Merge branch 'topic/asoc' into for-linus

* topic/asoc: (135 commits)
  ASoC: Apostrophe patrol
  ASoC: codec tlv320aic23 fix bogus divide by 0 message
  ASoC: fix NULL pointer dereference in soc_suspend()
  ASoC: Fix build error in twl4030.c
  ASoC: SSM2602: assign last substream to the master when shutting down
  ASoC: Blackfin: document how anomaly 05000250 is handled
  ASoC: Blackfin: set the transfer size according the ac97_frame size
  ASoC: SSM2602: remove unsupported sample rates
  ASoC: TWL4030: Check the interface format for 4 channel mode
  ASoC: TWL4030: Use reg_cache in twl4030_init_chip
  ASoC: Initialise dev for the dummy S/PDIF DAI
  ASoC: Add dummy S/PDIF codec support
  ASoC: correct print specifiers for unsigneds
  ASoC: Modify mpc5200 AC97 driver to use V9 of spin_event_timeout()
  ASoC: Switch FSL SSI DAI over to symmetric_rates
  ASoC: Mark MPC5200 AC97 as BROKEN until PowerPC merge issues are resolved
  ASoC: Fabric bindings for STAC9766 on the Efika
  ASoC: Support for AC97 on Phytec pmc030 base board.
  ASoC: AC97 driver for mpc5200
  ASoC: Main rewite of the mpc5200 audio DMA code
  ...

94 files changed, 12367 insertions, 1614 deletions

diff --git a/sound/aoa/fabrics/layout.c b/sound/aoa/fabrics/layout.c
index fbf5c933baa..586965f9605 100644
--- a/sound/aoa/fabrics/layout.c
+++ b/sound/aoa/fabrics/layout.c
@@ -1037,7 +1037,7 @@ static int aoa_fabric_layout_probe(struct soundbus_dev *sdev)
 	}
 	ldev->selfptr_headphone.ptr = ldev;
 	ldev->selfptr_lineout.ptr = ldev;
-	sdev->ofdev.dev.driver_data = ldev;
+	dev_set_drvdata(&sdev->ofdev.dev, ldev);
 	list_add(&ldev->list, &layouts_list);
 	layouts_list_items++;
 
@@ -1081,7 +1081,7 @@ static int aoa_fabric_layout_probe(struct soundbus_dev *sdev)
 
 static int aoa_fabric_layout_remove(struct soundbus_dev *sdev)
 {
-	struct layout_dev *ldev = sdev->ofdev.dev.driver_data;
+	struct layout_dev *ldev = dev_get_drvdata(&sdev->ofdev.dev);
 	int i;
 
 	for (i=0; i<MAX_CODECS_PER_BUS; i++) {
@@ -1114,7 +1114,7 @@ static int aoa_fabric_layout_remove(struct soundbus_dev *sdev)
 #ifdef CONFIG_PM
 static int aoa_fabric_layout_suspend(struct soundbus_dev *sdev, pm_message_t state)
 {
-	struct layout_dev *ldev = sdev->ofdev.dev.driver_data;
+	struct layout_dev *ldev = dev_get_drvdata(&sdev->ofdev.dev);
 
 	if (ldev->gpio.methods && ldev->gpio.methods->all_amps_off)
 		ldev->gpio.methods->all_amps_off(&ldev->gpio);
@@ -1124,7 +1124,7 @@ static int aoa_fabric_layout_suspend(struct soundbus_dev *sdev, pm_message_t sta
 
 static int aoa_fabric_layout_resume(struct soundbus_dev *sdev)
 {
-	struct layout_dev *ldev = sdev->ofdev.dev.driver_data;
+	struct layout_dev *ldev = dev_get_drvdata(&sdev->ofdev.dev);
 
 	if (ldev->gpio.methods && ldev->gpio.methods->all_amps_off)
 		ldev->gpio.methods->all_amps_restore(&ldev->gpio);
diff --git a/sound/aoa/soundbus/i2sbus/core.c b/sound/aoa/soundbus/i2sbus/core.c
index 418c84c99d6..4e3b819d499 100644
--- a/sound/aoa/soundbus/i2sbus/core.c
+++ b/sound/aoa/soundbus/i2sbus/core.c
@@ -358,14 +358,14 @@ static int i2sbus_probe(struct macio_dev* dev, const struct of_device_id *match)
 		return -ENODEV;
 	}
 
