init android origin source code

7 files changed, 3468 insertions, 0 deletions

diff --git a/ANDROID_3.4.5/sound/mips/Kconfig b/ANDROID_3.4.5/sound/mips/Kconfig
new file mode 100644
index 00000000..d2f615ab
--- /dev/null
+++ b/ANDROID_3.4.5/sound/mips/Kconfig
@@ -0,0 +1,37 @@
+# ALSA MIPS drivers
+
+menuconfig SND_MIPS
+	bool "MIPS sound devices"
+	depends on MIPS
+	default y
+	help
+	  Support for sound devices of MIPS architectures.
+
+if SND_MIPS
+
+config SND_SGI_O2
+	tristate "SGI O2 Audio"
+	depends on SGI_IP32
+        help
+                Sound support for the SGI O2 Workstation. 
+
+config SND_SGI_HAL2
+        tristate "SGI HAL2 Audio"
+        depends on SGI_HAS_HAL2
+        help
+                Sound support for the SGI Indy and Indigo2 Workstation.
+
+
+config SND_AU1X00
+	tristate "Au1x00 AC97 Port Driver (DEPRECATED)"
+	depends on MIPS_ALCHEMY
+	select SND_PCM
+	select SND_AC97_CODEC
+	help
+	  ALSA Sound driver for the Au1x00's AC97 port.
+
+	  Newer drivers for ASoC are available, please do not use
+	  this driver as it will be removed in the future.
+
+endif	# SND_MIPS
+
diff --git a/ANDROID_3.4.5/sound/mips/Makefile b/ANDROID_3.4.5/sound/mips/Makefile
new file mode 100644
index 00000000..861ec0a5
--- /dev/null
+++ b/ANDROID_3.4.5/sound/mips/Makefile
@@ -0,0 +1,12 @@
+#
+# Makefile for ALSA
+#
+
+snd-au1x00-objs := au1x00.o
+snd-sgi-o2-objs := sgio2audio.o ad1843.o
+snd-sgi-hal2-objs := hal2.o
+
+# Toplevel Module Dependency
+obj-$(CONFIG_SND_AU1X00) += snd-au1x00.o
+obj-$(CONFIG_SND_SGI_O2) += snd-sgi-o2.o
+obj-$(CONFIG_SND_SGI_HAL2) += snd-sgi-hal2.o
diff --git a/ANDROID_3.4.5/sound/mips/ad1843.c b/ANDROID_3.4.5/sound/mips/ad1843.c
new file mode 100644
index 00000000..c624510e
--- /dev/null
+++ b/ANDROID_3.4.5/sound/mips/ad1843.c
@@ -0,0 +1,561 @@
+/*
+ *   AD1843 low level driver
+ *
+ *   Copyright 2003 Vivien Chappelier <vivien.chappelier@linux-mips.org>
+ *   Copyright 2008 Thomas Bogendoerfer <tsbogend@alpha.franken.de>
+ *
+ *   inspired from vwsnd.c (SGI VW audio driver)
+ *     Copyright 1999 Silicon Graphics, Inc.  All rights reserved.
+ *
+ *   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., 59 Temple Place, Suite 330, Boston, MA  02111-1307 USA
+ *
+ */
+
+#include <linux/init.h>
+#include <linux/sched.h>
+#include <linux/errno.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/ad1843.h>
+
+/*
+ * AD1843 bitfield definitions.  All are named as in the AD1843 data
+ * sheet, with ad1843_ prepended and individual bit numbers removed.
+ *
+ * E.g., bits LSS0 through LSS2 become ad1843_LSS.
+ *
+ * Only the bitfields we need are defined.
+ */
+
+struct ad1843_bitfield {
+	char reg;
+	char lo_bit;
+	char nbits;
+};
+
+static const struct ad1843_bitfield
+	ad1843_PDNO   = {  0, 14,  1 },	/* Converter Power-Down Flag */
+	ad1843_INIT   = {  0, 15,  1 },	/* Clock Initialization Flag */
+	ad1843_RIG    = {  2,  0,  4 },	/* Right ADC Input Gain */
+	ad1843_RMGE   = {  2,  4,  1 },	/* Right ADC Mic Gain Enable */
+	ad1843_RSS    = {  2,  5,  3 },	/* Right ADC Source Select */
+	ad1843_LIG    = {  2,  8,  4 },	/* Left ADC Input Gain */
+	ad1843_LMGE   = {  2, 12,  1 },	/* Left ADC Mic Gain Enable */
+	ad1843_LSS    = {  2, 13,  3 },	/* Left ADC Source Select */
+	ad1843_RD2M   = {  3,  0,  5 },	/* Right DAC 2 Mix Gain/Atten */
+	ad1843_RD2MM  = {  3,  7,  1 },	/* Right DAC 2 Mix Mute */
+	ad1843_LD2M   = {  3,  8,  5 },	/* Left DAC 2 Mix Gain/Atten */
+	ad1843_LD2MM  = {  3, 15,  1 },	/* Left DAC 2 Mix Mute */
+	ad1843_RX1M   = {  4,  0,  5 },	/* Right Aux 1 Mix Gain/Atten */
+	ad1843_RX1MM  = {  4,  7,  1 },	/* Right Aux 1 Mix Mute */
+	ad1843_LX1M   = {  4,  8,  5 },	/* Left Aux 1 Mix Gain/Atten */
+	ad1843_LX1MM  = {  4, 15,  1 },	/* Left Aux 1 Mix Mute */
+	ad1843_RX2M   = {  5,  0,  5 },	/* Right Aux 2 Mix Gain/Atten */
+	ad1843_RX2MM  = {  5,  7,  1 },	/* Right Aux 2 Mix Mute */
+	ad1843_LX2M   = {  5,  8,  5 },	/* Left Aux 2 Mix Gain/Atten */
+	ad1843_LX2MM  = {  5, 15,  1 },	/* Left Aux 2 Mix Mute */
+	ad1843_RMCM   = {  7,  0,  5 },	/* Right Mic Mix Gain/Atten */
+	ad1843_RMCMM  = {  7,  7,  1 },	/* Right Mic Mix Mute */
+	ad1843_LMCM   = {  7,  8,  5 },	/* Left Mic Mix Gain/Atten */
+	ad1843_LMCMM  = {  7, 15,  1 },	/* Left Mic Mix Mute */
+	ad1843_HPOS   = {  8,  4,  1 },	/* Headphone Output Voltage Swing */
+	ad1843_HPOM   = {  8,  5,  1 },	/* Headphone Output Mute */
+	ad1843_MPOM   = {  8,  6,  1 },	/* Mono Output Mute */
+	ad1843_RDA1G  = {  9,  0,  6 },	/* Right DAC1 Analog/Digital Gain */
+	ad1843_RDA1GM = {  9,  7,  1 },	/* Right DAC1 Analog Mute */
+	ad1843_LDA1G  = {  9,  8,  6 },	/* Left DAC1 Analog/Digital Gain */
+	ad1843_LDA1GM = {  9, 15,  1 },	/* Left DAC1 Analog Mute */
+	ad1843_RDA2G  = { 10,  0,  6 },	/* Right DAC2 Analog/Digital Gain */
+	ad1843_RDA2GM = { 10,  7,  1 },	/* Right DAC2 Analog Mute */
+	ad1843_LDA2G  = { 10,  8,  6 },	/* Left DAC2 Analog/Digital Gain */
+	ad1843_LDA2GM = { 10, 15,  1 },	/* Left DAC2 Analog Mute */
+	ad1843_RDA1AM = { 11,  7,  1 },	/* Right DAC1 Digital Mute */
+	ad1843_LDA1AM = { 11, 15,  1 },	/* Left DAC1 Digital Mute */
+	ad1843_RDA2AM = { 12,  7,  1 },	/* Right DAC2 Digital Mute */
+	ad1843_LDA2AM = { 12, 15,  1 },	/* Left DAC2 Digital Mute */
+	ad1843_ADLC   = { 15,  0,  2 },	/* ADC Left Sample Rate Source */
+	ad1843_ADRC   = { 15,  2,  2 },	/* ADC Right Sample Rate Source */
+	ad1843_DA1C   = { 15,  8,  2 },	/* DAC1 Sample Rate Source */
+	ad1843_DA2C   = { 15, 10,  2 },	/* DAC2 Sample Rate Source */
+	ad1843_C1C    = { 17,  0, 16 },	/* Clock 1 Sample Rate Select */
+	ad1843_C2C    = { 20,  0, 16 },	/* Clock 2 Sample Rate Select */
+	ad1843_C3C    = { 23,  0, 16 },	/* Clock 3 Sample Rate Select */
+	ad1843_DAADL  = { 25,  4,  2 },	/* Digital ADC Left Source Select */
+	ad1843_DAADR  = { 25,  6,  2 },	/* Digital ADC Right Source Select */
+	ad1843_DAMIX  = { 25, 14,  1 },	/* DAC Digital Mix Enable */
+	ad1843_DRSFLT = { 25, 15,  1 },	/* Digital Reampler Filter Mode */
+	ad1843_ADLF   = { 26,  0,  2 }, /* ADC Left Channel Data Format */
+	ad1843_ADRF   = { 26,  2,  2 }, /* ADC Right Channel Data Format */
+	ad1843_ADTLK  = { 26,  4,  1 },	/* ADC Transmit Lock Mode Select */
+	ad1843_SCF    = { 26,  7,  1 },	/* SCLK Frequency Select */
+	ad1843_DA1F   = { 26,  8,  2 },	/* DAC1 Data Format Select */
+	ad1843_DA2F   = { 26, 10,  2 },	/* DAC2 Data Format Select */
+	ad1843_DA1SM  = { 26, 14,  1 },	/* DAC1 Stereo/Mono Mode Select */
+	ad1843_DA2SM  = { 26, 15,  1 },	/* DAC2 Stereo/Mono Mode Select */
+	ad1843_ADLEN  = { 27,  0,  1 },	/* ADC Left Channel Enable */
+	ad1843_ADREN  = { 27,  1,  1 },	/* ADC Right Channel Enable */
+	ad1843_AAMEN  = { 27,  4,  1 },	/* Analog to Analog Mix Enable */
+	ad1843_ANAEN  = { 27,  7,  1 },	/* Analog Channel Enable */
+	ad1843_DA1EN  = { 27,  8,  1 },	/* DAC1 Enable */
+	ad1843_DA2EN  = { 27,  9,  1 },	/* DAC2 Enable */
+	ad1843_DDMEN  = { 27, 12,  1 },	/* DAC2 to DAC1 Mix  Enable */
+	ad1843_C1EN   = { 28, 11,  1 },	/* Clock Generator 1 Enable */
+	ad1843_C2EN   = { 28, 12,  1 },	/* Clock Generator 2 Enable */
+	ad1843_C3EN   = { 28, 13,  1 },	/* Clock Generator 3 Enable */
+	ad1843_PDNI   = { 28, 15,  1 };	/* Converter Power Down */
+
+/*
+ * The various registers of the AD1843 use three different formats for
+ * specifying gain.  The ad1843_gain structure parameterizes the
+ * formats.
+ */
+
+struct ad1843_gain {
+	int	negative;		/* nonzero if gain is negative. */
+	const struct ad1843_bitfield *lfield;
+	const struct ad1843_bitfield *rfield;
+	const struct ad1843_bitfield *lmute;
+	const struct ad1843_bitfield *rmute;
+};
+
+static const struct ad1843_gain ad1843_gain_RECLEV = {
+	.negative = 0,
+	.lfield   = &ad1843_LIG,
+	.rfield   = &ad1843_RIG
+};
+static const struct ad1843_gain ad1843_gain_LINE = {
+	.negative = 1,
+	.lfield   = &ad1843_LX1M,
+	.rfield   = &ad1843_RX1M,
+	.lmute    = &ad1843_LX1MM,
+	.rmute    = &ad1843_RX1MM
+};
+static const struct ad1843_gain ad1843_gain_LINE_2 = {
+	.negative = 1,
+	.lfield   = &ad1843_LDA2G,
+	.rfield   = &ad1843_RDA2G,
+	.lmute    = &ad1843_LDA2GM,
+	.rmute    = &ad1843_RDA2GM
+};
+static const struct ad1843_gain ad1843_gain_MIC = {
+	.negative = 1,
+	.lfield   = &ad1843_LMCM,
+	.rfield   = &ad1843_RMCM,
+	.lmute    = &ad1843_LMCMM,
+	.rmute    = &ad1843_RMCMM
+};
+static const struct ad1843_gain ad1843_gain_PCM_0 = {
+	.negative = 1,
+	.lfield   = &ad1843_LDA1G,
+	.rfield   = &ad1843_RDA1G,
+	.lmute    = &ad1843_LDA1GM,
+	.rmute    = &ad1843_RDA1GM
+};
+static const struct ad1843_gain ad1843_gain_PCM_1 = {
+	.negative = 1,
+	.lfield   = &ad1843_LD2M,
+	.rfield   = &ad1843_RD2M,
+	.lmute    = &ad1843_LD2MM,
+	.rmute    = &ad1843_RD2MM
+};
+
+static const struct ad1843_gain *ad1843_gain[AD1843_GAIN_SIZE] =
+{
+	&ad1843_gain_RECLEV,
+	&ad1843_gain_LINE,
+	&ad1843_gain_LINE_2,
+	&ad1843_gain_MIC,
+	&ad1843_gain_PCM_0,
+	&ad1843_gain_PCM_1,
+};
+
+/* read the current value of an AD1843 bitfield. */
+
+static int ad1843_read_bits(struct snd_ad1843 *ad1843,
+			    const struct ad1843_bitfield *field)
+{
+	int w;
+
+	w = ad1843->read(ad1843->chip, field->reg);
+	return w >> field->lo_bit & ((1 << field->nbits) - 1);
+}
+
+/*
+ * write a new value to an AD1843 bitfield and return the old value.
+ */
+
+static int ad1843_write_bits(struct snd_ad1843 *ad1843,
+			     const struct ad1843_bitfield *field,
+			     int newval)
+{
+	int w, mask, oldval, newbits;
+
+	w = ad1843->read(ad1843->chip, field->reg);
+	mask = ((1 << field->nbits) - 1) << field->lo_bit;
+	oldval = (w & mask) >> field->lo_bit;
+	newbits = (newval << field->lo_bit) & mask;
+	w = (w & ~mask) | newbits;
+	ad1843->write(ad1843->chip, field->reg, w);
+
+	return oldval;
+}
+
+/*
+ * ad1843_read_multi reads multiple bitfields from the same AD1843
+ * register.  It uses a single read cycle to do it.  (Reading the
+ * ad1843 requires 256 bit times at 12.288 MHz, or nearly 20
+ * microseconds.)
+ *
+ * Called like this.
+ *
+ *  ad1843_read_multi(ad1843, nfields,
+ *		      &ad1843_FIELD1, &val1,
+ *		      &ad1843_FIELD2, &val2, ...);
+ */
+
+static void ad1843_read_multi(struct snd_ad1843 *ad1843, int argcount, ...)
+{
+	va_list ap;
+	const struct ad1843_bitfield *fp;
+	int w = 0, mask, *value, reg = -1;
+
+	va_start(ap, argcount);
+	while (--argcount >= 0) {
+		fp = va_arg(ap, const struct ad1843_bitfield *);
+		value = va_arg(ap, int *);
+		if (reg == -1) {
+			reg = fp->reg;
+			w = ad1843->read(ad1843->chip, reg);
+		}
+
+		mask = (1 << fp->nbits) - 1;
+		*value = w >> fp->lo_bit & mask;
+	}
+	va_end(ap);
+}
+
+/*
+ * ad1843_write_multi stores multiple bitfields into the same AD1843
+ * register.  It uses one read and one write cycle to do it.
+ *
+ * Called like this.
+ *
+ *  ad1843_write_multi(ad1843, nfields,
+ *		       &ad1843_FIELD1, val1,
+ *		       &ad1843_FIELF2, val2, ...);
+ */
+
+static void ad1843_write_multi(struct snd_ad1843 *ad1843, int argcount, ...)
+{
+	va_list ap;
+	int reg;
+	const struct ad1843_bitfield *fp;
+	int value;
+	int w, m, mask, bits;
+
+	mask = 0;
+	bits = 0;
+	reg = -1;
+
+	va_start(ap, argcount);
+	while (--argcount >= 0) {
+		fp = va_arg(ap, const struct ad1843_bitfield *);
+		value = va_arg(ap, int);
+		if (reg == -1)
+			reg = fp->reg;
+		else
+			BUG_ON(reg != fp->reg);
+		m = ((1 << fp->nbits) - 1) << fp->lo_bit;
+		mask |= m;
+		bits |= (value << fp->lo_bit) & m;
+	}
+	va_end(ap);
+
+	if (~mask & 0xFFFF)
+		w = ad1843->read(ad1843->chip, reg);
+	else
+		w = 0;
+	w = (w & ~mask) | bits;
+	ad1843->write(ad1843->chip, reg, w);
+}
+
+int ad1843_get_gain_max(struct snd_ad1843 *ad1843, int id)
+{
+	const struct ad1843_gain *gp = ad1843_gain[id];
+	int ret;
+
+	ret = (1 << gp->lfield->nbits);
+	if (!gp->lmute)
+		ret -= 1;
+	return ret;
+}
+
+/*
+ * ad1843_get_gain reads the specified register and extracts the gain value
+ * using the supplied gain type.
