Merge commit 'v2.6.30-rc5' into core/iommu

Merge reason: core/iommu was on an .30-rc1 base,
              update it to .30-rc5 to refresh.

Signed-off-by: Ingo Molnar <mingo@elte.hu>

21 files changed, 117 insertions, 58 deletions

diff --git a/sound/soc/au1x/dbdma2.c b/sound/soc/au1x/dbdma2.c
index 30490a259148..594c6c5b7838 100644
--- a/sound/soc/au1x/dbdma2.c
+++ b/sound/soc/au1x/dbdma2.c
@@ -82,7 +82,7 @@ static struct au1xpsc_audio_dmadata *au1xpsc_audio_pcmdma[2];
 /* PCM hardware DMA capabilities - platform specific */
 static const struct snd_pcm_hardware au1xpsc_pcm_hardware = {
 	.info		  = SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID |
-			    SNDRV_PCM_INFO_INTERLEAVED,
+			    SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BATCH,
 	.formats	  = AU1XPSC_PCM_FMTS,
 	.period_bytes_min = AU1XPSC_PERIOD_MIN_BYTES,
 	.period_bytes_max = 4096 * 1024 - 1,
diff --git a/sound/soc/codecs/Makefile b/sound/soc/codecs/Makefile
index 030d2454725f..f2653803ede8 100644
--- a/sound/soc/codecs/Makefile
+++ b/sound/soc/codecs/Makefile
@@ -56,7 +56,6 @@ 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_WM8990)	+= snd-soc-wm8990.o
-obj-$(CONFIG_SND_SOC_WM8991)	+= snd-soc-wm8991.o
 obj-$(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/twl4030.c b/sound/soc/codecs/twl4030.c
index 921b205de28a..df7c8c281d2f 100644
--- a/sound/soc/codecs/twl4030.c
+++ b/sound/soc/codecs/twl4030.c
@@ -836,6 +836,12 @@ static DECLARE_TLV_DB_SCALE(analog_tlv, -2400, 200, 0);
 static DECLARE_TLV_DB_SCALE(output_tvl, -1200, 600, 1);
 
 /*
+ * Gain control for earpiece amplifier
+ * 0 dB to 12 dB in 6 dB steps (mute instead of -6)
+ */
+static DECLARE_TLV_DB_SCALE(output_ear_tvl, -600, 600, 1);
+
+/*
  * Capture gain after the ADCs
  * from 0 dB to 31 dB in 1 dB steps
  */
@@ -900,7 +906,7 @@ static const struct snd_kcontrol_new twl4030_snd_controls[] = {
 		4, 3, 0, output_tvl),
 
 	SOC_SINGLE_TLV_TWL4030("Earpiece Playback Volume",
-		TWL4030_REG_EAR_CTL, 4, 3, 0, output_tvl),
+		TWL4030_REG_EAR_CTL, 4, 3, 0, output_ear_tvl),
 
 	/* Common capture gain controls */
 	SOC_DOUBLE_R_TLV("TX1 Digital Capture Volume",
diff --git a/sound/soc/codecs/wm8350.c b/sound/soc/codecs/wm8350.c
index 3b1d0993bed9..0275321ff8ab 100644
--- a/sound/soc/codecs/wm8350.c
+++ b/sound/soc/codecs/wm8350.c
@@ -968,7 +968,7 @@ static int wm8350_pcm_trigger(struct snd_pcm_substream *substream,
 	 * required for LRC in master mode. The DACs or ADCs need a
 	 * valid audio path i.e. pin -> ADC or DAC -> pin before
 	 * the LRC will be enabled in master mode. */
-	if (!master && cmd != SNDRV_PCM_TRIGGER_START)
+	if (!master || cmd != SNDRV_PCM_TRIGGER_START)
 		return 0;
 
 	if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) {
diff --git a/sound/soc/codecs/wm8580.c b/sound/soc/codecs/wm8580.c
index 442ea6f160fc..9f6be3d31ac0 100644
--- a/sound/soc/codecs/wm8580.c
+++ b/sound/soc/codecs/wm8580.c
@@ -268,9 +268,11 @@ static const DECLARE_TLV_DB_SCALE(dac_tlv, -12750, 50, 1);
 static int wm8580_out_vu(struct snd_kcontrol *kcontrol,
 			 struct snd_ctl_elem_value *ucontrol)
 {
+	struct soc_mixer_control *mc =
+		(struct soc_mixer_control *)kcontrol->private_value;
 	struct snd_soc_codec *codec = snd_kcontrol_chip(kcontrol);
-	int reg = kcontrol->private_value & 0xff;
-	int reg2 = (kcontrol->private_value >> 24) & 0xff;
+	unsigned int reg = mc->reg;
+	unsigned int reg2 = mc->rreg;
 	int ret;
 	u16 val;
 
