely exact and rounds upwards to be on the safe side; during
 * resampling, the number of samples generated depends on the current state
 * of the resampler).
 */
/* dsp_input_size MUST be called afterwards */
int dsp_output_count(struct dsp_config *dsp, int count)
{
    if (dsp->tdspeed_active)
        count = tdspeed_est_output_size();
    if (dsp->resample)
    {
        count = (int)(((unsigned long)count * NATIVE_FREQUENCY
                    + (dsp->frequency - 1)) / dsp->frequency);
    }

    /* Now we have the resampled sample count which must not exceed
     * RESAMPLE_BUF_RIGHT_CHANNEL to avoid resample buffer overflow. One
     * must call dsp_input_count() to get the correct input sample
     * count.
     */
    if (count > RESAMPLE_BUF_RIGHT_CHANNEL)
        count = RESAMPLE_BUF_RIGHT_CHANNEL;
        
    return count;
}

/* Given count output samples, calculate number of input samples
 * that would be consumed in order to fill the output buffer.
 */
int dsp_input_count(struct dsp_config *dsp, int count)
{
    /* count is now the number of resampled input samples. Convert to
       original input samples. */
    if (dsp->resample)
    {
        /* Use the real resampling delta =
         * dsp->frequency * 65536 / NATIVE_FREQUENCY, and
         * round towards zero to avoid buffer overflows. */
        count = (int)(((unsigned long)count *
                      dsp->data.resample_data.delta) >> 16);
    }

    if (dsp->tdspeed_active)
        count = tdspeed_est_input_size(count);

    return count;
}

static void dsp_set_gain_var(long *var, long value)
{
    *var = value;
    new_gain = true;
}

static void dsp_update_functions(struct dsp_config *dsp)
{
    sample_input_new_format(dsp);
    sample_output_new_format(dsp);
    if (dsp == &AUDIO_DSP)
        dsp_set_crossfeed(crossfeed_enabled);
}

intptr_t dsp_configure(struct dsp_config *dsp, int setting, intptr_t value)
{
    switch (setting)
    {
    case DSP_MYDSP:
        switch (value)
        {
        case CODEC_IDX_AUDIO:
            return (intptr_t)&AUDIO_DSP;
        case CODEC_IDX_VOICE:
            return (intptr_t)&VOICE_DSP;
        default:
            return (intptr_t)NULL;
        }

    case DSP_SET_FREQUENCY:
        memset(&dsp->data.resample_data, 0, sizeof (dsp->data.resample_data));
        /* Fall through!!! */
    case DSP_SWITCH_FREQUENCY:
        dsp->codec_frequency = (value == 0) ? NATIVE_FREQUENCY : value;
        /* Account for playback speed adjustment when setting dsp->frequency
           if we're called from the main audio thread. Voice UI thread should
           not need this feature.
         */
        if (dsp == &AUDIO_DSP)
            dsp->frequency = pitch_ratio * dsp->codec_frequency / PITCH_SPEED_100;
        else
            dsp->frequency = dsp->codec_frequency;

        resampler_new_delta(dsp);
        tdspeed_setup(dsp);
        break;

    case DSP_SET_SAMPLE_DEPTH:
        dsp->sample_depth = value;

        if (dsp->sample_depth <= NATIVE_DEPTH)
        {
            dsp->frac_bits = WORD_FRACBITS;
            dsp->sample_bytes = sizeof (int16_t); /* samples are 16 bits */
            dsp->data.clip_max =  ((1 << WORD_FRACBITS) - 1);
            dsp->data.clip_min = -((1 << WORD_FRACBITS));
        }
        else
        {
            dsp->frac_bits = value;
            dsp->sample_bytes = sizeof (int32_t); /* samples are 32 bits */
            dsp->data.clip_max = (1 << value) - 1;
            dsp->data.clip_min = -(1 << value);
        }

