* [RFC 0/7] snd-aoa: add snd-aoa
@ 2006-05-28 19:00 Johannes Berg
2006-05-28 19:00 ` Johannes Berg
` (7 more replies)
0 siblings, 8 replies; 45+ messages in thread
From: Johannes Berg @ 2006-05-28 19:00 UTC (permalink / raw)
To: alsa-devel; +Cc: linuxppc-dev
The following patches would add snd-aoa to the Linux source tree. I'm
posting them here in the spirit of getting review fairly early, it works
fine for those machines it handles but has some rough edges like not
providing headphone detection at the moment.
There are plans to fix this, but they might be influenced by your comments
so I'm holding off there a bit. One thing that will probably be done is
moving all the alsa controls for various inputs from the codecs to the
fabric, the fabric needs to know these anyway and code can be simplified and
prepared for headphone/... detection if the codecs simply provide hooks to
turn on/off the various in- and outputs.
Any comments are appreciated.
Thanks,
johannes
^ permalink raw reply [flat|nested] 45+ messages in thread
* [RFC 1/7] snd-aoa: add aoa header files
2006-05-28 19:00 [RFC 0/7] snd-aoa: add snd-aoa Johannes Berg
@ 2006-05-28 19:00 ` Johannes Berg
2006-05-28 19:00 ` Johannes Berg
` (6 subsequent siblings)
7 siblings, 0 replies; 45+ messages in thread
From: Johannes Berg @ 2006-05-28 19:00 UTC (permalink / raw)
To: alsa-devel; +Cc: linuxppc-dev
[-- Attachment #1: 01-add-headers.patch --]
[-- Type: text/plain, Size: 5488 bytes --]
This patch adds header files to use for communication between the various
parts of aoa.
--- /dev/null
+++ b/sound/aoa/aoa-gpio.h
@@ -0,0 +1,44 @@
+/*
+ * 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 <asm/prom.h>
+
+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);
+};
+
+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;
+};
+
+#endif /* __AOA_GPIO_H */
--- /dev/null
+++ b/sound/aoa/aoa.h
@@ -0,0 +1,130 @@
+/*
+ * 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 *map_gpio_methods; */
+
+#endif /* __AOA_H */
--
^ permalink raw reply [flat|nested] 45+ messages in thread
* [RFC 1/7] snd-aoa: add aoa header files
@ 2006-05-28 19:00 ` Johannes Berg
0 siblings, 0 replies; 45+ messages in thread
From: Johannes Berg @ 2006-05-28 19:00 UTC (permalink / raw)
To: alsa-devel; +Cc: linuxppc-dev
This patch adds header files to use for communication between the various
parts of aoa.
--- /dev/null
+++ b/sound/aoa/aoa-gpio.h
@@ -0,0 +1,44 @@
+/*
+ * 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 <asm/prom.h>
+
+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);
+};
+
+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;
+};
+
+#endif /* __AOA_GPIO_H */
--- /dev/null
+++ b/sound/aoa/aoa.h
@@ -0,0 +1,130 @@
+/*
+ * 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 *map_gpio_methods; */
+
+#endif /* __AOA_H */
--
^ permalink raw reply [flat|nested] 45+ messages in thread
* [RFC 2/7] snd-aoa: add aoa core
2006-05-28 19:00 [RFC 0/7] snd-aoa: add snd-aoa Johannes Berg
@ 2006-05-28 19:00 ` Johannes Berg
2006-05-28 19:00 ` Johannes Berg
` (6 subsequent siblings)
7 siblings, 0 replies; 45+ messages in thread
From: Johannes Berg @ 2006-05-28 19:00 UTC (permalink / raw)
To: alsa-devel; +Cc: linuxppc-dev
[-- Attachment #1: 02-add-core.patch --]
[-- Type: text/plain, Size: 8963 bytes --]
This patch adds the core of aoa, in itself pretty useless, but providing
useful functions to other modules.
--- /dev/null
+++ b/sound/aoa/core/Makefile
@@ -0,0 +1,4 @@
+obj-$(CONFIG_SND_AOA) += snd-aoa.o
+snd-aoa-objs := snd-aoa-core.o \
+ snd-aoa-alsa.o \
+ snd-aoa-gpio-pmf.o
--- /dev/null
+++ b/sound/aoa/core/snd-aoa-alsa.c
@@ -0,0 +1,91 @@
+/*
+ * 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 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(-1, name, mod, sizeof(struct aoa_card));
+ if (!alsa_card)
+ return -ENOMEM;
+ aoa_card = alsa_card->private_data;
+ aoa_card->alsa_card = alsa_card;
+ strncpy(alsa_card->driver, "AppleOnbdAudio", sizeof(alsa_card->driver));
+ strncpy(alsa_card->shortname, name, sizeof(alsa_card->shortname));
+ strncpy(alsa_card->longname, name, sizeof(alsa_card->longname));
+ strncpy(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);
--- /dev/null
+++ b/sound/aoa/core/snd-aoa-alsa.h
@@ -0,0 +1,17 @@
+/*
+ * 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
+/* FIXME */
+#include "../aoa.h"
+
+extern int aoa_alsa_init(char *name, struct module *mod);
+extern void aoa_alsa_cleanup(void);
+
+#endif /* __SND_AOA_ALSA_H */
--- /dev/null
+++ b/sound/aoa/core/snd-aoa-core.c
@@ -0,0 +1,153 @@
+/*
+ * 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 void attach_codec_to_fabric(struct aoa_codec *c)
+{
+ int err;
+
+ if (!try_module_get(c->owner))
+ return;
+ /* found_codec has to be assigned */
+ err = -ENOENT;
+ if (fabric->found_codec)
+ err = fabric->found_codec(c);
+ if (err) {
+ module_put(c->owner);
+ printk("snd-aoa: fabric didn't like codec %s\n", c->name);
+ return;
+ }
+ c->fabric = fabric;
+
+ err = 0;
+ if (c->init)
+ err = c->init(c);
+ if (err) {
+ printk("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;
+ }
+ if (fabric->attached_codec)
+ fabric->attached_codec(c);
+}
+
+int aoa_codec_register(struct aoa_codec *codec)
+{
+ list_add(&codec->list, &codec_list);
+ if (fabric)
+ attach_codec_to_fabric(codec);
+ return 0;
+}
+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);
--- /dev/null
+++ b/sound/aoa/core/snd-aoa-gpio-pmf.c
@@ -0,0 +1,91 @@
+/*
+ * 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_gpio_init(struct gpio_runtime *rt)
+{
+ pmf_gpio_all_amps_off(rt);
+ rt->implementation_private = 0;
+}
+
+static void pmf_gpio_exit(struct gpio_runtime *rt)
+{
+ pmf_gpio_all_amps_off(rt);
+ rt->implementation_private = 0;
+}
+
+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,
+};
+
+struct gpio_methods *pmf_gpio_methods = &methods;
+EXPORT_SYMBOL_GPL(pmf_gpio_methods);
--
^ permalink raw reply [flat|nested] 45+ messages in thread
* [RFC 2/7] snd-aoa: add aoa core
@ 2006-05-28 19:00 ` Johannes Berg
0 siblings, 0 replies; 45+ messages in thread
From: Johannes Berg @ 2006-05-28 19:00 UTC (permalink / raw)
To: alsa-devel; +Cc: linuxppc-dev
This patch adds the core of aoa, in itself pretty useless, but providing
useful functions to other modules.
--- /dev/null
+++ b/sound/aoa/core/Makefile
@@ -0,0 +1,4 @@
+obj-$(CONFIG_SND_AOA) += snd-aoa.o
+snd-aoa-objs := snd-aoa-core.o \
+ snd-aoa-alsa.o \
+ snd-aoa-gpio-pmf.o
--- /dev/null
+++ b/sound/aoa/core/snd-aoa-alsa.c
@@ -0,0 +1,91 @@
+/*
+ * 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 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(-1, name, mod, sizeof(struct aoa_card));
+ if (!alsa_card)
+ return -ENOMEM;
+ aoa_card = alsa_card->private_data;
+ aoa_card->alsa_card = alsa_card;
+ strncpy(alsa_card->driver, "AppleOnbdAudio", sizeof(alsa_card->driver));
+ strncpy(alsa_card->shortname, name, sizeof(alsa_card->shortname));
+ strncpy(alsa_card->longname, name, sizeof(alsa_card->longname));
+ strncpy(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);
--- /dev/null
+++ b/sound/aoa/core/snd-aoa-alsa.h
@@ -0,0 +1,17 @@
+/*
+ * 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
+/* FIXME */
+#include "../aoa.h"
+
+extern int aoa_alsa_init(char *name, struct module *mod);
+extern void aoa_alsa_cleanup(void);
+
+#endif /* __SND_AOA_ALSA_H */
--- /dev/null
+++ b/sound/aoa/core/snd-aoa-core.c
@@ -0,0 +1,153 @@
+/*
+ * 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 void attach_codec_to_fabric(struct aoa_codec *c)
+{
+ int err;
+
+ if (!try_module_get(c->owner))
+ return;
+ /* found_codec has to be assigned */
+ err = -ENOENT;
+ if (fabric->found_codec)
+ err = fabric->found_codec(c);
+ if (err) {
+ module_put(c->owner);
+ printk("snd-aoa: fabric didn't like codec %s\n", c->name);
+ return;
+ }
+ c->fabric = fabric;
+
+ err = 0;
+ if (c->init)
+ err = c->init(c);
+ if (err) {
+ printk("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;
+ }
+ if (fabric->attached_codec)
+ fabric->attached_codec(c);
+}
+
+int aoa_codec_register(struct aoa_codec *codec)
+{
+ list_add(&codec->list, &codec_list);
+ if (fabric)
+ attach_codec_to_fabric(codec);
+ return 0;
+}
+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);
--- /dev/null
+++ b/sound/aoa/core/snd-aoa-gpio-pmf.c
@@ -0,0 +1,91 @@
+/*
+ * 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_gpio_init(struct gpio_runtime *rt)
+{
+ pmf_gpio_all_amps_off(rt);
+ rt->implementation_private = 0;
+}
+
+static void pmf_gpio_exit(struct gpio_runtime *rt)
+{
+ pmf_gpio_all_amps_off(rt);
+ rt->implementation_private = 0;
+}
+
+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,
+};
+
+struct gpio_methods *pmf_gpio_methods = &methods;
+EXPORT_SYMBOL_GPL(pmf_gpio_methods);
--
^ permalink raw reply [flat|nested] 45+ messages in thread
* [RFC 3/7] snd-aoa: add soundbus and i2sbus
2006-05-28 19:00 [RFC 0/7] snd-aoa: add snd-aoa Johannes Berg
2006-05-28 19:00 ` Johannes Berg
2006-05-28 19:00 ` Johannes Berg
@ 2006-05-28 19:00 ` Johannes Berg
2006-05-28 19:00 ` Johannes Berg
` (4 subsequent siblings)
7 siblings, 0 replies; 45+ messages in thread
From: Johannes Berg @ 2006-05-28 19:00 UTC (permalink / raw)
To: alsa-devel; +Cc: linuxppc-dev
This patch adds the mostly generic soundbus that will be used for older
things (davbus comes to mind) as well as adding i2sbus, the module handling
the i2s bus on Apple mac-io chips.
--- /dev/null
+++ b/sound/aoa/soundbus/Kconfig
@@ -0,0 +1,14 @@
+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.
--- /dev/null
+++ b/sound/aoa/soundbus/Makefile
@@ -0,0 +1,3 @@
+obj-$(CONFIG_SND_AOA_SOUNDBUS) += snd-aoa-soundbus.o
+snd-aoa-soundbus-objs := core.o sysfs.o
+obj-$(CONFIG_SND_AOA_SOUNDBUS_I2S) += i2sbus/
--- /dev/null
+++ b/sound/aoa/soundbus/core.c
@@ -0,0 +1,250 @@
+/*
+ * 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);
--- /dev/null
+++ b/sound/aoa/soundbus/i2sbus/Makefile
@@ -0,0 +1,2 @@
+obj-$(CONFIG_SND_AOA_SOUNDBUS_I2S) += snd-aoa-i2sbus.o
+snd-aoa-i2sbus-objs := i2sbus-core.o i2sbus-pcm.o i2sbus-control.o
--- /dev/null
+++ b/sound/aoa/soundbus/i2sbus/i2sbus-control.c
@@ -0,0 +1,213 @@
+/*
+ * 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;
+ /* HACK HACK HACK
+ * And one that doesn't even work to boot.
+ * We should be using the feature calls for this, but
+ * since no one else uses the registers we currently hit
+ * this is fine for now. */
+ (*c)->rsrc.end = (*c)->rsrc.start + sizeof(struct i2s_control_regs);
+ (*c)->allocated_rsrc =
+ request_mem_region((*c)->rsrc.start,
+ (*c)->rsrc.end - (*c)->rsrc.start + 1,
+ "i2s control");
+ if (!(*c)->allocated_rsrc) {
+ printk(KERN_WARNING "i2sbus: !!!!! WARNING !!!!!\n");
+ printk(KERN_WARNING "i2sbus: control: couldn't allocate resource\n");
+ printk(KERN_WARNING "i2sbus: going ahead anyway\n");
+ /* we should not ignore this error, but
+ * some other device (via-pmu?) claims our memory region...
