From 74426fbff66eea8e8d1f42c8238c268d1e63a832 Mon Sep 17 00:00:00 2001 From: Robert Jarzmik Date: Sat, 2 Sep 2017 21:54:04 +0200 Subject: ALSA: ac97: add an ac97 bus AC97 is a bus for sound usage. It enables for a AC97 AC-Link to link one controller to 0 to 4 AC97 codecs. The goal of this new implementation is to implement a device/driver model for AC97, with an automatic scan of the bus and automatic discovery of AC97 codec devices. Signed-off-by: Robert Jarzmik Reviewed-by: Takashi Iwai Signed-off-by: Mark Brown --- sound/ac97/Kconfig | 19 ++ sound/ac97/Makefile | 8 + sound/ac97/ac97_core.h | 16 ++ sound/ac97/bus.c | 539 +++++++++++++++++++++++++++++++++++++++++++ sound/ac97/codec.c | 15 ++ sound/ac97/snd_ac97_compat.c | 108 +++++++++ 6 files changed, 705 insertions(+) create mode 100644 sound/ac97/Kconfig create mode 100644 sound/ac97/Makefile create mode 100644 sound/ac97/ac97_core.h create mode 100644 sound/ac97/bus.c create mode 100644 sound/ac97/codec.c create mode 100644 sound/ac97/snd_ac97_compat.c (limited to 'sound') diff --git a/sound/ac97/Kconfig b/sound/ac97/Kconfig new file mode 100644 index 000000000000..f8a64e15e5bf --- /dev/null +++ b/sound/ac97/Kconfig @@ -0,0 +1,19 @@ +# +# AC97 configuration +# + + +config AC97_BUS_NEW + tristate + select AC97 + help + This is the new AC97 bus type, successor of AC97_BUS. The ported + drivers which benefit from the AC97 automatic probing should "select" + this instead of the AC97_BUS. + Say Y here if you want to have AC97 devices, which are sound oriented + devices around an AC-Link. + +config AC97_BUS_COMPAT + bool + depends on AC97_BUS_NEW + depends on !AC97_BUS diff --git a/sound/ac97/Makefile b/sound/ac97/Makefile new file mode 100644 index 000000000000..f9c2640bfb59 --- /dev/null +++ b/sound/ac97/Makefile @@ -0,0 +1,8 @@ +# +# make for AC97 bus drivers +# + +obj-$(CONFIG_AC97_BUS_NEW) += ac97.o + +ac97-y += bus.o codec.o +ac97-$(CONFIG_AC97_BUS_COMPAT) += snd_ac97_compat.o diff --git a/sound/ac97/ac97_core.h b/sound/ac97/ac97_core.h new file mode 100644 index 000000000000..08441a4fda7c --- /dev/null +++ b/sound/ac97/ac97_core.h @@ -0,0 +1,16 @@ +/* + * Copyright (C) 2016 Robert Jarzmik + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +unsigned int snd_ac97_bus_scan_one(struct ac97_controller *ac97, + unsigned int codec_num); + +static inline bool ac97_ids_match(unsigned int id1, unsigned int id2, + unsigned int mask) +{ + return (id1 & mask) == (id2 & mask); +} diff --git a/sound/ac97/bus.c b/sound/ac97/bus.c new file mode 100644 index 000000000000..31f858eceffc --- /dev/null +++ b/sound/ac97/bus.c @@ -0,0 +1,539 @@ +/* + * Copyright (C) 2016 Robert Jarzmik + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include + +#include "ac97_core.h" + +/* + * Protects ac97_controllers and each ac97_controller structure. + */ +static DEFINE_MUTEX(ac97_controllers_mutex); +static DEFINE_IDR(ac97_adapter_idr); +static LIST_HEAD(ac97_controllers); + +static struct bus_type ac97_bus_type; + +static inline struct ac97_controller* +to_ac97_controller(struct device *ac97_adapter) +{ + return container_of(ac97_adapter, struct ac97_controller, adap); +} + +static int ac97_unbound_ctrl_write(struct ac97_controller *adrv, int slot, + unsigned short reg, unsigned short val) +{ + return -ENODEV; +} + +static int ac97_unbound_ctrl_read(struct ac97_controller *adrv, int slot, + unsigned short reg) +{ + return -ENODEV; +} + +static const struct ac97_controller_ops ac97_unbound_ctrl_ops = { + .write = ac97_unbound_ctrl_write, + .read = ac97_unbound_ctrl_read, +}; + +static struct ac97_controller ac97_unbound_ctrl = { + .ops = &ac97_unbound_ctrl_ops, +}; + +static struct ac97_codec_device * +ac97_codec_find(struct ac97_controller *ac97_ctrl, unsigned int codec_num) +{ + if (codec_num >= AC97_BUS_MAX_CODECS) + return ERR_PTR(-EINVAL); + + return ac97_ctrl->codecs[codec_num]; +} + +static void ac97_codec_release(struct device *dev) +{ + struct ac97_codec_device *adev; + struct ac97_controller *ac97_ctrl; + + adev = to_ac97_device(dev); + ac97_ctrl = adev->ac97_ctrl; + ac97_ctrl->codecs[adev->num] = NULL; + kfree(adev); +} + +static int ac97_codec_add(struct ac97_controller *ac97_ctrl, int idx, + unsigned int vendor_id) +{ + struct ac97_codec_device *codec; + int ret; + + codec = kzalloc(sizeof(*codec), GFP_KERNEL); + if (!codec) + return -ENOMEM; + ac97_ctrl->codecs[idx] = codec; + codec->vendor_id = vendor_id; + codec->dev.release = ac97_codec_release; + codec->dev.bus = &ac97_bus_type; + codec->dev.parent = &ac97_ctrl->adap; + codec->num = idx; + codec->ac97_ctrl = ac97_ctrl; + + device_initialize(&codec->dev); + dev_set_name(&codec->dev, "%s:%u", dev_name(ac97_ctrl->parent), idx); + + ret = device_add(&codec->dev); + if (ret) + goto err_free_codec; + + return 0; +err_free_codec: + put_device(&codec->dev); + kfree(codec); + ac97_ctrl->codecs[idx] = NULL; + + return ret; +} + +unsigned int snd_ac97_bus_scan_one(struct ac97_controller *adrv, + unsigned int codec_num) +{ + unsigned short vid1, vid2; + int ret; + + ret = adrv->ops->read(adrv, codec_num, AC97_VENDOR_ID1); + vid1 = (ret & 0xffff); + if (ret < 0) + return 0; + + ret = adrv->ops->read(adrv, codec_num, AC97_VENDOR_ID2); + vid2 = (ret & 0xffff); + if (ret < 0) + return 0; + + dev_dbg(&adrv->adap, "%s(codec_num=%u): vendor_id=0x%08x\n", + __func__, codec_num, AC97_ID(vid1, vid2)); + return AC97_ID(vid1, vid2); +} + +static int ac97_bus_scan(struct ac97_controller *ac97_ctrl) +{ + int ret, i; + unsigned int vendor_id; + + for (i = 0; i < AC97_BUS_MAX_CODECS; i++) { + if (ac97_codec_find(ac97_ctrl, i)) + continue; + if (!(ac97_ctrl->slots_available & BIT(i))) + continue; + vendor_id = snd_ac97_bus_scan_one(ac97_ctrl, i); + if (!vendor_id) + continue; + + ret = ac97_codec_add(ac97_ctrl, i, vendor_id); + if (ret < 0) + return ret; + } + return 0; +} + +static int ac97_bus_reset(struct ac97_controller *ac97_ctrl) +{ + ac97_ctrl->ops->reset(ac97_ctrl); + + return 0; +} + +/** + * snd_ac97_codec_driver_register - register an AC97 codec driver + * @dev: AC97 driver codec to register + * + * Register an AC97 codec driver to the ac97 bus driver, aka. the AC97 digital + * controller. + * + * Returns 0 on success or error code + */ +int snd_ac97_codec_driver_register(struct ac97_codec_driver *drv) +{ + drv->driver.bus = &ac97_bus_type; + return driver_register(&drv->driver); +} +EXPORT_SYMBOL_GPL(snd_ac97_codec_driver_register); + +/** + * snd_ac97_codec_driver_unregister - unregister an AC97 codec driver + * @dev: AC97 codec driver to unregister + * + * Unregister a previously registered ac97 codec driver. + */ +void snd_ac97_codec_driver_unregister(struct ac97_codec_driver *drv) +{ + driver_unregister(&drv->driver); +} +EXPORT_SYMBOL_GPL(snd_ac97_codec_driver_unregister); + +/** + * snd_ac97_codec_get_platdata - get platform_data + * @adev: the ac97 codec device + * + * For legacy platforms, in order to have platform_data in codec drivers + * available, while ac97 device are auto-created upon probe, this retrieves the + * platdata which was setup on ac97 controller registration. + * + * Returns the platform data pointer + */ +void *snd_ac97_codec_get_platdata(const struct ac97_codec_device *adev) +{ + struct ac97_controller *ac97_ctrl = adev->ac97_ctrl; + + return ac97_ctrl->codecs_pdata[adev->num]; +} +EXPORT_SYMBOL_GPL(snd_ac97_codec_get_platdata); + +static void ac97_ctrl_codecs_unregister(struct ac97_controller *ac97_ctrl) +{ + int i; + + for (i = 0; i < AC97_BUS_MAX_CODECS; i++) + if (ac97_ctrl->codecs[i]) { + ac97_ctrl->codecs[i]->ac97_ctrl = &ac97_unbound_ctrl; + device_unregister(&ac97_ctrl->codecs[i]->dev); + } +} + +static ssize_t cold_reset_store(struct device *dev, + struct device_attribute *attr, const char *buf, + size_t len) +{ + struct ac97_controller *ac97_ctrl; + + mutex_lock(&ac97_controllers_mutex); + ac97_ctrl = to_ac97_controller(dev); + ac97_ctrl->ops->reset(ac97_ctrl); + mutex_unlock(&ac97_controllers_mutex); + return len; +} +static DEVICE_ATTR_WO(cold_reset); + +static ssize_t warm_reset_store(struct device *dev, + struct device_attribute *attr, const char *buf, + size_t len) +{ + struct ac97_controller *ac97_ctrl; + + if (!dev) + return -ENODEV; + + mutex_lock(&ac97_controllers_mutex); + ac97_ctrl = to_ac97_controller(dev); + ac97_ctrl->ops->warm_reset(ac97_ctrl); + mutex_unlock(&ac97_controllers_mutex); + return len; +} +static DEVICE_ATTR_WO(warm_reset); + +static struct attribute *ac97_controller_device_attrs[] = { + &dev_attr_cold_reset.attr, + &dev_attr_warm_reset.attr, + NULL +}; + +static struct attribute_group ac97_adapter_attr_group = { + .name = "ac97_operations", + .attrs = ac97_controller_device_attrs, +}; + +static const struct attribute_group *ac97_adapter_groups[] = { + &ac97_adapter_attr_group, + NULL, +}; + +static void ac97_del_adapter(struct ac97_controller *ac97_ctrl) +{ + mutex_lock(&ac97_controllers_mutex); + ac97_ctrl_codecs_unregister(ac97_ctrl); + list_del(&ac97_ctrl->controllers); + mutex_unlock(&ac97_controllers_mutex); + + device_unregister(&ac97_ctrl->adap); +} + +static void ac97_adapter_release(struct device *dev) +{ + struct ac97_controller *ac97_ctrl; + + ac97_ctrl = to_ac97_controller(dev); + idr_remove(&ac97_adapter_idr, ac97_ctrl->nr); + dev_dbg(&ac97_ctrl->adap, "adapter unregistered by %s\n", + dev_name(ac97_ctrl->parent)); +} + +static const struct device_type ac97_adapter_type = { + .groups = ac97_adapter_groups, + .release = ac97_adapter_release, +}; + +static int ac97_add_adapter(struct ac97_controller *ac97_ctrl) +{ + int ret; + + mutex_lock(&ac97_controllers_mutex); + ret = idr_alloc(&ac97_adapter_idr, ac97_ctrl, 0, 0, GFP_KERNEL); + ac97_ctrl->nr = ret; + if (ret >= 0) { + dev_set_name(&ac97_ctrl->adap, "ac97-%d", ret); + ac97_ctrl->adap.type = &ac97_adapter_type; + ac97_ctrl->adap.parent = ac97_ctrl->parent; + ret = device_register(&ac97_ctrl->adap); + if (ret) + put_device(&ac97_ctrl->adap); + } + if (!ret) + list_add(&ac97_ctrl->controllers, &ac97_controllers); + mutex_unlock(&ac97_controllers_mutex); + + if (!ret) + dev_dbg(&ac97_ctrl->adap, "adapter registered by %s\n", + dev_name(ac97_ctrl->parent)); + return ret; +} + +/** + * snd_ac97_controller_register - register an ac97 controller + * @ops: the ac97 bus operations + * @dev: the device providing the ac97 DC function + * @slots_available: mask of the ac97 codecs