File: [cvs.NetBSD.org] / src / sys / dev / isa / gus.c (download)
Revision 1.41, Sun Aug 24 22:31:33 1997 UTC (26 years, 8 months ago) by augustss
Branch: MAIN
Changes since 1.40: +22 -25
lines
- Change audio_hw_if a little: set_param now sets the play and record modes
at the same time instead by using two different calls. This enables
it to check more easily if the combined mode is all right.
- Improve the error checking in audio.c.
- Add a new audio property, AUDIO_PROP_INDEPENDENT, show if the
play and record settings are independent.
- Fix some buglets in audio.c.
|
/* $NetBSD: gus.c,v 1.41 1997/08/24 22:31:33 augustss Exp $ */
/*-
* Copyright (c) 1996 The NetBSD Foundation, Inc.
* All rights reserved.
*
* This code is derived from software contributed to The NetBSD Foundation
* by Ken Hornstein and John Kohl.
*
* 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 NetBSD
* Foundation, Inc. and its contributors.
* 4. Neither the name of The NetBSD Foundation nor the names of its
* contributors may be used to endorse or promote products derived
* from this software without specific prior written permission.
*
* THIS SOFTWARE IS PROVIDED BY THE NETBSD FOUNDATION, INC. 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.
*/
/*
*
* TODO:
* . figure out why mixer activity while sound is playing causes problems
* (phantom interrupts?)
* . figure out a better deinterleave strategy that avoids sucking up
* CPU, memory and cache bandwidth. (Maybe a special encoding?
* Maybe use the double-speed sampling/hardware deinterleave trick
* from the GUS SDK?) A 486/33 isn't quite fast enough to keep
* up with 44.1kHz 16-bit stereo output without some drop-outs.
* . use CS4231 for 16-bit sampling, for a-law and mu-law playback.
* . actually test full-duplex sampling(recording) and playback.
*/
/*
* Gravis UltraSound driver
*
* For more detailed information, see the GUS developers' kit
* available on the net at:
*
* ftp://freedom.nmsu.edu/pub/ultrasound/gravis/util/
* gusdkXXX.zip (developers' kit--get rev 2.22 or later)
* See ultrawrd.doc inside--it's MS Word (ick), but it's the bible
*
*/
/*
* The GUS Max has a slightly strange set of connections between the CS4231
* and the GF1 and the DMA interconnects. It's set up so that the CS4231 can
* be playing while the GF1 is loading patches from the system.
*
* Here's a recreation of the DMA interconnect diagram:
*
* GF1
* +---------+ digital
* | | record ASIC
* | |--------------+
* | | | +--------+
* | | play (dram) | +----+ | |
* | |--------------(------|-\ | | +-+ |
* +---------+ | | >-|----|---|C|--|------ dma chan 1
* | +---|-/ | | +-+ |
* | | +----+ | | |
* | | +----+ | | |
* +---------+ +-+ +--(---|-\ | | | |
* | | play |8| | | >-|----|----+---|------ dma chan 2
* | ---C----|--------|/|------(---|-/ | | |
* | ^ |record |1| | +----+ | |
* | | | /----|6|------+ +--------+
* | ---+----|--/ +-+
* +---------+
* CS4231 8-to-16 bit bus conversion, if needed
*
*
* "C" is an optional combiner.
*
*/
#include "gus.h"
#if NGUS > 0
#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 <sys/fcntl.h>
#include <sys/malloc.h>
#include <sys/kernel.h>
#include <machine/cpu.h>
#include <machine/intr.h>
#include <machine/bus.h>
#include <machine/pio.h>
#include <machine/cpufunc.h>
#include <sys/audioio.h>
#include <dev/audio_if.h>
#include <dev/mulaw.h>
#include <dev/auconv.h>
#include <dev/isa/isavar.h>
#include <dev/isa/isadmavar.h>
#include <i386/isa/icu.h>
#include <dev/ic/ics2101reg.h>
#include <dev/ic/cs4231reg.h>
#include <dev/ic/ad1848reg.h>
#include <dev/isa/ics2101var.h>
#include <dev/isa/ad1848var.h>
#include <dev/isa/cs4231var.h>
#include "gusreg.h"
#ifdef AUDIO_DEBUG
#define STATIC /* empty; for debugging symbols */
#else
#define STATIC static
#endif
/*
* Software state of a single "voice" on the GUS
*/
struct gus_voice {
/*
* Various control bits
*/
unsigned char voccntl; /* State of voice control register */
unsigned char volcntl; /* State of volume control register */
unsigned char pan_pos; /* Position of volume panning (4 bits) */
int rate; /* Sample rate of voice being played back */
/*
* Address of the voice data into the GUS's DRAM. 20 bits each
*/
u_long start_addr; /* Starting address of voice data loop area */
u_long end_addr; /* Ending address of voice data loop */
u_long current_addr; /* Beginning address of voice data
(start playing here) */
/*
* linear volume values for the GUS's volume ramp. 0-511 (9 bits).
* These values must be translated into the logarithmic values using
* gus_log_volumes[]
*/
int start_volume; /* Starting position of volume ramp */
int current_volume; /* Current position of volume on volume ramp */
int end_volume; /* Ending position of volume on volume ramp */
};
/*
* Software state of GUS
*/
struct gus_softc {
struct device sc_dev; /* base device */
struct isadev sc_id; /* ISA device */
void *sc_ih; /* interrupt vector */
bus_space_tag_t sc_iot; /* tag */
bus_space_handle_t sc_ioh; /* handle */
int sc_iobase; /* I/O base address */
int sc_irq; /* IRQ used */
int sc_drq; /* DMA channel for play */
int sc_recdrq; /* DMA channel for recording */
int sc_flags; /* Various flags about the GUS */
#define GUS_MIXER_INSTALLED 0x01 /* An ICS mixer is installed */
#define GUS_LOCKED 0x02 /* GUS is busy doing multi-phase DMA */
#define GUS_CODEC_INSTALLED 0x04 /* CS4231 installed/MAX */
#define GUS_PLAYING 0x08 /* GUS is playing a voice */
#define GUS_DMAOUT_ACTIVE 0x10 /* GUS is busy doing audio DMA */
#define GUS_DMAIN_ACTIVE 0x20 /* GUS is busy sampling */
#define GUS_OPEN 0x100 /* GUS is open */
int sc_dsize; /* Size of GUS DRAM */
int sc_voices; /* Number of active voices */
u_char sc_revision; /* Board revision of GUS */
u_char sc_mixcontrol; /* Value of GUS_MIX_CONTROL register */
u_long sc_orate; /* Output sampling rate */
u_long sc_irate; /* Input sampling rate */
int sc_encoding; /* Current data encoding type */
int sc_precision; /* # of bits of precision */
int sc_channels; /* Number of active channels */
int sc_blocksize; /* Current blocksize */
int sc_chanblocksize; /* Current blocksize for each in-use
channel */
short sc_nbufs; /* how many on-GUS bufs per-channel */
short sc_bufcnt; /* how many need to be played */
void *sc_deintr_buf; /* deinterleave buffer for stereo */
int sc_ogain; /* Output gain control */
u_char sc_out_port; /* Current out port (generic only) */
u_char sc_in_port; /* keep track of it when no codec */
void (*sc_dmaoutintr) __P((void*)); /* DMA completion intr handler */
void *sc_outarg; /* argument for sc_dmaoutintr() */
u_char *sc_dmaoutaddr; /* for isa_dmadone */
u_long sc_gusaddr; /* where did we just put it? */
int sc_dmaoutcnt; /* for isa_dmadone */
void (*sc_dmainintr) __P((void*)); /* DMA completion intr handler */
void *sc_inarg; /* argument for sc_dmaoutintr() */
u_char *sc_dmainaddr; /* for isa_dmadone */
int sc_dmaincnt; /* for isa_dmadone */
struct stereo_dma_intr {
void (*intr)__P((void *));
void *arg;
u_char *buffer;
u_long dmabuf;
int size;
int flags;
} sc_stereo;
/*
* State information for linear audio layer
*/
int sc_dmabuf; /* Which ring buffer we're DMA'ing to */
int sc_playbuf; /* Which ring buffer we're playing */
/*
* Voice information array. All voice-specific information is stored
* here
*/
struct gus_voice sc_voc[32]; /* Voice data for each voice */
union {
struct ics2101_softc sc_mixer_u;
struct ad1848_softc sc_codec_u;
} u;
#define sc_mixer u.sc_mixer_u
#define sc_codec u.sc_codec_u
};
struct ics2101_volume {
u_char left;
u_char right;
};
#define HAS_CODEC(sc) ((sc)->sc_flags & GUS_CODEC_INSTALLED)
#define HAS_MIXER(sc) ((sc)->sc_flags & GUS_MIXER_INSTALLED)
/*
* Mixer devices for ICS2101
*/
/* MIC IN mute, line in mute, line out mute are first since they can be done
even if no ICS mixer. */
#define GUSICS_MIC_IN_MUTE 0
#define GUSICS_LINE_IN_MUTE 1
#define GUSICS_MASTER_MUTE 2
#define GUSICS_CD_MUTE 3
#define GUSICS_DAC_MUTE 4
#define GUSICS_MIC_IN_LVL 5
#define GUSICS_LINE_IN_LVL 6
#define GUSICS_CD_LVL 7
#define GUSICS_DAC_LVL 8
#define GUSICS_MASTER_LVL 9
#define GUSICS_RECORD_SOURCE 10
/* Classes */
#define GUSICS_INPUT_CLASS 11
#define GUSICS_OUTPUT_CLASS 12
#define GUSICS_RECORD_CLASS 13
/*
* Mixer & MUX devices for CS4231
*/
#define GUSMAX_MONO_LVL 0 /* mic input to MUX;
also mono mixer input */
#define GUSMAX_DAC_LVL 1 /* input to MUX; also mixer input */
#define GUSMAX_LINE_IN_LVL 2 /* input to MUX; also mixer input */
#define GUSMAX_CD_LVL 3 /* mixer input only */
#define GUSMAX_MONITOR_LVL 4 /* digital mix (?) */
#define GUSMAX_OUT_LVL 5 /* output level. (?) */
#define GUSMAX_SPEAKER_LVL 6 /* pseudo-device for mute */
#define GUSMAX_LINE_IN_MUTE 7 /* pre-mixer */
#define GUSMAX_DAC_MUTE 8 /* pre-mixer */
#define GUSMAX_CD_MUTE 9 /* pre-mixer */
#define GUSMAX_MONO_MUTE 10 /* pre-mixer--microphone/mono */
#define GUSMAX_MONITOR_MUTE 11 /* post-mixer level/mute */
#define GUSMAX_SPEAKER_MUTE 12 /* speaker mute */
#define GUSMAX_REC_LVL 13 /* post-MUX gain */
#define GUSMAX_RECORD_SOURCE 14
/* Classes */
#define GUSMAX_INPUT_CLASS 15
#define GUSMAX_RECORD_CLASS 16
#define GUSMAX_MONITOR_CLASS 17
#define GUSMAX_OUTPUT_CLASS 18
#ifdef AUDIO_DEBUG
#define GUSPLAYDEBUG /*XXX*/
#define DPRINTF(x) if (gusdebug) printf x
#define DMAPRINTF(x) if (gusdmadebug) printf x
int gusdebug = 0;
int gusdmadebug = 0;
#else
#define DPRINTF(x)
#define DMAPRINTF(x)
#endif
int gus_dostereo = 1;
#define NDMARECS 2048
#ifdef GUSPLAYDEBUG
int gusstats = 0;
struct dma_record {
struct timeval tv;
u_long gusaddr;
caddr_t bsdaddr;
u_short count;
u_char channel;
u_char direction;
} dmarecords[NDMARECS];
int dmarecord_index = 0;
#endif
/*
* local routines
*/
int gusopen __P((void *, int));
void gusclose __P((void *));
void gusmax_close __P((void *));
int gusintr __P((void *));
int gus_set_in_gain __P((caddr_t, u_int, u_char));
int gus_get_in_gain __P((caddr_t));
int gus_set_out_gain __P((caddr_t, u_int, u_char));
int gus_get_out_gain __P((caddr_t));
int gus_set_params __P((void *, int, int, struct audio_params *, struct audio_params *));
int gusmax_set_params __P((void *, int, int, struct audio_params *, struct audio_params *));
int gus_round_blocksize __P((void *, int));
int gus_set_out_port __P((void *, int));
int gus_get_out_port __P((void *));
int gus_set_in_port __P((void *, int));
int gus_get_in_port __P((void *));
int gus_commit_settings __P((void *));
int gus_dma_output __P((void *, void *, int, void (*)(void *), void *));
int gus_dma_input __P((void *, void *, int, void (*)(void *), void *));
int gus_halt_out_dma __P((void *));
int gus_halt_in_dma __P((void *));
int gus_cont_out_dma __P((void *));
int gus_cont_in_dma __P((void *));
int gus_speaker_ctl __P((void *, int));
int gusmaxopen __P((void *, int));
int gusmax_round_blocksize __P((void *, int));
int gusmax_commit_settings __P((void *));
int gusmax_dma_output __P((void *, void *, int, void (*)(void *), void *));
int gusmax_dma_input __P((void *, void *, int, void (*)(void *), void *));
int gusmax_halt_out_dma __P((void *));
int gusmax_halt_in_dma __P((void *));
int gusmax_cont_out_dma __P((void *));
int gusmax_cont_in_dma __P((void *));
int gusmax_speaker_ctl __P((void *, int));
int gusmax_set_out_port __P((void *, int));
int gusmax_get_out_port __P((void *));
int gusmax_set_in_port __P((void *, int));
int gusmax_get_in_port __P((void *));
int gus_getdev __P((void *, struct audio_device *));
STATIC void gus_deinterleave __P((struct gus_softc *, void *, int));
STATIC int gus_mic_ctl __P((void *, int));
STATIC int gus_linein_ctl __P((void *, int));
STATIC int gus_test_iobase __P((int));
STATIC void guspoke __P((int, long, u_char));
STATIC void gusdmaout __P((struct gus_softc *, int, u_long, caddr_t, int));
STATIC void gus_init_cs4231 __P((struct gus_softc *));
STATIC void gus_init_ics2101 __P((struct gus_softc *));
STATIC void gus_set_chan_addrs __P((struct gus_softc *));
STATIC void gusreset __P((struct gus_softc *, int));
STATIC void gus_set_voices __P((struct gus_softc *, int));
STATIC void gus_set_volume __P((struct gus_softc *, int, int));
STATIC void gus_set_samprate __P((struct gus_softc *, int, int));
STATIC void gus_set_recrate __P((struct gus_softc *, u_long));
STATIC void gus_start_voice __P((struct gus_softc *, int, int));
STATIC void gus_stop_voice __P((struct gus_softc *, int, int));
STATIC void gus_set_endaddr __P((struct gus_softc *, int, u_long));
#ifdef GUSPLAYDEBUG
STATIC void gus_set_curaddr __P((struct gus_softc *, int, u_long));
STATIC u_long gus_get_curaddr __P((struct gus_softc *, int));
#endif
STATIC int gus_dmaout_intr __P((struct gus_softc *));
STATIC void gus_dmaout_dointr __P((struct gus_softc *));
STATIC void gus_dmaout_timeout __P((void *));
STATIC int gus_dmain_intr __P((struct gus_softc *));
STATIC int gus_voice_intr __P((struct gus_softc *));
STATIC void gus_start_playing __P((struct gus_softc *, int));
STATIC int gus_continue_playing __P((struct gus_softc *, int));
STATIC u_char guspeek __P((int, u_long));
STATIC u_long convert_to_16bit __P((u_long));
STATIC int gus_mixer_set_port __P((void *, mixer_ctrl_t *));
STATIC int gus_mixer_get_port __P((void *, mixer_ctrl_t *));
STATIC int gusmax_mixer_set_port __P((void *, mixer_ctrl_t *));
STATIC int gusmax_mixer_get_port __P((void *, mixer_ctrl_t *));
STATIC int gus_mixer_query_devinfo __P((void *, mixer_devinfo_t *));
STATIC int gusmax_mixer_query_devinfo __P((void *, mixer_devinfo_t *));
STATIC int gus_query_encoding __P((void *, struct audio_encoding *));
STATIC int gus_get_props __P((void *));
STATIC int gusmax_get_props __P((void *));
STATIC void gusics_master_mute __P((struct ics2101_softc *, int));
STATIC void gusics_dac_mute __P((struct ics2101_softc *, int));
STATIC void gusics_mic_mute __P((struct ics2101_softc *, int));
STATIC void gusics_linein_mute __P((struct ics2101_softc *, int));
STATIC void gusics_cd_mute __P((struct ics2101_softc *, int));
STATIC __inline int gus_to_vol __P((mixer_ctrl_t *, struct ad1848_volume *));
STATIC __inline int gus_from_vol __P((mixer_ctrl_t *, struct ad1848_volume *));
void stereo_dmaintr __P((void *));
/*
* ISA bus driver routines
*/
int gusprobe __P((struct device *, void *, void *));
void gusattach __P((struct device *, struct device *, void *));
struct cfattach gus_ca = {
sizeof(struct gus_softc), gusprobe, gusattach,
};
struct cfdriver gus_cd = {
NULL, "gus", DV_DULL
};
/*
* A mapping from IRQ/DRQ values to the values used in the GUS's internal
* registers. A zero means that the referenced IRQ/DRQ is invalid
*/
static int gus_irq_map[] = {
IRQUNK, IRQUNK, 1, 3, IRQUNK, 2, IRQUNK, 4, IRQUNK, 1, IRQUNK, 5, 6,
IRQUNK, IRQUNK, 7
};
static int gus_drq_map[] = {
DRQUNK, 1, DRQUNK, 2, DRQUNK, 3, 4, 5
};
/*
* A list of valid base addresses for the GUS
*/
static int gus_base_addrs[] = {
0x210, 0x220, 0x230, 0x240, 0x250, 0x260
};
static int gus_addrs = sizeof(gus_base_addrs) / sizeof(gus_base_addrs[0]);
/*
* Maximum frequency values of the GUS based on the number of currently active
* voices. Since the GUS samples a voice every 1.6 us, the maximum frequency
* is dependent on the number of active voices. Yes, it is pretty weird.
*/
static int gus_max_frequency[] = {
44100, /* 14 voices */
41160, /* 15 voices */
38587, /* 16 voices */
36317, /* 17 voices */
34300, /* 18 voices */
32494, /* 19 voices */
30870, /* 20 voices */
29400, /* 21 voices */
28063, /* 22 voices */
26843, /* 23 voices */
25725, /* 24 voices */
24696, /* 25 voices */
23746, /* 26 voices */
22866, /* 27 voices */
22050, /* 28 voices */
21289, /* 29 voices */
20580, /* 30 voices */
19916, /* 31 voices */
19293 /* 32 voices */
};
/*
* A mapping of linear volume levels to the logarithmic volume values used
* by the GF1 chip on the GUS. From GUS SDK vol1.c.
