Merge branch 'master' of master.kernel.org:/pub/scm/linux/kernel/git/torvalds/linux-2.6

Conflicts:
	Documentation/feature-removal-schedule.txt
	drivers/scsi/fcoe/fcoe.c
	net/core/drop_monitor.c
	net/core/net-traces.c

35 files changed, 7392 insertions, 388 deletions

diff --git a/sound/soc/codecs/Kconfig b/sound/soc/codecs/Kconfig
index b6c7f7a01cb..bbc97fd7664 100644
--- a/sound/soc/codecs/Kconfig
+++ b/sound/soc/codecs/Kconfig
@@ -18,7 +18,9 @@ config SND_SOC_ALL_CODECS
 	select SND_SOC_AK4535 if I2C
 	select SND_SOC_CS4270 if I2C
 	select SND_SOC_PCM3008
+	select SND_SOC_SPDIF
 	select SND_SOC_SSM2602 if I2C
+	select SND_SOC_STAC9766 if SND_SOC_AC97_BUS
 	select SND_SOC_TLV320AIC23 if I2C
 	select SND_SOC_TLV320AIC26 if SPI_MASTER
 	select SND_SOC_TLV320AIC3X if I2C
@@ -35,8 +37,12 @@ config SND_SOC_ALL_CODECS
 	select SND_SOC_WM8753 if SND_SOC_I2C_AND_SPI
 	select SND_SOC_WM8900 if I2C
 	select SND_SOC_WM8903 if I2C
+	select SND_SOC_WM8940 if I2C
+	select SND_SOC_WM8960 if I2C
 	select SND_SOC_WM8971 if I2C
+	select SND_SOC_WM8988 if SND_SOC_I2C_AND_SPI
 	select SND_SOC_WM8990 if I2C
+	select SND_SOC_WM9081 if I2C
 	select SND_SOC_WM9705 if SND_SOC_AC97_BUS
 	select SND_SOC_WM9712 if SND_SOC_AC97_BUS
 	select SND_SOC_WM9713 if SND_SOC_AC97_BUS
@@ -86,9 +92,15 @@ config SND_SOC_L3
 config SND_SOC_PCM3008
        tristate
 
+config SND_SOC_SPDIF
+	tristate
+
 config SND_SOC_SSM2602
 	tristate
 
+config SND_SOC_STAC9766
+	tristate
+
 config SND_SOC_TLV320AIC23
 	tristate
 
@@ -138,12 +150,24 @@ config SND_SOC_WM8900
 config SND_SOC_WM8903
 	tristate
 
+config SND_SOC_WM8940
+        tristate
+
+config SND_SOC_WM8960
+	tristate
+
 config SND_SOC_WM8971
 	tristate
 
