Fixed MTP to work with TWRP

sound/soc/omap/Kconfig

@@ -0,0 +1,119 @@
+config SND_OMAP_SOC
+	tristate "SoC Audio for the Texas Instruments OMAP chips"
+	depends on (ARCH_OMAP && DMA_OMAP) || (ARM && COMPILE_TEST)
+	select SND_DMAENGINE_PCM
+
+config SND_OMAP_SOC_DMIC
+	tristate
+
+config SND_OMAP_SOC_MCBSP
+	tristate
+
+config SND_OMAP_SOC_MCPDM
+	tristate
+
+config SND_OMAP_SOC_HDMI
+	tristate
+
+config SND_OMAP_SOC_N810
+	tristate "SoC Audio support for Nokia N810"
+	depends on SND_OMAP_SOC && MACH_NOKIA_N810 && I2C
+	depends on OMAP_MUX
+	select SND_OMAP_SOC_MCBSP
+	select SND_SOC_TLV320AIC3X
+	help
+	  Say Y if you want to add support for SoC audio on Nokia N810.
+
+config SND_OMAP_SOC_RX51
+	tristate "SoC Audio support for Nokia RX-51"
+	depends on SND_OMAP_SOC && ARM && (MACH_NOKIA_RX51 || COMPILE_TEST) && I2C
+	select SND_OMAP_SOC_MCBSP
+	select SND_SOC_TLV320AIC3X
+	select SND_SOC_TPA6130A2
+	depends on GPIOLIB
+	help
+	  Say Y if you want to add support for SoC audio on Nokia RX-51
+	  hardware. This is also known as Nokia N900 product.
+
+config SND_OMAP_SOC_AMS_DELTA
+	tristate "SoC Audio support for Amstrad E3 (Delta) videophone"
+	depends on SND_OMAP_SOC && MACH_AMS_DELTA && TTY
+	select SND_OMAP_SOC_MCBSP
+	select SND_SOC_CX20442
+	help
+	  Say Y  if you want to add support  for SoC audio device  connected to
+	  a handset and a speakerphone found on Amstrad E3 (Delta) videophone.
+
+	  Note that in order to get those devices fully supported,  you have to
+	  build  the kernel  with  standard  serial port  driver  included  and
+	  configured for at least 4 ports.  Then, from userspace, you must load
+	  a line discipline #19 on the modem (ttyS3) serial line.  The simplest
+	  way to achieve this is to install util-linux-ng  and use the included
+	  ldattach  utility.  This  can be  started  automatically  from  udev,
+	  a simple rule like this one should do the trick (it does for me):
+	  	ACTION=="add", KERNEL=="controlC0", \
+				RUN+="/usr/sbin/ldattach 19 /dev/ttyS3"
+
+config SND_OMAP_SOC_OSK5912
+	tristate "SoC Audio support for omap osk5912"
+	depends on SND_OMAP_SOC && MACH_OMAP_OSK && I2C
+	select SND_OMAP_SOC_MCBSP
+	select SND_SOC_TLV320AIC23_I2C
+	help
+	  Say Y if you want to add support for SoC audio on osk5912.
+
+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_I2C
+	help
+	  Say Y if you want to add support for SoC audio on the OMAP3517 / AM3517
+	  EVM.
+
+config SND_OMAP_SOC_OMAP_TWL4030
+	tristate "SoC Audio support for TI SoC based boards with twl4030 codec"
+	depends on TWL4030_CORE && SND_OMAP_SOC
+	select SND_OMAP_SOC_MCBSP
+	select SND_SOC_TWL4030
+	help
+	  Say Y if you want to add support for SoC audio on TI SoC based boards
+	  using twl4030 as c codec. This driver currently supports:
+	  - Beagleboard or Devkit8000
+	  - Gumstix Overo or CompuLab CM-T35/CM-T3730
+	  - IGEP v2
+	  - OMAP3EVM
+	  - SDP3430
+	  - Zoom2
+
+config SND_OMAP_SOC_OMAP_ABE_TWL6040
+	tristate "SoC Audio support for OMAP boards using ABE and twl6040 codec"
+	depends on TWL6040_CORE && SND_OMAP_SOC && (ARCH_OMAP4 || COMPILE_TEST)
+	select SND_OMAP_SOC_DMIC
+	select SND_OMAP_SOC_MCPDM
+	select SND_SOC_TWL6040
+	select SND_SOC_DMIC
+	help
+	  Say Y if you want to add support for SoC audio on OMAP boards using
+	  ABE and twl6040 codec. This driver currently supports:
+	  - SDP4430/Blaze boards
+	  - PandaBoard (4430)
+	  - PandaBoardES (4460)
+
+config SND_OMAP_SOC_OMAP_HDMI
+	tristate "SoC Audio support for Texas Instruments OMAP HDMI"
+	depends on SND_OMAP_SOC && OMAP4_DSS_HDMI && OMAP2_DSS
+	select SND_OMAP_SOC_HDMI
+	select SND_SOC_HDMI_CODEC
+	select OMAP4_DSS_HDMI_AUDIO
+	help
+	  Say Y if you want to add support for SoC HDMI audio on Texas Instruments
+	  OMAP4 chips
+
+config SND_OMAP_SOC_OMAP3_PANDORA
+	tristate "SoC Audio support for OMAP3 Pandora"
+	depends on TWL4030_CORE && SND_OMAP_SOC && MACH_OMAP3_PANDORA
+	select SND_OMAP_SOC_MCBSP
+	select SND_SOC_TWL4030
+	help
+	  Say Y if you want to add support for SoC audio on the OMAP3 Pandora.

sound/soc/omap/Makefile

@@ -0,0 +1,33 @@
+# OMAP Platform Support
+snd-soc-omap-objs := omap-pcm.o
+snd-soc-omap-dmic-objs := omap-dmic.o
+snd-soc-omap-mcbsp-objs := omap-mcbsp.o mcbsp.o
+snd-soc-omap-mcpdm-objs := omap-mcpdm.o
+snd-soc-omap-hdmi-objs := omap-hdmi.o
+
+obj-$(CONFIG_SND_OMAP_SOC) += snd-soc-omap.o
+obj-$(CONFIG_SND_OMAP_SOC_DMIC) += snd-soc-omap-dmic.o
+obj-$(CONFIG_SND_OMAP_SOC_MCBSP) += snd-soc-omap-mcbsp.o
+obj-$(CONFIG_SND_OMAP_SOC_MCPDM) += snd-soc-omap-mcpdm.o
+obj-$(CONFIG_SND_OMAP_SOC_HDMI) += snd-soc-omap-hdmi.o
+
+# OMAP Machine Support
+snd-soc-n810-objs := n810.o
+snd-soc-rx51-objs := rx51.o
+snd-soc-ams-delta-objs := ams-delta.o
+snd-soc-osk5912-objs := osk5912.o
+snd-soc-am3517evm-objs := am3517evm.o
+snd-soc-omap-abe-twl6040-objs := omap-abe-twl6040.o
+snd-soc-omap-twl4030-objs := omap-twl4030.o
+snd-soc-omap3pandora-objs := omap3pandora.o
+snd-soc-omap-hdmi-card-objs := omap-hdmi-card.o
+
+obj-$(CONFIG_SND_OMAP_SOC_N810) += snd-soc-n810.o
+obj-$(CONFIG_SND_OMAP_SOC_RX51) += snd-soc-rx51.o
+obj-$(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_AM3517EVM) += snd-soc-am3517evm.o
+obj-$(CONFIG_SND_OMAP_SOC_OMAP_ABE_TWL6040) += snd-soc-omap-abe-twl6040.o
+obj-$(CONFIG_SND_OMAP_SOC_OMAP_TWL4030) += snd-soc-omap-twl4030.o
+obj-$(CONFIG_SND_OMAP_SOC_OMAP3_PANDORA) += snd-soc-omap3pandora.o
+obj-$(CONFIG_SND_OMAP_SOC_OMAP_HDMI) += snd-soc-omap-hdmi-card.o

sound/soc/omap/am3517evm.c

@@ -0,0 +1,141 @@
+/*
+ * 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 <linux/module.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/soc.h>
+
+#include <asm/mach-types.h>
+#include <linux/platform_data/asoc-ti-mcbsp.h>
+
+#include "omap-mcbsp.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->codec_dai;
+	int 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;
+}
+
+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"},
+};
+
+/* Digital audio interface glue - connects codec <--> CPU */
+static struct snd_soc_dai_link am3517evm_dai = {
+	.name = "TLV320AIC23",
+	.stream_name = "AIC23",
+	.cpu_dai_name = "omap-mcbsp.1",
+	.codec_dai_name = "tlv320aic23-hifi",
+	.platform_name = "omap-mcbsp.1",
+	.codec_name = "tlv320aic23-codec.2-001a",
+	.dai_fmt = SND_SOC_DAIFMT_DSP_B | SND_SOC_DAIFMT_NB_NF |
+		   SND_SOC_DAIFMT_CBM_CFM,
+	.ops = &am3517evm_ops,
+};
+
+/* Audio machine driver */
+static struct snd_soc_card snd_soc_am3517evm = {
+	.name = "am3517evm",
+	.owner = THIS_MODULE,
+	.dai_link = &am3517evm_dai,
+	.num_links = 1,
+
+	.dapm_widgets = tlv320aic23_dapm_widgets,
+	.num_dapm_widgets = ARRAY_SIZE(tlv320aic23_dapm_widgets),
+	.dapm_routes = audio_map,
+	.num_dapm_routes = ARRAY_SIZE(audio_map),
+};
+
+static struct platform_device *am3517evm_snd_device;
+
+static int __init am3517evm_soc_init(void)
+{
+	int ret;
+
+	if (!machine_is_omap3517evm())
+		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, &snd_soc_am3517evm);
+
+	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");

sound/soc/omap/ams-delta.c

@@ -0,0 +1,613 @@
+/*
+ * 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 <linux/module.h>
+
+#include <sound/soc.h>
+#include <sound/jack.h>
+
+#include <asm/mach-types.h>
+
+#include <mach/board-ams-delta.h>
+#include <linux/platform_data/asoc-ti-mcbsp.h>
+
+#include "omap-mcbsp.h"
+#include "../codecs/cx20442.h"
+
+/* Board specific 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),
+	/* 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))
+
+static const 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;
+
+/*
+ * Used for passing a codec structure pointer
+ * from the board initialization code to the tty line discipline.
+ */
+static struct snd_soc_codec *cx20442_codec;
+
+static int ams_delta_set_audio_mode(struct snd_kcontrol *kcontrol,
+					struct snd_ctl_elem_value *ucontrol)
+{
+	struct snd_soc_card *card = snd_kcontrol_chip(kcontrol);
+	struct snd_soc_dapm_context *dapm = &card->dapm;
+	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 (!cx20442_codec->hw_write)
+		return -EUNATCH;
+
+	if (ucontrol->value.enumerated.item[0] >= control->items)
+		return -EINVAL;
+
+	snd_soc_dapm_mutex_lock(dapm);
+
+	/* 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(dapm, "Mouthpiece")) {
+		changed = 1;
+		if (pin)
+			snd_soc_dapm_enable_pin_unlocked(dapm, "Mouthpiece");
+		else
+			snd_soc_dapm_disable_pin_unlocked(dapm, "Mouthpiece");
+	}
+	pin = !!(pins & (1 << AMS_DELTA_EARPIECE));
+	if (pin != snd_soc_dapm_get_pin_status(dapm, "Earpiece")) {
+		changed = 1;
+		if (pin)
+			snd_soc_dapm_enable_pin_unlocked(dapm, "Earpiece");
+		else
+			snd_soc_dapm_disable_pin_unlocked(dapm, "Earpiece");
+	}
+	pin = !!(pins & (1 << AMS_DELTA_MICROPHONE));
+	if (pin != snd_soc_dapm_get_pin_status(dapm, "Microphone")) {
+		changed = 1;
+		if (pin)
+			snd_soc_dapm_enable_pin_unlocked(dapm, "Microphone");
+		else
+			snd_soc_dapm_disable_pin_unlocked(dapm, "Microphone");
+	}
+	pin = !!(pins & (1 << AMS_DELTA_SPEAKER));
+	if (pin != snd_soc_dapm_get_pin_status(dapm, "Speaker")) {
+		changed = 1;
+		if (pin)
+			snd_soc_dapm_enable_pin_unlocked(dapm, "Speaker");
+		else
+			snd_soc_dapm_disable_pin_unlocked(dapm, "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_unlocked(dapm, "AGCIN");
+		else
+			snd_soc_dapm_disable_pin_unlocked(dapm, "AGCIN");
+	}
+
+	if (changed)
+		snd_soc_dapm_sync_unlocked(dapm);
+
+	snd_soc_dapm_mutex_unlock(dapm);
+
+	return changed;
+}
+
+static int ams_delta_get_audio_mode(struct snd_kcontrol *kcontrol,
+					struct snd_ctl_elem_value *ucontrol)
+{
+	struct snd_soc_card *card = snd_kcontrol_chip(kcontrol);
+	struct snd_soc_dapm_context *dapm = &card->dapm;
+	unsigned short pins, mode;
+
+	pins = ((snd_soc_dapm_get_pin_status(dapm, "Mouthpiece") <<
+							AMS_DELTA_MOUTHPIECE) |
+			(snd_soc_dapm_get_pin_status(dapm, "Earpiece") <<
+							AMS_DELTA_EARPIECE));
+	if (pins)
+		pins |= (snd_soc_dapm_get_pin_status(dapm, "Microphone") <<
+							AMS_DELTA_MICROPHONE);
+	else
+		pins = ((snd_soc_dapm_get_pin_status(dapm, "Microphone") <<
+							AMS_DELTA_MICROPHONE) |
+			(snd_soc_dapm_get_pin_status(dapm, "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 SOC_ENUM_SINGLE_EXT_DECL(ams_delta_audio_enum,
+				      ams_delta_audio_mode);
+
+static const struct snd_kcontrol_new ams_delta_audio_controls[] = {
+	SOC_ENUM_EXT("Audio Mode", ams_delta_audio_enum,
+			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 careful 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)
+{
+	int ret;
+
+	if (!cx20442_codec)
+		return -ENODEV;
+
+	/*
+	 * Pass the codec structure pointer for use by other ldisc callbacks,
+	 * both the card and the codec specific parts.
+	 */
+	tty->disc_data = cx20442_codec;
+
+	ret = v253_ops.open(tty);
+
+	if (ret < 0)
+		tty->disc_data = NULL;
+
+	return ret;
+}
+
+/* Line discipline .close() */
+static void cx81801_close(struct tty_struct *tty)
+{
+	struct snd_soc_codec *codec = tty->disc_data;
+	struct snd_soc_dapm_context *dapm = &codec->component.card->dapm;
+
+	del_timer_sync(&cx81801_timer);
+
+	/* Prevent the hook switch from further changing the DAPM pins */
+	INIT_LIST_HEAD(&ams_delta_hook_switch.pins);
+
+	if (!codec)
+		return;
+
+	v253_ops.close(tty);
+
+	/* Revert back to default audio input/output constellation */
+	snd_soc_dapm_mutex_lock(dapm);
+
+	snd_soc_dapm_disable_pin_unlocked(dapm, "Mouthpiece");
+	snd_soc_dapm_enable_pin_unlocked(dapm, "Earpiece");
+	snd_soc_dapm_enable_pin_unlocked(dapm, "Microphone");
+	snd_soc_dapm_disable_pin_unlocked(dapm, "Speaker");
+	snd_soc_dapm_disable_pin_unlocked(dapm, "AGCIN");
+
+	snd_soc_dapm_sync_unlocked(dapm);
+
+	snd_soc_dapm_mutex_unlock(dapm);
+}
+
+/* Line discipline .hangup() */
+static int cx81801_hangup(struct tty_struct *tty)
+{
+	cx81801_close(tty);
+	return 0;
+}
+
+/* Line discipline .receive_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)
+		return;
+
+	if (!codec->hw_write) {
+		/* 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->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 useful, 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 issuing 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->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,
+};
+
+
+/* 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 const 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_pcm_runtime *rtd)
+{
+	struct snd_soc_dai *codec_dai = rtd->codec_dai;
+	struct snd_soc_card *card = rtd->card;
+	struct snd_soc_dapm_context *dapm = &card->dapm;
+	int ret;
+	/* Codec is ready, now add/activate board specific controls */
+
+	/* Store a pointer to the codec structure for tty ldisc use */
+	cx20442_codec = rtd->codec;
+
+	/* Set up digital mute if not provided by the codec */
+	if (!codec_dai->driver->ops) {
+		codec_dai->driver->ops = &ams_delta_dai_ops;
+	} else {
+		ams_delta_ops.startup = ams_delta_startup;
+		ams_delta_ops.shutdown = ams_delta_shutdown;
+	}
+
+	/* Add hook switch - can be used to control the codec from userspace
+	 * even if line discipline fails */
+	ret = snd_soc_jack_new(rtd->codec, "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;
+	}
+
+	/* Set up initial pin constellation */
+	snd_soc_dapm_disable_pin(dapm, "Mouthpiece");
+	snd_soc_dapm_disable_pin(dapm, "Speaker");
+	snd_soc_dapm_disable_pin(dapm, "AGCIN");
+	snd_soc_dapm_disable_pin(dapm, "AGCOUT");
+
+	return 0;
+}
+
+static int ams_delta_card_remove(struct snd_soc_card *card)
+{
+	snd_soc_jack_free_gpios(&ams_delta_hook_switch,
+			ARRAY_SIZE(ams_delta_hook_switch_gpios),
+			ams_delta_hook_switch_gpios);
+
+	return 0;
+}
+
+/* DAI glue - connects codec <--> CPU */
+static struct snd_soc_dai_link ams_delta_dai_link = {
+	.name = "CX20442",
+	.stream_name = "CX20442",
+	.cpu_dai_name = "omap-mcbsp.1",
+	.codec_dai_name = "cx20442-voice",
+	.init = ams_delta_cx20442_init,
+	.platform_name = "omap-mcbsp.1",
+	.codec_name = "cx20442-codec",
+	.ops = &ams_delta_ops,
+};
+
+/* Audio card driver */
+static struct snd_soc_card ams_delta_audio_card = {
+	.name = "AMS_DELTA",
+	.owner = THIS_MODULE,
+	.remove = ams_delta_card_remove,
+	.dai_link = &ams_delta_dai_link,
+	.num_links = 1,
+
+	.controls = ams_delta_audio_controls,
+	.num_controls = ARRAY_SIZE(ams_delta_audio_controls),
+	.dapm_widgets = ams_delta_dapm_widgets,
+	.num_dapm_widgets = ARRAY_SIZE(ams_delta_dapm_widgets),
+	.dapm_routes = ams_delta_audio_map,
+	.num_dapm_routes = ARRAY_SIZE(ams_delta_audio_map),
+};
+
+/* Module init/exit */
+static int ams_delta_probe(struct platform_device *pdev)
+{
+	struct snd_soc_card *card = &ams_delta_audio_card;
+	int ret;
+
+	card->dev = &pdev->dev;
+
+	ret = snd_soc_register_card(card);
+	if (ret) {
+		dev_err(&pdev->dev, "snd_soc_register_card failed (%d)\n", ret);
+		card->dev = NULL;
+		return ret;
+	}
+	return 0;
+}
+
+static int ams_delta_remove(struct platform_device *pdev)
+{
+	struct snd_soc_card *card = platform_get_drvdata(pdev);
+
+	if (tty_unregister_ldisc(N_V253) != 0)
+		dev_warn(&pdev->dev,
+			"failed to unregister V253 line discipline\n");
+
+	snd_soc_unregister_card(card);
+	card->dev = NULL;
+	return 0;
+}
+
+#define DRV_NAME "ams-delta-audio"
+
+static struct platform_driver ams_delta_driver = {
+	.driver = {
+		.name = DRV_NAME,
+		.owner = THIS_MODULE,
+	},
+	.probe = ams_delta_probe,
+	.remove = ams_delta_remove,
+};
+
+module_platform_driver(ams_delta_driver);
+
+MODULE_AUTHOR("Janusz Krzysztofik <jkrzyszt@tis.icnet.pl>");
+MODULE_DESCRIPTION("ALSA SoC driver for Amstrad E3 (Delta) videophone");
+MODULE_LICENSE("GPL");
+MODULE_ALIAS("platform:" DRV_NAME);

