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: (294 commits)
  S3C64XX: Staticise platform data for PCM devices
  ASoC: Rename controls with a / in wm_hubs
  snd-fm801: autodetect SF64-PCR (tuner-only) card
  ALSA: tea575x-tuner: fix mute
  ASoC: au1x: dbdma2: plug memleak in pcm device creation error path
  ASoC: au1x: dbdma2: fix oops on soc device removal.
  ALSA: hda - Fix memory leaks in the previous patch
  ALSA: hda - Add ALC661/259, ALC892/888VD support
  ALSA: opti9xx: remove snd_opti9xx fields
  ALSA: aaci - Clean up duplicate code
  ALSA: usb - Fix mixer map for Hercules Gamesurround Muse Pocket LT
  ALSA: hda - Add position_fix quirk for HP dv3
  ALSA: hda - Add a pin-fix for FSC Amilo Pi1505
  ALSA: hda - Fix Cxt5047 test mode
  ASoC: pxa/raumfeld: adopt new snd_soc_dai_set_pll() API
  ASoC: sh: fsi: Add runtime PM support
  sh: ms7724se: Add runtime PM support for FSI
  ALSA: hda - Add a position_fix quirk for MSI Wind U115
  ALSA: opti-miro: add PnP detection
  ALSA: opti-miro: separate comon probing code
  ...

9 files changed, 432 insertions, 47 deletions

diff --git a/sound/soc/omap/Kconfig b/sound/soc/omap/Kconfig
index 653a362425d..61952aa6cd5 100644
--- a/sound/soc/omap/Kconfig
+++ b/sound/soc/omap/Kconfig
@@ -43,12 +43,13 @@ config SND_OMAP_SOC_OSK5912
 	  Say Y if you want to add support for SoC audio on osk5912.
 
 config SND_OMAP_SOC_OVERO
-	tristate "SoC Audio support for Gumstix Overo"
-	depends on TWL4030_CORE && SND_OMAP_SOC && MACH_OVERO
+	tristate "SoC Audio support for Gumstix Overo and CompuLab CM-T35"
+	depends on TWL4030_CORE && SND_OMAP_SOC && (MACH_OVERO || MACH_CM_T35)
 	select SND_OMAP_SOC_MCBSP
 	select SND_SOC_TWL4030
 	help
-	  Say Y if you want to add support for SoC audio on the Gumstix Overo.
+	  Say Y if you want to add support for SoC audio on the
+	  Gumstix Overo or CompuLab CM-T35
 
 config SND_OMAP_SOC_OMAP2EVM
 	tristate "SoC Audio support for OMAP2EVM board"
@@ -66,6 +67,15 @@ config SND_OMAP_SOC_OMAP3EVM
 	help
 	  Say Y if you want to add support for SoC audio on the omap3evm board.
 
+config SND_OMAP_SOC_AM3517EVM
+	tristate "SoC Audio support for OMAP3517 / AM3517 EVM"
+	depends on SND_OMAP_SOC && MACH_OMAP3517EVM && I2C
+	select SND_OMAP_SOC_MCBSP
+	select SND_SOC_TLV320AIC23
+	help
+	  Say Y if you want to add support for SoC audio on the OMAP3517 / AM3517
+	  EVM.
+
 config SND_OMAP_SOC_SDP3430
 	tristate "SoC Audio support for Texas Instruments SDP3430"
 	depends on TWL4030_CORE && SND_OMAP_SOC && MACH_OMAP_3430SDP
@@ -99,3 +109,10 @@ config SND_OMAP_SOC_ZOOM2
 	help
 	  Say Y if you want to add support for Soc audio on Zoom2 board.
 
