From 6b9331165e9827e055389e22d1cbdb5fe3cff835 Mon Sep 17 00:00:00 2001
From: Mark Brown <broonie@opensource.wolfsonmicro.com>
Date: Thu, 16 Oct 2008 11:00:07 +0100
Subject: ALSA: ASoC: Remove snd_soc_dapm_connect_input()

This was marked as deprecated in 2.6.27 and all users except for
playpaq_wm8510 fixed in that release.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 include/sound/soc-dapm.h | 2 --
 1 file changed, 2 deletions(-)

(limited to 'include/sound')

diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h
index ca699a3017f..7ee2f70ca42 100644
--- a/include/sound/soc-dapm.h
+++ b/include/sound/soc-dapm.h
@@ -221,8 +221,6 @@ int snd_soc_dapm_new_controls(struct snd_soc_codec *codec,
 	int num);
 
 /* dapm path setup */
-int  __deprecated snd_soc_dapm_connect_input(struct snd_soc_codec *codec,
-	const char *sink_name, const char *control_name, const char *src_name);
 int snd_soc_dapm_new_widgets(struct snd_soc_codec *codec);
 void snd_soc_dapm_free(struct snd_soc_device *socdev);
 int snd_soc_dapm_add_routes(struct snd_soc_codec *codec,
-- 
cgit v1.2.3-18-g5258


From bbaf5e97337287479eb78dbc3822d9560bbfd2e2 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Fri, 24 Oct 2008 18:16:50 +0200
Subject: ALSA: Add hrtimer backend for ALSA timer interface

Added the hrtimer backend for ALSA timer interface.
It can be used for the sequencer timer source.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 include/sound/asound.h | 1 +
 1 file changed, 1 insertion(+)

(limited to 'include/sound')

diff --git a/include/sound/asound.h b/include/sound/asound.h
index 2c4dc908a54..1c02ed1d7c4 100644
--- a/include/sound/asound.h
+++ b/include/sound/asound.h
@@ -575,6 +575,7 @@ enum {
 #define SNDRV_TIMER_GLOBAL_SYSTEM	0
 #define SNDRV_TIMER_GLOBAL_RTC		1
 #define SNDRV_TIMER_GLOBAL_HPET		2
+#define SNDRV_TIMER_GLOBAL_HRTIMER	3
 
 /* info flags */
 #define SNDRV_TIMER_FLG_SLAVE		(1<<0)	/* cannot be controlled */
-- 
cgit v1.2.3-18-g5258


From a53ccab3ccac9e8676a683df9822a2daec83ef54 Mon Sep 17 00:00:00 2001
From: Matthew Ranostay <mranostay@embeddedalley.com>
Date: Sat, 25 Oct 2008 01:05:04 -0400
Subject: ALSA: jack: lineout support to jack abstraction layer

This patch introduces support for reporting SW_LINEOUT_INSERT detection events
via the jack abstraction layer.

Also adds a SND_JACK_LINEOUT define to the input system header.

Signed-off-by: Matthew Ranostay <mranostay@embeddedalley.com>
Cc: Dmitry Torokhov <dtor@mail.ru>
Acked-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 include/sound/jack.h | 1 +
 1 file changed, 1 insertion(+)

(limited to 'include/sound')

diff --git a/include/sound/jack.h b/include/sound/jack.h
index b1b2b8b59ad..7cb25f4b50b 100644
--- a/include/sound/jack.h
+++ b/include/sound/jack.h
@@ -35,6 +35,7 @@ enum snd_jack_types {
 	SND_JACK_HEADPHONE	= 0x0001,
 	SND_JACK_MICROPHONE	= 0x0002,
 	SND_JACK_HEADSET	= SND_JACK_HEADPHONE | SND_JACK_MICROPHONE,
+	SND_JACK_LINEOUT	= 0x0004,
 };
 
 struct snd_jack {
-- 
cgit v1.2.3-18-g5258


From 9cbbb3ac628227ec5b65fc043539949db606cd17 Mon Sep 17 00:00:00 2001
From: Jaroslav Kysela <perex@perex.cz>
Date: Wed, 29 Oct 2008 13:41:35 +0100
Subject: ALSA: Release v1.0.18

Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 include/sound/version.h | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

(limited to 'include/sound')

diff --git a/include/sound/version.h b/include/sound/version.h
index 4aafeda8863..eae31557356 100644
--- a/include/sound/version.h
+++ b/include/sound/version.h
@@ -1,3 +1,3 @@
 /* include/version.h */
-#define CONFIG_SND_VERSION "1.0.18rc3"
+#define CONFIG_SND_VERSION "1.0.18"
 #define CONFIG_SND_DATE ""
-- 
cgit v1.2.3-18-g5258


From 12ef193d5817504621e503e78d641265f6a86ac4 Mon Sep 17 00:00:00 2001
From: Troy Kisky <troy.kisky@boundarydevices.com>
Date: Mon, 13 Oct 2008 17:42:14 -0700
Subject: ASoC: Allow setting codec register with debugfs filesystem

i.e. echo 6 59 >/sys/kernel/debug/soc-audio.0/codec_reg
will set register 0x06 to a value of 0x59.
Also, pop_time debugfs interface setup is moved so that it
is setup in the same function as codec_reg

Signed-off-by: Troy Kisky <troy.kisky@boundarydevices.com>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 include/sound/soc.h | 4 ++++
 1 file changed, 4 insertions(+)

(limited to 'include/sound')

diff --git a/include/sound/soc.h b/include/sound/soc.h
index a1e0357a84d..d33825d624a 100644
--- a/include/sound/soc.h
+++ b/include/sound/soc.h
@@ -425,6 +425,7 @@ struct snd_soc_codec {
 	short reg_cache_step;
 
 	/* dapm */
+	u32 pop_time;
 	struct list_head dapm_widgets;
 	struct list_head dapm_paths;
 	enum snd_soc_bias_level bias_level;
@@ -516,6 +517,9 @@ struct snd_soc_device {
 	struct delayed_work delayed_work;
 	struct work_struct deferred_resume_work;
 	void *codec_data;
+#ifdef CONFIG_DEBUG_FS
+	struct dentry	*debugfs_root;
+#endif
 };
 
 /* runtime channel data */
-- 
cgit v1.2.3-18-g5258


From ea913940c39a61214c799cc7093d7b20fe11a94c Mon Sep 17 00:00:00 2001
From: Mark Brown <broonie@opensource.wolfsonmicro.com>
Date: Wed, 5 Nov 2008 11:13:21 +0000
Subject: ASoC: Remove core version number

Rather than try to remember to keep the core version number updated
(which hasn't been happening) just remove it.  It was much more useful
when ASoC was out of tree.

Signed-off-by: Mark brown <broonie@opensource.wolfsonmicro.com>
---
 include/sound/soc.h | 2 --
 1 file changed, 2 deletions(-)

(limited to 'include/sound')

diff --git a/include/sound/soc.h b/include/sound/soc.h
index da0040b69c2..fa1b99b4589 100644
--- a/include/sound/soc.h
+++ b/include/sound/soc.h
@@ -21,8 +21,6 @@
 #include <sound/control.h>
 #include <sound/ac97_codec.h>
 
-#define SND_SOC_VERSION "0.13.2"
-
 /*
  * Convenience kcontrol builders
  */
-- 
cgit v1.2.3-18-g5258


From 945eed02cd619f525e097319cd3d18c58d01da87 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Fri, 7 Nov 2008 09:08:15 +0100
Subject: ALSA: Evaluate condition in snd_BUG_ON() in non-debugging case

Change snd_BUG_ON() to evaluate the given condition, at least, in syntax
for avoiding compile warnings such as unused variables.  The compiler
should optimize out the condition evaluation in the real code, though.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 include/sound/core.h | 4 ++--
 1 file changed, 2 insertions(+), 2 deletions(-)

(limited to 'include/sound')

diff --git a/include/sound/core.h b/include/sound/core.h
index 7e558947268..6fa4c7b1970 100644
--- a/include/sound/core.h
+++ b/include/sound/core.h
@@ -390,11 +390,11 @@ void snd_verbose_printd(const char *file, int line, const char *format, ...)
 