+ */
+
+int ad1843_get_gain(struct snd_ad1843 *ad1843, int id)
+{
+	int lg, rg, lm, rm;
+	const struct ad1843_gain *gp = ad1843_gain[id];
+	unsigned short mask = (1 << gp->lfield->nbits) - 1;
+
+	ad1843_read_multi(ad1843, 2, gp->lfield, &lg, gp->rfield, &rg);
+	if (gp->negative) {
+		lg = mask - lg;
+		rg = mask - rg;
+	}
+	if (gp->lmute) {
+		ad1843_read_multi(ad1843, 2, gp->lmute, &lm, gp->rmute, &rm);
+		if (lm)
+			lg = 0;
+		if (rm)
+			rg = 0;
+	}
+	return lg << 0 | rg << 8;
+}
+
+/*
+ * Set an audio channel's gain.
+ *
+ * Returns the new gain, which may be lower than the old gain.
+ */
+
+int ad1843_set_gain(struct snd_ad1843 *ad1843, int id, int newval)
+{
+	const struct ad1843_gain *gp = ad1843_gain[id];
+	unsigned short mask = (1 << gp->lfield->nbits) - 1;
+
+	int lg = (newval >> 0) & mask;
+	int rg = (newval >> 8) & mask;
+	int lm = (lg == 0) ? 1 : 0;
+	int rm = (rg == 0) ? 1 : 0;
+
+	if (gp->negative) {
+		lg = mask - lg;
+		rg = mask - rg;
+	}
+	if (gp->lmute)
+		ad1843_write_multi(ad1843, 2, gp->lmute, lm, gp->rmute, rm);
+	ad1843_write_multi(ad1843, 2, gp->lfield, lg, gp->rfield, rg);
+	return ad1843_get_gain(ad1843, id);
+}
+
+/* Returns the current recording source */
+
+int ad1843_get_recsrc(struct snd_ad1843 *ad1843)
+{
+	int val = ad1843_read_bits(ad1843, &ad1843_LSS);
+
+	if (val < 0 || val > 2) {
+		val = 2;
+		ad1843_write_multi(ad1843, 2,
+				   &ad1843_LSS, val, &ad1843_RSS, val);
+	}
+	return val;
+}
+
+/*
+ * Set recording source.
+ *
+ * Returns newsrc on success, -errno on failure.
+ */
+
+int ad1843_set_recsrc(struct snd_ad1843 *ad1843, int newsrc)
+{
+	if (newsrc < 0 || newsrc > 2)
+		return -EINVAL;
+
+	ad1843_write_multi(ad1843, 2, &ad1843_LSS, newsrc, &ad1843_RSS, newsrc);
+	return newsrc;
+}
+
+/* Setup ad1843 for D/A conversion. */
+
+void ad1843_setup_dac(struct snd_ad1843 *ad1843,
+		      unsigned int id,
+		      unsigned int framerate,
+		      snd_pcm_format_t fmt,
+		      unsigned int channels)
+{
+	int ad_fmt = 0, ad_mode = 0;
+
+	switch (fmt) {
+	case SNDRV_PCM_FORMAT_S8:
+		ad_fmt = 0;
+		break;
+	case SNDRV_PCM_FORMAT_U8:
+		ad_fmt = 0;
+		break;
+	case SNDRV_PCM_FORMAT_S16_LE:
+		ad_fmt = 1;
+		break;
+	case SNDRV_PCM_FORMAT_MU_LAW:
+		ad_fmt = 2;
+		break;
+	case SNDRV_PCM_FORMAT_A_LAW:
+		ad_fmt = 3;
+		break;
+	default:
+		break;
+	}
+
+	switch (channels) {
+	case 2:
+		ad_mode = 0;
+		break;
+	case 1:
+		ad_mode = 1;
+		break;
+	default:
+		break;
+	}
+
+	if (id) {
+		ad1843_write_bits(ad1843, &ad1843_C2C, framerate);
+		ad1843_write_multi(ad1843, 2,
+				   &ad1843_DA2SM, ad_mode,
+				   &ad1843_DA2F, ad_fmt);
+	} else {
+		ad1843_write_bits(ad1843, &ad1843_C1C, framerate);
+		ad1843_write_multi(ad1843, 2,
+				   &ad1843_DA1SM, ad_mode,
+				   &ad1843_DA1F, ad_fmt);
+	}
+}
+
+void ad1843_shutdown_dac(struct snd_ad1843 *ad1843, unsigned int id)
+{
+	if (id)
+		ad1843_write_bits(ad1843, &ad1843_DA2F, 1);
+	else
+		ad1843_write_bits(ad1843, &ad1843_DA1F, 1);
+}
+
+void ad1843_setup_adc(struct snd_ad1843 *ad1843,
+		      unsigned int framerate,
+		      snd_pcm_format_t fmt,
+		      unsigned int channels)
+{
+	int da_fmt = 0;
+
+	switch (fmt) {
+	case SNDRV_PCM_FORMAT_S8:	da_fmt = 0; break;
+	case SNDRV_PCM_FORMAT_U8:	da_fmt = 0; break;
+	case SNDRV_PCM_FORMAT_S16_LE:	da_fmt = 1; break;
+	case SNDRV_PCM_FORMAT_MU_LAW:	da_fmt = 2; break;
+	case SNDRV_PCM_FORMAT_A_LAW:	da_fmt = 3; break;
+	default:		break;
+	}
+
+	ad1843_write_bits(ad1843, &ad1843_C3C, framerate);
+	ad1843_write_multi(ad1843, 2,
+			   &ad1843_ADLF, da_fmt, &ad1843_ADRF, da_fmt);
+}
+
+void ad1843_shutdown_adc(struct snd_ad1843 *ad1843)
+{
+	/* nothing to do */
+}
+
+/*
+ * Fully initialize the ad1843.  As described in the AD1843 data
+ * sheet, section "START-UP SEQUENCE".  The numbered comments are
+ * subsection headings from the data sheet.  See the data sheet, pages
+ * 52-54, for more info.
+ *
+ * return 0 on success, -errno on failure.  */
+
+int ad1843_init(struct snd_ad1843 *ad1843)
+{
+	unsigned long later;
+
+	if (ad1843_read_bits(ad1843, &ad1843_INIT) != 0) {
+		printk(KERN_ERR "ad1843: AD1843 won't initialize\n");
+		return -EIO;
+	}
+
+	ad1843_write_bits(ad1843, &ad1843_SCF, 1);
+
+	/* 4. Put the conversion resources into standby. */
+	ad1843_write_bits(ad1843, &ad1843_PDNI, 0);
+	later = jiffies + msecs_to_jiffies(500);
+
+	while (ad1843_read_bits(ad1843, &ad1843_PDNO)) {
+		if (time_after(jiffies, later)) {
+			printk(KERN_ERR
+			       "ad1843: AD1843 won't power up\n");
+			return -EIO;
+		}
+		schedule_timeout_interruptible(5);
+	}
+
+	/* 5. Power up the clock generators and enable clock output pins. */
+	ad1843_write_multi(ad1843, 3,
+			   &ad1843_C1EN, 1,
+			   &ad1843_C2EN, 1,
+			   &ad1843_C3EN, 1);
+
+	/* 6. Configure conversion resources while they are in standby. */
+
+	/* DAC1/2 use clock 1/2 as source, ADC uses clock 3.  Always. */
+	ad1843_write_multi(ad1843, 4,
+			   &ad1843_DA1C, 1,
+			   &ad1843_DA2C, 2,
+			   &ad1843_ADLC, 3,
+			   &ad1843_ADRC, 3);
+
+	/* 7. Enable conversion resources. */
+	ad1843_write_bits(ad1843, &ad1843_ADTLK, 1);
+	ad1843_write_multi(ad1843, 7,
+			   &ad1843_ANAEN, 1,
+			   &ad1843_AAMEN, 1,
+			   &ad1843_DA1EN, 1,
+			   &ad1843_DA2EN, 1,
+			   &ad1843_DDMEN, 1,
+			   &ad1843_ADLEN, 1,
+			   &ad1843_ADREN, 1);
+
+	/* 8. Configure conversion resources while they are enabled. */
+
+	/* set gain to 0 for all channels */
+	ad1843_set_gain(ad1843, AD1843_GAIN_RECLEV, 0);
+	ad1843_set_gain(ad1843, AD1843_GAIN_LINE, 0);
+	ad1843_set_gain(ad1843, AD1843_GAIN_LINE_2, 0);
+	ad1843_set_gain(ad1843, AD1843_GAIN_MIC, 0);
+	ad1843_set_gain(ad1843, AD1843_GAIN_PCM_0, 0);
+	ad1843_set_gain(ad1843, AD1843_GAIN_PCM_1, 0);
+
+	/* Unmute all channels. */
+	/* DAC1 */
+	ad1843_write_multi(ad1843, 2, &ad1843_LDA1GM, 0, &ad1843_RDA1GM, 0);
+	/* DAC2 */
+	ad1843_write_multi(ad1843, 2, &ad1843_LDA2GM, 0, &ad1843_RDA2GM, 0);
+
+	/* Set default recording source to Line In and set
+	 * mic gain to +20 dB.
+	 */
+	ad1843_set_recsrc(ad1843, 2);
+	ad1843_write_multi(ad1843, 2, &ad1843_LMGE, 1, &ad1843_RMGE, 1);
+
+	/* Set Speaker Out level to +/- 4V and unmute it. */
+	ad1843_write_multi(ad1843, 3,
+			   &ad1843_HPOS, 1,
+			   &ad1843_HPOM, 0,
+			   &ad1843_MPOM, 0);
+
+	return 0;
+}
diff --git a/ANDROID_3.4.5/sound/mips/au1x00.c b/ANDROID_3.4.5/sound/mips/au1x00.c
new file mode 100644
index 00000000..3f3ec0be
--- /dev/null
+++ b/ANDROID_3.4.5/sound/mips/au1x00.c
@@ -0,0 +1,696 @@
+/*
+ * BRIEF MODULE DESCRIPTION
+ *  Driver for AMD Au1000 MIPS Processor, AC'97 Sound Port
+ *
+ * Copyright 2004 Cooper Street Innovations Inc.
+ * Author: Charles Eidsness	<charles@cooper-street.com>
+ *
+ *  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  SOFTWARE  IS PROVIDED   ``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 AUTHOR  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.
+ *
+ *  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:
+ *
+ * 2004-09-09 Charles Eidsness	-- Original verion -- based on
+ * 				  sa11xx-uda1341.c ALSA driver and the
+ *				  au1000.c OSS driver.
+ * 2004-09-09 Matt Porter	-- Added support for ALSA 1.0.6
+ *
+ */
+
+#include <linux/ioport.h>
+#include <linux/interrupt.h>
+#include <linux/init.h>
+#include <linux/slab.h>
+#include <linux/module.h>
+#include <sound/core.h>
+#include <sound/initval.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/ac97_codec.h>
+#include <asm/mach-au1x00/au1000.h>
+#include <asm/mach-au1x00/au1000_dma.h>
+
+MODULE_AUTHOR("Charles Eidsness <charles@cooper-street.com>");
+MODULE_DESCRIPTION("Au1000 AC'97 ALSA Driver");
+MODULE_LICENSE("GPL");
+MODULE_SUPPORTED_DEVICE("{{AMD,Au1000 AC'97}}");
+
+#define PLAYBACK 0
+#define CAPTURE 1
+#define AC97_SLOT_3 0x01
+#define AC97_SLOT_4 0x02
+#define AC97_SLOT_6 0x08
+#define AC97_CMD_IRQ 31
+#define READ 0
+#define WRITE 1
+#define READ_WAIT 2
+#define RW_DONE 3
+
+struct au1000_period
+{
+	u32 start;
+	u32 relative_end;	/*realtive to start of buffer*/
+	struct au1000_period * next;
+};
+
+/*Au1000 AC97 Port Control Reisters*/
+struct au1000_ac97_reg {
+	u32 volatile config;
+	u32 volatile status;
+	u32 volatile data;
+	u32 volatile cmd;
+	u32 volatile cntrl;
+};
+
+struct audio_stream {
+	struct snd_pcm_substream *substream;
+	int dma;
+	spinlock_t dma_lock;
+	struct au1000_period * buffer;
+	unsigned int period_size;
+	unsigned int periods;
+};
+
+struct snd_au1000 {
+	struct snd_card *card;
+	struct au1000_ac97_reg volatile *ac97_ioport;
+
+	struct resource *ac97_res_port;
+	spinlock_t ac97_lock;
+	struct snd_ac97 *ac97;
+
+	struct snd_pcm *pcm;
+	struct audio_stream *stream[2];	/* playback & capture */
+};
+
+/*--------------------------- Local Functions --------------------------------*/
+static void
+au1000_set_ac97_xmit_slots(struct snd_au1000 *au1000, long xmit_slots)
+{
+	u32 volatile ac97_config;
+
+	spin_lock(&au1000->ac97_lock);
+	ac97_config = au1000->ac97_ioport->config;
+	ac97_config = ac97_config & ~AC97C_XMIT_SLOTS_MASK;
+	ac97_config |= (xmit_slots << AC97C_XMIT_SLOTS_BIT);
+	au1000->ac97_ioport->config = ac97_config;
+	spin_unlock(&au1000->ac97_lock);
+}
+
+static void
+au1000_set_ac97_recv_slots(struct snd_au1000 *au1000, long recv_slots)
+{
+	u32 volatile ac97_config;
+
+	spin_lock(&au1000->ac97_lock);
+	ac97_config = au1000->ac97_ioport->config;
+	ac97_config = ac97_config & ~AC97C_RECV_SLOTS_MASK;
+	ac97_config |= (recv_slots << AC97C_RECV_SLOTS_BIT);
+	au1000->ac97_ioport->config = ac97_config;
+	spin_unlock(&au1000->ac97_lock);
+}
+
+
+static void
+au1000_release_dma_link(struct audio_stream *stream)
+{
+	struct au1000_period * pointer;
+	struct au1000_period * pointer_next;
+
+	stream->period_size = 0;
+	stream->periods = 0;
+	pointer = stream->buffer;
+	if (! pointer)
+		return;
+	do {
+		pointer_next = pointer->next;
+		kfree(pointer);
+		pointer = pointer_next;
+	} while (pointer != stream->buffer);
+	stream->buffer = NULL;
+}
+
+static int
+au1000_setup_dma_link(struct audio_stream *stream, unsigned int period_bytes,
+		      unsigned int periods)
+{
+	struct snd_pcm_substream *substream = stream->substream;
+	struct snd_pcm_runtime *runtime = substream->runtime;
+	struct au1000_period *pointer;
+	unsigned long dma_start;
+	int i;
+
+	dma_start = virt_to_phys(runtime->dma_area);
+
+	if (stream->period_size == period_bytes &&
+	    stream->periods == periods)
+		return 0; /* not changed */
+
+	au1000_release_dma_link(stream);
+
+	stream->period_size = period_bytes;
+	stream->periods = periods;
+
+	stream->buffer = kmalloc(sizeof(struct au1000_period), GFP_KERNEL);
+	if (! stream->buffer)
+		return -ENOMEM;
+	pointer = stream->buffer;
+	for (i = 0; i < periods; i++) {
+		pointer->start = (u32)(dma_start + (i * period_bytes));
+		pointer->relative_end = (u32) (((i+1) * period_bytes) - 0x1);
+		if (i < periods - 1) {
+			pointer->next = kmalloc(sizeof(struct au1000_period), GFP_KERNEL);
+			if (! pointer->next) {
+				au1000_release_dma_link(stream);
+				return -ENOMEM;
+			}
+			pointer = pointer->next;
+		}
+	}
+	pointer->next = stream->buffer;
+	return 0;
+}
+
+static void
+au1000_dma_stop(struct audio_stream *stream)
+{
+	if (snd_BUG_ON(!stream->buffer))
+		return;
+	disable_dma(stream->dma);
+}
+
+static void
+au1000_dma_start(struct audio_stream *stream)
+{
+	if (snd_BUG_ON(!stream->buffer))
+		return;
+
+	init_dma(stream->dma);
+	if (get_dma_active_buffer(stream->dma) == 0) {
+		clear_dma_done0(stream->dma);
+		set_dma_addr0(stream->dma, stream->buffer->start);
+		set_dma_count0(stream->dma, stream->period_size >> 1);
+		set_dma_addr1(stream->dma, stream->buffer->next->start);
+		set_dma_count1(stream->dma, stream->period_size >> 1);
+	} else {
+		clear_dma_done1(stream->dma);
+		set_dma_addr1(stream->dma, stream->buffer->start);
+		set_dma_count1(stream->dma, stream->period_size >> 1);
+		set_dma_addr0(stream->dma, stream->buffer->next->start);
+		set_dma_count0(stream->dma, stream->period_size >> 1);
+	}
+	enable_dma_buffers(stream->dma);
+	start_dma(stream->dma);
+}
+
+static irqreturn_t
+au1000_dma_interrupt(int irq, void *dev_id)
+{
+	struct audio_stream *stream = (struct audio_stream *) dev_id;
+	struct snd_pcm_substream *substream = stream->substream;
+
+	spin_lock(&stream->dma_lock);
+	switch (get_dma_buffer_done(stream->dma)) {
+	case DMA_D0:
+		stream->buffer = stream->buffer->next;
+		clear_dma_done0(stream->dma);
+		set_dma_addr0(stream->dma, stream->buffer->next->start);
+		set_dma_count0(stream->dma, stream->period_size >> 1);
+		enable_dma_buffer0(stream->dma);
+		break;
+	case DMA_D1:
+		stream->buffer = stream->buffer->next;
+		clear_dma_done1(stream->dma);
+		set_dma_addr1(stream->dma, stream->buffer->next->start);
+		set_dma_count1(stream->dma, stream->period_size >> 1);
+		enable_dma_buffer1(stream->dma);
+		break;
+	case (DMA_D0 | DMA_D1):
+		printk(KERN_ERR "DMA %d missed interrupt.\n",stream->dma);
+		au1000_dma_stop(stream);
+		au1000_dma_start(stream);
+		break;
+	case (~DMA_D0 & ~DMA_D1):
+		printk(KERN_ERR "DMA %d empty irq.\n",stream->dma);
