File-copy from v4.4.100
This is the result of 'cp' from a linux-stable tree with the 'v4.4.100'
tag checked out (commit 26d6298789e695c9f627ce49a7bbd2286405798a) on
git://git.kernel.org/pub/scm/linux/kernel/git/stable/linux.git
Please refer to that tree for all history prior to this point.
Change-Id: I8a9ee2aea93cd29c52c847d0ce33091a73ae6afe
diff --git a/sound/mips/Kconfig b/sound/mips/Kconfig
new file mode 100644
index 0000000..2153d31
--- /dev/null
+++ b/sound/mips/Kconfig
@@ -0,0 +1,39 @@
+# 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
+ select SND_PCM
+ help
+ Sound support for the SGI O2 Workstation.
+
+config SND_SGI_HAL2
+ tristate "SGI HAL2 Audio"
+ depends on SGI_HAS_HAL2
+ select SND_PCM
+ 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/sound/mips/Makefile b/sound/mips/Makefile
new file mode 100644
index 0000000..861ec0a
--- /dev/null
+++ b/sound/mips/Makefile
@@ -0,0 +1,12 @@
+#
+# Makefile for ALSA
+#
+
+snd-au1x00-objs := au1x00.o
+snd-sgi-o2-objs := sgio2audio.o ad1843.o
+snd-sgi-hal2-objs := hal2.o
+
+# Toplevel Module Dependency
+obj-$(CONFIG_SND_AU1X00) += snd-au1x00.o
+obj-$(CONFIG_SND_SGI_O2) += snd-sgi-o2.o
+obj-$(CONFIG_SND_SGI_HAL2) += snd-sgi-hal2.o
diff --git a/sound/mips/ad1843.c b/sound/mips/ad1843.c
new file mode 100644
index 0000000..5869075
--- /dev/null
+++ b/sound/mips/ad1843.c
@@ -0,0 +1,561 @@
+/*
+ * AD1843 low level driver
+ *
+ * Copyright 2003 Vivien Chappelier <vivien.chappelier@linux-mips.org>
+ * Copyright 2008 Thomas Bogendoerfer <tsbogend@alpha.franken.de>
+ *
+ * inspired from vwsnd.c (SGI VW audio driver)
+ * Copyright 1999 Silicon Graphics, Inc. All rights reserved.
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ *
+ */
+
+#include <linux/init.h>
+#include <linux/sched.h>
+#include <linux/errno.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/ad1843.h>
+
+/*
+ * AD1843 bitfield definitions. All are named as in the AD1843 data
+ * sheet, with ad1843_ prepended and individual bit numbers removed.
+ *
+ * E.g., bits LSS0 through LSS2 become ad1843_LSS.
+ *
+ * Only the bitfields we need are defined.
+ */
+
+struct ad1843_bitfield {
+ char reg;
+ char lo_bit;
+ char nbits;
+};
+
+static const struct ad1843_bitfield
+ ad1843_PDNO = { 0, 14, 1 }, /* Converter Power-Down Flag */
+ ad1843_INIT = { 0, 15, 1 }, /* Clock Initialization Flag */
+ ad1843_RIG = { 2, 0, 4 }, /* Right ADC Input Gain */
+ ad1843_RMGE = { 2, 4, 1 }, /* Right ADC Mic Gain Enable */
+ ad1843_RSS = { 2, 5, 3 }, /* Right ADC Source Select */
+ ad1843_LIG = { 2, 8, 4 }, /* Left ADC Input Gain */
+ ad1843_LMGE = { 2, 12, 1 }, /* Left ADC Mic Gain Enable */
+ ad1843_LSS = { 2, 13, 3 }, /* Left ADC Source Select */
+ ad1843_RD2M = { 3, 0, 5 }, /* Right DAC 2 Mix Gain/Atten */
+ ad1843_RD2MM = { 3, 7, 1 }, /* Right DAC 2 Mix Mute */
+ ad1843_LD2M = { 3, 8, 5 }, /* Left DAC 2 Mix Gain/Atten */
+ ad1843_LD2MM = { 3, 15, 1 }, /* Left DAC 2 Mix Mute */
+ ad1843_RX1M = { 4, 0, 5 }, /* Right Aux 1 Mix Gain/Atten */
+ ad1843_RX1MM = { 4, 7, 1 }, /* Right Aux 1 Mix Mute */
+ ad1843_LX1M = { 4, 8, 5 }, /* Left Aux 1 Mix Gain/Atten */
+ ad1843_LX1MM = { 4, 15, 1 }, /* Left Aux 1 Mix Mute */
+ ad1843_RX2M = { 5, 0, 5 }, /* Right Aux 2 Mix Gain/Atten */
+ ad1843_RX2MM = { 5, 7, 1 }, /* Right Aux 2 Mix Mute */
