Adding upstream version 6.1.76.upstream/6.1.76

Signed-off-by: Daniel Baumann <daniel.baumann@progress-linux.org>

31 files changed, 12537 insertions, 0 deletions

diff --git a/sound/soc/ti/Kconfig b/sound/soc/ti/Kconfig
new file mode 100644
index 000000000..40110e9a9
--- /dev/null
+++ b/sound/soc/ti/Kconfig
@@ -0,0 +1,234 @@
+# SPDX-License-Identifier: GPL-2.0-only
+menu "Audio support for Texas Instruments SoCs"
+depends on DMA_OMAP || TI_EDMA || TI_K3_UDMA || COMPILE_TEST
+
+config SND_SOC_TI_EDMA_PCM
+	tristate
+	select SND_SOC_GENERIC_DMAENGINE_PCM
+
+config SND_SOC_TI_SDMA_PCM
+	tristate
+	select SND_SOC_GENERIC_DMAENGINE_PCM
+
+config SND_SOC_TI_UDMA_PCM
+	tristate
+	select SND_SOC_GENERIC_DMAENGINE_PCM
+
+comment "Texas Instruments DAI support for:"
+config SND_SOC_DAVINCI_ASP
+	tristate "daVinci Audio Serial Port (ASP) or McBSP support"
+	depends on ARCH_DAVINCI || COMPILE_TEST
+	select SND_SOC_TI_EDMA_PCM
+	help
+	  Say Y or M here if you want audio support via daVinci ASP or McBSP.
+	  The driver only implements the ASP support which is a subset of
+	  daVinci McBSP (w/o the multichannel support).
+
+config SND_SOC_DAVINCI_MCASP
+	tristate "Multichannel Audio Serial Port (McASP) support"
+	depends on COMMON_CLK
+	select SND_SOC_TI_EDMA_PCM
+	select SND_SOC_TI_SDMA_PCM
+	select SND_SOC_TI_UDMA_PCM
+	help
+	  Say Y or M here if you want to have support for McASP IP found in
+	  various Texas Instruments SoCs like:
+	  - daVinci devices
+	  - Sitara line of SoCs (AM335x, AM438x, etc)
+	  - OMAP4
+	  - DRA7x devices
+	  - Keystone devices
+	  - K3 devices (am654, j721e)
+
+config SND_SOC_DAVINCI_VCIF
+	tristate "daVinci Voice Interface (VCIF) support"
+	depends on ARCH_DAVINCI || COMPILE_TEST
+	select SND_SOC_TI_EDMA_PCM
+	help
+	  Say Y or M here if you want audio support via daVinci VCIF.
+
+config SND_SOC_OMAP_DMIC
+	tristate "Digital Microphone Module (DMIC) support"
+	depends on ARCH_OMAP4 || SOC_OMAP5 || COMPILE_TEST && COMMON_CLK
+	select SND_SOC_TI_SDMA_PCM
+	help
+	  Say Y or M here if you want to have support for DMIC IP found in
+	  OMAP4 and OMAP5.
+
+config SND_SOC_OMAP_MCBSP
+	tristate "Multichannel Buffered Serial Port (McBSP) support"
+	depends on ARCH_OMAP || ARCH_OMAP1 || COMPILE_TEST && COMMON_CLK
+	select SND_SOC_TI_SDMA_PCM
+	help
+	  Say Y or M here if you want to have support for McBSP IP found in
+	  Texas Instruments OMAP1/2/3/4/5 SoCs.
+
+config SND_SOC_OMAP_MCPDM
+	tristate "Multichannel PDM Controller (McPDM) support"
+	depends on ARCH_OMAP4 || SOC_OMAP5 || COMPILE_TEST
+	select SND_SOC_TI_SDMA_PCM
+	help
+	  Say Y or M here if you want to have support for McPDM IP found in
+	  OMAP4 and OMAP5.
+
+comment "Audio support for boards with Texas Instruments SoCs"
+config SND_SOC_NOKIA_N810
+	tristate "SoC Audio support for Nokia N810"
+	depends on MACH_NOKIA_N810 && I2C
+	select SND_SOC_OMAP_MCBSP
+	select SND_SOC_TLV320AIC3X
+	help
+	  Say Y or M if you want to add support for SoC audio on Nokia N810.
+
+config SND_SOC_NOKIA_RX51
+	tristate "SoC Audio support for Nokia RX-51"
+	depends on ARCH_OMAP3 && I2C && GPIOLIB
+	select SND_SOC_OMAP_MCBSP
+	select SND_SOC_TLV320AIC3X
+	select SND_SOC_TPA6130A2
+	help
+	  Say Y or M if you want to add support for SoC audio on Nokia RX-51
+	  hardware. This is also known as Nokia N900 product.
+
+config SND_SOC_OMAP3_PANDORA
+	tristate "SoC Audio support for OMAP3 Pandora"
+	depends on ARCH_OMAP3
+	depends on TWL4030_CORE
+	select SND_SOC_OMAP_MCBSP
+	select SND_SOC_TWL4030
+	help
+	  Say Y or M if you want to add support for SoC audio on the OMAP3 Pandora.
+
+config SND_SOC_OMAP3_TWL4030
+	tristate "SoC Audio support for OMAP3 based boards with twl4030 codec"
+	depends on ARCH_OMAP3 || COMPILE_TEST && COMMON_CLK
+	depends on TWL4030_CORE
+	select SND_SOC_OMAP_MCBSP
+	select SND_SOC_TWL4030
+	help
+	  Say Y or M if you want to add support for SoC audio on OMAP3 based
+	  boards using twl4030 as codec. This driver currently supports:
+	  - Beagleboard or Devkit8000
+	  - Gumstix Overo or CompuLab CM-T35/CM-T3730
+	  - IGEP v2
+	  - OMAP3EVM
+	  - SDP3430
+	  - Zoom2
+
+config SND_SOC_OMAP_ABE_TWL6040
+	tristate "SoC Audio support for OMAP boards using ABE and twl6040 codec"
+	depends on TWL6040_CORE && COMMON_CLK
+	depends on ARCH_OMAP4 || (SOC_OMAP5 && MFD_PALMAS) || COMPILE_TEST
+	select SND_SOC_OMAP_DMIC
+	select SND_SOC_OMAP_MCPDM
+	select SND_SOC_TWL6040
+	help
+	  Say Y or M 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)
+	  - OMAP5 uEVM
+
+config SND_SOC_OMAP_AMS_DELTA
+	tristate "SoC Audio support for Amstrad E3 (Delta) videophone"
+	depends on MACH_AMS_DELTA && TTY
+	select SND_SOC_OMAP_MCBSP
+	select SND_SOC_CX20442
+	help
+	  Say Y  or M 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_SOC_OMAP_HDMI
+	tristate "OMAP4/5 HDMI audio support"
+	depends on OMAP4_DSS_HDMI || OMAP5_DSS_HDMI || COMPILE_TEST
+	select SND_SOC_TI_SDMA_PCM
+	help
+	  For HDMI audio to work OMAPDSS HDMI support should be
+	  enabled.
+	  The hdmi audio driver implements cpu-dai component using the
+	  callbacks provided by OMAPDSS and registers the component
+	  under DSS HDMI device. Omap-pcm is registered for platform
+	  component also under DSS HDMI device. Dummy codec is used as
+	  as codec component. The hdmi audio driver implements also
+	  the card and registers it under its own platform device.
+	  The device for the driver is registered by OMAPDSS hdmi
+	  driver.
+
+config SND_SOC_OMAP_OSK5912
+	tristate "SoC Audio support for omap osk5912"
+	depends on MACH_OMAP_OSK && I2C
+	select SND_SOC_OMAP_MCBSP
+	select SND_SOC_TLV320AIC23_I2C
+	help
+	  Say Y or M if you want to add support for SoC audio on osk5912.
+
+config SND_SOC_DAVINCI_EVM
+	tristate "SoC Audio support for DaVinci EVMs"
+	depends on ARCH_DAVINCI && I2C
+	select SND_SOC_DAVINCI_ASP if MACH_DAVINCI_DM355_EVM
+	select SND_SOC_DAVINCI_ASP if SND_SOC_DM365_AIC3X_CODEC
+	select SND_SOC_DAVINCI_VCIF if SND_SOC_DM365_VOICE_CODEC
+	select SND_SOC_DAVINCI_ASP if MACH_DAVINCI_EVM # DM6446
+	select SND_SOC_DAVINCI_MCASP if MACH_DAVINCI_DM6467_EVM
+	select SND_SOC_SPDIF if MACH_DAVINCI_DM6467_EVM
+	select SND_SOC_DAVINCI_MCASP if MACH_DAVINCI_DA830_EVM
+	select SND_SOC_DAVINCI_MCASP if MACH_DAVINCI_DA850_EVM
+	select SND_SOC_TLV320AIC3X
+	help
+	  Say Y if you want to add support for SoC audio on the following TI
+	  DaVinci EVM platforms:
+	  - DM355
+	  - DM365
+	  - DM6446
+	  - DM6447
+	  - DM830
+	  - DM850
+
+choice
+	prompt "DM365 codec select"
+	depends on SND_SOC_DAVINCI_EVM
+	depends on MACH_DAVINCI_DM365_EVM
+
+config SND_SOC_DM365_AIC3X_CODEC
+	bool "Audio Codec - AIC3101"
+	help
+	  Say Y if you want to add support for AIC3101 audio codec
+
+config SND_SOC_DM365_VOICE_CODEC
+	bool "Voice Codec - CQ93VC"
+	help
+	  Say Y if you want to add support for SoC On-chip voice codec
+endchoice
+
+config SND_SOC_DM365_SELECT_VOICE_CODECS
+	def_tristate y
+	depends on SND_SOC_DM365_VOICE_CODEC && SND_SOC
+	select MFD_DAVINCI_VOICECODEC
+	select SND_SOC_CQ0093VC
+	help
+	  The is an internal symbol needed to ensure that the codec
+	  and MFD driver can be built as loadable modules if necessary.
+
+config SND_SOC_J721E_EVM
+	tristate "SoC Audio support for j721e EVM"
+	depends on ARCH_K3 || COMPILE_TEST && COMMON_CLK
+	depends on I2C
+	select SND_SOC_PCM3168A_I2C
+	select SND_SOC_DAVINCI_MCASP
+	help
+	  Say Y if you want to add support for SoC audio on j721e Common
+	  Processor Board and Infotainment expansion board.
+endmenu
+
diff --git a/sound/soc/ti/Makefile b/sound/soc/ti/Makefile
new file mode 100644
index 000000000..a21e5b006
--- /dev/null
+++ b/sound/soc/ti/Makefile
@@ -0,0 +1,48 @@
+# SPDX-License-Identifier: GPL-2.0
+
+# Platform drivers
+snd-soc-ti-edma-objs := edma-pcm.o
+snd-soc-ti-sdma-objs := sdma-pcm.o
+snd-soc-ti-udma-objs := udma-pcm.o
+
+obj-$(CONFIG_SND_SOC_TI_EDMA_PCM) += snd-soc-ti-edma.o
+obj-$(CONFIG_SND_SOC_TI_SDMA_PCM) += snd-soc-ti-sdma.o
+obj-$(CONFIG_SND_SOC_TI_UDMA_PCM) += snd-soc-ti-udma.o
+
+# CPU DAI drivers
+snd-soc-davinci-asp-objs := davinci-i2s.o
+snd-soc-davinci-mcasp-objs := davinci-mcasp.o
+snd-soc-davinci-vcif-objs := davinci-vcif.o
+snd-soc-omap-dmic-objs := omap-dmic.o
+snd-soc-omap-mcbsp-objs := omap-mcbsp.o omap-mcbsp-st.o
+snd-soc-omap-mcpdm-objs := omap-mcpdm.o
+
+obj-$(CONFIG_SND_SOC_DAVINCI_ASP) += snd-soc-davinci-asp.o
+obj-$(CONFIG_SND_SOC_DAVINCI_MCASP) += snd-soc-davinci-mcasp.o
+obj-$(CONFIG_SND_SOC_DAVINCI_VCIF) += snd-soc-davinci-vcif.o
+obj-$(CONFIG_SND_SOC_OMAP_DMIC) += snd-soc-omap-dmic.o
+obj-$(CONFIG_SND_SOC_OMAP_MCBSP) += snd-soc-omap-mcbsp.o
+obj-$(CONFIG_SND_SOC_OMAP_MCPDM) += snd-soc-omap-mcpdm.o
+
+# Machine drivers
+snd-soc-davinci-evm-objs := davinci-evm.o
+snd-soc-n810-objs := n810.o
+snd-soc-rx51-objs := rx51.o
+snd-soc-omap3pandora-objs := omap3pandora.o
+snd-soc-omap-twl4030-objs := omap-twl4030.o
+snd-soc-omap-abe-twl6040-objs := omap-abe-twl6040.o
+snd-soc-ams-delta-objs := ams-delta.o
+snd-soc-omap-hdmi-objs := omap-hdmi.o
+snd-soc-osk5912-objs := osk5912.o
+snd-soc-j721e-evm-objs := j721e-evm.o
+
+obj-$(CONFIG_SND_SOC_DAVINCI_EVM) += snd-soc-davinci-evm.o
+obj-$(CONFIG_SND_SOC_NOKIA_N810) += snd-soc-n810.o
+obj-$(CONFIG_SND_SOC_NOKIA_RX51) += snd-soc-rx51.o
+obj-$(CONFIG_SND_SOC_OMAP3_PANDORA) += snd-soc-omap3pandora.o
+obj-$(CONFIG_SND_SOC_OMAP3_TWL4030) += snd-soc-omap-twl4030.o
+obj-$(CONFIG_SND_SOC_OMAP_ABE_TWL6040) += snd-soc-omap-abe-twl6040.o
+obj-$(CONFIG_SND_SOC_OMAP_AMS_DELTA) += snd-soc-ams-delta.o
+obj-$(CONFIG_SND_SOC_OMAP_HDMI) += snd-soc-omap-hdmi.o
+obj-$(CONFIG_SND_SOC_OMAP_OSK5912) += snd-soc-osk5912.o
+obj-$(CONFIG_SND_SOC_J721E_EVM) += snd-soc-j721e-evm.o
diff --git a/sound/soc/ti/ams-delta.c b/sound/soc/ti/ams-delta.c
new file mode 100644
index 000000000..1acc4505a
--- /dev/null
+++ b/sound/soc/ti/ams-delta.c
@@ -0,0 +1,609 @@
+// SPDX-License-Identifier: GPL-2.0-only
+/*
+ * 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
+ */
+
+#include <linux/gpio/consumer.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 <linux/platform_data/asoc-ti-mcbsp.h>
+
+#include "omap-mcbsp.h"
+#include "../codecs/cx20442.h"
+
+static struct gpio_desc *handset_mute;
+static struct gpio_desc *handsfree_mute;
+
+static int ams_delta_event_handset(struct snd_soc_dapm_widget *w,
+				   struct snd_kcontrol *k, int event)
+{
+	gpiod_set_value_cansleep(handset_mute, !SND_SOC_DAPM_EVENT_ON(event));
+	return 0;
+}
+
+static int ams_delta_event_handsfree(struct snd_soc_dapm_widget *w,
+				     struct snd_kcontrol *k, int event)
+{
+	gpiod_set_value_cansleep(handsfree_mute, !SND_SOC_DAPM_EVENT_ON(event));
+	return 0;
+}
+
+/* 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", ams_delta_event_handset),
+	/* Handsfree/Speakerphone */
+	SND_SOC_DAPM_MIC("Microphone", NULL),
+	SND_SOC_DAPM_SPK("Speaker", ams_delta_event_handsfree),
+};
+
+/* 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_component *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->card->pop_time)
+		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 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[] = {
+	{
+		.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 struct gpio_desc *gpiod_modem_codec;
+
+static void cx81801_timeout(struct timer_list *unused)
+{
+	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)
+		gpiod_set_value(gpiod_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_component *component = tty->disc_data;
+	struct snd_soc_dapm_context *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 (!component)
+		return;
+
+	v253_ops.close(tty);
+
+	dapm = &component->card->dapm;
+
+	/* 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 void cx81801_hangup(struct tty_struct *tty)
+{
+	cx81801_close(tty);
+}
+
+/* Line discipline .receive_buf() */
+static void cx81801_receive(struct tty_struct *tty, const unsigned char *cp,
+		const char *fp, int count)
+{
+	struct snd_soc_component *component = tty->disc_data;
+	const unsigned char *c;
+	int apply, ret;
+
+	if (!component)
+		return;
+
+	if (!component->card->pop_time) {
+		/* First modem response, complete setup procedure */
+
+		/* Initialize timer used for config pulse generation */
+		timer_setup(&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(component->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)
+			gpiod_set_value(gpiod_modem_codec, 1);
+		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 = {
+	.name = "cx81801",
+	.num = N_V253,
+	.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 functionality activated.  You can still control its audio input/output
+ * constellation and speakerphone gain from userspace by issuing AT commands
+ * over the modem port.
+ */
+
+static struct snd_soc_ops ams_delta_ops;
+
+
+/* Digital mute implemented using modem/CPU multiplexer.
+ * Shares hardware with codec config pulse generation */
+static bool ams_delta_muted = 1;
+
+static int ams_delta_mute(struct snd_soc_dai *dai, int mute, int direction)
+{
+	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)
+		gpiod_set_value(gpiod_modem_codec, !!mute);
+	return 0;
+}
+
+/* Our codec DAI probably doesn't have its own .ops structure */
+static const struct snd_soc_dai_ops ams_delta_dai_ops = {
+	.mute_stream = ams_delta_mute,
+	.no_capture_mute = 1,
+};
+
+/* 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_mute(NULL, 0, substream->stream);
+}
+
+static void ams_delta_shutdown(struct snd_pcm_substream *substream)
+{
+	ams_delta_mute(NULL, 1, substream->stream);
+}
+
+
+/*
+ * Card initialization
+ */
+
+static int ams_delta_cx20442_init(struct snd_soc_pcm_runtime *rtd)
+{
+	struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0);
+	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 = asoc_rtd_to_codec(rtd, 0)->component;
+
+	/* Add hook switch - can be used to control the codec from userspace
+	 * even if line discipline fails */
+	ret = snd_soc_card_jack_new_pins(card, "hook_switch", SND_JACK_HEADSET,
+					 &ams_delta_hook_switch, NULL, 0);
+	if (ret)
+		dev_warn(card->dev,
+				"Failed to allocate resources for hook switch, "
+				"will continue without one.\n");
+	else {
+		ret = snd_soc_jack_add_gpiods(card->dev, &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");
+	}
+
+	gpiod_modem_codec = devm_gpiod_get(card->dev, "modem_codec",
+					   GPIOD_OUT_HIGH);
+	if (IS_ERR(gpiod_modem_codec)) {
+		dev_warn(card->dev, "Failed to obtain modem_codec GPIO\n");
+		return 0;
+	}
+
+	/* 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;
+	}
+
+	/* Register optional line discipline for over the modem control */
+	ret = tty_register_ldisc(&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;
+}
+
+/* DAI glue - connects codec <--> CPU */
+SND_SOC_DAILINK_DEFS(cx20442,
+	DAILINK_COMP_ARRAY(COMP_CPU("omap-mcbsp.1")),
+	DAILINK_COMP_ARRAY(COMP_CODEC("cx20442-codec", "cx20442-voice")),
+	DAILINK_COMP_ARRAY(COMP_PLATFORM("omap-mcbsp.1")));
+
+static struct snd_soc_dai_link ams_delta_dai_link = {
+	.name = "CX20442",
+	.stream_name = "CX20442",
+	.init = ams_delta_cx20442_init,
+	.ops = &ams_delta_ops,
+	.dai_fmt = SND_SOC_DAIFMT_DSP_A | SND_SOC_DAIFMT_NB_NF |
+		   SND_SOC_DAIFMT_CBM_CFM,
+	SND_SOC_DAILINK_REG(cx20442),
+};
+
+/* Audio card driver */
+static struct snd_soc_card ams_delta_audio_card = {
+	.name = "AMS_DELTA",
+	.owner = THIS_MODULE,
+	.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;
+
+	handset_mute = devm_gpiod_get(card->dev, "handset_mute",
+				      GPIOD_OUT_HIGH);
+	if (IS_ERR(handset_mute))
+		return PTR_ERR(handset_mute);
+
+	handsfree_mute = devm_gpiod_get(card->dev, "handsfree_mute",
+					GPIOD_OUT_HIGH);
+	if (IS_ERR(handsfree_mute))
+		return PTR_ERR(handsfree_mute);
+
+	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);
+
+	tty_unregister_ldisc(&cx81801_ops);
+
+	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,
+	},
+	.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);
diff --git a/sound/soc/ti/davinci-evm.c b/sound/soc/ti/davinci-evm.c
new file mode 100644
index 000000000..68d69e326
--- /dev/null
+++ b/sound/soc/ti/davinci-evm.c
@@ -0,0 +1,536 @@
+// SPDX-License-Identifier: GPL-2.0-only
+/*
+ * ASoC driver for TI DAVINCI EVM platform
+ *
+ * Author:      Vladimir Barinov, <vbarinov@embeddedalley.com>
+ * Copyright:   (C) 2007 MontaVista Software, Inc., <source@mvista.com>
+ */
+
+#include <linux/module.h>
+#include <linux/moduleparam.h>
+#include <linux/timer.h>
+#include <linux/interrupt.h>
+#include <linux/platform_device.h>
+#include <linux/i2c.h>
+#include <linux/of_platform.h>
+#include <linux/clk.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/soc.h>
+
+#include <asm/dma.h>
+#include <asm/mach-types.h>
+
+struct snd_soc_card_drvdata_davinci {
+	struct clk *mclk;
+	unsigned sysclk;
+};
+
+static int evm_startup(struct snd_pcm_substream *substream)
+{
+	struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream);
+	struct snd_soc_card *soc_card = rtd->card;
+	struct snd_soc_card_drvdata_davinci *drvdata =
+		snd_soc_card_get_drvdata(soc_card);
+
+	if (drvdata->mclk)
+		return clk_prepare_enable(drvdata->mclk);
+
+	return 0;
+}
+
+static void evm_shutdown(struct snd_pcm_substream *substream)
+{
+	struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream);
+	struct snd_soc_card *soc_card = rtd->card;
+	struct snd_soc_card_drvdata_davinci *drvdata =
+		snd_soc_card_get_drvdata(soc_card);
+
+	clk_disable_unprepare(drvdata->mclk);
+}
+
+static int evm_hw_params(struct snd_pcm_substream *substream,
+			 struct snd_pcm_hw_params *params)
+{
+	struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream);
+	struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0);
+	struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0);
+	struct snd_soc_card *soc_card = rtd->card;
+	int ret = 0;
+	unsigned sysclk = ((struct snd_soc_card_drvdata_davinci *)
+			   snd_soc_card_get_drvdata(soc_card))->sysclk;
+
+	/* set the codec system clock */
+	ret = snd_soc_dai_set_sysclk(codec_dai, 0, sysclk, SND_SOC_CLOCK_OUT);
+	if (ret < 0)
+		return ret;
+
+	/* set the CPU system clock */
+	ret = snd_soc_dai_set_sysclk(cpu_dai, 0, sysclk, SND_SOC_CLOCK_OUT);
+	if (ret < 0 && ret != -ENOTSUPP)
+		return ret;
+
+	return 0;
+}
+
+static const struct snd_soc_ops evm_ops = {
+	.startup = evm_startup,
+	.shutdown = evm_shutdown,
+	.hw_params = evm_hw_params,
+};
+
+/* davinci-evm machine dapm widgets */
+static const struct snd_soc_dapm_widget aic3x_dapm_widgets[] = {
+	SND_SOC_DAPM_HP("Headphone Jack", NULL),
+	SND_SOC_DAPM_LINE("Line Out", NULL),
+	SND_SOC_DAPM_MIC("Mic Jack", NULL),
+	SND_SOC_DAPM_LINE("Line In", NULL),
+};
+
+/* davinci-evm machine audio_mapnections to the codec pins */
+static const struct snd_soc_dapm_route audio_map[] = {
+	/* Headphone connected to HPLOUT, HPROUT */
+	{"Headphone Jack", NULL, "HPLOUT"},
+	{"Headphone Jack", NULL, "HPROUT"},
+
+	/* Line Out connected to LLOUT, RLOUT */
+	{"Line Out", NULL, "LLOUT"},
+	{"Line Out", NULL, "RLOUT"},
+
+	/* Mic connected to (MIC3L | MIC3R) */
+	{"MIC3L", NULL, "Mic Bias"},
+	{"MIC3R", NULL, "Mic Bias"},
+	{"Mic Bias", NULL, "Mic Jack"},
+
+	/* Line In connected to (LINE1L | LINE2L), (LINE1R | LINE2R) */
+	{"LINE1L", NULL, "Line In"},
+	{"LINE2L", NULL, "Line In"},
+	{"LINE1R", NULL, "Line In"},
+	{"LINE2R", NULL, "Line In"},
+};
+
+/* Logic for a aic3x as connected on a davinci-evm */
+static int evm_aic3x_init(struct snd_soc_pcm_runtime *rtd)
+{
+	struct snd_soc_card *card = rtd->card;
+	struct device_node *np = card->dev->of_node;
+	int ret;
+
+	/* Add davinci-evm specific widgets */
+	snd_soc_dapm_new_controls(&card->dapm, aic3x_dapm_widgets,
+				  ARRAY_SIZE(aic3x_dapm_widgets));
+
+	if (np) {
+		ret = snd_soc_of_parse_audio_routing(card, "ti,audio-routing");
+		if (ret)
+			return ret;
+	} else {
+		/* Set up davinci-evm specific audio path audio_map */
+		snd_soc_dapm_add_routes(&card->dapm, audio_map,
+					ARRAY_SIZE(audio_map));
+	}
+
+	/* not connected */
+	snd_soc_dapm_nc_pin(&card->dapm, "MONO_LOUT");
+	snd_soc_dapm_nc_pin(&card->dapm, "HPLCOM");
+	snd_soc_dapm_nc_pin(&card->dapm, "HPRCOM");
+
+	return 0;
+}
+
+/* davinci-evm digital audio interface glue - connects codec <--> CPU */
+SND_SOC_DAILINK_DEFS(dm6446,
+	DAILINK_COMP_ARRAY(COMP_CPU("davinci-mcbsp")),
+	DAILINK_COMP_ARRAY(COMP_CODEC("tlv320aic3x-codec.1-001b",
+				      "tlv320aic3x-hifi")),
+	DAILINK_COMP_ARRAY(COMP_PLATFORM("davinci-mcbsp")));
+
+static struct snd_soc_dai_link dm6446_evm_dai = {
+	.name = "TLV320AIC3X",
+	.stream_name = "AIC3X",
+	.init = evm_aic3x_init,
+	.ops = &evm_ops,
+	.dai_fmt = SND_SOC_DAIFMT_DSP_B | SND_SOC_DAIFMT_CBM_CFM |
+		   SND_SOC_DAIFMT_IB_NF,
+	SND_SOC_DAILINK_REG(dm6446),
+};
+
+SND_SOC_DAILINK_DEFS(dm355,
+	DAILINK_COMP_ARRAY(COMP_CPU("davinci-mcbsp.1")),
+	DAILINK_COMP_ARRAY(COMP_CODEC("tlv320aic3x-codec.1-001b",
+				      "tlv320aic3x-hifi")),
+	DAILINK_COMP_ARRAY(COMP_PLATFORM("davinci-mcbsp.1")));
+
+static struct snd_soc_dai_link dm355_evm_dai = {
+	.name = "TLV320AIC3X",
+	.stream_name = "AIC3X",
+	.init = evm_aic3x_init,
+	.ops = &evm_ops,
+	.dai_fmt = SND_SOC_DAIFMT_DSP_B | SND_SOC_DAIFMT_CBM_CFM |
+		   SND_SOC_DAIFMT_IB_NF,
+	SND_SOC_DAILINK_REG(dm355),
+};
+
+#ifdef CONFIG_SND_SOC_DM365_AIC3X_CODEC
+SND_SOC_DAILINK_DEFS(dm365,
+	DAILINK_COMP_ARRAY(COMP_CPU("davinci-mcbsp")),
+	DAILINK_COMP_ARRAY(COMP_CODEC("tlv320aic3x-codec.1-0018",
+				      "tlv320aic3x-hifi")),
+	DAILINK_COMP_ARRAY(COMP_PLATFORM("davinci-mcbsp")));
+#elif defined(CONFIG_SND_SOC_DM365_VOICE_CODEC)
+SND_SOC_DAILINK_DEFS(dm365,
+	DAILINK_COMP_ARRAY(COMP_CPU("davinci-vcif")),
+	DAILINK_COMP_ARRAY(COMP_CODEC("cq93vc-codec", "cq93vc-hifi")),
+	DAILINK_COMP_ARRAY(COMP_PLATFORM("davinci-vcif")));
+#endif
+
+static struct snd_soc_dai_link dm365_evm_dai = {
+#ifdef CONFIG_SND_SOC_DM365_AIC3X_CODEC
+	.name = "TLV320AIC3X",
+	.stream_name = "AIC3X",
+	.init = evm_aic3x_init,
+	.ops = &evm_ops,
+	.dai_fmt = SND_SOC_DAIFMT_DSP_B | SND_SOC_DAIFMT_CBM_CFM |
+		   SND_SOC_DAIFMT_IB_NF,
+	SND_SOC_DAILINK_REG(dm365),
+#elif defined(CONFIG_SND_SOC_DM365_VOICE_CODEC)
+	.name = "Voice Codec - CQ93VC",
+	.stream_name = "CQ93",
+	SND_SOC_DAILINK_REG(dm365),
+#endif
+};
+
+SND_SOC_DAILINK_DEFS(dm6467_aic3x,
+	DAILINK_COMP_ARRAY(COMP_CPU("davinci-mcasp.0")),
+	DAILINK_COMP_ARRAY(COMP_CODEC("tlv320aic3x-codec.0-001a",
+				      "tlv320aic3x-hifi")),
+	DAILINK_COMP_ARRAY(COMP_PLATFORM("davinci-mcasp.0")));
+
+SND_SOC_DAILINK_DEFS(dm6467_spdif,
+	DAILINK_COMP_ARRAY(COMP_CPU("davinci-mcasp.1")),
+	DAILINK_COMP_ARRAY(COMP_CODEC("spdif_dit", "dit-hifi")),
+	DAILINK_COMP_ARRAY(COMP_PLATFORM("davinci-mcasp.1")));
+
+static struct snd_soc_dai_link dm6467_evm_dai[] = {
+	{
+		.name = "TLV320AIC3X",
+		.stream_name = "AIC3X",
+		.init = evm_aic3x_init,
+		.ops = &evm_ops,
+		.dai_fmt = SND_SOC_DAIFMT_DSP_B | SND_SOC_DAIFMT_CBM_CFM |
+			   SND_SOC_DAIFMT_IB_NF,
+		SND_SOC_DAILINK_REG(dm6467_aic3x),
+	},
+	{
+		.name = "McASP",
+		.stream_name = "spdif",
+		.dai_fmt = SND_SOC_DAIFMT_DSP_B | SND_SOC_DAIFMT_CBM_CFM |
+			   SND_SOC_DAIFMT_IB_NF,
+		SND_SOC_DAILINK_REG(dm6467_spdif),
+	},
+};
+
+SND_SOC_DAILINK_DEFS(da830,
+	DAILINK_COMP_ARRAY(COMP_CPU("davinci-mcasp.1")),
+	DAILINK_COMP_ARRAY(COMP_CODEC("tlv320aic3x-codec.1-0018",
+				      "tlv320aic3x-hifi")),
+	DAILINK_COMP_ARRAY(COMP_PLATFORM("davinci-mcasp.1")));
+
+static struct snd_soc_dai_link da830_evm_dai = {
+	.name = "TLV320AIC3X",
+	.stream_name = "AIC3X",
+	.init = evm_aic3x_init,
+	.ops = &evm_ops,
+	.dai_fmt = SND_SOC_DAIFMT_DSP_B | SND_SOC_DAIFMT_CBM_CFM |
+		   SND_SOC_DAIFMT_IB_NF,
+	SND_SOC_DAILINK_REG(da830),
+};
+
+SND_SOC_DAILINK_DEFS(da850,
+	DAILINK_COMP_ARRAY(COMP_CPU("davinci-mcasp.0")),
+	DAILINK_COMP_ARRAY(COMP_CODEC("tlv320aic3x-codec.1-0018",
+				      "tlv320aic3x-hifi")),
+	DAILINK_COMP_ARRAY(COMP_PLATFORM("davinci-mcasp.0")));
+
+static struct snd_soc_dai_link da850_evm_dai = {
+	.name = "TLV320AIC3X",
+	.stream_name = "AIC3X",
+	.init = evm_aic3x_init,
+	.ops = &evm_ops,
+	.dai_fmt = SND_SOC_DAIFMT_DSP_B | SND_SOC_DAIFMT_CBM_CFM |
+		   SND_SOC_DAIFMT_IB_NF,
+	SND_SOC_DAILINK_REG(da850),
+};
+
+/* davinci dm6446 evm audio machine driver */
+/*
+ * ASP0 in DM6446 EVM is clocked by U55, as configured by
+ * board-dm644x-evm.c using GPIOs from U18.  There are six
+ * options; here we "know" we use a 48 KHz sample rate.
+ */
+static struct snd_soc_card_drvdata_davinci dm6446_snd_soc_card_drvdata = {
+	.sysclk = 12288000,
+};
+
+static struct snd_soc_card dm6446_snd_soc_card_evm = {
+	.name = "DaVinci DM6446 EVM",
+	.owner = THIS_MODULE,
+	.dai_link = &dm6446_evm_dai,
+	.num_links = 1,
+	.drvdata = &dm6446_snd_soc_card_drvdata,
+};
+
+/* davinci dm355 evm audio machine driver */
+/* ASP1 on DM355 EVM is clocked by an external oscillator */
+static struct snd_soc_card_drvdata_davinci dm355_snd_soc_card_drvdata = {
+	.sysclk = 27000000,
+};
+
+static struct snd_soc_card dm355_snd_soc_card_evm = {
+	.name = "DaVinci DM355 EVM",
+	.owner = THIS_MODULE,
+	.dai_link = &dm355_evm_dai,
+	.num_links = 1,
+	.drvdata = &dm355_snd_soc_card_drvdata,
+};
+
+/* davinci dm365 evm audio machine driver */
+static struct snd_soc_card_drvdata_davinci dm365_snd_soc_card_drvdata = {
+	.sysclk = 27000000,
+};
+
+static struct snd_soc_card dm365_snd_soc_card_evm = {
+	.name = "DaVinci DM365 EVM",
+	.owner = THIS_MODULE,
+	.dai_link = &dm365_evm_dai,
+	.num_links = 1,
+	.drvdata = &dm365_snd_soc_card_drvdata,
+};
+
+/* davinci dm6467 evm audio machine driver */
+static struct snd_soc_card_drvdata_davinci dm6467_snd_soc_card_drvdata = {
+	.sysclk = 27000000,
+};
+
+static struct snd_soc_card dm6467_snd_soc_card_evm = {
+	.name = "DaVinci DM6467 EVM",
+	.owner = THIS_MODULE,
+	.dai_link = dm6467_evm_dai,
+	.num_links = ARRAY_SIZE(dm6467_evm_dai),
+	.drvdata = &dm6467_snd_soc_card_drvdata,
+};
+
+static struct snd_soc_card_drvdata_davinci da830_snd_soc_card_drvdata = {
+	.sysclk = 24576000,
+};
+
+static struct snd_soc_card da830_snd_soc_card = {
+	.name = "DA830/OMAP-L137 EVM",
+	.owner = THIS_MODULE,
+	.dai_link = &da830_evm_dai,
+	.num_links = 1,
+	.drvdata = &da830_snd_soc_card_drvdata,
+};
+
+static struct snd_soc_card_drvdata_davinci da850_snd_soc_card_drvdata = {
+	.sysclk = 24576000,
+};
+
+static struct snd_soc_card da850_snd_soc_card = {
+	.name = "DA850/OMAP-L138 EVM",
+	.owner = THIS_MODULE,
+	.dai_link = &da850_evm_dai,
+	.num_links = 1,
+	.drvdata = &da850_snd_soc_card_drvdata,
+};
+
+#if defined(CONFIG_OF)
+
+/*
+ * The struct is used as place holder. It will be completely
+ * filled with data from dt node.
+ */
+SND_SOC_DAILINK_DEFS(evm,
+	DAILINK_COMP_ARRAY(COMP_EMPTY()),
+	DAILINK_COMP_ARRAY(COMP_CODEC(NULL, "tlv320aic3x-hifi")),
+	DAILINK_COMP_ARRAY(COMP_EMPTY()));
+
+static struct snd_soc_dai_link evm_dai_tlv320aic3x = {
+	.name		= "TLV320AIC3X",
+	.stream_name	= "AIC3X",
+	.ops            = &evm_ops,
+	.init           = evm_aic3x_init,
+	.dai_fmt = SND_SOC_DAIFMT_DSP_B | SND_SOC_DAIFMT_CBM_CFM |
+		   SND_SOC_DAIFMT_IB_NF,
+	SND_SOC_DAILINK_REG(evm),
+};
+
+static const struct of_device_id davinci_evm_dt_ids[] = {
+	{
+		.compatible = "ti,da830-evm-audio",
+		.data = (void *) &evm_dai_tlv320aic3x,
+	},
+	{ /* sentinel */ }
+};
+MODULE_DEVICE_TABLE(of, davinci_evm_dt_ids);
+
+/* davinci evm audio machine driver */
+static struct snd_soc_card evm_soc_card = {
+	.owner = THIS_MODULE,
+	.num_links = 1,
+};
+
+static int davinci_evm_probe(struct platform_device *pdev)
+{
+	struct device_node *np = pdev->dev.of_node;
+	const struct of_device_id *match;
+	struct snd_soc_dai_link *dai;
+	struct snd_soc_card_drvdata_davinci *drvdata = NULL;
+	struct clk *mclk;
+	int ret = 0;
+
+	match = of_match_device(of_match_ptr(davinci_evm_dt_ids), &pdev->dev);
+	if (!match) {
+		dev_err(&pdev->dev, "Error: No device match found\n");
+		return -ENODEV;
+	}
+
+	dai = (struct snd_soc_dai_link *) match->data;
+
+	evm_soc_card.dai_link = dai;
+
+	dai->codecs->of_node = of_parse_phandle(np, "ti,audio-codec", 0);
+	if (!dai->codecs->of_node)
+		return -EINVAL;
+
+	dai->cpus->of_node = of_parse_phandle(np, "ti,mcasp-controller", 0);
+	if (!dai->cpus->of_node)
+		return -EINVAL;
+
+	dai->platforms->of_node = dai->cpus->of_node;
+
+	evm_soc_card.dev = &pdev->dev;
+	ret = snd_soc_of_parse_card_name(&evm_soc_card, "ti,model");
+	if (ret)
+		return ret;
+
+	mclk = devm_clk_get(&pdev->dev, "mclk");
+	if (PTR_ERR(mclk) == -EPROBE_DEFER) {
+		return -EPROBE_DEFER;
+	} else if (IS_ERR(mclk)) {
+		dev_dbg(&pdev->dev, "mclk not found.\n");
+		mclk = NULL;
+	}
+
+	drvdata = devm_kzalloc(&pdev->dev, sizeof(*drvdata), GFP_KERNEL);
+	if (!drvdata)
+		return -ENOMEM;
+
+	drvdata->mclk = mclk;
+
+	ret = of_property_read_u32(np, "ti,codec-clock-rate", &drvdata->sysclk);
+
+	if (ret < 0) {
+		if (!drvdata->mclk) {
+			dev_err(&pdev->dev,
+				"No clock or clock rate defined.\n");
+			return -EINVAL;
+		}
+		drvdata->sysclk = clk_get_rate(drvdata->mclk);
+	} else if (drvdata->mclk) {
+		unsigned int requestd_rate = drvdata->sysclk;
+		clk_set_rate(drvdata->mclk, drvdata->sysclk);
+		drvdata->sysclk = clk_get_rate(drvdata->mclk);
+		if (drvdata->sysclk != requestd_rate)
+			dev_warn(&pdev->dev,
+				 "Could not get requested rate %u using %u.\n",
+				 requestd_rate, drvdata->sysclk);
+	}
+
+	snd_soc_card_set_drvdata(&evm_soc_card, drvdata);
+	ret = devm_snd_soc_register_card(&pdev->dev, &evm_soc_card);
+
+	if (ret)
+		dev_err(&pdev->dev, "snd_soc_register_card failed (%d)\n", ret);
+
+	return ret;
+}
+
+static struct platform_driver davinci_evm_driver = {
+	.probe		= davinci_evm_probe,
+	.driver		= {
+		.name	= "davinci_evm",
+		.pm	= &snd_soc_pm_ops,
+		.of_match_table = of_match_ptr(davinci_evm_dt_ids),
+	},
+};
+#endif
+
+static struct platform_device *evm_snd_device;
+
+static int __init evm_init(void)
+{
+	struct snd_soc_card *evm_snd_dev_data;
+	int index;
+	int ret;
+
+	/*
+	 * If dtb is there, the devices will be created dynamically.
+	 * Only register platfrom driver structure.
+	 */
+#if defined(CONFIG_OF)
+	if (of_have_populated_dt())
+		return platform_driver_register(&davinci_evm_driver);
+#endif
+
+	if (machine_is_davinci_evm()) {
+		evm_snd_dev_data = &dm6446_snd_soc_card_evm;
+		index = 0;
+	} else if (machine_is_davinci_dm355_evm()) {
+		evm_snd_dev_data = &dm355_snd_soc_card_evm;
+		index = 1;
+	} else if (machine_is_davinci_dm365_evm()) {
+		evm_snd_dev_data = &dm365_snd_soc_card_evm;
+		index = 0;
+	} else if (machine_is_davinci_dm6467_evm()) {
+		evm_snd_dev_data = &dm6467_snd_soc_card_evm;
+		index = 0;
+	} else if (machine_is_davinci_da830_evm()) {
+		evm_snd_dev_data = &da830_snd_soc_card;
+		index = 1;
+	} else if (machine_is_davinci_da850_evm()) {
+		evm_snd_dev_data = &da850_snd_soc_card;
+		index = 0;
+	} else
+		return -EINVAL;
+
+	evm_snd_device = platform_device_alloc("soc-audio", index);
+	if (!evm_snd_device)
+		return -ENOMEM;
+
+	platform_set_drvdata(evm_snd_device, evm_snd_dev_data);
+	ret = platform_device_add(evm_snd_device);
+	if (ret)
+		platform_device_put(evm_snd_device);
+
+	return ret;
+}
+
+static void __exit evm_exit(void)
+{
+#if defined(CONFIG_OF)
+	if (of_have_populated_dt()) {
+		platform_driver_unregister(&davinci_evm_driver);
+		return;
+	}
+#endif
+
+	platform_device_unregister(evm_snd_device);
+}
+
+module_init(evm_init);
+module_exit(evm_exit);
+
+MODULE_AUTHOR("Vladimir Barinov");
+MODULE_DESCRIPTION("TI DAVINCI EVM ASoC driver");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/ti/davinci-i2s.c b/sound/soc/ti/davinci-i2s.c
new file mode 100644
index 000000000..e6e77a5f3
--- /dev/null
+++ b/sound/soc/ti/davinci-i2s.c
@@ -0,0 +1,773 @@
+// SPDX-License-Identifier: GPL-2.0-only
+/*
+ * ALSA SoC I2S (McBSP) Audio Layer for TI DAVINCI processor
+ *
+ * Author:      Vladimir Barinov, <vbarinov@embeddedalley.com>
+ * Copyright:   (C) 2007 MontaVista Software, Inc., <source@mvista.com>
+ *
+ * DT support	(c) 2016 Petr Kulhavy, Barix AG <petr@barix.com>
+ *		based on davinci-mcasp.c DT support
+ *
+ * TODO:
+ * on DA850 implement HW FIFOs instead of DMA into DXR and DRR registers
+ */
+
+#include <linux/init.h>
+#include <linux/module.h>
+#include <linux/device.h>
+#include <linux/slab.h>
+#include <linux/delay.h>
+#include <linux/io.h>
+#include <linux/clk.h>
+#include <linux/platform_data/davinci_asp.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 "edma-pcm.h"
+#include "davinci-i2s.h"
+
+#define DRV_NAME "davinci-i2s"
+
+/*
+ * NOTE:  terminology here is confusing.