+config SND_SOC_WM8988
+	tristate
+
 config SND_SOC_WM8990
 	tristate
 
+config SND_SOC_WM9081
+	tristate
+
 config SND_SOC_WM9705
 	tristate
 
diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile
index f2653803ede..8b7530546f4 100644
--- a/sound/soc/codecs/Makefile
+++ b/sound/soc/codecs/Makefile
@@ -6,7 +6,9 @@ snd-soc-ak4535-objs := ak4535.o
 snd-soc-cs4270-objs := cs4270.o
 snd-soc-l3-objs := l3.o
 snd-soc-pcm3008-objs := pcm3008.o
+snd-soc-spdif-objs := spdif_transciever.o
 snd-soc-ssm2602-objs := ssm2602.o
+snd-soc-stac9766-objs := stac9766.o
 snd-soc-tlv320aic23-objs := tlv320aic23.o
 snd-soc-tlv320aic26-objs := tlv320aic26.o
 snd-soc-tlv320aic3x-objs := tlv320aic3x.o
@@ -23,8 +25,12 @@ snd-soc-wm8750-objs := wm8750.o
 snd-soc-wm8753-objs := wm8753.o
 snd-soc-wm8900-objs := wm8900.o
 snd-soc-wm8903-objs := wm8903.o
+snd-soc-wm8940-objs := wm8940.o
+snd-soc-wm8960-objs := wm8960.o
 snd-soc-wm8971-objs := wm8971.o
+snd-soc-wm8988-objs := wm8988.o
 snd-soc-wm8990-objs := wm8990.o
+snd-soc-wm9081-objs := wm9081.o
 snd-soc-wm9705-objs := wm9705.o
 snd-soc-wm9712-objs := wm9712.o
 snd-soc-wm9713-objs := wm9713.o
@@ -37,7 +43,9 @@ obj-$(CONFIG_SND_SOC_AK4535)	+= snd-soc-ak4535.o
 obj-$(CONFIG_SND_SOC_CS4270)	+= snd-soc-cs4270.o
 obj-$(CONFIG_SND_SOC_L3)	+= snd-soc-l3.o
 obj-$(CONFIG_SND_SOC_PCM3008)	+= snd-soc-pcm3008.o
+obj-$(CONFIG_SND_SOC_SPDIF)	+= snd-soc-spdif.o
 obj-$(CONFIG_SND_SOC_SSM2602)	+= snd-soc-ssm2602.o
+obj-$(CONFIG_SND_SOC_STAC9766)	+= snd-soc-stac9766.o
 obj-$(CONFIG_SND_SOC_TLV320AIC23)	+= snd-soc-tlv320aic23.o
 obj-$(CONFIG_SND_SOC_TLV320AIC26)	+= snd-soc-tlv320aic26.o
 obj-$(CONFIG_SND_SOC_TLV320AIC3X)	+= snd-soc-tlv320aic3x.o
@@ -55,7 +63,11 @@ obj-$(CONFIG_SND_SOC_WM8753)	+= snd-soc-wm8753.o
 obj-$(CONFIG_SND_SOC_WM8900)	+= snd-soc-wm8900.o
 obj-$(CONFIG_SND_SOC_WM8903)	+= snd-soc-wm8903.o
 obj-$(CONFIG_SND_SOC_WM8971)	+= snd-soc-wm8971.o
+obj-$(CONFIG_SND_SOC_WM8940)	+= snd-soc-wm8940.o
+obj-$(CONFIG_SND_SOC_WM8960)	+= snd-soc-wm8960.o
+obj-$(CONFIG_SND_SOC_WM8988)	+= snd-soc-wm8988.o
 obj-$(CONFIG_SND_SOC_WM8990)	+= snd-soc-wm8990.o
+obj-$(CONFIG_SND_SOC_WM9081)	+= snd-soc-wm9081.o
 obj-$(CONFIG_SND_SOC_WM9705)	+= snd-soc-wm9705.o
 obj-$(CONFIG_SND_SOC_WM9712)	+= snd-soc-wm9712.o
 obj-$(CONFIG_SND_SOC_WM9713)	+= snd-soc-wm9713.o
diff --git a/sound/soc/codecs/ac97.c b/sound/soc/codecs/ac97.c
index b0d4af145b8..932299bb5d1 100644
--- a/sound/soc/codecs/ac97.c
+++ b/sound/soc/codecs/ac97.c
@@ -53,13 +53,13 @@ struct snd_soc_dai ac97_dai = {
 		.channels_min = 1,
 		.channels_max = 2,
 		.rates = STD_AC97_RATES,
-		.formats = SNDRV_PCM_FMTBIT_S16_LE,},
+		.formats = SND_SOC_STD_AC97_FMTS,},
 	.capture = {
 		.stream_name = "AC97 Capture",
 		.channels_min = 1,
 		.channels_max = 2,
 		.rates = STD_AC97_RATES,
-		.formats = SNDRV_PCM_FMTBIT_S16_LE,},
+		.formats = SND_SOC_STD_AC97_FMTS,},
 	.ops = &ac97_dai_ops,
 };
 EXPORT_SYMBOL_GPL(ac97_dai);
diff --git a/sound/soc/codecs/ad1980.c b/sound/soc/codecs/ad1980.c
index ddb3b08ac23..d7440a982d2 100644
--- a/sound/soc/codecs/ad1980.c
+++ b/sound/soc/codecs/ad1980.c
@@ -137,13 +137,13 @@ struct snd_soc_dai ad1980_dai = {
 		.channels_min = 2,
 		.channels_max = 6,
 		.rates = SNDRV_PCM_RATE_48000,
-		.formats = SNDRV_PCM_FMTBIT_S16_LE, },
+		.formats = SND_SOC_STD_AC97_FMTS, },
 	.capture = {
 		.stream_name = "Capture",
 		.channels_min = 2,
 		.channels_max = 2,
 		.rates = SNDRV_PCM_RATE_48000,
-		.formats = SNDRV_PCM_FMTBIT_S16_LE, },
+		.formats = SND_SOC_STD_AC97_FMTS, },
 };
 EXPORT_SYMBOL_GPL(ad1980_dai);
 
diff --git a/sound/soc/codecs/cs4270.c b/sound/soc/codecs/cs4270.c
index 7fa09a38762..a32b8226c8a 100644
--- a/sound/soc/codecs/cs4270.c
+++ b/sound/soc/codecs/cs4270.c
@@ -18,7 +18,7 @@
  * - The machine driver's 'startup' function must call
  *   cs4270_set_dai_sysclk() with the value of MCLK.
  * - Only I2S and left-justified modes are supported
- * - Power management is not supported
+ * - Power management is supported
  */
 