sound/soc/omap/mcbsp.h

@@ -0,0 +1,354 @@
+/*
+ * sound/soc/omap/mcbsp.h
+ *
+ * OMAP Multi-Channel Buffered Serial Port
+ *
+ * Contact: Jarkko Nikula <jarkko.nikula@bitmer.com>
+ *          Peter Ujfalusi <peter.ujfalusi@ti.com>
+ *
+ * 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; either version 2 of the License, or
+ * (at your option) any later version.
+ *
+ * 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., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
+ *
+ */
+#ifndef __ASOC_MCBSP_H
+#define __ASOC_MCBSP_H
+
+#ifdef CONFIG_ARCH_OMAP1
+#define mcbsp_omap1()	1
+#else
+#define mcbsp_omap1()	0
+#endif
+
+#include <sound/dmaengine_pcm.h>
+
+/* McBSP register numbers. Register address offset = num * reg_step */
+enum {
+	/* Common registers */
+	OMAP_MCBSP_REG_SPCR2 = 4,
+	OMAP_MCBSP_REG_SPCR1,
+	OMAP_MCBSP_REG_RCR2,
+	OMAP_MCBSP_REG_RCR1,
+	OMAP_MCBSP_REG_XCR2,
+	OMAP_MCBSP_REG_XCR1,
+	OMAP_MCBSP_REG_SRGR2,
+	OMAP_MCBSP_REG_SRGR1,
+	OMAP_MCBSP_REG_MCR2,
+	OMAP_MCBSP_REG_MCR1,
+	OMAP_MCBSP_REG_RCERA,
+	OMAP_MCBSP_REG_RCERB,
+	OMAP_MCBSP_REG_XCERA,
+	OMAP_MCBSP_REG_XCERB,
+	OMAP_MCBSP_REG_PCR0,
+	OMAP_MCBSP_REG_RCERC,
+	OMAP_MCBSP_REG_RCERD,
+	OMAP_MCBSP_REG_XCERC,
+	OMAP_MCBSP_REG_XCERD,
+	OMAP_MCBSP_REG_RCERE,
+	OMAP_MCBSP_REG_RCERF,
+	OMAP_MCBSP_REG_XCERE,
+	OMAP_MCBSP_REG_XCERF,
+	OMAP_MCBSP_REG_RCERG,
+	OMAP_MCBSP_REG_RCERH,
+	OMAP_MCBSP_REG_XCERG,
+	OMAP_MCBSP_REG_XCERH,
+
+	/* OMAP1-OMAP2420 registers */
+	OMAP_MCBSP_REG_DRR2 = 0,
+	OMAP_MCBSP_REG_DRR1,
+	OMAP_MCBSP_REG_DXR2,
+	OMAP_MCBSP_REG_DXR1,
+
+	/* OMAP2430 and onwards */
+	OMAP_MCBSP_REG_DRR = 0,
+	OMAP_MCBSP_REG_DXR = 2,
+	OMAP_MCBSP_REG_SYSCON =	35,
+	OMAP_MCBSP_REG_THRSH2,
+	OMAP_MCBSP_REG_THRSH1,
+	OMAP_MCBSP_REG_IRQST = 40,
+	OMAP_MCBSP_REG_IRQEN,
+	OMAP_MCBSP_REG_WAKEUPEN,
+	OMAP_MCBSP_REG_XCCR,
+	OMAP_MCBSP_REG_RCCR,
+	OMAP_MCBSP_REG_XBUFFSTAT,
+	OMAP_MCBSP_REG_RBUFFSTAT,
+	OMAP_MCBSP_REG_SSELCR,
+};
+
+/* OMAP3 sidetone control registers */
+#define OMAP_ST_REG_REV		0x00
+#define OMAP_ST_REG_SYSCONFIG	0x10
+#define OMAP_ST_REG_IRQSTATUS	0x18
+#define OMAP_ST_REG_IRQENABLE	0x1C
+#define OMAP_ST_REG_SGAINCR	0x24
+#define OMAP_ST_REG_SFIRCR	0x28
+#define OMAP_ST_REG_SSELCR	0x2C
+
+/************************** McBSP SPCR1 bit definitions ***********************/
+#define RRST			BIT(0)
+#define RRDY			BIT(1)
+#define RFULL			BIT(2)
+#define RSYNC_ERR		BIT(3)
+#define RINTM(value)		(((value) & 0x3) << 4)	/* bits 4:5 */
+#define ABIS			BIT(6)
+#define DXENA			BIT(7)
+#define CLKSTP(value)		(((value) & 0x3) << 11)	/* bits 11:12 */
+#define RJUST(value)		(((value) & 0x3) << 13)	/* bits 13:14 */
+#define ALB			BIT(15)
+#define DLB			BIT(15)
+
+/************************** McBSP SPCR2 bit definitions ***********************/
+#define XRST			BIT(0)
+#define XRDY			BIT(1)
+#define XEMPTY			BIT(2)
+#define XSYNC_ERR		BIT(3)
+#define XINTM(value)		(((value) & 0x3) << 4)	/* bits 4:5 */
+#define GRST			BIT(6)
+#define FRST			BIT(7)
+#define SOFT			BIT(8)
+#define FREE			BIT(9)
+
+/************************** McBSP PCR bit definitions *************************/
+#define CLKRP			BIT(0)
+#define CLKXP			BIT(1)
+#define FSRP			BIT(2)
+#define FSXP			BIT(3)
+#define DR_STAT			BIT(4)
+#define DX_STAT			BIT(5)
+#define CLKS_STAT		BIT(6)
+#define SCLKME			BIT(7)
+#define CLKRM			BIT(8)
+#define CLKXM			BIT(9)
+#define FSRM			BIT(10)
+#define FSXM			BIT(11)
+#define RIOEN			BIT(12)
+#define XIOEN			BIT(13)
+#define IDLE_EN			BIT(14)
+
+/************************** McBSP RCR1 bit definitions ************************/
+#define RWDLEN1(value)		(((value) & 0x7) << 5)	/* Bits 5:7 */
+#define RFRLEN1(value)		(((value) & 0x7f) << 8)	/* Bits 8:14 */
+
+/************************** McBSP XCR1 bit definitions ************************/
+#define XWDLEN1(value)		(((value) & 0x7) << 5)	/* Bits 5:7 */
+#define XFRLEN1(value)		(((value) & 0x7f) << 8)	/* Bits 8:14 */
+
+/*************************** McBSP RCR2 bit definitions ***********************/
+#define RDATDLY(value)		((value) & 0x3)		/* Bits 0:1 */
+#define RFIG			BIT(2)
+#define RCOMPAND(value)		(((value) & 0x3) << 3)	/* Bits 3:4 */
+#define RWDLEN2(value)		(((value) & 0x7) << 5)	/* Bits 5:7 */
+#define RFRLEN2(value)		(((value) & 0x7f) << 8)	/* Bits 8:14 */
+#define RPHASE			BIT(15)
+
+/*************************** McBSP XCR2 bit definitions ***********************/
+#define XDATDLY(value)		((value) & 0x3)		/* Bits 0:1 */
+#define XFIG			BIT(2)
+#define XCOMPAND(value)		(((value) & 0x3) << 3)	/* Bits 3:4 */
+#define XWDLEN2(value)		(((value) & 0x7) << 5)	/* Bits 5:7 */
+#define XFRLEN2(value)		(((value) & 0x7f) << 8)	/* Bits 8:14 */
+#define XPHASE			BIT(15)
+
+/************************* McBSP SRGR1 bit definitions ************************/
+#define CLKGDV(value)		((value) & 0x7f)		/* Bits 0:7 */
+#define FWID(value)		(((value) & 0xff) << 8)	/* Bits 8:15 */
+
+/************************* McBSP SRGR2 bit definitions ************************/
+#define FPER(value)		((value) & 0x0fff)	/* Bits 0:11 */
+#define FSGM			BIT(12)
+#define CLKSM			BIT(13)
+#define CLKSP			BIT(14)
+#define GSYNC			BIT(15)
+
+/************************* McBSP MCR1 bit definitions *************************/
+#define RMCM			BIT(0)
+#define RCBLK(value)		(((value) & 0x7) << 2)	/* Bits 2:4 */
+#define RPABLK(value)		(((value) & 0x3) << 5)	/* Bits 5:6 */
+#define RPBBLK(value)		(((value) & 0x3) << 7)	/* Bits 7:8 */
+
+/************************* McBSP MCR2 bit definitions *************************/
+#define XMCM(value)		((value) & 0x3)		/* Bits 0:1 */
+#define XCBLK(value)		(((value) & 0x7) << 2)	/* Bits 2:4 */
+#define XPABLK(value)		(((value) & 0x3) << 5)	/* Bits 5:6 */
+#define XPBBLK(value)		(((value) & 0x3) << 7)	/* Bits 7:8 */
+
+/*********************** McBSP XCCR bit definitions *************************/
+#define XDISABLE		BIT(0)
+#define XDMAEN			BIT(3)
+#define DILB			BIT(5)
+#define XFULL_CYCLE		BIT(11)
+#define DXENDLY(value)		(((value) & 0x3) << 12)	/* Bits 12:13 */
+#define PPCONNECT		BIT(14)
+#define EXTCLKGATE		BIT(15)
+
+/********************** McBSP RCCR bit definitions *************************/
+#define RDISABLE		BIT(0)
+#define RDMAEN			BIT(3)
+#define RFULL_CYCLE		BIT(11)
+
+/********************** McBSP SYSCONFIG bit definitions ********************/
+#define SOFTRST			BIT(1)
+#define ENAWAKEUP		BIT(2)
+#define SIDLEMODE(value)	(((value) & 0x3) << 3)
+#define CLOCKACTIVITY(value)	(((value) & 0x3) << 8)
+
+/********************** McBSP SSELCR bit definitions ***********************/
+#define SIDETONEEN		BIT(10)
+
+/********************** McBSP Sidetone SYSCONFIG bit definitions ***********/
+#define ST_AUTOIDLE		BIT(0)
+
+/********************** McBSP Sidetone SGAINCR bit definitions *************/
+#define ST_CH0GAIN(value)	((value) & 0xffff)	/* Bits 0:15 */
+#define ST_CH1GAIN(value)	(((value) & 0xffff) << 16) /* Bits 16:31 */
+
+/********************** McBSP Sidetone SFIRCR bit definitions **************/
+#define ST_FIRCOEFF(value)	((value) & 0xffff)	/* Bits 0:15 */
+
+/********************** McBSP Sidetone SSELCR bit definitions **************/
+#define ST_SIDETONEEN		BIT(0)
+#define ST_COEFFWREN		BIT(1)
+#define ST_COEFFWRDONE		BIT(2)
+
+/********************** McBSP DMA operating modes **************************/
+#define MCBSP_DMA_MODE_ELEMENT		0
+#define MCBSP_DMA_MODE_THRESHOLD	1
+
+/********************** McBSP WAKEUPEN/IRQST/IRQEN bit definitions *********/
+#define RSYNCERREN		BIT(0)
+#define RFSREN			BIT(1)
+#define REOFEN			BIT(2)
+#define RRDYEN			BIT(3)
+#define RUNDFLEN		BIT(4)
+#define ROVFLEN			BIT(5)
+#define XSYNCERREN		BIT(7)
+#define XFSXEN			BIT(8)
+#define XEOFEN			BIT(9)
+#define XRDYEN			BIT(10)
+#define XUNDFLEN		BIT(11)
+#define XOVFLEN			BIT(12)
+#define XEMPTYEOFEN		BIT(14)
+
+/* Clock signal muxing options */
+#define CLKR_SRC_CLKR		0 /* CLKR signal is from the CLKR pin */
+#define CLKR_SRC_CLKX		1 /* CLKR signal is from the CLKX pin */
+#define FSR_SRC_FSR		2 /* FSR signal is from the FSR pin */
+#define FSR_SRC_FSX		3 /* FSR signal is from the FSX pin */
+
+/* McBSP functional clock sources */
+#define MCBSP_CLKS_PRCM_SRC	0
+#define MCBSP_CLKS_PAD_SRC	1
+
+/* we don't do multichannel for now */
+struct omap_mcbsp_reg_cfg {
+	u16 spcr2;
+	u16 spcr1;
+	u16 rcr2;
+	u16 rcr1;
+	u16 xcr2;
+	u16 xcr1;
+	u16 srgr2;
+	u16 srgr1;
+	u16 mcr2;
+	u16 mcr1;
+	u16 pcr0;
+	u16 rcerc;
+	u16 rcerd;
+	u16 xcerc;
+	u16 xcerd;
+	u16 rcere;
+	u16 rcerf;
+	u16 xcere;
+	u16 xcerf;
+	u16 rcerg;
+	u16 rcerh;
+	u16 xcerg;
+	u16 xcerh;
+	u16 xccr;
+	u16 rccr;
+};
+
+struct omap_mcbsp_st_data {
+	void __iomem *io_base_st;
+	bool running;
+	bool enabled;
+	s16 taps[128];	/* Sidetone filter coefficients */
+	int nr_taps;	/* Number of filter coefficients in use */
+	s16 ch0gain;
+	s16 ch1gain;
+};
+
+struct omap_mcbsp {
+	struct device *dev;
+	struct clk *fclk;
+	spinlock_t lock;
+	unsigned long phys_base;
+	unsigned long phys_dma_base;
+	void __iomem *io_base;
+	u8 id;
+	/*
+	 * Flags indicating is the bus already activated and configured by
+	 * another substream
+	 */
+	int active;
+	int configured;
+	u8 free;
+
+	int irq;
+	int rx_irq;
+	int tx_irq;
+
+	/* Protect the field .free, while checking if the mcbsp is in use */
+	struct omap_mcbsp_platform_data *pdata;
+	struct omap_mcbsp_st_data *st_data;
+	struct omap_mcbsp_reg_cfg cfg_regs;
+	struct snd_dmaengine_dai_dma_data dma_data[2];
+	unsigned int dma_req[2];
+	int dma_op_mode;
+	u16 max_tx_thres;
+	u16 max_rx_thres;
+	void *reg_cache;
+	int reg_cache_size;
+
+	unsigned int fmt;
+	unsigned int in_freq;
+	int clk_div;
+	int wlen;
+};
+
+void omap_mcbsp_config(struct omap_mcbsp *mcbsp,
+		       const struct omap_mcbsp_reg_cfg *config);
+void omap_mcbsp_set_tx_threshold(struct omap_mcbsp *mcbsp, u16 threshold);
+void omap_mcbsp_set_rx_threshold(struct omap_mcbsp *mcbsp, u16 threshold);
+u16 omap_mcbsp_get_tx_delay(struct omap_mcbsp *mcbsp);
+u16 omap_mcbsp_get_rx_delay(struct omap_mcbsp *mcbsp);
+int omap_mcbsp_get_dma_op_mode(struct omap_mcbsp *mcbsp);
+int omap_mcbsp_request(struct omap_mcbsp *mcbsp);
+void omap_mcbsp_free(struct omap_mcbsp *mcbsp);
+void omap_mcbsp_start(struct omap_mcbsp *mcbsp, int tx, int rx);
+void omap_mcbsp_stop(struct omap_mcbsp *mcbsp, int tx, int rx);
+
+/* McBSP functional clock source changing function */
+int omap2_mcbsp_set_clks_src(struct omap_mcbsp *mcbsp, u8 fck_src_id);
+
+/* Sidetone specific API */
+int omap_st_set_chgain(struct omap_mcbsp *mcbsp, int channel, s16 chgain);
+int omap_st_get_chgain(struct omap_mcbsp *mcbsp, int channel, s16 *chgain);
+int omap_st_enable(struct omap_mcbsp *mcbsp);
+int omap_st_disable(struct omap_mcbsp *mcbsp);
+int omap_st_is_enabled(struct omap_mcbsp *mcbsp);
+
+int omap_mcbsp_init(struct platform_device *pdev);
+void omap_mcbsp_sysfs_remove(struct omap_mcbsp *mcbsp);
+
+#endif /* __ASOC_MCBSP_H */

sound/soc/omap/n810.c

@@ -0,0 +1,391 @@
+/*
+ * n810.c  --  SoC audio for Nokia N810
+ *
+ * Copyright (C) 2008 Nokia Corporation
+ *
+ * Contact: Jarkko Nikula <jarkko.nikula@bitmer.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/i2c.h>
+#include <linux/platform_device.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/soc.h>
+
+#include <asm/mach-types.h>
+#include <linux/gpio.h>
+#include <linux/module.h>
+#include <linux/platform_data/asoc-ti-mcbsp.h>
+
+#include "omap-mcbsp.h"
+
+#define N810_HEADSET_AMP_GPIO	10
+#define N810_SPEAKER_AMP_GPIO	101
+
+enum {
+	N810_JACK_DISABLED,
+	N810_JACK_HP,
+	N810_JACK_HS,
+	N810_JACK_MIC,
+};
+
+static struct clk *sys_clkout2;
+static struct clk *sys_clkout2_src;
+static struct clk *func96m_clk;
+
+static int n810_spk_func;
+static int n810_jack_func;
+static int n810_dmic_func;
+
+static void n810_ext_control(struct snd_soc_dapm_context *dapm)
+{
+	int hp = 0, line1l = 0;
+
+	switch (n810_jack_func) {
+	case N810_JACK_HS:
+		line1l = 1;
+	case N810_JACK_HP:
+		hp = 1;
+		break;
+	case N810_JACK_MIC:
+		line1l = 1;
+		break;
+	}
+
+	snd_soc_dapm_mutex_lock(dapm);
+
+	if (n810_spk_func)
+		snd_soc_dapm_enable_pin_unlocked(dapm, "Ext Spk");
+	else
+		snd_soc_dapm_disable_pin_unlocked(dapm, "Ext Spk");
+
+	if (hp)
+		snd_soc_dapm_enable_pin_unlocked(dapm, "Headphone Jack");
+	else
+		snd_soc_dapm_disable_pin_unlocked(dapm, "Headphone Jack");
+	if (line1l)
+		snd_soc_dapm_enable_pin_unlocked(dapm, "LINE1L");
+	else
+		snd_soc_dapm_disable_pin_unlocked(dapm, "LINE1L");
+
+	if (n810_dmic_func)
+		snd_soc_dapm_enable_pin_unlocked(dapm, "DMic");
+	else
+		snd_soc_dapm_disable_pin_unlocked(dapm, "DMic");
+
+	snd_soc_dapm_sync_unlocked(dapm);
+
+	snd_soc_dapm_mutex_unlock(dapm);
+}
+
+static int n810_startup(struct snd_pcm_substream *substream)
+{
+	struct snd_pcm_runtime *runtime = substream->runtime;
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_codec *codec = rtd->codec;
+
+	snd_pcm_hw_constraint_minmax(runtime,
+				     SNDRV_PCM_HW_PARAM_CHANNELS, 2, 2);
+
+	n810_ext_control(&codec->dapm);
+	return clk_prepare_enable(sys_clkout2);
+}
+
+static void n810_shutdown(struct snd_pcm_substream *substream)
+{
+	clk_disable_unprepare(sys_clkout2);
+}
+
+static int n810_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->codec_dai;
+	int err;
+
+	/* Set the codec system clock for DAC and ADC */
+	err = snd_soc_dai_set_sysclk(codec_dai, 0, 12000000,
+					    SND_SOC_CLOCK_IN);
+
+	return err;
+}
+
+static struct snd_soc_ops n810_ops = {
+	.startup = n810_startup,
+	.hw_params = n810_hw_params,
+	.shutdown = n810_shutdown,
+};
+
+static int n810_get_spk(struct snd_kcontrol *kcontrol,
+			struct snd_ctl_elem_value *ucontrol)
+{
+	ucontrol->value.integer.value[0] = n810_spk_func;
+
+	return 0;
+}
+
+static int n810_set_spk(struct snd_kcontrol *kcontrol,
+			struct snd_ctl_elem_value *ucontrol)
+{
+	struct snd_soc_card *card =  snd_kcontrol_chip(kcontrol);
+
+	if (n810_spk_func == ucontrol->value.integer.value[0])
+		return 0;
+
+	n810_spk_func = ucontrol->value.integer.value[0];
+	n810_ext_control(&card->dapm);
+
+	return 1;
+}
+
+static int n810_get_jack(struct snd_kcontrol *kcontrol,
+			 struct snd_ctl_elem_value *ucontrol)
+{
+	ucontrol->value.integer.value[0] = n810_jack_func;
+
+	return 0;
+}
+
+static int n810_set_jack(struct snd_kcontrol *kcontrol,
+			 struct snd_ctl_elem_value *ucontrol)
+{
+	struct snd_soc_card *card =  snd_kcontrol_chip(kcontrol);
+
+	if (n810_jack_func == ucontrol->value.integer.value[0])
+		return 0;
+
+	n810_jack_func = ucontrol->value.integer.value[0];
+	n810_ext_control(&card->dapm);
+
+	return 1;
+}
+
+static int n810_get_input(struct snd_kcontrol *kcontrol,
+			  struct snd_ctl_elem_value *ucontrol)
+{
+	ucontrol->value.integer.value[0] = n810_dmic_func;
+
+	return 0;
+}
+
+static int n810_set_input(struct snd_kcontrol *kcontrol,
+			  struct snd_ctl_elem_value *ucontrol)
+{
+	struct snd_soc_card *card =  snd_kcontrol_chip(kcontrol);
+
+	if (n810_dmic_func == ucontrol->value.integer.value[0])
+		return 0;
+
+	n810_dmic_func = ucontrol->value.integer.value[0];
+	n810_ext_control(&card->dapm);
+
+	return 1;
+}
+
+static int n810_spk_event(struct snd_soc_dapm_widget *w,
+			  struct snd_kcontrol *k, int event)
+{
+	if (SND_SOC_DAPM_EVENT_ON(event))
+		gpio_set_value(N810_SPEAKER_AMP_GPIO, 1);
+	else
+		gpio_set_value(N810_SPEAKER_AMP_GPIO, 0);
+
+	return 0;
+}
+
+static int n810_jack_event(struct snd_soc_dapm_widget *w,
+			   struct snd_kcontrol *k, int event)
+{
+	if (SND_SOC_DAPM_EVENT_ON(event))
+		gpio_set_value(N810_HEADSET_AMP_GPIO, 1);
+	else
+		gpio_set_value(N810_HEADSET_AMP_GPIO, 0);
+
+	return 0;
+}
+
+static const struct snd_soc_dapm_widget aic33_dapm_widgets[] = {
+	SND_SOC_DAPM_SPK("Ext Spk", n810_spk_event),
+	SND_SOC_DAPM_HP("Headphone Jack", n810_jack_event),
+	SND_SOC_DAPM_MIC("DMic", NULL),
+};
+
+static const struct snd_soc_dapm_route audio_map[] = {
+	{"Headphone Jack", NULL, "HPLOUT"},
+	{"Headphone Jack", NULL, "HPROUT"},
+
+	{"Ext Spk", NULL, "LLOUT"},
+	{"Ext Spk", NULL, "RLOUT"},
+
+	{"DMic Rate 64", NULL, "Mic Bias"},
+	{"Mic Bias", NULL, "DMic"},
+};
+
+static const char *spk_function[] = {"Off", "On"};
+static const char *jack_function[] = {"Off", "Headphone", "Headset", "Mic"};
+static const char *input_function[] = {"ADC", "Digital Mic"};
+static const struct soc_enum n810_enum[] = {
+	SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(spk_function), spk_function),
+	SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(jack_function), jack_function),
+	SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(input_function), input_function),
+};
+
+static const struct snd_kcontrol_new aic33_n810_controls[] = {
+	SOC_ENUM_EXT("Speaker Function", n810_enum[0],
+		     n810_get_spk, n810_set_spk),
+	SOC_ENUM_EXT("Jack Function", n810_enum[1],
+		     n810_get_jack, n810_set_jack),
+	SOC_ENUM_EXT("Input Select",  n810_enum[2],
+		     n810_get_input, n810_set_input),
+};
+
+static int n810_aic33_init(struct snd_soc_pcm_runtime *rtd)
+{
+	struct snd_soc_codec *codec = rtd->codec;
+	struct snd_soc_dapm_context *dapm = &codec->dapm;
+
+	/* Not connected */
+	snd_soc_dapm_nc_pin(dapm, "MONO_LOUT");
+	snd_soc_dapm_nc_pin(dapm, "HPLCOM");
+	snd_soc_dapm_nc_pin(dapm, "HPRCOM");
+	snd_soc_dapm_nc_pin(dapm, "MIC3L");
+	snd_soc_dapm_nc_pin(dapm, "MIC3R");
+	snd_soc_dapm_nc_pin(dapm, "LINE1R");
+	snd_soc_dapm_nc_pin(dapm, "LINE2L");
+	snd_soc_dapm_nc_pin(dapm, "LINE2R");
+
+	return 0;
+}
+
+/* Digital audio interface glue - connects codec <--> CPU */
+static struct snd_soc_dai_link n810_dai = {
+	.name = "TLV320AIC33",
+	.stream_name = "AIC33",
+	.cpu_dai_name = "omap-mcbsp.2",
+	.platform_name = "omap-mcbsp.2",
+	.codec_name = "tlv320aic3x-codec.2-0018",
+	.codec_dai_name = "tlv320aic3x-hifi",
+	.dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF |
+		   SND_SOC_DAIFMT_CBM_CFM,
+	.init = n810_aic33_init,
+	.ops = &n810_ops,
+};
+
+/* Audio machine driver */
+static struct snd_soc_card snd_soc_n810 = {
+	.name = "N810",
+	.owner = THIS_MODULE,
+	.dai_link = &n810_dai,
+	.num_links = 1,
+
+	.controls = aic33_n810_controls,
+	.num_controls = ARRAY_SIZE(aic33_n810_controls),
+	.dapm_widgets = aic33_dapm_widgets,
+	.num_dapm_widgets = ARRAY_SIZE(aic33_dapm_widgets),
+	.dapm_routes = audio_map,
+	.num_dapm_routes = ARRAY_SIZE(audio_map),
+};
+
+static struct platform_device *n810_snd_device;
+
+static int __init n810_soc_init(void)
+{
+	int err;
+	struct device *dev;
+
+	if (!of_have_populated_dt() ||
+	    (!of_machine_is_compatible("nokia,n810") &&
+	     !of_machine_is_compatible("nokia,n810-wimax")))
+		return -ENODEV;
+
+	n810_snd_device = platform_device_alloc("soc-audio", -1);
+	if (!n810_snd_device)
+		return -ENOMEM;
+
+	platform_set_drvdata(n810_snd_device, &snd_soc_n810);
+	err = platform_device_add(n810_snd_device);
+	if (err)
+		goto err1;
+
+	dev = &n810_snd_device->dev;
+
+	sys_clkout2_src = clk_get(dev, "sys_clkout2_src");
+	if (IS_ERR(sys_clkout2_src)) {
+		dev_err(dev, "Could not get sys_clkout2_src clock\n");
+		err = PTR_ERR(sys_clkout2_src);
+		goto err2;
+	}
+	sys_clkout2 = clk_get(dev, "sys_clkout2");
+	if (IS_ERR(sys_clkout2)) {
+		dev_err(dev, "Could not get sys_clkout2\n");
+		err = PTR_ERR(sys_clkout2);
+		goto err3;
+	}
+	/*
+	 * Configure 12 MHz output on SYS_CLKOUT2. Therefore we must use
+	 * 96 MHz as its parent in order to get 12 MHz
+	 */
+	func96m_clk = clk_get(dev, "func_96m_ck");
+	if (IS_ERR(func96m_clk)) {
+		dev_err(dev, "Could not get func 96M clock\n");
+		err = PTR_ERR(func96m_clk);
+		goto err4;
+	}
+	clk_set_parent(sys_clkout2_src, func96m_clk);
+	clk_set_rate(sys_clkout2, 12000000);
+
+	if (WARN_ON((gpio_request(N810_HEADSET_AMP_GPIO, "hs_amp") < 0) ||
+		    (gpio_request(N810_SPEAKER_AMP_GPIO, "spk_amp") < 0))) {
+		err = -EINVAL;
+		goto err4;
+	}
+
+	gpio_direction_output(N810_HEADSET_AMP_GPIO, 0);
+	gpio_direction_output(N810_SPEAKER_AMP_GPIO, 0);
+
+	return 0;
+err4:
+	clk_put(sys_clkout2);
+err3:
+	clk_put(sys_clkout2_src);
+err2:
+	platform_device_del(n810_snd_device);
+err1:
+	platform_device_put(n810_snd_device);
+
+	return err;
+}
+
+static void __exit n810_soc_exit(void)
+{
+	gpio_free(N810_SPEAKER_AMP_GPIO);
+	gpio_free(N810_HEADSET_AMP_GPIO);
+	clk_put(sys_clkout2_src);
+	clk_put(sys_clkout2);
+	clk_put(func96m_clk);
+
+	platform_device_unregister(n810_snd_device);
+}
+
+module_init(n810_soc_init);
+module_exit(n810_soc_exit);
+
+MODULE_AUTHOR("Jarkko Nikula <jarkko.nikula@bitmer.com>");
+MODULE_DESCRIPTION("ALSA SoC Nokia N810");
+MODULE_LICENSE("GPL");