 #define snd_printd(fmt, args...)	do { } while (0)
 #define snd_BUG()			do { } while (0)
-static inline int __snd_bug_on(void)
+static inline int __snd_bug_on(int cond)
 {
 	return 0;
 }
-#define snd_BUG_ON(cond)		__snd_bug_on()  /* always false */
+#define snd_BUG_ON(cond)	__snd_bug_on(0 && (cond))  /* always false */
 
 #endif /* CONFIG_SND_DEBUG */
 
-- 
cgit v1.2.3-18-g5258


From f66fcedc84dd06d42a0dba3894d238498509e8b7 Mon Sep 17 00:00:00 2001
From: Takashi Iwai <tiwai@suse.de>
Date: Fri, 7 Nov 2008 09:37:22 +0100
Subject: ALSA: Document debug macros

Add descriptions of snd_BUG() and snd_BUG_ON().
Also fixed a typo in the comment of snd_printk(), too.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 include/sound/core.h | 24 +++++++++++++++++++++++-
 1 file changed, 23 insertions(+), 1 deletion(-)

(limited to 'include/sound')

diff --git a/include/sound/core.h b/include/sound/core.h
index 6fa4c7b1970..b2f3bbe9e56 100644
--- a/include/sound/core.h
+++ b/include/sound/core.h
@@ -356,7 +356,7 @@ void snd_verbose_printd(const char *file, int line, const char *format, ...)
  * snd_printk - printk wrapper
  * @fmt: format string
  *
- * Works like print() but prints the file and the line of the caller
+ * Works like printk() but prints the file and the line of the caller
  * when configured with CONFIG_SND_VERBOSE_PRINTK.
  */
 #define snd_printk(fmt, args...) \
@@ -383,7 +383,29 @@ void snd_verbose_printd(const char *file, int line, const char *format, ...)
 	printk(fmt ,##args)
 #endif
 
+/**
+ * snd_BUG - give a BUG warning message and stack trace
+ *
+ * Calls WARN() if CONFIG_SND_DEBUG is set.
+ * Ignored when CONFIG_SND_DEBUG is not set.
+ */
 #define snd_BUG()		WARN(1, "BUG?\n")
+
+/**
+ * snd_BUG_ON - debugging check macro
+ * @cond: condition to evaluate
+ *
+ * When CONFIG_SND_DEBUG is set, this macro evaluates the given condition,
+ * and call WARN() and returns the value if it's non-zero.
+ * 
+ * When CONFIG_SND_DEBUG is not set, this just returns zero, and the given
+ * condition is ignored.
+ *
+ * NOTE: the argument won't be evaluated at all when CONFIG_SND_DEBUG=n.
+ * Thus, don't put any statement that influences on the code behavior,
+ * such as pre/post increment, to the argument of this macro.
+ * If you want to evaluate and give a warning, use standard WARN_ON().
+ */
 #define snd_BUG_ON(cond)	WARN((cond), "BUG? (%s)\n", __stringify(cond))
 
 #else /* !CONFIG_SND_DEBUG */
-- 
cgit v1.2.3-18-g5258


From a7d26f84cb76b962217ce277c3e3a9748a26ac11 Mon Sep 17 00:00:00 2001
From: Jaroslav Kysela <perex@perex.cz>
Date: Wed, 12 Nov 2008 11:02:25 +0100
Subject: ALSA: Release v1.0.18a

Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 include/sound/version.h | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

(limited to 'include/sound')

diff --git a/include/sound/version.h b/include/sound/version.h
index eae31557356..2b48237e23b 100644
--- a/include/sound/version.h
+++ b/include/sound/version.h
@@ -1,3 +1,3 @@
 /* include/version.h */
-#define CONFIG_SND_VERSION "1.0.18"
+#define CONFIG_SND_VERSION "1.0.18a"
 #define CONFIG_SND_DATE ""
-- 
cgit v1.2.3-18-g5258


From c2eb9c4ea383aee154e7139395872c4da629e715 Mon Sep 17 00:00:00 2001
From: Jaroslav Kysela <perex@perex.cz>
Date: Wed, 12 Nov 2008 16:31:37 +0100
Subject: ALSA: when card identification is changed, change also /proc/asound
 symlink

Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 include/sound/info.h | 2 ++
 1 file changed, 2 insertions(+)

(limited to 'include/sound')

diff --git a/include/sound/info.h b/include/sound/info.h
index 8ae72e74f89..46f702a76e4 100644
--- a/include/sound/info.h
+++ b/include/sound/info.h
@@ -126,6 +126,7 @@ int snd_info_card_create(struct snd_card * card);
 int snd_info_card_register(struct snd_card * card);
 int snd_info_card_free(struct snd_card * card);
 void snd_info_card_disconnect(struct snd_card * card);
+void snd_info_card_id_change(struct snd_card * card);
 int snd_info_register(struct snd_info_entry * entry);
 
 /* for card drivers */
@@ -160,6 +161,7 @@ static inline int snd_info_card_create(struct snd_card * card) { return 0; }
 static inline int snd_info_card_register(struct snd_card * card) { return 0; }
 static inline int snd_info_card_free(struct snd_card * card) { return 0; }
 static inline void snd_info_card_disconnect(struct snd_card * card) { }
+static inline void snd_info_card_id_change(struct snd_card * card) { }
 static inline int snd_info_register(struct snd_info_entry * entry) { return 0; }
 
 static inline int snd_card_proc_new(struct snd_card *card, const char *name,
-- 
cgit v1.2.3-18-g5258


From fd64138c0eff8351b93ef99f7da929bb8a49b9ed Mon Sep 17 00:00:00 2001
From: Jaroslav Kysela <perex@perex.cz>
Date: Wed, 12 Nov 2008 16:53:47 +0100
Subject: ALSA: include/sound/info.h - coding style changed

Change coding style to be more acceptable by checkpatch.pl.

Signed-off-by: Jaroslav Kysela <perex@perex.cz>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 include/sound/info.h | 108 +++++++++++++++++++++++++++------------------------
 1 file changed, 57 insertions(+), 51 deletions(-)

(limited to 'include/sound')

diff --git a/include/sound/info.h b/include/sound/info.h
index 46f702a76e4..7c2ee1a21b0 100644
--- a/include/sound/info.h
+++ b/include/sound/info.h
@@ -40,30 +40,34 @@ struct snd_info_buffer {
 struct snd_info_entry;
 
 struct snd_info_entry_text {
-	void (*read) (struct snd_info_entry *entry, struct snd_info_buffer *buffer);
-	void (*write) (struct snd_info_entry *entry, struct snd_info_buffer *buffer);
+	void (*read)(struct snd_info_entry *entry,
+		     struct snd_info_buffer *buffer);
+	void (*write)(struct snd_info_entry *entry,
+		      struct snd_info_buffer *buffer);
 };
 
 struct snd_info_entry_ops {
-	int (*open) (struct snd_info_entry *entry,
-		     unsigned short mode, void **file_private_data);
-	int (*release) (struct snd_info_entry * entry,
-			unsigned short mode, void *file_private_data);
-	long (*read) (struct snd_info_entry *entry, void *file_private_data,
-		      struct file * file, char __user *buf,
+	int (*open)(struct snd_info_entry *entry,
+		    unsigned short mode, void **file_private_data);
+	int (*release)(struct snd_info_entry *entry,
+		       unsigned short mode, void *file_private_data);
+	long (*read)(struct snd_info_entry *entry, void *file_private_data,
+		     struct file *file, char __user *buf,
+		     unsigned long count, unsigned long pos);
+	long (*write)(struct snd_info_entry *entry, void *file_private_data,
+		      struct file *file, const char __user *buf,
 		      unsigned long count, unsigned long pos);
-	long (*write) (struct snd_info_entry *entry, void *file_private_data,
-		       struct file * file, const char __user *buf,
-		       unsigned long count, unsigned long pos);
-	long long (*llseek) (struct snd_info_entry *entry, void *file_private_data,
-			    struct file * file, long long offset, int orig);
-	unsigned int (*poll) (struct snd_info_entry *entry, void *file_private_data,
-			      struct file * file, poll_table * wait);
-	int (*ioctl) (struct snd_info_entry *entry, void *file_private_data,
-		      struct file * file, unsigned int cmd, unsigned long arg);
-	int (*mmap) (struct snd_info_entry *entry, void *file_private_data,
-		     struct inode * inode, struct file * file,
-		     struct vm_area_struct * vma);
+	long long (*llseek)(struct snd_info_entry *entry,
+			    void *file_private_data, struct file *file,
+			    long long offset, int orig);
+	unsigned int(*poll)(struct snd_info_entry *entry,
+			    void *file_private_data, struct file *file,
+			    poll_table *wait);
+	int (*ioctl)(struct snd_info_entry *entry, void *file_private_data,
+		     struct file *file, unsigned int cmd, unsigned long arg);
+	int (*mmap)(struct snd_info_entry *entry, void *file_private_data,
+		    struct inode *inode, struct file *file,
+		    struct vm_area_struct *vma);
 };
 
 struct snd_info_entry {
@@ -106,35 +110,37 @@ void snd_card_info_read_oss(struct snd_info_buffer *buffer);
 static inline void snd_card_info_read_oss(struct snd_info_buffer *buffer) {}
 #endif
 
-int snd_iprintf(struct snd_info_buffer * buffer, char *fmt,...) __attribute__ ((format (printf, 2, 3)));
+int snd_iprintf(struct snd_info_buffer *buffer, char *fmt, ...) \
+				__attribute__ ((format (printf, 2, 3)));
 int snd_info_init(void);
 int snd_info_done(void);
 
