ASoC: OMAP4 ABE DSP - Add support for the OMAP4 ABE DSP

This patch adds the OMAP4 ABE platform driver. This driver defines and
exports control for the DSP Frontend and Backend routing.

Signed-off-by: Liam Girdwood <lrg@ti.com>

5 files changed, 2574 insertions, 0 deletions

diff --git a/include/sound/omap-abe-dsp.h b/include/sound/omap-abe-dsp.h
new file mode 100644
index 0000000..60c405d
--- /dev/null
+++ b/include/sound/omap-abe-dsp.h
@@ -0,0 +1,19 @@
+/*
+ * omap-aess  --  OMAP4 ABE DSP
+ *
+ * Author: Liam Girdwood <lrg@slimlogic.co.uk>
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2 as
+ * published by the Free Software Foundation.
+ */
+
+#ifndef _OMAP4_ABE_DSP_H
+#define _OMAP4_ABE_DSP_H
+
+struct omap4_abe_dsp_pdata {
+	/* Return context loss count due to PM states changing */
+        int (*get_context_loss_count)(struct device *dev);
+};
+
+#endif
diff --git a/sound/soc/omap/Kconfig b/sound/soc/omap/Kconfig
index 61441fd..43156b9 100644
--- a/sound/soc/omap/Kconfig
+++ b/sound/soc/omap/Kconfig
@@ -2,6 +2,10 @@ config SND_OMAP_SOC
 	tristate "SoC Audio for the Texas Instruments OMAP chips"
 	depends on ARCH_OMAP
 
+config SND_OMAP_SOC_ABE_DSP
+	tristate
+	select SND_DYNAMIC_MINORS
+
 config SND_OMAP_SOC_MCBSP
 	tristate
 	select OMAP_MCBSP
diff --git a/sound/soc/omap/Makefile b/sound/soc/omap/Makefile
index cafdbe6..0d90f64 100644
--- a/sound/soc/omap/Makefile
+++ b/sound/soc/omap/Makefile
@@ -3,11 +3,13 @@ snd-soc-omap-objs := omap-pcm.o
 snd-soc-omap-mcbsp-objs := omap-mcbsp.o
 snd-soc-omap-mcpdm-objs := omap-mcpdm.o
 snd-soc-omap-abe-objs := omap-abe.o
+snd-soc-omap-abe-dsp-objs := omap-abe-dsp.o
 
 obj-$(CONFIG_SND_OMAP_SOC) += snd-soc-omap.o
 obj-$(CONFIG_SND_OMAP_SOC_MCBSP) += snd-soc-omap-mcbsp.o
 obj-$(CONFIG_SND_OMAP_SOC_MCPDM) += snd-soc-omap-mcpdm.o
 obj-$(CONFIG_SND_OMAP_SOC_ABE) += snd-soc-omap-abe.o
+obj-$(CONFIG_SND_OMAP_SOC_ABE_DSP) += snd-soc-omap-abe-dsp.o abe/
 
 # OMAP Machine Support
 snd-soc-n810-objs := n810.o
diff --git a/sound/soc/omap/omap-abe-dsp.c b/sound/soc/omap/omap-abe-dsp.c
new file mode 100644
index 0000000..d2112ca
--- /dev/null
+++ b/sound/soc/omap/omap-abe-dsp.c
@@ -0,0 +1,2386 @@
+/*
+ * omap-abe-dsp.c
+ *
+ * This program is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU General Public License
+ * version 2 as published by the Free Software Foundation.
+ *
+ * This program is distributed in the hope that it will be useful, but
+ * WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA
+ * 02110-1301 USA
+ *
+ * Copyright (C) 2010 Texas Instruments Inc.
+ *
+ * Authors: Liam Girdwood <lrg@ti.com>
+ *          Misael Lopez Cruz <misael.lopez@ti.com>
+ *          Sebastien Guiriec <s-guiriec@ti.com>
+ *
+ */
+
+#include <linux/module.h>
+#include <linux/moduleparam.h>
+#include <linux/init.h>
+#include <linux/delay.h>
+#include <linux/pm.h>
+#include <linux/i2c.h>
+#include <linux/gpio.h>
+#include <linux/platform_device.h>
+#include <linux/workqueue.h>
+#include <linux/i2c/twl.h>
+#include <linux/clk.h>
+#include <linux/err.h>
+#include <linux/slab.h>
+#include <linux/pm_runtime.h>
+#include <linux/dma-mapping.h>
+#include <linux/wait.h>
+#include <linux/firmware.h>
+#include <linux/debugfs.h>
+
+#include <plat/omap_hwmod.h>
+#include <plat/omap_device.h>
+#include <plat/dma.h>
+
+#include <sound/core.h>
+#include <sound/pcm.h>
+#include <sound/pcm_params.h>
+#include <sound/soc.h>
+#include <sound/soc-dapm.h>
+#include <sound/initval.h>
+#include <sound/tlv.h>
+#include <sound/omap-abe-dsp.h>
+
+#include "omap-abe-dsp.h"
+#include "omap-abe.h"
+#include "abe/abe_main.h"
+#include "abe/port_mgr.h"
+
+#warning need omap_device_set_rate
+#define omap_device_set_rate(x, y, z)
+
+static const char *abe_memory_bank[5] = {
+	"dmem",
+	"cmem",
+	"smem",
+	"pmem",
+	"mpu"
+};
+
+
+/*
+ * ABE loadable coefficients.
+ * The coefficient and their mixer configurations are loaded with the firmware
+ * blob duing probe().
+ */
+
+struct coeff_config {
+	char name[ABE_COEFF_NAME_SIZE];
+	u32 count;
+	u32 coeff;
+	char texts[ABE_COEFF_NUM_TEXTS][ABE_COEFF_TEXT_SIZE];
+};
+
+/*
+ * ABE Firmware Header.
+ * The ABE firmware blob has a header that describes each data section. This
+ * way we can store coefficients etc in the firmware.
+ */
+struct fw_header {
+	u32 magic;			/* magic number */
+	u32 crc;			/* optional crc */
+	u32 firmware_size;	/* payload size */
+	u32 coeff_size;		/* payload size */
+	u32 coeff_version;	/* coefficent version */
+	u32 firmware_version;	/* min version of ABE firmware required */
+	u32 num_equ;		/* number of equalizers */
+};
+
+/*
+ * ABE private data.
+ */
+struct abe_data {
+	struct omap4_abe_dsp_pdata *abe_pdata;
+	struct device *dev;
+	struct snd_soc_platform *platform;
+	struct delayed_work delayed_work;
+	struct mutex mutex;
+	struct mutex opp_mutex;
+	struct clk *clk;
+	void __iomem *io_base[5];
+	int irq;
+	int opp;
+	int active;
+
+	/* coefficients */
+	struct fw_header hdr;
+	s32 *equ[ABE_MAX_EQU];
+	int equ_profile[ABE_MAX_EQU];
+	struct soc_enum equalizer_enum[ABE_MAX_EQU];
+	struct snd_kcontrol_new equalizer_control[ABE_MAX_EQU];
+	struct coeff_config *equ_texts;
+
+	/* DAPM mixer config - TODO: some of this can be replaced with HAL update */
+	u32 widget_opp[ABE_NUM_DAPM_REG + 1];
+
+	u16 router[16];
+	int loss_count;
+
+	struct snd_pcm_substream *ping_pong_substream;
+	int first_irq;
+
+	struct snd_pcm_substream *psubs;
+
+#ifdef CONFIG_DEBUG_FS
+	/* ABE runtime debug config */
+
+	/* its intended we can switch on/off individual debug items */
+	u32 dbg_format1; /* TODO: match flag names here to debug format flags */
+	u32 dbg_format2;
+	u32 dbg_format3;
+
+	u32 dbg_buffer_bytes;
+	u32 dbg_circular;
+	u32 dbg_buffer_msecs;  /* size of buffer in secs */
+	u32 dbg_elem_bytes;
+	dma_addr_t dbg_buffer_addr;
+	wait_queue_head_t wait;
+	int dbg_reader_offset;
+	int dbg_dma_offset;
+	int dbg_complete;
+	struct dentry *debugfs_root;
+	struct dentry *debugfs_fmt1;
+	struct dentry *debugfs_fmt2;
+	struct dentry *debugfs_fmt3;
+	struct dentry *debugfs_size;
+	struct dentry *debugfs_data;
+	struct dentry *debugfs_circ;
+	struct dentry *debugfs_elem_bytes;
+	struct dentry *debugfs_opp_level;
+	char *dbg_buffer;
+	struct omap_pcm_dma_data *dma_data;
+	int dma_ch;
+	int dma_req;
+#endif
+};
+
+static struct abe_data *the_abe;
+
+// TODO: map to the new version of HAL
+static unsigned int abe_dsp_read(struct snd_soc_platform *platform,
+		unsigned int reg)
+{
+	struct abe_data *abe = snd_soc_platform_get_drvdata(platform);
+
+	BUG_ON(reg > ABE_NUM_DAPM_REG);
+	return abe->widget_opp[reg];
+}
+
+static int abe_dsp_write(struct snd_soc_platform *platform, unsigned int reg,
+		unsigned int val)
+{
+	struct abe_data *abe = snd_soc_platform_get_drvdata(platform);
+
+	BUG_ON(reg > ABE_NUM_DAPM_REG);
+	abe->widget_opp[reg] = val;
+	return 0;
+}
+
+static void abe_irq_pingpong_subroutine(u32 *data)
+{
+	struct abe_data *abe = (struct abe_data *)data;
+	u32 dst, n_bytes;
+
+	abe_read_next_ping_pong_buffer(MM_DL_PORT, &dst, &n_bytes);
+	abe_set_ping_pong_buffer(MM_DL_PORT, n_bytes);
+
+	/* Do not call ALSA function for first IRQ */
+	if (the_abe->first_irq) {
+		the_abe->first_irq = 0;
+	} else {
+		if (the_abe->ping_pong_substream)
+			snd_pcm_period_elapsed(the_abe->ping_pong_substream);
+	}
+}
+
+static irqreturn_t abe_irq_handler(int irq, void *dev_id)
+{
+	struct abe_data *abe = dev_id;
+
+	/* TODO: handle underruns/overruns/errors */
+	pm_runtime_get_sync(abe->dev);
+	abe_clear_irq();  // TODO: why is IRQ not cleared after processing ?
+	abe_irq_processing();
+	pm_runtime_put_sync(abe->dev);
+	return IRQ_HANDLED;
+}
+
+// TODO: these should really be called internally since we will know the McPDM state
+void abe_dsp_pm_get(void)
+{
+	pm_runtime_get_sync(the_abe->dev);
+}
+EXPORT_SYMBOL_GPL(abe_dsp_pm_get);
+
+void abe_dsp_pm_put(void)
+{
+	pm_runtime_put_sync(the_abe->dev);
+}
+EXPORT_SYMBOL_GPL(abe_dsp_pm_put);
+
+void abe_dsp_shutdown(void)
+{
+	if (!the_abe->active && !abe_check_activity()) {
+		abe_set_opp_processing(ABE_OPP25);
+		the_abe->opp = 25;
+		abe_stop_event_generator();
+		udelay(250);
+		omap_device_set_rate(the_abe->dev, the_abe->dev, 0);
+	}
+}
+EXPORT_SYMBOL_GPL(abe_dsp_shutdown);
+
+/*
+ * These TLV settings will need fine tuned for each individual control
+ */
+
+/* Media DL1 volume control from -120 to 30 dB in 1 dB steps */
+static DECLARE_TLV_DB_SCALE(mm_dl1_tlv, -12000, 100, 3000);
+
+/* Media DL1 volume control from -120 to 30 dB in 1 dB steps */
+static DECLARE_TLV_DB_SCALE(tones_dl1_tlv, -12000, 100, 3000);
+
+/* Media DL1 volume control from -120 to 30 dB in 1 dB steps */
+static DECLARE_TLV_DB_SCALE(voice_dl1_tlv, -12000, 100, 3000);
+
+/* Media DL1 volume control from -120 to 30 dB in 1 dB steps */
+static DECLARE_TLV_DB_SCALE(capture_dl1_tlv, -12000, 100, 3000);
+
+/* Media DL2 volume control from -120 to 30 dB in 1 dB steps */
+static DECLARE_TLV_DB_SCALE(mm_dl2_tlv, -12000, 100, 3000);
+
+/* Media DL2 volume control from -120 to 30 dB in 1 dB steps */
+static DECLARE_TLV_DB_SCALE(tones_dl2_tlv, -12000, 100, 3000);
+
+/* Media DL2 volume control from -120 to 30 dB in 1 dB steps */
+static DECLARE_TLV_DB_SCALE(voice_dl2_tlv, -12000, 100, 3000);
+
+/* Media DL2 volume control from -120 to 30 dB in 1 dB steps */
+static DECLARE_TLV_DB_SCALE(capture_dl2_tlv, -12000, 100, 3000);
+
+/* SDT volume control from -120 to 30 dB in 1 dB steps */
+static DECLARE_TLV_DB_SCALE(sdt_ul_tlv, -12000, 100, 3000);
+
+/* SDT volume control from -120 to 30 dB in 1 dB steps */
