mirror of
https://github.com/AuxXxilium/linux_dsm_epyc7002.git
synced 2024-12-23 20:45:20 +07:00
f6af5df0c7
Update the aoa-soundbus framework to use dev_pm_ops rather than the deprecated legacy suspend and resume callbacks. Since there isn't anything special to do at the bus level the bus driver does not have to implement any callbacks. The device driver core will automatically pick up and execute the device's PM ops. As there is only a single aoa-soundbus driver implementing suspend and resume, update both the core and driver at the same time to avoid unnecessary code churn. Signed-off-by: Lars-Peter Clausen <lars@metafoo.de> Signed-off-by: Takashi Iwai <tiwai@suse.de>
203 lines
6.3 KiB
C
203 lines
6.3 KiB
C
/*
|
|
* 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 */
|