On Sat, Jun 28, 2008 at 02:09:16AM +0200, Thomas Bogendoerfer wrote:
> the dma ring buffer instead of using get_free_pages(). The driver
> is still missing cleanups and features.
v2 is now checkpatch clean and I switched over to use a platform device.
The next step is now to get capture and 2nd playback going.
Thomas.
arch/mips/sgi-ip32/ip32-platform.c | 18 +
include/sound/ad1843.h | 47 +++
sound/mips/Kconfig | 6 +
sound/mips/Makefile | 2 +
sound/mips/ad1843.c | 644 ++++++++++++++++++++++++++++++++
sound/mips/sgio2audio.c | 714 ++++++++++++++++++++++++++++++++++++
6 files changed, 1431 insertions(+), 0 deletions(-)
diff --git a/arch/mips/sgi-ip32/ip32-platform.c b/arch/mips/sgi-ip32/ip32-platform.c
index 89a71f4..8b624b9 100644
--- a/arch/mips/sgi-ip32/ip32-platform.c
+++ b/arch/mips/sgi-ip32/ip32-platform.c
@@ -65,6 +65,24 @@ static __init int meth_devinit(void)
device_initcall(meth_devinit);
+static __init int sgio2audio_devinit(void)
+{
+ struct platform_device *pd;
+ int ret;
+
+ pd = platform_device_alloc("sgio2audio", -1);
+ if (!pd)
+ return -ENOMEM;
+
+ ret = platform_device_add(pd);
+ if (ret)
+ platform_device_put(pd);
+
+ return ret;
+}
+
+device_initcall(sgio2audio_devinit);
+
MODULE_AUTHOR("Ralf Baechle <ralf@xxxxxxxxxxxxxx>");
MODULE_LICENSE("GPL");
MODULE_DESCRIPTION("8250 UART probe driver for SGI IP32 aka O2");
diff --git a/include/sound/ad1843.h b/include/sound/ad1843.h
new file mode 100644
index 0000000..492d10c
--- /dev/null
+++ b/include/sound/ad1843.h
@@ -0,0 +1,47 @@
+/*
+ * This file is subject to the terms and conditions of the GNU General Public
+ * License. See the file "COPYING" in the main directory of this archive
+ * for more details.
+ *
+ * Copyright 2003 Vivien Chappelier <vivien.chappelier@xxxxxxxxxxxxxx>
+ */
+
+#ifndef __SOUND_AD1843_H
+#define __SOUND_AD1843_H
+
+struct snd_ad1843 {
+ void *chip;
+ int (*read)(void *chip, int reg);
+ int (*write)(void *chip, int reg, int val);
+};
+
+#define AD1843_GAIN_RECLEV 0
+#define AD1843_GAIN_LINE 1
+#define AD1843_GAIN_CD 2
+#define AD1843_GAIN_MIC 3
+#define AD1843_GAIN_PCM_0 4
+#define AD1843_GAIN_PCM_1 5
+#define AD1843_GAIN_SIZE (AD1843_GAIN_PCM_1+1)
+
+int ad1843_get_gain(struct snd_ad1843 *ad1843, int id);
+int ad1843_set_gain(struct snd_ad1843 *ad1843, int id, int newval);
+int ad1843_get_recsrc(struct snd_ad1843 *ad1843);
+void ad1843_set_resample_mode(struct snd_ad1843 *ad1843, int onoff);
+int ad1843_set_recsrc(struct snd_ad1843 *ad1843, int newsrc);
+int ad1843_get_outsrc(struct snd_ad1843 *ad1843);
+int ad1843_set_outsrc(struct snd_ad1843 *ad1843, int mask);
+void ad1843_setup_dac(struct snd_ad1843 *ad1843,
+ unsigned int id,
+ unsigned int framerate,
+ snd_pcm_format_t fmt,
+ unsigned int channels);
+void ad1843_shutdown_dac(struct snd_ad1843 *ad1843,
+ unsigned int id);
+void ad1843_setup_adc(struct snd_ad1843 *ad1843,
+ unsigned int framerate,
+ snd_pcm_format_t fmt,
+ unsigned int channels);
+void ad1843_shutdown_adc(struct snd_ad1843 *ad1843);
+int ad1843_init(struct snd_ad1843 *ad1843);
+
+#endif /* __SOUND_AD1843_H */
diff --git a/sound/mips/Kconfig b/sound/mips/Kconfig
index 531f8ba..a3e202e 100644
--- a/sound/mips/Kconfig
+++ b/sound/mips/Kconfig
@@ -11,5 +11,11 @@ config SND_AU1X00
help
ALSA Sound driver for the Au1x00's AC97 port.