+ *
+ *  - This driver supports the "Audio Serial Port" (ASP),
+ *    found on dm6446, dm355, and other DaVinci chips.
+ *
+ *  - But it labels it a "Multi-channel Buffered Serial Port"
+ *    (McBSP) as on older chips like the dm642 ... which was
+ *    backward-compatible, possibly explaining that confusion.
+ *
+ *  - OMAP chips have a controller called McBSP, which is
+ *    incompatible with the DaVinci flavor of McBSP.
+ *
+ *  - Newer DaVinci chips have a controller called McASP,
+ *    incompatible with ASP and with either McBSP.
+ *
+ * In short:  this uses ASP to implement I2S, not McBSP.
+ * And it won't be the only DaVinci implemention of I2S.
+ */
+#define DAVINCI_MCBSP_DRR_REG	0x00
+#define DAVINCI_MCBSP_DXR_REG	0x04
+#define DAVINCI_MCBSP_SPCR_REG	0x08
+#define DAVINCI_MCBSP_RCR_REG	0x0c
+#define DAVINCI_MCBSP_XCR_REG	0x10
+#define DAVINCI_MCBSP_SRGR_REG	0x14
+#define DAVINCI_MCBSP_PCR_REG	0x24
+
+#define DAVINCI_MCBSP_SPCR_RRST		(1 << 0)
+#define DAVINCI_MCBSP_SPCR_RINTM(v)	((v) << 4)
+#define DAVINCI_MCBSP_SPCR_XRST		(1 << 16)
+#define DAVINCI_MCBSP_SPCR_XINTM(v)	((v) << 20)
+#define DAVINCI_MCBSP_SPCR_GRST		(1 << 22)
+#define DAVINCI_MCBSP_SPCR_FRST		(1 << 23)
+#define DAVINCI_MCBSP_SPCR_FREE		(1 << 25)
+
+#define DAVINCI_MCBSP_RCR_RWDLEN1(v)	((v) << 5)
+#define DAVINCI_MCBSP_RCR_RFRLEN1(v)	((v) << 8)
+#define DAVINCI_MCBSP_RCR_RDATDLY(v)	((v) << 16)
+#define DAVINCI_MCBSP_RCR_RFIG		(1 << 18)
+#define DAVINCI_MCBSP_RCR_RWDLEN2(v)	((v) << 21)
+#define DAVINCI_MCBSP_RCR_RFRLEN2(v)	((v) << 24)
+#define DAVINCI_MCBSP_RCR_RPHASE	BIT(31)
+
+#define DAVINCI_MCBSP_XCR_XWDLEN1(v)	((v) << 5)
+#define DAVINCI_MCBSP_XCR_XFRLEN1(v)	((v) << 8)
+#define DAVINCI_MCBSP_XCR_XDATDLY(v)	((v) << 16)
+#define DAVINCI_MCBSP_XCR_XFIG		(1 << 18)
+#define DAVINCI_MCBSP_XCR_XWDLEN2(v)	((v) << 21)
+#define DAVINCI_MCBSP_XCR_XFRLEN2(v)	((v) << 24)
+#define DAVINCI_MCBSP_XCR_XPHASE	BIT(31)
+
+#define DAVINCI_MCBSP_SRGR_FWID(v)	((v) << 8)
+#define DAVINCI_MCBSP_SRGR_FPER(v)	((v) << 16)
+#define DAVINCI_MCBSP_SRGR_FSGM		(1 << 28)
+#define DAVINCI_MCBSP_SRGR_CLKSM	BIT(29)
+
+#define DAVINCI_MCBSP_PCR_CLKRP		(1 << 0)
+#define DAVINCI_MCBSP_PCR_CLKXP		(1 << 1)
+#define DAVINCI_MCBSP_PCR_FSRP		(1 << 2)
+#define DAVINCI_MCBSP_PCR_FSXP		(1 << 3)
+#define DAVINCI_MCBSP_PCR_SCLKME	(1 << 7)
+#define DAVINCI_MCBSP_PCR_CLKRM		(1 << 8)
+#define DAVINCI_MCBSP_PCR_CLKXM		(1 << 9)
+#define DAVINCI_MCBSP_PCR_FSRM		(1 << 10)
+#define DAVINCI_MCBSP_PCR_FSXM		(1 << 11)
+
+enum {
+	DAVINCI_MCBSP_WORD_8 = 0,
+	DAVINCI_MCBSP_WORD_12,
+	DAVINCI_MCBSP_WORD_16,
+	DAVINCI_MCBSP_WORD_20,
+	DAVINCI_MCBSP_WORD_24,
+	DAVINCI_MCBSP_WORD_32,
+};
+
+static const unsigned char data_type[SNDRV_PCM_FORMAT_S32_LE + 1] = {
+	[SNDRV_PCM_FORMAT_S8]		= 1,
+	[SNDRV_PCM_FORMAT_S16_LE]	= 2,
+	[SNDRV_PCM_FORMAT_S32_LE]	= 4,
+};
+
+static const unsigned char asp_word_length[SNDRV_PCM_FORMAT_S32_LE + 1] = {
+	[SNDRV_PCM_FORMAT_S8]		= DAVINCI_MCBSP_WORD_8,
+	[SNDRV_PCM_FORMAT_S16_LE]	= DAVINCI_MCBSP_WORD_16,
+	[SNDRV_PCM_FORMAT_S32_LE]	= DAVINCI_MCBSP_WORD_32,
+};
+
+static const unsigned char double_fmt[SNDRV_PCM_FORMAT_S32_LE + 1] = {
+	[SNDRV_PCM_FORMAT_S8]		= SNDRV_PCM_FORMAT_S16_LE,
+	[SNDRV_PCM_FORMAT_S16_LE]	= SNDRV_PCM_FORMAT_S32_LE,
+};
+
+struct davinci_mcbsp_dev {
+	struct device *dev;
+	struct snd_dmaengine_dai_dma_data dma_data[2];
+	int dma_request[2];
+	void __iomem			*base;
+#define MOD_DSP_A	0
+#define MOD_DSP_B	1
+	int				mode;
+	u32				pcr;
+	struct clk			*clk;
+	/*
+	 * Combining both channels into 1 element will at least double the
+	 * amount of time between servicing the dma channel, increase
+	 * effiency, and reduce the chance of overrun/underrun. But,
+	 * it will result in the left & right channels being swapped.
+	 *
+	 * If relabeling the left and right channels is not possible,
+	 * you may want to let the codec know to swap them back.
+	 *
+	 * It may allow x10 the amount of time to service dma requests,
+	 * if the codec is master and is using an unnecessarily fast bit clock
+	 * (ie. tlvaic23b), independent of the sample rate. So, having an
+	 * entire frame at once means it can be serviced at the sample rate
+	 * instead of the bit clock rate.
+	 *
+	 * In the now unlikely case that an underrun still
+	 * occurs, both the left and right samples will be repeated
+	 * so that no pops are heard, and the left and right channels
+	 * won't end up being swapped because of the underrun.
+	 */
+	unsigned enable_channel_combine:1;
+
+	unsigned int fmt;
+	int clk_div;
+	int clk_input_pin;
+	bool i2s_accurate_sck;
+};
+
+static inline void davinci_mcbsp_write_reg(struct davinci_mcbsp_dev *dev,
+					   int reg, u32 val)
+{
+	__raw_writel(val, dev->base + reg);
+}
+
+static inline u32 davinci_mcbsp_read_reg(struct davinci_mcbsp_dev *dev, int reg)
+{
+	return __raw_readl(dev->base + reg);
+}
+
+static void toggle_clock(struct davinci_mcbsp_dev *dev, int playback)
+{
+	u32 m = playback ? DAVINCI_MCBSP_PCR_CLKXP : DAVINCI_MCBSP_PCR_CLKRP;
+	/* The clock needs to toggle to complete reset.
+	 * So, fake it by toggling the clk polarity.
+	 */
+	davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_PCR_REG, dev->pcr ^ m);
+	davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_PCR_REG, dev->pcr);
+}
+
+static void davinci_mcbsp_start(struct davinci_mcbsp_dev *dev,
+		struct snd_pcm_substream *substream)
+{
+	int playback = (substream->stream == SNDRV_PCM_STREAM_PLAYBACK);
+	u32 spcr;
+	u32 mask = playback ? DAVINCI_MCBSP_SPCR_XRST : DAVINCI_MCBSP_SPCR_RRST;
+
+	/* Enable transmitter or receiver */
+	spcr = davinci_mcbsp_read_reg(dev, DAVINCI_MCBSP_SPCR_REG);
+	spcr |= mask;
+
+	if (dev->pcr & (DAVINCI_MCBSP_PCR_FSXM | DAVINCI_MCBSP_PCR_FSRM)) {
+		/* Start frame sync */
+		spcr |= DAVINCI_MCBSP_SPCR_FRST;
+	}
+	davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_SPCR_REG, spcr);
+}
+
+static void davinci_mcbsp_stop(struct davinci_mcbsp_dev *dev, int playback)
+{
+	u32 spcr;
+
+	/* Reset transmitter/receiver and sample rate/frame sync generators */
+	spcr = davinci_mcbsp_read_reg(dev, DAVINCI_MCBSP_SPCR_REG);
+	spcr &= ~(DAVINCI_MCBSP_SPCR_GRST | DAVINCI_MCBSP_SPCR_FRST);
+	spcr &= playback ? ~DAVINCI_MCBSP_SPCR_XRST : ~DAVINCI_MCBSP_SPCR_RRST;
+	davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_SPCR_REG, spcr);
+	toggle_clock(dev, playback);
+}
+
+#define DEFAULT_BITPERSAMPLE	16
+
+static int davinci_i2s_set_dai_fmt(struct snd_soc_dai *cpu_dai,
+				   unsigned int fmt)
+{
+	struct davinci_mcbsp_dev *dev = snd_soc_dai_get_drvdata(cpu_dai);
+	unsigned int pcr;
+	unsigned int srgr;
+	bool inv_fs = false;
+	/* Attention srgr is updated by hw_params! */
+	srgr = DAVINCI_MCBSP_SRGR_FSGM |
+		DAVINCI_MCBSP_SRGR_FPER(DEFAULT_BITPERSAMPLE * 2 - 1) |
+		DAVINCI_MCBSP_SRGR_FWID(DEFAULT_BITPERSAMPLE - 1);
+
+	dev->fmt = fmt;
+	/* set master/slave audio interface */
+	switch (fmt & SND_SOC_DAIFMT_CLOCK_PROVIDER_MASK) {
+	case SND_SOC_DAIFMT_BP_FP:
+		/* cpu is master */
+		pcr = DAVINCI_MCBSP_PCR_FSXM |
+			DAVINCI_MCBSP_PCR_FSRM |
+			DAVINCI_MCBSP_PCR_CLKXM |
+			DAVINCI_MCBSP_PCR_CLKRM;
+		break;
+	case SND_SOC_DAIFMT_BC_FP:
+		pcr = DAVINCI_MCBSP_PCR_FSRM | DAVINCI_MCBSP_PCR_FSXM;
+		/*
+		 * Selection of the clock input pin that is the
+		 * input for the Sample Rate Generator.
+		 * McBSP FSR and FSX are driven by the Sample Rate
+		 * Generator.
+		 */
+		switch (dev->clk_input_pin) {
+		case MCBSP_CLKS:
+			pcr |= DAVINCI_MCBSP_PCR_CLKXM |
+				DAVINCI_MCBSP_PCR_CLKRM;
+			break;
+		case MCBSP_CLKR:
+			pcr |= DAVINCI_MCBSP_PCR_SCLKME;
+			break;
+		default:
+			dev_err(dev->dev, "bad clk_input_pin\n");
+			return -EINVAL;
+		}
+
+		break;
+	case SND_SOC_DAIFMT_BC_FC:
+		/* codec is master */
+		pcr = 0;
+		break;
+	default:
+		printk(KERN_ERR "%s:bad master\n", __func__);
+		return -EINVAL;
+	}
+
+	/* interface format */
+	switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
+	case SND_SOC_DAIFMT_I2S:
+		/* Davinci doesn't support TRUE I2S, but some codecs will have
+		 * the left and right channels contiguous. This allows
+		 * dsp_a mode to be used with an inverted normal frame clk.
+		 * If your codec is master and does not have contiguous
+		 * channels, then you will have sound on only one channel.
+		 * Try using a different mode, or codec as slave.
+		 *
+		 * The TLV320AIC33 is an example of a codec where this works.
+		 * It has a variable bit clock frequency allowing it to have
+		 * valid data on every bit clock.
+		 *
+		 * The TLV320AIC23 is an example of a codec where this does not
+		 * work. It has a fixed bit clock frequency with progressively
+		 * more empty bit clock slots between channels as the sample
+		 * rate is lowered.
+		 */
+		inv_fs = true;
+		fallthrough;
+	case SND_SOC_DAIFMT_DSP_A:
+		dev->mode = MOD_DSP_A;
+		break;
+	case SND_SOC_DAIFMT_DSP_B:
+		dev->mode = MOD_DSP_B;
+		break;
+	default:
+		printk(KERN_ERR "%s:bad format\n", __func__);
+		return -EINVAL;
+	}
+
+	switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
+	case SND_SOC_DAIFMT_NB_NF:
+		/* CLKRP Receive clock polarity,
+		 *	1 - sampled on rising edge of CLKR
+		 *	valid on rising edge
+		 * CLKXP Transmit clock polarity,
+		 *	1 - clocked on falling edge of CLKX
+		 *	valid on rising edge
+		 * FSRP  Receive frame sync pol, 0 - active high
+		 * FSXP  Transmit frame sync pol, 0 - active high
+		 */
+		pcr |= (DAVINCI_MCBSP_PCR_CLKXP | DAVINCI_MCBSP_PCR_CLKRP);
+		break;
+	case SND_SOC_DAIFMT_IB_IF:
+		/* CLKRP Receive clock polarity,
+		 *	0 - sampled on falling edge of CLKR
+		 *	valid on falling edge
+		 * CLKXP Transmit clock polarity,
+		 *	0 - clocked on rising edge of CLKX
+		 *	valid on falling edge
+		 * FSRP  Receive frame sync pol, 1 - active low
+		 * FSXP  Transmit frame sync pol, 1 - active low
+		 */
+		pcr |= (DAVINCI_MCBSP_PCR_FSXP | DAVINCI_MCBSP_PCR_FSRP);
+		break;
+	case SND_SOC_DAIFMT_NB_IF:
+		/* CLKRP Receive clock polarity,
+		 *	1 - sampled on rising edge of CLKR
+		 *	valid on rising edge
+		 * CLKXP Transmit clock polarity,
+		 *	1 - clocked on falling edge of CLKX
+		 *	valid on rising edge
+		 * FSRP  Receive frame sync pol, 1 - active low
+		 * FSXP  Transmit frame sync pol, 1 - active low
+		 */
+		pcr |= (DAVINCI_MCBSP_PCR_CLKXP | DAVINCI_MCBSP_PCR_CLKRP |
+			DAVINCI_MCBSP_PCR_FSXP | DAVINCI_MCBSP_PCR_FSRP);
+		break;
+	case SND_SOC_DAIFMT_IB_NF:
+		/* CLKRP Receive clock polarity,
+		 *	0 - sampled on falling edge of CLKR
+		 *	valid on falling edge
+		 * CLKXP Transmit clock polarity,
+		 *	0 - clocked on rising edge of CLKX
+		 *	valid on falling edge
+		 * FSRP  Receive frame sync pol, 0 - active high
+		 * FSXP  Transmit frame sync pol, 0 - active high
+		 */
+		break;
+	default:
+		return -EINVAL;
+	}
+	if (inv_fs == true)
+		pcr ^= (DAVINCI_MCBSP_PCR_FSXP | DAVINCI_MCBSP_PCR_FSRP);
+	davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_SRGR_REG, srgr);
+	dev->pcr = pcr;
+	davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_PCR_REG, pcr);
+	return 0;
+}
+
+static int davinci_i2s_dai_set_clkdiv(struct snd_soc_dai *cpu_dai,
+				int div_id, int div)
+{
+	struct davinci_mcbsp_dev *dev = snd_soc_dai_get_drvdata(cpu_dai);
+
+	if (div_id != DAVINCI_MCBSP_CLKGDV)
+		return -ENODEV;
+
+	dev->clk_div = div;
+	return 0;
+}
+
+static int davinci_i2s_hw_params(struct snd_pcm_substream *substream,
+				 struct snd_pcm_hw_params *params,
+				 struct snd_soc_dai *dai)
+{
+	struct davinci_mcbsp_dev *dev = snd_soc_dai_get_drvdata(dai);
+	struct snd_interval *i = NULL;
+	int mcbsp_word_length, master;
+	unsigned int rcr, xcr, srgr, clk_div, freq, framesize;
+	u32 spcr;
+	snd_pcm_format_t fmt;
+	unsigned element_cnt = 1;
+
+	/* general line settings */
+	spcr = davinci_mcbsp_read_reg(dev, DAVINCI_MCBSP_SPCR_REG);
+	if (substream->stream == SNDRV_PCM_STREAM_CAPTURE) {
+		spcr |= DAVINCI_MCBSP_SPCR_RINTM(3) | DAVINCI_MCBSP_SPCR_FREE;
+		davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_SPCR_REG, spcr);
+	} else {
+		spcr |= DAVINCI_MCBSP_SPCR_XINTM(3) | DAVINCI_MCBSP_SPCR_FREE;
+		davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_SPCR_REG, spcr);
+	}
+
+	master = dev->fmt & SND_SOC_DAIFMT_CLOCK_PROVIDER_MASK;
+	fmt = params_format(params);
+	mcbsp_word_length = asp_word_length[fmt];
+
+	switch (master) {
+	case SND_SOC_DAIFMT_BP_FP:
+		freq = clk_get_rate(dev->clk);
+		srgr = DAVINCI_MCBSP_SRGR_FSGM |
+		       DAVINCI_MCBSP_SRGR_CLKSM;
+		srgr |= DAVINCI_MCBSP_SRGR_FWID(mcbsp_word_length *
+						8 - 1);
+		if (dev->i2s_accurate_sck) {
+			clk_div = 256;
+			do {
+				framesize = (freq / (--clk_div)) /
+				params->rate_num *
+					params->rate_den;
+			} while (((framesize < 33) || (framesize > 4095)) &&
+				 (clk_div));
+			clk_div--;
+			srgr |= DAVINCI_MCBSP_SRGR_FPER(framesize - 1);
+		} else {
+			/* symmetric waveforms */
+			clk_div = freq / (mcbsp_word_length * 16) /
+				  params->rate_num * params->rate_den;
+			srgr |= DAVINCI_MCBSP_SRGR_FPER(mcbsp_word_length *
+							16 - 1);
+		}
+		clk_div &= 0xFF;
+		srgr |= clk_div;
+		break;
+	case SND_SOC_DAIFMT_BC_FP:
+		srgr = DAVINCI_MCBSP_SRGR_FSGM;
+		clk_div = dev->clk_div - 1;
+		srgr |= DAVINCI_MCBSP_SRGR_FWID(mcbsp_word_length * 8 - 1);
+		srgr |= DAVINCI_MCBSP_SRGR_FPER(mcbsp_word_length * 16 - 1);
+		clk_div &= 0xFF;
+		srgr |= clk_div;
+		break;
+	case SND_SOC_DAIFMT_BC_FC:
+		/* Clock and frame sync given from external sources */
+		i = hw_param_interval(params, SNDRV_PCM_HW_PARAM_SAMPLE_BITS);
+		srgr = DAVINCI_MCBSP_SRGR_FSGM;
+		srgr |= DAVINCI_MCBSP_SRGR_FWID(snd_interval_value(i) - 1);
+		pr_debug("%s - %d  FWID set: re-read srgr = %X\n",
+			__func__, __LINE__, snd_interval_value(i) - 1);
+
+		i = hw_param_interval(params, SNDRV_PCM_HW_PARAM_FRAME_BITS);
+		srgr |= DAVINCI_MCBSP_SRGR_FPER(snd_interval_value(i) - 1);
+		break;
+	default:
+		return -EINVAL;
+	}
+	davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_SRGR_REG, srgr);
+
+	rcr = DAVINCI_MCBSP_RCR_RFIG;
+	xcr = DAVINCI_MCBSP_XCR_XFIG;
+	if (dev->mode == MOD_DSP_B) {
+		rcr |= DAVINCI_MCBSP_RCR_RDATDLY(0);
+		xcr |= DAVINCI_MCBSP_XCR_XDATDLY(0);
+	} else {
+		rcr |= DAVINCI_MCBSP_RCR_RDATDLY(1);
+		xcr |= DAVINCI_MCBSP_XCR_XDATDLY(1);
+	}
+	/* Determine xfer data type */
+	fmt = params_format(params);
+	if ((fmt > SNDRV_PCM_FORMAT_S32_LE) || !data_type[fmt]) {
+		printk(KERN_WARNING "davinci-i2s: unsupported PCM format\n");
+		return -EINVAL;
+	}
+
+	if (params_channels(params) == 2) {
+		element_cnt = 2;
+		if (double_fmt[fmt] && dev->enable_channel_combine) {
+			element_cnt = 1;
+			fmt = double_fmt[fmt];
+		}
+		switch (master) {
+		case SND_SOC_DAIFMT_BP_FP:
+		case SND_SOC_DAIFMT_BP_FC:
+			rcr |= DAVINCI_MCBSP_RCR_RFRLEN2(0);
+			xcr |= DAVINCI_MCBSP_XCR_XFRLEN2(0);
+			rcr |= DAVINCI_MCBSP_RCR_RPHASE;
+			xcr |= DAVINCI_MCBSP_XCR_XPHASE;
+			break;
+		case SND_SOC_DAIFMT_BC_FC:
+		case SND_SOC_DAIFMT_BC_FP:
+			rcr |= DAVINCI_MCBSP_RCR_RFRLEN2(element_cnt - 1);
+			xcr |= DAVINCI_MCBSP_XCR_XFRLEN2(element_cnt - 1);
+			break;
+		default:
+			return -EINVAL;
+		}
+	}
+	mcbsp_word_length = asp_word_length[fmt];
+
+	switch (master) {
+	case SND_SOC_DAIFMT_BP_FP:
+	case SND_SOC_DAIFMT_BP_FC:
+		rcr |= DAVINCI_MCBSP_RCR_RFRLEN1(0);
+		xcr |= DAVINCI_MCBSP_XCR_XFRLEN1(0);
+		break;
+	case SND_SOC_DAIFMT_BC_FC:
+	case SND_SOC_DAIFMT_BC_FP:
+		rcr |= DAVINCI_MCBSP_RCR_RFRLEN1(element_cnt - 1);
+		xcr |= DAVINCI_MCBSP_XCR_XFRLEN1(element_cnt - 1);
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	rcr |= DAVINCI_MCBSP_RCR_RWDLEN1(mcbsp_word_length) |
+		DAVINCI_MCBSP_RCR_RWDLEN2(mcbsp_word_length);
+	xcr |= DAVINCI_MCBSP_XCR_XWDLEN1(mcbsp_word_length) |
+		DAVINCI_MCBSP_XCR_XWDLEN2(mcbsp_word_length);
+
+	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+		davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_XCR_REG, xcr);
+	else
+		davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_RCR_REG, rcr);
+
+	pr_debug("%s - %d  srgr=%X\n", __func__, __LINE__, srgr);
+	pr_debug("%s - %d  xcr=%X\n", __func__, __LINE__, xcr);
+	pr_debug("%s - %d  rcr=%X\n", __func__, __LINE__, rcr);
+	return 0;
+}
+
+static int davinci_i2s_prepare(struct snd_pcm_substream *substream,
+		struct snd_soc_dai *dai)
+{
+	struct davinci_mcbsp_dev *dev = snd_soc_dai_get_drvdata(dai);
+	int playback = (substream->stream == SNDRV_PCM_STREAM_PLAYBACK);
+	u32 spcr;
+	u32 mask = playback ? DAVINCI_MCBSP_SPCR_XRST : DAVINCI_MCBSP_SPCR_RRST;
+
+	davinci_mcbsp_stop(dev, playback);
+
+	spcr = davinci_mcbsp_read_reg(dev, DAVINCI_MCBSP_SPCR_REG);
+	if (spcr & mask) {
+		/* start off disabled */
+		davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_SPCR_REG,
+					spcr & ~mask);
+		toggle_clock(dev, playback);
+	}
+	if (dev->pcr & (DAVINCI_MCBSP_PCR_FSXM | DAVINCI_MCBSP_PCR_FSRM |
+			DAVINCI_MCBSP_PCR_CLKXM | DAVINCI_MCBSP_PCR_CLKRM)) {
+		/* Start the sample generator */
+		spcr |= DAVINCI_MCBSP_SPCR_GRST;
+		davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_SPCR_REG, spcr);
+	}
+
+	if (playback) {
+		/* Enable the transmitter */
+		spcr = davinci_mcbsp_read_reg(dev, DAVINCI_MCBSP_SPCR_REG);
+		spcr |= DAVINCI_MCBSP_SPCR_XRST;
+		davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_SPCR_REG, spcr);
+
+		/* wait for any unexpected frame sync error to occur */
+		udelay(100);
+
+		/* Disable the transmitter to clear any outstanding XSYNCERR */
+		spcr = davinci_mcbsp_read_reg(dev, DAVINCI_MCBSP_SPCR_REG);
+		spcr &= ~DAVINCI_MCBSP_SPCR_XRST;
+		davinci_mcbsp_write_reg(dev, DAVINCI_MCBSP_SPCR_REG, spcr);
+		toggle_clock(dev, playback);
+	}
+
+	return 0;
+}
+
+static int davinci_i2s_trigger(struct snd_pcm_substream *substream, int cmd,
+			       struct snd_soc_dai *dai)
+{
+	struct davinci_mcbsp_dev *dev = snd_soc_dai_get_drvdata(dai);
+	int ret = 0;
+	int playback = (substream->stream == SNDRV_PCM_STREAM_PLAYBACK);
+
+	switch (cmd) {
+	case SNDRV_PCM_TRIGGER_START:
+	case SNDRV_PCM_TRIGGER_RESUME:
+	case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
+		davinci_mcbsp_start(dev, substream);
+		break;
+	case SNDRV_PCM_TRIGGER_STOP:
+	case SNDRV_PCM_TRIGGER_SUSPEND:
+	case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
+		davinci_mcbsp_stop(dev, playback);
+		break;
+	default:
+		ret = -EINVAL;
+	}
+	return ret;
+}
+
+static void davinci_i2s_shutdown(struct snd_pcm_substream *substream,
+		struct snd_soc_dai *dai)
+{
+	struct davinci_mcbsp_dev *dev = snd_soc_dai_get_drvdata(dai);
+	int playback = (substream->stream == SNDRV_PCM_STREAM_PLAYBACK);
+	davinci_mcbsp_stop(dev, playback);
+}
+
+#define DAVINCI_I2S_RATES	SNDRV_PCM_RATE_8000_96000
+#define DAVINCI_I2S_FORMATS	(SNDRV_PCM_FMTBIT_S16_LE | \
+				 SNDRV_PCM_FMTBIT_S32_LE)
+
+static const struct snd_soc_dai_ops davinci_i2s_dai_ops = {
+	.shutdown	= davinci_i2s_shutdown,
+	.prepare	= davinci_i2s_prepare,
+	.trigger	= davinci_i2s_trigger,
+	.hw_params	= davinci_i2s_hw_params,
+	.set_fmt	= davinci_i2s_set_dai_fmt,
+	.set_clkdiv	= davinci_i2s_dai_set_clkdiv,
+
+};
+
+static int davinci_i2s_dai_probe(struct snd_soc_dai *dai)
+{
+	struct davinci_mcbsp_dev *dev = snd_soc_dai_get_drvdata(dai);
+
+	dai->playback_dma_data = &dev->dma_data[SNDRV_PCM_STREAM_PLAYBACK];
+	dai->capture_dma_data = &dev->dma_data[SNDRV_PCM_STREAM_CAPTURE];
+
+	return 0;
+}
+
+static struct snd_soc_dai_driver davinci_i2s_dai = {
+	.probe = davinci_i2s_dai_probe,
+	.playback = {
+		.channels_min = 2,
+		.channels_max = 2,
+		.rates = DAVINCI_I2S_RATES,
+		.formats = DAVINCI_I2S_FORMATS,
+	},
+	.capture = {
+		.channels_min = 2,
+		.channels_max = 2,
+		.rates = DAVINCI_I2S_RATES,
+		.formats = DAVINCI_I2S_FORMATS,
+	},
+	.ops = &davinci_i2s_dai_ops,
+
+};
+
+static const struct snd_soc_component_driver davinci_i2s_component = {
+	.name			= DRV_NAME,
+	.legacy_dai_naming	= 1,
+};
+
+static int davinci_i2s_probe(struct platform_device *pdev)
+{
+	struct snd_dmaengine_dai_dma_data *dma_data;
+	struct davinci_mcbsp_dev *dev;
+	struct resource *mem, *res;
+	void __iomem *io_base;
+	int *dma;
+	int ret;
+
+	mem = platform_get_resource_byname(pdev, IORESOURCE_MEM, "mpu");
+	if (!mem) {
+		dev_warn(&pdev->dev,
+			 "\"mpu\" mem resource not found, using index 0\n");
+		mem = platform_get_resource(pdev, IORESOURCE_MEM, 0);
+		if (!mem) {
+			dev_err(&pdev->dev, "no mem resource?\n");
+			return -ENODEV;
+		}
+	}
+
+	io_base = devm_ioremap_resource(&pdev->dev, mem);
+	if (IS_ERR(io_base))
+		return PTR_ERR(io_base);
+
+	dev = devm_kzalloc(&pdev->dev, sizeof(struct davinci_mcbsp_dev),
+			   GFP_KERNEL);
+	if (!dev)
+		return -ENOMEM;
+
+	dev->base = io_base;
+
+	/* setup DMA, first TX, then RX */
+	dma_data = &dev->dma_data[SNDRV_PCM_STREAM_PLAYBACK];
+	dma_data->addr = (dma_addr_t)(mem->start + DAVINCI_MCBSP_DXR_REG);
+
+	res = platform_get_resource(pdev, IORESOURCE_DMA, 0);
+	if (res) {
+		dma = &dev->dma_request[SNDRV_PCM_STREAM_PLAYBACK];
+		*dma = res->start;
+		dma_data->filter_data = dma;
+	} else if (IS_ENABLED(CONFIG_OF) && pdev->dev.of_node) {
+		dma_data->filter_data = "tx";
+	} else {
+		dev_err(&pdev->dev, "Missing DMA tx resource\n");
+		return -ENODEV;
+	}
+
+	dma_data = &dev->dma_data[SNDRV_PCM_STREAM_CAPTURE];
+	dma_data->addr = (dma_addr_t)(mem->start + DAVINCI_MCBSP_DRR_REG);
+
+	res = platform_get_resource(pdev, IORESOURCE_DMA, 1);
+	if (res) {
+		dma = &dev->dma_request[SNDRV_PCM_STREAM_CAPTURE];
+		*dma = res->start;
+		dma_data->filter_data = dma;
+	} else if (IS_ENABLED(CONFIG_OF) && pdev->dev.of_node) {
+		dma_data->filter_data = "rx";
+	} else {
+		dev_err(&pdev->dev, "Missing DMA rx resource\n");
+		return -ENODEV;
+	}
+
+	dev->clk = clk_get(&pdev->dev, NULL);
+	if (IS_ERR(dev->clk))
+		return -ENODEV;
+	ret = clk_enable(dev->clk);
+	if (ret)
+		goto err_put_clk;
+
+	dev->dev = &pdev->dev;
+	dev_set_drvdata(&pdev->dev, dev);
+
+	ret = snd_soc_register_component(&pdev->dev, &davinci_i2s_component,
+					 &davinci_i2s_dai, 1);
+	if (ret != 0)
+		goto err_release_clk;
+
+	ret = edma_pcm_platform_register(&pdev->dev);
+	if (ret) {
+		dev_err(&pdev->dev, "register PCM failed: %d\n", ret);
+		goto err_unregister_component;
+	}
+
+	return 0;
+
+err_unregister_component:
+	snd_soc_unregister_component(&pdev->dev);
+err_release_clk:
+	clk_disable(dev->clk);
+err_put_clk:
+	clk_put(dev->clk);
+	return ret;
+}
+
+static int davinci_i2s_remove(struct platform_device *pdev)
+{
+	struct davinci_mcbsp_dev *dev = dev_get_drvdata(&pdev->dev);
+
+	snd_soc_unregister_component(&pdev->dev);
+
+	clk_disable(dev->clk);
+	clk_put(dev->clk);
+	dev->clk = NULL;
+
+	return 0;
+}
+
+static const struct of_device_id davinci_i2s_match[] __maybe_unused = {
+	{ .compatible = "ti,da850-mcbsp" },
+	{},
+};
+MODULE_DEVICE_TABLE(of, davinci_i2s_match);
+
+static struct platform_driver davinci_mcbsp_driver = {
+	.probe		= davinci_i2s_probe,
+	.remove		= davinci_i2s_remove,
+	.driver		= {
+		.name	= "davinci-mcbsp",
+		.of_match_table = of_match_ptr(davinci_i2s_match),
+	},
+};
+
+module_platform_driver(davinci_mcbsp_driver);
+
+MODULE_AUTHOR("Vladimir Barinov");
+MODULE_DESCRIPTION("TI DAVINCI I2S (McBSP) SoC Interface");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/ti/davinci-i2s.h b/sound/soc/ti/davinci-i2s.h
new file mode 100644
index 000000000..88d4df1d1
--- /dev/null
+++ b/sound/soc/ti/davinci-i2s.h
@@ -0,0 +1,17 @@
+/* SPDX-License-Identifier: GPL-2.0-only */
+/*
+ * ALSA SoC I2S (McBSP) Audio Layer for TI DAVINCI processor
+ *
+ * Author:      Vladimir Barinov, <vbarinov@embeddedalley.com>
+ * Copyright:   (C) 2007 MontaVista Software, Inc., <source@mvista.com>
+ */
+
+#ifndef _DAVINCI_I2S_H
+#define _DAVINCI_I2S_H
+
+/* McBSP dividers */
+enum davinci_mcbsp_div {
+	DAVINCI_MCBSP_CLKGDV,              /* Sample rate generator divider */
+};
+
+#endif
diff --git a/sound/soc/ti/davinci-mcasp.c b/sound/soc/ti/davinci-mcasp.c
new file mode 100644
index 000000000..ca5d1bb6a
--- /dev/null
+++ b/sound/soc/ti/davinci-mcasp.c
@@ -0,0 +1,2546 @@
+// SPDX-License-Identifier: GPL-2.0-only
+/*
+ * ALSA SoC McASP Audio Layer for TI DAVINCI processor
+ *
+ * Multi-channel Audio Serial Port Driver
+ *
+ * Author: Nirmal Pandey <n-pandey@ti.com>,
+ *         Suresh Rajashekara <suresh.r@ti.com>
+ *         Steve Chen <schen@.mvista.com>
+ *
+ * Copyright:   (C) 2009 MontaVista Software, Inc., <source@mvista.com>
+ * Copyright:   (C) 2009  Texas Instruments, India
+ */
+
+#include <linux/init.h>
+#include <linux/module.h>
+#include <linux/device.h>
+#include <linux/slab.h>
+#include <linux/delay.h>
+#include <linux/io.h>
+#include <linux/clk.h>
+#include <linux/pm_runtime.h>
+#include <linux/of.h>
+#include <linux/of_platform.h>
+#include <linux/of_device.h>
+#include <linux/platform_data/davinci_asp.h>
+#include <linux/math64.h>
+#include <linux/bitmap.h>
+#include <linux/gpio/driver.h>
+
+#include <sound/asoundef.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 "edma-pcm.h"
+#include "sdma-pcm.h"
+#include "udma-pcm.h"
+#include "davinci-mcasp.h"
+
+#define MCASP_MAX_AFIFO_DEPTH	64
+
+#ifdef CONFIG_PM
+static u32 context_regs[] = {
+	DAVINCI_MCASP_TXFMCTL_REG,
+	DAVINCI_MCASP_RXFMCTL_REG,
+	DAVINCI_MCASP_TXFMT_REG,
+	DAVINCI_MCASP_RXFMT_REG,
+	DAVINCI_MCASP_ACLKXCTL_REG,
+	DAVINCI_MCASP_ACLKRCTL_REG,
+	DAVINCI_MCASP_AHCLKXCTL_REG,
+	DAVINCI_MCASP_AHCLKRCTL_REG,
+	DAVINCI_MCASP_PDIR_REG,
+	DAVINCI_MCASP_PFUNC_REG,
+	DAVINCI_MCASP_RXMASK_REG,
+	DAVINCI_MCASP_TXMASK_REG,
+	DAVINCI_MCASP_RXTDM_REG,
+	DAVINCI_MCASP_TXTDM_REG,
+};
+
+struct davinci_mcasp_context {
+	u32	config_regs[ARRAY_SIZE(context_regs)];
+	u32	afifo_regs[2]; /* for read/write fifo control registers */
+	u32	*xrsr_regs; /* for serializer configuration */
+	bool	pm_state;
+};
+#endif
+
+struct davinci_mcasp_ruledata {
+	struct davinci_mcasp *mcasp;
+	int serializers;
+};
+
+struct davinci_mcasp {
+	struct snd_dmaengine_dai_dma_data dma_data[2];
+	struct davinci_mcasp_pdata *pdata;
+	void __iomem *base;
+	u32 fifo_base;
+	struct device *dev;
+	struct snd_pcm_substream *substreams[2];
+	unsigned int dai_fmt;
+
+	u32 iec958_status;
+
+	/* Audio can not be enabled due to missing parameter(s) */
+	bool	missing_audio_param;
+
+	/* McASP specific data */
+	int	tdm_slots;
+	u32	tdm_mask[2];
+	int	slot_width;
+	u8	op_mode;
+	u8	dismod;
+	u8	num_serializer;
+	u8	*serial_dir;
+	u8	version;
+	u8	bclk_div;
+	int	streams;
+	u32	irq_request[2];
+
+	int	sysclk_freq;
+	bool	bclk_master;
+	u32	auxclk_fs_ratio;
+
+	unsigned long pdir; /* Pin direction bitfield */
+
+	/* McASP FIFO related */
+	u8	txnumevt;
+	u8	rxnumevt;
+
+	bool	dat_port;
+
+	/* Used for comstraint setting on the second stream */
+	u32	channels;
+	int	max_format_width;
+	u8	active_serializers[2];
+
+#ifdef CONFIG_GPIOLIB
+	struct gpio_chip gpio_chip;
+#endif
+
+#ifdef CONFIG_PM
+	struct davinci_mcasp_context context;
+#endif
+
+	struct davinci_mcasp_ruledata ruledata[2];
+	struct snd_pcm_hw_constraint_list chconstr[2];
+};
+
+static inline void mcasp_set_bits(struct davinci_mcasp *mcasp, u32 offset,
+				  u32 val)
+{
+	void __iomem *reg = mcasp->base + offset;
+	__raw_writel(__raw_readl(reg) | val, reg);
+}
+
+static inline void mcasp_clr_bits(struct davinci_mcasp *mcasp, u32 offset,
+				  u32 val)
+{
+	void __iomem *reg = mcasp->base + offset;
+	__raw_writel((__raw_readl(reg) & ~(val)), reg);
+}
+
+static inline void mcasp_mod_bits(struct davinci_mcasp *mcasp, u32 offset,
+				  u32 val, u32 mask)
+{
+	void __iomem *reg = mcasp->base + offset;
+	__raw_writel((__raw_readl(reg) & ~mask) | val, reg);
+}
+
+static inline void mcasp_set_reg(struct davinci_mcasp *mcasp, u32 offset,
+				 u32 val)
+{
+	__raw_writel(val, mcasp->base + offset);
+}
+
+static inline u32 mcasp_get_reg(struct davinci_mcasp *mcasp, u32 offset)
+{
+	return (u32)__raw_readl(mcasp->base + offset);
+}
+
+static void mcasp_set_ctl_reg(struct davinci_mcasp *mcasp, u32 ctl_reg, u32 val)
+{
+	int i = 0;
+
+	mcasp_set_bits(mcasp, ctl_reg, val);
+
+	/* programming GBLCTL needs to read back from GBLCTL and verfiy */
+	/* loop count is to avoid the lock-up */
+	for (i = 0; i < 1000; i++) {
+		if ((mcasp_get_reg(mcasp, ctl_reg) & val) == val)
+			break;
+	}
+
+	if (i == 1000 && ((mcasp_get_reg(mcasp, ctl_reg) & val) != val))
+		printk(KERN_ERR "GBLCTL write error\n");
+}
+
+static bool mcasp_is_synchronous(struct davinci_mcasp *mcasp)
+{
+	u32 rxfmctl = mcasp_get_reg(mcasp, DAVINCI_MCASP_RXFMCTL_REG);
+	u32 aclkxctl = mcasp_get_reg(mcasp, DAVINCI_MCASP_ACLKXCTL_REG);
+
+	return !(aclkxctl & TX_ASYNC) && rxfmctl & AFSRE;
+}
+
+static inline void mcasp_set_clk_pdir(struct davinci_mcasp *mcasp, bool enable)
+{
+	u32 bit = PIN_BIT_AMUTE;
+
+	for_each_set_bit_from(bit, &mcasp->pdir, PIN_BIT_AFSR + 1) {
+		if (enable)
+			mcasp_set_bits(mcasp, DAVINCI_MCASP_PDIR_REG, BIT(bit));
+		else
+			mcasp_clr_bits(mcasp, DAVINCI_MCASP_PDIR_REG, BIT(bit));
+	}
+}
+
+static inline void mcasp_set_axr_pdir(struct davinci_mcasp *mcasp, bool enable)
+{
+	u32 bit;
+
+	for_each_set_bit(bit, &mcasp->pdir, PIN_BIT_AMUTE) {
+		if (enable)
+			mcasp_set_bits(mcasp, DAVINCI_MCASP_PDIR_REG, BIT(bit));
+		else
+			mcasp_clr_bits(mcasp, DAVINCI_MCASP_PDIR_REG, BIT(bit));
+	}
+}
+
+static void mcasp_start_rx(struct davinci_mcasp *mcasp)
+{
+	if (mcasp->rxnumevt) {	/* enable FIFO */
+		u32 reg = mcasp->fifo_base + MCASP_RFIFOCTL_OFFSET;
+
+		mcasp_clr_bits(mcasp, reg, FIFO_ENABLE);
+		mcasp_set_bits(mcasp, reg, FIFO_ENABLE);
+	}
+
+	/* Start clocks */
+	mcasp_set_ctl_reg(mcasp, DAVINCI_MCASP_GBLCTLR_REG, RXHCLKRST);
+	mcasp_set_ctl_reg(mcasp, DAVINCI_MCASP_GBLCTLR_REG, RXCLKRST);
+	/*
+	 * When ASYNC == 0 the transmit and receive sections operate
+	 * synchronously from the transmit clock and frame sync. We need to make
+	 * sure that the TX signlas are enabled when starting reception.