 #include <linux/module.h>
@@ -27,6 +27,7 @@
 #include <sound/soc.h>
 #include <sound/initval.h>
 #include <linux/i2c.h>
+#include <linux/delay.h>
 
 #include "cs4270.h"
 
@@ -56,6 +57,7 @@
 #define CS4270_FIRSTREG	0x01
 #define CS4270_LASTREG	0x08
 #define CS4270_NUMREGS	(CS4270_LASTREG - CS4270_FIRSTREG + 1)
+#define CS4270_I2C_INCR	0x80
 
 /* Bit masks for the CS4270 registers */
 #define CS4270_CHIPID_ID	0xF0
@@ -64,6 +66,8 @@
 #define CS4270_PWRCTL_PDN_ADC	0x20
 #define CS4270_PWRCTL_PDN_DAC	0x02
 #define CS4270_PWRCTL_PDN	0x01
+#define CS4270_PWRCTL_PDN_ALL	\
+	(CS4270_PWRCTL_PDN_ADC | CS4270_PWRCTL_PDN_DAC | CS4270_PWRCTL_PDN)
 #define CS4270_MODE_SPEED_MASK	0x30
 #define CS4270_MODE_1X		0x00
 #define CS4270_MODE_2X		0x10
@@ -109,6 +113,7 @@ struct cs4270_private {
 	unsigned int mclk; /* Input frequency of the MCLK pin */
 	unsigned int mode; /* The mode (I2S or left-justified) */
 	unsigned int slave_mode;
+	unsigned int manual_mute;
 };
 
 /**
@@ -295,7 +300,7 @@ static int cs4270_fill_cache(struct snd_soc_codec *codec)
 	s32 length;
 
 	length = i2c_smbus_read_i2c_block_data(i2c_client,
-		CS4270_FIRSTREG | 0x80, CS4270_NUMREGS, cache);
+		CS4270_FIRSTREG | CS4270_I2C_INCR, CS4270_NUMREGS, cache);
 
 	if (length != CS4270_NUMREGS) {
 		dev_err(codec->dev, "i2c read failure, addr=0x%x\n",
@@ -453,7 +458,7 @@ static int cs4270_hw_params(struct snd_pcm_substream *substream,
 }
 
 /**
- * cs4270_mute - enable/disable the CS4270 external mute
+ * cs4270_dai_mute - enable/disable the CS4270 external mute
  * @dai: the SOC DAI
  * @mute: 0 = disable mute, 1 = enable mute
  *
@@ -462,21 +467,52 @@ static int cs4270_hw_params(struct snd_pcm_substream *substream,
  * board does not have the MUTEA or MUTEB pins connected to such circuitry,
  * then this function will do nothing.
  */
-static int cs4270_mute(struct snd_soc_dai *dai, int mute)
+static int cs4270_dai_mute(struct snd_soc_dai *dai, int mute)
 {
 	struct snd_soc_codec *codec = dai->codec;
+	struct cs4270_private *cs4270 = codec->private_data;
 	int reg6;
 
 	reg6 = snd_soc_read(codec, CS4270_MUTE);
 
 	if (mute)
 		reg6 |= CS4270_MUTE_DAC_A | CS4270_MUTE_DAC_B;
-	else
+	else {
 		reg6 &= ~(CS4270_MUTE_DAC_A | CS4270_MUTE_DAC_B);
+		reg6 |= cs4270->manual_mute;
+	}
 
 	return snd_soc_write(codec, CS4270_MUTE, reg6);
 }
 
+/**
+ * cs4270_soc_put_mute - put callback for the 'Master Playback switch'
+ * 			 alsa control.
+ * @kcontrol: mixer control
+ * @ucontrol: control element information
+ *
+ * This function basically passes the arguments on to the generic
+ * snd_soc_put_volsw() function and saves the mute information in
+ * our private data structure. This is because we want to prevent
+ * cs4270_dai_mute() neglecting the user's decision to manually
+ * mute the codec's output.
+ *
+ * Returns 0 for success.
+ */
+static int cs4270_soc_put_mute(struct snd_kcontrol *kcontrol,
+				struct snd_ctl_elem_value *ucontrol)
+{
+	struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
+	struct cs4270_private *cs4270 = codec->private_data;
+	int left = !ucontrol->value.integer.value[0];
+	int right = !ucontrol->value.integer.value[1];
+
+	cs4270->manual_mute = (left ? CS4270_MUTE_DAC_A : 0) |
+			      (right ? CS4270_MUTE_DAC_B : 0);
+
+	return snd_soc_put_volsw(kcontrol, ucontrol);
+}
+
 /* A list of non-DAPM controls that the CS4270 supports */
 static const struct snd_kcontrol_new cs4270_snd_controls[] = {
 	SOC_DOUBLE_R("Master Playback Volume",
@@ -486,7 +522,9 @@ static const struct snd_kcontrol_new cs4270_snd_controls[] = {
 	SOC_SINGLE("Zero Cross Switch", CS4270_TRANS, 5, 1, 0),
 	SOC_SINGLE("Popguard Switch", CS4270_MODE, 0, 1, 1),
 	SOC_SINGLE("Auto-Mute Switch", CS4270_MUTE, 5, 1, 0),
-	SOC_DOUBLE("Master Capture Switch", CS4270_MUTE, 3, 4, 1, 0)
+	SOC_DOUBLE("Master Capture Switch", CS4270_MUTE, 3, 4, 1, 1),
+	SOC_DOUBLE_EXT("Master Playback Switch", CS4270_MUTE, 0, 1, 1, 1,
+		snd_soc_get_volsw, cs4270_soc_put_mute),
 };
 