-	dev->ofdev.dev.driver_data = control;
+	dev_set_drvdata(&dev->ofdev.dev, control);
 
 	return 0;
 }
 
 static int i2sbus_remove(struct macio_dev* dev)
 {
-	struct i2sbus_control *control = dev->ofdev.dev.driver_data;
+	struct i2sbus_control *control = dev_get_drvdata(&dev->ofdev.dev);
 	struct i2sbus_dev *i2sdev, *tmp;
 
 	list_for_each_entry_safe(i2sdev, tmp, &control->list, item)
@@ -377,7 +377,7 @@ static int i2sbus_remove(struct macio_dev* dev)
 #ifdef CONFIG_PM
 static int i2sbus_suspend(struct macio_dev* dev, pm_message_t state)
 {
-	struct i2sbus_control *control = dev->ofdev.dev.driver_data;
+	struct i2sbus_control *control = dev_get_drvdata(&dev->ofdev.dev);
 	struct codec_info_item *cii;
 	struct i2sbus_dev* i2sdev;
 	int err, ret = 0;
@@ -407,7 +407,7 @@ static int i2sbus_suspend(struct macio_dev* dev, pm_message_t state)
 
 static int i2sbus_resume(struct macio_dev* dev)
 {
-	struct i2sbus_control *control = dev->ofdev.dev.driver_data;
+	struct i2sbus_control *control = dev_get_drvdata(&dev->ofdev.dev);
 	struct codec_info_item *cii;
 	struct i2sbus_dev* i2sdev;
 	int err, ret = 0;
diff --git a/sound/soc/Kconfig b/sound/soc/Kconfig
index 3d2bb6fc6dc..d3e786a9a0a 100644
--- a/sound/soc/Kconfig
+++ b/sound/soc/Kconfig
@@ -32,7 +32,9 @@ source "sound/soc/fsl/Kconfig"
 source "sound/soc/omap/Kconfig"
 source "sound/soc/pxa/Kconfig"
 source "sound/soc/s3c24xx/Kconfig"
+source "sound/soc/s6000/Kconfig"
 source "sound/soc/sh/Kconfig"
+source "sound/soc/txx9/Kconfig"
 
 # Supported codecs
 source "sound/soc/codecs/Kconfig"
diff --git a/sound/soc/Makefile b/sound/soc/Makefile
index 0237879fd41..6f1e28de23c 100644
--- a/sound/soc/Makefile
+++ b/sound/soc/Makefile
@@ -10,4 +10,6 @@ obj-$(CONFIG_SND_SOC)	+= fsl/
 obj-$(CONFIG_SND_SOC)	+= omap/
 obj-$(CONFIG_SND_SOC)	+= pxa/
 obj-$(CONFIG_SND_SOC)	+= s3c24xx/
+obj-$(CONFIG_SND_SOC)	+= s6000/
 obj-$(CONFIG_SND_SOC)	+= sh/
+obj-$(CONFIG_SND_SOC)	+= txx9/
diff --git a/sound/soc/atmel/Kconfig b/sound/soc/atmel/Kconfig
index a608d7009db..e720d5e6f04 100644
--- a/sound/soc/atmel/Kconfig
+++ b/sound/soc/atmel/Kconfig
@@ -41,3 +41,11 @@ config SND_AT32_SOC_PLAYPAQ_SLAVE
           and FRAME signals on the PlayPaq.  Unless you want to play
           with the AT32 as the SSC master, you probably want to say N here,
           as this will give you better sound quality.
+
+config SND_AT91_SOC_AFEB9260
+	tristate "SoC Audio support for AFEB9260 board"
+	depends on ARCH_AT91 && MACH_AFEB9260 && SND_ATMEL_SOC
+	select SND_ATMEL_SOC_SSC
+	select SND_SOC_TLV320AIC23
+	help
+	  Say Y here to support sound on AFEB9260 board.
diff --git a/sound/soc/atmel/Makefile b/sound/soc/atmel/Makefile
index f54a7cc68e6..e7ea56bd5f8 100644
--- a/sound/soc/atmel/Makefile
+++ b/sound/soc/atmel/Makefile
@@ -13,3 +13,4 @@ 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
+obj-$(CONFIG_SND_AT91_SOC_AFEB9260) += snd-soc-afeb9260.o
diff --git a/sound/soc/atmel/playpaq_wm8510.c b/sound/soc/atmel/playpaq_wm8510.c
index 70657534e6b..9eb610c2ba9 100644
--- a/sound/soc/atmel/playpaq_wm8510.c
+++ b/sound/soc/atmel/playpaq_wm8510.c
@@ -117,7 +117,7 @@ static struct ssc_clock_data playpaq_wm8510_calc_ssc_clock(
 	 * Find actual rate, compare to requested rate
 	 */
 	actual_rate = (cd.ssc_rate / (cd.cmr_div * 2)) / (2 * (cd.period + 1));
-	pr_debug("playpaq_wm8510: Request rate = %d, actual rate = %d\n",
+	pr_debug("playpaq_wm8510: Request rate = %u, actual rate = %u\n",
 		 rate, actual_rate);
 