+	}
+	spin_unlock(&stream->dma_lock);
+	snd_pcm_period_elapsed(substream);
+	return IRQ_HANDLED;
+}
+
+/*-------------------------- PCM Audio Streams -------------------------------*/
+
+static unsigned int rates[] = {8000, 11025, 16000, 22050};
+static struct snd_pcm_hw_constraint_list hw_constraints_rates = {
+	.count	= ARRAY_SIZE(rates),
+	.list	= rates,
+	.mask	= 0,
+};
+
+static struct snd_pcm_hardware snd_au1000_hw =
+{
+	.info			= (SNDRV_PCM_INFO_INTERLEAVED | \
+				SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID),
+	.formats		= SNDRV_PCM_FMTBIT_S16_LE,
+	.rates			= (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 |
+				SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_22050),
+	.rate_min		= 8000,
+	.rate_max		= 22050,
+	.channels_min		= 1,
+	.channels_max		= 2,
+	.buffer_bytes_max	= 128*1024,
+	.period_bytes_min	= 32,
+	.period_bytes_max	= 16*1024,
+	.periods_min		= 8,
+	.periods_max		= 255,
+	.fifo_size		= 16,
+};
+
+static int
+snd_au1000_playback_open(struct snd_pcm_substream *substream)
+{
+	struct snd_au1000 *au1000 = substream->pcm->private_data;
+
+	au1000->stream[PLAYBACK]->substream = substream;
+	au1000->stream[PLAYBACK]->buffer = NULL;
+	substream->private_data = au1000->stream[PLAYBACK];
+	substream->runtime->hw = snd_au1000_hw;
+	return (snd_pcm_hw_constraint_list(substream->runtime, 0,
+		SNDRV_PCM_HW_PARAM_RATE, &hw_constraints_rates) < 0);
+}
+
+static int
+snd_au1000_capture_open(struct snd_pcm_substream *substream)
+{
+	struct snd_au1000 *au1000 = substream->pcm->private_data;
+
+	au1000->stream[CAPTURE]->substream = substream;
+	au1000->stream[CAPTURE]->buffer = NULL;
+	substream->private_data = au1000->stream[CAPTURE];
+	substream->runtime->hw = snd_au1000_hw;
+	return (snd_pcm_hw_constraint_list(substream->runtime, 0,
+		SNDRV_PCM_HW_PARAM_RATE, &hw_constraints_rates) < 0);
+}
+
+static int
+snd_au1000_playback_close(struct snd_pcm_substream *substream)
+{
+	struct snd_au1000 *au1000 = substream->pcm->private_data;
+
+	au1000->stream[PLAYBACK]->substream = NULL;
+	return 0;
+}
+
+static int
+snd_au1000_capture_close(struct snd_pcm_substream *substream)
+{
+	struct snd_au1000 *au1000 = substream->pcm->private_data;
+
+	au1000->stream[CAPTURE]->substream = NULL;
+	return 0;
+}
+
+static int
+snd_au1000_hw_params(struct snd_pcm_substream *substream,
+					struct snd_pcm_hw_params *hw_params)
+{
+	struct audio_stream *stream = substream->private_data;
+	int err;
+
+	err = snd_pcm_lib_malloc_pages(substream,
+				       params_buffer_bytes(hw_params));
+	if (err < 0)
+		return err;
+	return au1000_setup_dma_link(stream,
+				     params_period_bytes(hw_params),
+				     params_periods(hw_params));
+}
+
+static int
+snd_au1000_hw_free(struct snd_pcm_substream *substream)
+{
+	struct audio_stream *stream = substream->private_data;
+	au1000_release_dma_link(stream);
+	return snd_pcm_lib_free_pages(substream);
+}
+
+static int
+snd_au1000_playback_prepare(struct snd_pcm_substream *substream)
+{
+	struct snd_au1000 *au1000 = substream->pcm->private_data;
+	struct snd_pcm_runtime *runtime = substream->runtime;
+
+	if (runtime->channels == 1)
+		au1000_set_ac97_xmit_slots(au1000, AC97_SLOT_4);
+	else
+		au1000_set_ac97_xmit_slots(au1000, AC97_SLOT_3 | AC97_SLOT_4);
+	snd_ac97_set_rate(au1000->ac97, AC97_PCM_FRONT_DAC_RATE, runtime->rate);
+	return 0;
+}
+
+static int
+snd_au1000_capture_prepare(struct snd_pcm_substream *substream)
+{
+	struct snd_au1000 *au1000 = substream->pcm->private_data;
+	struct snd_pcm_runtime *runtime = substream->runtime;
+
+	if (runtime->channels == 1)
+		au1000_set_ac97_recv_slots(au1000, AC97_SLOT_4);
+	else
+		au1000_set_ac97_recv_slots(au1000, AC97_SLOT_3 | AC97_SLOT_4);
+	snd_ac97_set_rate(au1000->ac97, AC97_PCM_LR_ADC_RATE, runtime->rate);
+	return 0;
+}
+
+static int
+snd_au1000_trigger(struct snd_pcm_substream *substream, int cmd)
+{
+	struct audio_stream *stream = substream->private_data;
+	int err = 0;
+
+	spin_lock(&stream->dma_lock);
+	switch (cmd) {
+	case SNDRV_PCM_TRIGGER_START:
+		au1000_dma_start(stream);
+		break;
+	case SNDRV_PCM_TRIGGER_STOP:
+		au1000_dma_stop(stream);
+		break;
+	default:
+		err = -EINVAL;
+		break;
+	}
+	spin_unlock(&stream->dma_lock);
+	return err;
+}
+
+static snd_pcm_uframes_t
+snd_au1000_pointer(struct snd_pcm_substream *substream)
+{
+	struct audio_stream *stream = substream->private_data;
+	struct snd_pcm_runtime *runtime = substream->runtime;
+	long location;
+
+	spin_lock(&stream->dma_lock);
+	location = get_dma_residue(stream->dma);
+	spin_unlock(&stream->dma_lock);
+	location = stream->buffer->relative_end - location;
+	if (location == -1)
+		location = 0;
+	return bytes_to_frames(runtime,location);
+}
+
+static struct snd_pcm_ops snd_card_au1000_playback_ops = {
+	.open			= snd_au1000_playback_open,
+	.close			= snd_au1000_playback_close,
+	.ioctl			= snd_pcm_lib_ioctl,
+	.hw_params	        = snd_au1000_hw_params,
+	.hw_free	        = snd_au1000_hw_free,
+	.prepare		= snd_au1000_playback_prepare,
+	.trigger		= snd_au1000_trigger,
+	.pointer		= snd_au1000_pointer,
+};
+
+static struct snd_pcm_ops snd_card_au1000_capture_ops = {
+	.open			= snd_au1000_capture_open,
+	.close			= snd_au1000_capture_close,
+	.ioctl			= snd_pcm_lib_ioctl,
+	.hw_params	        = snd_au1000_hw_params,
+	.hw_free	        = snd_au1000_hw_free,
+	.prepare		= snd_au1000_capture_prepare,
+	.trigger		= snd_au1000_trigger,
+	.pointer		= snd_au1000_pointer,
+};
+
+static int __devinit
+snd_au1000_pcm_new(struct snd_au1000 *au1000)
+{
+	struct snd_pcm *pcm;
+	int err;
+	unsigned long flags;
+
+	if ((err = snd_pcm_new(au1000->card, "AU1000 AC97 PCM", 0, 1, 1, &pcm)) < 0)
+		return err;
+
+	snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_CONTINUOUS,
+		snd_dma_continuous_data(GFP_KERNEL), 128*1024, 128*1024);
+
+	snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK,
+		&snd_card_au1000_playback_ops);
+	snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE,
+		&snd_card_au1000_capture_ops);
+
+	pcm->private_data = au1000;
+	pcm->info_flags = 0;
+	strcpy(pcm->name, "Au1000 AC97 PCM");
+
+	spin_lock_init(&au1000->stream[PLAYBACK]->dma_lock);
+	spin_lock_init(&au1000->stream[CAPTURE]->dma_lock);
+
+	flags = claim_dma_lock();
+	if ((au1000->stream[PLAYBACK]->dma = request_au1000_dma(DMA_ID_AC97C_TX,
+			"AC97 TX", au1000_dma_interrupt, 0,
+			au1000->stream[PLAYBACK])) < 0) {
+		release_dma_lock(flags);
+		return -EBUSY;
+	}
+	if ((au1000->stream[CAPTURE]->dma = request_au1000_dma(DMA_ID_AC97C_RX,
+			"AC97 RX", au1000_dma_interrupt, 0,
+			au1000->stream[CAPTURE])) < 0){
+		release_dma_lock(flags);
+		return -EBUSY;
+	}
+	/* enable DMA coherency in read/write DMA channels */
+	set_dma_mode(au1000->stream[PLAYBACK]->dma,
+		     get_dma_mode(au1000->stream[PLAYBACK]->dma) & ~DMA_NC);
+	set_dma_mode(au1000->stream[CAPTURE]->dma,
+		     get_dma_mode(au1000->stream[CAPTURE]->dma) & ~DMA_NC);
+	release_dma_lock(flags);
+	au1000->pcm = pcm;
+	return 0;
+}
+
+
+/*-------------------------- AC97 CODEC Control ------------------------------*/
+
+static unsigned short
+snd_au1000_ac97_read(struct snd_ac97 *ac97, unsigned short reg)
+{
+	struct snd_au1000 *au1000 = ac97->private_data;
+	u32 volatile cmd;
+	u16 volatile data;
+	int             i;
+
+	spin_lock(&au1000->ac97_lock);
+/* would rather use the interrupt than this polling but it works and I can't
+get the interrupt driven case to work efficiently */
+	for (i = 0; i < 0x5000; i++)
+		if (!(au1000->ac97_ioport->status & AC97C_CP))
+			break;
+	if (i == 0x5000)
+		printk(KERN_ERR "au1000 AC97: AC97 command read timeout\n");
+
+	cmd = (u32) reg & AC97C_INDEX_MASK;
+	cmd |= AC97C_READ;
+	au1000->ac97_ioport->cmd = cmd;
+
+	/* now wait for the data */
+	for (i = 0; i < 0x5000; i++)
+		if (!(au1000->ac97_ioport->status & AC97C_CP))
+			break;
+	if (i == 0x5000) {
+		printk(KERN_ERR "au1000 AC97: AC97 command read timeout\n");
+		spin_unlock(&au1000->ac97_lock);
+		return 0;
+	}
+
+	data = au1000->ac97_ioport->cmd & 0xffff;
+	spin_unlock(&au1000->ac97_lock);
+
+	return data;
+
+}
+
+
+static void
+snd_au1000_ac97_write(struct snd_ac97 *ac97, unsigned short reg, unsigned short val)
+{
+	struct snd_au1000 *au1000 = ac97->private_data;
+	u32 cmd;
+	int i;
+
+	spin_lock(&au1000->ac97_lock);
+/* would rather use the interrupt than this polling but it works and I can't
+get the interrupt driven case to work efficiently */
+	for (i = 0; i < 0x5000; i++)
+		if (!(au1000->ac97_ioport->status & AC97C_CP))
+			break;
+	if (i == 0x5000)
+		printk(KERN_ERR "au1000 AC97: AC97 command write timeout\n");
+
+	cmd = (u32) reg & AC97C_INDEX_MASK;
+	cmd &= ~AC97C_READ;
+	cmd |= ((u32) val << AC97C_WD_BIT);
+	au1000->ac97_ioport->cmd = cmd;
+	spin_unlock(&au1000->ac97_lock);
+}
+
+static int __devinit
+snd_au1000_ac97_new(struct snd_au1000 *au1000)
+{
+	int err;
+	struct snd_ac97_bus *pbus;
+	struct snd_ac97_template ac97;
+ 	static struct snd_ac97_bus_ops ops = {
+		.write = snd_au1000_ac97_write,
+		.read = snd_au1000_ac97_read,
+	};
+
+	if ((au1000->ac97_res_port = request_mem_region(CPHYSADDR(AC97C_CONFIG),
+	       		0x100000, "Au1x00 AC97")) == NULL) {
+		snd_printk(KERN_ERR "ALSA AC97: can't grap AC97 port\n");
+		return -EBUSY;
+	}
+	au1000->ac97_ioport = (struct au1000_ac97_reg *)
+		KSEG1ADDR(au1000->ac97_res_port->start);
+
+	spin_lock_init(&au1000->ac97_lock);
+
+	/* configure pins for AC'97
+	TODO: move to board_setup.c */
+	au_writel(au_readl(SYS_PINFUNC) & ~0x02, SYS_PINFUNC);
+
+	/* Initialise Au1000's AC'97 Control Block */
+	au1000->ac97_ioport->cntrl = AC97C_RS | AC97C_CE;
+	udelay(10);
+	au1000->ac97_ioport->cntrl = AC97C_CE;
+	udelay(10);
+
+	/* Initialise External CODEC -- cold reset */
+	au1000->ac97_ioport->config = AC97C_RESET;
+	udelay(10);
+	au1000->ac97_ioport->config = 0x0;
+	mdelay(5);
+
+	/* Initialise AC97 middle-layer */
+	if ((err = snd_ac97_bus(au1000->card, 0, &ops, au1000, &pbus)) < 0)
+ 		return err;
+
+	memset(&ac97, 0, sizeof(ac97));
+	ac97.private_data = au1000;
+	if ((err = snd_ac97_mixer(pbus, &ac97, &au1000->ac97)) < 0)
+		return err;
+
+	return 0;
+}
+
+/*------------------------------ Setup / Destroy ----------------------------*/
+
+void
+snd_au1000_free(struct snd_card *card)
+{
+	struct snd_au1000 *au1000 = card->private_data;
+
+	if (au1000->ac97_res_port) {
+		/* put internal AC97 block into reset */
+		au1000->ac97_ioport->cntrl = AC97C_RS;
+		au1000->ac97_ioport = NULL;
+		release_and_free_resource(au1000->ac97_res_port);
+	}
+
+	if (au1000->stream[PLAYBACK]) {
+	  	if (au1000->stream[PLAYBACK]->dma >= 0)
+			free_au1000_dma(au1000->stream[PLAYBACK]->dma);
+		kfree(au1000->stream[PLAYBACK]);
+	}
+
+	if (au1000->stream[CAPTURE]) {
+		if (au1000->stream[CAPTURE]->dma >= 0)
+			free_au1000_dma(au1000->stream[CAPTURE]->dma);
+		kfree(au1000->stream[CAPTURE]);
+	}
+}
+
+
+static struct snd_card *au1000_card;
+
+static int __init
+au1000_init(void)
+{
+	int err;
+	struct snd_card *card;
+	struct snd_au1000 *au1000;
+
+	err = snd_card_create(-1, "AC97", THIS_MODULE,
+			      sizeof(struct snd_au1000), &card);
+	if (err < 0)
+		return err;
+
+	card->private_free = snd_au1000_free;
+	au1000 = card->private_data;
+	au1000->card = card;
+
+	au1000->stream[PLAYBACK] = kmalloc(sizeof(struct audio_stream), GFP_KERNEL);
+	au1000->stream[CAPTURE ] = kmalloc(sizeof(struct audio_stream), GFP_KERNEL);
+	/* so that snd_au1000_free will work as intended */
+ 	au1000->ac97_res_port = NULL;
+	if (au1000->stream[PLAYBACK])
+		au1000->stream[PLAYBACK]->dma = -1;
+	if (au1000->stream[CAPTURE ])
+		au1000->stream[CAPTURE ]->dma = -1;
+
+	if (au1000->stream[PLAYBACK] == NULL ||
+	    au1000->stream[CAPTURE ] == NULL) {
+		snd_card_free(card);
+		return -ENOMEM;
+	}
+
+	if ((err = snd_au1000_ac97_new(au1000)) < 0 ) {
+		snd_card_free(card);
+		return err;
+	}
+
+	if ((err = snd_au1000_pcm_new(au1000)) < 0) {
+		snd_card_free(card);
+		return err;
+	}
+
+	strcpy(card->driver, "Au1000-AC97");
+	strcpy(card->shortname, "AMD Au1000-AC97");
+	sprintf(card->longname, "AMD Au1000--AC97 ALSA Driver");
+
+	if ((err = snd_card_register(card)) < 0) {
+		snd_card_free(card);
+		return err;
+	}
+
+	printk(KERN_INFO "ALSA AC97: Driver Initialized\n");
+	au1000_card = card;
+	return 0;
+}
+
+static void __exit au1000_exit(void)
+{
+	snd_card_free(au1000_card);
+}
+
+module_init(au1000_init);
+module_exit(au1000_exit);
+
diff --git a/ANDROID_3.4.5/sound/mips/hal2.c b/ANDROID_3.4.5/sound/mips/hal2.c
new file mode 100644
index 00000000..5f88d1f0
--- /dev/null
+++ b/ANDROID_3.4.5/sound/mips/hal2.c
@@ -0,0 +1,938 @@
+/*
+ *  Driver for A2 audio system used in SGI machines
+ *  Copyright (c) 2008 Thomas Bogendoerfer <tsbogend@alpha.fanken.de>
+ *
+ *  Based on OSS code from Ladislav Michl <ladis@linux-mips.org>, which
+ *  was based on code from Ulf Carlsson
+ *
+ *  This program is free software; you can redistribute it and/or modify
+ *  it under the terms of the GNU General Public License version 2 as
+ *  published by the Free Software Foundation.
