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diff --git a/sound/oss/maestro.c b/sound/oss/maestro.c
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+/*****************************************************************************
+ *
+ * ESS Maestro/Maestro-2/Maestro-2E driver for Linux 2.[23].x
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
+ *
+ * (c) Copyright 1999 Alan Cox <alan.cox@linux.org>
+ *
+ * Based heavily on SonicVibes.c:
+ * Copyright (C) 1998-1999 Thomas Sailer (sailer@ife.ee.ethz.ch)
+ *
+ * Heavily modified by Zach Brown <zab@zabbo.net> based on lunch
+ * with ESS engineers. Many thanks to Howard Kim for providing
+ * contacts and hardware. Honorable mention goes to Eric
+ * Brombaugh for all sorts of things. Best regards to the
+ * proprietors of Hack Central for fine lodging.
+ *
+ * Supported devices:
+ * /dev/dsp0-3 standard /dev/dsp device, (mostly) OSS compatible
+ * /dev/mixer standard /dev/mixer device, (mostly) OSS compatible
+ *
+ * Hardware Description
+ *
+ * A working Maestro setup contains the Maestro chip wired to a
+ * codec or 2. In the Maestro we have the APUs, the ASSP, and the
+ * Wavecache. The APUs can be though of as virtual audio routing
+ * channels. They can take data from a number of sources and perform
+ * basic encodings of the data. The wavecache is a storehouse for
+ * PCM data. Typically it deals with PCI and interracts with the
+ * APUs. The ASSP is a wacky DSP like device that ESS is loth
+ * to release docs on. Thankfully it isn't required on the Maestro
+ * until you start doing insane things like FM emulation and surround
+ * encoding. The codecs are almost always AC-97 compliant codecs,
+ * but it appears that early Maestros may have had PT101 (an ESS
+ * part?) wired to them. The only real difference in the Maestro
+ * families is external goop like docking capability, memory for
+ * the ASSP, and initialization differences.
+ *
+ * Driver Operation
+ *
+ * We only drive the APU/Wavecache as typical DACs and drive the
+ * mixers in the codecs. There are 64 APUs. We assign 6 to each
+ * /dev/dsp? device. 2 channels for output, and 4 channels for
+ * input.
+ *
+ * Each APU can do a number of things, but we only really use
+ * 3 basic functions. For playback we use them to convert PCM
+ * data fetched over PCI by the wavecahche into analog data that
+ * is handed to the codec. One APU for mono, and a pair for stereo.
+ * When in stereo, the combination of smarts in the APU and Wavecache
+ * decide which wavecache gets the left or right channel.
+ *
+ * For record we still use the old overly mono system. For each in
+ * coming channel the data comes in from the codec, through a 'input'
+ * APU, through another rate converter APU, and then into memory via
+ * the wavecache and PCI. If its stereo, we mash it back into LRLR in
+ * software. The pass between the 2 APUs is supposedly what requires us
+ * to have a 512 byte buffer sitting around in wavecache/memory.
+ *
+ * The wavecache makes our life even more fun. First off, it can
+ * only address the first 28 bits of PCI address space, making it
+ * useless on quite a few architectures. Secondly, its insane.
+ * It claims to fetch from 4 regions of PCI space, each 4 meg in length.
+ * But that doesn't really work. You can only use 1 region. So all our
+ * allocations have to be in 4meg of each other. Booo. Hiss.
+ * So we have a module parameter, dsps_order, that is the order of
+ * the number of dsps to provide. All their buffer space is allocated
+ * on open time. The sonicvibes OSS routines we inherited really want
+ * power of 2 buffers, so we have all those next to each other, then
+ * 512 byte regions for the recording wavecaches. This ends up
+ * wasting quite a bit of memory. The only fixes I can see would be
+ * getting a kernel allocator that could work in zones, or figuring out
+ * just how to coerce the WP into doing what we want.
+ *
+ * The indirection of the various registers means we have to spinlock
+ * nearly all register accesses. We have the main register indirection
+ * like the wave cache, maestro registers, etc. Then we have beasts
+ * like the APU interface that is indirect registers gotten at through
+ * the main maestro indirection. Ouch. We spinlock around the actual
+ * ports on a per card basis. This means spinlock activity at each IO
+ * operation, but the only IO operation clusters are in non critical
+ * paths and it makes the code far easier to follow. Interrupts are
+ * blocked while holding the locks because the int handler has to
+ * get at some of them :(. The mixer interface doesn't, however.
+ * We also have an OSS state lock that is thrown around in a few
+ * places.
+ *
+ * This driver has brute force APM suspend support. We catch suspend
+ * notifications and stop all work being done on the chip. Any people
+ * that try between this shutdown and the real suspend operation will
+ * be put to sleep. When we resume we restore our software state on
+ * the chip and wake up the people that were using it. The code thats
+ * being used now is quite dirty and assumes we're on a uni-processor
+ * machine. Much of it will need to be cleaned up for SMP ACPI or
+ * similar.
+ *
+ * We also pay attention to PCI power management now. The driver
+ * will power down units of the chip that it knows aren't needed.
+ * The WaveProcessor and company are only powered on when people
+ * have /dev/dsp*s open. On removal the driver will
+ * power down the maestro entirely. There could still be
+ * trouble with BIOSen that magically change power states
+ * themselves, but we'll see.
+ *
+ * History
+ * v0.15 - May 21 2001 - Marcus Meissner <mm@caldera.de>
+ * Ported to Linux 2.4 PCI API. Some clean ups, global devs list
+ * removed (now using pci device driver data).
+ * PM needs to be polished still. Bumped version.
