Merge branch 'topic/asoc' into for-linus

* topic/asoc: (226 commits)
  ASoC: au1x: PSC-AC97 bugfixes
  ASoC: Fix WM835x Out4 capture enumeration
  ASoC: Remove unuused hw_read_t
  ASoC: fix pxa2xx-ac97.c breakage
  ASoC: Fully specify DC servo bits to update in wm_hubs
  ASoC: Debugged improper setting of PLL fields in WM8580 driver
  ASoC: new board driver to connect bfin-5xx with ad1836 codec
  ASoC: OMAP: Add functionality to set CLKR and FSR sources in McBSP DAI
  ASoC: davinci: i2c device creation moved into board files
  ASoC: Don't reconfigure WM8350 FLL if not needed
  ASoC: Fix s3c-i2s-v2 build
  ASoC: Make platform data optional for TLV320AIC3x
  ASoC: Add S3C24xx dependencies for Simtec machines
  ASoC: SDP3430: Fix TWL GPIO6 pin mux request
  ASoC: S3C platform: Fix s3c2410_dma_started() called at improper time
  ARM: OMAP: McBSP: Merge two functions into omap_mcbsp_start/_stop
  ASoC: OMAP: Fix setup of XCCR and RCCR registers in McBSP DAI
  OMAP: McBSP: Use textual values in DMA operating mode sysfs files
  ARM: OMAP: DMA: Add support for DMA channel self linking on OMAP1510
  ASoC: Select core DMA when building for S3C64xx
  ...

8 files changed, 238 insertions, 23 deletions

diff --git a/include/linux/tty.h b/include/linux/tty.h
index e8c6c9136c97..0d3974f59c53 100644
--- a/include/linux/tty.h
+++ b/include/linux/tty.h
@@ -23,7 +23,7 @@
  */
 #define NR_UNIX98_PTY_DEFAULT	4096      /* Default maximum for Unix98 ptys */
 #define NR_UNIX98_PTY_MAX	(1 << MINORBITS) /* Absolute limit */
-#define NR_LDISCS		19
+#define NR_LDISCS		20
 
 /* line disciplines */
 #define N_TTY		0
@@ -47,6 +47,8 @@
 #define N_SLCAN		17	/* Serial / USB serial CAN Adaptors */
 #define N_PPS		18	/* Pulse per Second */
 
+#define N_V253		19	/* Codec control over voice modem */
+
 /*
  * This character is the same as _POSIX_VDISABLE: it cannot be used as
  * a c_cc[] character, but indicates that a particular special character
diff --git a/include/sound/ac97_codec.h b/include/sound/ac97_codec.h
index 251fc1cd5002..3dae3f799b9b 100644
--- a/include/sound/ac97_codec.h
+++ b/include/sound/ac97_codec.h
@@ -32,6 +32,9 @@
 #include "control.h"
 #include "info.h"
 
+/* maximum number of devices on the AC97 bus */
+#define	AC97_BUS_MAX_DEVICES	4
+
 /*
  *  AC'97 codec registers
  */
@@ -642,4 +645,10 @@ int snd_ac97_pcm_double_rate_rules(struct snd_pcm_runtime *runtime);
 /* ad hoc AC97 device driver access */
 extern struct bus_type ac97_bus_type;
 