*/
static unsigned short gus_log_volumes[512] = {
0x0000,
0x0700, 0x07ff, 0x0880, 0x08ff, 0x0940, 0x0980, 0x09c0, 0x09ff, 0x0a20,
0x0a40, 0x0a60, 0x0a80, 0x0aa0, 0x0ac0, 0x0ae0, 0x0aff, 0x0b10, 0x0b20,
0x0b30, 0x0b40, 0x0b50, 0x0b60, 0x0b70, 0x0b80, 0x0b90, 0x0ba0, 0x0bb0,
0x0bc0, 0x0bd0, 0x0be0, 0x0bf0, 0x0bff, 0x0c08, 0x0c10, 0x0c18, 0x0c20,
0x0c28, 0x0c30, 0x0c38, 0x0c40, 0x0c48, 0x0c50, 0x0c58, 0x0c60, 0x0c68,
0x0c70, 0x0c78, 0x0c80, 0x0c88, 0x0c90, 0x0c98, 0x0ca0, 0x0ca8, 0x0cb0,
0x0cb8, 0x0cc0, 0x0cc8, 0x0cd0, 0x0cd8, 0x0ce0, 0x0ce8, 0x0cf0, 0x0cf8,
0x0cff, 0x0d04, 0x0d08, 0x0d0c, 0x0d10, 0x0d14, 0x0d18, 0x0d1c, 0x0d20,
0x0d24, 0x0d28, 0x0d2c, 0x0d30, 0x0d34, 0x0d38, 0x0d3c, 0x0d40, 0x0d44,
0x0d48, 0x0d4c, 0x0d50, 0x0d54, 0x0d58, 0x0d5c, 0x0d60, 0x0d64, 0x0d68,
0x0d6c, 0x0d70, 0x0d74, 0x0d78, 0x0d7c, 0x0d80, 0x0d84, 0x0d88, 0x0d8c,
0x0d90, 0x0d94, 0x0d98, 0x0d9c, 0x0da0, 0x0da4, 0x0da8, 0x0dac, 0x0db0,
0x0db4, 0x0db8, 0x0dbc, 0x0dc0, 0x0dc4, 0x0dc8, 0x0dcc, 0x0dd0, 0x0dd4,
0x0dd8, 0x0ddc, 0x0de0, 0x0de4, 0x0de8, 0x0dec, 0x0df0, 0x0df4, 0x0df8,
0x0dfc, 0x0dff, 0x0e02, 0x0e04, 0x0e06, 0x0e08, 0x0e0a, 0x0e0c, 0x0e0e,
0x0e10, 0x0e12, 0x0e14, 0x0e16, 0x0e18, 0x0e1a, 0x0e1c, 0x0e1e, 0x0e20,
0x0e22, 0x0e24, 0x0e26, 0x0e28, 0x0e2a, 0x0e2c, 0x0e2e, 0x0e30, 0x0e32,
0x0e34, 0x0e36, 0x0e38, 0x0e3a, 0x0e3c, 0x0e3e, 0x0e40, 0x0e42, 0x0e44,
0x0e46, 0x0e48, 0x0e4a, 0x0e4c, 0x0e4e, 0x0e50, 0x0e52, 0x0e54, 0x0e56,
0x0e58, 0x0e5a, 0x0e5c, 0x0e5e, 0x0e60, 0x0e62, 0x0e64, 0x0e66, 0x0e68,
0x0e6a, 0x0e6c, 0x0e6e, 0x0e70, 0x0e72, 0x0e74, 0x0e76, 0x0e78, 0x0e7a,
0x0e7c, 0x0e7e, 0x0e80, 0x0e82, 0x0e84, 0x0e86, 0x0e88, 0x0e8a, 0x0e8c,
0x0e8e, 0x0e90, 0x0e92, 0x0e94, 0x0e96, 0x0e98, 0x0e9a, 0x0e9c, 0x0e9e,
0x0ea0, 0x0ea2, 0x0ea4, 0x0ea6, 0x0ea8, 0x0eaa, 0x0eac, 0x0eae, 0x0eb0,
0x0eb2, 0x0eb4, 0x0eb6, 0x0eb8, 0x0eba, 0x0ebc, 0x0ebe, 0x0ec0, 0x0ec2,
0x0ec4, 0x0ec6, 0x0ec8, 0x0eca, 0x0ecc, 0x0ece, 0x0ed0, 0x0ed2, 0x0ed4,
0x0ed6, 0x0ed8, 0x0eda, 0x0edc, 0x0ede, 0x0ee0, 0x0ee2, 0x0ee4, 0x0ee6,
0x0ee8, 0x0eea, 0x0eec, 0x0eee, 0x0ef0, 0x0ef2, 0x0ef4, 0x0ef6, 0x0ef8,
0x0efa, 0x0efc, 0x0efe, 0x0eff, 0x0f01, 0x0f02, 0x0f03, 0x0f04, 0x0f05,
0x0f06, 0x0f07, 0x0f08, 0x0f09, 0x0f0a, 0x0f0b, 0x0f0c, 0x0f0d, 0x0f0e,
0x0f0f, 0x0f10, 0x0f11, 0x0f12, 0x0f13, 0x0f14, 0x0f15, 0x0f16, 0x0f17,
0x0f18, 0x0f19, 0x0f1a, 0x0f1b, 0x0f1c, 0x0f1d, 0x0f1e, 0x0f1f, 0x0f20,
0x0f21, 0x0f22, 0x0f23, 0x0f24, 0x0f25, 0x0f26, 0x0f27, 0x0f28, 0x0f29,
0x0f2a, 0x0f2b, 0x0f2c, 0x0f2d, 0x0f2e, 0x0f2f, 0x0f30, 0x0f31, 0x0f32,
0x0f33, 0x0f34, 0x0f35, 0x0f36, 0x0f37, 0x0f38, 0x0f39, 0x0f3a, 0x0f3b,
0x0f3c, 0x0f3d, 0x0f3e, 0x0f3f, 0x0f40, 0x0f41, 0x0f42, 0x0f43, 0x0f44,
0x0f45, 0x0f46, 0x0f47, 0x0f48, 0x0f49, 0x0f4a, 0x0f4b, 0x0f4c, 0x0f4d,
0x0f4e, 0x0f4f, 0x0f50, 0x0f51, 0x0f52, 0x0f53, 0x0f54, 0x0f55, 0x0f56,
0x0f57, 0x0f58, 0x0f59, 0x0f5a, 0x0f5b, 0x0f5c, 0x0f5d, 0x0f5e, 0x0f5f,
0x0f60, 0x0f61, 0x0f62, 0x0f63, 0x0f64, 0x0f65, 0x0f66, 0x0f67, 0x0f68,
0x0f69, 0x0f6a, 0x0f6b, 0x0f6c, 0x0f6d, 0x0f6e, 0x0f6f, 0x0f70, 0x0f71,
0x0f72, 0x0f73, 0x0f74, 0x0f75, 0x0f76, 0x0f77, 0x0f78, 0x0f79, 0x0f7a,
0x0f7b, 0x0f7c, 0x0f7d, 0x0f7e, 0x0f7f, 0x0f80, 0x0f81, 0x0f82, 0x0f83,
0x0f84, 0x0f85, 0x0f86, 0x0f87, 0x0f88, 0x0f89, 0x0f8a, 0x0f8b, 0x0f8c,
0x0f8d, 0x0f8e, 0x0f8f, 0x0f90, 0x0f91, 0x0f92, 0x0f93, 0x0f94, 0x0f95,
0x0f96, 0x0f97, 0x0f98, 0x0f99, 0x0f9a, 0x0f9b, 0x0f9c, 0x0f9d, 0x0f9e,
0x0f9f, 0x0fa0, 0x0fa1, 0x0fa2, 0x0fa3, 0x0fa4, 0x0fa5, 0x0fa6, 0x0fa7,
0x0fa8, 0x0fa9, 0x0faa, 0x0fab, 0x0fac, 0x0fad, 0x0fae, 0x0faf, 0x0fb0,
0x0fb1, 0x0fb2, 0x0fb3, 0x0fb4, 0x0fb5, 0x0fb6, 0x0fb7, 0x0fb8, 0x0fb9,
0x0fba, 0x0fbb, 0x0fbc, 0x0fbd, 0x0fbe, 0x0fbf, 0x0fc0, 0x0fc1, 0x0fc2,
0x0fc3, 0x0fc4, 0x0fc5, 0x0fc6, 0x0fc7, 0x0fc8, 0x0fc9, 0x0fca, 0x0fcb,
0x0fcc, 0x0fcd, 0x0fce, 0x0fcf, 0x0fd0, 0x0fd1, 0x0fd2, 0x0fd3, 0x0fd4,
0x0fd5, 0x0fd6, 0x0fd7, 0x0fd8, 0x0fd9, 0x0fda, 0x0fdb, 0x0fdc, 0x0fdd,
0x0fde, 0x0fdf, 0x0fe0, 0x0fe1, 0x0fe2, 0x0fe3, 0x0fe4, 0x0fe5, 0x0fe6,
0x0fe7, 0x0fe8, 0x0fe9, 0x0fea, 0x0feb, 0x0fec, 0x0fed, 0x0fee, 0x0fef,
0x0ff0, 0x0ff1, 0x0ff2, 0x0ff3, 0x0ff4, 0x0ff5, 0x0ff6, 0x0ff7, 0x0ff8,
0x0ff9, 0x0ffa, 0x0ffb, 0x0ffc, 0x0ffd, 0x0ffe, 0x0fff};
#define SELECT_GUS_REG(port,x) outb(port+GUS_REG_SELECT,x)
#define WHICH_GUS_REG(port) inb(port+GUS_REG_SELECT)
#define ADDR_HIGH(x) (unsigned int) ((x >> 7L) & 0x1fffL)
#define ADDR_LOW(x) (unsigned int) ((x & 0x7fL) << 9L)
#define GUS_MIN_VOICES 14 /* Minimum possible number of voices */
#define GUS_MAX_VOICES 32 /* Maximum possible number of voices */
#define GUS_VOICE_LEFT 0 /* Voice used for left (and mono) playback */
#define GUS_VOICE_RIGHT 1 /* Voice used for right playback */
#define GUS_MEM_OFFSET 32 /* Offset into GUS memory to begin of buffer */
#define GUS_BUFFER_MULTIPLE 1024 /* Audio buffers are multiples of this */
#define GUS_MEM_FOR_BUFFERS 131072 /* use this many bytes on-GUS */
#define GUS_LEFT_RIGHT_OFFSET (sc->sc_nbufs * sc->sc_chanblocksize + GUS_MEM_OFFSET)
#define GUS_PREC_BYTES (sc->sc_precision >> 3) /* precision to bytes */
/* splgus() must be splaudio() */
#define splgus splaudio
/*
* Interface to higher level audio driver
*/
struct audio_hw_if gus_hw_if = {
gusopen,
gusclose,
NULL, /* drain */
gus_query_encoding,
gus_set_params,
gus_round_blocksize,
gus_set_out_port,
gus_get_out_port,
gus_set_in_port,
gus_get_in_port,
gus_commit_settings,
NULL,
NULL,
gus_dma_output,
gus_dma_input,
gus_halt_out_dma,
gus_halt_in_dma,
gus_cont_out_dma,
gus_cont_in_dma,
gus_speaker_ctl,
gus_getdev,
NULL,
gus_mixer_set_port,
gus_mixer_get_port,
gus_mixer_query_devinfo,
NULL,
NULL,
NULL,
NULL,
gus_get_props,
};
/*
* Some info about the current audio device
*/
struct audio_device gus_device = {
"UltraSound",
"",
"gus",
};
#define FLIP_REV 5 /* This rev has flipped mixer chans */
int
gusprobe(parent, match, aux)
struct device *parent;
void *match, *aux;
{
struct gus_softc *sc = match;
struct isa_attach_args *ia = aux;
struct cfdata *cf = sc->sc_dev.dv_cfdata;
int iobase = ia->ia_iobase;
int recdrq = cf->cf_flags;
sc->sc_iot = ia->ia_iot;
/*
* Before we do anything else, make sure requested IRQ and DRQ are
* valid for this card.
*/
if (gus_irq_map[ia->ia_irq] == IRQUNK) {
printf("gus: invalid irq %d, card not probed\n", ia->ia_irq);
return(0);
}
if (gus_drq_map[ia->ia_drq] == DRQUNK) {
printf("gus: invalid drq %d, card not probed\n", ia->ia_drq);
return(0);
}
if (recdrq != 0x00) {
if (recdrq > 7 || gus_drq_map[recdrq] == DRQUNK) {
printf("gus: invalid flag given for second DMA channel (0x%x), card not probed\n", recdrq);
return(0);
}
} else
recdrq = ia->ia_drq;
if (iobase == IOBASEUNK) {
int i;
for(i = 0; i < gus_addrs; i++)
if (gus_test_iobase(gus_base_addrs[i])) {
iobase = gus_base_addrs[i];
goto done;
}
return 0;
} else if (! gus_test_iobase(iobase))
return 0;
done:
sc->sc_iobase = iobase;
sc->sc_irq = ia->ia_irq;
sc->sc_drq = ia->ia_drq;
sc->sc_recdrq = recdrq;
ia->ia_iobase = sc->sc_iobase;
ia->ia_iosize = 16; /* XXX */
return(1);
}
/*
* Test to see if a particular I/O base is valid for the GUS. Return true
* if it is.
*/
STATIC int
gus_test_iobase (int iobase)
{
int i = splgus();
u_char s1, s2;
/*
* Reset GUS to an initial state before we do anything.
*/
delay(500);
SELECT_GUS_REG(iobase, GUSREG_RESET);
outb(iobase+GUS_DATA_HIGH, 0x00);
delay(500);
SELECT_GUS_REG(iobase, GUSREG_RESET);
outb(iobase+GUS_DATA_HIGH, GUSMASK_MASTER_RESET);
delay(500);
splx(i);
/*
* See if we can write to the board's memory
*/
s1 = guspeek(iobase, 0L);
s2 = guspeek(iobase, 1L);
guspoke(iobase, 0L, 0xaa);
guspoke(iobase, 1L, 0x55);
if ((i=(int)guspeek(iobase, 0L)) != 0xaa) {
return(0);
}
guspoke(iobase, 0L, s1);
guspoke(iobase, 1L, s2);
return 1;
}
/*
* Setup the GUS for use; called shortly after probe
*/
void
gusattach(parent, self, aux)
struct device *parent, *self;
void *aux;
{
struct gus_softc *sc = (void *) self;
struct isa_attach_args *ia = aux;
int port = ia->ia_iobase;
int i;
unsigned char c,d,m;
/*
* Figure out our board rev, and see if we need to initialize the
* mixer
*/
delay(500);
c = inb(port+GUS_BOARD_REV);
if (c != 0xff)
sc->sc_revision = c;
else
sc->sc_revision = 0;
SELECT_GUS_REG(port, GUSREG_RESET);
outb(port+GUS_DATA_HIGH, 0x00);
gusreset(sc, GUS_MAX_VOICES); /* initialize all voices */
gusreset(sc, GUS_MIN_VOICES); /* then set to just the ones we use */
/*
* Setup the IRQ and DRQ lines in software, using values from
* config file
*/
m = GUSMASK_LINE_IN|GUSMASK_LINE_OUT; /* disable all */
c = ((unsigned char) gus_irq_map[ia->ia_irq]) | GUSMASK_BOTH_RQ;
if (sc->sc_recdrq == sc->sc_drq)
d = (unsigned char) (gus_drq_map[sc->sc_drq] |
GUSMASK_BOTH_RQ);
else
d = (unsigned char) (gus_drq_map[sc->sc_drq] |
gus_drq_map[sc->sc_recdrq] << 3);
/*
* Program the IRQ and DMA channels on the GUS. Note that we hardwire
* the GUS to only use one IRQ channel, but we give the user the
* option of using two DMA channels (the other one given by the flags
* option in the config file). Two DMA channels are needed for full-
* duplex operation.
*
* The order of these operations is very magical.
*/
disable_intr();
outb(port+GUS_REG_CONTROL, GUS_REG_IRQCTL);
outb(port+GUS_MIX_CONTROL, m);
outb(port+GUS_IRQCTL_CONTROL, 0x00);
outb(port+0x0f, 0x00);
outb(port+GUS_MIX_CONTROL, m);
outb(port+GUS_DMA_CONTROL, d | 0x80); /* magic reset? */
outb(port+GUS_MIX_CONTROL, m | GUSMASK_CONTROL_SEL);
outb(port+GUS_IRQ_CONTROL, c);
outb(port+GUS_MIX_CONTROL, m);
outb(port+GUS_DMA_CONTROL, d);
outb(port+GUS_MIX_CONTROL, m | GUSMASK_CONTROL_SEL);
outb(port+GUS_IRQ_CONTROL, c);
outb(port+GUS_VOICE_SELECT, 0x00);
/* enable line in, line out. leave mic disabled. */
outb(port+GUS_MIX_CONTROL,
(m | GUSMASK_LATCHES) & ~(GUSMASK_LINE_OUT|GUSMASK_LINE_IN));
outb(port+GUS_VOICE_SELECT, 0x00);
enable_intr();
sc->sc_mixcontrol =
(m | GUSMASK_LATCHES) & ~(GUSMASK_LINE_OUT|GUSMASK_LINE_IN);
/* XXX WILL THIS ALWAYS WORK THE WAY THEY'RE OVERLAYED?! */
sc->sc_codec.sc_isa = sc->sc_dev.dv_parent;
if (sc->sc_revision >= 5 && sc->sc_revision <= 9) {
sc->sc_flags |= GUS_MIXER_INSTALLED;
gus_init_ics2101(sc);
}
if (sc->sc_revision >= 0xa) {
gus_init_cs4231(sc);
}
SELECT_GUS_REG(port, GUSREG_RESET);
/*
* Check to see how much memory we have on this card; see if any
* "mirroring" occurs. We're assuming at least 256K already exists
* on the card; otherwise the initial probe would have failed
*/
guspoke(port, 0L, 0x00);
for(i = 1; i < 1024; i++) {
u_long loc;
/*
* See if we've run into mirroring yet
*/
if (guspeek(port, 0L) != 0)
break;
loc = i << 10;
guspoke(port, loc, 0xaa);
if (guspeek(port, loc) != 0xaa)
break;
}
sc->sc_dsize = i;
sprintf(gus_device.version, "3.%d", sc->sc_revision);
printf("\n <Gravis UltraSound version 3.%d, %dKB DRAM, ",
sc->sc_revision, sc->sc_dsize);
if (HAS_MIXER(sc))
printf("ICS2101 mixer, ");
if (HAS_CODEC(sc))
printf("%s codec/mixer, ", sc->sc_codec.chip_name);
if (sc->sc_recdrq == sc->sc_drq) {
printf("half-duplex");
} else {
printf("full-duplex, record drq %d", sc->sc_recdrq);
}
printf(">\n");
/*
* Setup a default interrupt handler
*/
/* XXX we shouldn't have to use splgus == splclock, nor should
* we use IPL_CLOCK.