+ * (still kind of v0.14) May 13 2001 - Ben Pfaff <pfaffben@msu.edu>
+ * Add support for 978 docking and basic hardware volume control
+ * (still kind of v0.14) Nov 23 - Alan Cox <alan@redhat.com>
+ * Add clocking= for people with seriously warped hardware
+ * (still v0.14) Nov 10 2000 - Bartlomiej Zolnierkiewicz <bkz@linux-ide.org>
+ * add __init to maestro_ac97_init() and maestro_install()
+ * (still based on v0.14) Mar 29 2000 - Zach Brown <zab@redhat.com>
+ * move to 2.3 power management interface, which
+ * required hacking some suspend/resume/check paths
+ * make static compilation work
+ * v0.14 - Jan 28 2000 - Zach Brown <zab@redhat.com>
+ * add PCI power management through ACPI regs.
+ * we now shut down on machine reboot/halt
+ * leave scary PCI config items alone (isa stuff, mostly)
+ * enable 1921s, it seems only mine was broke.
+ * fix swapped left/right pcm dac. har har.
+ * up bob freq, increase buffers, fix pointers at underflow
+ * silly compilation problems
+ * v0.13 - Nov 18 1999 - Zach Brown <zab@redhat.com>
+ * fix nec Versas? man would that be cool.
+ * v0.12 - Nov 12 1999 - Zach Brown <zab@redhat.com>
+ * brown bag volume max fix..
+ * v0.11 - Nov 11 1999 - Zach Brown <zab@redhat.com>
+ * use proper stereo apu decoding, mmap/write should work.
+ * make volume sliders more useful, tweak rate calculation.
+ * fix lame 8bit format reporting bug. duh. apm apu saving buglet also
+ * fix maestro 1 clock freq "bug", remove pt101 support
+ * v0.10 - Oct 28 1999 - Zach Brown <zab@redhat.com>
+ * aha, so, sometimes the WP writes a status word to offset 0
+ * from one of the PCMBARs. rearrange allocation accordingly..
+ * cheers again to Eric for being a good hacker in investigating this.
+ * Jeroen Hoogervorst submits 7500 fix out of nowhere. yay. :)
+ * v0.09 - Oct 23 1999 - Zach Brown <zab@redhat.com>
+ * added APM support.
+ * re-order something such that some 2Es now work. Magic!
+ * new codec reset routine. made some codecs come to life.
+ * fix clear_advance, sync some control with ESS.
+ * now write to all base regs to be paranoid.
+ * v0.08 - Oct 20 1999 - Zach Brown <zab@redhat.com>
+ * Fix initial buflen bug. I am so smart. also smp compiling..
+ * I owe Eric yet another beer: fixed recmask, igain,
+ * muting, and adc sync consistency. Go Team.
+ * v0.07 - Oct 4 1999 - Zach Brown <zab@redhat.com>
+ * tweak adc/dac, formating, and stuff to allow full duplex
+ * allocate dsps memory at open() so we can fit in the wavecache window
+ * fix wavecache braindamage. again. no more scribbling?
+ * fix ess 1921 codec bug on some laptops.
+ * fix dumb pci scanning bug
+ * started 2.3 cleanup, redid spinlocks, little cleanups
+ * v0.06 - Sep 20 1999 - Zach Brown <zab@redhat.com>
+ * fix wavecache thinkos. limit to 1 /dev/dsp.
+ * eric is wearing his thinking toque this week.
+ * spotted apu mode bugs and gain ramping problem
+ * don't touch weird mixer regs, make recmask optional
+ * fixed igain inversion, defaults for mixers, clean up rec_start
+ * make mono recording work.
+ * report subsystem stuff, please send reports.
+ * littles: parallel out, amp now
+ * v0.05 - Sep 17 1999 - Zach Brown <zab@redhat.com>
+ * merged and fixed up Eric's initial recording code
+ * munged format handling to catch misuse, needs rewrite.
+ * revert ring bus init, fixup shared int, add pci busmaster setting
+ * fix mixer oss interface, fix mic mute and recmask
+ * mask off unsupported mixers, reset with all 1s, modularize defaults
+ * make sure bob is running while we need it
+ * got rid of device limit, initial minimal apm hooks
+ * pull out dead code/includes, only allow multimedia/audio maestros
+ * v0.04 - Sep 01 1999 - Zach Brown <zab@redhat.com>
+ * copied memory leak fix from sonicvibes driver
+ * different ac97 reset, play with 2.0 ac97, simplify ring bus setup
+ * bob freq code, region sanity, jitter sync fix; all from Eric
+ *
+ * TODO
+ * fix bob frequency
+ * endianness
+ * do smart things with ac97 2.0 bits.
+ * dual codecs
+ * leave 54->61 open
+ *
+ * it also would be fun to have a mode that would not use pci dma at all
+ * but would copy into the wavecache on board memory and use that
+ * on architectures that don't like the maestro's pci dma ickiness.