+ ad1843_LX2M = { 5, 8, 5 }, /* Left Aux 2 Mix Gain/Atten */
+ ad1843_LX2MM = { 5, 15, 1 }, /* Left Aux 2 Mix Mute */
+ ad1843_RMCM = { 7, 0, 5 }, /* Right Mic Mix Gain/Atten */
+ ad1843_RMCMM = { 7, 7, 1 }, /* Right Mic Mix Mute */
+ ad1843_LMCM = { 7, 8, 5 }, /* Left Mic Mix Gain/Atten */
+ ad1843_LMCMM = { 7, 15, 1 }, /* Left Mic Mix Mute */
+ ad1843_HPOS = { 8, 4, 1 }, /* Headphone Output Voltage Swing */
+ ad1843_HPOM = { 8, 5, 1 }, /* Headphone Output Mute */
+ ad1843_MPOM = { 8, 6, 1 }, /* Mono Output Mute */
+ ad1843_RDA1G = { 9, 0, 6 }, /* Right DAC1 Analog/Digital Gain */
+ ad1843_RDA1GM = { 9, 7, 1 }, /* Right DAC1 Analog Mute */
+ ad1843_LDA1G = { 9, 8, 6 }, /* Left DAC1 Analog/Digital Gain */
+ ad1843_LDA1GM = { 9, 15, 1 }, /* Left DAC1 Analog Mute */
+ ad1843_RDA2G = { 10, 0, 6 }, /* Right DAC2 Analog/Digital Gain */
+ ad1843_RDA2GM = { 10, 7, 1 }, /* Right DAC2 Analog Mute */
+ ad1843_LDA2G = { 10, 8, 6 }, /* Left DAC2 Analog/Digital Gain */
+ ad1843_LDA2GM = { 10, 15, 1 }, /* Left DAC2 Analog Mute */
+ ad1843_RDA1AM = { 11, 7, 1 }, /* Right DAC1 Digital Mute */
+ ad1843_LDA1AM = { 11, 15, 1 }, /* Left DAC1 Digital Mute */
+ ad1843_RDA2AM = { 12, 7, 1 }, /* Right DAC2 Digital Mute */
+ ad1843_LDA2AM = { 12, 15, 1 }, /* Left DAC2 Digital Mute */
+ ad1843_ADLC = { 15, 0, 2 }, /* ADC Left Sample Rate Source */
+ ad1843_ADRC = { 15, 2, 2 }, /* ADC Right Sample Rate Source */
+ ad1843_DA1C = { 15, 8, 2 }, /* DAC1 Sample Rate Source */
+ ad1843_DA2C = { 15, 10, 2 }, /* DAC2 Sample Rate Source */
+ ad1843_C1C = { 17, 0, 16 }, /* Clock 1 Sample Rate Select */
+ ad1843_C2C = { 20, 0, 16 }, /* Clock 2 Sample Rate Select */
+ ad1843_C3C = { 23, 0, 16 }, /* Clock 3 Sample Rate Select */
+ ad1843_DAADL = { 25, 4, 2 }, /* Digital ADC Left Source Select */
+ ad1843_DAADR = { 25, 6, 2 }, /* Digital ADC Right Source Select */
+ ad1843_DAMIX = { 25, 14, 1 }, /* DAC Digital Mix Enable */
+ ad1843_DRSFLT = { 25, 15, 1 }, /* Digital Reampler Filter Mode */
+ ad1843_ADLF = { 26, 0, 2 }, /* ADC Left Channel Data Format */
+ ad1843_ADRF = { 26, 2, 2 }, /* ADC Right Channel Data Format */
+ ad1843_ADTLK = { 26, 4, 1 }, /* ADC Transmit Lock Mode Select */
+ ad1843_SCF = { 26, 7, 1 }, /* SCLK Frequency Select */
+ ad1843_DA1F = { 26, 8, 2 }, /* DAC1 Data Format Select */
+ ad1843_DA2F = { 26, 10, 2 }, /* DAC2 Data Format Select */
+ ad1843_DA1SM = { 26, 14, 1 }, /* DAC1 Stereo/Mono Mode Select */
+ ad1843_DA2SM = { 26, 15, 1 }, /* DAC2 Stereo/Mono Mode Select */
+ ad1843_ADLEN = { 27, 0, 1 }, /* ADC Left Channel Enable */
+ ad1843_ADREN = { 27, 1, 1 }, /* ADC Right Channel Enable */
+ ad1843_AAMEN = { 27, 4, 1 }, /* Analog to Analog Mix Enable */
+ ad1843_ANAEN = { 27, 7, 1 }, /* Analog Channel Enable */
+ ad1843_DA1EN = { 27, 8, 1 }, /* DAC1 Enable */
+ ad1843_DA2EN = { 27, 9, 1 }, /* DAC2 Enable */
+ ad1843_DDMEN = { 27, 12, 1 }, /* DAC2 to DAC1 Mix Enable */
+ ad1843_C1EN = { 28, 11, 1 }, /* Clock Generator 1 Enable */
+ ad1843_C2EN = { 28, 12, 1 }, /* Clock Generator 2 Enable */
+ ad1843_C3EN = { 28, 13, 1 }, /* Clock Generator 3 Enable */
+ ad1843_PDNI = { 28, 15, 1 }; /* Converter Power Down */
+
+/*
+ * The various registers of the AD1843 use three different formats for
+ * specifying gain. The ad1843_gain structure parameterizes the
+ * formats.