*/
sc->sc_ih = isa_intr_establish(ia->ia_ic, ia->ia_irq, IST_EDGE,
IPL_AUDIO, gusintr, sc /* sc->sc_gusdsp */);
/*
* Set some default values
*/
sc->sc_irate = sc->sc_orate = 44100;
sc->sc_encoding = AUDIO_ENCODING_SLINEAR_LE;
sc->sc_precision = 16;
sc->sc_voc[GUS_VOICE_LEFT].voccntl |= GUSMASK_DATA_SIZE16;
sc->sc_voc[GUS_VOICE_RIGHT].voccntl |= GUSMASK_DATA_SIZE16;
sc->sc_channels = 1;
sc->sc_ogain = 340;
gus_commit_settings(sc);
/*
* We always put the left channel full left & right channel
* full right.
* For mono playback, we set up both voices playing the same buffer.
*/
outb(sc->sc_iobase+GUS_VOICE_SELECT, (unsigned char) GUS_VOICE_LEFT);
SELECT_GUS_REG(sc->sc_iobase, GUSREG_PAN_POS);
outb(sc->sc_iobase+GUS_DATA_HIGH, GUS_PAN_FULL_LEFT);
outb(sc->sc_iobase+GUS_VOICE_SELECT, (unsigned char) GUS_VOICE_RIGHT);
SELECT_GUS_REG(sc->sc_iobase, GUSREG_PAN_POS);
outb(sc->sc_iobase+GUS_DATA_HIGH, GUS_PAN_FULL_RIGHT);
/*
* Attach to the generic audio layer
*/
audio_attach_mi(&gus_hw_if, 0, HAS_CODEC(sc) ? (void *)&sc->sc_codec : (void *)sc, &sc->sc_dev);
}
int
gusopen(addr, flags)
void *addr;
int flags;
{
struct gus_softc *sc = addr;
DPRINTF(("gusopen() called\n"));
if (sc->sc_flags & GUS_OPEN)
return EBUSY;
/*
* Some initialization
*/
sc->sc_flags |= GUS_OPEN;
sc->sc_dmabuf = 0;
sc->sc_playbuf = -1;
sc->sc_bufcnt = 0;
sc->sc_voc[GUS_VOICE_LEFT].start_addr = GUS_MEM_OFFSET - 1;
sc->sc_voc[GUS_VOICE_LEFT].current_addr = GUS_MEM_OFFSET;
if (HAS_CODEC(sc)) {
ad1848_open(&sc->sc_codec, flags);
sc->sc_codec.aux1_mute = 0;
ad1848_mute_aux1(&sc->sc_codec, 0); /* turn on DAC output */
if (flags & FREAD) {
sc->sc_codec.mono_mute = 0;
cs4231_mute_mono(&sc->sc_codec, 0);
}
} else if (flags & FREAD) {
/* enable/unmute the microphone */
if (HAS_MIXER(sc)) {
gusics_mic_mute(&sc->sc_mixer, 0);
} else
gus_mic_ctl(sc, SPKR_ON);
}
if (sc->sc_nbufs == 0)
gus_round_blocksize(sc, GUS_BUFFER_MULTIPLE); /* default blksiz */
return 0;
}
int
gusmaxopen(addr, flags)
void *addr;
int flags;
{
struct ad1848_softc *ac = addr;
return gusopen(ac->parent, flags);
}
STATIC void
gus_deinterleave(sc, buf, size)
struct gus_softc *sc;
void *buf;
int size;
{
/* deinterleave the stereo data. We can use sc->sc_deintr_buf
for scratch space. */
int i;
if (size > sc->sc_blocksize) {
printf("gus: deinterleave %d > %d\n", size, sc->sc_blocksize);
return;
} else if (size < sc->sc_blocksize) {
DPRINTF(("gus: deinterleave %d < %d\n", size, sc->sc_blocksize));
}
/*
* size is in bytes.
*/
if (sc->sc_precision == 16) {
u_short *dei = sc->sc_deintr_buf;
u_short *sbuf = buf;
size >>= 1; /* bytecnt to shortcnt */
/* copy 2nd of each pair of samples to the staging area, while
compacting the 1st of each pair into the original area. */
for (i = 0; i < size/2-1; i++) {
dei[i] = sbuf[i*2+1];
sbuf[i+1] = sbuf[i*2+2];
}
/*
* this has copied one less sample than half of the
* buffer. The first sample of the 1st stream was
* already in place and didn't need copying.
* Therefore, we've moved all of the 1st stream's
* samples into place. We have one sample from 2nd
* stream in the last slot of original area, not
* copied to the staging area (But we don't need to!).
* Copy the remainder of the original stream into place.
*/
bcopy(dei, &sbuf[size/2], i * sizeof(short));
} else {
u_char *dei = sc->sc_deintr_buf;
u_char *sbuf = buf;
for (i = 0; i < size/2-1; i++) {
dei[i] = sbuf[i*2+1];
sbuf[i+1] = sbuf[i*2+2];
}
bcopy(dei, &sbuf[size/2], i);
}
}
/*
* Actually output a buffer to the DSP chip
*/
int
gusmax_dma_output(addr, buf, size, intr, arg)
void * addr;
void *buf;
int size;
void (*intr) __P((void *));
void *arg;
{
struct ad1848_softc *ac = addr;
return gus_dma_output(ac->parent, buf, size, intr, arg);
}
/*
* called at splgus() from interrupt handler.
*/
void
stereo_dmaintr(arg)
void *arg;
{
struct gus_softc *sc = arg;
struct stereo_dma_intr *sa = &sc->sc_stereo;
DMAPRINTF(("stereo_dmaintr"));
/*
* Put other half in its place, then call the real interrupt routine :)
*/
sc->sc_dmaoutintr = sa->intr;
sc->sc_outarg = sa->arg;
#ifdef GUSPLAYDEBUG
if (gusstats) {
microtime(&dmarecords[dmarecord_index].tv);
dmarecords[dmarecord_index].gusaddr = sa->dmabuf;
dmarecords[dmarecord_index].bsdaddr = sa->buffer;
dmarecords[dmarecord_index].count = sa->size;
dmarecords[dmarecord_index].channel = 1;
dmarecords[dmarecord_index].direction = 1;
dmarecord_index = ++dmarecord_index % NDMARECS;
}
#endif
gusdmaout(sc, sa->flags, sa->dmabuf, (caddr_t) sa->buffer, sa->size);
sa->flags = 0;
sa->dmabuf = 0;
sa->buffer = 0;
sa->size = 0;
sa->intr = 0;
sa->arg = 0;
}
/*
* Start up DMA output to the card.
* Called at splgus/splaudio already, either from intr handler or from
* generic audio code.
*/
int
gus_dma_output(addr, buf, size, intr, arg)
void * addr;
void *buf;
int size;
void (*intr) __P((void *));
void *arg;
{
struct gus_softc *sc = addr;
u_char *buffer = buf;
u_long boarddma;
int flags;
DMAPRINTF(("gus_dma_output %d @ %p\n", size, buf));
if (size != sc->sc_blocksize) {
DPRINTF(("gus_dma_output reqsize %d not sc_blocksize %d\n",
size, sc->sc_blocksize));
return EINVAL;
}
flags = GUSMASK_DMA_WRITE;
if (sc->sc_precision == 16)
flags |= GUSMASK_DMA_DATA_SIZE;
if (sc->sc_encoding == AUDIO_ENCODING_ULAW ||
sc->sc_encoding == AUDIO_ENCODING_ALAW ||
sc->sc_encoding == AUDIO_ENCODING_ULINEAR_BE ||
sc->sc_encoding == AUDIO_ENCODING_ULINEAR_LE)
flags |= GUSMASK_DMA_INVBIT;
if (sc->sc_channels == 2) {
if (sc->sc_precision == 16) {
if (size & 3) {
DPRINTF(("gus_dma_output: unpaired 16bit samples"));
size &= 3;
}
} else if (size & 1) {
DPRINTF(("gus_dma_output: unpaired samples"));
size &= 1;
}
if (size == 0)
return 0;
gus_deinterleave(sc, (void *)buffer, size);
size >>= 1;
boarddma = size * sc->sc_dmabuf + GUS_MEM_OFFSET;
sc->sc_stereo.intr = intr;
sc->sc_stereo.arg = arg;
sc->sc_stereo.size = size;
sc->sc_stereo.dmabuf = boarddma + GUS_LEFT_RIGHT_OFFSET;
sc->sc_stereo.buffer = buffer + size;
sc->sc_stereo.flags = flags;
if (gus_dostereo) {
intr = stereo_dmaintr;
arg = sc;
}
} else
boarddma = size * sc->sc_dmabuf + GUS_MEM_OFFSET;
sc->sc_flags |= GUS_LOCKED;
sc->sc_dmaoutintr = intr;
sc->sc_outarg = arg;
#ifdef GUSPLAYDEBUG
if (gusstats) {
microtime(&dmarecords[dmarecord_index].tv);
dmarecords[dmarecord_index].gusaddr = boarddma;
dmarecords[dmarecord_index].bsdaddr = buffer;
dmarecords[dmarecord_index].count = size;
dmarecords[dmarecord_index].channel = 0;
dmarecords[dmarecord_index].direction = 1;
dmarecord_index = ++dmarecord_index % NDMARECS;
}
#endif
gusdmaout(sc, flags, boarddma, (caddr_t) buffer, size);
return 0;
}
void
gusmax_close(addr)
void *addr;
{
struct ad1848_softc *ac = addr;
struct gus_softc *sc = ac->parent;
#if 0
ac->aux1_mute = 1;
ad1848_mute_aux1(ac, 1); /* turn off DAC output */
#endif
ad1848_close(ac);
gusclose(sc);
}
/*
* Close out device stuff. Called at splgus() from generic audio layer.
*/
void
gusclose(addr)
void *addr;
{
struct gus_softc *sc = addr;
DPRINTF(("gus_close: sc=%p\n", sc));
/* if (sc->sc_flags & GUS_DMAOUT_ACTIVE) */ {
gus_halt_out_dma(sc);
}
/* if (sc->sc_flags & GUS_DMAIN_ACTIVE) */ {
gus_halt_in_dma(sc);
}
sc->sc_flags &= ~(GUS_OPEN|GUS_LOCKED|GUS_DMAOUT_ACTIVE|GUS_DMAIN_ACTIVE);
if (sc->sc_deintr_buf) {
FREE(sc->sc_deintr_buf, M_DEVBUF);
sc->sc_deintr_buf = NULL;
}
/* turn off speaker, etc. */
/* make sure the voices shut up: */
gus_stop_voice(sc, GUS_VOICE_LEFT, 1);
gus_stop_voice(sc, GUS_VOICE_RIGHT, 0);
}
/*
* Service interrupts. Farm them off to helper routines if we are using the
* GUS for simple playback/record
*/
#ifdef DIAGNOSTIC
int gusintrcnt;
int gusdmaintrcnt;
int gusvocintrcnt;
#endif
int
gusintr(arg)
void *arg;
{
struct gus_softc *sc = arg;
unsigned char intr;
int port = sc->sc_iobase;
int retval = 0;
DPRINTF(("gusintr\n"));
#ifdef DIAGNOSTIC
gusintrcnt++;
#endif
if (HAS_CODEC(sc))
retval = ad1848_intr(&sc->sc_codec);
if ((intr = inb(port+GUS_IRQ_STATUS)) & GUSMASK_IRQ_DMATC) {
DMAPRINTF(("gusintr dma flags=%x\n", sc->sc_flags));
#ifdef DIAGNOSTIC
gusdmaintrcnt++;
#endif
retval += gus_dmaout_intr(sc);
if (sc->sc_flags & GUS_DMAIN_ACTIVE) {
SELECT_GUS_REG(port, GUSREG_SAMPLE_CONTROL);
intr = inb(port+GUS_DATA_HIGH);
if (intr & GUSMASK_SAMPLE_DMATC) {
retval += gus_dmain_intr(sc);
}
}
}
if (intr & (GUSMASK_IRQ_VOICE | GUSMASK_IRQ_VOLUME)) {
DMAPRINTF(("gusintr voice flags=%x\n", sc->sc_flags));
#ifdef DIAGNOSTIC
gusvocintrcnt++;
#endif
retval += gus_voice_intr(sc);
}
if (retval)
return 1;
return retval;
}
int gus_bufcnt[GUS_MEM_FOR_BUFFERS / GUS_BUFFER_MULTIPLE];
int gus_restart; /* how many restarts? */
int gus_stops; /* how many times did voice stop? */
int gus_falsestops; /* stopped but not done? */
int gus_continues;
struct playcont {
struct timeval tv;
u_int playbuf;
u_int dmabuf;
u_char bufcnt;
u_char vaction;
u_char voccntl;
u_char volcntl;
u_long curaddr;
u_long endaddr;
} playstats[NDMARECS];
int playcntr;
STATIC void
gus_dmaout_timeout(arg)
void *arg;
{
struct gus_softc *sc = arg;
int port = sc->sc_iobase;
int s;
printf("%s: dmaout timeout\n", sc->sc_dev.dv_xname);
/*
* Stop any DMA.
*/
s = splgus();
SELECT_GUS_REG(port, GUSREG_DMA_CONTROL);
outb(sc->sc_iobase+GUS_DATA_HIGH, 0);
#if 0
/* XXX we will dmadone below? */
isa_dmaabort(sc->sc_dev.dv_parent, sc->sc_drq);
#endif
gus_dmaout_dointr(sc);
splx(s);
}
/*
* Service DMA interrupts. This routine will only get called if we're doing
* a DMA transfer for playback/record requests from the audio layer.
*/
STATIC int
gus_dmaout_intr(sc)
struct gus_softc *sc;
{
int port = sc->sc_iobase;
/*
* If we got a DMA transfer complete from the GUS DRAM, then deal
* with it.
*/
SELECT_GUS_REG(port, GUSREG_DMA_CONTROL);
if (inb(port+GUS_DATA_HIGH) & GUSMASK_DMA_IRQPEND) {
untimeout(gus_dmaout_timeout, sc);
gus_dmaout_dointr(sc);
return 1;
}
return 0;
}
STATIC void
gus_dmaout_dointr(sc)
struct gus_softc *sc;
{
int port = sc->sc_iobase;
/* sc->sc_dmaoutcnt - 1 because DMA controller counts from zero?. */
isa_dmadone(sc->sc_dev.dv_parent, sc->sc_drq);
sc->sc_flags &= ~GUS_DMAOUT_ACTIVE; /* pending DMA is done */
DMAPRINTF(("gus_dmaout_dointr %d @ %p\n", sc->sc_dmaoutcnt,
sc->sc_dmaoutaddr));
/*
* to prevent clicking, we need to copy last sample
* from last buffer to scratch area just before beginning of
* buffer. However, if we're doing formats that are converted by
* the card during the DMA process, we need to pick up the converted
* byte rather than the one we have in memory.
*/
if (sc->sc_dmabuf == sc->sc_nbufs - 1) {
int i;
switch (sc->sc_encoding) {
case AUDIO_ENCODING_SLINEAR_LE:
case AUDIO_ENCODING_SLINEAR_BE:
if (sc->sc_precision == 8)
goto byte;
/* we have the native format */
for (i = 1; i <= 2; i++)
guspoke(port, sc->sc_gusaddr -
(sc->sc_nbufs - 1) * sc->sc_chanblocksize - i,
sc->sc_dmaoutaddr[sc->sc_dmaoutcnt-i]);
break;
case AUDIO_ENCODING_ULINEAR_LE:
case AUDIO_ENCODING_ULINEAR_BE:
guspoke(port, sc->sc_gusaddr -
(sc->sc_nbufs - 1) * sc->sc_chanblocksize - 2,
guspeek(port,
sc->sc_gusaddr + sc->sc_chanblocksize - 2));
case AUDIO_ENCODING_ALAW:
case AUDIO_ENCODING_ULAW:
byte:
/* we need to fetch the translated byte, then stuff it. */
guspoke(port, sc->sc_gusaddr -
(sc->sc_nbufs - 1) * sc->sc_chanblocksize - 1,
guspeek(port,
sc->sc_gusaddr + sc->sc_chanblocksize - 1));
break;
}
}
/*
* If this is the first half of stereo, "ignore" this one
* and copy out the second half.
*/
if (sc->sc_dmaoutintr == stereo_dmaintr) {
(*sc->sc_dmaoutintr)(sc->sc_outarg);
return;
}
/*
* If the voice is stopped, then start it. Reset the loop
* and roll bits. Call the audio layer routine, since if
* we're starting a stopped voice, that means that the next
* buffer can be filled
*/
sc->sc_flags &= ~GUS_LOCKED;
if (sc->sc_voc[GUS_VOICE_LEFT].voccntl &
GUSMASK_VOICE_STOPPED) {
if (sc->sc_flags & GUS_PLAYING) {
printf("%s: playing yet stopped?\n", sc->sc_dev.dv_xname);
}
sc->sc_bufcnt++; /* another yet to be played */
gus_start_playing(sc, sc->sc_dmabuf);
gus_restart++;
} else {
/*
* set the sound action based on which buffer we
* just transferred. If we just transferred buffer 0
* we want the sound to loop when it gets to the nth
* buffer; if we just transferred
* any other buffer, we want the sound to roll over
* at least one more time. The voice interrupt
* handlers will take care of accounting &
* setting control bits if it's not caught up to us
* yet.
*/
if (++sc->sc_bufcnt == 2) {
/*
* XXX
* If we're too slow in reaction here,
* the voice could be just approaching the
* end of its run. It should be set to stop,
* so these adjustments might not DTRT.
*/
if (sc->sc_dmabuf == 0 &&
sc->sc_playbuf == sc->sc_nbufs - 1) {
/* player is just at the last buf, we're at the
first. Turn on looping, turn off rolling. */
sc->sc_voc[GUS_VOICE_LEFT].voccntl |= GUSMASK_LOOP_ENABLE;
sc->sc_voc[GUS_VOICE_LEFT].volcntl &= ~GUSMASK_VOICE_ROLL;
playstats[playcntr].vaction = 3;
} else {
/* player is at previous buf:
turn on rolling, turn off looping */
sc->sc_voc[GUS_VOICE_LEFT].voccntl &= ~GUSMASK_LOOP_ENABLE;
sc->sc_voc[GUS_VOICE_LEFT].volcntl |= GUSMASK_VOICE_ROLL;
playstats[playcntr].vaction = 4;
}
#ifdef GUSPLAYDEBUG
if (gusstats) {
microtime(&playstats[playcntr].tv);
playstats[playcntr].endaddr = sc->sc_voc[GUS_VOICE_LEFT].end_addr;
playstats[playcntr].voccntl = sc->sc_voc[GUS_VOICE_LEFT].voccntl;
playstats[playcntr].volcntl = sc->sc_voc[GUS_VOICE_LEFT].volcntl;
playstats[playcntr].playbuf = sc->sc_playbuf;
playstats[playcntr].dmabuf = sc->sc_dmabuf;
playstats[playcntr].bufcnt = sc->sc_bufcnt;
playstats[playcntr].curaddr = gus_get_curaddr(sc, GUS_VOICE_LEFT);
playcntr = ++playcntr % NDMARECS;
}
#endif
outb(port+GUS_VOICE_SELECT, GUS_VOICE_LEFT);
SELECT_GUS_REG(port, GUSREG_VOICE_CNTL);
outb(port+GUS_DATA_HIGH, sc->sc_voc[GUS_VOICE_LEFT].voccntl);
SELECT_GUS_REG(port, GUSREG_VOLUME_CONTROL);
outb(port+GUS_DATA_HIGH, sc->sc_voc[GUS_VOICE_LEFT].volcntl);
}
}
gus_bufcnt[sc->sc_bufcnt-1]++;
/*
* flip to the next DMA buffer
*/
sc->sc_dmabuf = ++sc->sc_dmabuf % sc->sc_nbufs;
/*
* See comments below about DMA admission control strategy.