+static DECLARE_TLV_DB_SCALE(sdt_dl_tlv, -12000, 100, 3000);
+
+/* AUDUL volume control from -120 to 30 dB in 1 dB steps */
+static DECLARE_TLV_DB_SCALE(audul_mm_tlv, -12000, 100, 3000);
+
+/* AUDUL volume control from -120 to 30 dB in 1 dB steps */
+static DECLARE_TLV_DB_SCALE(audul_tones_tlv, -12000, 100, 3000);
+
+/* AUDUL volume control from -120 to 30 dB in 1 dB steps */
+static DECLARE_TLV_DB_SCALE(audul_vx_ul_tlv, -12000, 100, 3000);
+
+/* AUDUL volume control from -120 to 30 dB in 1 dB steps */
+static DECLARE_TLV_DB_SCALE(audul_vx_dl_tlv, -12000, 100, 3000);
+
+/* VXREC volume control from -120 to 30 dB in 1 dB steps */
+static DECLARE_TLV_DB_SCALE(vxrec_mm_dl_tlv, -12000, 100, 3000);
+
+/* VXREC volume control from -120 to 30 dB in 1 dB steps */
+static DECLARE_TLV_DB_SCALE(vxrec_tones_tlv, -12000, 100, 3000);
+
+/* VXREC volume control from -120 to 30 dB in 1 dB steps */
+static DECLARE_TLV_DB_SCALE(vxrec_vx_dl_tlv, -12000, 100, 3000);
+
+/* VXREC volume control from -120 to 30 dB in 1 dB steps */
+static DECLARE_TLV_DB_SCALE(vxrec_vx_ul_tlv, -12000, 100, 3000);
+
+/* DMIC volume control from -120 to 30 dB in 1 dB steps */
+static DECLARE_TLV_DB_SCALE(dmic_tlv, -12000, 100, 3000);
+
+/* BT UL volume control from -120 to 30 dB in 1 dB steps */
+static DECLARE_TLV_DB_SCALE(btul_tlv, -12000, 100, 3000);
+
+/* AMIC volume control from -120 to 30 dB in 1 dB steps */
+static DECLARE_TLV_DB_SCALE(amic_tlv, -12000, 100, 3000);
+
+//TODO: we have to use the shift value atm to represent register id due to current HAL
+static int dl1_put_mixer(struct snd_kcontrol *kcontrol,
+	struct snd_ctl_elem_value *ucontrol)
+{
+	struct snd_soc_dapm_widget_list *wlist = snd_kcontrol_chip(kcontrol);
+	struct snd_soc_dapm_widget *widget = wlist->widgets[0];
+	struct soc_mixer_control *mc =
+		(struct soc_mixer_control *)kcontrol->private_value;
+
+	pm_runtime_get_sync(the_abe->dev);
+
+	// TODO: optimise all of these to call HAL abe_enable_gain(mixer, enable)
+	if (ucontrol->value.integer.value[0]) {
+		the_abe->widget_opp[mc->shift] = ucontrol->value.integer.value[0];
+		snd_soc_dapm_mixer_update_power(widget, kcontrol, 1);
+		abe_enable_gain(MIXDL1, mc->reg);
+	} else {
+		the_abe->widget_opp[mc->shift] = ucontrol->value.integer.value[0];
+		snd_soc_dapm_mixer_update_power(widget, kcontrol, 0);
+		abe_disable_gain(MIXDL1, mc->reg);
+	}
+	pm_runtime_put_sync(the_abe->dev);
+
+	return 1;
+}
+
+static int dl2_put_mixer(struct snd_kcontrol *kcontrol,
+	struct snd_ctl_elem_value *ucontrol)
+{
+	struct snd_soc_dapm_widget_list *wlist = snd_kcontrol_chip(kcontrol);
+	struct snd_soc_dapm_widget *widget = wlist->widgets[0];
+	struct soc_mixer_control *mc =
+		(struct soc_mixer_control *)kcontrol->private_value;
+
+	pm_runtime_get_sync(the_abe->dev);
+
+	if (ucontrol->value.integer.value[0]) {
+		the_abe->widget_opp[mc->shift] = ucontrol->value.integer.value[0];
+		snd_soc_dapm_mixer_update_power(widget, kcontrol, 1);
+		abe_enable_gain(MIXDL2, mc->reg);
+	} else {
+		the_abe->widget_opp[mc->shift] = ucontrol->value.integer.value[0];
+		snd_soc_dapm_mixer_update_power(widget, kcontrol, 0);
+		abe_disable_gain(MIXDL2, mc->reg);
+	}
+
+	pm_runtime_put_sync(the_abe->dev);
+	return 1;
+}
+
+static int audio_ul_put_mixer(struct snd_kcontrol *kcontrol,
+	struct snd_ctl_elem_value *ucontrol)
+{
+	struct snd_soc_dapm_widget_list *wlist = snd_kcontrol_chip(kcontrol);
+	struct snd_soc_dapm_widget *widget = wlist->widgets[0];
+	struct soc_mixer_control *mc =
+		(struct soc_mixer_control *)kcontrol->private_value;
+
+	pm_runtime_get_sync(the_abe->dev);
+
+	if (ucontrol->value.integer.value[0]) {
+		the_abe->widget_opp[mc->shift] = ucontrol->value.integer.value[0];
+		snd_soc_dapm_mixer_update_power(widget, kcontrol, 1);
+		abe_enable_gain(MIXAUDUL, mc->reg);
+	} else {
+		the_abe->widget_opp[mc->shift] = ucontrol->value.integer.value[0];
+		snd_soc_dapm_mixer_update_power(widget, kcontrol, 0);
+		abe_disable_gain(MIXAUDUL, mc->reg);
+	}
+	pm_runtime_put_sync(the_abe->dev);
+
+	return 1;
+}
+
+static int vxrec_put_mixer(struct snd_kcontrol *kcontrol,
+	struct snd_ctl_elem_value *ucontrol)
+{
+	struct snd_soc_dapm_widget_list *wlist = snd_kcontrol_chip(kcontrol);
+	struct snd_soc_dapm_widget *widget = wlist->widgets[0];
+	struct soc_mixer_control *mc =
+		(struct soc_mixer_control *)kcontrol->private_value;
+
+	pm_runtime_get_sync(the_abe->dev);
+
+	if (ucontrol->value.integer.value[0]) {
+		the_abe->widget_opp[mc->shift] = ucontrol->value.integer.value[0];
+		snd_soc_dapm_mixer_update_power(widget, kcontrol, 1);
+		abe_enable_gain(MIXVXREC, mc->reg);
+	} else {
+		the_abe->widget_opp[mc->shift] = ucontrol->value.integer.value[0];
+		snd_soc_dapm_mixer_update_power(widget, kcontrol, 0);
+		abe_disable_gain(MIXVXREC, mc->reg);
+	}
+	pm_runtime_put_sync(the_abe->dev);
+
+	return 1;
+}
+
+static int sdt_put_mixer(struct snd_kcontrol *kcontrol,
+	struct snd_ctl_elem_value *ucontrol)
+{
+	struct snd_soc_dapm_widget_list *wlist = snd_kcontrol_chip(kcontrol);
+	struct snd_soc_dapm_widget *widget = wlist->widgets[0];
+	struct soc_mixer_control *mc =
+		(struct soc_mixer_control *)kcontrol->private_value;
+
+	pm_runtime_get_sync(the_abe->dev);
+
+	if (ucontrol->value.integer.value[0]) {
+		the_abe->widget_opp[mc->shift] = ucontrol->value.integer.value[0];
+		snd_soc_dapm_mixer_update_power(widget, kcontrol, 1);
+		abe_enable_gain(MIXSDT, mc->reg);
+	} else {
+		the_abe->widget_opp[mc->shift] = ucontrol->value.integer.value[0];
+		snd_soc_dapm_mixer_update_power(widget, kcontrol, 0);
+		abe_disable_gain(MIXSDT, mc->reg);
+	}
+	pm_runtime_put_sync(the_abe->dev);
+
+	return 1;
+}
+
+static int abe_get_mixer(struct snd_kcontrol *kcontrol,
+	struct snd_ctl_elem_value *ucontrol)
+{
+	struct soc_mixer_control *mc =
+		(struct soc_mixer_control *)kcontrol->private_value;
+
+	ucontrol->value.integer.value[0] = the_abe->widget_opp[mc->shift];
+	return 0;
+}
+
+/* router IDs that match our mixer strings */
+static const abe_router_t router[] = {
+		ZERO_labelID, /* strangely this is not 0 */
+		DMIC1_L_labelID, DMIC1_R_labelID,
+		DMIC2_L_labelID, DMIC2_R_labelID,
+		DMIC3_L_labelID, DMIC3_R_labelID,
+		BT_UL_L_labelID, BT_UL_R_labelID,
+		MM_EXT_IN_L_labelID, MM_EXT_IN_R_labelID,
+		AMIC_L_labelID, AMIC_R_labelID,
+		VX_REC_L_labelID, VX_REC_R_labelID,
+};
+
+static int ul_mux_put_route(struct snd_kcontrol *kcontrol,
+	struct snd_ctl_elem_value *ucontrol)
+{
+	struct snd_soc_dapm_widget_list *wlist = snd_kcontrol_chip(kcontrol);
+	struct snd_soc_dapm_widget *widget = wlist->widgets[0];
+	struct soc_enum *e = (struct soc_enum *)kcontrol->private_value;
+	int mux = ucontrol->value.enumerated.item[0];
+	int reg = e->reg - ABE_MUX(0);
+
+	pm_runtime_get_sync(the_abe->dev);
+
+	if (mux > ABE_ROUTES_UL)
+		return 0;
+
+	// TODO: get all this via firmware
+	if (reg < 8) {
+		/* 0  .. 9   = MM_UL */
+		the_abe->router[reg] = router[mux];
+	} else if (reg < 12) {
+		/* 10 .. 11  = MM_UL2 */
+		/* 12 .. 13  = VX_UL */
+		the_abe->router[reg + 2] = router[mux];
+	}
+
+	/* 2nd arg here is unused */
+	abe_set_router_configuration(UPROUTE, 0, (u32 *)the_abe->router);
+
+	if (router[mux] != ZERO_labelID)
+		the_abe->widget_opp[e->reg] = e->shift_l;
+	else
+		the_abe->widget_opp[e->reg] = 0;
+
+	snd_soc_dapm_mux_update_power(widget, kcontrol, 1, mux, e);
+	pm_runtime_put_sync(the_abe->dev);
+
+	return 1;
+}
+
+static int ul_mux_get_route(struct snd_kcontrol *kcontrol,
+	struct snd_ctl_elem_value *ucontrol)
+{
+	struct soc_enum *e =
+		(struct soc_enum *)kcontrol->private_value;
+	int reg = e->reg - ABE_MUX(0), i, rval = 0;
+
+	// TODO: get all this via firmware
+	if (reg < 8) {
+		/* 0  .. 9   = MM_UL */
+		rval = the_abe->router[reg];
+	} else if (reg < 12) {
+		/* 10 .. 11  = MM_UL2 */
+		/* 12 .. 13  = VX_UL */
+		rval = the_abe->router[reg + 2];
+	}
+
+	for (i = 0; i < ARRAY_SIZE(router); i++) {
+		if (router[i] == rval) {
+			ucontrol->value.integer.value[0] = i;
+			return 0;
+		}
+	}
+
+	return 1;
+}
+
+
+static int abe_put_switch(struct snd_kcontrol *kcontrol,
+	struct snd_ctl_elem_value *ucontrol)
+{
+	struct snd_soc_dapm_widget_list *wlist = snd_kcontrol_chip(kcontrol);
+	struct snd_soc_dapm_widget *widget = wlist->widgets[0];
+	struct soc_mixer_control *mc =
+		(struct soc_mixer_control *)kcontrol->private_value;
+
+	pm_runtime_get_sync(the_abe->dev);
+
+	if (ucontrol->value.integer.value[0]) {
+		the_abe->widget_opp[mc->shift] = ucontrol->value.integer.value[0];
+		snd_soc_dapm_mixer_update_power(widget, kcontrol, 1);
+	} else {
+		the_abe->widget_opp[mc->shift] = ucontrol->value.integer.value[0];
+		snd_soc_dapm_mixer_update_power(widget, kcontrol, 0);
+	}
+	pm_runtime_put_sync(the_abe->dev);
+
+	return 1;
+}
+
+
+static int volume_put_sdt_mixer(struct snd_kcontrol *kcontrol,
+	struct snd_ctl_elem_value *ucontrol)
+{
+	struct soc_mixer_control *mc =
+		(struct soc_mixer_control *)kcontrol->private_value;
+
+	pm_runtime_get_sync(the_abe->dev);
+
+	abe_write_mixer(MIXSDT, abe_val_to_gain(ucontrol->value.integer.value[0]),
+				RAMP_0MS, mc->reg);
+	pm_runtime_put_sync(the_abe->dev);
+
+	return 1;
+}
+
+static int volume_put_audul_mixer(struct snd_kcontrol *kcontrol,
+	struct snd_ctl_elem_value *ucontrol)
+{
+	struct soc_mixer_control *mc =
+		(struct soc_mixer_control *)kcontrol->private_value;
+
+	pm_runtime_get_sync(the_abe->dev);
+	abe_write_mixer(MIXAUDUL, abe_val_to_gain(ucontrol->value.integer.value[0]),
+				RAMP_0MS, mc->reg);
+	pm_runtime_put_sync(the_abe->dev);
+
+	return 1;
+}
+
+static int volume_put_vxrec_mixer(struct snd_kcontrol *kcontrol,