+/* AC97 platform_data adding function */
+static inline void snd_ac97_dev_add_pdata(struct snd_ac97 *ac97, void *data)
+{
+	ac97->dev.platform_data = data;
+}
+
 #endif /* __SOUND_AC97_CODEC_H */
diff --git a/include/sound/sh_fsi.h b/include/sound/sh_fsi.h
new file mode 100644
index 000000000000..c0227361a876
--- /dev/null
+++ b/include/sound/sh_fsi.h
@@ -0,0 +1,83 @@
+#ifndef __SOUND_FSI_H
+#define __SOUND_FSI_H
+
+/*
+ * Fifo-attached Serial Interface (FSI) support for SH7724
+ *
+ * Copyright (C) 2009 Renesas Solutions Corp.
+ * Kuninori Morimoto <morimoto.kuninori@renesas.com>
+ *
+ * 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.
+ */
+
+/* flags format
+
+ * 0xABCDEEFF
+ *
+ * A:  channel size for TDM (input)
+ * B:  channel size for TDM (ooutput)
+ * C:  inversion
+ * D:  mode
+ * E:  input format
+ * F:  output format
+ */
+
+#include <linux/clk.h>
+#include <sound/soc.h>
+
+/* TDM channel */
+#define SH_FSI_SET_CH_I(x)	((x & 0xF) << 28)
+#define SH_FSI_SET_CH_O(x)	((x & 0xF) << 24)
+
+#define SH_FSI_CH_IMASK		0xF0000000
+#define SH_FSI_CH_OMASK		0x0F000000
+#define SH_FSI_GET_CH_I(x)	((x & SH_FSI_CH_IMASK) >> 28)
+#define SH_FSI_GET_CH_O(x)	((x & SH_FSI_CH_OMASK) >> 24)
+
+/* clock inversion */
+#define SH_FSI_INVERSION_MASK	0x00F00000
+#define SH_FSI_LRM_INV		(1 << 20)
+#define SH_FSI_BRM_INV		(1 << 21)
+#define SH_FSI_LRS_INV		(1 << 22)
+#define SH_FSI_BRS_INV		(1 << 23)
+
+/* mode */
+#define SH_FSI_MODE_MASK	0x000F0000
+#define SH_FSI_IN_SLAVE_MODE	(1 << 16)  /* default master mode */
+#define SH_FSI_OUT_SLAVE_MODE	(1 << 17)  /* default master mode */
+
+/* DI format */
+#define SH_FSI_FMT_MASK		0x000000FF
+#define SH_FSI_IFMT(x)		(((SH_FSI_FMT_ ## x) & SH_FSI_FMT_MASK) << 8)
+#define SH_FSI_OFMT(x)		(((SH_FSI_FMT_ ## x) & SH_FSI_FMT_MASK) << 0)
+#define SH_FSI_GET_IFMT(x)	((x >> 8) & SH_FSI_FMT_MASK)
+#define SH_FSI_GET_OFMT(x)	((x >> 0) & SH_FSI_FMT_MASK)
+
+#define SH_FSI_FMT_MONO		(1 << 0)
+#define SH_FSI_FMT_MONO_DELAY	(1 << 1)
+#define SH_FSI_FMT_PCM		(1 << 2)
+#define SH_FSI_FMT_I2S		(1 << 3)
+#define SH_FSI_FMT_TDM		(1 << 4)
+#define SH_FSI_FMT_TDM_DELAY	(1 << 5)
+
+#define SH_FSI_IFMT_TDM_CH(x) \
+	(SH_FSI_IFMT(TDM)	| SH_FSI_SET_CH_I(x))
+#define SH_FSI_IFMT_TDM_DELAY_CH(x) \
+	(SH_FSI_IFMT(TDM_DELAY)	| SH_FSI_SET_CH_I(x))
+
+#define SH_FSI_OFMT_TDM_CH(x) \
+	(SH_FSI_OFMT(TDM)	| SH_FSI_SET_CH_O(x))
+#define SH_FSI_OFMT_TDM_DELAY_CH(x) \
+	(SH_FSI_OFMT(TDM_DELAY)	| SH_FSI_SET_CH_O(x))
+
+struct sh_fsi_platform_info {
+	unsigned long porta_flags;
+	unsigned long portb_flags;
+};
+
+extern struct snd_soc_dai fsi_soc_dai[2];
+extern struct snd_soc_platform fsi_soc_platform;
+
+#endif /* __SOUND_FSI_H */
diff --git a/include/sound/soc-dai.h b/include/sound/soc-dai.h
index 352d7eee9b6d..97ca9af414dc 100644
--- a/include/sound/soc-dai.h
+++ b/include/sound/soc-dai.h
@@ -27,8 +27,8 @@ struct snd_pcm_substream;
 #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_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 */
@@ -38,7 +38,7 @@ struct snd_pcm_substream;
 /*
  * DAI Clock gating.
  *
- * DAI bit clocks can be be gated (disabled) when not the DAI is not
+ * DAI bit clocks can be be gated (disabled) when 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 */
@@ -51,21 +51,21 @@ struct snd_pcm_substream;
  * 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 */
+#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
+ * 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_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_CBS_CFS		(3 << 12) /* codec clk & FRM slave */
 