* We can call the upper level here if we have an
* idle buffer (not currently playing) to DMA into.
*/
if (sc->sc_dmaoutintr && sc->sc_bufcnt < sc->sc_nbufs) {
/* clean out to prevent double calls */
void (*pfunc) __P((void *)) = sc->sc_dmaoutintr;
void *arg = sc->sc_outarg;
sc->sc_outarg = 0;
sc->sc_dmaoutintr = 0;
(*pfunc)(arg);
}
}
/*
* Service voice interrupts
*/
STATIC int
gus_voice_intr(sc)
struct gus_softc *sc;
{
int port = sc->sc_iobase;
int ignore = 0, voice, rval = 0;
unsigned char intr, status;
/*
* The point of this may not be obvious at first. A voice can
* interrupt more than once; according to the GUS SDK we are supposed
* to ignore multiple interrupts for the same voice.
*/
while(1) {
SELECT_GUS_REG(port, GUSREG_IRQ_STATUS);
intr = inb(port+GUS_DATA_HIGH);
if ((intr & (GUSMASK_WIRQ_VOLUME | GUSMASK_WIRQ_VOICE))
== (GUSMASK_WIRQ_VOLUME | GUSMASK_WIRQ_VOICE))
/*
* No more interrupts, time to return
*/
return rval;
if ((intr & GUSMASK_WIRQ_VOICE) == 0) {
/*
* We've got a voice interrupt. Ignore previous
* interrupts by the same voice.
*/
rval = 1;
voice = intr & GUSMASK_WIRQ_VOICEMASK;
if ((1 << voice) & ignore)
break;
ignore |= 1 << voice;
/*
* If the voice is stopped, then force it to stop
* (this stops it from continuously generating IRQs)
*/
SELECT_GUS_REG(port, GUSREG_VOICE_CNTL+0x80);
status = inb(port+GUS_DATA_HIGH);
if (status & GUSMASK_VOICE_STOPPED) {
if (voice != GUS_VOICE_LEFT) {
DMAPRINTF(("%s: spurious voice %d stop?\n",
sc->sc_dev.dv_xname, voice));
gus_stop_voice(sc, voice, 0);
continue;
}
gus_stop_voice(sc, voice, 1);
/* also kill right voice */
gus_stop_voice(sc, GUS_VOICE_RIGHT, 0);
sc->sc_bufcnt--; /* it finished a buffer */
if (sc->sc_bufcnt > 0) {
/*
* probably a race to get here: the voice
* stopped while the DMA code was just trying to
* get the next buffer in place.
* Start the voice again.
*/
printf("%s: stopped voice not drained? (%x)\n",
sc->sc_dev.dv_xname, sc->sc_bufcnt);
gus_falsestops++;
sc->sc_playbuf = ++sc->sc_playbuf % sc->sc_nbufs;
gus_start_playing(sc, sc->sc_playbuf);
} else if (sc->sc_bufcnt < 0) {
#ifdef DDB
printf("%s: negative bufcnt in stopped voice\n",
sc->sc_dev.dv_xname);
Debugger();
#else
panic("%s: negative bufcnt in stopped voice",
sc->sc_dev.dv_xname);
#endif
} else {
sc->sc_playbuf = -1; /* none are active */
gus_stops++;
}
/* fall through to callback and admit another
buffer.... */
} else if (sc->sc_bufcnt != 0) {
/*
* This should always be taken if the voice
* is not stopped.
*/
gus_continues++;
if (gus_continue_playing(sc, voice)) {
/*
* we shouldn't have continued--active DMA
* is in the way in the ring, for
* some as-yet undebugged reason.
*/
gus_stop_voice(sc, GUS_VOICE_LEFT, 1);
/* also kill right voice */
gus_stop_voice(sc, GUS_VOICE_RIGHT, 0);
sc->sc_playbuf = -1;
gus_stops++;
}
}
/*
* call the upper level to send on down another
* block. We do admission rate control as follows:
*
* When starting up output (in the first N
* blocks), call the upper layer after the DMA is
* complete (see above in gus_dmaout_intr()).
*
* When output is already in progress and we have
* no more GUS buffers to use for DMA, the DMA
* output routines do not call the upper layer.
* Instead, we call the DMA completion routine
* here, after the voice interrupts indicating
* that it's finished with a buffer.
*
* However, don't call anything here if the DMA
* output flag is set, (which shouldn't happen)
* because we'll squish somebody else's DMA if
* that's the case. When DMA is done, it will
* call back if there is a spare buffer.
*/
if (sc->sc_dmaoutintr && !(sc->sc_flags & GUS_LOCKED)) {
if (sc->sc_dmaoutintr == stereo_dmaintr)
printf("gusdmaout botch?\n");
else {
/* clean out to avoid double calls */
void (*pfunc) __P((void *)) = sc->sc_dmaoutintr;
void *arg = sc->sc_outarg;
sc->sc_outarg = 0;
sc->sc_dmaoutintr = 0;
(*pfunc)(arg);
}
}
}
/*
* Ignore other interrupts for now
*/
}
return 0;
}
STATIC void
gus_start_playing(sc, bufno)
struct gus_softc *sc;
int bufno;
{
int port = sc->sc_iobase;
/*
* Start the voices playing, with buffer BUFNO.
*/
/*
* Loop or roll if we have buffers ready.
*/
if (sc->sc_bufcnt == 1) {
sc->sc_voc[GUS_VOICE_LEFT].voccntl &= ~(GUSMASK_LOOP_ENABLE);
sc->sc_voc[GUS_VOICE_LEFT].volcntl &= ~(GUSMASK_VOICE_ROLL);
} else {
if (bufno == sc->sc_nbufs - 1) {
sc->sc_voc[GUS_VOICE_LEFT].voccntl |= GUSMASK_LOOP_ENABLE;
sc->sc_voc[GUS_VOICE_LEFT].volcntl &= ~(GUSMASK_VOICE_ROLL);
} else {
sc->sc_voc[GUS_VOICE_LEFT].voccntl &= ~GUSMASK_LOOP_ENABLE;
sc->sc_voc[GUS_VOICE_LEFT].volcntl |= GUSMASK_VOICE_ROLL;
}
}
outb(port+GUS_VOICE_SELECT, GUS_VOICE_LEFT);
SELECT_GUS_REG(port, GUSREG_VOICE_CNTL);
outb(port+GUS_DATA_HIGH, sc->sc_voc[GUS_VOICE_LEFT].voccntl);
SELECT_GUS_REG(port, GUSREG_VOLUME_CONTROL);
outb(port+GUS_DATA_HIGH, sc->sc_voc[GUS_VOICE_LEFT].volcntl);
sc->sc_voc[GUS_VOICE_LEFT].current_addr =
GUS_MEM_OFFSET + sc->sc_chanblocksize * bufno;
sc->sc_voc[GUS_VOICE_LEFT].end_addr =
sc->sc_voc[GUS_VOICE_LEFT].current_addr + sc->sc_chanblocksize - 1;
sc->sc_voc[GUS_VOICE_RIGHT].current_addr =
sc->sc_voc[GUS_VOICE_LEFT].current_addr +
(gus_dostereo && sc->sc_channels == 2 ? GUS_LEFT_RIGHT_OFFSET : 0);
/*
* set up right channel to just loop forever, no interrupts,
* starting at the buffer we just filled. We'll feed it data
* at the same time as left channel.
*/
sc->sc_voc[GUS_VOICE_RIGHT].voccntl |= GUSMASK_LOOP_ENABLE;
sc->sc_voc[GUS_VOICE_RIGHT].volcntl &= ~(GUSMASK_VOICE_ROLL);
#ifdef GUSPLAYDEBUG
if (gusstats) {
microtime(&playstats[playcntr].tv);
playstats[playcntr].curaddr = sc->sc_voc[GUS_VOICE_LEFT].current_addr;
playstats[playcntr].voccntl = sc->sc_voc[GUS_VOICE_LEFT].voccntl;
playstats[playcntr].volcntl = sc->sc_voc[GUS_VOICE_LEFT].volcntl;
playstats[playcntr].endaddr = sc->sc_voc[GUS_VOICE_LEFT].end_addr;
playstats[playcntr].playbuf = bufno;
playstats[playcntr].dmabuf = sc->sc_dmabuf;
playstats[playcntr].bufcnt = sc->sc_bufcnt;
playstats[playcntr].vaction = 5;
playcntr = ++playcntr % NDMARECS;
}
#endif
outb(port+GUS_VOICE_SELECT, GUS_VOICE_RIGHT);
SELECT_GUS_REG(port, GUSREG_VOICE_CNTL);
outb(port+GUS_DATA_HIGH, sc->sc_voc[GUS_VOICE_RIGHT].voccntl);
SELECT_GUS_REG(port, GUSREG_VOLUME_CONTROL);
outb(port+GUS_DATA_HIGH, sc->sc_voc[GUS_VOICE_RIGHT].volcntl);
gus_start_voice(sc, GUS_VOICE_RIGHT, 0);
gus_start_voice(sc, GUS_VOICE_LEFT, 1);
if (sc->sc_playbuf == -1)
/* mark start of playing */
sc->sc_playbuf = bufno;
}
STATIC int
gus_continue_playing(sc, voice)
struct gus_softc *sc;
int voice;
{
int port = sc->sc_iobase;
/*
* stop this voice from interrupting while we work.
*/
SELECT_GUS_REG(port, GUSREG_VOICE_CNTL);
outb(port+GUS_DATA_HIGH, sc->sc_voc[voice].voccntl & ~(GUSMASK_VOICE_IRQ));
/*
* update playbuf to point to the buffer the hardware just started
* playing
*/
sc->sc_playbuf = ++sc->sc_playbuf % sc->sc_nbufs;
/*
* account for buffer just finished
*/
if (--sc->sc_bufcnt == 0) {
DPRINTF(("gus: bufcnt 0 on continuing voice?\n"));
}
if (sc->sc_playbuf == sc->sc_dmabuf && (sc->sc_flags & GUS_LOCKED)) {
printf("%s: continue into active dmabuf?\n", sc->sc_dev.dv_xname);
return 1;
}
/*
* Select the end of the buffer based on the currently active
* buffer, [plus extra contiguous buffers (if ready)].
*/
/*
* set endpoint at end of buffer we just started playing.
*
* The total gets -1 because end addrs are one less than you might
* think (the end_addr is the address of the last sample to play)
*/
gus_set_endaddr(sc, voice, GUS_MEM_OFFSET +
sc->sc_chanblocksize * (sc->sc_playbuf + 1) - 1);
if (sc->sc_bufcnt < 2) {
/*
* Clear out the loop and roll flags, and rotate the currently
* playing buffer. That way, if we don't manage to get more
* data before this buffer finishes, we'll just stop.
*/
sc->sc_voc[voice].voccntl &= ~GUSMASK_LOOP_ENABLE;
sc->sc_voc[voice].volcntl &= ~GUSMASK_VOICE_ROLL;
playstats[playcntr].vaction = 0;
} else {
/*
* We have some buffers to play. set LOOP if we're on the
* last buffer in the ring, otherwise set ROLL.
*/
if (sc->sc_playbuf == sc->sc_nbufs - 1) {
sc->sc_voc[voice].voccntl |= GUSMASK_LOOP_ENABLE;
sc->sc_voc[voice].volcntl &= ~GUSMASK_VOICE_ROLL;
playstats[playcntr].vaction = 1;
} else {
sc->sc_voc[voice].voccntl &= ~GUSMASK_LOOP_ENABLE;
sc->sc_voc[voice].volcntl |= GUSMASK_VOICE_ROLL;
playstats[playcntr].vaction = 2;
}
}
#ifdef GUSPLAYDEBUG
if (gusstats) {
microtime(&playstats[playcntr].tv);
playstats[playcntr].curaddr = gus_get_curaddr(sc, voice);
playstats[playcntr].voccntl = sc->sc_voc[voice].voccntl;
playstats[playcntr].volcntl = sc->sc_voc[voice].volcntl;
playstats[playcntr].endaddr = sc->sc_voc[voice].end_addr;
playstats[playcntr].playbuf = sc->sc_playbuf;
playstats[playcntr].dmabuf = sc->sc_dmabuf;
playstats[playcntr].bufcnt = sc->sc_bufcnt;
playcntr = ++playcntr % NDMARECS;
}
#endif
/*
* (re-)set voice parameters. This will reenable interrupts from this
* voice.
*/
SELECT_GUS_REG(port, GUSREG_VOICE_CNTL);
outb(port+GUS_DATA_HIGH, sc->sc_voc[voice].voccntl);
SELECT_GUS_REG(port, GUSREG_VOLUME_CONTROL);
outb(port+GUS_DATA_HIGH, sc->sc_voc[voice].volcntl);
return 0;
}
/*
* Send/receive data into GUS's DRAM using DMA. Called at splgus()
*/
STATIC void
gusdmaout(sc, flags, gusaddr, buffaddr, length)
struct gus_softc *sc;
int flags, length;
u_long gusaddr;
caddr_t buffaddr;
{
unsigned char c = (unsigned char) flags;
int port = sc->sc_iobase;
DMAPRINTF(("gusdmaout flags=%x scflags=%x\n", flags, sc->sc_flags));
sc->sc_gusaddr = gusaddr;
/*
* If we're using a 16 bit DMA channel, we have to jump through some
* extra hoops; this includes translating the DRAM address a bit
*/
if (sc->sc_drq >= 4) {
c |= GUSMASK_DMA_WIDTH;
gusaddr = convert_to_16bit(gusaddr);
}
/*
* Add flag bits that we always set - fast DMA, enable IRQ
*/
c |= GUSMASK_DMA_ENABLE | GUSMASK_DMA_R0 | GUSMASK_DMA_IRQ;
/*
* Make sure the GUS _isn't_ setup for DMA
*/
SELECT_GUS_REG(port, GUSREG_DMA_CONTROL);
outb(port+GUS_DATA_HIGH, 0);
/*
* Tell the PC DMA controller to start doing DMA
*/
sc->sc_dmaoutaddr = (u_char *) buffaddr;
sc->sc_dmaoutcnt = length;
isa_dmastart(sc->sc_dev.dv_parent, sc->sc_drq, buffaddr, length,
NULL, DMAMODE_WRITE, BUS_DMA_NOWAIT);
/*
* Set up DMA address - use the upper 16 bits ONLY
*/
sc->sc_flags |= GUS_DMAOUT_ACTIVE;
SELECT_GUS_REG(port, GUSREG_DMA_START);
outw(port+GUS_DATA_LOW, (int) (gusaddr >> 4));
/*
* Tell the GUS to start doing DMA
*/
SELECT_GUS_REG(port, GUSREG_DMA_CONTROL);
outb(port+GUS_DATA_HIGH, c);
/*
* XXX If we don't finish in one second, give up...
*/
untimeout(gus_dmaout_timeout, sc); /* flush old one, if there is one */
timeout(gus_dmaout_timeout, sc, hz);
}
/*
* Start a voice playing on the GUS. Called from interrupt handler at
* splgus().
*/
STATIC void
gus_start_voice(sc, voice, intrs)
struct gus_softc *sc;
int voice;
int intrs;
{
int port = sc->sc_iobase;
u_long start;
u_long current;
u_long end;
/*
* Pick all the values for the voice out of the gus_voice struct
* and use those to program the voice
*/
start = sc->sc_voc[voice].start_addr;
current = sc->sc_voc[voice].current_addr;
end = sc->sc_voc[voice].end_addr;
/*
* If we're using 16 bit data, mangle the addresses a bit
*/
if (sc->sc_voc[voice].voccntl & GUSMASK_DATA_SIZE16) {
/* -1 on start so that we get onto sample boundary--other
code always sets it for 1-byte rollover protection */
start = convert_to_16bit(start-1);
current = convert_to_16bit(current);
end = convert_to_16bit(end);
}
/*
* Select the voice we want to use, and program the data addresses
*/
outb(port+GUS_VOICE_SELECT, (unsigned char) voice);
SELECT_GUS_REG(port, GUSREG_START_ADDR_HIGH);
outw(port+GUS_DATA_LOW, ADDR_HIGH(start));
SELECT_GUS_REG(port, GUSREG_START_ADDR_LOW);
outw(port+GUS_DATA_LOW, ADDR_LOW(start));
SELECT_GUS_REG(port, GUSREG_CUR_ADDR_HIGH);
outw(port+GUS_DATA_LOW, ADDR_HIGH(current));
SELECT_GUS_REG(port, GUSREG_CUR_ADDR_LOW);
outw(port+GUS_DATA_LOW, ADDR_LOW(current));
SELECT_GUS_REG(port, GUSREG_END_ADDR_HIGH);
outw(port+GUS_DATA_LOW, ADDR_HIGH(end));
SELECT_GUS_REG(port, GUSREG_END_ADDR_LOW);
outw(port+GUS_DATA_LOW, ADDR_LOW(end));
/*
* (maybe) enable interrupts, disable voice stopping
*/
if (intrs) {
sc->sc_flags |= GUS_PLAYING; /* playing is about to start */
sc->sc_voc[voice].voccntl |= GUSMASK_VOICE_IRQ;
DMAPRINTF(("gus voice playing=%x\n", sc->sc_flags));
} else
sc->sc_voc[voice].voccntl &= ~GUSMASK_VOICE_IRQ;
sc->sc_voc[voice].voccntl &= ~(GUSMASK_VOICE_STOPPED |
GUSMASK_STOP_VOICE);
/*
* Tell the GUS about it. Note that we're doing volume ramping here
* from 0 up to the set volume to help reduce clicks.
