Merge branch 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6

* 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6: (33 commits)
  ALSA: ASoC codec: remove unused #include <version.h>
  ALSA: ASoC: update email address for Liam Girdwood
  ALSA: hda: corrected invalid mixer values
  ALSA: hda: add mixers for analog mixer on 92hd75xx codecs
  ALSA: ASoC: Add destination and source port for DMA on OMAP1
  ALSA: ASoC: Drop device registration from GTA01 lm4857 driver
  ALSA: ASoC: Fix build of GTA01 audio driver
  ALSA: ASoC: Add widgets before setting endpoints on GTA01
  ALSA: ASoC: Fix inverted input PGA mute bits in WM8903
  ALSA: ASoC: OMAP: Set DMA stream name at runtime in McBSP DAI driver
  ALSA: ASoC: OMAP: Add support for OMAP2430 and OMAP34xx in McBSP DAI driver
  ALSA: ASoC: OMAP: Add multilink support to McBSP DAI driver
  ALSA: ASoC: Make TLV320AIC26 user-visible
  ALSA: ASoC - clean up Kconfig for TLV320AIC2
  ALSA: ASoC: Make WM8510 microphone input a DAPM mixer
  ALSA: ASoC: Implement WM8510 bias level control
  ALSA: ASoC: Remove unused AUDIO_NAME define from codec drivers
  ALSA: ASoC: tlv320aic3x: Use uniform tlv320aic naming
  ALSA: ASoC: Add WM8510 SPI support
  ALSA: ASoC: Add WM8753 SPI support
  ...

sound/soc/omap/Kconfig

@@ -13,3 +13,11 @@ config SND_OMAP_SOC_N810
 	select SND_SOC_TLV320AIC3X
 	help
 	  Say Y if you want to add support for SoC audio on Nokia N810.
+
+config SND_OMAP_SOC_OSK5912
+	tristate "SoC Audio support for omap osk5912"
+	depends on SND_OMAP_SOC && MACH_OMAP_OSK
+	select SND_OMAP_SOC_MCBSP
+	select SND_SOC_TLV320AIC23
+	help
+	  Say Y if you want to add support for SoC audio on osk5912.

sound/soc/omap/Makefile

@@ -7,5 +7,7 @@ obj-$(CONFIG_SND_OMAP_SOC_MCBSP) += snd-soc-omap-mcbsp.o
 
 # OMAP Machine Support
 snd-soc-n810-objs := n810.o
+snd-soc-osk5912-objs := osk5912.o
 
 obj-$(CONFIG_SND_OMAP_SOC_N810) += snd-soc-n810.o
+obj-$(CONFIG_SND_OMAP_SOC_OSK5912) += snd-soc-osk5912.o

sound/soc/omap/n810.c

@@ -247,9 +247,9 @@ static int n810_aic33_init(struct snd_soc_codec *codec)
 	int i, err;
 
 	/* Not connected */
-	snd_soc_dapm_disable_pin(codec, "MONO_LOUT");
-	snd_soc_dapm_disable_pin(codec, "HPLCOM");
-	snd_soc_dapm_disable_pin(codec, "HPRCOM");
+	snd_soc_dapm_nc_pin(codec, "MONO_LOUT");
+	snd_soc_dapm_nc_pin(codec, "HPLCOM");
+	snd_soc_dapm_nc_pin(codec, "HPRCOM");
 
 	/* Add N810 specific controls */
 	for (i = 0; i < ARRAY_SIZE(aic33_n810_controls); i++) {

sound/soc/omap/omap-mcbsp.c

@@ -59,12 +59,7 @@ static struct omap_mcbsp_data mcbsp_data[NUM_LINKS];
  * Stream DMA parameters. DMA request line and port address are set runtime
  * since they are different between OMAP1 and later OMAPs
  */
-static struct omap_pcm_dma_data omap_mcbsp_dai_dma_params[NUM_LINKS][2] = {
-{
-	{ .name		= "I2S PCM Stereo out", },
-	{ .name		= "I2S PCM Stereo in", },
-},
-};
+static struct omap_pcm_dma_data omap_mcbsp_dai_dma_params[NUM_LINKS][2];
 