+ *
+ *  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.
+ *
+ */
+#include <linux/kernel.h>
+#include <linux/init.h>
+#include <linux/interrupt.h>
+#include <linux/dma-mapping.h>
+#include <linux/platform_device.h>
+#include <linux/io.h>
+#include <linux/slab.h>
+#include <linux/module.h>
+
+#include <asm/sgi/hpc3.h>
+#include <asm/sgi/ip22.h>
+
+#include <sound/core.h>
+#include <sound/control.h>
+#include <sound/pcm.h>
+#include <sound/pcm-indirect.h>
+#include <sound/initval.h>
+
+#include "hal2.h"
+
+static int index = SNDRV_DEFAULT_IDX1;  /* Index 0-MAX */
+static char *id = SNDRV_DEFAULT_STR1;   /* ID for this card */
+
+module_param(index, int, 0444);
+MODULE_PARM_DESC(index, "Index value for SGI HAL2 soundcard.");
+module_param(id, charp, 0444);
+MODULE_PARM_DESC(id, "ID string for SGI HAL2 soundcard.");
+MODULE_DESCRIPTION("ALSA driver for SGI HAL2 audio");
+MODULE_AUTHOR("Thomas Bogendoerfer");
+MODULE_LICENSE("GPL");
+
+
+#define H2_BLOCK_SIZE	1024
+#define H2_BUF_SIZE	16384
+
+struct hal2_pbus {
+	struct hpc3_pbus_dmacregs *pbus;
+	int pbusnr;
+	unsigned int ctrl;		/* Current state of pbus->pbdma_ctrl */
+};
+
+struct hal2_desc {
+	struct hpc_dma_desc desc;
+	u32 pad;			/* padding */
+};
+
+struct hal2_codec {
+	struct snd_pcm_indirect pcm_indirect;
+	struct snd_pcm_substream *substream;
+
+	unsigned char *buffer;
+	dma_addr_t buffer_dma;
+	struct hal2_desc *desc;
+	dma_addr_t desc_dma;
+	int desc_count;
+	struct hal2_pbus pbus;
+	int voices;			/* mono/stereo */
+	unsigned int sample_rate;
+	unsigned int master;		/* Master frequency */
+	unsigned short mod;		/* MOD value */
+	unsigned short inc;		/* INC value */
+};
+
+#define H2_MIX_OUTPUT_ATT	0
+#define H2_MIX_INPUT_GAIN	1
+
+struct snd_hal2 {
+	struct snd_card *card;
+
+	struct hal2_ctl_regs *ctl_regs;	/* HAL2 ctl registers */
+	struct hal2_aes_regs *aes_regs;	/* HAL2 aes registers */
+	struct hal2_vol_regs *vol_regs;	/* HAL2 vol registers */
+	struct hal2_syn_regs *syn_regs;	/* HAL2 syn registers */
+
+	struct hal2_codec dac;
+	struct hal2_codec adc;
+};
+
+#define H2_INDIRECT_WAIT(regs)	while (hal2_read(&regs->isr) & H2_ISR_TSTATUS);
+
+#define H2_READ_ADDR(addr)	(addr | (1<<7))
+#define H2_WRITE_ADDR(addr)	(addr)
+
+static inline u32 hal2_read(u32 *reg)
+{
+	return __raw_readl(reg);
+}
+
+static inline void hal2_write(u32 val, u32 *reg)
+{
+	__raw_writel(val, reg);
+}
+
+
+static u32 hal2_i_read32(struct snd_hal2 *hal2, u16 addr)
+{
+	u32 ret;
+	struct hal2_ctl_regs *regs = hal2->ctl_regs;
+
+	hal2_write(H2_READ_ADDR(addr), &regs->iar);
+	H2_INDIRECT_WAIT(regs);
+	ret = hal2_read(&regs->idr0) & 0xffff;
+	hal2_write(H2_READ_ADDR(addr) | 0x1, &regs->iar);
+	H2_INDIRECT_WAIT(regs);
+	ret |= (hal2_read(&regs->idr0) & 0xffff) << 16;
+	return ret;
+}
+
+static void hal2_i_write16(struct snd_hal2 *hal2, u16 addr, u16 val)
+{
+	struct hal2_ctl_regs *regs = hal2->ctl_regs;
+
+	hal2_write(val, &regs->idr0);
+	hal2_write(0, &regs->idr1);
+	hal2_write(0, &regs->idr2);
+	hal2_write(0, &regs->idr3);
+	hal2_write(H2_WRITE_ADDR(addr), &regs->iar);
+	H2_INDIRECT_WAIT(regs);
+}
+
+static void hal2_i_write32(struct snd_hal2 *hal2, u16 addr, u32 val)
+{
+	struct hal2_ctl_regs *regs = hal2->ctl_regs;
+
+	hal2_write(val & 0xffff, &regs->idr0);
+	hal2_write(val >> 16, &regs->idr1);
+	hal2_write(0, &regs->idr2);
+	hal2_write(0, &regs->idr3);
+	hal2_write(H2_WRITE_ADDR(addr), &regs->iar);
+	H2_INDIRECT_WAIT(regs);
+}
+
+static void hal2_i_setbit16(struct snd_hal2 *hal2, u16 addr, u16 bit)
+{
+	struct hal2_ctl_regs *regs = hal2->ctl_regs;
+
+	hal2_write(H2_READ_ADDR(addr), &regs->iar);
+	H2_INDIRECT_WAIT(regs);
+	hal2_write((hal2_read(&regs->idr0) & 0xffff) | bit, &regs->idr0);
+	hal2_write(0, &regs->idr1);
+	hal2_write(0, &regs->idr2);
+	hal2_write(0, &regs->idr3);
+	hal2_write(H2_WRITE_ADDR(addr), &regs->iar);
+	H2_INDIRECT_WAIT(regs);
+}
+
+static void hal2_i_clearbit16(struct snd_hal2 *hal2, u16 addr, u16 bit)
+{
+	struct hal2_ctl_regs *regs = hal2->ctl_regs;
+
+	hal2_write(H2_READ_ADDR(addr), &regs->iar);
+	H2_INDIRECT_WAIT(regs);
+	hal2_write((hal2_read(&regs->idr0) & 0xffff) & ~bit, &regs->idr0);
+	hal2_write(0, &regs->idr1);
+	hal2_write(0, &regs->idr2);
+	hal2_write(0, &regs->idr3);
+	hal2_write(H2_WRITE_ADDR(addr), &regs->iar);
+	H2_INDIRECT_WAIT(regs);
+}
+
+static int hal2_gain_info(struct snd_kcontrol *kcontrol,
+			       struct snd_ctl_elem_info *uinfo)
+{
+	uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
+	uinfo->count = 2;
+	uinfo->value.integer.min = 0;
+	switch ((int)kcontrol->private_value) {
+	case H2_MIX_OUTPUT_ATT:
+		uinfo->value.integer.max = 31;
+		break;
+	case H2_MIX_INPUT_GAIN:
+		uinfo->value.integer.max = 15;
+		break;
+	}
+	return 0;
+}
+
+static int hal2_gain_get(struct snd_kcontrol *kcontrol,
+			       struct snd_ctl_elem_value *ucontrol)
+{
+	struct snd_hal2 *hal2 = snd_kcontrol_chip(kcontrol);
+	u32 tmp;
+	int l, r;
+
+	switch ((int)kcontrol->private_value) {
+	case H2_MIX_OUTPUT_ATT:
+		tmp = hal2_i_read32(hal2, H2I_DAC_C2);
+		if (tmp & H2I_C2_MUTE) {
+			l = 0;
+			r = 0;
+		} else {
+			l = 31 - ((tmp >> H2I_C2_L_ATT_SHIFT) & 31);
+			r = 31 - ((tmp >> H2I_C2_R_ATT_SHIFT) & 31);
+		}
+		break;
+	case H2_MIX_INPUT_GAIN:
+		tmp = hal2_i_read32(hal2, H2I_ADC_C2);
+		l = (tmp >> H2I_C2_L_GAIN_SHIFT) & 15;
+		r = (tmp >> H2I_C2_R_GAIN_SHIFT) & 15;
+		break;
+	}
+	ucontrol->value.integer.value[0] = l;
+	ucontrol->value.integer.value[1] = r;
+
+	return 0;
+}
+
+static int hal2_gain_put(struct snd_kcontrol *kcontrol,
+			 struct snd_ctl_elem_value *ucontrol)
+{
+	struct snd_hal2 *hal2 = snd_kcontrol_chip(kcontrol);
+	u32 old, new;
+	int l, r;
+
+	l = ucontrol->value.integer.value[0];
+	r = ucontrol->value.integer.value[1];
+
+	switch ((int)kcontrol->private_value) {
+	case H2_MIX_OUTPUT_ATT:
+		old = hal2_i_read32(hal2, H2I_DAC_C2);
+		new = old & ~(H2I_C2_L_ATT_M | H2I_C2_R_ATT_M | H2I_C2_MUTE);
+		if (l | r) {
+			l = 31 - l;
+			r = 31 - r;
+			new |= (l << H2I_C2_L_ATT_SHIFT);
+			new |= (r << H2I_C2_R_ATT_SHIFT);
+		} else
+			new |= H2I_C2_L_ATT_M | H2I_C2_R_ATT_M | H2I_C2_MUTE;
+		hal2_i_write32(hal2, H2I_DAC_C2, new);
+		break;
+	case H2_MIX_INPUT_GAIN:
+		old = hal2_i_read32(hal2, H2I_ADC_C2);
+		new = old & ~(H2I_C2_L_GAIN_M | H2I_C2_R_GAIN_M);
+		new |= (l << H2I_C2_L_GAIN_SHIFT);
+		new |= (r << H2I_C2_R_GAIN_SHIFT);
+		hal2_i_write32(hal2, H2I_ADC_C2, new);
+		break;
+	}
+	return old != new;
+}
+
+static struct snd_kcontrol_new hal2_ctrl_headphone __devinitdata = {
+	.iface          = SNDRV_CTL_ELEM_IFACE_MIXER,
+	.name           = "Headphone Playback Volume",
+	.access         = SNDRV_CTL_ELEM_ACCESS_READWRITE,
+	.private_value  = H2_MIX_OUTPUT_ATT,
+	.info           = hal2_gain_info,
+	.get            = hal2_gain_get,
+	.put            = hal2_gain_put,
+};
+
+static struct snd_kcontrol_new hal2_ctrl_mic __devinitdata = {
+	.iface          = SNDRV_CTL_ELEM_IFACE_MIXER,
+	.name           = "Mic Capture Volume",
+	.access         = SNDRV_CTL_ELEM_ACCESS_READWRITE,
+	.private_value  = H2_MIX_INPUT_GAIN,
+	.info           = hal2_gain_info,
+	.get            = hal2_gain_get,
+	.put            = hal2_gain_put,
+};
+
+static int __devinit hal2_mixer_create(struct snd_hal2 *hal2)
+{
+	int err;
+
+	/* mute DAC */
+	hal2_i_write32(hal2, H2I_DAC_C2,
+		       H2I_C2_L_ATT_M | H2I_C2_R_ATT_M | H2I_C2_MUTE);
+	/* mute ADC */
+	hal2_i_write32(hal2, H2I_ADC_C2, 0);
+
+	err = snd_ctl_add(hal2->card,
+			  snd_ctl_new1(&hal2_ctrl_headphone, hal2));
+	if (err < 0)
+		return err;
+
+	err = snd_ctl_add(hal2->card,
+			  snd_ctl_new1(&hal2_ctrl_mic, hal2));
+	if (err < 0)
+		return err;
+
+	return 0;
+}
+
+static irqreturn_t hal2_interrupt(int irq, void *dev_id)
+{
+	struct snd_hal2 *hal2 = dev_id;
+	irqreturn_t ret = IRQ_NONE;
+
+	/* decide what caused this interrupt */
+	if (hal2->dac.pbus.pbus->pbdma_ctrl & HPC3_PDMACTRL_INT) {
+		snd_pcm_period_elapsed(hal2->dac.substream);
+		ret = IRQ_HANDLED;
+	}
+	if (hal2->adc.pbus.pbus->pbdma_ctrl & HPC3_PDMACTRL_INT) {
+		snd_pcm_period_elapsed(hal2->adc.substream);
+		ret = IRQ_HANDLED;
+	}
+	return ret;
+}
+
+static int hal2_compute_rate(struct hal2_codec *codec, unsigned int rate)
+{
+	unsigned short mod;
+
+	if (44100 % rate < 48000 % rate) {
+		mod = 4 * 44100 / rate;
+		codec->master = 44100;
+	} else {
+		mod = 4 * 48000 / rate;
+		codec->master = 48000;
+	}
+
+	codec->inc = 4;
+	codec->mod = mod;
+	rate = 4 * codec->master / mod;
+
+	return rate;
+}
+
+static void hal2_set_dac_rate(struct snd_hal2 *hal2)
+{
+	unsigned int master = hal2->dac.master;
+	int inc = hal2->dac.inc;
+	int mod = hal2->dac.mod;
+
+	hal2_i_write16(hal2, H2I_BRES1_C1, (master == 44100) ? 1 : 0);
+	hal2_i_write32(hal2, H2I_BRES1_C2,
+		       ((0xffff & (inc - mod - 1)) << 16) | inc);
+}
+
+static void hal2_set_adc_rate(struct snd_hal2 *hal2)
+{
+	unsigned int master = hal2->adc.master;
+	int inc = hal2->adc.inc;
+	int mod = hal2->adc.mod;
+
+	hal2_i_write16(hal2, H2I_BRES2_C1, (master == 44100) ? 1 : 0);
+	hal2_i_write32(hal2, H2I_BRES2_C2,
+		       ((0xffff & (inc - mod - 1)) << 16) | inc);
+}
+
+static void hal2_setup_dac(struct snd_hal2 *hal2)
+{
+	unsigned int fifobeg, fifoend, highwater, sample_size;
+	struct hal2_pbus *pbus = &hal2->dac.pbus;
+
+	/* Now we set up some PBUS information. The PBUS needs information about
+	 * what portion of the fifo it will use. If it's receiving or
+	 * transmitting, and finally whether the stream is little endian or big
+	 * endian. The information is written later, on the start call.
+	 */
+	sample_size = 2 * hal2->dac.voices;
+	/* Fifo should be set to hold exactly four samples. Highwater mark
+	 * should be set to two samples. */
+	highwater = (sample_size * 2) >> 1;	/* halfwords */
+	fifobeg = 0;				/* playback is first */
+	fifoend = (sample_size * 4) >> 3;	/* doublewords */
+	pbus->ctrl = HPC3_PDMACTRL_RT | HPC3_PDMACTRL_LD |
+		     (highwater << 8) | (fifobeg << 16) | (fifoend << 24);
+	/* We disable everything before we do anything at all */
+	pbus->pbus->pbdma_ctrl = HPC3_PDMACTRL_LD;
+	hal2_i_clearbit16(hal2, H2I_DMA_PORT_EN, H2I_DMA_PORT_EN_CODECTX);
+	/* Setup the HAL2 for playback */
+	hal2_set_dac_rate(hal2);
+	/* Set endianess */
+	hal2_i_clearbit16(hal2, H2I_DMA_END, H2I_DMA_END_CODECTX);
+	/* Set DMA bus */
+	hal2_i_setbit16(hal2, H2I_DMA_DRV, (1 << pbus->pbusnr));
+	/* We are using 1st Bresenham clock generator for playback */
+	hal2_i_write16(hal2, H2I_DAC_C1, (pbus->pbusnr << H2I_C1_DMA_SHIFT)
+			| (1 << H2I_C1_CLKID_SHIFT)
+			| (hal2->dac.voices << H2I_C1_DATAT_SHIFT));
+}
+
+static void hal2_setup_adc(struct snd_hal2 *hal2)
+{
+	unsigned int fifobeg, fifoend, highwater, sample_size;
+	struct hal2_pbus *pbus = &hal2->adc.pbus;
+
+	sample_size = 2 * hal2->adc.voices;
+	highwater = (sample_size * 2) >> 1;		/* halfwords */
+	fifobeg = (4 * 4) >> 3;				/* record is second */
+	fifoend = (4 * 4 + sample_size * 4) >> 3;	/* doublewords */
+	pbus->ctrl = HPC3_PDMACTRL_RT | HPC3_PDMACTRL_RCV | HPC3_PDMACTRL_LD |
+		     (highwater << 8) | (fifobeg << 16) | (fifoend << 24);
+	pbus->pbus->pbdma_ctrl = HPC3_PDMACTRL_LD;
+	hal2_i_clearbit16(hal2, H2I_DMA_PORT_EN, H2I_DMA_PORT_EN_CODECR);
+	/* Setup the HAL2 for record */
+	hal2_set_adc_rate(hal2);
+	/* Set endianess */
+	hal2_i_clearbit16(hal2, H2I_DMA_END, H2I_DMA_END_CODECR);
+	/* Set DMA bus */
+	hal2_i_setbit16(hal2, H2I_DMA_DRV, (1 << pbus->pbusnr));
+	/* We are using 2nd Bresenham clock generator for record */
+	hal2_i_write16(hal2, H2I_ADC_C1, (pbus->pbusnr << H2I_C1_DMA_SHIFT)
+			| (2 << H2I_C1_CLKID_SHIFT)
+			| (hal2->adc.voices << H2I_C1_DATAT_SHIFT));
+}
+
+static void hal2_start_dac(struct snd_hal2 *hal2)
+{
+	struct hal2_pbus *pbus = &hal2->dac.pbus;
+
+	pbus->pbus->pbdma_dptr = hal2->dac.desc_dma;
+	pbus->pbus->pbdma_ctrl = pbus->ctrl | HPC3_PDMACTRL_ACT;
+	/* enable DAC */
+	hal2_i_setbit16(hal2, H2I_DMA_PORT_EN, H2I_DMA_PORT_EN_CODECTX);
+}
+
+static void hal2_start_adc(struct snd_hal2 *hal2)
+{
+	struct hal2_pbus *pbus = &hal2->adc.pbus;
+
+	pbus->pbus->pbdma_dptr = hal2->adc.desc_dma;
+	pbus->pbus->pbdma_ctrl = pbus->ctrl | HPC3_PDMACTRL_ACT;
+	/* enable ADC */
+	hal2_i_setbit16(hal2, H2I_DMA_PORT_EN, H2I_DMA_PORT_EN_CODECR);
+}
+
+static inline void hal2_stop_dac(struct snd_hal2 *hal2)
+{
+	hal2->dac.pbus.pbus->pbdma_ctrl = HPC3_PDMACTRL_LD;
+	/* The HAL2 itself may remain enabled safely */
+}
+
+static inline void hal2_stop_adc(struct snd_hal2 *hal2)
+{
+	hal2->adc.pbus.pbus->pbdma_ctrl = HPC3_PDMACTRL_LD;
+}
+
+static int hal2_alloc_dmabuf(struct hal2_codec *codec)
+{
+	struct hal2_desc *desc;
+	dma_addr_t desc_dma, buffer_dma;
+	int count = H2_BUF_SIZE / H2_BLOCK_SIZE;
+	int i;
+
+	codec->buffer = dma_alloc_noncoherent(NULL, H2_BUF_SIZE,
+					      &buffer_dma, GFP_KERNEL);
+	if (!codec->buffer)
+		return -ENOMEM;
+	desc = dma_alloc_noncoherent(NULL, count * sizeof(struct hal2_desc),
+				     &desc_dma, GFP_KERNEL);
+	if (!desc) {
+		dma_free_noncoherent(NULL, H2_BUF_SIZE,
+				     codec->buffer, buffer_dma);
+		return -ENOMEM;
+	}
+	codec->buffer_dma = buffer_dma;
+	codec->desc_dma = desc_dma;
+	codec->desc = desc;
+	for (i = 0; i < count; i++) {
+		desc->desc.pbuf = buffer_dma + i * H2_BLOCK_SIZE;
+		desc->desc.cntinfo = HPCDMA_XIE | H2_BLOCK_SIZE;
+		desc->desc.pnext = (i == count - 1) ?