sound/soc/omap/omap-abe-twl6040.c

@@ -0,0 +1,370 @@
+/*
+ * omap-abe-twl6040.c  --  SoC audio for TI OMAP based boards with ABE and
+ *			   twl6040 codec
+ *
+ * Author: Misael Lopez Cruz <misael.lopez@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 <linux/mfd/twl6040.h>
+#include <linux/module.h>
+#include <linux/of.h>
+
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/soc.h>
+#include <sound/jack.h>
+
+#include "omap-dmic.h"
+#include "omap-mcpdm.h"
+#include "../codecs/twl6040.h"
+
+struct abe_twl6040 {
+	int	jack_detection;	/* board can detect jack events */
+	int	mclk_freq;	/* MCLK frequency speed for twl6040 */
+
+	struct platform_device *dmic_codec_dev;
+};
+
+static int omap_abe_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->codec_dai;
+	struct snd_soc_card *card = rtd->card;
+	struct abe_twl6040 *priv = snd_soc_card_get_drvdata(card);
+	int clk_id, freq;
+	int ret;
+
+	clk_id = twl6040_get_clk_id(rtd->codec);
+	if (clk_id == TWL6040_SYSCLK_SEL_HPPLL)
+		freq = priv->mclk_freq;
+	else if (clk_id == TWL6040_SYSCLK_SEL_LPPLL)
+		freq = 32768;
+	else
+		return -EINVAL;
+
+	/* set the codec mclk */
+	ret = snd_soc_dai_set_sysclk(codec_dai, clk_id, freq,
+				SND_SOC_CLOCK_IN);
+	if (ret) {
+		printk(KERN_ERR "can't set codec system clock\n");
+		return ret;
+	}
+	return ret;
+}
+
+static struct snd_soc_ops omap_abe_ops = {
+	.hw_params = omap_abe_hw_params,
+};
+
+static int omap_abe_dmic_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 *cpu_dai = rtd->cpu_dai;
+	int ret = 0;
+
+	ret = snd_soc_dai_set_sysclk(cpu_dai, OMAP_DMIC_SYSCLK_PAD_CLKS,
+				     19200000, SND_SOC_CLOCK_IN);
+	if (ret < 0) {
+		printk(KERN_ERR "can't set DMIC cpu system clock\n");
+		return ret;
+	}
+	ret = snd_soc_dai_set_sysclk(cpu_dai, OMAP_DMIC_ABE_DMIC_CLK, 2400000,
+				     SND_SOC_CLOCK_OUT);
+	if (ret < 0) {
+		printk(KERN_ERR "can't set DMIC output clock\n");
+		return ret;
+	}
+	return 0;
+}
+
+static struct snd_soc_ops omap_abe_dmic_ops = {
+	.hw_params = omap_abe_dmic_hw_params,
+};
+
+/* Headset jack */
+static struct snd_soc_jack hs_jack;
+
+/*Headset jack detection DAPM pins */
+static struct snd_soc_jack_pin hs_jack_pins[] = {
+	{
+		.pin = "Headset Mic",
+		.mask = SND_JACK_MICROPHONE,
+	},
+	{
+		.pin = "Headset Stereophone",
+		.mask = SND_JACK_HEADPHONE,
+	},
+};
+
+/* SDP4430 machine DAPM */
+static const struct snd_soc_dapm_widget twl6040_dapm_widgets[] = {
+	/* Outputs */
+	SND_SOC_DAPM_HP("Headset Stereophone", NULL),
+	SND_SOC_DAPM_SPK("Earphone Spk", NULL),
+	SND_SOC_DAPM_SPK("Ext Spk", NULL),
+	SND_SOC_DAPM_LINE("Line Out", NULL),
+	SND_SOC_DAPM_SPK("Vibrator", NULL),
+
+	/* Inputs */
+	SND_SOC_DAPM_MIC("Headset Mic", NULL),
+	SND_SOC_DAPM_MIC("Main Handset Mic", NULL),
+	SND_SOC_DAPM_MIC("Sub Handset Mic", NULL),
+	SND_SOC_DAPM_LINE("Line In", NULL),
+
+	/* Digital microphones */
+	SND_SOC_DAPM_MIC("Digital Mic", NULL),
+};
+
+static const struct snd_soc_dapm_route audio_map[] = {
+	/* Routings for outputs */
+	{"Headset Stereophone", NULL, "HSOL"},
+	{"Headset Stereophone", NULL, "HSOR"},
+
+	{"Earphone Spk", NULL, "EP"},
+
+	{"Ext Spk", NULL, "HFL"},
+	{"Ext Spk", NULL, "HFR"},
+
+	{"Line Out", NULL, "AUXL"},
+	{"Line Out", NULL, "AUXR"},
+
+	{"Vibrator", NULL, "VIBRAL"},
+	{"Vibrator", NULL, "VIBRAR"},
+
+	/* Routings for inputs */
+	{"HSMIC", NULL, "Headset Mic"},
+	{"Headset Mic", NULL, "Headset Mic Bias"},
+
+	{"MAINMIC", NULL, "Main Handset Mic"},
+	{"Main Handset Mic", NULL, "Main Mic Bias"},
+
+	{"SUBMIC", NULL, "Sub Handset Mic"},
+	{"Sub Handset Mic", NULL, "Main Mic Bias"},
+
+	{"AFML", NULL, "Line In"},
+	{"AFMR", NULL, "Line In"},
+};
+
+static int omap_abe_twl6040_init(struct snd_soc_pcm_runtime *rtd)
+{
+	struct snd_soc_codec *codec = rtd->codec;
+	struct snd_soc_card *card = rtd->card;
+	struct abe_twl6040 *priv = snd_soc_card_get_drvdata(card);
+	int hs_trim;
+	int ret = 0;
+
+	/*
+	 * Configure McPDM offset cancellation based on the HSOTRIM value from
+	 * twl6040.
+	 */
+	hs_trim = twl6040_get_trim_value(codec, TWL6040_TRIM_HSOTRIM);
+	omap_mcpdm_configure_dn_offsets(rtd, TWL6040_HSF_TRIM_LEFT(hs_trim),
+					TWL6040_HSF_TRIM_RIGHT(hs_trim));
+
+	/* Headset jack detection only if it is supported */
+	if (priv->jack_detection) {
+		ret = snd_soc_jack_new(codec, "Headset Jack",
+					SND_JACK_HEADSET, &hs_jack);
+		if (ret)
+			return ret;
+
+		ret = snd_soc_jack_add_pins(&hs_jack, ARRAY_SIZE(hs_jack_pins),
+					hs_jack_pins);
+		twl6040_hs_jack_detect(codec, &hs_jack, SND_JACK_HEADSET);
+	}
+
+	return ret;
+}
+
+static const struct snd_soc_dapm_route dmic_audio_map[] = {
+	{"DMic", NULL, "Digital Mic"},
+	{"Digital Mic", NULL, "Digital Mic1 Bias"},
+};
+
+static int omap_abe_dmic_init(struct snd_soc_pcm_runtime *rtd)
+{
+	struct snd_soc_dapm_context *dapm = &rtd->card->dapm;
+
+	return snd_soc_dapm_add_routes(dapm, dmic_audio_map,
+				ARRAY_SIZE(dmic_audio_map));
+}
+
+/* Digital audio interface glue - connects codec <--> CPU */
+static struct snd_soc_dai_link abe_twl6040_dai_links[] = {
+	{
+		.name = "TWL6040",
+		.stream_name = "TWL6040",
+		.codec_dai_name = "twl6040-legacy",
+		.codec_name = "twl6040-codec",
+		.init = omap_abe_twl6040_init,
+		.ops = &omap_abe_ops,
+	},
+	{
+		.name = "DMIC",
+		.stream_name = "DMIC Capture",
+		.codec_dai_name = "dmic-hifi",
+		.codec_name = "dmic-codec",
+		.init = omap_abe_dmic_init,
+		.ops = &omap_abe_dmic_ops,
+	},
+};
+
+/* Audio machine driver */
+static struct snd_soc_card omap_abe_card = {
+	.owner = THIS_MODULE,
+
+	.dapm_widgets = twl6040_dapm_widgets,
+	.num_dapm_widgets = ARRAY_SIZE(twl6040_dapm_widgets),
+	.dapm_routes = audio_map,
+	.num_dapm_routes = ARRAY_SIZE(audio_map),
+};
+
+static int omap_abe_probe(struct platform_device *pdev)
+{
+	struct device_node *node = pdev->dev.of_node;
+	struct snd_soc_card *card = &omap_abe_card;
+	struct device_node *dai_node;
+	struct abe_twl6040 *priv;
+	int num_links = 0;
+	int ret = 0;
+
+	if (!node) {
+		dev_err(&pdev->dev, "of node is missing.\n");
+		return -ENODEV;
+	}
+
+	card->dev = &pdev->dev;
+
+	priv = devm_kzalloc(&pdev->dev, sizeof(struct abe_twl6040), GFP_KERNEL);
+	if (priv == NULL)
+		return -ENOMEM;
+
+	priv->dmic_codec_dev = ERR_PTR(-EINVAL);
+
+	if (snd_soc_of_parse_card_name(card, "ti,model")) {
+		dev_err(&pdev->dev, "Card name is not provided\n");
+		return -ENODEV;
+	}
+
+	ret = snd_soc_of_parse_audio_routing(card, "ti,audio-routing");
+	if (ret) {
+		dev_err(&pdev->dev, "Error while parsing DAPM routing\n");
+		return ret;
+	}
+
+	dai_node = of_parse_phandle(node, "ti,mcpdm", 0);
+	if (!dai_node) {
+		dev_err(&pdev->dev, "McPDM node is not provided\n");
+		return -EINVAL;
+	}
+	abe_twl6040_dai_links[0].cpu_of_node = dai_node;
+	abe_twl6040_dai_links[0].platform_of_node = dai_node;
+
+	dai_node = of_parse_phandle(node, "ti,dmic", 0);
+	if (dai_node) {
+		num_links = 2;
+		abe_twl6040_dai_links[1].cpu_of_node = dai_node;
+		abe_twl6040_dai_links[1].platform_of_node = dai_node;
+
+		priv->dmic_codec_dev = platform_device_register_simple(
+						"dmic-codec", -1, NULL, 0);
+		if (IS_ERR(priv->dmic_codec_dev)) {
+			dev_err(&pdev->dev, "Can't instantiate dmic-codec\n");
+			return PTR_ERR(priv->dmic_codec_dev);
+		}
+	} else {
+		num_links = 1;
+	}
+
+	priv->jack_detection = of_property_read_bool(node, "ti,jack-detection");
+	of_property_read_u32(node, "ti,mclk-freq", &priv->mclk_freq);
+	if (!priv->mclk_freq) {
+		dev_err(&pdev->dev, "MCLK frequency not provided\n");
+		ret = -EINVAL;
+		goto err_unregister;
+	}
+
+	card->fully_routed = 1;
+
+	if (!priv->mclk_freq) {
+		dev_err(&pdev->dev, "MCLK frequency missing\n");
+		ret = -ENODEV;
+		goto err_unregister;
+	}
+
+	card->dai_link = abe_twl6040_dai_links;
+	card->num_links = num_links;
+
+	snd_soc_card_set_drvdata(card, priv);
+
+	ret = snd_soc_register_card(card);
+	if (ret) {
+		dev_err(&pdev->dev, "snd_soc_register_card() failed: %d\n",
+			ret);
+		goto err_unregister;
+	}
+
+	return 0;
+
+err_unregister:
+	if (!IS_ERR(priv->dmic_codec_dev))
+		platform_device_unregister(priv->dmic_codec_dev);
+
+	return ret;
+}
+
+static int omap_abe_remove(struct platform_device *pdev)
+{
+	struct snd_soc_card *card = platform_get_drvdata(pdev);
+	struct abe_twl6040 *priv = snd_soc_card_get_drvdata(card);
+
+	snd_soc_unregister_card(card);
+
+	if (!IS_ERR(priv->dmic_codec_dev))
+		platform_device_unregister(priv->dmic_codec_dev);
+
+	return 0;
+}
+
+static const struct of_device_id omap_abe_of_match[] = {
+	{.compatible = "ti,abe-twl6040", },
+	{ },
+};
+MODULE_DEVICE_TABLE(of, omap_abe_of_match);
+
+static struct platform_driver omap_abe_driver = {
+	.driver = {
+		.name = "omap-abe-twl6040",
+		.owner = THIS_MODULE,
+		.pm = &snd_soc_pm_ops,
+		.of_match_table = omap_abe_of_match,
+	},
+	.probe = omap_abe_probe,
+	.remove = omap_abe_remove,
+};
+
+module_platform_driver(omap_abe_driver);
+
+MODULE_AUTHOR("Misael Lopez Cruz <misael.lopez@ti.com>");
+MODULE_DESCRIPTION("ALSA SoC for OMAP boards with ABE and twl6040 codec");
+MODULE_LICENSE("GPL");
+MODULE_ALIAS("platform:omap-abe-twl6040");

sound/soc/omap/omap-dmic.c

@@ -0,0 +1,523 @@
+/*
+ * omap-dmic.c  --  OMAP ASoC DMIC DAI driver
+ *
+ * Copyright (C) 2010 - 2011 Texas Instruments
+ *
+ * Author: David Lambert <dlambert@ti.com>
+ *	   Misael Lopez Cruz <misael.lopez@ti.com>
+ *	   Liam Girdwood <lrg@ti.com>
+ *	   Peter Ujfalusi <peter.ujfalusi@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/init.h>
+#include <linux/module.h>
+#include <linux/platform_device.h>
+#include <linux/err.h>
+#include <linux/clk.h>
+#include <linux/io.h>
+#include <linux/slab.h>
+#include <linux/pm_runtime.h>
+#include <linux/of_device.h>
+
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/initval.h>
+#include <sound/soc.h>
+#include <sound/dmaengine_pcm.h>
+#include <sound/omap-pcm.h>
+
+#include "omap-dmic.h"
+
+struct omap_dmic {
+	struct device *dev;
+	void __iomem *io_base;
+	struct clk *fclk;
+	int fclk_freq;
+	int out_freq;
+	int clk_div;
+	int sysclk;
+	int threshold;
+	u32 ch_enabled;
+	bool active;
+	struct mutex mutex;
+
+	struct snd_dmaengine_dai_dma_data dma_data;
+};
+
+static inline void omap_dmic_write(struct omap_dmic *dmic, u16 reg, u32 val)
+{
+	writel_relaxed(val, dmic->io_base + reg);
+}
+
+static inline int omap_dmic_read(struct omap_dmic *dmic, u16 reg)
+{
+	return readl_relaxed(dmic->io_base + reg);
+}
+
+static inline void omap_dmic_start(struct omap_dmic *dmic)
+{
+	u32 ctrl = omap_dmic_read(dmic, OMAP_DMIC_CTRL_REG);
+
+	/* Configure DMA controller */
+	omap_dmic_write(dmic, OMAP_DMIC_DMAENABLE_SET_REG,
+			OMAP_DMIC_DMA_ENABLE);
+
+	omap_dmic_write(dmic, OMAP_DMIC_CTRL_REG, ctrl | dmic->ch_enabled);
+}
+
+static inline void omap_dmic_stop(struct omap_dmic *dmic)
+{
+	u32 ctrl = omap_dmic_read(dmic, OMAP_DMIC_CTRL_REG);
+	omap_dmic_write(dmic, OMAP_DMIC_CTRL_REG,
+			ctrl & ~OMAP_DMIC_UP_ENABLE_MASK);
+
+	/* Disable DMA request generation */
+	omap_dmic_write(dmic, OMAP_DMIC_DMAENABLE_CLR_REG,
+			OMAP_DMIC_DMA_ENABLE);
+
+}
+
+static inline int dmic_is_enabled(struct omap_dmic *dmic)
+{
+	return omap_dmic_read(dmic, OMAP_DMIC_CTRL_REG) &
+						OMAP_DMIC_UP_ENABLE_MASK;
+}
+
+static int omap_dmic_dai_startup(struct snd_pcm_substream *substream,
+				  struct snd_soc_dai *dai)
+{
+	struct omap_dmic *dmic = snd_soc_dai_get_drvdata(dai);
+	int ret = 0;
+
+	mutex_lock(&dmic->mutex);
+
+	if (!dai->active)
+		dmic->active = 1;
+	else
+		ret = -EBUSY;
+
+	mutex_unlock(&dmic->mutex);
+
+	return ret;
+}
+
+static void omap_dmic_dai_shutdown(struct snd_pcm_substream *substream,
+				    struct snd_soc_dai *dai)
+{
+	struct omap_dmic *dmic = snd_soc_dai_get_drvdata(dai);
+
+	mutex_lock(&dmic->mutex);
+
+	if (!dai->active)
+		dmic->active = 0;
+
+	mutex_unlock(&dmic->mutex);
+}
+
+static int omap_dmic_select_divider(struct omap_dmic *dmic, int sample_rate)
+{
+	int divider = -EINVAL;
+
+	/*
+	 * 192KHz rate is only supported with 19.2MHz/3.84MHz clock
+	 * configuration.
+	 */
+	if (sample_rate == 192000) {
+		if (dmic->fclk_freq == 19200000 && dmic->out_freq == 3840000)
+			divider = 0x6; /* Divider: 5 (192KHz sampling rate) */
+		else
+			dev_err(dmic->dev,
+				"invalid clock configuration for 192KHz\n");
+
+		return divider;
+	}
+
+	switch (dmic->out_freq) {
+	case 1536000:
+		if (dmic->fclk_freq != 24576000)
+			goto div_err;
+		divider = 0x4; /* Divider: 16 */
+		break;
+	case 2400000:
+		switch (dmic->fclk_freq) {
+		case 12000000:
+			divider = 0x5; /* Divider: 5 */
+			break;
+		case 19200000:
+			divider = 0x0; /* Divider: 8 */
+			break;
+		case 24000000:
+			divider = 0x2; /* Divider: 10 */
+			break;
+		default:
+			goto div_err;
+		}
+		break;
+	case 3072000:
+		if (dmic->fclk_freq != 24576000)
+			goto div_err;
+		divider = 0x3; /* Divider: 8 */
+		break;
+	case 3840000:
+		if (dmic->fclk_freq != 19200000)
+			goto div_err;
+		divider = 0x1; /* Divider: 5 (96KHz sampling rate) */
+		break;
+	default:
+		dev_err(dmic->dev, "invalid out frequency: %dHz\n",
+			dmic->out_freq);
+		break;
+	}
+
+	return divider;
+
+div_err:
+	dev_err(dmic->dev, "invalid out frequency %dHz for %dHz input\n",
+		dmic->out_freq, dmic->fclk_freq);
+	return -EINVAL;
+}
+
+static int omap_dmic_dai_hw_params(struct snd_pcm_substream *substream,
+				    struct snd_pcm_hw_params *params,
+				    struct snd_soc_dai *dai)
+{
+	struct omap_dmic *dmic = snd_soc_dai_get_drvdata(dai);
+	struct snd_dmaengine_dai_dma_data *dma_data;
+	int channels;
+
+	dmic->clk_div = omap_dmic_select_divider(dmic, params_rate(params));
+	if (dmic->clk_div < 0) {
+		dev_err(dmic->dev, "no valid divider for %dHz from %dHz\n",
+			dmic->out_freq, dmic->fclk_freq);
+		return -EINVAL;
+	}
+
+	dmic->ch_enabled = 0;
+	channels = params_channels(params);
+	switch (channels) {
+	case 6:
+		dmic->ch_enabled |= OMAP_DMIC_UP3_ENABLE;
+	case 4:
+		dmic->ch_enabled |= OMAP_DMIC_UP2_ENABLE;
+	case 2:
+		dmic->ch_enabled |= OMAP_DMIC_UP1_ENABLE;
+		break;
+	default:
+		dev_err(dmic->dev, "invalid number of legacy channels\n");
+		return -EINVAL;
+	}
+
+	/* packet size is threshold * channels */
+	dma_data = snd_soc_dai_get_dma_data(dai, substream);
+	dma_data->maxburst = dmic->threshold * channels;
+
+	return 0;
+}
+
+static int omap_dmic_dai_prepare(struct snd_pcm_substream *substream,
+				  struct snd_soc_dai *dai)
+{
+	struct omap_dmic *dmic = snd_soc_dai_get_drvdata(dai);
+	u32 ctrl;
+
+	/* Configure uplink threshold */
+	omap_dmic_write(dmic, OMAP_DMIC_FIFO_CTRL_REG, dmic->threshold);
+
+	ctrl = omap_dmic_read(dmic, OMAP_DMIC_CTRL_REG);
+
+	/* Set dmic out format */
+	ctrl &= ~(OMAP_DMIC_FORMAT | OMAP_DMIC_POLAR_MASK);
+	ctrl |= (OMAP_DMICOUTFORMAT_LJUST | OMAP_DMIC_POLAR1 |
+		 OMAP_DMIC_POLAR2 | OMAP_DMIC_POLAR3);
+
+	/* Configure dmic clock divider */
+	ctrl &= ~OMAP_DMIC_CLK_DIV_MASK;
+	ctrl |= OMAP_DMIC_CLK_DIV(dmic->clk_div);
+
+	omap_dmic_write(dmic, OMAP_DMIC_CTRL_REG, ctrl);
+
+	omap_dmic_write(dmic, OMAP_DMIC_CTRL_REG,
+			ctrl | OMAP_DMICOUTFORMAT_LJUST | OMAP_DMIC_POLAR1 |
+			OMAP_DMIC_POLAR2 | OMAP_DMIC_POLAR3);
+
+	return 0;
+}
+
+static int omap_dmic_dai_trigger(struct snd_pcm_substream *substream,
+				  int cmd, struct snd_soc_dai *dai)
+{
+	struct omap_dmic *dmic = snd_soc_dai_get_drvdata(dai);
+
+	switch (cmd) {
+	case SNDRV_PCM_TRIGGER_START:
+		omap_dmic_start(dmic);
+		break;
+	case SNDRV_PCM_TRIGGER_STOP:
+		omap_dmic_stop(dmic);
+		break;
+	default:
+		break;
+	}
+
+	return 0;
+}
+
+static int omap_dmic_select_fclk(struct omap_dmic *dmic, int clk_id,
+				 unsigned int freq)
+{
+	struct clk *parent_clk;
+	char *parent_clk_name;
+	int ret = 0;
+
+	switch (freq) {
+	case 12000000:
+	case 19200000:
+	case 24000000:
+	case 24576000:
+		break;
+	default:
+		dev_err(dmic->dev, "invalid input frequency: %dHz\n", freq);
+		dmic->fclk_freq = 0;
+		return -EINVAL;
+	}
+
+	if (dmic->sysclk == clk_id) {
+		dmic->fclk_freq = freq;
+		return 0;
+	}
+
+	/* re-parent not allowed if a stream is ongoing */
+	if (dmic->active && dmic_is_enabled(dmic)) {
+		dev_err(dmic->dev, "can't re-parent when DMIC active\n");
+		return -EBUSY;
+	}
+
+	switch (clk_id) {
+	case OMAP_DMIC_SYSCLK_PAD_CLKS:
+		parent_clk_name = "pad_clks_ck";
+		break;
+	case OMAP_DMIC_SYSCLK_SLIMBLUS_CLKS:
+		parent_clk_name = "slimbus_clk";
+		break;
+	case OMAP_DMIC_SYSCLK_SYNC_MUX_CLKS:
+		parent_clk_name = "dmic_sync_mux_ck";
+		break;
+	default:
+		dev_err(dmic->dev, "fclk clk_id (%d) not supported\n", clk_id);
+		return -EINVAL;
+	}
+
+	parent_clk = clk_get(dmic->dev, parent_clk_name);
+	if (IS_ERR(parent_clk)) {
+		dev_err(dmic->dev, "can't get %s\n", parent_clk_name);
+		return -ENODEV;
+	}
+
+	mutex_lock(&dmic->mutex);
+	if (dmic->active) {
+		/* disable clock while reparenting */
+		pm_runtime_put_sync(dmic->dev);
+		ret = clk_set_parent(dmic->fclk, parent_clk);
+		pm_runtime_get_sync(dmic->dev);
+	} else {
+		ret = clk_set_parent(dmic->fclk, parent_clk);
+	}
+	mutex_unlock(&dmic->mutex);
+
+	if (ret < 0) {
+		dev_err(dmic->dev, "re-parent failed\n");
+		goto err_busy;
+	}
+
+	dmic->sysclk = clk_id;
+	dmic->fclk_freq = freq;
+
+err_busy:
+	clk_put(parent_clk);
+
+	return ret;
+}
+
+static int omap_dmic_select_outclk(struct omap_dmic *dmic, int clk_id,
+				    unsigned int freq)
+{
+	int ret = 0;
+
+	if (clk_id != OMAP_DMIC_ABE_DMIC_CLK) {
+		dev_err(dmic->dev, "output clk_id (%d) not supported\n",
+			clk_id);
+		return -EINVAL;
+	}
+
+	switch (freq) {
+	case 1536000:
+	case 2400000:
+	case 3072000:
+	case 3840000:
+		dmic->out_freq = freq;
+		break;
+	default:
+		dev_err(dmic->dev, "invalid out frequency: %dHz\n", freq);
+		dmic->out_freq = 0;
+		ret = -EINVAL;
+	}
+
+	return ret;
+}
+
+static int omap_dmic_set_dai_sysclk(struct snd_soc_dai *dai, int clk_id,
+				    unsigned int freq, int dir)
+{
+	struct omap_dmic *dmic = snd_soc_dai_get_drvdata(dai);
+
+	if (dir == SND_SOC_CLOCK_IN)
+		return omap_dmic_select_fclk(dmic, clk_id, freq);
+	else if (dir == SND_SOC_CLOCK_OUT)
+		return omap_dmic_select_outclk(dmic, clk_id, freq);
+
+	dev_err(dmic->dev, "invalid clock direction (%d)\n", dir);
+	return -EINVAL;
+}
+
+static const struct snd_soc_dai_ops omap_dmic_dai_ops = {
+	.startup	= omap_dmic_dai_startup,
+	.shutdown	= omap_dmic_dai_shutdown,
+	.hw_params	= omap_dmic_dai_hw_params,
+	.prepare	= omap_dmic_dai_prepare,
+	.trigger	= omap_dmic_dai_trigger,
+	.set_sysclk	= omap_dmic_set_dai_sysclk,
+};
+
+static int omap_dmic_probe(struct snd_soc_dai *dai)
+{
+	struct omap_dmic *dmic = snd_soc_dai_get_drvdata(dai);
+
+	pm_runtime_enable(dmic->dev);
+
+	/* Disable lines while request is ongoing */
+	pm_runtime_get_sync(dmic->dev);
+	omap_dmic_write(dmic, OMAP_DMIC_CTRL_REG, 0x00);
+	pm_runtime_put_sync(dmic->dev);
+
+	/* Configure DMIC threshold value */
+	dmic->threshold = OMAP_DMIC_THRES_MAX - 3;
+
+	snd_soc_dai_init_dma_data(dai, NULL, &dmic->dma_data);
+
+	return 0;
+}
+
+static int omap_dmic_remove(struct snd_soc_dai *dai)
+{
+	struct omap_dmic *dmic = snd_soc_dai_get_drvdata(dai);
+
+	pm_runtime_disable(dmic->dev);
+
+	return 0;
+}
+
+static struct snd_soc_dai_driver omap_dmic_dai = {
+	.name = "omap-dmic",
+	.probe = omap_dmic_probe,
+	.remove = omap_dmic_remove,
+	.capture = {
+		.channels_min = 2,
+		.channels_max = 6,
+		.rates = SNDRV_PCM_RATE_96000 | SNDRV_PCM_RATE_192000,
+		.formats = SNDRV_PCM_FMTBIT_S32_LE,
+		.sig_bits = 24,
+	},
+	.ops = &omap_dmic_dai_ops,
+};
+
+static const struct snd_soc_component_driver omap_dmic_component = {
+	.name		= "omap-dmic",
+};
+
+static int asoc_dmic_probe(struct platform_device *pdev)
+{
+	struct omap_dmic *dmic;
+	struct resource *res;
+	int ret;
+
+	dmic = devm_kzalloc(&pdev->dev, sizeof(struct omap_dmic), GFP_KERNEL);
+	if (!dmic)
+		return -ENOMEM;
+
+	platform_set_drvdata(pdev, dmic);
+	dmic->dev = &pdev->dev;
+	dmic->sysclk = OMAP_DMIC_SYSCLK_SYNC_MUX_CLKS;
+
+	mutex_init(&dmic->mutex);
+
+	dmic->fclk = devm_clk_get(dmic->dev, "fck");
+	if (IS_ERR(dmic->fclk)) {
+		dev_err(dmic->dev, "cant get fck\n");
+		return -ENODEV;
+	}
+
+	res = platform_get_resource_byname(pdev, IORESOURCE_MEM, "dma");
+	if (!res) {
+		dev_err(dmic->dev, "invalid dma memory resource\n");
+		return -ENODEV;
+	}
+	dmic->dma_data.addr = res->start + OMAP_DMIC_DATA_REG;
+
+	dmic->dma_data.filter_data = "up_link";
+
+	res = platform_get_resource_byname(pdev, IORESOURCE_MEM, "mpu");
+	dmic->io_base = devm_ioremap_resource(&pdev->dev, res);
+	if (IS_ERR(dmic->io_base))
+		return PTR_ERR(dmic->io_base);
+
+
+	ret = devm_snd_soc_register_component(&pdev->dev,
+					      &omap_dmic_component,
+					      &omap_dmic_dai, 1);
+	if (ret)
+		return ret;
+
+	ret = omap_pcm_platform_register(&pdev->dev);
+	if (ret)
+		return ret;
+
+	return 0;
+}
+
+static const struct of_device_id omap_dmic_of_match[] = {
+	{ .compatible = "ti,omap4-dmic", },
+	{ }
+};
+MODULE_DEVICE_TABLE(of, omap_dmic_of_match);
+
+static struct platform_driver asoc_dmic_driver = {
+	.driver = {
+		.name = "omap-dmic",
+		.owner = THIS_MODULE,
+		.of_match_table = omap_dmic_of_match,
+	},
+	.probe = asoc_dmic_probe,
+};
+
+module_platform_driver(asoc_dmic_driver);
+
+MODULE_ALIAS("platform:omap-dmic");
+MODULE_AUTHOR("Peter Ujfalusi <peter.ujfalusi@ti.com>");
+MODULE_DESCRIPTION("OMAP DMIC ASoC Interface");
+MODULE_LICENSE("GPL");

