Merge branch 'linus' into tracing/hw-breakpoints

Conflicts:
	arch/x86/kernel/process_64.c

Semantic conflict fixed in:
	arch/x86/kvm/x86.c

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

10 files changed, 1159 insertions, 32 deletions

diff --git a/sound/soc/omap/Kconfig b/sound/soc/omap/Kconfig
index b771238662b..2dee9839be8 100644
--- a/sound/soc/omap/Kconfig
+++ b/sound/soc/omap/Kconfig
@@ -15,6 +15,14 @@ config SND_OMAP_SOC_N810
 	help
 	  Say Y if you want to add support for SoC audio on Nokia N810.
 
+config SND_OMAP_SOC_AMS_DELTA
+	tristate "SoC Audio support for Amstrad E3 (Delta) videophone"
+	depends on SND_OMAP_SOC && MACH_AMS_DELTA
+	select SND_OMAP_SOC_MCBSP
+	select SND_SOC_CX20442
+	help
+	  Say Y if you want to add support for SoC audio on Amstrad Delta.
+
 config SND_OMAP_SOC_OSK5912
 	tristate "SoC Audio support for omap osk5912"
 	depends on SND_OMAP_SOC && MACH_OMAP_OSK && I2C
@@ -72,4 +80,11 @@ config SND_OMAP_SOC_OMAP3_BEAGLE
 	help
 	  Say Y if you want to add support for SoC audio on the Beagleboard.
 
+config SND_OMAP_SOC_ZOOM2
+	tristate "SoC Audio support for Zoom2"
+	depends on TWL4030_CORE && SND_OMAP_SOC && MACH_OMAP_ZOOM2
+	select SND_OMAP_SOC_MCBSP
+	select SND_SOC_TWL4030
+	help
+	  Say Y if you want to add support for Soc audio on Zoom2 board.
 
diff --git a/sound/soc/omap/Makefile b/sound/soc/omap/Makefile
index a37f4986238..02d69471dcb 100644
--- a/sound/soc/omap/Makefile
+++ b/sound/soc/omap/Makefile
@@ -7,6 +7,7 @@ obj-$(CONFIG_SND_OMAP_SOC_MCBSP) += snd-soc-omap-mcbsp.o
 
 # OMAP Machine Support
 snd-soc-n810-objs := n810.o
+snd-soc-ams-delta-objs := ams-delta.o
 snd-soc-osk5912-objs := osk5912.o
 snd-soc-overo-objs := overo.o
 snd-soc-omap2evm-objs := omap2evm.o
@@ -14,8 +15,10 @@ snd-soc-omap3evm-objs := omap3evm.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
 