+ * Until we somehow handle that we ignore the error */
+ /*goto err;*/
+ }
+ (*c)->controlregs = ioremap((*c)->rsrc.start,
+ (*c)->rsrc.end - (*c)->rsrc.start + 1);
+ if (!(*c)->controlregs)
+ goto release;
+
+ return 0;
+ release:
+ if ((*c)->allocated_rsrc)
+ release_resource((*c)->allocated_rsrc);
+ err:
+ kfree(*c);
+ *c = NULL;
+ return -ENODEV;
+}
+
+void i2sbus_control_destroy(struct i2sbus_control *c)
+{
+ iounmap(c->controlregs);
+ if (c->allocated_rsrc)
+ release_resource(c->allocated_rsrc);
+ 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;
+}
--- /dev/null
+++ b/sound/aoa/soundbus/i2sbus/i2sbus-control.h
@@ -0,0 +1,37 @@
+/*
+ * 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 */
--- /dev/null
+++ b/sound/aoa/soundbus/i2sbus/i2sbus-core.c
@@ -0,0 +1,370 @@
+/*
+ * 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 = pci_alloc_consistent(macio_get_pci_dev(i2sdev->macio),
+ r->size,
+ &r->bus_addr);
+
+ 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;
+
+ pci_free_consistent(macio_get_pci_dev(i2sdev->macio),
+ 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_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);
+
+ printk(KERN_INFO "i2sbus: interrupt, intr reg is 0x%x!\n", intreg);
+
+ /* acknowledge interrupt reasons */
+ out_le32(&dev->intfregs->intr_ctl, intreg);
+
+ spin_unlock(&dev->low_lock);
+
+ return IRQ_HANDLED;
+}
+
+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);
+ }
+ }
+
+ 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_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);
--- /dev/null
+++ b/sound/aoa/soundbus/i2sbus/i2sbus-interface.h
@@ -0,0 +1,187 @@
+/*
+ * 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 */
--- /dev/null
+++ b/sound/aoa/soundbus/i2sbus/i2sbus-pcm.c
@@ -0,0 +1,1008 @@
+/*
+ * 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 == 0) {
+ 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 == 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 == 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;
+ 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;
+
+ 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;
+ 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->substream) {
+ printk(KERN_INFO "i2sbus: got %s irq while not active!\n",
+ in ? "rx" : "tx");
+ 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 transfers, 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 + 1) % 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 is to correct the fact that the frame_count is too low at the beginning
+ * due to the dbdma's buffer. */
+ delta = 0;
+ }
+ }
+
+ pi->frame_count = fc - delta;
+ pi->current_period = 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");
+ }
+
+ 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 */
+ /* FIXME FIXME FIXME: If this fails, we BUG() when the alsa layer
+ * later tries to allocate memory. Apparently we should be setting
+ * some device pointer for that ...
+ */
+ 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);
+ }
+}
--- /dev/null
+++ b/sound/aoa/soundbus/i2sbus/i2sbus.h
@@ -0,0 +1,115 @@
+/*
+ * 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 "i2sbus-interface.h"
+#include "i2sbus-control.h"
+/* FIXME */
+#include "../soundbus.h"
+/* should go away, see comment in i2sbus_add_dev */
+#include <asm/prom.h>
+
+struct i2sbus_control {
+ volatile struct i2s_control_regs __iomem *controlregs;
+ struct resource rsrc;
+ struct resource *allocated_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 */
--- /dev/null
+++ b/sound/aoa/soundbus/soundbus.h
@@ -0,0 +1,202 @@
+/*
+ * 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 */
--- /dev/null
+++ b/sound/aoa/soundbus/sysfs.c
@@ -0,0 +1,67 @@
+#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
+compatible_show (struct device *dev, struct device_attribute *attr, char *buf)
+{
+ struct of_device *of;
+ char *compat;
+ int cplen;
+ int length = 0;
+
+ of = &to_soundbus_device (dev)->ofdev;
+ compat = (char *) get_property(of->node, "compatible", &cplen);
+ if (!compat) {
+ *buf = '\0';
+ return 0;
+ }
+ while (cplen > 0) {
+ int l;
+ length += sprintf (buf, "%s\n", compat);
+ buf += length;
+ l = strlen (compat) + 1;
+ compat += l;
+ cplen -= l;
+ }
+
+ return length;
+}
+
+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 (strlen(sdev->modalias)) {
+ length = snprintf (buf, 34, "%s\n", sdev->modalias);
+ } 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(compatible),
+ __ATTR_RO(modalias),
+ __ATTR_NULL
+};
--
^ permalink raw reply [flat|nested] 45+ messages in thread
* [RFC 4/7] snd-aoa: add codecs
2006-05-28 19:00 [RFC 0/7] snd-aoa: add snd-aoa Johannes Berg
@ 2006-05-28 19:00 ` Johannes Berg
2006-05-28 19:00 ` Johannes Berg
` (6 subsequent siblings)
7 siblings, 0 replies; 45+ messages in thread
From: Johannes Berg @ 2006-05-28 19:00 UTC (permalink / raw)
To: alsa-devel; +Cc: linuxppc-dev
[-- Attachment #1: 04-add-codecs.patch --]
[-- Type: text/plain, Size: 58673 bytes --]
This patch adds two of the most important codecs Apple uses in newer
machines, namely the Onyx and the tas3004.
--- /dev/null
+++ b/sound/aoa/codecs/Kconfig
@@ -0,0 +1,24 @@
+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.
--- /dev/null
+++ b/sound/aoa/codecs/Makefile
@@ -0,0 +1,2 @@
+obj-$(CONFIG_SND_AOA_ONYX) += snd-aoa-codec-onyx.o
+obj-$(CONFIG_SND_AOA_TAS) += snd-aoa-codec-tas.o
--- /dev/null
+++ b/sound/aoa/codecs/snd-aoa-codec-onyx.c
@@ -0,0 +1,1069 @@
+/*
+ * 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"
+/* FIXME */
+#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,
+};
+
+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+128;
+ uinfo->value.integer.max = -1+128;
+ 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+128;
+ ucontrol->value.integer.value[1] = r+128;
+
+ 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);
+
+ mutex_lock(&onyx->mutex);
+ onyx_write_register(onyx, ONYX_REG_DAC_ATTEN_LEFT, ucontrol->value.integer.value[0]-128);
+ onyx_write_register(onyx, ONYX_REG_DAC_ATTEN_RIGHT, ucontrol->value.integer.value[1]-128);
+ /* FIXME: we could be checking if anything changed */
+ 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,
+};
+
+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 = 3-3;
+ uinfo->value.integer.max = 28-3;
+ 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)-3;
+
+ 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 val;
+
+ mutex_lock(&onyx->mutex);
+ onyx_read_register(onyx, ONYX_REG_ADC_CONTROL, &val);
+ val &= ~ONYX_ADC_PGA_GAIN_MASK;
+ val |= (ucontrol->value.integer.value[0]+3) & ONYX_ADC_PGA_GAIN_MASK;
+ onyx_write_register(onyx, ONYX_REG_ADC_CONTROL, val);
+ mutex_unlock(&onyx->mutex);
+
+ return 1;
+}
+
+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 c;
+ int err = -EBUSY;
+
+ mutex_lock(&onyx->mutex);
+ if (onyx->analog_locked)
+ goto out_unlock;
+
+ onyx_read_register(onyx, ONYX_REG_DAC_CONTROL, &c);
+ 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);
+ if (err)
+ goto out_unlock;
+ /* FIXME: we could be checking if anything changed */
+ out_unlock:
+ mutex_unlock(&onyx->mutex);
+
+ return !err ? 1 : 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 c;
+ 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, &c);
+ c &= ~(mask);
+ if (!!ucontrol->value.integer.value[0] ^ polarity)
+ c |= mask;
+ err = onyx_write_register(onyx, address, c);
+ if (err)
+ goto out_unlock;
+ /* FIXME: we could be checking if anything changed */
+ out_unlock:
+ mutex_unlock(&onyx->mutex);
+
+ return !err ? 1 : 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[] = {
+ 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;
+
+ 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; \
+ aoa_snd_ctl_add(ctl); \
+ } \
+ } 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;
+}
+
+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;
+ strncpy(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;
+ }
+
+ strncpy(onyx->codec.name, "onyx", MAX_CODEC_NAME_LEN);
+ 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);
--- /dev/null
+++ b/sound/aoa/codecs/snd-aoa-codec-onyx.h
@@ -0,0 +1,76 @@
+/*
+ * 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 */
--- /dev/null
+++ b/sound/aoa/codecs/snd-aoa-codec-tas-gain-table.h
@@ -0,0 +1,209 @@
+/*
+ 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 */
+};
+
--- /dev/null
+++ b/sound/aoa/codecs/snd-aoa-codec-tas.c
@@ -0,0 +1,669 @@
+/*
+ * 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"
+/* FIXME */
+#include "../aoa.h"
+#include "../soundbus/soundbus.h"
+
+
+#define PFX "snd-aoa-codec-tas: "
+
+/* we use these to determine which codec we got */
+#define TAS_PRIMARY_MAGIC 0x74617331 /* tas1 */
+#define TAS_SECONDARY_MAGIC 0x32746173 /* 2tas */
+#define TAS_TERTIARY_MAGIC 0x74336173 /* t3as */
+
+struct tas {
+ u32 primary_magic;
+ struct aoa_codec codec;
+ /* see comment at top of file */
+ u32 secondary_magic;
+ struct aoa_codec secondary;
+ 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)
+{
+ u32 *tmp = (u32*)codec;
+ switch (*(tmp-1)) {
+ case TAS_PRIMARY_MAGIC:
+ return container_of(codec, struct tas, codec);
+ case TAS_SECONDARY_MAGIC:
+ return container_of(codec, struct tas, secondary);
+ default:
+ return NULL;
+ }
+}
+
+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);
+
+ 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);
+
+ 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;
+
+ 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(pcm2, "PCM2", 1);
+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);
+
+ tas->acr &= ~TAS_ACR_INPUT_B;
+ if (ucontrol->value.enumerated.item[0])
+ tas->acr |= TAS_ACR_INPUT_B;
+ 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;
+/*
+ char write[8];
+ union i2c_smbus_data read = { 0 };
+ int r1, r2;
+*/
+ 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;
+/* I need help here!
+ /* This is a bit tricky, but serves to detect if there really
+ * is a tas codec present.
+ * First, we set the volume register to 00,00,01 (on both channels).
+ * This is almost muted. Then, we read back the last 6 bytes we
+ * wrote to the chip, and check if they are the same.
+ *
+ write[0] = 7;
+ write[1] = TAS_REG_VOL;
+ write[2] = write[3] = 0;
+ write[4] = 1;
+ write[5] = write[6] = 0;
+ write[7] = 1;
+ r1 = tas_write_reg(tas, TAS_REG_VOL, 6, &write[1]);
+ /* Hmm, how am I supposed to do the i2c sequence that
+ * is mentioned on page 45 of the tas3004 datasheet?
+ * This doesn't cut it: *
+ read.block[0] = 7;
+ r2 = i2c_smbus_xfer(tas->i2c.adapter, tas->i2c.addr, tas->i2c.flags,
+ I2C_SMBUS_READ, TAS_REG_VOL,
+ I2C_SMBUS_BLOCK_DATA, &read);
+
+ printk(KERN_DEBUG "r1 = %d, r2 = %d, read=%x %x %x %x %x %x %x %x\n", r1, r2, read.block[0], read.block[1], read.block[2], read.block[3], read.block[4], read.block[5], read.block[6], read.block[7]);
+
+ if (r1 || r2 || memcmp(write, read.block, 8))
+ 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 primary = codec == &tas->codec;
+
+ 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;
+ }
+ aoa_snd_ctl_add(snd_ctl_new1(&volume_control, tas));
+ aoa_snd_ctl_add(snd_ctl_new1(&mute_control, tas));
+ if (primary) {
+ aoa_snd_ctl_add(snd_ctl_new1(&pcm1_control, tas));
+ } else {
+ aoa_snd_ctl_add(snd_ctl_new1(&pcm2_control, tas));
+ }
+ aoa_snd_ctl_add(snd_ctl_new1(&monitor_control, tas));
+ aoa_snd_ctl_add(snd_ctl_new1(&capture_source_control, tas));
+ return 0;
+}
+
+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->primary_magic = TAS_PRIMARY_MAGIC;
+ tas->secondary_magic = TAS_SECONDARY_MAGIC;
+ tas->i2c.driver = &tas_driver;
+ tas->i2c.adapter = adapter;
+ tas->i2c.addr = (addr >> 1) & 0x7f;
+ strncpy(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;
+ }
+
+ strncpy(tas->codec.name, "tas", MAX_CODEC_NAME_LEN);
+ 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)
+ return -ENODEV;
+ return tas_create(adapter, dev, *addr);
+ }
+ }
+
+ /* if that didn't work, try desperate mode for older
+ * machines that have stuff missing from the device tree */
+
+ if (!device_is_compatible(busnode, "keywest-i2c-bus"))
+ return -ENODEV;
+
+ printk(KERN_DEBUG PFX "found keywest-i2c-bus, checking if tas chip is on it\n");
+ /* can actually have address 0x34 as well... */
+ return tas_create(adapter, NULL, 0x35);
+}
+
+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);
--- /dev/null
+++ b/sound/aoa/codecs/snd-aoa-codec-tas.h
@@ -0,0 +1,47 @@
+/*
+ * 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 */
--
^ permalink raw reply [flat|nested] 45+ messages in thread
* [RFC 4/7] snd-aoa: add codecs
@ 2006-05-28 19:00 ` Johannes Berg
0 siblings, 0 replies; 45+ messages in thread
From: Johannes Berg @ 2006-05-28 19:00 UTC (permalink / raw)
To: alsa-devel; +Cc: linuxppc-dev
This patch adds two of the most important codecs Apple uses in newer
machines, namely the Onyx and the tas3004.