sound/soc/omap/omap-dmic.h

@@ -0,0 +1,69 @@
+/*
+ * omap-dmic.h  --  OMAP Digital Microphone Controller
+ *
+ * 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.
+ */
+
+#ifndef _OMAP_DMIC_H
+#define _OMAP_DMIC_H
+
+#define OMAP_DMIC_REVISION_REG		0x00
+#define OMAP_DMIC_SYSCONFIG_REG		0x10
+#define OMAP_DMIC_IRQSTATUS_RAW_REG	0x24
+#define OMAP_DMIC_IRQSTATUS_REG		0x28
+#define OMAP_DMIC_IRQENABLE_SET_REG	0x2C
+#define OMAP_DMIC_IRQENABLE_CLR_REG	0x30
+#define OMAP_DMIC_IRQWAKE_EN_REG	0x34
+#define OMAP_DMIC_DMAENABLE_SET_REG	0x38
+#define OMAP_DMIC_DMAENABLE_CLR_REG	0x3C
+#define OMAP_DMIC_DMAWAKEEN_REG		0x40
+#define OMAP_DMIC_CTRL_REG		0x44
+#define OMAP_DMIC_DATA_REG		0x48
+#define OMAP_DMIC_FIFO_CTRL_REG		0x4C
+#define OMAP_DMIC_FIFO_DMIC1R_DATA_REG	0x50
+#define OMAP_DMIC_FIFO_DMIC1L_DATA_REG	0x54
+#define OMAP_DMIC_FIFO_DMIC2R_DATA_REG	0x58
+#define OMAP_DMIC_FIFO_DMIC2L_DATA_REG	0x5C
+#define OMAP_DMIC_FIFO_DMIC3R_DATA_REG	0x60
+#define OMAP_DMIC_FIFO_DMIC3L_DATA_REG	0x64
+
+/* IRQSTATUS_RAW, IRQSTATUS, IRQENABLE_SET, IRQENABLE_CLR bit fields */
+#define OMAP_DMIC_IRQ			(1 << 0)
+#define OMAP_DMIC_IRQ_FULL		(1 << 1)
+#define OMAP_DMIC_IRQ_ALMST_EMPTY	(1 << 2)
+#define OMAP_DMIC_IRQ_EMPTY		(1 << 3)
+#define OMAP_DMIC_IRQ_MASK		0x07
+
+/* DMIC_DMAENABLE bit fields */
+#define OMAP_DMIC_DMA_ENABLE		0x1
+
+/* DMIC_CTRL bit fields */
+#define OMAP_DMIC_UP1_ENABLE		(1 << 0)
+#define OMAP_DMIC_UP2_ENABLE		(1 << 1)
+#define OMAP_DMIC_UP3_ENABLE		(1 << 2)
+#define OMAP_DMIC_UP_ENABLE_MASK	0x7
+#define OMAP_DMIC_FORMAT		(1 << 3)
+#define OMAP_DMIC_POLAR1		(1 << 4)
+#define OMAP_DMIC_POLAR2		(1 << 5)
+#define OMAP_DMIC_POLAR3		(1 << 6)
+#define OMAP_DMIC_POLAR_MASK		(0x7 << 4)
+#define OMAP_DMIC_CLK_DIV(x)		(((x) & 0x7) << 7)
+#define OMAP_DMIC_CLK_DIV_MASK		(0x7 << 7)
+#define	OMAP_DMIC_RESET			(1 << 10)
+
+#define OMAP_DMICOUTFORMAT_LJUST	(0 << 3)
+#define OMAP_DMICOUTFORMAT_RJUST	(1 << 3)
+
+/* DMIC_FIFO_CTRL bit fields */
+#define OMAP_DMIC_THRES_MAX		0xF
+
+enum omap_dmic_clk {
+	OMAP_DMIC_SYSCLK_PAD_CLKS,		/* PAD_CLKS */
+	OMAP_DMIC_SYSCLK_SLIMBLUS_CLKS,		/* SLIMBUS_CLK */
+	OMAP_DMIC_SYSCLK_SYNC_MUX_CLKS,		/* DMIC_SYNC_MUX_CLK */
+	OMAP_DMIC_ABE_DMIC_CLK,			/* abe_dmic_clk */
+};
+
+#endif

sound/soc/omap/omap-hdmi-card.c

@@ -0,0 +1,87 @@
+/*
+ * omap-hdmi-card.c
+ *
+ * OMAP ALSA SoC machine driver for TI OMAP HDMI
+ * Copyright (C) 2011 Texas Instruments Incorporated - http://www.ti.com/
+ * Author: Ricardo Neri <ricardo.neri@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/module.h>
+#include <sound/pcm.h>
+#include <sound/soc.h>
+#include <asm/mach-types.h>
+#include <video/omapdss.h>
+
+#define DRV_NAME "omap-hdmi-audio"
+
+static struct snd_soc_dai_link omap_hdmi_dai = {
+	.name = "HDMI",
+	.stream_name = "HDMI",
+	.cpu_dai_name = "omap-hdmi-audio-dai",
+	.platform_name = "omap-hdmi-audio-dai",
+	.codec_name = "hdmi-audio-codec",
+	.codec_dai_name = "hdmi-hifi",
+};
+
+static struct snd_soc_card snd_soc_omap_hdmi = {
+	.name = "OMAPHDMI",
+	.owner = THIS_MODULE,
+	.dai_link = &omap_hdmi_dai,
+	.num_links = 1,
+};
+
+static int omap_hdmi_probe(struct platform_device *pdev)
+{
+	struct snd_soc_card *card = &snd_soc_omap_hdmi;
+	int ret;
+
+	card->dev = &pdev->dev;
+
+	ret = snd_soc_register_card(card);
+	if (ret) {
+		dev_err(&pdev->dev, "snd_soc_register_card failed (%d)\n", ret);
+		card->dev = NULL;
+		return ret;
+	}
+	return 0;
+}
+
+static int omap_hdmi_remove(struct platform_device *pdev)
+{
+	struct snd_soc_card *card = platform_get_drvdata(pdev);
+
+	snd_soc_unregister_card(card);
+	card->dev = NULL;
+	return 0;
+}
+
+static struct platform_driver omap_hdmi_driver = {
+	.driver = {
+		.name = DRV_NAME,
+		.owner = THIS_MODULE,
+	},
+	.probe = omap_hdmi_probe,
+	.remove = omap_hdmi_remove,
+};
+
+module_platform_driver(omap_hdmi_driver);
+
+MODULE_AUTHOR("Ricardo Neri <ricardo.neri@ti.com>");
+MODULE_DESCRIPTION("OMAP HDMI machine ASoC driver");
+MODULE_LICENSE("GPL");
+MODULE_ALIAS("platform:" DRV_NAME);

sound/soc/omap/omap-hdmi.c

@@ -0,0 +1,364 @@
+/*
+ * omap-hdmi.c
+ *
+ * OMAP ALSA SoC DAI driver for HDMI audio on OMAP4 processors.
+ * Copyright (C) 2010-2011 Texas Instruments Incorporated - http://www.ti.com/
+ * Authors: Jorge Candelaria <jorge.candelaria@ti.com>
+ *          Ricardo Neri <ricardo.neri@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/init.h>
+#include <linux/module.h>
+#include <linux/device.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/initval.h>
+#include <sound/soc.h>
+#include <sound/asound.h>
+#include <sound/asoundef.h>
+#include <sound/dmaengine_pcm.h>
+#include <video/omapdss.h>
+#include <sound/omap-pcm.h>
+
+#include "omap-hdmi.h"
+
+#define DRV_NAME "omap-hdmi-audio-dai"
+
+struct hdmi_priv {
+	struct snd_dmaengine_dai_dma_data dma_data;
+	unsigned int dma_req;
+	struct omap_dss_audio dss_audio;
+	struct snd_aes_iec958 iec;
+	struct snd_cea_861_aud_if cea;
+	struct omap_dss_device *dssdev;
+};
+
+static int omap_hdmi_dai_startup(struct snd_pcm_substream *substream,
+				  struct snd_soc_dai *dai)
+{
+	struct hdmi_priv *priv = snd_soc_dai_get_drvdata(dai);
+	int err;
+	/*
+	 * Make sure that the period bytes are multiple of the DMA packet size.
+	 * Largest packet size we use is 32 32-bit words = 128 bytes
+	 */
+	err = snd_pcm_hw_constraint_step(substream->runtime, 0,
+				 SNDRV_PCM_HW_PARAM_PERIOD_BYTES, 128);
+	if (err < 0) {
+		dev_err(dai->dev, "could not apply constraint\n");
+		return err;
+	}
+
+	if (!priv->dssdev->driver->audio_supported(priv->dssdev)) {
+		dev_err(dai->dev, "audio not supported\n");
+		return -ENODEV;
+	}
+
+	snd_soc_dai_set_dma_data(dai, substream, &priv->dma_data);
+
+	return 0;
+}
+
+static int omap_hdmi_dai_prepare(struct snd_pcm_substream *substream,
+				struct snd_soc_dai *dai)
+{
+	struct hdmi_priv *priv = snd_soc_dai_get_drvdata(dai);
+
+	return priv->dssdev->driver->audio_enable(priv->dssdev);
+}
+
+static int omap_hdmi_dai_hw_params(struct snd_pcm_substream *substream,
+				    struct snd_pcm_hw_params *params,
+				    struct snd_soc_dai *dai)
+{
+	struct hdmi_priv *priv = snd_soc_dai_get_drvdata(dai);
+	struct snd_aes_iec958 *iec = &priv->iec;
+	struct snd_cea_861_aud_if *cea = &priv->cea;
+	int err = 0;
+
+	switch (params_format(params)) {
+	case SNDRV_PCM_FORMAT_S16_LE:
+		priv->dma_data.maxburst = 16;
+		break;
+	case SNDRV_PCM_FORMAT_S24_LE:
+		priv->dma_data.maxburst = 32;
+		break;
+	default:
+		dev_err(dai->dev, "format not supported!\n");
+		return -EINVAL;
+	}
+
+	/*
+	 * fill the IEC-60958 channel status word
+	 */
+	/* initialize the word bytes */
+	memset(iec->status, 0, sizeof(iec->status));
+
+	/* specify IEC-60958-3 (commercial use) */
+	iec->status[0] &= ~IEC958_AES0_PROFESSIONAL;
+
+	/* specify that the audio is LPCM*/
+	iec->status[0] &= ~IEC958_AES0_NONAUDIO;
+
+	iec->status[0] |= IEC958_AES0_CON_NOT_COPYRIGHT;
+
+	iec->status[0] |= IEC958_AES0_CON_EMPHASIS_NONE;
+
+	iec->status[0] |= IEC958_AES1_PRO_MODE_NOTID;
+
+	iec->status[1] = IEC958_AES1_CON_GENERAL;
+
+	iec->status[2] |= IEC958_AES2_CON_SOURCE_UNSPEC;
+
+	iec->status[2] |= IEC958_AES2_CON_CHANNEL_UNSPEC;
+
+	switch (params_rate(params)) {
+	case 32000:
+		iec->status[3] |= IEC958_AES3_CON_FS_32000;
+		break;
+	case 44100:
+		iec->status[3] |= IEC958_AES3_CON_FS_44100;
+		break;
+	case 48000:
+		iec->status[3] |= IEC958_AES3_CON_FS_48000;
+		break;
+	case 88200:
+		iec->status[3] |= IEC958_AES3_CON_FS_88200;
+		break;
+	case 96000:
+		iec->status[3] |= IEC958_AES3_CON_FS_96000;
+		break;
+	case 176400:
+		iec->status[3] |= IEC958_AES3_CON_FS_176400;
+		break;
+	case 192000:
+		iec->status[3] |= IEC958_AES3_CON_FS_192000;
+		break;
+	default:
+		dev_err(dai->dev, "rate not supported!\n");
+		return -EINVAL;
+	}
+
+	/* specify the clock accuracy */
+	iec->status[3] |= IEC958_AES3_CON_CLOCK_1000PPM;
+
+	/*
+	 * specify the word length. The same word length value can mean
+	 * two different lengths. Hence, we need to specify the maximum
+	 * word length as well.
+	 */
+	switch (params_format(params)) {
+	case SNDRV_PCM_FORMAT_S16_LE:
+		iec->status[4] |= IEC958_AES4_CON_WORDLEN_20_16;
+		iec->status[4] &= ~IEC958_AES4_CON_MAX_WORDLEN_24;
+		break;
+	case SNDRV_PCM_FORMAT_S24_LE:
+		iec->status[4] |= IEC958_AES4_CON_WORDLEN_24_20;
+		iec->status[4] |= IEC958_AES4_CON_MAX_WORDLEN_24;
+		break;
+	default:
+		dev_err(dai->dev, "format not supported!\n");
+		return -EINVAL;
+	}
+
+	/*
+	 * Fill the CEA-861 audio infoframe (see spec for details)
+	 */
+
+	cea->db1_ct_cc = (params_channels(params) - 1)
+		& CEA861_AUDIO_INFOFRAME_DB1CC;
+	cea->db1_ct_cc |= CEA861_AUDIO_INFOFRAME_DB1CT_FROM_STREAM;
+
+	cea->db2_sf_ss = CEA861_AUDIO_INFOFRAME_DB2SF_FROM_STREAM;
+	cea->db2_sf_ss |= CEA861_AUDIO_INFOFRAME_DB2SS_FROM_STREAM;
+
+	cea->db3 = 0; /* not used, all zeros */
+
+	/*
+	 * The OMAP HDMI IP requires to use the 8-channel channel code when
+	 * transmitting more than two channels.
+	 */
+	if (params_channels(params) == 2)
+		cea->db4_ca = 0x0;
+	else
+		cea->db4_ca = 0x13;
+
+	cea->db5_dminh_lsv = CEA861_AUDIO_INFOFRAME_DB5_DM_INH_PROHIBITED;
+	/* the expression is trivial but makes clear what we are doing */
+	cea->db5_dminh_lsv |= (0 & CEA861_AUDIO_INFOFRAME_DB5_LSV);
+
+	priv->dss_audio.iec = iec;
+	priv->dss_audio.cea = cea;
+
+	err = priv->dssdev->driver->audio_config(priv->dssdev,
+						 &priv->dss_audio);
+
+	return err;
+}
+
+static int omap_hdmi_dai_trigger(struct snd_pcm_substream *substream, int cmd,
+				struct snd_soc_dai *dai)
+{
+	struct hdmi_priv *priv = snd_soc_dai_get_drvdata(dai);
+	int err = 0;
+
+	switch (cmd) {
+	case SNDRV_PCM_TRIGGER_START:
+	case SNDRV_PCM_TRIGGER_RESUME:
+	case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
+		err = priv->dssdev->driver->audio_start(priv->dssdev);
+		break;
+	case SNDRV_PCM_TRIGGER_STOP:
+	case SNDRV_PCM_TRIGGER_SUSPEND:
+	case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
+		priv->dssdev->driver->audio_stop(priv->dssdev);
+		break;
+	default:
+		err = -EINVAL;
+	}
+	return err;
+}
+
+static void omap_hdmi_dai_shutdown(struct snd_pcm_substream *substream,
+				struct snd_soc_dai *dai)
+{
+	struct hdmi_priv *priv = snd_soc_dai_get_drvdata(dai);
+
+	priv->dssdev->driver->audio_disable(priv->dssdev);
+}
+
+static const struct snd_soc_dai_ops omap_hdmi_dai_ops = {
+	.startup	= omap_hdmi_dai_startup,
+	.hw_params	= omap_hdmi_dai_hw_params,
+	.prepare	= omap_hdmi_dai_prepare,
+	.trigger	= omap_hdmi_dai_trigger,
+	.shutdown	= omap_hdmi_dai_shutdown,
+};
+
+static struct snd_soc_dai_driver omap_hdmi_dai = {
+	.playback = {
+		.channels_min = 2,
+		.channels_max = 8,
+		.rates = OMAP_HDMI_RATES,
+		.formats = OMAP_HDMI_FORMATS,
+	},
+	.ops = &omap_hdmi_dai_ops,
+};
+
+static const struct snd_soc_component_driver omap_hdmi_component = {
+	.name		= DRV_NAME,
+};
+
+static int omap_hdmi_probe(struct platform_device *pdev)
+{
+	int ret;
+	struct resource *hdmi_rsrc;
+	struct hdmi_priv *hdmi_data;
+	bool hdmi_dev_found = false;
+
+	hdmi_data = devm_kzalloc(&pdev->dev, sizeof(*hdmi_data), GFP_KERNEL);
+	if (hdmi_data == NULL) {
+		dev_err(&pdev->dev, "Cannot allocate memory for HDMI data\n");
+		return -ENOMEM;
+	}
+
+	hdmi_rsrc = platform_get_resource(pdev, IORESOURCE_MEM, 0);
+	if (!hdmi_rsrc) {
+		dev_err(&pdev->dev, "Cannot obtain IORESOURCE_MEM HDMI\n");
+		return -ENODEV;
+	}
+
+	hdmi_data->dma_data.addr = hdmi_rsrc->start + OMAP_HDMI_AUDIO_DMA_PORT;
+
+	hdmi_rsrc = platform_get_resource(pdev, IORESOURCE_DMA, 0);
+	if (!hdmi_rsrc) {
+		dev_err(&pdev->dev, "Cannot obtain IORESOURCE_DMA HDMI\n");
+		return -ENODEV;
+	}
+
+	hdmi_data->dma_req = hdmi_rsrc->start;
+	hdmi_data->dma_data.filter_data = &hdmi_data->dma_req;
+	hdmi_data->dma_data.addr_width = DMA_SLAVE_BUSWIDTH_4_BYTES;
+
+	/*
+	 * TODO: We assume that there is only one DSS HDMI device. Future
+	 * OMAP implementations may support more than one HDMI devices and
+	 * we should provided separate audio support for all of them.
+	 */
+	/* Find an HDMI device. */
+	for_each_dss_dev(hdmi_data->dssdev) {
+		omap_dss_get_device(hdmi_data->dssdev);
+
+		if (!hdmi_data->dssdev->driver) {
+			omap_dss_put_device(hdmi_data->dssdev);
+			continue;
+		}
+
+		if (hdmi_data->dssdev->type == OMAP_DISPLAY_TYPE_HDMI) {
+			hdmi_dev_found = true;
+			break;
+		}
+	}
+
+	if (!hdmi_dev_found) {
+		dev_err(&pdev->dev, "no driver for HDMI display found\n");
+		return -ENODEV;
+	}
+
+	dev_set_drvdata(&pdev->dev, hdmi_data);
+	ret = snd_soc_register_component(&pdev->dev, &omap_hdmi_component,
+					 &omap_hdmi_dai, 1);
+
+	if (ret)
+		return ret;
+
+	return omap_pcm_platform_register(&pdev->dev);
+}
+
+static int omap_hdmi_remove(struct platform_device *pdev)
+{
+	struct hdmi_priv *hdmi_data = dev_get_drvdata(&pdev->dev);
+
+	snd_soc_unregister_component(&pdev->dev);
+
+	if (hdmi_data == NULL) {
+		dev_err(&pdev->dev, "cannot obtain HDMi data\n");
+		return -ENODEV;
+	}
+
+	omap_dss_put_device(hdmi_data->dssdev);
+	return 0;
+}
+
+static struct platform_driver hdmi_dai_driver = {
+	.driver = {
+		.name = DRV_NAME,
+		.owner = THIS_MODULE,
+	},
+	.probe = omap_hdmi_probe,
+	.remove = omap_hdmi_remove,
+};
+
+module_platform_driver(hdmi_dai_driver);
+
+MODULE_AUTHOR("Jorge Candelaria <jorge.candelaria@ti.com>");
+MODULE_AUTHOR("Ricardo Neri <ricardo.neri@ti.com>");
+MODULE_DESCRIPTION("OMAP HDMI SoC Interface");
+MODULE_LICENSE("GPL");
+MODULE_ALIAS("platform:" DRV_NAME);

sound/soc/omap/omap-hdmi.h

@@ -0,0 +1,38 @@
+/*
+ * omap-hdmi.h
+ *
+ * Definitions for OMAP ALSA SoC DAI driver for HDMI audio on OMAP4 processors.
+ * Copyright (C) 2010-2011 Texas Instruments Incorporated - http://www.ti.com/
+ * Authors: Jorge Candelaria <jorge.candelaria@ti.com>
+ *          Ricardo Neri <ricardo.neri@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
+ *
+ */
+
+#ifndef __OMAP_HDMI_H__
+#define __OMAP_HDMI_H__
+
+#define OMAP_HDMI_AUDIO_DMA_PORT 0x8c
+
+#define OMAP_HDMI_RATES	(SNDRV_PCM_RATE_32000 | \
+				SNDRV_PCM_RATE_44100 | SNDRV_PCM_RATE_48000 | \
+				SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000 | \
+				SNDRV_PCM_RATE_176400 | SNDRV_PCM_RATE_192000)
+
+#define OMAP_HDMI_FORMATS (SNDRV_PCM_FMTBIT_S16_LE | \
+				SNDRV_PCM_FMTBIT_S24_LE)
+
+#endif

