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-rw-r--r--ANDROID_3.4.5/sound/mips/Kconfig37
-rw-r--r--ANDROID_3.4.5/sound/mips/Makefile12
-rw-r--r--ANDROID_3.4.5/sound/mips/ad1843.c561
-rw-r--r--ANDROID_3.4.5/sound/mips/au1x00.c696
-rw-r--r--ANDROID_3.4.5/sound/mips/hal2.c938
-rw-r--r--ANDROID_3.4.5/sound/mips/hal2.h245
-rw-r--r--ANDROID_3.4.5/sound/mips/sgio2audio.c979
7 files changed, 0 insertions, 3468 deletions
diff --git a/ANDROID_3.4.5/sound/mips/Kconfig b/ANDROID_3.4.5/sound/mips/Kconfig
deleted file mode 100644
index d2f615ab..00000000
--- a/ANDROID_3.4.5/sound/mips/Kconfig
+++ /dev/null
@@ -1,37 +0,0 @@
-# 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
deleted file mode 100644
index 861ec0a5..00000000
--- a/ANDROID_3.4.5/sound/mips/Makefile
+++ /dev/null
@@ -1,12 +0,0 @@
-#
-# 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
deleted file mode 100644
index c624510e..00000000
--- a/ANDROID_3.4.5/sound/mips/ad1843.c
+++ /dev/null
@@ -1,561 +0,0 @@
-/*
- * 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
deleted file mode 100644
index 3f3ec0be..00000000
--- a/ANDROID_3.4.5/sound/mips/au1x00.c
+++ /dev/null
@@ -1,696 +0,0 @@
-/*
- * 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
deleted file mode 100644
index 5f88d1f0..00000000
--- a/ANDROID_3.4.5/sound/mips/hal2.c
+++ /dev/null
@@ -1,938 +0,0 @@
-/*
- * 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
deleted file mode 100644
index f19828bc..00000000
--- a/ANDROID_3.4.5/sound/mips/hal2.h
+++ /dev/null
@@ -1,245 +0,0 @@
-#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
deleted file mode 100644
index ceaa593e..00000000
--- a/ANDROID_3.4.5/sound/mips/sgio2audio.c
+++ /dev/null
@@ -1,979 +0,0 @@
-/*
- * 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);