 obj-$(CONFIG_SND_OMAP_SOC_N810) += snd-soc-n810.o
+obj-$(CONFIG_SND_OMAP_SOC_AMS_DELTA) += snd-soc-ams-delta.o
 obj-$(CONFIG_SND_OMAP_SOC_OSK5912) += snd-soc-osk5912.o
 obj-$(CONFIG_SND_OMAP_SOC_OVERO) += snd-soc-overo.o
 obj-$(CONFIG_MACH_OMAP2EVM) += snd-soc-omap2evm.o
@@ -23,3 +26,4 @@ obj-$(CONFIG_MACH_OMAP3EVM) += snd-soc-omap3evm.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
diff --git a/sound/soc/omap/ams-delta.c b/sound/soc/omap/ams-delta.c
new file mode 100644
index 00000000000..5a5166ac727
--- /dev/null
+++ b/sound/soc/omap/ams-delta.c
@@ -0,0 +1,646 @@
+/*
+ * ams-delta.c  --  SoC audio for Amstrad E3 (Delta) videophone
+ *
+ * Copyright (C) 2009 Janusz Krzysztofik <jkrzyszt@tis.icnet.pl>
+ *
+ * Initially based on sound/soc/omap/osk5912.x
+ * Copyright (C) 2008 Mistral Solutions
+ *
+ * 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/gpio.h>
+#include <linux/spinlock.h>
+#include <linux/tty.h>
+
+#include <sound/soc-dapm.h>
+#include <sound/jack.h>
+
+#include <asm/mach-types.h>
+
+#include <mach/board-ams-delta.h>
+#include <mach/mcbsp.h>
+
+#include "omap-mcbsp.h"
+#include "omap-pcm.h"
+#include "../codecs/cx20442.h"
+
+
+/* Board specific DAPM widgets */
+ const struct snd_soc_dapm_widget ams_delta_dapm_widgets[] = {
+	/* Handset */
+	SND_SOC_DAPM_MIC("Mouthpiece", NULL),
+	SND_SOC_DAPM_HP("Earpiece", NULL),
+	/* Handsfree/Speakerphone */
+	SND_SOC_DAPM_MIC("Microphone", NULL),
+	SND_SOC_DAPM_SPK("Speaker", NULL),
+};
+
+/* How they are connected to codec pins */
+static const struct snd_soc_dapm_route ams_delta_audio_map[] = {
+	{"TELIN", NULL, "Mouthpiece"},
+	{"Earpiece", NULL, "TELOUT"},
+
+	{"MIC", NULL, "Microphone"},
+	{"Speaker", NULL, "SPKOUT"},
+};
+
+/*
+ * Controls, functional after the modem line discipline is activated.
+ */
+
+/* Virtual switch: audio input/output constellations */
+static const char *ams_delta_audio_mode[] =
+	{"Mixed", "Handset", "Handsfree", "Speakerphone"};
+
+/* Selection <-> pin translation */
+#define AMS_DELTA_MOUTHPIECE	0
+#define AMS_DELTA_EARPIECE	1
+#define AMS_DELTA_MICROPHONE	2
+#define AMS_DELTA_SPEAKER	3
+#define AMS_DELTA_AGC		4
+
+#define AMS_DELTA_MIXED		((1 << AMS_DELTA_EARPIECE) | \
+						(1 << AMS_DELTA_MICROPHONE))
+#define AMS_DELTA_HANDSET	((1 << AMS_DELTA_MOUTHPIECE) | \
+						(1 << AMS_DELTA_EARPIECE))
+#define AMS_DELTA_HANDSFREE	((1 << AMS_DELTA_MICROPHONE) | \
+						(1 << AMS_DELTA_SPEAKER))
+#define AMS_DELTA_SPEAKERPHONE	(AMS_DELTA_HANDSFREE | (1 << AMS_DELTA_AGC))
+
+unsigned short ams_delta_audio_mode_pins[] = {
+	AMS_DELTA_MIXED,
+	AMS_DELTA_HANDSET,
+	AMS_DELTA_HANDSFREE,
+	AMS_DELTA_SPEAKERPHONE,
+};
+
+static unsigned short ams_delta_audio_agc;
+
+static int ams_delta_set_audio_mode(struct snd_kcontrol *kcontrol,
+					struct snd_ctl_elem_value *ucontrol)
+{
+	struct snd_soc_codec *codec =  snd_kcontrol_chip(kcontrol);
+	struct soc_enum *control = (struct soc_enum *)kcontrol->private_value;
+	unsigned short pins;
+	int pin, changed = 0;
+
+	/* Refuse any mode changes if we are not able to control the codec. */
+	if (!codec->control_data)
+		return -EUNATCH;
+
+	if (ucontrol->value.enumerated.item[0] >= control->max)
+		return -EINVAL;
+
+	mutex_lock(&codec->mutex);
+
+	/* Translate selection to bitmap */
+	pins = ams_delta_audio_mode_pins[ucontrol->value.enumerated.item[0]];
+
+	/* Setup pins after corresponding bits if changed */
+	pin = !!(pins & (1 << AMS_DELTA_MOUTHPIECE));
+	if (pin != snd_soc_dapm_get_pin_status(codec, "Mouthpiece")) {
+		changed = 1;
+		if (pin)
+			snd_soc_dapm_enable_pin(codec, "Mouthpiece");
+		else
+			snd_soc_dapm_disable_pin(codec, "Mouthpiece");
+	}
+	pin = !!(pins & (1 << AMS_DELTA_EARPIECE));
+	if (pin != snd_soc_dapm_get_pin_status(codec, "Earpiece")) {
+		changed = 1;
+		if (pin)
+			snd_soc_dapm_enable_pin(codec, "Earpiece");
+		else
+			snd_soc_dapm_disable_pin(codec, "Earpiece");
+	}
+	pin = !!(pins & (1 << AMS_DELTA_MICROPHONE));
+	if (pin != snd_soc_dapm_get_pin_status(codec, "Microphone")) {
+		changed = 1;
+		if (pin)
+			snd_soc_dapm_enable_pin(codec, "Microphone");
+		else
+			snd_soc_dapm_disable_pin(codec, "Microphone");
+	}
+	pin = !!(pins & (1 << AMS_DELTA_SPEAKER));
+	if (pin != snd_soc_dapm_get_pin_status(codec, "Speaker")) {
+		changed = 1;
+		if (pin)
+			snd_soc_dapm_enable_pin(codec, "Speaker");
+		else
+			snd_soc_dapm_disable_pin(codec, "Speaker");
+	}
+	pin = !!(pins & (1 << AMS_DELTA_AGC));
+	if (pin != ams_delta_audio_agc) {
+		ams_delta_audio_agc = pin;
+		changed = 1;
+		if (pin)
+			snd_soc_dapm_enable_pin(codec, "AGCIN");
+		else
+			snd_soc_dapm_disable_pin(codec, "AGCIN");
+	}
+	if (changed)
+		snd_soc_dapm_sync(codec);
+
+	mutex_unlock(&codec->mutex);
+
+	return changed;
+}
+
+static int ams_delta_get_audio_mode(struct snd_kcontrol *kcontrol,
+					struct snd_ctl_elem_value *ucontrol)
+{
+	struct snd_soc_codec *codec =  snd_kcontrol_chip(kcontrol);
+	unsigned short pins, mode;
+
+	pins = ((snd_soc_dapm_get_pin_status(codec, "Mouthpiece") <<
+							AMS_DELTA_MOUTHPIECE) |
+			(snd_soc_dapm_get_pin_status(codec, "Earpiece") <<
+							AMS_DELTA_EARPIECE));
+	if (pins)
+		pins |= (snd_soc_dapm_get_pin_status(codec, "Microphone") <<
+							AMS_DELTA_MICROPHONE);
+	else
+		pins = ((snd_soc_dapm_get_pin_status(codec, "Microphone") <<
+							AMS_DELTA_MICROPHONE) |
+			(snd_soc_dapm_get_pin_status(codec, "Speaker") <<
+							AMS_DELTA_SPEAKER) |
+			(ams_delta_audio_agc << AMS_DELTA_AGC));
+
+	for (mode = 0; mode < ARRAY_SIZE(ams_delta_audio_mode); mode++)
+		if (pins == ams_delta_audio_mode_pins[mode])
+			break;
+
+	if (mode >= ARRAY_SIZE(ams_delta_audio_mode))
+		return -EINVAL;
+
+	ucontrol->value.enumerated.item[0] = mode;
+
+	return 0;
+}
+
+static const struct soc_enum ams_delta_audio_enum[] = {
+	SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(ams_delta_audio_mode),
+						ams_delta_audio_mode),
+};
+
+static const struct snd_kcontrol_new ams_delta_audio_controls[] = {
+	SOC_ENUM_EXT("Audio Mode", ams_delta_audio_enum[0],
+			ams_delta_get_audio_mode, ams_delta_set_audio_mode),
+};
+
+/* Hook switch */
+static struct snd_soc_jack ams_delta_hook_switch;
+static struct snd_soc_jack_gpio ams_delta_hook_switch_gpios[] = {
+	{
+		.gpio = 4,
+		.name = "hook_switch",
+		.report = SND_JACK_HEADSET,
+		.invert = 1,
+		.debounce_time = 150,
+	}
+};
+
+/* After we are able to control the codec over the modem,
+ * the hook switch can be used for dynamic DAPM reconfiguration. */
+static struct snd_soc_jack_pin ams_delta_hook_switch_pins[] = {
+	/* Handset */
+	{
+		.pin = "Mouthpiece",
+		.mask = SND_JACK_MICROPHONE,
+	},
+	{
+		.pin = "Earpiece",
+		.mask = SND_JACK_HEADPHONE,
+	},
+	/* Handsfree */
+	{
+		.pin = "Microphone",
+		.mask = SND_JACK_MICROPHONE,
+		.invert = 1,
+	},
+	{
+		.pin = "Speaker",
+		.mask = SND_JACK_HEADPHONE,
+		.invert = 1,
+	},
+};
+
+
+/*
+ * Modem line discipline, required for making above controls functional.
+ * Activated from userspace with ldattach, possibly invoked from udev rule.
+ */
+
+/* To actually apply any modem controlled configuration changes to the codec,
+ * we must connect codec DAI pins to the modem for a moment.  Be carefull not
+ * to interfere with our digital mute function that shares the same hardware. */
+static struct timer_list cx81801_timer;
+static bool cx81801_cmd_pending;
+static bool ams_delta_muted;
+static DEFINE_SPINLOCK(ams_delta_lock);
+
+static void cx81801_timeout(unsigned long data)
+{
+	int muted;
+
+	spin_lock(&ams_delta_lock);
+	cx81801_cmd_pending = 0;
+	muted = ams_delta_muted;
+	spin_unlock(&ams_delta_lock);
+
+	/* Reconnect the codec DAI back from the modem to the CPU DAI
+	 * only if digital mute still off */
+	if (!muted)
+		ams_delta_latch2_write(AMS_DELTA_LATCH2_MODEM_CODEC, 0);
+}
+
+/* Line discipline .open() */
+static int cx81801_open(struct tty_struct *tty)
+{
+	return v253_ops.open(tty);
+}
+
+/* Line discipline .close() */
+static void cx81801_close(struct tty_struct *tty)
+{
+	struct snd_soc_codec *codec = tty->disc_data;
+
+	del_timer_sync(&cx81801_timer);
+
+	v253_ops.close(tty);
+
+	/* Prevent the hook switch from further changing the DAPM pins */
+	INIT_LIST_HEAD(&ams_delta_hook_switch.pins);
+
+	/* Revert back to default audio input/output constellation */
+	snd_soc_dapm_disable_pin(codec, "Mouthpiece");
+	snd_soc_dapm_enable_pin(codec, "Earpiece");
+	snd_soc_dapm_enable_pin(codec, "Microphone");
+	snd_soc_dapm_disable_pin(codec, "Speaker");
+	snd_soc_dapm_disable_pin(codec, "AGCIN");
+	snd_soc_dapm_sync(codec);
+}
+
+/* Line discipline .hangup() */
+static int cx81801_hangup(struct tty_struct *tty)
+{
+	cx81801_close(tty);
+	return 0;
+}
+
+/* Line discipline .recieve_buf() */
+static void cx81801_receive(struct tty_struct *tty,
+				const unsigned char *cp, char *fp, int count)
+{
+	struct snd_soc_codec *codec = tty->disc_data;
+	const unsigned char *c;
+	int apply, ret;
+
+	if (!codec->control_data) {
+		/* First modem response, complete setup procedure */
+
+		/* Initialize timer used for config pulse generation */
+		setup_timer(&cx81801_timer, cx81801_timeout, 0);
+
+		v253_ops.receive_buf(tty, cp, fp, count);
+
+		/* Link hook switch to DAPM pins */
+		ret = snd_soc_jack_add_pins(&ams_delta_hook_switch,
+					ARRAY_SIZE(ams_delta_hook_switch_pins),