+ */
+
+/*****************************************************************************/
+
+#include <linux/module.h>
+#include <linux/sched.h>
+#include <linux/smp_lock.h>
+#include <linux/string.h>
+#include <linux/ctype.h>
+#include <linux/ioport.h>
+#include <linux/delay.h>
+#include <linux/sound.h>
+#include <linux/slab.h>
+#include <linux/soundcard.h>
+#include <linux/pci.h>
+#include <linux/spinlock.h>
+#include <linux/init.h>
+#include <linux/interrupt.h>
+#include <linux/poll.h>
+#include <linux/reboot.h>
+#include <linux/bitops.h>
+#include <linux/wait.h>
+
+#include <asm/current.h>
+#include <asm/dma.h>
+#include <asm/io.h>
+#include <asm/page.h>
+#include <asm/uaccess.h>
+
+#include <linux/pm.h>
+static int maestro_pm_callback(struct pm_dev *dev, pm_request_t rqst, void *d);
+
+#include "maestro.h"
+
+static struct pci_driver maestro_pci_driver;
+
+/* --------------------------------------------------------------------- */
+
+#define M_DEBUG 1
+
+#ifdef M_DEBUG
+static int debug;
+#define M_printk(args...) {if (debug) printk(args);}
+#else
+#define M_printk(x)
+#endif
+
+/* we try to setup 2^(dsps_order) /dev/dsp devices */
+static int dsps_order;
+/* whether or not we mess around with power management */
+static int use_pm=2; /* set to 1 for force */
+/* clocking for broken hardware - a few laptops seem to use a 50Khz clock
+ ie insmod with clocking=50000 or so */
+
+static int clocking=48000;
+
+MODULE_AUTHOR("Zach Brown <zab@zabbo.net>, Alan Cox <alan@redhat.com>");
+MODULE_DESCRIPTION("ESS Maestro Driver");
+MODULE_LICENSE("GPL");
+
+#ifdef M_DEBUG
+module_param(debug, bool, 0644);
+#endif
+module_param(dsps_order, int, 0);
+module_param(use_pm, int, 0);
+module_param(clocking, int, 0);
+
+/* --------------------------------------------------------------------- */
+#define DRIVER_VERSION "0.15"
+
+#ifndef PCI_VENDOR_ESS
+#define PCI_VENDOR_ESS 0x125D
+#define PCI_DEVICE_ID_ESS_ESS1968 0x1968 /* Maestro 2 */
+#define PCI_DEVICE_ID_ESS_ESS1978 0x1978 /* Maestro 2E */
+
+#define PCI_VENDOR_ESS_OLD 0x1285 /* Platform Tech,
+ the people the maestro
+ was bought from */
+#define PCI_DEVICE_ID_ESS_ESS0100 0x0100 /* maestro 1 */
+#endif /* PCI_VENDOR_ESS */
+
+#define ESS_CHAN_HARD 0x100
+
+/* NEC Versas ? */
+#define NEC_VERSA_SUBID1 0x80581033
+#define NEC_VERSA_SUBID2 0x803c1033
+
+
+/* changed so that I could actually find all the
+ references and fix them up. it's a little more readable now. */
+#define ESS_FMT_STEREO 0x01
+#define ESS_FMT_16BIT 0x02
+#define ESS_FMT_MASK 0x03
+#define ESS_DAC_SHIFT 0
+#define ESS_ADC_SHIFT 4
+
+#define ESS_STATE_MAGIC 0x125D1968
+#define ESS_CARD_MAGIC 0x19283746
+
+#define DAC_RUNNING 1
+#define ADC_RUNNING 2
+
+#define MAX_DSP_ORDER 2
+#define MAX_DSPS (1<<MAX_DSP_ORDER)
+#define NR_DSPS (1<<dsps_order)
+#define NR_IDRS 32
+
+#define NR_APUS 64
+#define NR_APU_REGS 16
+
+/* acpi states */
+enum {
+ ACPI_D0=0,
+ ACPI_D1,
+ ACPI_D2,
+ ACPI_D3
+};
+
+/* bits in the acpi masks */
+#define ACPI_12MHZ ( 1 << 15)
+#define ACPI_24MHZ ( 1 << 14)
+#define ACPI_978 ( 1 << 13)
+#define ACPI_SPDIF ( 1 << 12)
+#define ACPI_GLUE ( 1 << 11)
+#define ACPI__10 ( 1 << 10) /* reserved */
+#define ACPI_PCIINT ( 1 << 9)
+#define ACPI_HV ( 1 << 8) /* hardware volume */
+#define ACPI_GPIO ( 1 << 7)
+#define ACPI_ASSP ( 1 << 6)
+#define ACPI_SB ( 1 << 5) /* sb emul */
+#define ACPI_FM ( 1 << 4) /* fm emul */
+#define ACPI_RB ( 1 << 3) /* ringbus / aclink */
+#define ACPI_MIDI ( 1 << 2)
+#define ACPI_GP ( 1 << 1) /* game port */
+#define ACPI_WP ( 1 << 0) /* wave processor */
+
+#define ACPI_ALL (0xffff)
+#define ACPI_SLEEP (~(ACPI_SPDIF|ACPI_ASSP|ACPI_SB|ACPI_FM| \
+ ACPI_MIDI|ACPI_GP|ACPI_WP))
+#define ACPI_NONE (ACPI__10)
+
+/* these masks indicate which units we care about at
+ which states */
+static u16 acpi_state_mask[] = {
+ [ACPI_D0] = ACPI_ALL,
+ [ACPI_D1] = ACPI_SLEEP,
+ [ACPI_D2] = ACPI_SLEEP,
+ [ACPI_D3] = ACPI_NONE
+};
+
+static char version[] __devinitdata =
+KERN_INFO "maestro: version " DRIVER_VERSION " time " __TIME__ " " __DATE__ "\n";
+
+
+
+static const unsigned sample_size[] = { 1, 2, 2, 4 };