-int snd_info_get_line(struct snd_info_buffer * buffer, char *line, int len);
+int snd_info_get_line(struct snd_info_buffer *buffer, char *line, int len);
 char *snd_info_get_str(char *dest, char *src, int len);
-struct snd_info_entry *snd_info_create_module_entry(struct module * module,
+struct snd_info_entry *snd_info_create_module_entry(struct module *module,
 					       const char *name,
-					       struct snd_info_entry * parent);
-struct snd_info_entry *snd_info_create_card_entry(struct snd_card * card,
+					       struct snd_info_entry *parent);
+struct snd_info_entry *snd_info_create_card_entry(struct snd_card *card,
 					     const char *name,
-					     struct snd_info_entry * parent);
-void snd_info_free_entry(struct snd_info_entry * entry);
-int snd_info_store_text(struct snd_info_entry * entry);
-int snd_info_restore_text(struct snd_info_entry * entry);
-
-int snd_info_card_create(struct snd_card * card);
-int snd_info_card_register(struct snd_card * card);
-int snd_info_card_free(struct snd_card * card);
-void snd_info_card_disconnect(struct snd_card * card);
-void snd_info_card_id_change(struct snd_card * card);
-int snd_info_register(struct snd_info_entry * entry);
+					     struct snd_info_entry *parent);
+void snd_info_free_entry(struct snd_info_entry *entry);
+int snd_info_store_text(struct snd_info_entry *entry);
+int snd_info_restore_text(struct snd_info_entry *entry);
+
+int snd_info_card_create(struct snd_card *card);
+int snd_info_card_register(struct snd_card *card);
+int snd_info_card_free(struct snd_card *card);
+void snd_info_card_disconnect(struct snd_card *card);
+void snd_info_card_id_change(struct snd_card *card);
+int snd_info_register(struct snd_info_entry *entry);
 
 /* for card drivers */
-int snd_card_proc_new(struct snd_card *card, const char *name, struct snd_info_entry **entryp);
+int snd_card_proc_new(struct snd_card *card, const char *name,
+		      struct snd_info_entry **entryp);
 
 static inline void snd_info_set_text_ops(struct snd_info_entry *entry, 
-					 void *private_data,
-					 void (*read)(struct snd_info_entry *, struct snd_info_buffer *))
+	void *private_data,
+	void (*read)(struct snd_info_entry *, struct snd_info_buffer *))
 {
 	entry->private_data = private_data;
 	entry->c.text.read = read;
@@ -147,22 +153,22 @@ int snd_info_check_reserved_words(const char *str);
 #define snd_seq_root NULL
 #define snd_oss_root NULL
 
-static inline int snd_iprintf(struct snd_info_buffer * buffer, char *fmt,...) { return 0; }
+static inline int snd_iprintf(struct snd_info_buffer *buffer, char *fmt, ...) { return 0; }
 static inline int snd_info_init(void) { return 0; }
 static inline int snd_info_done(void) { return 0; }
 
-static inline int snd_info_get_line(struct snd_info_buffer * buffer, char *line, int len) { return 0; }
+static inline int snd_info_get_line(struct snd_info_buffer *buffer, char *line, int len) { return 0; }
 static inline char *snd_info_get_str(char *dest, char *src, int len) { return NULL; }
-static inline struct snd_info_entry *snd_info_create_module_entry(struct module * module, const char *name, struct snd_info_entry * parent) { return NULL; }
-static inline struct snd_info_entry *snd_info_create_card_entry(struct snd_card * card, const char *name, struct snd_info_entry * parent) { return NULL; }
-static inline void snd_info_free_entry(struct snd_info_entry * entry) { ; }
-
-static inline int snd_info_card_create(struct snd_card * card) { return 0; }
-static inline int snd_info_card_register(struct snd_card * card) { return 0; }
-static inline int snd_info_card_free(struct snd_card * card) { return 0; }
-static inline void snd_info_card_disconnect(struct snd_card * card) { }
-static inline void snd_info_card_id_change(struct snd_card * card) { }
-static inline int snd_info_register(struct snd_info_entry * entry) { return 0; }
+static inline struct snd_info_entry *snd_info_create_module_entry(struct module *module, const char *name, struct snd_info_entry *parent) { return NULL; }
+static inline struct snd_info_entry *snd_info_create_card_entry(struct snd_card *card, const char *name, struct snd_info_entry *parent) { return NULL; }
+static inline void snd_info_free_entry(struct snd_info_entry *entry) { ; }
+
+static inline int snd_info_card_create(struct snd_card *card) { return 0; }
+static inline int snd_info_card_register(struct snd_card *card) { return 0; }
+static inline int snd_info_card_free(struct snd_card *card) { return 0; }
+static inline void snd_info_card_disconnect(struct snd_card *card) { }
+static inline void snd_info_card_id_change(struct snd_card *card) { }
+static inline int snd_info_register(struct snd_info_entry *entry) { return 0; }
 
 static inline int snd_card_proc_new(struct snd_card *card, const char *name,
 				    struct snd_info_entry **entryp) { return -EINVAL; }
-- 
cgit v1.2.3-18-g5258


From 1cad1de1b216b355a60d907c103b2daf1a285345 Mon Sep 17 00:00:00 2001
From: Christian Pellegrin <chripell@gmail.com>
Date: Sat, 15 Nov 2008 08:58:16 +0100
Subject: ASoC: UDA134x codec driver

Signed-off-by: Christian Pellegrin <chripell@fsfe.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 include/sound/l3.h      | 18 ++++++++++++++++++
 include/sound/uda134x.h | 26 ++++++++++++++++++++++++++
 2 files changed, 44 insertions(+)
 create mode 100644 include/sound/l3.h
 create mode 100644 include/sound/uda134x.h

(limited to 'include/sound')

diff --git a/include/sound/l3.h b/include/sound/l3.h
new file mode 100644
index 00000000000..423a08f0f1b
--- /dev/null
+++ b/include/sound/l3.h
@@ -0,0 +1,18 @@
+#ifndef _L3_H_
+#define _L3_H_ 1
+
+struct l3_pins {
+	void (*setdat)(int);
+	void (*setclk)(int);
+	void (*setmode)(int);
+	int data_hold;
+	int data_setup;
+	int clock_high;
+	int mode_hold;
+	int mode;
+	int mode_setup;
+};
+
+int l3_write(struct l3_pins *adap, u8 addr, u8 *data, int len);
+
+#endif
diff --git a/include/sound/uda134x.h b/include/sound/uda134x.h
new file mode 100644
index 00000000000..475ef8bb7dc
--- /dev/null
+++ b/include/sound/uda134x.h
@@ -0,0 +1,26 @@
+/*
+ * uda134x.h  --  UDA134x ALSA SoC Codec driver
+ *
+ * Copyright 2007 Dension Audio Systems Ltd.
+ * Author: Zoltan Devai
+ *
+ * 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.
+ */
+
+#ifndef _UDA134X_H
+#define _UDA134X_H
+
+#include <sound/l3.h>
+
+struct uda134x_platform_data {
+	struct l3_pins l3;
+	void (*power) (int);
+	int model;
+#define UDA134X_UDA1340 1
+#define UDA134X_UDA1341 2
+#define UDA134X_UDA1344 3
+};
+
+#endif /* _UDA134X_H */
-- 
cgit v1.2.3-18-g5258


From 7ad933d7a6677c20ce1bdb17425e732cf1ebee8a Mon Sep 17 00:00:00 2001
From: Christian Pellegrin <chripell@gmail.com>
Date: Sat, 15 Nov 2008 08:58:32 +0100
Subject: ASoC: Machine driver for for s3c24xx with uda134x

Signed-off-by: Christian Pellegrin <chripell@fsfe.org>
Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 include/sound/s3c24xx_uda134x.h | 14 ++++++++++++++
 1 file changed, 14 insertions(+)
 create mode 100644 include/sound/s3c24xx_uda134x.h

(limited to 'include/sound')

diff --git a/include/sound/s3c24xx_uda134x.h b/include/sound/s3c24xx_uda134x.h
new file mode 100644
index 00000000000..33df4cb909d
--- /dev/null
+++ b/include/sound/s3c24xx_uda134x.h
@@ -0,0 +1,14 @@
+#ifndef _S3C24XX_UDA134X_H_
+#define _S3C24XX_UDA134X_H_ 1
+
+#include <sound/uda134x.h>
+
+struct s3c24xx_uda134x_platform_data {
+	int l3_clk;
+	int l3_mode;
+	int l3_data;
+	void (*power) (int);
+	int model;
+};
+
+#endif
-- 
cgit v1.2.3-18-g5258


From ca3ea02e90d63a6a91c1c2a445d6d71f9031a44a Mon Sep 17 00:00:00 2001
From: Mark Brown <broonie@opensource.wolfsonmicro.com>
Date: Tue, 18 Nov 2008 20:40:36 +0000
Subject: ASoC: Remove unused snd_soc_machine_config declaration

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 include/sound/soc.h | 1 -
 1 file changed, 1 deletion(-)

(limited to 'include/sound')

diff --git a/include/sound/soc.h b/include/sound/soc.h
index fa1b99b4589..077dfe4e51f 100644
--- a/include/sound/soc.h
+++ b/include/sound/soc.h
@@ -228,7 +228,6 @@ struct snd_soc_dai_mode;
 struct snd_soc_pcm_runtime;
 struct snd_soc_dai;
 struct snd_soc_codec;
-struct snd_soc_machine_config;
 struct soc_enum;
 struct snd_soc_ac97_ops;
 struct snd_soc_clock_info;
-- 
cgit v1.2.3-18-g5258


From 875065491fba8eb13219f16c36e79a6fb4e15c68 Mon Sep 17 00:00:00 2001
From: Mark Brown <broonie@opensource.wolfsonmicro.com>
Date: Tue, 18 Nov 2008 20:50:34 +0000
Subject: ASoC: Rename snd_soc_card to snd_soc_machine

One of the issues with the ASoC v1 API which has been addressed in the
ASoC v2 work that Liam Girdwood has done is that the ALSA card provided
by ASoC is distributed around the ASoC structures. For example, machine
wide data such as the struct snd_card are maintained as part of the
CODEC data structure, preventing the use of multiple codecs. This has
been addressed by refactoring the data structures so that all the data
for the ALSA card is contained in a single structure snd_soc_card which
replaces the existing snd_soc_machine and snd_soc_device.

