/* * sound/awe_wave.c * * The low level driver for the AWE32/Sound Blaster 32 wave table synth. * version 0.4.2c; Oct. 7, 1997 * * Copyright (C) 1996,1997 Takashi Iwai * * 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 program 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 General Public License for more details. * * You should have received a copy of the GNU General Public License * along with this program; if not, write to the Free Software * Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA. * * $DragonFly: src/sys/i386/gnu/isa/sound/Attic/awe_wave.c,v 1.3 2004/02/12 23:33:26 joerg Exp $ */ #include #if defined(__DragonFly__) || defined(__FreeBSD__) # include "awe_config.h" #else # include "awe_config.h" #endif /*----------------------------------------------------------------*/ #ifdef CONFIG_AWE32_SYNTH #if defined(__DragonFly__) || defined(__FreeBSD__) # include "awe_hw.h" # include "awe_version.h" # include "awe_voice.h" #else # include "awe_hw.h" # include "awe_version.h" # include "awe_voice.h" #endif #ifdef AWE_HAS_GUS_COMPATIBILITY /* include finetune table */ #if defined(__DragonFly__) || defined(__FreeBSD__) # ifdef AWE_OBSOLETE_VOXWARE # define SEQUENCER_C # endif # include #else # ifdef AWE_OBSOLETE_VOXWARE # include "tuning.h" # else # include "../tuning.h" # endif #endif #ifdef linux # include #elif defined(__DragonFly__) || defined(__FreeBSD__) # include #endif #endif /* AWE_HAS_GUS_COMPATIBILITY */ /*---------------------------------------------------------------- * debug message *----------------------------------------------------------------*/ static int debug_mode = 0; #ifdef AWE_DEBUG_ON #define AWE_DEBUG(LVL,XXX) {if (debug_mode > LVL) { XXX; }} #define ERRMSG(XXX) {if (debug_mode) { XXX; }} #define FATALERR(XXX) XXX #else #define AWE_DEBUG(LVL,XXX) /**/ #define ERRMSG(XXX) XXX #define FATALERR(XXX) XXX #endif /*---------------------------------------------------------------- * bank and voice record *----------------------------------------------------------------*/ /* soundfont record */ typedef struct _sf_list { unsigned short sf_id; unsigned short type; int num_info; /* current info table index */ int num_sample; /* current sample table index */ int mem_ptr; /* current word byte pointer */ int infos; int samples; /*char name[AWE_PATCH_NAME_LEN];*/ } sf_list; /* bank record */ typedef struct _awe_voice_list { int next; /* linked list with same sf_id */ unsigned char bank, instr; /* preset number information */ char type, disabled; /* type=normal/mapped, disabled=boolean */ awe_voice_info v; /* voice information */ int next_instr; /* preset table list */ int next_bank; /* preset table list */ } awe_voice_list; /* voice list type */ #define V_ST_NORMAL 0 #define V_ST_MAPPED 1 typedef struct _awe_sample_list { int next; /* linked list with same sf_id */ awe_sample_info v; /* sample information */ } awe_sample_list; /* sample and information table */ static int current_sf_id = 0; static int locked_sf_id = 0; static int max_sfs; static sf_list *sflists = NULL; #define awe_free_mem_ptr() (current_sf_id <= 0 ? 0 : sflists[current_sf_id-1].mem_ptr) #define awe_free_info() (current_sf_id <= 0 ? 0 : sflists[current_sf_id-1].num_info) #define awe_free_sample() (current_sf_id <= 0 ? 0 : sflists[current_sf_id-1].num_sample) static int max_samples; static awe_sample_list *samples = NULL; static int max_infos; static awe_voice_list *infos = NULL; #define AWE_MAX_PRESETS 256 #define AWE_DEFAULT_PRESET 0 #define AWE_DEFAULT_BANK 0 #define AWE_DEFAULT_DRUM 0 #define AWE_DRUM_BANK 128 #define MAX_LAYERS AWE_MAX_VOICES /* preset table index */ static int preset_table[AWE_MAX_PRESETS]; /*---------------------------------------------------------------- * voice table *----------------------------------------------------------------*/ /* effects table */ typedef struct FX_Rec { /* channel effects */ unsigned char flags[AWE_FX_END]; short val[AWE_FX_END]; } FX_Rec; /* channel parameters */ typedef struct _awe_chan_info { int channel; /* channel number */ int bank; /* current tone bank */ int instr; /* current program */ int bender; /* midi pitchbend (-8192 - 8192) */ int bender_range; /* midi bender range (x100) */ int panning; /* panning (0-127) */ int main_vol; /* channel volume (0-127) */ int expression_vol; /* midi expression (0-127) */ int chan_press; /* channel pressure */ int vrec; /* instrument list */ int def_vrec; /* default instrument list */ int sustained; /* sustain status in MIDI */ FX_Rec fx; /* effects */ FX_Rec fx_layer[MAX_LAYERS]; /* layer effects */ } awe_chan_info; /* voice parameters */ typedef struct _voice_info { int state; #define AWE_ST_OFF (1<<0) /* no sound */ #define AWE_ST_ON (1<<1) /* playing */ #define AWE_ST_STANDBY (1<<2) /* stand by for playing */ #define AWE_ST_SUSTAINED (1<<3) /* sustained */ #define AWE_ST_MARK (1<<4) /* marked for allocation */ #define AWE_ST_DRAM (1<<5) /* DRAM read/write */ #define AWE_ST_FM (1<<6) /* reserved for FM */ #define AWE_ST_RELEASED (1<<7) /* released */ int ch; /* midi channel */ int key; /* internal key for search */ int layer; /* layer number (for channel mode only) */ int time; /* allocated time */ awe_chan_info *cinfo; /* channel info */ int note; /* midi key (0-127) */ int velocity; /* midi velocity (0-127) */ int sostenuto; /* sostenuto on/off */ awe_voice_info *sample; /* assigned voice */ /* EMU8000 parameters */ int apitch; /* pitch parameter */ int avol; /* volume parameter */ int apan; /* panning parameter */ } voice_info; /* voice information */ static voice_info *voices; #define IS_NO_SOUND(v) (voices[v].state & (AWE_ST_OFF|AWE_ST_RELEASED|AWE_ST_STANDBY|AWE_ST_SUSTAINED)) #define IS_NO_EFFECT(v) (voices[v].state != AWE_ST_ON) #define IS_PLAYING(v) (voices[v].state & (AWE_ST_ON|AWE_ST_SUSTAINED|AWE_ST_RELEASED)) #define IS_EMPTY(v) (voices[v].state & (AWE_ST_OFF|AWE_ST_MARK|AWE_ST_DRAM|AWE_ST_FM)) /* MIDI channel effects information (for hw control) */ static awe_chan_info *channels; /*---------------------------------------------------------------- * global variables *----------------------------------------------------------------*/ #ifndef AWE_DEFAULT_BASE_ADDR #define AWE_DEFAULT_BASE_ADDR 0 /* autodetect */ #endif #ifndef AWE_DEFAULT_MEM_SIZE #define AWE_DEFAULT_MEM_SIZE 0 /* autodetect */ #endif /* awe32 base address (overwritten at initialization) */ static int awe_base = AWE_DEFAULT_BASE_ADDR; /* memory byte size */ static int awe_mem_size = AWE_DEFAULT_MEM_SIZE; /* DRAM start offset */ static int awe_mem_start = AWE_DRAM_OFFSET; /* maximum channels for playing */ static int awe_max_voices = AWE_MAX_VOICES; static int patch_opened = 0; /* sample already loaded? */ static int reverb_mode = 4; /* reverb mode */ static int chorus_mode = 2; /* chorus mode */ static short init_atten = AWE_DEFAULT_ATTENUATION; /* 12dB below */ static int awe_present = FALSE; /* awe device present? */ static int awe_busy = FALSE; /* awe device opened? */ #define DEFAULT_DRUM_FLAGS ((1 << 9) | (1 << 25)) #define IS_DRUM_CHANNEL(c) (drum_flags & (1 << (c))) #define DRUM_CHANNEL_ON(c) (drum_flags |= (1 << (c))) #define DRUM_CHANNEL_OFF(c) (drum_flags &= ~(1 << (c))) static unsigned int drum_flags = DEFAULT_DRUM_FLAGS; /* channel flags */ static int playing_mode = AWE_PLAY_INDIRECT; #define SINGLE_LAYER_MODE() (playing_mode == AWE_PLAY_INDIRECT || playing_mode == AWE_PLAY_DIRECT) #define MULTI_LAYER_MODE() (playing_mode == AWE_PLAY_MULTI || playing_mode == AWE_PLAY_MULTI2) static int current_alloc_time = 0; /* voice allocation index for channel mode */ static struct MiscModeDef { int value; int init_each_time; } misc_modes_default[AWE_MD_END] = { {0,0}, {0,0}, /* <-- not used */ {AWE_VERSION_NUMBER, FALSE}, {TRUE, TRUE}, /* exclusive */ {TRUE, TRUE}, /* realpan */ {AWE_DEFAULT_BANK, TRUE}, /* gusbank */ {FALSE, TRUE}, /* keep effect */ {AWE_DEFAULT_ATTENUATION, FALSE}, /* zero_atten */ {FALSE, TRUE}, /* chn_prior */ {AWE_DEFAULT_MOD_SENSE, TRUE}, /* modwheel sense */ {AWE_DEFAULT_PRESET, TRUE}, /* def_preset */ {AWE_DEFAULT_BANK, TRUE}, /* def_bank */ {AWE_DEFAULT_DRUM, TRUE}, /* def_drum */ {FALSE, TRUE}, /* toggle_drum_bank */ }; static int misc_modes[AWE_MD_END]; static int awe_bass_level = 5; static int awe_treble_level = 9; static struct synth_info awe_info = { "AWE32 Synth", /* name */ 0, /* device */ SYNTH_TYPE_SAMPLE, /* synth_type */ SAMPLE_TYPE_AWE32, /* synth_subtype */ 0, /* perc_mode (obsolete) */ AWE_MAX_VOICES, /* nr_voices */ 0, /* nr_drums (obsolete) */ AWE_MAX_INFOS /* instr_bank_size */ }; static struct voice_alloc_info *voice_alloc; /* set at initialization */ /*---------------------------------------------------------------- * function prototypes *----------------------------------------------------------------*/ #if defined(linux) && !defined(AWE_OBSOLETE_VOXWARE) static int awe_check_port(void); static void awe_request_region(void); static void awe_release_region(void); #endif static void awe_reset_samples(void); /* emu8000 chip i/o access */ static void awe_poke(unsigned short cmd, unsigned short port, unsigned short data); static void awe_poke_dw(unsigned short cmd, unsigned short port, unsigned int data); static unsigned short awe_peek(unsigned short cmd, unsigned short port); static unsigned int awe_peek_dw(unsigned short cmd, unsigned short port); static void awe_wait(unsigned short delay); /* initialize emu8000 chip */ static void awe_initialize(void); /* set voice parameters */ static void awe_init_misc_modes(int init_all); static void awe_init_voice_info(awe_voice_info *vp); static void awe_init_voice_parm(awe_voice_parm *pp); #ifdef AWE_HAS_GUS_COMPATIBILITY static int freq_to_note(int freq); static int calc_rate_offset(int Hz); /*static int calc_parm_delay(int msec);*/ static int calc_parm_hold(int msec); static int calc_parm_attack(int msec); static int calc_parm_decay(int msec); static int calc_parm_search(int msec, short *table); #endif /* turn on/off note */ static void awe_note_on(int voice); static void awe_note_off(int voice); static void awe_terminate(int voice); static void awe_exclusive_off(int voice); static void awe_note_off_all(int do_sustain); /* calculate voice parameters */ typedef void (*fx_affect_func)(int voice, int forced); static void awe_set_pitch(int voice, int forced); static void awe_set_voice_pitch(int voice, int forced); static void awe_set_volume(int voice, int forced); static void awe_set_voice_vol(int voice, int forced); static void awe_set_pan(int voice, int forced); static void awe_fx_fmmod(int voice, int forced); static void awe_fx_tremfrq(int voice, int forced); static void awe_fx_fm2frq2(int voice, int forced); static void awe_fx_filterQ(int voice, int forced); static void awe_calc_pitch(int voice); #ifdef AWE_HAS_GUS_COMPATIBILITY static void awe_calc_pitch_from_freq(int voice, int freq); #endif static void awe_calc_volume(int voice); static void awe_voice_init(int voice, int init_all); static void awe_channel_init(int ch, int init_all); static void awe_fx_init(int ch); /* sequencer interface */ static int awe_open(int dev, int mode); static void awe_close(int dev); static int awe_ioctl(int dev, unsigned int cmd, caddr_t arg); static int awe_kill_note(int dev, int voice, int note, int velocity); static int awe_start_note(int dev, int v, int note_num, int volume); static int awe_set_instr(int dev, int voice, int instr_no); static int awe_set_instr_2(int dev, int voice, int instr_no); static void awe_reset(int dev); static void awe_hw_control(int dev, unsigned char *event); static int awe_load_patch(int dev, int format, const char *addr, int offs, int count, int pmgr_flag); static void awe_aftertouch(int dev, int voice, int pressure); static void awe_controller(int dev, int voice, int ctrl_num, int value); static void awe_panning(int dev, int voice, int value); static void awe_volume_method(int dev, int mode); #ifndef AWE_NO_PATCHMGR static int awe_patchmgr(int dev, struct patmgr_info *rec); #endif static void awe_bender(int dev, int voice, int value); static int awe_alloc(int dev, int chn, int note, struct voice_alloc_info *alloc); static void awe_setup_voice(int dev, int voice, int chn); /* hardware controls */ #ifdef AWE_HAS_GUS_COMPATIBILITY static void awe_hw_gus_control(int dev, int cmd, unsigned char *event); #endif static void awe_hw_awe_control(int dev, int cmd, unsigned char *event); static void awe_voice_change(int voice, fx_affect_func func); static void awe_sostenuto_on(int voice, int forced); static void awe_sustain_off(int voice, int forced); static void awe_terminate_and_init(int voice, int forced); /* voice search */ static int awe_search_instr(int bank, int preset); static int awe_search_multi_voices(int rec, int note, int velocity, awe_voice_info **vlist); static void awe_alloc_multi_voices(int ch, int note, int velocity, int key); static void awe_alloc_one_voice(int voice, int note, int velocity); static int awe_clear_voice(void); /* load / remove patches */ static int awe_open_patch(awe_patch_info *patch, const char *addr, int count); static int awe_close_patch(awe_patch_info *patch, const char *addr, int count); static int awe_unload_patch(awe_patch_info *patch, const char *addr, int count); static int awe_load_info(awe_patch_info *patch, const char *addr, int count); static int awe_load_data(awe_patch_info *patch, const char *addr, int count); static int awe_replace_data(awe_patch_info *patch, const char *addr, int count); static int awe_load_map(awe_patch_info *patch, const char *addr, int count); #ifdef AWE_HAS_GUS_COMPATIBILITY static int awe_load_guspatch(const char *addr, int offs, int size, int pmgr_flag); #endif static int check_patch_opened(int type, char *name); static int awe_write_wave_data(const char *addr, int offset, awe_sample_info *sp, int channels); static void add_sf_info(int rec); static void add_sf_sample(int rec); static void purge_old_list(int rec, int next); static void add_info_list(int rec); static void awe_remove_samples(int sf_id); static void rebuild_preset_list(void); static short awe_set_sample(awe_voice_info *vp); /* lowlevel functions */ static void awe_init_audio(void); static void awe_init_dma(void); static void awe_init_array(void); static void awe_send_array(unsigned short *data); static void awe_tweak_voice(int voice); static void awe_tweak(void); static void awe_init_fm(void); static int awe_open_dram_for_write(int offset, int channels); static void awe_open_dram_for_check(void); static void awe_close_dram(void); static void awe_write_dram(unsigned short c); static int awe_detect_base(int addr); static int awe_detect(void); static int awe_check_dram(void); static int awe_load_chorus_fx(awe_patch_info *patch, const char *addr, int count); static void awe_set_chorus_mode(int mode); static int awe_load_reverb_fx(awe_patch_info *patch, const char *addr, int count); static void awe_set_reverb_mode(int mode); static void awe_equalizer(int bass, int treble); #ifdef CONFIG_AWE32_MIXER static int awe_mixer_ioctl(int dev, unsigned int cmd, caddr_t arg); #endif /* define macros for compatibility */ #if defined(__DragonFly__) || defined(__FreeBSD__) # include "awe_compat.h" #else # include "awe_compat.h" #endif /*---------------------------------------------------------------- * synth operation table *----------------------------------------------------------------*/ static struct synth_operations awe_operations = { #ifdef AWE_OSS38 "EMU8K", #endif &awe_info, 0, SYNTH_TYPE_SAMPLE, SAMPLE_TYPE_AWE32, awe_open, awe_close, awe_ioctl, awe_kill_note, awe_start_note, awe_set_instr_2, awe_reset, awe_hw_control, awe_load_patch, awe_aftertouch, awe_controller, awe_panning, awe_volume_method, #ifndef AWE_NO_PATCHMGR awe_patchmgr, #endif awe_bender, awe_alloc, awe_setup_voice }; #ifdef CONFIG_AWE32_MIXER static struct mixer_operations awe_mixer_operations = { #if !defined(__DragonFly__) && !defined(__FreeBSD__) "AWE32", #endif "AWE32 Equalizer", awe_mixer_ioctl, }; #endif /*================================================================ * attach / unload interface *================================================================*/ #ifdef AWE_OBSOLETE_VOXWARE #define ATTACH_DECL static #else #define ATTACH_DECL /**/ #endif #if (defined(__DragonFly__) || defined(__FreeBSD__)) && !defined(AWE_OBSOLETE_VOXWARE) # define ATTACH_RET void attach_awe(struct address_info *hw_config) #else # define ATTACH_RET ret ATTACH_DECL int attach_awe(void) #endif { int ret = 0; /* check presence of AWE32 card */ if (! awe_detect()) { printk("AWE32: not detected\n"); return ATTACH_RET; } /* check AWE32 ports are available */ if (awe_check_port()) { printk("AWE32: I/O area already used.