ac97_codec.c

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/*
 * ac97_codec.c: Generic AC97 mixer/modem module
 *
 * Derived from ac97 mixer in maestro and trident driver.
 *
 * Copyright 2000 Silicon Integrated System Corporation
 *
 *	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.
 *
 * History
 * v0.4 Mar 15 2000 Ollie Lho
 *	dual codecs support verified with 4 channels output
 * v0.3 Feb 22 2000 Ollie Lho
 *	bug fix for record mask setting
 * v0.2 Feb 10 2000 Ollie Lho
 *	add ac97_read_proc for /proc/driver/{vendor}/ac97
 * v0.1 Jan 14 2000 Ollie Lho <[e-mail]> 
 *	Isolated from trident.c to support multiple ac97 codec
 */
#include <linux/module.h>
#include <linux/version.h>
#include <linux/kernel.h>
#include <linux/string.h>
#include <linux/errno.h>
#include <linux/bitops.h>
#include <linux/delay.h>
#include <linux/ac97_codec.h>
#include <asm/uaccess.h>

static int ac97_read_mixer(struct ac97_codec *codec, int oss_channel);
static void ac97_write_mixer(struct ac97_codec *codec, int oss_channel, 
			     unsigned int left, unsigned int right);
static void ac97_set_mixer(struct ac97_codec *codec, unsigned int oss_mixer, unsigned int val );
static int ac97_recmask_io(struct ac97_codec *codec, int rw, int mask);
static int ac97_mixer_ioctl(struct ac97_codec *codec, unsigned int cmd, unsigned long arg);

static int ac97_init_mixer(struct ac97_codec *codec);

static int sigmatel_init(struct ac97_codec *codec);
static int enable_eapd(struct ac97_codec *codec);

#define arraysize(x)   (sizeof(x)/sizeof((x)[0]))

static struct {
	unsigned int id;
	char *name;
	int  (*init)  (struct ac97_codec *codec);
} ac97_codec_ids[] = {
	{0x414B4D00, "Asahi Kasei AK4540 rev 0", NULL},
	{0x414B4D01, "Asahi Kasei AK4540 rev 1", NULL},
	{0x41445340, "Analog Devices AD1881"  , NULL},
	{0x41445360, "Analog Devices AD1885"  , enable_eapd},
	{0x43525900, "Cirrus Logic CS4297"    , NULL},
	{0x43525903, "Cirrus Logic CS4297"  ,	NULL},
	{0x43525913, "Cirrus Logic CS4297A"   , NULL},
	{0x43525923, "Cirrus Logic CS4298"    , NULL},
	{0x4352592B, "Cirrus Logic CS4294"    , NULL},
	{0x43525931, "Cirrus Logic CS4299"    , NULL},
	{0x43525934, "Cirrus Logic CS4299"    , NULL},
	{0x4e534331, "National Semiconductor LM4549" ,	NULL},
	{0x53494c22, "Silicon Laboratory Si3036"     ,	NULL},
	{0x53494c23, "Silicon Laboratory Si3038"     ,  NULL},
	{0x83847600, "SigmaTel STAC????"      , NULL},
	{0x83847604, "SigmaTel STAC9701/3/4/5", NULL},
	{0x83847605, "SigmaTel STAC9704"      , NULL},
	{0x83847608, "SigmaTel STAC9708"      , NULL},
	{0x83847609, "SigmaTel STAC9721/23"   , sigmatel_init},
	{0x54524103, "TriTech TR?????"	      , NULL},
	{0x54524106, "TriTech TR28026"        , NULL},
	{0x54524108, "TriTech TR28028"        , NULL},
	{0x54524123, "TriTech TR?????"	      , NULL},	
	{0x574D4C00, "Wolfson WM9704"         , NULL},
	{0x00000000, NULL, NULL}
};