+ */
+
+struct ad1843_gain {
+ int negative; /* nonzero if gain is negative. */
+ const struct ad1843_bitfield *lfield;
+ const struct ad1843_bitfield *rfield;
+ const struct ad1843_bitfield *lmute;
+ const struct ad1843_bitfield *rmute;
+};
+
+static const struct ad1843_gain ad1843_gain_RECLEV = {
+ .negative = 0,
+ .lfield = &ad1843_LIG,
+ .rfield = &ad1843_RIG
+};
+static const struct ad1843_gain ad1843_gain_LINE = {
+ .negative = 1,
+ .lfield = &ad1843_LX1M,
+ .rfield = &ad1843_RX1M,
+ .lmute = &ad1843_LX1MM,
+ .rmute = &ad1843_RX1MM
+};
+static const struct ad1843_gain ad1843_gain_LINE_2 = {
+ .negative = 1,
+ .lfield = &ad1843_LDA2G,
+ .rfield = &ad1843_RDA2G,
+ .lmute = &ad1843_LDA2GM,
+ .rmute = &ad1843_RDA2GM
+};
+static const struct ad1843_gain ad1843_gain_MIC = {
+ .negative = 1,
+ .lfield = &ad1843_LMCM,
+ .rfield = &ad1843_RMCM,
+ .lmute = &ad1843_LMCMM,
+ .rmute = &ad1843_RMCMM
+};
+static const struct ad1843_gain ad1843_gain_PCM_0 = {
+ .negative = 1,
+ .lfield = &ad1843_LDA1G,
+ .rfield = &ad1843_RDA1G,
+ .lmute = &ad1843_LDA1GM,
+ .rmute = &ad1843_RDA1GM
+};
+static const struct ad1843_gain ad1843_gain_PCM_1 = {
+ .negative = 1,
+ .lfield = &ad1843_LD2M,
+ .rfield = &ad1843_RD2M,
+ .lmute = &ad1843_LD2MM,
+ .rmute = &ad1843_RD2MM
+};
+
+static const struct ad1843_gain *ad1843_gain[AD1843_GAIN_SIZE] =
+{
+ &ad1843_gain_RECLEV,
+ &ad1843_gain_LINE,
+ &ad1843_gain_LINE_2,
+ &ad1843_gain_MIC,
+ &ad1843_gain_PCM_0,
+ &ad1843_gain_PCM_1,
+};
+
+/* read the current value of an AD1843 bitfield. */
+
+static int ad1843_read_bits(struct snd_ad1843 *ad1843,
+ const struct ad1843_bitfield *field)
+{
+ int w;
+
+ w = ad1843->read(ad1843->chip, field->reg);
+ return w >> field->lo_bit & ((1 << field->nbits) - 1);
+}
+
+/*
+ * write a new value to an AD1843 bitfield and return the old value.
+ */
+
+static int ad1843_write_bits(struct snd_ad1843 *ad1843,
+ const struct ad1843_bitfield *field,
+ int newval)
+{
+ int w, mask, oldval, newbits;
+
+ w = ad1843->read(ad1843->chip, field->reg);
+ mask = ((1 << field->nbits) - 1) << field->lo_bit;
+ oldval = (w & mask) >> field->lo_bit;
+ newbits = (newval << field->lo_bit) & mask;
+ w = (w & ~mask) | newbits;
+ ad1843->write(ad1843->chip, field->reg, w);
+
+ return oldval;
+}
+
+/*
+ * ad1843_read_multi reads multiple bitfields from the same AD1843
+ * register. It uses a single read cycle to do it. (Reading the
+ * ad1843 requires 256 bit times at 12.288 MHz, or nearly 20
+ * microseconds.)
+ *
+ * Called like this.