Begin the process of backporting this by renaming struct snd_soc_machine
to struct snd_soc_card, better reflecting its function and bringing it
closer to standard ALSA terminology.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 include/sound/soc.h | 8 ++++----
 1 file changed, 4 insertions(+), 4 deletions(-)

(limited to 'include/sound')

diff --git a/include/sound/soc.h b/include/sound/soc.h
index 077dfe4e51f..3be17b3c650 100644
--- a/include/sound/soc.h
+++ b/include/sound/soc.h
@@ -482,8 +482,8 @@ struct snd_soc_dai_link  {
 	struct snd_pcm *pcm;
 };
 
-/* SoC machine */
-struct snd_soc_machine {
+/* SoC card */
+struct snd_soc_card {
 	char *name;
 
 	int (*probe)(struct platform_device *pdev);
@@ -497,7 +497,7 @@ struct snd_soc_machine {
 	int (*resume_post)(struct platform_device *pdev);
 
 	/* callbacks */
-	int (*set_bias_level)(struct snd_soc_machine *,
+	int (*set_bias_level)(struct snd_soc_card *,
 			      enum snd_soc_bias_level level);
 
 	/* CPU <--> Codec DAI links  */
@@ -508,7 +508,7 @@ struct snd_soc_machine {
 /* SoC Device - the audio subsystem */
 struct snd_soc_device {
 	struct device *dev;
-	struct snd_soc_machine *machine;
+	struct snd_soc_card *card;
 	struct snd_soc_platform *platform;
 	struct snd_soc_codec *codec;
 	struct snd_soc_codec_device *codec_dev;
-- 
cgit v1.2.3-18-g5258


From a47cbe7263236691ee0bbc392f7fd4ec0da1159f Mon Sep 17 00:00:00 2001
From: Mark Brown <broonie@opensource.wolfsonmicro.com>
Date: Wed, 23 Jul 2008 14:03:07 +0100
Subject: ASoC: Move DAI structure definitions into new soc-dai.h

ASoC v2 factors most of the contents of soc.h out into separate headers,
including soc-dai.h for the DAI. Factor the existing DAI API out into
this file in order to prepare for backporting of the ASoC v2 DAI API.
Also backport some of Liam's improvements to the documentation.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 include/sound/soc-dai.h | 209 ++++++++++++++++++++++++++++++++++++++++++++++++
 include/sound/soc.h     | 148 +---------------------------------
 2 files changed, 211 insertions(+), 146 deletions(-)
 create mode 100644 include/sound/soc-dai.h

(limited to 'include/sound')

diff --git a/include/sound/soc-dai.h b/include/sound/soc-dai.h
new file mode 100644
index 00000000000..08b8f7025c6
--- /dev/null
+++ b/include/sound/soc-dai.h
@@ -0,0 +1,209 @@
+/*
+ * linux/sound/soc-dai.h -- ALSA SoC Layer
+ *
+ * Copyright:	2005-2008 Wolfson Microelectronics. PLC.
+ *
+ * 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.
+ *
+ * Digital Audio Interface (DAI) API.
+ */
+
+#ifndef __LINUX_SND_SOC_DAI_H
+#define __LINUX_SND_SOC_DAI_H
+
+
+#include <linux/list.h>
+
+struct snd_pcm_substream;
+
+/*
+ * DAI hardware audio formats.
+ *
+ * Describes the physical PCM data formating and clocking. Add new formats
+ * to the end.
+ */
+#define SND_SOC_DAIFMT_I2S		0 /* I2S mode */
+#define SND_SOC_DAIFMT_RIGHT_J		1 /* Right Justified mode */
+#define SND_SOC_DAIFMT_LEFT_J		2 /* Left Justified mode */
+#define SND_SOC_DAIFMT_DSP_A		3 /* L data msb after FRM LRC */
+#define SND_SOC_DAIFMT_DSP_B		4 /* L data msb during FRM LRC */
+#define SND_SOC_DAIFMT_AC97		5 /* AC97 */
+
+/* left and right justified also known as MSB and LSB respectively */
+#define SND_SOC_DAIFMT_MSB		SND_SOC_DAIFMT_LEFT_J
+#define SND_SOC_DAIFMT_LSB		SND_SOC_DAIFMT_RIGHT_J
+
+/*
+ * DAI Clock gating.
+ *
+ * DAI bit clocks can be be gated (disabled) when not the DAI is not
+ * sending or receiving PCM data in a frame. This can be used to save power.
+ */
+#define SND_SOC_DAIFMT_CONT		(0 << 4) /* continuous clock */
+#define SND_SOC_DAIFMT_GATED		(1 << 4) /* clock is gated */
+
+/*
+ * DAI Left/Right Clocks.
+ *
+ * Specifies whether the DAI can support different samples for similtanious
+ * playback and capture. This usually requires a seperate physical frame
+ * clock for playback and capture.
+ */
+#define SND_SOC_DAIFMT_SYNC		(0 << 5) /* Tx FRM = Rx FRM */
+#define SND_SOC_DAIFMT_ASYNC		(1 << 5) /* Tx FRM ~ Rx FRM */
+
+/*
+ * TDM
+ *
+ * Time Division Multiplexing. Allows PCM data to be multplexed with other
+ * data on the DAI.
+ */
+#define SND_SOC_DAIFMT_TDM		(1 << 6)
+
+/*
+ * DAI hardware signal inversions.
+ *
+ * Specifies whether the DAI can also support inverted clocks for the specified
+ * format.
+ */
+#define SND_SOC_DAIFMT_NB_NF		(0 << 8) /* normal bit clock + frame */
+#define SND_SOC_DAIFMT_NB_IF		(1 << 8) /* normal bclk + inv frm */
+#define SND_SOC_DAIFMT_IB_NF		(2 << 8) /* invert bclk + nor frm */
+#define SND_SOC_DAIFMT_IB_IF		(3 << 8) /* invert bclk + frm */
+
+/*
+ * DAI hardware clock masters.
+ *
+ * This is wrt the codec, the inverse is true for the interface
+ * i.e. if the codec is clk and frm master then the interface is
+ * clk and frame slave.
+ */
+#define SND_SOC_DAIFMT_CBM_CFM		(0 << 12) /* codec clk & frm master */
+#define SND_SOC_DAIFMT_CBS_CFM		(1 << 12) /* codec clk slave & frm master */
+#define SND_SOC_DAIFMT_CBM_CFS		(2 << 12) /* codec clk master & frame slave */
+#define SND_SOC_DAIFMT_CBS_CFS		(3 << 12) /* codec clk & frm slave */
+
+#define SND_SOC_DAIFMT_FORMAT_MASK	0x000f
+#define SND_SOC_DAIFMT_CLOCK_MASK	0x00f0
+#define SND_SOC_DAIFMT_INV_MASK		0x0f00
+#define SND_SOC_DAIFMT_MASTER_MASK	0xf000
+
+/*
+ * Master Clock Directions
+ */
+#define SND_SOC_CLOCK_IN		0
+#define SND_SOC_CLOCK_OUT		1
+
+struct snd_soc_dai_ops;
+struct snd_soc_dai;
+struct snd_ac97_bus_ops;
+
+/* Digital Audio Interface clocking API.*/
+int snd_soc_dai_set_sysclk(struct snd_soc_dai *dai, int clk_id,
+	unsigned int freq, int dir);
+
+int snd_soc_dai_set_clkdiv(struct snd_soc_dai *dai,
+	int div_id, int div);
+
+int snd_soc_dai_set_pll(struct snd_soc_dai *dai,
+	int pll_id, unsigned int freq_in, unsigned int freq_out);
+
+/* Digital Audio interface formatting */
+int snd_soc_dai_set_fmt(struct snd_soc_dai *dai, unsigned int fmt);
+
+int snd_soc_dai_set_tdm_slot(struct snd_soc_dai *dai,
+	unsigned int mask, int slots);
+
+int snd_soc_dai_set_tristate(struct snd_soc_dai *dai, int tristate);
+
+/* Digital Audio Interface mute */
+int snd_soc_dai_digital_mute(struct snd_soc_dai *dai, int mute);
+
+/*
+ * Digital Audio Interface.
+ *
+ * Describes the Digital Audio Interface in terms of it's ALSA, DAI and AC97
+ * operations an capabilities. Codec and platfom drivers will register a this
+ * structure for every DAI they have.
+ *
+ * This structure covers the clocking, formating and ALSA operations for each
+ * interface a
+ */
+struct snd_soc_dai_ops {
+	/*
+	 * DAI clocking configuration, all optional.
+	 * Called by soc_card drivers, normally in their hw_params.
+	 */
+	int (*set_sysclk)(struct snd_soc_dai *dai,
+		int clk_id, unsigned int freq, int dir);
+	int (*set_pll)(struct snd_soc_dai *dai,
+		int pll_id, unsigned int freq_in, unsigned int freq_out);
+	int (*set_clkdiv)(struct snd_soc_dai *dai, int div_id, int div);
+
+	/*
+	 * DAI format configuration
+	 * Called by soc_card drivers, normally in their hw_params.
+	 */
+	int (*set_fmt)(struct snd_soc_dai *dai, unsigned int fmt);
+	int (*set_tdm_slot)(struct snd_soc_dai *dai,
+		unsigned int mask, int slots);
+	int (*set_tristate)(struct snd_soc_dai *dai, int tristate);
+
+	/*
+	 * DAI digital mute - optional.
+	 * Called by soc-core to minimise any pops.
+	 */
+	int (*digital_mute)(struct snd_soc_dai *dai, int mute);
+};
+
+/*
+ * Digital Audio Interface runtime data.
+ *
+ * Holds runtime data for a DAI.
+ */
+struct snd_soc_dai {
+	/* DAI description */
+	char *name;
+	unsigned int id;
+	unsigned char type;
+
+	/* DAI callbacks */
+	int (*probe)(struct platform_device *pdev,
+		     struct snd_soc_dai *dai);
+	void (*remove)(struct platform_device *pdev,
+		       struct snd_soc_dai *dai);
+	int (*suspend)(struct platform_device *pdev,
+		struct snd_soc_dai *dai);
+	int (*resume)(struct platform_device *pdev,
+		struct snd_soc_dai *dai);
+
+	/* ops */
+	struct snd_soc_ops ops;
+	struct snd_soc_dai_ops dai_ops;
+
+	/* DAI capabilities */
+	struct snd_soc_pcm_stream capture;
+	struct snd_soc_pcm_stream playback;
+
+	/* DAI runtime info */
+	struct snd_pcm_runtime *runtime;
+	struct snd_soc_codec *codec;
+	unsigned int active;
+	unsigned char pop_wait:1;
+	void *dma_data;
+
+	/* DAI private data */
+	void *private_data;
+
+	/* parent codec/platform */
+	union {
+		struct snd_soc_codec *codec;
+		struct snd_soc_platform *platform;
+	};
+
+	struct list_head list;
+};
+
+#endif
diff --git a/include/sound/soc.h b/include/sound/soc.h
index 3be17b3c650..e4465f73aa4 100644
--- a/include/sound/soc.h
+++ b/include/sound/soc.h
@@ -151,76 +151,6 @@ enum snd_soc_bias_level {
 #define SND_SOC_DAI_PCM		0x4
 #define SND_SOC_DAI_AC97_BUS	0x8	/* for custom i.e. non ac97_codec.c */
 