+static const unsigned sample_shift[] = { 0, 1, 1, 2 };
+
+enum card_types_t {
+ TYPE_MAESTRO,
+ TYPE_MAESTRO2,
+ TYPE_MAESTRO2E
+};
+
+static const char *card_names[]={
+ [TYPE_MAESTRO] = "ESS Maestro",
+ [TYPE_MAESTRO2] = "ESS Maestro 2",
+ [TYPE_MAESTRO2E] = "ESS Maestro 2E"
+};
+
+static int clock_freq[]={
+ [TYPE_MAESTRO] = (49152000L / 1024L),
+ [TYPE_MAESTRO2] = (50000000L / 1024L),
+ [TYPE_MAESTRO2E] = (50000000L / 1024L)
+};
+
+static int maestro_notifier(struct notifier_block *nb, unsigned long event, void *buf);
+
+static struct notifier_block maestro_nb = {maestro_notifier, NULL, 0};
+
+/* --------------------------------------------------------------------- */
+
+struct ess_state {
+ unsigned int magic;
+ /* FIXME: we probably want submixers in here, but only one record pair */
+ u8 apu[6]; /* l/r output, l/r intput converters, l/r input apus */
+ u8 apu_mode[6]; /* Running mode for this APU */
+ u8 apu_pan[6]; /* Panning setup for this APU */
+ u32 apu_base[6]; /* base address for this apu */
+ struct ess_card *card; /* Card info */
+ /* wave stuff */
+ unsigned int rateadc, ratedac;
+ unsigned char fmt, enable;
+
+ int index;
+
+ /* this locks around the oss state in the driver */
+ spinlock_t lock;
+ /* only let 1 be opening at a time */
+ struct semaphore open_sem;
+ wait_queue_head_t open_wait;
+ mode_t open_mode;
+
+ /* soundcore stuff */
+ int dev_audio;
+
+ struct dmabuf {
+ void *rawbuf;
+ unsigned buforder;
+ unsigned numfrag;
+ unsigned fragshift;
+ /* XXX zab - swptr only in here so that it can be referenced by
+ clear_advance, as far as I can tell :( */
+ unsigned hwptr, swptr;
+ unsigned total_bytes;
+ int count;
+ unsigned error; /* over/underrun */
+ wait_queue_head_t wait;
+ /* redundant, but makes calculations easier */
+ unsigned fragsize;
+ unsigned dmasize;
+ unsigned fragsamples;
+ /* OSS stuff */
+ unsigned mapped:1;
+ unsigned ready:1; /* our oss buffers are ready to go */
+ unsigned endcleared:1;
+ unsigned ossfragshift;
+ int ossmaxfrags;
+ unsigned subdivision;
+ u16 base; /* Offset for ptr */
+ } dma_dac, dma_adc;
+
+ /* pointer to each dsp?s piece of the apu->src buffer page */
+ void *mixbuf;
+
+};
+
+struct ess_card {
+ unsigned int magic;
+
+ /* We keep maestro cards in a linked list */
+ struct ess_card *next;
+
+ int dev_mixer;
+
+ int card_type;
+
+ /* as most of this is static,
+ perhaps it should be a pointer to a global struct */
+ struct mixer_goo {
+ int modcnt;
+ int supported_mixers;
+ int stereo_mixers;
+ int record_sources;
+ /* the caller must guarantee arg sanity before calling these */
+/* int (*read_mixer)(struct ess_card *card, int index);*/
+ void (*write_mixer)(struct ess_card *card,int mixer, unsigned int left,unsigned int right);
+ int (*recmask_io)(struct ess_card *card,int rw,int mask);
+ unsigned int mixer_state[SOUND_MIXER_NRDEVICES];
+ } mix;
+
+ int power_regs;
+
+ int in_suspend;
+ wait_queue_head_t suspend_queue;
+
+ struct ess_state channels[MAX_DSPS];
+ u16 maestro_map[NR_IDRS]; /* Register map */
+ /* we have to store this junk so that we can come back from a
+ suspend */
+ u16 apu_map[NR_APUS][NR_APU_REGS]; /* contents of apu regs */
+
+ /* this locks around the physical registers on the card */
+ spinlock_t lock;
+
+ /* memory for this card.. wavecache limited :(*/
+ void *dmapages;
+ int dmaorder;
+
+ /* hardware resources */
+ struct pci_dev *pcidev;
+ u32 iobase;
+ u32 irq;
+
+ int bob_freq;
+ char dsps_open;
+
+ int dock_mute_vol;
+};
+
+static void set_mixer(struct ess_card *card,unsigned int mixer, unsigned int val );
+
+static unsigned
+ld2(unsigned int x)
+{
+ unsigned r = 0;
+
+ if (x >= 0x10000) {
+ x >>= 16;
+ r += 16;
+ }
+ if (x >= 0x100) {
+ x >>= 8;
+ r += 8;
+ }
+ if (x >= 0x10) {
+ x >>= 4;
+ r += 4;
+ }
+ if (x >= 4) {
+ x >>= 2;
+ r += 2;
+ }
+ if (x >= 2)
+ r++;
+ return r;
+}
+
+
+/* --------------------------------------------------------------------- */
+
+static void check_suspend(struct ess_card *card);
+
+/* --------------------------------------------------------------------- */
+
+
+/*
+ * ESS Maestro AC97 codec programming interface.