+					ams_delta_hook_switch_pins);
+		if (ret)
+			dev_warn(codec->socdev->card->dev,
+				"Failed to link hook switch to DAPM pins, "
+				"will continue with hook switch unlinked.\n");
+
+		return;
+	}
+
+	v253_ops.receive_buf(tty, cp, fp, count);
+
+	for (c = &cp[count - 1]; c >= cp; c--) {
+		if (*c != '\r')
+			continue;
+		/* Complete modem response received, apply config to codec */
+
+		spin_lock_bh(&ams_delta_lock);
+		mod_timer(&cx81801_timer, jiffies + msecs_to_jiffies(150));
+		apply = !ams_delta_muted && !cx81801_cmd_pending;
+		cx81801_cmd_pending = 1;
+		spin_unlock_bh(&ams_delta_lock);
+
+		/* Apply config pulse by connecting the codec to the modem
+		 * if not already done */
+		if (apply)
+			ams_delta_latch2_write(AMS_DELTA_LATCH2_MODEM_CODEC,
+						AMS_DELTA_LATCH2_MODEM_CODEC);
+		break;
+	}
+}
+
+/* Line discipline .write_wakeup() */
+static void cx81801_wakeup(struct tty_struct *tty)
+{
+	v253_ops.write_wakeup(tty);
+}
+
+static struct tty_ldisc_ops cx81801_ops = {
+	.magic = TTY_LDISC_MAGIC,
+	.name = "cx81801",
+	.owner = THIS_MODULE,
+	.open = cx81801_open,
+	.close = cx81801_close,
+	.hangup = cx81801_hangup,
+	.receive_buf = cx81801_receive,
+	.write_wakeup = cx81801_wakeup,
+};
+
+
+/*
+ * Even if not very usefull, the sound card can still work without any of the
+ * above functonality activated.  You can still control its audio input/output
+ * constellation and speakerphone gain from userspace by issueing AT commands
+ * over the modem port.
+ */
+
+static int ams_delta_hw_params(struct snd_pcm_substream *substream,
+			 struct snd_pcm_hw_params *params)
+{
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+
+	/* Set cpu DAI configuration */
+	return snd_soc_dai_set_fmt(rtd->dai->cpu_dai,
+				   SND_SOC_DAIFMT_DSP_A |
+				   SND_SOC_DAIFMT_NB_NF |
+				   SND_SOC_DAIFMT_CBM_CFM);
+}
+
+static struct snd_soc_ops ams_delta_ops = {
+	.hw_params = ams_delta_hw_params,
+};
+
+
+/* Board specific codec bias level control */
+static int ams_delta_set_bias_level(struct snd_soc_card *card,
+					enum snd_soc_bias_level level)
+{
+	struct snd_soc_codec *codec = card->codec;
+
+	switch (level) {
+	case SND_SOC_BIAS_ON:
+	case SND_SOC_BIAS_PREPARE:
+	case SND_SOC_BIAS_STANDBY:
+		if (codec->bias_level == SND_SOC_BIAS_OFF)
+			ams_delta_latch2_write(AMS_DELTA_LATCH2_MODEM_NRESET,
+						AMS_DELTA_LATCH2_MODEM_NRESET);
+		break;
+	case SND_SOC_BIAS_OFF:
+		if (codec->bias_level != SND_SOC_BIAS_OFF)
+			ams_delta_latch2_write(AMS_DELTA_LATCH2_MODEM_NRESET,
+						0);
+	}
+	codec->bias_level = level;
+
+	return 0;
+}
+
+/* Digital mute implemented using modem/CPU multiplexer.
+ * Shares hardware with codec config pulse generation */
+static bool ams_delta_muted = 1;
+
+static int ams_delta_digital_mute(struct snd_soc_dai *dai, int mute)
+{
+	int apply;
+
+	if (ams_delta_muted == mute)
+		return 0;
+
+	spin_lock_bh(&ams_delta_lock);
+	ams_delta_muted = mute;
+	apply = !cx81801_cmd_pending;
+	spin_unlock_bh(&ams_delta_lock);
+
+	if (apply)
+		ams_delta_latch2_write(AMS_DELTA_LATCH2_MODEM_CODEC,
+				mute ? AMS_DELTA_LATCH2_MODEM_CODEC : 0);
+	return 0;
+}
+
+/* Our codec DAI probably doesn't have its own .ops structure */
+static struct snd_soc_dai_ops ams_delta_dai_ops = {
+	.digital_mute = ams_delta_digital_mute,
+};
+
+/* Will be used if the codec ever has its own digital_mute function */
+static int ams_delta_startup(struct snd_pcm_substream *substream)
+{
+	return ams_delta_digital_mute(NULL, 0);
+}
+
+static void ams_delta_shutdown(struct snd_pcm_substream *substream)
+{
+	ams_delta_digital_mute(NULL, 1);
+}
+
+
+/*
+ * Card initialization
+ */
+
+static int ams_delta_cx20442_init(struct snd_soc_codec *codec)
+{
+	struct snd_soc_dai *codec_dai = codec->dai;
+	struct snd_soc_card *card = codec->socdev->card;
+	int ret;
+	/* Codec is ready, now add/activate board specific controls */
+
+	/* Set up digital mute if not provided by the codec */
+	if (!codec_dai->ops) {
+		codec_dai->ops = &ams_delta_dai_ops;
+	} else if (!codec_dai->ops->digital_mute) {
+		codec_dai->ops->digital_mute = ams_delta_digital_mute;
+	} else {
+		ams_delta_ops.startup = ams_delta_startup;
+		ams_delta_ops.shutdown = ams_delta_shutdown;
+	}
+
+	/* Set codec bias level */
+	ams_delta_set_bias_level(card, SND_SOC_BIAS_STANDBY);
+
+	/* Add hook switch - can be used to control the codec from userspace
+	 * even if line discipline fails */
+	ret = snd_soc_jack_new(card, "hook_switch",
+				SND_JACK_HEADSET, &ams_delta_hook_switch);
+	if (ret)
+		dev_warn(card->dev,
+				"Failed to allocate resources for hook switch, "
+				"will continue without one.\n");
+	else {
+		ret = snd_soc_jack_add_gpios(&ams_delta_hook_switch,
+					ARRAY_SIZE(ams_delta_hook_switch_gpios),
+					ams_delta_hook_switch_gpios);
+		if (ret)
+			dev_warn(card->dev,
+				"Failed to set up hook switch GPIO line, "
+				"will continue with hook switch inactive.\n");
+	}
+
+	/* Register optional line discipline for over the modem control */
+	ret = tty_register_ldisc(N_V253, &cx81801_ops);
+	if (ret) {
+		dev_warn(card->dev,
+				"Failed to register line discipline, "
+				"will continue without any controls.\n");
+		return 0;
+	}
+
+	/* Add board specific DAPM widgets and routes */
+	ret = snd_soc_dapm_new_controls(codec, ams_delta_dapm_widgets,
+					ARRAY_SIZE(ams_delta_dapm_widgets));
+	if (ret) {
+		dev_warn(card->dev,
+				"Failed to register DAPM controls, "
+				"will continue without any.\n");
+		return 0;
+	}
+
+	ret = snd_soc_dapm_add_routes(codec, ams_delta_audio_map,
+					ARRAY_SIZE(ams_delta_audio_map));
+	if (ret) {
+		dev_warn(card->dev,
+				"Failed to set up DAPM routes, "
+				"will continue with codec default map.\n");
+		return 0;
+	}
+
+	/* Set up initial pin constellation */
+	snd_soc_dapm_disable_pin(codec, "Mouthpiece");
+	snd_soc_dapm_enable_pin(codec, "Earpiece");
+	snd_soc_dapm_enable_pin(codec, "Microphone");
+	snd_soc_dapm_disable_pin(codec, "Speaker");
+	snd_soc_dapm_disable_pin(codec, "AGCIN");
+	snd_soc_dapm_disable_pin(codec, "AGCOUT");
+	snd_soc_dapm_sync(codec);
+
+	/* Add virtual switch */
+	ret = snd_soc_add_controls(codec, ams_delta_audio_controls,
+					ARRAY_SIZE(ams_delta_audio_controls));
+	if (ret)
+		dev_warn(card->dev,
+				"Failed to register audio mode control, "
+				"will continue without it.\n");
+
+	return 0;
+}
+
+/* DAI glue - connects codec <--> CPU */
+static struct snd_soc_dai_link ams_delta_dai_link = {
+	.name = "CX20442",
+	.stream_name = "CX20442",
+	.cpu_dai = &omap_mcbsp_dai[0],
+	.codec_dai = &cx20442_dai,
+	.init = ams_delta_cx20442_init,
+	.ops = &ams_delta_ops,
+};
+
+/* Audio card driver */
+static struct snd_soc_card ams_delta_audio_card = {
+	.name = "AMS_DELTA",
+	.platform = &omap_soc_platform,
+	.dai_link = &ams_delta_dai_link,
+	.num_links = 1,
+	.set_bias_level = ams_delta_set_bias_level,
+};
+
+/* Audio subsystem */
+static struct snd_soc_device ams_delta_snd_soc_device = {
+	.card = &ams_delta_audio_card,
+	.codec_dev = &cx20442_codec_dev,
+};
+
+/* Module init/exit */
+static struct platform_device *ams_delta_audio_platform_device;
+static struct platform_device *cx20442_platform_device;
+
+static int __init ams_delta_module_init(void)
+{
+	int ret;
+
+	if (!(machine_is_ams_delta()))
+		return -ENODEV;
+
+	ams_delta_audio_platform_device =
+			platform_device_alloc("soc-audio", -1);
+	if (!ams_delta_audio_platform_device)
+		return -ENOMEM;
+
+	platform_set_drvdata(ams_delta_audio_platform_device,
+				&ams_delta_snd_soc_device);
+	ams_delta_snd_soc_device.dev = &ams_delta_audio_platform_device->dev;
+	*(unsigned int *)ams_delta_dai_link.cpu_dai->private_data = OMAP_MCBSP1;
+
+	ret = platform_device_add(ams_delta_audio_platform_device);
+	if (ret)
+		goto err;
+
+	/*
+	 * Codec platform device could be registered from elsewhere (board?),
+	 * but I do it here as it makes sense only if used with the card.
+	 */
+	cx20442_platform_device = platform_device_register_simple("cx20442",
+								-1, NULL, 0);
+	return 0;
+err:
+	platform_device_put(ams_delta_audio_platform_device);
+	return ret;
+}
+module_init(ams_delta_module_init);
+
+static void __exit ams_delta_module_exit(void)
+{
+	struct snd_soc_codec *codec;
+	struct tty_struct *tty;
+
+	if (ams_delta_audio_card.codec) {
+		codec = ams_delta_audio_card.codec;
+
+		if (codec->control_data) {
+			tty = codec->control_data;
+
+			tty_hangup(tty);
+		}
+	}
+
+	if (tty_unregister_ldisc(N_V253) != 0)
+		dev_warn(&ams_delta_audio_platform_device->dev,
+			"failed to unregister V253 line discipline\n");
+
+	snd_soc_jack_free_gpios(&ams_delta_hook_switch,
+			ARRAY_SIZE(ams_delta_hook_switch_gpios),
+			ams_delta_hook_switch_gpios);
+
+	/* Keep modem power on */
+	ams_delta_set_bias_level(&ams_delta_audio_card, SND_SOC_BIAS_STANDBY);
+
+	platform_device_unregister(cx20442_platform_device);
+	platform_device_unregister(ams_delta_audio_platform_device);
+}
+module_exit(ams_delta_module_exit);
+
+MODULE_AUTHOR("Janusz Krzysztofik <jkrzyszt@tis.icnet.pl>");
+MODULE_DESCRIPTION("ALSA SoC driver for Amstrad E3 (Delta) videophone");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/omap/n810.c b/sound/soc/omap/n810.c
index b60b1dfbc43..0a505938e42 100644
--- a/sound/soc/omap/n810.c
+++ b/sound/soc/omap/n810.c
@@ -22,6 +22,7 @@
  */
 