+config SND_OMAP_SOC_IGEP0020
+	tristate "SoC Audio support for IGEP v2"
+	depends on TWL4030_CORE && SND_OMAP_SOC && MACH_IGEP0020
+	select SND_OMAP_SOC_MCBSP
+	select SND_SOC_TWL4030
+	help
+	  Say Y if you want to add support for Soc audio on IGEP v2 board.
diff --git a/sound/soc/omap/Makefile b/sound/soc/omap/Makefile
index 02d69471dcb..d49458a29bb 100644
--- a/sound/soc/omap/Makefile
+++ b/sound/soc/omap/Makefile
@@ -12,10 +12,12 @@ snd-soc-osk5912-objs := osk5912.o
 snd-soc-overo-objs := overo.o
 snd-soc-omap2evm-objs := omap2evm.o
 snd-soc-omap3evm-objs := omap3evm.o
+snd-soc-am3517evm-objs := am3517evm.o
 snd-soc-sdp3430-objs := sdp3430.o
 snd-soc-omap3pandora-objs := omap3pandora.o
 snd-soc-omap3beagle-objs := omap3beagle.o
 snd-soc-zoom2-objs := zoom2.o
+snd-soc-igep0020-objs := igep0020.o
 
 obj-$(CONFIG_SND_OMAP_SOC_N810) += snd-soc-n810.o
 obj-$(CONFIG_SND_OMAP_SOC_AMS_DELTA) += snd-soc-ams-delta.o
@@ -23,7 +25,9 @@ obj-$(CONFIG_SND_OMAP_SOC_OSK5912) += snd-soc-osk5912.o
 obj-$(CONFIG_SND_OMAP_SOC_OVERO) += snd-soc-overo.o
 obj-$(CONFIG_MACH_OMAP2EVM) += snd-soc-omap2evm.o
 obj-$(CONFIG_MACH_OMAP3EVM) += snd-soc-omap3evm.o
+obj-$(CONFIG_MACH_OMAP3517EVM) += snd-soc-am3517evm.o
 obj-$(CONFIG_SND_OMAP_SOC_SDP3430) += snd-soc-sdp3430.o
 obj-$(CONFIG_SND_OMAP_SOC_OMAP3_PANDORA) += snd-soc-omap3pandora.o
 obj-$(CONFIG_SND_OMAP_SOC_OMAP3_BEAGLE) += snd-soc-omap3beagle.o
 obj-$(CONFIG_SND_OMAP_SOC_ZOOM2) += snd-soc-zoom2.o
+obj-$(CONFIG_SND_OMAP_SOC_IGEP0020) += snd-soc-igep0020.o
diff --git a/sound/soc/omap/am3517evm.c b/sound/soc/omap/am3517evm.c
new file mode 100644
index 00000000000..135901b2ea1
--- /dev/null
+++ b/sound/soc/omap/am3517evm.c
@@ -0,0 +1,202 @@
+/*
+ * am3517evm.c  -- ALSA SoC support for OMAP3517 / AM3517 EVM
+ *
+ * Author: Anuj Aggarwal <anuj.aggarwal@ti.com>
+ *
+ * Based on sound/soc/omap/beagle.c by Steve Sakoman
+ *
+ * Copyright (C) 2009 Texas Instruments Incorporated
+ *
+ * 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 version 2.
+ *
+ * This program is distributed "as is" WITHOUT ANY WARRANTY of any kind,
+ * whether express or implied; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * General Public License for more details.
+ */
+
+#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 <mach/gpio.h>
+#include <plat/mcbsp.h>
+
+#include "omap-mcbsp.h"
+#include "omap-pcm.h"
+
+#include "../codecs/tlv320aic23.h"
+
+#define CODEC_CLOCK 	12000000
+
+static int am3517evm_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 ret;
+
+	/* Set codec DAI configuration */
+	ret = snd_soc_dai_set_fmt(codec_dai,
+				  SND_SOC_DAIFMT_DSP_B |
+				  SND_SOC_DAIFMT_NB_NF |
+				  SND_SOC_DAIFMT_CBM_CFM);
+	if (ret < 0) {
+		printk(KERN_ERR "can't set codec DAI configuration\n");
+		return ret;
+	}
+
+	/* Set cpu DAI configuration */
+	ret = snd_soc_dai_set_fmt(cpu_dai,
+				  SND_SOC_DAIFMT_DSP_B |
+				  SND_SOC_DAIFMT_NB_NF |
+				  SND_SOC_DAIFMT_CBM_CFM);
+	if (ret < 0) {
+		printk(KERN_ERR "can't set cpu DAI configuration\n");
+		return ret;
+	}
+
+	/* Set the codec system clock for DAC and ADC */
+	ret = snd_soc_dai_set_sysclk(codec_dai, 0,
+			CODEC_CLOCK, SND_SOC_CLOCK_IN);
+	if (ret < 0) {
+		printk(KERN_ERR "can't set codec system clock\n");
+		return ret;
+	}
+
+	ret = snd_soc_dai_set_sysclk(cpu_dai, OMAP_MCBSP_CLKR_SRC_CLKX, 0,
+				SND_SOC_CLOCK_IN);
+	if (ret < 0) {