@@ -292,15 +294,17 @@ static int wm8580_out_vu(struct snd_kcontrol *kcontrol,
 	return 0;
 }
 
-#define SOC_WM8580_OUT_DOUBLE_R_TLV(xname, reg_left, reg_right, shift, max, invert, tlv_array) \
+#define SOC_WM8580_OUT_DOUBLE_R_TLV(xname, reg_left, reg_right, xshift, xmax, \
+				    xinvert, 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 = snd_soc_get_volsw_2r, .put = wm8580_out_vu, \
-	.private_value = (reg_left) | ((shift) << 8)  |		\
-		((max) << 12) | ((invert) << 20) | ((reg_right) << 24) }
+	.private_value = (unsigned long)&(struct soc_mixer_control) \
+		{.reg = reg_left, .rreg = reg_right, .shift = xshift, \
+		.max = xmax, .invert = xinvert} }
 
 static const struct snd_kcontrol_new wm8580_snd_controls[] = {
 SOC_WM8580_OUT_DOUBLE_R_TLV("DAC1 Playback Volume",
@@ -522,7 +526,7 @@ static int wm8580_set_dai_pll(struct snd_soc_dai *codec_dai,
 	reg = wm8580_read(codec, WM8580_PLLA4 + offset);
 	reg &= ~0x3f;
 	reg |= pll_div.prescale | pll_div.postscale << 1 |
-		pll_div.freqmode << 4;
+		pll_div.freqmode << 3;
 
 	wm8580_write(codec, WM8580_PLLA4 + offset, reg);
 
diff --git a/sound/soc/codecs/wm9705.c b/sound/soc/codecs/wm9705.c
index 6e23a81dba78..c2d1a7a18fa3 100644
--- a/sound/soc/codecs/wm9705.c
+++ b/sound/soc/codecs/wm9705.c
@@ -318,7 +318,7 @@ static int wm9705_reset(struct snd_soc_codec *codec)
 }
 
 #ifdef CONFIG_PM
-static int wm9705_soc_suspend(struct platform_device *pdev)
+static int wm9705_soc_suspend(struct platform_device *pdev, pm_message_t msg)
 {
 	struct snd_soc_device *socdev = platform_get_drvdata(pdev);
 	struct snd_soc_codec *codec = socdev->card->codec;
diff --git a/sound/soc/fsl/mpc5200_psc_i2s.c b/sound/soc/fsl/mpc5200_psc_i2s.c
index 3aa729df27b5..1111c710118a 100644
--- a/sound/soc/fsl/mpc5200_psc_i2s.c
+++ b/sound/soc/fsl/mpc5200_psc_i2s.c
@@ -504,7 +504,8 @@ static struct snd_soc_dai psc_i2s_dai_template = {
 
 static const struct snd_pcm_hardware psc_i2s_pcm_hardware = {
 	.info = SNDRV_PCM_INFO_MMAP | SNDRV_PCM_INFO_MMAP_VALID |
-		SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BLOCK_TRANSFER,
+		SNDRV_PCM_INFO_INTERLEAVED | SNDRV_PCM_INFO_BLOCK_TRANSFER |
+		SNDRV_PCM_INFO_BATCH,
 	.formats = SNDRV_PCM_FMTBIT_S8 | SNDRV_PCM_FMTBIT_S16_BE |
 		   SNDRV_PCM_FMTBIT_S24_BE | SNDRV_PCM_FMTBIT_S32_BE,
 	.rate_min = 8000,
diff --git a/sound/soc/omap/n810.c b/sound/soc/omap/n810.c
index a6d1178ce128..91ef17992de5 100644
--- a/sound/soc/omap/n810.c
+++ b/sound/soc/omap/n810.c
@@ -3,7 +3,7 @@
  *
  * Copyright (C) 2008 Nokia Corporation
  *
- * Contact: Jarkko Nikula <jarkko.nikula@nokia.com>
+ * Contact: Jarkko Nikula <jhnikula@gmail.com>
  *
  * This program is free software; you can redistribute it and/or
  * modify it under the terms of the GNU General Public License
@@ -417,6 +417,6 @@ static void __exit n810_soc_exit(void)
 module_init(n810_soc_init);
 module_exit(n810_soc_exit);
 