        dsp->data.output_scale = dsp->frac_bits + 1 - NATIVE_DEPTH;
        sample_input_new_format(dsp);
        dither_init(dsp);
        break;

    case DSP_SET_STEREO_MODE:
        dsp->stereo_mode = value;
        dsp->data.num_channels = value == STEREO_MONO ? 1 : 2;
        dsp_update_functions(dsp);
        tdspeed_setup(dsp);
        break;

    case DSP_RESET:
        dsp->stereo_mode = STEREO_NONINTERLEAVED;
        dsp->data.num_channels = 2;
        dsp->sample_depth = NATIVE_DEPTH;
        dsp->frac_bits = WORD_FRACBITS;
        dsp->sample_bytes = sizeof (int16_t);
        dsp->data.output_scale = dsp->frac_bits + 1 - NATIVE_DEPTH;
        dsp->data.clip_max =  ((1 << WORD_FRACBITS) - 1);
        dsp->data.clip_min = -((1 << WORD_FRACBITS));
        dsp->codec_frequency = dsp->frequency = NATIVE_FREQUENCY;

        if (dsp == &AUDIO_DSP)
        {
            track_gain = 0;
            album_gain = 0;
            track_peak = 0;
            album_peak = 0;
            new_gain   = true;
        }

        dsp_update_functions(dsp);
        resampler_new_delta(dsp);
        tdspeed_setup(dsp);
        if (dsp == &AUDIO_DSP)
            release_gain = UNITY;
        break;

    case DSP_FLUSH:
        memset(&dsp->data.resample_data, 0,
               sizeof (dsp->data.resample_data));
        resampler_new_delta(dsp);
        dither_init(dsp);
        tdspeed_setup(dsp);
        if (dsp == &AUDIO_DSP)
            release_gain = UNITY;
        break;

    case DSP_SET_TRACK_GAIN:
        if (dsp == &AUDIO_DSP)
            dsp_set_gain_var(&track_gain, value);
        break;

    case DSP_SET_ALBUM_GAIN:
        if (dsp == &AUDIO_DSP)
            dsp_set_gain_var(&album_gain, value);
        break;

    case DSP_SET_TRACK_PEAK:
        if (dsp == &AUDIO_DSP)
            dsp_set_gain_var(&track_peak, value);
        break;

    case DSP_SET_ALBUM_PEAK:
        if (dsp == &AUDIO_DSP)
            dsp_set_gain_var(&album_peak, value);
        break;

    default:
        return 0;
    }

    return 1;
}

int get_replaygain_mode(bool have_track_gain, bool have_album_gain)
{
    int type;

    bool track = ((global_settings.replaygain_type == REPLAYGAIN_TRACK)
        || ((global_settings.replaygain_type == REPLAYGAIN_SHUFFLE)
            && global_settings.playlist_shuffle));

    type = (!track && have_album_gain) ? REPLAYGAIN_ALBUM 
        : have_track_gain ? REPLAYGAIN_TRACK : -1;
    
    return type;
}

void dsp_set_replaygain(void)
{
    long gain = 0;

    new_gain = false;

    if ((global_settings.replaygain_type != REPLAYGAIN_OFF) ||
            global_settings.replaygain_noclip)
    {
        bool track_mode = get_replaygain_mode(track_gain != 0,
            album_gain != 0) == REPLAYGAIN_TRACK;
        long peak = (track_mode || !album_peak) ? track_peak : album_peak;

        if (global_settings.replaygain_type != REPLAYGAIN_OFF)
        {
            gain = (track_mode || !album_gain) ? track_gain : album_gain;

            if (global_settings.replaygain_preamp)
            {
                long preamp = get_replaygain_int(
                    global_settings.replaygain_preamp * 10);

                gain = (long) (((int64_t) gain * preamp) >> 24);
            }
        }

        if (gain == 0)
        {
            /* So that noclip can work even with no gain information. */
            gain = DEFAULT_GAIN;
        }

        if (global_settings.replaygain_noclip && (peak != 0)
            && ((((int64_t) gain * peak) >> 24) >= DEFAULT_GAIN))
        {
            gain = (((int64_t) DEFAULT_GAIN << 24) / peak);
        }

        if (gain == DEFAULT_GAIN)
        {
            /* Nothing to do, disable processing. */
            gain = 0;
        }
    }

