[ALSA] snd-aoa: add snd-aoa
This large patch adds all of snd-aoa.
Consisting of many modules, it currently replaces snd-powermac
for all layout-id based machines and handles many more (for
example new powerbooks and powermacs with digital output that
previously couldn't be used at all).
It also has support for all layout-IDs that Apple has (judging
from their Info.plist file) but not all are tested.
The driver currently has 2 known regressions over snd-powermac:
* it doesn't handle powermac 7,2 and 7,3
* it doesn't have a DRC control on snapper-based machines
I will fix those during the 2.6.18 development cycle.
Signed-off-by: Johannes Berg <johannes@sipsolutions.net>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
diff --git a/sound/aoa/soundbus/soundbus.h b/sound/aoa/soundbus/soundbus.h
new file mode 100644
index 0000000..5c27297
--- /dev/null
+++ b/sound/aoa/soundbus/soundbus.h
@@ -0,0 +1,202 @@
+/*
+ * soundbus generic definitions
+ *
+ * Copyright 2006 Johannes Berg <johannes@sipsolutions.net>
+ *
+ * GPL v2, can be found in COPYING.
+ */
+#ifndef __SOUNDBUS_H
+#define __SOUNDBUS_H
+
+#include <asm/of_device.h>
+#include <sound/pcm.h>
+#include <linux/list.h>
+
+
+/* When switching from master to slave or the other way around,
+ * you don't want to have the codec chip acting as clock source
+ * while the bus still is.
+ * More importantly, while switch from slave to master, you need
+ * to turn off the chip's master function first, but then there's
+ * no clock for a while and other chips might reset, so we notify
+ * their drivers after having switched.
+ * The constants here are codec-point of view, so when we switch
+ * the soundbus to master we tell the codec we're going to switch
+ * and give it CLOCK_SWITCH_PREPARE_SLAVE!
+ */
+enum clock_switch {
+ CLOCK_SWITCH_PREPARE_SLAVE,
+ CLOCK_SWITCH_PREPARE_MASTER,
+ CLOCK_SWITCH_SLAVE,
+ CLOCK_SWITCH_MASTER,
+ CLOCK_SWITCH_NOTIFY,
+};
+
+/* information on a transfer the codec can take */
+struct transfer_info {
+ u64 formats; /* SNDRV_PCM_FMTBIT_* */
+ unsigned int rates; /* SNDRV_PCM_RATE_* */
+ /* flags */
+ u32 transfer_in:1, /* input = 1, output = 0 */
+ must_be_clock_source:1;
+ /* for codecs to distinguish among their TIs */
+ int tag;
+};
+
+struct codec_info_item {
+ struct codec_info *codec;
+ void *codec_data;
+ struct soundbus_dev *sdev;
+ /* internal, to be used by the soundbus provider */
+ struct list_head list;
+};
+
+/* for prepare, where the codecs need to know
+ * what we're going to drive the bus with */
+struct bus_info {
+ /* see below */
+ int sysclock_factor;
+ int bus_factor;
+};
+
+/* information on the codec itself, plus function pointers */
+struct codec_info {
+ /* the module this lives in */
+ struct module *owner;
+
+ /* supported transfer possibilities, array terminated by
+ * formats or rates being 0. */
+ struct transfer_info *transfers;
+
+ /* Master clock speed factor
+ * to be used (master clock speed = sysclock_factor * sampling freq)
+ * Unused if the soundbus provider has no such notion.
+ */
+ int sysclock_factor;
+
+ /* Bus factor, bus clock speed = bus_factor * sampling freq)
+ * Unused if the soundbus provider has no such notion.
+ */
+ int bus_factor;
+
+ /* operations */
+ /* clock switching, see above */
+ int (*switch_clock)(struct codec_info_item *cii,
+ enum clock_switch clock);
+
+ /* called for each transfer_info when the user
+ * opens the pcm device to determine what the
+ * hardware can support at this point in time.
+ * That can depend on other user-switchable controls.
+ * Return 1 if usable, 0 if not.
+ * out points to another instance of a transfer_info
+ * which is initialised to the values in *ti, and
+ * it's format and rate values can be modified by
+ * the callback if it is necessary to further restrict
+ * the formats that can be used at the moment, for
+ * example when one codec has multiple logical codec
+ * info structs for multiple inputs.
