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Commit f3d9478b authored by Johannes Berg's avatar Johannes Berg Committed by Jaroslav Kysela
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[ALSA] snd-aoa: add snd-aoa 

This large patch adds all of snd-aoa.
Consisting of many modules, it currently replaces snd-powermac
for all layout-id based machines and handles many more (for
example new powerbooks and powermacs with digital output that
previously couldn't be used at all).
It also has support for all layout-IDs that Apple has (judging
from their Info.plist file) but not all are tested.
The driver currently has 2 known regressions over snd-powermac:
 * it doesn't handle powermac 7,2 and 7,3
 * it doesn't have a DRC control on snapper-based machines
I will fix those during the 2.6.18 development cycle.
Signed-off-by: default avatarJohannes Berg <johannes@sipsolutions.net>
Signed-off-by: default avatarTakashi Iwai <tiwai@suse.de>
parent 41f0cd3a
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sound/Kconfig
View file @ f3d9478b
......@@ -58,6 +58,8 @@ source "sound/pci/Kconfig"
source "sound/ppc/Kconfig"
source "sound/aoa/Kconfig"
source "sound/arm/Kconfig"
source "sound/mips/Kconfig"
......
sound/Makefile
View file @ f3d9478b
......@@ -4,7 +4,7 @@
obj-$(CONFIG_SOUND) += soundcore.o
obj-$(CONFIG_SOUND_PRIME) += oss/
obj-$(CONFIG_DMASOUND) += oss/
obj-$(CONFIG_SND) += core/ i2c/ drivers/ isa/ pci/ ppc/ arm/ synth/ usb/ sparc/ parisc/ pcmcia/ mips/
obj-$(CONFIG_SND) += core/ i2c/ drivers/ isa/ pci/ ppc/ arm/ synth/ usb/ sparc/ parisc/ pcmcia/ mips/ aoa/
ifeq ($(CONFIG_SND),y)
obj-y += last.o
......
sound/aoa/Kconfig 0 → 100644
View file @ f3d9478b
menu "Apple Onboard Audio driver"
depends on SND != n && PPC
config SND_AOA
tristate "Apple Onboard Audio driver"
depends on SOUND && SND_PCM
---help---
This option enables the new driver for the various
Apple Onboard Audio components.
source "sound/aoa/fabrics/Kconfig"
source "sound/aoa/codecs/Kconfig"
source "sound/aoa/soundbus/Kconfig"
endmenu
sound/aoa/Makefile 0 → 100644
View file @ f3d9478b
obj-$(CONFIG_SND_AOA) += core/
obj-$(CONFIG_SND_AOA) += codecs/
obj-$(CONFIG_SND_AOA) += fabrics/
obj-$(CONFIG_SND_AOA_SOUNDBUS) += soundbus/
sound/aoa/aoa-gpio.h 0 → 100644
View file @ f3d9478b
/*
* Apple Onboard Audio GPIO definitions
*
* Copyright 2006 Johannes Berg <johannes@sipsolutions.net>
*
* GPL v2, can be found in COPYING.
*/
#ifndef __AOA_GPIO_H
#define __AOA_GPIO_H
#include <linux/workqueue.h>
#include <linux/mutex.h>
#include <asm/prom.h>
typedef void (*notify_func_t)(void *data);
enum notify_type {
AOA_NOTIFY_HEADPHONE,
AOA_NOTIFY_LINE_IN,
AOA_NOTIFY_LINE_OUT,
};
struct gpio_runtime;
struct gpio_methods {
/* for initialisation/de-initialisation of the GPIO layer */
void (*init)(struct gpio_runtime *rt);
void (*exit)(struct gpio_runtime *rt);
/* turn off headphone, speakers, lineout */
void (*all_amps_off)(struct gpio_runtime *rt);
/* turn headphone, speakers, lineout back to previous setting */
void (*all_amps_restore)(struct gpio_runtime *rt);
void (*set_headphone)(struct gpio_runtime *rt, int on);
void (*set_speakers)(struct gpio_runtime *rt, int on);
void (*set_lineout)(struct gpio_runtime *rt, int on);
int (*get_headphone)(struct gpio_runtime *rt);
int (*get_speakers)(struct gpio_runtime *rt);
int (*get_lineout)(struct gpio_runtime *rt);
void (*set_hw_reset)(struct gpio_runtime *rt, int on);
/* use this to be notified of any events. The notification
* function is passed the data, and is called in process
* context by the use of schedule_work.
* The interface for it is that setting a function to NULL
* removes it, and they return 0 if the operation succeeded,
* and -EBUSY if the notification is already assigned by
* someone else. */
int (*set_notify)(struct gpio_runtime *rt,
enum notify_type type,
notify_func_t notify,
void *data);
/* returns 0 if not plugged in, 1 if plugged in
* or a negative error code */
int (*get_detect)(struct gpio_runtime *rt,
enum notify_type type);
};
struct gpio_notification {
notify_func_t notify;
void *data;
void *gpio_private;
struct work_struct work;
struct mutex mutex;
};
struct gpio_runtime {
/* to be assigned by fabric */
struct device_node *node;
/* since everyone needs this pointer anyway... */
struct gpio_methods *methods;
/* to be used by the gpio implementation */
int implementation_private;
struct gpio_notification headphone_notify;
struct gpio_notification line_in_notify;
struct gpio_notification line_out_notify;
};
#endif /* __AOA_GPIO_H */
sound/aoa/aoa.h 0 → 100644
View file @ f3d9478b
/*
* Apple Onboard Audio definitions
*
* Copyright 2006 Johannes Berg <johannes@sipsolutions.net>
*
* GPL v2, can be found in COPYING.
*/
#ifndef __AOA_H
#define __AOA_H
#include <asm/prom.h>
#include <linux/module.h>
/* So apparently there's a reason for requiring driver.h to be included first! */
#include <sound/driver.h>
#include <sound/core.h>
#include <sound/asound.h>
#include <sound/control.h>
#include "aoa-gpio.h"
#include "soundbus/soundbus.h"
#define MAX_CODEC_NAME_LEN 32
struct aoa_codec {
char name[MAX_CODEC_NAME_LEN];
struct module *owner;
/* called when the fabric wants to init this codec.
* Do alsa card manipulations from here. */
int (*init)(struct aoa_codec *codec);
/* called when the fabric is done with the codec.
* The alsa card will be cleaned up so don't bother. */
void (*exit)(struct aoa_codec *codec);
/* May be NULL, but can be used by the fabric.
* Refcounting is the codec driver's responsibility */
struct device_node *node;
/* assigned by fabric before init() is called, points
* to the soundbus device. Cannot be NULL. */
struct soundbus_dev *soundbus_dev;
/* assigned by the fabric before init() is called, points
* to the fabric's gpio runtime record for the relevant
* device. */
struct gpio_runtime *gpio;
/* assigned by the fabric before init() is called, contains
* a codec specific bitmask of what outputs and inputs are
* actually connected */
u32 connected;
/* data the fabric can associate with this structure */
void *fabric_data;
/* private! */
struct list_head list;
struct aoa_fabric *fabric;
};
/* return 0 on success */
extern int
aoa_codec_register(struct aoa_codec *codec);
extern void
aoa_codec_unregister(struct aoa_codec *codec);
#define MAX_LAYOUT_NAME_LEN 32
struct aoa_fabric {
char name[MAX_LAYOUT_NAME_LEN];
struct module *owner;
/* once codecs register, they are passed here after.
* They are of course not initialised, since the
* fabric is responsible for initialising some fields
* in the codec structure! */
int (*found_codec)(struct aoa_codec *codec);
/* called for each codec when it is removed,
* also in the case that aoa_fabric_unregister
* is called and all codecs are removed
* from this fabric.
* Also called if found_codec returned 0 but
* the codec couldn't initialise. */
void (*remove_codec)(struct aoa_codec *codec);
/* If found_codec returned 0, and the codec
* could be initialised, this is called. */
void (*attached_codec)(struct aoa_codec *codec);
};
/* return 0 on success, -EEXIST if another fabric is
* registered, -EALREADY if the same fabric is registered.
* Passing NULL can be used to test for the presence
* of another fabric, if -EALREADY is returned there is
* no other fabric present.
* In the case that the function returns -EALREADY
* and the fabric passed is not NULL, all codecs
* that are not assigned yet are passed to the fabric
* again for reconsideration. */
extern int
aoa_fabric_register(struct aoa_fabric *fabric);
/* it is vital to call this when the fabric exits!
* When calling, the remove_codec will be called
* for all codecs, unless it is NULL. */
extern void
aoa_fabric_unregister(struct aoa_fabric *fabric);
/* if for some reason you want to get rid of a codec
* before the fabric is removed, use this.
* Note that remove_codec is called for it! */
extern void
aoa_fabric_unlink_codec(struct aoa_codec *codec);
/* alsa help methods */
struct aoa_card {
struct snd_card *alsa_card;
};
extern int aoa_snd_device_new(snd_device_type_t type,
void * device_data, struct snd_device_ops * ops);
extern struct snd_card *aoa_get_card(void);
extern int aoa_snd_ctl_add(struct snd_kcontrol* control);
/* GPIO stuff */
extern struct gpio_methods *pmf_gpio_methods;
extern struct gpio_methods *ftr_gpio_methods;
/* extern struct gpio_methods *map_gpio_methods; */
#endif /* __AOA_H */
sound/aoa/codecs/Kconfig 0 → 100644
View file @ f3d9478b
config SND_AOA_ONYX
tristate "support Onyx chip"
depends on SND_AOA
---help---
This option enables support for the Onyx (pcm3052)
codec chip found in the latest Apple machines
(most of those with digital audio output).
#config SND_AOA_TOPAZ
# tristate "support Topaz chips"
# depends on SND_AOA
# ---help---
# This option enables support for the Topaz (CS84xx)
# codec chips found in the latest Apple machines,
# these chips do the digital input and output on
# some PowerMacs.
config SND_AOA_TAS
tristate "support TAS chips"
depends on SND_AOA
---help---
This option enables support for the tas chips
found in a lot of Apple Machines, especially
iBooks and PowerBooks without digital.
config SND_AOA_TOONIE
tristate "support Toonie chip"
depends on SND_AOA
---help---
This option enables support for the toonie codec
found in the Mac Mini. If you have a Mac Mini and
want to hear sound, select this option.
sound/aoa/codecs/Makefile 0 → 100644
View file @ f3d9478b
obj-$(CONFIG_SND_AOA_ONYX) += snd-aoa-codec-onyx.o
obj-$(CONFIG_SND_AOA_TAS) += snd-aoa-codec-tas.o
obj-$(CONFIG_SND_AOA_TOONIE) += snd-aoa-codec-toonie.o
sound/aoa/codecs/snd-aoa-codec-onyx.c 0 → 100644
View file @ f3d9478b
/*
* Apple Onboard Audio driver for Onyx codec
*
* Copyright 2006 Johannes Berg <johannes@sipsolutions.net>
*
* GPL v2, can be found in COPYING.
*
*
* This is a driver for the pcm3052 codec chip (codenamed Onyx)
* that is present in newer Apple hardware (with digital output).
*
* The Onyx codec has the following connections (listed by the bit
* to be used in aoa_codec.connected):
* 0: analog output
* 1: digital output
* 2: line input
* 3: microphone input
* Note that even though I know of no machine that has for example
* the digital output connected but not the analog, I have handled
* all the different cases in the code so that this driver may serve
* as a good example of what to do.
*
* NOTE: This driver assumes that there's at most one chip to be
* used with one alsa card, in form of creating all kinds
* of mixer elements without regard for their existence.
* But snd-aoa assumes that there's at most one card, so
* this means you can only have one onyx on a system. This
* should probably be fixed by changing the assumption of
* having just a single card on a system, and making the
* 'card' pointer accessible to anyone who needs it instead
* of hiding it in the aoa_snd_* functions...
*
*/
#include <linux/delay.h>
#include <linux/module.h>
MODULE_AUTHOR("Johannes Berg <johannes@sipsolutions.net>");
MODULE_LICENSE("GPL");
MODULE_DESCRIPTION("pcm3052 (onyx) codec driver for snd-aoa");
#include "snd-aoa-codec-onyx.h"
#include "../aoa.h"
#include "../soundbus/soundbus.h"
#define PFX "snd-aoa-codec-onyx: "
struct onyx {
/* cache registers 65 to 80, they are write-only! */
u8 cache[16];
struct i2c_client i2c;
struct aoa_codec codec;
u32 initialised:1,
spdif_locked:1,
analog_locked:1,
original_mute:2;
int open_count;
struct codec_info *codec_info;
/* mutex serializes concurrent access to the device
* and this structure.
*/
struct mutex mutex;
};
#define codec_to_onyx(c) container_of(c, struct onyx, codec)
/* both return 0 if all ok, else on error */
static int onyx_read_register(struct onyx *onyx, u8 reg, u8 *value)
{
s32 v;
if (reg != ONYX_REG_CONTROL) {
*value = onyx->cache[reg-FIRSTREGISTER];
return 0;
}
v = i2c_smbus_read_byte_data(&onyx->i2c, reg);
if (v < 0)
return -1;
*value = (u8)v;
onyx->cache[ONYX_REG_CONTROL-FIRSTREGISTER] = *value;
return 0;
}
static int onyx_write_register(struct onyx *onyx, u8 reg, u8 value)
{
int result;
result = i2c_smbus_write_byte_data(&onyx->i2c, reg, value);
if (!result)
onyx->cache[reg-FIRSTREGISTER] = value;
return result;
}
/* alsa stuff */
static int onyx_dev_register(struct snd_device *dev)
{
return 0;
}
static struct snd_device_ops ops = {
.dev_register = onyx_dev_register,
};
/* this is necessary because most alsa mixer programs
* can't properly handle the negative range */
#define VOLUME_RANGE_SHIFT 128
static int onyx_snd_vol_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 = -128 + VOLUME_RANGE_SHIFT;
uinfo->value.integer.max = -1 + VOLUME_RANGE_SHIFT;
return 0;
}
static int onyx_snd_vol_get(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
struct onyx *onyx = snd_kcontrol_chip(kcontrol);
s8 l, r;
mutex_lock(&onyx->mutex);
onyx_read_register(onyx, ONYX_REG_DAC_ATTEN_LEFT, &l);
onyx_read_register(onyx, ONYX_REG_DAC_ATTEN_RIGHT, &r);
mutex_unlock(&onyx->mutex);
ucontrol->value.integer.value[0] = l + VOLUME_RANGE_SHIFT;
ucontrol->value.integer.value[1] = r + VOLUME_RANGE_SHIFT;
return 0;
}
static int onyx_snd_vol_put(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
struct onyx *onyx = snd_kcontrol_chip(kcontrol);
s8 l, r;
mutex_lock(&onyx->mutex);
onyx_read_register(onyx, ONYX_REG_DAC_ATTEN_LEFT, &l);
onyx_read_register(onyx, ONYX_REG_DAC_ATTEN_RIGHT, &r);
if (l + VOLUME_RANGE_SHIFT == ucontrol->value.integer.value[0] &&
r + VOLUME_RANGE_SHIFT == ucontrol->value.integer.value[1]) {
mutex_unlock(&onyx->mutex);
return 0;
}
onyx_write_register(onyx, ONYX_REG_DAC_ATTEN_LEFT,
ucontrol->value.integer.value[0]
- VOLUME_RANGE_SHIFT);
onyx_write_register(onyx, ONYX_REG_DAC_ATTEN_RIGHT,
ucontrol->value.integer.value[1]
- VOLUME_RANGE_SHIFT);
mutex_unlock(&onyx->mutex);
return 1;
}
static struct snd_kcontrol_new volume_control = {
.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
.name = "Master Playback Volume",
.access = SNDRV_CTL_ELEM_ACCESS_READWRITE,
.info = onyx_snd_vol_info,
.get = onyx_snd_vol_get,
.put = onyx_snd_vol_put,
};
/* like above, this is necessary because a lot
* of alsa mixer programs don't handle ranges
* that don't start at 0 properly.
* even alsamixer is one of them... */
#define INPUTGAIN_RANGE_SHIFT (-3)
static int onyx_snd_inputgain_info(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_info *uinfo)
{
uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
uinfo->count = 1;
uinfo->value.integer.min = 3 + INPUTGAIN_RANGE_SHIFT;
uinfo->value.integer.max = 28 + INPUTGAIN_RANGE_SHIFT;
return 0;
}
static int onyx_snd_inputgain_get(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
struct onyx *onyx = snd_kcontrol_chip(kcontrol);
u8 ig;
mutex_lock(&onyx->mutex);
onyx_read_register(onyx, ONYX_REG_ADC_CONTROL, &ig);
mutex_unlock(&onyx->mutex);
ucontrol->value.integer.value[0] =
(ig & ONYX_ADC_PGA_GAIN_MASK) + INPUTGAIN_RANGE_SHIFT;
return 0;
}
static int onyx_snd_inputgain_put(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
struct onyx *onyx = snd_kcontrol_chip(kcontrol);
u8 v, n;
mutex_lock(&onyx->mutex);
onyx_read_register(onyx, ONYX_REG_ADC_CONTROL, &v);
n = v;
n &= ~ONYX_ADC_PGA_GAIN_MASK;
n |= (ucontrol->value.integer.value[0] - INPUTGAIN_RANGE_SHIFT)
& ONYX_ADC_PGA_GAIN_MASK;
onyx_write_register(onyx, ONYX_REG_ADC_CONTROL, n);
mutex_unlock(&onyx->mutex);
return n != v;
}
static struct snd_kcontrol_new inputgain_control = {
.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
.name = "Master Capture Volume",
.access = SNDRV_CTL_ELEM_ACCESS_READWRITE,
.info = onyx_snd_inputgain_info,
.get = onyx_snd_inputgain_get,
.put = onyx_snd_inputgain_put,
};
static int onyx_snd_capture_source_info(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_info *uinfo)
{
static char *texts[] = { "Line-In", "Microphone" };
uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
uinfo->count = 1;
uinfo->value.enumerated.items = 2;
if (uinfo->value.enumerated.item > 1)
uinfo->value.enumerated.item = 1;
strcpy(uinfo->value.enumerated.name, texts[uinfo->value.enumerated.item]);
return 0;
}
static int onyx_snd_capture_source_get(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
struct onyx *onyx = snd_kcontrol_chip(kcontrol);
s8 v;
mutex_lock(&onyx->mutex);
onyx_read_register(onyx, ONYX_REG_ADC_CONTROL, &v);
mutex_unlock(&onyx->mutex);
ucontrol->value.enumerated.item[0] = !!(v&ONYX_ADC_INPUT_MIC);
return 0;
}
static void onyx_set_capture_source(struct onyx *onyx, int mic)
{
s8 v;
mutex_lock(&onyx->mutex);
onyx_read_register(onyx, ONYX_REG_ADC_CONTROL, &v);
v &= ~ONYX_ADC_INPUT_MIC;
if (mic)
v |= ONYX_ADC_INPUT_MIC;
onyx_write_register(onyx, ONYX_REG_ADC_CONTROL, v);
mutex_unlock(&onyx->mutex);
}
static int onyx_snd_capture_source_put(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
onyx_set_capture_source(snd_kcontrol_chip(kcontrol),
ucontrol->value.enumerated.item[0]);
return 1;
}
static struct snd_kcontrol_new capture_source_control = {
.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
/* If we name this 'Input Source', it properly shows up in
* alsamixer as a selection, * but it's shown under the
* 'Playback' category.
* If I name it 'Capture Source', it shows up in strange
* ways (two bools of which one can be selected at a
* time) but at least it's shown in the 'Capture'
* category.
* I was told that this was due to backward compatibility,
* but I don't understand then why the mangling is *not*
* done when I name it "Input Source".....
*/
.name = "Capture Source",
.access = SNDRV_CTL_ELEM_ACCESS_READWRITE,
.info = onyx_snd_capture_source_info,
.get = onyx_snd_capture_source_get,
.put = onyx_snd_capture_source_put,
};
static int onyx_snd_mute_info(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_info *uinfo)
{
uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN;
uinfo->count = 2;
uinfo->value.integer.min = 0;
uinfo->value.integer.max = 1;
return 0;
}
static int onyx_snd_mute_get(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
struct onyx *onyx = snd_kcontrol_chip(kcontrol);
u8 c;
mutex_lock(&onyx->mutex);
onyx_read_register(onyx, ONYX_REG_DAC_CONTROL, &c);
mutex_unlock(&onyx->mutex);
ucontrol->value.integer.value[0] = !(c & ONYX_MUTE_LEFT);
ucontrol->value.integer.value[1] = !(c & ONYX_MUTE_RIGHT);
return 0;
}
static int onyx_snd_mute_put(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
struct onyx *onyx = snd_kcontrol_chip(kcontrol);
u8 v = 0, c = 0;
int err = -EBUSY;
mutex_lock(&onyx->mutex);
if (onyx->analog_locked)
goto out_unlock;
onyx_read_register(onyx, ONYX_REG_DAC_CONTROL, &v);
c = v;
c &= ~(ONYX_MUTE_RIGHT | ONYX_MUTE_LEFT);
if (!ucontrol->value.integer.value[0])
c |= ONYX_MUTE_LEFT;
if (!ucontrol->value.integer.value[1])
c |= ONYX_MUTE_RIGHT;
err = onyx_write_register(onyx, ONYX_REG_DAC_CONTROL, c);
out_unlock:
mutex_unlock(&onyx->mutex);
return !err ? (v != c) : err;
}
static struct snd_kcontrol_new mute_control = {
.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
.name = "Master Playback Switch",
.access = SNDRV_CTL_ELEM_ACCESS_READWRITE,
.info = onyx_snd_mute_info,
.get = onyx_snd_mute_get,
.put = onyx_snd_mute_put,
};
static int onyx_snd_single_bit_info(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_info *uinfo)
{
uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN;
uinfo->count = 1;
uinfo->value.integer.min = 0;
uinfo->value.integer.max = 1;
return 0;
}
#define FLAG_POLARITY_INVERT 1
#define FLAG_SPDIFLOCK 2
static int onyx_snd_single_bit_get(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
struct onyx *onyx = snd_kcontrol_chip(kcontrol);
u8 c;
long int pv = kcontrol->private_value;
u8 polarity = (pv >> 16) & FLAG_POLARITY_INVERT;
u8 address = (pv >> 8) & 0xff;
u8 mask = pv & 0xff;
mutex_lock(&onyx->mutex);
onyx_read_register(onyx, address, &c);
mutex_unlock(&onyx->mutex);
ucontrol->value.integer.value[0] = !!(c & mask) ^ polarity;
return 0;
}
static int onyx_snd_single_bit_put(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
struct onyx *onyx = snd_kcontrol_chip(kcontrol);
u8 v = 0, c = 0;
int err;
long int pv = kcontrol->private_value;
u8 polarity = (pv >> 16) & FLAG_POLARITY_INVERT;
u8 spdiflock = (pv >> 16) & FLAG_SPDIFLOCK;
u8 address = (pv >> 8) & 0xff;
u8 mask = pv & 0xff;
mutex_lock(&onyx->mutex);
if (spdiflock && onyx->spdif_locked) {
/* even if alsamixer doesn't care.. */
err = -EBUSY;
goto out_unlock;
}
onyx_read_register(onyx, address, &v);
c = v;
c &= ~(mask);
if (!!ucontrol->value.integer.value[0] ^ polarity)
c |= mask;
err = onyx_write_register(onyx, address, c);
out_unlock:
mutex_unlock(&onyx->mutex);
return !err ? (v != c) : err;
}
#define SINGLE_BIT(n, type, description, address, mask, flags) \
static struct snd_kcontrol_new n##_control = { \
.iface = SNDRV_CTL_ELEM_IFACE_##type, \
.name = description, \
.access = SNDRV_CTL_ELEM_ACCESS_READWRITE, \
.info = onyx_snd_single_bit_info, \
.get = onyx_snd_single_bit_get, \
.put = onyx_snd_single_bit_put, \
.private_value = (flags << 16) | (address << 8) | mask \
}
SINGLE_BIT(spdif,
MIXER,
SNDRV_CTL_NAME_IEC958("", PLAYBACK, SWITCH),
ONYX_REG_DIG_INFO4,
ONYX_SPDIF_ENABLE,
FLAG_SPDIFLOCK);
SINGLE_BIT(ovr1,
MIXER,
"Oversampling Rate",
ONYX_REG_DAC_CONTROL,
ONYX_OVR1,
0);
SINGLE_BIT(flt0,
MIXER,
"Fast Digital Filter Rolloff",
ONYX_REG_DAC_FILTER,
ONYX_ROLLOFF_FAST,
FLAG_POLARITY_INVERT);
SINGLE_BIT(hpf,
MIXER,
"Highpass Filter",
ONYX_REG_ADC_HPF_BYPASS,
ONYX_HPF_DISABLE,
FLAG_POLARITY_INVERT);
SINGLE_BIT(dm12,
MIXER,
"Digital De-Emphasis",
ONYX_REG_DAC_DEEMPH,
ONYX_DIGDEEMPH_CTRL,
0);
static int onyx_spdif_info(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_info *uinfo)
{
uinfo->type = SNDRV_CTL_ELEM_TYPE_IEC958;
uinfo->count = 1;
return 0;
}
static int onyx_spdif_mask_get(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
/* datasheet page 30, all others are 0 */
ucontrol->value.iec958.status[0] = 0x3e;
ucontrol->value.iec958.status[1] = 0xff;
ucontrol->value.iec958.status[3] = 0x3f;
ucontrol->value.iec958.status[4] = 0x0f;
return 0;
}
static struct snd_kcontrol_new onyx_spdif_mask = {
.access = SNDRV_CTL_ELEM_ACCESS_READ,
.iface = SNDRV_CTL_ELEM_IFACE_PCM,
.name = SNDRV_CTL_NAME_IEC958("",PLAYBACK,CON_MASK),
.info = onyx_spdif_info,
.get = onyx_spdif_mask_get,
};
static int onyx_spdif_get(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
struct onyx *onyx = snd_kcontrol_chip(kcontrol);
u8 v;
mutex_lock(&onyx->mutex);
onyx_read_register(onyx, ONYX_REG_DIG_INFO1, &v);
ucontrol->value.iec958.status[0] = v & 0x3e;
onyx_read_register(onyx, ONYX_REG_DIG_INFO2, &v);
ucontrol->value.iec958.status[1] = v;
onyx_read_register(onyx, ONYX_REG_DIG_INFO3, &v);
ucontrol->value.iec958.status[3] = v & 0x3f;
onyx_read_register(onyx, ONYX_REG_DIG_INFO4, &v);
ucontrol->value.iec958.status[4] = v & 0x0f;
mutex_unlock(&onyx->mutex);
return 0;
}
static int onyx_spdif_put(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
struct onyx *onyx = snd_kcontrol_chip(kcontrol);
u8 v;
mutex_lock(&onyx->mutex);
onyx_read_register(onyx, ONYX_REG_DIG_INFO1, &v);
v = (v & ~0x3e) | (ucontrol->value.iec958.status[0] & 0x3e);
onyx_write_register(onyx, ONYX_REG_DIG_INFO1, v);
v = ucontrol->value.iec958.status[1];
onyx_write_register(onyx, ONYX_REG_DIG_INFO2, v);
onyx_read_register(onyx, ONYX_REG_DIG_INFO3, &v);
v = (v & ~0x3f) | (ucontrol->value.iec958.status[3] & 0x3f);
onyx_write_register(onyx, ONYX_REG_DIG_INFO3, v);
onyx_read_register(onyx, ONYX_REG_DIG_INFO4, &v);
v = (v & ~0x0f) | (ucontrol->value.iec958.status[4] & 0x0f);
onyx_write_register(onyx, ONYX_REG_DIG_INFO4, v);
mutex_unlock(&onyx->mutex);
return 1;
}
static struct snd_kcontrol_new onyx_spdif_ctrl = {
.access = SNDRV_CTL_ELEM_ACCESS_READWRITE,
.iface = SNDRV_CTL_ELEM_IFACE_PCM,
.name = SNDRV_CTL_NAME_IEC958("",PLAYBACK,DEFAULT),
.info = onyx_spdif_info,
.get = onyx_spdif_get,
.put = onyx_spdif_put,
};
/* our registers */
static u8 register_map[] = {
ONYX_REG_DAC_ATTEN_LEFT,
ONYX_REG_DAC_ATTEN_RIGHT,
ONYX_REG_CONTROL,
ONYX_REG_DAC_CONTROL,
ONYX_REG_DAC_DEEMPH,
ONYX_REG_DAC_FILTER,
ONYX_REG_DAC_OUTPHASE,
ONYX_REG_ADC_CONTROL,
ONYX_REG_ADC_HPF_BYPASS,
ONYX_REG_DIG_INFO1,
ONYX_REG_DIG_INFO2,
ONYX_REG_DIG_INFO3,
ONYX_REG_DIG_INFO4
};
static u8 initial_values[ARRAY_SIZE(register_map)] = {
0x80, 0x80, /* muted */
ONYX_MRST | ONYX_SRST, /* but handled specially! */
ONYX_MUTE_LEFT | ONYX_MUTE_RIGHT,
0, /* no deemphasis */
ONYX_DAC_FILTER_ALWAYS,
ONYX_OUTPHASE_INVERTED,
(-1 /*dB*/ + 8) & 0xF, /* line in selected, -1 dB gain*/
ONYX_ADC_HPF_ALWAYS,
(1<<2), /* pcm audio */
2, /* category: pcm coder */
0, /* sampling frequency 44.1 kHz, clock accuracy level II */
1 /* 24 bit depth */
};
/* reset registers of chip, either to initial or to previous values */
static int onyx_register_init(struct onyx *onyx)
{
int i;
u8 val;
u8 regs[sizeof(initial_values)];
if (!onyx->initialised) {
memcpy(regs, initial_values, sizeof(initial_values));
if (onyx_read_register(onyx, ONYX_REG_CONTROL, &val))
return -1;
val &= ~ONYX_SILICONVERSION;
val |= initial_values[3];
regs[3] = val;
} else {
for (i=0; i<sizeof(register_map); i++)
regs[i] = onyx->cache[register_map[i]-FIRSTREGISTER];
}
for (i=0; i<sizeof(register_map); i++) {
if (onyx_write_register(onyx, register_map[i], regs[i]))
return -1;
}
onyx->initialised = 1;
return 0;
}
static struct transfer_info onyx_transfers[] = {
/* this is first so we can skip it if no input is present...