static const char *ac97_stereo_enhancements[] =
{
	/*   0 */ "No 3D Stereo Enhancement",
	/*   1 */ "Analog Devices Phat Stereo",
	/*   2 */ "Creative Stereo Enhancement",
	/*   3 */ "National Semi 3D Stereo Enhancement",
	/*   4 */ "YAMAHA Ymersion",
	/*   5 */ "BBE 3D Stereo Enhancement",
	/*   6 */ "Crystal Semi 3D Stereo Enhancement",
	/*   7 */ "Qsound QXpander",
	/*   8 */ "Spatializer 3D Stereo Enhancement",
	/*   9 */ "SRS 3D Stereo Enhancement",
	/*  10 */ "Platform Tech 3D Stereo Enhancement",
	/*  11 */ "AKM 3D Audio",
	/*  12 */ "Aureal Stereo Enhancement",
	/*  13 */ "Aztech 3D Enhancement",
	/*  14 */ "Binaura 3D Audio Enhancement",
	/*  15 */ "ESS Technology Stereo Enhancement",
	/*  16 */ "Harman International VMAx",
	/*  17 */ "Nvidea 3D Stereo Enhancement",
	/*  18 */ "Philips Incredible Sound",
	/*  19 */ "Texas Instruments 3D Stereo Enhancement",
	/*  20 */ "VLSI Technology 3D Stereo Enhancement",
	/*  21 */ "TriTech 3D Stereo Enhancement",
	/*  22 */ "Realtek 3D Stereo Enhancement",
	/*  23 */ "Samsung 3D Stereo Enhancement",
	/*  24 */ "Wolfson Microelectronics 3D Enhancement",
	/*  25 */ "Delta Integration 3D Enhancement",
	/*  26 */ "SigmaTel 3D Enhancement",
	/*  27 */ "Reserved 27",
	/*  28 */ "Rockwell 3D Stereo Enhancement",
	/*  29 */ "Reserved 29",
	/*  30 */ "Reserved 30",
	/*  31 */ "Reserved 31"
};

/* this table has default mixer values for all OSS mixers. */
static struct mixer_defaults {
	int mixer;
	unsigned int value;
} mixer_defaults[SOUND_MIXER_NRDEVICES] = {
	/* all values 0 -> 100 in bytes */
	{SOUND_MIXER_VOLUME,	0x4343},
	{SOUND_MIXER_BASS,	0x4343},
	{SOUND_MIXER_TREBLE,	0x4343},
	{SOUND_MIXER_PCM,	0x4343},
	{SOUND_MIXER_SPEAKER,	0x4343},
	{SOUND_MIXER_LINE,	0x4343},
	{SOUND_MIXER_MIC,	0x0000},
	{SOUND_MIXER_CD,	0x4343},
	{SOUND_MIXER_ALTPCM,	0x4343},
	{SOUND_MIXER_IGAIN,	0x4343},
	{SOUND_MIXER_LINE1,	0x4343},
	{SOUND_MIXER_PHONEIN,	0x4343},
	{SOUND_MIXER_PHONEOUT,	0x4343},
	{SOUND_MIXER_VIDEO,	0x4343},
	{-1,0}
};

/* table to scale scale from OSS mixer value to AC97 mixer register value */	
static struct ac97_mixer_hw {
	unsigned char offset;
	int scale;
} ac97_hw[SOUND_MIXER_NRDEVICES]= {
	[SOUND_MIXER_VOLUME]	=	{AC97_MASTER_VOL_STEREO,64},
	[SOUND_MIXER_BASS]	=	{AC97_MASTER_TONE,	16},
	[SOUND_MIXER_TREBLE]	=	{AC97_MASTER_TONE,	16},
	[SOUND_MIXER_PCM]	=	{AC97_PCMOUT_VOL,	32},
	[SOUND_MIXER_SPEAKER]	=	{AC97_PCBEEP_VOL,	16},
	[SOUND_MIXER_LINE]	=	{AC97_LINEIN_VOL,	32},
	[SOUND_MIXER_MIC]	=	{AC97_MIC_VOL,		32},
	[SOUND_MIXER_CD]	=	{AC97_CD_VOL,		32},
	[SOUND_MIXER_ALTPCM]	=	{AC97_HEADPHONE_VOL,	64},
	[SOUND_MIXER_IGAIN]	=	{AC97_RECORD_GAIN,	16},
	[SOUND_MIXER_LINE1]	=	{AC97_AUX_VOL,		32},
	[SOUND_MIXER_PHONEIN]	= 	{AC97_PHONE_VOL,	32},
	[SOUND_MIXER_PHONEOUT]	= 	{AC97_MASTER_VOL_MONO,	64},
	[SOUND_MIXER_VIDEO]	=	{AC97_VIDEO_VOL,	32},
};