sound/soc/omap/omap-mcbsp.c

@@ -0,0 +1,847 @@
+/*
+ * omap-mcbsp.c  --  OMAP ALSA SoC DAI driver using McBSP port
+ *
+ * Copyright (C) 2008 Nokia Corporation
+ *
+ * Contact: Jarkko Nikula <jarkko.nikula@bitmer.com>
+ *          Peter Ujfalusi <peter.ujfalusi@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/init.h>
+#include <linux/module.h>
+#include <linux/device.h>
+#include <linux/pm_runtime.h>
+#include <linux/of.h>
+#include <linux/of_device.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/initval.h>
+#include <sound/soc.h>
+#include <sound/dmaengine_pcm.h>
+#include <sound/omap-pcm.h>
+
+#include <linux/platform_data/asoc-ti-mcbsp.h>
+#include "mcbsp.h"
+#include "omap-mcbsp.h"
+
+#define OMAP_MCBSP_RATES	(SNDRV_PCM_RATE_8000_96000)
+
+#define OMAP_MCBSP_SOC_SINGLE_S16_EXT(xname, xmin, xmax, \
+	xhandler_get, xhandler_put) \
+{	.iface = SNDRV_CTL_ELEM_IFACE_MIXER, .name = xname, \
+	.info = omap_mcbsp_st_info_volsw, \
+	.get = xhandler_get, .put = xhandler_put, \
+	.private_value = (unsigned long) &(struct soc_mixer_control) \
+	{.min = xmin, .max = xmax} }
+
+enum {
+	OMAP_MCBSP_WORD_8 = 0,
+	OMAP_MCBSP_WORD_12,
+	OMAP_MCBSP_WORD_16,
+	OMAP_MCBSP_WORD_20,
+	OMAP_MCBSP_WORD_24,
+	OMAP_MCBSP_WORD_32,
+};
+
+/*
+ * Stream DMA parameters. DMA request line and port address are set runtime
+ * since they are different between OMAP1 and later OMAPs
+ */
+static void omap_mcbsp_set_threshold(struct snd_pcm_substream *substream,
+		unsigned int packet_size)
+{
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
+	struct omap_mcbsp *mcbsp = snd_soc_dai_get_drvdata(cpu_dai);
+	int words;
+
+	/*
+	 * Configure McBSP threshold based on either:
+	 * packet_size, when the sDMA is in packet mode, or based on the
+	 * period size in THRESHOLD mode, otherwise use McBSP threshold = 1
+	 * for mono streams.
+	 */
+	if (packet_size)
+		words = packet_size;
+	else
+		words = 1;
+
+	/* Configure McBSP internal buffer usage */
+	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+		omap_mcbsp_set_tx_threshold(mcbsp, words);
+	else
+		omap_mcbsp_set_rx_threshold(mcbsp, words);
+}
+
+static int omap_mcbsp_hwrule_min_buffersize(struct snd_pcm_hw_params *params,
+				    struct snd_pcm_hw_rule *rule)
+{
+	struct snd_interval *buffer_size = hw_param_interval(params,
+					SNDRV_PCM_HW_PARAM_BUFFER_SIZE);
+	struct snd_interval *channels = hw_param_interval(params,
+					SNDRV_PCM_HW_PARAM_CHANNELS);
+	struct omap_mcbsp *mcbsp = rule->private;
+	struct snd_interval frames;
+	int size;
+
+	snd_interval_any(&frames);
+	size = mcbsp->pdata->buffer_size;
+
+	frames.min = size / channels->min;
+	frames.integer = 1;
+	return snd_interval_refine(buffer_size, &frames);
+}
+
+static int omap_mcbsp_dai_startup(struct snd_pcm_substream *substream,
+				  struct snd_soc_dai *cpu_dai)
+{
+	struct omap_mcbsp *mcbsp = snd_soc_dai_get_drvdata(cpu_dai);
+	int err = 0;
+
+	if (!cpu_dai->active)
+		err = omap_mcbsp_request(mcbsp);
+
+	/*
+	 * OMAP3 McBSP FIFO is word structured.
+	 * McBSP2 has 1024 + 256 = 1280 word long buffer,
+	 * McBSP1,3,4,5 has 128 word long buffer
+	 * This means that the size of the FIFO depends on the sample format.
+	 * For example on McBSP3:
+	 * 16bit samples: size is 128 * 2 = 256 bytes
+	 * 32bit samples: size is 128 * 4 = 512 bytes
+	 * It is simpler to place constraint for buffer and period based on
+	 * channels.
+	 * McBSP3 as example again (16 or 32 bit samples):
+	 * 1 channel (mono): size is 128 frames (128 words)
+	 * 2 channels (stereo): size is 128 / 2 = 64 frames (2 * 64 words)
+	 * 4 channels: size is 128 / 4 = 32 frames (4 * 32 words)
+	 */
+	if (mcbsp->pdata->buffer_size) {
+		/*
+		* Rule for the buffer size. We should not allow
+		* smaller buffer than the FIFO size to avoid underruns.
+		* This applies only for the playback stream.
+		*/
+		if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+			snd_pcm_hw_rule_add(substream->runtime, 0,
+					    SNDRV_PCM_HW_PARAM_BUFFER_SIZE,
+					    omap_mcbsp_hwrule_min_buffersize,
+					    mcbsp,
+					    SNDRV_PCM_HW_PARAM_CHANNELS, -1);
+
+		/* Make sure, that the period size is always even */
+		snd_pcm_hw_constraint_step(substream->runtime, 0,
+					   SNDRV_PCM_HW_PARAM_PERIOD_SIZE, 2);
+	}
+
+	return err;
+}
+
+static void omap_mcbsp_dai_shutdown(struct snd_pcm_substream *substream,
+				    struct snd_soc_dai *cpu_dai)
+{
+	struct omap_mcbsp *mcbsp = snd_soc_dai_get_drvdata(cpu_dai);
+
+	if (!cpu_dai->active) {
+		omap_mcbsp_free(mcbsp);
+		mcbsp->configured = 0;
+	}
+}
+
+static int omap_mcbsp_dai_trigger(struct snd_pcm_substream *substream, int cmd,
+				  struct snd_soc_dai *cpu_dai)
+{
+	struct omap_mcbsp *mcbsp = snd_soc_dai_get_drvdata(cpu_dai);
+	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:
+		mcbsp->active++;
+		omap_mcbsp_start(mcbsp, play, !play);
+		break;
+
+	case SNDRV_PCM_TRIGGER_STOP:
+	case SNDRV_PCM_TRIGGER_SUSPEND:
+	case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
+		omap_mcbsp_stop(mcbsp, play, !play);
+		mcbsp->active--;
+		break;
+	default:
+		err = -EINVAL;
+	}
+
+	return err;
+}
+
+static snd_pcm_sframes_t omap_mcbsp_dai_delay(
+			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->cpu_dai;
+	struct omap_mcbsp *mcbsp = snd_soc_dai_get_drvdata(cpu_dai);
+	u16 fifo_use;
+	snd_pcm_sframes_t delay;
+
+	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+		fifo_use = omap_mcbsp_get_tx_delay(mcbsp);
+	else
+		fifo_use = omap_mcbsp_get_rx_delay(mcbsp);
+
+	/*
+	 * Divide the used locations with the channel count to get the
+	 * FIFO usage in samples (don't care about partial samples in the
+	 * buffer).
+	 */
+	delay = fifo_use / substream->runtime->channels;
+
+	return delay;
+}
+
+static int omap_mcbsp_dai_hw_params(struct snd_pcm_substream *substream,
+				    struct snd_pcm_hw_params *params,
+				    struct snd_soc_dai *cpu_dai)
+{
+	struct omap_mcbsp *mcbsp = snd_soc_dai_get_drvdata(cpu_dai);
+	struct omap_mcbsp_reg_cfg *regs = &mcbsp->cfg_regs;
+	struct snd_dmaengine_dai_dma_data *dma_data;
+	int wlen, channels, wpf;
+	int pkt_size = 0;
+	unsigned int format, div, framesize, master;
+
+	dma_data = snd_soc_dai_get_dma_data(cpu_dai, substream);
+	channels = params_channels(params);
+
+	switch (params_format(params)) {
+	case SNDRV_PCM_FORMAT_S16_LE:
+		wlen = 16;
+		break;
+	case SNDRV_PCM_FORMAT_S32_LE:
+		wlen = 32;
+		break;
+	default:
+		return -EINVAL;
+	}
+	if (mcbsp->pdata->buffer_size) {
+		if (mcbsp->dma_op_mode == MCBSP_DMA_MODE_THRESHOLD) {
+			int period_words, max_thrsh;
+			int divider = 0;
+
+			period_words = params_period_bytes(params) / (wlen / 8);
+			if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+				max_thrsh = mcbsp->max_tx_thres;
+			else
+				max_thrsh = mcbsp->max_rx_thres;
+			/*
+			 * Use sDMA packet mode if McBSP is in threshold mode:
+			 * If period words less than the FIFO size the packet
+			 * size is set to the number of period words, otherwise
+			 * Look for the biggest threshold value which divides
+			 * the period size evenly.
+			 */
+			divider = period_words / max_thrsh;
+			if (period_words % max_thrsh)
+				divider++;
+			while (period_words % divider &&
+				divider < period_words)
+				divider++;
+			if (divider == period_words)
+				return -EINVAL;
+
+			pkt_size = period_words / divider;
+		} else if (channels > 1) {
+			/* Use packet mode for non mono streams */
+			pkt_size = channels;
+		}
+		omap_mcbsp_set_threshold(substream, pkt_size);
+	}
+
+	dma_data->maxburst = pkt_size;
+
+	if (mcbsp->configured) {
+		/* McBSP already configured by another stream */
+		return 0;
+	}
+
+	regs->rcr2	&= ~(RPHASE | RFRLEN2(0x7f) | RWDLEN2(7));
+	regs->xcr2	&= ~(RPHASE | XFRLEN2(0x7f) | XWDLEN2(7));
+	regs->rcr1	&= ~(RFRLEN1(0x7f) | RWDLEN1(7));
+	regs->xcr1	&= ~(XFRLEN1(0x7f) | XWDLEN1(7));
+	format = mcbsp->fmt & SND_SOC_DAIFMT_FORMAT_MASK;
+	wpf = channels;
+	if (channels == 2 && (format == SND_SOC_DAIFMT_I2S ||
+			      format == SND_SOC_DAIFMT_LEFT_J)) {
+		/* 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 */
+		regs->rcr2	|= RWDLEN2(OMAP_MCBSP_WORD_16);
+		regs->rcr1	|= RWDLEN1(OMAP_MCBSP_WORD_16);
+		regs->xcr2	|= XWDLEN2(OMAP_MCBSP_WORD_16);
+		regs->xcr1	|= XWDLEN1(OMAP_MCBSP_WORD_16);
+		break;
+	case SNDRV_PCM_FORMAT_S32_LE:
+		/* Set word lengths */
+		regs->rcr2	|= RWDLEN2(OMAP_MCBSP_WORD_32);
+		regs->rcr1	|= RWDLEN1(OMAP_MCBSP_WORD_32);
+		regs->xcr2	|= XWDLEN2(OMAP_MCBSP_WORD_32);
+		regs->xcr1	|= XWDLEN1(OMAP_MCBSP_WORD_32);
+		break;
+	default:
+		/* Unsupported PCM format */
+		return -EINVAL;
+	}
+
+	/* In McBSP master modes, FRAME (i.e. sample rate) is generated
+	 * by _counting_ BCLKs. Calculate frame size in BCLKs */
+	master = mcbsp->fmt & SND_SOC_DAIFMT_MASTER_MASK;
+	if (master ==	SND_SOC_DAIFMT_CBS_CFS) {
+		div = mcbsp->clk_div ? mcbsp->clk_div : 1;
+		framesize = (mcbsp->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 */
+	regs->srgr2	&= ~FPER(0xfff);
+	regs->srgr1	&= ~FWID(0xff);
+	switch (format) {
+	case SND_SOC_DAIFMT_I2S:
+	case SND_SOC_DAIFMT_LEFT_J:
+		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(framesize - 1);
+		regs->srgr1	|= FWID(0);
+		break;
+	}
+
+	omap_mcbsp_config(mcbsp, &mcbsp->cfg_regs);
+	mcbsp->wlen = wlen;
+	mcbsp->configured = 1;
+
+	return 0;
+}
+
+/*
+ * This must be called before _set_clkdiv and _set_sysclk since McBSP register
+ * cache is initialized here
+ */
+static int omap_mcbsp_dai_set_dai_fmt(struct snd_soc_dai *cpu_dai,
+				      unsigned int fmt)
+{
+	struct omap_mcbsp *mcbsp = snd_soc_dai_get_drvdata(cpu_dai);
+	struct omap_mcbsp_reg_cfg *regs = &mcbsp->cfg_regs;
+	bool inv_fs = false;
+
+	if (mcbsp->configured)
+		return 0;
+
+	mcbsp->fmt = fmt;
+	memset(regs, 0, sizeof(*regs));
+	/* Generic McBSP register settings */
+	regs->spcr2	|= XINTM(3) | FREE;
+	regs->spcr1	|= RINTM(3);
+	/* RFIG and XFIG are not defined in 2430 and on OMAP3+ */
+	if (!mcbsp->pdata->has_ccr) {
+		regs->rcr2	|= RFIG;
+		regs->xcr2	|= XFIG;
+	}
+
+	/* Configure XCCR/RCCR only for revisions which have ccr registers */
+	if (mcbsp->pdata->has_ccr) {
+		regs->xccr = DXENDLY(1) | XDMAEN | XDISABLE;
+		regs->rccr = RFULL_CYCLE | RDMAEN | RDISABLE;
+	}
+
+	switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
+	case SND_SOC_DAIFMT_I2S:
+		/* 1-bit data delay */
+		regs->rcr2	|= RDATDLY(1);
+		regs->xcr2	|= XDATDLY(1);
+		break;
+	case SND_SOC_DAIFMT_LEFT_J:
+		/* 0-bit data delay */
+		regs->rcr2	|= RDATDLY(0);
+		regs->xcr2	|= XDATDLY(0);
+		regs->spcr1	|= RJUST(2);
+		/* Invert FS polarity configuration */
+		inv_fs = true;
+		break;
+	case SND_SOC_DAIFMT_DSP_A:
+		/* 1-bit data delay */
+		regs->rcr2      |= RDATDLY(1);
+		regs->xcr2      |= XDATDLY(1);
+		/* Invert FS polarity configuration */
+		inv_fs = true;
+		break;
+	case SND_SOC_DAIFMT_DSP_B:
+		/* 0-bit data delay */
+		regs->rcr2      |= RDATDLY(0);
+		regs->xcr2      |= XDATDLY(0);
+		/* Invert FS polarity configuration */
+		inv_fs = true;
+		break;
+	default:
+		/* Unsupported data format */
+		return -EINVAL;
+	}
+
+	switch (fmt & SND_SOC_DAIFMT_MASTER_MASK) {
+	case SND_SOC_DAIFMT_CBS_CFS:
+		/* McBSP master. Set FS and bit clocks as outputs */
+		regs->pcr0	|= FSXM | FSRM |
+				   CLKXM | CLKRM;
+		/* Sample rate generator drives the FS */
+		regs->srgr2	|= FSGM;
+		break;
+	case SND_SOC_DAIFMT_CBM_CFS:
+		/* McBSP slave. FS clock as output */
+		regs->srgr2	|= FSGM;
+		regs->pcr0	|= FSXM | FSRM;
+		break;
+	case SND_SOC_DAIFMT_CBM_CFM:
+		/* McBSP slave */
+		break;
+	default:
+		/* Unsupported master/slave configuration */
+		return -EINVAL;
+	}
+
+	/* Set bit clock (CLKX/CLKR) and FS polarities */
+	switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
+	case SND_SOC_DAIFMT_NB_NF:
+		/*
+		 * Normal BCLK + FS.
+		 * FS active low. TX data driven on falling edge of bit clock
+		 * and RX data sampled on rising edge of bit clock.
+		 */
+		regs->pcr0	|= FSXP | FSRP |
+				   CLKXP | CLKRP;
+		break;
+	case SND_SOC_DAIFMT_NB_IF:
+		regs->pcr0	|= CLKXP | CLKRP;
+		break;
+	case SND_SOC_DAIFMT_IB_NF:
+		regs->pcr0	|= FSXP | FSRP;
+		break;
+	case SND_SOC_DAIFMT_IB_IF:
+		break;
+	default:
+		return -EINVAL;
+	}
+	if (inv_fs == true)
+		regs->pcr0 ^= FSXP | FSRP;
+
+	return 0;
+}
+
+static int omap_mcbsp_dai_set_clkdiv(struct snd_soc_dai *cpu_dai,
+				     int div_id, int div)
+{
+	struct omap_mcbsp *mcbsp = snd_soc_dai_get_drvdata(cpu_dai);
+	struct omap_mcbsp_reg_cfg *regs = &mcbsp->cfg_regs;
+
+	if (div_id != OMAP_MCBSP_CLKGDV)
+		return -ENODEV;
+
+	mcbsp->clk_div = div;
+	regs->srgr1	&= ~CLKGDV(0xff);
+	regs->srgr1	|= CLKGDV(div - 1);
+
+	return 0;
+}
+
+static int omap_mcbsp_dai_set_dai_sysclk(struct snd_soc_dai *cpu_dai,
+					 int clk_id, unsigned int freq,
+					 int dir)
+{
+	struct omap_mcbsp *mcbsp = snd_soc_dai_get_drvdata(cpu_dai);
+	struct omap_mcbsp_reg_cfg *regs = &mcbsp->cfg_regs;
+	int err = 0;
+
+	if (mcbsp->active) {
+		if (freq == mcbsp->in_freq)
+			return 0;
+		else
+			return -EBUSY;
+	}
+
+	mcbsp->in_freq = freq;
+	regs->srgr2 &= ~CLKSM;
+	regs->pcr0 &= ~SCLKME;
+
+	switch (clk_id) {
+	case OMAP_MCBSP_SYSCLK_CLK:
+		regs->srgr2	|= CLKSM;
+		break;
+	case OMAP_MCBSP_SYSCLK_CLKS_FCLK:
+		if (mcbsp_omap1()) {
+			err = -EINVAL;
+			break;
+		}
+		err = omap2_mcbsp_set_clks_src(mcbsp,
+					       MCBSP_CLKS_PRCM_SRC);
+		break;
+	case OMAP_MCBSP_SYSCLK_CLKS_EXT:
+		if (mcbsp_omap1()) {
+			err = 0;
+			break;
+		}
+		err = omap2_mcbsp_set_clks_src(mcbsp,
+					       MCBSP_CLKS_PAD_SRC);
+		break;
+
+	case OMAP_MCBSP_SYSCLK_CLKX_EXT:
+		regs->srgr2	|= CLKSM;
+	case OMAP_MCBSP_SYSCLK_CLKR_EXT:
+		regs->pcr0	|= SCLKME;
+		break;
+	default:
+		err = -ENODEV;
+	}
+
+	return err;
+}
+
+static const struct snd_soc_dai_ops mcbsp_dai_ops = {
+	.startup	= omap_mcbsp_dai_startup,
+	.shutdown	= omap_mcbsp_dai_shutdown,
+	.trigger	= omap_mcbsp_dai_trigger,
+	.delay		= omap_mcbsp_dai_delay,
+	.hw_params	= omap_mcbsp_dai_hw_params,
+	.set_fmt	= omap_mcbsp_dai_set_dai_fmt,
+	.set_clkdiv	= omap_mcbsp_dai_set_clkdiv,
+	.set_sysclk	= omap_mcbsp_dai_set_dai_sysclk,
+};
+
+static int omap_mcbsp_probe(struct snd_soc_dai *dai)
+{
+	struct omap_mcbsp *mcbsp = snd_soc_dai_get_drvdata(dai);
+
+	pm_runtime_enable(mcbsp->dev);
+
+	snd_soc_dai_init_dma_data(dai,
+				  &mcbsp->dma_data[SNDRV_PCM_STREAM_PLAYBACK],
+				  &mcbsp->dma_data[SNDRV_PCM_STREAM_CAPTURE]);
+
+	return 0;
+}
+
+static int omap_mcbsp_remove(struct snd_soc_dai *dai)
+{
+	struct omap_mcbsp *mcbsp = snd_soc_dai_get_drvdata(dai);
+
+	pm_runtime_disable(mcbsp->dev);
+
+	return 0;
+}
+
+static struct snd_soc_dai_driver omap_mcbsp_dai = {
+	.probe = omap_mcbsp_probe,
+	.remove = omap_mcbsp_remove,
+	.playback = {
+		.channels_min = 1,
+		.channels_max = 16,
+		.rates = OMAP_MCBSP_RATES,
+		.formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S32_LE,
+	},
+	.capture = {
+		.channels_min = 1,
+		.channels_max = 16,
+		.rates = OMAP_MCBSP_RATES,
+		.formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S32_LE,
+	},
+	.ops = &mcbsp_dai_ops,
+};
+
+static const struct snd_soc_component_driver omap_mcbsp_component = {
+	.name		= "omap-mcbsp",
+};
+
+static int omap_mcbsp_st_info_volsw(struct snd_kcontrol *kcontrol,
+			struct snd_ctl_elem_info *uinfo)
+{
+	struct soc_mixer_control *mc =
+		(struct soc_mixer_control *)kcontrol->private_value;
+	int max = mc->max;
+	int min = mc->min;
+
+	uinfo->type = SNDRV_CTL_ELEM_TYPE_INTEGER;
+	uinfo->count = 1;
+	uinfo->value.integer.min = min;
+	uinfo->value.integer.max = max;
+	return 0;
+}
+
+#define OMAP_MCBSP_ST_CHANNEL_VOLUME(channel)				\
+static int								\
+omap_mcbsp_set_st_ch##channel##_volume(struct snd_kcontrol *kc,		\
+					struct snd_ctl_elem_value *uc)	\
+{									\
+	struct snd_soc_dai *cpu_dai = snd_kcontrol_chip(kc);		\
+	struct omap_mcbsp *mcbsp = snd_soc_dai_get_drvdata(cpu_dai);	\
+	struct soc_mixer_control *mc =					\
+		(struct soc_mixer_control *)kc->private_value;		\
+	int max = mc->max;						\
+	int min = mc->min;						\
+	int val = uc->value.integer.value[0];				\
+									\
+	if (val < min || val > max)					\
+		return -EINVAL;						\
+									\
+	/* OMAP McBSP implementation uses index values 0..4 */		\
+	return omap_st_set_chgain(mcbsp, channel, val);			\
+}									\
+									\
+static int								\
+omap_mcbsp_get_st_ch##channel##_volume(struct snd_kcontrol *kc,		\
+					struct snd_ctl_elem_value *uc)	\
+{									\
+	struct snd_soc_dai *cpu_dai = snd_kcontrol_chip(kc);		\
+	struct omap_mcbsp *mcbsp = snd_soc_dai_get_drvdata(cpu_dai);	\
+	s16 chgain;							\
+									\
+	if (omap_st_get_chgain(mcbsp, channel, &chgain))		\
+		return -EAGAIN;						\
+									\
+	uc->value.integer.value[0] = chgain;				\
+	return 0;							\
+}
+
+OMAP_MCBSP_ST_CHANNEL_VOLUME(0)
+OMAP_MCBSP_ST_CHANNEL_VOLUME(1)
+
+static int omap_mcbsp_st_put_mode(struct snd_kcontrol *kcontrol,
+				struct snd_ctl_elem_value *ucontrol)
+{
+	struct snd_soc_dai *cpu_dai = snd_kcontrol_chip(kcontrol);
+	struct omap_mcbsp *mcbsp = snd_soc_dai_get_drvdata(cpu_dai);
+	u8 value = ucontrol->value.integer.value[0];
+
+	if (value == omap_st_is_enabled(mcbsp))
+		return 0;
+
+	if (value)
+		omap_st_enable(mcbsp);
+	else
+		omap_st_disable(mcbsp);
+
+	return 1;
+}
+
+static int omap_mcbsp_st_get_mode(struct snd_kcontrol *kcontrol,
+				struct snd_ctl_elem_value *ucontrol)
+{
+	struct snd_soc_dai *cpu_dai = snd_kcontrol_chip(kcontrol);
+	struct omap_mcbsp *mcbsp = snd_soc_dai_get_drvdata(cpu_dai);
+
+	ucontrol->value.integer.value[0] = omap_st_is_enabled(mcbsp);
+	return 0;
+}
+
+#define OMAP_MCBSP_ST_CONTROLS(port)					  \
+static const struct snd_kcontrol_new omap_mcbsp##port##_st_controls[] = { \
+SOC_SINGLE_EXT("McBSP" #port " Sidetone Switch", 1, 0, 1, 0,		  \
+	       omap_mcbsp_st_get_mode, omap_mcbsp_st_put_mode),		  \
+OMAP_MCBSP_SOC_SINGLE_S16_EXT("McBSP" #port " Sidetone Channel 0 Volume", \
+			      -32768, 32767,				  \
+			      omap_mcbsp_get_st_ch0_volume,		  \
+			      omap_mcbsp_set_st_ch0_volume),		  \
+OMAP_MCBSP_SOC_SINGLE_S16_EXT("McBSP" #port " Sidetone Channel 1 Volume", \
+			      -32768, 32767,				  \
+			      omap_mcbsp_get_st_ch1_volume,		  \
+			      omap_mcbsp_set_st_ch1_volume),		  \
+}
+
+OMAP_MCBSP_ST_CONTROLS(2);
+OMAP_MCBSP_ST_CONTROLS(3);
+
+int omap_mcbsp_st_add_controls(struct snd_soc_pcm_runtime *rtd, int port_id)
+{
+	struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
+	struct omap_mcbsp *mcbsp = snd_soc_dai_get_drvdata(cpu_dai);
+
+	if (!mcbsp->st_data) {
+		dev_warn(mcbsp->dev, "No sidetone data for port\n");
+		return 0;
+	}
+
+	switch (port_id) {
+	case 2: /* McBSP 2 */
+		return snd_soc_add_dai_controls(cpu_dai,
+					omap_mcbsp2_st_controls,
+					ARRAY_SIZE(omap_mcbsp2_st_controls));
+	case 3: /* McBSP 3 */
+		return snd_soc_add_dai_controls(cpu_dai,
+					omap_mcbsp3_st_controls,
+					ARRAY_SIZE(omap_mcbsp3_st_controls));
+	default:
+		dev_err(mcbsp->dev, "Port %d not supported\n", port_id);
+		break;
+	}
+
+	return -EINVAL;
+}
+EXPORT_SYMBOL_GPL(omap_mcbsp_st_add_controls);
+
+static struct omap_mcbsp_platform_data omap2420_pdata = {
+	.reg_step = 4,
+	.reg_size = 2,
+};
+
+static struct omap_mcbsp_platform_data omap2430_pdata = {
+	.reg_step = 4,
+	.reg_size = 4,
+	.has_ccr = true,
+};
+
+static struct omap_mcbsp_platform_data omap3_pdata = {
+	.reg_step = 4,
+	.reg_size = 4,
+	.has_ccr = true,
+	.has_wakeup = true,
+};
+
+static struct omap_mcbsp_platform_data omap4_pdata = {
+	.reg_step = 4,
+	.reg_size = 4,
+	.has_ccr = true,
+	.has_wakeup = true,
+};
+
+static const struct of_device_id omap_mcbsp_of_match[] = {
+	{
+		.compatible = "ti,omap2420-mcbsp",
+		.data = &omap2420_pdata,
+	},
+	{
+		.compatible = "ti,omap2430-mcbsp",
+		.data = &omap2430_pdata,
+	},
+	{
+		.compatible = "ti,omap3-mcbsp",
+		.data = &omap3_pdata,
+	},
+	{
+		.compatible = "ti,omap4-mcbsp",
+		.data = &omap4_pdata,
+	},
+	{ },
+};
+MODULE_DEVICE_TABLE(of, omap_mcbsp_of_match);
+
+static int asoc_mcbsp_probe(struct platform_device *pdev)
+{
+	struct omap_mcbsp_platform_data *pdata = dev_get_platdata(&pdev->dev);
+	struct omap_mcbsp *mcbsp;
+	const struct of_device_id *match;
+	int ret;
+
+	match = of_match_device(omap_mcbsp_of_match, &pdev->dev);
+	if (match) {
+		struct device_node *node = pdev->dev.of_node;
+		int buffer_size;
+
+		pdata = devm_kzalloc(&pdev->dev,
+				     sizeof(struct omap_mcbsp_platform_data),
+				     GFP_KERNEL);
+		if (!pdata)
+			return -ENOMEM;
+
+		memcpy(pdata, match->data, sizeof(*pdata));
+		if (!of_property_read_u32(node, "ti,buffer-size", &buffer_size))
+			pdata->buffer_size = buffer_size;
+	} else if (!pdata) {
+		dev_err(&pdev->dev, "missing platform data.\n");
+		return -EINVAL;
+	}
+	mcbsp = devm_kzalloc(&pdev->dev, sizeof(struct omap_mcbsp), GFP_KERNEL);
+	if (!mcbsp)
+		return -ENOMEM;
+
+	mcbsp->id = pdev->id;
+	mcbsp->pdata = pdata;
+	mcbsp->dev = &pdev->dev;
+	platform_set_drvdata(pdev, mcbsp);
+
+	ret = omap_mcbsp_init(pdev);
+	if (ret)
+		return ret;
+
+	ret = devm_snd_soc_register_component(&pdev->dev,
+					      &omap_mcbsp_component,
+					      &omap_mcbsp_dai, 1);
+	if (ret)
+		return ret;
+
+	return omap_pcm_platform_register(&pdev->dev);
+}
+
+static int asoc_mcbsp_remove(struct platform_device *pdev)
+{
+	struct omap_mcbsp *mcbsp = platform_get_drvdata(pdev);
+
+	if (mcbsp->pdata->ops && mcbsp->pdata->ops->free)
+		mcbsp->pdata->ops->free(mcbsp->id);
+
+	omap_mcbsp_sysfs_remove(mcbsp);
+
+	clk_put(mcbsp->fclk);
+
+	return 0;
+}
+
+static struct platform_driver asoc_mcbsp_driver = {
+	.driver = {
+			.name = "omap-mcbsp",
+			.owner = THIS_MODULE,
+			.of_match_table = omap_mcbsp_of_match,
+	},
+
+	.probe = asoc_mcbsp_probe,
+	.remove = asoc_mcbsp_remove,
+};
+
+module_platform_driver(asoc_mcbsp_driver);
+
+MODULE_AUTHOR("Jarkko Nikula <jarkko.nikula@bitmer.com>");
+MODULE_DESCRIPTION("OMAP I2S SoC Interface");
+MODULE_LICENSE("GPL");
+MODULE_ALIAS("platform:omap-mcbsp");

sound/soc/omap/omap-mcbsp.h

@@ -0,0 +1,44 @@
+/*
+ * omap-mcbsp.h
+ *
+ * Copyright (C) 2008 Nokia Corporation
+ *
+ * Contact: Jarkko Nikula <jarkko.nikula@bitmer.com>
+ *          Peter Ujfalusi <peter.ujfalusi@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
+ *
+ */
+
+#ifndef __OMAP_I2S_H__
+#define __OMAP_I2S_H__
+
+/* Source clocks for McBSP sample rate generator */
+enum omap_mcbsp_clksrg_clk {
+	OMAP_MCBSP_SYSCLK_CLKS_FCLK,	/* Internal FCLK */
+	OMAP_MCBSP_SYSCLK_CLKS_EXT,	/* External CLKS pin */
+	OMAP_MCBSP_SYSCLK_CLK,		/* Internal ICLK */
+	OMAP_MCBSP_SYSCLK_CLKX_EXT,	/* External CLKX pin */
+	OMAP_MCBSP_SYSCLK_CLKR_EXT,	/* External CLKR pin */
+};
+
+/* McBSP dividers */
+enum omap_mcbsp_div {
+	OMAP_MCBSP_CLKGDV,		/* Sample rate generator divider */
+};
+
+int omap_mcbsp_st_add_controls(struct snd_soc_pcm_runtime *rtd, int port_id);
+
+#endif