+		      desc_dma : desc_dma + (i + 1) * sizeof(struct hal2_desc);
+		desc++;
+	}
+	dma_cache_sync(NULL, codec->desc, count * sizeof(struct hal2_desc),
+		       DMA_TO_DEVICE);
+	codec->desc_count = count;
+	return 0;
+}
+
+static void hal2_free_dmabuf(struct hal2_codec *codec)
+{
+	dma_free_noncoherent(NULL, codec->desc_count * sizeof(struct hal2_desc),
+			     codec->desc, codec->desc_dma);
+	dma_free_noncoherent(NULL, H2_BUF_SIZE, codec->buffer,
+			     codec->buffer_dma);
+}
+
+static struct snd_pcm_hardware hal2_pcm_hw = {
+	.info = (SNDRV_PCM_INFO_MMAP |
+		 SNDRV_PCM_INFO_MMAP_VALID |
+		 SNDRV_PCM_INFO_INTERLEAVED |
+		 SNDRV_PCM_INFO_BLOCK_TRANSFER),
+	.formats =          SNDRV_PCM_FMTBIT_S16_BE,
+	.rates =            SNDRV_PCM_RATE_8000_48000,
+	.rate_min =         8000,
+	.rate_max =         48000,
+	.channels_min =     2,
+	.channels_max =     2,
+	.buffer_bytes_max = 65536,
+	.period_bytes_min = 1024,
+	.period_bytes_max = 65536,
+	.periods_min =      2,
+	.periods_max =      1024,
+};
+
+static int hal2_pcm_hw_params(struct snd_pcm_substream *substream,
+			      struct snd_pcm_hw_params *params)
+{
+	int err;
+
+	err = snd_pcm_lib_malloc_pages(substream, params_buffer_bytes(params));
+	if (err < 0)
+		return err;
+
+	return 0;
+}
+
+static int hal2_pcm_hw_free(struct snd_pcm_substream *substream)
+{
+	return snd_pcm_lib_free_pages(substream);
+}
+
+static int hal2_playback_open(struct snd_pcm_substream *substream)
+{
+	struct snd_pcm_runtime *runtime = substream->runtime;
+	struct snd_hal2 *hal2 = snd_pcm_substream_chip(substream);
+	int err;
+
+	runtime->hw = hal2_pcm_hw;
+
+	err = hal2_alloc_dmabuf(&hal2->dac);
+	if (err)
+		return err;
+	return 0;
+}
+
+static int hal2_playback_close(struct snd_pcm_substream *substream)
+{
+	struct snd_hal2 *hal2 = snd_pcm_substream_chip(substream);
+
+	hal2_free_dmabuf(&hal2->dac);
+	return 0;
+}
+
+static int hal2_playback_prepare(struct snd_pcm_substream *substream)
+{
+	struct snd_hal2 *hal2 = snd_pcm_substream_chip(substream);
+	struct snd_pcm_runtime *runtime = substream->runtime;
+	struct hal2_codec *dac = &hal2->dac;
+
+	dac->voices = runtime->channels;
+	dac->sample_rate = hal2_compute_rate(dac, runtime->rate);
+	memset(&dac->pcm_indirect, 0, sizeof(dac->pcm_indirect));
+	dac->pcm_indirect.hw_buffer_size = H2_BUF_SIZE;
+	dac->pcm_indirect.sw_buffer_size = snd_pcm_lib_buffer_bytes(substream);
+	dac->substream = substream;
+	hal2_setup_dac(hal2);
+	return 0;
+}
+
+static int hal2_playback_trigger(struct snd_pcm_substream *substream, int cmd)
+{
+	struct snd_hal2 *hal2 = snd_pcm_substream_chip(substream);
+
+	switch (cmd) {
+	case SNDRV_PCM_TRIGGER_START:
+		hal2->dac.pcm_indirect.hw_io = hal2->dac.buffer_dma;
+		hal2->dac.pcm_indirect.hw_data = 0;
+		substream->ops->ack(substream);
+		hal2_start_dac(hal2);
+		break;
+	case SNDRV_PCM_TRIGGER_STOP:
+		hal2_stop_dac(hal2);
+		break;
+	default:
+		return -EINVAL;
+	}
+	return 0;
+}
+
+static snd_pcm_uframes_t
+hal2_playback_pointer(struct snd_pcm_substream *substream)
+{
+	struct snd_hal2 *hal2 = snd_pcm_substream_chip(substream);
+	struct hal2_codec *dac = &hal2->dac;
+
+	return snd_pcm_indirect_playback_pointer(substream, &dac->pcm_indirect,
+						 dac->pbus.pbus->pbdma_bptr);
+}
+
+static void hal2_playback_transfer(struct snd_pcm_substream *substream,
+				   struct snd_pcm_indirect *rec, size_t bytes)
+{
+	struct snd_hal2 *hal2 = snd_pcm_substream_chip(substream);
+	unsigned char *buf = hal2->dac.buffer + rec->hw_data;
+
+	memcpy(buf, substream->runtime->dma_area + rec->sw_data, bytes);
+	dma_cache_sync(NULL, buf, bytes, DMA_TO_DEVICE);
+
+}
+
+static int hal2_playback_ack(struct snd_pcm_substream *substream)
+{
+	struct snd_hal2 *hal2 = snd_pcm_substream_chip(substream);
+	struct hal2_codec *dac = &hal2->dac;
+
+	dac->pcm_indirect.hw_queue_size = H2_BUF_SIZE / 2;
+	snd_pcm_indirect_playback_transfer(substream,
+					   &dac->pcm_indirect,
+					   hal2_playback_transfer);
+	return 0;
+}
+
+static int hal2_capture_open(struct snd_pcm_substream *substream)
+{
+	struct snd_pcm_runtime *runtime = substream->runtime;
+	struct snd_hal2 *hal2 = snd_pcm_substream_chip(substream);
+	struct hal2_codec *adc = &hal2->adc;
+	int err;
+
+	runtime->hw = hal2_pcm_hw;
+
+	err = hal2_alloc_dmabuf(adc);
+	if (err)
+		return err;
+	return 0;
+}
+
+static int hal2_capture_close(struct snd_pcm_substream *substream)
+{
+	struct snd_hal2 *hal2 = snd_pcm_substream_chip(substream);
+
+	hal2_free_dmabuf(&hal2->adc);
+	return 0;
+}
+
+static int hal2_capture_prepare(struct snd_pcm_substream *substream)
+{
+	struct snd_hal2 *hal2 = snd_pcm_substream_chip(substream);
+	struct snd_pcm_runtime *runtime = substream->runtime;
+	struct hal2_codec *adc = &hal2->adc;
+
+	adc->voices = runtime->channels;
+	adc->sample_rate = hal2_compute_rate(adc, runtime->rate);
+	memset(&adc->pcm_indirect, 0, sizeof(adc->pcm_indirect));
+	adc->pcm_indirect.hw_buffer_size = H2_BUF_SIZE;
+	adc->pcm_indirect.hw_queue_size = H2_BUF_SIZE / 2;
+	adc->pcm_indirect.sw_buffer_size = snd_pcm_lib_buffer_bytes(substream);
+	adc->substream = substream;
+	hal2_setup_adc(hal2);
+	return 0;
+}
+
+static int hal2_capture_trigger(struct snd_pcm_substream *substream, int cmd)
+{
+	struct snd_hal2 *hal2 = snd_pcm_substream_chip(substream);
+
+	switch (cmd) {
+	case SNDRV_PCM_TRIGGER_START:
+		hal2->adc.pcm_indirect.hw_io = hal2->adc.buffer_dma;
+		hal2->adc.pcm_indirect.hw_data = 0;
+		printk(KERN_DEBUG "buffer_dma %x\n", hal2->adc.buffer_dma);
+		hal2_start_adc(hal2);
+		break;
+	case SNDRV_PCM_TRIGGER_STOP:
+		hal2_stop_adc(hal2);
+		break;
+	default:
+		return -EINVAL;
+	}
+	return 0;
+}
+
+static snd_pcm_uframes_t
+hal2_capture_pointer(struct snd_pcm_substream *substream)
+{
+	struct snd_hal2 *hal2 = snd_pcm_substream_chip(substream);
+	struct hal2_codec *adc = &hal2->adc;
+
+	return snd_pcm_indirect_capture_pointer(substream, &adc->pcm_indirect,
+						adc->pbus.pbus->pbdma_bptr);
+}
+
+static void hal2_capture_transfer(struct snd_pcm_substream *substream,
+				  struct snd_pcm_indirect *rec, size_t bytes)
+{
+	struct snd_hal2 *hal2 = snd_pcm_substream_chip(substream);
+	unsigned char *buf = hal2->adc.buffer + rec->hw_data;
+
+	dma_cache_sync(NULL, buf, bytes, DMA_FROM_DEVICE);
+	memcpy(substream->runtime->dma_area + rec->sw_data, buf, bytes);
+}
+
+static int hal2_capture_ack(struct snd_pcm_substream *substream)
+{
+	struct snd_hal2 *hal2 = snd_pcm_substream_chip(substream);
+	struct hal2_codec *adc = &hal2->adc;
+
+	snd_pcm_indirect_capture_transfer(substream,
+					  &adc->pcm_indirect,
+					  hal2_capture_transfer);
+	return 0;
+}
+
+static struct snd_pcm_ops hal2_playback_ops = {
+	.open =        hal2_playback_open,
+	.close =       hal2_playback_close,
+	.ioctl =       snd_pcm_lib_ioctl,
+	.hw_params =   hal2_pcm_hw_params,
+	.hw_free =     hal2_pcm_hw_free,
+	.prepare =     hal2_playback_prepare,
+	.trigger =     hal2_playback_trigger,
+	.pointer =     hal2_playback_pointer,
+	.ack =         hal2_playback_ack,
+};
+
+static struct snd_pcm_ops hal2_capture_ops = {
+	.open =        hal2_capture_open,
+	.close =       hal2_capture_close,
+	.ioctl =       snd_pcm_lib_ioctl,
+	.hw_params =   hal2_pcm_hw_params,
+	.hw_free =     hal2_pcm_hw_free,
+	.prepare =     hal2_capture_prepare,
+	.trigger =     hal2_capture_trigger,
+	.pointer =     hal2_capture_pointer,
+	.ack =         hal2_capture_ack,
+};
+
+static int __devinit hal2_pcm_create(struct snd_hal2 *hal2)
+{
+	struct snd_pcm *pcm;
+	int err;
+
+	/* create first pcm device with one outputs and one input */
+	err = snd_pcm_new(hal2->card, "SGI HAL2 Audio", 0, 1, 1, &pcm);
+	if (err < 0)
+		return err;
+
+	pcm->private_data = hal2;
+	strcpy(pcm->name, "SGI HAL2");
+
+	/* set operators */
+	snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK,
+			&hal2_playback_ops);
+	snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE,
+			&hal2_capture_ops);
+	snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_CONTINUOUS,
+					   snd_dma_continuous_data(GFP_KERNEL),
+					   0, 1024 * 1024);
+
+	return 0;
+}
+
+static int hal2_dev_free(struct snd_device *device)
+{
+	struct snd_hal2 *hal2 = device->device_data;
+
+	free_irq(SGI_HPCDMA_IRQ, hal2);
+	kfree(hal2);
+	return 0;
+}
+
+static struct snd_device_ops hal2_ops = {
+	.dev_free = hal2_dev_free,
+};
+
+static void hal2_init_codec(struct hal2_codec *codec, struct hpc3_regs *hpc3,
+			    int index)
+{
+	codec->pbus.pbusnr = index;
+	codec->pbus.pbus = &hpc3->pbdma[index];
+}
+
+static int hal2_detect(struct snd_hal2 *hal2)
+{
+	unsigned short board, major, minor;
+	unsigned short rev;
+
+	/* reset HAL2 */
+	hal2_write(0, &hal2->ctl_regs->isr);
+
+	/* release reset */
+	hal2_write(H2_ISR_GLOBAL_RESET_N | H2_ISR_CODEC_RESET_N,
+		   &hal2->ctl_regs->isr);
+
+
+	hal2_i_write16(hal2, H2I_RELAY_C, H2I_RELAY_C_STATE);
+	rev = hal2_read(&hal2->ctl_regs->rev);
+	if (rev & H2_REV_AUDIO_PRESENT)
+		return -ENODEV;
+
+	board = (rev & H2_REV_BOARD_M) >> 12;
+	major = (rev & H2_REV_MAJOR_CHIP_M) >> 4;
+	minor = (rev & H2_REV_MINOR_CHIP_M);
+
+	printk(KERN_INFO "SGI HAL2 revision %i.%i.%i\n",
+	       board, major, minor);
+
+	return 0;
+}
+
+static int hal2_create(struct snd_card *card, struct snd_hal2 **rchip)
+{
+	struct snd_hal2 *hal2;
+	struct hpc3_regs *hpc3 = hpc3c0;
+	int err;
+
+	hal2 = kzalloc(sizeof(struct snd_hal2), GFP_KERNEL);
+	if (!hal2)
+		return -ENOMEM;
+
+	hal2->card = card;
+
+	if (request_irq(SGI_HPCDMA_IRQ, hal2_interrupt, IRQF_SHARED,
+			"SGI HAL2", hal2)) {
+		printk(KERN_ERR "HAL2: Can't get irq %d\n", SGI_HPCDMA_IRQ);
+		kfree(hal2);
+		return -EAGAIN;
+	}
+
+	hal2->ctl_regs = (struct hal2_ctl_regs *)hpc3->pbus_extregs[0];
+	hal2->aes_regs = (struct hal2_aes_regs *)hpc3->pbus_extregs[1];
+	hal2->vol_regs = (struct hal2_vol_regs *)hpc3->pbus_extregs[2];
+	hal2->syn_regs = (struct hal2_syn_regs *)hpc3->pbus_extregs[3];
+
+	if (hal2_detect(hal2) < 0) {
+		kfree(hal2);
+		return -ENODEV;
+	}
+
+	hal2_init_codec(&hal2->dac, hpc3, 0);
+	hal2_init_codec(&hal2->adc, hpc3, 1);
+
+	/*
+	 * All DMA channel interfaces in HAL2 are designed to operate with
+	 * PBUS programmed for 2 cycles in D3, 2 cycles in D4 and 2 cycles
+	 * in D5. HAL2 is a 16-bit device which can accept both big and little
+	 * endian format. It assumes that even address bytes are on high
+	 * portion of PBUS (15:8) and assumes that HPC3 is programmed to
+	 * accept a live (unsynchronized) version of P_DREQ_N from HAL2.