* No hardware exists with that, but it's here as an example
* of what to do :) */
{
/* analog input */
.formats = SNDRV_PCM_FMTBIT_S8 |
SNDRV_PCM_FMTBIT_S16_BE |
SNDRV_PCM_FMTBIT_S24_BE,
.rates = SNDRV_PCM_RATE_8000_96000,
.transfer_in = 1,
.must_be_clock_source = 0,
.tag = 0,
},
{
/* if analog and digital are currently off, anything should go,
* so this entry describes everything we can do... */
.formats = SNDRV_PCM_FMTBIT_S8 |
SNDRV_PCM_FMTBIT_S16_BE |
SNDRV_PCM_FMTBIT_S24_BE
#ifdef SNDRV_PCM_FMTBIT_COMPRESSED_16BE
| SNDRV_PCM_FMTBIT_COMPRESSED_16BE
#endif
,
.rates = SNDRV_PCM_RATE_8000_96000,
.tag = 0,
},
{
/* analog output */
.formats = SNDRV_PCM_FMTBIT_S8 |
SNDRV_PCM_FMTBIT_S16_BE |
SNDRV_PCM_FMTBIT_S24_BE,
.rates = SNDRV_PCM_RATE_8000_96000,
.transfer_in = 0,
.must_be_clock_source = 0,
.tag = 1,
},
{
/* digital pcm output, also possible for analog out */
.formats = SNDRV_PCM_FMTBIT_S8 |
SNDRV_PCM_FMTBIT_S16_BE |
SNDRV_PCM_FMTBIT_S24_BE,
.rates = SNDRV_PCM_RATE_32000 |
SNDRV_PCM_RATE_44100 |
SNDRV_PCM_RATE_48000,
.transfer_in = 0,
.must_be_clock_source = 0,
.tag = 2,
},
#ifdef SNDRV_PCM_FMTBIT_COMPRESSED_16BE
Once alsa gets supports for this kind of thing we can add it...
{
/* digital compressed output */
.formats = SNDRV_PCM_FMTBIT_COMPRESSED_16BE,
.rates = SNDRV_PCM_RATE_32000 |
SNDRV_PCM_RATE_44100 |
SNDRV_PCM_RATE_48000,
.tag = 2,
},
#endif
{}
};
static int onyx_usable(struct codec_info_item *cii,
struct transfer_info *ti,
struct transfer_info *out)
{
u8 v;
struct onyx *onyx = cii->codec_data;
int spdif_enabled, analog_enabled;
mutex_lock(&onyx->mutex);
onyx_read_register(onyx, ONYX_REG_DIG_INFO4, &v);
spdif_enabled = !!(v & ONYX_SPDIF_ENABLE);
onyx_read_register(onyx, ONYX_REG_DAC_CONTROL, &v);
analog_enabled =
(v & (ONYX_MUTE_RIGHT|ONYX_MUTE_LEFT))
!= (ONYX_MUTE_RIGHT|ONYX_MUTE_LEFT);
mutex_unlock(&onyx->mutex);
switch (ti->tag) {
case 0: return 1;
case 1: return analog_enabled;
case 2: return spdif_enabled;
}
return 1;
}
static int onyx_prepare(struct codec_info_item *cii,
struct bus_info *bi,
struct snd_pcm_substream *substream)
{
u8 v;
struct onyx *onyx = cii->codec_data;
int err = -EBUSY;
mutex_lock(&onyx->mutex);
#ifdef SNDRV_PCM_FMTBIT_COMPRESSED_16BE
if (substream->runtime->format == SNDRV_PCM_FMTBIT_COMPRESSED_16BE) {
/* mute and lock analog output */
onyx_read_register(onyx, ONYX_REG_DAC_CONTROL, &v);
if (onyx_write_register(onyx
ONYX_REG_DAC_CONTROL,
v | ONYX_MUTE_RIGHT | ONYX_MUTE_LEFT))
goto out_unlock;
onyx->analog_locked = 1;
err = 0;
goto out_unlock;
}
#endif
switch (substream->runtime->rate) {
case 32000:
case 44100:
case 48000:
/* these rates are ok for all outputs */
/* FIXME: program spdif channel control bits here so that
* userspace doesn't have to if it only plays pcm! */
err = 0;
goto out_unlock;
default:
/* got some rate that the digital output can't do,
* so disable and lock it */
onyx_read_register(cii->codec_data, ONYX_REG_DIG_INFO4, &v);
if (onyx_write_register(onyx,
ONYX_REG_DIG_INFO4,
v & ~ONYX_SPDIF_ENABLE))
goto out_unlock;
onyx->spdif_locked = 1;
err = 0;
goto out_unlock;
}
out_unlock:
mutex_unlock(&onyx->mutex);
return err;
}
static int onyx_open(struct codec_info_item *cii,
struct snd_pcm_substream *substream)
{
struct onyx *onyx = cii->codec_data;
mutex_lock(&onyx->mutex);
onyx->open_count++;
mutex_unlock(&onyx->mutex);
return 0;
}
static int onyx_close(struct codec_info_item *cii,
struct snd_pcm_substream *substream)
{
struct onyx *onyx = cii->codec_data;
mutex_lock(&onyx->mutex);
onyx->open_count--;
if (!onyx->open_count)
onyx->spdif_locked = onyx->analog_locked = 0;
mutex_unlock(&onyx->mutex);
return 0;
}
static int onyx_switch_clock(struct codec_info_item *cii,
enum clock_switch what)
{
struct onyx *onyx = cii->codec_data;
mutex_lock(&onyx->mutex);
/* this *MUST* be more elaborate later... */
switch (what) {
case CLOCK_SWITCH_PREPARE_SLAVE:
onyx->codec.gpio->methods->all_amps_off(onyx->codec.gpio);
break;
case CLOCK_SWITCH_SLAVE:
onyx->codec.gpio->methods->all_amps_restore(onyx->codec.gpio);
break;
default: /* silence warning */
break;
}
mutex_unlock(&onyx->mutex);
return 0;
}
#ifdef CONFIG_PM
static int onyx_suspend(struct codec_info_item *cii, pm_message_t state)
{
struct onyx *onyx = cii->codec_data;
u8 v;
int err = -ENXIO;
mutex_lock(&onyx->mutex);
if (onyx_read_register(onyx, ONYX_REG_CONTROL, &v))
goto out_unlock;
onyx_write_register(onyx, ONYX_REG_CONTROL, v | ONYX_ADPSV | ONYX_DAPSV);
/* Apple does a sleep here but the datasheet says to do it on resume */
err = 0;
out_unlock:
mutex_unlock(&onyx->mutex);
return err;
}
static int onyx_resume(struct codec_info_item *cii)
{
struct onyx *onyx = cii->codec_data;
u8 v;
int err = -ENXIO;
mutex_lock(&onyx->mutex);
/* take codec out of suspend */
if (onyx_read_register(onyx, ONYX_REG_CONTROL, &v))
goto out_unlock;
onyx_write_register(onyx, ONYX_REG_CONTROL, v & ~(ONYX_ADPSV | ONYX_DAPSV));
/* FIXME: should divide by sample rate, but 8k is the lowest we go */
msleep(2205000/8000);
/* reset all values */
onyx_register_init(onyx);
err = 0;
out_unlock:
mutex_unlock(&onyx->mutex);
return err;
}
#endif /* CONFIG_PM */
static struct codec_info onyx_codec_info = {
.transfers = onyx_transfers,
.sysclock_factor = 256,
.bus_factor = 64,
.owner = THIS_MODULE,
.usable = onyx_usable,
.prepare = onyx_prepare,
.open = onyx_open,
.close = onyx_close,
.switch_clock = onyx_switch_clock,
#ifdef CONFIG_PM
.suspend = onyx_suspend,
.resume = onyx_resume,
#endif
};
static int onyx_init_codec(struct aoa_codec *codec)
{
struct onyx *onyx = codec_to_onyx(codec);
struct snd_kcontrol *ctl;
struct codec_info *ci = &onyx_codec_info;
u8 v;
int err;
if (!onyx->codec.gpio || !onyx->codec.gpio->methods) {
printk(KERN_ERR PFX "gpios not assigned!!\n");
return -EINVAL;
}
onyx->codec.gpio->methods->set_hw_reset(onyx->codec.gpio, 0);
msleep(1);
onyx->codec.gpio->methods->set_hw_reset(onyx->codec.gpio, 1);
msleep(1);
onyx->codec.gpio->methods->set_hw_reset(onyx->codec.gpio, 0);
msleep(1);
if (onyx_register_init(onyx)) {
printk(KERN_ERR PFX "failed to initialise onyx registers\n");
return -ENODEV;
}
if (aoa_snd_device_new(SNDRV_DEV_LOWLEVEL, onyx, &ops)) {
printk(KERN_ERR PFX "failed to create onyx snd device!\n");
return -ENODEV;
}
/* nothing connected? what a joke! */
if ((onyx->codec.connected & 0xF) == 0)
return -ENOTCONN;
/* if no inputs are present... */
if ((onyx->codec.connected & 0xC) == 0) {
if (!onyx->codec_info)
onyx->codec_info = kmalloc(sizeof(struct codec_info), GFP_KERNEL);
if (!onyx->codec_info)
return -ENOMEM;
ci = onyx->codec_info;
*ci = onyx_codec_info;
ci->transfers++;
}
/* if no outputs are present... */
if ((onyx->codec.connected & 3) == 0) {
if (!onyx->codec_info)
onyx->codec_info = kmalloc(sizeof(struct codec_info), GFP_KERNEL);
if (!onyx->codec_info)
return -ENOMEM;
ci = onyx->codec_info;
/* this is fine as there have to be inputs
* if we end up in this part of the code */
*ci = onyx_codec_info;
ci->transfers[1].formats = 0;
}
if (onyx->codec.soundbus_dev->attach_codec(onyx->codec.soundbus_dev,
aoa_get_card(),
ci, onyx)) {
printk(KERN_ERR PFX "error creating onyx pcm\n");
return -ENODEV;
}
#define ADDCTL(n) \
do { \
ctl = snd_ctl_new1(&n, onyx); \
if (ctl) { \
ctl->id.device = \
onyx->codec.soundbus_dev->pcm->device; \
err = aoa_snd_ctl_add(ctl); \
if (err) \
goto error; \
} \
} while (0)
if (onyx->codec.soundbus_dev->pcm) {
/* give the user appropriate controls
* depending on what inputs are connected */
if ((onyx->codec.connected & 0xC) == 0xC)
ADDCTL(capture_source_control);
else if (onyx->codec.connected & 4)
onyx_set_capture_source(onyx, 0);
else
onyx_set_capture_source(onyx, 1);
if (onyx->codec.connected & 0xC)
ADDCTL(inputgain_control);
/* depending on what output is connected,
* give the user appropriate controls */
if (onyx->codec.connected & 1) {
ADDCTL(volume_control);
ADDCTL(mute_control);
ADDCTL(ovr1_control);
ADDCTL(flt0_control);
ADDCTL(hpf_control);
ADDCTL(dm12_control);
/* spdif control defaults to off */
}
if (onyx->codec.connected & 2) {
ADDCTL(onyx_spdif_mask);
ADDCTL(onyx_spdif_ctrl);
}
if ((onyx->codec.connected & 3) == 3)
ADDCTL(spdif_control);
/* if only S/PDIF is connected, enable it unconditionally */
if ((onyx->codec.connected & 3) == 2) {
onyx_read_register(onyx, ONYX_REG_DIG_INFO4, &v);
v |= ONYX_SPDIF_ENABLE;
onyx_write_register(onyx, ONYX_REG_DIG_INFO4, v);
}
}
#undef ADDCTL
printk(KERN_INFO PFX "attached to onyx codec via i2c\n");
return 0;
error:
onyx->codec.soundbus_dev->detach_codec(onyx->codec.soundbus_dev, onyx);
snd_device_free(aoa_get_card(), onyx);
return err;
}
static void onyx_exit_codec(struct aoa_codec *codec)
{
struct onyx *onyx = codec_to_onyx(codec);
if (!onyx->codec.soundbus_dev) {
printk(KERN_ERR PFX "onyx_exit_codec called without soundbus_dev!\n");
return;
}
onyx->codec.soundbus_dev->detach_codec(onyx->codec.soundbus_dev, onyx);
}
static struct i2c_driver onyx_driver;
static int onyx_create(struct i2c_adapter *adapter,
struct device_node *node,
int addr)
{
struct onyx *onyx;
u8 dummy;
onyx = kzalloc(sizeof(struct onyx), GFP_KERNEL);
if (!onyx)
return -ENOMEM;
mutex_init(&onyx->mutex);
onyx->i2c.driver = &onyx_driver;
onyx->i2c.adapter = adapter;
onyx->i2c.addr = addr & 0x7f;
strlcpy(onyx->i2c.name, "onyx audio codec", I2C_NAME_SIZE-1);
if (i2c_attach_client(&onyx->i2c)) {
printk(KERN_ERR PFX "failed to attach to i2c\n");
goto fail;
}
/* we try to read from register ONYX_REG_CONTROL
* to check if the codec is present */
if (onyx_read_register(onyx, ONYX_REG_CONTROL, &dummy) != 0) {
i2c_detach_client(&onyx->i2c);
printk(KERN_ERR PFX "failed to read control register\n");
goto fail;
}
strlcpy(onyx->codec.name, "onyx", MAX_CODEC_NAME_LEN-1);
onyx->codec.owner = THIS_MODULE;
onyx->codec.init = onyx_init_codec;
onyx->codec.exit = onyx_exit_codec;
onyx->codec.node = of_node_get(node);
if (aoa_codec_register(&onyx->codec)) {
i2c_detach_client(&onyx->i2c);
goto fail;
}
printk(KERN_DEBUG PFX "created and attached onyx instance\n");
return 0;
fail:
kfree(onyx);
return -EINVAL;
}
static int onyx_i2c_attach(struct i2c_adapter *adapter)
{
struct device_node *busnode, *dev = NULL;
struct pmac_i2c_bus *bus;
bus = pmac_i2c_adapter_to_bus(adapter);
if (bus == NULL)
return -ENODEV;
busnode = pmac_i2c_get_bus_node(bus);
while ((dev = of_get_next_child(busnode, dev)) != NULL) {
if (device_is_compatible(dev, "pcm3052")) {
u32 *addr;
printk(KERN_DEBUG PFX "found pcm3052\n");
addr = (u32 *) get_property(dev, "reg", NULL);
if (!addr)
return -ENODEV;
return onyx_create(adapter, dev, (*addr)>>1);
}
}
/* if that didn't work, try desperate mode for older
* machines that have stuff missing from the device tree */
if (!device_is_compatible(busnode, "k2-i2c"))
return -ENODEV;
printk(KERN_DEBUG PFX "found k2-i2c, checking if onyx chip is on it\n");
/* probe both possible addresses for the onyx chip */
if (onyx_create(adapter, NULL, 0x46) == 0)
return 0;
return onyx_create(adapter, NULL, 0x47);
}
static int onyx_i2c_detach(struct i2c_client *client)
{
struct onyx *onyx = container_of(client, struct onyx, i2c);
int err;
if ((err = i2c_detach_client(client)))
return err;
aoa_codec_unregister(&onyx->codec);
of_node_put(onyx->codec.node);
if (onyx->codec_info)
kfree(onyx->codec_info);
kfree(onyx);
return 0;
}
static struct i2c_driver onyx_driver = {
.driver = {
.name = "aoa_codec_onyx",
.owner = THIS_MODULE,
},
.attach_adapter = onyx_i2c_attach,
.detach_client = onyx_i2c_detach,
};
static int __init onyx_init(void)
{
return i2c_add_driver(&onyx_driver);
}
static void __exit onyx_exit(void)
{
i2c_del_driver(&onyx_driver);
}
module_init(onyx_init);
module_exit(onyx_exit);
sound/aoa/codecs/snd-aoa-codec-onyx.h 0 → 100644
View file @ f3d9478b
/*
* Apple Onboard Audio driver for Onyx codec (header)
*
* Copyright 2006 Johannes Berg <johannes@sipsolutions.net>
*
* GPL v2, can be found in COPYING.
*/
#ifndef __SND_AOA_CODEC_ONYX_H
#define __SND_AOA_CODEC_ONYX_H
#include <stddef.h>
#include <linux/i2c.h>
#include <linux/i2c-dev.h>
#include <asm/pmac_low_i2c.h>
#include <asm/prom.h>
/* PCM3052 register definitions */
/* the attenuation registers take values from
* -1 (0dB) to -127 (-63.0 dB) or others (muted) */
#define ONYX_REG_DAC_ATTEN_LEFT 65
#define FIRSTREGISTER ONYX_REG_DAC_ATTEN_LEFT
#define ONYX_REG_DAC_ATTEN_RIGHT 66
#define ONYX_REG_CONTROL 67
# define ONYX_MRST (1<<7)
# define ONYX_SRST (1<<6)
# define ONYX_ADPSV (1<<5)
# define ONYX_DAPSV (1<<4)
# define ONYX_SILICONVERSION (1<<0)
/* all others reserved */
#define ONYX_REG_DAC_CONTROL 68
# define ONYX_OVR1 (1<<6)
# define ONYX_MUTE_RIGHT (1<<1)
# define ONYX_MUTE_LEFT (1<<0)
#define ONYX_REG_DAC_DEEMPH 69
# define ONYX_DIGDEEMPH_SHIFT 5
# define ONYX_DIGDEEMPH_MASK (3<<ONYX_DIGDEEMPH_SHIFT)
# define ONYX_DIGDEEMPH_CTRL (1<<4)
#define ONYX_REG_DAC_FILTER 70
# define ONYX_ROLLOFF_FAST (1<<5)
# define ONYX_DAC_FILTER_ALWAYS (1<<2)
#define ONYX_REG_DAC_OUTPHASE 71
# define ONYX_OUTPHASE_INVERTED (1<<0)
#define ONYX_REG_ADC_CONTROL 72
# define ONYX_ADC_INPUT_MIC (1<<5)
/* 8 + input gain in dB, valid range for input gain is -4 .. 20 dB */
# define ONYX_ADC_PGA_GAIN_MASK 0x1f
#define ONYX_REG_ADC_HPF_BYPASS 75
# define ONYX_HPF_DISABLE (1<<3)
# define ONYX_ADC_HPF_ALWAYS (1<<2)
#define ONYX_REG_DIG_INFO1 77
# define ONYX_MASK_DIN_TO_BPZ (1<<7)
/* bits 1-5 control channel bits 1-5 */
# define ONYX_DIGOUT_DISABLE (1<<0)
#define ONYX_REG_DIG_INFO2 78
/* controls channel bits 8-15 */
#define ONYX_REG_DIG_INFO3 79
/* control channel bits 24-29, high 2 bits reserved */
#define ONYX_REG_DIG_INFO4 80
# define ONYX_VALIDL (1<<7)
# define ONYX_VALIDR (1<<6)
# define ONYX_SPDIF_ENABLE (1<<5)
/* lower 4 bits control bits 32-35 of channel control and word length */
# define ONYX_WORDLEN_MASK (0xF)
#endif /* __SND_AOA_CODEC_ONYX_H */
sound/aoa/codecs/snd-aoa-codec-tas-gain-table.h 0 → 100644
View file @ f3d9478b
/*
This is the program used to generate below table.
#include <stdio.h>
#include <math.h>
int main() {
int dB2;
printf("/" "* This file is only included exactly once!\n");
printf(" *\n");
printf(" * If they'd only tell us that generating this table was\n");
printf(" * as easy as calculating\n");
printf(" * hwvalue = 1048576.0*exp(0.057564628*dB*2)\n");
printf(" * :) *" "/\n");
printf("static int tas_gaintable[] = {\n");
printf(" 0x000000, /" "* -infinity dB *" "/\n");
for (dB2=-140;dB2<=36;dB2++)
printf(" 0x%.6x, /" "* %-02.1f dB *" "/\n", (int)(1048576.0*exp(0.057564628*dB2)), dB2/2.0);
printf("};\n\n");
}
*/
/* This file is only included exactly once!
*
* If they'd only tell us that generating this table was
* as easy as calculating
* hwvalue = 1048576.0*exp(0.057564628*dB*2)
* :) */
static int tas_gaintable[] = {
0x000000, /* -infinity dB */
0x00014b, /* -70.0 dB */
0x00015f, /* -69.5 dB */
0x000174, /* -69.0 dB */
0x00018a, /* -68.5 dB */
0x0001a1, /* -68.0 dB */
0x0001ba, /* -67.5 dB */
0x0001d4, /* -67.0 dB */
0x0001f0, /* -66.5 dB */
0x00020d, /* -66.0 dB */
0x00022c, /* -65.5 dB */
0x00024d, /* -65.0 dB */
0x000270, /* -64.5 dB */
0x000295, /* -64.0 dB */
0x0002bc, /* -63.5 dB */
0x0002e6, /* -63.0 dB */
0x000312, /* -62.5 dB */
0x000340, /* -62.0 dB */
0x000372, /* -61.5 dB */
0x0003a6, /* -61.0 dB */
0x0003dd, /* -60.5 dB */
0x000418, /* -60.0 dB */
0x000456, /* -59.5 dB */
0x000498, /* -59.0 dB */
0x0004de, /* -58.5 dB */
0x000528, /* -58.0 dB */
0x000576, /* -57.5 dB */
0x0005c9, /* -57.0 dB */
0x000620, /* -56.5 dB */
0x00067d, /* -56.0 dB */
0x0006e0, /* -55.5 dB */
0x000748, /* -55.0 dB */
0x0007b7, /* -54.5 dB */
0x00082c, /* -54.0 dB */
0x0008a8, /* -53.5 dB */
0x00092b, /* -53.0 dB */
0x0009b6, /* -52.5 dB */
0x000a49, /* -52.0 dB */
0x000ae5, /* -51.5 dB */
0x000b8b, /* -51.0 dB */
0x000c3a, /* -50.5 dB */
0x000cf3, /* -50.0 dB */
0x000db8, /* -49.5 dB */
0x000e88, /* -49.0 dB */
0x000f64, /* -48.5 dB */
0x00104e, /* -48.0 dB */
0x001145, /* -47.5 dB */
0x00124b, /* -47.0 dB */
0x001361, /* -46.5 dB */
0x001487, /* -46.0 dB */
0x0015be, /* -45.5 dB */
0x001708, /* -45.0 dB */
0x001865, /* -44.5 dB */
0x0019d8, /* -44.0 dB */
0x001b60, /* -43.5 dB */
0x001cff, /* -43.0 dB */
0x001eb7, /* -42.5 dB */
0x002089, /* -42.0 dB */
0x002276, /* -41.5 dB */
0x002481, /* -41.0 dB */
0x0026ab, /* -40.5 dB */
0x0028f5, /* -40.0 dB */
0x002b63, /* -39.5 dB */
0x002df5, /* -39.0 dB */
0x0030ae, /* -38.5 dB */
0x003390, /* -38.0 dB */
0x00369e, /* -37.5 dB */
0x0039db, /* -37.0 dB */
0x003d49, /* -36.5 dB */
0x0040ea, /* -36.0 dB */
0x0044c3, /* -35.5 dB */
0x0048d6, /* -35.0 dB */
0x004d27, /* -34.5 dB */
0x0051b9, /* -34.0 dB */
0x005691, /* -33.5 dB */
0x005bb2, /* -33.0 dB */
0x006121, /* -32.5 dB */
0x0066e3, /* -32.0 dB */
0x006cfb, /* -31.5 dB */
0x007370, /* -31.0 dB */
0x007a48, /* -30.5 dB */
0x008186, /* -30.0 dB */
0x008933, /* -29.5 dB */
0x009154, /* -29.0 dB */
0x0099f1, /* -28.5 dB */
0x00a310, /* -28.0 dB */
0x00acba, /* -27.5 dB */
0x00b6f6, /* -27.0 dB */
0x00c1cd, /* -26.5 dB */
0x00cd49, /* -26.0 dB */
0x00d973, /* -25.5 dB */
0x00e655, /* -25.0 dB */
0x00f3fb, /* -24.5 dB */
0x010270, /* -24.0 dB */
0x0111c0, /* -23.5 dB */
0x0121f9, /* -23.0 dB */
0x013328, /* -22.5 dB */
0x01455b, /* -22.0 dB */
0x0158a2, /* -21.5 dB */
0x016d0e, /* -21.0 dB */
0x0182af, /* -20.5 dB */
0x019999, /* -20.0 dB */
0x01b1de, /* -19.5 dB */
0x01cb94, /* -19.0 dB */
0x01e6cf, /* -18.5 dB */
0x0203a7, /* -18.0 dB */
0x022235, /* -17.5 dB */
0x024293, /* -17.0 dB */
0x0264db, /* -16.5 dB */
0x02892c, /* -16.0 dB */
0x02afa3, /* -15.5 dB */
0x02d862, /* -15.0 dB */
0x03038a, /* -14.5 dB */
0x033142, /* -14.0 dB */
0x0361af, /* -13.5 dB */
0x0394fa, /* -13.0 dB */
0x03cb50, /* -12.5 dB */
0x0404de, /* -12.0 dB */
0x0441d5, /* -11.5 dB */
0x048268, /* -11.0 dB */
0x04c6d0, /* -10.5 dB */
0x050f44, /* -10.0 dB */
0x055c04, /* -9.5 dB */
0x05ad50, /* -9.0 dB */
0x06036e, /* -8.5 dB */
0x065ea5, /* -8.0 dB */
0x06bf44, /* -7.5 dB */
0x07259d, /* -7.0 dB */
0x079207, /* -6.5 dB */
0x0804dc, /* -6.0 dB */
0x087e80, /* -5.5 dB */
0x08ff59, /* -5.0 dB */
0x0987d5, /* -4.5 dB */
0x0a1866, /* -4.0 dB */
0x0ab189, /* -3.5 dB */
0x0b53be, /* -3.0 dB */
0x0bff91, /* -2.5 dB */
0x0cb591, /* -2.0 dB */
0x0d765a, /* -1.5 dB */
0x0e4290, /* -1.0 dB */
0x0f1adf, /* -0.5 dB */
0x100000, /* 0.0 dB */
0x10f2b4, /* 0.5 dB */
0x11f3c9, /* 1.0 dB */
0x13041a, /* 1.5 dB */
0x14248e, /* 2.0 dB */
0x15561a, /* 2.5 dB */
0x1699c0, /* 3.0 dB */
0x17f094, /* 3.5 dB */
0x195bb8, /* 4.0 dB */
0x1adc61, /* 4.5 dB */
0x1c73d5, /* 5.0 dB */
0x1e236d, /* 5.5 dB */
0x1fec98, /* 6.0 dB */
0x21d0d9, /* 6.5 dB */
0x23d1cd, /* 7.0 dB */
0x25f125, /* 7.5 dB */
0x2830af, /* 8.0 dB */
0x2a9254, /* 8.5 dB */
0x2d1818, /* 9.0 dB */
0x2fc420, /* 9.5 dB */
0x3298b0, /* 10.0 dB */
0x35982f, /* 10.5 dB */
0x38c528, /* 11.0 dB */
0x3c224c, /* 11.5 dB */
0x3fb278, /* 12.0 dB */
0x4378b0, /* 12.5 dB */
0x477829, /* 13.0 dB */
0x4bb446, /* 13.5 dB */
0x5030a1, /* 14.0 dB */
0x54f106, /* 14.5 dB */
0x59f980, /* 15.0 dB */
0x5f4e52, /* 15.5 dB */
0x64f403, /* 16.0 dB */
0x6aef5e, /* 16.5 dB */
0x714575, /* 17.0 dB */
0x77fbaa, /* 17.5 dB */
0x7f17af, /* 18.0 dB */
};
sound/aoa/codecs/snd-aoa-codec-tas.c 0 → 100644
View file @ f3d9478b
/*
* Apple Onboard Audio driver for tas codec
*
* Copyright 2006 Johannes Berg <johannes@sipsolutions.net>
*
* GPL v2, can be found in COPYING.
