Merge branch 'topic/asoc' into to-push

Documentation/sound/alsa/soc/machine.txt

@@ -9,7 +9,7 @@ the audio subsystem with the kernel as a platform device and is represented by
 the following struct:-
 
 /* SoC machine */
-struct snd_soc_machine {
+struct snd_soc_card {
 	char *name;
 
 	int (*probe)(struct platform_device *pdev);
@@ -67,10 +67,10 @@ static struct snd_soc_dai_link corgi_dai = {
 	.ops = &corgi_ops,
 };
 
-struct snd_soc_machine then sets up the machine with it's DAIs. e.g.
+struct snd_soc_card then sets up the machine with it's DAIs. e.g.
 
 /* corgi audio machine driver */
-static struct snd_soc_machine snd_soc_machine_corgi = {
+static struct snd_soc_card snd_soc_corgi = {
 	.name = "Corgi",
 	.dai_link = &corgi_dai,
 	.num_links = 1,
@@ -90,7 +90,7 @@ static struct wm8731_setup_data corgi_wm8731_setup = {
 
 /* corgi audio subsystem */
 static struct snd_soc_device corgi_snd_devdata = {
-	.machine = &snd_soc_machine_corgi,
+	.machine = &snd_soc_corgi,
 	.platform = &pxa2xx_soc_platform,
 	.codec_dev = &soc_codec_dev_wm8731,
 	.codec_data = &corgi_wm8731_setup,

MAINTAINERS

@@ -3977,7 +3977,7 @@ M:	tiwai@suse.de
 L:	alsa-devel@alsa-project.org (subscribers-only)
 S:	Maintained
 
-SOUND - SOC LAYER / DYNAMIC AUDIO POWER MANAGEMENT
+SOUND - SOC LAYER / DYNAMIC AUDIO POWER MANAGEMENT (ASoC)
 P:	Liam Girdwood
 M:	lrg@slimlogic.co.uk
 P:	Mark Brown

arch/arm/mach-pxa/include/mach/palmasoc.h

+#ifndef _INCLUDE_PALMASOC_H_
+#define _INCLUDE_PALMASOC_H_
+struct palm27x_asoc_info {
+	int	jack_gpio;
+};
+
+#ifdef CONFIG_SND_PXA2XX_SOC_PALM27X
+void __init palm27x_asoc_set_pdata(struct palm27x_asoc_info *data);
+#else
+static inline void palm27x_asoc_set_pdata(struct palm27x_asoc_info *data) {}
+#endif
+
+#endif

include/linux/mfd/wm8350/audio.h

 /*
  * 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

include/sound/l3.h

+#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

include/sound/s3c24xx_uda134x.h

+#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

include/sound/soc-dai.h

+/*
+ * 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

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,

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

include/sound/uda134x.h

+/*
+ * 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 */

sound/soc/Kconfig

@@ -22,17 +22,16 @@ if SND_SOC
 config SND_SOC_AC97_BUS
 	bool
 
-# All the supported Soc's
-source "sound/soc/at32/Kconfig"
-source "sound/soc/at91/Kconfig"
+# All the supported SoCs
+source "sound/soc/atmel/Kconfig"
 source "sound/soc/au1x/Kconfig"
+source "sound/soc/blackfin/Kconfig"
+source "sound/soc/davinci/Kconfig"
+source "sound/soc/fsl/Kconfig"
+source "sound/soc/omap/Kconfig"
 source "sound/soc/pxa/Kconfig"
 source "sound/soc/s3c24xx/Kconfig"
 source "sound/soc/sh/Kconfig"
-source "sound/soc/fsl/Kconfig"
-source "sound/soc/davinci/Kconfig"
-source "sound/soc/omap/Kconfig"
-source "sound/soc/blackfin/Kconfig"
 
 # Supported codecs
 source "sound/soc/codecs/Kconfig"

sound/soc/Makefile

 snd-soc-core-objs := soc-core.o soc-dapm.o
 
 obj-$(CONFIG_SND_SOC)	+= snd-soc-core.o
-obj-$(CONFIG_SND_SOC)	+= codecs/ at32/ at91/ pxa/ s3c24xx/ sh/ fsl/ davinci/
-obj-$(CONFIG_SND_SOC)	+= omap/ au1x/ blackfin/
+obj-$(CONFIG_SND_SOC)	+= codecs/
+obj-$(CONFIG_SND_SOC)	+= atmel/
+obj-$(CONFIG_SND_SOC)	+= au1x/
+obj-$(CONFIG_SND_SOC)	+= blackfin/
+obj-$(CONFIG_SND_SOC)	+= davinci/
+obj-$(CONFIG_SND_SOC)	+= fsl/
+obj-$(CONFIG_SND_SOC)	+= omap/
+obj-$(CONFIG_SND_SOC)	+= pxa/
+obj-$(CONFIG_SND_SOC)	+= s3c24xx/
+obj-$(CONFIG_SND_SOC)	+= sh/

sound/soc/at32/Makefile

-# AT32 Platform Support
-snd-soc-at32-objs := at32-pcm.o
-snd-soc-at32-ssc-objs := at32-ssc.o
-
-obj-$(CONFIG_SND_AT32_SOC) += snd-soc-at32.o
-obj-$(CONFIG_SND_AT32_SOC_SSC) += snd-soc-at32-ssc.o
-
-# AT32 Machine Support
-snd-soc-playpaq-objs := playpaq_wm8510.o
-
-obj-$(CONFIG_SND_AT32_SOC_PLAYPAQ) += snd-soc-playpaq.o

sound/soc/at32/at32-pcm.h

-/* sound/soc/at32/at32-pcm.h
- * ASoC PCM interface for Atmel AT32 SoC
- *
- * Copyright (C) 2008 Long Range Systems
- *    Geoffrey Wossum <gwossum@acm.org>
- *
- * 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 __SOUND_SOC_AT32_AT32_PCM_H
-#define __SOUND_SOC_AT32_AT32_PCM_H __FILE__
-
-#include <linux/atmel-ssc.h>
-
-
-/*
- * Registers and status bits that are required by the PCM driver
- * TODO: Is ptcr really used?
- */
-struct at32_pdc_regs {
-	u32 xpr;		/* PDC RX/TX pointer */
-	u32 xcr;		/* PDC RX/TX counter */
-	u32 xnpr;		/* PDC next RX/TX pointer */
-	u32 xncr;		/* PDC next RX/TX counter */
-	u32 ptcr;		/* PDC transfer control */
-};
-
-
-
-/*
- * SSC mask info
- */
-struct at32_ssc_mask {
-	u32 ssc_enable;		/* SSC RX/TX enable */
-	u32 ssc_disable;	/* SSC RX/TX disable */
-	u32 ssc_endx;		/* SSC ENDTX or ENDRX */
-	u32 ssc_endbuf;		/* SSC TXBUFF or RXBUFF */
-	u32 pdc_enable;		/* PDC RX/TX enable */
-	u32 pdc_disable;	/* PDC RX/TX disable */
-};
-
-
-
-/*
- * This structure, shared between the PCM driver and the interface,
- * contains all information required by the PCM driver to perform the
- * PDC DMA operation.  All fields except dma_intr_handler() are initialized
- * by the interface.  The dms_intr_handler() pointer is set by the PCM
- * driver and called by the interface SSC interrupt handler if it is
- * non-NULL.
- */
-struct at32_pcm_dma_params {
-	char *name;		/* stream identifier */
-	int pdc_xfer_size;	/* PDC counter increment in bytes */
-	struct ssc_device *ssc;	/* SSC device for stream */
-	struct at32_pdc_regs *pdc;	/* PDC register info */
-	struct at32_ssc_mask *mask;	/* SSC mask info */
-	struct snd_pcm_substream *substream;
-	void (*dma_intr_handler) (u32, struct snd_pcm_substream *);
-};
-
-
-
-/*
- * The AT32 ASoC platform driver
- */
-extern struct snd_soc_platform at32_soc_platform;
-
-
-
-/*
- * SSC register access (since ssc_writel() / ssc_readl() require literal name)
- */
-#define ssc_readx(base, reg)            (__raw_readl((base) + (reg)))
-#define ssc_writex(base, reg, value)    __raw_writel((value), (base) + (reg))
-
-#endif /* __SOUND_SOC_AT32_AT32_PCM_H */