+		printk(KERN_ERR "can't set CPU system clock OMAP_MCBSP_CLKR_SRC_CLKX\n");
+		return ret;
+	}
+
+	snd_soc_dai_set_sysclk(cpu_dai, OMAP_MCBSP_FSR_SRC_FSX, 0,
+				SND_SOC_CLOCK_IN);
+	if (ret < 0) {
+		printk(KERN_ERR "can't set CPU system clock OMAP_MCBSP_FSR_SRC_FSX\n");
+		return ret;
+	}
+
+	return 0;
+}
+
+static struct snd_soc_ops am3517evm_ops = {
+	.hw_params = am3517evm_hw_params,
+};
+
+/* am3517evm machine dapm widgets */
+static const struct snd_soc_dapm_widget tlv320aic23_dapm_widgets[] = {
+	SND_SOC_DAPM_HP("Line Out", NULL),
+	SND_SOC_DAPM_LINE("Line In", NULL),
+	SND_SOC_DAPM_MIC("Mic In", NULL),
+};
+
+static const struct snd_soc_dapm_route audio_map[] = {
+	/* Line Out connected to LLOUT, RLOUT */
+	{"Line Out", NULL, "LOUT"},
+	{"Line Out", NULL, "ROUT"},
+
+	{"LLINEIN", NULL, "Line In"},
+	{"RLINEIN", NULL, "Line In"},
+
+	{"MICIN", NULL, "Mic In"},
+};
+
+static int am3517evm_aic23_init(struct snd_soc_codec *codec)
+{
+	/* Add am3517-evm specific widgets */
+	snd_soc_dapm_new_controls(codec, tlv320aic23_dapm_widgets,
+				  ARRAY_SIZE(tlv320aic23_dapm_widgets));
+
+	/* Set up davinci-evm specific audio path audio_map */
+	snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map));
+
+	/* always connected */
+	snd_soc_dapm_enable_pin(codec, "Line Out");
+	snd_soc_dapm_enable_pin(codec, "Line In");
+	snd_soc_dapm_enable_pin(codec, "Mic In");
+
+	snd_soc_dapm_sync(codec);
+
+	return 0;
+}
+
+/* Digital audio interface glue - connects codec <--> CPU */
+static struct snd_soc_dai_link am3517evm_dai = {
+	.name = "TLV320AIC23",
+	.stream_name = "AIC23",
+	.cpu_dai = &omap_mcbsp_dai[0],
+	.codec_dai = &tlv320aic23_dai,
+	.init = am3517evm_aic23_init,
+	.ops = &am3517evm_ops,
+};
+
+/* Audio machine driver */
+static struct snd_soc_card snd_soc_am3517evm = {
+	.name = "am3517evm",
+	.platform = &omap_soc_platform,
+	.dai_link = &am3517evm_dai,
+	.num_links = 1,
+};
+
+/* Audio subsystem */
+static struct snd_soc_device am3517evm_snd_devdata = {
+	.card = &snd_soc_am3517evm,
+	.codec_dev = &soc_codec_dev_tlv320aic23,
+};
+
+static struct platform_device *am3517evm_snd_device;
+
+static int __init am3517evm_soc_init(void)
+{
+	int ret;
+
+	if (!machine_is_omap3517evm()) {
+		pr_err("Not OMAP3517 / AM3517 EVM!\n");
+		return -ENODEV;
+	}
+	pr_info("OMAP3517 / AM3517 EVM SoC init\n");
+
+	am3517evm_snd_device = platform_device_alloc("soc-audio", -1);
+	if (!am3517evm_snd_device) {
+		printk(KERN_ERR "Platform device allocation failed\n");
+		return -ENOMEM;
+	}
+
+	platform_set_drvdata(am3517evm_snd_device, &am3517evm_snd_devdata);
+	am3517evm_snd_devdata.dev = &am3517evm_snd_device->dev;
+	*(unsigned int *)am3517evm_dai.cpu_dai->private_data = 0; /* McBSP1 */
+
+	ret = platform_device_add(am3517evm_snd_device);
+	if (ret)
+		goto err1;
+
+	return 0;
+
+err1:
+	printk(KERN_ERR "Unable to add platform device\n");
+	platform_device_put(am3517evm_snd_device);
+
+	return ret;
+}
+
+static void __exit am3517evm_soc_exit(void)
+{
+	platform_device_unregister(am3517evm_snd_device);
+}
+
+module_init(am3517evm_soc_init);
+module_exit(am3517evm_soc_exit);
+
+MODULE_AUTHOR("Anuj Aggarwal <anuj.aggarwal@ti.com>");
+MODULE_DESCRIPTION("ALSA SoC OMAP3517 / AM3517 EVM");
+MODULE_LICENSE("GPL v2");
diff --git a/sound/soc/omap/ams-delta.c b/sound/soc/omap/ams-delta.c
index 5a5166ac727..ae0fc9b135d 100644
--- a/sound/soc/omap/ams-delta.c
+++ b/sound/soc/omap/ams-delta.c
@@ -40,7 +40,7 @@
 