    /* Store in S7.24 format to simplify calculations. */
    replaygain = gain;
    set_gain(&AUDIO_DSP);
}

/** SET COMPRESSOR
 *  Called by the menu system to configure the compressor process */
void dsp_set_compressor(int c_threshold, int c_gain, int c_ratio,
                        int c_knee, int c_release)
{
    bool changed = false;
    bool active = (c_threshold < 0);
    bool new_auto_gain = (c_gain == 1);
    const int comp_ratio[] = {2, 4, 6, 10, 0};
    int  new_ratio = comp_ratio[c_ratio];
    bool new_knee = (c_knee == 1);
    int  new_release = c_release * NATIVE_FREQUENCY / 1000;
    
    if (c_menu.threshold != c_threshold)
    {
        changed = true;
        c_menu.threshold = c_threshold;
        logf("   Compressor Threshold: %d dB\tEnabled: %s",
            c_menu.threshold, active ? "Yes" : "No");
    }

    if (c_menu.auto_gain != new_auto_gain)
    {
        changed = true;
        c_menu.auto_gain = new_auto_gain;
        logf("   Compressor Makeup Gain: %s",
            c_menu.auto_gain ? "Auto" : "Off");
    }
    
    if (c_menu.ratio != new_ratio)
    {
        changed = true;
        c_menu.ratio = new_ratio;
        if (c_menu.ratio)
            { logf("   Compressor Ratio: %d:1", c_menu.ratio); }
        else
            { logf("   Compressor Ratio: Limit"); }
    }
    
    if (c_menu.soft_knee != new_knee)
    {
        changed = true;
        c_menu.soft_knee = new_knee;
        logf("   Compressor Knee: %s", c_menu.soft_knee==1?"Soft":"Hard");
    }
    
    if (c_menu.release != new_release)
    {
        changed = true;
        c_menu.release = new_release;
        logf("   Compressor Release: %d", c_menu.release);
    }

    if (changed && active)
    {
        /* configure variables for compressor operation */
        int i;
        const int32_t db[] ={0x000000,   /* positive db equivalents in S15.16 format */
         0x241FA4, 0x1E1A5E, 0x1A94C8, 0x181518, 0x1624EA, 0x148F82, 0x1338BD, 0x120FD2,
         0x1109EB, 0x101FA4, 0x0F4BB6, 0x0E8A3C, 0x0DD840, 0x0D3377, 0x0C9A0E, 0x0C0A8C,
         0x0B83BE, 0x0B04A5, 0x0A8C6C, 0x0A1A5E, 0x09ADE1, 0x094670, 0x08E398, 0x0884F6,
         0x082A30, 0x07D2FA, 0x077F0F, 0x072E31, 0x06E02A, 0x0694C8, 0x064BDF, 0x060546,
         0x05C0DA, 0x057E78, 0x053E03, 0x04FF5F, 0x04C273, 0x048726, 0x044D64, 0x041518,
         0x03DE30, 0x03A89B, 0x037448, 0x03412A, 0x030F32, 0x02DE52, 0x02AE80, 0x027FB0,
         0x0251D6, 0x0224EA, 0x01F8E2, 0x01CDB4, 0x01A359, 0x0179C9, 0x0150FC, 0x0128EB,
         0x010190, 0x00DAE4, 0x00B4E1, 0x008F82, 0x006AC1, 0x004699, 0x002305};
        
        struct curve_point
        {
            int32_t db;     /* S15.16 format */
            int32_t offset; /* S15.16 format */
        } db_curve[5];
        