sound/soc/at32/at32-ssc.h

-/* sound/soc/at32/at32-ssc.h
- * ASoC SSC interface for Atmel AT32 SoC
- *
- * Copyright (C) 2008 Long Range Systems
- *    Geoffrey Wossum <gwossum@acm.org>
- *
- * 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 __SOUND_SOC_AT32_AT32_SSC_H
-#define __SOUND_SOC_AT32_AT32_SSC_H __FILE__
-
-#include <linux/types.h>
-#include <linux/atmel-ssc.h>
-
-#include "at32-pcm.h"
-
-
-
-struct at32_ssc_state {
-	u32 ssc_cmr;
-	u32 ssc_rcmr;
-	u32 ssc_rfmr;
-	u32 ssc_tcmr;
-	u32 ssc_tfmr;
-	u32 ssc_sr;
-	u32 ssc_imr;
-};
-
-
-
-struct at32_ssc_info {
-	char *name;
-	struct ssc_device *ssc;
-	spinlock_t lock;	/* lock for dir_mask */
-	unsigned short dir_mask;	/* 0=unused, 1=playback, 2=capture */
-	unsigned short initialized;	/* true if SSC has been initialized */
-	unsigned short daifmt;
-	unsigned short cmr_div;
-	unsigned short tcmr_period;
-	unsigned short rcmr_period;
-	struct at32_pcm_dma_params *dma_params[2];
-	struct at32_ssc_state ssc_state;
-};
-
-
-/* SSC divider ids */
-#define AT32_SSC_CMR_DIV        0	/* MCK divider for BCLK */
-#define AT32_SSC_TCMR_PERIOD    1	/* BCLK divider for transmit FS */
-#define AT32_SSC_RCMR_PERIOD    2	/* BCLK divider for receive FS */
-
-
-extern struct snd_soc_dai at32_ssc_dai[];
-
-
-
-#endif /* __SOUND_SOC_AT32_AT32_SSC_H */

sound/soc/at91/Kconfig

-config SND_AT91_SOC
-	tristate "SoC Audio for the Atmel AT91 System-on-Chip"
-	depends on ARCH_AT91
-	help
-	  Say Y or M if you want to add support for codecs attached to
-	  the AT91 SSC interface. You will also need
-	  to select the audio interfaces to support below.
-
-config SND_AT91_SOC_SSC
-	tristate

sound/soc/at91/Makefile

-# AT91 Platform Support
-snd-soc-at91-objs := at91-pcm.o
-snd-soc-at91-ssc-objs := at91-ssc.o
-
-obj-$(CONFIG_SND_AT91_SOC) += snd-soc-at91.o
-obj-$(CONFIG_SND_AT91_SOC_SSC) += snd-soc-at91-ssc.o

sound/soc/at91/at91-ssc.h

-/*
- * at91-ssc.h - ALSA SSC interface for the Atmel AT91 SoC
- *
- * Author:	Frank Mandarino <fmandarino@endrelia.com>
- *		Endrelia Technologies Inc.
- * Created:	Jan 9, 2007
- *
- * 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 _AT91_SSC_H
-#define _AT91_SSC_H
-
-/* SSC system clock ids */
-#define AT91_SYSCLK_MCK		0 /* SSC uses AT91 MCK as system clock */
-
-/* SSC divider ids */
-#define AT91SSC_CMR_DIV		0 /* MCK divider for BCLK */
-#define AT91SSC_TCMR_PERIOD	1 /* BCLK divider for transmit FS */
-#define AT91SSC_RCMR_PERIOD	2 /* BCLK divider for receive FS */
-
-extern struct snd_soc_dai at91_ssc_dai[];
-
-#endif /* _AT91_SSC_H */
-

sound/soc/at32/Kconfig → sound/soc/atmel/Kconfig

-config SND_AT32_SOC
-        tristate "SoC Audio for the Atmel AT32 System-on-a-Chip"
-        depends on AVR32 && SND_SOC
-        help
-          Say Y or M if you want to add support for codecs attached to 
-          the AT32 SSC interface.  You will also need to
-          to select the audio interfaces to support below.
-
-
-config SND_AT32_SOC_SSC
-        tristate
-
-
+config SND_ATMEL_SOC
+	tristate "SoC Audio for the Atmel System-on-Chip"
+	depends on ARCH_AT91 || AVR32
+	help
+	  Say Y or M if you want to add support for codecs attached to
+	  the ATMEL SSC interface. You will also need
+	  to select the audio interfaces to support below.
+
+config SND_ATMEL_SOC_SSC
+	tristate
+	depends on SND_ATMEL_SOC
+	help
+	  Say Y or M if you want to add support for codecs the
+	  ATMEL SSC interface. You will also needs to select the individual
+	  machine drivers to support below.
+
+config SND_AT91_SOC_SAM9G20_WM8731
+	tristate "SoC Audio support for WM8731-based At91sam9g20 evaluation board"
+	depends on ATMEL_SSC && ARCH_AT91SAM9G20 && SND_ATMEL_SOC
+	select SND_ATMEL_SOC_SSC
+	select SND_SOC_WM8731
+	help
+	  Say Y if you want to add support for SoC audio on WM8731-based
+	  AT91sam9g20 evaluation board.
 
 config SND_AT32_SOC_PLAYPAQ
         tristate "SoC Audio support for PlayPaq with WM8510"
-        depends on SND_AT32_SOC && BOARD_PLAYPAQ
-        select SND_AT32_SOC_SSC
+        depends on SND_ATMEL_SOC && BOARD_PLAYPAQ
+        select SND_ATMEL_SOC_SSC
         select SND_SOC_WM8510
         help
           Say Y or M here if you want to add support for SoC audio
           on the LRS PlayPaq.
 