+	struct snd_ctl_elem_value *ucontrol)
+{
+	struct soc_mixer_control *mc =
+		(struct soc_mixer_control *)kcontrol->private_value;
+
+	pm_runtime_get_sync(the_abe->dev);
+	abe_write_mixer(MIXVXREC, abe_val_to_gain(ucontrol->value.integer.value[0]),
+				RAMP_0MS, mc->reg);
+	pm_runtime_put_sync(the_abe->dev);
+
+	return 1;
+}
+
+static int volume_put_dl1_mixer(struct snd_kcontrol *kcontrol,
+	struct snd_ctl_elem_value *ucontrol)
+{
+	struct soc_mixer_control *mc =
+		(struct soc_mixer_control *)kcontrol->private_value;
+
+	pm_runtime_get_sync(the_abe->dev);
+	abe_write_mixer(MIXDL1, abe_val_to_gain(ucontrol->value.integer.value[0]),
+				RAMP_0MS, mc->reg);
+	pm_runtime_put_sync(the_abe->dev);
+
+	return 1;
+}
+
+static int volume_put_dl2_mixer(struct snd_kcontrol *kcontrol,
+	struct snd_ctl_elem_value *ucontrol)
+{
+	struct soc_mixer_control *mc =
+		(struct soc_mixer_control *)kcontrol->private_value;
+
+	pm_runtime_get_sync(the_abe->dev);
+	abe_write_mixer(MIXDL2, abe_val_to_gain(ucontrol->value.integer.value[0]),
+				RAMP_0MS, mc->reg);
+	pm_runtime_put_sync(the_abe->dev);
+
+	return 1;
+}
+
+static int volume_put_gain(struct snd_kcontrol *kcontrol,
+	struct snd_ctl_elem_value *ucontrol)
+{
+	struct soc_mixer_control *mc =
+		(struct soc_mixer_control *)kcontrol->private_value;
+
+	pm_runtime_get_sync(the_abe->dev);
+	abe_write_gain(mc->reg,
+		       abe_val_to_gain(ucontrol->value.integer.value[0]),
+		       RAMP_20MS, mc->shift);
+	abe_write_gain(mc->reg,
+		       -12000 + (ucontrol->value.integer.value[1] * 100),
+		       RAMP_20MS, mc->rshift);
+	pm_runtime_put_sync(the_abe->dev);
+
+	return 1;
+}
+
+static int volume_get_dl1_mixer(struct snd_kcontrol *kcontrol,
+	struct snd_ctl_elem_value *ucontrol)
+{
+	struct soc_mixer_control *mc =
+		(struct soc_mixer_control *)kcontrol->private_value;
+	u32 val;
+
+	pm_runtime_get_sync(the_abe->dev);
+	abe_read_mixer(MIXDL1, &val, mc->reg);
+	ucontrol->value.integer.value[0] = abe_gain_to_val(val);
+	pm_runtime_put_sync(the_abe->dev);
+
+	return 0;
+}
+
+static int volume_get_dl2_mixer(struct snd_kcontrol *kcontrol,
+	struct snd_ctl_elem_value *ucontrol)
+{
+	struct soc_mixer_control *mc =
+		(struct soc_mixer_control *)kcontrol->private_value;
+	u32 val;
+
+	pm_runtime_get_sync(the_abe->dev);
+	abe_read_mixer(MIXDL2, &val, mc->reg);
+	ucontrol->value.integer.value[0] = abe_gain_to_val(val);
+	pm_runtime_put_sync(the_abe->dev);
+
+	return 0;
+}
+
+static int volume_get_audul_mixer(struct snd_kcontrol *kcontrol,
+	struct snd_ctl_elem_value *ucontrol)
+{
+	struct soc_mixer_control *mc =
+		(struct soc_mixer_control *)kcontrol->private_value;
+	u32 val;
+
+	pm_runtime_get_sync(the_abe->dev);
+	abe_read_mixer(MIXAUDUL, &val, mc->reg);
+	ucontrol->value.integer.value[0] = abe_gain_to_val(val);
+	pm_runtime_put_sync(the_abe->dev);
+
+	return 0;
+}
+
+static int volume_get_vxrec_mixer(struct snd_kcontrol *kcontrol,
+	struct snd_ctl_elem_value *ucontrol)
+{
+	struct soc_mixer_control *mc =
+		(struct soc_mixer_control *)kcontrol->private_value;
+	u32 val;
+
+	pm_runtime_get_sync(the_abe->dev);
+	abe_read_mixer(MIXVXREC, &val, mc->reg);
+	ucontrol->value.integer.value[0] = abe_gain_to_val(val);
+	pm_runtime_put_sync(the_abe->dev);
+
+	return 0;
+}
+
+static int volume_get_sdt_mixer(struct snd_kcontrol *kcontrol,
+	struct snd_ctl_elem_value *ucontrol)
+{
+	struct soc_mixer_control *mc =
+		(struct soc_mixer_control *)kcontrol->private_value;
+	u32 val;
+
+	pm_runtime_get_sync(the_abe->dev);
+	abe_read_mixer(MIXSDT, &val, mc->reg);
+	ucontrol->value.integer.value[0] = abe_gain_to_val(val);
+	pm_runtime_put_sync(the_abe->dev);
+
+	return 0;
+}
+
+static int volume_get_gain(struct snd_kcontrol *kcontrol,
+	struct snd_ctl_elem_value *ucontrol)
+{
+	struct soc_mixer_control *mc =
+		(struct soc_mixer_control *)kcontrol->private_value;
+	u32 val;
+
+	pm_runtime_get_sync(the_abe->dev);
+	abe_read_gain(mc->reg, &val, mc->shift);
+	ucontrol->value.integer.value[0] = abe_gain_to_val(val);
+	abe_read_gain(mc->reg, &val, mc->rshift);
+	ucontrol->value.integer.value[1] = abe_gain_to_val(val);
+	pm_runtime_put_sync(the_abe->dev);
+
+	return 0;
+}
+
+static int abe_get_equalizer(struct snd_kcontrol *kcontrol,
+			struct snd_ctl_elem_value *ucontrol)
+{
+#if defined(CONFIG_SND_OMAP_SOC_ABE_DSP_MODULE)
+	struct soc_enum *eqc = (struct soc_enum *)kcontrol->private_value;
+
+	ucontrol->value.integer.value[0] = the_abe->equ_profile[eqc->reg];
+#endif
+	return 0;
+}
+
+static int abe_put_equalizer(struct snd_kcontrol *kcontrol,
+			struct snd_ctl_elem_value *ucontrol)
+{
+#if defined(CONFIG_SND_OMAP_SOC_ABE_DSP_MODULE)
+	struct soc_enum *eqc = (struct soc_enum *)kcontrol->private_value;
+	u16 val = ucontrol->value.enumerated.item[0];
+	abe_equ_t equ_params;
+	int size;
+
+	if (val >= the_abe->hdr.num_equ)
+		return -EINVAL;
+
+	equ_params.equ_length = the_abe->equ_texts[eqc->reg].coeff;
+	size = the_abe->equ_texts[eqc->reg].coeff * sizeof(s32);
+	memcpy(equ_params.coef.type1, the_abe->equ[eqc->reg] + val * size, size);
+	the_abe->equ_profile[eqc->reg] = val;
+
+	pm_runtime_get_sync(the_abe->dev);
+	abe_write_equalizer(eqc->reg, &equ_params);
+	pm_runtime_put_sync(the_abe->dev);
+#endif
+	return 1;
+}
+
+int snd_soc_info_enum_ext1(struct snd_kcontrol *kcontrol,
+	struct snd_ctl_elem_info *uinfo)
+{
+	struct soc_enum *e = (struct soc_enum *)kcontrol->private_value;
+
+	uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
+	uinfo->count = 1;
+	uinfo->value.enumerated.items = e->max;
+
+	if (uinfo->value.enumerated.item > e->max - 1)
+		uinfo->value.enumerated.item = e->max - 1;
+	strcpy(uinfo->value.enumerated.name,
+		snd_soc_get_enum_text(e, uinfo->value.enumerated.item));
+
+	return 0;
+}
+
+static const char *route_ul_texts[] = {
+	"None", "DMic0L", "DMic0R", "DMic1L", "DMic1R", "DMic2L", "DMic2R",
+	"BT Left", "BT Right", "MMExt Left", "MMExt Right", "AMic0", "AMic1",
+	"VX Left", "VX Right"
+};
+
+/* ROUTE_UL Mux table */
+static const struct soc_enum abe_enum[] = {
+		SOC_ENUM_SINGLE(MUX_MM_UL10, 0, 15, route_ul_texts),
+		SOC_ENUM_SINGLE(MUX_MM_UL11, 0, 15, route_ul_texts),
+		SOC_ENUM_SINGLE(MUX_MM_UL12, 0, 15, route_ul_texts),
+		SOC_ENUM_SINGLE(MUX_MM_UL13, 0, 15, route_ul_texts),
+		SOC_ENUM_SINGLE(MUX_MM_UL14, 0, 15, route_ul_texts),
+		SOC_ENUM_SINGLE(MUX_MM_UL15, 0, 15, route_ul_texts),
+		SOC_ENUM_SINGLE(MUX_MM_UL16, 0, 15, route_ul_texts),
+		SOC_ENUM_SINGLE(MUX_MM_UL17, 0, 15, route_ul_texts),
+		SOC_ENUM_SINGLE(MUX_MM_UL20, 0, 15, route_ul_texts),
+		SOC_ENUM_SINGLE(MUX_MM_UL21, 0, 15, route_ul_texts),
+		SOC_ENUM_SINGLE(MUX_VX_UL0, 0, 15, route_ul_texts),
+		SOC_ENUM_SINGLE(MUX_VX_UL1, 0, 15, route_ul_texts),
+};
+
+static const struct snd_kcontrol_new mm_ul00_control =
+	SOC_DAPM_ENUM_EXT("Route", abe_enum[0],
+	ul_mux_get_route, ul_mux_put_route);
+
+static const struct snd_kcontrol_new mm_ul01_control =
+	SOC_DAPM_ENUM_EXT("Route", abe_enum[1],
+	ul_mux_get_route, ul_mux_put_route);
+
+static const struct snd_kcontrol_new mm_ul02_control =
+	SOC_DAPM_ENUM_EXT("Route", abe_enum[2],
+	ul_mux_get_route, ul_mux_put_route);
+
+static const struct snd_kcontrol_new mm_ul03_control =
+	SOC_DAPM_ENUM_EXT("Route", abe_enum[3],
+	ul_mux_get_route, ul_mux_put_route);
+
+static const struct snd_kcontrol_new mm_ul04_control =
+	SOC_DAPM_ENUM_EXT("Route", abe_enum[4],
+	ul_mux_get_route, ul_mux_put_route);
+
+static const struct snd_kcontrol_new mm_ul05_control =
+	SOC_DAPM_ENUM_EXT("Route", abe_enum[5],
+	ul_mux_get_route, ul_mux_put_route);
+
+static const struct snd_kcontrol_new mm_ul06_control =
+	SOC_DAPM_ENUM_EXT("Route", abe_enum[6],
+	ul_mux_get_route, ul_mux_put_route);
+
+static const struct snd_kcontrol_new mm_ul07_control =
+	SOC_DAPM_ENUM_EXT("Route", abe_enum[7],
+	ul_mux_get_route, ul_mux_put_route);
+
+static const struct snd_kcontrol_new mm_ul10_control =
+	SOC_DAPM_ENUM_EXT("Route", abe_enum[8],
+	ul_mux_get_route, ul_mux_put_route);
+
+static const struct snd_kcontrol_new mm_ul11_control =
+	SOC_DAPM_ENUM_EXT("Route", abe_enum[9],
+	ul_mux_get_route, ul_mux_put_route);
+
+static const struct snd_kcontrol_new mm_vx0_control =
+	SOC_DAPM_ENUM_EXT("Route", abe_enum[10],
+	ul_mux_get_route, ul_mux_put_route);
+
+static const struct snd_kcontrol_new mm_vx1_control =
+	SOC_DAPM_ENUM_EXT("Route", abe_enum[11],
+	ul_mux_get_route, ul_mux_put_route);
+
+/* DL1 mixer paths */
+static const struct snd_kcontrol_new dl1_mixer_controls[] = {
+	SOC_SINGLE_EXT("Tones", MIX_DL1_INPUT_TONES, MIX_DL1_TONES, 1, 0,
+		abe_get_mixer, dl1_put_mixer),
+	SOC_SINGLE_EXT("Voice", MIX_DL1_INPUT_VX_DL, MIX_DL1_VOICE, 1, 0,
+		abe_get_mixer, dl1_put_mixer),
+	SOC_SINGLE_EXT("Capture", MIX_DL1_INPUT_MM_UL2, MIX_DL1_CAPTURE, 1, 0,
+		abe_get_mixer, dl1_put_mixer),
+	SOC_SINGLE_EXT("Multimedia", MIX_DL1_INPUT_MM_DL, MIX_DL1_MEDIA, 1, 0,
+		abe_get_mixer, dl1_put_mixer),
+};
+
+/* DL2 mixer paths */
+static const struct snd_kcontrol_new dl2_mixer_controls[] = {
+	SOC_SINGLE_EXT("Tones", MIX_DL2_INPUT_TONES, MIX_DL2_TONES, 1, 0,
+		abe_get_mixer, dl2_put_mixer),
+	SOC_SINGLE_EXT("Voice", MIX_DL2_INPUT_VX_DL, MIX_DL2_VOICE, 1, 0,
+		abe_get_mixer, dl2_put_mixer),
+	SOC_SINGLE_EXT("Capture", MIX_DL2_INPUT_MM_UL2, MIX_DL2_CAPTURE, 1, 0,
+		abe_get_mixer, dl2_put_mixer),
+	SOC_SINGLE_EXT("Multimedia", MIX_DL2_INPUT_MM_DL, MIX_DL2_MEDIA, 1, 0,
+		abe_get_mixer, dl2_put_mixer),
+};
+
+/* AUDUL ("Voice Capture Mixer") mixer paths */
+static const struct snd_kcontrol_new audio_ul_mixer_controls[] = {