+	 */
+	if (mcasp_is_synchronous(mcasp)) {
+		mcasp_set_ctl_reg(mcasp, DAVINCI_MCASP_GBLCTLX_REG, TXHCLKRST);
+		mcasp_set_ctl_reg(mcasp, DAVINCI_MCASP_GBLCTLX_REG, TXCLKRST);
+		mcasp_set_clk_pdir(mcasp, true);
+	}
+
+	/* Activate serializer(s) */
+	mcasp_set_reg(mcasp, DAVINCI_MCASP_RXSTAT_REG, 0xFFFFFFFF);
+	mcasp_set_ctl_reg(mcasp, DAVINCI_MCASP_GBLCTLR_REG, RXSERCLR);
+	/* Release RX state machine */
+	mcasp_set_ctl_reg(mcasp, DAVINCI_MCASP_GBLCTLR_REG, RXSMRST);
+	/* Release Frame Sync generator */
+	mcasp_set_ctl_reg(mcasp, DAVINCI_MCASP_GBLCTLR_REG, RXFSRST);
+	if (mcasp_is_synchronous(mcasp))
+		mcasp_set_ctl_reg(mcasp, DAVINCI_MCASP_GBLCTLX_REG, TXFSRST);
+
+	/* enable receive IRQs */
+	mcasp_set_bits(mcasp, DAVINCI_MCASP_EVTCTLR_REG,
+		       mcasp->irq_request[SNDRV_PCM_STREAM_CAPTURE]);
+}
+
+static void mcasp_start_tx(struct davinci_mcasp *mcasp)
+{
+	u32 cnt;
+
+	if (mcasp->txnumevt) {	/* enable FIFO */
+		u32 reg = mcasp->fifo_base + MCASP_WFIFOCTL_OFFSET;
+
+		mcasp_clr_bits(mcasp, reg, FIFO_ENABLE);
+		mcasp_set_bits(mcasp, reg, FIFO_ENABLE);
+	}
+
+	/* Start clocks */
+	mcasp_set_ctl_reg(mcasp, DAVINCI_MCASP_GBLCTLX_REG, TXHCLKRST);
+	mcasp_set_ctl_reg(mcasp, DAVINCI_MCASP_GBLCTLX_REG, TXCLKRST);
+	mcasp_set_clk_pdir(mcasp, true);
+
+	/* Activate serializer(s) */
+	mcasp_set_reg(mcasp, DAVINCI_MCASP_TXSTAT_REG, 0xFFFFFFFF);
+	mcasp_set_ctl_reg(mcasp, DAVINCI_MCASP_GBLCTLX_REG, TXSERCLR);
+
+	/* wait for XDATA to be cleared */
+	cnt = 0;
+	while ((mcasp_get_reg(mcasp, DAVINCI_MCASP_TXSTAT_REG) & XRDATA) &&
+	       (cnt < 100000))
+		cnt++;
+
+	mcasp_set_axr_pdir(mcasp, true);
+
+	/* Release TX state machine */
+	mcasp_set_ctl_reg(mcasp, DAVINCI_MCASP_GBLCTLX_REG, TXSMRST);
+	/* Release Frame Sync generator */
+	mcasp_set_ctl_reg(mcasp, DAVINCI_MCASP_GBLCTLX_REG, TXFSRST);
+
+	/* enable transmit IRQs */
+	mcasp_set_bits(mcasp, DAVINCI_MCASP_EVTCTLX_REG,
+		       mcasp->irq_request[SNDRV_PCM_STREAM_PLAYBACK]);
+}
+
+static void davinci_mcasp_start(struct davinci_mcasp *mcasp, int stream)
+{
+	mcasp->streams++;
+
+	if (stream == SNDRV_PCM_STREAM_PLAYBACK)
+		mcasp_start_tx(mcasp);
+	else
+		mcasp_start_rx(mcasp);
+}
+
+static void mcasp_stop_rx(struct davinci_mcasp *mcasp)
+{
+	/* disable IRQ sources */
+	mcasp_clr_bits(mcasp, DAVINCI_MCASP_EVTCTLR_REG,
+		       mcasp->irq_request[SNDRV_PCM_STREAM_CAPTURE]);
+
+	/*
+	 * In synchronous mode stop the TX clocks if no other stream is
+	 * running
+	 */
+	if (mcasp_is_synchronous(mcasp) && !mcasp->streams) {
+		mcasp_set_clk_pdir(mcasp, false);
+		mcasp_set_reg(mcasp, DAVINCI_MCASP_GBLCTLX_REG, 0);
+	}
+
+	mcasp_set_reg(mcasp, DAVINCI_MCASP_GBLCTLR_REG, 0);
+	mcasp_set_reg(mcasp, DAVINCI_MCASP_RXSTAT_REG, 0xFFFFFFFF);
+
+	if (mcasp->rxnumevt) {	/* disable FIFO */
+		u32 reg = mcasp->fifo_base + MCASP_RFIFOCTL_OFFSET;
+
+		mcasp_clr_bits(mcasp, reg, FIFO_ENABLE);
+	}
+}
+
+static void mcasp_stop_tx(struct davinci_mcasp *mcasp)
+{
+	u32 val = 0;
+
+	/* disable IRQ sources */
+	mcasp_clr_bits(mcasp, DAVINCI_MCASP_EVTCTLX_REG,
+		       mcasp->irq_request[SNDRV_PCM_STREAM_PLAYBACK]);
+
+	/*
+	 * In synchronous mode keep TX clocks running if the capture stream is
+	 * still running.
+	 */
+	if (mcasp_is_synchronous(mcasp) && mcasp->streams)
+		val =  TXHCLKRST | TXCLKRST | TXFSRST;
+	else
+		mcasp_set_clk_pdir(mcasp, false);
+
+
+	mcasp_set_reg(mcasp, DAVINCI_MCASP_GBLCTLX_REG, val);
+	mcasp_set_reg(mcasp, DAVINCI_MCASP_TXSTAT_REG, 0xFFFFFFFF);
+
+	if (mcasp->txnumevt) {	/* disable FIFO */
+		u32 reg = mcasp->fifo_base + MCASP_WFIFOCTL_OFFSET;
+
+		mcasp_clr_bits(mcasp, reg, FIFO_ENABLE);
+	}
+
+	mcasp_set_axr_pdir(mcasp, false);
+}
+
+static void davinci_mcasp_stop(struct davinci_mcasp *mcasp, int stream)
+{
+	mcasp->streams--;
+
+	if (stream == SNDRV_PCM_STREAM_PLAYBACK)
+		mcasp_stop_tx(mcasp);
+	else
+		mcasp_stop_rx(mcasp);
+}
+
+static irqreturn_t davinci_mcasp_tx_irq_handler(int irq, void *data)
+{
+	struct davinci_mcasp *mcasp = (struct davinci_mcasp *)data;
+	struct snd_pcm_substream *substream;
+	u32 irq_mask = mcasp->irq_request[SNDRV_PCM_STREAM_PLAYBACK];
+	u32 handled_mask = 0;
+	u32 stat;
+
+	stat = mcasp_get_reg(mcasp, DAVINCI_MCASP_TXSTAT_REG);
+	if (stat & XUNDRN & irq_mask) {
+		dev_warn(mcasp->dev, "Transmit buffer underflow\n");
+		handled_mask |= XUNDRN;
+
+		substream = mcasp->substreams[SNDRV_PCM_STREAM_PLAYBACK];
+		if (substream)
+			snd_pcm_stop_xrun(substream);
+	}
+
+	if (!handled_mask)
+		dev_warn(mcasp->dev, "unhandled tx event. txstat: 0x%08x\n",
+			 stat);
+
+	if (stat & XRERR)
+		handled_mask |= XRERR;
+
+	/* Ack the handled event only */
+	mcasp_set_reg(mcasp, DAVINCI_MCASP_TXSTAT_REG, handled_mask);
+
+	return IRQ_RETVAL(handled_mask);
+}
+
+static irqreturn_t davinci_mcasp_rx_irq_handler(int irq, void *data)
+{
+	struct davinci_mcasp *mcasp = (struct davinci_mcasp *)data;
+	struct snd_pcm_substream *substream;
+	u32 irq_mask = mcasp->irq_request[SNDRV_PCM_STREAM_CAPTURE];
+	u32 handled_mask = 0;
+	u32 stat;
+
+	stat = mcasp_get_reg(mcasp, DAVINCI_MCASP_RXSTAT_REG);
+	if (stat & ROVRN & irq_mask) {
+		dev_warn(mcasp->dev, "Receive buffer overflow\n");
+		handled_mask |= ROVRN;
+
+		substream = mcasp->substreams[SNDRV_PCM_STREAM_CAPTURE];
+		if (substream)
+			snd_pcm_stop_xrun(substream);
+	}
+
+	if (!handled_mask)
+		dev_warn(mcasp->dev, "unhandled rx event. rxstat: 0x%08x\n",
+			 stat);
+
+	if (stat & XRERR)
+		handled_mask |= XRERR;
+
+	/* Ack the handled event only */
+	mcasp_set_reg(mcasp, DAVINCI_MCASP_RXSTAT_REG, handled_mask);
+
+	return IRQ_RETVAL(handled_mask);
+}
+
+static irqreturn_t davinci_mcasp_common_irq_handler(int irq, void *data)
+{
+	struct davinci_mcasp *mcasp = (struct davinci_mcasp *)data;
+	irqreturn_t ret = IRQ_NONE;
+
+	if (mcasp->substreams[SNDRV_PCM_STREAM_PLAYBACK])
+		ret = davinci_mcasp_tx_irq_handler(irq, data);
+
+	if (mcasp->substreams[SNDRV_PCM_STREAM_CAPTURE])
+		ret |= davinci_mcasp_rx_irq_handler(irq, data);
+
+	return ret;
+}
+
+static int davinci_mcasp_set_dai_fmt(struct snd_soc_dai *cpu_dai,
+					 unsigned int fmt)
+{
+	struct davinci_mcasp *mcasp = snd_soc_dai_get_drvdata(cpu_dai);
+	int ret = 0;
+	u32 data_delay;
+	bool fs_pol_rising;
+	bool inv_fs = false;
+
+	if (!fmt)
+		return 0;
+
+	pm_runtime_get_sync(mcasp->dev);
+	switch (fmt & SND_SOC_DAIFMT_FORMAT_MASK) {
+	case SND_SOC_DAIFMT_DSP_A:
+		mcasp_clr_bits(mcasp, DAVINCI_MCASP_TXFMCTL_REG, FSXDUR);
+		mcasp_clr_bits(mcasp, DAVINCI_MCASP_RXFMCTL_REG, FSRDUR);
+		/* 1st data bit occur one ACLK cycle after the frame sync */
+		data_delay = 1;
+		break;
+	case SND_SOC_DAIFMT_DSP_B:
+	case SND_SOC_DAIFMT_AC97:
+		mcasp_clr_bits(mcasp, DAVINCI_MCASP_TXFMCTL_REG, FSXDUR);
+		mcasp_clr_bits(mcasp, DAVINCI_MCASP_RXFMCTL_REG, FSRDUR);
+		/* No delay after FS */
+		data_delay = 0;
+		break;
+	case SND_SOC_DAIFMT_I2S:
+		/* configure a full-word SYNC pulse (LRCLK) */
+		mcasp_set_bits(mcasp, DAVINCI_MCASP_TXFMCTL_REG, FSXDUR);
+		mcasp_set_bits(mcasp, DAVINCI_MCASP_RXFMCTL_REG, FSRDUR);
+		/* 1st data bit occur one ACLK cycle after the frame sync */
+		data_delay = 1;
+		/* FS need to be inverted */
+		inv_fs = true;
+		break;
+	case SND_SOC_DAIFMT_RIGHT_J:
+	case SND_SOC_DAIFMT_LEFT_J:
+		/* configure a full-word SYNC pulse (LRCLK) */
+		mcasp_set_bits(mcasp, DAVINCI_MCASP_TXFMCTL_REG, FSXDUR);
+		mcasp_set_bits(mcasp, DAVINCI_MCASP_RXFMCTL_REG, FSRDUR);
+		/* No delay after FS */
+		data_delay = 0;
+		break;
+	default:
+		ret = -EINVAL;
+		goto out;
+	}
+
+	mcasp_mod_bits(mcasp, DAVINCI_MCASP_TXFMT_REG, FSXDLY(data_delay),
+		       FSXDLY(3));
+	mcasp_mod_bits(mcasp, DAVINCI_MCASP_RXFMT_REG, FSRDLY(data_delay),
+		       FSRDLY(3));
+
+	switch (fmt & SND_SOC_DAIFMT_CLOCK_PROVIDER_MASK) {
+	case SND_SOC_DAIFMT_BP_FP:
+		/* codec is clock and frame slave */
+		mcasp_set_bits(mcasp, DAVINCI_MCASP_ACLKXCTL_REG, ACLKXE);
+		mcasp_set_bits(mcasp, DAVINCI_MCASP_TXFMCTL_REG, AFSXE);
+
+		mcasp_set_bits(mcasp, DAVINCI_MCASP_ACLKRCTL_REG, ACLKRE);
+		mcasp_set_bits(mcasp, DAVINCI_MCASP_RXFMCTL_REG, AFSRE);
+
+		/* BCLK */
+		set_bit(PIN_BIT_ACLKX, &mcasp->pdir);
+		set_bit(PIN_BIT_ACLKR, &mcasp->pdir);
+		/* Frame Sync */
+		set_bit(PIN_BIT_AFSX, &mcasp->pdir);
+		set_bit(PIN_BIT_AFSR, &mcasp->pdir);
+
+		mcasp->bclk_master = 1;
+		break;
+	case SND_SOC_DAIFMT_BP_FC:
+		/* codec is clock slave and frame master */
+		mcasp_set_bits(mcasp, DAVINCI_MCASP_ACLKXCTL_REG, ACLKXE);
+		mcasp_clr_bits(mcasp, DAVINCI_MCASP_TXFMCTL_REG, AFSXE);
+
+		mcasp_set_bits(mcasp, DAVINCI_MCASP_ACLKRCTL_REG, ACLKRE);
+		mcasp_clr_bits(mcasp, DAVINCI_MCASP_RXFMCTL_REG, AFSRE);
+
+		/* BCLK */
+		set_bit(PIN_BIT_ACLKX, &mcasp->pdir);
+		set_bit(PIN_BIT_ACLKR, &mcasp->pdir);
+		/* Frame Sync */
+		clear_bit(PIN_BIT_AFSX, &mcasp->pdir);
+		clear_bit(PIN_BIT_AFSR, &mcasp->pdir);
+
+		mcasp->bclk_master = 1;
+		break;
+	case SND_SOC_DAIFMT_BC_FP:
+		/* codec is clock master and frame slave */
+		mcasp_clr_bits(mcasp, DAVINCI_MCASP_ACLKXCTL_REG, ACLKXE);
+		mcasp_set_bits(mcasp, DAVINCI_MCASP_TXFMCTL_REG, AFSXE);
+
+		mcasp_clr_bits(mcasp, DAVINCI_MCASP_ACLKRCTL_REG, ACLKRE);
+		mcasp_set_bits(mcasp, DAVINCI_MCASP_RXFMCTL_REG, AFSRE);
+
+		/* BCLK */
+		clear_bit(PIN_BIT_ACLKX, &mcasp->pdir);
+		clear_bit(PIN_BIT_ACLKR, &mcasp->pdir);
+		/* Frame Sync */
+		set_bit(PIN_BIT_AFSX, &mcasp->pdir);
+		set_bit(PIN_BIT_AFSR, &mcasp->pdir);
+
+		mcasp->bclk_master = 0;
+		break;
+	case SND_SOC_DAIFMT_BC_FC:
+		/* codec is clock and frame master */
+		mcasp_clr_bits(mcasp, DAVINCI_MCASP_ACLKXCTL_REG, ACLKXE);
+		mcasp_clr_bits(mcasp, DAVINCI_MCASP_TXFMCTL_REG, AFSXE);
+
+		mcasp_clr_bits(mcasp, DAVINCI_MCASP_ACLKRCTL_REG, ACLKRE);
+		mcasp_clr_bits(mcasp, DAVINCI_MCASP_RXFMCTL_REG, AFSRE);
+
+		/* BCLK */
+		clear_bit(PIN_BIT_ACLKX, &mcasp->pdir);
+		clear_bit(PIN_BIT_ACLKR, &mcasp->pdir);
+		/* Frame Sync */
+		clear_bit(PIN_BIT_AFSX, &mcasp->pdir);
+		clear_bit(PIN_BIT_AFSR, &mcasp->pdir);
+
+		mcasp->bclk_master = 0;
+		break;
+	default:
+		ret = -EINVAL;
+		goto out;
+	}
+
+	switch (fmt & SND_SOC_DAIFMT_INV_MASK) {
+	case SND_SOC_DAIFMT_IB_NF:
+		mcasp_clr_bits(mcasp, DAVINCI_MCASP_ACLKXCTL_REG, ACLKXPOL);
+		mcasp_clr_bits(mcasp, DAVINCI_MCASP_ACLKRCTL_REG, ACLKRPOL);
+		fs_pol_rising = true;
+		break;
+	case SND_SOC_DAIFMT_NB_IF:
+		mcasp_set_bits(mcasp, DAVINCI_MCASP_ACLKXCTL_REG, ACLKXPOL);
+		mcasp_set_bits(mcasp, DAVINCI_MCASP_ACLKRCTL_REG, ACLKRPOL);
+		fs_pol_rising = false;
+		break;
+	case SND_SOC_DAIFMT_IB_IF:
+		mcasp_clr_bits(mcasp, DAVINCI_MCASP_ACLKXCTL_REG, ACLKXPOL);
+		mcasp_clr_bits(mcasp, DAVINCI_MCASP_ACLKRCTL_REG, ACLKRPOL);
+		fs_pol_rising = false;
+		break;
+	case SND_SOC_DAIFMT_NB_NF:
+		mcasp_set_bits(mcasp, DAVINCI_MCASP_ACLKXCTL_REG, ACLKXPOL);
+		mcasp_set_bits(mcasp, DAVINCI_MCASP_ACLKRCTL_REG, ACLKRPOL);
+		fs_pol_rising = true;
+		break;
+	default:
+		ret = -EINVAL;
+		goto out;
+	}
+
+	if (inv_fs)
+		fs_pol_rising = !fs_pol_rising;
+
+	if (fs_pol_rising) {
+		mcasp_clr_bits(mcasp, DAVINCI_MCASP_TXFMCTL_REG, FSXPOL);
+		mcasp_clr_bits(mcasp, DAVINCI_MCASP_RXFMCTL_REG, FSRPOL);
+	} else {
+		mcasp_set_bits(mcasp, DAVINCI_MCASP_TXFMCTL_REG, FSXPOL);
+		mcasp_set_bits(mcasp, DAVINCI_MCASP_RXFMCTL_REG, FSRPOL);
+	}
+
+	mcasp->dai_fmt = fmt;
+out:
+	pm_runtime_put(mcasp->dev);
+	return ret;
+}
+
+static int __davinci_mcasp_set_clkdiv(struct davinci_mcasp *mcasp, int div_id,
+				      int div, bool explicit)
+{
+	pm_runtime_get_sync(mcasp->dev);
+	switch (div_id) {
+	case MCASP_CLKDIV_AUXCLK:			/* MCLK divider */
+		mcasp_mod_bits(mcasp, DAVINCI_MCASP_AHCLKXCTL_REG,
+			       AHCLKXDIV(div - 1), AHCLKXDIV_MASK);
+		mcasp_mod_bits(mcasp, DAVINCI_MCASP_AHCLKRCTL_REG,
+			       AHCLKRDIV(div - 1), AHCLKRDIV_MASK);
+		break;
+
+	case MCASP_CLKDIV_BCLK:			/* BCLK divider */
+		mcasp_mod_bits(mcasp, DAVINCI_MCASP_ACLKXCTL_REG,
+			       ACLKXDIV(div - 1), ACLKXDIV_MASK);
+		mcasp_mod_bits(mcasp, DAVINCI_MCASP_ACLKRCTL_REG,
+			       ACLKRDIV(div - 1), ACLKRDIV_MASK);
+		if (explicit)
+			mcasp->bclk_div = div;
+		break;
+
+	case MCASP_CLKDIV_BCLK_FS_RATIO:
+		/*
+		 * BCLK/LRCLK ratio descries how many bit-clock cycles
+		 * fit into one frame. The clock ratio is given for a
+		 * full period of data (for I2S format both left and
+		 * right channels), so it has to be divided by number
+		 * of tdm-slots (for I2S - divided by 2).
+		 * Instead of storing this ratio, we calculate a new
+		 * tdm_slot width by dividing the ratio by the
+		 * number of configured tdm slots.
+		 */
+		mcasp->slot_width = div / mcasp->tdm_slots;
+		if (div % mcasp->tdm_slots)
+			dev_warn(mcasp->dev,
+				 "%s(): BCLK/LRCLK %d is not divisible by %d tdm slots",
+				 __func__, div, mcasp->tdm_slots);
+		break;
+
+	default:
+		return -EINVAL;
+	}
+
+	pm_runtime_put(mcasp->dev);
+	return 0;
+}
+
+static int davinci_mcasp_set_clkdiv(struct snd_soc_dai *dai, int div_id,
+				    int div)
+{
+	struct davinci_mcasp *mcasp = snd_soc_dai_get_drvdata(dai);
+
+	return __davinci_mcasp_set_clkdiv(mcasp, div_id, div, 1);
+}
+
+static int davinci_mcasp_set_sysclk(struct snd_soc_dai *dai, int clk_id,
+				    unsigned int freq, int dir)
+{
+	struct davinci_mcasp *mcasp = snd_soc_dai_get_drvdata(dai);
+
+	pm_runtime_get_sync(mcasp->dev);
+
+	if (dir == SND_SOC_CLOCK_IN) {
+		switch (clk_id) {
+		case MCASP_CLK_HCLK_AHCLK:
+			mcasp_clr_bits(mcasp, DAVINCI_MCASP_AHCLKXCTL_REG,
+				       AHCLKXE);
+			mcasp_clr_bits(mcasp, DAVINCI_MCASP_AHCLKRCTL_REG,
+				       AHCLKRE);
+			clear_bit(PIN_BIT_AHCLKX, &mcasp->pdir);
+			break;
+		case MCASP_CLK_HCLK_AUXCLK:
+			mcasp_set_bits(mcasp, DAVINCI_MCASP_AHCLKXCTL_REG,
+				       AHCLKXE);
+			mcasp_set_bits(mcasp, DAVINCI_MCASP_AHCLKRCTL_REG,
+				       AHCLKRE);
+			set_bit(PIN_BIT_AHCLKX, &mcasp->pdir);
+			break;
+		default:
+			dev_err(mcasp->dev, "Invalid clk id: %d\n", clk_id);
+			goto out;
+		}
+	} else {
+		/* Select AUXCLK as HCLK */
+		mcasp_set_bits(mcasp, DAVINCI_MCASP_AHCLKXCTL_REG, AHCLKXE);
+		mcasp_set_bits(mcasp, DAVINCI_MCASP_AHCLKRCTL_REG, AHCLKRE);
+		set_bit(PIN_BIT_AHCLKX, &mcasp->pdir);
+	}
+	/*
+	 * When AHCLK X/R is selected to be output it means that the HCLK is
+	 * the same clock - coming via AUXCLK.
+	 */
+	mcasp->sysclk_freq = freq;
+out:
+	pm_runtime_put(mcasp->dev);
+	return 0;
+}
+
+/* All serializers must have equal number of channels */
+static int davinci_mcasp_ch_constraint(struct davinci_mcasp *mcasp, int stream,
+				       int serializers)
+{
+	struct snd_pcm_hw_constraint_list *cl = &mcasp->chconstr[stream];
+	unsigned int *list = (unsigned int *) cl->list;
+	int slots = mcasp->tdm_slots;
+	int i, count = 0;
+
+	if (mcasp->tdm_mask[stream])
+		slots = hweight32(mcasp->tdm_mask[stream]);
+
+	for (i = 1; i <= slots; i++)
+		list[count++] = i;
+
+	for (i = 2; i <= serializers; i++)
+		list[count++] = i*slots;
+
+	cl->count = count;
+
+	return 0;
+}
+
+static int davinci_mcasp_set_ch_constraints(struct davinci_mcasp *mcasp)
+{
+	int rx_serializers = 0, tx_serializers = 0, ret, i;
+
+	for (i = 0; i < mcasp->num_serializer; i++)
+		if (mcasp->serial_dir[i] == TX_MODE)
+			tx_serializers++;
+		else if (mcasp->serial_dir[i] == RX_MODE)
+			rx_serializers++;
+
+	ret = davinci_mcasp_ch_constraint(mcasp, SNDRV_PCM_STREAM_PLAYBACK,
+					  tx_serializers);
+	if (ret)
+		return ret;
+
+	ret = davinci_mcasp_ch_constraint(mcasp, SNDRV_PCM_STREAM_CAPTURE,
+					  rx_serializers);
+
+	return ret;
+}
+
+
+static int davinci_mcasp_set_tdm_slot(struct snd_soc_dai *dai,
+				      unsigned int tx_mask,
+				      unsigned int rx_mask,
+				      int slots, int slot_width)
+{
+	struct davinci_mcasp *mcasp = snd_soc_dai_get_drvdata(dai);
+
+	if (mcasp->op_mode == DAVINCI_MCASP_DIT_MODE)
+		return 0;
+
+	dev_dbg(mcasp->dev,
+		 "%s() tx_mask 0x%08x rx_mask 0x%08x slots %d width %d\n",
+		 __func__, tx_mask, rx_mask, slots, slot_width);
+
+	if (tx_mask >= (1<<slots) || rx_mask >= (1<<slots)) {
+		dev_err(mcasp->dev,
+			"Bad tdm mask tx: 0x%08x rx: 0x%08x slots %d\n",
+			tx_mask, rx_mask, slots);
+		return -EINVAL;
+	}
+
+	if (slot_width &&
+	    (slot_width < 8 || slot_width > 32 || slot_width % 4 != 0)) {
+		dev_err(mcasp->dev, "%s: Unsupported slot_width %d\n",
+			__func__, slot_width);
+		return -EINVAL;
+	}
+
+	mcasp->tdm_slots = slots;
+	mcasp->tdm_mask[SNDRV_PCM_STREAM_PLAYBACK] = tx_mask;
+	mcasp->tdm_mask[SNDRV_PCM_STREAM_CAPTURE] = rx_mask;
+	mcasp->slot_width = slot_width;
+
+	return davinci_mcasp_set_ch_constraints(mcasp);
+}
+
+static int davinci_config_channel_size(struct davinci_mcasp *mcasp,
+				       int sample_width)
+{
+	u32 fmt;
+	u32 tx_rotate, rx_rotate, slot_width;
+	u32 mask = (1ULL << sample_width) - 1;
+
+	if (mcasp->slot_width)
+		slot_width = mcasp->slot_width;
+	else if (mcasp->max_format_width)
+		slot_width = mcasp->max_format_width;
+	else
+		slot_width = sample_width;
+	/*
+	 * TX rotation:
+	 * right aligned formats: rotate w/ slot_width
+	 * left aligned formats: rotate w/ sample_width
+	 *
+	 * RX rotation:
+	 * right aligned formats: no rotation needed
+	 * left aligned formats: rotate w/ (slot_width - sample_width)
+	 */
+	if ((mcasp->dai_fmt & SND_SOC_DAIFMT_FORMAT_MASK) ==
+	    SND_SOC_DAIFMT_RIGHT_J) {
+		tx_rotate = (slot_width / 4) & 0x7;
+		rx_rotate = 0;
+	} else {
+		tx_rotate = (sample_width / 4) & 0x7;
+		rx_rotate = (slot_width - sample_width) / 4;
+	}
+
+	/* mapping of the XSSZ bit-field as described in the datasheet */
+	fmt = (slot_width >> 1) - 1;
+
+	if (mcasp->op_mode != DAVINCI_MCASP_DIT_MODE) {
+		mcasp_mod_bits(mcasp, DAVINCI_MCASP_RXFMT_REG, RXSSZ(fmt),
+			       RXSSZ(0x0F));
+		mcasp_mod_bits(mcasp, DAVINCI_MCASP_TXFMT_REG, TXSSZ(fmt),
+			       TXSSZ(0x0F));
+		mcasp_mod_bits(mcasp, DAVINCI_MCASP_TXFMT_REG, TXROT(tx_rotate),
+			       TXROT(7));
+		mcasp_mod_bits(mcasp, DAVINCI_MCASP_RXFMT_REG, RXROT(rx_rotate),
+			       RXROT(7));
+		mcasp_set_reg(mcasp, DAVINCI_MCASP_RXMASK_REG, mask);
+	} else {
+		/*
+		 * according to the TRM it should be TXROT=0, this one works:
+		 * 16 bit to 23-8 (TXROT=6, rotate 24 bits)
+		 * 24 bit to 23-0 (TXROT=0, rotate 0 bits)
+		 *
+		 * TXROT = 0 only works with 24bit samples
+		 */
+		tx_rotate = (sample_width / 4 + 2) & 0x7;
+
+		mcasp_mod_bits(mcasp, DAVINCI_MCASP_TXFMT_REG, TXROT(tx_rotate),
+			       TXROT(7));
+		mcasp_mod_bits(mcasp, DAVINCI_MCASP_TXFMT_REG, TXSSZ(15),
+			       TXSSZ(0x0F));
+	}
+
+	mcasp_set_reg(mcasp, DAVINCI_MCASP_TXMASK_REG, mask);
+
+	return 0;
+}
+
+static int mcasp_common_hw_param(struct davinci_mcasp *mcasp, int stream,
+				 int period_words, int channels)
+{
+	struct snd_dmaengine_dai_dma_data *dma_data = &mcasp->dma_data[stream];
+	int i;
+	u8 tx_ser = 0;
+	u8 rx_ser = 0;
+	u8 slots = mcasp->tdm_slots;
+	u8 max_active_serializers, max_rx_serializers, max_tx_serializers;
+	int active_serializers, numevt;
+	u32 reg;
+
+	/* In DIT mode we only allow maximum of one serializers for now */
+	if (mcasp->op_mode == DAVINCI_MCASP_DIT_MODE)
+		max_active_serializers = 1;
+	else
+		max_active_serializers = (channels + slots - 1) / slots;
+
+	/* Default configuration */
+	if (mcasp->version < MCASP_VERSION_3)
+		mcasp_set_bits(mcasp, DAVINCI_MCASP_PWREMUMGT_REG, MCASP_SOFT);
+
+	if (stream == SNDRV_PCM_STREAM_PLAYBACK) {
+		mcasp_set_reg(mcasp, DAVINCI_MCASP_TXSTAT_REG, 0xFFFFFFFF);
+		mcasp_clr_bits(mcasp, DAVINCI_MCASP_XEVTCTL_REG, TXDATADMADIS);
+		max_tx_serializers = max_active_serializers;
+		max_rx_serializers =
+			mcasp->active_serializers[SNDRV_PCM_STREAM_CAPTURE];
+	} else {
+		mcasp_set_reg(mcasp, DAVINCI_MCASP_RXSTAT_REG, 0xFFFFFFFF);
+		mcasp_clr_bits(mcasp, DAVINCI_MCASP_REVTCTL_REG, RXDATADMADIS);
+		max_tx_serializers =
+			mcasp->active_serializers[SNDRV_PCM_STREAM_PLAYBACK];
+		max_rx_serializers = max_active_serializers;
+	}
+
+	for (i = 0; i < mcasp->num_serializer; i++) {
+		mcasp_set_bits(mcasp, DAVINCI_MCASP_XRSRCTL_REG(i),
+			       mcasp->serial_dir[i]);
+		if (mcasp->serial_dir[i] == TX_MODE &&
+					tx_ser < max_tx_serializers) {
+			mcasp_mod_bits(mcasp, DAVINCI_MCASP_XRSRCTL_REG(i),
+				       mcasp->dismod, DISMOD_MASK);
+			set_bit(PIN_BIT_AXR(i), &mcasp->pdir);
+			tx_ser++;
+		} else if (mcasp->serial_dir[i] == RX_MODE &&
+					rx_ser < max_rx_serializers) {
+			clear_bit(PIN_BIT_AXR(i), &mcasp->pdir);
+			rx_ser++;
+		} else {
+			/* Inactive or unused pin, set it to inactive */
+			mcasp_mod_bits(mcasp, DAVINCI_MCASP_XRSRCTL_REG(i),
+				       SRMOD_INACTIVE, SRMOD_MASK);
+			/* If unused, set DISMOD for the pin */
+			if (mcasp->serial_dir[i] != INACTIVE_MODE)
+				mcasp_mod_bits(mcasp,
+					       DAVINCI_MCASP_XRSRCTL_REG(i),
+					       mcasp->dismod, DISMOD_MASK);
+			clear_bit(PIN_BIT_AXR(i), &mcasp->pdir);
+		}
+	}
+
+	if (stream == SNDRV_PCM_STREAM_PLAYBACK) {
+		active_serializers = tx_ser;
+		numevt = mcasp->txnumevt;
+		reg = mcasp->fifo_base + MCASP_WFIFOCTL_OFFSET;
+	} else {
+		active_serializers = rx_ser;
+		numevt = mcasp->rxnumevt;
+		reg = mcasp->fifo_base + MCASP_RFIFOCTL_OFFSET;
+	}
+
+	if (active_serializers < max_active_serializers) {
+		dev_warn(mcasp->dev, "stream has more channels (%d) than are "
+			 "enabled in mcasp (%d)\n", channels,
+			 active_serializers * slots);
+		return -EINVAL;
+	}
+
+	/* AFIFO is not in use */
+	if (!numevt) {
+		/* Configure the burst size for platform drivers */
+		if (active_serializers > 1) {
+			/*
+			 * If more than one serializers are in use we have one
+			 * DMA request to provide data for all serializers.
+			 * For example if three serializers are enabled the DMA
+			 * need to transfer three words per DMA request.
+			 */
+			dma_data->maxburst = active_serializers;
+		} else {
+			dma_data->maxburst = 0;
+		}
+
+		goto out;
+	}
+
+	if (period_words % active_serializers) {
+		dev_err(mcasp->dev, "Invalid combination of period words and "
+			"active serializers: %d, %d\n", period_words,
+			active_serializers);
+		return -EINVAL;
+	}
+
+	/*
+	 * Calculate the optimal AFIFO depth for platform side:
+	 * The number of words for numevt need to be in steps of active
+	 * serializers.
+	 */
+	numevt = (numevt / active_serializers) * active_serializers;
+
+	while (period_words % numevt && numevt > 0)
+		numevt -= active_serializers;
+	if (numevt <= 0)
+		numevt = active_serializers;
+
+	mcasp_mod_bits(mcasp, reg, active_serializers, NUMDMA_MASK);
+	mcasp_mod_bits(mcasp, reg, NUMEVT(numevt), NUMEVT_MASK);
+
+	/* Configure the burst size for platform drivers */
+	if (numevt == 1)
+		numevt = 0;
+	dma_data->maxburst = numevt;
+
+out:
+	mcasp->active_serializers[stream] = active_serializers;
+
+	return 0;
+}
+
+static int mcasp_i2s_hw_param(struct davinci_mcasp *mcasp, int stream,
+			      int channels)
+{
+	int i, active_slots;
+	int total_slots;
+	int active_serializers;
+	u32 mask = 0;
+	u32 busel = 0;
+
+	total_slots = mcasp->tdm_slots;
+
+	/*
+	 * If more than one serializer is needed, then use them with
+	 * all the specified tdm_slots. Otherwise, one serializer can
+	 * cope with the transaction using just as many slots as there
+	 * are channels in the stream.
+	 */
+	if (mcasp->tdm_mask[stream]) {
+		active_slots = hweight32(mcasp->tdm_mask[stream]);
+		active_serializers = (channels + active_slots - 1) /
+			active_slots;
+		if (active_serializers == 1)
+			active_slots = channels;
+		for (i = 0; i < total_slots; i++) {
+			if ((1 << i) & mcasp->tdm_mask[stream]) {
+				mask |= (1 << i);
+				if (--active_slots <= 0)
+					break;
+			}
+		}
+	} else {
+		active_serializers = (channels + total_slots - 1) / total_slots;
+		if (active_serializers == 1)
+			active_slots = channels;
+		else
+			active_slots = total_slots;
+
+		for (i = 0; i < active_slots; i++)
+			mask |= (1 << i);
+	}
+
+	mcasp_clr_bits(mcasp, DAVINCI_MCASP_ACLKXCTL_REG, TX_ASYNC);
+
+	if (!mcasp->dat_port)
+		busel = TXSEL;
+
+	if (stream == SNDRV_PCM_STREAM_PLAYBACK) {
+		mcasp_set_reg(mcasp, DAVINCI_MCASP_TXTDM_REG, mask);
+		mcasp_set_bits(mcasp, DAVINCI_MCASP_TXFMT_REG, busel | TXORD);
+		mcasp_mod_bits(mcasp, DAVINCI_MCASP_TXFMCTL_REG,
+			       FSXMOD(total_slots), FSXMOD(0x1FF));
+	} else if (stream == SNDRV_PCM_STREAM_CAPTURE) {
+		mcasp_set_reg(mcasp, DAVINCI_MCASP_RXTDM_REG, mask);
+		mcasp_set_bits(mcasp, DAVINCI_MCASP_RXFMT_REG, busel | RXORD);
+		mcasp_mod_bits(mcasp, DAVINCI_MCASP_RXFMCTL_REG,
+			       FSRMOD(total_slots), FSRMOD(0x1FF));
+		/*
+		 * If McASP is set to be TX/RX synchronous and the playback is
+		 * not running already we need to configure the TX slots in
+		 * order to have correct FSX on the bus
+		 */
+		if (mcasp_is_synchronous(mcasp) && !mcasp->channels)
+			mcasp_mod_bits(mcasp, DAVINCI_MCASP_TXFMCTL_REG,
+				       FSXMOD(total_slots), FSXMOD(0x1FF));
+	}
+
+	return 0;
+}
+
+/* S/PDIF */
+static int mcasp_dit_hw_param(struct davinci_mcasp *mcasp,
+			      unsigned int rate)
+{
+	u8 *cs_bytes = (u8 *)&mcasp->iec958_status;
+
+	if (!mcasp->dat_port)
+		mcasp_set_bits(mcasp, DAVINCI_MCASP_TXFMT_REG, TXSEL);
+	else
+		mcasp_clr_bits(mcasp, DAVINCI_MCASP_TXFMT_REG, TXSEL);
+
+	/* Set TX frame synch : DIT Mode, 1 bit width, internal, rising edge */
+	mcasp_set_reg(mcasp, DAVINCI_MCASP_TXFMCTL_REG, AFSXE | FSXMOD(0x180));
+
+	mcasp_set_reg(mcasp, DAVINCI_MCASP_TXMASK_REG, 0xFFFF);
+
+	/* Set the TX tdm : for all the slots */
+	mcasp_set_reg(mcasp, DAVINCI_MCASP_TXTDM_REG, 0xFFFFFFFF);
+
+	/* Set the TX clock controls : div = 1 and internal */
+	mcasp_set_bits(mcasp, DAVINCI_MCASP_ACLKXCTL_REG, ACLKXE | TX_ASYNC);
+
+	mcasp_clr_bits(mcasp, DAVINCI_MCASP_XEVTCTL_REG, TXDATADMADIS);
+
+	/* Set S/PDIF channel status bits */
+	cs_bytes[3] &= ~IEC958_AES3_CON_FS;
+	switch (rate) {
+	case 22050:
+		cs_bytes[3] |= IEC958_AES3_CON_FS_22050;
+		break;
+	case 24000:
+		cs_bytes[3] |= IEC958_AES3_CON_FS_24000;
+		break;
+	case 32000:
+		cs_bytes[3] |= IEC958_AES3_CON_FS_32000;
+		break;
+	case 44100:
+		cs_bytes[3] |= IEC958_AES3_CON_FS_44100;
+		break;
+	case 48000:
+		cs_bytes[3] |= IEC958_AES3_CON_FS_48000;
+		break;
+	case 88200:
+		cs_bytes[3] |= IEC958_AES3_CON_FS_88200;
+		break;
+	case 96000:
+		cs_bytes[3] |= IEC958_AES3_CON_FS_96000;
+		break;
+	case 176400:
+		cs_bytes[3] |= IEC958_AES3_CON_FS_176400;
+		break;
+	case 192000:
+		cs_bytes[3] |= IEC958_AES3_CON_FS_192000;
+		break;
+	default:
+		dev_err(mcasp->dev, "unsupported sampling rate: %d\n", rate);
+		return -EINVAL;
+	}
+
+	mcasp_set_reg(mcasp, DAVINCI_MCASP_DITCSRA_REG, mcasp->iec958_status);
+	mcasp_set_reg(mcasp, DAVINCI_MCASP_DITCSRB_REG, mcasp->iec958_status);
+
+	/* Enable the DIT */
+	mcasp_set_bits(mcasp, DAVINCI_MCASP_TXDITCTL_REG, DITEN);
+
+	return 0;
+}
+
+static int davinci_mcasp_calc_clk_div(struct davinci_mcasp *mcasp,
+				      unsigned int sysclk_freq,
+				      unsigned int bclk_freq, bool set)
+{
+	u32 reg = mcasp_get_reg(mcasp, DAVINCI_MCASP_AHCLKXCTL_REG);
+	int div = sysclk_freq / bclk_freq;
+	int rem = sysclk_freq % bclk_freq;
+	int error_ppm;
+	int aux_div = 1;
+
+	if (div > (ACLKXDIV_MASK + 1)) {
+		if (reg & AHCLKXE) {
+			aux_div = div / (ACLKXDIV_MASK + 1);
+			if (div % (ACLKXDIV_MASK + 1))
+				aux_div++;
+
+			sysclk_freq /= aux_div;
+			div = sysclk_freq / bclk_freq;
+			rem = sysclk_freq % bclk_freq;
+		} else if (set) {
+			dev_warn(mcasp->dev, "Too fast reference clock (%u)\n",
+				 sysclk_freq);
+		}
+	}
+
+	if (rem != 0) {
+		if (div == 0 ||
+		    ((sysclk_freq / div) - bclk_freq) >
+		    (bclk_freq - (sysclk_freq / (div+1)))) {
+			div++;
+			rem = rem - bclk_freq;
+		}
+	}
+	error_ppm = (div*1000000 + (int)div64_long(1000000LL*rem,
+		     (int)bclk_freq)) / div - 1000000;
+
+	if (set) {
+		if (error_ppm)
+			dev_info(mcasp->dev, "Sample-rate is off by %d PPM\n",
+				 error_ppm);
+
+		__davinci_mcasp_set_clkdiv(mcasp, MCASP_CLKDIV_BCLK, div, 0);
+		if (reg & AHCLKXE)
+			__davinci_mcasp_set_clkdiv(mcasp, MCASP_CLKDIV_AUXCLK,
+						   aux_div, 0);
+	}
+
+	return error_ppm;
+}
+
+static inline u32 davinci_mcasp_tx_delay(struct davinci_mcasp *mcasp)
+{
+	if (!mcasp->txnumevt)
+		return 0;
+
+	return mcasp_get_reg(mcasp, mcasp->fifo_base + MCASP_WFIFOSTS_OFFSET);
+}
+
+static inline u32 davinci_mcasp_rx_delay(struct davinci_mcasp *mcasp)
+{
+	if (!mcasp->rxnumevt)
+		return 0;
+
+	return mcasp_get_reg(mcasp, mcasp->fifo_base + MCASP_RFIFOSTS_OFFSET);
+}
+
+static snd_pcm_sframes_t davinci_mcasp_delay(
+			struct snd_pcm_substream *substream,
+			struct snd_soc_dai *cpu_dai)
+{
+	struct davinci_mcasp *mcasp = snd_soc_dai_get_drvdata(cpu_dai);
+	u32 fifo_use;
+
+	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+		fifo_use = davinci_mcasp_tx_delay(mcasp);
+	else
+		fifo_use = davinci_mcasp_rx_delay(mcasp);
+
+	/*
+	 * Divide the used locations with the channel count to get the
+	 * FIFO usage in samples (don't care about partial samples in the
+	 * buffer).
+	 */
+	return fifo_use / substream->runtime->channels;
+}
+
+static int davinci_mcasp_hw_params(struct snd_pcm_substream *substream,
+					struct snd_pcm_hw_params *params,
+					struct snd_soc_dai *cpu_dai)
+{
+	struct davinci_mcasp *mcasp = snd_soc_dai_get_drvdata(cpu_dai);
+	int word_length;
+	int channels = params_channels(params);
+	int period_size = params_period_size(params);
+	int ret;
+
+	switch (params_format(params)) {
+	case SNDRV_PCM_FORMAT_U8:
+	case SNDRV_PCM_FORMAT_S8:
+		word_length = 8;
+		break;
+
+	case SNDRV_PCM_FORMAT_U16_LE:
+	case SNDRV_PCM_FORMAT_S16_LE:
+		word_length = 16;
+		break;
+
+	case SNDRV_PCM_FORMAT_U24_3LE:
+	case SNDRV_PCM_FORMAT_S24_3LE:
+		word_length = 24;
+		break;
+
+	case SNDRV_PCM_FORMAT_U24_LE:
+	case SNDRV_PCM_FORMAT_S24_LE:
+		word_length = 24;
+		break;
+
+	case SNDRV_PCM_FORMAT_U32_LE:
+	case SNDRV_PCM_FORMAT_S32_LE:
+		word_length = 32;
+		break;
+
+	default:
+		printk(KERN_WARNING "davinci-mcasp: unsupported PCM format");
+		return -EINVAL;
+	}
+
+	ret = davinci_mcasp_set_dai_fmt(cpu_dai, mcasp->dai_fmt);
+	if (ret)
+		return ret;
+
+	/*
+	 * If mcasp is BCLK master, and a BCLK divider was not provided by
+	 * the machine driver, we need to calculate the ratio.