-
-
 config SND_AT32_SOC_PLAYPAQ_SLAVE
         bool "Run CODEC on PlayPaq in slave mode"
         depends on SND_AT32_SOC_PLAYPAQ

sound/soc/atmel/Makefile

+# AT91 Platform Support
+snd-soc-atmel-pcm-objs := atmel-pcm.o
+snd-soc-atmel_ssc_dai-objs := atmel_ssc_dai.o
+
+obj-$(CONFIG_SND_ATMEL_SOC) += snd-soc-atmel-pcm.o
+obj-$(CONFIG_SND_ATMEL_SOC_SSC) += snd-soc-atmel_ssc_dai.o
+
+# AT91 Machine Support
+snd-soc-sam9g20-wm8731-objs := sam9g20_wm8731.o
+
+# AT32 Machine Support
+snd-soc-playpaq-objs := playpaq_wm8510.o
+
+obj-$(CONFIG_SND_AT91_SOC_SAM9G20_WM8731) += snd-soc-sam9g20-wm8731.o
+obj-$(CONFIG_SND_AT32_SOC_PLAYPAQ) += snd-soc-playpaq.o

sound/soc/at91/at91-pcm.h → sound/soc/atmel/atmel-pcm.h

 /*
- * at91-pcm.h - ALSA PCM interface for the Atmel AT91 SoC
+ * at91-pcm.h - ALSA PCM interface for the Atmel AT91 SoC.
  *
- * Author:	Frank Mandarino <fmandarino@endrelia.com>
- *		Endrelia Technologies Inc.
- * Created:	Mar 3, 2006
+ *  Copyright (C) 2005 SAN People
+ *  Copyright (C) 2008 Atmel
  *
- * Based on pxa2xx-pcm.h by:
+ * Authors: Sedji Gaouaou <sedji.gaouaou@atmel.com>
+ *
+ * Based on at91-pcm. by:
+ * Frank Mandarino <fmandarino@endrelia.com>
+ * Copyright 2006 Endrelia Technologies Inc.
+ *
+ * Based on pxa2xx-pcm.c by:
  *
  * Author:	Nicolas Pitre
  * Created:	Nov 30, 2004
- * Copyright:	MontaVista Software, Inc.
+ * Copyright:	(C) 2004 MontaVista Software, Inc.
  *
  * 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.
+ * it under the terms of the GNU General Public License as published by
+ * the Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA  02111-1307  USA
  */
 
-#ifndef _AT91_PCM_H
-#define _AT91_PCM_H
+#ifndef _ATMEL_PCM_H
+#define _ATMEL_PCM_H
 
-#include <mach/hardware.h>
-
-struct at91_ssc_periph {
-	void __iomem	*base;
-	u32		pid;
-};
+#include <linux/atmel-ssc.h>
 
 /*
  * Registers and status bits that are required by the PCM driver.
  */
-struct at91_pdc_regs {
+struct atmel_pdc_regs {
 	unsigned int	xpr;		/* PDC recv/trans pointer */
 	unsigned int	xcr;		/* PDC recv/trans counter */
 	unsigned int	xnpr;		/* PDC next recv/trans pointer */
@@ -37,7 +47,7 @@ struct at91_pdc_regs {
 	unsigned int	ptcr;		/* PDC transfer control */
 };
 
-struct at91_ssc_mask {
+struct atmel_ssc_mask {
 	u32	ssc_enable;		/* SSC recv/trans enable */
 	u32	ssc_disable;		/* SSC recv/trans disable */
 	u32	ssc_endx;		/* SSC ENDTX or ENDRX */
@@ -54,19 +64,23 @@ struct at91_ssc_mask {
  * driver and called by the interface SSC interrupt handler if it is
  * non-NULL.
  */
-struct at91_pcm_dma_params {
+struct atmel_pcm_dma_params {
 	char *name;			/* stream identifier */
 	int pdc_xfer_size;		/* PDC counter increment in bytes */
-	void __iomem *ssc_base;		/* SSC base address */
-	struct at91_pdc_regs *pdc; /* PDC receive or transmit registers */
-	struct at91_ssc_mask *mask;/* SSC & PDC status bits */
+	struct ssc_device *ssc;		/* SSC device for stream */
+	struct atmel_pdc_regs *pdc;	/* PDC receive or transmit registers */
+	struct atmel_ssc_mask *mask;	/* SSC & PDC status bits */
 	struct snd_pcm_substream *substream;
 	void (*dma_intr_handler)(u32, struct snd_pcm_substream *);
 };
 
-extern struct snd_soc_platform at91_soc_platform;
+extern struct snd_soc_platform atmel_soc_platform;
+
 
-#define at91_ssc_read(a)	((unsigned long) __raw_readl(a))
-#define at91_ssc_write(a,v)	__raw_writel((v),(a))
+/*
+ * SSC register access (since ssc_writel() / ssc_readl() require literal name)
+ */
+#define ssc_readx(base, reg)            (__raw_readl((base) + (reg)))
+#define ssc_writex(base, reg, value)    __raw_writel((value), (base) + (reg))
 
-#endif /* _AT91_PCM_H */
+#endif /* _ATMEL_PCM_H */

sound/soc/atmel/atmel_ssc_dai.h

+/*
+ * atmel_ssc_dai.h - ALSA SSC interface for the Atmel  SoC
+ *
+ * Copyright (C) 2005 SAN People
+ * Copyright (C) 2008 Atmel
+ *
+ * Author: Sedji Gaouaou <sedji.gaouaou@atmel.com>
+ *         ATMEL CORP.
+ *
+ * Based on at91-ssc.c by
+ * Frank Mandarino <fmandarino@endrelia.com>
+ * Based on pxa2xx Platform drivers by
+ * Liam Girdwood <liam.girdwood@wolfsonmicro.com>
+ *
+ * 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 Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA  02111-1307  USA
+ */
+
+#ifndef _ATMEL_SSC_DAI_H
+#define _ATMEL_SSC_DAI_H
+
+#include <linux/types.h>
+#include <linux/atmel-ssc.h>
+
+#include "atmel-pcm.h"
+
+/* SSC system clock ids */
+#define ATMEL_SYSCLK_MCK	0 /* SSC uses AT91 MCK as system clock */
+
+/* SSC divider ids */
+#define ATMEL_SSC_CMR_DIV	0 /* MCK divider for BCLK */
+#define ATMEL_SSC_TCMR_PERIOD	1 /* BCLK divider for transmit FS */
+#define ATMEL_SSC_RCMR_PERIOD	2 /* BCLK divider for receive FS */
+/*
+ * SSC direction masks
+ */
+#define SSC_DIR_MASK_UNUSED	0
+#define SSC_DIR_MASK_PLAYBACK	1
+#define SSC_DIR_MASK_CAPTURE	2
+
+/*
+ * SSC register values that Atmel left out of <linux/atmel-ssc.h>.  These
+ * are expected to be used with SSC_BF
+ */
+/* START bit field values */
+#define SSC_START_CONTINUOUS	0
+#define SSC_START_TX_RX		1
+#define SSC_START_LOW_RF	2
+#define SSC_START_HIGH_RF	3
+#define SSC_START_FALLING_RF	4
+#define SSC_START_RISING_RF	5
+#define SSC_START_LEVEL_RF	6
+#define SSC_START_EDGE_RF	7
+#define SSS_START_COMPARE_0	8
+
+/* CKI bit field values */
+#define SSC_CKI_FALLING		0
+#define SSC_CKI_RISING		1
+
+/* CKO bit field values */
+#define SSC_CKO_NONE		0
+#define SSC_CKO_CONTINUOUS	1
+#define SSC_CKO_TRANSFER	2
+
+/* CKS bit field values */
+#define SSC_CKS_DIV		0
+#define SSC_CKS_CLOCK		1
+#define SSC_CKS_PIN		2
+
+/* FSEDGE bit field values */
+#define SSC_FSEDGE_POSITIVE	0
+#define SSC_FSEDGE_NEGATIVE	1
+
+/* FSOS bit field values */
+#define SSC_FSOS_NONE		0
+#define SSC_FSOS_NEGATIVE	1
+#define SSC_FSOS_POSITIVE	2
+#define SSC_FSOS_LOW		3
+#define SSC_FSOS_HIGH		4
+#define SSC_FSOS_TOGGLE		5
+
+#define START_DELAY		1
+
+struct atmel_ssc_state {
+	u32 ssc_cmr;
+	u32 ssc_rcmr;
+	u32 ssc_rfmr;
+	u32 ssc_tcmr;
+	u32 ssc_tfmr;
+	u32 ssc_sr;
+	u32 ssc_imr;
+};
+
+
+struct atmel_ssc_info {
+	char *name;
+	struct ssc_device *ssc;
+	spinlock_t lock;	/* lock for dir_mask */
+	unsigned short dir_mask;	/* 0=unused, 1=playback, 2=capture */
+	unsigned short initialized;	/* true if SSC has been initialized */
+	unsigned short daifmt;
+	unsigned short cmr_div;
+	unsigned short tcmr_period;
+	unsigned short rcmr_period;
+	struct atmel_pcm_dma_params *dma_params[2];
+	struct atmel_ssc_state ssc_state;
+};
+extern struct snd_soc_dai atmel_ssc_dai[];
+
+#endif /* _AT91_SSC_DAI_H */