/* 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_rm2oss[] = {
	[AC97_REC_MIC] 	 = SOUND_MIXER_MIC,
	[AC97_REC_CD] 	 = SOUND_MIXER_CD,
	[AC97_REC_VIDEO] = SOUND_MIXER_VIDEO,
	[AC97_REC_AUX] 	 = SOUND_MIXER_LINE1,
	[AC97_REC_LINE]  = SOUND_MIXER_LINE,
	[AC97_REC_STEREO]= SOUND_MIXER_IGAIN,
	[AC97_REC_PHONE] = SOUND_MIXER_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_IGAIN]	= AC97_REC_STEREO,
	[SOUND_MIXER_PHONEIN] 	= AC97_REC_PHONE
};

/* 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 ac97_codec *codec, int oss_channel) 
{
	u16 val;
	int ret = 0;
	int scale;
	struct ac97_mixer_hw *mh = &ac97_hw[oss_channel];

	val = codec->codec_read(codec , mh->offset);

	if (val & AC97_MUTE) {
		ret = 0;
	} else if (AC97_STEREO_MASK & (1 << oss_channel)) {
		/* nice stereo mixers .. */
		int left,right;

		left = (val >> 8)  & 0x7f;
		right = val  & 0x7f;

		if (oss_channel == SOUND_MIXER_IGAIN) {
			right = (right * 100) / mh->scale;
			left = (left * 100) / mh->scale;
		} else {
			/* these may have 5 or 6 bit resolution */
			if(oss_channel == SOUND_MIXER_VOLUME || oss_channel == SOUND_MIXER_ALTPCM)
				scale = (1 << codec->bit_resolution);
			else
				scale = mh->scale;

			right = 100 - ((right * 100) / scale);
			left = 100 - ((left * 100) / scale);
		}
		ret = left | (right << 8);
	} else if (oss_channel == SOUND_MIXER_SPEAKER) {
		ret = 100 - ((((val & 0x1e)>>1) * 100) / mh->scale);
	} else if (oss_channel == SOUND_MIXER_PHONEIN) {
		ret = 100 - (((val & 0x1f) * 100) / mh->scale);
	} else if (oss_channel == SOUND_MIXER_PHONEOUT) {
		scale = (1 << codec->bit_resolution);
		ret = 100 - (((val & 0x1f) * 100) / scale);
	} else if (oss_channel == SOUND_MIXER_MIC) {
		ret = 100 - (((val & 0x1f) * 100) / mh->scale);
		/*  the low bit is optional in the tone sliders and masking
		    it lets us avoid the 0xf 'bypass'.. */
	} else if (oss_channel == SOUND_MIXER_BASS) {
		ret = 100 - ((((val >> 8) & 0xe) * 100) / mh->scale);
	} else if (oss_channel == SOUND_MIXER_TREBLE) {
		ret = 100 - (((val & 0xe) * 100) / mh->scale);
	}

#ifdef DEBUG
	printk("ac97_codec: read OSS mixer %2d (%s ac97 register 0x%02x), "
	       "0x%04x -> 0x%04x\n",
	       oss_channel, codec->id ? "Secondary" : "Primary",
	       mh->offset, val, ret);
#endif

	return ret;
}

/* 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 */
static void ac97_write_mixer(struct ac97_codec *codec, int oss_channel,
		      unsigned int left, unsigned int right)
{
	u16 val = 0;
	int scale;
	struct ac97_mixer_hw *mh = &ac97_hw[oss_channel];