+	 */
+	if (mcasp->bclk_master && mcasp->bclk_div == 0 && mcasp->sysclk_freq) {
+		int slots = mcasp->tdm_slots;
+		int rate = params_rate(params);
+		int sbits = params_width(params);
+		unsigned int bclk_target;
+
+		if (mcasp->slot_width)
+			sbits = mcasp->slot_width;
+
+		if (mcasp->op_mode == DAVINCI_MCASP_IIS_MODE)
+			bclk_target = rate * sbits * slots;
+		else
+			bclk_target = rate * 128;
+
+		davinci_mcasp_calc_clk_div(mcasp, mcasp->sysclk_freq,
+					   bclk_target, true);
+	}
+
+	ret = mcasp_common_hw_param(mcasp, substream->stream,
+				    period_size * channels, channels);
+	if (ret)
+		return ret;
+
+	if (mcasp->op_mode == DAVINCI_MCASP_DIT_MODE)
+		ret = mcasp_dit_hw_param(mcasp, params_rate(params));
+	else
+		ret = mcasp_i2s_hw_param(mcasp, substream->stream,
+					 channels);
+
+	if (ret)
+		return ret;
+
+	davinci_config_channel_size(mcasp, word_length);
+
+	if (mcasp->op_mode == DAVINCI_MCASP_IIS_MODE) {
+		mcasp->channels = channels;
+		if (!mcasp->max_format_width)
+			mcasp->max_format_width = word_length;
+	}
+
+	return 0;
+}
+
+static int davinci_mcasp_trigger(struct snd_pcm_substream *substream,
+				     int cmd, struct snd_soc_dai *cpu_dai)
+{
+	struct davinci_mcasp *mcasp = snd_soc_dai_get_drvdata(cpu_dai);
+	int ret = 0;
+
+	switch (cmd) {
+	case SNDRV_PCM_TRIGGER_RESUME:
+	case SNDRV_PCM_TRIGGER_START:
+	case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
+		davinci_mcasp_start(mcasp, substream->stream);
+		break;
+	case SNDRV_PCM_TRIGGER_SUSPEND:
+	case SNDRV_PCM_TRIGGER_STOP:
+	case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
+		davinci_mcasp_stop(mcasp, substream->stream);
+		break;
+
+	default:
+		ret = -EINVAL;
+	}
+
+	return ret;
+}
+
+static int davinci_mcasp_hw_rule_slot_width(struct snd_pcm_hw_params *params,
+					    struct snd_pcm_hw_rule *rule)
+{
+	struct davinci_mcasp_ruledata *rd = rule->private;
+	struct snd_mask *fmt = hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT);
+	struct snd_mask nfmt;
+	int i, slot_width;
+
+	snd_mask_none(&nfmt);
+	slot_width = rd->mcasp->slot_width;
+
+	for (i = 0; i <= SNDRV_PCM_FORMAT_LAST; i++) {
+		if (snd_mask_test(fmt, i)) {
+			if (snd_pcm_format_width(i) <= slot_width) {
+				snd_mask_set(&nfmt, i);
+			}
+		}
+	}
+
+	return snd_mask_refine(fmt, &nfmt);
+}
+
+static int davinci_mcasp_hw_rule_format_width(struct snd_pcm_hw_params *params,
+					      struct snd_pcm_hw_rule *rule)
+{
+	struct davinci_mcasp_ruledata *rd = rule->private;
+	struct snd_mask *fmt = hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT);
+	struct snd_mask nfmt;
+	int i, format_width;
+
+	snd_mask_none(&nfmt);
+	format_width = rd->mcasp->max_format_width;
+
+	for (i = 0; i <= SNDRV_PCM_FORMAT_LAST; i++) {
+		if (snd_mask_test(fmt, i)) {
+			if (snd_pcm_format_width(i) == format_width) {
+				snd_mask_set(&nfmt, i);
+			}
+		}
+	}
+
+	return snd_mask_refine(fmt, &nfmt);
+}
+
+static const unsigned int davinci_mcasp_dai_rates[] = {
+	8000, 11025, 16000, 22050, 32000, 44100, 48000, 64000,
+	88200, 96000, 176400, 192000,
+};
+
+#define DAVINCI_MAX_RATE_ERROR_PPM 1000
+
+static int davinci_mcasp_hw_rule_rate(struct snd_pcm_hw_params *params,
+				      struct snd_pcm_hw_rule *rule)
+{
+	struct davinci_mcasp_ruledata *rd = rule->private;
+	struct snd_interval *ri =
+		hw_param_interval(params, SNDRV_PCM_HW_PARAM_RATE);
+	int sbits = params_width(params);
+	int slots = rd->mcasp->tdm_slots;
+	struct snd_interval range;
+	int i;
+
+	if (rd->mcasp->slot_width)
+		sbits = rd->mcasp->slot_width;
+
+	snd_interval_any(&range);
+	range.empty = 1;
+
+	for (i = 0; i < ARRAY_SIZE(davinci_mcasp_dai_rates); i++) {
+		if (snd_interval_test(ri, davinci_mcasp_dai_rates[i])) {
+			uint bclk_freq = sbits * slots *
+					 davinci_mcasp_dai_rates[i];
+			unsigned int sysclk_freq;
+			int ppm;
+
+			if (rd->mcasp->auxclk_fs_ratio)
+				sysclk_freq =  davinci_mcasp_dai_rates[i] *
+					       rd->mcasp->auxclk_fs_ratio;
+			else
+				sysclk_freq = rd->mcasp->sysclk_freq;
+
+			ppm = davinci_mcasp_calc_clk_div(rd->mcasp, sysclk_freq,
+							 bclk_freq, false);
+			if (abs(ppm) < DAVINCI_MAX_RATE_ERROR_PPM) {
+				if (range.empty) {
+					range.min = davinci_mcasp_dai_rates[i];
+					range.empty = 0;
+				}
+				range.max = davinci_mcasp_dai_rates[i];
+			}
+		}
+	}
+
+	dev_dbg(rd->mcasp->dev,
+		"Frequencies %d-%d -> %d-%d for %d sbits and %d tdm slots\n",
+		ri->min, ri->max, range.min, range.max, sbits, slots);
+
+	return snd_interval_refine(hw_param_interval(params, rule->var),
+				   &range);
+}
+
+static int davinci_mcasp_hw_rule_format(struct snd_pcm_hw_params *params,
+					struct snd_pcm_hw_rule *rule)
+{
+	struct davinci_mcasp_ruledata *rd = rule->private;
+	struct snd_mask *fmt = hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT);
+	struct snd_mask nfmt;
+	int rate = params_rate(params);
+	int slots = rd->mcasp->tdm_slots;
+	int i, count = 0;
+
+	snd_mask_none(&nfmt);
+
+	for (i = 0; i <= SNDRV_PCM_FORMAT_LAST; i++) {
+		if (snd_mask_test(fmt, i)) {
+			uint sbits = snd_pcm_format_width(i);
+			unsigned int sysclk_freq;
+			int ppm;
+
+			if (rd->mcasp->auxclk_fs_ratio)
+				sysclk_freq =  rate *
+					       rd->mcasp->auxclk_fs_ratio;
+			else
+				sysclk_freq = rd->mcasp->sysclk_freq;
+
+			if (rd->mcasp->slot_width)
+				sbits = rd->mcasp->slot_width;
+
+			ppm = davinci_mcasp_calc_clk_div(rd->mcasp, sysclk_freq,
+							 sbits * slots * rate,
+							 false);
+			if (abs(ppm) < DAVINCI_MAX_RATE_ERROR_PPM) {
+				snd_mask_set(&nfmt, i);
+				count++;
+			}
+		}
+	}
+	dev_dbg(rd->mcasp->dev,
+		"%d possible sample format for %d Hz and %d tdm slots\n",
+		count, rate, slots);
+
+	return snd_mask_refine(fmt, &nfmt);
+}
+
+static int davinci_mcasp_hw_rule_min_periodsize(
+		struct snd_pcm_hw_params *params, struct snd_pcm_hw_rule *rule)
+{
+	struct snd_interval *period_size = hw_param_interval(params,
+						SNDRV_PCM_HW_PARAM_PERIOD_SIZE);
+	struct snd_interval frames;
+
+	snd_interval_any(&frames);
+	frames.min = 64;
+	frames.integer = 1;
+
+	return snd_interval_refine(period_size, &frames);
+}
+
+static int davinci_mcasp_startup(struct snd_pcm_substream *substream,
+				 struct snd_soc_dai *cpu_dai)
+{
+	struct davinci_mcasp *mcasp = snd_soc_dai_get_drvdata(cpu_dai);
+	struct davinci_mcasp_ruledata *ruledata =
+					&mcasp->ruledata[substream->stream];
+	u32 max_channels = 0;
+	int i, dir, ret;
+	int tdm_slots = mcasp->tdm_slots;
+
+	/* Do not allow more then one stream per direction */
+	if (mcasp->substreams[substream->stream])
+		return -EBUSY;
+
+	mcasp->substreams[substream->stream] = substream;
+
+	if (mcasp->tdm_mask[substream->stream])
+		tdm_slots = hweight32(mcasp->tdm_mask[substream->stream]);
+
+	if (mcasp->op_mode == DAVINCI_MCASP_DIT_MODE)
+		return 0;
+
+	/*
+	 * Limit the maximum allowed channels for the first stream:
+	 * number of serializers for the direction * tdm slots per serializer
+	 */
+	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+		dir = TX_MODE;
+	else
+		dir = RX_MODE;
+
+	for (i = 0; i < mcasp->num_serializer; i++) {
+		if (mcasp->serial_dir[i] == dir)
+			max_channels++;
+	}
+	ruledata->serializers = max_channels;
+	ruledata->mcasp = mcasp;
+	max_channels *= tdm_slots;
+	/*
+	 * If the already active stream has less channels than the calculated
+	 * limit based on the seirializers * tdm_slots, and only one serializer
+	 * is in use we need to use that as a constraint for the second stream.
+	 * Otherwise (first stream or less allowed channels or more than one
+	 * serializer in use) we use the calculated constraint.
+	 */
+	if (mcasp->channels && mcasp->channels < max_channels &&
+	    ruledata->serializers == 1)
+		max_channels = mcasp->channels;
+	/*
+	 * But we can always allow channels upto the amount of
+	 * the available tdm_slots.
+	 */
+	if (max_channels < tdm_slots)
+		max_channels = tdm_slots;
+
+	snd_pcm_hw_constraint_minmax(substream->runtime,
+				     SNDRV_PCM_HW_PARAM_CHANNELS,
+				     0, max_channels);
+
+	snd_pcm_hw_constraint_list(substream->runtime,
+				   0, SNDRV_PCM_HW_PARAM_CHANNELS,
+				   &mcasp->chconstr[substream->stream]);
+
+	if (mcasp->max_format_width) {
+		/*
+		 * Only allow formats which require same amount of bits on the
+		 * bus as the currently running stream
+		 */
+		ret = snd_pcm_hw_rule_add(substream->runtime, 0,
+					  SNDRV_PCM_HW_PARAM_FORMAT,
+					  davinci_mcasp_hw_rule_format_width,
+					  ruledata,
+					  SNDRV_PCM_HW_PARAM_FORMAT, -1);
+		if (ret)
+			return ret;
+	}
+	else if (mcasp->slot_width) {
+		/* Only allow formats require <= slot_width bits on the bus */
+		ret = snd_pcm_hw_rule_add(substream->runtime, 0,
+					  SNDRV_PCM_HW_PARAM_FORMAT,
+					  davinci_mcasp_hw_rule_slot_width,
+					  ruledata,
+					  SNDRV_PCM_HW_PARAM_FORMAT, -1);
+		if (ret)
+			return ret;
+	}
+
+	/*
+	 * If we rely on implicit BCLK divider setting we should
+	 * set constraints based on what we can provide.
+	 */
+	if (mcasp->bclk_master && mcasp->bclk_div == 0 && mcasp->sysclk_freq) {
+		ret = snd_pcm_hw_rule_add(substream->runtime, 0,
+					  SNDRV_PCM_HW_PARAM_RATE,
+					  davinci_mcasp_hw_rule_rate,
+					  ruledata,
+					  SNDRV_PCM_HW_PARAM_FORMAT, -1);
+		if (ret)
+			return ret;
+		ret = snd_pcm_hw_rule_add(substream->runtime, 0,
+					  SNDRV_PCM_HW_PARAM_FORMAT,
+					  davinci_mcasp_hw_rule_format,
+					  ruledata,
+					  SNDRV_PCM_HW_PARAM_RATE, -1);
+		if (ret)
+			return ret;
+	}
+
+	snd_pcm_hw_rule_add(substream->runtime, 0,
+			    SNDRV_PCM_HW_PARAM_PERIOD_SIZE,
+			    davinci_mcasp_hw_rule_min_periodsize, NULL,
+			    SNDRV_PCM_HW_PARAM_PERIOD_SIZE, -1);
+
+	return 0;
+}
+
+static void davinci_mcasp_shutdown(struct snd_pcm_substream *substream,
+				   struct snd_soc_dai *cpu_dai)
+{
+	struct davinci_mcasp *mcasp = snd_soc_dai_get_drvdata(cpu_dai);
+
+	mcasp->substreams[substream->stream] = NULL;
+	mcasp->active_serializers[substream->stream] = 0;
+
+	if (mcasp->op_mode == DAVINCI_MCASP_DIT_MODE)
+		return;
+
+	if (!snd_soc_dai_active(cpu_dai)) {
+		mcasp->channels = 0;
+		mcasp->max_format_width = 0;
+	}
+}
+
+static const struct snd_soc_dai_ops davinci_mcasp_dai_ops = {
+	.startup	= davinci_mcasp_startup,
+	.shutdown	= davinci_mcasp_shutdown,
+	.trigger	= davinci_mcasp_trigger,
+	.delay		= davinci_mcasp_delay,
+	.hw_params	= davinci_mcasp_hw_params,
+	.set_fmt	= davinci_mcasp_set_dai_fmt,
+	.set_clkdiv	= davinci_mcasp_set_clkdiv,
+	.set_sysclk	= davinci_mcasp_set_sysclk,
+	.set_tdm_slot	= davinci_mcasp_set_tdm_slot,
+};
+
+static int davinci_mcasp_iec958_info(struct snd_kcontrol *kcontrol,
+				     struct snd_ctl_elem_info *uinfo)
+{
+	uinfo->type = SNDRV_CTL_ELEM_TYPE_IEC958;
+	uinfo->count = 1;
+
+	return 0;
+}
+
+static int davinci_mcasp_iec958_get(struct snd_kcontrol *kcontrol,
+				    struct snd_ctl_elem_value *uctl)
+{
+	struct snd_soc_dai *cpu_dai = snd_kcontrol_chip(kcontrol);
+	struct davinci_mcasp *mcasp = snd_soc_dai_get_drvdata(cpu_dai);
+
+	memcpy(uctl->value.iec958.status, &mcasp->iec958_status,
+	       sizeof(mcasp->iec958_status));
+
+	return 0;
+}
+
+static int davinci_mcasp_iec958_put(struct snd_kcontrol *kcontrol,
+				    struct snd_ctl_elem_value *uctl)
+{
+	struct snd_soc_dai *cpu_dai = snd_kcontrol_chip(kcontrol);
+	struct davinci_mcasp *mcasp = snd_soc_dai_get_drvdata(cpu_dai);
+
+	memcpy(&mcasp->iec958_status, uctl->value.iec958.status,
+	       sizeof(mcasp->iec958_status));
+
+	return 0;
+}
+
+static int davinci_mcasp_iec958_con_mask_get(struct snd_kcontrol *kcontrol,
+					     struct snd_ctl_elem_value *ucontrol)
+{
+	struct snd_soc_dai *cpu_dai = snd_kcontrol_chip(kcontrol);
+	struct davinci_mcasp *mcasp = snd_soc_dai_get_drvdata(cpu_dai);
+
+	memset(ucontrol->value.iec958.status, 0xff, sizeof(mcasp->iec958_status));
+	return 0;
+}
+
+static const struct snd_kcontrol_new davinci_mcasp_iec958_ctls[] = {
+	{
+		.access = (SNDRV_CTL_ELEM_ACCESS_READWRITE |
+			   SNDRV_CTL_ELEM_ACCESS_VOLATILE),
+		.iface = SNDRV_CTL_ELEM_IFACE_PCM,
+		.name = SNDRV_CTL_NAME_IEC958("", PLAYBACK, DEFAULT),
+		.info = davinci_mcasp_iec958_info,
+		.get = davinci_mcasp_iec958_get,
+		.put = davinci_mcasp_iec958_put,
+	}, {
+		.access = SNDRV_CTL_ELEM_ACCESS_READ,
+		.iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+		.name = SNDRV_CTL_NAME_IEC958("", PLAYBACK, CON_MASK),
+		.info = davinci_mcasp_iec958_info,
+		.get = davinci_mcasp_iec958_con_mask_get,
+	},
+};
+
+static void davinci_mcasp_init_iec958_status(struct davinci_mcasp *mcasp)
+{
+	unsigned char *cs = (u8 *)&mcasp->iec958_status;
+
+	cs[0] = IEC958_AES0_CON_NOT_COPYRIGHT | IEC958_AES0_CON_EMPHASIS_NONE;
+	cs[1] = IEC958_AES1_CON_PCM_CODER;
+	cs[2] = IEC958_AES2_CON_SOURCE_UNSPEC | IEC958_AES2_CON_CHANNEL_UNSPEC;
+	cs[3] = IEC958_AES3_CON_CLOCK_1000PPM;
+}
+
+static int davinci_mcasp_dai_probe(struct snd_soc_dai *dai)
+{
+	struct davinci_mcasp *mcasp = snd_soc_dai_get_drvdata(dai);
+
+	dai->playback_dma_data = &mcasp->dma_data[SNDRV_PCM_STREAM_PLAYBACK];
+	dai->capture_dma_data = &mcasp->dma_data[SNDRV_PCM_STREAM_CAPTURE];
+
+	if (mcasp->op_mode == DAVINCI_MCASP_DIT_MODE) {
+		davinci_mcasp_init_iec958_status(mcasp);
+		snd_soc_add_dai_controls(dai, davinci_mcasp_iec958_ctls,
+					 ARRAY_SIZE(davinci_mcasp_iec958_ctls));
+	}
+
+	return 0;
+}
+
+#define DAVINCI_MCASP_RATES	SNDRV_PCM_RATE_8000_192000
+
+#define DAVINCI_MCASP_PCM_FMTS (SNDRV_PCM_FMTBIT_S8 | \
+				SNDRV_PCM_FMTBIT_U8 | \
+				SNDRV_PCM_FMTBIT_S16_LE | \
+				SNDRV_PCM_FMTBIT_U16_LE | \
+				SNDRV_PCM_FMTBIT_S24_LE | \
+				SNDRV_PCM_FMTBIT_U24_LE | \
+				SNDRV_PCM_FMTBIT_S24_3LE | \
+				SNDRV_PCM_FMTBIT_U24_3LE | \
+				SNDRV_PCM_FMTBIT_S32_LE | \
+				SNDRV_PCM_FMTBIT_U32_LE)
+
+static struct snd_soc_dai_driver davinci_mcasp_dai[] = {
+	{
+		.name		= "davinci-mcasp.0",
+		.probe		= davinci_mcasp_dai_probe,
+		.playback	= {
+			.stream_name = "IIS Playback",
+			.channels_min	= 1,
+			.channels_max	= 32 * 16,
+			.rates 		= DAVINCI_MCASP_RATES,
+			.formats	= DAVINCI_MCASP_PCM_FMTS,
+		},
+		.capture 	= {
+			.stream_name = "IIS Capture",
+			.channels_min 	= 1,
+			.channels_max	= 32 * 16,
+			.rates 		= DAVINCI_MCASP_RATES,
+			.formats	= DAVINCI_MCASP_PCM_FMTS,
+		},
+		.ops 		= &davinci_mcasp_dai_ops,
+
+		.symmetric_rate		= 1,
+	},
+	{
+		.name		= "davinci-mcasp.1",
+		.probe		= davinci_mcasp_dai_probe,
+		.playback 	= {
+			.stream_name = "DIT Playback",
+			.channels_min	= 1,
+			.channels_max	= 384,
+			.rates		= DAVINCI_MCASP_RATES,
+			.formats	= SNDRV_PCM_FMTBIT_S16_LE |
+					  SNDRV_PCM_FMTBIT_S24_LE,
+		},
+		.ops 		= &davinci_mcasp_dai_ops,
+	},
+
+};
+
+static const struct snd_soc_component_driver davinci_mcasp_component = {
+	.name			= "davinci-mcasp",
+	.legacy_dai_naming	= 1,
+};
+
+/* Some HW specific values and defaults. The rest is filled in from DT. */
+static struct davinci_mcasp_pdata dm646x_mcasp_pdata = {
+	.tx_dma_offset = 0x400,
+	.rx_dma_offset = 0x400,
+	.version = MCASP_VERSION_1,
+};
+
+static struct davinci_mcasp_pdata da830_mcasp_pdata = {
+	.tx_dma_offset = 0x2000,
+	.rx_dma_offset = 0x2000,
+	.version = MCASP_VERSION_2,
+};
+
+static struct davinci_mcasp_pdata am33xx_mcasp_pdata = {
+	.tx_dma_offset = 0,
+	.rx_dma_offset = 0,
+	.version = MCASP_VERSION_3,
+};
+
+static struct davinci_mcasp_pdata dra7_mcasp_pdata = {
+	/* The CFG port offset will be calculated if it is needed */
+	.tx_dma_offset = 0,
+	.rx_dma_offset = 0,
+	.version = MCASP_VERSION_4,
+};
+
+static struct davinci_mcasp_pdata omap_mcasp_pdata = {
+	.tx_dma_offset = 0x200,
+	.rx_dma_offset = 0,
+	.version = MCASP_VERSION_OMAP,
+};
+
+static const struct of_device_id mcasp_dt_ids[] = {
+	{
+		.compatible = "ti,dm646x-mcasp-audio",
+		.data = &dm646x_mcasp_pdata,
+	},
+	{
+		.compatible = "ti,da830-mcasp-audio",
+		.data = &da830_mcasp_pdata,
+	},
+	{
+		.compatible = "ti,am33xx-mcasp-audio",
+		.data = &am33xx_mcasp_pdata,
+	},
+	{
+		.compatible = "ti,dra7-mcasp-audio",
+		.data = &dra7_mcasp_pdata,
+	},
+	{
+		.compatible = "ti,omap4-mcasp-audio",
+		.data = &omap_mcasp_pdata,
+	},
+	{ /* sentinel */ }
+};
+MODULE_DEVICE_TABLE(of, mcasp_dt_ids);
+
+static int mcasp_reparent_fck(struct platform_device *pdev)
+{
+	struct device_node *node = pdev->dev.of_node;
+	struct clk *gfclk, *parent_clk;
+	const char *parent_name;
+	int ret;
+
+	if (!node)
+		return 0;
+
+	parent_name = of_get_property(node, "fck_parent", NULL);
+	if (!parent_name)
+		return 0;
+
+	dev_warn(&pdev->dev, "Update the bindings to use assigned-clocks!\n");
+
+	gfclk = clk_get(&pdev->dev, "fck");
+	if (IS_ERR(gfclk)) {
+		dev_err(&pdev->dev, "failed to get fck\n");
+		return PTR_ERR(gfclk);
+	}
+
+	parent_clk = clk_get(NULL, parent_name);
+	if (IS_ERR(parent_clk)) {
+		dev_err(&pdev->dev, "failed to get parent clock\n");
+		ret = PTR_ERR(parent_clk);
+		goto err1;
+	}
+
+	ret = clk_set_parent(gfclk, parent_clk);
+	if (ret) {
+		dev_err(&pdev->dev, "failed to reparent fck\n");
+		goto err2;
+	}
+
+err2:
+	clk_put(parent_clk);
+err1:
+	clk_put(gfclk);
+	return ret;
+}
+
+static bool davinci_mcasp_have_gpiochip(struct davinci_mcasp *mcasp)
+{
+#ifdef CONFIG_OF_GPIO
+	return of_property_read_bool(mcasp->dev->of_node, "gpio-controller");
+#else
+	return false;
+#endif
+}
+
+static int davinci_mcasp_get_config(struct davinci_mcasp *mcasp,
+				    struct platform_device *pdev)
+{
+	const struct of_device_id *match = of_match_device(mcasp_dt_ids, &pdev->dev);
+	struct device_node *np = pdev->dev.of_node;
+	struct davinci_mcasp_pdata *pdata = NULL;
+	const u32 *of_serial_dir32;
+	u32 val;
+	int i;
+
+	if (pdev->dev.platform_data) {
+		pdata = pdev->dev.platform_data;
+		pdata->dismod = DISMOD_LOW;
+		goto out;
+	} else if (match) {
+		pdata = devm_kmemdup(&pdev->dev, match->data, sizeof(*pdata),
+				     GFP_KERNEL);
+		if (!pdata)
+			return -ENOMEM;
+	} else {
+		dev_err(&pdev->dev, "No compatible match found\n");
+		return -EINVAL;
+	}
+
+	if (of_property_read_u32(np, "op-mode", &val) == 0) {
+		pdata->op_mode = val;
+	} else {
+		mcasp->missing_audio_param = true;
+		goto out;
+	}
+
+	if (of_property_read_u32(np, "tdm-slots", &val) == 0) {
+		if (val < 2 || val > 32) {
+			dev_err(&pdev->dev, "tdm-slots must be in rage [2-32]\n");
+			return -EINVAL;
+		}
+
+		pdata->tdm_slots = val;
+	} else if (pdata->op_mode == DAVINCI_MCASP_IIS_MODE) {
+		mcasp->missing_audio_param = true;
+		goto out;
+	}
+
+	of_serial_dir32 = of_get_property(np, "serial-dir", &val);
+	val /= sizeof(u32);
+	if (of_serial_dir32) {
+		u8 *of_serial_dir = devm_kzalloc(&pdev->dev,
+						 (sizeof(*of_serial_dir) * val),
+						 GFP_KERNEL);
+		if (!of_serial_dir)
+			return -ENOMEM;
+
+		for (i = 0; i < val; i++)
+			of_serial_dir[i] = be32_to_cpup(&of_serial_dir32[i]);
+
+		pdata->num_serializer = val;
+		pdata->serial_dir = of_serial_dir;
+	} else {
+		mcasp->missing_audio_param = true;
+		goto out;
+	}
+
+	if (of_property_read_u32(np, "tx-num-evt", &val) == 0)
+		pdata->txnumevt = val;
+
+	if (of_property_read_u32(np, "rx-num-evt", &val) == 0)
+		pdata->rxnumevt = val;
+
+	if (of_property_read_u32(np, "auxclk-fs-ratio", &val) == 0)
+		mcasp->auxclk_fs_ratio = val;
+
+	if (of_property_read_u32(np, "dismod", &val) == 0) {
+		if (val == 0 || val == 2 || val == 3) {
+			pdata->dismod = DISMOD_VAL(val);
+		} else {
+			dev_warn(&pdev->dev, "Invalid dismod value: %u\n", val);
+			pdata->dismod = DISMOD_LOW;
+		}
+	} else {
+		pdata->dismod = DISMOD_LOW;
+	}
+
+out:
+	mcasp->pdata = pdata;
+
+	if (mcasp->missing_audio_param) {
+		if (davinci_mcasp_have_gpiochip(mcasp)) {
+			dev_dbg(&pdev->dev, "Missing DT parameter(s) for audio\n");
+			return 0;
+		}
+
+		dev_err(&pdev->dev, "Insufficient DT parameter(s)\n");
+		return -ENODEV;
+	}
+
+	mcasp->op_mode = pdata->op_mode;
+	/* sanity check for tdm slots parameter */
+	if (mcasp->op_mode == DAVINCI_MCASP_IIS_MODE) {
+		if (pdata->tdm_slots < 2) {
+			dev_warn(&pdev->dev, "invalid tdm slots: %d\n",
+				 pdata->tdm_slots);
+			mcasp->tdm_slots = 2;
+		} else if (pdata->tdm_slots > 32) {
+			dev_warn(&pdev->dev, "invalid tdm slots: %d\n",
+				 pdata->tdm_slots);
+			mcasp->tdm_slots = 32;
+		} else {
+			mcasp->tdm_slots = pdata->tdm_slots;
+		}
+	} else {
+		mcasp->tdm_slots = 32;
+	}
+
+	mcasp->num_serializer = pdata->num_serializer;
+#ifdef CONFIG_PM
+	mcasp->context.xrsr_regs = devm_kcalloc(&pdev->dev,
+						mcasp->num_serializer, sizeof(u32),
+						GFP_KERNEL);
+	if (!mcasp->context.xrsr_regs)
+		return -ENOMEM;
+#endif
+	mcasp->serial_dir = pdata->serial_dir;
+	mcasp->version = pdata->version;
+	mcasp->txnumevt = pdata->txnumevt;
+	mcasp->rxnumevt = pdata->rxnumevt;
+	mcasp->dismod = pdata->dismod;
+
+	return 0;
+}
+
+enum {
+	PCM_EDMA,
+	PCM_SDMA,
+	PCM_UDMA,
+};
+static const char *sdma_prefix = "ti,omap";
+
+static int davinci_mcasp_get_dma_type(struct davinci_mcasp *mcasp)
+{
+	struct dma_chan *chan;
+	const char *tmp;
+	int ret = PCM_EDMA;
+
+	if (!mcasp->dev->of_node)
+		return PCM_EDMA;
+
+	tmp = mcasp->dma_data[SNDRV_PCM_STREAM_PLAYBACK].filter_data;
+	chan = dma_request_chan(mcasp->dev, tmp);
+	if (IS_ERR(chan))
+		return dev_err_probe(mcasp->dev, PTR_ERR(chan),
+				     "Can't verify DMA configuration\n");
+	if (WARN_ON(!chan->device || !chan->device->dev)) {
+		dma_release_channel(chan);
+		return -EINVAL;
+	}
+
+	if (chan->device->dev->of_node)
+		ret = of_property_read_string(chan->device->dev->of_node,
+					      "compatible", &tmp);
+	else
+		dev_dbg(mcasp->dev, "DMA controller has no of-node\n");
+
+	dma_release_channel(chan);
+	if (ret)
+		return ret;
+
+	dev_dbg(mcasp->dev, "DMA controller compatible = \"%s\"\n", tmp);
+	if (!strncmp(tmp, sdma_prefix, strlen(sdma_prefix)))
+		return PCM_SDMA;
+	else if (strstr(tmp, "udmap"))
+		return PCM_UDMA;
+	else if (strstr(tmp, "bcdma"))
+		return PCM_UDMA;
+
+	return PCM_EDMA;
+}
+
+static u32 davinci_mcasp_txdma_offset(struct davinci_mcasp_pdata *pdata)
+{
+	int i;
+	u32 offset = 0;
+
+	if (pdata->version != MCASP_VERSION_4)
+		return pdata->tx_dma_offset;
+
+	for (i = 0; i < pdata->num_serializer; i++) {
+		if (pdata->serial_dir[i] == TX_MODE) {
+			if (!offset) {
+				offset = DAVINCI_MCASP_TXBUF_REG(i);
+			} else {
+				pr_err("%s: Only one serializer allowed!\n",
+				       __func__);
+				break;
+			}
+		}
+	}
+
+	return offset;
+}
+
+static u32 davinci_mcasp_rxdma_offset(struct davinci_mcasp_pdata *pdata)
+{
+	int i;
+	u32 offset = 0;
+
+	if (pdata->version != MCASP_VERSION_4)
+		return pdata->rx_dma_offset;
+
+	for (i = 0; i < pdata->num_serializer; i++) {
+		if (pdata->serial_dir[i] == RX_MODE) {
+			if (!offset) {
+				offset = DAVINCI_MCASP_RXBUF_REG(i);
+			} else {
+				pr_err("%s: Only one serializer allowed!\n",
+				       __func__);
+				break;
+			}
+		}
+	}
+
+	return offset;
+}
+
+#ifdef CONFIG_GPIOLIB
+static int davinci_mcasp_gpio_request(struct gpio_chip *chip, unsigned offset)
+{
+	struct davinci_mcasp *mcasp = gpiochip_get_data(chip);
+
+	if (mcasp->num_serializer && offset < mcasp->num_serializer &&
+	    mcasp->serial_dir[offset] != INACTIVE_MODE) {
+		dev_err(mcasp->dev, "AXR%u pin is  used for audio\n", offset);
+		return -EBUSY;
+	}
+
+	/* Do not change the PIN yet */
+	return pm_runtime_resume_and_get(mcasp->dev);
+}
+
+static void davinci_mcasp_gpio_free(struct gpio_chip *chip, unsigned offset)
+{
+	struct davinci_mcasp *mcasp = gpiochip_get_data(chip);
+
+	/* Set the direction to input */
+	mcasp_clr_bits(mcasp, DAVINCI_MCASP_PDIR_REG, BIT(offset));
+
+	/* Set the pin as McASP pin */
+	mcasp_clr_bits(mcasp, DAVINCI_MCASP_PFUNC_REG, BIT(offset));
+
+	pm_runtime_put_sync(mcasp->dev);
+}
+
+static int davinci_mcasp_gpio_direction_out(struct gpio_chip *chip,
+					    unsigned offset, int value)
+{
+	struct davinci_mcasp *mcasp = gpiochip_get_data(chip);
+	u32 val;
+
+	if (value)
+		mcasp_set_bits(mcasp, DAVINCI_MCASP_PDOUT_REG, BIT(offset));
+	else
+		mcasp_clr_bits(mcasp, DAVINCI_MCASP_PDOUT_REG, BIT(offset));
+
+	val = mcasp_get_reg(mcasp, DAVINCI_MCASP_PFUNC_REG);
+	if (!(val & BIT(offset))) {
+		/* Set the pin as GPIO pin */
+		mcasp_set_bits(mcasp, DAVINCI_MCASP_PFUNC_REG, BIT(offset));
+
+		/* Set the direction to output */
+		mcasp_set_bits(mcasp, DAVINCI_MCASP_PDIR_REG, BIT(offset));
+	}
+
+	return 0;
+}
+
+static void davinci_mcasp_gpio_set(struct gpio_chip *chip, unsigned offset,
+				  int value)
+{
+	struct davinci_mcasp *mcasp = gpiochip_get_data(chip);
+
+	if (value)
+		mcasp_set_bits(mcasp, DAVINCI_MCASP_PDOUT_REG, BIT(offset));
+	else
+		mcasp_clr_bits(mcasp, DAVINCI_MCASP_PDOUT_REG, BIT(offset));
+}
+
+static int davinci_mcasp_gpio_direction_in(struct gpio_chip *chip,
+					   unsigned offset)
+{
+	struct davinci_mcasp *mcasp = gpiochip_get_data(chip);
+	u32 val;
+
+	val = mcasp_get_reg(mcasp, DAVINCI_MCASP_PFUNC_REG);
+	if (!(val & BIT(offset))) {
+		/* Set the direction to input */
+		mcasp_clr_bits(mcasp, DAVINCI_MCASP_PDIR_REG, BIT(offset));
+
+		/* Set the pin as GPIO pin */
+		mcasp_set_bits(mcasp, DAVINCI_MCASP_PFUNC_REG, BIT(offset));
+	}
+
+	return 0;
+}
+
+static int davinci_mcasp_gpio_get(struct gpio_chip *chip, unsigned offset)
+{
+	struct davinci_mcasp *mcasp = gpiochip_get_data(chip);
+	u32 val;
+
+	val = mcasp_get_reg(mcasp, DAVINCI_MCASP_PDSET_REG);
+	if (val & BIT(offset))
+		return 1;
+
+	return 0;
+}
+
+static int davinci_mcasp_gpio_get_direction(struct gpio_chip *chip,
+					    unsigned offset)
+{
+	struct davinci_mcasp *mcasp = gpiochip_get_data(chip);
+	u32 val;
+
+	val = mcasp_get_reg(mcasp, DAVINCI_MCASP_PDIR_REG);
+	if (val & BIT(offset))
+		return 0;
+
+	return 1;
+}
+
+static const struct gpio_chip davinci_mcasp_template_chip = {
+	.owner			= THIS_MODULE,
+	.request		= davinci_mcasp_gpio_request,
+	.free			= davinci_mcasp_gpio_free,
+	.direction_output	= davinci_mcasp_gpio_direction_out,
+	.set			= davinci_mcasp_gpio_set,
+	.direction_input	= davinci_mcasp_gpio_direction_in,
+	.get			= davinci_mcasp_gpio_get,
+	.get_direction		= davinci_mcasp_gpio_get_direction,
+	.base			= -1,
+	.ngpio			= 32,
+};
+
+static int davinci_mcasp_init_gpiochip(struct davinci_mcasp *mcasp)
+{
+	if (!davinci_mcasp_have_gpiochip(mcasp))
+		return 0;
+
+	mcasp->gpio_chip = davinci_mcasp_template_chip;
+	mcasp->gpio_chip.label = dev_name(mcasp->dev);
+	mcasp->gpio_chip.parent = mcasp->dev;
+
+	return devm_gpiochip_add_data(mcasp->dev, &mcasp->gpio_chip, mcasp);
+}
+
+#else /* CONFIG_GPIOLIB */
+static inline int davinci_mcasp_init_gpiochip(struct davinci_mcasp *mcasp)
+{
+	return 0;
+}
+#endif /* CONFIG_GPIOLIB */
+
+static int davinci_mcasp_probe(struct platform_device *pdev)
+{
+	struct snd_dmaengine_dai_dma_data *dma_data;
+	struct resource *mem, *dat;
+	struct davinci_mcasp *mcasp;
+	char *irq_name;
+	int irq;
+	int ret;
+
+	if (!pdev->dev.platform_data && !pdev->dev.of_node) {
+		dev_err(&pdev->dev, "No platform data supplied\n");
+		return -EINVAL;
+	}
+
+	mcasp = devm_kzalloc(&pdev->dev, sizeof(struct davinci_mcasp),
+			   GFP_KERNEL);
+	if (!mcasp)
+		return	-ENOMEM;
+
+	mem = platform_get_resource_byname(pdev, IORESOURCE_MEM, "mpu");
+	if (!mem) {
+		dev_warn(&pdev->dev,
+			 "\"mpu\" mem resource not found, using index 0\n");
+		mem = platform_get_resource(pdev, IORESOURCE_MEM, 0);
+		if (!mem) {
+			dev_err(&pdev->dev, "no mem resource?\n");
+			return -ENODEV;
+		}
+	}
+
+	mcasp->base = devm_ioremap_resource(&pdev->dev, mem);
+	if (IS_ERR(mcasp->base))
+		return PTR_ERR(mcasp->base);
+
+	dev_set_drvdata(&pdev->dev, mcasp);
+	pm_runtime_enable(&pdev->dev);
+
+	mcasp->dev = &pdev->dev;
+	ret = davinci_mcasp_get_config(mcasp, pdev);
+	if (ret)
+		goto err;
+
+	/* All PINS as McASP */
+	pm_runtime_get_sync(mcasp->dev);
+	mcasp_set_reg(mcasp, DAVINCI_MCASP_PFUNC_REG, 0x00000000);
+	pm_runtime_put(mcasp->dev);
+
+	/* Skip audio related setup code if the configuration is not adequat */
+	if (mcasp->missing_audio_param)
+		goto no_audio;
+
+	irq = platform_get_irq_byname_optional(pdev, "common");
+	if (irq > 0) {
+		irq_name = devm_kasprintf(&pdev->dev, GFP_KERNEL, "%s_common",
+					  dev_name(&pdev->dev));
+		if (!irq_name) {
+			ret = -ENOMEM;
+			goto err;
+		}
+		ret = devm_request_threaded_irq(&pdev->dev, irq, NULL,
+						davinci_mcasp_common_irq_handler,
+						IRQF_ONESHOT | IRQF_SHARED,
+						irq_name, mcasp);
+		if (ret) {
+			dev_err(&pdev->dev, "common IRQ request failed\n");
+			goto err;
+		}
+
+		mcasp->irq_request[SNDRV_PCM_STREAM_PLAYBACK] = XUNDRN;
+		mcasp->irq_request[SNDRV_PCM_STREAM_CAPTURE] = ROVRN;
+	}
+
+	irq = platform_get_irq_byname_optional(pdev, "rx");
+	if (irq > 0) {
+		irq_name = devm_kasprintf(&pdev->dev, GFP_KERNEL, "%s_rx",
+					  dev_name(&pdev->dev));
+		if (!irq_name) {
+			ret = -ENOMEM;
+			goto err;
+		}
+		ret = devm_request_threaded_irq(&pdev->dev, irq, NULL,
+						davinci_mcasp_rx_irq_handler,
+						IRQF_ONESHOT, irq_name, mcasp);
+		if (ret) {
+			dev_err(&pdev->dev, "RX IRQ request failed\n");
+			goto err;
+		}
+
+		mcasp->irq_request[SNDRV_PCM_STREAM_CAPTURE] = ROVRN;
+	}
+
+	irq = platform_get_irq_byname_optional(pdev, "tx");
+	if (irq > 0) {
+		irq_name = devm_kasprintf(&pdev->dev, GFP_KERNEL, "%s_tx",
+					  dev_name(&pdev->dev));
+		if (!irq_name) {
+			ret = -ENOMEM;
+			goto err;
+		}
+		ret = devm_request_threaded_irq(&pdev->dev, irq, NULL,
+						davinci_mcasp_tx_irq_handler,
+						IRQF_ONESHOT, irq_name, mcasp);
+		if (ret) {
+			dev_err(&pdev->dev, "TX IRQ request failed\n");
+			goto err;
+		}
+
+		mcasp->irq_request[SNDRV_PCM_STREAM_PLAYBACK] = XUNDRN;
+	}
+
+	dat = platform_get_resource_byname(pdev, IORESOURCE_MEM, "dat");
+	if (dat)
+		mcasp->dat_port = true;
+
+	dma_data = &mcasp->dma_data[SNDRV_PCM_STREAM_PLAYBACK];
+	dma_data->filter_data = "tx";
+	if (dat) {
+		dma_data->addr = dat->start;
+		/*
+		 * According to the TRM there should be 0x200 offset added to
+		 * the DAT port address
+		 */
+		if (mcasp->version == MCASP_VERSION_OMAP)
+			dma_data->addr += davinci_mcasp_txdma_offset(mcasp->pdata);
+	} else {
+		dma_data->addr = mem->start + davinci_mcasp_txdma_offset(mcasp->pdata);
+	}
+
+
+	/* RX is not valid in DIT mode */
+	if (mcasp->op_mode != DAVINCI_MCASP_DIT_MODE) {
+		dma_data = &mcasp->dma_data[SNDRV_PCM_STREAM_CAPTURE];
+		dma_data->filter_data = "rx";
+		if (dat)
+			dma_data->addr = dat->start;
+		else
+			dma_data->addr =
+				mem->start + davinci_mcasp_rxdma_offset(mcasp->pdata);
+	}
+
+	if (mcasp->version < MCASP_VERSION_3) {
+		mcasp->fifo_base = DAVINCI_MCASP_V2_AFIFO_BASE;
+		/* dma_params->dma_addr is pointing to the data port address */
+		mcasp->dat_port = true;
+	} else {
+		mcasp->fifo_base = DAVINCI_MCASP_V3_AFIFO_BASE;
+	}
+
+	/* Allocate memory for long enough list for all possible
+	 * scenarios. Maximum number tdm slots is 32 and there cannot
+	 * be more serializers than given in the configuration.  The
+	 * serializer directions could be taken into account, but it
+	 * would make code much more complex and save only couple of
+	 * bytes.