--- /dev/null
+++ b/sound/aoa/codecs/Kconfig
@@ -0,0 +1,24 @@
+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.
--- /dev/null
+++ b/sound/aoa/codecs/Makefile
@@ -0,0 +1,2 @@
+obj-$(CONFIG_SND_AOA_ONYX) += snd-aoa-codec-onyx.o
+obj-$(CONFIG_SND_AOA_TAS) += snd-aoa-codec-tas.o
--- /dev/null
+++ b/sound/aoa/codecs/snd-aoa-codec-onyx.c
@@ -0,0 +1,1069 @@
+/*
+ * 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"
+/* FIXME */
+#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,
+};
+
+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+128;
+ uinfo->value.integer.max = -1+128;
+ 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+128;
+ ucontrol->value.integer.value[1] = r+128;
+
+ 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);
+
+ mutex_lock(&onyx->mutex);
+ onyx_write_register(onyx, ONYX_REG_DAC_ATTEN_LEFT, ucontrol->value.integer.value[0]-128);
+ onyx_write_register(onyx, ONYX_REG_DAC_ATTEN_RIGHT, ucontrol->value.integer.value[1]-128);
+ /* FIXME: we could be checking if anything changed */
+ 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,
+};
+
+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 = 3-3;
+ uinfo->value.integer.max = 28-3;
+ 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)-3;
+
+ 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 val;
+
+ mutex_lock(&onyx->mutex);
+ onyx_read_register(onyx, ONYX_REG_ADC_CONTROL, &val);
+ val &= ~ONYX_ADC_PGA_GAIN_MASK;
+ val |= (ucontrol->value.integer.value[0]+3) & ONYX_ADC_PGA_GAIN_MASK;
+ onyx_write_register(onyx, ONYX_REG_ADC_CONTROL, val);
+ mutex_unlock(&onyx->mutex);
+
+ return 1;
+}
+
+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 c;
+ int err = -EBUSY;
+
+ mutex_lock(&onyx->mutex);
+ if (onyx->analog_locked)
+ goto out_unlock;
+
+ onyx_read_register(onyx, ONYX_REG_DAC_CONTROL, &c);
+ 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);
+ if (err)
+ goto out_unlock;
+ /* FIXME: we could be checking if anything changed */
+ out_unlock:
+ mutex_unlock(&onyx->mutex);
+
+ return !err ? 1 : 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 c;
+ 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, &c);
+ c &= ~(mask);
+ if (!!ucontrol->value.integer.value[0] ^ polarity)
+ c |= mask;
+ err = onyx_write_register(onyx, address, c);
+ if (err)
+ goto out_unlock;
+ /* FIXME: we could be checking if anything changed */
+ out_unlock:
+ mutex_unlock(&onyx->mutex);
+
+ return !err ? 1 : 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[] = {
+ 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;
+
+ 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; \
+ aoa_snd_ctl_add(ctl); \
+ } \
+ } 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;
+}
+
+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;
+ strncpy(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;
+ }
+
+ strncpy(onyx->codec.name, "onyx", MAX_CODEC_NAME_LEN);
+ 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);
--- /dev/null
+++ b/sound/aoa/codecs/snd-aoa-codec-onyx.h
@@ -0,0 +1,76 @@
+/*
+ * 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 */
--- /dev/null
+++ b/sound/aoa/codecs/snd-aoa-codec-tas-gain-table.h
@@ -0,0 +1,209 @@
+/*
+ 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 */
+};
+
--- /dev/null
+++ b/sound/aoa/codecs/snd-aoa-codec-tas.c
@@ -0,0 +1,669 @@
+/*
+ * 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"
+/* FIXME */
+#include "../aoa.h"
+#include "../soundbus/soundbus.h"
+
+
+#define PFX "snd-aoa-codec-tas: "
+
+/* we use these to determine which codec we got */
+#define TAS_PRIMARY_MAGIC 0x74617331 /* tas1 */
+#define TAS_SECONDARY_MAGIC 0x32746173 /* 2tas */
+#define TAS_TERTIARY_MAGIC 0x74336173 /* t3as */
+
+struct tas {
+ u32 primary_magic;
+ struct aoa_codec codec;
+ /* see comment at top of file */
+ u32 secondary_magic;
+ struct aoa_codec secondary;
+ 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)
+{
+ u32 *tmp = (u32*)codec;
+ switch (*(tmp-1)) {
+ case TAS_PRIMARY_MAGIC:
+ return container_of(codec, struct tas, codec);
+ case TAS_SECONDARY_MAGIC:
+ return container_of(codec, struct tas, secondary);
+ default:
+ return NULL;
+ }
+}
+
+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);
+
+ 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);
+
+ 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;
+
+ 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(pcm2, "PCM2", 1);
+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);
+
+ tas->acr &= ~TAS_ACR_INPUT_B;
+ if (ucontrol->value.enumerated.item[0])
+ tas->acr |= TAS_ACR_INPUT_B;
+ 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;
+/*
+ char write[8];
+ union i2c_smbus_data read = { 0 };
+ int r1, r2;
+*/
+ 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;
+/* I need help here!
+ /* This is a bit tricky, but serves to detect if there really
+ * is a tas codec present.
+ * First, we set the volume register to 00,00,01 (on both channels).
+ * This is almost muted. Then, we read back the last 6 bytes we
+ * wrote to the chip, and check if they are the same.
+ *
+ write[0] = 7;
+ write[1] = TAS_REG_VOL;
+ write[2] = write[3] = 0;
+ write[4] = 1;
+ write[5] = write[6] = 0;
+ write[7] = 1;
+ r1 = tas_write_reg(tas, TAS_REG_VOL, 6, &write[1]);
+ /* Hmm, how am I supposed to do the i2c sequence that
+ * is mentioned on page 45 of the tas3004 datasheet?
+ * This doesn't cut it: *
+ read.block[0] = 7;
+ r2 = i2c_smbus_xfer(tas->i2c.adapter, tas->i2c.addr, tas->i2c.flags,
+ I2C_SMBUS_READ, TAS_REG_VOL,
+ I2C_SMBUS_BLOCK_DATA, &read);
+
+ printk(KERN_DEBUG "r1 = %d, r2 = %d, read=%x %x %x %x %x %x %x %x\n", r1, r2, read.block[0], read.block[1], read.block[2], read.block[3], read.block[4], read.block[5], read.block[6], read.block[7]);
+
+ if (r1 || r2 || memcmp(write, read.block, 8))
+ 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 primary = codec == &tas->codec;
+
+ 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;
+ }
+ aoa_snd_ctl_add(snd_ctl_new1(&volume_control, tas));
+ aoa_snd_ctl_add(snd_ctl_new1(&mute_control, tas));
+ if (primary) {
+ aoa_snd_ctl_add(snd_ctl_new1(&pcm1_control, tas));
+ } else {
+ aoa_snd_ctl_add(snd_ctl_new1(&pcm2_control, tas));
+ }
+ aoa_snd_ctl_add(snd_ctl_new1(&monitor_control, tas));
+ aoa_snd_ctl_add(snd_ctl_new1(&capture_source_control, tas));
+ return 0;
+}
+
+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->primary_magic = TAS_PRIMARY_MAGIC;
+ tas->secondary_magic = TAS_SECONDARY_MAGIC;
+ tas->i2c.driver = &tas_driver;
+ tas->i2c.adapter = adapter;
+ tas->i2c.addr = (addr >> 1) & 0x7f;
+ strncpy(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;
+ }
+
+ strncpy(tas->codec.name, "tas", MAX_CODEC_NAME_LEN);
+ 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)
+ return -ENODEV;
+ return tas_create(adapter, dev, *addr);
+ }
+ }
+
+ /* if that didn't work, try desperate mode for older
+ * machines that have stuff missing from the device tree */
+
+ if (!device_is_compatible(busnode, "keywest-i2c-bus"))
+ return -ENODEV;
+
+ printk(KERN_DEBUG PFX "found keywest-i2c-bus, checking if tas chip is on it\n");
+ /* can actually have address 0x34 as well... */
+ return tas_create(adapter, NULL, 0x35);
+}
+
+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);
--- /dev/null
+++ b/sound/aoa/codecs/snd-aoa-codec-tas.h
@@ -0,0 +1,47 @@
+/*
+ * 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 */
--
^ permalink raw reply [flat|nested] 45+ messages in thread
* [RFC 5/7] snd-aoa: add layout-id fabric
2006-05-28 19:00 [RFC 0/7] snd-aoa: add snd-aoa Johannes Berg
@ 2006-05-28 19:00 ` Johannes Berg
2006-05-28 19:00 ` Johannes Berg
` (6 subsequent siblings)
7 siblings, 0 replies; 45+ messages in thread
From: Johannes Berg @ 2006-05-28 19:00 UTC (permalink / raw)
To: alsa-devel; +Cc: linuxppc-dev
[-- Attachment #1: 05-add-fabrics.patch --]
[-- Type: text/plain, Size: 16126 bytes --]
The 'fabric' is the thing that pulls all of snd-aoa together, this patch
adds the 'layout' or 'layout-id' fabric that keys off the 'layout-id'
property in the device-tree to pull in the proper modules. It itself loads
if an i2sbus is present with a layout-id (exported in modalias) that it can
use.
--- /dev/null
+++ b/sound/aoa/fabrics/Kconfig
@@ -0,0 +1,12 @@
+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.
--- /dev/null
+++ b/sound/aoa/fabrics/Makefile
@@ -0,0 +1 @@
+obj-$(CONFIG_SND_AOA_FABRIC_LAYOUT) += snd-aoa-fabric-layout.o
--- /dev/null
+++ b/sound/aoa/fabrics/snd-aoa-fabric-layout.c
@@ -0,0 +1,605 @@
+/*
+ * 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-82");
+MODULE_ALIAS("sound-layout-45");
+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-86");
+MODULE_ALIAS("sound-layout-84");
+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 */
+};
+
+/* 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 */
+};
+
+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 = NULL /* TBD */,
+ },
+ },
+ /* 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 = NULL, /* TBD */
+ },
+ },
+ /* 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 = NULL /* TBD */,
+ },
+ },
+ /* PowerMac8,1 */
+ { .layout_id = 45,
+ .codecs[0] = {
+ .name = "onyx",
+ .connections = onyx_connections_noheadphones,
+ },
+ .codecs[1] = {
+ .name = "topaz",
+ .connections = NULL /* TBD */,
+ },
+ },
+ /* 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 = NULL /* TBD */,
+ },
+ .busname = "digital in", .pcmid = 1 },
+ /* Early 2005 PowerBook */
+ { .layout_id = 70,
+ .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 = NULL /* TBD */,
+ },
+ },
+ /* PowerBook6,7 */
+ { .layout_id = 92,
+ .codecs[0] = {
+ .name = "tas",
+ .connections = tas_connections_nolineout,
+ },
+ },
+ {}
+};
+
+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 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;
+};
+
+static LIST_HEAD(layouts_list);
+static int layouts_list_items;
+
+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 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 (codec->node) {
+ snprintf(propname, sizeof(propname), "platform-%s-codec-ref", codec->name);
+ ref = (u32*)get_property(ldev->sound, propname, NULL);
+ /* if the codec has a node, we require a reference */
+ 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. eek!\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_attached_codec(struct aoa_codec *codec)
+{
+ struct codec_connection *cc;
+ struct snd_kcontrol *ctl;
+
+ /* need to add this codec to our codec array! */
+
+ cc = codec->fabric_data;
+
+ while (cc->connected) {
+ if (cc->connected & CC_SPEAKERS)
+ aoa_snd_ctl_add(snd_ctl_new1(&speakers_ctl, codec->gpio));
+ if (cc->connected & CC_HEADPHONE)
+ aoa_snd_ctl_add(snd_ctl_new1(&headphone_ctl, codec->gpio));
+ if (cc->connected & CC_LINEOUT) {
+ 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));
+ aoa_snd_ctl_add(ctl);
+ }
+ cc++;
+ }
+}
+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;
+
+ /* 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;
+
+ ldev->sdev = sdev;
+ ldev->sound = sound;
+ ldev->layout = layout;
+ ldev->gpio.node = sound->parent;
+ ldev->gpio.methods = pmf_gpio_methods;
+ 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";
+ }
+
+ 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;
+ }
+
+ ldev->gpio.methods->init(&ldev->gpio);
+ use_layout(layout);
+ return 0;
+ outlistdel:
+ /* 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);
+ 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->exit(&ldev->gpio);
+ 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);
--
^ permalink raw reply [flat|nested] 45+ messages in thread
* [RFC 5/7] snd-aoa: add layout-id fabric
@ 2006-05-28 19:00 ` Johannes Berg
0 siblings, 0 replies; 45+ messages in thread
From: Johannes Berg @ 2006-05-28 19:00 UTC (permalink / raw)
To: alsa-devel; +Cc: linuxppc-dev
The 'fabric' is the thing that pulls all of snd-aoa together, this patch
adds the 'layout' or 'layout-id' fabric that keys off the 'layout-id'
property in the device-tree to pull in the proper modules. It itself loads
if an i2sbus is present with a layout-id (exported in modalias) that it can
use.