+ */
+
+static void maestro_ac97_set(struct ess_card *card, u8 cmd, u16 val)
+{
+ int io = card->iobase;
+ int i;
+ /*
+ * Wait for the codec bus to be free
+ */
+
+ check_suspend(card);
+
+ for(i=0;i<10000;i++)
+ {
+ if(!(inb(io+ESS_AC97_INDEX)&1))
+ break;
+ }
+ /*
+ * Write the bus
+ */
+ outw(val, io+ESS_AC97_DATA);
+ mdelay(1);
+ outb(cmd, io+ESS_AC97_INDEX);
+ mdelay(1);
+}
+
+static u16 maestro_ac97_get(struct ess_card *card, u8 cmd)
+{
+ int io = card->iobase;
+ int sanity=10000;
+ u16 data;
+ int i;
+
+ check_suspend(card);
+ /*
+ * Wait for the codec bus to be free
+ */
+
+ for(i=0;i<10000;i++)
+ {
+ if(!(inb(io+ESS_AC97_INDEX)&1))
+ break;
+ }
+
+ outb(cmd|0x80, io+ESS_AC97_INDEX);
+ mdelay(1);
+
+ while(inb(io+ESS_AC97_INDEX)&1)
+ {
+ sanity--;
+ if(!sanity)
+ {
+ printk(KERN_ERR "maestro: ac97 codec timeout reading 0x%x.\n",cmd);
+ return 0;
+ }
+ }
+ data=inw(io+ESS_AC97_DATA);
+ mdelay(1);
+ return data;
+}
+
+/* OSS interface to the ac97s.. */
+
+#define AC97_STEREO_MASK (SOUND_MASK_VOLUME|\
+ SOUND_MASK_PCM|SOUND_MASK_LINE|SOUND_MASK_CD|\
+ SOUND_MASK_VIDEO|SOUND_MASK_LINE1|SOUND_MASK_IGAIN)
+
+#define AC97_SUPPORTED_MASK (AC97_STEREO_MASK | \
+ SOUND_MASK_BASS|SOUND_MASK_TREBLE|SOUND_MASK_MIC|\
+ SOUND_MASK_SPEAKER)
+
+#define AC97_RECORD_MASK (SOUND_MASK_MIC|\
+ SOUND_MASK_CD| SOUND_MASK_VIDEO| SOUND_MASK_LINE1| SOUND_MASK_LINE|\
+ SOUND_MASK_PHONEIN)
+
+#define supported_mixer(CARD,FOO) ( CARD->mix.supported_mixers & (1<<FOO) )
+
+/* this table has default mixer values for all OSS mixers.
+ be sure to fill it in if you add oss mixers
+ to anyone's supported mixer defines */
+
+static unsigned int mixer_defaults[SOUND_MIXER_NRDEVICES] = {
+ [SOUND_MIXER_VOLUME] = 0x3232,
+ [SOUND_MIXER_BASS] = 0x3232,
+ [SOUND_MIXER_TREBLE] = 0x3232,
+ [SOUND_MIXER_SPEAKER] = 0x3232,
+ [SOUND_MIXER_MIC] = 0x8000, /* annoying */
+ [SOUND_MIXER_LINE] = 0x3232,
+ [SOUND_MIXER_CD] = 0x3232,
+ [SOUND_MIXER_VIDEO] = 0x3232,
+ [SOUND_MIXER_LINE1] = 0x3232,
+ [SOUND_MIXER_PCM] = 0x3232,
+ [SOUND_MIXER_IGAIN] = 0x3232
+};
+
+static struct ac97_mixer_hw {
+ unsigned char offset;
+ int scale;
+} ac97_hw[SOUND_MIXER_NRDEVICES]= {
+ [SOUND_MIXER_VOLUME] = {0x02,63},
+ [SOUND_MIXER_BASS] = {0x08,15},
+ [SOUND_MIXER_TREBLE] = {0x08,15},
+ [SOUND_MIXER_SPEAKER] = {0x0a,15},
+ [SOUND_MIXER_MIC] = {0x0e,31},
+ [SOUND_MIXER_LINE] = {0x10,31},
+ [SOUND_MIXER_CD] = {0x12,31},
+ [SOUND_MIXER_VIDEO] = {0x14,31},
+ [SOUND_MIXER_LINE1] = {0x16,31},
+ [SOUND_MIXER_PCM] = {0x18,31},
+ [SOUND_MIXER_IGAIN] = {0x1c,15}
+};
+
+#if 0 /* *shrug* removed simply because we never used it.
+ feel free to implement again if needed */
+
+/* reads the given OSS mixer from the ac97
+ the caller must have insured that the ac97 knows
+ about that given mixer, and should be holding a
+ spinlock for the card */
+static int ac97_read_mixer(struct ess_card *card, int mixer)
+{
+ u16 val;
+ int ret=0;
+ struct ac97_mixer_hw *mh = &ac97_hw[mixer];
+
+ val = maestro_ac97_get(card, mh->offset);
+
+ if(AC97_STEREO_MASK & (1<<mixer)) {
+ /* nice stereo mixers .. */
+ int left,right;
+
+ left = (val >> 8) & 0x7f;
+ right = val & 0x7f;
+
+ if (mixer == SOUND_MIXER_IGAIN) {
+ right = (right * 100) / mh->scale;
+ left = (left * 100) / mh->scale;
+ } else {
+ right = 100 - ((right * 100) / mh->scale);
+ left = 100 - ((left * 100) / mh->scale);
+ }
+
+ ret = left | (right << 8);
+ } else if (mixer == SOUND_MIXER_SPEAKER) {
+ ret = 100 - ((((val & 0x1e)>>1) * 100) / mh->scale);
+ } else if (mixer == SOUND_MIXER_MIC) {
+ ret = 100 - (((val & 0x1f) * 100) / mh->scale);
+ /* the low bit is optional in the tone sliders and masking
+ it lets is avoid the 0xf 'bypass'.. */
+ } else if (mixer == SOUND_MIXER_BASS) {
+ ret = 100 - ((((val >> 8) & 0xe) * 100) / mh->scale);
+ } else if (mixer == SOUND_MIXER_TREBLE) {
+ ret = 100 - (((val & 0xe) * 100) / mh->scale);
+ }
+
+ M_printk("read mixer %d (0x%x) %x -> %x\n",mixer,mh->offset,val,ret);
+
+ return ret;
+}
+#endif
+
+/* write the OSS encoded volume to the given OSS encoded mixer,
+ again caller's job to make sure all is well in arg land,
+ call with spinlock held */
+
+/* linear scale -> log */
+static unsigned char lin2log[101] =
