diff options
Diffstat (limited to '')
33 files changed, 7549 insertions, 0 deletions
diff --git a/sound/aoa/Kconfig b/sound/aoa/Kconfig new file mode 100644 index 000000000..c081e18b9 --- /dev/null +++ b/sound/aoa/Kconfig @@ -0,0 +1,17 @@ +menuconfig SND_AOA + tristate "Apple Onboard Audio driver" + depends on PPC_PMAC + select SND_PCM + ---help--- + This option enables the new driver for the various + Apple Onboard Audio components. + +if SND_AOA + +source "sound/aoa/fabrics/Kconfig" + +source "sound/aoa/codecs/Kconfig" + +source "sound/aoa/soundbus/Kconfig" + +endif # SND_AOA diff --git a/sound/aoa/Makefile b/sound/aoa/Makefile new file mode 100644 index 000000000..a8c037f90 --- /dev/null +++ b/sound/aoa/Makefile @@ -0,0 +1,4 @@ +obj-$(CONFIG_SND_AOA) += core/ +obj-$(CONFIG_SND_AOA_SOUNDBUS) += soundbus/ +obj-$(CONFIG_SND_AOA) += fabrics/ +obj-$(CONFIG_SND_AOA) += codecs/ diff --git a/sound/aoa/aoa-gpio.h b/sound/aoa/aoa-gpio.h new file mode 100644 index 000000000..6065b0344 --- /dev/null +++ b/sound/aoa/aoa-gpio.h @@ -0,0 +1,83 @@ +/* + * Apple Onboard Audio GPIO definitions + * + * Copyright 2006 Johannes Berg <johannes@sipsolutions.net> + * + * GPL v2, can be found in COPYING. + */ + +#ifndef __AOA_GPIO_H +#define __AOA_GPIO_H +#include <linux/workqueue.h> +#include <linux/mutex.h> +#include <asm/prom.h> + +typedef void (*notify_func_t)(void *data); + +enum notify_type { + AOA_NOTIFY_HEADPHONE, + AOA_NOTIFY_LINE_IN, + AOA_NOTIFY_LINE_OUT, +}; + +struct gpio_runtime; +struct gpio_methods { + /* for initialisation/de-initialisation of the GPIO layer */ + void (*init)(struct gpio_runtime *rt); + void (*exit)(struct gpio_runtime *rt); + + /* turn off headphone, speakers, lineout */ + void (*all_amps_off)(struct gpio_runtime *rt); + /* turn headphone, speakers, lineout back to previous setting */ + void (*all_amps_restore)(struct gpio_runtime *rt); + + void (*set_headphone)(struct gpio_runtime *rt, int on); + void (*set_speakers)(struct gpio_runtime *rt, int on); + void (*set_lineout)(struct gpio_runtime *rt, int on); + void (*set_master)(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); + int (*get_master)(struct gpio_runtime *rt); + + void (*set_hw_reset)(struct gpio_runtime *rt, int on); + + /* use this to be notified of any events. The notification + * function is passed the data, and is called in process + * context by the use of schedule_work. + * The interface for it is that setting a function to NULL + * removes it, and they return 0 if the operation succeeded, + * and -EBUSY if the notification is already assigned by + * someone else. */ + int (*set_notify)(struct gpio_runtime *rt, + enum notify_type type, + notify_func_t notify, + void *data); + /* returns 0 if not plugged in, 1 if plugged in + * or a negative error code */ + int (*get_detect)(struct gpio_runtime *rt, + enum notify_type type); +}; + +struct gpio_notification { + struct delayed_work work; + notify_func_t notify; + void *data; + void *gpio_private; + struct mutex mutex; +}; + +struct gpio_runtime { + /* to be assigned by fabric */ + struct device_node *node; + /* since everyone needs this pointer anyway... */ + struct gpio_methods *methods; + /* to be used by the gpio implementation */ + int implementation_private; + struct gpio_notification headphone_notify; + struct gpio_notification line_in_notify; + struct gpio_notification line_out_notify; +}; + +#endif /* __AOA_GPIO_H */ diff --git a/sound/aoa/aoa.h b/sound/aoa/aoa.h new file mode 100644 index 000000000..34c668f27 --- /dev/null +++ b/sound/aoa/aoa.h @@ -0,0 +1,129 @@ +/* + * 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> +#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, struct device *dev); + +/* 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(enum snd_device_type type, + void * device_data, struct snd_device_ops * ops); +extern struct snd_card *aoa_get_card(void); +extern int aoa_snd_ctl_add(struct snd_kcontrol* control); + +/* GPIO stuff */ +extern struct gpio_methods *pmf_gpio_methods; +extern struct gpio_methods *ftr_gpio_methods; +/* extern struct gpio_methods *map_gpio_methods; */ + +#endif /* __AOA_H */ diff --git a/sound/aoa/codecs/Kconfig b/sound/aoa/codecs/Kconfig new file mode 100644 index 000000000..0c68e3283 --- /dev/null +++ b/sound/aoa/codecs/Kconfig @@ -0,0 +1,24 @@ +config SND_AOA_ONYX + tristate "support Onyx chip" + select I2C + select I2C_POWERMAC + ---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_TAS + tristate "support TAS chips" + select I2C + select I2C_POWERMAC + ---help--- + This option enables support for the tas chips + found in a lot of Apple Machines, especially + iBooks and PowerBooks without digital. + +config SND_AOA_TOONIE + tristate "support Toonie chip" + ---help--- + This option enables support for the toonie codec + found in the Mac Mini. If you have a Mac Mini and + want to hear sound, select this option. diff --git a/sound/aoa/codecs/Makefile b/sound/aoa/codecs/Makefile new file mode 100644 index 000000000..95f4c3849 --- /dev/null +++ b/sound/aoa/codecs/Makefile @@ -0,0 +1,8 @@ +# SPDX-License-Identifier: GPL-2.0 +snd-aoa-codec-onyx-objs := onyx.o +snd-aoa-codec-tas-objs := tas.o +snd-aoa-codec-toonie-objs := toonie.o + +obj-$(CONFIG_SND_AOA_ONYX) += snd-aoa-codec-onyx.o +obj-$(CONFIG_SND_AOA_TAS) += snd-aoa-codec-tas.o +obj-$(CONFIG_SND_AOA_TOONIE) += snd-aoa-codec-toonie.o diff --git a/sound/aoa/codecs/onyx.c b/sound/aoa/codecs/onyx.c new file mode 100644 index 000000000..6224fd3bb --- /dev/null +++ b/sound/aoa/codecs/onyx.c @@ -0,0 +1,1061 @@ +/* + * 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> +#include <linux/slab.h> +MODULE_AUTHOR("Johannes Berg <johannes@sipsolutions.net>"); +MODULE_LICENSE("GPL"); +MODULE_DESCRIPTION("pcm3052 (onyx) codec driver for snd-aoa"); + +#include "onyx.h" +#include "../aoa.h" +#include "../soundbus/soundbus.h" + + +#define PFX "snd-aoa-codec-onyx: " + +struct onyx { + /* cache registers 65 to 80, they are write-only! */ + u8 cache[16]; + struct i2c_client *i2c; + struct aoa_codec codec; + u32 initialised:1, + spdif_locked:1, + analog_locked:1, + original_mute:2; + int open_count; + struct codec_info *codec_info; + + /* mutex serializes concurrent access to the device + * and this structure. + */ + struct mutex mutex; +}; +#define codec_to_onyx(c) container_of(c, struct onyx, codec) + +/* both return 0 if all ok, else on error */ +static int onyx_read_register(struct onyx *onyx, u8 reg, u8 *value) +{ + s32 v; + + if (reg != ONYX_REG_CONTROL) { + *value = onyx->cache[reg-FIRSTREGISTER]; + return 0; + } + v = i2c_smbus_read_byte_data(onyx->i2c, reg); + if (v < 0) { + *value = 0; + return -1; + } + *value = (u8)v; + onyx->cache[ONYX_REG_CONTROL-FIRSTREGISTER] = *value; + return 0; +} + +static int onyx_write_register(struct onyx *onyx, u8 reg, u8 value) +{ + int result; + + result = i2c_smbus_write_byte_data(onyx->i2c, reg, value); + if (!result) + onyx->cache[reg-FIRSTREGISTER] = value; + return result; +} + +/* alsa stuff */ + +static int onyx_dev_register(struct snd_device *dev) +{ + return 0; +} + +static struct snd_device_ops ops = { + .dev_register = onyx_dev_register, +}; + +/* this is necessary because most alsa mixer programs + * can't properly handle the negative range */ +#define VOLUME_RANGE_SHIFT 128 + +static int onyx_snd_vol_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + uinfo->count = 2; + uinfo->value.integer.min = -128 + VOLUME_RANGE_SHIFT; + uinfo->value.integer.max = -1 + VOLUME_RANGE_SHIFT; + return 0; +} + +static int onyx_snd_vol_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct onyx *onyx = snd_kcontrol_chip(kcontrol); + s8 l, r; + + mutex_lock(&onyx->mutex); + onyx_read_register(onyx, ONYX_REG_DAC_ATTEN_LEFT, &l); + onyx_read_register(onyx, ONYX_REG_DAC_ATTEN_RIGHT, &r); + mutex_unlock(&onyx->mutex); + + ucontrol->value.integer.value[0] = l + VOLUME_RANGE_SHIFT; + ucontrol->value.integer.value[1] = r + VOLUME_RANGE_SHIFT; + + return 0; +} + +static int onyx_snd_vol_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct onyx *onyx = snd_kcontrol_chip(kcontrol); + s8 l, r; + + if (ucontrol->value.integer.value[0] < -128 + VOLUME_RANGE_SHIFT || + ucontrol->value.integer.value[0] > -1 + VOLUME_RANGE_SHIFT) + return -EINVAL; + if (ucontrol->value.integer.value[1] < -128 + VOLUME_RANGE_SHIFT || + ucontrol->value.integer.value[1] > -1 + VOLUME_RANGE_SHIFT) + return -EINVAL; + + mutex_lock(&onyx->mutex); + onyx_read_register(onyx, ONYX_REG_DAC_ATTEN_LEFT, &l); + onyx_read_register(onyx, ONYX_REG_DAC_ATTEN_RIGHT, &r); + + if (l + VOLUME_RANGE_SHIFT == ucontrol->value.integer.value[0] && + r + VOLUME_RANGE_SHIFT == ucontrol->value.integer.value[1]) { + mutex_unlock(&onyx->mutex); + return 0; + } + + onyx_write_register(onyx, ONYX_REG_DAC_ATTEN_LEFT, + ucontrol->value.integer.value[0] + - VOLUME_RANGE_SHIFT); + onyx_write_register(onyx, ONYX_REG_DAC_ATTEN_RIGHT, + ucontrol->value.integer.value[1] + - VOLUME_RANGE_SHIFT); + mutex_unlock(&onyx->mutex); + + return 1; +} + +static const struct snd_kcontrol_new volume_control = { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Master Playback Volume", + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, + .info = onyx_snd_vol_info, + .get = onyx_snd_vol_get, + .put = onyx_snd_vol_put, +}; + +/* like above, this is necessary because a lot + * of alsa mixer programs don't handle ranges + * that don't start at 0 properly. + * even alsamixer is one of them... */ +#define INPUTGAIN_RANGE_SHIFT (-3) + +static int onyx_snd_inputgain_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER; + uinfo->count = 1; + uinfo->value.integer.min = 3 + INPUTGAIN_RANGE_SHIFT; + uinfo->value.integer.max = 28 + INPUTGAIN_RANGE_SHIFT; + return 0; +} + +static int onyx_snd_inputgain_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct onyx *onyx = snd_kcontrol_chip(kcontrol); + u8 ig; + + mutex_lock(&onyx->mutex); + onyx_read_register(onyx, ONYX_REG_ADC_CONTROL, &ig); + mutex_unlock(&onyx->mutex); + + ucontrol->value.integer.value[0] = + (ig & ONYX_ADC_PGA_GAIN_MASK) + INPUTGAIN_RANGE_SHIFT; + + return 0; +} + +static int onyx_snd_inputgain_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct onyx *onyx = snd_kcontrol_chip(kcontrol); + u8 v, n; + + if (ucontrol->value.integer.value[0] < 3 + INPUTGAIN_RANGE_SHIFT || + ucontrol->value.integer.value[0] > 28 + INPUTGAIN_RANGE_SHIFT) + return -EINVAL; + mutex_lock(&onyx->mutex); + onyx_read_register(onyx, ONYX_REG_ADC_CONTROL, &v); + n = v; + n &= ~ONYX_ADC_PGA_GAIN_MASK; + n |= (ucontrol->value.integer.value[0] - INPUTGAIN_RANGE_SHIFT) + & ONYX_ADC_PGA_GAIN_MASK; + onyx_write_register(onyx, ONYX_REG_ADC_CONTROL, n); + mutex_unlock(&onyx->mutex); + + return n != v; +} + +static const 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 const char * const texts[] = { "Line-In", "Microphone" }; + + return snd_ctl_enum_info(uinfo, 1, 2, texts); +} + +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) +{ + if (ucontrol->value.enumerated.item[0] > 1) + return -EINVAL; + onyx_set_capture_source(snd_kcontrol_chip(kcontrol), + ucontrol->value.enumerated.item[0]); + return 1; +} + +static const 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, +}; + +#define onyx_snd_mute_info snd_ctl_boolean_stereo_info + +static int onyx_snd_mute_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct onyx *onyx = snd_kcontrol_chip(kcontrol); + u8 c; + + mutex_lock(&onyx->mutex); + onyx_read_register(onyx, ONYX_REG_DAC_CONTROL, &c); + mutex_unlock(&onyx->mutex); + + ucontrol->value.integer.value[0] = !(c & ONYX_MUTE_LEFT); + ucontrol->value.integer.value[1] = !(c & ONYX_MUTE_RIGHT); + + return 0; +} + +static int onyx_snd_mute_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct onyx *onyx = snd_kcontrol_chip(kcontrol); + u8 v = 0, c = 0; + int err = -EBUSY; + + mutex_lock(&onyx->mutex); + if (onyx->analog_locked) + goto out_unlock; + + onyx_read_register(onyx, ONYX_REG_DAC_CONTROL, &v); + c = v; + c &= ~(ONYX_MUTE_RIGHT | ONYX_MUTE_LEFT); + if (!ucontrol->value.integer.value[0]) + c |= ONYX_MUTE_LEFT; + if (!ucontrol->value.integer.value[1]) + c |= ONYX_MUTE_RIGHT; + err = onyx_write_register(onyx, ONYX_REG_DAC_CONTROL, c); + + out_unlock: + mutex_unlock(&onyx->mutex); + + return !err ? (v != c) : err; +} + +static const 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, +}; + + +#define onyx_snd_single_bit_info snd_ctl_boolean_mono_info + +#define FLAG_POLARITY_INVERT 1 +#define FLAG_SPDIFLOCK 2 + +static int onyx_snd_single_bit_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct onyx *onyx = snd_kcontrol_chip(kcontrol); + u8 c; + long int pv = kcontrol->private_value; + u8 polarity = (pv >> 16) & FLAG_POLARITY_INVERT; + u8 address = (pv >> 8) & 0xff; + u8 mask = pv & 0xff; + + mutex_lock(&onyx->mutex); + onyx_read_register(onyx, address, &c); + mutex_unlock(&onyx->mutex); + + ucontrol->value.integer.value[0] = !!(c & mask) ^ polarity; + + return 0; +} + +static int onyx_snd_single_bit_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct onyx *onyx = snd_kcontrol_chip(kcontrol); + u8 v = 0, c = 0; + int err; + long int pv = kcontrol->private_value; + u8 polarity = (pv >> 16) & FLAG_POLARITY_INVERT; + u8 spdiflock = (pv >> 16) & FLAG_SPDIFLOCK; + u8 address = (pv >> 8) & 0xff; + u8 mask = pv & 0xff; + + mutex_lock(&onyx->mutex); + if (spdiflock && onyx->spdif_locked) { + /* even if alsamixer doesn't care.. */ + err = -EBUSY; + goto out_unlock; + } + onyx_read_register(onyx, address, &v); + c = v; + c &= ~(mask); + if (!!ucontrol->value.integer.value[0] ^ polarity) + c |= mask; + err = onyx_write_register(onyx, address, c); + + out_unlock: + mutex_unlock(&onyx->mutex); + + return !err ? (v != c) : err; +} + +#define SINGLE_BIT(n, type, description, address, mask, flags) \ +static struct snd_kcontrol_new n##_control = { \ + .iface = SNDRV_CTL_ELEM_IFACE_##type, \ + .name = description, \ + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, \ + .info = onyx_snd_single_bit_info, \ + .get = onyx_snd_single_bit_get, \ + .put = onyx_snd_single_bit_put, \ + .private_value = (flags << 16) | (address << 8) | mask \ +} + +SINGLE_BIT(spdif, + MIXER, + SNDRV_CTL_NAME_IEC958("", PLAYBACK, SWITCH), + ONYX_REG_DIG_INFO4, + ONYX_SPDIF_ENABLE, + FLAG_SPDIFLOCK); +SINGLE_BIT(ovr1, + MIXER, + "Oversampling Rate", + ONYX_REG_DAC_CONTROL, + ONYX_OVR1, + 0); +SINGLE_BIT(flt0, + MIXER, + "Fast Digital Filter Rolloff", + ONYX_REG_DAC_FILTER, + ONYX_ROLLOFF_FAST, + FLAG_POLARITY_INVERT); +SINGLE_BIT(hpf, + MIXER, + "Highpass Filter", + ONYX_REG_ADC_HPF_BYPASS, + ONYX_HPF_DISABLE, + FLAG_POLARITY_INVERT); +SINGLE_BIT(dm12, + MIXER, + "Digital De-Emphasis", + ONYX_REG_DAC_DEEMPH, + ONYX_DIGDEEMPH_CTRL, + 0); + +static int onyx_spdif_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + uinfo->type = SNDRV_CTL_ELEM_TYPE_IEC958; + uinfo->count = 1; + return 0; +} + +static int onyx_spdif_mask_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + /* datasheet page 30, all others are 0 */ + ucontrol->value.iec958.status[0] = 0x3e; + ucontrol->value.iec958.status[1] = 0xff; + + ucontrol->value.iec958.status[3] = 0x3f; + ucontrol->value.iec958.status[4] = 0x0f; + + return 0; +} + +static const 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 const struct snd_kcontrol_new onyx_spdif_ctrl = { + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, + .iface = SNDRV_CTL_ELEM_IFACE_PCM, + .name = SNDRV_CTL_NAME_IEC958("",PLAYBACK,DEFAULT), + .info = onyx_spdif_info, + .get = onyx_spdif_get, + .put = onyx_spdif_put, +}; + +/* our registers */ + +static u8 register_map[] = { + ONYX_REG_DAC_ATTEN_LEFT, + ONYX_REG_DAC_ATTEN_RIGHT, + ONYX_REG_CONTROL, + ONYX_REG_DAC_CONTROL, + ONYX_REG_DAC_DEEMPH, + ONYX_REG_DAC_FILTER, + ONYX_REG_DAC_OUTPHASE, + ONYX_REG_ADC_CONTROL, + ONYX_REG_ADC_HPF_BYPASS, + ONYX_REG_DIG_INFO1, + ONYX_REG_DIG_INFO2, + ONYX_REG_DIG_INFO3, + ONYX_REG_DIG_INFO4 +}; + +static u8 initial_values[ARRAY_SIZE(register_map)] = { + 0x80, 0x80, /* muted */ + ONYX_MRST | ONYX_SRST, /* but handled specially! */ + ONYX_MUTE_LEFT | ONYX_MUTE_RIGHT, + 0, /* no deemphasis */ + ONYX_DAC_FILTER_ALWAYS, + ONYX_OUTPHASE_INVERTED, + (-1 /*dB*/ + 8) & 0xF, /* line in selected, -1 dB gain*/ + ONYX_ADC_HPF_ALWAYS, + (1<<2), /* pcm audio */ + 2, /* category: pcm coder */ + 0, /* sampling frequency 44.1 kHz, clock accuracy level II */ + 1 /* 24 bit depth */ +}; + +/* reset registers of chip, either to initial or to previous values */ +static int onyx_register_init(struct onyx *onyx) +{ + int i; + u8 val; + u8 regs[sizeof(initial_values)]; + + if (!onyx->initialised) { + memcpy(regs, initial_values, sizeof(initial_values)); + if (onyx_read_register(onyx, ONYX_REG_CONTROL, &val)) + return -1; + val &= ~ONYX_SILICONVERSION; + val |= initial_values[3]; + regs[3] = val; + } else { + for (i=0; i<sizeof(register_map); i++) + regs[i] = onyx->cache[register_map[i]-FIRSTREGISTER]; + } + + for (i=0; i<sizeof(register_map); i++) { + if (onyx_write_register(onyx, register_map[i], regs[i])) + return -1; + } + onyx->initialised = 1; + return 0; +} + +static struct transfer_info onyx_transfers[] = { + /* this is first so we can skip it if no input is present... + * No hardware exists with that, but it's here as an example + * of what to do :) */ + { + /* analog input */ + .formats = SNDRV_PCM_FMTBIT_S8 | + SNDRV_PCM_FMTBIT_S16_BE | + SNDRV_PCM_FMTBIT_S24_BE, + .rates = SNDRV_PCM_RATE_8000_96000, + .transfer_in = 1, + .must_be_clock_source = 0, + .tag = 0, + }, + { + /* if analog and digital are currently off, anything should go, + * so this entry describes everything we can do... */ + .formats = SNDRV_PCM_FMTBIT_S8 | + SNDRV_PCM_FMTBIT_S16_BE | + SNDRV_PCM_FMTBIT_S24_BE +#ifdef SNDRV_PCM_FMTBIT_COMPRESSED_16BE + | SNDRV_PCM_FMTBIT_COMPRESSED_16BE +#endif + , + .rates = SNDRV_PCM_RATE_8000_96000, + .tag = 0, + }, + { + /* analog output */ + .formats = SNDRV_PCM_FMTBIT_S8 | + SNDRV_PCM_FMTBIT_S16_BE | + SNDRV_PCM_FMTBIT_S24_BE, + .rates = SNDRV_PCM_RATE_8000_96000, + .transfer_in = 0, + .must_be_clock_source = 0, + .tag = 1, + }, + { + /* digital pcm output, also possible for analog out */ + .formats = SNDRV_PCM_FMTBIT_S8 | + SNDRV_PCM_FMTBIT_S16_BE | + SNDRV_PCM_FMTBIT_S24_BE, + .rates = SNDRV_PCM_RATE_32000 | + SNDRV_PCM_RATE_44100 | + SNDRV_PCM_RATE_48000, + .transfer_in = 0, + .must_be_clock_source = 0, + .tag = 2, + }, +#ifdef SNDRV_PCM_FMTBIT_COMPRESSED_16BE + /* Once alsa gets supports for this kind of thing we can add it... */ + { + /* digital compressed output */ + .formats = SNDRV_PCM_FMTBIT_COMPRESSED_16BE, + .rates = SNDRV_PCM_RATE_32000 | + SNDRV_PCM_RATE_44100 | + SNDRV_PCM_RATE_48000, + .tag = 2, + }, +#endif + {} +}; + +static int onyx_usable(struct codec_info_item *cii, + struct transfer_info *ti, + struct transfer_info *out) +{ + u8 v; + struct onyx *onyx = cii->codec_data; + int spdif_enabled, analog_enabled; + + mutex_lock(&onyx->mutex); + onyx_read_register(onyx, ONYX_REG_DIG_INFO4, &v); + spdif_enabled = !!