+config SND_SGI_O2
+ tristate "SGI O2 Audio"
+ depends on SND && SGI_IP32
+ help
+ Sound support for the SGI O2 Workstation.
+
endmenu
diff --git a/sound/mips/Makefile b/sound/mips/Makefile
index 47afed9..55624d8 100644
--- a/sound/mips/Makefile
+++ b/sound/mips/Makefile
@@ -2,7 +2,9 @@
# Makefile for ALSA
#
+snd-sgi-o2-objs := sgio2audio.o ad1843.o
snd-au1x00-objs := au1x00.o
# Toplevel Module Dependency
obj-$(CONFIG_SND_AU1X00) += snd-au1x00.o
+obj-$(CONFIG_SND_SGI_O2) += snd-sgi-o2.o
diff --git a/sound/mips/ad1843.c b/sound/mips/ad1843.c
new file mode 100644
index 0000000..fca21d2
--- /dev/null
+++ b/sound/mips/ad1843.c
@@ -0,0 +1,644 @@
+/*
+ * AD1843 low level driver
+ *
+ * Copyright 2003 Vivien Chappelier <vivien.chappelier@xxxxxxxxxxxxxx>
+ *
+ * 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/slab.h>
+
+#include <linux/init.h>
+#include <linux/jiffies.h>
+#include <linux/sched.h>
+#include <linux/errno.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/initval.h>
+#include <sound/ad1843.h>
+
+/* TEMP: OSS stuff */
+#include <linux/soundcard.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_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 = { 9, 0, 6 }, /* Right DAC2 Analog/Digital Gain */
+ ad1843_RDA2GM = { 9, 7, 1 }, /* Right DAC2 Analog Mute */
+ ad1843_LDA2G = { 9, 8, 6 }, /* Left DAC2 Analog/Digital Gain */
+ ad1843_LDA2GM = { 9, 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 = { 11, 7, 1 }, /* Right DAC1 Digital Mute */
+ ad1843_LDA2AM = { 11, 15, 1 }, /* Left DAC1 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_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_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_gain ad1843_gain_RECLEV = {
+ 0, &ad1843_LIG, &ad1843_RIG
+};
+const struct ad1843_gain ad1843_gain_LINE = {
+ 1, &ad1843_LX1M, &ad1843_RX1M
+};
+const struct ad1843_gain ad1843_gain_CD = {
+ 1, &ad1843_LX2M, &ad1843_RX2M
+};
+const struct ad1843_gain ad1843_gain_MIC = {
+ 1, &ad1843_LMCM, &ad1843_RMCM
+};
+const struct ad1843_gain ad1843_gain_PCM_0 = {
+ 1, &ad1843_LDA1G, &ad1843_RDA1G
+};
+const struct ad1843_gain ad1843_gain_PCM_1 = {
+ 1, &ad1843_LDA2G, &ad1843_RDA2G
+};
+
+const struct ad1843_gain *ad1843_gain[AD1843_GAIN_SIZE] =
+{
+ &ad1843_gain_RECLEV,
+ &ad1843_gain_LINE,
+ &ad1843_gain_CD,
+ &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 = ad1843->read(ad1843->chip, field->reg);
+ int val = w >> field->lo_bit & ((1 << field->nbits) - 1);
+
+ return val;
+}
+
+/*
+ * 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 = ad1843->read(ad1843->chip, field->reg);
+ int mask = ((1 << field->nbits) - 1) << field->lo_bit;
+ int oldval = (w & mask) >> field->lo_bit;
+ int newbits = (newval << field->lo_bit) & mask;
+ w = (w & ~mask) | newbits;
+ (void) 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;
+
+ 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;
+ (void) ad1843->write(ad1843->chip, reg, w);
+}
+
+/*
+ * ad1843_get_gain reads the specified register and extracts the gain value
+ * using the supplied gain type. It returns the gain in OSS format.