 /*
@@ -506,7 +544,7 @@ static struct snd_soc_dai_ops cs4270_dai_ops = {
 	.hw_params	= cs4270_hw_params,
 	.set_sysclk	= cs4270_set_dai_sysclk,
 	.set_fmt	= cs4270_set_dai_fmt,
-	.digital_mute	= cs4270_mute,
+	.digital_mute	= cs4270_dai_mute,
 };
 
 struct snd_soc_dai cs4270_dai = {
@@ -753,6 +791,57 @@ static struct i2c_device_id cs4270_id[] = {
 };
 MODULE_DEVICE_TABLE(i2c, cs4270_id);
 
+#ifdef CONFIG_PM
+
+/* This suspend/resume implementation can handle both - a simple standby
+ * where the codec remains powered, and a full suspend, where the voltage
+ * domain the codec is connected to is teared down and/or any other hardware
+ * reset condition is asserted.
+ *
+ * The codec's own power saving features are enabled in the suspend callback,
+ * and all registers are written back to the hardware when resuming.
+ */
+
+static int cs4270_i2c_suspend(struct i2c_client *client, pm_message_t mesg)
+{
+	struct cs4270_private *cs4270 = i2c_get_clientdata(client);
+	struct snd_soc_codec *codec = &cs4270->codec;
+	int reg = snd_soc_read(codec, CS4270_PWRCTL) | CS4270_PWRCTL_PDN_ALL;
+
+	return snd_soc_write(codec, CS4270_PWRCTL, reg);
+}
+
+static int cs4270_i2c_resume(struct i2c_client *client)
+{
+	struct cs4270_private *cs4270 = i2c_get_clientdata(client);
+	struct snd_soc_codec *codec = &cs4270->codec;
+	int reg;
+
+	/* In case the device was put to hard reset during sleep, we need to
+	 * wait 500ns here before any I2C communication. */
+	ndelay(500);
+
+	/* first restore the entire register cache ... */
+	for (reg = CS4270_FIRSTREG; reg <= CS4270_LASTREG; reg++) {
+		u8 val = snd_soc_read(codec, reg);
+
+		if (i2c_smbus_write_byte_data(client, reg, val)) {
+			dev_err(codec->dev, "i2c write failed\n");
+			return -EIO;
+		}
+	}
+
+	/* ... then disable the power-down bits */
+	reg = snd_soc_read(codec, CS4270_PWRCTL);
+	reg &= ~CS4270_PWRCTL_PDN_ALL;
+
+	return snd_soc_write(codec, CS4270_PWRCTL, reg);
+}
+#else
+#define cs4270_i2c_suspend	NULL
+#define cs4270_i2c_resume	NULL
+#endif /* CONFIG_PM */
+
 /*
  * cs4270_i2c_driver - I2C device identification
  *
@@ -767,6 +856,8 @@ static struct i2c_driver cs4270_i2c_driver = {
 	.id_table = cs4270_id,
 	.probe = cs4270_i2c_probe,
 	.remove = cs4270_i2c_remove,
+	.suspend = cs4270_i2c_suspend,
+	.resume = cs4270_i2c_resume,
 };
 