*
* Open questions:
* - How to distinguish between 3004 and versions?
*
* FIXMEs:
* - This codec driver doesn't honour the 'connected'
* property of the aoa_codec struct, hence if
* it is used in machines where not everything is
* connected it will display wrong mixer elements.
* - Driver assumes that the microphone is always
* monaureal and connected to the right channel of
* the input. This should also be a codec-dependent
* flag, maybe the codec should have 3 different
* bits for the three different possibilities how
* it can be hooked up...
* But as long as I don't see any hardware hooked
* up that way...
* - As Apple notes in their code, the tas3004 seems
* to delay the right channel by one sample. You can
* see this when for example recording stereo in
* audacity, or recording the tas output via cable
* on another machine (use a sinus generator or so).
* I tried programming the BiQuads but couldn't
* make the delay work, maybe someone can read the
* datasheet and fix it. The relevant Apple comment
* is in AppleTAS3004Audio.cpp lines 1637 ff. Note
* that their comment describing how they program
* the filters sucks...
*
* Other things:
* - this should actually register *two* aoa_codec
* structs since it has two inputs. Then it must
* use the prepare callback to forbid running the
* secondary output on a different clock.
* Also, whatever bus knows how to do this must
* provide two soundbus_dev devices and the fabric
* must be able to link them correctly.
*
* I don't even know if Apple ever uses the second
* port on the tas3004 though, I don't think their
* i2s controllers can even do it. OTOH, they all
* derive the clocks from common clocks, so it
* might just be possible. The framework allows the
* codec to refine the transfer_info items in the
* usable callback, so we can simply remove the
* rates the second instance is not using when it
* actually is in use.
* Maybe we'll need to make the sound busses have
* a 'clock group id' value so the codec can
* determine if the two outputs can be driven at
* the same time. But that is likely overkill, up
* to the fabric to not link them up incorrectly,
* and up to the hardware designer to not wire
* them up in some weird unusable way.
*/
#include <stddef.h>
#include <linux/i2c.h>
#include <linux/i2c-dev.h>
#include <asm/pmac_low_i2c.h>
#include <asm/prom.h>
#include <linux/delay.h>
#include <linux/module.h>
MODULE_AUTHOR("Johannes Berg <johannes@sipsolutions.net>");
MODULE_LICENSE("GPL");
MODULE_DESCRIPTION("tas codec driver for snd-aoa");
#include "snd-aoa-codec-tas.h"
#include "snd-aoa-codec-tas-gain-table.h"
#include "../aoa.h"
#include "../soundbus/soundbus.h"
#define PFX "snd-aoa-codec-tas: "
struct tas {
struct aoa_codec codec;
struct i2c_client i2c;
u32 muted_l:1, muted_r:1,
controls_created:1;
u8 cached_volume_l, cached_volume_r;
u8 mixer_l[3], mixer_r[3];
u8 acr;
};
static struct tas *codec_to_tas(struct aoa_codec *codec)
{
return container_of(codec, struct tas, codec);
}
static inline int tas_write_reg(struct tas *tas, u8 reg, u8 len, u8 *data)
{
if (len == 1)
return i2c_smbus_write_byte_data(&tas->i2c, reg, *data);
else
return i2c_smbus_write_i2c_block_data(&tas->i2c, reg, len, data);
}
static void tas_set_volume(struct tas *tas)
{
u8 block[6];
int tmp;
u8 left, right;
left = tas->cached_volume_l;
right = tas->cached_volume_r;
if (left > 177) left = 177;
if (right > 177) right = 177;
if (tas->muted_l) left = 0;
if (tas->muted_r) right = 0;
/* analysing the volume and mixer tables shows
* that they are similar enough when we shift
* the mixer table down by 4 bits. The error
* is miniscule, in just one item the error
* is 1, at a value of 0x07f17b (mixer table
* value is 0x07f17a) */
tmp = tas_gaintable[left];
block[0] = tmp>>20;
block[1] = tmp>>12;
block[2] = tmp>>4;
tmp = tas_gaintable[right];
block[3] = tmp>>20;
block[4] = tmp>>12;
block[5] = tmp>>4;
tas_write_reg(tas, TAS_REG_VOL, 6, block);
}
static void tas_set_mixer(struct tas *tas)
{
u8 block[9];
int tmp, i;
u8 val;
for (i=0;i<3;i++) {
val = tas->mixer_l[i];
if (val > 177) val = 177;
tmp = tas_gaintable[val];
block[3*i+0] = tmp>>16;
block[3*i+1] = tmp>>8;
block[3*i+2] = tmp;
}
tas_write_reg(tas, TAS_REG_LMIX, 9, block);
for (i=0;i<3;i++) {
val = tas->mixer_r[i];
if (val > 177) val = 177;
tmp = tas_gaintable[val];
block[3*i+0] = tmp>>16;
block[3*i+1] = tmp>>8;
block[3*i+2] = tmp;
}
tas_write_reg(tas, TAS_REG_RMIX, 9, block);
}
/* alsa stuff */
static int tas_dev_register(struct snd_device *dev)
{
return 0;
}
static struct snd_device_ops ops = {
.dev_register = tas_dev_register,
};
static int tas_snd_vol_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;
uinfo->value.integer.max = 177;
return 0;
}
static int tas_snd_vol_get(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
struct tas *tas = snd_kcontrol_chip(kcontrol);
ucontrol->value.integer.value[0] = tas->cached_volume_l;
ucontrol->value.integer.value[1] = tas->cached_volume_r;
return 0;
}
static int tas_snd_vol_put(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
struct tas *tas = snd_kcontrol_chip(kcontrol);
if (tas->cached_volume_l == ucontrol->value.integer.value[0]
&& tas->cached_volume_r == ucontrol->value.integer.value[1])
return 0;
tas->cached_volume_l = ucontrol->value.integer.value[0];
tas->cached_volume_r = ucontrol->value.integer.value[1];
tas_set_volume(tas);
return 1;
}
static struct snd_kcontrol_new volume_control = {
.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
.name = "Master Playback Volume",
.access = SNDRV_CTL_ELEM_ACCESS_READWRITE,
.info = tas_snd_vol_info,
.get = tas_snd_vol_get,
.put = tas_snd_vol_put,
};
static int tas_snd_mute_info(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_info *uinfo)
{
uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN;
uinfo->count = 2;
uinfo->value.integer.min = 0;
uinfo->value.integer.max = 1;
return 0;
}
static int tas_snd_mute_get(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
struct tas *tas = snd_kcontrol_chip(kcontrol);
ucontrol->value.integer.value[0] = !tas->muted_l;
ucontrol->value.integer.value[1] = !tas->muted_r;
return 0;
}
static int tas_snd_mute_put(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
struct tas *tas = snd_kcontrol_chip(kcontrol);
if (tas->muted_l == !ucontrol->value.integer.value[0]
&& tas->muted_r == !ucontrol->value.integer.value[1])
return 0;
tas->muted_l = !ucontrol->value.integer.value[0];
tas->muted_r = !ucontrol->value.integer.value[1];
tas_set_volume(tas);
return 1;
}
static struct snd_kcontrol_new mute_control = {
.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
.name = "Master Playback Switch",
.access = SNDRV_CTL_ELEM_ACCESS_READWRITE,
.info = tas_snd_mute_info,
.get = tas_snd_mute_get,
.put = tas_snd_mute_put,
};
static int tas_snd_mixer_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;
uinfo->value.integer.max = 177;
return 0;
}
static int tas_snd_mixer_get(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
struct tas *tas = snd_kcontrol_chip(kcontrol);
int idx = kcontrol->private_value;
ucontrol->value.integer.value[0] = tas->mixer_l[idx];
ucontrol->value.integer.value[1] = tas->mixer_r[idx];
return 0;
}
static int tas_snd_mixer_put(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
struct tas *tas = snd_kcontrol_chip(kcontrol);
int idx = kcontrol->private_value;
if (tas->mixer_l[idx] == ucontrol->value.integer.value[0]
&& tas->mixer_r[idx] == ucontrol->value.integer.value[1])
return 0;
tas->mixer_l[idx] = ucontrol->value.integer.value[0];
tas->mixer_r[idx] = ucontrol->value.integer.value[1];
tas_set_mixer(tas);
return 1;
}
#define MIXER_CONTROL(n,descr,idx) \
static struct snd_kcontrol_new n##_control = { \
.iface = SNDRV_CTL_ELEM_IFACE_MIXER, \
.name = descr " Playback Volume", \
.access = SNDRV_CTL_ELEM_ACCESS_READWRITE, \
.info = tas_snd_mixer_info, \
.get = tas_snd_mixer_get, \
.put = tas_snd_mixer_put, \
.private_value = idx, \
}
MIXER_CONTROL(pcm1, "PCM1", 0);
MIXER_CONTROL(monitor, "Monitor", 2);
static int tas_snd_capture_source_info(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_info *uinfo)
{
static char *texts[] = { "Line-In", "Microphone" };
uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
uinfo->count = 1;
uinfo->value.enumerated.items = 2;
if (uinfo->value.enumerated.item > 1)
uinfo->value.enumerated.item = 1;
strcpy(uinfo->value.enumerated.name, texts[uinfo->value.enumerated.item]);
return 0;
}
static int tas_snd_capture_source_get(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
struct tas *tas = snd_kcontrol_chip(kcontrol);
ucontrol->value.enumerated.item[0] = !!(tas->acr & TAS_ACR_INPUT_B);
return 0;
}
static int tas_snd_capture_source_put(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
struct tas *tas = snd_kcontrol_chip(kcontrol);
int oldacr = tas->acr;
tas->acr &= ~TAS_ACR_INPUT_B;
if (ucontrol->value.enumerated.item[0])
tas->acr |= TAS_ACR_INPUT_B;
if (oldacr == tas->acr)
return 0;
tas_write_reg(tas, TAS_REG_ACR, 1, &tas->acr);
return 1;
}
static struct snd_kcontrol_new capture_source_control = {
.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
/* If we name this 'Input Source', it properly shows up in
* alsamixer as a selection, * but it's shown under the
* 'Playback' category.
* If I name it 'Capture Source', it shows up in strange
* ways (two bools of which one can be selected at a
* time) but at least it's shown in the 'Capture'
* category.
* I was told that this was due to backward compatibility,
* but I don't understand then why the mangling is *not*
* done when I name it "Input Source".....
*/
.name = "Capture Source",
.access = SNDRV_CTL_ELEM_ACCESS_READWRITE,
.info = tas_snd_capture_source_info,
.get = tas_snd_capture_source_get,
.put = tas_snd_capture_source_put,
};
static struct transfer_info tas_transfers[] = {
{
/* input */
.formats = SNDRV_PCM_FMTBIT_S16_BE | SNDRV_PCM_FMTBIT_S16_BE |
SNDRV_PCM_FMTBIT_S24_BE | SNDRV_PCM_FMTBIT_S24_BE,
.rates = SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000,
.transfer_in = 1,
},
{
/* output */
.formats = SNDRV_PCM_FMTBIT_S16_BE | SNDRV_PCM_FMTBIT_S16_BE |
SNDRV_PCM_FMTBIT_S24_BE | SNDRV_PCM_FMTBIT_S24_BE,
.rates = SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000,
.transfer_in = 0,
},
{}
};
static int tas_usable(struct codec_info_item *cii,
struct transfer_info *ti,
struct transfer_info *out)
{
return 1;
}
static int tas_reset_init(struct tas *tas)
{
u8 tmp;
tas->codec.gpio->methods->set_hw_reset(tas->codec.gpio, 0);
msleep(1);
tas->codec.gpio->methods->set_hw_reset(tas->codec.gpio, 1);
msleep(1);
tas->codec.gpio->methods->set_hw_reset(tas->codec.gpio, 0);
msleep(1);
tas->acr &= ~TAS_ACR_ANALOG_PDOWN;
tas->acr |= TAS_ACR_B_MONAUREAL | TAS_ACR_B_MON_SEL_RIGHT;
if (tas_write_reg(tas, TAS_REG_ACR, 1, &tas->acr))
return -ENODEV;
tmp = TAS_MCS_SCLK64 | TAS_MCS_SPORT_MODE_I2S | TAS_MCS_SPORT_WL_24BIT;
if (tas_write_reg(tas, TAS_REG_MCS, 1, &tmp))
return -ENODEV;
tmp = 0;
if (tas_write_reg(tas, TAS_REG_MCS2, 1, &tmp))
return -ENODEV;
return 0;
}
/* we are controlled via i2c and assume that is always up
* If that wasn't the case, we'd have to suspend once
* our i2c device is suspended, and then take note of that! */
static int tas_suspend(struct tas *tas)
{
tas->acr |= TAS_ACR_ANALOG_PDOWN;
tas_write_reg(tas, TAS_REG_ACR, 1, &tas->acr);
return 0;
}
static int tas_resume(struct tas *tas)
{
/* reset codec */
tas_reset_init(tas);
tas_set_volume(tas);
tas_set_mixer(tas);
return 0;
}
#ifdef CONFIG_PM
static int _tas_suspend(struct codec_info_item *cii, pm_message_t state)
{
return tas_suspend(cii->codec_data);
}
static int _tas_resume(struct codec_info_item *cii)
{
return tas_resume(cii->codec_data);
}
#endif
static struct codec_info tas_codec_info = {
.transfers = tas_transfers,
/* in theory, we can drive it at 512 too...
* but so far the framework doesn't allow
* for that and I don't see much point in it. */
.sysclock_factor = 256,
/* same here, could be 32 for just one 16 bit format */
.bus_factor = 64,
.owner = THIS_MODULE,
.usable = tas_usable,
#ifdef CONFIG_PM
.suspend = _tas_suspend,
.resume = _tas_resume,
#endif
};
static int tas_init_codec(struct aoa_codec *codec)
{
struct tas *tas = codec_to_tas(codec);
int err;
if (!tas->codec.gpio || !tas->codec.gpio->methods) {
printk(KERN_ERR PFX "gpios not assigned!!\n");
return -EINVAL;
}
if (tas_reset_init(tas)) {
printk(KERN_ERR PFX "tas failed to initialise\n");
return -ENXIO;
}
if (tas->codec.soundbus_dev->attach_codec(tas->codec.soundbus_dev,
aoa_get_card(),
&tas_codec_info, tas)) {
printk(KERN_ERR PFX "error attaching tas to soundbus\n");
return -ENODEV;
}
if (aoa_snd_device_new(SNDRV_DEV_LOWLEVEL, tas, &ops)) {
printk(KERN_ERR PFX "failed to create tas snd device!\n");
return -ENODEV;
}
err = aoa_snd_ctl_add(snd_ctl_new1(&volume_control, tas));
if (err)
goto error;
err = aoa_snd_ctl_add(snd_ctl_new1(&mute_control, tas));
if (err)
goto error;
err = aoa_snd_ctl_add(snd_ctl_new1(&pcm1_control, tas));
if (err)
goto error;
err = aoa_snd_ctl_add(snd_ctl_new1(&monitor_control, tas));
if (err)
goto error;
err = aoa_snd_ctl_add(snd_ctl_new1(&capture_source_control, tas));
if (err)
goto error;
return 0;
error:
tas->codec.soundbus_dev->detach_codec(tas->codec.soundbus_dev, tas);
snd_device_free(aoa_get_card(), tas);
return err;
}
static void tas_exit_codec(struct aoa_codec *codec)
{
struct tas *tas = codec_to_tas(codec);
if (!tas->codec.soundbus_dev)
return;
tas->codec.soundbus_dev->detach_codec(tas->codec.soundbus_dev, tas);
}
static struct i2c_driver tas_driver;
static int tas_create(struct i2c_adapter *adapter,
struct device_node *node,
int addr)
{
struct tas *tas;
tas = kzalloc(sizeof(struct tas), GFP_KERNEL);
if (!tas)
return -ENOMEM;
tas->i2c.driver = &tas_driver;
tas->i2c.adapter = adapter;
tas->i2c.addr = addr;
strlcpy(tas->i2c.name, "tas audio codec", I2C_NAME_SIZE-1);
if (i2c_attach_client(&tas->i2c)) {
printk(KERN_ERR PFX "failed to attach to i2c\n");
goto fail;
}
strlcpy(tas->codec.name, "tas", MAX_CODEC_NAME_LEN-1);
tas->codec.owner = THIS_MODULE;
tas->codec.init = tas_init_codec;
tas->codec.exit = tas_exit_codec;
tas->codec.node = of_node_get(node);
if (aoa_codec_register(&tas->codec)) {
goto detach;
}
printk(KERN_DEBUG "snd-aoa-codec-tas: created and attached tas instance\n");
return 0;
detach:
i2c_detach_client(&tas->i2c);
fail:
kfree(tas);
return -EINVAL;
}
static int tas_i2c_attach(struct i2c_adapter *adapter)
{
struct device_node *busnode, *dev = NULL;
struct pmac_i2c_bus *bus;
bus = pmac_i2c_adapter_to_bus(adapter);
if (bus == NULL)
return -ENODEV;
busnode = pmac_i2c_get_bus_node(bus);
while ((dev = of_get_next_child(busnode, dev)) != NULL) {
if (device_is_compatible(dev, "tas3004")) {
u32 *addr;
printk(KERN_DEBUG PFX "found tas3004\n");
addr = (u32 *) get_property(dev, "reg", NULL);
if (!addr)
continue;
return tas_create(adapter, dev, ((*addr) >> 1) & 0x7f);
}
/* older machines have no 'codec' node with a 'compatible'
* property that says 'tas3004', they just have a 'deq'
* node without any such property... */
if (strcmp(dev->name, "deq") == 0) {
u32 *_addr, addr;
printk(KERN_DEBUG PFX "found 'deq' node\n");
_addr = (u32 *) get_property(dev, "i2c-address", NULL);
if (!_addr)
continue;
addr = ((*_addr) >> 1) & 0x7f;
/* now, if the address doesn't match any of the two
* that a tas3004 can have, we cannot handle this.
* I doubt it ever happens but hey. */
if (addr != 0x34 && addr != 0x35)
continue;
return tas_create(adapter, dev, addr);
}
}
return -ENODEV;
}
static int tas_i2c_detach(struct i2c_client *client)
{
struct tas *tas = container_of(client, struct tas, i2c);
int err;
u8 tmp = TAS_ACR_ANALOG_PDOWN;
if ((err = i2c_detach_client(client)))
return err;
aoa_codec_unregister(&tas->codec);
of_node_put(tas->codec.node);
/* power down codec chip */
tas_write_reg(tas, TAS_REG_ACR, 1, &tmp);
kfree(tas);
return 0;
}
static struct i2c_driver tas_driver = {
.driver = {
.name = "aoa_codec_tas",
.owner = THIS_MODULE,
},
.attach_adapter = tas_i2c_attach,
.detach_client = tas_i2c_detach,
};
static int __init tas_init(void)
{
return i2c_add_driver(&tas_driver);
}
static void __exit tas_exit(void)
{
i2c_del_driver(&tas_driver);
}
module_init(tas_init);
module_exit(tas_exit);
sound/aoa/codecs/snd-aoa-codec-tas.h 0 → 100644
View file @ f3d9478b
/*
* Apple Onboard Audio driver for tas codec (header)
*
* Copyright 2006 Johannes Berg <johannes@sipsolutions.net>
*
* GPL v2, can be found in COPYING.
*/
#ifndef __SND_AOA_CODECTASH
#define __SND_AOA_CODECTASH
#define TAS_REG_MCS 0x01 /* main control */
# define TAS_MCS_FASTLOAD (1<<7)
# define TAS_MCS_SCLK64 (1<<6)
# define TAS_MCS_SPORT_MODE_MASK (3<<4)
# define TAS_MCS_SPORT_MODE_I2S (2<<4)
# define TAS_MCS_SPORT_MODE_RJ (1<<4)
# define TAS_MCS_SPORT_MODE_LJ (0<<4)
# define TAS_MCS_SPORT_WL_MASK (3<<0)
# define TAS_MCS_SPORT_WL_16BIT (0<<0)
# define TAS_MCS_SPORT_WL_18BIT (1<<0)
# define TAS_MCS_SPORT_WL_20BIT (2<<0)
# define TAS_MCS_SPORT_WL_24BIT (3<<0)
#define TAS_REG_DRC 0x02
#define TAS_REG_VOL 0x04
#define TAS_REG_TREBLE 0x05
#define TAS_REG_BASS 0x06
#define TAS_REG_LMIX 0x07
#define TAS_REG_RMIX 0x08
#define TAS_REG_ACR 0x40 /* analog control */
# define TAS_ACR_B_MONAUREAL (1<<7)
# define TAS_ACR_B_MON_SEL_RIGHT (1<<6)
# define TAS_ACR_DEEMPH_MASK (3<<2)
# define TAS_ACR_DEEMPH_OFF (0<<2)
# define TAS_ACR_DEEMPH_48KHz (1<<2)
# define TAS_ACR_DEEMPH_44KHz (2<<2)
# define TAS_ACR_INPUT_B (1<<1)
# define TAS_ACR_ANALOG_PDOWN (1<<0)
#define TAS_REG_MCS2 0x43 /* main control 2 */
# define TAS_MCS2_ALLPASS (1<<1)
#define TAS_REG_LEFT_BIQUAD6 0x10
#define TAS_REG_RIGHT_BIQUAD6 0x19
#endif /* __SND_AOA_CODECTASH */
sound/aoa/codecs/snd-aoa-codec-toonie.c 0 → 100644
View file @ f3d9478b
/*
* Apple Onboard Audio driver for Toonie codec
*
* Copyright 2006 Johannes Berg <johannes@sipsolutions.net>
*
* GPL v2, can be found in COPYING.
*
*
* This is a driver for the toonie codec chip. This chip is present
* on the Mac Mini and is nothing but a DAC.
*/
#include <linux/delay.h>
#include <linux/module.h>
MODULE_AUTHOR("Johannes Berg <johannes@sipsolutions.net>");
MODULE_LICENSE("GPL");
MODULE_DESCRIPTION("toonie codec driver for snd-aoa");
#include "../aoa.h"
#include "../soundbus/soundbus.h"
#define PFX "snd-aoa-codec-toonie: "
struct toonie {
struct aoa_codec codec;
};
#define codec_to_toonie(c) container_of(c, struct toonie, codec)
static int toonie_dev_register(struct snd_device *dev)
{
return 0;
}
static struct snd_device_ops ops = {
.dev_register = toonie_dev_register,
};
static struct transfer_info toonie_transfers[] = {
/* This thing *only* has analog output,
* the rates are taken from Info.plist
* from Darwin. */
{
.formats = SNDRV_PCM_FMTBIT_S16_BE |
SNDRV_PCM_FMTBIT_S24_BE,
.rates = SNDRV_PCM_RATE_32000 |
SNDRV_PCM_RATE_44100 |
SNDRV_PCM_RATE_48000 |
SNDRV_PCM_RATE_88200 |
SNDRV_PCM_RATE_96000,
},
{}
};
#ifdef CONFIG_PM
static int toonie_suspend(struct codec_info_item *cii, pm_message_t state)
{
/* can we turn it off somehow? */
return 0;
}
static int toonie_resume(struct codec_info_item *cii)
{
return 0;
}
#endif /* CONFIG_PM */
static struct codec_info toonie_codec_info = {
.transfers = toonie_transfers,
.sysclock_factor = 256,
.bus_factor = 64,
.owner = THIS_MODULE,
#ifdef CONFIG_PM
.suspend = toonie_suspend,
.resume = toonie_resume,
#endif
};
static int toonie_init_codec(struct aoa_codec *codec)
{
struct toonie *toonie = codec_to_toonie(codec);
if (aoa_snd_device_new(SNDRV_DEV_LOWLEVEL, toonie, &ops)) {
printk(KERN_ERR PFX "failed to create toonie snd device!\n");
return -ENODEV;
}
/* nothing connected? what a joke! */
if (toonie->codec.connected != 1)
return -ENOTCONN;
if (toonie->codec.soundbus_dev->attach_codec(toonie->codec.soundbus_dev,
aoa_get_card(),
&toonie_codec_info, toonie)) {
printk(KERN_ERR PFX "error creating toonie pcm\n");
return -ENODEV;
}
return 0;
}
static void toonie_exit_codec(struct aoa_codec *codec)
{
struct toonie *toonie = codec_to_toonie(codec);
if (!toonie->codec.soundbus_dev) {
printk(KERN_ERR PFX "toonie_exit_codec called without soundbus_dev!\n");
return;
}
toonie->codec.soundbus_dev->detach_codec(toonie->codec.soundbus_dev, toonie);
}
static struct toonie *toonie;
static int __init toonie_init(void)
{
toonie = kzalloc(sizeof(struct toonie), GFP_KERNEL);
if (!toonie)
return -ENOMEM;
strlcpy(toonie->codec.name, "toonie", sizeof(toonie->codec.name));
toonie->codec.owner = THIS_MODULE;
toonie->codec.init = toonie_init_codec;
toonie->codec.exit = toonie_exit_codec;
if (aoa_codec_register(&toonie->codec)) {
kfree(toonie);
return -EINVAL;
}
return 0;
}
static void __exit toonie_exit(void)
{
aoa_codec_unregister(&toonie->codec);
kfree(toonie);
}
module_init(toonie_init);
module_exit(toonie_exit);
sound/aoa/core/Makefile 0 → 100644
View file @ f3d9478b
obj-$(CONFIG_SND_AOA) += snd-aoa.o
snd-aoa-objs := snd-aoa-core.o \
snd-aoa-alsa.o \
snd-aoa-gpio-pmf.o \
snd-aoa-gpio-feature.o
sound/aoa/core/snd-aoa-alsa.c 0 → 100644
View file @ f3d9478b
/*
* Apple Onboard Audio Alsa helpers
*
* Copyright 2006 Johannes Berg <johannes@sipsolutions.net>
*
* GPL v2, can be found in COPYING.