+	SOC_SINGLE_EXT("Tones Playback", MIX_AUDUL_INPUT_TONES, MIX_AUDUL_TONES, 1, 0,
+		abe_get_mixer, audio_ul_put_mixer),
+	SOC_SINGLE_EXT("Media Playback", MIX_AUDUL_INPUT_MM_DL, MIX_AUDUL_MEDIA, 1, 0,
+		abe_get_mixer, audio_ul_put_mixer),
+	SOC_SINGLE_EXT("Capture", MIX_AUDUL_INPUT_UPLINK, MIX_AUDUL_CAPTURE, 1, 0,
+		abe_get_mixer, audio_ul_put_mixer),
+};
+
+/* VXREC ("Capture Mixer")  mixer paths */
+static const struct snd_kcontrol_new vx_rec_mixer_controls[] = {
+	SOC_SINGLE_EXT("Tones", MIX_VXREC_INPUT_TONES, MIX_VXREC_TONES, 1, 0,
+		abe_get_mixer, vxrec_put_mixer),
+	SOC_SINGLE_EXT("Voice Playback", MIX_VXREC_INPUT_VX_DL,
+		MIX_VXREC_VOICE_PLAYBACK, 1, 0, abe_get_mixer, vxrec_put_mixer),
+	SOC_SINGLE_EXT("Voice Capture", MIX_VXREC_INPUT_VX_UL,
+		MIX_VXREC_VOICE_CAPTURE, 1, 0, abe_get_mixer, vxrec_put_mixer),
+	SOC_SINGLE_EXT("Media Playback", MIX_VXREC_INPUT_MM_DL,
+		MIX_VXREC_MEDIA, 1, 0, abe_get_mixer, vxrec_put_mixer),
+};
+
+/* SDT ("Sidetone Mixer") mixer paths */
+static const struct snd_kcontrol_new sdt_mixer_controls[] = {
+	SOC_SINGLE_EXT("Capture", MIX_SDT_INPUT_UP_MIXER, MIX_SDT_CAPTURE, 1, 0,
+		abe_get_mixer, sdt_put_mixer),
+	SOC_SINGLE_EXT("Playback", MIX_SDT_INPUT_DL1_MIXER, MIX_SDT_PLAYBACK, 1, 0,
+		abe_get_mixer, sdt_put_mixer),
+};
+
+/* Virtual PDM_DL Switch */
+static const struct snd_kcontrol_new pdm_dl1_switch_controls =
+	SOC_SINGLE_EXT("Switch", ABE_VIRTUAL_SWITCH, MIX_SWITCH_PDM_DL, 1, 0,
+			abe_get_mixer, abe_put_switch);
+
+/* Virtual BT_VX_DL Switch */
+static const struct snd_kcontrol_new bt_vx_dl_switch_controls =
+	SOC_SINGLE_EXT("Switch", ABE_VIRTUAL_SWITCH, MIX_SWITCH_BT_VX_DL, 1, 0,
+			abe_get_mixer, abe_put_switch);
+
+/* Virtual MM_EXT_DL Switch */
+static const struct snd_kcontrol_new mm_ext_dl_switch_controls =
+	SOC_SINGLE_EXT("Switch", ABE_VIRTUAL_SWITCH, MIX_SWITCH_MM_EXT_DL, 1, 0,
+			abe_get_mixer, abe_put_switch);
+
+static const struct snd_kcontrol_new abe_controls[] = {
+	/* DL1 mixer gains */
+	SOC_SINGLE_EXT_TLV("DL1 Media Playback Volume",
+		MIX_DL1_INPUT_MM_DL, 0, 149, 0,
+		volume_get_dl1_mixer, volume_put_dl1_mixer, mm_dl1_tlv),
+	SOC_SINGLE_EXT_TLV("DL1 Tones Playback Volume",
+		MIX_DL1_INPUT_TONES, 0, 149, 0,
+		volume_get_dl1_mixer, volume_put_dl1_mixer, tones_dl1_tlv),
+	SOC_SINGLE_EXT_TLV("DL1 Voice Playback Volume",
+		MIX_DL1_INPUT_VX_DL, 0, 149, 0,
+		volume_get_dl1_mixer, volume_put_dl1_mixer, voice_dl1_tlv),
+	SOC_SINGLE_EXT_TLV("DL1 Capture Playback Volume",
+		MIX_DL1_INPUT_MM_UL2, 0, 149, 0,
+		volume_get_dl1_mixer, volume_put_dl1_mixer, capture_dl1_tlv),
+
+	/* DL2 mixer gains */
+	SOC_SINGLE_EXT_TLV("DL2 Media Playback Volume",
+		MIX_DL2_INPUT_MM_DL, 0, 149, 0,
+		volume_get_dl2_mixer, volume_put_dl2_mixer, mm_dl2_tlv),
+	SOC_SINGLE_EXT_TLV("DL2 Tones Playback Volume",
+		MIX_DL2_INPUT_TONES, 0, 149, 0,
+		volume_get_dl2_mixer, volume_put_dl2_mixer, tones_dl2_tlv),
+	SOC_SINGLE_EXT_TLV("DL2 Voice Playback Volume",
+		MIX_DL2_INPUT_VX_DL, 0, 149, 0,
+		volume_get_dl2_mixer, volume_put_dl2_mixer, voice_dl2_tlv),
+	SOC_SINGLE_EXT_TLV("DL2 Capture Playback Volume",
+		MIX_DL2_INPUT_MM_UL2, 0, 149, 0,
+		volume_get_dl2_mixer, volume_put_dl2_mixer, capture_dl2_tlv),
+
+	/* VXREC mixer gains */
+	SOC_SINGLE_EXT_TLV("VXREC Media Volume",
+		MIX_VXREC_INPUT_MM_DL, 0, 149, 0,
+		volume_get_vxrec_mixer, volume_put_vxrec_mixer, vxrec_mm_dl_tlv),
+	SOC_SINGLE_EXT_TLV("VXREC Tones Volume",
+		MIX_VXREC_INPUT_TONES, 0, 149, 0,
+		volume_get_vxrec_mixer, volume_put_vxrec_mixer, vxrec_tones_tlv),
+	SOC_SINGLE_EXT_TLV("VXREC Voice DL Volume",
+		MIX_VXREC_INPUT_VX_UL, 0, 149, 0,
+		volume_get_vxrec_mixer, volume_put_vxrec_mixer, vxrec_vx_dl_tlv),
+	SOC_SINGLE_EXT_TLV("VXREC Voice UL Volume",
+		MIX_VXREC_INPUT_VX_DL, 0, 149, 0,
+		volume_get_vxrec_mixer, volume_put_vxrec_mixer, vxrec_vx_ul_tlv),
+
+	/* AUDUL mixer gains */
+	SOC_SINGLE_EXT_TLV("AUDUL Media Volume",
+		MIX_AUDUL_INPUT_MM_DL, 0, 149, 0,
+		volume_get_audul_mixer, volume_put_audul_mixer, audul_mm_tlv),
+	SOC_SINGLE_EXT_TLV("AUDUL Tones Volume",
+		MIX_AUDUL_INPUT_TONES, 0, 149, 0,
+		volume_get_audul_mixer, volume_put_audul_mixer, audul_tones_tlv),
+	SOC_SINGLE_EXT_TLV("AUDUL Voice UL Volume",
+		MIX_AUDUL_INPUT_UPLINK, 0, 149, 0,
+		volume_get_audul_mixer, volume_put_audul_mixer, audul_vx_ul_tlv),
+	SOC_SINGLE_EXT_TLV("AUDUL Voice DL Volume",
+		MIX_AUDUL_INPUT_VX_DL, 0, 149, 0,
+		volume_get_audul_mixer, volume_put_audul_mixer, audul_vx_dl_tlv),
+
+	/* SDT mixer gains */
+	SOC_SINGLE_EXT_TLV("SDT UL Volume",
+		MIX_SDT_INPUT_UP_MIXER, 0, 149, 0,
+		volume_get_sdt_mixer, volume_put_sdt_mixer, sdt_ul_tlv),
+	SOC_SINGLE_EXT_TLV("SDT DL Volume",
+		MIX_SDT_INPUT_DL1_MIXER, 0, 149, 0,
+		volume_get_sdt_mixer, volume_put_sdt_mixer, sdt_dl_tlv),
+
+	/* DMIC gains */
+	SOC_DOUBLE_EXT_TLV("DMIC1 UL Volume",
+		GAINS_DMIC1, GAIN_LEFT_OFFSET, GAIN_RIGHT_OFFSET, 149, 0,
+		volume_get_gain, volume_put_gain, dmic_tlv),
+
+	SOC_DOUBLE_EXT_TLV("DMIC2 UL Volume",
+		GAINS_DMIC2, GAIN_LEFT_OFFSET, GAIN_RIGHT_OFFSET, 149, 0,
+		volume_get_gain, volume_put_gain, dmic_tlv),
+
+	SOC_DOUBLE_EXT_TLV("DMIC3 UL Volume",
+		GAINS_DMIC3, GAIN_LEFT_OFFSET, GAIN_RIGHT_OFFSET, 149, 0,
+		volume_get_gain, volume_put_gain, dmic_tlv),
+
+	SOC_DOUBLE_EXT_TLV("AMIC UL Volume",
+		GAINS_AMIC, GAIN_LEFT_OFFSET, GAIN_RIGHT_OFFSET, 149, 0,
+		volume_get_gain, volume_put_gain, amic_tlv),
+
+	SOC_DOUBLE_EXT_TLV("BT UL Volume",
+		GAINS_BTUL, GAIN_LEFT_OFFSET, GAIN_RIGHT_OFFSET, 149, 0,
+		volume_get_gain, volume_put_gain, btul_tlv),
+};
+
+static const struct snd_soc_dapm_widget abe_dapm_widgets[] = {
+
+	/* Frontend AIFs */
+	SND_SOC_DAPM_AIF_IN("TONES_DL", "Tones Playback", 0,
+			W_AIF_TONES_DL, ABE_OPP_25, 0),
+	SND_SOC_DAPM_AIF_IN("VX_DL", "Voice Playback", 0,
+			W_AIF_VX_DL, ABE_OPP_50, 0),
+	SND_SOC_DAPM_AIF_OUT("VX_UL", "Voice Capture", 0,
+			W_AIF_VX_UL, ABE_OPP_50, 0),
+	/* the MM_UL mapping is intentional */
+	SND_SOC_DAPM_AIF_OUT("MM_UL1", "MultiMedia1 Capture", 0,
+			W_AIF_MM_UL1, ABE_OPP_100, 0),
+	SND_SOC_DAPM_AIF_OUT("MM_UL2", "MultiMedia2 Capture", 0,
+			W_AIF_MM_UL2, ABE_OPP_50, 0),
+	SND_SOC_DAPM_AIF_IN("MM_DL", " MultiMedia1 Playback", 0,
+			W_AIF_MM_DL, ABE_OPP_25, 0),
+	SND_SOC_DAPM_AIF_IN("MM_DL_LP", " MultiMedia1 LP Playback", 0,
+			W_AIF_MM_DL_LP, ABE_OPP_25, 0),
+	SND_SOC_DAPM_AIF_IN("VIB_DL", "Vibra Playback", 0,
+			W_AIF_VIB_DL, ABE_OPP_100, 0),
+	SND_SOC_DAPM_AIF_IN("MODEM_DL", "MODEM Playback", 0,
+			W_AIF_MODEM_DL, ABE_OPP_50, 0),
+	SND_SOC_DAPM_AIF_OUT("MODEM_UL", "MODEM Capture", 0,
+			W_AIF_MODEM_UL, ABE_OPP_50, 0),
+
+	/* Backend DAIs  */
+	SND_SOC_DAPM_AIF_IN("PDM_UL1", "Analog Capture", 0,
+			W_AIF_PDM_UL1, ABE_OPP_50, 0),
+	SND_SOC_DAPM_AIF_OUT("PDM_DL1", "HS Playback", 0,
+			W_AIF_PDM_DL1, ABE_OPP_25, 0),
+	SND_SOC_DAPM_AIF_OUT("PDM_DL2", "HF Playback", 0,
+			W_AIF_PDM_DL2, ABE_OPP_100, 0),
+	SND_SOC_DAPM_AIF_OUT("PDM_VIB", "Vibra Playback", 0,
+			W_AIF_PDM_VIB, ABE_OPP_100, 0),
+	SND_SOC_DAPM_AIF_IN("BT_VX_UL", "BT Capture", 0,
+			W_AIF_BT_VX_UL, ABE_OPP_50, 0),
+	SND_SOC_DAPM_AIF_OUT("BT_VX_DL", "BT Playback", 0,
+			W_AIF_BT_VX_DL, ABE_OPP_50, 0),
+	SND_SOC_DAPM_AIF_IN("MM_EXT_UL", "FM Capture", 0,
+			W_AIF_MM_EXT_UL, ABE_OPP_50, 0),
+	SND_SOC_DAPM_AIF_OUT("MM_EXT_DL", "FM Playback", 0,
+			W_AIF_MM_EXT_DL, ABE_OPP_25, 0),
+	SND_SOC_DAPM_AIF_IN("DMIC0", "DMIC0 Capture", 0,
+			W_AIF_DMIC0, ABE_OPP_50, 0),
+	SND_SOC_DAPM_AIF_IN("DMIC1", "DMIC1 Capture", 0,
+			W_AIF_DMIC1, ABE_OPP_50, 0),
+	SND_SOC_DAPM_AIF_IN("DMIC2", "DMIC2 Capture", 0,
+			W_AIF_DMIC2, ABE_OPP_50, 0),
+
+	/* ROUTE_UL Capture Muxes */
+	SND_SOC_DAPM_MUX("MUX_UL00",
+			W_MUX_UL00, ABE_OPP_50, 0, &mm_ul00_control),
+	SND_SOC_DAPM_MUX("MUX_UL01",
+			W_MUX_UL01, ABE_OPP_50, 0, &mm_ul01_control),
+	SND_SOC_DAPM_MUX("MUX_UL02",
+			W_MUX_UL02, ABE_OPP_50, 0, &mm_ul02_control),
+	SND_SOC_DAPM_MUX("MUX_UL03",
+			W_MUX_UL03, ABE_OPP_50, 0, &mm_ul03_control),
+	SND_SOC_DAPM_MUX("MUX_UL04",
+			W_MUX_UL04, ABE_OPP_50, 0, &mm_ul04_control),
+	SND_SOC_DAPM_MUX("MUX_UL05",
+			W_MUX_UL05, ABE_OPP_50, 0, &mm_ul05_control),
+	SND_SOC_DAPM_MUX("MUX_UL06",
+			W_MUX_UL06, ABE_OPP_50, 0, &mm_ul06_control),
+	SND_SOC_DAPM_MUX("MUX_UL07",
+			W_MUX_UL07, ABE_OPP_50, 0, &mm_ul07_control),
+	SND_SOC_DAPM_MUX("MUX_UL10",
+			W_MUX_UL10, ABE_OPP_50, 0, &mm_ul10_control),
+	SND_SOC_DAPM_MUX("MUX_UL11",
+			W_MUX_UL11, ABE_OPP_50, 0, &mm_ul11_control),
+	SND_SOC_DAPM_MUX("MUX_VX0",
+			W_MUX_VX00, ABE_OPP_50, 0, &mm_vx0_control),
+	SND_SOC_DAPM_MUX("MUX_VX1",
+			W_MUX_VX01, ABE_OPP_50, 0, &mm_vx1_control),
+
+	/* DL1 & DL2 Playback Mixers */
+	SND_SOC_DAPM_MIXER("DL1 Mixer",
+			W_MIXER_DL1, ABE_OPP_25, 0, dl1_mixer_controls,
+			ARRAY_SIZE(dl1_mixer_controls)),
+	SND_SOC_DAPM_MIXER("DL2 Mixer",
+			W_MIXER_DL2, ABE_OPP_100, 0, dl2_mixer_controls,
+			ARRAY_SIZE(dl2_mixer_controls)),
+
+	/* DL1 Mixer Input volumes ?????*/
+	SND_SOC_DAPM_PGA("DL1 Media Volume",
+			W_VOLUME_DL1, 0, 0, NULL, 0),
+
+	/* AUDIO_UL_MIXER */
+	SND_SOC_DAPM_MIXER("Voice Capture Mixer",
+			W_MIXER_AUDIO_UL, ABE_OPP_50, 0, audio_ul_mixer_controls,
+			ARRAY_SIZE(audio_ul_mixer_controls)),
+
+	/* VX_REC_MIXER */
+	SND_SOC_DAPM_MIXER("Capture Mixer",
+			W_MIXER_VX_REC, ABE_OPP_50, 0, vx_rec_mixer_controls,
+			ARRAY_SIZE(vx_rec_mixer_controls)),
+
+	/* SDT_MIXER  - TODO: shoult this not be OPP25 ??? */
+	SND_SOC_DAPM_MIXER("Sidetone Mixer",
+			W_MIXER_SDT, ABE_OPP_25, 0, sdt_mixer_controls,
+			ARRAY_SIZE(sdt_mixer_controls)),
+
+	/*
+	 * The Following three are virtual switches to select the output port
+	 * after DL1 Gain.