 #if defined(CONFIG_ARCH_OMAP15XX) || defined(CONFIG_ARCH_OMAP16XX)
 static const int omap1_dma_reqs[][2] = {
@@ -84,11 +79,22 @@ static const unsigned long omap1_mcbsp_port[][2] = {
 static const int omap1_dma_reqs[][2] = {};
 static const unsigned long omap1_mcbsp_port[][2] = {};
 #endif
-#if defined(CONFIG_ARCH_OMAP2420)
-static const int omap2420_dma_reqs[][2] = {
+
+#if defined(CONFIG_ARCH_OMAP24XX) || defined(CONFIG_ARCH_OMAP34XX)
+static const int omap24xx_dma_reqs[][2] = {
 	{ OMAP24XX_DMA_MCBSP1_TX, OMAP24XX_DMA_MCBSP1_RX },
 	{ OMAP24XX_DMA_MCBSP2_TX, OMAP24XX_DMA_MCBSP2_RX },
+#if defined(CONFIG_ARCH_OMAP2430) || defined(CONFIG_ARCH_OMAP34XX)
+	{ OMAP24XX_DMA_MCBSP3_TX, OMAP24XX_DMA_MCBSP3_RX },
+	{ OMAP24XX_DMA_MCBSP4_TX, OMAP24XX_DMA_MCBSP4_RX },
+	{ OMAP24XX_DMA_MCBSP5_TX, OMAP24XX_DMA_MCBSP5_RX },
+#endif
 };
+#else
+static const int omap24xx_dma_reqs[][2] = {};
+#endif
+
+#if defined(CONFIG_ARCH_OMAP2420)
 static const unsigned long omap2420_mcbsp_port[][2] = {
 	{ OMAP24XX_MCBSP1_BASE + OMAP_MCBSP_REG_DXR1,
 	  OMAP24XX_MCBSP1_BASE + OMAP_MCBSP_REG_DRR1 },
@@ -96,10 +102,43 @@ static const unsigned long omap2420_mcbsp_port[][2] = {
 	  OMAP24XX_MCBSP2_BASE + OMAP_MCBSP_REG_DRR1 },
 };
 #else
-static const int omap2420_dma_reqs[][2] = {};
 static const unsigned long omap2420_mcbsp_port[][2] = {};
 #endif
 
+#if defined(CONFIG_ARCH_OMAP2430)
+static const unsigned long omap2430_mcbsp_port[][2] = {
+	{ OMAP24XX_MCBSP1_BASE + OMAP_MCBSP_REG_DXR,
+	  OMAP24XX_MCBSP1_BASE + OMAP_MCBSP_REG_DRR },
+	{ OMAP24XX_MCBSP2_BASE + OMAP_MCBSP_REG_DXR,
+	  OMAP24XX_MCBSP2_BASE + OMAP_MCBSP_REG_DRR },
+	{ OMAP2430_MCBSP3_BASE + OMAP_MCBSP_REG_DXR,
+	  OMAP2430_MCBSP3_BASE + OMAP_MCBSP_REG_DRR },
+	{ OMAP2430_MCBSP4_BASE + OMAP_MCBSP_REG_DXR,
+	  OMAP2430_MCBSP4_BASE + OMAP_MCBSP_REG_DRR },
+	{ OMAP2430_MCBSP5_BASE + OMAP_MCBSP_REG_DXR,
+	  OMAP2430_MCBSP5_BASE + OMAP_MCBSP_REG_DRR },
+};
+#else
+static const unsigned long omap2430_mcbsp_port[][2] = {};
+#endif
+
+#if defined(CONFIG_ARCH_OMAP34XX)
+static const unsigned long omap34xx_mcbsp_port[][2] = {
+	{ OMAP34XX_MCBSP1_BASE + OMAP_MCBSP_REG_DXR,
+	  OMAP34XX_MCBSP1_BASE + OMAP_MCBSP_REG_DRR },
+	{ OMAP34XX_MCBSP2_BASE + OMAP_MCBSP_REG_DXR,
+	  OMAP34XX_MCBSP2_BASE + OMAP_MCBSP_REG_DRR },
+	{ OMAP34XX_MCBSP3_BASE + OMAP_MCBSP_REG_DXR,
+	  OMAP34XX_MCBSP3_BASE + OMAP_MCBSP_REG_DRR },
+	{ OMAP34XX_MCBSP4_BASE + OMAP_MCBSP_REG_DXR,
+	  OMAP34XX_MCBSP4_BASE + OMAP_MCBSP_REG_DRR },
+	{ OMAP34XX_MCBSP5_BASE + OMAP_MCBSP_REG_DXR,
+	  OMAP34XX_MCBSP5_BASE + OMAP_MCBSP_REG_DRR },
+};
+#else
+static const unsigned long omap34xx_mcbsp_port[][2] = {};
+#endif
+
 static int omap_mcbsp_dai_startup(struct snd_pcm_substream *substream)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
@@ -167,14 +206,19 @@ static int omap_mcbsp_dai_hw_params(struct snd_pcm_substream *substream,
 		dma = omap1_dma_reqs[bus_id][substream->stream];
 		port = omap1_mcbsp_port[bus_id][substream->stream];
 	} else if (cpu_is_omap2420()) {
-		dma = omap2420_dma_reqs[bus_id][substream->stream];
+		dma = omap24xx_dma_reqs[bus_id][substream->stream];
 		port = omap2420_mcbsp_port[bus_id][substream->stream];
+	} else if (cpu_is_omap2430()) {
+		dma = omap24xx_dma_reqs[bus_id][substream->stream];
+		port = omap2430_mcbsp_port[bus_id][substream->stream];
+	} else if (cpu_is_omap343x()) {
+		dma = omap24xx_dma_reqs[bus_id][substream->stream];
+		port = omap34xx_mcbsp_port[bus_id][substream->stream];
 	} else {
-		/*
-		 * TODO: Add support for 2430 and 3430
-		 */
 		return -ENODEV;
 	}
+	omap_mcbsp_dai_dma_params[id][substream->stream].name =
+		substream->stream ? "Audio Capture" : "Audio Playback";
 	omap_mcbsp_dai_dma_params[id][substream->stream].dma_req = dma;
 	omap_mcbsp_dai_dma_params[id][substream->stream].port_addr = port;
 	cpu_dai->dma_data = &omap_mcbsp_dai_dma_params[id][substream->stream];
@@ -245,6 +289,11 @@ static int omap_mcbsp_dai_set_dai_fmt(struct snd_soc_dai *cpu_dai,
 		regs->rcr2	|= RDATDLY(1);
 		regs->xcr2	|= XDATDLY(1);
 		break;
+	case SND_SOC_DAIFMT_DSP_A:
+		/* 0-bit data delay */
+		regs->rcr2      |= RDATDLY(0);
+		regs->xcr2      |= XDATDLY(0);
+		break;
 	default:
 		/* Unsupported data format */
 		return -EINVAL;
@@ -310,7 +359,7 @@ static int omap_mcbsp_dai_set_clks_src(struct omap_mcbsp_data *mcbsp_data,
 				       int clk_id)
 {
 	int sel_bit;
-	u16 reg;
+	u16 reg, reg_devconf1 = OMAP243X_CONTROL_DEVCONF1;
 