*/
#include <linux/module.h>
#include "snd-aoa-alsa.h"
static int index = -1;
module_param(index, int, 0444);
MODULE_PARM_DESC(index, "index for AOA sound card.");
static struct aoa_card *aoa_card;
int aoa_alsa_init(char *name, struct module *mod)
{
struct snd_card *alsa_card;
int err;
if (aoa_card)
/* cannot be EEXIST due to usage in aoa_fabric_register */
return -EBUSY;
alsa_card = snd_card_new(index, name, mod, sizeof(struct aoa_card));
if (!alsa_card)
return -ENOMEM;
aoa_card = alsa_card->private_data;
aoa_card->alsa_card = alsa_card;
strlcpy(alsa_card->driver, "AppleOnbdAudio", sizeof(alsa_card->driver));
strlcpy(alsa_card->shortname, name, sizeof(alsa_card->shortname));
strlcpy(alsa_card->longname, name, sizeof(alsa_card->longname));
strlcpy(alsa_card->mixername, name, sizeof(alsa_card->mixername));
err = snd_card_register(aoa_card->alsa_card);
if (err < 0) {
printk(KERN_ERR "snd-aoa: couldn't register alsa card\n");
snd_card_free(aoa_card->alsa_card);
aoa_card = NULL;
return err;
}
return 0;
}
struct snd_card *aoa_get_card(void)
{
if (aoa_card)
return aoa_card->alsa_card;
return NULL;
}
EXPORT_SYMBOL_GPL(aoa_get_card);
void aoa_alsa_cleanup(void)
{
if (aoa_card) {
snd_card_free(aoa_card->alsa_card);
aoa_card = NULL;
}
}
int aoa_snd_device_new(snd_device_type_t type,
void * device_data, struct snd_device_ops * ops)
{
struct snd_card *card = aoa_get_card();
int err;
if (!card) return -ENOMEM;
err = snd_device_new(card, type, device_data, ops);
if (err) {
printk(KERN_ERR "snd-aoa: failed to create snd device (%d)\n", err);
return err;
}
err = snd_device_register(card, device_data);
if (err) {
printk(KERN_ERR "snd-aoa: failed to register "
"snd device (%d)\n", err);
printk(KERN_ERR "snd-aoa: have you forgotten the "
"dev_register callback?\n");
snd_device_free(card, device_data);
}
return err;
}
EXPORT_SYMBOL_GPL(aoa_snd_device_new);
int aoa_snd_ctl_add(struct snd_kcontrol* control)
{
int err;
if (!aoa_card) return -ENODEV;
err = snd_ctl_add(aoa_card->alsa_card, control);
if (err)
printk(KERN_ERR "snd-aoa: failed to add alsa control (%d)\n",
err);
return err;
}
EXPORT_SYMBOL_GPL(aoa_snd_ctl_add);
sound/aoa/core/snd-aoa-alsa.h 0 → 100644
View file @ f3d9478b
/*
* Apple Onboard Audio Alsa private helpers
*
* Copyright 2006 Johannes Berg <johannes@sipsolutions.net>
*
* GPL v2, can be found in COPYING.
*/
#ifndef __SND_AOA_ALSA_H
#define __SND_AOA_ALSA_H
#include "../aoa.h"
extern int aoa_alsa_init(char *name, struct module *mod);
extern void aoa_alsa_cleanup(void);
#endif /* __SND_AOA_ALSA_H */
sound/aoa/core/snd-aoa-core.c 0 → 100644
View file @ f3d9478b
/*
* Apple Onboard Audio driver core
*
* Copyright 2006 Johannes Berg <johannes@sipsolutions.net>
*
* GPL v2, can be found in COPYING.
*/
#include <linux/init.h>
#include <linux/module.h>
#include <linux/list.h>
#include "../aoa.h"
#include "snd-aoa-alsa.h"
MODULE_DESCRIPTION("Apple Onboard Audio Sound Driver");
MODULE_AUTHOR("Johannes Berg <johannes@sipsolutions.net>");
MODULE_LICENSE("GPL");
/* We allow only one fabric. This simplifies things,
* and more don't really make that much sense */
static struct aoa_fabric *fabric;
static LIST_HEAD(codec_list);
static int attach_codec_to_fabric(struct aoa_codec *c)
{
int err;
if (!try_module_get(c->owner))
return -EBUSY;
/* found_codec has to be assigned */
err = -ENOENT;
if (fabric->found_codec)
err = fabric->found_codec(c);
if (err) {
module_put(c->owner);
printk(KERN_ERR "snd-aoa: fabric didn't like codec %s\n",
c->name);
return err;
}
c->fabric = fabric;
err = 0;
if (c->init)
err = c->init(c);
if (err) {
printk(KERN_ERR "snd-aoa: codec %s didn't init\n", c->name);
c->fabric = NULL;
if (fabric->remove_codec)
fabric->remove_codec(c);
module_put(c->owner);
return err;
}
if (fabric->attached_codec)
fabric->attached_codec(c);
return 0;
}
int aoa_codec_register(struct aoa_codec *codec)
{
int err = 0;
/* if there's a fabric already, we can tell if we
* will want to have this codec, so propagate error
* through. Otherwise, this will happen later... */
if (fabric)
err = attach_codec_to_fabric(codec);
if (!err)
list_add(&codec->list, &codec_list);
return err;
}
EXPORT_SYMBOL_GPL(aoa_codec_register);
void aoa_codec_unregister(struct aoa_codec *codec)
{
list_del(&codec->list);
if (codec->fabric && codec->exit)
codec->exit(codec);
if (fabric && fabric->remove_codec)
fabric->remove_codec(codec);
codec->fabric = NULL;
module_put(codec->owner);
}
EXPORT_SYMBOL_GPL(aoa_codec_unregister);
int aoa_fabric_register(struct aoa_fabric *new_fabric)
{
struct aoa_codec *c;
int err;
/* allow querying for presence of fabric
* (i.e. do this test first!) */
if (new_fabric == fabric) {
err = -EALREADY;
goto attach;
}
if (fabric)
return -EEXIST;
if (!new_fabric)
return -EINVAL;
err = aoa_alsa_init(new_fabric->name, new_fabric->owner);
if (err)
return err;
fabric = new_fabric;
attach:
list_for_each_entry(c, &codec_list, list) {
if (c->fabric != fabric)
attach_codec_to_fabric(c);
}
return err;
}
EXPORT_SYMBOL_GPL(aoa_fabric_register);
void aoa_fabric_unregister(struct aoa_fabric *old_fabric)
{
struct aoa_codec *c;
if (fabric != old_fabric)
return;
list_for_each_entry(c, &codec_list, list) {
if (c->fabric)
aoa_fabric_unlink_codec(c);
}
aoa_alsa_cleanup();
fabric = NULL;
}
EXPORT_SYMBOL_GPL(aoa_fabric_unregister);
void aoa_fabric_unlink_codec(struct aoa_codec *codec)
{
if (!codec->fabric) {
printk(KERN_ERR "snd-aoa: fabric unassigned "
"in aoa_fabric_unlink_codec\n");
dump_stack();
return;
}
if (codec->exit)
codec->exit(codec);
if (codec->fabric->remove_codec)
codec->fabric->remove_codec(codec);
codec->fabric = NULL;
module_put(codec->owner);
}
EXPORT_SYMBOL_GPL(aoa_fabric_unlink_codec);
static int __init aoa_init(void)
{
return 0;
}
static void __exit aoa_exit(void)
{
aoa_alsa_cleanup();
}
module_init(aoa_init);
module_exit(aoa_exit);
sound/aoa/core/snd-aoa-gpio-feature.c 0 → 100644
View file @ f3d9478b
/*
* Apple Onboard Audio feature call GPIO control
*
* Copyright 2006 Johannes Berg <johannes@sipsolutions.net>
*
* GPL v2, can be found in COPYING.
*
* This file contains the GPIO control routines for
* direct (through feature calls) access to the GPIO
* registers.
*/
#include <asm/pmac_feature.h>
#include <linux/interrupt.h>
#include "../aoa.h"
/* TODO: these are 20 global variables
* that aren't used on most machines...
* Move them into a dynamically allocated
* structure and use that.
*/
/* these are the GPIO numbers (register addresses as offsets into
* the GPIO space) */
static int headphone_mute_gpio;
static int amp_mute_gpio;
static int lineout_mute_gpio;
static int hw_reset_gpio;
static int lineout_detect_gpio;
static int headphone_detect_gpio;
static int linein_detect_gpio;
/* see the SWITCH_GPIO macro */
static int headphone_mute_gpio_activestate;
static int amp_mute_gpio_activestate;
static int lineout_mute_gpio_activestate;
static int hw_reset_gpio_activestate;
static int lineout_detect_gpio_activestate;
static int headphone_detect_gpio_activestate;
static int linein_detect_gpio_activestate;
/* node pointers that we save when getting the GPIO number
* to get the interrupt later */
static struct device_node *lineout_detect_node;
static struct device_node *linein_detect_node;
static struct device_node *headphone_detect_node;
static int lineout_detect_irq;
static int linein_detect_irq;
static int headphone_detect_irq;
static struct device_node *get_gpio(char *name,
char *altname,
int *gpioptr,
int *gpioactiveptr)
{
struct device_node *np, *gpio;
u32 *reg;
char *audio_gpio;
*gpioptr = -1;
/* check if we can get it the easy way ... */
np = of_find_node_by_name(NULL, name);
if (!np) {
/* some machines have only gpioX/extint-gpioX nodes,
* and an audio-gpio property saying what it is ...
* So what we have to do is enumerate all children
* of the gpio node and check them all. */
gpio = of_find_node_by_name(NULL, "gpio");
if (!gpio)
return NULL;
while ((np = of_get_next_child(gpio, np))) {
audio_gpio = get_property(np, "audio-gpio", NULL);
if (!audio_gpio)
continue;
if (strcmp(audio_gpio, name) == 0)
break;
if (altname && (strcmp(audio_gpio, altname) == 0))
break;
}
/* still not found, assume not there */
if (!np)
return NULL;
}
reg = (u32 *)get_property(np, "reg", NULL);
if (!reg)
return NULL;
*gpioptr = *reg;
/* this is a hack, usually the GPIOs 'reg' property
* should have the offset based from the GPIO space
* which is at 0x50, but apparently not always... */
if (*gpioptr < 0x50)
*gpioptr += 0x50;
reg = (u32 *)get_property(np, "audio-gpio-active-state", NULL);
if (!reg)
/* Apple seems to default to 1, but
* that doesn't seem right at least on most
* machines. So until proven that the opposite
* is necessary, we default to 0
* (which, incidentally, snd-powermac also does...) */
*gpioactiveptr = 0;
else
*gpioactiveptr = *reg;
return np;
}
static void get_irq(struct device_node * np, int *irqptr)
{
*irqptr = -1;
if (!np)
return;
if (np->n_intrs != 1)
return;
*irqptr = np->intrs[0].line;
}
/* 0x4 is outenable, 0x1 is out, thus 4 or 5 */
#define SWITCH_GPIO(name, v, on) \
(((v)&~1) | ((on)? \
(name##_gpio_activestate==0?4:5): \
(name##_gpio_activestate==0?5:4)))
#define FTR_GPIO(name, bit) \
static void ftr_gpio_set_##name(struct gpio_runtime *rt, int on)\
{ \
int v; \
\
if (unlikely(!rt)) return; \
\
if (name##_mute_gpio < 0) \
return; \
\
v = pmac_call_feature(PMAC_FTR_READ_GPIO, NULL, \
name##_mute_gpio, \
0); \
\
/* muted = !on... */ \
v = SWITCH_GPIO(name##_mute, v, !on); \
\
pmac_call_feature(PMAC_FTR_WRITE_GPIO, NULL, \
name##_mute_gpio, v); \
\
rt->implementation_private &= ~(1<<bit); \
rt->implementation_private |= (!!on << bit); \
} \
static int ftr_gpio_get_##name(struct gpio_runtime *rt) \
{ \
if (unlikely(!rt)) return 0; \
return (rt->implementation_private>>bit)&1; \
}
FTR_GPIO(headphone, 0);
FTR_GPIO(amp, 1);
FTR_GPIO(lineout, 2);
static void ftr_gpio_set_hw_reset(struct gpio_runtime *rt, int on)
{
int v;
if (unlikely(!rt)) return;
if (hw_reset_gpio < 0)
return;
v = pmac_call_feature(PMAC_FTR_READ_GPIO, NULL,
hw_reset_gpio, 0);
v = SWITCH_GPIO(hw_reset, v, on);
pmac_call_feature(PMAC_FTR_WRITE_GPIO, NULL,
hw_reset_gpio, v);
}
static void ftr_gpio_all_amps_off(struct gpio_runtime *rt)
{
int saved;
if (unlikely(!rt)) return;
saved = rt->implementation_private;
ftr_gpio_set_headphone(rt, 0);
ftr_gpio_set_amp(rt, 0);
ftr_gpio_set_lineout(rt, 0);
rt->implementation_private = saved;
}
static void ftr_gpio_all_amps_restore(struct gpio_runtime *rt)
{
int s;
if (unlikely(!rt)) return;
s = rt->implementation_private;
ftr_gpio_set_headphone(rt, (s>>0)&1);
ftr_gpio_set_amp(rt, (s>>1)&1);
ftr_gpio_set_lineout(rt, (s>>2)&1);
}
static void ftr_handle_notify(void *data)
{
struct gpio_notification *notif = data;
mutex_lock(&notif->mutex);
if (notif->notify)
notif->notify(notif->data);
mutex_unlock(&notif->mutex);
}
static void ftr_gpio_init(struct gpio_runtime *rt)
{
get_gpio("headphone-mute", NULL,
&headphone_mute_gpio,
&headphone_mute_gpio_activestate);
get_gpio("amp-mute", NULL,
&amp_mute_gpio,
&amp_mute_gpio_activestate);
get_gpio("lineout-mute", NULL,
&lineout_mute_gpio,
&lineout_mute_gpio_activestate);
get_gpio("hw-reset", "audio-hw-reset",
&hw_reset_gpio,
&hw_reset_gpio_activestate);
headphone_detect_node = get_gpio("headphone-detect", NULL,
&headphone_detect_gpio,
&headphone_detect_gpio_activestate);
/* go Apple, and thanks for giving these different names
* across the board... */
lineout_detect_node = get_gpio("lineout-detect", "line-output-detect",
&lineout_detect_gpio,
&lineout_detect_gpio_activestate);
linein_detect_node = get_gpio("linein-detect", "line-input-detect",
&linein_detect_gpio,
&linein_detect_gpio_activestate);
get_irq(headphone_detect_node, &headphone_detect_irq);
get_irq(lineout_detect_node, &lineout_detect_irq);
get_irq(linein_detect_node, &linein_detect_irq);
ftr_gpio_all_amps_off(rt);
rt->implementation_private = 0;
INIT_WORK(&rt->headphone_notify.work, ftr_handle_notify,
&rt->headphone_notify);
INIT_WORK(&rt->line_in_notify.work, ftr_handle_notify,
&rt->line_in_notify);
INIT_WORK(&rt->line_out_notify.work, ftr_handle_notify,
&rt->line_out_notify);
mutex_init(&rt->headphone_notify.mutex);
mutex_init(&rt->line_in_notify.mutex);
mutex_init(&rt->line_out_notify.mutex);
}
static void ftr_gpio_exit(struct gpio_runtime *rt)
{
ftr_gpio_all_amps_off(rt);
rt->implementation_private = 0;
if (rt->headphone_notify.notify)
free_irq(headphone_detect_irq, &rt->headphone_notify);
if (rt->line_in_notify.gpio_private)
free_irq(linein_detect_irq, &rt->line_in_notify);
if (rt->line_out_notify.gpio_private)
free_irq(lineout_detect_irq, &rt->line_out_notify);
cancel_delayed_work(&rt->headphone_notify.work);
cancel_delayed_work(&rt->line_in_notify.work);
cancel_delayed_work(&rt->line_out_notify.work);
flush_scheduled_work();
mutex_destroy(&rt->headphone_notify.mutex);
mutex_destroy(&rt->line_in_notify.mutex);
mutex_destroy(&rt->line_out_notify.mutex);
}
static irqreturn_t ftr_handle_notify_irq(int xx,
void *data,
struct pt_regs *regs)
{
struct gpio_notification *notif = data;
schedule_work(&notif->work);
return IRQ_HANDLED;
}
static int ftr_set_notify(struct gpio_runtime *rt,
enum notify_type type,
notify_func_t notify,
void *data)
{
struct gpio_notification *notif;
notify_func_t old;
int irq;
char *name;
int err = -EBUSY;
switch (type) {
case AOA_NOTIFY_HEADPHONE:
notif = &rt->headphone_notify;
name = "headphone-detect";
irq = headphone_detect_irq;
break;
case AOA_NOTIFY_LINE_IN:
notif = &rt->line_in_notify;
name = "linein-detect";
irq = linein_detect_irq;
break;
case AOA_NOTIFY_LINE_OUT:
notif = &rt->line_out_notify;
name = "lineout-detect";
irq = lineout_detect_irq;
break;
default:
return -EINVAL;
}
if (irq == -1)
return -ENODEV;
mutex_lock(&notif->mutex);
old = notif->notify;
if (!old && !notify) {
err = 0;
goto out_unlock;
}
if (old && notify) {
if (old == notify && notif->data == data)
err = 0;
goto out_unlock;
}
if (old && !notify)
free_irq(irq, notif);
if (!old && notify) {
err = request_irq(irq, ftr_handle_notify_irq, 0, name, notif);
if (err)
goto out_unlock;
}
notif->notify = notify;
notif->data = data;
err = 0;
out_unlock:
mutex_unlock(&notif->mutex);
return err;
}
static int ftr_get_detect(struct gpio_runtime *rt,
enum notify_type type)
{
int gpio, ret, active;
switch (type) {
case AOA_NOTIFY_HEADPHONE:
gpio = headphone_detect_gpio;
active = headphone_detect_gpio_activestate;
break;
case AOA_NOTIFY_LINE_IN:
gpio = linein_detect_gpio;
active = linein_detect_gpio_activestate;
break;
case AOA_NOTIFY_LINE_OUT:
gpio = lineout_detect_gpio;
active = lineout_detect_gpio_activestate;
break;
default:
return -EINVAL;
}
if (gpio == -1)
return -ENODEV;
ret = pmac_call_feature(PMAC_FTR_READ_GPIO, NULL, gpio, 0);
if (ret < 0)
return ret;
return ((ret >> 1) & 1) == active;
}
static struct gpio_methods methods = {
.init = ftr_gpio_init,
.exit = ftr_gpio_exit,
.all_amps_off = ftr_gpio_all_amps_off,
.all_amps_restore = ftr_gpio_all_amps_restore,
.set_headphone = ftr_gpio_set_headphone,
.set_speakers = ftr_gpio_set_amp,
.set_lineout = ftr_gpio_set_lineout,
.set_hw_reset = ftr_gpio_set_hw_reset,
.get_headphone = ftr_gpio_get_headphone,
.get_speakers = ftr_gpio_get_amp,
.get_lineout = ftr_gpio_get_lineout,
.set_notify = ftr_set_notify,
.get_detect = ftr_get_detect,
};
struct gpio_methods *ftr_gpio_methods = &methods;
EXPORT_SYMBOL_GPL(ftr_gpio_methods);
sound/aoa/core/snd-aoa-gpio-pmf.c 0 → 100644
View file @ f3d9478b
/*
* Apple Onboard Audio pmf GPIOs
*
* Copyright 2006 Johannes Berg <johannes@sipsolutions.net>
*
* GPL v2, can be found in COPYING.
*/
#include <asm/pmac_feature.h>
#include <asm/pmac_pfunc.h>
#include "../aoa.h"
#define PMF_GPIO(name, bit) \
static void pmf_gpio_set_##name(struct gpio_runtime *rt, int on)\
{ \
struct pmf_args args = { .count = 1, .u[0].v = !on }; \
\
if (unlikely(!rt)) return; \
pmf_call_function(rt->node, #name "-mute", &args); \
rt->implementation_private &= ~(1<<bit); \
rt->implementation_private |= (!!on << bit); \
} \
static int pmf_gpio_get_##name(struct gpio_runtime *rt) \
{ \
if (unlikely(!rt)) return 0; \
return (rt->implementation_private>>bit)&1; \
}
PMF_GPIO(headphone, 0);
PMF_GPIO(amp, 1);
PMF_GPIO(lineout, 2);
static void pmf_gpio_set_hw_reset(struct gpio_runtime *rt, int on)
{
struct pmf_args args = { .count = 1, .u[0].v = !!on };
if (unlikely(!rt)) return;
pmf_call_function(rt->node, "hw-reset", &args);
}
static void pmf_gpio_all_amps_off(struct gpio_runtime *rt)
{
int saved;
if (unlikely(!rt)) return;
saved = rt->implementation_private;
pmf_gpio_set_headphone(rt, 0);
pmf_gpio_set_amp(rt, 0);
pmf_gpio_set_lineout(rt, 0);
rt->implementation_private = saved;
}
static void pmf_gpio_all_amps_restore(struct gpio_runtime *rt)
{
int s;
if (unlikely(!rt)) return;
s = rt->implementation_private;
pmf_gpio_set_headphone(rt, (s>>0)&1);
pmf_gpio_set_amp(rt, (s>>1)&1);
pmf_gpio_set_lineout(rt, (s>>2)&1);
}
static void pmf_handle_notify(void *data)
{
struct gpio_notification *notif = data;
mutex_lock(&notif->mutex);
if (notif->notify)
notif->notify(notif->data);
mutex_unlock(&notif->mutex);
}
static void pmf_gpio_init(struct gpio_runtime *rt)
{
pmf_gpio_all_amps_off(rt);
rt->implementation_private = 0;
INIT_WORK(&rt->headphone_notify.work, pmf_handle_notify,
&rt->headphone_notify);
INIT_WORK(&rt->line_in_notify.work, pmf_handle_notify,
&rt->line_in_notify);
INIT_WORK(&rt->line_out_notify.work, pmf_handle_notify,
&rt->line_out_notify);
mutex_init(&rt->headphone_notify.mutex);
mutex_init(&rt->line_in_notify.mutex);
mutex_init(&rt->line_out_notify.mutex);
}
static void pmf_gpio_exit(struct gpio_runtime *rt)
{
pmf_gpio_all_amps_off(rt);
rt->implementation_private = 0;
if (rt->headphone_notify.gpio_private)
pmf_unregister_irq_client(rt->headphone_notify.gpio_private);
if (rt->line_in_notify.gpio_private)
pmf_unregister_irq_client(rt->line_in_notify.gpio_private);
if (rt->line_out_notify.gpio_private)
pmf_unregister_irq_client(rt->line_out_notify.gpio_private);
/* make sure no work is pending before freeing
* all things */
cancel_delayed_work(&rt->headphone_notify.work);
cancel_delayed_work(&rt->line_in_notify.work);
cancel_delayed_work(&rt->line_out_notify.work);
flush_scheduled_work();
mutex_destroy(&rt->headphone_notify.mutex);
mutex_destroy(&rt->line_in_notify.mutex);
mutex_destroy(&rt->line_out_notify.mutex);
if (rt->headphone_notify.gpio_private)
kfree(rt->headphone_notify.gpio_private);
if (rt->line_in_notify.gpio_private)
kfree(rt->line_in_notify.gpio_private);
if (rt->line_out_notify.gpio_private)
kfree(rt->line_out_notify.gpio_private);
}
static void pmf_handle_notify_irq(void *data)
{
struct gpio_notification *notif = data;
schedule_work(&notif->work);
}
static int pmf_set_notify(struct gpio_runtime *rt,
enum notify_type type,
notify_func_t notify,
void *data)
{
struct gpio_notification *notif;
notify_func_t old;
struct pmf_irq_client *irq_client;
char *name;
int err = -EBUSY;
switch (type) {
case AOA_NOTIFY_HEADPHONE:
notif = &rt->headphone_notify;
name = "headphone-detect";
break;
case AOA_NOTIFY_LINE_IN:
notif = &rt->line_in_notify;
name = "linein-detect";
break;
case AOA_NOTIFY_LINE_OUT:
notif = &rt->line_out_notify;
name = "lineout-detect";
break;
default:
return -EINVAL;
}
mutex_lock(&notif->mutex);
old = notif->notify;
if (!old && !notify) {
err = 0;
goto out_unlock;
}
if (old && notify) {
if (old == notify && notif->data == data)
err = 0;
goto out_unlock;
}
if (old && !notify) {
irq_client = notif->gpio_private;
pmf_unregister_irq_client(irq_client);
kfree(irq_client);
notif->gpio_private = NULL;
}
if (!old && notify) {
irq_client = kzalloc(sizeof(struct pmf_irq_client),
GFP_KERNEL);
irq_client->data = notif;
irq_client->handler = pmf_handle_notify_irq;
irq_client->owner = THIS_MODULE;
err = pmf_register_irq_client(rt->node,
name,
irq_client);
if (err) {
printk(KERN_ERR "snd-aoa: gpio layer failed to"
" register %s irq (%d)\n", name, err);
kfree(irq_client);
goto out_unlock;
}
notif->gpio_private = irq_client;
}
notif->notify = notify;
notif->data = data;
err = 0;
out_unlock:
mutex_unlock(&notif->mutex);
return err;
}
static int pmf_get_detect(struct gpio_runtime *rt,
enum notify_type type)
{
char *name;
int err = -EBUSY, ret;
struct pmf_args args = { .count = 1, .u[0].p = &ret };
switch (type) {
case AOA_NOTIFY_HEADPHONE:
name = "headphone-detect";
break;
case AOA_NOTIFY_LINE_IN:
name = "linein-detect";
break;
case AOA_NOTIFY_LINE_OUT:
name = "lineout-detect";
break;
default:
return -EINVAL;
}
err = pmf_call_function(rt->node, name, &args);
if (err)
return err;
return ret;
}
static struct gpio_methods methods = {
.init = pmf_gpio_init,
.exit = pmf_gpio_exit,
.all_amps_off = pmf_gpio_all_amps_off,
.all_amps_restore = pmf_gpio_all_amps_restore,
.set_headphone = pmf_gpio_set_headphone,
.set_speakers = pmf_gpio_set_amp,
.set_lineout = pmf_gpio_set_lineout,
.set_hw_reset = pmf_gpio_set_hw_reset,
.get_headphone = pmf_gpio_get_headphone,
.get_speakers = pmf_gpio_get_amp,
.get_lineout = pmf_gpio_get_lineout,
.set_notify = pmf_set_notify,
.get_detect = pmf_get_detect,
};
struct gpio_methods *pmf_gpio_methods = &methods;
EXPORT_SYMBOL_GPL(pmf_gpio_methods);
sound/aoa/fabrics/Kconfig 0 → 100644
View file @ f3d9478b
config SND_AOA_FABRIC_LAYOUT
tristate "layout-id fabric"
depends SND_AOA
select SND_AOA_SOUNDBUS
select SND_AOA_SOUNDBUS_I2S
---help---
This enables the layout-id fabric for the Apple Onboard
Audio driver, the module holding it all together
based on the device-tree's layout-id property.
If you are unsure and have a later Apple machine,
compile it as a module.
sound/aoa/fabrics/Makefile 0 → 100644
View file @ f3d9478b
obj-$(CONFIG_SND_AOA_FABRIC_LAYOUT) += snd-aoa-fabric-layout.o
sound/aoa/fabrics/snd-aoa-fabric-layout.c 0 → 100644
View file @ f3d9478b
/*
* Apple Onboard Audio driver -- layout fabric
*
* Copyright 2006 Johannes Berg <johannes@sipsolutions.net>
*
* GPL v2, can be found in COPYING.
*
*
* This fabric module looks for sound codecs
* based on the layout-id property in the device tree.