 #define SND_SOC_DAIFMT_FORMAT_MASK	0x000f
 #define SND_SOC_DAIFMT_CLOCK_MASK	0x00f0
@@ -78,7 +78,13 @@ struct snd_pcm_substream;
 #define SND_SOC_CLOCK_IN		0
 #define SND_SOC_CLOCK_OUT		1
 
-#define SND_SOC_STD_AC97_FMTS (SNDRV_PCM_FMTBIT_S16_LE |\
+#define SND_SOC_STD_AC97_FMTS (SNDRV_PCM_FMTBIT_S8 |\
+			       SNDRV_PCM_FMTBIT_S16_LE |\
+			       SNDRV_PCM_FMTBIT_S16_BE |\
+			       SNDRV_PCM_FMTBIT_S20_3LE |\
+			       SNDRV_PCM_FMTBIT_S20_3BE |\
+			       SNDRV_PCM_FMTBIT_S24_3LE |\
+			       SNDRV_PCM_FMTBIT_S24_3BE |\
                                SNDRV_PCM_FMTBIT_S32_LE |\
                                SNDRV_PCM_FMTBIT_S32_BE)
 
@@ -106,7 +112,7 @@ int snd_soc_dai_set_pll(struct snd_soc_dai *dai,
 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);
+	unsigned int tx_mask, unsigned int rx_mask, int slots, int slot_width);
 
 int snd_soc_dai_set_tristate(struct snd_soc_dai *dai, int tristate);
 
@@ -116,12 +122,12 @@ 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
+ * Describes the Digital Audio Interface in terms of its ALSA, DAI and AC97
+ * operations and capabilities. Codec and platform drivers will register this
  * structure for every DAI they have.
  *
  * This structure covers the clocking, formating and ALSA operations for each
- * interface a
+ * interface.
  */
 struct snd_soc_dai_ops {
 	/*
@@ -140,7 +146,8 @@ struct snd_soc_dai_ops {
 	 */
 	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);
+		unsigned int tx_mask, unsigned int rx_mask,
+		int slots, int slot_width);
 	int (*set_tristate)(struct snd_soc_dai *dai, int tristate);
 
 	/*
@@ -179,6 +186,7 @@ struct snd_soc_dai {
 	int ac97_control;
 
 	struct device *dev;
+	void *ac97_pdata;	/* platform_data for the ac97 codec */
 
 	/* DAI callbacks */
 	int (*probe)(struct platform_device *pdev,
diff --git a/include/sound/soc-dapm.h b/include/sound/soc-dapm.h
index ec8a45f9a069..c1410e3191e3 100644
--- a/include/sound/soc-dapm.h
+++ b/include/sound/soc-dapm.h
@@ -137,6 +137,12 @@
 	.event_flags = SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_POST_PMD}
 
 /* stream domain */
+#define SND_SOC_DAPM_AIF_IN(wname, stname, wslot, wreg, wshift, winvert) \
+{	.id = snd_soc_dapm_aif_in, .name = wname, .sname = stname, \
+	.reg = wreg, .shift = wshift, .invert = winvert }
+#define SND_SOC_DAPM_AIF_OUT(wname, stname, wslot, wreg, wshift, winvert) \
+{	.id = snd_soc_dapm_aif_out, .name = wname, .sname = stname, \
+	.reg = wreg, .shift = wshift, .invert = winvert }
 #define SND_SOC_DAPM_DAC(wname, stname, wreg, wshift, winvert) \
 {	.id = snd_soc_dapm_dac, .name = wname, .sname = stname, .reg = wreg, \
 	.shift = wshift, .invert = winvert}
@@ -279,9 +285,11 @@ int snd_soc_dapm_add_routes(struct snd_soc_codec *codec,
 /* dapm events */
 int snd_soc_dapm_stream_event(struct snd_soc_codec *codec, char *stream,
 	int event);
+void snd_soc_dapm_shutdown(struct snd_soc_device *socdev);
 