 #include <linux/clk.h>
+#include <linux/i2c.h>
 #include <linux/platform_device.h>
 #include <sound/core.h>
 #include <sound/pcm.h>
@@ -322,8 +323,6 @@ static struct snd_soc_card snd_soc_n810 = {
 
 /* Audio private data */
 static struct aic3x_setup_data n810_aic33_setup = {
-	.i2c_bus = 2,
-	.i2c_address = 0x18,
 	.gpio_func[0] = AIC3X_GPIO1_FUNC_DISABLED,
 	.gpio_func[1] = AIC3X_GPIO2_FUNC_DIGITAL_MIC_INPUT,
 };
@@ -337,6 +336,13 @@ static struct snd_soc_device n810_snd_devdata = {
 
 static struct platform_device *n810_snd_device;
 
+/* temporary i2c device creation until this can be moved into the machine
+ * support file.
+*/
+static struct i2c_board_info i2c_device[] = {
+	{ I2C_BOARD_INFO("tlv320aic3x", 0x1b), }
+};
+
 static int __init n810_soc_init(void)
 {
 	int err;
@@ -345,6 +351,8 @@ static int __init n810_soc_init(void)
 	if (!(machine_is_nokia_n810() || machine_is_nokia_n810_wimax()))
 		return -ENODEV;
 
+	i2c_register_board_info(1, i2c_device, ARRAY_SIZE(i2c_device));
+
 	n810_snd_device = platform_device_alloc("soc-audio", -1);
 	if (!n810_snd_device)
 		return -ENOMEM;
diff --git a/sound/soc/omap/omap-mcbsp.c b/sound/soc/omap/omap-mcbsp.c
index a5d46a7b196..3341f49402c 100644
--- a/sound/soc/omap/omap-mcbsp.c
+++ b/sound/soc/omap/omap-mcbsp.c
@@ -139,27 +139,67 @@ static const unsigned long omap34xx_mcbsp_port[][2] = {
 static const unsigned long omap34xx_mcbsp_port[][2] = {};
 #endif
 