+ */
+
+int ad1843_get_gain(struct snd_ad1843 *ad1843, int id)
+{
+ int lg, rg;
+ 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;
+ }
+ lg = (lg * 100 + (mask >> 1)) / mask;
+ rg = (rg * 100 + (mask >> 1)) / mask;
+ return lg << 0 | rg << 8;
+}
+
+/*
+ * Set an audio channel's gain. Converts from OSS format to AD1843's
+ * format.
+ *
+ * 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 & 0xFF;
+ int rg = newval >> 8;
+ if (lg < 0 || lg > 100 || rg < 0 || rg > 100)
+ return -EINVAL;
+ lg = (lg * mask + (mask >> 1)) / 100;
+ rg = (rg * mask + (mask >> 1)) / 100;
+ if (gp->negative) {
+ lg = mask - lg;
+ rg = mask - rg;
+ }
+ ad1843_write_multi(ad1843, 2, gp->lfield, lg, gp->rfield, rg);
+ return ad1843_get_gain(ad1843, id);
+}
+
+/* Returns the current recording source, in OSS format. */
+
+int ad1843_get_recsrc(struct snd_ad1843 *ad1843)
+{
+ int ls = ad1843_read_bits(ad1843, &ad1843_LSS);
+
+ switch (ls) {
+ case 1:
+ return SOUND_MASK_MIC;
+ case 2:
+ return SOUND_MASK_LINE;
+ case 3:
+ return SOUND_MASK_CD;
+ case 6:
+ return SOUND_MASK_PCM;
+ default:
+ return -1;
+ }
+}
+
+/*
+ * Enable/disable digital resample mode in the AD1843.
+ *
+ * The AD1843 requires that ADL, ADR, DA1 and DA2 be powered down
+ * while switching modes. So we save DA's state, power them down,
+ * switch into/out of resample mode, power them up, and restore state.
+ *
+ * This will cause audible glitches if D/A or A/D is going on, so the
+ * driver disallows that (in mixer_write_ioctl()).
+ *
+ * The open question is, is this worth doing? I'm leaving it in,
+ * because it's written, but...
+ */
+
+void ad1843_set_resample_mode(struct snd_ad1843 *ad1843, int onoff)
+{
+ /* Save DA's mute and gain (addr 9/10 is analog gain/attenuation) */
+ int save_da1 = ad1843->read(ad1843->chip, 9);
+ int save_da2 = ad1843->read(ad1843->chip, 10);
+
+ /* Power down A/D and D/A. */
+ ad1843_write_multi(ad1843, 4,
+ &ad1843_DA1EN, 0,
+ &ad1843_DA2EN, 0,
+ &ad1843_ADLEN, 0,
+ &ad1843_ADREN, 0);
+
+ /* Switch mode */
+ ad1843_write_bits(ad1843, &ad1843_DRSFLT, onoff);
+
+ /* Power up A/D and D/A. */
+ ad1843_write_multi(ad1843, 3,
+ &ad1843_DA1EN, 1,
+ &ad1843_DA2EN, 1,
+ &ad1843_ADLEN, 1,
+ &ad1843_ADREN, 1);
+
+ /* Restore DA's mute and gain. */
+ ad1843->write(ad1843->chip, 9, save_da1);
+ ad1843->write(ad1843->chip, 10, save_da2);
+}
+
+/*
+ * Set recording source. Arg newsrc specifies an OSS channel mask.
+ *
+ * The complication is that when we switch into/out of loopback mode
+ * (i.e., src = SOUND_MASK_PCM), we change the AD1843 into/out of
+ * digital resampling mode.
+ *
+ * Returns newsrc on success, -errno on failure.