 
 /* Board specific DAPM widgets */
- const struct snd_soc_dapm_widget ams_delta_dapm_widgets[] = {
+static const struct snd_soc_dapm_widget ams_delta_dapm_widgets[] = {
 	/* Handset */
 	SND_SOC_DAPM_MIC("Mouthpiece", NULL),
 	SND_SOC_DAPM_HP("Earpiece", NULL),
@@ -81,7 +81,7 @@ static const char *ams_delta_audio_mode[] =
 						(1 << AMS_DELTA_SPEAKER))
 #define AMS_DELTA_SPEAKERPHONE	(AMS_DELTA_HANDSFREE | (1 << AMS_DELTA_AGC))
 
-unsigned short ams_delta_audio_mode_pins[] = {
+static const unsigned short ams_delta_audio_mode_pins[] = {
 	AMS_DELTA_MIXED,
 	AMS_DELTA_HANDSET,
 	AMS_DELTA_HANDSFREE,
diff --git a/sound/soc/omap/igep0020.c b/sound/soc/omap/igep0020.c
new file mode 100644
index 00000000000..3583c429f9b
--- /dev/null
+++ b/sound/soc/omap/igep0020.c
@@ -0,0 +1,148 @@
+/*
+ * igep0020.c  --  SoC audio for IGEP v2
+ *
+ * Based on sound/soc/omap/overo.c by Steve Sakoman
+ *
+ * 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 <mach/gpio.h>
+#include <plat/mcbsp.h>
+
+#include "omap-mcbsp.h"
+#include "omap-pcm.h"
+#include "../codecs/twl4030.h"
+
+static int igep2_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 ret;
+
+	/* Set codec DAI configuration */
+	ret = snd_soc_dai_set_fmt(codec_dai,
+				  SND_SOC_DAIFMT_I2S |
+				  SND_SOC_DAIFMT_NB_NF |
+				  SND_SOC_DAIFMT_CBM_CFM);
+	if (ret < 0) {
+		printk(KERN_ERR "can't set codec DAI configuration\n");
+		return ret;
+	}
+
+	/* Set cpu DAI configuration */
+	ret = snd_soc_dai_set_fmt(cpu_dai,
+				  SND_SOC_DAIFMT_I2S |
+				  SND_SOC_DAIFMT_NB_NF |
+				  SND_SOC_DAIFMT_CBM_CFM);
+	if (ret < 0) {
+		printk(KERN_ERR "can't set cpu DAI configuration\n");
+		return ret;
+	}
+
+	/* Set the codec system clock for DAC and ADC */
+	ret = snd_soc_dai_set_sysclk(codec_dai, 0, 26000000,
+					    SND_SOC_CLOCK_IN);
+	if (ret < 0) {
+		printk(KERN_ERR "can't set codec system clock\n");
+		return ret;
+	}
+
+	return 0;
+}
+
+static struct snd_soc_ops igep2_ops = {
+	.hw_params = igep2_hw_params,
+};
+
+/* Digital audio interface glue - connects codec <--> CPU */
+static struct snd_soc_dai_link igep2_dai = {
+	.name = "TWL4030",
+	.stream_name = "TWL4030",
+	.cpu_dai = &omap_mcbsp_dai[0],
+	.codec_dai = &twl4030_dai[TWL4030_DAI_HIFI],
+	.ops = &igep2_ops,
+};
+
+/* Audio machine driver */