+static void omap_mcbsp_set_threshold(struct snd_pcm_substream *substream)
+{
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;
+	struct omap_mcbsp_data *mcbsp_data = to_mcbsp(cpu_dai->private_data);
+	int dma_op_mode = omap_mcbsp_get_dma_op_mode(mcbsp_data->bus_id);
+	int samples;
+
+	/* TODO: Currently, MODE_ELEMENT == MODE_FRAME */
+	if (dma_op_mode == MCBSP_DMA_MODE_THRESHOLD)
+		samples = snd_pcm_lib_period_bytes(substream) >> 1;
+	else
+		samples = 1;
+
+	/* Configure McBSP internal buffer usage */
+	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+		omap_mcbsp_set_tx_threshold(mcbsp_data->bus_id, samples - 1);
+	else
+		omap_mcbsp_set_rx_threshold(mcbsp_data->bus_id, samples - 1);
+}
+
 static int omap_mcbsp_dai_startup(struct snd_pcm_substream *substream,
 				  struct snd_soc_dai *dai)
 {
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;
 	struct omap_mcbsp_data *mcbsp_data = to_mcbsp(cpu_dai->private_data);
+	int bus_id = mcbsp_data->bus_id;
 	int err = 0;
 
-	if (cpu_is_omap343x() && mcbsp_data->bus_id == 1) {
+	if (!cpu_dai->active)
+		err = omap_mcbsp_request(bus_id);
+
+	if (cpu_is_omap343x()) {
+		int dma_op_mode = omap_mcbsp_get_dma_op_mode(bus_id);
+		int max_period;
+
 		/*
 		 * McBSP2 in OMAP3 has 1024 * 32-bit internal audio buffer.
 		 * Set constraint for minimum buffer size to the same than FIFO
 		 * size in order to avoid underruns in playback startup because
 		 * HW is keeping the DMA request active until FIFO is filled.
 		 */
-		snd_pcm_hw_constraint_minmax(substream->runtime,
-			SNDRV_PCM_HW_PARAM_BUFFER_BYTES, 4096, UINT_MAX);
-	}
+		if (bus_id == 1)
+			snd_pcm_hw_constraint_minmax(substream->runtime,
+					SNDRV_PCM_HW_PARAM_BUFFER_BYTES,
+					4096, UINT_MAX);
 
-	if (!cpu_dai->active)
-		err = omap_mcbsp_request(mcbsp_data->bus_id);
+		if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+			max_period = omap_mcbsp_get_max_tx_threshold(bus_id);
+		else
+			max_period = omap_mcbsp_get_max_rx_threshold(bus_id);
+
+		max_period++;
+		max_period <<= 1;
+
+		if (dma_op_mode == MCBSP_DMA_MODE_THRESHOLD)
+			snd_pcm_hw_constraint_minmax(substream->runtime,
+						SNDRV_PCM_HW_PARAM_PERIOD_BYTES,
+						32, max_period);
+	}
 
 	return err;
 }
@@ -183,21 +223,21 @@ static int omap_mcbsp_dai_trigger(struct snd_pcm_substream *substream, int cmd,
 	struct snd_soc_pcm_runtime *rtd = substream->private_data;
 	struct snd_soc_dai *cpu_dai = rtd->dai->cpu_dai;
 	struct omap_mcbsp_data *mcbsp_data = to_mcbsp(cpu_dai->private_data);
-	int err = 0;
+	int err = 0, play = (substream->stream == SNDRV_PCM_STREAM_PLAYBACK);
 
 	switch (cmd) {
 	case SNDRV_PCM_TRIGGER_START:
 	case SNDRV_PCM_TRIGGER_RESUME:
 	case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
-		if (!mcbsp_data->active++)
-			omap_mcbsp_start(mcbsp_data->bus_id);
+		mcbsp_data->active++;
+		omap_mcbsp_start(mcbsp_data->bus_id, play, !play);
 		break;
 
 	case SNDRV_PCM_TRIGGER_STOP:
 	case SNDRV_PCM_TRIGGER_SUSPEND:
 	case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
-		if (!--mcbsp_data->active)
-			omap_mcbsp_stop(mcbsp_data->bus_id);
+		omap_mcbsp_stop(mcbsp_data->bus_id, play, !play);
+		mcbsp_data->active--;
 		break;
 	default:
 		err = -EINVAL;
@@ -215,7 +255,7 @@ static int omap_mcbsp_dai_hw_params(struct snd_pcm_substream *substream,
 	struct omap_mcbsp_data *mcbsp_data = to_mcbsp(cpu_dai->private_data);
 	struct omap_mcbsp_reg_cfg *regs = &mcbsp_data->regs;
 	int dma, bus_id = mcbsp_data->bus_id, id = cpu_dai->id;
-	int wlen, channels, wpf;
+	int wlen, channels, wpf, sync_mode = OMAP_DMA_SYNC_ELEMENT;
 	unsigned long port;
 	unsigned int format;
 
@@ -231,6 +271,12 @@ static int omap_mcbsp_dai_hw_params(struct snd_pcm_substream *substream,
 	} else if (cpu_is_omap343x()) {
 		dma = omap24xx_dma_reqs[bus_id][substream->stream];
 		port = omap34xx_mcbsp_port[bus_id][substream->stream];
+		omap_mcbsp_dai_dma_params[id][substream->stream].set_threshold =
+						omap_mcbsp_set_threshold;
+		/* TODO: Currently, MODE_ELEMENT == MODE_FRAME */
+		if (omap_mcbsp_get_dma_op_mode(bus_id) ==
+						MCBSP_DMA_MODE_THRESHOLD)
+			sync_mode = OMAP_DMA_SYNC_FRAME;
 	} else {
 		return -ENODEV;
 	}
@@ -238,6 +284,7 @@ static int omap_mcbsp_dai_hw_params(struct snd_pcm_substream *substream,
 		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;
+	omap_mcbsp_dai_dma_params[id][substream->stream].sync_mode = sync_mode;
 	cpu_dai->dma_data = &omap_mcbsp_dai_dma_params[id][substream->stream];
 
 	if (mcbsp_data->configured) {
@@ -321,11 +368,14 @@ static int omap_mcbsp_dai_set_dai_fmt(struct snd_soc_dai *cpu_dai,
 	/* Generic McBSP register settings */
 	regs->spcr2	|= XINTM(3) | FREE;
 	regs->spcr1	|= RINTM(3);
-	regs->rcr2	|= RFIG;
-	regs->xcr2	|= XFIG;
+	/* RFIG and XFIG are not defined in 34xx */
+	if (!cpu_is_omap34xx()) {
+		regs->rcr2	|= RFIG;
+		regs->xcr2	|= XFIG;
+	}
 	if (cpu_is_omap2430() || cpu_is_omap34xx()) {
-		regs->xccr = DXENDLY(1) | XDMAEN;
-		regs->rccr = RFULL_CYCLE | RDMAEN;
+		regs->xccr = DXENDLY(1) | XDMAEN | XDISABLE;
+		regs->rccr = RFULL_CYCLE | RDMAEN | RDISABLE;
 	}
 
 	switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
@@ -462,6 +512,40 @@ static int omap_mcbsp_dai_set_clks_src(struct omap_mcbsp_data *mcbsp_data,
 	return 0;
 }
 