+{
+0, 0 , 15 , 23 , 30 , 34 , 38 , 42 , 45 , 47 ,
+50 , 52 , 53 , 55 , 57 , 58 , 60 , 61 , 62 ,
+63 , 65 , 66 , 67 , 68 , 69 , 69 , 70 , 71 ,
+72 , 73 , 73 , 74 , 75 , 75 , 76 , 77 , 77 ,
+78 , 78 , 79 , 80 , 80 , 81 , 81 , 82 , 82 ,
+83 , 83 , 84 , 84 , 84 , 85 , 85 , 86 , 86 ,
+87 , 87 , 87 , 88 , 88 , 88 , 89 , 89 , 89 ,
+90 , 90 , 90 , 91 , 91 , 91 , 92 , 92 , 92 ,
+93 , 93 , 93 , 94 , 94 , 94 , 94 , 95 , 95 ,
+95 , 95 , 96 , 96 , 96 , 96 , 97 , 97 , 97 ,
+97 , 98 , 98 , 98 , 98 , 99 , 99 , 99 , 99 , 99
+};
+
+static void ac97_write_mixer(struct ess_card *card,int mixer, unsigned int left, unsigned int right)
+{
+ u16 val=0;
+ struct ac97_mixer_hw *mh = &ac97_hw[mixer];
+
+ M_printk("wrote mixer %d (0x%x) %d,%d",mixer,mh->offset,left,right);
+
+ if(AC97_STEREO_MASK & (1<<mixer)) {
+ /* stereo mixers, mute them if we can */
+
+ if (mixer == SOUND_MIXER_IGAIN) {
+ /* igain's slider is reversed.. */
+ right = (right * mh->scale) / 100;
+ left = (left * mh->scale) / 100;
+ if ((left == 0) && (right == 0))
+ val |= 0x8000;
+ } else if (mixer == SOUND_MIXER_PCM || mixer == SOUND_MIXER_CD) {
+ /* log conversion seems bad for them */
+ if ((left == 0) && (right == 0))
+ val = 0x8000;
+ right = ((100 - right) * mh->scale) / 100;
+ left = ((100 - left) * mh->scale) / 100;
+ } else {
+ /* log conversion for the stereo controls */
+ if((left == 0) && (right == 0))
+ val = 0x8000;
+ right = ((100 - lin2log[right]) * mh->scale) / 100;
+ left = ((100 - lin2log[left]) * mh->scale) / 100;
+ }
+
+ val |= (left << 8) | right;
+
+ } else if (mixer == SOUND_MIXER_SPEAKER) {
+ val = (((100 - left) * mh->scale) / 100) << 1;
+ } else if (mixer == SOUND_MIXER_MIC) {
+ val = maestro_ac97_get(card, mh->offset) & ~0x801f;
+ val |= (((100 - left) * mh->scale) / 100);
+ /* the low bit is optional in the tone sliders and masking
+ it lets is avoid the 0xf 'bypass'.. */
+ } else if (mixer == SOUND_MIXER_BASS) {
+ val = maestro_ac97_get(card , mh->offset) & ~0x0f00;
+ val |= ((((100 - left) * mh->scale) / 100) << 8) & 0x0e00;
+ } else if (mixer == SOUND_MIXER_TREBLE) {
+ val = maestro_ac97_get(card , mh->offset) & ~0x000f;
+ val |= (((100 - left) * mh->scale) / 100) & 0x000e;
+ }
+
+ maestro_ac97_set(card , mh->offset, val);
+
+ M_printk(" -> %x\n",val);
+}
+
+/* the following tables allow us to go from
+ OSS <-> ac97 quickly. */
+
+enum ac97_recsettings {
+ AC97_REC_MIC=0,
+ AC97_REC_CD,
+ AC97_REC_VIDEO,
+ AC97_REC_AUX,
+ AC97_REC_LINE,
+ AC97_REC_STEREO, /* combination of all enabled outputs.. */
+ AC97_REC_MONO, /*.. or the mono equivalent */
+ AC97_REC_PHONE
+};
+
+static unsigned int ac97_oss_mask[] = {
+ [AC97_REC_MIC] = SOUND_MASK_MIC,
+ [AC97_REC_CD] = SOUND_MASK_CD,
+ [AC97_REC_VIDEO] = SOUND_MASK_VIDEO,
+ [AC97_REC_AUX] = SOUND_MASK_LINE1,
+ [AC97_REC_LINE] = SOUND_MASK_LINE,
+ [AC97_REC_PHONE] = SOUND_MASK_PHONEIN
+};
+
+/* indexed by bit position */
+static unsigned int ac97_oss_rm[] = {
+ [SOUND_MIXER_MIC] = AC97_REC_MIC,
+ [SOUND_MIXER_CD] = AC97_REC_CD,
+ [SOUND_MIXER_VIDEO] = AC97_REC_VIDEO,
+ [SOUND_MIXER_LINE1] = AC97_REC_AUX,
+ [SOUND_MIXER_LINE] = AC97_REC_LINE,
+ [SOUND_MIXER_PHONEIN] = AC97_REC_PHONE
+};
+
+/* read or write the recmask
+ the ac97 can really have left and right recording
+ inputs independently set, but OSS doesn't seem to
+ want us to express that to the user.
+ the caller guarantees that we have a supported bit set,
+ and they must be holding the card's spinlock */
+static int
+ac97_recmask_io(struct ess_card *card, int read, int mask)
+{
+ unsigned int val = ac97_oss_mask[ maestro_ac97_get(card, 0x1a) & 0x7 ];
+
+ if (read) return val;
+
+ /* oss can have many inputs, maestro can't. try
+ to pick the 'new' one */
+
+ if (mask != val) mask &= ~val;
+
+ val = ffs(mask) - 1;
+ val = ac97_oss_rm[val];
+ val |= val << 8; /* set both channels */
+
+ M_printk("maestro: setting ac97 recmask to 0x%x\n",val);
+
+ maestro_ac97_set(card,0x1a,val);
+
+ return 0;
+};
+
+/*
+ * The Maestro can be wired to a standard AC97 compliant codec
+ * (see www.intel.com for the pdf's on this), or to a PT101 codec
+ * which appears to be the ES1918 (data sheet on the esstech.com.tw site)
+ *
+ * The PT101 setup is untested.