*
*/
#include <asm/prom.h>
#include <linux/list.h>
#include <linux/module.h>
#include "../aoa.h"
#include "../soundbus/soundbus.h"
MODULE_AUTHOR("Johannes Berg <johannes@sipsolutions.net>");
MODULE_LICENSE("GPL");
MODULE_DESCRIPTION("Layout-ID fabric for snd-aoa");
#define MAX_CODECS_PER_BUS 2
/* These are the connections the layout fabric
* knows about. It doesn't really care about the
* input ones, but I thought I'd separate them
* to give them proper names. The thing is that
* Apple usually will distinguish the active output
* by GPIOs, while the active input is set directly
* on the codec. Hence we here tell the codec what
* we think is connected. This information is hard-
* coded below ... */
#define CC_SPEAKERS (1<<0)
#define CC_HEADPHONE (1<<1)
#define CC_LINEOUT (1<<2)
#define CC_DIGITALOUT (1<<3)
#define CC_LINEIN (1<<4)
#define CC_MICROPHONE (1<<5)
#define CC_DIGITALIN (1<<6)
/* pretty bogus but users complain...
* This is a flag saying that the LINEOUT
* should be renamed to HEADPHONE.
* be careful with input detection! */
#define CC_LINEOUT_LABELLED_HEADPHONE (1<<7)
struct codec_connection {
/* CC_ flags from above */
int connected;
/* codec dependent bit to be set in the aoa_codec.connected field.
* This intentionally doesn't have any generic flags because the
* fabric has to know the codec anyway and all codecs might have
* different connectors */
int codec_bit;
};
struct codec_connect_info {
char *name;
struct codec_connection *connections;
};
#define LAYOUT_FLAG_COMBO_LINEOUT_SPDIF (1<<0)
struct layout {
unsigned int layout_id;
struct codec_connect_info codecs[MAX_CODECS_PER_BUS];
int flags;
/* if busname is not assigned, we use 'Master' below,
* so that our layout table doesn't need to be filled
* too much.
* We only assign these two if we expect to find more
* than one soundbus, i.e. on those machines with
* multiple layout-ids */
char *busname;
int pcmid;
};
MODULE_ALIAS("sound-layout-41");
MODULE_ALIAS("sound-layout-45");
MODULE_ALIAS("sound-layout-51");
MODULE_ALIAS("sound-layout-58");
MODULE_ALIAS("sound-layout-60");
MODULE_ALIAS("sound-layout-61");
MODULE_ALIAS("sound-layout-64");
MODULE_ALIAS("sound-layout-65");
MODULE_ALIAS("sound-layout-68");
MODULE_ALIAS("sound-layout-69");
MODULE_ALIAS("sound-layout-70");
MODULE_ALIAS("sound-layout-72");
MODULE_ALIAS("sound-layout-80");
MODULE_ALIAS("sound-layout-82");
MODULE_ALIAS("sound-layout-84");
MODULE_ALIAS("sound-layout-86");
MODULE_ALIAS("sound-layout-92");
/* onyx with all but microphone connected */
static struct codec_connection onyx_connections_nomic[] = {
{
.connected = CC_SPEAKERS | CC_HEADPHONE | CC_LINEOUT,
.codec_bit = 0,
},
{
.connected = CC_DIGITALOUT,
.codec_bit = 1,
},
{
.connected = CC_LINEIN,
.codec_bit = 2,
},
{} /* terminate array by .connected == 0 */
};
/* onyx on machines without headphone */
static struct codec_connection onyx_connections_noheadphones[] = {
{
.connected = CC_SPEAKERS | CC_LINEOUT |
CC_LINEOUT_LABELLED_HEADPHONE,
.codec_bit = 0,
},
{
.connected = CC_DIGITALOUT,
.codec_bit = 1,
},
/* FIXME: are these correct? probably not for all the machines
* below ... If not this will need separating. */
{
.connected = CC_LINEIN,
.codec_bit = 2,
},
{
.connected = CC_MICROPHONE,
.codec_bit = 3,
},
{} /* terminate array by .connected == 0 */
};
/* onyx on machines with real line-out */
static struct codec_connection onyx_connections_reallineout[] = {
{
.connected = CC_SPEAKERS | CC_LINEOUT | CC_HEADPHONE,
.codec_bit = 0,
},
{
.connected = CC_DIGITALOUT,
.codec_bit = 1,
},
{
.connected = CC_LINEIN,
.codec_bit = 2,
},
{} /* terminate array by .connected == 0 */
};
/* tas on machines without line out */
static struct codec_connection tas_connections_nolineout[] = {
{
.connected = CC_SPEAKERS | CC_HEADPHONE,
.codec_bit = 0,
},
{
.connected = CC_LINEIN,
.codec_bit = 2,
},
{
.connected = CC_MICROPHONE,
.codec_bit = 3,
},
{} /* terminate array by .connected == 0 */
};
/* tas on machines with neither line out nor line in */
static struct codec_connection tas_connections_noline[] = {
{
.connected = CC_SPEAKERS | CC_HEADPHONE,
.codec_bit = 0,
},
{
.connected = CC_MICROPHONE,
.codec_bit = 3,
},
{} /* terminate array by .connected == 0 */
};
/* tas on machines without microphone */
static struct codec_connection tas_connections_nomic[] = {
{
.connected = CC_SPEAKERS | CC_HEADPHONE | CC_LINEOUT,
.codec_bit = 0,
},
{
.connected = CC_LINEIN,
.codec_bit = 2,
},
{} /* terminate array by .connected == 0 */
};
/* tas on machines with everything connected */
static struct codec_connection tas_connections_all[] = {
{
.connected = CC_SPEAKERS | CC_HEADPHONE | CC_LINEOUT,
.codec_bit = 0,
},
{
.connected = CC_LINEIN,
.codec_bit = 2,
},
{
.connected = CC_MICROPHONE,
.codec_bit = 3,
},
{} /* terminate array by .connected == 0 */
};
static struct codec_connection toonie_connections[] = {
{
.connected = CC_SPEAKERS | CC_HEADPHONE,
.codec_bit = 0,
},
{} /* terminate array by .connected == 0 */
};
static struct codec_connection topaz_input[] = {
{
.connected = CC_DIGITALIN,
.codec_bit = 0,
},
{} /* terminate array by .connected == 0 */
};
static struct codec_connection topaz_output[] = {
{
.connected = CC_DIGITALOUT,
.codec_bit = 1,
},
{} /* terminate array by .connected == 0 */
};
static struct codec_connection topaz_inout[] = {
{
.connected = CC_DIGITALIN,
.codec_bit = 0,
},
{
.connected = CC_DIGITALOUT,
.codec_bit = 1,
},
{} /* terminate array by .connected == 0 */
};
static struct layout layouts[] = {
/* last PowerBooks (15" Oct 2005) */
{ .layout_id = 82,
.flags = LAYOUT_FLAG_COMBO_LINEOUT_SPDIF,
.codecs[0] = {
.name = "onyx",
.connections = onyx_connections_noheadphones,
},
.codecs[1] = {
.name = "topaz",
.connections = topaz_input,
},
},
/* PowerMac9,1 */
{ .layout_id = 60,
.codecs[0] = {
.name = "onyx",
.connections = onyx_connections_reallineout,
},
},
/* PowerMac9,1 */
{ .layout_id = 61,
.codecs[0] = {
.name = "topaz",
.connections = topaz_input,
},
},
/* PowerBook5,7 */
{ .layout_id = 64,
.flags = LAYOUT_FLAG_COMBO_LINEOUT_SPDIF,
.codecs[0] = {
.name = "onyx",
.connections = onyx_connections_noheadphones,
},
},
/* PowerBook5,7 */
{ .layout_id = 65,
.codecs[0] = {
.name = "topaz",
.connections = topaz_input,
},
},
/* PowerBook5,9 [17" Oct 2005] */
{ .layout_id = 84,
.flags = LAYOUT_FLAG_COMBO_LINEOUT_SPDIF,
.codecs[0] = {
.name = "onyx",
.connections = onyx_connections_noheadphones,
},
.codecs[1] = {
.name = "topaz",
.connections = topaz_input,
},
},
/* PowerMac8,1 */
{ .layout_id = 45,
.codecs[0] = {
.name = "onyx",
.connections = onyx_connections_noheadphones,
},
.codecs[1] = {
.name = "topaz",
.connections = topaz_input,
},
},
/* Quad PowerMac (analog in, analog/digital out) */
{ .layout_id = 68,
.codecs[0] = {
.name = "onyx",
.connections = onyx_connections_nomic,
},
},
/* Quad PowerMac (digital in) */
{ .layout_id = 69,
.codecs[0] = {
.name = "topaz",
.connections = topaz_input,
},
.busname = "digital in", .pcmid = 1 },
/* Early 2005 PowerBook (PowerBook 5,6) */
{ .layout_id = 70,
.codecs[0] = {
.name = "tas",
.connections = tas_connections_nolineout,
},
},
/* PowerBook 5,4 */
{ .layout_id = 51,
.codecs[0] = {
.name = "tas",
.connections = tas_connections_nolineout,
},
},
/* PowerBook6,7 */
{ .layout_id = 80,
.codecs[0] = {
.name = "tas",
.connections = tas_connections_noline,
},
},
/* PowerBook6,8 */
{ .layout_id = 72,
.codecs[0] = {
.name = "tas",
.connections = tas_connections_nolineout,
},
},
/* PowerMac8,2 */
{ .layout_id = 86,
.codecs[0] = {
.name = "onyx",
.connections = onyx_connections_nomic,
},
.codecs[1] = {
.name = "topaz",
.connections = topaz_input,
},
},
/* PowerBook6,7 */
{ .layout_id = 92,
.codecs[0] = {
.name = "tas",
.connections = tas_connections_nolineout,
},
},
/* PowerMac10,1 (Mac Mini) */
{ .layout_id = 58,
.codecs[0] = {
.name = "toonie",
.connections = toonie_connections,
},
},
/* unknown, untested, but this comes from Apple */
{ .layout_id = 41,
.codecs[0] = {
.name = "tas",
.connections = tas_connections_all,
},
},
{ .layout_id = 36,
.codecs[0] = {
.name = "tas",
.connections = tas_connections_nomic,
},
.codecs[1] = {
.name = "topaz",
.connections = topaz_inout,
},
},
{ .layout_id = 47,
.codecs[0] = {
.name = "onyx",
.connections = onyx_connections_noheadphones,
},
},
{ .layout_id = 48,
.codecs[0] = {
.name = "topaz",
.connections = topaz_input,
},
},
{ .layout_id = 49,
.codecs[0] = {
.name = "onyx",
.connections = onyx_connections_nomic,
},
},
{ .layout_id = 50,
.codecs[0] = {
.name = "topaz",
.connections = topaz_input,
},
},
{ .layout_id = 56,
.codecs[0] = {
.name = "onyx",
.connections = onyx_connections_noheadphones,
},
},
{ .layout_id = 57,
.codecs[0] = {
.name = "topaz",
.connections = topaz_input,
},
},
{ .layout_id = 62,
.codecs[0] = {
.name = "onyx",
.connections = onyx_connections_noheadphones,
},
.codecs[1] = {
.name = "topaz",
.connections = topaz_output,
},
},
{ .layout_id = 66,
.codecs[0] = {
.name = "onyx",
.connections = onyx_connections_noheadphones,
},
},
{ .layout_id = 67,
.codecs[0] = {
.name = "topaz",
.connections = topaz_input,
},
},
{ .layout_id = 76,
.codecs[0] = {
.name = "tas",
.connections = tas_connections_nomic,
},
.codecs[1] = {
.name = "topaz",
.connections = topaz_inout,
},
},
{ .layout_id = 90,
.codecs[0] = {
.name = "tas",
.connections = tas_connections_noline,
},
},
{ .layout_id = 94,
.codecs[0] = {
.name = "onyx",
/* but it has an external mic?? how to select? */
.connections = onyx_connections_noheadphones,
},
},
{ .layout_id = 96,
.codecs[0] = {
.name = "onyx",
.connections = onyx_connections_noheadphones,
},
},
{ .layout_id = 98,
.codecs[0] = {
.name = "toonie",
.connections = toonie_connections,
},
},
{ .layout_id = 100,
.codecs[0] = {
.name = "topaz",
.connections = topaz_input,
},
.codecs[1] = {
.name = "onyx",
.connections = onyx_connections_noheadphones,
},
},
{}
};
static struct layout *find_layout_by_id(unsigned int id)
{
struct layout *l;
l = layouts;
while (l->layout_id) {
if (l->layout_id == id)
return l;
l++;
}
return NULL;
}
static void use_layout(struct layout *l)
{
int i;
for (i=0; i<MAX_CODECS_PER_BUS; i++) {
if (l->codecs[i].name) {
request_module("snd-aoa-codec-%s", l->codecs[i].name);
}
}
/* now we wait for the codecs to call us back */
}
struct layout_dev;
struct layout_dev_ptr {
struct layout_dev *ptr;
};
struct layout_dev {
struct list_head list;
struct soundbus_dev *sdev;
struct device_node *sound;
struct aoa_codec *codecs[MAX_CODECS_PER_BUS];
struct layout *layout;
struct gpio_runtime gpio;
/* we need these for headphone/lineout detection */
struct snd_kcontrol *headphone_ctrl;
struct snd_kcontrol *lineout_ctrl;
struct snd_kcontrol *speaker_ctrl;
struct snd_kcontrol *headphone_detected_ctrl;
struct snd_kcontrol *lineout_detected_ctrl;
struct layout_dev_ptr selfptr_headphone;
struct layout_dev_ptr selfptr_lineout;
u32 have_lineout_detect:1,
have_headphone_detect:1,
switch_on_headphone:1,
switch_on_lineout:1;
};
static LIST_HEAD(layouts_list);
static int layouts_list_items;
/* this can go away but only if we allow multiple cards,
* make the fabric handle all the card stuff, etc... */
static struct layout_dev *layout_device;
static int control_info(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_info *uinfo)
{
uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN;
uinfo->count = 1;
uinfo->value.integer.min = 0;
uinfo->value.integer.max = 1;
return 0;
}
#define AMP_CONTROL(n, description) \
static int n##_control_get(struct snd_kcontrol *kcontrol, \
struct snd_ctl_elem_value *ucontrol) \
{ \
struct gpio_runtime *gpio = snd_kcontrol_chip(kcontrol); \
if (gpio->methods && gpio->methods->get_##n) \
ucontrol->value.integer.value[0] = \
gpio->methods->get_##n(gpio); \
return 0; \
} \
static int n##_control_put(struct snd_kcontrol *kcontrol, \
struct snd_ctl_elem_value *ucontrol) \
{ \
struct gpio_runtime *gpio = snd_kcontrol_chip(kcontrol); \
if (gpio->methods && gpio->methods->get_##n) \
gpio->methods->set_##n(gpio, \
ucontrol->value.integer.value[0]); \
return 1; \
} \
static struct snd_kcontrol_new n##_ctl = { \
.iface = SNDRV_CTL_ELEM_IFACE_MIXER, \
.name = description, \
.access = SNDRV_CTL_ELEM_ACCESS_READWRITE, \
.info = control_info, \
.get = n##_control_get, \
.put = n##_control_put, \
}
AMP_CONTROL(headphone, "Headphone Switch");
AMP_CONTROL(speakers, "Speakers Switch");
AMP_CONTROL(lineout, "Line-Out Switch");
static int detect_choice_get(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
struct layout_dev *ldev = snd_kcontrol_chip(kcontrol);
switch (kcontrol->private_value) {
case 0:
ucontrol->value.integer.value[0] = ldev->switch_on_headphone;
break;
case 1:
ucontrol->value.integer.value[0] = ldev->switch_on_lineout;
break;
default:
return -ENODEV;
}
return 0;
}
static int detect_choice_put(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
struct layout_dev *ldev = snd_kcontrol_chip(kcontrol);
switch (kcontrol->private_value) {
case 0:
ldev->switch_on_headphone = !!ucontrol->value.integer.value[0];
break;
case 1:
ldev->switch_on_lineout = !!ucontrol->value.integer.value[0];
break;
default:
return -ENODEV;
}
return 1;
}
static struct snd_kcontrol_new headphone_detect_choice = {
.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
.name = "Headphone Detect Autoswitch",
.info = control_info,
.access = SNDRV_CTL_ELEM_ACCESS_READWRITE,
.get = detect_choice_get,
.put = detect_choice_put,
.private_value = 0,
};
static struct snd_kcontrol_new lineout_detect_choice = {
.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
.name = "Line-Out Detect Autoswitch",
.info = control_info,
.access = SNDRV_CTL_ELEM_ACCESS_READWRITE,
.get = detect_choice_get,
.put = detect_choice_put,
.private_value = 1,
};
static int detected_get(struct snd_kcontrol *kcontrol,
struct snd_ctl_elem_value *ucontrol)
{
struct layout_dev *ldev = snd_kcontrol_chip(kcontrol);
int v;
switch (kcontrol->private_value) {
case 0:
v = ldev->gpio.methods->get_detect(&ldev->gpio,
AOA_NOTIFY_HEADPHONE);
break;
case 1:
v = ldev->gpio.methods->get_detect(&ldev->gpio,
AOA_NOTIFY_LINE_OUT);
break;
default:
return -ENODEV;
}
ucontrol->value.integer.value[0] = v;
return 0;
}
static struct snd_kcontrol_new headphone_detected = {
.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
.name = "Headphone Detected",
.info = control_info,
.access = SNDRV_CTL_ELEM_ACCESS_READ,
.get = detected_get,
.private_value = 0,
};
static struct snd_kcontrol_new lineout_detected = {
.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
.name = "Line-Out Detected",
.info = control_info,
.access = SNDRV_CTL_ELEM_ACCESS_READ,
.get = detected_get,
.private_value = 1,
};
static int check_codec(struct aoa_codec *codec,
struct layout_dev *ldev,
struct codec_connect_info *cci)
{
u32 *ref;
char propname[32];
struct codec_connection *cc;
/* if the codec has a 'codec' node, we require a reference */
if (codec->node && (strcmp(codec->node->name, "codec") == 0)) {
snprintf(propname, sizeof(propname),
"platform-%s-codec-ref", codec->name);
ref = (u32*)get_property(ldev->sound, propname, NULL);
if (!ref) {
printk(KERN_INFO "snd-aoa-fabric-layout: "
"required property %s not present\n", propname);
return -ENODEV;
}
if (*ref != codec->node->linux_phandle) {
printk(KERN_INFO "snd-aoa-fabric-layout: "
"%s doesn't match!\n", propname);
return -ENODEV;
}
} else {
if (layouts_list_items != 1) {
printk(KERN_INFO "snd-aoa-fabric-layout: "
"more than one soundbus, but no references.\n");
return -ENODEV;
}
}
codec->soundbus_dev = ldev->sdev;
codec->gpio = &ldev->gpio;
cc = cci->connections;
if (!cc)
return -EINVAL;
printk(KERN_INFO "snd-aoa-fabric-layout: can use this codec\n");
codec->connected = 0;
codec->fabric_data = cc;
while (cc->connected) {
codec->connected |= 1<<cc->codec_bit;
cc++;
}
return 0;
}
static int layout_found_codec(struct aoa_codec *codec)
{
struct layout_dev *ldev;
int i;
list_for_each_entry(ldev, &layouts_list, list) {
for (i=0; i<MAX_CODECS_PER_BUS; i++) {
if (!ldev->layout->codecs[i].name)
continue;
if (strcmp(ldev->layout->codecs[i].name, codec->name) == 0) {
if (check_codec(codec,
ldev,
&ldev->layout->codecs[i]) == 0)
return 0;
}
}
}
return -ENODEV;
}
static void layout_remove_codec(struct aoa_codec *codec)
{
int i;
/* here remove the codec from the layout dev's
* codec reference */
codec->soundbus_dev = NULL;
codec->gpio = NULL;
for (i=0; i<MAX_CODECS_PER_BUS; i++) {
}
}
static void layout_notify(void *data)
{
struct layout_dev_ptr *dptr = data;
struct layout_dev *ldev;
int v, update;
struct snd_kcontrol *detected, *c;
struct snd_card *card = aoa_get_card();
ldev = dptr->ptr;
if (data == &ldev->selfptr_headphone) {
v = ldev->gpio.methods->get_detect(&ldev->gpio, AOA_NOTIFY_HEADPHONE);
detected = ldev->headphone_detected_ctrl;
update = ldev->switch_on_headphone;
if (update) {
ldev->gpio.methods->set_speakers(&ldev->gpio, !v);
ldev->gpio.methods->set_headphone(&ldev->gpio, v);
ldev->gpio.methods->set_lineout(&ldev->gpio, 0);
}
} else if (data == &ldev->selfptr_lineout) {
v = ldev->gpio.methods->get_detect(&ldev->gpio, AOA_NOTIFY_LINE_OUT);
detected = ldev->lineout_detected_ctrl;
update = ldev->switch_on_lineout;
if (update) {
ldev->gpio.methods->set_speakers(&ldev->gpio, !v);
ldev->gpio.methods->set_headphone(&ldev->gpio, 0);
ldev->gpio.methods->set_lineout(&ldev->gpio, v);
}
} else
return;
if (detected)
snd_ctl_notify(card, SNDRV_CTL_EVENT_MASK_VALUE, &detected->id);
if (update) {
c = ldev->headphone_ctrl;
if (c)
snd_ctl_notify(card, SNDRV_CTL_EVENT_MASK_VALUE, &c->id);
c = ldev->speaker_ctrl;
if (c)
snd_ctl_notify(card, SNDRV_CTL_EVENT_MASK_VALUE, &c->id);
c = ldev->lineout_ctrl;
if (c)
snd_ctl_notify(card, SNDRV_CTL_EVENT_MASK_VALUE, &c->id);
}
}
static void layout_attached_codec(struct aoa_codec *codec)
{
struct codec_connection *cc;
struct snd_kcontrol *ctl;
int headphones, lineout;
struct layout_dev *ldev = layout_device;
/* need to add this codec to our codec array! */
cc = codec->fabric_data;
headphones = codec->gpio->methods->get_detect(codec->gpio,
AOA_NOTIFY_HEADPHONE);
lineout = codec->gpio->methods->get_detect(codec->gpio,
AOA_NOTIFY_LINE_OUT);
while (cc->connected) {
if (cc->connected & CC_SPEAKERS) {
if (headphones <= 0 && lineout <= 0)
ldev->gpio.methods->set_speakers(codec->gpio, 1);
ctl = snd_ctl_new1(&speakers_ctl, codec->gpio);
ldev->speaker_ctrl = ctl;
aoa_snd_ctl_add(ctl);
}
if (cc->connected & CC_HEADPHONE) {
if (headphones == 1)
ldev->gpio.methods->set_headphone(codec->gpio, 1);
ctl = snd_ctl_new1(&headphone_ctl, codec->gpio);
ldev->headphone_ctrl = ctl;
aoa_snd_ctl_add(ctl);
ldev->have_headphone_detect =
!ldev->gpio.methods
->set_notify(&ldev->gpio,
AOA_NOTIFY_HEADPHONE,
layout_notify,
&ldev->selfptr_headphone);
if (ldev->have_headphone_detect) {
ctl = snd_ctl_new1(&headphone_detect_choice,
ldev);
aoa_snd_ctl_add(ctl);
ctl = snd_ctl_new1(&headphone_detected,
ldev);
ldev->headphone_detected_ctrl = ctl;
aoa_snd_ctl_add(ctl);
}
}
if (cc->connected & CC_LINEOUT) {
if (lineout == 1)
ldev->gpio.methods->set_lineout(codec->gpio, 1);
ctl = snd_ctl_new1(&lineout_ctl, codec->gpio);
if (cc->connected & CC_LINEOUT_LABELLED_HEADPHONE)
strlcpy(ctl->id.name,
"Headphone Switch", sizeof(ctl->id.name));
ldev->lineout_ctrl = ctl;
aoa_snd_ctl_add(ctl);
ldev->have_lineout_detect =
!ldev->gpio.methods
->set_notify(&ldev->gpio,
AOA_NOTIFY_LINE_OUT,
layout_notify,
&ldev->selfptr_lineout);
if (ldev->have_lineout_detect) {
ctl = snd_ctl_new1(&lineout_detect_choice,
ldev);
if (cc->connected & CC_LINEOUT_LABELLED_HEADPHONE)
strlcpy(ctl->id.name,
"Headphone Detect Autoswitch",
sizeof(ctl->id.name));
aoa_snd_ctl_add(ctl);
ctl = snd_ctl_new1(&lineout_detected,
ldev);
if (cc->connected & CC_LINEOUT_LABELLED_HEADPHONE)
strlcpy(ctl->id.name,
"Headphone Detected",
sizeof(ctl->id.name));
ldev->lineout_detected_ctrl = ctl;
aoa_snd_ctl_add(ctl);
}
}
cc++;
}
/* now update initial state */
if (ldev->have_headphone_detect)
layout_notify(&ldev->selfptr_headphone);
if (ldev->have_lineout_detect)
layout_notify(&ldev->selfptr_lineout);
}
static struct aoa_fabric layout_fabric = {
.name = "SoundByLayout",
.owner = THIS_MODULE,
.found_codec = layout_found_codec,
.remove_codec = layout_remove_codec,
.attached_codec = layout_attached_codec,
};
static int aoa_fabric_layout_probe(struct soundbus_dev *sdev)
{
struct device_node *sound = NULL;
unsigned int *layout_id;
struct layout *layout;
struct layout_dev *ldev = NULL;
int err;
/* hm, currently we can only have one ... */
if (layout_device)
return -ENODEV;
/* by breaking out we keep a reference */
while ((sound = of_get_next_child(sdev->ofdev.node, sound))) {
if (sound->type && strcasecmp(sound->type, "soundchip") == 0)
break;
}
if (!sound) return -ENODEV;
layout_id = (unsigned int *) get_property(sound, "layout-id", NULL);
if (!layout_id)
goto outnodev;
printk(KERN_INFO "snd-aoa-fabric-layout: found bus with layout %d ", *layout_id);
layout = find_layout_by_id(*layout_id);
if (!layout) {
printk("(no idea how to handle)\n");
goto outnodev;
}
ldev = kzalloc(sizeof(struct layout_dev), GFP_KERNEL);
if (!ldev)
goto outnodev;
layout_device = ldev;
ldev->sdev = sdev;
ldev->sound = sound;
ldev->layout = layout;
ldev->gpio.node = sound->parent;
switch (layout->layout_id) {
case 41: /* that unknown machine no one seems to have */
case 51: /* PowerBook5,4 */
case 58: /* Mac Mini */
ldev->gpio.methods = ftr_gpio_methods;
break;
default:
ldev->gpio.methods = pmf_gpio_methods;
}
ldev->selfptr_headphone.ptr = ldev;
ldev->selfptr_lineout.ptr = ldev;
sdev->ofdev.dev.driver_data = ldev;
printk("(using)\n");
list_add(&ldev->list, &layouts_list);
layouts_list_items++;
/* assign these before registering ourselves, so
* callbacks that are done during registration
* already have the values */
sdev->pcmid = ldev->layout->pcmid;
if (ldev->layout->busname) {
sdev->pcmname = ldev->layout->busname;
} else {
sdev->pcmname = "Master";
}
ldev->gpio.methods->init(&ldev->gpio);
err = aoa_fabric_register(&layout_fabric);
if (err && err != -EALREADY) {
printk(KERN_INFO "snd-aoa-fabric-layout: can't use,"
" another fabric is active!\n");
goto outlistdel;
}
use_layout(layout);
ldev->switch_on_headphone = 1;
ldev->switch_on_lineout = 1;
return 0;
outlistdel:
/* we won't be using these then... */
ldev->gpio.methods->exit(&ldev->gpio);
/* reset if we didn't use it */
sdev->pcmname = NULL;
sdev->pcmid = -1;
list_del(&ldev->list);
layouts_list_items--;
outnodev:
if (sound) of_node_put(sound);
layout_device = NULL;
if (ldev) kfree(ldev);
return -ENODEV;
}
static int aoa_fabric_layout_remove(struct soundbus_dev *sdev)
{
struct layout_dev *ldev = sdev->ofdev.dev.driver_data;
int i;
for (i=0; i<MAX_CODECS_PER_BUS; i++) {
if (ldev->codecs[i]) {
aoa_fabric_unlink_codec(ldev->codecs[i]);
}
ldev->codecs[i] = NULL;
}
list_del(&ldev->list);
layouts_list_items--;
of_node_put(ldev->sound);
ldev->gpio.methods->set_notify(&ldev->gpio,
AOA_NOTIFY_HEADPHONE,
NULL,
NULL);
ldev->gpio.methods->set_notify(&ldev->gpio,
AOA_NOTIFY_LINE_OUT,
NULL,
NULL);
ldev->gpio.methods->exit(&ldev->gpio);
layout_device = NULL;
kfree(ldev);
sdev->pcmid = -1;
sdev->pcmname = NULL;
return 0;
}
#ifdef CONFIG_PM
static int aoa_fabric_layout_suspend(struct soundbus_dev *sdev, pm_message_t state)
{
struct layout_dev *ldev = sdev->ofdev.dev.driver_data;
printk("aoa_fabric_layout_suspend()\n");
if (ldev->gpio.methods && ldev->gpio.methods->all_amps_off)
ldev->gpio.methods->all_amps_off(&ldev->gpio);
return 0;
}
static int aoa_fabric_layout_resume(struct soundbus_dev *sdev)
{
struct layout_dev *ldev = sdev->ofdev.dev.driver_data;
printk("aoa_fabric_layout_resume()\n");
if (ldev->gpio.methods && ldev->gpio.methods->all_amps_off)
ldev->gpio.methods->all_amps_restore(&ldev->gpio);
return 0;
}
#endif
static struct soundbus_driver aoa_soundbus_driver = {
.name = "snd_aoa_soundbus_drv",
.owner = THIS_MODULE,
.probe = aoa_fabric_layout_probe,
.remove = aoa_fabric_layout_remove,
#ifdef CONFIG_PM
.suspend = aoa_fabric_layout_suspend,
.resume = aoa_fabric_layout_resume,
#endif
};
static int __init aoa_fabric_layout_init(void)
{
int err;
err = soundbus_register_driver(&aoa_soundbus_driver);
if (err)
return err;
return 0;
}
static void __exit aoa_fabric_layout_exit(void)
{
soundbus_unregister_driver(&aoa_soundbus_driver);
aoa_fabric_unregister(&layout_fabric);
}
module_init(aoa_fabric_layout_init);
module_exit(aoa_fabric_layout_exit);
sound/aoa/soundbus/Kconfig 0 → 100644
View file @ f3d9478b
config SND_AOA_SOUNDBUS
tristate "Apple Soundbus support"
depends on SOUND && SND_PCM && EXPERIMENTAL
---help---
This option enables the generic driver for the soundbus
support on Apple machines.