(v & ONYX_SPDIF_ENABLE); + onyx_read_register(onyx, ONYX_REG_DAC_CONTROL, &v); + analog_enabled = + (v & (ONYX_MUTE_RIGHT|ONYX_MUTE_LEFT)) + != (ONYX_MUTE_RIGHT|ONYX_MUTE_LEFT); + mutex_unlock(&onyx->mutex); + + switch (ti->tag) { + case 0: return 1; + case 1: return analog_enabled; + case 2: return spdif_enabled; + } + return 1; +} + +static int onyx_prepare(struct codec_info_item *cii, + struct bus_info *bi, + struct snd_pcm_substream *substream) +{ + u8 v; + struct onyx *onyx = cii->codec_data; + int err = -EBUSY; + + mutex_lock(&onyx->mutex); + +#ifdef SNDRV_PCM_FMTBIT_COMPRESSED_16BE + if (substream->runtime->format == SNDRV_PCM_FMTBIT_COMPRESSED_16BE) { + /* mute and lock analog output */ + onyx_read_register(onyx, ONYX_REG_DAC_CONTROL, &v); + if (onyx_write_register(onyx, + ONYX_REG_DAC_CONTROL, + v | ONYX_MUTE_RIGHT | ONYX_MUTE_LEFT)) + goto out_unlock; + onyx->analog_locked = 1; + err = 0; + goto out_unlock; + } +#endif + switch (substream->runtime->rate) { + case 32000: + case 44100: + case 48000: + /* these rates are ok for all outputs */ + /* FIXME: program spdif channel control bits here so that + * userspace doesn't have to if it only plays pcm! */ + err = 0; + goto out_unlock; + default: + /* got some rate that the digital output can't do, + * so disable and lock it */ + onyx_read_register(cii->codec_data, ONYX_REG_DIG_INFO4, &v); + if (onyx_write_register(onyx, + ONYX_REG_DIG_INFO4, + v & ~ONYX_SPDIF_ENABLE)) + goto out_unlock; + onyx->spdif_locked = 1; + err = 0; + goto out_unlock; + } + + out_unlock: + mutex_unlock(&onyx->mutex); + + return err; +} + +static int onyx_open(struct codec_info_item *cii, + struct snd_pcm_substream *substream) +{ + struct onyx *onyx = cii->codec_data; + + mutex_lock(&onyx->mutex); + onyx->open_count++; + mutex_unlock(&onyx->mutex); + + return 0; +} + +static int onyx_close(struct codec_info_item *cii, + struct snd_pcm_substream *substream) +{ + struct onyx *onyx = cii->codec_data; + + mutex_lock(&onyx->mutex); + onyx->open_count--; + if (!onyx->open_count) + onyx->spdif_locked = onyx->analog_locked = 0; + mutex_unlock(&onyx->mutex); + + return 0; +} + +static int onyx_switch_clock(struct codec_info_item *cii, + enum clock_switch what) +{ + struct onyx *onyx = cii->codec_data; + + mutex_lock(&onyx->mutex); + /* this *MUST* be more elaborate later... */ + switch (what) { + case CLOCK_SWITCH_PREPARE_SLAVE: + onyx->codec.gpio->methods->all_amps_off(onyx->codec.gpio); + break; + case CLOCK_SWITCH_SLAVE: + onyx->codec.gpio->methods->all_amps_restore(onyx->codec.gpio); + break; + default: /* silence warning */ + break; + } + mutex_unlock(&onyx->mutex); + + return 0; +} + +#ifdef CONFIG_PM + +static int onyx_suspend(struct codec_info_item *cii, pm_message_t state) +{ + struct onyx *onyx = cii->codec_data; + u8 v; + int err = -ENXIO; + + mutex_lock(&onyx->mutex); + if (onyx_read_register(onyx, ONYX_REG_CONTROL, &v)) + goto out_unlock; + onyx_write_register(onyx, ONYX_REG_CONTROL, v | ONYX_ADPSV | ONYX_DAPSV); + /* Apple does a sleep here but the datasheet says to do it on resume */ + err = 0; + out_unlock: + mutex_unlock(&onyx->mutex); + + return err; +} + +static int onyx_resume(struct codec_info_item *cii) +{ + struct onyx *onyx = cii->codec_data; + u8 v; + int err = -ENXIO; + + mutex_lock(&onyx->mutex); + + /* reset codec */ + 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); + + /* take codec out of suspend (if it still is after reset) */ + if (onyx_read_register(onyx, ONYX_REG_CONTROL, &v)) + goto out_unlock; + onyx_write_register(onyx, ONYX_REG_CONTROL, v & ~(ONYX_ADPSV | ONYX_DAPSV)); + /* FIXME: should divide by sample rate, but 8k is the lowest we go */ + msleep(2205000/8000); + /* reset all values */ + onyx_register_init(onyx); + err = 0; + out_unlock: + mutex_unlock(&onyx->mutex); + + return err; +} + +#endif /* CONFIG_PM */ + +static struct codec_info onyx_codec_info = { + .transfers = onyx_transfers, + .sysclock_factor = 256, + .bus_factor = 64, + .owner = THIS_MODULE, + .usable = onyx_usable, + .prepare = onyx_prepare, + .open = onyx_open, + .close = onyx_close, + .switch_clock = onyx_switch_clock, +#ifdef CONFIG_PM + .suspend = onyx_suspend, + .resume = onyx_resume, +#endif +}; + +static int onyx_init_codec(struct aoa_codec *codec) +{ + struct onyx *onyx = codec_to_onyx(codec); + struct snd_kcontrol *ctl; + struct codec_info *ci = &onyx_codec_info; + u8 v; + int err; + + if (!onyx->codec.gpio || !onyx->codec.gpio->methods) { + printk(KERN_ERR PFX "gpios not assigned!!\n"); + return -EINVAL; + } + + onyx->codec.gpio->methods->set_hw_reset(onyx->codec.gpio, 0); + msleep(1); + onyx->codec.gpio->methods->set_hw_reset(onyx->codec.gpio, 1); + msleep(1); + onyx->codec.gpio->methods->set_hw_reset(onyx->codec.gpio, 0); + msleep(1); + + if (onyx_register_init(onyx)) { + printk(KERN_ERR PFX "failed to initialise onyx registers\n"); + return -ENODEV; + } + + if (aoa_snd_device_new(SNDRV_DEV_CODEC, onyx, &ops)) { + printk(KERN_ERR PFX "failed to create onyx snd device!\n"); + return -ENODEV; + } + + /* nothing connected? what a joke! */ + if ((onyx->codec.connected & 0xF) == 0) + return -ENOTCONN; + + /* if no inputs are present... */ + if ((onyx->codec.connected & 0xC) == 0) { + if (!onyx->codec_info) + onyx->codec_info = kmalloc(sizeof(struct codec_info), GFP_KERNEL); + if (!onyx->codec_info) + return -ENOMEM; + ci = onyx->codec_info; + *ci = onyx_codec_info; + ci->transfers++; + } + + /* if no outputs are present... */ + if ((onyx->codec.connected & 3) == 0) { + if (!onyx->codec_info) + onyx->codec_info = kmalloc(sizeof(struct codec_info), GFP_KERNEL); + if (!onyx->codec_info) + return -ENOMEM; + ci = onyx->codec_info; + /* this is fine as there have to be inputs + * if we end up in this part of the code */ + *ci = onyx_codec_info; + ci->transfers[1].formats = 0; + } + + if (onyx->codec.soundbus_dev->attach_codec(onyx->codec.soundbus_dev, + aoa_get_card(), + ci, onyx)) { + printk(KERN_ERR PFX "error creating onyx pcm\n"); + return -ENODEV; + } +#define ADDCTL(n) \ + do { \ + ctl = snd_ctl_new1(&n, onyx); \ + if (ctl) { \ + ctl->id.device = \ + onyx->codec.soundbus_dev->pcm->device; \ + err = aoa_snd_ctl_add(ctl); \ + if (err) \ + goto error; \ + } \ + } while (0) + + if (onyx->codec.soundbus_dev->pcm) { + /* give the user appropriate controls + * depending on what inputs are connected */ + if ((onyx->codec.connected & 0xC) == 0xC) + ADDCTL(capture_source_control); + else if (onyx->codec.connected & 4) + onyx_set_capture_source(onyx, 0); + else + onyx_set_capture_source(onyx, 1); + if (onyx->codec.connected & 0xC) + ADDCTL(inputgain_control); + + /* depending on what output is connected, + * give the user appropriate controls */ + if (onyx->codec.connected & 1) { + ADDCTL(volume_control); + ADDCTL(mute_control); + ADDCTL(ovr1_control); + ADDCTL(flt0_control); + ADDCTL(hpf_control); + ADDCTL(dm12_control); + /* spdif control defaults to off */ + } + if (onyx->codec.connected & 2) { + ADDCTL(onyx_spdif_mask); + ADDCTL(onyx_spdif_ctrl); + } + if ((onyx->codec.connected & 3) == 3) + ADDCTL(spdif_control); + /* if only S/PDIF is connected, enable it unconditionally */ + if ((onyx->codec.connected & 3) == 2) { + onyx_read_register(onyx, ONYX_REG_DIG_INFO4, &v); + v |= ONYX_SPDIF_ENABLE; + onyx_write_register(onyx, ONYX_REG_DIG_INFO4, v); + } + } +#undef ADDCTL + printk(KERN_INFO PFX "attached to onyx codec via i2c\n"); + + return 0; + error: + onyx->codec.soundbus_dev->detach_codec(onyx->codec.soundbus_dev, onyx); + snd_device_free(aoa_get_card(), onyx); + return err; +} + +static void onyx_exit_codec(struct aoa_codec *codec) +{ + struct onyx *onyx = codec_to_onyx(codec); + + if (!onyx->codec.soundbus_dev) { + printk(KERN_ERR PFX "onyx_exit_codec called without soundbus_dev!\n"); + return; + } + onyx->codec.soundbus_dev->detach_codec(onyx->codec.soundbus_dev, onyx); +} + +static int onyx_i2c_probe(struct i2c_client *client, + const struct i2c_device_id *id) +{ + struct device_node *node = client->dev.of_node; + struct onyx *onyx; + u8 dummy; + + onyx = kzalloc(sizeof(struct onyx), GFP_KERNEL); + + if (!onyx) + return -ENOMEM; + + mutex_init(&onyx->mutex); + onyx->i2c = client; + i2c_set_clientdata(client, onyx); + + /* 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) { + printk(KERN_ERR PFX "failed to read control register\n"); + goto fail; + } + + strlcpy(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)) { + goto fail; + } + printk(KERN_DEBUG PFX "created and attached onyx instance\n"); + return 0; + fail: + kfree(onyx); + return -ENODEV; +} + +static int onyx_i2c_remove(struct i2c_client *client) +{ + struct onyx *onyx = i2c_get_clientdata(client); + + aoa_codec_unregister(&onyx->codec); + of_node_put(onyx->codec.node); + kfree(onyx->codec_info); + kfree(onyx); + return 0; +} + +static const struct i2c_device_id onyx_i2c_id[] = { + { "MAC,pcm3052", 0 }, + { } +}; +MODULE_DEVICE_TABLE(i2c,onyx_i2c_id); + +static struct i2c_driver onyx_driver = { + .driver = { + .name = "aoa_codec_onyx", + }, + .probe = onyx_i2c_probe, + .remove = onyx_i2c_remove, + .id_table = onyx_i2c_id, +}; + +module_i2c_driver(onyx_driver); diff --git a/sound/aoa/codecs/onyx.h b/sound/aoa/codecs/onyx.h new file mode 100644 index 000000000..ffd20254f --- /dev/null +++ b/sound/aoa/codecs/onyx.h @@ -0,0 +1,75 @@ +/* + * 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 <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 */ diff --git a/sound/aoa/codecs/tas-basstreble.h b/sound/aoa/codecs/tas-basstreble.h new file mode 100644 index 000000000..770935af6 --- /dev/null +++ b/sound/aoa/codecs/tas-basstreble.h @@ -0,0 +1,135 @@ +/* SPDX-License-Identifier: GPL-2.0 */ +/* + * This file is only included exactly once! + * + * The tables here are derived from the tas3004 datasheet, + * modulo typo corrections and some smoothing... + */ + +#define TAS3004_TREBLE_MIN 0 +#define TAS3004_TREBLE_MAX 72 +#define TAS3004_BASS_MIN 0 +#define TAS3004_BASS_MAX 72 +#define TAS3004_TREBLE_ZERO 36 +#define TAS3004_BASS_ZERO 36 + +static u8 tas3004_treble_table[] = { + 150, /* -18 dB */ + 149, + 148, + 147, + 146, + 145, + 144, + 143, + 142, + 141, + 140, + 139, + 138, + 137, + 136, + 135, + 134, + 133, + 132, + 131, + 130, + 129, + 128, + 127, + 126, + 125, + 124, + 123, + 122, + 121, + 120, + 119, + 118, + 117, + 116, + 115, + 114, /* 0 dB */ + 113, + 112, + 111, + 109, + 108, + 107, + 105, + 104, + 103, + 101, + 99, + 98, + 96, + 93, + 91, + 89, + 86, + 83, + 81, + 77, + 74, + 71, + 67, + 63, + 59, + 54, + 49, + 44, + 38, + 32, + 26, + 19, + 10, + 4, + 2, + 1, /* +18 dB */ +}; + +static inline u8 tas3004_treble(int idx) +{ + return tas3004_treble_table[idx]; +} + +/* I only save the difference here to the treble table + * so that the binary is smaller... + * I have also ignored completely differences of + * +/- 1 + */ +static s8 tas3004_bass_diff_to_treble[] = { + 2, /* 7 dB, offset 50 */ + 2, + 2, + 2, + 2, + 1, + 2, + 2, + 2, + 3, + 4, + 4, + 5, + 6, + 7, + 8, + 9, + 10, + 11, + 14, + 13, + 8, + 1, /* 18 dB */ +}; + +static inline u8 tas3004_bass(int idx) +{ + u8 result = tas3004_treble_table[idx]; + + if (idx >= 50) + result += tas3004_bass_diff_to_treble[idx-50]; + return result; +} diff --git a/sound/aoa/codecs/tas-gain-table.h b/sound/aoa/codecs/tas-gain-table.h new file mode 100644 index 000000000..77b8e7dc5 --- /dev/null +++ b/sound/aoa/codecs/tas-gain-table.h @@ -0,0 +1,210 @@ +/* SPDX-License-Identifier: GPL-2.0 */ +/* + 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 */ +}; + diff --git a/sound/aoa/codecs/tas.c b/sound/aoa/codecs/tas.c new file mode 100644 index 000000000..15c05755d --- /dev/null +++ b/sound/aoa/codecs/tas.c @@ -0,0 +1,948 @@ +/* + * 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 <asm/pmac_low_i2c.h> +#include <asm/prom.h> +#include <linux/delay.h> +#include <linux/module.h> +#include <linux/mutex.h> +#include <linux/slab.h> + +MODULE_AUTHOR("Johannes Berg <johannes@sipsolutions.net>"); +MODULE_LICENSE("GPL"); +MODULE_DESCRIPTION("tas codec driver for snd-aoa"); + +#include "tas.h" +#include "tas-gain-table.h" +#include "tas-basstreble.h" +#include "../aoa.h" +#include "../soundbus/soundbus.h" + +#define PFX "snd-aoa-codec-tas: " + + +struct tas { + struct aoa_codec codec; + struct i2c_client *i2c; + u32 mute_l:1, mute_r:1 , + controls_created:1 , + drc_enabled:1, + hw_enabled:1; + u8 cached_volume_l, cached_volume_r; + u8 mixer_l[3], mixer_r[3]; + u8 bass, treble; + u8 acr; + int drc_range; + /* protects hardware access against concurrency from + * userspace when hitting controls and during + * codec init/suspend/resume */ + struct mutex mtx; +}; + +static int tas_reset_init(struct tas *tas); + +static struct tas *codec_to_tas(struct aoa_codec *codec) +{ + return container_of(codec, struct tas, codec); +} + +static inline int tas_write_reg(struct tas *tas, u8 reg, u8 len, u8 *data) +{ + if (len == 1) + return i2c_smbus_write_byte_data(tas->i2c, reg, *data); + else + return i2c_smbus_write_i2c_block_data(tas->i2c, reg, len, data); +} + +static void tas3004_set_drc(struct tas *tas) +{ + unsigned char val[6]; + + if (tas->drc_enabled) + val[0] = 0x50; /* 3:1 above threshold */ + else + val[0] = 0x51; /* disabled */ + val[1] = 0x02; /* 1:1 below threshold */ + if (tas->drc_range > 0xef) + val[2] = 0xef; + else if (tas->drc_range < 0) + val[2] = 0x00; + else + val[2] = tas->drc_range; + val[3] = 0xb0; + val[4] = 0x60; + val[5] = 0xa0; + + tas_write_reg(tas, TAS_REG_DRC, 6, val); +} + +static void tas_set_treble(struct tas *tas) +{ + u8 tmp; + + tmp = tas3004_treble(tas->treble); + tas_write_reg(tas, TAS_REG_TREBLE, 1, &tmp); +} + +static void tas_set_bass(struct tas *tas) +{ + u8 tmp; + + tmp = tas3004_bass(tas->bass); + tas_write_reg(tas, TAS_REG_BASS, 1, &tmp); +} + +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->mute_l) left = 0; + if (tas->mute_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); + + mutex_lock(&tas->mtx); + ucontrol->value.integer.value[0] = tas->cached_volume_l; + ucontrol->value.integer.value[1] = tas->cached_volume_r; + mutex_unlock(&tas->mtx); + return 0; +} + +static int tas_snd_vol_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct tas *tas = snd_kcontrol_chip(kcontrol); + + if (ucontrol->value.integer.value[0] < 0 || + ucontrol->value.integer.value[0] > 177) + return -EINVAL; + if (ucontrol->value.integer.value[1] < 0 || + ucontrol->value.integer.value[1] > 177) + return -EINVAL; + + mutex_lock(&tas->mtx); + if (tas->cached_volume_l == ucontrol->value.integer.value[0] + && tas->cached_volume_r == ucontrol->value.integer.value[1]) { + mutex_unlock(&tas->mtx); + return 0; + } + + tas->cached_volume_l = ucontrol->value.integer.value[0]; + tas->cached_volume_r = ucontrol->value.integer.value[1]; + if (tas->hw_enabled) + tas_set_volume(tas); + mutex_unlock(&tas->mtx); + return 1; +} + +static const 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, +}; + +#define tas_snd_mute_info snd_ctl_boolean_stereo_info + +static int tas_snd_mute_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct tas *tas = snd_kcontrol_chip(kcontrol); + + mutex_lock(&tas->mtx); + ucontrol->value.integer.value[0] = !tas->mute_l; + ucontrol->value.integer.value[1] = !tas->mute_r; + mutex_unlock(&tas->mtx); + 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); + + mutex_lock(&tas->mtx); + if (tas->mute_l == !ucontrol->value.integer.value[0] + && tas->mute_r == !ucontrol->value.integer.value[1]) { + mutex_unlock(&tas->mtx); + return 0; + } + + tas->mute_l = !ucontrol->value.integer.value[0]; + tas->mute_r = !ucontrol->value.integer.value[1]; + if (tas->hw_enabled) + tas_set_volume(tas); + mutex_unlock(&tas->mtx); + return 1; +} + +static const 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; + + mutex_lock(&tas->mtx); + ucontrol->value.integer.value[0] = tas->mixer_l[idx]; + ucontrol->value.integer.value[1] = tas->mixer_r[idx]; + mutex_unlock(&tas->mtx); + + 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; + + mutex_lock(&tas->mtx); + if (tas->mixer_l[idx] == ucontrol->value.integer.value[0] + && tas->mixer_r[idx] == ucontrol->value.integer.value[1]) { + mutex_unlock(&tas->mtx); + return 0; + } + + tas->mixer_l[idx] = ucontrol->value.integer.value[0]; + tas->mixer_r[idx] = ucontrol->value.integer.value[1]; + + if (tas->hw_enabled) + tas_set_mixer(tas); + mutex_unlock(&tas->mtx); + 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, "PCM", 0); +MIXER_CONTROL(monitor, "Monitor", 2); + +static int tas_snd_drc_range_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 = 0; + uinfo->value.integer.max = TAS3004_DRC_MAX; + return 0; +} + +static int tas_snd_drc_range_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct tas *tas = snd_kcontrol_chip(kcontrol); + + mutex_lock(&tas->mtx); + ucontrol->value.integer.value[0] = tas->drc_range; + mutex_unlock(&tas->mtx); + return 0; +} + +static int tas_snd_drc_range_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct tas *tas = snd_kcontrol_chip(kcontrol); + + if (ucontrol->value.integer.value[0] < 0 || + ucontrol->value.integer.value[0] > TAS3004_DRC_MAX) + return -EINVAL; + + mutex_lock(&tas->mtx); + if (tas->drc_range == ucontrol->value.integer.value[0]) { + mutex_unlock(&tas->mtx); + return 0; + } + + tas->drc_range = ucontrol->value.integer.value[0]; + if (tas->hw_enabled) + tas3004_set_drc(tas); + mutex_unlock(&tas->mtx); + return 1; +} + +static const struct snd_kcontrol_new drc_range_control = { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "DRC Range", + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, + .info = tas_snd_drc_range_info, + .get = tas_snd_drc_range_get, + .put = tas_snd_drc_range_put, +}; + +#define tas_snd_drc_switch_info snd_ctl_boolean_mono_info + +static int tas_snd_drc_switch_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct tas *tas = snd_kcontrol_chip(kcontrol); + + mutex_lock(&tas->mtx); + ucontrol->value.integer.value[0] = tas->drc_enabled; + mutex_unlock(&tas->mtx); + return 0; +} + +static int tas_snd_drc_switch_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct tas *tas = snd_kcontrol_chip(kcontrol); + + mutex_lock(&tas->mtx); + if (tas->drc_enabled == ucontrol->value.integer.value[0]) { + mutex_unlock(&tas->mtx); + return 0; + } + + tas->drc_enabled = !!ucontrol->value.integer.value[0]; + if (tas->hw_enabled) + tas3004_set_drc(tas); + mutex_unlock(&tas->mtx); + return 1; +} + +static const struct snd_kcontrol_new drc_switch_control = { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "DRC Range Switch", + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, + .info = tas_snd_drc_switch_info, + .get = tas_snd_drc_switch_get, + .put = tas_snd_drc_switch_put, +}; + +static int tas_snd_capture_source_info(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_info *uinfo) +{ + static const char * const texts[] = { "Line-In", "Microphone" }; + + return snd_ctl_enum_info(uinfo, 1, 2, texts); +} + +static int tas_snd_capture_source_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct tas *tas = snd_kcontrol_chip(kcontrol); + + mutex_lock(&tas->mtx); + ucontrol->value.enumerated.item[0] = !!(tas->acr & TAS_ACR_INPUT_B); + mutex_unlock(&tas->mtx); + return 0; +} + +static int tas_snd_capture_source_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct tas *tas = snd_kcontrol_chip(kcontrol); + int oldacr; + + if (ucontrol->value.enumerated.item[0] > 1) + return -EINVAL; + mutex_lock(&tas->mtx); + oldacr = tas->acr; + + /* + * Despite what the data sheet says in one place, the + * TAS_ACR_B_MONAUREAL bit forces mono output even when + * input A (line in) is selected. + */ + tas->acr &= ~(TAS_ACR_INPUT_B | TAS_ACR_B_MONAUREAL); + if (ucontrol->value.enumerated.item[0]) + tas->acr |= TAS_ACR_INPUT_B | TAS_ACR_B_MONAUREAL | + TAS_ACR_B_MON_SEL_RIGHT; + if (oldacr == tas->acr) { + mutex_unlock(&tas->mtx); + return 0; + } + if (tas->hw_enabled) + tas_write_reg(tas, TAS_REG_ACR, 1, &tas->acr); + mutex_unlock(&tas->mtx); + return 1; +} + +static const 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 int tas_snd_treble_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 = TAS3004_TREBLE_MIN; + uinfo->value.integer.max = TAS3004_TREBLE_MAX; + return 0; +} + +static int tas_snd_treble_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct tas *tas = snd_kcontrol_chip(kcontrol); + + mutex_lock(&tas->mtx); + ucontrol->value.integer.value[0] = tas->treble; + mutex_unlock(&tas->mtx); + return 0; +} + +static int tas_snd_treble_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct tas *tas = snd_kcontrol_chip(kcontrol); + + if (ucontrol->value.integer.value[0] < TAS3004_TREBLE_MIN || + ucontrol->value.integer.value[0] > TAS3004_TREBLE_MAX) + return -EINVAL; + mutex_lock(&tas->mtx); + if (tas->treble == ucontrol->value.integer.value[0]) { + mutex_unlock(&tas->mtx); + return 0; + } + + tas->treble = ucontrol->value.integer.value[0]; + if (tas->hw_enabled) + tas_set_treble(tas); + mutex_unlock(&tas->mtx); + return 1; +} + +static const struct snd_kcontrol_new treble_control = { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Treble", + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, + .info = tas_snd_treble_info, + .get = tas_snd_treble_get, + .put = tas_snd_treble_put, +}; + +static int tas_snd_bass_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 = TAS3004_BASS_MIN; + uinfo->value.integer.max = TAS3004_BASS_MAX; + return 0; +} + +static int tas_snd_bass_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct tas *tas = snd_kcontrol_chip(kcontrol); + + mutex_lock(&tas->mtx); + ucontrol->value.integer.value[0] = tas->bass; + mutex_unlock(&tas->mtx); + return 0; +} + +static int tas_snd_bass_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct tas *tas = snd_kcontrol_chip(kcontrol); + + if (ucontrol->value.integer.value[0] < TAS3004_BASS_MIN || + ucontrol->value.integer.value[0] > TAS3004_BASS_MAX) + return -EINVAL; + mutex_lock(&tas->mtx); + if (tas->bass == ucontrol->value.integer.value[0]) { + mutex_unlock(&tas->mtx); + return 0; + } + + tas->bass = ucontrol->value.integer.value[0]; + if (tas->hw_enabled) + tas_set_bass(tas); + mutex_unlock(&tas->mtx); + return 1; +} + +static const struct snd_kcontrol_new bass_control = { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Bass", + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, + .info = tas_snd_bass_info, + .get = tas_snd_bass_get, + .put = tas_snd_bass_put, +}; + +static struct transfer_info tas_transfers[] = { + { + /* input */ + .formats = 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 = 1, + }, + { + /* output */ + .formats = 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, + }, + {} +}; + +static int tas_usable(struct codec_info_item *cii, + struct transfer_info *ti, + struct transfer_info *out) +{ + return 1; +} + +static int tas_reset_init(struct tas *tas) +{ + u8 tmp; + + tas->codec.gpio->methods->all_amps_off(tas->codec.gpio); + msleep(5); + tas->codec.gpio->methods->set_hw_reset(tas->codec.gpio, 0); + msleep(5); + tas->codec.gpio->methods->set_hw_reset(tas->codec.gpio, 1); + msleep(20); + tas->codec.gpio->methods->set_hw_reset(tas->codec.gpio, 0); + msleep(10); + tas->codec.gpio->methods->all_amps_restore(tas->codec.gpio); + + tmp = TAS_MCS_SCLK64 | TAS_MCS_SPORT_MODE_I2S | TAS_MCS_SPORT_WL_24BIT; + if (tas_write_reg(tas, TAS_REG_MCS, 1, &tmp)) + goto outerr; + + tas->acr |= TAS_ACR_ANALOG_PDOWN; + if (tas_write_reg(tas, TAS_REG_ACR, 1, &tas->acr)) + goto outerr; + + tmp = 0; + if (tas_write_reg(tas, TAS_REG_MCS2, 1, &tmp)) + goto outerr; + + tas3004_set_drc(tas); + + /* Set treble & bass to 0dB */ + tas->treble = TAS3004_TREBLE_ZERO; + tas->bass = TAS3004_BASS_ZERO; + tas_set_treble(tas); + tas_set_bass(tas); + + tas->acr &= ~TAS_ACR_ANALOG_PDOWN; + if (tas_write_reg(tas, TAS_REG_ACR, 1, &tas->acr)) + goto outerr; + + return 0; + outerr: + return -ENODEV; +} + +static int tas_switch_clock(struct codec_info_item *cii, enum clock_switch clock) +{ + struct tas *tas = cii->codec_data; + + switch(clock) { + case CLOCK_SWITCH_PREPARE_SLAVE: + /* Clocks are going away, mute mute mute */ + tas->codec.gpio->methods->all_amps_off(tas->codec.gpio); + tas->hw_enabled = 0; + break; + case CLOCK_SWITCH_SLAVE: + /* Clocks are back, re-init the codec */ + mutex_lock(&tas->mtx); + tas_reset_init(tas); + tas_set_volume(tas); + tas_set_mixer(tas); + tas->hw_enabled = 1; + tas->codec.gpio->methods->all_amps_restore(tas->codec.gpio); + mutex_unlock(&tas->mtx); + break; + default: + /* doesn't happen as of now */ + return -EINVAL; + } + return 0; +} + +#ifdef CONFIG_PM +/* 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) +{ + mutex_lock(&tas->mtx); + tas->hw_enabled = 0; + tas->acr |= TAS_ACR_ANALOG_PDOWN; + tas_write_reg(tas, TAS_REG_ACR, 1, &tas->acr); + mutex_unlock(&tas->mtx); + return 0; +} + +static int tas_resume(struct tas *tas) +{ + /* reset codec */ + mutex_lock(&tas->mtx); + tas_reset_init(tas); + tas_set_volume(tas); + tas_set_mixer(tas); + tas->hw_enabled = 1; + mutex_unlock(&tas->mtx); + return 0; +} + +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); +} +#else /* CONFIG_PM */ +#define _tas_suspend NULL +#define _tas_resume NULL +#endif /* CONFIG_PM */ + +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, + .switch_clock = tas_switch_clock, + .suspend = _tas_suspend, + .resume = _tas_resume, +}; + +static int tas_init_codec(struct aoa_codec *codec) +{ + struct tas *tas = codec_to_tas(codec); + int err; + + if (!tas->codec.gpio || !tas->codec.gpio->methods) { + printk(KERN_ERR PFX "gpios not assigned!!\n"); + return -EINVAL; + } + + mutex_lock(&tas->mtx); + if (tas_reset_init(tas)) { + printk(KERN_ERR PFX "tas failed to initialise\n"); + mutex_unlock(&tas->mtx); + return -ENXIO; + } + tas->hw_enabled = 1; + mutex_unlock(&tas->mtx); + + 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_CODEC, tas, &ops)) { + printk(KERN_ERR PFX "failed to create tas snd device!\n"); + return -ENODEV; + } + err = aoa_snd_ctl_add(snd_ctl_new1(&volume_control, tas)); + if (err) + goto error; + + err = aoa_snd_ctl_add(snd_ctl_new1(&mute_control, tas)); + if (err) + goto error; + + err = aoa_snd_ctl_add(snd_ctl_new1(&pcm1_control, tas)); + if (err) + goto error; + + err = aoa_snd_ctl_add(snd_ctl_new1(&monitor_control, tas)); + if (err) + goto error; + + err = aoa_snd_ctl_add(snd_ctl_new1(&capture_source_control, tas)); + if (err) + goto error; + + err = aoa_snd_ctl_add(snd_ctl_new1(&drc_range_control, tas)); + if (err) + goto error; + + err = aoa_snd_ctl_add(snd_ctl_new1(&drc_switch_control, tas)); + if (err) + goto error; + + err = aoa_snd_ctl_add(snd_ctl_new1(&treble_control, tas)); + if (err) + goto error; + + err = aoa_snd_ctl_add(snd_ctl_new1(&bass_control, tas)); + if (err) + goto error; + + return 0; + error: + tas->codec.soundbus_dev->detach_codec(tas->codec.soundbus_dev, tas); + snd_device_free(aoa_get_card(), tas); + return err; +} + +static void tas_exit_codec(struct aoa_codec *codec) +{ + struct tas *tas = codec_to_tas(codec); + + if (!tas->codec.soundbus_dev) + return; + tas->codec.soundbus_dev->detach_codec(tas->codec.soundbus_dev, tas); +} + + +static int tas_i2c_probe(struct i2c_client *client, + const struct i2c_device_id *id) +{ + struct device_node *node = client->dev.of_node; + struct tas *tas; + + tas = kzalloc(sizeof(struct tas), GFP_KERNEL); + + if (!tas) + return -ENOMEM; + + mutex_init(&tas->mtx); + tas->i2c = client; + i2c_set_clientdata(client, tas); + + /* seems that half is a saner default */ + tas->drc_range = TAS3004_DRC_MAX / 2; + + strlcpy(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 fail; + } + printk(KERN_DEBUG + "snd-aoa-codec-tas: tas found, addr 0x%02x on %pOF\n", + (unsigned int)client->addr, node); + return 0; + fail: + mutex_destroy(&tas->mtx); + kfree(tas); + return -EINVAL; +} + +static int tas_i2c_remove(struct i2c_client *client) +{ + struct tas *tas = i2c_get_clientdata(client); + u8 tmp = TAS_ACR_ANALOG_PDOWN; + + aoa_codec_unregister(&tas->codec); + of_node_put(tas->codec.node); + + /* power down codec chip */ + tas_write_reg(tas, TAS_REG_ACR, 1, &tmp); + + mutex_destroy(&tas->mtx); + kfree(tas); + return 0; +} + +static const struct i2c_device_id tas_i2c_id[] = { + { "MAC,tas3004", 0 }, + { } +}; +MODULE_DEVICE_TABLE(i2c,tas_i2c_id); + +static struct i2c_driver tas_driver = { + .driver = { + .name = "aoa_codec_tas", + }, + .probe = tas_i2c_probe, + .remove = tas_i2c_remove, + .id_table = tas_i2c_id, +}; + +module_i2c_driver(tas_driver); diff --git a/sound/aoa/codecs/tas.h b/sound/aoa/codecs/tas.h new file mode 100644 index 000000000..ae177e346 --- /dev/null +++ b/sound/aoa/codecs/tas.h @@ -0,0 +1,55 @@ +/* + * 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 + +#define TAS_REG_LEFT_LOUDNESS 0x21 +#define TAS_REG_RIGHT_LOUDNESS 0x22 +#define TAS_REG_LEFT_LOUDNESS_GAIN 0x23 +#define TAS_REG_RIGHT_LOUDNESS_GAIN 0x24 + +#define TAS3001_DRC_MAX 0x5f +#define TAS3004_DRC_MAX 0xef + +#endif /* __SND_AOA_CODECTASH */ diff --git a/sound/aoa/codecs/toonie.c b/sound/aoa/codecs/toonie.c new file mode 100644 index 000000000..7e8c3417c --- /dev/null +++ b/sound/aoa/codecs/toonie.c @@ -0,0 +1,151 @@ +/* + * Apple Onboard Audio driver for Toonie codec + * + * Copyright 2006 Johannes Berg <johannes@sipsolutions.net> + * + * GPL v2, can be found in COPYING. + * + * + * This is a driver for the toonie codec chip. This chip is present + * on the Mac Mini and is nothing but a DAC. + */ +#include <linux/delay.h> +#include <linux/module.h> +#include <linux/slab.h> +MODULE_AUTHOR("Johannes Berg <johannes@sipsolutions.net>"); +MODULE_LICENSE("GPL"); +MODULE_DESCRIPTION("toonie codec driver for snd-aoa"); + +#include "../aoa.h" +#include "../soundbus/soundbus.h" + + +#define PFX "snd-aoa-codec-toonie: " + +struct toonie { + struct aoa_codec codec; +}; +#define codec_to_toonie(c) container_of(c, struct toonie, codec) + +static int toonie_dev_register(struct snd_device *dev) +{ + return 0; +} + +static struct snd_device_ops ops = { + .dev_register = toonie_dev_register, +}; + +static struct transfer_info toonie_transfers[] = { + /* This thing *only* has analog output, + * the rates are taken from Info.plist + * from Darwin. */ + { + .formats = SNDRV_PCM_FMTBIT_S16_BE | + SNDRV_PCM_FMTBIT_S24_BE, + .rates = SNDRV_PCM_RATE_32000 | + SNDRV_PCM_RATE_44100 | + SNDRV_PCM_RATE_48000 | + SNDRV_PCM_RATE_88200 | + SNDRV_PCM_RATE_96000, + }, + {} +}; + +static int toonie_usable(struct codec_info_item *cii, + struct transfer_info *ti, + struct transfer_info *out) +{ + return 1; +} + +#ifdef CONFIG_PM +static int toonie_suspend(struct codec_info_item *cii, pm_message_t state) +{ + /* can we turn it off somehow? */ + return 0; +} + +static int toonie_resume(struct codec_info_item *cii) +{ + return 0; +} +#endif /* CONFIG_PM */ + +static struct codec_info toonie_codec_info = { + .transfers = toonie_transfers, + .sysclock_factor = 256, + .bus_factor = 64, + .owner = THIS_MODULE, + .usable = toonie_usable, +#ifdef CONFIG_PM + .suspend = toonie_suspend, + .resume = toonie_resume, +#endif +}; + +static int toonie_init_codec(struct aoa_codec *codec) +{ + struct toonie *toonie = codec_to_toonie(codec); + + /* nothing connected? what a joke! */ + if (toonie->codec.connected != 1) + return -ENOTCONN; + + if (aoa_snd_device_new(SNDRV_DEV_CODEC, toonie, &ops)) { + printk(KERN_ERR PFX "failed to create toonie snd device!