 	if (cpu_class_is_omap1()) {
 		/* OMAP1's can use only external source clock */
@@ -320,6 +369,12 @@ static int omap_mcbsp_dai_set_clks_src(struct omap_mcbsp_data *mcbsp_data,
 			return 0;
 	}
 
+	if (cpu_is_omap2420() && mcbsp_data->bus_id > 1)
+		return -EINVAL;
+
+	if (cpu_is_omap343x())
+		reg_devconf1 = OMAP343X_CONTROL_DEVCONF1;
+
 	switch (mcbsp_data->bus_id) {
 	case 0:
 		reg = OMAP2_CONTROL_DEVCONF0;
@@ -329,20 +384,26 @@ static int omap_mcbsp_dai_set_clks_src(struct omap_mcbsp_data *mcbsp_data,
 		reg = OMAP2_CONTROL_DEVCONF0;
 		sel_bit = 6;
 		break;
-	/* TODO: Support for ports 3 - 5 in OMAP2430 and OMAP34xx */
+	case 2:
+		reg = reg_devconf1;
+		sel_bit = 0;
+		break;
+	case 3:
+		reg = reg_devconf1;
+		sel_bit = 2;
+		break;
+	case 4:
+		reg = reg_devconf1;
+		sel_bit = 4;
+		break;
 	default:
 		return -EINVAL;
 	}
 
-	if (cpu_class_is_omap2()) {
-		if (clk_id == OMAP_MCBSP_SYSCLK_CLKS_FCLK) {
-			omap_ctrl_writel(omap_ctrl_readl(reg) &
-					 ~(1 << sel_bit), reg);
-		} else {
-			omap_ctrl_writel(omap_ctrl_readl(reg) |
-					 (1 << sel_bit), reg);
-		}
-	}
+	if (clk_id == OMAP_MCBSP_SYSCLK_CLKS_FCLK)
+		omap_ctrl_writel(omap_ctrl_readl(reg) & ~(1 << sel_bit), reg);
+	else
+		omap_ctrl_writel(omap_ctrl_readl(reg) | (1 << sel_bit), reg);
 
 	return 0;
 }
@@ -376,37 +437,49 @@ static int omap_mcbsp_dai_set_dai_sysclk(struct snd_soc_dai *cpu_dai,
 	return err;
 }
 