sound/soc/at32/playpaq_wm8510.c → sound/soc/atmel/playpaq_wm8510.c

@@ -22,7 +22,6 @@
 
 #include <linux/module.h>
 #include <linux/moduleparam.h>
-#include <linux/version.h>
 #include <linux/kernel.h>
 #include <linux/errno.h>
 #include <linux/clk.h>
@@ -40,8 +39,8 @@
 #include <mach/portmux.h>
 
 #include "../codecs/wm8510.h"
-#include "at32-pcm.h"
-#include "at32-ssc.h"
+#include "atmel-pcm.h"
+#include "atmel_ssc_dai.h"
 
 
 /*-------------------------------------------------------------------------*\
@@ -362,8 +361,9 @@ static struct snd_soc_dai_link playpaq_wm8510_dai = {
 
 
 
-static struct snd_soc_machine snd_soc_machine_playpaq = {
+static struct snd_soc_card snd_soc_playpaq = {
 	.name = "LRS_PlayPaq_WM8510",
+	.platform = &at32_soc_platform,
 	.dai_link = &playpaq_wm8510_dai,
 	.num_links = 1,
 };
@@ -378,8 +378,7 @@ static struct wm8510_setup_data playpaq_wm8510_setup = {
 
 
 static struct snd_soc_device playpaq_wm8510_snd_devdata = {
-	.machine = &snd_soc_machine_playpaq,
-	.platform = &at32_soc_platform,
+	.card = &snd_soc_playpaq,
 	.codec_dev = &soc_codec_dev_wm8510,
 	.codec_data = &playpaq_wm8510_setup,
 };