+	 */
+
+	/* Virtual PDM_DL1 Switch */
+	SND_SOC_DAPM_MIXER("DL1 PDM",
+			W_VSWITCH_DL1_PDM, ABE_OPP_25, 0, &pdm_dl1_switch_controls, 1),
+
+	/* Virtual BT_VX_DL Switch */
+	SND_SOC_DAPM_MIXER("DL1 BT_VX",
+			W_VSWITCH_DL1_BT_VX, ABE_OPP_50, 0, &bt_vx_dl_switch_controls, 1),
+
+	/* Virtual MM_EXT_DL Switch TODO: confrm OPP level here */
+	SND_SOC_DAPM_MIXER("DL1 MM_EXT",
+			W_VSWITCH_DL1_MM_EXT, ABE_OPP_50, 0, &mm_ext_dl_switch_controls, 1),
+
+	/* Virtuals to join our capture sources */
+	SND_SOC_DAPM_MIXER("Sidetone Capture VMixer", SND_SOC_NOPM, 0, 0, NULL, 0),
+	SND_SOC_DAPM_MIXER("Voice Capture VMixer", SND_SOC_NOPM, 0, 0, NULL, 0),
+	SND_SOC_DAPM_MIXER("DL1 Capture VMixer", SND_SOC_NOPM, 0, 0, NULL, 0),
+	SND_SOC_DAPM_MIXER("DL2 Capture VMixer", SND_SOC_NOPM, 0, 0, NULL, 0),
+
+	/* Join our MM_DL and MM_DL_LP playback */
+	SND_SOC_DAPM_MIXER("MM_DL VMixer", SND_SOC_NOPM, 0, 0, NULL, 0),
+
+	/* Virtual MODEM and VX_UL mixer */
+	SND_SOC_DAPM_MIXER("VX UL VMixer", SND_SOC_NOPM, 0, 0, NULL, 0),
+	SND_SOC_DAPM_MIXER("VX DL VMixer", SND_SOC_NOPM, 0, 0, NULL, 0),
+
+	/* Virtual Pins to force backends ON atm */
+	SND_SOC_DAPM_OUTPUT("BE_OUT"),
+	SND_SOC_DAPM_INPUT("BE_IN"),
+};
+
+static const struct snd_soc_dapm_route intercon[] = {
+
+	/* MUX_UL00 - ROUTE_UL - Chan 0  */
+	{"MUX_UL00", "DMic0L", "DMIC0"},
+	{"MUX_UL00", "DMic0R", "DMIC0"},
+	{"MUX_UL00", "DMic1L", "DMIC1"},
+	{"MUX_UL00", "DMic1R", "DMIC1"},
+	{"MUX_UL00", "DMic2L", "DMIC2"},
+	{"MUX_UL00", "DMic2R", "DMIC2"},
+	{"MUX_UL00", "BT Left", "BT_VX_UL"},
+	{"MUX_UL00", "BT Right", "BT_VX_UL"},
+	{"MUX_UL00", "MMExt Left", "MM_EXT_UL"},
+	{"MUX_UL00", "MMExt Right", "MM_EXT_UL"},
+	{"MUX_UL00", "AMic0", "PDM_UL1"},
+	{"MUX_UL00", "AMic1", "PDM_UL1"},
+	{"MUX_UL00", "VX Left", "Capture Mixer"},
+	{"MUX_UL00", "VX Right", "Capture Mixer"},
+	{"MM_UL1", NULL, "MUX_UL00"},
+
+	/* MUX_UL01 - ROUTE_UL - Chan 1  */
+	{"MUX_UL01", "DMic0L", "DMIC0"},
+	{"MUX_UL01", "DMic0R", "DMIC0"},
+	{"MUX_UL01", "DMic1L", "DMIC1"},
+	{"MUX_UL01", "DMic1R", "DMIC1"},
+	{"MUX_UL01", "DMic2L", "DMIC2"},
+	{"MUX_UL01", "DMic2R", "DMIC2"},
+	{"MUX_UL01", "BT Left", "BT_VX_UL"},
+	{"MUX_UL01", "BT Right", "BT_VX_UL"},
+	{"MUX_UL01", "MMExt Left", "MM_EXT_UL"},
+	{"MUX_UL01", "MMExt Right", "MM_EXT_UL"},
+	{"MUX_UL01", "AMic0", "PDM_UL1"},
+	{"MUX_UL01", "AMic1", "PDM_UL1"},
+	{"MUX_UL01", "VX Left", "Capture Mixer"},
+	{"MUX_UL01", "VX Right", "Capture Mixer"},
+	{"MM_UL1", NULL, "MUX_UL01"},
+
+	/* MUX_UL02 - ROUTE_UL - Chan 2  */
+	{"MUX_UL02", "DMic0L", "DMIC0"},
+	{"MUX_UL02", "DMic0R", "DMIC0"},
+	{"MUX_UL02", "DMic1L", "DMIC1"},
+	{"MUX_UL02", "DMic1R", "DMIC1"},
+	{"MUX_UL02", "DMic2L", "DMIC2"},
+	{"MUX_UL02", "DMic2R", "DMIC2"},
+	{"MUX_UL02", "BT Left", "BT_VX_UL"},
+	{"MUX_UL02", "BT Right", "BT_VX_UL"},
+	{"MUX_UL02", "MMExt Left", "MM_EXT_UL"},
+	{"MUX_UL02", "MMExt Right", "MM_EXT_UL"},
+	{"MUX_UL02", "AMic0", "PDM_UL1"},
+	{"MUX_UL02", "AMic1", "PDM_UL1"},
+	{"MUX_UL02", "VX Left", "Capture Mixer"},
+	{"MUX_UL02", "VX Right", "Capture Mixer"},
+	{"MM_UL1", NULL, "MUX_UL02"},
+
+	/* MUX_UL03 - ROUTE_UL - Chan 3  */
+	{"MUX_UL03", "DMic0L", "DMIC0"},
+	{"MUX_UL03", "DMic0R", "DMIC0"},
+	{"MUX_UL03", "DMic1L", "DMIC1"},
+	{"MUX_UL03", "DMic1R", "DMIC1"},
+	{"MUX_UL03", "DMic2L", "DMIC2"},
+	{"MUX_UL03", "DMic2R", "DMIC2"},
+	{"MUX_UL03", "BT Left", "BT_VX_UL"},
+	{"MUX_UL03", "BT Right", "BT_VX_UL"},
+	{"MUX_UL03", "MMExt Left", "MM_EXT_UL"},
+	{"MUX_UL03", "MMExt Right", "MM_EXT_UL"},
+	{"MUX_UL03", "AMic0", "PDM_UL1"},
+	{"MUX_UL03", "AMic1", "PDM_UL1"},
+	{"MUX_UL03", "VX Left", "Capture Mixer"},
+	{"MUX_UL03", "VX Right", "Capture Mixer"},
+	{"MM_UL1", NULL, "MUX_UL03"},
+
+	/* MUX_UL04 - ROUTE_UL - Chan 4  */
+	{"MUX_UL04", "DMic0L", "DMIC0"},
+	{"MUX_UL04", "DMic0R", "DMIC0"},
+	{"MUX_UL04", "DMic1L", "DMIC1"},
+	{"MUX_UL04", "DMic1R", "DMIC1"},
+	{"MUX_UL04", "DMic2L", "DMIC2"},
+	{"MUX_UL04", "DMic2R", "DMIC2"},
+	{"MUX_UL04", "BT Left", "BT_VX_UL"},
+	{"MUX_UL04", "BT Right", "BT_VX_UL"},
+	{"MUX_UL04", "MMExt Left", "MM_EXT_UL"},
+	{"MUX_UL04", "MMExt Right", "MM_EXT_UL"},
+	{"MUX_UL04", "AMic0", "PDM_UL1"},
+	{"MUX_UL04", "AMic1", "PDM_UL1"},
+	{"MUX_UL04", "VX Left", "Capture Mixer"},
+	{"MUX_UL04", "VX Right", "Capture Mixer"},
+	{"MM_UL1", NULL, "MUX_UL04"},
+
+	/* MUX_UL05 - ROUTE_UL - Chan 5  */
+	{"MUX_UL05", "DMic0L", "DMIC0"},
+	{"MUX_UL05", "DMic0R", "DMIC0"},
+	{"MUX_UL05", "DMic1L", "DMIC1"},
+	{"MUX_UL05", "DMic1R", "DMIC1"},
+	{"MUX_UL05", "DMic2L", "DMIC2"},
+	{"MUX_UL05", "DMic2R", "DMIC2"},
+	{"MUX_UL05", "BT Left", "BT_VX_UL"},
+	{"MUX_UL05", "BT Right", "BT_VX_UL"},
+	{"MUX_UL05", "MMExt Left", "MM_EXT_UL"},
+	{"MUX_UL05", "MMExt Right", "MM_EXT_UL"},
+	{"MUX_UL05", "AMic0", "PDM_UL1"},
+	{"MUX_UL05", "AMic1", "PDM_UL1"},
+	{"MUX_UL05", "VX Left", "Capture Mixer"},
+	{"MUX_UL05", "VX Right", "Capture Mixer"},
+	{"MM_UL1", NULL, "MUX_UL05"},
+
+	/* MUX_UL06 - ROUTE_UL - Chan 6  */
+	{"MUX_UL06", "DMic0L", "DMIC0"},
+	{"MUX_UL06", "DMic0R", "DMIC0"},
+	{"MUX_UL06", "DMic1L", "DMIC1"},
+	{"MUX_UL06", "DMic1R", "DMIC1"},
+	{"MUX_UL06", "DMic2L", "DMIC2"},
+	{"MUX_UL06", "DMic2R", "DMIC2"},
+	{"MUX_UL06", "BT Left", "BT_VX_UL"},
+	{"MUX_UL06", "BT Right", "BT_VX_UL"},
+	{"MUX_UL06", "MMExt Left", "MM_EXT_UL"},
+	{"MUX_UL06", "MMExt Right", "MM_EXT_UL"},
+	{"MUX_UL06", "AMic0", "PDM_UL1"},
+	{"MUX_UL06", "AMic1", "PDM_UL1"},
+	{"MUX_UL06", "VX Left", "Capture Mixer"},
+	{"MUX_UL06", "VX Right", "Capture Mixer"},
+	{"MM_UL1", NULL, "MUX_UL06"},
+
+	/* MUX_UL07 - ROUTE_UL - Chan 7  */
+	{"MUX_UL07", "DMic0L", "DMIC0"},
+	{"MUX_UL07", "DMic0R", "DMIC0"},
+	{"MUX_UL07", "DMic1L", "DMIC1"},
+	{"MUX_UL07", "DMic1R", "DMIC1"},
+	{"MUX_UL07", "DMic2L", "DMIC2"},
+	{"MUX_UL07", "DMic2R", "DMIC2"},
+	{"MUX_UL07", "BT Left", "BT_VX_UL"},
+	{"MUX_UL07", "BT Right", "BT_VX_UL"},
+	{"MUX_UL07", "MMExt Left", "MM_EXT_UL"},
+	{"MUX_UL07", "MMExt Right", "MM_EXT_UL"},
+	{"MUX_UL07", "AMic0", "PDM_UL1"},
+	{"MUX_UL07", "AMic1", "PDM_UL1"},
+	{"MUX_UL07", "VX Left", "Capture Mixer"},
+	{"MUX_UL07", "VX Right", "Capture Mixer"},
+	{"MM_UL1", NULL, "MUX_UL07"},
+
+	/* MUX_UL10 - ROUTE_UL - Chan 10  */
+	{"MUX_UL10", "DMic0L", "DMIC0"},
+	{"MUX_UL10", "DMic0R", "DMIC0"},
+	{"MUX_UL10", "DMic1L", "DMIC1"},
+	{"MUX_UL10", "DMic1R", "DMIC1"},
+	{"MUX_UL10", "DMic2L", "DMIC2"},
+	{"MUX_UL10", "DMic2R", "DMIC2"},
+	{"MUX_UL10", "BT Left", "BT_VX_UL"},
+	{"MUX_UL10", "BT Right", "BT_VX_UL"},
+	{"MUX_UL10", "MMExt Left", "MM_EXT_UL"},
+	{"MUX_UL10", "MMExt Right", "MM_EXT_UL"},
+	{"MUX_UL10", "AMic0", "PDM_UL1"},
+	{"MUX_UL10", "AMic1", "PDM_UL1"},
+	{"MUX_UL10", "VX Left", "Capture Mixer"},
+	{"MUX_UL10", "VX Right", "Capture Mixer"},
+	{"MM_UL2", NULL, "MUX_UL10"},
+
+	/* MUX_UL11 - ROUTE_UL - Chan 11  */
+	{"MUX_UL11", "DMic0L", "DMIC0"},
+	{"MUX_UL11", "DMic0R", "DMIC0"},
+	{"MUX_UL11", "DMic1L", "DMIC1"},
+	{"MUX_UL11", "DMic1R", "DMIC1"},
+	{"MUX_UL11", "DMic2L", "DMIC2"},
+	{"MUX_UL11", "DMic2R", "DMIC2"},
+	{"MUX_UL11", "BT Left", "BT_VX_UL"},
+	{"MUX_UL11", "BT Right", "BT_VX_UL"},
+	{"MUX_UL11", "MMExt Left", "MM_EXT_UL"},
+	{"MUX_UL11", "MMExt Right", "MM_EXT_UL"},
+	{"MUX_UL11", "AMic0", "PDM_UL1"},
+	{"MUX_UL11", "AMic1", "PDM_UL1"},
+	{"MUX_UL11", "VX Left", "Capture Mixer"},
+	{"MUX_UL11", "VX Right", "Capture Mixer"},
+	{"MM_UL2", NULL, "MUX_UL11"},
+
+	/* MUX_VX0 - ROUTE_UL - Chan 20  */
+	{"MUX_VX0", "DMic0L", "DMIC0"},
+	{"MUX_VX0", "DMic0R", "DMIC0"},
+	{"MUX_VX0", "DMic1L", "DMIC1"},
+	{"MUX_VX0", "DMic1R", "DMIC1"},
+	{"MUX_VX0", "DMic2L", "DMIC2"},
+	{"MUX_VX0", "DMic2R", "DMIC2"},
+	{"MUX_VX0", "BT Left", "BT_VX_UL"},
+	{"MUX_VX0", "BT Right", "BT_VX_UL"},
+	{"MUX_VX0", "MMExt Left", "MM_EXT_UL"},
+	{"MUX_VX0", "MMExt Right", "MM_EXT_UL"},
+	{"MUX_VX0", "AMic0", "PDM_UL1"},
+	{"MUX_VX0", "AMic1", "PDM_UL1"},
+	{"MUX_VX0", "VX Left", "Capture Mixer"},
+	{"MUX_VX0", "VX Right", "Capture Mixer"},
+
+	/* MUX_VX1 - ROUTE_UL - Chan 20  */
+	{"MUX_VX1", "DMic0L", "DMIC0"},
+	{"MUX_VX1", "DMic0R", "DMIC0"},
+	{"MUX_VX1", "DMic1L", "DMIC1"},
+	{"MUX_VX1", "DMic1R", "DMIC1"},
+	{"MUX_VX1", "DMic2L", "DMIC2"},
+	{"MUX_VX1", "DMic2R", "DMIC2"},
+	{"MUX_VX1", "BT Left", "BT_VX_UL"},
+	{"MUX_VX1", "BT Right", "BT_VX_UL"},
+	{"MUX_VX1", "MMExt Left", "MM_EXT_UL"},
+	{"MUX_VX1", "MMExt Right", "MM_EXT_UL"},
+	{"MUX_VX1", "AMic0", "PDM_UL1"},
+	{"MUX_VX1", "AMic1", "PDM_UL1"},
+	{"MUX_VX1", "VX Left", "Capture Mixer"},
+	{"MUX_VX1", "VX Right", "Capture Mixer"},
+
+	/* Headset (DL1)  playback path */
+	{"DL1 Mixer", "Tones", "TONES_DL"},
+	{"DL1 Mixer", "Voice", "VX DL VMixer"},
+	{"DL1 Mixer", "Capture", "DL1 Capture VMixer"},
+	{"DL1 Capture VMixer", NULL, "MUX_UL10"},
+	{"DL1 Capture VMixer", NULL, "MUX_UL11"},
+	{"DL1 Mixer", "Multimedia", "MM_DL VMixer"},
+	{"MM_DL VMixer", NULL, "MM_DL"},
+	{"MM_DL VMixer", NULL, "MM_DL_LP"},
+
+	/* Sidetone Mixer */
+	{"Sidetone Mixer", "Playback", "DL1 Mixer"},
+	{"Sidetone Mixer", "Capture", "Sidetone Capture VMixer"},
+	{"Sidetone Capture VMixer", NULL, "MUX_VX0"},
+	{"Sidetone Capture VMixer", NULL, "MUX_VX1"},
+
+	/* Playback Output selection after DL1 Gain */
+	{"DL1 BT_VX", "Switch", "Sidetone Mixer"},
+	{"DL1 MM_EXT", "Switch", "Sidetone Mixer"},
+	{"DL1 PDM", "Switch", "Sidetone Mixer"},
+	{"PDM_DL1", NULL, "DL1 PDM"},
+	{"BT_VX_DL", NULL, "DL1 BT_VX"},
+	{"MM_EXT_DL", NULL, "DL1 MM_EXT"},
+
+	/* Handsfree (DL2) playback path */
+	{"DL2 Mixer", "Tones", "TONES_DL"},
+	{"DL2 Mixer", "Voice", "VX DL VMixer"},
+	{"DL2 Mixer", "Capture", "DL2 Capture VMixer"},
+	{"DL2 Capture VMixer", NULL, "MUX_UL10"},
+	{"DL2 Capture VMixer", NULL, "MUX_UL11"},
+	{"DL2 Mixer", "Multimedia", "MM_DL VMixer"},
+	{"MM_DL VMixer", NULL, "MM_DL"},
+	{"MM_DL VMixer", NULL, "MM_DL_LP"},
+	{"PDM_DL2", NULL, "DL2 Mixer"},
+
+	/* VxREC Mixer */
+	{"Capture Mixer", "Tones", "TONES_DL"},
+	{"Capture Mixer", "Voice Playback", "VX DL VMixer"},
+	{"Capture Mixer", "Voice Capture", "VX UL VMixer"},
+	{"Capture Mixer", "Media Playback", "MM_DL VMixer"},
+	{"MM_DL VMixer", NULL, "MM_DL"},
+	{"MM_DL VMixer", NULL, "MM_DL_LP"},
+
+	/* Audio UL mixer */
+	{"Voice Capture Mixer", "Tones Playback", "TONES_DL"},
+	{"Voice Capture Mixer", "Media Playback", "MM_DL VMixer"},
+	{"MM_DL VMixer", NULL, "MM_DL"},
+	{"MM_DL VMixer", NULL, "MM_DL_LP"},
+	{"Voice Capture Mixer", "Capture", "Voice Capture VMixer"},
+	{"Voice Capture VMixer", NULL, "MUX_VX0"},
+	{"Voice Capture VMixer", NULL, "MUX_VX1"},
+
+	/* BT */
+	{"VX UL VMixer", NULL, "Voice Capture Mixer"},
+
+	/* Vibra */
+	{"PDM_VIB", NULL, "VIB_DL"},
+
+	/* VX and MODEM */
+	{"VX_UL", NULL, "VX UL VMixer"},
+	{"MODEM_UL", NULL, "VX UL VMixer"},
+	{"VX DL VMixer", NULL, "VX_DL"},
+	{"VX DL VMixer", NULL, "MODEM_DL"},
+
+	/* Backend Enablement - TODO: maybe re-work*/
+	{"BE_OUT", NULL, "PDM_DL1"},
+	{"BE_OUT", NULL, "PDM_DL2"},
+	{"BE_OUT", NULL, "PDM_VIB"},
+	{"BE_OUT", NULL, "MM_EXT_DL"},
+	{"BE_OUT", NULL, "BT_VX_DL"},
+	{"PDM_UL1", NULL, "BE_IN"},
+	{"BT_VX_UL", NULL, "BE_IN"},
+	{"MM_EXT_UL", NULL, "BE_IN"},
+	{"DMIC0", NULL, "BE_IN"},
+	{"DMIC1", NULL, "BE_IN"},
+	{"DMIC2", NULL, "BE_IN"},
+};
+
+#ifdef CONFIG_DEBUG_FS
+
+static int abe_dbg_get_dma_pos(struct abe_data *abe)
+{
+	return omap_get_dma_dst_pos(abe->dma_ch) - abe->dbg_buffer_addr;
+}
+
+static void abe_dbg_dma_irq(int ch, u16 stat, void *data)
+{
+}
+
+static int abe_dbg_start_dma(struct abe_data *abe, int circular)
+{
+	struct omap_dma_channel_params dma_params;
+	int err;
+
+	/* TODO: start the DMA in either :-
+	 *
+	 * 1) circular buffer mode where the DMA will restart when it get to
+	 *    the end of the buffer.