that can be scanned and probed + * bit0 => codec 0, bit1 => codec 1 ... bit 3 => codec 3 + * + * Register a digital controller which can control up to 4 ac97 codecs. This is + * the controller side of the AC97 AC-link, while the slave side are the codecs. + * + * Returns a valid controller upon success, negative pointer value upon error + */ +struct ac97_controller *snd_ac97_controller_register( + const struct ac97_controller_ops *ops, struct device *dev, + unsigned short slots_available, void **codecs_pdata) +{ + struct ac97_controller *ac97_ctrl; + int ret, i; + + ac97_ctrl = kzalloc(sizeof(*ac97_ctrl), GFP_KERNEL); + if (!ac97_ctrl) + return ERR_PTR(-ENOMEM); + + for (i = 0; i < AC97_BUS_MAX_CODECS && codecs_pdata; i++) + ac97_ctrl->codecs_pdata[i] = codecs_pdata[i]; + + ac97_ctrl->ops = ops; + ac97_ctrl->slots_available = slots_available; + ac97_ctrl->parent = dev; + ret = ac97_add_adapter(ac97_ctrl); + + if (ret) + goto err; + ac97_bus_reset(ac97_ctrl); + ac97_bus_scan(ac97_ctrl); + + return ac97_ctrl; +err: + kfree(ac97_ctrl); + return ERR_PTR(ret); +} +EXPORT_SYMBOL_GPL(snd_ac97_controller_register); + +/** + * snd_ac97_controller_unregister - unregister an ac97 controller + * @ac97_ctrl: the device previously provided to ac97_controller_register() + * + */ +void snd_ac97_controller_unregister(struct ac97_controller *ac97_ctrl) +{ + ac97_del_adapter(ac97_ctrl); +} +EXPORT_SYMBOL_GPL(snd_ac97_controller_unregister); + +#ifdef CONFIG_PM +static int ac97_pm_runtime_suspend(struct device *dev) +{ + struct ac97_codec_device *codec = to_ac97_device(dev); + int ret = pm_generic_runtime_suspend(dev); + + if (ret == 0 && dev->driver) { + if (pm_runtime_is_irq_safe(dev)) + clk_disable(codec->clk); + else + clk_disable_unprepare(codec->clk); + } + + return ret; +} + +static int ac97_pm_runtime_resume(struct device *dev) +{ + struct ac97_codec_device *codec = to_ac97_device(dev); + int ret; + + if (dev->driver) { + if (pm_runtime_is_irq_safe(dev)) + ret = clk_enable(codec->clk); + else + ret = clk_prepare_enable(codec->clk); + if (ret) + return ret; + } + + return pm_generic_runtime_resume(dev); +} +#endif /* CONFIG_PM */ + +static const struct dev_pm_ops ac97_pm = { + .suspend = pm_generic_suspend, + .resume = pm_generic_resume, + .freeze = pm_generic_freeze, + .thaw = pm_generic_thaw, + .poweroff = pm_generic_poweroff, + .restore = pm_generic_restore, + SET_RUNTIME_PM_OPS( + ac97_pm_runtime_suspend, + ac97_pm_runtime_resume, + NULL) +}; + +static int ac97_get_enable_clk(struct ac97_codec_device *adev) +{ + int ret; + + adev->clk = clk_get(&adev->dev, "ac97_clk"); + if (IS_ERR(adev->clk)) + return PTR_ERR(adev->clk); + + ret = clk_prepare_enable(adev->clk); + if (ret) + clk_put(adev->clk); + + return ret; +} + +static void ac97_put_disable_clk(struct ac97_codec_device *adev) +{ + clk_disable_unprepare(adev->clk); + clk_put(adev->clk); +} + +static ssize_t vendor_id_show(struct device *dev, + struct device_attribute *attr, char *buf) +{ + struct ac97_codec_device *codec = to_ac97_device(dev); + + return sprintf(buf, "%08x", codec->vendor_id); +} +DEVICE_ATTR_RO(vendor_id); + +static struct attribute *ac97_dev_attrs[] = { + &dev_attr_vendor_id.attr, + NULL, +}; +ATTRIBUTE_GROUPS(ac97_dev); + +static int ac97_bus_match(struct device *dev, struct device_driver *drv) +{ + struct ac97_codec_device *adev = to_ac97_device(dev); + struct ac97_codec_driver *adrv = to_ac97_driver(drv); + const struct ac97_id *id = adrv->id_table; + int i = 0; + + if (adev->vendor_id == 0x0 || adev->vendor_id == 0xffffffff) + return false; + + do { + if (ac97_ids_match(id[i].id, adev->vendor_id, id[i].mask)) + return true; + } while (id[i++].id); + + return false; +} + +static int ac97_bus_probe(struct device *dev) +{ + struct ac97_codec_device *adev = to_ac97_device(dev); + struct ac97_codec_driver *adrv = to_ac97_driver(dev->driver); + int ret; + + ret = ac97_get_enable_clk(adev); + if (ret) + return ret; + + pm_runtime_get_noresume(dev); + pm_runtime_set_active(dev); + pm_runtime_enable(dev); + + ret = adrv->probe(adev); + if (ret == 0) + return 0; + + pm_runtime_disable(dev); + pm_runtime_set_suspended(dev); + pm_runtime_put_noidle(dev); + ac97_put_disable_clk(adev); + + return ret; +} + +static int ac97_bus_remove(struct device *dev) +{ + struct ac97_codec_device *adev = to_ac97_device(dev); + struct ac97_codec_driver *adrv = to_ac97_driver(dev->driver); + int ret; + + ret = pm_runtime_get_sync(dev); + if (ret) + return ret; + + ret = adrv->remove(adev); + pm_runtime_put_noidle(dev); + if (ret == 0) + ac97_put_disable_clk(adev); + + return ret; +} + +static struct bus_type ac97_bus_type = { + .name = "ac97bus", + .dev_groups = ac97_dev_groups, + .match = ac97_bus_match, + .pm = &ac97_pm, + .probe = ac97_bus_probe, + .remove = ac97_bus_remove, +}; + +static int __init ac97_bus_init(void) +{ + return bus_register(&ac97_bus_type); +} +subsys_initcall(ac97_bus_init); + +static void __exit ac97_bus_exit(void) +{ + bus_unregister(&ac97_bus_type); +} +module_exit(ac97_bus_exit); + +MODULE_LICENSE("GPL"); +MODULE_AUTHOR("Robert Jarzmik "); diff --git a/sound/ac97/codec.c b/sound/ac97/codec.c new file mode 100644 index 000000000000..a835f03744bf --- /dev/null +++ b/sound/ac97/codec.c @@ -0,0 +1,15 @@ +/* + * Copyright (C) 2016 Robert Jarzmik + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#include +#include +#include +#include +#include +#include /* For compat_ac97_* */ + diff --git a/sound/ac97/snd_ac97_compat.c b/sound/ac97/snd_ac97_compat.c new file mode 100644 index 000000000000..61544e0d8de4 --- /dev/null +++ b/sound/ac97/snd_ac97_compat.c @@ -0,0 +1,108 @@ +/* + * Copyright (C) 2016 Robert Jarzmik + * + * This program is free software; you can redistribute it and/or modify + * it under the terms of the GNU General Public License version 2 as + * published by the Free Software Foundation. + */ + +#include +#include +#include +#include +#include +#include + +#include "ac97_core.h" + +static void compat_ac97_reset(struct snd_ac97 *ac97) +{ + struct ac97_codec_device *adev = to_ac97_device(ac97->private_data); + struct ac97_controller *actrl = adev->ac97_ctrl; + + if (actrl->ops->reset) + actrl->ops->reset(actrl); +} + +static void compat_ac97_warm_reset(struct snd_ac97 *ac97) +{ + struct ac97_codec_device *adev = to_ac97_device(ac97->private_data); + struct ac97_controller *actrl = adev->ac97_ctrl; + + if (actrl->ops->warm_reset) + actrl->ops->warm_reset(actrl); +} + +static void compat_ac97_write(struct snd_ac97 *ac97, unsigned short reg, + unsigned short val) +{ + struct ac97_codec_device *adev = to_ac97_device(ac97->private_data); + struct ac97_controller *actrl = adev->ac97_ctrl; + + actrl->ops->write(actrl, ac97->num, reg, val); +} + +static unsigned short compat_ac97_read(struct snd_ac97 *ac97, + unsigned short reg) +{ + struct ac97_codec_device *adev = to_ac97_device(ac97->private_data); + struct ac97_controller *actrl = adev->ac97_ctrl; + + return actrl->ops->read(actrl, ac97->num, reg); +} + +static struct snd_ac97_bus_ops compat_snd_ac97_bus_ops = { + .reset = compat_ac97_reset, + .warm_reset = compat_ac97_warm_reset, + .write = compat_ac97_write, + .read = compat_ac97_read, +}; + +static struct snd_ac97_bus compat_soc_ac97_bus = { + .ops = &compat_snd_ac97_bus_ops, +}; + +struct snd_ac97 *snd_ac97_compat_alloc(struct ac97_codec_device *adev) +{ + struct snd_ac97 *ac97; + + ac97 = kzalloc(sizeof(struct snd_ac97), GFP_KERNEL); + if (ac97 == NULL) + return ERR_PTR(-ENOMEM); + + ac97->dev = adev->dev; + ac97->private_data = adev; + ac97->bus = &compat_soc_ac97_bus; + return ac97; +} +EXPORT_SYMBOL_GPL(snd_ac97_compat_alloc); + +void snd_ac97_compat_release(struct snd_ac97 *ac97) +{ + kfree(ac97); +} +EXPORT_SYMBOL_GPL(snd_ac97_compat_release); + +int snd_ac97_reset(struct snd_ac97 *ac97, bool try_warm, unsigned int id, + unsigned int id_mask) +{ + struct ac97_codec_device *adev = to_ac97_device(ac97->private_data); + struct ac97_controller *actrl = adev->ac97_ctrl; + unsigned int scanned; + + if (try_warm) { + compat_ac97_warm_reset(ac97); + scanned = snd_ac97_bus_scan_one(actrl, adev->num); + if (ac97_ids_match(scanned, adev->vendor_id, id_mask)) + return 1; + } + + compat_ac97_reset(ac97); + compat_ac97_warm_reset(ac97); + scanned = snd_ac97_bus_scan_one(actrl, adev->num); + if (ac97_ids_match(scanned, adev->vendor_id, id_mask)) + return 0; + + return -ENODEV; +} +EXPORT_SYMBOL_GPL(snd_ac97_reset); -- cgit v1.2.3 From 8d43344108c9945456128b75b69beee594b64ed6 Mon Sep 17 00:00:00 2001 From: Robert Jarzmik Date: Sat, 2 Sep 2017 21:54:05 +0200 Subject: ASoC: add new ac97 bus support Add the new ac97 bus support, with ac97 bus automatic probing. Signed-off-by: Robert Jarzmik Acked-by: Charles Keepax Signed-off-by: Mark Brown --- sound/Kconfig | 2 ++ sound/Makefile | 1 + 2 files changed, 3 insertions(+) (limited to 'sound') diff --git a/sound/Kconfig b/sound/Kconfig index d7d2aac9542e..ed34218d38b8 100644 --- a/sound/Kconfig +++ b/sound/Kconfig @@ -80,6 +80,8 @@ source "sound/hda/Kconfig" source "sound/ppc/Kconfig" +source "sound/ac97/Kconfig" + source "sound/aoa/Kconfig" source "sound/arm/Kconfig" diff --git a/sound/Makefile b/sound/Makefile index 6de45d2c32f7..c03b0bed65d5 100644 --- a/sound/Makefile +++ b/sound/Makefile @@ -10,6 +10,7 @@ obj-$(CONFIG_SND_AOA) += aoa/ # This one must be compilable even if sound is configured out obj-$(CONFIG_AC97_BUS) += ac97_bus.o +obj-$(CONFIG_AC97_BUS_NEW) += ac97/ ifeq ($(CONFIG_SND),y) obj-y += last.o -- cgit v1.2.3 From 6f8acad646d29fbf5665a6e0c9adae71c3c2131e Mon Sep 17 00:00:00 2001 From: Robert Jarzmik Date: Sat, 2 Sep 2017 21:54:06 +0200 Subject: ASoC: arm: make pxa2xx-ac97-lib ac97 codec agnostic All pxa library functions don't use the input parameters for nothing but slot number. This simplifies their prototypes, and makes them usable by both the legacy ac97 bus and the new ac97 bus. Signed-off-by: Robert Jarzmik Signed-off-by: Mark Brown --- sound/arm/pxa2xx-ac97-lib.c | 37 +++++++++++++++++++++---------------- sound/arm/pxa2xx-ac97.c | 35 +++++++++++++++++++++++++++-------- sound/soc/pxa/pxa2xx-ac97.c | 32 ++++++++++++++++++++++++++------ 3 files changed, 74 insertions(+), 30 deletions(-) (limited to 'sound') diff --git a/sound/arm/pxa2xx-ac97-lib.c b/sound/arm/pxa2xx-ac97-lib.c index 39c3969ac1c7..5950a9e218d9 100644 --- a/sound/arm/pxa2xx-ac97-lib.c +++ b/sound/arm/pxa2xx-ac97-lib.c @@ -20,7 +20,6 @@ #include #include -#include #include #include @@ -46,38 +45,41 @@ extern void pxa27x_configure_ac97reset(int reset_gpio, bool to_gpio); * 1 jiffy timeout if interrupt never comes). */ -unsigned short pxa2xx_ac97_read(struct snd_ac97 *ac97, unsigned short reg) +int pxa2xx_ac97_read(int slot, unsigned short reg) { - unsigned short val = -1; + int val = -ENODEV; volatile u32 *reg_addr; + if (slot > 0) + return -ENODEV; + mutex_lock(&car_mutex); /* set up primary or secondary codec space */ if (cpu_is_pxa25x() && reg == AC97_GPIO_STATUS) - reg_addr = ac97->num ? &SMC_REG_BASE : &PMC_REG_BASE; + reg_addr = slot ? &SMC_REG_BASE : &PMC_REG_BASE; else - reg_addr = ac97->num ? &SAC_REG_BASE : &PAC_REG_BASE; + reg_addr = slot ? &SAC_REG_BASE : &PAC_REG_BASE; reg_addr += (reg >> 1); /* start read access across the ac97 link */ GSR = GSR_CDONE | GSR_SDONE; gsr_bits = 0; - val = *reg_addr; + val = (*reg_addr & 0xffff); if (reg == AC97_GPIO_STATUS) goto out; if (wait_event_timeout(gsr_wq, (GSR | gsr_bits) & GSR_SDONE, 1) <= 0 && !