diff options
| -rw-r--r-- | sys/arch/i386/conf/GENERIC | 6 | ||||
| -rw-r--r-- | sys/dev/isa/ad1848var.h | 3 | ||||
| -rw-r--r-- | sys/dev/isa/files.isa | 11 | ||||
| -rw-r--r-- | sys/dev/isa/pss.c | 1286 | ||||
| -rw-r--r-- | sys/dev/isa/pssreg.h | 165 |
5 files changed, 3 insertions, 1468 deletions
diff --git a/sys/arch/i386/conf/GENERIC b/sys/arch/i386/conf/GENERIC index a258198f689..a261e82a444 100644 --- a/sys/arch/i386/conf/GENERIC +++ b/sys/arch/i386/conf/GENERIC @@ -1,4 +1,4 @@ -# $OpenBSD: GENERIC,v 1.719 2011/06/26 23:19:10 tedu Exp $ +# $OpenBSD: GENERIC,v 1.720 2011/06/29 17:48:22 tedu Exp $ # # For further information on compiling OpenBSD kernels, see the config(8) # man page. @@ -706,9 +706,6 @@ ipgphy* at mii? # IC Plus IP1000A PHYs mlphy* at mii? # Micro Linear 6692 PHY ukphy* at mii? # "unknown" PHYs -pss0 at isa? port 0x220 irq 7 drq 6 # Personal Sound System -sp0 at pss0 port 0x530 irq 10 drq 0 # sound port driver - eap* at pci? # Ensoniq AudioPCI S5016 eso* at pci? # ESS Solo-1 PCI AudioDrive sv* at pci? # S3 SonicVibes (S3 617) @@ -754,7 +751,6 @@ spkr0 at pcppi? # PC speaker audio* at sb? audio* at gus? audio* at pas? -audio* at sp? audio* at ess? audio* at wss? audio* at ym? diff --git a/sys/dev/isa/ad1848var.h b/sys/dev/isa/ad1848var.h index b5d1e73bed9..0e193f77465 100644 --- a/sys/dev/isa/ad1848var.h +++ b/sys/dev/isa/ad1848var.h @@ -1,4 +1,4 @@ -/* $OpenBSD: ad1848var.h,v 1.13 2010/06/30 11:21:35 jakemsr Exp $ */ +/* $OpenBSD: ad1848var.h,v 1.14 2011/06/29 17:48:22 tedu Exp $ */ /* $NetBSD: ad1848var.h,v 1.22 1998/01/19 22:18:26 augustss Exp $ */ /* @@ -92,7 +92,6 @@ struct ad1848_softc { void *sc_parg; /* play arg for sc_intr() */ void *sc_rarg; /* rec arg for sc_intr() */ - /* Only used by pss XXX */ int sc_iobase; }; diff --git a/sys/dev/isa/files.isa b/sys/dev/isa/files.isa index 1b407e9d74a..bbc475d949d 100644 --- a/sys/dev/isa/files.isa +++ b/sys/dev/isa/files.isa @@ -1,4 +1,4 @@ -# $OpenBSD: files.isa,v 1.110 2011/06/28 20:19:19 matthieu Exp $ +# $OpenBSD: files.isa,v 1.111 2011/06/29 17:48:22 tedu Exp $ # $NetBSD: files.isa,v 1.21 1996/05/16 03:45:55 mycroft Exp $ # # Config file and device description for machine-independent ISA code. @@ -241,15 +241,6 @@ define ics2101 file dev/isa/ics2101.c ics2101 -# Audio systems based on Echo Speech Corp. ESC61[45] ASICs -device pss {[port = -1], [size = 0], - [iomem = -1], [iosiz = 0], - [irq = -1], [drq = -1]} -attach pss at isa -device sp: audio, isa_dma, ad1848, auconv -attach sp at pss -file dev/isa/pss.c pss needs-flag - # Microsoft Windows Sound System device wss: audio, isa_dma, ad1848, auconv file dev/isa/wss.c wss needs-flag diff --git a/sys/dev/isa/pss.c b/sys/dev/isa/pss.c deleted file mode 100644 index 57d0a2d8641..00000000000 --- a/sys/dev/isa/pss.c +++ /dev/null @@ -1,1286 +0,0 @@ -/* $OpenBSD: pss.c,v 1.25 2011/06/29 12:17:40 tedu Exp $ */ -/* $NetBSD: pss.c,v 1.38 1998/01/12 09:43:44 thorpej Exp $ */ - -/* - * Copyright (c) 1994 John Brezak - * Copyright (c) 1991-1993 Regents of the University of California. - * All rights reserved. - * - * Redistribution and use in source and binary forms, with or without - * modification, are permitted provided that the following conditions - * are met: - * 1. Redistributions of source code must retain the above copyright - * notice, this list of conditions and the following disclaimer. - * 2. Redistributions in binary form must reproduce the above copyright - * notice, this list of conditions and the following disclaimer in the - * documentation and/or other materials provided with the distribution. - * 3. All advertising materials mentioning features or use of this software - * must display the following acknowledgement: - * This product includes software developed by the Computer Systems - * Engineering Group at Lawrence Berkeley Laboratory. - * 4. Neither the name of the University nor of the Laboratory may be used - * to endorse or promote products derived from this software without - * specific prior written permission. - * - * THIS SOFTWARE IS PROVIDED BY THE REGENTS AND CONTRIBUTORS ``AS IS'' AND - * ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE - * IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE - * ARE DISCLAIMED. IN NO EVENT SHALL THE REGENTS OR CONTRIBUTORS BE LIABLE - * FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL - * DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS - * OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION) - * HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT - * LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY - * OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF - * SUCH DAMAGE. - * - */ - -/* - * Copyright (c) 1993 Analog Devices Inc. All rights reserved - * - * Portions provided by Marc.Hoffman@analog.com and - * Greg.Yukna@analog.com . - * - */ - -/* - * Todo: - * - Provide PSS driver to access DSP - * - Provide MIDI driver to access MPU - * - Finish support for CD drive (Sony and SCSI) - */ - -#include <sys/param.h> -#include <sys/systm.h> -#include <sys/errno.h> -#include <sys/ioctl.h> -#include <sys/syslog.h> -#include <sys/device.h> -#include <sys/proc.h> -#include <sys/buf.h> - -#include <machine/cpu.h> -#include <machine/intr.h> -#include <machine/bus.h> - -#include <sys/audioio.h> -#include <dev/audio_if.h> - -#include <dev/isa/isavar.h> -#include <dev/isa/isadmavar.h> - -#include <dev/isa/ad1848var.h> -#include <dev/isa/wssreg.h> -#include <dev/isa/pssreg.h> - -/* XXX Default WSS base */ -#define WSS_BASE_ADDRESS 0x0530 - -/* - * Mixer devices - */ -#define PSS_MIC_IN_LVL 0 -#define PSS_LINE_IN_LVL 1 -#define PSS_DAC_LVL 2 -#define PSS_REC_LVL 3 -#define PSS_MON_LVL 4 -#define PSS_MASTER_VOL 5 -#define PSS_MASTER_TREBLE 6 -#define PSS_MASTER_BASS 7 -#define PSS_MIC_IN_MUTE 8 -#define PSS_LINE_IN_MUTE 9 -#define PSS_DAC_MUTE 10 - -#define PSS_OUTPUT_MODE 11 -#define PSS_SPKR_MONO 0 -#define PSS_SPKR_STEREO 1 -#define PSS_SPKR_PSEUDO 2 -#define PSS_SPKR_SPATIAL 3 - -#define PSS_RECORD_SOURCE 12 - -/* Classes */ -#define PSS_INPUT_CLASS 13 -#define PSS_RECORD_CLASS 14 -#define PSS_MONITOR_CLASS 15 -#define PSS_OUTPUT_CLASS 16 - - -struct pss_softc { - struct device sc_dev; /* base device */ - void *sc_ih; /* interrupt vectoring */ - - int sc_iobase; /* I/O port base address */ - int sc_drq; /* dma channel */ - - struct ad1848_softc *ad1848_sc; - - int out_port; - - struct ad1848_volume master_volume; - int master_mode; - - int monitor_treble; - int monitor_bass; - - int mic_mute, cd_mute, dac_mute; -}; - -#ifdef AUDIO_DEBUG -#define DPRINTF(x) if (pssdebug) printf x -int pssdebug = 0; -#else -#define DPRINTF(x) -#endif - -int pssprobe(struct device *, void *, void *); -void pssattach(struct device *, struct device *, void *); - -int spprobe(struct device *, void *, void *); -void spattach(struct device *, struct device *, void *); - -int pssintr(void *); - -int pss_speaker_ctl(void *, int); - -int pss_getdev(void *, struct audio_device *); - -int pss_mixer_set_port(void *, mixer_ctrl_t *); -int pss_mixer_get_port(void *, mixer_ctrl_t *); -int pss_query_devinfo(void *, mixer_devinfo_t *); - -#ifdef PSS_DSP -void pss_dspwrite(struct pss_softc *, int); -#endif -void pss_setaddr(int, int); -int pss_setint(int, int); -int pss_setdma(int, int); -int pss_testirq(struct pss_softc *, int); -int pss_testdma(struct pss_softc *, int); -#ifdef AUDIO_DEBUG -void pss_dump_regs(struct pss_softc *); -#endif -int pss_set_master_gain(struct pss_softc *, struct ad1848_volume *); -int pss_set_master_mode(struct pss_softc *, int); -int pss_set_treble(struct pss_softc *, u_int); -int pss_set_bass(struct pss_softc *, u_int); -int pss_get_master_gain(struct pss_softc *, struct ad1848_volume *); -int pss_get_master_mode(struct pss_softc *, u_int *); -int pss_get_treble(struct pss_softc *, u_char *); -int pss_get_bass(struct pss_softc *, u_char *); - -#ifdef AUDIO_DEBUG -void wss_dump_regs(struct ad1848_softc *); -#endif - -/* - * Define our interface to the higher level audio driver. - */ - -struct audio_hw_if pss_audio_if = { - ad1848_open, - ad1848_close, - NULL, - ad1848_query_encoding, - ad1848_set_params, - ad1848_round_blocksize, - ad1848_commit_settings, - NULL, - NULL, - NULL, - NULL, - ad1848_halt_output, - ad1848_halt_input, - pss_speaker_ctl, - pss_getdev, - NULL, - pss_mixer_set_port, - pss_mixer_get_port, - pss_query_devinfo, - ad1848_malloc, - ad1848_free, - ad1848_round, - ad1848_mappage, - ad1848_get_props, - ad1848_trigger_output, - ad1848_trigger_input, - NULL -}; - - -/* Interrupt translation for WSS config */ -static u_char wss_interrupt_bits[16] = { - 0xff, 0xff, 0xff, 0xff, - 0xff, 0xff, 0xff, 0x08, - 0xff, 0x10, 0x18, 0x20, - 0xff, 0xff, 0xff, 0xff -}; -/* ditto for WSS DMA channel */ -static u_char wss_dma_bits[4] = {1, 2, 0, 3}; - -struct cfattach pss_ca = { - sizeof(struct pss_softc), pssprobe, pssattach -}; - -struct cfdriver pss_cd = { - NULL, "pss", DV_DULL, 1 -}; - -struct cfattach sp_ca = { - sizeof(struct ad1848_softc), spprobe, spattach -}; - -struct cfdriver sp_cd = { - NULL, "sp", DV_DULL -}; - -struct audio_device pss_device = { - "pss,ad1848", - "", - "PSS" -}; - -#ifdef PSS_DSP -void -pss_dspwrite(sc, data) - struct pss_softc *sc; - int data; -{ - int i; - int pss_base = sc->sc_iobase; - - /* - * Note! the i<5000000 is an emergency exit. The dsp_command() is sometimes - * called while interrupts are disabled. This means that the timer is - * disabled also. However the timeout situation is a abnormal condition. - * Normally the DSP should be ready to accept commands after just couple of - * loops. - */ - for (i = 0; i < 5000000; i++) { - if (inw(pss_base+PSS_STATUS) & PSS_WRITE_EMPTY) { - outw(pss_base+PSS_DATA, data); - return; - } - } - printf ("pss: DSP Command (%04x) Timeout.