\n"); + return -ENODEV; + } + + if (toonie->codec.soundbus_dev->attach_codec(toonie->codec.soundbus_dev, + aoa_get_card(), + &toonie_codec_info, toonie)) { + printk(KERN_ERR PFX "error creating toonie pcm\n"); + snd_device_free(aoa_get_card(), toonie); + return -ENODEV; + } + + return 0; +} + +static void toonie_exit_codec(struct aoa_codec *codec) +{ + struct toonie *toonie = codec_to_toonie(codec); + + if (!toonie->codec.soundbus_dev) { + printk(KERN_ERR PFX "toonie_exit_codec called without soundbus_dev!\n"); + return; + } + toonie->codec.soundbus_dev->detach_codec(toonie->codec.soundbus_dev, toonie); +} + +static struct toonie *toonie; + +static int __init toonie_init(void) +{ + toonie = kzalloc(sizeof(struct toonie), GFP_KERNEL); + + if (!toonie) + return -ENOMEM; + + strlcpy(toonie->codec.name, "toonie", sizeof(toonie->codec.name)); + toonie->codec.owner = THIS_MODULE; + toonie->codec.init = toonie_init_codec; + toonie->codec.exit = toonie_exit_codec; + + if (aoa_codec_register(&toonie->codec)) { + kfree(toonie); + return -EINVAL; + } + + return 0; +} + +static void __exit toonie_exit(void) +{ + aoa_codec_unregister(&toonie->codec); + kfree(toonie); +} + +module_init(toonie_init); +module_exit(toonie_exit); diff --git a/sound/aoa/core/Makefile b/sound/aoa/core/Makefile new file mode 100644 index 000000000..056d69683 --- /dev/null +++ b/sound/aoa/core/Makefile @@ -0,0 +1,6 @@ +# SPDX-License-Identifier: GPL-2.0 +obj-$(CONFIG_SND_AOA) += snd-aoa.o +snd-aoa-objs := core.o \ + alsa.o \ + gpio-pmf.o \ + gpio-feature.o diff --git a/sound/aoa/core/alsa.c b/sound/aoa/core/alsa.c new file mode 100644 index 000000000..4a7e4e6b7 --- /dev/null +++ b/sound/aoa/core/alsa.c @@ -0,0 +1,99 @@ +/* + * Apple Onboard Audio Alsa helpers + * + * Copyright 2006 Johannes Berg <johannes@sipsolutions.net> + * + * GPL v2, can be found in COPYING. + */ +#include <linux/module.h> +#include "alsa.h" + +static int index = -1; +module_param(index, int, 0444); +MODULE_PARM_DESC(index, "index for AOA sound card."); + +static struct aoa_card *aoa_card; + +int aoa_alsa_init(char *name, struct module *mod, struct device *dev) +{ + struct snd_card *alsa_card; + int err; + + if (aoa_card) + /* cannot be EEXIST due to usage in aoa_fabric_register */ + return -EBUSY; + + err = snd_card_new(dev, index, name, mod, sizeof(struct aoa_card), + &alsa_card); + if (err < 0) + return err; + aoa_card = alsa_card->private_data; + aoa_card->alsa_card = alsa_card; + strlcpy(alsa_card->driver, "AppleOnbdAudio", sizeof(alsa_card->driver)); + strlcpy(alsa_card->shortname, name, sizeof(alsa_card->shortname)); + strlcpy(alsa_card->longname, name, sizeof(alsa_card->longname)); + strlcpy(alsa_card->mixername, name, sizeof(alsa_card->mixername)); + err = snd_card_register(aoa_card->alsa_card); + if (err < 0) { + printk(KERN_ERR "snd-aoa: couldn't register alsa card\n"); + snd_card_free(aoa_card->alsa_card); + aoa_card = NULL; + return err; + } + return 0; +} + +struct snd_card *aoa_get_card(void) +{ + if (aoa_card) + return aoa_card->alsa_card; + return NULL; +} +EXPORT_SYMBOL_GPL(aoa_get_card); + +void aoa_alsa_cleanup(void) +{ + if (aoa_card) { + snd_card_free(aoa_card->alsa_card); + aoa_card = NULL; + } +} + +int aoa_snd_device_new(enum snd_device_type 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); diff --git a/sound/aoa/core/alsa.h b/sound/aoa/core/alsa.h new file mode 100644 index 000000000..9669e4489 --- /dev/null +++ b/sound/aoa/core/alsa.h @@ -0,0 +1,16 @@ +/* + * Apple Onboard Audio Alsa private helpers + * + * Copyright 2006 Johannes Berg <johannes@sipsolutions.net> + * + * GPL v2, can be found in COPYING. + */ + +#ifndef __SND_AOA_ALSA_H +#define __SND_AOA_ALSA_H +#include "../aoa.h" + +extern int aoa_alsa_init(char *name, struct module *mod, struct device *dev); +extern void aoa_alsa_cleanup(void); + +#endif /* __SND_AOA_ALSA_H */ diff --git a/sound/aoa/core/core.c b/sound/aoa/core/core.c new file mode 100644 index 000000000..10bec6c61 --- /dev/null +++ b/sound/aoa/core/core.c @@ -0,0 +1,162 @@ +/* + * 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 "alsa.h" + +MODULE_DESCRIPTION("Apple Onboard Audio Sound Driver"); +MODULE_AUTHOR("Johannes Berg <johannes@sipsolutions.net>"); +MODULE_LICENSE("GPL"); + +/* We allow only one fabric. This simplifies things, + * and more don't really make that much sense */ +static struct aoa_fabric *fabric; +static LIST_HEAD(codec_list); + +static int attach_codec_to_fabric(struct aoa_codec *c) +{ + int err; + + if (!try_module_get(c->owner)) + return -EBUSY; + /* found_codec has to be assigned */ + err = -ENOENT; + if (fabric->found_codec) + err = fabric->found_codec(c); + if (err) { + module_put(c->owner); + printk(KERN_ERR "snd-aoa: fabric didn't like codec %s\n", + c->name); + return err; + } + c->fabric = fabric; + + err = 0; + if (c->init) + err = c->init(c); + if (err) { + printk(KERN_ERR "snd-aoa: codec %s didn't init\n", c->name); + c->fabric = NULL; + if (fabric->remove_codec) + fabric->remove_codec(c); + module_put(c->owner); + return err; + } + if (fabric->attached_codec) + fabric->attached_codec(c); + return 0; +} + +int aoa_codec_register(struct aoa_codec *codec) +{ + int err = 0; + + /* if there's a fabric already, we can tell if we + * will want to have this codec, so propagate error + * through. Otherwise, this will happen later... */ + if (fabric) + err = attach_codec_to_fabric(codec); + if (!err) + list_add(&codec->list, &codec_list); + return err; +} +EXPORT_SYMBOL_GPL(aoa_codec_register); + +void aoa_codec_unregister(struct aoa_codec *codec) +{ + list_del(&codec->list); + if (codec->fabric && codec->exit) + codec->exit(codec); + if (fabric && fabric->remove_codec) + fabric->remove_codec(codec); + codec->fabric = NULL; + module_put(codec->owner); +} +EXPORT_SYMBOL_GPL(aoa_codec_unregister); + +int aoa_fabric_register(struct aoa_fabric *new_fabric, struct device *dev) +{ + 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, dev); + 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); diff --git a/sound/aoa/core/gpio-feature.c b/sound/aoa/core/gpio-feature.c new file mode 100644 index 000000000..65557421f --- /dev/null +++ b/sound/aoa/core/gpio-feature.c @@ -0,0 +1,425 @@ +/* + * Apple Onboard Audio feature call GPIO control + * + * Copyright 2006 Johannes Berg <johannes@sipsolutions.net> + * + * GPL v2, can be found in COPYING. + * + * This file contains the GPIO control routines for + * direct (through feature calls) access to the GPIO + * registers. + */ + +#include <linux/of_irq.h> +#include <linux/interrupt.h> +#include <asm/pmac_feature.h> +#include "../aoa.h" + +/* TODO: these are lots of global variables + * that aren't used on most machines... + * Move them into a dynamically allocated + * structure and use that. + */ + +/* these are the GPIO numbers (register addresses as offsets into + * the GPIO space) */ +static int headphone_mute_gpio; +static int master_mute_gpio; +static int amp_mute_gpio; +static int lineout_mute_gpio; +static int hw_reset_gpio; +static int lineout_detect_gpio; +static int headphone_detect_gpio; +static int linein_detect_gpio; + +/* see the SWITCH_GPIO macro */ +static int headphone_mute_gpio_activestate; +static int master_mute_gpio_activestate; +static int amp_mute_gpio_activestate; +static int lineout_mute_gpio_activestate; +static int hw_reset_gpio_activestate; +static int lineout_detect_gpio_activestate; +static int headphone_detect_gpio_activestate; +static int linein_detect_gpio_activestate; + +/* node pointers that we save when getting the GPIO number + * to get the interrupt later */ +static struct device_node *lineout_detect_node; +static struct device_node *linein_detect_node; +static struct device_node *headphone_detect_node; + +static int lineout_detect_irq; +static int linein_detect_irq; +static int headphone_detect_irq; + +static struct device_node *get_gpio(char *name, + char *altname, + int *gpioptr, + int *gpioactiveptr) +{ + struct device_node *np, *gpio; + const u32 *reg; + const char *audio_gpio; + + *gpioptr = -1; + + /* check if we can get it the easy way ... */ + np = of_find_node_by_name(NULL, name); + if (!np) { + /* some machines have only gpioX/extint-gpioX nodes, + * and an audio-gpio property saying what it is ... + * So what we have to do is enumerate all children + * of the gpio node and check them all. */ + gpio = of_find_node_by_name(NULL, "gpio"); + if (!gpio) + return NULL; + while ((np = of_get_next_child(gpio, np))) { + audio_gpio = of_get_property(np, "audio-gpio", NULL); + if (!audio_gpio) + continue; + if (strcmp(audio_gpio, name) == 0) + break; + if (altname && (strcmp(audio_gpio, altname) == 0)) + break; + } + /* still not found, assume not there */ + if (!np) + return NULL; + } + + reg = of_get_property(np, "reg", NULL); + if (!reg) { + of_node_put(np); + return NULL; + } + + *gpioptr = *reg; + + /* this is a hack, usually the GPIOs 'reg' property + * should have the offset based from the GPIO space + * which is at 0x50, but apparently not always... */ + if (*gpioptr < 0x50) + *gpioptr += 0x50; + + reg = of_get_property(np, "audio-gpio-active-state", NULL); + if (!reg) + /* Apple seems to default to 1, but + * that doesn't seem right at least on most + * machines. So until proven that the opposite + * is necessary, we default to 0 + * (which, incidentally, snd-powermac also does...) */ + *gpioactiveptr = 0; + else + *gpioactiveptr = *reg; + + return np; +} + +static void get_irq(struct device_node * np, int *irqptr) +{ + if (np) + *irqptr = irq_of_parse_and_map(np, 0); + else + *irqptr = 0; +} + +/* 0x4 is outenable, 0x1 is out, thus 4 or 5 */ +#define SWITCH_GPIO(name, v, on) \ + (((v)&~1) | ((on)? \ + (name##_gpio_activestate==0?4:5): \ + (name##_gpio_activestate==0?5:4))) + +#define FTR_GPIO(name, bit) \ +static void ftr_gpio_set_##name(struct gpio_runtime *rt, int on)\ +{ \ + int v; \ + \ + if (unlikely(!rt)) return; \ + \ + if (name##_mute_gpio < 0) \ + return; \ + \ + v = pmac_call_feature(PMAC_FTR_READ_GPIO, NULL, \ + name##_mute_gpio, \ + 0); \ + \ + /* muted = !on... */ \ + v = SWITCH_GPIO(name##_mute, v, !on); \ + \ + pmac_call_feature(PMAC_FTR_WRITE_GPIO, NULL, \ + name##_mute_gpio, v); \ + \ + rt->implementation_private &= ~(1<<bit); \ + rt->implementation_private |= (!!on << bit); \ +} \ +static int ftr_gpio_get_##name(struct gpio_runtime *rt) \ +{ \ + if (unlikely(!rt)) return 0; \ + return (rt->implementation_private>>bit)&1; \ +} + +FTR_GPIO(headphone, 0); +FTR_GPIO(amp, 1); +FTR_GPIO(lineout, 2); +FTR_GPIO(master, 3); + +static void ftr_gpio_set_hw_reset(struct gpio_runtime *rt, int on) +{ + int v; + + if (unlikely(!rt)) return; + if (hw_reset_gpio < 0) + return; + + v = pmac_call_feature(PMAC_FTR_READ_GPIO, NULL, + hw_reset_gpio, 0); + v = SWITCH_GPIO(hw_reset, v, on); + pmac_call_feature(PMAC_FTR_WRITE_GPIO, NULL, + hw_reset_gpio, v); +} + +static struct gpio_methods methods; + +static void ftr_gpio_all_amps_off(struct gpio_runtime *rt) +{ + int saved; + + if (unlikely(!rt)) return; + saved = rt->implementation_private; + ftr_gpio_set_headphone(rt, 0); + ftr_gpio_set_amp(rt, 0); + ftr_gpio_set_lineout(rt, 0); + if (methods.set_master) + ftr_gpio_set_master(rt, 0); + rt->implementation_private = saved; +} + +static void ftr_gpio_all_amps_restore(struct gpio_runtime *rt) +{ + int s; + + if (unlikely(!rt)) return; + s = rt->implementation_private; + ftr_gpio_set_headphone(rt, (s>>0)&1); + ftr_gpio_set_amp(rt, (s>>1)&1); + ftr_gpio_set_lineout(rt, (s>>2)&1); + if (methods.set_master) + ftr_gpio_set_master(rt, (s>>3)&1); +} + +static void ftr_handle_notify(struct work_struct *work) +{ + struct gpio_notification *notif = + container_of(work, struct gpio_notification, work.work); + + mutex_lock(¬if->mutex); + if (notif->notify) + notif->notify(notif->data); + mutex_unlock(¬if->mutex); +} + +static void gpio_enable_dual_edge(int gpio) +{ + int v; + + if (gpio == -1) + return; + v = pmac_call_feature(PMAC_FTR_READ_GPIO, NULL, gpio, 0); + v |= 0x80; /* enable dual edge */ + pmac_call_feature(PMAC_FTR_WRITE_GPIO, NULL, gpio, v); +} + +static void ftr_gpio_init(struct gpio_runtime *rt) +{ + get_gpio("headphone-mute", NULL, + &headphone_mute_gpio, + &headphone_mute_gpio_activestate); + get_gpio("amp-mute", NULL, + &_mute_gpio, + &_mute_gpio_activestate); + get_gpio("lineout-mute", NULL, + &lineout_mute_gpio, + &lineout_mute_gpio_activestate); + get_gpio("hw-reset", "audio-hw-reset", + &hw_reset_gpio, + &hw_reset_gpio_activestate); + if (get_gpio("master-mute", NULL, + &master_mute_gpio, + &master_mute_gpio_activestate)) { + methods.set_master = ftr_gpio_set_master; + methods.get_master = ftr_gpio_get_master; + } + + headphone_detect_node = get_gpio("headphone-detect", NULL, + &headphone_detect_gpio, + &headphone_detect_gpio_activestate); + /* go Apple, and thanks for giving these different names + * across the board... */ + lineout_detect_node = get_gpio("lineout-detect", "line-output-detect", + &lineout_detect_gpio, + &lineout_detect_gpio_activestate); + linein_detect_node = get_gpio("linein-detect", "line-input-detect", + &linein_detect_gpio, + &linein_detect_gpio_activestate); + + gpio_enable_dual_edge(headphone_detect_gpio); + gpio_enable_dual_edge(lineout_detect_gpio); + gpio_enable_dual_edge(linein_detect_gpio); + + get_irq(headphone_detect_node, &headphone_detect_irq); + get_irq(lineout_detect_node, &lineout_detect_irq); + get_irq(linein_detect_node, &linein_detect_irq); + + ftr_gpio_all_amps_off(rt); + rt->implementation_private = 0; + INIT_DELAYED_WORK(&rt->headphone_notify.work, ftr_handle_notify); + INIT_DELAYED_WORK(&rt->line_in_notify.work, ftr_handle_notify); + INIT_DELAYED_WORK(&rt->line_out_notify.work, ftr_handle_notify); + mutex_init(&rt->headphone_notify.mutex); + mutex_init(&rt->line_in_notify.mutex); + mutex_init(&rt->line_out_notify.mutex); +} + +static void ftr_gpio_exit(struct gpio_runtime *rt) +{ + ftr_gpio_all_amps_off(rt); + rt->implementation_private = 0; + if (rt->headphone_notify.notify) + free_irq(headphone_detect_irq, &rt->headphone_notify); + if (rt->line_in_notify.gpio_private) + free_irq(linein_detect_irq, &rt->line_in_notify); + if (rt->line_out_notify.gpio_private) + free_irq(lineout_detect_irq, &rt->line_out_notify); + cancel_delayed_work_sync(&rt->headphone_notify.work); + cancel_delayed_work_sync(&rt->line_in_notify.work); + cancel_delayed_work_sync(&rt->line_out_notify.work); + mutex_destroy(&rt->headphone_notify.mutex); + mutex_destroy(&rt->line_in_notify.mutex); + mutex_destroy(&rt->line_out_notify.mutex); +} + +static irqreturn_t ftr_handle_notify_irq(int xx, void *data) +{ + struct gpio_notification *notif = data; + + schedule_delayed_work(¬if->work, 0); + + return IRQ_HANDLED; +} + +static int ftr_set_notify(struct gpio_runtime *rt, + enum notify_type type, + notify_func_t notify, + void *data) +{ + struct gpio_notification *notif; + notify_func_t old; + int irq; + char *name; + int err = -EBUSY; + + switch (type) { + case AOA_NOTIFY_HEADPHONE: + notif = &rt->headphone_notify; + name = "headphone-detect"; + irq = headphone_detect_irq; + break; + case AOA_NOTIFY_LINE_IN: + notif = &rt->line_in_notify; + name = "linein-detect"; + irq = linein_detect_irq; + break; + case AOA_NOTIFY_LINE_OUT: + notif = &rt->line_out_notify; + name = "lineout-detect"; + irq = lineout_detect_irq; + break; + default: + return -EINVAL; + } + + if (!irq) + return -ENODEV; + + mutex_lock(¬if->mutex); + + old = notif->notify; + + if (!old && !notify) { + err = 0; + goto out_unlock; + } + + if (old && notify) { + if (old == notify && notif->data == data) + err = 0; + goto out_unlock; + } + + if (old && !notify) + free_irq(irq, notif); + + if (!old && notify) { + err = request_irq(irq, ftr_handle_notify_irq, 0, name, notif); + if (err) + goto out_unlock; + } + + notif->notify = notify; + notif->data = data; + + err = 0; + out_unlock: + mutex_unlock(¬if->mutex); + return err; +} + +static int ftr_get_detect(struct gpio_runtime *rt, + enum notify_type type) +{ + int gpio, ret, active; + + switch (type) { + case AOA_NOTIFY_HEADPHONE: + gpio = headphone_detect_gpio; + active = headphone_detect_gpio_activestate; + break; + case AOA_NOTIFY_LINE_IN: + gpio = linein_detect_gpio; + active = linein_detect_gpio_activestate; + break; + case AOA_NOTIFY_LINE_OUT: + gpio = lineout_detect_gpio; + active = lineout_detect_gpio_activestate; + break; + default: + return -EINVAL; + } + + if (gpio == -1) + return -ENODEV; + + ret = pmac_call_feature(PMAC_FTR_READ_GPIO, NULL, gpio, 0); + if (ret < 0) + return ret; + return ((ret >> 1) & 1) == active; +} + +static struct gpio_methods methods = { + .init = ftr_gpio_init, + .exit = ftr_gpio_exit, + .all_amps_off = ftr_gpio_all_amps_off, + .all_amps_restore = ftr_gpio_all_amps_restore, + .set_headphone = ftr_gpio_set_headphone, + .set_speakers = ftr_gpio_set_amp, + .set_lineout = ftr_gpio_set_lineout, + .set_hw_reset = ftr_gpio_set_hw_reset, + .get_headphone = ftr_gpio_get_headphone, + .get_speakers = ftr_gpio_get_amp, + .get_lineout = ftr_gpio_get_lineout, + .set_notify = ftr_set_notify, + .get_detect = ftr_get_detect, +}; + +struct gpio_methods *ftr_gpio_methods = &methods; +EXPORT_SYMBOL_GPL(ftr_gpio_methods); diff --git a/sound/aoa/core/gpio-pmf.c b/sound/aoa/core/gpio-pmf.c new file mode 100644 index 000000000..c8d8a1a6f --- /dev/null +++ b/sound/aoa/core/gpio-pmf.c @@ -0,0 +1,253 @@ +/* + * Apple Onboard Audio pmf GPIOs + * + * Copyright 2006 Johannes Berg <johannes@sipsolutions.net> + * + * GPL v2, can be found in COPYING. + */ + +#include <linux/slab.h> +#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 }; \ + int rc; \ + \ + if (unlikely(!rt)) return; \ + rc = pmf_call_function(rt->node, #name "-mute", &args); \ + if (rc && rc != -ENODEV) \ + printk(KERN_WARNING "pmf_gpio_set_" #name \ + " failed, rc: %d\n", rc); \ + 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 }; + int rc; + + if (unlikely(!rt)) return; + rc = pmf_call_function(rt->node, "hw-reset", &args); + if (rc) + printk(KERN_WARNING "pmf_gpio_set_hw_reset" + " failed, rc: %d\n", rc); +} + +static void pmf_gpio_all_amps_off(struct gpio_runtime *rt) +{ + int saved; + + if (unlikely(!rt)) return; + saved = rt->implementation_private; + pmf_gpio_set_headphone(rt, 0); + pmf_gpio_set_amp(rt, 0); + pmf_gpio_set_lineout(rt, 0); + rt->implementation_private = saved; +} + +static void pmf_gpio_all_amps_restore(struct gpio_runtime *rt) +{ + int s; + + if (unlikely(!rt)) return; + s = rt->implementation_private; + pmf_gpio_set_headphone(rt, (s>>0)&1); + pmf_gpio_set_amp(rt, (s>>1)&1); + pmf_gpio_set_lineout(rt, (s>>2)&1); +} + +static void pmf_handle_notify(struct work_struct *work) +{ + struct gpio_notification *notif = + container_of(work, struct gpio_notification, work.work); + + mutex_lock(¬if->mutex); + if (notif->notify) + notif->notify(notif->data); + mutex_unlock(¬if->mutex); +} + +static void pmf_gpio_init(struct gpio_runtime *rt) +{ + pmf_gpio_all_amps_off(rt); + rt->implementation_private = 0; + INIT_DELAYED_WORK(&rt->headphone_notify.work, pmf_handle_notify); + INIT_DELAYED_WORK(&rt->line_in_notify.work, pmf_handle_notify); + INIT_DELAYED_WORK(&rt->line_out_notify.work, pmf_handle_notify); + mutex_init(&rt->headphone_notify.mutex); + mutex_init(&rt->line_in_notify.mutex); + mutex_init(&rt->line_out_notify.mutex); +} + +static void pmf_gpio_exit(struct gpio_runtime *rt) +{ + pmf_gpio_all_amps_off(rt); + rt->implementation_private = 0; + + if (rt->headphone_notify.gpio_private) + pmf_unregister_irq_client(rt->headphone_notify.gpio_private); + if (rt->line_in_notify.gpio_private) + pmf_unregister_irq_client(rt->line_in_notify.gpio_private); + if (rt->line_out_notify.gpio_private) + pmf_unregister_irq_client(rt->line_out_notify.gpio_private); + + /* make sure no work is pending before freeing + * all things */ + cancel_delayed_work_sync(&rt->headphone_notify.work); + cancel_delayed_work_sync(&rt->line_in_notify.work); + cancel_delayed_work_sync(&rt->line_out_notify.work); + + mutex_destroy(&rt->headphone_notify.mutex); + mutex_destroy(&rt->line_in_notify.mutex); + mutex_destroy(&rt->line_out_notify.mutex); + + kfree(rt->headphone_notify.gpio_private); + kfree(rt->line_in_notify.gpio_private); + kfree(rt->line_out_notify.gpio_private); +} + +static void pmf_handle_notify_irq(void *data) +{ + struct gpio_notification *notif = data; + + schedule_delayed_work(¬if->work, 0); +} + +static int pmf_set_notify(struct gpio_runtime *rt, + enum notify_type type, + notify_func_t notify, + void *data) +{ + struct gpio_notification *notif; + notify_func_t old; + struct pmf_irq_client *irq_client; + char *name; + int err = -EBUSY; + + switch (type) { + case AOA_NOTIFY_HEADPHONE: + notif = &rt->headphone_notify; + name = "headphone-detect"; + break; + case AOA_NOTIFY_LINE_IN: + notif = &rt->line_in_notify; + name = "linein-detect"; + break; + case AOA_NOTIFY_LINE_OUT: + notif = &rt->line_out_notify; + name = "lineout-detect"; + break; + default: + return -EINVAL; + } + + mutex_lock(¬if->mutex); + + old = notif->notify; + + if (!old && !notify) { + err = 0; + goto out_unlock; + } + + if (old && notify) { + if (old == notify && notif->data == data) + err = 0; + goto out_unlock; + } + + if (old && !notify) { + irq_client = notif->gpio_private; + pmf_unregister_irq_client(irq_client); + kfree(irq_client); + notif->gpio_private = NULL; + } + if (!old && notify) { + irq_client = kzalloc(sizeof(struct pmf_irq_client), + GFP_KERNEL); + if (!irq_client) { + err = -ENOMEM; + goto out_unlock; + } + irq_client->data = notif; + irq_client->handler = pmf_handle_notify_irq; + irq_client->owner = THIS_MODULE; + err = pmf_register_irq_client(rt->node, + name, + irq_client); + if (err) { + printk(KERN_ERR "snd-aoa: gpio layer failed to" + " register %s irq (%d)\n", name, err); + kfree(irq_client); + goto out_unlock; + } + notif->gpio_private = irq_client; + } + notif->notify = notify; + notif->data = data; + + err = 0; + out_unlock: + mutex_unlock(¬if->mutex); + return err; +} + +static int pmf_get_detect(struct gpio_runtime *rt, + enum notify_type type) +{ + char *name; + int err = -EBUSY, ret; + struct pmf_args args = { .count = 1, .u[0].p = &ret }; + + switch (type) { + case AOA_NOTIFY_HEADPHONE: + name = "headphone-detect"; + break; + case AOA_NOTIFY_LINE_IN: + name = "linein-detect"; + break; + case AOA_NOTIFY_LINE_OUT: + name = "lineout-detect"; + break; + default: + return -EINVAL; + } + + err = pmf_call_function(rt->node, name, &args); + if (err) + return err; + return ret; +} + +static struct gpio_methods methods = { + .init = pmf_gpio_init, + .exit = pmf_gpio_exit, + .all_amps_off = pmf_gpio_all_amps_off, + .all_amps_restore = pmf_gpio_all_amps_restore, + .set_headphone = pmf_gpio_set_headphone, + .set_speakers = pmf_gpio_set_amp, + .set_lineout = pmf_gpio_set_lineout, + .set_hw_reset = pmf_gpio_set_hw_reset, + .get_headphone = pmf_gpio_get_headphone, + .get_speakers = pmf_gpio_get_amp, + .get_lineout = pmf_gpio_get_lineout, + .set_notify = pmf_set_notify, + .get_detect = pmf_get_detect, +}; + +struct gpio_methods *pmf_gpio_methods = &methods; +EXPORT_SYMBOL_GPL(pmf_gpio_methods); diff --git a/sound/aoa/fabrics/Kconfig b/sound/aoa/fabrics/Kconfig new file mode 100644 index 000000000..3ca475a88 --- /dev/null +++ b/sound/aoa/fabrics/Kconfig @@ -0,0 +1,11 @@ +config SND_AOA_FABRIC_LAYOUT + tristate "layout-id fabric" + 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. diff --git a/sound/aoa/fabrics/Makefile b/sound/aoa/fabrics/Makefile new file mode 100644 index 000000000..da37c10ec --- /dev/null +++ b/sound/aoa/fabrics/Makefile @@ -0,0 +1,3 @@ +snd-aoa-fabric-layout-objs += layout.o + +obj-$(CONFIG_SND_AOA_FABRIC_LAYOUT) += snd-aoa-fabric-layout.o diff --git a/sound/aoa/fabrics/layout.c b/sound/aoa/fabrics/layout.c new file mode 100644 index 000000000..1eddf8fa1 --- /dev/null +++ b/sound/aoa/fabrics/layout.c @@ -0,0 +1,1182 @@ +/* + * Apple Onboard Audio driver -- layout/machine id fabric + * + * Copyright 2006-2008 Johannes Berg <johannes@sipsolutions.net> + * + * GPL v2, can be found in COPYING. + * + * + * This fabric module looks for sound codecs based on the + * layout-id or device-id property in the device tree. + */ +#include <asm/prom.h> +#include <linux/list.h> +#include <linux/module.h> +#include <linux/slab.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, device_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-36"); +MODULE_ALIAS("sound-layout-41"); +MODULE_ALIAS("sound-layout-45"); +MODULE_ALIAS("sound-layout-47"); +MODULE_ALIAS("sound-layout-48"); +MODULE_ALIAS("sound-layout-49"); +MODULE_ALIAS("sound-layout-50"); +MODULE_ALIAS("sound-layout-51"); +MODULE_ALIAS("sound-layout-56"); +MODULE_ALIAS("sound-layout-57"); +MODULE_ALIAS("sound-layout-58"); +MODULE_ALIAS("sound-layout-60"); +MODULE_ALIAS("sound-layout-61"); +MODULE_ALIAS("sound-layout-62"); +MODULE_ALIAS("sound-layout-64"); +MODULE_ALIAS("sound-layout-65"); +MODULE_ALIAS("sound-layout-66"); +MODULE_ALIAS("sound-layout-67"); +MODULE_ALIAS("sound-layout-68"); +MODULE_ALIAS("sound-layout-69"); +MODULE_ALIAS("sound-layout-70"); +MODULE_ALIAS("sound-layout-72"); +MODULE_ALIAS("sound-layout-76"); +MODULE_ALIAS("sound-layout-80"); +MODULE_ALIAS("sound-layout-82"); +MODULE_ALIAS("sound-layout-84"); +MODULE_ALIAS("sound-layout-86"); +MODULE_ALIAS("sound-layout-90"); +MODULE_ALIAS("sound-layout-92"); +MODULE_ALIAS("sound-layout-94"); +MODULE_ALIAS("sound-layout-96"); +MODULE_ALIAS("sound-layout-98"); +MODULE_ALIAS("sound-layout-100"); + +MODULE_ALIAS("aoa-device-id-14"); +MODULE_ALIAS("aoa-device-id-22"); +MODULE_ALIAS("aoa-device-id-31"); +MODULE_ALIAS("aoa-device-id-35"); +MODULE_ALIAS("aoa-device-id-44"); + +/* onyx with all but microphone connected */ +static struct codec_connection onyx_connections_nomic[] = { + { + .connected = CC_SPEAKERS | CC_HEADPHONE | CC_LINEOUT, + .codec_bit = 0, + }, + { + .connected = CC_DIGITALOUT, + .codec_bit = 1, + }, + { + .connected = CC_LINEIN, + .codec_bit = 2, + }, + {} /* terminate array by .connected == 0 */ +}; + +/* onyx on machines without headphone */ +static struct codec_connection onyx_connections_noheadphones[] = { + { + .connected = CC_SPEAKERS | CC_LINEOUT | + CC_LINEOUT_LABELLED_HEADPHONE, + .codec_bit = 0, + }, + { + .connected = CC_DIGITALOUT, + .codec_bit = 1, + }, + /* FIXME: are these correct? probably not for all the machines + * below ... If not this will need separating. */ + { + .connected = CC_LINEIN, + .codec_bit = 2, + }, + { + .connected = CC_MICROPHONE, + .codec_bit = 3, + }, + {} /* terminate array by .connected == 0 */ +}; + +/* onyx on machines with real line-out */ +static struct codec_connection onyx_connections_reallineout[] = { + { + .connected = CC_SPEAKERS | CC_LINEOUT | CC_HEADPHONE, + .codec_bit = 0, + }, + { + .connected = CC_DIGITALOUT, + .codec_bit = 1, + }, + { + .connected = CC_LINEIN, + .codec_bit = 2, + }, + {} /* terminate array by .connected == 0 */ +}; + +/* tas on machines without line out */ +static struct codec_connection tas_connections_nolineout[] = { + { + .connected = CC_SPEAKERS | CC_HEADPHONE, + .codec_bit = 0, + }, + { + .connected = CC_LINEIN, + .codec_bit = 2, + }, + { + .connected = CC_MICROPHONE, + .codec_bit = 3, + }, + {} /* terminate array by .connected == 0 */ +}; + +/* tas on machines with neither line out nor line in */ +static struct codec_connection tas_connections_noline[] = { + { + .connected = CC_SPEAKERS | CC_HEADPHONE, + .codec_bit = 0, + }, + { + .connected = CC_MICROPHONE, + .codec_bit = 3, + }, + {} /* terminate array by .connected == 0 */ +}; + +/* tas on machines without microphone */ +static struct codec_connection tas_connections_nomic[] = { + { + .connected = CC_SPEAKERS | CC_HEADPHONE | CC_LINEOUT, + .codec_bit = 0, + }, + { + .connected = CC_LINEIN, + .codec_bit = 2, + }, + {} /* terminate array by .connected == 0 */ +}; + +/* tas on machines with everything connected */ +static struct codec_connection tas_connections_all[] = { + { + .connected = CC_SPEAKERS | CC_HEADPHONE | CC_LINEOUT, + .codec_bit = 0, + }, + { + .connected = CC_LINEIN, + .codec_bit = 2, + }, + { + .connected = CC_MICROPHONE, + .codec_bit = 3, + }, + {} /* terminate array by .connected == 0 */ +}; + +static struct codec_connection toonie_connections[] = { + { + .connected = CC_SPEAKERS | CC_HEADPHONE, + .codec_bit = 0, + }, + {} /* terminate array by .connected == 0 */ +}; + +static struct codec_connection topaz_input[] = { + { + .connected = CC_DIGITALIN, + .codec_bit = 0, + }, + {} /* terminate array by .connected == 0 */ +}; + +static struct codec_connection topaz_output[] = { + { + .connected = CC_DIGITALOUT, + .codec_bit = 1, + }, + {} /* terminate array by .connected == 0 */ +}; + +static struct codec_connection topaz_inout[] = { + { + .connected = CC_DIGITALIN, + .codec_bit = 0, + }, + { + .connected = CC_DIGITALOUT, + .codec_bit = 1, + }, + {} /* terminate array by .connected == 0 */ +}; + +static struct layout layouts[] = { + /* last PowerBooks (15" Oct 2005) */ + { .layout_id = 82, + .flags = LAYOUT_FLAG_COMBO_LINEOUT_SPDIF, + .codecs[0] = { + .name = "onyx", + .connections = onyx_connections_noheadphones, + }, + .codecs[1] = { + .name = "topaz", + .connections = topaz_input, + }, + }, + /* PowerMac9,1 */ + { .layout_id = 60, + .codecs[0] = { + .name = "onyx", + .connections = onyx_connections_reallineout, + }, + }, + /* PowerMac9,1 */ + { .layout_id = 61, + .codecs[0] = { + .name = "topaz", + .connections = topaz_input, + }, + }, + /* PowerBook5,7 */ + { .layout_id = 64, + .flags = LAYOUT_FLAG_COMBO_LINEOUT_SPDIF, + .codecs[0] = { + .name = "onyx", + .connections = onyx_connections_noheadphones, + }, + }, + /* PowerBook5,7 */ + { .layout_id = 65, + .codecs[0] = { + .name = "topaz", + .connections = topaz_input, + }, + }, + /* PowerBook5,9 [17" Oct 2005] */ + { .layout_id = 84, + .flags = LAYOUT_FLAG_COMBO_LINEOUT_SPDIF, + .codecs[0] = { + .name = "onyx", + .connections = onyx_connections_noheadphones, + }, + .codecs[1] = { + .name = "topaz", + .connections = topaz_input, + }, + }, + /* PowerMac8,1 */ + { .layout_id = 45, + .codecs[0] = { + .name = "onyx", + .connections = onyx_connections_noheadphones, + }, + .codecs[1] = { + .name = "topaz", + .connections = topaz_input, + }, + }, + /* Quad PowerMac (analog in, analog/digital out) */ + { .layout_id = 68, + .codecs[0] = { + .name = "onyx", + .connections = onyx_connections_nomic, + }, + }, + /* Quad PowerMac (digital in) */ + { .layout_id = 69, + .codecs[0] = { + .name = "topaz", + .connections = topaz_input, + }, + .busname = "digital in", .pcmid = 1 }, + /* Early 2005 PowerBook (PowerBook 5,6) */ + { .layout_id = 70, + .codecs[0] = { + .name = "tas", + .connections = tas_connections_nolineout, + }, + }, + /* PowerBook 5,4 */ + { .layout_id = 51, + .codecs[0] = { + .name = "tas", + .connections = tas_connections_nolineout, + }, + }, + /* PowerBook6,1 */ + { .device_id = 31, + .codecs[0] = { + .name = "tas", + .connections = tas_connections_nolineout, + }, + }, + /* PowerBook6,5 */ + { .device_id = 44, + .codecs[0] = { + .name = "tas", + .connections = tas_connections_all, + }, + }, + /* PowerBook6,7 */ + { .layout_id = 80, + .codecs[0] = { + .name = "tas", + .connections = tas_connections_noline, + }, + }, + /* PowerBook6,8 */ + { .layout_id = 72, + .codecs[0] = { + .name = "tas", + .connections = tas_connections_nolineout, + }, + }, + /* PowerMac8,2 */ + { .layout_id = 86, + .codecs[0] = { + .name = "onyx", + .connections = onyx_connections_nomic, + }, + .codecs[1] = { + .name = "topaz", + .connections = topaz_input, + }, + }, + /* PowerBook6,7 */ + { .layout_id = 92, + .codecs[0] = { + .name = "tas", + .connections = tas_connections_nolineout, + }, + }, + /* PowerMac10,1 (Mac Mini) */ + { .layout_id = 58, + .codecs[0] = { + .name = "toonie", + .connections = toonie_connections, + }, + }, + { + .layout_id = 96, + .codecs[0] = { + .name = "onyx", + .connections = onyx_connections_noheadphones, + }, + }, + /* unknown, untested, but this comes from Apple */ + { .layout_id = 41, + .codecs[0] = { + .name = "tas", + .connections = tas_connections_all, + }, + }, + { .layout_id = 36, + .codecs[0] = { + .name = "tas", + .connections = tas_connections_nomic, + }, + .codecs[1] = { + .name = "topaz", + .connections = topaz_inout, + }, + }, + { .layout_id = 47, + .codecs[0] = { + .name = "onyx", + .connections = onyx_connections_noheadphones, + }, + }, + { .layout_id = 48, + .codecs[0] = { + .name = "topaz", + .connections = topaz_input, + }, + }, + { .layout_id = 49, + .codecs[0] = { + .name = "onyx", + .connections = onyx_connections_nomic, + }, + }, + { .layout_id = 50, + .codecs[0] = { + .name = "topaz", + .connections = topaz_input, + }, + }, + { .layout_id = 56, + .codecs[0] = { + .name = "onyx", + .connections = onyx_connections_noheadphones, + }, + }, + { .layout_id = 57, + .codecs[0] = { + .name = "topaz", + .connections = topaz_input, + }, + }, + { .layout_id = 62, + .codecs[0] = { + .name = "onyx", + .connections = onyx_connections_noheadphones, + }, + .codecs[1] = { + .name = "topaz", + .connections = topaz_output, + }, + }, + { .layout_id = 66, + .codecs[0] = { + .name = "onyx", + .connections = onyx_connections_noheadphones, + }, + }, + { .layout_id = 67, + .codecs[0] = { + .name = "topaz", + .connections = topaz_input, + }, + }, + { .layout_id = 76, + .codecs[0] = { + .name = "tas", + .connections = tas_connections_nomic, + }, + .codecs[1] = { + .name = "topaz", + .connections = topaz_inout, + }, + }, + { .layout_id = 90, + .codecs[0] = { + .name = "tas", + .connections = tas_connections_noline, + }, + }, + { .layout_id = 94, + .codecs[0] = { + .name = "onyx", + /* but it has an external mic?? how to select? */ + .connections = onyx_connections_noheadphones, + }, + }, + { .layout_id = 98, + .codecs[0] = { + .name = "toonie", + .connections = toonie_connections, + }, + }, + { .layout_id = 100, + .codecs[0] = { + .name = "topaz", + .connections = topaz_input, + }, + .codecs[1] = { + .name = "onyx", + .connections = onyx_connections_noheadphones, + }, + }, + /* PowerMac3,4 */ + { .device_id = 14, + .codecs[0] = { + .name = "tas", + .connections = tas_connections_noline, + }, + }, + /* PowerMac3,6 */ + { .device_id = 22, + .codecs[0] = { + .name = "tas", + .connections = tas_connections_all, + }, + }, + /* PowerBook5,2 */ + { .device_id = 35, + .codecs[0] = { + .name = "tas", + .connections = tas_connections_all, + }, + }, + {} +}; + +static struct layout *find_layout_by_id(unsigned int id) +{ + struct layout *l; + + l = layouts; + while (l->codecs[0].name) { + if (l->layout_id == id) + return l; + l++; + } + return NULL; +} + +static struct layout *find_layout_by_device(unsigned int id) +{ + struct layout *l; + + l = layouts; + while (l->codecs[0].name) { + if (l->device_id == id) + return l; + l++; + } + return NULL; +} + +static void use_layout(struct layout *l) +{ + int i; + + for (i=0; i<MAX_CODECS_PER_BUS; i++) { + if (l->codecs[i].name) { + request_module("snd-aoa-codec-%s", l->codecs[i].name); + } + } + /* now we wait for the codecs to call us back */ +} + +struct layout_dev; + +struct layout_dev_ptr { + struct layout_dev *ptr; +}; + +struct layout_dev { + struct list_head list; + struct soundbus_dev *sdev; + struct device_node *sound; + struct aoa_codec *codecs[MAX_CODECS_PER_BUS]; + struct layout *layout; + struct gpio_runtime gpio; + + /* we need these for headphone/lineout detection */ + struct snd_kcontrol *headphone_ctrl; + struct snd_kcontrol *lineout_ctrl; + struct snd_kcontrol *speaker_ctrl; + struct snd_kcontrol *master_ctrl; + struct snd_kcontrol *headphone_detected_ctrl; + struct snd_kcontrol *lineout_detected_ctrl; + + struct layout_dev_ptr selfptr_headphone; + struct layout_dev_ptr selfptr_lineout; + + u32 have_lineout_detect:1, + have_headphone_detect:1, + switch_on_headphone:1, + switch_on_lineout:1; +}; + +static LIST_HEAD(layouts_list); +static int layouts_list_items; +/* this can go away but only if we allow multiple cards, + * make the fabric handle all the card stuff, etc... */ +static struct layout_dev *layout_device; + +#define control_info snd_ctl_boolean_mono_info + +#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->set_##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"); +AMP_CONTROL(master, "Master Switch"); + +static int detect_choice_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct layout_dev *ldev = snd_kcontrol_chip(kcontrol); + + switch (kcontrol->private_value) { + case 0: + ucontrol->value.integer.value[0] = ldev->switch_on_headphone; + break; + case 1: + ucontrol->value.integer.value[0] = ldev->switch_on_lineout; + break; + default: + return -ENODEV; + } + return 0; +} + +static int detect_choice_put(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct layout_dev *ldev = snd_kcontrol_chip(kcontrol); + + switch (kcontrol->private_value) { + case 0: + ldev->switch_on_headphone = !!ucontrol->value.integer.value[0]; + break; + case 1: + ldev->switch_on_lineout = !!ucontrol->value.integer.value[0]; + break; + default: + return -ENODEV; + } + return 1; +} + +static const struct snd_kcontrol_new headphone_detect_choice = { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Headphone Detect Autoswitch", + .info = control_info, + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, + .get = detect_choice_get, + .put = detect_choice_put, + .private_value = 0, +}; + +static const struct snd_kcontrol_new lineout_detect_choice = { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Line-Out Detect Autoswitch", + .info = control_info, + .access = SNDRV_CTL_ELEM_ACCESS_READWRITE, + .get = detect_choice_get, + .put = detect_choice_put, + .private_value = 1, +}; + +static int detected_get(struct snd_kcontrol *kcontrol, + struct snd_ctl_elem_value *ucontrol) +{ + struct layout_dev *ldev = snd_kcontrol_chip(kcontrol); + int v; + + switch (kcontrol->private_value) { + case 0: + v = ldev->gpio.methods->get_detect(&ldev->gpio, + AOA_NOTIFY_HEADPHONE); + break; + case 1: + v = ldev->gpio.methods->get_detect(&ldev->gpio, + AOA_NOTIFY_LINE_OUT); + break; + default: + return -ENODEV; + } + ucontrol->value.integer.value[0] = v; + return 0; +} + +static const struct snd_kcontrol_new headphone_detected = { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Headphone Detected", + .info = control_info, + .access = SNDRV_CTL_ELEM_ACCESS_READ, + .get = detected_get, + .private_value = 0, +}; + +static const struct snd_kcontrol_new lineout_detected = { + .iface = SNDRV_CTL_ELEM_IFACE_MIXER, + .name = "Line-Out Detected", + .info = control_info, + .access = SNDRV_CTL_ELEM_ACCESS_READ, + .get = detected_get, + .private_value = 1, +}; + +static int check_codec(struct aoa_codec *codec, + struct layout_dev *ldev, + struct codec_connect_info *cci) +{ + const u32 *ref; + char propname[32]; + struct codec_connection *cc; + + /* if the codec has a 'codec' node, we require a reference */ + if (codec->node && (strcmp(codec->node->name, "codec") == 0)) { + snprintf(propname, sizeof(propname), + "platform-%s-codec-ref", codec->name); + ref = of_get_property(ldev->sound, propname, NULL); + if (!ref) { + printk(KERN_INFO "snd-aoa-fabric-layout: " + "required property %s not present\n", propname); + return -ENODEV; + } + if (*ref != codec->node->phandle) { + printk(KERN_INFO "snd-aoa-fabric-layout: " + "%s doesn't match!\n", propname); + return -ENODEV; + } + } else { + if (layouts_list_items != 1) { + printk(KERN_INFO "snd-aoa-fabric-layout: " + "more than one soundbus, but no references.\n"); + return -ENODEV; + } + } + codec->soundbus_dev = ldev->sdev; + codec->gpio = &ldev->gpio; + + cc = cci->connections; + if (!cc) + return -EINVAL; + + printk(KERN_INFO "snd-aoa-fabric-layout: can use this codec\n"); + + codec->connected = 0; + codec->fabric_data = cc; + + while (cc->connected) { + codec->connected |= 1<<cc->codec_bit; + cc++; + } + + return 0; +} + +static int layout_found_codec(struct aoa_codec *codec) +{ + struct layout_dev *ldev; + int i; + + list_for_each_entry(ldev, &layouts_list, list) { + for (i=0; i<MAX_CODECS_PER_BUS; i++) { + if (!ldev->layout->codecs[i].name) + continue; + if (strcmp(ldev->layout->codecs[i].name, codec->name) == 0) { + if (check_codec(codec, + ldev, + &ldev->layout->codecs[i]) == 0) + return 0; + } + } + } + return -ENODEV; +} + +static void layout_remove_codec(struct aoa_codec *codec) +{ + int i; + /* here remove the codec from the layout dev's + * codec reference */ + + codec->soundbus_dev = NULL; + codec->gpio = NULL; + for (i=0; i<MAX_CODECS_PER_BUS; i++) { + } +} + +static void layout_notify(void *data) +{ + struct layout_dev_ptr *dptr = data; + struct layout_dev *ldev; + int v, update; + struct snd_kcontrol *detected, *c; + struct snd_card *card = aoa_get_card(); + + ldev = dptr->ptr; + if (data == &ldev->selfptr_headphone) { + v = ldev->gpio.methods->get_detect(&ldev->gpio, AOA_NOTIFY_HEADPHONE); + detected = ldev->headphone_detected_ctrl; + update = ldev->switch_on_headphone; + if (update) { + ldev->gpio.methods->set_speakers(&ldev->gpio, !v); + ldev->gpio.methods->set_headphone(&ldev->gpio, v); + ldev->gpio.methods->set_lineout(&ldev->gpio, 0); + } + } else if (data == &ldev->selfptr_lineout) { + v = ldev->gpio.methods->get_detect(&ldev->gpio, AOA_NOTIFY_LINE_OUT); + detected = ldev->lineout_detected_ctrl; + update = ldev->switch_on_lineout; + if (update) { + ldev->gpio.methods->set_speakers(&ldev->gpio, !v); + ldev->gpio.methods->set_headphone(&ldev->gpio, 0); + ldev->gpio.methods->set_lineout(&ldev->gpio, v); + } + } else + return; + + if (detected) + snd_ctl_notify(card, SNDRV_CTL_EVENT_MASK_VALUE, &detected->id); + if (update) { + c = ldev->headphone_ctrl; + if (c) + snd_ctl_notify(card, SNDRV_CTL_EVENT_MASK_VALUE, &c->id); + c = ldev->speaker_ctrl; + if (c) + snd_ctl_notify(card, SNDRV_CTL_EVENT_MASK_VALUE, &c->id); + c = ldev->lineout_ctrl; + if (c) + snd_ctl_notify(card, SNDRV_CTL_EVENT_MASK_VALUE, &c->id); + } +} + +static void layout_attached_codec(struct aoa_codec *codec) +{ + struct codec_connection *cc; + struct snd_kcontrol *ctl; + int headphones, lineout; + struct layout_dev *ldev = layout_device; + + /* need to add this codec to our codec array! */ + + cc = codec->fabric_data; + + headphones = codec->gpio->methods->get_detect(codec->gpio, + AOA_NOTIFY_HEADPHONE); + lineout = codec->gpio->methods->get_detect(codec->gpio, + AOA_NOTIFY_LINE_OUT); + + if (codec->gpio->methods->set_master) { + ctl = snd_ctl_new1(&master_ctl, codec->gpio); + ldev->master_ctrl = ctl; + aoa_snd_ctl_add(ctl); + } + while (cc->connected) { + if (cc->connected & CC_SPEAKERS) { + if (headphones <= 0 && lineout <= 0) + ldev->gpio.methods->set_speakers(codec->gpio, 1); + ctl = snd_ctl_new1(&speakers_ctl, codec->gpio); + ldev->speaker_ctrl = ctl; + aoa_snd_ctl_add(ctl); + } + if (cc->connected & CC_HEADPHONE) { + if (headphones == 1) + ldev->gpio.methods->set_headphone(codec->gpio, 1); + ctl = snd_ctl_new1(&headphone_ctl, codec->gpio); + ldev->headphone_ctrl = ctl; + aoa_snd_ctl_add(ctl); + ldev->have_headphone_detect = + !ldev->gpio.methods + ->set_notify(&ldev->gpio, + AOA_NOTIFY_HEADPHONE, + layout_notify, + &ldev->selfptr_headphone); + if (ldev->have_headphone_detect) { + ctl = snd_ctl_new1(&headphone_detect_choice, + ldev); + aoa_snd_ctl_add(ctl); + ctl = snd_ctl_new1(&headphone_detected, + ldev); + ldev->headphone_detected_ctrl = ctl; + aoa_snd_ctl_add(ctl); + } + } + if (cc->connected & CC_LINEOUT) { + if (lineout == 1) + ldev->gpio.methods->set_lineout(codec->gpio, 1); + ctl = snd_ctl_new1(&lineout_ctl, codec->gpio); + if (cc->connected & CC_LINEOUT_LABELLED_HEADPHONE) + strlcpy(ctl->id.name, + "Headphone Switch", sizeof(ctl->id.name)); + ldev->lineout_ctrl = ctl; + aoa_snd_ctl_add(ctl); + ldev->have_lineout_detect = + !ldev->gpio.methods + ->set_notify(&ldev->gpio, + AOA_NOTIFY_LINE_OUT, + layout_notify, + &ldev->selfptr_lineout); + if (ldev->have_lineout_detect) { + ctl = snd_ctl_new1(&lineout_detect_choice, + ldev); + if (cc->connected & CC_LINEOUT_LABELLED_HEADPHONE) + strlcpy(ctl->id.name, + "Headphone Detect Autoswitch", + sizeof(ctl->id.name)); + aoa_snd_ctl_add(ctl); + ctl = snd_ctl_new1(&lineout_detected, + ldev); + if (cc->connected & CC_LINEOUT_LABELLED_HEADPHONE) + strlcpy(ctl->id.name, + "Headphone Detected", + sizeof(ctl->id.name)); + ldev->lineout_detected_ctrl = ctl; + aoa_snd_ctl_add(ctl); + } + } + cc++; + } + /* now update initial state */ + if (ldev->have_headphone_detect) + layout_notify(&ldev->selfptr_headphone); + if (ldev->have_lineout_detect) + layout_notify(&ldev->selfptr_lineout); +} + +static struct aoa_fabric layout_fabric = { + .name = "SoundByLayout", + .owner = THIS_MODULE, + .found_codec = layout_found_codec, + .remove_codec = layout_remove_codec, + .attached_codec = layout_attached_codec, +}; + +static int aoa_fabric_layout_probe(struct soundbus_dev *sdev) +{ + struct device_node *sound = NULL; + const unsigned int *id; + struct layout *layout = NULL; + struct layout_dev *ldev = NULL; + int err; + + /* hm, currently we can only have one ... */ + if (layout_device) + return -ENODEV; + + /* by breaking out we keep a reference */ + while ((sound = of_get_next_child(sdev->ofdev.dev.of_node, sound))) { + if (sound->type && strcasecmp(sound->type, "soundchip") == 0) + break; + } + if (!sound) + return -ENODEV; + + id = of_get_property(sound, "layout-id", NULL); + if (id) { + layout = find_layout_by_id(*id); + } else { + id = of_get_property(sound, "device-id", NULL); + if (id) + layout = find_layout_by_device(*id); + } + + if (!layout) { + printk(KERN_ERR "snd-aoa-fabric-layout: unknown layout\n"); + goto outnodev; + } + + ldev = kzalloc(sizeof(struct layout_dev), GFP_KERNEL); + if (!ldev) + goto outnodev; + + layout_device = ldev; + ldev->sdev = sdev; + ldev->sound = sound; + ldev->layout = layout; + ldev->gpio.node = sound->parent; + switch (layout->layout_id) { + case 0: /* anything with device_id, not layout_id */ + case 41: /* that unknown machine no one seems to have */ + case 51: /* PowerBook5,4 */ + case 58: /* Mac Mini */ + ldev->gpio.methods = ftr_gpio_methods; + printk(KERN_DEBUG + "snd-aoa-fabric-layout: Using direct GPIOs\n"); + break; + default: + ldev->gpio.methods = pmf_gpio_methods; + printk(KERN_DEBUG + "snd-aoa-fabric-layout: Using PMF GPIOs\n"); + } + ldev->selfptr_headphone.ptr = ldev; + ldev->selfptr_lineout.ptr = ldev; + dev_set_drvdata(&sdev->ofdev.dev, ldev); + list_add(&ldev->list, &layouts_list); + layouts_list_items++; + + /* assign these before registering ourselves, so + * callbacks that are done during registration + * already have the values */ + sdev->pcmid = ldev->layout->pcmid; + if (ldev->layout->busname) { + sdev->pcmname = ldev->layout->busname; + } else { + sdev->pcmname = "Master"; + } + + ldev->gpio.methods->init(&ldev->gpio); + + err = aoa_fabric_register(&layout_fabric, &sdev->ofdev.dev); + if (err && err != -EALREADY) { + printk(KERN_INFO "snd-aoa-fabric-layout: can't use," + " another fabric is active!\n"); + goto outlistdel; + } + + use_layout(layout); + ldev->switch_on_headphone = 1; + ldev->switch_on_lineout = 1; + return 0; + outlistdel: + /* we won't be using these then... */ + ldev->gpio.methods->exit(&ldev->gpio); + /* reset if we didn't use it */ + sdev->pcmname = NULL; + sdev->pcmid = -1; + list_del(&ldev->list); + layouts_list_items--; + kfree(ldev); + outnodev: + of_node_put(sound); + layout_device = NULL; + return -ENODEV; +} + +static int aoa_fabric_layout_remove(struct soundbus_dev *sdev) +{ + struct layout_dev *ldev = dev_get_drvdata(&sdev->ofdev.dev); + int i; + + for (i=0; i<MAX_CODECS_PER_BUS; i++) { + if (ldev->codecs[i]) { + aoa_fabric_unlink_codec(ldev->codecs[i]); + } + ldev->codecs[i] = NULL; + } + list_del(&ldev->list); + layouts_list_items--; + of_node_put(ldev->sound); + + ldev->gpio.methods->set_notify(&ldev->gpio, + AOA_NOTIFY_HEADPHONE, + NULL, + NULL); + ldev->gpio.methods->set_notify(&ldev->gpio, + AOA_NOTIFY_LINE_OUT, + NULL, + NULL); + + ldev->gpio.methods->exit(&ldev->gpio); + layout_device = NULL; + kfree(ldev); + sdev->pcmid = -1; + sdev->pcmname = NULL; + return 0; +} + +#ifdef CONFIG_PM_SLEEP +static int aoa_fabric_layout_suspend(struct device *dev) +{ + struct layout_dev *ldev = dev_get_drvdata(dev); + + 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 device *dev) +{ + struct layout_dev *ldev = dev_get_drvdata(dev); + + if (ldev->gpio.methods && ldev->gpio.methods->all_amps_restore) + ldev->gpio.methods->all_amps_restore(&ldev->gpio); + + return 0; +} + +static SIMPLE_DEV_PM_OPS(aoa_fabric_layout_pm_ops, + aoa_fabric_layout_suspend, aoa_fabric_layout_resume); + +#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, + .driver = { + .owner = THIS_MODULE, +#ifdef CONFIG_PM_SLEEP + .pm = &aoa_fabric_layout_pm_ops, +#endif + } +}; + +static int __init aoa_fabric_layout_init(void) +{ + return soundbus_register_driver(&aoa_soundbus_driver); +} + +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); diff --git a/sound/aoa/soundbus/Kconfig b/sound/aoa/soundbus/Kconfig new file mode 100644 index 000000000..839d1137b --- /dev/null +++ b/sound/aoa/soundbus/Kconfig @@ -0,0 +1,14 @@ +config SND_AOA_SOUNDBUS + tristate "Apple Soundbus support" + select SND_PCM + ---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. diff --git a/sound/aoa/soundbus/Makefile b/sound/aoa/soundbus/Makefile new file mode 100644 index 000000000..0e61f5aa0 --- /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/ diff --git a/sound/aoa/soundbus/core.c b/sound/aoa/soundbus/core.c new file mode 100644 index 000000000..70bcaa7f9 --- /dev/null +++ b/sound/aoa/soundbus/core.c @@ -0,0 +1,193 @@ +/* + * 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, struct kobj_uevent_env *env) +{ + struct soundbus_dev * soundbus_dev; + struct platform_device * of; + const char *compat; + int retval = 0; + int cplen, 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 */ + retval = add_uevent_var(env, "OF_NAME=%s", of->dev.of_node->name); + if (retval) + return retval; + + retval = add_uevent_var(env, "OF_TYPE=%s", of->dev.of_node->type); + if (retval) + return retval; + + /* 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 = of_get_property(of->dev.of_node, "compatible", &cplen); + while (compat && cplen > 0) { + int tmp = env->buflen; + retval = add_uevent_var(env, "OF_COMPATIBLE_%d=%s", seen, compat); + if (retval) + return retval; + compat += env->buflen - tmp; + cplen -= env->buflen - tmp; + seen += 1; + } + + retval = add_uevent_var(env, "OF_COMPATIBLE_N=%d", seen); + if (retval) + return retval; + retval = add_uevent_var(env, "MODALIAS=%s", soundbus_dev->modalias); + + return retval; +} + +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); +} + +/* soundbus_dev_attrs is declared in sysfs.c */ +ATTRIBUTE_GROUPS(soundbus_dev); +static struct bus_type soundbus_bus_type = { + .name = "aoa-soundbus", + .probe = soundbus_probe, + .uevent = soundbus_uevent, + .remove = soundbus_device_remove, + .shutdown = soundbus_device_shutdown, + .dev_groups = soundbus_dev_groups, +}; + +int soundbus_add_one(struct soundbus_dev *dev) +{ + static int devcount; + + /* sanity checks */ + if (!dev->attach_codec || + !dev->ofdev.dev.of_node || + dev->pcmname || + dev->pcmid != -1) { + printk(KERN_ERR "soundbus: adding device failed sanity check!