Merge branch 'topic/asoc' into to-push

7 files changed, 359 insertions, 176 deletions

diff --git a/include/linux/mfd/wm8350/audio.h b/include/linux/mfd/wm8350/audio.h
index 217bb22ebb8e..af95a1d2f3a1 100644
--- a/include/linux/mfd/wm8350/audio.h
+++ b/include/linux/mfd/wm8350/audio.h
@@ -1,7 +1,7 @@
 /*
  * audio.h  --  Audio Driver for Wolfson WM8350 PMIC
  *
- * Copyright 2007 Wolfson Microelectronics PLC
+ * Copyright 2007, 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 as published by the
@@ -70,9 +70,9 @@
 #define WM8350_CODEC_ISEL_0_5                   3	/* x0.5 */
 
 #define WM8350_VMID_OFF                         0
-#define WM8350_VMID_500K                        1
-#define WM8350_VMID_100K                        2
-#define WM8350_VMID_10K                         3
+#define WM8350_VMID_300K                        1
+#define WM8350_VMID_50K                         2
+#define WM8350_VMID_5K                          3
 
 /*
  * R40 (0x28) - Clock Control 1
@@ -591,8 +591,38 @@
 #define WM8350_IRQ_CODEC_MICSCD			41
 #define WM8350_IRQ_CODEC_MICD			42
 
+/*
+ * WM8350 Platform data.
+ *
+ * This must be initialised per platform for best audio performance.
+ * Please see WM8350 datasheet for information.
+ */
+struct wm8350_audio_platform_data {
+	int vmid_discharge_msecs;	/* VMID --> OFF discharge time */
+	int drain_msecs;	/* OFF drain time */
+	int cap_discharge_msecs;	/* Cap ON (from OFF) discharge time */
+	int vmid_charge_msecs;	/* vmid power up time */
+	u32 vmid_s_curve:2;	/* vmid enable s curve speed */
+	u32 dis_out4:2;		/* out4 discharge speed */
+	u32 dis_out3:2;		/* out3 discharge speed */
+	u32 dis_out2:2;		/* out2 discharge speed */
+	u32 dis_out1:2;		/* out1 discharge speed */
+	u32 vroi_out4:1;	/* out4 tie off */
+	u32 vroi_out3:1;	/* out3 tie off */
+	u32 vroi_out2:1;	/* out2 tie off */
+	u32 vroi_out1:1;	/* out1 tie off */
+	u32 vroi_enable:1;	/* enable tie off */
+	u32 codec_current_on:2;	/* current level ON */
+	u32 codec_current_standby:2;	/* current level STANDBY */
+	u32 codec_current_charge:2;	/* codec current @ vmid charge */
+};
+
+struct snd_soc_codec;
+
 struct wm8350_codec {
 	struct platform_device *pdev;
+	struct snd_soc_codec *codec;
+	struct wm8350_audio_platform_data *platform_data;
 };
 
 #endif
diff --git a/include/sound/l3.h b/include/sound/l3.h
new file mode 100644
index 000000000000..423a08f0f1b0
--- /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/s3c24xx_uda134x.h b/include/sound/s3c24xx_uda134x.h
new file mode 100644
index 000000000000..33df4cb909d3
--- /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
diff --git a/include/sound/soc-dai.h b/include/sound/soc-dai.h
new file mode 100644
index 000000000000..24247f763608
--- /dev/null
+++ b/include/sound/soc-dai.h
@@ -0,0 +1,231 @@
+/*
+ * 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 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);
+
+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);
+
+	/*
+	 * 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 *);
+};
+
+/*
+ * Digital Audio Interface runtime data.
+ *
+ * Holds runtime data for a DAI.
+ */
+struct snd_soc_dai {
+	/* DAI description */
+	char *name;
+	unsigned int id;
+	int ac97_control;
+
+	struct device *dev;
+
+	/* 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 snd_soc_dai *dai);
+	int (*resume)(struct snd_soc_dai *dai);
+
+	/* ops */
+	struct snd_soc_dai_ops 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-dapm.h b/include/sound/soc-dapm.h
index ca699a3017f3..7ee2f70ca42e 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,
diff --git a/include/sound/soc.h b/include/sound/soc.h
index 5e0189876afd..f86e455d3828 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
  */
@@ -145,105 +143,31 @@ 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 */
-
-/*
- * 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;
 struct snd_soc_dai_mode;
 struct snd_soc_pcm_runtime;
 struct snd_soc_dai;
+struct snd_soc_platform;
 struct snd_soc_codec;
-struct snd_soc_machine_config;
 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);
 
 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);
 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,
@@ -263,27 +187,6 @@ 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
  */
@@ -341,66 +244,14 @@ 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;
 	struct module *owner;
 	struct mutex mutex;
+	struct device *dev;
+
+	struct list_head list;
 
 	/* callbacks */
 	int (*set_bias_level)(struct snd_soc_codec *,
@@ -426,6 +277,7 @@ 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;
@@ -435,6 +287,11 @@ 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 */
@@ -448,13 +305,12 @@ 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);
-	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 *,
@@ -484,9 +340,14 @@ struct snd_soc_dai_link  {
 	struct snd_pcm *pcm;
 };
 
-/* SoC machine */
-struct snd_soc_machine {
+/* 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);
@@ -499,23 +360,26 @@ 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  */
 	struct snd_soc_dai_link *dai_link;
 	int num_links;
+
+	struct snd_soc_device *socdev;
+
+	struct snd_soc_platform *platform;
+	struct delayed_work delayed_work;
+	struct work_struct deferred_resume_work;
 };
 
 /* SoC Device - the audio subsystem */
 struct snd_soc_device {
 	struct device *dev;
-	struct snd_soc_machine *machine;
-	struct snd_soc_platform *platform;
+	struct snd_soc_card *card;
 	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;
 };
 
@@ -542,4 +406,6 @@ struct soc_enum {
 	void *dapm;
 };
 
+#include <sound/soc-dai.h>
+
 #endif
diff --git a/include/sound/uda134x.h b/include/sound/uda134x.h
new file mode 100644
index 000000000000..475ef8bb7dcd
--- /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 */