\n"); return ATTACH_RET; } /* set buffers to NULL */ voices = NULL; channels = NULL; sflists = NULL; samples = NULL; infos = NULL; /* voice & channel info */ voices = (voice_info*)my_malloc(AWE_MAX_VOICES * sizeof(voice_info)); channels = (awe_chan_info*)my_malloc(AWE_MAX_CHANNELS * sizeof(awe_chan_info)); if (voices == NULL || channels == NULL) { my_free(voices); my_free(channels); printk("AWE32: can't allocate sample tables\n"); return ATTACH_RET; } /* allocate sample tables */ INIT_TABLE(sflists, max_sfs, AWE_MAX_SF_LISTS, sf_list); INIT_TABLE(samples, max_samples, AWE_MAX_SAMPLES, awe_sample_list); INIT_TABLE(infos, max_infos, AWE_MAX_INFOS, awe_voice_list); if (num_synths >= MAX_SYNTH_DEV) printk("AWE32 Error: too many synthesizers\n"); else { voice_alloc = &awe_operations.alloc; voice_alloc->max_voice = awe_max_voices; synth_devs[num_synths++] = &awe_operations; } #ifdef CONFIG_AWE32_MIXER if (num_mixers < MAX_MIXER_DEV) { mixer_devs[num_mixers++] = &awe_mixer_operations; } #endif /* reserve I/O ports for awedrv */ awe_request_region(); /* clear all samples */ awe_reset_samples(); /* intialize AWE32 hardware */ awe_initialize(); snprintf(awe_info.name, sizeof(awe_info.name), "AWE32-%s (RAM%dk)", AWEDRV_VERSION, awe_mem_size/1024); #if defined(__DragonFly__) || defined(__FreeBSD__) printk("awe0: ", awe_mem_size/1024); #elif defined(AWE_DEBUG_ON) printk("%s\n", awe_info.name); #endif /* set default values */ awe_init_misc_modes(TRUE); /* set reverb & chorus modes */ awe_set_reverb_mode(reverb_mode); awe_set_chorus_mode(chorus_mode); awe_present = TRUE; ret = 1; return ATTACH_RET; } #ifdef AWE_DYNAMIC_BUFFER static void free_tables(void) { my_free(sflists); sflists = NULL; max_sfs = 0; my_free(samples); samples = NULL; max_samples = 0; my_free(infos); infos = NULL; max_infos = 0; } #else #define free_buffers() /**/ #endif #ifdef linux ATTACH_DECL void unload_awe(void) { if (awe_present) { awe_reset_samples(); awe_release_region(); my_free(voices); my_free(channels); free_tables(); awe_present = FALSE; } } #endif /*---------------------------------------------------------------- * old type interface *----------------------------------------------------------------*/ #ifdef AWE_OBSOLETE_VOXWARE #if defined(__DragonFly__) || defined(__FreeBSD__) long attach_awe_obsolete(long mem_start, struct address_info *hw_config) #else int attach_awe_obsolete(int mem_start, struct address_info *hw_config) #endif { my_malloc_init(mem_start); if (! attach_awe()) return 0; return my_malloc_memptr(); } int probe_awe_obsolete(struct address_info *hw_config) { return 1; /*return awe_detect();*/ } #else #if defined(__DragonFly__) || defined(__FreeBSD__ ) int probe_awe(struct address_info *hw_config) { return 1; } #endif #endif /* AWE_OBSOLETE_VOXWARE */ /*================================================================ * clear sample tables *================================================================*/ static void awe_reset_samples(void) { int i; /* free all bank tables */ for (i = 0; i < AWE_MAX_PRESETS; i++) preset_table[i] = -1; free_tables(); current_sf_id = 0; locked_sf_id = 0; patch_opened = 0; } /*================================================================ * EMU register access *================================================================*/ /* select a given AWE32 pointer */ static int awe_cur_cmd = -1; #define awe_set_cmd(cmd) \ if (awe_cur_cmd != cmd) { OUTW(cmd, awe_base + 0x802); awe_cur_cmd = cmd; } #define awe_port(port) (awe_base - 0x620 + port) /* write 16bit data */ INLINE static void awe_poke(unsigned short cmd, unsigned short port, unsigned short data) { awe_set_cmd(cmd); OUTW(data, awe_port(port)); } /* write 32bit data */ INLINE static void awe_poke_dw(unsigned short cmd, unsigned short port, unsigned int data) { awe_set_cmd(cmd); OUTW(data, awe_port(port)); /* write lower 16 bits */ OUTW(data >> 16, awe_port(port)+2); /* write higher 16 bits */ } /* read 16bit data */ INLINE static unsigned short awe_peek(unsigned short cmd, unsigned short port) { unsigned short k; awe_set_cmd(cmd); k = inw(awe_port(port)); return k; } /* read 32bit data */ INLINE static unsigned int awe_peek_dw(unsigned short cmd, unsigned short port) { unsigned int k1, k2; awe_set_cmd(cmd); k1 = inw(awe_port(port)); k2 = inw(awe_port(port)+2); k1 |= k2 << 16; return k1; } /* wait delay number of AWE32 44100Hz clocks */ static void awe_wait(unsigned short delay) { unsigned short clock, target; unsigned short port = awe_port(AWE_WC_Port); int counter; /* sample counter */ awe_set_cmd(AWE_WC_Cmd); clock = (unsigned short)inw(port); target = clock + delay; counter = 0; if (target < clock) { for (; (unsigned short)inw(port) > target; counter++) if (counter > 65536) break; } for (; (unsigned short)inw(port) < target; counter++) if (counter > 65536) break; } /* write a word data */ INLINE static void awe_write_dram(unsigned short c) { awe_poke(AWE_SMLD, c); } #if defined(linux) && !defined(AWE_OBSOLETE_VOXWARE) /*================================================================ * port check / request * 0x620-622, 0xA20-A22, 0xE20-E22 *================================================================*/ static int awe_check_port(void) { return (check_region(awe_port(Data0), 4) || check_region(awe_port(Data1), 4) || check_region(awe_port(Data3), 4)); } static void awe_request_region(void) { request_region(awe_port(Data0), 4, "sound driver (AWE32)"); request_region(awe_port(Data1), 4, "sound driver (AWE32)"); request_region(awe_port(Data3), 4, "sound driver (AWE32)"); } static void awe_release_region(void) { release_region(awe_port(Data0), 4); release_region(awe_port(Data1), 4); release_region(awe_port(Data3), 4); } #endif /* !AWE_OBSOLETE_VOXWARE */ /*================================================================ * AWE32 initialization *================================================================*/ static void awe_initialize(void) { AWE_DEBUG(0,printk("AWE32: initializing..\n")); /* initialize hardware configuration */ awe_poke(AWE_HWCF1, 0x0059); awe_poke(AWE_HWCF2, 0x0020); /* disable audio; this seems to reduce a clicking noise a bit.. */ awe_poke(AWE_HWCF3, 0); /* initialize audio channels */ awe_init_audio(); /* initialize DMA */ awe_init_dma(); /* initialize init array */ awe_init_array(); /* check DRAM memory size */ awe_mem_size = awe_check_dram(); /* initialize the FM section of the AWE32 */ awe_init_fm(); /* set up voice envelopes */ awe_tweak(); /* enable audio */ awe_poke(AWE_HWCF3, 0x0004); /* set equalizer */ awe_equalizer(5, 9); } /*================================================================ * AWE32 voice parameters *================================================================*/ /* initialize voice_info record */ static void awe_init_voice_info(awe_voice_info *vp) { vp->sf_id = 0; /* normal mode */ vp->sample = 0; vp->rate_offset = 0; vp->start = 0; vp->end = 0; vp->loopstart = 0; vp->loopend = 0; vp->mode = 0; vp->root = 60; vp->tune = 0; vp->low = 0; vp->high = 127; vp->vellow = 0; vp->velhigh = 127; vp->fixkey = -1; vp->fixvel = -1; vp->fixpan = -1; vp->pan = -1; vp->exclusiveClass = 0; vp->amplitude = 127; vp->attenuation = 0; vp->scaleTuning = 100; awe_init_voice_parm(&vp->parm); } /* initialize voice_parm record: * Env1/2: delay=0, attack=0, hold=0, sustain=0, decay=0, release=0. * Vibrato and Tremolo effects are zero. * Cutoff is maximum. * Chorus and Reverb effects are zero. */ static void awe_init_voice_parm(awe_voice_parm *pp) { pp->moddelay = 0x8000; pp->modatkhld = 0x7f7f; pp->moddcysus = 0x7f7f; pp->modrelease = 0x807f; pp->modkeyhold = 0; pp->modkeydecay = 0; pp->voldelay = 0x8000; pp->volatkhld = 0x7f7f; pp->voldcysus = 0x7f7f; pp->volrelease = 0x807f; pp->volkeyhold = 0; pp->volkeydecay = 0; pp->lfo1delay = 0x8000; pp->lfo2delay = 0x8000; pp->pefe = 0; pp->fmmod = 0; pp->tremfrq = 0; pp->fm2frq2 = 0; pp->cutoff = 0xff; pp->filterQ = 0; pp->chorus = 0; pp->reverb = 0; } #ifdef AWE_HAS_GUS_COMPATIBILITY /* convert frequency mHz to abstract cents (= midi key * 100) */ static int freq_to_note(int mHz) { /* abscents = log(mHz/8176) / log(2) * 1200 */ unsigned int max_val = (unsigned int)0xffffffff / 10000; int i, times; unsigned int base; unsigned int freq; int note, tune; if (mHz == 0) return 0; if (mHz < 0) return 12799; /* maximum */ freq = mHz; note = 0; for (base = 8176 * 2; freq >= base; base *= 2) { note += 12; if (note >= 128) /* over maximum */ return 12799; } base /= 2; /* to avoid overflow... */ times = 10000; while (freq > max_val) { max_val *= 10; times /= 10; base /= 10; } freq = freq * times / base; for (i = 0; i < 12; i++) { if (freq < semitone_tuning[i+1]) break; note++; } tune = 0; freq = freq * 10000 / semitone_tuning[i]; for (i = 0; i < 100; i++) { if (freq < cent_tuning[i+1]) break; tune++; } return note * 100 + tune; } /* convert Hz to AWE32 rate offset: * sample pitch offset for the specified sample rate * rate=44100 is no offset, each 4096 is 1 octave (twice). * eg, when rate is 22050, this offset becomes -4096. */ static int calc_rate_offset(int Hz) { /* offset = log(Hz / 44100) / log(2) * 4096 */ int freq, base, i; /* maybe smaller than max (44100Hz) */ if (Hz <= 0 || Hz >= 44100) return 0; base = 0; for (freq = Hz * 2; freq < 44100; freq *= 2) base++; base *= 1200; freq = 44100 * 10000 / (freq/2); for (i = 0; i < 12; i++) { if (freq < semitone_tuning[i+1]) break; base += 100; } freq = freq * 10000 / semitone_tuning[i]; for (i = 0; i < 100; i++) { if (freq < cent_tuning[i+1]) break; base++; } return -base * 4096 / 1200; } /*---------------------------------------------------------------- * convert envelope time parameter to AWE32 raw parameter *----------------------------------------------------------------*/ /* attack & decay/release time table (msec) */ static short attack_time_tbl[128] = { 32767, 11878, 5939, 3959, 2969, 2375, 1979, 1696, 1484, 1319, 1187, 1079, 989, 913, 848, 791, 742, 698, 659, 625, 593, 565, 539, 516, 494, 475, 456, 439, 424, 409, 395, 383, 371, 359, 344, 330, 316, 302, 290, 277, 266, 255, 244, 233, 224, 214, 205, 196, 188, 180, 173, 165, 158, 152, 145, 139, 133, 127, 122, 117, 112, 107, 103, 98, 94, 90, 86, 83, 79, 76, 73, 69, 67, 64, 61, 58, 56, 54, 51, 49, 47, 45, 43, 41, 39, 38, 36, 35, 33, 32, 30, 29, 28, 27, 25, 24, 23, 22, 21, 20, 20, 19, 18, 17, 16, 16, 15, 14, 14, 13, 13, 12, 11, 11, 10, 10, 10, 9, 9, 8, 8, 8, 7, 7, 7, 6, 6, 0, }; static short decay_time_tbl[128] = { 32767, 32766, 4589, 4400, 4219, 4045, 3879, 3719, 3566, 3419, 3279, 3144, 3014, 2890, 2771, 2657, 2548, 2443, 2343, 2246, 2154, 2065, 1980, 1899, 1820, 1746, 1674, 1605, 1539, 1475, 1415, 1356, 1301, 1247, 1196, 1146, 1099, 1054, 1011, 969, 929, 891, 854, 819, 785, 753, 722, 692, 664, 636, 610, 585, 561, 538, 516, 494, 474, 455, 436, 418, 401, 384, 368, 353, 339, 325, 311, 298, 286, 274, 263, 252, 242, 232, 222, 213, 204, 196, 188, 180, 173, 166, 159, 152, 146, 140, 134, 129, 123, 118, 113, 109, 104, 100, 96, 92, 88, 84, 81, 77, 74, 71, 68, 65, 63, 60, 58, 55, 53, 51, 49, 47, 45, 43, 41, 39, 38, 36, 35, 33, 32, 30, 29, 28, 27, 26, 25, 24, }; /* static int calc_parm_delay(int msec) { return (0x8000 - msec * 1000 / 725); } */ /* delay time = 0x8000 - msec/92 */ static int calc_parm_hold(int msec) { int val = (0x7f * 92 - msec) / 92; if (val < 1) val = 1; if (val > 127) val = 127; return val; } /* attack time: search from time table */ static int calc_parm_attack(int msec) { return calc_parm_search(msec, attack_time_tbl); } /* decay/release time: search from time table */ static int calc_parm_decay(int msec) { return calc_parm_search(msec, decay_time_tbl); } /* search an index for specified time from given time table */ static int calc_parm_search(int msec, short *table) { int left = 1, right = 127, mid; while (left < right) { mid = (left + right) / 2; if (msec < (int)table[mid]) left = mid + 1; else right = mid; } return left; } #endif /* AWE_HAS_GUS_COMPATIBILITY */ /*================================================================ * effects table *================================================================*/ /* set an effect value */ #define FX_FLAG_OFF 0 #define FX_FLAG_SET 1 #define FX_FLAG_ADD 2 #define FX_SET(rec,type,value) \ ((rec)->flags[type] = FX_FLAG_SET, (rec)->val[type] = (value)) #define FX_ADD(rec,type,value) \ ((rec)->flags[type] = FX_FLAG_ADD, (rec)->val[type] = (value)) #define FX_UNSET(rec,type) \ ((rec)->flags[type] = FX_FLAG_OFF, (rec)->val[type] = 0) /* check the effect value is set */ #define FX_ON(rec,type) ((rec)->flags[type]) #define PARM_BYTE 0 #define PARM_WORD 1 static struct PARM_DEFS { int type; /* byte or word */ int low, high; /* value range */ fx_affect_func realtime; /* realtime paramater change */ } parm_defs[] = { {PARM_WORD, 0, 0x8000, NULL}, /* env1 delay */ {PARM_BYTE, 1, 0x7f, NULL}, /* env1 attack */ {PARM_BYTE, 0, 0x7e, NULL}, /* env1 hold */ {PARM_BYTE, 1, 0x7f, NULL}, /* env1 decay */ {PARM_BYTE, 1, 0x7f, NULL}, /* env1 release */ {PARM_BYTE, 0, 0x7f, NULL}, /* env1 sustain */ {PARM_BYTE, 0, 0xff, NULL}, /* env1 pitch */ {PARM_BYTE, 0, 0xff, NULL}, /* env1 cutoff */ {PARM_WORD, 0, 0x8000, NULL}, /* env2 delay */ {PARM_BYTE, 1, 0x7f, NULL}, /* env2 attack */ {PARM_BYTE, 0, 0x7e, NULL}, /* env2 hold */ {PARM_BYTE, 1, 0x7f, NULL}, /* env2 decay */ {PARM_BYTE, 1, 0x7f, NULL}, /* env2 release */ {PARM_BYTE, 0, 0x7f, NULL}, /* env2 sustain */ {PARM_WORD, 0, 0x8000, NULL}, /* lfo1 delay */ {PARM_BYTE, 0, 0xff, awe_fx_tremfrq}, /* lfo1 freq */ {PARM_BYTE, 0, 0x7f, awe_fx_tremfrq}, /* lfo1 volume (positive only)*/ {PARM_BYTE, 0, 0x7f, awe_fx_fmmod}, /* lfo1 pitch (positive only)*/ {PARM_BYTE, 0, 0xff, awe_fx_fmmod}, /* lfo1 cutoff (positive only)*/ {PARM_WORD, 0, 0x8000, NULL}, /* lfo2 delay */ {PARM_BYTE, 0, 0xff, awe_fx_fm2frq2}, /* lfo2 freq */ {PARM_BYTE, 0, 0x7f, awe_fx_fm2frq2}, /* lfo2 pitch (positive only)*/ {PARM_WORD, 0, 0xffff, awe_set_voice_pitch}, /* initial pitch */ {PARM_BYTE, 0, 0xff, NULL}, /* chorus */ {PARM_BYTE, 0, 0xff, NULL}, /* reverb */ {PARM_BYTE, 0, 0xff, awe_set_volume}, /* initial cutoff */ {PARM_BYTE, 0, 15, awe_fx_filterQ}, /* initial resonance */ {PARM_WORD, 0, 0xffff, NULL}, /* sample start */ {PARM_WORD, 0, 0xffff, NULL}, /* loop start */ {PARM_WORD, 0, 0xffff, NULL}, /* loop end */ {PARM_WORD, 0, 0xffff, NULL}, /* coarse sample start */ {PARM_WORD, 0, 0xffff, NULL}, /* coarse loop start */ {PARM_WORD, 0, 0xffff, NULL}, /* coarse loop end */ {PARM_BYTE, 0, 0xff, awe_set_volume}, /* initial attenuation */ }; static unsigned char FX_BYTE(FX_Rec *rec, FX_Rec *lay, int type, unsigned char value) { int effect = 0; int on = 0; if (lay && (on = FX_ON(lay, type)) != 0) effect = lay->val[type]; if (!on && (on = FX_ON(rec, type)) != 0) effect = rec->val[type]; if (on == FX_FLAG_ADD) effect += (int)value; if (on) { if (effect < parm_defs[type].low) effect = parm_defs[type].low; else if (effect > parm_defs[type].high) effect = parm_defs[type].high; return (unsigned char)effect; } return value; } /* get word effect value */ static unsigned short FX_WORD(FX_Rec *rec, FX_Rec *lay, int type, unsigned short value) { int effect = 0; int on = 0; if (lay && (on = FX_ON(lay, type)) != 0) effect = lay->val[type]; if (!on && (on = FX_ON(rec, type)) != 0) effect = rec->val[type]; if (on == FX_FLAG_ADD) effect += (int)value; if (on) { if (effect < parm_defs[type].low) effect = parm_defs[type].low; else if (effect > parm_defs[type].high) effect = parm_defs[type].high; return (unsigned short)effect; } return value; } /* get word (upper=type1/lower=type2) effect value */ static unsigned short FX_COMB(FX_Rec *rec, FX_Rec *lay, int type1, int type2, unsigned short value) { unsigned short tmp; tmp = FX_BYTE(rec, lay, type1, (unsigned char)(value >> 8)); tmp <<= 8; tmp |= FX_BYTE(rec, lay, type2, (unsigned char)(value & 0xff)); return tmp; } /* address offset */ static int FX_OFFSET(FX_Rec *rec, FX_Rec *lay, int lo, int hi, int mode) { int addr = 0; if (lay && FX_ON(lay, hi)) addr = (short)lay->val[hi]; else if (FX_ON(rec, hi)) addr = (short)rec->val[hi]; addr = addr << 15; if (lay && FX_ON(lay, lo)) addr += (short)lay->val[lo]; else if (FX_ON(rec, lo)) addr += (short)rec->val[lo]; if (!(mode & AWE_SAMPLE_8BITS)) addr /= 2; return addr; } /*================================================================ * turn on/off sample *================================================================*/ static void awe_note_on(int voice) { unsigned int temp; int addr; awe_voice_info *vp; FX_Rec *fx = &voices[voice].cinfo->fx; FX_Rec *fx_lay = NULL; if (voices[voice].layer < MAX_LAYERS) fx_lay = &voices[voice].cinfo->fx_layer[voices[voice].layer]; /* A voice sample must assigned before calling */ if ((vp = voices[voice].sample) == NULL || vp->index < 0) return; /* channel to be silent and idle */ awe_poke(AWE_DCYSUSV(voice), 0x0080); awe_poke(AWE_VTFT(voice), 0); awe_poke(AWE_CVCF(voice), 0); awe_poke(AWE_PTRX(voice), 0); awe_poke(AWE_CPF(voice), 0); /* modulation & volume envelope */ awe_poke(AWE_ENVVAL(voice), FX_WORD(fx, fx_lay, AWE_FX_ENV1_DELAY, vp->parm.moddelay)); awe_poke(AWE_ATKHLD(voice), FX_COMB(fx, fx_lay, AWE_FX_ENV1_HOLD, AWE_FX_ENV1_ATTACK, vp->parm.modatkhld)); awe_poke(AWE_DCYSUS(voice), FX_COMB(fx, fx_lay, AWE_FX_ENV1_SUSTAIN, AWE_FX_ENV1_DECAY, vp->parm.moddcysus)); awe_poke(AWE_ENVVOL(voice), FX_WORD(fx, fx_lay, AWE_FX_ENV2_DELAY, vp->parm.voldelay)); awe_poke(AWE_ATKHLDV(voice), FX_COMB(fx, fx_lay, AWE_FX_ENV2_HOLD, AWE_FX_ENV2_ATTACK, vp->parm.volatkhld)); /* decay/sustain parameter for volume envelope must be set at last */ /* pitch offset */ awe_set_pitch(voice, TRUE); /* cutoff and volume */ awe_set_volume(voice, TRUE); /* modulation envelope heights */ awe_poke(AWE_PEFE(voice), FX_COMB(fx, fx_lay, AWE_FX_ENV1_PITCH, AWE_FX_ENV1_CUTOFF, vp->parm.pefe)); /* lfo1/2 delay */ awe_poke(AWE_LFO1VAL(voice), FX_WORD(fx, fx_lay, AWE_FX_LFO1_DELAY, vp->parm.lfo1delay)); awe_poke(AWE_LFO2VAL(voice), FX_WORD(fx, fx_lay, AWE_FX_LFO2_DELAY, vp->parm.lfo2delay)); /* lfo1 pitch & cutoff shift */ awe_fx_fmmod(voice, TRUE); /* lfo1 volume & freq */ awe_fx_tremfrq(voice, TRUE); /* lfo2 pitch & freq */ awe_fx_fm2frq2(voice, TRUE); /* pan & loop start */ awe_set_pan(voice, TRUE); /* chorus & loop end (chorus 8bit, MSB) */ addr = vp->loopend - 1; addr += FX_OFFSET(fx, fx_lay, AWE_FX_LOOP_END, AWE_FX_COARSE_LOOP_END, vp->mode); temp = FX_BYTE(fx, fx_lay, AWE_FX_CHORUS, vp->parm.chorus); temp = (temp <<24) | (unsigned int)addr; awe_poke_dw(AWE_CSL(voice), temp); AWE_DEBUG(4,printk("AWE32: [-- loopend=%x/%x]\n", vp->loopend, addr)); /* Q & current address (Q 4bit value, MSB) */ addr = vp->start - 1; addr += FX_OFFSET(fx, fx_lay, AWE_FX_SAMPLE_START, AWE_FX_COARSE_SAMPLE_START, vp->mode); temp = FX_BYTE(fx, fx_lay, AWE_FX_FILTERQ, vp->parm.filterQ); temp = (temp<<28) | (unsigned int)addr; awe_poke_dw(AWE_CCCA(voice), temp); AWE_DEBUG(4,printk("AWE32: [-- startaddr=%x/%x]\n", vp->start, addr)); /* reset volume */ awe_poke_dw(AWE_VTFT(voice), 0x0000FFFF); awe_poke_dw(AWE_CVCF(voice), 0x0000FFFF); /* turn on envelope */ awe_poke(AWE_DCYSUSV(voice), FX_COMB(fx, fx_lay, AWE_FX_ENV2_SUSTAIN, AWE_FX_ENV2_DECAY, vp->parm.voldcysus)); /* set reverb */ temp = FX_BYTE(fx, fx_lay, AWE_FX_REVERB, vp->parm.reverb); temp = (awe_peek_dw(AWE_PTRX(voice)) & 0xffff0000) | (temp<<8); awe_poke_dw(AWE_PTRX(voice), temp); awe_poke_dw(AWE_CPF(voice), 0x40000000); voices[voice].state = AWE_ST_ON; /* clear voice position for the next note on this channel */ if (SINGLE_LAYER_MODE()) { FX_UNSET(fx, AWE_FX_SAMPLE_START); FX_UNSET(fx, AWE_FX_COARSE_SAMPLE_START); } } /* turn off the voice */ static void awe_note_off(int voice) { awe_voice_info *vp; unsigned short tmp; FX_Rec *fx = &voices[voice].cinfo->fx; FX_Rec *fx_lay = NULL; if (voices[voice].layer < MAX_LAYERS) fx_lay = &voices[voice].cinfo->fx_layer[voices[voice].layer]; if ((vp = voices[voice].sample) == NULL) { voices[voice].state = AWE_ST_OFF; return; } tmp = 0x8000 | FX_BYTE(fx, fx_lay, AWE_FX_ENV1_RELEASE, (unsigned char)vp->parm.modrelease); awe_poke(AWE_DCYSUS(voice), tmp); tmp = 0x8000 | FX_BYTE(fx, fx_lay, AWE_FX_ENV2_RELEASE, (unsigned char)vp->parm.volrelease); awe_poke(AWE_DCYSUSV(voice), tmp); voices[voice].state = AWE_ST_RELEASED; } /* force to terminate the voice (no releasing echo) */ static void awe_terminate(int voice) { awe_poke(AWE_DCYSUSV(voice), 0x807F); awe_tweak_voice(voice); voices[voice].state = AWE_ST_OFF; } /* turn off other voices with the same exclusive class (for drums) */ static void awe_exclusive_off(int voice) { int i, exclass; if (voices[voice].sample == NULL) return; if ((exclass = voices[voice].sample->exclusiveClass) == 0) return; /* not exclusive */ /* turn off voices with the same class */ for (i = 0; i < awe_max_voices; i++) { if (i != voice && IS_PLAYING(i) && voices[i].sample && voices[i].ch == voices[voice].ch && voices[i].sample->exclusiveClass == exclass) { AWE_DEBUG(4,printk("AWE32: [exoff(%d)]\n", i)); awe_terminate(i); awe_voice_init(i, TRUE); } } } /*================================================================ * change the parameters of an audible voice *================================================================*/ /* change pitch */ static void awe_set_pitch(int voice, int forced) { if (IS_NO_EFFECT(voice) && !forced) return; awe_poke(AWE_IP(voice), voices[voice].apitch); AWE_DEBUG(3,printk("AWE32: [-- pitch=%x]\n", voices[voice].apitch)); } /* calculate & change pitch */ static void awe_set_voice_pitch(int voice, int forced) { awe_calc_pitch(voice); awe_set_pitch(voice, forced); } /* change volume & cutoff */ static void awe_set_volume(int voice, int forced) { awe_voice_info *vp; unsigned short tmp2; FX_Rec *fx = &voices[voice].cinfo->fx; FX_Rec *fx_lay = NULL; if (voices[voice].layer < MAX_LAYERS) fx_lay = &voices[voice].cinfo->fx_layer[voices[voice].layer]; if (!IS_PLAYING(voice) && !forced) return; if ((vp = voices[voice].sample) == NULL || vp->index < 0) return; tmp2 = FX_BYTE(fx, fx_lay, AWE_FX_CUTOFF, vp->parm.cutoff); tmp2 = (tmp2 << 8); tmp2 |= FX_BYTE(fx, fx_lay, AWE_FX_ATTEN, (unsigned char)voices[voice].avol); awe_poke(AWE_IFATN(voice), tmp2); } /* calculate & change volume */ static void awe_set_voice_vol(int voice, int forced) { if (IS_EMPTY(voice)) return; awe_calc_volume(voice); awe_set_volume(voice, forced); } /* change pan; this could make a click noise.. */ static void awe_set_pan(int voice, int forced) { unsigned int temp; int addr; awe_voice_info *vp; FX_Rec *fx = &voices[voice].cinfo->fx; FX_Rec *fx_lay = NULL; if (voices[voice].layer < MAX_LAYERS) fx_lay = &voices[voice].cinfo->fx_layer[voices[voice].layer]; if (IS_NO_EFFECT(voice) && !forced) return; if ((vp = voices[voice].sample) == NULL || vp->index < 0) return; /* pan & loop start (pan 8bit, MSB, 0:right, 0xff:left) */ if (vp->fixpan > 0) /* 0-127 */ temp = 255 - (int)vp->fixpan * 2; else { int pos = 0; if (vp->pan >= 0) /* 0-127 */ pos = (int)vp->pan * 2 - 128; pos += voices[voice].cinfo->panning; /* -128 - 127 */ pos = 127 - pos; if (pos < 0) temp = 0; else if (pos > 255) temp = 255; else temp = pos; } if (forced || temp != voices[voice].apan) { addr = vp->loopstart - 1; addr += FX_OFFSET(fx, fx_lay, AWE_FX_LOOP_START, AWE_FX_COARSE_LOOP_START, vp->mode); temp = (temp<<24) | (unsigned int)addr; awe_poke_dw(AWE_PSST(voice), temp); voices[voice].apan = temp; AWE_DEBUG(4,printk("AWE32: [-- loopstart=%x/%x]\n", vp->loopstart, addr)); } } /* effects change during playing */ static void awe_fx_fmmod(int voice, int forced) { awe_voice_info *vp; FX_Rec *fx = &voices[voice].cinfo->fx; FX_Rec *fx_lay = NULL; if (voices[voice].layer < MAX_LAYERS) fx_lay = &voices[voice].cinfo->fx_layer[voices[voice].layer]; if (IS_NO_EFFECT(voice) && !forced) return; if ((vp = voices[voice].sample) == NULL || vp->index < 0) return; awe_poke(AWE_FMMOD(voice), FX_COMB(fx, fx_lay, AWE_FX_LFO1_PITCH, AWE_FX_LFO1_CUTOFF, vp->parm.fmmod)); } /* set tremolo (lfo1) volume & frequency */ static void awe_fx_tremfrq(int voice, int forced) { awe_voice_info *vp; FX_Rec *fx = &voices[voice].cinfo->fx; FX_Rec *fx_lay = NULL; if (voices[voice].layer < MAX_LAYERS) fx_lay = &voices[voice].cinfo->fx_layer[voices[voice].layer]; if (IS_NO_EFFECT(voice) && !forced) return; if ((vp = voices[voice].sample) == NULL || vp->index < 0) return; awe_poke(AWE_TREMFRQ(voice), FX_COMB(fx, fx_lay, AWE_FX_LFO1_VOLUME, AWE_FX_LFO1_FREQ, vp->parm.tremfrq)); } /* set lfo2 pitch & frequency */ static void awe_fx_fm2frq2(int voice, int forced) { awe_voice_info *vp; FX_Rec *fx = &voices[voice].cinfo->fx; FX_Rec *fx_lay = NULL; if (voices[voice].layer < MAX_LAYERS) fx_lay = &voices[voice].cinfo->fx_layer[voices[voice].layer]; if (IS_NO_EFFECT(voice) && !forced) return; if ((vp = voices[voice].sample) == NULL || vp->index < 0) return; awe_poke(AWE_FM2FRQ2(voice), FX_COMB(fx, fx_lay, AWE_FX_LFO2_PITCH, AWE_FX_LFO2_FREQ, vp->parm.fm2frq2)); } /* Q & current address (Q 4bit value, MSB) */ static void awe_fx_filterQ(int voice, int forced) { unsigned int addr; awe_voice_info *vp; FX_Rec *fx = &voices[voice].cinfo->fx; FX_Rec *fx_lay = NULL; if (voices[voice].layer < MAX_LAYERS) fx_lay = &voices[voice].cinfo->fx_layer[voices[voice].layer]; if (IS_NO_EFFECT(voice) && !forced) return; if ((vp = voices[voice].sample) == NULL || vp->index < 0) return; addr = awe_peek_dw(AWE_CCCA(voice)) & 0xffffff; addr |= (FX_BYTE(fx, fx_lay, AWE_FX_FILTERQ, vp->parm.filterQ) << 28); awe_poke_dw(AWE_CCCA(voice), addr); } /*================================================================ * calculate pitch offset *---------------------------------------------------------------- * 0xE000 is no pitch offset at 44100Hz sample. * Every 4096 is one octave. *================================================================*/ static void awe_calc_pitch(int voice) { voice_info *vp = &voices[voice]; awe_voice_info *ap; awe_chan_info *cp = voices[voice].cinfo; int offset; /* search voice information */ if ((ap = vp->sample) == NULL) return; if (ap->index < 0) { AWE_DEBUG(3,printk("AWE32: set sample (%d)\n", ap->sample)); if (awe_set_sample(ap) < 0) return; } /* calculate offset */ if (ap->fixkey >= 0) { AWE_DEBUG(3,printk("AWE32: p-> fixkey(%d) tune(%d)\n", ap->fixkey, ap->tune)); offset = (ap->fixkey - ap->root) * 4096 / 12; } else { AWE_DEBUG(3,printk("AWE32: p(%d)-> root(%d) tune(%d)\n", vp->note, ap->root, ap->tune)); offset = (vp->note - ap->root) * 4096 / 12; AWE_DEBUG(4,printk("AWE32: p-> ofs=%d\n", offset)); } offset = (offset * ap->scaleTuning) / 100; AWE_DEBUG(4,printk("AWE32: p-> scale* ofs=%d\n", offset)); offset += ap->tune * 4096 / 1200; AWE_DEBUG(4,printk("AWE32: p-> tune+ ofs=%d\n", offset)); if (cp->bender != 0) { AWE_DEBUG(3,printk("AWE32: p-> bend(%d) %d\n", voice, cp->bender)); /* (819200: 1 semitone) ==> (4096: 12 semitones) */ offset += cp->bender * cp->bender_range / 2400; } /* add initial pitch correction */ if (FX_ON(&cp->fx_layer[vp->layer], AWE_FX_INIT_PITCH)) offset += cp->fx_layer[vp->layer].val[AWE_FX_INIT_PITCH]; else if (FX_ON(&cp->fx, AWE_FX_INIT_PITCH)) offset += cp->fx.val[AWE_FX_INIT_PITCH]; /* 0xe000: root pitch */ vp->apitch = 0xe000 + ap->rate_offset + offset; AWE_DEBUG(4,printk("AWE32: p-> sum aofs=%x, rate_ofs=%d\n", vp->apitch, ap->rate_offset)); if (vp->apitch > 0xffff) vp->apitch = 0xffff; if (vp->apitch < 0) vp->apitch = 0; } #ifdef AWE_HAS_GUS_COMPATIBILITY /* calculate MIDI key and semitone from the specified frequency */ static void awe_calc_pitch_from_freq(int voice, int freq) { voice_info *vp = &voices[voice]; awe_voice_info *ap; FX_Rec *fx = &voices[voice].cinfo->fx; FX_Rec *fx_lay = NULL; int offset; int note; if (voices[voice].layer < MAX_LAYERS) fx_lay = &voices[voice].cinfo->fx_layer[voices[voice].layer]; /* search