-struct snd_soc_dai omap_mcbsp_dai[NUM_LINKS] = {
-{
-	.name = "omap-mcbsp-dai",
-	.id = 0,
-	.type = SND_SOC_DAI_I2S,
-	.playback = {
-		.channels_min = 2,
-		.channels_max = 2,
-		.rates = OMAP_MCBSP_RATES,
-		.formats = SNDRV_PCM_FMTBIT_S16_LE,
-	},
-	.capture = {
-		.channels_min = 2,
-		.channels_max = 2,
-		.rates = OMAP_MCBSP_RATES,
-		.formats = SNDRV_PCM_FMTBIT_S16_LE,
-	},
-	.ops = {
-		.startup = omap_mcbsp_dai_startup,
-		.shutdown = omap_mcbsp_dai_shutdown,
-		.trigger = omap_mcbsp_dai_trigger,
-		.hw_params = omap_mcbsp_dai_hw_params,
-	},
-	.dai_ops = {
-		.set_fmt = omap_mcbsp_dai_set_dai_fmt,
-		.set_clkdiv = omap_mcbsp_dai_set_clkdiv,
-		.set_sysclk = omap_mcbsp_dai_set_dai_sysclk,
-	},
-	.private_data = &mcbsp_data[0].bus_id,
-},
+#define OMAP_MCBSP_DAI_BUILDER(link_id)				\
+{								\
+	.name = "omap-mcbsp-dai-(link_id)",			\
+	.id = (link_id),					\
+	.type = SND_SOC_DAI_I2S,				\
+	.playback = {						\
+		.channels_min = 2,				\
+		.channels_max = 2,				\
+		.rates = OMAP_MCBSP_RATES,			\
+		.formats = SNDRV_PCM_FMTBIT_S16_LE,		\
+	},							\
+	.capture = {						\
+		.channels_min = 2,				\
+		.channels_max = 2,				\
+		.rates = OMAP_MCBSP_RATES,			\
+		.formats = SNDRV_PCM_FMTBIT_S16_LE,		\
+	},							\
+	.ops = {						\
+		.startup = omap_mcbsp_dai_startup,		\
+		.shutdown = omap_mcbsp_dai_shutdown,		\
+		.trigger = omap_mcbsp_dai_trigger,		\
+		.hw_params = omap_mcbsp_dai_hw_params,		\
+	},							\
+	.dai_ops = {						\
+		.set_fmt = omap_mcbsp_dai_set_dai_fmt,		\
+		.set_clkdiv = omap_mcbsp_dai_set_clkdiv,	\
+		.set_sysclk = omap_mcbsp_dai_set_dai_sysclk,	\
+	},							\
+	.private_data = &mcbsp_data[(link_id)].bus_id,		\
+}
+
+struct snd_soc_dai omap_mcbsp_dai[] = {
+	OMAP_MCBSP_DAI_BUILDER(0),
+	OMAP_MCBSP_DAI_BUILDER(1),
+#if NUM_LINKS >= 3
+	OMAP_MCBSP_DAI_BUILDER(2),
+#endif
+#if NUM_LINKS == 5
+	OMAP_MCBSP_DAI_BUILDER(3),
+	OMAP_MCBSP_DAI_BUILDER(4),
+#endif
 };
+
 EXPORT_SYMBOL_GPL(omap_mcbsp_dai);
 
 MODULE_AUTHOR("Jarkko Nikula <jarkko.nikula@nokia.com>");

sound/soc/omap/omap-mcbsp.h

@@ -38,11 +38,17 @@ enum omap_mcbsp_div {
 	OMAP_MCBSP_CLKGDV,		/* Sample rate generator divider */
 };
 
-/*
- * REVISIT: Preparation for the ASoC v2. Let the number of available links to
- * be same than number of McBSP ports found in OMAP(s) we are compiling for.
- */
-#define NUM_LINKS	1
+#if defined(CONFIG_ARCH_OMAP2420)
+#define NUM_LINKS	2
+#endif
+#if defined(CONFIG_ARCH_OMAP15XX) || defined(CONFIG_ARCH_OMAP16XX)
+#undef  NUM_LINKS
+#define NUM_LINKS	3
+#endif
+#if defined(CONFIG_ARCH_OMAP2430) || defined(CONFIG_ARCH_OMAP34XX)
+#undef  NUM_LINKS
+#define NUM_LINKS	5
+#endif
 
 extern struct snd_soc_dai omap_mcbsp_dai[NUM_LINKS];
 

sound/soc/omap/omap-pcm.c

@@ -97,7 +97,7 @@ static int omap_pcm_hw_params(struct snd_pcm_substream *substream,
 	prtd->dma_data = dma_data;
 	err = omap_request_dma(dma_data->dma_req, dma_data->name,
 			       omap_pcm_dma_irq, substream, &prtd->dma_ch);
-	if (!cpu_is_omap1510()) {
+	if (!err & !cpu_is_omap1510()) {
 		/*
 		 * Link channel with itself so DMA doesn't need any
 		 * reprogramming while looping the buffer
@@ -147,12 +147,14 @@ static int omap_pcm_prepare(struct snd_pcm_substream *substream)
 		dma_params.src_or_dst_synch	= OMAP_DMA_DST_SYNC;
 		dma_params.src_start		= runtime->dma_addr;
 		dma_params.dst_start		= dma_data->port_addr;
+		dma_params.dst_port		= OMAP_DMA_PORT_MPUI;
 	} else {
 		dma_params.src_amode		= OMAP_DMA_AMODE_CONSTANT;
 		dma_params.dst_amode		= OMAP_DMA_AMODE_POST_INC;
 		dma_params.src_or_dst_synch	= OMAP_DMA_SRC_SYNC;
 		dma_params.src_start		= dma_data->port_addr;
 		dma_params.dst_start		= runtime->dma_addr;
+		dma_params.src_port		= OMAP_DMA_PORT_MPUI;
 	}
 	/*
 	 * Set DMA transfer frame size equal to ALSA period size and frame