 
diff --git a/sound/soc/atmel/snd-soc-afeb9260.c b/sound/soc/atmel/snd-soc-afeb9260.c
new file mode 100644
index 00000000000..23349de2731
--- /dev/null
+++ b/sound/soc/atmel/snd-soc-afeb9260.c
@@ -0,0 +1,203 @@
+/*
+ * afeb9260.c  --  SoC audio for AFEB9260
+ *
+ * Copyright (C) 2009 Sergey Lapin <slapin@ossfans.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.
+ *
+ * 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., 51 Franklin St, Fifth Floor, Boston, MA
+ * 02110-1301 USA
+ *
+ */
+
+#include <linux/module.h>
+#include <linux/moduleparam.h>
+#include <linux/kernel.h>
+#include <linux/clk.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 <asm/mach-types.h>
+#include <mach/hardware.h>
+#include <linux/gpio.h>
+
+#include "../codecs/tlv320aic23.h"
+#include "atmel-pcm.h"
+#include "atmel_ssc_dai.h"
+
+#define CODEC_CLOCK 	12000000
+
+static int afeb9260_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;
+	int err;
+
+	/* Set codec DAI configuration */
+	err = snd_soc_dai_set_fmt(codec_dai,
+				  SND_SOC_DAIFMT_I2S|
+				  SND_SOC_DAIFMT_NB_IF |
+				  SND_SOC_DAIFMT_CBM_CFM);
+	if (err < 0) {
+		printk(KERN_ERR "can't set codec DAI configuration\n");
+		return err;
+	}
+
+	/* Set cpu DAI configuration */
+	err = snd_soc_dai_set_fmt(cpu_dai,
+				  SND_SOC_DAIFMT_I2S |
+				  SND_SOC_DAIFMT_NB_IF |
+				  SND_SOC_DAIFMT_CBM_CFM);
+	if (err < 0) {
+		printk(KERN_ERR "can't set cpu DAI configuration\n");
+		return err;
+	}
+
+	/* Set the codec system clock for DAC and ADC */
+	err =
+	    snd_soc_dai_set_sysclk(codec_dai, 0, CODEC_CLOCK, SND_SOC_CLOCK_IN);
+
+	if (err < 0) {
+		printk(KERN_ERR "can't set codec system clock\n");
+		return err;
+	}
+
+	return err;
+}
+
+static struct snd_soc_ops afeb9260_ops = {
+	.hw_params = afeb9260_hw_params,
+};
+
+static const struct snd_soc_dapm_widget tlv320aic23_dapm_widgets[] = {
+	SND_SOC_DAPM_HP("Headphone Jack", NULL),
+	SND_SOC_DAPM_LINE("Line In", NULL),
+	SND_SOC_DAPM_MIC("Mic Jack", NULL),
+};
+
+static const struct snd_soc_dapm_route audio_map[] = {
+	{"Headphone Jack", NULL, "LHPOUT"},
+	{"Headphone Jack", NULL, "RHPOUT"},
+
+	{"LLINEIN", NULL, "Line In"},
+	{"RLINEIN", NULL, "Line In"},
+
+	{"MICIN", NULL, "Mic Jack"},
+};
+
+static int afeb9260_tlv320aic23_init(struct snd_soc_codec *codec)
+{
+
+	/* Add afeb9260 specific widgets */
+	snd_soc_dapm_new_controls(codec, tlv320aic23_dapm_widgets,
+				  ARRAY_SIZE(tlv320aic23_dapm_widgets));
+
+	/* Set up afeb9260 specific audio path audio_map */
+	snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map));
+
+	snd_soc_dapm_enable_pin(codec, "Headphone Jack");
+	snd_soc_dapm_enable_pin(codec, "Line In");
+	snd_soc_dapm_enable_pin(codec, "Mic Jack");
+
+	snd_soc_dapm_sync(codec);
+
+	return 0;
+}
+
+/* Digital audio interface glue - connects codec <--> CPU */
+static struct snd_soc_dai_link afeb9260_dai = {
+	.name = "TLV320AIC23",
+	.stream_name = "AIC23",
+	.cpu_dai = &atmel_ssc_dai[0],
+	.codec_dai = &tlv320aic23_dai,
+	.init = afeb9260_tlv320aic23_init,
+	.ops = &afeb9260_ops,
+};
+
+/* Audio machine driver */
+static struct snd_soc_card snd_soc_machine_afeb9260 = {
+	.name = "AFEB9260",
+	.platform = &atmel_soc_platform,
+	.dai_link = &afeb9260_dai,
+	.num_links = 1,
+};
+
+/* Audio subsystem */
+static struct snd_soc_device afeb9260_snd_devdata = {
+	.card = &snd_soc_machine_afeb9260,
+	.codec_dev = &soc_codec_dev_tlv320aic23,
+};
+
+static struct platform_device *afeb9260_snd_device;
+
+static int __init afeb9260_soc_init(void)
+{
+	int err;
+	struct device *dev;
+	struct atmel_ssc_info *ssc_p = afeb9260_dai.cpu_dai->private_data;
+	struct ssc_device *ssc = NULL;
+