+ */
+
+int ad1843_set_recsrc(struct snd_ad1843 *ad1843, int newsrc)
+{
+ int bits;
+ int oldbits;
+
+ switch (newsrc) {
+ case SOUND_MASK_PCM:
+ bits = 6;
+ break;
+
+ case SOUND_MASK_MIC:
+ bits = 1;
+ break;
+
+ case SOUND_MASK_LINE:
+ bits = 2;
+ break;
+
+ case SOUND_MASK_CD:
+ bits = 3;
+ break;
+
+ default:
+ return -EINVAL;
+ }
+ oldbits = ad1843_read_bits(ad1843, &ad1843_LSS);
+ if (newsrc == SOUND_MASK_PCM && oldbits != 6) {
+
+ ad1843_set_resample_mode(ad1843, 1);
+ ad1843_write_multi(ad1843, 2,
+ &ad1843_DAADL, 2,
+ &ad1843_DAADR, 2);
+ } else if (newsrc != SOUND_MASK_PCM && oldbits == 6) {
+
+ ad1843_set_resample_mode(ad1843, 0);
+ ad1843_write_multi(ad1843, 2,
+ &ad1843_DAADL, 0,
+ &ad1843_DAADR, 0);
+ }
+ ad1843_write_multi(ad1843, 2, &ad1843_LSS, bits, &ad1843_RSS, bits);
+ return newsrc;
+}
+
+/*
+ * Return current output sources, in OSS format.
+ */
+
+int ad1843_get_outsrc(struct snd_ad1843 *ad1843)
+{
+ int pcm, line, mic, cd;
+
+ pcm = ad1843_read_bits(ad1843, &ad1843_LDA1GM) ? 0 : SOUND_MASK_PCM;
+ line = ad1843_read_bits(ad1843, &ad1843_LX1MM) ? 0 : SOUND_MASK_LINE;
+ cd = ad1843_read_bits(ad1843, &ad1843_LX2MM) ? 0 : SOUND_MASK_CD;
+ mic = ad1843_read_bits(ad1843, &ad1843_LMCMM) ? 0 : SOUND_MASK_MIC;
+
+ return pcm | line | cd | mic;
+}
+
+/*
+ * Set output sources. Arg is a mask of active sources in OSS format.
+ *
+ * Returns source mask on success, -errno on failure.
+ */
+
+int ad1843_set_outsrc(struct snd_ad1843 *ad1843, int mask)
+{
+ int pcm, line, mic, cd;
+
+ if (mask & ~(SOUND_MASK_PCM | SOUND_MASK_LINE |
+ SOUND_MASK_CD | SOUND_MASK_MIC))
+ return -EINVAL;
+ pcm = (mask & SOUND_MASK_PCM) ? 0 : 1;
+ line = (mask & SOUND_MASK_LINE) ? 0 : 1;
+ mic = (mask & SOUND_MASK_MIC) ? 0 : 1;
+ cd = (mask & SOUND_MASK_CD) ? 0 : 1;
+
+ ad1843_write_multi(ad1843, 2, &ad1843_LDA1GM, pcm, &ad1843_RDA1GM, pcm);
+ ad1843_write_multi(ad1843, 2, &ad1843_LX1MM, line, &ad1843_RX1MM, line);
+ ad1843_write_multi(ad1843, 2, &ad1843_LX2MM, cd, &ad1843_RX2MM, cd);
+ ad1843_write_multi(ad1843, 2, &ad1843_LMCMM, mic, &ad1843_RMCMM, mic);
+
+ return mask;
+}
+
+/* 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 + HZ / 2; /* roughly half a second */
+
+ 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();
+ }
+
+ /* 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, 3,
+ &ad1843_DA1C, 1,
+ &ad1843_DA1C, 2,
+ &ad1843_ADLC, 3,
+ &ad1843_ADRC, 3);
+
+ /* 7. Enable conversion resources. */
+ ad1843_write_bits(ad1843, &ad1843_ADTLK, 1);
+ ad1843_write_multi(ad1843, 5,
+ &ad1843_ANAEN, 1,
+ &ad1843_AAMEN, 1,
+ &ad1843_DA1EN, 1,
+ &ad1843_DA2EN, 1,
+ &ad1843_ADLEN, 1,
+ &ad1843_ADREN, 1);
+
+ /* 8. Configure conversion resources while they are enabled. */
+ ad1843_write_bits(ad1843, &ad1843_DA1C, 1);
+ ad1843_write_bits(ad1843, &ad1843_DA2C, 1);
+
+ /* Unmute all channels. */
+
+ ad1843_set_outsrc(ad1843,
+ (SOUND_MASK_PCM | SOUND_MASK_LINE |
+ SOUND_MASK_MIC | SOUND_MASK_CD));
+ ad1843_write_multi(ad1843, 4,
+ &ad1843_LDA1AM, 0,
+ &ad1843_RDA1AM, 0,
+ &ad1843_LDA2AM, 0,
+ &ad1843_RDA2AM, 0);
+
+ /* Set default recording source to Line In and set
+ * mic gain to +20 dB.