 /*
diff --git a/sound/soc/codecs/spdif_transciever.c b/sound/soc/codecs/spdif_transciever.c
new file mode 100644
index 00000000000..218b33adad9
--- /dev/null
+++ b/sound/soc/codecs/spdif_transciever.c
@@ -0,0 +1,71 @@
+/*
+ * ALSA SoC SPDIF DIT driver
+ *
+ *  This driver is used by controllers which can operate in DIT (SPDI/F) where
+ *  no codec is needed.  This file provides stub codec that can be used
+ *  in these configurations. TI DaVinci Audio controller uses this driver.
+ *
+ * Author:      Steve Chen,  <schen@mvista.com>
+ * Copyright:   (C) 2009 MontaVista Software, Inc., <source@mvista.com>
+ * Copyright:   (C) 2009  Texas Instruments, India
+ *
+ * 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.
+ */
+
+#include <linux/module.h>
+#include <linux/moduleparam.h>
+#include <sound/soc.h>
+#include <sound/pcm.h>
+
+#include "spdif_transciever.h"
+
+#define STUB_RATES	SNDRV_PCM_RATE_8000_96000
+#define STUB_FORMATS	SNDRV_PCM_FMTBIT_S16_LE
+
+struct snd_soc_dai dit_stub_dai = {
+	.name		= "DIT",
+	.playback 	= {
+		.stream_name	= "Playback",
+		.channels_min	= 1,
+		.channels_max	= 384,
+		.rates		= STUB_RATES,
+		.formats	= STUB_FORMATS,
+	},
+};
+
+static int spdif_dit_probe(struct platform_device *pdev)
+{
+	dit_stub_dai.dev = &pdev->dev;
+	return snd_soc_register_dai(&dit_stub_dai);
+}
+
+static int spdif_dit_remove(struct platform_device *pdev)
+{
+	snd_soc_unregister_dai(&dit_stub_dai);
+	return 0;
+}
+
+static struct platform_driver spdif_dit_driver = {
+	.probe		= spdif_dit_probe,
+	.remove		= spdif_dit_remove,
+	.driver		= {
+		.name	= "spdif-dit",
+		.owner	= THIS_MODULE,
+	},
+};
+
+static int __init dit_modinit(void)
+{
+	return platform_driver_register(&spdif_dit_driver);
+}
+
+static void __exit dit_exit(void)
+{
+	platform_driver_unregister(&spdif_dit_driver);
+}
+
+module_init(dit_modinit);
+module_exit(dit_exit);
+
diff --git a/sound/soc/codecs/spdif_transciever.h b/sound/soc/codecs/spdif_transciever.h
new file mode 100644
index 00000000000..296f2eb6c4e
--- /dev/null
+++ b/sound/soc/codecs/spdif_transciever.h
@@ -0,0 +1,17 @@
+/*
+ * ALSA SoC DIT/DIR driver header
+ *
+ * Author:      Steve Chen,  <schen@mvista.com>
+ * Copyright:   (C) 2008 MontaVista Software, Inc., <source@mvista.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.
+ */
+
+#ifndef CODEC_STUBS_H
+#define CODEC_STUBS_H
+
+extern struct snd_soc_dai dit_stub_dai;
+
+#endif /* CODEC_STUBS_H */
diff --git a/sound/soc/codecs/ssm2602.c b/sound/soc/codecs/ssm2602.c
index 87f606c7682..1fc4c8e0899 100644
--- a/sound/soc/codecs/ssm2602.c
+++ b/sound/soc/codecs/ssm2602.c
@@ -336,15 +336,17 @@ static int ssm2602_startup(struct snd_pcm_substream *substream,
 			master_runtime->sample_bits,
 			master_runtime->rate);
 
-		snd_pcm_hw_constraint_minmax(substream->runtime,
-					     SNDRV_PCM_HW_PARAM_RATE,
-					     master_runtime->rate,
-					     master_runtime->rate);
-
-		snd_pcm_hw_constraint_minmax(substream->runtime,
-					     SNDRV_PCM_HW_PARAM_SAMPLE_BITS,
-					     master_runtime->sample_bits,
-					     master_runtime->sample_bits);
+		if (master_runtime->rate != 0)
+			snd_pcm_hw_constraint_minmax(substream->runtime,
+						     SNDRV_PCM_HW_PARAM_RATE,
+						     master_runtime->rate,
+						     master_runtime->rate);
+
+		if (master_runtime->sample_bits != 0)
+			snd_pcm_hw_constraint_minmax(substream->runtime,
+						     SNDRV_PCM_HW_PARAM_SAMPLE_BITS,
+						     master_runtime->sample_bits,
+						     master_runtime->sample_bits);
 
 		ssm2602->slave_substream = substream;
 	} else
@@ -372,6 +374,11 @@ static void ssm2602_shutdown(struct snd_pcm_substream *substream,
 	struct snd_soc_device *socdev = rtd->socdev;
 	struct snd_soc_codec *codec = socdev->card->codec;
 	struct ssm2602_priv *ssm2602 = codec->private_data;
+
+	if (ssm2602->master_substream == substream)
+		ssm2602->master_substream = ssm2602->slave_substream;
+
+	ssm2602->slave_substream = NULL;
 	/* deactivate */
 	if (!codec->active)
 		ssm2602_write(codec, SSM2602_ACTIVE, 0);
@@ -497,11 +504,9 @@ static int ssm2602_set_bias_level(struct snd_soc_codec *codec,
 	return 0;
 }
 