+	 */
+	mcasp->chconstr[SNDRV_PCM_STREAM_PLAYBACK].list =
+		devm_kcalloc(mcasp->dev,
+			     32 + mcasp->num_serializer - 1,
+			     sizeof(unsigned int),
+			     GFP_KERNEL);
+
+	mcasp->chconstr[SNDRV_PCM_STREAM_CAPTURE].list =
+		devm_kcalloc(mcasp->dev,
+			     32 + mcasp->num_serializer - 1,
+			     sizeof(unsigned int),
+			     GFP_KERNEL);
+
+	if (!mcasp->chconstr[SNDRV_PCM_STREAM_PLAYBACK].list ||
+	    !mcasp->chconstr[SNDRV_PCM_STREAM_CAPTURE].list) {
+		ret = -ENOMEM;
+		goto err;
+	}
+
+	ret = davinci_mcasp_set_ch_constraints(mcasp);
+	if (ret)
+		goto err;
+
+	mcasp_reparent_fck(pdev);
+
+	ret = devm_snd_soc_register_component(&pdev->dev, &davinci_mcasp_component,
+					      &davinci_mcasp_dai[mcasp->op_mode], 1);
+
+	if (ret != 0)
+		goto err;
+
+	ret = davinci_mcasp_get_dma_type(mcasp);
+	switch (ret) {
+	case PCM_EDMA:
+		ret = edma_pcm_platform_register(&pdev->dev);
+		break;
+	case PCM_SDMA:
+		if (mcasp->op_mode == DAVINCI_MCASP_IIS_MODE)
+			ret = sdma_pcm_platform_register(&pdev->dev, "tx", "rx");
+		else
+			ret = sdma_pcm_platform_register(&pdev->dev, "tx", NULL);
+		break;
+	case PCM_UDMA:
+		ret = udma_pcm_platform_register(&pdev->dev);
+		break;
+	default:
+		dev_err(&pdev->dev, "No DMA controller found (%d)\n", ret);
+		fallthrough;
+	case -EPROBE_DEFER:
+		goto err;
+	}
+
+	if (ret) {
+		dev_err(&pdev->dev, "register PCM failed: %d\n", ret);
+		goto err;
+	}
+
+no_audio:
+	ret = davinci_mcasp_init_gpiochip(mcasp);
+	if (ret) {
+		dev_err(&pdev->dev, "gpiochip registration failed: %d\n", ret);
+		goto err;
+	}
+
+	return 0;
+err:
+	pm_runtime_disable(&pdev->dev);
+	return ret;
+}
+
+static int davinci_mcasp_remove(struct platform_device *pdev)
+{
+	pm_runtime_disable(&pdev->dev);
+
+	return 0;
+}
+
+#ifdef CONFIG_PM
+static int davinci_mcasp_runtime_suspend(struct device *dev)
+{
+	struct davinci_mcasp *mcasp = dev_get_drvdata(dev);
+	struct davinci_mcasp_context *context = &mcasp->context;
+	u32 reg;
+	int i;
+
+	for (i = 0; i < ARRAY_SIZE(context_regs); i++)
+		context->config_regs[i] = mcasp_get_reg(mcasp, context_regs[i]);
+
+	if (mcasp->txnumevt) {
+		reg = mcasp->fifo_base + MCASP_WFIFOCTL_OFFSET;
+		context->afifo_regs[0] = mcasp_get_reg(mcasp, reg);
+	}
+	if (mcasp->rxnumevt) {
+		reg = mcasp->fifo_base + MCASP_RFIFOCTL_OFFSET;
+		context->afifo_regs[1] = mcasp_get_reg(mcasp, reg);
+	}
+
+	for (i = 0; i < mcasp->num_serializer; i++)
+		context->xrsr_regs[i] = mcasp_get_reg(mcasp,
+						DAVINCI_MCASP_XRSRCTL_REG(i));
+
+	return 0;
+}
+
+static int davinci_mcasp_runtime_resume(struct device *dev)
+{
+	struct davinci_mcasp *mcasp = dev_get_drvdata(dev);
+	struct davinci_mcasp_context *context = &mcasp->context;
+	u32 reg;
+	int i;
+
+	for (i = 0; i < ARRAY_SIZE(context_regs); i++)
+		mcasp_set_reg(mcasp, context_regs[i], context->config_regs[i]);
+
+	if (mcasp->txnumevt) {
+		reg = mcasp->fifo_base + MCASP_WFIFOCTL_OFFSET;
+		mcasp_set_reg(mcasp, reg, context->afifo_regs[0]);
+	}
+	if (mcasp->rxnumevt) {
+		reg = mcasp->fifo_base + MCASP_RFIFOCTL_OFFSET;
+		mcasp_set_reg(mcasp, reg, context->afifo_regs[1]);
+	}
+
+	for (i = 0; i < mcasp->num_serializer; i++)
+		mcasp_set_reg(mcasp, DAVINCI_MCASP_XRSRCTL_REG(i),
+			      context->xrsr_regs[i]);
+
+	return 0;
+}
+
+#endif
+
+static const struct dev_pm_ops davinci_mcasp_pm_ops = {
+	SET_RUNTIME_PM_OPS(davinci_mcasp_runtime_suspend,
+			   davinci_mcasp_runtime_resume,
+			   NULL)
+};
+
+static struct platform_driver davinci_mcasp_driver = {
+	.probe		= davinci_mcasp_probe,
+	.remove		= davinci_mcasp_remove,
+	.driver		= {
+		.name	= "davinci-mcasp",
+		.pm     = &davinci_mcasp_pm_ops,
+		.of_match_table = mcasp_dt_ids,
+	},
+};
+
+module_platform_driver(davinci_mcasp_driver);
+
+MODULE_AUTHOR("Steve Chen");
+MODULE_DESCRIPTION("TI DAVINCI McASP SoC Interface");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/ti/davinci-mcasp.h b/sound/soc/ti/davinci-mcasp.h
new file mode 100644
index 000000000..5de2b8a31
--- /dev/null
+++ b/sound/soc/ti/davinci-mcasp.h
@@ -0,0 +1,307 @@
+/* SPDX-License-Identifier: GPL-2.0-only */
+/*
+ * ALSA SoC McASP Audio Layer for TI DAVINCI processor
+ *
+ * MCASP related definitions
+ *
+ * Author: Nirmal Pandey <n-pandey@ti.com>,
+ *         Suresh Rajashekara <suresh.r@ti.com>
+ *         Steve Chen <schen@.mvista.com>
+ *
+ * Copyright:   (C) 2009 MontaVista Software, Inc., <source@mvista.com>
+ * Copyright:   (C) 2009  Texas Instruments, India
+ */
+
+#ifndef DAVINCI_MCASP_H
+#define DAVINCI_MCASP_H
+
+/*
+ * McASP register definitions
+ */
+#define DAVINCI_MCASP_PID_REG		0x00
+#define DAVINCI_MCASP_PWREMUMGT_REG	0x04
+
+#define DAVINCI_MCASP_PFUNC_REG		0x10
+#define DAVINCI_MCASP_PDIR_REG		0x14
+#define DAVINCI_MCASP_PDOUT_REG		0x18
+#define DAVINCI_MCASP_PDSET_REG		0x1c
+
+#define DAVINCI_MCASP_PDCLR_REG		0x20
+
+#define DAVINCI_MCASP_TLGC_REG		0x30
+#define DAVINCI_MCASP_TLMR_REG		0x34
+
+#define DAVINCI_MCASP_GBLCTL_REG	0x44
+#define DAVINCI_MCASP_AMUTE_REG		0x48
+#define DAVINCI_MCASP_LBCTL_REG		0x4c
+
+#define DAVINCI_MCASP_TXDITCTL_REG	0x50
+
+#define DAVINCI_MCASP_GBLCTLR_REG	0x60
+#define DAVINCI_MCASP_RXMASK_REG	0x64
+#define DAVINCI_MCASP_RXFMT_REG		0x68
+#define DAVINCI_MCASP_RXFMCTL_REG	0x6c
+
+#define DAVINCI_MCASP_ACLKRCTL_REG	0x70
+#define DAVINCI_MCASP_AHCLKRCTL_REG	0x74
+#define DAVINCI_MCASP_RXTDM_REG		0x78
+#define DAVINCI_MCASP_EVTCTLR_REG	0x7c
+
+#define DAVINCI_MCASP_RXSTAT_REG	0x80
+#define DAVINCI_MCASP_RXTDMSLOT_REG	0x84
+#define DAVINCI_MCASP_RXCLKCHK_REG	0x88
+#define DAVINCI_MCASP_REVTCTL_REG	0x8c
+
+#define DAVINCI_MCASP_GBLCTLX_REG	0xa0
+#define DAVINCI_MCASP_TXMASK_REG	0xa4
+#define DAVINCI_MCASP_TXFMT_REG		0xa8
+#define DAVINCI_MCASP_TXFMCTL_REG	0xac
+
+#define DAVINCI_MCASP_ACLKXCTL_REG	0xb0
+#define DAVINCI_MCASP_AHCLKXCTL_REG	0xb4
+#define DAVINCI_MCASP_TXTDM_REG		0xb8
+#define DAVINCI_MCASP_EVTCTLX_REG	0xbc
+
+#define DAVINCI_MCASP_TXSTAT_REG	0xc0
+#define DAVINCI_MCASP_TXTDMSLOT_REG	0xc4
+#define DAVINCI_MCASP_TXCLKCHK_REG	0xc8
+#define DAVINCI_MCASP_XEVTCTL_REG	0xcc
+
+/* Left(even TDM Slot) Channel Status Register File */
+#define DAVINCI_MCASP_DITCSRA_REG	0x100
+/* Right(odd TDM slot) Channel Status Register File */
+#define DAVINCI_MCASP_DITCSRB_REG	0x118
+/* Left(even TDM slot) User Data Register File */
+#define DAVINCI_MCASP_DITUDRA_REG	0x130
+/* Right(odd TDM Slot) User Data Register File */
+#define DAVINCI_MCASP_DITUDRB_REG	0x148
+
+/* Serializer n Control Register */
+#define DAVINCI_MCASP_XRSRCTL_BASE_REG	0x180
+#define DAVINCI_MCASP_XRSRCTL_REG(n)	(DAVINCI_MCASP_XRSRCTL_BASE_REG + \
+						(n << 2))
+
+/* Transmit Buffer for Serializer n */
+#define DAVINCI_MCASP_TXBUF_REG(n)	(0x200 + (n << 2))
+/* Receive Buffer for Serializer n */
+#define DAVINCI_MCASP_RXBUF_REG(n)	(0x280 + (n << 2))
+
+/* McASP FIFO Registers */
+#define DAVINCI_MCASP_V2_AFIFO_BASE	(0x1010)
+#define DAVINCI_MCASP_V3_AFIFO_BASE	(0x1000)
+
+/* FIFO register offsets from AFIFO base */
+#define MCASP_WFIFOCTL_OFFSET		(0x0)
+#define MCASP_WFIFOSTS_OFFSET		(0x4)
+#define MCASP_RFIFOCTL_OFFSET		(0x8)
+#define MCASP_RFIFOSTS_OFFSET		(0xc)
+
+/*
+ * DAVINCI_MCASP_PWREMUMGT_REG - Power Down and Emulation Management
+ *     Register Bits
+ */
+#define MCASP_FREE	BIT(0)
+#define MCASP_SOFT	BIT(1)
+
+/*
+ * DAVINCI_MCASP_PFUNC_REG - Pin Function / GPIO Enable Register Bits
+ * DAVINCI_MCASP_PDIR_REG - Pin Direction Register Bits
+ * DAVINCI_MCASP_PDOUT_REG - Pin output in GPIO mode
+ * DAVINCI_MCASP_PDSET_REG - Pin input in GPIO mode
+ */
+#define PIN_BIT_AXR(n)	(n)
+#define PIN_BIT_AMUTE	25
+#define PIN_BIT_ACLKX	26
+#define PIN_BIT_AHCLKX	27
+#define PIN_BIT_AFSX	28
+#define PIN_BIT_ACLKR	29
+#define PIN_BIT_AHCLKR	30
+#define PIN_BIT_AFSR	31
+
+/*
+ * DAVINCI_MCASP_TXDITCTL_REG - Transmit DIT Control Register Bits
+ */
+#define DITEN	BIT(0)	/* Transmit DIT mode enable/disable */
+#define VA	BIT(2)
+#define VB	BIT(3)
+
+/*
+ * DAVINCI_MCASP_TXFMT_REG - Transmit Bitstream Format Register Bits
+ */
+#define TXROT(val)	(val)
+#define TXSEL		BIT(3)
+#define TXSSZ(val)	(val<<4)
+#define TXPBIT(val)	(val<<8)
+#define TXPAD(val)	(val<<13)
+#define TXORD		BIT(15)
+#define FSXDLY(val)	(val<<16)
+
+/*
+ * DAVINCI_MCASP_RXFMT_REG - Receive Bitstream Format Register Bits
+ */
+#define RXROT(val)	(val)
+#define RXSEL		BIT(3)
+#define RXSSZ(val)	(val<<4)
+#define RXPBIT(val)	(val<<8)
+#define RXPAD(val)	(val<<13)
+#define RXORD		BIT(15)
+#define FSRDLY(val)	(val<<16)
+
+/*
+ * DAVINCI_MCASP_TXFMCTL_REG -  Transmit Frame Control Register Bits
+ */
+#define FSXPOL		BIT(0)
+#define AFSXE		BIT(1)
+#define FSXDUR		BIT(4)
+#define FSXMOD(val)	(val<<7)
+
+/*
+ * DAVINCI_MCASP_RXFMCTL_REG - Receive Frame Control Register Bits
+ */
+#define FSRPOL		BIT(0)
+#define AFSRE		BIT(1)
+#define FSRDUR		BIT(4)
+#define FSRMOD(val)	(val<<7)
+
+/*
+ * DAVINCI_MCASP_ACLKXCTL_REG - Transmit Clock Control Register Bits
+ */
+#define ACLKXDIV(val)	(val)
+#define ACLKXE		BIT(5)
+#define TX_ASYNC	BIT(6)
+#define ACLKXPOL	BIT(7)
+#define ACLKXDIV_MASK	0x1f
+
+/*
+ * DAVINCI_MCASP_ACLKRCTL_REG Receive Clock Control Register Bits
+ */
+#define ACLKRDIV(val)	(val)
+#define ACLKRE		BIT(5)
+#define RX_ASYNC	BIT(6)
+#define ACLKRPOL	BIT(7)
+#define ACLKRDIV_MASK	0x1f
+
+/*
+ * DAVINCI_MCASP_AHCLKXCTL_REG - High Frequency Transmit Clock Control
+ *     Register Bits
+ */
+#define AHCLKXDIV(val)	(val)
+#define AHCLKXPOL	BIT(14)
+#define AHCLKXE		BIT(15)
+#define AHCLKXDIV_MASK	0xfff
+
+/*
+ * DAVINCI_MCASP_AHCLKRCTL_REG - High Frequency Receive Clock Control
+ *     Register Bits
+ */
+#define AHCLKRDIV(val)	(val)
+#define AHCLKRPOL	BIT(14)
+#define AHCLKRE		BIT(15)
+#define AHCLKRDIV_MASK	0xfff
+
+/*
+ * DAVINCI_MCASP_XRSRCTL_BASE_REG -  Serializer Control Register Bits
+ */
+#define MODE(val)	(val)
+#define DISMOD_3STATE	(0x0)
+#define DISMOD_LOW	(0x2 << 2)
+#define DISMOD_HIGH	(0x3 << 2)
+#define DISMOD_VAL(x)	((x) << 2)
+#define DISMOD_MASK	DISMOD_HIGH
+#define TXSTATE		BIT(4)
+#define RXSTATE		BIT(5)
+#define SRMOD_MASK	3
+#define SRMOD_INACTIVE	0
+
+/*
+ * DAVINCI_MCASP_LBCTL_REG - Loop Back Control Register Bits
+ */
+#define LBEN		BIT(0)
+#define LBORD		BIT(1)
+#define LBGENMODE(val)	(val<<2)
+
+/*
+ * DAVINCI_MCASP_TXTDMSLOT_REG - Transmit TDM Slot Register configuration
+ */
+#define TXTDMS(n)	(1<<n)
+
+/*
+ * DAVINCI_MCASP_RXTDMSLOT_REG - Receive TDM Slot Register configuration
+ */
+#define RXTDMS(n)	(1<<n)
+
+/*
+ * DAVINCI_MCASP_GBLCTL_REG -  Global Control Register Bits
+ */
+#define RXCLKRST	BIT(0)	/* Receiver Clock Divider Reset */
+#define RXHCLKRST	BIT(1)	/* Receiver High Frequency Clock Divider */
+#define RXSERCLR	BIT(2)	/* Receiver Serializer Clear */
+#define RXSMRST		BIT(3)	/* Receiver State Machine Reset */
+#define RXFSRST		BIT(4)	/* Frame Sync Generator Reset */
+#define TXCLKRST	BIT(8)	/* Transmitter Clock Divider Reset */
+#define TXHCLKRST	BIT(9)	/* Transmitter High Frequency Clock Divider*/
+#define TXSERCLR	BIT(10)	/* Transmit Serializer Clear */
+#define TXSMRST		BIT(11)	/* Transmitter State Machine Reset */
+#define TXFSRST		BIT(12)	/* Frame Sync Generator Reset */
+
+/*
+ * DAVINCI_MCASP_TXSTAT_REG - Transmitter Status Register Bits
+ * DAVINCI_MCASP_RXSTAT_REG - Receiver Status Register Bits
+ */
+#define XRERR		BIT(8) /* Transmit/Receive error */
+#define XRDATA		BIT(5) /* Transmit/Receive data ready */
+
+/*
+ * DAVINCI_MCASP_AMUTE_REG -  Mute Control Register Bits
+ */
+#define MUTENA(val)	(val)
+#define MUTEINPOL	BIT(2)
+#define MUTEINENA	BIT(3)
+#define MUTEIN		BIT(4)
+#define MUTER		BIT(5)
+#define MUTEX		BIT(6)
+#define MUTEFSR		BIT(7)
+#define MUTEFSX		BIT(8)
+#define MUTEBADCLKR	BIT(9)
+#define MUTEBADCLKX	BIT(10)
+#define MUTERXDMAERR	BIT(11)
+#define MUTETXDMAERR	BIT(12)
+
+/*
+ * DAVINCI_MCASP_REVTCTL_REG - Receiver DMA Event Control Register bits
+ */
+#define RXDATADMADIS	BIT(0)
+
+/*
+ * DAVINCI_MCASP_XEVTCTL_REG - Transmitter DMA Event Control Register bits
+ */
+#define TXDATADMADIS	BIT(0)
+
+/*
+ * DAVINCI_MCASP_EVTCTLR_REG - Receiver Interrupt Control Register Bits
+ */
+#define ROVRN		BIT(0)
+
+/*
+ * DAVINCI_MCASP_EVTCTLX_REG - Transmitter Interrupt Control Register Bits
+ */
+#define XUNDRN		BIT(0)
+
+/*
+ * DAVINCI_MCASP_W[R]FIFOCTL - Write/Read FIFO Control Register bits
+ */
+#define FIFO_ENABLE	BIT(16)
+#define NUMEVT_MASK	(0xFF << 8)
+#define NUMEVT(x)	(((x) & 0xFF) << 8)
+#define NUMDMA_MASK	(0xFF)
+
+/* Source of High-frequency transmit/receive clock */
+#define MCASP_CLK_HCLK_AHCLK		0 /* AHCLKX/R */
+#define MCASP_CLK_HCLK_AUXCLK		1 /* Internal functional clock */
+
+/* clock divider IDs */
+#define MCASP_CLKDIV_AUXCLK		0 /* HCLK divider from AUXCLK */
+#define MCASP_CLKDIV_BCLK		1 /* BCLK divider from HCLK */
+#define MCASP_CLKDIV_BCLK_FS_RATIO	2 /* to set BCLK FS ration */
+
+#endif	/* DAVINCI_MCASP_H */
diff --git a/sound/soc/ti/davinci-vcif.c b/sound/soc/ti/davinci-vcif.c
new file mode 100644
index 000000000..36fa97e2b
--- /dev/null
+++ b/sound/soc/ti/davinci-vcif.c
@@ -0,0 +1,247 @@
+// SPDX-License-Identifier: GPL-2.0-or-later
+/*
+ * ALSA SoC Voice Codec Interface for TI DAVINCI processor
+ *
+ * Copyright (C) 2010 Texas Instruments.
+ *
+ * Author: Miguel Aguilar <miguel.aguilar@ridgerun.com>
+ */
+
+#include <linux/init.h>
+#include <linux/module.h>
+#include <linux/device.h>
+#include <linux/delay.h>
+#include <linux/slab.h>
+#include <linux/io.h>
+#include <linux/mfd/davinci_voicecodec.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 "edma-pcm.h"
+#include "davinci-i2s.h"
+
+#define MOD_REG_BIT(val, mask, set) do { \
+	if (set) { \
+		val |= mask; \
+	} else { \
+		val &= ~mask; \
+	} \
+} while (0)
+
+struct davinci_vcif_dev {
+	struct davinci_vc *davinci_vc;
+	struct snd_dmaengine_dai_dma_data dma_data[2];
+	int dma_request[2];
+};
+
+static void davinci_vcif_start(struct snd_pcm_substream *substream)
+{
+	struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream);
+	struct davinci_vcif_dev *davinci_vcif_dev =
+			snd_soc_dai_get_drvdata(asoc_rtd_to_cpu(rtd, 0));
+	struct davinci_vc *davinci_vc = davinci_vcif_dev->davinci_vc;
+	u32 w;
+
+	/* Start the sample generator and enable transmitter/receiver */
+	w = readl(davinci_vc->base + DAVINCI_VC_CTRL);
+
+	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+		MOD_REG_BIT(w, DAVINCI_VC_CTRL_RSTDAC, 0);
+	else
+		MOD_REG_BIT(w, DAVINCI_VC_CTRL_RSTADC, 0);
+
+	writel(w, davinci_vc->base + DAVINCI_VC_CTRL);
+}
+
+static void davinci_vcif_stop(struct snd_pcm_substream *substream)
+{
+	struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream);
+	struct davinci_vcif_dev *davinci_vcif_dev =
+			snd_soc_dai_get_drvdata(asoc_rtd_to_cpu(rtd, 0));
+	struct davinci_vc *davinci_vc = davinci_vcif_dev->davinci_vc;
+	u32 w;
+
+	/* Reset transmitter/receiver and sample rate/frame sync generators */
+	w = readl(davinci_vc->base + DAVINCI_VC_CTRL);
+	if (substream->stream == SNDRV_PCM_STREAM_PLAYBACK)
+		MOD_REG_BIT(w, DAVINCI_VC_CTRL_RSTDAC, 1);
+	else
+		MOD_REG_BIT(w, DAVINCI_VC_CTRL_RSTADC, 1);
+
+	writel(w, davinci_vc->base + DAVINCI_VC_CTRL);
+}
+
+static int davinci_vcif_hw_params(struct snd_pcm_substream *substream,
+				  struct snd_pcm_hw_params *params,
+				  struct snd_soc_dai *dai)
+{
+	struct davinci_vcif_dev *davinci_vcif_dev = snd_soc_dai_get_drvdata(dai);
+	struct davinci_vc *davinci_vc = davinci_vcif_dev->davinci_vc;
+	u32 w;
+
+	/* Restart the codec before setup */
+	davinci_vcif_stop(substream);
+	davinci_vcif_start(substream);
+
+	/* General line settings */
+	writel(DAVINCI_VC_CTRL_MASK, davinci_vc->base + DAVINCI_VC_CTRL);
+
+	writel(DAVINCI_VC_INT_MASK, davinci_vc->base + DAVINCI_VC_INTCLR);
+
+	writel(DAVINCI_VC_INT_MASK, davinci_vc->base + DAVINCI_VC_INTEN);
+
+	w = readl(davinci_vc->base + DAVINCI_VC_CTRL);
+
+	/* Determine xfer data type */
+	switch (params_format(params)) {
+	case SNDRV_PCM_FORMAT_U8:
+		MOD_REG_BIT(w, DAVINCI_VC_CTRL_RD_BITS_8 |
+			    DAVINCI_VC_CTRL_RD_UNSIGNED |
+			    DAVINCI_VC_CTRL_WD_BITS_8 |
+			    DAVINCI_VC_CTRL_WD_UNSIGNED, 1);
+		break;
+	case SNDRV_PCM_FORMAT_S8:
+		MOD_REG_BIT(w, DAVINCI_VC_CTRL_RD_BITS_8 |
+			    DAVINCI_VC_CTRL_WD_BITS_8, 1);
+
+		MOD_REG_BIT(w, DAVINCI_VC_CTRL_RD_UNSIGNED |
+			    DAVINCI_VC_CTRL_WD_UNSIGNED, 0);
+		break;
+	case SNDRV_PCM_FORMAT_S16_LE:
+		MOD_REG_BIT(w, DAVINCI_VC_CTRL_RD_BITS_8 |
+			    DAVINCI_VC_CTRL_RD_UNSIGNED |
+			    DAVINCI_VC_CTRL_WD_BITS_8 |
+			    DAVINCI_VC_CTRL_WD_UNSIGNED, 0);
+		break;
+	default:
+		printk(KERN_WARNING "davinci-vcif: unsupported PCM format");
+		return -EINVAL;
+	}
+
+	writel(w, davinci_vc->base + DAVINCI_VC_CTRL);
+
+	return 0;
+}
+
+static int davinci_vcif_trigger(struct snd_pcm_substream *substream, int cmd,
+				struct snd_soc_dai *dai)
+{
+	int ret = 0;
+
+	switch (cmd) {
+	case SNDRV_PCM_TRIGGER_START:
+	case SNDRV_PCM_TRIGGER_RESUME:
+	case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
+		davinci_vcif_start(substream);
+		break;
+	case SNDRV_PCM_TRIGGER_STOP:
+	case SNDRV_PCM_TRIGGER_SUSPEND:
+	case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
+		davinci_vcif_stop(substream);
+		break;
+	default:
+		ret = -EINVAL;
+	}
+
+	return ret;
+}
+
+#define DAVINCI_VCIF_RATES	SNDRV_PCM_RATE_8000_48000
+
+static const struct snd_soc_dai_ops davinci_vcif_dai_ops = {
+	.trigger	= davinci_vcif_trigger,
+	.hw_params	= davinci_vcif_hw_params,
+};
+
+static int davinci_vcif_dai_probe(struct snd_soc_dai *dai)
+{
+	struct davinci_vcif_dev *dev = snd_soc_dai_get_drvdata(dai);
+
+	dai->playback_dma_data = &dev->dma_data[SNDRV_PCM_STREAM_PLAYBACK];
+	dai->capture_dma_data = &dev->dma_data[SNDRV_PCM_STREAM_CAPTURE];
+
+	return 0;
+}
+
+static struct snd_soc_dai_driver davinci_vcif_dai = {
+	.probe = davinci_vcif_dai_probe,
+	.playback = {
+		.channels_min = 1,
+		.channels_max = 2,
+		.rates = DAVINCI_VCIF_RATES,
+		.formats = SNDRV_PCM_FMTBIT_S16_LE,},
+	.capture = {
+		.channels_min = 1,
+		.channels_max = 2,
+		.rates = DAVINCI_VCIF_RATES,
+		.formats = SNDRV_PCM_FMTBIT_S16_LE,},
+	.ops = &davinci_vcif_dai_ops,
+
+};
+
+static const struct snd_soc_component_driver davinci_vcif_component = {
+	.name			= "davinci-vcif",
+	.legacy_dai_naming	= 1,
+};
+
+static int davinci_vcif_probe(struct platform_device *pdev)
+{
+	struct davinci_vc *davinci_vc = pdev->dev.platform_data;
+	struct davinci_vcif_dev *davinci_vcif_dev;
+	int ret;
+
+	davinci_vcif_dev = devm_kzalloc(&pdev->dev,
+					sizeof(struct davinci_vcif_dev),
+					GFP_KERNEL);
+	if (!davinci_vcif_dev)
+		return -ENOMEM;
+
+	/* DMA tx params */
+	davinci_vcif_dev->davinci_vc = davinci_vc;
+	davinci_vcif_dev->dma_data[SNDRV_PCM_STREAM_PLAYBACK].filter_data =
+				&davinci_vc->davinci_vcif.dma_tx_channel;
+	davinci_vcif_dev->dma_data[SNDRV_PCM_STREAM_PLAYBACK].addr =
+				davinci_vc->davinci_vcif.dma_tx_addr;
+
+	/* DMA rx params */
+	davinci_vcif_dev->dma_data[SNDRV_PCM_STREAM_CAPTURE].filter_data =
+				&davinci_vc->davinci_vcif.dma_rx_channel;
+	davinci_vcif_dev->dma_data[SNDRV_PCM_STREAM_CAPTURE].addr =
+				davinci_vc->davinci_vcif.dma_rx_addr;
+
+	dev_set_drvdata(&pdev->dev, davinci_vcif_dev);
+
+	ret = devm_snd_soc_register_component(&pdev->dev,
+					      &davinci_vcif_component,
+					      &davinci_vcif_dai, 1);
+	if (ret != 0) {
+		dev_err(&pdev->dev, "could not register dai\n");
+		return ret;
+	}
+
+	ret = edma_pcm_platform_register(&pdev->dev);
+	if (ret) {
+		dev_err(&pdev->dev, "register PCM failed: %d\n", ret);
+		return ret;
+	}
+
+	return 0;
+}
+
+static struct platform_driver davinci_vcif_driver = {
+	.probe		= davinci_vcif_probe,
+	.driver		= {
+		.name	= "davinci-vcif",
+	},
+};
+
+module_platform_driver(davinci_vcif_driver);
+
+MODULE_AUTHOR("Miguel Aguilar");
+MODULE_DESCRIPTION("Texas Instruments DaVinci ASoC Voice Codec Interface");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/ti/edma-pcm.c b/sound/soc/ti/edma-pcm.c
new file mode 100644
index 000000000..634b040b6
--- /dev/null
+++ b/sound/soc/ti/edma-pcm.c
@@ -0,0 +1,63 @@
+// SPDX-License-Identifier: GPL-2.0-only
+/*
+ * edma-pcm.c - eDMA PCM driver using dmaengine for AM3xxx, AM4xxx
+ *
+ * Copyright (C) 2014 Texas Instruments, Inc.
+ *
+ * Author: Peter Ujfalusi <peter.ujfalusi@ti.com>
+ *
+ * Based on: sound/soc/tegra/tegra_pcm.c
+ */
+
+#include <linux/module.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include <sound/dmaengine_pcm.h>
+
+#include "edma-pcm.h"
+
+static const struct snd_pcm_hardware edma_pcm_hardware = {
+	.info			= SNDRV_PCM_INFO_MMAP |
+				  SNDRV_PCM_INFO_MMAP_VALID |
+				  SNDRV_PCM_INFO_PAUSE | SNDRV_PCM_INFO_RESUME |
+				  SNDRV_PCM_INFO_NO_PERIOD_WAKEUP |
+				  SNDRV_PCM_INFO_INTERLEAVED,
+	.buffer_bytes_max	= 128 * 1024,
+	.period_bytes_min	= 32,
+	.period_bytes_max	= 64 * 1024,
+	.periods_min		= 2,
+	.periods_max		= 19, /* Limit by edma dmaengine driver */
+};
+
+static const struct snd_dmaengine_pcm_config edma_dmaengine_pcm_config = {
+	.pcm_hardware = &edma_pcm_hardware,
+	.prepare_slave_config = snd_dmaengine_pcm_prepare_slave_config,
+	.prealloc_buffer_size = 128 * 1024,
+};
+
+int edma_pcm_platform_register(struct device *dev)
+{
+	struct snd_dmaengine_pcm_config *config;
+
+	if (dev->of_node)
+		return devm_snd_dmaengine_pcm_register(dev,
+						&edma_dmaengine_pcm_config, 0);
+
+	config = devm_kzalloc(dev, sizeof(*config), GFP_KERNEL);
+	if (!config)
+		return -ENOMEM;
+
+	*config = edma_dmaengine_pcm_config;
+
+	config->chan_names[0] = "tx";
+	config->chan_names[1] = "rx";
+
+	return devm_snd_dmaengine_pcm_register(dev, config, 0);
+}
+EXPORT_SYMBOL_GPL(edma_pcm_platform_register);
+
+MODULE_AUTHOR("Peter Ujfalusi <peter.ujfalusi@ti.com>");
+MODULE_DESCRIPTION("eDMA PCM ASoC platform driver");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/ti/edma-pcm.h b/sound/soc/ti/edma-pcm.h
new file mode 100644
index 000000000..941d18527
--- /dev/null
+++ b/sound/soc/ti/edma-pcm.h
@@ -0,0 +1,24 @@
+/* SPDX-License-Identifier: GPL-2.0-only */
+/*
+ * edma-pcm.h - eDMA PCM driver using dmaengine for AM3xxx, AM4xxx
+ *
+ * Copyright (C) 2014 Texas Instruments, Inc.