+ */
+ ad1843_set_recsrc(ad1843, SOUND_MASK_LINE);
+ 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/sgio2audio.c b/sound/mips/sgio2audio.c
new file mode 100644
index 0000000..dea229b
--- /dev/null
+++ b/sound/mips/sgio2audio.c
@@ -0,0 +1,714 @@
+/*
+ * Sound driver for Silicon Graphics O2 Workstations A/V board audio.
+ *
+ * Copyright 2003 Vivien Chappelier <vivien.chappelier@xxxxxxxxxxxxxx>
+ *
+ * INCOMPLETE:
+ * - finish PCM,
+ * . finish/test 2nd DAC
+ * . recording
+ * - mixer and controls
+ * - /proc interface
+ *
+ * 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/gfp.h>
+#include <linux/vmalloc.h>
+#include <linux/interrupt.h>
+#include <linux/dma-mapping.h>
+#include <linux/platform_device.h>
+#include <linux/io.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@xxxxxxxxxxxxxx>");
+MODULE_DESCRIPTION("SGI O2 Audio");
+MODULE_LICENSE("GPL");
+MODULE_SUPPORTED_DEVICE("{{Silicon Graphics, O2 Audio}}");
+
+
+#define SGIO2AUDIO_MAX_VOLUME 1000
+
+#define AUDIO_CONTROL_RESET BIT(0) /* 1: reset audio interface */
+#define AUDIO_CONTROL_CODEC_PRESENT BIT(1) /* 1: codec detected */
+/* 2-8 : channel 1 write ptr alias */
+/* 9-15 : channel 2 read ptr alias */
+/* 16-22 : channel 3 read ptr alias */
+#define AUDIO_CONTROL_VOLUME_BUTTON_UP BIT(23) /* latched volume button */
+#define AUDIO_CONTROL_VOLUME_BUTTON_DOWN BIT(24) /* latched volume button */
+
+#define CODEC_CONTROL_WORD_SHIFT 0
+#define CODEC_CONTROL_WORD_MASK ((1 << 16) - 1)
+#define CODEC_CONTROL_READ BIT(16)
+#define CODEC_CONTROL_ADDRESS_SHIFT 17
+#define CODEC_CONTROL_ADDRESS_MASK ((1 << 7) - 1)
+
+#define CHANNEL_PTR_SHIFT 5
+#define CHANNEL_PTR_MASK ((1 << 6) - 1)
+#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_IN_BASE (0 << 12)
+#define CHANNEL_OUT1_BASE (1 << 12)
+#define CHANNEL_OUT2_BASE (2 << 12)
+
+#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
+
+/* definition of the chip-specific record */
+struct snd_sgio2audio {
+ struct snd_card *card;
+
+ struct snd_pcm *pcm;
+
+ /* codec */
+ struct snd_ad1843 *ad1843;
+ spinlock_t ad1843_lock;
+
+ /* channels */
+ struct {
+ struct snd_pcm_substream *substream;
+ void *buffer;
+ unsigned int pos;
+ snd_pcm_uframes_t size;
+ spinlock_t lock;
+ } channel[3];
+
+ /* properties */
+ int volume;
+ snd_pcm_format_t format;
+
+ /* resources */
+ void *ring_base;
+ dma_addr_t ring_base_dma;
+ int irq_start;
+ int irq_end;
+};
+
+/* PCM hardware definition */
+static struct snd_pcm_hardware snd_sgio2audio_playback_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,
+};
+
+/* 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 struct snd_ad1843 ad1843_ops = {
+ NULL, /* initialized in snd_sgio2_audio_create */
+ read_ad1843_reg,
+ write_ad1843_reg,
+};
+
+/* low-level audio interface DMA */
+
+/* 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;
+ unsigned long src_base, src_pos, src_mask;
+ 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 = (unsigned long) readq(&mace->perif.audio.chan[ch].write_ptr);
+ src_base = (unsigned long) chip->channel[ch].buffer;
+ src_pos = (unsigned long) 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);
+ unsigned int index = 1 + substream->number;