((GSR | gsr_bits) & GSR_SDONE)) { printk(KERN_ERR "%s: read error (ac97_reg=%d GSR=%#lx)\n", __func__, reg, GSR | gsr_bits); - val = -1; + val = -ETIMEDOUT; goto out; } /* valid data now */ GSR = GSR_CDONE | GSR_SDONE; gsr_bits = 0; - val = *reg_addr; + val = (*reg_addr & 0xffff); /* but we've just started another cycle... */ wait_event_timeout(gsr_wq, (GSR | gsr_bits) & GSR_SDONE, 1); @@ -86,29 +88,32 @@ out: mutex_unlock(&car_mutex); } EXPORT_SYMBOL_GPL(pxa2xx_ac97_read); -void pxa2xx_ac97_write(struct snd_ac97 *ac97, unsigned short reg, - unsigned short val) +int pxa2xx_ac97_write(int slot, unsigned short reg, unsigned short val) { volatile u32 *reg_addr; + int ret = 0; mutex_lock(&car_mutex); /* set up primary or secondary codec space */ if (cpu_is_pxa25x() && reg == AC97_GPIO_STATUS) - reg_addr = ac97->num ? &SMC_REG_BASE : &PMC_REG_BASE; + reg_addr = slot ? &SMC_REG_BASE : &PMC_REG_BASE; else - reg_addr = ac97->num ? &SAC_REG_BASE : &PAC_REG_BASE; + reg_addr = slot ? &SAC_REG_BASE : &PAC_REG_BASE; reg_addr += (reg >> 1); GSR = GSR_CDONE | GSR_SDONE; gsr_bits = 0; *reg_addr = val; if (wait_event_timeout(gsr_wq, (GSR | gsr_bits) & GSR_CDONE, 1) <= 0 && - !((GSR | gsr_bits) & GSR_CDONE)) + !((GSR | gsr_bits) & GSR_CDONE)) { printk(KERN_ERR "%s: write error (ac97_reg=%d GSR=%#lx)\n", __func__, reg, GSR | gsr_bits); + ret = -EIO; + } mutex_unlock(&car_mutex); + return ret; } EXPORT_SYMBOL_GPL(pxa2xx_ac97_write); @@ -188,7 +193,7 @@ static inline void pxa_ac97_cold_pxa3xx(void) } #endif -bool pxa2xx_ac97_try_warm_reset(struct snd_ac97 *ac97) +bool pxa2xx_ac97_try_warm_reset(void) { unsigned long gsr; unsigned int timeout = 100; @@ -225,7 +230,7 @@ bool pxa2xx_ac97_try_warm_reset(struct snd_ac97 *ac97) } EXPORT_SYMBOL_GPL(pxa2xx_ac97_try_warm_reset); -bool pxa2xx_ac97_try_cold_reset(struct snd_ac97 *ac97) +bool pxa2xx_ac97_try_cold_reset(void) { unsigned long gsr; unsigned int timeout = 1000; @@ -263,7 +268,7 @@ bool pxa2xx_ac97_try_cold_reset(struct snd_ac97 *ac97) EXPORT_SYMBOL_GPL(pxa2xx_ac97_try_cold_reset); -void pxa2xx_ac97_finish_reset(struct snd_ac97 *ac97) +void pxa2xx_ac97_finish_reset(void) { GCR &= ~(GCR_PRIRDY_IEN|GCR_SECRDY_IEN); GCR |= GCR_SDONE_IE|GCR_CDONE_IE; diff --git a/sound/arm/pxa2xx-ac97.c b/sound/arm/pxa2xx-ac97.c index fbd5dad0c484..4bc244c40f80 100644 --- a/sound/arm/pxa2xx-ac97.c +++ b/sound/arm/pxa2xx-ac97.c @@ -29,19 +29,38 @@ #include "pxa2xx-pcm.h" -static void pxa2xx_ac97_reset(struct snd_ac97 *ac97) +static void pxa2xx_ac97_legacy_reset(struct snd_ac97 *ac97) { - if (!pxa2xx_ac97_try_cold_reset(ac97)) { - pxa2xx_ac97_try_warm_reset(ac97); - } + if (!pxa2xx_ac97_try_cold_reset()) + pxa2xx_ac97_try_warm_reset(); + + pxa2xx_ac97_finish_reset(); +} + +static unsigned short pxa2xx_ac97_legacy_read(struct snd_ac97 *ac97, + unsigned short reg) +{ + int ret; + + ret = pxa2xx_ac97_read(ac97->num, reg); + if (ret < 0) + return 0; + else + return (unsigned short)(ret & 0xffff); +} + +static void pxa2xx_ac97_legacy_write(struct snd_ac97 *ac97, + unsigned short reg, unsigned short val) +{ + int __always_unused ret; - pxa2xx_ac97_finish_reset(ac97); + ret = pxa2xx_ac97_write(ac97->num, reg, val); } static struct snd_ac97_bus_ops pxa2xx_ac97_ops = { - .read = pxa2xx_ac97_read, - .write = pxa2xx_ac97_write, - .reset = pxa2xx_ac97_reset, + .read = pxa2xx_ac97_legacy_read, + .write = pxa2xx_ac97_legacy_write, + .reset = pxa2xx_ac97_legacy_reset, }; static struct pxad_param pxa2xx_ac97_pcm_out_req = { diff --git a/sound/soc/pxa/pxa2xx-ac97.c b/sound/soc/pxa/pxa2xx-ac97.c index f49bf02e5ec2..803818aabee9 100644 --- a/sound/soc/pxa/pxa2xx-ac97.c +++ b/sound/soc/pxa/pxa2xx-ac97.c @@ -29,21 +29,41 @@ static void pxa2xx_ac97_warm_reset(struct snd_ac97 *ac97) { - pxa2xx_ac97_try_warm_reset(ac97); + pxa2xx_ac97_try_warm_reset(); - pxa2xx_ac97_finish_reset(ac97); + pxa2xx_ac97_finish_reset(); } static void pxa2xx_ac97_cold_reset(struct snd_ac97 *ac97) { - pxa2xx_ac97_try_cold_reset(ac97); + pxa2xx_ac97_try_cold_reset(); - pxa2xx_ac97_finish_reset(ac97); + pxa2xx_ac97_finish_reset(); +} + +static unsigned short pxa2xx_ac97_legacy_read(struct snd_ac97 *ac97, + unsigned short reg) +{ + int ret; + + ret = pxa2xx_ac97_read(ac97->num, reg); + if (ret < 0) + return 0; + else + return (unsigned short)(ret & 0xffff); +} + +static void pxa2xx_ac97_legacy_write(struct snd_ac97 *ac97, + unsigned short reg, unsigned short val) +{ + int ret; + + ret = pxa2xx_ac97_write(ac97->num, reg, val); } static struct snd_ac97_bus_ops pxa2xx_ac97_ops = { - .read = pxa2xx_ac97_read, - .write = pxa2xx_ac97_write, + .read = pxa2xx_ac97_legacy_read, + .write = pxa2xx_ac97_legacy_write, .warm_reset = pxa2xx_ac97_warm_reset, .reset = pxa2xx_ac97_cold_reset, }; -- cgit v1.2.3 From 9e3f9f36a6f40bb6ba9b3844d709314121e4c106 Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Mon, 4 Sep 2017 16:41:48 +0100 Subject: ASoC: arizona: Add new common Arizona init function Currently the driver has quite a few small initialisation functions, in preparation for some refactoring add a new function arizona_init_common. This will be used bus probe level initialisation that is common across Arizona devices. For now just move the notifier chain initialisation in there. Signed-off-by: Charles Keepax Signed-off-by: Mark Brown --- sound/soc/codecs/arizona.c | 7 ++----- sound/soc/codecs/arizona.h | 3 ++- sound/soc/codecs/cs47l24.c | 3 ++- sound/soc/codecs/wm5102.c | 3 ++- sound/soc/codecs/wm5110.c | 3 ++- sound/soc/codecs/wm8997.c | 4 ++-- sound/soc/codecs/wm8998.c | 3 ++- 7 files changed, 14 insertions(+), 12 deletions(-) (limited to 'sound') diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c index a1149f6a8450..ba5f57a58219 100644 --- a/sound/soc/codecs/arizona.c +++ b/sound/soc/codecs/arizona.c @@ -293,16 +293,13 @@ int arizona_init_gpio(struct snd_soc_codec *codec) } EXPORT_SYMBOL_GPL(arizona_init_gpio); -int arizona_init_notifiers(struct snd_soc_codec *codec) +int arizona_init_common(struct arizona *arizona) { - struct arizona_priv *priv = snd_soc_codec_get_drvdata(codec); - struct arizona *arizona = priv->arizona; - BLOCKING_INIT_NOTIFIER_HEAD(&arizona->notifier); return 0; } -EXPORT_SYMBOL_GPL(arizona_init_notifiers); +EXPORT_SYMBOL_GPL(arizona_init_common); const char * const arizona_mixer_texts[ARIZONA_NUM_MIXER_INPUTS] = { "None", diff --git a/sound/soc/codecs/arizona.h b/sound/soc/codecs/arizona.h index 1822e3b3de80..292073ca3bd9 100644 --- a/sound/soc/codecs/arizona.h +++ b/sound/soc/codecs/arizona.h @@ -313,7 +313,8 @@ int arizona_set_fll(struct arizona_fll *fll, int source, int arizona_init_spk(struct snd_soc_codec *codec); int arizona_init_gpio(struct snd_soc_codec *codec); int arizona_init_mono(struct snd_soc_codec *codec); -int arizona_init_notifiers(struct snd_soc_codec *codec); + +int arizona_init_common(struct arizona *arizona); int arizona_init_spk_irqs(struct arizona *arizona); int arizona_free_spk_irqs(struct arizona *arizona); diff --git a/sound/soc/codecs/cs47l24.c b/sound/soc/codecs/cs47l24.c index e09fc8f037f1..fdcc7318993b 100644 --- a/sound/soc/codecs/cs47l24.c +++ b/sound/soc/codecs/cs47l24.c @@ -1130,7 +1130,6 @@ static int cs47l24_codec_probe(struct snd_soc_codec *codec) arizona_init_gpio(codec); arizona_init_mono(codec); - arizona_init_notifiers(codec); ret = wm_adsp2_codec_probe(&priv->core.adsp[1], codec); if (ret) @@ -1288,6 +1287,8 @@ static int cs47l24_probe(struct platform_device *pdev) return ret; } + arizona_init_common(arizona); + ret = arizona_init_spk_irqs(arizona); if (ret < 0) goto err_dsp_irq; diff --git a/sound/soc/codecs/wm5102.c b/sound/soc/codecs/wm5102.c index 72486bf072f2..8354bdf7fc15 100644 --- a/sound/soc/codecs/wm5102.c +++ b/sound/soc/codecs/wm5102.c @@ -1951,7 +1951,6 @@ static int wm5102_codec_probe(struct snd_soc_codec *codec) return ret; arizona_init_gpio(codec); - arizona_init_notifiers(codec); snd_soc_component_disable_pin(component, "HAPTICS"); @@ -2098,6 +2097,8 @@ static int wm5102_probe(struct platform_device *pdev) return ret; } + arizona_init_common(arizona); + ret = arizona_init_spk_irqs(arizona); if (ret < 0) goto err_dsp_irq; diff --git a/sound/soc/codecs/wm5110.c b/sound/soc/codecs/wm5110.c index 858a24fc28e8..0437df60be77 100644 --- a/sound/soc/codecs/wm5110.c +++ b/sound/soc/codecs/wm5110.c @@ -2290,7 +2290,6 @@ static int wm5110_codec_probe(struct snd_soc_codec *codec) arizona_init_gpio(codec); arizona_init_mono(codec); - arizona_init_notifiers(codec); for (i = 0; i < WM5110_NUM_ADSP; ++i) { ret = wm_adsp2_codec_probe(&priv->core.adsp[i], codec); @@ -2454,6 +2453,8 @@ static int wm5110_probe(struct platform_device *pdev) return ret; } + arizona_init_common(arizona); + ret = arizona_init_spk_irqs(arizona); if (ret < 0) goto err_dsp_irq; diff --git a/sound/soc/codecs/wm8997.c b/sound/soc/codecs/wm8997.c index 49401a8aae64..91c3c3e052d1 100644 --- a/sound/soc/codecs/wm8997.c +++ b/sound/soc/codecs/wm8997.c @@ -1068,8 +1068,6 @@ static int wm8997_codec_probe(struct snd_soc_codec *codec) if (ret < 0) return ret; - arizona_init_notifiers(codec); - snd_soc_component_disable_pin(component, "HAPTICS"); priv->core.arizona->dapm = dapm; @@ -1168,6 +1166,8 @@ static int wm8997_probe(struct platform_device *pdev) pm_runtime_enable(&pdev->dev); pm_runtime_idle(&pdev->dev); + arizona_init_common(arizona); + ret = arizona_init_spk_irqs(arizona); if (ret < 0) return ret; diff --git a/sound/soc/codecs/wm8998.c b/sound/soc/codecs/wm8998.c index 44f447136e22..27a8e1e75f28 100644 --- a/sound/soc/codecs/wm8998.c +++ b/sound/soc/codecs/wm8998.c @@ -1330,7 +1330,6 @@ static int wm8998_codec_probe(struct snd_soc_codec *codec) return ret; arizona_init_gpio(codec); - arizona_init_notifiers(codec); snd_soc_component_disable_pin(component, "HAPTICS"); @@ -1423,6 +1422,8 @@ static int wm8998_probe(struct platform_device *pdev) pm_runtime_enable(&pdev->dev); pm_runtime_idle(&pdev->dev); + arizona_init_common(arizona); + ret = arizona_init_spk_irqs(arizona); if (ret < 0) return ret; -- cgit v1.2.3 From 0a229b15d99e0a9761f9672f4ff7efeb18ce0ea1 Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Mon, 4 Sep 2017 16:41:49 +0100 Subject: ASoC: arizona: Add handling for audio related device tree entries Currently all the audio related device tree entries are handled by the MFD code, for most parts of the Arizona driver we group the device tree handling with the component that uses it and should do so here as well. Add handling in the ASoC code for the audio device tree entries, a later patch removes the MFD side handling but there is no harm in it being duplicated temporarily. Signed-off-by: Charles Keepax Signed-off-by: Mark Brown --- sound/soc/codecs/arizona.c | 136 +++++++++++++++++++++++++++++++++++++++++++++ sound/soc/codecs/arizona.h | 2 + sound/soc/codecs/cs47l24.c | 8 +++ sound/soc/codecs/wm5102.c | 8 +++ sound/soc/codecs/wm5110.c | 8 +++ sound/soc/codecs/wm8997.c | 8 +++ sound/soc/codecs/wm8998.c | 8 +++ 7 files changed, 178 insertions(+) (limited to 'sound') diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c index ba5f57a58219..e6967385dccb 100644 --- a/sound/soc/codecs/arizona.c +++ b/sound/soc/codecs/arizona.c @@ -13,6 +13,7 @@ #include #include #include +#include #include #include #include @@ -295,8 +296,78 @@ EXPORT_SYMBOL_GPL(arizona_init_gpio); int arizona_init_common(struct arizona *arizona) { + struct arizona_pdata *pdata = &arizona->pdata; + unsigned int val, mask; + int i; + BLOCKING_INIT_NOTIFIER_HEAD(&arizona->notifier); + for (i = 0; i < ARIZONA_MAX_OUTPUT; ++i) { + /* Default is 0 so noop with defaults */ + if (pdata->out_mono[i]) + val = ARIZONA_OUT1_MONO; + else + val = 0; + + regmap_update_bits(arizona->regmap, + ARIZONA_OUTPUT_PATH_CONFIG_1L + (i * 8), + ARIZONA_OUT1_MONO, val); + } + + for (i = 0; i < ARIZONA_MAX_PDM_SPK; i++) { + if (pdata->spk_mute[i]) + regmap_update_bits(arizona->regmap, + ARIZONA_PDM_SPK1_CTRL_1 + (i * 2), + ARIZONA_SPK1_MUTE_ENDIAN_MASK | + ARIZONA_SPK1_MUTE_SEQ1_MASK, + pdata->spk_mute[i]); + + if (pdata->spk_fmt[i]) + regmap_update_bits(arizona->regmap, + ARIZONA_PDM_SPK1_CTRL_2 + (i * 2), + ARIZONA_SPK1_FMT_MASK, + pdata->spk_fmt[i]); + } + + for (i = 0; i < ARIZONA_MAX_INPUT; i++) { + /* Default for both is 0 so noop with defaults */ + val = pdata->dmic_ref[i] << ARIZONA_IN1_DMIC_SUP_SHIFT; + if (pdata->inmode[i] & ARIZONA_INMODE_DMIC) + val |= 1 << ARIZONA_IN1_MODE_SHIFT; + + switch (arizona->type) { + case WM8998: + case WM1814: + regmap_update_bits(arizona->regmap, + ARIZONA_ADC_DIGITAL_VOLUME_1L + (i * 8), + ARIZONA_IN1L_SRC_SE_MASK, + (pdata->inmode[i] & ARIZONA_INMODE_SE) + << ARIZONA_IN1L_SRC_SE_SHIFT); + + regmap_update_bits(arizona->regmap, + ARIZONA_ADC_DIGITAL_VOLUME_1R + (i * 8), + ARIZONA_IN1R_SRC_SE_MASK, + (pdata->inmode[i] & ARIZONA_INMODE_SE) + << ARIZONA_IN1R_SRC_SE_SHIFT); + + mask = ARIZONA_IN1_DMIC_SUP_MASK | + ARIZONA_IN1_MODE_MASK; + break; + default: + if (pdata->inmode[i] & ARIZONA_INMODE_SE) + val |= 1 << ARIZONA_IN1_SINGLE_ENDED_SHIFT; + + mask = ARIZONA_IN1_DMIC_SUP_MASK | + ARIZONA_IN1_MODE_MASK | + ARIZONA_IN1_SINGLE_ENDED_MASK; + break; + } + + regmap_update_bits(arizona->regmap, + ARIZONA_IN1L_CONTROL + (i * 8), + mask, val); + } + return 0; } EXPORT_SYMBOL_GPL(arizona_init_common); @@ -2692,6 +2763,71 @@ int arizona_lhpf_coeff_put(struct snd_kcontrol *kcontrol, } EXPORT_SYMBOL_GPL(arizona_lhpf_coeff_put); +int arizona_of_get_audio_pdata(struct arizona *arizona) +{ + struct arizona_pdata *pdata = &arizona->pdata; + struct device_node *np = arizona->dev->of_node; + struct property *prop; + const __be32 *cur; + u32 val; + u32 pdm_val[ARIZONA_MAX_PDM_SPK]; + int ret; + int count = 0; + + count = 0; + of_property_for_each_u32(np, "wlf,inmode", prop, cur, val) { + if (count == ARRAY_SIZE(pdata->inmode)) + break; + + pdata->inmode[count] = val; + count++; + } + + count = 0; + of_property_for_each_u32(np, "wlf,dmic-ref", prop, cur, val) { + if (count == ARRAY_SIZE(pdata->dmic_ref)) + break; + + pdata->dmic_ref[count] = val; + count++; + } + + count = 0; + of_property_for_each_u32(np, "wlf,out-mono", prop, cur, val) { + if (count == ARRAY_SIZE(pdata->out_mono)) + break; + + pdata->out_mono[count] = !!val; + count++; + } + + count = 0; + of_property_for_each_u32(np, "wlf,max-channels-clocked", prop, cur, val) { + if (count == ARRAY_SIZE(pdata->max_channels_clocked)) + break; + + pdata->max_channels_clocked[count] = val; + count++; + } + + ret = of_property_read_u32_array(np, "wlf,spk-fmt", + pdm_val, ARRAY_SIZE(pdm_val)); + + if (ret >= 0) + for (count = 0; count < ARRAY_SIZE(pdata->spk_fmt); ++count) + pdata->spk_fmt[count] = pdm_val[count]; + + ret = of_property_read_u32_array(np, "wlf,spk-mute", + pdm_val, ARRAY_SIZE(pdm_val)); + + if (ret >= 0) + for (count = 0; count < ARRAY_SIZE(pdata->spk_mute); ++count) + pdata->spk_mute[count] = pdm_val[count]; + + return 0; +} +EXPORT_SYMBOL_GPL(arizona_of_get_audio_pdata); + MODULE_DESCRIPTION("ASoC Wolfson Arizona class device support"); MODULE_AUTHOR("Mark Brown "); MODULE_LICENSE("GPL"); diff --git a/sound/soc/codecs/arizona.h b/sound/soc/codecs/arizona.h index 292073ca3bd9..2d198fb2ce97 100644 --- a/sound/soc/codecs/arizona.h +++ b/sound/soc/codecs/arizona.h @@ -351,4 +351,6 @@ static inline int arizona_unregister_notifier(struct snd_soc_codec *codec, return blocking_notifier_chain_unregister(&arizona->notifier, nb); } +int arizona_of_get_audio_pdata(struct arizona *arizona); + #endif diff --git a/sound/soc/codecs/cs47l24.c b/sound/soc/codecs/cs47l24.c index fdcc7318993b..0fe7d7a87ff3 100644 --- a/sound/soc/codecs/cs47l24.c +++ b/sound/soc/codecs/cs47l24.c @@ -1229,6 +1229,14 @@ static int cs47l24_probe(struct platform_device *pdev) if (!cs47l24) return -ENOMEM; + if (IS_ENABLED(CONFIG_OF)) { + if (!dev_get_platdata(arizona->dev)) { + ret = arizona_of_get_audio_pdata(arizona); + if (ret < 0) + return ret; + } + } + platform_set_drvdata(pdev, cs47l24); cs47l24->core.arizona = arizona; diff --git a/sound/soc/codecs/wm5102.c b/sound/soc/codecs/wm5102.c index 8354bdf7fc15..5a917dd73f32 100644 --- a/sound/soc/codecs/wm5102.c +++ b/sound/soc/codecs/wm5102.c @@ -2042,6 +2042,14 @@ static int wm5102_probe(struct platform_device *pdev) return -ENOMEM; platform_set_drvdata(pdev, wm5102); + if (IS_ENABLED(CONFIG_OF)) { + if (!dev_get_platdata(arizona->dev)) { + ret = arizona_of_get_audio_pdata(arizona); + if (ret < 0) + return ret; + } + } + mutex_init(&arizona->dac_comp_lock); wm5102->core.arizona = arizona; diff --git a/sound/soc/codecs/wm5110.c b/sound/soc/codecs/wm5110.c index 0437df60be77..ba1e90ca8be4 100644 --- a/sound/soc/codecs/wm5110.c +++ b/sound/soc/codecs/wm5110.c @@ -2397,6 +2397,14 @@ static int wm5110_probe(struct platform_device *pdev) return -ENOMEM; platform_set_drvdata(pdev, wm5110); + if (IS_ENABLED(CONFIG_OF)) { + if (!dev_get_platdata(arizona->dev)) { + ret = arizona_of_get_audio_pdata(arizona); + if (ret < 0) + return ret; + } + } + wm5110->core.arizona = arizona; wm5110->core.num_inputs = 8; diff --git a/sound/soc/codecs/wm8997.c b/sound/soc/codecs/wm8997.c index 91c3c3e052d1..c5aef9ecdecc 100644 --- a/sound/soc/codecs/wm8997.c +++ b/sound/soc/codecs/wm8997.c @@ -1134,6 +1134,14 @@ static int wm8997_probe(struct platform_device *pdev) return -ENOMEM; platform_set_drvdata(pdev, wm8997); + if (IS_ENABLED(CONFIG_OF)) { + if (!dev_get_platdata(arizona->dev)) { + ret = arizona_of_get_audio_pdata(arizona); + if (ret < 0) + return ret; + } + } + wm8997->core.arizona = arizona; wm8997->core.num_inputs = 4; diff --git a/sound/soc/codecs/wm8998.c b/sound/soc/codecs/wm8998.c index 27a8e1e75f28..c59caaa75ba0 100644 --- a/sound/soc/codecs/wm8998.c +++ b/sound/soc/codecs/wm8998.c @@ -1398,6 +1398,14 @@ static int wm8998_probe(struct platform_device *pdev) return -ENOMEM; platform_set_drvdata(pdev, wm8998); + if (IS_ENABLED(CONFIG_OF)) { + if (!dev_get_platdata(arizona->dev)) { + ret = arizona_of_get_audio_pdata(arizona); + if (ret < 0) + return ret; + } + } + wm8998->core.arizona = arizona; wm8998->core.num_inputs = 3; /* IN1L, IN1R, IN2 */ -- cgit v1.2.3 From 85e7dd3f871b988702973c80d9ef128e10dd3dad Mon Sep 17 00:00:00 2001 From: Charles Keepax Date: Mon, 4 Sep 2017 16:41:53 +0100 Subject: ASoC: arizona: Add support for setting the output volume limits The output volume limits allow signals to be limited to specific levels appropriate for the hardware attached. As this is a property of the hardware itself these will be configured through device tree. Signed-off-by: Charles Keepax Acked-by: Lee Jones Signed-off-by: Mark Brown --- sound/soc/codecs/arizona.c | 25 +++++++++++++++++++++++++ sound/soc/codecs/arizona.h | 1 + sound/soc/codecs/cs47l24.c | 3 +++ sound/soc/codecs/wm5102.c | 3 +++ sound/soc/codecs/wm5110.c | 3 +++ sound/soc/codecs/wm8997.c | 3 +++ 6 files changed, 38 insertions(+) (limited to 'sound') diff --git a/sound/soc/codecs/arizona.c b/sound/soc/codecs/arizona.c index e6967385dccb..b3375e19598a 100644 --- a/sound/soc/codecs/arizona.c +++ b/sound/soc/codecs/arizona.c @@ -372,6 +372,22 @@ int arizona_init_common(struct arizona *arizona) } EXPORT_SYMBOL_GPL(arizona_init_common); +int arizona_init_vol_limit(struct arizona *arizona) +{ + int i; + + for (i = 0; i < ARRAY_SIZE(arizona->pdata.out_vol_limit); ++i) { + if (arizona->pdata.out_vol_limit[i]) + regmap_update_bits(arizona->regmap, + ARIZONA_DAC_VOLUME_LIMIT_1L + i * 4, + ARIZONA_OUT1L_VOL_LIM_MASK, + arizona->pdata.out_vol_limit[i]); + } + + return 0; +} +EXPORT_SYMBOL_GPL(arizona_init_vol_limit); + const char * const arizona_mixer_texts[ARIZONA_NUM_MIXER_INPUTS] = { "None", "Tone Generator 1", @@ -2810,6 +2826,15 @@ int arizona_of_get_audio_pdata(struct arizona *arizona) count++; } + count = 