--- /dev/null
+++ b/sound/aoa/fabrics/Kconfig
@@ -0,0 +1,12 @@
+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.
--- /dev/null
+++ b/sound/aoa/fabrics/Makefile
@@ -0,0 +1 @@
+obj-$(CONFIG_SND_AOA_FABRIC_LAYOUT) += snd-aoa-fabric-layout.o
--- /dev/null
+++ b/sound/aoa/fabrics/snd-aoa-fabric-layout.c
@@ -0,0 +1,605 @@
+/*
+ * 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-82");
+MODULE_ALIAS("sound-layout-45");
+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-86");
+MODULE_ALIAS("sound-layout-84");
+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 */
+};
+
+/* 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 */
+};
+
+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 = NULL /* TBD */,
+ },
+ },
+ /* 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 = NULL, /* TBD */
+ },
+ },
+ /* 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 = NULL /* TBD */,
+ },
+ },
+ /* PowerMac8,1 */
+ { .layout_id = 45,
+ .codecs[0] = {
+ .name = "onyx",
+ .connections = onyx_connections_noheadphones,
+ },
+ .codecs[1] = {
+ .name = "topaz",
+ .connections = NULL /* TBD */,
+ },
+ },
+ /* 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 = NULL /* TBD */,
+ },
+ .busname = "digital in", .pcmid = 1 },
+ /* Early 2005 PowerBook */
+ { .layout_id = 70,
+ .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 = NULL /* TBD */,
+ },
+ },
+ /* PowerBook6,7 */
+ { .layout_id = 92,
+ .codecs[0] = {
+ .name = "tas",
+ .connections = tas_connections_nolineout,
+ },
+ },
+ {}
+};
+
+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 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;
+};
+
+static LIST_HEAD(layouts_list);
+static int layouts_list_items;
+
+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 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 (codec->node) {
+ snprintf(propname, sizeof(propname), "platform-%s-codec-ref", codec->name);
+ ref = (u32*)get_property(ldev->sound, propname, NULL);
+ /* if the codec has a node, we require a reference */
+ 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. eek!\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_attached_codec(struct aoa_codec *codec)
+{
+ struct codec_connection *cc;
+ struct snd_kcontrol *ctl;
+
+ /* need to add this codec to our codec array! */
+
+ cc = codec->fabric_data;
+
+ while (cc->connected) {
+ if (cc->connected & CC_SPEAKERS)
+ aoa_snd_ctl_add(snd_ctl_new1(&speakers_ctl, codec->gpio));
+ if (cc->connected & CC_HEADPHONE)
+ aoa_snd_ctl_add(snd_ctl_new1(&headphone_ctl, codec->gpio));
+ if (cc->connected & CC_LINEOUT) {
+ 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));
+ aoa_snd_ctl_add(ctl);
+ }
+ cc++;
+ }
+}
+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;
+
+ /* 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;
+
+ ldev->sdev = sdev;
+ ldev->sound = sound;
+ ldev->layout = layout;
+ ldev->gpio.node = sound->parent;
+ ldev->gpio.methods = pmf_gpio_methods;
+ 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";
+ }
+
+ 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;
+ }
+
+ ldev->gpio.methods->init(&ldev->gpio);
+ use_layout(layout);
+ return 0;
+ outlistdel:
+ /* 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);
+ 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->exit(&ldev->gpio);
+ 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);
--
^ permalink raw reply [flat|nested] 45+ messages in thread
* [RFC 6/7] snd-aoa: add Kconfig and Makefile
2006-05-28 19:00 [RFC 0/7] snd-aoa: add snd-aoa Johannes Berg
@ 2006-05-28 19:00 ` Johannes Berg
2006-05-28 19:00 ` Johannes Berg
` (6 subsequent siblings)
7 siblings, 0 replies; 45+ messages in thread
From: Johannes Berg @ 2006-05-28 19:00 UTC (permalink / raw)
To: alsa-devel; +Cc: linuxppc-dev
[-- Attachment #1: 06-add-build-stuff.patch --]
[-- Type: text/plain, Size: 717 bytes --]
This patch adds the simple Kconfig and Makefile files for aoa.
--- /dev/null
+++ b/sound/aoa/Kconfig
@@ -0,0 +1,17 @@
+menu "Apple Onboard Audio driver"
+ depends on SND != n && PPC
+
+config SND_AOA
+ tristate "Apple Onboard Audio driver"
+ depends on SOUND && SND_PCM && EXPERIMENTAL
+ ---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
--- /dev/null
+++ b/sound/aoa/Makefile
@@ -0,0 +1,4 @@
+obj-$(CONFIG_SND_AOA) += core/
+obj-$(CONFIG_SND_AOA) += codecs/
+obj-$(CONFIG_SND_AOA) += fabrics/
+obj-$(CONFIG_SND_AOA_SOUNDBUS) += soundbus/
--
^ permalink raw reply [flat|nested] 45+ messages in thread
* [RFC 6/7] snd-aoa: add Kconfig and Makefile
@ 2006-05-28 19:00 ` Johannes Berg
0 siblings, 0 replies; 45+ messages in thread
From: Johannes Berg @ 2006-05-28 19:00 UTC (permalink / raw)
To: alsa-devel; +Cc: linuxppc-dev
This patch adds the simple Kconfig and Makefile files for aoa.
--- /dev/null
+++ b/sound/aoa/Kconfig
@@ -0,0 +1,17 @@
+menu "Apple Onboard Audio driver"
+ depends on SND != n && PPC
+
+config SND_AOA
+ tristate "Apple Onboard Audio driver"
+ depends on SOUND && SND_PCM && EXPERIMENTAL
+ ---help---
+ This option enables the new driver for the various
+ Apple Onboard Audio components.
+
+source "sound/aoa/fabrics/Kconfig"
+
+source "sound/aoa/codecs/Kconfig"
+
+source "sound/aoa/soundbus/Kconfig"
+
+endmenu
--- /dev/null
+++ b/sound/aoa/Makefile
@@ -0,0 +1,4 @@
+obj-$(CONFIG_SND_AOA) += core/
+obj-$(CONFIG_SND_AOA) += codecs/
+obj-$(CONFIG_SND_AOA) += fabrics/
+obj-$(CONFIG_SND_AOA_SOUNDBUS) += soundbus/
--
^ permalink raw reply [flat|nested] 45+ messages in thread
* [RFC 7/7] snd-aoa: wire up aoa in sound/
2006-05-28 19:00 [RFC 0/7] snd-aoa: add snd-aoa Johannes Berg
@ 2006-05-28 19:00 ` Johannes Berg
2006-05-28 19:00 ` Johannes Berg
` (6 subsequent siblings)
7 siblings, 0 replies; 45+ messages in thread
From: Johannes Berg @ 2006-05-28 19:00 UTC (permalink / raw)
To: alsa-devel; +Cc: linuxppc-dev
[-- Attachment #1: 07-wire-up.patch --]
[-- Type: text/plain, Size: 733 bytes --]
This patch adds the necessary Kconfig and Makefile hooks to make aoa
buildable inside the kernel tree.
--- a/sound/Kconfig
+++ b/sound/Kconfig
@@ -58,6 +58,8 @@ source "sound/pci/Kconfig"
source "sound/ppc/Kconfig"
+source "sound/aoa/Kconfig"
+
source "sound/arm/Kconfig"
source "sound/mips/Kconfig"
--- a/sound/Makefile
+++ b/sound/Makefile
@@ -4,7 +4,7 @@ #
obj-$(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 ($(CONFIG_SND),y)
obj-y += last.o
--
^ permalink raw reply [flat|nested] 45+ messages in thread
* [RFC 7/7] snd-aoa: wire up aoa in sound/
@ 2006-05-28 19:00 ` Johannes Berg
0 siblings, 0 replies; 45+ messages in thread
From: Johannes Berg @ 2006-05-28 19:00 UTC (permalink / raw)
To: alsa-devel; +Cc: linuxppc-dev
This patch adds the necessary Kconfig and Makefile hooks to make aoa
buildable inside the kernel tree.
--- a/sound/Kconfig
+++ b/sound/Kconfig
@@ -58,6 +58,8 @@ source "sound/pci/Kconfig"
source "sound/ppc/Kconfig"
+source "sound/aoa/Kconfig"
+
source "sound/arm/Kconfig"
source "sound/mips/Kconfig"
--- a/sound/Makefile
+++ b/sound/Makefile
@@ -4,7 +4,7 @@ #
obj-$(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
--
^ permalink raw reply [flat|nested] 45+ messages in thread
* Re: [RFC 0/7] snd-aoa: add snd-aoa
2006-05-28 19:00 [RFC 0/7] snd-aoa: add snd-aoa Johannes Berg
` (6 preceding siblings ...)
2006-05-28 19:00 ` Johannes Berg
@ 2006-05-28 19:10 ` Johannes Berg
2006-05-29 10:12 ` Takashi Iwai
7 siblings, 1 reply; 45+ messages in thread
From: Johannes Berg @ 2006-05-28 19:10 UTC (permalink / raw)
To: alsa-devel
[-- Attachment #1.1: Type: text/plain, Size: 103 bytes --]
Hmm. The most interesting, #3, was held for being too large. Can someone
free it please?
johannes
[-- Attachment #1.2: This is a digitally signed message part --]
[-- Type: application/pgp-signature, Size: 793 bytes --]
[-- Attachment #2: Type: text/plain, Size: 0 bytes --]
[-- Attachment #3: Type: text/plain, Size: 161 bytes --]
_______________________________________________
Alsa-devel mailing list
Alsa-devel@lists.sourceforge.net
https://lists.sourceforge.net/lists/listinfo/alsa-devel
^ permalink raw reply [flat|nested] 45+ messages in thread
* Re: [RFC 0/7] snd-aoa: add snd-aoa
2006-05-28 19:10 ` [RFC 0/7] snd-aoa: add snd-aoa Johannes Berg
@ 2006-05-29 10:12 ` Takashi Iwai
2006-05-30 12:37 ` Johannes Berg
0 siblings, 1 reply; 45+ messages in thread
From: Takashi Iwai @ 2006-05-29 10:12 UTC (permalink / raw)
To: Johannes Berg; +Cc: alsa-devel
At Sun, 28 May 2006 21:10:08 +0200,
Johannes Berg wrote:
>
> Hmm. The most interesting, #3, was held for being too large. Can someone
> free it please?
Not sure which filtered the post (sourceforge server?)...
Could you repost with a gzipped attachment?
thanks,
Takashi
^ permalink raw reply [flat|nested] 45+ messages in thread
* Re: [RFC 2/7] snd-aoa: add aoa core
2006-05-28 19:00 ` Johannes Berg
@ 2006-05-29 10:57 ` Takashi Iwai
-1 siblings, 0 replies; 45+ messages in thread
From: Takashi Iwai @ 2006-05-29 10:57 UTC (permalink / raw)
To: Johannes Berg; +Cc: linuxppc-dev, alsa-devel
At Sun, 28 May 2006 21:00:28 +0200,
Johannes Berg wrote:
> --- /dev/null
> +++ b/sound/aoa/core/snd-aoa-alsa.c
> @@ -0,0 +1,91 @@
> +/*
> + * 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 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(-1, name, mod, sizeof(struct aoa_card));
Usually the first argument here is the index module option so that
user may specify the card order (e.g. secondary card) independent from
the order of module loading.