+ *
+ * ad1843_read_multi(ad1843, nfields,
+ * &ad1843_FIELD1, &val1,
+ * &ad1843_FIELD2, &val2, ...);
+ */
+
+static void ad1843_read_multi(struct snd_ad1843 *ad1843, int argcount, ...)
+{
+ va_list ap;
+ const struct ad1843_bitfield *fp;
+ int w = 0, mask, *value, reg = -1;
+
+ va_start(ap, argcount);
+ while (--argcount >= 0) {
+ fp = va_arg(ap, const struct ad1843_bitfield *);
+ value = va_arg(ap, int *);
+ if (reg == -1) {
+ reg = fp->reg;
+ w = ad1843->read(ad1843->chip, reg);
+ }
+
+ mask = (1 << fp->nbits) - 1;
+ *value = w >> fp->lo_bit & mask;
+ }
+ va_end(ap);
+}
+
+/*
+ * ad1843_write_multi stores multiple bitfields into the same AD1843
+ * register. It uses one read and one write cycle to do it.
+ *
+ * Called like this.
+ *
+ * ad1843_write_multi(ad1843, nfields,
+ * &ad1843_FIELD1, val1,
+ * &ad1843_FIELF2, val2, ...);
+ */
+
+static void ad1843_write_multi(struct snd_ad1843 *ad1843, int argcount, ...)
+{
+ va_list ap;
+ int reg;
+ const struct ad1843_bitfield *fp;
+ int value;
+ int w, m, mask, bits;
+
+ mask = 0;
+ bits = 0;
+ reg = -1;
+
+ va_start(ap, argcount);
+ while (--argcount >= 0) {
+ fp = va_arg(ap, const struct ad1843_bitfield *);
+ value = va_arg(ap, int);
+ if (reg == -1)
+ reg = fp->reg;
+ else
+ WARN_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/sound/mips/au1x00.c b/sound/mips/au1x00.c
new file mode 100644
index 0000000..1e30e84
--- /dev/null
+++ b/sound/mips/au1x00.c
@@ -0,0 +1,734 @@
+/*
+ * 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/platform_device.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 */
+ int dmaid[2]; /* tx(0)/rx(1) DMA ids */
+};
+
+/*--------------------------- 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
+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();
+ au1000->stream[PLAYBACK]->dma = request_au1000_dma(au1000->dmaid[0],
+ "AC97 TX", au1000_dma_interrupt, 0,
+ au1000->stream[PLAYBACK]);
+ if (au1000->stream[PLAYBACK]->dma < 0) {
+ release_dma_lock(flags);
+ return -EBUSY;
+ }
+ au1000->stream[CAPTURE]->dma = request_au1000_dma(au1000->dmaid[1],
+ "AC97 RX", au1000_dma_interrupt, 0,
+ au1000->stream[CAPTURE]);
+ if (au1000->stream[CAPTURE]->dma < 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);
+}
+
+/*------------------------------ Setup / Destroy ----------------------------*/
+
+static void snd_au1000_free(struct snd_card *card)
+{
+ struct snd_au1000 *au1000 = card->private_data;
+
+ 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]);
+ }
+
+ if (au1000->ac97_res_port) {
+ /* put internal AC97 block into reset */
+ if (au1000->ac97_ioport) {
+ au1000->ac97_ioport->cntrl = AC97C_RS;
+ iounmap(au1000->ac97_ioport);
+ au1000->ac97_ioport = NULL;
+ }
+ release_and_free_resource(au1000->ac97_res_port);
+ au1000->ac97_res_port = NULL;
+ }
+}
+
+static struct snd_ac97_bus_ops ops = {
+ .write = snd_au1000_ac97_write,
+ .read = snd_au1000_ac97_read,
+};
+
+static int au1000_ac97_probe(struct platform_device *pdev)
+{
+ int err;
+ void __iomem *io;
+ struct resource *r;
+ struct snd_card *card;
+ struct snd_au1000 *au1000;
+ struct snd_ac97_bus *pbus;
+ struct snd_ac97_template ac97;
+
+ err = snd_card_new(&pdev->dev, -1, "AC97", THIS_MODULE,
+ sizeof(struct snd_au1000), &card);
+ if (err < 0)
+ return err;
+
+ au1000 = card->private_data;
+ au1000->card = card;
+ spin_lock_init(&au1000->ac97_lock);
+
+ /* from here on let ALSA call the special freeing function */
+ card->private_free = snd_au1000_free;
+
+ /* TX DMA ID */
+ r = platform_get_resource(pdev, IORESOURCE_DMA, 0);