-MODULE_AUTHOR("Jarkko Nikula <jarkko.nikula@nokia.com>");
+MODULE_AUTHOR("Jarkko Nikula <jhnikula@gmail.com>");
 MODULE_DESCRIPTION("ALSA SoC Nokia N810");
 MODULE_LICENSE("GPL");
diff --git a/sound/soc/omap/omap-mcbsp.c b/sound/soc/omap/omap-mcbsp.c
index 9c09b94f0cf8..912614283848 100644
--- a/sound/soc/omap/omap-mcbsp.c
+++ b/sound/soc/omap/omap-mcbsp.c
@@ -3,7 +3,8 @@
  *
  * Copyright (C) 2008 Nokia Corporation
  *
- * Contact: Jarkko Nikula <jarkko.nikula@nokia.com>
+ * Contact: Jarkko Nikula <jhnikula@gmail.com>
+ *          Peter Ujfalusi <peter.ujfalusi@nokia.com>
  *
  * This program is free software; you can redistribute it and/or
  * modify it under the terms of the GNU General Public License
@@ -283,7 +284,7 @@ static int omap_mcbsp_dai_hw_params(struct snd_pcm_substream *substream,
 		break;
 	case SND_SOC_DAIFMT_DSP_B:
 		regs->srgr2	|= FPER(wlen * channels - 1);
-		regs->srgr1	|= FWID(wlen * channels - 2);
+		regs->srgr1	|= FWID(0);
 		break;
 	}
 
@@ -302,6 +303,7 @@ static int omap_mcbsp_dai_set_dai_fmt(struct snd_soc_dai *cpu_dai,
 {
 	struct omap_mcbsp_data *mcbsp_data = to_mcbsp(cpu_dai->private_data);
 	struct omap_mcbsp_reg_cfg *regs = &mcbsp_data->regs;
+	unsigned int temp_fmt = fmt;
 
 	if (mcbsp_data->configured)
 		return 0;
@@ -328,6 +330,8 @@ static int omap_mcbsp_dai_set_dai_fmt(struct snd_soc_dai *cpu_dai,
 		/* 0-bit data delay */
 		regs->rcr2      |= RDATDLY(0);
 		regs->xcr2      |= XDATDLY(0);
+		/* Invert FS polarity configuration */
+		temp_fmt ^= SND_SOC_DAIFMT_NB_IF;
 		break;
 	default:
 		/* Unsupported data format */
@@ -351,7 +355,7 @@ static int omap_mcbsp_dai_set_dai_fmt(struct snd_soc_dai *cpu_dai,
 	}
 
 	/* Set bit clock (CLKX/CLKR) and FS polarities */
-	switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
+	switch (temp_fmt & SND_SOC_DAIFMT_INV_MASK) {
 	case SND_SOC_DAIFMT_NB_NF:
 		/*
 		 * Normal BCLK + FS.
@@ -529,6 +533,6 @@ static void __exit snd_omap_mcbsp_exit(void)
 }
 module_exit(snd_omap_mcbsp_exit);
 