\n", data); -} -#endif /* PSS_DSP */ - -void -pss_setaddr(addr, configAddr) - int addr; - int configAddr; -{ - int val; - - val = inw(configAddr); - val &= ADDR_MASK; - val |= (addr << 4); - outw(configAddr,val); -} - -/* pss_setint - * This function sets the correct bits in the - * configuration register to - * enable the chosen interrupt. - */ -int -pss_setint(intNum, configAddress) - int intNum; - int configAddress; -{ - int val; - - switch(intNum) { - case 3: - val = inw(configAddress); - val &= INT_MASK; - val |= INT_3_BITS; - break; - case 5: - val = inw(configAddress); - val &= INT_MASK; - val |= INT_5_BITS; - break; - case 7: - val = inw(configAddress); - val &= INT_MASK; - val |= INT_7_BITS; - break; - case 9: - val = inw(configAddress); - val &= INT_MASK; - val |= INT_9_BITS; - break; - case 10: - val = inw(configAddress); - val &= INT_MASK; - val |= INT_10_BITS; - break; - case 11: - val = inw(configAddress); - val &= INT_MASK; - val |= INT_11_BITS; - break; - case 12: - val = inw(configAddress); - val &= INT_MASK; - val |= INT_12_BITS; - break; - default: - DPRINTF(("pss_setint: invalid irq (%d)\n", intNum)); - return 1; - } - outw(configAddress,val); - return 0; -} - -int -pss_setdma(dmaNum, configAddress) - int dmaNum; - int configAddress; -{ - int val; - - switch(dmaNum) { - case 0: - val = inw(configAddress); - val &= DMA_MASK; - val |= DMA_0_BITS; - break; - case 1: - val = inw(configAddress); - val &= DMA_MASK; - val |= DMA_1_BITS; - break; - case 3: - val = inw(configAddress); - val &= DMA_MASK; - val |= DMA_3_BITS; - break; - case 5: - val = inw(configAddress); - val &= DMA_MASK; - val |= DMA_5_BITS; - break; - case 6: - val = inw(configAddress); - val &= DMA_MASK; - val |= DMA_6_BITS; - break; - case 7: - val = inw(configAddress); - val &= DMA_MASK; - val |= DMA_7_BITS; - break; - default: - DPRINTF(("pss_setdma: invalid drq (%d)\n", dmaNum)); - return 1; - } - outw(configAddress, val); - return 0; -} - -/* - * This function tests an interrupt number to see if - * it is available. It takes the interrupt button - * as its argument and returns TRUE if the interrupt - * is ok. -*/ -int -pss_testirq(struct pss_softc *sc, int intNum) -{ - int config = sc->sc_iobase + PSS_CONFIG; - int val; - int ret; - int i; - - /* Set the interrupt bits */ - switch(intNum) { - case 3: - val = inw(config); - val &= INT_MASK; /* Special: 0 */ - break; - case 5: - val = inw(config); - val &= INT_MASK; - val |= INT_TEST_BIT | INT_5_BITS; - break; - case 7: - val = inw(config); - val &= INT_MASK; - val |= INT_TEST_BIT | INT_7_BITS; - break; - case 9: - val = inw(config); - val &= INT_MASK; - val |= INT_TEST_BIT | INT_9_BITS; - break; - case 10: - val = inw(config); - val &= INT_MASK; - val |= INT_TEST_BIT | INT_10_BITS; - break; - case 11: - val = inw(config); - val &= INT_MASK; - val |= INT_TEST_BIT | INT_11_BITS; - break; - case 12: - val = inw(config); - val &= INT_MASK; - val |= INT_TEST_BIT | INT_12_BITS; - break; - default: - DPRINTF(("pss_testirq: invalid irq (%d)\n", intNum)); - return 0; - } - outw(config, val); - - /* Check if the interrupt is in use */ - /* Do it a few times in case there is a delay */ - ret = 0; - for (i = 0; i < 5; i++) { - val = inw(config); - if (val & INT_TEST_PASS) { - ret = 1; - break; - } - } - - /* Clear the Test bit and the interrupt bits */ - val = inw(config); - val &= INT_TEST_BIT_MASK & INT_MASK; - outw(config, val); - return(ret); -} - -/* - * This function tests a dma channel to see if - * it is available. It takes the DMA channel button - * as its argument and returns TRUE if the channel - * is ok. - */ -int -pss_testdma(sc, dmaNum) - struct pss_softc *sc; - int dmaNum; -{ - int config = sc->sc_iobase + PSS_CONFIG; - int val; - int i, ret; - - switch (dmaNum) { - case 0: - val = inw(config); - val &= DMA_MASK; - val |= DMA_TEST_BIT | DMA_0_BITS; - break; - case 1: - val = inw(config); - val &= DMA_MASK; - val |= DMA_TEST_BIT | DMA_1_BITS; - break; - case 3: - val = inw(config); - val &= DMA_MASK; - val |= DMA_TEST_BIT | DMA_3_BITS; - break; - case 5: - val = inw(config); - val &= DMA_MASK; - val |= DMA_TEST_BIT | DMA_5_BITS; - break; - case 6: - val = inw(config); - val &= DMA_MASK; - val |= DMA_TEST_BIT | DMA_6_BITS; - break; - case 7: - val = inw(config); - val &= DMA_MASK; - val |= DMA_TEST_BIT | DMA_7_BITS; - break; - default: - DPRINTF(("pss_testdma: invalid drq (%d)\n", dmaNum)); - return 0; - } - outw(config, val); - - /* Check if the DMA channel is in use */ - /* Do it a few times in case there is a delay */ - ret = 0; - for (i = 0; i < 3; i++) { - val = inw(config); - if (val & DMA_TEST_PASS) { - ret = 1; - break; - } - } - - /* Clear the Test bit and the DMA bits */ - val = inw(config); - val &= DMA_TEST_BIT_MASK & DMA_MASK; - outw(config, val); - return(ret); -} - -#ifdef AUDIO_DEBUG -void -wss_dump_regs(sc) - struct ad1848_softc *sc; -{ - - printf("WSS reg: status=%02x\n", - (u_char)inb(sc->sc_iobase-WSS_CODEC+WSS_STATUS)); -} - -void -pss_dump_regs(sc) - struct pss_softc *sc; -{ - - printf("PSS regs: status=%04x vers=%04x ", - (u_short)inw(sc->sc_iobase+PSS_STATUS), - (u_short)inw(sc->sc_iobase+PSS_ID_VERS)); - - printf("config=%04x wss_config=%04x\n", - (u_short)inw(sc->sc_iobase+PSS_CONFIG), - (u_short)inw(sc->sc_iobase+PSS_WSS_CONFIG)); -} -#endif - -/* - * Probe for the PSS hardware. - */ -int -pssprobe(parent, self, aux) - struct device *parent; - void *self; - void *aux; -{ - struct pss_softc *sc = self; - struct isa_attach_args *ia = aux; - int iobase = ia->ia_iobase; - - if (!PSS_BASE_VALID(iobase)) { - DPRINTF(("pss: configured iobase %x invalid\n", iobase)); - return 0; - } - - /* Need to probe for iobase when IOBASEUNK {0x220 0x240} */ - if (iobase == IOBASEUNK) { - - iobase = 0x220; - if ((inw(iobase+PSS_ID_VERS) & 0xff00) == 0x4500) - goto pss_found; - - iobase = 0x240; - if ((inw(iobase+PSS_ID_VERS) & 0xff00) == 0x4500) - goto pss_found; - - DPRINTF(("pss: no PSS found (at 0x220 or 0x240)\n")); - return 0; - } - else if ((inw(iobase+PSS_ID_VERS) & 0xff00) != 0x4500) { - DPRINTF(("pss: not a PSS - %x\n", inw(iobase+PSS_ID_VERS))); - return 0; - } - -pss_found: - sc->sc_iobase = iobase; - - /* Clear WSS config */ - pss_setaddr(WSS_BASE_ADDRESS, sc->sc_iobase+PSS_WSS_CONFIG); /* XXX! */ - outb(WSS_BASE_ADDRESS+WSS_CONFIG, 0); - - /* Clear config registers (POR reset state) */ - outw(sc->sc_iobase+PSS_CONFIG, 0); - outw(sc->sc_iobase+PSS_WSS_CONFIG, 0); - outw(sc->sc_iobase+SB_CONFIG, 0); - outw(sc->sc_iobase+MIDI_CONFIG, 0); - outw(sc->sc_iobase+CD_CONFIG, 0); - - if (ia->ia_irq == IRQUNK) { - int i; - for (i = 0; i < 16; i++) { - if (pss_testirq(sc, i) != 0) - break; - } - if (i == 16) { - DPRINTF(("pss: unable to locate free IRQ channel\n")); - return 0; - } - else { - ia->ia_irq = i; - DPRINTF(("pss: found IRQ %d free\n", i)); - } - } - else { - if (pss_testirq(sc, ia->ia_irq) == 0) { - DPRINTF(("pss: configured IRQ unavailable (%d)\n", ia->ia_irq)); - return 0; - } - } - - /* XXX Need to deal with DRQUNK */ - if (pss_testdma(sc, ia->ia_drq) == 0) { - DPRINTF(("pss: configured DMA channel unavailable (%d)\n", ia->ia_drq)); - return 0; - } - - ia->ia_iosize = PSS_NPORT; - - /* Initialize PSS irq and dma */ - pss_setint(ia->ia_irq, sc->sc_iobase+PSS_CONFIG); - pss_setdma(sc->sc_drq, sc->sc_iobase+PSS_CONFIG); - - return 1; -} - -/* - * Probe for the Soundport (ad1848) - */ -int -spprobe(parent, match, aux) - struct device *parent; - void *match, *aux; -{ - struct ad1848_softc *sc = match; - struct pss_softc *pc = (void *) parent; - struct cfdata *cf = (void *)sc->sc_dev.dv_cfdata; - struct isa_attach_args *ia = aux; - u_char bits; - int i; - - sc->sc_iot = ia->ia_iot; - sc->sc_iobase = cf->cf_iobase + WSS_CODEC; - - /* Set WSS io address */ - pss_setaddr(cf->cf_iobase, pc->sc_iobase+PSS_WSS_CONFIG); - - /* Is there an ad1848 chip at the WSS iobase ? */ - if (ad1848_probe(sc) == 0) { - DPRINTF(("sp: no ad1848 ? iobase=%x\n", sc->sc_iobase)); - return 0; - } - - /* Setup WSS interrupt and DMA if auto */ - if (cf->cf_irq == IRQUNK) { - - /* Find unused IRQ for WSS */ - for (i = 0; i < 12; i++) { - if (wss_interrupt_bits[i] != 0xff) { - if (pss_testirq(pc, i)) - break; - } - } - if (i == 12) { - DPRINTF(("sp: unable to locate free IRQ for WSS\n")); - return 0; - } - else { - cf->cf_irq = i; - sc->sc_irq = i; - DPRINTF(("sp: found IRQ %d free\n", i)); - } - } - else { - sc->sc_irq = cf->cf_irq; - if (pss_testirq(pc, sc->sc_irq) == 0) { - DPRINTF(("sp: configured IRQ unavailable (%d)\n", sc->sc_irq)); - return 0; - } - } - - if (cf->cf_drq == DRQUNK) { - /* Find unused DMA channel for WSS */ - for (i = 0; i < 4; i++) { - if (wss_dma_bits[i]) { - if (pss_testdma(pc, i)) - break; - } - } - if (i == 4) { - DPRINTF(("sp: unable to locate free DMA channel for WSS\n")); - return 0; - } - else { - sc->sc_drq = cf->cf_drq = i; - DPRINTF(("sp: found DMA %d free\n", i)); - } - } - else { - if (pss_testdma(pc, sc->sc_drq) == 0) { - DPRINTF(("sp: configured DMA channel unavailable (%d)\n", sc->sc_drq)); - return 0; - } - sc->sc_drq = cf->cf_drq; - } - sc->sc_recdrq = sc->sc_drq; - - /* Set WSS config registers */ - if ((bits = wss_interrupt_bits[sc->sc_irq]) == 0xff) { - DPRINTF(("sp: invalid interrupt configuration (irq=%d)\n", sc->sc_irq)); - return 0; - } - - outb(sc->sc_iobase+WSS_CONFIG, (bits | 0x40)); - if ((inb(sc->sc_iobase+WSS_STATUS) & 0x40) == 0) /* XXX What do these bits mean ? */ - DPRINTF(("sp: IRQ %x\n", inb(sc->sc_iobase+WSS_STATUS))); - - outb(sc->sc_iobase+WSS_CONFIG, (bits | wss_dma_bits[sc->sc_drq])); - - pc->ad1848_sc = sc; - sc->parent = pc; - - return 1; -} - -/* - * Attach hardware to driver, attach hardware driver to audio - * pseudo-device driver . - */ -void -pssattach(parent, self, aux) - struct device *parent, *self; - void *aux; -{ - struct pss_softc *sc = (struct pss_softc *)self; - struct isa_attach_args *ia = (struct isa_attach_args *)aux; - int iobase = ia->ia_iobase; - u_char vers; - struct ad1848_volume vol = {150, 150}; - - sc->sc_iobase = iobase; - sc->sc_drq = ia->ia_drq; - - /* Setup interrupt handler for PSS */ - sc->sc_ih = isa_intr_establish(ia->ia_ic, ia->ia_irq, IST_EDGE, IPL_AUDIO, - pssintr, sc, sc->sc_dev.dv_xname); - - vers = (inw(sc->sc_iobase+PSS_ID_VERS)&0xff) - 1; - printf(": ESC614%c\n", (vers > 0)?'A'+vers:' '); - - (void)config_found(self, ia->ia_ic, NULL); /* XXX */ - - sc->out_port = PSS_MASTER_VOL; - - (void)pss_set_master_mode(sc, PSS_SPKR_STEREO); - (void)pss_set_master_gain(sc, &vol); - (void)pss_set_treble(sc, AUDIO_MAX_GAIN/2); - (void)pss_set_bass(sc, AUDIO_MAX_GAIN/2); - - audio_attach_mi(&pss_audio_if, sc->ad1848_sc, &sc->ad1848_sc->sc_dev); -} - -void -spattach(parent, self, aux) - struct device *parent, *self; - void *aux; -{ - struct ad1848_softc *sc = (struct ad1848_softc *)self; - struct cfdata *cf = (void *)sc->sc_dev.dv_cfdata; - isa_chipset_tag_t ic = aux; /* XXX */ - int iobase = cf->cf_iobase; - - sc->sc_iobase = iobase; - sc->sc_drq = cf->cf_drq; - - sc->sc_ih = isa_intr_establish(ic, cf->cf_irq, IST_EDGE, IPL_AUDIO, - ad1848_intr, sc, sc->sc_dev.dv_xname); - - sc->sc_isa = parent->dv_parent; - - ad1848_attach(sc); - - printf("\n"); -} - -int -pss_set_master_gain(sc, gp) - struct pss_softc *sc; - struct ad1848_volume *gp; -{ - DPRINTF(("pss_set_master_gain: %d:%d\n", gp->left, gp->right)); - -#ifdef PSS_DSP - if (gp->left > PHILLIPS_VOL_MAX) - gp->left = PHILLIPS_VOL_MAX; - if (gp->left < PHILLIPS_VOL_MIN) - gp->left = PHILLIPS_VOL_MIN; - if (gp->right > PHILLIPS_VOL_MAX) - gp->right = PHILLIPS_VOL_MAX; - if (gp->right < PHILLIPS_VOL_MIN) - gp->right = PHILLIPS_VOL_MIN; - - pss_dspwrite(sc, SET_MASTER_COMMAND); - pss_dspwrite(sc, MASTER_VOLUME_LEFT|(PHILLIPS_VOL_CONSTANT + gp->left / PHILLIPS_VOL_STEP)); - pss_dspwrite(sc, SET_MASTER_COMMAND); - pss_dspwrite(sc, MASTER_VOLUME_RIGHT|(PHILLIPS_VOL_CONSTANT + gp->right / PHILLIPS_VOL_STEP)); -#endif - - sc->master_volume = *gp; - return(0); -} - -int -pss_set_master_mode(sc, mode) - struct pss_softc *sc; - int mode; -{ - short phillips_mode; - - DPRINTF(("pss_set_master_mode: %d\n", mode)); - - if (mode == PSS_SPKR_STEREO) - phillips_mode = PSS_STEREO; - else if (mode == PSS_SPKR_PSEUDO) - phillips_mode = PSS_PSEUDO; - else if (mode == PSS_SPKR_SPATIAL) - phillips_mode = PSS_SPATIAL; - else if (mode == PSS_SPKR_MONO) - phillips_mode = PSS_MONO; - else - return (EINVAL); - -#ifdef PSS_DSP - pss_dspwrite(sc, SET_MASTER_COMMAND); - pss_dspwrite(sc, MASTER_SWITCH | mode); -#endif - - sc->master_mode = mode; - - return(0); -} - -int -pss_set_treble(sc, treb) - struct pss_softc *sc; - u_int treb; -{ - DPRINTF(("pss_set_treble: %d\n", treb)); - -#ifdef PSS_DSP - if (treb > PHILLIPS_TREBLE_MAX) - treb = PHILLIPS_TREBLE_MAX; - if (treb < PHILLIPS_TREBLE_MIN) - treb = PHILLIPS_TREBLE_MIN; - pss_dspwrite(sc, SET_MASTER_COMMAND); - pss_dspwrite(sc, MASTER_TREBLE|(PHILLIPS_TREBLE_CONSTANT + treb / PHILLIPS_TREBLE_STEP)); -#endif - - sc->monitor_treble = treb; - - return(0); -} - -int -pss_set_bass(sc, bass) - struct pss_softc *sc; - u_int bass; -{ - DPRINTF(("pss_set_bass: %d\n", bass)); - -#ifdef PSS_DSP - if (bass > PHILLIPS_BASS_MAX) - bass = PHILLIPS_BASS_MAX; - if (bass < PHILLIPS_BASS_MIN) - bass = PHILLIPS_BASS_MIN; - pss_dspwrite(sc, SET_MASTER_COMMAND); - pss_dspwrite(sc, MASTER_BASS|(PHILLIPS_BASS_CONSTANT + bass / PHILLIPS_BASS_STEP)); -#endif - - sc->monitor_bass = bass; - - return(0); -} - -int -pss_get_master_gain(sc, gp) - struct pss_softc *sc; - struct ad1848_volume *gp; -{ - *gp = sc->master_volume; - return(0); -} - -int -pss_get_master_mode(sc, mode) - struct pss_softc *sc; - u_int *mode; -{ - *mode = sc->master_mode; - return(0); -} - -int -pss_get_treble(sc, tp) - struct pss_softc *sc; - u_char *tp; -{ - *tp = sc->monitor_treble; - return(0); -} - -int -pss_get_bass(sc, bp) - struct pss_softc *sc; - u_char *bp; -{ - *bp = sc->monitor_bass; - return(0); -} - -int -pss_speaker_ctl(addr, newstate) - void *addr; - int newstate; -{ - return(0); -} - -int -pssintr(arg) - void *arg; -{ - struct pss_softc *sc = arg; - u_short sr; - - sr = inw(sc->sc_iobase+PSS_STATUS); - - DPRINTF(("pssintr: sc=%p st=%x\n", sc, sr)); - - /* Acknowledge intr */ - outw(sc->sc_iobase+PSS_IRQ_ACK, 0); - - /* Is it one of ours ? */ - if (sr & (PSS_WRITE_EMPTY|PSS_READ_FULL|PSS_IRQ|PSS_DMQ_TC)) { - /* XXX do something */ - return 1; - } - - return 0; -} - -int -pss_getdev(addr, retp) - void *addr; - struct audio_device *retp; -{ - DPRINTF(("pss_getdev: retp=%p\n", retp)); - - *retp = pss_device; - return 0; -} - -static ad1848_devmap_t mappings[] = { -{ PSS_MIC_IN_LVL, AD1848_KIND_LVL, AD1848_AUX2_CHANNEL }, -{ PSS_LINE_IN_LVL, AD1848_KIND_LVL, AD1848_AUX1_CHANNEL }, -{ PSS_DAC_LVL, AD1848_KIND_LVL, AD1848_DAC_CHANNEL }, -{ PSS_MON_LVL, AD1848_KIND_LVL, AD1848_MONO_CHANNEL }, -{ PSS_MIC_IN_MUTE, AD1848_KIND_MUTE, AD1848_AUX2_CHANNEL }, -{ PSS_LINE_IN_MUTE, AD1848_KIND_MUTE, AD1848_AUX1_CHANNEL }, -{ PSS_DAC_MUTE, AD1848_KIND_MUTE, AD1848_DAC_CHANNEL }, -{ PSS_REC_LVL, AD1848_KIND_RECORDGAIN, -1 }, -{ PSS_RECORD_SOURCE, AD1848_KIND_RECORDSOURCE, -1} -}; - -static int nummap = sizeof(mappings) / sizeof(mappings[0]); - -int -pss_mixer_set_port(addr, cp) - void *addr; - mixer_ctrl_t *cp; -{ - struct ad1848_softc *ac = addr; - struct pss_softc *sc = ac->parent; - struct ad1848_volume vol; - int error = ad1848_mixer_set_port(ac, mappings, nummap, cp); - - if (error != ENXIO) - return (error); - - switch (cp->dev) { - case PSS_MASTER_VOL: /* master volume */ - if (cp->type == AUDIO_MIXER_VALUE) { - if (ad1848_to_vol(cp, &vol)) - error = pss_set_master_gain(sc, &vol); - } - break; - - case PSS_OUTPUT_MODE: - if (cp->type == AUDIO_MIXER_ENUM) - error = pss_set_master_mode(sc, cp->un.ord); - break; - - case PSS_MASTER_TREBLE: /* master treble */ - if (cp->type == AUDIO_MIXER_VALUE && cp->un.value.num_channels == 1) - error = pss_set_treble(sc, (u_char)cp->un.value.level[AUDIO_MIXER_LEVEL_MONO]); - break; - - case PSS_MASTER_BASS: /* master bass */ - if (cp->type == AUDIO_MIXER_VALUE && cp->un.value.num_channels == 1) - error = pss_set_bass(sc, (u_char)cp->un.value.level[AUDIO_MIXER_LEVEL_MONO]); - break; - - default: - return ENXIO; - /*NOTREACHED*/ - } - - return 0; -} - -int -pss_mixer_get_port(addr, cp) - void *addr; - mixer_ctrl_t *cp; -{ - struct ad1848_softc *ac = addr; - struct pss_softc *sc = ac->parent; - struct ad1848_volume vol; - u_char eq; - int error = ad1848_mixer_get_port(ac, mappings, nummap, cp); - - if (error != ENXIO) - return (error); - - error = EINVAL; - - switch (cp->dev) { - case PSS_MASTER_VOL: /* master volume */ - if (cp->type == AUDIO_MIXER_VALUE) { - error = pss_get_master_gain(sc, &vol); - if (!error) - ad1848_from_vol(cp, &vol); - } - break; - - case PSS_MASTER_TREBLE: /* master treble */ - if (cp->type == AUDIO_MIXER_VALUE && cp->un.value.num_channels == 1) { - error = pss_get_treble(sc, &eq); - if (!error) - cp->un.value.level[AUDIO_MIXER_LEVEL_MONO] = eq; - } - break; - - case PSS_MASTER_BASS: /* master bass */ - if (cp->type == AUDIO_MIXER_VALUE && cp->un.value.num_channels == 1) { - error = pss_get_bass(sc, &eq); - if (!error) - cp->un.value.level[AUDIO_MIXER_LEVEL_MONO] = eq; - } - break; - - case PSS_OUTPUT_MODE: - if (cp->type == AUDIO_MIXER_ENUM) - error = pss_get_master_mode(sc, &cp->un.ord); - break; - - default: - error = ENXIO; - break; - } - - return(error); -} - -int -pss_query_devinfo(addr, dip) - void *addr; - mixer_devinfo_t *dip; -{ - DPRINTF(("pss_query_devinfo: index=%d\n", dip->index)); - - switch(dip->index) { - case PSS_MIC_IN_LVL: /* Microphone */ - dip->type = AUDIO_MIXER_VALUE; - dip->mixer_class = PSS_INPUT_CLASS; - dip->prev = AUDIO_MIXER_LAST; - dip->next = PSS_MIC_IN_MUTE; - strlcpy(dip->label.name, AudioNmicrophone, sizeof dip->label.name); - dip->un.v.num_channels = 2; - strlcpy(dip->un.v.units.name, AudioNvolume, - sizeof dip->un.v.units.name); - break; - - case PSS_LINE_IN_LVL: /* line/CD */ - dip->type = AUDIO_MIXER_VALUE; - dip->mixer_class = PSS_INPUT_CLASS; - dip->prev = AUDIO_MIXER_LAST; - dip->next = PSS_LINE_IN_MUTE; - strlcpy(dip->label.name, AudioNcd, sizeof dip->label.name); - dip->un.v.num_channels = 2; - strlcpy(dip->un.v.units.name, AudioNvolume, - sizeof dip->un.v.units.name); - break; - - case PSS_DAC_LVL: /* dacout */ - dip->type = AUDIO_MIXER_VALUE; - dip->mixer_class = PSS_INPUT_CLASS; - dip->prev = AUDIO_MIXER_LAST; - dip->next = PSS_DAC_MUTE; - strlcpy(dip->label.name, AudioNdac, sizeof dip->label.name); - dip->un.v.num_channels = 2; - strlcpy(dip->un.v.units.name, AudioNvolume, - sizeof dip->un.v.units.name); - break; - - case PSS_REC_LVL: /* record level */ - dip->type = AUDIO_MIXER_VALUE; - dip->mixer_class = PSS_RECORD_CLASS; - dip->prev = AUDIO_MIXER_LAST; - dip->next = PSS_RECORD_SOURCE; - strlcpy(dip->label.name, AudioNrecord, sizeof dip->label.name); - dip->un.v.num_channels = 2; - strlcpy(dip->un.v.units.name, AudioNvolume, - sizeof dip->un.v.units.name); - break; - - case PSS_MON_LVL: /* monitor level */ - dip->type = AUDIO_MIXER_VALUE; - dip->mixer_class = PSS_MONITOR_CLASS; - dip->next = dip->prev = AUDIO_MIXER_LAST; - strlcpy(dip->label.name, AudioNmonitor, sizeof dip->label.name); - dip->un.v.num_channels = 1; - strlcpy(dip->un.v.units.name, AudioNvolume, - sizeof