+ */
+ int (*usable)(struct codec_info_item *cii,
+ struct transfer_info *ti,
+ struct transfer_info *out);
+
+ /* called when pcm stream is opened, probably not implemented
+ * most of the time since it isn't too useful */
+ int (*open)(struct codec_info_item *cii,
+ struct snd_pcm_substream *substream);
+
+ /* called when the pcm stream is closed, at this point
+ * the user choices can all be unlocked (see below) */
+ int (*close)(struct codec_info_item *cii,
+ struct snd_pcm_substream *substream);
+
+ /* if the codec must forbid some user choices because
+ * they are not valid with the substream/transfer info,
+ * it must do so here. Example: no digital output for
+ * incompatible framerate, say 8KHz, on Onyx.
+ * If the selected stuff in the substream is NOT
+ * compatible, you have to reject this call! */
+ int (*prepare)(struct codec_info_item *cii,
+ struct bus_info *bi,
+ struct snd_pcm_substream *substream);
+
+ /* start() is called before data is pushed to the codec.
+ * Note that start() must be atomic! */
+ int (*start)(struct codec_info_item *cii,
+ struct snd_pcm_substream *substream);
+
+ /* stop() is called after data is no longer pushed to the codec.
+ * Note that stop() must be atomic! */
+ int (*stop)(struct codec_info_item *cii,
+ struct snd_pcm_substream *substream);
+
+ int (*suspend)(struct codec_info_item *cii, pm_message_t state);
+ int (*resume)(struct codec_info_item *cii);
+};
+
+/* information on a soundbus device */
+struct soundbus_dev {
+ /* the bus it belongs to */
+ struct list_head onbuslist;
+
+ /* the of device it represents */
+ struct of_device ofdev;
+
+ /* what modules go by */
+ char modalias[32];
+
+ /* These fields must be before attach_codec can be called.
+ * They should be set by the owner of the alsa card object
+ * that is needed, and whoever sets them must make sure
+ * that they are unique within that alsa card object. */
+ char *pcmname;
+ int pcmid;
+
+ /* this is assigned by the soundbus provider in attach_codec */
+ struct snd_pcm *pcm;
+
+ /* operations */
+ /* attach a codec to this soundbus, give the alsa
+ * card object the PCMs for this soundbus should be in.
+ * The 'data' pointer must be unique, it is used as the
+ * key for detach_codec(). */
+ int (*attach_codec)(struct soundbus_dev *dev, struct snd_card *card,
+ struct codec_info *ci, void *data);
+ void (*detach_codec)(struct soundbus_dev *dev, void *data);
+ /* TODO: suspend/resume */
+
+ /* private for the soundbus provider */
+ struct list_head codec_list;
+ u32 have_out:1, have_in:1;
+};
+#define to_soundbus_device(d) container_of(d, struct soundbus_dev, ofdev.dev)
+#define of_to_soundbus_device(d) container_of(d, struct soundbus_dev, ofdev)
+
+extern int soundbus_add_one(struct soundbus_dev *dev);
+extern void soundbus_remove_one(struct soundbus_dev *dev);
+
+extern struct soundbus_dev *soundbus_dev_get(struct soundbus_dev *dev);
+extern void soundbus_dev_put(struct soundbus_dev *dev);
+
+struct soundbus_driver {
+ char *name;
+ struct module *owner;
+
+ /* we don't implement any matching at all */
+
+ int (*probe)(struct soundbus_dev* dev);
+ int (*remove)(struct soundbus_dev* dev);
+
+ int (*suspend)(struct soundbus_dev* dev, pm_message_t state);
+ int (*resume)(struct soundbus_dev* dev);
+ int (*shutdown)(struct soundbus_dev* dev);
+
+ struct device_driver driver;
+};
+#define to_soundbus_driver(drv) container_of(drv,struct soundbus_driver, driver)
+
+extern int soundbus_register_driver(struct soundbus_driver *drv);
+extern void soundbus_unregister_driver(struct soundbus_driver *drv);
+
+#endif /* __SOUNDBUS_H */