\n"); + return -EINVAL; + } + + dev_set_name(&dev->ofdev.dev, "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); + +static int __init soundbus_init(void) +{ + return bus_register(&soundbus_bus_type); +} + +static void __exit soundbus_exit(void) +{ + bus_unregister(&soundbus_bus_type); +} + +subsys_initcall(soundbus_init); +module_exit(soundbus_exit); diff --git a/sound/aoa/soundbus/i2sbus/Makefile b/sound/aoa/soundbus/i2sbus/Makefile new file mode 100644 index 000000000..1b949b2a4 --- /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 := core.o pcm.o control.o diff --git a/sound/aoa/soundbus/i2sbus/control.c b/sound/aoa/soundbus/i2sbus/control.c new file mode 100644 index 000000000..f4495decc --- /dev/null +++ b/sound/aoa/soundbus/i2sbus/control.c @@ -0,0 +1,194 @@ +/* + * i2sbus driver -- bus control routines + * + * Copyright 2006 Johannes Berg <johannes@sipsolutions.net> + * + * GPL v2, can be found in COPYING. + */ + +#include <linux/kernel.h> +#include <linux/delay.h> +#include <linux/slab.h> +#include <linux/io.h> + +#include <asm/prom.h> +#include <asm/macio.h> +#include <asm/pmac_feature.h> +#include <asm/pmac_pfunc.h> +#include <asm/keylargo.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); + + (*c)->macio = dev->bus->chip; + return 0; +} + +void i2sbus_control_destroy(struct i2sbus_control *c) +{ + 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.dev.of_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 }; + struct macio_chip *macio = c->macio; + + if (i2sdev->enable) + return pmf_call_one(i2sdev->enable, &args); + + if (macio == NULL || macio->base == NULL) + return -ENODEV; + + switch (i2sdev->bus_number) { + case 0: + /* these need to be locked or done through + * newly created feature calls! */ + MACIO_BIS(KEYLARGO_FCR1, KL1_I2S0_ENABLE); + break; + case 1: + MACIO_BIS(KEYLARGO_FCR1, KL1_I2S1_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 }; + struct macio_chip *macio = c->macio; + + 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; + } + + if (macio == NULL || macio->base == NULL) + return -ENODEV; + + switch (i2sdev->bus_number) { + case 0: + if (enable) + MACIO_BIS(KEYLARGO_FCR1, KL1_I2S0_CELL_ENABLE); + else + MACIO_BIC(KEYLARGO_FCR1, KL1_I2S0_CELL_ENABLE); + break; + case 1: + if (enable) + MACIO_BIS(KEYLARGO_FCR1, KL1_I2S1_CELL_ENABLE); + else + MACIO_BIC(KEYLARGO_FCR1, KL1_I2S1_CELL_ENABLE); + 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 }; + struct macio_chip *macio = c->macio; + + 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; + } + + if (macio == NULL || macio->base == NULL) + return -ENODEV; + + switch (i2sdev->bus_number) { + case 0: + if (enable) + MACIO_BIS(KEYLARGO_FCR1, KL1_I2S0_CLK_ENABLE_BIT); + else + MACIO_BIC(KEYLARGO_FCR1, KL1_I2S0_CLK_ENABLE_BIT); + break; + case 1: + if (enable) + MACIO_BIS(KEYLARGO_FCR1, KL1_I2S1_CLK_ENABLE_BIT); + else + MACIO_BIC(KEYLARGO_FCR1, KL1_I2S1_CLK_ENABLE_BIT); + break; + default: + return -ENODEV; + } + return 0; +} diff --git a/sound/aoa/soundbus/i2sbus/core.c b/sound/aoa/soundbus/i2sbus/core.c new file mode 100644 index 000000000..000b58522 --- /dev/null +++ b/sound/aoa/soundbus/i2sbus/core.c @@ -0,0 +1,458 @@ +/* + * i2sbus driver + * + * Copyright 2006-2008 Johannes Berg <johannes@sipsolutions.net> + * + * GPL v2, can be found in COPYING. + */ + +#include <linux/module.h> +#include <linux/slab.h> +#include <linux/pci.h> +#include <linux/interrupt.h> +#include <linux/dma-mapping.h> +#include <linux/of_address.h> +#include <linux/of_irq.h> + +#include <sound/core.h> + +#include <asm/macio.h> +#include <asm/dbdma.h> + +#include "../soundbus.h" +#include "i2sbus.h" + +MODULE_LICENSE("GPL"); +MODULE_AUTHOR("Johannes Berg <johannes@sipsolutions.net>"); +MODULE_DESCRIPTION("Apple Soundbus: I2S support"); + +static int force; +module_param(force, int, 0444); +MODULE_PARM_DESC(force, "Force loading i2sbus even when" + " no layout-id property is present"); + +static const struct of_device_id i2sbus_match[] = { + { .name = "i2s" }, + { } +}; + +MODULE_DEVICE_TABLE(of, i2sbus_match); + +static int alloc_dbdma_descriptor_ring(struct i2sbus_dev *i2sdev, + struct dbdma_command_mem *r, + int numcmds) +{ + /* one more for rounding, one for branch back, one for stop command */ + r->size = (numcmds + 3) * sizeof(struct dbdma_cmd); + /* We use the PCI APIs for now until the generic one gets fixed + * enough or until we get some macio-specific versions + */ + r->space = dma_zalloc_coherent(&macio_get_pci_dev(i2sdev->macio)->dev, + r->size, &r->bus_addr, GFP_KERNEL); + if (!r->space) + return -ENOMEM; + + r->cmds = (void*)DBDMA_ALIGN(r->space); + r->bus_cmd_start = r->bus_addr + + (dma_addr_t)((char*)r->cmds - (char*)r->space); + + return 0; +} + +static void free_dbdma_descriptor_ring(struct i2sbus_dev *i2sdev, + struct dbdma_command_mem *r) +{ + if (!r->space) return; + + dma_free_coherent(&macio_get_pci_dev(i2sdev->macio)->dev, + r->size, r->space, r->bus_addr); +} + +static void i2sbus_release_dev(struct device *dev) +{ + struct i2sbus_dev *i2sdev; + int i; + + i2sdev = container_of(dev, struct i2sbus_dev, sound.ofdev.dev); + iounmap(i2sdev->intfregs); + iounmap(i2sdev->out.dbdma); + iounmap(i2sdev->in.dbdma); + for (i = aoa_resource_i2smmio; i <= aoa_resource_rxdbdma; i++) + release_and_free_resource(i2sdev->allocated_resource[i]); + free_dbdma_descriptor_ring(i2sdev, &i2sdev->out.dbdma_ring); + free_dbdma_descriptor_ring(i2sdev, &i2sdev->in.dbdma_ring); + for (i = aoa_resource_i2smmio; i <= aoa_resource_rxdbdma; 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 i2sbus_dev *dev = devid; + u32 intreg; + + spin_lock(&dev->low_lock); + intreg = in_le32(&dev->intfregs->intr_ctl); + + /* acknowledge interrupt reasons */ + out_le32(&dev->intfregs->intr_ctl, intreg); + + spin_unlock(&dev->low_lock); + + return IRQ_HANDLED; +} + + +/* + * XXX FIXME: We test the layout_id's here to get the proper way of + * mapping in various registers, thanks to bugs in Apple device-trees. + * We could instead key off the machine model and the name of the i2s + * node (i2s-a). This we'll do when we move it all to macio_asic.c + * and have that export items for each sub-node too. + */ +static int i2sbus_get_and_fixup_rsrc(struct device_node *np, int index, + int layout, struct resource *res) +{ + struct device_node *parent; + int pindex, rc = -ENXIO; + const u32 *reg; + + /* Machines with layout 76 and 36 (K2 based) have a weird device + * tree what we need to special case. + * Normal machines just fetch the resource from the i2s-X node. + * Darwin further divides normal machines into old and new layouts + * with a subtely different code path but that doesn't seem necessary + * in practice, they just bloated it. In addition, even on our K2 + * case the i2s-modem node, if we ever want to handle it, uses the + * normal layout + */ + if (layout != 76 && layout != 36) + return of_address_to_resource(np, index, res); + + parent = of_get_parent(np); + pindex = (index == aoa_resource_i2smmio) ? 0 : 1; + rc = of_address_to_resource(parent, pindex, res); + if (rc) + goto bail; + reg = of_get_property(np, "reg", NULL); + if (reg == NULL) { + rc = -ENXIO; + goto bail; + } + res->start += reg[index * 2]; + res->end = res->start + reg[index * 2 + 1] - 1; + bail: + of_node_put(parent); + return rc; +} + +/* FIXME: look at device node refcounting */ +static int i2sbus_add_dev(struct macio_dev *macio, + struct i2sbus_control *control, + struct device_node *np) +{ + struct i2sbus_dev *dev; + struct device_node *child = NULL, *sound = NULL; + struct resource *r; + int i, layout = 0, rlen, ok = force; + static const char *rnames[] = { "i2sbus: %s (control)", + "i2sbus: %s (tx)", + "i2sbus: %s (rx)" }; + static irq_handler_t ints[] = { + i2sbus_bus_intr, + i2sbus_tx_intr, + i2sbus_rx_intr + }; + + if (strlen(np->name) != 5) + return 0; + if (strncmp(np->name, "i2s-", 4)) + 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) { + const u32 *id = of_get_property(sound, "layout-id", NULL); + + if (id) { + layout = *id; + snprintf(dev->sound.modalias, 32, + "sound-layout-%d", layout); + ok = 1; + } else { + id = of_get_property(sound, "device-id", NULL); + /* + * We probably cannot handle all device-id machines, + * so restrict to those we do handle for now. + */ + if (id && (*id == 22 || *id == 14 || *id == 35 || + *id == 31 || *id == 44)) { + snprintf(dev->sound.modalias, 32, + "aoa-device-id-%d", *id); + ok = 1; + layout = -1; + } + } + } + /* for the time being, until we can handle non-layout-id + * things in some fabric, refuse to attach if there is no + * layout-id property or we haven't been forced to attach. + * When there are two i2s busses and only one has a layout-id, + * then this depends on the order, but that isn't important + * either as the second one in that case is just a modem. */ + if (!ok) { + kfree(dev); + return -ENODEV; + } + + mutex_init(&dev->lock); + spin_lock_init(&dev->low_lock); + dev->sound.ofdev.archdata.dma_mask = macio->ofdev.archdata.dma_mask; + dev->sound.ofdev.dev.of_node = np; + dev->sound.ofdev.dev.dma_mask = &dev->sound.ofdev.archdata.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 = aoa_resource_i2smmio; i <= aoa_resource_rxdbdma; i++) { + dev->interrupts[i] = -1; + snprintf(dev->rnames[i], sizeof(dev->rnames[i]), + rnames[i], np->name); + } + for (i = aoa_resource_i2smmio; i <= aoa_resource_rxdbdma; i++) { + int irq = irq_of_parse_and_map(np, i); + if (request_irq(irq, ints[i], 0, dev->rnames[i], dev)) + goto err; + dev->interrupts[i] = irq; + } + + + /* Resource handling is problematic as some device-trees contain + * useless crap (ugh ugh ugh). We work around that here by calling + * specific functions for calculating the appropriate resources. + * + * This will all be moved to macio_asic.c at one point + */ + for (i = aoa_resource_i2smmio; i <= aoa_resource_rxdbdma; i++) { + if (i2sbus_get_and_fixup_rsrc(np,i,layout,&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, + resource_size(&dev->resources[i]), + dev->rnames[i]); + if (!dev->allocated_resource[i]) { + printk(KERN_ERR "i2sbus: failed to claim resource %d!\n", i); + goto err; + } + } + + r = &dev->resources[aoa_resource_i2smmio]; + rlen = resource_size(r); + if (rlen < sizeof(struct i2s_interface_regs)) + goto err; + dev->intfregs = ioremap(r->start, rlen); + + r = &dev->resources[aoa_resource_txdbdma]; + rlen = resource_size(r); + if (rlen < sizeof(struct dbdma_regs)) + goto err; + dev->out.dbdma = ioremap(r->start, rlen); + + r = &dev->resources[aoa_resource_rxdbdma]; + rlen = resource_size(r); + if (rlen < sizeof(struct dbdma_regs)) + goto err; + dev->in.dbdma = ioremap(r->start, rlen); + + 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); + iounmap(dev->intfregs); + iounmap(dev->out.dbdma); + iounmap(dev->in.dbdma); + for (i=0;i<3;i++) + release_and_free_resource(dev->allocated_resource[i]); + mutex_destroy(&dev->lock); + kfree(dev); + return 0; +} + +static int i2sbus_probe(struct macio_dev* dev, const struct of_device_id *match) +{ + struct device_node *np = NULL; + int got = 0, err; + struct i2sbus_control *control = NULL; + + err = i2sbus_control_init(dev, &control); + if (err) + return err; + if (!control) { + printk(KERN_ERR "i2sbus_control_init API breakage\n"); + return -ENODEV; + } + + while ((np = of_get_next_child(dev->ofdev.dev.of_node, np))) { + if (of_device_is_compatible(np, "i2sbus") || + of_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_set_drvdata(&dev->ofdev.dev, control); + + return 0; +} + +static int i2sbus_remove(struct macio_dev* dev) +{ + struct i2sbus_control *control = dev_get_drvdata(&dev->ofdev.dev); + 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_get_drvdata(&dev->ofdev.dev); + struct codec_info_item *cii; + struct i2sbus_dev* i2sdev; + int err, ret = 0; + + list_for_each_entry(i2sdev, &control->list, item) { + /* Notify Alsa */ + /* Suspend PCM streams */ + snd_pcm_suspend_all(i2sdev->sound.pcm); + + /* 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; + } + + /* wait until streams are stopped */ + i2sbus_wait_for_stop_both(i2sdev); + } + + return ret; +} + +static int i2sbus_resume(struct macio_dev* dev) +{ + struct i2sbus_control *control = dev_get_drvdata(&dev->ofdev.dev); + struct codec_info_item *cii; + struct i2sbus_dev* i2sdev; + int err, ret = 0; + + list_for_each_entry(i2sdev, &control->list, item) { + /* reset i2s bus format etc. */ + i2sbus_pcm_prepare_both(i2sdev); + + /* 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; + } + } + + return ret; +} +#endif /* CONFIG_PM */ + +static int i2sbus_shutdown(struct macio_dev* dev) +{ + return 0; +} + +static struct macio_driver i2sbus_drv = { + .driver = { + .name = "soundbus-i2s", + .owner = THIS_MODULE, + .of_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); diff --git a/sound/aoa/soundbus/i2sbus/i2sbus.h b/sound/aoa/soundbus/i2sbus/i2sbus.h new file mode 100644 index 000000000..befefd99e --- /dev/null +++ b/sound/aoa/soundbus/i2sbus/i2sbus.h @@ -0,0 +1,126 @@ +/* + * 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 <linux/interrupt.h> +#include <linux/spinlock.h> +#include <linux/mutex.h> +#include <linux/completion.h> + +#include <sound/pcm.h> + +#include <asm/prom.h> +#include <asm/pmac_feature.h> +#include <asm/dbdma.h> + +#include "interface.h" +#include "../soundbus.h" + +struct i2sbus_control { + struct list_head list; + struct macio_chip *macio; +}; + +#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; + u32 stopping: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 completion *stop_completion; +}; + +enum { + aoa_resource_i2smmio = 0, + aoa_resource_txdbdma, + aoa_resource_rxdbdma, +}; + +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); +extern irqreturn_t +i2sbus_rx_intr(int irq, void *devid); + +extern void i2sbus_wait_for_stop_both(struct i2sbus_dev *i2sdev); +extern void i2sbus_pcm_prepare_both(struct i2sbus_dev *i2sdev); + +/* 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 */ diff --git a/sound/aoa/soundbus/i2sbus/interface.h b/sound/aoa/soundbus/i2sbus/interface.h new file mode 100644 index 000000000..c6b5f5452 --- /dev/null +++ b/sound/aoa/soundbus/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 */ diff --git a/sound/aoa/soundbus/i2sbus/pcm.c b/sound/aoa/soundbus/i2sbus/pcm.c new file mode 100644 index 000000000..e61853175 --- /dev/null +++ b/sound/aoa/soundbus/i2sbus/pcm.c @@ -0,0 +1,1067 @@ +/* + * i2sbus driver -- pcm routines + * + * Copyright 2006 Johannes Berg <johannes@sipsolutions.net> + * + * GPL v2, can be found in COPYING. + */ + +#include <linux/io.h> +#include <linux/delay.h> +#include <linux/slab.h> +#include <sound/core.h> +#include <asm/macio.h> +#include <linux/pci.h> +#include <linux/module.h> +#include "../soundbus.h" +#include "i2sbus.h" + +static inline void get_pcm_info(struct i2sbus_dev *i2sdev, int in, + struct pcm_info **pi, struct pcm_info **other) +{ + if (in) { + if (pi) + *pi = &i2sdev->in; + if (other) + *other = &i2sdev->out; + } else { + if (pi) + *pi = &i2sdev->out; + if (other) + *other = &i2sdev->in; + } +} + +static int clock_and_divisors(int mclk, int sclk, int rate, int *out) +{ + /* sclk must be derived from mclk! */ + if (mclk % sclk) + return -1; + /* derive sclk register value */ + if (i2s_sf_sclkdiv(mclk / sclk, out)) + return -1; + + if (I2S_CLOCK_SPEED_18MHz % (rate * mclk) == 0) { + if (!i2s_sf_mclkdiv(I2S_CLOCK_SPEED_18MHz / (rate * mclk), out)) { + *out |= I2S_SF_CLOCK_SOURCE_18MHz; + return 0; + } + } + if (I2S_CLOCK_SPEED_45MHz % (rate * mclk) == 0) { + if (!i2s_sf_mclkdiv(I2S_CLOCK_SPEED_45MHz / (rate * mclk), out)) { + *out |= I2S_SF_CLOCK_SOURCE_45MHz; + return 0; + } + } + if (I2S_CLOCK_SPEED_49MHz % (rate * mclk) == 0) { + if (!i2s_sf_mclkdiv(I2S_CLOCK_SPEED_49MHz / (rate * mclk), out)) { + *out |= I2S_SF_CLOCK_SOURCE_49MHz; + return 0; + } + } + return -1; +} + +#define CHECK_RATE(rate) \ + do { if (rates & SNDRV_PCM_RATE_ ##rate) { \ + int dummy; \ + if (clock_and_divisors(sysclock_factor, \ + bus_factor, rate, &dummy)) \ + rates &= ~SNDRV_PCM_RATE_ ##rate; \ + } } while (0) + +static int i2sbus_pcm_open(struct i2sbus_dev *i2sdev, int in) +{ + struct pcm_info *pi, *other; + struct soundbus_dev *sdev; + int masks_inited = 0, err; + struct codec_info_item *cii, *rev; + struct snd_pcm_hardware *hw; + u64 formats = 0; + unsigned int rates = 0; + struct transfer_info v; + int result = 0; + int bus_factor = 0, sysclock_factor = 0; + int found_this; + + mutex_lock(&i2sdev->lock); + + get_pcm_info(i2sdev, in, &pi, &other); + + hw = &pi->substream->runtime->hw; + sdev = &i2sdev->sound; + + if (pi->active) { + /* alsa messed up */ + result = -EBUSY; + goto out_unlock; + } + + /* we now need to assign the hw */ + list_for_each_entry(cii, &sdev->codec_list, list) { + struct transfer_info *ti = cii->codec->transfers; + bus_factor = cii->codec->bus_factor; + sysclock_factor = cii->codec->sysclock_factor; + while (ti->formats && ti->rates) { + v = *ti; + if (ti->transfer_in == in + && cii->codec->usable(cii, ti, &v)) { + if (masks_inited) { + formats &= v.formats; + rates &= v.rates; + } else { + formats = v.formats; + rates = v.rates; + masks_inited = 1; + } + } + ti++; + } + } + if (!masks_inited || !bus_factor || !sysclock_factor) { + result = -ENODEV; + goto out_unlock; + } + /* bus dependent stuff */ + hw->info = SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID | + SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_RESUME | + SNDRV_PCM_INFO_JOINT_DUPLEX; + + 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 &= pcm_format_to_bits(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; + err = snd_pcm_hw_constraint_integer(pi->substream->runtime, + SNDRV_PCM_HW_PARAM_PERIODS); + if (err < 0) { + result = err; + goto out_unlock; + } + list_for_each_entry(cii, &sdev->codec_list, list) { + if (cii->codec->open) { + err = cii->codec->open(cii, pi->substream); + if (err) { + result = err; + /* unwind */ + found_this = 0; + list_for_each_entry_reverse(rev, + &sdev->codec_list, list) { + if (found_this && rev->codec->close) { + rev->codec->close(rev, + pi->substream); + } + if (rev == cii) + found_this = 1; + } + goto out_unlock; + } + } + } + + out_unlock: + mutex_unlock(&i2sdev->lock); + return result; +} + +#undef CHECK_RATE + +static int i2sbus_pcm_close(struct i2sbus_dev *i2sdev, int in) +{ + struct codec_info_item *cii; + struct pcm_info *pi; + int err = 0, tmp; + + mutex_lock(&i2sdev->lock); + + get_pcm_info(i2sdev, in, &pi, NULL); + + list_for_each_entry(cii, &i2sdev->sound.codec_list, list) { + if (cii->codec->close) { + tmp = cii->codec->close(cii, pi->substream); + if (tmp) + err = tmp; + } + } + + pi->substream = NULL; + pi->active = 0; + mutex_unlock(&i2sdev->lock); + return err; +} + +static void i2sbus_wait_for_stop(struct i2sbus_dev *i2sdev, + struct pcm_info *pi) +{ + unsigned long flags; + struct completion done; + long timeout; + + spin_lock_irqsave(&i2sdev->low_lock, flags); + if (pi->dbdma_ring.stopping) { + init_completion(&done); + pi->stop_completion = &done; + spin_unlock_irqrestore(&i2sdev->low_lock, flags); + timeout = wait_for_completion_timeout(&done, HZ); + spin_lock_irqsave(&i2sdev->low_lock, flags); + pi->stop_completion = NULL; + if (timeout == 0) { + /* timeout expired, stop dbdma forcefully */ + printk(KERN_ERR "i2sbus_wait_for_stop: timed out\n"); + /* make sure RUN, PAUSE and S0 bits are cleared */ + out_le32(&pi->dbdma->control, (RUN | PAUSE | 1) << 16); + pi->dbdma_ring.stopping = 0; + timeout = 10; + while (in_le32(&pi->dbdma->status) & ACTIVE) { + if (--timeout <= 0) + break; + udelay(1); + } + } + } + spin_unlock_irqrestore(&i2sdev->low_lock, flags); +} + +#ifdef CONFIG_PM +void i2sbus_wait_for_stop_both(struct i2sbus_dev *i2sdev) +{ + struct pcm_info *pi; + + get_pcm_info(i2sdev, 0, &pi, NULL); + i2sbus_wait_for_stop(i2sdev, pi); + get_pcm_info(i2sdev, 1, &pi, NULL); + i2sbus_wait_for_stop(i2sdev, pi); +} +#endif + +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 inline int i2sbus_hw_free(struct snd_pcm_substream *substream, int in) +{ + struct i2sbus_dev *i2sdev = snd_pcm_substream_chip(substream); + struct pcm_info *pi; + + get_pcm_info(i2sdev, in, &pi, NULL); + if (pi->dbdma_ring.stopping) + i2sbus_wait_for_stop(i2sdev, pi); + snd_pcm_lib_free_pages(substream); + return 0; +} + +static int i2sbus_playback_hw_free(struct snd_pcm_substream *substream) +{ + return i2sbus_hw_free(substream, 0); +} + +static int i2sbus_record_hw_free(struct snd_pcm_substream *substream) +{ + return i2sbus_hw_free(substream, 1); +} + +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, nperiods; + 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; + unsigned int cmd, stopaddr; + + mutex_lock(&i2sdev->lock); + + get_pcm_info(i2sdev, in, &pi, &other); + + if (pi->dbdma_ring.running) { + result = -EBUSY; + goto out_unlock; + } + if (pi->dbdma_ring.stopping) + i2sbus_wait_for_stop(i2sdev, pi); + + if (!pi->substream || !pi->substream->runtime) { + result = -EINVAL; + 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); + nperiods = pi->substream->runtime->periods; + pi->current_period = 0; + + /* generate dbdma command ring first */ + command = pi->dbdma_ring.cmds; + memset(command, 0, (nperiods + 2) * sizeof(struct dbdma_cmd)); + + /* commands to DMA to/from the ring */ + /* + * For input, we need to do a graceful stop; if we abort + * the DMA, we end up with leftover bytes that corrupt + * the next recording. To do this we set the S0 status + * bit and wait for the DMA controller to stop. Each + * command has a branch condition to + * make it branch to a stop command if S0 is set. + * On input we also need to wait for the S7 bit to be + * set before turning off the DMA controller. + * In fact we do the graceful stop for output as well. + */ + offset = runtime->dma_addr; + cmd = (in? INPUT_MORE: OUTPUT_MORE) | BR_IFSET | INTR_ALWAYS; + stopaddr = pi->dbdma_ring.bus_cmd_start + + (nperiods + 1) * sizeof(struct dbdma_cmd); + for (i = 0; i < nperiods; i++, command++, offset += periodsize) { + command->command = cpu_to_le16(cmd); + command->cmd_dep = cpu_to_le32(stopaddr); + command->phy_addr = cpu_to_le32(offset); + command->req_count = cpu_to_le16(periodsize); + } + + /* branch back to beginning of ring */ + command->command = cpu_to_le16(DBDMA_NOP | BR_ALWAYS); + command->cmd_dep = cpu_to_le32(pi->dbdma_ring.bus_cmd_start); + command++; + + /* set stop command */ + command->command = cpu_to_le16(DBDMA_STOP); + + /* 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; +} + +#ifdef CONFIG_PM +void i2sbus_pcm_prepare_both(struct i2sbus_dev *i2sdev) +{ + i2sbus_pcm_prepare(i2sdev, 0); + i2sbus_pcm_prepare(i2sdev, 1); +} +#endif + +static int i2sbus_pcm_trigger(struct i2sbus_dev *i2sdev, int in, int cmd) +{ + struct codec_info_item *cii; + struct pcm_info *pi; + 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; + + if (pi->dbdma_ring.stopping) { + /* Clear the S0 bit, then see if we stopped yet */ + out_le32(&pi->dbdma->control, 1 << 16); + if (in_le32(&pi->dbdma->status) & ACTIVE) { + /* possible race here? */ + udelay(10); + if (in_le32(&pi->dbdma->status) & ACTIVE) { + pi->dbdma_ring.stopping = 0; + goto out_unlock; /* keep running */ + } + } + } + + /* make sure RUN, PAUSE and S0 bits are cleared */ + out_le32(&pi->dbdma->control, (RUN | PAUSE | 1) << 16); + + /* set branch condition select register */ + out_le32(&pi->dbdma->br_sel, (1 << 16) | 1); + + /* write dma command buffer address to the dbdma chip */ + out_le32(&pi->dbdma->cmdptr, pi->dbdma_ring.bus_cmd_start); + + /* initialize the frame count and current period */ + pi->current_period = 0; + pi->frame_count = in_le32(&i2sdev->intfregs->frame_count); + + /* set the DMA controller running */ + out_le32(&pi->dbdma->control, (RUN << 16) | RUN); + + /* off you go! */ + break; + + case SNDRV_PCM_TRIGGER_STOP: + case SNDRV_PCM_TRIGGER_SUSPEND: + if (!pi->dbdma_ring.running) { + result = -EALREADY; + goto out_unlock; + } + pi->dbdma_ring.running = 0; + + /* Set the S0 bit to make the DMA branch to the stop cmd */ + out_le32(&pi->dbdma->control, (1 << 16) | 1); + pi->dbdma_ring.stopping = 1; + + 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; + + if (fc >= pi->substream->runtime->buffer_size) + fc %= pi->substream->runtime->buffer_size; + return fc; +} + +static inline void handle_interrupt(struct i2sbus_dev *i2sdev, int in) +{ + struct pcm_info *pi; + u32 fc, nframes; + u32 status; + int timeout, i; + int dma_stopped = 0; + struct snd_pcm_runtime *runtime; + + spin_lock(&i2sdev->low_lock); + get_pcm_info(i2sdev, in, &pi, NULL); + if (!pi->dbdma_ring.running && !pi->dbdma_ring.stopping) + goto out_unlock; + + i = pi->current_period; + runtime = pi->substream->runtime; + while (pi->dbdma_ring.cmds[i].xfer_status) { + if (le16_to_cpu(pi->dbdma_ring.cmds[i].xfer_status) & BT) + /* + * BT is the branch taken bit. If it took a branch + * it is because we set the S0 bit to make it + * branch to the stop command. + */ + dma_stopped = 1; + pi->dbdma_ring.cmds[i].xfer_status = 0; + + if (++i >= runtime->periods) { + i = 0; + pi->frame_count += runtime->buffer_size; + } + pi->current_period = i; + + /* + * Check the frame count. The DMA tends to get a bit + * ahead of the frame counter, which confuses the core. + */ + fc = in_le32(&i2sdev->intfregs->frame_count); + nframes = i * runtime->period_size; + if (fc < pi->frame_count + nframes) + pi->frame_count = fc - nframes; + } + + if (dma_stopped) { + timeout = 1000; + for (;;) { + status = in_le32(&pi->dbdma->status); + if (!(status & ACTIVE) && (!in || (status & 0x80))) + break; + if (--timeout <= 0) { + printk(KERN_ERR "i2sbus: timed out " + "waiting for DMA to stop!\n"); + break; + } + udelay(1); + } + + /* Turn off DMA controller, clear S0 bit */ + out_le32(&pi->dbdma->control, (RUN | PAUSE | 1) << 16); + + pi->dbdma_ring.stopping = 0; + if (pi->stop_completion) + complete(pi->stop_completion); + } + + if (!pi->dbdma_ring.running) + goto out_unlock; + 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) +{ + handle_interrupt((struct i2sbus_dev *)devid, 0); + return IRQ_HANDLED; +} + +irqreturn_t i2sbus_rx_intr(int irq, void *devid) +{ + 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 const 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_playback_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 const 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_record_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); +} + +int +i2sbus_attach_codec(struct soundbus_dev *dev, struct snd_card *card, + struct codec_info *ci, void *data) +{ + int err, in = 0, out = 0; + struct transfer_info *tmp; + struct i2sbus_dev *i2sdev = soundbus_dev_to_i2sbus_dev(dev); + struct codec_info_item *cii; + + if (!dev->pcmname || dev->pcmid == -1) { + printk(KERN_ERR "i2sbus: pcm name and id must be set!\n"); + return -EINVAL; + } + + list_for_each_entry(cii, &dev->codec_list, list) { + if (cii->codec_data == data) + return -EALREADY; + } + + if (!ci->transfers || !ci->transfers->formats + || !ci->transfers->rates || !ci->usable) + return -EINVAL; + + /* we currently code the i2s transfer on the clock, and support only + * 32 and 64 */ + if (ci->bus_factor != 32 && ci->bus_factor != 64) + return -EINVAL; + + /* If you want to fix this, you need to keep track of what transport infos + * are to be used, which codecs they belong to, and then fix all the + * sysclock/busclock stuff above to depend on which is usable */ + list_for_each_entry(cii, &dev->codec_list, list) { + if (cii->codec->sysclock_factor != ci->sysclock_factor) { + printk(KERN_DEBUG + "cannot yet handle multiple different sysclocks!\n"); + return -EINVAL; + } + if (cii->codec->bus_factor != ci->bus_factor) { + printk(KERN_DEBUG + "cannot yet handle multiple different bus clocks!\n"); + return -EINVAL; + } + } + + tmp = ci->transfers; + while (tmp->formats && tmp->rates) { + if (tmp->transfer_in) + in = 1; + else + out = 1; + tmp++; + } + + cii = kzalloc(sizeof(struct codec_info_item), GFP_KERNEL); + if (!cii) { + printk(KERN_DEBUG "i2sbus: failed to allocate cii\n"); + return -ENOMEM; + } + + /* use the private data to point to the codec info */ + cii->sdev = soundbus_dev_get(dev); + cii->codec = ci; + cii->codec_data = data; + + if (!cii->sdev) { + printk(KERN_DEBUG + "i2sbus: failed to get soundbus dev reference\n"); + err = -ENODEV; + goto out_free_cii; + } + + if (!try_module_get(THIS_MODULE)) { + printk(KERN_DEBUG "i2sbus: failed to get module reference!\n"); + err = -EBUSY; + goto out_put_sdev; + } + + if (!try_module_get(ci->owner)) { + printk(KERN_DEBUG + "i2sbus: failed to get module reference to codec owner!\n"); + err = -EBUSY; + goto out_put_this_module; + } + + 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"); + goto out_put_ci_module; + } + } + + /* 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"); + err = -EINVAL; + goto out_put_ci_module; + } + err = snd_pcm_new_stream(dev->pcm, SNDRV_PCM_STREAM_PLAYBACK, 1); + if (err) + goto out_put_ci_module; + snd_pcm_set_ops(dev->pcm, SNDRV_PCM_STREAM_PLAYBACK, + &i2sbus_playback_ops); + dev->pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].dev.parent = + &dev->ofdev.dev; + 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"); + err = -EINVAL; + goto out_put_ci_module; + } + err = snd_pcm_new_stream(dev->pcm, SNDRV_PCM_STREAM_CAPTURE, 1); + if (err) + goto out_put_ci_module; + snd_pcm_set_ops(dev->pcm, SNDRV_PCM_STREAM_CAPTURE, + &i2sbus_record_ops); + dev->pcm->streams[SNDRV_PCM_STREAM_CAPTURE].dev.parent = + &dev->ofdev.dev; + 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"); + goto out_put_ci_module; + } + /* no errors any more, so let's add this to our list */ + list_add(&cii->list, &dev->codec_list); + + dev->pcm->private_data = i2sdev; + dev->pcm->private_free = i2sbus_private_free; + + /* well, we really should support scatter/gather DMA */ + snd_pcm_lib_preallocate_pages_for_all( + dev->pcm, SNDRV_DMA_TYPE_DEV, + snd_dma_pci_data(macio_get_pci_dev(i2sdev->macio)), + 64 * 1024, 64 * 1024); + + return 0; + out_put_ci_module: + module_put(ci->owner); + out_put_this_module: + module_put(THIS_MODULE); + out_put_sdev: + soundbus_dev_put(dev); + out_free_cii: + kfree(cii); + return err; +} + +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); + } +} diff --git a/sound/aoa/soundbus/soundbus.h b/sound/aoa/soundbus/soundbus.h new file mode 100644 index 000000000..ae4022438 --- /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 <linux/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 platform_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 (*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); + +extern struct attribute *soundbus_dev_attrs[]; + +#endif /* __SOUNDBUS_H */ diff --git a/sound/aoa/soundbus/sysfs.c b/sound/aoa/soundbus/sysfs.c new file mode 100644 index 000000000..81da020bd --- /dev/null +++ b/sound/aoa/soundbus/sysfs.c @@ -0,0 +1,46 @@ +// SPDX-License-Identifier: GPL-2.0 +#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.dev.of_node->field); \ +} + +static ssize_t modalias_show(struct device *dev, struct device_attribute *attr, + char *buf) +{ + struct soundbus_dev *sdev = to_soundbus_device(dev); + struct platform_device *of = &sdev->ofdev; + int length; + + if (*sdev->modalias) { + strlcpy(buf, sdev->modalias, sizeof(sdev->modalias) + 1); + strcat(buf, "\n"); + length = strlen(buf); + } else { + length = sprintf(buf, "of:N%sT%s\n", + of->dev.of_node->name, of->dev.of_node->type); + } + + return length; +} +static DEVICE_ATTR_RO(modalias); + +soundbus_config_of_attr (name, "%s\n"); +static DEVICE_ATTR_RO(name); +soundbus_config_of_attr (type, "%s\n"); +static DEVICE_ATTR_RO(type); + +struct attribute *soundbus_dev_attrs[] = { + &dev_attr_name.attr, + &dev_attr_type.attr, + &dev_attr_modalias.attr, + NULL, +}; |