0; + of_property_for_each_u32(np, "wlf,out-volume-limit", prop, cur, val) { + if (count == ARRAY_SIZE(pdata->out_vol_limit)) + break; + + pdata->out_vol_limit[count] = val; + count++; + } + ret = of_property_read_u32_array(np, "wlf,spk-fmt", pdm_val, ARRAY_SIZE(pdm_val)); diff --git a/sound/soc/codecs/arizona.h b/sound/soc/codecs/arizona.h index 2d198fb2ce97..dfdf6d8c9687 100644 --- a/sound/soc/codecs/arizona.h +++ b/sound/soc/codecs/arizona.h @@ -315,6 +315,7 @@ int arizona_init_gpio(struct snd_soc_codec *codec); int arizona_init_mono(struct snd_soc_codec *codec); int arizona_init_common(struct arizona *arizona); +int arizona_init_vol_limit(struct arizona *arizona); int arizona_init_spk_irqs(struct arizona *arizona); int arizona_free_spk_irqs(struct arizona *arizona); diff --git a/sound/soc/codecs/cs47l24.c b/sound/soc/codecs/cs47l24.c index 0fe7d7a87ff3..94c0209977d0 100644 --- a/sound/soc/codecs/cs47l24.c +++ b/sound/soc/codecs/cs47l24.c @@ -1297,6 +1297,9 @@ static int cs47l24_probe(struct platform_device *pdev) arizona_init_common(arizona); + ret = arizona_init_vol_limit(arizona); + if (ret < 0) + goto err_dsp_irq; ret = arizona_init_spk_irqs(arizona); if (ret < 0) goto err_dsp_irq; diff --git a/sound/soc/codecs/wm5102.c b/sound/soc/codecs/wm5102.c index 5a917dd73f32..4f0481d3c7a7 100644 --- a/sound/soc/codecs/wm5102.c +++ b/sound/soc/codecs/wm5102.c @@ -2107,6 +2107,9 @@ static int wm5102_probe(struct platform_device *pdev) arizona_init_common(arizona); + ret = arizona_init_vol_limit(arizona); + if (ret < 0) + goto err_dsp_irq; ret = arizona_init_spk_irqs(arizona); if (ret < 0) goto err_dsp_irq; diff --git a/sound/soc/codecs/wm5110.c b/sound/soc/codecs/wm5110.c index ba1e90ca8be4..6ed1e1f9ce51 100644 --- a/sound/soc/codecs/wm5110.c +++ b/sound/soc/codecs/wm5110.c @@ -2463,6 +2463,9 @@ static int wm5110_probe(struct platform_device *pdev) arizona_init_common(arizona); + ret = arizona_init_vol_limit(arizona); + if (ret < 0) + goto err_dsp_irq; ret = arizona_init_spk_irqs(arizona); if (ret < 0) goto err_dsp_irq; diff --git a/sound/soc/codecs/wm8997.c b/sound/soc/codecs/wm8997.c index c5aef9ecdecc..77f512767273 100644 --- a/sound/soc/codecs/wm8997.c +++ b/sound/soc/codecs/wm8997.c @@ -1176,6 +1176,9 @@ static int wm8997_probe(struct platform_device *pdev) arizona_init_common(arizona); + ret = arizona_init_vol_limit(arizona); + if (ret < 0) + return ret; ret = arizona_init_spk_irqs(arizona); if (ret < 0) return ret; -- cgit v1.2.3 From a1b16aaa55b6425418d6d7a87d3dbbe40bac8c37 Mon Sep 17 00:00:00 2001 From: Vijendar Mukunda Date: Mon, 9 Oct 2017 16:36:08 -0400 Subject: ASoC: AMD: Added asic_type as ACP DMA driver platform data asic_type information is passed to ACP DMA Driver as platform data. Reviewed-by: Alex Deucher Signed-off-by: Vijendar Mukunda Signed-off-by: Alex Deucher --- sound/soc/amd/acp-pcm-dma.c | 8 ++------ sound/soc/amd/acp.h | 7 +++++++ 2 files changed, 9 insertions(+), 6 deletions(-) (limited to 'sound') diff --git a/sound/soc/amd/acp-pcm-dma.c b/sound/soc/amd/acp-pcm-dma.c index 08b1399d1da2..dcbf9973884d 100644 --- a/sound/soc/amd/acp-pcm-dma.c +++ b/sound/soc/amd/acp-pcm-dma.c @@ -73,12 +73,6 @@ static const struct snd_pcm_hardware acp_pcm_hardware_capture = { .periods_max = CAPTURE_MAX_NUM_PERIODS, }; -struct audio_drv_data { - struct snd_pcm_substream *play_stream; - struct snd_pcm_substream *capture_stream; - void __iomem *acp_mmio; -}; - static u32 acp_reg_read(void __iomem *acp_mmio, u32 reg) { return readl(acp_mmio + (reg * 4)); @@ -916,6 +910,7 @@ static int acp_audio_probe(struct platform_device *pdev) int status; struct audio_drv_data *audio_drv_data; struct resource *res; + const u32 *pdata = pdev->dev.platform_data; audio_drv_data = devm_kzalloc(&pdev->dev, sizeof(struct audio_drv_data), GFP_KERNEL); @@ -932,6 +927,7 @@ static int acp_audio_probe(struct platform_device *pdev) audio_drv_data->play_stream = NULL; audio_drv_data->capture_stream = NULL; + audio_drv_data->asic_type = *pdata; res = platform_get_resource(pdev, IORESOURCE_IRQ, 0); if (!res) { diff --git a/sound/soc/amd/acp.h b/sound/soc/amd/acp.h index 330832ef4e5e..28cf9140f49c 100644 --- a/sound/soc/amd/acp.h +++ b/sound/soc/amd/acp.h @@ -84,6 +84,13 @@ struct audio_substream_data { void __iomem *acp_mmio; }; +struct audio_drv_data { + struct snd_pcm_substream *play_stream; + struct snd_pcm_substream *capture_stream; + void __iomem *acp_mmio; + u32 asic_type; +}; + enum { ACP_TILE_P1 = 0, ACP_TILE_P2, -- cgit v1.2.3 From 607b39ef7f5be3036e4f66a932bedb334832722f Mon Sep 17 00:00:00 2001 From: Vijendar Mukunda Date: Wed, 18 Oct 2017 12:13:57 -0400 Subject: ASoC: AMD: disabling memory gating in stoney platform For Stoney platform, Memory gating is disabled.i.e SRAM Banks won't be turned off. By Default, SRAM Bank state set to ON. Added condition checks to skip SRAM Bank state set logic for Stoney platform. Acked-by: Mark Brown Reviewed-by: Alex Deucher Signed-off-by: Vijendar Mukunda Signed-off-by: Alex Deucher Signed-off-by: Mark Brown --- sound/soc/amd/acp-pcm-dma.c | 79 +++++++++++++++++++++++++++++++-------------- 1 file changed, 55 insertions(+), 24 deletions(-) (limited to 'sound') diff --git a/sound/soc/amd/acp-pcm-dma.c b/sound/soc/amd/acp-pcm-dma.c index dcbf9973884d..f00b6b92e076 100644 --- a/sound/soc/amd/acp-pcm-dma.c +++ b/sound/soc/amd/acp-pcm-dma.c @@ -20,7 +20,7 @@ #include #include - +#include #include "acp.h" #define PLAYBACK_MIN_NUM_PERIODS 2 @@ -419,7 +419,7 @@ static void acp_set_sram_bank_state(void __iomem *acp_mmio, u16 bank, } /* Initialize and bring ACP hardware to default state. */ -static int acp_init(void __iomem *acp_mmio) +static int acp_init(void __iomem *acp_mmio, u32 asic_type) { u16 bank; u32 val, count, sram_pte_offset; @@ -493,9 +493,14 @@ static int acp_init(void __iomem *acp_mmio) /* When ACP_TILE_P1 is turned on, all SRAM banks get turned on. * Now, turn off all of them. This can't be done in 'poweron' of * ACP pm domain, as this requires ACP to be initialized. + * For Stoney, Memory gating is disabled,i.e SRAM Banks + * won't be turned off. The default state for SRAM banks is ON. + * Setting SRAM bank state code skipped for STONEY platform. */ - for (bank = 1; bank < 48; bank++) - acp_set_sram_bank_state(acp_mmio, bank, false); + if (asic_type != CHIP_STONEY) { + for (bank = 1; bank < 48; bank++) + acp_set_sram_bank_state(acp_mmio, bank, false); + } return 0; } @@ -646,14 +651,22 @@ static int acp_dma_open(struct snd_pcm_substream *substream) if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { intr_data->play_stream = substream; - for (bank = 1; bank <= 4; bank++) - acp_set_sram_bank_state(intr_data->acp_mmio, bank, - true); + /* For Stoney, Memory gating is disabled,i.e SRAM Banks + * won't be turned off. The default state for SRAM banks is ON. + * Setting SRAM bank state code skipped for STONEY platform. + */ + if (intr_data->asic_type != CHIP_STONEY) { + for (bank = 1; bank <= 4; bank++) + acp_set_sram_bank_state(intr_data->acp_mmio, + bank, true); + } } else { intr_data->capture_stream = substream; - for (bank = 5; bank <= 8; bank++) - acp_set_sram_bank_state(intr_data->acp_mmio, bank, - true); + if (intr_data->asic_type != CHIP_STONEY) { + for (bank = 5; bank <= 8; bank++) + acp_set_sram_bank_state(intr_data->acp_mmio, + bank, true); + } } return 0; @@ -869,14 +882,23 @@ static int acp_dma_close(struct snd_pcm_substream *substream) if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { adata->play_stream = NULL; - for (bank = 1; bank <= 4; bank++) - acp_set_sram_bank_state(adata->acp_mmio, bank, - false); - } else { + /* For Stoney, Memory gating is disabled,i.e SRAM Banks + * won't be turned off. The default state for SRAM banks is ON. + * Setting SRAM bank state code skipped for STONEY platform. + * added condition checks for Carrizo platform only + */ + if (adata->asic_type != CHIP_STONEY) { + for (bank = 1; bank <= 4; bank++) + acp_set_sram_bank_state(adata->acp_mmio, bank, + false); + } + } else { adata->capture_stream = NULL; - for (bank = 5; bank <= 8; bank++) - acp_set_sram_bank_state(adata->acp_mmio, bank, - false); + if (adata->asic_type != CHIP_STONEY) { + for (bank = 5; bank <= 8; bank++) + acp_set_sram_bank_state(adata->acp_mmio, bank, + false); + } } /* Disable ACP irq, when the current stream is being closed and @@ -945,7 +967,7 @@ static int acp_audio_probe(struct platform_device *pdev) dev_set_drvdata(&pdev->dev, audio_drv_data); /* Initialize the ACP */ - acp_init(audio_drv_data->acp_mmio); + acp_init(audio_drv_data->acp_mmio, audio_drv_data->asic_type); status = snd_soc_register_platform(&pdev->dev, &acp_asoc_platform); if (status != 0) { @@ -976,19 +998,27 @@ static int acp_pcm_resume(struct device *dev) u16 bank; struct audio_drv_data *adata = dev_get_drvdata(dev); - acp_init(adata->acp_mmio); + acp_init(adata->acp_mmio, adata->asic_type); if (adata->play_stream && adata->play_stream->runtime) { - for (bank = 1; bank <= 4; bank++) - acp_set_sram_bank_state(adata->acp_mmio, bank, + /* For Stoney, Memory gating is disabled,i.e SRAM Banks + * won't be turned off. The default state for SRAM banks is ON. + * Setting SRAM bank state code skipped for STONEY platform. + */ + if (adata->asic_type != CHIP_STONEY) { + for (bank = 1; bank <= 4; bank++) + acp_set_sram_bank_state(adata->acp_mmio, bank, true); + } config_acp_dma(adata->acp_mmio, adata->play_stream->runtime->private_data); } if (adata->capture_stream && adata->capture_stream->runtime) { - for (bank = 5; bank <= 8; bank++) - acp_set_sram_bank_state(adata->acp_mmio, bank, + if (adata->asic_type != CHIP_STONEY) { + for (bank = 5; bank <= 8; bank++) + acp_set_sram_bank_state(adata->acp_mmio, bank, true); + } config_acp_dma(adata->acp_mmio, adata->capture_stream->runtime->private_data); } @@ -1009,7 +1039,7 @@ static int acp_pcm_runtime_resume(struct device *dev) { struct audio_drv_data *adata = dev_get_drvdata(dev); - acp_init(adata->acp_mmio); + acp_init(adata->acp_mmio, adata->asic_type); acp_reg_write(1, adata->acp_mmio, mmACP_EXTERNAL_INTR_ENB); return 0; } @@ -1031,6 +1061,7 @@ static struct platform_driver acp_dma_driver = { module_platform_driver(acp_dma_driver); +MODULE_AUTHOR("Vijendar.Mukunda@amd.com"); MODULE_AUTHOR("Maruthi.Bayyavarapu@amd.com"); MODULE_DESCRIPTION("AMD ACP PCM Driver"); MODULE_LICENSE("GPL v2"); -- cgit v1.2.3 From aac89748ee2746656848d30dd1855ab9804acd72 