sound/soc/omap/osk5912.c

+/*
+ * osk5912.c  --  SoC audio for OSK 5912
+ *
+ * Copyright (C) 2008 Mistral Solutions
+ *
+ * Contact: Arun KS  <arunks@mistralsolutions.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.
+ *
+ * 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/clk.h>
+#include <linux/platform_device.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+
+#include <asm/mach-types.h>
+#include <mach/hardware.h>
+#include <linux/gpio.h>
+#include <mach/mcbsp.h>
+
+#include "omap-mcbsp.h"
+#include "omap-pcm.h"
+#include "../codecs/tlv320aic23.h"
+
+#define CODEC_CLOCK 	12000000
+
+static struct clk *tlv320aic23_mclk;
+
+static int osk_startup(struct snd_pcm_substream *substream)
+{
+	return clk_enable(tlv320aic23_mclk);
+}
+
+static void osk_shutdown(struct snd_pcm_substream *substream)
+{
+	clk_disable(tlv320aic23_mclk);
+}
+
+static int osk_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_DSP_A |
+				  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_DSP_A |
+				  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 osk_ops = {
+	.startup = osk_startup,
+	.hw_params = osk_hw_params,
+	.shutdown = osk_shutdown,
+};
+
+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 osk_tlv320aic23_init(struct snd_soc_codec *codec)
+{
+
+	/* Add osk5912 specific widgets */
+	snd_soc_dapm_new_controls(codec, tlv320aic23_dapm_widgets,
+				  ARRAY_SIZE(tlv320aic23_dapm_widgets));
+
+	/* Set up osk5912 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 osk_dai = {
+	.name = "TLV320AIC23",
+	.stream_name = "AIC23",
+	.cpu_dai = &omap_mcbsp_dai[0],
+	.codec_dai = &tlv320aic23_dai,
+	.init = osk_tlv320aic23_init,
+	.ops = &osk_ops,
+};
+
+/* Audio machine driver */
+static struct snd_soc_machine snd_soc_machine_osk = {
+	.name = "OSK5912",
+	.dai_link = &osk_dai,
+	.num_links = 1,
+};
+
+/* Audio subsystem */
+static struct snd_soc_device osk_snd_devdata = {
+	.machine = &snd_soc_machine_osk,
+	.platform = &omap_soc_platform,
+	.codec_dev = &soc_codec_dev_tlv320aic23,
+};
+
+static struct platform_device *osk_snd_device;
+
+static int __init osk_soc_init(void)
+{
+	int err;
+	u32 curRate;
+	struct device *dev;
+
+	if (!(machine_is_omap_osk()))
+		return -ENODEV;
+
+	osk_snd_device = platform_device_alloc("soc-audio", -1);
+	if (!osk_snd_device)
+		return -ENOMEM;
+
+	platform_set_drvdata(osk_snd_device, &osk_snd_devdata);
+	osk_snd_devdata.dev = &osk_snd_device->dev;
+	*(unsigned int *)osk_dai.cpu_dai->private_data = 0;	/* McBSP1 */
+	err = platform_device_add(osk_snd_device);
+	if (err)
+		goto err1;
+
+	dev = &osk_snd_device->dev;
+
+	tlv320aic23_mclk = clk_get(dev, "mclk");
+	if (IS_ERR(tlv320aic23_mclk)) {
+		printk(KERN_ERR "Could not get mclk clock\n");
+		return -ENODEV;
+	}
+
+	if (clk_get_usecount(tlv320aic23_mclk) > 0) {
+		/* MCLK is already in use */
+		printk(KERN_WARNING
+		       "MCLK in use at %d Hz. We change it to %d Hz\n",
+		       (uint) clk_get_rate(tlv320aic23_mclk), CODEC_CLOCK);
+	}
+
+	/*
+	 * Configure 12 MHz output on MCLK.
+	 */
+	curRate = (uint) clk_get_rate(tlv320aic23_mclk);
+	if (curRate != CODEC_CLOCK) {
+		if (clk_set_rate(tlv320aic23_mclk, CODEC_CLOCK)) {
+			printk(KERN_ERR "Cannot set MCLK for AIC23 CODEC\n");
+			err = -ECANCELED;
+			goto err1;
+		}
+	}
+
+	printk(KERN_INFO "MCLK = %d [%d], usecount = %d\n",
+	       (uint) clk_get_rate(tlv320aic23_mclk), CODEC_CLOCK,
+	       clk_get_usecount(tlv320aic23_mclk));
+
+	return 0;
+err1:
+	clk_put(tlv320aic23_mclk);
+	platform_device_del(osk_snd_device);
+	platform_device_put(osk_snd_device);
+
+	return err;
+
+}
+
+static void __exit osk_soc_exit(void)
+{
+	platform_device_unregister(osk_snd_device);
+}
+
+module_init(osk_soc_init);
+module_exit(osk_soc_exit);
+
+MODULE_AUTHOR("Arun KS <arunks@mistralsolutions.com>");
+MODULE_DESCRIPTION("ALSA SoC OSK 5912");
+MODULE_LICENSE("GPL");