> + if (!alsa_card)
> + return -ENOMEM;
> + aoa_card = alsa_card->private_data;
> + aoa_card->alsa_card = alsa_card;
> + strncpy(alsa_card->driver, "AppleOnbdAudio", sizeof(alsa_card->driver));
> + strncpy(alsa_card->shortname, name, sizeof(alsa_card->shortname));
> + strncpy(alsa_card->longname, name, sizeof(alsa_card->longname));
> + strncpy(alsa_card->mixername, name, sizeof(alsa_card->mixername));
strlcpy is safer. Also, longname usually contains more verbose info.
> --- /dev/null
> +++ b/sound/aoa/core/snd-aoa-core.c
(snip)
> + err = aoa_alsa_init(new_fabric->name, new_fabric->owner);
> + if (err) {
> + return err;
> + }
Remove braces.
Takashi
^ permalink raw reply [flat|nested] 45+ messages in thread
* Re: [Alsa-devel] [RFC 2/7] snd-aoa: add aoa core
@ 2006-05-29 10:57 ` Takashi Iwai
0 siblings, 0 replies; 45+ messages in thread
From: Takashi Iwai @ 2006-05-29 10:57 UTC (permalink / raw)
To: Johannes Berg; +Cc: linuxppc-dev, alsa-devel
At Sun, 28 May 2006 21:00:28 +0200,
Johannes Berg wrote:
> --- /dev/null
> +++ b/sound/aoa/core/snd-aoa-alsa.c
> @@ -0,0 +1,91 @@
> +/*
> + * 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 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(-1, name, mod, sizeof(struct aoa_card));
Usually the first argument here is the index module option so that
user may specify the card order (e.g. secondary card) independent from
the order of module loading.
> + if (!alsa_card)
> + return -ENOMEM;
> + aoa_card = alsa_card->private_data;
> + aoa_card->alsa_card = alsa_card;
> + strncpy(alsa_card->driver, "AppleOnbdAudio", sizeof(alsa_card->driver));
> + strncpy(alsa_card->shortname, name, sizeof(alsa_card->shortname));
> + strncpy(alsa_card->longname, name, sizeof(alsa_card->longname));
> + strncpy(alsa_card->mixername, name, sizeof(alsa_card->mixername));
strlcpy is safer. Also, longname usually contains more verbose info.
> --- /dev/null
> +++ b/sound/aoa/core/snd-aoa-core.c
(snip)
> + err = aoa_alsa_init(new_fabric->name, new_fabric->owner);
> + if (err) {
> + return err;
> + }
Remove braces.
Takashi
^ permalink raw reply [flat|nested] 45+ messages in thread
* Re: [RFC 4/7] snd-aoa: add codecs
2006-05-28 19:00 ` Johannes Berg
@ 2006-05-29 12:08 ` Takashi Iwai
-1 siblings, 0 replies; 45+ messages in thread
From: Takashi Iwai @ 2006-05-29 12:08 UTC (permalink / raw)
To: Johannes Berg; +Cc: linuxppc-dev, alsa-devel
At Sun, 28 May 2006 21:00:30 +0200,
Johannes Berg wrote:
> --- /dev/null
> +++ b/sound/aoa/codecs/snd-aoa-codec-onyx.c
> +/* 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;
Isn't it "reg - FIRSTREGISTER" ?
> + 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,
> +};
> +
> +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+128;
> + uinfo->value.integer.max = -1+128;
I'd define a constant for 128.
> + 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+128;
> + ucontrol->value.integer.value[1] = r+128;
> +
> + 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);
> +
> + mutex_lock(&onyx->mutex);
> + onyx_write_register(onyx, ONYX_REG_DAC_ATTEN_LEFT, ucontrol->value.integer.value[0]-128);
> + onyx_write_register(onyx, ONYX_REG_DAC_ATTEN_RIGHT, ucontrol->value.integer.value[1]-128);
Fold lines to fit with 80 columns (heh, blaming other one's code is
easy :)
> + /* FIXME: we could be checking if anything changed */
> + mutex_unlock(&onyx->mutex);
> +
> + return 1;
The put callback is supposed to return 0 if the values are unchanged
(although most apps ignore the return value).
(snip)
> +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[] = {
Should have ARRAY_SIZE(register_map) since this size must be identical
with 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,
Too long lines ;)
(snip)
> + 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; \
> + aoa_snd_ctl_add(ctl); \
No error check?
> + /* 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;
> + }
> +
> + strncpy(onyx->codec.name, "onyx", MAX_CODEC_NAME_LEN);
Use strlcpy, or MAX_CODEC_NAME_LEN-1. Similar lines are found in
tas driver too.
> --- /dev/null
> +++ b/sound/aoa/codecs/snd-aoa-codec-tas.c
> @@ -0,0 +1,669 @@
(snip)
> +struct tas {
> + u32 primary_magic;
> + struct aoa_codec codec;
> + /* see comment at top of file */
> + u32 secondary_magic;
> + struct aoa_codec secondary;
> + 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)
> +{
> + u32 *tmp = (u32*)codec;
> + switch (*(tmp-1)) {
> + case TAS_PRIMARY_MAGIC:
> + return container_of(codec, struct tas, codec);
> + case TAS_SECONDARY_MAGIC:
> + return container_of(codec, struct tas, secondary);
> + default:
> + return NULL;
> + }
> +}
Looks a bit too hacky. IMO, it's better to define a struct
struct tas_codec {
struct aoa_codec codec;
struct tas *tas;
}
to make the above simpler like:
static struct tas *codec_to_tas(struct aoa_codec *codec)
{
return ((struct tas_codec *)codec)->tas;
}
The tas struct becomes:
struct as {
struct tas_codec primary;
struct tas_codec secondary;
...
}
and is initialized like:
tas->primary.tas = tas;
tas->secondary.tas = tas;
> +static int tas_snd_capture_source_info(struct snd_kcontrol *kcontrol,
> + struct snd_ctl_elem_info *uinfo)
> +{
> + static char* texts[] = { "Line-In", "Microphone" };
char *texts[]
> +static int tas_reset_init(struct tas *tas)
> +{
> + u8 tmp;
> +/*
> + char write[8];
> + union i2c_smbus_data read = { 0 };
> + int r1, r2;
> +*/
> + 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;
> +/* I need help here!
Use ifdef. Nested comments are bad.
> + /* This is a bit tricky, but serves to detect if there really
> + * is a tas codec present.
> + * First, we set the volume register to 00,00,01 (on both channels).
> + * This is almost muted. Then, we read back the last 6 bytes we
> + * wrote to the chip, and check if they are the same.
> + *
> + write[0] = 7;
> + write[1] = TAS_REG_VOL;
> + write[2] = write[3] = 0;
> + write[4] = 1;
> + write[5] = write[6] = 0;
> + write[7] = 1;
> + r1 = tas_write_reg(tas, TAS_REG_VOL, 6, &write[1]);
> + /* Hmm, how am I supposed to do the i2c sequence that
> + * is mentioned on page 45 of the tas3004 datasheet?
> + * This doesn't cut it: *
> + read.block[0] = 7;
> + r2 = i2c_smbus_xfer(tas->i2c.adapter, tas->i2c.addr, tas->i2c.flags,
> + I2C_SMBUS_READ, TAS_REG_VOL,
> + I2C_SMBUS_BLOCK_DATA, &read);
> +
> + printk(KERN_DEBUG "r1 = %d, r2 = %d, read=%x %x %x %x %x %x %x %x\n", r1, r2, read.block[0], read.block[1], read.block[2], read.block[3], read.block[4], read.block[5], read.block[6], read.block[7]);
> +
> + if (r1 || r2 || memcmp(write, read.block, 8))
> + return -ENODEV;
> +*/
> +
> + return 0;
> +}
> +static int tas_init_codec(struct aoa_codec *codec)
> +{
(snip)
> + aoa_snd_ctl_add(snd_ctl_new1(&volume_control, tas));
Error checks please.
Takashi
^ permalink raw reply [flat|nested] 45+ messages in thread
* Re: [Alsa-devel] [RFC 4/7] snd-aoa: add codecs
@ 2006-05-29 12:08 ` Takashi Iwai
0 siblings, 0 replies; 45+ messages in thread
From: Takashi Iwai @ 2006-05-29 12:08 UTC (permalink / raw)
To: Johannes Berg; +Cc: linuxppc-dev, alsa-devel
At Sun, 28 May 2006 21:00:30 +0200,
Johannes Berg wrote:
> --- /dev/null
> +++ b/sound/aoa/codecs/snd-aoa-codec-onyx.c
> +/* 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;
Isn't it "reg - FIRSTREGISTER" ?
> + 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,
> +};
> +
> +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+128;
> + uinfo->value.integer.max = -1+128;
I'd define a constant for 128.
> + 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+128;
> + ucontrol->value.integer.value[1] = r+128;
> +
> + 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);
> +
> + mutex_lock(&onyx->mutex);
> + onyx_write_register(onyx, ONYX_REG_DAC_ATTEN_LEFT, ucontrol->value.integer.value[0]-128);
> + onyx_write_register(onyx, ONYX_REG_DAC_ATTEN_RIGHT, ucontrol->value.integer.value[1]-128);
Fold lines to fit with 80 columns (heh, blaming other one's code is
easy :)
> + /* FIXME: we could be checking if anything changed */
> + mutex_unlock(&onyx->mutex);
> +
> + return 1;
The put callback is supposed to return 0 if the values are unchanged
(although most apps ignore the return value).
(snip)
> +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[] = {
Should have ARRAY_SIZE(register_map) since this size must be identical
with 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,
Too long lines ;)
(snip)
> + 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; \
> + aoa_snd_ctl_add(ctl); \
No error check?
> + /* 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;
> + }
> +
> + strncpy(onyx->codec.name, "onyx", MAX_CODEC_NAME_LEN);
Use strlcpy, or MAX_CODEC_NAME_LEN-1. Similar lines are found in
tas driver too.
> --- /dev/null
> +++ b/sound/aoa/codecs/snd-aoa-codec-tas.c
> @@ -0,0 +1,669 @@
(snip)
> +struct tas {
> + u32 primary_magic;
> + struct aoa_codec codec;
> + /* see comment at top of file */
> + u32 secondary_magic;
> + struct aoa_codec secondary;
> + 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)
> +{
> + u32 *tmp = (u32*)codec;
> + switch (*(tmp-1)) {
> + case TAS_PRIMARY_MAGIC:
> + return container_of(codec, struct tas, codec);
> + case TAS_SECONDARY_MAGIC:
> + return container_of(codec, struct tas, secondary);
> + default:
> + return NULL;
> + }
> +}
Looks a bit too hacky. IMO, it's better to define a struct
struct tas_codec {
struct aoa_codec codec;
struct tas *tas;
}
to make the above simpler like:
static struct tas *codec_to_tas(struct aoa_codec *codec)
{
return ((struct tas_codec *)codec)->tas;
}
The tas struct becomes:
struct as {
struct tas_codec primary;
struct tas_codec secondary;
...
}
and is initialized like:
tas->primary.tas = tas;
tas->secondary.tas = tas;
> +static int tas_snd_capture_source_info(struct snd_kcontrol *kcontrol,
> + struct snd_ctl_elem_info *uinfo)
> +{
> + static char* texts[] = { "Line-In", "Microphone" };
char *texts[]
> +static int tas_reset_init(struct tas *tas)
> +{
> + u8 tmp;
> +/*
> + char write[8];
> + union i2c_smbus_data read = { 0 };
> + int r1, r2;
> +*/
> + 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;
> +/* I need help here!
Use ifdef. Nested comments are bad.
> + /* This is a bit tricky, but serves to detect if there really
> + * is a tas codec present.
> + * First, we set the volume register to 00,00,01 (on both channels).
> + * This is almost muted. Then, we read back the last 6 bytes we
> + * wrote to the chip, and check if they are the same.
> + *
> + write[0] = 7;
> + write[1] = TAS_REG_VOL;
> + write[2] = write[3] = 0;
> + write[4] = 1;
> + write[5] = write[6] = 0;
> + write[7] = 1;
> + r1 = tas_write_reg(tas, TAS_REG_VOL, 6, &write[1]);
> + /* Hmm, how am I supposed to do the i2c sequence that
> + * is mentioned on page 45 of the tas3004 datasheet?
> + * This doesn't cut it: *
> + read.block[0] = 7;
> + r2 = i2c_smbus_xfer(tas->i2c.adapter, tas->i2c.addr, tas->i2c.flags,
> + I2C_SMBUS_READ, TAS_REG_VOL,
> + I2C_SMBUS_BLOCK_DATA, &read);
> +
> + printk(KERN_DEBUG "r1 = %d, r2 = %d, read=%x %x %x %x %x %x %x %x\n", r1, r2, read.block[0], read.block[1], read.block[2], read.block[3], read.block[4], read.block[5], read.block[6], read.block[7]);
> +
> + if (r1 || r2 || memcmp(write, read.block, 8))
> + return -ENODEV;
> +*/
> +
> + return 0;
> +}
> +static int tas_init_codec(struct aoa_codec *codec)
> +{
(snip)
> + aoa_snd_ctl_add(snd_ctl_new1(&volume_control, tas));
Error checks please.