+static struct snd_soc_card snd_soc_card_igep2 = {
+	.name = "igep2",
+	.platform = &omap_soc_platform,
+	.dai_link = &igep2_dai,
+	.num_links = 1,
+};
+
+/* Audio subsystem */
+static struct snd_soc_device igep2_snd_devdata = {
+	.card = &snd_soc_card_igep2,
+	.codec_dev = &soc_codec_dev_twl4030,
+};
+
+static struct platform_device *igep2_snd_device;
+
+static int __init igep2_soc_init(void)
+{
+	int ret;
+
+	if (!machine_is_igep0020()) {
+		pr_debug("Not IGEP v2!\n");
+		return -ENODEV;
+	}
+	printk(KERN_INFO "IGEP v2 SoC init\n");
+
+	igep2_snd_device = platform_device_alloc("soc-audio", -1);
+	if (!igep2_snd_device) {
+		printk(KERN_ERR "Platform device allocation failed\n");
+		return -ENOMEM;
+	}
+
+	platform_set_drvdata(igep2_snd_device, &igep2_snd_devdata);
+	igep2_snd_devdata.dev = &igep2_snd_device->dev;
+	*(unsigned int *)igep2_dai.cpu_dai->private_data = 1; /* McBSP2 */
+
+	ret = platform_device_add(igep2_snd_device);
+	if (ret)
+		goto err1;
+
+	return 0;
+
+err1:
+	printk(KERN_ERR "Unable to add platform device\n");
+	platform_device_put(igep2_snd_device);
+
+	return ret;
+}
+module_init(igep2_soc_init);
+
+static void __exit igep2_soc_exit(void)
+{
+	platform_device_unregister(igep2_snd_device);
+}
+module_exit(igep2_soc_exit);
+
+MODULE_AUTHOR("Enric Balletbo i Serra <eballetbo@iseebcn.com>");
+MODULE_DESCRIPTION("ALSA SoC IGEP v2");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/omap/omap-mcbsp.c b/sound/soc/omap/omap-mcbsp.c
index 3341f49402c..45be94201c8 100644
--- a/sound/soc/omap/omap-mcbsp.c
+++ b/sound/soc/omap/omap-mcbsp.c
@@ -49,6 +49,8 @@ struct omap_mcbsp_data {
 	 */
 	int				active;
 	int				configured;
+	unsigned int			in_freq;
+	int				clk_div;
 };
 
 #define to_mcbsp(priv)	container_of((priv), struct omap_mcbsp_data, bus_id)
@@ -257,7 +259,7 @@ static int omap_mcbsp_dai_hw_params(struct snd_pcm_substream *substream,
 	int dma, bus_id = mcbsp_data->bus_id, id = cpu_dai->id;
 	int wlen, channels, wpf, sync_mode = OMAP_DMA_SYNC_ELEMENT;
 	unsigned long port;
-	unsigned int format;
+	unsigned int format, div, framesize, master;
 
 	if (cpu_class_is_omap1()) {
 		dma = omap1_dma_reqs[bus_id][substream->stream];
@@ -294,28 +296,19 @@ static int omap_mcbsp_dai_hw_params(struct snd_pcm_substream *substream,
 