-/*
- * DAI hardware audio formats
- */
-#define SND_SOC_DAIFMT_I2S		0	/* I2S mode */
-#define SND_SOC_DAIFMT_RIGHT_J	1	/* Right justified mode */
-#define SND_SOC_DAIFMT_LEFT_J	2	/* Left Justified mode */
-#define SND_SOC_DAIFMT_DSP_A	3	/* L data msb after FRM or LRC */
-#define SND_SOC_DAIFMT_DSP_B	4	/* L data msb during FRM or LRC */
-#define SND_SOC_DAIFMT_AC97		5	/* AC97 */
-
-#define SND_SOC_DAIFMT_MSB 	SND_SOC_DAIFMT_LEFT_J
-#define SND_SOC_DAIFMT_LSB	SND_SOC_DAIFMT_RIGHT_J
-
-/*
- * DAI Gating
- */
-#define SND_SOC_DAIFMT_CONT			(0 << 4)	/* continuous clock */
-#define SND_SOC_DAIFMT_GATED		(1 << 4)	/* clock is gated when not Tx/Rx */
-
-/*
- * DAI Sync
- * Synchronous LR (Left Right) clocks and Frame signals.
- */
-#define SND_SOC_DAIFMT_SYNC		(0 << 5)	/* Tx FRM = Rx FRM */
-#define SND_SOC_DAIFMT_ASYNC		(1 << 5)	/* Tx FRM ~ Rx FRM */
-
-/*
- * TDM
- */
-#define SND_SOC_DAIFMT_TDM		(1 << 6)
-
-/*
- * DAI hardware signal inversions
- */
-#define SND_SOC_DAIFMT_NB_NF		(0 << 8)	/* normal bclk + frm */
-#define SND_SOC_DAIFMT_NB_IF		(1 << 8)	/* normal bclk + inv frm */
-#define SND_SOC_DAIFMT_IB_NF		(2 << 8)	/* invert bclk + nor frm */
-#define SND_SOC_DAIFMT_IB_IF		(3 << 8)	/* invert bclk + frm */
-
-/*
- * DAI hardware clock masters
- * This is wrt the codec, the inverse is true for the interface
- * i.e. if the codec is clk and frm master then the interface is
- * clk and frame slave.
- */
-#define SND_SOC_DAIFMT_CBM_CFM	(0 << 12) /* codec clk & frm master */
-#define SND_SOC_DAIFMT_CBS_CFM	(1 << 12) /* codec clk slave & frm master */
-#define SND_SOC_DAIFMT_CBM_CFS	(2 << 12) /* codec clk master & frame slave */
-#define SND_SOC_DAIFMT_CBS_CFS	(3 << 12) /* codec clk & frm slave */
-
-#define SND_SOC_DAIFMT_FORMAT_MASK		0x000f
-#define SND_SOC_DAIFMT_CLOCK_MASK		0x00f0
-#define SND_SOC_DAIFMT_INV_MASK			0x0f00
-#define SND_SOC_DAIFMT_MASTER_MASK		0xf000
-
-
-/*
- * Master Clock Directions
- */
-#define SND_SOC_CLOCK_IN	0
-#define SND_SOC_CLOCK_OUT	1
-
-/*
- * AC97 codec ID's bitmask
- */
-#define SND_SOC_DAI_AC97_ID0	(1 << 0)
-#define SND_SOC_DAI_AC97_ID1	(1 << 1)
-#define SND_SOC_DAI_AC97_ID2	(1 << 2)
-#define SND_SOC_DAI_AC97_ID3	(1 << 3)
-
 struct snd_soc_device;
 struct snd_soc_pcm_stream;
 struct snd_soc_ops;
@@ -260,27 +190,6 @@ int snd_soc_new_ac97_codec(struct snd_soc_codec *codec,
 	struct snd_ac97_bus_ops *ops, int num);
 void snd_soc_free_ac97_codec(struct snd_soc_codec *codec);
 
-/* Digital Audio Interface clocking API.*/
-int snd_soc_dai_set_sysclk(struct snd_soc_dai *dai, int clk_id,
-	unsigned int freq, int dir);
-
-int snd_soc_dai_set_clkdiv(struct snd_soc_dai *dai,
-	int div_id, int div);
-
-int snd_soc_dai_set_pll(struct snd_soc_dai *dai,
-	int pll_id, unsigned int freq_in, unsigned int freq_out);
-
-/* Digital Audio interface formatting */
-int snd_soc_dai_set_fmt(struct snd_soc_dai *dai, unsigned int fmt);
-
-int snd_soc_dai_set_tdm_slot(struct snd_soc_dai *dai,
-	unsigned int mask, int slots);
-
-int snd_soc_dai_set_tristate(struct snd_soc_dai *dai, int tristate);
-
-/* Digital Audio Interface mute */
-int snd_soc_dai_digital_mute(struct snd_soc_dai *dai, int mute);
-
 /*
  *Controls
  */
@@ -338,61 +247,6 @@ struct snd_soc_ops {
 	int (*trigger)(struct snd_pcm_substream *, int);
 };
 