-MODULE_AUTHOR("Jarkko Nikula <jarkko.nikula@nokia.com>");
+MODULE_AUTHOR("Jarkko Nikula <jhnikula@gmail.com>");
 MODULE_DESCRIPTION("OMAP I2S SoC Interface");
 MODULE_LICENSE("GPL");
diff --git a/sound/soc/omap/omap-mcbsp.h b/sound/soc/omap/omap-mcbsp.h
index df7ad13ba73d..c8147aace813 100644
--- a/sound/soc/omap/omap-mcbsp.h
+++ b/sound/soc/omap/omap-mcbsp.h
@@ -3,7 +3,8 @@
  *
  * Copyright (C) 2008 Nokia Corporation
  *
- * Contact: Jarkko Nikula <jarkko.nikula@nokia.com>
+ * Contact: Jarkko Nikula <jhnikula@gmail.com>
+ *          Peter Ujfalusi <peter.ujfalusi@nokia.com>
  *
  * This program is free software; you can redistribute it and/or
  * modify it under the terms of the GNU General Public License
diff --git a/sound/soc/omap/omap-pcm.c b/sound/soc/omap/omap-pcm.c
index 1bdbb0427183..07cf7f46b584 100644
--- a/sound/soc/omap/omap-pcm.c
+++ b/sound/soc/omap/omap-pcm.c
@@ -3,7 +3,8 @@
  *
  * Copyright (C) 2008 Nokia Corporation
  *
- * Contact: Jarkko Nikula <jarkko.nikula@nokia.com>
+ * Contact: Jarkko Nikula <jhnikula@gmail.com>
+ *          Peter Ujfalusi <peter.ujfalusi@nokia.com>
  *
  * This program is free software; you can redistribute it and/or
  * modify it under the terms of the GNU General Public License
@@ -367,6 +368,6 @@ static void __exit omap_soc_platform_exit(void)
 }
 module_exit(omap_soc_platform_exit);
 
-MODULE_AUTHOR("Jarkko Nikula <jarkko.nikula@nokia.com>");
+MODULE_AUTHOR("Jarkko Nikula <jhnikula@gmail.com>");
 MODULE_DESCRIPTION("OMAP PCM DMA module");
 MODULE_LICENSE("GPL");
diff --git a/sound/soc/omap/omap-pcm.h b/sound/soc/omap/omap-pcm.h
index e4369bdfd77d..8d9d26916b05 100644
--- a/sound/soc/omap/omap-pcm.h
+++ b/sound/soc/omap/omap-pcm.h
@@ -3,7 +3,8 @@
  *
  * Copyright (C) 2008 Nokia Corporation
  *
- * Contact: Jarkko Nikula <jarkko.nikula@nokia.com>
+ * Contact: Jarkko Nikula <jhnikula@gmail.com>
+ *          Peter Ujfalusi <peter.ujfalusi@nokia.com>
  *
  * This program is free software; you can redistribute it and/or
  * modify it under the terms of the GNU General Public License
diff --git a/sound/soc/omap/osk5912.c b/sound/soc/omap/osk5912.c
index a952a4eb3361..a4e149b7f0eb 100644
--- a/sound/soc/omap/osk5912.c
+++ b/sound/soc/omap/osk5912.c
@@ -62,7 +62,7 @@ static int osk_hw_params(struct snd_pcm_substream *substream,
 	/* Set codec DAI configuration */
 	err = snd_soc_dai_set_fmt(codec_dai,
 				  SND_SOC_DAIFMT_DSP_B |
-				  SND_SOC_DAIFMT_NB_IF |
+				  SND_SOC_DAIFMT_NB_NF |
 				  SND_SOC_DAIFMT_CBM_CFM);
 	if (err < 0) {
 		printk(KERN_ERR "can't set codec DAI configuration\n");
@@ -72,7 +72,7 @@ static int osk_hw_params(struct snd_pcm_substream *substream,
 	/* Set cpu DAI configuration */
 	err = snd_soc_dai_set_fmt(cpu_dai,
 				  SND_SOC_DAIFMT_DSP_B |
-				  SND_SOC_DAIFMT_NB_IF |
+				  SND_SOC_DAIFMT_NB_NF |
 				  SND_SOC_DAIFMT_CBM_CFM);
 	if (err < 0) {
 		printk(KERN_ERR "can't set cpu DAI configuration\n");
diff --git a/sound/soc/pxa/magician.c b/sound/soc/pxa/magician.c
index f7c4544f7859..0625c342a1c9 100644
--- a/sound/soc/pxa/magician.c
+++ b/sound/soc/pxa/magician.c
@@ -27,8 +27,6 @@
 #include <sound/soc.h>
 #include <sound/soc-dapm.h>
 
-#include <mach/pxa-regs.h>
-#include <mach/hardware.h>
 #include <mach/magician.h>
 #include <asm/mach-types.h>
 #include "../codecs/uda1380.h"
diff --git a/sound/soc/pxa/palm27x.c b/sound/soc/pxa/palm27x.c
index 48a73f64500b..44fcc4e01e08 100644
--- a/sound/soc/pxa/palm27x.c
+++ b/sound/soc/pxa/palm27x.c
@@ -200,7 +200,7 @@ static struct snd_soc_device palm27x_snd_devdata = {
 
 static struct platform_device *palm27x_snd_device;
 