 /* dapm sys fs - used by the core */
 int snd_soc_dapm_sys_add(struct device *dev);
+void snd_soc_dapm_debugfs_init(struct snd_soc_codec *codec);
 
 /* dapm audio pin control and status */
 int snd_soc_dapm_enable_pin(struct snd_soc_codec *codec, const char *pin);
@@ -311,6 +319,8 @@ enum snd_soc_dapm_type {
 	snd_soc_dapm_pre,			/* machine specific pre widget - exec first */
 	snd_soc_dapm_post,			/* machine specific post widget - exec last */
 	snd_soc_dapm_supply,		/* power/clock supply */
+	snd_soc_dapm_aif_in,		/* audio interface input */
+	snd_soc_dapm_aif_out,		/* audio interface output */
 };
 
 /*
diff --git a/include/sound/soc.h b/include/sound/soc.h
index cf6111d72b17..475cb7ed6bec 100644
--- a/include/sound/soc.h
+++ b/include/sound/soc.h
@@ -135,6 +135,28 @@
 	.info = snd_soc_info_volsw, \
 	.get = xhandler_get, .put = xhandler_put, \
 	.private_value = SOC_SINGLE_VALUE(xreg, xshift, xmax, xinvert) }
+#define SOC_DOUBLE_EXT_TLV(xname, xreg, shift_left, shift_right, xmax, xinvert,\
+	 xhandler_get, xhandler_put, tlv_array) \
+{	.iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname), \
+	.access = SNDRV_CTL_ELEM_ACCESS_TLV_READ | \
+		 SNDRV_CTL_ELEM_ACCESS_READWRITE, \
+	.tlv.p = (tlv_array), \
+	.info = snd_soc_info_volsw, \
+	.get = xhandler_get, .put = xhandler_put, \
+	.private_value = (unsigned long)&(struct soc_mixer_control) \
+		{.reg = xreg, .shift = shift_left, .rshift = shift_right, \
+		.max = xmax, .invert = xinvert} }
+#define SOC_DOUBLE_R_EXT_TLV(xname, reg_left, reg_right, xshift, xmax, xinvert,\
+	 xhandler_get, xhandler_put, tlv_array) \
+{	.iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = (xname), \
+	.access = SNDRV_CTL_ELEM_ACCESS_TLV_READ | \
+		 SNDRV_CTL_ELEM_ACCESS_READWRITE, \
+	.tlv.p = (tlv_array), \
+	.info = snd_soc_info_volsw_2r, \
+	.get = xhandler_get, .put = xhandler_put, \
+	.private_value = (unsigned long)&(struct soc_mixer_control) \
+		{.reg = reg_left, .rreg = reg_right, .shift = xshift, \
+		.max = xmax, .invert = xinvert} }
 #define SOC_SINGLE_BOOL_EXT(xname, xdata, xhandler_get, xhandler_put) \
 {	.iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \
 	.info = snd_soc_info_bool_ext, \
@@ -183,14 +205,28 @@ struct snd_soc_jack_gpio;
 #endif
 
 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;
 
+enum snd_soc_control_type {
+	SND_SOC_CUSTOM,
+	SND_SOC_I2C,
+	SND_SOC_SPI,
+};
+
 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);
+int snd_soc_codec_volatile_register(struct snd_soc_codec *codec, int reg);
+int snd_soc_codec_set_cache_io(struct snd_soc_codec *codec,
+			       int addr_bits, int data_bits,
+			       enum snd_soc_control_type control);
+
+#ifdef CONFIG_PM
+int snd_soc_suspend_device(struct device *dev);
+int snd_soc_resume_device(struct device *dev);
+#endif
 
 /* pcm <-> DAI connect */
 void snd_soc_free_pcms(struct snd_soc_device *socdev);
@@ -216,9 +252,9 @@ void snd_soc_jack_free_gpios(struct snd_soc_jack *jack, int count,
 