+	 * 2) default mode, where DMA stops at the end of the buffer.
+	 */
+
+	abe->dma_req = OMAP44XX_DMA_ABE_REQ_7;
+	err = omap_request_dma(abe->dma_req, "ABE debug",
+			       abe_dbg_dma_irq, abe, &abe->dma_ch);
+	if (abe->dbg_circular) {
+		/*
+		 * Link channel with itself so DMA doesn't need any
+		 * reprogramming while looping the buffer
+		 */
+		omap_dma_link_lch(abe->dma_ch, abe->dma_ch);
+	}
+
+	memset(&dma_params, 0, sizeof(dma_params));
+	dma_params.data_type = OMAP_DMA_DATA_TYPE_S32;
+	dma_params.trigger = abe->dma_req;
+	dma_params.sync_mode = OMAP_DMA_SYNC_FRAME;
+	dma_params.src_amode = OMAP_DMA_AMODE_DOUBLE_IDX;
+	dma_params.dst_amode = OMAP_DMA_AMODE_POST_INC;
+	dma_params.src_or_dst_synch = OMAP_DMA_SRC_SYNC;
+	dma_params.src_start = D_DEBUG_FIFO_ADDR + ABE_DMEM_BASE_ADDRESS_L3;
+	dma_params.dst_start = abe->dbg_buffer_addr;
+	dma_params.src_port = OMAP_DMA_PORT_MPUI;
+	dma_params.src_ei = 1;
+	dma_params.src_fi = 1 - abe->dbg_elem_bytes;
+
+	dma_params.elem_count = abe->dbg_elem_bytes >> 2; /* 128 bytes shifted into words */
+	dma_params.frame_count = abe->dbg_buffer_bytes / abe->dbg_elem_bytes;
+	omap_set_dma_params(abe->dma_ch, &dma_params);
+
+	omap_enable_dma_irq(abe->dma_ch, OMAP_DMA_FRAME_IRQ);
+	omap_set_dma_src_burst_mode(abe->dma_ch, OMAP_DMA_DATA_BURST_16);
+	omap_set_dma_dest_burst_mode(abe->dma_ch, OMAP_DMA_DATA_BURST_16);
+
+	abe->dbg_reader_offset = 0;
+
+	pm_runtime_get_sync(abe->dev);
+	omap_start_dma(abe->dma_ch);
+	return 0;
+}
+
+static void abe_dbg_stop_dma(struct abe_data *abe)
+{
+	while (omap_get_dma_active_status(abe->dma_ch))
+		omap_stop_dma(abe->dma_ch);
+
+	if (abe->dbg_circular)
+		omap_dma_unlink_lch(abe->dma_ch, abe->dma_ch);
+	omap_free_dma(abe->dma_ch);
+	pm_runtime_put_sync(abe->dev);
+}
+
+static int abe_open_data(struct inode *inode, struct file *file)
+{
+	struct abe_data *abe = inode->i_private;
+
+	abe->dbg_elem_bytes = 128; /* size of debug data per tick */
+
+	if (abe->dbg_format1)
+		abe->dbg_elem_bytes += ABE_DBG_FLAG1_SIZE;
+	if (abe->dbg_format2)
+		abe->dbg_elem_bytes += ABE_DBG_FLAG2_SIZE;
+	if (abe->dbg_format3)
+		abe->dbg_elem_bytes += ABE_DBG_FLAG3_SIZE;
+
+	abe->dbg_buffer_bytes = abe->dbg_elem_bytes * 4 *
+							abe->dbg_buffer_msecs;
+
+	abe->dbg_buffer = dma_alloc_writecombine(abe->dev,
+			abe->dbg_buffer_bytes, &abe->dbg_buffer_addr, GFP_KERNEL);
+	if (abe->dbg_buffer == NULL)
+		return -ENOMEM;
+
+	file->private_data = inode->i_private;
+	abe->dbg_complete = 0;
+	abe_dbg_start_dma(abe, abe->dbg_circular);
+
+	return 0;
+}
+
+static int abe_release_data(struct inode *inode, struct file *file)
+{
+	struct abe_data *abe = inode->i_private;
+
+	abe_dbg_stop_dma(abe);
+
+	dma_free_writecombine(abe->dev, abe->dbg_buffer_bytes,
+				      abe->dbg_buffer, abe->dbg_buffer_addr);
+	return 0;
+}
+
+static ssize_t abe_copy_to_user(struct abe_data *abe, char __user *user_buf,
+			       size_t count)
+{
+	/* check for reader buffer wrap */
+	if (abe->dbg_reader_offset + count > abe->dbg_buffer_bytes) {
+		int size = abe->dbg_buffer_bytes - abe->dbg_reader_offset;
+
+		/* wrap */
+		if (copy_to_user(user_buf,
+			abe->dbg_buffer + abe->dbg_reader_offset, size))
+			return -EFAULT;
+
+		/* need to just return if non circular */
+		if (!abe->dbg_circular) {
+			abe->dbg_complete = 1;
+			return count;
+		}
+
+		if (copy_to_user(user_buf,
+			abe->dbg_buffer, count - size))
+			return -EFAULT;
+		abe->dbg_reader_offset = count - size;
+		return count;
+	} else {
+		/* no wrap */
+		if (copy_to_user(user_buf,
+			abe->dbg_buffer + abe->dbg_reader_offset, count))
+			return -EFAULT;
+		abe->dbg_reader_offset += count;
+
+		if (!abe->dbg_circular &&
+				abe->dbg_reader_offset == abe->dbg_buffer_bytes)
+			abe->dbg_complete = 1;
+
+		return count;
+	}
+}
+
+static ssize_t abe_read_data(struct file *file, char __user *user_buf,
+			       size_t count, loff_t *ppos)
+{
+	ssize_t ret = 0;
+	struct abe_data *abe = file->private_data;
+	DECLARE_WAITQUEUE(wait, current);
+	int dma_offset, bytes;
+
+	add_wait_queue(&abe->wait, &wait);
+	do {
+		set_current_state(TASK_INTERRUPTIBLE);
+		/* TODO: Check if really needed. Or adjust sleep delay
+		 * If not delay trace is not working */
+		msleep_interruptible(1);
+		dma_offset = abe_dbg_get_dma_pos(abe);
+
+		/* is DMA finished ? */
+		if (abe->dbg_complete)
+			break;
+
+		/* get maximum amount of debug bytes we can read */
+		if (dma_offset >= abe->dbg_reader_offset) {
+			/* dma ptr is ahead of reader */
+			bytes = dma_offset - abe->dbg_reader_offset;
+		} else {
+			/* dma ptr is behind reader */
+			bytes = dma_offset + abe->dbg_buffer_bytes -
+				abe->dbg_reader_offset;
+		}
+
+		if (count > bytes)
+			count = bytes;
+
+		if (count > 0) {
+			ret = abe_copy_to_user(abe, user_buf, count);
+			break;
+		}
+
+		if (file->f_flags & O_NONBLOCK) {
+			ret = -EAGAIN;
+			break;
+		}
+
+		if (signal_pending(current)) {
+			ret = -ERESTARTSYS;
+			break;
+		}
+
+		schedule();
+
+	} while (1);
+
+	__set_current_state(TASK_RUNNING);
+	remove_wait_queue(&abe->wait, &wait);
+
+	return ret;
+}
+
+static const struct file_operations abe_data_fops = {
+	.open = abe_open_data,
+	.read = abe_read_data,
+	.release = abe_release_data,
+};
+
+static void abe_init_debugfs(struct abe_data *abe)
+{
+	abe->debugfs_root = debugfs_create_dir("omap4-abe", NULL);
+	if (!abe->debugfs_root) {
+		printk(KERN_WARNING "ABE: Failed to create debugfs directory\n");
+		return;
+	}
+
+	abe->debugfs_fmt1 = debugfs_create_bool("format1", 0644,
+						 abe->debugfs_root,
+						 &abe->dbg_format1);
+	if (!abe->debugfs_fmt1)
+		printk(KERN_WARNING "ABE: Failed to create format1 debugfs file\n");
+
+	abe->debugfs_fmt2 = debugfs_create_bool("format2", 0644,
+						 abe->debugfs_root,
+						 &abe->dbg_format2);
+	if (!abe->debugfs_fmt2)
+		printk(KERN_WARNING "ABE: Failed to create format2 debugfs file\n");
+
+	abe->debugfs_fmt3 = debugfs_create_bool("format3", 0644,
+						 abe->debugfs_root,
+						 &abe->dbg_format3);
+	if (!abe->debugfs_fmt3)
+		printk(KERN_WARNING "ABE: Failed to create format3 debugfs file\n");
+
+	abe->debugfs_elem_bytes = debugfs_create_u32("element_bytes", 0604,
+						 abe->debugfs_root,
+						 &abe->dbg_elem_bytes);
+	if (!abe->debugfs_elem_bytes)
+		printk(KERN_WARNING "ABE: Failed to create element size debugfs file\n");
+
+	abe->debugfs_size = debugfs_create_u32("msecs", 0644,
+						 abe->debugfs_root,
+						 &abe->dbg_buffer_msecs);
+	if (!abe->debugfs_size)
+		printk(KERN_WARNING "ABE: Failed to create buffer size debugfs file\n");
+
+	abe->debugfs_circ = debugfs_create_bool("circular", 0644,
+						 abe->debugfs_root,
+						 &abe->dbg_circular);
+	if (!abe->debugfs_size)
+		printk(KERN_WARNING "ABE: Failed to create circular mode debugfs file\n");
+
+	abe->debugfs_data = debugfs_create_file("debug", 0644,
+						 abe->debugfs_root,
+						 abe, &abe_data_fops);
+	if (!abe->debugfs_data)
+		printk(KERN_WARNING "ABE: Failed to create data debugfs file\n");
+
+	abe->debugfs_opp_level = debugfs_create_u32("opp_level", 0604,
+						 abe->debugfs_root,
+						 &abe->opp);
+	if (!abe->debugfs_opp_level)
+		printk(KERN_WARNING "ABE: Failed to create OPP level debugfs file\n");
+
+	abe->dbg_buffer_msecs = 500;
+	init_waitqueue_head(&abe->wait);
+}
+
+static void abe_cleanup_debugfs(struct abe_data *abe)
+{
+	debugfs_remove_recursive(abe->debugfs_root);
+}
+
+#else
+
+static inline void abe_init_debugfs(struct abe_data *abe)
+{
+}
+
+static inline void abe_cleanup_debugfs(struct abe_data *abe)
+{
+}
+#endif
+
+static const struct snd_pcm_hardware omap_abe_hardware = {
+	.info			= SNDRV_PCM_INFO_MMAP |
+				  SNDRV_PCM_INFO_MMAP_VALID |
+				  SNDRV_PCM_INFO_INTERLEAVED |
+				  SNDRV_PCM_INFO_BLOCK_TRANSFER |
+				  SNDRV_PCM_INFO_PAUSE |
+				  SNDRV_PCM_INFO_RESUME,
+	.formats		= SNDRV_PCM_FMTBIT_S16_LE |
+				  SNDRV_PCM_FMTBIT_S32_LE,
+	.period_bytes_min	= 4 * 1024,
+	.period_bytes_max	= 24 * 1024,
+	.periods_min		= 2,
+	.periods_max		= 2,
+	.buffer_bytes_max	= 24 * 1024 * 2,
+};
+
+static int aess_set_runtime_opp_level(struct abe_data *abe)
+{
+	int i, opp = 0;
+
+	mutex_lock(&abe->opp_mutex);
+
+	pm_runtime_get_sync(abe->dev);
+
+	/* now calculate OPP level based upon DAPM widget status */
+	for (i = 0; i < ABE_NUM_WIDGETS; i++) {
+		if (abe->widget_opp[ABE_WIDGET(i)]) {
+			dev_dbg(abe->dev, "OPP: id %d = %d%%\n", i,
+					abe->widget_opp[ABE_WIDGET(i)] * 25);
+			opp |= abe->widget_opp[ABE_WIDGET(i)];
+		}
+	}
+	opp = (1 << (fls(opp) - 1)) * 25;
+
+	if (abe->opp > opp) {
+		/* Decrease OPP mode - no need of OPP100% */
+		switch (opp) {
+		case 25:
+			abe_set_opp_processing(ABE_OPP25);
+			udelay(250);
+			omap_device_set_rate(abe->dev, abe->dev, 49150000);
+			break;
+		case 50:
+		default:
+			abe_set_opp_processing(ABE_OPP50);
+			udelay(250);
+			omap_device_set_rate(abe->dev, abe->dev, 98300000);
+			break;
+		}
+	} else if (abe->opp < opp) {
+		/* Increase OPP mode */
+		switch (opp) {
+		case 25:
+			omap_device_set_rate(abe->dev, abe->dev, 49000000);
+			abe_set_opp_processing(ABE_OPP25);
+			break;
+		case 50:
+			omap_device_set_rate(abe->dev, abe->dev, 98300000);
+			abe_set_opp_processing(ABE_OPP50);
+			break;
+		case 100:
+		default:
+			omap_device_set_rate(abe->dev, abe->dev, 196600000);
+			abe_set_opp_processing(ABE_OPP100);
+			break;
+		}
+	}
+	abe->opp = opp;
+	dev_dbg(abe->dev, "new OPP level is %d\n", opp);
+
+	pm_runtime_put_sync(abe->dev);
+
+	mutex_unlock(&abe->opp_mutex);
+
+	return 0;
+}
+
+static int aess_open(struct snd_pcm_substream *substream)
+{
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_platform *platform = rtd->platform;
+	struct abe_data *abe = snd_soc_platform_get_drvdata(platform);
+	struct snd_soc_dai *dai = rtd->cpu_dai;
+	int ret = 0;
+
+	mutex_lock(&abe->mutex);
+
+	dev_dbg(dai->dev, "%s: %s\n", __func__, dai->name);
+
+	pm_runtime_get_sync(abe->dev);
+
+	if (!abe->active++)
+		abe_wakeup();
+
+	switch (dai->id) {
+	case ABE_FRONTEND_DAI_MODEM:
+		break;
+	case ABE_FRONTEND_DAI_LP_MEDIA:
+		snd_soc_set_runtime_hwparams(substream, &omap_abe_hardware);
+		ret = snd_pcm_hw_constraint_step(substream->runtime, 0,
+					 SNDRV_PCM_HW_PARAM_BUFFER_BYTES, 1024);
+		break;
+	default:
+		break;
+	}
+
+	mutex_unlock(&abe->mutex);
+	return ret;
+}
+
+static int aess_hw_params(struct snd_pcm_substream *substream,
+	struct snd_pcm_hw_params *params)
+{
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_pcm_runtime *runtime = substream->runtime;
+	struct snd_soc_platform *platform = rtd->platform;
+	struct abe_data *abe = snd_soc_platform_get_drvdata(platform);
+	struct snd_soc_dai *dai = rtd->cpu_dai;
+	abe_data_format_t format;
+	size_t period_size;
+	u32 dst;
+
+	dev_dbg(dai->dev, "%s: %s\n", __func__, dai->name);
+
+	if (dai->id != ABE_FRONTEND_DAI_LP_MEDIA)
+		return 0;
+
+	/*Storing substream pointer for irq*/
+	abe->ping_pong_substream = substream;
+
+	format.f = params_rate(params);
+	if (params_format(params) == SNDRV_PCM_FORMAT_S32_LE)
+		format.samp_format = STEREO_MSB;
+	else
+		format.samp_format = STEREO_16_16;