sound/soc/pxa/corgi.c

@@ -4,7 +4,7 @@
  * Copyright 2005 Wolfson Microelectronics PLC.
  * Copyright 2005 Openedhand Ltd.
  *
- * Authors: Liam Girdwood <liam.girdwood@wolfsonmicro.com>
+ * Authors: Liam Girdwood <lrg@slimlogic.co.uk>
  *          Richard Purdie <richard@openedhand.com>
  *
  *  This program is free software; you can redistribute  it and/or modify it
@@ -281,8 +281,8 @@ static int corgi_wm8731_init(struct snd_soc_codec *codec)
 {
 	int i, err;
 
-	snd_soc_dapm_disable_pin(codec, "LLINEIN");
-	snd_soc_dapm_disable_pin(codec, "RLINEIN");
+	snd_soc_dapm_nc_pin(codec, "LLINEIN");
+	snd_soc_dapm_nc_pin(codec, "RLINEIN");
 
 	/* Add corgi specific controls */
 	for (i = 0; i < ARRAY_SIZE(wm8731_corgi_controls); i++) {

sound/soc/pxa/em-x270.c

@@ -9,7 +9,7 @@
  * Copyright 2005 Wolfson Microelectronics PLC.
  * Copyright 2005 Openedhand Ltd.
  *
- * Authors: Liam Girdwood <liam.girdwood@wolfsonmicro.com>
+ * Authors: Liam Girdwood <lrg@slimlogic.co.uk>
  *          Richard Purdie <richard@openedhand.com>
  *
  *  This program is free software; you can redistribute  it and/or modify it

sound/soc/pxa/poodle.c

@@ -4,7 +4,7 @@
  * Copyright 2005 Wolfson Microelectronics PLC.
  * Copyright 2005 Openedhand Ltd.
  *
- * Authors: Liam Girdwood <liam.girdwood@wolfsonmicro.com>
+ * Authors: Liam Girdwood <lrg@slimlogic.co.uk>
  *          Richard Purdie <richard@openedhand.com>
  *
  *  This program is free software; you can redistribute  it and/or modify it
@@ -242,8 +242,8 @@ static int poodle_wm8731_init(struct snd_soc_codec *codec)
 {
 	int i, err;
 
-	snd_soc_dapm_disable_pin(codec, "LLINEIN");
-	snd_soc_dapm_disable_pin(codec, "RLINEIN");
+	snd_soc_dapm_nc_pin(codec, "LLINEIN");
+	snd_soc_dapm_nc_pin(codec, "RLINEIN");
 	snd_soc_dapm_enable_pin(codec, "MICIN");
 
 	/* Add poodle specific controls */

sound/soc/pxa/pxa2xx-i2s.c

@@ -3,7 +3,7 @@
  *
  * Copyright 2005 Wolfson Microelectronics PLC.
  * Author: Liam Girdwood
- *         liam.girdwood@wolfsonmicro.com or linux@wolfsonmicro.com
+ *         lrg@slimlogic.co.uk
  *
  *  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
@@ -405,6 +405,6 @@ module_init(pxa2xx_i2s_init);
 module_exit(pxa2xx_i2s_exit);
 
 /* Module information */
-MODULE_AUTHOR("Liam Girdwood, liam.girdwood@wolfsonmicro.com, www.wolfsonmicro.com");
+MODULE_AUTHOR("Liam Girdwood, lrg@slimlogic.co.uk");
 MODULE_DESCRIPTION("pxa2xx I2S SoC Interface");
 MODULE_LICENSE("GPL");

sound/soc/pxa/spitz.c

@@ -4,7 +4,7 @@
  * Copyright 2005 Wolfson Microelectronics PLC.
  * Copyright 2005 Openedhand Ltd.
  *
- * Authors: Liam Girdwood <liam.girdwood@wolfsonmicro.com>
+ * Authors: Liam Girdwood <lrg@slimlogic.co.uk>
  *          Richard Purdie <richard@openedhand.com>
  *
  *  This program is free software; you can redistribute  it and/or modify it
@@ -281,13 +281,13 @@ static int spitz_wm8750_init(struct snd_soc_codec *codec)
 	int i, err;
 
 	/* NC codec pins */
-	snd_soc_dapm_disable_pin(codec, "RINPUT1");
-	snd_soc_dapm_disable_pin(codec, "LINPUT2");
-	snd_soc_dapm_disable_pin(codec, "RINPUT2");
-	snd_soc_dapm_disable_pin(codec, "LINPUT3");
-	snd_soc_dapm_disable_pin(codec, "RINPUT3");
-	snd_soc_dapm_disable_pin(codec, "OUT3");
-	snd_soc_dapm_disable_pin(codec, "MONO1");
+	snd_soc_dapm_nc_pin(codec, "RINPUT1");
+	snd_soc_dapm_nc_pin(codec, "LINPUT2");
+	snd_soc_dapm_nc_pin(codec, "RINPUT2");
+	snd_soc_dapm_nc_pin(codec, "LINPUT3");
+	snd_soc_dapm_nc_pin(codec, "RINPUT3");
+	snd_soc_dapm_nc_pin(codec, "OUT3");
+	snd_soc_dapm_nc_pin(codec, "MONO1");
 