+
+ /* reset DMA channel */
+ writeq(CHANNEL_CONTROL_RESET, &mace->perif.audio.chan[index].control);
+ udelay(10);
+ writeq(0, &mace->perif.audio.chan[index].control);
+
+ /* push a full buffer */
+ snd_sgio2audio_dma_push_frag(chip, index, 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[index].control);
+ return 0;
+}
+
+static int snd_sgio2audio_dma_stop(struct snd_pcm_substream *substream)
+{
+ unsigned int index = 1 + substream->number;
+
+ writeq(0, &mace->perif.audio.chan[index].control);
+ return 0;
+}
+
+static void snd_sgio2audio_dma_interrupt(struct snd_sgio2audio *chip, int ch)
+{
+ struct snd_pcm_substream *substream = chip->channel[ch].substream;
+ int count;
+
+ /* 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);
+}
+
+static irqreturn_t snd_sgio2audio_interrupt(int irq, void *dev_id)
+{
+ struct snd_sgio2audio *chip = dev_id;
+ unsigned long status;
+
+ switch (irq) {
+ /* volume changed */
+ case MACEISA_AUDIO_SC_IRQ:
+ status = readq(&mace->perif.audio.control);
+ if (status & AUDIO_CONTROL_VOLUME_BUTTON_UP) {
+ status &= ~AUDIO_CONTROL_VOLUME_BUTTON_UP;
+ writeq(status, &mace->perif.audio.control);
+ if (chip->volume < SGIO2AUDIO_MAX_VOLUME)
+ chip->volume++;
+ }
+ if (status & AUDIO_CONTROL_VOLUME_BUTTON_DOWN) {
+ status &= ~AUDIO_CONTROL_VOLUME_BUTTON_DOWN;
+ writeq(status, &mace->perif.audio.control);
+ if (chip->volume > 0)
+ chip->volume--;
+ }
+ /* program AD1843 with the new volume */
+ ad1843_set_gain(chip->ad1843, AD1843_GAIN_PCM_0,
+ chip->volume / 10);
+ ad1843_set_gain(chip->ad1843, AD1843_GAIN_PCM_1,
+ chip->volume / 10);
+ break;
+
+ /* dma ring ready */
+ case MACEISA_AUDIO1_DMAT_IRQ:
+ snd_sgio2audio_dma_interrupt(chip, 0);
+ break;
+ case MACEISA_AUDIO2_DMAT_IRQ:
+ snd_sgio2audio_dma_interrupt(chip, 1);
+ break;
+ case MACEISA_AUDIO3_DMAT_IRQ:
+ snd_sgio2audio_dma_interrupt(chip, 2);
+ break;
+ default:
+ printk(KERN_ERR "sgio2audio: unhandled interrupt %d\n", irq);
+ /* TEMP : stop DMA */
+ writeq(0, &mace->perif.audio.chan[0].control);
+ writeq(0, &mace->perif.audio.chan[1].control);
+ writeq(0, &mace->perif.audio.chan[2].control);
+ break;
+ }
+ return IRQ_HANDLED;
+}
+
+/* PCM part */
+
+/* PCM playback open callback */
+static int snd_sgio2audio_playback_open(struct snd_pcm_substream *substream)
+{
+ struct snd_pcm_runtime *runtime = substream->runtime;
+
+ runtime->hw = snd_sgio2audio_playback_hw;
+ return 0;
+}
+
+/* PCM playback close callback */
+static int snd_sgio2audio_playback_close(struct snd_pcm_substream *substream)
+{
+ /* TODO: hardware code */
+ 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_playback_hw;
+
+ /* initialize AD1843 */
+ ad1843_init(chip->ad1843);
+
+ return 0;
+}
+
+/* PCM capture close callback */
+static int snd_sgio2audio_capture_close(struct snd_pcm_substream *substream)
+{
+ return 0;
+}
+
+
+/* hw_params callback */
+static int snd_sgio2audio_pcm_hw_params(struct snd_pcm_substream *substream,
+ struct snd_pcm_hw_params *hw_params)
+{
+ /* alloc virtual 'dma' area */
+ if (substream->runtime->dma_area)
+ vfree(substream->runtime->dma_area);
+ substream->runtime->dma_area = vmalloc(params_buffer_bytes(hw_params));