sound/soc/atmel/sam9g20_wm8731.c

+/*
+ * sam9g20_wm8731  --  SoC audio for AT91SAM9G20-based
+ * 			ATMEL AT91SAM9G20ek board.
+ *
+ *  Copyright (C) 2005 SAN People
+ *  Copyright (C) 2008 Atmel
+ *
+ * Authors: Sedji Gaouaou <sedji.gaouaou@atmel.com>
+ *
+ * Based on ati_b1_wm8731.c by:
+ * Frank Mandarino <fmandarino@endrelia.com>
+ * Copyright 2006 Endrelia Technologies Inc.
+ * Based on corgi.c by:
+ * Copyright 2005 Wolfson Microelectronics PLC.
+ * Copyright 2005 Openedhand Ltd.
+ *
+ * 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 Free Software Foundation; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA  02111-1307  USA
+ */
+
+#include <linux/module.h>
+#include <linux/moduleparam.h>
+#include <linux/kernel.h>
+#include <linux/clk.h>
+#include <linux/timer.h>
+#include <linux/interrupt.h>
+#include <linux/platform_device.h>
+
+#include <linux/atmel-ssc.h>
+
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+
+#include <mach/hardware.h>
+#include <mach/gpio.h>
+
+#include "../codecs/wm8731.h"
+#include "atmel-pcm.h"
+#include "atmel_ssc_dai.h"
+
+
+static int at91sam9g20ek_startup(struct snd_pcm_substream *substream)
+{
+	struct snd_soc_pcm_runtime *rtd = snd_pcm_substream_chip(substream);
+	struct snd_soc_dai *codec_dai = rtd->dai->codec_dai;
+	int ret;
+
+	/* codec system clock is supplied by PCK0, set to 12MHz */
+	ret = snd_soc_dai_set_sysclk(codec_dai, WM8731_SYSCLK,
+		12000000, SND_SOC_CLOCK_IN);
+	if (ret < 0)
+		return ret;
+
+	return 0;
+}
+
+static void at91sam9g20ek_shutdown(struct snd_pcm_substream *substream)
+{
+	struct snd_soc_pcm_runtime *rtd = snd_pcm_substream_chip(substream);
+
+	dev_dbg(rtd->socdev->dev, "shutdown");
+}
+
+static int at91sam9g20ek_hw_params(struct snd_pcm_substream *substream,
+	struct snd_pcm_hw_params *params)
+{
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_dai *codec_dai = rtd->dai->codec_dai;
+	struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;
+	struct atmel_ssc_info *ssc_p = cpu_dai->private_data;
+	struct ssc_device *ssc = ssc_p->ssc;
+	int ret;
+
+	unsigned int rate;
+	int cmr_div, period;
+
+	if (ssc == NULL) {
+		printk(KERN_INFO "at91sam9g20ek_hw_params: ssc is NULL!\n");
+		return -EINVAL;
+	}
+
+	/* set codec DAI configuration */
+	ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S |
+		SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS);
+	if (ret < 0)
+		return ret;
+
+	/* set cpu DAI configuration */
+	ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S |
+		SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBS_CFS);
+	if (ret < 0)
+		return ret;
+
+	/*
+	 * The SSC clock dividers depend on the sample rate.  The CMR.DIV
+	 * field divides the system master clock MCK to drive the SSC TK
+	 * signal which provides the codec BCLK.  The TCMR.PERIOD and
+	 * RCMR.PERIOD fields further divide the BCLK signal to drive
+	 * the SSC TF and RF signals which provide the codec DACLRC and
+	 * ADCLRC clocks.
+	 *
+	 * The dividers were determined through trial and error, where a
+	 * CMR.DIV value is chosen such that the resulting BCLK value is
+	 * divisible, or almost divisible, by (2 * sample rate), and then
+	 * the TCMR.PERIOD or RCMR.PERIOD is BCLK / (2 * sample rate) - 1.
+	 */
+	rate = params_rate(params);
+
+	switch (rate) {
+	case 8000:
+		cmr_div = 55;	/* BCLK = 133MHz/(2*55) = 1.209MHz */
+		period = 74;	/* LRC = BCLK/(2*(74+1)) ~= 8060,6Hz */
+		break;
+	case 11025:
+		cmr_div = 67;	/* BCLK = 133MHz/(2*60) = 1.108MHz */
+		period = 45;	/* LRC = BCLK/(2*(49+1)) = 11083,3Hz */
+		break;
+	case 16000:
+		cmr_div = 63;	/* BCLK = 133MHz/(2*63) = 1.055MHz */
+		period = 32;	/* LRC = BCLK/(2*(32+1)) = 15993,2Hz */
+		break;
+	case 22050:
+		cmr_div = 52;	/* BCLK = 133MHz/(2*52) = 1.278MHz */
+		period = 28;	/* LRC = BCLK/(2*(28+1)) = 22049Hz */
+		break;
+	case 32000:
+		cmr_div = 66;	/* BCLK = 133MHz/(2*66) = 1.007MHz */
+		period = 15;	/* LRC = BCLK/(2*(15+1)) = 31486,742Hz */
+		break;
+	case 44100:
+		cmr_div = 29;	/* BCLK = 133MHz/(2*29) = 2.293MHz */
+		period = 25;	/* LRC = BCLK/(2*(25+1)) = 44098Hz */
+		break;
+	case 48000:
+		cmr_div = 33;	/* BCLK = 133MHz/(2*33) = 2.015MHz */
+		period = 20;	/* LRC = BCLK/(2*(20+1)) = 47979,79Hz */
+		break;
+	case 88200:
+		cmr_div = 29;	/* BCLK = 133MHz/(2*29) = 2.293MHz */
+		period = 12;	/* LRC = BCLK/(2*(12+1)) = 88196Hz */
+		break;
+	case 96000:
+		cmr_div = 23;	/* BCLK = 133MHz/(2*23) = 2.891MHz */
+		period = 14;	/* LRC = BCLK/(2*(14+1)) = 96376Hz */
+		break;
+	default:
+		printk(KERN_WARNING "unsupported rate %d"
+				" on at91sam9g20ek board\n", rate);
+		return -EINVAL;
+	}
+
+	/* set the MCK divider for BCLK */
+	ret = snd_soc_dai_set_clkdiv(cpu_dai, ATMEL_SSC_CMR_DIV, cmr_div);
+	if (ret < 0)
+		return ret;
+
+	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
+		/* set the BCLK divider for DACLRC */
+		ret = snd_soc_dai_set_clkdiv(cpu_dai,
+						ATMEL_SSC_TCMR_PERIOD, period);
+	} else {
+		/* set the BCLK divider for ADCLRC */
+		ret = snd_soc_dai_set_clkdiv(cpu_dai,
+						ATMEL_SSC_RCMR_PERIOD, period);
+	}
+	if (ret < 0)
+		return ret;
+
+	return 0;
+}
+
+static struct snd_soc_ops at91sam9g20ek_ops = {
+	.startup = at91sam9g20ek_startup,
+	.hw_params = at91sam9g20ek_hw_params,
+	.shutdown = at91sam9g20ek_shutdown,
+};
+
+
+static const struct snd_soc_dapm_widget at91sam9g20ek_dapm_widgets[] = {
+	SND_SOC_DAPM_MIC("Int Mic", NULL),
+	SND_SOC_DAPM_SPK("Ext Spk", NULL),
+};
+
+static const struct snd_soc_dapm_route intercon[] = {
+
+	/* speaker connected to LHPOUT */
+	{"Ext Spk", NULL, "LHPOUT"},
+
+	/* mic is connected to Mic Jack, with WM8731 Mic Bias */
+	{"MICIN", NULL, "Mic Bias"},
+	{"Mic Bias", NULL, "Int Mic"},
+};
+
+/*
+ * Logic for a wm8731 as connected on a at91sam9g20ek board.
+ */
+static int at91sam9g20ek_wm8731_init(struct snd_soc_codec *codec)
+{
+	printk(KERN_DEBUG
+			"at91sam9g20ek_wm8731 "
+			": at91sam9g20ek_wm8731_init() called\n");
+
+	/* Add specific widgets */
+	snd_soc_dapm_new_controls(codec, at91sam9g20ek_dapm_widgets,
+				  ARRAY_SIZE(at91sam9g20ek_dapm_widgets));
+	/* Set up specific audio path interconnects */
+	snd_soc_dapm_add_routes(codec, intercon, ARRAY_SIZE(intercon));
+
+	/* not connected */
+	snd_soc_dapm_disable_pin(codec, "RLINEIN");
+	snd_soc_dapm_disable_pin(codec, "LLINEIN");
+
+	/* always connected */
+	snd_soc_dapm_enable_pin(codec, "Int Mic");
+	snd_soc_dapm_enable_pin(codec, "Ext Spk");
+
+	snd_soc_dapm_sync(codec);
+
+	return 0;
+}
+
+static struct snd_soc_dai_link at91sam9g20ek_dai = {
+	.name = "WM8731",
+	.stream_name = "WM8731 PCM",
+	.cpu_dai = &atmel_ssc_dai[0],
+	.codec_dai = &wm8731_dai,
+	.init = at91sam9g20ek_wm8731_init,
+	.ops = &at91sam9g20ek_ops,
+};
+
+static struct snd_soc_card snd_soc_at91sam9g20ek = {
+	.name = "WM8731",
+	.platform = &atmel_soc_platform,
+	.dai_link = &at91sam9g20ek_dai,
+	.num_links = 1,
+};
+
+static struct wm8731_setup_data at91sam9g20ek_wm8731_setup = {
+	.i2c_bus = 0,
+	.i2c_address = 0x1b,
+};
+
+static struct snd_soc_device at91sam9g20ek_snd_devdata = {
+	.card = &snd_soc_at91sam9g20ek,
+	.codec_dev = &soc_codec_dev_wm8731,
+	.codec_data = &at91sam9g20ek_wm8731_setup,
+};
+
+static struct platform_device *at91sam9g20ek_snd_device;
+
+static int __init at91sam9g20ek_init(void)
+{
+	struct atmel_ssc_info *ssc_p = at91sam9g20ek_dai.cpu_dai->private_data;
+	struct ssc_device *ssc = NULL;
+	int ret;
+
+	/*
+	 * Request SSC device
+	 */
+	ssc = ssc_request(0);
+	if (IS_ERR(ssc)) {
+		ret = PTR_ERR(ssc);
+		ssc = NULL;
+		goto err_ssc;
+	}
+	ssc_p->ssc = ssc;
+
+	at91sam9g20ek_snd_device = platform_device_alloc("soc-audio", -1);
+	if (!at91sam9g20ek_snd_device) {
+		printk(KERN_DEBUG
+				"platform device allocation failed\n");
+		ret = -ENOMEM;
+	}
+
+	platform_set_drvdata(at91sam9g20ek_snd_device,
+			&at91sam9g20ek_snd_devdata);
+	at91sam9g20ek_snd_devdata.dev = &at91sam9g20ek_snd_device->dev;
+
+	ret = platform_device_add(at91sam9g20ek_snd_device);
+	if (ret) {
+		printk(KERN_DEBUG
+				"platform device allocation failed\n");
+		platform_device_put(at91sam9g20ek_snd_device);
+	}
+
+	return ret;
+
+err_ssc:
+	return ret;
+}
+
+static void __exit at91sam9g20ek_exit(void)
+{
+	struct atmel_ssc_info *ssc_p = at91sam9g20ek_dai.cpu_dai->private_data;
+	struct ssc_device *ssc;
+
+	if (ssc_p != NULL) {
+		ssc = ssc_p->ssc;
+		if (ssc != NULL)
+			ssc_free(ssc);
+		ssc_p->ssc = NULL;
+	}
+
+	platform_device_unregister(at91sam9g20ek_snd_device);
+	at91sam9g20ek_snd_device = NULL;
+}
+
+module_init(at91sam9g20ek_init);
+module_exit(at91sam9g20ek_exit);
+
+/* Module information */
+MODULE_AUTHOR("Sedji Gaouaou <sedji.gaouaou@atmel.com>");
+MODULE_DESCRIPTION("ALSA SoC AT91SAM9G20EK_WM8731");
+MODULE_LICENSE("GPL");