sound/soc/omap/omap-mcpdm.c

@@ -0,0 +1,527 @@
+/*
+ * omap-mcpdm.c  --  OMAP ALSA SoC DAI driver using McPDM port
+ *
+ * Copyright (C) 2009 - 2011 Texas Instruments
+ *
+ * Author: Misael Lopez Cruz <misael.lopez@ti.com>
+ * Contact: Jorge Eduardo Candelaria <x0107209@ti.com>
+ *          Margarita Olaya <magi.olaya@ti.com>
+ *          Peter Ujfalusi <peter.ujfalusi@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/init.h>
+#include <linux/module.h>
+#include <linux/platform_device.h>
+#include <linux/interrupt.h>
+#include <linux/err.h>
+#include <linux/io.h>
+#include <linux/irq.h>
+#include <linux/slab.h>
+#include <linux/pm_runtime.h>
+#include <linux/of_device.h>
+
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include <sound/dmaengine_pcm.h>
+#include <sound/omap-pcm.h>
+
+#include "omap-mcpdm.h"
+
+struct mcpdm_link_config {
+	u32 link_mask; /* channel mask for the direction */
+	u32 threshold; /* FIFO threshold */
+};
+
+struct omap_mcpdm {
+	struct device *dev;
+	unsigned long phys_base;
+	void __iomem *io_base;
+	int irq;
+
+	struct mutex mutex;
+
+	/* Playback/Capture configuration */
+	struct mcpdm_link_config config[2];
+
+	/* McPDM dn offsets for rx1, and 2 channels */
+	u32 dn_rx_offset;
+
+	/* McPDM needs to be restarted due to runtime reconfiguration */
+	bool restart;
+
+	struct snd_dmaengine_dai_dma_data dma_data[2];
+};
+
+/*
+ * Stream DMA parameters
+ */
+
+static inline void omap_mcpdm_write(struct omap_mcpdm *mcpdm, u16 reg, u32 val)
+{
+	writel_relaxed(val, mcpdm->io_base + reg);
+}
+
+static inline int omap_mcpdm_read(struct omap_mcpdm *mcpdm, u16 reg)
+{
+	return readl_relaxed(mcpdm->io_base + reg);
+}
+
+#ifdef DEBUG
+static void omap_mcpdm_reg_dump(struct omap_mcpdm *mcpdm)
+{
+	dev_dbg(mcpdm->dev, "***********************\n");
+	dev_dbg(mcpdm->dev, "IRQSTATUS_RAW:  0x%04x\n",
+			omap_mcpdm_read(mcpdm, MCPDM_REG_IRQSTATUS_RAW));
+	dev_dbg(mcpdm->dev, "IRQSTATUS:  0x%04x\n",
+			omap_mcpdm_read(mcpdm, MCPDM_REG_IRQSTATUS));
+	dev_dbg(mcpdm->dev, "IRQENABLE_SET:  0x%04x\n",
+			omap_mcpdm_read(mcpdm, MCPDM_REG_IRQENABLE_SET));
+	dev_dbg(mcpdm->dev, "IRQENABLE_CLR:  0x%04x\n",
+			omap_mcpdm_read(mcpdm, MCPDM_REG_IRQENABLE_CLR));
+	dev_dbg(mcpdm->dev, "IRQWAKE_EN: 0x%04x\n",
+			omap_mcpdm_read(mcpdm, MCPDM_REG_IRQWAKE_EN));
+	dev_dbg(mcpdm->dev, "DMAENABLE_SET: 0x%04x\n",
+			omap_mcpdm_read(mcpdm, MCPDM_REG_DMAENABLE_SET));
+	dev_dbg(mcpdm->dev, "DMAENABLE_CLR:  0x%04x\n",
+			omap_mcpdm_read(mcpdm, MCPDM_REG_DMAENABLE_CLR));
+	dev_dbg(mcpdm->dev, "DMAWAKEEN:  0x%04x\n",
+			omap_mcpdm_read(mcpdm, MCPDM_REG_DMAWAKEEN));
+	dev_dbg(mcpdm->dev, "CTRL:  0x%04x\n",
+			omap_mcpdm_read(mcpdm, MCPDM_REG_CTRL));
+	dev_dbg(mcpdm->dev, "DN_DATA:  0x%04x\n",
+			omap_mcpdm_read(mcpdm, MCPDM_REG_DN_DATA));
+	dev_dbg(mcpdm->dev, "UP_DATA: 0x%04x\n",
+			omap_mcpdm_read(mcpdm, MCPDM_REG_UP_DATA));
+	dev_dbg(mcpdm->dev, "FIFO_CTRL_DN: 0x%04x\n",
+			omap_mcpdm_read(mcpdm, MCPDM_REG_FIFO_CTRL_DN));
+	dev_dbg(mcpdm->dev, "FIFO_CTRL_UP:  0x%04x\n",
+			omap_mcpdm_read(mcpdm, MCPDM_REG_FIFO_CTRL_UP));
+	dev_dbg(mcpdm->dev, "***********************\n");
+}
+#else
+static void omap_mcpdm_reg_dump(struct omap_mcpdm *mcpdm) {}
+#endif
+
+/*
+ * Enables the transfer through the PDM interface to/from the Phoenix
+ * codec by enabling the corresponding UP or DN channels.
+ */
+static void omap_mcpdm_start(struct omap_mcpdm *mcpdm)
+{
+	u32 ctrl = omap_mcpdm_read(mcpdm, MCPDM_REG_CTRL);
+	u32 link_mask = mcpdm->config[0].link_mask | mcpdm->config[1].link_mask;
+
+	ctrl |= (MCPDM_SW_DN_RST | MCPDM_SW_UP_RST);
+	omap_mcpdm_write(mcpdm, MCPDM_REG_CTRL, ctrl);
+
+	ctrl |= link_mask;
+	omap_mcpdm_write(mcpdm, MCPDM_REG_CTRL, ctrl);
+
+	ctrl &= ~(MCPDM_SW_DN_RST | MCPDM_SW_UP_RST);
+	omap_mcpdm_write(mcpdm, MCPDM_REG_CTRL, ctrl);
+}
+
+/*
+ * Disables the transfer through the PDM interface to/from the Phoenix
+ * codec by disabling the corresponding UP or DN channels.
+ */
+static void omap_mcpdm_stop(struct omap_mcpdm *mcpdm)
+{
+	u32 ctrl = omap_mcpdm_read(mcpdm, MCPDM_REG_CTRL);
+	u32 link_mask = MCPDM_PDM_DN_MASK | MCPDM_PDM_UP_MASK;
+
+	ctrl |= (MCPDM_SW_DN_RST | MCPDM_SW_UP_RST);
+	omap_mcpdm_write(mcpdm, MCPDM_REG_CTRL, ctrl);
+
+	ctrl &= ~(link_mask);
+	omap_mcpdm_write(mcpdm, MCPDM_REG_CTRL, ctrl);
+
+	ctrl &= ~(MCPDM_SW_DN_RST | MCPDM_SW_UP_RST);
+	omap_mcpdm_write(mcpdm, MCPDM_REG_CTRL, ctrl);
+
+}
+
+/*
+ * Is the physical McPDM interface active.
+ */
+static inline int omap_mcpdm_active(struct omap_mcpdm *mcpdm)
+{
+	return omap_mcpdm_read(mcpdm, MCPDM_REG_CTRL) &
+					(MCPDM_PDM_DN_MASK | MCPDM_PDM_UP_MASK);
+}
+
+/*
+ * Configures McPDM uplink, and downlink for audio.
+ * This function should be called before omap_mcpdm_start.
+ */
+static void omap_mcpdm_open_streams(struct omap_mcpdm *mcpdm)
+{
+	omap_mcpdm_write(mcpdm, MCPDM_REG_IRQENABLE_SET,
+			MCPDM_DN_IRQ_EMPTY | MCPDM_DN_IRQ_FULL |
+			MCPDM_UP_IRQ_EMPTY | MCPDM_UP_IRQ_FULL);
+
+	/* Enable DN RX1/2 offset cancellation feature, if configured */
+	if (mcpdm->dn_rx_offset) {
+		u32 dn_offset = mcpdm->dn_rx_offset;
+
+		omap_mcpdm_write(mcpdm, MCPDM_REG_DN_OFFSET, dn_offset);
+		dn_offset |= (MCPDM_DN_OFST_RX1_EN | MCPDM_DN_OFST_RX2_EN);
+		omap_mcpdm_write(mcpdm, MCPDM_REG_DN_OFFSET, dn_offset);
+	}
+
+	omap_mcpdm_write(mcpdm, MCPDM_REG_FIFO_CTRL_DN,
+			 mcpdm->config[SNDRV_PCM_STREAM_PLAYBACK].threshold);
+	omap_mcpdm_write(mcpdm, MCPDM_REG_FIFO_CTRL_UP,
+			 mcpdm->config[SNDRV_PCM_STREAM_CAPTURE].threshold);
+
+	omap_mcpdm_write(mcpdm, MCPDM_REG_DMAENABLE_SET,
+			MCPDM_DMA_DN_ENABLE | MCPDM_DMA_UP_ENABLE);
+}
+
+/*
+ * Cleans McPDM uplink, and downlink configuration.
+ * This function should be called when the stream is closed.
+ */
+static void omap_mcpdm_close_streams(struct omap_mcpdm *mcpdm)
+{
+	/* Disable irq request generation for downlink */
+	omap_mcpdm_write(mcpdm, MCPDM_REG_IRQENABLE_CLR,
+			MCPDM_DN_IRQ_EMPTY | MCPDM_DN_IRQ_FULL);
+
+	/* Disable DMA request generation for downlink */
+	omap_mcpdm_write(mcpdm, MCPDM_REG_DMAENABLE_CLR, MCPDM_DMA_DN_ENABLE);
+
+	/* Disable irq request generation for uplink */
+	omap_mcpdm_write(mcpdm, MCPDM_REG_IRQENABLE_CLR,
+			MCPDM_UP_IRQ_EMPTY | MCPDM_UP_IRQ_FULL);
+
+	/* Disable DMA request generation for uplink */
+	omap_mcpdm_write(mcpdm, MCPDM_REG_DMAENABLE_CLR, MCPDM_DMA_UP_ENABLE);
+
+	/* Disable RX1/2 offset cancellation */
+	if (mcpdm->dn_rx_offset)
+		omap_mcpdm_write(mcpdm, MCPDM_REG_DN_OFFSET, 0);
+}
+
+static irqreturn_t omap_mcpdm_irq_handler(int irq, void *dev_id)
+{
+	struct omap_mcpdm *mcpdm = dev_id;
+	int irq_status;
+
+	irq_status = omap_mcpdm_read(mcpdm, MCPDM_REG_IRQSTATUS);
+
+	/* Acknowledge irq event */
+	omap_mcpdm_write(mcpdm, MCPDM_REG_IRQSTATUS, irq_status);
+
+	if (irq_status & MCPDM_DN_IRQ_FULL)
+		dev_dbg(mcpdm->dev, "DN (playback) FIFO Full\n");
+
+	if (irq_status & MCPDM_DN_IRQ_EMPTY)
+		dev_dbg(mcpdm->dev, "DN (playback) FIFO Empty\n");
+
+	if (irq_status & MCPDM_DN_IRQ)
+		dev_dbg(mcpdm->dev, "DN (playback) write request\n");
+
+	if (irq_status & MCPDM_UP_IRQ_FULL)
+		dev_dbg(mcpdm->dev, "UP (capture) FIFO Full\n");
+
+	if (irq_status & MCPDM_UP_IRQ_EMPTY)
+		dev_dbg(mcpdm->dev, "UP (capture) FIFO Empty\n");
+
+	if (irq_status & MCPDM_UP_IRQ)
+		dev_dbg(mcpdm->dev, "UP (capture) write request\n");
+
+	return IRQ_HANDLED;
+}
+
+static int omap_mcpdm_dai_startup(struct snd_pcm_substream *substream,
+				  struct snd_soc_dai *dai)
+{
+	struct omap_mcpdm *mcpdm = snd_soc_dai_get_drvdata(dai);
+
+	mutex_lock(&mcpdm->mutex);
+
+	if (!dai->active) {
+		u32 ctrl = omap_mcpdm_read(mcpdm, MCPDM_REG_CTRL);
+
+		omap_mcpdm_write(mcpdm, MCPDM_REG_CTRL, ctrl | MCPDM_WD_EN);
+		omap_mcpdm_open_streams(mcpdm);
+	}
+	mutex_unlock(&mcpdm->mutex);
+
+	return 0;
+}
+
+static void omap_mcpdm_dai_shutdown(struct snd_pcm_substream *substream,
+				  struct snd_soc_dai *dai)
+{
+	struct omap_mcpdm *mcpdm = snd_soc_dai_get_drvdata(dai);
+
+	mutex_lock(&mcpdm->mutex);
+
+	if (!dai->active) {
+		if (omap_mcpdm_active(mcpdm)) {
+			omap_mcpdm_stop(mcpdm);
+			omap_mcpdm_close_streams(mcpdm);
+			mcpdm->config[0].link_mask = 0;
+			mcpdm->config[1].link_mask = 0;
+		}
+	}
+
+	mutex_unlock(&mcpdm->mutex);
+}
+
+static int omap_mcpdm_dai_hw_params(struct snd_pcm_substream *substream,
+				    struct snd_pcm_hw_params *params,
+				    struct snd_soc_dai *dai)
+{
+	struct omap_mcpdm *mcpdm = snd_soc_dai_get_drvdata(dai);
+	int stream = substream->stream;
+	struct snd_dmaengine_dai_dma_data *dma_data;
+	u32 threshold;
+	int channels;
+	int link_mask = 0;
+
+	channels = params_channels(params);
+	switch (channels) {
+	case 5:
+		if (stream == SNDRV_PCM_STREAM_CAPTURE)
+			/* up to 3 channels for capture */
+			return -EINVAL;
+		link_mask |= 1 << 4;
+	case 4:
+		if (stream == SNDRV_PCM_STREAM_CAPTURE)
+			/* up to 3 channels for capture */
+			return -EINVAL;
+		link_mask |= 1 << 3;
+	case 3:
+		link_mask |= 1 << 2;
+	case 2:
+		link_mask |= 1 << 1;
+	case 1:
+		link_mask |= 1 << 0;
+		break;
+	default:
+		/* unsupported number of channels */
+		return -EINVAL;
+	}
+
+	dma_data = snd_soc_dai_get_dma_data(dai, substream);
+
+	threshold = mcpdm->config[stream].threshold;
+	/* Configure McPDM channels, and DMA packet size */
+	if (stream == SNDRV_PCM_STREAM_PLAYBACK) {
+		link_mask <<= 3;
+
+		/* If capture is not running assume a stereo stream to come */
+		if (!mcpdm->config[!stream].link_mask)
+			mcpdm->config[!stream].link_mask = 0x3;
+
+		dma_data->maxburst =
+				(MCPDM_DN_THRES_MAX - threshold) * channels;
+	} else {
+		/* If playback is not running assume a stereo stream to come */
+		if (!mcpdm->config[!stream].link_mask)
+			mcpdm->config[!stream].link_mask = (0x3 << 3);
+
+		dma_data->maxburst = threshold * channels;
+	}
+
+	/* Check if we need to restart McPDM with this stream */
+	if (mcpdm->config[stream].link_mask &&
+	    mcpdm->config[stream].link_mask != link_mask)
+		mcpdm->restart = true;
+
+	mcpdm->config[stream].link_mask = link_mask;
+
+	return 0;
+}
+
+static int omap_mcpdm_prepare(struct snd_pcm_substream *substream,
+				  struct snd_soc_dai *dai)
+{
+	struct omap_mcpdm *mcpdm = snd_soc_dai_get_drvdata(dai);
+
+	if (!omap_mcpdm_active(mcpdm)) {
+		omap_mcpdm_start(mcpdm);
+		omap_mcpdm_reg_dump(mcpdm);
+	} else if (mcpdm->restart) {
+		omap_mcpdm_stop(mcpdm);
+		omap_mcpdm_start(mcpdm);
+		mcpdm->restart = false;
+		omap_mcpdm_reg_dump(mcpdm);
+	}
+
+	return 0;
+}
+
+static const struct snd_soc_dai_ops omap_mcpdm_dai_ops = {
+	.startup	= omap_mcpdm_dai_startup,
+	.shutdown	= omap_mcpdm_dai_shutdown,
+	.hw_params	= omap_mcpdm_dai_hw_params,
+	.prepare	= omap_mcpdm_prepare,
+};
+
+static int omap_mcpdm_probe(struct snd_soc_dai *dai)
+{
+	struct omap_mcpdm *mcpdm = snd_soc_dai_get_drvdata(dai);
+	int ret;
+
+	pm_runtime_enable(mcpdm->dev);
+
+	/* Disable lines while request is ongoing */
+	pm_runtime_get_sync(mcpdm->dev);
+	omap_mcpdm_write(mcpdm, MCPDM_REG_CTRL, 0x00);
+
+	ret = devm_request_irq(mcpdm->dev, mcpdm->irq, omap_mcpdm_irq_handler,
+				0, "McPDM", (void *)mcpdm);
+
+	pm_runtime_put_sync(mcpdm->dev);
+
+	if (ret) {
+		dev_err(mcpdm->dev, "Request for IRQ failed\n");
+		pm_runtime_disable(mcpdm->dev);
+	}
+
+	/* Configure McPDM threshold values */
+	mcpdm->config[SNDRV_PCM_STREAM_PLAYBACK].threshold = 2;
+	mcpdm->config[SNDRV_PCM_STREAM_CAPTURE].threshold =
+							MCPDM_UP_THRES_MAX - 3;
+
+	snd_soc_dai_init_dma_data(dai,
+				  &mcpdm->dma_data[SNDRV_PCM_STREAM_PLAYBACK],
+				  &mcpdm->dma_data[SNDRV_PCM_STREAM_CAPTURE]);
+
+	return ret;
+}
+
+static int omap_mcpdm_remove(struct snd_soc_dai *dai)
+{
+	struct omap_mcpdm *mcpdm = snd_soc_dai_get_drvdata(dai);
+
+	pm_runtime_disable(mcpdm->dev);
+
+	return 0;
+}
+
+#define OMAP_MCPDM_RATES	(SNDRV_PCM_RATE_88200 | SNDRV_PCM_RATE_96000)
+#define OMAP_MCPDM_FORMATS	SNDRV_PCM_FMTBIT_S32_LE
+
+static struct snd_soc_dai_driver omap_mcpdm_dai = {
+	.probe = omap_mcpdm_probe,
+	.remove = omap_mcpdm_remove,
+	.probe_order = SND_SOC_COMP_ORDER_LATE,
+	.remove_order = SND_SOC_COMP_ORDER_EARLY,
+	.playback = {
+		.channels_min = 1,
+		.channels_max = 5,
+		.rates = OMAP_MCPDM_RATES,
+		.formats = OMAP_MCPDM_FORMATS,
+		.sig_bits = 24,
+	},
+	.capture = {
+		.channels_min = 1,
+		.channels_max = 3,
+		.rates = OMAP_MCPDM_RATES,
+		.formats = OMAP_MCPDM_FORMATS,
+		.sig_bits = 24,
+	},
+	.ops = &omap_mcpdm_dai_ops,
+};
+
+static const struct snd_soc_component_driver omap_mcpdm_component = {
+	.name		= "omap-mcpdm",
+};
+
+void omap_mcpdm_configure_dn_offsets(struct snd_soc_pcm_runtime *rtd,
+				    u8 rx1, u8 rx2)
+{
+	struct omap_mcpdm *mcpdm = snd_soc_dai_get_drvdata(rtd->cpu_dai);
+
+	mcpdm->dn_rx_offset = MCPDM_DNOFST_RX1(rx1) | MCPDM_DNOFST_RX2(rx2);
+}
+EXPORT_SYMBOL_GPL(omap_mcpdm_configure_dn_offsets);
+
+static int asoc_mcpdm_probe(struct platform_device *pdev)
+{
+	struct omap_mcpdm *mcpdm;
+	struct resource *res;
+	int ret;
+
+	mcpdm = devm_kzalloc(&pdev->dev, sizeof(struct omap_mcpdm), GFP_KERNEL);
+	if (!mcpdm)
+		return -ENOMEM;
+
+	platform_set_drvdata(pdev, mcpdm);
+
+	mutex_init(&mcpdm->mutex);
+
+	res = platform_get_resource_byname(pdev, IORESOURCE_MEM, "dma");
+	if (res == NULL)
+		return -ENOMEM;
+
+	mcpdm->dma_data[0].addr = res->start + MCPDM_REG_DN_DATA;
+	mcpdm->dma_data[1].addr = res->start + MCPDM_REG_UP_DATA;
+
+	mcpdm->dma_data[0].filter_data = "dn_link";
+	mcpdm->dma_data[1].filter_data = "up_link";
+
+	res = platform_get_resource_byname(pdev, IORESOURCE_MEM, "mpu");
+	mcpdm->io_base = devm_ioremap_resource(&pdev->dev, res);
+	if (IS_ERR(mcpdm->io_base))
+		return PTR_ERR(mcpdm->io_base);
+
+	mcpdm->irq = platform_get_irq(pdev, 0);
+	if (mcpdm->irq < 0)
+		return mcpdm->irq;
+
+	mcpdm->dev = &pdev->dev;
+
+	ret =  devm_snd_soc_register_component(&pdev->dev,
+					       &omap_mcpdm_component,
+					       &omap_mcpdm_dai, 1);
+	if (ret)
+		return ret;
+
+	return omap_pcm_platform_register(&pdev->dev);
+}
+
+static const struct of_device_id omap_mcpdm_of_match[] = {
+	{ .compatible = "ti,omap4-mcpdm", },
+	{ }
+};
+MODULE_DEVICE_TABLE(of, omap_mcpdm_of_match);
+
+static struct platform_driver asoc_mcpdm_driver = {
+	.driver = {
+		.name	= "omap-mcpdm",
+		.owner	= THIS_MODULE,
+		.of_match_table = omap_mcpdm_of_match,
+	},
+
+	.probe	= asoc_mcpdm_probe,
+};
+
+module_platform_driver(asoc_mcpdm_driver);
+
+MODULE_ALIAS("platform:omap-mcpdm");
+MODULE_AUTHOR("Misael Lopez Cruz <misael.lopez@ti.com>");
+MODULE_DESCRIPTION("OMAP PDM SoC Interface");
+MODULE_LICENSE("GPL");

sound/soc/omap/omap-mcpdm.h

@@ -0,0 +1,107 @@
+/*
+ * omap-mcpdm.h
+ *
+ * Copyright (C) 2009 - 2011 Texas Instruments
+ *
+ * Contact: Misael Lopez Cruz <misael.lopez@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
+ *
+ */
+
+#ifndef __OMAP_MCPDM_H__
+#define __OMAP_MCPDM_H__
+
+#define MCPDM_REG_REVISION		0x00
+#define MCPDM_REG_SYSCONFIG		0x10
+#define MCPDM_REG_IRQSTATUS_RAW		0x24
+#define MCPDM_REG_IRQSTATUS		0x28
+#define MCPDM_REG_IRQENABLE_SET		0x2C
+#define MCPDM_REG_IRQENABLE_CLR		0x30
+#define MCPDM_REG_IRQWAKE_EN		0x34
+#define MCPDM_REG_DMAENABLE_SET		0x38
+#define MCPDM_REG_DMAENABLE_CLR		0x3C
+#define MCPDM_REG_DMAWAKEEN		0x40
+#define MCPDM_REG_CTRL			0x44
+#define MCPDM_REG_DN_DATA		0x48
+#define MCPDM_REG_UP_DATA		0x4C
+#define MCPDM_REG_FIFO_CTRL_DN		0x50
+#define MCPDM_REG_FIFO_CTRL_UP		0x54
+#define MCPDM_REG_DN_OFFSET		0x58
+
+/*
+ * MCPDM_IRQ bit fields
+ * IRQSTATUS_RAW, IRQSTATUS, IRQENABLE_SET, IRQENABLE_CLR
+ */
+
+#define MCPDM_DN_IRQ			(1 << 0)
+#define MCPDM_DN_IRQ_EMPTY		(1 << 1)
+#define MCPDM_DN_IRQ_ALMST_EMPTY	(1 << 2)
+#define MCPDM_DN_IRQ_FULL		(1 << 3)
+
+#define MCPDM_UP_IRQ			(1 << 8)
+#define MCPDM_UP_IRQ_EMPTY		(1 << 9)
+#define MCPDM_UP_IRQ_ALMST_FULL		(1 << 10)
+#define MCPDM_UP_IRQ_FULL		(1 << 11)
+
+#define MCPDM_DOWNLINK_IRQ_MASK		0x00F
+#define MCPDM_UPLINK_IRQ_MASK		0xF00
+
+/*
+ * MCPDM_DMAENABLE bit fields
+ */
+
+#define MCPDM_DMA_DN_ENABLE		(1 << 0)
+#define MCPDM_DMA_UP_ENABLE		(1 << 1)
+
+/*
+ * MCPDM_CTRL bit fields
+ */
+
+#define MCPDM_PDM_UPLINK_EN(x)		(1 << (x - 1)) /* ch1 is at bit 0 */
+#define MCPDM_PDM_DOWNLINK_EN(x)	(1 << (x + 2)) /* ch1 is at bit 3 */
+#define MCPDM_PDMOUTFORMAT		(1 << 8)
+#define MCPDM_CMD_INT			(1 << 9)
+#define MCPDM_STATUS_INT		(1 << 10)
+#define MCPDM_SW_UP_RST			(1 << 11)
+#define MCPDM_SW_DN_RST			(1 << 12)
+#define MCPDM_WD_EN			(1 << 14)
+#define MCPDM_PDM_UP_MASK		0x7
+#define MCPDM_PDM_DN_MASK		(0x1f << 3)
+
+
+#define MCPDM_PDMOUTFORMAT_LJUST	(0 << 8)
+#define MCPDM_PDMOUTFORMAT_RJUST	(1 << 8)
+
+/*
+ * MCPDM_FIFO_CTRL bit fields
+ */
+
+#define MCPDM_UP_THRES_MAX		0xF
+#define MCPDM_DN_THRES_MAX		0xF
+
+/*
+ * MCPDM_DN_OFFSET bit fields
+ */
+
+#define MCPDM_DN_OFST_RX1_EN		(1 << 0)
+#define MCPDM_DNOFST_RX1(x)		((x & 0x1f) << 1)
+#define MCPDM_DN_OFST_RX2_EN		(1 << 8)
+#define MCPDM_DNOFST_RX2(x)		((x & 0x1f) << 9)
+
+void omap_mcpdm_configure_dn_offsets(struct snd_soc_pcm_runtime *rtd,
+				    u8 rx1, u8 rx2);
+
+#endif	/* End of __OMAP_MCPDM_H__ */