+ */
+
+static u16 __init maestro_ac97_init(struct ess_card *card)
+{
+ u16 vend1, vend2, caps;
+
+ card->mix.supported_mixers = AC97_SUPPORTED_MASK;
+ card->mix.stereo_mixers = AC97_STEREO_MASK;
+ card->mix.record_sources = AC97_RECORD_MASK;
+/* card->mix.read_mixer = ac97_read_mixer;*/
+ card->mix.write_mixer = ac97_write_mixer;
+ card->mix.recmask_io = ac97_recmask_io;
+
+ vend1 = maestro_ac97_get(card, 0x7c);
+ vend2 = maestro_ac97_get(card, 0x7e);
+
+ caps = maestro_ac97_get(card, 0x00);
+
+ printk(KERN_INFO "maestro: AC97 Codec detected: v: 0x%2x%2x caps: 0x%x pwr: 0x%x\n",
+ vend1,vend2,caps,maestro_ac97_get(card,0x26) & 0xf);
+
+ if (! (caps & 0x4) ) {
+ /* no bass/treble nobs */
+ card->mix.supported_mixers &= ~(SOUND_MASK_BASS|SOUND_MASK_TREBLE);
+ }
+
+ /* XXX endianness, dork head. */
+ /* vendor specifc bits.. */
+ switch ((long)(vend1 << 16) | vend2) {
+ case 0x545200ff: /* TriTech */
+ /* no idea what this does */
+ maestro_ac97_set(card,0x2a,0x0001);
+ maestro_ac97_set(card,0x2c,0x0000);
+ maestro_ac97_set(card,0x2c,0xffff);
+ break;
+#if 0 /* i thought the problems I was seeing were with
+ the 1921, but apparently they were with the pci board
+ it was on, so this code is commented out.
+ lets see if this holds true. */
+ case 0x83847609: /* ESS 1921 */
+ /* writing to 0xe (mic) or 0x1a (recmask) seems
+ to hang this codec */
+ card->mix.supported_mixers &= ~(SOUND_MASK_MIC);
+ card->mix.record_sources = 0;
+ card->mix.recmask_io = NULL;
+#if 0 /* don't ask. I have yet to see what these actually do. */
+ maestro_ac97_set(card,0x76,0xABBA); /* o/~ Take a chance on me o/~ */
+ udelay(20);
+ maestro_ac97_set(card,0x78,0x3002);
+ udelay(20);
+ maestro_ac97_set(card,0x78,0x3802);
+ udelay(20);
+#endif
+ break;
+#endif
+ default: break;
+ }
+
+ maestro_ac97_set(card, 0x1E, 0x0404);
+ /* null misc stuff */
+ maestro_ac97_set(card, 0x20, 0x0000);
+
+ return 0;
+}
+
+#if 0 /* there has been 1 person on the planet with a pt101 that we
+ know of. If they care, they can put this back in :) */
+static u16 maestro_pt101_init(struct ess_card *card,int iobase)
+{
+ printk(KERN_INFO "maestro: PT101 Codec detected, initializing but _not_ installing mixer device.\n");
+ /* who knows.. */
+ maestro_ac97_set(iobase, 0x2A, 0x0001);
+ maestro_ac97_set(iobase, 0x2C, 0x0000);
+ maestro_ac97_set(iobase, 0x2C, 0xFFFF);
+ maestro_ac97_set(iobase, 0x10, 0x9F1F);
+ maestro_ac97_set(iobase, 0x12, 0x0808);
+ maestro_ac97_set(iobase, 0x14, 0x9F1F);
+ maestro_ac97_set(iobase, 0x16, 0x9F1F);
+ maestro_ac97_set(iobase, 0x18, 0x0404);
+ maestro_ac97_set(iobase, 0x1A, 0x0000);
+ maestro_ac97_set(iobase, 0x1C, 0x0000);
+ maestro_ac97_set(iobase, 0x02, 0x0404);
+ maestro_ac97_set(iobase, 0x04, 0x0808);
+ maestro_ac97_set(iobase, 0x0C, 0x801F);
+ maestro_ac97_set(iobase, 0x0E, 0x801F);
+ return 0;
+}
+#endif
+
+/* this is very magic, and very slow.. */
+static void
+maestro_ac97_reset(int ioaddr, struct pci_dev *pcidev)
+{
+ u16 save_68;
+ u16 w;
+ u32 vend;
+
+ outw( inw(ioaddr + 0x38) & 0xfffc, ioaddr + 0x38);
+ outw( inw(ioaddr + 0x3a) & 0xfffc, ioaddr + 0x3a);
+ outw( inw(ioaddr + 0x3c) & 0xfffc, ioaddr + 0x3c);
+
+ /* reset the first codec */
+ outw(0x0000, ioaddr+0x36);
+ save_68 = inw(ioaddr+0x68);
+ pci_read_config_word(pcidev, 0x58, &w); /* something magical with gpio and bus arb. */
+ pci_read_config_dword(pcidev, PCI_SUBSYSTEM_VENDOR_ID, &vend);
+ if( w & 0x1)
+ save_68 |= 0x10;
+ outw(0xfffe, ioaddr + 0x64); /* tickly gpio 0.. */
+ outw(0x0001, ioaddr + 0x68);
+ outw(0x0000, ioaddr + 0x60);
+ udelay(20);
+ outw(0x0001, ioaddr + 0x60);
+ mdelay(20);
+
+ outw(save_68 | 0x1, ioaddr + 0x68); /* now restore .. */
+ outw( (inw(ioaddr + 0x38) & 0xfffc)|0x1, ioaddr + 0x38);