sound/soc/au1x/dbdma2.c

@@ -406,11 +406,12 @@ static int __init au1xpsc_audio_dbdma_init(void)
 {
 	au1xpsc_audio_pcmdma[PCM_TX] = NULL;
 	au1xpsc_audio_pcmdma[PCM_RX] = NULL;
-	return 0;
+	return snd_soc_register_platform(&au1xpsc_soc_platform);
 }
 
 static void __exit au1xpsc_audio_dbdma_exit(void)
 {
+	snd_soc_unregister_platform(&au1xpsc_soc_platform);
 }
 
 module_init(au1xpsc_audio_dbdma_init);

sound/soc/au1x/psc-ac97.c

@@ -160,7 +160,8 @@ struct snd_ac97_bus_ops soc_ac97_ops = {
 EXPORT_SYMBOL_GPL(soc_ac97_ops);
 
 static int au1xpsc_ac97_hw_params(struct snd_pcm_substream *substream,
-				  struct snd_pcm_hw_params *params)
+				  struct snd_pcm_hw_params *params,
+				  struct snd_soc_dai *dai)
 {
 	/* FIXME */
 	struct au1xpsc_audio_data *pscdata = au1xpsc_ac97_workdata;
@@ -210,7 +211,7 @@ static int au1xpsc_ac97_hw_params(struct snd_pcm_substream *substream,
 }
 
 static int au1xpsc_ac97_trigger(struct snd_pcm_substream *substream,
-				int cmd)
+				int cmd, struct snd_soc_dai *dai)
 {
 	/* FIXME */
 	struct au1xpsc_audio_data *pscdata = au1xpsc_ac97_workdata;
@@ -313,8 +314,7 @@ static void au1xpsc_ac97_remove(struct platform_device *pdev,
 	au1xpsc_ac97_workdata = NULL;
 }
 
-static int au1xpsc_ac97_suspend(struct platform_device *pdev,
-				struct snd_soc_dai *dai)
+static int au1xpsc_ac97_suspend(struct snd_soc_dai *dai)
 {
 	/* save interesting registers and disable PSC */
 	au1xpsc_ac97_workdata->pm[0] =
@@ -328,8 +328,7 @@ static int au1xpsc_ac97_suspend(struct platform_device *pdev,
 	return 0;
 }
 
-static int au1xpsc_ac97_resume(struct platform_device *pdev,
-			       struct snd_soc_dai *dai)
+static int au1xpsc_ac97_resume(struct snd_soc_dai *dai)
 {
 	/* restore PSC clock config */
 	au_writel(au1xpsc_ac97_workdata->pm[0] | PSC_SEL_PS_AC97MODE,
@@ -345,7 +344,7 @@ static int au1xpsc_ac97_resume(struct platform_device *pdev,
 
 struct snd_soc_dai au1xpsc_ac97_dai = {
 	.name			= "au1xpsc_ac97",
-	.type			= SND_SOC_DAI_AC97,
+	.ac97_control		= 1,
 	.probe			= au1xpsc_ac97_probe,
 	.remove			= au1xpsc_ac97_remove,
 	.suspend		= au1xpsc_ac97_suspend,
@@ -372,11 +371,12 @@ EXPORT_SYMBOL_GPL(au1xpsc_ac97_dai);
 static int __init au1xpsc_ac97_init(void)
 {
 	au1xpsc_ac97_workdata = NULL;
-	return 0;
+	return snd_soc_register_dai(&au1xpsc_ac97_dai);
 }
 
 static void __exit au1xpsc_ac97_exit(void)
 {
+	snd_soc_unregister_dai(&au1xpsc_ac97_dai);
 }
 
 module_init(au1xpsc_ac97_init);

sound/soc/au1x/psc-i2s.c

@@ -116,7 +116,8 @@ static int au1xpsc_i2s_set_fmt(struct snd_soc_dai *cpu_dai,
 }
 
 static int au1xpsc_i2s_hw_params(struct snd_pcm_substream *substream,
-				 struct snd_pcm_hw_params *params)
+				 struct snd_pcm_hw_params *params,
+				 struct snd_soc_dai *dai)
 {
 	struct au1xpsc_audio_data *pscdata = au1xpsc_i2s_workdata;
 
@@ -240,7 +241,8 @@ static int au1xpsc_i2s_stop(struct au1xpsc_audio_data *pscdata, int stype)
 	return 0;
 }
 
-static int au1xpsc_i2s_trigger(struct snd_pcm_substream *substream, int cmd)
+static int au1xpsc_i2s_trigger(struct snd_pcm_substream *substream, int cmd,
+			       struct snd_soc_dai *dai)
 {
 	struct au1xpsc_audio_data *pscdata = au1xpsc_i2s_workdata;
 	int ret, stype = SUBSTREAM_TYPE(substream);
@@ -337,8 +339,7 @@ static void au1xpsc_i2s_remove(struct platform_device *pdev,
 	au1xpsc_i2s_workdata = NULL;
 }
 
-static int au1xpsc_i2s_suspend(struct platform_device *pdev,
-			       struct snd_soc_dai *cpu_dai)
+static int au1xpsc_i2s_suspend(struct snd_soc_dai *cpu_dai)
 {
 	/* save interesting register and disable PSC */
 	au1xpsc_i2s_workdata->pm[0] =
@@ -352,8 +353,7 @@ static int au1xpsc_i2s_suspend(struct platform_device *pdev,
 	return 0;
 }
 
-static int au1xpsc_i2s_resume(struct platform_device *pdev,
-			      struct snd_soc_dai *cpu_dai)
+static int au1xpsc_i2s_resume(struct snd_soc_dai *cpu_dai)
 {
 	/* select I2S mode and PSC clock */
 	au_writel(PSC_CTRL_DISABLE, PSC_CTRL(au1xpsc_i2s_workdata));
@@ -369,7 +369,6 @@ static int au1xpsc_i2s_resume(struct platform_device *pdev,
 
 struct snd_soc_dai au1xpsc_i2s_dai = {
 	.name			= "au1xpsc_i2s",
-	.type			= SND_SOC_DAI_I2S,
 	.probe			= au1xpsc_i2s_probe,
 	.remove			= au1xpsc_i2s_remove,
 	.suspend		= au1xpsc_i2s_suspend,
@@ -389,8 +388,6 @@ struct snd_soc_dai au1xpsc_i2s_dai = {
 	.ops = {
 		.trigger	= au1xpsc_i2s_trigger,
 		.hw_params	= au1xpsc_i2s_hw_params,
-	},
-	.dai_ops = {
 		.set_fmt	= au1xpsc_i2s_set_fmt,
 	},
 };
@@ -399,11 +396,12 @@ EXPORT_SYMBOL(au1xpsc_i2s_dai);
 static int __init au1xpsc_i2s_init(void)
 {
 	au1xpsc_i2s_workdata = NULL;
-	return 0;
+	return snd_soc_register_dai(&au1xpsc_i2s_dai);
 }
 
 static void __exit au1xpsc_i2s_exit(void)
 {
+	snd_soc_unregister_dai(&au1xpsc_i2s_dai);
 }
 
 module_init(au1xpsc_i2s_init);

sound/soc/au1x/sample-ac97.c

@@ -42,14 +42,14 @@ static struct snd_soc_dai_link au1xpsc_sample_ac97_dai = {
 	.ops		= NULL,
 };
 