-static int __init palm27x_asoc_init(void)
+static int palm27x_asoc_probe(struct platform_device *pdev)
 {
 	int ret;
 
@@ -208,6 +208,10 @@ static int __init palm27x_asoc_init(void)
 		machine_is_palmld()))
 		return -ENODEV;
 
+	if (pdev->dev.platform_data)
+		palm27x_ep_gpio = ((struct palm27x_asoc_info *)
+			(pdev->dev.platform_data))->jack_gpio;
+
 	ret = gpio_request(palm27x_ep_gpio, "Headphone Jack");
 	if (ret)
 		return ret;
@@ -245,16 +249,31 @@ err_alloc:
 	return ret;
 }
 
-static void __exit palm27x_asoc_exit(void)
+static int __devexit palm27x_asoc_remove(struct platform_device *pdev)
 {
 	free_irq(gpio_to_irq(palm27x_ep_gpio), NULL);
 	gpio_free(palm27x_ep_gpio);
 	platform_device_unregister(palm27x_snd_device);
+	return 0;
 }
 
-void __init palm27x_asoc_set_pdata(struct palm27x_asoc_info *data)
+static struct platform_driver palm27x_wm9712_driver = {
+	.probe		= palm27x_asoc_probe,
+	.remove		= __devexit_p(palm27x_asoc_remove),
+	.driver		= {
+		.name		= "palm27x-asoc",
+		.owner		= THIS_MODULE,
+	},
+};
+
+static int __init palm27x_asoc_init(void)
+{
+	return platform_driver_register(&palm27x_wm9712_driver);
+}
+
+static void __exit palm27x_asoc_exit(void)
 {
-	palm27x_ep_gpio = data->jack_gpio;
+	platform_driver_unregister(&palm27x_wm9712_driver);
 }
 
 module_init(palm27x_asoc_init);
diff --git a/sound/soc/pxa/pxa-ssp.c b/sound/soc/pxa/pxa-ssp.c
index 308a657928d2..286be31545df 100644
--- a/sound/soc/pxa/pxa-ssp.c
+++ b/sound/soc/pxa/pxa-ssp.c
@@ -280,12 +280,33 @@ static int pxa_ssp_resume(struct snd_soc_dai *cpu_dai)
  * ssp_set_clkdiv - set SSP clock divider
  * @div: serial clock rate divider
  */
-static void ssp_set_scr(struct ssp_dev *dev, u32 div)
+static void ssp_set_scr(struct ssp_device *ssp, u32 div)
 {
-	struct ssp_device *ssp = dev->ssp;
-	u32 sscr0 = ssp_read_reg(dev->ssp, SSCR0) & ~SSCR0_SCR;
+	u32 sscr0 = ssp_read_reg(ssp, SSCR0);
+
+	if (cpu_is_pxa25x() && ssp->type == PXA25x_SSP) {
+		sscr0 &= ~0x0000ff00;
+		sscr0 |= ((div - 2)/2) << 8; /* 2..512 */
+	} else {
+		sscr0 &= ~0x000fff00;
+		sscr0 |= (div - 1) << 8;     /* 1..4096 */
+	}
+	ssp_write_reg(ssp, SSCR0, sscr0);
+}
+
+/**
+ * ssp_get_clkdiv - get SSP clock divider
+ */
+static u32 ssp_get_scr(struct ssp_device *ssp)
+{
+	u32 sscr0 = ssp_read_reg(ssp, SSCR0);
+	u32 div;
 
-	ssp_write_reg(ssp, SSCR0, (sscr0 | SSCR0_SerClkDiv(div)));
+	if (cpu_is_pxa25x() && ssp->type == PXA25x_SSP)
+		div = ((sscr0 >> 8) & 0xff) * 2 + 2;
+	else
+		div = ((sscr0 >> 8) & 0xfff) + 1;
+	return div;
 }
 