+ *
+ * Author: Peter Ujfalusi <peter.ujfalusi@ti.com>
+ *
+ * Based on: sound/soc/tegra/tegra_pcm.h
+ */
+
+#ifndef __EDMA_PCM_H__
+#define __EDMA_PCM_H__
+
+#if IS_ENABLED(CONFIG_SND_SOC_TI_EDMA_PCM)
+int edma_pcm_platform_register(struct device *dev);
+#else
+static inline int edma_pcm_platform_register(struct device *dev)
+{
+	return 0;
+}
+#endif /* CONFIG_SND_SOC_TI_EDMA_PCM */
+
+#endif /* __EDMA_PCM_H__ */
diff --git a/sound/soc/ti/j721e-evm.c b/sound/soc/ti/j721e-evm.c
new file mode 100644
index 000000000..6a969874c
--- /dev/null
+++ b/sound/soc/ti/j721e-evm.c
@@ -0,0 +1,935 @@
+// SPDX-License-Identifier: GPL-2.0
+/*
+ *  Copyright (C) 2020 Texas Instruments Incorporated - http://www.ti.com
+ *  Author: Peter Ujfalusi <peter.ujfalusi@ti.com>
+ */
+
+#include <linux/clk.h>
+#include <linux/module.h>
+#include <linux/of.h>
+#include <linux/platform_device.h>
+
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+
+#include "davinci-mcasp.h"
+
+/*
+ * Maximum number of configuration entries for prefixes:
+ * CPB: 2 (mcasp10 + codec)
+ * IVI: 3 (mcasp0 + 2x codec)
+ */
+#define J721E_CODEC_CONF_COUNT	5
+
+enum j721e_audio_domain_id {
+	J721E_AUDIO_DOMAIN_CPB = 0,
+	J721E_AUDIO_DOMAIN_IVI,
+	J721E_AUDIO_DOMAIN_LAST,
+};
+
+#define J721E_CLK_PARENT_48000	0
+#define J721E_CLK_PARENT_44100	1
+
+#define J721E_MAX_CLK_HSDIV	128
+#define PCM1368A_MAX_SYSCLK	36864000
+
+#define J721E_DAI_FMT		(SND_SOC_DAIFMT_RIGHT_J | \
+				 SND_SOC_DAIFMT_NB_NF |   \
+				 SND_SOC_DAIFMT_CBS_CFS)
+
+enum j721e_board_type {
+	J721E_BOARD_CPB = 1,
+	J721E_BOARD_CPB_IVI,
+};
+
+struct j721e_audio_match_data {
+	enum j721e_board_type board_type;
+	int num_links;
+	unsigned int pll_rates[2];
+};
+
+static unsigned int ratios_for_pcm3168a[] = {
+	256,
+	512,
+	768,
+};
+
+struct j721e_audio_clocks {
+	struct clk *target;
+	struct clk *parent[2];
+};
+
+struct j721e_audio_domain {
+	struct j721e_audio_clocks codec;
+	struct j721e_audio_clocks mcasp;
+	int parent_clk_id;
+
+	int active;
+	unsigned int active_link;
+	unsigned int rate;
+};
+
+struct j721e_priv {
+	struct device *dev;
+	struct snd_soc_card card;
+	struct snd_soc_dai_link *dai_links;
+	struct snd_soc_codec_conf codec_conf[J721E_CODEC_CONF_COUNT];
+	struct snd_interval rate_range;
+	const struct j721e_audio_match_data *match_data;
+	u32 pll_rates[2];
+	unsigned int hsdiv_rates[2];
+
+	struct j721e_audio_domain audio_domains[J721E_AUDIO_DOMAIN_LAST];
+
+	struct mutex mutex;
+};
+
+static const struct snd_soc_dapm_widget j721e_cpb_dapm_widgets[] = {
+	SND_SOC_DAPM_HP("CPB Stereo HP 1", NULL),
+	SND_SOC_DAPM_HP("CPB Stereo HP 2", NULL),
+	SND_SOC_DAPM_HP("CPB Stereo HP 3", NULL),
+	SND_SOC_DAPM_LINE("CPB Line Out", NULL),
+	SND_SOC_DAPM_MIC("CPB Stereo Mic 1", NULL),
+	SND_SOC_DAPM_MIC("CPB Stereo Mic 2", NULL),
+	SND_SOC_DAPM_LINE("CPB Line In", NULL),
+};
+
+static const struct snd_soc_dapm_route j721e_cpb_dapm_routes[] = {
+	{"CPB Stereo HP 1", NULL, "codec-1 AOUT1L"},
+	{"CPB Stereo HP 1", NULL, "codec-1 AOUT1R"},
+	{"CPB Stereo HP 2", NULL, "codec-1 AOUT2L"},
+	{"CPB Stereo HP 2", NULL, "codec-1 AOUT2R"},
+	{"CPB Stereo HP 3", NULL, "codec-1 AOUT3L"},
+	{"CPB Stereo HP 3", NULL, "codec-1 AOUT3R"},
+	{"CPB Line Out", NULL, "codec-1 AOUT4L"},
+	{"CPB Line Out", NULL, "codec-1 AOUT4R"},
+
+	{"codec-1 AIN1L", NULL, "CPB Stereo Mic 1"},
+	{"codec-1 AIN1R", NULL, "CPB Stereo Mic 1"},
+	{"codec-1 AIN2L", NULL, "CPB Stereo Mic 2"},
+	{"codec-1 AIN2R", NULL, "CPB Stereo Mic 2"},
+	{"codec-1 AIN3L", NULL, "CPB Line In"},
+	{"codec-1 AIN3R", NULL, "CPB Line In"},
+};
+
+static const struct snd_soc_dapm_widget j721e_ivi_codec_a_dapm_widgets[] = {
+	SND_SOC_DAPM_LINE("IVI A Line Out 1", NULL),
+	SND_SOC_DAPM_LINE("IVI A Line Out 2", NULL),
+	SND_SOC_DAPM_LINE("IVI A Line Out 3", NULL),
+	SND_SOC_DAPM_LINE("IVI A Line Out 4", NULL),
+	SND_SOC_DAPM_MIC("IVI A Stereo Mic 1", NULL),
+	SND_SOC_DAPM_MIC("IVI A Stereo Mic 2", NULL),
+	SND_SOC_DAPM_LINE("IVI A Line In", NULL),
+};
+
+static const struct snd_soc_dapm_route j721e_codec_a_dapm_routes[] = {
+	{"IVI A Line Out 1", NULL, "codec-a AOUT1L"},
+	{"IVI A Line Out 1", NULL, "codec-a AOUT1R"},
+	{"IVI A Line Out 2", NULL, "codec-a AOUT2L"},
+	{"IVI A Line Out 2", NULL, "codec-a AOUT2R"},
+	{"IVI A Line Out 3", NULL, "codec-a AOUT3L"},
+	{"IVI A Line Out 3", NULL, "codec-a AOUT3R"},
+	{"IVI A Line Out 4", NULL, "codec-a AOUT4L"},
+	{"IVI A Line Out 4", NULL, "codec-a AOUT4R"},
+
+	{"codec-a AIN1L", NULL, "IVI A Stereo Mic 1"},
+	{"codec-a AIN1R", NULL, "IVI A Stereo Mic 1"},
+	{"codec-a AIN2L", NULL, "IVI A Stereo Mic 2"},
+	{"codec-a AIN2R", NULL, "IVI A Stereo Mic 2"},
+	{"codec-a AIN3L", NULL, "IVI A Line In"},
+	{"codec-a AIN3R", NULL, "IVI A Line In"},
+};
+
+static const struct snd_soc_dapm_widget j721e_ivi_codec_b_dapm_widgets[] = {
+	SND_SOC_DAPM_LINE("IVI B Line Out 1", NULL),
+	SND_SOC_DAPM_LINE("IVI B Line Out 2", NULL),
+	SND_SOC_DAPM_LINE("IVI B Line Out 3", NULL),
+	SND_SOC_DAPM_LINE("IVI B Line Out 4", NULL),
+	SND_SOC_DAPM_MIC("IVI B Stereo Mic 1", NULL),
+	SND_SOC_DAPM_MIC("IVI B Stereo Mic 2", NULL),
+	SND_SOC_DAPM_LINE("IVI B Line In", NULL),
+};
+
+static const struct snd_soc_dapm_route j721e_codec_b_dapm_routes[] = {
+	{"IVI B Line Out 1", NULL, "codec-b AOUT1L"},
+	{"IVI B Line Out 1", NULL, "codec-b AOUT1R"},
+	{"IVI B Line Out 2", NULL, "codec-b AOUT2L"},
+	{"IVI B Line Out 2", NULL, "codec-b AOUT2R"},
+	{"IVI B Line Out 3", NULL, "codec-b AOUT3L"},
+	{"IVI B Line Out 3", NULL, "codec-b AOUT3R"},
+	{"IVI B Line Out 4", NULL, "codec-b AOUT4L"},
+	{"IVI B Line Out 4", NULL, "codec-b AOUT4R"},
+
+	{"codec-b AIN1L", NULL, "IVI B Stereo Mic 1"},
+	{"codec-b AIN1R", NULL, "IVI B Stereo Mic 1"},
+	{"codec-b AIN2L", NULL, "IVI B Stereo Mic 2"},
+	{"codec-b AIN2R", NULL, "IVI B Stereo Mic 2"},
+	{"codec-b AIN3L", NULL, "IVI B Line In"},
+	{"codec-b AIN3R", NULL, "IVI B Line In"},
+};
+
+static int j721e_configure_refclk(struct j721e_priv *priv,
+				  unsigned int audio_domain, unsigned int rate)
+{
+	struct j721e_audio_domain *domain = &priv->audio_domains[audio_domain];
+	unsigned int scki;
+	int ret = -EINVAL;
+	int i, clk_id;
+
+	if (!(rate % 8000) && priv->pll_rates[J721E_CLK_PARENT_48000])
+		clk_id = J721E_CLK_PARENT_48000;
+	else if (!(rate % 11025) && priv->pll_rates[J721E_CLK_PARENT_44100])
+		clk_id = J721E_CLK_PARENT_44100;
+	else
+		return ret;
+
+	for (i = 0; i < ARRAY_SIZE(ratios_for_pcm3168a); i++) {
+		scki = ratios_for_pcm3168a[i] * rate;
+
+		if (priv->pll_rates[clk_id] / scki <= J721E_MAX_CLK_HSDIV) {
+			ret = 0;
+			break;
+		}
+	}
+
+	if (ret) {
+		dev_err(priv->dev, "No valid clock configuration for %u Hz\n",
+			rate);
+		return ret;
+	}
+
+	if (domain->parent_clk_id == -1 || priv->hsdiv_rates[domain->parent_clk_id] != scki) {
+		dev_dbg(priv->dev,
+			"domain%u configuration for %u Hz: %s, %dxFS (SCKI: %u Hz)\n",
+			audio_domain, rate,
+			clk_id == J721E_CLK_PARENT_48000 ? "PLL4" : "PLL15",
+			ratios_for_pcm3168a[i], scki);
+
+		if (domain->parent_clk_id != clk_id) {
+			ret = clk_set_parent(domain->codec.target,
+					     domain->codec.parent[clk_id]);
+			if (ret)
+				return ret;
+
+			ret = clk_set_parent(domain->mcasp.target,
+					     domain->mcasp.parent[clk_id]);
+			if (ret)
+				return ret;
+
+			domain->parent_clk_id = clk_id;
+		}
+
+		ret = clk_set_rate(domain->codec.target, scki);
+		if (ret) {
+			dev_err(priv->dev, "codec set rate failed for %u Hz\n",
+				scki);
+			return ret;
+		}
+
+		ret = clk_set_rate(domain->mcasp.target, scki);
+		if (!ret) {
+			priv->hsdiv_rates[domain->parent_clk_id] = scki;
+		} else {
+			dev_err(priv->dev, "mcasp set rate failed for %u Hz\n",
+				scki);
+			return ret;
+		}
+	}
+
+	return ret;
+}
+
+static int j721e_rule_rate(struct snd_pcm_hw_params *params,
+			   struct snd_pcm_hw_rule *rule)
+{
+	struct snd_interval *t = rule->private;
+
+	return snd_interval_refine(hw_param_interval(params, rule->var), t);
+}
+
+static int j721e_audio_startup(struct snd_pcm_substream *substream)
+{
+	struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream);
+	struct j721e_priv *priv = snd_soc_card_get_drvdata(rtd->card);
+	unsigned int domain_id = rtd->dai_link->id;
+	struct j721e_audio_domain *domain = &priv->audio_domains[domain_id];
+	struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0);
+	struct snd_soc_dai *codec_dai;
+	unsigned int active_rate;
+	int ret = 0;
+	int i;
+
+	mutex_lock(&priv->mutex);
+
+	domain->active++;
+
+	for (i = 0; i < J721E_AUDIO_DOMAIN_LAST; i++) {
+		active_rate = priv->audio_domains[i].rate;
+		if (active_rate)
+			break;
+	}
+
+	if (active_rate)
+		ret = snd_pcm_hw_constraint_single(substream->runtime,
+						   SNDRV_PCM_HW_PARAM_RATE,
+						   active_rate);
+	else
+		ret = snd_pcm_hw_rule_add(substream->runtime, 0,
+					  SNDRV_PCM_HW_PARAM_RATE,
+					  j721e_rule_rate, &priv->rate_range,
+					  SNDRV_PCM_HW_PARAM_RATE, -1);
+
+
+	if (ret)
+		goto out;
+
+	/* Reset TDM slots to 32 */
+	ret = snd_soc_dai_set_tdm_slot(cpu_dai, 0x3, 0x3, 2, 32);
+	if (ret && ret != -ENOTSUPP)
+		goto out;
+
+	for_each_rtd_codec_dais(rtd, i, codec_dai) {
+		ret = snd_soc_dai_set_tdm_slot(codec_dai, 0x3, 0x3, 2, 32);
+		if (ret && ret != -ENOTSUPP)
+			goto out;
+	}
+
+	if (ret == -ENOTSUPP)
+		ret = 0;
+out:
+	if (ret)
+		domain->active--;
+	mutex_unlock(&priv->mutex);
+
+	return ret;
+}
+
+static int j721e_audio_hw_params(struct snd_pcm_substream *substream,
+				 struct snd_pcm_hw_params *params)
+{
+	struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream);
+	struct snd_soc_card *card = rtd->card;
+	struct j721e_priv *priv = snd_soc_card_get_drvdata(card);
+	unsigned int domain_id = rtd->dai_link->id;
+	struct j721e_audio_domain *domain = &priv->audio_domains[domain_id];
+	struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0);
+	struct snd_soc_dai *codec_dai;
+	unsigned int sysclk_rate;
+	int slot_width = 32;
+	int ret;
+	int i;
+
+	mutex_lock(&priv->mutex);
+
+	if (domain->rate && domain->rate != params_rate(params)) {
+		ret = -EINVAL;
+		goto out;
+	}
+
+	if (params_width(params) == 16)
+		slot_width = 16;
+
+	ret = snd_soc_dai_set_tdm_slot(cpu_dai, 0x3, 0x3, 2, slot_width);
+	if (ret && ret != -ENOTSUPP)
+		goto out;
+
+	for_each_rtd_codec_dais(rtd, i, codec_dai) {
+		ret = snd_soc_dai_set_tdm_slot(codec_dai, 0x3, 0x3, 2,
+					       slot_width);
+		if (ret && ret != -ENOTSUPP)
+			goto out;
+	}
+
+	ret = j721e_configure_refclk(priv, domain_id, params_rate(params));
+	if (ret)
+		goto out;
+
+	sysclk_rate = priv->hsdiv_rates[domain->parent_clk_id];
+	for_each_rtd_codec_dais(rtd, i, codec_dai) {
+		ret = snd_soc_dai_set_sysclk(codec_dai, 0, sysclk_rate,
+					     SND_SOC_CLOCK_IN);
+		if (ret && ret != -ENOTSUPP) {
+			dev_err(priv->dev,
+				"codec set_sysclk failed for %u Hz\n",
+				sysclk_rate);
+			goto out;
+		}
+	}
+
+	ret = snd_soc_dai_set_sysclk(cpu_dai, MCASP_CLK_HCLK_AUXCLK,
+				     sysclk_rate, SND_SOC_CLOCK_IN);
+
+	if (ret && ret != -ENOTSUPP) {
+		dev_err(priv->dev, "mcasp set_sysclk failed for %u Hz\n",
+			sysclk_rate);
+	} else {
+		domain->rate = params_rate(params);
+		ret = 0;
+	}
+
+out:
+	mutex_unlock(&priv->mutex);
+	return ret;
+}
+
+static void j721e_audio_shutdown(struct snd_pcm_substream *substream)
+{
+	struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream);
+	struct j721e_priv *priv = snd_soc_card_get_drvdata(rtd->card);
+	unsigned int domain_id = rtd->dai_link->id;
+	struct j721e_audio_domain *domain = &priv->audio_domains[domain_id];
+
+	mutex_lock(&priv->mutex);
+
+	domain->active--;
+	if (!domain->active) {
+		domain->rate = 0;
+		domain->active_link = 0;
+	}
+
+	mutex_unlock(&priv->mutex);
+}
+
+static const struct snd_soc_ops j721e_audio_ops = {
+	.startup = j721e_audio_startup,
+	.hw_params = j721e_audio_hw_params,
+	.shutdown = j721e_audio_shutdown,
+};
+
+static int j721e_audio_init(struct snd_soc_pcm_runtime *rtd)
+{
+	struct j721e_priv *priv = snd_soc_card_get_drvdata(rtd->card);
+	unsigned int domain_id = rtd->dai_link->id;
+	struct j721e_audio_domain *domain = &priv->audio_domains[domain_id];
+	struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0);
+	struct snd_soc_dai *codec_dai;
+	unsigned int sysclk_rate;
+	int i, ret;
+
+	/* Set up initial clock configuration */
+	ret = j721e_configure_refclk(priv, domain_id, 48000);
+	if (ret)
+		return ret;
+
+	sysclk_rate = priv->hsdiv_rates[domain->parent_clk_id];
+	for_each_rtd_codec_dais(rtd, i, codec_dai) {
+		ret = snd_soc_dai_set_sysclk(codec_dai, 0, sysclk_rate,
+					     SND_SOC_CLOCK_IN);
+		if (ret && ret != -ENOTSUPP)
+			return ret;
+	}
+
+	ret = snd_soc_dai_set_sysclk(cpu_dai, MCASP_CLK_HCLK_AUXCLK,
+				     sysclk_rate, SND_SOC_CLOCK_IN);
+	if (ret && ret != -ENOTSUPP)
+		return ret;
+
+	/* Set initial tdm slots */
+	ret = snd_soc_dai_set_tdm_slot(cpu_dai, 0x3, 0x3, 2, 32);
+	if (ret && ret != -ENOTSUPP)
+		return ret;
+
+	for_each_rtd_codec_dais(rtd, i, codec_dai) {
+		ret = snd_soc_dai_set_tdm_slot(codec_dai, 0x3, 0x3, 2, 32);
+		if (ret && ret != -ENOTSUPP)
+			return ret;
+	}
+
+	return 0;
+}
+
+static int j721e_audio_init_ivi(struct snd_soc_pcm_runtime *rtd)
+{
+	struct snd_soc_dapm_context *dapm = &rtd->card->dapm;
+
+	snd_soc_dapm_new_controls(dapm, j721e_ivi_codec_a_dapm_widgets,
+				  ARRAY_SIZE(j721e_ivi_codec_a_dapm_widgets));
+	snd_soc_dapm_add_routes(dapm, j721e_codec_a_dapm_routes,
+				ARRAY_SIZE(j721e_codec_a_dapm_routes));
+	snd_soc_dapm_new_controls(dapm, j721e_ivi_codec_b_dapm_widgets,
+				  ARRAY_SIZE(j721e_ivi_codec_b_dapm_widgets));
+	snd_soc_dapm_add_routes(dapm, j721e_codec_b_dapm_routes,
+				ARRAY_SIZE(j721e_codec_b_dapm_routes));
+
+	return j721e_audio_init(rtd);
+}
+
+static int j721e_get_clocks(struct device *dev,
+			    struct j721e_audio_clocks *clocks, char *prefix)
+{
+	struct clk *parent;
+	char *clk_name;
+	int ret;
+
+	clocks->target = devm_clk_get(dev, prefix);
+	if (IS_ERR(clocks->target))
+		return dev_err_probe(dev, PTR_ERR(clocks->target),
+				     "failed to acquire %s\n", prefix);
+
+	clk_name = kasprintf(GFP_KERNEL, "%s-48000", prefix);
+	if (clk_name) {
+		parent = devm_clk_get(dev, clk_name);
+		kfree(clk_name);
+		if (IS_ERR(parent)) {
+			ret = PTR_ERR(parent);
+			if (ret == -EPROBE_DEFER)
+				return ret;
+
+			dev_dbg(dev, "no 48KHz parent for %s: %d\n", prefix, ret);
+			parent = NULL;
+		}
+		clocks->parent[J721E_CLK_PARENT_48000] = parent;
+	} else {
+		return -ENOMEM;
+	}
+
+	clk_name = kasprintf(GFP_KERNEL, "%s-44100", prefix);
+	if (clk_name) {
+		parent = devm_clk_get(dev, clk_name);
+		kfree(clk_name);
+		if (IS_ERR(parent)) {
+			ret = PTR_ERR(parent);
+			if (ret == -EPROBE_DEFER)
+				return ret;
+
+			dev_dbg(dev, "no 44.1KHz parent for %s: %d\n", prefix, ret);
+			parent = NULL;
+		}
+		clocks->parent[J721E_CLK_PARENT_44100] = parent;
+	} else {
+		return -ENOMEM;
+	}
+
+	if (!clocks->parent[J721E_CLK_PARENT_44100] &&
+	    !clocks->parent[J721E_CLK_PARENT_48000]) {
+		dev_err(dev, "At least one parent clock is needed for %s\n",
+			prefix);
+		return -EINVAL;
+	}
+
+	return 0;
+}
+
+static const struct j721e_audio_match_data j721e_cpb_data = {
+	.board_type = J721E_BOARD_CPB,
+	.num_links = 2, /* CPB pcm3168a */
+	.pll_rates = {
+		[J721E_CLK_PARENT_44100] = 1083801600, /* PLL15 */
+		[J721E_CLK_PARENT_48000] = 1179648000, /* PLL4 */
+	},
+};
+
+static const struct j721e_audio_match_data j721e_cpb_ivi_data = {
+	.board_type = J721E_BOARD_CPB_IVI,
+	.num_links = 4, /* CPB pcm3168a + 2x pcm3168a on IVI */
+	.pll_rates = {
+		[J721E_CLK_PARENT_44100] = 1083801600, /* PLL15 */
+		[J721E_CLK_PARENT_48000] = 1179648000, /* PLL4 */
+	},
+};
+
+static const struct j721e_audio_match_data j7200_cpb_data = {
+	.board_type = J721E_BOARD_CPB,
+	.num_links = 2, /* CPB pcm3168a */
+	.pll_rates = {
+		[J721E_CLK_PARENT_48000] = 2359296000u, /* PLL4 */
+	},
+};
+
+static const struct of_device_id j721e_audio_of_match[] = {
+	{
+		.compatible = "ti,j721e-cpb-audio",
+		.data = &j721e_cpb_data,
+	}, {
+		.compatible = "ti,j721e-cpb-ivi-audio",
+		.data = &j721e_cpb_ivi_data,
+	}, {
+		.compatible = "ti,j7200-cpb-audio",
+		.data = &j7200_cpb_data,
+	},
+	{ },
+};
+MODULE_DEVICE_TABLE(of, j721e_audio_of_match);
+
+static int j721e_calculate_rate_range(struct j721e_priv *priv)
+{
+	const struct j721e_audio_match_data *match_data = priv->match_data;
+	struct j721e_audio_clocks *domain_clocks;
+	unsigned int min_rate, max_rate, pll_rate;
+	struct clk *pll;
+
+	domain_clocks = &priv->audio_domains[J721E_AUDIO_DOMAIN_CPB].mcasp;
+
+	pll = clk_get_parent(domain_clocks->parent[J721E_CLK_PARENT_44100]);
+	if (IS_ERR_OR_NULL(pll)) {
+		priv->pll_rates[J721E_CLK_PARENT_44100] =
+				match_data->pll_rates[J721E_CLK_PARENT_44100];
+	} else {
+		priv->pll_rates[J721E_CLK_PARENT_44100] = clk_get_rate(pll);
+		clk_put(pll);
+	}
+
+	pll = clk_get_parent(domain_clocks->parent[J721E_CLK_PARENT_48000]);
+	if (IS_ERR_OR_NULL(pll)) {
+		priv->pll_rates[J721E_CLK_PARENT_48000] =
+				match_data->pll_rates[J721E_CLK_PARENT_48000];
+	} else {
+		priv->pll_rates[J721E_CLK_PARENT_48000] = clk_get_rate(pll);
+		clk_put(pll);
+	}
+
+	if (!priv->pll_rates[J721E_CLK_PARENT_44100] &&
+	    !priv->pll_rates[J721E_CLK_PARENT_48000]) {
+		dev_err(priv->dev, "At least one PLL is needed\n");
+		return -EINVAL;
+	}
+
+	if (priv->pll_rates[J721E_CLK_PARENT_44100])
+		pll_rate = priv->pll_rates[J721E_CLK_PARENT_44100];
+	else
+		pll_rate = priv->pll_rates[J721E_CLK_PARENT_48000];
+
+	min_rate = pll_rate / J721E_MAX_CLK_HSDIV;
+	min_rate /= ratios_for_pcm3168a[ARRAY_SIZE(ratios_for_pcm3168a) - 1];
+
+	if (priv->pll_rates[J721E_CLK_PARENT_48000])
+		pll_rate = priv->pll_rates[J721E_CLK_PARENT_48000];
+	else
+		pll_rate = priv->pll_rates[J721E_CLK_PARENT_44100];
+
+	if (pll_rate > PCM1368A_MAX_SYSCLK)
+		pll_rate = PCM1368A_MAX_SYSCLK;
+
+	max_rate = pll_rate / ratios_for_pcm3168a[0];
+
+	snd_interval_any(&priv->rate_range);
+	priv->rate_range.min = min_rate;
+	priv->rate_range.max = max_rate;
+
+	return 0;
+}
+
+static int j721e_soc_probe_cpb(struct j721e_priv *priv, int *link_idx,
+			       int *conf_idx)
+{
+	struct device_node *node = priv->dev->of_node;
+	struct snd_soc_dai_link_component *compnent;
+	struct device_node *dai_node, *codec_node;
+	struct j721e_audio_domain *domain;
+	int comp_count, comp_idx;
+	int ret;
+
+	dai_node = of_parse_phandle(node, "ti,cpb-mcasp", 0);
+	if (!dai_node) {
+		dev_err(priv->dev, "CPB McASP node is not provided\n");
+		return -EINVAL;
+	}
+
+	codec_node = of_parse_phandle(node, "ti,cpb-codec", 0);
+	if (!codec_node) {
+		dev_err(priv->dev, "CPB codec node is not provided\n");
+		ret = -EINVAL;
+		goto put_dai_node;
+	}
+
+	domain = &priv->audio_domains[J721E_AUDIO_DOMAIN_CPB];
+	ret = j721e_get_clocks(priv->dev, &domain->codec, "cpb-codec-scki");
+	if (ret)
+		goto put_codec_node;
+
+	ret = j721e_get_clocks(priv->dev, &domain->mcasp, "cpb-mcasp-auxclk");
+	if (ret)
+		goto put_codec_node;
+
+	/*
+	 * Common Processor Board, two links
+	 * Link 1: McASP10 -> pcm3168a_1 DAC
+	 * Link 2: McASP10 <- pcm3168a_1 ADC
+	 */
+	comp_count = 6;
+	compnent = devm_kzalloc(priv->dev, comp_count * sizeof(*compnent),
+				GFP_KERNEL);
+	if (!compnent) {
+		ret = -ENOMEM;
+		goto put_codec_node;
+	}
+
+	comp_idx = 0;
+	priv->dai_links[*link_idx].cpus = &compnent[comp_idx++];
+	priv->dai_links[*link_idx].num_cpus = 1;
+	priv->dai_links[*link_idx].codecs = &compnent[comp_idx++];
+	priv->dai_links[*link_idx].num_codecs = 1;
+	priv->dai_links[*link_idx].platforms = &compnent[comp_idx++];
+	priv->dai_links[*link_idx].num_platforms = 1;
+
+	priv->dai_links[*link_idx].name = "CPB PCM3168A Playback";
+	priv->dai_links[*link_idx].stream_name = "CPB PCM3168A Analog";
+	priv->dai_links[*link_idx].cpus->of_node = dai_node;
+	priv->dai_links[*link_idx].platforms->of_node = dai_node;
+	priv->dai_links[*link_idx].codecs->of_node = codec_node;
+	priv->dai_links[*link_idx].codecs->dai_name = "pcm3168a-dac";
+	priv->dai_links[*link_idx].playback_only = 1;
+	priv->dai_links[*link_idx].id = J721E_AUDIO_DOMAIN_CPB;
+	priv->dai_links[*link_idx].dai_fmt = J721E_DAI_FMT;
+	priv->dai_links[*link_idx].init = j721e_audio_init;
+	priv->dai_links[*link_idx].ops = &j721e_audio_ops;
+	(*link_idx)++;
+
+	priv->dai_links[*link_idx].cpus = &compnent[comp_idx++];
+	priv->dai_links[*link_idx].num_cpus = 1;
+	priv->dai_links[*link_idx].codecs = &compnent[comp_idx++];
+	priv->dai_links[*link_idx].num_codecs = 1;
+	priv->dai_links[*link_idx].platforms = &compnent[comp_idx++];
+	priv->dai_links[*link_idx].num_platforms = 1;
+
+	priv->dai_links[*link_idx].name = "CPB PCM3168A Capture";
+	priv->dai_links[*link_idx].stream_name = "CPB PCM3168A Analog";
+	priv->dai_links[*link_idx].cpus->of_node = dai_node;
+	priv->dai_links[*link_idx].platforms->of_node = dai_node;
+	priv->dai_links[*link_idx].codecs->of_node = codec_node;
+	priv->dai_links[*link_idx].codecs->dai_name = "pcm3168a-adc";
+	priv->dai_links[*link_idx].capture_only = 1;
+	priv->dai_links[*link_idx].id = J721E_AUDIO_DOMAIN_CPB;
+	priv->dai_links[*link_idx].dai_fmt = J721E_DAI_FMT;
+	priv->dai_links[*link_idx].init = j721e_audio_init;
+	priv->dai_links[*link_idx].ops = &j721e_audio_ops;
+	(*link_idx)++;
+
+	priv->codec_conf[*conf_idx].dlc.of_node = codec_node;
+	priv->codec_conf[*conf_idx].name_prefix = "codec-1";
+	(*conf_idx)++;
+	priv->codec_conf[*conf_idx].dlc.of_node = dai_node;
+	priv->codec_conf[*conf_idx].name_prefix = "McASP10";
+	(*conf_idx)++;
+
+	return 0;
+
+put_codec_node:
+	of_node_put(codec_node);
+put_dai_node:
+	of_node_put(dai_node);
+	return ret;
+}
+
+static int j721e_soc_probe_ivi(struct j721e_priv *priv, int *link_idx,
+			       int *conf_idx)
+{
+	struct device_node *node = priv->dev->of_node;
+	struct snd_soc_dai_link_component *compnent;
+	struct device_node *dai_node, *codeca_node, *codecb_node;
+	struct j721e_audio_domain *domain;
+	int comp_count, comp_idx;
+	int ret;
+
+	if (priv->match_data->board_type != J721E_BOARD_CPB_IVI)
+		return 0;
+
+	dai_node = of_parse_phandle(node, "ti,ivi-mcasp", 0);
+	if (!dai_node) {
+		dev_err(priv->dev, "IVI McASP node is not provided\n");
+		return -EINVAL;
+	}
+
+	codeca_node = of_parse_phandle(node, "ti,ivi-codec-a", 0);
+	if (!codeca_node) {
+		dev_err(priv->dev, "IVI codec-a node is not provided\n");
+		ret = -EINVAL;
+		goto put_dai_node;
+	}
+
+	codecb_node = of_parse_phandle(node, "ti,ivi-codec-b", 0);
+	if (!codecb_node) {
+		dev_warn(priv->dev, "IVI codec-b node is not provided\n");
+		ret = 0;
+		goto put_codeca_node;
+	}
+
+	domain = &priv->audio_domains[J721E_AUDIO_DOMAIN_IVI];
+	ret = j721e_get_clocks(priv->dev, &domain->codec, "ivi-codec-scki");
+	if (ret)
+		goto put_codecb_node;
+
+	ret = j721e_get_clocks(priv->dev, &domain->mcasp, "ivi-mcasp-auxclk");
+	if (ret)
+		goto put_codecb_node;
+
+	/*
+	 * IVI extension, two links
+	 * Link 1: McASP0 -> pcm3168a_a DAC
+	 *		  \> pcm3168a_b DAC
+	 * Link 2: McASP0 <- pcm3168a_a ADC
+	 *		   \ pcm3168a_b ADC
+	 */
+	comp_count = 8;
+	compnent = devm_kzalloc(priv->dev, comp_count * sizeof(*compnent),
+				GFP_KERNEL);
+	if (!compnent) {
+		ret = -ENOMEM;
+		goto put_codecb_node;
+	}
+
+	comp_idx = 0;
+	priv->dai_links[*link_idx].cpus = &compnent[comp_idx++];
+	priv->dai_links[*link_idx].num_cpus = 1;
+	priv->dai_links[*link_idx].platforms = &compnent[comp_idx++];
+	priv->dai_links[*link_idx].num_platforms = 1;
+	priv->dai_links[*link_idx].codecs = &compnent[comp_idx];
+	priv->dai_links[*link_idx].num_codecs = 2;
+	comp_idx += 2;
+
+	priv->dai_links[*link_idx].name = "IVI 2xPCM3168A Playback";
+	priv->dai_links[*link_idx].stream_name = "IVI 2xPCM3168A Analog";
+	priv->dai_links[*link_idx].cpus->of_node = dai_node;
+	priv->dai_links[*link_idx].platforms->of_node = dai_node;
+	priv->dai_links[*link_idx].codecs[0].of_node = codeca_node;
+	priv->dai_links[*link_idx].codecs[0].dai_name = "pcm3168a-dac";
+	priv->dai_links[*link_idx].codecs[1].of_node = codecb_node;
+	priv->dai_links[*link_idx].codecs[1].dai_name = "pcm3168a-dac";
+	priv->dai_links[*link_idx].playback_only = 1;
+	priv->dai_links[*link_idx].id = J721E_AUDIO_DOMAIN_IVI;
+	priv->dai_links[*link_idx].dai_fmt = J721E_DAI_FMT;
+	priv->dai_links[*link_idx].init = j721e_audio_init_ivi;
+	priv->dai_links[*link_idx].ops = &j721e_audio_ops;
+	(*link_idx)++;
+
+	priv->dai_links[*link_idx].cpus = &compnent[comp_idx++];
+	priv->dai_links[*link_idx].num_cpus = 1;
+	priv->dai_links[*link_idx].platforms = &compnent[comp_idx++];
+	priv->dai_links[*link_idx].num_platforms = 1;
+	priv->dai_links[*link_idx].codecs = &compnent[comp_idx];
+	priv->dai_links[*link_idx].num_codecs = 2;
+
+	priv->dai_links[*link_idx].name = "IVI 2xPCM3168A Capture";
+	priv->dai_links[*link_idx].stream_name = "IVI 2xPCM3168A Analog";
+	priv->dai_links[*link_idx].cpus->of_node = dai_node;
+	priv->dai_links[*link_idx].platforms->of_node = dai_node;
+	priv->dai_links[*link_idx].codecs[0].of_node = codeca_node;
+	priv->dai_links[*link_idx].codecs[0].dai_name = "pcm3168a-adc";
+	priv->dai_links[*link_idx].codecs[1].of_node = codecb_node;
+	priv->dai_links[*link_idx].codecs[1].dai_name = "pcm3168a-adc";
+	priv->dai_links[*link_idx].capture_only = 1;
+	priv->dai_links[*link_idx].id = J721E_AUDIO_DOMAIN_IVI;
+	priv->dai_links[*link_idx].dai_fmt = J721E_DAI_FMT;
+	priv->dai_links[*link_idx].init = j721e_audio_init;
+	priv->dai_links[*link_idx].ops = &j721e_audio_ops;
+	(*link_idx)++;
+
+	priv->codec_conf[*conf_idx].dlc.of_node = codeca_node;
+	priv->codec_conf[*conf_idx].name_prefix = "codec-a";
+	(*conf_idx)++;
+
+	priv->codec_conf[*conf_idx].dlc.of_node = codecb_node;
+	priv->codec_conf[*conf_idx].name_prefix = "codec-b";
+	(*conf_idx)++;
+
+	priv->codec_conf[*conf_idx].dlc.of_node = dai_node;
+	priv->codec_conf[*conf_idx].name_prefix = "McASP0";
+	(*conf_idx)++;
+
+	return 0;
+
+
+put_codecb_node:
+	of_node_put(codecb_node);
+put_codeca_node:
+	of_node_put(codeca_node);
+put_dai_node:
+	of_node_put(dai_node);
+	return ret;
+}
+
+static int j721e_soc_probe(struct platform_device *pdev)
+{
+	struct device_node *node = pdev->dev.of_node;
+	struct snd_soc_card *card;
+	const struct of_device_id *match;
+	struct j721e_priv *priv;
+	int link_cnt, conf_cnt, ret, i;
+
+	if (!node) {
+		dev_err(&pdev->dev, "of node is missing.\n");
+		return -ENODEV;
+	}
+
+	match = of_match_node(j721e_audio_of_match, node);
+	if (!match) {
+		dev_err(&pdev->dev, "No compatible match found\n");
+		return -ENODEV;
+	}
+
+	priv = devm_kzalloc(&pdev->dev, sizeof(*priv), GFP_KERNEL);
+	if (!priv)
+		return -ENOMEM;
+
+	priv->match_data = match->data;
+
+	priv->dai_links = devm_kcalloc(&pdev->dev, priv->match_data->num_links,
+				       sizeof(*priv->dai_links), GFP_KERNEL);
+	if (!priv->dai_links)
+		return -ENOMEM;
+
+	for (i = 0; i < J721E_AUDIO_DOMAIN_LAST; i++)
+		priv->audio_domains[i].parent_clk_id = -1;
+
+	priv->dev = &pdev->dev;
+	card = &priv->card;
+	card->dev = &pdev->dev;
+	card->owner = THIS_MODULE;
+	card->dapm_widgets = j721e_cpb_dapm_widgets;
+	card->num_dapm_widgets = ARRAY_SIZE(j721e_cpb_dapm_widgets);
+	card->dapm_routes = j721e_cpb_dapm_routes;
+	card->num_dapm_routes = ARRAY_SIZE(j721e_cpb_dapm_routes);
+	card->fully_routed = 1;
+
+	if (snd_soc_of_parse_card_name(card, "model")) {
+		dev_err(&pdev->dev, "Card name is not provided\n");
+		return -ENODEV;
+	}
+
+	link_cnt = 0;
+	conf_cnt = 0;
+	ret = j721e_soc_probe_cpb(priv, &link_cnt, &conf_cnt);
+	if (ret)
+		return ret;
+
+	ret = j721e_soc_probe_ivi(priv, &link_cnt, &conf_cnt);
+	if (ret)
+		return ret;
+
+	card->dai_link = priv->dai_links;
+	card->num_links = link_cnt;
+
+	card->codec_conf = priv->codec_conf;
+	card->num_configs = conf_cnt;
+
+	ret = j721e_calculate_rate_range(priv);
+	if (ret)
+		return ret;
+
+	snd_soc_card_set_drvdata(card, priv);
+
+	mutex_init(&priv->mutex);
+	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;
+}
+
+static struct platform_driver j721e_soc_driver = {
+	.driver = {
+		.name = "j721e-audio",
+		.pm = &snd_soc_pm_ops,
+		.of_match_table = j721e_audio_of_match,
+	},
+	.probe = j721e_soc_probe,
+};
+
+module_platform_driver(j721e_soc_driver);
+
+MODULE_AUTHOR("Peter Ujfalusi <peter.ujfalusi@ti.com>");
+MODULE_DESCRIPTION("ASoC machine driver for j721e Common Processor Board");
+MODULE_LICENSE("GPL v2");
diff --git a/sound/soc/ti/n810.c b/sound/soc/ti/n810.c
new file mode 100644
index 000000000..ed217b34f
--- /dev/null
+++ b/sound/soc/ti/n810.c
@@ -0,0 +1,368 @@
+// SPDX-License-Identifier: GPL-2.0-only
+/*
+ * n810.c  --  SoC audio for Nokia N810
+ *
+ * Copyright (C) 2008 Nokia Corporation
+ *
+ * Contact: Jarkko Nikula <jarkko.nikula@bitmer.com>
+ */
+
+#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;
+		fallthrough;
+	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, "HS Mic");
+	else
+		snd_soc_dapm_disable_pin_unlocked(dapm, "HS Mic");
+
+	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 = asoc_substream_to_rtd(substream);
+
+	snd_pcm_hw_constraint_single(runtime, SNDRV_PCM_HW_PARAM_CHANNELS, 2);
+
+	n810_ext_control(&rtd->card->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 = asoc_substream_to_rtd(substream);
+	struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0);
+	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 const 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.enumerated.item[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.enumerated.item[0])
+		return 0;
+
+	n810_spk_func = ucontrol->value.enumerated.item[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.enumerated.item[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.enumerated.item[0])
+		return 0;
+
+	n810_jack_func = ucontrol->value.enumerated.item[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.enumerated.item[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.enumerated.item[0])
+		return 0;
+
+	n810_dmic_func = ucontrol->value.enumerated.item[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),
+	SND_SOC_DAPM_MIC("HS Mic", 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, "DMic"},
+	{"DMic", NULL, "Mic Bias"},
+
+	/*
+	 * Note that the mic bias is coming from Retu/Vilma and we don't have
+	 * control over it atm. The analog HS mic is not working. <- TODO
+	 */
+	{"LINE1L", NULL, "HS Mic"},
+};
+
+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),
+};
+
+/* Digital audio interface glue - connects codec <--> CPU */
+SND_SOC_DAILINK_DEFS(aic33,
+	DAILINK_COMP_ARRAY(COMP_CPU("48076000.mcbsp")),
+	DAILINK_COMP_ARRAY(COMP_CODEC("tlv320aic3x-codec.1-0018",
+				      "tlv320aic3x-hifi")),
+	DAILINK_COMP_ARRAY(COMP_PLATFORM("48076000.mcbsp")));
+
+static struct snd_soc_dai_link n810_dai = {
+	.name = "TLV320AIC33",
+	.stream_name = "AIC33",
+	.dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF |
+		   SND_SOC_DAIFMT_CBM_CFM,
+	.ops = &n810_ops,
+	SND_SOC_DAILINK_REG(aic33),
+};
+
+/* 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),
+	.fully_routed = true,
+};
+
+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");
diff --git a/sound/soc/ti/omap-abe-twl6040.c b/sound/soc/ti/omap-abe-twl6040.c
new file mode 100644
index 000000000..805ffbf89
--- /dev/null
+++ b/sound/soc/ti/omap-abe-twl6040.c
@@ -0,0 +1,354 @@
+// SPDX-License-Identifier: GPL-2.0-only
+/*
+ * omap-abe-twl6040.c  --  SoC audio for TI OMAP based boards with ABE and
+ *			   twl6040 codec
+ *
+ * Author: Misael Lopez Cruz <misael.lopez@ti.com>
+ */
+
+#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"
+
+SND_SOC_DAILINK_DEFS(link0,
+	DAILINK_COMP_ARRAY(COMP_EMPTY()),
+	DAILINK_COMP_ARRAY(COMP_CODEC("twl6040-codec",
+				      "twl6040-legacy")),
+	DAILINK_COMP_ARRAY(COMP_EMPTY()));
+
+SND_SOC_DAILINK_DEFS(link1,
+	DAILINK_COMP_ARRAY(COMP_EMPTY()),
+	DAILINK_COMP_ARRAY(COMP_CODEC("dmic-codec",
+				      "dmic-hifi")),
+	DAILINK_COMP_ARRAY(COMP_EMPTY()));
+
+struct abe_twl6040 {
+	struct snd_soc_card card;
+	struct snd_soc_dai_link dai_links[2];
+	int	jack_detection;	/* board can detect jack events */
+	int	mclk_freq;	/* MCLK frequency speed for twl6040 */
+};
+
+static 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 = asoc_substream_to_rtd(substream);
+	struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0);
+	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(codec_dai->component);
+	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 const 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 = asoc_substream_to_rtd(substream);
+	struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0);
+	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 const 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_component *component = asoc_rtd_to_codec(rtd, 0)->component;
+	struct snd_soc_card *card = rtd->card;
+	struct abe_twl6040 *priv = snd_soc_card_get_drvdata(card);
+	int hs_trim;
+	int ret;
+
+	/*
+	 * Configure McPDM offset cancellation based on the HSOTRIM value from
+	 * twl6040.