-static struct snd_soc_machine au1xpsc_sample_ac97_machine = {
+static struct snd_soc_card au1xpsc_sample_ac97_machine = {
 	.name		= "Au1xxx PSC AC97 Audio",
 	.dai_link	= &au1xpsc_sample_ac97_dai,
 	.num_links	= 1,
 };
 
 static struct snd_soc_device au1xpsc_sample_ac97_devdata = {
-	.machine	= &au1xpsc_sample_ac97_machine,
+	.card		= &au1xpsc_sample_ac97_machine,
 	.platform	= &au1xpsc_soc_platform, /* see dbdma2.c */
 	.codec_dev	= &soc_codec_dev_ac97,
 };

sound/soc/blackfin/Kconfig

 config SND_BF5XX_I2S
 	tristate "SoC I2S Audio for the ADI BF5xx chip"
-	depends on BLACKFIN && SND_SOC
+	depends on BLACKFIN
 	help
 	  Say Y or M if you want to add support for codecs attached to
 	  the Blackfin SPORT (synchronous serial ports) interface in I2S
@@ -13,7 +13,6 @@ config SND_BF5XX_SOC_SSM2602
 	select SND_BF5XX_SOC_I2S
 	select SND_SOC_SSM2602
 	select I2C
-	select I2C_BLACKFIN_TWI
 	help
 	  Say Y if you want to add support for SoC audio on BF527-EZKIT.
 
@@ -35,7 +34,7 @@ config SND_BFIN_AD73311_SE
 
 config SND_BF5XX_AC97
 	tristate "SoC AC97 Audio for the ADI BF5xx chip"
-	depends on BLACKFIN && SND_SOC
+	depends on BLACKFIN
 	help
 	  Say Y or M if you want to add support for codecs attached to
 	  the Blackfin SPORT (synchronous serial ports) interface in slot 16
@@ -47,7 +46,7 @@ config SND_BF5XX_AC97
 	  properly with this driver. This driver is known to work with the
 	  Analog Devices line of AC97 codecs.
 
-config SND_MMAP_SUPPORT
+config SND_BF5XX_MMAP_SUPPORT
 	bool "Enable MMAP Support"
 	depends on SND_BF5XX_AC97
 	default y
@@ -55,9 +54,17 @@ config SND_MMAP_SUPPORT
 	  Say y if you want AC97 driver to support mmap mode.
 	  We introduce an intermediate buffer to simulate mmap.
 
+config SND_BF5XX_MULTICHAN_SUPPORT
+	bool "Enable Multichannel Support"
+	depends on SND_BF5XX_AC97
+	default n
+	help
+	  Say y if you want AC97 driver to support up to 5.1 channel audio.
+	  this mode will consume much more memory for DMA.
+
 config SND_BF5XX_SOC_SPORT
 	tristate
-	
+
 config SND_BF5XX_SOC_I2S
 	tristate
 	select SND_BF5XX_SOC_SPORT
@@ -80,7 +87,7 @@ config SND_BF5XX_SPORT_NUM
 	int "Set a SPORT for Sound chip"
 	depends on (SND_BF5XX_I2S || SND_BF5XX_AC97)
 	range 0 3 if BF54x
-	range 0 1 if (BF53x || BF561)
+	range 0 1 if !BF54x
 	default 0
 	help
 	  Set the correct SPORT for sound chip.
@@ -90,12 +97,13 @@ config SND_BF5XX_HAVE_COLD_RESET
 	depends on SND_BF5XX_AC97
 	default y if BFIN548_EZKIT
 	default n if !BFIN548_EZKIT
-	
+
 config SND_BF5XX_RESET_GPIO_NUM
 	int "Set a GPIO for cold reset"
 	depends on SND_BF5XX_HAVE_COLD_RESET
 	range 0 159
 	default 19 if BFIN548_EZKIT
 	default 5 if BFIN537_STAMP
+	default 0
 	help
 	  Set the correct GPIO for RESET the sound chip.

sound/soc/blackfin/bf5xx-ac97.h

@@ -16,21 +16,46 @@ struct ac97_frame {
 	u16 ac97_tag;		/* slot 0 */
 	u16 ac97_addr;		/* slot 1 */
 	u16 ac97_data;		/* slot 2 */
-	u32 ac97_pcm;		/* slot 3 and 4: left and right pcm data */
+	u16 ac97_pcm_l;		/*slot 3:front left*/
+	u16 ac97_pcm_r;		/*slot 4:front left*/
+#if defined(CONFIG_SND_BF5XX_MULTICHAN_SUPPORT)
+	u16 ac97_mdm_l1;
+	u16 ac97_center;	/*slot 6:center*/
+	u16 ac97_sl;		/*slot 7:surround left*/
+	u16 ac97_sr;		/*slot 8:surround right*/
+	u16 ac97_lfe;		/*slot 9:lfe*/
+#endif
 } __attribute__ ((packed));
 
+/* Speaker location */
+#define SP_FL		0x0001
+#define SP_FR		0x0010
+#define SP_FC		0x0002
+#define SP_LFE		0x0020
+#define SP_SL		0x0004
+#define SP_SR		0x0040
+
+#define SP_STEREO	(SP_FL | SP_FR)
+#define SP_2DOT1	(SP_FL | SP_FR | SP_LFE)
+#define SP_QUAD		(SP_FL | SP_FR | SP_SL | SP_SR)
+#define SP_5DOT1	(SP_FL | SP_FR | SP_FC | SP_LFE | SP_SL | SP_SR)
+
 #define TAG_VALID		0x8000
 #define TAG_CMD			0x6000
 #define TAG_PCM_LEFT		0x1000
 #define TAG_PCM_RIGHT		0x0800
-#define TAG_PCM			(TAG_PCM_LEFT | TAG_PCM_RIGHT)
+#define TAG_PCM_MDM_L1		0x0400
+#define TAG_PCM_CENTER		0x0200
+#define TAG_PCM_SL		0x0100
+#define TAG_PCM_SR		0x0080
+#define TAG_PCM_LFE		0x0040
 
 extern struct snd_soc_dai bfin_ac97_dai;
 
-void bf5xx_pcm_to_ac97(struct ac97_frame *dst, const __u32 *src, \
-		size_t count);
+void bf5xx_pcm_to_ac97(struct ac97_frame *dst, const __u16 *src, \
+		size_t count, unsigned int chan_mask);
 