+ outw( (inw(ioaddr + 0x3a) & 0xfffc)|0x1, ioaddr + 0x3a);
+ outw( (inw(ioaddr + 0x3c) & 0xfffc)|0x1, ioaddr + 0x3c);
+
+ /* now the second codec */
+ outw(0x0000, ioaddr+0x36);
+ outw(0xfff7, ioaddr + 0x64);
+ save_68 = inw(ioaddr+0x68);
+ outw(0x0009, ioaddr + 0x68);
+ outw(0x0001, ioaddr + 0x60);
+ udelay(20);
+ outw(0x0009, ioaddr + 0x60);
+ mdelay(500); /* .. ouch.. */
+ outw( inw(ioaddr + 0x38) & 0xfffc, ioaddr + 0x38);
+ outw( inw(ioaddr + 0x3a) & 0xfffc, ioaddr + 0x3a);
+ outw( inw(ioaddr + 0x3c) & 0xfffc, ioaddr + 0x3c);
+
+#if 0 /* the loop here needs to be much better if we want it.. */
+ M_printk("trying software reset\n");
+ /* try and do a software reset */
+ outb(0x80|0x7c, ioaddr + 0x30);
+ for (w=0; ; w++) {
+ if ((inw(ioaddr+ 0x30) & 1) == 0) {
+ if(inb(ioaddr + 0x32) !=0) break;
+
+ outb(0x80|0x7d, ioaddr + 0x30);
+ if (((inw(ioaddr+ 0x30) & 1) == 0) && (inb(ioaddr + 0x32) !=0)) break;
+ outb(0x80|0x7f, ioaddr + 0x30);
+ if (((inw(ioaddr+ 0x30) & 1) == 0) && (inb(ioaddr + 0x32) !=0)) break;
+ }
+
+ if( w > 10000) {
+ outb( inb(ioaddr + 0x37) | 0x08, ioaddr + 0x37); /* do a software reset */
+ mdelay(500); /* oh my.. */
+ outb( inb(ioaddr + 0x37) & ~0x08, ioaddr + 0x37);
+ udelay(1);
+ outw( 0x80, ioaddr+0x30);
+ for(w = 0 ; w < 10000; w++) {
+ if((inw(ioaddr + 0x30) & 1) ==0) break;
+ }
+ }
+ }
+#endif
+ if ( vend == NEC_VERSA_SUBID1 || vend == NEC_VERSA_SUBID2) {
+ /* turn on external amp? */
+ outw(0xf9ff, ioaddr + 0x64);
+ outw(inw(ioaddr+0x68) | 0x600, ioaddr + 0x68);
+ outw(0x0209, ioaddr + 0x60);
+ }
+
+ /* Turn on the 978 docking chip.
+ First frob the "master output enable" bit,
+ then set most of the playback volume control registers to max. */
+ outb(inb(ioaddr+0xc0)|(1<<5), ioaddr+0xc0);
+ outb(0xff, ioaddr+0xc3);
+ outb(0xff, ioaddr+0xc4);
+ outb(0xff, ioaddr+0xc6);
+ outb(0xff, ioaddr+0xc8);
+ outb(0x3f, ioaddr+0xcf);
+ outb(0x3f, ioaddr+0xd0);
+}
+/*
+ * Indirect register access. Not all registers are readable so we
+ * need to keep register state ourselves
+ */
+
+#define WRITEABLE_MAP 0xEFFFFF
+#define READABLE_MAP 0x64003F
+
+/*
+ * The Maestro engineers were a little indirection happy. These indirected
+ * registers themselves include indirect registers at another layer
+ */
+
+static void __maestro_write(struct ess_card *card, u16 reg, u16 data)
+{
+ long ioaddr = card->iobase;
+
+ outw(reg, ioaddr+0x02);
+ outw(data, ioaddr+0x00);
+ if( reg >= NR_IDRS) printk("maestro: IDR %d out of bounds!\n",reg);
+ else card->maestro_map[reg]=data;
+
+}
+
+static void maestro_write(struct ess_state *s, u16 reg, u16 data)
+{
+ unsigned long flags;
+
+ check_suspend(s->card);
+ spin_lock_irqsave(&s->card->lock,flags);
+
+ __maestro_write(s->card,reg,data);
+
+ spin_unlock_irqrestore(&s->card->lock,flags);
+}
+
+static u16 __maestro_read(struct ess_card *card, u16 reg)
+{
+ long ioaddr = card->iobase;
+
+ outw(reg, ioaddr+0x02);
+ return card->maestro_map[reg]=inw(ioaddr+0x00);
+}
+
+static u16 maestro_read(struct ess_state *s, u16 reg)
+{
+ if(READABLE_MAP & (1<<reg))
+ {
+ unsigned long flags;
+ check_suspend(s->card);
+ spin_lock_irqsave(&s->card->lock,flags);
+
+ __maestro_read(s->card,reg);
+
+ spin_unlock_irqrestore(&s->card->lock,flags);
+ }
+ return s->card->maestro_map[reg];
+}
+
+/*
+ * These routines handle accessing the second level indirections to the
+ * wave ram.
+ */
+
+/*
+ * The register names are the ones ESS uses (see 104T31.ZIP)
+ */
+
+#define IDR0_DATA_PORT 0x00
+#define IDR1_CRAM_POINTER 0x01
+#define IDR2_CRAM_DATA 0x02
+#define IDR3_WAVE_DATA 0x03
+#define IDR4_WAVE_PTR_LOW 0x04
+#define IDR5_WAVE_PTR_HI 0x05
+#define IDR6_TIMER_CTRL 0x06
+#define IDR7_WAVE_ROMRAM 0x07
+
+stat