 	format = mcbsp_data->fmt & SND_SOC_DAIFMT_FORMAT_MASK;
 	wpf = channels = params_channels(params);
-	switch (channels) {
-	case 2:
-		if (format == SND_SOC_DAIFMT_I2S) {
-			/* Use dual-phase frames */
-			regs->rcr2	|= RPHASE;
-			regs->xcr2	|= XPHASE;
-			/* Set 1 word per (McBSP) frame for phase1 and phase2 */
-			wpf--;
-			regs->rcr2	|= RFRLEN2(wpf - 1);
-			regs->xcr2	|= XFRLEN2(wpf - 1);
-		}
-	case 1:
-	case 4:
-		/* Set word per (McBSP) frame for phase1 */
-		regs->rcr1	|= RFRLEN1(wpf - 1);
-		regs->xcr1	|= XFRLEN1(wpf - 1);
-		break;
-	default:
-		/* Unsupported number of channels */
-		return -EINVAL;
+	if (channels == 2 && format == SND_SOC_DAIFMT_I2S) {
+		/* Use dual-phase frames */
+		regs->rcr2	|= RPHASE;
+		regs->xcr2	|= XPHASE;
+		/* Set 1 word per (McBSP) frame for phase1 and phase2 */
+		wpf--;
+		regs->rcr2	|= RFRLEN2(wpf - 1);
+		regs->xcr2	|= XFRLEN2(wpf - 1);
 	}
 
+	regs->rcr1	|= RFRLEN1(wpf - 1);
+	regs->xcr1	|= XFRLEN1(wpf - 1);
+
 	switch (params_format(params)) {
 	case SNDRV_PCM_FORMAT_S16_LE:
 		/* Set word lengths */
@@ -330,15 +323,30 @@ static int omap_mcbsp_dai_hw_params(struct snd_pcm_substream *substream,
 		return -EINVAL;
 	}
 
+	/* In McBSP master modes, FRAME (i.e. sample rate) is generated
+	 * by _counting_ BCLKs. Calculate frame size in BCLKs */
+	master = mcbsp_data->fmt & SND_SOC_DAIFMT_MASTER_MASK;
+	if (master ==	SND_SOC_DAIFMT_CBS_CFS) {
+		div = mcbsp_data->clk_div ? mcbsp_data->clk_div : 1;
+		framesize = (mcbsp_data->in_freq / div) / params_rate(params);
+
+		if (framesize < wlen * channels) {
+			printk(KERN_ERR "%s: not enough bandwidth for desired rate and "
+					"channels\n", __func__);
+			return -EINVAL;
+		}
+	} else
+		framesize = wlen * channels;
+
 	/* Set FS period and length in terms of bit clock periods */
 	switch (format) {
 	case SND_SOC_DAIFMT_I2S:
-		regs->srgr2	|= FPER(wlen * channels - 1);
-		regs->srgr1	|= FWID(wlen - 1);
+		regs->srgr2	|= FPER(framesize - 1);
+		regs->srgr1	|= FWID((framesize >> 1) - 1);
 		break;
 	case SND_SOC_DAIFMT_DSP_A:
 	case SND_SOC_DAIFMT_DSP_B:
-		regs->srgr2	|= FPER(wlen * channels - 1);
+		regs->srgr2	|= FPER(framesize - 1);
 		regs->srgr1	|= FWID(0);
 		break;
 	}
@@ -454,6 +462,7 @@ static int omap_mcbsp_dai_set_clkdiv(struct snd_soc_dai *cpu_dai,
 	if (div_id != OMAP_MCBSP_CLKGDV)
 		return -ENODEV;
 
+	mcbsp_data->clk_div = div;
 	regs->srgr1	|= CLKGDV(div - 1);
 
 	return 0;
@@ -554,6 +563,8 @@ static int omap_mcbsp_dai_set_dai_sysclk(struct snd_soc_dai *cpu_dai,
 	struct omap_mcbsp_reg_cfg *regs = &mcbsp_data->regs;
 	int err = 0;
 
+	mcbsp_data->in_freq = freq;
+
 	switch (clk_id) {
 	case OMAP_MCBSP_SYSCLK_CLK:
 		regs->srgr2	|= CLKSM;
@@ -598,13 +609,13 @@ static struct snd_soc_dai_ops omap_mcbsp_dai_ops = {
 	.id = (link_id),					\
 	.playback = {						\
 		.channels_min = 1,				\
-		.channels_max = 4,				\
+		.channels_max = 16,				\
 		.rates = OMAP_MCBSP_RATES,			\
 		.formats = SNDRV_PCM_FMTBIT_S16_LE,		\
 	},							\
 	.capture = {						\
 		.channels_min = 1,				\
-		.channels_max = 4,				\
+		.channels_max = 16,				\
 		.rates = OMAP_MCBSP_RATES,			\
 		.formats = SNDRV_PCM_FMTBIT_S16_LE,		\
 	},							\
diff --git a/sound/soc/omap/omap3evm.c b/sound/soc/omap/omap3evm.c
index 13aa380de16..f484dcd6340 100644
--- a/sound/soc/omap/omap3evm.c
+++ b/sound/soc/omap/omap3evm.c
@@ -93,10 +93,17 @@ static struct snd_soc_card snd_soc_omap3evm = {
 	.num_links = 1,
 };
 