-void bf5xx_ac97_to_pcm(const struct ac97_frame *src, __u32 *dst, \
+void bf5xx_ac97_to_pcm(const struct ac97_frame *src, __u16 *dst, \
 		size_t count);
 
 #endif

sound/soc/blackfin/bf5xx-ad1980.c

@@ -43,7 +43,7 @@
 #include "bf5xx-ac97-pcm.h"
 #include "bf5xx-ac97.h"
 
-static struct snd_soc_machine bf5xx_board;
+static struct snd_soc_card bf5xx_board;
 
 static int bf5xx_board_startup(struct snd_pcm_substream *substream)
 {
@@ -67,15 +67,15 @@ static struct snd_soc_dai_link bf5xx_board_dai = {
 	.ops = &bf5xx_board_ops,
 };
 
-static struct snd_soc_machine bf5xx_board = {
+static struct snd_soc_card bf5xx_board = {
 	.name = "bf5xx-board",
+	.platform = &bf5xx_ac97_soc_platform,
 	.dai_link = &bf5xx_board_dai,
 	.num_links = 1,
 };
 
 static struct snd_soc_device bf5xx_board_snd_devdata = {
-	.machine = &bf5xx_board,
-	.platform = &bf5xx_ac97_soc_platform,
+	.card = &bf5xx_board,
 	.codec_dev = &soc_codec_dev_ad1980,
 };
 

sound/soc/blackfin/bf5xx-ad73311.c

@@ -65,7 +65,7 @@
 
 #define GPIO_SE CONFIG_SND_BFIN_AD73311_SE
 
-static struct snd_soc_machine bf5xx_ad73311;
+static struct snd_soc_card bf5xx_ad73311;
 
 static int snd_ad73311_startup(void)
 {
@@ -168,7 +168,7 @@ static int bf5xx_ad73311_hw_params(struct snd_pcm_substream *substream,
 		params_format(params));
 
 	/* set cpu DAI configuration */
-	ret = cpu_dai->dai_ops.set_fmt(cpu_dai, SND_SOC_DAIFMT_DSP_A |
+	ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_DSP_A |
 		SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM);
 	if (ret < 0)
 		return ret;
@@ -190,16 +190,16 @@ static struct snd_soc_dai_link bf5xx_ad73311_dai = {
 	.ops = &bf5xx_ad73311_ops,
 };
 
-static struct snd_soc_machine bf5xx_ad73311 = {
+static struct snd_soc_card bf5xx_ad73311 = {
 	.name = "bf5xx_ad73311",
+	.platform = &bf5xx_i2s_soc_platform,
 	.probe = bf5xx_probe,
 	.dai_link = &bf5xx_ad73311_dai,
 	.num_links = 1,
 };
 
 static struct snd_soc_device bf5xx_ad73311_snd_devdata = {
-	.machine = &bf5xx_ad73311,
-	.platform = &bf5xx_i2s_soc_platform,
+	.card = &bf5xx_ad73311,
 	.codec_dev = &soc_codec_dev_ad73311,
 };
 

sound/soc/blackfin/bf5xx-i2s-pcm.c

@@ -283,6 +283,18 @@ struct snd_soc_platform bf5xx_i2s_soc_platform = {
 };
 EXPORT_SYMBOL_GPL(bf5xx_i2s_soc_platform);
 
+static int __init bfin_i2s_init(void)
+{
+	return snd_soc_register_platform(&bf5xx_i2s_soc_platform);
+}
+module_init(bfin_i2s_init);
+
+static void __exit bfin_i2s_exit(void)
+{
+	snd_soc_unregister_platform(&bf5xx_i2s_soc_platform);
+}
+module_exit(bfin_i2s_exit);
+
 MODULE_AUTHOR("Cliff Cai");
 MODULE_DESCRIPTION("ADI Blackfin I2S PCM DMA module");
 MODULE_LICENSE("GPL");

sound/soc/blackfin/bf5xx-sport.h

@@ -116,7 +116,7 @@ struct sport_device {
 	void *err_data;
 	unsigned char *tx_dma_buf;
 	unsigned char *rx_dma_buf;
-#ifdef CONFIG_SND_MMAP_SUPPORT
+#ifdef CONFIG_SND_BF5XX_MMAP_SUPPORT
 	dma_addr_t tx_dma_phy;
 	dma_addr_t rx_dma_phy;
 	int tx_pos;/*pcm sample count*/

sound/soc/blackfin/bf5xx-ssm2602.c

@@ -44,7 +44,7 @@
 #include "bf5xx-i2s-pcm.h"
 #include "bf5xx-i2s.h"
 
-static struct snd_soc_machine bf5xx_ssm2602;
+static struct snd_soc_card bf5xx_ssm2602;
 
 static int bf5xx_ssm2602_startup(struct snd_pcm_substream *substream)
 {
@@ -92,17 +92,17 @@ static int bf5xx_ssm2602_hw_params(struct snd_pcm_substream *substream,
 	 */
 
 	/* set codec DAI configuration */
-	ret = codec_dai->dai_ops.set_fmt(codec_dai, SND_SOC_DAIFMT_I2S |
+	ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S |
 		SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM);
 	if (ret < 0)
 		return ret;
 	/* set cpu DAI configuration */
-	ret = cpu_dai->dai_ops.set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S |
+	ret = snd_soc_dai_set_fmt(cpu_dai, SND_SOC_DAIFMT_I2S |
 		SND_SOC_DAIFMT_NB_NF | SND_SOC_DAIFMT_CBM_CFM);
 	if (ret < 0)
 		return ret;
 
-	ret = codec_dai->dai_ops.set_sysclk(codec_dai, SSM2602_SYSCLK, clk,
+	ret = snd_soc_dai_set_sysclk(codec_dai, SSM2602_SYSCLK, clk,
 		SND_SOC_CLOCK_IN);
 	if (ret < 0)
 		return ret;
@@ -135,15 +135,15 @@ static struct ssm2602_setup_data bf5xx_ssm2602_setup = {
 	.i2c_address = 0x1b,
 };
 
-static struct snd_soc_machine bf5xx_ssm2602 = {
+static struct snd_soc_card bf5xx_ssm2602 = {
 	.name = "bf5xx_ssm2602",
+	.platform = &bf5xx_i2s_soc_platform,
 	.dai_link = &bf5xx_ssm2602_dai,
 	.num_links = 1,
 };
 
 static struct snd_soc_device bf5xx_ssm2602_snd_devdata = {
-	.machine = &bf5xx_ssm2602,
-	.platform = &bf5xx_i2s_soc_platform,
+	.card = &bf5xx_ssm2602,
 	.codec_dev = &soc_codec_dev_ssm2602,
 	.codec_data = &bf5xx_ssm2602_setup,
 };

sound/soc/codecs/wm8350.h

+/*
+ * wm8350.h - WM8903 audio codec interface
+ *
+ * Copyright 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
+ *  Free Software Foundation;  either version 2 of the  License, or (at your
+ *  option) any later version.
+ */
+
+#ifndef _WM8350_H
+#define _WM8350_H
+
+#include <sound/soc.h>
+
+extern struct snd_soc_dai wm8350_dai;
+extern struct snd_soc_codec_device soc_codec_dev_wm8350;
+
+#endif