voice information */ if ((ap = vp->sample) == NULL) return; if (ap->index < 0) { AWE_DEBUG(3,printk("AWE32: set sample (%d)\n", ap->sample)); if (awe_set_sample(ap) < 0) return; } note = freq_to_note(freq); offset = (note - ap->root * 100 + ap->tune) * 4096 / 1200; offset = (offset * ap->scaleTuning) / 100; if (fx_lay && FX_ON(fx_lay, AWE_FX_INIT_PITCH)) offset += fx_lay->val[AWE_FX_INIT_PITCH]; else if (FX_ON(fx, AWE_FX_INIT_PITCH)) offset += fx->val[AWE_FX_INIT_PITCH]; vp->apitch = 0xe000 + ap->rate_offset + offset; if (vp->apitch > 0xffff) vp->apitch = 0xffff; if (vp->apitch < 0) vp->apitch = 0; } #endif /* AWE_HAS_GUS_COMPATIBILITY */ /*================================================================ * calculate volume attenuation *---------------------------------------------------------------- * Voice volume is controlled by volume attenuation parameter. * So volume becomes maximum when avol is 0 (no attenuation), and * minimum when 255 (-96dB or silence). *================================================================*/ static int vol_table[128] = { 255,111,95,86,79,74,70,66,63,61,58,56,54,52,50,49, 47,46,45,43,42,41,40,39,38,37,36,35,34,34,33,32, 31,31,30,29,29,28,27,27,26,26,25,24,24,23,23,22, 22,21,21,21,20,20,19,19,18,18,18,17,17,16,16,16, 15,15,15,14,14,14,13,13,13,12,12,12,11,11,11,10, 10,10,10,9,9,9,8,8,8,8,7,7,7,7,6,6, 6,6,5,5,5,5,5,4,4,4,4,3,3,3,3,3, 2,2,2,2,2,1,1,1,1,1,0,0,0,0,0,0, }; static void awe_calc_volume(int voice) { voice_info *vp = &voices[voice]; awe_voice_info *ap; awe_chan_info *cp = voices[voice].cinfo; int vol; /* search voice information */ if ((ap = vp->sample) == NULL) return; ap = vp->sample; if (ap->index < 0) { AWE_DEBUG(3,printk("AWE32: set sample (%d)\n", ap->sample)); if (awe_set_sample(ap) < 0) return; } /* 0 - 127 */ vol = (vp->velocity * cp->main_vol * cp->expression_vol) / (127*127); vol = vol * ap->amplitude / 127; if (vol < 0) vol = 0; if (vol > 127) vol = 127; /* calc to attenuation */ vol = vol_table[vol]; vol = vol + (int)ap->attenuation + init_atten; if (vol > 255) vol = 255; vp->avol = vol; AWE_DEBUG(3,printk("AWE32: [-- voice(%d) vol=%x]\n", voice, vol)); } /* set sostenuto on */ static void awe_sostenuto_on(int voice, int forced) { if (IS_NO_EFFECT(voice) && !forced) return; voices[voice].sostenuto = 127; } /* drop sustain */ static void awe_sustain_off(int voice, int forced) { if (voices[voice].state == AWE_ST_SUSTAINED) { awe_note_off(voice); awe_fx_init(voices[voice].ch); awe_voice_init(voice, FALSE); } } /* terminate and initialize voice */ static void awe_terminate_and_init(int voice, int forced) { awe_terminate(voice); awe_fx_init(voices[voice].ch); awe_voice_init(voice, TRUE); } /*================================================================ * synth operation routines *================================================================*/ #define AWE_VOICE_KEY(v) (0x8000 | (v)) #define AWE_CHAN_KEY(c,n) (((c) << 8) | ((n) + 1)) #define KEY_CHAN_MATCH(key,c) (((key) >> 8) == (c)) /* initialize the voice */ static void awe_voice_init(int voice, int init_all) { voice_info *vp = &voices[voice]; /* reset voice search key */ if (playing_mode == AWE_PLAY_DIRECT) vp->key = AWE_VOICE_KEY(voice); else vp->key = 0; /* clear voice mapping */ voice_alloc->map[voice] = 0; /* touch the timing flag */ vp->time = current_alloc_time; /* initialize other parameters if necessary */ if (init_all) { vp->note = -1; vp->velocity = 0; vp->sostenuto = 0; vp->sample = NULL; vp->cinfo = &channels[voice]; vp->ch = voice; vp->state = AWE_ST_OFF; /* emu8000 parameters */ vp->apitch = 0; vp->avol = 255; vp->apan = -1; } } /* clear effects */ static void awe_fx_init(int ch) { if (SINGLE_LAYER_MODE() && !misc_modes[AWE_MD_KEEP_EFFECT]) { BZERO(&channels[ch].fx, sizeof(channels[ch].fx)); BZERO(&channels[ch].fx_layer, sizeof(&channels[ch].fx_layer)); } } /* initialize channel info */ static void awe_channel_init(int ch, int init_all) { awe_chan_info *cp = &channels[ch]; cp->channel = ch; if (init_all) { cp->panning = 0; /* zero center */ cp->bender_range = 200; /* sense * 100 */ cp->main_vol = 127; if (MULTI_LAYER_MODE() && IS_DRUM_CHANNEL(ch)) { cp->instr = misc_modes[AWE_MD_DEF_DRUM]; cp->bank = AWE_DRUM_BANK; } else { cp->instr = misc_modes[AWE_MD_DEF_PRESET]; cp->bank = misc_modes[AWE_MD_DEF_BANK]; } cp->vrec = -1; cp->def_vrec = -1; } cp->bender = 0; /* zero tune skew */ cp->expression_vol = 127; cp->chan_press = 0; cp->sustained = 0; if (! misc_modes[AWE_MD_KEEP_EFFECT]) { BZERO(&cp->fx, sizeof(cp->fx)); BZERO(&cp->fx_layer, sizeof(cp->fx_layer)); } } /* change the voice parameters; voice = channel */ static void awe_voice_change(int voice, fx_affect_func func) { int i; switch (playing_mode) { case AWE_PLAY_DIRECT: func(voice, FALSE); break; case AWE_PLAY_INDIRECT: for (i = 0; i < awe_max_voices; i++) if (voices[i].key == AWE_VOICE_KEY(voice)) func(i, FALSE); break; default: for (i = 0; i < awe_max_voices; i++) if (KEY_CHAN_MATCH(voices[i].key, voice)) func(i, FALSE); break; } } /*---------------------------------------------------------------- * device open / close *----------------------------------------------------------------*/ /* open device: * reset status of all voices, and clear sample position flag */ static int awe_open(int dev, int mode) { if (awe_busy) return RET_ERROR(EBUSY); awe_busy = TRUE; /* set default mode */ awe_init_misc_modes(FALSE); init_atten = misc_modes[AWE_MD_ZERO_ATTEN]; drum_flags = DEFAULT_DRUM_FLAGS; playing_mode = AWE_PLAY_INDIRECT; /* reset voices & channels */ awe_reset(dev); patch_opened = 0; return 0; } /* close device: * reset all voices again (terminate sounds) */ static void awe_close(int dev) { awe_reset(dev); awe_busy = FALSE; } /* set miscellaneous mode parameters */ static void awe_init_misc_modes(int init_all) { int i; for (i = 0; i < AWE_MD_END; i++) { if (init_all || misc_modes_default[i].init_each_time) misc_modes[i] = misc_modes_default[i].value; } } /* sequencer I/O control: */ static int awe_ioctl(int dev, unsigned int cmd, caddr_t arg) { switch (cmd) { case SNDCTL_SYNTH_INFO: if (playing_mode == AWE_PLAY_DIRECT) awe_info.nr_voices = awe_max_voices; else awe_info.nr_voices = AWE_MAX_CHANNELS; IOCTL_TO_USER((char*)arg, 0, &awe_info, sizeof(awe_info)); return 0; break; case SNDCTL_SEQ_RESETSAMPLES: awe_reset_samples(); awe_reset(dev); return 0; break; case SNDCTL_SEQ_PERCMODE: /* what's this? */ return 0; break; case SNDCTL_SYNTH_MEMAVL: return awe_mem_size - awe_free_mem_ptr() * 2; default: printk("AWE32: unsupported ioctl %d\n", cmd); return RET_ERROR(EINVAL); } } static int voice_in_range(int voice) { if (playing_mode == AWE_PLAY_DIRECT) { if (voice < 0 || voice >= awe_max_voices) return FALSE; } else { if (voice < 0 || voice >= AWE_MAX_CHANNELS) return FALSE; } return TRUE; } static void release_voice(int voice, int do_sustain) { if (IS_NO_SOUND(voice)) return; if (do_sustain && (voices[voice].cinfo->sustained == 127 || voices[voice].sostenuto == 127)) voices[voice].state = AWE_ST_SUSTAINED; else { awe_note_off(voice); awe_fx_init(voices[voice].ch); awe_voice_init(voice, FALSE); } } /* release all notes */ static void awe_note_off_all(int do_sustain) { int i; for (i = 0; i < awe_max_voices; i++) release_voice(i, do_sustain); } /* kill a voice: * not terminate, just release the voice. */ static int awe_kill_note(int dev, int voice, int note, int velocity) { int i, v2, key; AWE_DEBUG(2,printk("AWE32: [off(%d) nt=%d vl=%d]\n", voice, note, velocity)); if (! voice_in_range(voice)) return RET_ERROR(EINVAL); switch (playing_mode) { case AWE_PLAY_DIRECT: case AWE_PLAY_INDIRECT: key = AWE_VOICE_KEY(voice); break; case AWE_PLAY_MULTI2: v2 = voice_alloc->map[voice] >> 8; voice_alloc->map[voice] = 0; voice = v2; if (voice < 0 || voice >= AWE_MAX_CHANNELS) return RET_ERROR(EINVAL); /* continue to below */ default: key = AWE_CHAN_KEY(voice, note); break; } for (i = 0; i < awe_max_voices; i++) { if (voices[i].key == key) release_voice(i, TRUE); } return 0; } static void start_or_volume_change(int voice, int velocity) { voices[voice].velocity = velocity; awe_calc_volume(voice); if (voices[voice].state == AWE_ST_STANDBY) awe_note_on(voice); else if (voices[voice].state == AWE_ST_ON) awe_set_volume(voice, FALSE); } static void set_and_start_voice(int voice, int state) { /* calculate pitch & volume parameters */ voices[voice].state = state; awe_calc_pitch(voice); awe_calc_volume(voice); if (state == AWE_ST_ON) awe_note_on(voice); } /* start a voice: * if note is 255, identical with aftertouch function. * Otherwise, start a voice with specified not and volume. */ static int awe_start_note(int dev, int voice, int note, int velocity) { int i, key, state, volonly; AWE_DEBUG(2,printk("AWE32: [on(%d) nt=%d vl=%d]\n", voice, note, velocity)); if (! voice_in_range(voice)) return RET_ERROR(EINVAL); if (velocity == 0) state = AWE_ST_STANDBY; /* stand by for playing */ else state = AWE_ST_ON; /* really play */ volonly = FALSE; switch (playing_mode) { case AWE_PLAY_DIRECT: case AWE_PLAY_INDIRECT: key = AWE_VOICE_KEY(voice); if (note == 255) volonly = TRUE; break; case AWE_PLAY_MULTI2: voice = voice_alloc->map[voice] >> 8; if (voice < 0 || voice >= AWE_MAX_CHANNELS) return RET_ERROR(EINVAL); /* continue to below */ default: if (note >= 128) { /* key volume mode */ note -= 128; volonly = TRUE; } key = AWE_CHAN_KEY(voice, note); break; } /* dynamic volume change */ if (volonly) { for (i = 0; i < awe_max_voices; i++) { if (voices[i].key == key) start_or_volume_change(i, velocity); } return 0; } /* if the same note still playing, stop it */ for (i = 0; i < awe_max_voices; i++) if (voices[i].key == key) { if (voices[i].state == AWE_ST_ON) { awe_note_off(i); awe_voice_init(i, FALSE); } else if (voices[i].state == AWE_ST_STANDBY) awe_voice_init(i, TRUE); } /* allocate voices */ if (playing_mode == AWE_PLAY_DIRECT) awe_alloc_one_voice(voice, note, velocity); else awe_alloc_multi_voices(voice, note, velocity, key); /* turn off other voices exlusively (for drums) */ for (i = 0; i < awe_max_voices; i++) if (voices[i].key == key) awe_exclusive_off(i); /* set up pitch and volume parameters */ for (i = 0; i < awe_max_voices; i++) { if (voices[i].key == key && voices[i].state == AWE_ST_OFF) set_and_start_voice(i, state); } return 0; } /* search instrument from preset table with the specified bank */ static int awe_search_instr(int bank, int preset) { int i; for (i = preset_table[preset]; i >= 0; i = infos[i].next_bank) { if (infos[i].bank == bank) return i; } return -1; } /* assign the instrument to a voice */ static int awe_set_instr_2(int dev, int voice, int instr_no) { if (playing_mode == AWE_PLAY_MULTI2) { voice = voice_alloc->map[voice] >> 8; if (voice < 0 || voice >= AWE_MAX_CHANNELS) return RET_ERROR(EINVAL); } return awe_set_instr(dev, voice, instr_no); } /* assign the instrument to a channel; voice is the channel number */ static int awe_set_instr(int dev, int voice, int instr_no) { awe_chan_info *cinfo; int def_bank; if (! voice_in_range(voice)) return RET_ERROR(EINVAL); if (instr_no < 0 || instr_no >= AWE_MAX_PRESETS) return RET_ERROR(EINVAL); cinfo = &channels[voice]; if (MULTI_LAYER_MODE() && IS_DRUM_CHANNEL(voice)) def_bank = AWE_DRUM_BANK; /* always search drumset */ else def_bank = cinfo->bank; cinfo->vrec = -1; cinfo->def_vrec = -1; cinfo->vrec = awe_search_instr(def_bank, instr_no); if (def_bank == AWE_DRUM_BANK) /* search default drumset */ cinfo->def_vrec = awe_search_instr(def_bank, misc_modes[AWE_MD_DEF_DRUM]); else /* search default preset */ cinfo->def_vrec = awe_search_instr(misc_modes[AWE_MD_DEF_BANK], instr_no); if (cinfo->vrec < 0 && cinfo->def_vrec < 0) { AWE_DEBUG(1,printk("AWE32 Warning: can't find instrument %d\n", instr_no)); } cinfo->instr = instr_no; return 0; } /* reset all voices; terminate sounds and initialize parameters */ static void awe_reset(int dev) { int i; current_alloc_time = 0; /* don't turn off voice 31 and 32. they are used also for FM voices */ for (i = 0; i < awe_max_voices; i++) { awe_terminate(i); awe_voice_init(i, TRUE); } for (i = 0; i < AWE_MAX_CHANNELS; i++) awe_channel_init(i, TRUE); for (i = 0; i < 16; i++) { awe_operations.chn_info[i].controllers[CTL_MAIN_VOLUME] = 127; awe_operations.chn_info[i].controllers[CTL_EXPRESSION] = 127; } awe_init_fm(); awe_tweak(); } /* hardware specific control: * GUS specific and AWE32 specific controls are available. */ static void awe_hw_control(int dev, unsigned char *event) { int cmd = event[2]; if (cmd & _AWE_MODE_FLAG) awe_hw_awe_control(dev, cmd & _AWE_MODE_VALUE_MASK, event); #ifdef AWE_HAS_GUS_COMPATIBILITY else awe_hw_gus_control(dev, cmd & _AWE_MODE_VALUE_MASK, event); #endif } #ifdef AWE_HAS_GUS_COMPATIBILITY /* GUS compatible controls */ static void awe_hw_gus_control(int dev, int cmd, unsigned char *event) { int voice, i, key; unsigned short p1; short p2; int plong; if (MULTI_LAYER_MODE()) return; if (cmd == _GUS_NUMVOICES) return; voice = event[3]; if (! voice_in_range(voice)) return; p1 = *(unsigned short *) &event[4]; p2 = *(short *) &event[6]; plong = *(int*) &event[4]; switch (cmd) { case _GUS_VOICESAMPLE: awe_set_instr(dev, voice, p1); return; case _GUS_VOICEBALA: /* 0 to 15 --> -128 to 127 */ awe_panning(dev, voice, ((int)p1 << 4) - 128); return; case _GUS_VOICEVOL: case _GUS_VOICEVOL2: /* not supported yet */ return; case _GUS_RAMPRANGE: case _GUS_RAMPRATE: case _GUS_RAMPMODE: case _GUS_RAMPON: case _GUS_RAMPOFF: /* volume ramping not supported */ return; case _GUS_VOLUME_SCALE: return; case _GUS_VOICE_POS: FX_SET(&channels[voice].fx, AWE_FX_SAMPLE_START, (short)(plong & 0x7fff)); FX_SET(&channels[voice].fx, AWE_FX_COARSE_SAMPLE_START, (plong >> 15) & 0xffff); return; } key = AWE_VOICE_KEY(voice); for (i = 0; i < awe_max_voices; i++) { if (voices[i].key == key) { switch (cmd) { case _GUS_VOICEON: awe_note_on(i); break; case _GUS_VOICEOFF: awe_terminate(i); awe_fx_init(voices[i].ch); awe_voice_init(i, TRUE); break; case _GUS_VOICEFADE: awe_note_off(i); awe_fx_init(voices[i].ch); awe_voice_init(i, FALSE); break; case _GUS_VOICEFREQ: awe_calc_pitch_from_freq(i, plong); break; } } } } #endif /* AWE32 specific controls */ static void awe_hw_awe_control(int dev, int cmd, unsigned char *event) { int voice; unsigned short p1; short p2; awe_chan_info *cinfo; FX_Rec *fx; int i; voice = event[3]; if (! voice_in_range(voice)) return; if (playing_mode == AWE_PLAY_MULTI2) { voice = voice_alloc->map[voice] >> 8; if (voice < 0 || voice >= AWE_MAX_CHANNELS) return; } p1 = *(unsigned short *) &event[4]; p2 = *(short *) &event[6]; cinfo = &channels[voice]; switch (cmd) { case _AWE_DEBUG_MODE: debug_mode = p1; printk("AWE32: debug mode = %d\n", debug_mode); break; case _AWE_REVERB_MODE: awe_set_reverb_mode(p1); break; case _AWE_CHORUS_MODE: awe_set_chorus_mode(p1); break; case _AWE_REMOVE_LAST_SAMPLES: AWE_DEBUG(0,printk("AWE32: remove last samples\n")); if (locked_sf_id > 0) awe_remove_samples(locked_sf_id); break; case _AWE_INITIALIZE_CHIP: awe_initialize(); break; case _AWE_SEND_EFFECT: fx = &cinfo->fx; i = FX_FLAG_SET; if (p1 >= 0x100) { int layer = (p1 >> 8); if (layer >= 0 && layer < MAX_LAYERS) fx = &cinfo->fx_layer[layer]; p1 &= 0xff; } if (p1 & 0x40) i = FX_FLAG_OFF; if (p1 & 0x80) i = FX_FLAG_ADD; p1 &= 0x3f; if (p1 < AWE_FX_END) { AWE_DEBUG(0,printk("AWE32: effects (%d) %d %d\n", voice, p1, p2)); if (i == FX_FLAG_SET) FX_SET(fx, p1, p2); else if (i == FX_FLAG_ADD) FX_ADD(fx, p1, p2); else FX_UNSET(fx, p1); if (i != FX_FLAG_OFF && parm_defs[p1].realtime) { AWE_DEBUG(0,printk("AWE32: fx_realtime (%d)\n", voice)); awe_voice_change(voice, parm_defs[p1].realtime); } } break; case _AWE_RESET_CHANNEL: awe_channel_init(voice, !p1); break; case _AWE_TERMINATE_ALL: awe_reset(0); break; case _AWE_TERMINATE_CHANNEL: awe_voice_change(voice, awe_terminate_and_init); break; case _AWE_RELEASE_ALL: awe_note_off_all(FALSE); break; case _AWE_NOTEOFF_ALL: awe_note_off_all(TRUE); break; case _AWE_INITIAL_VOLUME: AWE_DEBUG(0,printk("AWE32: init attenuation %d\n", p1)); if (p2 == 0) /* absolute value */ init_atten = (short)p1; else /* relative value */ init_atten = misc_modes[AWE_MD_ZERO_ATTEN] + (short)p1; if (init_atten < 0) init_atten = 0; for (i = 0; i < awe_max_voices; i++) awe_set_voice_vol(i, TRUE); break; case _AWE_CHN_PRESSURE: cinfo->chan_press = p1; p1 = p1 * misc_modes[AWE_MD_MOD_SENSE] / 1200; FX_ADD(&cinfo->fx, AWE_FX_LFO1_PITCH, p1); awe_voice_change(voice, awe_fx_fmmod); FX_ADD(&cinfo->fx, AWE_FX_LFO2_PITCH, p1); awe_voice_change(voice, awe_fx_fm2frq2); break; case _AWE_CHANNEL_MODE: AWE_DEBUG(0,printk("AWE32: channel mode = %d\n", p1)); playing_mode = p1; awe_reset(0); break; case _AWE_DRUM_CHANNELS: AWE_DEBUG(0,printk("AWE32: drum flags = %x\n", p1)); drum_flags = *(unsigned int*)&event[4]; break; case _AWE_MISC_MODE: AWE_DEBUG(0,printk("AWE32: misc mode = %d %d\n", p1, p2)); if (p1 > AWE_MD_VERSION && p1 < AWE_MD_END) misc_modes[p1] = p2; break; case _AWE_EQUALIZER: awe_equalizer((int)p1, (int)p2); break; default: AWE_DEBUG(0,printk("AWE32: hw control cmd=%d voice=%d\n", cmd, voice)); break; } } /* voice pressure change */ static void awe_aftertouch(int dev, int voice, int pressure) { int note; AWE_DEBUG(2,printk("AWE32: [after(%d) %d]\n", voice, pressure)); if (! voice_in_range(voice)) return; switch (playing_mode) { case AWE_PLAY_DIRECT: case AWE_PLAY_INDIRECT: awe_start_note(dev, voice, 255, pressure); break; case AWE_PLAY_MULTI2: note = (voice_alloc->map[voice] & 0xff) - 1; awe_start_note(dev, voice, note + 0x80, pressure); break; } } /* voice control change */ static void awe_controller(int dev, int voice, int ctrl_num, int value) { int i; awe_chan_info *cinfo; if (! voice_in_range(voice)) return; if (playing_mode == AWE_PLAY_MULTI2) { voice = voice_alloc->map[voice] >> 8; if (voice < 0 || voice >= AWE_MAX_CHANNELS) return; } cinfo = &channels[voice]; switch (ctrl_num) { case CTL_BANK_SELECT: /* MIDI control #0 */ AWE_DEBUG(2,printk("AWE32: [bank(%d) %d]\n", voice, value)); if (MULTI_LAYER_MODE() && IS_DRUM_CHANNEL(voice) && !misc_modes[AWE_MD_TOGGLE_DRUM_BANK]) break; cinfo->bank = value; if (cinfo->bank == AWE_DRUM_BANK) DRUM_CHANNEL_ON(cinfo->channel); else DRUM_CHANNEL_OFF(cinfo->channel); awe_set_instr(dev, voice, cinfo->instr); break; case CTL_MODWHEEL: /* MIDI control #1 */ AWE_DEBUG(2,printk("AWE32: [modwheel(%d) %d]\n", voice, value)); i = value * misc_modes[AWE_MD_MOD_SENSE] / 1200; FX_ADD(&cinfo->fx, AWE_FX_LFO1_PITCH, i); awe_voice_change(voice, awe_fx_fmmod); FX_ADD(&cinfo->fx, AWE_FX_LFO2_PITCH, i); awe_voice_change(voice, awe_fx_fm2frq2); break; case CTRL_PITCH_BENDER: /* SEQ1 V2 contorl */ AWE_DEBUG(2,printk("AWE32: [bend(%d) %d]\n", voice, value)); /* zero centered */ cinfo->bender = value; awe_voice_change(voice, awe_set_voice_pitch); break; case CTRL_PITCH_BENDER_RANGE: /* SEQ1 V2 control */ AWE_DEBUG(2,printk("AWE32: [range(%d) %d]\n", voice, value)); /* value = sense x 100 */ cinfo->bender_range = value; /* no audible pitch change yet.. */ break; case CTL_EXPRESSION: /* MIDI control #11 */ if (SINGLE_LAYER_MODE()) value /= 128; case CTRL_EXPRESSION: /* SEQ1 V2 control */ AWE_DEBUG(2,printk("AWE32: [expr(%d) %d]\n", voice, value)); /* 0 - 127 */ cinfo->expression_vol = value; awe_voice_change(voice, awe_set_voice_vol); break; case CTL_PAN: /* MIDI control #10 */ AWE_DEBUG(2,printk("AWE32: [pan(%d) %d]\n", voice, value)); /* (0-127) -> signed 8bit */ cinfo->panning = value * 2 - 128; if (misc_modes[AWE_MD_REALTIME_PAN]) awe_voice_change(voice, awe_set_pan); break; case CTL_MAIN_VOLUME: /* MIDI control #7 */ if (SINGLE_LAYER_MODE()) value = (value * 100) / 16383; case CTRL_MAIN_VOLUME: /* SEQ1 V2 control */ AWE_DEBUG(2,printk("AWE32: [mainvol(%d) %d]\n", voice, value)); /* 0 - 127 */ cinfo->main_vol = value; awe_voice_change(voice, awe_set_voice_vol); break; case CTL_EXT_EFF_DEPTH: /* reverb effects: 0-127 */ AWE_DEBUG(2,printk("AWE32: [reverb(%d) %d]\n", voice, value)); FX_SET(&cinfo->fx, AWE_FX_REVERB, value * 2); break; case CTL_CHORUS_DEPTH: /* chorus effects: 0-127 */ AWE_DEBUG(2,printk("AWE32: [chorus(%d) %d]\n", voice, value)); FX_SET(&cinfo->fx, AWE_FX_CHORUS, value * 2); break; #ifdef AWE_ACCEPT_ALL_SOUNDS_CONTROLL case 120: /* all sounds off */ awe_note_off_all(FALSE); break; case 123: /* all notes off */ awe_note_off_all(TRUE); break; #endif case CTL_SUSTAIN: /* MIDI control #64 */ cinfo->sustained = value; if (value != 127) awe_voice_change(voice, awe_sustain_off); break; case CTL_SOSTENUTO: /* MIDI control #66 */ if (value == 127) awe_voice_change(voice, awe_sostenuto_on); else awe_voice_change(voice, awe_sustain_off); break; default: AWE_DEBUG(0,printk("AWE32: [control(%d) ctrl=%d val=%d]\n", voice, ctrl_num, value)); break; } } /* voice pan change (value = -128 - 127) */ static void awe_panning(int dev, int voice, int value) { awe_chan_info *cinfo; if (! voice_in_range(voice)) return; if (playing_mode == AWE_PLAY_MULTI2) { voice = voice_alloc->map[voice] >> 8; if (voice < 0 || voice >= AWE_MAX_CHANNELS) return; } cinfo = &channels[voice]; cinfo->panning = value; AWE_DEBUG(2,printk("AWE32: [pan(%d) %d]\n", voice, cinfo->panning)); if (misc_modes[AWE_MD_REALTIME_PAN]) awe_voice_change(voice, awe_set_pan); } /* volume mode change */ static void awe_volume_method(int dev, int mode) { /* not impremented */ AWE_DEBUG(0,printk("AWE32: [volmethod mode=%d]\n", mode)); } #ifndef AWE_NO_PATCHMGR /* patch manager */ static int awe_patchmgr(int dev, struct patmgr_info *rec) { printk("AWE32 Warning: patch manager control not supported\n"); return 0; } #endif /* pitch wheel change: 0-16384 */ static void awe_bender(int dev, int voice, int value) { awe_chan_info *cinfo; if (! voice_in_range(voice)) return; if (playing_mode == AWE_PLAY_MULTI2) { voice = voice_alloc->map[voice] >> 8; if (voice < 0 || voice >= AWE_MAX_CHANNELS) return; } /* convert to zero centered value */ cinfo = &channels[voice]; cinfo->bender = value - 8192; AWE_DEBUG(2,printk("AWE32: [bend(%d) %d]\n", voice, cinfo->bender)); awe_voice_change(voice, awe_set_voice_pitch); } /*---------------------------------------------------------------- * load a sound patch: * three types of patches are accepted: AWE, GUS, and SYSEX. *----------------------------------------------------------------*/ static int awe_load_patch(int dev, int format, const char *addr, int offs, int count, int pmgr_flag) { awe_patch_info patch; int rc = 0; #ifdef AWE_HAS_GUS_COMPATIBILITY if (format == GUS_PATCH) { return awe_load_guspatch(addr, offs, count, pmgr_flag); } else #endif if (format == SYSEX_PATCH) { /* no system exclusive message supported yet */ return 0; } else if (format != AWE_PATCH) { printk("AWE32 Error: Invalid patch format (key) 0x%x\n", format); return RET_ERROR(EINVAL); } if (count < AWE_PATCH_INFO_SIZE) { printk("AWE32 Error: Patch header too short\n"); return RET_ERROR(EINVAL); } COPY_FROM_USER(((char*)&patch) + offs, addr, offs, AWE_PATCH_INFO_SIZE - offs); count -= AWE_PATCH_INFO_SIZE; if (count < patch.len) { printk("AWE32: sample: Patch record too short (%d<%d)\n", count, patch.len); return RET_ERROR(EINVAL); } switch (patch.type) { case AWE_LOAD_INFO: rc = awe_load_info(&patch, addr, count); break; case AWE_LOAD_DATA: rc = awe_load_data(&patch, addr, count); break; case AWE_OPEN_PATCH: rc = awe_open_patch(&patch, addr, count); break; case AWE_CLOSE_PATCH: rc = awe_close_patch(&patch, addr, count); break; case AWE_UNLOAD_PATCH: rc = awe_unload_patch(&patch, addr, count); break; case AWE_REPLACE_DATA: rc = awe_replace_data(&patch, addr, count); break; case AWE_MAP_PRESET: rc = awe_load_map(&patch, addr, count); break; case AWE_LOAD_CHORUS_FX: rc = awe_load_chorus_fx(&patch, addr, count); break; case AWE_LOAD_REVERB_FX: rc = awe_load_reverb_fx(&patch, addr, count); break; default: printk("AWE32 Error: unknown patch format type %d\n", patch.type); rc = RET_ERROR(EINVAL); } return rc; } /* create an sflist record */ static int awe_create_sf(int type, char *name) { sf_list *rec; /* terminate sounds */ awe_reset(0); if (current_sf_id >= max_sfs) { int newsize = max_sfs + AWE_MAX_SF_LISTS; sf_list *newlist = my_realloc(sflists, sizeof(sf_list)*max_sfs, sizeof(sf_list)*newsize); if (newlist == NULL) return 1; sflists = newlist; max_sfs = newsize; } rec = &sflists[current_sf_id]; rec->sf_id = current_sf_id + 1; rec->type = type; if (current_sf_id == 0 || (type & AWE_PAT_LOCKED) != 0) locked_sf_id = current_sf_id + 1; /* if (name) MEMCPY(rec->name, name, AWE_PATCH_NAME_LEN); else BZERO(rec->name, AWE_PATCH_NAME_LEN); */ rec->num_info = awe_free_info(); rec->num_sample = awe_free_sample(); rec->mem_ptr = awe_free_mem_ptr(); rec->infos = -1; rec->samples = -1; current_sf_id++; return 0; } /* open patch; create sf list and set opened flag */ static int awe_open_patch(awe_patch_info *patch, const char *addr, int count) { awe_open_parm parm; COPY_FROM_USER(&parm, addr, AWE_PATCH_INFO_SIZE, sizeof(parm)); if (awe_create_sf(parm.type, parm.name)) { printk("AWE32: can't open: failed to alloc new list\n"); return RET_ERROR(ENOSPC); } patch_opened = TRUE; return current_sf_id; } /* check if the patch is already opened */ static int check_patch_opened(int type, char *name) { if (! patch_opened) { if (awe_create_sf(type, name)) { printk("AWE32: failed to alloc new list\n"); return RET_ERROR(ENOSPC); } patch_opened = TRUE; return current_sf_id; } return current_sf_id; } /* close the patch; if no voice is loaded, remove the patch */ static int awe_close_patch(awe_patch_info *patch, const char *addr, int count) { if (patch_opened && current_sf_id > 0) { /* if no voice is loaded, release the current patch */ if (sflists[current_sf_id-1].infos == -1) awe_remove_samples(current_sf_id - 1); } patch_opened = 0; return 0; } /* remove the latest patch */ static int awe_unload_patch(awe_patch_info *patch, const char *addr, int count) { if (current_sf_id > 0) awe_remove_samples(current_sf_id - 1); return 0; } /* allocate voice info list records */ static int alloc_new_info(int nvoices) { int newsize, free_info; awe_voice_list *newlist; free_info = awe_free_info(); if (free_info + nvoices >= max_infos) { do { newsize = max_infos + AWE_MAX_INFOS; } while (free_info + nvoices >= newsize); newlist = my_realloc(infos, sizeof(awe_voice_list)*max_infos, sizeof(awe_voice_list)*newsize); if (newlist == NULL) { printk("AWE32: can't alloc info table\n"); return RET_ERROR(ENOSPC); } infos = newlist; max_infos = newsize; } return 0; } /* allocate sample info list records */ static int alloc_new_sample(void) { int newsize, free_sample; awe_sample_list *newlist; free_sample = awe_free_sample(); if (free_sample >= max_samples) { newsize = max_samples + AWE_MAX_SAMPLES; newlist = my_realloc(samples, sizeof(awe_sample_list)*max_samples, sizeof(awe_sample_list)*newsize); if (newlist == NULL) { printk("AWE32: can't alloc sample table\n"); return RET_ERROR(ENOSPC); } samples = newlist; max_samples = newsize; } return 0; } /* load voice map */ static int awe_load_map(awe_patch_info *patch, const char *addr, int count) { awe_voice_map map; awe_voice_list *rec; int free_info; if (check_patch_opened(AWE_PAT_TYPE_MAP, NULL) < 0) return RET_ERROR(ENOSPC); if (alloc_new_info(1) < 0) return RET_ERROR(ENOSPC); COPY_FROM_USER(&map, addr, AWE_PATCH_INFO_SIZE, sizeof(map)); free_info = awe_free_info(); rec = &infos[free_info]; rec->bank = map.map_bank; rec->instr = map.map_instr; rec->type = V_ST_MAPPED; rec->disabled = FALSE; awe_init_voice_info(&rec->v); if (map.map_key >= 0) { rec->v.low = map.map_key; rec->v.high = map.map_key; } rec->v.start = map.src_instr; rec->v.end = map.src_bank; rec->v.fixkey = map.src_key; rec->v.sf_id = current_sf_id; add_info_list(free_info); add_sf_info(free_info); return 0; } /* load voice information data */ static int awe_load_info(awe_patch_info *patch, const char *addr, int count) { int offset; awe_voice_rec_hdr hdr; int i; int total_size; if (count < AWE_VOICE_REC_SIZE) { printk("AWE32 Error: invalid patch info length\n"); return RET_ERROR(EINVAL); } offset = AWE_PATCH_INFO_SIZE; COPY_FROM_USER((char*)&hdr, addr, offset, AWE_VOICE_REC_SIZE); offset += AWE_VOICE_REC_SIZE; if (hdr.nvoices <= 0 || hdr.nvoices >= 100) { printk("AWE32 Error: Illegal voice number %d\n", hdr.nvoices); return RET_ERROR(EINVAL); } total_size = AWE_VOICE_REC_SIZE + AWE_VOICE_INFO_SIZE * hdr.nvoices; if (count < total_size) { printk("AWE32 Error: patch length(%d) is smaller than nvoices(%d)\n", count, hdr.nvoices); return RET_ERROR(EINVAL); } if (check_patch_opened(AWE_PAT_TYPE_MISC, NULL) < 0) return RET_ERROR(ENOSPC); #if 0 /* it looks like not so useful.. */ /* check if the same preset already exists in the info list */ for (i = sflists[current_sf_id-1].infos; i >= 0; i = infos[i].next) { if (infos[i].disabled) continue; if (infos[i].bank == hdr.bank && infos[i].instr == hdr.instr) { /* in exclusive mode, do skip loading this */ if (hdr.write_mode == AWE_WR_EXCLUSIVE) return 0; /* in replace mode, disable the old data */ else if (hdr.write_mode == AWE_WR_REPLACE) infos[i].disabled = TRUE; } } if (hdr.write_mode == AWE_WR_REPLACE) rebuild_preset_list(); #endif if (alloc_new_info(hdr.nvoices) < 0) return RET_ERROR(ENOSPC); for (i = 0; i < hdr.nvoices; i++) { int rec = awe_free_info(); infos[rec].bank = hdr.bank; infos[rec].instr = hdr.instr; infos[rec].type = V_ST_NORMAL; infos[rec].disabled = FALSE; /* copy awe_voice_info parameters */ COPY_FROM_USER(&infos[rec].v, addr, offset, AWE_VOICE_INFO_SIZE); offset += AWE_VOICE_INFO_SIZE; infos[rec].v.sf_id = current_sf_id; if (infos[rec].v.mode & AWE_MODE_INIT_PARM) awe_init_voice_parm(&infos[rec].v.parm); awe_set_sample(&infos[rec].v); add_info_list(rec); add_sf_info(rec); } return 0; } /* load wave sample data */ static int awe_load_data(awe_patch_info *patch, const char *addr, int count) { int offset, size; int rc, free_sample; awe_sample_info *rec; if (check_patch_opened(AWE_PAT_TYPE_MISC, NULL) < 0) return RET_ERROR(ENOSPC); if (alloc_new_sample() < 0) return RET_ERROR(ENOSPC); free_sample = awe_free_sample(); rec = &samples[free_sample].v; size = (count - AWE_SAMPLE_INFO_SIZE) / 2; offset = AWE_PATCH_INFO_SIZE; COPY_FROM_USER(rec, addr, offset, AWE_SAMPLE_INFO_SIZE); offset += AWE_SAMPLE_INFO_SIZE; if (size != rec->size) { printk("AWE32: load: sample size differed (%d != %d)\n", rec->size, size); return RET_ERROR(EINVAL); } if (rec->size > 0) if ((rc = awe_write_wave_data(addr, offset, rec, -1)) != 0) return rc; rec->sf_id = current_sf_id; add_sf_sample(free_sample); return 0; } /* replace wave sample data */ static int awe_replace_data(awe_patch_info *patch, const char *addr, int count) { int offset; int size; int rc, i; int channels; awe_sample_info cursmp; int save_mem_ptr; if (! patch_opened) { printk("AWE32: replace: patch not opened\n"); return RET_ERROR(EINVAL); } size = (count - AWE_SAMPLE_INFO_SIZE) / 2; offset = AWE_PATCH_INFO_SIZE; COPY_FROM_USER(&cursmp, addr, offset, AWE_SAMPLE_INFO_SIZE); offset += AWE_SAMPLE_INFO_SIZE; if (cursmp.size == 0 || size != cursmp.size) { printk("AWE32: replace: illegal sample size (%d!