It is required for the sound bus implementations.
config SND_AOA_SOUNDBUS_I2S
tristate "I2S bus support"
depends on SND_AOA_SOUNDBUS && PCI
---help---
This option enables support for Apple I2S busses.
sound/aoa/soundbus/Makefile 0 → 100644
View file @ f3d9478b
obj-$(CONFIG_SND_AOA_SOUNDBUS) += snd-aoa-soundbus.o
snd-aoa-soundbus-objs := core.o sysfs.o
obj-$(CONFIG_SND_AOA_SOUNDBUS_I2S) += i2sbus/
sound/aoa/soundbus/core.c 0 → 100644
View file @ f3d9478b
/*
* soundbus
*
* Copyright 2006 Johannes Berg <johannes@sipsolutions.net>
*
* GPL v2, can be found in COPYING.
*/
#include <linux/module.h>
#include "soundbus.h"
MODULE_AUTHOR("Johannes Berg <johannes@sipsolutions.net>");
MODULE_LICENSE("GPL");
MODULE_DESCRIPTION("Apple Soundbus");
struct soundbus_dev *soundbus_dev_get(struct soundbus_dev *dev)
{
struct device *tmp;
if (!dev)
return NULL;
tmp = get_device(&dev->ofdev.dev);
if (tmp)
return to_soundbus_device(tmp);
else
return NULL;
}
EXPORT_SYMBOL_GPL(soundbus_dev_get);
void soundbus_dev_put(struct soundbus_dev *dev)
{
if (dev)
put_device(&dev->ofdev.dev);
}
EXPORT_SYMBOL_GPL(soundbus_dev_put);
static int soundbus_probe(struct device *dev)
{
int error = -ENODEV;
struct soundbus_driver *drv;
struct soundbus_dev *soundbus_dev;
drv = to_soundbus_driver(dev->driver);
soundbus_dev = to_soundbus_device(dev);
if (!drv->probe)
return error;
soundbus_dev_get(soundbus_dev);
error = drv->probe(soundbus_dev);
if (error)
soundbus_dev_put(soundbus_dev);
return error;
}
static int soundbus_uevent(struct device *dev, char **envp, int num_envp,
char *buffer, int buffer_size)
{
struct soundbus_dev * soundbus_dev;
struct of_device * of;
char *scratch, *compat, *compat2;
int i = 0;
int length, cplen, cplen2, seen = 0;
if (!dev)
return -ENODEV;
soundbus_dev = to_soundbus_device(dev);
if (!soundbus_dev)
return -ENODEV;
of = &soundbus_dev->ofdev;
/* stuff we want to pass to /sbin/hotplug */
envp[i++] = scratch = buffer;
length = scnprintf (scratch, buffer_size, "OF_NAME=%s", of->node->name);
++length;
buffer_size -= length;
if ((buffer_size <= 0) || (i >= num_envp))
return -ENOMEM;
scratch += length;
envp[i++] = scratch;
length = scnprintf (scratch, buffer_size, "OF_TYPE=%s", of->node->type);
++length;
buffer_size -= length;
if ((buffer_size <= 0) || (i >= num_envp))
return -ENOMEM;
scratch += length;
/* Since the compatible field can contain pretty much anything
* it's not really legal to split it out with commas. We split it
* up using a number of environment variables instead. */
compat = (char *) get_property(of->node, "compatible", &cplen);
compat2 = compat;
cplen2= cplen;
while (compat && cplen > 0) {
envp[i++] = scratch;
length = scnprintf (scratch, buffer_size,
"OF_COMPATIBLE_%d=%s", seen, compat);
++length;
buffer_size -= length;
if ((buffer_size <= 0) || (i >= num_envp))
return -ENOMEM;
scratch += length;
length = strlen (compat) + 1;
compat += length;
cplen -= length;
seen++;
}
envp[i++] = scratch;
length = scnprintf (scratch, buffer_size, "OF_COMPATIBLE_N=%d", seen);
++length;
buffer_size -= length;
if ((buffer_size <= 0) || (i >= num_envp))
return -ENOMEM;
scratch += length;
envp[i++] = scratch;
length = scnprintf (scratch, buffer_size, "MODALIAS=%s",
soundbus_dev->modalias);
buffer_size -= length;
if ((buffer_size <= 0) || (i >= num_envp))
return -ENOMEM;
envp[i] = NULL;
return 0;
}
static int soundbus_device_remove(struct device *dev)
{
struct soundbus_dev * soundbus_dev = to_soundbus_device(dev);
struct soundbus_driver * drv = to_soundbus_driver(dev->driver);
if (dev->driver && drv->remove)
drv->remove(soundbus_dev);
soundbus_dev_put(soundbus_dev);
return 0;
}
static void soundbus_device_shutdown(struct device *dev)
{
struct soundbus_dev * soundbus_dev = to_soundbus_device(dev);
struct soundbus_driver * drv = to_soundbus_driver(dev->driver);
if (dev->driver && drv->shutdown)
drv->shutdown(soundbus_dev);
}
#ifdef CONFIG_PM
static int soundbus_device_suspend(struct device *dev, pm_message_t state)
{
struct soundbus_dev * soundbus_dev = to_soundbus_device(dev);
struct soundbus_driver * drv = to_soundbus_driver(dev->driver);
if (dev->driver && drv->suspend)
return drv->suspend(soundbus_dev, state);
return 0;
}
static int soundbus_device_resume(struct device * dev)
{
struct soundbus_dev * soundbus_dev = to_soundbus_device(dev);
struct soundbus_driver * drv = to_soundbus_driver(dev->driver);
if (dev->driver && drv->resume)
return drv->resume(soundbus_dev);
return 0;
}
#endif /* CONFIG_PM */
extern struct device_attribute soundbus_dev_attrs[];
static struct bus_type soundbus_bus_type = {
.name = "aoa-soundbus",
.probe = soundbus_probe,
.uevent = soundbus_uevent,
.remove = soundbus_device_remove,
.shutdown = soundbus_device_shutdown,
#ifdef CONFIG_PM
.suspend = soundbus_device_suspend,
.resume = soundbus_device_resume,
#endif
.dev_attrs = soundbus_dev_attrs,
};
static int __init soundbus_init(void)
{
return bus_register(&soundbus_bus_type);
}
static void __exit soundbus_exit(void)
{
bus_unregister(&soundbus_bus_type);
}
int soundbus_add_one(struct soundbus_dev *dev)
{
static int devcount;
/* sanity checks */
if (!dev->attach_codec ||
!dev->ofdev.node ||
dev->pcmname ||
dev->pcmid != -1) {
printk(KERN_ERR "soundbus: adding device failed sanity check!\n");
return -EINVAL;
}
snprintf(dev->ofdev.dev.bus_id, BUS_ID_SIZE, "soundbus:%x", ++devcount);
dev->ofdev.dev.bus = &soundbus_bus_type;
return of_device_register(&dev->ofdev);
}
EXPORT_SYMBOL_GPL(soundbus_add_one);
void soundbus_remove_one(struct soundbus_dev *dev)
{
of_device_unregister(&dev->ofdev);
}
EXPORT_SYMBOL_GPL(soundbus_remove_one);
int soundbus_register_driver(struct soundbus_driver *drv)
{
/* initialize common driver fields */
drv->driver.name = drv->name;
drv->driver.bus = &soundbus_bus_type;
/* register with core */
return driver_register(&drv->driver);
}
EXPORT_SYMBOL_GPL(soundbus_register_driver);
void soundbus_unregister_driver(struct soundbus_driver *drv)
{
driver_unregister(&drv->driver);
}
EXPORT_SYMBOL_GPL(soundbus_unregister_driver);
module_init(soundbus_init);
module_exit(soundbus_exit);
sound/aoa/soundbus/i2sbus/Makefile 0 → 100644
View file @ f3d9478b
obj-$(CONFIG_SND_AOA_SOUNDBUS_I2S) += snd-aoa-i2sbus.o
snd-aoa-i2sbus-objs := i2sbus-core.o i2sbus-pcm.o i2sbus-control.o
sound/aoa/soundbus/i2sbus/i2sbus-control.c 0 → 100644
View file @ f3d9478b
/*
* i2sbus driver -- bus control routines
*
* Copyright 2006 Johannes Berg <johannes@sipsolutions.net>
*
* GPL v2, can be found in COPYING.
*/
#include <asm/io.h>
#include <linux/delay.h>
#include <asm/prom.h>
#include <asm/macio.h>
#include <asm/pmac_feature.h>
#include <asm/pmac_pfunc.h>
#include "i2sbus.h"
int i2sbus_control_init(struct macio_dev* dev, struct i2sbus_control **c)
{
*c = kzalloc(sizeof(struct i2sbus_control), GFP_KERNEL);
if (!*c)
return -ENOMEM;
INIT_LIST_HEAD(&(*c)->list);
if (of_address_to_resource(dev->ofdev.node, 0, &(*c)->rsrc))
goto err;
/* we really should be using feature calls instead of mapping
* these registers. It's safe for now since no one else is
* touching them... */
(*c)->controlregs = ioremap((*c)->rsrc.start,
sizeof(struct i2s_control_regs));
if (!(*c)->controlregs)
goto err;
return 0;
err:
kfree(*c);
*c = NULL;
return -ENODEV;
}
void i2sbus_control_destroy(struct i2sbus_control *c)
{
iounmap(c->controlregs);
kfree(c);
}
/* this is serialised externally */
int i2sbus_control_add_dev(struct i2sbus_control *c,
struct i2sbus_dev *i2sdev)
{
struct device_node *np;
np = i2sdev->sound.ofdev.node;
i2sdev->enable = pmf_find_function(np, "enable");
i2sdev->cell_enable = pmf_find_function(np, "cell-enable");
i2sdev->clock_enable = pmf_find_function(np, "clock-enable");
i2sdev->cell_disable = pmf_find_function(np, "cell-disable");
i2sdev->clock_disable = pmf_find_function(np, "clock-disable");
/* if the bus number is not 0 or 1 we absolutely need to use
* the platform functions -- there's nothing in Darwin that
* would allow seeing a system behind what the FCRs are then,
* and I don't want to go parsing a bunch of platform functions
* by hand to try finding a system... */
if (i2sdev->bus_number != 0 && i2sdev->bus_number != 1 &&
(!i2sdev->enable ||
!i2sdev->cell_enable || !i2sdev->clock_enable ||
!i2sdev->cell_disable || !i2sdev->clock_disable)) {
pmf_put_function(i2sdev->enable);
pmf_put_function(i2sdev->cell_enable);
pmf_put_function(i2sdev->clock_enable);
pmf_put_function(i2sdev->cell_disable);
pmf_put_function(i2sdev->clock_disable);
return -ENODEV;
}
list_add(&i2sdev->item, &c->list);
return 0;
}
void i2sbus_control_remove_dev(struct i2sbus_control *c,
struct i2sbus_dev *i2sdev)
{
/* this is serialised externally */
list_del(&i2sdev->item);
if (list_empty(&c->list))
i2sbus_control_destroy(c);
}
int i2sbus_control_enable(struct i2sbus_control *c,
struct i2sbus_dev *i2sdev)
{
struct pmf_args args = { .count = 0 };
int cc;
if (i2sdev->enable)
return pmf_call_one(i2sdev->enable, &args);
switch (i2sdev->bus_number) {
case 0:
cc = in_le32(&c->controlregs->cell_control);
out_le32(&c->controlregs->cell_control, cc | CTRL_CLOCK_INTF_0_ENABLE);
break;
case 1:
cc = in_le32(&c->controlregs->cell_control);
out_le32(&c->controlregs->cell_control, cc | CTRL_CLOCK_INTF_1_ENABLE);
break;
default:
return -ENODEV;
}
return 0;
}
int i2sbus_control_cell(struct i2sbus_control *c,
struct i2sbus_dev *i2sdev,
int enable)
{
struct pmf_args args = { .count = 0 };
int cc;
switch (enable) {
case 0:
if (i2sdev->cell_disable)
return pmf_call_one(i2sdev->cell_disable, &args);
break;
case 1:
if (i2sdev->cell_enable)
return pmf_call_one(i2sdev->cell_enable, &args);
break;
default:
printk(KERN_ERR "i2sbus: INVALID CELL ENABLE VALUE\n");
return -ENODEV;
}
switch (i2sdev->bus_number) {
case 0:
cc = in_le32(&c->controlregs->cell_control);
cc &= ~CTRL_CLOCK_CELL_0_ENABLE;
cc |= enable * CTRL_CLOCK_CELL_0_ENABLE;
out_le32(&c->controlregs->cell_control, cc);
break;
case 1:
cc = in_le32(&c->controlregs->cell_control);
cc &= ~CTRL_CLOCK_CELL_1_ENABLE;
cc |= enable * CTRL_CLOCK_CELL_1_ENABLE;
out_le32(&c->controlregs->cell_control, cc);
break;
default:
return -ENODEV;
}
return 0;
}
int i2sbus_control_clock(struct i2sbus_control *c,
struct i2sbus_dev *i2sdev,
int enable)
{
struct pmf_args args = { .count = 0 };
int cc;
switch (enable) {
case 0:
if (i2sdev->clock_disable)
return pmf_call_one(i2sdev->clock_disable, &args);
break;
case 1:
if (i2sdev->clock_enable)
return pmf_call_one(i2sdev->clock_enable, &args);
break;
default:
printk(KERN_ERR "i2sbus: INVALID CLOCK ENABLE VALUE\n");
return -ENODEV;
}
switch (i2sdev->bus_number) {
case 0:
cc = in_le32(&c->controlregs->cell_control);
cc &= ~CTRL_CLOCK_CLOCK_0_ENABLE;
cc |= enable * CTRL_CLOCK_CLOCK_0_ENABLE;
out_le32(&c->controlregs->cell_control, cc);
break;
case 1:
cc = in_le32(&c->controlregs->cell_control);
cc &= ~CTRL_CLOCK_CLOCK_1_ENABLE;
cc |= enable * CTRL_CLOCK_CLOCK_1_ENABLE;
out_le32(&c->controlregs->cell_control, cc);
break;
default:
return -ENODEV;
}
return 0;
}
sound/aoa/soundbus/i2sbus/i2sbus-control.h 0 → 100644
View file @ f3d9478b
/*
* i2sbus driver -- bus register definitions
*
* Copyright 2006 Johannes Berg <johannes@sipsolutions.net>
*
* GPL v2, can be found in COPYING.
*/
#ifndef __I2SBUS_CONTROLREGS_H
#define __I2SBUS_CONTROLREGS_H
/* i2s control registers, at least what we know about them */
#define __PAD(m,n) u8 __pad##m[n]
#define _PAD(line, n) __PAD(line, n)
#define PAD(n) _PAD(__LINE__, (n))
struct i2s_control_regs {
PAD(0x38);
__le32 fcr0; /* 0x38 (unknown) */
__le32 cell_control; /* 0x3c (fcr1) */
__le32 fcr2; /* 0x40 (unknown) */
__le32 fcr3; /* 0x44 (fcr3) */
__le32 clock_control; /* 0x48 (unknown) */
PAD(4);
/* total size: 0x50 bytes */
} __attribute__((__packed__));
#define CTRL_CLOCK_CELL_0_ENABLE (1<<10)
#define CTRL_CLOCK_CLOCK_0_ENABLE (1<<12)
#define CTRL_CLOCK_SWRESET_0 (1<<11)
#define CTRL_CLOCK_INTF_0_ENABLE (1<<13)
#define CTRL_CLOCK_CELL_1_ENABLE (1<<17)
#define CTRL_CLOCK_CLOCK_1_ENABLE (1<<18)
#define CTRL_CLOCK_SWRESET_1 (1<<19)
#define CTRL_CLOCK_INTF_1_ENABLE (1<<20)
#endif /* __I2SBUS_CONTROLREGS_H */
sound/aoa/soundbus/i2sbus/i2sbus-core.c 0 → 100644
View file @ f3d9478b
/*
* i2sbus driver
*
* Copyright 2006 Johannes Berg <johannes@sipsolutions.net>
*
* GPL v2, can be found in COPYING.
*/
#include <linux/module.h>
#include <asm/macio.h>
#include <asm/dbdma.h>
#include <linux/pci.h>
#include <linux/interrupt.h>
#include <sound/driver.h>
#include <sound/core.h>
#include <linux/dma-mapping.h>
#include "../soundbus.h"
#include "i2sbus.h"
MODULE_LICENSE("GPL");
MODULE_AUTHOR("Johannes Berg <johannes@sipsolutions.net>");
MODULE_DESCRIPTION("Apple Soundbus: I2S support");
/* for auto-loading, declare that we handle this weird
* string that macio puts into the relevant device */
MODULE_ALIAS("of:Ni2sTi2sC");
static struct of_device_id i2sbus_match[] = {
{ .name = "i2s" },
{ }
};
static int alloc_dbdma_descriptor_ring(struct i2sbus_dev *i2sdev,
struct dbdma_command_mem *r,
int numcmds)
{
/* one more for rounding */
r->size = (numcmds+1) * sizeof(struct dbdma_cmd);
/* We use the PCI APIs for now until the generic one gets fixed
* enough or until we get some macio-specific versions
*/
r->space = dma_alloc_coherent(
&macio_get_pci_dev(i2sdev->macio)->dev,
r->size,
&r->bus_addr,
GFP_KERNEL);
if (!r->space) return -ENOMEM;
memset(r->space, 0, r->size);
r->cmds = (void*)DBDMA_ALIGN(r->space);
r->bus_cmd_start = r->bus_addr +
(dma_addr_t)((char*)r->cmds - (char*)r->space);
return 0;
}
static void free_dbdma_descriptor_ring(struct i2sbus_dev *i2sdev,
struct dbdma_command_mem *r)
{
if (!r->space) return;
dma_free_coherent(&macio_get_pci_dev(i2sdev->macio)->dev,
r->size, r->space, r->bus_addr);
}
static void i2sbus_release_dev(struct device *dev)
{
struct i2sbus_dev *i2sdev;
int i;
i2sdev = container_of(dev, struct i2sbus_dev, sound.ofdev.dev);
if (i2sdev->intfregs) iounmap(i2sdev->intfregs);
if (i2sdev->out.dbdma) iounmap(i2sdev->out.dbdma);
if (i2sdev->in.dbdma) iounmap(i2sdev->in.dbdma);
for (i=0;i<3;i++)
if (i2sdev->allocated_resource[i])
release_and_free_resource(i2sdev->allocated_resource[i]);
free_dbdma_descriptor_ring(i2sdev, &i2sdev->out.dbdma_ring);
free_dbdma_descriptor_ring(i2sdev, &i2sdev->in.dbdma_ring);
for (i=0;i<3;i++)
free_irq(i2sdev->interrupts[i], i2sdev);
i2sbus_control_remove_dev(i2sdev->control, i2sdev);
mutex_destroy(&i2sdev->lock);
kfree(i2sdev);
}
static irqreturn_t i2sbus_bus_intr(int irq, void *devid, struct pt_regs *regs)
{
struct i2sbus_dev *dev = devid;
u32 intreg;
spin_lock(&dev->low_lock);
intreg = in_le32(&dev->intfregs->intr_ctl);
/* acknowledge interrupt reasons */
out_le32(&dev->intfregs->intr_ctl, intreg);
spin_unlock(&dev->low_lock);
return IRQ_HANDLED;
}
static int force;
module_param(force, int, 0444);
MODULE_PARM_DESC(force, "Force loading i2sbus even when"
" no layout-id property is present");
/* FIXME: look at device node refcounting */
static int i2sbus_add_dev(struct macio_dev *macio,
struct i2sbus_control *control,
struct device_node *np)
{
struct i2sbus_dev *dev;
struct device_node *child = NULL, *sound = NULL;
int i;
static const char *rnames[] = { "i2sbus: %s (control)",
"i2sbus: %s (tx)",
"i2sbus: %s (rx)" };
static irqreturn_t (*ints[])(int irq, void *devid,
struct pt_regs *regs) = {
i2sbus_bus_intr,
i2sbus_tx_intr,
i2sbus_rx_intr
};
if (strlen(np->name) != 5)
return 0;
if (strncmp(np->name, "i2s-", 4))
return 0;
if (np->n_intrs != 3)
return 0;
dev = kzalloc(sizeof(struct i2sbus_dev), GFP_KERNEL);
if (!dev)
return 0;
i = 0;
while ((child = of_get_next_child(np, child))) {
if (strcmp(child->name, "sound") == 0) {
i++;
sound = child;
}
}
if (i == 1) {
u32 *layout_id;
layout_id = (u32*) get_property(sound, "layout-id", NULL);
if (layout_id) {
snprintf(dev->sound.modalias, 32,
"sound-layout-%d", *layout_id);
force = 1;
}
}
/* for the time being, until we can handle non-layout-id
* things in some fabric, refuse to attach if there is no
* layout-id property or we haven't been forced to attach.
* When there are two i2s busses and only one has a layout-id,
* then this depends on the order, but that isn't important
* either as the second one in that case is just a modem. */
if (!force) {
kfree(dev);
return -ENODEV;
}
mutex_init(&dev->lock);
spin_lock_init(&dev->low_lock);
dev->sound.ofdev.node = np;
dev->sound.ofdev.dma_mask = macio->ofdev.dma_mask;
dev->sound.ofdev.dev.dma_mask = &dev->sound.ofdev.dma_mask;
dev->sound.ofdev.dev.parent = &macio->ofdev.dev;
dev->sound.ofdev.dev.release = i2sbus_release_dev;
dev->sound.attach_codec = i2sbus_attach_codec;
dev->sound.detach_codec = i2sbus_detach_codec;
dev->sound.pcmid = -1;
dev->macio = macio;
dev->control = control;
dev->bus_number = np->name[4] - 'a';
INIT_LIST_HEAD(&dev->sound.codec_list);
for (i=0;i<3;i++) {
dev->interrupts[i] = -1;
snprintf(dev->rnames[i], sizeof(dev->rnames[i]), rnames[i], np->name);
}
for (i=0;i<3;i++) {
if (request_irq(np->intrs[i].line, ints[i], 0, dev->rnames[i], dev))
goto err;
dev->interrupts[i] = np->intrs[i].line;
}
for (i=0;i<3;i++) {
if (of_address_to_resource(np, i, &dev->resources[i]))
goto err;
/* if only we could use our resource dev->resources[i]...
* but request_resource doesn't know about parents and
* contained resources... */
dev->allocated_resource[i] =
request_mem_region(dev->resources[i].start,
dev->resources[i].end -
dev->resources[i].start + 1,
dev->rnames[i]);
if (!dev->allocated_resource[i]) {
printk(KERN_ERR "i2sbus: failed to claim resource %d!\n", i);
goto err;
}
}
/* should do sanity checking here about length of them */
dev->intfregs = ioremap(dev->resources[0].start,
dev->resources[0].end-dev->resources[0].start+1);
dev->out.dbdma = ioremap(dev->resources[1].start,
dev->resources[1].end-dev->resources[1].start+1);
dev->in.dbdma = ioremap(dev->resources[2].start,
dev->resources[2].end-dev->resources[2].start+1);
if (!dev->intfregs || !dev->out.dbdma || !dev->in.dbdma)
goto err;
if (alloc_dbdma_descriptor_ring(dev, &dev->out.dbdma_ring,
MAX_DBDMA_COMMANDS))
goto err;
if (alloc_dbdma_descriptor_ring(dev, &dev->in.dbdma_ring,
MAX_DBDMA_COMMANDS))
goto err;
if (i2sbus_control_add_dev(dev->control, dev)) {
printk(KERN_ERR "i2sbus: control layer didn't like bus\n");
goto err;
}
if (soundbus_add_one(&dev->sound)) {
printk(KERN_DEBUG "i2sbus: device registration error!\n");
goto err;
}
/* enable this cell */
i2sbus_control_cell(dev->control, dev, 1);
i2sbus_control_enable(dev->control, dev);
i2sbus_control_clock(dev->control, dev, 1);
return 1;
err:
for (i=0;i<3;i++)
if (dev->interrupts[i] != -1)
free_irq(dev->interrupts[i], dev);
free_dbdma_descriptor_ring(dev, &dev->out.dbdma_ring);
free_dbdma_descriptor_ring(dev, &dev->in.dbdma_ring);
if (dev->intfregs) iounmap(dev->intfregs);
if (dev->out.dbdma) iounmap(dev->out.dbdma);
if (dev->in.dbdma) iounmap(dev->in.dbdma);
for (i=0;i<3;i++)
if (dev->allocated_resource[i])
release_and_free_resource(dev->allocated_resource[i]);
mutex_destroy(&dev->lock);
kfree(dev);
return 0;
}
static int i2sbus_probe(struct macio_dev* dev, const struct of_device_id *match)
{
struct device_node *np = NULL;
int got = 0, err;
struct i2sbus_control *control = NULL;
err = i2sbus_control_init(dev, &control);
if (err)
return err;
if (!control) {
printk(KERN_ERR "i2sbus_control_init API breakage\n");
return -ENODEV;
}
while ((np = of_get_next_child(dev->ofdev.node, np))) {
if (device_is_compatible(np, "i2sbus") ||
device_is_compatible(np, "i2s-modem")) {
got += i2sbus_add_dev(dev, control, np);
}
}
if (!got) {
/* found none, clean up */
i2sbus_control_destroy(control);
return -ENODEV;
}
dev->ofdev.dev.driver_data = control;
return 0;
}
static int i2sbus_remove(struct macio_dev* dev)
{
struct i2sbus_control *control = dev->ofdev.dev.driver_data;
struct i2sbus_dev *i2sdev, *tmp;
list_for_each_entry_safe(i2sdev, tmp, &control->list, item)
soundbus_remove_one(&i2sdev->sound);
return 0;
}
#ifdef CONFIG_PM
static int i2sbus_suspend(struct macio_dev* dev, pm_message_t state)
{
struct i2sbus_control *control = dev->ofdev.dev.driver_data;
struct codec_info_item *cii;
struct i2sbus_dev* i2sdev;
int err, ret = 0;
list_for_each_entry(i2sdev, &control->list, item) {
/* Notify Alsa */
if (i2sdev->sound.pcm) {
/* Suspend PCM streams */
snd_pcm_suspend_all(i2sdev->sound.pcm);
/* Probably useless as we handle
* power transitions ourselves */
snd_power_change_state(i2sdev->sound.pcm->card,
SNDRV_CTL_POWER_D3hot);
}
/* Notify codecs */
list_for_each_entry(cii, &i2sdev->sound.codec_list, list) {
err = 0;
if (cii->codec->suspend)
err = cii->codec->suspend(cii, state);
if (err)
ret = err;
}
}
return ret;
}
static int i2sbus_resume(struct macio_dev* dev)
{
struct i2sbus_control *control = dev->ofdev.dev.driver_data;
struct codec_info_item *cii;
struct i2sbus_dev* i2sdev;
int err, ret = 0;
list_for_each_entry(i2sdev, &control->list, item) {
/* Notify codecs so they can re-initialize */
list_for_each_entry(cii, &i2sdev->sound.codec_list, list) {
err = 0;
if (cii->codec->resume)
err = cii->codec->resume(cii);
if (err)
ret = err;
}
/* Notify Alsa */
if (i2sdev->sound.pcm) {
/* Same comment as above, probably useless */
snd_power_change_state(i2sdev->sound.pcm->card,
SNDRV_CTL_POWER_D0);
}
}
return ret;
}
#endif /* CONFIG_PM */
static int i2sbus_shutdown(struct macio_dev* dev)
{
return 0;
}
static struct macio_driver i2sbus_drv = {
.name = "soundbus-i2s",
.owner = THIS_MODULE,
.match_table = i2sbus_match,
.probe = i2sbus_probe,
.remove = i2sbus_remove,
#ifdef CONFIG_PM
.suspend = i2sbus_suspend,
.resume = i2sbus_resume,
#endif
.shutdown = i2sbus_shutdown,
};
static int __init soundbus_i2sbus_init(void)
{
return macio_register_driver(&i2sbus_drv);
}
static void __exit soundbus_i2sbus_exit(void)
{
macio_unregister_driver(&i2sbus_drv);
}
module_init(soundbus_i2sbus_init);
module_exit(soundbus_i2sbus_exit);
sound/aoa/soundbus/i2sbus/i2sbus-interface.h 0 → 100644
View file @ f3d9478b
/*
* i2sbus driver -- interface register definitions
*
* Copyright 2006 Johannes Berg <johannes@sipsolutions.net>
*
* GPL v2, can be found in COPYING.