Mon Sep 17 00:00:00 2001 From: Vijendar Mukunda Date: Wed, 18 Oct 2017 12:13:58 -0400 Subject: ASoC: AMD: DMA driver changes for Stoney Platform Added DMA driver changes for Stoney platform. Below are the key differences between Stoney and CZ In Stoney, Memory Gating is disabled.SRAM Banks won't be turned off.No Of SRAM Banks reduced to 6. DAGB Garlic Interface used and 16 bit resolution is supported. SRAM bank 1 & SRAM bank 2 will be used for playback scenario. SRAM Bank 3 & SRAM Bank 4 will be used for Capture scenario. Acked-by: Mark Brown Reviewed-by: Alex Deucher Signed-off-by: Vijendar Mukunda Signed-off-by: Alex Deucher Signed-off-by: Mark Brown --- sound/soc/amd/acp-pcm-dma.c | 87 +++++++++++++++++++++++++++++++++------------ sound/soc/amd/acp.h | 2 ++ 2 files changed, 67 insertions(+), 22 deletions(-) (limited to 'sound') diff --git a/sound/soc/amd/acp-pcm-dma.c b/sound/soc/amd/acp-pcm-dma.c index f00b6b92e076..f16e0b8e2ed7 100644 --- a/sound/soc/amd/acp-pcm-dma.c +++ b/sound/soc/amd/acp-pcm-dma.c @@ -137,8 +137,8 @@ static void config_dma_descriptor_in_sram(void __iomem *acp_mmio, * system memory <-> ACP SRAM */ static void set_acp_sysmem_dma_descriptors(void __iomem *acp_mmio, - u32 size, int direction, - u32 pte_offset) + u32 size, int direction, + u32 pte_offset, u32 asic_type) { u16 i; u16 dma_dscr_idx = PLAYBACK_START_DMA_DESCR_CH12; @@ -152,20 +152,42 @@ static void set_acp_sysmem_dma_descriptors(void __iomem *acp_mmio, (size / 2) - (i * (size/2)); dmadscr[i].src = ACP_INTERNAL_APERTURE_WINDOW_0_ADDRESS + (pte_offset * SZ_4K) + (i * (size/2)); - dmadscr[i].xfer_val |= - (ACP_DMA_ATTRIBUTES_DAGB_ONION_TO_SHAREDMEM << 16) | - (size / 2); + switch (asic_type) { + case CHIP_STONEY: + dmadscr[i].xfer_val |= + (ACP_DMA_ATTRIBUTES_DAGB_GARLIC_TO_SHAREDMEM << 16) | + (size / 2); + break; + default: + dmadscr[i].xfer_val |= + (ACP_DMA_ATTRIBUTES_DAGB_ONION_TO_SHAREDMEM << 16) | + (size / 2); + } } else { dma_dscr_idx = CAPTURE_START_DMA_DESCR_CH14 + i; - dmadscr[i].src = ACP_SHARED_RAM_BANK_5_ADDRESS + - (i * (size/2)); - dmadscr[i].dest = ACP_INTERNAL_APERTURE_WINDOW_0_ADDRESS - + (pte_offset * SZ_4K) + - (i * (size/2)); - dmadscr[i].xfer_val |= - BIT(22) | - (ACP_DMA_ATTRIBUTES_SHAREDMEM_TO_DAGB_ONION << 16) | - (size / 2); + switch (asic_type) { + case CHIP_STONEY: + dmadscr[i].src = ACP_SHARED_RAM_BANK_3_ADDRESS + + (i * (size/2)); + dmadscr[i].dest = + ACP_INTERNAL_APERTURE_WINDOW_0_ADDRESS + + (pte_offset * SZ_4K) + (i * (size/2)); + dmadscr[i].xfer_val |= + BIT(22) | + (ACP_DMA_ATTRIBUTES_SHARED_MEM_TO_DAGB_GARLIC << 16) | + (size / 2); + break; + default: + dmadscr[i].src = ACP_SHARED_RAM_BANK_5_ADDRESS + + (i * (size/2)); + dmadscr[i].dest = + ACP_INTERNAL_APERTURE_WINDOW_0_ADDRESS + + (pte_offset * SZ_4K) + (i * (size/2)); + dmadscr[i].xfer_val |= + BIT(22) | + (ACP_DMA_ATTRIBUTES_SHAREDMEM_TO_DAGB_ONION << 16) | + (size / 2); + } } config_dma_descriptor_in_sram(acp_mmio, dma_dscr_idx, &dmadscr[i]); @@ -186,7 +208,8 @@ static void set_acp_sysmem_dma_descriptors(void __iomem *acp_mmio, * ACP SRAM <-> I2S */ static void set_acp_to_i2s_dma_descriptors(void __iomem *acp_mmio, - u32 size, int direction) + u32 size, int direction, + u32 asic_type) { u16 i; @@ -207,8 +230,17 @@ static void set_acp_to_i2s_dma_descriptors(void __iomem *acp_mmio, dma_dscr_idx = CAPTURE_START_DMA_DESCR_CH15 + i; /* dmadscr[i].src is unused by hardware. */ dmadscr[i].src = 0; - dmadscr[i].dest = ACP_SHARED_RAM_BANK_5_ADDRESS + + switch (asic_type) { + case CHIP_STONEY: + dmadscr[i].dest = + ACP_SHARED_RAM_BANK_3_ADDRESS + (i * (size / 2)); + break; + default: + dmadscr[i].dest = + ACP_SHARED_RAM_BANK_5_ADDRESS + + (i * (size / 2)); + } dmadscr[i].xfer_val |= BIT(22) | (FROM_ACP_I2S_1 << 16) | (size / 2); } @@ -264,7 +296,8 @@ static void acp_pte_config(void __iomem *acp_mmio, struct page *pg, } static void config_acp_dma(void __iomem *acp_mmio, - struct audio_substream_data *audio_config) + struct audio_substream_data *audio_config, + u32 asic_type) { u32 pte_offset; @@ -278,11 +311,11 @@ static void config_acp_dma(void __iomem *acp_mmio, /* Configure System memory <-> ACP SRAM DMA descriptors */ set_acp_sysmem_dma_descriptors(acp_mmio, audio_config->size, - audio_config->direction, pte_offset); + audio_config->direction, pte_offset, asic_type); /* Configure ACP SRAM <-> I2S DMA descriptors */ set_acp_to_i2s_dma_descriptors(acp_mmio, audio_config->size, - audio_config->direction); + audio_config->direction, asic_type); } /* Start a given DMA channel transfer */ @@ -502,6 +535,12 @@ static int acp_init(void __iomem *acp_mmio, u32 asic_type) acp_set_sram_bank_state(acp_mmio, bank, false); } + /* Stoney supports 16bit resolution */ + if (asic_type == CHIP_STONEY) { + val = acp_reg_read(acp_mmio, mmACP_I2S_16BIT_RESOLUTION_EN); + val |= 0x03; + acp_reg_write(val, acp_mmio, mmACP_I2S_16BIT_RESOLUTION_EN); + } return 0; } @@ -680,6 +719,8 @@ static int acp_dma_hw_params(struct snd_pcm_substream *substream, struct page *pg; struct snd_pcm_runtime *runtime; struct audio_substream_data *rtd; + struct snd_soc_pcm_runtime *prtd = substream->private_data; + struct audio_drv_data *adata = dev_get_drvdata(prtd->platform->dev); runtime = substream->runtime; rtd = runtime->private_data; @@ -707,7 +748,7 @@ static int acp_dma_hw_params(struct snd_pcm_substream *substream, rtd->num_of_pages = PAGE_ALIGN(size) >> PAGE_SHIFT; rtd->direction = substream->stream; - config_acp_dma(rtd->acp_mmio, rtd); + config_acp_dma(rtd->acp_mmio, rtd, adata->asic_type); status = 0; } else { status = -ENOMEM; @@ -1011,7 +1052,8 @@ static int acp_pcm_resume(struct device *dev) true); } config_acp_dma(adata->acp_mmio, - adata->play_stream->runtime->private_data); + adata->play_stream->runtime->private_data, + adata->asic_type); } if (adata->capture_stream && adata->capture_stream->runtime) { if (adata->asic_type != CHIP_STONEY) { @@ -1020,7 +1062,8 @@ static int acp_pcm_resume(struct device *dev) true); } config_acp_dma(adata->acp_mmio, - adata->capture_stream->runtime->private_data); + adata->capture_stream->runtime->private_data, + adata->asic_type); } acp_reg_write(1, adata->acp_mmio, mmACP_EXTERNAL_INTR_ENB); return 0; diff --git a/sound/soc/amd/acp.h b/sound/soc/amd/acp.h index 28cf9140f49c..a330a99bfff8 100644 --- a/sound/soc/amd/acp.h +++ b/sound/soc/amd/acp.h @@ -19,6 +19,7 @@ /* Capture SRAM address (as a source in dma descriptor) */ #define ACP_SHARED_RAM_BANK_5_ADDRESS 0x400A000 +#define ACP_SHARED_RAM_BANK_3_ADDRESS 0x4006000 #define ACP_DMA_RESET_TIME 10000 #define ACP_CLOCK_EN_TIME_OUT_VALUE 0x000000FF @@ -67,6 +68,7 @@ #define CAPTURE_START_DMA_DESCR_CH15 6 #define CAPTURE_END_DMA_DESCR_CH15 7 +#define mmACP_I2S_16BIT_RESOLUTION_EN 0x5209 enum acp_dma_priority_level { /* 0x0 Specifies the DMA channel is given normal priority */ ACP_DMA_PRIORITY_LEVEL_NORMAL = 0x0, -- cgit v1.2.3 From 9c7d6fabf22b2782d4ab2c03f9e8df2beee6d063 Mon Sep 17 00:00:00 2001 From: Vijendar Mukunda Date: Wed, 18 Oct 2017 12:13:59 -0400 Subject: ASoC: AMD: Audio buffer related changes for Stoney Stoney uses 16kb SRAM memory for playback and 16Kb for capture.Modified Max buffer size to have the correct mapping between System Memory and SRAM. Added snd_pcm_hardware structures for playback and capture for Stoney. Reviewed-by: Alex Deucher Signed-off-by: Vijendar Mukunda Signed-off-by: Alex Deucher Signed-off-by: Mark Brown --- sound/soc/amd/acp-pcm-dma.c | 83 ++++++++++++++++++++++++++++++++++++++++++--- 1 file changed, 78 insertions(+), 5 deletions(-) (limited to 'sound') diff --git a/sound/soc/amd/acp-pcm-dma.c b/sound/soc/amd/acp-pcm-dma.c index f16e0b8e2ed7..73b58ee00383 100644 --- a/sound/soc/amd/acp-pcm-dma.c +++ b/sound/soc/amd/acp-pcm-dma.c @@ -35,6 +35,11 @@ #define MAX_BUFFER (PLAYBACK_MAX_PERIOD_SIZE * PLAYBACK_MAX_NUM_PERIODS) #define MIN_BUFFER MAX_BUFFER +#define ST_PLAYBACK_MAX_PERIOD_SIZE 8192 +#define ST_CAPTURE_MAX_PERIOD_SIZE ST_PLAYBACK_MAX_PERIOD_SIZE +#define ST_MAX_BUFFER (ST_PLAYBACK_MAX_PERIOD_SIZE * PLAYBACK_MAX_NUM_PERIODS) +#define ST_MIN_BUFFER ST_MAX_BUFFER + static const struct snd_pcm_hardware acp_pcm_hardware_playback = { .info = SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BLOCK_TRANSFER | SNDRV_PCM_INFO_MMAP | @@ -73,6 +78,44 @@ static const struct snd_pcm_hardware acp_pcm_hardware_capture = { .periods_max = CAPTURE_MAX_NUM_PERIODS, }; +static const struct snd_pcm_hardware acp_st_pcm_hardware_playback = { + .info = SNDRV_PCM_INFO_INTERLEAVED | + SNDRV_PCM_INFO_BLOCK_TRANSFER | SNDRV_PCM_INFO_MMAP | + SNDRV_PCM_INFO_MMAP_VALID | SNDRV_PCM_INFO_BATCH | + SNDRV_PCM_INFO_PAUSE | SNDRV_PCM_INFO_RESUME, + .formats = SNDRV_PCM_FMTBIT_S16_LE | + SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE, + .channels_min = 1, + .channels_max = 8, + .rates = SNDRV_PCM_RATE_8000_96000, + .rate_min = 8000, + .rate_max = 96000, + .buffer_bytes_max = ST_MAX_BUFFER, + .period_bytes_min = PLAYBACK_MIN_PERIOD_SIZE, + .period_bytes_max = ST_PLAYBACK_MAX_PERIOD_SIZE, + .periods_min = PLAYBACK_MIN_NUM_PERIODS, + .periods_max = PLAYBACK_MAX_NUM_PERIODS, +}; + +static const struct snd_pcm_hardware acp_st_pcm_hardware_capture = { + .info = SNDRV_PCM_INFO_INTERLEAVED | + SNDRV_PCM_INFO_BLOCK_TRANSFER | SNDRV_PCM_INFO_MMAP | + SNDRV_PCM_INFO_MMAP_VALID | SNDRV_PCM_INFO_BATCH | + SNDRV_PCM_INFO_PAUSE | SNDRV_PCM_INFO_RESUME, + .formats = SNDRV_PCM_FMTBIT_S16_LE | + SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE, + .channels_min = 1, + .channels_max = 2, + .rates = SNDRV_PCM_RATE_8000_48000, + .rate_min = 8000, + .rate_max = 48000, + .buffer_bytes_max = ST_MAX_BUFFER, + .period_bytes_min = CAPTURE_MIN_PERIOD_SIZE, + .period_bytes_max = ST_CAPTURE_MAX_PERIOD_SIZE, + .periods_min = CAPTURE_MIN_NUM_PERIODS, + .periods_max = CAPTURE_MAX_NUM_PERIODS, +}; + static u32 acp_reg_read(void __iomem *acp_mmio, u32 reg) { return readl(acp_mmio + (reg * 4)); @@ -664,10 +707,23 @@ static int acp_dma_open(struct snd_pcm_substream *substream) if (adata == NULL) return -ENOMEM; - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) - runtime->hw = acp_pcm_hardware_playback; - else - runtime->hw = acp_pcm_hardware_capture; + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + switch (intr_data->asic_type) { + case CHIP_STONEY: + runtime->hw = acp_st_pcm_hardware_playback; + break; + default: + runtime->hw = acp_pcm_hardware_playback; + } + } else { + switch (intr_data->asic_type) { + case CHIP_STONEY: + runtime->hw = acp_st_pcm_hardware_capture; + break; + default: + runtime->hw = acp_pcm_hardware_capture; + } + } ret = snd_pcm_hw_constraint_integer(runtime, SNDRV_PCM_HW_PARAM_PERIODS); @@ -905,10 +961,27 @@ static int acp_dma_trigger(struct snd_pcm_substream *substream, int cmd) static int acp_dma_new(struct snd_soc_pcm_runtime *rtd) { - return snd_pcm_lib_preallocate_pages_for_all(rtd->pcm, + int ret; + struct audio_drv_data *adata = dev_get_drvdata(rtd->platform->dev); + + switch (adata->asic_type) { + case CHIP_STONEY: + ret = snd_pcm_lib_preallocate_pages_for_all(rtd->pcm, + SNDRV_DMA_TYPE_DEV, + NULL, ST_MIN_BUFFER, + ST_MAX_BUFFER); + break; + default: + ret = snd_pcm_lib_preallocate_pages_for_all(rtd->pcm, SNDRV_DMA_TYPE_DEV, NULL, MIN_BUFFER, MAX_BUFFER); + break; + } + if (ret < 0) + dev_err(rtd->platform->dev, + "buffer preallocation failer error:%d\n", ret); + return ret; } static int acp_dma_close(struct snd_pcm_substream *substream) -- cgit v1.2.3 From 566a1847fb37f1b12d997f85623cbf8658c87394 Mon Sep 17 00:00:00 2001 From: Akshu Agrawal Date: Wed, 18 Oct 2017 12:14:00 -0400 Subject: ASoC: AMD: Add machine driver for cz rt5650 The driver is used for AMD board using rt5650 codec. Reviewed-by: Alex Deucher Signed-off-by: Akshu Agrawal Signed-off-by: Alex Deucher Signed-off-by: Dylan Reid Signed-off-by: Mark Brown --- sound/soc/amd/Kconfig | 7 ++ sound/soc/amd/Makefile | 2 + sound/soc/amd/acp-rt5645.c | 199 +++++++++++++++++++++++++++++++++++++++++++++ 3 files changed, 208 insertions(+) create mode 100644 sound/soc/amd/acp-rt5645.c (limited to 'sound') diff --git a/sound/soc/amd/Kconfig b/sound/soc/amd/Kconfig index 78187eb24f56..d5838402f667 100644 --- a/sound/soc/amd/Kconfig +++ b/sound/soc/amd/Kconfig @@ -2,3 +2,10 @@ config SND_SOC_AMD_ACP tristate "AMD Audio Coprocessor support" help This option enables ACP DMA support on AMD platform. + +config SND_SOC_AMD_CZ_RT5645_MACH + tristate "AMD CZ support for RT5645" + select SND_SOC_RT5645 + depends on SND_SOC_AMD_ACP && I2C + help + This option enables machine driver for rt5645. diff --git a/sound/soc/amd/Makefile b/sound/soc/amd/Makefile index 1a66ec0366b2..eed64ff6c73e 100644 --- a/sound/soc/amd/Makefile +++ b/sound/soc/amd/Makefile @@ -1,3 +1,5 @@ snd-soc-acp-pcm-objs := acp-pcm-dma.o +snd-soc-acp-rt5645-mach-objs := acp-rt5645.o obj-$(CONFIG_SND_SOC_AMD_ACP) += snd-soc-acp-pcm.o +obj-$(CONFIG_SND_SOC_AMD_CZ_RT5645_MACH) += snd-soc-acp-rt5645-mach.o diff --git a/sound/soc/amd/acp-rt5645.c b/sound/soc/amd/acp-rt5645.c new file mode 100644 index 000000000000..941aed6bb364 --- /dev/null +++ b/sound/soc/amd/acp-rt5645.c @@ -0,0 +1,199 @@ +/* + * Machine driver for AMD ACP Audio engine using Realtek RT5645 codec + * + * Copyright 2017 Advanced Micro Devices, Inc. + * + * This file is modified from rt288 machine driver + * + * Permission is hereby granted, free of charge, to any person obtaining a + * copy of this software and associated documentation files (the "Software"), + * to deal in the Software without restriction, including without limitation + * the rights to use, copy, modify, merge, publish, distribute, sublicense, + * and/or sell copies of the Software, and to permit persons to whom the + * Software is furnished to do so, subject to the following conditions: + * + * The above copyright notice and this permission notice shall be included in + * all copies or substantial portions of the Software. + * + * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR + * IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY, + * FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL + * THE COPYRIGHT HOLDER(S) OR AUTHOR(S) BE LIABLE FOR ANY CLAIM, DAMAGES OR + * OTHER LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, + * ARISING FROM, OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR + * OTHER DEALINGS IN THE SOFTWARE. + * + * + */ + +#include +#include +#include +#include +#include +#include +#include +#include +#include +#include + +#include "../codecs/rt5645.h" + +#define CZ_PLAT_CLK 24000000 + +static struct snd_soc_jack cz_jack; + +static int cz_aif1_hw_params(struct snd_pcm_substream *substream, + struct snd_pcm_hw_params *params) +{ + int ret = 0; + struct snd_soc_pcm_runtime *rtd = substream->private_data; + struct snd_soc_dai *codec_dai = rtd->codec_dai; + + ret = snd_soc_dai_set_pll(codec_dai, 0, RT5645_PLL1_S_MCLK, + CZ_PLAT_CLK, params_rate(params) * 512); + if (ret < 0) { + dev_err(rtd->dev, "can't set codec pll: %d\n", ret); + return ret; + } + + ret = snd_soc_dai_set_sysclk(codec_dai, RT5645_SCLK_S_PLL1, + params_rate(params) * 512, SND_SOC_CLOCK_OUT); + if (ret < 0) { + dev_err(rtd->dev, "can't set codec sysclk: %d\n", ret); + return ret; + } + + return ret; +} + +static int cz_init(struct snd_soc_pcm_runtime *rtd) +{ + int ret; + struct snd_soc_card *card; + struct snd_soc_codec *codec; + + codec = rtd->codec; + card = rtd->card; + + ret = snd_soc_card_jack_new(card, "Headset Jack", + SND_JACK_HEADPHONE | SND_JACK_MICROPHONE | + SND_JACK_BTN_0 | SND_JACK_BTN_1 | + SND_JACK_BTN_2 | SND_JACK_BTN_3, + &cz_jack, NULL, 0); + if (ret) { + dev_err(card->dev, "HP jack creation failed %d\n", ret); + return ret; + } + + rt5645_set_jack_detect(codec, &cz_jack, &cz_jack, &cz_jack); + + return 0; +} + +static struct snd_soc_ops cz_aif1_ops = { + .hw_params = cz_aif1_hw_params, +}; + +static struct snd_soc_dai_link cz_dai_rt5650[] = { + { + .name = "amd-rt5645-play", + .stream_name = "RT5645_AIF1", + .platform_name = "acp_audio_dma.0.auto", + .cpu_dai_name = "designware-i2s.1.auto", + .codec_dai_name = "rt5645-aif1", + .codec_name = "i2c-10EC5650:00", + .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF + | SND_SOC_DAIFMT_CBM_CFM, + .init = cz_init, + .ops = &cz_aif1_ops, + }, + { + .name = "amd-rt5645-cap", + .stream_name = "RT5645_AIF1", + .platform_name = "acp_audio_dma.0.auto", + .cpu_dai_name = "designware-i2s.2.auto", + .codec_dai_name = "rt5645-aif1", + .codec_name = "i2c-10EC5650:00", + .dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF + | SND_SOC_DAIFMT_CBM_CFM, + .ops = &cz_aif1_ops, + }, +}; + +static const struct snd_soc_dapm_widget cz_widgets[] = { + SND_SOC_DAPM_HP("Headphones", NULL), + SND_SOC_DAPM_SPK("Speakers", NULL), + SND_SOC_DAPM_MIC("Headset Mic", NULL), + SND_SOC_DAPM_MIC("Int Mic", NULL), +}; + +static const struct snd_soc_dapm_route cz_audio_route[] = { + {"Headphones", NULL, "HPOL"}, + {"Headphones", NULL, "HPOR"}, + {"RECMIXL", NULL, "Headset Mic"}, + {"RECMIXR", NULL, "Headset Mic"}, + {"Speakers", NULL, "SPOL"}, + {"Speakers", NULL, "SPOR"}, + {"DMIC L2", NULL, "Int Mic"}, + {"DMIC R2", NULL, "Int Mic"}, +}; + +static const struct snd_kcontrol_new cz_mc_controls[] = { + SOC_DAPM_PIN_SWITCH("Headphones"), + SOC_DAPM_PIN_SWITCH("Speakers"), + SOC_DAPM_PIN_SWITCH("Headset Mic"), + SOC_DAPM_PIN_SWITCH("Int Mic"), +}; + +static struct snd_soc_card cz_card = { + .name = "acprt5650", + .owner = THIS_MODULE, + .dai_link = cz_dai_rt5650, + .num_links = ARRAY_SIZE(cz_dai_rt5650), + .dapm_widgets = cz_widgets, + .num_dapm_widgets = ARRAY_SIZE(cz_widgets), + .dapm_routes = cz_audio_route, + .num_dapm_routes = ARRAY_SIZE(cz_audio_route), + .controls = cz_mc_controls, + .num_controls = ARRAY_SIZE(cz_mc_controls), +}; + +static int cz_probe(struct platform_device *pdev) +{ + int ret; + struct snd_soc_card *card; + + card = &cz_card; + cz_card.dev = &pdev->dev; + platform_set_drvdata(pdev, card); + ret = devm_snd_soc_register_card(&pdev->dev, &cz_card); + if (ret) { + dev_err(&pdev->dev, + "devm_snd_soc_register_card(%s) failed: %d\n", + cz_card.name, ret); + return ret; + } + return 0; +} + +static const struct acpi_device_id cz_audio_acpi_match[] = { + { "AMDI1002", 0 }, + {}, +}; +MODULE_DEVICE_TABLE(acpi, cz_audio_acpi_match); + +static struct platform_driver cz_pcm_driver = { + .driver = { + .name = "cz-rt5645", + .acpi_match_table = ACPI_PTR(cz_audio_acpi_match), + .pm = &snd_soc_pm_ops, + }, + .probe = cz_probe, +}; + +module_platform_driver(cz_pcm_driver); + +MODULE_AUTHOR("akshu.agrawal@amd.com"); +MODULE_DESCRIPTION("cz-rt5645 audio support"); +MODULE_LICENSE("GPL v2"); -- cgit v1.2.3 From 61add8147942d23519b91f0edc30980d7c14482c Mon Sep 17 00:00:00 2001 From: Vijendar Mukunda Date: Fri, 3 Nov 2017 16:35:43 -0400 Subject: ASoC: amd: Report accurate hw_ptr during dma Using hw register to read transmitted byte count and report accordingly the hw pointer. TEST= modprobe snd-soc-acp-pcm.ko modprobe snd-soc-acp-rt5645.ko aplay Signed-off-by: Vijendar Mukunda Signed-off-by: Akshu Agrawal Tested-by: Akshu Agrawal Reviewed-by: Jason Clinton Signed-off-by: Alex Deucher Signed-off-by: Mark Brown --- sound/soc/amd/acp-pcm-dma.c | 71 ++++++++++++++++++++++++++++----------------- sound/soc/amd/acp.h | 10 +++++++ 2 files changed, 55 insertions(+), 26 deletions(-) (limited to 'sound') diff --git a/sound/soc/amd/acp-pcm-dma.c b/sound/soc/amd/acp-pcm-dma.c index 73b58ee00383..e19f281afeaa 100644 --- a/sound/soc/amd/acp-pcm-dma.c +++ b/sound/soc/amd/acp-pcm-dma.c @@ -817,39 +817,48 @@ static int acp_dma_hw_free(struct snd_pcm_substream *substream) return snd_pcm_lib_free_pages(substream); } +static u64 acp_get_byte_count(void __iomem *acp_mmio, int stream) +{ + union acp_dma_count playback_dma_count; + union acp_dma_count capture_dma_count; + u64 bytescount = 0; + + if (stream == SNDRV_PCM_STREAM_PLAYBACK) { + playback_dma_count.bcount.high = acp_reg_read(acp_mmio, + mmACP_I2S_TRANSMIT_BYTE_CNT_HIGH); + playback_dma_count.bcount.low = acp_reg_read(acp_mmio, + mmACP_I2S_TRANSMIT_BYTE_CNT_LOW); + bytescount = playback_dma_count.bytescount; + } else { + capture_dma_count.bcount.high = acp_reg_read(acp_mmio, + mmACP_I2S_RECEIVED_BYTE_CNT_HIGH); + capture_dma_count.bcount.low = acp_reg_read(acp_mmio, + mmACP_I2S_RECEIVED_BYTE_CNT_LOW); + bytescount = capture_dma_count.bytescount; + } + return bytescount; +} + static snd_pcm_uframes_t acp_dma_pointer(struct snd_pcm_substream *substream) { - u16 dscr; - u32 mul, dma_config, period_bytes; + u32 buffersize; u32 pos = 0; + u64 bytescount = 0; struct snd_pcm_runtime *runtime = substream->runtime; struct audio_substream_data *rtd = runtime->private_data; - period_bytes = frames_to_bytes(runtime, runtime->period_size); - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { - dscr = acp_reg_read(rtd->acp_mmio, mmACP_DMA_CUR_DSCR_13); + buffersize = frames_to_bytes(runtime, runtime->buffer_size); + bytescount = acp_get_byte_count(rtd->acp_mmio, substream->stream); - if (dscr == PLAYBACK_START_DMA_DESCR_CH13) - mul = 0; - else - mul = 1; - pos = (mul * period_bytes); + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + if (bytescount > rtd->renderbytescount) + bytescount = bytescount - rtd->renderbytescount; + pos = bytescount % buffersize; } else { - dma_config = acp_reg_read(rtd->acp_mmio, mmACP_DMA_CNTL_14); - if (dma_config != 0) { - dscr = acp_reg_read(rtd->acp_mmio, - mmACP_DMA_CUR_DSCR_14); - if (dscr == CAPTURE_START_DMA_DESCR_CH14) - mul = 1; - else - mul = 2; - pos = (mul * period_bytes); - } - - if (pos >= (2 * period_bytes)) - pos = 0; - + if (bytescount > rtd->capturebytescount) + bytescount = bytescount - rtd->capturebytescount; + pos = bytescount % buffersize; } return bytes_to_frames(runtime, pos); } @@ -904,6 +913,7 @@ static int acp_dma_trigger(struct snd_pcm_substream *substream, int cmd) { int ret; u32 loops = 1000; + u64 bytescount = 0; struct snd_pcm_runtime *runtime = substream->runtime; struct snd_soc_pcm_runtime *prtd = substream->private_data; @@ -915,7 +925,11 @@ static int acp_dma_trigger(struct snd_pcm_substream *substream, int cmd) case SNDRV_PCM_TRIGGER_START: case SNDRV_PCM_TRIGGER_PAUSE_RELEASE: case SNDRV_PCM_TRIGGER_RESUME: + bytescount = acp_get_byte_count(rtd->acp_mmio, + substream->stream); if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { + if (rtd->renderbytescount == 0) + rtd->renderbytescount = bytescount; acp_dma_start(rtd->acp_mmio, SYSRAM_TO_ACP_CH_NUM, false); while (acp_reg_read(rtd->acp_mmio, mmACP_DMA_CH_STS) & @@ -932,6 +946,8 @@ static int acp_dma_trigger(struct snd_pcm_substream *substream, int cmd) ACP_TO_I2S_DMA_CH_NUM, true); } else { + if (rtd->capturebytescount == 0) + rtd->capturebytescount = bytescount; acp_dma_start(rtd->acp_mmio, I2S_TO_ACP_DMA_CH_NUM, true); } @@ -945,12 +961,15 @@ static int acp_dma_trigger(struct snd_pcm_substream *substream, int cmd) * channels will stopped automatically after its transfer * completes : SYSRAM_TO_ACP_CH_NUM / ACP_TO_SYSRAM_CH_NUM */ - if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) + if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { ret = acp_dma_stop(rtd->acp_mmio, ACP_TO_I2S_DMA_CH_NUM); - else + rtd->renderbytescount = 0; + } else { ret = acp_dma_stop(rtd->acp_mmio, I2S_TO_ACP_DMA_CH_NUM); + rtd->capturebytescount = 0; + } break; default: ret = -EINVAL; diff --git a/sound/soc/amd/acp.h b/sound/soc/amd/acp.h index a330a99bfff8..de08ff077ac7 100644 --- a/sound/soc/amd/acp.h +++ b/sound/soc/amd/acp.h @@ -83,6 +83,8 @@ struct audio_substream_data { u16 num_of_pages; u16 direction; uint64_t size; + u64 renderbytescount; + u64 capturebytescount; void __iomem *acp_mmio; }; @@ -93,6 +95,14 @@ struct audio_drv_data { u32 asic_type; }; +union acp_dma_count { + struct { + u32 low; + u32 high; + } bcount; + u64 bytescount; +}; + enum { ACP_TILE_P1 = 0, ACP_TILE_P2, -- cgit v1.2.3 From bdd2a858afd55cc11723df9dd2841241a4c49ce5 Mon Sep 17 00:00:00 2001 From: Akshu Agrawal Date: Wed, 8 Nov 2017 12:24:02 -0500 Subject: ASoC: amd: Make the driver name consistent across files This fixes the issue of driver not getting auto loaded with MODULE_ALIAS. find /sys/devices -name modalias -print0 | xargs -0 grep 'audio' /sys/devices/pci0000:00/0000:00:01.0/acp_audio_dma.0.auto/modalias:platform:acp_audio_dma TEST=boot and check for device in lsmod [Removed yet more ChromeOS crap from the changelog -- broonie] Signed-off-by: Akshu Agrawal Tested-by: Jason Clinton Reviewed-by: Jason Clinton Signed-off-by: Alex Deucher Signed-off-by: Mark Brown --- sound/soc/amd/Makefile | 4 ++-- sound/soc/amd/acp-pcm-dma.c | 6 ++++-- 2 files changed, 6 insertions(+), 4 deletions(-) (limited to 'sound') diff --git a/sound/soc/amd/Makefile b/sound/soc/amd/Makefile index eed64ff6c73e..f07fd2e2870a 100644 --- a/sound/soc/amd/Makefile +++ b/sound/soc/amd/Makefile @@ -1,5 +1,5 @@ -snd-soc-acp-pcm-objs := acp-pcm-dma.o +acp_audio_dma-objs := acp-pcm-dma.o snd-soc-acp-rt5645-mach-objs := acp-rt5645.o -obj-$(CONFIG_SND_SOC_AMD_ACP) += snd-soc-acp-pcm.o +obj-$(CONFIG_SND_SOC_AMD_ACP) += acp_audio_dma.o obj-$(CONFIG_SND_SOC_AMD_CZ_RT5645_MACH) += snd-soc-acp-rt5645-mach.o diff --git a/sound/soc/amd/acp-pcm-dma.c b/sound/soc/amd/acp-pcm-dma.c index e19f281afeaa..13d040a4d26f 100644 --- a/sound/soc/amd/acp-pcm-dma.c +++ b/sound/soc/amd/acp-pcm-dma.c @@ -40,6 +40,8 @@ #define ST_MAX_BUFFER (ST_PLAYBACK_MAX_PERIOD_SIZE * PLAYBACK_MAX_NUM_PERIODS) #define ST_MIN_BUFFER ST_MAX_BUFFER +#define DRV_NAME "acp_audio_dma" + static const struct snd_pcm_hardware acp_pcm_hardware_playback = { .info = SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BLOCK_TRANSFER | SNDRV_PCM_INFO_MMAP | @@ -1189,7 +1191,7 @@ static struct platform_driver acp_dma_driver = { .probe = acp_audio_probe, .remove = acp_audio_remove, .driver = { - .name = "acp_audio_dma", + .name = DRV_NAME, .pm = &acp_pm_ops, }, }; @@ -1200,4 +1202,4 @@ MODULE_AUTHOR("Vijendar.Mukunda@amd.com"); MODULE_AUTHOR("Maruthi.Bayyavarapu@amd.com"); MODULE_DESCRIPTION("AMD ACP PCM Driver"); MODULE_LICENSE("GPL v2"); -MODULE_ALIAS("platform:acp-dma-audio"); +MODULE_ALIAS("platform:"DRV_NAME); -- cgit v1.2.3 From 7db08b2cb36cbfbcb06c44dc8e48ccb6a119466f Mon Sep 17 00:00:00 2001 From: Guenter Roeck Date: Wed, 8 Nov 2017 16:34:54 -0500 Subject: ASoC: amd: use do_div rather than 64 bit division to fix 32 bit builds ERROR: "__aeabi_uldivmod" [sound/soc/amd/snd-soc-acp-pcm.ko] undefined! 64-bit divides require special operations to avoid build errors on 32-bit systems. [Reword the commit message to make it clearer - Alex] fixes: 61add8147942 (ASoC: amd: Report accurate hw_ptr during dma) Signed-off-by: Guenter Roeck Reviewed-on: https://chromium-review.googlesource.com/678919 Reviewed-by: Jason Clinton Reviewed-on: https://chromium-review.googlesource.com/681618 Signed-off-by: Alex Deucher Signed-off-by: Mark Brown --- sound/soc/amd/acp-pcm-dma.c | 3 +-- 1 file changed, 1 insertion(+), 2 deletions(-) (limited to 'sound') diff --git a/sound/soc/amd/acp-pcm-dma.c b/sound/soc/amd/acp-pcm-dma.c index 13d040a4d26f..ef7e98ad960c 100644 --- a/sound/soc/amd/acp-pcm-dma.c +++ b/sound/soc/amd/acp-pcm-dma.c @@ -856,12 +856,11 @@ static snd_pcm_uframes_t acp_dma_pointer(struct snd_pcm_substream *substream) if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) { if (bytescount > rtd->renderbytescount) bytescount = bytescount - rtd->renderbytescount; - pos = bytescount % buffersize; } else { if (bytescount > rtd->capturebytescount) bytescount = bytescount - rtd->capturebytescount; - pos = bytescount % buffersize; } + pos = do_div(bytescount, buffersize); return bytes_to_frames(runtime, pos); } -- cgit v1.2.3 From 31c45b3e8d0ecc3a5cbfbf3dfe18adeab2f17a48 Mon Sep 17 00:00:00 2001 From: Vijendar Mukunda Date: Thu, 9 Nov 2017 12:35:52 -0500 Subject: ASoC: amd: Modified DMA transfer Mechanism for Playback Before rendering starts, DMA driver copies full buffer valid data to ACP SRAM for the first time, after that ACP SRAM to I2S FIFO DMA will be initiated. After rendering first half of ACP SRAM, IOC will be raised then Audio data will be copied from first half of System Memory to first half of ACP SRAM. Similarly after rendering second half of ACP SRAM, IOC will be raised then Audio Data will be copied from second half of the System Memory to second half of the ACP SRAM in ping-pong way till rendering stops. Old design introducing latency issues resulting stutter sound observed during playback. Signed-off-by: Vijendar Mukunda Signed-off-by: Akshu Agrawal Signed-off-by: Alex Deucher Signed-off-by: Mark Brown --- sound/soc/amd/acp-pcm-dma.c | 27 +++++---------------------- 1 file changed, 5 insertions(+), 22 deletions(-) (limited to 'sound') diff --git a/sound/soc/amd/acp-pcm-dma.c b/sound/soc/amd/acp-pcm-dma.c index ef7e98ad960c..9f521a55d610 100644 --- a/sound/soc/amd/acp-pcm-dma.c +++ b/sound/soc/amd/acp-pcm-dma.c @@ -193,8 +193,8 @@ static void set_acp_sysmem_dma_descriptors(void __iomem *acp_mmio, dmadscr[i].xfer_val = 0; if (direction == SNDRV_PCM_STREAM_PLAYBACK) { dma_dscr_idx = PLAYBACK_START_DMA_DESCR_CH12 + i; - dmadscr[i].dest = ACP_SHARED_RAM_BANK_1_ADDRESS + - (size / 2) - (i * (size/2)); + dmadscr[i].dest = ACP_SHARED_RAM_BANK_1_ADDRESS + + (i * (size/2)); dmadscr[i].src = ACP_INTERNAL_APERTURE_WINDOW_0_ADDRESS + (pte_offset * SZ_4K) + (i * (size/2)); switch (asic_type) { @@ -655,9 +655,9 @@ static irqreturn_t dma_irq_handler(int irq, void *arg) valid_irq = true; if (acp_reg_read(acp_mmio, mmACP_DMA_CUR_DSCR_13) == PLAYBACK_START_DMA_DESCR_CH13) - dscr_idx = PLAYBACK_START_DMA_DESCR_CH12; - else dscr_idx = PLAYBACK_END_DMA_DESCR_CH12; + else + dscr_idx = PLAYBACK_START_DMA_DESCR_CH12; config_acp_dma_channel(acp_mmio, SYSRAM_TO_ACP_CH_NUM, dscr_idx, 1, 0); acp_dma_start(acp_mmio, SYSRAM_TO_ACP_CH_NUM, false); @@ -882,23 +882,6 @@ static int acp_dma_prepare(struct snd_pcm_substream *substream) config_acp_dma_channel(rtd->acp_mmio, ACP_TO_I2S_DMA_CH_NUM, PLAYBACK_START_DMA_DESCR_CH13, NUM_DSCRS_PER_CHANNEL, 0); - /* Fill ACP SRAM (2 periods) with zeros from System RAM - * which is zero-ed in hw_params - */ - acp_dma_start(rtd->acp_mmio, SYSRAM_TO_ACP_CH_NUM, false); - - /* ACP SRAM (2 periods of buffer size) is intially filled with - * zeros. Before rendering starts, 2nd half of SRAM will be - * filled with valid audio data DMA'ed from first half of system - * RAM and 1st half of SRAM will be filled with Zeros. This is - * the initial scenario when redering starts from SRAM. Later - * on, 2nd half of system memory will be DMA'ed to 1st half of - * SRAM, 1st half of system memory will be DMA'ed to 2nd half of - * SRAM in ping-pong way till rendering stops. - */ - config_acp_dma_channel(rtd->acp_mmio, SYSRAM_TO_ACP_CH_NUM, - PLAYBACK_START_DMA_DESCR_CH12, - 1, 0); } else { config_acp_dma_channel(rtd->acp_mmio, ACP_TO_SYSRAM_CH_NUM, CAPTURE_START_DMA_DESCR_CH14, @@ -913,7 +896,7 @@ static int acp_dma_prepare(struct snd_pcm_substream *substream) static int acp_dma_trigger(struct snd_pcm_substream *substream, int cmd) { int ret; - u32 loops = 1000; + u32 loops = 4000; u64 bytescount = 0; struct snd_pcm_runtime *runtime = substream->runtime; -- cgit v1.2.3