-#define SSM2602_RATES (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_11025 |\
-		SNDRV_PCM_RATE_16000 | SNDRV_PCM_RATE_22050 |\
-		SNDRV_PCM_RATE_32000 | SNDRV_PCM_RATE_44100 |\
-		SNDRV_PCM_RATE_48000 | SNDRV_PCM_RATE_88200 |\
-		SNDRV_PCM_RATE_96000)
+#define SSM2602_RATES (SNDRV_PCM_RATE_8000 | SNDRV_PCM_RATE_32000 |\
+		SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 |\
+		SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000)
 
 #define SSM2602_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S20_3LE |\
 		SNDRV_PCM_FMTBIT_S24_LE | SNDRV_PCM_FMTBIT_S32_LE)
diff --git a/sound/soc/codecs/stac9766.c b/sound/soc/codecs/stac9766.c
new file mode 100644
index 00000000000..8ad4b7b3e3b
--- /dev/null
+++ b/sound/soc/codecs/stac9766.c
@@ -0,0 +1,463 @@
+/*
+ * stac9766.c  --  ALSA SoC STAC9766 codec support
+ *
+ * Copyright 2009 Jon Smirl, Digispeaker
+ * Author: Jon Smirl <jonsmirl@gmail.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.
+ *
+ *  Features:-
+ *
+ *   o Support for AC97 Codec, S/PDIF
+ */
+
+#include <linux/init.h>
+#include <linux/module.h>
+#include <linux/device.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/ac97_codec.h>
+#include <sound/initval.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include <sound/tlv.h>
+#include <sound/soc-of-simple.h>
+
+#include "stac9766.h"
+
+#define STAC9766_VERSION "0.10"
+
+/*
+ * STAC9766 register cache
+ */
+static const u16 stac9766_reg[] = {
+	0x6A90, 0x8000, 0x8000, 0x8000, /* 6 */
+	0x0000, 0x0000, 0x8008, 0x8008, /* e */
+	0x8808, 0x8808, 0x8808, 0x8808, /* 16 */
+	0x8808, 0x0000, 0x8000, 0x0000, /* 1e */
+	0x0000, 0x0000, 0x0000, 0x000f, /* 26 */
+	0x0a05, 0x0400, 0xbb80, 0x0000, /* 2e */
+	0x0000, 0xbb80, 0x0000, 0x0000, /* 36 */
+	0x0000, 0x2000, 0x0000, 0x0100, /* 3e */
+	0x0000, 0x0000, 0x0080, 0x0000, /* 46 */
+	0x0000, 0x0000, 0x0003, 0xffff, /* 4e */
+	0x0000, 0x0000, 0x0000, 0x0000, /* 56 */
+	0x4000, 0x0000, 0x0000, 0x0000, /* 5e */
+	0x1201, 0xFFFF, 0xFFFF, 0x0000, /* 66 */
+	0x0000, 0x0000, 0x0000, 0x0000, /* 6e */
+	0x0000, 0x0000, 0x0000, 0x0006, /* 76 */
+	0x0000, 0x0000, 0x0000, 0x0000, /* 7e */
+};
+
+static const char *stac9766_record_mux[] = {"Mic", "CD", "Video", "AUX",
+			"Line", "Stereo Mix", "Mono Mix", "Phone"};
+static const char *stac9766_mono_mux[] = {"Mix", "Mic"};
+static const char *stac9766_mic_mux[] = {"Mic1", "Mic2"};
+static const char *stac9766_SPDIF_mux[] = {"PCM", "ADC Record"};
+static const char *stac9766_popbypass_mux[] = {"Normal", "Bypass Mixer"};
+static const char *stac9766_record_all_mux[] = {"All analog",
+	"Analog plus DAC"};
+static const char *stac9766_boost1[] = {"0dB", "10dB"};
+static const char *stac9766_boost2[] = {"0dB", "20dB"};
+static const char *stac9766_stereo_mic[] = {"Off", "On"};
+
+static const struct soc_enum stac9766_record_enum =
+	SOC_ENUM_DOUBLE(AC97_REC_SEL, 8, 0, 8, stac9766_record_mux);
+static const struct soc_enum stac9766_mono_enum =
+	SOC_ENUM_SINGLE(AC97_GENERAL_PURPOSE, 9, 2, stac9766_mono_mux);
+static const struct soc_enum stac9766_mic_enum =