 	/* Add spitz specific controls */
 	for (i = 0; i < ARRAY_SIZE(wm8750_spitz_controls); i++) {

sound/soc/pxa/tosa.c

@@ -4,7 +4,7 @@
  * Copyright 2005 Wolfson Microelectronics PLC.
  * Copyright 2005 Openedhand Ltd.
  *
- * Authors: Liam Girdwood <liam.girdwood@wolfsonmicro.com>
+ * Authors: Liam Girdwood <lrg@slimlogic.co.uk>
  *          Richard Purdie <richard@openedhand.com>
  *
  *  This program is free software; you can redistribute  it and/or modify it
@@ -190,8 +190,8 @@ static int tosa_ac97_init(struct snd_soc_codec *codec)
 {
 	int i, err;
 
-	snd_soc_dapm_disable_pin(codec, "OUT3");
-	snd_soc_dapm_disable_pin(codec, "MONOOUT");
+	snd_soc_dapm_nc_pin(codec, "OUT3");
+	snd_soc_dapm_nc_pin(codec, "MONOOUT");
 
 	/* add tosa specific controls */
 	for (i = 0; i < ARRAY_SIZE(tosa_controls); i++) {

sound/soc/s3c24xx/neo1973_wm8753.c

@@ -511,21 +511,20 @@ static int neo1973_wm8753_init(struct snd_soc_codec *codec)
 	DBG("Entered %s\n", __func__);
 
 	/* set up NC codec pins */
-	snd_soc_dapm_disable_pin(codec, "LOUT2");
-	snd_soc_dapm_disable_pin(codec, "ROUT2");
-	snd_soc_dapm_disable_pin(codec, "OUT3");
-	snd_soc_dapm_disable_pin(codec, "OUT4");
-	snd_soc_dapm_disable_pin(codec, "LINE1");
-	snd_soc_dapm_disable_pin(codec, "LINE2");
-
-
-	/* set endpoints to default mode */
-	set_scenario_endpoints(codec, NEO_AUDIO_OFF);
+	snd_soc_dapm_nc_pin(codec, "LOUT2");
+	snd_soc_dapm_nc_pin(codec, "ROUT2");
+	snd_soc_dapm_nc_pin(codec, "OUT3");
+	snd_soc_dapm_nc_pin(codec, "OUT4");
+	snd_soc_dapm_nc_pin(codec, "LINE1");
+	snd_soc_dapm_nc_pin(codec, "LINE2");
 
 	/* Add neo1973 specific widgets */
 	snd_soc_dapm_new_controls(codec, wm8753_dapm_widgets,
 				  ARRAY_SIZE(wm8753_dapm_widgets));
 
+	/* set endpoints to default mode */
+	set_scenario_endpoints(codec, NEO_AUDIO_OFF);
+
 	/* add neo1973 specific controls */
 	for (i = 0; i < ARRAY_SIZE(wm8753_neo1973_controls); i++) {
 		err = snd_ctl_add(codec->card,
@@ -603,6 +602,8 @@ static int lm4857_i2c_probe(struct i2c_client *client,
 {
 	DBG("Entered %s\n", __func__);
 
+	i2c = client;
+
 	lm4857_write_regs();
 	return 0;
 }
@@ -611,6 +612,8 @@ static int lm4857_i2c_remove(struct i2c_client *client)
 {
 	DBG("Entered %s\n", __func__);
 
+	i2c = NULL;
+
 	return 0;
 }
 
@@ -650,7 +653,7 @@ static void lm4857_shutdown(struct i2c_client *dev)
 }
 
 static const struct i2c_device_id lm4857_i2c_id[] = {
-	{ "neo1973_lm4857", 0 }
+	{ "neo1973_lm4857", 0 },
 	{ }
 };
 