-/* ASoC DAI ops */
-struct snd_soc_dai_ops {
-	/* DAI clocking configuration */
-	int (*set_sysclk)(struct snd_soc_dai *dai,
-		int clk_id, unsigned int freq, int dir);
-	int (*set_pll)(struct snd_soc_dai *dai,
-		int pll_id, unsigned int freq_in, unsigned int freq_out);
-	int (*set_clkdiv)(struct snd_soc_dai *dai, int div_id, int div);
-
-	/* DAI format configuration */
-	int (*set_fmt)(struct snd_soc_dai *dai, unsigned int fmt);
-	int (*set_tdm_slot)(struct snd_soc_dai *dai,
-		unsigned int mask, int slots);
-	int (*set_tristate)(struct snd_soc_dai *dai, int tristate);
-
-	/* digital mute */
-	int (*digital_mute)(struct snd_soc_dai *dai, int mute);
-};
-
-/* SoC  DAI (Digital Audio Interface) */
-struct snd_soc_dai {
-	/* DAI description */
-	char *name;
-	unsigned int id;
-	unsigned char type;
-
-	/* DAI callbacks */
-	int (*probe)(struct platform_device *pdev,
-		     struct snd_soc_dai *dai);
-	void (*remove)(struct platform_device *pdev,
-		       struct snd_soc_dai *dai);
-	int (*suspend)(struct platform_device *pdev,
-		struct snd_soc_dai *dai);
-	int (*resume)(struct platform_device *pdev,
-		struct snd_soc_dai *dai);
-
-	/* ops */
-	struct snd_soc_ops ops;
-	struct snd_soc_dai_ops dai_ops;
-
-	/* DAI capabilities */
-	struct snd_soc_pcm_stream capture;
-	struct snd_soc_pcm_stream playback;
-
-	/* DAI runtime info */
-	struct snd_pcm_runtime *runtime;
-	struct snd_soc_codec *codec;
-	unsigned int active;
-	unsigned char pop_wait:1;
-	void *dma_data;
-
-	/* DAI private data */
-	void *private_data;
-};
-
 /* SoC Audio Codec */
 struct snd_soc_codec {
 	char *name;
@@ -543,4 +397,6 @@ struct soc_enum {
 	void *dapm;
 };
 
+#include <sound/soc-dai.h>
+
 #endif
-- 
cgit v1.2.3-18-g5258


From dee89c4d94433520e4e3977ae203d4cfbfe385fb Mon Sep 17 00:00:00 2001
From: Mark Brown <broonie@opensource.wolfsonmicro.com>
Date: Tue, 18 Nov 2008 22:11:38 +0000
Subject: ASoC: Merge snd_soc_ops into snd_soc_dai_ops

Liam Girdwood's ASoC v2 work avoids having two different ops structures
for DAIs by merging the members of struct snd_soc_ops into struct
snd_soc_dai_ops, allowing per DAI configuration for everything.
Backport this change.

This paves the way for future work allowing any combination of DAIs to
be connected rather than having fixed purpose CODEC and CPU DAIs and
only allowing CODEC<->CPU interconnections.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 include/sound/soc-dai.h | 20 ++++++++++++++++++--
 1 file changed, 18 insertions(+), 2 deletions(-)

(limited to 'include/sound')

diff --git a/include/sound/soc-dai.h b/include/sound/soc-dai.h
index 08b8f7025c6..f51cb55902f 100644
--- a/include/sound/soc-dai.h
+++ b/include/sound/soc-dai.h
@@ -156,6 +156,23 @@ struct snd_soc_dai_ops {
 	 * Called by soc-core to minimise any pops.
 	 */
 	int (*digital_mute)(struct snd_soc_dai *dai, int mute);
+
+	/*
+	 * ALSA PCM audio operations - all optional.
+	 * Called by soc-core during audio PCM operations.
+	 */
+	int (*startup)(struct snd_pcm_substream *,
+		struct snd_soc_dai *);
+	void (*shutdown)(struct snd_pcm_substream *,
+		struct snd_soc_dai *);
+	int (*hw_params)(struct snd_pcm_substream *,
+		struct snd_pcm_hw_params *, struct snd_soc_dai *);
+	int (*hw_free)(struct snd_pcm_substream *,
+		struct snd_soc_dai *);
+	int (*prepare)(struct snd_pcm_substream *,
+		struct snd_soc_dai *);
+	int (*trigger)(struct snd_pcm_substream *, int,
+		struct snd_soc_dai *);
 };
 
 /*
@@ -180,8 +197,7 @@ struct snd_soc_dai {
 		struct snd_soc_dai *dai);
 
 	/* ops */
-	struct snd_soc_ops ops;
-	struct snd_soc_dai_ops dai_ops;
+	struct snd_soc_dai_ops ops;
 
 	/* DAI capabilities */
 	struct snd_soc_pcm_stream capture;
-- 
cgit v1.2.3-18-g5258


From 3ba9e10a6d3b6abf5f5952572cff8f8d5a35ae54 Mon Sep 17 00:00:00 2001
From: Mark Brown <broonie@opensource.wolfsonmicro.com>
Date: Mon, 24 Nov 2008 18:01:05 +0000
Subject: ASoC: Remove DAI type information

DAI type information is only ever used within ASoC in order to special
case AC97 and for diagnostic purposes. Since modern CPUs and codecs
support multi function DAIs which can be configured for several modes
it is more trouble than it's worth to maintain anything other than a
flag identifying AC97 DAIs so remove the type field and replace it with
an ac97_control flag.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 include/sound/soc-dai.h | 2 +-
 include/sound/soc.h     | 8 --------
 2 files changed, 1 insertion(+), 9 deletions(-)

(limited to 'include/sound')

diff --git a/include/sound/soc-dai.h b/include/sound/soc-dai.h
index f51cb55902f..a01a24b1019 100644
--- a/include/sound/soc-dai.h
+++ b/include/sound/soc-dai.h
@@ -184,7 +184,7 @@ struct snd_soc_dai {
 	/* DAI description */
 	char *name;
 	unsigned int id;
-	unsigned char type;
+	int ac97_control;
 
 	/* DAI callbacks */
 	int (*probe)(struct platform_device *pdev,
diff --git a/include/sound/soc.h b/include/sound/soc.h
index e4465f73aa4..444f9c21137 100644
--- a/include/sound/soc.h
+++ b/include/sound/soc.h
@@ -143,14 +143,6 @@ enum snd_soc_bias_level {
 	SND_SOC_BIAS_OFF,
 };
 
-/*
- * Digital Audio Interface (DAI) types
- */
-#define SND_SOC_DAI_AC97	0x1
-#define SND_SOC_DAI_I2S		0x2
-#define SND_SOC_DAI_PCM		0x4
-#define SND_SOC_DAI_AC97_BUS	0x8	/* for custom i.e. non ac97_codec.c */
-
 struct snd_soc_device;
 struct snd_soc_pcm_stream;
 struct snd_soc_ops;
-- 
cgit v1.2.3-18-g5258


From 968a6025aa9f909d487988efb542217a126023a0 Mon Sep 17 00:00:00 2001
From: Mark Brown <broonie@opensource.wolfsonmicro.com>
Date: Fri, 28 Nov 2008 11:49:07 +0000
Subject: ASoC: Rename snd_soc_register_card() to snd_soc_init_card()

Currently ASoC card initialisation is completed by a function called
snd_soc_register_card().  As part of the work to allow independant
registration of cards, codecs and machines in ASoC v2 a new function of
the same name has been added so rename the existing function to
facilitate the merge of v2.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 include/sound/soc.h | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

(limited to 'include/sound')

diff --git a/include/sound/soc.h b/include/sound/soc.h
index 444f9c21137..9356c1ce98c 100644
--- a/include/sound/soc.h
+++ b/include/sound/soc.h
@@ -162,7 +162,7 @@ extern struct snd_ac97_bus_ops soc_ac97_ops;
 /* pcm <-> DAI connect */
 void snd_soc_free_pcms(struct snd_soc_device *socdev);
 int snd_soc_new_pcms(struct snd_soc_device *socdev, int idx, const char *xid);
-int snd_soc_register_card(struct snd_soc_device *socdev);
+int snd_soc_init_card(struct snd_soc_device *socdev);
 
 /* set runtime hw params */
 int snd_soc_set_runtime_hwparams(struct snd_pcm_substream *substream,
-- 
cgit v1.2.3-18-g5258


From 6308419a199eed66086cd756ab8dc81b88d54a6b Mon Sep 17 00:00:00 2001
From: Mark Brown <broonie@opensource.wolfsonmicro.com>
Date: Tue, 2 Dec 2008 15:08:03 +0000
Subject: ASoC: Push workqueue data into snd_soc_card

ASoC v2 does not use the struct snd_soc_device at runtime, using struct
snd_soc_card as the root of the card.  Begin removing data from
snd_soc_device by pushing the workqueue data into snd_soc_card, using a
backpointer to the snd_soc_device to keep things going for the time
being.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 include/sound/soc.h | 7 +++++--
 1 file changed, 5 insertions(+), 2 deletions(-)