+ if (substream->runtime->dma_area == NULL)
+ return -ENOMEM;
+ return 0;
+}
+
+/* hw_free callback */
+static int snd_sgio2audio_pcm_hw_free(struct snd_pcm_substream *substream)
+{
+ if (substream->runtime->dma_area)
+ vfree(substream->runtime->dma_area);
+ substream->runtime->dma_area = NULL;
+ return 0;
+}
+
+/* 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;
+ unsigned int index = 1 + substream->number;
+ unsigned long flags;
+
+ spin_lock_irqsave(&chip->channel[index].lock, flags);
+
+ /* Setup the pseudo-dma transfer pointers. */
+ chip->channel[index].buffer = runtime->dma_area;
+ chip->channel[index].pos = 0;
+ chip->channel[index].size = 0;
+ chip->channel[index].substream = substream;
+
+ /* set AD1843 format */
+ /* hardware format is always S16_LE */
+ if (index) {
+ /* playback */
+ ad1843_setup_dac(chip->ad1843,
+ index - 1,
+ runtime->rate,
+ SNDRV_PCM_FORMAT_S16_LE,
+ runtime->channels);
+ } else {
+ /* capture */
+ ad1843_setup_adc(chip->ad1843,
+ runtime->rate,
+ SNDRV_PCM_FORMAT_S16_LE,
+ runtime->channels);
+ }
+
+ spin_unlock_irqrestore(&chip->channel[index].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);
+ unsigned int index = 1 + substream->number;
+ snd_pcm_uframes_t current_ptr;
+
+ /* get the current hardware pointer */
+ current_ptr = bytes_to_frames(substream->runtime,
+ chip->channel[index].pos);
+
+ return current_ptr;
+}
+
+/* get the physical page pointer on the given offset */
+static struct page *snd_sgio2audio_page(struct snd_pcm_substream *substream,
+ unsigned long offset)
+{
+ return vmalloc_to_page(substream->runtime->dma_area + offset);
+}
+
+/* operators */
+static struct snd_pcm_ops snd_sgio2audio_playback_ops = {
+ .open = snd_sgio2audio_playback_open,
+ .close = snd_sgio2audio_playback_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_sgio2audio_page,
+};
+
+static struct snd_pcm_ops snd_sgio2audio_capture_ops = {
+ .open = snd_sgio2audio_capture_open,
+ .close = snd_sgio2audio_capture_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_sgio2audio_page,
+};
+
+/*
+ * definitions of capture are omitted here...
+ */
+
+/* create a pcm device */
+static int __devinit snd_sgio2audio_new_pcm(struct snd_sgio2audio *chip)
+{
+ struct snd_pcm *pcm;
+ int err;
+
+ /* create a new pcm device with two outputs and one input */
+ err = snd_pcm_new(chip->card, "SGI O2 Audio", 0, 2, 1, &pcm);
+ if (err < 0)
+ return err;
+
+ pcm->private_data = chip;
+ strcpy(pcm->name, "SGI O2 Audio");
+ chip->pcm = pcm;
+
+ /* set operators */
+ snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK,
+ &snd_sgio2audio_playback_ops);
+ snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE,
+ &snd_sgio2audio_capture_ops);
+ return 0;
+}
+
+/* ALSA driver */
+
+static int snd_sgio2audio_free(struct snd_sgio2audio *chip)
+{
+ int irq;
+
+ /* reset interface */
+ writeq(AUDIO_CONTROL_RESET, &mace->perif.audio.control);
+ udelay(1);
+ writeq(0, &mace->perif.audio.control);
+
+ /* release IRQ's */
+ for (irq = MACEISA_AUDIO_SW_IRQ; irq <= MACEISA_AUDIO3_MERR_IRQ; irq++)
+ free_irq(irq, chip);
+
+ 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;