+	SOC_ENUM_SINGLE(AC97_GENERAL_PURPOSE, 8, 2, stac9766_mic_mux);
+static const struct soc_enum stac9766_SPDIF_enum =
+	SOC_ENUM_SINGLE(AC97_STAC_DA_CONTROL, 1, 2, stac9766_SPDIF_mux);
+static const struct soc_enum stac9766_popbypass_enum =
+	SOC_ENUM_SINGLE(AC97_GENERAL_PURPOSE, 15, 2, stac9766_popbypass_mux);
+static const struct soc_enum stac9766_record_all_enum =
+	SOC_ENUM_SINGLE(AC97_STAC_ANALOG_SPECIAL, 12, 2,
+			stac9766_record_all_mux);
+static const struct soc_enum stac9766_boost1_enum =
+	SOC_ENUM_SINGLE(AC97_MIC, 6, 2, stac9766_boost1); /* 0/10dB */
+static const struct soc_enum stac9766_boost2_enum =
+	SOC_ENUM_SINGLE(AC97_STAC_ANALOG_SPECIAL, 2, 2, stac9766_boost2); /* 0/20dB */
+static const struct soc_enum stac9766_stereo_mic_enum =
+	SOC_ENUM_SINGLE(AC97_STAC_STEREO_MIC, 2, 1, stac9766_stereo_mic);
+
+static const DECLARE_TLV_DB_LINEAR(master_tlv, -4600, 0);
+static const DECLARE_TLV_DB_LINEAR(record_tlv, 0, 2250);
+static const DECLARE_TLV_DB_LINEAR(beep_tlv, -4500, 0);
+static const DECLARE_TLV_DB_LINEAR(mix_tlv, -3450, 1200);
+
+static const struct snd_kcontrol_new stac9766_snd_ac97_controls[] = {
+	SOC_DOUBLE_TLV("Speaker Volume", AC97_MASTER, 8, 0, 31, 1, master_tlv),
+	SOC_SINGLE("Speaker Switch", AC97_MASTER, 15, 1, 1),
+	SOC_DOUBLE_TLV("Headphone Volume", AC97_HEADPHONE, 8, 0, 31, 1,
+		       master_tlv),
+	SOC_SINGLE("Headphone Switch", AC97_HEADPHONE, 15, 1, 1),
+	SOC_SINGLE_TLV("Mono Out Volume", AC97_MASTER_MONO, 0, 31, 1,
+		       master_tlv),
+	SOC_SINGLE("Mono Out Switch", AC97_MASTER_MONO, 15, 1, 1),
+
+	SOC_DOUBLE_TLV("Record Volume", AC97_REC_GAIN, 8, 0, 15, 0, record_tlv),
+	SOC_SINGLE("Record Switch", AC97_REC_GAIN, 15, 1, 1),
+
+
+	SOC_SINGLE_TLV("Beep Volume", AC97_PC_BEEP, 1, 15, 1, beep_tlv),
+	SOC_SINGLE("Beep Switch", AC97_PC_BEEP, 15, 1, 1),
+	SOC_SINGLE("Beep Frequency", AC97_PC_BEEP, 5, 127, 1),
+	SOC_SINGLE_TLV("Phone Volume", AC97_PHONE, 0, 31, 1, mix_tlv),
+	SOC_SINGLE("Phone Switch", AC97_PHONE, 15, 1, 1),
+
+	SOC_ENUM("Mic Boost1", stac9766_boost1_enum),
+	SOC_ENUM("Mic Boost2", stac9766_boost2_enum),
+	SOC_SINGLE_TLV("Mic Volume", AC97_MIC, 0, 31, 1, mix_tlv),
+	SOC_SINGLE("Mic Switch", AC97_MIC, 15, 1, 1),
+	SOC_ENUM("Stereo Mic", stac9766_stereo_mic_enum),
+
+	SOC_DOUBLE_TLV("Line Volume", AC97_LINE, 8, 0, 31, 1, mix_tlv),
+	SOC_SINGLE("Line Switch", AC97_LINE, 15, 1, 1),
+	SOC_DOUBLE_TLV("CD Volume", AC97_CD, 8, 0, 31, 1, mix_tlv),
+	SOC_SINGLE("CD Switch", AC97_CD, 15, 1, 1),
+	SOC_DOUBLE_TLV("AUX Volume", AC97_AUX, 8, 0, 31, 1, mix_tlv),
+	SOC_SINGLE("AUX Switch", AC97_AUX, 15, 1, 1),
+	SOC_DOUBLE_TLV("Video Volume", AC97_VIDEO, 8, 0, 31, 1, mix_tlv),
+	SOC_SINGLE("Video Switch", AC97_VIDEO, 15, 1, 1),
+
+	SOC_DOUBLE_TLV("DAC Volume", AC97_PCM, 8, 0, 31, 1, mix_tlv),
+	SOC_SINGLE("DAC Switch", AC97_PCM, 15, 1, 1),
+	SOC_SINGLE("Loopback Test Switch", AC97_GENERAL_PURPOSE, 7, 1, 0),
+	SOC_SINGLE("3D Volume", AC97_3D_CONTROL, 3, 2, 1),
+	SOC_SINGLE("3D Switch", AC97_GENERAL_PURPOSE, 13, 1, 0),
+
+	SOC_ENUM("SPDIF Mux", stac9766_SPDIF_enum),
+	SOC_ENUM("Mic1/2 Mux", stac9766_mic_enum),