@@ -668,48 +671,6 @@ static struct i2c_driver lm4857_i2c_driver = {
 };
 
 static struct platform_device *neo1973_snd_device;
-static struct i2c_client *lm4857_client;
-
-static int __init neo1973_add_lm4857_device(struct platform_device *pdev,
-					    int i2c_bus,
-					    unsigned short i2c_address)
-{
-	struct i2c_board_info info;
-	struct i2c_adapter *adapter;
-	struct i2c_client *client;
-	int ret;
-
-	ret = i2c_add_driver(&lm4857_i2c_driver);
-	if (ret != 0) {
-		dev_err(&pdev->dev, "can't add lm4857 driver\n");
-		return ret;
-	}
-
-	memset(&info, 0, sizeof(struct i2c_board_info));
-	info.addr = i2c_address;
-	strlcpy(info.type, "neo1973_lm4857", I2C_NAME_SIZE);
-
-	adapter = i2c_get_adapter(i2c_bus);
-	if (!adapter) {
-		dev_err(&pdev->dev, "can't get i2c adapter %d\n", i2c_bus);
-		goto err_driver;
-	}
-
-	client = i2c_new_device(adapter, &info);
-	i2c_put_adapter(adapter);
-	if (!client) {
-		dev_err(&pdev->dev, "can't add lm4857 device at 0x%x\n",
-			(unsigned int)info.addr);
-		goto err_driver;
-	}
-
-	lm4857_client = client;
-	return 0;
-
-err_driver:
-	i2c_del_driver(&lm4857_i2c_driver);
-	return -ENODEV;
-}
 
 static int __init neo1973_init(void)
 {
@@ -736,8 +697,8 @@ static int __init neo1973_init(void)
 		return ret;
 	}
 
-	ret = neo1973_add_lm4857_device(neo1973_snd_device,
-					neo1973_wm8753_setup, 0x7C);
+	ret = i2c_add_driver(&lm4857_i2c_driver);
+
 	if (ret != 0)
 		platform_device_unregister(neo1973_snd_device);
 
@@ -748,7 +709,6 @@ static void __exit neo1973_exit(void)
 {
 	DBG("Entered %s\n", __func__);
 
-	i2c_unregister_device(lm4857_client);
 	i2c_del_driver(&lm4857_i2c_driver);
 	platform_device_unregister(neo1973_snd_device);
 }

sound/soc/soc-core.c

@@ -4,8 +4,7 @@
  * Copyright 2005 Wolfson Microelectronics PLC.
  * Copyright 2005 Openedhand Ltd.
  *
- * Author: Liam Girdwood
- *         liam.girdwood@wolfsonmicro.com or linux@wolfsonmicro.com
+ * Author: Liam Girdwood <lrg@slimlogic.co.uk>
  *         with code, comments and ideas from :-
  *         Richard Purdie <richard@openedhand.com>
  *
@@ -1886,7 +1885,7 @@ module_init(snd_soc_init);
 module_exit(snd_soc_exit);
 
 /* Module information */
-MODULE_AUTHOR("Liam Girdwood, liam.girdwood@wolfsonmicro.com, www.wolfsonmicro.com");
+MODULE_AUTHOR("Liam Girdwood, lrg@slimlogic.co.uk");
 MODULE_DESCRIPTION("ALSA SoC Core");
 MODULE_LICENSE("GPL");
 MODULE_ALIAS("platform:soc-audio");

sound/soc/soc-dapm.c

@@ -2,8 +2,7 @@
  * soc-dapm.c  --  ALSA SoC Dynamic Audio Power Management
  *
  * Copyright 2005 Wolfson Microelectronics PLC.
- * Author: Liam Girdwood
- *         liam.girdwood@wolfsonmicro.com or linux@wolfsonmicro.com
+ * Author: Liam Girdwood <lrg@slimlogic.co.uk>
  *
  *  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
@@ -1483,6 +1482,26 @@ int snd_soc_dapm_disable_pin(struct snd_soc_codec *codec, char *pin)
 }
 EXPORT_SYMBOL_GPL(snd_soc_dapm_disable_pin);
 
+/**
+ * snd_soc_dapm_nc_pin - permanently disable pin.
+ * @codec: SoC codec
+ * @pin: pin name
+ *
+ * Marks the specified pin as being not connected, disabling it along
+ * any parent or child widgets.  At present this is identical to
+ * snd_soc_dapm_disable_pin() but in future it will be extended to do
+ * additional things such as disabling controls which only affect
+ * paths through the pin.
+ *
+ * NOTE: snd_soc_dapm_sync() needs to be called after this for DAPM to
+ * do any widget power switching.
+ */
+int snd_soc_dapm_nc_pin(struct snd_soc_codec *codec, char *pin)
+{
+	return snd_soc_dapm_set_pin(codec, pin, 0);
+}
+EXPORT_SYMBOL_GPL(snd_soc_dapm_nc_pin);
+
 /**
  * snd_soc_dapm_get_pin_status - get audio pin status
  * @codec: audio codec
@@ -1521,6 +1540,6 @@ void snd_soc_dapm_free(struct snd_soc_device *socdev)
 EXPORT_SYMBOL_GPL(snd_soc_dapm_free);
 
 /* Module information */
-MODULE_AUTHOR("Liam Girdwood, liam.girdwood@wolfsonmicro.com, www.wolfsonmicro.com");
+MODULE_AUTHOR("Liam Girdwood, lrg@slimlogic.co.uk");
 MODULE_DESCRIPTION("Dynamic Audio Power Management core for ALSA SoC");
 MODULE_LICENSE("GPL");