        /** Set up the shape of the compression curve first as decibel values*/
        /* db_curve[0] = bottom of knee
                   [1] = threshold
                   [2] = top of knee
                   [3] = 0 db input
                   [4] = ~+12db input (2 bits clipping overhead) */
        
        db_curve[1].db = c_menu.threshold << 16;
        if (c_menu.soft_knee)
        {
            /* bottom of knee is 3dB below the threshold for soft knee*/
            db_curve[0].db = db_curve[1].db - (3 << 16);
            /* top of knee is 3dB above the threshold for soft knee */
            db_curve[2].db = db_curve[1].db + (3 << 16);
            if (c_menu.ratio)
                /* offset = -3db * (ratio - 1) / ratio */
                db_curve[2].offset = (int32_t)((long long)(-3 << 16)
                    * (c_menu.ratio - 1) / c_menu.ratio);
            else
                /* offset = -3db for hard limit */
                db_curve[2].offset = (-3 << 16);
        }
        else
        {
            /* bottom of knee is at the threshold for hard knee */
            db_curve[0].db = c_menu.threshold << 16;
            /* top of knee is at the threshold for hard knee */
            db_curve[2].db = c_menu.threshold << 16;
            db_curve[2].offset = 0;
        }
        
        /* Calculate 0db and ~+12db offsets */
        db_curve[4].db = 0xC0A8C; /* db of 2 bits clipping */
        if (c_menu.ratio)
        {
            /* offset = threshold * (ratio - 1) / ratio */
            db_curve[3].offset = (int32_t)((long long)(c_menu.threshold << 16)
                * (c_menu.ratio - 1) / c_menu.ratio);
            db_curve[4].offset = (int32_t)((long long)-db_curve[4].db
                * (c_menu.ratio - 1) / c_menu.ratio) + db_curve[3].offset;
        }
        else
        {
            /* offset = threshold for hard limit */
            db_curve[3].offset = (c_menu.threshold << 16);
            db_curve[4].offset = -db_curve[4].db + db_curve[3].offset;
        }
        
        /** Now set up the comp_curve table with compression offsets in the form
            of gain factors in S7.24 format */
        /* comp_curve[0] is 0 (-infinity db) input */
        comp_curve[0] = UNITY;
        /* comp_curve[1 to 63] are intermediate compression values corresponding
           to the 6 MSB of the input values of a non-clipped signal */
        for (i = 1; i < 64; i++)
        {
            /* db constants are stored as positive numbers;
               make them negative here */
            int32_t this_db = -db[i];
            
            /* no compression below the knee */
            if (this_db <= db_curve[0].db)
                comp_curve[i] = UNITY;
            
            /* if soft knee and below top of knee,
               interpolate along soft knee slope */
            else if (c_menu.soft_knee && (this_db <= db_curve[2].db))
                comp_curve[i] = fp_factor(fp_mul(
                    ((this_db - db_curve[0].db) / 6),
                    db_curve[2].offset, 16), 16) << 8;
            
            /* interpolate along ratio slope above the knee */
            else
                comp_curve[i] = fp_factor(fp_mul(
                    fp_div((db_curve[1].db - this_db), db_curve[1].db, 16),
                    db_curve[3].offset, 16), 16) << 8;
        }
        /* comp_curve[64] is the compression level of a maximum level,
           non-clipped signal */
        comp_curve[64] = fp_factor(db_curve[3].offset, 16) << 8;
        
        /* comp_curve[65] is the compression level of a maximum level,
           clipped signal */
        comp_curve[65] = fp_factor(db_curve[4].offset, 16) << 8;
        