(limited to 'include/sound')

diff --git a/include/sound/soc.h b/include/sound/soc.h
index 9356c1ce98c..359ec49f8d0 100644
--- a/include/sound/soc.h
+++ b/include/sound/soc.h
@@ -349,6 +349,11 @@ struct snd_soc_card {
 	/* CPU <--> Codec DAI links  */
 	struct snd_soc_dai_link *dai_link;
 	int num_links;
+
+	struct snd_soc_device *socdev;
+
+	struct delayed_work delayed_work;
+	struct work_struct deferred_resume_work;
 };
 
 /* SoC Device - the audio subsystem */
@@ -358,8 +363,6 @@ struct snd_soc_device {
 	struct snd_soc_platform *platform;
 	struct snd_soc_codec *codec;
 	struct snd_soc_codec_device *codec_dev;
-	struct delayed_work delayed_work;
-	struct work_struct deferred_resume_work;
 	void *codec_data;
 #ifdef CONFIG_DEBUG_FS
 	struct dentry	*debugfs_root;
-- 
cgit v1.2.3-18-g5258


From 87689d567a45f80416feea0a2aa6d3a2a6b8963a Mon Sep 17 00:00:00 2001
From: Mark Brown <broonie@opensource.wolfsonmicro.com>
Date: Tue, 2 Dec 2008 16:01:14 +0000
Subject: ASoC: Push platform registration down into the card

As part of the deprecation of snd_soc_device push the registration of
the platform down into the card structure.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 include/sound/soc.h | 2 +-
 1 file changed, 1 insertion(+), 1 deletion(-)

(limited to 'include/sound')

diff --git a/include/sound/soc.h b/include/sound/soc.h
index 359ec49f8d0..ad8141acd6b 100644
--- a/include/sound/soc.h
+++ b/include/sound/soc.h
@@ -352,6 +352,7 @@ struct snd_soc_card {
 
 	struct snd_soc_device *socdev;
 
+	struct snd_soc_platform *platform;
 	struct delayed_work delayed_work;
 	struct work_struct deferred_resume_work;
 };
@@ -360,7 +361,6 @@ struct snd_soc_card {
 struct snd_soc_device {
 	struct device *dev;
 	struct snd_soc_card *card;
-	struct snd_soc_platform *platform;
 	struct snd_soc_codec *codec;
 	struct snd_soc_codec_device *codec_dev;
 	void *codec_data;
-- 
cgit v1.2.3-18-g5258


From 384c89e2e4cb5879b86a38414d1b3bb2b23ec8ee Mon Sep 17 00:00:00 2001
From: Mark Brown <broonie@opensource.wolfsonmicro.com>
Date: Wed, 3 Dec 2008 17:34:03 +0000
Subject: ASoC: Push debugfs files out of the snd_soc_device structure

This is in preparation for the removal of struct snd_soc_device.

The pop time configuration should really be a property of the card not
the codec but since DAPM currently uses the codec rather than the card
using the codec is fine for now.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 include/sound/soc.h | 8 +++++---
 1 file changed, 5 insertions(+), 3 deletions(-)

(limited to 'include/sound')

diff --git a/include/sound/soc.h b/include/sound/soc.h
index ad8141acd6b..3ee608dce2f 100644
--- a/include/sound/soc.h
+++ b/include/sound/soc.h
@@ -279,6 +279,11 @@ struct snd_soc_codec {
 	/* codec DAI's */
 	struct snd_soc_dai *dai;
 	unsigned int num_dai;
+
+#ifdef CONFIG_DEBUG_FS
+	struct dentry *debugfs_reg;
+	struct dentry *debugfs_pop_time;
+#endif
 };
 
 /* codec device */
@@ -364,9 +369,6 @@ struct snd_soc_device {
 	struct snd_soc_codec *codec;
 	struct snd_soc_codec_device *codec_dev;
 	void *codec_data;
-#ifdef CONFIG_DEBUG_FS
-	struct dentry	*debugfs_root;
-#endif
 };
 
 /* runtime channel data */
-- 
cgit v1.2.3-18-g5258


From 07c84d0409f3551b79d676630d8ee76bb551598d Mon Sep 17 00:00:00 2001
From: Mark Brown <broonie@opensource.wolfsonmicro.com>
Date: Wed, 3 Dec 2008 18:17:28 +0000
Subject: ASoC: Remove device from platform suspend and resume operations

None of the platforms are actually using the SoC device so remove it
(only atmel actually has a suspend method).

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 include/sound/soc.h | 6 ++----
 1 file changed, 2 insertions(+), 4 deletions(-)

(limited to 'include/sound')

diff --git a/include/sound/soc.h b/include/sound/soc.h
index 3ee608dce2f..8ec63c02dc1 100644
--- a/include/sound/soc.h
+++ b/include/sound/soc.h
@@ -300,10 +300,8 @@ struct snd_soc_platform {
 
 	int (*probe)(struct platform_device *pdev);
 	int (*remove)(struct platform_device *pdev);
-	int (*suspend)(struct platform_device *pdev,
-		struct snd_soc_dai *dai);
-	int (*resume)(struct platform_device *pdev,
-		struct snd_soc_dai *dai);
+	int (*suspend)(struct snd_soc_dai *dai);
+	int (*resume)(struct snd_soc_dai *dai);
 
 	/* pcm creation and destruction */
 	int (*pcm_new)(struct snd_card *, struct snd_soc_dai *,
-- 
cgit v1.2.3-18-g5258


From dc7d7b830ee1f4111696e73d1c25da683b461548 Mon Sep 17 00:00:00 2001
From: Mark Brown <broonie@opensource.wolfsonmicro.com>
Date: Wed, 3 Dec 2008 18:21:52 +0000
Subject: ASoC: Remove platform device from DAI suspend and resume operations

None of the DAIs use it except s3c2412-i2s which only uses it for
dev_() printouts.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 include/sound/soc-dai.h | 6 ++----
 1 file changed, 2 insertions(+), 4 deletions(-)

(limited to 'include/sound')

diff --git a/include/sound/soc-dai.h b/include/sound/soc-dai.h
index a01a24b1019..e2d5f76838c 100644
--- a/include/sound/soc-dai.h
+++ b/include/sound/soc-dai.h
@@ -191,10 +191,8 @@ struct snd_soc_dai {
 		     struct snd_soc_dai *dai);
 	void (*remove)(struct platform_device *pdev,
 		       struct snd_soc_dai *dai);
-	int (*suspend)(struct platform_device *pdev,
-		struct snd_soc_dai *dai);
-	int (*resume)(struct platform_device *pdev,
-		struct snd_soc_dai *dai);
+	int (*suspend)(struct snd_soc_dai *dai);
+	int (*resume)(struct snd_soc_dai *dai);
 
 	/* ops */
 	struct snd_soc_dai_ops ops;
-- 
cgit v1.2.3-18-g5258


From 32c8dabc97d436582298ebd0e33af041c69f5a4b Mon Sep 17 00:00:00 2001
From: Mark Brown <broonie@opensource.wolfsonmicro.com>
Date: Wed, 3 Dec 2008 19:41:13 +0000
Subject: ASoC: Remove obsolete declaration of struct snd_soc_clock_info

The struct is never defined.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 include/sound/soc.h | 1 -
 1 file changed, 1 deletion(-)

(limited to 'include/sound')

diff --git a/include/sound/soc.h b/include/sound/soc.h
index 8ec63c02dc1..79d855d2bdd 100644
--- a/include/sound/soc.h
+++ b/include/sound/soc.h
@@ -152,7 +152,6 @@ struct snd_soc_dai;
 struct snd_soc_codec;
 struct soc_enum;
 struct snd_soc_ac97_ops;
-struct snd_soc_clock_info;
 
 typedef int (*hw_write_t)(void *,const char* ,int);
 typedef int (*hw_read_t)(void *,char* ,int);
-- 
cgit v1.2.3-18-g5258


From c5af3a2e192d333997d1e191f3eba7fd2f869681 Mon Sep 17 00:00:00 2001
From: Mark Brown <broonie@opensource.wolfsonmicro.com>
Date: Fri, 28 Nov 2008 13:29:45 +0000
Subject: ASoC: Add card registration API

ASoC v2 allows cards, codecs and platforms to instantiate separately,
with the overall ASoC device only being instantiated once all the
required components have registered. As part of backporting Liam's work
introduce an initial version of the card registration functions. At
present these do nothing active and are internal only, they will be
exposed to machine drivers after further backporting.  Adding this now
allows the datastructures used for dynamic card instantiation to be
built up gradually.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 include/sound/soc.h | 5 +++++
 1 file changed, 5 insertions(+)

(limited to 'include/sound')

diff --git a/include/sound/soc.h b/include/sound/soc.h
index 79d855d2bdd..4a578b5d855 100644
--- a/include/sound/soc.h
+++ b/include/sound/soc.h
@@ -333,6 +333,11 @@ struct snd_soc_dai_link  {
 /* SoC card */
 struct snd_soc_card {
 	char *name;
+	struct device *dev;
+
+	struct list_head list;
+
+	int instantiated;
 
 	int (*probe)(struct platform_device *pdev);
 	int (*remove)(struct platform_device *pdev);
-- 
cgit v1.2.3-18-g5258