*/
SELECT_GUS_REG(port, GUSREG_START_VOLUME);
outb(port+GUS_DATA_HIGH, 0x00);
SELECT_GUS_REG(port, GUSREG_END_VOLUME);
outb(port+GUS_DATA_HIGH, sc->sc_voc[voice].current_volume >> 4);
SELECT_GUS_REG(port, GUSREG_CUR_VOLUME);
outw(port+GUS_DATA_LOW, 0x00);
SELECT_GUS_REG(port, GUSREG_VOLUME_RATE);
outb(port+GUS_DATA_HIGH, 63);
SELECT_GUS_REG(port, GUSREG_VOICE_CNTL);
outb(port+GUS_DATA_HIGH, sc->sc_voc[voice].voccntl);
SELECT_GUS_REG(port, GUSREG_VOLUME_CONTROL);
outb(port+GUS_DATA_HIGH, 0x00);
delay(50);
SELECT_GUS_REG(port, GUSREG_VOICE_CNTL);
outb(port+GUS_DATA_HIGH, sc->sc_voc[voice].voccntl);
SELECT_GUS_REG(port, GUSREG_VOLUME_CONTROL);
outb(port+GUS_DATA_HIGH, 0x00);
}
/*
* Stop a given voice. called at splgus()
*/
STATIC void
gus_stop_voice(sc, voice, intrs_too)
struct gus_softc *sc;
int voice;
int intrs_too;
{
int port = sc->sc_iobase;
sc->sc_voc[voice].voccntl |= GUSMASK_VOICE_STOPPED |
GUSMASK_STOP_VOICE;
if (intrs_too) {
sc->sc_voc[voice].voccntl &= ~(GUSMASK_VOICE_IRQ);
/* no more DMA to do */
sc->sc_flags &= ~GUS_PLAYING;
}
DMAPRINTF(("gusintr voice notplaying=%x\n", sc->sc_flags));
guspoke(port, 0L, 0);
outb(port+GUS_VOICE_SELECT, (unsigned char) voice);
SELECT_GUS_REG(port, GUSREG_CUR_VOLUME);
outw(port+GUS_DATA_LOW, 0x0000);
SELECT_GUS_REG(port, GUSREG_VOICE_CNTL);
outb(port+GUS_DATA_HIGH, sc->sc_voc[voice].voccntl);
delay(100);
SELECT_GUS_REG(port, GUSREG_CUR_VOLUME);
outw(port+GUS_DATA_LOW, 0x0000);
SELECT_GUS_REG(port, GUSREG_VOICE_CNTL);
outb(port+GUS_DATA_HIGH, sc->sc_voc[voice].voccntl);
SELECT_GUS_REG(port, GUSREG_CUR_ADDR_HIGH);
outw(port+GUS_DATA_LOW, 0x0000);
SELECT_GUS_REG(port, GUSREG_CUR_ADDR_LOW);
outw(port+GUS_DATA_LOW, 0x0000);
}
/*
* Set the volume of a given voice. Called at splgus().
*/
STATIC void
gus_set_volume(sc, voice, volume)
struct gus_softc *sc;
int voice, volume;
{
int port = sc->sc_iobase;
unsigned int gusvol;
gusvol = gus_log_volumes[volume < 512 ? volume : 511];
sc->sc_voc[voice].current_volume = gusvol;
outb(port+GUS_VOICE_SELECT, (unsigned char) voice);
SELECT_GUS_REG(port, GUSREG_START_VOLUME);
outb(port+GUS_DATA_HIGH, (unsigned char) (gusvol >> 4));
SELECT_GUS_REG(port, GUSREG_END_VOLUME);
outb(port+GUS_DATA_HIGH, (unsigned char) (gusvol >> 4));
SELECT_GUS_REG(port, GUSREG_CUR_VOLUME);
outw(port+GUS_DATA_LOW, gusvol << 4);
delay(500);
outw(port+GUS_DATA_LOW, gusvol << 4);
}
/*
* Interface to the audio layer.
*/
int
gusmax_set_params(addr, setmode, usemode, p, r)
void *addr;
int setmode, usemode;
struct audio_params *p, *r;
{
struct ad1848_softc *ac = addr;
struct gus_softc *sc = ac->parent;
int error;
error = ad1848_set_params(ac, setmode, usemode, p, r);
if (error)
return error;
error = gus_set_params(sc, setmode, usemode, p, r);
return error;
}
int
gus_set_params(addr, setmode, usemode, p, r)
void *addr;
int setmode, usemode;
struct audio_params *p, *r;
{
struct gus_softc *sc = addr;
int s;
switch (p->encoding) {
case AUDIO_ENCODING_ULAW:
case AUDIO_ENCODING_ALAW:
case AUDIO_ENCODING_SLINEAR_LE:
case AUDIO_ENCODING_ULINEAR_LE:
case AUDIO_ENCODING_SLINEAR_BE:
case AUDIO_ENCODING_ULINEAR_BE:
break;
default:
return (EINVAL);
}
s = splaudio();
if (p->precision == 8) {
sc->sc_voc[GUS_VOICE_LEFT].voccntl &= ~GUSMASK_DATA_SIZE16;
sc->sc_voc[GUS_VOICE_RIGHT].voccntl &= ~GUSMASK_DATA_SIZE16;
} else {
sc->sc_voc[GUS_VOICE_LEFT].voccntl |= GUSMASK_DATA_SIZE16;
sc->sc_voc[GUS_VOICE_RIGHT].voccntl |= GUSMASK_DATA_SIZE16;
}
sc->sc_encoding = p->encoding;
sc->sc_precision = p->precision;
sc->sc_channels = p->channels;
splx(s);
if (p->sample_rate > gus_max_frequency[sc->sc_voices - GUS_MIN_VOICES])
p->sample_rate = gus_max_frequency[sc->sc_voices - GUS_MIN_VOICES];
if (setmode & AUMODE_RECORD)
sc->sc_irate = p->sample_rate;
if (setmode & AUMODE_PLAY)
sc->sc_orate = p->sample_rate;
switch (p->encoding) {
case AUDIO_ENCODING_ULAW:
p->sw_code = mulaw_to_ulinear8;
r->sw_code = ulinear8_to_mulaw;
break;
case AUDIO_ENCODING_ALAW:
p->sw_code = alaw_to_ulinear8;
r->sw_code = ulinear8_to_alaw;
break;
case AUDIO_ENCODING_ULINEAR_BE:
case AUDIO_ENCODING_SLINEAR_BE:
r->sw_code = p->sw_code = swap_bytes;
break;
}
return 0;
}
/*
* Interface to the audio layer - set the blocksize to the correct number
* of units
*/
int
gusmax_round_blocksize(addr, blocksize)
void * addr;
int blocksize;
{
struct ad1848_softc *ac = addr;
struct gus_softc *sc = ac->parent;
/* blocksize = ad1848_round_blocksize(ac, blocksize);*/
return gus_round_blocksize(sc, blocksize);
}
int
gus_round_blocksize(addr, blocksize)
void * addr;
int blocksize;
{
struct gus_softc *sc = addr;
DPRINTF(("gus_round_blocksize called\n"));
if ((sc->sc_encoding == AUDIO_ENCODING_ULAW ||
sc->sc_encoding == AUDIO_ENCODING_ALAW) && blocksize > 32768)
blocksize = 32768;
else if (blocksize > 65536)
blocksize = 65536;
if ((blocksize % GUS_BUFFER_MULTIPLE) != 0)
blocksize = (blocksize / GUS_BUFFER_MULTIPLE + 1) *
GUS_BUFFER_MULTIPLE;
/* set up temporary buffer to hold the deinterleave, if necessary
for stereo output */
if (sc->sc_deintr_buf) {
FREE(sc->sc_deintr_buf, M_DEVBUF);
sc->sc_deintr_buf = NULL;
}
MALLOC(sc->sc_deintr_buf, void *, blocksize>>1, M_DEVBUF, M_WAITOK);
sc->sc_blocksize = blocksize;
/* multi-buffering not quite working yet. */
sc->sc_nbufs = /*GUS_MEM_FOR_BUFFERS / blocksize*/ 2;
gus_set_chan_addrs(sc);
return blocksize;
}
int
gus_get_out_gain(addr)
caddr_t addr;
{
struct gus_softc *sc = (struct gus_softc *) addr;
DPRINTF(("gus_get_out_gain called\n"));
return sc->sc_ogain / 2;
}
STATIC inline void gus_set_voices(sc, voices)
struct gus_softc *sc;
int voices;
{
int port = sc->sc_iobase;
/*
* Select the active number of voices
*/
SELECT_GUS_REG(port, GUSREG_ACTIVE_VOICES);
outb(port+GUS_DATA_HIGH, (voices-1) | 0xc0);
sc->sc_voices = voices;
}
/*
* Actually set the settings of various values on the card
*/
int
gusmax_commit_settings(addr)
void * addr;
{
struct ad1848_softc *ac = addr;
struct gus_softc *sc = ac->parent;
int error;
error = ad1848_commit_settings(ac);
if (error)
return error;
return gus_commit_settings(sc);
}
/*
* Commit the settings. Called at normal IPL.
*/
int
gus_commit_settings(addr)
void * addr;
{
struct gus_softc *sc = addr;
int s;
DPRINTF(("gus_commit_settings called (gain = %d)\n",sc->sc_ogain));
s = splgus();
gus_set_recrate(sc, sc->sc_irate);
gus_set_volume(sc, GUS_VOICE_LEFT, sc->sc_ogain);
gus_set_volume(sc, GUS_VOICE_RIGHT, sc->sc_ogain);
gus_set_samprate(sc, GUS_VOICE_LEFT, sc->sc_orate);
gus_set_samprate(sc, GUS_VOICE_RIGHT, sc->sc_orate);
splx(s);
gus_set_chan_addrs(sc);
return 0;
}
STATIC void
gus_set_chan_addrs(sc)
struct gus_softc *sc;
{
/*
* We use sc_nbufs * blocksize bytes of storage in the on-board GUS
* ram.
* For mono, each of the sc_nbufs buffers is DMA'd to in one chunk,
* and both left & right channels play the same buffer.
*
* For stereo, each channel gets a contiguous half of the memory,
* and each has sc_nbufs buffers of size blocksize/2.
* Stereo data are deinterleaved in main memory before the DMA out
* routines are called to queue the output.
*
* The blocksize per channel is kept in sc_chanblocksize.
*/
if (sc->sc_channels == 2)
sc->sc_chanblocksize = sc->sc_blocksize/2;
else
sc->sc_chanblocksize = sc->sc_blocksize;
sc->sc_voc[GUS_VOICE_LEFT].start_addr = GUS_MEM_OFFSET - 1;
sc->sc_voc[GUS_VOICE_RIGHT].start_addr =
(gus_dostereo && sc->sc_channels == 2 ? GUS_LEFT_RIGHT_OFFSET : 0)
+ GUS_MEM_OFFSET - 1;
sc->sc_voc[GUS_VOICE_RIGHT].current_addr =
sc->sc_voc[GUS_VOICE_RIGHT].start_addr + 1;
sc->sc_voc[GUS_VOICE_RIGHT].end_addr =
sc->sc_voc[GUS_VOICE_RIGHT].start_addr +
sc->sc_nbufs * sc->sc_chanblocksize;
}
/*
* Set the sample rate of the given voice. Called at splgus().
*/
STATIC void
gus_set_samprate(sc, voice, freq)
struct gus_softc *sc;
int voice, freq;
{
int port = sc->sc_iobase;
unsigned int fc;
u_long temp, f = (u_long) freq;
/*
* calculate fc based on the number of active voices;
* we need to use longs to preserve enough bits
*/
temp = (u_long) gus_max_frequency[sc->sc_voices-GUS_MIN_VOICES];
fc = (unsigned int)(((f << 9L) + (temp >> 1L)) / temp);
fc <<= 1;
/*
* Program the voice frequency, and set it in the voice data record
*/
outb(port+GUS_VOICE_SELECT, (unsigned char) voice);
SELECT_GUS_REG(port, GUSREG_FREQ_CONTROL);
outw(port+GUS_DATA_LOW, fc);
sc->sc_voc[voice].rate = freq;
}
/*
* Set the sample rate of the recording frequency. Formula is from the GUS
* SDK. Called at splgus().
*/
STATIC void
gus_set_recrate(sc, rate)
struct gus_softc *sc;
u_long rate;
{
int port = sc->sc_iobase;
u_char realrate;
DPRINTF(("gus_set_recrate %lu\n", rate));
#if 0
realrate = 9878400/(16*(rate+2)); /* formula from GUS docs */
#endif
realrate = (9878400 >> 4)/rate - 2; /* formula from code, sigh. */
SELECT_GUS_REG(port, GUSREG_SAMPLE_FREQ);
outb(port+GUS_DATA_HIGH, realrate);
}
/*
* Interface to the audio layer - turn the output on or off. Note that some
* of these bits are flipped in the register
*/
int
gusmax_speaker_ctl(addr, newstate)
void * addr;
int newstate;
{
struct ad1848_softc *sc = addr;
return gus_speaker_ctl(sc->parent, newstate);
}
int
gus_speaker_ctl(addr, newstate)
void * addr;
int newstate;
{
struct gus_softc *sc = (struct gus_softc *) addr;
/* Line out bit is flipped: 0 enables, 1 disables */
if ((newstate == SPKR_ON) &&
(sc->sc_mixcontrol & GUSMASK_LINE_OUT)) {
sc->sc_mixcontrol &= ~GUSMASK_LINE_OUT;
outb(sc->sc_iobase+GUS_MIX_CONTROL, sc->sc_mixcontrol);
}
if ((newstate == SPKR_OFF) &&
(sc->sc_mixcontrol & GUSMASK_LINE_OUT) == 0) {
sc->sc_mixcontrol |= GUSMASK_LINE_OUT;
outb(sc->sc_iobase+GUS_MIX_CONTROL, sc->sc_mixcontrol);
}
return 0;
}
STATIC int
gus_linein_ctl(addr, newstate)
void * addr;
int newstate;
{
struct gus_softc *sc = (struct gus_softc *) addr;
/* Line in bit is flipped: 0 enables, 1 disables */
if ((newstate == SPKR_ON) &&
(sc->sc_mixcontrol & GUSMASK_LINE_IN)) {
sc->sc_mixcontrol &= ~GUSMASK_LINE_IN;
outb(sc->sc_iobase+GUS_MIX_CONTROL, sc->sc_mixcontrol);
}
if ((newstate == SPKR_OFF) &&
(sc->sc_mixcontrol & GUSMASK_LINE_IN) == 0) {
sc->sc_mixcontrol |= GUSMASK_LINE_IN;
outb(sc->sc_iobase+GUS_MIX_CONTROL, sc->sc_mixcontrol);
}
return 0;
}
STATIC int
gus_mic_ctl(addr, newstate)
void * addr;
int newstate;
{
struct gus_softc *sc = (struct gus_softc *) addr;
/* Mic bit is normal: 1 enables, 0 disables */
if ((newstate == SPKR_ON) &&
(sc->sc_mixcontrol & GUSMASK_MIC_IN) == 0) {
sc->sc_mixcontrol |= GUSMASK_MIC_IN;
outb(sc->sc_iobase+GUS_MIX_CONTROL, sc->sc_mixcontrol);
}
if ((newstate == SPKR_OFF) &&
(sc->sc_mixcontrol & GUSMASK_MIC_IN)) {
sc->sc_mixcontrol &= ~GUSMASK_MIC_IN;
outb(sc->sc_iobase+GUS_MIX_CONTROL, sc->sc_mixcontrol);
}
return 0;
}
/*
* Set the end address of a give voice. Called at splgus()
*/
STATIC void
gus_set_endaddr(sc, voice, addr)
struct gus_softc *sc;
int voice;
u_long addr;
{
int port = sc->sc_iobase;
sc->sc_voc[voice].end_addr = addr;
if (sc->sc_voc[voice].voccntl & GUSMASK_DATA_SIZE16)
addr = convert_to_16bit(addr);
SELECT_GUS_REG(port, GUSREG_END_ADDR_HIGH);
outw(port+GUS_DATA_LOW, ADDR_HIGH(addr));
SELECT_GUS_REG(port, GUSREG_END_ADDR_LOW);
outw(port+GUS_DATA_LOW, ADDR_LOW(addr));
}
#ifdef GUSPLAYDEBUG
/*
* Set current address. called at splgus()
*/
STATIC void
gus_set_curaddr(sc, voice, addr)
struct gus_softc *sc;
int voice;
u_long addr;
{
int port = sc->sc_iobase;
sc->sc_voc[voice].current_addr = addr;
if (sc->sc_voc[voice].voccntl & GUSMASK_DATA_SIZE16)
addr = convert_to_16bit(addr);
outb(port+GUS_VOICE_SELECT, (unsigned char) voice);
SELECT_GUS_REG(port, GUSREG_CUR_ADDR_HIGH);
outw(port+GUS_DATA_LOW, ADDR_HIGH(addr));
SELECT_GUS_REG(port, GUSREG_CUR_ADDR_LOW);
outw(port+GUS_DATA_LOW, ADDR_LOW(addr));
}
/*
* Get current GUS playback address. Called at splgus().