+static int omap_mcbsp_dai_set_rcvr_src(struct omap_mcbsp_data *mcbsp_data,
+				       int clk_id)
+{
+	int sel_bit, set = 0;
+	u16 reg = OMAP2_CONTROL_DEVCONF0;
+
+	if (cpu_class_is_omap1())
+		return -EINVAL; /* TODO: Can this be implemented for OMAP1? */
+	if (mcbsp_data->bus_id != 0)
+		return -EINVAL;
+
+	switch (clk_id) {
+	case OMAP_MCBSP_CLKR_SRC_CLKX:
+		set = 1;
+	case OMAP_MCBSP_CLKR_SRC_CLKR:
+		sel_bit = 3;
+		break;
+	case OMAP_MCBSP_FSR_SRC_FSX:
+		set = 1;
+	case OMAP_MCBSP_FSR_SRC_FSR:
+		sel_bit = 4;
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	if (set)
+		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;
+}
+
 static int omap_mcbsp_dai_set_dai_sysclk(struct snd_soc_dai *cpu_dai,
 					 int clk_id, unsigned int freq,
 					 int dir)
@@ -484,6 +568,13 @@ static int omap_mcbsp_dai_set_dai_sysclk(struct snd_soc_dai *cpu_dai,
 	case OMAP_MCBSP_SYSCLK_CLKR_EXT:
 		regs->pcr0	|= SCLKME;
 		break;
+
+	case OMAP_MCBSP_CLKR_SRC_CLKR:
+	case OMAP_MCBSP_CLKR_SRC_CLKX:
+	case OMAP_MCBSP_FSR_SRC_FSR:
+	case OMAP_MCBSP_FSR_SRC_FSX:
+		err = omap_mcbsp_dai_set_rcvr_src(mcbsp_data, clk_id);
+		break;
 	default:
 		err = -ENODEV;
 	}
diff --git a/sound/soc/omap/omap-mcbsp.h b/sound/soc/omap/omap-mcbsp.h
index c8147aace81..647d2f981ab 100644
--- a/sound/soc/omap/omap-mcbsp.h
+++ b/sound/soc/omap/omap-mcbsp.h
@@ -32,6 +32,10 @@ enum omap_mcbsp_clksrg_clk {
 	OMAP_MCBSP_SYSCLK_CLK,		/* Internal ICLK */
 	OMAP_MCBSP_SYSCLK_CLKX_EXT,	/* External CLKX pin */
 	OMAP_MCBSP_SYSCLK_CLKR_EXT,	/* External CLKR pin */
+	OMAP_MCBSP_CLKR_SRC_CLKR,	/* CLKR from CLKR pin */
+	OMAP_MCBSP_CLKR_SRC_CLKX,	/* CLKR from CLKX pin */
+	OMAP_MCBSP_FSR_SRC_FSR,		/* FSR from FSR pin */
+	OMAP_MCBSP_FSR_SRC_FSX,		/* FSR from FSX pin */
 };
 
 /* McBSP dividers */
diff --git a/sound/soc/omap/omap-pcm.c b/sound/soc/omap/omap-pcm.c
index 84a1950880e..5735945788b 100644
--- a/sound/soc/omap/omap-pcm.c
+++ b/sound/soc/omap/omap-pcm.c
@@ -59,16 +59,31 @@ static void omap_pcm_dma_irq(int ch, u16 stat, void *data)
 	struct omap_runtime_data *prtd = runtime->private_data;
 	unsigned long flags;
 
-	if (cpu_is_omap1510()) {
+	if ((cpu_is_omap1510()) &&
+			(substream->stream == SNDRV_PCM_STREAM_PLAYBACK)) {
 		/*
-		 * OMAP1510 doesn't support DMA chaining so have to restart
-		 * the transfer after all periods are transferred
+		 * OMAP1510 doesn't fully support DMA progress counter
+		 * and there is no software emulation implemented yet,
+		 * so have to maintain our own playback progress counter
+		 * that can be used by omap_pcm_pointer() instead.
 		 */
 		spin_lock_irqsave(&prtd->lock, flags);
+		if ((stat == OMAP_DMA_LAST_IRQ) &&
+				(prtd->period_index == runtime->periods - 1)) {
+			/* we are in sync, do nothing */
+			spin_unlock_irqrestore(&prtd->lock, flags);
+			return;
+		}
 		if (prtd->period_index >= 0) {
-			if (++prtd->period_index == runtime->periods) {
+			if (stat & OMAP_DMA_BLOCK_IRQ) {
+				/* end of buffer reached, loop back */
+				prtd->period_index = 0;
+			} else if (stat & OMAP_DMA_LAST_IRQ) {
+				/* update the counter for the last period */
+				prtd->period_index = runtime->periods - 1;
+			} else if (++prtd->period_index >= runtime->periods) {
+				/* end of buffer missed? loop back */
 				prtd->period_index = 0;
-				omap_start_dma(prtd->dma_ch);
 			}
 		}
 		spin_unlock_irqrestore(&prtd->lock, flags);
@@ -100,7 +115,7 @@ 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 (!err && !cpu_is_omap1510()) {
+	if (!err) {
 		/*
 		 * Link channel with itself so DMA doesn't need any
 		 * reprogramming while looping the buffer
@@ -119,8 +134,7 @@ static int omap_pcm_hw_free(struct snd_pcm_substream *substream)
 	if (prtd->dma_data == NULL)
 		return 0;
 
-	if (!cpu_is_omap1510())
-		omap_dma_unlink_lch(prtd->dma_ch, prtd->dma_ch);
+	omap_dma_unlink_lch(prtd->dma_ch, prtd->dma_ch);
 	omap_free_dma(prtd->dma_ch);
 	prtd->dma_data = NULL;
 
@@ -148,7 +162,7 @@ static int omap_pcm_prepare(struct snd_pcm_substream *substream)
 	 */
 	dma_params.data_type			= OMAP_DMA_DATA_TYPE_S16;
 	dma_params.trigger			= dma_data->dma_req;
-	dma_params.sync_mode			= OMAP_DMA_SYNC_ELEMENT;
+	dma_params.sync_mode			= dma_data->sync_mode;
 	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK) {
 		dma_params.src_amode		= OMAP_DMA_AMODE_POST_INC;
 		dma_params.dst_amode		= OMAP_DMA_AMODE_CONSTANT;
@@ -174,7 +188,15 @@ static int omap_pcm_prepare(struct snd_pcm_substream *substream)
 	dma_params.frame_count	= runtime->periods;
 	omap_set_dma_params(prtd->dma_ch, &dma_params);
 
-	omap_enable_dma_irq(prtd->dma_ch, OMAP_DMA_FRAME_IRQ);
+	if ((cpu_is_omap1510()) &&
+			(substream->stream == SNDRV_PCM_STREAM_PLAYBACK))
+		omap_enable_dma_irq(prtd->dma_ch, OMAP_DMA_FRAME_IRQ |
+			      OMAP_DMA_LAST_IRQ | OMAP_DMA_BLOCK_IRQ);
+	else
+		omap_enable_dma_irq(prtd->dma_ch, OMAP_DMA_FRAME_IRQ);
+
+	omap_set_dma_src_burst_mode(prtd->dma_ch, OMAP_DMA_DATA_BURST_16);
+	omap_set_dma_dest_burst_mode(prtd->dma_ch, OMAP_DMA_DATA_BURST_16);
 
 	return 0;
 }
@@ -183,6 +205,7 @@ static int omap_pcm_trigger(struct snd_pcm_substream *substream, int cmd)
 {
 	struct snd_pcm_runtime *runtime = substream->runtime;
 	struct omap_runtime_data *prtd = runtime->private_data;
+	struct omap_pcm_dma_data *dma_data = prtd->dma_data;
 	unsigned long flags;
 	int ret = 0;
 