+ if (!r) {
+ err = -ENODEV;
+ snd_printk(KERN_INFO "no TX DMA platform resource!\n");
+ goto out;
+ }
+ au1000->dmaid[0] = r->start;
+
+ /* RX DMA ID */
+ r = platform_get_resource(pdev, IORESOURCE_DMA, 1);
+ if (!r) {
+ err = -ENODEV;
+ snd_printk(KERN_INFO "no RX DMA platform resource!\n");
+ goto out;
+ }
+ au1000->dmaid[1] = r->start;
+
+ au1000->stream[PLAYBACK] = kmalloc(sizeof(struct audio_stream),
+ GFP_KERNEL);
+ if (!au1000->stream[PLAYBACK]) {
+ err = -ENOMEM;
+ goto out;
+ }
+ au1000->stream[PLAYBACK]->dma = -1;
+
+ au1000->stream[CAPTURE] = kmalloc(sizeof(struct audio_stream),
+ GFP_KERNEL);
+ if (!au1000->stream[CAPTURE]) {
+ err = -ENOMEM;
+ goto out;
+ }
+ au1000->stream[CAPTURE]->dma = -1;
+
+ r = platform_get_resource(pdev, IORESOURCE_MEM, 0);
+ if (!r) {
+ err = -ENODEV;
+ goto out;
+ }
+
+ err = -EBUSY;
+ au1000->ac97_res_port = request_mem_region(r->start, resource_size(r),
+ pdev->name);
+ if (!au1000->ac97_res_port) {
+ snd_printk(KERN_ERR "ALSA AC97: can't grab AC97 port\n");
+ goto out;
+ }
+
+ io = ioremap(r->start, resource_size(r));
+ if (!io)
+ goto out;
+
+ au1000->ac97_ioport = (struct au1000_ac97_reg *)io;
+
+ /* 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 */
+ err = snd_ac97_bus(au1000->card, 0, &ops, au1000, &pbus);
+ if (err < 0)
+ goto out;
+
+ memset(&ac97, 0, sizeof(ac97));
+ ac97.private_data = au1000;
+ err = snd_ac97_mixer(pbus, &ac97, &au1000->ac97);
+ if (err < 0)
+ goto out;
+
+ err = snd_au1000_pcm_new(au1000);
+ if (err < 0)
+ goto out;
+
+ strcpy(card->driver, "Au1000-AC97");
+ strcpy(card->shortname, "AMD Au1000-AC97");
+ sprintf(card->longname, "AMD Au1000--AC97 ALSA Driver");
+
+ err = snd_card_register(card);
+ if (err < 0)
+ goto out;
+
+ printk(KERN_INFO "ALSA AC97: Driver Initialized\n");
+
+ platform_set_drvdata(pdev, card);
+
+ return 0;
+
+ out:
+ snd_card_free(card);
+ return err;
+}
+
+static int au1000_ac97_remove(struct platform_device *pdev)
+{
+ return snd_card_free(platform_get_drvdata(pdev));
+}
+
+struct platform_driver au1000_ac97c_driver = {
+ .driver = {
+ .name = "au1000-ac97c",
+ .owner = THIS_MODULE,
+ },
+ .probe = au1000_ac97_probe,
+ .remove = au1000_ac97_remove,
+};
+
+module_platform_driver(au1000_ac97c_driver);
diff --git a/sound/mips/hal2.c b/sound/mips/hal2.c
new file mode 100644
index 0000000..ede449f
--- /dev/null
+++ b/sound/mips/hal2.c
@@ -0,0 +1,935 @@
+/*
+ * 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(®s->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), ®s->iar);
+ H2_INDIRECT_WAIT(regs);
+ ret = hal2_read(®s->idr0) & 0xffff;
+ hal2_write(H2_READ_ADDR(addr) | 0x1, ®s->iar);
+ H2_INDIRECT_WAIT(regs);
+ ret |= (hal2_read(®s->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, ®s->idr0);
+ hal2_write(0, ®s->idr1);
+ hal2_write(0, ®s->idr2);
+ hal2_write(0, ®s->idr3);
+ hal2_write(H2_WRITE_ADDR(addr), ®s->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, ®s->idr0);
+ hal2_write(val >> 16, ®s->idr1);
+ hal2_write(0, ®s->idr2);
+ hal2_write(0, ®s->idr3);
+ hal2_write(H2_WRITE_ADDR(addr), ®s->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), ®s->iar);
+ H2_INDIRECT_WAIT(regs);
+ hal2_write((hal2_read(®s->idr0) & 0xffff) | bit, ®s->idr0);
+ hal2_write(0, ®s->idr1);
+ hal2_write(0, ®s->idr2);
+ hal2_write(0, ®s->idr3);
+ hal2_write(H2_WRITE_ADDR(addr), ®s->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), ®s->iar);
+ H2_INDIRECT_WAIT(regs);
+ hal2_write((hal2_read(®s->idr0) & 0xffff) & ~bit, ®s->idr0);
+ hal2_write(0, ®s->idr1);
+ hal2_write(0, ®s->idr2);