*/
STATIC u_long
gus_get_curaddr(sc, voice)
struct gus_softc *sc;
int voice;
{
int port = sc->sc_iobase;
u_long addr;
outb(port+GUS_VOICE_SELECT, (unsigned char) voice);
SELECT_GUS_REG(port, GUSREG_CUR_ADDR_HIGH|GUSREG_READ);
addr = (inw(port+GUS_DATA_LOW) & 0x1fff) << 7;
SELECT_GUS_REG(port, GUSREG_CUR_ADDR_LOW|GUSREG_READ);
addr |= (inw(port+GUS_DATA_LOW) >> 9L) & 0x7f;
if (sc->sc_voc[voice].voccntl & GUSMASK_DATA_SIZE16)
addr = (addr & 0xc0000) | ((addr & 0x1ffff) << 1); /* undo 16-bit change */
DPRINTF(("gus voice %d curaddr %ld end_addr %ld\n",
voice, addr, sc->sc_voc[voice].end_addr));
/* XXX sanity check the address? */
return(addr);
}
#endif
/*
* Convert an address value to a "16 bit" value - why this is necessary I
* have NO idea
*/
STATIC u_long
convert_to_16bit(address)
u_long address;
{
u_long old_address;
old_address = address;
address >>= 1;
address &= 0x0001ffffL;
address |= (old_address & 0x000c0000L);
return (address);
}
/*
* Write a value into the GUS's DRAM
*/
STATIC void
guspoke(port, address, value)
int port;
long address;
unsigned char value;
{
/*
* Select the DRAM address
*/
SELECT_GUS_REG(port, GUSREG_DRAM_ADDR_LOW);
outw(port+GUS_DATA_LOW, (unsigned int) (address & 0xffff));
SELECT_GUS_REG(port, GUSREG_DRAM_ADDR_HIGH);
outb(port+GUS_DATA_HIGH, (unsigned char) ((address >> 16) & 0xff));
/*
* Actually write the data
*/
outb(port+GUS_DRAM_DATA, value);
}
/*
* Read a value from the GUS's DRAM
*/
STATIC unsigned char
guspeek(port, address)
int port;
u_long address;
{
/*
* Select the DRAM address
*/
SELECT_GUS_REG(port, GUSREG_DRAM_ADDR_LOW);
outw(port+GUS_DATA_LOW, (unsigned int) (address & 0xffff));
SELECT_GUS_REG(port, GUSREG_DRAM_ADDR_HIGH);
outb(port+GUS_DATA_HIGH, (unsigned char) ((address >> 16) & 0xff));
/*
* Read in the data from the board
*/
return (unsigned char) inb(port+GUS_DRAM_DATA);
}
/*
* Reset the Gravis UltraSound card, completely
*/
STATIC void
gusreset(sc, voices)
struct gus_softc *sc;
int voices;
{
int port = sc->sc_iobase;
int i,s;
s = splgus();
/*
* Reset the GF1 chip
*/
SELECT_GUS_REG(port, GUSREG_RESET);
outb(port+GUS_DATA_HIGH, 0x00);
delay(500);
/*
* Release reset
*/
SELECT_GUS_REG(port, GUSREG_RESET);
outb(port+GUS_DATA_HIGH, GUSMASK_MASTER_RESET);
delay(500);
/*
* Reset MIDI port as well
*/
outb(GUS_MIDI_CONTROL,MIDI_RESET);
delay(500);
outb(GUS_MIDI_CONTROL,0x00);
/*
* Clear interrupts
*/
SELECT_GUS_REG(port, GUSREG_DMA_CONTROL);
outb(port+GUS_DATA_HIGH, 0x00);
SELECT_GUS_REG(port, GUSREG_TIMER_CONTROL);
outb(port+GUS_DATA_HIGH, 0x00);
SELECT_GUS_REG(port, GUSREG_SAMPLE_CONTROL);
outb(port+GUS_DATA_HIGH, 0x00);
gus_set_voices(sc, voices);
inb(port+GUS_IRQ_STATUS);
SELECT_GUS_REG(port, GUSREG_DMA_CONTROL);
inb(port+GUS_DATA_HIGH);
SELECT_GUS_REG(port, GUSREG_SAMPLE_CONTROL);
inb(port+GUS_DATA_HIGH);
SELECT_GUS_REG(port, GUSREG_IRQ_STATUS);
inb(port+GUS_DATA_HIGH);
/*
* Reset voice specific information
*/
for(i = 0; i < voices; i++) {
outb(port+GUS_VOICE_SELECT, (unsigned char) i);
SELECT_GUS_REG(port, GUSREG_VOICE_CNTL);
sc->sc_voc[i].voccntl = GUSMASK_VOICE_STOPPED |
GUSMASK_STOP_VOICE;
outb(port+GUS_DATA_HIGH, sc->sc_voc[i].voccntl);
sc->sc_voc[i].volcntl = GUSMASK_VOLUME_STOPPED |
GUSMASK_STOP_VOLUME;
SELECT_GUS_REG(port, GUSREG_VOLUME_CONTROL);
outb(port+GUS_DATA_HIGH, sc->sc_voc[i].volcntl);
delay(100);
gus_set_samprate(sc, i, 8000);
SELECT_GUS_REG(port, GUSREG_START_ADDR_HIGH);
outw(port+GUS_DATA_LOW, 0x0000);
SELECT_GUS_REG(port, GUSREG_START_ADDR_LOW);
outw(port+GUS_DATA_LOW, 0x0000);
SELECT_GUS_REG(port, GUSREG_END_ADDR_HIGH);
outw(port+GUS_DATA_LOW, 0x0000);
SELECT_GUS_REG(port, GUSREG_END_ADDR_LOW);
outw(port+GUS_DATA_LOW, 0x0000);
SELECT_GUS_REG(port, GUSREG_VOLUME_RATE);
outb(port+GUS_DATA_HIGH, 0x01);
SELECT_GUS_REG(port, GUSREG_START_VOLUME);
outb(port+GUS_DATA_HIGH, 0x10);
SELECT_GUS_REG(port, GUSREG_END_VOLUME);
outb(port+GUS_DATA_HIGH, 0xe0);
SELECT_GUS_REG(port, GUSREG_CUR_VOLUME);
outw(port+GUS_DATA_LOW, 0x0000);
SELECT_GUS_REG(port, GUSREG_CUR_ADDR_HIGH);
outw(port+GUS_DATA_LOW, 0x0000);
SELECT_GUS_REG(port, GUSREG_CUR_ADDR_LOW);
outw(port+GUS_DATA_LOW, 0x0000);
SELECT_GUS_REG(port, GUSREG_PAN_POS);
outb(port+GUS_DATA_HIGH, 0x07);
}
/*
* Clear out any pending IRQs
*/
inb(port+GUS_IRQ_STATUS);
SELECT_GUS_REG(port, GUSREG_DMA_CONTROL);
inb(port+GUS_DATA_HIGH);
SELECT_GUS_REG(port, GUSREG_SAMPLE_CONTROL);
inb(port+GUS_DATA_HIGH);
SELECT_GUS_REG(port, GUSREG_IRQ_STATUS);
inb(port+GUS_DATA_HIGH);
SELECT_GUS_REG(port, GUSREG_RESET);
outb(port+GUS_DATA_HIGH, GUSMASK_MASTER_RESET | GUSMASK_DAC_ENABLE |
GUSMASK_IRQ_ENABLE);
splx(s);
}
STATIC void
gus_init_cs4231(sc)
struct gus_softc *sc;
{
int port = sc->sc_iobase;
u_char ctrl;
ctrl = (port & 0xf0) >> 4; /* set port address middle nibble */
/*
* The codec is a bit weird--swapped dma channels.
*/
ctrl |= GUS_MAX_CODEC_ENABLE;
if (sc->sc_drq >= 4)
ctrl |= GUS_MAX_RECCHAN16;
if (sc->sc_recdrq >= 4)
ctrl |= GUS_MAX_PLAYCHAN16;
outb(port+GUS_MAX_CTRL, ctrl);
sc->sc_codec.sc_iot = sc->sc_iot;
sc->sc_codec.sc_iobase = port+GUS_MAX_CODEC_BASE;
if (ad1848_probe(&sc->sc_codec) == 0) {
sc->sc_flags &= ~GUS_CODEC_INSTALLED;
} else {
struct ad1848_volume vol = {AUDIO_MAX_GAIN, AUDIO_MAX_GAIN};
static struct audio_hw_if gusmax_hw_if = {
gusmaxopen,
gusmax_close,
NULL, /* drain */
gus_query_encoding, /* query encoding */
gusmax_set_params,
gusmax_round_blocksize,
gusmax_set_out_port,
gusmax_get_out_port,
gusmax_set_in_port,
gusmax_get_in_port,
gusmax_commit_settings,
NULL,
NULL,
gusmax_dma_output,
gusmax_dma_input,
gusmax_halt_out_dma,
gusmax_halt_in_dma,
gusmax_cont_out_dma,
gusmax_cont_in_dma,
gusmax_speaker_ctl,
gus_getdev,
NULL,
gusmax_mixer_set_port,
gusmax_mixer_get_port,
gusmax_mixer_query_devinfo,
NULL,
NULL,
NULL,
NULL,
gusmax_get_props,
};
sc->sc_flags |= GUS_CODEC_INSTALLED;
sc->sc_codec.parent = sc;
sc->sc_codec.sc_drq = sc->sc_recdrq;
sc->sc_codec.sc_recdrq = sc->sc_drq;
gus_hw_if = gusmax_hw_if;
/* enable line in and mic in the GUS mixer; the codec chip
will do the real mixing for them. */
sc->sc_mixcontrol &= ~GUSMASK_LINE_IN; /* 0 enables. */
sc->sc_mixcontrol |= GUSMASK_MIC_IN; /* 1 enables. */
outb(sc->sc_iobase+GUS_MIX_CONTROL, sc->sc_mixcontrol);
ad1848_attach(&sc->sc_codec);
/* turn on pre-MUX microphone gain. */
ad1848_set_mic_gain(&sc->sc_codec, &vol);
}
}
/*
* Return info about the audio device, for the AUDIO_GETINFO ioctl
*/
int
gus_getdev(addr, dev)
void * addr;
struct audio_device *dev;
{
*dev = gus_device;
return 0;
}
/*
* stubs (XXX)
*/
int
gus_set_in_gain(addr, gain, balance)
caddr_t addr;
u_int gain;
u_char balance;
{
DPRINTF(("gus_set_in_gain called\n"));
return 0;
}
int
gus_get_in_gain(addr)
caddr_t addr;
{
DPRINTF(("gus_get_in_gain called\n"));
return 0;
}
int
gusmax_set_out_port(addr, port)
void * addr;
int port;
{
struct ad1848_softc *sc = addr;
return gus_set_out_port(sc->parent, port);
}
int
gus_set_out_port(addr, port)
void * addr;
int port;
{
struct gus_softc *sc = addr;
DPRINTF(("gus_set_out_port called\n"));
sc->sc_out_port = port;
return 0;
}
int
gusmax_get_out_port(addr)
void * addr;
{
struct ad1848_softc *sc = addr;
return gus_get_out_port(sc->parent);
}
int
gus_get_out_port(addr)
void * addr;
{
struct gus_softc *sc = addr;
DPRINTF(("gus_get_out_port() called\n"));
return sc->sc_out_port;
}
int
gusmax_set_in_port(addr, port)
void * addr;
int port;
{
struct ad1848_softc *sc = addr;
DPRINTF(("gusmax_set_in_port: %d\n", port));
switch(port) {
case MIC_IN_PORT:
case LINE_IN_PORT:
case AUX1_IN_PORT:
case DAC_IN_PORT:
break;
default:
return(EINVAL);
/*NOTREACHED*/
}
return(ad1848_set_rec_port(sc, port));
}
int
gusmax_get_in_port(addr)
void * addr;
{
struct ad1848_softc *sc = addr;
int port;
port = ad1848_get_rec_port(sc);
DPRINTF(("gusmax_get_in_port: %d\n", port));
return(port);
}
int
gus_set_in_port(addr, port)
void * addr;
int port;
{
struct gus_softc *sc = addr;
DPRINTF(("gus_set_in_port called\n"));
/*
* On the GUS with ICS mixer, the ADC input is after the mixer stage,
* so we can't set the input port.
*
* On the GUS with CS4231 codec/mixer, see gusmax_set_in_port().
*/
sc->sc_in_port = port;
return 0;
}
int
gus_get_in_port(addr)
void * addr;
{
struct gus_softc *sc = addr;
DPRINTF(("gus_get_in_port called\n"));
return sc->sc_in_port;
}
int
gusmax_dma_input(addr, buf, size, callback, arg)
void * addr;
void *buf;
int size;
void (*callback) __P((void *));
void *arg;
{
struct ad1848_softc *sc = addr;
return gus_dma_input(sc->parent, buf, size, callback, arg);
}
/*
* Start sampling the input source into the requested DMA buffer.
* Called at splgus(), either from top-half or from interrupt handler.
*/
int
gus_dma_input(addr, buf, size, callback, arg)
void * addr;
void *buf;
int size;
void (*callback) __P((void *));
void *arg;
{
struct gus_softc *sc = addr;
int port = sc->sc_iobase;
u_char dmac;
DMAPRINTF(("gus_dma_input called\n"));
/*
* Sample SIZE bytes of data from the card, into buffer at BUF.
*/
if (sc->sc_precision == 16)
return EINVAL; /* XXX */
/* set DMA modes */
dmac = GUSMASK_SAMPLE_IRQ|GUSMASK_SAMPLE_START;
if (sc->sc_recdrq >= 4)
dmac |= GUSMASK_SAMPLE_DATA16;
if (sc->sc_encoding == AUDIO_ENCODING_ULAW ||
sc->sc_encoding == AUDIO_ENCODING_ALAW ||
sc->sc_encoding == AUDIO_ENCODING_ULINEAR_LE ||
sc->sc_encoding == AUDIO_ENCODING_ULINEAR_BE)
dmac |= GUSMASK_SAMPLE_INVBIT;
if (sc->sc_channels == 2)
dmac |= GUSMASK_SAMPLE_STEREO;
isa_dmastart(sc->sc_dev.dv_parent, sc->sc_recdrq, buf, size,
NULL, DMAMODE_READ, BUS_DMA_NOWAIT);
DMAPRINTF(("gus_dma_input isa_dmastarted\n"));
sc->sc_flags |= GUS_DMAIN_ACTIVE;
sc->sc_dmainintr = callback;
sc->sc_inarg = arg;
sc->sc_dmaincnt = size;
sc->sc_dmainaddr = buf;
SELECT_GUS_REG(port, GUSREG_SAMPLE_CONTROL);
outb(port+GUS_DATA_HIGH, dmac); /* Go! */
DMAPRINTF(("gus_dma_input returning\n"));
return 0;
}
STATIC int
gus_dmain_intr(sc)
struct gus_softc *sc;
{
void (*callback) __P((void *));
void *arg;
DMAPRINTF(("gus_dmain_intr called\n"));
if (sc->sc_dmainintr) {
isa_dmadone(sc->sc_dev.dv_parent, sc->sc_recdrq);
callback = sc->sc_dmainintr;
arg = sc->sc_inarg;
sc->sc_dmainaddr = 0;
sc->sc_dmaincnt = 0;
sc->sc_dmainintr = 0;
sc->sc_inarg = 0;
sc->sc_flags &= ~GUS_DMAIN_ACTIVE;
DMAPRINTF(("calling dmain_intr callback %p(%p)\n", callback, arg));
(*callback)(arg);
return 1;
} else {
DMAPRINTF(("gus_dmain_intr false?\n"));
return 0; /* XXX ??? */
}
}
int
gusmax_halt_out_dma(addr)
void * addr;
{
struct ad1848_softc *sc = addr;
return gus_halt_out_dma(sc->parent);
}
int
gusmax_halt_in_dma(addr)
void * addr;
{
struct ad1848_softc *sc = addr;
return gus_halt_in_dma(sc->parent);
}
int
gusmax_cont_out_dma(addr)
void * addr;
{
struct ad1848_softc *sc = addr;
return gus_cont_out_dma(sc->parent);
}
int
gusmax_cont_in_dma(addr)
void * addr;
{
struct ad1848_softc *sc = addr;
return gus_cont_in_dma(sc->parent);
}
/*
* Stop any DMA output. Called at splgus().
*/
int
gus_halt_out_dma(addr)
void * addr;
{
struct gus_softc *sc = addr;
int port = sc->sc_iobase;
DMAPRINTF(("gus_halt_out_dma called\n"));
/*
* Make sure the GUS _isn't_ setup for DMA
*/
SELECT_GUS_REG(port, GUSREG_DMA_CONTROL);
outb(sc->sc_iobase+GUS_DATA_HIGH, 0);
untimeout(gus_dmaout_timeout, sc);
isa_dmaabort(sc->sc_dev.dv_parent, sc->sc_drq);
sc->sc_flags &= ~(GUS_DMAOUT_ACTIVE|GUS_LOCKED);
sc->sc_dmaoutintr = 0;
sc->sc_outarg = 0;
sc->sc_dmaoutaddr = 0;
sc->sc_dmaoutcnt = 0;
sc->sc_dmabuf = 0;
sc->sc_bufcnt = 0;
sc->sc_playbuf = -1;
/* also stop playing */
gus_stop_voice(sc, GUS_VOICE_LEFT, 1);
gus_stop_voice(sc, GUS_VOICE_RIGHT, 0);
return 0;
}
/*
* Stop any DMA output. Called at splgus().