+	 */
+#define HAL2_PBUS_DMACFG ((0 << HPC3_DMACFG_D3R_SHIFT) | \
+			  (2 << HPC3_DMACFG_D4R_SHIFT) | \
+			  (2 << HPC3_DMACFG_D5R_SHIFT) | \
+			  (0 << HPC3_DMACFG_D3W_SHIFT) | \
+			  (2 << HPC3_DMACFG_D4W_SHIFT) | \
+			  (2 << HPC3_DMACFG_D5W_SHIFT) | \
+				HPC3_DMACFG_DS16 | \
+				HPC3_DMACFG_EVENHI | \
+				HPC3_DMACFG_RTIME | \
+			  (8 << HPC3_DMACFG_BURST_SHIFT) | \
+				HPC3_DMACFG_DRQLIVE)
+	/*
+	 * Ignore what's mentioned in the specification and write value which
+	 * works in The Real World (TM)
+	 */
+	hpc3->pbus_dmacfg[hal2->dac.pbus.pbusnr][0] = 0x8208844;
+	hpc3->pbus_dmacfg[hal2->adc.pbus.pbusnr][0] = 0x8208844;
+
+	err = snd_device_new(card, SNDRV_DEV_LOWLEVEL, hal2, &hal2_ops);
+	if (err < 0) {
+		free_irq(SGI_HPCDMA_IRQ, hal2);
+		kfree(hal2);
+		return err;
+	}
+	*rchip = hal2;
+	return 0;
+}
+
+static int __devinit hal2_probe(struct platform_device *pdev)
+{
+	struct snd_card *card;
+	struct snd_hal2 *chip;
+	int err;
+
+	err = snd_card_create(index, id, THIS_MODULE, 0, &card);
+	if (err < 0)
+		return err;
+
+	err = hal2_create(card, &chip);
+	if (err < 0) {
+		snd_card_free(card);
+		return err;
+	}
+	snd_card_set_dev(card, &pdev->dev);
+
+	err = hal2_pcm_create(chip);
+	if (err < 0) {
+		snd_card_free(card);
+		return err;
+	}
+	err = hal2_mixer_create(chip);
+	if (err < 0) {
+		snd_card_free(card);
+		return err;
+	}
+
+	strcpy(card->driver, "SGI HAL2 Audio");
+	strcpy(card->shortname, "SGI HAL2 Audio");
+	sprintf(card->longname, "%s irq %i",
+		card->shortname,
+		SGI_HPCDMA_IRQ);
+
+	err = snd_card_register(card);
+	if (err < 0) {
+		snd_card_free(card);
+		return err;
+	}
+	platform_set_drvdata(pdev, card);
+	return 0;
+}
+
+static int __devexit hal2_remove(struct platform_device *pdev)
+{
+	struct snd_card *card = platform_get_drvdata(pdev);
+
+	snd_card_free(card);
+	platform_set_drvdata(pdev, NULL);
+	return 0;
+}
+
+static struct platform_driver hal2_driver = {
+	.probe	= hal2_probe,
+	.remove	= __devexit_p(hal2_remove),
+	.driver = {
+		.name	= "sgihal2",
+		.owner	= THIS_MODULE,
+	}
+};
+
+module_platform_driver(hal2_driver);
diff --git a/ANDROID_3.4.5/sound/mips/hal2.h b/ANDROID_3.4.5/sound/mips/hal2.h
new file mode 100644
index 00000000..f19828bc
--- /dev/null
+++ b/ANDROID_3.4.5/sound/mips/hal2.h
@@ -0,0 +1,245 @@
+#ifndef __HAL2_H
+#define __HAL2_H
+
+/*
+ *  Driver for HAL2 sound processors
+ *  Copyright (c) 1999 Ulf Carlsson <ulfc@bun.falkenberg.se>
+ *  Copyright (c) 2001, 2002, 2003 Ladislav Michl <ladis@linux-mips.org>
+ *
+ *  This program is free software; you can redistribute it and/or modify
+ *  it under the terms of the GNU General Public License version 2 as
+ *  published by the Free Software Foundation.
+ *
+ *  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.
+ *
+ */
+
+#include <linux/types.h>
+
+/* Indirect status register */
+
+#define H2_ISR_TSTATUS		0x01	/* RO: transaction status 1=busy */
+#define H2_ISR_USTATUS		0x02	/* RO: utime status bit 1=armed */
+#define H2_ISR_QUAD_MODE	0x04	/* codec mode 0=indigo 1=quad */
+#define H2_ISR_GLOBAL_RESET_N	0x08	/* chip global reset 0=reset */
+#define H2_ISR_CODEC_RESET_N	0x10	/* codec/synth reset 0=reset  */
+
+/* Revision register */
+
+#define H2_REV_AUDIO_PRESENT	0x8000	/* RO: audio present 0=present */
+#define H2_REV_BOARD_M		0x7000	/* RO: bits 14:12, board revision */
+#define H2_REV_MAJOR_CHIP_M	0x00F0	/* RO: bits 7:4, major chip revision */
+#define H2_REV_MINOR_CHIP_M	0x000F	/* RO: bits 3:0, minor chip revision */
+
+/* Indirect address register */
+
+/*
+ * Address of indirect internal register to be accessed. A write to this
+ * register initiates read or write access to the indirect registers in the
+ * HAL2. Note that there af four indirect data registers for write access to
+ * registers larger than 16 byte.
+ */
+
+#define H2_IAR_TYPE_M		0xF000	/* bits 15:12, type of functional */
+					/* block the register resides in */
+					/* 1=DMA Port */
+					/* 9=Global DMA Control */
+					/* 2=Bresenham */
+					/* 3=Unix Timer */
+#define H2_IAR_NUM_M		0x0F00	/* bits 11:8 instance of the */
+					/* blockin which the indirect */
+					/* register resides */
+					/* If IAR_TYPE_M=DMA Port: */
+					/* 1=Synth In */
+					/* 2=AES In */
+					/* 3=AES Out */
+					/* 4=DAC Out */
+					/* 5=ADC Out */
+					/* 6=Synth Control */
+					/* If IAR_TYPE_M=Global DMA Control: */
+					/* 1=Control */
+					/* If IAR_TYPE_M=Bresenham: */
+					/* 1=Bresenham Clock Gen 1 */
+					/* 2=Bresenham Clock Gen 2 */
+					/* 3=Bresenham Clock Gen 3 */
+					/* If IAR_TYPE_M=Unix Timer: */
+					/* 1=Unix Timer */
+#define H2_IAR_ACCESS_SELECT	0x0080	/* 1=read 0=write */
+#define H2_IAR_PARAM		0x000C	/* Parameter Select */
+#define H2_IAR_RB_INDEX_M	0x0003	/* Read Back Index */
+					/* 00:word0 */
+					/* 01:word1 */
+					/* 10:word2 */
+					/* 11:word3 */
+/*
+ * HAL2 internal addressing
+ *
+ * The HAL2 has "indirect registers" (idr) which are accessed by writing to the
+ * Indirect Data registers. Write the address to the Indirect Address register
+ * to transfer the data.
+ *
+ * We define the H2IR_* to the read address and H2IW_* to the write address and
+ * H2I_* to be fields in whatever register is referred to.
+ *
+ * When we write to indirect registers which are larger than one word (16 bit)
+ * we have to fill more than one indirect register before writing. When we read
+ * back however we have to read several times, each time with different Read
+ * Back Indexes (there are defs for doing this easily).
+ */
+
+/*
+ * Relay Control
+ */
+#define H2I_RELAY_C		0x9100
+#define H2I_RELAY_C_STATE	0x01		/* state of RELAY pin signal */
+
+/* DMA port enable */
+
+#define H2I_DMA_PORT_EN		0x9104
+#define H2I_DMA_PORT_EN_SY_IN	0x01		/* Synth_in DMA port */
+#define H2I_DMA_PORT_EN_AESRX	0x02		/* AES receiver DMA port */
+#define H2I_DMA_PORT_EN_AESTX	0x04		/* AES transmitter DMA port */
+#define H2I_DMA_PORT_EN_CODECTX	0x08		/* CODEC transmit DMA port */
+#define H2I_DMA_PORT_EN_CODECR	0x10		/* CODEC receive DMA port */
+
+#define H2I_DMA_END		0x9108 		/* global dma endian select */
+#define H2I_DMA_END_SY_IN	0x01		/* Synth_in DMA port */
+#define H2I_DMA_END_AESRX	0x02		/* AES receiver DMA port */
+#define H2I_DMA_END_AESTX	0x04		/* AES transmitter DMA port */
+#define H2I_DMA_END_CODECTX	0x08		/* CODEC transmit DMA port */
+#define H2I_DMA_END_CODECR	0x10		/* CODEC receive DMA port */
+						/* 0=b_end 1=l_end */
+
+#define H2I_DMA_DRV		0x910C  	/* global PBUS DMA enable */
+
+#define H2I_SYNTH_C		0x1104		/* Synth DMA control */
+
+#define H2I_AESRX_C		0x1204	 	/* AES RX dma control */
+
+#define H2I_C_TS_EN		0x20		/* Timestamp enable */
+#define H2I_C_TS_FRMT		0x40		/* Timestamp format */
+#define H2I_C_NAUDIO		0x80		/* Sign extend */
+
+/* AESRX CTL, 16 bit */
+
+#define H2I_AESTX_C		0x1304		/* AES TX DMA control */
+#define H2I_AESTX_C_CLKID_SHIFT	3		/* Bresenham Clock Gen 1-3 */
+#define H2I_AESTX_C_CLKID_M	0x18
+#define H2I_AESTX_C_DATAT_SHIFT	8		/* 1=mono 2=stereo (3=quad) */
+#define H2I_AESTX_C_DATAT_M	0x300
+
+/* CODEC registers */
+
+#define H2I_DAC_C1		0x1404 		/* DAC DMA control, 16 bit */
+#define H2I_DAC_C2		0x1408		/* DAC DMA control, 32 bit */
+#define H2I_ADC_C1		0x1504 		/* ADC DMA control, 16 bit */
+#define H2I_ADC_C2		0x1508		/* ADC DMA control, 32 bit */
+
+/* Bits in CTL1 register */
+
+#define H2I_C1_DMA_SHIFT	0		/* DMA channel */
+#define H2I_C1_DMA_M		0x7
+#define H2I_C1_CLKID_SHIFT	3		/* Bresenham Clock Gen 1-3 */
+#define H2I_C1_CLKID_M		0x18
+#define H2I_C1_DATAT_SHIFT	8		/* 1=mono 2=stereo (3=quad) */
+#define H2I_C1_DATAT_M		0x300
+
+/* Bits in CTL2 register */
+
+#define H2I_C2_R_GAIN_SHIFT	0		/* right a/d input gain */
+#define H2I_C2_R_GAIN_M		0xf
+#define H2I_C2_L_GAIN_SHIFT	4		/* left a/d input gain */
+#define H2I_C2_L_GAIN_M		0xf0
+#define H2I_C2_R_SEL		0x100		/* right input select */
+#define H2I_C2_L_SEL		0x200		/* left input select */
+#define H2I_C2_MUTE		0x400		/* mute */
+#define H2I_C2_DO1		0x00010000	/* digital output port bit 0 */
+#define H2I_C2_DO2		0x00020000	/* digital output port bit 1 */
+#define H2I_C2_R_ATT_SHIFT	18		/* right d/a output - */
+#define H2I_C2_R_ATT_M		0x007c0000	/* attenuation */
+#define H2I_C2_L_ATT_SHIFT	23		/* left d/a output - */
+#define H2I_C2_L_ATT_M		0x0f800000	/* attenuation */
+
+#define H2I_SYNTH_MAP_C		0x1104		/* synth dma handshake ctrl */
+
+/* Clock generator CTL 1, 16 bit */
+
+#define H2I_BRES1_C1		0x2104
+#define H2I_BRES2_C1		0x2204
+#define H2I_BRES3_C1		0x2304
+
+#define H2I_BRES_C1_SHIFT	0		/* 0=48.0 1=44.1 2=aes_rx */
+#define H2I_BRES_C1_M		0x03
+
+/* Clock generator CTL 2, 32 bit */
+
+#define H2I_BRES1_C2		0x2108
+#define H2I_BRES2_C2		0x2208
+#define H2I_BRES3_C2		0x2308
+
+#define H2I_BRES_C2_INC_SHIFT	0		/* increment value */
+#define H2I_BRES_C2_INC_M	0xffff
+#define H2I_BRES_C2_MOD_SHIFT	16		/* modcontrol value */
+#define H2I_BRES_C2_MOD_M	0xffff0000	/* modctrl=0xffff&(modinc-1) */
+
+/* Unix timer, 64 bit */
+
+#define H2I_UTIME		0x3104
+#define H2I_UTIME_0_LD		0xffff		/* microseconds, LSB's */
+#define H2I_UTIME_1_LD0		0x0f		/* microseconds, MSB's */
+#define H2I_UTIME_1_LD1		0xf0		/* tenths of microseconds */
+#define H2I_UTIME_2_LD		0xffff		/* seconds, LSB's */
+#define H2I_UTIME_3_LD		0xffff		/* seconds, MSB's */
+
+struct hal2_ctl_regs {
+	u32 _unused0[4];
+	u32 isr;		/* 0x10 Status Register */
+	u32 _unused1[3];
+	u32 rev;		/* 0x20 Revision Register */
+	u32 _unused2[3];
+	u32 iar;		/* 0x30 Indirect Address Register */
+	u32 _unused3[3];
+	u32 idr0;		/* 0x40 Indirect Data Register 0 */
+	u32 _unused4[3];
+	u32 idr1;		/* 0x50 Indirect Data Register 1 */
+	u32 _unused5[3];
+	u32 idr2;		/* 0x60 Indirect Data Register 2 */
+	u32 _unused6[3];
+	u32 idr3;		/* 0x70 Indirect Data Register 3 */
+};
+
+struct hal2_aes_regs {
+	u32 rx_stat[2];	/* Status registers */
+	u32 rx_cr[2];		/* Control registers */
+	u32 rx_ud[4];		/* User data window */
+	u32 rx_st[24];		/* Channel status data */
+
+	u32 tx_stat[1];	/* Status register */
+	u32 tx_cr[3];		/* Control registers */
+	u32 tx_ud[4];		/* User data window */
+	u32 tx_st[24];		/* Channel status data */
+};
+
+struct hal2_vol_regs {
+	u32 right;		/* Right volume */
+	u32 left;		/* Left volume */
+};
+
+struct hal2_syn_regs {
+	u32 _unused0[2];
+	u32 page;		/* DOC Page register */
+	u32 regsel;		/* DOC Register selection */
+	u32 dlow;		/* DOC Data low */
+	u32 dhigh;		/* DOC Data high */
+	u32 irq;		/* IRQ Status */
+	u32 dram;		/* DRAM Access */
+};
+
+#endif	/* __HAL2_H */
diff --git a/ANDROID_3.4.5/sound/mips/sgio2audio.c b/ANDROID_3.4.5/sound/mips/sgio2audio.c
new file mode 100644
index 00000000..ceaa593e
--- /dev/null
+++ b/ANDROID_3.4.5/sound/mips/sgio2audio.c
@@ -0,0 +1,979 @@
+/*
+ *   Sound driver for Silicon Graphics O2 Workstations A/V board audio.
+ *
+ *   Copyright 2003 Vivien Chappelier <vivien.chappelier@linux-mips.org>
+ *   Copyright 2008 Thomas Bogendoerfer <tsbogend@alpha.franken.de>
+ *   Mxier part taken from mace_audio.c:
+ *   Copyright 2007 Thorben Jändling <tj.trevelyan@gmail.com>
+ *
+ *   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., 59 Temple Place, Suite 330, Boston, MA  02111-1307 USA
+ *
+ */
+
+#include <linux/init.h>
+#include <linux/delay.h>
+#include <linux/spinlock.h>
+#include <linux/interrupt.h>
+#include <linux/dma-mapping.h>
+#include <linux/platform_device.h>
+#include <linux/io.h>
+#include <linux/slab.h>
+#include <linux/module.h>
+
+#include <asm/ip32/ip32_ints.h>
+#include <asm/ip32/mace.h>
+
+#include <sound/core.h>
+#include <sound/control.h>
+#include <sound/pcm.h>
+#define SNDRV_GET_ID
+#include <sound/initval.h>
+#include <sound/ad1843.h>
+
+
+MODULE_AUTHOR("Vivien Chappelier <vivien.chappelier@linux-mips.org>");
+MODULE_DESCRIPTION("SGI O2 Audio");
+MODULE_LICENSE("GPL");
+MODULE_SUPPORTED_DEVICE("{{Silicon Graphics, O2 Audio}}");
+
+static int index = SNDRV_DEFAULT_IDX1;  /* Index 0-MAX */
+static char *id = SNDRV_DEFAULT_STR1;   /* ID for this card */
+
+module_param(index, int, 0444);
+MODULE_PARM_DESC(index, "Index value for SGI O2 soundcard.");
+module_param(id, charp, 0444);
+MODULE_PARM_DESC(id, "ID string for SGI O2 soundcard.");
+
+
+#define AUDIO_CONTROL_RESET              BIT(0) /* 1: reset audio interface */
+#define AUDIO_CONTROL_CODEC_PRESENT      BIT(1) /* 1: codec detected */
+
+#define CODEC_CONTROL_WORD_SHIFT        0
+#define CODEC_CONTROL_READ              BIT(16)
+#define CODEC_CONTROL_ADDRESS_SHIFT     17
+
+#define CHANNEL_CONTROL_RESET           BIT(10) /* 1: reset channel */
+#define CHANNEL_DMA_ENABLE              BIT(9)  /* 1: enable DMA transfer */
+#define CHANNEL_INT_THRESHOLD_DISABLED  (0 << 5) /* interrupt disabled */
+#define CHANNEL_INT_THRESHOLD_25        (1 << 5) /* int on buffer >25% full */
+#define CHANNEL_INT_THRESHOLD_50        (2 << 5) /* int on buffer >50% full */
+#define CHANNEL_INT_THRESHOLD_75        (3 << 5) /* int on buffer >75% full */
+#define CHANNEL_INT_THRESHOLD_EMPTY     (4 << 5) /* int on buffer empty */
+#define CHANNEL_INT_THRESHOLD_NOT_EMPTY (5 << 5) /* int on buffer !empty */
+#define CHANNEL_INT_THRESHOLD_FULL      (6 << 5) /* int on buffer empty */
+#define CHANNEL_INT_THRESHOLD_NOT_FULL  (7 << 5) /* int on buffer !empty */
+
+#define CHANNEL_RING_SHIFT              12
+#define CHANNEL_RING_SIZE               (1 << CHANNEL_RING_SHIFT)
+#define CHANNEL_RING_MASK               (CHANNEL_RING_SIZE - 1)
+
+#define CHANNEL_LEFT_SHIFT 40
+#define CHANNEL_RIGHT_SHIFT 8
+
+struct snd_sgio2audio_chan {
+	int idx;
+	struct snd_pcm_substream *substream;
+	int pos;
+	snd_pcm_uframes_t size;
+	spinlock_t lock;
+};
+
+/* definition of the chip-specific record */
+struct snd_sgio2audio {
+	struct snd_card *card;
+
+	/* codec */
+	struct snd_ad1843 ad1843;
+	spinlock_t ad1843_lock;
+
+	/* channels */
+	struct snd_sgio2audio_chan channel[3];
+
+	/* resources */
+	void *ring_base;
+	dma_addr_t ring_base_dma;
+};
+
+/* AD1843 access */
+
+/*
+ * read_ad1843_reg returns the current contents of a 16 bit AD1843 register.