=%d)\n", cursmp.size, size); return RET_ERROR(EINVAL); } channels = patch->optarg; if (channels <= 0 || channels > AWE_NORMAL_VOICES) { printk("AWE32: replace: illegal channels %d\n", channels); return RET_ERROR(EINVAL); } for (i = sflists[current_sf_id-1].samples; i >= 0; i = samples[i].next) { if (samples[i].v.sample == cursmp.sample) break; } if (i < 0) { printk("AWE32: replace: cannot find existing sample data %d\n", cursmp.sample); return RET_ERROR(EINVAL); } if (samples[i].v.size != cursmp.size) { printk("AWE32: replace: exiting size differed (%d!=%d)\n", samples[i].v.size, cursmp.size); return RET_ERROR(EINVAL); } save_mem_ptr = awe_free_mem_ptr(); sflists[current_sf_id-1].mem_ptr = samples[i].v.start - awe_mem_start; MEMCPY(&samples[i].v, &cursmp, sizeof(cursmp)); if ((rc = awe_write_wave_data(addr, offset, &samples[i].v, channels)) != 0) return rc; sflists[current_sf_id-1].mem_ptr = save_mem_ptr; samples[i].v.sf_id = current_sf_id; return 0; } /*----------------------------------------------------------------*/ static const char *readbuf_addr; static int readbuf_offs; static int readbuf_flags; #if defined(__DragonFly__) || defined(__FreeBSD__) static unsigned short *readbuf_loop; static int readbuf_loopstart, readbuf_loopend; #endif /* initialize read buffer */ static int readbuf_init(const char *addr, int offset, awe_sample_info *sp) { #if defined(__DragonFly__) || defined(__FreeBSD__) readbuf_loop = NULL; readbuf_loopstart = sp->loopstart; readbuf_loopend = sp->loopend; if (sp->mode_flags & (AWE_SAMPLE_BIDIR_LOOP|AWE_SAMPLE_REVERSE_LOOP)) { int looplen = sp->loopend - sp->loopstart; readbuf_loop = my_malloc(looplen * 2); if (readbuf_loop == NULL) { printk("AWE32: can't malloc temp buffer\n"); return RET_ERROR(ENOSPC); } } #endif readbuf_addr = addr; readbuf_offs = offset; readbuf_flags = sp->mode_flags; return 0; } /* read directly from user buffer */ static unsigned short readbuf_word(int pos) { unsigned short c; /* read from user buffer */ if (readbuf_flags & AWE_SAMPLE_8BITS) { unsigned char cc; GET_BYTE_FROM_USER(cc, readbuf_addr, readbuf_offs + pos); c = cc << 8; /* convert 8bit -> 16bit */ } else { GET_SHORT_FROM_USER(c, readbuf_addr, readbuf_offs + pos * 2); } if (readbuf_flags & AWE_SAMPLE_UNSIGNED) c ^= 0x8000; /* unsigned -> signed */ #if defined(__DragonFly__) || defined(__FreeBSD__) /* write on cache for reverse loop */ if (readbuf_flags & (AWE_SAMPLE_BIDIR_LOOP|AWE_SAMPLE_REVERSE_LOOP)) { if (pos >= readbuf_loopstart && pos < readbuf_loopend) readbuf_loop[pos - readbuf_loopstart] = c; } #endif return c; } #if defined(__DragonFly__) || defined(__FreeBSD__) /* read from cache */ static unsigned short readbuf_word_cache(int pos) { if (pos >= readbuf_loopstart && pos < readbuf_loopend) return readbuf_loop[pos - readbuf_loopstart]; return 0; } static void readbuf_end(void) { if (readbuf_loop) { my_free(readbuf_loop); } readbuf_loop = NULL; } #else #define readbuf_word_cache readbuf_word #define readbuf_end() /**/ #endif /*----------------------------------------------------------------*/ #define BLANK_LOOP_START 8 #define BLANK_LOOP_END 40 #define BLANK_LOOP_SIZE 48 /* loading onto memory */ static int awe_write_wave_data(const char *addr, int offset, awe_sample_info *sp, int channels) { int i, truesize, dram_offset; int rc; /* be sure loop points start < end */ if (sp->loopstart > sp->loopend) { int tmp = sp->loopstart; sp->loopstart = sp->loopend; sp->loopend = tmp; } /* compute true data size to be loaded */ truesize = sp->size; if (sp->mode_flags & AWE_SAMPLE_BIDIR_LOOP) truesize += sp->loopend - sp->loopstart; if (sp->mode_flags & AWE_SAMPLE_NO_BLANK) truesize += BLANK_LOOP_SIZE; if (awe_free_mem_ptr() + truesize >= awe_mem_size/2) { printk("AWE32 Error: Sample memory full\n"); return RET_ERROR(ENOSPC); } /* recalculate address offset */ sp->end -= sp->start; sp->loopstart -= sp->start; sp->loopend -= sp->start; dram_offset = awe_free_mem_ptr() + awe_mem_start; sp->start = dram_offset; sp->end += dram_offset; sp->loopstart += dram_offset; sp->loopend += dram_offset; /* set the total size (store onto obsolete checksum value) */ if (sp->size == 0) sp->checksum = 0; else sp->checksum = truesize; if ((rc = awe_open_dram_for_write(dram_offset, channels)) != 0) return rc; if (readbuf_init(addr, offset, sp) < 0) return RET_ERROR(ENOSPC); for (i = 0; i < sp->size; i++) { unsigned short c; c = readbuf_word(i); awe_write_dram(c); if (i == sp->loopend && (sp->mode_flags & (AWE_SAMPLE_BIDIR_LOOP|AWE_SAMPLE_REVERSE_LOOP))) { int looplen = sp->loopend - sp->loopstart; /* copy reverse loop */ int k; for (k = 1; k <= looplen; k++) { c = readbuf_word_cache(i - k); awe_write_dram(c); } if (sp->mode_flags & AWE_SAMPLE_BIDIR_LOOP) { sp->end += looplen; } else { sp->start += looplen; sp->end += looplen; } } } readbuf_end(); /* if no blank loop is attached in the sample, add it */ if (sp->mode_flags & AWE_SAMPLE_NO_BLANK) { for (i = 0; i < BLANK_LOOP_SIZE; i++) awe_write_dram(0); if (sp->mode_flags & AWE_SAMPLE_SINGLESHOT) { sp->loopstart = sp->end + BLANK_LOOP_START; sp->loopend = sp->end + BLANK_LOOP_END; } } sflists[current_sf_id-1].mem_ptr += truesize; awe_close_dram(); /* initialize FM */ awe_init_fm(); return 0; } /*----------------------------------------------------------------*/ #ifdef AWE_HAS_GUS_COMPATIBILITY /* calculate GUS envelope time: * is this correct? i have no idea.. */ static int calc_gus_envelope_time(int rate, int start, int end) { int r, p, t; r = (3 - ((rate >> 6) & 3)) * 3; p = rate & 0x3f; t = end - start; if (t < 0) t = -t; if (13 > r) t = t << (13 - r); else t = t >> (r - 13); return (t * 10) / (p * 441); } #define calc_gus_sustain(val) (0x7f - vol_table[(val)/2]) #define calc_gus_attenuation(val) vol_table[(val)/2] /* load GUS patch */ static int awe_load_guspatch(const char *addr, int offs, int size, int pmgr_flag) { struct patch_info patch; awe_voice_info *rec; awe_sample_info *smp; int sizeof_patch; int note, free_sample, free_info; int rc; sizeof_patch = offsetof(struct patch_info, data); /* header size */ if (size < sizeof_patch) { printk("AWE32 Error: Patch header too short\n"); return RET_ERROR(EINVAL); } COPY_FROM_USER(((char*)&patch) + offs, addr, offs, sizeof_patch - offs); size -= sizeof_patch; if (size < patch.len) { printk("AWE32 Warning: Patch record too short (%d<%ld)\n", size, patch.len); return RET_ERROR(EINVAL); } if (check_patch_opened(AWE_PAT_TYPE_GUS, NULL) < 0) return RET_ERROR(ENOSPC); if (alloc_new_sample() < 0) return RET_ERROR(ENOSPC); if (alloc_new_info(1)) return RET_ERROR(ENOSPC); free_sample = awe_free_sample(); smp = &samples[free_sample].v; smp->sample = free_sample; smp->start = 0; smp->end = patch.len; smp->loopstart = patch.loop_start; smp->loopend = patch.loop_end; smp->size = patch.len; /* set up mode flags */ smp->mode_flags = 0; if (!(patch.mode & WAVE_16_BITS)) smp->mode_flags |= AWE_SAMPLE_8BITS; if (patch.mode & WAVE_UNSIGNED) smp->mode_flags |= AWE_SAMPLE_UNSIGNED; smp->mode_flags |= AWE_SAMPLE_NO_BLANK; if (!(patch.mode & (WAVE_LOOPING|WAVE_BIDIR_LOOP|WAVE_LOOP_BACK))) smp->mode_flags |= AWE_SAMPLE_SINGLESHOT; if (patch.mode & WAVE_BIDIR_LOOP) smp->mode_flags |= AWE_SAMPLE_BIDIR_LOOP; if (patch.mode & WAVE_LOOP_BACK) smp->mode_flags |= AWE_SAMPLE_REVERSE_LOOP; AWE_DEBUG(0,printk("AWE32: [sample %d mode %x]\n", patch.instr_no, smp->mode_flags)); if (patch.mode & WAVE_16_BITS) { /* convert to word offsets */ smp->size /= 2; smp->end /= 2; smp->loopstart /= 2; smp->loopend /= 2; } smp->checksum_flag = 0; smp->checksum = 0; if ((rc = awe_write_wave_data(addr, sizeof_patch, smp, -1)) != 0) return rc; smp->sf_id = current_sf_id; add_sf_sample(free_sample); /* set up voice info */ free_info = awe_free_info(); rec = &infos[free_info].v; awe_init_voice_info(rec); rec->sample = free_sample; /* the last sample */ rec->rate_offset = calc_rate_offset(patch.base_freq); note = freq_to_note(patch.base_note); rec->root = note / 100; rec->tune = -(note % 100); rec->low = freq_to_note(patch.low_note) / 100; rec->high = freq_to_note(patch.high_note) / 100; AWE_DEBUG(1,printk("AWE32: [gus base offset=%d, note=%d, range=%d-%d(%lu-%lu)]\n", rec->rate_offset, note, rec->low, rec->high, patch.low_note, patch.high_note)); /* panning position; -128 - 127 => 0-127 */ rec->pan = (patch.panning + 128) / 2; /* detuning is ignored */ /* 6points volume envelope */ if (patch.mode & WAVE_ENVELOPES) { int attack, hold, decay, release; attack = calc_gus_envelope_time (patch.env_rate[0], 0, patch.env_offset[0]); hold = calc_gus_envelope_time (patch.env_rate[1], patch.env_offset[0], patch.env_offset[1]); decay = calc_gus_envelope_time (patch.env_rate[2], patch.env_offset[1], patch.env_offset[2]); release = calc_gus_envelope_time (patch.env_rate[3], patch.env_offset[1], patch.env_offset[4]); release += calc_gus_envelope_time (patch.env_rate[4], patch.env_offset[3], patch.env_offset[4]); release += calc_gus_envelope_time (patch.env_rate[5], patch.env_offset[4], patch.env_offset[5]); rec->parm.volatkhld = (calc_parm_attack(attack) << 8) | calc_parm_hold(hold); rec->parm.voldcysus = (calc_gus_sustain(patch.env_offset[2]) << 8) | calc_parm_decay(decay); rec->parm.volrelease = 0x8000 | calc_parm_decay(release); AWE_DEBUG(2,printk("AWE32: [gusenv atk=%d, hld=%d, dcy=%d, rel=%d]\n", attack, hold, decay, release)); rec->attenuation = calc_gus_attenuation(patch.env_offset[0]); } /* tremolo effect */ if (patch.mode & WAVE_TREMOLO) { int rate = (patch.tremolo_rate * 1000 / 38) / 42; rec->parm.tremfrq = ((patch.tremolo_depth / 2) << 8) | rate; AWE_DEBUG(2,printk("AWE32: [gusenv tremolo rate=%d, dep=%d, tremfrq=%x]\n", patch.tremolo_rate, patch.tremolo_depth, rec->parm.tremfrq)); } /* vibrato effect */ if (patch.mode & WAVE_VIBRATO) { int rate = (patch.vibrato_rate * 1000 / 38) / 42; rec->parm.fm2frq2 = ((patch.vibrato_depth / 6) << 8) | rate; AWE_DEBUG(2,printk("AWE32: [gusenv vibrato rate=%d, dep=%d, tremfrq=%x]\n", patch.tremolo_rate, patch.tremolo_depth, rec->parm.tremfrq)); } /* scale_freq, scale_factor, volume, and fractions not implemented */ /* append to the tail of the list */ infos[free_info].bank = misc_modes[AWE_MD_GUS_BANK]; infos[free_info].instr = patch.instr_no; infos[free_info].disabled = FALSE; infos[free_info].type = V_ST_NORMAL; infos[free_info].v.sf_id = current_sf_id; add_info_list(free_info); add_sf_info(free_info); /* set the voice index */ awe_set_sample(rec); return 0; } #endif /* AWE_HAS_GUS_COMPATIBILITY */ /*---------------------------------------------------------------- * sample and voice list handlers *----------------------------------------------------------------*/ /* append this to the sf list */ static void add_sf_info(int rec) { int sf_id = infos[rec].v.sf_id; if (sf_id == 0) return; sf_id--; if (sflists[sf_id].infos < 0) sflists[sf_id].infos = rec; else { int i, prev; prev = sflists[sf_id].infos; while ((i = infos[prev].next) >= 0) prev = i; infos[prev].next = rec; } infos[rec].next = -1; sflists[sf_id].num_info++; } /* prepend this sample to sf list */ static void add_sf_sample(int rec) { int sf_id = samples[rec].v.sf_id; if (sf_id == 0) return; sf_id--; samples[rec].next = sflists[sf_id].samples; sflists[sf_id].samples = rec; sflists[sf_id].num_sample++; } /* purge the old records which don't belong with the same file id */ static void purge_old_list(int rec, int next) { infos[rec].next_instr = next; if (infos[rec].bank == AWE_DRUM_BANK) { /* remove samples with the same note range */ int cur, *prevp = &infos[rec].next_instr; int low = infos[rec].v.low; int high = infos[rec].v.high; for (cur = next; cur >= 0; cur = infos[cur].next_instr) { if (infos[cur].v.low == low && infos[cur].v.high == high && infos[cur].v.sf_id != infos[rec].v.sf_id) *prevp = infos[cur].next_instr; prevp = &infos[cur].next_instr; } } else { if (infos[next].v.sf_id != infos[rec].v.sf_id) infos[rec].next_instr = -1; } } /* prepend to top of the preset table */ static void add_info_list(int rec) { int *prevp, cur; int instr = infos[rec].instr; int bank = infos[rec].bank; if (infos[rec].disabled) return; prevp = &preset_table[instr]; cur = *prevp; while (cur >= 0) { /* search the first record with the same bank number */ if (infos[cur].bank == bank) { /* replace the list with the new record */ infos[rec].next_bank = infos[cur].next_bank; *prevp = rec; purge_old_list(rec, cur); return; } prevp = &infos[cur].next_bank; cur = infos[cur].next_bank; } /* this is the first bank record.. just add this */ infos[rec].next_instr = -1; infos[rec].next_bank = preset_table[instr]; preset_table[instr] = rec; } /* remove samples later than the specified sf_id */ static void awe_remove_samples(int sf_id) { if (sf_id <= 0) { awe_reset_samples(); return; } /* already removed? */ if (current_sf_id <= sf_id) return; current_sf_id = sf_id; if (locked_sf_id > sf_id) locked_sf_id = sf_id; rebuild_preset_list(); } /* rebuild preset search list */ static void rebuild_preset_list(void) { int i, j; for (i = 0; i < AWE_MAX_PRESETS; i++) preset_table[i] = -1; for (i = 0; i < current_sf_id; i++) { for (j = sflists[i].infos; j >= 0; j = infos[j].next) add_info_list(j); } } /* search the specified sample */ static short awe_set_sample(awe_voice_info *vp) { int i; vp->index = -1; for (i = sflists[vp->sf_id-1].samples; i >= 0; i = samples[i].next) { if (samples[i].v.sample == vp->sample) { /* set the actual sample offsets */ vp->start += samples[i].v.start; vp->end += samples[i].v.end; vp->loopstart += samples[i].v.loopstart; vp->loopend += samples[i].v.loopend; /* copy mode flags */ vp->mode = samples[i].v.mode_flags; /* set index */ vp->index = i; return i; } } return -1; } /*---------------------------------------------------------------- * voice allocation *----------------------------------------------------------------*/ /* look for all voices associated with the specified note & velocity */ static int awe_search_multi_voices(int rec, int note, int velocity, awe_voice_info **vlist) { int nvoices; nvoices = 0; for (; rec >= 0; rec = infos[rec].next_instr) { if (note >= infos[rec].v.low && note <= infos[rec].v.high && velocity >= infos[rec].v.vellow && velocity <= infos[rec].v.velhigh) { vlist[nvoices] = &infos[rec].v; if (infos[rec].type == V_ST_MAPPED) /* mapper */ return -1; nvoices++; if (nvoices >= AWE_MAX_VOICES) break; } } return nvoices; } /* store the voice list from the specified note and velocity. if the preset is mapped, seek for the destination preset, and rewrite the note number if necessary. */ static int really_alloc_voices(int vrec, int def_vrec, int *note, int velocity, awe_voice_info **vlist, int level) { int nvoices; nvoices = awe_search_multi_voices(vrec, *note, velocity, vlist); if (nvoices == 0) nvoices = awe_search_multi_voices(def_vrec, *note, velocity, vlist); if (nvoices < 0) { /* mapping */ int preset = vlist[0]->start; int bank = vlist[0]->end; int key = vlist[0]->fixkey; if (level > 5) { printk("AWE32: too deep mapping level\n"); return 0; } vrec = awe_search_instr(bank, preset); if (bank == AWE_DRUM_BANK) def_vrec = awe_search_instr(bank, 0); else def_vrec = awe_search_instr(0, preset); if (key >= 0) *note = key; return really_alloc_voices(vrec, def_vrec, note, velocity, vlist, level+1); } return nvoices; } /* allocate voices corresponding note and velocity; supports multiple insts. */ static void awe_alloc_multi_voices(int ch, int note, int velocity, int key) { int i, v, nvoices; awe_voice_info *vlist[AWE_MAX_VOICES]; if (channels[ch].vrec < 0 && channels[ch].def_vrec < 0) awe_set_instr(0, ch, channels[ch].instr); /* check the possible voices; note may be changeable if mapped */ nvoices = really_alloc_voices(channels[ch].vrec, channels[ch].def_vrec, ¬e, velocity, vlist, 0); /* set the voices */ current_alloc_time++; for (i = 0; i < nvoices; i++) { v = awe_clear_voice(); voices[v].key = key; voices[v].ch = ch; voices[v].note = note; voices[v].velocity = velocity; voices[v].time = current_alloc_time; voices[v].cinfo = &channels[ch]; voices[v].sample = vlist[i]; voices[v].state = AWE_ST_MARK; voices[v].layer = nvoices - i - 1; /* in reverse order */ } /* clear the mark in allocated voices */ for (i = 0; i < awe_max_voices; i++) { if (voices[i].state == AWE_ST_MARK) voices[i].state = AWE_ST_OFF; } } /* search the best voice from the specified status condition */ static int search_best_voice(int condition) { int i, time, best; best = -1; time = current_alloc_time + 1; for (i = 0; i < awe_max_voices; i++) { if ((voices[i].state & condition) && (best < 0 || voices[i].time < time)) { best = i; time = voices[i].time; } } /* clear voice */ if (best >= 0) { if (voices[best].state != AWE_ST_OFF) awe_terminate(best); awe_voice_init(best, TRUE); } return best; } /* search an empty voice. if no empty voice is found, at least terminate a voice */ static int awe_clear_voice(void) { int best; /* looking for the oldest empty voice */ if ((best = search_best_voice(AWE_ST_OFF)) >= 0) return best; if ((best = search_best_voice(AWE_ST_RELEASED)) >= 0) return best; /* looking for the oldest sustained voice */ if ((best = search_best_voice(AWE_ST_SUSTAINED)) >= 0) return best; #ifdef AWE_LOOKUP_MIDI_PRIORITY if (MULTI_LAYER_MODE() && misc_modes[AWE_MD_CHN_PRIOR]) { int ch = -1; int time = current_alloc_time + 1; int i; /* looking for the voices from high channel (except drum ch) */ for (i = 0; i < awe_max_voices; i++) { if (IS_DRUM_CHANNEL(voices[i].ch)) continue; if (voices[i].ch < ch) continue; if (voices[i].state != AWE_ST_MARK && (voices[i].ch > ch || voices[i].time < time)) { best = i; time = voices[i].time; ch = voices[i].ch; } } } #endif if (best < 0) best = search_best_voice(~AWE_ST_MARK); if (best >= 0) return best; return 0; } /* search sample for the specified note & velocity and set it on the voice; * note that voice is the voice index (not channel index) */ static void awe_alloc_one_voice(int voice, int note, int velocity) { int ch, nvoices; awe_voice_info *vlist[AWE_MAX_VOICES]; ch = voices[voice].ch; if (channels[ch].vrec < 0 && channels[ch].def_vrec < 0) awe_set_instr(0, ch, channels[ch].instr); nvoices = really_alloc_voices(voices[voice].cinfo->vrec, voices[voice].cinfo->def_vrec, ¬e, velocity, vlist, 0); if (nvoices > 0) { voices[voice].time = ++current_alloc_time; voices[voice].sample = vlist[0]; /* use the first one */ voices[voice].layer = 0; voices[voice].note = note; voices[voice].velocity = velocity; } } /*---------------------------------------------------------------- * sequencer2 functions *----------------------------------------------------------------*/ /* search an empty voice; used by sequencer2 */ static int awe_alloc(int dev, int chn, int note, struct voice_alloc_info *alloc) { playing_mode = AWE_PLAY_MULTI2; awe_info.nr_voices = AWE_MAX_CHANNELS; return awe_clear_voice(); } /* set up voice; used by sequencer2 */ static void awe_setup_voice(int dev, int voice, int chn) { struct channel_info *info; if (synth_devs[dev] == NULL || (info = &synth_devs[dev]->chn_info[chn]) == NULL) return; if (voice < 0 || voice >= awe_max_voices) return; AWE_DEBUG(2,printk("AWE32: [setup(%d) ch=%d]\n", voice, chn)); channels[chn].expression_vol = info->controllers[CTL_EXPRESSION]; channels[chn].main_vol = info->controllers[CTL_MAIN_VOLUME]; channels[chn].panning = info->controllers[CTL_PAN] * 2 - 128; /* signed 8bit */ channels[chn].bender = info->bender_value; /* zero center */ channels[chn].bank = info->controllers[CTL_BANK_SELECT]; channels[chn].sustained = info->controllers[CTL_SUSTAIN]; if (info->controllers[CTL_EXT_EFF_DEPTH]) { FX_SET(&channels[chn].fx, AWE_FX_REVERB, info->controllers[CTL_EXT_EFF_DEPTH] * 2); } if (info->controllers[CTL_CHORUS_DEPTH]) { FX_SET(&channels[chn].fx, AWE_FX_CHORUS, info->controllers[CTL_CHORUS_DEPTH] * 2); } awe_set_instr(dev, chn, info->pgm_num); } #ifdef CONFIG_AWE32_MIXER /*================================================================ * AWE32 mixer device control *================================================================*/ static int awe_mixer_ioctl(int dev, unsigned int cmd, caddr_t arg) { int i, level; if (((cmd >> 8) & 0xff) != 'M') return RET_ERROR(EINVAL); level = (int)IOCTL_IN(arg); level = ((level & 0xff) + (level >> 8)) / 2; AWE_DEBUG(0,printk("AWEMix: cmd=%x val=%d\n", cmd & 0xff, level)); if (IO_WRITE_CHECK(cmd)) { switch (cmd & 0xff) { case SOUND_MIXER_BASS: awe_bass_level = level * 12 / 100; if (awe_bass_level >= 12) awe_bass_level = 11; awe_equalizer(awe_bass_level, awe_treble_level); break; case SOUND_MIXER_TREBLE: awe_treble_level = level * 12 / 100; if (awe_treble_level >= 12) awe_treble_level = 11; awe_equalizer(awe_bass_level, awe_treble_level); break; case SOUND_MIXER_VOLUME: level = level * 127 / 100; if (level >= 128) level = 127; init_atten = vol_table[level]; for (i = 0; i < awe_max_voices; i++) awe_set_voice_vol(i, TRUE); break; } } switch (cmd & 0xff) { case SOUND_MIXER_BASS: level = awe_bass_level * 100 / 24; level = (level << 8) | level; break; case SOUND_MIXER_TREBLE: level = awe_treble_level * 100 / 24; level = (level << 8) | level; break; case SOUND_MIXER_VOLUME: for (i = 127; i > 0; i--) { if (init_atten <= vol_table[i]) break; } level = i * 100 / 127; level = (level << 8) | level; break; case SOUND_MIXER_DEVMASK: level = SOUND_MASK_BASS|SOUND_MASK_TREBLE|SOUND_MASK_VOLUME; break; default: level = 0; break; } return IOCTL_OUT(arg, level); } #endif /* CONFIG_AWE32_MIXER */ /*================================================================ * initialization of AWE32 *================================================================*/ /* intiailize audio channels */ static void awe_init_audio(void) { int ch; /* turn off envelope engines */ for (ch = 0; ch < AWE_MAX_VOICES; ch++) { awe_poke(AWE_DCYSUSV(ch), 0x80); } /* reset all other parameters to zero */ for (ch = 0; ch < AWE_MAX_VOICES; ch++) { awe_poke(AWE_ENVVOL(ch), 0); awe_poke(AWE_ENVVAL(ch), 0); awe_poke(AWE_DCYSUS(ch), 0); awe_poke(AWE_ATKHLDV(ch), 0); awe_poke(AWE_LFO1VAL(ch), 0); awe_poke(AWE_ATKHLD(ch), 0); awe_poke(AWE_LFO2VAL(ch), 0); awe_poke(AWE_IP(ch), 0); awe_poke(AWE_IFATN(ch), 0); awe_poke(AWE_PEFE(ch), 0); awe_poke(AWE_FMMOD(ch), 0); awe_poke(AWE_TREMFRQ(ch), 0); awe_poke(AWE_FM2FRQ2(ch), 0); awe_poke_dw(AWE_PTRX(ch), 0); awe_poke_dw(AWE_VTFT(ch), 0); awe_poke_dw(AWE_PSST(ch), 0); awe_poke_dw(AWE_CSL(ch), 0); awe_poke_dw(AWE_CCCA(ch), 0); } for (ch = 0; ch < AWE_MAX_VOICES; ch++) { awe_poke_dw(AWE_CPF(ch), 0); awe_poke_dw(AWE_CVCF(ch), 0); } } /* initialize DMA address */ static void awe_init_dma(void) { awe_poke_dw(AWE_SMALR, 0); awe_poke_dw(AWE_SMARR, 0); awe_poke_dw(AWE_SMALW, 0); awe_poke_dw(AWE_SMARW, 0); } /* initialization arrays; from ADIP */ static unsigned short init1[128] = { 0x03ff, 0x0030, 0x07ff, 0x0130, 0x0bff, 0x0230, 0x0fff, 0x0330, 0x13ff, 0x0430, 0x17ff, 0x0530, 0x1bff, 0x0630, 0x1fff, 0x0730, 0x23ff, 0x0830, 0x27ff, 0x0930, 0x2bff, 0x0a30, 0x2fff, 0x0b30, 0x33ff, 0x0c30, 0x37ff, 0x0d30, 0x3bff, 0x0e30, 0x3fff, 0x0f30, 0x43ff, 0x0030, 0x47ff, 0x0130, 0x4bff, 0x0230, 0x4fff, 0x0330, 0x53ff, 0x0430, 0x57ff, 0x0530, 0x5bff, 0x0630, 0x5fff, 0x0730, 0x63ff, 0x0830, 0x67ff, 0x0930, 0x6bff, 0x0a30, 0x6fff, 0x0b30, 0x73ff, 0x0c30, 0x77ff, 0x0d30, 0x7bff, 0x0e30, 0x7fff, 0x0f30, 0x83ff, 0x0030, 0x87ff, 0x0130, 0x8bff, 0x0230, 0x8fff, 0x0330, 0x93ff, 0x0430, 0x97ff, 0x0530, 0x9bff, 0x0630, 0x9fff, 0x0730, 0xa3ff, 0x0830, 0xa7ff, 0x0930, 0xabff, 0x0a30, 0xafff, 0x0b30, 0xb3ff, 0x0c30, 0xb7ff, 0x0d30, 0xbbff, 0x0e30, 0xbfff, 0x0f30, 0xc3ff, 0x0030, 0xc7ff, 0x0130, 0xcbff, 0x0230, 0xcfff, 0x0330, 0xd3ff, 0x0430, 0xd7ff, 0x0530, 0xdbff, 0x0630, 0xdfff, 0x0730, 0xe3ff, 0x0830, 0xe7ff, 0x0930, 0xebff, 0x0a30, 0xefff, 0x0b30, 0xf3ff, 0x0c30, 0xf7ff, 0x0d30, 0xfbff, 0x0e30, 0xffff, 0x0f30, }; static unsigned short init2[128] = { 0x03ff, 0x8030, 0x07ff, 0x8130, 0x0bff, 0x8230, 0x0fff, 0x8330, 0x13ff, 0x8430, 0x17ff, 0x8530, 0x1bff, 0x8630, 0x1fff, 0x8730, 0x23ff, 0x8830, 0x27ff, 0x8930, 0x2bff, 0x8a30, 0x2fff, 0x8b30, 0x33ff, 0x8c30, 0x37ff, 0x8d30, 0x3bff, 0x8e30, 0x3fff, 0x8f30, 0x43ff, 0x8030, 0x47ff, 0x8130, 0x4bff, 0x8230, 0x4fff, 0x8330, 0x53ff, 0x8430, 0x57ff, 0x8530, 0x5bff, 0x8630, 0x5fff, 0x8730, 0x63ff, 0x8830, 0x67ff, 0x8930, 0x6bff, 0x8a30, 0x6fff, 0x8b30, 0x73ff, 0x8c30, 0x77ff, 0x8d30, 0x7bff, 0x8e30, 0x7fff, 0x8f30, 0x83ff, 0x8030, 0x87ff, 0x8130, 0x8bff, 0x8230, 0x8fff, 0x8330, 0x93ff, 0x8430, 0x97ff, 0x8530, 0x9bff, 0x8630, 0x9fff, 0x8730, 0xa3ff, 0x8830, 0xa7ff, 0x8930, 0xabff, 0x8a30, 0xafff, 0x8b30, 0xb3ff, 0x8c30, 0xb7ff, 0x8d30, 0xbbff, 0x8e30, 0xbfff, 0x8f30, 0xc3ff, 0x8030, 0xc7ff, 0x8130, 0xcbff, 0x8230, 0xcfff, 0x8330, 0xd3ff, 0x8430, 0xd7ff, 0x8530, 0xdbff, 0x8630, 0xdfff, 0x8730, 0xe3ff, 0x8830, 0xe7ff, 0x8930, 0xebff, 0x8a30, 0xefff, 0x8b30, 0xf3ff, 0x8c30, 0xf7ff, 0x8d30, 0xfbff, 0x8e30, 0xffff, 0x8f30, }; static unsigned short init3[128] = { 0x0C10, 0x8470, 0x14FE, 0xB488, 0x167F, 0xA470, 0x18E7, 0x84B5, 0x1B6E, 0x842A, 0x1F1D, 0x852A, 0x0DA3, 0x8F7C, 0x167E, 0xF254, 0x0000, 0x842A, 0x0001, 0x852A, 0x18E6, 0x8BAA, 0x1B6D, 0xF234, 0x229F, 0x8429, 0x2746, 0x8529, 0x1F1C, 0x86E7, 0x229E, 0xF224, 0x0DA4, 0x8429, 0x2C29, 0x8529, 0x2745, 0x87F6, 0x2C28, 0xF254, 0x383B, 0x8428, 0x320F, 0x8528, 0x320E, 0x8F02, 0x1341, 0xF264, 0x3EB6, 0x8428, 0x3EB9, 0x8528, 0x383A, 0x8FA9, 0x3EB5, 0xF294, 0x3EB7, 0x8474, 0x3EBA, 0x8575, 0x3EB8, 0xC4C3, 0x3EBB, 0xC5C3, 0x0000, 0xA404, 0x0001, 0xA504, 0x141F, 0x8671, 0x14FD, 0x8287, 0x3EBC, 0xE610, 0x3EC8, 0x8C7B, 0x031A, 0x87E6, 0x3EC8, 0x86F7, 0x3EC0, 0x821E, 0x3EBE, 0xD208, 0x3EBD, 0x821F, 0x3ECA, 0x8386, 0x3EC1, 0x8C03, 0x3EC9, 0x831E, 0x3ECA, 0x8C4C, 0x3EBF, 0x8C55, 0x3EC9, 0xC208, 0x3EC4, 0xBC84, 0x3EC8, 0x8EAD, 0x3EC8, 0xD308, 0x3EC2, 0x8F7E, 0x3ECB, 0x8219, 0x3ECB, 0xD26E, 0x3EC5, 0x831F, 0x3EC6, 0xC308, 0x3EC3, 0xB2FF, 0x3EC9, 0x8265, 0x3EC9, 0x8319, 0x1342, 0xD36E, 0x3EC7, 0xB3FF, 0x0000, 0x8365, 0x1420, 0x9570, }; static unsigned short init4[128] = { 0x0C10, 0x8470, 0x14FE, 0xB488, 0x167F, 0xA470, 0x18E7, 0x84B5, 0x1B6E, 0x842A, 0x1F1D, 0x852A, 0x0DA3, 0x0F7C, 0x167E, 0x7254, 0x0000, 0x842A, 0x0001, 0x852A, 0x18E6, 0x0BAA, 0x1B6D, 0x7234, 0x229F, 0x8429, 0x2746, 0x8529, 0x1F1C, 0x06E7, 0x229E, 0x7224, 0x0DA4, 0x8429, 0x2C29, 0x8529, 0x2745, 0x07F6, 0x2C28, 0x7254, 0x383B, 0x8428, 0x320F, 0x8528, 0x320E, 0x0F02, 0x1341, 0x7264, 0x3EB6, 0x8428, 0x3EB9, 0x8528, 0x383A, 0x0FA9, 0x3EB5, 0x7294, 0x3EB7, 0x8474, 0x3EBA, 0x8575, 0x3EB8, 0x44C3, 0x3EBB, 0x45C3, 0x0000, 0xA404, 0x0001, 0xA504, 0x141F, 0x0671, 0x14FD, 0x0287, 0x3EBC, 0xE610, 0x3EC8, 0x0C7B, 0x031A, 0x07E6, 0x3EC8, 0x86F7, 0x3EC0, 0x821E, 0x3EBE, 0xD208, 0x3EBD, 0x021F, 0x3ECA, 0x0386, 0x3EC1, 0x0C03, 0x3EC9, 0x031E, 0x3ECA, 0x8C4C, 0x3EBF, 0x0C55, 0x3EC9, 0xC208, 0x3EC4, 0xBC84, 0x3EC8, 0x0EAD, 0x3EC8, 0xD308, 0x3EC2, 0x8F7E, 0x3ECB, 0x0219, 0x3ECB, 0xD26E, 0x3EC5, 0x031F, 0x3EC6, 0xC308, 0x3EC3, 0x32FF, 0x3EC9, 0x0265, 0x3EC9, 0x8319, 0x1342, 0xD36E, 0x3EC7, 0x33FF, 0x0000, 0x8365, 0x1420, 0x9570, }; /* send initialization arrays to start up */ static void awe_init_array(void) { awe_send_array(init1); awe_wait(1024); awe_send_array(init2); awe_send_array(init3); awe_poke_dw(AWE_HWCF4, 0); awe_poke_dw(AWE_HWCF5, 0x83); awe_poke_dw(AWE_HWCF6, 0x8000); awe_send_array(init4); } /* send an initialization array */ static void awe_send_array(unsigned short *data) { int i; unsigned short *p; p = data; for (i = 0; i < AWE_MAX_VOICES; i++, p++) awe_poke(AWE_INIT1(i), *p); for (i = 0; i < AWE_MAX_VOICES; i++, p++) awe_poke(AWE_INIT2(i), *p); for (i = 0; i < AWE_MAX_VOICES; i++, p++) awe_poke(AWE_INIT3(i), *p); for (i = 0; i < AWE_MAX_VOICES; i++, p++) awe_poke(AWE_INIT4(i), *p); } /* * set up awe32 channels to some known state. */ /* set the envelope & LFO parameters to the default values; see ADIP */ static void awe_tweak_voice(int i) { /* set all mod/vol envelope shape to minimum */ awe_poke(AWE_ENVVOL(i), 0x8000); awe_poke(AWE_ENVVAL(i), 0x8000); awe_poke(AWE_DCYSUS(i), 0x7F7F); awe_poke(AWE_ATKHLDV(i), 0x7F7F); awe_poke(AWE_ATKHLD(i), 0x7F7F); awe_poke(AWE_PEFE(i), 0); /* mod envelope height to zero */ awe_poke(AWE_LFO1VAL(i), 0x8000); /* no delay for LFO1 */ awe_poke(AWE_LFO2VAL(i), 