#if defined(ROCKBOX_HAS_LOGF) && defined(LOGF_ENABLE)
        logf("\n   *** Compression Offsets ***");
        /* some settings for display only, not used in calculations */
        db_curve[0].offset = 0;
        db_curve[1].offset = 0;
        db_curve[3].db = 0;
        
        for (i = 0; i <= 4; i++)
        {
            logf("Curve[%d]: db: % 6.2f\toffset: % 6.2f", i,
                (float)db_curve[i].db / (1 << 16),
                (float)db_curve[i].offset / (1 << 16));
        }
        
        logf("\nGain factors:");
        for (i = 1; i <= 65; i++)
        {
            debugf("%02d: %.6f  ", i, (float)comp_curve[i] / UNITY);
            if (i % 4 == 0) debugf("\n");
        }
        debugf("\n");
#endif
        
        /* if using auto peak, then makeup gain is max offset - .1dB headroom */
        comp_makeup_gain = c_menu.auto_gain ?
            fp_factor(-(db_curve[3].offset) - 0x199A, 16) << 8 : UNITY;
        logf("Makeup gain:\t%.6f", (float)comp_makeup_gain / UNITY);

        /* calculate per-sample gain change a rate of 10db over release time */
        comp_rel_slope = 0xAF0BB2 / c_menu.release;
        logf("Release slope:\t%.6f", (float)comp_rel_slope / UNITY);
        
        release_gain = UNITY;
    }
    
    /* enable/disable the compressor */
    AUDIO_DSP.compressor_process = active ? compressor_process : NULL;
}

/** GET COMPRESSION GAIN
 *  Returns the required gain factor in S7.24 format in order to compress the
 *  sample in accordance with the compression curve.  Always 1 or less.
 */
static inline int32_t get_compression_gain(int32_t sample)
{
    const int frac_bits_offset = AUDIO_DSP.frac_bits - 15;
    
    /* sample must be positive */
    if (sample < 0)
        sample = -(sample + 1);
        
    /* shift sample into 15 frac bit range */
    if (frac_bits_offset > 0)
        sample >>= frac_bits_offset;
    if (frac_bits_offset < 0)
        sample <<= -frac_bits_offset;
    
    /* normal case: sample isn't clipped */
    if (sample < (1 << 15))
    {
        /* index is 6 MSB, rem is 9 LSB */
        int index = sample >> 9;
        int32_t rem = (sample & 0x1FF) << 22;
        
        /* interpolate from the compression curve:
            higher gain - ((rem / (1 << 31)) * (higher gain - lower gain)) */
        return comp_curve[index] - (FRACMUL(rem,
            (comp_curve[index] - comp_curve[index + 1])));
    }
    /* sample is somewhat clipped, up to 2 bits of overhead */
    if (sample < (1 << 17))
    {
        /* straight interpolation:
            higher gain - ((clipped portion of sample * 4/3
            / (1 << 31)) * (higher gain - lower gain)) */
        return comp_curve[64] - (FRACMUL(((sample - (1 << 15)) / 3) << 16,
            (comp_curve[64] - comp_curve[65])));
    }
    
    /* sample is too clipped, return invalid value */
    return -1;
}

/** COMPRESSOR PROCESS
 *  Changes the gain of the samples according to the compressor curve
 */
static void compressor_process(int count, int32_t *buf[])
{
    const int num_chan = AUDIO_DSP.data.num_channels;
    int32_t *in_buf[2] = {buf[0], buf[1]};
    
    while (count-- > 0)
    {
        int ch;
        /* use lowest (most compressed) gain factor of the output buffer
           sample pair for both samples (mono is also handled correctly here) */
        int32_t sample_gain = UNITY;
        for (ch = 0; ch < num_chan; ch++)
        {
            int32_t this_gain = get_compression_gain(*in_buf[ch]);
            if (this_gain < sample_gain)
                sample_gain = this_gain;
        }
        
        /* perform release slope; skip if no compression and no release slope */
        if ((sample_gain != UNITY) || (release_gain != UNITY))
        {
            /* if larger offset than previous slope, start new release slope */
            if ((sample_gain <= release_gain) && (sample_gain > 0))
            {
                release_gain = sample_gain;
            }
            else