+ *
+ * Returns unsigned register value on success, -errno on failure.
+ */
+static int read_ad1843_reg(void *priv, int reg)
+{
+	struct snd_sgio2audio *chip = priv;
+	int val;
+	unsigned long flags;
+
+	spin_lock_irqsave(&chip->ad1843_lock, flags);
+
+	writeq((reg << CODEC_CONTROL_ADDRESS_SHIFT) |
+	       CODEC_CONTROL_READ, &mace->perif.audio.codec_control);
+	wmb();
+	val = readq(&mace->perif.audio.codec_control); /* flush bus */
+	udelay(200);
+
+	val = readq(&mace->perif.audio.codec_read);
+
+	spin_unlock_irqrestore(&chip->ad1843_lock, flags);
+	return val;
+}
+
+/*
+ * write_ad1843_reg writes the specified value to a 16 bit AD1843 register.
+ */
+static int write_ad1843_reg(void *priv, int reg, int word)
+{
+	struct snd_sgio2audio *chip = priv;
+	int val;
+	unsigned long flags;
+
+	spin_lock_irqsave(&chip->ad1843_lock, flags);
+
+	writeq((reg << CODEC_CONTROL_ADDRESS_SHIFT) |
+	       (word << CODEC_CONTROL_WORD_SHIFT),
+	       &mace->perif.audio.codec_control);
+	wmb();
+	val = readq(&mace->perif.audio.codec_control); /* flush bus */
+	udelay(200);
+
+	spin_unlock_irqrestore(&chip->ad1843_lock, flags);
+	return 0;
+}
+
+static int sgio2audio_gain_info(struct snd_kcontrol *kcontrol,
+			       struct snd_ctl_elem_info *uinfo)
+{
+	struct snd_sgio2audio *chip = snd_kcontrol_chip(kcontrol);
+
+	uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
+	uinfo->count = 2;
+	uinfo->value.integer.min = 0;
+	uinfo->value.integer.max = ad1843_get_gain_max(&chip->ad1843,
+					     (int)kcontrol->private_value);
+	return 0;
+}
+
+static int sgio2audio_gain_get(struct snd_kcontrol *kcontrol,
+			       struct snd_ctl_elem_value *ucontrol)
+{
+	struct snd_sgio2audio *chip = snd_kcontrol_chip(kcontrol);
+	int vol;
+
+	vol = ad1843_get_gain(&chip->ad1843, (int)kcontrol->private_value);
+
+	ucontrol->value.integer.value[0] = (vol >> 8) & 0xFF;
+	ucontrol->value.integer.value[1] = vol & 0xFF;
+
+	return 0;
+}
+
+static int sgio2audio_gain_put(struct snd_kcontrol *kcontrol,
+			struct snd_ctl_elem_value *ucontrol)
+{
+	struct snd_sgio2audio *chip = snd_kcontrol_chip(kcontrol);
+	int newvol, oldvol;
+
+	oldvol = ad1843_get_gain(&chip->ad1843, kcontrol->private_value);
+	newvol = (ucontrol->value.integer.value[0] << 8) |
+		ucontrol->value.integer.value[1];
+
+	newvol = ad1843_set_gain(&chip->ad1843, kcontrol->private_value,
+		newvol);
+
+	return newvol != oldvol;
+}
+
+static int sgio2audio_source_info(struct snd_kcontrol *kcontrol,
+			       struct snd_ctl_elem_info *uinfo)
+{
+	static const char *texts[3] = {
+		"Cam Mic", "Mic", "Line"
+	};
+	uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
+	uinfo->count = 1;
+	uinfo->value.enumerated.items = 3;
+	if (uinfo->value.enumerated.item >= 3)
+		uinfo->value.enumerated.item = 1;
+	strcpy(uinfo->value.enumerated.name,
+	       texts[uinfo->value.enumerated.item]);
+	return 0;
+}
+
+static int sgio2audio_source_get(struct snd_kcontrol *kcontrol,
+			       struct snd_ctl_elem_value *ucontrol)
+{
+	struct snd_sgio2audio *chip = snd_kcontrol_chip(kcontrol);
+
+	ucontrol->value.enumerated.item[0] = ad1843_get_recsrc(&chip->ad1843);
+	return 0;
+}
+
+static int sgio2audio_source_put(struct snd_kcontrol *kcontrol,
+			struct snd_ctl_elem_value *ucontrol)
+{
+	struct snd_sgio2audio *chip = snd_kcontrol_chip(kcontrol);
+	int newsrc, oldsrc;
+
+	oldsrc = ad1843_get_recsrc(&chip->ad1843);
+	newsrc = ad1843_set_recsrc(&chip->ad1843,
+				   ucontrol->value.enumerated.item[0]);
+
+	return newsrc != oldsrc;
+}
+
+/* dac1/pcm0 mixer control */
+static struct snd_kcontrol_new sgio2audio_ctrl_pcm0 __devinitdata = {
+	.iface          = SNDRV_CTL_ELEM_IFACE_MIXER,
+	.name           = "PCM Playback Volume",
+	.index          = 0,
+	.access         = SNDRV_CTL_ELEM_ACCESS_READWRITE,
+	.private_value  = AD1843_GAIN_PCM_0,
+	.info           = sgio2audio_gain_info,
+	.get            = sgio2audio_gain_get,
+	.put            = sgio2audio_gain_put,
+};
+
+/* dac2/pcm1 mixer control */
+static struct snd_kcontrol_new sgio2audio_ctrl_pcm1 __devinitdata = {
+	.iface          = SNDRV_CTL_ELEM_IFACE_MIXER,
+	.name           = "PCM Playback Volume",
+	.index          = 1,
+	.access         = SNDRV_CTL_ELEM_ACCESS_READWRITE,
+	.private_value  = AD1843_GAIN_PCM_1,
+	.info           = sgio2audio_gain_info,
+	.get            = sgio2audio_gain_get,
+	.put            = sgio2audio_gain_put,
+};
+
+/* record level mixer control */
+static struct snd_kcontrol_new sgio2audio_ctrl_reclevel __devinitdata = {
+	.iface          = SNDRV_CTL_ELEM_IFACE_MIXER,
+	.name           = "Capture Volume",
+	.access         = SNDRV_CTL_ELEM_ACCESS_READWRITE,
+	.private_value  = AD1843_GAIN_RECLEV,
+	.info           = sgio2audio_gain_info,
+	.get            = sgio2audio_gain_get,
+	.put            = sgio2audio_gain_put,
+};
+
+/* record level source control */
+static struct snd_kcontrol_new sgio2audio_ctrl_recsource __devinitdata = {
+	.iface          = SNDRV_CTL_ELEM_IFACE_MIXER,
+	.name           = "Capture Source",
+	.access         = SNDRV_CTL_ELEM_ACCESS_READWRITE,
+	.info           = sgio2audio_source_info,
+	.get            = sgio2audio_source_get,
+	.put            = sgio2audio_source_put,
+};
+
+/* line mixer control */
+static struct snd_kcontrol_new sgio2audio_ctrl_line __devinitdata = {
+	.iface          = SNDRV_CTL_ELEM_IFACE_MIXER,
+	.name           = "Line Playback Volume",
+	.index          = 0,
+	.access         = SNDRV_CTL_ELEM_ACCESS_READWRITE,
+	.private_value  = AD1843_GAIN_LINE,
+	.info           = sgio2audio_gain_info,
+	.get            = sgio2audio_gain_get,
+	.put            = sgio2audio_gain_put,
+};
+
+/* cd mixer control */
+static struct snd_kcontrol_new sgio2audio_ctrl_cd __devinitdata = {
+	.iface          = SNDRV_CTL_ELEM_IFACE_MIXER,
+	.name           = "Line Playback Volume",
+	.index          = 1,
+	.access         = SNDRV_CTL_ELEM_ACCESS_READWRITE,
+	.private_value  = AD1843_GAIN_LINE_2,
+	.info           = sgio2audio_gain_info,
+	.get            = sgio2audio_gain_get,
+	.put            = sgio2audio_gain_put,
+};
+
+/* mic mixer control */
+static struct snd_kcontrol_new sgio2audio_ctrl_mic __devinitdata = {
+	.iface          = SNDRV_CTL_ELEM_IFACE_MIXER,
+	.name           = "Mic Playback Volume",
+	.access         = SNDRV_CTL_ELEM_ACCESS_READWRITE,
+	.private_value  = AD1843_GAIN_MIC,
+	.info           = sgio2audio_gain_info,
+	.get            = sgio2audio_gain_get,
+	.put            = sgio2audio_gain_put,
+};
+
+
+static int __devinit snd_sgio2audio_new_mixer(struct snd_sgio2audio *chip)
+{
+	int err;
+
+	err = snd_ctl_add(chip->card,
+			  snd_ctl_new1(&sgio2audio_ctrl_pcm0, chip));
+	if (err < 0)
+		return err;
+
+	err = snd_ctl_add(chip->card,
+			  snd_ctl_new1(&sgio2audio_ctrl_pcm1, chip));
+	if (err < 0)
+		return err;
+
+	err = snd_ctl_add(chip->card,
+			  snd_ctl_new1(&sgio2audio_ctrl_reclevel, chip));
+	if (err < 0)
+		return err;
+
+	err = snd_ctl_add(chip->card,
+			  snd_ctl_new1(&sgio2audio_ctrl_recsource, chip));
+	if (err < 0)
+		return err;
+	err = snd_ctl_add(chip->card,
+			  snd_ctl_new1(&sgio2audio_ctrl_line, chip));
+	if (err < 0)
+		return err;
+
+	err = snd_ctl_add(chip->card,
+			  snd_ctl_new1(&sgio2audio_ctrl_cd, chip));
+	if (err < 0)
+		return err;
+
+	err = snd_ctl_add(chip->card,
+			  snd_ctl_new1(&sgio2audio_ctrl_mic, chip));
+	if (err < 0)
+		return err;
+
+	return 0;
+}
+
+/* low-level audio interface DMA */
+
+/* get data out of bounce buffer, count must be a multiple of 32 */
+/* returns 1 if a period has elapsed */
+static int snd_sgio2audio_dma_pull_frag(struct snd_sgio2audio *chip,
+					unsigned int ch, unsigned int count)
+{
+	int ret;
+	unsigned long src_base, src_pos, dst_mask;
+	unsigned char *dst_base;
+	int dst_pos;
+	u64 *src;
+	s16 *dst;
+	u64 x;
+	unsigned long flags;
+	struct snd_pcm_runtime *runtime = chip->channel[ch].substream->runtime;
+
+	spin_lock_irqsave(&chip->channel[ch].lock, flags);
+
+	src_base = (unsigned long) chip->ring_base | (ch << CHANNEL_RING_SHIFT);
+	src_pos = readq(&mace->perif.audio.chan[ch].read_ptr);
+	dst_base = runtime->dma_area;
+	dst_pos = chip->channel[ch].pos;
+	dst_mask = frames_to_bytes(runtime, runtime->buffer_size) - 1;
+
+	/* check if a period has elapsed */
+	chip->channel[ch].size += (count >> 3); /* in frames */
+	ret = chip->channel[ch].size >= runtime->period_size;
+	chip->channel[ch].size %= runtime->period_size;
+
+	while (count) {
+		src = (u64 *)(src_base + src_pos);
+		dst = (s16 *)(dst_base + dst_pos);
+
+		x = *src;
+		dst[0] = (x >> CHANNEL_LEFT_SHIFT) & 0xffff;
+		dst[1] = (x >> CHANNEL_RIGHT_SHIFT) & 0xffff;
+
+		src_pos = (src_pos + sizeof(u64)) & CHANNEL_RING_MASK;
+		dst_pos = (dst_pos + 2 * sizeof(s16)) & dst_mask;
+		count -= sizeof(u64);
+	}
+
+	writeq(src_pos, &mace->perif.audio.chan[ch].read_ptr); /* in bytes */
+	chip->channel[ch].pos = dst_pos;
+
+	spin_unlock_irqrestore(&chip->channel[ch].lock, flags);
+	return ret;
+}
+
+/* put some DMA data in bounce buffer, count must be a multiple of 32 */
+/* returns 1 if a period has elapsed */
+static int snd_sgio2audio_dma_push_frag(struct snd_sgio2audio *chip,
+					unsigned int ch, unsigned int count)
+{
+	int ret;
+	s64 l, r;
+	unsigned long dst_base, dst_pos, src_mask;
+	unsigned char *src_base;
+	int src_pos;
+	u64 *dst;
+	s16 *src;
+	unsigned long flags;
+	struct snd_pcm_runtime *runtime = chip->channel[ch].substream->runtime;
+
+	spin_lock_irqsave(&chip->channel[ch].lock, flags);
+
+	dst_base = (unsigned long)chip->ring_base | (ch << CHANNEL_RING_SHIFT);
+	dst_pos = readq(&mace->perif.audio.chan[ch].write_ptr);
+	src_base = runtime->dma_area;
+	src_pos = chip->channel[ch].pos;
+	src_mask = frames_to_bytes(runtime, runtime->buffer_size) - 1;
+
+	/* check if a period has elapsed */
+	chip->channel[ch].size += (count >> 3); /* in frames */
+	ret = chip->channel[ch].size >= runtime->period_size;
+	chip->channel[ch].size %= runtime->period_size;
+
+	while (count) {
+		src = (s16 *)(src_base + src_pos);
+		dst = (u64 *)(dst_base + dst_pos);
+
+		l = src[0]; /* sign extend */
+		r = src[1]; /* sign extend */
+
+		*dst = ((l & 0x00ffffff) << CHANNEL_LEFT_SHIFT) |
+			((r & 0x00ffffff) << CHANNEL_RIGHT_SHIFT);
+
+		dst_pos = (dst_pos + sizeof(u64)) & CHANNEL_RING_MASK;
+		src_pos = (src_pos + 2 * sizeof(s16)) & src_mask;
+		count -= sizeof(u64);
+	}
+
+	writeq(dst_pos, &mace->perif.audio.chan[ch].write_ptr); /* in bytes */
+	chip->channel[ch].pos = src_pos;
+
+	spin_unlock_irqrestore(&chip->channel[ch].lock, flags);
+	return ret;
+}
+
+static int snd_sgio2audio_dma_start(struct snd_pcm_substream *substream)
+{
+	struct snd_sgio2audio *chip = snd_pcm_substream_chip(substream);
+	struct snd_sgio2audio_chan *chan = substream->runtime->private_data;
+	int ch = chan->idx;
+
+	/* reset DMA channel */
+	writeq(CHANNEL_CONTROL_RESET, &mace->perif.audio.chan[ch].control);
+	udelay(10);
+	writeq(0, &mace->perif.audio.chan[ch].control);
+
+	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+		/* push a full buffer */
+		snd_sgio2audio_dma_push_frag(chip, ch, CHANNEL_RING_SIZE - 32);
+	}
+	/* set DMA to wake on 50% empty and enable interrupt */
+	writeq(CHANNEL_DMA_ENABLE | CHANNEL_INT_THRESHOLD_50,
+	       &mace->perif.audio.chan[ch].control);
+	return 0;
+}
+
+static int snd_sgio2audio_dma_stop(struct snd_pcm_substream *substream)
+{
+	struct snd_sgio2audio_chan *chan = substream->runtime->private_data;
+
+	writeq(0, &mace->perif.audio.chan[chan->idx].control);
+	return 0;
+}
+
+static irqreturn_t snd_sgio2audio_dma_in_isr(int irq, void *dev_id)
+{
+	struct snd_sgio2audio_chan *chan = dev_id;
+	struct snd_pcm_substream *substream;
+	struct snd_sgio2audio *chip;
+	int count, ch;
+
+	substream = chan->substream;
+	chip = snd_pcm_substream_chip(substream);
+	ch = chan->idx;
+
+	/* empty the ring */
+	count = CHANNEL_RING_SIZE -
+		readq(&mace->perif.audio.chan[ch].depth) - 32;
+	if (snd_sgio2audio_dma_pull_frag(chip, ch, count))
+		snd_pcm_period_elapsed(substream);
+
+	return IRQ_HANDLED;
+}
+
+static irqreturn_t snd_sgio2audio_dma_out_isr(int irq, void *dev_id)
+{
+	struct snd_sgio2audio_chan *chan = dev_id;
+	struct snd_pcm_substream *substream;
+	struct snd_sgio2audio *chip;
+	int count, ch;
+
+	substream = chan->substream;
+	chip = snd_pcm_substream_chip(substream);
+	ch = chan->idx;