Takashi
^ permalink raw reply [flat|nested] 45+ messages in thread
* Re: [RFC 5/7] snd-aoa: add layout-id fabric
2006-05-28 19:00 ` Johannes Berg
@ 2006-05-29 12:14 ` Takashi Iwai
-1 siblings, 0 replies; 45+ messages in thread
From: Takashi Iwai @ 2006-05-29 12:14 UTC (permalink / raw)
To: Johannes Berg; +Cc: linuxppc-dev, alsa-devel
At Sun, 28 May 2006 21:00:31 +0200,
Johannes Berg wrote:
>
> The 'fabric' is the thing that pulls all of snd-aoa together, this patch
> adds the 'layout' or 'layout-id' fabric that keys off the 'layout-id'
> property in the device-tree to pull in the proper modules. It itself loads
> if an i2sbus is present with a layout-id (exported in modalias) that it can
> use.
(snip)
> +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);
Just from curiosity: Does this work properly?
I remember that there was a deadlock when you call request_module()
during module probe.
Takashi
^ permalink raw reply [flat|nested] 45+ messages in thread
* Re: [Alsa-devel] [RFC 5/7] snd-aoa: add layout-id fabric
@ 2006-05-29 12:14 ` Takashi Iwai
0 siblings, 0 replies; 45+ messages in thread
From: Takashi Iwai @ 2006-05-29 12:14 UTC (permalink / raw)
To: Johannes Berg; +Cc: linuxppc-dev, alsa-devel
At Sun, 28 May 2006 21:00:31 +0200,
Johannes Berg wrote:
>
> The 'fabric' is the thing that pulls all of snd-aoa together, this patch
> adds the 'layout' or 'layout-id' fabric that keys off the 'layout-id'
> property in the device-tree to pull in the proper modules. It itself loads
> if an i2sbus is present with a layout-id (exported in modalias) that it can
> use.
(snip)
> +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);
Just from curiosity: Does this work properly?
I remember that there was a deadlock when you call request_module()
during module probe.
Takashi
^ permalink raw reply [flat|nested] 45+ messages in thread
* Re: [RFC 0/7] snd-aoa: add snd-aoa
2006-05-29 10:12 ` Takashi Iwai
@ 2006-05-30 12:37 ` Johannes Berg
0 siblings, 0 replies; 45+ messages in thread
From: Johannes Berg @ 2006-05-30 12:37 UTC (permalink / raw)
To: Takashi Iwai; +Cc: alsa-devel
[-- Attachment #1.1: Type: text/plain, Size: 318 bytes --]
On Mon, 2006-05-29 at 12:12 +0200, Takashi Iwai wrote:
> Not sure which filtered the post (sourceforge server?)...
Yeah, sf.net, you can look at it on their mailman interface.
> Could you repost with a gzipped attachment?
I'll instead split it up into two which are both smaller, can be done.
johannes
[-- Attachment #1.2: This is a digitally signed message part --]
[-- Type: application/pgp-signature, Size: 793 bytes --]
[-- Attachment #2: Type: text/plain, Size: 0 bytes --]
[-- Attachment #3: Type: text/plain, Size: 161 bytes --]
_______________________________________________
Alsa-devel mailing list
Alsa-devel@lists.sourceforge.net
https://lists.sourceforge.net/lists/listinfo/alsa-devel
^ permalink raw reply [flat|nested] 45+ messages in thread
* Re: [RFC 5/7] snd-aoa: add layout-id fabric
2006-05-29 12:14 ` [Alsa-devel] " Takashi Iwai
@ 2006-05-30 12:37 ` Johannes Berg
-1 siblings, 0 replies; 45+ messages in thread
From: Johannes Berg @ 2006-05-30 12:37 UTC (permalink / raw)
To: Takashi Iwai; +Cc: linuxppc-dev, alsa-devel
[-- Attachment #1.1: Type: text/plain, Size: 323 bytes --]
On Mon, 2006-05-29 at 14:14 +0200, Takashi Iwai wrote:
> Just from curiosity: Does this work properly?
> I remember that there was a deadlock when you call request_module()
> during module probe.
Yes, it works, but I don't think this is actually done during module
probe (would have to check though)
johannes
[-- Attachment #1.2: This is a digitally signed message part --]
[-- Type: application/pgp-signature, Size: 793 bytes --]
[-- Attachment #2: Type: text/plain, Size: 0 bytes --]
[-- Attachment #3: Type: text/plain, Size: 161 bytes --]
_______________________________________________
Alsa-devel mailing list
Alsa-devel@lists.sourceforge.net
https://lists.sourceforge.net/lists/listinfo/alsa-devel
^ permalink raw reply [flat|nested] 45+ messages in thread
* Re: [Alsa-devel] [RFC 5/7] snd-aoa: add layout-id fabric
@ 2006-05-30 12:37 ` Johannes Berg
0 siblings, 0 replies; 45+ messages in thread
From: Johannes Berg @ 2006-05-30 12:37 UTC (permalink / raw)
To: Takashi Iwai; +Cc: linuxppc-dev, alsa-devel
[-- Attachment #1: Type: text/plain, Size: 323 bytes --]
On Mon, 2006-05-29 at 14:14 +0200, Takashi Iwai wrote:
> Just from curiosity: Does this work properly?
> I remember that there was a deadlock when you call request_module()
> during module probe.
Yes, it works, but I don't think this is actually done during module
probe (would have to check though)
johannes
[-- Attachment #2: This is a digitally signed message part --]
[-- Type: application/pgp-signature, Size: 793 bytes --]
^ permalink raw reply [flat|nested] 45+ messages in thread
* Re: [RFC 4/7] snd-aoa: add codecs
2006-05-29 12:08 ` [Alsa-devel] " Takashi Iwai
(?)
(?)
@ 2006-05-30 12:43 ` Johannes Berg
-1 siblings, 0 replies; 45+ messages in thread
From: Johannes Berg @ 2006-05-30 12:43 UTC (permalink / raw)
To: Takashi Iwai; +Cc: linuxppc-dev, alsa-devel
[-- Attachment #1.1: Type: text/plain, Size: 2268 bytes --]
On Mon, 2006-05-29 at 14:08 +0200, Takashi Iwai wrote:
> > + 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;
>
> Isn't it "reg - FIRSTREGISTER" ?
Nah, look at the first line I quoted :)
> I'd define a constant for 128.
#define OFFSET_BECAUSE_ALSA_USERSPACE_PROGRAMS_SUCK 128
;)
Just kidding, will do.
> Fold lines to fit with 80 columns (heh, blaming other one's code is
> easy :)
Heh. I thought I'd Lindent'ed most of it.
> > + /* FIXME: we could be checking if anything changed */
> > + mutex_unlock(&onyx->mutex);
> > +
> > + return 1;
>
> The put callback is supposed to return 0 if the values are unchanged
> (although most apps ignore the return value).
Does it have to? This way there's an event, but...
> > +static u8 initial_values[] = {
>
> Should have ARRAY_SIZE(register_map) since this size must be identical
> with register_map.
Good point.
> > +#define ADDCTL(n) \
> > + do { \
> > + ctl = snd_ctl_new1(&n, onyx); \
> > + if (ctl) { \
> > + ctl->id.device = \
> > + onyx->codec.soundbus_dev->pcm->device; \
> > + aoa_snd_ctl_add(ctl); \
>
> No error check?
Hmm, yeah, I guess it should have error checks. Will revisit that.
> Use strlcpy, or MAX_CODEC_NAME_LEN-1. Similar lines are found in
> tas driver too.
Right.
> Looks a bit too hacky. IMO, it's better to define a struct
Maybe I'll just get rid of it completely, it's never hooked up as far as
I know. But yeah, I guess you're right.
> > +static int tas_snd_capture_source_info(struct snd_kcontrol *kcontrol,
> > + struct snd_ctl_elem_info *uinfo)
> > +{
> > + static char* texts[] = { "Line-In", "Microphone" };
>
> char *texts[]
Any particular reason?
> > +/* I need help here!
>
> Use ifdef. Nested comments are bad.
I know. Temporary code, probably going away completely.
> > + aoa_snd_ctl_add(snd_ctl_new1(&volume_control, tas));
>
> Error checks please.
What should it do on such errors?
Thanks,
johannes
[-- Attachment #1.2: This is a digitally signed message part --]
[-- Type: application/pgp-signature, Size: 793 bytes --]
[-- Attachment #2: Type: text/plain, Size: 0 bytes --]
[-- Attachment #3: Type: text/plain, Size: 161 bytes --]
_______________________________________________
Alsa-devel mailing list
Alsa-devel@lists.sourceforge.net
https://lists.sourceforge.net/lists/listinfo/alsa-devel
^ permalink raw reply [flat|nested] 45+ messages in thread
* Re: [Alsa-devel] [RFC 4/7] snd-aoa: add codecs
2006-05-29 12:08 ` [Alsa-devel] " Takashi Iwai
(?)
@ 2006-05-30 12:43 ` Johannes Berg
2006-05-30 13:19 ` [Alsa-devel] " Takashi Iwai
-1 siblings, 1 reply; 45+ messages in thread
From: Johannes Berg @ 2006-05-30 12:43 UTC (permalink / raw)
To: Takashi Iwai; +Cc: linuxppc-dev, alsa-devel
[-- Attachment #1: Type: text/plain, Size: 2268 bytes --]
On Mon, 2006-05-29 at 14:08 +0200, Takashi Iwai wrote:
> > + 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;
>
> Isn't it "reg - FIRSTREGISTER" ?
Nah, look at the first line I quoted :)
> I'd define a constant for 128.
#define OFFSET_BECAUSE_ALSA_USERSPACE_PROGRAMS_SUCK 128
;)
Just kidding, will do.
> Fold lines to fit with 80 columns (heh, blaming other one's code is
> easy :)
Heh. I thought I'd Lindent'ed most of it.
> > + /* FIXME: we could be checking if anything changed */
> > + mutex_unlock(&onyx->mutex);
> > +
> > + return 1;
>
> The put callback is supposed to return 0 if the values are unchanged
> (although most apps ignore the return value).
Does it have to? This way there's an event, but...
> > +static u8 initial_values[] = {
>
> Should have ARRAY_SIZE(register_map) since this size must be identical
> with register_map.
Good point.
> > +#define ADDCTL(n) \
> > + do { \
> > + ctl = snd_ctl_new1(&n, onyx); \
> > + if (ctl) { \
> > + ctl->id.device = \
> > + onyx->codec.soundbus_dev->pcm->device; \
> > + aoa_snd_ctl_add(ctl); \
>
> No error check?
Hmm, yeah, I guess it should have error checks. Will revisit that.
> Use strlcpy, or MAX_CODEC_NAME_LEN-1. Similar lines are found in
> tas driver too.
Right.
> Looks a bit too hacky. IMO, it's better to define a struct
Maybe I'll just get rid of it completely, it's never hooked up as far as
I know. But yeah, I guess you're right.
> > +static int tas_snd_capture_source_info(struct snd_kcontrol *kcontrol,
> > + struct snd_ctl_elem_info *uinfo)
> > +{
> > + static char* texts[] = { "Line-In", "Microphone" };
>
> char *texts[]
Any particular reason?
> > +/* I need help here!
>
> Use ifdef. Nested comments are bad.
I know. Temporary code, probably going away completely.
> > + aoa_snd_ctl_add(snd_ctl_new1(&volume_control, tas));
>
> Error checks please.
What should it do on such errors?
Thanks,
johannes
[-- Attachment #2: This is a digitally signed message part --]
[-- Type: application/pgp-signature, Size: 793 bytes --]
^ permalink raw reply [flat|nested] 45+ messages in thread
* Re: [RFC 2/7] snd-aoa: add aoa core
2006-05-29 10:57 ` [Alsa-devel] " Takashi Iwai
(?)
(?)
@ 2006-05-30 12:44 ` Johannes Berg
-1 siblings, 0 replies; 45+ messages in thread
From: Johannes Berg @ 2006-05-30 12:44 UTC (permalink / raw)
To: Takashi Iwai; +Cc: linuxppc-dev, alsa-devel
[-- Attachment #1.1: Type: text/plain, Size: 390 bytes --]
On Mon, 2006-05-29 at 12:57 +0200, Takashi Iwai wrote:
> > + alsa_card = snd_card_new(-1, name, mod, sizeof(struct aoa_card));
>
> Usually the first argument here is the index module option so that
> user may specify the card order (e.g. secondary card) independent from
> the order of module loading.