sound/soc/omap/omap-pcm.c

@@ -0,0 +1,244 @@
+/*
+ * omap-pcm.c  --  ALSA PCM interface for the OMAP SoC
+ *
+ * Copyright (C) 2008 Nokia Corporation
+ *
+ * Contact: Jarkko Nikula <jarkko.nikula@bitmer.com>
+ *          Peter Ujfalusi <peter.ujfalusi@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/dma-mapping.h>
+#include <linux/slab.h>
+#include <linux/module.h>
+#include <linux/omap-dma.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/dmaengine_pcm.h>
+#include <sound/soc.h>
+#include <sound/omap-pcm.h>
+
+#ifdef CONFIG_ARCH_OMAP1
+#define pcm_omap1510()	cpu_is_omap1510()
+#else
+#define pcm_omap1510()	0
+#endif
+
+static const struct snd_pcm_hardware omap_pcm_hardware = {
+	.info			= SNDRV_PCM_INFO_MMAP |
+				  SNDRV_PCM_INFO_MMAP_VALID |
+				  SNDRV_PCM_INFO_INTERLEAVED |
+				  SNDRV_PCM_INFO_PAUSE |
+				  SNDRV_PCM_INFO_RESUME |
+				  SNDRV_PCM_INFO_NO_PERIOD_WAKEUP,
+	.period_bytes_min	= 32,
+	.period_bytes_max	= 64 * 1024,
+	.periods_min		= 2,
+	.periods_max		= 255,
+	.buffer_bytes_max	= 128 * 1024,
+};
+
+/* this may get called several times by oss emulation */
+static int omap_pcm_hw_params(struct snd_pcm_substream *substream,
+			      struct snd_pcm_hw_params *params)
+{
+	struct snd_pcm_runtime *runtime = substream->runtime;
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct omap_pcm_dma_data *dma_data;
+	struct dma_slave_config config;
+	struct dma_chan *chan;
+	int err = 0;
+
+	dma_data = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream);
+
+	/* return if this is a bufferless transfer e.g.
+	 * codec <--> BT codec or GSM modem -- lg FIXME */
+	if (!dma_data)
+		return 0;
+
+	snd_pcm_set_runtime_buffer(substream, &substream->dma_buffer);
+	runtime->dma_bytes = params_buffer_bytes(params);
+
+	chan = snd_dmaengine_pcm_get_chan(substream);
+	if (!chan)
+		return -EINVAL;
+
+	/* fills in addr_width and direction */
+	err = snd_hwparams_to_dma_slave_config(substream, params, &config);
+	if (err)
+		return err;
+
+	snd_dmaengine_pcm_set_config_from_dai_data(substream,
+			snd_soc_dai_get_dma_data(rtd->cpu_dai, substream),
+			&config);
+
+	return dmaengine_slave_config(chan, &config);
+}
+
+static int omap_pcm_hw_free(struct snd_pcm_substream *substream)
+{
+	snd_pcm_set_runtime_buffer(substream, NULL);
+	return 0;
+}
+
+static snd_pcm_uframes_t omap_pcm_pointer(struct snd_pcm_substream *substream)
+{
+	snd_pcm_uframes_t offset;
+
+	if (pcm_omap1510())
+		offset = snd_dmaengine_pcm_pointer_no_residue(substream);
+	else
+		offset = snd_dmaengine_pcm_pointer(substream);
+
+	return offset;
+}
+
+static int omap_pcm_open(struct snd_pcm_substream *substream)
+{
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_dmaengine_dai_dma_data *dma_data;
+	int ret;
+
+	snd_soc_set_runtime_hwparams(substream, &omap_pcm_hardware);
+
+	dma_data = snd_soc_dai_get_dma_data(rtd->cpu_dai, substream);
+
+	/* DT boot: filter_data is the DMA name */
+	if (rtd->cpu_dai->dev->of_node) {
+		struct dma_chan *chan;
+
+		chan = dma_request_slave_channel(rtd->cpu_dai->dev,
+						 dma_data->filter_data);
+		ret = snd_dmaengine_pcm_open(substream, chan);
+	} else {
+		ret = snd_dmaengine_pcm_open_request_chan(substream,
+							  omap_dma_filter_fn,
+							  dma_data->filter_data);
+	}
+	return ret;
+}
+
+static int omap_pcm_mmap(struct snd_pcm_substream *substream,
+	struct vm_area_struct *vma)
+{
+	struct snd_pcm_runtime *runtime = substream->runtime;
+
+	return dma_mmap_writecombine(substream->pcm->card->dev, vma,
+				     runtime->dma_area,
+				     runtime->dma_addr,
+				     runtime->dma_bytes);
+}
+
+static struct snd_pcm_ops omap_pcm_ops = {
+	.open		= omap_pcm_open,
+	.close		= snd_dmaengine_pcm_close_release_chan,
+	.ioctl		= snd_pcm_lib_ioctl,
+	.hw_params	= omap_pcm_hw_params,
+	.hw_free	= omap_pcm_hw_free,
+	.trigger	= snd_dmaengine_pcm_trigger,
+	.pointer	= omap_pcm_pointer,
+	.mmap		= omap_pcm_mmap,
+};
+
+static int omap_pcm_preallocate_dma_buffer(struct snd_pcm *pcm,
+	int stream)
+{
+	struct snd_pcm_substream *substream = pcm->streams[stream].substream;
+	struct snd_dma_buffer *buf = &substream->dma_buffer;
+	size_t size = omap_pcm_hardware.buffer_bytes_max;
+
+	buf->dev.type = SNDRV_DMA_TYPE_DEV;
+	buf->dev.dev = pcm->card->dev;
+	buf->private_data = NULL;
+	buf->area = dma_alloc_writecombine(pcm->card->dev, size,
+					   &buf->addr, GFP_KERNEL);
+	if (!buf->area)
+		return -ENOMEM;
+
+	buf->bytes = size;
+	return 0;
+}
+
+static void omap_pcm_free_dma_buffers(struct snd_pcm *pcm)
+{
+	struct snd_pcm_substream *substream;
+	struct snd_dma_buffer *buf;
+	int stream;
+
+	for (stream = 0; stream < 2; stream++) {
+		substream = pcm->streams[stream].substream;
+		if (!substream)
+			continue;
+
+		buf = &substream->dma_buffer;
+		if (!buf->area)
+			continue;
+
+		dma_free_writecombine(pcm->card->dev, buf->bytes,
+				      buf->area, buf->addr);
+		buf->area = NULL;
+	}
+}
+
+static int omap_pcm_new(struct snd_soc_pcm_runtime *rtd)
+{
+	struct snd_card *card = rtd->card->snd_card;
+	struct snd_pcm *pcm = rtd->pcm;
+	int ret;
+
+	ret = dma_coerce_mask_and_coherent(card->dev, DMA_BIT_MASK(32));
+	if (ret)
+		return ret;
+
+	if (pcm->streams[SNDRV_PCM_STREAM_PLAYBACK].substream) {
+		ret = omap_pcm_preallocate_dma_buffer(pcm,
+			SNDRV_PCM_STREAM_PLAYBACK);
+		if (ret)
+			goto out;
+	}
+
+	if (pcm->streams[SNDRV_PCM_STREAM_CAPTURE].substream) {
+		ret = omap_pcm_preallocate_dma_buffer(pcm,
+			SNDRV_PCM_STREAM_CAPTURE);
+		if (ret)
+			goto out;
+	}
+
+out:
+	/* free preallocated buffers in case of error */
+	if (ret)
+		omap_pcm_free_dma_buffers(pcm);
+
+	return ret;
+}
+
+static struct snd_soc_platform_driver omap_soc_platform = {
+	.ops		= &omap_pcm_ops,
+	.pcm_new	= omap_pcm_new,
+	.pcm_free	= omap_pcm_free_dma_buffers,
+};
+
+int omap_pcm_platform_register(struct device *dev)
+{
+	return devm_snd_soc_register_platform(dev, &omap_soc_platform);
+}
+EXPORT_SYMBOL_GPL(omap_pcm_platform_register);
+
+MODULE_AUTHOR("Jarkko Nikula <jarkko.nikula@bitmer.com>");
+MODULE_DESCRIPTION("OMAP PCM DMA module");
+MODULE_LICENSE("GPL");

sound/soc/omap/omap-twl4030.c

@@ -0,0 +1,390 @@
+/*
+ * omap-twl4030.c  --  SoC audio for TI SoC based boards with twl4030 codec
+ *
+ * Copyright (C) 2012 Texas Instruments Incorporated - http://www.ti.com
+ * All rights reserved.
+ *
+ * Author: Peter Ujfalusi <peter.ujfalusi@ti.com>
+ *
+ * This driver replaces the following machine drivers:
+ * omap3beagle (Author: Steve Sakoman <steve@sakoman.com>)
+ * omap3evm (Author: Anuj Aggarwal <anuj.aggarwal@ti.com>)
+ * overo (Author: Steve Sakoman <steve@sakoman.com>)
+ * igep0020 (Author: Enric Balletbo i Serra <eballetbo@iseebcn.com>)
+ * zoom2 (Author: Misael Lopez Cruz <misael.lopez@ti.com>)
+ * sdp3430 (Author: Misael Lopez Cruz <misael.lopez@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/platform_device.h>
+#include <linux/platform_data/omap-twl4030.h>
+#include <linux/module.h>
+#include <linux/of.h>
+#include <linux/gpio.h>
+#include <linux/of_gpio.h>
+
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/soc.h>
+#include <sound/jack.h>
+
+#include "omap-mcbsp.h"
+
+struct omap_twl4030 {
+	int jack_detect;	/* board can detect jack events */
+	struct snd_soc_jack hs_jack;
+};
+
+static int omap_twl4030_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->codec_dai;
+	struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
+	struct snd_soc_card *card = rtd->card;
+	unsigned int fmt;
+	int ret;
+
+	switch (params_channels(params)) {
+	case 2: /* Stereo I2S mode */
+		fmt =	SND_SOC_DAIFMT_I2S |
+			SND_SOC_DAIFMT_NB_NF |
+			SND_SOC_DAIFMT_CBM_CFM;
+		break;
+	case 4: /* Four channel TDM mode */
+		fmt =	SND_SOC_DAIFMT_DSP_A |
+			SND_SOC_DAIFMT_IB_NF |
+			SND_SOC_DAIFMT_CBM_CFM;
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	/* Set codec DAI configuration */
+	ret = snd_soc_dai_set_fmt(codec_dai, fmt);
+	if (ret < 0) {
+		dev_err(card->dev, "can't set codec DAI configuration\n");
+		return ret;
+	}
+
+	/* Set cpu DAI configuration */
+	ret = snd_soc_dai_set_fmt(cpu_dai, fmt);
+	if (ret < 0) {
+		dev_err(card->dev, "can't set cpu DAI configuration\n");
+		return ret;
+	}
+
+	return 0;
+}
+
+static struct snd_soc_ops omap_twl4030_ops = {
+	.hw_params = omap_twl4030_hw_params,
+};
+
+static const struct snd_soc_dapm_widget dapm_widgets[] = {
+	SND_SOC_DAPM_SPK("Earpiece Spk", NULL),
+	SND_SOC_DAPM_SPK("Handsfree Spk", NULL),
+	SND_SOC_DAPM_HP("Headset Stereophone", NULL),
+	SND_SOC_DAPM_SPK("Ext Spk", NULL),
+	SND_SOC_DAPM_SPK("Carkit Spk", NULL),
+
+	SND_SOC_DAPM_MIC("Main Mic", NULL),
+	SND_SOC_DAPM_MIC("Sub Mic", NULL),
+	SND_SOC_DAPM_MIC("Headset Mic", NULL),
+	SND_SOC_DAPM_MIC("Carkit Mic", NULL),
+	SND_SOC_DAPM_MIC("Digital0 Mic", NULL),
+	SND_SOC_DAPM_MIC("Digital1 Mic", NULL),
+	SND_SOC_DAPM_LINE("Line In", NULL),
+};
+
+static const struct snd_soc_dapm_route audio_map[] = {
+	/* Headset Stereophone:  HSOL, HSOR */
+	{"Headset Stereophone", NULL, "HSOL"},
+	{"Headset Stereophone", NULL, "HSOR"},
+	/* External Speakers: HFL, HFR */
+	{"Handsfree Spk", NULL, "HFL"},
+	{"Handsfree Spk", NULL, "HFR"},
+	/* External Speakers: PredrivL, PredrivR */
+	{"Ext Spk", NULL, "PREDRIVEL"},
+	{"Ext Spk", NULL, "PREDRIVER"},
+	/* Carkit speakers:  CARKITL, CARKITR */
+	{"Carkit Spk", NULL, "CARKITL"},
+	{"Carkit Spk", NULL, "CARKITR"},
+	/* Earpiece */
+	{"Earpiece Spk", NULL, "EARPIECE"},
+
+	/* External Mics: MAINMIC, SUBMIC with bias */
+	{"MAINMIC", NULL, "Main Mic"},
+	{"Main Mic", NULL, "Mic Bias 1"},
+	{"SUBMIC", NULL, "Sub Mic"},
+	{"Sub Mic", NULL, "Mic Bias 2"},
+	/* Headset Mic: HSMIC with bias */
+	{"HSMIC", NULL, "Headset Mic"},
+	{"Headset Mic", NULL, "Headset Mic Bias"},
+	/* Digital Mics: DIGIMIC0, DIGIMIC1 with bias */
+	{"DIGIMIC0", NULL, "Digital0 Mic"},
+	{"Digital0 Mic", NULL, "Mic Bias 1"},
+	{"DIGIMIC1", NULL, "Digital1 Mic"},
+	{"Digital1 Mic", NULL, "Mic Bias 2"},
+	/* Carkit In: CARKITMIC */
+	{"CARKITMIC", NULL, "Carkit Mic"},
+	/* Aux In: AUXL, AUXR */
+	{"AUXL", NULL, "Line In"},
+	{"AUXR", NULL, "Line In"},
+};
+
+/* Headset jack detection DAPM pins */
+static struct snd_soc_jack_pin hs_jack_pins[] = {
+	{
+		.pin = "Headset Mic",
+		.mask = SND_JACK_MICROPHONE,
+	},
+	{
+		.pin = "Headset Stereophone",
+		.mask = SND_JACK_HEADPHONE,
+	},
+};
+
+/* Headset jack detection gpios */
+static struct snd_soc_jack_gpio hs_jack_gpios[] = {
+	{
+		.name = "hsdet-gpio",
+		.report = SND_JACK_HEADSET,
+		.debounce_time = 200,
+	},
+};
+
+static inline void twl4030_disconnect_pin(struct snd_soc_dapm_context *dapm,
+					  int connected, char *pin)
+{
+	if (!connected)
+		snd_soc_dapm_disable_pin(dapm, pin);
+}
+
+static int omap_twl4030_init(struct snd_soc_pcm_runtime *rtd)
+{
+	struct snd_soc_codec *codec = rtd->codec;
+	struct snd_soc_card *card = rtd->card;
+	struct snd_soc_dapm_context *dapm = &codec->dapm;
+	struct omap_tw4030_pdata *pdata = dev_get_platdata(card->dev);
+	struct omap_twl4030 *priv = snd_soc_card_get_drvdata(card);
+	int ret = 0;
+
+	/* Headset jack detection only if it is supported */
+	if (priv->jack_detect > 0) {
+		hs_jack_gpios[0].gpio = priv->jack_detect;
+
+		ret = snd_soc_jack_new(codec, "Headset Jack", SND_JACK_HEADSET,
+				       &priv->hs_jack);
+		if (ret)
+			return ret;
+
+		ret = snd_soc_jack_add_pins(&priv->hs_jack,
+					    ARRAY_SIZE(hs_jack_pins),
+					    hs_jack_pins);
+		if (ret)
+			return ret;
+
+		ret = snd_soc_jack_add_gpios(&priv->hs_jack,
+					     ARRAY_SIZE(hs_jack_gpios),
+					     hs_jack_gpios);
+		if (ret)
+			return ret;
+	}
+
+	/*
+	 * NULL pdata means we booted with DT. In this case the routing is
+	 * provided and the card is fully routed, no need to mark pins.
+	 */
+	if (!pdata || !pdata->custom_routing)
+		return ret;
+
+	/* Disable not connected paths if not used */
+	twl4030_disconnect_pin(dapm, pdata->has_ear, "Earpiece Spk");
+	twl4030_disconnect_pin(dapm, pdata->has_hf, "Handsfree Spk");
+	twl4030_disconnect_pin(dapm, pdata->has_hs, "Headset Stereophone");
+	twl4030_disconnect_pin(dapm, pdata->has_predriv, "Ext Spk");
+	twl4030_disconnect_pin(dapm, pdata->has_carkit, "Carkit Spk");
+
+	twl4030_disconnect_pin(dapm, pdata->has_mainmic, "Main Mic");
+	twl4030_disconnect_pin(dapm, pdata->has_submic, "Sub Mic");
+	twl4030_disconnect_pin(dapm, pdata->has_hsmic, "Headset Mic");
+	twl4030_disconnect_pin(dapm, pdata->has_carkitmic, "Carkit Mic");
+	twl4030_disconnect_pin(dapm, pdata->has_digimic0, "Digital0 Mic");
+	twl4030_disconnect_pin(dapm, pdata->has_digimic1, "Digital1 Mic");
+	twl4030_disconnect_pin(dapm, pdata->has_linein, "Line In");
+
+	return ret;
+}
+
+static int omap_twl4030_card_remove(struct snd_soc_card *card)
+{
+	struct omap_twl4030 *priv = snd_soc_card_get_drvdata(card);
+
+	if (priv->jack_detect > 0)
+		snd_soc_jack_free_gpios(&priv->hs_jack,
+					ARRAY_SIZE(hs_jack_gpios),
+					hs_jack_gpios);
+
+	return 0;
+}
+
+/* Digital audio interface glue - connects codec <--> CPU */
+static struct snd_soc_dai_link omap_twl4030_dai_links[] = {
+	{
+		.name = "TWL4030 HiFi",
+		.stream_name = "TWL4030 HiFi",
+		.cpu_dai_name = "omap-mcbsp.2",
+		.codec_dai_name = "twl4030-hifi",
+		.platform_name = "omap-mcbsp.2",
+		.codec_name = "twl4030-codec",
+		.init = omap_twl4030_init,
+		.ops = &omap_twl4030_ops,
+	},
+	{
+		.name = "TWL4030 Voice",
+		.stream_name = "TWL4030 Voice",
+		.cpu_dai_name = "omap-mcbsp.3",
+		.codec_dai_name = "twl4030-voice",
+		.platform_name = "omap-mcbsp.3",
+		.codec_name = "twl4030-codec",
+		.dai_fmt = SND_SOC_DAIFMT_DSP_A | SND_SOC_DAIFMT_IB_NF |
+			   SND_SOC_DAIFMT_CBM_CFM,
+	},
+};
+
+/* Audio machine driver */
+static struct snd_soc_card omap_twl4030_card = {
+	.owner = THIS_MODULE,
+	.remove = omap_twl4030_card_remove,
+	.dai_link = omap_twl4030_dai_links,
+	.num_links = ARRAY_SIZE(omap_twl4030_dai_links),
+
+	.dapm_widgets = dapm_widgets,
+	.num_dapm_widgets = ARRAY_SIZE(dapm_widgets),
+	.dapm_routes = audio_map,
+	.num_dapm_routes = ARRAY_SIZE(audio_map),
+};
+
+static int omap_twl4030_probe(struct platform_device *pdev)
+{
+	struct omap_tw4030_pdata *pdata = dev_get_platdata(&pdev->dev);
+	struct device_node *node = pdev->dev.of_node;
+	struct snd_soc_card *card = &omap_twl4030_card;
+	struct omap_twl4030 *priv;
+	int ret = 0;
+
+	card->dev = &pdev->dev;
+
+	priv = devm_kzalloc(&pdev->dev, sizeof(struct omap_twl4030), GFP_KERNEL);
+	if (priv == NULL)
+		return -ENOMEM;
+
+	if (node) {
+		struct device_node *dai_node;
+		struct property *prop;
+
+		if (snd_soc_of_parse_card_name(card, "ti,model")) {
+			dev_err(&pdev->dev, "Card name is not provided\n");
+			return -ENODEV;
+		}
+
+		dai_node = of_parse_phandle(node, "ti,mcbsp", 0);
+		if (!dai_node) {
+			dev_err(&pdev->dev, "McBSP node is not provided\n");
+			return -EINVAL;
+		}
+		omap_twl4030_dai_links[0].cpu_dai_name  = NULL;
+		omap_twl4030_dai_links[0].cpu_of_node = dai_node;
+
+		omap_twl4030_dai_links[0].platform_name  = NULL;
+		omap_twl4030_dai_links[0].platform_of_node = dai_node;
+
+		dai_node = of_parse_phandle(node, "ti,mcbsp-voice", 0);
+		if (!dai_node) {
+			card->num_links = 1;
+		} else {
+			omap_twl4030_dai_links[1].cpu_dai_name  = NULL;
+			omap_twl4030_dai_links[1].cpu_of_node = dai_node;
+
+			omap_twl4030_dai_links[1].platform_name  = NULL;
+			omap_twl4030_dai_links[1].platform_of_node = dai_node;
+		}
+
+		priv->jack_detect = of_get_named_gpio(node,
+						      "ti,jack-det-gpio", 0);
+
+		/* Optional: audio routing can be provided */
+		prop = of_find_property(node, "ti,audio-routing", NULL);
+		if (prop) {
+			ret = snd_soc_of_parse_audio_routing(card,
+							    "ti,audio-routing");
+			if (ret)
+				return ret;
+
+			card->fully_routed = 1;
+		}
+	} else if (pdata) {
+		if (pdata->card_name) {
+			card->name = pdata->card_name;
+		} else {
+			dev_err(&pdev->dev, "Card name is not provided\n");
+			return -ENODEV;
+		}
+
+		if (!pdata->voice_connected)
+			card->num_links = 1;
+
+		priv->jack_detect = pdata->jack_detect;
+	} else {
+		dev_err(&pdev->dev, "Missing pdata\n");
+		return -ENODEV;
+	}
+
+	snd_soc_card_set_drvdata(card, priv);
+	ret = devm_snd_soc_register_card(&pdev->dev, card);
+	if (ret) {
+		dev_err(&pdev->dev, "devm_snd_soc_register_card() failed: %d\n",
+			ret);
+		return ret;
+	}
+
+	return 0;
+}
+
+static const struct of_device_id omap_twl4030_of_match[] = {
+	{.compatible = "ti,omap-twl4030", },
+	{ },
+};
+MODULE_DEVICE_TABLE(of, omap_twl4030_of_match);
+
+static struct platform_driver omap_twl4030_driver = {
+	.driver = {
+		.name = "omap-twl4030",
+		.owner = THIS_MODULE,
+		.pm = &snd_soc_pm_ops,
+		.of_match_table = omap_twl4030_of_match,
+	},
+	.probe = omap_twl4030_probe,
+};
+
+module_platform_driver(omap_twl4030_driver);
+
+MODULE_AUTHOR("Peter Ujfalusi <peter.ujfalusi@ti.com>");
+MODULE_DESCRIPTION("ALSA SoC for TI SoC based boards with twl4030 codec");
+MODULE_LICENSE("GPL");
+MODULE_ALIAS("platform:omap-twl4030");

sound/soc/omap/omap3pandora.c

@@ -0,0 +1,317 @@
+/*
+ * omap3pandora.c  --  SoC audio for Pandora Handheld Console
+ *
+ * Author: Gražvydas Ignotas <notasas@gmail.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 <linux/gpio.h>
+#include <linux/delay.h>
+#include <linux/regulator/consumer.h>
+#include <linux/module.h>
+
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/soc.h>
+
+#include <asm/mach-types.h>
+#include <linux/platform_data/asoc-ti-mcbsp.h>
+
+#include "omap-mcbsp.h"
+
+#define OMAP3_PANDORA_DAC_POWER_GPIO	118
+#define OMAP3_PANDORA_AMP_POWER_GPIO	14
+
+#define PREFIX "ASoC omap3pandora: "
+
+static struct regulator *omap3pandora_dac_reg;
+
+static int omap3pandora_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->codec_dai;
+	struct snd_soc_dai *cpu_dai = rtd->cpu_dai;
+	int 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) {
+		pr_err(PREFIX "can't set codec system clock\n");
+		return ret;
+	}
+
+	/* Set McBSP clock to external */
+	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;
+	}
+
+	ret = snd_soc_dai_set_clkdiv(cpu_dai, OMAP_MCBSP_CLKGDV, 8);
+	if (ret < 0) {
+		pr_err(PREFIX "can't set SRG clock divider\n");
+		return ret;
+	}
+
+	return 0;
+}
+
+static int omap3pandora_dac_event(struct snd_soc_dapm_widget *w,
+	struct snd_kcontrol *k, int event)
+{
+	int ret;
+
+	/*
+	 * The PCM1773 DAC datasheet requires 1ms delay between switching
+	 * VCC power on/off and /PD pin high/low
+	 */
+	if (SND_SOC_DAPM_EVENT_ON(event)) {
+		ret = regulator_enable(omap3pandora_dac_reg);
+		if (ret) {
+			dev_err(w->dapm->dev, "Failed to power DAC: %d\n", ret);
+			return ret;
+		}
+		mdelay(1);
+		gpio_set_value(OMAP3_PANDORA_DAC_POWER_GPIO, 1);
+	} else {
+		gpio_set_value(OMAP3_PANDORA_DAC_POWER_GPIO, 0);
+		mdelay(1);
+		regulator_disable(omap3pandora_dac_reg);
+	}
+
+	return 0;
+}
+
+static int omap3pandora_hp_event(struct snd_soc_dapm_widget *w,
+	struct snd_kcontrol *k, int event)
+{
+	if (SND_SOC_DAPM_EVENT_ON(event))
+		gpio_set_value(OMAP3_PANDORA_AMP_POWER_GPIO, 1);
+	else
+		gpio_set_value(OMAP3_PANDORA_AMP_POWER_GPIO, 0);
+
+	return 0;
+}
+
+/*
+ * Audio paths on Pandora board:
+ *
+ *  |O| ---> PCM DAC +-> AMP -> Headphone Jack
+ *  |M|         A    +--------> Line Out
+ *  |A| <~~clk~~+
+ *  |P| <--- TWL4030 <--------- Line In and MICs
+ */
+static const struct snd_soc_dapm_widget omap3pandora_dapm_widgets[] = {
+	SND_SOC_DAPM_DAC_E("PCM DAC", "HiFi Playback", SND_SOC_NOPM,
+			   0, 0, omap3pandora_dac_event,
+			   SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD),
+	SND_SOC_DAPM_PGA_E("Headphone Amplifier", SND_SOC_NOPM,
+			   0, 0, NULL, 0, omap3pandora_hp_event,
+			   SND_SOC_DAPM_POST_PMU | SND_SOC_DAPM_PRE_PMD),
+	SND_SOC_DAPM_HP("Headphone Jack", NULL),
+	SND_SOC_DAPM_LINE("Line Out", NULL),
+
+	SND_SOC_DAPM_MIC("Mic (internal)", NULL),
+	SND_SOC_DAPM_MIC("Mic (external)", NULL),
+	SND_SOC_DAPM_LINE("Line In", NULL),
+};
+
+static const struct snd_soc_dapm_route omap3pandora_map[] = {
+	{"PCM DAC", NULL, "APLL Enable"},
+	{"Headphone Amplifier", NULL, "PCM DAC"},
+	{"Line Out", NULL, "PCM DAC"},
+	{"Headphone Jack", NULL, "Headphone Amplifier"},
+
+	{"AUXL", NULL, "Line In"},
+	{"AUXR", NULL, "Line In"},
+
+	{"MAINMIC", NULL, "Mic (internal)"},
+	{"Mic (internal)", NULL, "Mic Bias 1"},
+
+	{"SUBMIC", NULL, "Mic (external)"},
+	{"Mic (external)", NULL, "Mic Bias 2"},
+};
+
+static int omap3pandora_out_init(struct snd_soc_pcm_runtime *rtd)
+{
+	struct snd_soc_codec *codec = rtd->codec;
+	struct snd_soc_dapm_context *dapm = &codec->dapm;
+
+	/* All TWL4030 output pins are floating */
+	snd_soc_dapm_nc_pin(dapm, "EARPIECE");
+	snd_soc_dapm_nc_pin(dapm, "PREDRIVEL");
+	snd_soc_dapm_nc_pin(dapm, "PREDRIVER");
+	snd_soc_dapm_nc_pin(dapm, "HSOL");
+	snd_soc_dapm_nc_pin(dapm, "HSOR");
+	snd_soc_dapm_nc_pin(dapm, "CARKITL");
+	snd_soc_dapm_nc_pin(dapm, "CARKITR");
+	snd_soc_dapm_nc_pin(dapm, "HFL");
+	snd_soc_dapm_nc_pin(dapm, "HFR");
+	snd_soc_dapm_nc_pin(dapm, "VIBRA");
+
+	return 0;
+}
+
+static int omap3pandora_in_init(struct snd_soc_pcm_runtime *rtd)
+{
+	struct snd_soc_codec *codec = rtd->codec;
+	struct snd_soc_dapm_context *dapm = &codec->dapm;
+
+	/* Not comnnected */
+	snd_soc_dapm_nc_pin(dapm, "HSMIC");
+	snd_soc_dapm_nc_pin(dapm, "CARKITMIC");
+	snd_soc_dapm_nc_pin(dapm, "DIGIMIC0");
+	snd_soc_dapm_nc_pin(dapm, "DIGIMIC1");
+
+	return 0;
+}
+
+static struct snd_soc_ops omap3pandora_ops = {
+	.hw_params = omap3pandora_hw_params,
+};
+
+/* Digital audio interface glue - connects codec <--> CPU */
+static struct snd_soc_dai_link omap3pandora_dai[] = {
+	{
+		.name = "PCM1773",
+		.stream_name = "HiFi Out",
+		.cpu_dai_name = "omap-mcbsp.2",
+		.codec_dai_name = "twl4030-hifi",
+		.platform_name = "omap-mcbsp.2",
+		.codec_name = "twl4030-codec",
+		.dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF |
+			   SND_SOC_DAIFMT_CBS_CFS,
+		.ops = &omap3pandora_ops,
+		.init = omap3pandora_out_init,
+	}, {
+		.name = "TWL4030",
+		.stream_name = "Line/Mic In",
+		.cpu_dai_name = "omap-mcbsp.4",
+		.codec_dai_name = "twl4030-hifi",
+		.platform_name = "omap-mcbsp.4",
+		.codec_name = "twl4030-codec",
+		.dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF |
+			   SND_SOC_DAIFMT_CBS_CFS,
+		.ops = &omap3pandora_ops,
+		.init = omap3pandora_in_init,
+	}
+};
+
+/* SoC card */
+static struct snd_soc_card snd_soc_card_omap3pandora = {
+	.name = "omap3pandora",
+	.owner = THIS_MODULE,
+	.dai_link = omap3pandora_dai,
+	.num_links = ARRAY_SIZE(omap3pandora_dai),
+
+	.dapm_widgets = omap3pandora_dapm_widgets,
+	.num_dapm_widgets = ARRAY_SIZE(omap3pandora_dapm_widgets),
+	.dapm_routes = omap3pandora_map,
+	.num_dapm_routes = ARRAY_SIZE(omap3pandora_map),
+};
+
+static struct platform_device *omap3pandora_snd_device;
+
+static int __init omap3pandora_soc_init(void)
+{
+	int ret;
+
+	if (!machine_is_omap3_pandora())
+		return -ENODEV;
+
+	pr_info("OMAP3 Pandora SoC init\n");
+
+	ret = gpio_request(OMAP3_PANDORA_DAC_POWER_GPIO, "dac_power");
+	if (ret) {
+		pr_err(PREFIX "Failed to get DAC power GPIO\n");
+		return ret;
+	}
+
+	ret = gpio_direction_output(OMAP3_PANDORA_DAC_POWER_GPIO, 0);
+	if (ret) {
+		pr_err(PREFIX "Failed to set DAC power GPIO direction\n");
+		goto fail0;
+	}
+
+	ret = gpio_request(OMAP3_PANDORA_AMP_POWER_GPIO, "amp_power");
+	if (ret) {
+		pr_err(PREFIX "Failed to get amp power GPIO\n");
+		goto fail0;
+	}
+
+	ret = gpio_direction_output(OMAP3_PANDORA_AMP_POWER_GPIO, 0);
+	if (ret) {
+		pr_err(PREFIX "Failed to set amp power GPIO direction\n");
+		goto fail1;
+	}
+
+	omap3pandora_snd_device = platform_device_alloc("soc-audio", -1);
+	if (omap3pandora_snd_device == NULL) {
+		pr_err(PREFIX "Platform device allocation failed\n");
+		ret = -ENOMEM;
+		goto fail1;
+	}
+
+	platform_set_drvdata(omap3pandora_snd_device, &snd_soc_card_omap3pandora);
+
+	ret = platform_device_add(omap3pandora_snd_device);
+	if (ret) {
+		pr_err(PREFIX "Unable to add platform device\n");
+		goto fail2;
+	}
+
+	omap3pandora_dac_reg = regulator_get(&omap3pandora_snd_device->dev, "vcc");
+	if (IS_ERR(omap3pandora_dac_reg)) {
+		pr_err(PREFIX "Failed to get DAC regulator from %s: %ld\n",
+			dev_name(&omap3pandora_snd_device->dev),
+			PTR_ERR(omap3pandora_dac_reg));
+		ret = PTR_ERR(omap3pandora_dac_reg);
+		goto fail3;
+	}
+
+	return 0;
+
+fail3:
+	platform_device_del(omap3pandora_snd_device);
+fail2:
+	platform_device_put(omap3pandora_snd_device);
+fail1:
+	gpio_free(OMAP3_PANDORA_AMP_POWER_GPIO);
+fail0:
+	gpio_free(OMAP3_PANDORA_DAC_POWER_GPIO);
+	return ret;
+}
+module_init(omap3pandora_soc_init);
+
+static void __exit omap3pandora_soc_exit(void)
+{
+	regulator_put(omap3pandora_dac_reg);
+	platform_device_unregister(omap3pandora_snd_device);
+	gpio_free(OMAP3_PANDORA_AMP_POWER_GPIO);
+	gpio_free(OMAP3_PANDORA_DAC_POWER_GPIO);
+}
+module_exit(omap3pandora_soc_exit);
+
+MODULE_AUTHOR("Grazvydas Ignotas <notasas@gmail.com>");
+MODULE_DESCRIPTION("ALSA SoC OMAP3 Pandora");
+MODULE_LICENSE("GPL");