+ /* TODO: component destructor */
+ return snd_sgio2audio_free(chip);
+}
+
+static struct snd_device_ops ops = {
+ .dev_free = snd_sgio2audio_dev_free,
+};
+
+static int __devinit snd_sgio2audio_create(struct snd_card *card,
+ struct snd_sgio2audio **rchip)
+{
+ struct snd_sgio2audio *chip;
+ int i, err, irq;
+
+ *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->irq_start = MACEISA_AUDIO_SW_IRQ;
+ chip->irq_end = MACEISA_AUDIO3_MERR_IRQ;
+
+ 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);
+
+ chip->volume = SGIO2AUDIO_MAX_VOLUME;
+
+ /* initialize channels */
+ for (i = 0; i < 3; i++) {
+ spin_lock_init(&chip->channel[i].lock);
+ chip->channel[i].substream = NULL;
+ chip->channel[i].buffer = NULL;
+ chip->channel[i].pos = 0;
+ chip->channel[i].size = 0;
+ }
+
+ /* allocate IRQs */
+ for (irq = chip->irq_start; irq <= chip->irq_end; irq++) {
+ if (request_irq(irq, snd_sgio2audio_interrupt,
+ IRQF_SHARED, "SGI O2 Audio",
+ (void *)chip)) {
+ snd_sgio2audio_free(chip);
+ printk(KERN_ERR
+ "sgio2audio: cannot allocate irq %d\n", irq);
+ return -EBUSY;
+ }
+ }
+
+ /* reset the interface */
+ writeq(AUDIO_CONTROL_RESET, &mace->perif.audio.control);
+ udelay(1);
+ writeq(0, &mace->perif.audio.control);
+
+ /* set ring base */
+ writeq(chip->ring_base_dma, &mace->perif.ctrl.ringbase);
+
+ /* attach the AD1843 codec */
+ chip->ad1843 = &ad1843_ops;
+ chip->ad1843->chip = (void *) chip;
+
+ /* initialize the AD1843 codec */
+ err = ad1843_init(chip->ad1843);
+ if (err < 0) {
+ snd_sgio2audio_free(chip);
+ return err;
+ }
+
+ err = snd_device_new(card, SNDRV_DEV_LOWLEVEL, chip, &ops);
+ if (err < 0) {
+ snd_sgio2audio_free(chip);
+ return err;
+ }
+ *rchip = chip;
+ return 0;
+}
+
+static int __devinit snd_sgio2audio_probe(struct platform_device *pdev)
+{
+ struct snd_card *card;
+ struct snd_sgio2audio *chip;
+ int err;
+
+ card = snd_card_new(0, 0, THIS_MODULE, 0);
+ if (card == NULL)
+ return -ENOMEM;
+
+ err = snd_sgio2audio_create(card, &chip);
+ if (err < 0) {
+ snd_card_free(card);
+ return err;
+ }
+
+ /* TODO : finish PCM, do mixer and /proc */
+ err = snd_sgio2audio_new_pcm(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,
+ chip->irq_start, chip->irq_end);
+
+ err = snd_card_register(card);
+ if (err < 0) {
+ snd_card_free(card);
+ return err;
+ }
+ platform_set_drvdata(pdev, card);
+ return 0;
+}
+
+static int __exit snd_sgio2audio_remove(struct platform_device *pdev)
+{
+ struct snd_card *card = platform_get_drvdata(pdev);
+
+ snd_card_free(card);
+ platform_set_drvdata(pdev, NULL);
+ return 0;
+}
+
+static struct platform_driver sgio2audio_driver = {
+ .probe = snd_sgio2audio_probe,
+ .remove = __devexit_p(snd_sgio2audio_remove),
+ .driver = {
+ .name = "sgio2audio",
+ .owner = THIS_MODULE,
+ }
+};
+
+static int __init alsa_card_sgio2audio_init(void)
+{
+ return platform_driver_register(&sgio2audio_driver);
+}
+
+static void __exit alsa_card_sgio2audio_exit(void)
+{
+ platform_driver_unregister(&sgio2audio_driver);
+}
+
+module_init(alsa_card_sgio2audio_init)
+module_exit(alsa_card_sgio2audio_exit)
--
Crap can work. Given enough thrust pigs will fly, but it's not necessary a
good idea. [ RFC1925, 2.3 ]