*/
#ifndef __I2SBUS_INTERFACE_H
#define __I2SBUS_INTERFACE_H
/* i2s bus control registers, at least what we know about them */
#define __PAD(m,n) u8 __pad##m[n]
#define _PAD(line, n) __PAD(line, n)
#define PAD(n) _PAD(__LINE__, (n))
struct i2s_interface_regs {
__le32 intr_ctl; /* 0x00 */
PAD(12);
__le32 serial_format; /* 0x10 */
PAD(12);
__le32 codec_msg_out; /* 0x20 */
PAD(12);
__le32 codec_msg_in; /* 0x30 */
PAD(12);
__le32 frame_count; /* 0x40 */
PAD(12);
__le32 frame_match; /* 0x50 */
PAD(12);
__le32 data_word_sizes; /* 0x60 */
PAD(12);
__le32 peak_level_sel; /* 0x70 */
PAD(12);
__le32 peak_level_in0; /* 0x80 */
PAD(12);
__le32 peak_level_in1; /* 0x90 */
PAD(12);
/* total size: 0x100 bytes */
} __attribute__((__packed__));
/* interrupt register is just a bitfield with
* interrupt enable and pending bits */
#define I2S_REG_INTR_CTL 0x00
# define I2S_INT_FRAME_COUNT (1<<31)
# define I2S_PENDING_FRAME_COUNT (1<<30)
# define I2S_INT_MESSAGE_FLAG (1<<29)
# define I2S_PENDING_MESSAGE_FLAG (1<<28)
# define I2S_INT_NEW_PEAK (1<<27)
# define I2S_PENDING_NEW_PEAK (1<<26)
# define I2S_INT_CLOCKS_STOPPED (1<<25)
# define I2S_PENDING_CLOCKS_STOPPED (1<<24)
# define I2S_INT_EXTERNAL_SYNC_ERROR (1<<23)
# define I2S_PENDING_EXTERNAL_SYNC_ERROR (1<<22)
# define I2S_INT_EXTERNAL_SYNC_OK (1<<21)
# define I2S_PENDING_EXTERNAL_SYNC_OK (1<<20)
# define I2S_INT_NEW_SAMPLE_RATE (1<<19)
# define I2S_PENDING_NEW_SAMPLE_RATE (1<<18)
# define I2S_INT_STATUS_FLAG (1<<17)
# define I2S_PENDING_STATUS_FLAG (1<<16)
/* serial format register is more interesting :)
* It contains:
* - clock source
* - MClk divisor
* - SClk divisor
* - SClk master flag
* - serial format (sony, i2s 64x, i2s 32x, dav, silabs)
* - external sample frequency interrupt (don't understand)
* - external sample frequency
*/
#define I2S_REG_SERIAL_FORMAT 0x10
/* clock source. You get either 18.432, 45.1584 or 49.1520 MHz */
# define I2S_SF_CLOCK_SOURCE_SHIFT 30
# define I2S_SF_CLOCK_SOURCE_MASK (3<<I2S_SF_CLOCK_SOURCE_SHIFT)
# define I2S_SF_CLOCK_SOURCE_18MHz (0<<I2S_SF_CLOCK_SOURCE_SHIFT)
# define I2S_SF_CLOCK_SOURCE_45MHz (1<<I2S_SF_CLOCK_SOURCE_SHIFT)
# define I2S_SF_CLOCK_SOURCE_49MHz (2<<I2S_SF_CLOCK_SOURCE_SHIFT)
/* also, let's define the exact clock speeds here, in Hz */
#define I2S_CLOCK_SPEED_18MHz 18432000
#define I2S_CLOCK_SPEED_45MHz 45158400
#define I2S_CLOCK_SPEED_49MHz 49152000
/* MClk is the clock that drives the codec, usually called its 'system clock'.
* It is derived by taking only every 'divisor' tick of the clock.
*/
# define I2S_SF_MCLKDIV_SHIFT 24
# define I2S_SF_MCLKDIV_MASK (0x1F<<I2S_SF_MCLKDIV_SHIFT)
# define I2S_SF_MCLKDIV_1 (0x14<<I2S_SF_MCLKDIV_SHIFT)
# define I2S_SF_MCLKDIV_3 (0x13<<I2S_SF_MCLKDIV_SHIFT)
# define I2S_SF_MCLKDIV_5 (0x12<<I2S_SF_MCLKDIV_SHIFT)
# define I2S_SF_MCLKDIV_14 (0x0E<<I2S_SF_MCLKDIV_SHIFT)
# define I2S_SF_MCLKDIV_OTHER(div) (((div/2-1)<<I2S_SF_MCLKDIV_SHIFT)&I2S_SF_MCLKDIV_MASK)
static inline int i2s_sf_mclkdiv(int div, int *out)
{
int d;
switch(div) {
case 1: *out |= I2S_SF_MCLKDIV_1; return 0;
case 3: *out |= I2S_SF_MCLKDIV_3; return 0;
case 5: *out |= I2S_SF_MCLKDIV_5; return 0;
case 14: *out |= I2S_SF_MCLKDIV_14; return 0;
default:
if (div%2) return -1;
d = div/2-1;
if (d == 0x14 || d == 0x13 || d == 0x12 || d == 0x0E)
return -1;
*out |= I2S_SF_MCLKDIV_OTHER(div);
return 0;
}
}
/* SClk is the clock that drives the i2s wire bus. Note that it is
* derived from the MClk above by taking only every 'divisor' tick
* of MClk.
*/
# define I2S_SF_SCLKDIV_SHIFT 20
# define I2S_SF_SCLKDIV_MASK (0xF<<I2S_SF_SCLKDIV_SHIFT)
# define I2S_SF_SCLKDIV_1 (8<<I2S_SF_SCLKDIV_SHIFT)
# define I2S_SF_SCLKDIV_3 (9<<I2S_SF_SCLKDIV_SHIFT)
# define I2S_SF_SCLKDIV_OTHER(div) (((div/2-1)<<I2S_SF_SCLKDIV_SHIFT)&I2S_SF_SCLKDIV_MASK)
static inline int i2s_sf_sclkdiv(int div, int *out)
{
int d;
switch(div) {
case 1: *out |= I2S_SF_SCLKDIV_1; return 0;
case 3: *out |= I2S_SF_SCLKDIV_3; return 0;
default:
if (div%2) return -1;
d = div/2-1;
if (d == 8 || d == 9) return -1;
*out |= I2S_SF_SCLKDIV_OTHER(div);
return 0;
}
}
# define I2S_SF_SCLK_MASTER (1<<19)
/* serial format is the way the data is put to the i2s wire bus */
# define I2S_SF_SERIAL_FORMAT_SHIFT 16
# define I2S_SF_SERIAL_FORMAT_MASK (7<<I2S_SF_SERIAL_FORMAT_SHIFT)
# define I2S_SF_SERIAL_FORMAT_SONY (0<<I2S_SF_SERIAL_FORMAT_SHIFT)
# define I2S_SF_SERIAL_FORMAT_I2S_64X (1<<I2S_SF_SERIAL_FORMAT_SHIFT)
# define I2S_SF_SERIAL_FORMAT_I2S_32X (2<<I2S_SF_SERIAL_FORMAT_SHIFT)
# define I2S_SF_SERIAL_FORMAT_I2S_DAV (4<<I2S_SF_SERIAL_FORMAT_SHIFT)
# define I2S_SF_SERIAL_FORMAT_I2S_SILABS (5<<I2S_SF_SERIAL_FORMAT_SHIFT)
/* unknown */
# define I2S_SF_EXT_SAMPLE_FREQ_INT_SHIFT 12
# define I2S_SF_EXT_SAMPLE_FREQ_INT_MASK (0xF<<I2S_SF_SAMPLE_FREQ_INT_SHIFT)
/* probably gives external frequency? */
# define I2S_SF_EXT_SAMPLE_FREQ_MASK 0xFFF
/* used to send codec messages, but how isn't clear */
#define I2S_REG_CODEC_MSG_OUT 0x20
/* used to receive codec messages, but how isn't clear */
#define I2S_REG_CODEC_MSG_IN 0x30
/* frame count reg isn't clear to me yet, but probably useful */
#define I2S_REG_FRAME_COUNT 0x40
/* program to some value, and get interrupt if frame count reaches it */
#define I2S_REG_FRAME_MATCH 0x50
/* this register describes how the bus transfers data */
#define I2S_REG_DATA_WORD_SIZES 0x60
/* number of interleaved input channels */
# define I2S_DWS_NUM_CHANNELS_IN_SHIFT 24
# define I2S_DWS_NUM_CHANNELS_IN_MASK (0x1F<<I2S_DWS_NUM_CHANNELS_IN_SHIFT)
/* word size of input data */
# define I2S_DWS_DATA_IN_SIZE_SHIFT 16
# define I2S_DWS_DATA_IN_16BIT (0<<I2S_DWS_DATA_IN_SIZE_SHIFT)
# define I2S_DWS_DATA_IN_24BIT (3<<I2S_DWS_DATA_IN_SIZE_SHIFT)
/* number of interleaved output channels */
# define I2S_DWS_NUM_CHANNELS_OUT_SHIFT 8
# define I2S_DWS_NUM_CHANNELS_OUT_MASK (0x1F<<I2S_DWS_NUM_CHANNELS_OUT_SHIFT)
/* word size of output data */
# define I2S_DWS_DATA_OUT_SIZE_SHIFT 0
# define I2S_DWS_DATA_OUT_16BIT (0<<I2S_DWS_DATA_OUT_SIZE_SHIFT)
# define I2S_DWS_DATA_OUT_24BIT (3<<I2S_DWS_DATA_OUT_SIZE_SHIFT)
/* unknown */
#define I2S_REG_PEAK_LEVEL_SEL 0x70
/* unknown */
#define I2S_REG_PEAK_LEVEL_IN0 0x80
/* unknown */
#define I2S_REG_PEAK_LEVEL_IN1 0x90
#endif /* __I2SBUS_INTERFACE_H */
sound/aoa/soundbus/i2sbus/i2sbus-pcm.c 0 → 100644
View file @ f3d9478b
/*
* i2sbus driver -- pcm routines
*
* Copyright 2006 Johannes Berg <johannes@sipsolutions.net>
*
* GPL v2, can be found in COPYING.
*/
#include <asm/io.h>
#include <linux/delay.h>
/* So apparently there's a reason for requiring driver.h
* to be included first, even if I don't know it... */
#include <sound/driver.h>
#include <sound/core.h>
#include <asm/macio.h>
#include <linux/pci.h>
#include "../soundbus.h"
#include "i2sbus.h"
static inline void get_pcm_info(struct i2sbus_dev *i2sdev, int in,
struct pcm_info **pi, struct pcm_info **other)
{
if (in) {
if (pi)
*pi = &i2sdev->in;
if (other)
*other = &i2sdev->out;
} else {
if (pi)
*pi = &i2sdev->out;
if (other)
*other = &i2sdev->in;
}
}
static int clock_and_divisors(int mclk, int sclk, int rate, int *out)
{
/* sclk must be derived from mclk! */
if (mclk % sclk)
return -1;
/* derive sclk register value */
if (i2s_sf_sclkdiv(mclk / sclk, out))
return -1;
if (I2S_CLOCK_SPEED_18MHz % (rate * mclk) == 0) {
if (!i2s_sf_mclkdiv(I2S_CLOCK_SPEED_18MHz / (rate * mclk), out)) {
*out |= I2S_SF_CLOCK_SOURCE_18MHz;
return 0;
}
}
if (I2S_CLOCK_SPEED_45MHz % (rate * mclk) == 0) {
if (!i2s_sf_mclkdiv(I2S_CLOCK_SPEED_45MHz / (rate * mclk), out)) {
*out |= I2S_SF_CLOCK_SOURCE_45MHz;
return 0;
}
}
if (I2S_CLOCK_SPEED_49MHz % (rate * mclk) == 0) {
if (!i2s_sf_mclkdiv(I2S_CLOCK_SPEED_49MHz / (rate * mclk), out)) {
*out |= I2S_SF_CLOCK_SOURCE_49MHz;
return 0;
}
}
return -1;
}
#define CHECK_RATE(rate) \
do { if (rates & SNDRV_PCM_RATE_ ##rate) { \
int dummy; \
if (clock_and_divisors(sysclock_factor, \
bus_factor, rate, &dummy)) \
rates &= ~SNDRV_PCM_RATE_ ##rate; \
} } while (0)
static int i2sbus_pcm_open(struct i2sbus_dev *i2sdev, int in)
{
struct pcm_info *pi, *other;
struct soundbus_dev *sdev;
int masks_inited = 0, err;
struct codec_info_item *cii, *rev;
struct snd_pcm_hardware *hw;
u64 formats = 0;
unsigned int rates = 0;
struct transfer_info v;
int result = 0;
int bus_factor = 0, sysclock_factor = 0;
int found_this;
mutex_lock(&i2sdev->lock);
get_pcm_info(i2sdev, in, &pi, &other);
hw = &pi->substream->runtime->hw;
sdev = &i2sdev->sound;
if (pi->active) {
/* alsa messed up */
result = -EBUSY;
goto out_unlock;
}
/* we now need to assign the hw */
list_for_each_entry(cii, &sdev->codec_list, list) {
struct transfer_info *ti = cii->codec->transfers;
bus_factor = cii->codec->bus_factor;
sysclock_factor = cii->codec->sysclock_factor;
while (ti->formats && ti->rates) {
v = *ti;
if (ti->transfer_in == in
&& cii->codec->usable(cii, ti, &v)) {
if (masks_inited) {
formats &= v.formats;
rates &= v.rates;
} else {
formats = v.formats;
rates = v.rates;
masks_inited = 1;
}
}
ti++;
}
}
if (!masks_inited || !bus_factor || !sysclock_factor) {
result = -ENODEV;
goto out_unlock;
}
/* bus dependent stuff */
hw->info = SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID |
SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_RESUME;
CHECK_RATE(5512);
CHECK_RATE(8000);
CHECK_RATE(11025);
CHECK_RATE(16000);
CHECK_RATE(22050);
CHECK_RATE(32000);
CHECK_RATE(44100);
CHECK_RATE(48000);
CHECK_RATE(64000);
CHECK_RATE(88200);
CHECK_RATE(96000);
CHECK_RATE(176400);
CHECK_RATE(192000);
hw->rates = rates;
/* well. the codec might want 24 bits only, and we'll
* ever only transfer 24 bits, but they are top-aligned!
* So for alsa, we claim that we're doing full 32 bit
* while in reality we'll ignore the lower 8 bits of
* that when doing playback (they're transferred as 0
* as far as I know, no codecs we have are 32-bit capable
* so I can't really test) and when doing recording we'll
* always have those lower 8 bits recorded as 0 */
if (formats & SNDRV_PCM_FMTBIT_S24_BE)
formats |= SNDRV_PCM_FMTBIT_S32_BE;
if (formats & SNDRV_PCM_FMTBIT_U24_BE)
formats |= SNDRV_PCM_FMTBIT_U32_BE;
/* now mask off what we can support. I suppose we could
* also support S24_3LE and some similar formats, but I
* doubt there's a codec that would be able to use that,
* so we don't support it here. */
hw->formats = formats & (SNDRV_PCM_FMTBIT_S16_BE |
SNDRV_PCM_FMTBIT_U16_BE |
SNDRV_PCM_FMTBIT_S32_BE |
SNDRV_PCM_FMTBIT_U32_BE);
/* we need to set the highest and lowest rate possible.
* These are the highest and lowest rates alsa can
* support properly in its bitfield.
* Below, we'll use that to restrict to the rate
* currently in use (if any). */
hw->rate_min = 5512;
hw->rate_max = 192000;
/* if the other stream is active, then we can only
* support what it is currently using.
* FIXME: I lied. This comment is wrong. We can support
* anything that works with the same serial format, ie.
* when recording 24 bit sound we can well play 16 bit
* sound at the same time iff using the same transfer mode.
*/
if (other->active) {
/* FIXME: is this guaranteed by the alsa api? */
hw->formats &= (1ULL << i2sdev->format);
/* see above, restrict rates to the one we already have */
hw->rate_min = i2sdev->rate;
hw->rate_max = i2sdev->rate;
}
hw->channels_min = 2;
hw->channels_max = 2;
/* these are somewhat arbitrary */
hw->buffer_bytes_max = 131072;
hw->period_bytes_min = 256;
hw->period_bytes_max = 16384;
hw->periods_min = 3;
hw->periods_max = MAX_DBDMA_COMMANDS;
list_for_each_entry(cii, &sdev->codec_list, list) {
if (cii->codec->open) {
err = cii->codec->open(cii, pi->substream);
if (err) {
result = err;
/* unwind */
found_this = 0;
list_for_each_entry_reverse(rev,
&sdev->codec_list, list) {
if (found_this && rev->codec->close) {
rev->codec->close(rev,
pi->substream);
}
if (rev == cii)
found_this = 1;
}
goto out_unlock;
}
}
}
out_unlock:
mutex_unlock(&i2sdev->lock);
return result;
}
#undef CHECK_RATE
static int i2sbus_pcm_close(struct i2sbus_dev *i2sdev, int in)
{
struct codec_info_item *cii;
struct pcm_info *pi;
int err = 0, tmp;
mutex_lock(&i2sdev->lock);
get_pcm_info(i2sdev, in, &pi, NULL);
list_for_each_entry(cii, &i2sdev->sound.codec_list, list) {
if (cii->codec->close) {
tmp = cii->codec->close(cii, pi->substream);
if (tmp)
err = tmp;
}
}
pi->substream = NULL;
pi->active = 0;
mutex_unlock(&i2sdev->lock);
return err;
}
static int i2sbus_hw_params(struct snd_pcm_substream *substream,
struct snd_pcm_hw_params *params)
{
return snd_pcm_lib_malloc_pages(substream, params_buffer_bytes(params));
}
static int i2sbus_hw_free(struct snd_pcm_substream *substream)
{
snd_pcm_lib_free_pages(substream);
return 0;
}
static int i2sbus_pcm_prepare(struct i2sbus_dev *i2sdev, int in)
{
/* whee. Hard work now. The user has selected a bitrate
* and bit format, so now we have to program our
* I2S controller appropriately. */
struct snd_pcm_runtime *runtime;
struct dbdma_cmd *command;
int i, periodsize;
dma_addr_t offset;
struct bus_info bi;
struct codec_info_item *cii;
int sfr = 0; /* serial format register */
int dws = 0; /* data word sizes reg */
int input_16bit;
struct pcm_info *pi, *other;
int cnt;
int result = 0;
mutex_lock(&i2sdev->lock);
get_pcm_info(i2sdev, in, &pi, &other);
if (pi->dbdma_ring.running) {
result = -EBUSY;
goto out_unlock;
}
runtime = pi->substream->runtime;
pi->active = 1;
if (other->active &&
((i2sdev->format != runtime->format)
|| (i2sdev->rate != runtime->rate))) {
result = -EINVAL;
goto out_unlock;
}
i2sdev->format = runtime->format;
i2sdev->rate = runtime->rate;
periodsize = snd_pcm_lib_period_bytes(pi->substream);
pi->current_period = 0;
/* generate dbdma command ring first */
command = pi->dbdma_ring.cmds;
offset = runtime->dma_addr;
for (i = 0; i < pi->substream->runtime->periods;
i++, command++, offset += periodsize) {
memset(command, 0, sizeof(struct dbdma_cmd));
command->command =
cpu_to_le16((in ? INPUT_MORE : OUTPUT_MORE) | INTR_ALWAYS);
command->phy_addr = cpu_to_le32(offset);
command->req_count = cpu_to_le16(periodsize);
command->xfer_status = cpu_to_le16(0);
}
/* last one branches back to first */
command--;
command->command |= cpu_to_le16(BR_ALWAYS);
command->cmd_dep = cpu_to_le32(pi->dbdma_ring.bus_cmd_start);
/* ok, let's set the serial format and stuff */
switch (runtime->format) {
/* 16 bit formats */
case SNDRV_PCM_FORMAT_S16_BE:
case SNDRV_PCM_FORMAT_U16_BE:
/* FIXME: if we add different bus factors we need to
* do more here!! */
bi.bus_factor = 0;
list_for_each_entry(cii, &i2sdev->sound.codec_list, list) {
bi.bus_factor = cii->codec->bus_factor;
break;
}
if (!bi.bus_factor) {
result = -ENODEV;
goto out_unlock;
}
input_16bit = 1;
break;
case SNDRV_PCM_FORMAT_S32_BE:
case SNDRV_PCM_FORMAT_U32_BE:
/* force 64x bus speed, otherwise the data cannot be
* transferred quickly enough! */
bi.bus_factor = 64;
input_16bit = 0;
break;
default:
result = -EINVAL;
goto out_unlock;
}
/* we assume all sysclocks are the same! */
list_for_each_entry(cii, &i2sdev->sound.codec_list, list) {
bi.sysclock_factor = cii->codec->sysclock_factor;
break;
}
if (clock_and_divisors(bi.sysclock_factor,
bi.bus_factor,
runtime->rate,
&sfr) < 0) {
result = -EINVAL;
goto out_unlock;
}
switch (bi.bus_factor) {
case 32:
sfr |= I2S_SF_SERIAL_FORMAT_I2S_32X;
break;
case 64:
sfr |= I2S_SF_SERIAL_FORMAT_I2S_64X;
break;
}
/* FIXME: THIS ASSUMES MASTER ALL THE TIME */
sfr |= I2S_SF_SCLK_MASTER;
list_for_each_entry(cii, &i2sdev->sound.codec_list, list) {
int err = 0;
if (cii->codec->prepare)
err = cii->codec->prepare(cii, &bi, pi->substream);
if (err) {
result = err;
goto out_unlock;
}
}
/* codecs are fine with it, so set our clocks */
if (input_16bit)
dws = (2 << I2S_DWS_NUM_CHANNELS_IN_SHIFT) |
(2 << I2S_DWS_NUM_CHANNELS_OUT_SHIFT) |
I2S_DWS_DATA_IN_16BIT | I2S_DWS_DATA_OUT_16BIT;
else
dws = (2 << I2S_DWS_NUM_CHANNELS_IN_SHIFT) |
(2 << I2S_DWS_NUM_CHANNELS_OUT_SHIFT) |
I2S_DWS_DATA_IN_24BIT | I2S_DWS_DATA_OUT_24BIT;
/* early exit if already programmed correctly */
/* not locking these is fine since we touch them only in this function */
if (in_le32(&i2sdev->intfregs->serial_format) == sfr
&& in_le32(&i2sdev->intfregs->data_word_sizes) == dws)
goto out_unlock;
/* let's notify the codecs about clocks going away.