#ifdef DEBUG
	printk("ac97_codec: wrote OSS mixer %2d (%s ac97 register 0x%02x), "
	       "left vol:%2d, right vol:%2d:",
	       oss_channel, codec->id ? "Secondary" : "Primary",
	       mh->offset, left, right);
#endif

	if (AC97_STEREO_MASK & (1 << oss_channel)) {
		/* stereo mixers */
		if (left == 0 && right == 0) {
			val = AC97_MUTE;
		} else {
			if (oss_channel == SOUND_MIXER_IGAIN) {
				right = (right * mh->scale) / 100;
				left = (left * mh->scale) / 100;
				if (right >= mh->scale)
					right = mh->scale-1;
				if (left >= mh->scale)
					left = mh->scale-1;
			} else {
				/* these may have 5 or 6 bit resolution */
				if (oss_channel == SOUND_MIXER_VOLUME ||
				    oss_channel == SOUND_MIXER_ALTPCM)
					scale = (1 << codec->bit_resolution);
				else
					scale = mh->scale;

				right = ((100 - right) * scale) / 100;
				left = ((100 - left) * scale) / 100;
				if (right >= scale)
					right = scale-1;
				if (left >= scale)
					left = scale-1;
			}
			val = (left << 8) | right;
		}
	} else if (oss_channel == SOUND_MIXER_BASS) {
		val = codec->codec_read(codec , mh->offset) & ~0x0f00;
		left = ((100 - left) * mh->scale) / 100;
		if (left >= mh->scale)
			left = mh->scale-1;
		val |= (left << 8) & 0x0e00;
	} else if (oss_channel == SOUND_MIXER_TREBLE) {
		val = codec->codec_read(codec , mh->offset) & ~0x000f;
		left = ((100 - left) * mh->scale) / 100;
		if (left >= mh->scale)
			left = mh->scale-1;
		val |= left & 0x000e;
	} else if(left == 0) {
		val = AC97_MUTE;
	} else if (oss_channel == SOUND_MIXER_SPEAKER) {
		left = ((100 - left) * mh->scale) / 100;
		if (left >= mh->scale)
			left = mh->scale-1;
		val = left << 1;
	} else if (oss_channel == SOUND_MIXER_PHONEIN) {
		left = ((100 - left) * mh->scale) / 100;
		if (left >= mh->scale)
			left = mh->scale-1;
		val = left;
	} else if (oss_channel == SOUND_MIXER_PHONEOUT) {
		scale = (1 << codec->bit_resolution);
		left = ((100 - left) * scale) / 100;
		if (left >= mh->scale)
			left = mh->scale-1;
		val = left;
	} else if (oss_channel == SOUND_MIXER_MIC) {
		val = codec->codec_read(codec , mh->offset) & ~0x801f;
		left = ((100 - left) * mh->scale) / 100;
		if (left >= mh->scale)
			left = mh->scale-1;
		val |= left;
		/*  the low bit is optional in the tone sliders and masking
		    it lets us avoid the 0xf 'bypass'.. */
	}
#ifdef DEBUG
	printk(" 0x%04x", val);
#endif

	codec->codec_write(codec, mh->offset, val);

#ifdef DEBUG
	val = codec->codec_read(codec, mh->offset);
	printk(" -> 0x%04x\n", val);
#endif
}

/* a thin wrapper for write_mixer */
static void ac97_set_mixer(struct ac97_codec *codec, unsigned int oss_mixer, unsigned int val ) 
{
	unsigned int left,right;

	/* cleanse input a little */
	right = ((val >> 8)  & 0xff) ;
	left = (val  & 0xff) ;

	if (right > 100) right = 100;
	if (left > 100) left = 100;

	codec->mixer_state[oss_mixer] = (right << 8) | left;
	codec->write_mixer(codec, oss_mixer, left, right);
}