11 files changed, 362 insertions, 183 deletions

diff --git a/include/sound/ac97_codec.h b/include/sound/ac97_codec.h
index 9c309daf492b..251fc1cd5002 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 */
diff --git a/include/sound/core.h b/include/sound/core.h
index e5eec5f73502..f632484bc743 100644
--- a/include/sound/core.h
+++ b/include/sound/core.h
@@ -43,9 +43,6 @@
 #ifdef CONFIG_PCI
 struct pci_dev;
 #endif
-#ifdef CONFIG_SBUS
-struct sbus_dev;
-#endif
 
 /* device allocation stuff */
 
@@ -356,7 +353,7 @@ 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...) \
@@ -383,14 +380,40 @@ 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 */
 
-#define snd_printd(fmt, args...)        /* nothing */
-#define snd_BUG()                       /* nothing */
-#define snd_BUG_ON(cond)        ({/*(void)(cond);*/ 0;})  /* always false */
+#define snd_printd(fmt, args...)        do { } while (0)
+#define snd_BUG()                       do { } while (0)
+static inline int __snd_bug_on(int cond)
+{
+        return 0;
+}
+#define snd_BUG_ON(cond)        __snd_bug_on(0 && (cond))  /* always false */
 
 #endif /* CONFIG_SND_DEBUG */
 
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/memalloc.h b/include/sound/memalloc.h
index d787a6b4a101..7ccce94a5255 100644
--- a/include/sound/memalloc.h
+++ b/include/sound/memalloc.h
@@ -37,7 +37,6 @@ struct snd_dma_device {
 #ifndef snd_dma_pci_data
 #define snd_dma_pci_data(pci)   (&(pci)->dev)
 #define snd_dma_isa_data()      NULL
-#define snd_dma_sbus_data(sbus) ((struct device *)(sbus))
 #define snd_dma_continuous_data(x)      ((struct device *)(unsigned long)(x))
 #endif
 
@@ -49,7 +48,6 @@ struct snd_dma_device {
 #define SNDRV_DMA_TYPE_CONTINUOUS       1       /* continuous no-DMA memory */
 #define SNDRV_DMA_TYPE_DEV              2       /* generic device continuous */
 #define SNDRV_DMA_TYPE_DEV_SG           3       /* generic device SG-buffer */
-#define SNDRV_DMA_TYPE_SBUS             4       /* SBUS continuous */
 
 /*
  * info for buffer allocation
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 c1b26fcc0b5c..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,
@@ -240,6 +238,7 @@ int snd_soc_dapm_sys_add(struct device *dev);
 /* dapm audio pin control and status */
 int snd_soc_dapm_enable_pin(struct snd_soc_codec *codec, char *pin);
 int snd_soc_dapm_disable_pin(struct snd_soc_codec *codec, char *pin);
+int snd_soc_dapm_nc_pin(struct snd_soc_codec *codec, char *pin);
 int snd_soc_dapm_get_pin_status(struct snd_soc_codec *codec, char *pin);
 int snd_soc_dapm_sync(struct snd_soc_codec *codec);
 
diff --git a/include/sound/soc.h b/include/sound/soc.h
index a1e0357a84d7..f86e455d3828 100644
--- a/include/sound/soc.h
+++ b/include/sound/soc.h
@@ -21,14 +21,13 @@
 #include <sound/control.h>
 #include <sound/ac97_codec.h>
 
-#define SND_SOC_VERSION "0.13.2"
-
 /*
  * Convenience kcontrol builders
  */
 #define SOC_SINGLE_VALUE(xreg, xshift, xmax, xinvert) \
         ((unsigned long)&(struct soc_mixer_control) \
-        {.reg = xreg, .shift = xshift, .max = xmax, .invert = xinvert})
+        {.reg = xreg, .shift = xshift, .rshift = xshift, .max = xmax, \
+        .invert = xinvert})
 #define SOC_SINGLE_VALUE_EXT(xreg, xmax, xinvert) \
         ((unsigned long)&(struct soc_mixer_control) \
         {.reg = xreg, .max = xmax, .invert = xinvert})
@@ -144,105 +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,
@@ -262,27 +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
  */
@@ -340,66 +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 *,
@@ -425,6 +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;
@@ -434,6 +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 */
@@ -447,13 +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 *,
@@ -483,9 +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);
@@ -498,23 +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;
 };
 
@@ -541,4 +406,6 @@ struct soc_enum {
         void *dapm;
 };
 
+#include <sound/soc-dai.h>
+
 #endif
diff --git a/include/sound/tea575x-tuner.h b/include/sound/tea575x-tuner.h
index b62ce3e077f9..b6870cbaf2b3 100644
--- a/include/sound/tea575x-tuner.h
+++ b/include/sound/tea575x-tuner.h
@@ -43,6 +43,7 @@ struct snd_tea575x {
         unsigned int freq_fixup;        /* crystal onboard */
         unsigned int val;               /* hw value */
         unsigned long freq;             /* frequency */
+        unsigned long in_use;           /* set if the device is in use */
         struct snd_tea575x_ops *ops;
         void *private_data;
 };
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 */
diff --git a/include/sound/version.h b/include/sound/version.h
index 4aafeda88634..2b48237e23bf 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 "1.0.18a"
 #define CONFIG_SND_DATE ""