0x8000); awe_poke(AWE_IP(i), 0xE000); /* no pitch shift */ awe_poke(AWE_IFATN(i), 0xFF00); /* volume to minimum */ awe_poke(AWE_FMMOD(i), 0); awe_poke(AWE_TREMFRQ(i), 0); awe_poke(AWE_FM2FRQ2(i), 0); } static void awe_tweak(void) { int i; /* reset all channels */ for (i = 0; i < awe_max_voices; i++) awe_tweak_voice(i); } /* * initializes the FM section of AWE32; * see Vince Vu's unofficial AWE32 programming guide */ static void awe_init_fm(void) { #ifndef AWE_ALWAYS_INIT_FM /* if no extended memory is on board.. */ if (awe_mem_size <= 0) return; #endif AWE_DEBUG(3,printk("AWE32: initializing FM\n")); /* Initialize the last two channels for DRAM refresh and producing the reverb and chorus effects for Yamaha OPL-3 synthesizer */ /* 31: FM left channel, 0xffffe0-0xffffe8 */ awe_poke(AWE_DCYSUSV(30), 0x80); awe_poke_dw(AWE_PSST(30), 0xFFFFFFE0); /* full left */ awe_poke_dw(AWE_CSL(30), 0x00FFFFE8 | (DEF_FM_CHORUS_DEPTH << 24)); awe_poke_dw(AWE_PTRX(30), (DEF_FM_REVERB_DEPTH << 8)); awe_poke_dw(AWE_CPF(30), 0); awe_poke_dw(AWE_CCCA(30), 0x00FFFFE3); /* 32: FM right channel, 0xfffff0-0xfffff8 */ awe_poke(AWE_DCYSUSV(31), 0x80); awe_poke_dw(AWE_PSST(31), 0x00FFFFF0); /* full right */ awe_poke_dw(AWE_CSL(31), 0x00FFFFF8 | (DEF_FM_CHORUS_DEPTH << 24)); awe_poke_dw(AWE_PTRX(31), (DEF_FM_REVERB_DEPTH << 8)); awe_poke_dw(AWE_CPF(31), 0x8000); awe_poke_dw(AWE_CCCA(31), 0x00FFFFF3); /* skew volume & cutoff */ awe_poke_dw(AWE_VTFT(30), 0x8000FFFF); awe_poke_dw(AWE_VTFT(31), 0x8000FFFF); voices[30].state = AWE_ST_FM; voices[31].state = AWE_ST_FM; /* change maximum channels to 30 */ awe_max_voices = AWE_NORMAL_VOICES; if (playing_mode == AWE_PLAY_DIRECT) awe_info.nr_voices = awe_max_voices; else awe_info.nr_voices = AWE_MAX_CHANNELS; voice_alloc->max_voice = awe_max_voices; } /* * AWE32 DRAM access routines */ /* open DRAM write accessing mode */ static int awe_open_dram_for_write(int offset, int channels) { int vidx[AWE_NORMAL_VOICES]; int i; if (channels < 0 || channels >= AWE_NORMAL_VOICES) { channels = AWE_NORMAL_VOICES; for (i = 0; i < AWE_NORMAL_VOICES; i++) vidx[i] = i; } else { for (i = 0; i < channels; i++) vidx[i] = awe_clear_voice(); } /* use all channels for DMA transfer */ for (i = 0; i < channels; i++) { if (vidx[i] < 0) continue; awe_poke(AWE_DCYSUSV(vidx[i]), 0x80); awe_poke_dw(AWE_VTFT(vidx[i]), 0); awe_poke_dw(AWE_CVCF(vidx[i]), 0); awe_poke_dw(AWE_PTRX(vidx[i]), 0x40000000); awe_poke_dw(AWE_CPF(vidx[i]), 0x40000000); awe_poke_dw(AWE_PSST(vidx[i]), 0); awe_poke_dw(AWE_CSL(vidx[i]), 0); awe_poke_dw(AWE_CCCA(vidx[i]), 0x06000000); voices[vidx[i]].state = AWE_ST_DRAM; } /* point channels 31 & 32 to ROM samples for DRAM refresh */ awe_poke_dw(AWE_VTFT(30), 0); awe_poke_dw(AWE_PSST(30), 0x1d8); awe_poke_dw(AWE_CSL(30), 0x1e0); awe_poke_dw(AWE_CCCA(30), 0x1d8); awe_poke_dw(AWE_VTFT(31), 0); awe_poke_dw(AWE_PSST(31), 0x1d8); awe_poke_dw(AWE_CSL(31), 0x1e0); awe_poke_dw(AWE_CCCA(31), 0x1d8); voices[30].state = AWE_ST_FM; voices[31].state = AWE_ST_FM; /* if full bit is on, not ready to write on */ if (awe_peek_dw(AWE_SMALW) & 0x80000000) { for (i = 0; i < channels; i++) { awe_poke_dw(AWE_CCCA(vidx[i]), 0); voices[i].state = AWE_ST_OFF; } return RET_ERROR(ENOSPC); } /* set address to write */ awe_poke_dw(AWE_SMALW, offset); return 0; } /* open DRAM for RAM size detection */ static void awe_open_dram_for_check(void) { int i; for (i = 0; i < AWE_NORMAL_VOICES; i++) { awe_poke(AWE_DCYSUSV(i), 0x80); awe_poke_dw(AWE_VTFT(i), 0); awe_poke_dw(AWE_CVCF(i), 0); awe_poke_dw(AWE_PTRX(i), 0x40000000); awe_poke_dw(AWE_CPF(i), 0x40000000); awe_poke_dw(AWE_PSST(i), 0); awe_poke_dw(AWE_CSL(i), 0); if (i & 1) /* DMA write */ awe_poke_dw(AWE_CCCA(i), 0x06000000); else /* DMA read */ awe_poke_dw(AWE_CCCA(i), 0x04000000); voices[i].state = AWE_ST_DRAM; } } /* close dram access */ static void awe_close_dram(void) { int i; /* wait until FULL bit in SMAxW register be false */ for (i = 0; i < 10000; i++) { if (!(awe_peek_dw(AWE_SMALW) & 0x80000000)) break; awe_wait(10); } for (i = 0; i < AWE_NORMAL_VOICES; i++) { if (voices[i].state == AWE_ST_DRAM) { awe_poke_dw(AWE_CCCA(i), 0); awe_poke(AWE_DCYSUSV(i), 0x807F); voices[i].state = AWE_ST_OFF; } } } /*================================================================ * detect presence of AWE32 and check memory size *================================================================*/ /* detect emu8000 chip on the specified address; from VV's guide */ static int awe_detect_base(int addr) { awe_base = addr; if ((awe_peek(AWE_U1) & 0x000F) != 0x000C) return 0; if ((awe_peek(AWE_HWCF1) & 0x007E) != 0x0058) return 0; if ((awe_peek(AWE_HWCF2) & 0x0003) != 0x0003) return 0; AWE_DEBUG(0,printk("AWE32 found at %x\n", awe_base)); return 1; } static int awe_detect(void) { int base; if (awe_base == 0) { for (base = 0x620; base <= 0x680; base += 0x20) if (awe_detect_base(base)) return 1; AWE_DEBUG(0,printk("AWE32 not found\n")); return 0; } return 1; } /*================================================================ * check dram size on AWE board *================================================================*/ /* any three numbers you like */ #define UNIQUE_ID1 0x1234 #define UNIQUE_ID2 0x4321 #define UNIQUE_ID3 0xFFFF static int awe_check_dram(void) { if (awe_mem_size > 0) { awe_mem_size *= 1024; /* convert to Kbytes */ return awe_mem_size; } awe_open_dram_for_check(); awe_mem_size = 0; /* set up unique two id numbers */ awe_poke_dw(AWE_SMALW, AWE_DRAM_OFFSET); awe_poke(AWE_SMLD, UNIQUE_ID1); awe_poke(AWE_SMLD, UNIQUE_ID2); while (awe_mem_size < AWE_MAX_DRAM_SIZE) { awe_wait(2); /* read a data on the DRAM start address */ awe_poke_dw(AWE_SMALR, AWE_DRAM_OFFSET); awe_peek(AWE_SMLD); /* discard stale data */ if (awe_peek(AWE_SMLD) != UNIQUE_ID1) break; if (awe_peek(AWE_SMLD) != UNIQUE_ID2) break; awe_mem_size += 32; /* increment 32 Kbytes */ /* Write a unique data on the test address; * if the address is out of range, the data is written on * 0x200000(=AWE_DRAM_OFFSET). Then the two id words are * broken by this data. */ awe_poke_dw(AWE_SMALW, AWE_DRAM_OFFSET + awe_mem_size*512L); awe_poke(AWE_SMLD, UNIQUE_ID3); awe_wait(2); /* read a data on the just written DRAM address */ awe_poke_dw(AWE_SMALR, AWE_DRAM_OFFSET + awe_mem_size*512L); awe_peek(AWE_SMLD); /* discard stale data */ if (awe_peek(AWE_SMLD) != UNIQUE_ID3) break; } awe_close_dram(); AWE_DEBUG(0,printk("AWE32: %d Kbytes memory detected\n", awe_mem_size)); /* convert to Kbytes */ awe_mem_size *= 1024; return awe_mem_size; } /*================================================================ * chorus and reverb controls; from VV's guide *================================================================*/ /* 5 parameters for each chorus mode; 3 x 16bit, 2 x 32bit */ static char chorus_defined[AWE_CHORUS_NUMBERS]; static awe_chorus_fx_rec chorus_parm[AWE_CHORUS_NUMBERS] = { {0xE600, 0x03F6, 0xBC2C ,0x00000000, 0x0000006D}, /* chorus 1 */ {0xE608, 0x031A, 0xBC6E, 0x00000000, 0x0000017C}, /* chorus 2 */ {0xE610, 0x031A, 0xBC84, 0x00000000, 0x00000083}, /* chorus 3 */ {0xE620, 0x0269, 0xBC6E, 0x00000000, 0x0000017C}, /* chorus 4 */ {0xE680, 0x04D3, 0xBCA6, 0x00000000, 0x0000005B}, /* feedback */ {0xE6E0, 0x044E, 0xBC37, 0x00000000, 0x00000026}, /* flanger */ {0xE600, 0x0B06, 0xBC00, 0x0000E000, 0x00000083}, /* short delay */ {0xE6C0, 0x0B06, 0xBC00, 0x0000E000, 0x00000083}, /* short delay + feedback */ }; static int awe_load_chorus_fx(awe_patch_info *patch, const char *addr, int count) { if (patch->optarg < AWE_CHORUS_PREDEFINED || patch->optarg >= AWE_CHORUS_NUMBERS) { printk("AWE32 Error: illegal chorus mode %d for uploading\n", patch->optarg); return RET_ERROR(EINVAL); } if (count < sizeof(awe_chorus_fx_rec)) { printk("AWE32 Error: too short chorus fx parameters\n"); return RET_ERROR(EINVAL); } COPY_FROM_USER(&chorus_parm[patch->optarg], addr, AWE_PATCH_INFO_SIZE, sizeof(awe_chorus_fx_rec)); chorus_defined[patch->optarg] = TRUE; return 0; } static void awe_set_chorus_mode(int effect) { if (effect < 0 || effect >= AWE_CHORUS_NUMBERS || (effect >= AWE_CHORUS_PREDEFINED && !chorus_defined[effect])) return; awe_poke(AWE_INIT3(9), chorus_parm[effect].feedback); awe_poke(AWE_INIT3(12), chorus_parm[effect].delay_offset); awe_poke(AWE_INIT4(3), chorus_parm[effect].lfo_depth); awe_poke_dw(AWE_HWCF4, chorus_parm[effect].delay); awe_poke_dw(AWE_HWCF5, chorus_parm[effect].lfo_freq); awe_poke_dw(AWE_HWCF6, 0x8000); awe_poke_dw(AWE_HWCF7, 0x0000); chorus_mode = effect; } /*----------------------------------------------------------------*/ /* reverb mode settings; write the following 28 data of 16 bit length * on the corresponding ports in the reverb_cmds array */ static char reverb_defined[AWE_CHORUS_NUMBERS]; static awe_reverb_fx_rec reverb_parm[AWE_REVERB_NUMBERS] = { {{ /* room 1 */ 0xB488, 0xA450, 0x9550, 0x84B5, 0x383A, 0x3EB5, 0x72F4, 0x72A4, 0x7254, 0x7204, 0x7204, 0x7204, 0x4416, 0x4516, 0xA490, 0xA590, 0x842A, 0x852A, 0x842A, 0x852A, 0x8429, 0x8529, 0x8429, 0x8529, 0x8428, 0x8528, 0x8428, 0x8528, }}, {{ /* room 2 */ 0xB488, 0xA458, 0x9558, 0x84B5, 0x383A, 0x3EB5, 0x7284, 0x7254, 0x7224, 0x7224, 0x7254, 0x7284, 0x4448, 0x4548, 0xA440, 0xA540, 0x842A, 0x852A, 0x842A, 0x852A, 0x8429, 0x8529, 0x8429, 0x8529, 0x8428, 0x8528, 0x8428, 0x8528, }}, {{ /* room 3 */ 0xB488, 0xA460, 0x9560, 0x84B5, 0x383A, 0x3EB5, 0x7284, 0x7254, 0x7224, 0x7224, 0x7254, 0x7284, 0x4416, 0x4516, 0xA490, 0xA590, 0x842C, 0x852C, 0x842C, 0x852C, 0x842B, 0x852B, 0x842B, 0x852B, 0x842A, 0x852A, 0x842A, 0x852A, }}, {{ /* hall 1 */ 0xB488, 0xA470, 0x9570, 0x84B5, 0x383A, 0x3EB5, 0x7284, 0x7254, 0x7224, 0x7224, 0x7254, 0x7284, 0x4448, 0x4548, 0xA440, 0xA540, 0x842B, 0x852B, 0x842B, 0x852B, 0x842A, 0x852A, 0x842A, 0x852A, 0x8429, 0x8529, 0x8429, 0x8529, }}, {{ /* hall 2 */ 0xB488, 0xA470, 0x9570, 0x84B5, 0x383A, 0x3EB5, 0x7254, 0x7234, 0x7224, 0x7254, 0x7264, 0x7294, 0x44C3, 0x45C3, 0xA404, 0xA504, 0x842A, 0x852A, 0x842A, 0x852A, 0x8429, 0x8529, 0x8429, 0x8529, 0x8428, 0x8528, 0x8428, 0x8528, }}, {{ /* plate */ 0xB4FF, 0xA470, 0x9570, 0x84B5, 0x383A, 0x3EB5, 0x7234, 0x7234, 0x7234, 0x7234, 0x7234, 0x7234, 0x4448, 0x4548, 0xA440, 0xA540, 0x842A, 0x852A, 0x842A, 0x852A, 0x8429, 0x8529, 0x8429, 0x8529, 0x8428, 0x8528, 0x8428, 0x8528, }}, {{ /* delay */ 0xB4FF, 0xA470, 0x9500, 0x84B5, 0x333A, 0x39B5, 0x7204, 0x7204, 0x7204, 0x7204, 0x7204, 0x72F4, 0x4400, 0x4500, 0xA4FF, 0xA5FF, 0x8420, 0x8520, 0x8420, 0x8520, 0x8420, 0x8520, 0x8420, 0x8520, 0x8420, 0x8520, 0x8420, 0x8520, }}, {{ /* panning delay */ 0xB4FF, 0xA490, 0x9590, 0x8474, 0x333A, 0x39B5, 0x7204, 0x7204, 0x7204, 0x7204, 0x7204, 0x72F4, 0x4400, 0x4500, 0xA4FF, 0xA5FF, 0x8420, 0x8520, 0x8420, 0x8520, 0x8420, 0x8520, 0x8420, 0x8520, 0x8420, 0x8520, 0x8420, 0x8520, }}, }; static struct ReverbCmdPair { unsigned short cmd, port; } reverb_cmds[28] = { {AWE_INIT1(0x03)}, {AWE_INIT1(0x05)}, {AWE_INIT4(0x1F)}, {AWE_INIT1(0x07)}, {AWE_INIT2(0x14)}, {AWE_INIT2(0x16)}, {AWE_INIT1(0x0F)}, {AWE_INIT1(0x17)}, {AWE_INIT1(0x1F)}, {AWE_INIT2(0x07)}, {AWE_INIT2(0x0F)}, {AWE_INIT2(0x17)}, {AWE_INIT2(0x1D)}, {AWE_INIT2(0x1F)}, {AWE_INIT3(0x01)}, {AWE_INIT3(0x03)}, {AWE_INIT1(0x09)}, {AWE_INIT1(0x0B)}, {AWE_INIT1(0x11)}, {AWE_INIT1(0x13)}, {AWE_INIT1(0x19)}, {AWE_INIT1(0x1B)}, {AWE_INIT2(0x01)}, {AWE_INIT2(0x03)}, {AWE_INIT2(0x09)}, {AWE_INIT2(0x0B)}, {AWE_INIT2(0x11)}, {AWE_INIT2(0x13)}, }; static int awe_load_reverb_fx(awe_patch_info *patch, const char *addr, int count) { if (patch->optarg < AWE_REVERB_PREDEFINED || patch->optarg >= AWE_REVERB_NUMBERS) { printk("AWE32 Error: illegal reverb mode %d for uploading\n", patch->optarg); return RET_ERROR(EINVAL); } if (count < sizeof(awe_reverb_fx_rec)) { printk("AWE32 Error: too short reverb fx parameters\n"); return RET_ERROR(EINVAL); } COPY_FROM_USER(&reverb_parm[patch->optarg], addr, AWE_PATCH_INFO_SIZE, sizeof(awe_reverb_fx_rec)); reverb_defined[patch->optarg] = TRUE; return 0; } static void awe_set_reverb_mode(int effect) { int i; if (effect < 0 || effect >= AWE_REVERB_NUMBERS || (effect >= AWE_REVERB_PREDEFINED && !reverb_defined[effect])) return; for (i = 0; i < 28; i++) awe_poke(reverb_cmds[i].cmd, reverb_cmds[i].port, reverb_parm[effect].parms[i]); reverb_mode = effect; } /*================================================================ * treble/bass equalizer control *================================================================*/ static unsigned short bass_parm[12][3] = { {0xD26A, 0xD36A, 0x0000}, /* -12 dB */ {0xD25B, 0xD35B, 0x0000}, /* -8 */ {0xD24C, 0xD34C, 0x0000}, /* -6 */ {0xD23D, 0xD33D, 0x0000}, /* -4 */ {0xD21F, 0xD31F, 0x0000}, /* -2 */ {0xC208, 0xC308, 0x0001}, /* 0 (HW default) */ {0xC219, 0xC319, 0x0001}, /* +2 */ {0xC22A, 0xC32A, 0x0001}, /* +4 */ {0xC24C, 0xC34C, 0x0001}, /* +6 */ {0xC26E, 0xC36E, 0x0001}, /* +8 */ {0xC248, 0xC348, 0x0002}, /* +10 */ {0xC26A, 0xC36A, 0x0002}, /* +12 dB */ }; static unsigned short treble_parm[12][9] = { {0x821E, 0xC26A, 0x031E, 0xC36A, 0x021E, 0xD208, 0x831E, 0xD308, 0x0001}, /* -12 dB */ {0x821E, 0xC25B, 0x031E, 0xC35B, 0x021E, 0xD208, 0x831E, 0xD308, 0x0001}, {0x821E, 0xC24C, 0x031E, 0xC34C, 0x021E, 0xD208, 0x831E, 0xD308, 0x0001}, {0x821E, 0xC23D, 0x031E, 0xC33D, 0x021E, 0xD208, 0x831E, 0xD308, 0x0001}, {0x821E, 0xC21F, 0x031E, 0xC31F, 0x021E, 0xD208, 0x831E, 0xD308, 0x0001}, {0x821E, 0xD208, 0x031E, 0xD308, 0x021E, 0xD208, 0x831E, 0xD308, 0x0002}, {0x821E, 0xD208, 0x031E, 0xD308, 0x021D, 0xD219, 0x831D, 0xD319, 0x0002}, {0x821E, 0xD208, 0x031E, 0xD308, 0x021C, 0xD22A, 0x831C, 0xD32A, 0x0002}, {0x821E, 0xD208, 0x031E, 0xD308, 0x021A, 0xD24C, 0x831A, 0xD34C, 0x0002}, {0x821E, 0xD208, 0x031E, 0xD308, 0x0219, 0xD26E, 0x8319, 0xD36E, 0x0002}, /* +8 (HW default) */ {0x821D, 0xD219, 0x031D, 0xD319, 0x0219, 0xD26E, 0x8319, 0xD36E, 0x0002}, {0x821C, 0xD22A, 0x031C, 0xD32A, 0x0219, 0xD26E, 0x8319, 0xD36E, 0x0002}, /* +12 dB */ }; /* * set Emu8000 digital equalizer; from 0 to 11 [-12dB - 12dB] */ static void awe_equalizer(int bass, int treble) { unsigned short w; if (bass < 0 || bass > 11 || treble < 0 || treble > 11) return; awe_bass_level = bass; awe_treble_level = treble; awe_poke(AWE_INIT4(0x01), bass_parm[bass][0]); awe_poke(AWE_INIT4(0x11), bass_parm[bass][1]); awe_poke(AWE_INIT3(0x11), treble_parm[treble][0]); awe_poke(AWE_INIT3(0x13), treble_parm[treble][1]); awe_poke(AWE_INIT3(0x1B), treble_parm[treble][2]); awe_poke(AWE_INIT4(0x07), treble_parm[treble][3]); awe_poke(AWE_INIT4(0x0B), treble_parm[treble][4]); awe_poke(AWE_INIT4(0x0D), treble_parm[treble][5]); awe_poke(AWE_INIT4(0x17), treble_parm[treble][6]); awe_poke(AWE_INIT4(0x19), treble_parm[treble][7]); w = bass_parm[bass][2] + treble_parm[treble][8]; awe_poke(AWE_INIT4(0x15), (unsigned short)(w + 0x0262)); awe_poke(AWE_INIT4(0x1D), (unsigned short)(w + 0x8362)); } #endif /* CONFIG_AWE32_SYNTH */