* For now we only do mastering on the i2s cell... */
list_for_each_entry(cii, &i2sdev->sound.codec_list, list)
if (cii->codec->switch_clock)
cii->codec->switch_clock(cii, CLOCK_SWITCH_PREPARE_SLAVE);
i2sbus_control_enable(i2sdev->control, i2sdev);
i2sbus_control_cell(i2sdev->control, i2sdev, 1);
out_le32(&i2sdev->intfregs->intr_ctl, I2S_PENDING_CLOCKS_STOPPED);
i2sbus_control_clock(i2sdev->control, i2sdev, 0);
msleep(1);
/* wait for clock stopped. This can apparently take a while... */
cnt = 100;
while (cnt-- &&
!(in_le32(&i2sdev->intfregs->intr_ctl) & I2S_PENDING_CLOCKS_STOPPED)) {
msleep(5);
}
out_le32(&i2sdev->intfregs->intr_ctl, I2S_PENDING_CLOCKS_STOPPED);
/* not locking these is fine since we touch them only in this function */
out_le32(&i2sdev->intfregs->serial_format, sfr);
out_le32(&i2sdev->intfregs->data_word_sizes, dws);
i2sbus_control_enable(i2sdev->control, i2sdev);
i2sbus_control_cell(i2sdev->control, i2sdev, 1);
i2sbus_control_clock(i2sdev->control, i2sdev, 1);
msleep(1);
list_for_each_entry(cii, &i2sdev->sound.codec_list, list)
if (cii->codec->switch_clock)
cii->codec->switch_clock(cii, CLOCK_SWITCH_SLAVE);
out_unlock:
mutex_unlock(&i2sdev->lock);
return result;
}
static struct dbdma_cmd STOP_CMD = {
.command = __constant_cpu_to_le16(DBDMA_STOP),
};
static int i2sbus_pcm_trigger(struct i2sbus_dev *i2sdev, int in, int cmd)
{
struct codec_info_item *cii;
struct pcm_info *pi;
int timeout;
struct dbdma_cmd tmp;
int result = 0;
unsigned long flags;
spin_lock_irqsave(&i2sdev->low_lock, flags);
get_pcm_info(i2sdev, in, &pi, NULL);
switch (cmd) {
case SNDRV_PCM_TRIGGER_START:
case SNDRV_PCM_TRIGGER_RESUME:
if (pi->dbdma_ring.running) {
result = -EALREADY;
goto out_unlock;
}
list_for_each_entry(cii, &i2sdev->sound.codec_list, list)
if (cii->codec->start)
cii->codec->start(cii, pi->substream);
pi->dbdma_ring.running = 1;
/* reset dma engine */
out_le32(&pi->dbdma->control,
0 | (RUN | PAUSE | FLUSH | WAKE) << 16);
timeout = 100;
while (in_le32(&pi->dbdma->status) & RUN && timeout--)
udelay(1);
if (timeout <= 0) {
printk(KERN_ERR
"i2sbus: error waiting for dma reset\n");
result = -ENXIO;
goto out_unlock;
}
/* write dma command buffer address to the dbdma chip */
out_le32(&pi->dbdma->cmdptr, pi->dbdma_ring.bus_cmd_start);
/* post PCI write */
mb();
(void)in_le32(&pi->dbdma->status);
/* change first command to STOP */
tmp = *pi->dbdma_ring.cmds;
*pi->dbdma_ring.cmds = STOP_CMD;
/* set running state, remember that the first command is STOP */
out_le32(&pi->dbdma->control, RUN | (RUN << 16));
timeout = 100;
/* wait for STOP to be executed */
while (in_le32(&pi->dbdma->status) & ACTIVE && timeout--)
udelay(1);
if (timeout <= 0) {
printk(KERN_ERR "i2sbus: error waiting for dma stop\n");
result = -ENXIO;
goto out_unlock;
}
/* again, write dma command buffer address to the dbdma chip,
* this time of the first real command */
*pi->dbdma_ring.cmds = tmp;
out_le32(&pi->dbdma->cmdptr, pi->dbdma_ring.bus_cmd_start);
/* post write */
mb();
(void)in_le32(&pi->dbdma->status);
/* reset dma engine again */
out_le32(&pi->dbdma->control,
0 | (RUN | PAUSE | FLUSH | WAKE) << 16);
timeout = 100;
while (in_le32(&pi->dbdma->status) & RUN && timeout--)
udelay(1);
if (timeout <= 0) {
printk(KERN_ERR
"i2sbus: error waiting for dma reset\n");
result = -ENXIO;
goto out_unlock;
}
/* wake up the chip with the next descriptor */
out_le32(&pi->dbdma->control,
(RUN | WAKE) | ((RUN | WAKE) << 16));
/* get the frame count */
pi->frame_count = in_le32(&i2sdev->intfregs->frame_count);
/* off you go! */
break;
case SNDRV_PCM_TRIGGER_STOP:
case SNDRV_PCM_TRIGGER_SUSPEND:
if (!pi->dbdma_ring.running) {
result = -EALREADY;
goto out_unlock;
}
/* turn off all relevant bits */
out_le32(&pi->dbdma->control,
(RUN | WAKE | FLUSH | PAUSE) << 16);
{
/* FIXME: move to own function */
int timeout = 5000;
while ((in_le32(&pi->dbdma->status) & RUN)
&& --timeout > 0)
udelay(1);
if (!timeout)
printk(KERN_ERR
"i2sbus: timed out turning "
"off dbdma engine!\n");
}
pi->dbdma_ring.running = 0;
list_for_each_entry(cii, &i2sdev->sound.codec_list, list)
if (cii->codec->stop)
cii->codec->stop(cii, pi->substream);
break;
default:
result = -EINVAL;
goto out_unlock;
}
out_unlock:
spin_unlock_irqrestore(&i2sdev->low_lock, flags);
return result;
}
static snd_pcm_uframes_t i2sbus_pcm_pointer(struct i2sbus_dev *i2sdev, int in)
{
struct pcm_info *pi;
u32 fc;
get_pcm_info(i2sdev, in, &pi, NULL);
fc = in_le32(&i2sdev->intfregs->frame_count);
fc = fc - pi->frame_count;
return (bytes_to_frames(pi->substream->runtime,
pi->current_period *
snd_pcm_lib_period_bytes(pi->substream))
+ fc) % pi->substream->runtime->buffer_size;
}
static inline void handle_interrupt(struct i2sbus_dev *i2sdev, int in)
{
struct pcm_info *pi;
u32 fc;
u32 delta;
spin_lock(&i2sdev->low_lock);
get_pcm_info(i2sdev, in, &pi, NULL);
if (!pi->dbdma_ring.running) {
/* there was still an interrupt pending
* while we stopped. or maybe another
* processor (not the one that was stopping
* the DMA engine) was spinning above
* waiting for the lock. */
goto out_unlock;
}
fc = in_le32(&i2sdev->intfregs->frame_count);
/* a counter overflow does not change the calculation. */
delta = fc - pi->frame_count;
/* update current_period */
while (delta >= pi->substream->runtime->period_size) {
pi->current_period++;
delta = delta - pi->substream->runtime->period_size;
}
if (unlikely(delta)) {
/* Some interrupt came late, so check the dbdma.
* This special case exists to syncronize the frame_count with
* the dbdma transfer, but is hit every once in a while. */
int period;
period = (in_le32(&pi->dbdma->cmdptr)
- pi->dbdma_ring.bus_cmd_start)
/ sizeof(struct dbdma_cmd);
pi->current_period = pi->current_period
% pi->substream->runtime->periods;
while (pi->current_period != period) {
pi->current_period++;
pi->current_period %= pi->substream->runtime->periods;
/* Set delta to zero, as the frame_count value is too
* high (otherwise the code path will not be executed).
* This corrects the fact that the frame_count is too
* low at the beginning due to buffering. */
delta = 0;
}
}
pi->frame_count = fc - delta;
pi->current_period %= pi->substream->runtime->periods;
spin_unlock(&i2sdev->low_lock);
/* may call _trigger again, hence needs to be unlocked */
snd_pcm_period_elapsed(pi->substream);
return;
out_unlock:
spin_unlock(&i2sdev->low_lock);
}
irqreturn_t i2sbus_tx_intr(int irq, void *devid, struct pt_regs *regs)
{
handle_interrupt((struct i2sbus_dev *)devid, 0);
return IRQ_HANDLED;
}
irqreturn_t i2sbus_rx_intr(int irq, void *devid, struct pt_regs * regs)
{
handle_interrupt((struct i2sbus_dev *)devid, 1);
return IRQ_HANDLED;
}
static int i2sbus_playback_open(struct snd_pcm_substream *substream)
{
struct i2sbus_dev *i2sdev = snd_pcm_substream_chip(substream);
if (!i2sdev)
return -EINVAL;
i2sdev->out.substream = substream;
return i2sbus_pcm_open(i2sdev, 0);
}
static int i2sbus_playback_close(struct snd_pcm_substream *substream)
{
struct i2sbus_dev *i2sdev = snd_pcm_substream_chip(substream);
int err;
if (!i2sdev)
return -EINVAL;
if (i2sdev->out.substream != substream)
return -EINVAL;
err = i2sbus_pcm_close(i2sdev, 0);
if (!err)
i2sdev->out.substream = NULL;
return err;
}
static int i2sbus_playback_prepare(struct snd_pcm_substream *substream)
{
struct i2sbus_dev *i2sdev = snd_pcm_substream_chip(substream);
if (!i2sdev)
return -EINVAL;
if (i2sdev->out.substream != substream)
return -EINVAL;
return i2sbus_pcm_prepare(i2sdev, 0);
}
static int i2sbus_playback_trigger(struct snd_pcm_substream *substream, int cmd)
{
struct i2sbus_dev *i2sdev = snd_pcm_substream_chip(substream);
if (!i2sdev)
return -EINVAL;
if (i2sdev->out.substream != substream)
return -EINVAL;
return i2sbus_pcm_trigger(i2sdev, 0, cmd);
}
static snd_pcm_uframes_t i2sbus_playback_pointer(struct snd_pcm_substream
*substream)
{
struct i2sbus_dev *i2sdev = snd_pcm_substream_chip(substream);
if (!i2sdev)
return -EINVAL;
if (i2sdev->out.substream != substream)
return 0;
return i2sbus_pcm_pointer(i2sdev, 0);
}
static struct snd_pcm_ops i2sbus_playback_ops = {
.open = i2sbus_playback_open,
.close = i2sbus_playback_close,
.ioctl = snd_pcm_lib_ioctl,
.hw_params = i2sbus_hw_params,
.hw_free = i2sbus_hw_free,
.prepare = i2sbus_playback_prepare,
.trigger = i2sbus_playback_trigger,
.pointer = i2sbus_playback_pointer,
};
static int i2sbus_record_open(struct snd_pcm_substream *substream)
{
struct i2sbus_dev *i2sdev = snd_pcm_substream_chip(substream);
if (!i2sdev)
return -EINVAL;
i2sdev->in.substream = substream;
return i2sbus_pcm_open(i2sdev, 1);
}
static int i2sbus_record_close(struct snd_pcm_substream *substream)
{
struct i2sbus_dev *i2sdev = snd_pcm_substream_chip(substream);
int err;
if (!i2sdev)
return -EINVAL;
if (i2sdev->in.substream != substream)
return -EINVAL;
err = i2sbus_pcm_close(i2sdev, 1);
if (!err)
i2sdev->in.substream = NULL;
return err;
}
static int i2sbus_record_prepare(struct snd_pcm_substream *substream)
{
struct i2sbus_dev *i2sdev = snd_pcm_substream_chip(substream);
if (!i2sdev)
return -EINVAL;
if (i2sdev->in.substream != substream)
return -EINVAL;
return i2sbus_pcm_prepare(i2sdev, 1);
}
static int i2sbus_record_trigger(struct snd_pcm_substream *substream, int cmd)
{
struct i2sbus_dev *i2sdev = snd_pcm_substream_chip(substream);
if (!i2sdev)
return -EINVAL;
if (i2sdev->in.substream != substream)
return -EINVAL;
return i2sbus_pcm_trigger(i2sdev, 1, cmd);
}
static snd_pcm_uframes_t i2sbus_record_pointer(struct snd_pcm_substream
*substream)
{
struct i2sbus_dev *i2sdev = snd_pcm_substream_chip(substream);
if (!i2sdev)
return -EINVAL;
if (i2sdev->in.substream != substream)
return 0;
return i2sbus_pcm_pointer(i2sdev, 1);
}
static struct snd_pcm_ops i2sbus_record_ops = {
.open = i2sbus_record_open,
.close = i2sbus_record_close,
.ioctl = snd_pcm_lib_ioctl,
.hw_params = i2sbus_hw_params,
.hw_free = i2sbus_hw_free,
.prepare = i2sbus_record_prepare,
.trigger = i2sbus_record_trigger,
.pointer = i2sbus_record_pointer,
};
static void i2sbus_private_free(struct snd_pcm *pcm)
{
struct i2sbus_dev *i2sdev = snd_pcm_chip(pcm);
struct codec_info_item *p, *tmp;
i2sdev->sound.pcm = NULL;
i2sdev->out.created = 0;
i2sdev->in.created = 0;
list_for_each_entry_safe(p, tmp, &i2sdev->sound.codec_list, list) {
printk(KERN_ERR "i2sbus: a codec didn't unregister!\n");
list_del(&p->list);
module_put(p->codec->owner);
kfree(p);
}
soundbus_dev_put(&i2sdev->sound);
module_put(THIS_MODULE);
}
/* FIXME: this function needs an error handling strategy with labels */
int
i2sbus_attach_codec(struct soundbus_dev *dev, struct snd_card *card,
struct codec_info *ci, void *data)
{
int err, in = 0, out = 0;
struct transfer_info *tmp;
struct i2sbus_dev *i2sdev = soundbus_dev_to_i2sbus_dev(dev);
struct codec_info_item *cii;
if (!dev->pcmname || dev->pcmid == -1) {
printk(KERN_ERR "i2sbus: pcm name and id must be set!\n");
return -EINVAL;
}
list_for_each_entry(cii, &dev->codec_list, list) {
if (cii->codec_data == data)
return -EALREADY;
}
if (!ci->transfers || !ci->transfers->formats
|| !ci->transfers->rates || !ci->usable)
return -EINVAL;
/* we currently code the i2s transfer on the clock, and support only
* 32 and 64 */
if (ci->bus_factor != 32 && ci->bus_factor != 64)
return -EINVAL;
/* If you want to fix this, you need to keep track of what transport infos
* are to be used, which codecs they belong to, and then fix all the
* sysclock/busclock stuff above to depend on which is usable */
list_for_each_entry(cii, &dev->codec_list, list) {
if (cii->codec->sysclock_factor != ci->sysclock_factor) {
printk(KERN_DEBUG
"cannot yet handle multiple different sysclocks!\n");
return -EINVAL;
}
if (cii->codec->bus_factor != ci->bus_factor) {
printk(KERN_DEBUG
"cannot yet handle multiple different bus clocks!\n");
return -EINVAL;
}
}
tmp = ci->transfers;
while (tmp->formats && tmp->rates) {
if (tmp->transfer_in)
in = 1;
else
out = 1;
tmp++;
}
cii = kzalloc(sizeof(struct codec_info_item), GFP_KERNEL);
if (!cii) {
printk(KERN_DEBUG "i2sbus: failed to allocate cii\n");
return -ENOMEM;
}
/* use the private data to point to the codec info */
cii->sdev = soundbus_dev_get(dev);
cii->codec = ci;
cii->codec_data = data;
if (!cii->sdev) {
printk(KERN_DEBUG
"i2sbus: failed to get soundbus dev reference\n");
kfree(cii);
return -ENODEV;
}
if (!try_module_get(THIS_MODULE)) {
printk(KERN_DEBUG "i2sbus: failed to get module reference!\n");
soundbus_dev_put(dev);
kfree(cii);
return -EBUSY;
}
if (!try_module_get(ci->owner)) {
printk(KERN_DEBUG
"i2sbus: failed to get module reference to codec owner!\n");
module_put(THIS_MODULE);
soundbus_dev_put(dev);
kfree(cii);
return -EBUSY;
}
if (!dev->pcm) {
err = snd_pcm_new(card,
dev->pcmname,
dev->pcmid,
0,
0,
&dev->pcm);
if (err) {
printk(KERN_DEBUG "i2sbus: failed to create pcm\n");
kfree(cii);
module_put(ci->owner);
soundbus_dev_put(dev);
module_put(THIS_MODULE);
return err;
}
}
/* ALSA yet again sucks.
* If it is ever fixed, remove this line. See below. */
out = in = 1;
if (!i2sdev->out.created && out) {
if (dev->pcm->card != card) {
/* eh? */
printk(KERN_ERR
"Can't attach same bus to different cards!\n");
module_put(ci->owner);
kfree(cii);
soundbus_dev_put(dev);
module_put(THIS_MODULE);
return -EINVAL;
}
if ((err =
snd_pcm_new_stream(dev->pcm, SNDRV_PCM_STREAM_PLAYBACK, 1))) {
module_put(ci->owner);
kfree(cii);
soundbus_dev_put(dev);
module_put(THIS_MODULE);
return err;
}
snd_pcm_set_ops(dev->pcm, SNDRV_PCM_STREAM_PLAYBACK,
&i2sbus_playback_ops);
i2sdev->out.created = 1;
}
if (!i2sdev->in.created && in) {
if (dev->pcm->card != card) {
printk(KERN_ERR
"Can't attach same bus to different cards!\n");
module_put(ci->owner);
kfree(cii);
soundbus_dev_put(dev);
module_put(THIS_MODULE);
return -EINVAL;
}
if ((err =
snd_pcm_new_stream(dev->pcm, SNDRV_PCM_STREAM_CAPTURE, 1))) {
module_put(ci->owner);
kfree(cii);
soundbus_dev_put(dev);
module_put(THIS_MODULE);
return err;
}
snd_pcm_set_ops(dev->pcm, SNDRV_PCM_STREAM_CAPTURE,
&i2sbus_record_ops);
i2sdev->in.created = 1;
}
/* so we have to register the pcm after adding any substream
* to it because alsa doesn't create the devices for the
* substreams when we add them later.
* Therefore, force in and out on both busses (above) and
* register the pcm now instead of just after creating it.
*/
err = snd_device_register(card, dev->pcm);
if (err) {
printk(KERN_ERR "i2sbus: error registering new pcm\n");
module_put(ci->owner);
kfree(cii);
soundbus_dev_put(dev);
module_put(THIS_MODULE);
return err;
}
/* no errors any more, so let's add this to our list */
list_add(&cii->list, &dev->codec_list);
dev->pcm->private_data = i2sdev;
dev->pcm->private_free = i2sbus_private_free;
/* well, we really should support scatter/gather DMA */
snd_pcm_lib_preallocate_pages_for_all(
dev->pcm, SNDRV_DMA_TYPE_DEV,
snd_dma_pci_data(macio_get_pci_dev(i2sdev->macio)),
64 * 1024, 64 * 1024);
return 0;
}
void i2sbus_detach_codec(struct soundbus_dev *dev, void *data)
{
struct codec_info_item *cii = NULL, *i;
list_for_each_entry(i, &dev->codec_list, list) {
if (i->codec_data == data) {
cii = i;
break;
}
}
if (cii) {
list_del(&cii->list);
module_put(cii->codec->owner);
kfree(cii);
}
/* no more codecs, but still a pcm? */
if (list_empty(&dev->codec_list) && dev->pcm) {
/* the actual cleanup is done by the callback above! */
snd_device_free(dev->pcm->card, dev->pcm);
}
}
sound/aoa/soundbus/i2sbus/i2sbus.h 0 → 100644
View file @ f3d9478b
/*
* i2sbus driver -- private definitions
*
* Copyright 2006 Johannes Berg <johannes@sipsolutions.net>
*
* GPL v2, can be found in COPYING.
*/
#ifndef __I2SBUS_H
#define __I2SBUS_H
#include <asm/dbdma.h>
#include <linux/interrupt.h>
#include <sound/pcm.h>
#include <linux/spinlock.h>
#include <linux/mutex.h>
#include <asm/prom.h>
#include "i2sbus-interface.h"
#include "i2sbus-control.h"
#include "../soundbus.h"
struct i2sbus_control {
volatile struct i2s_control_regs __iomem *controlregs;
struct resource rsrc;
struct list_head list;
};
#define MAX_DBDMA_COMMANDS 32
struct dbdma_command_mem {
dma_addr_t bus_addr;
dma_addr_t bus_cmd_start;
struct dbdma_cmd *cmds;
void *space;
int size;
u32 running:1;
};
struct pcm_info {
u32 created:1, /* has this direction been created with alsa? */
active:1; /* is this stream active? */
/* runtime information */
struct snd_pcm_substream *substream;
int current_period;
u32 frame_count;
struct dbdma_command_mem dbdma_ring;
volatile struct dbdma_regs __iomem *dbdma;
};
struct i2sbus_dev {
struct soundbus_dev sound;
struct macio_dev *macio;
struct i2sbus_control *control;
volatile struct i2s_interface_regs __iomem *intfregs;
struct resource resources[3];
struct resource *allocated_resource[3];
int interrupts[3];
char rnames[3][32];
/* info about currently active substreams */
struct pcm_info out, in;
snd_pcm_format_t format;
unsigned int rate;
/* list for a single controller */
struct list_head item;
/* number of bus on controller */
int bus_number;
/* for use by control layer */
struct pmf_function *enable,
*cell_enable,
*cell_disable,
*clock_enable,
*clock_disable;
/* locks */
/* spinlock for low-level interrupt locking */
spinlock_t low_lock;
/* mutex for high-level consistency */
struct mutex lock;
};
#define soundbus_dev_to_i2sbus_dev(sdev) \
container_of(sdev, struct i2sbus_dev, sound)
/* pcm specific functions */
extern int
i2sbus_attach_codec(struct soundbus_dev *dev, struct snd_card *card,
struct codec_info *ci, void *data);
extern void
i2sbus_detach_codec(struct soundbus_dev *dev, void *data);
extern irqreturn_t
i2sbus_tx_intr(int irq, void *devid, struct pt_regs *regs);
extern irqreturn_t
i2sbus_rx_intr(int irq, void *devid, struct pt_regs *regs);
/* control specific functions */
extern int i2sbus_control_init(struct macio_dev* dev,
struct i2sbus_control **c);
extern void i2sbus_control_destroy(struct i2sbus_control *c);
extern int i2sbus_control_add_dev(struct i2sbus_control *c,
struct i2sbus_dev *i2sdev);
extern void i2sbus_control_remove_dev(struct i2sbus_control *c,
struct i2sbus_dev *i2sdev);
extern int i2sbus_control_enable(struct i2sbus_control *c,
struct i2sbus_dev *i2sdev);
extern int i2sbus_control_cell(struct i2sbus_control *c,
struct i2sbus_dev *i2sdev,
int enable);
extern int i2sbus_control_clock(struct i2sbus_control *c,
struct i2sbus_dev *i2sdev,
int enable);
#endif /* __I2SBUS_H */
sound/aoa/soundbus/soundbus.h 0 → 100644
View file @ f3d9478b
/*
* soundbus generic definitions
*
* Copyright 2006 Johannes Berg <johannes@sipsolutions.net>
*
* GPL v2, can be found in COPYING.
*/
#ifndef __SOUNDBUS_H
#define __SOUNDBUS_H
#include <asm/of_device.h>
#include <sound/pcm.h>
#include <linux/list.h>
/* When switching from master to slave or the other way around,
* you don't want to have the codec chip acting as clock source
* while the bus still is.
* More importantly, while switch from slave to master, you need
* to turn off the chip's master function first, but then there's
* no clock for a while and other chips might reset, so we notify
* their drivers after having switched.
* The constants here are codec-point of view, so when we switch
* the soundbus to master we tell the codec we're going to switch
* and give it CLOCK_SWITCH_PREPARE_SLAVE!
*/
enum clock_switch {
CLOCK_SWITCH_PREPARE_SLAVE,
CLOCK_SWITCH_PREPARE_MASTER,
CLOCK_SWITCH_SLAVE,
CLOCK_SWITCH_MASTER,
CLOCK_SWITCH_NOTIFY,
};
/* information on a transfer the codec can take */
struct transfer_info {
u64 formats; /* SNDRV_PCM_FMTBIT_* */
unsigned int rates; /* SNDRV_PCM_RATE_* */
/* flags */
u32 transfer_in:1, /* input = 1, output = 0 */
must_be_clock_source:1;
/* for codecs to distinguish among their TIs */
int tag;
};
struct codec_info_item {
struct codec_info *codec;
void *codec_data;
struct soundbus_dev *sdev;
/* internal, to be used by the soundbus provider */
struct list_head list;
};
/* for prepare, where the codecs need to know
* what we're going to drive the bus with */
struct bus_info {
/* see below */
int sysclock_factor;
int bus_factor;
};
/* information on the codec itself, plus function pointers */
struct codec_info {
/* the module this lives in */
struct module *owner;
/* supported transfer possibilities, array terminated by
* formats or rates being 0. */
struct transfer_info *transfers;
/* Master clock speed factor
* to be used (master clock speed = sysclock_factor * sampling freq)
* Unused if the soundbus provider has no such notion.
*/
int sysclock_factor;
/* Bus factor, bus clock speed = bus_factor * sampling freq)
* Unused if the soundbus provider has no such notion.
*/
int bus_factor;
/* operations */
/* clock switching, see above */
int (*switch_clock)(struct codec_info_item *cii,
enum clock_switch clock);
/* called for each transfer_info when the user
* opens the pcm device to determine what the
* hardware can support at this point in time.
* That can depend on other user-switchable controls.
* Return 1 if usable, 0 if not.
* out points to another instance of a transfer_info
* which is initialised to the values in *ti, and
* it's format and rate values can be modified by
* the callback if it is necessary to further restrict
* the formats that can be used at the moment, for
* example when one codec has multiple logical codec
* info structs for multiple inputs.
*/
int (*usable)(struct codec_info_item *cii,
struct transfer_info *ti,
struct transfer_info *out);
/* called when pcm stream is opened, probably not implemented
* most of the time since it isn't too useful */
int (*open)(struct codec_info_item *cii,
struct snd_pcm_substream *substream);
/* called when the pcm stream is closed, at this point
* the user choices can all be unlocked (see below) */
int (*close)(struct codec_info_item *cii,
struct snd_pcm_substream *substream);
/* if the codec must forbid some user choices because
* they are not valid with the substream/transfer info,
* it must do so here. Example: no digital output for
* incompatible framerate, say 8KHz, on Onyx.
* If the selected stuff in the substream is NOT
* compatible, you have to reject this call! */
int (*prepare)(struct codec_info_item *cii,
struct bus_info *bi,
struct snd_pcm_substream *substream);
/* start() is called before data is pushed to the codec.
* Note that start() must be atomic! */
int (*start)(struct codec_info_item *cii,
struct snd_pcm_substream *substream);
/* stop() is called after data is no longer pushed to the codec.
* Note that stop() must be atomic! */
int (*stop)(struct codec_info_item *cii,
struct snd_pcm_substream *substream);
int (*suspend)(struct codec_info_item *cii, pm_message_t state);
int (*resume)(struct codec_info_item *cii);
};
/* information on a soundbus device */
struct soundbus_dev {
/* the bus it belongs to */
struct list_head onbuslist;
/* the of device it represents */
struct of_device ofdev;
/* what modules go by */
char modalias[32];
/* These fields must be before attach_codec can be called.
* They should be set by the owner of the alsa card object
* that is needed, and whoever sets them must make sure
* that they are unique within that alsa card object. */
char *pcmname;
int pcmid;
/* this is assigned by the soundbus provider in attach_codec */
struct snd_pcm *pcm;
/* operations */
/* attach a codec to this soundbus, give the alsa
* card object the PCMs for this soundbus should be in.
* The 'data' pointer must be unique, it is used as the
* key for detach_codec(). */
int (*attach_codec)(struct soundbus_dev *dev, struct snd_card *card,
struct codec_info *ci, void *data);
void (*detach_codec)(struct soundbus_dev *dev, void *data);
/* TODO: suspend/resume */
/* private for the soundbus provider */
struct list_head codec_list;
u32 have_out:1, have_in:1;
};
#define to_soundbus_device(d) container_of(d, struct soundbus_dev, ofdev.dev)
#define of_to_soundbus_device(d) container_of(d, struct soundbus_dev, ofdev)
extern int soundbus_add_one(struct soundbus_dev *dev);
extern void soundbus_remove_one(struct soundbus_dev *dev);
extern struct soundbus_dev *soundbus_dev_get(struct soundbus_dev *dev);
extern void soundbus_dev_put(struct soundbus_dev *dev);
struct soundbus_driver {
char *name;
struct module *owner;
/* we don't implement any matching at all */
int (*probe)(struct soundbus_dev* dev);
int (*remove)(struct soundbus_dev* dev);
int (*suspend)(struct soundbus_dev* dev, pm_message_t state);
int (*resume)(struct soundbus_dev* dev);
int (*shutdown)(struct soundbus_dev* dev);
struct device_driver driver;
};
#define to_soundbus_driver(drv) container_of(drv,struct soundbus_driver, driver)
extern int soundbus_register_driver(struct soundbus_driver *drv);
extern void soundbus_unregister_driver(struct soundbus_driver *drv);
#endif /* __SOUNDBUS_H */
sound/aoa/soundbus/sysfs.c 0 → 100644
View file @ f3d9478b
#include <linux/config.h>
#include <linux/kernel.h>
#include <linux/stat.h>
/* FIX UP */
#include "soundbus.h"
#define soundbus_config_of_attr(field, format_string) \
static ssize_t \
field##_show (struct device *dev, struct device_attribute *attr, \
char *buf) \
{ \
struct soundbus_dev *mdev = to_soundbus_device (dev); \
return sprintf (buf, format_string, mdev->ofdev.node->field); \
}
static ssize_t modalias_show(struct device *dev, struct device_attribute *attr,
char *buf)
{
struct soundbus_dev *sdev = to_soundbus_device(dev);
struct of_device *of = &sdev->ofdev;
int length;
if (*sdev->modalias) {
strlcpy(buf, sdev->modalias, sizeof(sdev->modalias) + 1);
strcat(buf, "\n");
length = strlen(buf);
} else {
length = sprintf(buf, "of:N%sT%s\n",
of->node->name, of->node->type);
}
return length;
}
soundbus_config_of_attr (name, "%s\n");
soundbus_config_of_attr (type, "%s\n");
struct device_attribute soundbus_dev_attrs[] = {
__ATTR_RO(name),
__ATTR_RO(type),
__ATTR_RO(modalias),
__ATTR_NULL
};
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