*/
int
gus_halt_in_dma(addr)
void * addr;
{
struct gus_softc *sc = addr;
int port = sc->sc_iobase;
DMAPRINTF(("gus_halt_in_dma called\n"));
/*
* Make sure the GUS _isn't_ setup for DMA
*/
SELECT_GUS_REG(port, GUSREG_SAMPLE_CONTROL);
outb(port+GUS_DATA_HIGH,
inb(port+GUS_DATA_HIGH) & ~(GUSMASK_SAMPLE_START|GUSMASK_SAMPLE_IRQ));
isa_dmaabort(sc->sc_dev.dv_parent, sc->sc_recdrq);
sc->sc_flags &= ~GUS_DMAIN_ACTIVE;
sc->sc_dmainintr = 0;
sc->sc_inarg = 0;
sc->sc_dmainaddr = 0;
sc->sc_dmaincnt = 0;
return 0;
}
int
gus_cont_out_dma(addr)
void * addr;
{
DPRINTF(("gus_cont_out_dma called\n"));
return EOPNOTSUPP;
}
int
gus_cont_in_dma(addr)
void * addr;
{
DPRINTF(("gus_cont_in_dma called\n"));
return EOPNOTSUPP;
}
STATIC __inline int
gus_to_vol(cp, vol)
mixer_ctrl_t *cp;
struct ad1848_volume *vol;
{
if (cp->un.value.num_channels == 1) {
vol->left = vol->right = cp->un.value.level[AUDIO_MIXER_LEVEL_MONO];
return(1);
}
else if (cp->un.value.num_channels == 2) {
vol->left = cp->un.value.level[AUDIO_MIXER_LEVEL_LEFT];
vol->right = cp->un.value.level[AUDIO_MIXER_LEVEL_RIGHT];
return(1);
}
return(0);
}
STATIC __inline int
gus_from_vol(cp, vol)
mixer_ctrl_t *cp;
struct ad1848_volume *vol;
{
if (cp->un.value.num_channels == 1) {
cp->un.value.level[AUDIO_MIXER_LEVEL_MONO] = vol->left;
return(1);
}
else if (cp->un.value.num_channels == 2) {
cp->un.value.level[AUDIO_MIXER_LEVEL_LEFT] = vol->left;
cp->un.value.level[AUDIO_MIXER_LEVEL_RIGHT] = vol->right;
return(1);
}
return(0);
}
STATIC int
gusmax_mixer_get_port(addr, cp)
void *addr;
mixer_ctrl_t *cp;
{
struct ad1848_softc *ac = addr;
struct gus_softc *sc = ac->parent;
struct ad1848_volume vol;
int error = EINVAL;
DPRINTF(("gusmax_mixer_get_port: port=%d\n", cp->dev));
switch (cp->dev) {
#if 0 /* use mono level instead */
case GUSMAX_MIC_IN_LVL: /* Microphone */
if (cp->type == AUDIO_MIXER_VALUE) {
error = ad1848_get_mic_gain(ac, &vol);
if (!error)
gus_from_vol(cp, &vol);
}
break;
#endif
case GUSMAX_DAC_LVL: /* dac out */
if (cp->type == AUDIO_MIXER_VALUE) {
error = ad1848_get_aux1_gain(ac, &vol);
if (!error)
gus_from_vol(cp, &vol);
}
break;
case GUSMAX_LINE_IN_LVL: /* line in */
if (cp->type == AUDIO_MIXER_VALUE) {
error = cs4231_get_linein_gain(ac, &vol);
if (!error)
gus_from_vol(cp, &vol);
}
break;
case GUSMAX_MONO_LVL: /* mono */
if (cp->type == AUDIO_MIXER_VALUE &&
cp->un.value.num_channels == 1) {
error = cs4231_get_mono_gain(ac, &vol);
if (!error)
gus_from_vol(cp, &vol);
}
break;
case GUSMAX_CD_LVL: /* CD */
if (cp->type == AUDIO_MIXER_VALUE) {
error = ad1848_get_aux2_gain(ac, &vol);
if (!error)
gus_from_vol(cp, &vol);
}
break;
case GUSMAX_MONITOR_LVL: /* monitor level */
if (cp->type == AUDIO_MIXER_VALUE &&
cp->un.value.num_channels == 1) {
error = ad1848_get_mon_gain(ac, &vol);
if (!error)
cp->un.value.level[AUDIO_MIXER_LEVEL_MONO] =
vol.left;
}
break;
case GUSMAX_OUT_LVL: /* output level */
if (cp->type == AUDIO_MIXER_VALUE) {
error = ad1848_get_out_gain(ac, &vol);
if (!error)
gus_from_vol(cp, &vol);
}
break;
case GUSMAX_SPEAKER_LVL: /* fake speaker for mute naming */
if (cp->type == AUDIO_MIXER_VALUE) {
if (sc->sc_mixcontrol & GUSMASK_LINE_OUT)
vol.left = vol.right = AUDIO_MAX_GAIN;
else
vol.left = vol.right = AUDIO_MIN_GAIN;
error = 0;
gus_from_vol(cp, &vol);
}
break;
case GUSMAX_LINE_IN_MUTE:
if (cp->type == AUDIO_MIXER_ENUM) {
cp->un.ord = ac->line_mute;
error = 0;
}
break;
case GUSMAX_DAC_MUTE:
if (cp->type == AUDIO_MIXER_ENUM) {
cp->un.ord = ac->aux1_mute;
error = 0;
}
break;
case GUSMAX_CD_MUTE:
if (cp->type == AUDIO_MIXER_ENUM) {
cp->un.ord = ac->aux2_mute;
error = 0;
}
break;
case GUSMAX_MONO_MUTE:
if (cp->type == AUDIO_MIXER_ENUM) {
cp->un.ord = ac->mono_mute;
error = 0;
}
break;
case GUSMAX_MONITOR_MUTE:
if (cp->type == AUDIO_MIXER_ENUM) {
cp->un.ord = ac->mon_mute;
error = 0;
}
break;
case GUSMAX_SPEAKER_MUTE:
if (cp->type == AUDIO_MIXER_ENUM) {
cp->un.ord = sc->sc_mixcontrol & GUSMASK_LINE_OUT ? 1 : 0;
error = 0;
}
break;
case GUSMAX_REC_LVL: /* record level */
if (cp->type == AUDIO_MIXER_VALUE) {
error = ad1848_get_rec_gain(ac, &vol);
if (!error)
gus_from_vol(cp, &vol);
}
break;
case GUSMAX_RECORD_SOURCE:
if (cp->type == AUDIO_MIXER_ENUM) {
cp->un.ord = ad1848_get_rec_port(ac);
error = 0;
}
break;
default:
error = ENXIO;
break;
}
return(error);
}
STATIC int
gus_mixer_get_port(addr, cp)
void *addr;
mixer_ctrl_t *cp;
{
struct gus_softc *sc = addr;
struct ics2101_softc *ic = &sc->sc_mixer;
struct ad1848_volume vol;
int error = EINVAL;
DPRINTF(("gus_mixer_get_port: dev=%d type=%d\n", cp->dev, cp->type));
if (!HAS_MIXER(sc) && cp->dev > GUSICS_MASTER_MUTE)
return ENXIO;
switch (cp->dev) {
case GUSICS_MIC_IN_MUTE: /* Microphone */
if (cp->type == AUDIO_MIXER_ENUM) {
if (HAS_MIXER(sc))
cp->un.ord = ic->sc_mute[GUSMIX_CHAN_MIC][ICSMIX_LEFT];
else
cp->un.ord =
sc->sc_mixcontrol & GUSMASK_MIC_IN ? 0 : 1;
error = 0;
}
break;
case GUSICS_LINE_IN_MUTE:
if (cp->type == AUDIO_MIXER_ENUM) {
if (HAS_MIXER(sc))
cp->un.ord = ic->sc_mute[GUSMIX_CHAN_LINE][ICSMIX_LEFT];
else
cp->un.ord =
sc->sc_mixcontrol & GUSMASK_LINE_IN ? 1 : 0;
error = 0;
}
break;
case GUSICS_MASTER_MUTE:
if (cp->type == AUDIO_MIXER_ENUM) {
if (HAS_MIXER(sc))
cp->un.ord = ic->sc_mute[GUSMIX_CHAN_MASTER][ICSMIX_LEFT];
else
cp->un.ord =
sc->sc_mixcontrol & GUSMASK_LINE_OUT ? 1 : 0;
error = 0;
}
break;
case GUSICS_DAC_MUTE:
if (cp->type == AUDIO_MIXER_ENUM) {
cp->un.ord = ic->sc_mute[GUSMIX_CHAN_DAC][ICSMIX_LEFT];
error = 0;
}
break;
case GUSICS_CD_MUTE:
if (cp->type == AUDIO_MIXER_ENUM) {
cp->un.ord = ic->sc_mute[GUSMIX_CHAN_CD][ICSMIX_LEFT];
error = 0;
}
break;
case GUSICS_MASTER_LVL:
if (cp->type == AUDIO_MIXER_VALUE) {
vol.left = ic->sc_setting[GUSMIX_CHAN_MASTER][ICSMIX_LEFT];
vol.right = ic->sc_setting[GUSMIX_CHAN_MASTER][ICSMIX_RIGHT];
if (gus_from_vol(cp, &vol))
error = 0;
}
break;
case GUSICS_MIC_IN_LVL: /* Microphone */
if (cp->type == AUDIO_MIXER_VALUE) {
vol.left = ic->sc_setting[GUSMIX_CHAN_MIC][ICSMIX_LEFT];
vol.right = ic->sc_setting[GUSMIX_CHAN_MIC][ICSMIX_RIGHT];
if (gus_from_vol(cp, &vol))
error = 0;
}
break;
case GUSICS_LINE_IN_LVL: /* line in */
if (cp->type == AUDIO_MIXER_VALUE) {
vol.left = ic->sc_setting[GUSMIX_CHAN_LINE][ICSMIX_LEFT];
vol.right = ic->sc_setting[GUSMIX_CHAN_LINE][ICSMIX_RIGHT];
if (gus_from_vol(cp, &vol))
error = 0;
}
break;
case GUSICS_CD_LVL:
if (cp->type == AUDIO_MIXER_VALUE) {
vol.left = ic->sc_setting[GUSMIX_CHAN_CD][ICSMIX_LEFT];
vol.right = ic->sc_setting[GUSMIX_CHAN_CD][ICSMIX_RIGHT];
if (gus_from_vol(cp, &vol))
error = 0;
}
break;
case GUSICS_DAC_LVL: /* dac out */
if (cp->type == AUDIO_MIXER_VALUE) {
vol.left = ic->sc_setting[GUSMIX_CHAN_DAC][ICSMIX_LEFT];
vol.right = ic->sc_setting[GUSMIX_CHAN_DAC][ICSMIX_RIGHT];
if (gus_from_vol(cp, &vol))
error = 0;
}
break;
case GUSICS_RECORD_SOURCE:
if (cp->type == AUDIO_MIXER_ENUM) {
/* Can't set anything else useful, sigh. */
cp->un.ord = 0;
}
break;
default:
return ENXIO;
/*NOTREACHED*/
}
return error;
}
STATIC void
gusics_master_mute(ic, mute)
struct ics2101_softc *ic;
int mute;
{
ics2101_mix_mute(ic, GUSMIX_CHAN_MASTER, ICSMIX_LEFT, mute);
ics2101_mix_mute(ic, GUSMIX_CHAN_MASTER, ICSMIX_RIGHT, mute);
}
STATIC void
gusics_mic_mute(ic, mute)
struct ics2101_softc *ic;
int mute;
{
ics2101_mix_mute(ic, GUSMIX_CHAN_MIC, ICSMIX_LEFT, mute);
ics2101_mix_mute(ic, GUSMIX_CHAN_MIC, ICSMIX_RIGHT, mute);
}
STATIC void
gusics_linein_mute(ic, mute)
struct ics2101_softc *ic;
int mute;
{
ics2101_mix_mute(ic, GUSMIX_CHAN_LINE, ICSMIX_LEFT, mute);
ics2101_mix_mute(ic, GUSMIX_CHAN_LINE, ICSMIX_RIGHT, mute);
}
STATIC void
gusics_cd_mute(ic, mute)
struct ics2101_softc *ic;
int mute;
{
ics2101_mix_mute(ic, GUSMIX_CHAN_CD, ICSMIX_LEFT, mute);
ics2101_mix_mute(ic, GUSMIX_CHAN_CD, ICSMIX_RIGHT, mute);
}
STATIC void
gusics_dac_mute(ic, mute)
struct ics2101_softc *ic;
int mute;
{
ics2101_mix_mute(ic, GUSMIX_CHAN_DAC, ICSMIX_LEFT, mute);
ics2101_mix_mute(ic, GUSMIX_CHAN_DAC, ICSMIX_RIGHT, mute);
}
STATIC int
gusmax_mixer_set_port(addr, cp)
void *addr;
mixer_ctrl_t *cp;
{
struct ad1848_softc *ac = addr;
struct gus_softc *sc = ac->parent;
struct ad1848_volume vol;
int error = EINVAL;
DPRINTF(("gusmax_mixer_set_port: dev=%d type=%d\n", cp->dev, cp->type));
switch (cp->dev) {
#if 0
case GUSMAX_MIC_IN_LVL: /* Microphone */
if (cp->type == AUDIO_MIXER_VALUE &&
cp->un.value.num_channels == 1) {
/* XXX enable/disable pre-MUX fixed gain */
if (gus_to_vol(cp, &vol))
error = ad1848_set_mic_gain(ac, &vol);
}
break;
#endif
case GUSMAX_DAC_LVL: /* dac out */
if (cp->type == AUDIO_MIXER_VALUE) {
if (gus_to_vol(cp, &vol))
error = ad1848_set_aux1_gain(ac, &vol);
}
break;
case GUSMAX_LINE_IN_LVL: /* line in */
if (cp->type == AUDIO_MIXER_VALUE) {
if (gus_to_vol(cp, &vol))
error = cs4231_set_linein_gain(ac, &vol);
}
break;
case GUSMAX_MONO_LVL: /* mic/mono in */
if (cp->type == AUDIO_MIXER_VALUE &&
cp->un.value.num_channels == 1) {
if (gus_to_vol(cp, &vol))
error = cs4231_set_mono_gain(ac, &vol);
}
break;
case GUSMAX_CD_LVL: /* CD: AUX2 */
if (cp->type == AUDIO_MIXER_VALUE) {
if (gus_to_vol(cp, &vol))
error = ad1848_set_aux2_gain(ac, &vol);
}
break;
case GUSMAX_MONITOR_LVL:
if (cp->type == AUDIO_MIXER_VALUE &&
cp->un.value.num_channels == 1) {
vol.left = cp->un.value.level[AUDIO_MIXER_LEVEL_MONO];
error = ad1848_set_mon_gain(ac, &vol);
}
break;
case GUSMAX_OUT_LVL: /* output volume */
if (cp->type == AUDIO_MIXER_VALUE) {
if (gus_to_vol(cp, &vol))
error = ad1848_set_out_gain(ac, &vol);
}
break;
case GUSMAX_SPEAKER_LVL:
if (cp->type == AUDIO_MIXER_VALUE &&
cp->un.value.num_channels == 1) {
if (gus_to_vol(cp, &vol)) {
gus_speaker_ctl(sc, vol.left > AUDIO_MIN_GAIN ?
SPKR_ON : SPKR_OFF);
error = 0;
}
}
break;
case GUSMAX_LINE_IN_MUTE:
if (cp->type == AUDIO_MIXER_ENUM) {
ac->line_mute = cp->un.ord ? 1 : 0;
DPRINTF(("line mute %d\n", cp->un.ord));
cs4231_mute_line(ac, ac->line_mute);
gus_linein_ctl(sc, ac->line_mute ? SPKR_OFF : SPKR_ON);
error = 0;
}
break;
case GUSMAX_DAC_MUTE:
if (cp->type == AUDIO_MIXER_ENUM) {
ac->aux1_mute = cp->un.ord ? 1 : 0;
DPRINTF(("dac mute %d\n", cp->un.ord));
ad1848_mute_aux1(ac, ac->aux1_mute);
error = 0;
}
break;
case GUSMAX_CD_MUTE:
if (cp->type == AUDIO_MIXER_ENUM) {
ac->aux2_mute = cp->un.ord ? 1 : 0;
DPRINTF(("cd mute %d\n", cp->un.ord));
ad1848_mute_aux2(ac, ac->aux2_mute);
error = 0;
}
break;
case GUSMAX_MONO_MUTE: /* Microphone */
if (cp->type == AUDIO_MIXER_ENUM) {
ac->mono_mute = cp->un.ord ? 1 : 0;
DPRINTF(("mono mute %d\n", cp->un.ord));
cs4231_mute_mono(ac, ac->mono_mute);
gus_mic_ctl(sc, ac->mono_mute ? SPKR_OFF : SPKR_ON);
error = 0;
}
break;
case GUSMAX_MONITOR_MUTE:
if (cp->type == AUDIO_MIXER_ENUM) {
ac->mon_mute = cp->un.ord ? 1 : 0;
DPRINTF(("mono mute %d\n", cp->un.ord));
cs4231_mute_monitor(ac, ac->mon_mute);
error = 0;
}
break;
case GUSMAX_SPEAKER_MUTE:
if (cp->type == AUDIO_MIXER_ENUM) {
gus_speaker_ctl(sc, cp->un.ord ? SPKR_OFF : SPKR_ON);
error = 0;
}
break;
case GUSMAX_REC_LVL: /* record level */
if (cp->type == AUDIO_MIXER_VALUE) {
if (gus_to_vol(cp, &vol))
error = ad1848_set_rec_gain(ac, &vol);
}
break;
case GUSMAX_RECORD_SOURCE:
if (cp->type == AUDIO_MIXER_ENUM) {
error = ad1848_set_rec_port(ac, cp->un.ord);
}
break;
default:
return ENXIO;
/*NOTREACHED*/
}
return error;
}
STATIC int
gus_mixer_set_port(addr, cp)
void *addr;
mixer_ctrl_t *cp;
{
struct gus_softc *sc = addr;
struct ics2101_softc *ic = &sc->sc_mixer;
struct ad1848_volume vol;
int error = EINVAL;
DPRINTF(("gus_mixer_set_port: dev=%d type=%d\n", cp->dev, cp->type));
if (!HAS_MIXER(sc) && cp->dev > GUSICS_MASTER_MUTE)
return ENXIO;
switch (cp->dev) {
case GUSICS_MIC_IN_MUTE: /* Microphone */
if (cp->type == AUDIO_MIXER_ENUM) {
DPRINTF(("mic mute %d\n", cp->un.ord));
if (HAS_MIXER(sc)) {
gusics_mic_mute(ic, cp->un.ord);
}
gus_mic_ctl(sc, cp->un.ord ? SPKR_OFF : SPKR_ON);
error = 0;
}
break;
case GUSICS_LINE_IN_MUTE:
if (cp->type == AUDIO_MIXER_ENUM) {
DPRINTF(("linein mute %d\n", cp->un.ord));
if (HAS_MIXER(sc)) {
gusics_linein_mute(ic, cp->un.ord);
}
gus_linein_ctl(sc, cp->un.ord ? SPKR_OFF : SPKR_ON);
error = 0;
}
break;
case GUSICS_MASTER_MUTE:
if (cp->type == AUDIO_MIXER_ENUM) {
DPRINTF(("master mute %d\n", cp->un.ord));
if (HAS_MIXER(sc)) {
gusics_master_mute(ic, cp->un.ord);
}
gus_speaker_ctl(sc, cp->un.ord ? SPKR_OFF : SPKR_ON);
error = 0;
}
break;
case GUSICS_DAC_MUTE:
if (cp->type == AUDIO_MIXER_ENUM) {
gusics_dac_mute(ic, cp->un.ord);
error = 0;
}
break;
case GUSICS_CD_MUTE:
if (cp->type == AUDIO_MIXER_ENUM) {
gusics_cd_mute(ic, cp->un.ord);
error = 0;
}
break;
case GUSICS_MASTER_LVL:
if (cp->type == AUDIO_MIXER_VALUE) {
if (gus_to_vol(cp, &vol)) {
ics2101_mix_attenuate(ic,
GUSMIX_CHAN_MASTER,
ICSMIX_LEFT,
vol.left);
ics2101_mix_attenuate(ic,
GUSMIX_CHAN_MASTER,
ICSMIX_RIGHT,
vol.right);
error = 0;
}
}
break;
case GUSICS_MIC_IN_LVL: /* Microphone */
if (cp->type == AUDIO_MIXER_VALUE) {
if (gus_to_vol(cp, &vol)) {
ics2101_mix_attenuate(ic,
GUSMIX_CHAN_MIC,
ICSMIX_LEFT,
vol.left);
ics2101_mix_attenuate(ic,
GUSMIX_CHAN_MIC,
ICSMIX_RIGHT,
vol.right);
error = 0;
}
}
break;
case GUSICS_LINE_IN_LVL: /* line in */
if (cp->type == AUDIO_MIXER_VALUE) {
if (gus_to_vol(cp, &vol)) {
ics2101_mix_attenuate(ic,
GUSMIX_CHAN_LINE,
ICSMIX_LEFT,
vol.left);
ics2101_mix_attenuate(ic,