+	SOC_ENUM("Record All Mux", stac9766_record_all_enum),
+	SOC_ENUM("Record Mux", stac9766_record_enum),
+	SOC_ENUM("Mono Mux", stac9766_mono_enum),
+	SOC_ENUM("Pop Bypass Mux", stac9766_popbypass_enum),
+};
+
+static int stac9766_ac97_write(struct snd_soc_codec *codec, unsigned int reg,
+			       unsigned int val)
+{
+	u16 *cache = codec->reg_cache;
+
+	if (reg > AC97_STAC_PAGE0) {
+		stac9766_ac97_write(codec, AC97_INT_PAGING, 0);
+		soc_ac97_ops.write(codec->ac97, reg, val);
+		stac9766_ac97_write(codec, AC97_INT_PAGING, 1);
+		return 0;
+	}
+	if (reg / 2 > ARRAY_SIZE(stac9766_reg))
+		return -EIO;
+
+	soc_ac97_ops.write(codec->ac97, reg, val);
+	cache[reg / 2] = val;
+	return 0;
+}
+
+static unsigned int stac9766_ac97_read(struct snd_soc_codec *codec,
+				       unsigned int reg)
+{
+	u16 val = 0, *cache = codec->reg_cache;
+
+	if (reg > AC97_STAC_PAGE0) {
+		stac9766_ac97_write(codec, AC97_INT_PAGING, 0);
+		val = soc_ac97_ops.read(codec->ac97, reg - AC97_STAC_PAGE0);
+		stac9766_ac97_write(codec, AC97_INT_PAGING, 1);
+		return val;
+	}
+	if (reg / 2 > ARRAY_SIZE(stac9766_reg))
+		return -EIO;
+
+	if (reg == AC97_RESET || reg == AC97_GPIO_STATUS ||
+		reg == AC97_INT_PAGING || reg == AC97_VENDOR_ID1 ||
+		reg == AC97_VENDOR_ID2) {
+
+		val = soc_ac97_ops.read(codec->ac97, reg);
+		return val;
+	}
+	return cache[reg / 2];
+}
+
+static int ac97_analog_prepare(struct snd_pcm_substream *substream,
+			       struct snd_soc_dai *dai)
+{
+	struct snd_soc_codec *codec = dai->codec;
+	struct snd_pcm_runtime *runtime = substream->runtime;
+	unsigned short reg, vra;
+
+	vra = stac9766_ac97_read(codec, AC97_EXTENDED_STATUS);
+
+	vra |= 0x1; /* enable variable rate audio */
+
+	stac9766_ac97_write(codec, AC97_EXTENDED_STATUS, vra);
+
+	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+		reg = AC97_PCM_FRONT_DAC_RATE;
+	else
+		reg = AC97_PCM_LR_ADC_RATE;
+
+	return stac9766_ac97_write(codec, reg, runtime->rate);
+}
+
+static int ac97_digital_prepare(struct snd_pcm_substream *substream,
+				struct snd_soc_dai *dai)
+{
+	struct snd_soc_codec *codec = dai->codec;
+	struct snd_pcm_runtime *runtime = substream->runtime;
+	unsigned short reg, vra;
+
+	stac9766_ac97_write(codec, AC97_SPDIF, 0x2002);
+
+	vra = stac9766_ac97_read(codec, AC97_EXTENDED_STATUS);
+	vra |= 0x5; /* Enable VRA and SPDIF out */
+
+	stac9766_ac97_write(codec, AC97_EXTENDED_STATUS, vra);
+
+	reg = AC97_PCM_FRONT_DAC_RATE;
+
+	return stac9766_ac97_write(codec, reg, runtime->rate);
+}
+
+static int ac97_digital_trigger(struct snd_pcm_substream *substream,
+				int cmd, struct snd_soc_dai *dai)
+{
+	struct snd_soc_codec *codec = dai->codec;
+	unsigned short vra;
+
+	switch (cmd) {
+	case SNDRV_PCM_TRIGGER_STOP:
+		vra = stac9766_ac97_read(codec, AC97_EXTENDED_STATUS);
+		vra &= !0x04;
+		stac9766_ac97_write(codec, AC97_EXTENDED_STATUS, vra);
+		break;
+	}
+	return 0;
+}
+
+static int stac9766_set_bias_level(struct snd_soc_codec *codec,
+				   enum snd_soc_bias_level level)
+{
+	switch (level) {
+	case SND_SOC_BIAS_ON: /* full On */
+	case SND_SOC_BIAS_P