+/* twl4030 setup */
+static struct twl4030_setup_data twl4030_setup = {
+	.ramp_delay_value = 4,
+	.sysclk = 26000,
+};
+
 /* Audio subsystem */
 static struct snd_soc_device omap3evm_snd_devdata = {
 	.card = &snd_soc_omap3evm,
 	.codec_dev = &soc_codec_dev_twl4030,
+	.codec_data = &twl4030_setup,
 };
 
 static struct platform_device *omap3evm_snd_device;
diff --git a/sound/soc/omap/omap3pandora.c b/sound/soc/omap/omap3pandora.c
index 0cd06f5dd35..71b2c161158 100644
--- a/sound/soc/omap/omap3pandora.c
+++ b/sound/soc/omap/omap3pandora.c
@@ -40,9 +40,12 @@
 
 #define PREFIX "ASoC omap3pandora: "
 
-static int omap3pandora_cmn_hw_params(struct snd_soc_dai *codec_dai,
-	struct snd_soc_dai *cpu_dai, unsigned int fmt)
+static int omap3pandora_cmn_hw_params(struct snd_pcm_substream *substream,
+	struct snd_pcm_hw_params *params, unsigned int fmt)
 {
+	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 ret;
 
 	/* Set codec DAI configuration */
@@ -68,8 +71,9 @@ static int omap3pandora_cmn_hw_params(struct snd_soc_dai *codec_dai,
 	}
 
 	/* Set McBSP clock to external */
-	ret = snd_soc_dai_set_sysclk(cpu_dai, OMAP_MCBSP_SYSCLK_CLKS_EXT, 0,
-					    SND_SOC_CLOCK_IN);
+	ret = snd_soc_dai_set_sysclk(cpu_dai, OMAP_MCBSP_SYSCLK_CLKS_EXT,
+				     256 * params_rate(params),
+				     SND_SOC_CLOCK_IN);
 	if (ret < 0) {
 		pr_err(PREFIX "can't set cpu system clock\n");
 		return ret;
@@ -87,11 +91,7 @@ static int omap3pandora_cmn_hw_params(struct snd_soc_dai *codec_dai,
 static int omap3pandora_out_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;
-
-	return omap3pandora_cmn_hw_params(codec_dai, cpu_dai,
+	return omap3pandora_cmn_hw_params(substream, params,
 					  SND_SOC_DAIFMT_I2S |
 					  SND_SOC_DAIFMT_IB_NF |
 					  SND_SOC_DAIFMT_CBS_CFS);
@@ -100,11 +100,7 @@ static int omap3pandora_out_hw_params(struct snd_pcm_substream *substream,
 static int omap3pandora_in_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;
-
-	return omap3pandora_cmn_hw_params(codec_dai, cpu_dai,
+	return omap3pandora_cmn_hw_params(substream, params,
 					  SND_SOC_DAIFMT_I2S |
 					  SND_SOC_DAIFMT_NB_NF |
 					  SND_SOC_DAIFMT_CBS_CFS);
diff --git a/sound/soc/omap/overo.c b/sound/soc/omap/overo.c
index ec4f8fd8b3a..97a4d6308bd 100644
--- a/sound/soc/omap/overo.c
+++ b/sound/soc/omap/overo.c
@@ -107,8 +107,8 @@ static int __init overo_soc_init(void)
 {
 	int ret;
 
-	if (!machine_is_overo()) {
-		pr_debug("Not Overo!\n");
+	if (!(machine_is_overo() || machine_is_cm_t35())) {
+		pr_debug("Incomatible machine!\n");
 		return -ENODEV;
 	}
 	printk(KERN_INFO "overo SoC init\n");