 /*
@@ -326,7 +347,7 @@ static int pxa_ssp_set_dai_sysclk(struct snd_soc_dai *cpu_dai,
 		break;
 	case PXA_SSP_CLK_AUDIO:
 		priv->sysclk = 0;
-		ssp_set_scr(&priv->dev, 1);
+		ssp_set_scr(ssp, 1);
 		sscr0 |= SSCR0_ACS;
 		break;
 	default:
@@ -387,7 +408,7 @@ static int pxa_ssp_set_dai_clkdiv(struct snd_soc_dai *cpu_dai,
 		ssp_write_reg(ssp, SSACD, val);
 		break;
 	case PXA_SSP_DIV_SCR:
-		ssp_set_scr(&priv->dev, div);
+		ssp_set_scr(ssp, div);
 		break;
 	default:
 		return -ENODEV;
@@ -674,8 +695,7 @@ static int pxa_ssp_hw_params(struct snd_pcm_substream *substream,
 	case SND_SOC_DAIFMT_I2S:
 	       sspsp = ssp_read_reg(ssp, SSPSP);
 
-		if (((sscr0 & SSCR0_SCR) == SSCR0_SerClkDiv(4)) &&
-		     (width == 16)) {
+		if ((ssp_get_scr(ssp) == 4) && (width == 16)) {
 			/* This is a special case where the bitclk is 64fs
 			* and we're not dealing with 2*32 bits of audio
 			* samples.
@@ -806,6 +826,7 @@ static int pxa_ssp_probe(struct platform_device *pdev,
 		goto err_priv;
 	}
 
+	priv->dai_fmt = (unsigned int) -1;
 	dai->private_data = priv;
 
 	return 0;
diff --git a/sound/soc/s3c24xx/Kconfig b/sound/soc/s3c24xx/Kconfig
index 2f3a21eee051..df494d1e346f 100644
--- a/sound/soc/s3c24xx/Kconfig
+++ b/sound/soc/s3c24xx/Kconfig
@@ -1,10 +1,10 @@
 config SND_S3C24XX_SOC
 	tristate "SoC Audio for the Samsung S3CXXXX chips"
-	depends on ARCH_S3C2410 || ARCH_S3C64XX
+	depends on ARCH_S3C2410
 	help
 	  Say Y or M if you want to add support for codecs attached to
-	  the S3C24XX and S3C64XX AC97, I2S or SSP interface. You will
-	  also need to select the audio interfaces to support below.
+	  the S3C24XX AC97 or I2S interfaces. You will also need to
+	  select the audio interfaces to support below.
 
 config SND_S3C24XX_SOC_I2S
 	tristate
diff --git a/sound/soc/s3c24xx/jive_wm8750.c b/sound/soc/s3c24xx/jive_wm8750.c
index 32063790d95b..93e6c87b7399 100644
--- a/sound/soc/s3c24xx/jive_wm8750.c
+++ b/sound/soc/s3c24xx/jive_wm8750.c
@@ -69,8 +69,8 @@ static int jive_hw_params(struct snd_pcm_substream *substream,
 		break;
 	}
 
-	s3c_i2sv2_calc_rate(&div, NULL, params_rate(params),
-			    s3c2412_get_iisclk());
+	s3c_i2sv2_iis_calc_rate(&div, NULL, params_rate(params),
+				s3c2412_get_iisclk());
 
 	/* set codec DAI configuration */
 	ret = snd_soc_dai_set_fmt(codec_dai, SND_SOC_DAIFMT_I2S |
@@ -145,8 +145,9 @@ static struct snd_soc_dai_link jive_dai = {
 };
 
 /* jive audio machine driver */
-static struct snd_soc_machine snd_soc_machine_jive = {
+static struct snd_soc_card snd_soc_machine_jive = {
 	.name		= "Jive",
+	.platform	= &s3c24xx_soc_platform,
 	.dai_link	= &jive_dai,
 	.num_links	= 1,
 };
@@ -157,9 +158,8 @@ static struct wm8750_setup_data jive_wm8750_setup = {
 