 /* codec register bit access */
 int snd_soc_update_bits(struct snd_soc_codec *codec, unsigned short reg,
-				unsigned short mask, unsigned short value);
+				unsigned int mask, unsigned int value);
 int snd_soc_test_bits(struct snd_soc_codec *codec, unsigned short reg,
-				unsigned short mask, unsigned short value);
+				unsigned int mask, unsigned int value);
 
 int snd_soc_new_ac97_codec(struct snd_soc_codec *codec,
 	struct snd_ac97_bus_ops *ops, int num);
@@ -356,8 +392,10 @@ struct snd_soc_codec {
 	int (*write)(struct snd_soc_codec *, unsigned int, unsigned int);
 	int (*display_register)(struct snd_soc_codec *, char *,
 				size_t, unsigned int);
+	int (*volatile_register)(unsigned int);
+	int (*readable_register)(unsigned int);
 	hw_write_t hw_write;
-	hw_read_t hw_read;
+	unsigned int (*hw_read)(struct snd_soc_codec *, unsigned int);
 	void *reg_cache;
 	short reg_cache_size;
 	short reg_cache_step;
@@ -369,8 +407,6 @@ struct snd_soc_codec {
 	enum snd_soc_bias_level bias_level;
 	enum snd_soc_bias_level suspend_bias_level;
 	struct delayed_work delayed_work;
-	struct list_head up_list;
-	struct list_head down_list;
 
 	/* codec DAI's */
 	struct snd_soc_dai *dai;
@@ -379,6 +415,7 @@ struct snd_soc_codec {
 #ifdef CONFIG_DEBUG_FS
 	struct dentry *debugfs_reg;
 	struct dentry *debugfs_pop_time;
+	struct dentry *debugfs_dapm;
 #endif
 };
 
diff --git a/include/sound/uda1380.h b/include/sound/uda1380.h
new file mode 100644
index 000000000000..381319c7000c
--- /dev/null
+++ b/include/sound/uda1380.h
@@ -0,0 +1,22 @@
+/*
+ * UDA1380 ALSA SoC Codec driver
+ *
+ * Copyright 2009 Philipp Zabel
+ *
+ * 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 __UDA1380_H
+#define __UDA1380_H
+
+struct uda1380_platform_data {
+	int gpio_power;
+	int gpio_reset;
+	int dac_clk;
+#define UDA1380_DAC_CLK_SYSCLK 0
+#define UDA1380_DAC_CLK_WSPLL  1
+};
+
+#endif /* __UDA1380_H */
diff --git a/include/sound/wm8993.h b/include/sound/wm8993.h
new file mode 100644
index 000000000000..9c661f2f8cda
--- /dev/null
+++ b/include/sound/wm8993.h
@@ -0,0 +1,44 @@
+/*
+ * linux/sound/wm8993.h -- Platform data for WM8993
+ *
+ * Copyright 2009 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.
+ */
+
+#ifndef __LINUX_SND_WM8993_H
+#define __LINUX_SND_WM8993_H
+
+/* Note that EQ1 only contains the enable/disable bit so will be
+   ignored but is included for simplicity.
+ */
+struct wm8993_retune_mobile_setting {
+	const char *name;
+	unsigned int rate;
+	u16 config[24];
+};
+
+struct wm8993_platform_data {
+	struct wm8993_retune_mobile_setting *retune_configs;
+	int num_retune_configs;
+
+	/* LINEOUT can be differential or single ended */
+	unsigned int lineout1_diff:1;
+	unsigned int lineout2_diff:1;
+
+	/* Common mode feedback */
+	unsigned int lineout1fb:1;
+	unsigned int lineout2fb:1;
+
+	/* Microphone biases: 0=0.9*AVDD1 1=0.65*AVVD1 */
+	unsigned int micbias1_lvl:1;
+	unsigned int micbias2_lvl:1;
+
+	/* Jack detect threashold levels, see datasheet for values */
+	unsigned int jd_scthr:2;
+	unsigned int jd_thr:2;
+};
+
+#endif