sound/soc/omap/osk5912.c

@@ -0,0 +1,187 @@
+/*
+ * 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 <asm/mach-types.h>
+#include <linux/gpio.h>
+#include <linux/module.h>
+#include <linux/platform_data/asoc-ti-mcbsp.h>
+
+#include "omap-mcbsp.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->codec_dai;
+	int 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"},
+};
+
+/* Digital audio interface glue - connects codec <--> CPU */
+static struct snd_soc_dai_link osk_dai = {
+	.name = "TLV320AIC23",
+	.stream_name = "AIC23",
+	.cpu_dai_name = "omap-mcbsp.1",
+	.codec_dai_name = "tlv320aic23-hifi",
+	.platform_name = "omap-mcbsp.1",
+	.codec_name = "tlv320aic23-codec",
+	.dai_fmt = SND_SOC_DAIFMT_DSP_B | SND_SOC_DAIFMT_NB_NF |
+		   SND_SOC_DAIFMT_CBM_CFM,
+	.ops = &osk_ops,
+};
+
+/* Audio machine driver */
+static struct snd_soc_card snd_soc_card_osk = {
+	.name = "OSK5912",
+	.owner = THIS_MODULE,
+	.dai_link = &osk_dai,
+	.num_links = 1,
+
+	.dapm_widgets = tlv320aic23_dapm_widgets,
+	.num_dapm_widgets = ARRAY_SIZE(tlv320aic23_dapm_widgets),
+	.dapm_routes = audio_map,
+	.num_dapm_routes = ARRAY_SIZE(audio_map),
+};
+
+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, &snd_soc_card_osk);
+	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");
+		err = PTR_ERR(tlv320aic23_mclk);
+		goto err2;
+	}
+
+	/*
+	 * 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 err3;
+		}
+	}
+
+	printk(KERN_INFO "MCLK = %d [%d]\n",
+	       (uint) clk_get_rate(tlv320aic23_mclk), CODEC_CLOCK);
+
+	return 0;
+
+err3:
+	clk_put(tlv320aic23_mclk);
+err2:
+	platform_device_del(osk_snd_device);
+err1:
+	platform_device_put(osk_snd_device);
+
+	return err;
+
+}
+
+static void __exit osk_soc_exit(void)
+{
+	clk_put(tlv320aic23_mclk);
+	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/omap/rx51.c

@@ -0,0 +1,533 @@
+/*
+ * rx51.c  --  SoC audio for Nokia RX-51
+ *
+ * Copyright (C) 2008 - 2009 Nokia Corporation
+ *
+ * Contact: Peter Ujfalusi <peter.ujfalusi@ti.com>
+ *          Eduardo Valentin <eduardo.valentin@nokia.com>
+ *          Jarkko Nikula <jarkko.nikula@bitmer.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/delay.h>
+#include <linux/gpio.h>
+#include <linux/platform_device.h>
+#include <linux/gpio/consumer.h>
+#include <linux/module.h>
+#include <sound/core.h>
+#include <sound/jack.h>
+#include <sound/pcm.h>
+#include <sound/soc.h>
+#include <linux/platform_data/asoc-ti-mcbsp.h>
+#include "../codecs/tpa6130a2.h"
+
+#include <asm/mach-types.h>
+
+#include "omap-mcbsp.h"
+
+enum {
+	RX51_JACK_DISABLED,
+	RX51_JACK_TVOUT,		/* tv-out with stereo output */
+	RX51_JACK_HP,			/* headphone: stereo output, no mic */
+	RX51_JACK_HS,			/* headset: stereo output with mic */
+};
+
+struct rx51_audio_pdata {
+	struct gpio_desc *tvout_selection_gpio;
+	struct gpio_desc *jack_detection_gpio;
+	struct gpio_desc *eci_sw_gpio;
+	struct gpio_desc *speaker_amp_gpio;
+};
+
+static int rx51_spk_func;
+static int rx51_dmic_func;
+static int rx51_jack_func;
+
+static void rx51_ext_control(struct snd_soc_dapm_context *dapm)
+{
+	struct snd_soc_card *card = dapm->card;
+	struct rx51_audio_pdata *pdata = snd_soc_card_get_drvdata(card);
+	int hp = 0, hs = 0, tvout = 0;
+
+	switch (rx51_jack_func) {
+	case RX51_JACK_TVOUT:
+		tvout = 1;
+		hp = 1;
+		break;
+	case RX51_JACK_HS:
+		hs = 1;
+	case RX51_JACK_HP:
+		hp = 1;
+		break;
+	}
+
+	snd_soc_dapm_mutex_lock(dapm);
+
+	if (rx51_spk_func)
+		snd_soc_dapm_enable_pin_unlocked(dapm, "Ext Spk");
+	else
+		snd_soc_dapm_disable_pin_unlocked(dapm, "Ext Spk");
+	if (rx51_dmic_func)
+		snd_soc_dapm_enable_pin_unlocked(dapm, "DMic");
+	else
+		snd_soc_dapm_disable_pin_unlocked(dapm, "DMic");
+	if (hp)
+		snd_soc_dapm_enable_pin_unlocked(dapm, "Headphone Jack");
+	else
+		snd_soc_dapm_disable_pin_unlocked(dapm, "Headphone Jack");
+	if (hs)
+		snd_soc_dapm_enable_pin_unlocked(dapm, "HS Mic");
+	else
+		snd_soc_dapm_disable_pin_unlocked(dapm, "HS Mic");
+
+	gpiod_set_value(pdata->tvout_selection_gpio, tvout);
+
+	snd_soc_dapm_sync_unlocked(dapm);
+
+	snd_soc_dapm_mutex_unlock(dapm);
+}
+
+static int rx51_startup(struct snd_pcm_substream *substream)
+{
+	struct snd_pcm_runtime *runtime = substream->runtime;
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_card *card = rtd->card;
+
+	snd_pcm_hw_constraint_minmax(runtime,
+				     SNDRV_PCM_HW_PARAM_CHANNELS, 2, 2);
+	rx51_ext_control(&card->dapm);
+
+	return 0;
+}
+
+static int rx51_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->codec_dai;
+
+	/* Set the codec system clock for DAC and ADC */
+	return snd_soc_dai_set_sysclk(codec_dai, 0, 19200000,
+				      SND_SOC_CLOCK_IN);
+}
+
+static struct snd_soc_ops rx51_ops = {
+	.startup = rx51_startup,
+	.hw_params = rx51_hw_params,
+};
+
+static int rx51_get_spk(struct snd_kcontrol *kcontrol,
+			struct snd_ctl_elem_value *ucontrol)
+{
+	ucontrol->value.integer.value[0] = rx51_spk_func;
+
+	return 0;
+}
+
+static int rx51_set_spk(struct snd_kcontrol *kcontrol,
+			struct snd_ctl_elem_value *ucontrol)
+{
+	struct snd_soc_card *card = snd_kcontrol_chip(kcontrol);
+
+	if (rx51_spk_func == ucontrol->value.integer.value[0])
+		return 0;
+
+	rx51_spk_func = ucontrol->value.integer.value[0];
+	rx51_ext_control(&card->dapm);
+
+	return 1;
+}
+
+static int rx51_spk_event(struct snd_soc_dapm_widget *w,
+			  struct snd_kcontrol *k, int event)
+{
+	struct snd_soc_dapm_context *dapm = w->dapm;
+	struct snd_soc_card *card = dapm->card;
+	struct rx51_audio_pdata *pdata = snd_soc_card_get_drvdata(card);
+
+	gpiod_set_raw_value_cansleep(pdata->speaker_amp_gpio,
+				     !!SND_SOC_DAPM_EVENT_ON(event));
+
+	return 0;
+}
+
+static int rx51_hp_event(struct snd_soc_dapm_widget *w,
+			 struct snd_kcontrol *k, int event)
+{
+	struct snd_soc_codec *codec = snd_soc_dapm_to_codec(w->dapm);
+
+	if (SND_SOC_DAPM_EVENT_ON(event))
+		tpa6130a2_stereo_enable(codec, 1);
+	else
+		tpa6130a2_stereo_enable(codec, 0);
+
+	return 0;
+}
+
+static int rx51_get_input(struct snd_kcontrol *kcontrol,
+			  struct snd_ctl_elem_value *ucontrol)
+{
+	ucontrol->value.integer.value[0] = rx51_dmic_func;
+
+	return 0;
+}
+
+static int rx51_set_input(struct snd_kcontrol *kcontrol,
+			  struct snd_ctl_elem_value *ucontrol)
+{
+	struct snd_soc_card *card = snd_kcontrol_chip(kcontrol);
+
+	if (rx51_dmic_func == ucontrol->value.integer.value[0])
+		return 0;
+
+	rx51_dmic_func = ucontrol->value.integer.value[0];
+	rx51_ext_control(&card->dapm);
+
+	return 1;
+}
+
+static int rx51_get_jack(struct snd_kcontrol *kcontrol,
+			 struct snd_ctl_elem_value *ucontrol)
+{
+	ucontrol->value.integer.value[0] = rx51_jack_func;
+
+	return 0;
+}
+
+static int rx51_set_jack(struct snd_kcontrol *kcontrol,
+			 struct snd_ctl_elem_value *ucontrol)
+{
+	struct snd_soc_card *card = snd_kcontrol_chip(kcontrol);
+
+	if (rx51_jack_func == ucontrol->value.integer.value[0])
+		return 0;
+
+	rx51_jack_func = ucontrol->value.integer.value[0];
+	rx51_ext_control(&card->dapm);
+
+	return 1;
+}
+
+static struct snd_soc_jack rx51_av_jack;
+
+static struct snd_soc_jack_gpio rx51_av_jack_gpios[] = {
+	{
+		.name = "avdet-gpio",
+		.report = SND_JACK_HEADSET,
+		.invert = 1,
+		.debounce_time = 200,
+	},
+};
+
+static const struct snd_soc_dapm_widget aic34_dapm_widgets[] = {
+	SND_SOC_DAPM_SPK("Ext Spk", rx51_spk_event),
+	SND_SOC_DAPM_MIC("DMic", NULL),
+	SND_SOC_DAPM_HP("Headphone Jack", rx51_hp_event),
+	SND_SOC_DAPM_MIC("HS Mic", NULL),
+	SND_SOC_DAPM_LINE("FM Transmitter", NULL),
+	SND_SOC_DAPM_SPK("Earphone", NULL),
+};
+
+static const struct snd_soc_dapm_route audio_map[] = {
+	{"Ext Spk", NULL, "HPLOUT"},
+	{"Ext Spk", NULL, "HPROUT"},
+	{"Headphone Jack", NULL, "LLOUT"},
+	{"Headphone Jack", NULL, "RLOUT"},
+	{"FM Transmitter", NULL, "LLOUT"},
+	{"FM Transmitter", NULL, "RLOUT"},
+
+	{"DMic Rate 64", NULL, "Mic Bias"},
+	{"Mic Bias", NULL, "DMic"},
+
+	{"b LINE2R", NULL, "MONO_LOUT"},
+	{"Earphone", NULL, "b HPLOUT"},
+
+	{"LINE1L", NULL, "b Mic Bias"},
+	{"b Mic Bias", NULL, "HS Mic"}
+};
+
+static const char * const spk_function[] = {"Off", "On"};
+static const char * const input_function[] = {"ADC", "Digital Mic"};
+static const char * const jack_function[] = {
+	"Off", "TV-OUT", "Headphone", "Headset"
+};
+
+static const struct soc_enum rx51_enum[] = {
+	SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(spk_function), spk_function),
+	SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(input_function), input_function),
+	SOC_ENUM_SINGLE_EXT(ARRAY_SIZE(jack_function), jack_function),
+};
+
+static const struct snd_kcontrol_new aic34_rx51_controls[] = {
+	SOC_ENUM_EXT("Speaker Function", rx51_enum[0],
+		     rx51_get_spk, rx51_set_spk),
+	SOC_ENUM_EXT("Input Select",  rx51_enum[1],
+		     rx51_get_input, rx51_set_input),
+	SOC_ENUM_EXT("Jack Function", rx51_enum[2],
+		     rx51_get_jack, rx51_set_jack),
+	SOC_DAPM_PIN_SWITCH("FM Transmitter"),
+	SOC_DAPM_PIN_SWITCH("Earphone"),
+};
+
+static int rx51_aic34_init(struct snd_soc_pcm_runtime *rtd)
+{
+	struct snd_soc_codec *codec = rtd->codec;
+	struct snd_soc_card *card = rtd->card;
+	struct rx51_audio_pdata *pdata = snd_soc_card_get_drvdata(card);
+
+	struct snd_soc_dapm_context *dapm = &codec->dapm;
+	int err;
+
+	/* Set up NC codec pins */
+	snd_soc_dapm_nc_pin(dapm, "MIC3L");
+	snd_soc_dapm_nc_pin(dapm, "MIC3R");
+	snd_soc_dapm_nc_pin(dapm, "LINE1R");
+
+	err = tpa6130a2_add_controls(codec);
+	if (err < 0) {
+		dev_err(card->dev, "Failed to add TPA6130A2 controls\n");
+		return err;
+	}
+	snd_soc_limit_volume(codec, "TPA6130A2 Headphone Playback Volume", 42);
+
+	err = omap_mcbsp_st_add_controls(rtd, 2);
+	if (err < 0) {
+		dev_err(card->dev, "Failed to add MCBSP controls\n");
+		return err;
+	}
+
+	/* AV jack detection */
+	err = snd_soc_jack_new(codec, "AV Jack",
+			       SND_JACK_HEADSET | SND_JACK_VIDEOOUT,
+			       &rx51_av_jack);
+	if (err) {
+		dev_err(card->dev, "Failed to add AV Jack\n");
+		return err;
+	}
+
+	/* prepare gpio for snd_soc_jack_add_gpios */
+	rx51_av_jack_gpios[0].gpio = desc_to_gpio(pdata->jack_detection_gpio);
+	devm_gpiod_put(card->dev, pdata->jack_detection_gpio);
+
+	err = snd_soc_jack_add_gpios(&rx51_av_jack,
+				     ARRAY_SIZE(rx51_av_jack_gpios),
+				     rx51_av_jack_gpios);
+	if (err) {
+		dev_err(card->dev, "Failed to add GPIOs\n");
+		return err;
+	}
+
+	return err;
+}
+
+static int rx51_card_remove(struct snd_soc_card *card)
+{
+	snd_soc_jack_free_gpios(&rx51_av_jack, ARRAY_SIZE(rx51_av_jack_gpios),
+				rx51_av_jack_gpios);
+
+	return 0;
+}
+
+/* Digital audio interface glue - connects codec <--> CPU */
+static struct snd_soc_dai_link rx51_dai[] = {
+	{
+		.name = "TLV320AIC34",
+		.stream_name = "AIC34",
+		.cpu_dai_name = "omap-mcbsp.2",
+		.codec_dai_name = "tlv320aic3x-hifi",
+		.platform_name = "omap-mcbsp.2",
+		.codec_name = "tlv320aic3x-codec.2-0018",
+		.dai_fmt = SND_SOC_DAIFMT_DSP_A | SND_SOC_DAIFMT_IB_NF |
+			   SND_SOC_DAIFMT_CBM_CFM,
+		.init = rx51_aic34_init,
+		.ops = &rx51_ops,
+	},
+};
+
+static struct snd_soc_aux_dev rx51_aux_dev[] = {
+	{
+		.name = "TLV320AIC34b",
+		.codec_name = "tlv320aic3x-codec.2-0019",
+	},
+};
+
+static struct snd_soc_codec_conf rx51_codec_conf[] = {
+	{
+		.dev_name = "tlv320aic3x-codec.2-0019",
+		.name_prefix = "b",
+	},
+};
+
+/* Audio card */
+static struct snd_soc_card rx51_sound_card = {
+	.name = "RX-51",
+	.owner = THIS_MODULE,
+	.remove = rx51_card_remove,
+	.dai_link = rx51_dai,
+	.num_links = ARRAY_SIZE(rx51_dai),
+	.aux_dev = rx51_aux_dev,
+	.num_aux_devs = ARRAY_SIZE(rx51_aux_dev),
+	.codec_conf = rx51_codec_conf,
+	.num_configs = ARRAY_SIZE(rx51_codec_conf),
+
+	.controls = aic34_rx51_controls,
+	.num_controls = ARRAY_SIZE(aic34_rx51_controls),
+	.dapm_widgets = aic34_dapm_widgets,
+	.num_dapm_widgets = ARRAY_SIZE(aic34_dapm_widgets),
+	.dapm_routes = audio_map,
+	.num_dapm_routes = ARRAY_SIZE(audio_map),
+};
+
+static int rx51_soc_probe(struct platform_device *pdev)
+{
+	struct rx51_audio_pdata *pdata;
+	struct device_node *np = pdev->dev.of_node;
+	struct snd_soc_card *card = &rx51_sound_card;
+	int err;
+
+	if (!machine_is_nokia_rx51() && !of_machine_is_compatible("nokia,omap3-n900"))
+		return -ENODEV;
+
+	card->dev = &pdev->dev;
+
+	if (np) {
+		struct device_node *dai_node;
+
+		dai_node = of_parse_phandle(np, "nokia,cpu-dai", 0);
+		if (!dai_node) {
+			dev_err(&pdev->dev, "McBSP node is not provided\n");
+			return -EINVAL;
+		}
+		rx51_dai[0].cpu_dai_name = NULL;
+		rx51_dai[0].platform_name = NULL;
+		rx51_dai[0].cpu_of_node = dai_node;
+		rx51_dai[0].platform_of_node = dai_node;
+
+		dai_node = of_parse_phandle(np, "nokia,audio-codec", 0);
+		if (!dai_node) {
+			dev_err(&pdev->dev, "Codec node is not provided\n");
+			return -EINVAL;
+		}
+		rx51_dai[0].codec_name = NULL;
+		rx51_dai[0].codec_of_node = dai_node;
+
+		dai_node = of_parse_phandle(np, "nokia,audio-codec", 1);
+		if (!dai_node) {
+			dev_err(&pdev->dev, "Auxiliary Codec node is not provided\n");
+			return -EINVAL;
+		}
+		rx51_aux_dev[0].codec_name = NULL;
+		rx51_aux_dev[0].codec_of_node = dai_node;
+		rx51_codec_conf[0].dev_name = NULL;
+		rx51_codec_conf[0].of_node = dai_node;
+
+		dai_node = of_parse_phandle(np, "nokia,headphone-amplifier", 0);
+		if (!dai_node) {
+			dev_err(&pdev->dev, "Headphone amplifier node is not provided\n");
+			return -EINVAL;
+		}
+
+		/* TODO: tpa6130a2a driver supports only a single instance, so
+		 * this driver ignores the headphone-amplifier node for now.
+		 * It's already mandatory in the DT binding to be future proof.
+		 */
+	}
+
+	pdata = devm_kzalloc(&pdev->dev, sizeof(*pdata), GFP_KERNEL);
+	if (pdata == NULL) {
+		dev_err(card->dev, "failed to create private data\n");
+		return -ENOMEM;
+	}
+	snd_soc_card_set_drvdata(card, pdata);
+
+	pdata->tvout_selection_gpio = devm_gpiod_get(card->dev,
+						     "tvout-selection");
+	if (IS_ERR(pdata->tvout_selection_gpio)) {
+		dev_err(card->dev, "could not get tvout selection gpio\n");
+		return PTR_ERR(pdata->tvout_selection_gpio);
+	}
+
+	err = gpiod_direction_output(pdata->tvout_selection_gpio, 0);
+	if (err) {
+		dev_err(card->dev, "could not setup tvout selection gpio\n");
+		return err;
+	}
+
+	pdata->jack_detection_gpio = devm_gpiod_get(card->dev,
+						    "jack-detection");
+	if (IS_ERR(pdata->jack_detection_gpio)) {
+		dev_err(card->dev, "could not get jack detection gpio\n");
+		return PTR_ERR(pdata->jack_detection_gpio);
+	}
+
+	pdata->eci_sw_gpio = devm_gpiod_get(card->dev, "eci-switch");
+	if (IS_ERR(pdata->eci_sw_gpio)) {
+		dev_err(card->dev, "could not get eci switch gpio\n");
+		return PTR_ERR(pdata->eci_sw_gpio);
+	}
+
+	err = gpiod_direction_output(pdata->eci_sw_gpio, 1);
+	if (err) {
+		dev_err(card->dev, "could not setup eci switch gpio\n");
+		return err;
+	}
+
+	pdata->speaker_amp_gpio = devm_gpiod_get(card->dev,
+						 "speaker-amplifier");
+	if (IS_ERR(pdata->speaker_amp_gpio)) {
+		dev_err(card->dev, "could not get speaker enable gpio\n");
+		return PTR_ERR(pdata->speaker_amp_gpio);
+	}
+
+	err = gpiod_direction_output(pdata->speaker_amp_gpio, 0);
+	if (err) {
+		dev_err(card->dev, "could not setup speaker enable gpio\n");
+		return err;
+	}
+
+	err = devm_snd_soc_register_card(card->dev, card);
+	if (err) {
+		dev_err(&pdev->dev, "snd_soc_register_card failed (%d)\n", err);
+		return err;
+	}
+
+	return 0;
+}
+
+#if defined(CONFIG_OF)
+static const struct of_device_id rx51_audio_of_match[] = {
+	{ .compatible = "nokia,n900-audio", },
+	{},
+};
+MODULE_DEVICE_TABLE(of, rx51_audio_of_match);
+#endif
+
+static struct platform_driver rx51_soc_driver = {
+	.driver = {
+		.name = "rx51-audio",
+		.owner = THIS_MODULE,
+		.of_match_table = of_match_ptr(rx51_audio_of_match),
+	},
+	.probe = rx51_soc_probe,
+};
+
+module_platform_driver(rx51_soc_driver);
+
+MODULE_AUTHOR("Nokia Corporation");
+MODULE_DESCRIPTION("ALSA SoC Nokia RX-51");
+MODULE_LICENSE("GPL");
+MODULE_ALIAS("platform:rx51-audio");