Are you saying I should add an index module option and pass that?
johannes
[-- Attachment #1.2: This is a digitally signed message part --]
[-- Type: application/pgp-signature, Size: 793 bytes --]
[-- Attachment #2: Type: text/plain, Size: 0 bytes --]
[-- Attachment #3: Type: text/plain, Size: 161 bytes --]
_______________________________________________
Alsa-devel mailing list
Alsa-devel@lists.sourceforge.net
https://lists.sourceforge.net/lists/listinfo/alsa-devel
^ permalink raw reply [flat|nested] 45+ messages in thread
* Re: [Alsa-devel] [RFC 2/7] snd-aoa: add aoa core
2006-05-29 10:57 ` [Alsa-devel] " Takashi Iwai
(?)
@ 2006-05-30 12:44 ` Johannes Berg
2006-05-30 13:20 ` [Alsa-devel] " Takashi Iwai
-1 siblings, 1 reply; 45+ messages in thread
From: Johannes Berg @ 2006-05-30 12:44 UTC (permalink / raw)
To: Takashi Iwai; +Cc: linuxppc-dev, alsa-devel
[-- Attachment #1: Type: text/plain, Size: 390 bytes --]
On Mon, 2006-05-29 at 12:57 +0200, Takashi Iwai wrote:
> > + alsa_card = snd_card_new(-1, name, mod, sizeof(struct aoa_card));
>
> Usually the first argument here is the index module option so that
> user may specify the card order (e.g. secondary card) independent from
> the order of module loading.
Are you saying I should add an index module option and pass that?
johannes
[-- Attachment #2: This is a digitally signed message part --]
[-- Type: application/pgp-signature, Size: 793 bytes --]
^ permalink raw reply [flat|nested] 45+ messages in thread
* Re: [RFC 4/7] snd-aoa: add codecs
2006-05-30 12:43 ` Johannes Berg
@ 2006-05-30 13:19 ` Takashi Iwai
0 siblings, 0 replies; 45+ messages in thread
From: Takashi Iwai @ 2006-05-30 13:19 UTC (permalink / raw)
To: Johannes Berg; +Cc: linuxppc-dev, alsa-devel
At Tue, 30 May 2006 14:43:16 +0200,
Johannes Berg wrote:
>
> On Mon, 2006-05-29 at 14:08 +0200, Takashi Iwai wrote:
> > > + 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;
> >
> > Isn't it "reg - FIRSTREGISTER" ?
>
> Nah, look at the first line I quoted :)
Ah, OK it's anyway same...
> > > + /* FIXME: we could be checking if anything changed */
> > > + mutex_unlock(&onyx->mutex);
> > > +
> > > + return 1;
> >
> > The put callback is supposed to return 0 if the values are unchanged
> > (although most apps ignore the return value).
>
> Does it have to? This way there's an event, but...
Yes, in principle. As mentioned, it works even without the check,
though.
> > > +static int tas_snd_capture_source_info(struct snd_kcontrol *kcontrol,
> > > + struct snd_ctl_elem_info *uinfo)
> > > +{
> > > + static char* texts[] = { "Line-In", "Microphone" };
> >
> > char *texts[]
>
> Any particular reason?
Well, I meant the position of asterisk to follow the conventional C
coding style. Of course it should be static.
> > > + aoa_snd_ctl_add(snd_ctl_new1(&volume_control, tas));
> >
> > Error checks please.
>
> What should it do on such errors?
The driver should give up the initialization and fail to load.
Usually the error from snd_ctl_add() is critical, either no memory or
a duplicated control element is found.
Takashi
^ permalink raw reply [flat|nested] 45+ messages in thread
* Re: [Alsa-devel] [RFC 4/7] snd-aoa: add codecs
@ 2006-05-30 13:19 ` Takashi Iwai
0 siblings, 0 replies; 45+ messages in thread
From: Takashi Iwai @ 2006-05-30 13:19 UTC (permalink / raw)
To: Johannes Berg; +Cc: linuxppc-dev, alsa-devel
At Tue, 30 May 2006 14:43:16 +0200,
Johannes Berg wrote:
>
> On Mon, 2006-05-29 at 14:08 +0200, Takashi Iwai wrote:
> > > + 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;
> >
> > Isn't it "reg - FIRSTREGISTER" ?
>
> Nah, look at the first line I quoted :)
Ah, OK it's anyway same...
> > > + /* FIXME: we could be checking if anything changed */
> > > + mutex_unlock(&onyx->mutex);
> > > +
> > > + return 1;
> >
> > The put callback is supposed to return 0 if the values are unchanged
> > (although most apps ignore the return value).
>
> Does it have to? This way there's an event, but...
Yes, in principle. As mentioned, it works even without the check,
though.
> > > +static int tas_snd_capture_source_info(struct snd_kcontrol *kcontrol,
> > > + struct snd_ctl_elem_info *uinfo)
> > > +{
> > > + static char* texts[] = { "Line-In", "Microphone" };
> >
> > char *texts[]
>
> Any particular reason?
Well, I meant the position of asterisk to follow the conventional C
coding style. Of course it should be static.
> > > + aoa_snd_ctl_add(snd_ctl_new1(&volume_control, tas));
> >
> > Error checks please.
>
> What should it do on such errors?
The driver should give up the initialization and fail to load.
Usually the error from snd_ctl_add() is critical, either no memory or
a duplicated control element is found.
Takashi
^ permalink raw reply [flat|nested] 45+ messages in thread
* Re: [RFC 2/7] snd-aoa: add aoa core
2006-05-30 12:44 ` Johannes Berg
@ 2006-05-30 13:20 ` Takashi Iwai
0 siblings, 0 replies; 45+ messages in thread
From: Takashi Iwai @ 2006-05-30 13:20 UTC (permalink / raw)
To: Johannes Berg; +Cc: linuxppc-dev, alsa-devel
At Tue, 30 May 2006 14:44:31 +0200,
Johannes Berg wrote:
>
> On Mon, 2006-05-29 at 12:57 +0200, Takashi Iwai wrote:
> > > + alsa_card = snd_card_new(-1, name, mod, sizeof(struct aoa_card));
> >
> > Usually the first argument here is the index module option so that
> > user may specify the card order (e.g. secondary card) independent from
> > the order of module loading.
>
> Are you saying I should add an index module option and pass that?
Yep, all other drivers do so.
Takashi
^ permalink raw reply [flat|nested] 45+ messages in thread
* Re: [Alsa-devel] [RFC 2/7] snd-aoa: add aoa core
@ 2006-05-30 13:20 ` Takashi Iwai
0 siblings, 0 replies; 45+ messages in thread
From: Takashi Iwai @ 2006-05-30 13:20 UTC (permalink / raw)
To: Johannes Berg; +Cc: linuxppc-dev, alsa-devel
At Tue, 30 May 2006 14:44:31 +0200,
Johannes Berg wrote:
>
> On Mon, 2006-05-29 at 12:57 +0200, Takashi Iwai wrote:
> > > + alsa_card = snd_card_new(-1, name, mod, sizeof(struct aoa_card));
> >
> > Usually the first argument here is the index module option so that
> > user may specify the card order (e.g. secondary card) independent from
> > the order of module loading.
>
> Are you saying I should add an index module option and pass that?
Yep, all other drivers do so.
Takashi
^ permalink raw reply [flat|nested] 45+ messages in thread
* Re: [RFC 4/7] snd-aoa: add codecs
2006-05-30 13:19 ` [Alsa-devel] " Takashi Iwai
@ 2006-05-30 13:26 ` Johannes Berg
-1 siblings, 0 replies; 45+ messages in thread
From: Johannes Berg @ 2006-05-30 13:26 UTC (permalink / raw)
To: Takashi Iwai; +Cc: linuxppc-dev, alsa-devel
[-- Attachment #1.1: Type: text/plain, Size: 1035 bytes --]
> > > > + return 1;
> > >
> > > The put callback is supposed to return 0 if the values are unchanged
> > > (although most apps ignore the return value).
> >
> > Does it have to? This way there's an event, but...
>
> Yes, in principle. As mentioned, it works even without the check,
> though.
Hmm. I thought it worked like this:
0 -> no change
1 -> change applied, and other apps are notified, 0 returned to the app
-err -> error returned to app
Is that wrong?
> Well, I meant the position of asterisk to follow the conventional C
> coding style. Of course it should be static.
Ah. Yeah, I'll see to that and probably just Lindent the files (cleaning
up after it...)
> The driver should give up the initialization and fail to load.
> Usually the error from snd_ctl_add() is critical, either no memory or
> a duplicated control element is found.
Hmm, that's a bit hard to do since the driver here consists of more than
one module, but I suppose I can do something about it.
johannes
[-- Attachment #1.2: This is a digitally signed message part --]
[-- Type: application/pgp-signature, Size: 793 bytes --]
[-- Attachment #2: Type: text/plain, Size: 0 bytes --]
[-- Attachment #3: Type: text/plain, Size: 161 bytes --]
_______________________________________________
Alsa-devel mailing list
Alsa-devel@lists.sourceforge.net
https://lists.sourceforge.net/lists/listinfo/alsa-devel
^ permalink raw reply [flat|nested] 45+ messages in thread
* Re: [Alsa-devel] [RFC 4/7] snd-aoa: add codecs
@ 2006-05-30 13:26 ` Johannes Berg
0 siblings, 0 replies; 45+ messages in thread
From: Johannes Berg @ 2006-05-30 13:26 UTC (permalink / raw)
To: Takashi Iwai; +Cc: linuxppc-dev, alsa-devel
[-- Attachment #1: Type: text/plain, Size: 1035 bytes --]
> > > > + return 1;
> > >
> > > The put callback is supposed to return 0 if the values are unchanged
> > > (although most apps ignore the return value).
> >
> > Does it have to? This way there's an event, but...
>
> Yes, in principle. As mentioned, it works even without the check,
> though.
Hmm. I thought it worked like this:
0 -> no change
1 -> change applied, and other apps are notified, 0 returned to the app
-err -> error returned to app
Is that wrong?
> Well, I meant the position of asterisk to follow the conventional C
> coding style. Of course it should be static.
Ah. Yeah, I'll see to that and probably just Lindent the files (cleaning
up after it...)
> The driver should give up the initialization and fail to load.
> Usually the error from snd_ctl_add() is critical, either no memory or
> a duplicated control element is found.
Hmm, that's a bit hard to do since the driver here consists of more than
one module, but I suppose I can do something about it.
johannes
[-- Attachment #2: This is a digitally signed message part --]
[-- Type: application/pgp-signature, Size: 793 bytes --]
^ permalink raw reply [flat|nested] 45+ messages in thread
* Re: [RFC 4/7] snd-aoa: add codecs
2006-05-30 13:26 ` [Alsa-devel] " Johannes Berg
(?)
@ 2006-05-30 13:35 ` Takashi Iwai
-1 siblings, 0 replies; 45+ messages in thread
From: Takashi Iwai @ 2006-05-30 13:35 UTC (permalink / raw)
To: Johannes Berg; +Cc: linuxppc-dev, alsa-devel
At Tue, 30 May 2006 15:26:10 +0200,
Johannes Berg wrote:
>
> > > > > + return 1;
> > > >
> > > > The put callback is supposed to return 0 if the values are unchanged
> > > > (although most apps ignore the return value).
> > >
> > > Does it have to? This way there's an event, but...
> >
> > Yes, in principle. As mentioned, it works even without the check,
> > though.
>
> Hmm. I thought it worked like this:
> 0 -> no change
> 1 -> change applied, and other apps are notified, 0 returned to the app
> -err -> error returned to app
>
> Is that wrong?
It's correct.
And I thought the put callbacks in your code always return either 1 or
-err?
Takashi
^ permalink raw reply [flat|nested] 45+ messages in thread
* Re: [Alsa-devel] [RFC 4/7] snd-aoa: add codecs
2006-05-30 13:26 ` [Alsa-devel] " Johannes Berg
(?)
(?)
@ 2006-05-30 13:35 ` Takashi Iwai
2006-05-30 13:39 ` [Alsa-devel] " Johannes Berg
-1 siblings, 1 reply; 45+ messages in thread
From: Takashi Iwai @ 2006-05-30 13:35 UTC (permalink / raw)
To: Johannes Berg; +Cc: linuxppc-dev, alsa-devel
At Tue, 30 May 2006 15:26:10 +0200,
Johannes Berg wrote:
>
> > > > > + return 1;
> > > >
> > > > The put callback is supposed to return 0 if the values are unchanged
> > > > (although most apps ignore the return value).
> > >
> > > Does it have to? This way there's an event, but...
> >
> > Yes, in principle. As mentioned, it works even without the check,
> > though.
>
> Hmm. I thought it worked like this:
> 0 -> no change
> 1 -> change applied, and other apps are notified, 0 returned to the app
> -err -> error returned to app
>
> Is that wrong?
It's correct.
And I thought the put callbacks in your code always return either 1 or
-err?
