[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>

1 files changed, 202 insertions, 0 deletions

diff --git a/sound/aoa/soundbus/soundbus.h b/sound/aoa/soundbus/soundbus.h
new file mode 100644
index 00000000000..5c27297835d
--- /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 */