+	 */
+	hs_trim = twl6040_get_trim_value(component, 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_card_jack_new_pins(rtd->card, "Headset Jack",
+						 SND_JACK_HEADSET, &hs_jack,
+						 hs_jack_pins,
+						 ARRAY_SIZE(hs_jack_pins));
+		if (ret)
+			return ret;
+
+		twl6040_hs_jack_detect(component, &hs_jack, SND_JACK_HEADSET);
+	}
+
+	return 0;
+}
+
+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));
+}
+
+static int omap_abe_probe(struct platform_device *pdev)
+{
+	struct device_node *node = pdev->dev.of_node;
+	struct snd_soc_card *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;
+	}
+
+	priv = devm_kzalloc(&pdev->dev, sizeof(struct abe_twl6040), GFP_KERNEL);
+	if (priv == NULL)
+		return -ENOMEM;
+
+	card = &priv->card;
+	card->dev = &pdev->dev;
+	card->owner = THIS_MODULE;
+	card->dapm_widgets = twl6040_dapm_widgets;
+	card->num_dapm_widgets = ARRAY_SIZE(twl6040_dapm_widgets);
+	card->dapm_routes = audio_map;
+	card->num_dapm_routes = ARRAY_SIZE(audio_map);
+
+	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;
+	}
+
+	priv->dai_links[0].name = "DMIC";
+	priv->dai_links[0].stream_name = "TWL6040";
+	priv->dai_links[0].cpus = link0_cpus;
+	priv->dai_links[0].num_cpus = 1;
+	priv->dai_links[0].cpus->of_node = dai_node;
+	priv->dai_links[0].platforms = link0_platforms;
+	priv->dai_links[0].num_platforms = 1;
+	priv->dai_links[0].platforms->of_node = dai_node;
+	priv->dai_links[0].codecs = link0_codecs;
+	priv->dai_links[0].num_codecs = 1;
+	priv->dai_links[0].init = omap_abe_twl6040_init;
+	priv->dai_links[0].ops = &omap_abe_ops;
+
+	dai_node = of_parse_phandle(node, "ti,dmic", 0);
+	if (dai_node) {
+		num_links = 2;
+		priv->dai_links[1].name = "TWL6040";
+		priv->dai_links[1].stream_name = "DMIC Capture";
+		priv->dai_links[1].cpus = link1_cpus;
+		priv->dai_links[1].num_cpus = 1;
+		priv->dai_links[1].cpus->of_node = dai_node;
+		priv->dai_links[1].platforms = link1_platforms;
+		priv->dai_links[1].num_platforms = 1;
+		priv->dai_links[1].platforms->of_node = dai_node;
+		priv->dai_links[1].codecs = link1_codecs;
+		priv->dai_links[1].num_codecs = 1;
+		priv->dai_links[1].init = omap_abe_dmic_init;
+		priv->dai_links[1].ops = &omap_abe_dmic_ops;
+	} 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");
+		return -EINVAL;
+	}
+
+	card->fully_routed = 1;
+
+	card->dai_link = priv->dai_links;
+	card->num_links = num_links;
+
+	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;
+}
+
+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",
+		.pm = &snd_soc_pm_ops,
+		.of_match_table = omap_abe_of_match,
+	},
+	.probe = omap_abe_probe,
+};
+
+static int __init omap_abe_init(void)
+{
+	int ret;
+
+	dmic_codec_dev = platform_device_register_simple("dmic-codec", -1, NULL,
+							 0);
+	if (IS_ERR(dmic_codec_dev)) {
+		pr_err("%s: dmic-codec device registration failed\n", __func__);
+		return PTR_ERR(dmic_codec_dev);
+	}
+
+	ret = platform_driver_register(&omap_abe_driver);
+	if (ret) {
+		pr_err("%s: platform driver registration failed\n", __func__);
+		platform_device_unregister(dmic_codec_dev);
+	}
+
+	return ret;
+}
+module_init(omap_abe_init);
+
+static void __exit omap_abe_exit(void)
+{
+	platform_driver_unregister(&omap_abe_driver);
+	platform_device_unregister(dmic_codec_dev);
+}
+module_exit(omap_abe_exit);
+
+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");
diff --git a/sound/soc/ti/omap-dmic.c b/sound/soc/ti/omap-dmic.c
new file mode 100644
index 000000000..825c70a44
--- /dev/null
+++ b/sound/soc/ti/omap-dmic.c
@@ -0,0 +1,529 @@
+// SPDX-License-Identifier: GPL-2.0-only
+/*
+ * 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>
+ */
+
+#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 "omap-dmic.h"
+#include "sdma-pcm.h"
+
+struct omap_dmic {
+	struct device *dev;
+	void __iomem *io_base;
+	struct clk *fclk;
+	struct pm_qos_request pm_qos_req;
+	int latency;
+	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 (!snd_soc_dai_active(dai))
+		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);
+
+	cpu_latency_qos_remove_request(&dmic->pm_qos_req);
+
+	if (!snd_soc_dai_active(dai))
+		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;
+		fallthrough;
+	case 4:
+		dmic->ch_enabled |= OMAP_DMIC_UP2_ENABLE;
+		fallthrough;
+	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;
+	dmic->latency = (OMAP_DMIC_THRES_MAX - dmic->threshold) * USEC_PER_SEC /
+			params_rate(params);
+
+	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;
+
+	if (cpu_latency_qos_request_active(&dmic->pm_qos_req))
+		cpu_latency_qos_update_request(&dmic->pm_qos_req,
+					       dmic->latency);
+
+	/* 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, *mux;
+	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;
+	}
+
+	mux = clk_get_parent(dmic->fclk);
+	if (IS_ERR(mux)) {
+		dev_err(dmic->dev, "can't get fck mux parent\n");
+		clk_put(parent_clk);
+		return -ENODEV;
+	}
+
+	mutex_lock(&dmic->mutex);
+	if (dmic->active) {
+		/* disable clock while reparenting */
+		pm_runtime_put_sync(dmic->dev);
+		ret = clk_set_parent(mux, parent_clk);
+		pm_runtime_get_sync(dmic->dev);
+	} else {
+		ret = clk_set_parent(mux, 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(mux);
+	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",
+	.legacy_dai_naming	= 1,
+};
+
+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, "can't 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 = sdma_pcm_platform_register(&pdev->dev, NULL, "up_link");
+	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",
+		.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");
diff --git a/sound/soc/ti/omap-dmic.h b/sound/soc/ti/omap-dmic.h
new file mode 100644
index 000000000..472cdbd9a
--- /dev/null
+++ b/sound/soc/ti/omap-dmic.h
@@ -0,0 +1,66 @@
+/* SPDX-License-Identifier: GPL-2.0-only */
+/*
+ * omap-dmic.h  --  OMAP Digital Microphone Controller
+ */
+
+#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
diff --git a/sound/soc/ti/omap-hdmi.c b/sound/soc/ti/omap-hdmi.c
new file mode 100644
index 000000000..0dc047567
--- /dev/null
+++ b/sound/soc/ti/omap-hdmi.c
@@ -0,0 +1,422 @@
+// SPDX-License-Identifier: GPL-2.0-only
+/*
+ * omap-hdmi-audio.c -- OMAP4+ DSS HDMI audio support library
+ *
+ * Copyright (C) 2014 Texas Instruments Incorporated - https://www.ti.com
+ *
+ * Author: Jyri Sarha <jsarha@ti.com>
+ */
+
+#include <linux/kernel.h>
+#include <linux/module.h>
+#include <linux/err.h>
+#include <linux/string.h>
+#include <linux/platform_device.h>
+#include <sound/soc.h>
+#include <sound/pcm_params.h>
+#include <sound/dmaengine_pcm.h>
+#include <uapi/sound/asound.h>
+#include <sound/asoundef.h>
+#include <sound/omap-hdmi-audio.h>
+
+#include "sdma-pcm.h"
+
+#define DRV_NAME "omap-hdmi-audio"
+
+struct hdmi_audio_data {
+	struct snd_soc_card *card;
+
+	const struct omap_hdmi_audio_ops *ops;
+	struct device *dssdev;
+	struct snd_dmaengine_dai_dma_data dma_data;
+	struct omap_dss_audio dss_audio;
+	struct snd_aes_iec958 iec;
+	struct snd_cea_861_aud_if cea;
+
+	struct mutex current_stream_lock;
+	struct snd_pcm_substream *current_stream;
+};
+
+static
+struct hdmi_audio_data *card_drvdata_substream(struct snd_pcm_substream *ss)
+{
+	struct snd_soc_pcm_runtime *rtd = ss->private_data;
+
+	return snd_soc_card_get_drvdata(rtd->card);
+}
+
+static void hdmi_dai_abort(struct device *dev)
+{
+	struct hdmi_audio_data *ad = dev_get_drvdata(dev);
+
+	mutex_lock(&ad->current_stream_lock);
+	if (ad->current_stream && ad->current_stream->runtime &&
+	    snd_pcm_running(ad->current_stream)) {
+		dev_err(dev, "HDMI display disabled, aborting playback\n");
+		snd_pcm_stream_lock_irq(ad->current_stream);
+		snd_pcm_stop(ad->current_stream, SNDRV_PCM_STATE_DISCONNECTED);
+		snd_pcm_stream_unlock_irq(ad->current_stream);
+	}
+	mutex_unlock(&ad->current_stream_lock);
+}
+
+static int hdmi_dai_startup(struct snd_pcm_substream *substream,
+			    struct snd_soc_dai *dai)
+{
+	struct hdmi_audio_data *ad = card_drvdata_substream(substream);
+	int ret;
+	/*
+	 * 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
+	 */
+	ret = snd_pcm_hw_constraint_step(substream->runtime, 0,
+					 SNDRV_PCM_HW_PARAM_PERIOD_BYTES, 128);
+	if (ret < 0) {
+		dev_err(dai->dev, "Could not apply period constraint: %d\n",
+			ret);
+		return ret;
+	}
+	ret = snd_pcm_hw_constraint_step(substream->runtime, 0,
+					 SNDRV_PCM_HW_PARAM_BUFFER_BYTES, 128);
+	if (ret < 0) {
+		dev_err(dai->dev, "Could not apply buffer constraint: %d\n",
+			ret);
+		return ret;
+	}
+
+	snd_soc_dai_set_dma_data(dai, substream, &ad->dma_data);
+
+	mutex_lock(&ad->current_stream_lock);
+	ad->current_stream = substream;
+	mutex_unlock(&ad->current_stream_lock);
+
+	ret = ad->ops->audio_startup(ad->dssdev, hdmi_dai_abort);
+
+	if (ret) {
+		mutex_lock(&ad->current_stream_lock);
+		ad->current_stream = NULL;
+		mutex_unlock(&ad->current_stream_lock);
+	}
+
+	return ret;
+}
+
+static int hdmi_dai_hw_params(struct snd_pcm_substream *substream,
+			      struct snd_pcm_hw_params *params,
+			      struct snd_soc_dai *dai)
+{
+	struct hdmi_audio_data *ad = card_drvdata_substream(substream);
+	struct snd_aes_iec958 *iec = &ad->iec;
+	struct snd_cea_861_aud_if *cea = &ad->cea;
+
+	WARN_ON(ad->current_stream != substream);
+
+	switch (params_format(params)) {
+	case SNDRV_PCM_FORMAT_S16_LE:
+		ad->dma_data.maxburst = 16;
+		break;
+	case SNDRV_PCM_FORMAT_S24_LE:
+		ad->dma_data.maxburst = 32;
+		break;
+	default:
+		dev_err(dai->dev, "format not supported!\n");
+		return -EINVAL;
+	}
+
+	ad->dss_audio.iec = iec;
+	ad->dss_audio.cea = cea;
+	/*
+	 * 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[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 */
+
+	if (params_channels(params) == 2)
+		cea->db4_ca = 0x0;
+	else if (params_channels(params) == 6)
+		cea->db4_ca = 0xb;
+	else
+		cea->db4_ca = 0x13;
+
+	if (cea->db4_ca == 0x00)
+		cea->db5_dminh_lsv = CEA861_AUDIO_INFOFRAME_DB5_DM_INH_PERMITTED;
+	else
+		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);
+
+	return ad->ops->audio_config(ad->dssdev, &ad->dss_audio);
+}
+
+static int hdmi_dai_trigger(struct snd_pcm_substream *substream, int cmd,
+			    struct snd_soc_dai *dai)
+{
+	struct hdmi_audio_data *ad = card_drvdata_substream(substream);
+	int err = 0;
+
+	WARN_ON(ad->current_stream != substream);
+
+	switch (cmd) {
+	case SNDRV_PCM_TRIGGER_START:
+	case SNDRV_PCM_TRIGGER_RESUME:
+	case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
+		err = ad->ops->audio_start(ad->dssdev);
+		break;
+	case SNDRV_PCM_TRIGGER_STOP:
+	case SNDRV_PCM_TRIGGER_SUSPEND:
+	case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
+		ad->ops->audio_stop(ad->dssdev);
+		break;
+	default:
+		err = -EINVAL;
+	}
+	return err;
+}
+
+static void hdmi_dai_shutdown(struct snd_pcm_substream *substream,
+			      struct snd_soc_dai *dai)
+{
+	struct hdmi_audio_data *ad = card_drvdata_substream(substream);
+
+	WARN_ON(ad->current_stream != substream);
+
+	ad->ops->audio_shutdown(ad->dssdev);
+
+	mutex_lock(&ad->current_stream_lock);
+	ad->current_stream = NULL;
+	mutex_unlock(&ad->current_stream_lock);
+}
+
+static const struct snd_soc_dai_ops hdmi_dai_ops = {
+	.startup	= hdmi_dai_startup,
+	.hw_params	= hdmi_dai_hw_params,
+	.trigger	= hdmi_dai_trigger,
+	.shutdown	= hdmi_dai_shutdown,
+};
+
+static const struct snd_soc_component_driver omap_hdmi_component = {
+	.name = "omapdss_hdmi",
+	.legacy_dai_naming = 1,
+};
+
+static struct snd_soc_dai_driver omap5_hdmi_dai = {
+	.name = "omap5-hdmi-dai",
+	.playback = {
+		.channels_min = 2,
+		.channels_max = 8,
+		.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),
+		.formats = SNDRV_PCM_FMTBIT_S16_LE,
+	},
+	.ops = &hdmi_dai_ops,
+};
+
+static struct snd_soc_dai_driver omap4_hdmi_dai = {
+	.name = "omap4-hdmi-dai",
+	.playback = {
+		.channels_min = 2,
+		.channels_max = 8,
+		.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),
+		.formats = SNDRV_PCM_FMTBIT_S16_LE | SNDRV_PCM_FMTBIT_S24_LE,
+	},
+	.ops = &hdmi_dai_ops,
+};
+
+static int omap_hdmi_audio_probe(struct platform_device *pdev)
+{
+	struct omap_hdmi_audio_pdata *ha = pdev->dev.platform_data;
+	struct device *dev = &pdev->dev;
+	struct hdmi_audio_data *ad;
+	struct snd_soc_dai_driver *dai_drv;
+	struct snd_soc_card *card;
+	struct snd_soc_dai_link_component *compnent;
+	int ret;
+
+	if (!ha) {
+		dev_err(dev, "No platform data\n");
+		return -EINVAL;
+	}
+
+	ad = devm_kzalloc(dev, sizeof(*ad), GFP_KERNEL);
+	if (!ad)
+		return -ENOMEM;
+	ad->dssdev = ha->dev;
+	ad->ops = ha->ops;
+	ad->dma_data.addr = ha->audio_dma_addr;
+	ad->dma_data.filter_data = "audio_tx";
+	ad->dma_data.addr_width = DMA_SLAVE_BUSWIDTH_4_BYTES;
+	mutex_init(&ad->current_stream_lock);
+
+	switch (ha->version) {
+	case 4:
+		dai_drv = &omap4_hdmi_dai;
+		break;
+	case 5:
+		dai_drv = &omap5_hdmi_dai;
+		break;
+	default:
+		return -EINVAL;
+	}
+	ret = devm_snd_soc_register_component(ad->dssdev, &omap_hdmi_component,
+					 dai_drv, 1);
+	if (ret)
+		return ret;
+
+	ret = sdma_pcm_platform_register(ad->dssdev, "audio_tx", NULL);
+	if (ret)
+		return ret;
+
+	card = devm_kzalloc(dev, sizeof(*card), GFP_KERNEL);
+	if (!card)
+		return -ENOMEM;
+
+	card->name = devm_kasprintf(dev, GFP_KERNEL,
+				    "HDMI %s", dev_name(ad->dssdev));
+	if (!card->name)
+		return -ENOMEM;
+
+	card->owner = THIS_MODULE;
+	card->dai_link =
+		devm_kzalloc(dev, sizeof(*(card->dai_link)), GFP_KERNEL);
+	if (!card->dai_link)
+		return -ENOMEM;
+
+	compnent = devm_kzalloc(dev, 3 * sizeof(*compnent), GFP_KERNEL);
+	if (!compnent)
+		return -ENOMEM;
+	card->dai_link->cpus		= &compnent[0];
+	card->dai_link->num_cpus	= 1;
+	card->dai_link->codecs		= &compnent[1];
+	card->dai_link->num_codecs	= 1;
+	card->dai_link->platforms	= &compnent[2];
+	card->dai_link->num_platforms	= 1;
+
+	card->dai_link->name = card->name;
+	card->dai_link->stream_name = card->name;
+	card->dai_link->cpus->dai_name = dev_name(ad->dssdev);
+	card->dai_link->platforms->name = dev_name(ad->dssdev);
+	card->dai_link->codecs->name = "snd-soc-dummy";
+	card->dai_link->codecs->dai_name = "snd-soc-dummy-dai";
+	card->num_links = 1;
+	card->dev = dev;
+
+	ret = snd_soc_register_card(card);
+	if (ret) {
+		dev_err(dev, "snd_soc_register_card failed (%d)\n", ret);
+		return ret;
+	}
+
+	ad->card = card;
+	snd_soc_card_set_drvdata(card, ad);
+
+	dev_set_drvdata(dev, ad);
+
+	return 0;
+}
+
+static int omap_hdmi_audio_remove(struct platform_device *pdev)
+{
+	struct hdmi_audio_data *ad = platform_get_drvdata(pdev);
+
+	snd_soc_unregister_card(ad->card);
+	return 0;
+}
+
+static struct platform_driver hdmi_audio_driver = {
+	.driver = {
+		.name = DRV_NAME,
+	},
+	.probe = omap_hdmi_audio_probe,
+	.remove = omap_hdmi_audio_remove,
+};
+
+module_platform_driver(hdmi_audio_driver);
+
+MODULE_AUTHOR("Jyri Sarha <jsarha@ti.com>");
+MODULE_DESCRIPTION("OMAP HDMI Audio Driver");
+MODULE_LICENSE("GPL");
+MODULE_ALIAS("platform:" DRV_NAME);
diff --git a/sound/soc/ti/omap-mcbsp-priv.h b/sound/soc/ti/omap-mcbsp-priv.h
new file mode 100644
index 000000000..da519ea1f
--- /dev/null
+++ b/sound/soc/ti/omap-mcbsp-priv.h
@@ -0,0 +1,322 @@
+/* SPDX-License-Identifier: GPL-2.0 */
+/*
+ * OMAP Multi-Channel Buffered Serial Port
+ *
+ * Contact: Jarkko Nikula <jarkko.nikula@bitmer.com>
+ *          Peter Ujfalusi <peter.ujfalusi@ti.com>
+ */
+
+#ifndef __OMAP_MCBSP_PRIV_H__
+#define __OMAP_MCBSP_PRIV_H__
+
+#include <linux/platform_data/asoc-ti-mcbsp.h>
+
+#ifdef CONFIG_ARCH_OMAP1
+#define mcbsp_omap1()	1
+#else
+#define mcbsp_omap1()	0
+#endif
+
+/* 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,
+};
+
+/************************** 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 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;
+
+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;
+	unsigned int latency[2];
+	int clk_div;
+	int wlen;
+
+	struct pm_qos_request pm_qos_req;
+};
+
+static inline void omap_mcbsp_write(struct omap_mcbsp *mcbsp, u16 reg, u32 val)
+{
+	void __iomem *addr = mcbsp->io_base + reg * mcbsp->pdata->reg_step;
+
+	if (mcbsp->pdata->reg_size == 2) {
+		((u16 *)mcbsp->reg_cache)[reg] = (u16)val;
+		writew_relaxed((u16)val, addr);
+	} else {
+		((u32 *)mcbsp->reg_cache)[reg] = val;
+		writel_relaxed(val, addr);
+	}
+}
+
+static inline int omap_mcbsp_read(struct omap_mcbsp *mcbsp, u16 reg,
+				  bool from_cache)
+{
+	void __iomem *addr = mcbsp->io_base + reg * mcbsp->pdata->reg_step;
+
+	if (mcbsp->pdata->reg_size == 2) {
+		return !from_cache ? readw_relaxed(addr) :
+				     ((u16 *)mcbsp->reg_cache)[reg];
+	} else {
+		return !from_cache ? readl_relaxed(addr) :
+				     ((u32 *)mcbsp->reg_cache)[reg];
+	}
+}
+
+#define MCBSP_READ(mcbsp, reg) \
+		omap_mcbsp_read(mcbsp, OMAP_MCBSP_REG_##reg, 0)
+#define MCBSP_WRITE(mcbsp, reg, val) \
+		omap_mcbsp_write(mcbsp, OMAP_MCBSP_REG_##reg, val)
+#define MCBSP_READ_CACHE(mcbsp, reg) \
+		omap_mcbsp_read(mcbsp, OMAP_MCBSP_REG_##reg, 1)
+
+
+/* Sidetone specific API */
+int omap_mcbsp_st_init(struct platform_device *pdev);
+int omap_mcbsp_st_start(struct omap_mcbsp *mcbsp);
+int omap_mcbsp_st_stop(struct omap_mcbsp *mcbsp);
+
+#endif /* __OMAP_MCBSP_PRIV_H__ */
diff --git a/sound/soc/ti/omap-mcbsp-st.c b/sound/soc/ti/omap-mcbsp-st.c
new file mode 100644
index 000000000..8163f453b
--- /dev/null
+++ b/sound/soc/ti/omap-mcbsp-st.c
@@ -0,0 +1,503 @@
+// SPDX-License-Identifier: GPL-2.0
+/*
+ * McBSP Sidetone support
+ *
+ * Copyright (C) 2004 Nokia Corporation
+ * Author: Samuel Ortiz <samuel.ortiz@nokia.com>
+ *
+ * Contact: Jarkko Nikula <jarkko.nikula@bitmer.com>
+ *          Peter Ujfalusi <peter.ujfalusi@ti.com>
+ */
+
+#include <linux/module.h>
+#include <linux/init.h>
+#include <linux/device.h>
+#include <linux/platform_device.h>
+#include <linux/interrupt.h>
+#include <linux/err.h>
+#include <linux/clk.h>
+#include <linux/delay.h>
+#include <linux/io.h>
+#include <linux/slab.h>
+#include <linux/pm_runtime.h>
+
+#include "omap-mcbsp.h"
+#include "omap-mcbsp-priv.h"
+
+/* 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 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)
+
+struct omap_mcbsp_st_data {
+	void __iomem *io_base_st;
+	struct clk *mcbsp_iclk;
+	bool running;
+	bool enabled;
+	s16 taps[128];	/* Sidetone filter coefficients */
+	int nr_taps;	/* Number of filter coefficients in use */
+	s16 ch0gain;
+	s16 ch1gain;
+};
+
+static void omap_mcbsp_st_write(struct omap_mcbsp *mcbsp, u16 reg, u32 val)
+{
+	writel_relaxed(val, mcbsp->st_data->io_base_st + reg);
+}
+
+static int omap_mcbsp_st_read(struct omap_mcbsp *mcbsp, u16 reg)
+{
+	return readl_relaxed(mcbsp->st_data->io_base_st + reg);
+}
+
+#define MCBSP_ST_READ(mcbsp, reg) omap_mcbsp_st_read(mcbsp, OMAP_ST_REG_##reg)
+#define MCBSP_ST_WRITE(mcbsp, reg, val) \
+			omap_mcbsp_st_write(mcbsp, OMAP_ST_REG_##reg, val)
+
+static void omap_mcbsp_st_on(struct omap_mcbsp *mcbsp)
+{
+	unsigned int w;
+
+	if (mcbsp->pdata->force_ick_on)
+		mcbsp->pdata->force_ick_on(mcbsp->st_data->mcbsp_iclk, true);
+
+	/* Disable Sidetone clock auto-gating for normal operation */
+	w = MCBSP_ST_READ(mcbsp, SYSCONFIG);
+	MCBSP_ST_WRITE(mcbsp, SYSCONFIG, w & ~(ST_AUTOIDLE));
+
+	/* Enable McBSP Sidetone */
+	w = MCBSP_READ(mcbsp, SSELCR);
+	MCBSP_WRITE(mcbsp, SSELCR, w | SIDETONEEN);
+
+	/* Enable Sidetone from Sidetone Core */
+	w = MCBSP_ST_READ(mcbsp, SSELCR);
+	MCBSP_ST_WRITE(mcbsp, SSELCR, w | ST_SIDETONEEN);
+}
+
+static void omap_mcbsp_st_off(struct omap_mcbsp *mcbsp)
+{
+	unsigned int w;
+
+	w = MCBSP_ST_READ(mcbsp, SSELCR);
+	MCBSP_ST_WRITE(mcbsp, SSELCR, w & ~(ST_SIDETONEEN));
+
+	w = MCBSP_READ(mcbsp, SSELCR);
+	MCBSP_WRITE(mcbsp, SSELCR, w & ~(SIDETONEEN));
+
+	/* Enable Sidetone clock auto-gating to reduce power consumption */
+	w = MCBSP_ST_READ(mcbsp, SYSCONFIG);
+	MCBSP_ST_WRITE(mcbsp, SYSCONFIG, w | ST_AUTOIDLE);
+
+	if (mcbsp->pdata->force_ick_on)
+		mcbsp->pdata->force_ick_on(mcbsp->st_data->mcbsp_iclk, false);
+}
+
+static void omap_mcbsp_st_fir_write(struct omap_mcbsp *mcbsp, s16 *fir)
+{
+	u16 val, i;
+
+	val = MCBSP_ST_READ(mcbsp, SSELCR);
+
+	if (val & ST_COEFFWREN)
+		MCBSP_ST_WRITE(mcbsp, SSELCR, val & ~(ST_COEFFWREN));
+
+	MCBSP_ST_WRITE(mcbsp, SSELCR, val | ST_COEFFWREN);
+
+	for (i = 0; i < 128; i++)
+		MCBSP_ST_WRITE(mcbsp, SFIRCR, fir[i]);
+
+	i = 0;
+
+	val = MCBSP_ST_READ(mcbsp, SSELCR);
+	while (!(val & ST_COEFFWRDONE) && (++i < 1000))
+		val = MCBSP_ST_READ(mcbsp, SSELCR);
+
+	MCBSP_ST_WRITE(mcbsp, SSELCR, val & ~(ST_COEFFWREN));
+
+	if (i == 1000)
+		dev_err(mcbsp->dev, "McBSP FIR load error!\n");
+}
+
+static void omap_mcbsp_st_chgain(struct omap_mcbsp *mcbsp)
+{
+	struct omap_mcbsp_st_data *st_data = mcbsp->st_data;
+
+	MCBSP_ST_WRITE(mcbsp, SGAINCR, ST_CH0GAIN(st_data->ch0gain) |
+		       ST_CH1GAIN(st_data->ch1gain));
+}
+
+static int omap_mcbsp_st_set_chgain(struct omap_mcbsp *mcbsp, int channel,
+				    s16 chgain)
+{
+	struct omap_mcbsp_st_data *st_data = mcbsp->st_data;
+	int ret = 0;
+
+	if (!st_data)
+		return -ENOENT;
+
+	spin_lock_irq(&mcbsp->lock);
+	if (channel == 0)
+		st_data->ch0gain = chgain;
+	else if (channel == 1)
+		st_data->ch1gain = chgain;
+	else
+		ret = -EINVAL;
+
+	if (st_data->enabled)
+		omap_mcbsp_st_chgain(mcbsp);
+	spin_unlock_irq(&mcbsp->lock);
+
+	return ret;
+}
+
+static int omap_mcbsp_st_get_chgain(struct omap_mcbsp *mcbsp, int channel,
+				    s16 *chgain)
+{
+	struct omap_mcbsp_st_data *st_data = mcbsp->st_data;
+	int ret = 0;
+
+	if (!st_data)
+		return -ENOENT;
+
+	spin_lock_irq(&mcbsp->lock);
+	if (channel == 0)
+		*chgain = st_data->ch0gain;
+	else if (channel == 1)
+		*chgain = st_data->ch1gain;
+	else
+		ret = -EINVAL;
+	spin_unlock_irq(&mcbsp->lock);
+
+	return ret;
+}
+
+static int omap_mcbsp_st_enable(struct omap_mcbsp *mcbsp)
+{
+	struct omap_mcbsp_st_data *st_data = mcbsp->st_data;
+
+	if (!st_data)
+		return -ENODEV;
+
+	spin_lock_irq(&mcbsp->lock);
+	st_data->enabled = 1;
+	omap_mcbsp_st_start(mcbsp);
+	spin_unlock_irq(&mcbsp->lock);
+
+	return 0;
+}
+
+static int omap_mcbsp_st_disable(struct omap_mcbsp *mcbsp)
+{
+	struct omap_mcbsp_st_data *st_data = mcbsp->st_data;
+	int ret = 0;
+
+	if (!st_data)
+		return -ENODEV;
+
+	spin_lock_irq(&mcbsp->lock);
+	omap_mcbsp_st_stop(mcbsp);
+	st_data->enabled = 0;
+	spin_unlock_irq(&mcbsp->lock);
+
+	return ret;
+}
+
+static int omap_mcbsp_st_is_enabled(struct omap_mcbsp *mcbsp)
+{
+	struct omap_mcbsp_st_data *st_data = mcbsp->st_data;
+
+	if (!st_data)
+		return -ENODEV;
+
+	return st_data->enabled;
+}
+
+static ssize_t st_taps_show(struct device *dev,
+			    struct device_attribute *attr, char *buf)
+{
+	struct omap_mcbsp *mcbsp = dev_get_drvdata(dev);
+	struct omap_mcbsp_st_data *st_data = mcbsp->st_data;
+	ssize_t status = 0;
+	int i;
+
+	spin_lock_irq(&mcbsp->lock);
+	for (i = 0; i < st_data->nr_taps; i++)
+		status += sysfs_emit_at(buf, status, (i ? ", %d" : "%d"),
+					st_data->taps[i]);
+	if (i)
+		status += sysfs_emit_at(buf, status, "\n");
+	spin_unlock_irq(&mcbsp->lock);
+
+	return status;
+}
+
+static ssize_t st_taps_store(struct device *dev,
+			     struct device_attribute *attr,
+			     const char *buf, size_t size)
+{
+	struct omap_mcbsp *mcbsp = dev_get_drvdata(dev);
+	struct omap_mcbsp_st_data *st_data = mcbsp->st_data;
+	int val, tmp, status, i = 0;
+
+	spin_lock_irq(&mcbsp->lock);
+	memset(st_data->taps, 0, sizeof(st_data->taps));
+	st_data->nr_taps = 0;
+
+	do {
+		status = sscanf(buf, "%d%n", &val, &tmp);
+		if (status < 0 || status == 0) {
+			size = -EINVAL;
+			goto out;
+		}
+		if (val < -32768 || val > 32767) {
+			size = -EINVAL;
+			goto out;
+		}
+		st_data->taps[i++] = val;
+		buf += tmp;
+		if (*buf != ',')
+			break;
+		buf++;
+	} while (1);
+
+	st_data->nr_taps = i;
+
+out:
+	spin_unlock_irq(&mcbsp->lock);
+
+	return size;
+}
+
+static DEVICE_ATTR_RW(st_taps);
+
+static const struct attribute *sidetone_attrs[] = {
+	&dev_attr_st_taps.attr,
+	NULL,
+};
+
+static const struct attribute_group sidetone_attr_group = {
+	.attrs = (struct attribute **)sidetone_attrs,
+};
+
+int omap_mcbsp_st_start(struct omap_mcbsp *mcbsp)
+{
+	struct omap_mcbsp_st_data *st_data = mcbsp->st_data;
+
+	if (st_data->enabled && !st_data->running) {
+		omap_mcbsp_st_fir_write(mcbsp, st_data->taps);
+		omap_mcbsp_st_chgain(mcbsp);
+
+		if (!mcbsp->free) {
+			omap_mcbsp_st_on(mcbsp);
+			st_data->running = 1;
+		}
+	}
+
+	return 0;
+}
+
+int omap_mcbsp_st_stop(struct omap_mcbsp *mcbsp)
+{
+	struct omap_mcbsp_st_data *st_data = mcbsp->st_data;
+
+	if (st_data->running) {
+		if (!mcbsp->free) {
+			omap_mcbsp_st_off(mcbsp);
+			st_data->running = 0;
+		}
+	}
+
+	return 0;
+}
+
+int omap_mcbsp_st_init(struct platform_device *pdev)
+{
+	struct omap_mcbsp *mcbsp = platform_get_drvdata(pdev);
+	struct omap_mcbsp_st_data *st_data;
+	struct resource *res;
+	int ret;
+
+	res = platform_get_resource_byname(pdev, IORESOURCE_MEM, "sidetone");
+	if (!res)
+		return 0;
+
+	st_data = devm_kzalloc(mcbsp->dev, sizeof(*mcbsp->st_data), GFP_KERNEL);
+	if (!st_data)
+		return -ENOMEM;
+
+	st_data->mcbsp_iclk = devm_clk_get(mcbsp->dev, "ick");
+	if (IS_ERR(st_data->mcbsp_iclk)) {
+		dev_warn(mcbsp->dev,
+			 "Failed to get ick, sidetone might be broken\n");
+		st_data->mcbsp_iclk = NULL;
+	}
+
+	st_data->io_base_st = devm_ioremap(mcbsp->dev, res->start,
+					   resource_size(res));
+	if (!st_data->io_base_st)
+		return -ENOMEM;
+
+	ret = devm_device_add_group(mcbsp->dev, &sidetone_attr_group);
+	if (ret)
+		return ret;
+
+	mcbsp->st_data = st_data;
+
+	return 0;
+}
+
+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_mcbsp_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_mcbsp_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_mcbsp_st_is_enabled(mcbsp))
+		return 0;
+
+	if (value)
+		omap_mcbsp_st_enable(mcbsp);
+	else
+		omap_mcbsp_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_mcbsp_st_is_enabled(mcbsp);
+	return 0;
+}
+
+#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} }
+
+#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 = asoc_rtd_to_cpu(rtd, 0);
+	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);
diff --git a/sound/soc/ti/omap-mcbsp.c b/sound/soc/ti/omap-mcbsp.c
new file mode 100644
index 000000000..4b8aac1a3
--- /dev/null
+++ b/sound/soc/ti/omap-mcbsp.c
@@ -0,0 +1,1445 @@
+// SPDX-License-Identifier: GPL-2.0-only
+/*
+ * 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>
+ */
+
+#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 "omap-mcbsp-priv.h"
+#include "omap-mcbsp.h"
+#include "sdma-pcm.h"
+
+#define OMAP_MCBSP_RATES	(SNDRV_PCM_RATE_8000_96000)
+
+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,
+};
+
+static void omap_mcbsp_dump_reg(struct omap_mcbsp *mcbsp)
+{
+	dev_dbg(mcbsp->dev, "**** McBSP%d regs ****\n", mcbsp->id);
+	dev_dbg(mcbsp->dev, "DRR2:  0x%04x\n", MCBSP_READ(mcbsp, DRR2));
+	dev_dbg(mcbsp->dev, "DRR1:  0x%04x\n", MCBSP_READ(mcbsp, DRR1));
+	dev_dbg(mcbsp->dev, "DXR2:  0x%04x\n", MCBSP_READ(mcbsp, DXR2));
+	dev_dbg(mcbsp->dev, "DXR1:  0x%04x\n", MCBSP_READ(mcbsp, DXR1));
+	dev_dbg(mcbsp->dev, "SPCR2: 0x%04x\n", MCBSP_READ(mcbsp, SPCR2));
+	dev_dbg(mcbsp->dev, "SPCR1: 0x%04x\n", MCBSP_READ(mcbsp, SPCR1));
+	dev_dbg(mcbsp->dev, "RCR2:  0x%04x\n", MCBSP_READ(mcbsp, RCR2));
+	dev_dbg(mcbsp->dev, "RCR1:  0x%04x\n", MCBSP_READ(mcbsp, RCR1));
+	dev_dbg(mcbsp->dev, "XCR2:  0x%04x\n", MCBSP_READ(mcbsp, XCR2));
+	dev_dbg(mcbsp->dev, "XCR1:  0x%04x\n", MCBSP_READ(mcbsp, XCR1));
+	dev_dbg(mcbsp->dev, "SRGR2: 0x%04x\n", MCBSP_READ(mcbsp, SRGR2));
+	dev_dbg(mcbsp->dev, "SRGR1: 0x%04x\n", MCBSP_READ(mcbsp, SRGR1));
+	dev_dbg(mcbsp->dev, "PCR0:  0x%04x\n", MCBSP_READ(mcbsp, PCR0));
+	dev_dbg(mcbsp->dev, "***********************\n");
+}
+
+static int omap2_mcbsp_set_clks_src(struct omap_mcbsp *mcbsp, u8 fck_src_id)
+{
+	struct clk *fck_src;
+	const char *src;
+	int r;
+
+	if (fck_src_id == MCBSP_CLKS_PAD_SRC)
+		src = "pad_fck";
+	else if (fck_src_id == MCBSP_CLKS_PRCM_SRC)
+		src = "prcm_fck";
+	else
+		return -EINVAL;
+
+	fck_src = clk_get(mcbsp->dev, src);
+	if (IS_ERR(fck_src)) {
+		dev_err(mcbsp->dev, "CLKS: could not clk_get() %s\n", src);
+		return -EINVAL;
+	}
+
+	if (mcbsp->active)
+		pm_runtime_put_sync(mcbsp->dev);
+
+	r = clk_set_parent(mcbsp->fclk, fck_src);
+	if (r)
+		dev_err(mcbsp->dev, "CLKS: could not clk_set_parent() to %s\n",
+			src);
+
+	if (mcbsp->active)
+		pm_runtime_get_sync(mcbsp->dev);
+
+	clk_put(fck_src);
+
+	return r;
+}
+
+static irqreturn_t omap_mcbsp_irq_handler(int irq, void *data)
+{
+	struct omap_mcbsp *mcbsp = data;
+	u16 irqst;
+
+	irqst = MCBSP_READ(mcbsp, IRQST);
+	dev_dbg(mcbsp->dev, "IRQ callback : 0x%x\n", irqst);
+
+	if (irqst & RSYNCERREN)
+		dev_err(mcbsp->dev, "RX Frame Sync Error!\n");
+	if (irqst & RFSREN)
+		dev_dbg(mcbsp->dev, "RX Frame Sync\n");
+	if (irqst & REOFEN)
+		dev_dbg(mcbsp->dev, "RX End Of Frame\n");
+	if (irqst & RRDYEN)
+		dev_dbg(mcbsp->dev, "RX Buffer Threshold Reached\n");
+	if (irqst & RUNDFLEN)
+		dev_err(mcbsp->dev, "RX Buffer Underflow!\n");
+	if (irqst & ROVFLEN)
+		dev_err(mcbsp->dev, "RX Buffer Overflow!\n");
+
+	if (irqst & XSYNCERREN)
+		dev_err(mcbsp->dev, "TX Frame Sync Error!\n");
+	if (irqst & XFSXEN)
+		dev_dbg(mcbsp->dev, "TX Frame Sync\n");
+	if (irqst & XEOFEN)
+		dev_dbg(mcbsp->dev, "TX End Of Frame\n");
+	if (irqst & XRDYEN)
+		dev_dbg(mcbsp->dev, "TX Buffer threshold Reached\n");
+	if (irqst & XUNDFLEN)
+		dev_err(mcbsp->dev, "TX Buffer Underflow!\n");
+	if (irqst & XOVFLEN)
+		dev_err(mcbsp->dev, "TX Buffer Overflow!\n");
+	if (irqst & XEMPTYEOFEN)
+		dev_dbg(mcbsp->dev, "TX Buffer empty at end of frame\n");
+
+	MCBSP_WRITE(mcbsp, IRQST, irqst);
+
+	return IRQ_HANDLED;
+}
+
+static irqreturn_t omap_mcbsp_tx_irq_handler(int irq, void *data)
+{
+	struct omap_mcbsp *mcbsp = data;
+	u16 irqst_spcr2;
+
+	irqst_spcr2 = MCBSP_READ(mcbsp, SPCR2);
+	dev_dbg(mcbsp->dev, "TX IRQ callback : 0x%x\n", irqst_spcr2);
+
+	if (irqst_spcr2 & XSYNC_ERR) {
+		dev_err(mcbsp->dev, "TX Frame Sync Error! : 0x%x\n",
+			irqst_spcr2);
+		/* Writing zero to XSYNC_ERR clears the IRQ */
+		MCBSP_WRITE(mcbsp, SPCR2, MCBSP_READ_CACHE(mcbsp, SPCR2));
+	}
+
+	return IRQ_HANDLED;
+}
+
+static irqreturn_t omap_mcbsp_rx_irq_handler(int irq, void *data)
+{
+	struct omap_mcbsp *mcbsp = data;
+	u16 irqst_spcr1;
+
+	irqst_spcr1 = MCBSP_READ(mcbsp, SPCR1);
+	dev_dbg(mcbsp->dev, "RX IRQ callback : 0x%x\n", irqst_spcr1);
+
+	if (irqst_spcr1 & RSYNC_ERR) {
+		dev_err(mcbsp->dev, "RX Frame Sync Error! : 0x%x\n",
+			irqst_spcr1);
+		/* Writing zero to RSYNC_ERR clears the IRQ */
+		MCBSP_WRITE(mcbsp, SPCR1, MCBSP_READ_CACHE(mcbsp, SPCR1));
+	}
+
+	return IRQ_HANDLED;
+}
+
+/*
+ * omap_mcbsp_config simply write a config to the
+ * appropriate McBSP.
+ * You either call this function or set the McBSP registers
+ * by yourself before calling omap_mcbsp_start().
+ */
+static void omap_mcbsp_config(struct omap_mcbsp *mcbsp,
+			      const struct omap_mcbsp_reg_cfg *config)
+{
+	dev_dbg(mcbsp->dev, "Configuring McBSP%d  phys_base: 0x%08lx\n",
+		mcbsp->id, mcbsp->phys_base);
+
+	/* We write the given config */
+	MCBSP_WRITE(mcbsp, SPCR2, config->spcr2);
+	MCBSP_WRITE(mcbsp, SPCR1, config->spcr1);
+	MCBSP_WRITE(mcbsp, RCR2, config->rcr2);
+	MCBSP_WRITE(mcbsp, RCR1, config->rcr1);
+	MCBSP_WRITE(mcbsp, XCR2, config->xcr2);
+	MCBSP_WRITE(mcbsp, XCR1, config->xcr1);
+	MCBSP_WRITE(mcbsp, SRGR2, config->srgr2);
+	MCBSP_WRITE(mcbsp, SRGR1, config->srgr1);
+	MCBSP_WRITE(mcbsp, MCR2, config->mcr2);
+	MCBSP_WRITE(mcbsp, MCR1, config->mcr1);
+	MCBSP_WRITE(mcbsp, PCR0, config->pcr0);
+	if (mcbsp->pdata->has_ccr) {
+		MCBSP_WRITE(mcbsp, XCCR, config->xccr);
+		MCBSP_WRITE(mcbsp, RCCR, config->rccr);
+	}
+	/* Enable wakeup behavior */
+	if (mcbsp->pdata->has_wakeup)
+		MCBSP_WRITE(mcbsp, WAKEUPEN, XRDYEN | RRDYEN);
+
+	/* Enable TX/RX sync error interrupts by default */
+	if (mcbsp->irq)
+		MCBSP_WRITE(mcbsp, IRQEN, RSYNCERREN | XSYNCERREN |
+			    RUNDFLEN | ROVFLEN | XUNDFLEN | XOVFLEN);
+}
+
+/**
+ * omap_mcbsp_dma_reg_params - returns the address of mcbsp data register
+ * @mcbsp: omap_mcbsp struct for the McBSP instance
+ * @stream: Stream direction (playback/capture)
+ *
+ * Returns the address of mcbsp data transmit register or data receive register
+ * to be used by DMA for transferring/receiving data
+ */
+static int omap_mcbsp_dma_reg_params(struct omap_mcbsp *mcbsp,
+				     unsigned int stream)
+{
+	int data_reg;
+
+	if (stream == SNDRV_PCM_STREAM_PLAYBACK) {
+		if (mcbsp->pdata->reg_size == 2)
+			data_reg = OMAP_MCBSP_REG_DXR1;
+		else
+			data_reg = OMAP_MCBSP_REG_DXR;
+	} else {
+		if (mcbsp->pdata->reg_size == 2)
+			data_reg = OMAP_MCBSP_REG_DRR1;
+		else
+			data_reg = OMAP_MCBSP_REG_DRR;
+	}
+
+	return mcbsp->phys_dma_base + data_reg * mcbsp->pdata->reg_step;
+}
+
+/*
+ * omap_mcbsp_set_rx_threshold configures the transmit threshold in words.
+ * The threshold parameter is 1 based, and it is converted (threshold - 1)
+ * for the THRSH2 register.
+ */
+static void omap_mcbsp_set_tx_threshold(struct omap_mcbsp *mcbsp, u16 threshold)
+{
+	if (threshold && threshold <= mcbsp->max_tx_thres)
+		MCBSP_WRITE(mcbsp, THRSH2, threshold - 1);
+}
+
+/*
+ * omap_mcbsp_set_rx_threshold configures the receive threshold in words.
+ * The threshold parameter is 1 based, and it is converted (threshold - 1)
+ * for the THRSH1 register.
+ */
+static void omap_mcbsp_set_rx_threshold(struct omap_mcbsp *mcbsp, u16 threshold)
+{
+	if (threshold && threshold <= mcbsp->max_rx_thres)
+		MCBSP_WRITE(mcbsp, THRSH1, threshold - 1);
+}
+
+/*
+ * omap_mcbsp_get_tx_delay returns the number of used slots in the McBSP FIFO
+ */
+static u16 omap_mcbsp_get_tx_delay(struct omap_mcbsp *mcbsp)
+{
+	u16 buffstat;
+
+	/* Returns the number of free locations in the buffer */
+	buffstat = MCBSP_READ(mcbsp, XBUFFSTAT);
+
+	/* Number of slots are different in McBSP ports */
+	return mcbsp->pdata->buffer_size - buffstat;
+}
+
+/*
+ * omap_mcbsp_get_rx_delay returns the number of free slots in the McBSP FIFO
+ * to reach the threshold value (when the DMA will be triggered to read it)
+ */
+static u16 omap_mcbsp_get_rx_delay(struct omap_mcbsp *mcbsp)
+{
+	u16 buffstat, threshold;
+
+	/* Returns the number of used locations in the buffer */
+	buffstat = MCBSP_READ(mcbsp, RBUFFSTAT);
+	/* RX threshold */
+	threshold = MCBSP_READ(mcbsp, THRSH1);
+
+	/* Return the number of location till we reach the threshold limit */
+	if (threshold <= buffstat)
+		return 0;
+	else
+		return threshold - buffstat;
+}
+
+static int omap_mcbsp_request(struct omap_mcbsp *mcbsp)
+{
+	void *reg_cache;
+	int err;
+
+	reg_cache = kzalloc(mcbsp->reg_cache_size, GFP_KERNEL);
+	if (!reg_cache)
+		return -ENOMEM;
+
+	spin_lock(&mcbsp->lock);
+	if (!mcbsp->free) {
+		dev_err(mcbsp->dev, "McBSP%d is currently in use\n", mcbsp->id);
+		err = -EBUSY;
+		goto err_kfree;
+	}
+
+	mcbsp->free = false;
+	mcbsp->reg_cache = reg_cache;
+	spin_unlock(&mcbsp->lock);
+
+	if(mcbsp->pdata->ops && mcbsp->pdata->ops->request)
+		mcbsp->pdata->ops->request(mcbsp->id - 1);
+
+	/*
+	 * Make sure that transmitter, receiver and sample-rate generator are
+	 * not running before activating IRQs.
+	 */
+	MCBSP_WRITE(mcbsp, SPCR1, 0);
+	MCBSP_WRITE(mcbsp, SPCR2, 0);
+
+	if (mcbsp->irq) {
+		err = request_irq(mcbsp->irq, omap_mcbsp_irq_handler, 0,
+				  "McBSP", (void *)mcbsp);
+		if (err != 0) {
+			dev_err(mcbsp->dev, "Unable to request IRQ\n");
+			goto err_clk_disable;
+		}
+	} else {
+		err = request_irq(mcbsp->tx_irq, omap_mcbsp_tx_irq_handler, 0,
+				  "McBSP TX", (void *)mcbsp);
+		if (err != 0) {
+			dev_err(mcbsp->dev, "Unable to request TX IRQ\n");
+			goto err_clk_disable;
+		}
+
+		err = request_irq(mcbsp->rx_irq, omap_mcbsp_rx_irq_handler, 0,
+				  "McBSP RX", (void *)mcbsp);
+		if (err != 0) {
+			dev_err(mcbsp->dev, "Unable to request RX IRQ\n");
+			goto err_free_irq;
+		}
+	}
+
+	return 0;
+err_free_irq:
+	free_irq(mcbsp->tx_irq, (void *)mcbsp);
+err_clk_disable:
+	if(mcbsp->pdata->ops && mcbsp->pdata->ops->free)
+		mcbsp->pdata->ops->free(mcbsp->id - 1);
+
+	/* Disable wakeup behavior */
+	if (mcbsp->pdata->has_wakeup)
+		MCBSP_WRITE(mcbsp, WAKEUPEN, 0);
+
+	spin_lock(&mcbsp->lock);
+	mcbsp->free = true;
+	mcbsp->reg_cache = NULL;
+err_kfree:
+	spin_unlock(&mcbsp->lock);
+	kfree(reg_cache);
+
+	return err;
+}
+
+static void omap_mcbsp_free(struct omap_mcbsp *mcbsp)
+{
+	void *reg_cache;
+
+	if(mcbsp->pdata->ops && mcbsp->pdata->ops->free)
+		mcbsp->pdata->ops->free(mcbsp->id - 1);
+
+	/* Disable wakeup behavior */
+	if (mcbsp->pdata->has_wakeup)
+		MCBSP_WRITE(mcbsp, WAKEUPEN, 0);
+
+	/* Disable interrupt requests */
+	if (mcbsp->irq) {
+		MCBSP_WRITE(mcbsp, IRQEN, 0);
+
+		free_irq(mcbsp->irq, (void *)mcbsp);
+	} else {
+		free_irq(mcbsp->rx_irq, (void *)mcbsp);
+		free_irq(mcbsp->tx_irq, (void *)mcbsp);
+	}
+
+	reg_cache = mcbsp->reg_cache;
+
+	/*
+	 * Select CLKS source from internal source unconditionally before
+	 * marking the McBSP port as free.
+	 * If the external clock source via MCBSP_CLKS pin has been selected the
+	 * system will refuse to enter idle if the CLKS pin source is not reset
+	 * back to internal source.
+	 */
+	if (!mcbsp_omap1())
+		omap2_mcbsp_set_clks_src(mcbsp, MCBSP_CLKS_PRCM_SRC);
+
+	spin_lock(&mcbsp->lock);
+	if (mcbsp->free)
+		dev_err(mcbsp->dev, "McBSP%d was not reserved\n", mcbsp->id);
+	else
+		mcbsp->free = true;
+	mcbsp->reg_cache = NULL;
+	spin_unlock(&mcbsp->lock);
+
+	kfree(reg_cache);
+}
+
+/*
+ * Here we start the McBSP, by enabling transmitter, receiver or both.
+ * If no transmitter or receiver is active prior calling, then sample-rate
+ * generator and frame sync are started.
+ */
+static void omap_mcbsp_start(struct omap_mcbsp *mcbsp, int stream)
+{
+	int tx = (stream == SNDRV_PCM_STREAM_PLAYBACK);
+	int rx = !tx;
+	int enable_srg = 0;
+	u16 w;
+
+	if (mcbsp->st_data)
+		omap_mcbsp_st_start(mcbsp);
+
+	/* Only enable SRG, if McBSP is master */
+	w = MCBSP_READ_CACHE(mcbsp, PCR0);
+	if (w & (FSXM | FSRM | CLKXM | CLKRM))
+		enable_srg = !((MCBSP_READ_CACHE(mcbsp, SPCR2) |
+				MCBSP_READ_CACHE(mcbsp, SPCR1)) & 1);
+
+	if (enable_srg) {
+		/* Start the sample generator */
+		w = MCBSP_READ_CACHE(mcbsp, SPCR2);
+		MCBSP_WRITE(mcbsp, SPCR2, w | (1 << 6));
+	}
+
+	/* Enable transmitter and receiver */
+	tx &= 1;
+	w = MCBSP_READ_CACHE(mcbsp, SPCR2);
+	MCBSP_WRITE(mcbsp, SPCR2, w | tx);
+
+	rx &= 1;
+	w = MCBSP_READ_CACHE(mcbsp, SPCR1);
+	MCBSP_WRITE(mcbsp, SPCR1, w | rx);
+
+	/*
+	 * Worst case: CLKSRG*2 = 8000khz: (1/8000) * 2 * 2 usec
+	 * REVISIT: 100us may give enough time for two CLKSRG, however
+	 * due to some unknown PM related, clock gating etc. reason it
+	 * is now at 500us.