+
+	if (format.f == 44100)
+		abe_write_event_generator(EVENT_44100);
+
+	period_size = params_period_bytes(params);
+
+	/*Adding ping pong buffer subroutine*/
+	abe_plug_subroutine(&abe_irq_pingpong_player_id,
+				(abe_subroutine2) abe_irq_pingpong_subroutine,
+				SUB_1_PARAM, (u32 *)abe);
+
+	/* Connect a Ping-Pong cache-flush protocol to MM_DL port */
+	abe_connect_irq_ping_pong_port(MM_DL_PORT, &format,
+				abe_irq_pingpong_player_id,
+				period_size, &dst,
+				PING_PONG_WITH_MCU_IRQ);
+
+	/* Memory mapping for hw params */
+	runtime->dma_area  = abe->io_base[0] + dst;
+	runtime->dma_addr  = 0;
+	runtime->dma_bytes = period_size * 2;
+
+	/* Need to set the first buffer in order to get interrupt */
+	abe_set_ping_pong_buffer(MM_DL_PORT, period_size);
+	abe->first_irq = 1;
+
+	return 0;
+}
+
+static int aess_prepare(struct snd_pcm_substream *substream)
+{
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_platform *platform = rtd->platform;
+	struct abe_data *abe = snd_soc_platform_get_drvdata(platform);
+	struct snd_soc_dai *dai = rtd->cpu_dai;
+
+	mutex_lock(&abe->mutex);
+	dev_dbg(dai->dev, "%s: %s\n", __func__, dai->name);
+	aess_set_runtime_opp_level(abe);
+	mutex_unlock(&abe->mutex);
+	return 0;
+}
+
+static int aess_close(struct snd_pcm_substream *substream)
+{
+	struct snd_soc_pcm_runtime *rtd = substream->private_data;
+	struct snd_soc_platform *platform = rtd->platform;
+	struct abe_data *abe = snd_soc_platform_get_drvdata(platform);
+	struct snd_soc_dai *dai = rtd->cpu_dai;
+
+	mutex_lock(&abe->mutex);
+	aess_set_runtime_opp_level(abe);
+
+	dev_dbg(dai->dev, "%s: %s\n", __func__, dai->name);
+
+	if (!--abe->active) {
+		abe_disable_irq();
+		abe_dsp_shutdown();
+		pm_runtime_put_sync(abe->dev);
+	}
+
+	mutex_unlock(&abe->mutex);
+	return 0;
+}
+
+static int aess_mmap(struct snd_pcm_substream *substream,
+	struct vm_area_struct *vma)
+{
+	struct snd_soc_pcm_runtime  *rtd = substream->private_data;
+	struct snd_soc_dai *dai = rtd->cpu_dai;
+	int offset, size, err;
+
+	if (dai->id != ABE_FRONTEND_DAI_LP_MEDIA)
+		return -EINVAL;
+
+	vma->vm_flags |= VM_IO | VM_RESERVED;
+	vma->vm_page_prot = pgprot_noncached(vma->vm_page_prot);
+	size = vma->vm_end - vma->vm_start;
+	offset = vma->vm_pgoff << PAGE_SHIFT;
+
+	err = io_remap_pfn_range(vma, vma->vm_start,
+			(ABE_DMEM_BASE_ADDRESS_MPU +
+			ABE_DMEM_BASE_OFFSET_PING_PONG + offset) >> PAGE_SHIFT,
+			size, vma->vm_page_prot);
+
+	if (err)
+		return -EAGAIN;
+
+	return 0;
+}
+
+static snd_pcm_uframes_t aess_pointer(struct snd_pcm_substream *substream)
+{
+	snd_pcm_uframes_t offset;
+	u32 pingpong;
+
+	abe_read_offset_from_ping_buffer(MM_DL_PORT, &pingpong);
+	offset = (snd_pcm_uframes_t)pingpong;
+
+	return offset;
+}
+
+static struct snd_pcm_ops omap_aess_pcm_ops = {
+	.open           = aess_open,
+	.hw_params	= aess_hw_params,
+	.prepare	= aess_prepare,
+	.close	        = aess_close,
+	.pointer	= aess_pointer,
+	.mmap		= aess_mmap,
+};
+
+#if CONFIG_PM
+static int aess_suspend(struct device *dev)
+{
+	struct abe_data *abe = dev_get_drvdata(dev);
+	struct omap4_abe_dsp_pdata *pdata = abe->abe_pdata;
+
+	pm_runtime_get_sync(abe->dev);
+
+	if (abe->active && abe_check_activity()) {
+		dev_dbg(abe->dev, "Suspend in a middle of ABE activity!\n");
+		goto no_suspend;
+	}
+
+	/* TODO: Find a better way to save/retore gains after dor OFF mode */
+	abe_mute_gain(MIXSDT, MIX_SDT_INPUT_UP_MIXER);
+	abe_mute_gain(MIXSDT, MIX_SDT_INPUT_DL1_MIXER);
+	abe_mute_gain(MIXECHO, MIX_ECHO_DL1);
+	abe_mute_gain(MIXECHO, MIX_ECHO_DL2);
+	abe_mute_gain(MIXAUDUL, MIX_AUDUL_INPUT_MM_DL);
+	abe_mute_gain(MIXAUDUL, MIX_AUDUL_INPUT_TONES);
+	abe_mute_gain(MIXAUDUL, MIX_AUDUL_INPUT_UPLINK);
+	abe_mute_gain(MIXAUDUL, MIX_AUDUL_INPUT_VX_DL);
+	abe_mute_gain(MIXVXREC, MIX_VXREC_INPUT_TONES);
+	abe_mute_gain(MIXVXREC, MIX_VXREC_INPUT_VX_DL);
+	abe_mute_gain(MIXVXREC, MIX_VXREC_INPUT_MM_DL);
+	abe_mute_gain(MIXVXREC, MIX_VXREC_INPUT_VX_UL);
+
+no_suspend:
+	pm_runtime_put_sync(abe->dev);
+
+	/*
+	 * force setting OPP after suspend/resume to ensure
+	 * ABE freq/volt are set to proper values
+	 */
+	abe->opp = 0;
+
+	if (pdata->get_context_loss_count)
+		abe->loss_count = pdata->get_context_loss_count(dev);
+
+	return 0;
+}
+
+static int aess_resume(struct device *dev)
+{
+	struct abe_data *abe = dev_get_drvdata(dev);
+	struct omap4_abe_dsp_pdata *pdata = abe->abe_pdata;
+	int loss_count = 0;
+
+	if (pdata->get_context_loss_count)
+		loss_count = pdata->get_context_loss_count(dev);
+
+	pm_runtime_get_sync(abe->dev);
+
+	if (abe->active && abe_check_activity()) {
+		dev_dbg(abe->dev, "Resume in a middle of ABE activity!\n");
+		goto no_resume;
+	}
+
+	if (loss_count != abe->loss_count)
+		abe_reload_fw();
+
+	/* TODO: Find a better way to save/retore gains after dor OFF mode */
+	abe_unmute_gain(MIXSDT, MIX_SDT_INPUT_UP_MIXER);
+	abe_unmute_gain(MIXSDT, MIX_SDT_INPUT_DL1_MIXER);
+	abe_unmute_gain(MIXECHO, MIX_ECHO_DL1);
+	abe_unmute_gain(MIXECHO, MIX_ECHO_DL2);
+	abe_unmute_gain(MIXAUDUL, MIX_AUDUL_INPUT_MM_DL);
+	abe_unmute_gain(MIXAUDUL, MIX_AUDUL_INPUT_TONES);
+	abe_unmute_gain(MIXAUDUL, MIX_AUDUL_INPUT_UPLINK);
+	abe_unmute_gain(MIXAUDUL, MIX_AUDUL_INPUT_VX_DL);
+	abe_unmute_gain(MIXVXREC, MIX_VXREC_INPUT_TONES);
+	abe_unmute_gain(MIXVXREC, MIX_VXREC_INPUT_VX_DL);
+	abe_unmute_gain(MIXVXREC, MIX_VXREC_INPUT_MM_DL);
+	abe_unmute_gain(MIXVXREC, MIX_VXREC_INPUT_VX_UL);
+//	abe_dsp_set_equalizer(EQ1, abe->dl1_equ_profile);
+//	abe_dsp_set_equalizer(EQ2L, abe->dl20_equ_profile);
+//	abe_dsp_set_equalizer(EQ2R, abe->dl21_equ_profile);
+//	abe_dsp_set_equalizer(EQAMIC, abe->amic_equ_profile);
+//	abe_dsp_set_equalizer(EQDMIC, abe->dmic_equ_profile);
+//	abe_dsp_set_equalizer(EQSDT, abe->sdt_equ_profile);
+
+no_resume:
+	pm_runtime_put_sync(abe->dev);
+
+	return 0;
+}
+
+#else
+#define aess_suspend	NULL
+#define aess_resume	NULL
+#endif
+
+static const struct dev_pm_ops aess_pm_ops = {
+	.suspend = aess_suspend,
+	.resume = aess_resume,
+};
+
+static int aess_stream_event(struct snd_soc_dapm_context *dapm)
+{
+	struct snd_soc_platform *platform = dapm->platform;
+	struct abe_data *abe = snd_soc_platform_get_drvdata(platform);
+
+	pm_runtime_get_sync(abe->dev);
+
+	if (abe->active)
+		aess_set_runtime_opp_level(abe);
+
+	pm_runtime_put_sync(abe->dev);
+
+	return 0;
+}
+
+static int abe_add_widgets(struct snd_soc_platform *platform)
+{
+	struct abe_data *abe = snd_soc_platform_get_drvdata(platform);
+	struct fw_header *hdr = &abe->hdr;
+	int i, j;
+
+#if defined(CONFIG_SND_OMAP_SOC_ABE_DSP_MODULE)
+	/* create equalizer controls */
+	for (i = 0; i < hdr->num_equ; i++) {
+		struct soc_enum *equalizer_enum = &abe->equalizer_enum[i];
+		struct snd_kcontrol_new *equalizer_control =
+				&abe->equalizer_control[i];
+
+		equalizer_enum->reg = i;
+		equalizer_enum->max = abe->equ_texts[i].count;
+		for (j = 0; j < abe->equ_texts[i].count; j++)
+			equalizer_enum->dtexts[j] = abe->equ_texts[i].texts[j];
+
+		equalizer_control->name = abe->equ_texts[i].name;
+		equalizer_control->private_value = (unsigned long)equalizer_enum;
+		equalizer_control->get = abe_get_equalizer;
+		equalizer_control->put = abe_put_equalizer;
+		equalizer_control->info = snd_soc_info_enum_ext1;
+		equalizer_control->iface = SNDRV_CTL_ELEM_IFACE_MIXER;
+
+		dev_dbg(platform->dev, "added EQU mixer: %s profiles %d\n",
+				abe->equ_texts[i].name, abe->equ_texts[i].count);
+
+		for (j = 0; j < abe->equ_texts[i].count; j++)
+			dev_dbg(platform->dev, " %s\n", equalizer_enum->dtexts[j]);
+	}
+
+	snd_soc_add_platform_controls(platform, abe->equalizer_control,
+			hdr->num_equ);
+#endif
+
+	snd_soc_add_platform_controls(platform, abe_controls,
+			ARRAY_SIZE(abe_controls));
+
+	snd_soc_dapm_new_controls(&platform->dapm, abe_dapm_widgets,
+				 ARRAY_SIZE(abe_dapm_widgets));
+
+	snd_soc_dapm_add_routes(&platform->dapm, intercon, ARRAY_SIZE(intercon));
+
+	snd_soc_dapm_new_widgets(&platform->dapm);
+
+	return 0;
+}
+
+static int abe_probe(struct snd_soc_platform *platform)
+{
+	struct abe_data *abe = snd_soc_platform_get_drvdata(platform);
+	const struct firmware *fw;
+#ifndef CONFIG_PM_RUNTIME
+	struct omap4_abe_dsp_pdata *pdata = priv->abe_pdata;
+#endif
+	int ret = 0, i, offset = 0;
+
+	abe->platform = platform;
+
+	pm_runtime_enable(abe->dev);
+
+#if defined(CONFIG_SND_OMAP_SOC_ABE_DSP_MODULE)
+	/* request firmware & coefficients */
+	ret = request_firmware(&fw, "omap4_abe", platform->dev);
+	if (ret != 0) {
+		dev_err(abe->dev, "Failed to load firmware: %d\n", ret);
+		return ret;
+	}
+
+	/* get firmware and coefficients header info */
+	memcpy(&abe->hdr, fw->data, sizeof(struct fw_header));
+	if (abe->hdr.firmware_size > ABE_MAX_FW_SIZE) {
+			dev_err(abe->dev, "Firmware too large at %d bytes: %d\n",
+					abe->hdr.firmware_size, ret);
+			ret = -EINVAL;
+			goto err_fw;
+	}
+	dev_dbg(abe->dev, "ABE firmware size %d bytes\n", abe->hdr.firmware_size);
+
+	if (abe->hdr.coeff_size > ABE_MAX_COEFF_SIZE) {
+		dev_err(abe->dev, "Coefficients too large at %d bytes: %d\n",
+					abe->hdr.coeff_size, ret);
+			ret = -EINVAL;
+			goto err_fw;
+	}
+	dev_dbg(abe->dev, "ABE coefficients size %d bytes\n", abe->hdr.coeff_size);
+
+	/* get coefficient EQU mixer strings */
+	if (abe->hdr.num_equ >= ABE_MAX_EQU) {
+		dev_err(abe->dev, "Too many equalizers got %d\n", abe->hdr.num_equ);
+		ret = -EINVAL;
+		goto err_fw;
+	}
+	abe->equ_texts = kzalloc(abe->hdr.num_equ * sizeof(struct coeff_config),
+			GFP_KERNEL);
+	if (abe->equ_texts == NULL) {
+		ret = -ENOMEM;
+		goto err_fw;
+	}
+	offset = sizeof(struct fw_header);
+	memcpy(abe->equ_texts, fw->data + offset,
+			abe->hdr.num_equ * sizeof(struct coeff_config));
+
+	/* get coefficients from firmware */
+	abe->equ[0] = kmalloc(abe->hdr.coeff_size, GFP_KERNEL);
+	if (abe->equ[0] == NULL) {
+		ret = -ENOMEM;
+		goto err_equ;
+	}
+	offset += abe->hdr.num_equ * sizeof(struct coeff_config);
+	memcpy(abe->equ[0], fw->data + offset, abe->hdr.coeff_size);
+
+	/* allocate coefficient mixer texts */
+	dev_dbg(abe->dev, "loaded %d equalizers\n", abe->hdr.num_equ);
+	for (i = 0; i < abe->hdr.num_equ; i++) {
+		dev_dbg(abe->dev, "equ %d: %s profiles %d\n", i,
+				abe->equ_texts[i].name, abe->equ_texts[i].count);
+		if (abe->equ_texts[i].count >= ABE_MAX_PROFILES) {
+			dev_err(abe->dev, "Too many profiles got %d for equ %d\n",
+					abe->equ_texts[i].count, i);
+			ret = -EINVAL;
+			goto err_texts;
+		}
+		abe->equalizer_enum[i].dtexts =
+				kzalloc(abe->equ_texts[i].count * sizeof(char *), GFP_KERNEL);
+		if (abe->equalizer_enum[i].dtexts == NULL) {
+			ret = -ENOMEM;
+			goto err_texts;
+		}
+	}
+
+	/* initialise coefficient equalizers */
+	for (i = 1; i < abe->hdr.num_equ; i++) {
+		abe->equ[i] = abe->equ[i - 1] +
+			abe->equ_texts[i - 1].count * abe->equ_texts[i - 1].coeff * sizeof(s32);
+	}
+#endif
+	ret = request_irq(abe->irq, abe_irq_handler, 0, "ABE", (void *)abe);
+	if (ret) {
+		dev_err(platform->dev, "request for ABE IRQ %d failed %d\n",
+				abe->irq, ret);
+		goto err_texts;
+	}
+
+	/* aess_clk has to be enabled to access hal register.