+	/* fill the ring */
+	count = CHANNEL_RING_SIZE -
+		readq(&mace->perif.audio.chan[ch].depth) - 32;
+	if (snd_sgio2audio_dma_push_frag(chip, ch, count))
+		snd_pcm_period_elapsed(substream);
+
+	return IRQ_HANDLED;
+}
+
+static irqreturn_t snd_sgio2audio_error_isr(int irq, void *dev_id)
+{
+	struct snd_sgio2audio_chan *chan = dev_id;
+	struct snd_pcm_substream *substream;
+
+	substream = chan->substream;
+	snd_sgio2audio_dma_stop(substream);
+	snd_sgio2audio_dma_start(substream);
+	return IRQ_HANDLED;
+}
+
+/* PCM part */
+/* PCM hardware definition */
+static struct snd_pcm_hardware snd_sgio2audio_pcm_hw = {
+	.info = (SNDRV_PCM_INFO_MMAP |
+		 SNDRV_PCM_INFO_MMAP_VALID |
+		 SNDRV_PCM_INFO_INTERLEAVED |
+		 SNDRV_PCM_INFO_BLOCK_TRANSFER),
+	.formats =          SNDRV_PCM_FMTBIT_S16_BE,
+	.rates =            SNDRV_PCM_RATE_8000_48000,
+	.rate_min =         8000,
+	.rate_max =         48000,
+	.channels_min =     2,
+	.channels_max =     2,
+	.buffer_bytes_max = 65536,
+	.period_bytes_min = 32768,
+	.period_bytes_max = 65536,
+	.periods_min =      1,
+	.periods_max =      1024,
+};
+
+/* PCM playback open callback */
+static int snd_sgio2audio_playback1_open(struct snd_pcm_substream *substream)
+{
+	struct snd_sgio2audio *chip = snd_pcm_substream_chip(substream);
+	struct snd_pcm_runtime *runtime = substream->runtime;
+
+	runtime->hw = snd_sgio2audio_pcm_hw;
+	runtime->private_data = &chip->channel[1];
+	return 0;
+}
+
+static int snd_sgio2audio_playback2_open(struct snd_pcm_substream *substream)
+{
+	struct snd_sgio2audio *chip = snd_pcm_substream_chip(substream);
+	struct snd_pcm_runtime *runtime = substream->runtime;
+
+	runtime->hw = snd_sgio2audio_pcm_hw;
+	runtime->private_data = &chip->channel[2];
+	return 0;
+}
+
+/* PCM capture open callback */
+static int snd_sgio2audio_capture_open(struct snd_pcm_substream *substream)
+{
+	struct snd_sgio2audio *chip = snd_pcm_substream_chip(substream);
+	struct snd_pcm_runtime *runtime = substream->runtime;
+
+	runtime->hw = snd_sgio2audio_pcm_hw;
+	runtime->private_data = &chip->channel[0];
+	return 0;
+}
+
+/* PCM close callback */
+static int snd_sgio2audio_pcm_close(struct snd_pcm_substream *substream)
+{
+	struct snd_pcm_runtime *runtime = substream->runtime;
+
+	runtime->private_data = NULL;
+	return 0;
+}
+
+
+/* hw_params callback */
+static int snd_sgio2audio_pcm_hw_params(struct snd_pcm_substream *substream,
+					struct snd_pcm_hw_params *hw_params)
+{
+	return snd_pcm_lib_alloc_vmalloc_buffer(substream,
+						params_buffer_bytes(hw_params));
+}
+
+/* hw_free callback */
+static int snd_sgio2audio_pcm_hw_free(struct snd_pcm_substream *substream)
+{
+	return snd_pcm_lib_free_vmalloc_buffer(substream);
+}
+
+/* prepare callback */
+static int snd_sgio2audio_pcm_prepare(struct snd_pcm_substream *substream)
+{
+	struct snd_sgio2audio *chip = snd_pcm_substream_chip(substream);
+	struct snd_pcm_runtime *runtime = substream->runtime;
+	struct snd_sgio2audio_chan *chan = substream->runtime->private_data;
+	int ch = chan->idx;
+	unsigned long flags;
+
+	spin_lock_irqsave(&chip->channel[ch].lock, flags);
+
+	/* Setup the pseudo-dma transfer pointers.  */
+	chip->channel[ch].pos = 0;
+	chip->channel[ch].size = 0;
+	chip->channel[ch].substream = substream;
+
+	/* set AD1843 format */
+	/* hardware format is always S16_LE */
+	switch (substream->stream) {
+	case SNDRV_PCM_STREAM_PLAYBACK:
+		ad1843_setup_dac(&chip->ad1843,
+				 ch - 1,
+				 runtime->rate,
+				 SNDRV_PCM_FORMAT_S16_LE,
+				 runtime->channels);
+		break;
+	case SNDRV_PCM_STREAM_CAPTURE:
+		ad1843_setup_adc(&chip->ad1843,
+				 runtime->rate,
+				 SNDRV_PCM_FORMAT_S16_LE,
+				 runtime->channels);
+		break;
+	}
+	spin_unlock_irqrestore(&chip->channel[ch].lock, flags);
+	return 0;
+}
+
+/* trigger callback */
+static int snd_sgio2audio_pcm_trigger(struct snd_pcm_substream *substream,
+				      int cmd)
+{
+	switch (cmd) {
+	case SNDRV_PCM_TRIGGER_START:
+		/* start the PCM engine */
+		snd_sgio2audio_dma_start(substream);
+		break;
+	case SNDRV_PCM_TRIGGER_STOP:
+		/* stop the PCM engine */
+		snd_sgio2audio_dma_stop(substream);
+		break;
+	default:
+		return -EINVAL;
+	}
+	return 0;
+}
+
+/* pointer callback */
+static snd_pcm_uframes_t
+snd_sgio2audio_pcm_pointer(struct snd_pcm_substream *substream)
+{
+	struct snd_sgio2audio *chip = snd_pcm_substream_chip(substream);
+	struct snd_sgio2audio_chan *chan = substream->runtime->private_data;
+
+	/* get the current hardware pointer */
+	return bytes_to_frames(substream->runtime,
+			       chip->channel[chan->idx].pos);
+}
+
+/* operators */
+static struct snd_pcm_ops snd_sgio2audio_playback1_ops = {
+	.open =        snd_sgio2audio_playback1_open,
+	.close =       snd_sgio2audio_pcm_close,
+	.ioctl =       snd_pcm_lib_ioctl,
+	.hw_params =   snd_sgio2audio_pcm_hw_params,
+	.hw_free =     snd_sgio2audio_pcm_hw_free,
+	.prepare =     snd_sgio2audio_pcm_prepare,
+	.trigger =     snd_sgio2audio_pcm_trigger,
+	.pointer =     snd_sgio2audio_pcm_pointer,
+	.page =        snd_pcm_lib_get_vmalloc_page,
+	.mmap =        snd_pcm_lib_mmap_vmalloc,
+};
+
+static struct snd_pcm_ops snd_sgio2audio_playback2_ops = {
+	.open =        snd_sgio2audio_playback2_open,
+	.close =       snd_sgio2audio_pcm_close,
+	.ioctl =       snd_pcm_lib_ioctl,
+	.hw_params =   snd_sgio2audio_pcm_hw_params,
+	.hw_free =     snd_sgio2audio_pcm_hw_free,
+	.prepare =     snd_sgio2audio_pcm_prepare,
+	.trigger =     snd_sgio2audio_pcm_trigger,
+	.pointer =     snd_sgio2audio_pcm_pointer,
+	.page =        snd_pcm_lib_get_vmalloc_page,
+	.mmap =        snd_pcm_lib_mmap_vmalloc,
+};
+
+static struct snd_pcm_ops snd_sgio2audio_capture_ops = {
+	.open =        snd_sgio2audio_capture_open,
+	.close =       snd_sgio2audio_pcm_close,
+	.ioctl =       snd_pcm_lib_ioctl,
+	.hw_params =   snd_sgio2audio_pcm_hw_params,
+	.hw_free =     snd_sgio2audio_pcm_hw_free,
+	.prepare =     snd_sgio2audio_pcm_prepare,
+	.trigger =     snd_sgio2audio_pcm_trigger,
+	.pointer =     snd_sgio2audio_pcm_pointer,
+	.page =        snd_pcm_lib_get_vmalloc_page,
+	.mmap =        snd_pcm_lib_mmap_vmalloc,
+};
+
+/*
+ *  definitions of capture are omitted here...
+ */
+
+/* create a pcm device */
+static int __devinit snd_sgio2audio_new_pcm(struct snd_sgio2audio *chip)
+{
+	struct snd_pcm *pcm;
+	int err;
+
+	/* create first pcm device with one outputs and one input */
+	err = snd_pcm_new(chip->card, "SGI O2 Audio", 0, 1, 1, &pcm);
+	if (err < 0)
+		return err;
+
+	pcm->private_data = chip;
+	strcpy(pcm->name, "SGI O2 DAC1");
+
+	/* set operators */
+	snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK,
+			&snd_sgio2audio_playback1_ops);
+	snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE,
+			&snd_sgio2audio_capture_ops);
+
+	/* create second  pcm device with one outputs and no input */
+	err = snd_pcm_new(chip->card, "SGI O2 Audio", 1, 1, 0, &pcm);
+	if (err < 0)
+		return err;
+
+	pcm->private_data = chip;
+	strcpy(pcm->name, "SGI O2 DAC2");
+
+	/* set operators */
+	snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK,
+			&snd_sgio2audio_playback2_ops);
+
+	return 0;
+}
+
+static struct {
+	int idx;
+	int irq;
+	irqreturn_t (*isr)(int, void *);
+	const char *desc;
+} snd_sgio2_isr_table[] = {
+	{
+		.idx = 0,
+		.irq = MACEISA_AUDIO1_DMAT_IRQ,
+		.isr = snd_sgio2audio_dma_in_isr,
+		.desc = "Capture DMA Channel 0"
+	}, {
+		.idx = 0,
+		.irq = MACEISA_AUDIO1_OF_IRQ,
+		.isr = snd_sgio2audio_error_isr,
+		.desc = "Capture Overflow"
+	}, {
+		.idx = 1,
+		.irq = MACEISA_AUDIO2_DMAT_IRQ,
+		.isr = snd_sgio2audio_dma_out_isr,
+		.desc = "Playback DMA Channel 1"
+	}, {
+		.idx = 1,
+		.irq = MACEISA_AUDIO2_MERR_IRQ,
+		.isr = snd_sgio2audio_error_isr,
+		.desc = "Memory Error Channel 1"
+	}, {
+		.idx = 2,
+		.irq = MACEISA_AUDIO3_DMAT_IRQ,
+		.isr = snd_sgio2audio_dma_out_isr,
+		.desc = "Playback DMA Channel 2"
+	}, {
+		.idx = 2,
+		.irq = MACEISA_AUDIO3_MERR_IRQ,
+		.isr = snd_sgio2audio_error_isr,
+		.desc = "Memory Error Channel 2"
+	}
+};
+
+/* ALSA driver */
+
+static int snd_sgio2audio_free(struct snd_sgio2audio *chip)
+{
+	int i;
+
+	/* reset interface */
+	writeq(AUDIO_CONTROL_RESET, &mace->perif.audio.control);
+	udelay(1);
+	writeq(0, &mace->perif.audio.control);
+
+	/* release IRQ's */
+	for (i = 0; i < ARRAY_SIZE(snd_sgio2_isr_table); i++)
+		free_irq(snd_sgio2_isr_table[i].irq,
+			 &chip->channel[snd_sgio2_isr_table[i].idx]);
+
+	dma_free_coherent(NULL, MACEISA_RINGBUFFERS_SIZE,
+			  chip->ring_base, chip->ring_base_dma);
+
+	/* release card data */
+	kfree(chip);
+	return 0;
+}
+
+static int snd_sgio2audio_dev_free(struct snd_device *device)
+{
+	struct snd_sgio2audio *chip = device->device_data;
+
+	return snd_sgio2audio_free(chip);
+}
+
+static struct snd_device_ops ops = {
+	.dev_free = snd_sgio2audio_dev_free,
+};
+
+static int __devinit snd_sgio2audio_create(struct snd_card *card,
+					   struct snd_sgio2audio **rchip)
+{
+	struct snd_sgio2audio *chip;
+	int i, err;
+
+	*rchip = NULL;
+
+	/* check if a codec is attached to the interface */
+	/* (Audio or Audio/Video board present) */
+	if (!(readq(&mace->perif.audio.control) & AUDIO_CONTROL_CODEC_PRESENT))
+		return -ENOENT;
+
+	chip = kzalloc(sizeof(struct snd_sgio2audio), GFP_KERNEL);
+	if (chip == NULL)
+		return -ENOMEM;
+
+	chip->card = card;
+
+	chip->ring_base = dma_alloc_coherent(NULL, MACEISA_RINGBUFFERS_SIZE,
+					     &chip->ring_base_dma, GFP_USER);
+	if (chip->ring_base == NULL) {
+		printk(KERN_ERR
+		       "sgio2audio: could not allocate ring buffers\n");
+		kfree(chip);
+		return -ENOMEM;
+	}
+
+	spin_lock_init(&chip->ad1843_lock);
+
+	/* initialize channels */
+	for (i = 0; i < 3; i++) {
+		spin_lock_init(&chip->channel[i].lock);
+		chip->channel[i].idx = i;
+	}
+
+	/* allocate IRQs */
+	for (i = 0; i < ARRAY_SIZE(snd_sgio2_isr_table); i++) {
+		if (request_irq(snd_sgio2_isr_table[i].irq,
+				snd_sgio2_isr_table[i].isr,
+				0,
+				snd_sgio2_isr_table[i].desc,
+				&chip->channel[snd_sgio2_isr_table[i].idx])) {
+			snd_sgio2audio_free(chip);
+			printk(KERN_ERR "sgio2audio: cannot allocate irq %d\n",
+			       snd_sgio2_isr_table[i].irq);
+			return -EBUSY;
+		}
+	}
+
+	/* reset the interface */
+	writeq(AUDIO_CONTROL_RESET, &mace->perif.audio.control);
+	udelay(1);
+	writeq(0, &mace->perif.audio.control);
+	msleep_interruptible(1); /* give time to recover */
+
+	/* set ring base */
+	writeq(chip->ring_base_dma, &mace->perif.ctrl.ringbase);
+
+	/* attach the AD1843 codec */
+	chip->ad1843.read = read_ad1843_reg;
+	chip->ad1843.write = write_ad1843_reg;
+	chip->ad1843.chip = chip;
+
+	/* initialize the AD1843 codec */
+	err = ad1843_init(&chip->ad1843);
+	if (err < 0) {
+		snd_sgio2audio_free(chip);
+		return err;
+	}
+
+	err = snd_device_new(card, SNDRV_DEV_LOWLEVEL, chip, &ops);
+	if (err < 0) {
+		snd_sgio2audio_free(chip);
+		return err;
+	}
+	*rchip = chip;
+	return 0;
+}
+
+static int __devinit snd_sgio2audio_probe(struct platform_device *pdev)
+{
+	struct snd_card *card;
+	struct snd_sgio2audio *chip;
+	int err;
+
+	err = snd_card_create(index, id, THIS_MODULE, 0, &card);
+	if (err < 0)
+		return err;
+
+	err = snd_sgio2audio_create(card, &chip);
+	if (err < 0) {
+		snd_card_free(card);
+		return err;
+	}
+	snd_card_set_dev(card, &pdev->dev);
+
+	err = snd_sgio2audio_new_pcm(chip);
+	if (err < 0) {
+		snd_card_free(card);
+		return err;
+	}
+	err = snd_sgio2audio_new_mixer(chip);
+	if (err < 0) {
+		snd_card_free(card);
+		return err;
+	}
+
+	strcpy(card->driver, "SGI O2 Audio");
+	strcpy(card->shortname, "SGI O2 Audio");
+	sprintf(card->longname, "%s irq %i-%i",
+		card->shortname,
+		MACEISA_AUDIO1_DMAT_IRQ,
+		MACEISA_AUDIO3_MERR_IRQ);
+
+	err = snd_card_register(card);
+	if (err < 0) {
+		snd_card_free(card);
+		return err;
+	}
+	platform_set_drvdata(pdev, card);
+	return 0;
+}
+
+static int __devexit snd_sgio2audio_remove(struct platform_device *pdev)
+{
+	struct snd_card *card = platform_get_drvdata(pdev);
+
+	snd_card_free(card);
+	platform_set_drvdata(pdev, NULL);
+	return 0;
+}
+
+static struct platform_driver sgio2audio_driver = {
+	.probe	= snd_sgio2audio_probe,
+	.remove	= __devexit_p(snd_sgio2audio_remove),
+	.driver = {
+		.name	= "sgio2audio",
+		.owner	= THIS_MODULE,
+	}
+};
+
+module_platform_driver(sgio2audio_driver);