+	 */
+	udelay(500);
+
+	if (enable_srg) {
+		/* Start frame sync */
+		w = MCBSP_READ_CACHE(mcbsp, SPCR2);
+		MCBSP_WRITE(mcbsp, SPCR2, w | (1 << 7));
+	}
+
+	if (mcbsp->pdata->has_ccr) {
+		/* Release the transmitter and receiver */
+		w = MCBSP_READ_CACHE(mcbsp, XCCR);
+		w &= ~(tx ? XDISABLE : 0);
+		MCBSP_WRITE(mcbsp, XCCR, w);
+		w = MCBSP_READ_CACHE(mcbsp, RCCR);
+		w &= ~(rx ? RDISABLE : 0);
+		MCBSP_WRITE(mcbsp, RCCR, w);
+	}
+
+	/* Dump McBSP Regs */
+	omap_mcbsp_dump_reg(mcbsp);
+}
+
+static void omap_mcbsp_stop(struct omap_mcbsp *mcbsp, int stream)
+{
+	int tx = (stream == SNDRV_PCM_STREAM_PLAYBACK);
+	int rx = !tx;
+	int idle;
+	u16 w;
+
+	/* Reset transmitter */
+	tx &= 1;
+	if (mcbsp->pdata->has_ccr) {
+		w = MCBSP_READ_CACHE(mcbsp, XCCR);
+		w |= (tx ? XDISABLE : 0);
+		MCBSP_WRITE(mcbsp, XCCR, w);
+	}
+	w = MCBSP_READ_CACHE(mcbsp, SPCR2);
+	MCBSP_WRITE(mcbsp, SPCR2, w & ~tx);
+
+	/* Reset receiver */
+	rx &= 1;
+	if (mcbsp->pdata->has_ccr) {
+		w = MCBSP_READ_CACHE(mcbsp, RCCR);
+		w |= (rx ? RDISABLE : 0);
+		MCBSP_WRITE(mcbsp, RCCR, w);
+	}
+	w = MCBSP_READ_CACHE(mcbsp, SPCR1);
+	MCBSP_WRITE(mcbsp, SPCR1, w & ~rx);
+
+	idle = !((MCBSP_READ_CACHE(mcbsp, SPCR2) |
+			MCBSP_READ_CACHE(mcbsp, SPCR1)) & 1);
+
+	if (idle) {
+		/* Reset the sample rate generator */
+		w = MCBSP_READ_CACHE(mcbsp, SPCR2);
+		MCBSP_WRITE(mcbsp, SPCR2, w & ~(1 << 6));
+	}
+
+	if (mcbsp->st_data)
+		omap_mcbsp_st_stop(mcbsp);
+}
+
+#define max_thres(m)			(mcbsp->pdata->buffer_size)
+#define valid_threshold(m, val)		((val) <= max_thres(m))
+#define THRESHOLD_PROP_BUILDER(prop)					\
+static ssize_t prop##_show(struct device *dev,				\
+			struct device_attribute *attr, char *buf)	\
+{									\
+	struct omap_mcbsp *mcbsp = dev_get_drvdata(dev);		\
+									\
+	return sysfs_emit(buf, "%u\n", mcbsp->prop);			\
+}									\
+									\
+static ssize_t prop##_store(struct device *dev,				\
+				struct device_attribute *attr,		\
+				const char *buf, size_t size)		\
+{									\
+	struct omap_mcbsp *mcbsp = dev_get_drvdata(dev);		\
+	unsigned long val;						\
+	int status;							\
+									\
+	status = kstrtoul(buf, 0, &val);				\
+	if (status)							\
+		return status;						\
+									\
+	if (!valid_threshold(mcbsp, val))				\
+		return -EDOM;						\
+									\
+	mcbsp->prop = val;						\
+	return size;							\
+}									\
+									\
+static DEVICE_ATTR_RW(prop)
+
+THRESHOLD_PROP_BUILDER(max_tx_thres);
+THRESHOLD_PROP_BUILDER(max_rx_thres);
+
+static const char * const dma_op_modes[] = {
+	"element", "threshold",
+};
+
+static ssize_t dma_op_mode_show(struct device *dev,
+				struct device_attribute *attr, char *buf)
+{
+	struct omap_mcbsp *mcbsp = dev_get_drvdata(dev);
+	int dma_op_mode, i = 0;
+	ssize_t len = 0;
+	const char * const *s;
+
+	dma_op_mode = mcbsp->dma_op_mode;
+
+	for (s = &dma_op_modes[i]; i < ARRAY_SIZE(dma_op_modes); s++, i++) {
+		if (dma_op_mode == i)
+			len += sysfs_emit_at(buf, len, "[%s] ", *s);
+		else
+			len += sysfs_emit_at(buf, len, "%s ", *s);
+	}
+	len += sysfs_emit_at(buf, len, "\n");
+
+	return len;
+}
+
+static ssize_t dma_op_mode_store(struct device *dev,
+				 struct device_attribute *attr, const char *buf,
+				 size_t size)
+{
+	struct omap_mcbsp *mcbsp = dev_get_drvdata(dev);
+	int i;
+
+	i = sysfs_match_string(dma_op_modes, buf);
+	if (i < 0)
+		return i;
+
+	spin_lock_irq(&mcbsp->lock);
+	if (!mcbsp->free) {
+		size = -EBUSY;
+		goto unlock;
+	}
+	mcbsp->dma_op_mode = i;
+
+unlock:
+	spin_unlock_irq(&mcbsp->lock);
+
+	return size;
+}
+
+static DEVICE_ATTR_RW(dma_op_mode);
+
+static const struct attribute *additional_attrs[] = {
+	&dev_attr_max_tx_thres.attr,
+	&dev_attr_max_rx_thres.attr,
+	&dev_attr_dma_op_mode.attr,
+	NULL,
+};
+
+static const struct attribute_group additional_attr_group = {
+	.attrs = (struct attribute **)additional_attrs,
+};
+
+/*
+ * McBSP1 and McBSP3 are directly mapped on 1610 and 1510.
+ * 730 has only 2 McBSP, and both of them are MPU peripherals.
+ */
+static int omap_mcbsp_init(struct platform_device *pdev)
+{
+	struct omap_mcbsp *mcbsp = platform_get_drvdata(pdev);
+	struct resource *res;
+	int ret;
+
+	spin_lock_init(&mcbsp->lock);
+	mcbsp->free = true;
+
+	res = platform_get_resource_byname(pdev, IORESOURCE_MEM, "mpu");
+	if (!res)
+		res = platform_get_resource(pdev, IORESOURCE_MEM, 0);
+
+	mcbsp->io_base = devm_ioremap_resource(&pdev->dev, res);
+	if (IS_ERR(mcbsp->io_base))
+		return PTR_ERR(mcbsp->io_base);
+
+	mcbsp->phys_base = res->start;
+	mcbsp->reg_cache_size = resource_size(res);
+
+	res = platform_get_resource_byname(pdev, IORESOURCE_MEM, "dma");
+	if (!res)
+		mcbsp->phys_dma_base = mcbsp->phys_base;
+	else
+		mcbsp->phys_dma_base = res->start;
+
+	/*
+	 * OMAP1, 2 uses two interrupt lines: TX, RX
+	 * OMAP2430, OMAP3 SoC have combined IRQ line as well.
+	 * OMAP4 and newer SoC only have the combined IRQ line.
+	 * Use the combined IRQ if available since it gives better debugging
+	 * possibilities.
+	 */
+	mcbsp->irq = platform_get_irq_byname(pdev, "common");
+	if (mcbsp->irq == -ENXIO) {
+		mcbsp->tx_irq = platform_get_irq_byname(pdev, "tx");
+
+		if (mcbsp->tx_irq == -ENXIO) {
+			mcbsp->irq = platform_get_irq(pdev, 0);
+			mcbsp->tx_irq = 0;
+		} else {
+			mcbsp->rx_irq = platform_get_irq_byname(pdev, "rx");
+			mcbsp->irq = 0;
+		}
+	}
+
+	if (!pdev->dev.of_node) {
+		res = platform_get_resource_byname(pdev, IORESOURCE_DMA, "tx");
+		if (!res) {
+			dev_err(&pdev->dev, "invalid tx DMA channel\n");
+			return -ENODEV;
+		}
+		mcbsp->dma_req[0] = res->start;
+		mcbsp->dma_data[0].filter_data = &mcbsp->dma_req[0];
+
+		res = platform_get_resource_byname(pdev, IORESOURCE_DMA, "rx");
+		if (!res) {
+			dev_err(&pdev->dev, "invalid rx DMA channel\n");
+			return -ENODEV;
+		}
+		mcbsp->dma_req[1] = res->start;
+		mcbsp->dma_data[1].filter_data = &mcbsp->dma_req[1];
+	} else {
+		mcbsp->dma_data[0].filter_data = "tx";
+		mcbsp->dma_data[1].filter_data = "rx";
+	}
+
+	mcbsp->dma_data[0].addr = omap_mcbsp_dma_reg_params(mcbsp,
+						SNDRV_PCM_STREAM_PLAYBACK);
+	mcbsp->dma_data[1].addr = omap_mcbsp_dma_reg_params(mcbsp,
+						SNDRV_PCM_STREAM_CAPTURE);
+
+	mcbsp->fclk = devm_clk_get(&pdev->dev, "fck");
+	if (IS_ERR(mcbsp->fclk)) {
+		ret = PTR_ERR(mcbsp->fclk);
+		dev_err(mcbsp->dev, "unable to get fck: %d\n", ret);
+		return ret;
+	}
+
+	mcbsp->dma_op_mode = MCBSP_DMA_MODE_ELEMENT;
+	if (mcbsp->pdata->buffer_size) {
+		/*
+		 * Initially configure the maximum thresholds to a safe value.
+		 * The McBSP FIFO usage with these values should not go under
+		 * 16 locations.
+		 * If the whole FIFO without safety buffer is used, than there
+		 * is a possibility that the DMA will be not able to push the
+		 * new data on time, causing channel shifts in runtime.
+		 */
+		mcbsp->max_tx_thres = max_thres(mcbsp) - 0x10;
+		mcbsp->max_rx_thres = max_thres(mcbsp) - 0x10;
+
+		ret = devm_device_add_group(mcbsp->dev, &additional_attr_group);
+		if (ret) {
+			dev_err(mcbsp->dev,
+				"Unable to create additional controls\n");
+			return ret;
+		}
+	}
+
+	return omap_mcbsp_st_init(pdev);
+}
+
+/*
+ * 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 = asoc_substream_to_rtd(substream);
+	struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0);
+	struct omap_mcbsp *mcbsp = snd_soc_dai_get_drvdata(cpu_dai);
+	int words;
+
+	/* No need to proceed further if McBSP does not have FIFO */
+	if (mcbsp->pdata->buffer_size == 0)
+		return;
+
+	/*
+	 * 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 (!snd_soc_dai_active(cpu_dai))
+		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);
+	int tx = (substream->stream == SNDRV_PCM_STREAM_PLAYBACK);
+	int stream1 = tx ? SNDRV_PCM_STREAM_PLAYBACK : SNDRV_PCM_STREAM_CAPTURE;
+	int stream2 = tx ? SNDRV_PCM_STREAM_CAPTURE : SNDRV_PCM_STREAM_PLAYBACK;
+
+	if (mcbsp->latency[stream2])
+		cpu_latency_qos_update_request(&mcbsp->pm_qos_req,
+					       mcbsp->latency[stream2]);
+	else if (mcbsp->latency[stream1])
+		cpu_latency_qos_remove_request(&mcbsp->pm_qos_req);
+
+	mcbsp->latency[stream1] = 0;
+
+	if (!snd_soc_dai_active(cpu_dai)) {
+		omap_mcbsp_free(mcbsp);
+		mcbsp->configured = 0;
+	}
+}
+
+static int omap_mcbsp_dai_prepare(struct snd_pcm_substream *substream,
+				  struct snd_soc_dai *cpu_dai)
+{
+	struct omap_mcbsp *mcbsp = snd_soc_dai_get_drvdata(cpu_dai);
+	struct pm_qos_request *pm_qos_req = &mcbsp->pm_qos_req;
+	int tx = (substream->stream == SNDRV_PCM_STREAM_PLAYBACK);
+	int stream1 = tx ? SNDRV_PCM_STREAM_PLAYBACK : SNDRV_PCM_STREAM_CAPTURE;
+	int stream2 = tx ? SNDRV_PCM_STREAM_CAPTURE : SNDRV_PCM_STREAM_PLAYBACK;
+	int latency = mcbsp->latency[stream2];
+
+	/* Prevent omap hardware from hitting off between FIFO fills */
+	if (!latency || mcbsp->latency[stream1] < latency)
+		latency = mcbsp->latency[stream1];
+
+	if (cpu_latency_qos_request_active(pm_qos_req))
+		cpu_latency_qos_update_request(pm_qos_req, latency);
+	else if (latency)
+		cpu_latency_qos_add_request(pm_qos_req, latency);
+
+	return 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);
+
+	switch (cmd) {
+	case SNDRV_PCM_TRIGGER_START:
+	case SNDRV_PCM_TRIGGER_RESUME:
+	case SNDRV_PCM_TRIGGER_PAUSE_RELEASE:
+		mcbsp->active++;
+		omap_mcbsp_start(mcbsp, substream->stream);
+		break;
+
+	case SNDRV_PCM_TRIGGER_STOP:
+	case SNDRV_PCM_TRIGGER_SUSPEND:
+	case SNDRV_PCM_TRIGGER_PAUSE_PUSH:
+		omap_mcbsp_stop(mcbsp, substream->stream);
+		mcbsp->active--;
+		break;
+	default:
+		return -EINVAL;
+	}
+
+	return 0;
+}
+
+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 = asoc_substream_to_rtd(substream);
+	struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0);
+	struct omap_mcbsp *mcbsp = snd_soc_dai_get_drvdata(cpu_dai);
+	u16 fifo_use;
+	snd_pcm_sframes_t delay;
+
+	/* No need to proceed further if McBSP does not have FIFO */
+	if (mcbsp->pdata->buffer_size == 0)
+		return 0;
+
+	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;
+	unsigned int buffer_size = mcbsp->pdata->buffer_size;
+
+	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 (buffer_size) {
+		int latency;
+
+		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;
+		}
+
+		latency = (buffer_size - pkt_size) / channels;
+		latency = latency * USEC_PER_SEC /
+			  (params->rate_num / params->rate_den);
+		mcbsp->latency[substream->stream] = latency;
+
+		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_CLOCK_PROVIDER_MASK;
+	if (master == SND_SOC_DAIFMT_BP_FP) {
+		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_CLOCK_PROVIDER_MASK) {
+	case SND_SOC_DAIFMT_BP_FP:
+		/* 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_BC_FP:
+		/* McBSP slave. FS clock as output */
+		regs->srgr2	|= FSGM;
+		regs->pcr0	|= FSXM | FSRM;
+		break;
+	case SND_SOC_DAIFMT_BC_FC:
+		/* 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)
+		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;
+		regs->pcr0	|= SCLKME;
+		/*
+		 * If McBSP is master but yet the CLKX/CLKR pin drives the SRG,
+		 * disable output on those pins. This enables to inject the
+		 * reference clock through CLKX/CLKR. For this to work
+		 * set_dai_sysclk() _needs_ to be called after set_dai_fmt().
+		 */
+		regs->pcr0	&= ~CLKXM;
+		break;
+	case OMAP_MCBSP_SYSCLK_CLKR_EXT:
+		regs->pcr0	|= SCLKME;
+		/* Disable ouput on CLKR pin in master mode */
+		regs->pcr0	&= ~CLKRM;
+		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,
+	.prepare	= omap_mcbsp_dai_prepare,
+	.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",
+	.legacy_dai_naming	= 1,
+};
+
+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;
+		struct omap_mcbsp_platform_data *pdata_quirk = pdata;
+		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;
+		if (pdata_quirk)
+			pdata->force_ick_on = pdata_quirk->force_ick_on;
+	} 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;
+
+	if (mcbsp->pdata->reg_size == 2) {
+		omap_mcbsp_dai.playback.formats = SNDRV_PCM_FMTBIT_S16_LE;
+		omap_mcbsp_dai.capture.formats = SNDRV_PCM_FMTBIT_S16_LE;
+	}
+
+	ret = devm_snd_soc_register_component(&pdev->dev,
+					      &omap_mcbsp_component,
+					      &omap_mcbsp_dai, 1);
+	if (ret)
+		return ret;
+
+	return sdma_pcm_platform_register(&pdev->dev, "tx", "rx");
+}
+
+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);
+
+	if (cpu_latency_qos_request_active(&mcbsp->pm_qos_req))
+		cpu_latency_qos_remove_request(&mcbsp->pm_qos_req);
+
+	return 0;
+}
+
+static struct platform_driver asoc_mcbsp_driver = {
+	.driver = {
+			.name = "omap-mcbsp",
+			.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");
diff --git a/sound/soc/ti/omap-mcbsp.h b/sound/soc/ti/omap-mcbsp.h
new file mode 100644
index 000000000..9fdba86ba
--- /dev/null
+++ b/sound/soc/ti/omap-mcbsp.h
@@ -0,0 +1,32 @@
+/* SPDX-License-Identifier: GPL-2.0-only */
+/*
+ * omap-mcbsp.h
+ *
+ * Copyright (C) 2008 Nokia Corporation
+ *
+ * Contact: Jarkko Nikula <jarkko.nikula@bitmer.com>
+ *          Peter Ujfalusi <peter.ujfalusi@ti.com>
+ */
+
+#ifndef __OMAP_MCBSP_H__
+#define __OMAP_MCBSP_H__
+
+#include <sound/dmaengine_pcm.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 /* __OMAP_MCBSP_H__ */
diff --git a/sound/soc/ti/omap-mcpdm.c b/sound/soc/ti/omap-mcpdm.c
new file mode 100644
index 000000000..0b18a7bfd
--- /dev/null
+++ b/sound/soc/ti/omap-mcpdm.c
@@ -0,0 +1,606 @@
+// SPDX-License-Identifier: GPL-2.0-only
+/*
+ * 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>
+ */
+
+#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 "omap-mcpdm.h"
+#include "sdma-pcm.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 pm_qos_request pm_qos_req;
+	int latency[2];
+
+	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;
+
+	/* pm state for suspend/resume handling */
+	int pm_active_count;
+
+	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)
+{
+	u32 ctrl = omap_mcpdm_read(mcpdm, MCPDM_REG_CTRL);
+
+	omap_mcpdm_write(mcpdm, MCPDM_REG_CTRL, ctrl | MCPDM_WD_EN);
+
+	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 (!snd_soc_dai_active(dai))
+		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);
+	int tx = (substream->stream == SNDRV_PCM_STREAM_PLAYBACK);
+	int stream1 = tx ? SNDRV_PCM_STREAM_PLAYBACK : SNDRV_PCM_STREAM_CAPTURE;
+	int stream2 = tx ? SNDRV_PCM_STREAM_CAPTURE : SNDRV_PCM_STREAM_PLAYBACK;
+
+	mutex_lock(&mcpdm->mutex);
+
+	if (!snd_soc_dai_active(dai)) {
+		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;
+		}
+	}
+
+	if (mcpdm->latency[stream2])
+		cpu_latency_qos_update_request(&mcpdm->pm_qos_req,
+					       mcpdm->latency[stream2]);
+	else if (mcpdm->latency[stream1])
+		cpu_latency_qos_remove_request(&mcpdm->pm_qos_req);
+
+	mcpdm->latency[stream1] = 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, latency;
+	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;
+		fallthrough;
+	case 4:
+		if (stream == SNDRV_PCM_STREAM_CAPTURE)
+			/* up to 3 channels for capture */
+			return -EINVAL;
+		link_mask |= 1 << 3;
+		fallthrough;
+	case 3:
+		link_mask |= 1 << 2;
+		fallthrough;
+	case 2:
+		link_mask |= 1 << 1;
+		fallthrough;
+	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;
+		latency = threshold;
+	} 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;
+		latency = (MCPDM_DN_THRES_MAX - threshold);
+	}
+
+	/*
+	 * The DMA must act to a DMA request within latency time (usec) to avoid
+	 * under/overflow
+	 */
+	mcpdm->latency[stream] = latency * USEC_PER_SEC / params_rate(params);
+
+	if (!mcpdm->latency[stream])
+		mcpdm->latency[stream] = 10;
+
+	/* 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);
+	struct pm_qos_request *pm_qos_req = &mcpdm->pm_qos_req;
+	int tx = (substream->stream == SNDRV_PCM_STREAM_PLAYBACK);
+	int stream1 = tx ? SNDRV_PCM_STREAM_PLAYBACK : SNDRV_PCM_STREAM_CAPTURE;
+	int stream2 = tx ? SNDRV_PCM_STREAM_CAPTURE : SNDRV_PCM_STREAM_PLAYBACK;
+	int latency = mcpdm->latency[stream2];
+
+	/* Prevent omap hardware from hitting off between FIFO fills */
+	if (!latency || mcpdm->latency[stream1] < latency)
+		latency = mcpdm->latency[stream1];
+
+	if (cpu_latency_qos_request_active(pm_qos_req))
+		cpu_latency_qos_update_request(pm_qos_req, latency);
+	else if (latency)
+		cpu_latency_qos_add_request(pm_qos_req, latency);
+
+	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 = request_irq(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);
+
+	free_irq(mcpdm->irq, (void *)mcpdm);
+	pm_runtime_disable(mcpdm->dev);
+
+	if (cpu_latency_qos_request_active(&mcpdm->pm_qos_req))
+		cpu_latency_qos_remove_request(&mcpdm->pm_qos_req);
+
+	return 0;
+}
+
+#ifdef CONFIG_PM_SLEEP
+static int omap_mcpdm_suspend(struct snd_soc_component *component)
+{
+	struct omap_mcpdm *mcpdm = snd_soc_component_get_drvdata(component);
+
+	if (snd_soc_component_active(component)) {
+		omap_mcpdm_stop(mcpdm);
+		omap_mcpdm_close_streams(mcpdm);
+	}
+
+	mcpdm->pm_active_count = 0;
+	while (pm_runtime_active(mcpdm->dev)) {
+		pm_runtime_put_sync(mcpdm->dev);
+		mcpdm->pm_active_count++;
+	}
+
+	return 0;
+}
+
+static int omap_mcpdm_resume(struct snd_soc_component *component)
+{
+	struct omap_mcpdm *mcpdm = snd_soc_component_get_drvdata(component);
+
+	if (mcpdm->pm_active_count) {
+		while (mcpdm->pm_active_count--)
+			pm_runtime_get_sync(mcpdm->dev);
+
+		if (snd_soc_component_active(component)) {
+			omap_mcpdm_open_streams(mcpdm);
+			omap_mcpdm_start(mcpdm);
+		}
+	}
+
+
+	return 0;
+}
+#else
+#define omap_mcpdm_suspend NULL
+#define omap_mcpdm_resume NULL
+#endif
+
+#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",
+	.suspend		= omap_mcpdm_suspend,
+	.resume			= omap_mcpdm_resume,
+	.legacy_dai_naming	= 1,
+};
+
+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(asoc_rtd_to_cpu(rtd, 0));
+
+	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 sdma_pcm_platform_register(&pdev->dev, "dn_link", "up_link");
+}
+
+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",
+		.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");
diff --git a/sound/soc/ti/omap-mcpdm.h b/sound/soc/ti/omap-mcpdm.h
new file mode 100644
index 000000000..e9956b4ef
--- /dev/null
+++ b/sound/soc/ti/omap-mcpdm.h
@@ -0,0 +1,93 @@
+/* SPDX-License-Identifier: GPL-2.0-only */
+/*
+ * omap-mcpdm.h
+ *
+ * Copyright (C) 2009 - 2011 Texas Instruments
+ *
+ * Contact: Misael Lopez Cruz <misael.lopez@ti.com>
+ */
+
+#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__ */
diff --git a/sound/soc/ti/omap-twl4030.c b/sound/soc/ti/omap-twl4030.c
new file mode 100644
index 000000000..950eec445
--- /dev/null
+++ b/sound/soc/ti/omap-twl4030.c
@@ -0,0 +1,343 @@
+// SPDX-License-Identifier: GPL-2.0-only
+/*
+ * omap-twl4030.c  --  SoC audio for TI SoC based boards with twl4030 codec
+ *
+ * Copyright (C) 2012 Texas Instruments Incorporated - https://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>)
+ */
+
+#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 = asoc_substream_to_rtd(substream);
+	unsigned int fmt;
+
+	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;
+	}
+
+	return snd_soc_runtime_set_dai_fmt(rtd, fmt);
+}
+
+static const 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_card *card = rtd->card;
+	struct snd_soc_dapm_context *dapm = &card->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_card_jack_new_pins(rtd->card, "Headset Jack",
+						 SND_JACK_HEADSET,
+						 &priv->hs_jack, hs_jack_pins,
+						 ARRAY_SIZE(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;
+}
+
+/* Digital audio interface glue - connects codec <--> CPU */
+SND_SOC_DAILINK_DEFS(hifi,
+	DAILINK_COMP_ARRAY(COMP_CPU("omap-mcbsp.2")),
+	DAILINK_COMP_ARRAY(COMP_CODEC("twl4030-codec", "twl4030-hifi")),
+	DAILINK_COMP_ARRAY(COMP_PLATFORM("omap-mcbsp.2")));
+
+SND_SOC_DAILINK_DEFS(voice,
+	DAILINK_COMP_ARRAY(COMP_CPU("omap-mcbsp.3")),
+	DAILINK_COMP_ARRAY(COMP_CODEC("twl4030-codec", "twl4030-voice")),
+	DAILINK_COMP_ARRAY(COMP_PLATFORM("omap-mcbsp.3")));
+
+static struct snd_soc_dai_link omap_twl4030_dai_links[] = {
+	{
+		.name = "TWL4030 HiFi",
+		.stream_name = "TWL4030 HiFi",
+		.init = omap_twl4030_init,
+		.ops = &omap_twl4030_ops,
+		SND_SOC_DAILINK_REG(hifi),
+	},
+	{
+		.name = "TWL4030 Voice",
+		.stream_name = "TWL4030 Voice",
+		.dai_fmt = SND_SOC_DAIFMT_DSP_A | SND_SOC_DAIFMT_IB_NF |
+			   SND_SOC_DAIFMT_CBM_CFM,
+		SND_SOC_DAILINK_REG(voice),
+	},
+};
+
+/* Audio machine driver */
+static struct snd_soc_card omap_twl4030_card = {
+	.owner = THIS_MODULE,
+	.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].cpus->dai_name  = NULL;
+		omap_twl4030_dai_links[0].cpus->of_node = dai_node;
+
+		omap_twl4030_dai_links[0].platforms->name  = NULL;
+		omap_twl4030_dai_links[0].platforms->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].cpus->dai_name  = NULL;
+			omap_twl4030_dai_links[1].cpus->of_node = dai_node;
+
+			omap_twl4030_dai_links[1].platforms->name  = NULL;
+			omap_twl4030_dai_links[1].platforms->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",
+		.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");
diff --git a/sound/soc/ti/omap3pandora.c b/sound/soc/ti/omap3pandora.c
new file mode 100644
index 000000000..a287e9747
--- /dev/null
+++ b/sound/soc/ti/omap3pandora.c
@@ -0,0 +1,305 @@
+// SPDX-License-Identifier: GPL-2.0-only
+/*
+ * omap3pandora.c  --  SoC audio for Pandora Handheld Console
+ *
+ * Author: Gražvydas Ignotas <notasas@gmail.com>
+ */
+
+#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 = asoc_substream_to_rtd(substream);
+	struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0);
+	struct snd_soc_dai *cpu_dai = asoc_rtd_to_cpu(rtd, 0);
+	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_dapm_context *dapm = &rtd->card->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_dapm_context *dapm = &rtd->card->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 const struct snd_soc_ops omap3pandora_ops = {
+	.hw_params = omap3pandora_hw_params,
+};
+
+/* Digital audio interface glue - connects codec <--> CPU */
+SND_SOC_DAILINK_DEFS(out,
+	DAILINK_COMP_ARRAY(COMP_CPU("omap-mcbsp.2")),
+	DAILINK_COMP_ARRAY(COMP_CODEC("twl4030-codec", "twl4030-hifi")),
+	DAILINK_COMP_ARRAY(COMP_PLATFORM("omap-mcbsp.2")));
+
+SND_SOC_DAILINK_DEFS(in,
+	DAILINK_COMP_ARRAY(COMP_CPU("omap-mcbsp.4")),
+	DAILINK_COMP_ARRAY(COMP_CODEC("twl4030-codec", "twl4030-hifi")),
+	DAILINK_COMP_ARRAY(COMP_PLATFORM("omap-mcbsp.4")));
+
+static struct snd_soc_dai_link omap3pandora_dai[] = {
+	{
+		.name = "PCM1773",
+		.stream_name = "HiFi Out",
+		.dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF |
+			   SND_SOC_DAIFMT_CBS_CFS,
+		.ops = &omap3pandora_ops,
+		.init = omap3pandora_out_init,
+		SND_SOC_DAILINK_REG(out),
+	}, {
+		.name = "TWL4030",
+		.stream_name = "Line/Mic In",
+		.dai_fmt = SND_SOC_DAIFMT_I2S | SND_SOC_DAIFMT_NB_NF |
+			   SND_SOC_DAIFMT_CBS_CFS,
+		.ops = &omap3pandora_ops,
+		.init = omap3pandora_in_init,
+		SND_SOC_DAILINK_REG(in),
+	}
+};
+
+/* 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");
diff --git a/sound/soc/ti/osk5912.c b/sound/soc/ti/osk5912.c
new file mode 100644
index 000000000..2790c8915
--- /dev/null
+++ b/sound/soc/ti/osk5912.c
@@ -0,0 +1,176 @@
+// SPDX-License-Identifier: GPL-2.0-only
+/*
+ * osk5912.c  --  SoC audio for OSK 5912
+ *
+ * Copyright (C) 2008 Mistral Solutions
+ *
+ * Contact: Arun KS  <arunks@mistralsolutions.com>
+ */
+
+#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_prepare_enable(tlv320aic23_mclk);
+}
+
+static void osk_shutdown(struct snd_pcm_substream *substream)
+{
+	clk_disable_unprepare(tlv320aic23_mclk);
+}
+
+static int osk_hw_params(struct snd_pcm_substream *substream,
+			 struct snd_pcm_hw_params *params)
+{
+	struct snd_soc_pcm_runtime *rtd = asoc_substream_to_rtd(substream);
+	struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0);
+	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 const 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 */
+SND_SOC_DAILINK_DEFS(aic23,
+	DAILINK_COMP_ARRAY(COMP_CPU("omap-mcbsp.1")),
+	DAILINK_COMP_ARRAY(COMP_CODEC("tlv320aic23-codec",
+				      "tlv320aic23-hifi")),
+	DAILINK_COMP_ARRAY(COMP_PLATFORM("omap-mcbsp.1")));
+
+static struct snd_soc_dai_link osk_dai = {
+	.name = "TLV320AIC23",
+	.stream_name = "AIC23",
+	.dai_fmt = SND_SOC_DAIFMT_DSP_B | SND_SOC_DAIFMT_NB_NF |
+		   SND_SOC_DAIFMT_CBM_CFM,
+	.ops = &osk_ops,
+	SND_SOC_DAILINK_REG(aic23),
+};
+
+/* 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");
diff --git a/sound/soc/ti/rx51.c b/sound/soc/ti/rx51.c
new file mode 100644
index 000000000..322c398d2
--- /dev/null
+++ b/sound/soc/ti/rx51.c
@@ -0,0 +1,481 @@
+// SPDX-License-Identifier: GPL-2.0-only
+/*
+ * 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>
+ */
+
+#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 <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;
+		fallthrough;
+	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 = asoc_substream_to_rtd(substream);
+	struct snd_soc_card *card = rtd->card;
+
+	snd_pcm_hw_constraint_single(runtime, SNDRV_PCM_HW_PARAM_CHANNELS, 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 = asoc_substream_to_rtd(substream);
+	struct snd_soc_dai *codec_dai = asoc_rtd_to_codec(rtd, 0);
+
+	/* Set the codec system clock for DAC and ADC */
+	return snd_soc_dai_set_sysclk(codec_dai, 0, 19200000,
+				      SND_SOC_CLOCK_IN);
+}
+
+static const 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.enumerated.item[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.enumerated.item[0])
+		return 0;
+
+	rx51_spk_func = ucontrol->value.enumerated.item[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_get_input(struct snd_kcontrol *kcontrol,
+			  struct snd_ctl_elem_value *ucontrol)
+{
+	ucontrol->value.enumerated.item[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.enumerated.item[0])
+		return 0;
+
+	rx51_dmic_func = ucontrol->value.enumerated.item[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.enumerated.item[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.enumerated.item[0])
+		return 0;
+
+	rx51_jack_func = ucontrol->value.enumerated.item[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", NULL),
+	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"},
+	{"Ext Spk", NULL, "HPLCOM"},
+	{"Ext Spk", NULL, "HPRCOM"},
+	{"FM Transmitter", NULL, "LLOUT"},
+	{"FM Transmitter", NULL, "RLOUT"},
+
+	{"Headphone Jack", NULL, "TPA6130A2 HPLEFT"},
+	{"Headphone Jack", NULL, "TPA6130A2 HPRIGHT"},
+	{"TPA6130A2 LEFTIN", NULL, "LLOUT"},
+	{"TPA6130A2 RIGHTIN", NULL, "RLOUT"},
+
+	{"DMic Rate 64", NULL, "DMic"},
+	{"DMic", NULL, "Mic Bias"},
+
+	{"b LINE2R", NULL, "MONO_LOUT"},
+	{"Earphone", NULL, "b HPLOUT"},
+
+	{"LINE1L", NULL, "HS Mic"},
+	{"HS Mic", NULL, "b Mic Bias"},
+};
+
+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_card *card = rtd->card;
+	struct rx51_audio_pdata *pdata = snd_soc_card_get_drvdata(card);
+	int err;
+
+	snd_soc_limit_volume(card, "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_card_jack_new(rtd->card, "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;
+}
+
+/* Digital audio interface glue - connects codec <--> CPU */
+SND_SOC_DAILINK_DEFS(aic34,
+	DAILINK_COMP_ARRAY(COMP_CPU("omap-mcbsp.2")),
+	DAILINK_COMP_ARRAY(COMP_CODEC("tlv320aic3x-codec.2-0018",
+				      "tlv320aic3x-hifi")),
+	DAILINK_COMP_ARRAY(COMP_PLATFORM("omap-mcbsp.2")));
+
+static struct snd_soc_dai_link rx51_dai[] = {
+	{
+		.name = "TLV320AIC34",
+		.stream_name = "AIC34",
+		.dai_fmt = SND_SOC_DAIFMT_DSP_A | SND_SOC_DAIFMT_IB_NF |
+			   SND_SOC_DAIFMT_CBM_CFM,
+		.init = rx51_aic34_init,
+		.ops = &rx51_ops,
+		SND_SOC_DAILINK_REG(aic34),
+	},
+};
+
+static struct snd_soc_aux_dev rx51_aux_dev[] = {
+	{
+		.dlc = COMP_AUX("tlv320aic3x-codec.2-0019"),
+	},
+	{
+		.dlc = COMP_AUX("tpa6130a2.2-0060"),
+	},
+};
+
+static struct snd_soc_codec_conf rx51_codec_conf[] = {
+	{
+		.dlc = COMP_CODEC_CONF("tlv320aic3x-codec.2-0019"),
+		.name_prefix = "b",
+	},
+	{
+		.dlc = COMP_CODEC_CONF("tpa6130a2.2-0060"),
+		.name_prefix = "TPA6130A2",
+	},
+};
+
+/* Audio card */
+static struct snd_soc_card rx51_sound_card = {
+	.name = "RX-51",
+	.owner = THIS_MODULE,
+	.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),
+	.fully_routed = true,
+
+	.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].cpus->dai_name = NULL;
+		rx51_dai[0].platforms->name = NULL;
+		rx51_dai[0].cpus->of_node = dai_node;
+		rx51_dai[0].platforms->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].codecs->name = NULL;
+		rx51_dai[0].codecs->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].dlc.name = NULL;
+		rx51_aux_dev[0].dlc.of_node = dai_node;
+		rx51_codec_conf[0].dlc.name = NULL;
+		rx51_codec_conf[0].dlc.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;
+		}
+		rx51_aux_dev[1].dlc.name = NULL;
+		rx51_aux_dev[1].dlc.of_node = dai_node;
+		rx51_codec_conf[1].dlc.name = NULL;
+		rx51_codec_conf[1].dlc.of_node = dai_node;
+	}
+
+	pdata = devm_kzalloc(&pdev->dev, sizeof(*pdata), GFP_KERNEL);
+	if (pdata == NULL)
+		return -ENOMEM;
+
+	snd_soc_card_set_drvdata(card, pdata);
+
+	pdata->tvout_selection_gpio = devm_gpiod_get(card->dev,
+						     "tvout-selection",
+						     GPIOD_OUT_LOW);
+	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);
+	}
+
+	pdata->jack_detection_gpio = devm_gpiod_get(card->dev,
+						    "jack-detection",
+						    GPIOD_ASIS);
+	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",
+					    GPIOD_OUT_HIGH);
+	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);
+	}
+
+	pdata->speaker_amp_gpio = devm_gpiod_get(card->dev,
+						 "speaker-amplifier",
+						 GPIOD_OUT_LOW);
+	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 = 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",
+		.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");
diff --git a/sound/soc/ti/sdma-pcm.c b/sound/soc/ti/sdma-pcm.c
new file mode 100644
index 000000000..9e7691103
--- /dev/null
+++ b/sound/soc/ti/sdma-pcm.c
@@ -0,0 +1,71 @@
+// SPDX-License-Identifier: GPL-2.0
+/*
+ *  Copyright (C) 2018 Texas Instruments Incorporated - https://www.ti.com
+ *  Author: Peter Ujfalusi <peter.ujfalusi@ti.com>
+ */
+
+#include <linux/device.h>
+#include <linux/module.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include <sound/dmaengine_pcm.h>
+
+#include "sdma-pcm.h"
+
+static const struct snd_pcm_hardware sdma_pcm_hardware = {
+	.info			= SNDRV_PCM_INFO_MMAP |
+				  SNDRV_PCM_INFO_MMAP_VALID |
+				  SNDRV_PCM_INFO_PAUSE | SNDRV_PCM_INFO_RESUME |
+				  SNDRV_PCM_INFO_NO_PERIOD_WAKEUP |
+				  SNDRV_PCM_INFO_INTERLEAVED,
+	.period_bytes_min	= 32,
+	.period_bytes_max	= 64 * 1024,
+	.buffer_bytes_max	= 128 * 1024,
+	.periods_min		= 2,
+	.periods_max		= 255,
+};
+
+static const struct snd_dmaengine_pcm_config sdma_dmaengine_pcm_config = {
+	.pcm_hardware = &sdma_pcm_hardware,
+	.prepare_slave_config = snd_dmaengine_pcm_prepare_slave_config,
+	.prealloc_buffer_size = 128 * 1024,
+};
+
+int sdma_pcm_platform_register(struct device *dev,
+			       char *txdmachan, char *rxdmachan)
+{
+	struct snd_dmaengine_pcm_config *config;
+	unsigned int flags = 0;
+
+	/* Standard names for the directions: 'tx' and 'rx' */
+	if (!txdmachan && !rxdmachan)
+		return devm_snd_dmaengine_pcm_register(dev,
+						&sdma_dmaengine_pcm_config, 0);
+
+	config = devm_kzalloc(dev, sizeof(*config), GFP_KERNEL);
+	if (!config)
+		return -ENOMEM;
+
+	*config = sdma_dmaengine_pcm_config;
+
+	if (!txdmachan || !rxdmachan) {
+		/* One direction only PCM */
+		flags |= SND_DMAENGINE_PCM_FLAG_HALF_DUPLEX;
+		if (!txdmachan) {
+			txdmachan = rxdmachan;
+			rxdmachan = NULL;
+		}
+	}
+
+	config->chan_names[0] = txdmachan;
+	config->chan_names[1] = rxdmachan;
+
+	return devm_snd_dmaengine_pcm_register(dev, config, flags);
+}
+EXPORT_SYMBOL_GPL(sdma_pcm_platform_register);
+
+MODULE_AUTHOR("Peter Ujfalusi <peter.ujfalusi@ti.com>");
+MODULE_DESCRIPTION("sDMA PCM ASoC platform driver");
+MODULE_LICENSE("GPL v2");
diff --git a/sound/soc/ti/sdma-pcm.h b/sound/soc/ti/sdma-pcm.h
new file mode 100644
index 000000000..c19efb4c0
--- /dev/null
+++ b/sound/soc/ti/sdma-pcm.h
@@ -0,0 +1,21 @@
+/* SPDX-License-Identifier: GPL-2.0 */
+/*
+ *  Copyright (C) 2018 Texas Instruments Incorporated - https://www.ti.com
+ *  Author: Peter Ujfalusi <peter.ujfalusi@ti.com>
+ */
+
+#ifndef __SDMA_PCM_H__
+#define __SDMA_PCM_H__
+
+#if IS_ENABLED(CONFIG_SND_SOC_TI_SDMA_PCM)
+int sdma_pcm_platform_register(struct device *dev,
+			       char *txdmachan, char *rxdmachan);
+#else
+static inline int sdma_pcm_platform_register(struct device *dev,
+					     char *txdmachan, char *rxdmachan)
+{
+	return -ENODEV;
+}
+#endif /* CONFIG_SND_SOC_TI_SDMA_PCM */
+
+#endif /* __SDMA_PCM_H__ */
diff --git a/sound/soc/ti/udma-pcm.c b/sound/soc/ti/udma-pcm.c
new file mode 100644
index 000000000..2ff0f518a
--- /dev/null
+++ b/sound/soc/ti/udma-pcm.c
@@ -0,0 +1,43 @@
+// SPDX-License-Identifier: GPL-2.0
+/*
+ *  Copyright (C) 2020 Texas Instruments Incorporated - https://www.ti.com
+ *  Author: Peter Ujfalusi <peter.ujfalusi@ti.com>
+ */
+
+#include <linux/module.h>
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include <sound/dmaengine_pcm.h>
+
+#include "udma-pcm.h"
+
+static const struct snd_pcm_hardware udma_pcm_hardware = {
+	.info			= SNDRV_PCM_INFO_MMAP |
+				  SNDRV_PCM_INFO_MMAP_VALID |
+				  SNDRV_PCM_INFO_PAUSE | SNDRV_PCM_INFO_RESUME |
+				  SNDRV_PCM_INFO_NO_PERIOD_WAKEUP |
+				  SNDRV_PCM_INFO_INTERLEAVED,
+	.buffer_bytes_max	= SIZE_MAX,
+	.period_bytes_min	= 32,
+	.period_bytes_max	= SZ_64K,
+	.periods_min		= 2,
+	.periods_max		= UINT_MAX,
+};
+
+static const struct snd_dmaengine_pcm_config udma_dmaengine_pcm_config = {
+	.pcm_hardware = &udma_pcm_hardware,
+	.prepare_slave_config = snd_dmaengine_pcm_prepare_slave_config,
+};
+
+int udma_pcm_platform_register(struct device *dev)
+{
+	return devm_snd_dmaengine_pcm_register(dev, &udma_dmaengine_pcm_config,
+					       0);
+}
+EXPORT_SYMBOL_GPL(udma_pcm_platform_register);
+
+MODULE_AUTHOR("Peter Ujfalusi <peter.ujfalusi@ti.com>");
+MODULE_DESCRIPTION("UDMA PCM ASoC platform driver");
+MODULE_LICENSE("GPL v2");
diff --git a/sound/soc/ti/udma-pcm.h b/sound/soc/ti/udma-pcm.h
new file mode 100644
index 000000000..9ed588fd7
--- /dev/null
+++ b/sound/soc/ti/udma-pcm.h
@@ -0,0 +1,18 @@
+/* SPDX-License-Identifier: GPL-2.0 */
+/*
+ *  Copyright (C) 2018 Texas Instruments Incorporated - https://www.ti.com
+ */
+
+#ifndef __UDMA_PCM_H__
+#define __UDMA_PCM_H__
+
+#if IS_ENABLED(CONFIG_SND_SOC_TI_UDMA_PCM)
+int udma_pcm_platform_register(struct device *dev);
+#else
+static inline int udma_pcm_platform_register(struct device *dev)
+{
+	return 0;
+}
+#endif /* CONFIG_SND_SOC_TI_UDMA_PCM */
+
+#endif /* __UDMA_PCM_H__ */