@@ -192,6 +215,10 @@ static int omap_pcm_trigger(struct snd_pcm_substream *substream, int cmd)
 	case SNDRV_PCM_TRIGGER_RESUME:
 	case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
 		prtd->period_index = 0;
+		/* Configure McBSP internal buffer usage */
+		if (dma_data->set_threshold)
+			dma_data->set_threshold(substream);
+
 		omap_start_dma(prtd->dma_ch);
 		break;
 
@@ -288,7 +315,7 @@ static struct snd_pcm_ops omap_pcm_ops = {
 	.mmap		= omap_pcm_mmap,
 };
 
-static u64 omap_pcm_dmamask = DMA_BIT_MASK(32);
+static u64 omap_pcm_dmamask = DMA_BIT_MASK(64);
 
 static int omap_pcm_preallocate_dma_buffer(struct snd_pcm *pcm,
 	int stream)
@@ -330,7 +357,7 @@ static void omap_pcm_free_dma_buffers(struct snd_pcm *pcm)
 	}
 }
 
-int omap_pcm_new(struct snd_card *card, struct snd_soc_dai *dai,
+static int omap_pcm_new(struct snd_card *card, struct snd_soc_dai *dai,
 		 struct snd_pcm *pcm)
 {
 	int ret = 0;
@@ -338,7 +365,7 @@ int omap_pcm_new(struct snd_card *card, struct snd_soc_dai *dai,
 	if (!card->dev->dma_mask)
 		card->dev->dma_mask = &omap_pcm_dmamask;
 	if (!card->dev->coherent_dma_mask)
-		card->dev->coherent_dma_mask = DMA_BIT_MASK(32);
+		card->dev->coherent_dma_mask = DMA_BIT_MASK(64);
 
 	if (dai->playback.channels_min) {
 		ret = omap_pcm_preallocate_dma_buffer(pcm,
diff --git a/sound/soc/omap/omap-pcm.h b/sound/soc/omap/omap-pcm.h
index 8d9d26916b0..38a821dd411 100644
--- a/sound/soc/omap/omap-pcm.h
+++ b/sound/soc/omap/omap-pcm.h
@@ -29,6 +29,8 @@ struct omap_pcm_dma_data {
 	char		*name;		/* stream identifier */
 	int		dma_req;	/* DMA request line */
 	unsigned long	port_addr;	/* transmit/receive register */
+	int		sync_mode;	/* DMA sync mode */
+	void (*set_threshold)(struct snd_pcm_substream *substream);
 };
 
 extern struct snd_soc_platform omap_soc_platform;
diff --git a/sound/soc/omap/sdp3430.c b/sound/soc/omap/sdp3430.c
index b719e5db4f5..4a3f62d1f29 100644
--- a/sound/soc/omap/sdp3430.c
+++ b/sound/soc/omap/sdp3430.c
@@ -24,6 +24,7 @@
 
 #include <linux/clk.h>
 #include <linux/platform_device.h>
+#include <linux/i2c/twl4030.h>
 #include <sound/core.h>
 #include <sound/pcm.h>
 #include <sound/soc.h>
@@ -39,6 +40,11 @@
 #include "omap-pcm.h"
 #include "../codecs/twl4030.h"
 
+/* TWL4030 PMBR1 Register */
+#define TWL4030_INTBR_PMBR1		0x0D
+/* TWL4030 PMBR1 Register GPIO6 mux bit */
+#define TWL4030_GPIO6_PWM0_MUTE(value)	(value << 2)
+
 static struct snd_soc_card snd_soc_sdp3430;
 
 static int sdp3430_hw_params(struct snd_pcm_substream *substream,
@@ -96,7 +102,7 @@ static int sdp3430_hw_voice_params(struct snd_pcm_substream *substream,
 	ret = snd_soc_dai_set_fmt(codec_dai,
 				SND_SOC_DAIFMT_DSP_A |
 				SND_SOC_DAIFMT_IB_NF |
-				SND_SOC_DAIFMT_CBS_CFM);
+				SND_SOC_DAIFMT_CBM_CFM);
 	if (ret) {
 		printk(KERN_ERR "can't set codec DAI configuration\n");
 		return ret;
@@ -280,6 +286,7 @@ static struct snd_soc_card snd_soc_sdp3430 = {
 static struct twl4030_setup_data twl4030_setup = {
 	.ramp_delay_value = 3,
 	.sysclk = 26000,
+	.hs_extmute = 1,
 };
 
 /* Audio subsystem */
@@ -294,6 +301,7 @@ static struct platform_device *sdp3430_snd_device;
 static int __init sdp3430_soc_init(void)
 {
 	int ret;
+	u8 pin_mux;
 
 	if (!machine_is_omap_3430sdp()) {
 		pr_debug("Not SDP3430!\n");
@@ -312,6 +320,14 @@ static int __init sdp3430_soc_init(void)
 	*(unsigned int *)sdp3430_dai[0].cpu_dai->private_data = 1; /* McBSP2 */
 	*(unsigned int *)sdp3430_dai[1].cpu_dai->private_data = 2; /* McBSP3 */
 