Takashi
^ permalink raw reply [flat|nested] 45+ messages in thread
* Re: [RFC 4/7] snd-aoa: add codecs
2006-05-30 13:35 ` [Alsa-devel] " Takashi Iwai
@ 2006-05-30 13:39 ` Johannes Berg
0 siblings, 0 replies; 45+ messages in thread
From: Johannes Berg @ 2006-05-30 13:39 UTC (permalink / raw)
To: Takashi Iwai; +Cc: linuxppc-dev, alsa-devel
[-- Attachment #1.1: Type: text/plain, Size: 394 bytes --]
On Tue, 2006-05-30 at 15:35 +0200, Takashi Iwai wrote:
> And I thought the put callbacks in your code always return either 1 or
> -err?
Yeah, they return 1 if the change was applied (or no change) and -err if
there's an error about the change (only in very few cases).
I figured the extra notification wouldn't hurt, and to the application
there wouldn't be a problem.
johannes
[-- Attachment #1.2: This is a digitally signed message part --]
[-- Type: application/pgp-signature, Size: 793 bytes --]
[-- Attachment #2: Type: text/plain, Size: 0 bytes --]
[-- Attachment #3: Type: text/plain, Size: 161 bytes --]
_______________________________________________
Alsa-devel mailing list
Alsa-devel@lists.sourceforge.net
https://lists.sourceforge.net/lists/listinfo/alsa-devel
^ permalink raw reply [flat|nested] 45+ messages in thread
* Re: [Alsa-devel] [RFC 4/7] snd-aoa: add codecs
@ 2006-05-30 13:39 ` Johannes Berg
0 siblings, 0 replies; 45+ messages in thread
From: Johannes Berg @ 2006-05-30 13:39 UTC (permalink / raw)
To: Takashi Iwai; +Cc: linuxppc-dev, alsa-devel
[-- Attachment #1: Type: text/plain, Size: 394 bytes --]
On Tue, 2006-05-30 at 15:35 +0200, Takashi Iwai wrote:
> And I thought the put callbacks in your code always return either 1 or
> -err?
Yeah, they return 1 if the change was applied (or no change) and -err if
there's an error about the change (only in very few cases).
I figured the extra notification wouldn't hurt, and to the application
there wouldn't be a problem.
johannes
[-- Attachment #2: This is a digitally signed message part --]
[-- Type: application/pgp-signature, Size: 793 bytes --]
^ permalink raw reply [flat|nested] 45+ messages in thread
* Re: [RFC 4/7] snd-aoa: add codecs
2006-05-30 13:39 ` [Alsa-devel] " Johannes Berg
@ 2006-05-30 13:48 ` Takashi Iwai
-1 siblings, 0 replies; 45+ messages in thread
From: Takashi Iwai @ 2006-05-30 13:48 UTC (permalink / raw)
To: Johannes Berg; +Cc: linuxppc-dev, alsa-devel
At Tue, 30 May 2006 15:39:09 +0200,
Johannes Berg wrote:
>
> On Tue, 2006-05-30 at 15:35 +0200, Takashi Iwai wrote:
>
> > And I thought the put callbacks in your code always return either 1 or
> > -err?
>
> Yeah, they return 1 if the change was applied (or no change) and -err if
> there's an error about the change (only in very few cases).
>
> I figured the extra notification wouldn't hurt, and to the application
> there wouldn't be a problem.
Actually the extra notification won't hurt much in practice.
I've seen that some drivers also don't check the unchanged case and
return always 1, so apparently the apps work. But, it's of course
better to check the state on the driver side.
Takashi
^ permalink raw reply [flat|nested] 45+ messages in thread
* Re: [Alsa-devel] [RFC 4/7] snd-aoa: add codecs
@ 2006-05-30 13:48 ` Takashi Iwai
0 siblings, 0 replies; 45+ messages in thread
From: Takashi Iwai @ 2006-05-30 13:48 UTC (permalink / raw)
To: Johannes Berg; +Cc: linuxppc-dev, alsa-devel
At Tue, 30 May 2006 15:39:09 +0200,
Johannes Berg wrote:
>
> On Tue, 2006-05-30 at 15:35 +0200, Takashi Iwai wrote:
>
> > And I thought the put callbacks in your code always return either 1 or
> > -err?
>
> Yeah, they return 1 if the change was applied (or no change) and -err if
> there's an error about the change (only in very few cases).
>
> I figured the extra notification wouldn't hurt, and to the application
> there wouldn't be a problem.
Actually the extra notification won't hurt much in practice.
I've seen that some drivers also don't check the unchanged case and
return always 1, so apparently the apps work. But, it's of course
better to check the state on the driver side.
Takashi
^ permalink raw reply [flat|nested] 45+ messages in thread
* Re: [RFC 4/7] snd-aoa: add codecs
2006-05-30 13:48 ` [Alsa-devel] " Takashi Iwai
@ 2006-05-30 13:53 ` Johannes Berg
-1 siblings, 0 replies; 45+ messages in thread
From: Johannes Berg @ 2006-05-30 13:53 UTC (permalink / raw)
To: Takashi Iwai; +Cc: linuxppc-dev, alsa-devel
[-- Attachment #1.1: Type: text/plain, Size: 581 bytes --]
On Tue, 2006-05-30 at 15:48 +0200, Takashi Iwai wrote:
> Actually the extra notification won't hurt much in practice.
> I've seen that some drivers also don't check the unchanged case and
> return always 1, so apparently the apps work. But, it's of course
> better to check the state on the driver side.
Yeah, agreed. I'll fix it.
How about making alsamixer actually handle the error cases? In the onyx
codec, there's some things that prohibit changing the s/pdif output if
you dno't use a bitrate that can handle, but alsamixer 'changes' it
anyway...
johannes
[-- Attachment #1.2: This is a digitally signed message part --]
[-- Type: application/pgp-signature, Size: 793 bytes --]
[-- Attachment #2: Type: text/plain, Size: 0 bytes --]
[-- Attachment #3: Type: text/plain, Size: 161 bytes --]
_______________________________________________
Alsa-devel mailing list
Alsa-devel@lists.sourceforge.net
https://lists.sourceforge.net/lists/listinfo/alsa-devel
^ permalink raw reply [flat|nested] 45+ messages in thread
* Re: [Alsa-devel] [RFC 4/7] snd-aoa: add codecs
@ 2006-05-30 13:53 ` Johannes Berg
0 siblings, 0 replies; 45+ messages in thread
From: Johannes Berg @ 2006-05-30 13:53 UTC (permalink / raw)
To: Takashi Iwai; +Cc: linuxppc-dev, alsa-devel
[-- Attachment #1: Type: text/plain, Size: 581 bytes --]
On Tue, 2006-05-30 at 15:48 +0200, Takashi Iwai wrote:
> Actually the extra notification won't hurt much in practice.
> I've seen that some drivers also don't check the unchanged case and
> return always 1, so apparently the apps work. But, it's of course
> better to check the state on the driver side.
Yeah, agreed. I'll fix it.
How about making alsamixer actually handle the error cases? In the onyx
codec, there's some things that prohibit changing the s/pdif output if
you dno't use a bitrate that can handle, but alsamixer 'changes' it
anyway...
johannes
[-- Attachment #2: This is a digitally signed message part --]
[-- Type: application/pgp-signature, Size: 793 bytes --]
^ permalink raw reply [flat|nested] 45+ messages in thread
* Re: [RFC 4/7] snd-aoa: add codecs
2006-05-30 13:53 ` [Alsa-devel] " Johannes Berg
@ 2006-05-30 13:56 ` Takashi Iwai
-1 siblings, 0 replies; 45+ messages in thread
From: Takashi Iwai @ 2006-05-30 13:56 UTC (permalink / raw)
To: Johannes Berg; +Cc: linuxppc-dev, alsa-devel
At Tue, 30 May 2006 15:53:25 +0200,
Johannes Berg wrote:
>
> On Tue, 2006-05-30 at 15:48 +0200, Takashi Iwai wrote:
>
> > Actually the extra notification won't hurt much in practice.
> > I've seen that some drivers also don't check the unchanged case and
> > return always 1, so apparently the apps work. But, it's of course
> > better to check the state on the driver side.
>
> Yeah, agreed. I'll fix it.
>
> How about making alsamixer actually handle the error cases? In the onyx
> codec, there's some things that prohibit changing the s/pdif output if
> you dno't use a bitrate that can handle, but alsamixer 'changes' it
> anyway...
Oh, it should be called as a "bug"... ;)
Takashi
^ permalink raw reply [flat|nested] 45+ messages in thread
* Re: [Alsa-devel] [RFC 4/7] snd-aoa: add codecs
@ 2006-05-30 13:56 ` Takashi Iwai
0 siblings, 0 replies; 45+ messages in thread
From: Takashi Iwai @ 2006-05-30 13:56 UTC (permalink / raw)
To: Johannes Berg; +Cc: linuxppc-dev, alsa-devel
At Tue, 30 May 2006 15:53:25 +0200,
Johannes Berg wrote:
>
> On Tue, 2006-05-30 at 15:48 +0200, Takashi Iwai wrote:
>
> > Actually the extra notification won't hurt much in practice.
> > I've seen that some drivers also don't check the unchanged case and
> > return always 1, so apparently the apps work. But, it's of course
> > better to check the state on the driver side.
>
> Yeah, agreed. I'll fix it.
>
> How about making alsamixer actually handle the error cases? In the onyx
> codec, there's some things that prohibit changing the s/pdif output if
> you dno't use a bitrate that can handle, but alsamixer 'changes' it
> anyway...
Oh, it should be called as a "bug"... ;)
Takashi
^ permalink raw reply [flat|nested] 45+ messages in thread
end of thread, other threads:[~2006-05-30 13:56 UTC | newest]
Thread overview: 45+ messages (download: mbox.gz / follow: Atom feed)
-- links below jump to the message on this page --
2006-05-28 19:00 [RFC 0/7] snd-aoa: add snd-aoa Johannes Berg
2006-05-28 19:00 ` [RFC 1/7] snd-aoa: add aoa header files Johannes Berg
2006-05-28 19:00 ` Johannes Berg
2006-05-28 19:00 ` [RFC 2/7] snd-aoa: add aoa core Johannes Berg
2006-05-28 19:00 ` Johannes Berg
2006-05-29 10:57 ` Takashi Iwai
2006-05-29 10:57 ` [Alsa-devel] " Takashi Iwai
2006-05-30 12:44 ` Johannes Berg
2006-05-30 13:20 ` Takashi Iwai
2006-05-30 13:20 ` [Alsa-devel] " Takashi Iwai
2006-05-30 12:44 ` Johannes Berg
2006-05-28 19:00 ` [RFC 3/7] snd-aoa: add soundbus and i2sbus Johannes Berg
2006-05-28 19:00 ` [RFC 4/7] snd-aoa: add codecs Johannes Berg
2006-05-28 19:00 ` Johannes Berg
2006-05-29 12:08 ` Takashi Iwai
2006-05-29 12:08 ` [Alsa-devel] " Takashi Iwai
2006-05-30 12:43 ` Johannes Berg
2006-05-30 13:19 ` Takashi Iwai
2006-05-30 13:19 ` [Alsa-devel] " Takashi Iwai
2006-05-30 13:26 ` Johannes Berg
2006-05-30 13:26 ` [Alsa-devel] " Johannes Berg
2006-05-30 13:35 ` Takashi Iwai
2006-05-30 13:35 ` [Alsa-devel] " Takashi Iwai
2006-05-30 13:39 ` Johannes Berg
2006-05-30 13:39 ` [Alsa-devel] " Johannes Berg
2006-05-30 13:48 ` Takashi Iwai
2006-05-30 13:48 ` [Alsa-devel] " Takashi Iwai
2006-05-30 13:53 ` Johannes Berg
2006-05-30 13:53 ` [Alsa-devel] " Johannes Berg
2006-05-30 13:56 ` Takashi Iwai
2006-05-30 13:56 ` [Alsa-devel] " Takashi Iwai
2006-05-30 12:43 ` Johannes Berg
2006-05-28 19:00 ` [RFC 5/7] snd-aoa: add layout-id fabric Johannes Berg
2006-05-28 19:00 ` Johannes Berg
2006-05-29 12:14 ` Takashi Iwai
2006-05-29 12:14 ` [Alsa-devel] " Takashi Iwai
2006-05-30 12:37 ` Johannes Berg
2006-05-30 12:37 ` [Alsa-devel] " Johannes Berg
2006-05-28 19:00 ` [RFC 6/7] snd-aoa: add Kconfig and Makefile Johannes Berg
2006-05-28 19:00 ` Johannes Berg
2006-05-28 19:00 ` [RFC 7/7] snd-aoa: wire up aoa in sound/ Johannes Berg
2006-05-28 19:00 ` Johannes Berg
2006-05-28 19:10 ` [RFC 0/7] snd-aoa: add snd-aoa Johannes Berg
2006-05-29 10:12 ` Takashi Iwai
2006-05-30 12:37 ` Johannes Berg
This is an external index of several public inboxes,
see mirroring instructions on how to clone and mirror
all data and code used by this external index.