+	 * Disable the clk after it has been used.
+	 */
+	pm_runtime_get_sync(abe->dev);
+
+	abe_init_mem(abe->io_base);
+
+	abe_reset_hal();
+
+#if 0
+#warning fixup load fw args
+	//abe_load_fw(fw->data + sizeof(struct fw_header) + abe->hdr.coeff_size);
+#else
+	abe_load_fw();
+#endif
+	/* Config OPP 100 for now */
+	abe_set_opp_processing(ABE_OPP100);
+
+	/* "tick" of the audio engine */
+	abe_write_event_generator(EVENT_TIMER);
+	/* Stop the engine */
+	abe_stop_event_generator();
+	abe_disable_irq();
+
+	pm_runtime_put_sync(abe->dev);
+	abe_add_widgets(platform);
+
+#if defined(CONFIG_SND_OMAP_SOC_ABE_DSP_MODULE)
+	release_firmware(fw);
+#endif
+	return ret;
+
+err_texts:
+#if defined(CONFIG_SND_OMAP_SOC_ABE_DSP_MODULE)
+	for (i = 0; i < abe->hdr.num_equ; i++)
+		kfree(abe->equalizer_enum[i].texts);
+	kfree(abe->equ[0]);
+err_equ:
+	kfree(abe->equ_texts);
+err_fw:
+	release_firmware(fw);
+#endif
+	return ret;
+}
+
+static int abe_remove(struct snd_soc_platform *platform)
+{
+	struct abe_data *abe = snd_soc_platform_get_drvdata(platform);
+	int i;
+
+	free_irq(abe->irq, (void *)abe);
+
+#if defined(CONFIG_SND_OMAP_SOC_ABE_DSP_MODULE)
+	for (i = 0; i < abe->hdr.num_equ; i++)
+		kfree(abe->equalizer_enum[i].texts);
+
+	kfree(abe->equ[0]);
+	kfree(abe->equ_texts);
+#endif
+	pm_runtime_disable(abe->dev);
+
+	return 0;
+}
+
+static struct snd_soc_platform_driver omap_aess_platform = {
+	.ops		= &omap_aess_pcm_ops,
+	.probe		= abe_probe,
+	.remove		= abe_remove,
+	.read		= abe_dsp_read,
+	.write		= abe_dsp_write,
+	.stream_event = aess_stream_event,
+};
+
+static int __devinit abe_engine_probe(struct platform_device *pdev)
+{
+	struct resource *res;
+	struct omap4_abe_dsp_pdata *pdata = pdev->dev.platform_data;
+	struct abe_data *abe;
+	int ret = -EINVAL, i, k;
+
+	abe = kzalloc(sizeof(struct abe_data), GFP_KERNEL);
+	if (abe == NULL)
+		return -ENOMEM;
+	dev_set_drvdata(&pdev->dev, abe);
+	the_abe = abe;
+
+	/* ZERO_labelID should really be 0 */
+	for (i = 0; i < ABE_ROUTES_UL + 2; i++)
+		abe->router[i] = ZERO_labelID;
+
+	for (i = 0; i < 5; i++) {
+		res = platform_get_resource_byname(pdev, IORESOURCE_MEM,
+						   abe_memory_bank[i]);
+		if (res == NULL) {
+			dev_err(&pdev->dev, "no resource %s\n",
+				abe_memory_bank[i]);
+			goto err;
+		}
+		abe->io_base[i] = ioremap(res->start, resource_size(res));
+		if (!abe->io_base[i]) {
+			ret = -ENOMEM;
+			goto err;
+		}
+	}
+
+	abe->irq = platform_get_irq(pdev, 0);
+	if (abe->irq < 0) {
+		ret = abe->irq;
+		goto err;
+	}
+
+	abe->abe_pdata = pdata;
+	abe->dev = &pdev->dev;
+	mutex_init(&abe->mutex);
+	mutex_init(&abe->opp_mutex);
+
+	ret = snd_soc_register_platform(abe->dev,
+			&omap_aess_platform);
+	if (ret < 0)
+		return ret;
+
+	abe_init_debugfs(abe);
+	return ret;
+
+err:
+	for (--i; i >= 0; i--)
+		iounmap(abe->io_base[i]);
+	kfree(abe);
+	return ret;
+}
+
+static int __devexit abe_engine_remove(struct platform_device *pdev)
+{
+	struct abe_data *abe = dev_get_drvdata(&pdev->dev);
+	int i;
+
+	abe_cleanup_debugfs(abe);
+	snd_soc_unregister_platform(&pdev->dev);
+	for (i = 0; i < 5; i++)
+		iounmap(abe->io_base[i]);
+	kfree(abe);
+	return 0;
+}
+
+static struct platform_driver omap_aess_driver = {
+	.driver = {
+		.name = "aess",
+		.owner = THIS_MODULE,
+		.pm = &aess_pm_ops,
+	},
+	.probe = abe_engine_probe,
+	.remove = __devexit_p(abe_engine_remove),
+};
+
+static int __init abe_engine_init(void)
+{
+	return platform_driver_register(&omap_aess_driver);
+}
+module_init(abe_engine_init);
+
+static void __exit abe_engine_exit(void)
+{
+	platform_driver_unregister(&omap_aess_driver);
+}
+module_exit(abe_engine_exit);
+
+MODULE_DESCRIPTION("ASoC OMAP4 ABE");
+MODULE_AUTHOR("Liam Girdwood <lrg@ti.com>");
+MODULE_LICENSE("GPL");
diff --git a/sound/soc/omap/omap-abe-dsp.h b/sound/soc/omap/omap-abe-dsp.h
new file mode 100644
index 0000000..5d7016e
--- /dev/null
+++ b/sound/soc/omap/omap-abe-dsp.h
@@ -0,0 +1,163 @@
+/*
+ * omap-abe-dsp.h
+ *
+ * Copyright (C) 2010 Texas Instruments
+ *
+ * Contact: Liam Girdwood <lrg@ti.com>
+ *
+ * This program is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU General Public License
+ * version 2 as published by the Free Software Foundation.
+ *
+ * This program is distributed in the hope that it will be useful, but
+ * WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA
+ * 02110-1301 USA
+ *
+ */
+
+#ifndef __OMAP_ABE_DSP_H__
+#define __OMAP_ABE_DSP_H__
+
+#define ABE_MIXER(x)		(x)
+
+#define MIX_DL1_TONES		ABE_MIXER(0)
+#define MIX_DL1_VOICE		ABE_MIXER(1)
+#define MIX_DL1_CAPTURE		ABE_MIXER(2)
+#define MIX_DL1_MEDIA		ABE_MIXER(3)
+#define MIX_DL2_TONES		ABE_MIXER(4)
+#define MIX_DL2_VOICE		ABE_MIXER(5)
+#define MIX_DL2_CAPTURE		ABE_MIXER(6)
+#define MIX_DL2_MEDIA		ABE_MIXER(7)
+#define MIX_AUDUL_TONES		ABE_MIXER(8)
+#define MIX_AUDUL_MEDIA		ABE_MIXER(9)
+#define MIX_AUDUL_CAPTURE		ABE_MIXER(10)
+#define MIX_VXREC_TONES		ABE_MIXER(11)
+#define MIX_VXREC_VOICE_PLAYBACK		ABE_MIXER(12)
+#define MIX_VXREC_VOICE_CAPTURE		ABE_MIXER(13)
+#define MIX_VXREC_MEDIA		ABE_MIXER(14)
+#define MIX_SDT_CAPTURE		ABE_MIXER(15)
+#define MIX_SDT_PLAYBACK		ABE_MIXER(16)
+#define MIX_SWITCH_PDM_DL		ABE_MIXER(17)
+#define MIX_SWITCH_BT_VX_DL		ABE_MIXER(18)
+#define MIX_SWITCH_MM_EXT_DL		ABE_MIXER(19)
+
+#define ABE_NUM_MIXERS		(MIX_SWITCH_MM_EXT_DL + 1)
+
+#define ABE_MUX(x)		(x + ABE_NUM_MIXERS)
+
+#define MUX_MM_UL10		ABE_MUX(0)
+#define MUX_MM_UL11		ABE_MUX(1)
+#define MUX_MM_UL12		ABE_MUX(2)
+#define MUX_MM_UL13		ABE_MUX(3)
+#define MUX_MM_UL14		ABE_MUX(4)
+#define MUX_MM_UL15		ABE_MUX(5)
+#define MUX_MM_UL16		ABE_MUX(6)
+#define MUX_MM_UL17		ABE_MUX(7)
+#define MUX_MM_UL20		ABE_MUX(8)
+#define MUX_MM_UL21		ABE_MUX(9)
+#define MUX_VX_UL0		ABE_MUX(10)
+#define MUX_VX_UL1		ABE_MUX(11)
+
+#define ABE_NUM_MUXES		(MUX_VX_UL1 - MUX_MM_UL10)
+
+#define ABE_WIDGET(x)		(x + ABE_NUM_MIXERS + ABE_NUM_MUXES)
+
+/* ABE AIF Frontend Widgets */
+#define W_AIF_TONES_DL		ABE_WIDGET(0)
+#define W_AIF_VX_DL		ABE_WIDGET(1)
+#define W_AIF_VX_UL		ABE_WIDGET(2)
+#define W_AIF_MM_UL1		ABE_WIDGET(3)
+#define W_AIF_MM_UL2		ABE_WIDGET(4)
+#define W_AIF_MM_DL		ABE_WIDGET(5)
+#define W_AIF_MM_DL_LP		W_AIF_MM_DL
+#define W_AIF_VIB_DL		ABE_WIDGET(6)
+#define W_AIF_MODEM_DL		ABE_WIDGET(7)
+#define W_AIF_MODEM_UL		ABE_WIDGET(8)
+
+/* ABE AIF Backend Widgets */
+#define W_AIF_PDM_UL1		ABE_WIDGET(9)
+#define W_AIF_PDM_DL1		ABE_WIDGET(10)
+#define W_AIF_PDM_DL2		ABE_WIDGET(11)
+#define W_AIF_PDM_VIB		ABE_WIDGET(12)
+#define W_AIF_BT_VX_UL		ABE_WIDGET(13)
+#define W_AIF_BT_VX_DL		ABE_WIDGET(14)
+#define W_AIF_MM_EXT_UL	ABE_WIDGET(15)
+#define W_AIF_MM_EXT_DL	ABE_WIDGET(16)
+#define W_AIF_DMIC0		ABE_WIDGET(17)
+#define W_AIF_DMIC1		ABE_WIDGET(18)
+#define W_AIF_DMIC2		ABE_WIDGET(19)
+
+/* ABE ROUTE_UL MUX Widgets */
+#define W_MUX_UL00		ABE_WIDGET(20)
+#define W_MUX_UL01		ABE_WIDGET(21)
+#define W_MUX_UL02		ABE_WIDGET(22)
+#define W_MUX_UL03		ABE_WIDGET(23)
+#define W_MUX_UL04		ABE_WIDGET(24)
+#define W_MUX_UL05		ABE_WIDGET(25)
+#define W_MUX_UL06		ABE_WIDGET(26)
+#define W_MUX_UL07		ABE_WIDGET(27)
+#define W_MUX_UL10		ABE_WIDGET(28)
+#define W_MUX_UL11		ABE_WIDGET(29)
+#define W_MUX_VX00		ABE_WIDGET(30)
+#define W_MUX_VX01		ABE_WIDGET(31)
+
+/* ABE Volume and Mixer Widgets */
+#define W_MIXER_DL1		ABE_WIDGET(32)
+#define W_MIXER_DL2		ABE_WIDGET(33)
+#define W_VOLUME_DL1		ABE_WIDGET(34)
+#define W_MIXER_AUDIO_UL	ABE_WIDGET(35)
+#define W_MIXER_VX_REC		ABE_WIDGET(36)
+#define W_MIXER_SDT		ABE_WIDGET(37)
+#define W_VSWITCH_DL1_PDM	ABE_WIDGET(38)
+#define W_VSWITCH_DL1_BT_VX	ABE_WIDGET(39)
+#define W_VSWITCH_DL1_MM_EXT	ABE_WIDGET(40)
+
+#define ABE_NUM_WIDGETS		(W_VSWITCH_DL1_MM_EXT - W_AIF_TONES_DL)
+#define ABE_WIDGET_LAST		W_VSWITCH_DL1_MM_EXT
+
+#define ABE_NUM_DAPM_REG		\
+	(ABE_NUM_MIXERS + ABE_NUM_MUXES + ABE_NUM_WIDGETS)
+
+#define ABE_VIRTUAL_SWITCH	0
+#define ABE_ROUTES_UL		14
+
+// TODO: OPP bitmask - Use HAL version after update
+#define ABE_OPP_25		0
+#define ABE_OPP_50		1
+#define ABE_OPP_100		2
+
+/* TODO: size in bytes of debug options */
+#define ABE_DBG_FLAG1_SIZE	0
+#define ABE_DBG_FLAG2_SIZE	0
+#define ABE_DBG_FLAG3_SIZE	0
+
+/* TODO: Pong start offset of DMEM */
+/* Ping pong buffer DMEM offset */
+#define ABE_DMEM_BASE_OFFSET_PING_PONG	0x4000
+
+/* Gain value conversion */
+#define ABE_MAX_GAIN		12000
+#define ABE_GAIN_SCALE		100
+#define abe_gain_to_val(gain)	((val + ABE_MAX_GAIN) / ABE_GAIN_SCALE)
+#define abe_val_to_gain(val) (-ABE_MAX_GAIN + (val * ABE_GAIN_SCALE))
+
+/* Firmware coefficients and equalizers */
+#define ABE_MAX_FW_SIZE		(1024 * 128)
+#define ABE_MAX_COEFF_SIZE	(1024 * 4)
+#define ABE_COEFF_NAME_SIZE	20
+#define ABE_COEFF_TEXT_SIZE	20
+#define ABE_COEFF_NUM_TEXTS	10
+#define ABE_MAX_EQU		10
+#define ABE_MAX_PROFILES	30
+
+void abe_dsp_shutdown(void);
+void abe_dsp_pm_get(void);
+void abe_dsp_pm_put(void);
+
+#endif	/* End of __OMAP_ABE_DSP_H__ */