+	/* Set TWL4030 GPIO6 as EXTMUTE signal */
+	twl4030_i2c_read_u8(TWL4030_MODULE_INTBR, &pin_mux,
+						TWL4030_INTBR_PMBR1);
+	pin_mux &= ~TWL4030_GPIO6_PWM0_MUTE(0x03);
+	pin_mux |= TWL4030_GPIO6_PWM0_MUTE(0x02);
+	twl4030_i2c_write_u8(TWL4030_MODULE_INTBR, pin_mux,
+						TWL4030_INTBR_PMBR1);
+
 	ret = platform_device_add(sdp3430_snd_device);
 	if (ret)
 		goto err1;
diff --git a/sound/soc/omap/zoom2.c b/sound/soc/omap/zoom2.c
new file mode 100644
index 00000000000..f90b45f5622
--- /dev/null
+++ b/sound/soc/omap/zoom2.c
@@ -0,0 +1,314 @@
+/*
+ * zoom2.c  --  SoC audio for Zoom2
+ *
+ * Author: Misael Lopez Cruz <x0052729@ti.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 <mach/gpio.h>
+#include <mach/mcbsp.h>
+
+#include "omap-mcbsp.h"
+#include "omap-pcm.h"
+#include "../codecs/twl4030.h"
+
+#define ZOOM2_HEADSET_MUX_GPIO		(OMAP_MAX_GPIO_LINES + 15)
+#define ZOOM2_HEADSET_EXTMUTE_GPIO	153
+
+static int zoom2_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 zoom2_ops = {
+	.hw_params = zoom2_hw_params,
+};
+
+static int zoom2_hw_voice_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_A |
+				SND_SOC_DAIFMT_IB_NF |
+				SND_SOC_DAIFMT_CBM_CFM);
+	if (ret) {
+		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_A |
+				SND_SOC_DAIFMT_IB_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 zoom2_voice_ops = {
+	.hw_params = zoom2_hw_voice_params,
+};
+
+/* Zoom2 machine DAPM */
+static const struct snd_soc_dapm_widget zoom2_twl4030_dapm_widgets[] = {
+	SND_SOC_DAPM_MIC("Ext Mic", NULL),
+	SND_SOC_DAPM_SPK("Ext Spk", NULL),
+	SND_SOC_DAPM_MIC("Headset Mic", NULL),
+	SND_SOC_DAPM_HP("Headset Stereophone", NULL),
+	SND_SOC_DAPM_LINE("Aux In", NULL),
+};
+
+static const struct snd_soc_dapm_route audio_map[] = {
+	/* External Mics: MAINMIC, SUBMIC with bias*/
+	{"MAINMIC", NULL, "Mic Bias 1"},
+	{"SUBMIC", NULL, "Mic Bias 2"},
+	{"Mic Bias 1", NULL, "Ext Mic"},
+	{"Mic Bias 2", NULL, "Ext Mic"},
+
+	/* External Speakers: HFL, HFR */
+	{"Ext Spk", NULL, "HFL"},
+	{"Ext Spk", NULL, "HFR"},
+
+	/* Headset Stereophone:  HSOL, HSOR */
+	{"Headset Stereophone", NULL, "HSOL"},
+	{"Headset Stereophone", NULL, "HSOR"},
+
+	/* Headset Mic: HSMIC with bias */
+	{"HSMIC", NULL, "Headset Mic Bias"},
+	{"Headset Mic Bias", NULL, "Headset Mic"},
+
+	/* Aux In: AUXL, AUXR */
+	{"Aux In", NULL, "AUXL"},
+	{"Aux In", NULL, "AUXR"},
+};
+
+static int zoom2_twl4030_init(struct snd_soc_codec *codec)
+{
+	int ret;
+
+	/* Add Zoom2 specific widgets */
+	ret = snd_soc_dapm_new_controls(codec, zoom2_twl4030_dapm_widgets,
+				ARRAY_SIZE(zoom2_twl4030_dapm_widgets));
+	if (ret)
+		return ret;
+
+	/* Set up Zoom2 specific audio path audio_map */
+	snd_soc_dapm_add_routes(codec, audio_map, ARRAY_SIZE(audio_map));
+
+	/* Zoom2 connected pins */
+	snd_soc_dapm_enable_pin(codec, "Ext Mic");
+	snd_soc_dapm_enable_pin(codec, "Ext Spk");
+	snd_soc_dapm_enable_pin(codec, "Headset Mic");
+	snd_soc_dapm_enable_pin(codec, "Headset Stereophone");
+	snd_soc_dapm_enable_pin(codec, "Aux In");
+
+	/* TWL4030 not connected pins */
+	snd_soc_dapm_nc_pin(codec, "CARKITMIC");
+	snd_soc_dapm_nc_pin(codec, "DIGIMIC0");
+	snd_soc_dapm_nc_pin(codec, "DIGIMIC1");
+
+	snd_soc_dapm_nc_pin(codec, "OUTL");
+	snd_soc_dapm_nc_pin(codec, "OUTR");
+	snd_soc_dapm_nc_pin(codec, "EARPIECE");
+	snd_soc_dapm_nc_pin(codec, "PREDRIVEL");
+	snd_soc_dapm_nc_pin(codec, "PREDRIVER");
+	snd_soc_dapm_nc_pin(codec, "CARKITL");
+	snd_soc_dapm_nc_pin(codec, "CARKITR");
+
+	ret = snd_soc_dapm_sync(codec);
+
+	return ret;
+}
+
+static int zoom2_twl4030_voice_init(struct snd_soc_codec *codec)
+{
+	unsigned short reg;
+
+	/* Enable voice interface */
+	reg = codec->read(codec, TWL4030_REG_VOICE_IF);
+	reg |= TWL4030_VIF_DIN_EN | TWL4030_VIF_DOUT_EN | TWL4030_VIF_EN;
+	codec->write(codec, TWL4030_REG_VOICE_IF, reg);
+
+	return 0;
+}
+
+/* Digital audio interface glue - connects codec <--> CPU */
+static struct snd_soc_dai_link zoom2_dai[] = {
+	{
+		.name = "TWL4030 I2S",
+		.stream_name = "TWL4030 Audio",
+		.cpu_dai = &omap_mcbsp_dai[0],
+		.codec_dai = &twl4030_dai[TWL4030_DAI_HIFI],
+		.init = zoom2_twl4030_init,
+		.ops = &zoom2_ops,
+	},
+	{
+		.name = "TWL4030 PCM",
+		.stream_name = "TWL4030 Voice",
+		.cpu_dai = &omap_mcbsp_dai[1],
+		.codec_dai = &twl4030_dai[TWL4030_DAI_VOICE],
+		.init = zoom2_twl4030_voice_init,
+		.ops = &zoom2_voice_ops,
+	},
+};
+
+/* Audio machine driver */
+static struct snd_soc_card snd_soc_zoom2 = {
+	.name = "Zoom2",
+	.platform = &omap_soc_platform,
+	.dai_link = zoom2_dai,
+	.num_links = ARRAY_SIZE(zoom2_dai),
+};
+
+/* EXTMUTE callback function */
+void zoom2_set_hs_extmute(int mute)
+{
+	gpio_set_value(ZOOM2_HEADSET_EXTMUTE_GPIO, mute);
+}
+
+/* twl4030 setup */
+static struct twl4030_setup_data twl4030_setup = {
+	.ramp_delay_value = 3,	/* 161 ms */
+	.sysclk = 26000,
+	.hs_extmute = 1,
+	.set_hs_extmute = zoom2_set_hs_extmute,
+};
+
+/* Audio subsystem */
+static struct snd_soc_device zoom2_snd_devdata = {
+	.card = &snd_soc_zoom2,
+	.codec_dev = &soc_codec_dev_twl4030,
+	.codec_data = &twl4030_setup,
+};
+
+static struct platform_device *zoom2_snd_device;
+
+static int __init zoom2_soc_init(void)
+{
+	int ret;
+
+	if (!machine_is_omap_zoom2()) {
+		pr_debug("Not Zoom2!\n");
+		return -ENODEV;
+	}
+	printk(KERN_INFO "Zoom2 SoC init\n");
+
+	zoom2_snd_device = platform_device_alloc("soc-audio", -1);
+	if (!zoom2_snd_device) {
+		printk(KERN_ERR "Platform device allocation failed\n");
+		return -ENOMEM;
+	}
+
+	platform_set_drvdata(zoom2_snd_device, &zoom2_snd_devdata);
+	zoom2_snd_devdata.dev = &zoom2_snd_device->dev;
+	*(unsigned int *)zoom2_dai[0].cpu_dai->private_data = 1; /* McBSP2 */
+	*(unsigned int *)zoom2_dai[1].cpu_dai->private_data = 2; /* McBSP3 */
+
+	ret = platform_device_add(zoom2_snd_device);
+	if (ret)
+		goto err1;
+
+	BUG_ON(gpio_request(ZOOM2_HEADSET_MUX_GPIO, "hs_mux") < 0);
+	gpio_direction_output(ZOOM2_HEADSET_MUX_GPIO, 0);
+
+	BUG_ON(gpio_request(ZOOM2_HEADSET_EXTMUTE_GPIO, "ext_mute") < 0);
+	gpio_direction_output(ZOOM2_HEADSET_EXTMUTE_GPIO, 0);
+
+	return 0;
+
+err1:
+	printk(KERN_ERR "Unable to add platform device\n");
+	platform_device_put(zoom2_snd_device);
+
+	return ret;
+}
+module_init(zoom2_soc_init);
+
+static void __exit zoom2_soc_exit(void)
+{
+	gpio_free(ZOOM2_HEADSET_MUX_GPIO);
+	gpio_free(ZOOM2_HEADSET_EXTMUTE_GPIO);
+
+	platform_device_unregister(zoom2_snd_device);
+}
+module_exit(zoom2_soc_exit);
+
+MODULE_AUTHOR("Misael Lopez Cruz <x0052729@ti.com>");
+MODULE_DESCRIPTION("ALSA SoC Zoom2");
+MODULE_LICENSE("GPL");
+