 /* jive audio subsystem */
 static struct snd_soc_device jive_snd_devdata = {
-	.machine	= &snd_soc_machine_jive,
-	.platform	= &s3c24xx_soc_platform,
-	.codec_dev	= &soc_codec_dev_wm8750_spi,
+	.card		= &snd_soc_machine_jive,
+	.codec_dev	= &soc_codec_dev_wm8750,
 	.codec_data	= &jive_wm8750_setup,
 };
 
diff --git a/sound/soc/s3c24xx/s3c-i2s-v2.c b/sound/soc/s3c24xx/s3c-i2s-v2.c
index 295a4c910262..ab680aac3fcb 100644
--- a/sound/soc/s3c24xx/s3c-i2s-v2.c
+++ b/sound/soc/s3c24xx/s3c-i2s-v2.c
@@ -473,9 +473,9 @@ static int s3c2412_i2s_set_clkdiv(struct snd_soc_dai *cpu_dai,
 /* default table of all avaialable root fs divisors */
 static unsigned int iis_fs_tab[] = { 256, 512, 384, 768 };
 
-int s3c2412_iis_calc_rate(struct s3c_i2sv2_rate_calc *info,
-			  unsigned int *fstab,
-			  unsigned int rate, struct clk *clk)
+int s3c_i2sv2_iis_calc_rate(struct s3c_i2sv2_rate_calc *info,
+			    unsigned int *fstab,
+			    unsigned int rate, struct clk *clk)
 {
 	unsigned long clkrate = clk_get_rate(clk);
 	unsigned int div;
@@ -531,7 +531,7 @@ int s3c2412_iis_calc_rate(struct s3c_i2sv2_rate_calc *info,
 
 	return 0;
 }
-EXPORT_SYMBOL_GPL(s3c2412_iis_calc_rate);
+EXPORT_SYMBOL_GPL(s3c_i2sv2_iis_calc_rate);
 
 int s3c_i2sv2_probe(struct platform_device *pdev,
 		    struct snd_soc_dai *dai,
@@ -624,15 +624,18 @@ static int s3c2412_i2s_resume(struct snd_soc_dai *dai)
 
 int s3c_i2sv2_register_dai(struct snd_soc_dai *dai)
 {
-	dai->ops.trigger = s3c2412_i2s_trigger;
-	dai->ops.hw_params = s3c2412_i2s_hw_params;
-	dai->ops.set_fmt = s3c2412_i2s_set_fmt;
-	dai->ops.set_clkdiv = s3c2412_i2s_set_clkdiv;
+	struct snd_soc_dai_ops *ops = dai->ops;
+
+	ops->trigger = s3c2412_i2s_trigger;
+	ops->hw_params = s3c2412_i2s_hw_params;
+	ops->set_fmt = s3c2412_i2s_set_fmt;
+	ops->set_clkdiv = s3c2412_i2s_set_clkdiv;
 
 	dai->suspend = s3c2412_i2s_suspend;
 	dai->resume = s3c2412_i2s_resume;
 
 	return snd_soc_register_dai(dai);
 }
-
 EXPORT_SYMBOL_GPL(s3c_i2sv2_register_dai);
+
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/s3c24xx/s3c2412-i2s.c b/sound/soc/s3c24xx/s3c2412-i2s.c
index 1ca3cdaa8213..b7e0b3f0bfc8 100644
--- a/sound/soc/s3c24xx/s3c2412-i2s.c
+++ b/sound/soc/s3c24xx/s3c2412-i2s.c
@@ -33,8 +33,8 @@
 
 #include <plat/regs-s3c2412-iis.h>
 
-#include <plat/regs-gpio.h>
 #include <plat/audio.h>
+#include <mach/regs-gpio.h>
 #include <mach/dma.h>
 
 #include "s3c24xx-pcm.h"
diff --git a/sound/soc/sh/dma-sh7760.c b/sound/soc/sh/dma-sh7760.c
index 0dad3a0bb920..baddb1242c71 100644
--- a/sound/soc/sh/dma-sh7760.c
+++ b/sound/soc/sh/dma-sh7760.c
@@ -103,7 +103,8 @@ static struct snd_pcm_hardware camelot_pcm_hardware = {
 	.info = (SNDRV_PCM_INFO_MMAP |
 		SNDRV_PCM_INFO_INTERLEAVED |
 		SNDRV_PCM_INFO_BLOCK_TRANSFER |
-		SNDRV_PCM_INFO_MMAP_VALID),
+		SNDRV_PCM_INFO_MMAP_VALID |
+		SNDRV_PCM_INFO_BATCH),
 	.formats =	DMABRG_FMTS,
 	.rates =	DMABRG_RATES,
 	.rate_min =		8000,