+ hal2_write(0, ®s->idr3);
+ hal2_write(H2_WRITE_ADDR(addr), ®s->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 = {
+ .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 = {
+ .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 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 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 hal2_probe(struct platform_device *pdev)
+{
+ struct snd_card *card;
+ struct snd_hal2 *chip;
+ int err;
+
+ err = snd_card_new(&pdev->dev, 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;
+ }
+
+ 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 hal2_remove(struct platform_device *pdev)
+{
+ struct snd_card *card = platform_get_drvdata(pdev);
+
+ snd_card_free(card);
+ return 0;
+}
+
+static struct platform_driver hal2_driver = {
+ .probe = hal2_probe,
+ .remove = hal2_remove,
+ .driver = {
+ .name = "sgihal2",
+ }
+};
+
+module_platform_driver(hal2_driver);
diff --git a/sound/mips/hal2.h b/sound/mips/hal2.h
new file mode 100644
index 0000000..f19828b
--- /dev/null
+++ b/sound/mips/hal2.h
@@ -0,0 +1,245 @@
+#ifndef __HAL2_H
+#define __HAL2_H
+
+/*
+ * Driver for HAL2 sound processors
+ * Copyright (c) 1999 Ulf Carlsson <ulfc@bun.falkenberg.se>
+ * Copyright (c) 2001, 2002, 2003 Ladislav Michl <ladis@linux-mips.org>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
+ *
+ */
+
+#include <linux/types.h>
+
+/* Indirect status register */
+
+#define H2_ISR_TSTATUS 0x01 /* RO: transaction status 1=busy */
+#define H2_ISR_USTATUS 0x02 /* RO: utime status bit 1=armed */
+#define H2_ISR_QUAD_MODE 0x04 /* codec mode 0=indigo 1=quad */
+#define H2_ISR_GLOBAL_RESET_N 0x08 /* chip global reset 0=reset */
+#define H2_ISR_CODEC_RESET_N 0x10 /* codec/synth reset 0=reset */
+
+/* Revision register */
+
+#define H2_REV_AUDIO_PRESENT 0x8000 /* RO: audio present 0=present */
+#define H2_REV_BOARD_M 0x7000 /* RO: bits 14:12, board revision */
+#define H2_REV_MAJOR_CHIP_M 0x00F0 /* RO: bits 7:4, major chip revision */
+#define H2_REV_MINOR_CHIP_M 0x000F /* RO: bits 3:0, minor chip revision */
+
+/* Indirect address register */
+
+/*
+ * Address of indirect internal register to be accessed. A write to this
+ * register initiates read or write access to the indirect registers in the
+ * HAL2. Note that there af four indirect data registers for write access to
+ * registers larger than 16 byte.
+ */
+
+#define H2_IAR_TYPE_M 0xF000 /* bits 15:12, type of functional */
+ /* block the register resides in */
+ /* 1=DMA Port */
+ /* 9=Global DMA Control */
+ /* 2=Bresenham */
+ /* 3=Unix Timer */
+#define H2_IAR_NUM_M 0x0F00 /* bits 11:8 instance of the */
+ /* blockin which the indirect */
+ /* register resides */
+ /* If IAR_TYPE_M=DMA Port: */
+ /* 1=Synth In */
+ /* 2=AES In */
+ /* 3=AES Out */
+ /* 4=DAC Out */
+ /* 5=ADC Out */
+ /* 6=Synth Control */
+ /* If IAR_TYPE_M=Global DMA Control: */
+ /* 1=Control */
+ /* If IAR_TYPE_M=Bresenham: */
+ /* 1=Bresenham Clock Gen 1 */
+ /* 2=Bresenham Clock Gen 2 */
+ /* 3=Bresenham Clock Gen 3 */
+ /* If IAR_TYPE_M=Unix Timer: */
+ /* 1=Unix Timer */
+#define H2_IAR_ACCESS_SELECT 0x0080 /* 1=read 0=write */
+#define H2_IAR_PARAM 0x000C /* Parameter Select */
+#define H2_IAR_RB_INDEX_M 0x0003 /* Read Back Index */
+ /* 00:word0 */
+ /* 01:word1 */
+ /* 10:word2 */
+ /* 11:word3 */
+/*
+ * HAL2 internal addressing
+ *
+ * The HAL2 has "indirect registers" (idr) which are accessed by writing to the
+ * Indirect Data registers. Write the address to the Indirect Address register
+ * to transfer the data.