From 9115171a6b79b6b4d5c6697f123556b6efc37f1f Mon Sep 17 00:00:00 2001
From: Mark Brown <broonie@opensource.wolfsonmicro.com>
Date: Sun, 30 Nov 2008 23:31:24 +0000
Subject: ASoC: Add DAI registration API

Add API calls to register and unregister DAIs with the core.  Currently
these APIs are ineffective.  Since multiple DAIs for a given device are
a common case bulk variants are provided.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 include/sound/soc-dai.h | 8 ++++++++
 1 file changed, 8 insertions(+)

(limited to 'include/sound')

diff --git a/include/sound/soc-dai.h b/include/sound/soc-dai.h
index e2d5f76838c..24247f76360 100644
--- a/include/sound/soc-dai.h
+++ b/include/sound/soc-dai.h
@@ -100,6 +100,12 @@ struct snd_soc_dai_ops;
 struct snd_soc_dai;
 struct snd_ac97_bus_ops;
 
+/* Digital Audio Interface registration */
+int snd_soc_register_dai(struct snd_soc_dai *dai);
+void snd_soc_unregister_dai(struct snd_soc_dai *dai);
+int snd_soc_register_dais(struct snd_soc_dai *dai, size_t count);
+void snd_soc_unregister_dais(struct snd_soc_dai *dai, size_t count);
+
 /* Digital Audio Interface clocking API.*/
 int snd_soc_dai_set_sysclk(struct snd_soc_dai *dai, int clk_id,
 	unsigned int freq, int dir);
@@ -186,6 +192,8 @@ struct snd_soc_dai {
 	unsigned int id;
 	int ac97_control;
 
+	struct device *dev;
+
 	/* DAI callbacks */
 	int (*probe)(struct platform_device *pdev,
 		     struct snd_soc_dai *dai);
-- 
cgit v1.2.3-18-g5258


From 12a48a8c0087ba39d926cf1d63938ccbdb9752c3 Mon Sep 17 00:00:00 2001
From: Mark Brown <broonie@opensource.wolfsonmicro.com>
Date: Wed, 3 Dec 2008 19:40:30 +0000
Subject: ASoC: Add platform registration API

ASoC v2 allows platform drivers to instantiate independantly of the
overall ASoC card. This API allows drivers to notify the core when
they are registered.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 include/sound/soc.h | 5 +++++
 1 file changed, 5 insertions(+)

(limited to 'include/sound')

diff --git a/include/sound/soc.h b/include/sound/soc.h
index 4a578b5d855..ce3661d07c2 100644
--- a/include/sound/soc.h
+++ b/include/sound/soc.h
@@ -149,6 +149,7 @@ struct snd_soc_ops;
 struct snd_soc_dai_mode;
 struct snd_soc_pcm_runtime;
 struct snd_soc_dai;
+struct snd_soc_platform;
 struct snd_soc_codec;
 struct soc_enum;
 struct snd_soc_ac97_ops;
@@ -158,6 +159,9 @@ typedef int (*hw_read_t)(void *,char* ,int);
 
 extern struct snd_ac97_bus_ops soc_ac97_ops;
 
+int snd_soc_register_platform(struct snd_soc_platform *platform);
+void snd_soc_unregister_platform(struct snd_soc_platform *platform);
+
 /* pcm <-> DAI connect */
 void snd_soc_free_pcms(struct snd_soc_device *socdev);
 int snd_soc_new_pcms(struct snd_soc_device *socdev, int idx, const char *xid);
@@ -296,6 +300,7 @@ struct snd_soc_codec_device {
 /* SoC platform interface */
 struct snd_soc_platform {
 	char *name;
+	struct list_head list;
 
 	int (*probe)(struct platform_device *pdev);
 	int (*remove)(struct platform_device *pdev);
-- 
cgit v1.2.3-18-g5258


From cdc693643271b2e6a693cf8f6afb258cce01f058 Mon Sep 17 00:00:00 2001
From: Mark Brown <broonie@opensource.wolfsonmicro.com>
Date: Wed, 10 Dec 2008 13:55:49 +0000
Subject: ALSA: Add support for mechanical jack insertion

Some systems support both mechanical and electrical jack detection,
allowing them to report that a jack is physically present but does
not have any functioning connections. Add a new jack type for these,
allowing user space to report faulty connections.

Thanks to Guillem Jover for the suggestion.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 include/sound/jack.h | 1 +
 1 file changed, 1 insertion(+)

(limited to 'include/sound')

diff --git a/include/sound/jack.h b/include/sound/jack.h
index 7cb25f4b50b..2e0315cdd0d 100644
--- a/include/sound/jack.h
+++ b/include/sound/jack.h
@@ -36,6 +36,7 @@ enum snd_jack_types {
 	SND_JACK_MICROPHONE	= 0x0002,
 	SND_JACK_HEADSET	= SND_JACK_HEADPHONE | SND_JACK_MICROPHONE,
 	SND_JACK_LINEOUT	= 0x0004,
+	SND_JACK_MECHANICAL	= 0x0008, /* If detected separately */
 };
 
 struct snd_jack {
-- 
cgit v1.2.3-18-g5258


From 0d0cf00a7fc63cee9a4c4a3b8612879b4f7f42ba Mon Sep 17 00:00:00 2001
From: Mark Brown <broonie@opensource.wolfsonmicro.com>
Date: Wed, 10 Dec 2008 14:32:45 +0000
Subject: ASoC: Add codec registration API

Another part of the backporting of Liam's ASoC v2 work. Using this is
more complicated than the other registration types since currently the
codec is instantiated during the probe of the ASoC device so we can't
currently readily wait for the codec to register.

Signed-off-by: Mark Brown <broonie@opensource.wolfsonmicro.com>
---
 include/sound/soc.h | 5 +++++
 1 file changed, 5 insertions(+)

(limited to 'include/sound')

diff --git a/include/sound/soc.h b/include/sound/soc.h
index ce3661d07c2..f86e455d382 100644
--- a/include/sound/soc.h
+++ b/include/sound/soc.h
@@ -161,6 +161,8 @@ extern struct snd_ac97_bus_ops soc_ac97_ops;
 
 int snd_soc_register_platform(struct snd_soc_platform *platform);
 void snd_soc_unregister_platform(struct snd_soc_platform *platform);
+int snd_soc_register_codec(struct snd_soc_codec *codec);
+void snd_soc_unregister_codec(struct snd_soc_codec *codec);
 
 /* pcm <-> DAI connect */
 void snd_soc_free_pcms(struct snd_soc_device *socdev);
@@ -247,6 +249,9 @@ struct snd_soc_codec {
 	char *name;
 	struct module *owner;
 	struct mutex mutex;
+	struct device *dev;
+
+	struct list_head list;
 
 	/* callbacks */
 	int (*set_bias_level)(struct snd_soc_codec *,
-- 
cgit v1.2.3-18-g5258


From 0bed7b292d68f82316bfb8cd521e16c867689efe Mon Sep 17 00:00:00 2001
From: Andres Salomon <dilinger@queued.net>
Date: Wed, 5 Nov 2008 17:29:53 -0500
Subject: ALSA: cs5535audio: stick AD1888 bitshift values into a header file

We'd like to use the High Pass Filter and V_REFOUT bitshift values elsewhere,
so stick them into a ac97_codec.h.

Signed-off-by: Andres Salomon <dilinger@debian.org>
Signed-off-by: Takashi Iwai <tiwai@suse.de>
---
 include/sound/ac97_codec.h | 2 ++
 1 file changed, 2 insertions(+)

(limited to 'include/sound')

diff --git a/include/sound/ac97_codec.h b/include/sound/ac97_codec.h
index 9c309daf492..251fc1cd500 100644
--- a/include/sound/ac97_codec.h
+++ b/include/sound/ac97_codec.h
@@ -281,10 +281,12 @@
 /* specific - Analog Devices */
 #define AC97_AD_TEST		0x5a	/* test register */
 #define AC97_AD_TEST2		0x5c	/* undocumented test register 2 */
+#define AC97_AD_HPFD_SHIFT	12	/* High Pass Filter Disable */
 #define AC97_AD_CODEC_CFG	0x70	/* codec configuration */
 #define AC97_AD_JACK_SPDIF	0x72	/* Jack Sense & S/PDIF */
 #define AC97_AD_SERIAL_CFG	0x74	/* Serial Configuration */
 #define AC97_AD_MISC		0x76	/* Misc Control Bits */
+#define AC97_AD_VREFD_SHIFT	2	/* V_REFOUT Disable (AD1888) */
 
 /* specific - Cirrus Logic */
 #define AC97_CSR_ACMODE		0x5e	/* AC Mode Register */
-- 
cgit v1.2.3-18-g5258