GUSMIX_CHAN_LINE,
ICSMIX_RIGHT,
vol.right);
error = 0;
}
}
break;
case GUSICS_CD_LVL:
if (cp->type == AUDIO_MIXER_VALUE) {
if (gus_to_vol(cp, &vol)) {
ics2101_mix_attenuate(ic,
GUSMIX_CHAN_CD,
ICSMIX_LEFT,
vol.left);
ics2101_mix_attenuate(ic,
GUSMIX_CHAN_CD,
ICSMIX_RIGHT,
vol.right);
error = 0;
}
}
break;
case GUSICS_DAC_LVL: /* dac out */
if (cp->type == AUDIO_MIXER_VALUE) {
if (gus_to_vol(cp, &vol)) {
ics2101_mix_attenuate(ic,
GUSMIX_CHAN_DAC,
ICSMIX_LEFT,
vol.left);
ics2101_mix_attenuate(ic,
GUSMIX_CHAN_DAC,
ICSMIX_RIGHT,
vol.right);
error = 0;
}
}
break;
case GUSICS_RECORD_SOURCE:
if (cp->type == AUDIO_MIXER_ENUM && cp->un.ord == 0) {
/* Can't set anything else useful, sigh. */
error = 0;
}
break;
default:
return ENXIO;
/*NOTREACHED*/
}
return error;
}
STATIC int
gus_get_props(addr)
void *addr;
{
struct gus_softc *sc = addr;
return sc->sc_recdrq == sc->sc_drq ? 0 : AUDIO_PROP_FULLDUPLEX;
}
STATIC int
gusmax_get_props(addr)
void *addr;
{
struct ad1848_softc *ac = addr;
return gus_get_props(ac->parent);
}
STATIC int
gusmax_mixer_query_devinfo(addr, dip)
void *addr;
mixer_devinfo_t *dip;
{
DPRINTF(("gusmax_query_devinfo: index=%d\n", dip->index));
switch(dip->index) {
#if 0
case GUSMAX_MIC_IN_LVL: /* Microphone */
dip->type = AUDIO_MIXER_VALUE;
dip->mixer_class = GUSMAX_INPUT_CLASS;
dip->prev = AUDIO_MIXER_LAST;
dip->next = GUSMAX_MIC_IN_MUTE;
strcpy(dip->label.name, AudioNmicrophone);
dip->un.v.num_channels = 2;
strcpy(dip->un.v.units.name, AudioNvolume);
break;
#endif
case GUSMAX_MONO_LVL: /* mono/microphone mixer */
dip->type = AUDIO_MIXER_VALUE;
dip->mixer_class = GUSMAX_INPUT_CLASS;
dip->prev = AUDIO_MIXER_LAST;
dip->next = GUSMAX_MONO_MUTE;
strcpy(dip->label.name, AudioNmicrophone);
dip->un.v.num_channels = 1;
strcpy(dip->un.v.units.name, AudioNvolume);
break;
case GUSMAX_DAC_LVL: /* dacout */
dip->type = AUDIO_MIXER_VALUE;
dip->mixer_class = GUSMAX_INPUT_CLASS;
dip->prev = AUDIO_MIXER_LAST;
dip->next = GUSMAX_DAC_MUTE;
strcpy(dip->label.name, AudioNdac);
dip->un.v.num_channels = 2;
strcpy(dip->un.v.units.name, AudioNvolume);
break;
case GUSMAX_LINE_IN_LVL: /* line */
dip->type = AUDIO_MIXER_VALUE;
dip->mixer_class = GUSMAX_INPUT_CLASS;
dip->prev = AUDIO_MIXER_LAST;
dip->next = GUSMAX_LINE_IN_MUTE;
strcpy(dip->label.name, AudioNline);
dip->un.v.num_channels = 2;
strcpy(dip->un.v.units.name, AudioNvolume);
break;
case GUSMAX_CD_LVL: /* cd */
dip->type = AUDIO_MIXER_VALUE;
dip->mixer_class = GUSMAX_INPUT_CLASS;
dip->prev = AUDIO_MIXER_LAST;
dip->next = GUSMAX_CD_MUTE;
strcpy(dip->label.name, AudioNcd);
dip->un.v.num_channels = 2;
strcpy(dip->un.v.units.name, AudioNvolume);
break;
case GUSMAX_MONITOR_LVL: /* monitor level */
dip->type = AUDIO_MIXER_VALUE;
dip->mixer_class = GUSMAX_MONITOR_CLASS;
dip->next = GUSMAX_MONITOR_MUTE;
dip->prev = AUDIO_MIXER_LAST;
strcpy(dip->label.name, AudioNmonitor);
dip->un.v.num_channels = 1;
strcpy(dip->un.v.units.name, AudioNvolume);
break;
case GUSMAX_OUT_LVL: /* cs4231 output volume: not useful? */
dip->type = AUDIO_MIXER_VALUE;
dip->mixer_class = GUSMAX_MONITOR_CLASS;
dip->prev = dip->next = AUDIO_MIXER_LAST;
strcpy(dip->label.name, AudioNoutput);
dip->un.v.num_channels = 2;
strcpy(dip->un.v.units.name, AudioNvolume);
break;
case GUSMAX_SPEAKER_LVL: /* fake speaker volume */
dip->type = AUDIO_MIXER_VALUE;
dip->mixer_class = GUSMAX_MONITOR_CLASS;
dip->prev = AUDIO_MIXER_LAST;
dip->next = GUSMAX_SPEAKER_MUTE;
strcpy(dip->label.name, AudioNspeaker);
dip->un.v.num_channels = 2;
strcpy(dip->un.v.units.name, AudioNvolume);
break;
case GUSMAX_LINE_IN_MUTE:
dip->mixer_class = GUSMAX_INPUT_CLASS;
dip->type = AUDIO_MIXER_ENUM;
dip->prev = GUSMAX_LINE_IN_LVL;
dip->next = AUDIO_MIXER_LAST;
goto mute;
case GUSMAX_DAC_MUTE:
dip->mixer_class = GUSMAX_INPUT_CLASS;
dip->type = AUDIO_MIXER_ENUM;
dip->prev = GUSMAX_DAC_LVL;
dip->next = AUDIO_MIXER_LAST;
goto mute;
case GUSMAX_CD_MUTE:
dip->mixer_class = GUSMAX_INPUT_CLASS;
dip->type = AUDIO_MIXER_ENUM;
dip->prev = GUSMAX_CD_LVL;
dip->next = AUDIO_MIXER_LAST;
goto mute;
case GUSMAX_MONO_MUTE:
dip->mixer_class = GUSMAX_INPUT_CLASS;
dip->type = AUDIO_MIXER_ENUM;
dip->prev = GUSMAX_MONO_LVL;
dip->next = AUDIO_MIXER_LAST;
goto mute;
case GUSMAX_MONITOR_MUTE:
dip->mixer_class = GUSMAX_OUTPUT_CLASS;
dip->type = AUDIO_MIXER_ENUM;
dip->prev = GUSMAX_MONITOR_LVL;
dip->next = AUDIO_MIXER_LAST;
goto mute;
case GUSMAX_SPEAKER_MUTE:
dip->mixer_class = GUSMAX_OUTPUT_CLASS;
dip->type = AUDIO_MIXER_ENUM;
dip->prev = GUSMAX_SPEAKER_LVL;
dip->next = AUDIO_MIXER_LAST;
mute:
strcpy(dip->label.name, AudioNmute);
dip->un.e.num_mem = 2;
strcpy(dip->un.e.member[0].label.name, AudioNoff);
dip->un.e.member[0].ord = 0;
strcpy(dip->un.e.member[1].label.name, AudioNon);
dip->un.e.member[1].ord = 1;
break;
case GUSMAX_REC_LVL: /* record level */
dip->type = AUDIO_MIXER_VALUE;
dip->mixer_class = GUSMAX_RECORD_CLASS;
dip->prev = AUDIO_MIXER_LAST;
dip->next = GUSMAX_RECORD_SOURCE;
strcpy(dip->label.name, AudioNrecord);
dip->un.v.num_channels = 2;
strcpy(dip->un.v.units.name, AudioNvolume);
break;
case GUSMAX_RECORD_SOURCE:
dip->mixer_class = GUSMAX_RECORD_CLASS;
dip->type = AUDIO_MIXER_ENUM;
dip->prev = GUSMAX_REC_LVL;
dip->next = AUDIO_MIXER_LAST;
strcpy(dip->label.name, AudioNsource);
dip->un.e.num_mem = 4;
strcpy(dip->un.e.member[0].label.name, AudioNoutput);
dip->un.e.member[0].ord = DAC_IN_PORT;
strcpy(dip->un.e.member[1].label.name, AudioNmicrophone);
dip->un.e.member[1].ord = MIC_IN_PORT;
strcpy(dip->un.e.member[2].label.name, AudioNdac);
dip->un.e.member[2].ord = AUX1_IN_PORT;
strcpy(dip->un.e.member[3].label.name, AudioNline);
dip->un.e.member[3].ord = LINE_IN_PORT;
break;
case GUSMAX_INPUT_CLASS: /* input class descriptor */
dip->type = AUDIO_MIXER_CLASS;
dip->mixer_class = GUSMAX_INPUT_CLASS;
dip->next = dip->prev = AUDIO_MIXER_LAST;
strcpy(dip->label.name, AudioCInputs);
break;
case GUSMAX_OUTPUT_CLASS: /* output class descriptor */
dip->type = AUDIO_MIXER_CLASS;
dip->mixer_class = GUSMAX_OUTPUT_CLASS;
dip->next = dip->prev = AUDIO_MIXER_LAST;
strcpy(dip->label.name, AudioCOutputs);
break;
case GUSMAX_MONITOR_CLASS: /* monitor class descriptor */
dip->type = AUDIO_MIXER_CLASS;
dip->mixer_class = GUSMAX_MONITOR_CLASS;
dip->next = dip->prev = AUDIO_MIXER_LAST;
strcpy(dip->label.name, AudioCMonitor);
break;
case GUSMAX_RECORD_CLASS: /* record source class */
dip->type = AUDIO_MIXER_CLASS;
dip->mixer_class = GUSMAX_RECORD_CLASS;
dip->next = dip->prev = AUDIO_MIXER_LAST;
strcpy(dip->label.name, AudioCRecord);
break;
default:
return ENXIO;
/*NOTREACHED*/
}
DPRINTF(("AUDIO_MIXER_DEVINFO: name=%s\n", dip->label.name));
return 0;
}
STATIC int
gus_mixer_query_devinfo(addr, dip)
void *addr;
mixer_devinfo_t *dip;
{
struct gus_softc *sc = addr;
DPRINTF(("gusmax_query_devinfo: index=%d\n", dip->index));
if (!HAS_MIXER(sc) && dip->index > GUSICS_MASTER_MUTE)
return ENXIO;
switch(dip->index) {
case GUSICS_MIC_IN_LVL: /* Microphone */
dip->type = AUDIO_MIXER_VALUE;
dip->mixer_class = GUSICS_INPUT_CLASS;
dip->prev = AUDIO_MIXER_LAST;
dip->next = GUSICS_MIC_IN_MUTE;
strcpy(dip->label.name, AudioNmicrophone);
dip->un.v.num_channels = 2;
strcpy(dip->un.v.units.name, AudioNvolume);
break;
case GUSICS_LINE_IN_LVL: /* line */
dip->type = AUDIO_MIXER_VALUE;
dip->mixer_class = GUSICS_INPUT_CLASS;
dip->prev = AUDIO_MIXER_LAST;
dip->next = GUSICS_LINE_IN_MUTE;
strcpy(dip->label.name, AudioNline);
dip->un.v.num_channels = 2;
strcpy(dip->un.v.units.name, AudioNvolume);
break;
case GUSICS_CD_LVL: /* cd */
dip->type = AUDIO_MIXER_VALUE;
dip->mixer_class = GUSICS_INPUT_CLASS;
dip->prev = AUDIO_MIXER_LAST;
dip->next = GUSICS_CD_MUTE;
strcpy(dip->label.name, AudioNcd);
dip->un.v.num_channels = 2;
strcpy(dip->un.v.units.name, AudioNvolume);
break;
case GUSICS_DAC_LVL: /* dacout */
dip->type = AUDIO_MIXER_VALUE;
dip->mixer_class = GUSICS_INPUT_CLASS;
dip->prev = AUDIO_MIXER_LAST;
dip->next = GUSICS_DAC_MUTE;
strcpy(dip->label.name, AudioNdac);
dip->un.v.num_channels = 2;
strcpy(dip->un.v.units.name, AudioNvolume);
break;
case GUSICS_MASTER_LVL: /* master output */
dip->type = AUDIO_MIXER_VALUE;
dip->mixer_class = GUSICS_OUTPUT_CLASS;
dip->prev = AUDIO_MIXER_LAST;
dip->next = GUSICS_MASTER_MUTE;
strcpy(dip->label.name, AudioNvolume);
dip->un.v.num_channels = 2;
strcpy(dip->un.v.units.name, AudioNvolume);
break;
case GUSICS_LINE_IN_MUTE:
dip->mixer_class = GUSICS_INPUT_CLASS;
dip->type = AUDIO_MIXER_ENUM;
dip->prev = GUSICS_LINE_IN_LVL;
dip->next = AUDIO_MIXER_LAST;
goto mute;
case GUSICS_DAC_MUTE:
dip->mixer_class = GUSICS_INPUT_CLASS;
dip->type = AUDIO_MIXER_ENUM;
dip->prev = GUSICS_DAC_LVL;
dip->next = AUDIO_MIXER_LAST;
goto mute;
case GUSICS_CD_MUTE:
dip->mixer_class = GUSICS_INPUT_CLASS;
dip->type = AUDIO_MIXER_ENUM;
dip->prev = GUSICS_CD_LVL;
dip->next = AUDIO_MIXER_LAST;
goto mute;
case GUSICS_MIC_IN_MUTE:
dip->mixer_class = GUSICS_INPUT_CLASS;
dip->type = AUDIO_MIXER_ENUM;
dip->prev = GUSICS_MIC_IN_LVL;
dip->next = AUDIO_MIXER_LAST;
goto mute;
case GUSICS_MASTER_MUTE:
dip->mixer_class = GUSICS_OUTPUT_CLASS;
dip->type = AUDIO_MIXER_ENUM;
dip->prev = GUSICS_MASTER_LVL;
dip->next = AUDIO_MIXER_LAST;
mute:
strcpy(dip->label.name, AudioNmute);
dip->un.e.num_mem = 2;
strcpy(dip->un.e.member[0].label.name, AudioNoff);
dip->un.e.member[0].ord = 0;
strcpy(dip->un.e.member[1].label.name, AudioNon);
dip->un.e.member[1].ord = 1;
break;
case GUSICS_RECORD_SOURCE:
dip->mixer_class = GUSICS_RECORD_CLASS;
dip->type = AUDIO_MIXER_ENUM;
dip->prev = dip->next = AUDIO_MIXER_LAST;
strcpy(dip->label.name, AudioNsource);
dip->un.e.num_mem = 1;
strcpy(dip->un.e.member[0].label.name, AudioNoutput);
dip->un.e.member[0].ord = GUSICS_MASTER_LVL;
break;
case GUSICS_INPUT_CLASS:
dip->type = AUDIO_MIXER_CLASS;
dip->mixer_class = GUSICS_INPUT_CLASS;
dip->next = dip->prev = AUDIO_MIXER_LAST;
strcpy(dip->label.name, AudioCInputs);
break;
case GUSICS_OUTPUT_CLASS:
dip->type = AUDIO_MIXER_CLASS;
dip->mixer_class = GUSICS_OUTPUT_CLASS;
dip->next = dip->prev = AUDIO_MIXER_LAST;
strcpy(dip->label.name, AudioCOutputs);
break;
case GUSICS_RECORD_CLASS:
dip->type = AUDIO_MIXER_CLASS;
dip->mixer_class = GUSICS_RECORD_CLASS;
dip->next = dip->prev = AUDIO_MIXER_LAST;
strcpy(dip->label.name, AudioCRecord);
break;
default:
return ENXIO;
/*NOTREACHED*/
}
DPRINTF(("AUDIO_MIXER_DEVINFO: name=%s\n", dip->label.name));
return 0;
}
STATIC int
gus_query_encoding(addr, fp)
void *addr;
struct audio_encoding *fp;
{
switch (fp->index) {
case 0:
strcpy(fp->name, AudioEmulaw);
fp->encoding = AUDIO_ENCODING_ULAW;
fp->precision = 8;
fp->flags = AUDIO_ENCODINGFLAG_EMULATED;
break;
case 1:
strcpy(fp->name, AudioElinear);
fp->encoding = AUDIO_ENCODING_SLINEAR;
fp->precision = 8;
fp->flags = 0;
break;
case 2:
strcpy(fp->name, AudioElinear_le);
fp->encoding = AUDIO_ENCODING_SLINEAR_LE;
fp->precision = 16;
fp->flags = 0;
break;
case 3:
strcpy(fp->name, AudioEulinear);
fp->encoding = AUDIO_ENCODING_ULINEAR;
fp->precision = 8;
fp->flags = 0;
break;
case 4:
strcpy(fp->name, AudioEulinear_le);
fp->encoding = AUDIO_ENCODING_ULINEAR_LE;
fp->precision = 16;
fp->flags = 0;
break;
case 5:
strcpy(fp->name, AudioElinear_be);
fp->encoding = AUDIO_ENCODING_SLINEAR_BE;
fp->precision = 16;
fp->flags = AUDIO_ENCODINGFLAG_EMULATED;
break;
case 6:
strcpy(fp->name, AudioEulinear_be);
fp->encoding = AUDIO_ENCODING_ULINEAR_BE;
fp->precision = 16;
fp->flags = AUDIO_ENCODINGFLAG_EMULATED;
break;
case 7:
strcpy(fp->name, AudioEalaw);
fp->encoding = AUDIO_ENCODING_ALAW;
fp->precision = 8;
fp->flags = AUDIO_ENCODINGFLAG_EMULATED;
break;
default:
return(EINVAL);
/*NOTREACHED*/
}
return (0);
}
/*
* Setup the ICS mixer in "transparent" mode: reset everything to a sensible
* level. Levels as suggested by GUS SDK code.
*/
STATIC void
gus_init_ics2101(sc)
struct gus_softc *sc;
{
int port = sc->sc_iobase;
struct ics2101_softc *ic = &sc->sc_mixer;
sc->sc_mixer.sc_selio = port+GUS_MIXER_SELECT;
sc->sc_mixer.sc_dataio = port+GUS_MIXER_DATA;
sc->sc_mixer.sc_flags = (sc->sc_revision == 5) ? ICS_FLIP : 0;
ics2101_mix_attenuate(ic,
GUSMIX_CHAN_MIC,
ICSMIX_LEFT,
ICSMIX_MIN_ATTN);
ics2101_mix_attenuate(ic,
GUSMIX_CHAN_MIC,
ICSMIX_RIGHT,
ICSMIX_MIN_ATTN);
/*
* Start with microphone muted by the mixer...
*/
gusics_mic_mute(ic, 1);
/* ... and enabled by the GUS master mix control */
gus_mic_ctl(sc, SPKR_ON);
ics2101_mix_attenuate(ic,
GUSMIX_CHAN_LINE,
ICSMIX_LEFT,
ICSMIX_MIN_ATTN);
ics2101_mix_attenuate(ic,
GUSMIX_CHAN_LINE,
ICSMIX_RIGHT,
ICSMIX_MIN_ATTN);
ics2101_mix_attenuate(ic,
GUSMIX_CHAN_CD,
ICSMIX_LEFT,
ICSMIX_MIN_ATTN);
ics2101_mix_attenuate(ic,
GUSMIX_CHAN_CD,
ICSMIX_RIGHT,
ICSMIX_MIN_ATTN);
ics2101_mix_attenuate(ic,
GUSMIX_CHAN_DAC,
ICSMIX_LEFT,
ICSMIX_MIN_ATTN);
ics2101_mix_attenuate(ic,
GUSMIX_CHAN_DAC,
ICSMIX_RIGHT,
ICSMIX_MIN_ATTN);
ics2101_mix_attenuate(ic,
ICSMIX_CHAN_4,
ICSMIX_LEFT,
ICSMIX_MAX_ATTN);
ics2101_mix_attenuate(ic,
ICSMIX_CHAN_4,
ICSMIX_RIGHT,
ICSMIX_MAX_ATTN);
ics2101_mix_attenuate(ic,
GUSMIX_CHAN_MASTER,
ICSMIX_LEFT,
ICSMIX_MIN_ATTN);
ics2101_mix_attenuate(ic,
GUSMIX_CHAN_MASTER,
ICSMIX_RIGHT,
ICSMIX_MIN_ATTN);
/* unmute other stuff: */
gusics_cd_mute(ic, 0);
gusics_dac_mute(ic, 0);
gusics_linein_mute(ic, 0);
return;
}
#endif /* NGUS */
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