+ *
+ * We define the H2IR_* to the read address and H2IW_* to the write address and
+ * H2I_* to be fields in whatever register is referred to.
+ *
+ * When we write to indirect registers which are larger than one word (16 bit)
+ * we have to fill more than one indirect register before writing. When we read
+ * back however we have to read several times, each time with different Read
+ * Back Indexes (there are defs for doing this easily).
+ */
+
+/*
+ * Relay Control
+ */
+#define H2I_RELAY_C 0x9100
+#define H2I_RELAY_C_STATE 0x01 /* state of RELAY pin signal */
+
+/* DMA port enable */
+
+#define H2I_DMA_PORT_EN 0x9104
+#define H2I_DMA_PORT_EN_SY_IN 0x01 /* Synth_in DMA port */
+#define H2I_DMA_PORT_EN_AESRX 0x02 /* AES receiver DMA port */
+#define H2I_DMA_PORT_EN_AESTX 0x04 /* AES transmitter DMA port */
+#define H2I_DMA_PORT_EN_CODECTX 0x08 /* CODEC transmit DMA port */
+#define H2I_DMA_PORT_EN_CODECR 0x10 /* CODEC receive DMA port */
+
+#define H2I_DMA_END 0x9108 /* global dma endian select */
+#define H2I_DMA_END_SY_IN 0x01 /* Synth_in DMA port */
+#define H2I_DMA_END_AESRX 0x02 /* AES receiver DMA port */
+#define H2I_DMA_END_AESTX 0x04 /* AES transmitter DMA port */
+#define H2I_DMA_END_CODECTX 0x08 /* CODEC transmit DMA port */
+#define H2I_DMA_END_CODECR 0x10 /* CODEC receive DMA port */
+ /* 0=b_end 1=l_end */
+
+#define H2I_DMA_DRV 0x910C /* global PBUS DMA enable */
+
+#define H2I_SYNTH_C 0x1104 /* Synth DMA control */
+
+#define H2I_AESRX_C 0x1204 /* AES RX dma control */
+
+#define H2I_C_TS_EN 0x20 /* Timestamp enable */
+#define H2I_C_TS_FRMT 0x40 /* Timestamp format */
+#define H2I_C_NAUDIO 0x80 /* Sign extend */
+
+/* AESRX CTL, 16 bit */
+
+#define H2I_AESTX_C 0x1304 /* AES TX DMA control */
+#define H2I_AESTX_C_CLKID_SHIFT 3 /* Bresenham Clock Gen 1-3 */
+#define H2I_AESTX_C_CLKID_M 0x18
+#define H2I_AESTX_C_DATAT_SHIFT 8 /* 1=mono 2=stereo (3=quad) */
+#define H2I_AESTX_C_DATAT_M 0x300
+
+/* CODEC registers */
+
+#define H2I_DAC_C1 0x1404 /* DAC DMA control, 16 bit */
+#define H2I_DAC_C2 0x1408 /* DAC DMA control, 32 bit */
+#define H2I_ADC_C1 0x1504 /* ADC DMA control, 16 bit */
+#define H2I_ADC_C2 0x1508 /* ADC DMA control, 32 bit */
+
+/* Bits in CTL1 register */
+
+#define H2I_C1_DMA_SHIFT 0 /* DMA channel */
+#define H2I_C1_DMA_M 0x7
+#define H2I_C1_CLKID_SHIFT 3 /* Bresenham Clock Gen 1-3 */
+#define H2I_C1_CLKID_M 0x18
+#define H2I_C1_DATAT_SHIFT 8 /* 1=mono 2=stereo (3=quad) */
+#define H2I_C1_DATAT_M 0x300
+
+/* Bits in CTL2 register */
+
+#define H2I_C2_R_GAIN_SHIFT 0 /* right a/d input gain */
+#define H2I_C2_R_GAIN_M 0xf
+#define H2I_C2_L_GAIN_SHIFT 4 /* left a/d input gain */
+#define H2I_C2_L_GAIN_M 0xf0
+#define H2I_C2_R_SEL 0x100 /* right input select */
+#define H2I_C2_L_SEL 0x200 /* left input select */
+#define H2I_C2_MUTE 0x400 /* mute */
+#define H2I_C2_DO1 0x00010000 /* digital output port bit 0 */
+#define H2I_C2_DO2 0x00020000 /* digital output port bit 1 */
+#define H2I_C2_R_ATT_SHIFT 18 /* right d/a output - */
+#define H2I_C2_R_ATT_M 0x007c0000 /* attenuation */
+#define H2I_C2_L_ATT_SHIFT 23 /* left d/a output - */
+#define H2I_C2_L_ATT_M 0x0f800000 /* attenuation */
+
+#define H2I_SYNTH_MAP_C 0x1104 /* synth dma handshake ctrl */
+
+/* Clock generator CTL 1, 16 bit */
+
+#define H2I_BRES1_C1 0x2104
+#define H2I_BRES2_C1 0x2204
+#define H2I_BRES3_C1 0x2304
+
+#define H2I_BRES_C1_SHIFT 0 /* 0=48.0 1=44.1 2=aes_rx */
+#define H2I_BRES_C1_M 0x03
+
+/* Clock generator CTL 2, 32 bit */