dip->un.v.units.name); - break; - - case PSS_MASTER_VOL: /* master volume */ - dip->type = AUDIO_MIXER_VALUE; - dip->mixer_class = PSS_OUTPUT_CLASS; - dip->prev = AUDIO_MIXER_LAST; - dip->next = PSS_OUTPUT_MODE; - strlcpy(dip->label.name, AudioNmaster, sizeof dip->label.name); - dip->un.v.num_channels = 2; - strlcpy(dip->un.v.units.name, AudioNvolume, - sizeof dip->un.v.units.name); - break; - - case PSS_MASTER_TREBLE: /* master treble */ - dip->type = AUDIO_MIXER_VALUE; - dip->mixer_class = PSS_OUTPUT_CLASS; - dip->next = dip->prev = AUDIO_MIXER_LAST; - strlcpy(dip->label.name, AudioNtreble, sizeof dip->label.name); - dip->un.v.num_channels = 1; - strlcpy(dip->un.v.units.name, AudioNtreble, - sizeof dip->un.v.units.name); - break; - - case PSS_MASTER_BASS: /* master bass */ - dip->type = AUDIO_MIXER_VALUE; - dip->mixer_class = PSS_OUTPUT_CLASS; - dip->next = dip->prev = AUDIO_MIXER_LAST; - strlcpy(dip->label.name, AudioNbass, sizeof dip->label.name); - dip->un.v.num_channels = 1; - strlcpy(dip->un.v.units.name, AudioNbass, sizeof dip->un.v.units.name); - break; - - case PSS_OUTPUT_CLASS: /* output class descriptor */ - dip->type = AUDIO_MIXER_CLASS; - dip->mixer_class = PSS_OUTPUT_CLASS; - dip->next = dip->prev = AUDIO_MIXER_LAST; - strlcpy(dip->label.name, AudioCoutputs, sizeof dip->label.name); - break; - - case PSS_INPUT_CLASS: /* input class descriptor */ - dip->type = AUDIO_MIXER_CLASS; - dip->mixer_class = PSS_INPUT_CLASS; - dip->next = dip->prev = AUDIO_MIXER_LAST; - strlcpy(dip->label.name, AudioCinputs, sizeof dip->label.name); - break; - - case PSS_MONITOR_CLASS: /* monitor class descriptor */ - dip->type = AUDIO_MIXER_CLASS; - dip->mixer_class = PSS_MONITOR_CLASS; - dip->next = dip->prev = AUDIO_MIXER_LAST; - strlcpy(dip->label.name, AudioCmonitor, sizeof dip->label.name); - break; - - case PSS_RECORD_CLASS: /* record source class */ - dip->type = AUDIO_MIXER_CLASS; - dip->mixer_class = PSS_RECORD_CLASS; - dip->next = dip->prev = AUDIO_MIXER_LAST; - strlcpy(dip->label.name, AudioCrecord, sizeof dip->label.name); - break; - - case PSS_MIC_IN_MUTE: - dip->mixer_class = PSS_INPUT_CLASS; - dip->type = AUDIO_MIXER_ENUM; - dip->prev = PSS_MIC_IN_LVL; - dip->next = AUDIO_MIXER_LAST; - goto mute; - - case PSS_LINE_IN_MUTE: - dip->mixer_class = PSS_INPUT_CLASS; - dip->type = AUDIO_MIXER_ENUM; - dip->prev = PSS_LINE_IN_LVL; - dip->next = AUDIO_MIXER_LAST; - goto mute; - - case PSS_DAC_MUTE: - dip->mixer_class = PSS_INPUT_CLASS; - dip->type = AUDIO_MIXER_ENUM; - dip->prev = PSS_DAC_LVL; - dip->next = AUDIO_MIXER_LAST; - mute: - strlcpy(dip->label.name, AudioNmute, sizeof dip->label.name); - dip->un.e.num_mem = 2; - strlcpy(dip->un.e.member[0].label.name, AudioNoff, - sizeof dip->un.e.member[0].label.name); - dip->un.e.member[0].ord = 0; - strlcpy(dip->un.e.member[1].label.name, AudioNon, - sizeof dip->un.e.member[1].label.name); - dip->un.e.member[1].ord = 1; - break; - - case PSS_OUTPUT_MODE: - dip->mixer_class = PSS_OUTPUT_CLASS; - dip->type = AUDIO_MIXER_ENUM; - dip->prev = PSS_MASTER_VOL; - dip->next = AUDIO_MIXER_LAST; - strlcpy(dip->label.name, AudioNmode, sizeof dip->label.name); - dip->un.e.num_mem = 4; - strlcpy(dip->un.e.member[0].label.name, AudioNmono, - sizeof dip->un.e.member[0].label.name); - dip->un.e.member[0].ord = PSS_SPKR_MONO; - strlcpy(dip->un.e.member[1].label.name, AudioNstereo, - sizeof dip->un.e.member[1].label.name); - dip->un.e.member[1].ord = PSS_SPKR_STEREO; - strlcpy(dip->un.e.member[2].label.name, AudioNpseudo, - sizeof dip->un.e.member[2].label.name); - dip->un.e.member[2].ord = PSS_SPKR_PSEUDO; - strlcpy(dip->un.e.member[3].label.name, AudioNspatial, - sizeof dip->un.e.member[3].label.name); - dip->un.e.member[3].ord = PSS_SPKR_SPATIAL; - break; - - case PSS_RECORD_SOURCE: - dip->mixer_class = PSS_RECORD_CLASS; - dip->type = AUDIO_MIXER_ENUM; - dip->prev = PSS_REC_LVL; - dip->next = AUDIO_MIXER_LAST; - strlcpy(dip->label.name, AudioNsource, sizeof dip->label.name); - dip->un.e.num_mem = 3; - strlcpy(dip->un.e.member[0].label.name, AudioNmicrophone, - sizeof dip->un.e.member[0].label.name); - dip->un.e.member[0].ord = PSS_MIC_IN_LVL; - strlcpy(dip->un.e.member[1].label.name, AudioNcd, - sizeof dip->un.e.member[1].label.name); - dip->un.e.member[1].ord = PSS_LINE_IN_LVL; - strlcpy(dip->un.e.member[2].label.name, AudioNdac, - sizeof dip->un.e.member[2].label.name); - dip->un.e.member[2].ord = PSS_DAC_LVL; - break; - - default: - return ENXIO; - /*NOTREACHED*/ - } - DPRINTF(("AUDIO_MIXER_DEVINFO: name=%s\n", dip->label.name)); - - return 0; -} diff --git a/sys/dev/isa/pssreg.h b/sys/dev/isa/pssreg.h deleted file mode 100644 index 4899c184ff6..00000000000 --- a/sys/dev/isa/pssreg.h +++ /dev/null @@ -1,165 +0,0 @@ -/* $OpenBSD: pssreg.h,v 1.3 2007/10/26 15:00:49 martin Exp $ */ -/* $NetBSD: pssreg.h,v 1.2 1995/05/08 22:02:09 brezak Exp $ */ - -/* - * Copyright (c) 1994 John Brezak - * Copyright (c) 1991-1993 Regents of the University of California. - * All rights reserved. - * - * Redistribution and use in source and binary forms, with or without - * modification, are permitted provided that the following conditions - * are met: - * 1. Redistributions of source code must retain the above copyright - * notice, this list of conditions and the following disclaimer. - * 2. Redistributions in binary form must reproduce the above copyright - * notice, this list of conditions and the following disclaimer in the - * documentation and/or other materials provided with the distribution. - * 3. All advertising materials mentioning features or use of this software - * must display the following acknowledgement: - * This product includes software developed by the Computer Systems - * Engineering Group at Lawrence Berkeley Laboratory. - * 4. Neither the name of the University nor of the Laboratory may be used - * to endorse or promote products derived from this software without - * specific prior written permission. - * - * THIS SOFTWARE IS PROVIDED BY THE REGENTS AND CONTRIBUTORS ``AS IS'' AND - * ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE - * IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE - * ARE DISCLAIMED. IN NO EVENT SHALL THE REGENTS OR CONTRIBUTORS BE LIABLE - * FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, EXEMPLARY, OR CONSEQUENTIAL - * DAMAGES (INCLUDING, BUT NOT LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS - * OR SERVICES; LOSS OF USE, DATA, OR PROFITS; OR BUSINESS INTERRUPTION) - * HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT - * LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY - * OUT OF THE USE OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF - * SUCH DAMAGE. - * - */ -/* - * Copyright (c) 1993 Analog Devices Inc. All rights reserved - */ - -/* - * Macros to detect valid hardware configuration data. - */ -#define PSS_BASE_VALID(base) ((base) == 0x220 || (base) == 0x240) - -/* - * ESC614 Interface chip - */ -#define ADDR_MASK 0x003f - -#define INT_MASK 0xffc7 -#define INT_3_BITS 0x0008 -#define INT_5_BITS 0x0010 -#define INT_7_BITS 0x0018 -#define INT_9_BITS 0x0020 -#define INT_10_BITS 0x0028 -#define INT_11_BITS 0x0030 -#define INT_12_BITS 0x0038 - -#define INT_TEST_BIT 0x0200 -#define INT_TEST_PASS 0x0100 -#define INT_TEST_BIT_MASK 0xFDFF - -#define DMA_MASK 0xfff8 -#define DMA_0_BITS 0x0001 -#define DMA_1_BITS 0x0002 -#define DMA_3_BITS 0x0003 -#define DMA_5_BITS 0x0004 -#define DMA_6_BITS 0x0005 -#define DMA_7_BITS 0x0006 - -#define DMA_TEST_BIT 0x0080 -#define DMA_TEST_PASS 0x0040 -#define DMA_TEST_BIT_MASK 0xFF7F - -/* Echo DSP Flags */ -#define DSP_FLAG3 0x10 -#define DSP_FLAG2 0x08 -#define DSP_FLAG1 0x80 -#define DSP_FLAG0 0x40 - -/* ESC614 register offsets */ -#define PSS_NPORT 32 - -#define PSS_DATA 0x00 -#define PSS_STATUS 0x02 -#define PSS_CONTROL 0x02 -#define PSS_ID_VERS 0x04 -#define PSS_IRQ_ACK 0x04 - -#define PSS_CONFIG 0x10 -#define PSS_WSS_CONFIG 0x12 -#define SB_CONFIG 0x14 -#define CD_CONFIG 0x16 -#define MIDI_CONFIG 0x18 -#define UART_CONFIG 0x1a - -/* PSS control register */ -#define PSS_WEIE 0x8000 -#define PSS_RFIE 0x4000 -#define PSS_RESET 0x2000 -#define PSS_FLAG1 0x1000 -#define PSS_FLAG0 0x0800 - -/* PSS status register */ -#define PSS_WRITE_EMPTY 0x8000 -#define PSS_READ_FULL 0x4000 -#define PSS_IRQ 0x2000 -#define PSS_DMQ_TC 0x1000 -#define PSS_FLAG3 0x0800 -#define PSS_FLAG2 0x0400 - -/* Game control register */ -#define GAME_BIT 0x0400 -#define GAME_BIT_MASK 0xfbff - -/* MPU registers */ -#define MIDI_NPORT 8 - -#define MIDI_DATA_REG 0x00 -#define MIDI_STATUS_REG 0x01 -#define MIDI_COMMAND_REG 0x01 - -#define MIDI_SR_RF 0x80 -#define MIDI_SR_TE 0x40 - -/* CD Interface registers */ -#define CD_NPORT 16 - -#define CD_POL_MASK 0xFFBF -#define CD_POL_BIT 0x0040 - -/* Philips amplifier controls: only via DSP */ -/* DSP commands */ -#define SET_MASTER_COMMAND 0x0010 -#define MASTER_VOLUME_LEFT 0x0000 -#define MASTER_VOLUME_RIGHT 0x0100 -#define MASTER_BASS 0x0200 -#define MASTER_TREBLE 0x0300 -#define MASTER_SWITCH 0x0800 - -#define PSS_STEREO 0x00ce -#define PSS_PSEUDO 0x00d6 -#define PSS_SPATIAL 0x00de -#define PSS_MONO 0x00c6 - -#define PHILLIPS_VOL_MIN -64 -#define PHILLIPS_VOL_MAX 6 -#define PHILLIPS_VOL_DELTA 70 -#define PHILLIPS_VOL_INITIAL -20 -#define PHILLIPS_VOL_CONSTANT 252 -#define PHILLIPS_VOL_STEP 2 -#define PHILLIPS_BASS_MIN -12 -#define PHILLIPS_BASS_MAX 15 -#define PHILLIPS_BASS_DELTA 27 -#define PHILLIPS_BASS_INITIAL 0 -#define PHILLIPS_BASS_CONSTANT 246 -#define PHILLIPS_BASS_STEP 2 -#define PHILLIPS_TREBLE_MIN -12 -#define PHILLIPS_TREBLE_MAX 12 -#define PHILLIPS_TREBLE_DELTA 24 -#define PHILLIPS_TREBLE_INITIAL 0 -#define PHILLIPS_TREBLE_CONSTANT 246 -#define PHILLIPS_TREBLE_STEP 2 |