+
+#define H2I_BRES1_C2 0x2108
+#define H2I_BRES2_C2 0x2208
+#define H2I_BRES3_C2 0x2308
+
+#define H2I_BRES_C2_INC_SHIFT 0 /* increment value */
+#define H2I_BRES_C2_INC_M 0xffff
+#define H2I_BRES_C2_MOD_SHIFT 16 /* modcontrol value */
+#define H2I_BRES_C2_MOD_M 0xffff0000 /* modctrl=0xffff&(modinc-1) */
+
+/* Unix timer, 64 bit */
+
+#define H2I_UTIME 0x3104
+#define H2I_UTIME_0_LD 0xffff /* microseconds, LSB's */
+#define H2I_UTIME_1_LD0 0x0f /* microseconds, MSB's */
+#define H2I_UTIME_1_LD1 0xf0 /* tenths of microseconds */
+#define H2I_UTIME_2_LD 0xffff /* seconds, LSB's */
+#define H2I_UTIME_3_LD 0xffff /* seconds, MSB's */
+
+struct hal2_ctl_regs {
+ u32 _unused0[4];
+ u32 isr; /* 0x10 Status Register */
+ u32 _unused1[3];
+ u32 rev; /* 0x20 Revision Register */
+ u32 _unused2[3];
+ u32 iar; /* 0x30 Indirect Address Register */
+ u32 _unused3[3];
+ u32 idr0; /* 0x40 Indirect Data Register 0 */
+ u32 _unused4[3];
+ u32 idr1; /* 0x50 Indirect Data Register 1 */
+ u32 _unused5[3];
+ u32 idr2; /* 0x60 Indirect Data Register 2 */
+ u32 _unused6[3];
+ u32 idr3; /* 0x70 Indirect Data Register 3 */
+};
+
+struct hal2_aes_regs {
+ u32 rx_stat[2]; /* Status registers */
+ u32 rx_cr[2]; /* Control registers */
+ u32 rx_ud[4]; /* User data window */
+ u32 rx_st[24]; /* Channel status data */
+
+ u32 tx_stat[1]; /* Status register */
+ u32 tx_cr[3]; /* Control registers */
+ u32 tx_ud[4]; /* User data window */
+ u32 tx_st[24]; /* Channel status data */
+};
+
+struct hal2_vol_regs {
+ u32 right; /* Right volume */
+ u32 left; /* Left volume */
+};
+
+struct hal2_syn_regs {
+ u32 _unused0[2];
+ u32 page; /* DOC Page register */
+ u32 regsel; /* DOC Register selection */
+ u32 dlow; /* DOC Data low */
+ u32 dhigh; /* DOC Data high */
+ u32 irq; /* IRQ Status */
+ u32 dram; /* DRAM Access */
+};
+
+#endif /* __HAL2_H */
diff --git a/sound/mips/sgio2audio.c b/sound/mips/sgio2audio.c
new file mode 100644
index 0000000..f07aa39
--- /dev/null
+++ b/sound/mips/sgio2audio.c
@@ -0,0 +1,969 @@
+/*
+ * 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 * const texts[3] = {
+ "Cam Mic", "Mic", "Line"
+ };
+ return snd_ctl_enum_info(uinfo, 1, 3, texts);
+}
+
+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 = {
+ .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 = {
+ .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 = {
+ .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 = {
+ .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 = {
+ .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 = {
+ .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 = {
+ .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 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 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 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 snd_sgio2audio_probe(struct platform_device *pdev)
+{
+ struct snd_card *card;
+ struct snd_sgio2audio *chip;
+ int err;
+
+ err = snd_card_new(&pdev->dev, 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;
+ }
+
+ 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 snd_sgio2audio_remove(struct platform_device *pdev)
+{
+ struct snd_card *card = platform_get_drvdata(pdev);
+
+ snd_card_free(card);
+ return 0;
+}
+
+static struct platform_driver sgio2audio_driver = {
+ .probe = snd_sgio2audio_probe,
+ .remove = snd_sgio2audio_remove,
+ .driver = {
+ .name = "sgio2audio",
+ }
+};
+
+module_platform_driver(sgio2audio_driver);